diff options
292 files changed, 22045 insertions, 8037 deletions
diff --git a/Documentation/devicetree/bindings/sound/imx-audio-sgtl5000.txt b/Documentation/devicetree/bindings/sound/imx-audio-sgtl5000.txt new file mode 100644 index 00000000000..e4acdd891e4 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/imx-audio-sgtl5000.txt @@ -0,0 +1,49 @@ +Freescale i.MX audio complex with SGTL5000 codec + +Required properties: +- compatible : "fsl,imx-audio-sgtl5000" +- model : The user-visible name of this sound complex +- ssi-controller : The phandle of the i.MX SSI controller +- audio-codec : The phandle of the SGTL5000 audio codec +- audio-routing : A list of the connections between audio components. + Each entry is a pair of strings, the first being the connection's sink, + the second being the connection's source. Valid names could be power + supplies, SGTL5000 pins, and the jacks on the board: + + Power supplies: + * Mic Bias + + SGTL5000 pins: + * MIC_IN + * LINE_IN + * HP_OUT + * LINE_OUT + + Board connectors: + * Mic Jack + * Line In Jack + * Headphone Jack + * Line Out Jack + * Ext Spk + +- mux-int-port : The internal port of the i.MX audio muxer (AUDMUX) +- mux-ext-port : The external port of the i.MX audio muxer + +Note: The AUDMUX port numbering should start at 1, which is consistent with +hardware manual. + +Example: + +sound { + compatible = "fsl,imx51-babbage-sgtl5000", + "fsl,imx-audio-sgtl5000"; + model = "imx51-babbage-sgtl5000"; + ssi-controller = <&ssi1>; + audio-codec = <&sgtl5000>; + audio-routing = + "MIC_IN", "Mic Jack", + "Mic Jack", "Mic Bias", + "Headphone Jack", "HP_OUT"; + mux-int-port = <1>; + mux-ext-port = <3>; +}; diff --git a/Documentation/devicetree/bindings/sound/mxs-audio-sgtl5000.txt b/Documentation/devicetree/bindings/sound/mxs-audio-sgtl5000.txt new file mode 100644 index 00000000000..601c518edda --- /dev/null +++ b/Documentation/devicetree/bindings/sound/mxs-audio-sgtl5000.txt @@ -0,0 +1,17 @@ +* Freescale MXS audio complex with SGTL5000 codec + +Required properties: +- compatible: "fsl,mxs-audio-sgtl5000" +- model: The user-visible name of this sound complex +- saif-controllers: The phandle list of the MXS SAIF controller +- audio-codec: The phandle of the SGTL5000 audio codec + +Example: + +sound { + compatible = "fsl,imx28-evk-sgtl5000", + "fsl,mxs-audio-sgtl5000"; + model = "imx28-evk-sgtl5000"; + saif-controllers = <&saif0 &saif1>; + audio-codec = <&sgtl5000>; +}; diff --git a/Documentation/devicetree/bindings/sound/mxs-saif.txt b/Documentation/devicetree/bindings/sound/mxs-saif.txt new file mode 100644 index 00000000000..c37ba6143d9 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/mxs-saif.txt @@ -0,0 +1,36 @@ +* Freescale MXS Serial Audio Interface (SAIF) + +Required properties: +- compatible: Should be "fsl,<chip>-saif" +- reg: Should contain registers location and length +- interrupts: Should contain ERROR and DMA interrupts +- fsl,saif-dma-channel: APBX DMA channel for the SAIF + +Optional properties: +- fsl,saif-master: phandle to the master SAIF. It's only required for + the slave SAIF. + +Note: Each SAIF controller should have an alias correctly numbered +in "aliases" node. + +Example: + +aliases { + saif0 = &saif0; + saif1 = &saif1; +}; + +saif0: saif@80042000 { + compatible = "fsl,imx28-saif"; + reg = <0x80042000 2000>; + interrupts = <59 80>; + fsl,saif-dma-channel = <4>; +}; + +saif1: saif@80046000 { + compatible = "fsl,imx28-saif"; + reg = <0x80046000 2000>; + interrupts = <58 81>; + fsl,saif-dma-channel = <5>; + fsl,saif-master = <&saif0>; +}; diff --git a/Documentation/devicetree/bindings/sound/nvidia,tegra30-ahub.txt b/Documentation/devicetree/bindings/sound/nvidia,tegra30-ahub.txt new file mode 100644 index 00000000000..1ac7b164218 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/nvidia,tegra30-ahub.txt @@ -0,0 +1,32 @@ +NVIDIA Tegra30 AHUB (Audio Hub) + +Required properties: +- compatible : "nvidia,tegra30-ahub" +- reg : Should contain the register physical address and length for each of + the AHUB's APBIF registers and the AHUB's own registers. +- interrupts : Should contain AHUB interrupt +- nvidia,dma-request-selector : The Tegra DMA controller's phandle and + request selector for the first APBIF channel. +- ranges : The bus address mapping for the configlink register bus. + Can be empty since the mapping is 1:1. +- #address-cells : For the configlink bus. Should be <1>; +- #size-cells : For the configlink bus. Should be <1>. + +AHUB client modules need to specify the IDs of their CIFs (Client InterFaces). +For RX CIFs, the numbers indicate the register number within AHUB routing +register space (APBIF 0..3 RX, I2S 0..5 RX, DAM 0..2 RX 0..1, SPDIF RX 0..1). +For TX CIFs, the numbers indicate the bit position within the AHUB routing +registers (APBIF 0..3 TX, I2S 0..5 TX, DAM 0..2 TX, SPDIF TX 0..1). + +Example: + +ahub@70080000 { + compatible = "nvidia,tegra30-ahub"; + reg = <0x70080000 0x200 0x70080200 0x100>; + interrupts = < 0 103 0x04 >; + nvidia,dma-request-selector = <&apbdma 1>; + + ranges; + #address-cells = <1>; + #size-cells = <1>; +}; diff --git a/Documentation/devicetree/bindings/sound/nvidia,tegra30-i2s.txt b/Documentation/devicetree/bindings/sound/nvidia,tegra30-i2s.txt new file mode 100644 index 00000000000..dfa6c037124 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/nvidia,tegra30-i2s.txt @@ -0,0 +1,15 @@ +NVIDIA Tegra30 I2S controller + +Required properties: +- compatible : "nvidia,tegra30-i2s" +- reg : Should contain I2S registers location and length +- nvidia,ahub-cif-ids : The list of AHUB CIF IDs for this port, rx (playback) + first, tx (capture) second. See nvidia,tegra30-ahub.txt for values. + +Example: + +i2s@70002800 { + compatible = "nvidia,tegra30-i2s"; + reg = <0x70080300 0x100>; + nvidia,ahub-cif-ids = <4 4>; +}; diff --git a/Documentation/devicetree/bindings/sound/tegra-audio-trimslice.txt b/Documentation/devicetree/bindings/sound/tegra-audio-trimslice.txt new file mode 100644 index 00000000000..04b14cfb1f1 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/tegra-audio-trimslice.txt @@ -0,0 +1,14 @@ +NVIDIA Tegra audio complex for TrimSlice + +Required properties: +- compatible : "nvidia,tegra-audio-trimslice" +- nvidia,i2s-controller : The phandle of the Tegra I2S1 controller +- nvidia,audio-codec : The phandle of the WM8903 audio codec + +Example: + +sound { + compatible = "nvidia,tegra-audio-trimslice"; + nvidia,i2s-controller = <&tegra_i2s1>; + nvidia,audio-codec = <&codec>; +}; diff --git a/Documentation/devicetree/bindings/sound/tegra-audio-wm8753.txt b/Documentation/devicetree/bindings/sound/tegra-audio-wm8753.txt new file mode 100644 index 00000000000..c4dd39ce616 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/tegra-audio-wm8753.txt @@ -0,0 +1,54 @@ +NVIDIA Tegra audio complex + +Required properties: +- compatible : "nvidia,tegra-audio-wm8753" +- nvidia,model : The user-visible name of this sound complex. +- nvidia,audio-routing : A list of the connections between audio components. + Each entry is a pair of strings, the first being the connection's sink, + the second being the connection's source. Valid names for sources and + sinks are the WM8753's pins, and the jacks on the board: + + WM8753 pins: + + * LOUT1 + * LOUT2 + * ROUT1 + * ROUT2 + * MONO1 + * MONO2 + * OUT3 + * OUT4 + * LINE1 + * LINE2 + * RXP + * RXN + * ACIN + * ACOP + * MIC1N + * MIC1 + * MIC2N + * MIC2 + * Mic Bias + + Board connectors: + + * Headphone Jack + * Mic Jack + +- nvidia,i2s-controller : The phandle of the Tegra I2S1 controller +- nvidia,audio-codec : The phandle of the WM8753 audio codec +Example: + +sound { + compatible = "nvidia,tegra-audio-wm8753-whistler", + "nvidia,tegra-audio-wm8753" + nvidia,model = "tegra-wm8753-harmony"; + + nvidia,audio-routing = + "Headphone Jack", "LOUT1", + "Headphone Jack", "ROUT1"; + + nvidia,i2s-controller = <&i2s1>; + nvidia,audio-codec = <&wm8753>; +}; + diff --git a/Documentation/sound/alsa/ALSA-Configuration.txt b/Documentation/sound/alsa/ALSA-Configuration.txt index 8c16d50f6cb..221b81016db 100644 --- a/Documentation/sound/alsa/ALSA-Configuration.txt +++ b/Documentation/sound/alsa/ALSA-Configuration.txt @@ -1545,7 +1545,7 @@ Prior to version 0.9.0rc4 options had a 'snd_' prefix. This was removed. Module for sound cards based on the C-Media CMI8786/8787/8788 chip: * Asound A-8788 - * Asus Xonar DG + * Asus Xonar DG/DGX * AuzenTech X-Meridian * AuzenTech X-Meridian 2G * Bgears b-Enspirer diff --git a/MAINTAINERS b/MAINTAINERS index 9c293bf340b..61dd168d349 100644 --- a/MAINTAINERS +++ b/MAINTAINERS @@ -6691,6 +6691,12 @@ F: drivers/misc/tifm* F: drivers/mmc/host/tifm_sd.c F: include/linux/tifm.h +TI LM49xxx FAMILY ASoC CODEC DRIVERS +M: M R Swami Reddy <mr.swami.reddy@ti.com> +L: alsa-devel@alsa-project.org (moderated for non-subscribers) +S: Maintained +F: sound/soc/codecs/lm49453* + TI TWL4030 SERIES SOC CODEC DRIVER M: Peter Ujfalusi <peter.ujfalusi@ti.com> L: alsa-devel@alsa-project.org (moderated for non-subscribers) diff --git a/arch/arm/mach-imx/Kconfig b/arch/arm/mach-imx/Kconfig index 7d6322ce522..cca8c0c7479 100644 --- a/arch/arm/mach-imx/Kconfig +++ b/arch/arm/mach-imx/Kconfig @@ -151,6 +151,7 @@ config MACH_MX25_3DS select IMX_HAVE_PLATFORM_IMX2_WDT select IMX_HAVE_PLATFORM_IMXDI_RTC select IMX_HAVE_PLATFORM_IMX_I2C + select IMX_HAVE_PLATFORM_IMX_SSI select IMX_HAVE_PLATFORM_IMX_FB select IMX_HAVE_PLATFORM_IMX_KEYPAD select IMX_HAVE_PLATFORM_IMX_UART @@ -495,6 +496,7 @@ config MACH_MX31MOBOARD select IMX_HAVE_PLATFORM_FSL_USB2_UDC select IMX_HAVE_PLATFORM_IMX2_WDT select IMX_HAVE_PLATFORM_IMX_I2C + select IMX_HAVE_PLATFORM_IMX_SSI select IMX_HAVE_PLATFORM_IMX_UART select IMX_HAVE_PLATFORM_IPU_CORE select IMX_HAVE_PLATFORM_MXC_EHCI diff --git a/arch/arm/mach-imx/mach-mx31_3ds.c b/arch/arm/mach-imx/mach-mx31_3ds.c index 4d1aab15440..4eafdf275ea 100644 --- a/arch/arm/mach-imx/mach-mx31_3ds.c +++ b/arch/arm/mach-imx/mach-mx31_3ds.c @@ -156,6 +156,11 @@ static int mx31_3ds_pins[] = { MX31_PIN_CSI_VSYNC__CSI_VSYNC, MX31_PIN_CSI_D5__GPIO3_5, /* CMOS PWDN */ IOMUX_MODE(MX31_PIN_RI_DTE1, IOMUX_CONFIG_GPIO), /* CMOS reset */ + /* SSI */ + MX31_PIN_STXD4__STXD4, + MX31_PIN_SRXD4__SRXD4, + MX31_PIN_SCK4__SCK4, + MX31_PIN_SFS4__SFS4, }; /* @@ -488,12 +493,23 @@ static struct mc13xxx_regulator_init_data mx31_3ds_regulators[] = { }; /* MC13783 */ +static struct mc13xxx_codec_platform_data mx31_3ds_codec = { + .dac_ssi_port = MC13783_SSI1_PORT, + .adc_ssi_port = MC13783_SSI1_PORT, +}; + static struct mc13xxx_platform_data mc13783_pdata = { .regulators = { .regulators = mx31_3ds_regulators, .num_regulators = ARRAY_SIZE(mx31_3ds_regulators), }, - .flags = MC13XXX_USE_TOUCHSCREEN | MC13XXX_USE_RTC, + .codec = &mx31_3ds_codec, + .flags = MC13XXX_USE_TOUCHSCREEN | MC13XXX_USE_RTC | MC13XXX_USE_CODEC, + +}; + +static struct imx_ssi_platform_data mx31_3ds_ssi_pdata = { + .flags = IMX_SSI_DMA | IMX_SSI_NET, }; /* SPI */ @@ -741,6 +757,10 @@ static void __init mx31_3ds_init(void) } mx31_3ds_init_camera(); + + imx31_add_imx_ssi(0, &mx31_3ds_ssi_pdata); + + imx_add_platform_device("imx_mc13783", 0, NULL, 0, NULL, 0); } static void __init mx31_3ds_timer_init(void) diff --git a/arch/arm/mach-imx/mach-mx31moboard.c b/arch/arm/mach-imx/mach-mx31moboard.c index 1dfe3c7a7be..016791f038b 100644 --- a/arch/arm/mach-imx/mach-mx31moboard.c +++ b/arch/arm/mach-imx/mach-mx31moboard.c @@ -47,6 +47,7 @@ #include <mach/hardware.h> #include <mach/iomux-mx3.h> #include <mach/ulpi.h> +#include <mach/ssi.h> #include "devices-imx31.h" @@ -102,6 +103,9 @@ static unsigned int moboard_pins[] = { MX31_PIN_CSPI3_MOSI__MOSI, MX31_PIN_CSPI3_MISO__MISO, MX31_PIN_CSPI3_SCLK__SCLK, MX31_PIN_CSPI3_SPI_RDY__SPI_RDY, MX31_PIN_CSPI2_SS1__CSPI3_SS1, + /* SSI */ + MX31_PIN_STXD4__STXD4, MX31_PIN_SRXD4__SRXD4, + MX31_PIN_SCK4__SCK4, MX31_PIN_SFS4__SFS4, }; static struct physmap_flash_data mx31moboard_flash_data = { @@ -276,6 +280,11 @@ static struct mc13xxx_buttons_platform_data moboard_buttons = { .b1on_key = KEY_POWER, }; +static struct mc13xxx_codec_platform_data moboard_codec = { + .dac_ssi_port = MC13783_SSI1_PORT, + .adc_ssi_port = MC13783_SSI1_PORT, +}; + static struct mc13xxx_platform_data moboard_pmic = { .regulators = { .regulators = moboard_regulators, @@ -283,7 +292,12 @@ static struct mc13xxx_platform_data moboard_pmic = { }, .leds = &moboard_leds, .buttons = &moboard_buttons, - .flags = MC13XXX_USE_RTC | MC13XXX_USE_ADC, + .codec = &moboard_codec, + .flags = MC13XXX_USE_RTC | MC13XXX_USE_ADC | MC13XXX_USE_CODEC, +}; + +static struct imx_ssi_platform_data moboard_ssi_pdata = { + .flags = IMX_SSI_DMA | IMX_SSI_NET, }; static struct spi_board_info moboard_spi_board_info[] __initdata = { @@ -554,6 +568,10 @@ static void __init mx31moboard_init(void) moboard_usbh2_init(); + imx31_add_imx_ssi(0, &moboard_ssi_pdata); + + imx_add_platform_device("imx_mc13783", 0, NULL, 0, NULL, 0); + pm_power_off = mx31moboard_poweroff; switch (mx31moboard_baseboard) { diff --git a/arch/arm/mach-shmobile/Kconfig b/arch/arm/mach-shmobile/Kconfig index 98327b7a503..f31383c32f9 100644 --- a/arch/arm/mach-shmobile/Kconfig +++ b/arch/arm/mach-shmobile/Kconfig @@ -64,6 +64,7 @@ config MACH_AP4EVB depends on ARCH_SH7372 select ARCH_REQUIRE_GPIOLIB select SH_LCD_MIPI_DSI + select SND_SOC_AK4642 if SND_SIMPLE_CARD choice prompt "AP4EVB LCD panel selection" @@ -88,6 +89,7 @@ config MACH_MACKEREL bool "mackerel board" depends on ARCH_SH7372 select ARCH_REQUIRE_GPIOLIB + select SND_SOC_AK4642 if SND_SIMPLE_CARD config MACH_KOTA2 bool "KOTA2 board" diff --git a/arch/arm/mach-shmobile/board-ap4evb.c b/arch/arm/mach-shmobile/board-ap4evb.c index 0c3caeba2f3..b540b8eb20c 100644 --- a/arch/arm/mach-shmobile/board-ap4evb.c +++ b/arch/arm/mach-shmobile/board-ap4evb.c @@ -50,6 +50,7 @@ #include <media/soc_camera.h> #include <sound/sh_fsi.h> +#include <sound/simple_card.h> #include <video/sh_mobile_hdmi.h> #include <video/sh_mobile_lcdc.h> @@ -785,17 +786,25 @@ static struct platform_device fsi_device = { }, }; -static struct fsi_ak4642_info fsi2_ak4643_info = { +static struct asoc_simple_dai_init_info fsi2_ak4643_init_info = { + .fmt = SND_SOC_DAIFMT_LEFT_J, + .codec_daifmt = SND_SOC_DAIFMT_CBM_CFM, + .cpu_daifmt = SND_SOC_DAIFMT_CBS_CFS, + .sysclk = 11289600, +}; + +static struct asoc_simple_card_info fsi2_ak4643_info = { .name = "AK4643", .card = "FSI2A-AK4643", .cpu_dai = "fsia-dai", .codec = "ak4642-codec.0-0013", .platform = "sh_fsi2", - .id = FSI_PORT_A, + .codec_dai = "ak4642-hifi", + .init = &fsi2_ak4643_init_info, }; static struct platform_device fsi_ak4643_device = { - .name = "fsi-ak4642-audio", + .name = "asoc-simple-card", .dev = { .platform_data = &fsi2_ak4643_info, }, @@ -900,8 +909,26 @@ static struct platform_device lcdc1_device = { }, }; +static struct asoc_simple_dai_init_info fsi2_hdmi_init_info = { + .cpu_daifmt = SND_SOC_DAIFMT_CBM_CFM, +}; + +static struct asoc_simple_card_info fsi2_hdmi_info = { + .name = "HDMI", + .card = "FSI2B-HDMI", + .cpu_dai = "fsib-dai", + .codec = "sh-mobile-hdmi", + .platform = "sh_fsi2", + .codec_dai = "sh_mobile_hdmi-hifi", + .init = &fsi2_hdmi_init_info, +}; + static struct platform_device fsi_hdmi_device = { - .name = "sh_fsi2_b_hdmi", + .name = "asoc-simple-card", + .id = 1, + .dev = { + .platform_data = &fsi2_hdmi_info, + }, }; static struct gpio_led ap4evb_leds[] = { diff --git a/arch/arm/mach-shmobile/board-mackerel.c b/arch/arm/mach-shmobile/board-mackerel.c index aae2e24fde4..50c67b22d08 100644 --- a/arch/arm/mach-shmobile/board-mackerel.c +++ b/arch/arm/mach-shmobile/board-mackerel.c @@ -53,6 +53,7 @@ #include <media/soc_camera.h> #include <media/soc_camera_platform.h> #include <sound/sh_fsi.h> +#include <sound/simple_card.h> #include <mach/common.h> #include <mach/irqs.h> @@ -502,8 +503,26 @@ static struct platform_device hdmi_lcdc_device = { }, }; +static struct asoc_simple_dai_init_info fsi2_hdmi_init_info = { + .cpu_daifmt = SND_SOC_DAIFMT_CBM_CFM, +}; + +static struct asoc_simple_card_info fsi2_hdmi_info = { + .name = "HDMI", + .card = "FSI2B-HDMI", + .cpu_dai = "fsib-dai", + .codec = "sh-mobile-hdmi", + .platform = "sh_fsi2", + .codec_dai = "sh_mobile_hdmi-hifi", + .init = &fsi2_hdmi_init_info, +}; + static struct platform_device fsi_hdmi_device = { - .name = "sh_fsi2_b_hdmi", + .name = "asoc-simple-card", + .id = 1, + .dev = { + .platform_data = &fsi2_hdmi_info, + }, }; static void __init hdmi_init_pm_clock(void) @@ -945,17 +964,25 @@ static struct platform_device fsi_device = { }, }; -static struct fsi_ak4642_info fsi2_ak4643_info = { +static struct asoc_simple_dai_init_info fsi2_ak4643_init_info = { + .fmt = SND_SOC_DAIFMT_LEFT_J, + .codec_daifmt = SND_SOC_DAIFMT_CBM_CFM, + .cpu_daifmt = SND_SOC_DAIFMT_CBS_CFS, + .sysclk = 11289600, +}; + +static struct asoc_simple_card_info fsi2_ak4643_info = { .name = "AK4643", .card = "FSI2A-AK4643", .cpu_dai = "fsia-dai", .codec = "ak4642-codec.0-0013", .platform = "sh_fsi2", - .id = FSI_PORT_A, + .codec_dai = "ak4642-hifi", + .init = &fsi2_ak4643_init_info, }; static struct platform_device fsi_ak4643_device = { - .name = "fsi-ak4642-audio", + .name = "asoc-simple-card", .dev = { .platform_data = &fsi2_ak4643_info, }, diff --git a/arch/arm/mach-tegra/board-dt-tegra20.c b/arch/arm/mach-tegra/board-dt-tegra20.c index 8351c4c147a..fac3eb1af17 100644 --- a/arch/arm/mach-tegra/board-dt-tegra20.c +++ b/arch/arm/mach-tegra/board-dt-tegra20.c @@ -55,9 +55,9 @@ struct of_dev_auxdata tegra20_auxdata_lookup[] __initdata = { OF_DEV_AUXDATA("nvidia,tegra20-i2c", TEGRA_I2C2_BASE, "tegra-i2c.1", NULL), OF_DEV_AUXDATA("nvidia,tegra20-i2c", TEGRA_I2C3_BASE, "tegra-i2c.2", NULL), OF_DEV_AUXDATA("nvidia,tegra20-i2c-dvc", TEGRA_DVC_BASE, "tegra-i2c.3", NULL), - OF_DEV_AUXDATA("nvidia,tegra20-i2s", TEGRA_I2S1_BASE, "tegra-i2s.0", NULL), - OF_DEV_AUXDATA("nvidia,tegra20-i2s", TEGRA_I2S2_BASE, "tegra-i2s.1", NULL), - OF_DEV_AUXDATA("nvidia,tegra20-das", TEGRA_APB_MISC_DAS_BASE, "tegra-das", NULL), + OF_DEV_AUXDATA("nvidia,tegra20-i2s", TEGRA_I2S1_BASE, "tegra20-i2s.0", NULL), + OF_DEV_AUXDATA("nvidia,tegra20-i2s", TEGRA_I2S2_BASE, "tegra20-i2s.1", NULL), + OF_DEV_AUXDATA("nvidia,tegra20-das", TEGRA_APB_MISC_DAS_BASE, "tegra20-das", NULL), OF_DEV_AUXDATA("nvidia,tegra20-ehci", TEGRA_USB_BASE, "tegra-ehci.0", &tegra_ehci1_pdata), OF_DEV_AUXDATA("nvidia,tegra20-ehci", TEGRA_USB2_BASE, "tegra-ehci.1", diff --git a/arch/arm/mach-tegra/board-harmony.c b/arch/arm/mach-tegra/board-harmony.c index 222182e0022..b906b3b6077 100644 --- a/arch/arm/mach-tegra/board-harmony.c +++ b/arch/arm/mach-tegra/board-harmony.c @@ -124,7 +124,6 @@ static struct platform_device *harmony_devices[] __initdata = { &tegra_ehci3_device, &tegra_i2s_device1, &tegra_das_device, - &tegra_pcm_device, &harmony_audio_device, }; diff --git a/arch/arm/mach-tegra/board-seaboard.c b/arch/arm/mach-tegra/board-seaboard.c index 20743bcec03..79064c7a790 100644 --- a/arch/arm/mach-tegra/board-seaboard.c +++ b/arch/arm/mach-tegra/board-seaboard.c @@ -156,7 +156,6 @@ static struct platform_device *seaboard_devices[] __initdata = { &seaboard_gpio_keys_device, &tegra_i2s_device1, &tegra_das_device, - &tegra_pcm_device, &seaboard_audio_device, }; diff --git a/arch/arm/mach-tegra/board-trimslice.c b/arch/arm/mach-tegra/board-trimslice.c index 0a00183feee..bc59b379c6f 100644 --- a/arch/arm/mach-tegra/board-trimslice.c +++ b/arch/arm/mach-tegra/board-trimslice.c @@ -89,7 +89,6 @@ static struct platform_device *trimslice_devices[] __initdata = { &tegra_sdhci_device4, &tegra_i2s_device1, &tegra_das_device, - &tegra_pcm_device, &trimslice_audio_device, }; diff --git a/arch/arm/mach-tegra/devices.c b/arch/arm/mach-tegra/devices.c index bd3035e0cea..2d8dfa2faf8 100644 --- a/arch/arm/mach-tegra/devices.c +++ b/arch/arm/mach-tegra/devices.c @@ -674,14 +674,14 @@ static struct resource i2s_resource2[] = { }; struct platform_device tegra_i2s_device1 = { - .name = "tegra-i2s", + .name = "tegra20-i2s", .id = 0, .resource = i2s_resource1, .num_resources = ARRAY_SIZE(i2s_resource1), }; struct platform_device tegra_i2s_device2 = { - .name = "tegra-i2s", + .name = "tegra20-i2s", .id = 1, .resource = i2s_resource2, .num_resources = ARRAY_SIZE(i2s_resource2), @@ -696,13 +696,8 @@ static struct resource tegra_das_resources[] = { }; struct platform_device tegra_das_device = { - .name = "tegra-das", + .name = "tegra20-das", .id = -1, .num_resources = ARRAY_SIZE(tegra_das_resources), .resource = tegra_das_resources, }; - -struct platform_device tegra_pcm_device = { - .name = "tegra-pcm-audio", - .id = -1, -}; diff --git a/arch/arm/mach-tegra/devices.h b/arch/arm/mach-tegra/devices.h index ec455679b21..138c642e59f 100644 --- a/arch/arm/mach-tegra/devices.h +++ b/arch/arm/mach-tegra/devices.h @@ -52,6 +52,5 @@ extern struct platform_device tegra_pmu_device; extern struct platform_device tegra_i2s_device1; extern struct platform_device tegra_i2s_device2; extern struct platform_device tegra_das_device; -extern struct platform_device tegra_pcm_device; #endif diff --git a/arch/arm/mach-tegra/tegra2_clocks.c b/arch/arm/mach-tegra/tegra2_clocks.c index 2cae5cbc20b..bae09b85989 100644 --- a/arch/arm/mach-tegra/tegra2_clocks.c +++ b/arch/arm/mach-tegra/tegra2_clocks.c @@ -2148,8 +2148,8 @@ static struct clk tegra_list_clks[] = { PERIPH_CLK("apbdma", "tegra-dma", NULL, 34, 0, 108000000, mux_pclk, 0), PERIPH_CLK("rtc", "rtc-tegra", NULL, 4, 0, 32768, mux_clk_32k, PERIPH_NO_RESET), PERIPH_CLK("timer", "timer", NULL, 5, 0, 26000000, mux_clk_m, 0), - PERIPH_CLK("i2s1", "tegra-i2s.0", NULL, 11, 0x100, 26000000, mux_pllaout0_audio2x_pllp_clkm, MUX | DIV_U71), - PERIPH_CLK("i2s2", "tegra-i2s.1", NULL, 18, 0x104, 26000000, mux_pllaout0_audio2x_pllp_clkm, MUX | DIV_U71), + PERIPH_CLK("i2s1", "tegra20-i2s.0", NULL, 11, 0x100, 26000000, mux_pllaout0_audio2x_pllp_clkm, MUX | DIV_U71), + PERIPH_CLK("i2s2", "tegra20-i2s.1", NULL, 18, 0x104, 26000000, mux_pllaout0_audio2x_pllp_clkm, MUX | DIV_U71), PERIPH_CLK("spdif_out", "spdif_out", NULL, 10, 0x108, 100000000, mux_pllaout0_audio2x_pllp_clkm, MUX | DIV_U71), PERIPH_CLK("spdif_in", "spdif_in", NULL, 10, 0x10c, 100000000, mux_pllp_pllc_pllm, MUX | DIV_U71), PERIPH_CLK("pwm", "pwm", NULL, 17, 0x110, 432000000, mux_pllp_pllc_audio_clkm_clk32, MUX | DIV_U71), diff --git a/arch/powerpc/configs/86xx/mpc8610_hpcd_defconfig b/arch/powerpc/configs/86xx/mpc8610_hpcd_defconfig index 0db9ba0423f..c09598b31de 100644 --- a/arch/powerpc/configs/86xx/mpc8610_hpcd_defconfig +++ b/arch/powerpc/configs/86xx/mpc8610_hpcd_defconfig @@ -100,6 +100,7 @@ CONFIG_SND_MIXER_OSS=y CONFIG_SND_PCM_OSS=y # CONFIG_SND_SUPPORT_OLD_API is not set CONFIG_SND_SOC=y +CONFIG_SND_POWERPC_SOC=y CONFIG_RTC_CLASS=y CONFIG_RTC_DRV_CMOS=y CONFIG_EXT2_FS=y diff --git a/arch/powerpc/configs/mpc85xx_defconfig b/arch/powerpc/configs/mpc85xx_defconfig index d6b6df5e874..62bb723c5b5 100644 --- a/arch/powerpc/configs/mpc85xx_defconfig +++ b/arch/powerpc/configs/mpc85xx_defconfig @@ -141,6 +141,7 @@ CONFIG_SND_INTEL8X0=y # CONFIG_SND_PPC is not set # CONFIG_SND_USB is not set CONFIG_SND_SOC=y +CONFIG_SND_POWERPC_SOC=y CONFIG_HID_A4TECH=y CONFIG_HID_APPLE=y CONFIG_HID_BELKIN=y diff --git a/arch/powerpc/configs/mpc85xx_smp_defconfig b/arch/powerpc/configs/mpc85xx_smp_defconfig index 5b0e2926bec..d1828427ae5 100644 --- a/arch/powerpc/configs/mpc85xx_smp_defconfig +++ b/arch/powerpc/configs/mpc85xx_smp_defconfig @@ -143,6 +143,7 @@ CONFIG_SND_INTEL8X0=y # CONFIG_SND_PPC is not set # CONFIG_SND_USB is not set CONFIG_SND_SOC=y +CONFIG_SND_POWERPC_SOC=y CONFIG_HID_A4TECH=y CONFIG_HID_APPLE=y CONFIG_HID_BELKIN=y diff --git a/arch/sh/boards/Kconfig b/arch/sh/boards/Kconfig index 3a74b10922e..1f56b35d324 100644 --- a/arch/sh/boards/Kconfig +++ b/arch/sh/boards/Kconfig @@ -54,6 +54,7 @@ config SH_7724_SOLUTION_ENGINE select SOLUTION_ENGINE depends on CPU_SUBTYPE_SH7724 select ARCH_REQUIRE_GPIOLIB + select SND_SOC_AK4642 if SND_SIMPLE_CARD help Select 7724 SolutionEngine if configuring for a Hitachi SH7724 evaluation board. @@ -224,6 +225,7 @@ config SH_ECOVEC bool "EcoVec" depends on CPU_SUBTYPE_SH7724 select ARCH_REQUIRE_GPIOLIB + select SND_SOC_DA7210 if SND_SIMPLE_CARD help Renesas "R0P7724LC0011/21RL (EcoVec)" support. diff --git a/arch/sh/boards/mach-ecovec24/setup.c b/arch/sh/boards/mach-ecovec24/setup.c index 63002c8a0ec..4158d70c0de 100644 --- a/arch/sh/boards/mach-ecovec24/setup.c +++ b/arch/sh/boards/mach-ecovec24/setup.c @@ -33,6 +33,7 @@ #include <linux/videodev2.h> #include <video/sh_mobile_lcdc.h> #include <sound/sh_fsi.h> +#include <sound/simple_card.h> #include <media/sh_mobile_ceu.h> #include <media/soc_camera.h> #include <media/tw9910.h> @@ -811,6 +812,30 @@ static struct platform_device fsi_device = { }, }; +static struct asoc_simple_dai_init_info fsi_da7210_init_info = { + .fmt = SND_SOC_DAIFMT_I2S, + .codec_daifmt = SND_SOC_DAIFMT_CBM_CFM, + .cpu_daifmt = SND_SOC_DAIFMT_CBS_CFS, +}; + +static struct asoc_simple_card_info fsi_da7210_info = { + .name = "DA7210", + .card = "FSIB-DA7210", + .cpu_dai = "fsib-dai", + .codec = "da7210.0-001a", + .platform = "sh_fsi.0", + .codec_dai = "da7210-hifi", + .init = &fsi_da7210_init_info, +}; + +static struct platform_device fsi_da7210_device = { + .name = "asoc-simple-card", + .dev = { + .platform_data = &fsi_da7210_info, + }, +}; + + /* IrDA */ static struct resource irda_resources[] = { [0] = { @@ -947,6 +972,7 @@ static struct platform_device *ecovec_devices[] __initdata = { &camera_devices[1], &camera_devices[2], &fsi_device, + &fsi_da7210_device, &irda_device, &vou_device, #if defined(CONFIG_MMC_SH_MMCIF) || defined(CONFIG_MMC_SH_MMCIF_MODULE) diff --git a/arch/sh/boards/mach-se/7724/setup.c b/arch/sh/boards/mach-se/7724/setup.c index dd931e36daf..ffbf5bc7366 100644 --- a/arch/sh/boards/mach-se/7724/setup.c +++ b/arch/sh/boards/mach-se/7724/setup.c @@ -29,6 +29,7 @@ #include <video/sh_mobile_lcdc.h> #include <media/sh_mobile_ceu.h> #include <sound/sh_fsi.h> +#include <sound/simple_card.h> #include <asm/io.h> #include <asm/heartbeat.h> #include <asm/clock.h> @@ -305,17 +306,25 @@ static struct platform_device fsi_device = { }, }; -static struct fsi_ak4642_info fsi_ak4642_info = { +static struct asoc_simple_dai_init_info fsi2_ak4642_init_info = { + .fmt = SND_SOC_DAIFMT_LEFT_J, + .codec_daifmt = SND_SOC_DAIFMT_CBM_CFM, + .cpu_daifmt = SND_SOC_DAIFMT_CBS_CFS, + .sysclk = 11289600, +}; + +static struct asoc_simple_card_info fsi_ak4642_info = { .name = "AK4642", .card = "FSIA-AK4642", .cpu_dai = "fsia-dai", .codec = "ak4642-codec.0-0012", .platform = "sh_fsi.0", - .id = FSI_PORT_A, + .codec_dai = "ak4642-hifi", + .init = &fsi2_ak4642_init_info, }; static struct platform_device fsi_ak4642_device = { - .name = "fsi-ak4642-audio", + .name = "asoc-simple-card", .dev = { .platform_data = &fsi_ak4642_info, }, diff --git a/drivers/mfd/mc13xxx-core.c b/drivers/mfd/mc13xxx-core.c index 9fd4f63c45c..738722cdeca 100644 --- a/drivers/mfd/mc13xxx-core.c +++ b/drivers/mfd/mc13xxx-core.c @@ -813,7 +813,8 @@ err_revision: mc13xxx_add_subdevice(mc13xxx, "%s-adc"); if (mc13xxx->flags & MC13XXX_USE_CODEC) - mc13xxx_add_subdevice(mc13xxx, "%s-codec"); + mc13xxx_add_subdevice_pdata(mc13xxx, "%s-codec", + pdata->codec, sizeof(*pdata->codec)); if (mc13xxx->flags & MC13XXX_USE_RTC) mc13xxx_add_subdevice(mc13xxx, "%s-rtc"); diff --git a/include/linux/mfd/mc13xxx.h b/include/linux/mfd/mc13xxx.h index 10e038bac8d..bf070755982 100644 --- a/include/linux/mfd/mc13xxx.h +++ b/include/linux/mfd/mc13xxx.h @@ -170,6 +170,16 @@ struct mc13xxx_ts_platform_data { bool atox; }; +enum mc13783_ssi_port { + MC13783_SSI1_PORT, + MC13783_SSI2_PORT, +}; + +struct mc13xxx_codec_platform_data { + enum mc13783_ssi_port adc_ssi_port; + enum mc13783_ssi_port dac_ssi_port; +}; + struct mc13xxx_platform_data { #define MC13XXX_USE_TOUCHSCREEN (1 << 0) #define MC13XXX_USE_CODEC (1 << 1) @@ -181,6 +191,7 @@ struct mc13xxx_platform_data { struct mc13xxx_leds_platform_data *leds; struct mc13xxx_buttons_platform_data *buttons; struct mc13xxx_ts_platform_data touch; + struct mc13xxx_codec_platform_data *codec; }; #define MC13XXX_ADC_MODE_TS 1 diff --git a/include/sound/asound.h b/include/sound/asound.h index a2e4ff5ba9e..0876a1e76ae 100644 --- a/include/sound/asound.h +++ b/include/sound/asound.h @@ -70,6 +70,20 @@ struct snd_aes_iec958 { /**************************************************************************** * * + * CEA-861 Audio InfoFrame. Used in HDMI and DisplayPort * + * * + ****************************************************************************/ + +struct snd_cea_861_aud_if { + unsigned char db1_ct_cc; /* coding type and channel count */ + unsigned char db2_sf_ss; /* sample frequency and size */ + unsigned char db3; /* not used, all zeros */ + unsigned char db4_ca; /* channel allocation code */ + unsigned char db5_dminh_lsv; /* downmix inhibit & level-shit values */ +}; + +/**************************************************************************** + * * * Section for driver hardware dependent interface - /dev/snd/hw? * * * ****************************************************************************/ diff --git a/include/sound/asoundef.h b/include/sound/asoundef.h index 20ebf3298eb..bb05c02f89b 100644 --- a/include/sound/asoundef.h +++ b/include/sound/asoundef.h @@ -170,6 +170,47 @@ #define IEC958_AES5_CON_CGMSA_COPYNOMORE (2<<0) /* condition not be used */ #define IEC958_AES5_CON_CGMSA_COPYNEVER (3<<0) /* no copying is permitted */ +/**************************************************************************** + * * + * CEA-861 Audio InfoFrame. Used in HDMI and DisplayPort * + * * + ****************************************************************************/ +#define CEA861_AUDIO_INFOFRAME_DB1CC (7<<0) /* mask - channel count */ +#define CEA861_AUDIO_INFOFRAME_DB1CT (0xf<<4) /* mask - coding type */ +#define CEA861_AUDIO_INFOFRAME_DB1CT_FROM_STREAM (0<<4) /* refer to stream */ +#define CEA861_AUDIO_INFOFRAME_DB1CT_IEC60958 (1<<4) /* IEC-60958 L-PCM */ +#define CEA861_AUDIO_INFOFRAME_DB1CT_AC3 (2<<4) /* AC-3 */ +#define CEA861_AUDIO_INFOFRAME_DB1CT_MPEG1 (3<<4) /* MPEG1 Layers 1 & 2 */ +#define CEA861_AUDIO_INFOFRAME_DB1CT_MP3 (4<<4) /* MPEG1 Layer 3 */ +#define CEA861_AUDIO_INFOFRAME_DB1CT_MPEG2_MULTICH (5<<4) /* MPEG2 Multichannel */ +#define CEA861_AUDIO_INFOFRAME_DB1CT_AAC (6<<4) /* AAC */ +#define CEA861_AUDIO_INFOFRAME_DB1CT_DTS (7<<4) /* DTS */ +#define CEA861_AUDIO_INFOFRAME_DB1CT_ATRAC (8<<4) /* ATRAC */ +#define CEA861_AUDIO_INFOFRAME_DB1CT_ONEBIT (9<<4) /* One Bit Audio */ +#define CEA861_AUDIO_INFOFRAME_DB1CT_DOLBY_DIG_PLUS (10<<4) /* Dolby Digital + */ +#define CEA861_AUDIO_INFOFRAME_DB1CT_DTS_HD (11<<4) /* DTS-HD */ +#define CEA861_AUDIO_INFOFRAME_DB1CT_MAT (12<<4) /* MAT (MLP) */ +#define CEA861_AUDIO_INFOFRAME_DB1CT_DST (13<<4) /* DST */ +#define CEA861_AUDIO_INFOFRAME_DB1CT_WMA_PRO (14<<4) /* WMA Pro */ +#define CEA861_AUDIO_INFOFRAME_DB2SF (7<<2) /* mask - sample frequency */ +#define CEA861_AUDIO_INFOFRAME_DB2SF_FROM_STREAM (0<<2) /* refer to stream */ +#define CEA861_AUDIO_INFOFRAME_DB2SF_32000 (1<<2) /* 32kHz */ +#define CEA861_AUDIO_INFOFRAME_DB2SF_44100 (2<<2) /* 44.1kHz */ +#define CEA861_AUDIO_INFOFRAME_DB2SF_48000 (3<<2) /* 48kHz */ +#define CEA861_AUDIO_INFOFRAME_DB2SF_88200 (4<<2) /* 88.2kHz */ +#define CEA861_AUDIO_INFOFRAME_DB2SF_96000 (5<<2) /* 96kHz */ +#define CEA861_AUDIO_INFOFRAME_DB2SF_176400 (6<<2) /* 176.4kHz */ +#define CEA861_AUDIO_INFOFRAME_DB2SF_192000 (7<<2) /* 192kHz */ +#define CEA861_AUDIO_INFOFRAME_DB2SS (3<<0) /* mask - sample size */ +#define CEA861_AUDIO_INFOFRAME_DB2SS_FROM_STREAM (0<<0) /* refer to stream */ +#define CEA861_AUDIO_INFOFRAME_DB2SS_16BIT (1<<0) /* 16 bits */ +#define CEA861_AUDIO_INFOFRAME_DB2SS_20BIT (2<<0) /* 20 bits */ +#define CEA861_AUDIO_INFOFRAME_DB2SS_24BIT (3<<0) /* 24 bits */ +#define CEA861_AUDIO_INFOFRAME_DB5_DM_INH (1<<7) /* mask - inhibit downmixing */ +#define CEA861_AUDIO_INFOFRAME_DB5_DM_INH_PERMITTED (0<<7) /* stereo downmix permitted */ +#define CEA861_AUDIO_INFOFRAME_DB5_DM_INH_PROHIBITED (1<<7) /* stereo downmis prohibited */ +#define CEA861_AUDIO_INFOFRAME_DB5_LSV (0xf<<3) /* mask - level-shift values */ + /***************************************************************************** * * * MIDI v1.0 interface * diff --git a/include/sound/cs42l52.h b/include/sound/cs42l52.h new file mode 100644 index 00000000000..4c68955f733 --- /dev/null +++ b/include/sound/cs42l52.h @@ -0,0 +1,36 @@ +/* + * linux/sound/cs42l52.h -- Platform data for CS42L52 + * + * Copyright (c) 2012 Cirrus Logic Inc. + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License version 2 as + * published by the Free Software Foundation. + */ + +#ifndef __CS42L52_H +#define __CS42L52_H + +struct cs42l52_platform_data { + + /* MICBIAS Level. Check datasheet Pg48 */ + unsigned int micbias_lvl; + + /* MICA mode selection 0=Single 1=Differential */ + unsigned int mica_cfg; + + /* MICB mode selection 0=Single 1=Differential */ + unsigned int micb_cfg; + + /* MICA Select 0=MIC1A 1=MIC2A */ + unsigned int mica_sel; + + /* MICB Select 0=MIC2A 1=MIC2B */ + unsigned int micb_sel; + + /* Charge Pump Freq. Check datasheet Pg73 */ + unsigned int chgfreq; + +}; + +#endif /* __CS42L52_H */ diff --git a/include/sound/max98095.h b/include/sound/max98095.h index 7513a42dd4a..e87ae67b0a5 100644 --- a/include/sound/max98095.h +++ b/include/sound/max98095.h @@ -49,6 +49,18 @@ struct max98095_pdata { */ unsigned int digmic_left_mode:1; unsigned int digmic_right_mode:1; + + /* Pin5 is the mechanical method of sensing jack insertion + * but it is something that might not be supported. + * 0 = PIN5 not supported + * 1 = PIN5 supported + */ + unsigned int jack_detect_pin5en:1; + + /* Slew amount for jack detection. Calculated as 4 * (delay + 1). + * Default delay is 24 to get a time of 100ms. + */ + unsigned int jack_detect_delay; }; #endif diff --git a/include/sound/sh_fsi.h b/include/sound/sh_fsi.h index b457e87fbd0..906010344dd 100644 --- a/include/sound/sh_fsi.h +++ b/include/sound/sh_fsi.h @@ -21,10 +21,11 @@ /* * flags format * - * 0x000000BA + * 0x00000CBA * * A: inversion * B: format mode + * C: chip specific */ /* A: clock inversion */ @@ -39,6 +40,9 @@ #define SH_FSI_FMT_DAI (0 << 4) #define SH_FSI_FMT_SPDIF (1 << 4) +/* C: chip specific */ +#define SH_FSI_OPTION_MASK 0x00000F00 +#define SH_FSI_ENABLE_STREAM_MODE (1 << 8) /* for 16bit data */ /* * set_rate return value @@ -84,16 +88,4 @@ struct sh_fsi_platform_info { struct sh_fsi_port_info port_b; }; -/* - * for fsi-ak4642 - */ -struct fsi_ak4642_info { - const char *name; - const char *card; - const char *cpu_dai; - const char *codec; - const char *platform; - int id; -}; - #endif /* __SOUND_FSI_H */ diff --git a/include/sound/simple_card.h b/include/sound/simple_card.h new file mode 100644 index 00000000000..4b62b8dc6a4 --- /dev/null +++ b/include/sound/simple_card.h @@ -0,0 +1,38 @@ +/* + * ASoC simple sound card support + * + * Copyright (C) 2012 Renesas Solutions Corp. + * Kuninori Morimoto <kuninori.morimoto.gx@renesas.com> + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License version 2 as + * published by the Free Software Foundation. + */ + +#ifndef __SIMPLE_CARD_H +#define __SIMPLE_CARD_H + +#include <sound/soc.h> + +struct asoc_simple_dai_init_info { + unsigned int fmt; + unsigned int cpu_daifmt; + unsigned int codec_daifmt; + unsigned int sysclk; +}; + +struct asoc_simple_card_info { + const char *name; + const char *card; + const char *cpu_dai; + const char *codec; + const char *platform; + const char *codec_dai; + struct asoc_simple_dai_init_info *init; /* for snd_link.init */ + + /* used in simple-card.c */ + struct snd_soc_dai_link snd_link; + struct snd_soc_card snd_card; +}; + +#endif /* __SIMPLE_CARD_H */ diff --git a/include/sound/soc-dai.h b/include/sound/soc-dai.h index c429f248cf4..1f69e0af294 100644 --- a/include/sound/soc-dai.h +++ b/include/sound/soc-dai.h @@ -173,6 +173,8 @@ struct snd_soc_dai_ops { struct snd_soc_dai *); int (*trigger)(struct snd_pcm_substream *, int, struct snd_soc_dai *); + int (*bespoke_trigger)(struct snd_pcm_substream *, int, + struct snd_soc_dai *); /* * For hardware based FIFO caused delay reporting. * Optional. @@ -196,6 +198,7 @@ struct snd_soc_dai_driver { const char *name; unsigned int id; int ac97_control; + unsigned int base; /* DAI driver callbacks */ int (*probe)(struct snd_soc_dai *dai); @@ -241,6 +244,7 @@ struct snd_soc_dai { struct snd_soc_dapm_widget *playback_widget; struct snd_soc_dapm_widget *capture_widget; + struct snd_soc_dapm_context dapm; /* DAI DMA data */ void *playback_dma_data; diff --git a/include/sound/soc-dapm.h b/include/sound/soc-dapm.h index 8da3c240906..e3833d9f191 100644 --- a/include/sound/soc-dapm.h +++ b/include/sound/soc-dapm.h @@ -141,10 +141,6 @@ struct device; { .id = snd_soc_dapm_mixer, .name = wname, .reg = wreg, .shift = wshift, \ .invert = winvert, .kcontrol_news = wcontrols, \ .num_kcontrols = wncontrols, .event = wevent, .event_flags = wflags} -#define SND_SOC_DAPM_MICBIAS_E(wname, wreg, wshift, winvert, wevent, wflags) \ -{ .id = snd_soc_dapm_micbias, .name = wname, .reg = wreg, .shift = wshift, \ - .invert = winvert, .kcontrol_news = NULL, .num_kcontrols = 0, \ - .event = wevent, .event_flags = wflags} #define SND_SOC_DAPM_SWITCH_E(wname, wreg, wshift, winvert, wcontrols, \ wevent, wflags) \ { .id = snd_soc_dapm_switch, .name = wname, .reg = wreg, .shift = wshift, \ @@ -324,6 +320,8 @@ struct snd_soc_dapm_path; struct snd_soc_dapm_pin; struct snd_soc_dapm_route; struct snd_soc_dapm_context; +struct regulator; +struct snd_soc_dapm_widget_list; int dapm_reg_event(struct snd_soc_dapm_widget *w, struct snd_kcontrol *kcontrol, int event); @@ -359,6 +357,10 @@ int snd_soc_dapm_new_controls(struct snd_soc_dapm_context *dapm, int snd_soc_dapm_new_dai_widgets(struct snd_soc_dapm_context *dapm, struct snd_soc_dai *dai); int snd_soc_dapm_link_dai_widgets(struct snd_soc_card *card); +int snd_soc_dapm_new_pcm(struct snd_soc_card *card, + const struct snd_soc_pcm_stream *params, + struct snd_soc_dapm_widget *source, + struct snd_soc_dapm_widget *sink); /* dapm path setup */ int snd_soc_dapm_new_widgets(struct snd_soc_dapm_context *dapm); @@ -369,8 +371,8 @@ int snd_soc_dapm_weak_routes(struct snd_soc_dapm_context *dapm, const struct snd_soc_dapm_route *route, int num); /* dapm events */ -int snd_soc_dapm_stream_event(struct snd_soc_pcm_runtime *rtd, int stream, - struct snd_soc_dai *dai, int event); +void snd_soc_dapm_stream_event(struct snd_soc_pcm_runtime *rtd, int stream, + int event); void snd_soc_dapm_shutdown(struct snd_soc_card *card); /* external DAPM widget events */ @@ -402,6 +404,10 @@ void snd_soc_dapm_auto_nc_codec_pins(struct snd_soc_codec *codec); /* Mostly internal - should not normally be used */ void dapm_mark_dirty(struct snd_soc_dapm_widget *w, const char *reason); +/* dapm path query */ +int snd_soc_dapm_dai_get_connected_widgets(struct snd_soc_dai *dai, int stream, + struct snd_soc_dapm_widget_list **list); + /* dapm widget types */ enum snd_soc_dapm_type { snd_soc_dapm_input = 0, /* input pin */ @@ -430,6 +436,12 @@ enum snd_soc_dapm_type { snd_soc_dapm_aif_out, /* audio interface output */ snd_soc_dapm_siggen, /* signal generator */ snd_soc_dapm_dai, /* link to DAI structure */ + snd_soc_dapm_dai_link, /* link between two DAI structures */ +}; + +enum snd_soc_dapm_subclass { + SND_SOC_DAPM_CLASS_INIT = 0, + SND_SOC_DAPM_CLASS_RUNTIME = 1, }; /* @@ -482,9 +494,11 @@ struct snd_soc_dapm_widget { struct snd_soc_dapm_context *dapm; void *priv; /* widget specific data */ + struct regulator *regulator; /* attached regulator */ + const struct snd_soc_pcm_stream *params; /* params for dai links */ /* dapm control */ - short reg; /* negative reg = no direct dapm */ + int reg; /* negative reg = no direct dapm */ unsigned char shift; /* bits to shift */ unsigned int saved_value; /* widget saved value */ unsigned int value; /* widget current value */ diff --git a/include/sound/soc-dpcm.h b/include/sound/soc-dpcm.h new file mode 100644 index 00000000000..04598f1efd7 --- /dev/null +++ b/include/sound/soc-dpcm.h @@ -0,0 +1,138 @@ +/* + * linux/sound/soc-dpcm.h -- ALSA SoC Dynamic PCM Support + * + * Author: Liam Girdwood <lrg@ti.com> + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License version 2 as + * published by the Free Software Foundation. + */ + +#ifndef __LINUX_SND_SOC_DPCM_H +#define __LINUX_SND_SOC_DPCM_H + +#include <linux/list.h> +#include <sound/pcm.h> + +struct snd_soc_pcm_runtime; + +/* + * Types of runtime_update to perform. e.g. originated from FE PCM ops + * or audio route changes triggered by muxes/mixers. + */ +enum snd_soc_dpcm_update { + SND_SOC_DPCM_UPDATE_NO = 0, + SND_SOC_DPCM_UPDATE_BE, + SND_SOC_DPCM_UPDATE_FE, +}; + +/* + * Dynamic PCM Frontend -> Backend link management states. + */ +enum snd_soc_dpcm_link_state { + SND_SOC_DPCM_LINK_STATE_NEW = 0, /* newly created link */ + SND_SOC_DPCM_LINK_STATE_FREE, /* link to be dismantled */ +}; + +/* + * Dynamic PCM Frontend -> Backend link PCM states. + */ +enum snd_soc_dpcm_state { + SND_SOC_DPCM_STATE_NEW = 0, + SND_SOC_DPCM_STATE_OPEN, + SND_SOC_DPCM_STATE_HW_PARAMS, + SND_SOC_DPCM_STATE_PREPARE, + SND_SOC_DPCM_STATE_START, + SND_SOC_DPCM_STATE_STOP, + SND_SOC_DPCM_STATE_PAUSED, + SND_SOC_DPCM_STATE_SUSPEND, + SND_SOC_DPCM_STATE_HW_FREE, + SND_SOC_DPCM_STATE_CLOSE, +}; + +/* + * Dynamic PCM trigger ordering. Triggering flexibility is required as some + * DSPs require triggering before/after their CPU platform and DAIs. + * + * i.e. some clients may want to manually order this call in their PCM + * trigger() whilst others will just use the regular core ordering. + */ +enum snd_soc_dpcm_trigger { + SND_SOC_DPCM_TRIGGER_PRE = 0, + SND_SOC_DPCM_TRIGGER_POST, + SND_SOC_DPCM_TRIGGER_BESPOKE, +}; + +/* + * Dynamic PCM link + * This links together a FE and BE DAI at runtime and stores the link + * state information and the hw_params configuration. + */ +struct snd_soc_dpcm { + /* FE and BE DAIs*/ + struct snd_soc_pcm_runtime *be; + struct snd_soc_pcm_runtime *fe; + + /* link state */ + enum snd_soc_dpcm_link_state state; + + /* list of BE and FE for this DPCM link */ + struct list_head list_be; + struct list_head list_fe; + + /* hw params for this link - may be different for each link */ + struct snd_pcm_hw_params hw_params; +#ifdef CONFIG_DEBUG_FS + struct dentry *debugfs_state; +#endif +}; + +/* + * Dynamic PCM runtime data. + */ +struct snd_soc_dpcm_runtime { + struct list_head be_clients; + struct list_head fe_clients; + + int users; + struct snd_pcm_runtime *runtime; + struct snd_pcm_hw_params hw_params; + + /* state and update */ + enum snd_soc_dpcm_update runtime_update; + enum snd_soc_dpcm_state state; +}; + +/* can this BE stop and free */ +int snd_soc_dpcm_can_be_free_stop(struct snd_soc_pcm_runtime *fe, + struct snd_soc_pcm_runtime *be, int stream); + +/* can this BE perform a hw_params() */ +int snd_soc_dpcm_can_be_params(struct snd_soc_pcm_runtime *fe, + struct snd_soc_pcm_runtime *be, int stream); + +/* is the current PCM operation for this FE ? */ +int snd_soc_dpcm_fe_can_update(struct snd_soc_pcm_runtime *fe, int stream); + +/* is the current PCM operation for this BE ? */ +int snd_soc_dpcm_be_can_update(struct snd_soc_pcm_runtime *fe, + struct snd_soc_pcm_runtime *be, int stream); + +/* get the substream for this BE */ +struct snd_pcm_substream * + snd_soc_dpcm_get_substream(struct snd_soc_pcm_runtime *be, int stream); + +/* get the BE runtime state */ +enum snd_soc_dpcm_state + snd_soc_dpcm_be_get_state(struct snd_soc_pcm_runtime *be, int stream); + +/* set the BE runtime state */ +void snd_soc_dpcm_be_set_state(struct snd_soc_pcm_runtime *be, int stream, + enum snd_soc_dpcm_state state); + +/* internal use only */ +int soc_dpcm_be_digital_mute(struct snd_soc_pcm_runtime *fe, int mute); +int soc_dpcm_debugfs_add(struct snd_soc_pcm_runtime *rtd); +int soc_dpcm_runtime_update(struct snd_soc_dapm_widget *); + +#endif diff --git a/include/sound/soc.h b/include/sound/soc.h index 2ebf7877c14..c703871f5f6 100644 --- a/include/sound/soc.h +++ b/include/sound/soc.h @@ -55,6 +55,18 @@ .info = snd_soc_info_volsw, .get = snd_soc_get_volsw,\ .put = snd_soc_put_volsw, \ .private_value = SOC_SINGLE_VALUE(reg, shift, max, invert) } +#define SOC_SINGLE_SX_TLV(xname, xreg, xshift, xmin, xmax, tlv_array) \ +{ .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = xname, \ + .access = SNDRV_CTL_ELEM_ACCESS_TLV_READ | \ + SNDRV_CTL_ELEM_ACCESS_READWRITE, \ + .tlv.p = (tlv_array),\ + .info = snd_soc_info_volsw, \ + .get = snd_soc_get_volsw_sx,\ + .put = snd_soc_put_volsw_sx, \ + .private_value = (unsigned long)&(struct soc_mixer_control) \ + {.reg = xreg, .rreg = xreg, \ + .shift = xshift, .rshift = xshift, \ + .max = xmax, .min = xmin} } #define SOC_DOUBLE(xname, reg, shift_left, shift_right, max, invert) \ { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = (xname),\ .info = snd_soc_info_volsw, .get = snd_soc_get_volsw, \ @@ -85,6 +97,18 @@ .get = snd_soc_get_volsw, .put = snd_soc_put_volsw, \ .private_value = SOC_DOUBLE_R_VALUE(reg_left, reg_right, xshift, \ xmax, xinvert) } +#define SOC_DOUBLE_R_SX_TLV(xname, xreg, xrreg, xshift, xmin, xmax, tlv_array) \ +{ .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = (xname), \ + .access = SNDRV_CTL_ELEM_ACCESS_TLV_READ | \ + SNDRV_CTL_ELEM_ACCESS_READWRITE, \ + .tlv.p = (tlv_array), \ + .info = snd_soc_info_volsw, \ + .get = snd_soc_get_volsw_sx, \ + .put = snd_soc_put_volsw_sx, \ + .private_value = (unsigned long)&(struct soc_mixer_control) \ + {.reg = xreg, .rreg = xrreg, \ + .shift = xshift, .rshift = xshift, \ + .max = xmax, .min = xmin} } #define SOC_DOUBLE_S8_TLV(xname, xreg, xmin, xmax, tlv_array) \ { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = (xname), \ .access = SNDRV_CTL_ELEM_ACCESS_TLV_READ | \ @@ -171,20 +195,6 @@ .get = xhandler_get, .put = xhandler_put, \ .private_value = (unsigned long)&xenum } -#define SOC_DOUBLE_R_SX_TLV(xname, xreg_left, xreg_right, xshift,\ - xmin, xmax, tlv_array) \ -{ .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = (xname), \ - .access = SNDRV_CTL_ELEM_ACCESS_TLV_READ | \ - SNDRV_CTL_ELEM_ACCESS_READWRITE, \ - .tlv.p = (tlv_array), \ - .info = snd_soc_info_volsw_2r_sx, \ - .get = snd_soc_get_volsw_2r_sx, \ - .put = snd_soc_put_volsw_2r_sx, \ - .private_value = (unsigned long)&(struct soc_mixer_control) \ - {.reg = xreg_left, \ - .rreg = xreg_right, .shift = xshift, \ - .min = xmin, .max = xmax} } - #define SND_SOC_BYTES(xname, xbase, xregs) \ { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = xname, \ .info = snd_soc_bytes_info, .get = snd_soc_bytes_get, \ @@ -200,6 +210,19 @@ {.base = xbase, .num_regs = xregs, \ .mask = xmask }) } +#define SOC_SINGLE_XR_SX(xname, xregbase, xregcount, xnbits, \ + xmin, xmax, xinvert) \ +{ .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = (xname), \ + .info = snd_soc_info_xr_sx, .get = snd_soc_get_xr_sx, \ + .put = snd_soc_put_xr_sx, \ + .private_value = (unsigned long)&(struct soc_mreg_control) \ + {.regbase = xregbase, .regcount = xregcount, .nbits = xnbits, \ + .invert = xinvert, .min = xmin, .max = xmax} } + +#define SOC_SINGLE_STROBE(xname, xreg, xshift, xinvert) \ + SOC_SINGLE_EXT(xname, xreg, xshift, 1, xinvert, \ + snd_soc_get_strobe, snd_soc_put_strobe) + /* * Simplified versions of above macros, declaring a struct and calculating * ARRAY_SIZE internally @@ -264,6 +287,7 @@ struct snd_soc_jack_zone; struct snd_soc_jack_pin; struct snd_soc_cache_ops; #include <sound/soc-dapm.h> +#include <sound/soc-dpcm.h> #ifdef CONFIG_GPIOLIB struct snd_soc_jack_gpio; @@ -288,6 +312,11 @@ enum snd_soc_pcm_subclass { SND_SOC_PCM_CLASS_BE = 1, }; +enum snd_soc_card_subclass { + SND_SOC_CARD_CLASS_INIT = 0, + SND_SOC_CARD_CLASS_RUNTIME = 1, +}; + int snd_soc_codec_set_sysclk(struct snd_soc_codec *codec, int clk_id, int source, unsigned int freq, int dir); int snd_soc_codec_set_pll(struct snd_soc_codec *codec, int pll_id, int source, @@ -333,6 +362,11 @@ int snd_soc_platform_write(struct snd_soc_platform *platform, unsigned int reg, unsigned int val); int soc_new_pcm(struct snd_soc_pcm_runtime *rtd, int num); +struct snd_pcm_substream *snd_soc_get_dai_substream(struct snd_soc_card *card, + const char *dai_link, int stream); +struct snd_soc_pcm_runtime *snd_soc_get_pcm_runtime(struct snd_soc_card *card, + const char *dai_link); + /* Utility functions to get clock rates from various things */ int snd_soc_calc_frame_size(int sample_size, int channels, int tdm_slots); int snd_soc_params_to_frame_size(struct snd_pcm_hw_params *params); @@ -343,6 +377,9 @@ int snd_soc_params_to_bclk(struct snd_pcm_hw_params *parms); int snd_soc_set_runtime_hwparams(struct snd_pcm_substream *substream, const struct snd_pcm_hardware *hw); +int snd_soc_platform_trigger(struct snd_pcm_substream *substream, + int cmd, struct snd_soc_platform *platform); + /* Jack reporting */ int snd_soc_jack_new(struct snd_soc_codec *codec, const char *id, int type, struct snd_soc_jack *jack); @@ -413,6 +450,10 @@ int snd_soc_put_volsw(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol); #define snd_soc_get_volsw_2r snd_soc_get_volsw #define snd_soc_put_volsw_2r snd_soc_put_volsw +int snd_soc_get_volsw_sx(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol); +int snd_soc_put_volsw_sx(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol); int snd_soc_info_volsw_s8(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_info *uinfo); int snd_soc_get_volsw_s8(struct snd_kcontrol *kcontrol, @@ -421,19 +462,22 @@ int snd_soc_put_volsw_s8(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol); int snd_soc_limit_volume(struct snd_soc_codec *codec, const char *name, int max); -int snd_soc_info_volsw_2r_sx(struct snd_kcontrol *kcontrol, - struct snd_ctl_elem_info *uinfo); -int snd_soc_get_volsw_2r_sx(struct snd_kcontrol *kcontrol, - struct snd_ctl_elem_value *ucontrol); -int snd_soc_put_volsw_2r_sx(struct snd_kcontrol *kcontrol, - struct snd_ctl_elem_value *ucontrol); int snd_soc_bytes_info(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_info *uinfo); int snd_soc_bytes_get(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol); int snd_soc_bytes_put(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol); - +int snd_soc_info_xr_sx(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_info *uinfo); +int snd_soc_get_xr_sx(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol); +int snd_soc_put_xr_sx(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol); +int snd_soc_get_strobe(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol); +int snd_soc_put_strobe(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol); /** * struct snd_soc_reg_access - Describes whether a given register is @@ -513,6 +557,7 @@ struct snd_soc_jack_gpio { #endif struct snd_soc_jack { + struct mutex mutex; struct snd_jack *jack; struct snd_soc_codec *codec; struct list_head pins; @@ -711,6 +756,7 @@ struct snd_soc_platform_driver { /* platform IO - used for platform DAPM */ unsigned int (*read)(struct snd_soc_platform *, unsigned int); int (*write)(struct snd_soc_platform *, unsigned int, unsigned int); + int (*bespoke_trigger)(struct snd_pcm_substream *, int); }; struct snd_soc_platform { @@ -746,21 +792,36 @@ struct snd_soc_dai_link { const char *cpu_dai_name; const struct device_node *cpu_dai_of_node; const char *codec_dai_name; + int be_id; /* optional ID for machine driver BE identification */ + + const struct snd_soc_pcm_stream *params; unsigned int dai_fmt; /* format to set on init */ + enum snd_soc_dpcm_trigger trigger[2]; /* trigger type for DPCM */ + /* Keep DAI active over suspend */ unsigned int ignore_suspend:1; /* Symmetry requirements */ unsigned int symmetric_rates:1; + /* Do not create a PCM for this DAI link (Backend link) */ + unsigned int no_pcm:1; + + /* This DAI link can route to other DAI links at runtime (Frontend)*/ + unsigned int dynamic:1; + /* pmdown_time is ignored at stop */ unsigned int ignore_pmdown_time:1; /* codec/machine specific init - e.g. add machine controls */ int (*init)(struct snd_soc_pcm_runtime *rtd); + /* optional hw_params re-writing for BE and FE sync */ + int (*be_hw_params_fixup)(struct snd_soc_pcm_runtime *rtd, + struct snd_pcm_hw_params *params); + /* machine stream operations */ struct snd_soc_ops *ops; }; @@ -800,6 +861,7 @@ struct snd_soc_card { struct list_head list; struct mutex mutex; + struct mutex dapm_mutex; bool instantiated; @@ -889,9 +951,11 @@ struct snd_soc_pcm_runtime { enum snd_soc_pcm_subclass pcm_subclass; struct snd_pcm_ops ops; - unsigned int complete:1; unsigned int dev_registered:1; + /* Dynamic PCM BE runtime data */ + struct snd_soc_dpcm_runtime dpcm[2]; + long pmdown_time; /* runtime devices */ @@ -902,6 +966,10 @@ struct snd_soc_pcm_runtime { struct snd_soc_dai *cpu_dai; struct delayed_work delayed_work; +#ifdef CONFIG_DEBUG_FS + struct dentry *debugfs_dpcm_root; + struct dentry *debugfs_dpcm_state; +#endif }; /* mixer control */ @@ -916,6 +984,12 @@ struct soc_bytes { u32 mask; }; +/* multi register control */ +struct soc_mreg_control { + long min, max; + unsigned int regbase, regcount, nbits, invert; +}; + /* enumerated kcontrol */ struct soc_enum { unsigned short reg; diff --git a/include/trace/events/asoc.h b/include/trace/events/asoc.h index ab26f8aa3c7..5fc2dcdd21c 100644 --- a/include/trace/events/asoc.h +++ b/include/trace/events/asoc.h @@ -7,6 +7,8 @@ #include <linux/ktime.h> #include <linux/tracepoint.h> +#define DAPM_DIRECT "(direct)" + struct snd_soc_jack; struct snd_soc_codec; struct snd_soc_platform; @@ -241,6 +243,84 @@ TRACE_EVENT(snd_soc_dapm_walk_done, (int)__entry->path_checks, (int)__entry->neighbour_checks) ); +TRACE_EVENT(snd_soc_dapm_output_path, + + TP_PROTO(struct snd_soc_dapm_widget *widget, + struct snd_soc_dapm_path *path), + + TP_ARGS(widget, path), + + TP_STRUCT__entry( + __string( wname, widget->name ) + __string( pname, path->name ? path->name : DAPM_DIRECT) + __string( psname, path->sink->name ) + __field( int, path_sink ) + __field( int, path_connect ) + ), + + TP_fast_assign( + __assign_str(wname, widget->name); + __assign_str(pname, path->name ? path->name : DAPM_DIRECT); + __assign_str(psname, path->sink->name); + __entry->path_connect = path->connect; + __entry->path_sink = (long)path->sink; + ), + + TP_printk("%c%s -> %s -> %s\n", + (int) __entry->path_sink && + (int) __entry->path_connect ? '*' : ' ', + __get_str(wname), __get_str(pname), __get_str(psname)) +); + +TRACE_EVENT(snd_soc_dapm_input_path, + + TP_PROTO(struct snd_soc_dapm_widget *widget, + struct snd_soc_dapm_path *path), + + TP_ARGS(widget, path), + + TP_STRUCT__entry( + __string( wname, widget->name ) + __string( pname, path->name ? path->name : DAPM_DIRECT) + __string( psname, path->source->name ) + __field( int, path_source ) + __field( int, path_connect ) + ), + + TP_fast_assign( + __assign_str(wname, widget->name); + __assign_str(pname, path->name ? path->name : DAPM_DIRECT); + __assign_str(psname, path->source->name); + __entry->path_connect = path->connect; + __entry->path_source = (long)path->source; + ), + + TP_printk("%c%s <- %s <- %s\n", + (int) __entry->path_source && + (int) __entry->path_connect ? '*' : ' ', + __get_str(wname), __get_str(pname), __get_str(psname)) +); + +TRACE_EVENT(snd_soc_dapm_connected, + + TP_PROTO(int paths, int stream), + + TP_ARGS(paths, stream), + + TP_STRUCT__entry( + __field( int, paths ) + __field( int, stream ) + ), + + TP_fast_assign( + __entry->paths = paths; + __entry->stream = stream; + ), + + TP_printk("%s: found %d paths\n", + __entry->stream ? "capture" : "playback", __entry->paths) +); + TRACE_EVENT(snd_soc_jack_irq, TP_PROTO(const char *name), diff --git a/sound/atmel/ac97c.c b/sound/atmel/ac97c.c index 115313ef54d..f5ded640b39 100644 --- a/sound/atmel/ac97c.c +++ b/sound/atmel/ac97c.c @@ -991,6 +991,8 @@ static int __devinit atmel_ac97c_probe(struct platform_device *pdev) gpio_direction_output(pdata->reset_pin, 1); chip->reset_pin = pdata->reset_pin; } + } else { + chip->reset_pin = -EINVAL; } snd_card_set_dev(card, &pdev->dev); diff --git a/sound/core/jack.c b/sound/core/jack.c index 471e1e3b0a9..a06b1651fcb 100644 --- a/sound/core/jack.c +++ b/sound/core/jack.c @@ -155,7 +155,7 @@ EXPORT_SYMBOL(snd_jack_new); * @jack: The jack to configure * @parent: The device to set as parent for the jack. * - * Set the parent for the jack input device in the device tree. This + * Set the parent for the jack devices in the device tree. This * function is only valid prior to registration of the jack. If no * parent is configured then the parent device will be the sound card. */ @@ -179,6 +179,9 @@ EXPORT_SYMBOL(snd_jack_set_parent); * mapping is provided but keys are enabled in the jack type then * BTN_n numeric buttons will be reported. * + * If jacks are not reporting via the input API this call will have no + * effect. + * * Note that this is intended to be use by simple devices with small * numbers of keys that can be reported. It is also possible to * access the input device directly - devices with complex input diff --git a/sound/core/pcm_lib.c b/sound/core/pcm_lib.c index 4d18941178e..faedb1481b2 100644 --- a/sound/core/pcm_lib.c +++ b/sound/core/pcm_lib.c @@ -1894,6 +1894,7 @@ static snd_pcm_sframes_t snd_pcm_lib_write1(struct snd_pcm_substream *substream, struct snd_pcm_runtime *runtime = substream->runtime; snd_pcm_uframes_t xfer = 0; snd_pcm_uframes_t offset = 0; + snd_pcm_uframes_t avail; int err = 0; if (size == 0) @@ -1917,13 +1918,12 @@ static snd_pcm_sframes_t snd_pcm_lib_write1(struct snd_pcm_substream *substream, } runtime->twake = runtime->control->avail_min ? : 1; + if (runtime->status->state == SNDRV_PCM_STATE_RUNNING) + snd_pcm_update_hw_ptr(substream); + avail = snd_pcm_playback_avail(runtime); while (size > 0) { snd_pcm_uframes_t frames, appl_ptr, appl_ofs; - snd_pcm_uframes_t avail; snd_pcm_uframes_t cont; - if (runtime->status->state == SNDRV_PCM_STATE_RUNNING) - snd_pcm_update_hw_ptr(substream); - avail = snd_pcm_playback_avail(runtime); if (!avail) { if (nonblock) { err = -EAGAIN; @@ -1971,6 +1971,7 @@ static snd_pcm_sframes_t snd_pcm_lib_write1(struct snd_pcm_substream *substream, offset += frames; size -= frames; xfer += frames; + avail -= frames; if (runtime->status->state == SNDRV_PCM_STATE_PREPARED && snd_pcm_playback_hw_avail(runtime) >= (snd_pcm_sframes_t)runtime->start_threshold) { err = snd_pcm_start(substream); @@ -2111,6 +2112,7 @@ static snd_pcm_sframes_t snd_pcm_lib_read1(struct snd_pcm_substream *substream, struct snd_pcm_runtime *runtime = substream->runtime; snd_pcm_uframes_t xfer = 0; snd_pcm_uframes_t offset = 0; + snd_pcm_uframes_t avail; int err = 0; if (size == 0) @@ -2141,13 +2143,12 @@ static snd_pcm_sframes_t snd_pcm_lib_read1(struct snd_pcm_substream *substream, } runtime->twake = runtime->control->avail_min ? : 1; + if (runtime->status->state == SNDRV_PCM_STATE_RUNNING) + snd_pcm_update_hw_ptr(substream); + avail = snd_pcm_capture_avail(runtime); while (size > 0) { snd_pcm_uframes_t frames, appl_ptr, appl_ofs; - snd_pcm_uframes_t avail; snd_pcm_uframes_t cont; - if (runtime->status->state == SNDRV_PCM_STATE_RUNNING) - snd_pcm_update_hw_ptr(substream); - avail = snd_pcm_capture_avail(runtime); if (!avail) { if (runtime->status->state == SNDRV_PCM_STATE_DRAINING) { @@ -2202,6 +2203,7 @@ static snd_pcm_sframes_t snd_pcm_lib_read1(struct snd_pcm_substream *substream, offset += frames; size -= frames; xfer += frames; + avail -= frames; } _end_unlock: runtime->twake = 0; diff --git a/sound/core/pcm_native.c b/sound/core/pcm_native.c index 3fe99e644eb..53b5ada8f7c 100644 --- a/sound/core/pcm_native.c +++ b/sound/core/pcm_native.c @@ -1360,7 +1360,14 @@ static int snd_pcm_prepare(struct snd_pcm_substream *substream, static int snd_pcm_pre_drain_init(struct snd_pcm_substream *substream, int state) { - substream->runtime->trigger_master = substream; + struct snd_pcm_runtime *runtime = substream->runtime; + switch (runtime->status->state) { + case SNDRV_PCM_STATE_OPEN: + case SNDRV_PCM_STATE_DISCONNECTED: + case SNDRV_PCM_STATE_SUSPENDED: + return -EBADFD; + } + runtime->trigger_master = substream; return 0; } @@ -1379,6 +1386,9 @@ static int snd_pcm_do_drain_init(struct snd_pcm_substream *substream, int state) case SNDRV_PCM_STATE_RUNNING: runtime->status->state = SNDRV_PCM_STATE_DRAINING; break; + case SNDRV_PCM_STATE_XRUN: + runtime->status->state = SNDRV_PCM_STATE_SETUP; + break; default: break; } diff --git a/sound/core/sound_oss.c b/sound/core/sound_oss.c index c7009204306..e9528333e36 100644 --- a/sound/core/sound_oss.c +++ b/sound/core/sound_oss.c @@ -35,7 +35,7 @@ #include <linux/sound.h> #include <linux/mutex.h> -#define SNDRV_OSS_MINORS 128 +#define SNDRV_OSS_MINORS 256 static struct snd_minor *snd_oss_minors[SNDRV_OSS_MINORS]; static DEFINE_MUTEX(sound_oss_mutex); @@ -111,7 +111,7 @@ int snd_register_oss_device(int type, struct snd_card *card, int dev, int register1 = -1, register2 = -1; struct device *carddev = snd_card_get_device_link(card); - if (card && card->number >= 8) + if (card && card->number >= SNDRV_MINOR_OSS_DEVICES) return 0; /* ignore silently */ if (minor < 0) return minor; @@ -170,7 +170,7 @@ int snd_unregister_oss_device(int type, struct snd_card *card, int dev) int track2 = -1; struct snd_minor *mptr; - if (card && card->number >= 8) + if (card && card->number >= SNDRV_MINOR_OSS_DEVICES) return 0; if (minor < 0) return minor; diff --git a/sound/drivers/aloop.c b/sound/drivers/aloop.c index ad079b63b8b..8b5c36f4d30 100644 --- a/sound/drivers/aloop.c +++ b/sound/drivers/aloop.c @@ -117,6 +117,7 @@ struct loopback_pcm { /* timer stuff */ unsigned int irq_pos; /* fractional IRQ position */ unsigned int period_size_frac; + unsigned int last_drift; unsigned long last_jiffies; struct timer_list timer; }; @@ -264,6 +265,7 @@ static int loopback_trigger(struct snd_pcm_substream *substream, int cmd) return err; dpcm->last_jiffies = jiffies; dpcm->pcm_rate_shift = 0; + dpcm->last_drift = 0; spin_lock(&cable->lock); cable->running |= stream; cable->pause &= ~stream; @@ -444,34 +446,30 @@ static void copy_play_buf(struct loopback_pcm *play, } } -#define BYTEPOS_UPDATE_POSONLY 0 -#define BYTEPOS_UPDATE_CLEAR 1 -#define BYTEPOS_UPDATE_COPY 2 - -static void loopback_bytepos_update(struct loopback_pcm *dpcm, - unsigned int delta, - unsigned int cmd) +static inline unsigned int bytepos_delta(struct loopback_pcm *dpcm, + unsigned int jiffies_delta) { - unsigned int count; unsigned long last_pos; + unsigned int delta; last_pos = byte_pos(dpcm, dpcm->irq_pos); - dpcm->irq_pos += delta * dpcm->pcm_bps; - count = byte_pos(dpcm, dpcm->irq_pos) - last_pos; - if (!count) - return; - if (cmd == BYTEPOS_UPDATE_CLEAR) - clear_capture_buf(dpcm, count); - else if (cmd == BYTEPOS_UPDATE_COPY) - copy_play_buf(dpcm->cable->streams[SNDRV_PCM_STREAM_PLAYBACK], - dpcm->cable->streams[SNDRV_PCM_STREAM_CAPTURE], - count); - dpcm->buf_pos += count; - dpcm->buf_pos %= dpcm->pcm_buffer_size; + dpcm->irq_pos += jiffies_delta * dpcm->pcm_bps; + delta = byte_pos(dpcm, dpcm->irq_pos) - last_pos; + if (delta >= dpcm->last_drift) + delta -= dpcm->last_drift; + dpcm->last_drift = 0; if (dpcm->irq_pos >= dpcm->period_size_frac) { dpcm->irq_pos %= dpcm->period_size_frac; dpcm->period_update_pending = 1; } + return delta; +} + +static inline void bytepos_finish(struct loopback_pcm *dpcm, + unsigned int delta) +{ + dpcm->buf_pos += delta; + dpcm->buf_pos %= dpcm->pcm_buffer_size; } static unsigned int loopback_pos_update(struct loopback_cable *cable) @@ -481,7 +479,7 @@ static unsigned int loopback_pos_update(struct loopback_cable *cable) struct loopback_pcm *dpcm_capt = cable->streams[SNDRV_PCM_STREAM_CAPTURE]; unsigned long delta_play = 0, delta_capt = 0; - unsigned int running; + unsigned int running, count1, count2; unsigned long flags; spin_lock_irqsave(&cable->lock, flags); @@ -500,12 +498,13 @@ static unsigned int loopback_pos_update(struct loopback_cable *cable) goto unlock; if (delta_play > delta_capt) { - loopback_bytepos_update(dpcm_play, delta_play - delta_capt, - BYTEPOS_UPDATE_POSONLY); + count1 = bytepos_delta(dpcm_play, delta_play - delta_capt); + bytepos_finish(dpcm_play, count1); delta_play = delta_capt; } else if (delta_play < delta_capt) { - loopback_bytepos_update(dpcm_capt, delta_capt - delta_play, - BYTEPOS_UPDATE_CLEAR); + count1 = bytepos_delta(dpcm_capt, delta_capt - delta_play); + clear_capture_buf(dpcm_capt, count1); + bytepos_finish(dpcm_capt, count1); delta_capt = delta_play; } @@ -513,8 +512,17 @@ static unsigned int loopback_pos_update(struct loopback_cable *cable) goto unlock; /* note delta_capt == delta_play at this moment */ - loopback_bytepos_update(dpcm_capt, delta_capt, BYTEPOS_UPDATE_COPY); - loopback_bytepos_update(dpcm_play, delta_play, BYTEPOS_UPDATE_POSONLY); + count1 = bytepos_delta(dpcm_play, delta_play); + count2 = bytepos_delta(dpcm_capt, delta_capt); + if (count1 < count2) { + dpcm_capt->last_drift = count2 - count1; + count1 = count2; + } else if (count1 > count2) { + dpcm_play->last_drift = count1 - count2; + } + copy_play_buf(dpcm_play, dpcm_capt, count1); + bytepos_finish(dpcm_play, count1); + bytepos_finish(dpcm_capt, count1); unlock: spin_unlock_irqrestore(&cable->lock, flags); return running; diff --git a/sound/firewire/amdtp.c b/sound/firewire/amdtp.c index 87657dd7714..ea995af6d04 100644 --- a/sound/firewire/amdtp.c +++ b/sound/firewire/amdtp.c @@ -31,6 +31,8 @@ #define INTERRUPT_INTERVAL 16 #define QUEUE_LENGTH 48 +static void pcm_period_tasklet(unsigned long data); + /** * amdtp_out_stream_init - initialize an AMDTP output stream structure * @s: the AMDTP output stream to initialize @@ -47,6 +49,7 @@ int amdtp_out_stream_init(struct amdtp_out_stream *s, struct fw_unit *unit, s->flags = flags; s->context = ERR_PTR(-1); mutex_init(&s->mutex); + tasklet_init(&s->period_tasklet, pcm_period_tasklet, (unsigned long)s); s->packet_index = 0; return 0; @@ -164,6 +167,21 @@ void amdtp_out_stream_set_pcm_format(struct amdtp_out_stream *s, } EXPORT_SYMBOL(amdtp_out_stream_set_pcm_format); +/** + * amdtp_out_stream_pcm_prepare - prepare PCM device for running + * @s: the AMDTP output stream + * + * This function should be called from the PCM device's .prepare callback. + */ +void amdtp_out_stream_pcm_prepare(struct amdtp_out_stream *s) +{ + tasklet_kill(&s->period_tasklet); + s->pcm_buffer_pointer = 0; + s->pcm_period_pointer = 0; + s->pointer_flush = true; +} +EXPORT_SYMBOL(amdtp_out_stream_pcm_prepare); + static unsigned int calculate_data_blocks(struct amdtp_out_stream *s) { unsigned int phase, data_blocks; @@ -376,11 +394,21 @@ static void queue_out_packet(struct amdtp_out_stream *s, unsigned int cycle) s->pcm_period_pointer += data_blocks; if (s->pcm_period_pointer >= pcm->runtime->period_size) { s->pcm_period_pointer -= pcm->runtime->period_size; - snd_pcm_period_elapsed(pcm); + s->pointer_flush = false; + tasklet_hi_schedule(&s->period_tasklet); } } } +static void pcm_period_tasklet(unsigned long data) +{ + struct amdtp_out_stream *s = (void *)data; + struct snd_pcm_substream *pcm = ACCESS_ONCE(s->pcm); + + if (pcm) + snd_pcm_period_elapsed(pcm); +} + static void out_packet_callback(struct fw_iso_context *context, u32 cycle, size_t header_length, void *header, void *data) { @@ -506,6 +534,24 @@ err_unlock: EXPORT_SYMBOL(amdtp_out_stream_start); /** + * amdtp_out_stream_pcm_pointer - get the PCM buffer position + * @s: the AMDTP output stream that transports the PCM data + * + * Returns the current buffer position, in frames. + */ +unsigned long amdtp_out_stream_pcm_pointer(struct amdtp_out_stream *s) +{ + /* this optimization is allowed to be racy */ + if (s->pointer_flush) + fw_iso_context_flush_completions(s->context); + else + s->pointer_flush = true; + + return ACCESS_ONCE(s->pcm_buffer_pointer); +} +EXPORT_SYMBOL(amdtp_out_stream_pcm_pointer); + +/** * amdtp_out_stream_update - update the stream after a bus reset * @s: the AMDTP output stream */ @@ -532,6 +578,7 @@ void amdtp_out_stream_stop(struct amdtp_out_stream *s) return; } + tasklet_kill(&s->period_tasklet); fw_iso_context_stop(s->context); fw_iso_context_destroy(s->context); s->context = ERR_PTR(-1); diff --git a/sound/firewire/amdtp.h b/sound/firewire/amdtp.h index 537a9cb8358..b680c5ef01d 100644 --- a/sound/firewire/amdtp.h +++ b/sound/firewire/amdtp.h @@ -1,6 +1,7 @@ #ifndef SOUND_FIREWIRE_AMDTP_H_INCLUDED #define SOUND_FIREWIRE_AMDTP_H_INCLUDED +#include <linux/interrupt.h> #include <linux/mutex.h> #include <linux/spinlock.h> #include "packets-buffer.h" @@ -55,6 +56,7 @@ struct amdtp_out_stream { struct iso_packets_buffer buffer; struct snd_pcm_substream *pcm; + struct tasklet_struct period_tasklet; int packet_index; unsigned int data_block_counter; @@ -66,6 +68,7 @@ struct amdtp_out_stream { unsigned int pcm_buffer_pointer; unsigned int pcm_period_pointer; + bool pointer_flush; }; int amdtp_out_stream_init(struct amdtp_out_stream *s, struct fw_unit *unit, @@ -81,6 +84,8 @@ void amdtp_out_stream_stop(struct amdtp_out_stream *s); void amdtp_out_stream_set_pcm_format(struct amdtp_out_stream *s, snd_pcm_format_t format); +void amdtp_out_stream_pcm_prepare(struct amdtp_out_stream *s); +unsigned long amdtp_out_stream_pcm_pointer(struct amdtp_out_stream *s); void amdtp_out_stream_pcm_abort(struct amdtp_out_stream *s); /** @@ -123,18 +128,6 @@ static inline bool amdtp_out_streaming_error(struct amdtp_out_stream *s) } /** - * amdtp_out_stream_pcm_prepare - prepare PCM device for running - * @s: the AMDTP output stream - * - * This function should be called from the PCM device's .prepare callback. - */ -static inline void amdtp_out_stream_pcm_prepare(struct amdtp_out_stream *s) -{ - s->pcm_buffer_pointer = 0; - s->pcm_period_pointer = 0; -} - -/** * amdtp_out_stream_pcm_trigger - start/stop playback from a PCM device * @s: the AMDTP output stream * @pcm: the PCM device to be started, or %NULL to stop the current device @@ -149,18 +142,6 @@ static inline void amdtp_out_stream_pcm_trigger(struct amdtp_out_stream *s, ACCESS_ONCE(s->pcm) = pcm; } -/** - * amdtp_out_stream_pcm_pointer - get the PCM buffer position - * @s: the AMDTP output stream that transports the PCM data - * - * Returns the current buffer position, in frames. - */ -static inline unsigned long -amdtp_out_stream_pcm_pointer(struct amdtp_out_stream *s) -{ - return ACCESS_ONCE(s->pcm_buffer_pointer); -} - static inline bool cip_sfc_is_base_44100(enum cip_sfc sfc) { return sfc & 1; diff --git a/sound/pci/Kconfig b/sound/pci/Kconfig index 5ca0939e422..ff3af6e77d6 100644 --- a/sound/pci/Kconfig +++ b/sound/pci/Kconfig @@ -228,7 +228,7 @@ config SND_OXYGEN Say Y here to include support for sound cards based on the C-Media CMI8788 (Oxygen HD Audio) chip: * Asound A-8788 - * Asus Xonar DG + * Asus Xonar DG/DGX * AuzenTech X-Meridian * AuzenTech X-Meridian 2G * Bgears b-Enspirer diff --git a/sound/pci/ad1889.c b/sound/pci/ad1889.c index 9d91d61902b..e672ff4df2d 100644 --- a/sound/pci/ad1889.c +++ b/sound/pci/ad1889.c @@ -1062,17 +1062,4 @@ static struct pci_driver ad1889_pci_driver = { .remove = __devexit_p(snd_ad1889_remove), }; -static int __init -alsa_ad1889_init(void) -{ - return pci_register_driver(&ad1889_pci_driver); -} - -static void __exit -alsa_ad1889_fini(void) -{ - pci_unregister_driver(&ad1889_pci_driver); -} - -module_init(alsa_ad1889_init); -module_exit(alsa_ad1889_fini); +module_pci_driver(ad1889_pci_driver); diff --git a/sound/pci/ali5451/ali5451.c b/sound/pci/ali5451/ali5451.c index bdd6164e9c7..9dfc27bf6cc 100644 --- a/sound/pci/ali5451/ali5451.c +++ b/sound/pci/ali5451/ali5451.c @@ -2294,7 +2294,7 @@ static void __devexit snd_ali_remove(struct pci_dev *pci) pci_set_drvdata(pci, NULL); } -static struct pci_driver driver = { +static struct pci_driver ali5451_driver = { .name = KBUILD_MODNAME, .id_table = snd_ali_ids, .probe = snd_ali_probe, @@ -2305,15 +2305,4 @@ static struct pci_driver driver = { #endif }; -static int __init alsa_card_ali_init(void) -{ - return pci_register_driver(&driver); -} - -static void __exit alsa_card_ali_exit(void) -{ - pci_unregister_driver(&driver); -} - -module_init(alsa_card_ali_init) -module_exit(alsa_card_ali_exit) +module_pci_driver(ali5451_driver); diff --git a/sound/pci/als300.c b/sound/pci/als300.c index 8196e229b2d..59d65388faf 100644 --- a/sound/pci/als300.c +++ b/sound/pci/als300.c @@ -852,7 +852,7 @@ static int __devinit snd_als300_probe(struct pci_dev *pci, return 0; } -static struct pci_driver driver = { +static struct pci_driver als300_driver = { .name = KBUILD_MODNAME, .id_table = snd_als300_ids, .probe = snd_als300_probe, @@ -863,15 +863,4 @@ static struct pci_driver driver = { #endif }; -static int __init alsa_card_als300_init(void) -{ - return pci_register_driver(&driver); -} - -static void __exit alsa_card_als300_exit(void) -{ - pci_unregister_driver(&driver); -} - -module_init(alsa_card_als300_init) -module_exit(alsa_card_als300_exit) +module_pci_driver(als300_driver); diff --git a/sound/pci/als4000.c b/sound/pci/als4000.c index 3269b8011ea..7d7f2598c74 100644 --- a/sound/pci/als4000.c +++ b/sound/pci/als4000.c @@ -1036,7 +1036,7 @@ static int snd_als4000_resume(struct pci_dev *pci) #endif /* CONFIG_PM */ -static struct pci_driver driver = { +static struct pci_driver als4000_driver = { .name = KBUILD_MODNAME, .id_table = snd_als4000_ids, .probe = snd_card_als4000_probe, @@ -1047,15 +1047,4 @@ static struct pci_driver driver = { #endif }; -static int __init alsa_card_als4000_init(void) -{ - return pci_register_driver(&driver); -} - -static void __exit alsa_card_als4000_exit(void) -{ - pci_unregister_driver(&driver); -} - -module_init(alsa_card_als4000_init) -module_exit(alsa_card_als4000_exit) +module_pci_driver(als4000_driver); diff --git a/sound/pci/atiixp.c b/sound/pci/atiixp.c index 590682f115e..156a94f8a12 100644 --- a/sound/pci/atiixp.c +++ b/sound/pci/atiixp.c @@ -1700,7 +1700,7 @@ static void __devexit snd_atiixp_remove(struct pci_dev *pci) pci_set_drvdata(pci, NULL); } -static struct pci_driver driver = { +static struct pci_driver atiixp_driver = { .name = KBUILD_MODNAME, .id_table = snd_atiixp_ids, .probe = snd_atiixp_probe, @@ -1711,16 +1711,4 @@ static struct pci_driver driver = { #endif }; - -static int __init alsa_card_atiixp_init(void) -{ - return pci_register_driver(&driver); -} - -static void __exit alsa_card_atiixp_exit(void) -{ - pci_unregister_driver(&driver); -} - -module_init(alsa_card_atiixp_init) -module_exit(alsa_card_atiixp_exit) +module_pci_driver(atiixp_driver); diff --git a/sound/pci/atiixp_modem.c b/sound/pci/atiixp_modem.c index 524d35f3123..30a4fd96ce7 100644 --- a/sound/pci/atiixp_modem.c +++ b/sound/pci/atiixp_modem.c @@ -1331,7 +1331,7 @@ static void __devexit snd_atiixp_remove(struct pci_dev *pci) pci_set_drvdata(pci, NULL); } -static struct pci_driver driver = { +static struct pci_driver atiixp_modem_driver = { .name = KBUILD_MODNAME, .id_table = snd_atiixp_ids, .probe = snd_atiixp_probe, @@ -1342,16 +1342,4 @@ static struct pci_driver driver = { #endif }; - -static int __init alsa_card_atiixp_init(void) -{ - return pci_register_driver(&driver); -} - -static void __exit alsa_card_atiixp_exit(void) -{ - pci_unregister_driver(&driver); -} - -module_init(alsa_card_atiixp_init) -module_exit(alsa_card_atiixp_exit) +module_pci_driver(atiixp_modem_driver); diff --git a/sound/pci/au88x0/au88x0.c b/sound/pci/au88x0/au88x0.c index f13ad536b2d..ffc376f9f4e 100644 --- a/sound/pci/au88x0/au88x0.c +++ b/sound/pci/au88x0/au88x0.c @@ -375,24 +375,11 @@ static void __devexit snd_vortex_remove(struct pci_dev *pci) } // pci_driver definition -static struct pci_driver driver = { +static struct pci_driver vortex_driver = { .name = KBUILD_MODNAME, .id_table = snd_vortex_ids, .probe = snd_vortex_probe, .remove = __devexit_p(snd_vortex_remove), }; -// initialization of the module -static int __init alsa_card_vortex_init(void) -{ - return pci_register_driver(&driver); -} - -// clean up the module -static void __exit alsa_card_vortex_exit(void) -{ - pci_unregister_driver(&driver); -} - -module_init(alsa_card_vortex_init) -module_exit(alsa_card_vortex_exit) +module_pci_driver(vortex_driver); diff --git a/sound/pci/aw2/aw2-alsa.c b/sound/pci/aw2/aw2-alsa.c index 1c523193146..0f804741825 100644 --- a/sound/pci/aw2/aw2-alsa.c +++ b/sound/pci/aw2/aw2-alsa.c @@ -112,8 +112,6 @@ struct aw2 { /********************************* * FUNCTION DECLARATIONS ********************************/ -static int __init alsa_card_aw2_init(void); -static void __exit alsa_card_aw2_exit(void); static int snd_aw2_dev_free(struct snd_device *device); static int __devinit snd_aw2_create(struct snd_card *card, struct pci_dev *pci, struct aw2 **rchip); @@ -171,13 +169,15 @@ static DEFINE_PCI_DEVICE_TABLE(snd_aw2_ids) = { MODULE_DEVICE_TABLE(pci, snd_aw2_ids); /* pci_driver definition */ -static struct pci_driver driver = { +static struct pci_driver aw2_driver = { .name = KBUILD_MODNAME, .id_table = snd_aw2_ids, .probe = snd_aw2_probe, .remove = __devexit_p(snd_aw2_remove), }; +module_pci_driver(aw2_driver); + /* operators for playback PCM alsa interface */ static struct snd_pcm_ops snd_aw2_playback_ops = { .open = snd_aw2_pcm_playback_open, @@ -217,23 +217,6 @@ static struct snd_kcontrol_new aw2_control __devinitdata = { * FUNCTION IMPLEMENTATIONS ********************************/ -/* initialization of the module */ -static int __init alsa_card_aw2_init(void) -{ - snd_printdd(KERN_DEBUG "aw2: Load aw2 module\n"); - return pci_register_driver(&driver); -} - -/* clean up the module */ -static void __exit alsa_card_aw2_exit(void) -{ - snd_printdd(KERN_DEBUG "aw2: Unload aw2 module\n"); - pci_unregister_driver(&driver); -} - -module_init(alsa_card_aw2_init); -module_exit(alsa_card_aw2_exit); - /* component-destructor */ static int snd_aw2_dev_free(struct snd_device *device) { diff --git a/sound/pci/azt3328.c b/sound/pci/azt3328.c index 496f14c1a73..f0b4d7493af 100644 --- a/sound/pci/azt3328.c +++ b/sound/pci/azt3328.c @@ -2862,7 +2862,7 @@ snd_azf3328_resume(struct pci_dev *pci) #endif /* CONFIG_PM */ -static struct pci_driver driver = { +static struct pci_driver azf3328_driver = { .name = KBUILD_MODNAME, .id_table = snd_azf3328_ids, .probe = snd_azf3328_probe, @@ -2873,23 +2873,4 @@ static struct pci_driver driver = { #endif }; -static int __init -alsa_card_azf3328_init(void) -{ - int err; - snd_azf3328_dbgcallenter(); - err = pci_register_driver(&driver); - snd_azf3328_dbgcallleave(); - return err; -} - -static void __exit -alsa_card_azf3328_exit(void) -{ - snd_azf3328_dbgcallenter(); - pci_unregister_driver(&driver); - snd_azf3328_dbgcallleave(); -} - -module_init(alsa_card_azf3328_init) -module_exit(alsa_card_azf3328_exit) +module_pci_driver(azf3328_driver); diff --git a/sound/pci/bt87x.c b/sound/pci/bt87x.c index 62d6163fc9d..b6a95eeca09 100644 --- a/sound/pci/bt87x.c +++ b/sound/pci/bt87x.c @@ -836,8 +836,6 @@ static struct { {0x7063, 0x2000}, /* pcHDTV HD-2000 TV */ }; -static struct pci_driver driver; - /* return the id of the card, or a negative value if it's blacklisted */ static int __devinit snd_bt87x_detect_card(struct pci_dev *pci) { @@ -964,24 +962,11 @@ static DEFINE_PCI_DEVICE_TABLE(snd_bt87x_default_ids) = { { } }; -static struct pci_driver driver = { +static struct pci_driver bt87x_driver = { .name = KBUILD_MODNAME, .id_table = snd_bt87x_ids, .probe = snd_bt87x_probe, .remove = __devexit_p(snd_bt87x_remove), }; -static int __init alsa_card_bt87x_init(void) -{ - if (load_all) - driver.id_table = snd_bt87x_default_ids; - return pci_register_driver(&driver); -} - -static void __exit alsa_card_bt87x_exit(void) -{ - pci_unregister_driver(&driver); -} - -module_init(alsa_card_bt87x_init) -module_exit(alsa_card_bt87x_exit) +module_pci_driver(bt87x_driver); diff --git a/sound/pci/ca0106/ca0106_main.c b/sound/pci/ca0106/ca0106_main.c index 08d6ebfe5a6..e76d68a7081 100644 --- a/sound/pci/ca0106/ca0106_main.c +++ b/sound/pci/ca0106/ca0106_main.c @@ -1932,7 +1932,7 @@ static DEFINE_PCI_DEVICE_TABLE(snd_ca0106_ids) = { MODULE_DEVICE_TABLE(pci, snd_ca0106_ids); // pci_driver definition -static struct pci_driver driver = { +static struct pci_driver ca0106_driver = { .name = KBUILD_MODNAME, .id_table = snd_ca0106_ids, .probe = snd_ca0106_probe, @@ -1943,17 +1943,4 @@ static struct pci_driver driver = { #endif }; -// initialization of the module -static int __init alsa_card_ca0106_init(void) -{ - return pci_register_driver(&driver); -} - -// clean up the module -static void __exit alsa_card_ca0106_exit(void) -{ - pci_unregister_driver(&driver); -} - -module_init(alsa_card_ca0106_init) -module_exit(alsa_card_ca0106_exit) +module_pci_driver(ca0106_driver); diff --git a/sound/pci/cmipci.c b/sound/pci/cmipci.c index 19b06269adc..3815bd4c677 100644 --- a/sound/pci/cmipci.c +++ b/sound/pci/cmipci.c @@ -3398,7 +3398,7 @@ static int snd_cmipci_resume(struct pci_dev *pci) } #endif /* CONFIG_PM */ -static struct pci_driver driver = { +static struct pci_driver cmipci_driver = { .name = KBUILD_MODNAME, .id_table = snd_cmipci_ids, .probe = snd_cmipci_probe, @@ -3409,15 +3409,4 @@ static struct pci_driver driver = { #endif }; -static int __init alsa_card_cmipci_init(void) -{ - return pci_register_driver(&driver); -} - -static void __exit alsa_card_cmipci_exit(void) -{ - pci_unregister_driver(&driver); -} - -module_init(alsa_card_cmipci_init) -module_exit(alsa_card_cmipci_exit) +module_pci_driver(cmipci_driver); diff --git a/sound/pci/cs4281.c b/sound/pci/cs4281.c index a9f368f60df..33506ee569b 100644 --- a/sound/pci/cs4281.c +++ b/sound/pci/cs4281.c @@ -2084,7 +2084,7 @@ static int cs4281_resume(struct pci_dev *pci) } #endif /* CONFIG_PM */ -static struct pci_driver driver = { +static struct pci_driver cs4281_driver = { .name = KBUILD_MODNAME, .id_table = snd_cs4281_ids, .probe = snd_cs4281_probe, @@ -2095,15 +2095,4 @@ static struct pci_driver driver = { #endif }; -static int __init alsa_card_cs4281_init(void) -{ - return pci_register_driver(&driver); -} - -static void __exit alsa_card_cs4281_exit(void) -{ - pci_unregister_driver(&driver); -} - -module_init(alsa_card_cs4281_init) -module_exit(alsa_card_cs4281_exit) +module_pci_driver(cs4281_driver); diff --git a/sound/pci/cs46xx/cs46xx.c b/sound/pci/cs46xx/cs46xx.c index 819d79d0586..6cc7404e0e8 100644 --- a/sound/pci/cs46xx/cs46xx.c +++ b/sound/pci/cs46xx/cs46xx.c @@ -161,7 +161,7 @@ static void __devexit snd_card_cs46xx_remove(struct pci_dev *pci) pci_set_drvdata(pci, NULL); } -static struct pci_driver driver = { +static struct pci_driver cs46xx_driver = { .name = KBUILD_MODNAME, .id_table = snd_cs46xx_ids, .probe = snd_card_cs46xx_probe, @@ -172,15 +172,4 @@ static struct pci_driver driver = { #endif }; -static int __init alsa_card_cs46xx_init(void) -{ - return pci_register_driver(&driver); -} - -static void __exit alsa_card_cs46xx_exit(void) -{ - pci_unregister_driver(&driver); -} - -module_init(alsa_card_cs46xx_init) -module_exit(alsa_card_cs46xx_exit) +module_pci_driver(cs46xx_driver); diff --git a/sound/pci/cs5530.c b/sound/pci/cs5530.c index c47cabff2bf..f1e4229993a 100644 --- a/sound/pci/cs5530.c +++ b/sound/pci/cs5530.c @@ -291,23 +291,11 @@ static int __devinit snd_cs5530_probe(struct pci_dev *pci, return 0; } -static struct pci_driver driver = { +static struct pci_driver cs5530_driver = { .name = KBUILD_MODNAME, .id_table = snd_cs5530_ids, .probe = snd_cs5530_probe, .remove = __devexit_p(snd_cs5530_remove), }; -static int __init alsa_card_cs5530_init(void) -{ - return pci_register_driver(&driver); -} - -static void __exit alsa_card_cs5530_exit(void) -{ - pci_unregister_driver(&driver); -} - -module_init(alsa_card_cs5530_init) -module_exit(alsa_card_cs5530_exit) - +module_pci_driver(cs5530_driver); diff --git a/sound/pci/cs5535audio/cs5535audio.c b/sound/pci/cs5535audio/cs5535audio.c index a2fb2173e98..2c9697cf0a1 100644 --- a/sound/pci/cs5535audio/cs5535audio.c +++ b/sound/pci/cs5535audio/cs5535audio.c @@ -394,7 +394,7 @@ static void __devexit snd_cs5535audio_remove(struct pci_dev *pci) pci_set_drvdata(pci, NULL); } -static struct pci_driver driver = { +static struct pci_driver cs5535audio_driver = { .name = KBUILD_MODNAME, .id_table = snd_cs5535audio_ids, .probe = snd_cs5535audio_probe, @@ -405,18 +405,7 @@ static struct pci_driver driver = { #endif }; -static int __init alsa_card_cs5535audio_init(void) -{ - return pci_register_driver(&driver); -} - -static void __exit alsa_card_cs5535audio_exit(void) -{ - pci_unregister_driver(&driver); -} - -module_init(alsa_card_cs5535audio_init) -module_exit(alsa_card_cs5535audio_exit) +module_pci_driver(cs5535audio_driver); MODULE_AUTHOR("Jaya Kumar"); MODULE_LICENSE("GPL"); diff --git a/sound/pci/ctxfi/xfi.c b/sound/pci/ctxfi/xfi.c index 15d95d2bace..75aa2c33841 100644 --- a/sound/pci/ctxfi/xfi.c +++ b/sound/pci/ctxfi/xfi.c @@ -154,15 +154,4 @@ static struct pci_driver ct_driver = { #endif }; -static int __init ct_card_init(void) -{ - return pci_register_driver(&ct_driver); -} - -static void __exit ct_card_exit(void) -{ - pci_unregister_driver(&ct_driver); -} - -module_init(ct_card_init) -module_exit(ct_card_exit) +module_pci_driver(ct_driver); diff --git a/sound/pci/echoaudio/echoaudio.c b/sound/pci/echoaudio/echoaudio.c index 595c11f904b..0f8eda1dafd 100644 --- a/sound/pci/echoaudio/echoaudio.c +++ b/sound/pci/echoaudio/echoaudio.c @@ -2328,7 +2328,7 @@ static void __devexit snd_echo_remove(struct pci_dev *pci) ******************************************************************************/ /* pci_driver definition */ -static struct pci_driver driver = { +static struct pci_driver echo_driver = { .name = KBUILD_MODNAME, .id_table = snd_echo_ids, .probe = snd_echo_probe, @@ -2339,22 +2339,4 @@ static struct pci_driver driver = { #endif /* CONFIG_PM */ }; - - -/* initialization of the module */ -static int __init alsa_card_echo_init(void) -{ - return pci_register_driver(&driver); -} - - - -/* clean up the module */ -static void __exit alsa_card_echo_exit(void) -{ - pci_unregister_driver(&driver); -} - - -module_init(alsa_card_echo_init) -module_exit(alsa_card_echo_exit) +module_pci_driver(echo_driver); diff --git a/sound/pci/emu10k1/emu10k1.c b/sound/pci/emu10k1/emu10k1.c index 790c65d980c..7fdbbe4d996 100644 --- a/sound/pci/emu10k1/emu10k1.c +++ b/sound/pci/emu10k1/emu10k1.c @@ -263,7 +263,7 @@ static int snd_emu10k1_resume(struct pci_dev *pci) } #endif -static struct pci_driver driver = { +static struct pci_driver emu10k1_driver = { .name = KBUILD_MODNAME, .id_table = snd_emu10k1_ids, .probe = snd_card_emu10k1_probe, @@ -274,15 +274,4 @@ static struct pci_driver driver = { #endif }; -static int __init alsa_card_emu10k1_init(void) -{ - return pci_register_driver(&driver); -} - -static void __exit alsa_card_emu10k1_exit(void) -{ - pci_unregister_driver(&driver); -} - -module_init(alsa_card_emu10k1_init) -module_exit(alsa_card_emu10k1_exit) +module_pci_driver(emu10k1_driver); diff --git a/sound/pci/emu10k1/emu10k1x.c b/sound/pci/emu10k1/emu10k1x.c index 47a651cb6e8..5c8978b2c4d 100644 --- a/sound/pci/emu10k1/emu10k1x.c +++ b/sound/pci/emu10k1/emu10k1x.c @@ -1612,24 +1612,11 @@ static DEFINE_PCI_DEVICE_TABLE(snd_emu10k1x_ids) = { MODULE_DEVICE_TABLE(pci, snd_emu10k1x_ids); // pci_driver definition -static struct pci_driver driver = { +static struct pci_driver emu10k1x_driver = { .name = KBUILD_MODNAME, .id_table = snd_emu10k1x_ids, .probe = snd_emu10k1x_probe, .remove = __devexit_p(snd_emu10k1x_remove), }; -// initialization of the module -static int __init alsa_card_emu10k1x_init(void) -{ - return pci_register_driver(&driver); -} - -// clean up the module -static void __exit alsa_card_emu10k1x_exit(void) -{ - pci_unregister_driver(&driver); -} - -module_init(alsa_card_emu10k1x_init) -module_exit(alsa_card_emu10k1x_exit) +module_pci_driver(emu10k1x_driver); diff --git a/sound/pci/ens1370.c b/sound/pci/ens1370.c index 47a245e8419..3821c81d1c9 100644 --- a/sound/pci/ens1370.c +++ b/sound/pci/ens1370.c @@ -2488,7 +2488,7 @@ static void __devexit snd_audiopci_remove(struct pci_dev *pci) pci_set_drvdata(pci, NULL); } -static struct pci_driver driver = { +static struct pci_driver ens137x_driver = { .name = KBUILD_MODNAME, .id_table = snd_audiopci_ids, .probe = snd_audiopci_probe, @@ -2499,15 +2499,4 @@ static struct pci_driver driver = { #endif }; -static int __init alsa_card_ens137x_init(void) -{ - return pci_register_driver(&driver); -} - -static void __exit alsa_card_ens137x_exit(void) -{ - pci_unregister_driver(&driver); -} - -module_init(alsa_card_ens137x_init) -module_exit(alsa_card_ens137x_exit) +module_pci_driver(ens137x_driver); diff --git a/sound/pci/es1938.c b/sound/pci/es1938.c index 53eb76b4110..82c8d8c5c52 100644 --- a/sound/pci/es1938.c +++ b/sound/pci/es1938.c @@ -1882,7 +1882,7 @@ static void __devexit snd_es1938_remove(struct pci_dev *pci) pci_set_drvdata(pci, NULL); } -static struct pci_driver driver = { +static struct pci_driver es1938_driver = { .name = KBUILD_MODNAME, .id_table = snd_es1938_ids, .probe = snd_es1938_probe, @@ -1893,15 +1893,4 @@ static struct pci_driver driver = { #endif }; -static int __init alsa_card_es1938_init(void) -{ - return pci_register_driver(&driver); -} - -static void __exit alsa_card_es1938_exit(void) -{ - pci_unregister_driver(&driver); -} - -module_init(alsa_card_es1938_init) -module_exit(alsa_card_es1938_exit) +module_pci_driver(es1938_driver); diff --git a/sound/pci/es1968.c b/sound/pci/es1968.c index a8faae1c85e..67f47d89195 100644 --- a/sound/pci/es1968.c +++ b/sound/pci/es1968.c @@ -2898,7 +2898,7 @@ static void __devexit snd_es1968_remove(struct pci_dev *pci) pci_set_drvdata(pci, NULL); } -static struct pci_driver driver = { +static struct pci_driver es1968_driver = { .name = KBUILD_MODNAME, .id_table = snd_es1968_ids, .probe = snd_es1968_probe, @@ -2909,15 +2909,4 @@ static struct pci_driver driver = { #endif }; -static int __init alsa_card_es1968_init(void) -{ - return pci_register_driver(&driver); -} - -static void __exit alsa_card_es1968_exit(void) -{ - pci_unregister_driver(&driver); -} - -module_init(alsa_card_es1968_init) -module_exit(alsa_card_es1968_exit) +module_pci_driver(es1968_driver); diff --git a/sound/pci/fm801.c b/sound/pci/fm801.c index a416ea8af3e..f6966232275 100644 --- a/sound/pci/fm801.c +++ b/sound/pci/fm801.c @@ -1416,7 +1416,7 @@ static int snd_fm801_resume(struct pci_dev *pci) } #endif -static struct pci_driver driver = { +static struct pci_driver fm801_driver = { .name = KBUILD_MODNAME, .id_table = snd_fm801_ids, .probe = snd_card_fm801_probe, @@ -1427,15 +1427,4 @@ static struct pci_driver driver = { #endif }; -static int __init alsa_card_fm801_init(void) -{ - return pci_register_driver(&driver); -} - -static void __exit alsa_card_fm801_exit(void) -{ - pci_unregister_driver(&driver); -} - -module_init(alsa_card_fm801_init) -module_exit(alsa_card_fm801_exit) +module_pci_driver(fm801_driver); diff --git a/sound/pci/hda/Makefile b/sound/pci/hda/Makefile index ace157cc3d1..bd4149f1aaf 100644 --- a/sound/pci/hda/Makefile +++ b/sound/pci/hda/Makefile @@ -1,6 +1,6 @@ snd-hda-intel-objs := hda_intel.o -snd-hda-codec-y := hda_codec.o hda_jack.o +snd-hda-codec-y := hda_codec.o hda_jack.o hda_auto_parser.o snd-hda-codec-$(CONFIG_SND_HDA_GENERIC) += hda_generic.o snd-hda-codec-$(CONFIG_PROC_FS) += hda_proc.o snd-hda-codec-$(CONFIG_SND_HDA_HWDEP) += hda_hwdep.o diff --git a/sound/pci/hda/hda_auto_parser.c b/sound/pci/hda/hda_auto_parser.c new file mode 100644 index 00000000000..6e9ef3e2509 --- /dev/null +++ b/sound/pci/hda/hda_auto_parser.c @@ -0,0 +1,760 @@ +/* + * BIOS auto-parser helper functions for HD-audio + * + * Copyright (c) 2012 Takashi Iwai <tiwai@suse.de> + * + * This driver is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License as published by + * the Free Software Foundation; either version 2 of the License, or + * (at your option) any later version. + */ + +#include <linux/slab.h> +#include <linux/export.h> +#include <sound/core.h> +#include "hda_codec.h" +#include "hda_local.h" +#include "hda_auto_parser.h" + +#define SFX "hda_codec: " + +/* + * Helper for automatic pin configuration + */ + +static int is_in_nid_list(hda_nid_t nid, const hda_nid_t *list) +{ + for (; *list; list++) + if (*list == nid) + return 1; + return 0; +} + + +/* + * Sort an associated group of pins according to their sequence numbers. + */ +static void sort_pins_by_sequence(hda_nid_t *pins, short *sequences, + int num_pins) +{ + int i, j; + short seq; + hda_nid_t nid; + + for (i = 0; i < num_pins; i++) { + for (j = i + 1; j < num_pins; j++) { + if (sequences[i] > sequences[j]) { + seq = sequences[i]; + sequences[i] = sequences[j]; + sequences[j] = seq; + nid = pins[i]; + pins[i] = pins[j]; + pins[j] = nid; + } + } + } +} + + +/* add the found input-pin to the cfg->inputs[] table */ +static void add_auto_cfg_input_pin(struct auto_pin_cfg *cfg, hda_nid_t nid, + int type) +{ + if (cfg->num_inputs < AUTO_CFG_MAX_INS) { + cfg->inputs[cfg->num_inputs].pin = nid; + cfg->inputs[cfg->num_inputs].type = type; + cfg->num_inputs++; + } +} + +/* sort inputs in the order of AUTO_PIN_* type */ +static void sort_autocfg_input_pins(struct auto_pin_cfg *cfg) +{ + int i, j; + + for (i = 0; i < cfg->num_inputs; i++) { + for (j = i + 1; j < cfg->num_inputs; j++) { + if (cfg->inputs[i].type > cfg->inputs[j].type) { + struct auto_pin_cfg_item tmp; + tmp = cfg->inputs[i]; + cfg->inputs[i] = cfg->inputs[j]; + cfg->inputs[j] = tmp; + } + } + } +} + +/* Reorder the surround channels + * ALSA sequence is front/surr/clfe/side + * HDA sequence is: + * 4-ch: front/surr => OK as it is + * 6-ch: front/clfe/surr + * 8-ch: front/clfe/rear/side|fc + */ +static void reorder_outputs(unsigned int nums, hda_nid_t *pins) +{ + hda_nid_t nid; + + switch (nums) { + case 3: + case 4: + nid = pins[1]; + pins[1] = pins[2]; + pins[2] = nid; + break; + } +} + +/* + * Parse all pin widgets and store the useful pin nids to cfg + * + * The number of line-outs or any primary output is stored in line_outs, + * and the corresponding output pins are assigned to line_out_pins[], + * in the order of front, rear, CLFE, side, ... + * + * If more extra outputs (speaker and headphone) are found, the pins are + * assisnged to hp_pins[] and speaker_pins[], respectively. If no line-out jack + * is detected, one of speaker of HP pins is assigned as the primary + * output, i.e. to line_out_pins[0]. So, line_outs is always positive + * if any analog output exists. + * + * The analog input pins are assigned to inputs array. + * The digital input/output pins are assigned to dig_in_pin and dig_out_pin, + * respectively. + */ +int snd_hda_parse_pin_defcfg(struct hda_codec *codec, + struct auto_pin_cfg *cfg, + const hda_nid_t *ignore_nids, + unsigned int cond_flags) +{ + hda_nid_t nid, end_nid; + short seq, assoc_line_out; + short sequences_line_out[ARRAY_SIZE(cfg->line_out_pins)]; + short sequences_speaker[ARRAY_SIZE(cfg->speaker_pins)]; + short sequences_hp[ARRAY_SIZE(cfg->hp_pins)]; + int i; + + memset(cfg, 0, sizeof(*cfg)); + + memset(sequences_line_out, 0, sizeof(sequences_line_out)); + memset(sequences_speaker, 0, sizeof(sequences_speaker)); + memset(sequences_hp, 0, sizeof(sequences_hp)); + assoc_line_out = 0; + + codec->ignore_misc_bit = true; + end_nid = codec->start_nid + codec->num_nodes; + for (nid = codec->start_nid; nid < end_nid; nid++) { + unsigned int wid_caps = get_wcaps(codec, nid); + unsigned int wid_type = get_wcaps_type(wid_caps); + unsigned int def_conf; + short assoc, loc, conn, dev; + + /* read all default configuration for pin complex */ + if (wid_type != AC_WID_PIN) + continue; + /* ignore the given nids (e.g. pc-beep returns error) */ + if (ignore_nids && is_in_nid_list(nid, ignore_nids)) + continue; + + def_conf = snd_hda_codec_get_pincfg(codec, nid); + if (!(get_defcfg_misc(snd_hda_codec_get_pincfg(codec, nid)) & + AC_DEFCFG_MISC_NO_PRESENCE)) + codec->ignore_misc_bit = false; + conn = get_defcfg_connect(def_conf); + if (conn == AC_JACK_PORT_NONE) + continue; + loc = get_defcfg_location(def_conf); + dev = get_defcfg_device(def_conf); + + /* workaround for buggy BIOS setups */ + if (dev == AC_JACK_LINE_OUT) { + if (conn == AC_JACK_PORT_FIXED) + dev = AC_JACK_SPEAKER; + } + + switch (dev) { + case AC_JACK_LINE_OUT: + seq = get_defcfg_sequence(def_conf); + assoc = get_defcfg_association(def_conf); + + if (!(wid_caps & AC_WCAP_STEREO)) + if (!cfg->mono_out_pin) + cfg->mono_out_pin = nid; + if (!assoc) + continue; + if (!assoc_line_out) + assoc_line_out = assoc; + else if (assoc_line_out != assoc) + continue; + if (cfg->line_outs >= ARRAY_SIZE(cfg->line_out_pins)) + continue; + cfg->line_out_pins[cfg->line_outs] = nid; + sequences_line_out[cfg->line_outs] = seq; + cfg->line_outs++; + break; + case AC_JACK_SPEAKER: + seq = get_defcfg_sequence(def_conf); + assoc = get_defcfg_association(def_conf); + if (cfg->speaker_outs >= ARRAY_SIZE(cfg->speaker_pins)) + continue; + cfg->speaker_pins[cfg->speaker_outs] = nid; + sequences_speaker[cfg->speaker_outs] = (assoc << 4) | seq; + cfg->speaker_outs++; + break; + case AC_JACK_HP_OUT: + seq = get_defcfg_sequence(def_conf); + assoc = get_defcfg_association(def_conf); + if (cfg->hp_outs >= ARRAY_SIZE(cfg->hp_pins)) + continue; + cfg->hp_pins[cfg->hp_outs] = nid; + sequences_hp[cfg->hp_outs] = (assoc << 4) | seq; + cfg->hp_outs++; + break; + case AC_JACK_MIC_IN: + add_auto_cfg_input_pin(cfg, nid, AUTO_PIN_MIC); + break; + case AC_JACK_LINE_IN: + add_auto_cfg_input_pin(cfg, nid, AUTO_PIN_LINE_IN); + break; + case AC_JACK_CD: + add_auto_cfg_input_pin(cfg, nid, AUTO_PIN_CD); + break; + case AC_JACK_AUX: + add_auto_cfg_input_pin(cfg, nid, AUTO_PIN_AUX); + break; + case AC_JACK_SPDIF_OUT: + case AC_JACK_DIG_OTHER_OUT: + if (cfg->dig_outs >= ARRAY_SIZE(cfg->dig_out_pins)) + continue; + cfg->dig_out_pins[cfg->dig_outs] = nid; + cfg->dig_out_type[cfg->dig_outs] = + (loc == AC_JACK_LOC_HDMI) ? + HDA_PCM_TYPE_HDMI : HDA_PCM_TYPE_SPDIF; + cfg->dig_outs++; + break; + case AC_JACK_SPDIF_IN: + case AC_JACK_DIG_OTHER_IN: + cfg->dig_in_pin = nid; + if (loc == AC_JACK_LOC_HDMI) + cfg->dig_in_type = HDA_PCM_TYPE_HDMI; + else + cfg->dig_in_type = HDA_PCM_TYPE_SPDIF; + break; + } + } + + /* FIX-UP: + * If no line-out is defined but multiple HPs are found, + * some of them might be the real line-outs. + */ + if (!cfg->line_outs && cfg->hp_outs > 1 && + !(cond_flags & HDA_PINCFG_NO_HP_FIXUP)) { + int i = 0; + while (i < cfg->hp_outs) { + /* The real HPs should have the sequence 0x0f */ + if ((sequences_hp[i] & 0x0f) == 0x0f) { + i++; + continue; + } + /* Move it to the line-out table */ + cfg->line_out_pins[cfg->line_outs] = cfg->hp_pins[i]; + sequences_line_out[cfg->line_outs] = sequences_hp[i]; + cfg->line_outs++; + cfg->hp_outs--; + memmove(cfg->hp_pins + i, cfg->hp_pins + i + 1, + sizeof(cfg->hp_pins[0]) * (cfg->hp_outs - i)); + memmove(sequences_hp + i, sequences_hp + i + 1, + sizeof(sequences_hp[0]) * (cfg->hp_outs - i)); + } + memset(cfg->hp_pins + cfg->hp_outs, 0, + sizeof(hda_nid_t) * (AUTO_CFG_MAX_OUTS - cfg->hp_outs)); + if (!cfg->hp_outs) + cfg->line_out_type = AUTO_PIN_HP_OUT; + + } + + /* sort by sequence */ + sort_pins_by_sequence(cfg->line_out_pins, sequences_line_out, + cfg->line_outs); + sort_pins_by_sequence(cfg->speaker_pins, sequences_speaker, + cfg->speaker_outs); + sort_pins_by_sequence(cfg->hp_pins, sequences_hp, + cfg->hp_outs); + + /* + * FIX-UP: if no line-outs are detected, try to use speaker or HP pin + * as a primary output + */ + if (!cfg->line_outs && + !(cond_flags & HDA_PINCFG_NO_LO_FIXUP)) { + if (cfg->speaker_outs) { + cfg->line_outs = cfg->speaker_outs; + memcpy(cfg->line_out_pins, cfg->speaker_pins, + sizeof(cfg->speaker_pins)); + cfg->speaker_outs = 0; + memset(cfg->speaker_pins, 0, sizeof(cfg->speaker_pins)); + cfg->line_out_type = AUTO_PIN_SPEAKER_OUT; + } else if (cfg->hp_outs) { + cfg->line_outs = cfg->hp_outs; + memcpy(cfg->line_out_pins, cfg->hp_pins, + sizeof(cfg->hp_pins)); + cfg->hp_outs = 0; + memset(cfg->hp_pins, 0, sizeof(cfg->hp_pins)); + cfg->line_out_type = AUTO_PIN_HP_OUT; + } + } + + reorder_outputs(cfg->line_outs, cfg->line_out_pins); + reorder_outputs(cfg->hp_outs, cfg->hp_pins); + reorder_outputs(cfg->speaker_outs, cfg->speaker_pins); + + sort_autocfg_input_pins(cfg); + + /* + * debug prints of the parsed results + */ + snd_printd("autoconfig: line_outs=%d (0x%x/0x%x/0x%x/0x%x/0x%x) type:%s\n", + cfg->line_outs, cfg->line_out_pins[0], cfg->line_out_pins[1], + cfg->line_out_pins[2], cfg->line_out_pins[3], + cfg->line_out_pins[4], + cfg->line_out_type == AUTO_PIN_HP_OUT ? "hp" : + (cfg->line_out_type == AUTO_PIN_SPEAKER_OUT ? + "speaker" : "line")); + snd_printd(" speaker_outs=%d (0x%x/0x%x/0x%x/0x%x/0x%x)\n", + cfg->speaker_outs, cfg->speaker_pins[0], + cfg->speaker_pins[1], cfg->speaker_pins[2], + cfg->speaker_pins[3], cfg->speaker_pins[4]); + snd_printd(" hp_outs=%d (0x%x/0x%x/0x%x/0x%x/0x%x)\n", + cfg->hp_outs, cfg->hp_pins[0], + cfg->hp_pins[1], cfg->hp_pins[2], + cfg->hp_pins[3], cfg->hp_pins[4]); + snd_printd(" mono: mono_out=0x%x\n", cfg->mono_out_pin); + if (cfg->dig_outs) + snd_printd(" dig-out=0x%x/0x%x\n", + cfg->dig_out_pins[0], cfg->dig_out_pins[1]); + snd_printd(" inputs:"); + for (i = 0; i < cfg->num_inputs; i++) { + snd_printd(" %s=0x%x", + hda_get_autocfg_input_label(codec, cfg, i), + cfg->inputs[i].pin); + } + snd_printd("\n"); + if (cfg->dig_in_pin) + snd_printd(" dig-in=0x%x\n", cfg->dig_in_pin); + + return 0; +} +EXPORT_SYMBOL_HDA(snd_hda_parse_pin_defcfg); + +int snd_hda_get_input_pin_attr(unsigned int def_conf) +{ + unsigned int loc = get_defcfg_location(def_conf); + unsigned int conn = get_defcfg_connect(def_conf); + if (conn == AC_JACK_PORT_NONE) + return INPUT_PIN_ATTR_UNUSED; + /* Windows may claim the internal mic to be BOTH, too */ + if (conn == AC_JACK_PORT_FIXED || conn == AC_JACK_PORT_BOTH) + return INPUT_PIN_ATTR_INT; + if ((loc & 0x30) == AC_JACK_LOC_INTERNAL) + return INPUT_PIN_ATTR_INT; + if ((loc & 0x30) == AC_JACK_LOC_SEPARATE) + return INPUT_PIN_ATTR_DOCK; + if (loc == AC_JACK_LOC_REAR) + return INPUT_PIN_ATTR_REAR; + if (loc == AC_JACK_LOC_FRONT) + return INPUT_PIN_ATTR_FRONT; + return INPUT_PIN_ATTR_NORMAL; +} +EXPORT_SYMBOL_HDA(snd_hda_get_input_pin_attr); + +/** + * hda_get_input_pin_label - Give a label for the given input pin + * + * When check_location is true, the function checks the pin location + * for mic and line-in pins, and set an appropriate prefix like "Front", + * "Rear", "Internal". + */ + +static const char *hda_get_input_pin_label(struct hda_codec *codec, + hda_nid_t pin, bool check_location) +{ + unsigned int def_conf; + static const char * const mic_names[] = { + "Internal Mic", "Dock Mic", "Mic", "Front Mic", "Rear Mic", + }; + int attr; + + def_conf = snd_hda_codec_get_pincfg(codec, pin); + + switch (get_defcfg_device(def_conf)) { + case AC_JACK_MIC_IN: + if (!check_location) + return "Mic"; + attr = snd_hda_get_input_pin_attr(def_conf); + if (!attr) + return "None"; + return mic_names[attr - 1]; + case AC_JACK_LINE_IN: + if (!check_location) + return "Line"; + attr = snd_hda_get_input_pin_attr(def_conf); + if (!attr) + return "None"; + if (attr == INPUT_PIN_ATTR_DOCK) + return "Dock Line"; + return "Line"; + case AC_JACK_AUX: + return "Aux"; + case AC_JACK_CD: + return "CD"; + case AC_JACK_SPDIF_IN: + return "SPDIF In"; + case AC_JACK_DIG_OTHER_IN: + return "Digital In"; + default: + return "Misc"; + } +} + +/* Check whether the location prefix needs to be added to the label. + * If all mic-jacks are in the same location (e.g. rear panel), we don't + * have to put "Front" prefix to each label. In such a case, returns false. + */ +static int check_mic_location_need(struct hda_codec *codec, + const struct auto_pin_cfg *cfg, + int input) +{ + unsigned int defc; + int i, attr, attr2; + + defc = snd_hda_codec_get_pincfg(codec, cfg->inputs[input].pin); + attr = snd_hda_get_input_pin_attr(defc); + /* for internal or docking mics, we need locations */ + if (attr <= INPUT_PIN_ATTR_NORMAL) + return 1; + + attr = 0; + for (i = 0; i < cfg->num_inputs; i++) { + defc = snd_hda_codec_get_pincfg(codec, cfg->inputs[i].pin); + attr2 = snd_hda_get_input_pin_attr(defc); + if (attr2 >= INPUT_PIN_ATTR_NORMAL) { + if (attr && attr != attr2) + return 1; /* different locations found */ + attr = attr2; + } + } + return 0; +} + +/** + * hda_get_autocfg_input_label - Get a label for the given input + * + * Get a label for the given input pin defined by the autocfg item. + * Unlike hda_get_input_pin_label(), this function checks all inputs + * defined in autocfg and avoids the redundant mic/line prefix as much as + * possible. + */ +const char *hda_get_autocfg_input_label(struct hda_codec *codec, + const struct auto_pin_cfg *cfg, + int input) +{ + int type = cfg->inputs[input].type; + int has_multiple_pins = 0; + + if ((input > 0 && cfg->inputs[input - 1].type == type) || + (input < cfg->num_inputs - 1 && cfg->inputs[input + 1].type == type)) + has_multiple_pins = 1; + if (has_multiple_pins && type == AUTO_PIN_MIC) + has_multiple_pins &= check_mic_location_need(codec, cfg, input); + return hda_get_input_pin_label(codec, cfg->inputs[input].pin, + has_multiple_pins); +} +EXPORT_SYMBOL_HDA(hda_get_autocfg_input_label); + +/* return the position of NID in the list, or -1 if not found */ +static int find_idx_in_nid_list(hda_nid_t nid, const hda_nid_t *list, int nums) +{ + int i; + for (i = 0; i < nums; i++) + if (list[i] == nid) + return i; + return -1; +} + +/* get a unique suffix or an index number */ +static const char *check_output_sfx(hda_nid_t nid, const hda_nid_t *pins, + int num_pins, int *indexp) +{ + static const char * const channel_sfx[] = { + " Front", " Surround", " CLFE", " Side" + }; + int i; + + i = find_idx_in_nid_list(nid, pins, num_pins); + if (i < 0) + return NULL; + if (num_pins == 1) + return ""; + if (num_pins > ARRAY_SIZE(channel_sfx)) { + if (indexp) + *indexp = i; + return ""; + } + return channel_sfx[i]; +} + +static int fill_audio_out_name(struct hda_codec *codec, hda_nid_t nid, + const struct auto_pin_cfg *cfg, + const char *name, char *label, int maxlen, + int *indexp) +{ + unsigned int def_conf = snd_hda_codec_get_pincfg(codec, nid); + int attr = snd_hda_get_input_pin_attr(def_conf); + const char *pfx = "", *sfx = ""; + + /* handle as a speaker if it's a fixed line-out */ + if (!strcmp(name, "Line Out") && attr == INPUT_PIN_ATTR_INT) + name = "Speaker"; + /* check the location */ + switch (attr) { + case INPUT_PIN_ATTR_DOCK: + pfx = "Dock "; + break; + case INPUT_PIN_ATTR_FRONT: + pfx = "Front "; + break; + } + if (cfg) { + /* try to give a unique suffix if needed */ + sfx = check_output_sfx(nid, cfg->line_out_pins, cfg->line_outs, + indexp); + if (!sfx) + sfx = check_output_sfx(nid, cfg->speaker_pins, cfg->speaker_outs, + indexp); + if (!sfx) { + /* don't add channel suffix for Headphone controls */ + int idx = find_idx_in_nid_list(nid, cfg->hp_pins, + cfg->hp_outs); + if (idx >= 0) + *indexp = idx; + sfx = ""; + } + } + snprintf(label, maxlen, "%s%s%s", pfx, name, sfx); + return 1; +} + +/** + * snd_hda_get_pin_label - Get a label for the given I/O pin + * + * Get a label for the given pin. This function works for both input and + * output pins. When @cfg is given as non-NULL, the function tries to get + * an optimized label using hda_get_autocfg_input_label(). + * + * This function tries to give a unique label string for the pin as much as + * possible. For example, when the multiple line-outs are present, it adds + * the channel suffix like "Front", "Surround", etc (only when @cfg is given). + * If no unique name with a suffix is available and @indexp is non-NULL, the + * index number is stored in the pointer. + */ +int snd_hda_get_pin_label(struct hda_codec *codec, hda_nid_t nid, + const struct auto_pin_cfg *cfg, + char *label, int maxlen, int *indexp) +{ + unsigned int def_conf = snd_hda_codec_get_pincfg(codec, nid); + const char *name = NULL; + int i; + + if (indexp) + *indexp = 0; + if (get_defcfg_connect(def_conf) == AC_JACK_PORT_NONE) + return 0; + + switch (get_defcfg_device(def_conf)) { + case AC_JACK_LINE_OUT: + return fill_audio_out_name(codec, nid, cfg, "Line Out", + label, maxlen, indexp); + case AC_JACK_SPEAKER: + return fill_audio_out_name(codec, nid, cfg, "Speaker", + label, maxlen, indexp); + case AC_JACK_HP_OUT: + return fill_audio_out_name(codec, nid, cfg, "Headphone", + label, maxlen, indexp); + case AC_JACK_SPDIF_OUT: + case AC_JACK_DIG_OTHER_OUT: + if (get_defcfg_location(def_conf) == AC_JACK_LOC_HDMI) + name = "HDMI"; + else + name = "SPDIF"; + if (cfg && indexp) { + i = find_idx_in_nid_list(nid, cfg->dig_out_pins, + cfg->dig_outs); + if (i >= 0) + *indexp = i; + } + break; + default: + if (cfg) { + for (i = 0; i < cfg->num_inputs; i++) { + if (cfg->inputs[i].pin != nid) + continue; + name = hda_get_autocfg_input_label(codec, cfg, i); + if (name) + break; + } + } + if (!name) + name = hda_get_input_pin_label(codec, nid, true); + break; + } + if (!name) + return 0; + strlcpy(label, name, maxlen); + return 1; +} +EXPORT_SYMBOL_HDA(snd_hda_get_pin_label); + +int snd_hda_gen_add_verbs(struct hda_gen_spec *spec, + const struct hda_verb *list) +{ + const struct hda_verb **v; + snd_array_init(&spec->verbs, sizeof(struct hda_verb *), 8); + v = snd_array_new(&spec->verbs); + if (!v) + return -ENOMEM; + *v = list; + return 0; +} +EXPORT_SYMBOL_HDA(snd_hda_gen_add_verbs); + +void snd_hda_gen_apply_verbs(struct hda_codec *codec) +{ + struct hda_gen_spec *spec = codec->spec; + int i; + for (i = 0; i < spec->verbs.used; i++) { + struct hda_verb **v = snd_array_elem(&spec->verbs, i); + snd_hda_sequence_write(codec, *v); + } +} +EXPORT_SYMBOL_HDA(snd_hda_gen_apply_verbs); + +void snd_hda_apply_pincfgs(struct hda_codec *codec, + const struct hda_pintbl *cfg) +{ + for (; cfg->nid; cfg++) + snd_hda_codec_set_pincfg(codec, cfg->nid, cfg->val); +} +EXPORT_SYMBOL_HDA(snd_hda_apply_pincfgs); + +void snd_hda_apply_fixup(struct hda_codec *codec, int action) +{ + struct hda_gen_spec *spec = codec->spec; + int id = spec->fixup_id; +#ifdef CONFIG_SND_DEBUG_VERBOSE + const char *modelname = spec->fixup_name; +#endif + int depth = 0; + + if (!spec->fixup_list) + return; + + while (id >= 0) { + const struct hda_fixup *fix = spec->fixup_list + id; + + switch (fix->type) { + case HDA_FIXUP_PINS: + if (action != HDA_FIXUP_ACT_PRE_PROBE || !fix->v.pins) + break; + snd_printdd(KERN_INFO SFX + "%s: Apply pincfg for %s\n", + codec->chip_name, modelname); + snd_hda_apply_pincfgs(codec, fix->v.pins); + break; + case HDA_FIXUP_VERBS: + if (action != HDA_FIXUP_ACT_PROBE || !fix->v.verbs) + break; + snd_printdd(KERN_INFO SFX + "%s: Apply fix-verbs for %s\n", + codec->chip_name, modelname); + snd_hda_gen_add_verbs(codec->spec, fix->v.verbs); + break; + case HDA_FIXUP_FUNC: + if (!fix->v.func) + break; + snd_printdd(KERN_INFO SFX + "%s: Apply fix-func for %s\n", + codec->chip_name, modelname); + fix->v.func(codec, fix, action); + break; + default: + snd_printk(KERN_ERR SFX + "%s: Invalid fixup type %d\n", + codec->chip_name, fix->type); + break; + } + if (!fix->chained) + break; + if (++depth > 10) + break; + id = fix->chain_id; + } +} +EXPORT_SYMBOL_HDA(snd_hda_apply_fixup); + +void snd_hda_pick_fixup(struct hda_codec *codec, + const struct hda_model_fixup *models, + const struct snd_pci_quirk *quirk, + const struct hda_fixup *fixlist) +{ + struct hda_gen_spec *spec = codec->spec; + const struct snd_pci_quirk *q; + int id = -1; + const char *name = NULL; + + /* when model=nofixup is given, don't pick up any fixups */ + if (codec->modelname && !strcmp(codec->modelname, "nofixup")) { + spec->fixup_list = NULL; + spec->fixup_id = -1; + return; + } + + if (codec->modelname && models) { + while (models->name) { + if (!strcmp(codec->modelname, models->name)) { + id = models->id; + name = models->name; + break; + } + models++; + } + } + if (id < 0) { + q = snd_pci_quirk_lookup(codec->bus->pci, quirk); + if (q) { + id = q->value; +#ifdef CONFIG_SND_DEBUG_VERBOSE + name = q->name; +#endif + } + } + if (id < 0) { + for (q = quirk; q->subvendor; q++) { + unsigned int vendorid = + q->subdevice | (q->subvendor << 16); + if (vendorid == codec->subsystem_id) { + id = q->value; +#ifdef CONFIG_SND_DEBUG_VERBOSE + name = q->name; +#endif + break; + } + } + } + + spec->fixup_id = id; + if (id >= 0) { + spec->fixup_list = fixlist; + spec->fixup_name = name; + } +} +EXPORT_SYMBOL_HDA(snd_hda_pick_fixup); diff --git a/sound/pci/hda/hda_auto_parser.h b/sound/pci/hda/hda_auto_parser.h new file mode 100644 index 00000000000..2a7889dfbd1 --- /dev/null +++ b/sound/pci/hda/hda_auto_parser.h @@ -0,0 +1,160 @@ +/* + * BIOS auto-parser helper functions for HD-audio + * + * Copyright (c) 2012 Takashi Iwai <tiwai@suse.de> + * + * This driver is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License as published by + * the Free Software Foundation; either version 2 of the License, or + * (at your option) any later version. + */ + +#ifndef __SOUND_HDA_AUTO_PARSER_H +#define __SOUND_HDA_AUTO_PARSER_H + +/* + * Helper for automatic pin configuration + */ + +enum { + AUTO_PIN_MIC, + AUTO_PIN_LINE_IN, + AUTO_PIN_CD, + AUTO_PIN_AUX, + AUTO_PIN_LAST +}; + +enum { + AUTO_PIN_LINE_OUT, + AUTO_PIN_SPEAKER_OUT, + AUTO_PIN_HP_OUT +}; + +#define AUTO_CFG_MAX_OUTS HDA_MAX_OUTS +#define AUTO_CFG_MAX_INS 8 + +struct auto_pin_cfg_item { + hda_nid_t pin; + int type; +}; + +struct auto_pin_cfg; +const char *hda_get_autocfg_input_label(struct hda_codec *codec, + const struct auto_pin_cfg *cfg, + int input); +int snd_hda_get_pin_label(struct hda_codec *codec, hda_nid_t nid, + const struct auto_pin_cfg *cfg, + char *label, int maxlen, int *indexp); + +enum { + INPUT_PIN_ATTR_UNUSED, /* pin not connected */ + INPUT_PIN_ATTR_INT, /* internal mic/line-in */ + INPUT_PIN_ATTR_DOCK, /* docking mic/line-in */ + INPUT_PIN_ATTR_NORMAL, /* mic/line-in jack */ + INPUT_PIN_ATTR_FRONT, /* mic/line-in jack in front */ + INPUT_PIN_ATTR_REAR, /* mic/line-in jack in rear */ +}; + +int snd_hda_get_input_pin_attr(unsigned int def_conf); + +struct auto_pin_cfg { + int line_outs; + /* sorted in the order of Front/Surr/CLFE/Side */ + hda_nid_t line_out_pins[AUTO_CFG_MAX_OUTS]; + int speaker_outs; + hda_nid_t speaker_pins[AUTO_CFG_MAX_OUTS]; + int hp_outs; + int line_out_type; /* AUTO_PIN_XXX_OUT */ + hda_nid_t hp_pins[AUTO_CFG_MAX_OUTS]; + int num_inputs; + struct auto_pin_cfg_item inputs[AUTO_CFG_MAX_INS]; + int dig_outs; + hda_nid_t dig_out_pins[2]; + hda_nid_t dig_in_pin; + hda_nid_t mono_out_pin; + int dig_out_type[2]; /* HDA_PCM_TYPE_XXX */ + int dig_in_type; /* HDA_PCM_TYPE_XXX */ +}; + +/* bit-flags for snd_hda_parse_pin_def_config() behavior */ +#define HDA_PINCFG_NO_HP_FIXUP (1 << 0) /* no HP-split */ +#define HDA_PINCFG_NO_LO_FIXUP (1 << 1) /* don't take other outs as LO */ + +int snd_hda_parse_pin_defcfg(struct hda_codec *codec, + struct auto_pin_cfg *cfg, + const hda_nid_t *ignore_nids, + unsigned int cond_flags); + +/* older function */ +#define snd_hda_parse_pin_def_config(codec, cfg, ignore) \ + snd_hda_parse_pin_defcfg(codec, cfg, ignore, 0) + +/* + */ + +struct hda_gen_spec { + /* fix-up list */ + int fixup_id; + const struct hda_fixup *fixup_list; + const char *fixup_name; + + /* additional init verbs */ + struct snd_array verbs; +}; + + +/* + * Fix-up pin default configurations and add default verbs + */ + +struct hda_pintbl { + hda_nid_t nid; + u32 val; +}; + +struct hda_model_fixup { + const int id; + const char *name; +}; + +struct hda_fixup { + int type; + bool chained; + int chain_id; + union { + const struct hda_pintbl *pins; + const struct hda_verb *verbs; + void (*func)(struct hda_codec *codec, + const struct hda_fixup *fix, + int action); + } v; +}; + +/* fixup types */ +enum { + HDA_FIXUP_INVALID, + HDA_FIXUP_PINS, + HDA_FIXUP_VERBS, + HDA_FIXUP_FUNC, +}; + +/* fixup action definitions */ +enum { + HDA_FIXUP_ACT_PRE_PROBE, + HDA_FIXUP_ACT_PROBE, + HDA_FIXUP_ACT_INIT, + HDA_FIXUP_ACT_BUILD, +}; + +int snd_hda_gen_add_verbs(struct hda_gen_spec *spec, + const struct hda_verb *list); +void snd_hda_gen_apply_verbs(struct hda_codec *codec); +void snd_hda_apply_pincfgs(struct hda_codec *codec, + const struct hda_pintbl *cfg); +void snd_hda_apply_fixup(struct hda_codec *codec, int action); +void snd_hda_pick_fixup(struct hda_codec *codec, + const struct hda_model_fixup *models, + const struct snd_pci_quirk *quirk, + const struct hda_fixup *fixlist); + +#endif /* __SOUND_HDA_AUTO_PARSER_H */ diff --git a/sound/pci/hda/hda_codec.c b/sound/pci/hda/hda_codec.c index 841475cc13b..eb09a334832 100644 --- a/sound/pci/hda/hda_codec.c +++ b/sound/pci/hda/hda_codec.c @@ -334,78 +334,67 @@ static hda_nid_t *lookup_conn_list(struct snd_array *array, hda_nid_t nid) return NULL; } +/* read the connection and add to the cache */ +static int read_and_add_raw_conns(struct hda_codec *codec, hda_nid_t nid) +{ + hda_nid_t list[HDA_MAX_CONNECTIONS]; + int len; + + len = snd_hda_get_raw_connections(codec, nid, list, ARRAY_SIZE(list)); + if (len < 0) + return len; + return snd_hda_override_conn_list(codec, nid, len, list); +} + /** - * snd_hda_get_conn_list - get connection list + * snd_hda_get_connections - copy connection list * @codec: the HDA codec * @nid: NID to parse - * @listp: the pointer to store NID list + * @conn_list: connection list array; when NULL, checks only the size + * @max_conns: max. number of connections to store * * Parses the connection list of the given widget and stores the list * of NIDs. * * Returns the number of connections, or a negative error code. */ -int snd_hda_get_conn_list(struct hda_codec *codec, hda_nid_t nid, - const hda_nid_t **listp) +int snd_hda_get_connections(struct hda_codec *codec, hda_nid_t nid, + hda_nid_t *conn_list, int max_conns) { struct snd_array *array = &codec->conn_lists; - int len, err; - hda_nid_t list[HDA_MAX_CONNECTIONS]; + int len; hda_nid_t *p; bool added = false; again: + mutex_lock(&codec->hash_mutex); + len = -1; /* if the connection-list is already cached, read it */ p = lookup_conn_list(array, nid); if (p) { - if (listp) - *listp = p + 2; - return p[1]; + len = p[1]; + if (conn_list && len > max_conns) { + snd_printk(KERN_ERR "hda_codec: " + "Too many connections %d for NID 0x%x\n", + len, nid); + mutex_unlock(&codec->hash_mutex); + return -EINVAL; + } + if (conn_list && len) + memcpy(conn_list, p + 2, len * sizeof(hda_nid_t)); } + mutex_unlock(&codec->hash_mutex); + if (len >= 0) + return len; if (snd_BUG_ON(added)) return -EINVAL; - /* read the connection and add to the cache */ - len = snd_hda_get_raw_connections(codec, nid, list, HDA_MAX_CONNECTIONS); + len = read_and_add_raw_conns(codec, nid); if (len < 0) return len; - err = snd_hda_override_conn_list(codec, nid, len, list); - if (err < 0) - return err; added = true; goto again; } -EXPORT_SYMBOL_HDA(snd_hda_get_conn_list); - -/** - * snd_hda_get_connections - copy connection list - * @codec: the HDA codec - * @nid: NID to parse - * @conn_list: connection list array - * @max_conns: max. number of connections to store - * - * Parses the connection list of the given widget and stores the list - * of NIDs. - * - * Returns the number of connections, or a negative error code. - */ -int snd_hda_get_connections(struct hda_codec *codec, hda_nid_t nid, - hda_nid_t *conn_list, int max_conns) -{ - const hda_nid_t *list; - int len = snd_hda_get_conn_list(codec, nid, &list); - - if (len <= 0) - return len; - if (len > max_conns) { - snd_printk(KERN_ERR "hda_codec: " - "Too many connections %d for NID 0x%x\n", - len, nid); - return -EINVAL; - } - memcpy(conn_list, list, len * sizeof(hda_nid_t)); - return len; -} EXPORT_SYMBOL_HDA(snd_hda_get_connections); /** @@ -543,6 +532,7 @@ int snd_hda_override_conn_list(struct hda_codec *codec, hda_nid_t nid, int len, hda_nid_t *p; int i, old_used; + mutex_lock(&codec->hash_mutex); p = lookup_conn_list(array, nid); if (p) *p = -1; /* invalidate the old entry */ @@ -553,10 +543,12 @@ int snd_hda_override_conn_list(struct hda_codec *codec, hda_nid_t nid, int len, for (i = 0; i < len; i++) if (!add_conn_list(array, list[i])) goto error_add; + mutex_unlock(&codec->hash_mutex); return 0; error_add: array->used = old_used; + mutex_unlock(&codec->hash_mutex); return -ENOMEM; } EXPORT_SYMBOL_HDA(snd_hda_override_conn_list); @@ -1255,6 +1247,7 @@ int /*__devinit*/ snd_hda_codec_new(struct hda_bus *bus, codec->addr = codec_addr; mutex_init(&codec->spdif_mutex); mutex_init(&codec->control_mutex); + mutex_init(&codec->hash_mutex); init_hda_cache(&codec->amp_cache, sizeof(struct hda_amp_info)); init_hda_cache(&codec->cmd_cache, sizeof(struct hda_cache_head)); snd_array_init(&codec->mixers, sizeof(struct hda_nid_item), 32); @@ -1264,15 +1257,9 @@ int /*__devinit*/ snd_hda_codec_new(struct hda_bus *bus, snd_array_init(&codec->cvt_setups, sizeof(struct hda_cvt_setup), 8); snd_array_init(&codec->conn_lists, sizeof(hda_nid_t), 64); snd_array_init(&codec->spdif_out, sizeof(struct hda_spdif_out), 16); - if (codec->bus->modelname) { - codec->modelname = kstrdup(codec->bus->modelname, GFP_KERNEL); - if (!codec->modelname) { - snd_hda_codec_free(codec); - return -ENODEV; - } - } #ifdef CONFIG_SND_HDA_POWER_SAVE + spin_lock_init(&codec->power_lock); INIT_DELAYED_WORK(&codec->power_work, hda_power_work); /* snd_hda_codec_new() marks the codec as power-up, and leave it as is. * the caller has to power down appropriatley after initialization @@ -1281,6 +1268,14 @@ int /*__devinit*/ snd_hda_codec_new(struct hda_bus *bus, hda_keep_power_on(codec); #endif + if (codec->bus->modelname) { + codec->modelname = kstrdup(codec->bus->modelname, GFP_KERNEL); + if (!codec->modelname) { + snd_hda_codec_free(codec); + return -ENODEV; + } + } + list_add_tail(&codec->list, &bus->codec_list); bus->caddr_tbl[codec_addr] = codec; @@ -1603,6 +1598,60 @@ get_alloc_amp_hash(struct hda_codec *codec, u32 key) return (struct hda_amp_info *)get_alloc_hash(&codec->amp_cache, key); } +/* overwrite the value with the key in the caps hash */ +static int write_caps_hash(struct hda_codec *codec, u32 key, unsigned int val) +{ + struct hda_amp_info *info; + + mutex_lock(&codec->hash_mutex); + info = get_alloc_amp_hash(codec, key); + if (!info) { + mutex_unlock(&codec->hash_mutex); + return -EINVAL; + } + info->amp_caps = val; + info->head.val |= INFO_AMP_CAPS; + mutex_unlock(&codec->hash_mutex); + return 0; +} + +/* query the value from the caps hash; if not found, fetch the current + * value from the given function and store in the hash + */ +static unsigned int +query_caps_hash(struct hda_codec *codec, hda_nid_t nid, int dir, u32 key, + unsigned int (*func)(struct hda_codec *, hda_nid_t, int)) +{ + struct hda_amp_info *info; + unsigned int val; + + mutex_lock(&codec->hash_mutex); + info = get_alloc_amp_hash(codec, key); + if (!info) { + mutex_unlock(&codec->hash_mutex); + return 0; + } + if (!(info->head.val & INFO_AMP_CAPS)) { + mutex_unlock(&codec->hash_mutex); /* for reentrance */ + val = func(codec, nid, dir); + write_caps_hash(codec, key, val); + } else { + val = info->amp_caps; + mutex_unlock(&codec->hash_mutex); + } + return val; +} + +static unsigned int read_amp_cap(struct hda_codec *codec, hda_nid_t nid, + int direction) +{ + if (!(get_wcaps(codec, nid) & AC_WCAP_AMP_OVRD)) + nid = codec->afg; + return snd_hda_param_read(codec, nid, + direction == HDA_OUTPUT ? + AC_PAR_AMP_OUT_CAP : AC_PAR_AMP_IN_CAP); +} + /** * query_amp_caps - query AMP capabilities * @codec: the HD-auio codec @@ -1617,22 +1666,9 @@ get_alloc_amp_hash(struct hda_codec *codec, u32 key) */ u32 query_amp_caps(struct hda_codec *codec, hda_nid_t nid, int direction) { - struct hda_amp_info *info; - - info = get_alloc_amp_hash(codec, HDA_HASH_KEY(nid, direction, 0)); - if (!info) - return 0; - if (!(info->head.val & INFO_AMP_CAPS)) { - if (!(get_wcaps(codec, nid) & AC_WCAP_AMP_OVRD)) - nid = codec->afg; - info->amp_caps = snd_hda_param_read(codec, nid, - direction == HDA_OUTPUT ? - AC_PAR_AMP_OUT_CAP : - AC_PAR_AMP_IN_CAP); - if (info->amp_caps) - info->head.val |= INFO_AMP_CAPS; - } - return info->amp_caps; + return query_caps_hash(codec, nid, direction, + HDA_HASH_KEY(nid, direction, 0), + read_amp_cap); } EXPORT_SYMBOL_HDA(query_amp_caps); @@ -1652,34 +1688,12 @@ EXPORT_SYMBOL_HDA(query_amp_caps); int snd_hda_override_amp_caps(struct hda_codec *codec, hda_nid_t nid, int dir, unsigned int caps) { - struct hda_amp_info *info; - - info = get_alloc_amp_hash(codec, HDA_HASH_KEY(nid, dir, 0)); - if (!info) - return -EINVAL; - info->amp_caps = caps; - info->head.val |= INFO_AMP_CAPS; - return 0; + return write_caps_hash(codec, HDA_HASH_KEY(nid, dir, 0), caps); } EXPORT_SYMBOL_HDA(snd_hda_override_amp_caps); -static unsigned int -query_caps_hash(struct hda_codec *codec, hda_nid_t nid, u32 key, - unsigned int (*func)(struct hda_codec *, hda_nid_t)) -{ - struct hda_amp_info *info; - - info = get_alloc_amp_hash(codec, key); - if (!info) - return 0; - if (!info->head.val) { - info->head.val |= INFO_AMP_CAPS; - info->amp_caps = func(codec, nid); - } - return info->amp_caps; -} - -static unsigned int read_pin_cap(struct hda_codec *codec, hda_nid_t nid) +static unsigned int read_pin_cap(struct hda_codec *codec, hda_nid_t nid, + int dir) { return snd_hda_param_read(codec, nid, AC_PAR_PIN_CAP); } @@ -1697,7 +1711,7 @@ static unsigned int read_pin_cap(struct hda_codec *codec, hda_nid_t nid) */ u32 snd_hda_query_pin_caps(struct hda_codec *codec, hda_nid_t nid) { - return query_caps_hash(codec, nid, HDA_HASH_PINCAP_KEY(nid), + return query_caps_hash(codec, nid, 0, HDA_HASH_PINCAP_KEY(nid), read_pin_cap); } EXPORT_SYMBOL_HDA(snd_hda_query_pin_caps); @@ -1715,41 +1729,47 @@ EXPORT_SYMBOL_HDA(snd_hda_query_pin_caps); int snd_hda_override_pin_caps(struct hda_codec *codec, hda_nid_t nid, unsigned int caps) { - struct hda_amp_info *info; - info = get_alloc_amp_hash(codec, HDA_HASH_PINCAP_KEY(nid)); - if (!info) - return -ENOMEM; - info->amp_caps = caps; - info->head.val |= INFO_AMP_CAPS; - return 0; + return write_caps_hash(codec, HDA_HASH_PINCAP_KEY(nid), caps); } EXPORT_SYMBOL_HDA(snd_hda_override_pin_caps); -/* - * read the current volume to info - * if the cache exists, read the cache value. +/* read or sync the hash value with the current value; + * call within hash_mutex */ -static unsigned int get_vol_mute(struct hda_codec *codec, - struct hda_amp_info *info, hda_nid_t nid, - int ch, int direction, int index) +static struct hda_amp_info * +update_amp_hash(struct hda_codec *codec, hda_nid_t nid, int ch, + int direction, int index) { - u32 val, parm; - - if (info->head.val & INFO_AMP_VOL(ch)) - return info->vol[ch]; + struct hda_amp_info *info; + unsigned int parm, val = 0; + bool val_read = false; - parm = ch ? AC_AMP_GET_RIGHT : AC_AMP_GET_LEFT; - parm |= direction == HDA_OUTPUT ? AC_AMP_GET_OUTPUT : AC_AMP_GET_INPUT; - parm |= index; - val = snd_hda_codec_read(codec, nid, 0, + retry: + info = get_alloc_amp_hash(codec, HDA_HASH_KEY(nid, direction, index)); + if (!info) + return NULL; + if (!(info->head.val & INFO_AMP_VOL(ch))) { + if (!val_read) { + mutex_unlock(&codec->hash_mutex); + parm = ch ? AC_AMP_GET_RIGHT : AC_AMP_GET_LEFT; + parm |= direction == HDA_OUTPUT ? + AC_AMP_GET_OUTPUT : AC_AMP_GET_INPUT; + parm |= index; + val = snd_hda_codec_read(codec, nid, 0, AC_VERB_GET_AMP_GAIN_MUTE, parm); - info->vol[ch] = val & 0xff; - info->head.val |= INFO_AMP_VOL(ch); - return info->vol[ch]; + val &= 0xff; + val_read = true; + mutex_lock(&codec->hash_mutex); + goto retry; + } + info->vol[ch] = val; + info->head.val |= INFO_AMP_VOL(ch); + } + return info; } /* - * write the current volume in info to the h/w and update the cache + * write the current volume in info to the h/w */ static void put_vol_mute(struct hda_codec *codec, struct hda_amp_info *info, hda_nid_t nid, int ch, int direction, int index, @@ -1766,7 +1786,6 @@ static void put_vol_mute(struct hda_codec *codec, struct hda_amp_info *info, else parm |= val; snd_hda_codec_write(codec, nid, 0, AC_VERB_SET_AMP_GAIN_MUTE, parm); - info->vol[ch] = val; } /** @@ -1783,10 +1802,14 @@ int snd_hda_codec_amp_read(struct hda_codec *codec, hda_nid_t nid, int ch, int direction, int index) { struct hda_amp_info *info; - info = get_alloc_amp_hash(codec, HDA_HASH_KEY(nid, direction, index)); - if (!info) - return 0; - return get_vol_mute(codec, info, nid, ch, direction, index); + unsigned int val = 0; + + mutex_lock(&codec->hash_mutex); + info = update_amp_hash(codec, nid, ch, direction, index); + if (info) + val = info->vol[ch]; + mutex_unlock(&codec->hash_mutex); + return val; } EXPORT_SYMBOL_HDA(snd_hda_codec_amp_read); @@ -1808,15 +1831,23 @@ int snd_hda_codec_amp_update(struct hda_codec *codec, hda_nid_t nid, int ch, { struct hda_amp_info *info; - info = get_alloc_amp_hash(codec, HDA_HASH_KEY(nid, direction, idx)); - if (!info) - return 0; if (snd_BUG_ON(mask & ~0xff)) mask &= 0xff; val &= mask; - val |= get_vol_mute(codec, info, nid, ch, direction, idx) & ~mask; - if (info->vol[ch] == val) + + mutex_lock(&codec->hash_mutex); + info = update_amp_hash(codec, nid, ch, direction, idx); + if (!info) { + mutex_unlock(&codec->hash_mutex); + return 0; + } + val |= info->vol[ch] & ~mask; + if (info->vol[ch] == val) { + mutex_unlock(&codec->hash_mutex); return 0; + } + info->vol[ch] = val; + mutex_unlock(&codec->hash_mutex); put_vol_mute(codec, info, nid, ch, direction, idx, val); return 1; } @@ -2263,7 +2294,10 @@ int snd_hda_codec_reset(struct hda_codec *codec) /* OK, let it free */ #ifdef CONFIG_SND_HDA_POWER_SAVE - cancel_delayed_work(&codec->power_work); + cancel_delayed_work_sync(&codec->power_work); + codec->power_on = 0; + codec->power_transition = 0; + codec->power_jiffies = jiffies; flush_workqueue(codec->bus->workq); #endif snd_hda_ctls_clear(codec); @@ -2859,12 +2893,15 @@ static int snd_hda_spdif_default_get(struct snd_kcontrol *kcontrol, { struct hda_codec *codec = snd_kcontrol_chip(kcontrol); int idx = kcontrol->private_value; - struct hda_spdif_out *spdif = snd_array_elem(&codec->spdif_out, idx); + struct hda_spdif_out *spdif; + mutex_lock(&codec->spdif_mutex); + spdif = snd_array_elem(&codec->spdif_out, idx); ucontrol->value.iec958.status[0] = spdif->status & 0xff; ucontrol->value.iec958.status[1] = (spdif->status >> 8) & 0xff; ucontrol->value.iec958.status[2] = (spdif->status >> 16) & 0xff; ucontrol->value.iec958.status[3] = (spdif->status >> 24) & 0xff; + mutex_unlock(&codec->spdif_mutex); return 0; } @@ -2950,12 +2987,14 @@ static int snd_hda_spdif_default_put(struct snd_kcontrol *kcontrol, { struct hda_codec *codec = snd_kcontrol_chip(kcontrol); int idx = kcontrol->private_value; - struct hda_spdif_out *spdif = snd_array_elem(&codec->spdif_out, idx); - hda_nid_t nid = spdif->nid; + struct hda_spdif_out *spdif; + hda_nid_t nid; unsigned short val; int change; mutex_lock(&codec->spdif_mutex); + spdif = snd_array_elem(&codec->spdif_out, idx); + nid = spdif->nid; spdif->status = ucontrol->value.iec958.status[0] | ((unsigned int)ucontrol->value.iec958.status[1] << 8) | ((unsigned int)ucontrol->value.iec958.status[2] << 16) | @@ -2977,9 +3016,12 @@ static int snd_hda_spdif_out_switch_get(struct snd_kcontrol *kcontrol, { struct hda_codec *codec = snd_kcontrol_chip(kcontrol); int idx = kcontrol->private_value; - struct hda_spdif_out *spdif = snd_array_elem(&codec->spdif_out, idx); + struct hda_spdif_out *spdif; + mutex_lock(&codec->spdif_mutex); + spdif = snd_array_elem(&codec->spdif_out, idx); ucontrol->value.integer.value[0] = spdif->ctls & AC_DIG1_ENABLE; + mutex_unlock(&codec->spdif_mutex); return 0; } @@ -2999,12 +3041,14 @@ static int snd_hda_spdif_out_switch_put(struct snd_kcontrol *kcontrol, { struct hda_codec *codec = snd_kcontrol_chip(kcontrol); int idx = kcontrol->private_value; - struct hda_spdif_out *spdif = snd_array_elem(&codec->spdif_out, idx); - hda_nid_t nid = spdif->nid; + struct hda_spdif_out *spdif; + hda_nid_t nid; unsigned short val; int change; mutex_lock(&codec->spdif_mutex); + spdif = snd_array_elem(&codec->spdif_out, idx); + nid = spdif->nid; val = spdif->ctls & ~AC_DIG1_ENABLE; if (ucontrol->value.integer.value[0]) val |= AC_DIG1_ENABLE; @@ -3092,6 +3136,9 @@ int snd_hda_create_spdif_out_ctls(struct hda_codec *codec, } EXPORT_SYMBOL_HDA(snd_hda_create_spdif_out_ctls); +/* get the hda_spdif_out entry from the given NID + * call within spdif_mutex lock + */ struct hda_spdif_out *snd_hda_spdif_out_of_nid(struct hda_codec *codec, hda_nid_t nid) { @@ -3108,9 +3155,10 @@ EXPORT_SYMBOL_HDA(snd_hda_spdif_out_of_nid); void snd_hda_spdif_ctls_unassign(struct hda_codec *codec, int idx) { - struct hda_spdif_out *spdif = snd_array_elem(&codec->spdif_out, idx); + struct hda_spdif_out *spdif; mutex_lock(&codec->spdif_mutex); + spdif = snd_array_elem(&codec->spdif_out, idx); spdif->nid = (u16)-1; mutex_unlock(&codec->spdif_mutex); } @@ -3118,10 +3166,11 @@ EXPORT_SYMBOL_HDA(snd_hda_spdif_ctls_unassign); void snd_hda_spdif_ctls_assign(struct hda_codec *codec, int idx, hda_nid_t nid) { - struct hda_spdif_out *spdif = snd_array_elem(&codec->spdif_out, idx); + struct hda_spdif_out *spdif; unsigned short val; mutex_lock(&codec->spdif_mutex); + spdif = snd_array_elem(&codec->spdif_out, idx); if (spdif->nid != nid) { spdif->nid = nid; val = spdif->ctls; @@ -3486,11 +3535,14 @@ static void hda_call_codec_suspend(struct hda_codec *codec) codec->afg ? codec->afg : codec->mfg, AC_PWRST_D3); #ifdef CONFIG_SND_HDA_POWER_SAVE - snd_hda_update_power_acct(codec); cancel_delayed_work(&codec->power_work); + spin_lock(&codec->power_lock); + snd_hda_update_power_acct(codec); + trace_hda_power_down(codec); codec->power_on = 0; codec->power_transition = 0; codec->power_jiffies = jiffies; + spin_unlock(&codec->power_lock); #endif } @@ -3499,6 +3551,10 @@ static void hda_call_codec_suspend(struct hda_codec *codec) */ static void hda_call_codec_resume(struct hda_codec *codec) { + /* set as if powered on for avoiding re-entering the resume + * in the resume / power-save sequence + */ + hda_keep_power_on(codec); hda_set_power_state(codec, codec->afg ? codec->afg : codec->mfg, AC_PWRST_D0); @@ -3514,6 +3570,7 @@ static void hda_call_codec_resume(struct hda_codec *codec) snd_hda_codec_resume_amp(codec); snd_hda_codec_resume_cache(codec); } + snd_hda_power_down(codec); /* flag down before returning */ } #endif /* CONFIG_PM */ @@ -3665,7 +3722,8 @@ unsigned int snd_hda_calc_stream_format(unsigned int rate, } EXPORT_SYMBOL_HDA(snd_hda_calc_stream_format); -static unsigned int get_pcm_param(struct hda_codec *codec, hda_nid_t nid) +static unsigned int get_pcm_param(struct hda_codec *codec, hda_nid_t nid, + int dir) { unsigned int val = 0; if (nid != codec->afg && @@ -3680,11 +3738,12 @@ static unsigned int get_pcm_param(struct hda_codec *codec, hda_nid_t nid) static unsigned int query_pcm_param(struct hda_codec *codec, hda_nid_t nid) { - return query_caps_hash(codec, nid, HDA_HASH_PARPCM_KEY(nid), + return query_caps_hash(codec, nid, 0, HDA_HASH_PARPCM_KEY(nid), get_pcm_param); } -static unsigned int get_stream_param(struct hda_codec *codec, hda_nid_t nid) +static unsigned int get_stream_param(struct hda_codec *codec, hda_nid_t nid, + int dir) { unsigned int streams = snd_hda_param_read(codec, nid, AC_PAR_STREAM); if (!streams || streams == -1) @@ -3696,7 +3755,7 @@ static unsigned int get_stream_param(struct hda_codec *codec, hda_nid_t nid) static unsigned int query_stream_param(struct hda_codec *codec, hda_nid_t nid) { - return query_caps_hash(codec, nid, HDA_HASH_PARSTR_KEY(nid), + return query_caps_hash(codec, nid, 0, HDA_HASH_PARSTR_KEY(nid), get_stream_param); } @@ -3775,11 +3834,13 @@ int snd_hda_query_supported_pcm(struct hda_codec *codec, hda_nid_t nid, bps = 20; } } +#if 0 /* FIXME: CS4206 doesn't work, which is the only codec supporting float */ if (streams & AC_SUPFMT_FLOAT32) { formats |= SNDRV_PCM_FMTBIT_FLOAT_LE; if (!bps) bps = 32; } +#endif if (streams == AC_SUPFMT_AC3) { /* should be exclusive */ /* temporary hack: we have still no proper support @@ -4283,12 +4344,18 @@ static void hda_power_work(struct work_struct *work) container_of(work, struct hda_codec, power_work.work); struct hda_bus *bus = codec->bus; + spin_lock(&codec->power_lock); + if (codec->power_transition > 0) { /* during power-up sequence? */ + spin_unlock(&codec->power_lock); + return; + } if (!codec->power_on || codec->power_count) { codec->power_transition = 0; + spin_unlock(&codec->power_lock); return; } + spin_unlock(&codec->power_lock); - trace_hda_power_down(codec); hda_call_codec_suspend(codec); if (bus->ops.pm_notify) bus->ops.pm_notify(bus); @@ -4296,9 +4363,11 @@ static void hda_power_work(struct work_struct *work) static void hda_keep_power_on(struct hda_codec *codec) { + spin_lock(&codec->power_lock); codec->power_count++; codec->power_on = 1; codec->power_jiffies = jiffies; + spin_unlock(&codec->power_lock); } /* update the power on/off account with the current jiffies */ @@ -4323,19 +4392,31 @@ void snd_hda_power_up(struct hda_codec *codec) { struct hda_bus *bus = codec->bus; + spin_lock(&codec->power_lock); codec->power_count++; - if (codec->power_on || codec->power_transition) + if (codec->power_on || codec->power_transition > 0) { + spin_unlock(&codec->power_lock); return; + } + spin_unlock(&codec->power_lock); + cancel_delayed_work_sync(&codec->power_work); + + spin_lock(&codec->power_lock); trace_hda_power_up(codec); snd_hda_update_power_acct(codec); codec->power_on = 1; codec->power_jiffies = jiffies; + codec->power_transition = 1; /* avoid reentrance */ + spin_unlock(&codec->power_lock); + if (bus->ops.pm_notify) bus->ops.pm_notify(bus); hda_call_codec_resume(codec); - cancel_delayed_work(&codec->power_work); + + spin_lock(&codec->power_lock); codec->power_transition = 0; + spin_unlock(&codec->power_lock); } EXPORT_SYMBOL_HDA(snd_hda_power_up); @@ -4351,14 +4432,18 @@ EXPORT_SYMBOL_HDA(snd_hda_power_up); */ void snd_hda_power_down(struct hda_codec *codec) { + spin_lock(&codec->power_lock); --codec->power_count; - if (!codec->power_on || codec->power_count || codec->power_transition) + if (!codec->power_on || codec->power_count || codec->power_transition) { + spin_unlock(&codec->power_lock); return; + } if (power_save(codec)) { - codec->power_transition = 1; /* avoid reentrance */ + codec->power_transition = -1; /* avoid reentrance */ queue_delayed_work(codec->bus->workq, &codec->power_work, msecs_to_jiffies(power_save(codec) * 1000)); } + spin_unlock(&codec->power_lock); } EXPORT_SYMBOL_HDA(snd_hda_power_down); @@ -4710,11 +4795,11 @@ int snd_hda_multi_out_analog_prepare(struct hda_codec *codec, { const hda_nid_t *nids = mout->dac_nids; int chs = substream->runtime->channels; - struct hda_spdif_out *spdif = - snd_hda_spdif_out_of_nid(codec, mout->dig_out_nid); + struct hda_spdif_out *spdif; int i; mutex_lock(&codec->spdif_mutex); + spdif = snd_hda_spdif_out_of_nid(codec, mout->dig_out_nid); if (mout->dig_out_nid && mout->share_spdif && mout->dig_out_used != HDA_DIG_EXCLUSIVE) { if (chs == 2 && @@ -4795,601 +4880,58 @@ int snd_hda_multi_out_analog_cleanup(struct hda_codec *codec, } EXPORT_SYMBOL_HDA(snd_hda_multi_out_analog_cleanup); -/* - * Helper for automatic pin configuration - */ - -static int is_in_nid_list(hda_nid_t nid, const hda_nid_t *list) -{ - for (; *list; list++) - if (*list == nid) - return 1; - return 0; -} - - -/* - * Sort an associated group of pins according to their sequence numbers. - */ -static void sort_pins_by_sequence(hda_nid_t *pins, short *sequences, - int num_pins) -{ - int i, j; - short seq; - hda_nid_t nid; - - for (i = 0; i < num_pins; i++) { - for (j = i + 1; j < num_pins; j++) { - if (sequences[i] > sequences[j]) { - seq = sequences[i]; - sequences[i] = sequences[j]; - sequences[j] = seq; - nid = pins[i]; - pins[i] = pins[j]; - pins[j] = nid; - } - } - } -} - - -/* add the found input-pin to the cfg->inputs[] table */ -static void add_auto_cfg_input_pin(struct auto_pin_cfg *cfg, hda_nid_t nid, - int type) -{ - if (cfg->num_inputs < AUTO_CFG_MAX_INS) { - cfg->inputs[cfg->num_inputs].pin = nid; - cfg->inputs[cfg->num_inputs].type = type; - cfg->num_inputs++; - } -} - -/* sort inputs in the order of AUTO_PIN_* type */ -static void sort_autocfg_input_pins(struct auto_pin_cfg *cfg) -{ - int i, j; - - for (i = 0; i < cfg->num_inputs; i++) { - for (j = i + 1; j < cfg->num_inputs; j++) { - if (cfg->inputs[i].type > cfg->inputs[j].type) { - struct auto_pin_cfg_item tmp; - tmp = cfg->inputs[i]; - cfg->inputs[i] = cfg->inputs[j]; - cfg->inputs[j] = tmp; - } - } - } -} - -/* Reorder the surround channels - * ALSA sequence is front/surr/clfe/side - * HDA sequence is: - * 4-ch: front/surr => OK as it is - * 6-ch: front/clfe/surr - * 8-ch: front/clfe/rear/side|fc - */ -static void reorder_outputs(unsigned int nums, hda_nid_t *pins) -{ - hda_nid_t nid; - - switch (nums) { - case 3: - case 4: - nid = pins[1]; - pins[1] = pins[2]; - pins[2] = nid; - break; - } -} - -/* - * Parse all pin widgets and store the useful pin nids to cfg - * - * The number of line-outs or any primary output is stored in line_outs, - * and the corresponding output pins are assigned to line_out_pins[], - * in the order of front, rear, CLFE, side, ... - * - * If more extra outputs (speaker and headphone) are found, the pins are - * assisnged to hp_pins[] and speaker_pins[], respectively. If no line-out jack - * is detected, one of speaker of HP pins is assigned as the primary - * output, i.e. to line_out_pins[0]. So, line_outs is always positive - * if any analog output exists. - * - * The analog input pins are assigned to inputs array. - * The digital input/output pins are assigned to dig_in_pin and dig_out_pin, - * respectively. - */ -int snd_hda_parse_pin_defcfg(struct hda_codec *codec, - struct auto_pin_cfg *cfg, - const hda_nid_t *ignore_nids, - unsigned int cond_flags) -{ - hda_nid_t nid, end_nid; - short seq, assoc_line_out; - short sequences_line_out[ARRAY_SIZE(cfg->line_out_pins)]; - short sequences_speaker[ARRAY_SIZE(cfg->speaker_pins)]; - short sequences_hp[ARRAY_SIZE(cfg->hp_pins)]; - int i; - - memset(cfg, 0, sizeof(*cfg)); - - memset(sequences_line_out, 0, sizeof(sequences_line_out)); - memset(sequences_speaker, 0, sizeof(sequences_speaker)); - memset(sequences_hp, 0, sizeof(sequences_hp)); - assoc_line_out = 0; - - codec->ignore_misc_bit = true; - end_nid = codec->start_nid + codec->num_nodes; - for (nid = codec->start_nid; nid < end_nid; nid++) { - unsigned int wid_caps = get_wcaps(codec, nid); - unsigned int wid_type = get_wcaps_type(wid_caps); - unsigned int def_conf; - short assoc, loc, conn, dev; - - /* read all default configuration for pin complex */ - if (wid_type != AC_WID_PIN) - continue; - /* ignore the given nids (e.g. pc-beep returns error) */ - if (ignore_nids && is_in_nid_list(nid, ignore_nids)) - continue; - - def_conf = snd_hda_codec_get_pincfg(codec, nid); - if (!(get_defcfg_misc(snd_hda_codec_get_pincfg(codec, nid)) & - AC_DEFCFG_MISC_NO_PRESENCE)) - codec->ignore_misc_bit = false; - conn = get_defcfg_connect(def_conf); - if (conn == AC_JACK_PORT_NONE) - continue; - loc = get_defcfg_location(def_conf); - dev = get_defcfg_device(def_conf); - - /* workaround for buggy BIOS setups */ - if (dev == AC_JACK_LINE_OUT) { - if (conn == AC_JACK_PORT_FIXED) - dev = AC_JACK_SPEAKER; - } - - switch (dev) { - case AC_JACK_LINE_OUT: - seq = get_defcfg_sequence(def_conf); - assoc = get_defcfg_association(def_conf); - - if (!(wid_caps & AC_WCAP_STEREO)) - if (!cfg->mono_out_pin) - cfg->mono_out_pin = nid; - if (!assoc) - continue; - if (!assoc_line_out) - assoc_line_out = assoc; - else if (assoc_line_out != assoc) - continue; - if (cfg->line_outs >= ARRAY_SIZE(cfg->line_out_pins)) - continue; - cfg->line_out_pins[cfg->line_outs] = nid; - sequences_line_out[cfg->line_outs] = seq; - cfg->line_outs++; - break; - case AC_JACK_SPEAKER: - seq = get_defcfg_sequence(def_conf); - assoc = get_defcfg_association(def_conf); - if (cfg->speaker_outs >= ARRAY_SIZE(cfg->speaker_pins)) - continue; - cfg->speaker_pins[cfg->speaker_outs] = nid; - sequences_speaker[cfg->speaker_outs] = (assoc << 4) | seq; - cfg->speaker_outs++; - break; - case AC_JACK_HP_OUT: - seq = get_defcfg_sequence(def_conf); - assoc = get_defcfg_association(def_conf); - if (cfg->hp_outs >= ARRAY_SIZE(cfg->hp_pins)) - continue; - cfg->hp_pins[cfg->hp_outs] = nid; - sequences_hp[cfg->hp_outs] = (assoc << 4) | seq; - cfg->hp_outs++; - break; - case AC_JACK_MIC_IN: - add_auto_cfg_input_pin(cfg, nid, AUTO_PIN_MIC); - break; - case AC_JACK_LINE_IN: - add_auto_cfg_input_pin(cfg, nid, AUTO_PIN_LINE_IN); - break; - case AC_JACK_CD: - add_auto_cfg_input_pin(cfg, nid, AUTO_PIN_CD); - break; - case AC_JACK_AUX: - add_auto_cfg_input_pin(cfg, nid, AUTO_PIN_AUX); - break; - case AC_JACK_SPDIF_OUT: - case AC_JACK_DIG_OTHER_OUT: - if (cfg->dig_outs >= ARRAY_SIZE(cfg->dig_out_pins)) - continue; - cfg->dig_out_pins[cfg->dig_outs] = nid; - cfg->dig_out_type[cfg->dig_outs] = - (loc == AC_JACK_LOC_HDMI) ? - HDA_PCM_TYPE_HDMI : HDA_PCM_TYPE_SPDIF; - cfg->dig_outs++; - break; - case AC_JACK_SPDIF_IN: - case AC_JACK_DIG_OTHER_IN: - cfg->dig_in_pin = nid; - if (loc == AC_JACK_LOC_HDMI) - cfg->dig_in_type = HDA_PCM_TYPE_HDMI; - else - cfg->dig_in_type = HDA_PCM_TYPE_SPDIF; - break; - } - } - - /* FIX-UP: - * If no line-out is defined but multiple HPs are found, - * some of them might be the real line-outs. - */ - if (!cfg->line_outs && cfg->hp_outs > 1 && - !(cond_flags & HDA_PINCFG_NO_HP_FIXUP)) { - int i = 0; - while (i < cfg->hp_outs) { - /* The real HPs should have the sequence 0x0f */ - if ((sequences_hp[i] & 0x0f) == 0x0f) { - i++; - continue; - } - /* Move it to the line-out table */ - cfg->line_out_pins[cfg->line_outs] = cfg->hp_pins[i]; - sequences_line_out[cfg->line_outs] = sequences_hp[i]; - cfg->line_outs++; - cfg->hp_outs--; - memmove(cfg->hp_pins + i, cfg->hp_pins + i + 1, - sizeof(cfg->hp_pins[0]) * (cfg->hp_outs - i)); - memmove(sequences_hp + i, sequences_hp + i + 1, - sizeof(sequences_hp[0]) * (cfg->hp_outs - i)); - } - memset(cfg->hp_pins + cfg->hp_outs, 0, - sizeof(hda_nid_t) * (AUTO_CFG_MAX_OUTS - cfg->hp_outs)); - if (!cfg->hp_outs) - cfg->line_out_type = AUTO_PIN_HP_OUT; - - } - - /* sort by sequence */ - sort_pins_by_sequence(cfg->line_out_pins, sequences_line_out, - cfg->line_outs); - sort_pins_by_sequence(cfg->speaker_pins, sequences_speaker, - cfg->speaker_outs); - sort_pins_by_sequence(cfg->hp_pins, sequences_hp, - cfg->hp_outs); - - /* - * FIX-UP: if no line-outs are detected, try to use speaker or HP pin - * as a primary output - */ - if (!cfg->line_outs && - !(cond_flags & HDA_PINCFG_NO_LO_FIXUP)) { - if (cfg->speaker_outs) { - cfg->line_outs = cfg->speaker_outs; - memcpy(cfg->line_out_pins, cfg->speaker_pins, - sizeof(cfg->speaker_pins)); - cfg->speaker_outs = 0; - memset(cfg->speaker_pins, 0, sizeof(cfg->speaker_pins)); - cfg->line_out_type = AUTO_PIN_SPEAKER_OUT; - } else if (cfg->hp_outs) { - cfg->line_outs = cfg->hp_outs; - memcpy(cfg->line_out_pins, cfg->hp_pins, - sizeof(cfg->hp_pins)); - cfg->hp_outs = 0; - memset(cfg->hp_pins, 0, sizeof(cfg->hp_pins)); - cfg->line_out_type = AUTO_PIN_HP_OUT; - } - } - - reorder_outputs(cfg->line_outs, cfg->line_out_pins); - reorder_outputs(cfg->hp_outs, cfg->hp_pins); - reorder_outputs(cfg->speaker_outs, cfg->speaker_pins); - - sort_autocfg_input_pins(cfg); - - /* - * debug prints of the parsed results - */ - snd_printd("autoconfig: line_outs=%d (0x%x/0x%x/0x%x/0x%x/0x%x) type:%s\n", - cfg->line_outs, cfg->line_out_pins[0], cfg->line_out_pins[1], - cfg->line_out_pins[2], cfg->line_out_pins[3], - cfg->line_out_pins[4], - cfg->line_out_type == AUTO_PIN_HP_OUT ? "hp" : - (cfg->line_out_type == AUTO_PIN_SPEAKER_OUT ? - "speaker" : "line")); - snd_printd(" speaker_outs=%d (0x%x/0x%x/0x%x/0x%x/0x%x)\n", - cfg->speaker_outs, cfg->speaker_pins[0], - cfg->speaker_pins[1], cfg->speaker_pins[2], - cfg->speaker_pins[3], cfg->speaker_pins[4]); - snd_printd(" hp_outs=%d (0x%x/0x%x/0x%x/0x%x/0x%x)\n", - cfg->hp_outs, cfg->hp_pins[0], - cfg->hp_pins[1], cfg->hp_pins[2], - cfg->hp_pins[3], cfg->hp_pins[4]); - snd_printd(" mono: mono_out=0x%x\n", cfg->mono_out_pin); - if (cfg->dig_outs) - snd_printd(" dig-out=0x%x/0x%x\n", - cfg->dig_out_pins[0], cfg->dig_out_pins[1]); - snd_printd(" inputs:"); - for (i = 0; i < cfg->num_inputs; i++) { - snd_printd(" %s=0x%x", - hda_get_autocfg_input_label(codec, cfg, i), - cfg->inputs[i].pin); - } - snd_printd("\n"); - if (cfg->dig_in_pin) - snd_printd(" dig-in=0x%x\n", cfg->dig_in_pin); - - return 0; -} -EXPORT_SYMBOL_HDA(snd_hda_parse_pin_defcfg); - -int snd_hda_get_input_pin_attr(unsigned int def_conf) -{ - unsigned int loc = get_defcfg_location(def_conf); - unsigned int conn = get_defcfg_connect(def_conf); - if (conn == AC_JACK_PORT_NONE) - return INPUT_PIN_ATTR_UNUSED; - /* Windows may claim the internal mic to be BOTH, too */ - if (conn == AC_JACK_PORT_FIXED || conn == AC_JACK_PORT_BOTH) - return INPUT_PIN_ATTR_INT; - if ((loc & 0x30) == AC_JACK_LOC_INTERNAL) - return INPUT_PIN_ATTR_INT; - if ((loc & 0x30) == AC_JACK_LOC_SEPARATE) - return INPUT_PIN_ATTR_DOCK; - if (loc == AC_JACK_LOC_REAR) - return INPUT_PIN_ATTR_REAR; - if (loc == AC_JACK_LOC_FRONT) - return INPUT_PIN_ATTR_FRONT; - return INPUT_PIN_ATTR_NORMAL; -} -EXPORT_SYMBOL_HDA(snd_hda_get_input_pin_attr); - -/** - * hda_get_input_pin_label - Give a label for the given input pin - * - * When check_location is true, the function checks the pin location - * for mic and line-in pins, and set an appropriate prefix like "Front", - * "Rear", "Internal". - */ - -static const char *hda_get_input_pin_label(struct hda_codec *codec, - hda_nid_t pin, bool check_location) -{ - unsigned int def_conf; - static const char * const mic_names[] = { - "Internal Mic", "Dock Mic", "Mic", "Front Mic", "Rear Mic", - }; - int attr; - - def_conf = snd_hda_codec_get_pincfg(codec, pin); - - switch (get_defcfg_device(def_conf)) { - case AC_JACK_MIC_IN: - if (!check_location) - return "Mic"; - attr = snd_hda_get_input_pin_attr(def_conf); - if (!attr) - return "None"; - return mic_names[attr - 1]; - case AC_JACK_LINE_IN: - if (!check_location) - return "Line"; - attr = snd_hda_get_input_pin_attr(def_conf); - if (!attr) - return "None"; - if (attr == INPUT_PIN_ATTR_DOCK) - return "Dock Line"; - return "Line"; - case AC_JACK_AUX: - return "Aux"; - case AC_JACK_CD: - return "CD"; - case AC_JACK_SPDIF_IN: - return "SPDIF In"; - case AC_JACK_DIG_OTHER_IN: - return "Digital In"; - default: - return "Misc"; - } -} - -/* Check whether the location prefix needs to be added to the label. - * If all mic-jacks are in the same location (e.g. rear panel), we don't - * have to put "Front" prefix to each label. In such a case, returns false. - */ -static int check_mic_location_need(struct hda_codec *codec, - const struct auto_pin_cfg *cfg, - int input) -{ - unsigned int defc; - int i, attr, attr2; - - defc = snd_hda_codec_get_pincfg(codec, cfg->inputs[input].pin); - attr = snd_hda_get_input_pin_attr(defc); - /* for internal or docking mics, we need locations */ - if (attr <= INPUT_PIN_ATTR_NORMAL) - return 1; - - attr = 0; - for (i = 0; i < cfg->num_inputs; i++) { - defc = snd_hda_codec_get_pincfg(codec, cfg->inputs[i].pin); - attr2 = snd_hda_get_input_pin_attr(defc); - if (attr2 >= INPUT_PIN_ATTR_NORMAL) { - if (attr && attr != attr2) - return 1; /* different locations found */ - attr = attr2; - } - } - return 0; -} - /** - * hda_get_autocfg_input_label - Get a label for the given input + * snd_hda_get_default_vref - Get the default (mic) VREF pin bits * - * Get a label for the given input pin defined by the autocfg item. - * Unlike hda_get_input_pin_label(), this function checks all inputs - * defined in autocfg and avoids the redundant mic/line prefix as much as - * possible. - */ -const char *hda_get_autocfg_input_label(struct hda_codec *codec, - const struct auto_pin_cfg *cfg, - int input) -{ - int type = cfg->inputs[input].type; - int has_multiple_pins = 0; - - if ((input > 0 && cfg->inputs[input - 1].type == type) || - (input < cfg->num_inputs - 1 && cfg->inputs[input + 1].type == type)) - has_multiple_pins = 1; - if (has_multiple_pins && type == AUTO_PIN_MIC) - has_multiple_pins &= check_mic_location_need(codec, cfg, input); - return hda_get_input_pin_label(codec, cfg->inputs[input].pin, - has_multiple_pins); -} -EXPORT_SYMBOL_HDA(hda_get_autocfg_input_label); - -/* return the position of NID in the list, or -1 if not found */ -static int find_idx_in_nid_list(hda_nid_t nid, const hda_nid_t *list, int nums) -{ - int i; - for (i = 0; i < nums; i++) - if (list[i] == nid) - return i; - return -1; -} - -/* get a unique suffix or an index number */ -static const char *check_output_sfx(hda_nid_t nid, const hda_nid_t *pins, - int num_pins, int *indexp) -{ - static const char * const channel_sfx[] = { - " Front", " Surround", " CLFE", " Side" - }; - int i; - - i = find_idx_in_nid_list(nid, pins, num_pins); - if (i < 0) - return NULL; - if (num_pins == 1) - return ""; - if (num_pins > ARRAY_SIZE(channel_sfx)) { - if (indexp) - *indexp = i; - return ""; - } - return channel_sfx[i]; -} - -static int fill_audio_out_name(struct hda_codec *codec, hda_nid_t nid, - const struct auto_pin_cfg *cfg, - const char *name, char *label, int maxlen, - int *indexp) -{ - unsigned int def_conf = snd_hda_codec_get_pincfg(codec, nid); - int attr = snd_hda_get_input_pin_attr(def_conf); - const char *pfx = "", *sfx = ""; - - /* handle as a speaker if it's a fixed line-out */ - if (!strcmp(name, "Line Out") && attr == INPUT_PIN_ATTR_INT) - name = "Speaker"; - /* check the location */ - switch (attr) { - case INPUT_PIN_ATTR_DOCK: - pfx = "Dock "; - break; - case INPUT_PIN_ATTR_FRONT: - pfx = "Front "; - break; - } - if (cfg) { - /* try to give a unique suffix if needed */ - sfx = check_output_sfx(nid, cfg->line_out_pins, cfg->line_outs, - indexp); - if (!sfx) - sfx = check_output_sfx(nid, cfg->speaker_pins, cfg->speaker_outs, - indexp); - if (!sfx) { - /* don't add channel suffix for Headphone controls */ - int idx = find_idx_in_nid_list(nid, cfg->hp_pins, - cfg->hp_outs); - if (idx >= 0) - *indexp = idx; - sfx = ""; + * Guess the suitable VREF pin bits to be set as the pin-control value. + * Note: the function doesn't set the AC_PINCTL_IN_EN bit. + */ +unsigned int snd_hda_get_default_vref(struct hda_codec *codec, hda_nid_t pin) +{ + unsigned int pincap; + unsigned int oldval; + oldval = snd_hda_codec_read(codec, pin, 0, + AC_VERB_GET_PIN_WIDGET_CONTROL, 0); + pincap = snd_hda_query_pin_caps(codec, pin); + pincap = (pincap & AC_PINCAP_VREF) >> AC_PINCAP_VREF_SHIFT; + /* Exception: if the default pin setup is vref50, we give it priority */ + if ((pincap & AC_PINCAP_VREF_80) && oldval != PIN_VREF50) + return AC_PINCTL_VREF_80; + else if (pincap & AC_PINCAP_VREF_50) + return AC_PINCTL_VREF_50; + else if (pincap & AC_PINCAP_VREF_100) + return AC_PINCTL_VREF_100; + else if (pincap & AC_PINCAP_VREF_GRD) + return AC_PINCTL_VREF_GRD; + return AC_PINCTL_VREF_HIZ; +} +EXPORT_SYMBOL_HDA(snd_hda_get_default_vref); + +int _snd_hda_set_pin_ctl(struct hda_codec *codec, hda_nid_t pin, + unsigned int val, bool cached) +{ + if (val) { + unsigned int cap = snd_hda_query_pin_caps(codec, pin); + if (cap && (val & AC_PINCTL_OUT_EN)) { + if (!(cap & AC_PINCAP_OUT)) + val &= ~(AC_PINCTL_OUT_EN | AC_PINCTL_HP_EN); + else if ((val & AC_PINCTL_HP_EN) && + !(cap & AC_PINCAP_HP_DRV)) + val &= ~AC_PINCTL_HP_EN; } - } - snprintf(label, maxlen, "%s%s%s", pfx, name, sfx); - return 1; -} - -/** - * snd_hda_get_pin_label - Get a label for the given I/O pin - * - * Get a label for the given pin. This function works for both input and - * output pins. When @cfg is given as non-NULL, the function tries to get - * an optimized label using hda_get_autocfg_input_label(). - * - * This function tries to give a unique label string for the pin as much as - * possible. For example, when the multiple line-outs are present, it adds - * the channel suffix like "Front", "Surround", etc (only when @cfg is given). - * If no unique name with a suffix is available and @indexp is non-NULL, the - * index number is stored in the pointer. - */ -int snd_hda_get_pin_label(struct hda_codec *codec, hda_nid_t nid, - const struct auto_pin_cfg *cfg, - char *label, int maxlen, int *indexp) -{ - unsigned int def_conf = snd_hda_codec_get_pincfg(codec, nid); - const char *name = NULL; - int i; - - if (indexp) - *indexp = 0; - if (get_defcfg_connect(def_conf) == AC_JACK_PORT_NONE) - return 0; - - switch (get_defcfg_device(def_conf)) { - case AC_JACK_LINE_OUT: - return fill_audio_out_name(codec, nid, cfg, "Line Out", - label, maxlen, indexp); - case AC_JACK_SPEAKER: - return fill_audio_out_name(codec, nid, cfg, "Speaker", - label, maxlen, indexp); - case AC_JACK_HP_OUT: - return fill_audio_out_name(codec, nid, cfg, "Headphone", - label, maxlen, indexp); - case AC_JACK_SPDIF_OUT: - case AC_JACK_DIG_OTHER_OUT: - if (get_defcfg_location(def_conf) == AC_JACK_LOC_HDMI) - name = "HDMI"; - else - name = "SPDIF"; - if (cfg && indexp) { - i = find_idx_in_nid_list(nid, cfg->dig_out_pins, - cfg->dig_outs); - if (i >= 0) - *indexp = i; - } - break; - default: - if (cfg) { - for (i = 0; i < cfg->num_inputs; i++) { - if (cfg->inputs[i].pin != nid) - continue; - name = hda_get_autocfg_input_label(codec, cfg, i); - if (name) - break; - } + if (cap && (val & AC_PINCTL_IN_EN)) { + if (!(cap & AC_PINCAP_IN)) + val &= ~(AC_PINCTL_IN_EN | AC_PINCTL_VREFEN); } - if (!name) - name = hda_get_input_pin_label(codec, nid, true); - break; } - if (!name) - return 0; - strlcpy(label, name, maxlen); - return 1; + if (cached) + return snd_hda_codec_update_cache(codec, pin, 0, + AC_VERB_SET_PIN_WIDGET_CONTROL, val); + else + return snd_hda_codec_write(codec, pin, 0, + AC_VERB_SET_PIN_WIDGET_CONTROL, val); } -EXPORT_SYMBOL_HDA(snd_hda_get_pin_label); +EXPORT_SYMBOL_HDA(_snd_hda_set_pin_ctl); /** * snd_hda_add_imux_item - Add an item to input_mux @@ -5444,8 +4986,6 @@ int snd_hda_suspend(struct hda_bus *bus) list_for_each_entry(codec, &bus->codec_list, list) { if (hda_codec_is_power_on(codec)) hda_call_codec_suspend(codec); - if (codec->patch_ops.post_suspend) - codec->patch_ops.post_suspend(codec); } return 0; } @@ -5465,10 +5005,7 @@ int snd_hda_resume(struct hda_bus *bus) struct hda_codec *codec; list_for_each_entry(codec, &bus->codec_list, list) { - if (codec->patch_ops.pre_resume) - codec->patch_ops.pre_resume(codec); - if (snd_hda_codec_needs_resume(codec)) - hda_call_codec_resume(codec); + hda_call_codec_resume(codec); } return 0; } diff --git a/sound/pci/hda/hda_codec.h b/sound/pci/hda/hda_codec.h index 56b4f74c0b1..54b52819fb4 100644 --- a/sound/pci/hda/hda_codec.h +++ b/sound/pci/hda/hda_codec.h @@ -704,8 +704,6 @@ struct hda_codec_ops { unsigned int power_state); #ifdef CONFIG_PM int (*suspend)(struct hda_codec *codec, pm_message_t state); - int (*post_suspend)(struct hda_codec *codec); - int (*pre_resume)(struct hda_codec *codec); int (*resume)(struct hda_codec *codec); #endif #ifdef CONFIG_SND_HDA_POWER_SAVE @@ -829,6 +827,7 @@ struct hda_codec { struct mutex spdif_mutex; struct mutex control_mutex; + struct mutex hash_mutex; struct snd_array spdif_out; unsigned int spdif_in_enable; /* SPDIF input enable? */ const hda_nid_t *slave_dig_outs; /* optional digital out slave widgets */ @@ -861,12 +860,13 @@ struct hda_codec { unsigned int no_jack_detect:1; /* Machine has no jack-detection */ #ifdef CONFIG_SND_HDA_POWER_SAVE unsigned int power_on :1; /* current (global) power-state */ - unsigned int power_transition :1; /* power-state in transition */ + int power_transition; /* power-state in transition */ int power_count; /* current (global) power refcount */ struct delayed_work power_work; /* delayed task for powerdown */ unsigned long power_on_acct; unsigned long power_off_acct; unsigned long power_jiffies; + spinlock_t power_lock; #endif /* codec-specific additional proc output */ @@ -911,10 +911,13 @@ int snd_hda_get_sub_nodes(struct hda_codec *codec, hda_nid_t nid, hda_nid_t *start_id); int snd_hda_get_connections(struct hda_codec *codec, hda_nid_t nid, hda_nid_t *conn_list, int max_conns); +static inline int +snd_hda_get_num_conns(struct hda_codec *codec, hda_nid_t nid) +{ + return snd_hda_get_connections(codec, nid, NULL, 0); +} int snd_hda_get_raw_connections(struct hda_codec *codec, hda_nid_t nid, hda_nid_t *conn_list, int max_conns); -int snd_hda_get_conn_list(struct hda_codec *codec, hda_nid_t nid, - const hda_nid_t **listp); int snd_hda_override_conn_list(struct hda_codec *codec, hda_nid_t nid, int nums, const hda_nid_t *list); int snd_hda_get_conn_index(struct hda_codec *codec, hda_nid_t mux, @@ -1051,12 +1054,10 @@ const char *snd_hda_get_jack_location(u32 cfg); #ifdef CONFIG_SND_HDA_POWER_SAVE void snd_hda_power_up(struct hda_codec *codec); void snd_hda_power_down(struct hda_codec *codec); -#define snd_hda_codec_needs_resume(codec) codec->power_count void snd_hda_update_power_acct(struct hda_codec *codec); #else static inline void snd_hda_power_up(struct hda_codec *codec) {} static inline void snd_hda_power_down(struct hda_codec *codec) {} -#define snd_hda_codec_needs_resume(codec) 1 #endif #ifdef CONFIG_SND_HDA_PATCH_LOADER diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c index 1f350522bed..4ab8102f87e 100644 --- a/sound/pci/hda/hda_intel.c +++ b/sound/pci/hda/hda_intel.c @@ -497,6 +497,7 @@ enum { AZX_DRIVER_NVIDIA, AZX_DRIVER_TERA, AZX_DRIVER_CTX, + AZX_DRIVER_CTHDA, AZX_DRIVER_GENERIC, AZX_NUM_DRIVERS, /* keep this as last entry */ }; @@ -518,6 +519,7 @@ enum { #define AZX_DCAPS_OLD_SSYNC (1 << 20) /* Old SSYNC reg for ICH */ #define AZX_DCAPS_BUFSIZE (1 << 21) /* no buffer size alignment */ #define AZX_DCAPS_ALIGN_BUFSIZE (1 << 22) /* buffer size alignment */ +#define AZX_DCAPS_4K_BDLE_BOUNDARY (1 << 23) /* BDLE in 4k boundary */ /* quirks for ATI SB / AMD Hudson */ #define AZX_DCAPS_PRESET_ATI_SB \ @@ -533,6 +535,9 @@ enum { (AZX_DCAPS_NVIDIA_SNOOP | AZX_DCAPS_RIRB_DELAY | AZX_DCAPS_NO_MSI |\ AZX_DCAPS_ALIGN_BUFSIZE) +#define AZX_DCAPS_PRESET_CTHDA \ + (AZX_DCAPS_NO_MSI | AZX_DCAPS_POSFIX_LPIB | AZX_DCAPS_4K_BDLE_BOUNDARY) + static char *driver_short_names[] __devinitdata = { [AZX_DRIVER_ICH] = "HDA Intel", [AZX_DRIVER_PCH] = "HDA Intel PCH", @@ -546,6 +551,7 @@ static char *driver_short_names[] __devinitdata = { [AZX_DRIVER_NVIDIA] = "HDA NVidia", [AZX_DRIVER_TERA] = "HDA Teradici", [AZX_DRIVER_CTX] = "HDA Creative", + [AZX_DRIVER_CTHDA] = "HDA Creative", [AZX_DRIVER_GENERIC] = "HD-Audio Generic", }; @@ -1285,7 +1291,8 @@ static irqreturn_t azx_interrupt(int irq, void *dev_id) /* * set up a BDL entry */ -static int setup_bdle(struct snd_pcm_substream *substream, +static int setup_bdle(struct azx *chip, + struct snd_pcm_substream *substream, struct azx_dev *azx_dev, u32 **bdlp, int ofs, int size, int with_ioc) { @@ -1304,6 +1311,12 @@ static int setup_bdle(struct snd_pcm_substream *substream, bdl[1] = cpu_to_le32(upper_32_bits(addr)); /* program the size field of the BDL entry */ chunk = snd_pcm_sgbuf_get_chunk_size(substream, ofs, size); + /* one BDLE cannot cross 4K boundary on CTHDA chips */ + if (chip->driver_caps & AZX_DCAPS_4K_BDLE_BOUNDARY) { + u32 remain = 0x1000 - (ofs & 0xfff); + if (chunk > remain) + chunk = remain; + } bdl[2] = cpu_to_le32(chunk); /* program the IOC to enable interrupt * only when the whole fragment is processed @@ -1356,7 +1369,7 @@ static int azx_setup_periods(struct azx *chip, bdl_pos_adj[chip->dev_index]); pos_adj = 0; } else { - ofs = setup_bdle(substream, azx_dev, + ofs = setup_bdle(chip, substream, azx_dev, &bdl, ofs, pos_adj, !substream->runtime->no_period_wakeup); if (ofs < 0) @@ -1366,10 +1379,10 @@ static int azx_setup_periods(struct azx *chip, pos_adj = 0; for (i = 0; i < periods; i++) { if (i == periods - 1 && pos_adj) - ofs = setup_bdle(substream, azx_dev, &bdl, ofs, + ofs = setup_bdle(chip, substream, azx_dev, &bdl, ofs, period_bytes - pos_adj, 0); else - ofs = setup_bdle(substream, azx_dev, &bdl, ofs, + ofs = setup_bdle(chip, substream, azx_dev, &bdl, ofs, period_bytes, !substream->runtime->no_period_wakeup); if (ofs < 0) @@ -2353,17 +2366,6 @@ static void azx_power_notify(struct hda_bus *bus) * power management */ -static int snd_hda_codecs_inuse(struct hda_bus *bus) -{ - struct hda_codec *codec; - - list_for_each_entry(codec, &bus->codec_list, list) { - if (snd_hda_codec_needs_resume(codec)) - return 1; - } - return 0; -} - static int azx_suspend(struct pci_dev *pci, pm_message_t state) { struct snd_card *card = pci_get_drvdata(pci); @@ -2410,8 +2412,7 @@ static int azx_resume(struct pci_dev *pci) return -EIO; azx_init_pci(chip); - if (snd_hda_codecs_inuse(chip->bus)) - azx_init_chip(chip, 1); + azx_init_chip(chip, 1); snd_hda_resume(chip->bus); snd_power_change_state(card, SNDRV_CTL_POWER_D0); @@ -2565,6 +2566,8 @@ static struct snd_pci_quirk probe_mask_list[] __devinitdata = { /* forced codec slots */ SND_PCI_QUIRK(0x1043, 0x1262, "ASUS W5Fm", 0x103), SND_PCI_QUIRK(0x1046, 0x1262, "ASUS W5F", 0x103), + /* WinFast VP200 H (Teradici) user reported broken communication */ + SND_PCI_QUIRK(0x3a21, 0x040d, "WinFast VP200 H", 0x101), {} }; @@ -3130,6 +3133,11 @@ static DEFINE_PCI_DEVICE_TABLE(azx_ids) = { .driver_data = AZX_DRIVER_CTX | AZX_DCAPS_CTX_WORKAROUND | AZX_DCAPS_RIRB_PRE_DELAY | AZX_DCAPS_POSFIX_LPIB }, #endif + /* CTHDA chips */ + { PCI_DEVICE(0x1102, 0x0010), + .driver_data = AZX_DRIVER_CTHDA | AZX_DCAPS_PRESET_CTHDA }, + { PCI_DEVICE(0x1102, 0x0012), + .driver_data = AZX_DRIVER_CTHDA | AZX_DCAPS_PRESET_CTHDA }, /* Vortex86MX */ { PCI_DEVICE(0x17f3, 0x3010), .driver_data = AZX_DRIVER_GENERIC }, /* VMware HDAudio */ @@ -3148,7 +3156,7 @@ static DEFINE_PCI_DEVICE_TABLE(azx_ids) = { MODULE_DEVICE_TABLE(pci, azx_ids); /* pci_driver definition */ -static struct pci_driver driver = { +static struct pci_driver azx_driver = { .name = KBUILD_MODNAME, .id_table = azx_ids, .probe = azx_probe, @@ -3159,15 +3167,4 @@ static struct pci_driver driver = { #endif }; -static int __init alsa_card_azx_init(void) -{ - return pci_register_driver(&driver); -} - -static void __exit alsa_card_azx_exit(void) -{ - pci_unregister_driver(&driver); -} - -module_init(alsa_card_azx_init) -module_exit(alsa_card_azx_exit) +module_pci_driver(azx_driver); diff --git a/sound/pci/hda/hda_jack.c b/sound/pci/hda/hda_jack.c index d68948499fb..2dd1c113a4c 100644 --- a/sound/pci/hda/hda_jack.c +++ b/sound/pci/hda/hda_jack.c @@ -17,6 +17,7 @@ #include <sound/jack.h> #include "hda_codec.h" #include "hda_local.h" +#include "hda_auto_parser.h" #include "hda_jack.h" bool is_jack_detectable(struct hda_codec *codec, hda_nid_t nid) diff --git a/sound/pci/hda/hda_jack.h b/sound/pci/hda/hda_jack.h index c66655cf413..8ae52465ec5 100644 --- a/sound/pci/hda/hda_jack.h +++ b/sound/pci/hda/hda_jack.h @@ -12,6 +12,8 @@ #ifndef __SOUND_HDA_JACK_H #define __SOUND_HDA_JACK_H +struct auto_pin_cfg; + struct hda_jack_tbl { hda_nid_t nid; unsigned char action; /* event action (0 = none) */ diff --git a/sound/pci/hda/hda_local.h b/sound/pci/hda/hda_local.h index 0ec9248165b..9a096a8e0fc 100644 --- a/sound/pci/hda/hda_local.h +++ b/sound/pci/hda/hda_local.h @@ -262,6 +262,8 @@ int snd_hda_input_mux_put(struct hda_codec *codec, const struct hda_input_mux *imux, struct snd_ctl_elem_value *ucontrol, hda_nid_t nid, unsigned int *cur_val); +int snd_hda_add_imux_item(struct hda_input_mux *imux, const char *label, + int index, int *type_index_ret); /* * Channel mode helper @@ -393,72 +395,7 @@ struct hda_bus_unsolicited { struct hda_bus *bus; }; -/* - * Helper for automatic pin configuration - */ - -enum { - AUTO_PIN_MIC, - AUTO_PIN_LINE_IN, - AUTO_PIN_CD, - AUTO_PIN_AUX, - AUTO_PIN_LAST -}; - -enum { - AUTO_PIN_LINE_OUT, - AUTO_PIN_SPEAKER_OUT, - AUTO_PIN_HP_OUT -}; - -#define AUTO_CFG_MAX_OUTS HDA_MAX_OUTS -#define AUTO_CFG_MAX_INS 8 - -struct auto_pin_cfg_item { - hda_nid_t pin; - int type; -}; - -struct auto_pin_cfg; -const char *hda_get_autocfg_input_label(struct hda_codec *codec, - const struct auto_pin_cfg *cfg, - int input); -int snd_hda_get_pin_label(struct hda_codec *codec, hda_nid_t nid, - const struct auto_pin_cfg *cfg, - char *label, int maxlen, int *indexp); -int snd_hda_add_imux_item(struct hda_input_mux *imux, const char *label, - int index, int *type_index_ret); - -enum { - INPUT_PIN_ATTR_UNUSED, /* pin not connected */ - INPUT_PIN_ATTR_INT, /* internal mic/line-in */ - INPUT_PIN_ATTR_DOCK, /* docking mic/line-in */ - INPUT_PIN_ATTR_NORMAL, /* mic/line-in jack */ - INPUT_PIN_ATTR_FRONT, /* mic/line-in jack in front */ - INPUT_PIN_ATTR_REAR, /* mic/line-in jack in rear */ -}; - -int snd_hda_get_input_pin_attr(unsigned int def_conf); - -struct auto_pin_cfg { - int line_outs; - /* sorted in the order of Front/Surr/CLFE/Side */ - hda_nid_t line_out_pins[AUTO_CFG_MAX_OUTS]; - int speaker_outs; - hda_nid_t speaker_pins[AUTO_CFG_MAX_OUTS]; - int hp_outs; - int line_out_type; /* AUTO_PIN_XXX_OUT */ - hda_nid_t hp_pins[AUTO_CFG_MAX_OUTS]; - int num_inputs; - struct auto_pin_cfg_item inputs[AUTO_CFG_MAX_INS]; - int dig_outs; - hda_nid_t dig_out_pins[2]; - hda_nid_t dig_in_pin; - hda_nid_t mono_out_pin; - int dig_out_type[2]; /* HDA_PCM_TYPE_XXX */ - int dig_in_type; /* HDA_PCM_TYPE_XXX */ -}; - +/* helper macros to retrieve pin default-config values */ #define get_defcfg_connect(cfg) \ ((cfg & AC_DEFCFG_PORT_CONN) >> AC_DEFCFG_PORT_CONN_SHIFT) #define get_defcfg_association(cfg) \ @@ -472,19 +409,6 @@ struct auto_pin_cfg { #define get_defcfg_misc(cfg) \ ((cfg & AC_DEFCFG_MISC) >> AC_DEFCFG_MISC_SHIFT) -/* bit-flags for snd_hda_parse_pin_def_config() behavior */ -#define HDA_PINCFG_NO_HP_FIXUP (1 << 0) /* no HP-split */ -#define HDA_PINCFG_NO_LO_FIXUP (1 << 1) /* don't take other outs as LO */ - -int snd_hda_parse_pin_defcfg(struct hda_codec *codec, - struct auto_pin_cfg *cfg, - const hda_nid_t *ignore_nids, - unsigned int cond_flags); - -/* older function */ -#define snd_hda_parse_pin_def_config(codec, cfg, ignore) \ - snd_hda_parse_pin_defcfg(codec, cfg, ignore, 0) - /* amp values */ #define AMP_IN_MUTE(idx) (0x7080 | ((idx)<<8)) #define AMP_IN_UNMUTE(idx) (0x7000 | ((idx)<<8)) @@ -502,6 +426,46 @@ int snd_hda_parse_pin_defcfg(struct hda_codec *codec, #define PIN_HP (AC_PINCTL_OUT_EN | AC_PINCTL_HP_EN) #define PIN_HP_AMP (AC_PINCTL_HP_EN) +unsigned int snd_hda_get_default_vref(struct hda_codec *codec, hda_nid_t pin); +int _snd_hda_set_pin_ctl(struct hda_codec *codec, hda_nid_t pin, + unsigned int val, bool cached); + +/** + * _snd_hda_set_pin_ctl - Set a pin-control value safely + * @codec: the codec instance + * @pin: the pin NID to set the control + * @val: the pin-control value (AC_PINCTL_* bits) + * + * This function sets the pin-control value to the given pin, but + * filters out the invalid pin-control bits when the pin has no such + * capabilities. For example, when PIN_HP is passed but the pin has no + * HP-drive capability, the HP bit is omitted. + * + * The function doesn't check the input VREF capability bits, though. + * Use snd_hda_get_default_vref() to guess the right value. + * Also, this function is only for analog pins, not for HDMI pins. + */ +static inline int +snd_hda_set_pin_ctl(struct hda_codec *codec, hda_nid_t pin, unsigned int val) +{ + return _snd_hda_set_pin_ctl(codec, pin, val, false); +} + +/** + * snd_hda_set_pin_ctl_cache - Set a pin-control value safely + * @codec: the codec instance + * @pin: the pin NID to set the control + * @val: the pin-control value (AC_PINCTL_* bits) + * + * Just like snd_hda_set_pin_ctl() but write to cache as well. + */ +static inline int +snd_hda_set_pin_ctl_cache(struct hda_codec *codec, hda_nid_t pin, + unsigned int val) +{ + return _snd_hda_set_pin_ctl(codec, pin, val, true); +} + /* * get widget capabilities */ diff --git a/sound/pci/hda/patch_analog.c b/sound/pci/hda/patch_analog.c index 7143393927d..d8b2d6dee98 100644 --- a/sound/pci/hda/patch_analog.c +++ b/sound/pci/hda/patch_analog.c @@ -28,6 +28,7 @@ #include <sound/core.h> #include "hda_codec.h" #include "hda_local.h" +#include "hda_auto_parser.h" #include "hda_beep.h" #include "hda_jack.h" @@ -1742,9 +1743,7 @@ static int ad1981_hp_master_sw_put(struct snd_kcontrol *kcontrol, if (! ad198x_eapd_put(kcontrol, ucontrol)) return 0; /* change speaker pin appropriately */ - snd_hda_codec_write(codec, 0x05, 0, - AC_VERB_SET_PIN_WIDGET_CONTROL, - spec->cur_eapd ? PIN_OUT : 0); + snd_hda_set_pin_ctl(codec, 0x05, spec->cur_eapd ? PIN_OUT : 0); /* toggle HP mute appropriately */ snd_hda_codec_amp_stereo(codec, 0x06, HDA_OUTPUT, 0, HDA_AMP_MUTE, @@ -3103,7 +3102,7 @@ static void ad1988_auto_set_output_and_unmute(struct hda_codec *codec, int dac_idx) { /* set as output */ - snd_hda_codec_write(codec, nid, 0, AC_VERB_SET_PIN_WIDGET_CONTROL, pin_type); + snd_hda_set_pin_ctl(codec, nid, pin_type); snd_hda_codec_write(codec, nid, 0, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE); switch (nid) { case 0x11: /* port-A - DAC 03 */ @@ -3157,6 +3156,7 @@ static void ad1988_auto_init_analog_input(struct hda_codec *codec) for (i = 0; i < cfg->num_inputs; i++) { hda_nid_t nid = cfg->inputs[i].pin; int type = cfg->inputs[i].type; + int val; switch (nid) { case 0x15: /* port-C */ snd_hda_codec_write(codec, 0x33, 0, AC_VERB_SET_CONNECT_SEL, 0x0); @@ -3165,8 +3165,10 @@ static void ad1988_auto_init_analog_input(struct hda_codec *codec) snd_hda_codec_write(codec, 0x34, 0, AC_VERB_SET_CONNECT_SEL, 0x0); break; } - snd_hda_codec_write(codec, nid, 0, AC_VERB_SET_PIN_WIDGET_CONTROL, - type == AUTO_PIN_MIC ? PIN_VREF80 : PIN_IN); + val = PIN_IN; + if (type == AUTO_PIN_MIC) + val |= snd_hda_get_default_vref(codec, nid); + snd_hda_set_pin_ctl(codec, nid, val); if (nid != AD1988_PIN_CD_NID) snd_hda_codec_write(codec, nid, 0, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE); diff --git a/sound/pci/hda/patch_ca0110.c b/sound/pci/hda/patch_ca0110.c index 09ccfabb4a1..19ae14f739c 100644 --- a/sound/pci/hda/patch_ca0110.c +++ b/sound/pci/hda/patch_ca0110.c @@ -26,6 +26,7 @@ #include <sound/core.h> #include "hda_codec.h" #include "hda_local.h" +#include "hda_auto_parser.h" /* */ @@ -341,8 +342,7 @@ static int ca0110_build_pcms(struct hda_codec *codec) static void init_output(struct hda_codec *codec, hda_nid_t pin, hda_nid_t dac) { if (pin) { - snd_hda_codec_write(codec, pin, 0, - AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP); + snd_hda_set_pin_ctl(codec, pin, PIN_HP); if (get_wcaps(codec, pin) & AC_WCAP_OUT_AMP) snd_hda_codec_write(codec, pin, 0, AC_VERB_SET_AMP_GAIN_MUTE, @@ -356,8 +356,8 @@ static void init_output(struct hda_codec *codec, hda_nid_t pin, hda_nid_t dac) static void init_input(struct hda_codec *codec, hda_nid_t pin, hda_nid_t adc) { if (pin) { - snd_hda_codec_write(codec, pin, 0, - AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80); + snd_hda_set_pin_ctl(codec, pin, PIN_IN | + snd_hda_get_default_vref(codec, pin)); if (get_wcaps(codec, pin) & AC_WCAP_IN_AMP) snd_hda_codec_write(codec, pin, 0, AC_VERB_SET_AMP_GAIN_MUTE, diff --git a/sound/pci/hda/patch_ca0132.c b/sound/pci/hda/patch_ca0132.c index 21d91d580da..d0d3540e39e 100644 --- a/sound/pci/hda/patch_ca0132.c +++ b/sound/pci/hda/patch_ca0132.c @@ -30,6 +30,7 @@ #include <sound/core.h> #include "hda_codec.h" #include "hda_local.h" +#include "hda_auto_parser.h" #define WIDGET_CHIP_CTRL 0x15 #define WIDGET_DSP_CTRL 0x16 @@ -239,8 +240,7 @@ enum get_set { static void init_output(struct hda_codec *codec, hda_nid_t pin, hda_nid_t dac) { if (pin) { - snd_hda_codec_write(codec, pin, 0, - AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP); + snd_hda_set_pin_ctl(codec, pin, PIN_HP); if (get_wcaps(codec, pin) & AC_WCAP_OUT_AMP) snd_hda_codec_write(codec, pin, 0, AC_VERB_SET_AMP_GAIN_MUTE, @@ -254,9 +254,8 @@ static void init_output(struct hda_codec *codec, hda_nid_t pin, hda_nid_t dac) static void init_input(struct hda_codec *codec, hda_nid_t pin, hda_nid_t adc) { if (pin) { - snd_hda_codec_write(codec, pin, 0, - AC_VERB_SET_PIN_WIDGET_CONTROL, - PIN_VREF80); + snd_hda_set_pin_ctl(codec, pin, PIN_IN | + snd_hda_get_default_vref(codec, pin)); if (get_wcaps(codec, pin) & AC_WCAP_IN_AMP) snd_hda_codec_write(codec, pin, 0, AC_VERB_SET_AMP_GAIN_MUTE, diff --git a/sound/pci/hda/patch_cirrus.c b/sound/pci/hda/patch_cirrus.c index c83ccdba1e5..9647ed4d792 100644 --- a/sound/pci/hda/patch_cirrus.c +++ b/sound/pci/hda/patch_cirrus.c @@ -26,6 +26,7 @@ #include <sound/core.h> #include "hda_codec.h" #include "hda_local.h" +#include "hda_auto_parser.h" #include "hda_jack.h" #include <sound/tlv.h> @@ -933,8 +934,7 @@ static void cs_automute(struct hda_codec *codec) pin_ctl = 0; nid = cfg->speaker_pins[i]; - snd_hda_codec_write(codec, nid, 0, - AC_VERB_SET_PIN_WIDGET_CONTROL, pin_ctl); + snd_hda_set_pin_ctl(codec, nid, pin_ctl); } if (spec->gpio_eapd_hp) { unsigned int gpio = hp_present ? @@ -948,16 +948,14 @@ static void cs_automute(struct hda_codec *codec) /* mute HPs if spdif jack (SENSE_B) is present */ for (i = 0; i < cfg->hp_outs; i++) { nid = cfg->hp_pins[i]; - snd_hda_codec_write(codec, nid, 0, - AC_VERB_SET_PIN_WIDGET_CONTROL, + snd_hda_set_pin_ctl(codec, nid, (spdif_present && spec->sense_b) ? 0 : PIN_HP); } /* SPDIF TX on/off */ if (cfg->dig_outs) { nid = cfg->dig_out_pins[0]; - snd_hda_codec_write(codec, nid, 0, - AC_VERB_SET_PIN_WIDGET_CONTROL, + snd_hda_set_pin_ctl(codec, nid, spdif_present ? PIN_OUT : 0); } @@ -1024,13 +1022,11 @@ static void init_output(struct hda_codec *codec) /* set appropriate pin controls */ for (i = 0; i < cfg->line_outs; i++) - snd_hda_codec_write(codec, cfg->line_out_pins[i], 0, - AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT); + snd_hda_set_pin_ctl(codec, cfg->line_out_pins[i], PIN_OUT); /* HP */ for (i = 0; i < cfg->hp_outs; i++) { hda_nid_t nid = cfg->hp_pins[i]; - snd_hda_codec_write(codec, nid, 0, - AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP); + snd_hda_set_pin_ctl(codec, nid, PIN_HP); if (!cfg->speaker_outs) continue; if (get_wcaps(codec, nid) & AC_WCAP_UNSOL_CAP) { @@ -1041,8 +1037,7 @@ static void init_output(struct hda_codec *codec) /* Speaker */ for (i = 0; i < cfg->speaker_outs; i++) - snd_hda_codec_write(codec, cfg->speaker_pins[i], 0, - AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT); + snd_hda_set_pin_ctl(codec, cfg->speaker_pins[i], PIN_OUT); /* SPDIF is enabled on presence detect for CS421x */ if (spec->hp_detect || spec->spdif_detect) @@ -1063,14 +1058,9 @@ static void init_input(struct hda_codec *codec) continue; /* set appropriate pin control and mute first */ ctl = PIN_IN; - if (cfg->inputs[i].type == AUTO_PIN_MIC) { - unsigned int caps = snd_hda_query_pin_caps(codec, pin); - caps >>= AC_PINCAP_VREF_SHIFT; - if (caps & AC_PINCAP_VREF_80) - ctl = PIN_VREF80; - } - snd_hda_codec_write(codec, pin, 0, - AC_VERB_SET_PIN_WIDGET_CONTROL, ctl); + if (cfg->inputs[i].type == AUTO_PIN_MIC) + ctl |= snd_hda_get_default_vref(codec, pin); + snd_hda_set_pin_ctl(codec, pin, ctl); snd_hda_codec_write(codec, spec->adc_nid[i], 0, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(spec->adc_idx[i])); diff --git a/sound/pci/hda/patch_cmedia.c b/sound/pci/hda/patch_cmedia.c index b6767b4ced4..c8fdaaefe70 100644 --- a/sound/pci/hda/patch_cmedia.c +++ b/sound/pci/hda/patch_cmedia.c @@ -29,6 +29,7 @@ #include <sound/core.h> #include "hda_codec.h" #include "hda_local.h" +#include "hda_auto_parser.h" #define NUM_PINS 11 diff --git a/sound/pci/hda/patch_conexant.c b/sound/pci/hda/patch_conexant.c index d906c5b74cf..3acb5824ad3 100644 --- a/sound/pci/hda/patch_conexant.c +++ b/sound/pci/hda/patch_conexant.c @@ -30,6 +30,7 @@ #include "hda_codec.h" #include "hda_local.h" +#include "hda_auto_parser.h" #include "hda_beep.h" #include "hda_jack.h" @@ -66,6 +67,7 @@ struct imux_info { }; struct conexant_spec { + struct hda_gen_spec gen; const struct snd_kcontrol_new *mixers[5]; int num_mixers; @@ -141,6 +143,7 @@ struct conexant_spec { unsigned int hp_laptop:1; unsigned int asus:1; unsigned int pin_eapd_ctrls:1; + unsigned int fixup_stereo_dmic:1; unsigned int adc_switching:1; @@ -1601,17 +1604,13 @@ static void cxt5051_update_speaker(struct hda_codec *codec) unsigned int pinctl; /* headphone pin */ pinctl = (spec->hp_present && spec->cur_eapd) ? PIN_HP : 0; - snd_hda_codec_write(codec, 0x16, 0, AC_VERB_SET_PIN_WIDGET_CONTROL, - pinctl); + snd_hda_set_pin_ctl(codec, 0x16, pinctl); /* speaker pin */ pinctl = (!spec->hp_present && spec->cur_eapd) ? PIN_OUT : 0; - snd_hda_codec_write(codec, 0x1a, 0, AC_VERB_SET_PIN_WIDGET_CONTROL, - pinctl); + snd_hda_set_pin_ctl(codec, 0x1a, pinctl); /* on ideapad there is an additional speaker (subwoofer) to mute */ if (spec->ideapad) - snd_hda_codec_write(codec, 0x1b, 0, - AC_VERB_SET_PIN_WIDGET_CONTROL, - pinctl); + snd_hda_set_pin_ctl(codec, 0x1b, pinctl); } /* turn on/off EAPD (+ mute HP) as a master switch */ @@ -1996,8 +1995,7 @@ static void cxt5066_update_speaker(struct hda_codec *codec) /* Port A (HP) */ pinctl = (hp_port_a_present(spec) && spec->cur_eapd) ? PIN_HP : 0; - snd_hda_codec_write(codec, 0x19, 0, AC_VERB_SET_PIN_WIDGET_CONTROL, - pinctl); + snd_hda_set_pin_ctl(codec, 0x19, pinctl); /* Port D (HP/LO) */ pinctl = spec->cur_eapd ? spec->port_d_mode : 0; @@ -2010,13 +2008,11 @@ static void cxt5066_update_speaker(struct hda_codec *codec) if (!hp_port_d_present(spec)) pinctl = 0; } - snd_hda_codec_write(codec, 0x1c, 0, AC_VERB_SET_PIN_WIDGET_CONTROL, - pinctl); + snd_hda_set_pin_ctl(codec, 0x1c, pinctl); /* CLASS_D AMP */ pinctl = (!spec->hp_present && spec->cur_eapd) ? PIN_OUT : 0; - snd_hda_codec_write(codec, 0x1f, 0, AC_VERB_SET_PIN_WIDGET_CONTROL, - pinctl); + snd_hda_set_pin_ctl(codec, 0x1f, pinctl); } /* turn on/off EAPD (+ mute HP) as a master switch */ @@ -2047,8 +2043,7 @@ static int cxt5066_set_olpc_dc_bias(struct hda_codec *codec) /* Even though port F is the DC input, the bias is controlled on port B. * we also leave that port as an active input (but unselected) in DC mode * just in case that is necessary to make the bias setting take effect. */ - return snd_hda_codec_write_cache(codec, 0x1a, 0, - AC_VERB_SET_PIN_WIDGET_CONTROL, + return snd_hda_set_pin_ctl_cache(codec, 0x1a, cxt5066_olpc_dc_bias.items[spec->dc_input_bias].index); } @@ -2081,14 +2076,14 @@ static void cxt5066_olpc_select_mic(struct hda_codec *codec) } /* disable DC (port F) */ - snd_hda_codec_write(codec, 0x1e, 0, AC_VERB_SET_PIN_WIDGET_CONTROL, 0); + snd_hda_set_pin_ctl(codec, 0x1e, 0); /* external mic, port B */ - snd_hda_codec_write(codec, 0x1a, 0, AC_VERB_SET_PIN_WIDGET_CONTROL, + snd_hda_set_pin_ctl(codec, 0x1a, spec->ext_mic_present ? CXT5066_OLPC_EXT_MIC_BIAS : 0); /* internal mic, port C */ - snd_hda_codec_write(codec, 0x1b, 0, AC_VERB_SET_PIN_WIDGET_CONTROL, + snd_hda_set_pin_ctl(codec, 0x1b, spec->ext_mic_present ? 0 : PIN_VREF80); } @@ -3357,9 +3352,7 @@ static void do_automute(struct hda_codec *codec, int num_pins, struct conexant_spec *spec = codec->spec; int i; for (i = 0; i < num_pins; i++) - snd_hda_codec_write(codec, pins[i], 0, - AC_VERB_SET_PIN_WIDGET_CONTROL, - on ? PIN_OUT : 0); + snd_hda_set_pin_ctl(codec, pins[i], on ? PIN_OUT : 0); if (spec->pin_eapd_ctrls) cx_auto_turn_eapd(codec, num_pins, pins, on); } @@ -3976,8 +3969,7 @@ static void cx_auto_init_output(struct hda_codec *codec) if (snd_hda_query_pin_caps(codec, cfg->hp_pins[i]) & AC_PINCAP_HP_DRV) val |= AC_PINCTL_HP_EN; - snd_hda_codec_write(codec, cfg->hp_pins[i], 0, - AC_VERB_SET_PIN_WIDGET_CONTROL, val); + snd_hda_set_pin_ctl(codec, cfg->hp_pins[i], val); } mute_outputs(codec, cfg->hp_outs, cfg->hp_pins); mute_outputs(codec, cfg->line_outs, cfg->line_out_pins); @@ -4030,13 +4022,11 @@ static void cx_auto_init_input(struct hda_codec *codec) } for (i = 0; i < cfg->num_inputs; i++) { - unsigned int type; + hda_nid_t pin = cfg->inputs[i].pin; + unsigned int type = PIN_IN; if (cfg->inputs[i].type == AUTO_PIN_MIC) - type = PIN_VREF80; - else - type = PIN_IN; - snd_hda_codec_write(codec, cfg->inputs[i].pin, 0, - AC_VERB_SET_PIN_WIDGET_CONTROL, type); + type |= snd_hda_get_default_vref(codec, pin); + snd_hda_set_pin_ctl(codec, pin, type); } if (spec->auto_mic) { @@ -4063,11 +4053,9 @@ static void cx_auto_init_digital(struct hda_codec *codec) struct auto_pin_cfg *cfg = &spec->autocfg; if (spec->multiout.dig_out_nid) - snd_hda_codec_write(codec, cfg->dig_out_pins[0], 0, - AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT); + snd_hda_set_pin_ctl(codec, cfg->dig_out_pins[0], PIN_OUT); if (spec->dig_in_nid) - snd_hda_codec_write(codec, cfg->dig_in_pin, 0, - AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN); + snd_hda_set_pin_ctl(codec, cfg->dig_in_pin, PIN_IN); } static int cx_auto_init(struct hda_codec *codec) @@ -4084,9 +4072,9 @@ static int cx_auto_init(struct hda_codec *codec) static int cx_auto_add_volume_idx(struct hda_codec *codec, const char *basename, const char *dir, int cidx, - hda_nid_t nid, int hda_dir, int amp_idx) + hda_nid_t nid, int hda_dir, int amp_idx, int chs) { - static char name[32]; + static char name[44]; static struct snd_kcontrol_new knew[] = { HDA_CODEC_VOLUME(name, 0, 0, 0), HDA_CODEC_MUTE(name, 0, 0, 0), @@ -4096,7 +4084,7 @@ static int cx_auto_add_volume_idx(struct hda_codec *codec, const char *basename, for (i = 0; i < 2; i++) { struct snd_kcontrol *kctl; - knew[i].private_value = HDA_COMPOSE_AMP_VAL(nid, 3, amp_idx, + knew[i].private_value = HDA_COMPOSE_AMP_VAL(nid, chs, amp_idx, hda_dir); knew[i].subdevice = HDA_SUBDEV_AMP_FLAG; knew[i].index = cidx; @@ -4115,7 +4103,7 @@ static int cx_auto_add_volume_idx(struct hda_codec *codec, const char *basename, } #define cx_auto_add_volume(codec, str, dir, cidx, nid, hda_dir) \ - cx_auto_add_volume_idx(codec, str, dir, cidx, nid, hda_dir, 0) + cx_auto_add_volume_idx(codec, str, dir, cidx, nid, hda_dir, 0, 3) #define cx_auto_add_pb_volume(codec, nid, str, idx) \ cx_auto_add_volume(codec, str, " Playback", idx, nid, HDA_OUTPUT) @@ -4185,6 +4173,36 @@ static int cx_auto_build_output_controls(struct hda_codec *codec) return 0; } +/* Returns zero if this is a normal stereo channel, and non-zero if it should + be split in two independent channels. + dest_label must be at least 44 characters. */ +static int cx_auto_get_rightch_label(struct hda_codec *codec, const char *label, + char *dest_label, int nid) +{ + struct conexant_spec *spec = codec->spec; + int i; + + if (!spec->fixup_stereo_dmic) + return 0; + + for (i = 0; i < AUTO_CFG_MAX_INS; i++) { + int def_conf; + if (spec->autocfg.inputs[i].pin != nid) + continue; + + if (spec->autocfg.inputs[i].type != AUTO_PIN_MIC) + return 0; + def_conf = snd_hda_codec_get_pincfg(codec, nid); + if (snd_hda_get_input_pin_attr(def_conf) != INPUT_PIN_ATTR_INT) + return 0; + + /* Finally found the inverted internal mic! */ + snprintf(dest_label, 44, "Inverted %s", label); + return 1; + } + return 0; +} + static int cx_auto_add_capture_volume(struct hda_codec *codec, hda_nid_t nid, const char *label, const char *pfx, int cidx) @@ -4193,14 +4211,25 @@ static int cx_auto_add_capture_volume(struct hda_codec *codec, hda_nid_t nid, int i; for (i = 0; i < spec->num_adc_nids; i++) { + char rightch_label[44]; hda_nid_t adc_nid = spec->adc_nids[i]; int idx = get_input_connection(codec, adc_nid, nid); if (idx < 0) continue; if (codec->single_adc_amp) idx = 0; + + if (cx_auto_get_rightch_label(codec, label, rightch_label, nid)) { + /* Make two independent kcontrols for left and right */ + int err = cx_auto_add_volume_idx(codec, label, pfx, + cidx, adc_nid, HDA_INPUT, idx, 1); + if (err < 0) + return err; + return cx_auto_add_volume_idx(codec, rightch_label, pfx, + cidx, adc_nid, HDA_INPUT, idx, 2); + } return cx_auto_add_volume_idx(codec, label, pfx, - cidx, adc_nid, HDA_INPUT, idx); + cidx, adc_nid, HDA_INPUT, idx, 3); } return 0; } @@ -4213,9 +4242,19 @@ static int cx_auto_add_boost_volume(struct hda_codec *codec, int idx, int i, con; nid = spec->imux_info[idx].pin; - if (get_wcaps(codec, nid) & AC_WCAP_IN_AMP) + if (get_wcaps(codec, nid) & AC_WCAP_IN_AMP) { + char rightch_label[44]; + if (cx_auto_get_rightch_label(codec, label, rightch_label, nid)) { + int err = cx_auto_add_volume_idx(codec, label, " Boost", + cidx, nid, HDA_INPUT, 0, 1); + if (err < 0) + return err; + return cx_auto_add_volume_idx(codec, rightch_label, " Boost", + cidx, nid, HDA_INPUT, 0, 2); + } return cx_auto_add_volume(codec, label, " Boost", cidx, nid, HDA_INPUT); + } con = __select_input_connection(codec, spec->imux_info[idx].adc, nid, &mux, false, 0); if (con < 0) @@ -4370,37 +4409,21 @@ static const struct hda_codec_ops cx_auto_patch_ops = { /* * pin fix-up */ -struct cxt_pincfg { - hda_nid_t nid; - u32 val; -}; - -static void apply_pincfg(struct hda_codec *codec, const struct cxt_pincfg *cfg) -{ - for (; cfg->nid; cfg++) - snd_hda_codec_set_pincfg(codec, cfg->nid, cfg->val); - -} - -static void apply_pin_fixup(struct hda_codec *codec, - const struct snd_pci_quirk *quirk, - const struct cxt_pincfg **table) -{ - quirk = snd_pci_quirk_lookup(codec->bus->pci, quirk); - if (quirk) { - snd_printdd(KERN_INFO "hda_codec: applying pincfg for %s\n", - quirk->name); - apply_pincfg(codec, table[quirk->value]); - } -} - enum { CXT_PINCFG_LENOVO_X200, CXT_PINCFG_LENOVO_TP410, + CXT_FIXUP_STEREO_DMIC, }; +static void cxt_fixup_stereo_dmic(struct hda_codec *codec, + const struct hda_fixup *fix, int action) +{ + struct conexant_spec *spec = codec->spec; + spec->fixup_stereo_dmic = 1; +} + /* ThinkPad X200 & co with cxt5051 */ -static const struct cxt_pincfg cxt_pincfg_lenovo_x200[] = { +static const struct hda_pintbl cxt_pincfg_lenovo_x200[] = { { 0x16, 0x042140ff }, /* HP (seq# overridden) */ { 0x17, 0x21a11000 }, /* dock-mic */ { 0x19, 0x2121103f }, /* dock-HP */ @@ -4409,16 +4432,26 @@ static const struct cxt_pincfg cxt_pincfg_lenovo_x200[] = { }; /* ThinkPad 410/420/510/520, X201 & co with cxt5066 */ -static const struct cxt_pincfg cxt_pincfg_lenovo_tp410[] = { +static const struct hda_pintbl cxt_pincfg_lenovo_tp410[] = { { 0x19, 0x042110ff }, /* HP (seq# overridden) */ { 0x1a, 0x21a190f0 }, /* dock-mic */ { 0x1c, 0x212140ff }, /* dock-HP */ {} }; -static const struct cxt_pincfg *cxt_pincfg_tbl[] = { - [CXT_PINCFG_LENOVO_X200] = cxt_pincfg_lenovo_x200, - [CXT_PINCFG_LENOVO_TP410] = cxt_pincfg_lenovo_tp410, +static const struct hda_fixup cxt_fixups[] = { + [CXT_PINCFG_LENOVO_X200] = { + .type = HDA_FIXUP_PINS, + .v.pins = cxt_pincfg_lenovo_x200, + }, + [CXT_PINCFG_LENOVO_TP410] = { + .type = HDA_FIXUP_PINS, + .v.pins = cxt_pincfg_lenovo_tp410, + }, + [CXT_FIXUP_STEREO_DMIC] = { + .type = HDA_FIXUP_FUNC, + .v.func = cxt_fixup_stereo_dmic, + }, }; static const struct snd_pci_quirk cxt5051_fixups[] = { @@ -4432,6 +4465,7 @@ static const struct snd_pci_quirk cxt5066_fixups[] = { SND_PCI_QUIRK(0x17aa, 0x215f, "Lenovo T510", CXT_PINCFG_LENOVO_TP410), SND_PCI_QUIRK(0x17aa, 0x21ce, "Lenovo T420", CXT_PINCFG_LENOVO_TP410), SND_PCI_QUIRK(0x17aa, 0x21cf, "Lenovo T520", CXT_PINCFG_LENOVO_TP410), + SND_PCI_QUIRK(0x17aa, 0x3975, "Lenovo U300s", CXT_FIXUP_STEREO_DMIC), {} }; @@ -4471,13 +4505,16 @@ static int patch_conexant_auto(struct hda_codec *codec) case 0x14f15051: add_cx5051_fake_mutes(codec); codec->pin_amp_workaround = 1; - apply_pin_fixup(codec, cxt5051_fixups, cxt_pincfg_tbl); + snd_hda_pick_fixup(codec, NULL, cxt5051_fixups, cxt_fixups); break; default: codec->pin_amp_workaround = 1; - apply_pin_fixup(codec, cxt5066_fixups, cxt_pincfg_tbl); + snd_hda_pick_fixup(codec, NULL, cxt5066_fixups, cxt_fixups); + break; } + snd_hda_apply_fixup(codec, HDA_FIXUP_ACT_PRE_PROBE); + /* Show mute-led control only on HP laptops * This is a sort of white-list: on HP laptops, EAPD corresponds * only to the mute-LED without actualy amp function. Meanwhile, @@ -4556,6 +4593,12 @@ static const struct hda_codec_preset snd_hda_preset_conexant[] = { .patch = patch_conexant_auto }, { .id = 0x14f150b9, .name = "CX20665", .patch = patch_conexant_auto }, + { .id = 0x14f1510f, .name = "CX20751/2", + .patch = patch_conexant_auto }, + { .id = 0x14f15110, .name = "CX20751/2", + .patch = patch_conexant_auto }, + { .id = 0x14f15111, .name = "CX20753/4", + .patch = patch_conexant_auto }, {} /* terminator */ }; @@ -4576,6 +4619,9 @@ MODULE_ALIAS("snd-hda-codec-id:14f150ab"); MODULE_ALIAS("snd-hda-codec-id:14f150ac"); MODULE_ALIAS("snd-hda-codec-id:14f150b8"); MODULE_ALIAS("snd-hda-codec-id:14f150b9"); +MODULE_ALIAS("snd-hda-codec-id:14f1510f"); +MODULE_ALIAS("snd-hda-codec-id:14f15110"); +MODULE_ALIAS("snd-hda-codec-id:14f15111"); MODULE_LICENSE("GPL"); MODULE_DESCRIPTION("Conexant HD-audio codec"); diff --git a/sound/pci/hda/patch_hdmi.c b/sound/pci/hda/patch_hdmi.c index 83f345f3c96..ad319d4dc32 100644 --- a/sound/pci/hda/patch_hdmi.c +++ b/sound/pci/hda/patch_hdmi.c @@ -1592,10 +1592,10 @@ static int nvhdmi_8ch_7x_pcm_prepare(struct hda_pcm_stream *hinfo, unsigned int dataDCC2, channel_id; int i; struct hdmi_spec *spec = codec->spec; - struct hda_spdif_out *spdif = - snd_hda_spdif_out_of_nid(codec, spec->cvts[0].cvt_nid); + struct hda_spdif_out *spdif; mutex_lock(&codec->spdif_mutex); + spdif = snd_hda_spdif_out_of_nid(codec, spec->cvts[0].cvt_nid); chs = substream->runtime->channels; diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 708d47c294e..ff71dcef08e 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -32,6 +32,7 @@ #include <sound/jack.h> #include "hda_codec.h" #include "hda_local.h" +#include "hda_auto_parser.h" #include "hda_beep.h" #include "hda_jack.h" @@ -66,8 +67,6 @@ struct alc_customize_define { unsigned int fixup:1; /* Means that this sku is set by driver, not read from hw */ }; -struct alc_fixup; - struct alc_multi_io { hda_nid_t pin; /* multi-io widget pin NID */ hda_nid_t dac; /* DAC to be connected */ @@ -82,19 +81,33 @@ enum { #define MAX_VOL_NIDS 0x40 +/* make compatible with old code */ +#define alc_apply_pincfgs snd_hda_apply_pincfgs +#define alc_apply_fixup snd_hda_apply_fixup +#define alc_pick_fixup snd_hda_pick_fixup +#define alc_fixup hda_fixup +#define alc_pincfg hda_pintbl +#define alc_model_fixup hda_model_fixup + +#define ALC_FIXUP_PINS HDA_FIXUP_PINS +#define ALC_FIXUP_VERBS HDA_FIXUP_VERBS +#define ALC_FIXUP_FUNC HDA_FIXUP_FUNC + +#define ALC_FIXUP_ACT_PRE_PROBE HDA_FIXUP_ACT_PRE_PROBE +#define ALC_FIXUP_ACT_PROBE HDA_FIXUP_ACT_PROBE +#define ALC_FIXUP_ACT_INIT HDA_FIXUP_ACT_INIT +#define ALC_FIXUP_ACT_BUILD HDA_FIXUP_ACT_BUILD + + struct alc_spec { + struct hda_gen_spec gen; + /* codec parameterization */ const struct snd_kcontrol_new *mixers[5]; /* mixer arrays */ unsigned int num_mixers; const struct snd_kcontrol_new *cap_mixer; /* capture mixer */ unsigned int beep_amp; /* beep amp value, set via set_beep_amp() */ - const struct hda_verb *init_verbs[10]; /* initialization verbs - * don't forget NULL - * termination! - */ - unsigned int num_init_verbs; - char stream_name_analog[32]; /* analog PCM stream */ const struct hda_pcm_stream *stream_analog_playback; const struct hda_pcm_stream *stream_analog_capture; @@ -210,11 +223,6 @@ struct alc_spec { unsigned int pll_coef_idx, pll_coef_bit; unsigned int coef0; - /* fix-up list */ - int fixup_id; - const struct alc_fixup *fixup_list; - const char *fixup_name; - /* multi-io */ int multi_ios; struct alc_multi_io multi_io[4]; @@ -319,13 +327,16 @@ static int alc_mux_select(struct hda_codec *codec, unsigned int adc_idx, /* for shared I/O, change the pin-control accordingly */ if (spec->shared_mic_hp) { + unsigned int val; + hda_nid_t pin = spec->autocfg.inputs[1].pin; /* NOTE: this assumes that there are only two inputs, the * first is the real internal mic and the second is HP jack. */ - snd_hda_codec_write(codec, spec->autocfg.inputs[1].pin, 0, - AC_VERB_SET_PIN_WIDGET_CONTROL, - spec->cur_mux[adc_idx] ? - PIN_VREF80 : PIN_HP); + if (spec->cur_mux[adc_idx]) + val = snd_hda_get_default_vref(codec, pin) | PIN_IN; + else + val = PIN_HP; + snd_hda_set_pin_ctl(codec, pin, val); spec->automute_speaker = !spec->cur_mux[adc_idx]; call_update_outputs(codec); } @@ -338,7 +349,7 @@ static int alc_mux_select(struct hda_codec *codec, unsigned int adc_idx, nid = get_capsrc(spec, adc_idx); /* no selection? */ - num_conns = snd_hda_get_conn_list(codec, nid, NULL); + num_conns = snd_hda_get_num_conns(codec, nid); if (num_conns <= 1) return 1; @@ -376,25 +387,9 @@ static void alc_set_input_pin(struct hda_codec *codec, hda_nid_t nid, int auto_pin_type) { unsigned int val = PIN_IN; - - if (auto_pin_type == AUTO_PIN_MIC) { - unsigned int pincap; - unsigned int oldval; - oldval = snd_hda_codec_read(codec, nid, 0, - AC_VERB_GET_PIN_WIDGET_CONTROL, 0); - pincap = snd_hda_query_pin_caps(codec, nid); - pincap = (pincap & AC_PINCAP_VREF) >> AC_PINCAP_VREF_SHIFT; - /* if the default pin setup is vref50, we give it priority */ - if ((pincap & AC_PINCAP_VREF_80) && oldval != PIN_VREF50) - val = PIN_VREF80; - else if (pincap & AC_PINCAP_VREF_50) - val = PIN_VREF50; - else if (pincap & AC_PINCAP_VREF_100) - val = PIN_VREF100; - else if (pincap & AC_PINCAP_VREF_GRD) - val = PIN_VREFGRD; - } - snd_hda_codec_write(codec, nid, 0, AC_VERB_SET_PIN_WIDGET_CONTROL, val); + if (auto_pin_type == AUTO_PIN_MIC) + val |= snd_hda_get_default_vref(codec, nid); + snd_hda_set_pin_ctl(codec, nid, val); } /* @@ -409,13 +404,6 @@ static void add_mixer(struct alc_spec *spec, const struct snd_kcontrol_new *mix) spec->mixers[spec->num_mixers++] = mix; } -static void add_verb(struct alc_spec *spec, const struct hda_verb *verb) -{ - if (snd_BUG_ON(spec->num_init_verbs >= ARRAY_SIZE(spec->init_verbs))) - return; - spec->init_verbs[spec->num_init_verbs++] = verb; -} - /* * GPIO setup tables, used in initialization */ @@ -517,9 +505,7 @@ static void do_automute(struct hda_codec *codec, int num_pins, hda_nid_t *pins, } else val = 0; val |= pin_bits; - snd_hda_codec_write(codec, nid, 0, - AC_VERB_SET_PIN_WIDGET_CONTROL, - val); + snd_hda_set_pin_ctl(codec, nid, val); break; case ALC_AUTOMUTE_AMP: snd_hda_codec_amp_stereo(codec, nid, HDA_OUTPUT, 0, @@ -1200,6 +1186,16 @@ static void alc_auto_check_switches(struct hda_codec *codec) */ #define ALC_FIXUP_SKU_IGNORE (2) +static void alc_fixup_sku_ignore(struct hda_codec *codec, + const struct hda_fixup *fix, int action) +{ + struct alc_spec *spec = codec->spec; + if (action == HDA_FIXUP_ACT_PRE_PROBE) { + spec->cdefine.fixup = 1; + spec->cdefine.sku_cfg = ALC_FIXUP_SKU_IGNORE; + } +} + static int alc_auto_parse_customize_define(struct hda_codec *codec) { unsigned int ass, tmp, i; @@ -1403,178 +1399,6 @@ static void alc_ssid_check(struct hda_codec *codec, const hda_nid_t *ports) } /* - * Fix-up pin default configurations and add default verbs - */ - -struct alc_pincfg { - hda_nid_t nid; - u32 val; -}; - -struct alc_model_fixup { - const int id; - const char *name; -}; - -struct alc_fixup { - int type; - bool chained; - int chain_id; - union { - unsigned int sku; - const struct alc_pincfg *pins; - const struct hda_verb *verbs; - void (*func)(struct hda_codec *codec, - const struct alc_fixup *fix, - int action); - } v; -}; - -enum { - ALC_FIXUP_INVALID, - ALC_FIXUP_SKU, - ALC_FIXUP_PINS, - ALC_FIXUP_VERBS, - ALC_FIXUP_FUNC, -}; - -enum { - ALC_FIXUP_ACT_PRE_PROBE, - ALC_FIXUP_ACT_PROBE, - ALC_FIXUP_ACT_INIT, - ALC_FIXUP_ACT_BUILD, -}; - -static void alc_apply_pincfgs(struct hda_codec *codec, - const struct alc_pincfg *cfg) -{ - for (; cfg->nid; cfg++) - snd_hda_codec_set_pincfg(codec, cfg->nid, cfg->val); -} - -static void alc_apply_fixup(struct hda_codec *codec, int action) -{ - struct alc_spec *spec = codec->spec; - int id = spec->fixup_id; -#ifdef CONFIG_SND_DEBUG_VERBOSE - const char *modelname = spec->fixup_name; -#endif - int depth = 0; - - if (!spec->fixup_list) - return; - - while (id >= 0) { - const struct alc_fixup *fix = spec->fixup_list + id; - const struct alc_pincfg *cfg; - - switch (fix->type) { - case ALC_FIXUP_SKU: - if (action != ALC_FIXUP_ACT_PRE_PROBE || !fix->v.sku) - break; - snd_printdd(KERN_INFO "hda_codec: %s: " - "Apply sku override for %s\n", - codec->chip_name, modelname); - spec->cdefine.sku_cfg = fix->v.sku; - spec->cdefine.fixup = 1; - break; - case ALC_FIXUP_PINS: - cfg = fix->v.pins; - if (action != ALC_FIXUP_ACT_PRE_PROBE || !cfg) - break; - snd_printdd(KERN_INFO "hda_codec: %s: " - "Apply pincfg for %s\n", - codec->chip_name, modelname); - alc_apply_pincfgs(codec, cfg); - break; - case ALC_FIXUP_VERBS: - if (action != ALC_FIXUP_ACT_PROBE || !fix->v.verbs) - break; - snd_printdd(KERN_INFO "hda_codec: %s: " - "Apply fix-verbs for %s\n", - codec->chip_name, modelname); - add_verb(codec->spec, fix->v.verbs); - break; - case ALC_FIXUP_FUNC: - if (!fix->v.func) - break; - snd_printdd(KERN_INFO "hda_codec: %s: " - "Apply fix-func for %s\n", - codec->chip_name, modelname); - fix->v.func(codec, fix, action); - break; - default: - snd_printk(KERN_ERR "hda_codec: %s: " - "Invalid fixup type %d\n", - codec->chip_name, fix->type); - break; - } - if (!fix->chained) - break; - if (++depth > 10) - break; - id = fix->chain_id; - } -} - -static void alc_pick_fixup(struct hda_codec *codec, - const struct alc_model_fixup *models, - const struct snd_pci_quirk *quirk, - const struct alc_fixup *fixlist) -{ - struct alc_spec *spec = codec->spec; - const struct snd_pci_quirk *q; - int id = -1; - const char *name = NULL; - - /* when model=nofixup is given, don't pick up any fixups */ - if (codec->modelname && !strcmp(codec->modelname, "nofixup")) { - spec->fixup_list = NULL; - spec->fixup_id = -1; - return; - } - - if (codec->modelname && models) { - while (models->name) { - if (!strcmp(codec->modelname, models->name)) { - id = models->id; - name = models->name; - break; - } - models++; - } - } - if (id < 0) { - q = snd_pci_quirk_lookup(codec->bus->pci, quirk); - if (q) { - id = q->value; -#ifdef CONFIG_SND_DEBUG_VERBOSE - name = q->name; -#endif - } - } - if (id < 0) { - for (q = quirk; q->subvendor; q++) { - unsigned int vendorid = - q->subdevice | (q->subvendor << 16); - if (vendorid == codec->subsystem_id) { - id = q->value; -#ifdef CONFIG_SND_DEBUG_VERBOSE - name = q->name; -#endif - break; - } - } - } - - spec->fixup_id = id; - if (id >= 0) { - spec->fixup_list = fixlist; - spec->fixup_name = name; - } -} - -/* * COEF access helper functions */ static int alc_read_coef_idx(struct hda_codec *codec, @@ -1621,8 +1445,7 @@ static void alc_auto_init_digital(struct hda_codec *codec) pin = spec->autocfg.dig_out_pins[i]; if (!pin) continue; - snd_hda_codec_write(codec, pin, 0, - AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT); + snd_hda_set_pin_ctl(codec, pin, PIN_OUT); if (!i) dac = spec->multiout.dig_out_nid; else @@ -1635,9 +1458,7 @@ static void alc_auto_init_digital(struct hda_codec *codec) } pin = spec->autocfg.dig_in_pin; if (pin) - snd_hda_codec_write(codec, pin, 0, - AC_VERB_SET_PIN_WIDGET_CONTROL, - PIN_IN); + snd_hda_set_pin_ctl(codec, pin, PIN_IN); } /* parse digital I/Os and set up NIDs in BIOS auto-parse mode */ @@ -2068,7 +1889,6 @@ static void alc_auto_init_std(struct hda_codec *codec); static int alc_init(struct hda_codec *codec) { struct alc_spec *spec = codec->spec; - unsigned int i; if (spec->init_hook) spec->init_hook(codec); @@ -2076,8 +1896,6 @@ static int alc_init(struct hda_codec *codec) alc_fix_pll(codec); alc_auto_init_amp(codec, spec->init_amp); - for (i = 0; i < spec->num_init_verbs; i++) - snd_hda_sequence_write(codec, spec->init_verbs[i]); alc_init_special_input_src(codec); alc_auto_init_std(codec); @@ -2725,7 +2543,6 @@ static int alc_auto_fill_adc_caps(struct hda_codec *codec) nid = codec->start_nid; for (i = 0; i < codec->num_nodes; i++, nid++) { hda_nid_t src; - const hda_nid_t *list; unsigned int caps = get_wcaps(codec, nid); int type = get_wcaps_type(caps); @@ -2743,13 +2560,14 @@ static int alc_auto_fill_adc_caps(struct hda_codec *codec) cap_nids[nums] = src; break; } - n = snd_hda_get_conn_list(codec, src, &list); + n = snd_hda_get_num_conns(codec, src); if (n > 1) { cap_nids[nums] = src; break; } else if (n != 1) break; - src = *list; + if (snd_hda_get_connections(codec, src, &src, 1) != 1) + break; } if (++nums >= max_nums) break; @@ -2856,8 +2674,7 @@ static int alc_auto_create_shared_input(struct hda_codec *codec) static void alc_set_pin_output(struct hda_codec *codec, hda_nid_t nid, unsigned int pin_type) { - snd_hda_codec_write(codec, nid, 0, AC_VERB_SET_PIN_WIDGET_CONTROL, - pin_type); + snd_hda_set_pin_ctl(codec, nid, pin_type); /* unmute pin */ if (nid_has_mute(codec, nid, HDA_OUTPUT)) snd_hda_codec_write(codec, nid, 0, AC_VERB_SET_AMP_GAIN_MUTE, @@ -2891,7 +2708,7 @@ static void alc_auto_init_analog_input(struct hda_codec *codec) /* mute all loopback inputs */ if (spec->mixer_nid) { - int nums = snd_hda_get_conn_list(codec, spec->mixer_nid, NULL); + int nums = snd_hda_get_num_conns(codec, spec->mixer_nid); for (i = 0; i < nums; i++) snd_hda_codec_write(codec, spec->mixer_nid, 0, AC_VERB_SET_AMP_GAIN_MUTE, @@ -3521,7 +3338,7 @@ static int alc_auto_add_sw_ctl(struct hda_codec *codec, if (wid_type == AC_WID_PIN || wid_type == AC_WID_AUD_OUT) { type = ALC_CTL_WIDGET_MUTE; val = HDA_COMPOSE_AMP_VAL(nid, chs, 0, HDA_OUTPUT); - } else if (snd_hda_get_conn_list(codec, nid, NULL) == 1) { + } else if (snd_hda_get_num_conns(codec, nid) == 1) { type = ALC_CTL_WIDGET_MUTE; val = HDA_COMPOSE_AMP_VAL(nid, chs, 0, HDA_INPUT); } else { @@ -3998,9 +3815,7 @@ static int alc_set_multi_io(struct hda_codec *codec, int idx, bool output) snd_hda_codec_read(codec, nid, 0, AC_VERB_GET_PIN_WIDGET_CONTROL, 0); if (output) { - snd_hda_codec_update_cache(codec, nid, 0, - AC_VERB_SET_PIN_WIDGET_CONTROL, - PIN_OUT); + snd_hda_set_pin_ctl_cache(codec, nid, PIN_OUT); if (get_wcaps(codec, nid) & AC_WCAP_OUT_AMP) snd_hda_codec_amp_stereo(codec, nid, HDA_OUTPUT, 0, HDA_AMP_MUTE, 0); @@ -4009,9 +3824,8 @@ static int alc_set_multi_io(struct hda_codec *codec, int idx, bool output) if (get_wcaps(codec, nid) & AC_WCAP_OUT_AMP) snd_hda_codec_amp_stereo(codec, nid, HDA_OUTPUT, 0, HDA_AMP_MUTE, HDA_AMP_MUTE); - snd_hda_codec_update_cache(codec, nid, 0, - AC_VERB_SET_PIN_WIDGET_CONTROL, - spec->multi_io[idx].ctl_in); + snd_hda_set_pin_ctl_cache(codec, nid, + spec->multi_io[idx].ctl_in); } return 0; } @@ -4084,7 +3898,7 @@ static void alc_remove_invalid_adc_nids(struct hda_codec *codec) nums = 0; for (n = 0; n < spec->num_adc_nids; n++) { hda_nid_t cap = spec->private_capsrc_nids[n]; - int num_conns = snd_hda_get_conn_list(codec, cap, NULL); + int num_conns = snd_hda_get_num_conns(codec, cap); for (i = 0; i < imux->num_items; i++) { hda_nid_t pin = spec->imux_pins[i]; if (pin) { @@ -4213,7 +4027,7 @@ static void select_or_unmute_capsrc(struct hda_codec *codec, hda_nid_t cap, if (get_wcaps_type(get_wcaps(codec, cap)) == AC_WID_AUD_MIX) { snd_hda_codec_amp_stereo(codec, cap, HDA_INPUT, idx, HDA_AMP_MUTE, 0); - } else if (snd_hda_get_conn_list(codec, cap, NULL) > 1) { + } else if (snd_hda_get_num_conns(codec, cap) > 1) { snd_hda_codec_write_cache(codec, cap, 0, AC_VERB_SET_CONNECT_SEL, idx); } @@ -4427,6 +4241,25 @@ static int alc_parse_auto_config(struct hda_codec *codec, return 1; } +/* common preparation job for alc_spec */ +static int alc_alloc_spec(struct hda_codec *codec, hda_nid_t mixer_nid) +{ + struct alc_spec *spec = kzalloc(sizeof(*spec), GFP_KERNEL); + int err; + + if (!spec) + return -ENOMEM; + codec->spec = spec; + spec->mixer_nid = mixer_nid; + + err = alc_codec_rename_from_preset(codec); + if (err < 0) { + kfree(spec); + return err; + } + return 0; +} + static int alc880_parse_auto_config(struct hda_codec *codec) { static const hda_nid_t alc880_ignore[] = { 0x1d, 0 }; @@ -4808,13 +4641,11 @@ static int patch_alc880(struct hda_codec *codec) struct alc_spec *spec; int err; - spec = kzalloc(sizeof(*spec), GFP_KERNEL); - if (spec == NULL) - return -ENOMEM; - - codec->spec = spec; + err = alc_alloc_spec(codec, 0x0b); + if (err < 0) + return err; - spec->mixer_nid = 0x0b; + spec = codec->spec; spec->need_dac_fix = 1; alc_pick_fixup(codec, alc880_fixup_models, alc880_fixup_tbl, @@ -4890,7 +4721,7 @@ static void alc260_fixup_gpio1_toggle(struct hda_codec *codec, spec->autocfg.hp_pins[0] = 0x0f; /* copy it for automute */ snd_hda_jack_detect_enable(codec, 0x0f, ALC_HP_EVENT); spec->unsol_event = alc_sku_unsol_event; - add_verb(codec->spec, alc_gpio1_init_verbs); + snd_hda_gen_add_verbs(&spec->gen, alc_gpio1_init_verbs); } } @@ -5001,13 +4832,11 @@ static int patch_alc260(struct hda_codec *codec) struct alc_spec *spec; int err; - spec = kzalloc(sizeof(*spec), GFP_KERNEL); - if (spec == NULL) - return -ENOMEM; - - codec->spec = spec; + err = alc_alloc_spec(codec, 0x07); + if (err < 0) + return err; - spec->mixer_nid = 0x07; + spec = codec->spec; alc_pick_fixup(codec, NULL, alc260_fixup_tbl, alc260_fixups); alc_apply_fixup(codec, ALC_FIXUP_ACT_PRE_PROBE); @@ -5171,8 +5000,7 @@ static void alc889_fixup_mbp_vref(struct hda_codec *codec, val = snd_hda_codec_read(codec, nids[i], 0, AC_VERB_GET_PIN_WIDGET_CONTROL, 0); val |= AC_PINCTL_VREF_80; - snd_hda_codec_write(codec, nids[i], 0, - AC_VERB_SET_PIN_WIDGET_CONTROL, val); + snd_hda_set_pin_ctl(codec, nids[i], val); spec->keep_vref_in_automute = 1; break; } @@ -5193,8 +5021,7 @@ static void alc889_fixup_imac91_vref(struct hda_codec *codec, val = snd_hda_codec_read(codec, nids[i], 0, AC_VERB_GET_PIN_WIDGET_CONTROL, 0); val |= AC_PINCTL_VREF_50; - snd_hda_codec_write(codec, nids[i], 0, - AC_VERB_SET_PIN_WIDGET_CONTROL, val); + snd_hda_set_pin_ctl(codec, nids[i], val); } spec->keep_vref_in_automute = 1; } @@ -5225,8 +5052,8 @@ static const struct alc_fixup alc882_fixups[] = { } }, [ALC882_FIXUP_ACER_ASPIRE_7736] = { - .type = ALC_FIXUP_SKU, - .v.sku = ALC_FIXUP_SKU_IGNORE, + .type = ALC_FIXUP_FUNC, + .v.func = alc_fixup_sku_ignore, }, [ALC882_FIXUP_ASUS_W90V] = { .type = ALC_FIXUP_PINS, @@ -5476,13 +5303,11 @@ static int patch_alc882(struct hda_codec *codec) struct alc_spec *spec; int err; - spec = kzalloc(sizeof(*spec), GFP_KERNEL); - if (spec == NULL) - return -ENOMEM; - - codec->spec = spec; + err = alc_alloc_spec(codec, 0x0b); + if (err < 0) + return err; - spec->mixer_nid = 0x0b; + spec = codec->spec; switch (codec->vendor_id) { case 0x10ec0882: @@ -5494,10 +5319,6 @@ static int patch_alc882(struct hda_codec *codec) break; } - err = alc_codec_rename_from_preset(codec); - if (err < 0) - goto error; - alc_pick_fixup(codec, alc882_fixup_models, alc882_fixup_tbl, alc882_fixups); alc_apply_fixup(codec, ALC_FIXUP_ACT_PRE_PROBE); @@ -5621,13 +5442,11 @@ static int patch_alc262(struct hda_codec *codec) struct alc_spec *spec; int err; - spec = kzalloc(sizeof(*spec), GFP_KERNEL); - if (spec == NULL) - return -ENOMEM; - - codec->spec = spec; + err = alc_alloc_spec(codec, 0x0b); + if (err < 0) + return err; - spec->mixer_nid = 0x0b; + spec = codec->spec; #if 0 /* pshou 07/11/05 set a zero PCM sample to DAC when FIFO is @@ -5710,7 +5529,7 @@ static int alc268_parse_auto_config(struct hda_codec *codec) if (err > 0) { if (!spec->no_analog && spec->autocfg.speaker_pins[0] != 0x1d) { add_mixer(spec, alc268_beep_mixer); - add_verb(spec, alc268_beep_init_verbs); + snd_hda_gen_add_verbs(&spec->gen, alc268_beep_init_verbs); } } return err; @@ -5723,13 +5542,12 @@ static int patch_alc268(struct hda_codec *codec) struct alc_spec *spec; int i, has_beep, err; - spec = kzalloc(sizeof(*spec), GFP_KERNEL); - if (spec == NULL) - return -ENOMEM; - - codec->spec = spec; - /* ALC268 has no aa-loopback mixer */ + err = alc_alloc_spec(codec, 0); + if (err < 0) + return err; + + spec = codec->spec; /* automatic parse from the BIOS config */ err = alc268_parse_auto_config(codec); @@ -5946,9 +5764,7 @@ static void alc269_fixup_mic2_mute_hook(void *private_data, int enabled) { struct hda_codec *codec = private_data; unsigned int pinval = enabled ? 0x20 : 0x24; - snd_hda_codec_update_cache(codec, 0x19, 0, - AC_VERB_SET_PIN_WIDGET_CONTROL, - pinval); + snd_hda_set_pin_ctl_cache(codec, 0x19, pinval); } static void alc269_fixup_mic2_mute(struct hda_codec *codec, @@ -6015,8 +5831,8 @@ static const struct alc_fixup alc269_fixups[] = { } }, [ALC269_FIXUP_SKU_IGNORE] = { - .type = ALC_FIXUP_SKU, - .v.sku = ALC_FIXUP_SKU_IGNORE, + .type = ALC_FIXUP_FUNC, + .v.func = alc_fixup_sku_ignore, }, [ALC269_FIXUP_ASUS_G73JW] = { .type = ALC_FIXUP_PINS, @@ -6242,19 +6058,13 @@ static void alc269_fill_coef(struct hda_codec *codec) static int patch_alc269(struct hda_codec *codec) { struct alc_spec *spec; - int err = 0; - - spec = kzalloc(sizeof(*spec), GFP_KERNEL); - if (spec == NULL) - return -ENOMEM; - - codec->spec = spec; - - spec->mixer_nid = 0x0b; + int err; - err = alc_codec_rename_from_preset(codec); + err = alc_alloc_spec(codec, 0x0b); if (err < 0) - goto error; + return err; + + spec = codec->spec; if (codec->vendor_id == 0x10ec0269) { spec->codec_variant = ALC269_TYPE_ALC269VA; @@ -6346,8 +6156,7 @@ static void alc861_fixup_asus_amp_vref_0f(struct hda_codec *codec, if (!(val & (AC_PINCTL_IN_EN | AC_PINCTL_OUT_EN))) val |= AC_PINCTL_IN_EN; val |= AC_PINCTL_VREF_50; - snd_hda_codec_write(codec, 0x0f, 0, - AC_VERB_SET_PIN_WIDGET_CONTROL, val); + snd_hda_set_pin_ctl(codec, 0x0f, val); spec->keep_vref_in_automute = 1; } @@ -6401,13 +6210,11 @@ static int patch_alc861(struct hda_codec *codec) struct alc_spec *spec; int err; - spec = kzalloc(sizeof(*spec), GFP_KERNEL); - if (spec == NULL) - return -ENOMEM; - - codec->spec = spec; + err = alc_alloc_spec(codec, 0x15); + if (err < 0) + return err; - spec->mixer_nid = 0x15; + spec = codec->spec; alc_pick_fixup(codec, NULL, alc861_fixup_tbl, alc861_fixups); alc_apply_fixup(codec, ALC_FIXUP_ACT_PRE_PROBE); @@ -6504,13 +6311,11 @@ static int patch_alc861vd(struct hda_codec *codec) struct alc_spec *spec; int err; - spec = kzalloc(sizeof(*spec), GFP_KERNEL); - if (spec == NULL) - return -ENOMEM; - - codec->spec = spec; + err = alc_alloc_spec(codec, 0x0b); + if (err < 0) + return err; - spec->mixer_nid = 0x0b; + spec = codec->spec; alc_pick_fixup(codec, NULL, alc861vd_fixup_tbl, alc861vd_fixups); alc_apply_fixup(codec, ALC_FIXUP_ACT_PRE_PROBE); @@ -6522,7 +6327,7 @@ static int patch_alc861vd(struct hda_codec *codec) if (codec->vendor_id == 0x10ec0660) { /* always turn on EAPD */ - add_verb(spec, alc660vd_eapd_verbs); + snd_hda_gen_add_verbs(&spec->gen, alc660vd_eapd_verbs); } if (!spec->no_analog) { @@ -6635,8 +6440,8 @@ static const struct alc_fixup alc662_fixups[] = { } }, [ALC662_FIXUP_SKU_IGNORE] = { - .type = ALC_FIXUP_SKU, - .v.sku = ALC_FIXUP_SKU_IGNORE, + .type = ALC_FIXUP_FUNC, + .v.func = alc_fixup_sku_ignore, }, [ALC662_FIXUP_HP_RP5800] = { .type = ALC_FIXUP_PINS, @@ -6849,25 +6654,19 @@ static const struct alc_model_fixup alc662_fixup_models[] = { static int patch_alc662(struct hda_codec *codec) { struct alc_spec *spec; - int err = 0; - - spec = kzalloc(sizeof(*spec), GFP_KERNEL); - if (!spec) - return -ENOMEM; + int err; - codec->spec = spec; + err = alc_alloc_spec(codec, 0x0b); + if (err < 0) + return err; - spec->mixer_nid = 0x0b; + spec = codec->spec; /* handle multiple HPs as is */ spec->parse_flags = HDA_PINCFG_NO_HP_FIXUP; alc_fix_pll_init(codec, 0x20, 0x04, 15); - err = alc_codec_rename_from_preset(codec); - if (err < 0) - goto error; - if ((alc_get_coef0(codec) & (1 << 14)) && codec->bus->pci->subsystem_vendor == 0x1025 && spec->cdefine.platform_type == 1) { @@ -6930,16 +6729,12 @@ static int alc680_parse_auto_config(struct hda_codec *codec) */ static int patch_alc680(struct hda_codec *codec) { - struct alc_spec *spec; int err; - spec = kzalloc(sizeof(*spec), GFP_KERNEL); - if (spec == NULL) - return -ENOMEM; - - codec->spec = spec; - /* ALC680 has no aa-loopback mixer */ + err = alc_alloc_spec(codec, 0); + if (err < 0) + return err; /* automatic parse from the BIOS config */ err = alc680_parse_auto_config(codec); diff --git a/sound/pci/hda/patch_sigmatel.c b/sound/pci/hda/patch_sigmatel.c index 2cb1e08f962..7db8228f1b8 100644 --- a/sound/pci/hda/patch_sigmatel.c +++ b/sound/pci/hda/patch_sigmatel.c @@ -36,6 +36,7 @@ #include <sound/tlv.h> #include "hda_codec.h" #include "hda_local.h" +#include "hda_auto_parser.h" #include "hda_beep.h" #include "hda_jack.h" @@ -221,6 +222,7 @@ struct sigmatel_spec { unsigned char aloopback_shift; /* power management */ + unsigned int power_map_bits; unsigned int num_pwrs; const hda_nid_t *pwr_nids; const hda_nid_t *dac_list; @@ -314,6 +316,9 @@ struct sigmatel_spec { struct hda_vmaster_mute_hook vmaster_mute; }; +#define AC_VERB_IDT_SET_POWER_MAP 0x7ec +#define AC_VERB_IDT_GET_POWER_MAP 0xfec + static const hda_nid_t stac9200_adc_nids[1] = { 0x03, }; @@ -681,8 +686,7 @@ static int stac_vrefout_set(struct hda_codec *codec, pinctl &= ~AC_PINCTL_VREFEN; pinctl |= (new_vref & AC_PINCTL_VREFEN); - error = snd_hda_codec_write_cache(codec, nid, 0, - AC_VERB_SET_PIN_WIDGET_CONTROL, pinctl); + error = snd_hda_set_pin_ctl_cache(codec, nid, pinctl); if (error < 0) return error; @@ -706,8 +710,7 @@ static unsigned int stac92xx_vref_set(struct hda_codec *codec, else pincfg |= AC_PINCTL_IN_EN; - error = snd_hda_codec_write_cache(codec, nid, 0, - AC_VERB_SET_PIN_WIDGET_CONTROL, pincfg); + error = snd_hda_set_pin_ctl_cache(codec, nid, pincfg); if (error < 0) return error; else @@ -2505,27 +2508,10 @@ static int stac92xx_build_pcms(struct hda_codec *codec) return 0; } -static unsigned int stac92xx_get_default_vref(struct hda_codec *codec, - hda_nid_t nid) -{ - unsigned int pincap = snd_hda_query_pin_caps(codec, nid); - pincap = (pincap & AC_PINCAP_VREF) >> AC_PINCAP_VREF_SHIFT; - if (pincap & AC_PINCAP_VREF_100) - return AC_PINCTL_VREF_100; - if (pincap & AC_PINCAP_VREF_80) - return AC_PINCTL_VREF_80; - if (pincap & AC_PINCAP_VREF_50) - return AC_PINCTL_VREF_50; - if (pincap & AC_PINCAP_VREF_GRD) - return AC_PINCTL_VREF_GRD; - return 0; -} - static void stac92xx_auto_set_pinctl(struct hda_codec *codec, hda_nid_t nid, int pin_type) { - snd_hda_codec_write_cache(codec, nid, 0, - AC_VERB_SET_PIN_WIDGET_CONTROL, pin_type); + snd_hda_set_pin_ctl_cache(codec, nid, pin_type); } #define stac92xx_hp_switch_info snd_ctl_boolean_mono_info @@ -2594,7 +2580,7 @@ static int stac92xx_dc_bias_get(struct snd_kcontrol *kcontrol, hda_nid_t nid = kcontrol->private_value; unsigned int vref = stac92xx_vref_get(codec, nid); - if (vref == stac92xx_get_default_vref(codec, nid)) + if (vref == snd_hda_get_default_vref(codec, nid)) ucontrol->value.enumerated.item[0] = 0; else if (vref == AC_PINCTL_VREF_GRD) ucontrol->value.enumerated.item[0] = 1; @@ -2613,7 +2599,7 @@ static int stac92xx_dc_bias_put(struct snd_kcontrol *kcontrol, hda_nid_t nid = kcontrol->private_value; if (ucontrol->value.enumerated.item[0] == 0) - new_vref = stac92xx_get_default_vref(codec, nid); + new_vref = snd_hda_get_default_vref(codec, nid); else if (ucontrol->value.enumerated.item[0] == 1) new_vref = AC_PINCTL_VREF_GRD; else if (ucontrol->value.enumerated.item[0] == 2) @@ -2679,7 +2665,7 @@ static int stac92xx_io_switch_put(struct snd_kcontrol *kcontrol, struct snd_ctl_ else { unsigned int pinctl = AC_PINCTL_IN_EN; if (io_idx) /* set VREF for mic */ - pinctl |= stac92xx_get_default_vref(codec, nid); + pinctl |= snd_hda_get_default_vref(codec, nid); stac92xx_auto_set_pinctl(codec, nid, pinctl); } @@ -2847,7 +2833,7 @@ static inline int stac92xx_add_jack_mode_control(struct hda_codec *codec, char name[22]; if (snd_hda_get_input_pin_attr(def_conf) != INPUT_PIN_ATTR_INT) { - if (stac92xx_get_default_vref(codec, nid) == AC_PINCTL_VREF_GRD + if (snd_hda_get_default_vref(codec, nid) == AC_PINCTL_VREF_GRD && nid == spec->line_switch) control = STAC_CTL_WIDGET_IO_SWITCH; else if (snd_hda_query_pin_caps(codec, nid) @@ -4250,13 +4236,6 @@ static void stac_store_hints(struct hda_codec *codec) val = snd_hda_get_bool_hint(codec, "eapd_switch"); if (val >= 0) spec->eapd_switch = val; - get_int_hint(codec, "gpio_led_polarity", &spec->gpio_led_polarity); - if (get_int_hint(codec, "gpio_led", &spec->gpio_led)) { - spec->gpio_mask |= spec->gpio_led; - spec->gpio_dir |= spec->gpio_led; - if (spec->gpio_led_polarity) - spec->gpio_data |= spec->gpio_led; - } } static void stac_issue_unsol_events(struct hda_codec *codec, int num_pins, @@ -4354,7 +4333,7 @@ static int stac92xx_init(struct hda_codec *codec) unsigned int pinctl, conf; if (type == AUTO_PIN_MIC) { /* for mic pins, force to initialize */ - pinctl = stac92xx_get_default_vref(codec, nid); + pinctl = snd_hda_get_default_vref(codec, nid); pinctl |= AC_PINCTL_IN_EN; stac92xx_auto_set_pinctl(codec, nid, pinctl); } else { @@ -4390,10 +4369,18 @@ static int stac92xx_init(struct hda_codec *codec) hda_nid_t nid = spec->pwr_nids[i]; int pinctl, def_conf; + def_conf = snd_hda_codec_get_pincfg(codec, nid); + def_conf = get_defcfg_connect(def_conf); + if (def_conf == AC_JACK_PORT_NONE) { + /* power off unused ports */ + stac_toggle_power_map(codec, nid, 0); + continue; + } /* power on when no jack detection is available */ /* or when the VREF is used for controlling LED */ if (!spec->hp_detect || - spec->vref_mute_led_nid == nid) { + spec->vref_mute_led_nid == nid || + !is_jack_detectable(codec, nid)) { stac_toggle_power_map(codec, nid, 1); continue; } @@ -4411,15 +4398,6 @@ static int stac92xx_init(struct hda_codec *codec) stac_toggle_power_map(codec, nid, 1); continue; } - def_conf = snd_hda_codec_get_pincfg(codec, nid); - def_conf = get_defcfg_connect(def_conf); - /* skip any ports that don't have jacks since presence - * detection is useless */ - if (def_conf != AC_JACK_PORT_NONE && - !is_jack_detectable(codec, nid)) { - stac_toggle_power_map(codec, nid, 1); - continue; - } if (enable_pin_detect(codec, nid, STAC_PWR_EVENT)) { stac_issue_unsol_event(codec, nid); continue; @@ -4432,6 +4410,12 @@ static int stac92xx_init(struct hda_codec *codec) /* sync mute LED */ snd_hda_sync_vmaster_hook(&spec->vmaster_mute); + + /* sync the power-map */ + if (spec->num_pwrs) + snd_hda_codec_write(codec, codec->afg, 0, + AC_VERB_IDT_SET_POWER_MAP, + spec->power_map_bits); if (spec->dac_list) stac92xx_power_down(codec); return 0; @@ -4460,8 +4444,7 @@ static void stac92xx_shutup_pins(struct hda_codec *codec) struct hda_pincfg *pin = snd_array_elem(&codec->init_pins, i); def_conf = snd_hda_codec_get_pincfg(codec, pin->nid); if (get_defcfg_connect(def_conf) != AC_JACK_PORT_NONE) - snd_hda_codec_write(codec, pin->nid, 0, - AC_VERB_SET_PIN_WIDGET_CONTROL, 0); + snd_hda_set_pin_ctl(codec, pin->nid, 0); } } @@ -4517,9 +4500,7 @@ static void stac92xx_set_pinctl(struct hda_codec *codec, hda_nid_t nid, pin_ctl |= flag; if (old_ctl != pin_ctl) - snd_hda_codec_write_cache(codec, nid, 0, - AC_VERB_SET_PIN_WIDGET_CONTROL, - pin_ctl); + snd_hda_set_pin_ctl_cache(codec, nid, pin_ctl); } static void stac92xx_reset_pinctl(struct hda_codec *codec, hda_nid_t nid, @@ -4528,9 +4509,7 @@ static void stac92xx_reset_pinctl(struct hda_codec *codec, hda_nid_t nid, unsigned int pin_ctl = snd_hda_codec_read(codec, nid, 0, AC_VERB_GET_PIN_WIDGET_CONTROL, 0x00); if (pin_ctl & flag) - snd_hda_codec_write_cache(codec, nid, 0, - AC_VERB_SET_PIN_WIDGET_CONTROL, - pin_ctl & ~flag); + snd_hda_set_pin_ctl_cache(codec, nid, pin_ctl & ~flag); } static inline int get_pin_presence(struct hda_codec *codec, hda_nid_t nid) @@ -4682,14 +4661,18 @@ static void stac_toggle_power_map(struct hda_codec *codec, hda_nid_t nid, idx = 1 << idx; - val = snd_hda_codec_read(codec, codec->afg, 0, 0x0fec, 0x0) & 0xff; + val = spec->power_map_bits; if (enable) val &= ~idx; else val |= idx; /* power down unused output ports */ - snd_hda_codec_write(codec, codec->afg, 0, 0x7ec, val); + if (val != spec->power_map_bits) { + spec->power_map_bits = val; + snd_hda_codec_write(codec, codec->afg, 0, + AC_VERB_IDT_SET_POWER_MAP, val); + } } static void stac92xx_pin_sense(struct hda_codec *codec, hda_nid_t nid) @@ -4866,6 +4849,11 @@ static int find_mute_led_cfg(struct hda_codec *codec, int default_polarity) struct sigmatel_spec *spec = codec->spec; const struct dmi_device *dev = NULL; + if (get_int_hint(codec, "gpio_led", &spec->gpio_led)) { + get_int_hint(codec, "gpio_led_polarity", + &spec->gpio_led_polarity); + return 1; + } if ((codec->subsystem_id >> 16) == PCI_VENDOR_ID_HP) { while ((dev = dmi_find_device(DMI_DEV_TYPE_OEM_STRING, NULL, dev))) { @@ -4952,7 +4940,8 @@ static void stac92hd_proc_hook(struct snd_info_buffer *buffer, { if (nid == codec->afg) snd_iprintf(buffer, "Power-Map: 0x%02x\n", - snd_hda_codec_read(codec, nid, 0, 0x0fec, 0x0)); + snd_hda_codec_read(codec, nid, 0, + AC_VERB_IDT_GET_POWER_MAP, 0)); } static void analog_loop_proc_hook(struct snd_info_buffer *buffer, @@ -5009,20 +4998,6 @@ static int stac92xx_suspend(struct hda_codec *codec, pm_message_t state) return 0; } -static int stac92xx_pre_resume(struct hda_codec *codec) -{ - struct sigmatel_spec *spec = codec->spec; - - /* sync mute LED */ - if (spec->vref_mute_led_nid) - stac_vrefout_set(codec, spec->vref_mute_led_nid, - spec->vref_led); - else if (spec->gpio_led) - stac_gpio_set(codec, spec->gpio_mask, - spec->gpio_dir, spec->gpio_data); - return 0; -} - static void stac92xx_set_power_state(struct hda_codec *codec, hda_nid_t fg, unsigned int power_state) { @@ -5046,7 +5021,6 @@ static void stac92xx_set_power_state(struct hda_codec *codec, hda_nid_t fg, #else #define stac92xx_suspend NULL #define stac92xx_resume NULL -#define stac92xx_pre_resume NULL #define stac92xx_set_power_state NULL #endif /* CONFIG_PM */ @@ -5592,9 +5566,6 @@ again: codec->patch_ops.set_power_state = stac92xx_set_power_state; } -#ifdef CONFIG_PM - codec->patch_ops.pre_resume = stac92xx_pre_resume; -#endif } err = stac92xx_parse_auto_config(codec); @@ -5901,9 +5872,6 @@ again: codec->patch_ops.set_power_state = stac92xx_set_power_state; } -#ifdef CONFIG_PM - codec->patch_ops.pre_resume = stac92xx_pre_resume; -#endif } spec->multiout.dac_nids = spec->dac_nids; diff --git a/sound/pci/hda/patch_via.c b/sound/pci/hda/patch_via.c index 06214fdc948..82b368068e0 100644 --- a/sound/pci/hda/patch_via.c +++ b/sound/pci/hda/patch_via.c @@ -54,6 +54,7 @@ #include <sound/asoundef.h> #include "hda_codec.h" #include "hda_local.h" +#include "hda_auto_parser.h" #include "hda_jack.h" /* Pin Widget NID */ @@ -484,7 +485,7 @@ static void activate_output_mix(struct hda_codec *codec, struct nid_path *path, if (!path) return; - num = snd_hda_get_conn_list(codec, mix_nid, NULL); + num = snd_hda_get_num_conns(codec, mix_nid); for (i = 0; i < num; i++) { if (i == idx) val = AMP_IN_UNMUTE(i); @@ -532,8 +533,7 @@ static void init_output_pin(struct hda_codec *codec, hda_nid_t pin, { if (!pin) return; - snd_hda_codec_write(codec, pin, 0, AC_VERB_SET_PIN_WIDGET_CONTROL, - pin_type); + snd_hda_set_pin_ctl(codec, pin, pin_type); if (snd_hda_query_pin_caps(codec, pin) & AC_PINCAP_EAPD) snd_hda_codec_write(codec, pin, 0, AC_VERB_SET_EAPD_BTLENABLE, 0x02); @@ -662,12 +662,12 @@ static void via_auto_init_analog_input(struct hda_codec *codec) hda_nid_t nid = cfg->inputs[i].pin; if (spec->smart51_enabled && is_smart51_pins(codec, nid)) ctl = PIN_OUT; - else if (cfg->inputs[i].type == AUTO_PIN_MIC) - ctl = PIN_VREF50; - else + else { ctl = PIN_IN; - snd_hda_codec_write(codec, nid, 0, - AC_VERB_SET_PIN_WIDGET_CONTROL, ctl); + if (cfg->inputs[i].type == AUTO_PIN_MIC) + ctl |= snd_hda_get_default_vref(codec, nid); + } + snd_hda_set_pin_ctl(codec, nid, ctl); } /* init input-src */ @@ -1006,9 +1006,7 @@ static int via_smart51_put(struct snd_kcontrol *kcontrol, AC_VERB_GET_PIN_WIDGET_CONTROL, 0); parm &= ~(AC_PINCTL_IN_EN | AC_PINCTL_OUT_EN); parm |= out_in; - snd_hda_codec_write(codec, nid, 0, - AC_VERB_SET_PIN_WIDGET_CONTROL, - parm); + snd_hda_set_pin_ctl(codec, nid, parm); if (out_in == AC_PINCTL_OUT_EN) { mute_aa_path(codec, 1); notify_aa_path_ctls(codec); @@ -1647,8 +1645,7 @@ static void toggle_output_mutes(struct hda_codec *codec, int num_pins, parm &= ~AC_PINCTL_OUT_EN; else parm |= AC_PINCTL_OUT_EN; - snd_hda_codec_write(codec, pins[i], 0, - AC_VERB_SET_PIN_WIDGET_CONTROL, parm); + snd_hda_set_pin_ctl(codec, pins[i], parm); } } @@ -1709,8 +1706,7 @@ static void via_gpio_control(struct hda_codec *codec) if (gpio_data == 0x02) { /* unmute line out */ - snd_hda_codec_write(codec, spec->autocfg.line_out_pins[0], 0, - AC_VERB_SET_PIN_WIDGET_CONTROL, + snd_hda_set_pin_ctl(codec, spec->autocfg.line_out_pins[0], PIN_OUT); if (vol_counter & 0x20) { /* decrease volume */ @@ -1728,9 +1724,7 @@ static void via_gpio_control(struct hda_codec *codec) } } else if (!(gpio_data & 0x02)) { /* mute line out */ - snd_hda_codec_write(codec, spec->autocfg.line_out_pins[0], 0, - AC_VERB_SET_PIN_WIDGET_CONTROL, - 0); + snd_hda_set_pin_ctl(codec, spec->autocfg.line_out_pins[0], 0); } } @@ -2757,8 +2751,7 @@ static void via_auto_init_dig_in(struct hda_codec *codec) struct via_spec *spec = codec->spec; if (!spec->dig_in_nid) return; - snd_hda_codec_write(codec, spec->autocfg.dig_in_pin, 0, - AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN); + snd_hda_set_pin_ctl(codec, spec->autocfg.dig_in_pin, PIN_IN); } /* initialize the unsolicited events */ diff --git a/sound/pci/ice1712/ice1712.c b/sound/pci/ice1712/ice1712.c index 132a86e09d0..5be2e120a14 100644 --- a/sound/pci/ice1712/ice1712.c +++ b/sound/pci/ice1712/ice1712.c @@ -2803,22 +2803,11 @@ static void __devexit snd_ice1712_remove(struct pci_dev *pci) pci_set_drvdata(pci, NULL); } -static struct pci_driver driver = { +static struct pci_driver ice1712_driver = { .name = KBUILD_MODNAME, .id_table = snd_ice1712_ids, .probe = snd_ice1712_probe, .remove = __devexit_p(snd_ice1712_remove), }; -static int __init alsa_card_ice1712_init(void) -{ - return pci_register_driver(&driver); -} - -static void __exit alsa_card_ice1712_exit(void) -{ - pci_unregister_driver(&driver); -} - -module_init(alsa_card_ice1712_init) -module_exit(alsa_card_ice1712_exit) +module_pci_driver(ice1712_driver); diff --git a/sound/pci/ice1712/ice1724.c b/sound/pci/ice1712/ice1724.c index 812d10e43ae..a01a00d1cf4 100644 --- a/sound/pci/ice1712/ice1724.c +++ b/sound/pci/ice1712/ice1724.c @@ -2873,7 +2873,7 @@ static int snd_vt1724_resume(struct pci_dev *pci) } #endif -static struct pci_driver driver = { +static struct pci_driver vt1724_driver = { .name = KBUILD_MODNAME, .id_table = snd_vt1724_ids, .probe = snd_vt1724_probe, @@ -2884,15 +2884,4 @@ static struct pci_driver driver = { #endif }; -static int __init alsa_card_ice1724_init(void) -{ - return pci_register_driver(&driver); -} - -static void __exit alsa_card_ice1724_exit(void) -{ - pci_unregister_driver(&driver); -} - -module_init(alsa_card_ice1724_init) -module_exit(alsa_card_ice1724_exit) +module_pci_driver(vt1724_driver); diff --git a/sound/pci/intel8x0.c b/sound/pci/intel8x0.c index e0a4263baa2..f4e2dd4da8c 100644 --- a/sound/pci/intel8x0.c +++ b/sound/pci/intel8x0.c @@ -3338,7 +3338,7 @@ static void __devexit snd_intel8x0_remove(struct pci_dev *pci) pci_set_drvdata(pci, NULL); } -static struct pci_driver driver = { +static struct pci_driver intel8x0_driver = { .name = KBUILD_MODNAME, .id_table = snd_intel8x0_ids, .probe = snd_intel8x0_probe, @@ -3349,16 +3349,4 @@ static struct pci_driver driver = { #endif }; - -static int __init alsa_card_intel8x0_init(void) -{ - return pci_register_driver(&driver); -} - -static void __exit alsa_card_intel8x0_exit(void) -{ - pci_unregister_driver(&driver); -} - -module_init(alsa_card_intel8x0_init) -module_exit(alsa_card_intel8x0_exit) +module_pci_driver(intel8x0_driver); diff --git a/sound/pci/intel8x0m.c b/sound/pci/intel8x0m.c index d689913a61b..fc27a6a69e7 100644 --- a/sound/pci/intel8x0m.c +++ b/sound/pci/intel8x0m.c @@ -1324,7 +1324,7 @@ static void __devexit snd_intel8x0m_remove(struct pci_dev *pci) pci_set_drvdata(pci, NULL); } -static struct pci_driver driver = { +static struct pci_driver intel8x0m_driver = { .name = KBUILD_MODNAME, .id_table = snd_intel8x0m_ids, .probe = snd_intel8x0m_probe, @@ -1335,16 +1335,4 @@ static struct pci_driver driver = { #endif }; - -static int __init alsa_card_intel8x0m_init(void) -{ - return pci_register_driver(&driver); -} - -static void __exit alsa_card_intel8x0m_exit(void) -{ - pci_unregister_driver(&driver); -} - -module_init(alsa_card_intel8x0m_init) -module_exit(alsa_card_intel8x0m_exit) +module_pci_driver(intel8x0m_driver); diff --git a/sound/pci/korg1212/korg1212.c b/sound/pci/korg1212/korg1212.c index 8fea45ab588..e69ce5f9c31 100644 --- a/sound/pci/korg1212/korg1212.c +++ b/sound/pci/korg1212/korg1212.c @@ -2476,22 +2476,11 @@ static void __devexit snd_korg1212_remove(struct pci_dev *pci) pci_set_drvdata(pci, NULL); } -static struct pci_driver driver = { +static struct pci_driver korg1212_driver = { .name = KBUILD_MODNAME, .id_table = snd_korg1212_ids, .probe = snd_korg1212_probe, .remove = __devexit_p(snd_korg1212_remove), }; -static int __init alsa_card_korg1212_init(void) -{ - return pci_register_driver(&driver); -} - -static void __exit alsa_card_korg1212_exit(void) -{ - pci_unregister_driver(&driver); -} - -module_init(alsa_card_korg1212_init) -module_exit(alsa_card_korg1212_exit) +module_pci_driver(korg1212_driver); diff --git a/sound/pci/lola/lola.c b/sound/pci/lola/lola.c index 37598273685..ac15166bee6 100644 --- a/sound/pci/lola/lola.c +++ b/sound/pci/lola/lola.c @@ -770,22 +770,11 @@ static DEFINE_PCI_DEVICE_TABLE(lola_ids) = { MODULE_DEVICE_TABLE(pci, lola_ids); /* pci_driver definition */ -static struct pci_driver driver = { +static struct pci_driver lola_driver = { .name = KBUILD_MODNAME, .id_table = lola_ids, .probe = lola_probe, .remove = __devexit_p(lola_remove), }; -static int __init alsa_card_lola_init(void) -{ - return pci_register_driver(&driver); -} - -static void __exit alsa_card_lola_exit(void) -{ - pci_unregister_driver(&driver); -} - -module_init(alsa_card_lola_init) -module_exit(alsa_card_lola_exit) +module_pci_driver(lola_driver); diff --git a/sound/pci/lx6464es/lx6464es.c b/sound/pci/lx6464es/lx6464es.c index d94c0c292bd..d1ab4370673 100644 --- a/sound/pci/lx6464es/lx6464es.c +++ b/sound/pci/lx6464es/lx6464es.c @@ -1141,24 +1141,11 @@ static void __devexit snd_lx6464es_remove(struct pci_dev *pci) } -static struct pci_driver driver = { +static struct pci_driver lx6464es_driver = { .name = KBUILD_MODNAME, .id_table = snd_lx6464es_ids, .probe = snd_lx6464es_probe, .remove = __devexit_p(snd_lx6464es_remove), }; - -/* module initialization */ -static int __init mod_init(void) -{ - return pci_register_driver(&driver); -} - -static void __exit mod_exit(void) -{ - pci_unregister_driver(&driver); -} - -module_init(mod_init); -module_exit(mod_exit); +module_pci_driver(lx6464es_driver); diff --git a/sound/pci/maestro3.c b/sound/pci/maestro3.c index 78229b0dad2..deef2139958 100644 --- a/sound/pci/maestro3.c +++ b/sound/pci/maestro3.c @@ -2837,7 +2837,7 @@ static void __devexit snd_m3_remove(struct pci_dev *pci) pci_set_drvdata(pci, NULL); } -static struct pci_driver driver = { +static struct pci_driver m3_driver = { .name = KBUILD_MODNAME, .id_table = snd_m3_ids, .probe = snd_m3_probe, @@ -2848,15 +2848,4 @@ static struct pci_driver driver = { #endif }; -static int __init alsa_card_m3_init(void) -{ - return pci_register_driver(&driver); -} - -static void __exit alsa_card_m3_exit(void) -{ - pci_unregister_driver(&driver); -} - -module_init(alsa_card_m3_init) -module_exit(alsa_card_m3_exit) +module_pci_driver(m3_driver); diff --git a/sound/pci/mixart/mixart.c b/sound/pci/mixart/mixart.c index 487837c01c9..0762610c99c 100644 --- a/sound/pci/mixart/mixart.c +++ b/sound/pci/mixart/mixart.c @@ -1380,22 +1380,11 @@ static void __devexit snd_mixart_remove(struct pci_dev *pci) pci_set_drvdata(pci, NULL); } -static struct pci_driver driver = { +static struct pci_driver mixart_driver = { .name = KBUILD_MODNAME, .id_table = snd_mixart_ids, .probe = snd_mixart_probe, .remove = __devexit_p(snd_mixart_remove), }; -static int __init alsa_card_mixart_init(void) -{ - return pci_register_driver(&driver); -} - -static void __exit alsa_card_mixart_exit(void) -{ - pci_unregister_driver(&driver); -} - -module_init(alsa_card_mixart_init) -module_exit(alsa_card_mixart_exit) +module_pci_driver(mixart_driver); diff --git a/sound/pci/nm256/nm256.c b/sound/pci/nm256/nm256.c index ade2c64bd60..8159b05ee94 100644 --- a/sound/pci/nm256/nm256.c +++ b/sound/pci/nm256/nm256.c @@ -1742,7 +1742,7 @@ static void __devexit snd_nm256_remove(struct pci_dev *pci) } -static struct pci_driver driver = { +static struct pci_driver nm256_driver = { .name = KBUILD_MODNAME, .id_table = snd_nm256_ids, .probe = snd_nm256_probe, @@ -1753,16 +1753,4 @@ static struct pci_driver driver = { #endif }; - -static int __init alsa_card_nm256_init(void) -{ - return pci_register_driver(&driver); -} - -static void __exit alsa_card_nm256_exit(void) -{ - pci_unregister_driver(&driver); -} - -module_init(alsa_card_nm256_init) -module_exit(alsa_card_nm256_exit) +module_pci_driver(nm256_driver); diff --git a/sound/pci/oxygen/oxygen.c b/sound/pci/oxygen/oxygen.c index eab663eef11..610275bfbae 100644 --- a/sound/pci/oxygen/oxygen.c +++ b/sound/pci/oxygen/oxygen.c @@ -94,6 +94,7 @@ enum { MODEL_2CH_OUTPUT, MODEL_HG2PCI, MODEL_XONAR_DG, + MODEL_XONAR_DGX, }; static DEFINE_PCI_DEVICE_TABLE(oxygen_ids) = { @@ -109,6 +110,8 @@ static DEFINE_PCI_DEVICE_TABLE(oxygen_ids) = { { OXYGEN_PCI_SUBID(0x1a58, 0x0910), .driver_data = MODEL_CMEDIA_REF }, /* Asus Xonar DG */ { OXYGEN_PCI_SUBID(0x1043, 0x8467), .driver_data = MODEL_XONAR_DG }, + /* Asus Xonar DGX */ + { OXYGEN_PCI_SUBID(0x1043, 0x8521), .driver_data = MODEL_XONAR_DGX }, /* PCI 2.0 HD Audio */ { OXYGEN_PCI_SUBID(0x13f6, 0x8782), .driver_data = MODEL_2CH_OUTPUT }, /* Kuroutoshikou CMI8787-HG2PCI */ @@ -827,6 +830,11 @@ static int __devinit get_oxygen_model(struct oxygen *chip, break; case MODEL_XONAR_DG: chip->model = model_xonar_dg; + chip->model.shortname = "Xonar DG"; + break; + case MODEL_XONAR_DGX: + chip->model = model_xonar_dg; + chip->model.shortname = "Xonar DGX"; break; } if (id->driver_data == MODEL_MERIDIAN || @@ -870,15 +878,4 @@ static struct pci_driver oxygen_driver = { #endif }; -static int __init alsa_card_oxygen_init(void) -{ - return pci_register_driver(&oxygen_driver); -} - -static void __exit alsa_card_oxygen_exit(void) -{ - pci_unregister_driver(&oxygen_driver); -} - -module_init(alsa_card_oxygen_init) -module_exit(alsa_card_oxygen_exit) +module_pci_driver(oxygen_driver); diff --git a/sound/pci/oxygen/virtuoso.c b/sound/pci/oxygen/virtuoso.c index 3fdee495017..19962c6d38c 100644 --- a/sound/pci/oxygen/virtuoso.c +++ b/sound/pci/oxygen/virtuoso.c @@ -100,15 +100,4 @@ static struct pci_driver xonar_driver = { .shutdown = oxygen_pci_shutdown, }; -static int __init alsa_card_xonar_init(void) -{ - return pci_register_driver(&xonar_driver); -} - -static void __exit alsa_card_xonar_exit(void) -{ - pci_unregister_driver(&xonar_driver); -} - -module_init(alsa_card_xonar_init) -module_exit(alsa_card_xonar_exit) +module_pci_driver(xonar_driver); diff --git a/sound/pci/oxygen/xonar_dg.c b/sound/pci/oxygen/xonar_dg.c index 793bdf03d7e..77acd790ea4 100644 --- a/sound/pci/oxygen/xonar_dg.c +++ b/sound/pci/oxygen/xonar_dg.c @@ -1,5 +1,5 @@ /* - * card driver for the Xonar DG + * card driver for the Xonar DG/DGX * * Copyright (c) Clemens Ladisch <clemens@ladisch.de> * @@ -17,8 +17,8 @@ */ /* - * Xonar DG - * -------- + * Xonar DG/DGX + * ------------ * * CMI8788: * @@ -581,7 +581,6 @@ static void dump_cs4245_registers(struct oxygen *chip, } struct oxygen_model model_xonar_dg = { - .shortname = "Xonar DG", .longname = "C-Media Oxygen HD Audio", .chip = "CMI8786", .init = dg_init, diff --git a/sound/pci/pcxhr/pcxhr.c b/sound/pci/pcxhr/pcxhr.c index fd1809ab73b..0435f45e951 100644 --- a/sound/pci/pcxhr/pcxhr.c +++ b/sound/pci/pcxhr/pcxhr.c @@ -1607,22 +1607,11 @@ static void __devexit pcxhr_remove(struct pci_dev *pci) pci_set_drvdata(pci, NULL); } -static struct pci_driver driver = { +static struct pci_driver pcxhr_driver = { .name = KBUILD_MODNAME, .id_table = pcxhr_ids, .probe = pcxhr_probe, .remove = __devexit_p(pcxhr_remove), }; -static int __init pcxhr_module_init(void) -{ - return pci_register_driver(&driver); -} - -static void __exit pcxhr_module_exit(void) -{ - pci_unregister_driver(&driver); -} - -module_init(pcxhr_module_init) -module_exit(pcxhr_module_exit) +module_pci_driver(pcxhr_driver); diff --git a/sound/pci/riptide/riptide.c b/sound/pci/riptide/riptide.c index 0481d94aac9..cbeb3f77350 100644 --- a/sound/pci/riptide/riptide.c +++ b/sound/pci/riptide/riptide.c @@ -1837,8 +1837,7 @@ static int snd_riptide_free(struct snd_riptide *chip) } if (chip->irq >= 0) free_irq(chip->irq, chip); - if (chip->fw_entry) - release_firmware(chip->fw_entry); + release_firmware(chip->fw_entry); release_and_free_resource(chip->res_port); kfree(chip); return 0; diff --git a/sound/pci/rme32.c b/sound/pci/rme32.c index b4819d5e41d..46b3629dda2 100644 --- a/sound/pci/rme32.c +++ b/sound/pci/rme32.c @@ -1984,22 +1984,11 @@ static void __devexit snd_rme32_remove(struct pci_dev *pci) pci_set_drvdata(pci, NULL); } -static struct pci_driver driver = { +static struct pci_driver rme32_driver = { .name = KBUILD_MODNAME, .id_table = snd_rme32_ids, .probe = snd_rme32_probe, .remove = __devexit_p(snd_rme32_remove), }; -static int __init alsa_card_rme32_init(void) -{ - return pci_register_driver(&driver); -} - -static void __exit alsa_card_rme32_exit(void) -{ - pci_unregister_driver(&driver); -} - -module_init(alsa_card_rme32_init) -module_exit(alsa_card_rme32_exit) +module_pci_driver(rme32_driver); diff --git a/sound/pci/rme96.c b/sound/pci/rme96.c index ba894158e76..9b98dc40698 100644 --- a/sound/pci/rme96.c +++ b/sound/pci/rme96.c @@ -2395,22 +2395,11 @@ static void __devexit snd_rme96_remove(struct pci_dev *pci) pci_set_drvdata(pci, NULL); } -static struct pci_driver driver = { +static struct pci_driver rme96_driver = { .name = KBUILD_MODNAME, .id_table = snd_rme96_ids, .probe = snd_rme96_probe, .remove = __devexit_p(snd_rme96_remove), }; -static int __init alsa_card_rme96_init(void) -{ - return pci_register_driver(&driver); -} - -static void __exit alsa_card_rme96_exit(void) -{ - pci_unregister_driver(&driver); -} - -module_init(alsa_card_rme96_init) -module_exit(alsa_card_rme96_exit) +module_pci_driver(rme96_driver); diff --git a/sound/pci/rme9652/hdsp.c b/sound/pci/rme9652/hdsp.c index 0b2aea2ce17..0d6930c4f4b 100644 --- a/sound/pci/rme9652/hdsp.c +++ b/sound/pci/rme9652/hdsp.c @@ -5636,22 +5636,11 @@ static void __devexit snd_hdsp_remove(struct pci_dev *pci) pci_set_drvdata(pci, NULL); } -static struct pci_driver driver = { +static struct pci_driver hdsp_driver = { .name = KBUILD_MODNAME, .id_table = snd_hdsp_ids, .probe = snd_hdsp_probe, .remove = __devexit_p(snd_hdsp_remove), }; -static int __init alsa_card_hdsp_init(void) -{ - return pci_register_driver(&driver); -} - -static void __exit alsa_card_hdsp_exit(void) -{ - pci_unregister_driver(&driver); -} - -module_init(alsa_card_hdsp_init) -module_exit(alsa_card_hdsp_exit) +module_pci_driver(hdsp_driver); diff --git a/sound/pci/rme9652/hdspm.c b/sound/pci/rme9652/hdspm.c index bc030a2088d..0a5027b9471 100644 --- a/sound/pci/rme9652/hdspm.c +++ b/sound/pci/rme9652/hdspm.c @@ -6918,23 +6918,11 @@ static void __devexit snd_hdspm_remove(struct pci_dev *pci) pci_set_drvdata(pci, NULL); } -static struct pci_driver driver = { +static struct pci_driver hdspm_driver = { .name = KBUILD_MODNAME, .id_table = snd_hdspm_ids, .probe = snd_hdspm_probe, .remove = __devexit_p(snd_hdspm_remove), }; - -static int __init alsa_card_hdspm_init(void) -{ - return pci_register_driver(&driver); -} - -static void __exit alsa_card_hdspm_exit(void) -{ - pci_unregister_driver(&driver); -} - -module_init(alsa_card_hdspm_init) -module_exit(alsa_card_hdspm_exit) +module_pci_driver(hdspm_driver); diff --git a/sound/pci/rme9652/rme9652.c b/sound/pci/rme9652/rme9652.c index b737d1619cc..a15fc100ab0 100644 --- a/sound/pci/rme9652/rme9652.c +++ b/sound/pci/rme9652/rme9652.c @@ -2631,22 +2631,11 @@ static void __devexit snd_rme9652_remove(struct pci_dev *pci) pci_set_drvdata(pci, NULL); } -static struct pci_driver driver = { +static struct pci_driver rme9652_driver = { .name = KBUILD_MODNAME, .id_table = snd_rme9652_ids, .probe = snd_rme9652_probe, .remove = __devexit_p(snd_rme9652_remove), }; -static int __init alsa_card_hammerfall_init(void) -{ - return pci_register_driver(&driver); -} - -static void __exit alsa_card_hammerfall_exit(void) -{ - pci_unregister_driver(&driver); -} - -module_init(alsa_card_hammerfall_init) -module_exit(alsa_card_hammerfall_exit) +module_pci_driver(rme9652_driver); diff --git a/sound/pci/sis7019.c b/sound/pci/sis7019.c index ff500a87f76..1552642765d 100644 --- a/sound/pci/sis7019.c +++ b/sound/pci/sis7019.c @@ -1488,15 +1488,4 @@ static struct pci_driver sis7019_driver = { #endif }; -static int __init sis7019_init(void) -{ - return pci_register_driver(&sis7019_driver); -} - -static void __exit sis7019_exit(void) -{ - pci_unregister_driver(&sis7019_driver); -} - -module_init(sis7019_init); -module_exit(sis7019_exit); +module_pci_driver(sis7019_driver); diff --git a/sound/pci/sonicvibes.c b/sound/pci/sonicvibes.c index 54cc802050f..baa9946bedf 100644 --- a/sound/pci/sonicvibes.c +++ b/sound/pci/sonicvibes.c @@ -1530,22 +1530,11 @@ static void __devexit snd_sonic_remove(struct pci_dev *pci) pci_set_drvdata(pci, NULL); } -static struct pci_driver driver = { +static struct pci_driver sonicvibes_driver = { .name = KBUILD_MODNAME, .id_table = snd_sonic_ids, .probe = snd_sonic_probe, .remove = __devexit_p(snd_sonic_remove), }; -static int __init alsa_card_sonicvibes_init(void) -{ - return pci_register_driver(&driver); -} - -static void __exit alsa_card_sonicvibes_exit(void) -{ - pci_unregister_driver(&driver); -} - -module_init(alsa_card_sonicvibes_init) -module_exit(alsa_card_sonicvibes_exit) +module_pci_driver(sonicvibes_driver); diff --git a/sound/pci/trident/trident.c b/sound/pci/trident/trident.c index 5f1def7f45e..611983ec732 100644 --- a/sound/pci/trident/trident.c +++ b/sound/pci/trident/trident.c @@ -172,7 +172,7 @@ static void __devexit snd_trident_remove(struct pci_dev *pci) pci_set_drvdata(pci, NULL); } -static struct pci_driver driver = { +static struct pci_driver trident_driver = { .name = KBUILD_MODNAME, .id_table = snd_trident_ids, .probe = snd_trident_probe, @@ -183,15 +183,4 @@ static struct pci_driver driver = { #endif }; -static int __init alsa_card_trident_init(void) -{ - return pci_register_driver(&driver); -} - -static void __exit alsa_card_trident_exit(void) -{ - pci_unregister_driver(&driver); -} - -module_init(alsa_card_trident_init) -module_exit(alsa_card_trident_exit) +module_pci_driver(trident_driver); diff --git a/sound/pci/via82xx.c b/sound/pci/via82xx.c index 75630408c6d..b5afab48943 100644 --- a/sound/pci/via82xx.c +++ b/sound/pci/via82xx.c @@ -2619,7 +2619,7 @@ static void __devexit snd_via82xx_remove(struct pci_dev *pci) pci_set_drvdata(pci, NULL); } -static struct pci_driver driver = { +static struct pci_driver via82xx_driver = { .name = KBUILD_MODNAME, .id_table = snd_via82xx_ids, .probe = snd_via82xx_probe, @@ -2630,15 +2630,4 @@ static struct pci_driver driver = { #endif }; -static int __init alsa_card_via82xx_init(void) -{ - return pci_register_driver(&driver); -} - -static void __exit alsa_card_via82xx_exit(void) -{ - pci_unregister_driver(&driver); -} - -module_init(alsa_card_via82xx_init) -module_exit(alsa_card_via82xx_exit) +module_pci_driver(via82xx_driver); diff --git a/sound/pci/via82xx_modem.c b/sound/pci/via82xx_modem.c index 5efcbcac506..59fd47ed0a3 100644 --- a/sound/pci/via82xx_modem.c +++ b/sound/pci/via82xx_modem.c @@ -1223,7 +1223,7 @@ static void __devexit snd_via82xx_remove(struct pci_dev *pci) pci_set_drvdata(pci, NULL); } -static struct pci_driver driver = { +static struct pci_driver via82xx_modem_driver = { .name = KBUILD_MODNAME, .id_table = snd_via82xx_modem_ids, .probe = snd_via82xx_probe, @@ -1234,15 +1234,4 @@ static struct pci_driver driver = { #endif }; -static int __init alsa_card_via82xx_init(void) -{ - return pci_register_driver(&driver); -} - -static void __exit alsa_card_via82xx_exit(void) -{ - pci_unregister_driver(&driver); -} - -module_init(alsa_card_via82xx_init) -module_exit(alsa_card_via82xx_exit) +module_pci_driver(via82xx_modem_driver); diff --git a/sound/pci/vx222/vx222.c b/sound/pci/vx222/vx222.c index 6a534bfe127..1ea1f656a5d 100644 --- a/sound/pci/vx222/vx222.c +++ b/sound/pci/vx222/vx222.c @@ -289,7 +289,7 @@ static int snd_vx222_resume(struct pci_dev *pci) } #endif -static struct pci_driver driver = { +static struct pci_driver vx222_driver = { .name = KBUILD_MODNAME, .id_table = snd_vx222_ids, .probe = snd_vx222_probe, @@ -300,15 +300,4 @@ static struct pci_driver driver = { #endif }; -static int __init alsa_card_vx222_init(void) -{ - return pci_register_driver(&driver); -} - -static void __exit alsa_card_vx222_exit(void) -{ - pci_unregister_driver(&driver); -} - -module_init(alsa_card_vx222_init) -module_exit(alsa_card_vx222_exit) +module_pci_driver(vx222_driver); diff --git a/sound/pci/ymfpci/ymfpci.c b/sound/pci/ymfpci/ymfpci.c index 94ab728f5ca..9a1d01d653a 100644 --- a/sound/pci/ymfpci/ymfpci.c +++ b/sound/pci/ymfpci/ymfpci.c @@ -350,7 +350,7 @@ static void __devexit snd_card_ymfpci_remove(struct pci_dev *pci) pci_set_drvdata(pci, NULL); } -static struct pci_driver driver = { +static struct pci_driver ymfpci_driver = { .name = KBUILD_MODNAME, .id_table = snd_ymfpci_ids, .probe = snd_card_ymfpci_probe, @@ -361,15 +361,4 @@ static struct pci_driver driver = { #endif }; -static int __init alsa_card_ymfpci_init(void) -{ - return pci_register_driver(&driver); -} - -static void __exit alsa_card_ymfpci_exit(void) -{ - pci_unregister_driver(&driver); -} - -module_init(alsa_card_ymfpci_init) -module_exit(alsa_card_ymfpci_exit) +module_pci_driver(ymfpci_driver); diff --git a/sound/sh/sh_dac_audio.c b/sound/sh/sh_dac_audio.c index b11f82b5718..f8b01c77b29 100644 --- a/sound/sh/sh_dac_audio.c +++ b/sound/sh/sh_dac_audio.c @@ -433,7 +433,7 @@ probe_error: /* * "driver" definition */ -static struct platform_driver driver = { +static struct platform_driver sh_dac_driver = { .probe = snd_sh_dac_probe, .remove = snd_sh_dac_remove, .driver = { @@ -441,4 +441,4 @@ static struct platform_driver driver = { }, }; -module_platform_driver(driver); +module_platform_driver(sh_dac_driver); diff --git a/sound/soc/Kconfig b/sound/soc/Kconfig index 91c985599d3..40b2ad1bb1c 100644 --- a/sound/soc/Kconfig +++ b/sound/soc/Kconfig @@ -35,7 +35,6 @@ source "sound/soc/blackfin/Kconfig" source "sound/soc/davinci/Kconfig" source "sound/soc/ep93xx/Kconfig" source "sound/soc/fsl/Kconfig" -source "sound/soc/imx/Kconfig" source "sound/soc/jz4740/Kconfig" source "sound/soc/nuc900/Kconfig" source "sound/soc/omap/Kconfig" @@ -48,9 +47,13 @@ source "sound/soc/s6000/Kconfig" source "sound/soc/sh/Kconfig" source "sound/soc/tegra/Kconfig" source "sound/soc/txx9/Kconfig" +source "sound/soc/ux500/Kconfig" # Supported codecs source "sound/soc/codecs/Kconfig" +# generic frame-work +source "sound/soc/generic/Kconfig" + endif # SND_SOC diff --git a/sound/soc/Makefile b/sound/soc/Makefile index 2feaf376e94..70990f4017f 100644 --- a/sound/soc/Makefile +++ b/sound/soc/Makefile @@ -6,13 +6,13 @@ obj-$(CONFIG_SND_SOC_DMAENGINE_PCM) += snd-soc-dmaengine-pcm.o obj-$(CONFIG_SND_SOC) += snd-soc-core.o obj-$(CONFIG_SND_SOC) += codecs/ +obj-$(CONFIG_SND_SOC) += generic/ obj-$(CONFIG_SND_SOC) += atmel/ obj-$(CONFIG_SND_SOC) += au1x/ obj-$(CONFIG_SND_SOC) += blackfin/ obj-$(CONFIG_SND_SOC) += davinci/ obj-$(CONFIG_SND_SOC) += ep93xx/ obj-$(CONFIG_SND_SOC) += fsl/ -obj-$(CONFIG_SND_SOC) += imx/ obj-$(CONFIG_SND_SOC) += jz4740/ obj-$(CONFIG_SND_SOC) += mid-x86/ obj-$(CONFIG_SND_SOC) += mxs/ @@ -25,3 +25,4 @@ obj-$(CONFIG_SND_SOC) += s6000/ obj-$(CONFIG_SND_SOC) += sh/ obj-$(CONFIG_SND_SOC) += tegra/ obj-$(CONFIG_SND_SOC) += txx9/ +obj-$(CONFIG_SND_SOC) += ux500/ diff --git a/sound/soc/blackfin/bf5xx-ssm2602.c b/sound/soc/blackfin/bf5xx-ssm2602.c index b39ad356b92..7dbeef1099b 100644 --- a/sound/soc/blackfin/bf5xx-ssm2602.c +++ b/sound/soc/blackfin/bf5xx-ssm2602.c @@ -44,16 +44,8 @@ static struct snd_soc_card bf5xx_ssm2602; -static int bf5xx_ssm2602_hw_params(struct snd_pcm_substream *substream, - struct snd_pcm_hw_params *params) +static int bf5xx_ssm2602_dai_init(struct snd_soc_pcm_runtime *rtd) { - struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct snd_soc_dai *codec_dai = rtd->codec_dai; - unsigned int clk = 0; - int ret = 0; - - pr_debug("%s rate %d format %x\n", __func__, params_rate(params), - params_format(params)); /* * If you are using a crystal source which frequency is not 12MHz * then modify the below case statement with frequency of the crystal. @@ -61,31 +53,10 @@ static int bf5xx_ssm2602_hw_params(struct snd_pcm_substream *substream, * If you are using the SPORT to generate clocking then this is * where to do it. */ - - switch (params_rate(params)) { - case 8000: - case 16000: - case 48000: - case 96000: - case 11025: - case 22050: - case 44100: - clk = 12000000; - break; - } - - ret = snd_soc_dai_set_sysclk(codec_dai, SSM2602_SYSCLK, clk, + return snd_soc_dai_set_sysclk(rtd->codec_dai, SSM2602_SYSCLK, 12000000, SND_SOC_CLOCK_IN); - if (ret < 0) - return ret; - - return 0; } -static struct snd_soc_ops bf5xx_ssm2602_ops = { - .hw_params = bf5xx_ssm2602_hw_params, -}; - /* CODEC is master for BCLK and LRC in this configuration. */ #define BF5XX_SSM2602_DAIFMT (SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF | \ SND_SOC_DAIFMT_CBM_CFM) @@ -98,7 +69,7 @@ static struct snd_soc_dai_link bf5xx_ssm2602_dai[] = { .codec_dai_name = "ssm2602-hifi", .platform_name = "bfin-i2s-pcm-audio", .codec_name = "ssm2602.0-001b", - .ops = &bf5xx_ssm2602_ops, + .init = bf5xx_ssm2602_dai_init, .dai_fmt = BF5XX_SSM2602_DAIFMT, }, { @@ -108,7 +79,7 @@ static struct snd_soc_dai_link bf5xx_ssm2602_dai[] = { .codec_dai_name = "ssm2602-hifi", .platform_name = "bfin-i2s-pcm-audio", .codec_name = "ssm2602.0-001b", - .ops = &bf5xx_ssm2602_ops, + .init = bf5xx_ssm2602_dai_init, .dai_fmt = BF5XX_SSM2602_DAIFMT, }, }; diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig index 59d8efaa17e..1e1613a438d 100644 --- a/sound/soc/codecs/Kconfig +++ b/sound/soc/codecs/Kconfig @@ -29,6 +29,7 @@ config SND_SOC_ALL_CODECS select SND_SOC_ALC5632 if I2C select SND_SOC_CQ0093VC if MFD_DAVINCI_VOICECODEC select SND_SOC_CS42L51 if I2C + select SND_SOC_CS42L52 if I2C select SND_SOC_CS42L73 if I2C select SND_SOC_CS4270 if I2C select SND_SOC_CS4271 if SND_SOC_I2C_AND_SPI @@ -37,11 +38,15 @@ config SND_SOC_ALL_CODECS select SND_SOC_DFBMCS320 select SND_SOC_JZ4740_CODEC select SND_SOC_LM4857 if I2C + select SND_SOC_LM49453 if I2C select SND_SOC_MAX98088 if I2C select SND_SOC_MAX98095 if I2C select SND_SOC_MAX9850 if I2C select SND_SOC_MAX9768 if I2C select SND_SOC_MAX9877 if I2C + select SND_SOC_MC13783 if MFD_MC13XXX + select SND_SOC_ML26124 if I2C + select SND_SOC_OMAP_HDMI_CODEC if OMAP4_DSS_HDMI select SND_SOC_PCM3008 select SND_SOC_RT5631 if I2C select SND_SOC_SGTL5000 if I2C @@ -181,6 +186,9 @@ config SND_SOC_CQ0093VC config SND_SOC_CS42L51 tristate +config SND_SOC_CS42L52 + tristate + config SND_SOC_CS42L73 tristate @@ -217,6 +225,9 @@ config SND_SOC_DFBMCS320 config SND_SOC_DMIC tristate +config SND_SOC_LM49453 + tristate + config SND_SOC_MAX98088 tristate @@ -226,6 +237,9 @@ config SND_SOC_MAX98095 config SND_SOC_MAX9850 tristate +config SND_SOC_OMAP_HDMI_CODEC + tristate + config SND_SOC_PCM3008 tristate @@ -435,5 +449,11 @@ config SND_SOC_MAX9768 config SND_SOC_MAX9877 tristate +config SND_SOC_MC13783 + tristate + +config SND_SOC_ML26124 + tristate + config SND_SOC_TPA6130A2 tristate diff --git a/sound/soc/codecs/Makefile b/sound/soc/codecs/Makefile index 6662eb0cdcc..fc27fec3948 100644 --- a/sound/soc/codecs/Makefile +++ b/sound/soc/codecs/Makefile @@ -15,6 +15,7 @@ snd-soc-ak4642-objs := ak4642.o snd-soc-ak4671-objs := ak4671.o snd-soc-cq93vc-objs := cq93vc.o snd-soc-cs42l51-objs := cs42l51.o +snd-soc-cs42l52-objs := cs42l52.o snd-soc-cs42l73-objs := cs42l73.o snd-soc-cs4270-objs := cs4270.o snd-soc-cs4271-objs := cs4271.o @@ -25,10 +26,14 @@ snd-soc-dmic-objs := dmic.o snd-soc-jz4740-codec-objs := jz4740.o snd-soc-l3-objs := l3.o snd-soc-lm4857-objs := lm4857.o +snd-soc-lm49453-objs := lm49453.o snd-soc-max9768-objs := max9768.o snd-soc-max98088-objs := max98088.o snd-soc-max98095-objs := max98095.o snd-soc-max9850-objs := max9850.o +snd-soc-mc13783-objs := mc13783.o +snd-soc-ml26124-objs := ml26124.o +snd-soc-omap-hdmi-codec-objs := omap-hdmi.o snd-soc-pcm3008-objs := pcm3008.o snd-soc-rt5631-objs := rt5631.o snd-soc-sgtl5000-objs := sgtl5000.o @@ -121,6 +126,7 @@ obj-$(CONFIG_SND_SOC_ALC5623) += snd-soc-alc5623.o obj-$(CONFIG_SND_SOC_ALC5632) += snd-soc-alc5632.o obj-$(CONFIG_SND_SOC_CQ0093VC) += snd-soc-cq93vc.o obj-$(CONFIG_SND_SOC_CS42L51) += snd-soc-cs42l51.o +obj-$(CONFIG_SND_SOC_CS42L52) += snd-soc-cs42l52.o obj-$(CONFIG_SND_SOC_CS42L73) += snd-soc-cs42l73.o obj-$(CONFIG_SND_SOC_CS4270) += snd-soc-cs4270.o obj-$(CONFIG_SND_SOC_CS4271) += snd-soc-cs4271.o @@ -128,13 +134,17 @@ obj-$(CONFIG_SND_SOC_CX20442) += snd-soc-cx20442.o obj-$(CONFIG_SND_SOC_DA7210) += snd-soc-da7210.o obj-$(CONFIG_SND_SOC_DFBMCS320) += snd-soc-dfbmcs320.o obj-$(CONFIG_SND_SOC_DMIC) += snd-soc-dmic.o +obj-$(CONFIG_SND_SOC_JZ4740_CODEC) += snd-soc-jz4740-codec.o obj-$(CONFIG_SND_SOC_L3) += snd-soc-l3.o obj-$(CONFIG_SND_SOC_LM4857) += snd-soc-lm4857.o -obj-$(CONFIG_SND_SOC_JZ4740_CODEC) += snd-soc-jz4740-codec.o +obj-$(CONFIG_SND_SOC_LM49453) += snd-soc-lm49453.o obj-$(CONFIG_SND_SOC_MAX9768) += snd-soc-max9768.o obj-$(CONFIG_SND_SOC_MAX98088) += snd-soc-max98088.o obj-$(CONFIG_SND_SOC_MAX98095) += snd-soc-max98095.o obj-$(CONFIG_SND_SOC_MAX9850) += snd-soc-max9850.o +obj-$(CONFIG_SND_SOC_MC13783) += snd-soc-mc13783.o +obj-$(CONFIG_SND_SOC_ML26124) += snd-soc-ml26124.o +obj-$(CONFIG_SND_SOC_OMAP_HDMI_CODEC) += snd-soc-omap-hdmi-codec.o obj-$(CONFIG_SND_SOC_PCM3008) += snd-soc-pcm3008.o obj-$(CONFIG_SND_SOC_RT5631) += snd-soc-rt5631.o obj-$(CONFIG_SND_SOC_SGTL5000) += snd-soc-sgtl5000.o diff --git a/sound/soc/codecs/ac97.c b/sound/soc/codecs/ac97.c index 1bbad4c16d2..2023c749f23 100644 --- a/sound/soc/codecs/ac97.c +++ b/sound/soc/codecs/ac97.c @@ -26,13 +26,11 @@ static int ac97_prepare(struct snd_pcm_substream *substream, struct snd_soc_dai *dai) { - struct snd_pcm_runtime *runtime = substream->runtime; - struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct snd_soc_codec *codec = rtd->codec; + struct snd_soc_codec *codec = dai->codec; int reg = (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) ? AC97_PCM_FRONT_DAC_RATE : AC97_PCM_LR_ADC_RATE; - return snd_ac97_set_rate(codec->ac97, reg, runtime->rate); + return snd_ac97_set_rate(codec->ac97, reg, substream->runtime->rate); } #define STD_AC97_RATES (SNDRV_PCM_RATE_8000 | SNDRV_PCM_RATE_11025 |\ diff --git a/sound/soc/codecs/ad1836.c b/sound/soc/codecs/ad1836.c index 12e3b411855..c67b50d8b31 100644 --- a/sound/soc/codecs/ad1836.c +++ b/sound/soc/codecs/ad1836.c @@ -162,9 +162,7 @@ static int ad1836_hw_params(struct snd_pcm_substream *substream, struct snd_soc_dai *dai) { int word_len = 0; - - struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct snd_soc_codec *codec = rtd->codec; + struct snd_soc_codec *codec = dai->codec; /* bit size */ switch (params_format(params)) { diff --git a/sound/soc/codecs/ad193x.c b/sound/soc/codecs/ad193x.c index a4a6bef2c0b..13e62be4f99 100644 --- a/sound/soc/codecs/ad193x.c +++ b/sound/soc/codecs/ad193x.c @@ -245,9 +245,7 @@ static int ad193x_hw_params(struct snd_pcm_substream *substream, struct snd_soc_dai *dai) { int word_len = 0, master_rate = 0; - - struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct snd_soc_codec *codec = rtd->codec; + struct snd_soc_codec *codec = dai->codec; struct ad193x_priv *ad193x = snd_soc_codec_get_drvdata(codec); /* bit size */ diff --git a/sound/soc/codecs/adau1701.c b/sound/soc/codecs/adau1701.c index 78e9ce48bb9..3d50fc8646b 100644 --- a/sound/soc/codecs/adau1701.c +++ b/sound/soc/codecs/adau1701.c @@ -258,8 +258,7 @@ static int adau1701_set_playback_pcm_format(struct snd_soc_codec *codec, static int adau1701_hw_params(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *params, struct snd_soc_dai *dai) { - struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct snd_soc_codec *codec = rtd->codec; + struct snd_soc_codec *codec = dai->codec; snd_pcm_format_t format; unsigned int val; diff --git a/sound/soc/codecs/ak4104.c b/sound/soc/codecs/ak4104.c index ceb96ecf558..31d4483245d 100644 --- a/sound/soc/codecs/ak4104.c +++ b/sound/soc/codecs/ak4104.c @@ -88,8 +88,7 @@ static int ak4104_hw_params(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *params, struct snd_soc_dai *dai) { - struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct snd_soc_codec *codec = rtd->codec; + struct snd_soc_codec *codec = dai->codec; int val = 0; /* set the IEC958 bits: consumer mode, no copyright bit */ diff --git a/sound/soc/codecs/ak4535.c b/sound/soc/codecs/ak4535.c index 838ae8b22b5..618fdc30f73 100644 --- a/sound/soc/codecs/ak4535.c +++ b/sound/soc/codecs/ak4535.c @@ -262,8 +262,7 @@ static int ak4535_hw_params(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *params, struct snd_soc_dai *dai) { - struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct snd_soc_codec *codec = rtd->codec; + struct snd_soc_codec *codec = dai->codec; struct ak4535_priv *ak4535 = snd_soc_codec_get_drvdata(codec); u8 mode2 = snd_soc_read(codec, AK4535_MODE2) & ~(0x3 << 5); int rate = params_rate(params), fs = 256; diff --git a/sound/soc/codecs/ak4641.c b/sound/soc/codecs/ak4641.c index c4d165a4bdd..543a12f471b 100644 --- a/sound/soc/codecs/ak4641.c +++ b/sound/soc/codecs/ak4641.c @@ -296,8 +296,7 @@ static int ak4641_i2s_hw_params(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *params, struct snd_soc_dai *dai) { - struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct snd_soc_codec *codec = rtd->codec; + struct snd_soc_codec *codec = dai->codec; struct ak4641_priv *ak4641 = snd_soc_codec_get_drvdata(codec); int rate = params_rate(params), fs = 256; u8 mode2; @@ -517,67 +516,24 @@ static int ak4641_resume(struct snd_soc_codec *codec) static int ak4641_probe(struct snd_soc_codec *codec) { - struct ak4641_platform_data *pdata = codec->dev->platform_data; int ret; - - if (pdata) { - if (gpio_is_valid(pdata->gpio_power)) { - ret = gpio_request_one(pdata->gpio_power, - GPIOF_OUT_INIT_LOW, "ak4641 power"); - if (ret) - goto err_out; - } - if (gpio_is_valid(pdata->gpio_npdn)) { - ret = gpio_request_one(pdata->gpio_npdn, - GPIOF_OUT_INIT_LOW, "ak4641 npdn"); - if (ret) - goto err_gpio; - - udelay(1); /* > 150 ns */ - gpio_set_value(pdata->gpio_npdn, 1); - } - } - ret = snd_soc_codec_set_cache_io(codec, 8, 8, SND_SOC_I2C); if (ret != 0) { dev_err(codec->dev, "Failed to set cache I/O: %d\n", ret); - goto err_register; + return ret; } /* power on device */ ak4641_set_bias_level(codec, SND_SOC_BIAS_STANDBY); return 0; - -err_register: - if (pdata) { - if (gpio_is_valid(pdata->gpio_power)) - gpio_set_value(pdata->gpio_power, 0); - if (gpio_is_valid(pdata->gpio_npdn)) - gpio_free(pdata->gpio_npdn); - } -err_gpio: - if (pdata && gpio_is_valid(pdata->gpio_power)) - gpio_free(pdata->gpio_power); -err_out: - return ret; } static int ak4641_remove(struct snd_soc_codec *codec) { - struct ak4641_platform_data *pdata = codec->dev->platform_data; - ak4641_set_bias_level(codec, SND_SOC_BIAS_OFF); - if (pdata) { - if (gpio_is_valid(pdata->gpio_power)) { - gpio_set_value(pdata->gpio_power, 0); - gpio_free(pdata->gpio_power); - } - if (gpio_is_valid(pdata->gpio_npdn)) - gpio_free(pdata->gpio_npdn); - } return 0; } @@ -604,6 +560,7 @@ static struct snd_soc_codec_driver soc_codec_dev_ak4641 = { static int __devinit ak4641_i2c_probe(struct i2c_client *i2c, const struct i2c_device_id *id) { + struct ak4641_platform_data *pdata = i2c->dev.platform_data; struct ak4641_priv *ak4641; int ret; @@ -612,16 +569,62 @@ static int __devinit ak4641_i2c_probe(struct i2c_client *i2c, if (!ak4641) return -ENOMEM; + if (pdata) { + if (gpio_is_valid(pdata->gpio_power)) { + ret = gpio_request_one(pdata->gpio_power, + GPIOF_OUT_INIT_LOW, "ak4641 power"); + if (ret) + goto err_out; + } + if (gpio_is_valid(pdata->gpio_npdn)) { + ret = gpio_request_one(pdata->gpio_npdn, + GPIOF_OUT_INIT_LOW, "ak4641 npdn"); + if (ret) + goto err_gpio; + + udelay(1); /* > 150 ns */ + gpio_set_value(pdata->gpio_npdn, 1); + } + } + i2c_set_clientdata(i2c, ak4641); ret = snd_soc_register_codec(&i2c->dev, &soc_codec_dev_ak4641, ak4641_dai, ARRAY_SIZE(ak4641_dai)); + if (ret != 0) + goto err_gpio2; + + return 0; + +err_gpio2: + if (pdata) { + if (gpio_is_valid(pdata->gpio_power)) + gpio_set_value(pdata->gpio_power, 0); + if (gpio_is_valid(pdata->gpio_npdn)) + gpio_free(pdata->gpio_npdn); + } +err_gpio: + if (pdata && gpio_is_valid(pdata->gpio_power)) + gpio_free(pdata->gpio_power); +err_out: return ret; } static int __devexit ak4641_i2c_remove(struct i2c_client *i2c) { + struct ak4641_platform_data *pdata = i2c->dev.platform_data; + snd_soc_unregister_codec(&i2c->dev); + + if (pdata) { + if (gpio_is_valid(pdata->gpio_power)) { + gpio_set_value(pdata->gpio_power, 0); + gpio_free(pdata->gpio_power); + } + if (gpio_is_valid(pdata->gpio_npdn)) + gpio_free(pdata->gpio_npdn); + } + return 0; } @@ -641,23 +644,7 @@ static struct i2c_driver ak4641_i2c_driver = { .id_table = ak4641_i2c_id, }; -static int __init ak4641_modinit(void) -{ - int ret; - - ret = i2c_add_driver(&ak4641_i2c_driver); - if (ret != 0) - pr_err("Failed to register AK4641 I2C driver: %d\n", ret); - - return ret; -} -module_init(ak4641_modinit); - -static void __exit ak4641_exit(void) -{ - i2c_del_driver(&ak4641_i2c_driver); -} -module_exit(ak4641_exit); +module_i2c_driver(ak4641_i2c_driver); MODULE_DESCRIPTION("SoC AK4641 driver"); MODULE_AUTHOR("Harald Welte <laforge@gnufiish.org>"); diff --git a/sound/soc/codecs/alc5623.c b/sound/soc/codecs/alc5623.c index d47b62ddb21..1960478ce6b 100644 --- a/sound/soc/codecs/alc5623.c +++ b/sound/soc/codecs/alc5623.c @@ -705,8 +705,7 @@ static int alc5623_set_dai_fmt(struct snd_soc_dai *codec_dai, static int alc5623_pcm_hw_params(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *params, struct snd_soc_dai *dai) { - struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct snd_soc_codec *codec = rtd->codec; + struct snd_soc_codec *codec = dai->codec; struct alc5623_priv *alc5623 = snd_soc_codec_get_drvdata(codec); int coeff, rate; u16 iface; @@ -1084,25 +1083,7 @@ static struct i2c_driver alc5623_i2c_driver = { .id_table = alc5623_i2c_table, }; -static int __init alc5623_modinit(void) -{ - int ret; - - ret = i2c_add_driver(&alc5623_i2c_driver); - if (ret != 0) { - printk(KERN_ERR "%s: can't add i2c driver", __func__); - return ret; - } - - return ret; -} -module_init(alc5623_modinit); - -static void __exit alc5623_modexit(void) -{ - i2c_del_driver(&alc5623_i2c_driver); -} -module_exit(alc5623_modexit); +module_i2c_driver(alc5623_i2c_driver); MODULE_DESCRIPTION("ASoC alc5621/2/3 driver"); MODULE_AUTHOR("Arnaud Patard <arnaud.patard@rtp-net.org>"); diff --git a/sound/soc/codecs/alc5632.c b/sound/soc/codecs/alc5632.c index e2111e0ccad..7dd02420b36 100644 --- a/sound/soc/codecs/alc5632.c +++ b/sound/soc/codecs/alc5632.c @@ -861,8 +861,7 @@ static int alc5632_set_dai_fmt(struct snd_soc_dai *codec_dai, static int alc5632_pcm_hw_params(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *params, struct snd_soc_dai *dai) { - struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct snd_soc_codec *codec = rtd->codec; + struct snd_soc_codec *codec = dai->codec; int coeff, rate; u16 iface; @@ -1131,7 +1130,7 @@ static __devinit int alc5632_i2c_probe(struct i2c_client *client, i2c_set_clientdata(client, alc5632); - alc5632->regmap = regmap_init_i2c(client, &alc5632_regmap); + alc5632->regmap = devm_regmap_init_i2c(client, &alc5632_regmap); if (IS_ERR(alc5632->regmap)) { ret = PTR_ERR(alc5632->regmap); dev_err(&client->dev, "regmap_init() failed: %d\n", ret); @@ -1143,7 +1142,6 @@ static __devinit int alc5632_i2c_probe(struct i2c_client *client, if (ret1 != 0 || ret2 != 0) { dev_err(&client->dev, "Failed to read chip ID: ret1=%d, ret2=%d\n", ret1, ret2); - regmap_exit(alc5632->regmap); return -EIO; } @@ -1152,14 +1150,12 @@ static __devinit int alc5632_i2c_probe(struct i2c_client *client, if ((vid1 != 0x10EC) || (vid2 != id->driver_data)) { dev_err(&client->dev, "Device is not a ALC5632: VID1=0x%x, VID2=0x%x\n", vid1, vid2); - regmap_exit(alc5632->regmap); return -EINVAL; } ret = alc5632_reset(alc5632->regmap); if (ret < 0) { dev_err(&client->dev, "Failed to issue reset\n"); - regmap_exit(alc5632->regmap); return ret; } @@ -1177,7 +1173,6 @@ static __devinit int alc5632_i2c_probe(struct i2c_client *client, if (ret < 0) { dev_err(&client->dev, "Failed to register codec: %d\n", ret); - regmap_exit(alc5632->regmap); return ret; } @@ -1186,9 +1181,7 @@ static __devinit int alc5632_i2c_probe(struct i2c_client *client, static __devexit int alc5632_i2c_remove(struct i2c_client *client) { - struct alc5632_priv *alc5632 = i2c_get_clientdata(client); snd_soc_unregister_codec(&client->dev); - regmap_exit(alc5632->regmap); return 0; } @@ -1209,25 +1202,7 @@ static struct i2c_driver alc5632_i2c_driver = { .id_table = alc5632_i2c_table, }; -static int __init alc5632_modinit(void) -{ - int ret; - - ret = i2c_add_driver(&alc5632_i2c_driver); - if (ret != 0) { - printk(KERN_ERR "%s: can't add i2c driver", __func__); - return ret; - } - - return ret; -} -module_init(alc5632_modinit); - -static void __exit alc5632_modexit(void) -{ - i2c_del_driver(&alc5632_i2c_driver); -} -module_exit(alc5632_modexit); +module_i2c_driver(alc5632_i2c_driver); MODULE_DESCRIPTION("ASoC ALC5632 driver"); MODULE_AUTHOR("Leon Romanovsky <leon@leon.nu>"); diff --git a/sound/soc/codecs/cs4270.c b/sound/soc/codecs/cs4270.c index 1d672f52866..047917f0b8a 100644 --- a/sound/soc/codecs/cs4270.c +++ b/sound/soc/codecs/cs4270.c @@ -307,8 +307,7 @@ static int cs4270_hw_params(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *params, struct snd_soc_dai *dai) { - struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct snd_soc_codec *codec = rtd->codec; + struct snd_soc_codec *codec = dai->codec; struct cs4270_private *cs4270 = snd_soc_codec_get_drvdata(codec); int ret; unsigned int i; @@ -600,10 +599,12 @@ static int cs4270_soc_suspend(struct snd_soc_codec *codec) static int cs4270_soc_resume(struct snd_soc_codec *codec) { struct cs4270_private *cs4270 = snd_soc_codec_get_drvdata(codec); - int reg; + int reg, ret; - regulator_bulk_enable(ARRAY_SIZE(cs4270->supplies), - cs4270->supplies); + ret = regulator_bulk_enable(ARRAY_SIZE(cs4270->supplies), + cs4270->supplies); + if (ret != 0) + return ret; /* In case the device was put to hard reset during sleep, we need to * wait 500ns here before any I2C communication. */ diff --git a/sound/soc/codecs/cs4271.c b/sound/soc/codecs/cs4271.c index bf7141280a7..9eb01d7d58a 100644 --- a/sound/soc/codecs/cs4271.c +++ b/sound/soc/codecs/cs4271.c @@ -318,8 +318,7 @@ static int cs4271_hw_params(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *params, struct snd_soc_dai *dai) { - struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct snd_soc_codec *codec = rtd->codec; + struct snd_soc_codec *codec = dai->codec; struct cs4271_private *cs4271 = snd_soc_codec_get_drvdata(codec); int i, ret; unsigned int ratio, val; diff --git a/sound/soc/codecs/cs42l51.c b/sound/soc/codecs/cs42l51.c index a8bf588e874..091d0193f50 100644 --- a/sound/soc/codecs/cs42l51.c +++ b/sound/soc/codecs/cs42l51.c @@ -141,15 +141,15 @@ static const struct soc_enum cs42l51_chan_mix = static const struct snd_kcontrol_new cs42l51_snd_controls[] = { SOC_DOUBLE_R_SX_TLV("PCM Playback Volume", CS42L51_PCMA_VOL, CS42L51_PCMB_VOL, - 7, 0xffffff99, 0x18, adc_pcm_tlv), + 6, 0x19, 0x7F, adc_pcm_tlv), SOC_DOUBLE_R("PCM Playback Switch", CS42L51_PCMA_VOL, CS42L51_PCMB_VOL, 7, 1, 1), SOC_DOUBLE_R_SX_TLV("Analog Playback Volume", CS42L51_AOUTA_VOL, CS42L51_AOUTB_VOL, - 8, 0xffffff19, 0x18, aout_tlv), + 0, 0x34, 0xE4, aout_tlv), SOC_DOUBLE_R_SX_TLV("ADC Mixer Volume", CS42L51_ADCA_VOL, CS42L51_ADCB_VOL, - 7, 0xffffff99, 0x18, adc_pcm_tlv), + 6, 0x19, 0x7F, adc_pcm_tlv), SOC_DOUBLE_R("ADC Mixer Switch", CS42L51_ADCA_VOL, CS42L51_ADCB_VOL, 7, 1, 1), SOC_SINGLE("Playback Deemphasis Switch", CS42L51_DAC_CTL, 3, 1, 0), @@ -356,8 +356,7 @@ static int cs42l51_hw_params(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *params, struct snd_soc_dai *dai) { - struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct snd_soc_codec *codec = rtd->codec; + struct snd_soc_codec *codec = dai->codec; struct cs42l51_private *cs42l51 = snd_soc_codec_get_drvdata(codec); int ret; unsigned int i; diff --git a/sound/soc/codecs/cs42l52.c b/sound/soc/codecs/cs42l52.c new file mode 100644 index 00000000000..a7109413aef --- /dev/null +++ b/sound/soc/codecs/cs42l52.c @@ -0,0 +1,1295 @@ +/* + * cs42l52.c -- CS42L52 ALSA SoC audio driver + * + * Copyright 2012 CirrusLogic, Inc. + * + * Author: Georgi Vlaev <joe@nucleusys.com> + * Author: Brian Austin <brian.austin@cirrus.com> + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License version 2 as + * published by the Free Software Foundation. + * + */ + +#include <linux/module.h> +#include <linux/moduleparam.h> +#include <linux/version.h> +#include <linux/kernel.h> +#include <linux/init.h> +#include <linux/delay.h> +#include <linux/pm.h> +#include <linux/i2c.h> +#include <linux/input.h> +#include <linux/regmap.h> +#include <linux/slab.h> +#include <linux/workqueue.h> +#include <linux/platform_device.h> +#include <linux/slab.h> +#include <sound/core.h> +#include <sound/pcm.h> +#include <sound/pcm_params.h> +#include <sound/soc.h> +#include <sound/soc-dapm.h> +#include <sound/initval.h> +#include <sound/tlv.h> +#include <sound/cs42l52.h> +#include "cs42l52.h" + +struct sp_config { + u8 spc, format, spfs; + u32 srate; +}; + +struct cs42l52_private { + struct regmap *regmap; + struct snd_soc_codec *codec; + struct device *dev; + struct sp_config config; + struct cs42l52_platform_data pdata; + u32 sysclk; + u8 mclksel; + u32 mclk; + u8 flags; +#if defined(CONFIG_INPUT) || defined(CONFIG_INPUT_MODULE) + struct input_dev *beep; + struct work_struct beep_work; + int beep_rate; +#endif +}; + +static const struct reg_default cs42l52_reg_defaults[] = { + { CS42L52_PWRCTL1, 0x9F }, /* r02 PWRCTL 1 */ + { CS42L52_PWRCTL2, 0x07 }, /* r03 PWRCTL 2 */ + { CS42L52_PWRCTL3, 0xFF }, /* r04 PWRCTL 3 */ + { CS42L52_CLK_CTL, 0xA0 }, /* r05 Clocking Ctl */ + { CS42L52_IFACE_CTL1, 0x00 }, /* r06 Interface Ctl 1 */ + { CS42L52_ADC_PGA_A, 0x80 }, /* r08 Input A Select */ + { CS42L52_ADC_PGA_B, 0x80 }, /* r09 Input B Select */ + { CS42L52_ANALOG_HPF_CTL, 0xA5 }, /* r0A Analog HPF Ctl */ + { CS42L52_ADC_HPF_FREQ, 0x00 }, /* r0B ADC HPF Corner Freq */ + { CS42L52_ADC_MISC_CTL, 0x00 }, /* r0C Misc. ADC Ctl */ + { CS42L52_PB_CTL1, 0x60 }, /* r0D Playback Ctl 1 */ + { CS42L52_MISC_CTL, 0x02 }, /* r0E Misc. Ctl */ + { CS42L52_PB_CTL2, 0x00 }, /* r0F Playback Ctl 2 */ + { CS42L52_MICA_CTL, 0x00 }, /* r10 MICA Amp Ctl */ + { CS42L52_MICB_CTL, 0x00 }, /* r11 MICB Amp Ctl */ + { CS42L52_PGAA_CTL, 0x00 }, /* r12 PGAA Vol, Misc. */ + { CS42L52_PGAB_CTL, 0x00 }, /* r13 PGAB Vol, Misc. */ + { CS42L52_PASSTHRUA_VOL, 0x00 }, /* r14 Bypass A Vol */ + { CS42L52_PASSTHRUB_VOL, 0x00 }, /* r15 Bypass B Vol */ + { CS42L52_ADCA_VOL, 0x00 }, /* r16 ADCA Volume */ + { CS42L52_ADCB_VOL, 0x00 }, /* r17 ADCB Volume */ + { CS42L52_ADCA_MIXER_VOL, 0x80 }, /* r18 ADCA Mixer Volume */ + { CS42L52_ADCB_MIXER_VOL, 0x80 }, /* r19 ADCB Mixer Volume */ + { CS42L52_PCMA_MIXER_VOL, 0x00 }, /* r1A PCMA Mixer Volume */ + { CS42L52_PCMB_MIXER_VOL, 0x00 }, /* r1B PCMB Mixer Volume */ + { CS42L52_BEEP_FREQ, 0x00 }, /* r1C Beep Freq on Time */ + { CS42L52_BEEP_VOL, 0x00 }, /* r1D Beep Volume off Time */ + { CS42L52_BEEP_TONE_CTL, 0x00 }, /* r1E Beep Tone Cfg. */ + { CS42L52_TONE_CTL, 0x00 }, /* r1F Tone Ctl */ + { CS42L52_MASTERA_VOL, 0x88 }, /* r20 Master A Volume */ + { CS42L52_MASTERB_VOL, 0x00 }, /* r21 Master B Volume */ + { CS42L52_HPA_VOL, 0x00 }, /* r22 Headphone A Volume */ + { CS42L52_HPB_VOL, 0x00 }, /* r23 Headphone B Volume */ + { CS42L52_SPKA_VOL, 0x00 }, /* r24 Speaker A Volume */ + { CS42L52_SPKB_VOL, 0x00 }, /* r25 Speaker B Volume */ + { CS42L52_ADC_PCM_MIXER, 0x00 }, /* r26 Channel Mixer and Swap */ + { CS42L52_LIMITER_CTL1, 0x00 }, /* r27 Limit Ctl 1 Thresholds */ + { CS42L52_LIMITER_CTL2, 0x7F }, /* r28 Limit Ctl 2 Release Rate */ + { CS42L52_LIMITER_AT_RATE, 0xC0 }, /* r29 Limiter Attack Rate */ + { CS42L52_ALC_CTL, 0x00 }, /* r2A ALC Ctl 1 Attack Rate */ + { CS42L52_ALC_RATE, 0x3F }, /* r2B ALC Release Rate */ + { CS42L52_ALC_THRESHOLD, 0x3f }, /* r2C ALC Thresholds */ + { CS42L52_NOISE_GATE_CTL, 0x00 }, /* r2D Noise Gate Ctl */ + { CS42L52_CLK_STATUS, 0x00 }, /* r2E Overflow and Clock Status */ + { CS42L52_BATT_COMPEN, 0x00 }, /* r2F battery Compensation */ + { CS42L52_BATT_LEVEL, 0x00 }, /* r30 VP Battery Level */ + { CS42L52_SPK_STATUS, 0x00 }, /* r31 Speaker Status */ + { CS42L52_TEM_CTL, 0x3B }, /* r32 Temp Ctl */ + { CS42L52_THE_FOLDBACK, 0x00 }, /* r33 Foldback */ +}; + +static bool cs42l52_readable_register(struct device *dev, unsigned int reg) +{ + switch (reg) { + case CS42L52_CHIP: + case CS42L52_PWRCTL1: + case CS42L52_PWRCTL2: + case CS42L52_PWRCTL3: + case CS42L52_CLK_CTL: + case CS42L52_IFACE_CTL1: + case CS42L52_IFACE_CTL2: + case CS42L52_ADC_PGA_A: + case CS42L52_ADC_PGA_B: + case CS42L52_ANALOG_HPF_CTL: + case CS42L52_ADC_HPF_FREQ: + case CS42L52_ADC_MISC_CTL: + case CS42L52_PB_CTL1: + case CS42L52_MISC_CTL: + case CS42L52_PB_CTL2: + case CS42L52_MICA_CTL: + case CS42L52_MICB_CTL: + case CS42L52_PGAA_CTL: + case CS42L52_PGAB_CTL: + case CS42L52_PASSTHRUA_VOL: + case CS42L52_PASSTHRUB_VOL: + case CS42L52_ADCA_VOL: + case CS42L52_ADCB_VOL: + case CS42L52_ADCA_MIXER_VOL: + case CS42L52_ADCB_MIXER_VOL: + case CS42L52_PCMA_MIXER_VOL: + case CS42L52_PCMB_MIXER_VOL: + case CS42L52_BEEP_FREQ: + case CS42L52_BEEP_VOL: + case CS42L52_BEEP_TONE_CTL: + case CS42L52_TONE_CTL: + case CS42L52_MASTERA_VOL: + case CS42L52_MASTERB_VOL: + case CS42L52_HPA_VOL: + case CS42L52_HPB_VOL: + case CS42L52_SPKA_VOL: + case CS42L52_SPKB_VOL: + case CS42L52_ADC_PCM_MIXER: + case CS42L52_LIMITER_CTL1: + case CS42L52_LIMITER_CTL2: + case CS42L52_LIMITER_AT_RATE: + case CS42L52_ALC_CTL: + case CS42L52_ALC_RATE: + case CS42L52_ALC_THRESHOLD: + case CS42L52_NOISE_GATE_CTL: + case CS42L52_CLK_STATUS: + case CS42L52_BATT_COMPEN: + case CS42L52_BATT_LEVEL: + case CS42L52_SPK_STATUS: + case CS42L52_TEM_CTL: + case CS42L52_THE_FOLDBACK: + case CS42L52_CHARGE_PUMP: + return true; + default: + return false; + } +} + +static bool cs42l52_volatile_register(struct device *dev, unsigned int reg) +{ + switch (reg) { + case CS42L52_IFACE_CTL2: + case CS42L52_CLK_STATUS: + case CS42L52_BATT_LEVEL: + case CS42L52_SPK_STATUS: + case CS42L52_CHARGE_PUMP: + return 1; + default: + return 0; + } +} + +static DECLARE_TLV_DB_SCALE(hl_tlv, -10200, 50, 0); + +static DECLARE_TLV_DB_SCALE(hpd_tlv, -9600, 50, 1); + +static DECLARE_TLV_DB_SCALE(ipd_tlv, -9600, 100, 0); + +static DECLARE_TLV_DB_SCALE(mic_tlv, 1600, 100, 0); + +static DECLARE_TLV_DB_SCALE(pga_tlv, -600, 50, 0); + +static const unsigned int limiter_tlv[] = { + TLV_DB_RANGE_HEAD(2), + 0, 2, TLV_DB_SCALE_ITEM(-3000, 600, 0), + 3, 7, TLV_DB_SCALE_ITEM(-1200, 300, 0), +}; + +static const char * const cs42l52_adca_text[] = { + "Input1A", "Input2A", "Input3A", "Input4A", "PGA Input Left"}; + +static const char * const cs42l52_adcb_text[] = { + "Input1B", "Input2B", "Input3B", "Input4B", "PGA Input Right"}; + +static const struct soc_enum adca_enum = + SOC_ENUM_SINGLE(CS42L52_ADC_PGA_A, 5, + ARRAY_SIZE(cs42l52_adca_text), cs42l52_adca_text); + +static const struct soc_enum adcb_enum = + SOC_ENUM_SINGLE(CS42L52_ADC_PGA_B, 5, + ARRAY_SIZE(cs42l52_adcb_text), cs42l52_adcb_text); + +static const struct snd_kcontrol_new adca_mux = + SOC_DAPM_ENUM("Left ADC Input Capture Mux", adca_enum); + +static const struct snd_kcontrol_new adcb_mux = + SOC_DAPM_ENUM("Right ADC Input Capture Mux", adcb_enum); + +static const char * const mic_bias_level_text[] = { + "0.5 +VA", "0.6 +VA", "0.7 +VA", + "0.8 +VA", "0.83 +VA", "0.91 +VA" +}; + +static const struct soc_enum mic_bias_level_enum = + SOC_ENUM_SINGLE(CS42L52_IFACE_CTL1, 0, + ARRAY_SIZE(mic_bias_level_text), mic_bias_level_text); + +static const char * const cs42l52_mic_text[] = { "Single", "Differential" }; + +static const struct soc_enum mica_enum = + SOC_ENUM_SINGLE(CS42L52_MICA_CTL, 5, + ARRAY_SIZE(cs42l52_mic_text), cs42l52_mic_text); + +static const struct soc_enum micb_enum = + SOC_ENUM_SINGLE(CS42L52_MICB_CTL, 5, + ARRAY_SIZE(cs42l52_mic_text), cs42l52_mic_text); + +static const struct snd_kcontrol_new mica_mux = + SOC_DAPM_ENUM("Left Mic Input Capture Mux", mica_enum); + +static const struct snd_kcontrol_new micb_mux = + SOC_DAPM_ENUM("Right Mic Input Capture Mux", micb_enum); + +static const char * const digital_output_mux_text[] = {"ADC", "DSP"}; + +static const struct soc_enum digital_output_mux_enum = + SOC_ENUM_SINGLE(CS42L52_ADC_MISC_CTL, 6, + ARRAY_SIZE(digital_output_mux_text), + digital_output_mux_text); + +static const struct snd_kcontrol_new digital_output_mux = + SOC_DAPM_ENUM("Digital Output Mux", digital_output_mux_enum); + +static const char * const hp_gain_num_text[] = { + "0.3959", "0.4571", "0.5111", "0.6047", + "0.7099", "0.8399", "1.000", "1.1430" +}; + +static const struct soc_enum hp_gain_enum = + SOC_ENUM_SINGLE(CS42L52_PB_CTL1, 4, + ARRAY_SIZE(hp_gain_num_text), hp_gain_num_text); + +static const char * const beep_pitch_text[] = { + "C4", "C5", "D5", "E5", "F5", "G5", "A5", "B5", + "C6", "D6", "E6", "F6", "G6", "A6", "B6", "C7" +}; + +static const struct soc_enum beep_pitch_enum = + SOC_ENUM_SINGLE(CS42L52_BEEP_FREQ, 4, + ARRAY_SIZE(beep_pitch_text), beep_pitch_text); + +static const char * const beep_ontime_text[] = { + "86 ms", "430 ms", "780 ms", "1.20 s", "1.50 s", + "1.80 s", "2.20 s", "2.50 s", "2.80 s", "3.20 s", + "3.50 s", "3.80 s", "4.20 s", "4.50 s", "4.80 s", "5.20 s" +}; + +static const struct soc_enum beep_ontime_enum = + SOC_ENUM_SINGLE(CS42L52_BEEP_FREQ, 0, + ARRAY_SIZE(beep_ontime_text), beep_ontime_text); + +static const char * const beep_offtime_text[] = { + "1.23 s", "2.58 s", "3.90 s", "5.20 s", + "6.60 s", "8.05 s", "9.35 s", "10.80 s" +}; + +static const struct soc_enum beep_offtime_enum = + SOC_ENUM_SINGLE(CS42L52_BEEP_VOL, 5, + ARRAY_SIZE(beep_offtime_text), beep_offtime_text); + +static const char * const beep_config_text[] = { + "Off", "Single", "Multiple", "Continuous" +}; + +static const struct soc_enum beep_config_enum = + SOC_ENUM_SINGLE(CS42L52_BEEP_TONE_CTL, 6, + ARRAY_SIZE(beep_config_text), beep_config_text); + +static const char * const beep_bass_text[] = { + "50 Hz", "100 Hz", "200 Hz", "250 Hz" +}; + +static const struct soc_enum beep_bass_enum = + SOC_ENUM_SINGLE(CS42L52_BEEP_TONE_CTL, 1, + ARRAY_SIZE(beep_bass_text), beep_bass_text); + +static const char * const beep_treble_text[] = { + "5 kHz", "7 kHz", "10 kHz", " 15 kHz" +}; + +static const struct soc_enum beep_treble_enum = + SOC_ENUM_SINGLE(CS42L52_BEEP_TONE_CTL, 3, + ARRAY_SIZE(beep_treble_text), beep_treble_text); + +static const char * const ng_threshold_text[] = { + "-34dB", "-37dB", "-40dB", "-43dB", + "-46dB", "-52dB", "-58dB", "-64dB" +}; + +static const struct soc_enum ng_threshold_enum = + SOC_ENUM_SINGLE(CS42L52_NOISE_GATE_CTL, 2, + ARRAY_SIZE(ng_threshold_text), ng_threshold_text); + +static const char * const cs42l52_ng_delay_text[] = { + "50ms", "100ms", "150ms", "200ms"}; + +static const struct soc_enum ng_delay_enum = + SOC_ENUM_SINGLE(CS42L52_NOISE_GATE_CTL, 0, + ARRAY_SIZE(cs42l52_ng_delay_text), cs42l52_ng_delay_text); + +static const char * const cs42l52_ng_type_text[] = { + "Apply Specific", "Apply All" +}; + +static const struct soc_enum ng_type_enum = + SOC_ENUM_SINGLE(CS42L52_NOISE_GATE_CTL, 6, + ARRAY_SIZE(cs42l52_ng_type_text), cs42l52_ng_type_text); + +static const char * const left_swap_text[] = { + "Left", "LR 2", "Right"}; + +static const char * const right_swap_text[] = { + "Right", "LR 2", "Left"}; + +static const unsigned int swap_values[] = { 0, 1, 3 }; + +static const struct soc_enum adca_swap_enum = + SOC_VALUE_ENUM_SINGLE(CS42L52_ADC_PCM_MIXER, 2, 1, + ARRAY_SIZE(left_swap_text), + left_swap_text, + swap_values); + +static const struct snd_kcontrol_new adca_mixer = + SOC_DAPM_ENUM("Route", adca_swap_enum); + +static const struct soc_enum pcma_swap_enum = + SOC_VALUE_ENUM_SINGLE(CS42L52_ADC_PCM_MIXER, 6, 1, + ARRAY_SIZE(left_swap_text), + left_swap_text, + swap_values); + +static const struct snd_kcontrol_new pcma_mixer = + SOC_DAPM_ENUM("Route", pcma_swap_enum); + +static const struct soc_enum adcb_swap_enum = + SOC_VALUE_ENUM_SINGLE(CS42L52_ADC_PCM_MIXER, 0, 1, + ARRAY_SIZE(right_swap_text), + right_swap_text, + swap_values); + +static const struct snd_kcontrol_new adcb_mixer = + SOC_DAPM_ENUM("Route", adcb_swap_enum); + +static const struct soc_enum pcmb_swap_enum = + SOC_VALUE_ENUM_SINGLE(CS42L52_ADC_PCM_MIXER, 4, 1, + ARRAY_SIZE(right_swap_text), + right_swap_text, + swap_values); + +static const struct snd_kcontrol_new pcmb_mixer = + SOC_DAPM_ENUM("Route", pcmb_swap_enum); + + +static const struct snd_kcontrol_new passthrul_ctl = + SOC_DAPM_SINGLE("Switch", CS42L52_MISC_CTL, 6, 1, 0); + +static const struct snd_kcontrol_new passthrur_ctl = + SOC_DAPM_SINGLE("Switch", CS42L52_MISC_CTL, 7, 1, 0); + +static const struct snd_kcontrol_new spkl_ctl = + SOC_DAPM_SINGLE("Switch", CS42L52_PWRCTL3, 0, 1, 1); + +static const struct snd_kcontrol_new spkr_ctl = + SOC_DAPM_SINGLE("Switch", CS42L52_PWRCTL3, 2, 1, 1); + +static const struct snd_kcontrol_new hpl_ctl = + SOC_DAPM_SINGLE("Switch", CS42L52_PWRCTL3, 4, 1, 1); + +static const struct snd_kcontrol_new hpr_ctl = + SOC_DAPM_SINGLE("Switch", CS42L52_PWRCTL3, 6, 1, 1); + +static const struct snd_kcontrol_new cs42l52_snd_controls[] = { + + SOC_DOUBLE_R_SX_TLV("Master Volume", CS42L52_MASTERA_VOL, + CS42L52_MASTERB_VOL, 0, 0x34, 0xE4, hl_tlv), + + SOC_DOUBLE_R_SX_TLV("Headphone Volume", CS42L52_HPA_VOL, + CS42L52_HPB_VOL, 0, 0x34, 0xCC, hpd_tlv), + + SOC_ENUM("Headphone Analog Gain", hp_gain_enum), + + SOC_DOUBLE_R_SX_TLV("Speaker Volume", CS42L52_SPKA_VOL, + CS42L52_SPKB_VOL, 7, 0x1, 0xff, hl_tlv), + + SOC_DOUBLE_R_SX_TLV("Bypass Volume", CS42L52_PASSTHRUA_VOL, + CS42L52_PASSTHRUB_VOL, 6, 0x18, 0x90, pga_tlv), + + SOC_DOUBLE("Bypass Mute", CS42L52_MISC_CTL, 4, 5, 1, 0), + + SOC_DOUBLE_R_TLV("MIC Gain Volume", CS42L52_MICA_CTL, + CS42L52_MICB_CTL, 0, 0x10, 0, mic_tlv), + + SOC_ENUM("MIC Bias Level", mic_bias_level_enum), + + SOC_DOUBLE_R_SX_TLV("ADC Volume", CS42L52_ADCA_VOL, + CS42L52_ADCB_VOL, 7, 0x80, 0xA0, ipd_tlv), + SOC_DOUBLE_R_SX_TLV("ADC Mixer Volume", + CS42L52_ADCA_MIXER_VOL, CS42L52_ADCB_MIXER_VOL, + 6, 0x7f, 0x19, ipd_tlv), + + SOC_DOUBLE("ADC Switch", CS42L52_ADC_MISC_CTL, 0, 1, 1, 0), + + SOC_DOUBLE_R("ADC Mixer Switch", CS42L52_ADCA_MIXER_VOL, + CS42L52_ADCB_MIXER_VOL, 7, 1, 1), + + SOC_DOUBLE_R_SX_TLV("PGA Volume", CS42L52_PGAA_CTL, + CS42L52_PGAB_CTL, 0, 0x28, 0x30, pga_tlv), + + SOC_DOUBLE_R_SX_TLV("PCM Mixer Volume", + CS42L52_PCMA_MIXER_VOL, CS42L52_PCMB_MIXER_VOL, + 6, 0x7f, 0x19, hl_tlv), + SOC_DOUBLE_R("PCM Mixer Switch", + CS42L52_PCMA_MIXER_VOL, CS42L52_PCMB_MIXER_VOL, 7, 1, 1), + + SOC_ENUM("Beep Config", beep_config_enum), + SOC_ENUM("Beep Pitch", beep_pitch_enum), + SOC_ENUM("Beep on Time", beep_ontime_enum), + SOC_ENUM("Beep off Time", beep_offtime_enum), + SOC_SINGLE_TLV("Beep Volume", CS42L52_BEEP_VOL, 0, 0x1f, 0x07, hl_tlv), + SOC_SINGLE("Beep Mixer Switch", CS42L52_BEEP_TONE_CTL, 5, 1, 1), + SOC_ENUM("Beep Treble Corner Freq", beep_treble_enum), + SOC_ENUM("Beep Bass Corner Freq", beep_bass_enum), + + SOC_SINGLE("Tone Control Switch", CS42L52_BEEP_TONE_CTL, 0, 1, 1), + SOC_SINGLE_TLV("Treble Gain Volume", + CS42L52_TONE_CTL, 4, 15, 1, hl_tlv), + SOC_SINGLE_TLV("Bass Gain Volume", + CS42L52_TONE_CTL, 0, 15, 1, hl_tlv), + + /* Limiter */ + SOC_SINGLE_TLV("Limiter Max Threshold Volume", + CS42L52_LIMITER_CTL1, 5, 7, 0, limiter_tlv), + SOC_SINGLE_TLV("Limiter Cushion Threshold Volume", + CS42L52_LIMITER_CTL1, 2, 7, 0, limiter_tlv), + SOC_SINGLE_TLV("Limiter Release Rate Volume", + CS42L52_LIMITER_CTL2, 0, 63, 0, limiter_tlv), + SOC_SINGLE_TLV("Limiter Attack Rate Volume", + CS42L52_LIMITER_AT_RATE, 0, 63, 0, limiter_tlv), + + SOC_SINGLE("Limiter SR Switch", CS42L52_LIMITER_CTL1, 1, 1, 0), + SOC_SINGLE("Limiter ZC Switch", CS42L52_LIMITER_CTL1, 0, 1, 0), + SOC_SINGLE("Limiter Switch", CS42L52_LIMITER_CTL2, 7, 1, 0), + + /* ALC */ + SOC_SINGLE_TLV("ALC Attack Rate Volume", CS42L52_ALC_CTL, + 0, 63, 0, limiter_tlv), + SOC_SINGLE_TLV("ALC Release Rate Volume", CS42L52_ALC_RATE, + 0, 63, 0, limiter_tlv), + SOC_SINGLE_TLV("ALC Max Threshold Volume", CS42L52_ALC_THRESHOLD, + 5, 7, 0, limiter_tlv), + SOC_SINGLE_TLV("ALC Min Threshold Volume", CS42L52_ALC_THRESHOLD, + 2, 7, 0, limiter_tlv), + + SOC_DOUBLE_R("ALC SR Capture Switch", CS42L52_PGAA_CTL, + CS42L52_PGAB_CTL, 7, 1, 1), + SOC_DOUBLE_R("ALC ZC Capture Switch", CS42L52_PGAA_CTL, + CS42L52_PGAB_CTL, 6, 1, 1), + SOC_DOUBLE("ALC Capture Switch", CS42L52_ALC_CTL, 6, 7, 1, 0), + + /* Noise gate */ + SOC_ENUM("NG Type Switch", ng_type_enum), + SOC_SINGLE("NG Enable Switch", CS42L52_NOISE_GATE_CTL, 6, 1, 0), + SOC_SINGLE("NG Boost Switch", CS42L52_NOISE_GATE_CTL, 5, 1, 1), + SOC_ENUM("NG Threshold", ng_threshold_enum), + SOC_ENUM("NG Delay", ng_delay_enum), + + SOC_DOUBLE("HPF Switch", CS42L52_ANALOG_HPF_CTL, 5, 7, 1, 0), + + SOC_DOUBLE("Analog SR Switch", CS42L52_ANALOG_HPF_CTL, 1, 3, 1, 1), + SOC_DOUBLE("Analog ZC Switch", CS42L52_ANALOG_HPF_CTL, 0, 2, 1, 1), + SOC_SINGLE("Digital SR Switch", CS42L52_MISC_CTL, 1, 1, 0), + SOC_SINGLE("Digital ZC Switch", CS42L52_MISC_CTL, 0, 1, 0), + SOC_SINGLE("Deemphasis Switch", CS42L52_MISC_CTL, 2, 1, 0), + + SOC_SINGLE("Batt Compensation Switch", CS42L52_BATT_COMPEN, 7, 1, 0), + SOC_SINGLE("Batt VP Monitor Switch", CS42L52_BATT_COMPEN, 6, 1, 0), + SOC_SINGLE("Batt VP ref", CS42L52_BATT_COMPEN, 0, 0x0f, 0), + + SOC_SINGLE("PGA AIN1L Switch", CS42L52_ADC_PGA_A, 0, 1, 0), + SOC_SINGLE("PGA AIN1R Switch", CS42L52_ADC_PGA_B, 0, 1, 0), + SOC_SINGLE("PGA AIN2L Switch", CS42L52_ADC_PGA_A, 1, 1, 0), + SOC_SINGLE("PGA AIN2R Switch", CS42L52_ADC_PGA_B, 1, 1, 0), + + SOC_SINGLE("PGA AIN3L Switch", CS42L52_ADC_PGA_A, 2, 1, 0), + SOC_SINGLE("PGA AIN3R Switch", CS42L52_ADC_PGA_B, 2, 1, 0), + + SOC_SINGLE("PGA AIN4L Switch", CS42L52_ADC_PGA_A, 3, 1, 0), + SOC_SINGLE("PGA AIN4R Switch", CS42L52_ADC_PGA_B, 3, 1, 0), + + SOC_SINGLE("PGA MICA Switch", CS42L52_ADC_PGA_A, 4, 1, 0), + SOC_SINGLE("PGA MICB Switch", CS42L52_ADC_PGA_B, 4, 1, 0), + +}; + +static const struct snd_soc_dapm_widget cs42l52_dapm_widgets[] = { + + SND_SOC_DAPM_INPUT("AIN1L"), + SND_SOC_DAPM_INPUT("AIN1R"), + SND_SOC_DAPM_INPUT("AIN2L"), + SND_SOC_DAPM_INPUT("AIN2R"), + SND_SOC_DAPM_INPUT("AIN3L"), + SND_SOC_DAPM_INPUT("AIN3R"), + SND_SOC_DAPM_INPUT("AIN4L"), + SND_SOC_DAPM_INPUT("AIN4R"), + SND_SOC_DAPM_INPUT("MICA"), + SND_SOC_DAPM_INPUT("MICB"), + SND_SOC_DAPM_SIGGEN("Beep"), + + SND_SOC_DAPM_AIF_OUT("AIFOUTL", NULL, 0, + SND_SOC_NOPM, 0, 0), + SND_SOC_DAPM_AIF_OUT("AIFOUTR", NULL, 0, + SND_SOC_NOPM, 0, 0), + + SND_SOC_DAPM_MUX("MICA Mux", SND_SOC_NOPM, 0, 0, &mica_mux), + SND_SOC_DAPM_MUX("MICB Mux", SND_SOC_NOPM, 0, 0, &micb_mux), + + SND_SOC_DAPM_ADC("ADC Left", NULL, CS42L52_PWRCTL1, 1, 1), + SND_SOC_DAPM_ADC("ADC Right", NULL, CS42L52_PWRCTL1, 2, 1), + SND_SOC_DAPM_PGA("PGA Left", CS42L52_PWRCTL1, 3, 1, NULL, 0), + SND_SOC_DAPM_PGA("PGA Right", CS42L52_PWRCTL1, 4, 1, NULL, 0), + + SND_SOC_DAPM_MUX("ADC Left Mux", SND_SOC_NOPM, 0, 0, &adca_mux), + SND_SOC_DAPM_MUX("ADC Right Mux", SND_SOC_NOPM, 0, 0, &adcb_mux), + + SND_SOC_DAPM_MUX("ADC Left Swap", SND_SOC_NOPM, + 0, 0, &adca_mixer), + SND_SOC_DAPM_MUX("ADC Right Swap", SND_SOC_NOPM, + 0, 0, &adcb_mixer), + + SND_SOC_DAPM_MUX("Output Mux", SND_SOC_NOPM, + 0, 0, &digital_output_mux), + + SND_SOC_DAPM_PGA("PGA MICA", CS42L52_PWRCTL2, 1, 1, NULL, 0), + SND_SOC_DAPM_PGA("PGA MICB", CS42L52_PWRCTL2, 2, 1, NULL, 0), + + SND_SOC_DAPM_SUPPLY("Mic Bias", CS42L52_PWRCTL2, 0, 1, NULL, 0), + SND_SOC_DAPM_SUPPLY("Charge Pump", CS42L52_PWRCTL1, 7, 1, NULL, 0), + + SND_SOC_DAPM_AIF_IN("AIFINL", NULL, 0, + SND_SOC_NOPM, 0, 0), + SND_SOC_DAPM_AIF_IN("AIFINR", NULL, 0, + SND_SOC_NOPM, 0, 0), + + SND_SOC_DAPM_DAC("DAC Left", NULL, SND_SOC_NOPM, 0, 0), + SND_SOC_DAPM_DAC("DAC Right", NULL, SND_SOC_NOPM, 0, 0), + + SND_SOC_DAPM_SWITCH("Bypass Left", CS42L52_MISC_CTL, + 6, 0, &passthrul_ctl), + SND_SOC_DAPM_SWITCH("Bypass Right", CS42L52_MISC_CTL, + 7, 0, &passthrur_ctl), + + SND_SOC_DAPM_MUX("PCM Left Swap", SND_SOC_NOPM, + 0, 0, &pcma_mixer), + SND_SOC_DAPM_MUX("PCM Right Swap", SND_SOC_NOPM, + 0, 0, &pcmb_mixer), + + SND_SOC_DAPM_SWITCH("HP Left Amp", SND_SOC_NOPM, 0, 0, &hpl_ctl), + SND_SOC_DAPM_SWITCH("HP Right Amp", SND_SOC_NOPM, 0, 0, &hpr_ctl), + + SND_SOC_DAPM_SWITCH("SPK Left Amp", SND_SOC_NOPM, 0, 0, &spkl_ctl), + SND_SOC_DAPM_SWITCH("SPK Right Amp", SND_SOC_NOPM, 0, 0, &spkr_ctl), + + SND_SOC_DAPM_OUTPUT("HPOUTA"), + SND_SOC_DAPM_OUTPUT("HPOUTB"), + SND_SOC_DAPM_OUTPUT("SPKOUTA"), + SND_SOC_DAPM_OUTPUT("SPKOUTB"), + +}; + +static const struct snd_soc_dapm_route cs42l52_audio_map[] = { + + {"Capture", NULL, "AIFOUTL"}, + {"Capture", NULL, "AIFOUTL"}, + + {"AIFOUTL", NULL, "Output Mux"}, + {"AIFOUTR", NULL, "Output Mux"}, + + {"Output Mux", "ADC", "ADC Left"}, + {"Output Mux", "ADC", "ADC Right"}, + + {"ADC Left", NULL, "Charge Pump"}, + {"ADC Right", NULL, "Charge Pump"}, + + {"Charge Pump", NULL, "ADC Left Mux"}, + {"Charge Pump", NULL, "ADC Right Mux"}, + + {"ADC Left Mux", "Input1A", "AIN1L"}, + {"ADC Right Mux", "Input1B", "AIN1R"}, + {"ADC Left Mux", "Input2A", "AIN2L"}, + {"ADC Right Mux", "Input2B", "AIN2R"}, + {"ADC Left Mux", "Input3A", "AIN3L"}, + {"ADC Right Mux", "Input3B", "AIN3R"}, + {"ADC Left Mux", "Input4A", "AIN4L"}, + {"ADC Right Mux", "Input4B", "AIN4R"}, + {"ADC Left Mux", "PGA Input Left", "PGA Left"}, + {"ADC Right Mux", "PGA Input Right" , "PGA Right"}, + + {"PGA Left", "Switch", "AIN1L"}, + {"PGA Right", "Switch", "AIN1R"}, + {"PGA Left", "Switch", "AIN2L"}, + {"PGA Right", "Switch", "AIN2R"}, + {"PGA Left", "Switch", "AIN3L"}, + {"PGA Right", "Switch", "AIN3R"}, + {"PGA Left", "Switch", "AIN4L"}, + {"PGA Right", "Switch", "AIN4R"}, + + {"PGA Left", "Switch", "PGA MICA"}, + {"PGA MICA", NULL, "MICA"}, + + {"PGA Right", "Switch", "PGA MICB"}, + {"PGA MICB", NULL, "MICB"}, + + {"HPOUTA", NULL, "HP Left Amp"}, + {"HPOUTB", NULL, "HP Right Amp"}, + {"HP Left Amp", NULL, "Bypass Left"}, + {"HP Right Amp", NULL, "Bypass Right"}, + {"Bypass Left", "Switch", "PGA Left"}, + {"Bypass Right", "Switch", "PGA Right"}, + {"HP Left Amp", "Switch", "DAC Left"}, + {"HP Right Amp", "Switch", "DAC Right"}, + + {"SPKOUTA", NULL, "SPK Left Amp"}, + {"SPKOUTB", NULL, "SPK Right Amp"}, + + {"SPK Left Amp", NULL, "Beep"}, + {"SPK Right Amp", NULL, "Beep"}, + {"SPK Left Amp", "Switch", "Playback"}, + {"SPK Right Amp", "Switch", "Playback"}, + + {"DAC Left", NULL, "Beep"}, + {"DAC Right", NULL, "Beep"}, + {"DAC Left", NULL, "Playback"}, + {"DAC Right", NULL, "Playback"}, + + {"Output Mux", "DSP", "Playback"}, + {"Output Mux", "DSP", "Playback"}, + + {"AIFINL", NULL, "Playback"}, + {"AIFINR", NULL, "Playback"}, + +}; + +struct cs42l52_clk_para { + u32 mclk; + u32 rate; + u8 speed; + u8 group; + u8 videoclk; + u8 ratio; + u8 mclkdiv2; +}; + +static const struct cs42l52_clk_para clk_map_table[] = { + /*8k*/ + {12288000, 8000, CLK_QS_MODE, CLK_32K, CLK_NO_27M, CLK_R_128, 0}, + {18432000, 8000, CLK_QS_MODE, CLK_32K, CLK_NO_27M, CLK_R_128, 0}, + {12000000, 8000, CLK_QS_MODE, CLK_32K, CLK_NO_27M, CLK_R_125, 0}, + {24000000, 8000, CLK_QS_MODE, CLK_32K, CLK_NO_27M, CLK_R_125, 1}, + {27000000, 8000, CLK_QS_MODE, CLK_32K, CLK_27M_MCLK, CLK_R_125, 0}, + + /*11.025k*/ + {11289600, 11025, CLK_QS_MODE, CLK_NO_32K, CLK_NO_27M, CLK_R_128, 0}, + {16934400, 11025, CLK_QS_MODE, CLK_NO_32K, CLK_NO_27M, CLK_R_128, 0}, + + /*16k*/ + {12288000, 16000, CLK_HS_MODE, CLK_32K, CLK_NO_27M, CLK_R_128, 0}, + {18432000, 16000, CLK_HS_MODE, CLK_32K, CLK_NO_27M, CLK_R_128, 0}, + {12000000, 16000, CLK_HS_MODE, CLK_32K, CLK_NO_27M, CLK_R_125, 0}, + {24000000, 16000, CLK_HS_MODE, CLK_32K, CLK_NO_27M, CLK_R_125, 1}, + {27000000, 16000, CLK_HS_MODE, CLK_32K, CLK_27M_MCLK, CLK_R_125, 1}, + + /*22.05k*/ + {11289600, 22050, CLK_HS_MODE, CLK_NO_32K, CLK_NO_27M, CLK_R_128, 0}, + {16934400, 22050, CLK_HS_MODE, CLK_NO_32K, CLK_NO_27M, CLK_R_128, 0}, + + /* 32k */ + {12288000, 32000, CLK_SS_MODE, CLK_32K, CLK_NO_27M, CLK_R_128, 0}, + {18432000, 32000, CLK_SS_MODE, CLK_32K, CLK_NO_27M, CLK_R_128, 0}, + {12000000, 32000, CLK_SS_MODE, CLK_32K, CLK_NO_27M, CLK_R_125, 0}, + {24000000, 32000, CLK_SS_MODE, CLK_32K, CLK_NO_27M, CLK_R_125, 1}, + {27000000, 32000, CLK_SS_MODE, CLK_32K, CLK_27M_MCLK, CLK_R_125, 0}, + + /* 44.1k */ + {11289600, 44100, CLK_SS_MODE, CLK_NO_32K, CLK_NO_27M, CLK_R_128, 0}, + {16934400, 44100, CLK_SS_MODE, CLK_NO_32K, CLK_NO_27M, CLK_R_128, 0}, + + /* 48k */ + {12288000, 48000, CLK_SS_MODE, CLK_NO_32K, CLK_NO_27M, CLK_R_128, 0}, + {18432000, 48000, CLK_SS_MODE, CLK_NO_32K, CLK_NO_27M, CLK_R_128, 0}, + {12000000, 48000, CLK_SS_MODE, CLK_NO_32K, CLK_NO_27M, CLK_R_125, 0}, + {24000000, 48000, CLK_SS_MODE, CLK_NO_32K, CLK_NO_27M, CLK_R_125, 1}, + {27000000, 48000, CLK_SS_MODE, CLK_NO_32K, CLK_27M_MCLK, CLK_R_125, 1}, + + /* 88.2k */ + {11289600, 88200, CLK_DS_MODE, CLK_NO_32K, CLK_NO_27M, CLK_R_128, 0}, + {16934400, 88200, CLK_DS_MODE, CLK_NO_32K, CLK_NO_27M, CLK_R_128, 0}, + + /* 96k */ + {12288000, 96000, CLK_DS_MODE, CLK_NO_32K, CLK_NO_27M, CLK_R_128, 0}, + {18432000, 96000, CLK_DS_MODE, CLK_NO_32K, CLK_NO_27M, CLK_R_128, 0}, + {12000000, 96000, CLK_DS_MODE, CLK_NO_32K, CLK_NO_27M, CLK_R_125, 0}, + {24000000, 96000, CLK_DS_MODE, CLK_NO_32K, CLK_NO_27M, CLK_R_125, 1}, +}; + +static int cs42l52_get_clk(int mclk, int rate) +{ + int i, ret = 0; + u_int mclk1, mclk2 = 0; + + for (i = 0; i < ARRAY_SIZE(clk_map_table); i++) { + if (clk_map_table[i].rate == rate) { + mclk1 = clk_map_table[i].mclk; + if (abs(mclk - mclk1) < abs(mclk - mclk2)) { + mclk2 = mclk1; + ret = i; + } + } + } + if (ret > ARRAY_SIZE(clk_map_table)) + return -EINVAL; + return ret; +} + +static int cs42l52_set_sysclk(struct snd_soc_dai *codec_dai, + int clk_id, unsigned int freq, int dir) +{ + struct snd_soc_codec *codec = codec_dai->codec; + struct cs42l52_private *cs42l52 = snd_soc_codec_get_drvdata(codec); + + if ((freq >= CS42L52_MIN_CLK) && (freq <= CS42L52_MAX_CLK)) { + cs42l52->sysclk = freq; + } else { + dev_err(codec->dev, "Invalid freq paramter\n"); + return -EINVAL; + } + return 0; +} + +static int cs42l52_set_fmt(struct snd_soc_dai *codec_dai, unsigned int fmt) +{ + struct snd_soc_codec *codec = codec_dai->codec; + struct cs42l52_private *cs42l52 = snd_soc_codec_get_drvdata(codec); + int ret = 0; + u8 iface = 0; + + switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) { + case SND_SOC_DAIFMT_CBM_CFM: + iface = CS42L52_IFACE_CTL1_MASTER; + break; + case SND_SOC_DAIFMT_CBS_CFS: + iface = CS42L52_IFACE_CTL1_SLAVE; + break; + default: + return -EINVAL; + } + + /* interface format */ + switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) { + case SND_SOC_DAIFMT_I2S: + iface |= CS42L52_IFACE_CTL1_ADC_FMT_I2S | + CS42L52_IFACE_CTL1_DAC_FMT_I2S; + break; + case SND_SOC_DAIFMT_RIGHT_J: + iface |= CS42L52_IFACE_CTL1_DAC_FMT_RIGHT_J; + break; + case SND_SOC_DAIFMT_LEFT_J: + iface |= CS42L52_IFACE_CTL1_ADC_FMT_LEFT_J | + CS42L52_IFACE_CTL1_DAC_FMT_LEFT_J; + break; + case SND_SOC_DAIFMT_DSP_A: + iface |= CS42L52_IFACE_CTL1_DSP_MODE_EN; + break; + case SND_SOC_DAIFMT_DSP_B: + break; + default: + return -EINVAL; + } + + /* clock inversion */ + switch (fmt & SND_SOC_DAIFMT_INV_MASK) { + case SND_SOC_DAIFMT_NB_NF: + break; + case SND_SOC_DAIFMT_IB_IF: + iface |= CS42L52_IFACE_CTL1_INV_SCLK; + break; + case SND_SOC_DAIFMT_IB_NF: + iface |= CS42L52_IFACE_CTL1_INV_SCLK; + break; + case SND_SOC_DAIFMT_NB_IF: + break; + default: + ret = -EINVAL; + } + cs42l52->config.format = iface; + snd_soc_write(codec, CS42L52_IFACE_CTL1, cs42l52->config.format); + + return 0; +} + +static int cs42l52_digital_mute(struct snd_soc_dai *dai, int mute) +{ + struct snd_soc_codec *codec = dai->codec; + + if (mute) + snd_soc_update_bits(codec, CS42L52_PB_CTL1, + CS42L52_PB_CTL1_MUTE_MASK, + CS42L52_PB_CTL1_MUTE); + else + snd_soc_update_bits(codec, CS42L52_PB_CTL1, + CS42L52_PB_CTL1_MUTE_MASK, + CS42L52_PB_CTL1_UNMUTE); + + return 0; +} + +static int cs42l52_pcm_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params, + struct snd_soc_dai *dai) +{ + struct snd_soc_codec *codec = dai->codec; + struct cs42l52_private *cs42l52 = snd_soc_codec_get_drvdata(codec); + u32 clk = 0; + int index; + + index = cs42l52_get_clk(cs42l52->sysclk, params_rate(params)); + if (index >= 0) { + cs42l52->sysclk = clk_map_table[index].mclk; + + clk |= (clk_map_table[index].speed << CLK_SPEED_SHIFT) | + (clk_map_table[index].group << CLK_32K_SR_SHIFT) | + (clk_map_table[index].videoclk << CLK_27M_MCLK_SHIFT) | + (clk_map_table[index].ratio << CLK_RATIO_SHIFT) | + clk_map_table[index].mclkdiv2; + + snd_soc_write(codec, CS42L52_CLK_CTL, clk); + } else { + dev_err(codec->dev, "can't get correct mclk\n"); + return -EINVAL; + } + + return 0; +} + +static int cs42l52_set_bias_level(struct snd_soc_codec *codec, + enum snd_soc_bias_level level) +{ + struct cs42l52_private *cs42l52 = snd_soc_codec_get_drvdata(codec); + + switch (level) { + case SND_SOC_BIAS_ON: + break; + case SND_SOC_BIAS_PREPARE: + snd_soc_update_bits(codec, CS42L52_PWRCTL1, + CS42L52_PWRCTL1_PDN_CODEC, 0); + break; + case SND_SOC_BIAS_STANDBY: + if (codec->dapm.bias_level == SND_SOC_BIAS_OFF) { + regcache_cache_only(cs42l52->regmap, false); + regcache_sync(cs42l52->regmap); + } + snd_soc_write(codec, CS42L52_PWRCTL1, CS42L52_PWRCTL1_PDN_ALL); + break; + case SND_SOC_BIAS_OFF: + snd_soc_write(codec, CS42L52_PWRCTL1, CS42L52_PWRCTL1_PDN_ALL); + regcache_cache_only(cs42l52->regmap, true); + break; + } + codec->dapm.bias_level = level; + + return 0; +} + +#define CS42L52_RATES (SNDRV_PCM_RATE_8000_96000) + +#define CS42L52_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_U16_LE | \ + SNDRV_PCM_FMTBIT_S18_3LE | SNDRV_PCM_FMTBIT_U18_3LE | \ + SNDRV_PCM_FMTBIT_S20_3LE | SNDRV_PCM_FMTBIT_U20_3LE | \ + SNDRV_PCM_FMTBIT_S24_LE | SNDRV_PCM_FMTBIT_U24_LE) + +static struct snd_soc_dai_ops cs42l52_ops = { + .hw_params = cs42l52_pcm_hw_params, + .digital_mute = cs42l52_digital_mute, + .set_fmt = cs42l52_set_fmt, + .set_sysclk = cs42l52_set_sysclk, +}; + +static struct snd_soc_dai_driver cs42l52_dai = { + .name = "cs42l52", + .playback = { + .stream_name = "Playback", + .channels_min = 1, + .channels_max = 2, + .rates = CS42L52_RATES, + .formats = CS42L52_FORMATS, + }, + .capture = { + .stream_name = "Capture", + .channels_min = 1, + .channels_max = 2, + .rates = CS42L52_RATES, + .formats = CS42L52_FORMATS, + }, + .ops = &cs42l52_ops, +}; + +static int cs42l52_suspend(struct snd_soc_codec *codec) +{ + cs42l52_set_bias_level(codec, SND_SOC_BIAS_OFF); + + return 0; +} + +static int cs42l52_resume(struct snd_soc_codec *codec) +{ + cs42l52_set_bias_level(codec, SND_SOC_BIAS_STANDBY); + + return 0; +} + +#if defined(CONFIG_INPUT) || defined(CONFIG_INPUT_MODULE) +static int beep_rates[] = { + 261, 522, 585, 667, 706, 774, 889, 1000, + 1043, 1200, 1333, 1412, 1600, 1714, 2000, 2182 +}; + +static void cs42l52_beep_work(struct work_struct *work) +{ + struct cs42l52_private *cs42l52 = + container_of(work, struct cs42l52_private, beep_work); + struct snd_soc_codec *codec = cs42l52->codec; + struct snd_soc_dapm_context *dapm = &codec->dapm; + int i; + int val = 0; + int best = 0; + + if (cs42l52->beep_rate) { + for (i = 0; i < ARRAY_SIZE(beep_rates); i++) { + if (abs(cs42l52->beep_rate - beep_rates[i]) < + abs(cs42l52->beep_rate - beep_rates[best])) + best = i; + } + + dev_dbg(codec->dev, "Set beep rate %dHz for requested %dHz\n", + beep_rates[best], cs42l52->beep_rate); + + val = (best << CS42L52_BEEP_RATE_SHIFT); + + snd_soc_dapm_enable_pin(dapm, "Beep"); + } else { + dev_dbg(codec->dev, "Disabling beep\n"); + snd_soc_dapm_disable_pin(dapm, "Beep"); + } + + snd_soc_update_bits(codec, CS42L52_BEEP_FREQ, + CS42L52_BEEP_RATE_MASK, val); + + snd_soc_dapm_sync(dapm); +} + +/* For usability define a way of injecting beep events for the device - + * many systems will not have a keyboard. + */ +static int cs42l52_beep_event(struct input_dev *dev, unsigned int type, + unsigned int code, int hz) +{ + struct snd_soc_codec *codec = input_get_drvdata(dev); + struct cs42l52_private *cs42l52 = snd_soc_codec_get_drvdata(codec); + + dev_dbg(codec->dev, "Beep event %x %x\n", code, hz); + + switch (code) { + case SND_BELL: + if (hz) + hz = 261; + case SND_TONE: + break; + default: + return -1; + } + + /* Kick the beep from a workqueue */ + cs42l52->beep_rate = hz; + schedule_work(&cs42l52->beep_work); + return 0; +} + +static ssize_t cs42l52_beep_set(struct device *dev, + struct device_attribute *attr, + const char *buf, size_t count) +{ + struct cs42l52_private *cs42l52 = dev_get_drvdata(dev); + long int time; + int ret; + + ret = kstrtol(buf, 10, &time); + if (ret != 0) + return ret; + + input_event(cs42l52->beep, EV_SND, SND_TONE, time); + + return count; +} + +static DEVICE_ATTR(beep, 0200, NULL, cs42l52_beep_set); + +static void cs42l52_init_beep(struct snd_soc_codec *codec) +{ + struct cs42l52_private *cs42l52 = snd_soc_codec_get_drvdata(codec); + int ret; + + cs42l52->beep = input_allocate_device(); + if (!cs42l52->beep) { + dev_err(codec->dev, "Failed to allocate beep device\n"); + return; + } + + INIT_WORK(&cs42l52->beep_work, cs42l52_beep_work); + cs42l52->beep_rate = 0; + + cs42l52->beep->name = "CS42L52 Beep Generator"; + cs42l52->beep->phys = dev_name(codec->dev); + cs42l52->beep->id.bustype = BUS_I2C; + + cs42l52->beep->evbit[0] = BIT_MASK(EV_SND); + cs42l52->beep->sndbit[0] = BIT_MASK(SND_BELL) | BIT_MASK(SND_TONE); + cs42l52->beep->event = cs42l52_beep_event; + cs42l52->beep->dev.parent = codec->dev; + input_set_drvdata(cs42l52->beep, codec); + + ret = input_register_device(cs42l52->beep); + if (ret != 0) { + input_free_device(cs42l52->beep); + cs42l52->beep = NULL; + dev_err(codec->dev, "Failed to register beep device\n"); + } + + ret = device_create_file(codec->dev, &dev_attr_beep); + if (ret != 0) { + dev_err(codec->dev, "Failed to create keyclick file: %d\n", + ret); + } +} + +static void cs42l52_free_beep(struct snd_soc_codec *codec) +{ + struct cs42l52_private *cs42l52 = snd_soc_codec_get_drvdata(codec); + + device_remove_file(codec->dev, &dev_attr_beep); + input_unregister_device(cs42l52->beep); + cancel_work_sync(&cs42l52->beep_work); + cs42l52->beep = NULL; + + snd_soc_update_bits(codec, CS42L52_BEEP_TONE_CTL, + CS42L52_BEEP_EN_MASK, 0); +} +#else +static void cs42l52_init_beep(struct snd_soc_codec *codec) +{ +} + +static void cs42l52_free_beep(struct snd_soc_codec *codec) +{ +} +#endif + +static int cs42l52_probe(struct snd_soc_codec *codec) +{ + struct cs42l52_private *cs42l52 = snd_soc_codec_get_drvdata(codec); + int ret; + + codec->control_data = cs42l52->regmap; + ret = snd_soc_codec_set_cache_io(codec, 8, 8, SND_SOC_REGMAP); + if (ret < 0) { + dev_err(codec->dev, "Failed to set cache I/O: %d\n", ret); + return ret; + } + regcache_cache_only(cs42l52->regmap, true); + + cs42l52_init_beep(codec); + + cs42l52_set_bias_level(codec, SND_SOC_BIAS_STANDBY); + + cs42l52->sysclk = CS42L52_DEFAULT_CLK; + cs42l52->config.format = CS42L52_DEFAULT_FORMAT; + + /* Set Platform MICx CFG */ + snd_soc_update_bits(codec, CS42L52_MICA_CTL, + CS42L52_MIC_CTL_TYPE_MASK, + cs42l52->pdata.mica_cfg << + CS42L52_MIC_CTL_TYPE_SHIFT); + + snd_soc_update_bits(codec, CS42L52_MICB_CTL, + CS42L52_MIC_CTL_TYPE_MASK, + cs42l52->pdata.micb_cfg << + CS42L52_MIC_CTL_TYPE_SHIFT); + + /* if Single Ended, Get Mic_Select */ + if (cs42l52->pdata.mica_cfg) + snd_soc_update_bits(codec, CS42L52_MICA_CTL, + CS42L52_MIC_CTL_MIC_SEL_MASK, + cs42l52->pdata.mica_sel << + CS42L52_MIC_CTL_MIC_SEL_SHIFT); + if (cs42l52->pdata.micb_cfg) + snd_soc_update_bits(codec, CS42L52_MICB_CTL, + CS42L52_MIC_CTL_MIC_SEL_MASK, + cs42l52->pdata.micb_sel << + CS42L52_MIC_CTL_MIC_SEL_SHIFT); + + /* Set Platform Charge Pump Freq */ + snd_soc_update_bits(codec, CS42L52_CHARGE_PUMP, + CS42L52_CHARGE_PUMP_MASK, + cs42l52->pdata.chgfreq << + CS42L52_CHARGE_PUMP_SHIFT); + + /* Set Platform Bias Level */ + snd_soc_update_bits(codec, CS42L52_IFACE_CTL2, + CS42L52_IFACE_CTL2_BIAS_LVL, + cs42l52->pdata.micbias_lvl); + + return ret; +} + +static int cs42l52_remove(struct snd_soc_codec *codec) +{ + cs42l52_free_beep(codec); + cs42l52_set_bias_level(codec, SND_SOC_BIAS_OFF); + + return 0; +} + +static struct snd_soc_codec_driver soc_codec_dev_cs42l52 = { + .probe = cs42l52_probe, + .remove = cs42l52_remove, + .suspend = cs42l52_suspend, + .resume = cs42l52_resume, + .set_bias_level = cs42l52_set_bias_level, + + .dapm_widgets = cs42l52_dapm_widgets, + .num_dapm_widgets = ARRAY_SIZE(cs42l52_dapm_widgets), + .dapm_routes = cs42l52_audio_map, + .num_dapm_routes = ARRAY_SIZE(cs42l52_audio_map), + + .controls = cs42l52_snd_controls, + .num_controls = ARRAY_SIZE(cs42l52_snd_controls), +}; + +/* Current and threshold powerup sequence Pg37 */ +static const struct reg_default cs42l52_threshold_patch[] = { + + { 0x00, 0x99 }, + { 0x3E, 0xBA }, + { 0x47, 0x80 }, + { 0x32, 0xBB }, + { 0x32, 0x3B }, + { 0x00, 0x00 }, + +}; + +static struct regmap_config cs42l52_regmap = { + .reg_bits = 8, + .val_bits = 8, + + .max_register = CS42L52_MAX_REGISTER, + .reg_defaults = cs42l52_reg_defaults, + .num_reg_defaults = ARRAY_SIZE(cs42l52_reg_defaults), + .readable_reg = cs42l52_readable_register, + .volatile_reg = cs42l52_volatile_register, + .cache_type = REGCACHE_RBTREE, +}; + +static int cs42l52_i2c_probe(struct i2c_client *i2c_client, + const struct i2c_device_id *id) +{ + struct cs42l52_private *cs42l52; + int ret; + unsigned int devid = 0; + unsigned int reg; + + cs42l52 = devm_kzalloc(&i2c_client->dev, sizeof(struct cs42l52_private), + GFP_KERNEL); + if (cs42l52 == NULL) + return -ENOMEM; + cs42l52->dev = &i2c_client->dev; + + cs42l52->regmap = regmap_init_i2c(i2c_client, &cs42l52_regmap); + if (IS_ERR(cs42l52->regmap)) { + ret = PTR_ERR(cs42l52->regmap); + dev_err(&i2c_client->dev, "regmap_init() failed: %d\n", ret); + goto err; + } + + i2c_set_clientdata(i2c_client, cs42l52); + + if (dev_get_platdata(&i2c_client->dev)) + memcpy(&cs42l52->pdata, dev_get_platdata(&i2c_client->dev), + sizeof(cs42l52->pdata)); + + ret = regmap_register_patch(cs42l52->regmap, cs42l52_threshold_patch, + ARRAY_SIZE(cs42l52_threshold_patch)); + if (ret != 0) + dev_warn(cs42l52->dev, "Failed to apply regmap patch: %d\n", + ret); + + ret = regmap_read(cs42l52->regmap, CS42L52_CHIP, ®); + devid = reg & CS42L52_CHIP_ID_MASK; + if (devid != CS42L52_CHIP_ID) { + ret = -ENODEV; + dev_err(&i2c_client->dev, + "CS42L52 Device ID (%X). Expected %X\n", + devid, CS42L52_CHIP_ID); + goto err_regmap; + } + + regcache_cache_only(cs42l52->regmap, true); + + ret = snd_soc_register_codec(&i2c_client->dev, + &soc_codec_dev_cs42l52, &cs42l52_dai, 1); + if (ret < 0) + goto err_regmap; + return 0; + +err_regmap: + regmap_exit(cs42l52->regmap); + +err: + return ret; +} + +static int cs42l52_i2c_remove(struct i2c_client *client) +{ + struct cs42l52_private *cs42l52 = i2c_get_clientdata(client); + + snd_soc_unregister_codec(&client->dev); + regmap_exit(cs42l52->regmap); + + return 0; +} + +static const struct i2c_device_id cs42l52_id[] = { + { "cs42l52", 0 }, + { } +}; +MODULE_DEVICE_TABLE(i2c, cs42l52_id); + +static struct i2c_driver cs42l52_i2c_driver = { + .driver = { + .name = "cs42l52", + .owner = THIS_MODULE, + }, + .id_table = cs42l52_id, + .probe = cs42l52_i2c_probe, + .remove = __devexit_p(cs42l52_i2c_remove), +}; + +module_i2c_driver(cs42l52_i2c_driver); + +MODULE_DESCRIPTION("ASoC CS42L52 driver"); +MODULE_AUTHOR("Georgi Vlaev, Nucleus Systems Ltd, <joe@nucleusys.com>"); +MODULE_AUTHOR("Brian Austin, Cirrus Logic Inc, <brian.austin@cirrus.com>"); +MODULE_LICENSE("GPL"); diff --git a/sound/soc/codecs/cs42l52.h b/sound/soc/codecs/cs42l52.h new file mode 100644 index 00000000000..60985c05907 --- /dev/null +++ b/sound/soc/codecs/cs42l52.h @@ -0,0 +1,274 @@ +/* + * cs42l52.h -- CS42L52 ALSA SoC audio driver + * + * Copyright 2012 CirrusLogic, Inc. + * + * Author: Georgi Vlaev <joe@nucleusys.com> + * Author: Brian Austin <brian.austin@cirrus.com> + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License version 2 as + * published by the Free Software Foundation. + * + */ + +#ifndef __CS42L52_H__ +#define __CS42L52_H__ + +#define CS42L52_NAME "CS42L52" +#define CS42L52_DEFAULT_CLK 12000000 +#define CS42L52_MIN_CLK 11000000 +#define CS42L52_MAX_CLK 27000000 +#define CS42L52_DEFAULT_FORMAT SNDRV_PCM_FMTBIT_S16_LE +#define CS42L52_DEFAULT_MAX_CHANS 2 +#define CS42L52_SYSCLK 1 + +#define CS42L52_CHIP_SWICTH (1 << 17) +#define CS42L52_ALL_IN_ONE (1 << 16) +#define CS42L52_CHIP_ONE 0x00 +#define CS42L52_CHIP_TWO 0x01 +#define CS42L52_CHIP_THR 0x02 +#define CS42L52_CHIP_MASK 0x0f + +#define CS42L52_FIX_BITS_CTL 0x00 +#define CS42L52_CHIP 0x01 +#define CS42L52_CHIP_ID 0xE0 +#define CS42L52_CHIP_ID_MASK 0xF8 +#define CS42L52_CHIP_REV_A0 0x00 +#define CS42L52_CHIP_REV_A1 0x01 +#define CS42L52_CHIP_REV_B0 0x02 +#define CS42L52_CHIP_REV_MASK 0x03 + +#define CS42L52_PWRCTL1 0x02 +#define CS42L52_PWRCTL1_PDN_ALL 0x9F +#define CS42L52_PWRCTL1_PDN_CHRG 0x80 +#define CS42L52_PWRCTL1_PDN_PGAB 0x10 +#define CS42L52_PWRCTL1_PDN_PGAA 0x08 +#define CS42L52_PWRCTL1_PDN_ADCB 0x04 +#define CS42L52_PWRCTL1_PDN_ADCA 0x02 +#define CS42L52_PWRCTL1_PDN_CODEC 0x01 + +#define CS42L52_PWRCTL2 0x03 +#define CS42L52_PWRCTL2_OVRDB (1 << 4) +#define CS42L52_PWRCTL2_OVRDA (1 << 3) +#define CS42L52_PWRCTL2_PDN_MICB (1 << 2) +#define CS42L52_PWRCTL2_PDN_MICB_SHIFT 2 +#define CS42L52_PWRCTL2_PDN_MICA (1 << 1) +#define CS42L52_PWRCTL2_PDN_MICA_SHIFT 1 +#define CS42L52_PWRCTL2_PDN_MICBIAS (1 << 0) +#define CS42L52_PWRCTL2_PDN_MICBIAS_SHIFT 0 + +#define CS42L52_PWRCTL3 0x04 +#define CS42L52_PWRCTL3_HPB_PDN_SHIFT 6 +#define CS42L52_PWRCTL3_HPB_ON_LOW 0x00 +#define CS42L52_PWRCTL3_HPB_ON_HIGH 0x01 +#define CS42L52_PWRCTL3_HPB_ALWAYS_ON 0x02 +#define CS42L52_PWRCTL3_HPB_ALWAYS_OFF 0x03 +#define CS42L52_PWRCTL3_HPA_PDN_SHIFT 4 +#define CS42L52_PWRCTL3_HPA_ON_LOW 0x00 +#define CS42L52_PWRCTL3_HPA_ON_HIGH 0x01 +#define CS42L52_PWRCTL3_HPA_ALWAYS_ON 0x02 +#define CS42L52_PWRCTL3_HPA_ALWAYS_OFF 0x03 +#define CS42L52_PWRCTL3_SPKB_PDN_SHIFT 2 +#define CS42L52_PWRCTL3_SPKB_ON_LOW 0x00 +#define CS42L52_PWRCTL3_SPKB_ON_HIGH 0x01 +#define CS42L52_PWRCTL3_SPKB_ALWAYS_ON 0x02 +#define CS42L52_PWRCTL3_PDN_SPKB (1 << 2) +#define CS42L52_PWRCTL3_PDN_SPKA (1 << 0) +#define CS42L52_PWRCTL3_SPKA_PDN_SHIFT 0 +#define CS42L52_PWRCTL3_SPKA_ON_LOW 0x00 +#define CS42L52_PWRCTL3_SPKA_ON_HIGH 0x01 +#define CS42L52_PWRCTL3_SPKA_ALWAYS_ON 0x02 + +#define CS42L52_DEFAULT_OUTPUT_STATE 0x05 +#define CS42L52_PWRCTL3_CONF_MASK 0x03 + +#define CS42L52_CLK_CTL 0x05 +#define CLK_AUTODECT_ENABLE (1 << 7) +#define CLK_SPEED_SHIFT 5 +#define CLK_DS_MODE 0x00 +#define CLK_SS_MODE 0x01 +#define CLK_HS_MODE 0x02 +#define CLK_QS_MODE 0x03 +#define CLK_32K_SR_SHIFT 4 +#define CLK_32K 0x01 +#define CLK_NO_32K 0x00 +#define CLK_27M_MCLK_SHIFT 3 +#define CLK_27M_MCLK 0x01 +#define CLK_NO_27M 0x00 +#define CLK_RATIO_SHIFT 1 +#define CLK_R_128 0x00 +#define CLK_R_125 0x01 +#define CLK_R_132 0x02 +#define CLK_R_136 0x03 + +#define CS42L52_IFACE_CTL1 0x06 +#define CS42L52_IFACE_CTL1_MASTER (1 << 7) +#define CS42L52_IFACE_CTL1_SLAVE (0 << 7) +#define CS42L52_IFACE_CTL1_INV_SCLK (1 << 6) +#define CS42L52_IFACE_CTL1_ADC_FMT_I2S (1 << 5) +#define CS42L52_IFACE_CTL1_ADC_FMT_LEFT_J (0 << 5) +#define CS42L52_IFACE_CTL1_DSP_MODE_EN (1 << 4) +#define CS42L52_IFACE_CTL1_DAC_FMT_LEFT_J (0 << 2) +#define CS42L52_IFACE_CTL1_DAC_FMT_I2S (1 << 2) +#define CS42L52_IFACE_CTL1_DAC_FMT_RIGHT_J (2 << 2) +#define CS42L52_IFACE_CTL1_WL_32BIT (0x00) +#define CS42L52_IFACE_CTL1_WL_24BIT (0x01) +#define CS42L52_IFACE_CTL1_WL_20BIT (0x02) +#define CS42L52_IFACE_CTL1_WL_16BIT (0x03) +#define CS42L52_IFACE_CTL1_WL_MASK 0xFFFF + +#define CS42L52_IFACE_CTL2 0x07 +#define CS42L52_IFACE_CTL2_SC_MC_EQ (1 << 6) +#define CS42L52_IFACE_CTL2_LOOPBACK (1 << 5) +#define CS42L52_IFACE_CTL2_S_MODE_OUTPUT_EN (0 << 4) +#define CS42L52_IFACE_CTL2_S_MODE_OUTPUT_HIZ (1 << 4) +#define CS42L52_IFACE_CTL2_HP_SW_INV (1 << 3) +#define CS42L52_IFACE_CTL2_BIAS_LVL 0x07 + +#define CS42L52_ADC_PGA_A 0x08 +#define CS42L52_ADC_PGA_B 0x09 +#define CS42L52_ADC_SEL_SHIFT 5 +#define CS42L52_ADC_SEL_AIN1 0x00 +#define CS42L52_ADC_SEL_AIN2 0x01 +#define CS42L52_ADC_SEL_AIN3 0x02 +#define CS42L52_ADC_SEL_AIN4 0x03 +#define CS42L52_ADC_SEL_PGA 0x04 + +#define CS42L52_ANALOG_HPF_CTL 0x0A +#define CS42L52_HPF_CTL_ANLGSFTB (1 << 3) +#define CS42L52_HPF_CTL_ANLGSFTA (1 << 0) + +#define CS42L52_ADC_HPF_FREQ 0x0B +#define CS42L52_ADC_MISC_CTL 0x0C +#define CS42L52_ADC_MISC_CTL_SOURCE_DSP (1 << 6) + +#define CS42L52_PB_CTL1 0x0D +#define CS42L52_PB_CTL1_HP_GAIN_SHIFT 5 +#define CS42L52_PB_CTL1_HP_GAIN_03959 0x00 +#define CS42L52_PB_CTL1_HP_GAIN_04571 0x01 +#define CS42L52_PB_CTL1_HP_GAIN_05111 0x02 +#define CS42L52_PB_CTL1_HP_GAIN_06047 0x03 +#define CS42L52_PB_CTL1_HP_GAIN_07099 0x04 +#define CS42L52_PB_CTL1_HP_GAIN_08399 0x05 +#define CS42L52_PB_CTL1_HP_GAIN_10000 0x06 +#define CS42L52_PB_CTL1_HP_GAIN_11430 0x07 +#define CS42L52_PB_CTL1_INV_PCMB (1 << 3) +#define CS42L52_PB_CTL1_INV_PCMA (1 << 2) +#define CS42L52_PB_CTL1_MSTB_MUTE (1 << 1) +#define CS42L52_PB_CTL1_MSTA_MUTE (1 << 0) +#define CS42L52_PB_CTL1_MUTE_MASK 0xFFFD +#define CS42L52_PB_CTL1_MUTE 3 +#define CS42L52_PB_CTL1_UNMUTE 0 + +#define CS42L52_MISC_CTL 0x0E +#define CS42L52_MISC_CTL_DEEMPH (1 << 2) +#define CS42L52_MISC_CTL_DIGSFT (1 << 1) +#define CS42L52_MISC_CTL_DIGZC (1 << 0) + +#define CS42L52_PB_CTL2 0x0F +#define CS42L52_PB_CTL2_HPB_MUTE (1 << 7) +#define CS42L52_PB_CTL2_HPA_MUTE (1 << 6) +#define CS42L52_PB_CTL2_SPKB_MUTE (1 << 5) +#define CS42L52_PB_CTL2_SPKA_MUTE (1 << 4) +#define CS42L52_PB_CTL2_SPK_SWAP (1 << 2) +#define CS42L52_PB_CTL2_SPK_MONO (1 << 1) +#define CS42L52_PB_CTL2_SPK_MUTE50 (1 << 0) + +#define CS42L52_MICA_CTL 0x10 +#define CS42L52_MICB_CTL 0x11 +#define CS42L52_MIC_CTL_MIC_SEL_MASK 0xBF +#define CS42L52_MIC_CTL_MIC_SEL_SHIFT 6 +#define CS42L52_MIC_CTL_TYPE_MASK 0xDF +#define CS42L52_MIC_CTL_TYPE_SHIFT 5 + + +#define CS42L52_PGAA_CTL 0x12 +#define CS42L52_PGAB_CTL 0x13 +#define CS42L52_PGAX_CTL_VOL_12DB 24 +#define CS42L52_PGAX_CTL_VOL_6DB 12 /*step size 0.5db*/ + +#define CS42L52_PASSTHRUA_VOL 0x14 +#define CS42L52_PASSTHRUB_VOL 0x15 + +#define CS42L52_ADCA_VOL 0x16 +#define CS42L52_ADCB_VOL 0x17 +#define CS42L52_ADCX_VOL_24DB 24 /*step size 1db*/ +#define CS42L52_ADCX_VOL_12DB 12 +#define CS42L52_ADCX_VOL_6DB 6 + +#define CS42L52_ADCA_MIXER_VOL 0x18 +#define CS42L52_ADCB_MIXER_VOL 0x19 +#define CS42L52_ADC_MIXER_VOL_12DB 0x18 + +#define CS42L52_PCMA_MIXER_VOL 0x1A +#define CS42L52_PCMB_MIXER_VOL 0x1B + +#define CS42L52_BEEP_FREQ 0x1C +#define CS42L52_BEEP_VOL 0x1D +#define CS42L52_BEEP_TONE_CTL 0x1E +#define CS42L52_BEEP_RATE_SHIFT 4 +#define CS42L52_BEEP_RATE_MASK 0x0F + +#define CS42L52_TONE_CTL 0x1F +#define CS42L52_BEEP_EN_MASK 0x3F + +#define CS42L52_MASTERA_VOL 0x20 +#define CS42L52_MASTERB_VOL 0x21 + +#define CS42L52_HPA_VOL 0x22 +#define CS42L52_HPB_VOL 0x23 +#define CS42L52_DEFAULT_HP_VOL 0xF0 + +#define CS42L52_SPKA_VOL 0x24 +#define CS42L52_SPKB_VOL 0x25 +#define CS42L52_DEFAULT_SPK_VOL 0xF0 + +#define CS42L52_ADC_PCM_MIXER 0x26 + +#define CS42L52_LIMITER_CTL1 0x27 +#define CS42L52_LIMITER_CTL2 0x28 +#define CS42L52_LIMITER_AT_RATE 0x29 + +#define CS42L52_ALC_CTL 0x2A +#define CS42L52_ALC_CTL_ALCB_ENABLE_SHIFT 7 +#define CS42L52_ALC_CTL_ALCA_ENABLE_SHIFT 6 +#define CS42L52_ALC_CTL_FASTEST_ATTACK 0 + +#define CS42L52_ALC_RATE 0x2B +#define CS42L52_ALC_SLOWEST_RELEASE 0x3F + +#define CS42L52_ALC_THRESHOLD 0x2C +#define CS42L52_ALC_MAX_RATE_SHIFT 5 +#define CS42L52_ALC_MIN_RATE_SHIFT 2 +#define CS42L52_ALC_RATE_0DB 0 +#define CS42L52_ALC_RATE_3DB 1 +#define CS42L52_ALC_RATE_6DB 2 + +#define CS42L52_NOISE_GATE_CTL 0x2D +#define CS42L52_NG_ENABLE_SHIFT 6 +#define CS42L52_NG_THRESHOLD_SHIFT 2 +#define CS42L52_NG_MIN_70DB 2 +#define CS42L52_NG_DELAY_SHIFT 0 +#define CS42L52_NG_DELAY_100MS 1 + +#define CS42L52_CLK_STATUS 0x2E +#define CS42L52_BATT_COMPEN 0x2F + +#define CS42L52_BATT_LEVEL 0x30 +#define CS42L52_SPK_STATUS 0x31 +#define CS42L52_SPK_STATUS_PIN_SHIFT 3 +#define CS42L52_SPK_STATUS_PIN_HIGH 1 + +#define CS42L52_TEM_CTL 0x32 +#define CS42L52_TEM_CTL_SET 0x80 +#define CS42L52_THE_FOLDBACK 0x33 +#define CS42L52_CHARGE_PUMP 0x34 +#define CS42L52_CHARGE_PUMP_MASK 0xF0 +#define CS42L52_CHARGE_PUMP_SHIFT 4 +#define CS42L52_FIX_BITS1 0x3E +#define CS42L52_FIX_BITS2 0x47 + +#define CS42L52_MAX_REGISTER 0x34 + +#endif diff --git a/sound/soc/codecs/cs42l73.c b/sound/soc/codecs/cs42l73.c index 3686417f5ea..e0d45fdaa75 100644 --- a/sound/soc/codecs/cs42l73.c +++ b/sound/soc/codecs/cs42l73.c @@ -43,9 +43,6 @@ struct cs42l73_private { }; static const struct reg_default cs42l73_reg_defaults[] = { - { 1, 0x42 }, /* r01 - Device ID A&B */ - { 2, 0xA7 }, /* r02 - Device ID C&D */ - { 3, 0x30 }, /* r03 - Device ID E */ { 6, 0xF1 }, /* r06 - Power Ctl 1 */ { 7, 0xDF }, /* r07 - Power Ctl 2 */ { 8, 0x3F }, /* r08 - Power Ctl 3 */ @@ -402,37 +399,37 @@ static const struct snd_kcontrol_new ear_amp_ctl = static const struct snd_kcontrol_new cs42l73_snd_controls[] = { SOC_DOUBLE_R_SX_TLV("Headphone Analog Playback Volume", - CS42L73_HPAAVOL, CS42L73_HPBAVOL, 7, - 0xffffffC1, 0x0C, hpaloa_tlv), + CS42L73_HPAAVOL, CS42L73_HPBAVOL, 0, + 0x41, 0x4B, hpaloa_tlv), SOC_DOUBLE_R_SX_TLV("LineOut Analog Playback Volume", CS42L73_LOAAVOL, - CS42L73_LOBAVOL, 7, 0xffffffC1, 0x0C, hpaloa_tlv), + CS42L73_LOBAVOL, 0, 0x41, 0x4B, hpaloa_tlv), SOC_DOUBLE_R_SX_TLV("Input PGA Analog Volume", CS42L73_MICAPREPGAAVOL, - CS42L73_MICBPREPGABVOL, 5, 0xffffff35, - 0x34, micpga_tlv), + CS42L73_MICBPREPGABVOL, 5, 0x34, + 0x24, micpga_tlv), SOC_DOUBLE_R("MIC Preamp Switch", CS42L73_MICAPREPGAAVOL, CS42L73_MICBPREPGABVOL, 6, 1, 1), SOC_DOUBLE_R_SX_TLV("Input Path Digital Volume", CS42L73_IPADVOL, - CS42L73_IPBDVOL, 7, 0xffffffA0, 0xA0, ipd_tlv), + CS42L73_IPBDVOL, 0, 0xA0, 0x6C, ipd_tlv), SOC_DOUBLE_R_SX_TLV("HL Digital Playback Volume", - CS42L73_HLADVOL, CS42L73_HLBDVOL, 7, 0xffffffE5, - 0xE4, hl_tlv), + CS42L73_HLADVOL, CS42L73_HLBDVOL, + 0, 0x34, 0xE4, hl_tlv), SOC_SINGLE_TLV("ADC A Boost Volume", CS42L73_ADCIPC, 2, 0x01, 1, adc_boost_tlv), SOC_SINGLE_TLV("ADC B Boost Volume", - CS42L73_ADCIPC, 6, 0x01, 1, adc_boost_tlv), + CS42L73_ADCIPC, 6, 0x01, 1, adc_boost_tlv), - SOC_SINGLE_TLV("Speakerphone Digital Playback Volume", - CS42L73_SPKDVOL, 0, 0xE4, 1, hl_tlv), + SOC_SINGLE_SX_TLV("Speakerphone Digital Volume", + CS42L73_SPKDVOL, 0, 0x34, 0xE4, hl_tlv), - SOC_SINGLE_TLV("Ear Speaker Digital Playback Volume", - CS42L73_ESLDVOL, 0, 0xE4, 1, hl_tlv), + SOC_SINGLE_SX_TLV("Ear Speaker Digital Volume", + CS42L73_ESLDVOL, 0, 0x34, 0xE4, hl_tlv), SOC_DOUBLE_R("Headphone Analog Playback Switch", CS42L73_HPAAVOL, CS42L73_HPBAVOL, 7, 1, 1), @@ -599,17 +596,17 @@ static const struct snd_soc_dapm_widget cs42l73_dapm_widgets[] = { SND_SOC_DAPM_INPUT("MIC2"), SND_SOC_DAPM_SUPPLY("MIC2 Bias", CS42L73_PWRCTL2, 7, 1, NULL, 0), - SND_SOC_DAPM_AIF_OUT("XSPOUTL", "XSP Capture", 0, + SND_SOC_DAPM_AIF_OUT("XSPOUTL", NULL, 0, CS42L73_PWRCTL2, 1, 1), - SND_SOC_DAPM_AIF_OUT("XSPOUTR", "XSP Capture", 0, + SND_SOC_DAPM_AIF_OUT("XSPOUTR", NULL, 0, CS42L73_PWRCTL2, 1, 1), - SND_SOC_DAPM_AIF_OUT("ASPOUTL", "ASP Capture", 0, + SND_SOC_DAPM_AIF_OUT("ASPOUTL", NULL, 0, CS42L73_PWRCTL2, 3, 1), - SND_SOC_DAPM_AIF_OUT("ASPOUTR", "ASP Capture", 0, + SND_SOC_DAPM_AIF_OUT("ASPOUTR", NULL, 0, CS42L73_PWRCTL2, 3, 1), - SND_SOC_DAPM_AIF_OUT("VSPOUTL", "VSP Capture", 0, + SND_SOC_DAPM_AIF_OUT("VSPOUTL", NULL, 0, CS42L73_PWRCTL2, 4, 1), - SND_SOC_DAPM_AIF_OUT("VSPOUTR", "VSP Capture", 0, + SND_SOC_DAPM_AIF_OUT("VSPOUTR", NULL, 0, CS42L73_PWRCTL2, 4, 1), SND_SOC_DAPM_PGA("PGA Left", SND_SOC_NOPM, 0, 0, NULL, 0), @@ -638,21 +635,21 @@ static const struct snd_soc_dapm_widget cs42l73_dapm_widgets[] = { SND_SOC_DAPM_MIXER("VSPL Output Mixer", SND_SOC_NOPM, 0, 0, NULL, 0), SND_SOC_DAPM_MIXER("VSPR Output Mixer", SND_SOC_NOPM, 0, 0, NULL, 0), - SND_SOC_DAPM_AIF_IN("XSPINL", "XSP Playback", 0, + SND_SOC_DAPM_AIF_IN("XSPINL", NULL, 0, CS42L73_PWRCTL2, 0, 1), - SND_SOC_DAPM_AIF_IN("XSPINR", "XSP Playback", 0, + SND_SOC_DAPM_AIF_IN("XSPINR", NULL, 0, CS42L73_PWRCTL2, 0, 1), - SND_SOC_DAPM_AIF_IN("XSPINM", "XSP Playback", 0, + SND_SOC_DAPM_AIF_IN("XSPINM", NULL, 0, CS42L73_PWRCTL2, 0, 1), - SND_SOC_DAPM_AIF_IN("ASPINL", "ASP Playback", 0, + SND_SOC_DAPM_AIF_IN("ASPINL", NULL, 0, CS42L73_PWRCTL2, 2, 1), - SND_SOC_DAPM_AIF_IN("ASPINR", "ASP Playback", 0, + SND_SOC_DAPM_AIF_IN("ASPINR", NULL, 0, CS42L73_PWRCTL2, 2, 1), - SND_SOC_DAPM_AIF_IN("ASPINM", "ASP Playback", 0, + SND_SOC_DAPM_AIF_IN("ASPINM", NULL, 0, CS42L73_PWRCTL2, 2, 1), - SND_SOC_DAPM_AIF_IN("VSPIN", "VSP Playback", 0, + SND_SOC_DAPM_AIF_IN("VSPIN", NULL, 0, CS42L73_PWRCTL2, 4, 1), SND_SOC_DAPM_MIXER("HL Left Mixer", SND_SOC_NOPM, 0, 0, NULL, 0), @@ -776,6 +773,14 @@ static const struct snd_soc_dapm_route cs42l73_audio_map[] = { {"HL Left Mixer", NULL, "VSPIN"}, {"HL Right Mixer", NULL, "VSPIN"}, + {"ASPINL", NULL, "ASP Playback"}, + {"ASPINM", NULL, "ASP Playback"}, + {"ASPINR", NULL, "ASP Playback"}, + {"XSPINL", NULL, "XSP Playback"}, + {"XSPINM", NULL, "XSP Playback"}, + {"XSPINR", NULL, "XSP Playback"}, + {"VSPIN", NULL, "VSP Playback"}, + /* Capture Paths */ {"MIC1", NULL, "MIC1 Bias"}, {"PGA Left Mux", "Mic 1", "MIC1"}, @@ -822,6 +827,13 @@ static const struct snd_soc_dapm_route cs42l73_audio_map[] = { {"VSPOUTL", NULL, "VSPL Output Mixer"}, {"VSPOUTR", NULL, "VSPR Output Mixer"}, + + {"ASP Capture", NULL, "ASPOUTL"}, + {"ASP Capture", NULL, "ASPOUTR"}, + {"XSP Capture", NULL, "XSPOUTL"}, + {"XSP Capture", NULL, "XSPOUTR"}, + {"VSP Capture", NULL, "VSPOUTL"}, + {"VSP Capture", NULL, "VSPOUTR"}, }; struct cs42l73_mclk_div { @@ -1091,8 +1103,7 @@ static int cs42l73_pcm_hw_params(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *params, struct snd_soc_dai *dai) { - struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct snd_soc_codec *codec = rtd->codec; + struct snd_soc_codec *codec = dai->codec; struct cs42l73_private *priv = snd_soc_codec_get_drvdata(codec); int id = dai->id; int mclk_coeff; @@ -1429,25 +1440,7 @@ static struct i2c_driver cs42l73_i2c_driver = { }; -static int __init cs42l73_modinit(void) -{ - int ret; - ret = i2c_add_driver(&cs42l73_i2c_driver); - if (ret != 0) { - pr_err("Failed to register CS42L73 I2C driver: %d\n", ret); - return ret; - } - return 0; -} - -module_init(cs42l73_modinit); - -static void __exit cs42l73_exit(void) -{ - i2c_del_driver(&cs42l73_i2c_driver); -} - -module_exit(cs42l73_exit); +module_i2c_driver(cs42l73_i2c_driver); MODULE_DESCRIPTION("ASoC CS42L73 driver"); MODULE_AUTHOR("Georgi Vlaev, Nucleus Systems Ltd, <joe@nucleusys.com>"); diff --git a/sound/soc/codecs/da7210.c b/sound/soc/codecs/da7210.c index 7843711729b..af5db708051 100644 --- a/sound/soc/codecs/da7210.c +++ b/sound/soc/codecs/da7210.c @@ -17,6 +17,7 @@ #include <linux/delay.h> #include <linux/i2c.h> +#include <linux/spi/spi.h> #include <linux/regmap.h> #include <linux/slab.h> #include <linux/module.h> @@ -27,6 +28,7 @@ #include <sound/tlv.h> /* DA7210 register space */ +#define DA7210_PAGE_CONTROL 0x00 #define DA7210_CONTROL 0x01 #define DA7210_STATUS 0x02 #define DA7210_STARTUP1 0x03 @@ -146,6 +148,7 @@ #define DA7210_DAI_EN (1 << 7) /*PLL_DIV3 bit fields */ +#define DA7210_PLL_DIV_L_MASK (0xF << 0) #define DA7210_MCLK_RANGE_10_20_MHZ (1 << 4) #define DA7210_PLL_BYP (1 << 6) @@ -162,12 +165,16 @@ #define DA7210_PLL_FS_48000 (0xB << 0) #define DA7210_PLL_FS_88200 (0xE << 0) #define DA7210_PLL_FS_96000 (0xF << 0) +#define DA7210_MCLK_DET_EN (0x1 << 5) +#define DA7210_MCLK_SRM_EN (0x1 << 6) #define DA7210_PLL_EN (0x1 << 7) /* SOFTMUTE bit fields */ #define DA7210_RAMP_EN (1 << 6) /* CONTROL bit fields */ +#define DA7210_REG_EN (1 << 0) +#define DA7210_BIAS_EN (1 << 2) #define DA7210_NOISE_SUP_EN (1 << 3) /* IN_GAIN bit fields */ @@ -206,6 +213,47 @@ #define DA7210_OUT2_OUTMIX_L (1 << 6) #define DA7210_OUT2_EN (1 << 7) +struct pll_div { + int fref; + int fout; + u8 div1; + u8 div2; + u8 div3; + u8 mode; /* 0 = slave, 1 = master */ +}; + +/* PLL dividers table */ +static const struct pll_div da7210_pll_div[] = { + /* for MASTER mode, fs = 44.1Khz */ + { 12000000, 2822400, 0xE8, 0x6C, 0x2, 1}, /* MCLK=12Mhz */ + { 13000000, 2822400, 0xDF, 0x28, 0xC, 1}, /* MCLK=13Mhz */ + { 13500000, 2822400, 0xDB, 0x0A, 0xD, 1}, /* MCLK=13.5Mhz */ + { 14400000, 2822400, 0xD4, 0x5A, 0x2, 1}, /* MCLK=14.4Mhz */ + { 19200000, 2822400, 0xBB, 0x43, 0x9, 1}, /* MCLK=19.2Mhz */ + { 19680000, 2822400, 0xB9, 0x6D, 0xA, 1}, /* MCLK=19.68Mhz */ + { 19800000, 2822400, 0xB8, 0xFB, 0xB, 1}, /* MCLK=19.8Mhz */ + /* for MASTER mode, fs = 48Khz */ + { 12000000, 3072000, 0xF3, 0x12, 0x7, 1}, /* MCLK=12Mhz */ + { 13000000, 3072000, 0xE8, 0xFD, 0x5, 1}, /* MCLK=13Mhz */ + { 13500000, 3072000, 0xE4, 0x82, 0x3, 1}, /* MCLK=13.5Mhz */ + { 14400000, 3072000, 0xDD, 0x3A, 0x0, 1}, /* MCLK=14.4Mhz */ + { 19200000, 3072000, 0xC1, 0xEB, 0x8, 1}, /* MCLK=19.2Mhz */ + { 19680000, 3072000, 0xBF, 0xEC, 0x0, 1}, /* MCLK=19.68Mhz */ + { 19800000, 3072000, 0xBF, 0x70, 0x0, 1}, /* MCLK=19.8Mhz */ + /* for SLAVE mode with SRM */ + { 12000000, 2822400, 0xED, 0xBF, 0x5, 0}, /* MCLK=12Mhz */ + { 13000000, 2822400, 0xE4, 0x13, 0x0, 0}, /* MCLK=13Mhz */ + { 13500000, 2822400, 0xDF, 0xC6, 0x8, 0}, /* MCLK=13.5Mhz */ + { 14400000, 2822400, 0xD8, 0xCA, 0x1, 0}, /* MCLK=14.4Mhz */ + { 19200000, 2822400, 0xBE, 0x97, 0x9, 0}, /* MCLK=19.2Mhz */ + { 19680000, 2822400, 0xBC, 0xAC, 0xD, 0}, /* MCLK=19.68Mhz */ + { 19800000, 2822400, 0xBC, 0x35, 0xE, 0}, /* MCLK=19.8Mhz */ +}; + +enum clk_src { + DA7210_CLKSRC_MCLK +}; + #define DA7210_VERSION "0.0.1" /* @@ -628,9 +676,12 @@ static const struct snd_soc_dapm_route da7210_audio_map[] = { /* Codec private data */ struct da7210_priv { struct regmap *regmap; + unsigned int mclk_rate; + int master; }; static struct reg_default da7210_reg_defaults[] = { + { 0x00, 0x00 }, { 0x01, 0x11 }, { 0x03, 0x00 }, { 0x04, 0x00 }, @@ -713,10 +764,10 @@ static int da7210_hw_params(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *params, struct snd_soc_dai *dai) { - struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct snd_soc_codec *codec = rtd->codec; + struct snd_soc_codec *codec = dai->codec; + struct da7210_priv *da7210 = snd_soc_codec_get_drvdata(codec); u32 dai_cfg1; - u32 fs, bypass; + u32 fs, sysclk; /* set DAI source to Left and Right ADC */ snd_soc_write(codec, DA7210_DAI_SRC_SEL, @@ -749,43 +800,43 @@ static int da7210_hw_params(struct snd_pcm_substream *substream, switch (params_rate(params)) { case 8000: fs = DA7210_PLL_FS_8000; - bypass = DA7210_PLL_BYP; + sysclk = 3072000; break; case 11025: fs = DA7210_PLL_FS_11025; - bypass = 0; + sysclk = 2822400; break; case 12000: fs = DA7210_PLL_FS_12000; - bypass = DA7210_PLL_BYP; + sysclk = 3072000; break; case 16000: fs = DA7210_PLL_FS_16000; - bypass = DA7210_PLL_BYP; + sysclk = 3072000; break; case 22050: fs = DA7210_PLL_FS_22050; - bypass = 0; + sysclk = 2822400; break; case 32000: fs = DA7210_PLL_FS_32000; - bypass = DA7210_PLL_BYP; + sysclk = 3072000; break; case 44100: fs = DA7210_PLL_FS_44100; - bypass = 0; + sysclk = 2822400; break; case 48000: fs = DA7210_PLL_FS_48000; - bypass = DA7210_PLL_BYP; + sysclk = 3072000; break; case 88200: fs = DA7210_PLL_FS_88200; - bypass = 0; + sysclk = 2822400; break; case 96000: fs = DA7210_PLL_FS_96000; - bypass = DA7210_PLL_BYP; + sysclk = 3072000; break; default: return -EINVAL; @@ -795,8 +846,26 @@ static int da7210_hw_params(struct snd_pcm_substream *substream, snd_soc_update_bits(codec, DA7210_STARTUP1, DA7210_SC_MST_EN, 0); snd_soc_update_bits(codec, DA7210_PLL, DA7210_PLL_FS_MASK, fs); - snd_soc_update_bits(codec, DA7210_PLL_DIV3, DA7210_PLL_BYP, bypass); + if (da7210->mclk_rate && (da7210->mclk_rate != sysclk)) { + /* PLL mode, disable PLL bypass */ + snd_soc_update_bits(codec, DA7210_PLL_DIV3, DA7210_PLL_BYP, 0); + + if (!da7210->master) { + /* PLL slave mode, also enable SRM */ + snd_soc_update_bits(codec, DA7210_PLL, + (DA7210_MCLK_SRM_EN | + DA7210_MCLK_DET_EN), + (DA7210_MCLK_SRM_EN | + DA7210_MCLK_DET_EN)); + } + } else { + /* PLL bypass mode, enable PLL bypass and Auto Detection */ + snd_soc_update_bits(codec, DA7210_PLL, DA7210_MCLK_DET_EN, + DA7210_MCLK_DET_EN); + snd_soc_update_bits(codec, DA7210_PLL_DIV3, DA7210_PLL_BYP, + DA7210_PLL_BYP); + } /* Enable active mode */ snd_soc_update_bits(codec, DA7210_STARTUP1, DA7210_SC_MST_EN, DA7210_SC_MST_EN); @@ -810,17 +879,24 @@ static int da7210_hw_params(struct snd_pcm_substream *substream, static int da7210_set_dai_fmt(struct snd_soc_dai *codec_dai, u32 fmt) { struct snd_soc_codec *codec = codec_dai->codec; + struct da7210_priv *da7210 = snd_soc_codec_get_drvdata(codec); u32 dai_cfg1; u32 dai_cfg3; dai_cfg1 = 0x7f & snd_soc_read(codec, DA7210_DAI_CFG1); dai_cfg3 = 0xfc & snd_soc_read(codec, DA7210_DAI_CFG3); + if ((snd_soc_read(codec, DA7210_PLL) & DA7210_PLL_EN) && + (!(snd_soc_read(codec, DA7210_PLL_DIV3) & DA7210_PLL_BYP))) + return -EINVAL; + switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) { case SND_SOC_DAIFMT_CBM_CFM: + da7210->master = 1; dai_cfg1 |= DA7210_DAI_MODE_MASTER; break; case SND_SOC_DAIFMT_CBS_CFS: + da7210->master = 0; dai_cfg1 |= DA7210_DAI_MODE_SLAVE; break; default: @@ -872,10 +948,101 @@ static int da7210_mute(struct snd_soc_dai *dai, int mute) #define DA7210_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S20_3LE |\ SNDRV_PCM_FMTBIT_S24_LE | SNDRV_PCM_FMTBIT_S32_LE) +static int da7210_set_dai_sysclk(struct snd_soc_dai *codec_dai, + int clk_id, unsigned int freq, int dir) +{ + struct snd_soc_codec *codec = codec_dai->codec; + struct da7210_priv *da7210 = snd_soc_codec_get_drvdata(codec); + + switch (clk_id) { + case DA7210_CLKSRC_MCLK: + switch (freq) { + case 12000000: + case 13000000: + case 13500000: + case 14400000: + case 19200000: + case 19680000: + case 19800000: + da7210->mclk_rate = freq; + return 0; + default: + dev_err(codec_dai->dev, "Unsupported MCLK value %d\n", + freq); + return -EINVAL; + } + break; + default: + dev_err(codec_dai->dev, "Unknown clock source %d\n", clk_id); + return -EINVAL; + } +} + +/** + * da7210_set_dai_pll :Configure the codec PLL + * @param codec_dai : pointer to codec DAI + * @param pll_id : da7210 has only one pll, so pll_id is always zero + * @param fref : MCLK frequency, should be < 20MHz + * @param fout : FsDM value, Refer page 44 & 45 of datasheet + * @return int : Zero for success, negative error code for error + * + * Note: Supported PLL input frequencies are 12MHz, 13MHz, 13.5MHz, 14.4MHz, + * 19.2MHz, 19.6MHz and 19.8MHz + */ +static int da7210_set_dai_pll(struct snd_soc_dai *codec_dai, int pll_id, + int source, unsigned int fref, unsigned int fout) +{ + struct snd_soc_codec *codec = codec_dai->codec; + struct da7210_priv *da7210 = snd_soc_codec_get_drvdata(codec); + + u8 pll_div1, pll_div2, pll_div3, cnt; + + /* In slave mode, there is only one set of divisors */ + if (!da7210->master) + fout = 2822400; + + /* Search pll div array for correct divisors */ + for (cnt = 0; cnt < ARRAY_SIZE(da7210_pll_div); cnt++) { + /* check fref, mode and fout */ + if ((fref == da7210_pll_div[cnt].fref) && + (da7210->master == da7210_pll_div[cnt].mode) && + (fout == da7210_pll_div[cnt].fout)) { + /* all match, pick up divisors */ + pll_div1 = da7210_pll_div[cnt].div1; + pll_div2 = da7210_pll_div[cnt].div2; + pll_div3 = da7210_pll_div[cnt].div3; + break; + } + } + if (cnt >= ARRAY_SIZE(da7210_pll_div)) + goto err; + + /* Disable active mode */ + snd_soc_update_bits(codec, DA7210_STARTUP1, DA7210_SC_MST_EN, 0); + /* Write PLL dividers */ + snd_soc_write(codec, DA7210_PLL_DIV1, pll_div1); + snd_soc_write(codec, DA7210_PLL_DIV2, pll_div2); + snd_soc_update_bits(codec, DA7210_PLL_DIV3, + DA7210_PLL_DIV_L_MASK, pll_div3); + + /* Enable PLL */ + snd_soc_update_bits(codec, DA7210_PLL, DA7210_PLL_EN, DA7210_PLL_EN); + + /* Enable active mode */ + snd_soc_update_bits(codec, DA7210_STARTUP1, DA7210_SC_MST_EN, + DA7210_SC_MST_EN); + return 0; +err: + dev_err(codec_dai->dev, "Unsupported PLL input frequency %d\n", fref); + return -EINVAL; +} + /* DAI operations */ static const struct snd_soc_dai_ops da7210_dai_ops = { .hw_params = da7210_hw_params, .set_fmt = da7210_set_dai_fmt, + .set_sysclk = da7210_set_dai_sysclk, + .set_pll = da7210_set_dai_pll, .digital_mute = da7210_mute, }; @@ -915,24 +1082,11 @@ static int da7210_probe(struct snd_soc_codec *codec) return ret; } - /* FIXME - * - * This driver use fixed value here - * And below settings expects MCLK = 12.288MHz - * - * When you select different MCLK, please check... - * DA7210_PLL_DIV1 val - * DA7210_PLL_DIV2 val - * DA7210_PLL_DIV3 val - * DA7210_PLL_DIV3 :: DA7210_MCLK_RANGExxx - */ + da7210->mclk_rate = 0; /* This will be set from set_sysclk() */ + da7210->master = 0; /* This will be set from set_fmt() */ - /* - * make sure that DA7210 use bypass mode before start up - */ - snd_soc_write(codec, DA7210_STARTUP1, 0); - snd_soc_write(codec, DA7210_PLL_DIV3, - DA7210_MCLK_RANGE_10_20_MHZ | DA7210_PLL_BYP); + /* Enable internal regulator & bias current */ + snd_soc_write(codec, DA7210_CONTROL, DA7210_REG_EN | DA7210_BIAS_EN); /* * ADC settings @@ -1007,34 +1161,13 @@ static int da7210_probe(struct snd_soc_codec *codec) /* Enable Aux2 */ snd_soc_write(codec, DA7210_AUX2, DA7210_AUX2_EN); + /* Set PLL Master clock range 10-20 MHz, enable PLL bypass */ + snd_soc_write(codec, DA7210_PLL_DIV3, DA7210_MCLK_RANGE_10_20_MHZ | + DA7210_PLL_BYP); + /* Diable PLL and bypass it */ snd_soc_write(codec, DA7210_PLL, DA7210_PLL_FS_48000); - /* - * If 48kHz sound came, it use bypass mode, - * and when it is 44.1kHz, it use PLL. - * - * This time, this driver sets PLL always ON - * and controls bypass/PLL mode by switching - * DA7210_PLL_DIV3 :: DA7210_PLL_BYP bit. - * see da7210_hw_params - */ - snd_soc_write(codec, DA7210_PLL_DIV1, 0xE5); /* MCLK = 12.288MHz */ - snd_soc_write(codec, DA7210_PLL_DIV2, 0x99); - snd_soc_write(codec, DA7210_PLL_DIV3, 0x0A | - DA7210_MCLK_RANGE_10_20_MHZ | DA7210_PLL_BYP); - snd_soc_update_bits(codec, DA7210_PLL, DA7210_PLL_EN, DA7210_PLL_EN); - - /* As suggested by Dialog */ - /* unlock */ - regmap_write(da7210->regmap, DA7210_A_HID_UNLOCK, 0x8B); - regmap_write(da7210->regmap, DA7210_A_TEST_UNLOCK, 0xB4); - regmap_write(da7210->regmap, DA7210_A_PLL1, 0x01); - regmap_write(da7210->regmap, DA7210_A_CP_MODE, 0x7C); - /* re-lock */ - regmap_write(da7210->regmap, DA7210_A_HID_UNLOCK, 0x00); - regmap_write(da7210->regmap, DA7210_A_TEST_UNLOCK, 0x00); - /* Activate all enabled subsystem */ snd_soc_write(codec, DA7210_STARTUP1, DA7210_SC_MST_EN); @@ -1055,7 +1188,26 @@ static struct snd_soc_codec_driver soc_codec_dev_da7210 = { .num_dapm_routes = ARRAY_SIZE(da7210_audio_map), }; -static struct regmap_config da7210_regmap = { +#if defined(CONFIG_I2C) || defined(CONFIG_I2C_MODULE) + +static struct reg_default da7210_regmap_i2c_patch[] = { + + /* System controller master disable */ + { DA7210_STARTUP1, 0x00 }, + /* Set PLL Master clock range 10-20 MHz */ + { DA7210_PLL_DIV3, DA7210_MCLK_RANGE_10_20_MHZ }, + + /* to unlock */ + { DA7210_A_HID_UNLOCK, 0x8B}, + { DA7210_A_TEST_UNLOCK, 0xB4}, + { DA7210_A_PLL1, 0x01}, + { DA7210_A_CP_MODE, 0x7C}, + /* to re-lock */ + { DA7210_A_HID_UNLOCK, 0x00}, + { DA7210_A_TEST_UNLOCK, 0x00}, +}; + +static const struct regmap_config da7210_regmap_config_i2c = { .reg_bits = 8, .val_bits = 8, @@ -1066,7 +1218,6 @@ static struct regmap_config da7210_regmap = { .cache_type = REGCACHE_RBTREE, }; -#if defined(CONFIG_I2C) || defined(CONFIG_I2C_MODULE) static int __devinit da7210_i2c_probe(struct i2c_client *i2c, const struct i2c_device_id *id) { @@ -1080,13 +1231,18 @@ static int __devinit da7210_i2c_probe(struct i2c_client *i2c, i2c_set_clientdata(i2c, da7210); - da7210->regmap = regmap_init_i2c(i2c, &da7210_regmap); + da7210->regmap = regmap_init_i2c(i2c, &da7210_regmap_config_i2c); if (IS_ERR(da7210->regmap)) { ret = PTR_ERR(da7210->regmap); dev_err(&i2c->dev, "regmap_init() failed: %d\n", ret); return ret; } + ret = regmap_register_patch(da7210->regmap, da7210_regmap_i2c_patch, + ARRAY_SIZE(da7210_regmap_i2c_patch)); + if (ret != 0) + dev_warn(&i2c->dev, "Failed to apply regmap patch: %d\n", ret); + ret = snd_soc_register_codec(&i2c->dev, &soc_codec_dev_da7210, &da7210_dai, 1); if (ret < 0) { @@ -1119,7 +1275,7 @@ MODULE_DEVICE_TABLE(i2c, da7210_i2c_id); /* I2C codec control layer */ static struct i2c_driver da7210_i2c_driver = { .driver = { - .name = "da7210-codec", + .name = "da7210", .owner = THIS_MODULE, }, .probe = da7210_i2c_probe, @@ -1128,12 +1284,112 @@ static struct i2c_driver da7210_i2c_driver = { }; #endif +#if defined(CONFIG_SPI_MASTER) + +static struct reg_default da7210_regmap_spi_patch[] = { + /* Dummy read to give two pulses over nCS for SPI */ + { DA7210_AUX2, 0x00 }, + { DA7210_AUX2, 0x00 }, + + /* System controller master disable */ + { DA7210_STARTUP1, 0x00 }, + /* Set PLL Master clock range 10-20 MHz */ + { DA7210_PLL_DIV3, DA7210_MCLK_RANGE_10_20_MHZ }, + + /* to set PAGE1 of SPI register space */ + { DA7210_PAGE_CONTROL, 0x80 }, + /* to unlock */ + { DA7210_A_HID_UNLOCK, 0x8B}, + { DA7210_A_TEST_UNLOCK, 0xB4}, + { DA7210_A_PLL1, 0x01}, + { DA7210_A_CP_MODE, 0x7C}, + /* to re-lock */ + { DA7210_A_HID_UNLOCK, 0x00}, + { DA7210_A_TEST_UNLOCK, 0x00}, + /* to set back PAGE0 of SPI register space */ + { DA7210_PAGE_CONTROL, 0x00 }, +}; + +static const struct regmap_config da7210_regmap_config_spi = { + .reg_bits = 8, + .val_bits = 8, + .read_flag_mask = 0x01, + .write_flag_mask = 0x00, + + .reg_defaults = da7210_reg_defaults, + .num_reg_defaults = ARRAY_SIZE(da7210_reg_defaults), + .volatile_reg = da7210_volatile_register, + .readable_reg = da7210_readable_register, + .cache_type = REGCACHE_RBTREE, +}; + +static int __devinit da7210_spi_probe(struct spi_device *spi) +{ + struct da7210_priv *da7210; + int ret; + + da7210 = devm_kzalloc(&spi->dev, sizeof(struct da7210_priv), + GFP_KERNEL); + if (!da7210) + return -ENOMEM; + + spi_set_drvdata(spi, da7210); + da7210->regmap = devm_regmap_init_spi(spi, &da7210_regmap_config_spi); + if (IS_ERR(da7210->regmap)) { + ret = PTR_ERR(da7210->regmap); + dev_err(&spi->dev, "Failed to register regmap: %d\n", ret); + return ret; + } + + ret = regmap_register_patch(da7210->regmap, da7210_regmap_spi_patch, + ARRAY_SIZE(da7210_regmap_spi_patch)); + if (ret != 0) + dev_warn(&spi->dev, "Failed to apply regmap patch: %d\n", ret); + + ret = snd_soc_register_codec(&spi->dev, + &soc_codec_dev_da7210, &da7210_dai, 1); + if (ret < 0) + goto err_regmap; + + return ret; + +err_regmap: + regmap_exit(da7210->regmap); + + return ret; +} + +static int __devexit da7210_spi_remove(struct spi_device *spi) +{ + struct da7210_priv *da7210 = spi_get_drvdata(spi); + snd_soc_unregister_codec(&spi->dev); + regmap_exit(da7210->regmap); + return 0; +} + +static struct spi_driver da7210_spi_driver = { + .driver = { + .name = "da7210", + .owner = THIS_MODULE, + }, + .probe = da7210_spi_probe, + .remove = __devexit_p(da7210_spi_remove) +}; +#endif + static int __init da7210_modinit(void) { int ret = 0; #if defined(CONFIG_I2C) || defined(CONFIG_I2C_MODULE) ret = i2c_add_driver(&da7210_i2c_driver); #endif +#if defined(CONFIG_SPI_MASTER) + ret = spi_register_driver(&da7210_spi_driver); + if (ret) { + printk(KERN_ERR "Failed to register da7210 SPI driver: %d\n", + ret); + } +#endif return ret; } module_init(da7210_modinit); @@ -1143,6 +1399,9 @@ static void __exit da7210_exit(void) #if defined(CONFIG_I2C) || defined(CONFIG_I2C_MODULE) i2c_del_driver(&da7210_i2c_driver); #endif +#if defined(CONFIG_SPI_MASTER) + spi_unregister_driver(&da7210_spi_driver); +#endif } module_exit(da7210_exit); diff --git a/sound/soc/codecs/jz4740.c b/sound/soc/codecs/jz4740.c index 4624e752a18..85d9cabe6d5 100644 --- a/sound/soc/codecs/jz4740.c +++ b/sound/soc/codecs/jz4740.c @@ -164,8 +164,7 @@ static int jz4740_codec_hw_params(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *params, struct snd_soc_dai *dai) { uint32_t val; - struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct snd_soc_codec *codec =rtd->codec; + struct snd_soc_codec *codec = dai->codec; switch (params_rate(params)) { case 8000: diff --git a/sound/soc/codecs/lm49453.c b/sound/soc/codecs/lm49453.c new file mode 100644 index 00000000000..802b9f176b1 --- /dev/null +++ b/sound/soc/codecs/lm49453.c @@ -0,0 +1,1550 @@ +/* + * lm49453.c - LM49453 ALSA Soc Audio driver + * + * Copyright (c) 2012 Texas Instruments, Inc + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License as published by + * the Free Software Foundation; version 2 of the License. + * + * Initially based on sound/soc/codecs/wm8350.c + */ + +#include <linux/module.h> +#include <linux/moduleparam.h> +#include <linux/version.h> +#include <linux/kernel.h> +#include <linux/init.h> +#include <linux/delay.h> +#include <linux/pm.h> +#include <linux/i2c.h> +#include <linux/regmap.h> +#include <linux/slab.h> +#include <sound/core.h> +#include <sound/pcm.h> +#include <sound/pcm_params.h> +#include <sound/soc.h> +#include <sound/soc-dapm.h> +#include <sound/tlv.h> +#include <sound/jack.h> +#include <sound/initval.h> +#include <asm/div64.h> +#include "lm49453.h" + +static struct reg_default lm49453_reg_defs[] = { + { 0, 0x00 }, + { 1, 0x00 }, + { 2, 0x00 }, + { 3, 0x00 }, + { 4, 0x00 }, + { 5, 0x00 }, + { 6, 0x00 }, + { 7, 0x00 }, + { 8, 0x00 }, + { 9, 0x00 }, + { 10, 0x00 }, + { 11, 0x00 }, + { 12, 0x00 }, + { 13, 0x00 }, + { 14, 0x00 }, + { 15, 0x00 }, + { 16, 0x00 }, + { 17, 0x00 }, + { 18, 0x00 }, + { 19, 0x00 }, + { 20, 0x00 }, + { 21, 0x00 }, + { 22, 0x00 }, + { 23, 0x00 }, + { 32, 0x00 }, + { 33, 0x00 }, + { 35, 0x00 }, + { 36, 0x00 }, + { 37, 0x00 }, + { 46, 0x00 }, + { 48, 0x00 }, + { 49, 0x00 }, + { 51, 0x00 }, + { 56, 0x00 }, + { 58, 0x00 }, + { 59, 0x00 }, + { 60, 0x00 }, + { 61, 0x00 }, + { 62, 0x00 }, + { 63, 0x00 }, + { 64, 0x00 }, + { 65, 0x00 }, + { 66, 0x00 }, + { 67, 0x00 }, + { 68, 0x00 }, + { 69, 0x00 }, + { 70, 0x00 }, + { 71, 0x00 }, + { 72, 0x00 }, + { 73, 0x00 }, + { 74, 0x00 }, + { 75, 0x00 }, + { 76, 0x00 }, + { 77, 0x00 }, + { 78, 0x00 }, + { 79, 0x00 }, + { 80, 0x00 }, + { 81, 0x00 }, + { 82, 0x00 }, + { 83, 0x00 }, + { 85, 0x00 }, + { 85, 0x00 }, + { 86, 0x00 }, + { 87, 0x00 }, + { 88, 0x00 }, + { 89, 0x00 }, + { 90, 0x00 }, + { 91, 0x00 }, + { 92, 0x00 }, + { 93, 0x00 }, + { 94, 0x00 }, + { 95, 0x00 }, + { 96, 0x01 }, + { 97, 0x00 }, + { 98, 0x00 }, + { 99, 0x00 }, + { 100, 0x00 }, + { 101, 0x00 }, + { 102, 0x00 }, + { 103, 0x01 }, + { 105, 0x01 }, + { 106, 0x00 }, + { 107, 0x01 }, + { 107, 0x00 }, + { 108, 0x00 }, + { 109, 0x00 }, + { 110, 0x00 }, + { 111, 0x02 }, + { 112, 0x02 }, + { 113, 0x00 }, + { 121, 0x80 }, + { 122, 0xBB }, + { 123, 0x80 }, + { 124, 0xBB }, + { 128, 0x00 }, + { 130, 0x00 }, + { 131, 0x00 }, + { 132, 0x00 }, + { 133, 0x0A }, + { 134, 0x0A }, + { 135, 0x0A }, + { 136, 0x0F }, + { 137, 0x00 }, + { 138, 0x73 }, + { 139, 0x33 }, + { 140, 0x73 }, + { 141, 0x33 }, + { 142, 0x73 }, + { 143, 0x33 }, + { 144, 0x73 }, + { 145, 0x33 }, + { 146, 0x73 }, + { 147, 0x33 }, + { 148, 0x73 }, + { 149, 0x33 }, + { 150, 0x73 }, + { 151, 0x33 }, + { 152, 0x00 }, + { 153, 0x00 }, + { 154, 0x00 }, + { 155, 0x00 }, + { 176, 0x00 }, + { 177, 0x00 }, + { 178, 0x00 }, + { 179, 0x00 }, + { 180, 0x00 }, + { 181, 0x00 }, + { 182, 0x00 }, + { 183, 0x00 }, + { 184, 0x00 }, + { 185, 0x00 }, + { 186, 0x00 }, + { 189, 0x00 }, + { 188, 0x00 }, + { 194, 0x00 }, + { 195, 0x00 }, + { 196, 0x00 }, + { 197, 0x00 }, + { 200, 0x00 }, + { 201, 0x00 }, + { 202, 0x00 }, + { 203, 0x00 }, + { 204, 0x00 }, + { 205, 0x00 }, + { 208, 0x00 }, + { 209, 0x00 }, + { 210, 0x00 }, + { 211, 0x00 }, + { 213, 0x00 }, + { 214, 0x00 }, + { 215, 0x00 }, + { 216, 0x00 }, + { 217, 0x00 }, + { 218, 0x00 }, + { 219, 0x00 }, + { 221, 0x00 }, + { 222, 0x00 }, + { 224, 0x00 }, + { 225, 0x00 }, + { 226, 0x00 }, + { 227, 0x00 }, + { 228, 0x00 }, + { 229, 0x00 }, + { 230, 0x13 }, + { 231, 0x00 }, + { 232, 0x80 }, + { 233, 0x0C }, + { 234, 0xDD }, + { 235, 0x00 }, + { 236, 0x04 }, + { 237, 0x00 }, + { 238, 0x00 }, + { 239, 0x00 }, + { 240, 0x00 }, + { 241, 0x00 }, + { 242, 0x00 }, + { 243, 0x00 }, + { 244, 0x00 }, + { 245, 0x00 }, + { 248, 0x00 }, + { 249, 0x00 }, + { 254, 0x00 }, + { 255, 0x00 }, +}; + +/* codec private data */ +struct lm49453_priv { + struct regmap *regmap; + int fs_rate; +}; + +/* capture path controls */ + +static const char *lm49453_mic2mode_text[] = {"Single Ended", "Differential"}; + +static const SOC_ENUM_SINGLE_DECL(lm49453_mic2mode_enum, LM49453_P0_MICR_REG, 5, + lm49453_mic2mode_text); + +static const char *lm49453_dmic_cfg_text[] = {"DMICDAT1", "DMICDAT2"}; + +static const SOC_ENUM_SINGLE_DECL(lm49453_dmic12_cfg_enum, + LM49453_P0_DIGITAL_MIC1_CONFIG_REG, + 7, lm49453_dmic_cfg_text); + +static const SOC_ENUM_SINGLE_DECL(lm49453_dmic34_cfg_enum, + LM49453_P0_DIGITAL_MIC2_CONFIG_REG, + 7, lm49453_dmic_cfg_text); + +/* MUX Controls */ +static const char *lm49453_adcl_mux_text[] = { "MIC1", "Aux_L" }; + +static const char *lm49453_adcr_mux_text[] = { "MIC2", "Aux_R" }; + +static const struct soc_enum lm49453_adcl_enum = + SOC_ENUM_SINGLE(LM49453_P0_ANALOG_MIXER_ADC_REG, 0, + ARRAY_SIZE(lm49453_adcl_mux_text), + lm49453_adcl_mux_text); + +static const struct soc_enum lm49453_adcr_enum = + SOC_ENUM_SINGLE(LM49453_P0_ANALOG_MIXER_ADC_REG, 1, + ARRAY_SIZE(lm49453_adcr_mux_text), + lm49453_adcr_mux_text); + +static const struct snd_kcontrol_new lm49453_adcl_mux_control = + SOC_DAPM_ENUM("ADC Left Mux", lm49453_adcl_enum); + +static const struct snd_kcontrol_new lm49453_adcr_mux_control = + SOC_DAPM_ENUM("ADC Right Mux", lm49453_adcr_enum); + +static const struct snd_kcontrol_new lm49453_headset_left_mixer[] = { +SOC_DAPM_SINGLE("Port1_1 Switch", LM49453_P0_DACHPL1_REG, 0, 1, 0), +SOC_DAPM_SINGLE("Port1_2 Switch", LM49453_P0_DACHPL1_REG, 1, 1, 0), +SOC_DAPM_SINGLE("Port1_3 Switch", LM49453_P0_DACHPL1_REG, 2, 1, 0), +SOC_DAPM_SINGLE("Port1_4 Switch", LM49453_P0_DACHPL1_REG, 3, 1, 0), +SOC_DAPM_SINGLE("Port1_5 Switch", LM49453_P0_DACHPL1_REG, 4, 1, 0), +SOC_DAPM_SINGLE("Port1_6 Switch", LM49453_P0_DACHPL1_REG, 5, 1, 0), +SOC_DAPM_SINGLE("Port1_7 Switch", LM49453_P0_DACHPL1_REG, 6, 1, 0), +SOC_DAPM_SINGLE("Port1_8 Switch", LM49453_P0_DACHPL1_REG, 7, 1, 0), +SOC_DAPM_SINGLE("DMIC1L Switch", LM49453_P0_DACHPL2_REG, 0, 1, 0), +SOC_DAPM_SINGLE("DMIC1R Switch", LM49453_P0_DACHPL2_REG, 1, 1, 0), +SOC_DAPM_SINGLE("DMIC2L Switch", LM49453_P0_DACHPL2_REG, 2, 1, 0), +SOC_DAPM_SINGLE("DMIC2R Switch", LM49453_P0_DACHPL2_REG, 3, 1, 0), +SOC_DAPM_SINGLE("ADCL Switch", LM49453_P0_DACHPL2_REG, 4, 1, 0), +SOC_DAPM_SINGLE("ADCR Switch", LM49453_P0_DACHPL2_REG, 5, 1, 0), +SOC_DAPM_SINGLE("Port2_1 Switch", LM49453_P0_DACHPL2_REG, 6, 1, 0), +SOC_DAPM_SINGLE("Port2_2 Switch", LM49453_P0_DACHPL2_REG, 7, 1, 0), +SOC_DAPM_SINGLE("Sidetone Switch", LM49453_P0_STN_SEL_REG, 0, 0, 0), +}; + +static const struct snd_kcontrol_new lm49453_headset_right_mixer[] = { +SOC_DAPM_SINGLE("Port1_1 Switch", LM49453_P0_DACHPR1_REG, 0, 1, 0), +SOC_DAPM_SINGLE("Port1_2 Switch", LM49453_P0_DACHPR1_REG, 1, 1, 0), +SOC_DAPM_SINGLE("Port1_3 Switch", LM49453_P0_DACHPR1_REG, 2, 1, 0), +SOC_DAPM_SINGLE("Port1_4 Switch", LM49453_P0_DACHPR1_REG, 3, 1, 0), +SOC_DAPM_SINGLE("Port1_5 Switch", LM49453_P0_DACHPR1_REG, 4, 1, 0), +SOC_DAPM_SINGLE("Port1_6 Switch", LM49453_P0_DACHPR1_REG, 5, 1, 0), +SOC_DAPM_SINGLE("Port1_7 Switch", LM49453_P0_DACHPR1_REG, 6, 1, 0), +SOC_DAPM_SINGLE("Port1_8 Switch", LM49453_P0_DACHPR1_REG, 7, 1, 0), +SOC_DAPM_SINGLE("DMIC1L Switch", LM49453_P0_DACHPR2_REG, 0, 1, 0), +SOC_DAPM_SINGLE("DMIC1R Switch", LM49453_P0_DACHPR2_REG, 1, 1, 0), +SOC_DAPM_SINGLE("DMIC2L Switch", LM49453_P0_DACHPR2_REG, 2, 1, 0), +SOC_DAPM_SINGLE("DMIC2R Switch", LM49453_P0_DACHPR2_REG, 3, 1, 0), +SOC_DAPM_SINGLE("ADCL Switch", LM49453_P0_DACHPR2_REG, 4, 1, 0), +SOC_DAPM_SINGLE("ADCR Switch", LM49453_P0_DACHPR2_REG, 5, 1, 0), +SOC_DAPM_SINGLE("Port2_1 Switch", LM49453_P0_DACHPR2_REG, 6, 1, 0), +SOC_DAPM_SINGLE("Port2_2 Switch", LM49453_P0_DACHPR2_REG, 7, 1, 0), +SOC_DAPM_SINGLE("Sidetone Switch", LM49453_P0_STN_SEL_REG, 1, 0, 0), +}; + +static const struct snd_kcontrol_new lm49453_speaker_left_mixer[] = { +SOC_DAPM_SINGLE("Port1_1 Switch", LM49453_P0_DACLSL1_REG, 0, 1, 0), +SOC_DAPM_SINGLE("Port1_2 Switch", LM49453_P0_DACLSL1_REG, 1, 1, 0), +SOC_DAPM_SINGLE("Port1_3 Switch", LM49453_P0_DACLSL1_REG, 2, 1, 0), +SOC_DAPM_SINGLE("Port1_4 Switch", LM49453_P0_DACLSL1_REG, 3, 1, 0), +SOC_DAPM_SINGLE("Port1_5 Switch", LM49453_P0_DACLSL1_REG, 4, 1, 0), +SOC_DAPM_SINGLE("Port1_6 Switch", LM49453_P0_DACLSL1_REG, 5, 1, 0), +SOC_DAPM_SINGLE("Port1_7 Switch", LM49453_P0_DACLSL1_REG, 6, 1, 0), +SOC_DAPM_SINGLE("Port1_8 Switch", LM49453_P0_DACLSL1_REG, 7, 1, 0), +SOC_DAPM_SINGLE("DMIC1L Switch", LM49453_P0_DACLSL2_REG, 0, 1, 0), +SOC_DAPM_SINGLE("DMIC1R Switch", LM49453_P0_DACLSL2_REG, 1, 1, 0), +SOC_DAPM_SINGLE("DMIC2L Switch", LM49453_P0_DACLSL2_REG, 2, 1, 0), +SOC_DAPM_SINGLE("DMIC2R Switch", LM49453_P0_DACLSL2_REG, 3, 1, 0), +SOC_DAPM_SINGLE("ADCL Switch", LM49453_P0_DACLSL2_REG, 4, 1, 0), +SOC_DAPM_SINGLE("ADCR Switch", LM49453_P0_DACLSL2_REG, 5, 1, 0), +SOC_DAPM_SINGLE("Port2_1 Switch", LM49453_P0_DACLSL2_REG, 6, 1, 0), +SOC_DAPM_SINGLE("Port2_2 Switch", LM49453_P0_DACLSL2_REG, 7, 1, 0), +SOC_DAPM_SINGLE("Sidetone Switch", LM49453_P0_STN_SEL_REG, 2, 0, 0), +}; + +static const struct snd_kcontrol_new lm49453_speaker_right_mixer[] = { +SOC_DAPM_SINGLE("Port1_1 Switch", LM49453_P0_DACLSR1_REG, 0, 1, 0), +SOC_DAPM_SINGLE("Port1_2 Switch", LM49453_P0_DACLSR1_REG, 1, 1, 0), +SOC_DAPM_SINGLE("Port1_3 Switch", LM49453_P0_DACLSR1_REG, 2, 1, 0), +SOC_DAPM_SINGLE("Port1_4 Switch", LM49453_P0_DACLSR1_REG, 3, 1, 0), +SOC_DAPM_SINGLE("Port1_5 Switch", LM49453_P0_DACLSR1_REG, 4, 1, 0), +SOC_DAPM_SINGLE("Port1_6 Switch", LM49453_P0_DACLSR1_REG, 5, 1, 0), +SOC_DAPM_SINGLE("Port1_7 Switch", LM49453_P0_DACLSR1_REG, 6, 1, 0), +SOC_DAPM_SINGLE("Port1_8 Switch", LM49453_P0_DACLSR1_REG, 7, 1, 0), +SOC_DAPM_SINGLE("DMIC1L Switch", LM49453_P0_DACLSR2_REG, 0, 1, 0), +SOC_DAPM_SINGLE("DMIC1R Switch", LM49453_P0_DACLSR2_REG, 1, 1, 0), +SOC_DAPM_SINGLE("DMIC2L Switch", LM49453_P0_DACLSR2_REG, 2, 1, 0), +SOC_DAPM_SINGLE("DMIC2R Switch", LM49453_P0_DACLSR2_REG, 3, 1, 0), +SOC_DAPM_SINGLE("ADCL Switch", LM49453_P0_DACLSR2_REG, 4, 1, 0), +SOC_DAPM_SINGLE("ADCR Switch", LM49453_P0_DACLSR2_REG, 5, 1, 0), +SOC_DAPM_SINGLE("Port2_1 Switch", LM49453_P0_DACLSR2_REG, 6, 1, 0), +SOC_DAPM_SINGLE("Port2_2 Switch", LM49453_P0_DACLSR2_REG, 7, 1, 0), +SOC_DAPM_SINGLE("Sidetone Switch", LM49453_P0_STN_SEL_REG, 3, 0, 0), +}; + +static const struct snd_kcontrol_new lm49453_haptic_left_mixer[] = { +SOC_DAPM_SINGLE("Port1_1 Switch", LM49453_P0_DACHAL1_REG, 0, 1, 0), +SOC_DAPM_SINGLE("Port1_2 Switch", LM49453_P0_DACHAL1_REG, 1, 1, 0), +SOC_DAPM_SINGLE("Port1_3 Switch", LM49453_P0_DACHAL1_REG, 2, 1, 0), +SOC_DAPM_SINGLE("Port1_4 Switch", LM49453_P0_DACHAL1_REG, 3, 1, 0), +SOC_DAPM_SINGLE("Port1_5 Switch", LM49453_P0_DACHAL1_REG, 4, 1, 0), +SOC_DAPM_SINGLE("Port1_6 Switch", LM49453_P0_DACHAL1_REG, 5, 1, 0), +SOC_DAPM_SINGLE("Port1_7 Switch", LM49453_P0_DACHAL1_REG, 6, 1, 0), +SOC_DAPM_SINGLE("Port1_8 Switch", LM49453_P0_DACHAL1_REG, 7, 1, 0), +SOC_DAPM_SINGLE("DMIC1L Switch", LM49453_P0_DACHAL2_REG, 0, 1, 0), +SOC_DAPM_SINGLE("DMIC1R Switch", LM49453_P0_DACHAL2_REG, 1, 1, 0), +SOC_DAPM_SINGLE("DMIC2L Switch", LM49453_P0_DACHAL2_REG, 2, 1, 0), +SOC_DAPM_SINGLE("DMIC2R Switch", LM49453_P0_DACHAL2_REG, 3, 1, 0), +SOC_DAPM_SINGLE("ADCL Switch", LM49453_P0_DACHAL2_REG, 4, 1, 0), +SOC_DAPM_SINGLE("ADCR Switch", LM49453_P0_DACHAL2_REG, 5, 1, 0), +SOC_DAPM_SINGLE("Port2_1 Switch", LM49453_P0_DACHAL2_REG, 6, 1, 0), +SOC_DAPM_SINGLE("Port2_2 Switch", LM49453_P0_DACHAL2_REG, 7, 1, 0), +SOC_DAPM_SINGLE("Sidetone Switch", LM49453_P0_STN_SEL_REG, 4, 0, 0), +}; + +static const struct snd_kcontrol_new lm49453_haptic_right_mixer[] = { +SOC_DAPM_SINGLE("Port1_1 Switch", LM49453_P0_DACHAR1_REG, 0, 1, 0), +SOC_DAPM_SINGLE("Port1_2 Switch", LM49453_P0_DACHAR1_REG, 1, 1, 0), +SOC_DAPM_SINGLE("Port1_3 Switch", LM49453_P0_DACHAR1_REG, 2, 1, 0), +SOC_DAPM_SINGLE("Port1_4 Switch", LM49453_P0_DACHAR1_REG, 3, 1, 0), +SOC_DAPM_SINGLE("Port1_5 Switch", LM49453_P0_DACHAR1_REG, 4, 1, 0), +SOC_DAPM_SINGLE("Port1_6 Switch", LM49453_P0_DACHAR1_REG, 5, 1, 0), +SOC_DAPM_SINGLE("Port1_7 Switch", LM49453_P0_DACHAR1_REG, 6, 1, 0), +SOC_DAPM_SINGLE("Port1_8 Switch", LM49453_P0_DACHAR1_REG, 7, 1, 0), +SOC_DAPM_SINGLE("DMIC1L Switch", LM49453_P0_DACHAR2_REG, 0, 1, 0), +SOC_DAPM_SINGLE("DMIC1R Switch", LM49453_P0_DACHAR2_REG, 1, 1, 0), +SOC_DAPM_SINGLE("DMIC2L Switch", LM49453_P0_DACHAR2_REG, 2, 1, 0), +SOC_DAPM_SINGLE("DMIC2R Switch", LM49453_P0_DACHAR2_REG, 3, 1, 0), +SOC_DAPM_SINGLE("ADCL Switch", LM49453_P0_DACHAR2_REG, 4, 1, 0), +SOC_DAPM_SINGLE("ADCR Switch", LM49453_P0_DACHAR2_REG, 5, 1, 0), +SOC_DAPM_SINGLE("Port2_1 Switch", LM49453_P0_DACHAR2_REG, 6, 1, 0), +SOC_DAPM_SINGLE("Port2_2 Switch", LM49453_P0_DACHAR2_REG, 7, 1, 0), +SOC_DAPM_SINGLE("Sidetone Switch", LM49453_P0_STN_SEL_REG, 5, 0, 0), +}; + +static const struct snd_kcontrol_new lm49453_lineout_left_mixer[] = { +SOC_DAPM_SINGLE("Port1_1 Switch", LM49453_P0_DACLOL1_REG, 0, 1, 0), +SOC_DAPM_SINGLE("Port1_2 Switch", LM49453_P0_DACLOL1_REG, 1, 1, 0), +SOC_DAPM_SINGLE("Port1_3 Switch", LM49453_P0_DACLOL1_REG, 2, 1, 0), +SOC_DAPM_SINGLE("Port1_4 Switch", LM49453_P0_DACLOL1_REG, 3, 1, 0), +SOC_DAPM_SINGLE("Port1_5 Switch", LM49453_P0_DACLOL1_REG, 4, 1, 0), +SOC_DAPM_SINGLE("Port1_6 Switch", LM49453_P0_DACLOL1_REG, 5, 1, 0), +SOC_DAPM_SINGLE("Port1_7 Switch", LM49453_P0_DACLOL1_REG, 6, 1, 0), +SOC_DAPM_SINGLE("Port1_8 Switch", LM49453_P0_DACLOL1_REG, 7, 1, 0), +SOC_DAPM_SINGLE("DMIC1L Switch", LM49453_P0_DACLOL2_REG, 0, 1, 0), +SOC_DAPM_SINGLE("DMIC1R Switch", LM49453_P0_DACLOL2_REG, 1, 1, 0), +SOC_DAPM_SINGLE("DMIC2L Switch", LM49453_P0_DACLOL2_REG, 2, 1, 0), +SOC_DAPM_SINGLE("DMIC2R Switch", LM49453_P0_DACLOL2_REG, 3, 1, 0), +SOC_DAPM_SINGLE("ADCL Switch", LM49453_P0_DACLOL2_REG, 4, 1, 0), +SOC_DAPM_SINGLE("ADCR Switch", LM49453_P0_DACLOL2_REG, 5, 1, 0), +SOC_DAPM_SINGLE("Port2_1 Switch", LM49453_P0_DACLOL2_REG, 6, 1, 0), +SOC_DAPM_SINGLE("Port2_2 Switch", LM49453_P0_DACLOL2_REG, 7, 1, 0), +SOC_DAPM_SINGLE("Sidetone Switch", LM49453_P0_STN_SEL_REG, 6, 0, 0), +}; + +static const struct snd_kcontrol_new lm49453_lineout_right_mixer[] = { +SOC_DAPM_SINGLE("Port1_1 Switch", LM49453_P0_DACLOR1_REG, 0, 1, 0), +SOC_DAPM_SINGLE("Port1_2 Switch", LM49453_P0_DACLOR1_REG, 1, 1, 0), +SOC_DAPM_SINGLE("Port1_3 Switch", LM49453_P0_DACLOR1_REG, 2, 1, 0), +SOC_DAPM_SINGLE("Port1_4 Switch", LM49453_P0_DACLOR1_REG, 3, 1, 0), +SOC_DAPM_SINGLE("Port1_5 Switch", LM49453_P0_DACLOR1_REG, 4, 1, 0), +SOC_DAPM_SINGLE("Port1_6 Switch", LM49453_P0_DACLOR1_REG, 5, 1, 0), +SOC_DAPM_SINGLE("Port1_7 Switch", LM49453_P0_DACLOR1_REG, 6, 1, 0), +SOC_DAPM_SINGLE("Port1_8 Switch", LM49453_P0_DACLOR1_REG, 7, 1, 0), +SOC_DAPM_SINGLE("DMIC1L Switch", LM49453_P0_DACLOR2_REG, 0, 1, 0), +SOC_DAPM_SINGLE("DMIC1R Switch", LM49453_P0_DACLOR2_REG, 1, 1, 0), +SOC_DAPM_SINGLE("DMIC2L Switch", LM49453_P0_DACLOR2_REG, 2, 1, 0), +SOC_DAPM_SINGLE("DMIC2R Switch", LM49453_P0_DACLOR2_REG, 3, 1, 0), +SOC_DAPM_SINGLE("ADCL Switch", LM49453_P0_DACLOR2_REG, 4, 1, 0), +SOC_DAPM_SINGLE("ADCR Switch", LM49453_P0_DACLOR2_REG, 5, 1, 0), +SOC_DAPM_SINGLE("Port2_1 Switch", LM49453_P0_DACLOR2_REG, 6, 1, 0), +SOC_DAPM_SINGLE("Port2_2 Switch", LM49453_P0_DACLOR2_REG, 7, 1, 0), +SOC_DAPM_SINGLE("Sidetone Switch", LM49453_P0_STN_SEL_REG, 7, 0, 0), +}; + +static const struct snd_kcontrol_new lm49453_port1_tx1_mixer[] = { +SOC_DAPM_SINGLE("DMIC1L Switch", LM49453_P0_PORT1_TX1_REG, 0, 1, 0), +SOC_DAPM_SINGLE("DMIC1R Switch", LM49453_P0_PORT1_TX1_REG, 1, 1, 0), +SOC_DAPM_SINGLE("DMIC2L Switch", LM49453_P0_PORT1_TX1_REG, 2, 1, 0), +SOC_DAPM_SINGLE("DMIC2R Switch", LM49453_P0_PORT1_TX1_REG, 3, 1, 0), +SOC_DAPM_SINGLE("ADCL Switch", LM49453_P0_PORT1_TX1_REG, 4, 1, 0), +SOC_DAPM_SINGLE("ADCR Switch", LM49453_P0_PORT1_TX1_REG, 5, 1, 0), +SOC_DAPM_SINGLE("Port1_1 Switch", LM49453_P0_PORT1_TX1_REG, 6, 1, 0), +SOC_DAPM_SINGLE("Port2_1 Switch", LM49453_P0_PORT1_TX1_REG, 7, 1, 0), +}; + +static const struct snd_kcontrol_new lm49453_port1_tx2_mixer[] = { +SOC_DAPM_SINGLE("DMIC1L Switch", LM49453_P0_PORT1_TX2_REG, 0, 1, 0), +SOC_DAPM_SINGLE("DMIC1R Switch", LM49453_P0_PORT1_TX2_REG, 1, 1, 0), +SOC_DAPM_SINGLE("DMIC2L Switch", LM49453_P0_PORT1_TX2_REG, 2, 1, 0), +SOC_DAPM_SINGLE("DMIC2R Switch", LM49453_P0_PORT1_TX2_REG, 3, 1, 0), +SOC_DAPM_SINGLE("ADCL Switch", LM49453_P0_PORT1_TX2_REG, 4, 1, 0), +SOC_DAPM_SINGLE("ADCR Switch", LM49453_P0_PORT1_TX2_REG, 5, 1, 0), +SOC_DAPM_SINGLE("Port1_2 Switch", LM49453_P0_PORT1_TX2_REG, 6, 1, 0), +SOC_DAPM_SINGLE("Port2_2 Switch", LM49453_P0_PORT1_TX2_REG, 7, 1, 0), +}; + +static const struct snd_kcontrol_new lm49453_port1_tx3_mixer[] = { +SOC_DAPM_SINGLE("DMIC1L Switch", LM49453_P0_PORT1_TX3_REG, 0, 1, 0), +SOC_DAPM_SINGLE("DMIC1R Switch", LM49453_P0_PORT1_TX3_REG, 1, 1, 0), +SOC_DAPM_SINGLE("DMIC2L Switch", LM49453_P0_PORT1_TX3_REG, 2, 1, 0), +SOC_DAPM_SINGLE("DMIC2R Switch", LM49453_P0_PORT1_TX3_REG, 3, 1, 0), +SOC_DAPM_SINGLE("ADCL Switch", LM49453_P0_PORT1_TX3_REG, 4, 1, 0), +SOC_DAPM_SINGLE("ADCR Switch", LM49453_P0_PORT1_TX3_REG, 5, 1, 0), +SOC_DAPM_SINGLE("Port1_3 Switch", LM49453_P0_PORT1_TX3_REG, 6, 1, 0), +}; + +static const struct snd_kcontrol_new lm49453_port1_tx4_mixer[] = { +SOC_DAPM_SINGLE("DMIC1L Switch", LM49453_P0_PORT1_TX4_REG, 0, 1, 0), +SOC_DAPM_SINGLE("DMIC1R Switch", LM49453_P0_PORT1_TX4_REG, 1, 1, 0), +SOC_DAPM_SINGLE("DMIC2L Switch", LM49453_P0_PORT1_TX4_REG, 2, 1, 0), +SOC_DAPM_SINGLE("DMIC2R Switch", LM49453_P0_PORT1_TX4_REG, 3, 1, 0), +SOC_DAPM_SINGLE("ADCL Switch", LM49453_P0_PORT1_TX4_REG, 4, 1, 0), +SOC_DAPM_SINGLE("ADCR Switch", LM49453_P0_PORT1_TX4_REG, 5, 1, 0), +SOC_DAPM_SINGLE("Port1_4 Switch", LM49453_P0_PORT1_TX4_REG, 6, 1, 0), +}; + +static const struct snd_kcontrol_new lm49453_port1_tx5_mixer[] = { +SOC_DAPM_SINGLE("DMIC1L Switch", LM49453_P0_PORT1_TX5_REG, 0, 1, 0), +SOC_DAPM_SINGLE("DMIC1R Switch", LM49453_P0_PORT1_TX5_REG, 1, 1, 0), +SOC_DAPM_SINGLE("DMIC2L Switch", LM49453_P0_PORT1_TX5_REG, 2, 1, 0), +SOC_DAPM_SINGLE("DMIC2R Switch", LM49453_P0_PORT1_TX5_REG, 3, 1, 0), +SOC_DAPM_SINGLE("ADCL Switch", LM49453_P0_PORT1_TX5_REG, 4, 1, 0), +SOC_DAPM_SINGLE("ADCR Switch", LM49453_P0_PORT1_TX5_REG, 5, 1, 0), +SOC_DAPM_SINGLE("Port1_5 Switch", LM49453_P0_PORT1_TX5_REG, 6, 1, 0), +}; + +static const struct snd_kcontrol_new lm49453_port1_tx6_mixer[] = { +SOC_DAPM_SINGLE("DMIC1L Switch", LM49453_P0_PORT1_TX6_REG, 0, 1, 0), +SOC_DAPM_SINGLE("DMIC1R Switch", LM49453_P0_PORT1_TX6_REG, 1, 1, 0), +SOC_DAPM_SINGLE("DMIC2L Switch", LM49453_P0_PORT1_TX6_REG, 2, 1, 0), +SOC_DAPM_SINGLE("DMIC2R Switch", LM49453_P0_PORT1_TX6_REG, 3, 1, 0), +SOC_DAPM_SINGLE("ADCL Switch", LM49453_P0_PORT1_TX6_REG, 4, 1, 0), +SOC_DAPM_SINGLE("ADCR Switch", LM49453_P0_PORT1_TX6_REG, 5, 1, 0), +SOC_DAPM_SINGLE("Port1_6 Switch", LM49453_P0_PORT1_TX6_REG, 6, 1, 0), +}; + +static const struct snd_kcontrol_new lm49453_port1_tx7_mixer[] = { +SOC_DAPM_SINGLE("DMIC1L Switch", LM49453_P0_PORT1_TX7_REG, 0, 1, 0), +SOC_DAPM_SINGLE("DMIC1R Switch", LM49453_P0_PORT1_TX7_REG, 1, 1, 0), +SOC_DAPM_SINGLE("DMIC2L Switch", LM49453_P0_PORT1_TX7_REG, 2, 1, 0), +SOC_DAPM_SINGLE("DMIC2R Switch", LM49453_P0_PORT1_TX7_REG, 3, 1, 0), +SOC_DAPM_SINGLE("ADCL Switch", LM49453_P0_PORT1_TX7_REG, 4, 1, 0), +SOC_DAPM_SINGLE("ADCR Switch", LM49453_P0_PORT1_TX7_REG, 5, 1, 0), +SOC_DAPM_SINGLE("Port1_7 Switch", LM49453_P0_PORT1_TX7_REG, 6, 1, 0), +}; + +static const struct snd_kcontrol_new lm49453_port1_tx8_mixer[] = { +SOC_DAPM_SINGLE("DMIC1L Switch", LM49453_P0_PORT1_TX8_REG, 0, 1, 0), +SOC_DAPM_SINGLE("DMIC1R Switch", LM49453_P0_PORT1_TX8_REG, 1, 1, 0), +SOC_DAPM_SINGLE("DMIC2L Switch", LM49453_P0_PORT1_TX8_REG, 2, 1, 0), +SOC_DAPM_SINGLE("DMIC2R Switch", LM49453_P0_PORT1_TX8_REG, 3, 1, 0), +SOC_DAPM_SINGLE("ADCL Switch", LM49453_P0_PORT1_TX8_REG, 4, 1, 0), +SOC_DAPM_SINGLE("ADCR Switch", LM49453_P0_PORT1_TX8_REG, 5, 1, 0), +SOC_DAPM_SINGLE("Port1_8 Switch", LM49453_P0_PORT1_TX8_REG, 6, 1, 0), +}; + +static const struct snd_kcontrol_new lm49453_port2_tx1_mixer[] = { +SOC_DAPM_SINGLE("DMIC1L Switch", LM49453_P0_PORT2_TX1_REG, 0, 1, 0), +SOC_DAPM_SINGLE("DMIC1R Switch", LM49453_P0_PORT2_TX1_REG, 1, 1, 0), +SOC_DAPM_SINGLE("DMIC2L Switch", LM49453_P0_PORT2_TX1_REG, 2, 1, 0), +SOC_DAPM_SINGLE("DMIC2R Switch", LM49453_P0_PORT2_TX1_REG, 3, 1, 0), +SOC_DAPM_SINGLE("ADCL Switch", LM49453_P0_PORT2_TX1_REG, 4, 1, 0), +SOC_DAPM_SINGLE("ADCR Switch", LM49453_P0_PORT2_TX1_REG, 5, 1, 0), +SOC_DAPM_SINGLE("Port1_1 Switch", LM49453_P0_PORT2_TX1_REG, 6, 1, 0), +SOC_DAPM_SINGLE("Port2_1 Switch", LM49453_P0_PORT2_TX1_REG, 7, 1, 0), +}; + +static const struct snd_kcontrol_new lm49453_port2_tx2_mixer[] = { +SOC_DAPM_SINGLE("DMIC1L Switch", LM49453_P0_PORT2_TX2_REG, 0, 1, 0), +SOC_DAPM_SINGLE("DMIC1R Switch", LM49453_P0_PORT2_TX2_REG, 1, 1, 0), +SOC_DAPM_SINGLE("DMIC2L Switch", LM49453_P0_PORT2_TX2_REG, 2, 1, 0), +SOC_DAPM_SINGLE("DMIC2R Switch", LM49453_P0_PORT2_TX2_REG, 3, 1, 0), +SOC_DAPM_SINGLE("ADCL Switch", LM49453_P0_PORT2_TX2_REG, 4, 1, 0), +SOC_DAPM_SINGLE("ADCR Switch", LM49453_P0_PORT2_TX2_REG, 5, 1, 0), +SOC_DAPM_SINGLE("Port1_2 Switch", LM49453_P0_PORT2_TX2_REG, 6, 1, 0), +SOC_DAPM_SINGLE("Port2_2 Switch", LM49453_P0_PORT2_TX2_REG, 7, 1, 0), +}; + +/* TLV Declarations */ +static const DECLARE_TLV_DB_SCALE(digital_tlv, -7650, 150, 1); +static const DECLARE_TLV_DB_SCALE(port_tlv, 0, 600, 0); + +static const struct snd_kcontrol_new lm49453_sidetone_mixer_controls[] = { +/* Sidetone supports mono only */ +SOC_DAPM_SINGLE_TLV("Sidetone ADCL Volume", LM49453_P0_STN_VOL_ADCL_REG, + 0, 0x3F, 0, digital_tlv), +SOC_DAPM_SINGLE_TLV("Sidetone ADCR Volume", LM49453_P0_STN_VOL_ADCR_REG, + 0, 0x3F, 0, digital_tlv), +SOC_DAPM_SINGLE_TLV("Sidetone DMIC1L Volume", LM49453_P0_STN_VOL_DMIC1L_REG, + 0, 0x3F, 0, digital_tlv), +SOC_DAPM_SINGLE_TLV("Sidetone DMIC1R Volume", LM49453_P0_STN_VOL_DMIC1R_REG, + 0, 0x3F, 0, digital_tlv), +SOC_DAPM_SINGLE_TLV("Sidetone DMIC2L Volume", LM49453_P0_STN_VOL_DMIC2L_REG, + 0, 0x3F, 0, digital_tlv), +SOC_DAPM_SINGLE_TLV("Sidetone DMIC2R Volume", LM49453_P0_STN_VOL_DMIC2R_REG, + 0, 0x3F, 0, digital_tlv), +}; + +static const struct snd_kcontrol_new lm49453_snd_controls[] = { + /* mic1 and mic2 supports mono only */ + SOC_SINGLE_TLV("Mic1 Volume", LM49453_P0_ADC_LEVELL_REG, 0, 6, + 0, digital_tlv), + SOC_SINGLE_TLV("Mic2 Volume", LM49453_P0_ADC_LEVELR_REG, 0, 6, + 0, digital_tlv), + + SOC_DOUBLE_R_TLV("DMIC1 Volume", LM49453_P0_DMIC1_LEVELL_REG, + LM49453_P0_DMIC1_LEVELR_REG, 0, 6, 0, digital_tlv), + SOC_DOUBLE_R_TLV("DMIC2 Volume", LM49453_P0_DMIC2_LEVELL_REG, + LM49453_P0_DMIC2_LEVELR_REG, 0, 6, 0, digital_tlv), + + SOC_DAPM_ENUM("Mic2Mode", lm49453_mic2mode_enum), + SOC_DAPM_ENUM("DMIC12 SRC", lm49453_dmic12_cfg_enum), + SOC_DAPM_ENUM("DMIC34 SRC", lm49453_dmic34_cfg_enum), + + /* Capture path filter enable */ + SOC_SINGLE("DMIC1 HPFilter Switch", LM49453_P0_ADC_FX_ENABLES_REG, + 0, 1, 0), + SOC_SINGLE("DMIC2 HPFilter Switch", LM49453_P0_ADC_FX_ENABLES_REG, + 1, 1, 0), + SOC_SINGLE("ADC HPFilter Switch", LM49453_P0_ADC_FX_ENABLES_REG, + 2, 1, 0), + + SOC_DOUBLE_R_TLV("DAC HP Volume", LM49453_P0_DAC_HP_LEVELL_REG, + LM49453_P0_DAC_HP_LEVELR_REG, 0, 6, 0, digital_tlv), + SOC_DOUBLE_R_TLV("DAC LO Volume", LM49453_P0_DAC_LO_LEVELL_REG, + LM49453_P0_DAC_LO_LEVELR_REG, 0, 6, 0, digital_tlv), + SOC_DOUBLE_R_TLV("DAC LS Volume", LM49453_P0_DAC_LS_LEVELL_REG, + LM49453_P0_DAC_LS_LEVELR_REG, 0, 6, 0, digital_tlv), + SOC_DOUBLE_R_TLV("DAC HA Volume", LM49453_P0_DAC_HA_LEVELL_REG, + LM49453_P0_DAC_HA_LEVELR_REG, 0, 6, 0, digital_tlv), + + SOC_SINGLE_TLV("EP Volume", LM49453_P0_DAC_LS_LEVELL_REG, + 0, 6, 0, digital_tlv), + + SOC_SINGLE_TLV("PORT1_1_RX_LVL Volume", LM49453_P0_PORT1_RX_LVL1_REG, + 0, 3, 0, port_tlv), + SOC_SINGLE_TLV("PORT1_2_RX_LVL Volume", LM49453_P0_PORT1_RX_LVL1_REG, + 2, 3, 0, port_tlv), + SOC_SINGLE_TLV("PORT1_3_RX_LVL Volume", LM49453_P0_PORT1_RX_LVL1_REG, + 4, 3, 0, port_tlv), + SOC_SINGLE_TLV("PORT1_4_RX_LVL Volume", LM49453_P0_PORT1_RX_LVL1_REG, + 6, 3, 0, port_tlv), + SOC_SINGLE_TLV("PORT1_5_RX_LVL Volume", LM49453_P0_PORT1_RX_LVL2_REG, + 0, 3, 0, port_tlv), + SOC_SINGLE_TLV("PORT1_6_RX_LVL Volume", LM49453_P0_PORT1_RX_LVL2_REG, + 2, 3, 0, port_tlv), + SOC_SINGLE_TLV("PORT1_7_RX_LVL Volume", LM49453_P0_PORT1_RX_LVL2_REG, + 4, 3, 0, port_tlv), + SOC_SINGLE_TLV("PORT1_8_RX_LVL Volume", LM49453_P0_PORT1_RX_LVL2_REG, + 6, 3, 0, port_tlv), + + SOC_SINGLE_TLV("PORT2_1_RX_LVL Volume", LM49453_P0_PORT2_RX_LVL_REG, + 0, 3, 0, port_tlv), + SOC_SINGLE_TLV("PORT2_2_RX_LVL Volume", LM49453_P0_PORT2_RX_LVL_REG, + 2, 3, 0, port_tlv), + + SOC_SINGLE("Port1 Playback Switch", LM49453_P0_AUDIO_PORT1_BASIC_REG, + 1, 1, 0), + SOC_SINGLE("Port2 Playback Switch", LM49453_P0_AUDIO_PORT2_BASIC_REG, + 1, 1, 0), + SOC_SINGLE("Port1 Capture Switch", LM49453_P0_AUDIO_PORT1_BASIC_REG, + 2, 1, 0), + SOC_SINGLE("Port2 Capture Switch", LM49453_P0_AUDIO_PORT2_BASIC_REG, + 2, 1, 0) + +}; + +/* DAPM widgets */ +static const struct snd_soc_dapm_widget lm49453_dapm_widgets[] = { + + /* All end points HP,EP, LS, Lineout and Haptic */ + SND_SOC_DAPM_OUTPUT("HPOUTL"), + SND_SOC_DAPM_OUTPUT("HPOUTR"), + SND_SOC_DAPM_OUTPUT("EPOUT"), + SND_SOC_DAPM_OUTPUT("LSOUTL"), + SND_SOC_DAPM_OUTPUT("LSOUTR"), + SND_SOC_DAPM_OUTPUT("LOOUTR"), + SND_SOC_DAPM_OUTPUT("LOOUTL"), + SND_SOC_DAPM_OUTPUT("HAOUTL"), + SND_SOC_DAPM_OUTPUT("HAOUTR"), + + SND_SOC_DAPM_INPUT("AMIC1"), + SND_SOC_DAPM_INPUT("AMIC2"), + SND_SOC_DAPM_INPUT("DMIC1DAT"), + SND_SOC_DAPM_INPUT("DMIC2DAT"), + SND_SOC_DAPM_INPUT("AUXL"), + SND_SOC_DAPM_INPUT("AUXR"), + + SND_SOC_DAPM_PGA("PORT1_1_RX", SND_SOC_NOPM, 0, 0, NULL, 0), + SND_SOC_DAPM_PGA("PORT1_2_RX", SND_SOC_NOPM, 0, 0, NULL, 0), + SND_SOC_DAPM_PGA("PORT1_3_RX", SND_SOC_NOPM, 0, 0, NULL, 0), + SND_SOC_DAPM_PGA("PORT1_4_RX", SND_SOC_NOPM, 0, 0, NULL, 0), + SND_SOC_DAPM_PGA("PORT1_5_RX", SND_SOC_NOPM, 0, 0, NULL, 0), + SND_SOC_DAPM_PGA("PORT1_6_RX", SND_SOC_NOPM, 0, 0, NULL, 0), + SND_SOC_DAPM_PGA("PORT1_7_RX", SND_SOC_NOPM, 0, 0, NULL, 0), + SND_SOC_DAPM_PGA("PORT1_8_RX", SND_SOC_NOPM, 0, 0, NULL, 0), + SND_SOC_DAPM_PGA("PORT2_1_RX", SND_SOC_NOPM, 0, 0, NULL, 0), + SND_SOC_DAPM_PGA("PORT2_2_RX", SND_SOC_NOPM, 0, 0, NULL, 0), + + SND_SOC_DAPM_SUPPLY("AMIC1Bias", LM49453_P0_MICL_REG, 6, 0, NULL, 0), + SND_SOC_DAPM_SUPPLY("AMIC2Bias", LM49453_P0_MICR_REG, 6, 0, NULL, 0), + + /* playback path driver enables */ + SND_SOC_DAPM_OUT_DRV("Headset Switch", + LM49453_P0_PMC_SETUP_REG, 0, 0, NULL, 0), + SND_SOC_DAPM_OUT_DRV("Earpiece Switch", + LM49453_P0_EP_REG, 0, 0, NULL, 0), + SND_SOC_DAPM_OUT_DRV("Speaker Left Switch", + LM49453_P0_DIS_PKVL_FB_REG, 0, 1, NULL, 0), + SND_SOC_DAPM_OUT_DRV("Speaker Right Switch", + LM49453_P0_DIS_PKVL_FB_REG, 1, 1, NULL, 0), + SND_SOC_DAPM_OUT_DRV("Haptic Left Switch", + LM49453_P0_DIS_PKVL_FB_REG, 2, 1, NULL, 0), + SND_SOC_DAPM_OUT_DRV("Haptic Right Switch", + LM49453_P0_DIS_PKVL_FB_REG, 3, 1, NULL, 0), + + /* DAC */ + SND_SOC_DAPM_DAC("HPL DAC", "Headset", SND_SOC_NOPM, 0, 0), + SND_SOC_DAPM_DAC("HPR DAC", "Headset", SND_SOC_NOPM, 0, 0), + SND_SOC_DAPM_DAC("LSL DAC", "Speaker", SND_SOC_NOPM, 0, 0), + SND_SOC_DAPM_DAC("LSR DAC", "Speaker", SND_SOC_NOPM, 0, 0), + SND_SOC_DAPM_DAC("HAL DAC", "Haptic", SND_SOC_NOPM, 0, 0), + SND_SOC_DAPM_DAC("HAR DAC", "Haptic", SND_SOC_NOPM, 0, 0), + SND_SOC_DAPM_DAC("LOL DAC", "Lineout", SND_SOC_NOPM, 0, 0), + SND_SOC_DAPM_DAC("LOR DAC", "Lineout", SND_SOC_NOPM, 0, 0), + + + SND_SOC_DAPM_PGA("AUXL Input", + LM49453_P0_ANALOG_MIXER_ADC_REG, 2, 0, NULL, 0), + SND_SOC_DAPM_PGA("AUXR Input", + LM49453_P0_ANALOG_MIXER_ADC_REG, 3, 0, NULL, 0), + + SND_SOC_DAPM_PGA("Sidetone", SND_SOC_NOPM, 0, 0, NULL, 0), + + /* ADC */ + SND_SOC_DAPM_ADC("DMIC1 Left", "Capture", SND_SOC_NOPM, 1, 0), + SND_SOC_DAPM_ADC("DMIC1 Right", "Capture", SND_SOC_NOPM, 1, 0), + SND_SOC_DAPM_ADC("DMIC2 Left", "Capture", SND_SOC_NOPM, 1, 0), + SND_SOC_DAPM_ADC("DMIC2 Right", "Capture", SND_SOC_NOPM, 1, 0), + + SND_SOC_DAPM_ADC("ADC Left", "Capture", SND_SOC_NOPM, 1, 0), + SND_SOC_DAPM_ADC("ADC Right", "Capture", SND_SOC_NOPM, 0, 0), + + SND_SOC_DAPM_MUX("ADCL Mux", SND_SOC_NOPM, 0, 0, + &lm49453_adcl_mux_control), + SND_SOC_DAPM_MUX("ADCR Mux", SND_SOC_NOPM, 0, 0, + &lm49453_adcr_mux_control), + + SND_SOC_DAPM_MUX("Mic1 Input", + SND_SOC_NOPM, 0, 0, &lm49453_adcl_mux_control), + + SND_SOC_DAPM_MUX("Mic2 Input", + SND_SOC_NOPM, 0, 0, &lm49453_adcr_mux_control), + + /* AIF */ + SND_SOC_DAPM_AIF_IN("PORT1_SDI", NULL, 0, + LM49453_P0_PULL_CONFIG1_REG, 2, 0), + SND_SOC_DAPM_AIF_IN("PORT2_SDI", NULL, 0, + LM49453_P0_PULL_CONFIG1_REG, 6, 0), + + SND_SOC_DAPM_AIF_OUT("PORT1_SDO", NULL, 0, + LM49453_P0_PULL_CONFIG1_REG, 3, 0), + SND_SOC_DAPM_AIF_OUT("PORT2_SDO", NULL, 0, + LM49453_P0_PULL_CONFIG1_REG, 7, 0), + + /* Port1 TX controls */ + SND_SOC_DAPM_OUT_DRV("P1_1_TX", SND_SOC_NOPM, 0, 0, NULL, 0), + SND_SOC_DAPM_OUT_DRV("P1_2_TX", SND_SOC_NOPM, 0, 0, NULL, 0), + SND_SOC_DAPM_OUT_DRV("P1_3_TX", SND_SOC_NOPM, 0, 0, NULL, 0), + SND_SOC_DAPM_OUT_DRV("P1_4_TX", SND_SOC_NOPM, 0, 0, NULL, 0), + SND_SOC_DAPM_OUT_DRV("P1_5_TX", SND_SOC_NOPM, 0, 0, NULL, 0), + SND_SOC_DAPM_OUT_DRV("P1_6_TX", SND_SOC_NOPM, 0, 0, NULL, 0), + SND_SOC_DAPM_OUT_DRV("P1_7_TX", SND_SOC_NOPM, 0, 0, NULL, 0), + SND_SOC_DAPM_OUT_DRV("P1_8_TX", SND_SOC_NOPM, 0, 0, NULL, 0), + + /* Port2 TX controls */ + SND_SOC_DAPM_OUT_DRV("P2_1_TX", SND_SOC_NOPM, 0, 0, NULL, 0), + SND_SOC_DAPM_OUT_DRV("P2_2_TX", SND_SOC_NOPM, 0, 0, NULL, 0), + + /* Sidetone Mixer */ + SND_SOC_DAPM_MIXER("Sidetone Mixer", SND_SOC_NOPM, 0, 0, + lm49453_sidetone_mixer_controls, + ARRAY_SIZE(lm49453_sidetone_mixer_controls)), + + /* DAC MIXERS */ + SND_SOC_DAPM_MIXER("HPL Mixer", SND_SOC_NOPM, 0, 0, + lm49453_headset_left_mixer, + ARRAY_SIZE(lm49453_headset_left_mixer)), + SND_SOC_DAPM_MIXER("HPR Mixer", SND_SOC_NOPM, 0, 0, + lm49453_headset_right_mixer, + ARRAY_SIZE(lm49453_headset_right_mixer)), + SND_SOC_DAPM_MIXER("LOL Mixer", SND_SOC_NOPM, 0, 0, + lm49453_lineout_left_mixer, + ARRAY_SIZE(lm49453_lineout_left_mixer)), + SND_SOC_DAPM_MIXER("LOR Mixer", SND_SOC_NOPM, 0, 0, + lm49453_lineout_right_mixer, + ARRAY_SIZE(lm49453_lineout_right_mixer)), + SND_SOC_DAPM_MIXER("LSL Mixer", SND_SOC_NOPM, 0, 0, + lm49453_speaker_left_mixer, + ARRAY_SIZE(lm49453_speaker_left_mixer)), + SND_SOC_DAPM_MIXER("LSR Mixer", SND_SOC_NOPM, 0, 0, + lm49453_speaker_right_mixer, + ARRAY_SIZE(lm49453_speaker_right_mixer)), + SND_SOC_DAPM_MIXER("HAL Mixer", SND_SOC_NOPM, 0, 0, + lm49453_haptic_left_mixer, + ARRAY_SIZE(lm49453_haptic_left_mixer)), + SND_SOC_DAPM_MIXER("HAR Mixer", SND_SOC_NOPM, 0, 0, + lm49453_haptic_right_mixer, + ARRAY_SIZE(lm49453_haptic_right_mixer)), + + /* Capture Mixer */ + SND_SOC_DAPM_MIXER("Port1_1 Mixer", SND_SOC_NOPM, 0, 0, + lm49453_port1_tx1_mixer, + ARRAY_SIZE(lm49453_port1_tx1_mixer)), + SND_SOC_DAPM_MIXER("Port1_2 Mixer", SND_SOC_NOPM, 0, 0, + lm49453_port1_tx2_mixer, + ARRAY_SIZE(lm49453_port1_tx2_mixer)), + SND_SOC_DAPM_MIXER("Port1_3 Mixer", SND_SOC_NOPM, 0, 0, + lm49453_port1_tx3_mixer, + ARRAY_SIZE(lm49453_port1_tx3_mixer)), + SND_SOC_DAPM_MIXER("Port1_4 Mixer", SND_SOC_NOPM, 0, 0, + lm49453_port1_tx4_mixer, + ARRAY_SIZE(lm49453_port1_tx4_mixer)), + SND_SOC_DAPM_MIXER("Port1_5 Mixer", SND_SOC_NOPM, 0, 0, + lm49453_port1_tx5_mixer, + ARRAY_SIZE(lm49453_port1_tx5_mixer)), + SND_SOC_DAPM_MIXER("Port1_6 Mixer", SND_SOC_NOPM, 0, 0, + lm49453_port1_tx6_mixer, + ARRAY_SIZE(lm49453_port1_tx6_mixer)), + SND_SOC_DAPM_MIXER("Port1_7 Mixer", SND_SOC_NOPM, 0, 0, + lm49453_port1_tx7_mixer, + ARRAY_SIZE(lm49453_port1_tx7_mixer)), + SND_SOC_DAPM_MIXER("Port1_8 Mixer", SND_SOC_NOPM, 0, 0, + lm49453_port1_tx8_mixer, + ARRAY_SIZE(lm49453_port1_tx8_mixer)), + + SND_SOC_DAPM_MIXER("Port2_1 Mixer", SND_SOC_NOPM, 0, 0, + lm49453_port2_tx1_mixer, + ARRAY_SIZE(lm49453_port2_tx1_mixer)), + SND_SOC_DAPM_MIXER("Port2_2 Mixer", SND_SOC_NOPM, 0, 0, + lm49453_port2_tx2_mixer, + ARRAY_SIZE(lm49453_port2_tx2_mixer)), +}; + +static const struct snd_soc_dapm_route lm49453_audio_map[] = { + /* Port SDI mapping */ + { "PORT1_1_RX", "Port1 Playback Switch", "PORT1_SDI" }, + { "PORT1_2_RX", "Port1 Playback Switch", "PORT1_SDI" }, + { "PORT1_3_RX", "Port1 Playback Switch", "PORT1_SDI" }, + { "PORT1_4_RX", "Port1 Playback Switch", "PORT1_SDI" }, + { "PORT1_5_RX", "Port1 Playback Switch", "PORT1_SDI" }, + { "PORT1_6_RX", "Port1 Playback Switch", "PORT1_SDI" }, + { "PORT1_7_RX", "Port1 Playback Switch", "PORT1_SDI" }, + { "PORT1_8_RX", "Port1 Playback Switch", "PORT1_SDI" }, + + { "PORT2_1_RX", "Port2 Playback Switch", "PORT2_SDI" }, + { "PORT2_2_RX", "Port2 Playback Switch", "PORT2_SDI" }, + + /* HP mapping */ + { "HPL Mixer", "Port1_1 Switch", "PORT1_1_RX" }, + { "HPL Mixer", "Port1_2 Switch", "PORT1_2_RX" }, + { "HPL Mixer", "Port1_3 Switch", "PORT1_3_RX" }, + { "HPL Mixer", "Port1_4 Switch", "PORT1_4_RX" }, + { "HPL Mixer", "Port1_5 Switch", "PORT1_5_RX" }, + { "HPL Mixer", "Port1_6 Switch", "PORT1_6_RX" }, + { "HPL Mixer", "Port1_7 Switch", "PORT1_7_RX" }, + { "HPL Mixer", "Port1_8 Switch", "PORT1_8_RX" }, + + { "HPL Mixer", "Port2_1 Switch", "PORT2_1_RX" }, + { "HPL Mixer", "Port2_2 Switch", "PORT2_2_RX" }, + + { "HPL Mixer", "ADCL Switch", "ADC Left" }, + { "HPL Mixer", "ADCR Switch", "ADC Right" }, + { "HPL Mixer", "DMIC1L Switch", "DMIC1 Left" }, + { "HPL Mixer", "DMIC1R Switch", "DMIC1 Right" }, + { "HPL Mixer", "DMIC2L Switch", "DMIC2 Left" }, + { "HPL Mixer", "DMIC2R Switch", "DMIC2 Right" }, + { "HPL Mixer", "Sidetone Switch", "Sidetone" }, + + { "HPL DAC", NULL, "HPL Mixer" }, + + { "HPR Mixer", "Port1_1 Switch", "PORT1_1_RX" }, + { "HPR Mixer", "Port1_2 Switch", "PORT1_2_RX" }, + { "HPR Mixer", "Port1_3 Switch", "PORT1_3_RX" }, + { "HPR Mixer", "Port1_4 Switch", "PORT1_4_RX" }, + { "HPR Mixer", "Port1_5 Switch", "PORT1_5_RX" }, + { "HPR Mixer", "Port1_6 Switch", "PORT1_6_RX" }, + { "HPR Mixer", "Port1_7 Switch", "PORT1_7_RX" }, + { "HPR Mixer", "Port1_8 Switch", "PORT1_8_RX" }, + + /* Port 2 */ + { "HPR Mixer", "Port2_1 Switch", "PORT2_1_RX" }, + { "HPR Mixer", "Port2_2 Switch", "PORT2_2_RX" }, + + { "HPR Mixer", "ADCL Switch", "ADC Left" }, + { "HPR Mixer", "ADCR Switch", "ADC Right" }, + { "HPR Mixer", "DMIC1L Switch", "DMIC1 Left" }, + { "HPR Mixer", "DMIC1R Switch", "DMIC1 Right" }, + { "HPR Mixer", "DMIC2L Switch", "DMIC2 Left" }, + { "HPR Mixer", "DMIC2L Switch", "DMIC2 Right" }, + { "HPR Mixer", "Sidetone Switch", "Sidetone" }, + + { "HPR DAC", NULL, "HPR Mixer" }, + + { "HPOUTL", "Headset Switch", "HPL DAC"}, + { "HPOUTR", "Headset Switch", "HPR DAC"}, + + /* EP map */ + { "EPOUT", "Earpiece Switch", "HPL DAC" }, + + /* Speaker map */ + { "LSL Mixer", "Port1_1 Switch", "PORT1_1_RX" }, + { "LSL Mixer", "Port1_2 Switch", "PORT1_2_RX" }, + { "LSL Mixer", "Port1_3 Switch", "PORT1_3_RX" }, + { "LSL Mixer", "Port1_4 Switch", "PORT1_4_RX" }, + { "LSL Mixer", "Port1_5 Switch", "PORT1_5_RX" }, + { "LSL Mixer", "Port1_6 Switch", "PORT1_6_RX" }, + { "LSL Mixer", "Port1_7 Switch", "PORT1_7_RX" }, + { "LSL Mixer", "Port1_8 Switch", "PORT1_8_RX" }, + + /* Port 2 */ + { "LSL Mixer", "Port2_1 Switch", "PORT2_1_RX" }, + { "LSL Mixer", "Port2_2 Switch", "PORT2_2_RX" }, + + { "LSL Mixer", "ADCL Switch", "ADC Left" }, + { "LSL Mixer", "ADCR Switch", "ADC Right" }, + { "LSL Mixer", "DMIC1L Switch", "DMIC1 Left" }, + { "LSL Mixer", "DMIC1R Switch", "DMIC1 Right" }, + { "LSL Mixer", "DMIC2L Switch", "DMIC2 Left" }, + { "LSL Mixer", "DMIC2R Switch", "DMIC2 Right" }, + { "LSL Mixer", "Sidetone Switch", "Sidetone" }, + + { "LSL DAC", NULL, "LSL Mixer" }, + + { "LSR Mixer", "Port1_1 Switch", "PORT1_1_RX" }, + { "LSR Mixer", "Port1_2 Switch", "PORT1_2_RX" }, + { "LSR Mixer", "Port1_3 Switch", "PORT1_3_RX" }, + { "LSR Mixer", "Port1_4 Switch", "PORT1_4_RX" }, + { "LSR Mixer", "Port1_5 Switch", "PORT1_5_RX" }, + { "LSR Mixer", "Port1_6 Switch", "PORT1_6_RX" }, + { "LSR Mixer", "Port1_7 Switch", "PORT1_7_RX" }, + { "LSR Mixer", "Port1_8 Switch", "PORT1_8_RX" }, + + /* Port 2 */ + { "LSR Mixer", "Port2_1 Switch", "PORT2_1_RX" }, + { "LSR Mixer", "Port2_2 Switch", "PORT2_2_RX" }, + + { "LSR Mixer", "ADCL Switch", "ADC Left" }, + { "LSR Mixer", "ADCR Switch", "ADC Right" }, + { "LSR Mixer", "DMIC1L Switch", "DMIC1 Left" }, + { "LSR Mixer", "DMIC1R Switch", "DMIC1 Right" }, + { "LSR Mixer", "DMIC2L Switch", "DMIC2 Left" }, + { "LSR Mixer", "DMIC2R Switch", "DMIC2 Right" }, + { "LSR Mixer", "Sidetone Switch", "Sidetone" }, + + { "LSR DAC", NULL, "LSR Mixer" }, + + { "LSOUTL", "Speaker Left Switch", "LSL DAC"}, + { "LSOUTR", "Speaker Left Switch", "LSR DAC"}, + + /* Haptic map */ + { "HAL Mixer", "Port1_1 Switch", "PORT1_1_RX" }, + { "HAL Mixer", "Port1_2 Switch", "PORT1_2_RX" }, + { "HAL Mixer", "Port1_3 Switch", "PORT1_3_RX" }, + { "HAL Mixer", "Port1_4 Switch", "PORT1_4_RX" }, + { "HAL Mixer", "Port1_5 Switch", "PORT1_5_RX" }, + { "HAL Mixer", "Port1_6 Switch", "PORT1_6_RX" }, + { "HAL Mixer", "Port1_7 Switch", "PORT1_7_RX" }, + { "HAL Mixer", "Port1_8 Switch", "PORT1_8_RX" }, + + /* Port 2 */ + { "HAL Mixer", "Port2_1 Switch", "PORT2_1_RX" }, + { "HAL Mixer", "Port2_2 Switch", "PORT2_2_RX" }, + + { "HAL Mixer", "ADCL Switch", "ADC Left" }, + { "HAL Mixer", "ADCR Switch", "ADC Right" }, + { "HAL Mixer", "DMIC1L Switch", "DMIC1 Left" }, + { "HAL Mixer", "DMIC1R Switch", "DMIC1 Right" }, + { "HAL Mixer", "DMIC2L Switch", "DMIC2 Left" }, + { "HAL Mixer", "DMIC2R Switch", "DMIC2 Right" }, + { "HAL Mixer", "Sidetone Switch", "Sidetone" }, + + { "HAL DAC", NULL, "HAL Mixer" }, + + { "HAR Mixer", "Port1_1 Switch", "PORT1_1_RX" }, + { "HAR Mixer", "Port1_2 Switch", "PORT1_2_RX" }, + { "HAR Mixer", "Port1_3 Switch", "PORT1_3_RX" }, + { "HAR Mixer", "Port1_4 Switch", "PORT1_4_RX" }, + { "HAR Mixer", "Port1_5 Switch", "PORT1_5_RX" }, + { "HAR Mixer", "Port1_6 Switch", "PORT1_6_RX" }, + { "HAR Mixer", "Port1_7 Switch", "PORT1_7_RX" }, + { "HAR Mixer", "Port1_8 Switch", "PORT1_8_RX" }, + + /* Port 2 */ + { "HAR Mixer", "Port2_1 Switch", "PORT2_1_RX" }, + { "HAR Mixer", "Port2_2 Switch", "PORT2_2_RX" }, + + { "HAR Mixer", "ADCL Switch", "ADC Left" }, + { "HAR Mixer", "ADCR Switch", "ADC Right" }, + { "HAR Mixer", "DMIC1L Switch", "DMIC1 Left" }, + { "HAR Mixer", "DMIC1R Switch", "DMIC1 Right" }, + { "HAR Mixer", "DMIC2L Switch", "DMIC2 Left" }, + { "HAR Mixer", "DMIC2R Switch", "DMIC2 Right" }, + { "HAR Mixer", "Sideton Switch", "Sidetone" }, + + { "HAR DAC", NULL, "HAR Mixer" }, + + { "HAOUTL", "Haptic Left Switch", "HAL DAC" }, + { "HAOUTR", "Haptic Right Switch", "HAR DAC" }, + + /* Lineout map */ + { "LOL Mixer", "Port1_1 Switch", "PORT1_1_RX" }, + { "LOL Mixer", "Port1_2 Switch", "PORT1_2_RX" }, + { "LOL Mixer", "Port1_3 Switch", "PORT1_3_RX" }, + { "LOL Mixer", "Port1_4 Switch", "PORT1_4_RX" }, + { "LOL Mixer", "Port1_5 Switch", "PORT1_5_RX" }, + { "LOL Mixer", "Port1_6 Switch", "PORT1_6_RX" }, + { "LOL Mixer", "Port1_7 Switch", "PORT1_7_RX" }, + { "LOL Mixer", "Port1_8 Switch", "PORT1_8_RX" }, + + /* Port 2 */ + { "LOL Mixer", "Port2_1 Switch", "PORT2_1_RX" }, + { "LOL Mixer", "Port2_2 Switch", "PORT2_2_RX" }, + + { "LOL Mixer", "ADCL Switch", "ADC Left" }, + { "LOL Mixer", "ADCR Switch", "ADC Right" }, + { "LOL Mixer", "DMIC1L Switch", "DMIC1 Left" }, + { "LOL Mixer", "DMIC1R Switch", "DMIC1 Right" }, + { "LOL Mixer", "DMIC2L Switch", "DMIC2 Left" }, + { "LOL Mixer", "DMIC2R Switch", "DMIC2 Right" }, + { "LOL Mixer", "Sidetone Switch", "Sidetone" }, + + { "LOL DAC", NULL, "LOL Mixer" }, + + { "LOR Mixer", "Port1_1 Switch", "PORT1_1_RX" }, + { "LOR Mixer", "Port1_2 Switch", "PORT1_2_RX" }, + { "LOR Mixer", "Port1_3 Switch", "PORT1_3_RX" }, + { "LOR Mixer", "Port1_4 Switch", "PORT1_4_RX" }, + { "LOR Mixer", "Port1_5 Switch", "PORT1_5_RX" }, + { "LOR Mixer", "Port1_6 Switch", "PORT1_6_RX" }, + { "LOR Mixer", "Port1_7 Switch", "PORT1_7_RX" }, + { "LOR Mixer", "Port1_8 Switch", "PORT1_8_RX" }, + + /* Port 2 */ + { "LOR Mixer", "Port2_1 Switch", "PORT2_1_RX" }, + { "LOR Mixer", "Port2_2 Switch", "PORT2_2_RX" }, + + { "LOR Mixer", "ADCL Switch", "ADC Left" }, + { "LOR Mixer", "ADCR Switch", "ADC Right" }, + { "LOR Mixer", "DMIC1L Switch", "DMIC1 Left" }, + { "LOR Mixer", "DMIC1R Switch", "DMIC1 Right" }, + { "LOR Mixer", "DMIC2L Switch", "DMIC2 Left" }, + { "LOR Mixer", "DMIC2R Switch", "DMIC2 Right" }, + { "LOR Mixer", "Sidetone Switch", "Sidetone" }, + + { "LOR DAC", NULL, "LOR Mixer" }, + + { "LOOUTL", NULL, "LOL DAC" }, + { "LOOUTR", NULL, "LOR DAC" }, + + /* TX map */ + /* Port1 mappings */ + { "Port1_1 Mixer", "ADCL Switch", "ADC Left" }, + { "Port1_1 Mixer", "ADCR Switch", "ADC Right" }, + { "Port1_1 Mixer", "DMIC1L Switch", "DMIC1 Left" }, + { "Port1_1 Mixer", "DMIC1R Switch", "DMIC1 Right" }, + { "Port1_1 Mixer", "DMIC2L Switch", "DMIC2 Left" }, + { "Port1_1 Mixer", "DMIC2R Switch", "DMIC2 Right" }, + + { "Port1_2 Mixer", "ADCL Switch", "ADC Left" }, + { "Port1_2 Mixer", "ADCR Switch", "ADC Right" }, + { "Port1_2 Mixer", "DMIC1L Switch", "DMIC1 Left" }, + { "Port1_2 Mixer", "DMIC1R Switch", "DMIC1 Right" }, + { "Port1_2 Mixer", "DMIC2L Switch", "DMIC2 Left" }, + { "Port1_2 Mixer", "DMIC2R Switch", "DMIC2 Right" }, + + { "Port1_3 Mixer", "ADCL Switch", "ADC Left" }, + { "Port1_3 Mixer", "ADCR Switch", "ADC Right" }, + { "Port1_3 Mixer", "DMIC1L Switch", "DMIC1 Left" }, + { "Port1_3 Mixer", "DMIC1R Switch", "DMIC1 Right" }, + { "Port1_3 Mixer", "DMIC2L Switch", "DMIC2 Left" }, + { "Port1_3 Mixer", "DMIC2R Switch", "DMIC2 Right" }, + + { "Port1_4 Mixer", "ADCL Switch", "ADC Left" }, + { "Port1_4 Mixer", "ADCR Switch", "ADC Right" }, + { "Port1_4 Mixer", "DMIC1L Switch", "DMIC1 Left" }, + { "Port1_4 Mixer", "DMIC1R Switch", "DMIC1 Right" }, + { "Port1_4 Mixer", "DMIC2L Switch", "DMIC2 Left" }, + { "Port1_4 Mixer", "DMIC2R Switch", "DMIC2 Right" }, + + { "Port1_5 Mixer", "ADCL Switch", "ADC Left" }, + { "Port1_5 Mixer", "ADCR Switch", "ADC Right" }, + { "Port1_5 Mixer", "DMIC1L Switch", "DMIC1 Left" }, + { "Port1_5 Mixer", "DMIC1R Switch", "DMIC1 Right" }, + { "Port1_5 Mixer", "DMIC2L Switch", "DMIC2 Left" }, + { "Port1_5 Mixer", "DMIC2R Switch", "DMIC2 Right" }, + + { "Port1_6 Mixer", "ADCL Switch", "ADC Left" }, + { "Port1_6 Mixer", "ADCR Switch", "ADC Right" }, + { "Port1_6 Mixer", "DMIC1L Switch", "DMIC1 Left" }, + { "Port1_6 Mixer", "DMIC1R Switch", "DMIC1 Right" }, + { "Port1_6 Mixer", "DMIC2L Switch", "DMIC2 Left" }, + { "Port1_6 Mixer", "DMIC2R Switch", "DMIC2 Right" }, + + { "Port1_7 Mixer", "ADCL Switch", "ADC Left" }, + { "Port1_7 Mixer", "ADCR Switch", "ADC Right" }, + { "Port1_7 Mixer", "DMIC1L Switch", "DMIC1 Left" }, + { "Port1_7 Mixer", "DMIC1R Switch", "DMIC1 Right" }, + { "Port1_7 Mixer", "DMIC2L Switch", "DMIC2 Left" }, + { "Port1_7 Mixer", "DMIC2R Switch", "DMIC2 Right" }, + + { "Port1_8 Mixer", "ADCL Switch", "ADC Left" }, + { "Port1_8 Mixer", "ADCR Switch", "ADC Right" }, + { "Port1_8 Mixer", "DMIC1L Switch", "DMIC1 Left" }, + { "Port1_8 Mixer", "DMIC1R Switch", "DMIC1 Right" }, + { "Port1_8 Mixer", "DMIC2L Switch", "DMIC2 Left" }, + { "Port1_8 Mixer", "DMIC2R Switch", "DMIC2 Right" }, + + { "Port2_1 Mixer", "ADCL Switch", "ADC Left" }, + { "Port2_1 Mixer", "ADCR Switch", "ADC Right" }, + { "Port2_1 Mixer", "DMIC1L Switch", "DMIC1 Left" }, + { "Port2_1 Mixer", "DMIC1R Switch", "DMIC1 Right" }, + { "Port2_1 Mixer", "DMIC2L Switch", "DMIC2 Left" }, + { "Port2_1 Mixer", "DMIC2R Switch", "DMIC2 Right" }, + + { "Port2_2 Mixer", "ADCL Switch", "ADC Left" }, + { "Port2_2 Mixer", "ADCR Switch", "ADC Right" }, + { "Port2_2 Mixer", "DMIC1L Switch", "DMIC1 Left" }, + { "Port2_2 Mixer", "DMIC1R Switch", "DMIC1 Right" }, + { "Port2_2 Mixer", "DMIC2L Switch", "DMIC2 Left" }, + { "Port2_2 Mixer", "DMIC2R Switch", "DMIC2 Right" }, + + { "P1_1_TX", NULL, "Port1_1 Mixer" }, + { "P1_2_TX", NULL, "Port1_2 Mixer" }, + { "P1_3_TX", NULL, "Port1_3 Mixer" }, + { "P1_4_TX", NULL, "Port1_4 Mixer" }, + { "P1_5_TX", NULL, "Port1_5 Mixer" }, + { "P1_6_TX", NULL, "Port1_6 Mixer" }, + { "P1_7_TX", NULL, "Port1_7 Mixer" }, + { "P1_8_TX", NULL, "Port1_8 Mixer" }, + + { "P2_1_TX", NULL, "Port2_1 Mixer" }, + { "P2_2_TX", NULL, "Port2_2 Mixer" }, + + { "PORT1_SDO", "Port1 Capture Switch", "P1_1_TX"}, + { "PORT1_SDO", "Port1 Capture Switch", "P1_2_TX"}, + { "PORT1_SDO", "Port1 Capture Switch", "P1_3_TX"}, + { "PORT1_SDO", "Port1 Capture Switch", "P1_4_TX"}, + { "PORT1_SDO", "Port1 Capture Switch", "P1_5_TX"}, + { "PORT1_SDO", "Port1 Capture Switch", "P1_6_TX"}, + { "PORT1_SDO", "Port1 Capture Switch", "P1_7_TX"}, + { "PORT1_SDO", "Port1 Capture Switch", "P1_8_TX"}, + + { "PORT2_SDO", "Port2 Capture Switch", "P2_1_TX"}, + { "PORT2_SDO", "Port2 Capture Switch", "P2_2_TX"}, + + { "Mic1 Input", NULL, "AMIC1" }, + { "Mic2 Input", NULL, "AMIC2" }, + + { "AUXL Input", NULL, "AUXL" }, + { "AUXR Input", NULL, "AUXR" }, + + /* AUX connections */ + { "ADCL Mux", "Aux_L", "AUXL Input" }, + { "ADCL Mux", "MIC1", "Mic1 Input" }, + + { "ADCR Mux", "Aux_R", "AUXR Input" }, + { "ADCR Mux", "MIC2", "Mic2 Input" }, + + /* ADC connection */ + { "ADC Left", NULL, "ADCL Mux"}, + { "ADC Right", NULL, "ADCR Mux"}, + + { "DMIC1 Left", NULL, "DMIC1DAT"}, + { "DMIC1 Right", NULL, "DMIC1DAT"}, + { "DMIC2 Left", NULL, "DMIC2DAT"}, + { "DMIC2 Right", NULL, "DMIC2DAT"}, + + /* Sidetone map */ + { "Sidetone Mixer", NULL, "ADC Left" }, + { "Sidetone Mixer", NULL, "ADC Right" }, + { "Sidetone Mixer", NULL, "DMIC1 Left" }, + { "Sidetone Mixer", NULL, "DMIC1 Right" }, + { "Sidetone Mixer", NULL, "DMIC2 Left" }, + { "Sidetone Mixer", NULL, "DMIC2 Right" }, + + { "Sidetone", "Sidetone Switch", "Sidetone Mixer" }, +}; + +static int lm49453_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params, + struct snd_soc_dai *dai) +{ + struct snd_soc_codec *codec = dai->codec; + struct lm49453_priv *lm49453 = snd_soc_codec_get_drvdata(codec); + u16 clk_div = 0; + + lm49453->fs_rate = params_rate(params); + + /* Setting DAC clock dividers based on substream sample rate. */ + switch (lm49453->fs_rate) { + case 8000: + case 16000: + case 32000: + case 24000: + case 48000: + clk_div = 256; + break; + case 11025: + case 22050: + case 44100: + clk_div = 216; + break; + case 96000: + clk_div = 127; + break; + default: + return -EINVAL; + } + + snd_soc_write(codec, LM49453_P0_ADC_CLK_DIV_REG, clk_div); + snd_soc_write(codec, LM49453_P0_DAC_HP_CLK_DIV_REG, clk_div); + + return 0; +} + +static int lm49453_set_dai_fmt(struct snd_soc_dai *codec_dai, unsigned int fmt) +{ + struct snd_soc_codec *codec = codec_dai->codec; + + u16 aif_val; + int mode = 0; + int clk_phase = 0; + int clk_shift = 0; + + switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) { + case SND_SOC_DAIFMT_CBS_CFS: + aif_val = 0; + break; + case SND_SOC_DAIFMT_CBS_CFM: + aif_val = LM49453_AUDIO_PORT1_BASIC_SYNC_MS; + break; + case SND_SOC_DAIFMT_CBM_CFS: + aif_val = LM49453_AUDIO_PORT1_BASIC_CLK_MS; + break; + case SND_SOC_DAIFMT_CBM_CFM: + aif_val = LM49453_AUDIO_PORT1_BASIC_CLK_MS | + LM49453_AUDIO_PORT1_BASIC_SYNC_MS; + break; + default: + return -EINVAL; + } + + + switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) { + case SND_SOC_DAIFMT_I2S: + break; + case SND_SOC_DAIFMT_DSP_A: + mode = 1; + clk_phase = (1 << 5); + clk_shift = 1; + break; + case SND_SOC_DAIFMT_DSP_B: + mode = 1; + clk_phase = (1 << 5); + clk_shift = 0; + break; + default: + return -EINVAL; + } + + snd_soc_update_bits(codec, LM49453_P0_AUDIO_PORT1_BASIC_REG, + LM49453_AUDIO_PORT1_BASIC_FMT_MASK|BIT(1)|BIT(5), + (aif_val | mode | clk_phase)); + + snd_soc_write(codec, LM49453_P0_AUDIO_PORT1_RX_MSB_REG, clk_shift); + + return 0; +} + +static int lm49453_set_dai_sysclk(struct snd_soc_dai *dai, int clk_id, + unsigned int freq, int dir) +{ + struct snd_soc_codec *codec = dai->codec; + u16 pll_clk = 0; + + switch (freq) { + case 12288000: + case 26000000: + case 19200000: + /* pll clk slection */ + pll_clk = 0; + break; + case 48000: + case 32576: + /* fll clk slection */ + pll_clk = BIT(4); + return 0; + default: + return -EINVAL; + } + + snd_soc_update_bits(codec, LM49453_P0_PMC_SETUP_REG, BIT(4), pll_clk); + + return 0; +} + +static int lm49453_hp_mute(struct snd_soc_dai *dai, int mute) +{ + snd_soc_update_bits(dai->codec, LM49453_P0_DAC_DSP_REG, BIT(1)|BIT(0), + (mute ? (BIT(1)|BIT(0)) : 0)); + return 0; +} + +static int lm49453_lo_mute(struct snd_soc_dai *dai, int mute) +{ + snd_soc_update_bits(dai->codec, LM49453_P0_DAC_DSP_REG, BIT(3)|BIT(2), + (mute ? (BIT(3)|BIT(2)) : 0)); + return 0; +} + +static int lm49453_ls_mute(struct snd_soc_dai *dai, int mute) +{ + snd_soc_update_bits(dai->codec, LM49453_P0_DAC_DSP_REG, BIT(5)|BIT(4), + (mute ? (BIT(5)|BIT(4)) : 0)); + return 0; +} + +static int lm49453_ep_mute(struct snd_soc_dai *dai, int mute) +{ + snd_soc_update_bits(dai->codec, LM49453_P0_DAC_DSP_REG, BIT(4), + (mute ? BIT(4) : 0)); + return 0; +} + +static int lm49453_ha_mute(struct snd_soc_dai *dai, int mute) +{ + snd_soc_update_bits(dai->codec, LM49453_P0_DAC_DSP_REG, BIT(7)|BIT(6), + (mute ? (BIT(7)|BIT(6)) : 0)); + return 0; +} + +static int lm49453_set_bias_level(struct snd_soc_codec *codec, + enum snd_soc_bias_level level) +{ + struct lm49453_priv *lm49453 = snd_soc_codec_get_drvdata(codec); + + switch (level) { + case SND_SOC_BIAS_ON: + case SND_SOC_BIAS_PREPARE: + break; + + case SND_SOC_BIAS_STANDBY: + if (codec->dapm.bias_level == SND_SOC_BIAS_OFF) + regcache_sync(lm49453->regmap); + + snd_soc_update_bits(codec, LM49453_P0_PMC_SETUP_REG, + LM49453_PMC_SETUP_CHIP_EN, LM49453_CHIP_EN); + break; + + case SND_SOC_BIAS_OFF: + snd_soc_update_bits(codec, LM49453_P0_PMC_SETUP_REG, + LM49453_PMC_SETUP_CHIP_EN, 0); + break; + } + + codec->dapm.bias_level = level; + + return 0; +} + +/* Formates supported by LM49453 driver. */ +#define LM49453_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S20_3LE |\ + SNDRV_PCM_FMTBIT_S24_LE | SNDRV_PCM_FMTBIT_S32_LE) + +static struct snd_soc_dai_ops lm49453_headset_dai_ops = { + .hw_params = lm49453_hw_params, + .set_sysclk = lm49453_set_dai_sysclk, + .set_fmt = lm49453_set_dai_fmt, + .digital_mute = lm49453_hp_mute, +}; + +static struct snd_soc_dai_ops lm49453_speaker_dai_ops = { + .hw_params = lm49453_hw_params, + .set_sysclk = lm49453_set_dai_sysclk, + .set_fmt = lm49453_set_dai_fmt, + .digital_mute = lm49453_ls_mute, +}; + +static struct snd_soc_dai_ops lm49453_haptic_dai_ops = { + .hw_params = lm49453_hw_params, + .set_sysclk = lm49453_set_dai_sysclk, + .set_fmt = lm49453_set_dai_fmt, + .digital_mute = lm49453_ha_mute, +}; + +static struct snd_soc_dai_ops lm49453_ep_dai_ops = { + .hw_params = lm49453_hw_params, + .set_sysclk = lm49453_set_dai_sysclk, + .set_fmt = lm49453_set_dai_fmt, + .digital_mute = lm49453_ep_mute, +}; + +static struct snd_soc_dai_ops lm49453_lineout_dai_ops = { + .hw_params = lm49453_hw_params, + .set_sysclk = lm49453_set_dai_sysclk, + .set_fmt = lm49453_set_dai_fmt, + .digital_mute = lm49453_lo_mute, +}; + +/* LM49453 dai structure. */ +static const struct snd_soc_dai_driver lm49453_dai[] = { + { + .name = "LM49453 Headset", + .playback = { + .stream_name = "Headset", + .channels_min = 2, + .channels_max = 2, + .rates = SNDRV_PCM_RATE_8000_192000, + .formats = LM49453_FORMATS, + }, + .capture = { + .stream_name = "Capture", + .channels_min = 1, + .channels_max = 5, + .rates = SNDRV_PCM_RATE_8000_192000, + .formats = LM49453_FORMATS, + }, + .ops = &lm49453_headset_dai_ops, + .symmetric_rates = 1, + }, + { + .name = "LM49453 Speaker", + .playback = { + .stream_name = "Speaker", + .channels_min = 2, + .channels_max = 2, + .rates = SNDRV_PCM_RATE_8000_192000, + .formats = LM49453_FORMATS, + }, + .ops = &lm49453_speaker_dai_ops, + }, + { + .name = "LM49453 Haptic", + .playback = { + .stream_name = "Haptic", + .channels_min = 2, + .channels_max = 2, + .rates = SNDRV_PCM_RATE_8000_192000, + .formats = LM49453_FORMATS, + }, + .ops = &lm49453_haptic_dai_ops, + }, + { + .name = "LM49453 Earpiece", + .playback = { + .stream_name = "Earpiece", + .channels_min = 1, + .channels_max = 1, + .rates = SNDRV_PCM_RATE_8000_192000, + .formats = LM49453_FORMATS, + }, + .ops = &lm49453_ep_dai_ops, + }, + { + .name = "LM49453 line out", + .playback = { + .stream_name = "Lineout", + .channels_min = 2, + .channels_max = 2, + .rates = SNDRV_PCM_RATE_8000_192000, + .formats = LM49453_FORMATS, + }, + .ops = &lm49453_lineout_dai_ops, + }, +}; + +static int lm49453_suspend(struct snd_soc_codec *codec) +{ + lm49453_set_bias_level(codec, SND_SOC_BIAS_OFF); + return 0; +} + +static int lm49453_resume(struct snd_soc_codec *codec) +{ + lm49453_set_bias_level(codec, SND_SOC_BIAS_STANDBY); + return 0; +} + +static int lm49453_probe(struct snd_soc_codec *codec) +{ + struct lm49453_priv *lm49453 = snd_soc_codec_get_drvdata(codec); + int ret = 0; + + codec->control_data = lm49453->regmap; + + ret = snd_soc_codec_set_cache_io(codec, 8, 8, SND_SOC_REGMAP); + if (ret < 0) { + dev_err(codec->dev, "Failed to set cache I/O: %d\n", ret); + return ret; + } + + return 0; +} + +/* power down chip */ +static int lm49453_remove(struct snd_soc_codec *codec) +{ + lm49453_set_bias_level(codec, SND_SOC_BIAS_OFF); + return 0; +} + +static struct snd_soc_codec_driver soc_codec_dev_lm49453 = { + .probe = lm49453_probe, + .remove = lm49453_remove, + .suspend = lm49453_suspend, + .resume = lm49453_resume, + .set_bias_level = lm49453_set_bias_level, + .controls = lm49453_snd_controls, + .num_controls = ARRAY_SIZE(lm49453_snd_controls), + .dapm_widgets = lm49453_dapm_widgets, + .num_dapm_widgets = ARRAY_SIZE(lm49453_dapm_widgets), + .dapm_routes = lm49453_audio_map, + .num_dapm_routes = ARRAY_SIZE(lm49453_audio_map), + .idle_bias_off = true, +}; + +static const struct regmap_config lm49453_regmap_config = { + .reg_bits = 8, + .val_bits = 8, + + .max_register = LM49453_MAX_REGISTER, + .reg_defaults = lm49453_reg_defs, + .num_reg_defaults = ARRAY_SIZE(lm49453_reg_defs), + .cache_type = REGCACHE_RBTREE, +}; + +static __devinit int lm49453_i2c_probe(struct i2c_client *i2c, + const struct i2c_device_id *id) +{ + struct lm49453_priv *lm49453; + int ret = 0; + + lm49453 = devm_kzalloc(&i2c->dev, sizeof(struct lm49453_priv), + GFP_KERNEL); + + if (lm49453 == NULL) + return -ENOMEM; + + i2c_set_clientdata(i2c, lm49453); + + lm49453->regmap = regmap_init_i2c(i2c, &lm49453_regmap_config); + if (IS_ERR(lm49453->regmap)) { + ret = PTR_ERR(lm49453->regmap); + dev_err(&i2c->dev, "Failed to allocate register map: %d\n", + ret); + return ret; + } + + ret = snd_soc_register_codec(&i2c->dev, + &soc_codec_dev_lm49453, + lm49453_dai, ARRAY_SIZE(lm49453_dai)); + if (ret < 0) { + dev_err(&i2c->dev, "Failed to register codec: %d\n", ret); + regmap_exit(lm49453->regmap); + return ret; + } + + return ret; +} + +static int __devexit lm49453_i2c_remove(struct i2c_client *client) +{ + struct lm49453_priv *lm49453 = i2c_get_clientdata(client); + + snd_soc_unregister_codec(&client->dev); + regmap_exit(lm49453->regmap); + return 0; +} + +static const struct i2c_device_id lm49453_i2c_id[] = { + { "lm49453", 0 }, + { } +}; +MODULE_DEVICE_TABLE(i2c, lm49453_i2c_id); + +static struct i2c_driver lm49453_i2c_driver = { + .driver = { + .name = "lm49453", + .owner = THIS_MODULE, + }, + .probe = lm49453_i2c_probe, + .remove = __devexit_p(lm49453_i2c_remove), + .id_table = lm49453_i2c_id, +}; + +module_i2c_driver(lm49453_i2c_driver); + +MODULE_DESCRIPTION("ASoC LM49453 driver"); +MODULE_AUTHOR("M R Swami Reddy <MR.Swami.Reddy@ti.com>"); +MODULE_LICENSE("GPL v2"); diff --git a/sound/soc/codecs/lm49453.h b/sound/soc/codecs/lm49453.h new file mode 100644 index 00000000000..a63cfa5c088 --- /dev/null +++ b/sound/soc/codecs/lm49453.h @@ -0,0 +1,380 @@ +/* + * lm49453.h - LM49453 ALSA Soc Audio drive + * + * Copyright (c) 2012 Texas Instruments, Inc + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License as published by + * the Free Software Foundation; version 2 of the License. + * + */ + +#ifndef _LM49453_H +#define _LM49453_H + +#include <linux/bitops.h> + +/* LM49453_P0 register space for page0 */ +#define LM49453_P0_PMC_SETUP_REG 0x00 +#define LM49453_P0_PLL_CLK_SEL1_REG 0x01 +#define LM49453_P0_PLL_CLK_SEL2_REG 0x02 +#define LM49453_P0_PMC_CLK_DIV_REG 0x03 +#define LM49453_P0_HSDET_CLK_DIV_REG 0x04 +#define LM49453_P0_DMIC_CLK_DIV_REG 0x05 +#define LM49453_P0_ADC_CLK_DIV_REG 0x06 +#define LM49453_P0_DAC_OT_CLK_DIV_REG 0x07 +#define LM49453_P0_PLL_HF_M_REG 0x08 +#define LM49453_P0_PLL_LF_M_REG 0x09 +#define LM49453_P0_PLL_NL_REG 0x0A +#define LM49453_P0_PLL_N_MODL_REG 0x0B +#define LM49453_P0_PLL_N_MODH_REG 0x0C +#define LM49453_P0_PLL_P1_REG 0x0D +#define LM49453_P0_PLL_P2_REG 0x0E +#define LM49453_P0_FLL_REF_FREQL_REG 0x0F +#define LM49453_P0_FLL_REF_FREQH_REG 0x10 +#define LM49453_P0_VCO_TARGETLL_REG 0x11 +#define LM49453_P0_VCO_TARGETLH_REG 0x12 +#define LM49453_P0_VCO_TARGETHL_REG 0x13 +#define LM49453_P0_VCO_TARGETHH_REG 0x14 +#define LM49453_P0_PLL_CONFIG_REG 0x15 +#define LM49453_P0_DAC_CLK_SEL_REG 0x16 +#define LM49453_P0_DAC_HP_CLK_DIV_REG 0x17 + +/* Analog Mixer Input Stages */ +#define LM49453_P0_MICL_REG 0x20 +#define LM49453_P0_MICR_REG 0x21 +#define LM49453_P0_EP_REG 0x24 +#define LM49453_P0_DIS_PKVL_FB_REG 0x25 + +/* Analog Mixer Output Stages */ +#define LM49453_P0_ANALOG_MIXER_ADC_REG 0x2E + +/*ADC or DAC */ +#define LM49453_P0_ADC_DSP_REG 0x30 +#define LM49453_P0_DAC_DSP_REG 0x31 + +/* EFFECTS ENABLES */ +#define LM49453_P0_ADC_FX_ENABLES_REG 0x33 + +/* GPIO */ +#define LM49453_P0_GPIO1_REG 0x38 +#define LM49453_P0_GPIO2_REG 0x39 +#define LM49453_P0_GPIO3_REG 0x3A +#define LM49453_P0_HAP_CTL_REG 0x3B +#define LM49453_P0_HAP_FREQ_PROG_LEFTL_REG 0x3C +#define LM49453_P0_HAP_FREQ_PROG_LEFTH_REG 0x3D +#define LM49453_P0_HAP_FREQ_PROG_RIGHTL_REG 0x3E +#define LM49453_P0_HAP_FREQ_PROG_RIGHTH_REG 0x3F + +/* DIGITAL MIXER */ +#define LM49453_P0_DMIX_CLK_SEL_REG 0x40 +#define LM49453_P0_PORT1_RX_LVL1_REG 0x41 +#define LM49453_P0_PORT1_RX_LVL2_REG 0x42 +#define LM49453_P0_PORT2_RX_LVL_REG 0x43 +#define LM49453_P0_PORT1_TX1_REG 0x44 +#define LM49453_P0_PORT1_TX2_REG 0x45 +#define LM49453_P0_PORT1_TX3_REG 0x46 +#define LM49453_P0_PORT1_TX4_REG 0x47 +#define LM49453_P0_PORT1_TX5_REG 0x48 +#define LM49453_P0_PORT1_TX6_REG 0x49 +#define LM49453_P0_PORT1_TX7_REG 0x4A +#define LM49453_P0_PORT1_TX8_REG 0x4B +#define LM49453_P0_PORT2_TX1_REG 0x4C +#define LM49453_P0_PORT2_TX2_REG 0x4D +#define LM49453_P0_STN_SEL_REG 0x4F +#define LM49453_P0_DACHPL1_REG 0x50 +#define LM49453_P0_DACHPL2_REG 0x51 +#define LM49453_P0_DACHPR1_REG 0x52 +#define LM49453_P0_DACHPR2_REG 0x53 +#define LM49453_P0_DACLOL1_REG 0x54 +#define LM49453_P0_DACLOL2_REG 0x55 +#define LM49453_P0_DACLOR1_REG 0x56 +#define LM49453_P0_DACLOR2_REG 0x57 +#define LM49453_P0_DACLSL1_REG 0x58 +#define LM49453_P0_DACLSL2_REG 0x59 +#define LM49453_P0_DACLSR1_REG 0x5A +#define LM49453_P0_DACLSR2_REG 0x5B +#define LM49453_P0_DACHAL1_REG 0x5C +#define LM49453_P0_DACHAL2_REG 0x5D +#define LM49453_P0_DACHAR1_REG 0x5E +#define LM49453_P0_DACHAR2_REG 0x5F + +/* AUDIO PORT 1 (TDM) */ +#define LM49453_P0_AUDIO_PORT1_BASIC_REG 0x60 +#define LM49453_P0_AUDIO_PORT1_CLK_GEN1_REG 0x61 +#define LM49453_P0_AUDIO_PORT1_CLK_GEN2_REG 0x62 +#define LM49453_P0_AUDIO_PORT1_CLK_GEN3_REG 0x63 +#define LM49453_P0_AUDIO_PORT1_SYNC_RATE_REG 0x64 +#define LM49453_P0_AUDIO_PORT1_SYNC_SDO_SETUP_REG 0x65 +#define LM49453_P0_AUDIO_PORT1_DATA_WIDTH_REG 0x66 +#define LM49453_P0_AUDIO_PORT1_RX_MSB_REG 0x67 +#define LM49453_P0_AUDIO_PORT1_TX_MSB_REG 0x68 +#define LM49453_P0_AUDIO_PORT1_TDM_CHANNELS_REG 0x69 + +/* AUDIO PORT 2 */ +#define LM49453_P0_AUDIO_PORT2_BASIC_REG 0x6A +#define LM49453_P0_AUDIO_PORT2_CLK_GEN1_REG 0x6B +#define LM49453_P0_AUDIO_PORT2_CLK_GEN2_REG 0x6C +#define LM49453_P0_AUDIO_PORT2_SYNC_GEN_REG 0x6D +#define LM49453_P0_AUDIO_PORT2_DATA_WIDTH_REG 0x6E +#define LM49453_P0_AUDIO_PORT2_RX_MODE_REG 0x6F +#define LM49453_P0_AUDIO_PORT2_TX_MODE_REG 0x70 + +/* SAMPLE RATE */ +#define LM49453_P0_PORT1_SR_LSB_REG 0x79 +#define LM49453_P0_PORT1_SR_MSB_REG 0x7A +#define LM49453_P0_PORT2_SR_LSB_REG 0x7B +#define LM49453_P0_PORT2_SR_MSB_REG 0x7C + +/* EFFECTS - HPFs */ +#define LM49453_P0_HPF_REG 0x80 + +/* EFFECTS ADC ALC */ +#define LM49453_P0_ADC_ALC1_REG 0x82 +#define LM49453_P0_ADC_ALC2_REG 0x83 +#define LM49453_P0_ADC_ALC3_REG 0x84 +#define LM49453_P0_ADC_ALC4_REG 0x85 +#define LM49453_P0_ADC_ALC5_REG 0x86 +#define LM49453_P0_ADC_ALC6_REG 0x87 +#define LM49453_P0_ADC_ALC7_REG 0x88 +#define LM49453_P0_ADC_ALC8_REG 0x89 +#define LM49453_P0_DMIC1_LEVELL_REG 0x8A +#define LM49453_P0_DMIC1_LEVELR_REG 0x8B +#define LM49453_P0_DMIC2_LEVELL_REG 0x8C +#define LM49453_P0_DMIC2_LEVELR_REG 0x8D +#define LM49453_P0_ADC_LEVELL_REG 0x8E +#define LM49453_P0_ADC_LEVELR_REG 0x8F +#define LM49453_P0_DAC_HP_LEVELL_REG 0x90 +#define LM49453_P0_DAC_HP_LEVELR_REG 0x91 +#define LM49453_P0_DAC_LO_LEVELL_REG 0x92 +#define LM49453_P0_DAC_LO_LEVELR_REG 0x93 +#define LM49453_P0_DAC_LS_LEVELL_REG 0x94 +#define LM49453_P0_DAC_LS_LEVELR_REG 0x95 +#define LM49453_P0_DAC_HA_LEVELL_REG 0x96 +#define LM49453_P0_DAC_HA_LEVELR_REG 0x97 +#define LM49453_P0_SOFT_MUTE_REG 0x98 +#define LM49453_P0_DMIC_MUTE_CFG_REG 0x99 +#define LM49453_P0_ADC_MUTE_CFG_REG 0x9A +#define LM49453_P0_DAC_MUTE_CFG_REG 0x9B + +/*DIGITAL MIC1 */ +#define LM49453_P0_DIGITAL_MIC1_CONFIG_REG 0xB0 +#define LM49453_P0_DIGITAL_MIC1_DATA_DELAYL_REG 0xB1 +#define LM49453_P0_DIGITAL_MIC1_DATA_DELAYR_REG 0xB2 + +/*DIGITAL MIC2 */ +#define LM49453_P0_DIGITAL_MIC2_CONFIG_REG 0xB3 +#define LM49453_P0_DIGITAL_MIC2_DATA_DELAYL_REG 0xB4 +#define LM49453_P0_DIGITAL_MIC2_DATA_DELAYR_REG 0xB5 + +/* ADC DECIMATOR */ +#define LM49453_P0_ADC_DECIMATOR_REG 0xB6 + +/* DAC CONFIGURE */ +#define LM49453_P0_DAC_CONFIG_REG 0xB7 + +/* SIDETONE */ +#define LM49453_P0_STN_VOL_ADCL_REG 0xB8 +#define LM49453_P0_STN_VOL_ADCR_REG 0xB9 +#define LM49453_P0_STN_VOL_DMIC1L_REG 0xBA +#define LM49453_P0_STN_VOL_DMIC1R_REG 0xBB +#define LM49453_P0_STN_VOL_DMIC2L_REG 0xBC +#define LM49453_P0_STN_VOL_DMIC2R_REG 0xBD + +/* ADC/DAC CLIPPING MONITORS (Read Only/Write to Clear) */ +#define LM49453_P0_ADC_DEC_CLIP_REG 0xC2 +#define LM49453_P0_ADC_HPF_CLIP_REG 0xC3 +#define LM49453_P0_ADC_LVL_CLIP_REG 0xC4 +#define LM49453_P0_DAC_LVL_CLIP_REG 0xC5 + +/* ADC ALC EFFECT MONITORS (Read Only) */ +#define LM49453_P0_ADC_LVLMONL_REG 0xC8 +#define LM49453_P0_ADC_LVLMONR_REG 0xC9 +#define LM49453_P0_ADC_ALCMONL_REG 0xCA +#define LM49453_P0_ADC_ALCMONR_REG 0xCB +#define LM49453_P0_ADC_MUTED_REG 0xCC +#define LM49453_P0_DAC_MUTED_REG 0xCD + +/* HEADSET DETECT */ +#define LM49453_P0_HSD_PPB_LONG_CNT_LIMITL_REG 0xD0 +#define LM49453_P0_HSD_PPB_LONG_CNT_LIMITR_REG 0xD1 +#define LM49453_P0_HSD_PIN3_4_EX_LOOP_CNT_LIMITL_REG 0xD2 +#define LM49453_P0_HSD_PIN3_4_EX_LOOP_CNT_LIMITH_REG 0xD3 +#define LM49453_P0_HSD_TIMEOUT1_REG 0xD4 +#define LM49453_P0_HSD_TIMEOUT2_REG 0xD5 +#define LM49453_P0_HSD_TIMEOUT3_REG 0xD6 +#define LM49453_P0_HSD_PIN3_4_CFG_REG 0xD7 +#define LM49453_P0_HSD_IRQ1_REG 0xD8 +#define LM49453_P0_HSD_IRQ2_REG 0xD9 +#define LM49453_P0_HSD_IRQ3_REG 0xDA +#define LM49453_P0_HSD_IRQ4_REG 0xDB +#define LM49453_P0_HSD_IRQ_MASK1_REG 0xDC +#define LM49453_P0_HSD_IRQ_MASK2_REG 0xDD +#define LM49453_P0_HSD_IRQ_MASK3_REG 0xDE +#define LM49453_P0_HSD_R_HPLL_REG 0xE0 +#define LM49453_P0_HSD_R_HPLH_REG 0xE1 +#define LM49453_P0_HSD_R_HPLU_REG 0xE2 +#define LM49453_P0_HSD_R_HPRL_REG 0xE3 +#define LM49453_P0_HSD_R_HPRH_REG 0xE4 +#define LM49453_P0_HSD_R_HPRU_REG 0xE5 +#define LM49453_P0_HSD_VEL_L_FINALL_REG 0xE6 +#define LM49453_P0_HSD_VEL_L_FINALH_REG 0xE7 +#define LM49453_P0_HSD_VEL_L_FINALU_REG 0xE8 +#define LM49453_P0_HSD_RO_FINALL_REG 0xE9 +#define LM49453_P0_HSD_RO_FINALH_REG 0xEA +#define LM49453_P0_HSD_RO_FINALU_REG 0xEB +#define LM49453_P0_HSD_VMIC_BIAS_FINALL_REG 0xEC +#define LM49453_P0_HSD_VMIC_BIAS_FINALH_REG 0xED +#define LM49453_P0_HSD_VMIC_BIAS_FINALU_REG 0xEE +#define LM49453_P0_HSD_PIN_CONFIG_REG 0xEF +#define LM49453_P0_HSD_PLUG_DETECT_BB_IRQ_STATUS1_REG 0xF1 +#define LM49453_P0_HSD_PLUG_DETECT_BB_IRQ_STATUS2_REG 0xF2 +#define LM49453_P0_HSD_PLUG_DETECT_BB_IRQ_STATUS3_REG 0xF3 +#define LM49453_P0_HSD_PLUG_DETECT_BB_IRQ_STATEL_REG 0xF4 +#define LM49453_P0_HSD_PLUG_DETECT_BB_IRQ_STATEH_REG 0xF5 + +/* I/O PULLDOWN CONFIG */ +#define LM49453_P0_PULL_CONFIG1_REG 0xF8 +#define LM49453_P0_PULL_CONFIG2_REG 0xF9 +#define LM49453_P0_PULL_CONFIG3_REG 0xFA + +/* RESET */ +#define LM49453_P0_RESET_REG 0xFE + +/* PAGE */ +#define LM49453_PAGE_REG 0xFF + +#define LM49453_MAX_REGISTER (0xFF+1) + +/* LM49453_P0_PMC_SETUP_REG (0x00h) */ +#define LM49453_PMC_SETUP_CHIP_EN (BIT(1)|BIT(0)) +#define LM49453_PMC_SETUP_PLL_EN BIT(2) +#define LM49453_PMC_SETUP_PLL_P2_EN BIT(3) +#define LM49453_PMC_SETUP_PLL_FLL BIT(4) +#define LM49453_PMC_SETUP_MCLK_OVER BIT(5) +#define LM49453_PMC_SETUP_RTC_CLK_OVER BIT(6) +#define LM49453_PMC_SETUP_CHIP_ACTIVE BIT(7) + +/* Chip Enable bits */ +#define LM49453_CHIP_EN_SHUTDOWN 0x00 +#define LM49453_CHIP_EN 0x01 +#define LM49453_CHIP_EN_HSD_DETECT 0x02 +#define LM49453_CHIP_EN_INVALID_HSD 0x03 + +/* LM49453_P0_PLL_CLK_SEL1_REG (0x01h) */ +#define LM49453_CLK_SEL1_MCLK_SEL 0x11 +#define LM49453_CLK_SEL1_RTC_SEL 0x11 +#define LM49453_CLK_SEL1_PORT1_SEL 0x10 +#define LM49453_CLK_SEL1_PORT2_SEL 0x11 + +/* LM49453_P0_PLL_CLK_SEL2_REG (0x02h) */ +#define LM49453_CLK_SEL2_ADC_CLK_SEL 0x38 + +/* LM49453_P0_FLL_REF_FREQL_REG (0x0F) */ +#define LM49453_FLL_REF_FREQ_VAL 0x8ca0001 + +/* LM49453_P0_VCO_TARGETLL_REG (0x11) */ +#define LM49453_VCO_TARGET_VAL 0x8ca0001 + +/* LM49453_P0_ADC_DSP_REG (0x30h) */ +#define LM49453_ADC_DSP_ADC_MUTEL BIT(0) +#define LM49453_ADC_DSP_ADC_MUTER BIT(1) +#define LM49453_ADC_DSP_DMIC1_MUTEL BIT(2) +#define LM49453_ADC_DSP_DMIC1_MUTER BIT(3) +#define LM49453_ADC_DSP_DMIC2_MUTEL BIT(4) +#define LM49453_ADC_DSP_DMIC2_MUTER BIT(5) +#define LM49453_ADC_DSP_MUTE_ALL 0x3F + +/* LM49453_P0_DAC_DSP_REG (0x31h) */ +#define LM49453_DAC_DSP_MUTE_ALL 0xFF + +/* LM49453_P0_AUDIO_PORT1_BASIC_REG (0x60h) */ +#define LM49453_AUDIO_PORT1_BASIC_FMT_MASK (BIT(4)|BIT(3)) +#define LM49453_AUDIO_PORT1_BASIC_CLK_MS BIT(3) +#define LM49453_AUDIO_PORT1_BASIC_SYNC_MS BIT(4) + +/* LM49453_P0_RESET_REG (0xFEh) */ +#define LM49453_RESET_REG_RST BIT(0) + +/* Page select register bits (0xFF) */ +#define LM49453_PAGE0_SELECT 0x0 +#define LM49453_PAGE1_SELECT 0x1 + +/* LM49453_P0_HSD_PIN3_4_CFG_REG (Jack Pin config - 0xD7) */ +#define LM49453_JACK_DISABLE 0x00 +#define LM49453_JACK_CONFIG1 0x01 +#define LM49453_JACK_CONFIG2 0x02 +#define LM49453_JACK_CONFIG3 0x03 +#define LM49453_JACK_CONFIG4 0x04 +#define LM49453_JACK_CONFIG5 0x05 + +/* Page 1 REGISTERS */ + +/* SIDETONE */ +#define LM49453_P1_SIDETONE_SA0L_REG 0x80 +#define LM49453_P1_SIDETONE_SA0H_REG 0x81 +#define LM49453_P1_SIDETONE_SAB0U_REG 0x82 +#define LM49453_P1_SIDETONE_SB0L_REG 0x83 +#define LM49453_P1_SIDETONE_SB0H_REG 0x84 +#define LM49453_P1_SIDETONE_SH0L_REG 0x85 +#define LM49453_P1_SIDETONE_SH0H_REG 0x86 +#define LM49453_P1_SIDETONE_SH0U_REG 0x87 +#define LM49453_P1_SIDETONE_SA1L_REG 0x88 +#define LM49453_P1_SIDETONE_SA1H_REG 0x89 +#define LM49453_P1_SIDETONE_SAB1U_REG 0x8A +#define LM49453_P1_SIDETONE_SB1L_REG 0x8B +#define LM49453_P1_SIDETONE_SB1H_REG 0x8C +#define LM49453_P1_SIDETONE_SH1L_REG 0x8D +#define LM49453_P1_SIDETONE_SH1H_REG 0x8E +#define LM49453_P1_SIDETONE_SH1U_REG 0x8F +#define LM49453_P1_SIDETONE_SA2L_REG 0x90 +#define LM49453_P1_SIDETONE_SA2H_REG 0x91 +#define LM49453_P1_SIDETONE_SAB2U_REG 0x92 +#define LM49453_P1_SIDETONE_SB2L_REG 0x93 +#define LM49453_P1_SIDETONE_SB2H_REG 0x94 +#define LM49453_P1_SIDETONE_SH2L_REG 0x95 +#define LM49453_P1_SIDETONE_SH2H_REG 0x96 +#define LM49453_P1_SIDETONE_SH2U_REG 0x97 +#define LM49453_P1_SIDETONE_SA3L_REG 0x98 +#define LM49453_P1_SIDETONE_SA3H_REG 0x99 +#define LM49453_P1_SIDETONE_SAB3U_REG 0x9A +#define LM49453_P1_SIDETONE_SB3L_REG 0x9B +#define LM49453_P1_SIDETONE_SB3H_REG 0x9C +#define LM49453_P1_SIDETONE_SH3L_REG 0x9D +#define LM49453_P1_SIDETONE_SH3H_REG 0x9E +#define LM49453_P1_SIDETONE_SH3U_REG 0x9F +#define LM49453_P1_SIDETONE_SA4L_REG 0xA0 +#define LM49453_P1_SIDETONE_SA4H_REG 0xA1 +#define LM49453_P1_SIDETONE_SAB4U_REG 0xA2 +#define LM49453_P1_SIDETONE_SB4L_REG 0xA3 +#define LM49453_P1_SIDETONE_SB4H_REG 0xA4 +#define LM49453_P1_SIDETONE_SH4L_REG 0xA5 +#define LM49453_P1_SIDETONE_SH4H_REG 0xA6 +#define LM49453_P1_SIDETONE_SH4U_REG 0xA7 +#define LM49453_P1_SIDETONE_SA5L_REG 0xA8 +#define LM49453_P1_SIDETONE_SA5H_REG 0xA9 +#define LM49453_P1_SIDETONE_SAB5U_REG 0xAA +#define LM49453_P1_SIDETONE_SB5L_REG 0xAB +#define LM49453_P1_SIDETONE_SB5H_REG 0xAC +#define LM49453_P1_SIDETONE_SH5L_REG 0xAD +#define LM49453_P1_SIDETONE_SH5H_REG 0xAE +#define LM49453_P1_SIDETONE_SH5U_REG 0xAF + +/* CHARGE PUMP CONFIG */ +#define LM49453_P1_CP_CONFIG1_REG 0xB0 +#define LM49453_P1_CP_CONFIG2_REG 0xB1 +#define LM49453_P1_CP_CONFIG3_REG 0xB2 +#define LM49453_P1_CP_CONFIG4_REG 0xB3 +#define LM49453_P1_CP_LA_VTH1L_REG 0xB4 +#define LM49453_P1_CP_LA_VTH1M_REG 0xB5 +#define LM49453_P1_CP_LA_VTH2L_REG 0xB6 +#define LM49453_P1_CP_LA_VTH2M_REG 0xB7 +#define LM49453_P1_CP_LA_VTH3L_REG 0xB8 +#define LM49453_P1_CP_LA_VTH3H_REG 0xB9 +#define LM49453_P1_CP_CLK_DIV_REG 0xBA + +/* DAC */ +#define LM49453_P1_DAC_CHOP_REG 0xC0 + +#define LM49453_CLK_SRC_MCLK 1 +#endif diff --git a/sound/soc/codecs/max98095.c b/sound/soc/codecs/max98095.c index 0bb511a0388..35179e2c23c 100644 --- a/sound/soc/codecs/max98095.c +++ b/sound/soc/codecs/max98095.c @@ -24,6 +24,7 @@ #include <linux/slab.h> #include <asm/div64.h> #include <sound/max98095.h> +#include <sound/jack.h> #include "max98095.h" enum max98095_type { @@ -51,6 +52,8 @@ struct max98095_priv { u8 lin_state; unsigned int mic1pre; unsigned int mic2pre; + struct snd_soc_jack *headphone_jack; + struct snd_soc_jack *mic_jack; }; static const u8 max98095_reg_def[M98095_REG_CNT] = { @@ -2173,9 +2176,125 @@ static void max98095_handle_pdata(struct snd_soc_codec *codec) max98095_handle_bq_pdata(codec); } +static irqreturn_t max98095_report_jack(int irq, void *data) +{ + struct snd_soc_codec *codec = data; + struct max98095_priv *max98095 = snd_soc_codec_get_drvdata(codec); + unsigned int value; + int hp_report = 0; + int mic_report = 0; + + /* Read the Jack Status Register */ + value = snd_soc_read(codec, M98095_007_JACK_AUTO_STS); + + /* If ddone is not set, then detection isn't finished yet */ + if ((value & M98095_DDONE) == 0) + return IRQ_NONE; + + /* if hp, check its bit, and if set, clear it */ + if ((value & M98095_HP_IN || value & M98095_LO_IN) && + max98095->headphone_jack) + hp_report |= SND_JACK_HEADPHONE; + + /* if mic, check its bit, and if set, clear it */ + if ((value & M98095_MIC_IN) && max98095->mic_jack) + mic_report |= SND_JACK_MICROPHONE; + + if (max98095->headphone_jack == max98095->mic_jack) { + snd_soc_jack_report(max98095->headphone_jack, + hp_report | mic_report, + SND_JACK_HEADSET); + } else { + if (max98095->headphone_jack) + snd_soc_jack_report(max98095->headphone_jack, + hp_report, SND_JACK_HEADPHONE); + if (max98095->mic_jack) + snd_soc_jack_report(max98095->mic_jack, + mic_report, SND_JACK_MICROPHONE); + } + + return IRQ_HANDLED; +} + +int max98095_jack_detect_enable(struct snd_soc_codec *codec) +{ + struct max98095_priv *max98095 = snd_soc_codec_get_drvdata(codec); + int ret = 0; + int detect_enable = M98095_JDEN; + unsigned int slew = M98095_DEFAULT_SLEW_DELAY; + + if (max98095->pdata->jack_detect_pin5en) + detect_enable |= M98095_PIN5EN; + + if (max98095->pdata->jack_detect_delay) + slew = max98095->pdata->jack_detect_delay; + + ret = snd_soc_write(codec, M98095_08E_JACK_DC_SLEW, slew); + if (ret < 0) { + dev_err(codec->dev, "Failed to cfg auto detect %d\n", ret); + return ret; + } + + /* configure auto detection to be enabled */ + ret = snd_soc_write(codec, M98095_089_JACK_DET_AUTO, detect_enable); + if (ret < 0) { + dev_err(codec->dev, "Failed to cfg auto detect %d\n", ret); + return ret; + } + + return ret; +} + +int max98095_jack_detect_disable(struct snd_soc_codec *codec) +{ + int ret = 0; + + /* configure auto detection to be disabled */ + ret = snd_soc_write(codec, M98095_089_JACK_DET_AUTO, 0x0); + if (ret < 0) { + dev_err(codec->dev, "Failed to cfg auto detect %d\n", ret); + return ret; + } + + return ret; +} + +int max98095_jack_detect(struct snd_soc_codec *codec, + struct snd_soc_jack *hp_jack, struct snd_soc_jack *mic_jack) +{ + struct max98095_priv *max98095 = snd_soc_codec_get_drvdata(codec); + struct i2c_client *client = to_i2c_client(codec->dev); + int ret = 0; + + max98095->headphone_jack = hp_jack; + max98095->mic_jack = mic_jack; + + /* only progress if we have at least 1 jack pointer */ + if (!hp_jack && !mic_jack) + return -EINVAL; + + max98095_jack_detect_enable(codec); + + /* enable interrupts for headphone jack detection */ + ret = snd_soc_update_bits(codec, M98095_013_JACK_INT_EN, + M98095_IDDONE, M98095_IDDONE); + if (ret < 0) { + dev_err(codec->dev, "Failed to cfg jack irqs %d\n", ret); + return ret; + } + + max98095_report_jack(client->irq, codec); + return 0; +} + #ifdef CONFIG_PM static int max98095_suspend(struct snd_soc_codec *codec) { + struct max98095_priv *max98095 = snd_soc_codec_get_drvdata(codec); + + if (max98095->headphone_jack || max98095->mic_jack) + max98095_jack_detect_disable(codec); + max98095_set_bias_level(codec, SND_SOC_BIAS_OFF); return 0; @@ -2183,8 +2302,16 @@ static int max98095_suspend(struct snd_soc_codec *codec) static int max98095_resume(struct snd_soc_codec *codec) { + struct max98095_priv *max98095 = snd_soc_codec_get_drvdata(codec); + struct i2c_client *client = to_i2c_client(codec->dev); + max98095_set_bias_level(codec, SND_SOC_BIAS_STANDBY); + if (max98095->headphone_jack || max98095->mic_jack) { + max98095_jack_detect_enable(codec); + max98095_report_jack(client->irq, codec); + } + return 0; } #else @@ -2227,6 +2354,7 @@ static int max98095_probe(struct snd_soc_codec *codec) { struct max98095_priv *max98095 = snd_soc_codec_get_drvdata(codec); struct max98095_cdata *cdata; + struct i2c_client *client; int ret = 0; ret = snd_soc_codec_set_cache_io(codec, 8, 8, SND_SOC_I2C); @@ -2238,6 +2366,8 @@ static int max98095_probe(struct snd_soc_codec *codec) /* reset the codec, the DSP core, and disable all interrupts */ max98095_reset(codec); + client = to_i2c_client(codec->dev); + /* initialize private data */ max98095->sysclk = (unsigned)-1; @@ -2266,11 +2396,23 @@ static int max98095_probe(struct snd_soc_codec *codec) max98095->mic1pre = 0; max98095->mic2pre = 0; + if (client->irq) { + /* register an audio interrupt */ + ret = request_threaded_irq(client->irq, NULL, + max98095_report_jack, + IRQF_TRIGGER_FALLING | IRQF_TRIGGER_RISING, + "max98095", codec); + if (ret) { + dev_err(codec->dev, "Failed to request IRQ: %d\n", ret); + goto err_access; + } + } + ret = snd_soc_read(codec, M98095_0FF_REV_ID); if (ret < 0) { dev_err(codec->dev, "Failure reading hardware revision: %d\n", ret); - goto err_access; + goto err_irq; } dev_info(codec->dev, "Hardware revision: %c\n", ret - 0x40 + 'A'); @@ -2306,14 +2448,28 @@ static int max98095_probe(struct snd_soc_codec *codec) max98095_add_widgets(codec); + return 0; + +err_irq: + if (client->irq) + free_irq(client->irq, codec); err_access: return ret; } static int max98095_remove(struct snd_soc_codec *codec) { + struct max98095_priv *max98095 = snd_soc_codec_get_drvdata(codec); + struct i2c_client *client = to_i2c_client(codec->dev); + max98095_set_bias_level(codec, SND_SOC_BIAS_OFF); + if (max98095->headphone_jack || max98095->mic_jack) + max98095_jack_detect_disable(codec); + + if (client->irq) + free_irq(client->irq, codec); + return 0; } diff --git a/sound/soc/codecs/max98095.h b/sound/soc/codecs/max98095.h index 891584a0eb0..2ebbe4e894b 100644 --- a/sound/soc/codecs/max98095.h +++ b/sound/soc/codecs/max98095.h @@ -175,11 +175,23 @@ /* MAX98095 Registers Bit Fields */ +/* M98095_007_JACK_AUTO_STS */ + #define M98095_MIC_IN (1<<3) + #define M98095_LO_IN (1<<5) + #define M98095_HP_IN (1<<6) + #define M98095_DDONE (1<<7) + /* M98095_00F_HOST_CFG */ #define M98095_SEG (1<<0) #define M98095_XTEN (1<<1) #define M98095_MDLLEN (1<<2) +/* M98095_013_JACK_INT_EN */ + #define M98095_IMIC_IN (1<<3) + #define M98095_ILO_IN (1<<5) + #define M98095_IHP_IN (1<<6) + #define M98095_IDDONE (1<<7) + /* M98095_027_DAI1_CLKMODE, M98095_031_DAI2_CLKMODE, M98095_03B_DAI3_CLKMODE */ #define M98095_CLKMODE_MASK 0xFF @@ -255,6 +267,10 @@ #define M98095_EQ2EN (1<<1) #define M98095_EQ1EN (1<<0) +/* M98095_089_JACK_DET_AUTO */ + #define M98095_PIN5EN (1<<2) + #define M98095_JDEN (1<<7) + /* M98095_090_PWR_EN_IN */ #define M98095_INEN (1<<7) #define M98095_MB2EN (1<<3) @@ -296,4 +312,10 @@ #define M98095_174_DAI1_BQ_BASE 0x74 #define M98095_17E_DAI2_BQ_BASE 0x7E +/* Default Delay used in Slew Rate Calculation for Jack detection */ +#define M98095_DEFAULT_SLEW_DELAY 0x18 + +extern int max98095_jack_detect(struct snd_soc_codec *codec, + struct snd_soc_jack *hp_jack, struct snd_soc_jack *mic_jack); + #endif diff --git a/sound/soc/codecs/mc13783.c b/sound/soc/codecs/mc13783.c new file mode 100644 index 00000000000..6276e352125 --- /dev/null +++ b/sound/soc/codecs/mc13783.c @@ -0,0 +1,786 @@ +/* + * Copyright 2008 Juergen Beisert, kernel@pengutronix.de + * Copyright 2009 Sascha Hauer, s.hauer@pengutronix.de + * Copyright 2012 Philippe Retornaz, philippe.retornaz@epfl.ch + * + * Initial development of this code was funded by + * Phytec Messtechnik GmbH, http://www.phytec.de + * + * This program is free software; you can redistribute it and/or + * modify it under the terms of the GNU General Public License + * as published by the Free Software Foundation; either version 2 + * of the License, or (at your option) any later version. + * This program is distributed in the hope that it will be useful, + * but WITHOUT ANY WARRANTY; without even the implied warranty of + * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the + * GNU General Public License for more details. + * + * You should have received a copy of the GNU General Public License + * along with this program; if not, write to the Free Software + * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, + * MA 02110-1301, USA. + */ +#include <linux/module.h> +#include <linux/device.h> +#include <linux/mfd/mc13xxx.h> +#include <linux/slab.h> +#include <sound/core.h> +#include <sound/control.h> +#include <sound/pcm.h> +#include <sound/soc.h> +#include <sound/initval.h> +#include <sound/soc-dapm.h> + +#include "mc13783.h" + +#define MC13783_AUDIO_RX0 36 +#define MC13783_AUDIO_RX1 37 +#define MC13783_AUDIO_TX 38 +#define MC13783_SSI_NETWORK 39 +#define MC13783_AUDIO_CODEC 40 +#define MC13783_AUDIO_DAC 41 + +#define AUDIO_RX0_ALSPEN (1 << 5) +#define AUDIO_RX0_ALSPSEL (1 << 7) +#define AUDIO_RX0_ADDCDC (1 << 21) +#define AUDIO_RX0_ADDSTDC (1 << 22) +#define AUDIO_RX0_ADDRXIN (1 << 23) + +#define AUDIO_RX1_PGARXEN (1 << 0); +#define AUDIO_RX1_PGASTEN (1 << 5) +#define AUDIO_RX1_ARXINEN (1 << 10) + +#define AUDIO_TX_AMC1REN (1 << 5) +#define AUDIO_TX_AMC1LEN (1 << 7) +#define AUDIO_TX_AMC2EN (1 << 9) +#define AUDIO_TX_ATXINEN (1 << 11) +#define AUDIO_TX_RXINREC (1 << 13) + +#define SSI_NETWORK_CDCTXRXSLOT(x) (((x) & 0x3) << 2) +#define SSI_NETWORK_CDCTXSECSLOT(x) (((x) & 0x3) << 4) +#define SSI_NETWORK_CDCRXSECSLOT(x) (((x) & 0x3) << 6) +#define SSI_NETWORK_CDCRXSECGAIN(x) (((x) & 0x3) << 8) +#define SSI_NETWORK_CDCSUMGAIN(x) (1 << 10) +#define SSI_NETWORK_CDCFSDLY(x) (1 << 11) +#define SSI_NETWORK_DAC_SLOTS_8 (1 << 12) +#define SSI_NETWORK_DAC_SLOTS_4 (2 << 12) +#define SSI_NETWORK_DAC_SLOTS_2 (3 << 12) +#define SSI_NETWORK_DAC_SLOT_MASK (3 << 12) +#define SSI_NETWORK_DAC_RXSLOT_0_1 (0 << 14) +#define SSI_NETWORK_DAC_RXSLOT_2_3 (1 << 14) +#define SSI_NETWORK_DAC_RXSLOT_4_5 (2 << 14) +#define SSI_NETWORK_DAC_RXSLOT_6_7 (3 << 14) +#define SSI_NETWORK_DAC_RXSLOT_MASK (3 << 14) +#define SSI_NETWORK_STDCRXSECSLOT(x) (((x) & 0x3) << 16) +#define SSI_NETWORK_STDCRXSECGAIN(x) (((x) & 0x3) << 18) +#define SSI_NETWORK_STDCSUMGAIN (1 << 20) + +/* + * MC13783_AUDIO_CODEC and MC13783_AUDIO_DAC mostly share the same + * register layout + */ +#define AUDIO_SSI_SEL (1 << 0) +#define AUDIO_CLK_SEL (1 << 1) +#define AUDIO_CSM (1 << 2) +#define AUDIO_BCL_INV (1 << 3) +#define AUDIO_CFS_INV (1 << 4) +#define AUDIO_CFS(x) (((x) & 0x3) << 5) +#define AUDIO_CLK(x) (((x) & 0x7) << 7) +#define AUDIO_C_EN (1 << 11) +#define AUDIO_C_CLK_EN (1 << 12) +#define AUDIO_C_RESET (1 << 15) + +#define AUDIO_CODEC_CDCFS8K16K (1 << 10) +#define AUDIO_DAC_CFS_DLY_B (1 << 10) + +struct mc13783_priv { + struct snd_soc_codec codec; + struct mc13xxx *mc13xxx; + + enum mc13783_ssi_port adc_ssi_port; + enum mc13783_ssi_port dac_ssi_port; +}; + +static unsigned int mc13783_read(struct snd_soc_codec *codec, + unsigned int reg) +{ + struct mc13783_priv *priv = snd_soc_codec_get_drvdata(codec); + unsigned int value = 0; + + mc13xxx_lock(priv->mc13xxx); + + mc13xxx_reg_read(priv->mc13xxx, reg, &value); + + mc13xxx_unlock(priv->mc13xxx); + + return value; +} + +static int mc13783_write(struct snd_soc_codec *codec, + unsigned int reg, unsigned int value) +{ + struct mc13783_priv *priv = snd_soc_codec_get_drvdata(codec); + int ret; + + mc13xxx_lock(priv->mc13xxx); + + ret = mc13xxx_reg_write(priv->mc13xxx, reg, value); + + mc13xxx_unlock(priv->mc13xxx); + + return ret; +} + +/* Mapping between sample rates and register value */ +static unsigned int mc13783_rates[] = { + 8000, 11025, 12000, 16000, + 22050, 24000, 32000, 44100, + 48000, 64000, 96000 +}; + +static int mc13783_pcm_hw_params_dac(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params, + struct snd_soc_dai *dai) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_soc_codec *codec = rtd->codec; + unsigned int rate = params_rate(params); + int i; + + for (i = 0; i < ARRAY_SIZE(mc13783_rates); i++) { + if (rate == mc13783_rates[i]) { + snd_soc_update_bits(codec, MC13783_AUDIO_DAC, + 0xf << 17, i << 17); + return 0; + } + } + + return -EINVAL; +} + +static int mc13783_pcm_hw_params_codec(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params, + struct snd_soc_dai *dai) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_soc_codec *codec = rtd->codec; + unsigned int rate = params_rate(params); + unsigned int val; + + switch (rate) { + case 8000: + val = 0; + break; + case 16000: + val = AUDIO_CODEC_CDCFS8K16K; + break; + default: + return -EINVAL; + } + + snd_soc_update_bits(codec, MC13783_AUDIO_CODEC, AUDIO_CODEC_CDCFS8K16K, + val); + + return 0; +} + +static int mc13783_pcm_hw_params_sync(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params, + struct snd_soc_dai *dai) +{ + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) + return mc13783_pcm_hw_params_dac(substream, params, dai); + else + return mc13783_pcm_hw_params_codec(substream, params, dai); +} + +static int mc13783_set_fmt(struct snd_soc_dai *dai, unsigned int fmt, + unsigned int reg) +{ + struct snd_soc_codec *codec = dai->codec; + unsigned int val = 0; + unsigned int mask = AUDIO_CFS(3) | AUDIO_BCL_INV | AUDIO_CFS_INV | + AUDIO_CSM | AUDIO_C_CLK_EN | AUDIO_C_RESET; + + + /* DAI mode */ + switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) { + case SND_SOC_DAIFMT_I2S: + val |= AUDIO_CFS(2); + break; + case SND_SOC_DAIFMT_DSP_A: + val |= AUDIO_CFS(1); + break; + default: + return -EINVAL; + } + + /* DAI clock inversion */ + switch (fmt & SND_SOC_DAIFMT_INV_MASK) { + case SND_SOC_DAIFMT_NB_NF: + val |= AUDIO_BCL_INV; + break; + case SND_SOC_DAIFMT_NB_IF: + val |= AUDIO_BCL_INV | AUDIO_CFS_INV; + break; + case SND_SOC_DAIFMT_IB_NF: + break; + case SND_SOC_DAIFMT_IB_IF: + val |= AUDIO_CFS_INV; + break; + } + + /* DAI clock master masks */ + switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) { + case SND_SOC_DAIFMT_CBM_CFM: + val |= AUDIO_C_CLK_EN; + break; + case SND_SOC_DAIFMT_CBS_CFS: + val |= AUDIO_CSM; + break; + case SND_SOC_DAIFMT_CBM_CFS: + case SND_SOC_DAIFMT_CBS_CFM: + return -EINVAL; + } + + val |= AUDIO_C_RESET; + + snd_soc_update_bits(codec, reg, mask, val); + + return 0; +} + +static int mc13783_set_fmt_async(struct snd_soc_dai *dai, unsigned int fmt) +{ + if (dai->id == MC13783_ID_STEREO_DAC) + return mc13783_set_fmt(dai, fmt, MC13783_AUDIO_DAC); + else + return mc13783_set_fmt(dai, fmt, MC13783_AUDIO_CODEC); +} + +static int mc13783_set_fmt_sync(struct snd_soc_dai *dai, unsigned int fmt) +{ + int ret; + + ret = mc13783_set_fmt(dai, fmt, MC13783_AUDIO_DAC); + if (ret) + return ret; + + /* + * In synchronous mode force the voice codec into slave mode + * so that the clock / framesync from the stereo DAC is used + */ + fmt &= ~SND_SOC_DAIFMT_MASTER_MASK; + fmt |= SND_SOC_DAIFMT_CBS_CFS; + ret = mc13783_set_fmt(dai, fmt, MC13783_AUDIO_CODEC); + + return ret; +} + +static int mc13783_sysclk[] = { + 13000000, + 15360000, + 16800000, + -1, + 26000000, + -1, /* 12000000, invalid for voice codec */ + -1, /* 3686400, invalid for voice codec */ + 33600000, +}; + +static int mc13783_set_sysclk(struct snd_soc_dai *dai, + int clk_id, unsigned int freq, int dir, + unsigned int reg) +{ + struct snd_soc_codec *codec = dai->codec; + int clk; + unsigned int val = 0; + unsigned int mask = AUDIO_CLK(0x7) | AUDIO_CLK_SEL; + + for (clk = 0; clk < ARRAY_SIZE(mc13783_sysclk); clk++) { + if (mc13783_sysclk[clk] < 0) + continue; + if (mc13783_sysclk[clk] == freq) + break; + } + + if (clk == ARRAY_SIZE(mc13783_sysclk)) + return -EINVAL; + + if (clk_id == MC13783_CLK_CLIB) + val |= AUDIO_CLK_SEL; + + val |= AUDIO_CLK(clk); + + snd_soc_update_bits(codec, reg, mask, val); + + return 0; +} + +static int mc13783_set_sysclk_dac(struct snd_soc_dai *dai, + int clk_id, unsigned int freq, int dir) +{ + return mc13783_set_sysclk(dai, clk_id, freq, dir, MC13783_AUDIO_DAC); +} + +static int mc13783_set_sysclk_codec(struct snd_soc_dai *dai, + int clk_id, unsigned int freq, int dir) +{ + return mc13783_set_sysclk(dai, clk_id, freq, dir, MC13783_AUDIO_CODEC); +} + +static int mc13783_set_sysclk_sync(struct snd_soc_dai *dai, + int clk_id, unsigned int freq, int dir) +{ + int ret; + + ret = mc13783_set_sysclk(dai, clk_id, freq, dir, MC13783_AUDIO_DAC); + if (ret) + return ret; + + return mc13783_set_sysclk(dai, clk_id, freq, dir, MC13783_AUDIO_CODEC); +} + +static int mc13783_set_tdm_slot_dac(struct snd_soc_dai *dai, + unsigned int tx_mask, unsigned int rx_mask, int slots, + int slot_width) +{ + struct snd_soc_codec *codec = dai->codec; + unsigned int val = 0; + unsigned int mask = SSI_NETWORK_DAC_SLOT_MASK | + SSI_NETWORK_DAC_RXSLOT_MASK; + + switch (slots) { + case 2: + val |= SSI_NETWORK_DAC_SLOTS_2; + break; + case 4: + val |= SSI_NETWORK_DAC_SLOTS_4; + break; + case 8: + val |= SSI_NETWORK_DAC_SLOTS_8; + break; + default: + return -EINVAL; + } + + switch (rx_mask) { + case 0xfffffffc: + val |= SSI_NETWORK_DAC_RXSLOT_0_1; + break; + case 0xfffffff3: + val |= SSI_NETWORK_DAC_RXSLOT_2_3; + break; + case 0xffffffcf: + val |= SSI_NETWORK_DAC_RXSLOT_4_5; + break; + case 0xffffff3f: + val |= SSI_NETWORK_DAC_RXSLOT_6_7; + break; + default: + return -EINVAL; + }; + + snd_soc_update_bits(codec, MC13783_SSI_NETWORK, mask, val); + + return 0; +} + +static int mc13783_set_tdm_slot_codec(struct snd_soc_dai *dai, + unsigned int tx_mask, unsigned int rx_mask, int slots, + int slot_width) +{ + struct snd_soc_codec *codec = dai->codec; + unsigned int val = 0; + unsigned int mask = 0x3f; + + if (slots != 4) + return -EINVAL; + + if (tx_mask != 0xfffffffc) + return -EINVAL; + + val |= (0x00 << 2); /* primary timeslot RX/TX(?) is 0 */ + val |= (0x01 << 4); /* secondary timeslot TX is 1 */ + + snd_soc_update_bits(codec, MC13783_SSI_NETWORK, mask, val); + + return 0; +} + +static int mc13783_set_tdm_slot_sync(struct snd_soc_dai *dai, + unsigned int tx_mask, unsigned int rx_mask, int slots, + int slot_width) +{ + int ret; + + ret = mc13783_set_tdm_slot_dac(dai, tx_mask, rx_mask, slots, + slot_width); + if (ret) + return ret; + + ret = mc13783_set_tdm_slot_codec(dai, tx_mask, rx_mask, slots, + slot_width); + + return ret; +} + +static const struct snd_kcontrol_new mc1l_amp_ctl = + SOC_DAPM_SINGLE("Switch", 38, 7, 1, 0); + +static const struct snd_kcontrol_new mc1r_amp_ctl = + SOC_DAPM_SINGLE("Switch", 38, 5, 1, 0); + +static const struct snd_kcontrol_new mc2_amp_ctl = + SOC_DAPM_SINGLE("Switch", 38, 9, 1, 0); + +static const struct snd_kcontrol_new atx_amp_ctl = + SOC_DAPM_SINGLE("Switch", 38, 11, 1, 0); + + +/* Virtual mux. The chip does the input selection automatically + * as soon as we enable one input. */ +static const char * const adcl_enum_text[] = { + "MC1L", "RXINL", +}; + +static const struct soc_enum adcl_enum = + SOC_ENUM_SINGLE(0, 0, ARRAY_SIZE(adcl_enum_text), adcl_enum_text); + +static const struct snd_kcontrol_new left_input_mux = + SOC_DAPM_ENUM_VIRT("Route", adcl_enum); + +static const char * const adcr_enum_text[] = { + "MC1R", "MC2", "RXINR", "TXIN", +}; + +static const struct soc_enum adcr_enum = + SOC_ENUM_SINGLE(0, 0, ARRAY_SIZE(adcr_enum_text), adcr_enum_text); + +static const struct snd_kcontrol_new right_input_mux = + SOC_DAPM_ENUM_VIRT("Route", adcr_enum); + +static const struct snd_kcontrol_new samp_ctl = + SOC_DAPM_SINGLE("Switch", 36, 3, 1, 0); + +static const struct snd_kcontrol_new lamp_ctl = + SOC_DAPM_SINGLE("Switch", 36, 5, 1, 0); + +static const struct snd_kcontrol_new hlamp_ctl = + SOC_DAPM_SINGLE("Switch", 36, 10, 1, 0); + +static const struct snd_kcontrol_new hramp_ctl = + SOC_DAPM_SINGLE("Switch", 36, 9, 1, 0); + +static const struct snd_kcontrol_new llamp_ctl = + SOC_DAPM_SINGLE("Switch", 36, 16, 1, 0); + +static const struct snd_kcontrol_new lramp_ctl = + SOC_DAPM_SINGLE("Switch", 36, 15, 1, 0); + +static const struct snd_soc_dapm_widget mc13783_dapm_widgets[] = { +/* Input */ + SND_SOC_DAPM_INPUT("MC1LIN"), + SND_SOC_DAPM_INPUT("MC1RIN"), + SND_SOC_DAPM_INPUT("MC2IN"), + SND_SOC_DAPM_INPUT("RXINR"), + SND_SOC_DAPM_INPUT("RXINL"), + SND_SOC_DAPM_INPUT("TXIN"), + + SND_SOC_DAPM_SUPPLY("MC1 Bias", 38, 0, 0, NULL, 0), + SND_SOC_DAPM_SUPPLY("MC2 Bias", 38, 1, 0, NULL, 0), + + SND_SOC_DAPM_SWITCH("MC1L Amp", 38, 7, 0, &mc1l_amp_ctl), + SND_SOC_DAPM_SWITCH("MC1R Amp", 38, 5, 0, &mc1r_amp_ctl), + SND_SOC_DAPM_SWITCH("MC2 Amp", 38, 9, 0, &mc2_amp_ctl), + SND_SOC_DAPM_SWITCH("TXIN Amp", 38, 11, 0, &atx_amp_ctl), + + SND_SOC_DAPM_VIRT_MUX("PGA Left Input Mux", SND_SOC_NOPM, 0, 0, + &left_input_mux), + SND_SOC_DAPM_VIRT_MUX("PGA Right Input Mux", SND_SOC_NOPM, 0, 0, + &right_input_mux), + + SND_SOC_DAPM_PGA("PGA Left Input", SND_SOC_NOPM, 0, 0, NULL, 0), + SND_SOC_DAPM_PGA("PGA Right Input", SND_SOC_NOPM, 0, 0, NULL, 0), + + SND_SOC_DAPM_ADC("ADC", "Capture", 40, 11, 0), + SND_SOC_DAPM_SUPPLY("ADC_Reset", 40, 15, 0, NULL, 0), + +/* Output */ + SND_SOC_DAPM_SUPPLY("DAC_E", 41, 11, 0, NULL, 0), + SND_SOC_DAPM_SUPPLY("DAC_Reset", 41, 15, 0, NULL, 0), + SND_SOC_DAPM_OUTPUT("RXOUTL"), + SND_SOC_DAPM_OUTPUT("RXOUTR"), + SND_SOC_DAPM_OUTPUT("HSL"), + SND_SOC_DAPM_OUTPUT("HSR"), + SND_SOC_DAPM_OUTPUT("LSP"), + SND_SOC_DAPM_OUTPUT("SP"), + + SND_SOC_DAPM_SWITCH("Speaker Amp", 36, 3, 0, &samp_ctl), + SND_SOC_DAPM_SWITCH("Loudspeaker Amp", SND_SOC_NOPM, 0, 0, &lamp_ctl), + SND_SOC_DAPM_SWITCH("Headset Amp Left", 36, 10, 0, &hlamp_ctl), + SND_SOC_DAPM_SWITCH("Headset Amp Right", 36, 9, 0, &hramp_ctl), + SND_SOC_DAPM_SWITCH("Line out Amp Left", 36, 16, 0, &llamp_ctl), + SND_SOC_DAPM_SWITCH("Line out Amp Right", 36, 15, 0, &lramp_ctl), + SND_SOC_DAPM_DAC("DAC", "Playback", 36, 22, 0), + SND_SOC_DAPM_PGA("DAC PGA", 37, 5, 0, NULL, 0), +}; + +static struct snd_soc_dapm_route mc13783_routes[] = { +/* Input */ + { "MC1L Amp", NULL, "MC1LIN"}, + { "MC1R Amp", NULL, "MC1RIN" }, + { "MC2 Amp", NULL, "MC2IN" }, + { "TXIN Amp", NULL, "TXIN"}, + + { "PGA Left Input Mux", "MC1L", "MC1L Amp" }, + { "PGA Left Input Mux", "RXINL", "RXINL"}, + { "PGA Right Input Mux", "MC1R", "MC1R Amp" }, + { "PGA Right Input Mux", "MC2", "MC2 Amp"}, + { "PGA Right Input Mux", "TXIN", "TXIN Amp"}, + { "PGA Right Input Mux", "RXINR", "RXINR"}, + + { "PGA Left Input", NULL, "PGA Left Input Mux"}, + { "PGA Right Input", NULL, "PGA Right Input Mux"}, + + { "ADC", NULL, "PGA Left Input"}, + { "ADC", NULL, "PGA Right Input"}, + { "ADC", NULL, "ADC_Reset"}, + +/* Output */ + { "HSL", NULL, "Headset Amp Left" }, + { "HSR", NULL, "Headset Amp Right"}, + { "RXOUTL", NULL, "Line out Amp Left"}, + { "RXOUTR", NULL, "Line out Amp Right"}, + { "SP", NULL, "Speaker Amp"}, + { "Speaker Amp", NULL, "DAC PGA"}, + { "LSP", NULL, "DAC PGA"}, + { "Headset Amp Left", NULL, "DAC PGA"}, + { "Headset Amp Right", NULL, "DAC PGA"}, + { "Line out Amp Left", NULL, "DAC PGA"}, + { "Line out Amp Right", NULL, "DAC PGA"}, + { "DAC PGA", NULL, "DAC"}, + { "DAC", NULL, "DAC_E"}, +}; + +static const char * const mc13783_3d_mixer[] = {"Stereo", "Phase Mix", + "Mono", "Mono Mix"}; + +static const struct soc_enum mc13783_enum_3d_mixer = + SOC_ENUM_SINGLE(MC13783_AUDIO_RX1, 16, ARRAY_SIZE(mc13783_3d_mixer), + mc13783_3d_mixer); + +static struct snd_kcontrol_new mc13783_control_list[] = { + SOC_SINGLE("Loudspeaker enable", MC13783_AUDIO_RX0, 5, 1, 0), + SOC_SINGLE("PCM Playback Volume", MC13783_AUDIO_RX1, 6, 15, 0), + SOC_DOUBLE("PCM Capture Volume", MC13783_AUDIO_TX, 19, 14, 31, 0), + SOC_ENUM("3D Control", mc13783_enum_3d_mixer), +}; + +static int mc13783_probe(struct snd_soc_codec *codec) +{ + struct mc13783_priv *priv = snd_soc_codec_get_drvdata(codec); + + mc13xxx_lock(priv->mc13xxx); + + /* these are the reset values */ + mc13xxx_reg_write(priv->mc13xxx, MC13783_AUDIO_RX0, 0x25893); + mc13xxx_reg_write(priv->mc13xxx, MC13783_AUDIO_RX1, 0x00d35A); + mc13xxx_reg_write(priv->mc13xxx, MC13783_AUDIO_TX, 0x420000); + mc13xxx_reg_write(priv->mc13xxx, MC13783_SSI_NETWORK, 0x013060); + mc13xxx_reg_write(priv->mc13xxx, MC13783_AUDIO_CODEC, 0x180027); + mc13xxx_reg_write(priv->mc13xxx, MC13783_AUDIO_DAC, 0x0e0004); + + if (priv->adc_ssi_port == MC13783_SSI1_PORT) + mc13xxx_reg_rmw(priv->mc13xxx, MC13783_AUDIO_CODEC, + AUDIO_SSI_SEL, 0); + else + mc13xxx_reg_rmw(priv->mc13xxx, MC13783_AUDIO_CODEC, + 0, AUDIO_SSI_SEL); + + if (priv->dac_ssi_port == MC13783_SSI1_PORT) + mc13xxx_reg_rmw(priv->mc13xxx, MC13783_AUDIO_DAC, + AUDIO_SSI_SEL, 0); + else + mc13xxx_reg_rmw(priv->mc13xxx, MC13783_AUDIO_DAC, + 0, AUDIO_SSI_SEL); + + mc13xxx_unlock(priv->mc13xxx); + + return 0; +} + +static int mc13783_remove(struct snd_soc_codec *codec) +{ + struct mc13783_priv *priv = snd_soc_codec_get_drvdata(codec); + + mc13xxx_lock(priv->mc13xxx); + + /* Make sure VAUDIOON is off */ + mc13xxx_reg_rmw(priv->mc13xxx, MC13783_AUDIO_RX0, 0x3, 0); + + mc13xxx_unlock(priv->mc13xxx); + + return 0; +} + +#define MC13783_RATES_RECORD (SNDRV_PCM_RATE_8000 | SNDRV_PCM_RATE_16000) + +#define MC13783_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S20_3LE |\ + SNDRV_PCM_FMTBIT_S24_LE) + +static struct snd_soc_dai_ops mc13783_ops_dac = { + .hw_params = mc13783_pcm_hw_params_dac, + .set_fmt = mc13783_set_fmt_async, + .set_sysclk = mc13783_set_sysclk_dac, + .set_tdm_slot = mc13783_set_tdm_slot_dac, +}; + +static struct snd_soc_dai_ops mc13783_ops_codec = { + .hw_params = mc13783_pcm_hw_params_codec, + .set_fmt = mc13783_set_fmt_async, + .set_sysclk = mc13783_set_sysclk_codec, + .set_tdm_slot = mc13783_set_tdm_slot_codec, +}; + +/* + * The mc13783 has two SSI ports, both of them can be routed either + * to the voice codec or the stereo DAC. When two different SSI ports + * are used for the voice codec and the stereo DAC we can do different + * formats and sysclock settings for playback and capture + * (mc13783-hifi-playback and mc13783-hifi-capture). Using the same port + * forces us to use symmetric rates (mc13783-hifi). + */ +static struct snd_soc_dai_driver mc13783_dai_async[] = { + { + .name = "mc13783-hifi-playback", + .id = MC13783_ID_STEREO_DAC, + .playback = { + .stream_name = "Playback", + .channels_min = 1, + .channels_max = 2, + .rates = SNDRV_PCM_RATE_8000_96000, + .formats = MC13783_FORMATS, + }, + .ops = &mc13783_ops_dac, + }, { + .name = "mc13783-hifi-capture", + .id = MC13783_ID_STEREO_CODEC, + .capture = { + .stream_name = "Capture", + .channels_min = 1, + .channels_max = 2, + .rates = MC13783_RATES_RECORD, + .formats = MC13783_FORMATS, + }, + .ops = &mc13783_ops_codec, + }, +}; + +static struct snd_soc_dai_ops mc13783_ops_sync = { + .hw_params = mc13783_pcm_hw_params_sync, + .set_fmt = mc13783_set_fmt_sync, + .set_sysclk = mc13783_set_sysclk_sync, + .set_tdm_slot = mc13783_set_tdm_slot_sync, +}; + +static struct snd_soc_dai_driver mc13783_dai_sync[] = { + { + .name = "mc13783-hifi", + .id = MC13783_ID_SYNC, + .playback = { + .stream_name = "Playback", + .channels_min = 1, + .channels_max = 2, + .rates = SNDRV_PCM_RATE_8000_96000, + .formats = MC13783_FORMATS, + }, + .capture = { + .stream_name = "Capture", + .channels_min = 1, + .channels_max = 2, + .rates = MC13783_RATES_RECORD, + .formats = MC13783_FORMATS, + }, + .ops = &mc13783_ops_sync, + .symmetric_rates = 1, + } +}; + +static struct snd_soc_codec_driver soc_codec_dev_mc13783 = { + .probe = mc13783_probe, + .remove = mc13783_remove, + .read = mc13783_read, + .write = mc13783_write, + .controls = mc13783_control_list, + .num_controls = ARRAY_SIZE(mc13783_control_list), + .dapm_widgets = mc13783_dapm_widgets, + .num_dapm_widgets = ARRAY_SIZE(mc13783_dapm_widgets), + .dapm_routes = mc13783_routes, + .num_dapm_routes = ARRAY_SIZE(mc13783_routes), +}; + +static int mc13783_codec_probe(struct platform_device *pdev) +{ + struct mc13xxx *mc13xxx; + struct mc13783_priv *priv; + struct mc13xxx_codec_platform_data *pdata = pdev->dev.platform_data; + int ret; + + mc13xxx = dev_get_drvdata(pdev->dev.parent); + + + priv = devm_kzalloc(&pdev->dev, sizeof(*priv), GFP_KERNEL); + if (priv == NULL) + return -ENOMEM; + + dev_set_drvdata(&pdev->dev, priv); + priv->mc13xxx = mc13xxx; + if (pdata) { + priv->adc_ssi_port = pdata->adc_ssi_port; + priv->dac_ssi_port = pdata->dac_ssi_port; + } else { + priv->adc_ssi_port = MC13783_SSI1_PORT; + priv->dac_ssi_port = MC13783_SSI2_PORT; + } + + if (priv->adc_ssi_port == priv->dac_ssi_port) + ret = snd_soc_register_codec(&pdev->dev, &soc_codec_dev_mc13783, + mc13783_dai_sync, ARRAY_SIZE(mc13783_dai_sync)); + else + ret = snd_soc_register_codec(&pdev->dev, &soc_codec_dev_mc13783, + mc13783_dai_async, ARRAY_SIZE(mc13783_dai_async)); + + if (ret) + goto err_register_codec; + + return 0; + +err_register_codec: + dev_err(&pdev->dev, "register codec failed with %d\n", ret); + + return ret; +} + +static int mc13783_codec_remove(struct platform_device *pdev) +{ + snd_soc_unregister_codec(&pdev->dev); + + return 0; +} + +static struct platform_driver mc13783_codec_driver = { + .driver = { + .name = "mc13783-codec", + .owner = THIS_MODULE, + }, + .probe = mc13783_codec_probe, + .remove = __devexit_p(mc13783_codec_remove), +}; + +module_platform_driver(mc13783_codec_driver); + +MODULE_DESCRIPTION("ASoC MC13783 driver"); +MODULE_AUTHOR("Sascha Hauer, Pengutronix <s.hauer@pengutronix.de>"); +MODULE_AUTHOR("Philippe Retornaz <philippe.retornaz@epfl.ch>"); +MODULE_LICENSE("GPL"); diff --git a/sound/soc/codecs/mc13783.h b/sound/soc/codecs/mc13783.h new file mode 100644 index 00000000000..3a6d1993a21 --- /dev/null +++ b/sound/soc/codecs/mc13783.h @@ -0,0 +1,28 @@ +/* + * Copyright 2008 Juergen Beisert, kernel@pengutronix.de + * + * This program is free software; you can redistribute it and/or + * modify it under the terms of the GNU General Public License + * as published by the Free Software Foundation; either version 2 + * of the License, or (at your option) any later version. + * This program is distributed in the hope that it will be useful, + * but WITHOUT ANY WARRANTY; without even the implied warranty of + * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the + * GNU General Public License for more details. + * + * You should have received a copy of the GNU General Public License + * along with this program; if not, write to the Free Software Foundation, Inc. + * 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301, USA. + */ + +#ifndef MC13783_MIXER_H +#define MC13783_MIXER_H + +#define MC13783_CLK_CLIA 1 +#define MC13783_CLK_CLIB 2 + +#define MC13783_ID_STEREO_DAC 1 +#define MC13783_ID_STEREO_CODEC 2 +#define MC13783_ID_SYNC 3 + +#endif /* MC13783_MIXER_H */ diff --git a/sound/soc/codecs/ml26124.c b/sound/soc/codecs/ml26124.c new file mode 100644 index 00000000000..22cb5bf5927 --- /dev/null +++ b/sound/soc/codecs/ml26124.c @@ -0,0 +1,681 @@ +/* + * Copyright (C) 2011 LAPIS Semiconductor Co., Ltd. + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License as published by + * the Free Software Foundation; version 2 of the License. + * + * This program is distributed in the hope that it will be useful, + * but WITHOUT ANY WARRANTY; without even the implied warranty of + * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the + * GNU General Public License for more details. + * + * You should have received a copy of the GNU General Public License + * along with this program; if not, write to the Free Software + * Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307, USA. + */ + +#include <linux/module.h> +#include <linux/moduleparam.h> +#include <linux/init.h> +#include <linux/delay.h> +#include <linux/pm.h> +#include <linux/i2c.h> +#include <linux/slab.h> +#include <linux/platform_device.h> +#include <linux/regmap.h> +#include <sound/core.h> +#include <sound/pcm.h> +#include <sound/pcm_params.h> +#include <sound/soc.h> +#include <sound/tlv.h> +#include "ml26124.h" + +#define DVOL_CTL_DVMUTE_ON BIT(4) /* Digital volume MUTE On */ +#define DVOL_CTL_DVMUTE_OFF 0 /* Digital volume MUTE Off */ +#define ML26124_SAI_NO_DELAY BIT(1) +#define ML26124_SAI_FRAME_SYNC (BIT(5) | BIT(0)) /* For mono (Telecodec) */ +#define ML26134_CACHESIZE 212 +#define ML26124_VMID BIT(1) +#define ML26124_RATES (SNDRV_PCM_RATE_16000 | SNDRV_PCM_RATE_32000 |\ + SNDRV_PCM_RATE_48000) +#define ML26124_FORMATS (SNDRV_PCM_FMTBIT_U8 | SNDRV_PCM_FMTBIT_S16_LE |\ + SNDRV_PCM_FMTBIT_S32_LE) +#define ML26124_NUM_REGISTER ML26134_CACHESIZE + +struct ml26124_priv { + u32 mclk; + u32 rate; + struct regmap *regmap; + int clk_in; + struct snd_pcm_substream *substream; +}; + +struct clk_coeff { + u32 mclk; + u32 rate; + u8 pllnl; + u8 pllnh; + u8 pllml; + u8 pllmh; + u8 plldiv; +}; + +/* ML26124 configuration */ +static const DECLARE_TLV_DB_SCALE(digital_tlv, -7150, 50, 0); + +static const DECLARE_TLV_DB_SCALE(alclvl, -2250, 150, 0); +static const DECLARE_TLV_DB_SCALE(mingain, -1200, 600, 0); +static const DECLARE_TLV_DB_SCALE(maxgain, -675, 600, 0); +static const DECLARE_TLV_DB_SCALE(boost_vol, -1200, 75, 0); +static const DECLARE_TLV_DB_SCALE(ngth, -7650, 150, 0); + +static const char * const ml26124_companding[] = {"16bit PCM", "u-law", + "A-law"}; + +static const struct soc_enum ml26124_adc_companding_enum + = SOC_ENUM_SINGLE(ML26124_SAI_TRANS_CTL, 6, 3, ml26124_companding); + +static const struct soc_enum ml26124_dac_companding_enum + = SOC_ENUM_SINGLE(ML26124_SAI_RCV_CTL, 6, 3, ml26124_companding); + +static const struct snd_kcontrol_new ml26124_snd_controls[] = { + SOC_SINGLE_TLV("Capture Digital Volume", ML26124_RECORD_DIG_VOL, 0, + 0xff, 1, digital_tlv), + SOC_SINGLE_TLV("Playback Digital Volume", ML26124_PLBAK_DIG_VOL, 0, + 0xff, 1, digital_tlv), + SOC_SINGLE_TLV("Digital Boost Volume", ML26124_DIGI_BOOST_VOL, 0, + 0x3f, 0, boost_vol), + SOC_SINGLE_TLV("EQ Band0 Volume", ML26124_EQ_GAIN_BRAND0, 0, + 0xff, 1, digital_tlv), + SOC_SINGLE_TLV("EQ Band1 Volume", ML26124_EQ_GAIN_BRAND1, 0, + 0xff, 1, digital_tlv), + SOC_SINGLE_TLV("EQ Band2 Volume", ML26124_EQ_GAIN_BRAND2, 0, + 0xff, 1, digital_tlv), + SOC_SINGLE_TLV("EQ Band3 Volume", ML26124_EQ_GAIN_BRAND3, 0, + 0xff, 1, digital_tlv), + SOC_SINGLE_TLV("EQ Band4 Volume", ML26124_EQ_GAIN_BRAND4, 0, + 0xff, 1, digital_tlv), + SOC_SINGLE_TLV("ALC Target Level", ML26124_ALC_TARGET_LEV, 0, + 0xf, 1, alclvl), + SOC_SINGLE_TLV("ALC Min Input Volume", ML26124_ALC_MAXMIN_GAIN, 0, + 7, 0, mingain), + SOC_SINGLE_TLV("ALC Max Input Volume", ML26124_ALC_MAXMIN_GAIN, 4, + 7, 1, maxgain), + SOC_SINGLE_TLV("Playback Limiter Min Input Volume", + ML26124_PL_MAXMIN_GAIN, 0, 7, 0, mingain), + SOC_SINGLE_TLV("Playback Limiter Max Input Volume", + ML26124_PL_MAXMIN_GAIN, 4, 7, 1, maxgain), + SOC_SINGLE_TLV("Playback Boost Volume", ML26124_PLYBAK_BOST_VOL, 0, + 0x3f, 0, boost_vol), + SOC_SINGLE("DC High Pass Filter Switch", ML26124_FILTER_EN, 0, 1, 0), + SOC_SINGLE("Noise High Pass Filter Switch", ML26124_FILTER_EN, 1, 1, 0), + SOC_SINGLE("ZC Switch", ML26124_PW_ZCCMP_PW_MNG, 1, + 1, 0), + SOC_SINGLE("EQ Band0 Switch", ML26124_FILTER_EN, 2, 1, 0), + SOC_SINGLE("EQ Band1 Switch", ML26124_FILTER_EN, 3, 1, 0), + SOC_SINGLE("EQ Band2 Switch", ML26124_FILTER_EN, 4, 1, 0), + SOC_SINGLE("EQ Band3 Switch", ML26124_FILTER_EN, 5, 1, 0), + SOC_SINGLE("EQ Band4 Switch", ML26124_FILTER_EN, 6, 1, 0), + SOC_SINGLE("Play Limiter", ML26124_DVOL_CTL, 0, 1, 0), + SOC_SINGLE("Capture Limiter", ML26124_DVOL_CTL, 1, 1, 0), + SOC_SINGLE("Digital Volume Fade Switch", ML26124_DVOL_CTL, 3, 1, 0), + SOC_SINGLE("Digital Switch", ML26124_DVOL_CTL, 4, 1, 0), + SOC_ENUM("DAC Companding", ml26124_dac_companding_enum), + SOC_ENUM("ADC Companding", ml26124_adc_companding_enum), +}; + +static const struct snd_kcontrol_new ml26124_output_mixer_controls[] = { + SOC_DAPM_SINGLE("DAC Switch", ML26124_SPK_AMP_OUT, 1, 1, 0), + SOC_DAPM_SINGLE("Line in loopback Switch", ML26124_SPK_AMP_OUT, 3, 1, + 0), + SOC_DAPM_SINGLE("PGA Switch", ML26124_SPK_AMP_OUT, 5, 1, 0), +}; + +/* Input mux */ +static const char * const ml26124_input_select[] = {"Analog MIC SingleEnded in", + "Digital MIC in", "Analog MIC Differential in"}; + +static const struct soc_enum ml26124_insel_enum = + SOC_ENUM_SINGLE(ML26124_MIC_IF_CTL, 0, 3, ml26124_input_select); + +static const struct snd_kcontrol_new ml26124_input_mux_controls = + SOC_DAPM_ENUM("Input Select", ml26124_insel_enum); + +static const struct snd_kcontrol_new ml26124_line_control = + SOC_DAPM_SINGLE("Switch", ML26124_PW_LOUT_PW_MNG, 1, 1, 0); + +static const struct snd_soc_dapm_widget ml26124_dapm_widgets[] = { + SND_SOC_DAPM_SUPPLY("MCLKEN", ML26124_CLK_EN, 0, 0, NULL, 0), + SND_SOC_DAPM_SUPPLY("PLLEN", ML26124_CLK_EN, 1, 0, NULL, 0), + SND_SOC_DAPM_SUPPLY("PLLOE", ML26124_CLK_EN, 2, 0, NULL, 0), + SND_SOC_DAPM_SUPPLY("MICBIAS", ML26124_PW_REF_PW_MNG, 2, 0, NULL, 0), + SND_SOC_DAPM_MIXER("Output Mixer", SND_SOC_NOPM, 0, 0, + &ml26124_output_mixer_controls[0], + ARRAY_SIZE(ml26124_output_mixer_controls)), + SND_SOC_DAPM_DAC("DAC", "Playback", ML26124_PW_DAC_PW_MNG, 1, 0), + SND_SOC_DAPM_ADC("ADC", "Capture", ML26124_PW_IN_PW_MNG, 1, 0), + SND_SOC_DAPM_PGA("PGA", ML26124_PW_IN_PW_MNG, 3, 0, NULL, 0), + SND_SOC_DAPM_MUX("Input Mux", SND_SOC_NOPM, 0, 0, + &ml26124_input_mux_controls), + SND_SOC_DAPM_SWITCH("Line Out Enable", SND_SOC_NOPM, 0, 0, + &ml26124_line_control), + SND_SOC_DAPM_INPUT("MDIN"), + SND_SOC_DAPM_INPUT("MIN"), + SND_SOC_DAPM_INPUT("LIN"), + SND_SOC_DAPM_OUTPUT("SPOUT"), + SND_SOC_DAPM_OUTPUT("LOUT"), +}; + +static const struct snd_soc_dapm_route ml26124_intercon[] = { + /* Supply */ + {"DAC", NULL, "MCLKEN"}, + {"ADC", NULL, "MCLKEN"}, + {"DAC", NULL, "PLLEN"}, + {"ADC", NULL, "PLLEN"}, + {"DAC", NULL, "PLLOE"}, + {"ADC", NULL, "PLLOE"}, + + /* output mixer */ + {"Output Mixer", "DAC Switch", "DAC"}, + {"Output Mixer", "Line in loopback Switch", "LIN"}, + + /* outputs */ + {"LOUT", NULL, "Output Mixer"}, + {"SPOUT", NULL, "Output Mixer"}, + {"Line Out Enable", NULL, "LOUT"}, + + /* input */ + {"ADC", NULL, "Input Mux"}, + {"Input Mux", "Analog MIC SingleEnded in", "PGA"}, + {"Input Mux", "Analog MIC Differential in", "PGA"}, + {"PGA", NULL, "MIN"}, +}; + +/* PLLOutputFreq(Hz) = InputMclkFreq(Hz) * PLLM / (PLLN * PLLDIV) */ +static const struct clk_coeff coeff_div[] = { + {12288000, 16000, 0xc, 0x0, 0x20, 0x0, 0x4}, + {12288000, 32000, 0xc, 0x0, 0x20, 0x0, 0x4}, + {12288000, 48000, 0xc, 0x0, 0x30, 0x0, 0x4}, +}; + +static struct reg_default ml26124_reg[] = { + /* CLOCK control Register */ + {0x00, 0x00 }, /* Sampling Rate */ + {0x02, 0x00}, /* PLL NL */ + {0x04, 0x00}, /* PLLNH */ + {0x06, 0x00}, /* PLLML */ + {0x08, 0x00}, /* MLLMH */ + {0x0a, 0x00}, /* PLLDIV */ + {0x0c, 0x00}, /* Clock Enable */ + {0x0e, 0x00}, /* CLK Input/Output Control */ + + /* System Control Register */ + {0x10, 0x00}, /* Software RESET */ + {0x12, 0x00}, /* Record/Playback Run */ + {0x14, 0x00}, /* Mic Input/Output control */ + + /* Power Management Register */ + {0x20, 0x00}, /* Reference Power Management */ + {0x22, 0x00}, /* Input Power Management */ + {0x24, 0x00}, /* DAC Power Management */ + {0x26, 0x00}, /* SP-AMP Power Management */ + {0x28, 0x00}, /* LINEOUT Power Management */ + {0x2a, 0x00}, /* VIDEO Power Management */ + {0x2e, 0x00}, /* AC-CMP Power Management */ + + /* Analog reference Control Register */ + {0x30, 0x04}, /* MICBIAS Voltage Control */ + + /* Input/Output Amplifier Control Register */ + {0x32, 0x10}, /* MIC Input Volume */ + {0x38, 0x00}, /* Mic Boost Volume */ + {0x3a, 0x33}, /* Speaker AMP Volume */ + {0x48, 0x00}, /* AMP Volume Control Function Enable */ + {0x4a, 0x00}, /* Amplifier Volume Fader Control */ + + /* Analog Path Control Register */ + {0x54, 0x00}, /* Speaker AMP Output Control */ + {0x5a, 0x00}, /* Mic IF Control */ + {0xe8, 0x01}, /* Mic Select Control */ + + /* Audio Interface Control Register */ + {0x60, 0x00}, /* SAI-Trans Control */ + {0x62, 0x00}, /* SAI-Receive Control */ + {0x64, 0x00}, /* SAI Mode select */ + + /* DSP Control Register */ + {0x66, 0x01}, /* Filter Func Enable */ + {0x68, 0x00}, /* Volume Control Func Enable */ + {0x6A, 0x00}, /* Mixer & Volume Control*/ + {0x6C, 0xff}, /* Record Digital Volume */ + {0x70, 0xff}, /* Playback Digital Volume */ + {0x72, 0x10}, /* Digital Boost Volume */ + {0x74, 0xe7}, /* EQ gain Band0 */ + {0x76, 0xe7}, /* EQ gain Band1 */ + {0x78, 0xe7}, /* EQ gain Band2 */ + {0x7A, 0xe7}, /* EQ gain Band3 */ + {0x7C, 0xe7}, /* EQ gain Band4 */ + {0x7E, 0x00}, /* HPF2 CutOff*/ + {0x80, 0x00}, /* EQ Band0 Coef0L */ + {0x82, 0x00}, /* EQ Band0 Coef0H */ + {0x84, 0x00}, /* EQ Band0 Coef0L */ + {0x86, 0x00}, /* EQ Band0 Coef0H */ + {0x88, 0x00}, /* EQ Band1 Coef0L */ + {0x8A, 0x00}, /* EQ Band1 Coef0H */ + {0x8C, 0x00}, /* EQ Band1 Coef0L */ + {0x8E, 0x00}, /* EQ Band1 Coef0H */ + {0x90, 0x00}, /* EQ Band2 Coef0L */ + {0x92, 0x00}, /* EQ Band2 Coef0H */ + {0x94, 0x00}, /* EQ Band2 Coef0L */ + {0x96, 0x00}, /* EQ Band2 Coef0H */ + {0x98, 0x00}, /* EQ Band3 Coef0L */ + {0x9A, 0x00}, /* EQ Band3 Coef0H */ + {0x9C, 0x00}, /* EQ Band3 Coef0L */ + {0x9E, 0x00}, /* EQ Band3 Coef0H */ + {0xA0, 0x00}, /* EQ Band4 Coef0L */ + {0xA2, 0x00}, /* EQ Band4 Coef0H */ + {0xA4, 0x00}, /* EQ Band4 Coef0L */ + {0xA6, 0x00}, /* EQ Band4 Coef0H */ + + /* ALC Control Register */ + {0xb0, 0x00}, /* ALC Mode */ + {0xb2, 0x02}, /* ALC Attack Time */ + {0xb4, 0x03}, /* ALC Decay Time */ + {0xb6, 0x00}, /* ALC Hold Time */ + {0xb8, 0x0b}, /* ALC Target Level */ + {0xba, 0x70}, /* ALC Max/Min Gain */ + {0xbc, 0x00}, /* Noise Gate Threshold */ + {0xbe, 0x00}, /* ALC ZeroCross TimeOut */ + + /* Playback Limiter Control Register */ + {0xc0, 0x04}, /* PL Attack Time */ + {0xc2, 0x05}, /* PL Decay Time */ + {0xc4, 0x0d}, /* PL Target Level */ + {0xc6, 0x70}, /* PL Max/Min Gain */ + {0xc8, 0x10}, /* Playback Boost Volume */ + {0xca, 0x00}, /* PL ZeroCross TimeOut */ + + /* Video Amplifier Control Register */ + {0xd0, 0x01}, /* VIDEO AMP Gain Control */ + {0xd2, 0x01}, /* VIDEO AMP Setup 1 */ + {0xd4, 0x01}, /* VIDEO AMP Control2 */ +}; + +/* Get sampling rate value of sampling rate setting register (0x0) */ +static inline int get_srate(int rate) +{ + int srate; + + switch (rate) { + case 16000: + srate = 3; + break; + case 32000: + srate = 6; + break; + case 48000: + srate = 8; + break; + default: + return -EINVAL; + } + return srate; +} + +static inline int get_coeff(int mclk, int rate) +{ + int i; + + for (i = 0; i < ARRAY_SIZE(coeff_div); i++) { + if (coeff_div[i].rate == rate && coeff_div[i].mclk == mclk) + return i; + } + return -EINVAL; +} + +static int ml26124_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *hw_params, + struct snd_soc_dai *dai) +{ + struct snd_soc_codec *codec = dai->codec; + struct ml26124_priv *priv = snd_soc_codec_get_drvdata(codec); + int i = get_coeff(priv->mclk, params_rate(hw_params)); + + priv->substream = substream; + priv->rate = params_rate(hw_params); + + if (priv->clk_in) { + switch (priv->mclk / params_rate(hw_params)) { + case 256: + snd_soc_update_bits(codec, ML26124_CLK_CTL, + BIT(0) | BIT(1), 1); + break; + case 512: + snd_soc_update_bits(codec, ML26124_CLK_CTL, + BIT(0) | BIT(1), 2); + break; + case 1024: + snd_soc_update_bits(codec, ML26124_CLK_CTL, + BIT(0) | BIT(1), 3); + break; + default: + dev_err(codec->dev, "Unsupported MCLKI\n"); + break; + } + } else { + snd_soc_update_bits(codec, ML26124_CLK_CTL, + BIT(0) | BIT(1), 0); + } + + switch (params_rate(hw_params)) { + case 16000: + snd_soc_update_bits(codec, ML26124_SMPLING_RATE, 0xf, + get_srate(params_rate(hw_params))); + snd_soc_update_bits(codec, ML26124_PLLNL, 0xff, + coeff_div[i].pllnl); + snd_soc_update_bits(codec, ML26124_PLLNH, 0x1, + coeff_div[i].pllnh); + snd_soc_update_bits(codec, ML26124_PLLML, 0xff, + coeff_div[i].pllml); + snd_soc_update_bits(codec, ML26124_PLLMH, 0x3f, + coeff_div[i].pllmh); + snd_soc_update_bits(codec, ML26124_PLLDIV, 0x1f, + coeff_div[i].plldiv); + break; + case 32000: + snd_soc_update_bits(codec, ML26124_SMPLING_RATE, 0xf, + get_srate(params_rate(hw_params))); + snd_soc_update_bits(codec, ML26124_PLLNL, 0xff, + coeff_div[i].pllnl); + snd_soc_update_bits(codec, ML26124_PLLNH, 0x1, + coeff_div[i].pllnh); + snd_soc_update_bits(codec, ML26124_PLLML, 0xff, + coeff_div[i].pllml); + snd_soc_update_bits(codec, ML26124_PLLMH, 0x3f, + coeff_div[i].pllmh); + snd_soc_update_bits(codec, ML26124_PLLDIV, 0x1f, + coeff_div[i].plldiv); + break; + case 48000: + snd_soc_update_bits(codec, ML26124_SMPLING_RATE, 0xf, + get_srate(params_rate(hw_params))); + snd_soc_update_bits(codec, ML26124_PLLNL, 0xff, + coeff_div[i].pllnl); + snd_soc_update_bits(codec, ML26124_PLLNH, 0x1, + coeff_div[i].pllnh); + snd_soc_update_bits(codec, ML26124_PLLML, 0xff, + coeff_div[i].pllml); + snd_soc_update_bits(codec, ML26124_PLLMH, 0x3f, + coeff_div[i].pllmh); + snd_soc_update_bits(codec, ML26124_PLLDIV, 0x1f, + coeff_div[i].plldiv); + break; + default: + pr_err("%s:this rate is no support for ml26124\n", __func__); + return -EINVAL; + } + + return 0; +} + +static int ml26124_mute(struct snd_soc_dai *dai, int mute) +{ + struct snd_soc_codec *codec = dai->codec; + struct ml26124_priv *priv = snd_soc_codec_get_drvdata(codec); + + switch (priv->substream->stream) { + case SNDRV_PCM_STREAM_CAPTURE: + snd_soc_update_bits(codec, ML26124_REC_PLYBAK_RUN, BIT(0), 1); + break; + case SNDRV_PCM_STREAM_PLAYBACK: + snd_soc_update_bits(codec, ML26124_REC_PLYBAK_RUN, BIT(1), 2); + break; + } + + if (mute) + snd_soc_update_bits(codec, ML26124_DVOL_CTL, BIT(4), + DVOL_CTL_DVMUTE_ON); + else + snd_soc_update_bits(codec, ML26124_DVOL_CTL, BIT(4), + DVOL_CTL_DVMUTE_OFF); + + return 0; +} + +static int ml26124_set_dai_fmt(struct snd_soc_dai *codec_dai, + unsigned int fmt) +{ + unsigned char mode; + struct snd_soc_codec *codec = codec_dai->codec; + + /* set master/slave audio interface */ + switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) { + case SND_SOC_DAIFMT_CBM_CFM: + mode = 1; + break; + case SND_SOC_DAIFMT_CBS_CFS: + mode = 0; + break; + default: + return -EINVAL; + } + snd_soc_update_bits(codec, ML26124_SAI_MODE_SEL, BIT(0), mode); + + /* interface format */ + switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) { + case SND_SOC_DAIFMT_I2S: + break; + default: + return -EINVAL; + } + + /* clock inversion */ + switch (fmt & SND_SOC_DAIFMT_INV_MASK) { + case SND_SOC_DAIFMT_NB_NF: + break; + default: + return -EINVAL; + } + + return 0; +} + +static int ml26124_set_dai_sysclk(struct snd_soc_dai *codec_dai, + int clk_id, unsigned int freq, int dir) +{ + struct snd_soc_codec *codec = codec_dai->codec; + struct ml26124_priv *priv = snd_soc_codec_get_drvdata(codec); + + switch (clk_id) { + case ML26124_USE_PLLOUT: + priv->clk_in = ML26124_USE_PLLOUT; + break; + case ML26124_USE_MCLKI: + priv->clk_in = ML26124_USE_MCLKI; + break; + default: + return -EINVAL; + } + + priv->mclk = freq; + + return 0; +} + +static int ml26124_set_bias_level(struct snd_soc_codec *codec, + enum snd_soc_bias_level level) +{ + struct ml26124_priv *priv = snd_soc_codec_get_drvdata(codec); + + switch (level) { + case SND_SOC_BIAS_ON: + snd_soc_update_bits(codec, ML26124_PW_SPAMP_PW_MNG, + ML26124_R26_MASK, ML26124_BLT_PREAMP_ON); + msleep(100); + snd_soc_update_bits(codec, ML26124_PW_SPAMP_PW_MNG, + ML26124_R26_MASK, + ML26124_MICBEN_ON | ML26124_BLT_ALL_ON); + break; + case SND_SOC_BIAS_PREPARE: + break; + case SND_SOC_BIAS_STANDBY: + /* VMID ON */ + if (codec->dapm.bias_level == SND_SOC_BIAS_OFF) { + snd_soc_update_bits(codec, ML26124_PW_REF_PW_MNG, + ML26124_VMID, ML26124_VMID); + msleep(500); + regcache_sync(priv->regmap); + } + break; + case SND_SOC_BIAS_OFF: + /* VMID OFF */ + snd_soc_update_bits(codec, ML26124_PW_REF_PW_MNG, + ML26124_VMID, 0); + break; + } + codec->dapm.bias_level = level; + return 0; +} + +static const struct snd_soc_dai_ops ml26124_dai_ops = { + .hw_params = ml26124_hw_params, + .digital_mute = ml26124_mute, + .set_fmt = ml26124_set_dai_fmt, + .set_sysclk = ml26124_set_dai_sysclk, +}; + +static struct snd_soc_dai_driver ml26124_dai = { + .name = "ml26124-hifi", + .playback = { + .stream_name = "Playback", + .channels_min = 1, + .channels_max = 2, + .rates = ML26124_RATES, + .formats = ML26124_FORMATS,}, + .capture = { + .stream_name = "Capture", + .channels_min = 1, + .channels_max = 2, + .rates = ML26124_RATES, + .formats = ML26124_FORMATS,}, + .ops = &ml26124_dai_ops, + .symmetric_rates = 1, +}; + +#ifdef CONFIG_PM +static int ml26124_suspend(struct snd_soc_codec *codec) +{ + ml26124_set_bias_level(codec, SND_SOC_BIAS_OFF); + + return 0; +} + +static int ml26124_resume(struct snd_soc_codec *codec) +{ + ml26124_set_bias_level(codec, SND_SOC_BIAS_STANDBY); + + return 0; +} +#else +#define ml26124_suspend NULL +#define ml26124_resume NULL +#endif + +static int ml26124_probe(struct snd_soc_codec *codec) +{ + int ret; + struct ml26124_priv *priv = snd_soc_codec_get_drvdata(codec); + codec->control_data = priv->regmap; + + ret = snd_soc_codec_set_cache_io(codec, 7, 9, SND_SOC_REGMAP); + if (ret < 0) { + dev_err(codec->dev, "Failed to set cache I/O: %d\n", ret); + return ret; + } + + /* Software Reset */ + snd_soc_update_bits(codec, ML26124_SW_RST, 0x01, 1); + snd_soc_update_bits(codec, ML26124_SW_RST, 0x01, 0); + + ml26124_set_bias_level(codec, SND_SOC_BIAS_STANDBY); + + return 0; +} + +static struct snd_soc_codec_driver soc_codec_dev_ml26124 = { + .probe = ml26124_probe, + .suspend = ml26124_suspend, + .resume = ml26124_resume, + .set_bias_level = ml26124_set_bias_level, + .dapm_widgets = ml26124_dapm_widgets, + .num_dapm_widgets = ARRAY_SIZE(ml26124_dapm_widgets), + .dapm_routes = ml26124_intercon, + .num_dapm_routes = ARRAY_SIZE(ml26124_intercon), + .controls = ml26124_snd_controls, + .num_controls = ARRAY_SIZE(ml26124_snd_controls), +}; + +static const struct regmap_config ml26124_i2c_regmap = { + .val_bits = 8, + .reg_bits = 8, + .max_register = ML26124_NUM_REGISTER, + .reg_defaults = ml26124_reg, + .num_reg_defaults = ARRAY_SIZE(ml26124_reg), + .cache_type = REGCACHE_RBTREE, + .write_flag_mask = 0x01, +}; + +static __devinit int ml26124_i2c_probe(struct i2c_client *i2c, + const struct i2c_device_id *id) +{ + struct ml26124_priv *priv; + int ret; + + priv = devm_kzalloc(&i2c->dev, sizeof(*priv), GFP_KERNEL); + if (!priv) + return -ENOMEM; + + i2c_set_clientdata(i2c, priv); + + priv->regmap = regmap_init_i2c(i2c, &ml26124_i2c_regmap); + if (IS_ERR(priv->regmap)) { + ret = PTR_ERR(priv->regmap); + dev_err(&i2c->dev, "regmap_init_i2c() failed: %d\n", ret); + return ret; + } + + return snd_soc_register_codec(&i2c->dev, + &soc_codec_dev_ml26124, &ml26124_dai, 1); +} + +static __devexit int ml26124_i2c_remove(struct i2c_client *client) +{ + struct ml26124_priv *priv = i2c_get_clientdata(client); + + snd_soc_unregister_codec(&client->dev); + regmap_exit(priv->regmap); + return 0; +} + +static const struct i2c_device_id ml26124_i2c_id[] = { + { "ml26124", 0 }, + { } +}; +MODULE_DEVICE_TABLE(i2c, ml26124_i2c_id); + +static struct i2c_driver ml26124_i2c_driver = { + .driver = { + .name = "ml26124", + .owner = THIS_MODULE, + }, + .probe = ml26124_i2c_probe, + .remove = __devexit_p(ml26124_i2c_remove), + .id_table = ml26124_i2c_id, +}; + +module_i2c_driver(ml26124_i2c_driver); + +MODULE_AUTHOR("Tomoya MORINAGA <tomoya.rohm@gmail.com>"); +MODULE_DESCRIPTION("LAPIS Semiconductor ML26124 ALSA SoC codec driver"); +MODULE_LICENSE("GPL"); diff --git a/sound/soc/codecs/ml26124.h b/sound/soc/codecs/ml26124.h new file mode 100644 index 00000000000..5ea0cbb8c46 --- /dev/null +++ b/sound/soc/codecs/ml26124.h @@ -0,0 +1,184 @@ +/* + * Copyright (C) 2011 LAPIS Semiconductor Co., Ltd. + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License as published by + * the Free Software Foundation; version 2 of the License. + * + * This program is distributed in the hope that it will be useful, + * but WITHOUT ANY WARRANTY; without even the implied warranty of + * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the + * GNU General Public License for more details. + * + * You should have received a copy of the GNU General Public License + * along with this program; if not, write to the Free Software + * Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307, USA. + */ + +#ifndef ML26124_H +#define ML26124_H + +/* Clock Control Register */ +#define ML26124_SMPLING_RATE 0x00 +#define ML26124_PLLNL 0x02 +#define ML26124_PLLNH 0x04 +#define ML26124_PLLML 0x06 +#define ML26124_PLLMH 0x08 +#define ML26124_PLLDIV 0x0a +#define ML26124_CLK_EN 0x0c +#define ML26124_CLK_CTL 0x0e + +/* System Control Register */ +#define ML26124_SW_RST 0x10 +#define ML26124_REC_PLYBAK_RUN 0x12 +#define ML26124_MIC_TIM 0x14 + +/* Power Mnagement Register */ +#define ML26124_PW_REF_PW_MNG 0x20 +#define ML26124_PW_IN_PW_MNG 0x22 +#define ML26124_PW_DAC_PW_MNG 0x24 +#define ML26124_PW_SPAMP_PW_MNG 0x26 +#define ML26124_PW_LOUT_PW_MNG 0x28 +#define ML26124_PW_VOUT_PW_MNG 0x2a +#define ML26124_PW_ZCCMP_PW_MNG 0x2e + +/* Analog Reference Control Register */ +#define ML26124_PW_MICBIAS_VOL 0x30 + +/* Input/Output Amplifier Control Register */ +#define ML26124_PW_MIC_IN_VOL 0x32 +#define ML26124_PW_MIC_BOST_VOL 0x38 +#define ML26124_PW_SPK_AMP_VOL 0x3a +#define ML26124_PW_AMP_VOL_FUNC 0x48 +#define ML26124_PW_AMP_VOL_FADE 0x4a + +/* Analog Path Control Register */ +#define ML26124_SPK_AMP_OUT 0x54 +#define ML26124_MIC_IF_CTL 0x5a +#define ML26124_MIC_SELECT 0xe8 + +/* Audio Interface Control Register */ +#define ML26124_SAI_TRANS_CTL 0x60 +#define ML26124_SAI_RCV_CTL 0x62 +#define ML26124_SAI_MODE_SEL 0x64 + +/* DSP Control Register */ +#define ML26124_FILTER_EN 0x66 +#define ML26124_DVOL_CTL 0x68 +#define ML26124_MIXER_VOL_CTL 0x6a +#define ML26124_RECORD_DIG_VOL 0x6c +#define ML26124_PLBAK_DIG_VOL 0x70 +#define ML26124_DIGI_BOOST_VOL 0x72 +#define ML26124_EQ_GAIN_BRAND0 0x74 +#define ML26124_EQ_GAIN_BRAND1 0x76 +#define ML26124_EQ_GAIN_BRAND2 0x78 +#define ML26124_EQ_GAIN_BRAND3 0x7a +#define ML26124_EQ_GAIN_BRAND4 0x7c +#define ML26124_HPF2_CUTOFF 0x7e +#define ML26124_EQBRAND0_F0L 0x80 +#define ML26124_EQBRAND0_F0H 0x82 +#define ML26124_EQBRAND0_F1L 0x84 +#define ML26124_EQBRAND0_F1H 0x86 +#define ML26124_EQBRAND1_F0L 0x88 +#define ML26124_EQBRAND1_F0H 0x8a +#define ML26124_EQBRAND1_F1L 0x8c +#define ML26124_EQBRAND1_F1H 0x8e +#define ML26124_EQBRAND2_F0L 0x90 +#define ML26124_EQBRAND2_F0H 0x92 +#define ML26124_EQBRAND2_F1L 0x94 +#define ML26124_EQBRAND2_F1H 0x96 +#define ML26124_EQBRAND3_F0L 0x98 +#define ML26124_EQBRAND3_F0H 0x9a +#define ML26124_EQBRAND3_F1L 0x9c +#define ML26124_EQBRAND3_F1H 0x9e +#define ML26124_EQBRAND4_F0L 0xa0 +#define ML26124_EQBRAND4_F0H 0xa2 +#define ML26124_EQBRAND4_F1L 0xa4 +#define ML26124_EQBRAND4_F1H 0xa6 + +/* ALC Control Register */ +#define ML26124_ALC_MODE 0xb0 +#define ML26124_ALC_ATTACK_TIM 0xb2 +#define ML26124_ALC_DECAY_TIM 0xb4 +#define ML26124_ALC_HOLD_TIM 0xb6 +#define ML26124_ALC_TARGET_LEV 0xb8 +#define ML26124_ALC_MAXMIN_GAIN 0xba +#define ML26124_NOIS_GATE_THRSH 0xbc +#define ML26124_ALC_ZERO_TIMOUT 0xbe + +/* Playback Limiter Control Register */ +#define ML26124_PL_ATTACKTIME 0xc0 +#define ML26124_PL_DECAYTIME 0xc2 +#define ML26124_PL_TARGETTIME 0xc4 +#define ML26124_PL_MAXMIN_GAIN 0xc6 +#define ML26124_PLYBAK_BOST_VOL 0xc8 +#define ML26124_PL_0CROSS_TIMOUT 0xca + +/* Video Amplifer Control Register */ +#define ML26124_VIDEO_AMP_GAIN_CTL 0xd0 +#define ML26124_VIDEO_AMP_SETUP1 0xd2 +#define ML26124_VIDEO_AMP_CTL2 0xd4 + +/* Clock select for machine driver */ +#define ML26124_USE_PLL 0 +#define ML26124_USE_MCLKI_256FS 1 +#define ML26124_USE_MCLKI_512FS 2 +#define ML26124_USE_MCLKI_1024FS 3 + +/* Register Mask */ +#define ML26124_R0_MASK 0xf +#define ML26124_R2_MASK 0xff +#define ML26124_R4_MASK 0x1 +#define ML26124_R6_MASK 0xf +#define ML26124_R8_MASK 0x3f +#define ML26124_Ra_MASK 0x1f +#define ML26124_Rc_MASK 0x1f +#define ML26124_Re_MASK 0x7 +#define ML26124_R10_MASK 0x1 +#define ML26124_R12_MASK 0x17 +#define ML26124_R14_MASK 0x3f +#define ML26124_R20_MASK 0x47 +#define ML26124_R22_MASK 0xa +#define ML26124_R24_MASK 0x2 +#define ML26124_R26_MASK 0x1f +#define ML26124_R28_MASK 0x2 +#define ML26124_R2a_MASK 0x2 +#define ML26124_R2e_MASK 0x2 +#define ML26124_R30_MASK 0x7 +#define ML26124_R32_MASK 0x3f +#define ML26124_R38_MASK 0x38 +#define ML26124_R3a_MASK 0x3f +#define ML26124_R48_MASK 0x3 +#define ML26124_R4a_MASK 0x7 +#define ML26124_R54_MASK 0x2a +#define ML26124_R5a_MASK 0x3 +#define ML26124_Re8_MASK 0x3 +#define ML26124_R60_MASK 0xff +#define ML26124_R62_MASK 0xff +#define ML26124_R64_MASK 0x1 +#define ML26124_R66_MASK 0xff +#define ML26124_R68_MASK 0x3b +#define ML26124_R6a_MASK 0xf3 +#define ML26124_R6c_MASK 0xff +#define ML26124_R70_MASK 0xff + +#define ML26124_MCLKEN BIT(0) +#define ML26124_PLLEN BIT(1) +#define ML26124_PLLOE BIT(2) +#define ML26124_MCLKOE BIT(3) + +#define ML26124_BLT_ALL_ON 0x1f +#define ML26124_BLT_PREAMP_ON 0x13 + +#define ML26124_MICBEN_ON BIT(2) + +enum ml26124_regs { + ML26124_MCLK = 0, +}; + +enum ml26124_clk_in { + ML26124_USE_PLLOUT = 0, + ML26124_USE_MCLKI, +}; + +#endif diff --git a/sound/soc/codecs/omap-hdmi.c b/sound/soc/codecs/omap-hdmi.c new file mode 100644 index 00000000000..1bf5c74f5f9 --- /dev/null +++ b/sound/soc/codecs/omap-hdmi.c @@ -0,0 +1,69 @@ +/* + * ALSA SoC codec driver for HDMI audio on OMAP processors. + * Copyright (C) 2012 Texas Instruments Incorporated - http://www.ti.com/ + * Author: Ricardo Neri <ricardo.neri@ti.com> + * + * This program is free software; you can redistribute it and/or + * modify it under the terms of the GNU General Public License + * version 2 as published by the Free Software Foundation. + * + * This program is distributed in the hope that it will be useful, but + * WITHOUT ANY WARRANTY; without even the implied warranty of + * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU + * General Public License for more details. + * + * You should have received a copy of the GNU General Public License + * along with this program; if not, write to the Free Software + * Foundation, Inc., 51 Franklin St, Fifth Floor, Boston, MA + * 02110-1301 USA + * + */ +#include <linux/module.h> +#include <sound/soc.h> + +#define DRV_NAME "hdmi-audio-codec" + +static struct snd_soc_codec_driver omap_hdmi_codec; + +static struct snd_soc_dai_driver omap_hdmi_codec_dai = { + .name = "omap-hdmi-hifi", + .playback = { + .channels_min = 2, + .channels_max = 8, + .rates = SNDRV_PCM_RATE_32000 | + SNDRV_PCM_RATE_44100 | SNDRV_PCM_RATE_48000 | + SNDRV_PCM_RATE_88200 | SNDRV_PCM_RATE_96000 | + SNDRV_PCM_RATE_176400 | SNDRV_PCM_RATE_192000, + .formats = SNDRV_PCM_FMTBIT_S16_LE | + SNDRV_PCM_FMTBIT_S24_LE, + }, +}; + +static __devinit int omap_hdmi_codec_probe(struct platform_device *pdev) +{ + return snd_soc_register_codec(&pdev->dev, &omap_hdmi_codec, + &omap_hdmi_codec_dai, 1); +} + +static __devexit int omap_hdmi_codec_remove(struct platform_device *pdev) +{ + snd_soc_unregister_codec(&pdev->dev); + return 0; +} + +static struct platform_driver omap_hdmi_codec_driver = { + .driver = { + .name = DRV_NAME, + .owner = THIS_MODULE, + }, + + .probe = omap_hdmi_codec_probe, + .remove = __devexit_p(omap_hdmi_codec_remove), +}; + +module_platform_driver(omap_hdmi_codec_driver); + +MODULE_AUTHOR("Ricardo Neri <ricardo.neri@ti.com>"); +MODULE_DESCRIPTION("ASoC OMAP HDMI codec driver"); +MODULE_LICENSE("GPL"); +MODULE_ALIAS("platform:" DRV_NAME); diff --git a/sound/soc/codecs/rt5631.c b/sound/soc/codecs/rt5631.c index 20c324c7c34..960d0e93cce 100644 --- a/sound/soc/codecs/rt5631.c +++ b/sound/soc/codecs/rt5631.c @@ -18,7 +18,7 @@ #include <linux/delay.h> #include <linux/pm.h> #include <linux/i2c.h> -#include <linux/spi/spi.h> +#include <linux/regmap.h> #include <sound/core.h> #include <sound/pcm.h> #include <sound/pcm_params.h> @@ -30,6 +30,7 @@ #include "rt5631.h" struct rt5631_priv { + struct regmap *regmap; int codec_version; int master; int sysclk; @@ -38,33 +39,33 @@ struct rt5631_priv { int dmic_used_flag; }; -static const u16 rt5631_reg[RT5631_VENDOR_ID2 + 1] = { - [RT5631_SPK_OUT_VOL] = 0x8888, - [RT5631_HP_OUT_VOL] = 0x8080, - [RT5631_MONO_AXO_1_2_VOL] = 0xa080, - [RT5631_AUX_IN_VOL] = 0x0808, - [RT5631_ADC_REC_MIXER] = 0xf0f0, - [RT5631_VDAC_DIG_VOL] = 0x0010, - [RT5631_OUTMIXER_L_CTRL] = 0xffc0, - [RT5631_OUTMIXER_R_CTRL] = 0xffc0, - [RT5631_AXO1MIXER_CTRL] = 0x88c0, - [RT5631_AXO2MIXER_CTRL] = 0x88c0, - [RT5631_DIG_MIC_CTRL] = 0x3000, - [RT5631_MONO_INPUT_VOL] = 0x8808, - [RT5631_SPK_MIXER_CTRL] = 0xf8f8, - [RT5631_SPK_MONO_OUT_CTRL] = 0xfc00, - [RT5631_SPK_MONO_HP_OUT_CTRL] = 0x4440, - [RT5631_SDP_CTRL] = 0x8000, - [RT5631_MONO_SDP_CTRL] = 0x8000, - [RT5631_STEREO_AD_DA_CLK_CTRL] = 0x2010, - [RT5631_GEN_PUR_CTRL_REG] = 0x0e00, - [RT5631_INT_ST_IRQ_CTRL_2] = 0x071a, - [RT5631_MISC_CTRL] = 0x2040, - [RT5631_DEPOP_FUN_CTRL_2] = 0x8000, - [RT5631_SOFT_VOL_CTRL] = 0x07e0, - [RT5631_ALC_CTRL_1] = 0x0206, - [RT5631_ALC_CTRL_3] = 0x2000, - [RT5631_PSEUDO_SPATL_CTRL] = 0x0553, +static const struct reg_default rt5631_reg[] = { + { RT5631_SPK_OUT_VOL, 0x8888 }, + { RT5631_HP_OUT_VOL, 0x8080 }, + { RT5631_MONO_AXO_1_2_VOL, 0xa080 }, + { RT5631_AUX_IN_VOL, 0x0808 }, + { RT5631_ADC_REC_MIXER, 0xf0f0 }, + { RT5631_VDAC_DIG_VOL, 0x0010 }, + { RT5631_OUTMIXER_L_CTRL, 0xffc0 }, + { RT5631_OUTMIXER_R_CTRL, 0xffc0 }, + { RT5631_AXO1MIXER_CTRL, 0x88c0 }, + { RT5631_AXO2MIXER_CTRL, 0x88c0 }, + { RT5631_DIG_MIC_CTRL, 0x3000 }, + { RT5631_MONO_INPUT_VOL, 0x8808 }, + { RT5631_SPK_MIXER_CTRL, 0xf8f8 }, + { RT5631_SPK_MONO_OUT_CTRL, 0xfc00 }, + { RT5631_SPK_MONO_HP_OUT_CTRL, 0x4440 }, + { RT5631_SDP_CTRL, 0x8000 }, + { RT5631_MONO_SDP_CTRL, 0x8000 }, + { RT5631_STEREO_AD_DA_CLK_CTRL, 0x2010 }, + { RT5631_GEN_PUR_CTRL_REG, 0x0e00 }, + { RT5631_INT_ST_IRQ_CTRL_2, 0x071a }, + { RT5631_MISC_CTRL, 0x2040 }, + { RT5631_DEPOP_FUN_CTRL_2, 0x8000 }, + { RT5631_SOFT_VOL_CTRL, 0x07e0 }, + { RT5631_ALC_CTRL_1, 0x0206 }, + { RT5631_ALC_CTRL_3, 0x2000 }, + { RT5631_PSEUDO_SPATL_CTRL, 0x0553 }, }; /** @@ -96,8 +97,7 @@ static int rt5631_reset(struct snd_soc_codec *codec) return snd_soc_write(codec, RT5631_RESET, 0); } -static int rt5631_volatile_register(struct snd_soc_codec *codec, - unsigned int reg) +static bool rt5631_volatile_register(struct device *dev, unsigned int reg) { switch (reg) { case RT5631_RESET: @@ -111,8 +111,7 @@ static int rt5631_volatile_register(struct snd_soc_codec *codec, } } -static int rt5631_readable_register(struct snd_soc_codec *codec, - unsigned int reg) +static bool rt5631_readable_register(struct device *dev, unsigned int reg) { switch (reg) { case RT5631_RESET: @@ -1361,8 +1360,7 @@ static int get_coeff(int mclk, int rate, int timesofbclk) static int rt5631_hifi_pcm_params(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *params, struct snd_soc_dai *dai) { - struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct snd_soc_codec *codec = rtd->codec; + struct snd_soc_codec *codec = dai->codec; struct rt5631_priv *rt5631 = snd_soc_codec_get_drvdata(codec); int timesofbclk = 32, coeff; unsigned int iface = 0; @@ -1544,6 +1542,8 @@ static int rt5631_codec_set_dai_pll(struct snd_soc_dai *codec_dai, int pll_id, static int rt5631_set_bias_level(struct snd_soc_codec *codec, enum snd_soc_bias_level level) { + struct rt5631_priv *rt5631 = snd_soc_codec_get_drvdata(codec); + switch (level) { case SND_SOC_BIAS_ON: case SND_SOC_BIAS_PREPARE: @@ -1561,8 +1561,8 @@ static int rt5631_set_bias_level(struct snd_soc_codec *codec, snd_soc_update_bits(codec, RT5631_PWR_MANAG_ADD3, RT5631_PWR_FAST_VREF_CTRL, RT5631_PWR_FAST_VREF_CTRL); - codec->cache_only = false; - snd_soc_cache_sync(codec); + regcache_cache_only(rt5631->regmap, false); + regcache_sync(rt5631->regmap); } break; @@ -1587,7 +1587,9 @@ static int rt5631_probe(struct snd_soc_codec *codec) unsigned int val; int ret; - ret = snd_soc_codec_set_cache_io(codec, 8, 16, SND_SOC_I2C); + codec->control_data = rt5631->regmap; + + ret = snd_soc_codec_set_cache_io(codec, 8, 16, SND_SOC_REGMAP); if (ret != 0) { dev_err(codec->dev, "Failed to set cache I/O: %d\n", ret); return ret; @@ -1698,12 +1700,6 @@ static struct snd_soc_codec_driver soc_codec_dev_rt5631 = { .suspend = rt5631_suspend, .resume = rt5631_resume, .set_bias_level = rt5631_set_bias_level, - .reg_cache_size = RT5631_VENDOR_ID2 + 1, - .reg_word_size = sizeof(u16), - .reg_cache_default = rt5631_reg, - .volatile_register = rt5631_volatile_register, - .readable_register = rt5631_readable_register, - .reg_cache_step = 1, .controls = rt5631_snd_controls, .num_controls = ARRAY_SIZE(rt5631_snd_controls), .dapm_widgets = rt5631_dapm_widgets, @@ -1718,6 +1714,18 @@ static const struct i2c_device_id rt5631_i2c_id[] = { }; MODULE_DEVICE_TABLE(i2c, rt5631_i2c_id); +static const struct regmap_config rt5631_regmap_config = { + .reg_bits = 8, + .val_bits = 16, + + .readable_reg = rt5631_readable_register, + .volatile_reg = rt5631_volatile_register, + .max_register = RT5631_VENDOR_ID2, + .reg_defaults = rt5631_reg, + .num_reg_defaults = ARRAY_SIZE(rt5631_reg), + .cache_type = REGCACHE_RBTREE, +}; + static int rt5631_i2c_probe(struct i2c_client *i2c, const struct i2c_device_id *id) { @@ -1731,6 +1739,10 @@ static int rt5631_i2c_probe(struct i2c_client *i2c, i2c_set_clientdata(i2c, rt5631); + rt5631->regmap = devm_regmap_init_i2c(i2c, &rt5631_regmap_config); + if (IS_ERR(rt5631->regmap)) + return PTR_ERR(rt5631->regmap); + ret = snd_soc_register_codec(&i2c->dev, &soc_codec_dev_rt5631, rt5631_dai, ARRAY_SIZE(rt5631_dai)); return ret; @@ -1752,17 +1764,7 @@ static struct i2c_driver rt5631_i2c_driver = { .id_table = rt5631_i2c_id, }; -static int __init rt5631_modinit(void) -{ - return i2c_add_driver(&rt5631_i2c_driver); -} -module_init(rt5631_modinit); - -static void __exit rt5631_modexit(void) -{ - i2c_del_driver(&rt5631_i2c_driver); -} -module_exit(rt5631_modexit); +module_i2c_driver(rt5631_i2c_driver); MODULE_DESCRIPTION("ASoC RT5631 driver"); MODULE_AUTHOR("flove <flove@realtek.com>"); diff --git a/sound/soc/codecs/sgtl5000.c b/sound/soc/codecs/sgtl5000.c index c395ec37044..8af6a5245b1 100644 --- a/sound/soc/codecs/sgtl5000.c +++ b/sound/soc/codecs/sgtl5000.c @@ -84,8 +84,8 @@ static struct regulator_consumer_supply ldo_consumer[] = { static struct regulator_init_data ldo_init_data = { .constraints = { - .min_uV = 850000, - .max_uV = 1600000, + .min_uV = 1200000, + .max_uV = 1200000, .valid_modes_mask = REGULATOR_MODE_NORMAL, .valid_ops_mask = REGULATOR_CHANGE_STATUS, }, @@ -197,9 +197,9 @@ static const struct snd_soc_dapm_widget sgtl5000_dapm_widgets[] = { SND_SOC_DAPM_OUTPUT("HP_OUT"), SND_SOC_DAPM_OUTPUT("LINE_OUT"), - SND_SOC_DAPM_MICBIAS_E("Mic Bias", SGTL5000_CHIP_MIC_CTRL, 8, 0, - mic_bias_event, - SND_SOC_DAPM_POST_PMU | SND_SOC_DAPM_PRE_PMD), + SND_SOC_DAPM_SUPPLY("Mic Bias", SGTL5000_CHIP_MIC_CTRL, 8, 0, + mic_bias_event, + SND_SOC_DAPM_POST_PMU | SND_SOC_DAPM_PRE_PMD), SND_SOC_DAPM_PGA("HP", SGTL5000_CHIP_ANA_POWER, 4, 0, NULL, 0), SND_SOC_DAPM_PGA("LO", SGTL5000_CHIP_ANA_POWER, 0, 0, NULL, 0), @@ -665,8 +665,7 @@ static int sgtl5000_pcm_hw_params(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *params, struct snd_soc_dai *dai) { - struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct snd_soc_codec *codec = rtd->codec; + struct snd_soc_codec *codec = dai->codec; struct sgtl5000_priv *sgtl5000 = snd_soc_codec_get_drvdata(codec); int channels = params_channels(params); int i2s_ctl = 0; @@ -1455,17 +1454,7 @@ static struct i2c_driver sgtl5000_i2c_driver = { .id_table = sgtl5000_id, }; -static int __init sgtl5000_modinit(void) -{ - return i2c_add_driver(&sgtl5000_i2c_driver); -} -module_init(sgtl5000_modinit); - -static void __exit sgtl5000_exit(void) -{ - i2c_del_driver(&sgtl5000_i2c_driver); -} -module_exit(sgtl5000_exit); +module_i2c_driver(sgtl5000_i2c_driver); MODULE_DESCRIPTION("Freescale SGTL5000 ALSA SoC Codec Driver"); MODULE_AUTHOR("Zeng Zhaoming <zengzm.kernel@gmail.com>"); diff --git a/sound/soc/codecs/ssm2602.c b/sound/soc/codecs/ssm2602.c index de2b20544ce..079066fef42 100644 --- a/sound/soc/codecs/ssm2602.c +++ b/sound/soc/codecs/ssm2602.c @@ -33,6 +33,7 @@ #include <linux/pm.h> #include <linux/i2c.h> #include <linux/spi/spi.h> +#include <linux/regmap.h> #include <linux/slab.h> #include <sound/core.h> #include <sound/pcm.h> @@ -43,8 +44,6 @@ #include "ssm2602.h" -#define SSM2602_VERSION "0.1" - enum ssm2602_type { SSM2602, SSM2604, @@ -53,10 +52,12 @@ enum ssm2602_type { /* codec private data */ struct ssm2602_priv { unsigned int sysclk; - enum snd_soc_control_type control_type; + struct snd_pcm_hw_constraint_list *sysclk_constraints; struct snd_pcm_substream *master_substream; struct snd_pcm_substream *slave_substream; + struct regmap *regmap; + enum ssm2602_type type; unsigned int clk_out_pwr; }; @@ -73,7 +74,6 @@ static const u16 ssm2602_reg[SSM2602_CACHEREGNUM] = { 0x0000, 0x0000 }; -#define ssm2602_reset(c) snd_soc_write(c, SSM2602_RESET, 0) /*Appending several "None"s just for OSS mixer use*/ static const char *ssm2602_input_select[] = { @@ -195,6 +195,24 @@ static const struct snd_soc_dapm_route ssm2604_routes[] = { {"ADC", NULL, "Line Input"}, }; +static const unsigned int ssm2602_rates_12288000[] = { + 8000, 32000, 48000, 96000, +}; + +static struct snd_pcm_hw_constraint_list ssm2602_constraints_12288000 = { + .list = ssm2602_rates_12288000, + .count = ARRAY_SIZE(ssm2602_rates_12288000), +}; + +static const unsigned int ssm2602_rates_11289600[] = { + 8000, 44100, 88200, +}; + +static struct snd_pcm_hw_constraint_list ssm2602_constraints_11289600 = { + .list = ssm2602_rates_11289600, + .count = ARRAY_SIZE(ssm2602_rates_11289600), +}; + struct ssm2602_coeff { u32 mclk; u32 rate; @@ -254,11 +272,10 @@ static int ssm2602_hw_params(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *params, struct snd_soc_dai *dai) { - struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct snd_soc_codec *codec = rtd->codec; + struct snd_soc_codec *codec = dai->codec; struct ssm2602_priv *ssm2602 = snd_soc_codec_get_drvdata(codec); - u16 iface = snd_soc_read(codec, SSM2602_IFACE) & 0xfff3; int srate = ssm2602_get_coeff(ssm2602->sysclk, params_rate(params)); + unsigned int iface; if (substream == ssm2602->slave_substream) { dev_dbg(codec->dev, "Ignoring hw_params for slave substream\n"); @@ -268,31 +285,34 @@ static int ssm2602_hw_params(struct snd_pcm_substream *substream, if (srate < 0) return srate; - snd_soc_write(codec, SSM2602_SRATE, srate); + regmap_write(ssm2602->regmap, SSM2602_SRATE, srate); /* bit size */ switch (params_format(params)) { case SNDRV_PCM_FORMAT_S16_LE: + iface = 0x0; break; case SNDRV_PCM_FORMAT_S20_3LE: - iface |= 0x0004; + iface = 0x4; break; case SNDRV_PCM_FORMAT_S24_LE: - iface |= 0x0008; + iface = 0x8; break; case SNDRV_PCM_FORMAT_S32_LE: - iface |= 0x000c; + iface = 0xc; break; + default: + return -EINVAL; } - snd_soc_write(codec, SSM2602_IFACE, iface); + regmap_update_bits(ssm2602->regmap, SSM2602_IFACE, + IFACE_AUDIO_DATA_LEN, iface); return 0; } static int ssm2602_startup(struct snd_pcm_substream *substream, struct snd_soc_dai *dai) { - struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct snd_soc_codec *codec = rtd->codec; + struct snd_soc_codec *codec = dai->codec; struct ssm2602_priv *ssm2602 = snd_soc_codec_get_drvdata(codec); struct snd_pcm_runtime *master_runtime; @@ -322,14 +342,19 @@ static int ssm2602_startup(struct snd_pcm_substream *substream, } else ssm2602->master_substream = substream; + if (ssm2602->sysclk_constraints) { + snd_pcm_hw_constraint_list(substream->runtime, 0, + SNDRV_PCM_HW_PARAM_RATE, + ssm2602->sysclk_constraints); + } + return 0; } static void ssm2602_shutdown(struct snd_pcm_substream *substream, struct snd_soc_dai *dai) { - struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct snd_soc_codec *codec = rtd->codec; + struct snd_soc_codec *codec = dai->codec; struct ssm2602_priv *ssm2602 = snd_soc_codec_get_drvdata(codec); if (ssm2602->master_substream == substream) @@ -341,14 +366,14 @@ static void ssm2602_shutdown(struct snd_pcm_substream *substream, static int ssm2602_mute(struct snd_soc_dai *dai, int mute) { - struct snd_soc_codec *codec = dai->codec; + struct ssm2602_priv *ssm2602 = snd_soc_codec_get_drvdata(dai->codec); if (mute) - snd_soc_update_bits(codec, SSM2602_APDIGI, + regmap_update_bits(ssm2602->regmap, SSM2602_APDIGI, APDIGI_ENABLE_DAC_MUTE, APDIGI_ENABLE_DAC_MUTE); else - snd_soc_update_bits(codec, SSM2602_APDIGI, + regmap_update_bits(ssm2602->regmap, SSM2602_APDIGI, APDIGI_ENABLE_DAC_MUTE, 0); return 0; } @@ -364,16 +389,21 @@ static int ssm2602_set_dai_sysclk(struct snd_soc_dai *codec_dai, return -EINVAL; switch (freq) { - case 11289600: - case 12000000: case 12288000: - case 16934400: case 18432000: - ssm2602->sysclk = freq; + ssm2602->sysclk_constraints = &ssm2602_constraints_12288000; + break; + case 11289600: + case 16934400: + ssm2602->sysclk_constraints = &ssm2602_constraints_11289600; + break; + case 12000000: + ssm2602->sysclk_constraints = NULL; break; default: return -EINVAL; } + ssm2602->sysclk = freq; } else { unsigned int mask; @@ -393,7 +423,7 @@ static int ssm2602_set_dai_sysclk(struct snd_soc_dai *codec_dai, else ssm2602->clk_out_pwr &= ~mask; - snd_soc_update_bits(codec, SSM2602_PWR, + regmap_update_bits(ssm2602->regmap, SSM2602_PWR, PWR_CLK_OUT_PDN | PWR_OSC_PDN, ssm2602->clk_out_pwr); } @@ -403,8 +433,8 @@ static int ssm2602_set_dai_sysclk(struct snd_soc_dai *codec_dai, static int ssm2602_set_dai_fmt(struct snd_soc_dai *codec_dai, unsigned int fmt) { - struct snd_soc_codec *codec = codec_dai->codec; - u16 iface = 0; + struct ssm2602_priv *ssm2602 = snd_soc_codec_get_drvdata(codec_dai->codec); + unsigned int iface = 0; /* set master/slave audio interface */ switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) { @@ -455,7 +485,7 @@ static int ssm2602_set_dai_fmt(struct snd_soc_dai *codec_dai, } /* set iface */ - snd_soc_write(codec, SSM2602_IFACE, iface); + regmap_write(ssm2602->regmap, SSM2602_IFACE, iface); return 0; } @@ -467,7 +497,7 @@ static int ssm2602_set_bias_level(struct snd_soc_codec *codec, switch (level) { case SND_SOC_BIAS_ON: /* vref/mid on, osc and clkout on if enabled */ - snd_soc_update_bits(codec, SSM2602_PWR, + regmap_update_bits(ssm2602->regmap, SSM2602_PWR, PWR_POWER_OFF | PWR_CLK_OUT_PDN | PWR_OSC_PDN, ssm2602->clk_out_pwr); break; @@ -475,13 +505,13 @@ static int ssm2602_set_bias_level(struct snd_soc_codec *codec, break; case SND_SOC_BIAS_STANDBY: /* everything off except vref/vmid, */ - snd_soc_update_bits(codec, SSM2602_PWR, + regmap_update_bits(ssm2602->regmap, SSM2602_PWR, PWR_POWER_OFF | PWR_CLK_OUT_PDN | PWR_OSC_PDN, PWR_CLK_OUT_PDN | PWR_OSC_PDN); break; case SND_SOC_BIAS_OFF: /* everything off */ - snd_soc_update_bits(codec, SSM2602_PWR, + regmap_update_bits(ssm2602->regmap, SSM2602_PWR, PWR_POWER_OFF, PWR_POWER_OFF); break; @@ -540,12 +570,13 @@ static int ssm2602_resume(struct snd_soc_codec *codec) static int ssm2602_probe(struct snd_soc_codec *codec) { + struct ssm2602_priv *ssm2602 = snd_soc_codec_get_drvdata(codec); struct snd_soc_dapm_context *dapm = &codec->dapm; int ret; - snd_soc_update_bits(codec, SSM2602_LOUT1V, + regmap_update_bits(ssm2602->regmap, SSM2602_LOUT1V, LOUT1V_LRHP_BOTH, LOUT1V_LRHP_BOTH); - snd_soc_update_bits(codec, SSM2602_ROUT1V, + regmap_update_bits(ssm2602->regmap, SSM2602_ROUT1V, ROUT1V_RLHP_BOTH, ROUT1V_RLHP_BOTH); ret = snd_soc_add_codec_controls(codec, ssm2602_snd_controls, @@ -581,27 +612,26 @@ static int ssm260x_probe(struct snd_soc_codec *codec) struct ssm2602_priv *ssm2602 = snd_soc_codec_get_drvdata(codec); int ret; - pr_info("ssm2602 Audio Codec %s", SSM2602_VERSION); - - ret = snd_soc_codec_set_cache_io(codec, 7, 9, ssm2602->control_type); + codec->control_data = ssm2602->regmap; + ret = snd_soc_codec_set_cache_io(codec, 0, 0, SND_SOC_REGMAP); if (ret < 0) { dev_err(codec->dev, "Failed to set cache I/O: %d\n", ret); return ret; } - ret = ssm2602_reset(codec); + ret = regmap_write(ssm2602->regmap, SSM2602_RESET, 0); if (ret < 0) { dev_err(codec->dev, "Failed to issue reset: %d\n", ret); return ret; } /* set the update bits */ - snd_soc_update_bits(codec, SSM2602_LINVOL, + regmap_update_bits(ssm2602->regmap, SSM2602_LINVOL, LINVOL_LRIN_BOTH, LINVOL_LRIN_BOTH); - snd_soc_update_bits(codec, SSM2602_RINVOL, + regmap_update_bits(ssm2602->regmap, SSM2602_RINVOL, RINVOL_RLIN_BOTH, RINVOL_RLIN_BOTH); /*select Line in as default input*/ - snd_soc_write(codec, SSM2602_APANA, APANA_SELECT_DAC | + regmap_write(ssm2602->regmap, SSM2602_APANA, APANA_SELECT_DAC | APANA_ENABLE_MIC_BOOST); switch (ssm2602->type) { @@ -634,9 +664,6 @@ static struct snd_soc_codec_driver soc_codec_dev_ssm2602 = { .suspend = ssm2602_suspend, .resume = ssm2602_resume, .set_bias_level = ssm2602_set_bias_level, - .reg_cache_size = ARRAY_SIZE(ssm2602_reg), - .reg_word_size = sizeof(u16), - .reg_cache_default = ssm2602_reg, .controls = ssm260x_snd_controls, .num_controls = ARRAY_SIZE(ssm260x_snd_controls), @@ -646,6 +673,23 @@ static struct snd_soc_codec_driver soc_codec_dev_ssm2602 = { .num_dapm_routes = ARRAY_SIZE(ssm260x_routes), }; +static bool ssm2602_register_volatile(struct device *dev, unsigned int reg) +{ + return reg == SSM2602_RESET; +} + +static const struct regmap_config ssm2602_regmap_config = { + .val_bits = 9, + .reg_bits = 7, + + .max_register = SSM2602_RESET, + .volatile_reg = ssm2602_register_volatile, + + .cache_type = REGCACHE_RBTREE, + .reg_defaults_raw = ssm2602_reg, + .num_reg_defaults_raw = ARRAY_SIZE(ssm2602_reg), +}; + #if defined(CONFIG_SPI_MASTER) static int __devinit ssm2602_spi_probe(struct spi_device *spi) { @@ -658,9 +702,12 @@ static int __devinit ssm2602_spi_probe(struct spi_device *spi) return -ENOMEM; spi_set_drvdata(spi, ssm2602); - ssm2602->control_type = SND_SOC_SPI; ssm2602->type = SSM2602; + ssm2602->regmap = devm_regmap_init_spi(spi, &ssm2602_regmap_config); + if (IS_ERR(ssm2602->regmap)) + return PTR_ERR(ssm2602->regmap); + ret = snd_soc_register_codec(&spi->dev, &soc_codec_dev_ssm2602, &ssm2602_dai, 1); return ret; @@ -701,9 +748,12 @@ static int __devinit ssm2602_i2c_probe(struct i2c_client *i2c, return -ENOMEM; i2c_set_clientdata(i2c, ssm2602); - ssm2602->control_type = SND_SOC_I2C; ssm2602->type = id->driver_data; + ssm2602->regmap = devm_regmap_init_i2c(i2c, &ssm2602_regmap_config); + if (IS_ERR(ssm2602->regmap)) + return PTR_ERR(ssm2602->regmap); + ret = snd_soc_register_codec(&i2c->dev, &soc_codec_dev_ssm2602, &ssm2602_dai, 1); return ret; diff --git a/sound/soc/codecs/sta32x.c b/sound/soc/codecs/sta32x.c index 7db6fa51502..8d717f4b5a8 100644 --- a/sound/soc/codecs/sta32x.c +++ b/sound/soc/codecs/sta32x.c @@ -609,8 +609,7 @@ static int sta32x_hw_params(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *params, struct snd_soc_dai *dai) { - struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct snd_soc_codec *codec = rtd->codec; + struct snd_soc_codec *codec = dai->codec; struct sta32x_priv *sta32x = snd_soc_codec_get_drvdata(codec); unsigned int rate; int i, mcs = -1, ir = -1; diff --git a/sound/soc/codecs/tlv320aic23.c b/sound/soc/codecs/tlv320aic23.c index df1e07ffac3..31762ebdd77 100644 --- a/sound/soc/codecs/tlv320aic23.c +++ b/sound/soc/codecs/tlv320aic23.c @@ -34,8 +34,6 @@ #include "tlv320aic23.h" -#define AIC23_VERSION "0.1" - /* * AIC23 register cache */ @@ -325,8 +323,7 @@ static int tlv320aic23_hw_params(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *params, struct snd_soc_dai *dai) { - struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct snd_soc_codec *codec = rtd->codec; + struct snd_soc_codec *codec = dai->codec; u16 iface_reg; int ret; struct aic23 *aic23 = snd_soc_codec_get_drvdata(codec); @@ -371,8 +368,7 @@ static int tlv320aic23_hw_params(struct snd_pcm_substream *substream, static int tlv320aic23_pcm_prepare(struct snd_pcm_substream *substream, struct snd_soc_dai *dai) { - struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct snd_soc_codec *codec = rtd->codec; + struct snd_soc_codec *codec = dai->codec; /* set active */ snd_soc_write(codec, TLV320AIC23_ACTIVE, 0x0001); @@ -383,8 +379,7 @@ static int tlv320aic23_pcm_prepare(struct snd_pcm_substream *substream, static void tlv320aic23_shutdown(struct snd_pcm_substream *substream, struct snd_soc_dai *dai) { - struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct snd_soc_codec *codec = rtd->codec; + struct snd_soc_codec *codec = dai->codec; struct aic23 *aic23 = snd_soc_codec_get_drvdata(codec); /* deactivate */ @@ -548,8 +543,6 @@ static int tlv320aic23_probe(struct snd_soc_codec *codec) struct aic23 *aic23 = snd_soc_codec_get_drvdata(codec); int ret; - printk(KERN_INFO "AIC23 Audio Codec %s\n", AIC23_VERSION); - ret = snd_soc_codec_set_cache_io(codec, 7, 9, aic23->control_type); if (ret < 0) { dev_err(codec->dev, "Failed to set cache I/O: %d\n", ret); diff --git a/sound/soc/codecs/tlv320aic26.c b/sound/soc/codecs/tlv320aic26.c index 802064b5030..85944e95357 100644 --- a/sound/soc/codecs/tlv320aic26.c +++ b/sound/soc/codecs/tlv320aic26.c @@ -126,8 +126,7 @@ static int aic26_hw_params(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *params, struct snd_soc_dai *dai) { - struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct snd_soc_codec *codec = rtd->codec; + struct snd_soc_codec *codec = dai->codec; struct aic26 *aic26 = snd_soc_codec_get_drvdata(codec); int fsref, divisor, wlen, pval, jval, dval, qval; u16 reg; diff --git a/sound/soc/codecs/tlv320aic3x.c b/sound/soc/codecs/tlv320aic3x.c index 8d20f6ec20f..64d2a4fa34b 100644 --- a/sound/soc/codecs/tlv320aic3x.c +++ b/sound/soc/codecs/tlv320aic3x.c @@ -802,8 +802,7 @@ static int aic3x_hw_params(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *params, struct snd_soc_dai *dai) { - struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct snd_soc_codec *codec =rtd->codec; + struct snd_soc_codec *codec = dai->codec; struct aic3x_priv *aic3x = snd_soc_codec_get_drvdata(codec); int codec_clk = 0, bypass_pll = 0, fsref, last_clk = 0; u8 data, j, r, p, pll_q, pll_p = 1, pll_r = 1, pll_j = 1; @@ -1161,24 +1160,6 @@ static int aic3x_set_bias_level(struct snd_soc_codec *codec, return 0; } -void aic3x_set_headset_detection(struct snd_soc_codec *codec, int detect, - int headset_debounce, int button_debounce) -{ - u8 val; - - val = ((detect & AIC3X_HEADSET_DETECT_MASK) - << AIC3X_HEADSET_DETECT_SHIFT) | - ((headset_debounce & AIC3X_HEADSET_DEBOUNCE_MASK) - << AIC3X_HEADSET_DEBOUNCE_SHIFT) | - ((button_debounce & AIC3X_BUTTON_DEBOUNCE_MASK) - << AIC3X_BUTTON_DEBOUNCE_SHIFT); - - if (detect & AIC3X_HEADSET_DETECT_MASK) - val |= AIC3X_HEADSET_DETECT_ENABLED; - - snd_soc_write(codec, AIC3X_HEADSET_DETECT_CTRL_A, val); -} - #define AIC3X_RATES SNDRV_PCM_RATE_8000_96000 #define AIC3X_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S20_3LE | \ SNDRV_PCM_FMTBIT_S24_3LE | SNDRV_PCM_FMTBIT_S32_LE) diff --git a/sound/soc/codecs/tlv320dac33.c b/sound/soc/codecs/tlv320dac33.c index 4587ddd0fbf..0dd41077ab7 100644 --- a/sound/soc/codecs/tlv320dac33.c +++ b/sound/soc/codecs/tlv320dac33.c @@ -62,8 +62,10 @@ #define UTHR_FROM_PERIOD_SIZE(samples, playrate, burstrate) \ (((samples)*5000) / (((burstrate)*5000) / ((burstrate) - (playrate)))) -static void dac33_calculate_times(struct snd_pcm_substream *substream); -static int dac33_prepare_chip(struct snd_pcm_substream *substream); +static void dac33_calculate_times(struct snd_pcm_substream *substream, + struct snd_soc_codec *codec); +static int dac33_prepare_chip(struct snd_pcm_substream *substream, + struct snd_soc_codec *codec); enum dac33_state { DAC33_IDLE = 0, @@ -427,8 +429,8 @@ static int dac33_playback_event(struct snd_soc_dapm_widget *w, switch (event) { case SND_SOC_DAPM_PRE_PMU: if (likely(dac33->substream)) { - dac33_calculate_times(dac33->substream); - dac33_prepare_chip(dac33->substream); + dac33_calculate_times(dac33->substream, w->codec); + dac33_prepare_chip(dac33->substream, w->codec); } break; case SND_SOC_DAPM_POST_PMD: @@ -799,8 +801,7 @@ static void dac33_oscwait(struct snd_soc_codec *codec) static int dac33_startup(struct snd_pcm_substream *substream, struct snd_soc_dai *dai) { - struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct snd_soc_codec *codec = rtd->codec; + struct snd_soc_codec *codec = dai->codec; struct tlv320dac33_priv *dac33 = snd_soc_codec_get_drvdata(codec); /* Stream started, save the substream pointer */ @@ -812,8 +813,7 @@ static int dac33_startup(struct snd_pcm_substream *substream, static void dac33_shutdown(struct snd_pcm_substream *substream, struct snd_soc_dai *dai) { - struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct snd_soc_codec *codec = rtd->codec; + struct snd_soc_codec *codec = dai->codec; struct tlv320dac33_priv *dac33 = snd_soc_codec_get_drvdata(codec); dac33->substream = NULL; @@ -825,8 +825,7 @@ static int dac33_hw_params(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *params, struct snd_soc_dai *dai) { - struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct snd_soc_codec *codec = rtd->codec; + struct snd_soc_codec *codec = dai->codec; struct tlv320dac33_priv *dac33 = snd_soc_codec_get_drvdata(codec); /* Check parameters for validity */ @@ -868,10 +867,9 @@ static int dac33_hw_params(struct snd_pcm_substream *substream, * writes happens in different order, than dac33 might end up in unknown state. * Use the known, working sequence of register writes to initialize the dac33. */ -static int dac33_prepare_chip(struct snd_pcm_substream *substream) +static int dac33_prepare_chip(struct snd_pcm_substream *substream, + struct snd_soc_codec *codec) { - struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct snd_soc_codec *codec = rtd->codec; struct tlv320dac33_priv *dac33 = snd_soc_codec_get_drvdata(codec); unsigned int oscset, ratioset, pwr_ctrl, reg_tmp; u8 aictrl_a, aictrl_b, fifoctrl_a; @@ -1067,10 +1065,9 @@ static int dac33_prepare_chip(struct snd_pcm_substream *substream) return 0; } -static void dac33_calculate_times(struct snd_pcm_substream *substream) +static void dac33_calculate_times(struct snd_pcm_substream *substream, + struct snd_soc_codec *codec) { - struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct snd_soc_codec *codec = rtd->codec; struct tlv320dac33_priv *dac33 = snd_soc_codec_get_drvdata(codec); unsigned int period_size = substream->runtime->period_size; unsigned int rate = substream->runtime->rate; @@ -1128,8 +1125,7 @@ static void dac33_calculate_times(struct snd_pcm_substream *substream) static int dac33_pcm_trigger(struct snd_pcm_substream *substream, int cmd, struct snd_soc_dai *dai) { - struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct snd_soc_codec *codec = rtd->codec; + struct snd_soc_codec *codec = dai->codec; struct tlv320dac33_priv *dac33 = snd_soc_codec_get_drvdata(codec); int ret = 0; @@ -1161,8 +1157,7 @@ static snd_pcm_sframes_t dac33_dai_delay( struct snd_pcm_substream *substream, struct snd_soc_dai *dai) { - struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct snd_soc_codec *codec = rtd->codec; + struct snd_soc_codec *codec = dai->codec; struct tlv320dac33_priv *dac33 = snd_soc_codec_get_drvdata(codec); unsigned long long t0, t1, t_now; unsigned int time_delta, uthr; diff --git a/sound/soc/codecs/twl4030.c b/sound/soc/codecs/twl4030.c index 170cf9a8fc7..391fcfc7b63 100644 --- a/sound/soc/codecs/twl4030.c +++ b/sound/soc/codecs/twl4030.c @@ -1685,8 +1685,7 @@ static void twl4030_tdm_enable(struct snd_soc_codec *codec, int direction, static int twl4030_startup(struct snd_pcm_substream *substream, struct snd_soc_dai *dai) { - struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct snd_soc_codec *codec = rtd->codec; + struct snd_soc_codec *codec = dai->codec; struct twl4030_priv *twl4030 = snd_soc_codec_get_drvdata(codec); if (twl4030->master_substream) { @@ -1715,8 +1714,7 @@ static int twl4030_startup(struct snd_pcm_substream *substream, static void twl4030_shutdown(struct snd_pcm_substream *substream, struct snd_soc_dai *dai) { - struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct snd_soc_codec *codec = rtd->codec; + struct snd_soc_codec *codec = dai->codec; struct twl4030_priv *twl4030 = snd_soc_codec_get_drvdata(codec); if (twl4030->master_substream == substream) @@ -1740,8 +1738,7 @@ static int twl4030_hw_params(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *params, struct snd_soc_dai *dai) { - struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct snd_soc_codec *codec = rtd->codec; + struct snd_soc_codec *codec = dai->codec; struct twl4030_priv *twl4030 = snd_soc_codec_get_drvdata(codec); u8 mode, old_mode, format, old_format; @@ -1974,8 +1971,7 @@ static void twl4030_voice_enable(struct snd_soc_codec *codec, int direction, static int twl4030_voice_startup(struct snd_pcm_substream *substream, struct snd_soc_dai *dai) { - struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct snd_soc_codec *codec = rtd->codec; + struct snd_soc_codec *codec = dai->codec; struct twl4030_priv *twl4030 = snd_soc_codec_get_drvdata(codec); u8 mode; @@ -2007,8 +2003,7 @@ static int twl4030_voice_startup(struct snd_pcm_substream *substream, static void twl4030_voice_shutdown(struct snd_pcm_substream *substream, struct snd_soc_dai *dai) { - struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct snd_soc_codec *codec = rtd->codec; + struct snd_soc_codec *codec = dai->codec; /* Enable voice digital filters */ twl4030_voice_enable(codec, substream->stream, 0); @@ -2017,8 +2012,7 @@ static void twl4030_voice_shutdown(struct snd_pcm_substream *substream, static int twl4030_voice_hw_params(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *params, struct snd_soc_dai *dai) { - struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct snd_soc_codec *codec = rtd->codec; + struct snd_soc_codec *codec = dai->codec; struct twl4030_priv *twl4030 = snd_soc_codec_get_drvdata(codec); u8 old_mode, mode; diff --git a/sound/soc/codecs/twl6040.c b/sound/soc/codecs/twl6040.c index dc7509b9d53..a36e9fcdf18 100644 --- a/sound/soc/codecs/twl6040.c +++ b/sound/soc/codecs/twl6040.c @@ -46,17 +46,6 @@ #define TWL6040_OUTHF_0dB 0x03 #define TWL6040_OUTHF_M52dB 0x1D -#define TWL6040_RAMP_NONE 0 -#define TWL6040_RAMP_UP 1 -#define TWL6040_RAMP_DOWN 2 - -#define TWL6040_HSL_VOL_MASK 0x0F -#define TWL6040_HSL_VOL_SHIFT 0 -#define TWL6040_HSR_VOL_MASK 0xF0 -#define TWL6040_HSR_VOL_SHIFT 4 -#define TWL6040_HF_VOL_MASK 0x1F -#define TWL6040_HF_VOL_SHIFT 0 - /* Shadow register used by the driver */ #define TWL6040_REG_SW_SHADOW 0x2F #define TWL6040_CACHEREGNUM (TWL6040_REG_SW_SHADOW + 1) @@ -64,18 +53,6 @@ /* TWL6040_REG_SW_SHADOW (0x2F) fields */ #define TWL6040_EAR_PATH_ENABLE 0x01 -struct twl6040_output { - u16 active; - u16 left_vol; - u16 right_vol; - u16 left_step; - u16 right_step; - unsigned int step_delay; - u16 ramp; - struct delayed_work work; - struct completion ramp_done; -}; - struct twl6040_jack_data { struct snd_soc_jack *jack; struct delayed_work work; @@ -100,8 +77,6 @@ struct twl6040_data { struct snd_soc_codec *codec; struct workqueue_struct *workqueue; struct mutex mutex; - struct twl6040_output headset; - struct twl6040_output handsfree; }; /* @@ -311,318 +286,6 @@ static void twl6040_restore_regs(struct snd_soc_codec *codec) } } -/* - * Ramp HS PGA volume to minimise pops at stream startup and shutdown. - */ -static inline int twl6040_hs_ramp_step(struct snd_soc_codec *codec, - unsigned int left_step, unsigned int right_step) -{ - - struct twl6040_data *priv = snd_soc_codec_get_drvdata(codec); - struct twl6040_output *headset = &priv->headset; - int left_complete = 0, right_complete = 0; - u8 reg, val; - - /* left channel */ - left_step = (left_step > 0xF) ? 0xF : left_step; - reg = twl6040_read_reg_cache(codec, TWL6040_REG_HSGAIN); - val = (~reg & TWL6040_HSL_VOL_MASK); - - if (headset->ramp == TWL6040_RAMP_UP) { - /* ramp step up */ - if (val < headset->left_vol) { - if (val + left_step > headset->left_vol) - val = headset->left_vol; - else - val += left_step; - - reg &= ~TWL6040_HSL_VOL_MASK; - twl6040_write(codec, TWL6040_REG_HSGAIN, - (reg | (~val & TWL6040_HSL_VOL_MASK))); - } else { - left_complete = 1; - } - } else if (headset->ramp == TWL6040_RAMP_DOWN) { - /* ramp step down */ - if (val > 0x0) { - if ((int)val - (int)left_step < 0) - val = 0; - else - val -= left_step; - - reg &= ~TWL6040_HSL_VOL_MASK; - twl6040_write(codec, TWL6040_REG_HSGAIN, reg | - (~val & TWL6040_HSL_VOL_MASK)); - } else { - left_complete = 1; - } - } - - /* right channel */ - right_step = (right_step > 0xF) ? 0xF : right_step; - reg = twl6040_read_reg_cache(codec, TWL6040_REG_HSGAIN); - val = (~reg & TWL6040_HSR_VOL_MASK) >> TWL6040_HSR_VOL_SHIFT; - - if (headset->ramp == TWL6040_RAMP_UP) { - /* ramp step up */ - if (val < headset->right_vol) { - if (val + right_step > headset->right_vol) - val = headset->right_vol; - else - val += right_step; - - reg &= ~TWL6040_HSR_VOL_MASK; - twl6040_write(codec, TWL6040_REG_HSGAIN, - (reg | (~val << TWL6040_HSR_VOL_SHIFT))); - } else { - right_complete = 1; - } - } else if (headset->ramp == TWL6040_RAMP_DOWN) { - /* ramp step down */ - if (val > 0x0) { - if ((int)val - (int)right_step < 0) - val = 0; - else - val -= right_step; - - reg &= ~TWL6040_HSR_VOL_MASK; - twl6040_write(codec, TWL6040_REG_HSGAIN, - reg | (~val << TWL6040_HSR_VOL_SHIFT)); - } else { - right_complete = 1; - } - } - - return left_complete & right_complete; -} - -/* - * Ramp HF PGA volume to minimise pops at stream startup and shutdown. - */ -static inline int twl6040_hf_ramp_step(struct snd_soc_codec *codec, - unsigned int left_step, unsigned int right_step) -{ - struct twl6040_data *priv = snd_soc_codec_get_drvdata(codec); - struct twl6040_output *handsfree = &priv->handsfree; - int left_complete = 0, right_complete = 0; - u16 reg, val; - - /* left channel */ - left_step = (left_step > 0x1D) ? 0x1D : left_step; - reg = twl6040_read_reg_cache(codec, TWL6040_REG_HFLGAIN); - reg = 0x1D - reg; - val = (reg & TWL6040_HF_VOL_MASK); - if (handsfree->ramp == TWL6040_RAMP_UP) { - /* ramp step up */ - if (val < handsfree->left_vol) { - if (val + left_step > handsfree->left_vol) - val = handsfree->left_vol; - else - val += left_step; - - reg &= ~TWL6040_HF_VOL_MASK; - twl6040_write(codec, TWL6040_REG_HFLGAIN, - reg | (0x1D - val)); - } else { - left_complete = 1; - } - } else if (handsfree->ramp == TWL6040_RAMP_DOWN) { - /* ramp step down */ - if (val > 0) { - if ((int)val - (int)left_step < 0) - val = 0; - else - val -= left_step; - - reg &= ~TWL6040_HF_VOL_MASK; - twl6040_write(codec, TWL6040_REG_HFLGAIN, - reg | (0x1D - val)); - } else { - left_complete = 1; - } - } - - /* right channel */ - right_step = (right_step > 0x1D) ? 0x1D : right_step; - reg = twl6040_read_reg_cache(codec, TWL6040_REG_HFRGAIN); - reg = 0x1D - reg; - val = (reg & TWL6040_HF_VOL_MASK); - if (handsfree->ramp == TWL6040_RAMP_UP) { - /* ramp step up */ - if (val < handsfree->right_vol) { - if (val + right_step > handsfree->right_vol) - val = handsfree->right_vol; - else - val += right_step; - - reg &= ~TWL6040_HF_VOL_MASK; - twl6040_write(codec, TWL6040_REG_HFRGAIN, - reg | (0x1D - val)); - } else { - right_complete = 1; - } - } else if (handsfree->ramp == TWL6040_RAMP_DOWN) { - /* ramp step down */ - if (val > 0) { - if ((int)val - (int)right_step < 0) - val = 0; - else - val -= right_step; - - reg &= ~TWL6040_HF_VOL_MASK; - twl6040_write(codec, TWL6040_REG_HFRGAIN, - reg | (0x1D - val)); - } - } - - return left_complete & right_complete; -} - -/* - * This work ramps both output PGAs at stream start/stop time to - * minimise pop associated with DAPM power switching. - */ -static void twl6040_pga_hs_work(struct work_struct *work) -{ - struct twl6040_data *priv = - container_of(work, struct twl6040_data, headset.work.work); - struct snd_soc_codec *codec = priv->codec; - struct twl6040_output *headset = &priv->headset; - int i, headset_complete; - - /* do we need to ramp at all ? */ - if (headset->ramp == TWL6040_RAMP_NONE) - return; - - /* HS PGA gain range: 0x0 - 0xf (0 - 15) */ - for (i = 0; i < 16; i++) { - headset_complete = twl6040_hs_ramp_step(codec, - headset->left_step, - headset->right_step); - - /* ramp finished ? */ - if (headset_complete) - break; - - schedule_timeout_interruptible( - msecs_to_jiffies(headset->step_delay)); - } - - if (headset->ramp == TWL6040_RAMP_DOWN) { - headset->active = 0; - complete(&headset->ramp_done); - } else { - headset->active = 1; - } - headset->ramp = TWL6040_RAMP_NONE; -} - -static void twl6040_pga_hf_work(struct work_struct *work) -{ - struct twl6040_data *priv = - container_of(work, struct twl6040_data, handsfree.work.work); - struct snd_soc_codec *codec = priv->codec; - struct twl6040_output *handsfree = &priv->handsfree; - int i, handsfree_complete; - - /* do we need to ramp at all ? */ - if (handsfree->ramp == TWL6040_RAMP_NONE) - return; - - /* - * HF PGA gain range: 0x00 - 0x1d (0 - 29) */ - for (i = 0; i < 30; i++) { - handsfree_complete = twl6040_hf_ramp_step(codec, - handsfree->left_step, - handsfree->right_step); - - /* ramp finished ? */ - if (handsfree_complete) - break; - - schedule_timeout_interruptible( - msecs_to_jiffies(handsfree->step_delay)); - } - - - if (handsfree->ramp == TWL6040_RAMP_DOWN) { - handsfree->active = 0; - complete(&handsfree->ramp_done); - } else - handsfree->active = 1; - handsfree->ramp = TWL6040_RAMP_NONE; -} - -static int out_drv_event(struct snd_soc_dapm_widget *w, - struct snd_kcontrol *kcontrol, int event) -{ - struct snd_soc_codec *codec = w->codec; - struct twl6040_data *priv = snd_soc_codec_get_drvdata(codec); - struct twl6040_output *out; - struct delayed_work *work; - - switch (w->shift) { - case 2: /* Headset output driver */ - out = &priv->headset; - work = &out->work; - /* - * Make sure, that we do not mess up variables for already - * executing work. - */ - cancel_delayed_work_sync(work); - - out->left_step = priv->hs_left_step; - out->right_step = priv->hs_right_step; - out->step_delay = 5; /* 5 ms between volume ramp steps */ - break; - case 4: /* Handsfree output driver */ - out = &priv->handsfree; - work = &out->work; - /* - * Make sure, that we do not mess up variables for already - * executing work. - */ - cancel_delayed_work_sync(work); - - out->left_step = priv->hf_left_step; - out->right_step = priv->hf_right_step; - out->step_delay = 5; /* 5 ms between volume ramp steps */ - break; - default: - return -1; - } - - switch (event) { - case SND_SOC_DAPM_POST_PMU: - if (out->active) - break; - - /* don't use volume ramp for power-up */ - out->ramp = TWL6040_RAMP_UP; - out->left_step = out->left_vol; - out->right_step = out->right_vol; - - queue_delayed_work(priv->workqueue, work, msecs_to_jiffies(1)); - break; - - case SND_SOC_DAPM_PRE_PMD: - if (!out->active) - break; - - /* use volume ramp for power-down */ - out->ramp = TWL6040_RAMP_DOWN; - INIT_COMPLETION(out->ramp_done); - - queue_delayed_work(priv->workqueue, work, msecs_to_jiffies(1)); - - wait_for_completion_timeout(&out->ramp_done, - msecs_to_jiffies(2000)); - break; - } - - return 0; -} - /* set headset dac and driver power mode */ static int headset_power_mode(struct snd_soc_codec *codec, int high_perf) { @@ -747,71 +410,6 @@ static irqreturn_t twl6040_audio_handler(int irq, void *data) return IRQ_HANDLED; } -static int twl6040_put_volsw(struct snd_kcontrol *kcontrol, - struct snd_ctl_elem_value *ucontrol) -{ - struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol); - struct twl6040_data *twl6040_priv = snd_soc_codec_get_drvdata(codec); - struct twl6040_output *out = NULL; - struct soc_mixer_control *mc = - (struct soc_mixer_control *)kcontrol->private_value; - int ret; - - /* For HS and HF we shadow the values and only actually write - * them out when active in order to ensure the amplifier comes on - * as quietly as possible. */ - switch (mc->reg) { - case TWL6040_REG_HSGAIN: - out = &twl6040_priv->headset; - break; - case TWL6040_REG_HFLGAIN: - out = &twl6040_priv->handsfree; - break; - default: - dev_warn(codec->dev, "%s: Unexpected register: 0x%02x\n", - __func__, mc->reg); - return -EINVAL; - } - - out->left_vol = ucontrol->value.integer.value[0]; - out->right_vol = ucontrol->value.integer.value[1]; - if (!out->active) - return 1; - - ret = snd_soc_put_volsw(kcontrol, ucontrol); - if (ret < 0) - return ret; - - return 1; -} - -static int twl6040_get_volsw(struct snd_kcontrol *kcontrol, - struct snd_ctl_elem_value *ucontrol) -{ - struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol); - struct twl6040_data *twl6040_priv = snd_soc_codec_get_drvdata(codec); - struct twl6040_output *out = &twl6040_priv->headset; - struct soc_mixer_control *mc = - (struct soc_mixer_control *)kcontrol->private_value; - - switch (mc->reg) { - case TWL6040_REG_HSGAIN: - out = &twl6040_priv->headset; - break; - case TWL6040_REG_HFLGAIN: - out = &twl6040_priv->handsfree; - break; - default: - dev_warn(codec->dev, "%s: Unexpected register: 0x%02x\n", - __func__, mc->reg); - return -EINVAL; - } - - ucontrol->value.integer.value[0] = out->left_vol; - ucontrol->value.integer.value[1] = out->right_vol; - return 0; -} - static int twl6040_soc_dapm_put_vibra_enum(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) { @@ -1076,12 +674,10 @@ static const struct snd_kcontrol_new twl6040_snd_controls[] = { TWL6040_REG_LINEGAIN, 0, 3, 7, 0, afm_amp_tlv), /* Playback gains */ - SOC_DOUBLE_EXT_TLV("Headset Playback Volume", - TWL6040_REG_HSGAIN, 0, 4, 0xF, 1, twl6040_get_volsw, - twl6040_put_volsw, hs_tlv), - SOC_DOUBLE_R_EXT_TLV("Handsfree Playback Volume", - TWL6040_REG_HFLGAIN, TWL6040_REG_HFRGAIN, 0, 0x1D, 1, - twl6040_get_volsw, twl6040_put_volsw, hf_tlv), + SOC_DOUBLE_TLV("Headset Playback Volume", + TWL6040_REG_HSGAIN, 0, 4, 0xF, 1, hs_tlv), + SOC_DOUBLE_R_TLV("Handsfree Playback Volume", + TWL6040_REG_HFLGAIN, TWL6040_REG_HFRGAIN, 0, 0x1D, 1, hf_tlv), SOC_SINGLE_TLV("Earphone Playback Volume", TWL6040_REG_EARCTL, 1, 0xF, 1, ep_tlv), @@ -1180,22 +776,14 @@ static const struct snd_soc_dapm_widget twl6040_dapm_widgets[] = { &auxr_switch_control), /* Analog playback drivers */ - SND_SOC_DAPM_OUT_DRV_E("HF Left Driver", - TWL6040_REG_HFLCTL, 4, 0, NULL, 0, - out_drv_event, - SND_SOC_DAPM_POST_PMU | SND_SOC_DAPM_PRE_PMD), - SND_SOC_DAPM_OUT_DRV_E("HF Right Driver", - TWL6040_REG_HFRCTL, 4, 0, NULL, 0, - out_drv_event, - SND_SOC_DAPM_POST_PMU | SND_SOC_DAPM_PRE_PMD), - SND_SOC_DAPM_OUT_DRV_E("HS Left Driver", - TWL6040_REG_HSLCTL, 2, 0, NULL, 0, - out_drv_event, - SND_SOC_DAPM_POST_PMU | SND_SOC_DAPM_PRE_PMD), - SND_SOC_DAPM_OUT_DRV_E("HS Right Driver", - TWL6040_REG_HSRCTL, 2, 0, NULL, 0, - out_drv_event, - SND_SOC_DAPM_POST_PMU | SND_SOC_DAPM_PRE_PMD), + SND_SOC_DAPM_OUT_DRV("HF Left Driver", + TWL6040_REG_HFLCTL, 4, 0, NULL, 0), + SND_SOC_DAPM_OUT_DRV("HF Right Driver", + TWL6040_REG_HFRCTL, 4, 0, NULL, 0), + SND_SOC_DAPM_OUT_DRV("HS Left Driver", + TWL6040_REG_HSLCTL, 2, 0, NULL, 0), + SND_SOC_DAPM_OUT_DRV("HS Right Driver", + TWL6040_REG_HSRCTL, 2, 0, NULL, 0), SND_SOC_DAPM_OUT_DRV_E("Earphone Driver", TWL6040_REG_EARCTL, 0, 0, NULL, 0, twl6040_ep_drv_event, @@ -1339,8 +927,7 @@ static int twl6040_set_bias_level(struct snd_soc_codec *codec, static int twl6040_startup(struct snd_pcm_substream *substream, struct snd_soc_dai *dai) { - struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct snd_soc_codec *codec = rtd->codec; + struct snd_soc_codec *codec = dai->codec; struct twl6040_data *priv = snd_soc_codec_get_drvdata(codec); snd_pcm_hw_constraint_list(substream->runtime, 0, @@ -1354,8 +941,7 @@ static int twl6040_hw_params(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *params, struct snd_soc_dai *dai) { - struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct snd_soc_codec *codec = rtd->codec; + struct snd_soc_codec *codec = dai->codec; struct twl6040_data *priv = snd_soc_codec_get_drvdata(codec); int rate; @@ -1391,8 +977,7 @@ static int twl6040_hw_params(struct snd_pcm_substream *substream, static int twl6040_prepare(struct snd_pcm_substream *substream, struct snd_soc_dai *dai) { - struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct snd_soc_codec *codec = rtd->codec; + struct snd_soc_codec *codec = dai->codec; struct twl6040 *twl6040 = codec->control_data; struct twl6040_data *priv = snd_soc_codec_get_drvdata(codec); int ret; @@ -1570,14 +1155,9 @@ static int twl6040_probe(struct snd_soc_codec *codec) } INIT_DELAYED_WORK(&priv->hs_jack.work, twl6040_accessory_work); - INIT_DELAYED_WORK(&priv->headset.work, twl6040_pga_hs_work); - INIT_DELAYED_WORK(&priv->handsfree.work, twl6040_pga_hf_work); mutex_init(&priv->mutex); - init_completion(&priv->headset.ramp_done); - init_completion(&priv->handsfree.ramp_done); - ret = request_threaded_irq(priv->plug_irq, NULL, twl6040_audio_handler, 0, "twl6040_irq_plug", codec); if (ret) { diff --git a/sound/soc/codecs/uda134x.c b/sound/soc/codecs/uda134x.c index 797b0dde2c6..6c3d43b8ee8 100644 --- a/sound/soc/codecs/uda134x.c +++ b/sound/soc/codecs/uda134x.c @@ -159,8 +159,7 @@ static int uda134x_mute(struct snd_soc_dai *dai, int mute) static int uda134x_startup(struct snd_pcm_substream *substream, struct snd_soc_dai *dai) { - struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct snd_soc_codec *codec =rtd->codec; + struct snd_soc_codec *codec = dai->codec; struct uda134x_priv *uda134x = snd_soc_codec_get_drvdata(codec); struct snd_pcm_runtime *master_runtime; @@ -191,8 +190,7 @@ static int uda134x_startup(struct snd_pcm_substream *substream, static void uda134x_shutdown(struct snd_pcm_substream *substream, struct snd_soc_dai *dai) { - struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct snd_soc_codec *codec = rtd->codec; + struct snd_soc_codec *codec = dai->codec; struct uda134x_priv *uda134x = snd_soc_codec_get_drvdata(codec); if (uda134x->master_substream == substream) diff --git a/sound/soc/codecs/uda1380.c b/sound/soc/codecs/uda1380.c index 4f1b23d7e40..2502214b84a 100644 --- a/sound/soc/codecs/uda1380.c +++ b/sound/soc/codecs/uda1380.c @@ -502,8 +502,7 @@ static int uda1380_set_dai_fmt_capture(struct snd_soc_dai *codec_dai, static int uda1380_trigger(struct snd_pcm_substream *substream, int cmd, struct snd_soc_dai *dai) { - struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct snd_soc_codec *codec = rtd->codec; + struct snd_soc_codec *codec = dai->codec; struct uda1380_priv *uda1380 = snd_soc_codec_get_drvdata(codec); int mixer = uda1380_read_reg_cache(codec, UDA1380_MIXER); @@ -528,8 +527,7 @@ static int uda1380_pcm_hw_params(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *params, struct snd_soc_dai *dai) { - struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct snd_soc_codec *codec = rtd->codec; + struct snd_soc_codec *codec = dai->codec; u16 clk = uda1380_read_reg_cache(codec, UDA1380_CLK); /* set WSPLL power and divider if running from this clock */ diff --git a/sound/soc/codecs/wl1273.c b/sound/soc/codecs/wl1273.c index 3d868dc4009..7b24d6d192e 100644 --- a/sound/soc/codecs/wl1273.c +++ b/sound/soc/codecs/wl1273.c @@ -293,8 +293,7 @@ static const struct snd_kcontrol_new wl1273_controls[] = { static int wl1273_startup(struct snd_pcm_substream *substream, struct snd_soc_dai *dai) { - struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct snd_soc_codec *codec = rtd->codec; + struct snd_soc_codec *codec = dai->codec; struct wl1273_priv *wl1273 = snd_soc_codec_get_drvdata(codec); switch (wl1273->mode) { @@ -329,8 +328,7 @@ static int wl1273_hw_params(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *params, struct snd_soc_dai *dai) { - struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct wl1273_priv *wl1273 = snd_soc_codec_get_drvdata(rtd->codec); + struct wl1273_priv *wl1273 = snd_soc_codec_get_drvdata(dai->codec); struct wl1273_core *core = wl1273->core; unsigned int rate, width, r; diff --git a/sound/soc/codecs/wm1250-ev1.c b/sound/soc/codecs/wm1250-ev1.c index aefb4f89be0..e0b51e9f8b1 100644 --- a/sound/soc/codecs/wm1250-ev1.c +++ b/sound/soc/codecs/wm1250-ev1.c @@ -79,22 +79,65 @@ static const struct snd_soc_dapm_route wm1250_ev1_dapm_routes[] = { { "WM1250 Output", NULL, "DAC" }, }; +static int wm1250_ev1_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params, + struct snd_soc_dai *dai) +{ + struct wm1250_priv *wm1250 = snd_soc_codec_get_drvdata(dai->codec); + + switch (params_rate(params)) { + case 8000: + gpio_set_value(wm1250->gpios[WM1250_EV1_GPIO_CLK_SEL0].gpio, + 1); + gpio_set_value(wm1250->gpios[WM1250_EV1_GPIO_CLK_SEL1].gpio, + 1); + break; + case 16000: + gpio_set_value(wm1250->gpios[WM1250_EV1_GPIO_CLK_SEL0].gpio, + 0); + gpio_set_value(wm1250->gpios[WM1250_EV1_GPIO_CLK_SEL1].gpio, + 1); + break; + case 32000: + gpio_set_value(wm1250->gpios[WM1250_EV1_GPIO_CLK_SEL0].gpio, + 1); + gpio_set_value(wm1250->gpios[WM1250_EV1_GPIO_CLK_SEL1].gpio, + 0); + break; + case 64000: + gpio_set_value(wm1250->gpios[WM1250_EV1_GPIO_CLK_SEL0].gpio, + 0); + gpio_set_value(wm1250->gpios[WM1250_EV1_GPIO_CLK_SEL1].gpio, + 0); + break; + default: + return -EINVAL; + } + + return 0; +} + +static const struct snd_soc_dai_ops wm1250_ev1_ops = { + .hw_params = wm1250_ev1_hw_params, +}; + static struct snd_soc_dai_driver wm1250_ev1_dai = { .name = "wm1250-ev1", .playback = { .stream_name = "Playback", .channels_min = 1, - .channels_max = 1, + .channels_max = 2, .rates = SNDRV_PCM_RATE_8000, .formats = SNDRV_PCM_FMTBIT_S16_LE, }, .capture = { .stream_name = "Capture", .channels_min = 1, - .channels_max = 1, + .channels_max = 2, .rates = SNDRV_PCM_RATE_8000, .formats = SNDRV_PCM_FMTBIT_S16_LE, }, + .ops = &wm1250_ev1_ops, }; static struct snd_soc_codec_driver soc_codec_dev_wm1250_ev1 = { @@ -215,23 +258,7 @@ static struct i2c_driver wm1250_ev1_i2c_driver = { .id_table = wm1250_ev1_i2c_id, }; -static int __init wm1250_ev1_modinit(void) -{ - int ret = 0; - - ret = i2c_add_driver(&wm1250_ev1_i2c_driver); - if (ret != 0) - pr_err("Failed to register WM1250-EV1 I2C driver: %d\n", ret); - - return ret; -} -module_init(wm1250_ev1_modinit); - -static void __exit wm1250_ev1_exit(void) -{ - i2c_del_driver(&wm1250_ev1_i2c_driver); -} -module_exit(wm1250_ev1_exit); +module_i2c_driver(wm1250_ev1_i2c_driver); MODULE_AUTHOR("Mark Brown <broonie@opensource.wolfsonmicro.com>"); MODULE_DESCRIPTION("WM1250-EV1 audio I/O module driver"); diff --git a/sound/soc/codecs/wm5100-tables.c b/sound/soc/codecs/wm5100-tables.c index 9a18fae6820..e167207a19c 100644 --- a/sound/soc/codecs/wm5100-tables.c +++ b/sound/soc/codecs/wm5100-tables.c @@ -32,7 +32,18 @@ bool wm5100_volatile_register(struct device *dev, unsigned int reg) case WM5100_MIC_DETECT_3: return 1; default: - return 0; + if ((reg >= WM5100_DSP1_PM_0 && reg <= WM5100_DSP1_PM_1535) || + (reg >= WM5100_DSP1_ZM_0 && reg <= WM5100_DSP1_ZM_2047) || + (reg >= WM5100_DSP1_DM_0 && reg <= WM5100_DSP1_DM_511) || + (reg >= WM5100_DSP2_PM_0 && reg <= WM5100_DSP2_PM_1535) || + (reg >= WM5100_DSP2_ZM_0 && reg <= WM5100_DSP2_ZM_2047) || + (reg >= WM5100_DSP2_DM_0 && reg <= WM5100_DSP2_DM_511) || + (reg >= WM5100_DSP3_PM_0 && reg <= WM5100_DSP3_PM_1535) || + (reg >= WM5100_DSP3_ZM_0 && reg <= WM5100_DSP3_ZM_2047) || + (reg >= WM5100_DSP3_DM_0 && reg <= WM5100_DSP3_DM_511)) + return 1; + else + return 0; } } @@ -697,9 +708,110 @@ bool wm5100_readable_register(struct device *dev, unsigned int reg) case WM5100_HPLPF3_2: case WM5100_HPLPF4_1: case WM5100_HPLPF4_2: + case WM5100_DSP1_CONTROL_1: + case WM5100_DSP1_CONTROL_2: + case WM5100_DSP1_CONTROL_3: + case WM5100_DSP1_CONTROL_4: + case WM5100_DSP1_CONTROL_5: + case WM5100_DSP1_CONTROL_6: + case WM5100_DSP1_CONTROL_7: + case WM5100_DSP1_CONTROL_8: + case WM5100_DSP1_CONTROL_9: + case WM5100_DSP1_CONTROL_10: + case WM5100_DSP1_CONTROL_11: + case WM5100_DSP1_CONTROL_12: + case WM5100_DSP1_CONTROL_13: + case WM5100_DSP1_CONTROL_14: + case WM5100_DSP1_CONTROL_15: + case WM5100_DSP1_CONTROL_16: + case WM5100_DSP1_CONTROL_17: + case WM5100_DSP1_CONTROL_18: + case WM5100_DSP1_CONTROL_19: + case WM5100_DSP1_CONTROL_20: + case WM5100_DSP1_CONTROL_21: + case WM5100_DSP1_CONTROL_22: + case WM5100_DSP1_CONTROL_23: + case WM5100_DSP1_CONTROL_24: + case WM5100_DSP1_CONTROL_25: + case WM5100_DSP1_CONTROL_26: + case WM5100_DSP1_CONTROL_27: + case WM5100_DSP1_CONTROL_28: + case WM5100_DSP1_CONTROL_29: + case WM5100_DSP1_CONTROL_30: + case WM5100_DSP2_CONTROL_1: + case WM5100_DSP2_CONTROL_2: + case WM5100_DSP2_CONTROL_3: + case WM5100_DSP2_CONTROL_4: + case WM5100_DSP2_CONTROL_5: + case WM5100_DSP2_CONTROL_6: + case WM5100_DSP2_CONTROL_7: + case WM5100_DSP2_CONTROL_8: + case WM5100_DSP2_CONTROL_9: + case WM5100_DSP2_CONTROL_10: + case WM5100_DSP2_CONTROL_11: + case WM5100_DSP2_CONTROL_12: + case WM5100_DSP2_CONTROL_13: + case WM5100_DSP2_CONTROL_14: + case WM5100_DSP2_CONTROL_15: + case WM5100_DSP2_CONTROL_16: + case WM5100_DSP2_CONTROL_17: + case WM5100_DSP2_CONTROL_18: + case WM5100_DSP2_CONTROL_19: + case WM5100_DSP2_CONTROL_20: + case WM5100_DSP2_CONTROL_21: + case WM5100_DSP2_CONTROL_22: + case WM5100_DSP2_CONTROL_23: + case WM5100_DSP2_CONTROL_24: + case WM5100_DSP2_CONTROL_25: + case WM5100_DSP2_CONTROL_26: + case WM5100_DSP2_CONTROL_27: + case WM5100_DSP2_CONTROL_28: + case WM5100_DSP2_CONTROL_29: + case WM5100_DSP2_CONTROL_30: + case WM5100_DSP3_CONTROL_1: + case WM5100_DSP3_CONTROL_2: + case WM5100_DSP3_CONTROL_3: + case WM5100_DSP3_CONTROL_4: + case WM5100_DSP3_CONTROL_5: + case WM5100_DSP3_CONTROL_6: + case WM5100_DSP3_CONTROL_7: + case WM5100_DSP3_CONTROL_8: + case WM5100_DSP3_CONTROL_9: + case WM5100_DSP3_CONTROL_10: + case WM5100_DSP3_CONTROL_11: + case WM5100_DSP3_CONTROL_12: + case WM5100_DSP3_CONTROL_13: + case WM5100_DSP3_CONTROL_14: + case WM5100_DSP3_CONTROL_15: + case WM5100_DSP3_CONTROL_16: + case WM5100_DSP3_CONTROL_17: + case WM5100_DSP3_CONTROL_18: + case WM5100_DSP3_CONTROL_19: + case WM5100_DSP3_CONTROL_20: + case WM5100_DSP3_CONTROL_21: + case WM5100_DSP3_CONTROL_22: + case WM5100_DSP3_CONTROL_23: + case WM5100_DSP3_CONTROL_24: + case WM5100_DSP3_CONTROL_25: + case WM5100_DSP3_CONTROL_26: + case WM5100_DSP3_CONTROL_27: + case WM5100_DSP3_CONTROL_28: + case WM5100_DSP3_CONTROL_29: + case WM5100_DSP3_CONTROL_30: return 1; default: - return 0; + if ((reg >= WM5100_DSP1_PM_0 && reg <= WM5100_DSP1_PM_1535) || + (reg >= WM5100_DSP1_ZM_0 && reg <= WM5100_DSP1_ZM_2047) || + (reg >= WM5100_DSP1_DM_0 && reg <= WM5100_DSP1_DM_511) || + (reg >= WM5100_DSP2_PM_0 && reg <= WM5100_DSP2_PM_1535) || + (reg >= WM5100_DSP2_ZM_0 && reg <= WM5100_DSP2_ZM_2047) || + (reg >= WM5100_DSP2_DM_0 && reg <= WM5100_DSP2_DM_511) || + (reg >= WM5100_DSP3_PM_0 && reg <= WM5100_DSP3_PM_1535) || + (reg >= WM5100_DSP3_ZM_0 && reg <= WM5100_DSP3_ZM_2047) || + (reg >= WM5100_DSP3_DM_0 && reg <= WM5100_DSP3_DM_511)) + return 1; + else + return 0; } } @@ -1361,4 +1473,13 @@ struct reg_default wm5100_reg_defaults[WM5100_REGISTER_COUNT] = { { 0x0EC9, 0x0000 }, /* R3785 - HPLPF3_2 */ { 0x0ECC, 0x0000 }, /* R3788 - HPLPF4_1 */ { 0x0ECD, 0x0000 }, /* R3789 - HPLPF4_2 */ + { 0x0F02, 0x0000 }, /* R3842 - DSP1 Control 2 */ + { 0x0F03, 0x0000 }, /* R3843 - DSP1 Control 3 */ + { 0x0F04, 0x0000 }, /* R3844 - DSP1 Control 4 */ + { 0x1002, 0x0000 }, /* R4098 - DSP2 Control 2 */ + { 0x1003, 0x0000 }, /* R4099 - DSP2 Control 3 */ + { 0x1004, 0x0000 }, /* R4100 - DSP2 Control 4 */ + { 0x1102, 0x0000 }, /* R4354 - DSP3 Control 2 */ + { 0x1103, 0x0000 }, /* R4355 - DSP3 Control 3 */ + { 0x1104, 0x0000 }, /* R4356 - DSP3 Control 4 */ }; diff --git a/sound/soc/codecs/wm5100.c b/sound/soc/codecs/wm5100.c index b9c185ce64e..cb6d5372103 100644 --- a/sound/soc/codecs/wm5100.c +++ b/sound/soc/codecs/wm5100.c @@ -1265,29 +1265,12 @@ static const __devinitdata struct reg_default wm5100_reva_patches[] = { { WM5100_AUDIO_IF_3_19, 1 }, }; -static int wm5100_dai_to_base(struct snd_soc_dai *dai) -{ - switch (dai->id) { - case 0: - return WM5100_AUDIO_IF_1_1 - 1; - case 1: - return WM5100_AUDIO_IF_2_1 - 1; - case 2: - return WM5100_AUDIO_IF_3_1 - 1; - default: - BUG(); - return -EINVAL; - } -} - static int wm5100_set_fmt(struct snd_soc_dai *dai, unsigned int fmt) { struct snd_soc_codec *codec = dai->codec; int lrclk, bclk, mask, base; - base = wm5100_dai_to_base(dai); - if (base < 0) - return base; + base = dai->driver->base; lrclk = 0; bclk = 0; @@ -1414,9 +1397,7 @@ static int wm5100_hw_params(struct snd_pcm_substream *substream, int i, base, bclk, aif_rate, lrclk, wl, fl, sr; int *bclk_rates; - base = wm5100_dai_to_base(dai); - if (base < 0) - return base; + base = dai->driver->base; /* Data sizes if not using TDM */ wl = snd_pcm_format_width(params_format(params)); @@ -1897,6 +1878,7 @@ static int wm5100_set_fll(struct snd_soc_codec *codec, int fll_id, int source, static struct snd_soc_dai_driver wm5100_dai[] = { { .name = "wm5100-aif1", + .base = WM5100_AUDIO_IF_1_1 - 1, .playback = { .stream_name = "AIF1 Playback", .channels_min = 2, @@ -1916,6 +1898,7 @@ static struct snd_soc_dai_driver wm5100_dai[] = { { .name = "wm5100-aif2", .id = 1, + .base = WM5100_AUDIO_IF_2_1 - 1, .playback = { .stream_name = "AIF2 Playback", .channels_min = 2, @@ -1935,6 +1918,7 @@ static struct snd_soc_dai_driver wm5100_dai[] = { { .name = "wm5100-aif3", .id = 2, + .base = WM5100_AUDIO_IF_3_1 - 1, .playback = { .stream_name = "AIF3 Playback", .channels_min = 2, @@ -2454,7 +2438,7 @@ static __devinit int wm5100_i2c_probe(struct i2c_client *i2c, wm5100->dev = &i2c->dev; - wm5100->regmap = regmap_init_i2c(i2c, &wm5100_regmap); + wm5100->regmap = devm_regmap_init_i2c(i2c, &wm5100_regmap); if (IS_ERR(wm5100->regmap)) { ret = PTR_ERR(wm5100->regmap); dev_err(&i2c->dev, "Failed to allocate register map: %d\n", @@ -2479,7 +2463,7 @@ static __devinit int wm5100_i2c_probe(struct i2c_client *i2c, if (ret != 0) { dev_err(&i2c->dev, "Failed to request core supplies: %d\n", ret); - goto err_regmap; + goto err; } ret = regulator_bulk_enable(ARRAY_SIZE(wm5100->core_supplies), @@ -2487,7 +2471,7 @@ static __devinit int wm5100_i2c_probe(struct i2c_client *i2c, if (ret != 0) { dev_err(&i2c->dev, "Failed to enable core supplies: %d\n", ret); - goto err_regmap; + goto err; } if (wm5100->pdata.ldo_ena) { @@ -2660,8 +2644,6 @@ err_ldo: err_enable: regulator_bulk_disable(ARRAY_SIZE(wm5100->core_supplies), wm5100->core_supplies); -err_regmap: - regmap_exit(wm5100->regmap); err: return ret; } @@ -2682,7 +2664,6 @@ static __devexit int wm5100_i2c_remove(struct i2c_client *i2c) gpio_set_value_cansleep(wm5100->pdata.ldo_ena, 0); gpio_free(wm5100->pdata.ldo_ena); } - regmap_exit(wm5100->regmap); return 0; } @@ -2749,17 +2730,7 @@ static struct i2c_driver wm5100_i2c_driver = { .id_table = wm5100_i2c_id, }; -static int __init wm5100_modinit(void) -{ - return i2c_add_driver(&wm5100_i2c_driver); -} -module_init(wm5100_modinit); - -static void __exit wm5100_exit(void) -{ - i2c_del_driver(&wm5100_i2c_driver); -} -module_exit(wm5100_exit); +module_i2c_driver(wm5100_i2c_driver); MODULE_DESCRIPTION("ASoC WM5100 driver"); MODULE_AUTHOR("Mark Brown <broonie@opensource.wolfsonmicro.com>"); diff --git a/sound/soc/codecs/wm5100.h b/sound/soc/codecs/wm5100.h index 25cb6016f9d..935a9b7fb27 100644 --- a/sound/soc/codecs/wm5100.h +++ b/sound/soc/codecs/wm5100.h @@ -709,6 +709,96 @@ int wm5100_detect(struct snd_soc_codec *codec, struct snd_soc_jack *jack); #define WM5100_HPLPF3_2 0xEC9 #define WM5100_HPLPF4_1 0xECC #define WM5100_HPLPF4_2 0xECD +#define WM5100_DSP1_CONTROL_1 0xF00 +#define WM5100_DSP1_CONTROL_2 0xF02 +#define WM5100_DSP1_CONTROL_3 0xF03 +#define WM5100_DSP1_CONTROL_4 0xF04 +#define WM5100_DSP1_CONTROL_5 0xF06 +#define WM5100_DSP1_CONTROL_6 0xF07 +#define WM5100_DSP1_CONTROL_7 0xF08 +#define WM5100_DSP1_CONTROL_8 0xF09 +#define WM5100_DSP1_CONTROL_9 0xF0A +#define WM5100_DSP1_CONTROL_10 0xF0B +#define WM5100_DSP1_CONTROL_11 0xF0C +#define WM5100_DSP1_CONTROL_12 0xF0D +#define WM5100_DSP1_CONTROL_13 0xF0F +#define WM5100_DSP1_CONTROL_14 0xF10 +#define WM5100_DSP1_CONTROL_15 0xF11 +#define WM5100_DSP1_CONTROL_16 0xF12 +#define WM5100_DSP1_CONTROL_17 0xF13 +#define WM5100_DSP1_CONTROL_18 0xF14 +#define WM5100_DSP1_CONTROL_19 0xF16 +#define WM5100_DSP1_CONTROL_20 0xF17 +#define WM5100_DSP1_CONTROL_21 0xF18 +#define WM5100_DSP1_CONTROL_22 0xF1A +#define WM5100_DSP1_CONTROL_23 0xF1B +#define WM5100_DSP1_CONTROL_24 0xF1C +#define WM5100_DSP1_CONTROL_25 0xF1E +#define WM5100_DSP1_CONTROL_26 0xF20 +#define WM5100_DSP1_CONTROL_27 0xF21 +#define WM5100_DSP1_CONTROL_28 0xF22 +#define WM5100_DSP1_CONTROL_29 0xF23 +#define WM5100_DSP1_CONTROL_30 0xF24 +#define WM5100_DSP2_CONTROL_1 0x1000 +#define WM5100_DSP2_CONTROL_2 0x1002 +#define WM5100_DSP2_CONTROL_3 0x1003 +#define WM5100_DSP2_CONTROL_4 0x1004 +#define WM5100_DSP2_CONTROL_5 0x1006 +#define WM5100_DSP2_CONTROL_6 0x1007 +#define WM5100_DSP2_CONTROL_7 0x1008 +#define WM5100_DSP2_CONTROL_8 0x1009 +#define WM5100_DSP2_CONTROL_9 0x100A +#define WM5100_DSP2_CONTROL_10 0x100B +#define WM5100_DSP2_CONTROL_11 0x100C +#define WM5100_DSP2_CONTROL_12 0x100D +#define WM5100_DSP2_CONTROL_13 0x100F +#define WM5100_DSP2_CONTROL_14 0x1010 +#define WM5100_DSP2_CONTROL_15 0x1011 +#define WM5100_DSP2_CONTROL_16 0x1012 +#define WM5100_DSP2_CONTROL_17 0x1013 +#define WM5100_DSP2_CONTROL_18 0x1014 +#define WM5100_DSP2_CONTROL_19 0x1016 +#define WM5100_DSP2_CONTROL_20 0x1017 +#define WM5100_DSP2_CONTROL_21 0x1018 +#define WM5100_DSP2_CONTROL_22 0x101A +#define WM5100_DSP2_CONTROL_23 0x101B +#define WM5100_DSP2_CONTROL_24 0x101C +#define WM5100_DSP2_CONTROL_25 0x101E +#define WM5100_DSP2_CONTROL_26 0x1020 +#define WM5100_DSP2_CONTROL_27 0x1021 +#define WM5100_DSP2_CONTROL_28 0x1022 +#define WM5100_DSP2_CONTROL_29 0x1023 +#define WM5100_DSP2_CONTROL_30 0x1024 +#define WM5100_DSP3_CONTROL_1 0x1100 +#define WM5100_DSP3_CONTROL_2 0x1102 +#define WM5100_DSP3_CONTROL_3 0x1103 +#define WM5100_DSP3_CONTROL_4 0x1104 +#define WM5100_DSP3_CONTROL_5 0x1106 +#define WM5100_DSP3_CONTROL_6 0x1107 +#define WM5100_DSP3_CONTROL_7 0x1108 +#define WM5100_DSP3_CONTROL_8 0x1109 +#define WM5100_DSP3_CONTROL_9 0x110A +#define WM5100_DSP3_CONTROL_10 0x110B +#define WM5100_DSP3_CONTROL_11 0x110C +#define WM5100_DSP3_CONTROL_12 0x110D +#define WM5100_DSP3_CONTROL_13 0x110F +#define WM5100_DSP3_CONTROL_14 0x1110 +#define WM5100_DSP3_CONTROL_15 0x1111 +#define WM5100_DSP3_CONTROL_16 0x1112 +#define WM5100_DSP3_CONTROL_17 0x1113 +#define WM5100_DSP3_CONTROL_18 0x1114 +#define WM5100_DSP3_CONTROL_19 0x1116 +#define WM5100_DSP3_CONTROL_20 0x1117 +#define WM5100_DSP3_CONTROL_21 0x1118 +#define WM5100_DSP3_CONTROL_22 0x111A +#define WM5100_DSP3_CONTROL_23 0x111B +#define WM5100_DSP3_CONTROL_24 0x111C +#define WM5100_DSP3_CONTROL_25 0x111E +#define WM5100_DSP3_CONTROL_26 0x1120 +#define WM5100_DSP3_CONTROL_27 0x1121 +#define WM5100_DSP3_CONTROL_28 0x1122 +#define WM5100_DSP3_CONTROL_29 0x1123 +#define WM5100_DSP3_CONTROL_30 0x1124 #define WM5100_DSP1_DM_0 0x4000 #define WM5100_DSP1_DM_1 0x4001 #define WM5100_DSP1_DM_2 0x4002 @@ -4561,6 +4651,75 @@ int wm5100_detect(struct snd_soc_codec *codec, struct snd_soc_jack *jack); #define WM5100_LHPF4_COEFF_WIDTH 16 /* LHPF4_COEFF - [15:0] */ /* + * R4132 (0x1024) - DSP2 Control 30 + */ +#define WM5100_DSP2_RATE_MASK 0xC000 /* DSP2_RATE - [15:14] */ +#define WM5100_DSP2_RATE_SHIFT 14 /* DSP2_RATE - [15:14] */ +#define WM5100_DSP2_RATE_WIDTH 2 /* DSP2_RATE - [15:14] */ +#define WM5100_DSP2_DBG_CLK_ENA 0x0008 /* DSP2_DBG_CLK_ENA */ +#define WM5100_DSP2_DBG_CLK_ENA_MASK 0x0008 /* DSP2_DBG_CLK_ENA */ +#define WM5100_DSP2_DBG_CLK_ENA_SHIFT 3 /* DSP2_DBG_CLK_ENA */ +#define WM5100_DSP2_DBG_CLK_ENA_WIDTH 1 /* DSP2_DBG_CLK_ENA */ +#define WM5100_DSP2_SYS_ENA 0x0004 /* DSP2_SYS_ENA */ +#define WM5100_DSP2_SYS_ENA_MASK 0x0004 /* DSP2_SYS_ENA */ +#define WM5100_DSP2_SYS_ENA_SHIFT 2 /* DSP2_SYS_ENA */ +#define WM5100_DSP2_SYS_ENA_WIDTH 1 /* DSP2_SYS_ENA */ +#define WM5100_DSP2_CORE_ENA 0x0002 /* DSP2_CORE_ENA */ +#define WM5100_DSP2_CORE_ENA_MASK 0x0002 /* DSP2_CORE_ENA */ +#define WM5100_DSP2_CORE_ENA_SHIFT 1 /* DSP2_CORE_ENA */ +#define WM5100_DSP2_CORE_ENA_WIDTH 1 /* DSP2_CORE_ENA */ +#define WM5100_DSP2_START 0x0001 /* DSP2_START */ +#define WM5100_DSP2_START_MASK 0x0001 /* DSP2_START */ +#define WM5100_DSP2_START_SHIFT 0 /* DSP2_START */ +#define WM5100_DSP2_START_WIDTH 1 /* DSP2_START */ + +/* + * R3876 (0xF24) - DSP1 Control 30 + */ +#define WM5100_DSP1_RATE_MASK 0xC000 /* DSP1_RATE - [15:14] */ +#define WM5100_DSP1_RATE_SHIFT 14 /* DSP1_RATE - [15:14] */ +#define WM5100_DSP1_RATE_WIDTH 2 /* DSP1_RATE - [15:14] */ +#define WM5100_DSP1_DBG_CLK_ENA 0x0008 /* DSP1_DBG_CLK_ENA */ +#define WM5100_DSP1_DBG_CLK_ENA_MASK 0x0008 /* DSP1_DBG_CLK_ENA */ +#define WM5100_DSP1_DBG_CLK_ENA_SHIFT 3 /* DSP1_DBG_CLK_ENA */ +#define WM5100_DSP1_DBG_CLK_ENA_WIDTH 1 /* DSP1_DBG_CLK_ENA */ +#define WM5100_DSP1_SYS_ENA 0x0004 /* DSP1_SYS_ENA */ +#define WM5100_DSP1_SYS_ENA_MASK 0x0004 /* DSP1_SYS_ENA */ +#define WM5100_DSP1_SYS_ENA_SHIFT 2 /* DSP1_SYS_ENA */ +#define WM5100_DSP1_SYS_ENA_WIDTH 1 /* DSP1_SYS_ENA */ +#define WM5100_DSP1_CORE_ENA 0x0002 /* DSP1_CORE_ENA */ +#define WM5100_DSP1_CORE_ENA_MASK 0x0002 /* DSP1_CORE_ENA */ +#define WM5100_DSP1_CORE_ENA_SHIFT 1 /* DSP1_CORE_ENA */ +#define WM5100_DSP1_CORE_ENA_WIDTH 1 /* DSP1_CORE_ENA */ +#define WM5100_DSP1_START 0x0001 /* DSP1_START */ +#define WM5100_DSP1_START_MASK 0x0001 /* DSP1_START */ +#define WM5100_DSP1_START_SHIFT 0 /* DSP1_START */ +#define WM5100_DSP1_START_WIDTH 1 /* DSP1_START */ + +/* + * R4388 (0x1124) - DSP3 Control 30 + */ +#define WM5100_DSP3_RATE_MASK 0xC000 /* DSP3_RATE - [15:14] */ +#define WM5100_DSP3_RATE_SHIFT 14 /* DSP3_RATE - [15:14] */ +#define WM5100_DSP3_RATE_WIDTH 2 /* DSP3_RATE - [15:14] */ +#define WM5100_DSP3_DBG_CLK_ENA 0x0008 /* DSP3_DBG_CLK_ENA */ +#define WM5100_DSP3_DBG_CLK_ENA_MASK 0x0008 /* DSP3_DBG_CLK_ENA */ +#define WM5100_DSP3_DBG_CLK_ENA_SHIFT 3 /* DSP3_DBG_CLK_ENA */ +#define WM5100_DSP3_DBG_CLK_ENA_WIDTH 1 /* DSP3_DBG_CLK_ENA */ +#define WM5100_DSP3_SYS_ENA 0x0004 /* DSP3_SYS_ENA */ +#define WM5100_DSP3_SYS_ENA_MASK 0x0004 /* DSP3_SYS_ENA */ +#define WM5100_DSP3_SYS_ENA_SHIFT 2 /* DSP3_SYS_ENA */ +#define WM5100_DSP3_SYS_ENA_WIDTH 1 /* DSP3_SYS_ENA */ +#define WM5100_DSP3_CORE_ENA 0x0002 /* DSP3_CORE_ENA */ +#define WM5100_DSP3_CORE_ENA_MASK 0x0002 /* DSP3_CORE_ENA */ +#define WM5100_DSP3_CORE_ENA_SHIFT 1 /* DSP3_CORE_ENA */ +#define WM5100_DSP3_CORE_ENA_WIDTH 1 /* DSP3_CORE_ENA */ +#define WM5100_DSP3_START 0x0001 /* DSP3_START */ +#define WM5100_DSP3_START_MASK 0x0001 /* DSP3_START */ +#define WM5100_DSP3_START_SHIFT 0 /* DSP3_START */ +#define WM5100_DSP3_START_WIDTH 1 /* DSP3_START */ + +/* * R16384 (0x4000) - DSP1 DM 0 */ #define WM5100_DSP1_DM_START_1_MASK 0x00FF /* DSP1_DM_START - [7:0] */ diff --git a/sound/soc/codecs/wm8350.c b/sound/soc/codecs/wm8350.c index aa12c6b6bee..555ee146ae0 100644 --- a/sound/soc/codecs/wm8350.c +++ b/sound/soc/codecs/wm8350.c @@ -71,13 +71,6 @@ struct wm8350_data { int fll_freq_in; }; -static unsigned int wm8350_codec_cache_read(struct snd_soc_codec *codec, - unsigned int reg) -{ - struct wm8350 *wm8350 = codec->control_data; - return wm8350->reg_cache[reg]; -} - static unsigned int wm8350_codec_read(struct snd_soc_codec *codec, unsigned int reg) { @@ -99,7 +92,7 @@ static inline int wm8350_out1_ramp_step(struct snd_soc_codec *codec) { struct wm8350_data *wm8350_data = snd_soc_codec_get_drvdata(codec); struct wm8350_output *out1 = &wm8350_data->out1; - struct wm8350 *wm8350 = codec->control_data; + struct wm8350 *wm8350 = wm8350_data->wm8350; int left_complete = 0, right_complete = 0; u16 reg, val; @@ -165,7 +158,7 @@ static inline int wm8350_out2_ramp_step(struct snd_soc_codec *codec) { struct wm8350_data *wm8350_data = snd_soc_codec_get_drvdata(codec); struct wm8350_output *out2 = &wm8350_data->out2; - struct wm8350 *wm8350 = codec->control_data; + struct wm8350 *wm8350 = wm8350_data->wm8350; int left_complete = 0, right_complete = 0; u16 reg, val; @@ -360,8 +353,8 @@ static int wm8350_put_volsw_2r_vu(struct snd_kcontrol *kcontrol, return ret; /* now hit the volume update bits (always bit 8) */ - val = wm8350_codec_read(codec, reg); - wm8350_codec_write(codec, reg, val | WM8350_OUT1_VU); + val = snd_soc_read(codec, reg); + snd_soc_write(codec, reg, val | WM8350_OUT1_VU); return 1; } @@ -781,7 +774,8 @@ static int wm8350_set_dai_sysclk(struct snd_soc_dai *codec_dai, int clk_id, unsigned int freq, int dir) { struct snd_soc_codec *codec = codec_dai->codec; - struct wm8350 *wm8350 = codec->control_data; + struct wm8350_data *wm8350_data = snd_soc_codec_get_drvdata(codec); + struct wm8350 *wm8350 = wm8350_data->wm8350; u16 fll_4; switch (clk_id) { @@ -795,9 +789,9 @@ static int wm8350_set_dai_sysclk(struct snd_soc_dai *codec_dai, case WM8350_MCLK_SEL_PLL_32K: wm8350_set_bits(wm8350, WM8350_CLOCK_CONTROL_1, WM8350_MCLK_SEL); - fll_4 = wm8350_codec_read(codec, WM8350_FLL_CONTROL_4) & + fll_4 = snd_soc_read(codec, WM8350_FLL_CONTROL_4) & ~WM8350_FLL_CLK_SRC_MASK; - wm8350_codec_write(codec, WM8350_FLL_CONTROL_4, fll_4 | clk_id); + snd_soc_write(codec, WM8350_FLL_CONTROL_4, fll_4 | clk_id); break; } @@ -819,39 +813,39 @@ static int wm8350_set_clkdiv(struct snd_soc_dai *codec_dai, int div_id, int div) switch (div_id) { case WM8350_ADC_CLKDIV: - val = wm8350_codec_read(codec, WM8350_ADC_DIVIDER) & + val = snd_soc_read(codec, WM8350_ADC_DIVIDER) & ~WM8350_ADC_CLKDIV_MASK; - wm8350_codec_write(codec, WM8350_ADC_DIVIDER, val | div); + snd_soc_write(codec, WM8350_ADC_DIVIDER, val | div); break; case WM8350_DAC_CLKDIV: - val = wm8350_codec_read(codec, WM8350_DAC_CLOCK_CONTROL) & + val = snd_soc_read(codec, WM8350_DAC_CLOCK_CONTROL) & ~WM8350_DAC_CLKDIV_MASK; - wm8350_codec_write(codec, WM8350_DAC_CLOCK_CONTROL, val | div); + snd_soc_write(codec, WM8350_DAC_CLOCK_CONTROL, val | div); break; case WM8350_BCLK_CLKDIV: - val = wm8350_codec_read(codec, WM8350_CLOCK_CONTROL_1) & + val = snd_soc_read(codec, WM8350_CLOCK_CONTROL_1) & ~WM8350_BCLK_DIV_MASK; - wm8350_codec_write(codec, WM8350_CLOCK_CONTROL_1, val | div); + snd_soc_write(codec, WM8350_CLOCK_CONTROL_1, val | div); break; case WM8350_OPCLK_CLKDIV: - val = wm8350_codec_read(codec, WM8350_CLOCK_CONTROL_1) & + val = snd_soc_read(codec, WM8350_CLOCK_CONTROL_1) & ~WM8350_OPCLK_DIV_MASK; - wm8350_codec_write(codec, WM8350_CLOCK_CONTROL_1, val | div); + snd_soc_write(codec, WM8350_CLOCK_CONTROL_1, val | div); break; case WM8350_SYS_CLKDIV: - val = wm8350_codec_read(codec, WM8350_CLOCK_CONTROL_1) & + val = snd_soc_read(codec, WM8350_CLOCK_CONTROL_1) & ~WM8350_MCLK_DIV_MASK; - wm8350_codec_write(codec, WM8350_CLOCK_CONTROL_1, val | div); + snd_soc_write(codec, WM8350_CLOCK_CONTROL_1, val | div); break; case WM8350_DACLR_CLKDIV: - val = wm8350_codec_read(codec, WM8350_DAC_LR_RATE) & + val = snd_soc_read(codec, WM8350_DAC_LR_RATE) & ~WM8350_DACLRC_RATE_MASK; - wm8350_codec_write(codec, WM8350_DAC_LR_RATE, val | div); + snd_soc_write(codec, WM8350_DAC_LR_RATE, val | div); break; case WM8350_ADCLR_CLKDIV: - val = wm8350_codec_read(codec, WM8350_ADC_LR_RATE) & + val = snd_soc_read(codec, WM8350_ADC_LR_RATE) & ~WM8350_ADCLRC_RATE_MASK; - wm8350_codec_write(codec, WM8350_ADC_LR_RATE, val | div); + snd_soc_write(codec, WM8350_ADC_LR_RATE, val | div); break; default: return -EINVAL; @@ -863,13 +857,13 @@ static int wm8350_set_clkdiv(struct snd_soc_dai *codec_dai, int div_id, int div) static int wm8350_set_dai_fmt(struct snd_soc_dai *codec_dai, unsigned int fmt) { struct snd_soc_codec *codec = codec_dai->codec; - u16 iface = wm8350_codec_read(codec, WM8350_AI_FORMATING) & + u16 iface = snd_soc_read(codec, WM8350_AI_FORMATING) & ~(WM8350_AIF_BCLK_INV | WM8350_AIF_LRCLK_INV | WM8350_AIF_FMT_MASK); - u16 master = wm8350_codec_read(codec, WM8350_AI_DAC_CONTROL) & + u16 master = snd_soc_read(codec, WM8350_AI_DAC_CONTROL) & ~WM8350_BCLK_MSTR; - u16 dac_lrc = wm8350_codec_read(codec, WM8350_DAC_LR_RATE) & + u16 dac_lrc = snd_soc_read(codec, WM8350_DAC_LR_RATE) & ~WM8350_DACLRC_ENA; - u16 adc_lrc = wm8350_codec_read(codec, WM8350_ADC_LR_RATE) & + u16 adc_lrc = snd_soc_read(codec, WM8350_ADC_LR_RATE) & ~WM8350_ADCLRC_ENA; /* set master/slave audio interface */ @@ -922,42 +916,10 @@ static int wm8350_set_dai_fmt(struct snd_soc_dai *codec_dai, unsigned int fmt) return -EINVAL; } - wm8350_codec_write(codec, WM8350_AI_FORMATING, iface); - wm8350_codec_write(codec, WM8350_AI_DAC_CONTROL, master); - wm8350_codec_write(codec, WM8350_DAC_LR_RATE, dac_lrc); - wm8350_codec_write(codec, WM8350_ADC_LR_RATE, adc_lrc); - return 0; -} - -static int wm8350_pcm_trigger(struct snd_pcm_substream *substream, - int cmd, struct snd_soc_dai *codec_dai) -{ - struct snd_soc_codec *codec = codec_dai->codec; - int master = wm8350_codec_cache_read(codec, WM8350_AI_DAC_CONTROL) & - WM8350_BCLK_MSTR; - int enabled = 0; - - /* Check that the DACs or ADCs are enabled since they are - * required for LRC in master mode. The DACs or ADCs need a - * valid audio path i.e. pin -> ADC or DAC -> pin before - * the LRC will be enabled in master mode. */ - if (!master || cmd != SNDRV_PCM_TRIGGER_START) - return 0; - - if (substream->stream == SNDRV_PCM_STREAM_CAPTURE) { - enabled = wm8350_codec_cache_read(codec, WM8350_POWER_MGMT_4) & - (WM8350_ADCR_ENA | WM8350_ADCL_ENA); - } else { - enabled = wm8350_codec_cache_read(codec, WM8350_POWER_MGMT_4) & - (WM8350_DACR_ENA | WM8350_DACL_ENA); - } - - if (!enabled) { - dev_err(codec->dev, - "%s: invalid audio path - no clocks available\n", - __func__); - return -EINVAL; - } + snd_soc_write(codec, WM8350_AI_FORMATING, iface); + snd_soc_write(codec, WM8350_AI_DAC_CONTROL, master); + snd_soc_write(codec, WM8350_DAC_LR_RATE, dac_lrc); + snd_soc_write(codec, WM8350_ADC_LR_RATE, adc_lrc); return 0; } @@ -966,8 +928,9 @@ static int wm8350_pcm_hw_params(struct snd_pcm_substream *substream, struct snd_soc_dai *codec_dai) { struct snd_soc_codec *codec = codec_dai->codec; - struct wm8350 *wm8350 = codec->control_data; - u16 iface = wm8350_codec_read(codec, WM8350_AI_FORMATING) & + struct wm8350_data *wm8350_data = snd_soc_codec_get_drvdata(codec); + struct wm8350 *wm8350 = wm8350_data->wm8350; + u16 iface = snd_soc_read(codec, WM8350_AI_FORMATING) & ~WM8350_AIF_WL_MASK; /* bit size */ @@ -985,7 +948,7 @@ static int wm8350_pcm_hw_params(struct snd_pcm_substream *substream, break; } - wm8350_codec_write(codec, WM8350_AI_FORMATING, iface); + snd_soc_write(codec, WM8350_AI_FORMATING, iface); /* The sloping stopband filter is recommended for use with * lower sample rates to improve performance. @@ -1005,12 +968,15 @@ static int wm8350_pcm_hw_params(struct snd_pcm_substream *substream, static int wm8350_mute(struct snd_soc_dai *dai, int mute) { struct snd_soc_codec *codec = dai->codec; - struct wm8350 *wm8350 = codec->control_data; + unsigned int val; if (mute) - wm8350_set_bits(wm8350, WM8350_DAC_MUTE, WM8350_DAC_MUTE_ENA); + val = WM8350_DAC_MUTE_ENA; else - wm8350_clear_bits(wm8350, WM8350_DAC_MUTE, WM8350_DAC_MUTE_ENA); + val = 0; + + snd_soc_update_bits(codec, WM8350_DAC_MUTE, WM8350_DAC_MUTE_ENA, val); + return 0; } @@ -1079,8 +1045,8 @@ static int wm8350_set_fll(struct snd_soc_dai *codec_dai, unsigned int freq_out) { struct snd_soc_codec *codec = codec_dai->codec; - struct wm8350 *wm8350 = codec->control_data; struct wm8350_data *priv = snd_soc_codec_get_drvdata(codec); + struct wm8350 *wm8350 = priv->wm8350; struct _fll_div fll_div; int ret = 0; u16 fll_1, fll_4; @@ -1104,17 +1070,17 @@ static int wm8350_set_fll(struct snd_soc_dai *codec_dai, fll_div.ratio); /* set up N.K & dividers */ - fll_1 = wm8350_codec_read(codec, WM8350_FLL_CONTROL_1) & + fll_1 = snd_soc_read(codec, WM8350_FLL_CONTROL_1) & ~(WM8350_FLL_OUTDIV_MASK | WM8350_FLL_RSP_RATE_MASK | 0xc000); - wm8350_codec_write(codec, WM8350_FLL_CONTROL_1, + snd_soc_write(codec, WM8350_FLL_CONTROL_1, fll_1 | (fll_div.div << 8) | 0x50); - wm8350_codec_write(codec, WM8350_FLL_CONTROL_2, + snd_soc_write(codec, WM8350_FLL_CONTROL_2, (fll_div.ratio << 11) | (fll_div. n & WM8350_FLL_N_MASK)); - wm8350_codec_write(codec, WM8350_FLL_CONTROL_3, fll_div.k); - fll_4 = wm8350_codec_read(codec, WM8350_FLL_CONTROL_4) & + snd_soc_write(codec, WM8350_FLL_CONTROL_3, fll_div.k); + fll_4 = snd_soc_read(codec, WM8350_FLL_CONTROL_4) & ~(WM8350_FLL_FRAC | WM8350_FLL_SLOW_LOCK_REF); - wm8350_codec_write(codec, WM8350_FLL_CONTROL_4, + snd_soc_write(codec, WM8350_FLL_CONTROL_4, fll_4 | (fll_div.k ? WM8350_FLL_FRAC : 0) | (fll_div.ratio == 8 ? WM8350_FLL_SLOW_LOCK_REF : 0)); @@ -1131,8 +1097,8 @@ static int wm8350_set_fll(struct snd_soc_dai *codec_dai, static int wm8350_set_bias_level(struct snd_soc_codec *codec, enum snd_soc_bias_level level) { - struct wm8350 *wm8350 = codec->control_data; struct wm8350_data *priv = snd_soc_codec_get_drvdata(codec); + struct wm8350 *wm8350 = priv->wm8350; struct wm8350_audio_platform_data *platform = wm8350->codec.platform_data; u16 pm1; @@ -1339,35 +1305,36 @@ static void wm8350_hpr_work(struct work_struct *work) wm8350_hp_work(priv, &priv->hpr, WM8350_JACK_R_LVL); } -static irqreturn_t wm8350_hp_jack_handler(int irq, void *data) +static irqreturn_t wm8350_hpl_jack_handler(int irq, void *data) { struct wm8350_data *priv = data; struct wm8350 *wm8350 = priv->wm8350; - struct wm8350_jack_data *jack = NULL; - switch (irq - wm8350->irq_base) { - case WM8350_IRQ_CODEC_JCK_DET_L: #ifndef CONFIG_SND_SOC_WM8350_MODULE - trace_snd_soc_jack_irq("WM8350 HPL"); + trace_snd_soc_jack_irq("WM8350 HPL"); #endif - jack = &priv->hpl; - break; - case WM8350_IRQ_CODEC_JCK_DET_R: + if (device_may_wakeup(wm8350->dev)) + pm_wakeup_event(wm8350->dev, 250); + + schedule_delayed_work(&priv->hpl.work, 200); + + return IRQ_HANDLED; +} + +static irqreturn_t wm8350_hpr_jack_handler(int irq, void *data) +{ + struct wm8350_data *priv = data; + struct wm8350 *wm8350 = priv->wm8350; + #ifndef CONFIG_SND_SOC_WM8350_MODULE - trace_snd_soc_jack_irq("WM8350 HPR"); + trace_snd_soc_jack_irq("WM8350 HPR"); #endif - jack = &priv->hpr; - break; - - default: - BUG(); - } if (device_may_wakeup(wm8350->dev)) pm_wakeup_event(wm8350->dev, 250); - schedule_delayed_work(&jack->work, 200); + schedule_delayed_work(&priv->hpr.work, 200); return IRQ_HANDLED; } @@ -1387,7 +1354,7 @@ int wm8350_hp_jack_detect(struct snd_soc_codec *codec, enum wm8350_jack which, struct snd_soc_jack *jack, int report) { struct wm8350_data *priv = snd_soc_codec_get_drvdata(codec); - struct wm8350 *wm8350 = codec->control_data; + struct wm8350 *wm8350 = priv->wm8350; int irq; int ena; @@ -1418,7 +1385,14 @@ int wm8350_hp_jack_detect(struct snd_soc_codec *codec, enum wm8350_jack which, } /* Sync status */ - wm8350_hp_jack_handler(irq + wm8350->irq_base, priv); + switch (which) { + case WM8350_JDL: + wm8350_hpl_jack_handler(0, priv); + break; + case WM8350_JDR: + wm8350_hpr_jack_handler(0, priv); + break; + } return 0; } @@ -1463,7 +1437,7 @@ int wm8350_mic_jack_detect(struct snd_soc_codec *codec, int detect_report, int short_report) { struct wm8350_data *priv = snd_soc_codec_get_drvdata(codec); - struct wm8350 *wm8350 = codec->control_data; + struct wm8350 *wm8350 = priv->wm8350; priv->mic.jack = jack; priv->mic.report = detect_report; @@ -1491,7 +1465,6 @@ EXPORT_SYMBOL_GPL(wm8350_mic_jack_detect); static const struct snd_soc_dai_ops wm8350_dai_ops = { .hw_params = wm8350_pcm_hw_params, .digital_mute = wm8350_mute, - .trigger = wm8350_pcm_trigger, .set_fmt = wm8350_set_dai_fmt, .set_sysclk = wm8350_set_dai_sysclk, .set_pll = wm8350_set_fll, @@ -1559,9 +1532,9 @@ static int wm8350_codec_probe(struct snd_soc_codec *codec) wm8350_set_bits(wm8350, WM8350_POWER_MGMT_5, WM8350_CODEC_ENA); /* Enable robust clocking mode in ADC */ - wm8350_codec_write(codec, WM8350_SECURITY, 0xa7); - wm8350_codec_write(codec, 0xde, 0x13); - wm8350_codec_write(codec, WM8350_SECURITY, 0); + snd_soc_write(codec, WM8350_SECURITY, 0xa7); + snd_soc_write(codec, 0xde, 0x13); + snd_soc_write(codec, WM8350_SECURITY, 0); /* read OUT1 & OUT2 volumes */ out1 = &priv->out1; @@ -1601,10 +1574,10 @@ static int wm8350_codec_probe(struct snd_soc_codec *codec) WM8350_JDL_ENA | WM8350_JDR_ENA); wm8350_register_irq(wm8350, WM8350_IRQ_CODEC_JCK_DET_L, - wm8350_hp_jack_handler, 0, "Left jack detect", + wm8350_hpl_jack_handler, 0, "Left jack detect", priv); wm8350_register_irq(wm8350, WM8350_IRQ_CODEC_JCK_DET_R, - wm8350_hp_jack_handler, 0, "Right jack detect", + wm8350_hpr_jack_handler, 0, "Right jack detect", priv); wm8350_register_irq(wm8350, WM8350_IRQ_CODEC_MICSCD, wm8350_mic_handler, 0, "Microphone short", priv); diff --git a/sound/soc/codecs/wm8400.c b/sound/soc/codecs/wm8400.c index 898979d2301..5dc31ebcd0e 100644 --- a/sound/soc/codecs/wm8400.c +++ b/sound/soc/codecs/wm8400.c @@ -138,8 +138,8 @@ static int wm8400_outpga_put_volsw_vu(struct snd_kcontrol *kcontrol, return ret; /* now hit the volume update bits (always bit 8) */ - val = wm8400_read(codec, reg); - return wm8400_write(codec, reg, val | 0x0100); + val = snd_soc_read(codec, reg); + return snd_soc_write(codec, reg, val | 0x0100); } #define WM8400_OUTPGA_SINGLE_R_TLV(xname, reg, shift, max, invert, tlv_array) \ @@ -362,8 +362,8 @@ static int inmixer_event (struct snd_soc_dapm_widget *w, { u16 reg, fakepower; - reg = wm8400_read(w->codec, WM8400_POWER_MANAGEMENT_2); - fakepower = wm8400_read(w->codec, WM8400_INTDRIVBITS); + reg = snd_soc_read(w->codec, WM8400_POWER_MANAGEMENT_2); + fakepower = snd_soc_read(w->codec, WM8400_INTDRIVBITS); if (fakepower & ((1 << WM8400_INMIXL_PWR) | (1 << WM8400_AINLMUX_PWR))) { @@ -378,7 +378,7 @@ static int inmixer_event (struct snd_soc_dapm_widget *w, } else { reg &= ~WM8400_AINR_ENA; } - wm8400_write(w->codec, WM8400_POWER_MANAGEMENT_2, reg); + snd_soc_write(w->codec, WM8400_POWER_MANAGEMENT_2, reg); return 0; } @@ -394,7 +394,7 @@ static int outmixer_event (struct snd_soc_dapm_widget *w, switch (reg_shift) { case WM8400_SPEAKER_MIXER | (WM8400_LDSPK << 8) : - reg = wm8400_read(w->codec, WM8400_OUTPUT_MIXER1); + reg = snd_soc_read(w->codec, WM8400_OUTPUT_MIXER1); if (reg & WM8400_LDLO) { printk(KERN_WARNING "Cannot set as Output Mixer 1 LDLO Set\n"); @@ -402,7 +402,7 @@ static int outmixer_event (struct snd_soc_dapm_widget *w, } break; case WM8400_SPEAKER_MIXER | (WM8400_RDSPK << 8): - reg = wm8400_read(w->codec, WM8400_OUTPUT_MIXER2); + reg = snd_soc_read(w->codec, WM8400_OUTPUT_MIXER2); if (reg & WM8400_RDRO) { printk(KERN_WARNING "Cannot set as Output Mixer 2 RDRO Set\n"); @@ -410,7 +410,7 @@ static int outmixer_event (struct snd_soc_dapm_widget *w, } break; case WM8400_OUTPUT_MIXER1 | (WM8400_LDLO << 8): - reg = wm8400_read(w->codec, WM8400_SPEAKER_MIXER); + reg = snd_soc_read(w->codec, WM8400_SPEAKER_MIXER); if (reg & WM8400_LDSPK) { printk(KERN_WARNING "Cannot set as Speaker Mixer LDSPK Set\n"); @@ -418,7 +418,7 @@ static int outmixer_event (struct snd_soc_dapm_widget *w, } break; case WM8400_OUTPUT_MIXER2 | (WM8400_RDRO << 8): - reg = wm8400_read(w->codec, WM8400_SPEAKER_MIXER); + reg = snd_soc_read(w->codec, WM8400_SPEAKER_MIXER); if (reg & WM8400_RDSPK) { printk(KERN_WARNING "Cannot set as Speaker Mixer RDSPK Set\n"); @@ -1021,13 +1021,13 @@ static int wm8400_set_dai_pll(struct snd_soc_dai *codec_dai, int pll_id, wm8400->fll_in = freq_in; /* We *must* disable the FLL before any changes */ - reg = wm8400_read(codec, WM8400_POWER_MANAGEMENT_2); + reg = snd_soc_read(codec, WM8400_POWER_MANAGEMENT_2); reg &= ~WM8400_FLL_ENA; - wm8400_write(codec, WM8400_POWER_MANAGEMENT_2, reg); + snd_soc_write(codec, WM8400_POWER_MANAGEMENT_2, reg); - reg = wm8400_read(codec, WM8400_FLL_CONTROL_1); + reg = snd_soc_read(codec, WM8400_FLL_CONTROL_1); reg &= ~WM8400_FLL_OSC_ENA; - wm8400_write(codec, WM8400_FLL_CONTROL_1, reg); + snd_soc_write(codec, WM8400_FLL_CONTROL_1, reg); if (!freq_out) return 0; @@ -1035,15 +1035,15 @@ static int wm8400_set_dai_pll(struct snd_soc_dai *codec_dai, int pll_id, reg &= ~(WM8400_FLL_REF_FREQ | WM8400_FLL_FRATIO_MASK); reg |= WM8400_FLL_FRAC | factors.fratio; reg |= factors.freq_ref << WM8400_FLL_REF_FREQ_SHIFT; - wm8400_write(codec, WM8400_FLL_CONTROL_1, reg); + snd_soc_write(codec, WM8400_FLL_CONTROL_1, reg); - wm8400_write(codec, WM8400_FLL_CONTROL_2, factors.k); - wm8400_write(codec, WM8400_FLL_CONTROL_3, factors.n); + snd_soc_write(codec, WM8400_FLL_CONTROL_2, factors.k); + snd_soc_write(codec, WM8400_FLL_CONTROL_3, factors.n); - reg = wm8400_read(codec, WM8400_FLL_CONTROL_4); + reg = snd_soc_read(codec, WM8400_FLL_CONTROL_4); reg &= ~WM8400_FLL_OUTDIV_MASK; reg |= factors.outdiv; - wm8400_write(codec, WM8400_FLL_CONTROL_4, reg); + snd_soc_write(codec, WM8400_FLL_CONTROL_4, reg); return 0; } @@ -1057,8 +1057,8 @@ static int wm8400_set_dai_fmt(struct snd_soc_dai *codec_dai, struct snd_soc_codec *codec = codec_dai->codec; u16 audio1, audio3; - audio1 = wm8400_read(codec, WM8400_AUDIO_INTERFACE_1); - audio3 = wm8400_read(codec, WM8400_AUDIO_INTERFACE_3); + audio1 = snd_soc_read(codec, WM8400_AUDIO_INTERFACE_1); + audio3 = snd_soc_read(codec, WM8400_AUDIO_INTERFACE_3); /* set master/slave audio interface */ switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) { @@ -1099,8 +1099,8 @@ static int wm8400_set_dai_fmt(struct snd_soc_dai *codec_dai, return -EINVAL; } - wm8400_write(codec, WM8400_AUDIO_INTERFACE_1, audio1); - wm8400_write(codec, WM8400_AUDIO_INTERFACE_3, audio3); + snd_soc_write(codec, WM8400_AUDIO_INTERFACE_1, audio1); + snd_soc_write(codec, WM8400_AUDIO_INTERFACE_3, audio3); return 0; } @@ -1112,24 +1112,24 @@ static int wm8400_set_dai_clkdiv(struct snd_soc_dai *codec_dai, switch (div_id) { case WM8400_MCLK_DIV: - reg = wm8400_read(codec, WM8400_CLOCKING_2) & + reg = snd_soc_read(codec, WM8400_CLOCKING_2) & ~WM8400_MCLK_DIV_MASK; - wm8400_write(codec, WM8400_CLOCKING_2, reg | div); + snd_soc_write(codec, WM8400_CLOCKING_2, reg | div); break; case WM8400_DACCLK_DIV: - reg = wm8400_read(codec, WM8400_CLOCKING_2) & + reg = snd_soc_read(codec, WM8400_CLOCKING_2) & ~WM8400_DAC_CLKDIV_MASK; - wm8400_write(codec, WM8400_CLOCKING_2, reg | div); + snd_soc_write(codec, WM8400_CLOCKING_2, reg | div); break; case WM8400_ADCCLK_DIV: - reg = wm8400_read(codec, WM8400_CLOCKING_2) & + reg = snd_soc_read(codec, WM8400_CLOCKING_2) & ~WM8400_ADC_CLKDIV_MASK; - wm8400_write(codec, WM8400_CLOCKING_2, reg | div); + snd_soc_write(codec, WM8400_CLOCKING_2, reg | div); break; case WM8400_BCLK_DIV: - reg = wm8400_read(codec, WM8400_CLOCKING_1) & + reg = snd_soc_read(codec, WM8400_CLOCKING_1) & ~WM8400_BCLK_DIV_MASK; - wm8400_write(codec, WM8400_CLOCKING_1, reg | div); + snd_soc_write(codec, WM8400_CLOCKING_1, reg | div); break; default: return -EINVAL; @@ -1145,9 +1145,8 @@ static int wm8400_hw_params(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *params, struct snd_soc_dai *dai) { - struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct snd_soc_codec *codec = rtd->codec; - u16 audio1 = wm8400_read(codec, WM8400_AUDIO_INTERFACE_1); + struct snd_soc_codec *codec = dai->codec; + u16 audio1 = snd_soc_read(codec, WM8400_AUDIO_INTERFACE_1); audio1 &= ~WM8400_AIF_WL_MASK; /* bit size */ @@ -1165,19 +1164,19 @@ static int wm8400_hw_params(struct snd_pcm_substream *substream, break; } - wm8400_write(codec, WM8400_AUDIO_INTERFACE_1, audio1); + snd_soc_write(codec, WM8400_AUDIO_INTERFACE_1, audio1); return 0; } static int wm8400_mute(struct snd_soc_dai *dai, int mute) { struct snd_soc_codec *codec = dai->codec; - u16 val = wm8400_read(codec, WM8400_DAC_CTRL) & ~WM8400_DAC_MUTE; + u16 val = snd_soc_read(codec, WM8400_DAC_CTRL) & ~WM8400_DAC_MUTE; if (mute) - wm8400_write(codec, WM8400_DAC_CTRL, val | WM8400_DAC_MUTE); + snd_soc_write(codec, WM8400_DAC_CTRL, val | WM8400_DAC_MUTE); else - wm8400_write(codec, WM8400_DAC_CTRL, val); + snd_soc_write(codec, WM8400_DAC_CTRL, val); return 0; } @@ -1196,9 +1195,9 @@ static int wm8400_set_bias_level(struct snd_soc_codec *codec, case SND_SOC_BIAS_PREPARE: /* VMID=2*50k */ - val = wm8400_read(codec, WM8400_POWER_MANAGEMENT_1) & + val = snd_soc_read(codec, WM8400_POWER_MANAGEMENT_1) & ~WM8400_VMID_MODE_MASK; - wm8400_write(codec, WM8400_POWER_MANAGEMENT_1, val | 0x2); + snd_soc_write(codec, WM8400_POWER_MANAGEMENT_1, val | 0x2); break; case SND_SOC_BIAS_STANDBY: @@ -1212,74 +1211,74 @@ static int wm8400_set_bias_level(struct snd_soc_codec *codec, return ret; } - wm8400_write(codec, WM8400_POWER_MANAGEMENT_1, + snd_soc_write(codec, WM8400_POWER_MANAGEMENT_1, WM8400_CODEC_ENA | WM8400_SYSCLK_ENA); /* Enable POBCTRL, SOFT_ST, VMIDTOG and BUFDCOPEN */ - wm8400_write(codec, WM8400_ANTIPOP2, WM8400_SOFTST | + snd_soc_write(codec, WM8400_ANTIPOP2, WM8400_SOFTST | WM8400_BUFDCOPEN | WM8400_POBCTRL); msleep(50); /* Enable VREF & VMID at 2x50k */ - val = wm8400_read(codec, WM8400_POWER_MANAGEMENT_1); + val = snd_soc_read(codec, WM8400_POWER_MANAGEMENT_1); val |= 0x2 | WM8400_VREF_ENA; - wm8400_write(codec, WM8400_POWER_MANAGEMENT_1, val); + snd_soc_write(codec, WM8400_POWER_MANAGEMENT_1, val); /* Enable BUFIOEN */ - wm8400_write(codec, WM8400_ANTIPOP2, WM8400_SOFTST | + snd_soc_write(codec, WM8400_ANTIPOP2, WM8400_SOFTST | WM8400_BUFDCOPEN | WM8400_POBCTRL | WM8400_BUFIOEN); /* disable POBCTRL, SOFT_ST and BUFDCOPEN */ - wm8400_write(codec, WM8400_ANTIPOP2, WM8400_BUFIOEN); + snd_soc_write(codec, WM8400_ANTIPOP2, WM8400_BUFIOEN); } /* VMID=2*300k */ - val = wm8400_read(codec, WM8400_POWER_MANAGEMENT_1) & + val = snd_soc_read(codec, WM8400_POWER_MANAGEMENT_1) & ~WM8400_VMID_MODE_MASK; - wm8400_write(codec, WM8400_POWER_MANAGEMENT_1, val | 0x4); + snd_soc_write(codec, WM8400_POWER_MANAGEMENT_1, val | 0x4); break; case SND_SOC_BIAS_OFF: /* Enable POBCTRL and SOFT_ST */ - wm8400_write(codec, WM8400_ANTIPOP2, WM8400_SOFTST | + snd_soc_write(codec, WM8400_ANTIPOP2, WM8400_SOFTST | WM8400_POBCTRL | WM8400_BUFIOEN); /* Enable POBCTRL, SOFT_ST and BUFDCOPEN */ - wm8400_write(codec, WM8400_ANTIPOP2, WM8400_SOFTST | + snd_soc_write(codec, WM8400_ANTIPOP2, WM8400_SOFTST | WM8400_BUFDCOPEN | WM8400_POBCTRL | WM8400_BUFIOEN); /* mute DAC */ - val = wm8400_read(codec, WM8400_DAC_CTRL); - wm8400_write(codec, WM8400_DAC_CTRL, val | WM8400_DAC_MUTE); + val = snd_soc_read(codec, WM8400_DAC_CTRL); + snd_soc_write(codec, WM8400_DAC_CTRL, val | WM8400_DAC_MUTE); /* Enable any disabled outputs */ - val = wm8400_read(codec, WM8400_POWER_MANAGEMENT_1); + val = snd_soc_read(codec, WM8400_POWER_MANAGEMENT_1); val |= WM8400_SPK_ENA | WM8400_OUT3_ENA | WM8400_OUT4_ENA | WM8400_LOUT_ENA | WM8400_ROUT_ENA; - wm8400_write(codec, WM8400_POWER_MANAGEMENT_1, val); + snd_soc_write(codec, WM8400_POWER_MANAGEMENT_1, val); /* Disable VMID */ val &= ~WM8400_VMID_MODE_MASK; - wm8400_write(codec, WM8400_POWER_MANAGEMENT_1, val); + snd_soc_write(codec, WM8400_POWER_MANAGEMENT_1, val); msleep(300); /* Enable all output discharge bits */ - wm8400_write(codec, WM8400_ANTIPOP1, WM8400_DIS_LLINE | + snd_soc_write(codec, WM8400_ANTIPOP1, WM8400_DIS_LLINE | WM8400_DIS_RLINE | WM8400_DIS_OUT3 | WM8400_DIS_OUT4 | WM8400_DIS_LOUT | WM8400_DIS_ROUT); /* Disable VREF */ val &= ~WM8400_VREF_ENA; - wm8400_write(codec, WM8400_POWER_MANAGEMENT_1, val); + snd_soc_write(codec, WM8400_POWER_MANAGEMENT_1, val); /* disable POBCTRL, SOFT_ST and BUFDCOPEN */ - wm8400_write(codec, WM8400_ANTIPOP2, 0x0); + snd_soc_write(codec, WM8400_ANTIPOP2, 0x0); ret = regulator_bulk_disable(ARRAY_SIZE(power), &power[0]); @@ -1385,19 +1384,19 @@ static int wm8400_codec_probe(struct snd_soc_codec *codec) wm8400_codec_reset(codec); - reg = wm8400_read(codec, WM8400_POWER_MANAGEMENT_1); - wm8400_write(codec, WM8400_POWER_MANAGEMENT_1, reg | WM8400_CODEC_ENA); + reg = snd_soc_read(codec, WM8400_POWER_MANAGEMENT_1); + snd_soc_write(codec, WM8400_POWER_MANAGEMENT_1, reg | WM8400_CODEC_ENA); /* Latch volume update bits */ - reg = wm8400_read(codec, WM8400_LEFT_LINE_INPUT_1_2_VOLUME); - wm8400_write(codec, WM8400_LEFT_LINE_INPUT_1_2_VOLUME, + reg = snd_soc_read(codec, WM8400_LEFT_LINE_INPUT_1_2_VOLUME); + snd_soc_write(codec, WM8400_LEFT_LINE_INPUT_1_2_VOLUME, reg & WM8400_IPVU); - reg = wm8400_read(codec, WM8400_RIGHT_LINE_INPUT_1_2_VOLUME); - wm8400_write(codec, WM8400_RIGHT_LINE_INPUT_1_2_VOLUME, + reg = snd_soc_read(codec, WM8400_RIGHT_LINE_INPUT_1_2_VOLUME); + snd_soc_write(codec, WM8400_RIGHT_LINE_INPUT_1_2_VOLUME, reg & WM8400_IPVU); - wm8400_write(codec, WM8400_LEFT_OUTPUT_VOLUME, 0x50 | (1<<8)); - wm8400_write(codec, WM8400_RIGHT_OUTPUT_VOLUME, 0x50 | (1<<8)); + snd_soc_write(codec, WM8400_LEFT_OUTPUT_VOLUME, 0x50 | (1<<8)); + snd_soc_write(codec, WM8400_RIGHT_OUTPUT_VOLUME, 0x50 | (1<<8)); if (!schedule_work(&priv->work)) { ret = -EINVAL; @@ -1414,8 +1413,8 @@ static int wm8400_codec_remove(struct snd_soc_codec *codec) { u16 reg; - reg = wm8400_read(codec, WM8400_POWER_MANAGEMENT_1); - wm8400_write(codec, WM8400_POWER_MANAGEMENT_1, + reg = snd_soc_read(codec, WM8400_POWER_MANAGEMENT_1); + snd_soc_write(codec, WM8400_POWER_MANAGEMENT_1, reg & (~WM8400_CODEC_ENA)); regulator_bulk_free(ARRAY_SIZE(power), power); @@ -1428,7 +1427,7 @@ static struct snd_soc_codec_driver soc_codec_dev_wm8400 = { .remove = wm8400_codec_remove, .suspend = wm8400_suspend, .resume = wm8400_resume, - .read = wm8400_read, + .read = snd_soc_read, .write = wm8400_write, .set_bias_level = wm8400_set_bias_level, diff --git a/sound/soc/codecs/wm8510.c b/sound/soc/codecs/wm8510.c index 9166126bd31..56a049555e2 100644 --- a/sound/soc/codecs/wm8510.c +++ b/sound/soc/codecs/wm8510.c @@ -392,8 +392,7 @@ static int wm8510_pcm_hw_params(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *params, struct snd_soc_dai *dai) { - struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct snd_soc_codec *codec = rtd->codec; + struct snd_soc_codec *codec = dai->codec; u16 iface = snd_soc_read(codec, WM8510_IFACE) & 0x19f; u16 adn = snd_soc_read(codec, WM8510_ADD) & 0x1f1; diff --git a/sound/soc/codecs/wm8523.c b/sound/soc/codecs/wm8523.c index 7fea2c3bf7e..1c3ffb290cd 100644 --- a/sound/soc/codecs/wm8523.c +++ b/sound/soc/codecs/wm8523.c @@ -145,8 +145,7 @@ static int wm8523_hw_params(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *params, struct snd_soc_dai *dai) { - struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct snd_soc_codec *codec = rtd->codec; + struct snd_soc_codec *codec = dai->codec; struct wm8523_priv *wm8523 = snd_soc_codec_get_drvdata(codec); int i; u16 aifctrl1 = snd_soc_read(codec, WM8523_AIF_CTRL1); diff --git a/sound/soc/codecs/wm8728.c b/sound/soc/codecs/wm8728.c index fc3d59e4908..1467f97dce2 100644 --- a/sound/soc/codecs/wm8728.c +++ b/sound/soc/codecs/wm8728.c @@ -88,8 +88,7 @@ static int wm8728_hw_params(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *params, struct snd_soc_dai *dai) { - struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct snd_soc_codec *codec = rtd->codec; + struct snd_soc_codec *codec = dai->codec; u16 dac = snd_soc_read(codec, WM8728_DACCTL); dac &= ~0x18; diff --git a/sound/soc/codecs/wm8731.c b/sound/soc/codecs/wm8731.c index a32caa72bd7..9d1b9b0271f 100644 --- a/sound/soc/codecs/wm8731.c +++ b/sound/soc/codecs/wm8731.c @@ -635,16 +635,17 @@ static int __devinit wm8731_spi_probe(struct spi_device *spi) struct wm8731_priv *wm8731; int ret; - wm8731 = kzalloc(sizeof(struct wm8731_priv), GFP_KERNEL); + wm8731 = devm_kzalloc(&spi->dev, sizeof(struct wm8731_priv), + GFP_KERNEL); if (wm8731 == NULL) return -ENOMEM; - wm8731->regmap = regmap_init_spi(spi, &wm8731_regmap); + wm8731->regmap = devm_regmap_init_spi(spi, &wm8731_regmap); if (IS_ERR(wm8731->regmap)) { ret = PTR_ERR(wm8731->regmap); dev_err(&spi->dev, "Failed to allocate register map: %d\n", ret); - goto err; + return ret; } spi_set_drvdata(spi, wm8731); @@ -653,25 +654,15 @@ static int __devinit wm8731_spi_probe(struct spi_device *spi) &soc_codec_dev_wm8731, &wm8731_dai, 1); if (ret != 0) { dev_err(&spi->dev, "Failed to register CODEC: %d\n", ret); - goto err_regmap; + return ret; } return 0; - -err_regmap: - regmap_exit(wm8731->regmap); -err: - kfree(wm8731); - return ret; } static int __devexit wm8731_spi_remove(struct spi_device *spi) { - struct wm8731_priv *wm8731 = spi_get_drvdata(spi); - snd_soc_unregister_codec(&spi->dev); - regmap_exit(wm8731->regmap); - kfree(wm8731); return 0; } @@ -693,16 +684,17 @@ static __devinit int wm8731_i2c_probe(struct i2c_client *i2c, struct wm8731_priv *wm8731; int ret; - wm8731 = kzalloc(sizeof(struct wm8731_priv), GFP_KERNEL); + wm8731 = devm_kzalloc(&i2c->dev, sizeof(struct wm8731_priv), + GFP_KERNEL); if (wm8731 == NULL) return -ENOMEM; - wm8731->regmap = regmap_init_i2c(i2c, &wm8731_regmap); + wm8731->regmap = devm_regmap_init_i2c(i2c, &wm8731_regmap); if (IS_ERR(wm8731->regmap)) { ret = PTR_ERR(wm8731->regmap); dev_err(&i2c->dev, "Failed to allocate register map: %d\n", ret); - goto err; + return ret; } i2c_set_clientdata(i2c, wm8731); @@ -711,24 +703,15 @@ static __devinit int wm8731_i2c_probe(struct i2c_client *i2c, &soc_codec_dev_wm8731, &wm8731_dai, 1); if (ret != 0) { dev_err(&i2c->dev, "Failed to register CODEC: %d\n", ret); - goto err_regmap; + return ret; } return 0; - -err_regmap: - regmap_exit(wm8731->regmap); -err: - kfree(wm8731); - return ret; } static __devexit int wm8731_i2c_remove(struct i2c_client *client) { - struct wm8731_priv *wm8731 = i2c_get_clientdata(client); snd_soc_unregister_codec(&client->dev); - regmap_exit(wm8731->regmap); - kfree(wm8731); return 0; } diff --git a/sound/soc/codecs/wm8737.c b/sound/soc/codecs/wm8737.c index 4fe9d191e27..d0520124616 100644 --- a/sound/soc/codecs/wm8737.c +++ b/sound/soc/codecs/wm8737.c @@ -329,8 +329,7 @@ static int wm8737_hw_params(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *params, struct snd_soc_dai *dai) { - struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct snd_soc_codec *codec = rtd->codec; + struct snd_soc_codec *codec = dai->codec; struct wm8737_priv *wm8737 = snd_soc_codec_get_drvdata(codec); int i; u16 clocking = 0; diff --git a/sound/soc/codecs/wm8741.c b/sound/soc/codecs/wm8741.c index 3941f50bf18..6e849cb0424 100644 --- a/sound/soc/codecs/wm8741.c +++ b/sound/soc/codecs/wm8741.c @@ -203,8 +203,7 @@ static int wm8741_hw_params(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *params, struct snd_soc_dai *dai) { - struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct snd_soc_codec *codec = rtd->codec; + struct snd_soc_codec *codec = dai->codec; struct wm8741_priv *wm8741 = snd_soc_codec_get_drvdata(codec); u16 iface = snd_soc_read(codec, WM8741_FORMAT_CONTROL) & 0x1FC; int i; diff --git a/sound/soc/codecs/wm8750.c b/sound/soc/codecs/wm8750.c index e4c50ce7d9c..89151ca5e77 100644 --- a/sound/soc/codecs/wm8750.c +++ b/sound/soc/codecs/wm8750.c @@ -547,8 +547,7 @@ static int wm8750_pcm_hw_params(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *params, struct snd_soc_dai *dai) { - struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct snd_soc_codec *codec = rtd->codec; + struct snd_soc_codec *codec = dai->codec; struct wm8750_priv *wm8750 = snd_soc_codec_get_drvdata(codec); u16 iface = snd_soc_read(codec, WM8750_IFACE) & 0x1f3; u16 srate = snd_soc_read(codec, WM8750_SRATE) & 0x1c0; diff --git a/sound/soc/codecs/wm8753.c b/sound/soc/codecs/wm8753.c index e27e7b62b36..a26482cd765 100644 --- a/sound/soc/codecs/wm8753.c +++ b/sound/soc/codecs/wm8753.c @@ -931,8 +931,7 @@ static int wm8753_pcm_hw_params(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *params, struct snd_soc_dai *dai) { - struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct snd_soc_codec *codec = rtd->codec; + struct snd_soc_codec *codec = dai->codec; struct wm8753_priv *wm8753 = snd_soc_codec_get_drvdata(codec); u16 voice = snd_soc_read(codec, WM8753_PCM) & 0x01f3; u16 srate = snd_soc_read(codec, WM8753_SRATE1) & 0x017f; @@ -1161,8 +1160,7 @@ static int wm8753_i2s_hw_params(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *params, struct snd_soc_dai *dai) { - struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct snd_soc_codec *codec = rtd->codec; + struct snd_soc_codec *codec = dai->codec; struct wm8753_priv *wm8753 = snd_soc_codec_get_drvdata(codec); u16 srate = snd_soc_read(codec, WM8753_SRATE1) & 0x01c0; u16 hifi = snd_soc_read(codec, WM8753_HIFI) & 0x01f3; diff --git a/sound/soc/codecs/wm8900.c b/sound/soc/codecs/wm8900.c index f18c554efc9..077c9628c70 100644 --- a/sound/soc/codecs/wm8900.c +++ b/sound/soc/codecs/wm8900.c @@ -610,8 +610,7 @@ static int wm8900_hw_params(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *params, struct snd_soc_dai *dai) { - struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct snd_soc_codec *codec = rtd->codec; + struct snd_soc_codec *codec = dai->codec; u16 reg; reg = snd_soc_read(codec, WM8900_REG_AUDIO1) & ~0x60; diff --git a/sound/soc/codecs/wm8903.c b/sound/soc/codecs/wm8903.c index c91fb2f99c1..86b8a292659 100644 --- a/sound/soc/codecs/wm8903.c +++ b/sound/soc/codecs/wm8903.c @@ -1432,8 +1432,7 @@ static int wm8903_hw_params(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *params, struct snd_soc_dai *dai) { - struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct snd_soc_codec *codec =rtd->codec; + struct snd_soc_codec *codec = dai->codec; struct wm8903_priv *wm8903 = snd_soc_codec_get_drvdata(codec); int fs = params_rate(params); int bclk; diff --git a/sound/soc/codecs/wm8940.c b/sound/soc/codecs/wm8940.c index d2883affea3..481a3d9cfe4 100644 --- a/sound/soc/codecs/wm8940.c +++ b/sound/soc/codecs/wm8940.c @@ -371,8 +371,7 @@ static int wm8940_i2s_hw_params(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *params, struct snd_soc_dai *dai) { - struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct snd_soc_codec *codec = rtd->codec; + struct snd_soc_codec *codec = dai->codec; u16 iface = snd_soc_read(codec, WM8940_IFACE) & 0xFD9F; u16 addcntrl = snd_soc_read(codec, WM8940_ADDCNTRL) & 0xFFF1; u16 companding = snd_soc_read(codec, diff --git a/sound/soc/codecs/wm8960.c b/sound/soc/codecs/wm8960.c index 840d72086d0..8bc659d8dd2 100644 --- a/sound/soc/codecs/wm8960.c +++ b/sound/soc/codecs/wm8960.c @@ -505,8 +505,7 @@ static int wm8960_hw_params(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *params, struct snd_soc_dai *dai) { - struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct snd_soc_codec *codec = rtd->codec; + struct snd_soc_codec *codec = dai->codec; struct wm8960_priv *wm8960 = snd_soc_codec_get_drvdata(codec); u16 iface = snd_soc_read(codec, WM8960_IFACE1) & 0xfff3; int i; diff --git a/sound/soc/codecs/wm8962.c b/sound/soc/codecs/wm8962.c index 15d467ff91b..0cfce9999c8 100644 --- a/sound/soc/codecs/wm8962.c +++ b/sound/soc/codecs/wm8962.c @@ -1478,7 +1478,8 @@ static const DECLARE_TLV_DB_SCALE(eq_tlv, -1200, 100, 0); static int wm8962_dsp2_write_config(struct snd_soc_codec *codec) { - return 0; + return regcache_sync_region(codec->control_data, + WM8962_HDBASS_AI_1, WM8962_MAX_REGISTER); } static int wm8962_dsp2_set_enable(struct snd_soc_codec *codec, u16 val) @@ -1755,10 +1756,22 @@ SOC_DOUBLE_R_TLV("EQ4 Volume", WM8962_EQ3, WM8962_EQ23, SOC_DOUBLE_R_TLV("EQ5 Volume", WM8962_EQ3, WM8962_EQ23, WM8962_EQL_B5_GAIN_SHIFT, 31, 0, eq_tlv), +SOC_SINGLE("3D Switch", WM8962_THREED1, 0, 1, 0), +SND_SOC_BYTES_MASK("3D Coefficients", WM8962_THREED1, 4, WM8962_THREED_ENA), + +SOC_SINGLE("DF1 Switch", WM8962_DF1, 0, 1, 0), +SND_SOC_BYTES_MASK("DF1 Coefficients", WM8962_DF1, 7, WM8962_DF1_ENA), + +SOC_SINGLE("DRC Switch", WM8962_DRC_1, 0, 1, 0), +SND_SOC_BYTES_MASK("DRC Coefficients", WM8962_DRC_1, 5, WM8962_DRC_ENA), + WM8962_DSP2_ENABLE("VSS Switch", WM8962_VSS_ENA_SHIFT), +SND_SOC_BYTES("VSS Coefficients", WM8962_VSS_XHD2_1, 148), WM8962_DSP2_ENABLE("HPF1 Switch", WM8962_HPF1_ENA_SHIFT), WM8962_DSP2_ENABLE("HPF2 Switch", WM8962_HPF2_ENA_SHIFT), +SND_SOC_BYTES("HPF Coefficients", WM8962_LHPF2, 1), WM8962_DSP2_ENABLE("HD Bass Switch", WM8962_HDBASS_ENA_SHIFT), +SND_SOC_BYTES("HD Bass Coefficients", WM8962_HDBASS_AI_1, 30), }; static const struct snd_kcontrol_new wm8962_spk_mono_controls[] = { @@ -2519,8 +2532,7 @@ static int wm8962_hw_params(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *params, struct snd_soc_dai *dai) { - struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct snd_soc_codec *codec = rtd->codec; + struct snd_soc_codec *codec = dai->codec; struct wm8962_priv *wm8962 = snd_soc_codec_get_drvdata(codec); int i; int aif0 = 0; diff --git a/sound/soc/codecs/wm8971.c b/sound/soc/codecs/wm8971.c index 28fe59e3ce0..eef783f6b6d 100644 --- a/sound/soc/codecs/wm8971.c +++ b/sound/soc/codecs/wm8971.c @@ -478,8 +478,7 @@ static int wm8971_pcm_hw_params(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *params, struct snd_soc_dai *dai) { - struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct snd_soc_codec *codec = rtd->codec; + struct snd_soc_codec *codec = dai->codec; struct wm8971_priv *wm8971 = snd_soc_codec_get_drvdata(codec); u16 iface = snd_soc_read(codec, WM8971_IFACE) & 0x1f3; u16 srate = snd_soc_read(codec, WM8971_SRATE) & 0x1c0; diff --git a/sound/soc/codecs/wm8978.c b/sound/soc/codecs/wm8978.c index 72d5fdcd3cc..a5be3adecf7 100644 --- a/sound/soc/codecs/wm8978.c +++ b/sound/soc/codecs/wm8978.c @@ -723,8 +723,7 @@ static int wm8978_hw_params(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *params, struct snd_soc_dai *dai) { - struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct snd_soc_codec *codec = rtd->codec; + struct snd_soc_codec *codec = dai->codec; struct wm8978_priv *wm8978 = snd_soc_codec_get_drvdata(codec); /* Word length mask = 0x60 */ u16 iface_ctl = snd_soc_read(codec, WM8978_AUDIO_INTERFACE) & ~0x60; diff --git a/sound/soc/codecs/wm8988.c b/sound/soc/codecs/wm8988.c index 6cdf6a2bc28..1d4c5cf47b0 100644 --- a/sound/soc/codecs/wm8988.c +++ b/sound/soc/codecs/wm8988.c @@ -668,8 +668,7 @@ static int wm8988_pcm_hw_params(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *params, struct snd_soc_dai *dai) { - struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct snd_soc_codec *codec = rtd->codec; + struct snd_soc_codec *codec = dai->codec; struct wm8988_priv *wm8988 = snd_soc_codec_get_drvdata(codec); u16 iface = snd_soc_read(codec, WM8988_IFACE) & 0x1f3; u16 srate = snd_soc_read(codec, WM8988_SRATE) & 0x180; diff --git a/sound/soc/codecs/wm8990.c b/sound/soc/codecs/wm8990.c index 9d242351e6e..db63c97ddf5 100644 --- a/sound/soc/codecs/wm8990.c +++ b/sound/soc/codecs/wm8990.c @@ -1112,8 +1112,7 @@ static int wm8990_hw_params(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *params, struct snd_soc_dai *dai) { - struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct snd_soc_codec *codec = rtd->codec; + struct snd_soc_codec *codec = dai->codec; u16 audio1 = snd_soc_read(codec, WM8990_AUDIO_INTERFACE_1); audio1 &= ~WM8990_AIF_WL_MASK; diff --git a/sound/soc/codecs/wm8993.c b/sound/soc/codecs/wm8993.c index d256a934064..36acfccab99 100644 --- a/sound/soc/codecs/wm8993.c +++ b/sound/soc/codecs/wm8993.c @@ -218,7 +218,6 @@ struct wm8993_priv { unsigned int sysclk_rate; unsigned int fs; unsigned int bclk; - int class_w_users; unsigned int fll_fref; unsigned int fll_fout; int fll_src; @@ -824,84 +823,6 @@ static int clk_sys_event(struct snd_soc_dapm_widget *w, return 0; } -/* - * When used with DAC outputs only the WM8993 charge pump supports - * operation in class W mode, providing very low power consumption - * when used with digital sources. Enable and disable this mode - * automatically depending on the mixer configuration. - * - * Currently the only supported paths are the direct DAC->headphone - * paths (which provide minimum power consumption anyway). - */ -static int class_w_put(struct snd_kcontrol *kcontrol, - struct snd_ctl_elem_value *ucontrol) -{ - struct snd_soc_dapm_widget_list *wlist = snd_kcontrol_chip(kcontrol); - struct snd_soc_dapm_widget *widget = wlist->widgets[0]; - struct snd_soc_codec *codec = widget->codec; - struct wm8993_priv *wm8993 = snd_soc_codec_get_drvdata(codec); - int ret; - - /* Turn it off if we're using the main output mixer */ - if (ucontrol->value.integer.value[0] == 0) { - if (wm8993->class_w_users == 0) { - dev_dbg(codec->dev, "Disabling Class W\n"); - snd_soc_update_bits(codec, WM8993_CLASS_W_0, - WM8993_CP_DYN_FREQ | - WM8993_CP_DYN_V, - 0); - } - wm8993->class_w_users++; - wm8993->hubs_data.class_w = true; - } - - /* Implement the change */ - ret = snd_soc_dapm_put_enum_double(kcontrol, ucontrol); - - /* Enable it if we're using the direct DAC path */ - if (ucontrol->value.integer.value[0] == 1) { - if (wm8993->class_w_users == 1) { - dev_dbg(codec->dev, "Enabling Class W\n"); - snd_soc_update_bits(codec, WM8993_CLASS_W_0, - WM8993_CP_DYN_FREQ | - WM8993_CP_DYN_V, - WM8993_CP_DYN_FREQ | - WM8993_CP_DYN_V); - } - wm8993->class_w_users--; - wm8993->hubs_data.class_w = false; - } - - dev_dbg(codec->dev, "Indirect DAC use count now %d\n", - wm8993->class_w_users); - - return ret; -} - -#define SOC_DAPM_ENUM_W(xname, xenum) \ -{ .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = xname, \ - .info = snd_soc_info_enum_double, \ - .get = snd_soc_dapm_get_enum_double, \ - .put = class_w_put, \ - .private_value = (unsigned long)&xenum } - -static const char *hp_mux_text[] = { - "Mixer", - "DAC", -}; - -static const struct soc_enum hpl_enum = - SOC_ENUM_SINGLE(WM8993_OUTPUT_MIXER1, 8, 2, hp_mux_text); - -static const struct snd_kcontrol_new hpl_mux = - SOC_DAPM_ENUM_W("Left Headphone Mux", hpl_enum); - -static const struct soc_enum hpr_enum = - SOC_ENUM_SINGLE(WM8993_OUTPUT_MIXER2, 8, 2, hp_mux_text); - -static const struct snd_kcontrol_new hpr_mux = - SOC_DAPM_ENUM_W("Right Headphone Mux", hpr_enum); - static const struct snd_kcontrol_new left_speaker_mixer[] = { SOC_DAPM_SINGLE("Input Switch", WM8993_SPEAKER_MIXER, 7, 1, 0), SOC_DAPM_SINGLE("IN1LP Switch", WM8993_SPEAKER_MIXER, 5, 1, 0), @@ -988,8 +909,8 @@ SND_SOC_DAPM_MUX("DACR Sidetone", SND_SOC_NOPM, 0, 0, &sidetoner_mux), SND_SOC_DAPM_DAC("DACL", NULL, WM8993_POWER_MANAGEMENT_3, 1, 0), SND_SOC_DAPM_DAC("DACR", NULL, WM8993_POWER_MANAGEMENT_3, 0, 0), -SND_SOC_DAPM_MUX("Left Headphone Mux", SND_SOC_NOPM, 0, 0, &hpl_mux), -SND_SOC_DAPM_MUX("Right Headphone Mux", SND_SOC_NOPM, 0, 0, &hpr_mux), +SND_SOC_DAPM_MUX("Left Headphone Mux", SND_SOC_NOPM, 0, 0, &wm_hubs_hpl_mux), +SND_SOC_DAPM_MUX("Right Headphone Mux", SND_SOC_NOPM, 0, 0, &wm_hubs_hpr_mux), SND_SOC_DAPM_MIXER("SPKL", WM8993_POWER_MANAGEMENT_3, 8, 0, left_speaker_mixer, ARRAY_SIZE(left_speaker_mixer)), @@ -1579,9 +1500,6 @@ static int wm8993_probe(struct snd_soc_codec *codec) return ret; } - /* By default we're using the output mixers */ - wm8993->class_w_users = 2; - /* Latch volume update bits and default ZC on */ snd_soc_update_bits(codec, WM8993_RIGHT_DAC_DIGITAL_VOLUME, WM8993_DAC_VU, WM8993_DAC_VU); diff --git a/sound/soc/codecs/wm8994.c b/sound/soc/codecs/wm8994.c index 2de12ebe43b..993639d694c 100644 --- a/sound/soc/codecs/wm8994.c +++ b/sound/soc/codecs/wm8994.c @@ -70,8 +70,8 @@ static const struct wm8958_micd_rate micdet_rates[] = { static const struct wm8958_micd_rate jackdet_rates[] = { { 32768, true, 0, 1 }, { 32768, false, 0, 1 }, - { 44100 * 256, true, 7, 10 }, - { 44100 * 256, false, 7, 10 }, + { 44100 * 256, true, 10, 10 }, + { 44100 * 256, false, 7, 8 }, }; static void wm8958_micd_set_rate(struct snd_soc_codec *codec) @@ -82,7 +82,8 @@ static void wm8958_micd_set_rate(struct snd_soc_codec *codec) const struct wm8958_micd_rate *rates; int num_rates; - if (wm8994->jack_cb != wm8958_default_micdet) + if (!(wm8994->pdata && wm8994->pdata->micd_rates) && + wm8994->jack_cb != wm8958_default_micdet) return; idle = !wm8994->jack_mic; @@ -118,6 +119,10 @@ static void wm8958_micd_set_rate(struct snd_soc_codec *codec) val = rates[best].start << WM8958_MICD_BIAS_STARTTIME_SHIFT | rates[best].rate << WM8958_MICD_RATE_SHIFT; + dev_dbg(codec->dev, "MICD rate %d,%d for %dHz %s\n", + rates[best].start, rates[best].rate, sysclk, + idle ? "idle" : "active"); + snd_soc_update_bits(codec, WM8958_MIC_DETECT_1, WM8958_MICD_BIAS_STARTTIME_MASK | WM8958_MICD_RATE_MASK, val); @@ -398,7 +403,7 @@ static void wm8994_set_retune_mobile(struct snd_soc_codec *codec, int block) wm8994->dac_rates[iface]); /* The EQ will be disabled while reconfiguring it, remember the - * current configuration. + * current configuration. */ save = snd_soc_read(codec, base); save &= WM8994_AIF1DAC1_EQ_ENA; @@ -689,6 +694,9 @@ static void wm1811_jackdet_set_mode(struct snd_soc_codec *codec, u16 mode) if (!wm8994->jackdet || !wm8994->jack_cb) return; + if (!wm8994->jackdet || !wm8994->jack_cb) + return; + if (wm8994->active_refcount) mode = WM1811_JACKDET_MODE_AUDIO; @@ -784,7 +792,7 @@ static void vmid_reference(struct snd_soc_codec *codec) switch (wm8994->vmid_mode) { default: - WARN_ON(0 == "Invalid VMID mode"); + WARN_ON(NULL == "Invalid VMID mode"); case WM8994_VMID_NORMAL: /* Startup bias, VMID ramp & buffer */ snd_soc_update_bits(codec, WM8994_ANTIPOP_2, @@ -937,27 +945,12 @@ static int vmid_event(struct snd_soc_dapm_widget *w, return 0; } -static void wm8994_update_class_w(struct snd_soc_codec *codec) +static bool wm8994_check_class_w_digital(struct snd_soc_codec *codec) { - struct wm8994_priv *wm8994 = snd_soc_codec_get_drvdata(codec); - int enable = 1; int source = 0; /* GCC flow analysis can't track enable */ int reg, reg_r; - /* Only support direct DAC->headphone paths */ - reg = snd_soc_read(codec, WM8994_OUTPUT_MIXER_1); - if (!(reg & WM8994_DAC1L_TO_HPOUT1L)) { - dev_vdbg(codec->dev, "HPL connected to output mixer\n"); - enable = 0; - } - - reg = snd_soc_read(codec, WM8994_OUTPUT_MIXER_2); - if (!(reg & WM8994_DAC1R_TO_HPOUT1R)) { - dev_vdbg(codec->dev, "HPR connected to output mixer\n"); - enable = 0; - } - - /* We also need the same setting for L/R and only one path */ + /* We also need the same AIF source for L/R and only one path */ reg = snd_soc_read(codec, WM8994_DAC1_LEFT_MIXER_ROUTING); switch (reg) { case WM8994_AIF2DACL_TO_DAC1L: @@ -974,30 +967,20 @@ static void wm8994_update_class_w(struct snd_soc_codec *codec) break; default: dev_vdbg(codec->dev, "DAC mixer setting: %x\n", reg); - enable = 0; - break; + return false; } reg_r = snd_soc_read(codec, WM8994_DAC1_RIGHT_MIXER_ROUTING); if (reg_r != reg) { dev_vdbg(codec->dev, "Left and right DAC mixers different\n"); - enable = 0; + return false; } - if (enable) { - dev_dbg(codec->dev, "Class W enabled\n"); - snd_soc_update_bits(codec, WM8994_CLASS_W_1, - WM8994_CP_DYN_PWR | - WM8994_CP_DYN_SRC_SEL_MASK, - source | WM8994_CP_DYN_PWR); - wm8994->hubs.class_w = true; - - } else { - dev_dbg(codec->dev, "Class W disabled\n"); - snd_soc_update_bits(codec, WM8994_CLASS_W_1, - WM8994_CP_DYN_PWR, 0); - wm8994->hubs.class_w = false; - } + /* Set the source up */ + snd_soc_update_bits(codec, WM8994_CLASS_W_1, + WM8994_CP_DYN_SRC_SEL_MASK, source); + + return true; } static int aif1clk_ev(struct snd_soc_dapm_widget *w, @@ -1280,45 +1263,6 @@ static int dac_ev(struct snd_soc_dapm_widget *w, return 0; } -static const char *hp_mux_text[] = { - "Mixer", - "DAC", -}; - -#define WM8994_HP_ENUM(xname, xenum) \ -{ .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = xname, \ - .info = snd_soc_info_enum_double, \ - .get = snd_soc_dapm_get_enum_double, \ - .put = wm8994_put_hp_enum, \ - .private_value = (unsigned long)&xenum } - -static int wm8994_put_hp_enum(struct snd_kcontrol *kcontrol, - struct snd_ctl_elem_value *ucontrol) -{ - struct snd_soc_dapm_widget_list *wlist = snd_kcontrol_chip(kcontrol); - struct snd_soc_dapm_widget *w = wlist->widgets[0]; - struct snd_soc_codec *codec = w->codec; - int ret; - - ret = snd_soc_dapm_put_enum_double(kcontrol, ucontrol); - - wm8994_update_class_w(codec); - - return ret; -} - -static const struct soc_enum hpl_enum = - SOC_ENUM_SINGLE(WM8994_OUTPUT_MIXER_1, 8, 2, hp_mux_text); - -static const struct snd_kcontrol_new hpl_mux = - WM8994_HP_ENUM("Left Headphone Mux", hpl_enum); - -static const struct soc_enum hpr_enum = - SOC_ENUM_SINGLE(WM8994_OUTPUT_MIXER_2, 8, 2, hp_mux_text); - -static const struct snd_kcontrol_new hpr_mux = - WM8994_HP_ENUM("Right Headphone Mux", hpr_enum); - static const char *adc_mux_text[] = { "ADC", "DMIC", @@ -1430,7 +1374,7 @@ static int wm8994_put_class_w(struct snd_kcontrol *kcontrol, ret = snd_soc_dapm_put_volsw(kcontrol, ucontrol); - wm8994_update_class_w(codec); + wm_hubs_update_class_w(codec); return ret; } @@ -1524,7 +1468,7 @@ static const struct snd_kcontrol_new wm8958_aif3adc_mux = SOC_DAPM_ENUM("AIF3ADC Mux", wm8958_aif3adc_enum); static const char *mono_pcm_out_text[] = { - "None", "AIF2ADCL", "AIF2ADCR", + "None", "AIF2ADCL", "AIF2ADCR", }; static const struct soc_enum mono_pcm_out_enum = @@ -1573,9 +1517,9 @@ SND_SOC_DAPM_MIXER_E("SPKL", WM8994_POWER_MANAGEMENT_3, 8, 0, SND_SOC_DAPM_MIXER_E("SPKR", WM8994_POWER_MANAGEMENT_3, 9, 0, right_speaker_mixer, ARRAY_SIZE(right_speaker_mixer), late_enable_ev, SND_SOC_DAPM_PRE_PMU), -SND_SOC_DAPM_MUX_E("Left Headphone Mux", SND_SOC_NOPM, 0, 0, &hpl_mux, +SND_SOC_DAPM_MUX_E("Left Headphone Mux", SND_SOC_NOPM, 0, 0, &wm_hubs_hpl_mux, late_enable_ev, SND_SOC_DAPM_PRE_PMU), -SND_SOC_DAPM_MUX_E("Right Headphone Mux", SND_SOC_NOPM, 0, 0, &hpr_mux, +SND_SOC_DAPM_MUX_E("Right Headphone Mux", SND_SOC_NOPM, 0, 0, &wm_hubs_hpr_mux, late_enable_ev, SND_SOC_DAPM_PRE_PMU), SND_SOC_DAPM_POST("Late Disable PGA", late_disable_ev) @@ -1591,8 +1535,8 @@ SND_SOC_DAPM_MIXER("SPKL", WM8994_POWER_MANAGEMENT_3, 8, 0, left_speaker_mixer, ARRAY_SIZE(left_speaker_mixer)), SND_SOC_DAPM_MIXER("SPKR", WM8994_POWER_MANAGEMENT_3, 9, 0, right_speaker_mixer, ARRAY_SIZE(right_speaker_mixer)), -SND_SOC_DAPM_MUX("Left Headphone Mux", SND_SOC_NOPM, 0, 0, &hpl_mux), -SND_SOC_DAPM_MUX("Right Headphone Mux", SND_SOC_NOPM, 0, 0, &hpr_mux), +SND_SOC_DAPM_MUX("Left Headphone Mux", SND_SOC_NOPM, 0, 0, &wm_hubs_hpl_mux), +SND_SOC_DAPM_MUX("Right Headphone Mux", SND_SOC_NOPM, 0, 0, &wm_hubs_hpr_mux), }; static const struct snd_soc_dapm_widget wm8994_dac_revd_widgets[] = { @@ -1732,6 +1676,7 @@ SND_SOC_DAPM_MUX("AIF3ADC Mux", SND_SOC_NOPM, 0, 0, &wm8994_aif3adc_mux), }; static const struct snd_soc_dapm_widget wm8958_dapm_widgets[] = { +SND_SOC_DAPM_SUPPLY("AIF3", WM8994_POWER_MANAGEMENT_6, 5, 1, NULL, 0), SND_SOC_DAPM_MUX("Mono PCM Out Mux", SND_SOC_NOPM, 0, 0, &mono_pcm_out_mux), SND_SOC_DAPM_MUX("AIF2DACL Mux", SND_SOC_NOPM, 0, 0, &aif2dacl_src_mux), SND_SOC_DAPM_MUX("AIF2DACR Mux", SND_SOC_NOPM, 0, 0, &aif2dacr_src_mux), @@ -1972,6 +1917,9 @@ static const struct snd_soc_dapm_route wm8958_intercon[] = { { "AIF2DACR Mux", "AIF2", "AIF2DAC Mux" }, { "AIF2DACR Mux", "AIF3", "AIF3DACDAT" }, + { "AIF3DACDAT", NULL, "AIF3" }, + { "AIF3ADCDAT", NULL, "AIF3" }, + { "Mono PCM Out Mux", "AIF2ADCL", "AIF2ADCL" }, { "Mono PCM Out Mux", "AIF2ADCR", "AIF2ADCR" }, @@ -2068,24 +2016,20 @@ static int _wm8994_set_fll(struct snd_soc_codec *codec, int id, int src, struct wm8994 *control = wm8994->wm8994; int reg_offset, ret; struct fll_div fll; - u16 reg, aif1, aif2; + u16 reg, clk1, aif_reg, aif_src; unsigned long timeout; bool was_enabled; - aif1 = snd_soc_read(codec, WM8994_AIF1_CLOCKING_1) - & WM8994_AIF1CLK_ENA; - - aif2 = snd_soc_read(codec, WM8994_AIF2_CLOCKING_1) - & WM8994_AIF2CLK_ENA; - switch (id) { case WM8994_FLL1: reg_offset = 0; id = 0; + aif_src = 0x10; break; case WM8994_FLL2: reg_offset = 0x20; id = 1; + aif_src = 0x18; break; default: return -EINVAL; @@ -2127,16 +2071,33 @@ static int _wm8994_set_fll(struct snd_soc_codec *codec, int id, int src, if (ret < 0) return ret; - /* Gate the AIF clocks while we reclock */ - snd_soc_update_bits(codec, WM8994_AIF1_CLOCKING_1, - WM8994_AIF1CLK_ENA, 0); - snd_soc_update_bits(codec, WM8994_AIF2_CLOCKING_1, - WM8994_AIF2CLK_ENA, 0); + /* Make sure that we're not providing SYSCLK right now */ + clk1 = snd_soc_read(codec, WM8994_CLOCKING_1); + if (clk1 & WM8994_SYSCLK_SRC) + aif_reg = WM8994_AIF2_CLOCKING_1; + else + aif_reg = WM8994_AIF1_CLOCKING_1; + reg = snd_soc_read(codec, aif_reg); + + if ((reg & WM8994_AIF1CLK_ENA) && + (reg & WM8994_AIF1CLK_SRC_MASK) == aif_src) { + dev_err(codec->dev, "FLL%d is currently providing SYSCLK\n", + id + 1); + return -EBUSY; + } /* We always need to disable the FLL while reconfiguring */ snd_soc_update_bits(codec, WM8994_FLL1_CONTROL_1 + reg_offset, WM8994_FLL1_ENA, 0); + if (wm8994->fll_byp && src == WM8994_FLL_SRC_BCLK && + freq_in == freq_out && freq_out) { + dev_dbg(codec->dev, "Bypassing FLL%d\n", id + 1); + snd_soc_update_bits(codec, WM8994_FLL1_CONTROL_5 + reg_offset, + WM8958_FLL1_BYP, WM8958_FLL1_BYP); + goto out; + } + reg = (fll.outdiv << WM8994_FLL1_OUTDIV_SHIFT) | (fll.fll_fratio << WM8994_FLL1_FRATIO_SHIFT); snd_soc_update_bits(codec, WM8994_FLL1_CONTROL_2 + reg_offset, @@ -2151,6 +2112,7 @@ static int _wm8994_set_fll(struct snd_soc_codec *codec, int id, int src, fll.n << WM8994_FLL1_N_SHIFT); snd_soc_update_bits(codec, WM8994_FLL1_CONTROL_5 + reg_offset, + WM8958_FLL1_BYP | WM8994_FLL1_REFCLK_DIV_MASK | WM8994_FLL1_REFCLK_SRC_MASK, (fll.clk_ref_div << WM8994_FLL1_REFCLK_DIV_SHIFT) | @@ -2213,16 +2175,11 @@ static int _wm8994_set_fll(struct snd_soc_codec *codec, int id, int src, } } +out: wm8994->fll[id].in = freq_in; wm8994->fll[id].out = freq_out; wm8994->fll[id].src = src; - /* Enable any gated AIF clocks */ - snd_soc_update_bits(codec, WM8994_AIF1_CLOCKING_1, - WM8994_AIF1CLK_ENA, aif1); - snd_soc_update_bits(codec, WM8994_AIF2_CLOCKING_1, - WM8994_AIF2CLK_ENA, aif2); - configure_clock(codec); return 0; @@ -2290,7 +2247,7 @@ static int wm8994_set_dai_sysclk(struct snd_soc_dai *dai, case WM8994_SYSCLK_OPCLK: /* Special case - a division (times 10) is given and - * no effect on main clocking. + * no effect on main clocking. */ if (freq) { for (i = 0; i < ARRAY_SIZE(opclk_divs); i++) @@ -2792,33 +2749,6 @@ static int wm8994_aif3_hw_params(struct snd_pcm_substream *substream, return snd_soc_update_bits(codec, aif1_reg, WM8994_AIF1_WL_MASK, aif1); } -static void wm8994_aif_shutdown(struct snd_pcm_substream *substream, - struct snd_soc_dai *dai) -{ - struct snd_soc_codec *codec = dai->codec; - int rate_reg = 0; - - switch (dai->id) { - case 1: - rate_reg = WM8994_AIF1_RATE; - break; - case 2: - rate_reg = WM8994_AIF2_RATE; - break; - default: - break; - } - - /* If the DAI is idle then configure the divider tree for the - * lowest output rate to save a little power if the clock is - * still active (eg, because it is system clock). - */ - if (rate_reg && !dai->playback_active && !dai->capture_active) - snd_soc_update_bits(codec, rate_reg, - WM8994_AIF1_SR_MASK | - WM8994_AIF1CLK_RATE_MASK, 0x9); -} - static int wm8994_aif_mute(struct snd_soc_dai *codec_dai, int mute) { struct snd_soc_codec *codec = codec_dai->codec; @@ -2860,10 +2790,6 @@ static int wm8994_set_tristate(struct snd_soc_dai *codec_dai, int tristate) reg = WM8994_AIF2_MASTER_SLAVE; mask = WM8994_AIF2_TRI; break; - case 3: - reg = WM8994_POWER_MANAGEMENT_6; - mask = WM8994_AIF3_TRI; - break; default: return -EINVAL; } @@ -2900,7 +2826,6 @@ static const struct snd_soc_dai_ops wm8994_aif1_dai_ops = { .set_sysclk = wm8994_set_dai_sysclk, .set_fmt = wm8994_set_dai_fmt, .hw_params = wm8994_hw_params, - .shutdown = wm8994_aif_shutdown, .digital_mute = wm8994_aif_mute, .set_pll = wm8994_set_fll, .set_tristate = wm8994_set_tristate, @@ -2910,7 +2835,6 @@ static const struct snd_soc_dai_ops wm8994_aif2_dai_ops = { .set_sysclk = wm8994_set_dai_sysclk, .set_fmt = wm8994_set_dai_fmt, .hw_params = wm8994_hw_params, - .shutdown = wm8994_aif_shutdown, .digital_mute = wm8994_aif_mute, .set_pll = wm8994_set_fll, .set_tristate = wm8994_set_tristate, @@ -2918,7 +2842,6 @@ static const struct snd_soc_dai_ops wm8994_aif2_dai_ops = { static const struct snd_soc_dai_ops wm8994_aif3_dai_ops = { .hw_params = wm8994_aif3_hw_params, - .set_tristate = wm8994_set_tristate, }; static struct snd_soc_dai_driver wm8994_dai[] = { @@ -3126,14 +3049,14 @@ static void wm8994_handle_retune_mobile_pdata(struct wm8994_priv *wm8994) /* Expand the array... */ t = krealloc(wm8994->retune_mobile_texts, - sizeof(char *) * + sizeof(char *) * (wm8994->num_retune_mobile_texts + 1), GFP_KERNEL); if (t == NULL) continue; /* ...store the new entry... */ - t[wm8994->num_retune_mobile_texts] = + t[wm8994->num_retune_mobile_texts] = pdata->retune_mobile_cfgs[i].name; /* ...and remember the new version. */ @@ -3304,25 +3227,25 @@ int wm8994_mic_detect(struct snd_soc_codec *codec, struct snd_soc_jack *jack, } EXPORT_SYMBOL_GPL(wm8994_mic_detect); -static irqreturn_t wm8994_mic_irq(int irq, void *data) +static void wm8994_mic_work(struct work_struct *work) { - struct wm8994_priv *priv = data; - struct snd_soc_codec *codec = priv->codec; - int reg; + struct wm8994_priv *priv = container_of(work, + struct wm8994_priv, + mic_work.work); + struct regmap *regmap = priv->wm8994->regmap; + struct device *dev = priv->wm8994->dev; + unsigned int reg; + int ret; int report; -#ifndef CONFIG_SND_SOC_WM8994_MODULE - trace_snd_soc_jack_irq(dev_name(codec->dev)); -#endif - - reg = snd_soc_read(codec, WM8994_INTERRUPT_RAW_STATUS_2); - if (reg < 0) { - dev_err(codec->dev, "Failed to read microphone status: %d\n", - reg); - return IRQ_HANDLED; + ret = regmap_read(regmap, WM8994_INTERRUPT_RAW_STATUS_2, ®); + if (ret < 0) { + dev_err(dev, "Failed to read microphone status: %d\n", + ret); + return; } - dev_dbg(codec->dev, "Microphone status: %x\n", reg); + dev_dbg(dev, "Microphone status: %x\n", reg); report = 0; if (reg & WM8994_MIC1_DET_STS) { @@ -3361,6 +3284,20 @@ static irqreturn_t wm8994_mic_irq(int irq, void *data) snd_soc_jack_report(priv->micdet[1].jack, report, SND_JACK_HEADSET | SND_JACK_BTN_0); +} + +static irqreturn_t wm8994_mic_irq(int irq, void *data) +{ + struct wm8994_priv *priv = data; + struct snd_soc_codec *codec = priv->codec; + +#ifndef CONFIG_SND_SOC_WM8994_MODULE + trace_snd_soc_jack_irq(dev_name(codec->dev)); +#endif + + pm_wakeup_event(codec->dev, 300); + + schedule_delayed_work(&priv->mic_work, msecs_to_jiffies(250)); return IRQ_HANDLED; } @@ -3415,9 +3352,6 @@ static void wm8958_default_micdet(u16 status, void *data) wm8958_micd_set_rate(codec); - snd_soc_jack_report(wm8994->micdet[0].jack, SND_JACK_HEADPHONE, - SND_JACK_HEADSET); - /* If we have jackdet that will detect removal */ if (wm8994->jackdet) { mutex_lock(&wm8994->accdet_lock); @@ -3430,14 +3364,13 @@ static void wm8958_default_micdet(u16 status, void *data) mutex_unlock(&wm8994->accdet_lock); - if (wm8994->pdata->jd_ext_cap) { - mutex_lock(&codec->mutex); + if (wm8994->pdata->jd_ext_cap) snd_soc_dapm_disable_pin(&codec->dapm, "MICBIAS2"); - snd_soc_dapm_sync(&codec->dapm); - mutex_unlock(&codec->mutex); - } } + + snd_soc_jack_report(wm8994->micdet[0].jack, SND_JACK_HEADPHONE, + SND_JACK_HEADSET); } /* Report short circuit as a button */ @@ -3489,6 +3422,8 @@ static irqreturn_t wm1811_jackdet_irq(int irq, void *data) if (present) { dev_dbg(codec->dev, "Jack detected\n"); + wm8958_micd_set_rate(codec); + snd_soc_update_bits(codec, WM8958_MICBIAS2, WM8958_MICB2_DISCH, 0); @@ -3526,16 +3461,11 @@ static irqreturn_t wm1811_jackdet_irq(int irq, void *data) /* If required for an external cap force MICBIAS on */ if (wm8994->pdata->jd_ext_cap) { - mutex_lock(&codec->mutex); - if (present) snd_soc_dapm_force_enable_pin(&codec->dapm, "MICBIAS2"); else snd_soc_dapm_disable_pin(&codec->dapm, "MICBIAS2"); - - snd_soc_dapm_sync(&codec->dapm); - mutex_unlock(&codec->mutex); } if (present) @@ -3740,6 +3670,7 @@ static int wm8994_codec_probe(struct snd_soc_codec *codec) wm8994->codec = codec; mutex_init(&wm8994->accdet_lock); + INIT_DELAYED_WORK(&wm8994->mic_work, wm8994_mic_work); for (i = 0; i < ARRAY_SIZE(wm8994->fll_locked); i++) init_completion(&wm8994->fll_locked[i]); @@ -3783,13 +3714,22 @@ static int wm8994_codec_probe(struct snd_soc_codec *codec) case WM8958: wm8994->hubs.dcs_readback_mode = 1; wm8994->hubs.hp_startup_mode = 1; + + switch (wm8994->revision) { + case 0: + break; + default: + wm8994->fll_byp = true; + break; + } break; case WM1811: wm8994->hubs.dcs_readback_mode = 2; wm8994->hubs.no_series_update = 1; wm8994->hubs.hp_startup_mode = 1; - wm8994->hubs.no_cache_class_w = true; + wm8994->hubs.no_cache_dac_hp_direct = true; + wm8994->fll_byp = true; switch (wm8994->revision) { case 0: @@ -4010,7 +3950,8 @@ static int wm8994_codec_probe(struct snd_soc_codec *codec) break; } - wm8994_update_class_w(codec); + wm8994->hubs.check_class_w_digital = wm8994_check_class_w_digital; + wm_hubs_update_class_w(codec); wm8994_handle_pdata(wm8994); @@ -4075,7 +4016,6 @@ static int wm8994_codec_probe(struct snd_soc_codec *codec) ARRAY_SIZE(wm8994_dac_widgets)); break; } - wm_hubs_add_analogue_routes(codec, 0, 0); snd_soc_dapm_add_routes(dapm, intercon, ARRAY_SIZE(intercon)); @@ -4140,7 +4080,7 @@ err_irq: return ret; } -static int wm8994_codec_remove(struct snd_soc_codec *codec) +static int wm8994_codec_remove(struct snd_soc_codec *codec) { struct wm8994_priv *wm8994 = snd_soc_codec_get_drvdata(codec); struct wm8994 *control = wm8994->wm8994; @@ -4181,14 +4121,10 @@ static int wm8994_codec_remove(struct snd_soc_codec *codec) free_irq(wm8994->micdet_irq, wm8994); break; } - if (wm8994->mbc) - release_firmware(wm8994->mbc); - if (wm8994->mbc_vss) - release_firmware(wm8994->mbc_vss); - if (wm8994->enh_eq) - release_firmware(wm8994->enh_eq); + release_firmware(wm8994->mbc); + release_firmware(wm8994->mbc_vss); + release_firmware(wm8994->enh_eq); kfree(wm8994->retune_mobile_texts); - return 0; } diff --git a/sound/soc/codecs/wm8994.h b/sound/soc/codecs/wm8994.h index c724112998d..d77e06f0a67 100644 --- a/sound/soc/codecs/wm8994.h +++ b/sound/soc/codecs/wm8994.h @@ -12,6 +12,7 @@ #include <sound/soc.h> #include <linux/firmware.h> #include <linux/completion.h> +#include <linux/workqueue.h> #include "wm_hubs.h" @@ -79,6 +80,7 @@ struct wm8994_priv { struct wm8994_fll_config fll[2], fll_suspend[2]; struct completion fll_locked[2]; bool fll_locked_irq; + bool fll_byp; int vmid_refcount; int active_refcount; @@ -126,6 +128,7 @@ struct wm8994_priv { struct mutex accdet_lock; struct wm8994_micdet micdet[2]; + struct delayed_work mic_work; bool mic_detecting; bool jack_mic; int btn_mask; diff --git a/sound/soc/codecs/wm8996.c b/sound/soc/codecs/wm8996.c index 1fd63549404..8af422e38fd 100644 --- a/sound/soc/codecs/wm8996.c +++ b/sound/soc/codecs/wm8996.c @@ -1770,7 +1770,13 @@ static int wm8996_set_bias_level(struct snd_soc_codec *codec, switch (level) { case SND_SOC_BIAS_ON: + break; case SND_SOC_BIAS_PREPARE: + /* Put the MICBIASes into regulating mode */ + snd_soc_update_bits(codec, WM8996_MICBIAS_1, + WM8996_MICB1_MODE, 0); + snd_soc_update_bits(codec, WM8996_MICBIAS_2, + WM8996_MICB2_MODE, 0); break; case SND_SOC_BIAS_STANDBY: @@ -1793,6 +1799,12 @@ static int wm8996_set_bias_level(struct snd_soc_codec *codec, regcache_cache_only(codec->control_data, false); regcache_sync(codec->control_data); } + + /* Bypass the MICBIASes for lowest power */ + snd_soc_update_bits(codec, WM8996_MICBIAS_1, + WM8996_MICB1_MODE, WM8996_MICB1_MODE); + snd_soc_update_bits(codec, WM8996_MICBIAS_2, + WM8996_MICB2_MODE, WM8996_MICB2_MODE); break; case SND_SOC_BIAS_OFF: diff --git a/sound/soc/codecs/wm9081.c b/sound/soc/codecs/wm9081.c index 076c126ed9b..9328270df16 100644 --- a/sound/soc/codecs/wm9081.c +++ b/sound/soc/codecs/wm9081.c @@ -774,7 +774,7 @@ static const struct snd_soc_dapm_widget wm9081_dapm_widgets[] = { SND_SOC_DAPM_INPUT("IN1"), SND_SOC_DAPM_INPUT("IN2"), -SND_SOC_DAPM_DAC("DAC", "HiFi Playback", WM9081_POWER_MANAGEMENT, 0, 0), +SND_SOC_DAPM_DAC("DAC", NULL, WM9081_POWER_MANAGEMENT, 0, 0), SND_SOC_DAPM_MIXER_NAMED_CTL("Mixer", SND_SOC_NOPM, 0, 0, mixer, ARRAY_SIZE(mixer)), @@ -799,6 +799,7 @@ SND_SOC_DAPM_SUPPLY("TSENSE", WM9081_POWER_MANAGEMENT, 7, 0, NULL, 0), static const struct snd_soc_dapm_route wm9081_audio_paths[] = { { "DAC", NULL, "CLK_SYS" }, { "DAC", NULL, "CLK_DSP" }, + { "DAC", NULL, "AIF" }, { "Mixer", "IN1 Switch", "IN1" }, { "Mixer", "IN2 Switch", "IN2" }, @@ -1252,7 +1253,7 @@ static const struct snd_soc_dai_ops wm9081_dai_ops = { static struct snd_soc_dai_driver wm9081_dai = { .name = "wm9081-hifi", .playback = { - .stream_name = "HiFi Playback", + .stream_name = "AIF", .channels_min = 1, .channels_max = 2, .rates = WM9081_RATES, diff --git a/sound/soc/codecs/wm9705.c b/sound/soc/codecs/wm9705.c index cacc6a86b46..e8e782a0c78 100644 --- a/sound/soc/codecs/wm9705.c +++ b/sound/soc/codecs/wm9705.c @@ -236,9 +236,7 @@ static int ac97_write(struct snd_soc_codec *codec, unsigned int reg, static int ac97_prepare(struct snd_pcm_substream *substream, struct snd_soc_dai *dai) { - struct snd_pcm_runtime *runtime = substream->runtime; - struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct snd_soc_codec *codec = rtd->codec; + struct snd_soc_codec *codec = dai->codec; int reg; u16 vra; @@ -250,7 +248,7 @@ static int ac97_prepare(struct snd_pcm_substream *substream, else reg = AC97_PCM_LR_ADC_RATE; - return ac97_write(codec, reg, runtime->rate); + return ac97_write(codec, reg, substream->runtime->rate); } #define WM9705_AC97_RATES (SNDRV_PCM_RATE_8000 | SNDRV_PCM_RATE_11025 | \ diff --git a/sound/soc/codecs/wm9712.c b/sound/soc/codecs/wm9712.c index b342ae50bcd..a1541414d90 100644 --- a/sound/soc/codecs/wm9712.c +++ b/sound/soc/codecs/wm9712.c @@ -467,11 +467,10 @@ static int ac97_write(struct snd_soc_codec *codec, unsigned int reg, static int ac97_prepare(struct snd_pcm_substream *substream, struct snd_soc_dai *dai) { - struct snd_pcm_runtime *runtime = substream->runtime; - struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct snd_soc_codec *codec =rtd->codec; + struct snd_soc_codec *codec = dai->codec; int reg; u16 vra; + struct snd_pcm_runtime *runtime = substream->runtime; vra = ac97_read(codec, AC97_EXTENDED_STATUS); ac97_write(codec, AC97_EXTENDED_STATUS, vra | 0x1); @@ -487,10 +486,9 @@ static int ac97_prepare(struct snd_pcm_substream *substream, static int ac97_aux_prepare(struct snd_pcm_substream *substream, struct snd_soc_dai *dai) { - struct snd_pcm_runtime *runtime = substream->runtime; - struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct snd_soc_codec *codec = rtd->codec; + struct snd_soc_codec *codec = dai->codec; u16 vra, xsle; + struct snd_pcm_runtime *runtime = substream->runtime; vra = ac97_read(codec, AC97_EXTENDED_STATUS); ac97_write(codec, AC97_EXTENDED_STATUS, vra | 0x1); diff --git a/sound/soc/codecs/wm_hubs.c b/sound/soc/codecs/wm_hubs.c index 6c028c47060..dfe957a47f2 100644 --- a/sound/soc/codecs/wm_hubs.c +++ b/sound/soc/codecs/wm_hubs.c @@ -109,12 +109,103 @@ irqreturn_t wm_hubs_dcs_done(int irq, void *data) } EXPORT_SYMBOL_GPL(wm_hubs_dcs_done); +static bool wm_hubs_dac_hp_direct(struct snd_soc_codec *codec) +{ + int reg; + + /* If we're going via the mixer we'll need to do additional checks */ + reg = snd_soc_read(codec, WM8993_OUTPUT_MIXER1); + if (!(reg & WM8993_DACL_TO_HPOUT1L)) { + if (reg & ~WM8993_DACL_TO_MIXOUTL) { + dev_vdbg(codec->dev, "Analogue paths connected: %x\n", + reg & ~WM8993_DACL_TO_HPOUT1L); + return false; + } else { + dev_vdbg(codec->dev, "HPL connected to mixer\n"); + } + } else { + dev_vdbg(codec->dev, "HPL connected to DAC\n"); + } + + reg = snd_soc_read(codec, WM8993_OUTPUT_MIXER2); + if (!(reg & WM8993_DACR_TO_HPOUT1R)) { + if (reg & ~WM8993_DACR_TO_MIXOUTR) { + dev_vdbg(codec->dev, "Analogue paths connected: %x\n", + reg & ~WM8993_DACR_TO_HPOUT1R); + return false; + } else { + dev_vdbg(codec->dev, "HPR connected to mixer\n"); + } + } else { + dev_vdbg(codec->dev, "HPR connected to DAC\n"); + } + + return true; +} + +struct wm_hubs_dcs_cache { + struct list_head list; + unsigned int left; + unsigned int right; + u16 dcs_cfg; +}; + +static bool wm_hubs_dcs_cache_get(struct snd_soc_codec *codec, + struct wm_hubs_dcs_cache **entry) +{ + struct wm_hubs_data *hubs = snd_soc_codec_get_drvdata(codec); + struct wm_hubs_dcs_cache *cache; + unsigned int left, right; + + left = snd_soc_read(codec, WM8993_LEFT_OUTPUT_VOLUME); + left &= WM8993_HPOUT1L_VOL_MASK; + + right = snd_soc_read(codec, WM8993_RIGHT_OUTPUT_VOLUME); + right &= WM8993_HPOUT1R_VOL_MASK; + + list_for_each_entry(cache, &hubs->dcs_cache, list) { + if (cache->left != left || cache->right != right) + continue; + + *entry = cache; + return true; + } + + return false; +} + +static void wm_hubs_dcs_cache_set(struct snd_soc_codec *codec, u16 dcs_cfg) +{ + struct wm_hubs_data *hubs = snd_soc_codec_get_drvdata(codec); + struct wm_hubs_dcs_cache *cache; + + if (hubs->no_cache_dac_hp_direct) + return; + + cache = devm_kzalloc(codec->dev, sizeof(*cache), GFP_KERNEL); + if (!cache) { + dev_err(codec->dev, "Failed to allocate DCS cache entry\n"); + return; + } + + cache->left = snd_soc_read(codec, WM8993_LEFT_OUTPUT_VOLUME); + cache->left &= WM8993_HPOUT1L_VOL_MASK; + + cache->right = snd_soc_read(codec, WM8993_RIGHT_OUTPUT_VOLUME); + cache->right &= WM8993_HPOUT1R_VOL_MASK; + + cache->dcs_cfg = dcs_cfg; + + list_add_tail(&cache->list, &hubs->dcs_cache); +} + /* * Startup calibration of the DC servo */ static void calibrate_dc_servo(struct snd_soc_codec *codec) { struct wm_hubs_data *hubs = snd_soc_codec_get_drvdata(codec); + struct wm_hubs_dcs_cache *cache; s8 offset; u16 reg, reg_l, reg_r, dcs_cfg, dcs_reg; @@ -129,10 +220,11 @@ static void calibrate_dc_servo(struct snd_soc_codec *codec) /* If we're using a digital only path and have a previously * callibrated DC servo offset stored then use that. */ - if (hubs->class_w && hubs->class_w_dcs) { - dev_dbg(codec->dev, "Using cached DC servo offset %x\n", - hubs->class_w_dcs); - snd_soc_write(codec, dcs_reg, hubs->class_w_dcs); + if (wm_hubs_dac_hp_direct(codec) && + wm_hubs_dcs_cache_get(codec, &cache)) { + dev_dbg(codec->dev, "Using cached DCS offset %x for %d,%d\n", + cache->dcs_cfg, cache->left, cache->right); + snd_soc_write(codec, dcs_reg, cache->dcs_cfg); wait_for_dc_servo(codec, WM8993_DCS_TRIG_DAC_WR_0 | WM8993_DCS_TRIG_DAC_WR_1); @@ -207,8 +299,8 @@ static void calibrate_dc_servo(struct snd_soc_codec *codec) /* Save the callibrated offset if we're in class W mode and * therefore don't have any analogue signal mixed in. */ - if (hubs->class_w && !hubs->no_cache_class_w) - hubs->class_w_dcs = dcs_cfg; + if (wm_hubs_dac_hp_direct(codec)) + wm_hubs_dcs_cache_set(codec, dcs_cfg); } /* @@ -223,9 +315,6 @@ static int wm8993_put_dc_servo(struct snd_kcontrol *kcontrol, ret = snd_soc_put_volsw(kcontrol, ucontrol); - /* Updating the analogue gains invalidates the DC servo cache */ - hubs->class_w_dcs = 0; - /* If we're applying an offset correction then updating the * callibration would be likely to introduce further offsets. */ if (hubs->dcs_codes_l || hubs->dcs_codes_r || hubs->no_series_update) @@ -530,6 +619,86 @@ static int lineout_event(struct snd_soc_dapm_widget *w, return 0; } +void wm_hubs_update_class_w(struct snd_soc_codec *codec) +{ + struct wm_hubs_data *hubs = snd_soc_codec_get_drvdata(codec); + int enable = WM8993_CP_DYN_V | WM8993_CP_DYN_FREQ; + + if (!wm_hubs_dac_hp_direct(codec)) + enable = false; + + if (hubs->check_class_w_digital && !hubs->check_class_w_digital(codec)) + enable = false; + + dev_vdbg(codec->dev, "Class W %s\n", enable ? "enabled" : "disabled"); + + snd_soc_update_bits(codec, WM8993_CLASS_W_0, + WM8993_CP_DYN_V | WM8993_CP_DYN_FREQ, enable); +} +EXPORT_SYMBOL_GPL(wm_hubs_update_class_w); + +#define WM_HUBS_SINGLE_W(xname, reg, shift, max, invert) \ +{ .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = xname, \ + .info = snd_soc_info_volsw, \ + .get = snd_soc_dapm_get_volsw, .put = class_w_put_volsw, \ + .private_value = SOC_SINGLE_VALUE(reg, shift, max, invert) } + +static int class_w_put_volsw(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_dapm_widget_list *wlist = snd_kcontrol_chip(kcontrol); + struct snd_soc_dapm_widget *widget = wlist->widgets[0]; + struct snd_soc_codec *codec = widget->codec; + int ret; + + ret = snd_soc_dapm_put_volsw(kcontrol, ucontrol); + + wm_hubs_update_class_w(codec); + + return ret; +} + +#define WM_HUBS_ENUM_W(xname, xenum) \ +{ .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = xname, \ + .info = snd_soc_info_enum_double, \ + .get = snd_soc_dapm_get_enum_double, \ + .put = class_w_put_double, \ + .private_value = (unsigned long)&xenum } + +static int class_w_put_double(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_dapm_widget_list *wlist = snd_kcontrol_chip(kcontrol); + struct snd_soc_dapm_widget *widget = wlist->widgets[0]; + struct snd_soc_codec *codec = widget->codec; + int ret; + + ret = snd_soc_dapm_put_enum_double(kcontrol, ucontrol); + + wm_hubs_update_class_w(codec); + + return ret; +} + +static const char *hp_mux_text[] = { + "Mixer", + "DAC", +}; + +static const struct soc_enum hpl_enum = + SOC_ENUM_SINGLE(WM8993_OUTPUT_MIXER1, 8, 2, hp_mux_text); + +const struct snd_kcontrol_new wm_hubs_hpl_mux = + WM_HUBS_ENUM_W("Left Headphone Mux", hpl_enum); +EXPORT_SYMBOL_GPL(wm_hubs_hpl_mux); + +static const struct soc_enum hpr_enum = + SOC_ENUM_SINGLE(WM8993_OUTPUT_MIXER2, 8, 2, hp_mux_text); + +const struct snd_kcontrol_new wm_hubs_hpr_mux = + WM_HUBS_ENUM_W("Right Headphone Mux", hpr_enum); +EXPORT_SYMBOL_GPL(wm_hubs_hpr_mux); + static const struct snd_kcontrol_new in1l_pga[] = { SOC_DAPM_SINGLE("IN1LP Switch", WM8993_INPUT_MIXER2, 5, 1, 0), SOC_DAPM_SINGLE("IN1LN Switch", WM8993_INPUT_MIXER2, 4, 1, 0), @@ -561,25 +730,25 @@ SOC_DAPM_SINGLE("IN1R Switch", WM8993_INPUT_MIXER4, 5, 1, 0), }; static const struct snd_kcontrol_new left_output_mixer[] = { -SOC_DAPM_SINGLE("Right Input Switch", WM8993_OUTPUT_MIXER1, 7, 1, 0), -SOC_DAPM_SINGLE("Left Input Switch", WM8993_OUTPUT_MIXER1, 6, 1, 0), -SOC_DAPM_SINGLE("IN2RN Switch", WM8993_OUTPUT_MIXER1, 5, 1, 0), -SOC_DAPM_SINGLE("IN2LN Switch", WM8993_OUTPUT_MIXER1, 4, 1, 0), -SOC_DAPM_SINGLE("IN2LP Switch", WM8993_OUTPUT_MIXER1, 1, 1, 0), -SOC_DAPM_SINGLE("IN1R Switch", WM8993_OUTPUT_MIXER1, 3, 1, 0), -SOC_DAPM_SINGLE("IN1L Switch", WM8993_OUTPUT_MIXER1, 2, 1, 0), -SOC_DAPM_SINGLE("DAC Switch", WM8993_OUTPUT_MIXER1, 0, 1, 0), +WM_HUBS_SINGLE_W("Right Input Switch", WM8993_OUTPUT_MIXER1, 7, 1, 0), +WM_HUBS_SINGLE_W("Left Input Switch", WM8993_OUTPUT_MIXER1, 6, 1, 0), +WM_HUBS_SINGLE_W("IN2RN Switch", WM8993_OUTPUT_MIXER1, 5, 1, 0), +WM_HUBS_SINGLE_W("IN2LN Switch", WM8993_OUTPUT_MIXER1, 4, 1, 0), +WM_HUBS_SINGLE_W("IN2LP Switch", WM8993_OUTPUT_MIXER1, 1, 1, 0), +WM_HUBS_SINGLE_W("IN1R Switch", WM8993_OUTPUT_MIXER1, 3, 1, 0), +WM_HUBS_SINGLE_W("IN1L Switch", WM8993_OUTPUT_MIXER1, 2, 1, 0), +WM_HUBS_SINGLE_W("DAC Switch", WM8993_OUTPUT_MIXER1, 0, 1, 0), }; static const struct snd_kcontrol_new right_output_mixer[] = { -SOC_DAPM_SINGLE("Left Input Switch", WM8993_OUTPUT_MIXER2, 7, 1, 0), -SOC_DAPM_SINGLE("Right Input Switch", WM8993_OUTPUT_MIXER2, 6, 1, 0), -SOC_DAPM_SINGLE("IN2LN Switch", WM8993_OUTPUT_MIXER2, 5, 1, 0), -SOC_DAPM_SINGLE("IN2RN Switch", WM8993_OUTPUT_MIXER2, 4, 1, 0), -SOC_DAPM_SINGLE("IN1L Switch", WM8993_OUTPUT_MIXER2, 3, 1, 0), -SOC_DAPM_SINGLE("IN1R Switch", WM8993_OUTPUT_MIXER2, 2, 1, 0), -SOC_DAPM_SINGLE("IN2RP Switch", WM8993_OUTPUT_MIXER2, 1, 1, 0), -SOC_DAPM_SINGLE("DAC Switch", WM8993_OUTPUT_MIXER2, 0, 1, 0), +WM_HUBS_SINGLE_W("Left Input Switch", WM8993_OUTPUT_MIXER2, 7, 1, 0), +WM_HUBS_SINGLE_W("Right Input Switch", WM8993_OUTPUT_MIXER2, 6, 1, 0), +WM_HUBS_SINGLE_W("IN2LN Switch", WM8993_OUTPUT_MIXER2, 5, 1, 0), +WM_HUBS_SINGLE_W("IN2RN Switch", WM8993_OUTPUT_MIXER2, 4, 1, 0), +WM_HUBS_SINGLE_W("IN1L Switch", WM8993_OUTPUT_MIXER2, 3, 1, 0), +WM_HUBS_SINGLE_W("IN1R Switch", WM8993_OUTPUT_MIXER2, 2, 1, 0), +WM_HUBS_SINGLE_W("IN2RP Switch", WM8993_OUTPUT_MIXER2, 1, 1, 0), +WM_HUBS_SINGLE_W("DAC Switch", WM8993_OUTPUT_MIXER2, 0, 1, 0), }; static const struct snd_kcontrol_new earpiece_mixer[] = { @@ -943,6 +1112,7 @@ int wm_hubs_add_analogue_routes(struct snd_soc_codec *codec, struct wm_hubs_data *hubs = snd_soc_codec_get_drvdata(codec); struct snd_soc_dapm_context *dapm = &codec->dapm; + INIT_LIST_HEAD(&hubs->dcs_cache); init_completion(&hubs->dcs_done); snd_soc_dapm_add_routes(dapm, analogue_routes, diff --git a/sound/soc/codecs/wm_hubs.h b/sound/soc/codecs/wm_hubs.h index 5705276f494..da2dc899ce6 100644 --- a/sound/soc/codecs/wm_hubs.h +++ b/sound/soc/codecs/wm_hubs.h @@ -16,6 +16,8 @@ #include <linux/completion.h> #include <linux/interrupt.h> +#include <linux/list.h> +#include <sound/control.h> struct snd_soc_codec; @@ -30,9 +32,9 @@ struct wm_hubs_data { int series_startup; int no_series_update; - bool no_cache_class_w; - bool class_w; - u16 class_w_dcs; + bool no_cache_dac_hp_direct; + struct list_head dcs_cache; + bool (*check_class_w_digital)(struct snd_soc_codec *); bool lineout1_se; bool lineout1n_ena; @@ -58,5 +60,9 @@ extern irqreturn_t wm_hubs_dcs_done(int irq, void *data); extern void wm_hubs_vmid_ena(struct snd_soc_codec *codec); extern void wm_hubs_set_bias_level(struct snd_soc_codec *codec, enum snd_soc_bias_level level); +extern void wm_hubs_update_class_w(struct snd_soc_codec *codec); + +extern const struct snd_kcontrol_new wm_hubs_hpl_mux; +extern const struct snd_kcontrol_new wm_hubs_hpr_mux; #endif diff --git a/sound/soc/ep93xx/ep93xx-ac97.c b/sound/soc/ep93xx/ep93xx-ac97.c index 0678637abd6..bdffab33e16 100644 --- a/sound/soc/ep93xx/ep93xx-ac97.c +++ b/sound/soc/ep93xx/ep93xx-ac97.c @@ -87,17 +87,13 @@ * struct ep93xx_ac97_info - EP93xx AC97 controller info structure * @lock: mutex serializing access to the bus (slot 1 & 2 ops) * @dev: pointer to the platform device dev structure - * @mem: physical memory resource for the registers * @regs: mapped AC97 controller registers - * @irq: AC97 interrupt number * @done: bus ops wait here for an interrupt */ struct ep93xx_ac97_info { struct mutex lock; struct device *dev; - struct resource *mem; void __iomem *regs; - int irq; struct completion done; }; @@ -359,66 +355,50 @@ static struct snd_soc_dai_driver ep93xx_ac97_dai = { static int __devinit ep93xx_ac97_probe(struct platform_device *pdev) { struct ep93xx_ac97_info *info; + struct resource *res; + unsigned int irq; int ret; - info = kzalloc(sizeof(struct ep93xx_ac97_info), GFP_KERNEL); + info = devm_kzalloc(&pdev->dev, sizeof(*info), GFP_KERNEL); if (!info) return -ENOMEM; - dev_set_drvdata(&pdev->dev, info); - - mutex_init(&info->lock); - init_completion(&info->done); - info->dev = &pdev->dev; + res = platform_get_resource(pdev, IORESOURCE_MEM, 0); + if (!res) + return -ENODEV; - info->mem = platform_get_resource(pdev, IORESOURCE_MEM, 0); - if (!info->mem) { - ret = -ENXIO; - goto fail_free_info; - } + info->regs = devm_request_and_ioremap(&pdev->dev, res); + if (!info->regs) + return -ENXIO; - info->irq = platform_get_irq(pdev, 0); - if (!info->irq) { - ret = -ENXIO; - goto fail_free_info; - } + irq = platform_get_irq(pdev, 0); + if (!irq) + return -ENODEV; - if (!request_mem_region(info->mem->start, resource_size(info->mem), - pdev->name)) { - ret = -EBUSY; - goto fail_free_info; - } + ret = devm_request_irq(&pdev->dev, irq, ep93xx_ac97_interrupt, + IRQF_TRIGGER_HIGH, pdev->name, info); + if (ret) + goto fail; - info->regs = ioremap(info->mem->start, resource_size(info->mem)); - if (!info->regs) { - ret = -ENOMEM; - goto fail_release_mem; - } + dev_set_drvdata(&pdev->dev, info); - ret = request_irq(info->irq, ep93xx_ac97_interrupt, IRQF_TRIGGER_HIGH, - pdev->name, info); - if (ret) - goto fail_unmap_mem; + mutex_init(&info->lock); + init_completion(&info->done); + info->dev = &pdev->dev; ep93xx_ac97_info = info; platform_set_drvdata(pdev, info); ret = snd_soc_register_dai(&pdev->dev, &ep93xx_ac97_dai); if (ret) - goto fail_free_irq; + goto fail; return 0; -fail_free_irq: +fail: platform_set_drvdata(pdev, NULL); - free_irq(info->irq, info); -fail_unmap_mem: - iounmap(info->regs); -fail_release_mem: - release_mem_region(info->mem->start, resource_size(info->mem)); -fail_free_info: - kfree(info); - + ep93xx_ac97_info = NULL; + dev_set_drvdata(&pdev->dev, NULL); return ret; } @@ -431,11 +411,9 @@ static int __devexit ep93xx_ac97_remove(struct platform_device *pdev) /* disable the AC97 controller */ ep93xx_ac97_write_reg(info, AC97GCR, 0); - free_irq(info->irq, info); - iounmap(info->regs); - release_mem_region(info->mem->start, resource_size(info->mem)); platform_set_drvdata(pdev, NULL); - kfree(info); + ep93xx_ac97_info = NULL; + dev_set_drvdata(&pdev->dev, NULL); return 0; } diff --git a/sound/soc/ep93xx/ep93xx-i2s.c b/sound/soc/ep93xx/ep93xx-i2s.c index f7a62348e3f..8df8f6dc474 100644 --- a/sound/soc/ep93xx/ep93xx-i2s.c +++ b/sound/soc/ep93xx/ep93xx-i2s.c @@ -63,7 +63,6 @@ struct ep93xx_i2s_info { struct clk *sclk; struct clk *lrclk; struct ep93xx_pcm_dma_params *dma_params; - struct resource *mem; void __iomem *regs; }; @@ -373,38 +372,22 @@ static int ep93xx_i2s_probe(struct platform_device *pdev) struct resource *res; int err; - info = kzalloc(sizeof(struct ep93xx_i2s_info), GFP_KERNEL); - if (!info) { - err = -ENOMEM; - goto fail; - } - - dev_set_drvdata(&pdev->dev, info); - info->dma_params = ep93xx_i2s_dma_params; + info = devm_kzalloc(&pdev->dev, sizeof(*info), GFP_KERNEL); + if (!info) + return -ENOMEM; res = platform_get_resource(pdev, IORESOURCE_MEM, 0); - if (!res) { - err = -ENODEV; - goto fail_free_info; - } + if (!res) + return -ENODEV; - info->mem = request_mem_region(res->start, resource_size(res), - pdev->name); - if (!info->mem) { - err = -EBUSY; - goto fail_free_info; - } - - info->regs = ioremap(info->mem->start, resource_size(info->mem)); - if (!info->regs) { - err = -ENXIO; - goto fail_release_mem; - } + info->regs = devm_request_and_ioremap(&pdev->dev, res); + if (!info->regs) + return -ENXIO; info->mclk = clk_get(&pdev->dev, "mclk"); if (IS_ERR(info->mclk)) { err = PTR_ERR(info->mclk); - goto fail_unmap_mem; + goto fail; } info->sclk = clk_get(&pdev->dev, "sclk"); @@ -419,6 +402,9 @@ static int ep93xx_i2s_probe(struct platform_device *pdev) goto fail_put_sclk; } + dev_set_drvdata(&pdev->dev, info); + info->dma_params = ep93xx_i2s_dma_params; + err = snd_soc_register_dai(&pdev->dev, &ep93xx_i2s_dai); if (err) goto fail_put_lrclk; @@ -426,17 +412,12 @@ static int ep93xx_i2s_probe(struct platform_device *pdev) return 0; fail_put_lrclk: + dev_set_drvdata(&pdev->dev, NULL); clk_put(info->lrclk); fail_put_sclk: clk_put(info->sclk); fail_put_mclk: clk_put(info->mclk); -fail_unmap_mem: - iounmap(info->regs); -fail_release_mem: - release_mem_region(info->mem->start, resource_size(info->mem)); -fail_free_info: - kfree(info); fail: return err; } @@ -446,12 +427,10 @@ static int __devexit ep93xx_i2s_remove(struct platform_device *pdev) struct ep93xx_i2s_info *info = dev_get_drvdata(&pdev->dev); snd_soc_unregister_dai(&pdev->dev); + dev_set_drvdata(&pdev->dev, NULL); clk_put(info->lrclk); clk_put(info->sclk); clk_put(info->mclk); - iounmap(info->regs); - release_mem_region(info->mem->start, resource_size(info->mem)); - kfree(info); return 0; } diff --git a/sound/soc/fsl/Kconfig b/sound/soc/fsl/Kconfig index d754d34d68a..d70133086ac 100644 --- a/sound/soc/fsl/Kconfig +++ b/sound/soc/fsl/Kconfig @@ -1,18 +1,31 @@ -config SND_MPC52xx_DMA +config SND_SOC_FSL_SSI tristate -# ASoC platform support for the Freescale PowerPC SOCs that have an SSI and -# an Elo DMA controller, such as the MPC8610 and P1022. You will still need to -# select a platform driver and a codec driver. -config SND_SOC_POWERPC_SSI +config SND_SOC_FSL_UTILS tristate + +menuconfig SND_POWERPC_SOC + tristate "SoC Audio for Freescale PowerPC CPUs" depends on FSL_SOC + help + Say Y or M if you want to add support for codecs attached to + the PowerPC CPUs. + +if SND_POWERPC_SOC + +config SND_MPC52xx_DMA + tristate + +config SND_SOC_POWERPC_DMA + tristate config SND_SOC_MPC8610_HPCD tristate "ALSA SoC support for the Freescale MPC8610 HPCD board" # I2C is necessary for the CS4270 driver depends on MPC8610_HPCD && I2C - select SND_SOC_POWERPC_SSI + select SND_SOC_FSL_SSI + select SND_SOC_FSL_UTILS + select SND_SOC_POWERPC_DMA select SND_SOC_CS4270 select SND_SOC_CS4270_VD33_ERRATA default y if MPC8610_HPCD @@ -23,7 +36,9 @@ config SND_SOC_P1022_DS tristate "ALSA SoC support for the Freescale P1022 DS board" # I2C is necessary for the WM8776 driver depends on P1022_DS && I2C - select SND_SOC_POWERPC_SSI + select SND_SOC_FSL_SSI + select SND_SOC_FSL_UTILS + select SND_SOC_POWERPC_DMA select SND_SOC_WM8776 default y if P1022_DS help @@ -65,3 +80,103 @@ config SND_MPC52xx_SOC_EFIKA help Say Y if you want to add support for sound on the Efika. +endif # SND_POWERPC_SOC + +menuconfig SND_IMX_SOC + tristate "SoC Audio for Freescale i.MX CPUs" + depends on ARCH_MXC + help + Say Y or M if you want to add support for codecs attached to + the i.MX CPUs. + +if SND_IMX_SOC + +config SND_SOC_IMX_SSI + tristate + +config SND_SOC_IMX_PCM + tristate + +config SND_SOC_IMX_PCM_FIQ + tristate + select FIQ + select SND_SOC_IMX_PCM + +config SND_SOC_IMX_PCM_DMA + tristate + select SND_SOC_DMAENGINE_PCM + select SND_SOC_IMX_PCM + +config SND_SOC_IMX_AUDMUX + tristate + +config SND_MXC_SOC_WM1133_EV1 + tristate "Audio on the i.MX31ADS with WM1133-EV1 fitted" + depends on MACH_MX31ADS_WM1133_EV1 && EXPERIMENTAL + select SND_SOC_WM8350 + select SND_SOC_IMX_PCM_FIQ + select SND_SOC_IMX_AUDMUX + select SND_SOC_IMX_SSI + help + Enable support for audio on the i.MX31ADS with the WM1133-EV1 + PMIC board with WM8835x fitted. + +config SND_SOC_MX27VIS_AIC32X4 + tristate "SoC audio support for Visstrim M10 boards" + depends on MACH_IMX27_VISSTRIM_M10 && I2C + select SND_SOC_TLV320AIC32X4 + select SND_SOC_IMX_PCM_DMA + select SND_SOC_IMX_AUDMUX + select SND_SOC_IMX_SSI + help + Say Y if you want to add support for SoC audio on Visstrim SM10 + board with TLV320AIC32X4 codec. + +config SND_SOC_PHYCORE_AC97 + tristate "SoC Audio support for Phytec phyCORE (and phyCARD) boards" + depends on MACH_PCM043 || MACH_PCA100 + select SND_SOC_AC97_BUS + select SND_SOC_WM9712 + select SND_SOC_IMX_PCM_FIQ + select SND_SOC_IMX_AUDMUX + select SND_SOC_IMX_SSI + help + Say Y if you want to add support for SoC audio on Phytec phyCORE + and phyCARD boards in AC97 mode + +config SND_SOC_EUKREA_TLV320 + tristate "Eukrea TLV320" + depends on MACH_EUKREA_MBIMX27_BASEBOARD \ + || MACH_EUKREA_MBIMXSD25_BASEBOARD \ + || MACH_EUKREA_MBIMXSD35_BASEBOARD \ + || MACH_EUKREA_MBIMXSD51_BASEBOARD + depends on I2C + select SND_SOC_TLV320AIC23 + select SND_SOC_IMX_PCM_FIQ + select SND_SOC_IMX_AUDMUX + select SND_SOC_IMX_SSI + help + Enable I2S based access to the TLV320AIC23B codec attached + to the SSI interface + +config SND_SOC_IMX_SGTL5000 + tristate "SoC Audio support for i.MX boards with sgtl5000" + depends on OF && I2C + select SND_SOC_SGTL5000 + select SND_SOC_IMX_PCM_DMA + select SND_SOC_IMX_AUDMUX + select SND_SOC_FSL_SSI + select SND_SOC_FSL_UTILS + help + Say Y if you want to add support for SoC audio on an i.MX board with + a sgtl5000 codec. + +config SND_SOC_IMX_MC13783 + tristate "SoC Audio support for I.MX boards with mc13783" + depends on MFD_MC13783 + select SND_SOC_IMX_SSI + select SND_SOC_IMX_AUDMUX + select SND_SOC_MC13783 + select SND_SOC_IMX_PCM_DMA + +endif # SND_IMX_SOC diff --git a/sound/soc/fsl/Makefile b/sound/soc/fsl/Makefile index b4a38c0ac58..5f3cf3f52ea 100644 --- a/sound/soc/fsl/Makefile +++ b/sound/soc/fsl/Makefile @@ -8,8 +8,11 @@ obj-$(CONFIG_SND_SOC_P1022_DS) += snd-soc-p1022-ds.o # Freescale PowerPC SSI/DMA Platform Support snd-soc-fsl-ssi-objs := fsl_ssi.o +snd-soc-fsl-utils-objs := fsl_utils.o snd-soc-fsl-dma-objs := fsl_dma.o -obj-$(CONFIG_SND_SOC_POWERPC_SSI) += snd-soc-fsl-ssi.o snd-soc-fsl-dma.o +obj-$(CONFIG_SND_SOC_FSL_SSI) += snd-soc-fsl-ssi.o +obj-$(CONFIG_SND_SOC_FSL_UTILS) += snd-soc-fsl-utils.o +obj-$(CONFIG_SND_SOC_POWERPC_DMA) += snd-soc-fsl-dma.o # MPC5200 Platform Support obj-$(CONFIG_SND_MPC52xx_DMA) += mpc5200_dma.o @@ -20,3 +23,29 @@ obj-$(CONFIG_SND_SOC_MPC5200_AC97) += mpc5200_psc_ac97.o obj-$(CONFIG_SND_MPC52xx_SOC_PCM030) += pcm030-audio-fabric.o obj-$(CONFIG_SND_MPC52xx_SOC_EFIKA) += efika-audio-fabric.o +# i.MX Platform Support +snd-soc-imx-ssi-objs := imx-ssi.o +snd-soc-imx-audmux-objs := imx-audmux.o + +obj-$(CONFIG_SND_SOC_IMX_SSI) += snd-soc-imx-ssi.o +obj-$(CONFIG_SND_SOC_IMX_AUDMUX) += snd-soc-imx-audmux.o + +obj-$(CONFIG_SND_SOC_IMX_PCM) += snd-soc-imx-pcm.o +snd-soc-imx-pcm-y := imx-pcm.o +snd-soc-imx-pcm-$(CONFIG_SND_SOC_IMX_PCM_FIQ) += imx-pcm-fiq.o +snd-soc-imx-pcm-$(CONFIG_SND_SOC_IMX_PCM_DMA) += imx-pcm-dma.o + +# i.MX Machine Support +snd-soc-eukrea-tlv320-objs := eukrea-tlv320.o +snd-soc-phycore-ac97-objs := phycore-ac97.o +snd-soc-mx27vis-aic32x4-objs := mx27vis-aic32x4.o +snd-soc-wm1133-ev1-objs := wm1133-ev1.o +snd-soc-imx-sgtl5000-objs := imx-sgtl5000.o +snd-soc-imx-mc13783-objs := imx-mc13783.o + +obj-$(CONFIG_SND_SOC_EUKREA_TLV320) += snd-soc-eukrea-tlv320.o +obj-$(CONFIG_SND_SOC_PHYCORE_AC97) += snd-soc-phycore-ac97.o +obj-$(CONFIG_SND_SOC_MX27VIS_AIC32X4) += snd-soc-mx27vis-aic32x4.o +obj-$(CONFIG_SND_MXC_SOC_WM1133_EV1) += snd-soc-wm1133-ev1.o +obj-$(CONFIG_SND_SOC_IMX_SGTL5000) += snd-soc-imx-sgtl5000.o +obj-$(CONFIG_SND_SOC_IMX_MC13783) += snd-soc-imx-mc13783.o diff --git a/sound/soc/imx/eukrea-tlv320.c b/sound/soc/fsl/eukrea-tlv320.c index 7d4475cfdb2..efb9ede0120 100644 --- a/sound/soc/imx/eukrea-tlv320.c +++ b/sound/soc/fsl/eukrea-tlv320.c @@ -7,7 +7,7 @@ * which is Copyright 2009 Simtec Electronics * and on sound/soc/imx/phycore-ac97.c which is * Copyright 2009 Sascha Hauer, Pengutronix <s.hauer@pengutronix.de> - * + * * This program is free software; you can redistribute it and/or modify it * under the terms of the GNU General Public License as published by the * Free Software Foundation; either version 2 of the License, or (at your diff --git a/sound/soc/fsl/fsl_ssi.c b/sound/soc/fsl/fsl_ssi.c index 2eb407fa3b4..4ed2afd4778 100644 --- a/sound/soc/fsl/fsl_ssi.c +++ b/sound/soc/fsl/fsl_ssi.c @@ -11,11 +11,15 @@ */ #include <linux/init.h> +#include <linux/io.h> #include <linux/module.h> #include <linux/interrupt.h> +#include <linux/clk.h> #include <linux/device.h> #include <linux/delay.h> #include <linux/slab.h> +#include <linux/of_address.h> +#include <linux/of_irq.h> #include <linux/of_platform.h> #include <sound/core.h> @@ -25,6 +29,26 @@ #include <sound/soc.h> #include "fsl_ssi.h" +#include "imx-pcm.h" + +#ifdef PPC +#define read_ssi(addr) in_be32(addr) +#define write_ssi(val, addr) out_be32(addr, val) +#define write_ssi_mask(addr, clear, set) clrsetbits_be32(addr, clear, set) +#elif defined ARM +#define read_ssi(addr) readl(addr) +#define write_ssi(val, addr) writel(val, addr) +/* + * FIXME: Proper locking should be added at write_ssi_mask caller level + * to ensure this register read/modify/write sequence is race free. + */ +static inline void write_ssi_mask(u32 __iomem *addr, u32 clear, u32 set) +{ + u32 val = readl(addr); + val = (val & ~clear) | set; + writel(val, addr); +} +#endif /** * FSLSSI_I2S_RATES: sample rates supported by the I2S @@ -94,6 +118,13 @@ struct fsl_ssi_private { struct device_attribute dev_attr; struct platform_device *pdev; + bool new_binding; + bool ssi_on_imx; + struct clk *clk; + struct platform_device *imx_pcm_pdev; + struct imx_pcm_dma_params dma_params_tx; + struct imx_pcm_dma_params dma_params_rx; + struct { unsigned int rfrc; unsigned int tfrc; @@ -145,7 +176,7 @@ static irqreturn_t fsl_ssi_isr(int irq, void *dev_id) were interrupted for. We mask it with the Interrupt Enable register so that we only check for events that we're interested in. */ - sisr = in_be32(&ssi->sisr) & SIER_FLAGS; + sisr = read_ssi(&ssi->sisr) & SIER_FLAGS; if (sisr & CCSR_SSI_SISR_RFRC) { ssi_private->stats.rfrc++; @@ -260,7 +291,7 @@ static irqreturn_t fsl_ssi_isr(int irq, void *dev_id) /* Clear the bits that we set */ if (sisr2) - out_be32(&ssi->sisr, sisr2); + write_ssi(sisr2, &ssi->sisr); return ret; } @@ -295,7 +326,7 @@ static int fsl_ssi_startup(struct snd_pcm_substream *substream, * SSI needs to be disabled before updating the registers we set * here. */ - clrbits32(&ssi->scr, CCSR_SSI_SCR_SSIEN); + write_ssi_mask(&ssi->scr, CCSR_SSI_SCR_SSIEN, 0); /* * Program the SSI into I2S Slave Non-Network Synchronous mode. @@ -303,20 +334,18 @@ static int fsl_ssi_startup(struct snd_pcm_substream *substream, * * FIXME: Little-endian samples require a different shift dir */ - clrsetbits_be32(&ssi->scr, + write_ssi_mask(&ssi->scr, CCSR_SSI_SCR_I2S_MODE_MASK | CCSR_SSI_SCR_SYN, CCSR_SSI_SCR_TFR_CLK_DIS | CCSR_SSI_SCR_I2S_MODE_SLAVE | (synchronous ? CCSR_SSI_SCR_SYN : 0)); - out_be32(&ssi->stcr, - CCSR_SSI_STCR_TXBIT0 | CCSR_SSI_STCR_TFEN0 | + write_ssi(CCSR_SSI_STCR_TXBIT0 | CCSR_SSI_STCR_TFEN0 | CCSR_SSI_STCR_TFSI | CCSR_SSI_STCR_TEFS | - CCSR_SSI_STCR_TSCKP); + CCSR_SSI_STCR_TSCKP, &ssi->stcr); - out_be32(&ssi->srcr, - CCSR_SSI_SRCR_RXBIT0 | CCSR_SSI_SRCR_RFEN0 | + write_ssi(CCSR_SSI_SRCR_RXBIT0 | CCSR_SSI_SRCR_RFEN0 | CCSR_SSI_SRCR_RFSI | CCSR_SSI_SRCR_REFS | - CCSR_SSI_SRCR_RSCKP); + CCSR_SSI_SRCR_RSCKP, &ssi->srcr); /* * The DC and PM bits are only used if the SSI is the clock @@ -324,7 +353,7 @@ static int fsl_ssi_startup(struct snd_pcm_substream *substream, */ /* Enable the interrupts and DMA requests */ - out_be32(&ssi->sier, SIER_FLAGS); + write_ssi(SIER_FLAGS, &ssi->sier); /* * Set the watermark for transmit FIFI 0 and receive FIFO 0. We @@ -339,9 +368,9 @@ static int fsl_ssi_startup(struct snd_pcm_substream *substream, * make this value larger (and maybe we should), but this way * data will be written to memory as soon as it's available. */ - out_be32(&ssi->sfcsr, - CCSR_SSI_SFCSR_TFWM0(ssi_private->fifo_depth - 2) | - CCSR_SSI_SFCSR_RFWM0(ssi_private->fifo_depth - 2)); + write_ssi(CCSR_SSI_SFCSR_TFWM0(ssi_private->fifo_depth - 2) | + CCSR_SSI_SFCSR_RFWM0(ssi_private->fifo_depth - 2), + &ssi->sfcsr); /* * We keep the SSI disabled because if we enable it, then the @@ -393,6 +422,12 @@ static int fsl_ssi_startup(struct snd_pcm_substream *substream, ssi_private->second_stream = substream; } + if (ssi_private->ssi_on_imx) + snd_soc_dai_set_dma_data(dai, substream, + (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) ? + &ssi_private->dma_params_tx : + &ssi_private->dma_params_rx); + return 0; } @@ -417,7 +452,7 @@ static int fsl_ssi_hw_params(struct snd_pcm_substream *substream, unsigned int sample_size = snd_pcm_format_width(params_format(hw_params)); u32 wl = CCSR_SSI_SxCCR_WL(sample_size); - int enabled = in_be32(&ssi->scr) & CCSR_SSI_SCR_SSIEN; + int enabled = read_ssi(&ssi->scr) & CCSR_SSI_SCR_SSIEN; /* * If we're in synchronous mode, and the SSI is already enabled, @@ -439,9 +474,9 @@ static int fsl_ssi_hw_params(struct snd_pcm_substream *substream, /* In synchronous mode, the SSI uses STCCR for capture */ if ((substream->stream == SNDRV_PCM_STREAM_PLAYBACK) || ssi_private->cpu_dai_drv.symmetric_rates) - clrsetbits_be32(&ssi->stccr, CCSR_SSI_SxCCR_WL_MASK, wl); + write_ssi_mask(&ssi->stccr, CCSR_SSI_SxCCR_WL_MASK, wl); else - clrsetbits_be32(&ssi->srccr, CCSR_SSI_SxCCR_WL_MASK, wl); + write_ssi_mask(&ssi->srccr, CCSR_SSI_SxCCR_WL_MASK, wl); return 0; } @@ -466,19 +501,19 @@ static int fsl_ssi_trigger(struct snd_pcm_substream *substream, int cmd, case SNDRV_PCM_TRIGGER_START: case SNDRV_PCM_TRIGGER_PAUSE_RELEASE: if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) - setbits32(&ssi->scr, + write_ssi_mask(&ssi->scr, 0, CCSR_SSI_SCR_SSIEN | CCSR_SSI_SCR_TE); else - setbits32(&ssi->scr, + write_ssi_mask(&ssi->scr, 0, CCSR_SSI_SCR_SSIEN | CCSR_SSI_SCR_RE); break; case SNDRV_PCM_TRIGGER_STOP: case SNDRV_PCM_TRIGGER_PAUSE_PUSH: if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) - clrbits32(&ssi->scr, CCSR_SSI_SCR_TE); + write_ssi_mask(&ssi->scr, CCSR_SSI_SCR_TE, 0); else - clrbits32(&ssi->scr, CCSR_SSI_SCR_RE); + write_ssi_mask(&ssi->scr, CCSR_SSI_SCR_RE, 0); break; default: @@ -510,7 +545,7 @@ static void fsl_ssi_shutdown(struct snd_pcm_substream *substream, if (!ssi_private->first_stream) { struct ccsr_ssi __iomem *ssi = ssi_private->ssi; - clrbits32(&ssi->scr, CCSR_SSI_SCR_SSIEN); + write_ssi_mask(&ssi->scr, CCSR_SSI_SCR_SSIEN, 0); } } @@ -622,12 +657,6 @@ static int __devinit fsl_ssi_probe(struct platform_device *pdev) if (!of_device_is_available(np)) return -ENODEV; - /* Check for a codec-handle property. */ - if (!of_get_property(np, "codec-handle", NULL)) { - dev_err(&pdev->dev, "missing codec-handle property\n"); - return -ENODEV; - } - /* We only support the SSI in "I2S Slave" mode */ sprop = of_get_property(np, "fsl,mode", NULL); if (!sprop || strcmp(sprop, "i2s-slave")) { @@ -692,6 +721,50 @@ static int __devinit fsl_ssi_probe(struct platform_device *pdev) /* Older 8610 DTs didn't have the fifo-depth property */ ssi_private->fifo_depth = 8; + if (of_device_is_compatible(pdev->dev.of_node, "fsl,imx21-ssi")) { + u32 dma_events[2]; + ssi_private->ssi_on_imx = true; + + ssi_private->clk = clk_get(&pdev->dev, NULL); + if (IS_ERR(ssi_private->clk)) { + ret = PTR_ERR(ssi_private->clk); + dev_err(&pdev->dev, "could not get clock: %d\n", ret); + goto error_irq; + } + clk_prepare_enable(ssi_private->clk); + + /* + * We have burstsize be "fifo_depth - 2" to match the SSI + * watermark setting in fsl_ssi_startup(). + */ + ssi_private->dma_params_tx.burstsize = + ssi_private->fifo_depth - 2; + ssi_private->dma_params_rx.burstsize = + ssi_private->fifo_depth - 2; + ssi_private->dma_params_tx.dma_addr = + ssi_private->ssi_phys + offsetof(struct ccsr_ssi, stx0); + ssi_private->dma_params_rx.dma_addr = + ssi_private->ssi_phys + offsetof(struct ccsr_ssi, srx0); + /* + * TODO: This is a temporary solution and should be changed + * to use generic DMA binding later when the helplers get in. + */ + ret = of_property_read_u32_array(pdev->dev.of_node, + "fsl,ssi-dma-events", dma_events, 2); + if (ret) { + dev_err(&pdev->dev, "could not get dma events\n"); + goto error_clk; + } + ssi_private->dma_params_tx.dma = dma_events[0]; + ssi_private->dma_params_rx.dma = dma_events[1]; + + ssi_private->dma_params_tx.shared_peripheral = + of_device_is_compatible(of_get_parent(np), + "fsl,spba-bus"); + ssi_private->dma_params_rx.shared_peripheral = + ssi_private->dma_params_tx.shared_peripheral; + } + /* Initialize the the device_attribute structure */ dev_attr = &ssi_private->dev_attr; sysfs_attr_init(&dev_attr->attr); @@ -715,6 +788,26 @@ static int __devinit fsl_ssi_probe(struct platform_device *pdev) goto error_dev; } + if (ssi_private->ssi_on_imx) { + ssi_private->imx_pcm_pdev = + platform_device_register_simple("imx-pcm-audio", + -1, NULL, 0); + if (IS_ERR(ssi_private->imx_pcm_pdev)) { + ret = PTR_ERR(ssi_private->imx_pcm_pdev); + goto error_dev; + } + } + + /* + * If codec-handle property is missing from SSI node, we assume + * that the machine driver uses new binding which does not require + * SSI driver to trigger machine driver's probe. + */ + if (!of_get_property(np, "codec-handle", NULL)) { + ssi_private->new_binding = true; + goto done; + } + /* Trigger the machine driver's probe function. The platform driver * name of the machine driver is taken from /compatible property of the * device tree. We also pass the address of the CPU DAI driver @@ -736,15 +829,24 @@ static int __devinit fsl_ssi_probe(struct platform_device *pdev) goto error_dai; } +done: return 0; error_dai: + if (ssi_private->ssi_on_imx) + platform_device_unregister(ssi_private->imx_pcm_pdev); snd_soc_unregister_dai(&pdev->dev); error_dev: dev_set_drvdata(&pdev->dev, NULL); device_remove_file(&pdev->dev, dev_attr); +error_clk: + if (ssi_private->ssi_on_imx) { + clk_disable_unprepare(ssi_private->clk); + clk_put(ssi_private->clk); + } + error_irq: free_irq(ssi_private->irq, ssi_private); @@ -764,7 +866,13 @@ static int fsl_ssi_remove(struct platform_device *pdev) { struct fsl_ssi_private *ssi_private = dev_get_drvdata(&pdev->dev); - platform_device_unregister(ssi_private->pdev); + if (!ssi_private->new_binding) + platform_device_unregister(ssi_private->pdev); + if (ssi_private->ssi_on_imx) { + platform_device_unregister(ssi_private->imx_pcm_pdev); + clk_disable_unprepare(ssi_private->clk); + clk_put(ssi_private->clk); + } snd_soc_unregister_dai(&pdev->dev); device_remove_file(&pdev->dev, &ssi_private->dev_attr); @@ -779,6 +887,7 @@ static int fsl_ssi_remove(struct platform_device *pdev) static const struct of_device_id fsl_ssi_ids[] = { { .compatible = "fsl,mpc8610-ssi", }, + { .compatible = "fsl,imx21-ssi", }, {} }; MODULE_DEVICE_TABLE(of, fsl_ssi_ids); diff --git a/sound/soc/fsl/fsl_utils.c b/sound/soc/fsl/fsl_utils.c new file mode 100644 index 00000000000..b9e42b503a3 --- /dev/null +++ b/sound/soc/fsl/fsl_utils.c @@ -0,0 +1,91 @@ +/** + * Freescale ALSA SoC Machine driver utility + * + * Author: Timur Tabi <timur@freescale.com> + * + * Copyright 2010 Freescale Semiconductor, Inc. + * + * This file is licensed under the terms of the GNU General Public License + * version 2. This program is licensed "as is" without any warranty of any + * kind, whether express or implied. + */ + +#include <linux/module.h> +#include <linux/of_address.h> +#include <sound/soc.h> + +#include "fsl_utils.h" + +/** + * fsl_asoc_get_dma_channel - determine the dma channel for a SSI node + * + * @ssi_np: pointer to the SSI device tree node + * @name: name of the phandle pointing to the dma channel + * @dai: ASoC DAI link pointer to be filled with platform_name + * @dma_channel_id: dma channel id to be returned + * @dma_id: dma id to be returned + * + * This function determines the dma and channel id for given SSI node. It + * also discovers the platform_name for the ASoC DAI link. + */ +int fsl_asoc_get_dma_channel(struct device_node *ssi_np, + const char *name, + struct snd_soc_dai_link *dai, + unsigned int *dma_channel_id, + unsigned int *dma_id) +{ + struct resource res; + struct device_node *dma_channel_np, *dma_np; + const u32 *iprop; + int ret; + + dma_channel_np = of_parse_phandle(ssi_np, name, 0); + if (!dma_channel_np) + return -EINVAL; + + if (!of_device_is_compatible(dma_channel_np, "fsl,ssi-dma-channel")) { + of_node_put(dma_channel_np); + return -EINVAL; + } + + /* Determine the dev_name for the device_node. This code mimics the + * behavior of of_device_make_bus_id(). We need this because ASoC uses + * the dev_name() of the device to match the platform (DMA) device with + * the CPU (SSI) device. It's all ugly and hackish, but it works (for + * now). + * + * dai->platform name should already point to an allocated buffer. + */ + ret = of_address_to_resource(dma_channel_np, 0, &res); + if (ret) { + of_node_put(dma_channel_np); + return ret; + } + snprintf((char *)dai->platform_name, DAI_NAME_SIZE, "%llx.%s", + (unsigned long long) res.start, dma_channel_np->name); + + iprop = of_get_property(dma_channel_np, "cell-index", NULL); + if (!iprop) { + of_node_put(dma_channel_np); + return -EINVAL; + } + *dma_channel_id = be32_to_cpup(iprop); + + dma_np = of_get_parent(dma_channel_np); + iprop = of_get_property(dma_np, "cell-index", NULL); + if (!iprop) { + of_node_put(dma_np); + return -EINVAL; + } + *dma_id = be32_to_cpup(iprop); + + of_node_put(dma_np); + of_node_put(dma_channel_np); + + return 0; +} +EXPORT_SYMBOL(fsl_asoc_get_dma_channel); + +MODULE_AUTHOR("Timur Tabi <timur@freescale.com>"); +MODULE_DESCRIPTION("Freescale ASoC utility code"); +MODULE_LICENSE("GPL v2"); diff --git a/sound/soc/fsl/fsl_utils.h b/sound/soc/fsl/fsl_utils.h new file mode 100644 index 00000000000..b2951126527 --- /dev/null +++ b/sound/soc/fsl/fsl_utils.h @@ -0,0 +1,26 @@ +/** + * Freescale ALSA SoC Machine driver utility + * + * Author: Timur Tabi <timur@freescale.com> + * + * Copyright 2010 Freescale Semiconductor, Inc. + * + * This file is licensed under the terms of the GNU General Public License + * version 2. This program is licensed "as is" without any warranty of any + * kind, whether express or implied. + */ + +#ifndef _FSL_UTILS_H +#define _FSL_UTILS_H + +#define DAI_NAME_SIZE 32 + +struct snd_soc_dai_link; +struct device_node; + +int fsl_asoc_get_dma_channel(struct device_node *ssi_np, const char *name, + struct snd_soc_dai_link *dai, + unsigned int *dma_channel_id, + unsigned int *dma_id); + +#endif /* _FSL_UTILS_H */ diff --git a/sound/soc/imx/imx-audmux.c b/sound/soc/fsl/imx-audmux.c index f23700359c6..f23700359c6 100644 --- a/sound/soc/imx/imx-audmux.c +++ b/sound/soc/fsl/imx-audmux.c diff --git a/sound/soc/imx/imx-audmux.h b/sound/soc/fsl/imx-audmux.h index 04ebbab8d7b..04ebbab8d7b 100644 --- a/sound/soc/imx/imx-audmux.h +++ b/sound/soc/fsl/imx-audmux.h diff --git a/sound/soc/fsl/imx-mc13783.c b/sound/soc/fsl/imx-mc13783.c new file mode 100644 index 00000000000..f59c3494366 --- /dev/null +++ b/sound/soc/fsl/imx-mc13783.c @@ -0,0 +1,156 @@ +/* + * imx-mc13783.c -- SoC audio for imx based boards with mc13783 codec + * + * Copyright 2012 Philippe Retornaz, <philippe.retornaz@epfl.ch> + * + * Heavly based on phycore-mc13783: + * Copyright 2009 Sascha Hauer, Pengutronix <s.hauer@pengutronix.de> + * + * This program is free software; you can redistribute it and/or modify it + * under the terms of the GNU General Public License as published by the + * Free Software Foundation; either version 2 of the License, or (at your + * option) any later version. + * + */ + +#include <linux/module.h> +#include <linux/moduleparam.h> +#include <linux/device.h> +#include <sound/core.h> +#include <sound/pcm.h> +#include <sound/soc.h> +#include <sound/soc-dapm.h> +#include <asm/mach-types.h> + +#include "../codecs/mc13783.h" +#include "imx-ssi.h" +#include "imx-audmux.h" + +#define FMT_SSI (SND_SOC_DAIFMT_DSP_A | SND_SOC_DAIFMT_NB_NF | \ + SND_SOC_DAIFMT_CBM_CFM) + +static int imx_mc13783_hifi_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_soc_dai *cpu_dai = rtd->cpu_dai; + struct snd_soc_dai *codec_dai = rtd->codec_dai; + int ret; + + ret = snd_soc_dai_set_tdm_slot(codec_dai, 0xfffffffc, 0xfffffffc, + 4, 16); + if (ret) + return ret; + + ret = snd_soc_dai_set_sysclk(codec_dai, MC13783_CLK_CLIA, 26000000, 0); + if (ret) + return ret; + + ret = snd_soc_dai_set_tdm_slot(cpu_dai, 0x0, 0xfffffffc, 2, 16); + if (ret) + return ret; + + return 0; +} + +static struct snd_soc_ops imx_mc13783_hifi_ops = { + .hw_params = imx_mc13783_hifi_hw_params, +}; + +static struct snd_soc_dai_link imx_mc13783_dai_mc13783[] = { + { + .name = "MC13783", + .stream_name = "Sound", + .codec_dai_name = "mc13783-hifi", + .codec_name = "mc13783-codec", + .cpu_dai_name = "imx-ssi.0", + .platform_name = "imx-pcm-audio.0", + .ops = &imx_mc13783_hifi_ops, + .symmetric_rates = 1, + .dai_fmt = FMT_SSI, + }, +}; + +static const struct snd_soc_dapm_widget imx_mc13783_widget[] = { + SND_SOC_DAPM_MIC("Mic", NULL), + SND_SOC_DAPM_HP("Headphone", NULL), + SND_SOC_DAPM_SPK("Speaker", NULL), +}; + +static const struct snd_soc_dapm_route imx_mc13783_routes[] = { + {"Speaker", NULL, "LSP"}, + {"Headphone", NULL, "HSL"}, + {"Headphone", NULL, "HSR"}, + + {"MC1LIN", NULL, "MC1 Bias"}, + {"MC2IN", NULL, "MC2 Bias"}, + {"MC1 Bias", NULL, "Mic"}, + {"MC2 Bias", NULL, "Mic"}, +}; + +static struct snd_soc_card imx_mc13783 = { + .name = "imx_mc13783", + .dai_link = imx_mc13783_dai_mc13783, + .num_links = ARRAY_SIZE(imx_mc13783_dai_mc13783), + .dapm_widgets = imx_mc13783_widget, + .num_dapm_widgets = ARRAY_SIZE(imx_mc13783_widget), + .dapm_routes = imx_mc13783_routes, + .num_dapm_routes = ARRAY_SIZE(imx_mc13783_routes), +}; + +static int __devinit imx_mc13783_probe(struct platform_device *pdev) +{ + int ret; + + imx_mc13783.dev = &pdev->dev; + + ret = snd_soc_register_card(&imx_mc13783); + if (ret) { + dev_err(&pdev->dev, "snd_soc_register_card failed (%d)\n", + ret); + return ret; + } + + imx_audmux_v2_configure_port(MX31_AUDMUX_PORT4_SSI_PINS_4, + IMX_AUDMUX_V2_PTCR_SYN, + IMX_AUDMUX_V2_PDCR_RXDSEL(MX31_AUDMUX_PORT1_SSI0) | + IMX_AUDMUX_V2_PDCR_MODE(1) | + IMX_AUDMUX_V2_PDCR_INMMASK(0xfc)); + imx_audmux_v2_configure_port(MX31_AUDMUX_PORT1_SSI0, + IMX_AUDMUX_V2_PTCR_SYN | + IMX_AUDMUX_V2_PTCR_TFSDIR | + IMX_AUDMUX_V2_PTCR_TFSEL(MX31_AUDMUX_PORT4_SSI_PINS_4) | + IMX_AUDMUX_V2_PTCR_TCLKDIR | + IMX_AUDMUX_V2_PTCR_TCSEL(MX31_AUDMUX_PORT4_SSI_PINS_4) | + IMX_AUDMUX_V2_PTCR_RFSDIR | + IMX_AUDMUX_V2_PTCR_RFSEL(MX31_AUDMUX_PORT4_SSI_PINS_4) | + IMX_AUDMUX_V2_PTCR_RCLKDIR | + IMX_AUDMUX_V2_PTCR_RCSEL(MX31_AUDMUX_PORT4_SSI_PINS_4), + IMX_AUDMUX_V2_PDCR_RXDSEL(MX31_AUDMUX_PORT4_SSI_PINS_4)); + + return ret; +} + +static int __devexit imx_mc13783_remove(struct platform_device *pdev) +{ + snd_soc_unregister_card(&imx_mc13783); + + return 0; +} + +static struct platform_driver imx_mc13783_audio_driver = { + .driver = { + .name = "imx_mc13783", + .owner = THIS_MODULE, + }, + .probe = imx_mc13783_probe, + .remove = __devexit_p(imx_mc13783_remove) +}; + +module_platform_driver(imx_mc13783_audio_driver); + +MODULE_AUTHOR("Sascha Hauer <s.hauer@pengutronix.de>"); +MODULE_AUTHOR("Philippe Retornaz <philippe.retornaz@epfl.ch"); +MODULE_DESCRIPTION("imx with mc13783 codec ALSA SoC driver"); +MODULE_LICENSE("GPL"); +MODULE_ALIAS("platform:imx_mc13783"); diff --git a/sound/soc/imx/imx-pcm-dma-mx2.c b/sound/soc/fsl/imx-pcm-dma.c index 6b818de2fc0..f3c0a5ef35c 100644 --- a/sound/soc/imx/imx-pcm-dma-mx2.c +++ b/sound/soc/fsl/imx-pcm-dma.c @@ -109,7 +109,8 @@ static int snd_imx_open(struct snd_pcm_substream *substream) dma_params = snd_soc_dai_get_dma_data(rtd->cpu_dai, substream); dma_data = kzalloc(sizeof(*dma_data), GFP_KERNEL); - dma_data->peripheral_type = IMX_DMATYPE_SSI; + dma_data->peripheral_type = dma_params->shared_peripheral ? + IMX_DMATYPE_SSI_SP : IMX_DMATYPE_SSI; dma_data->priority = DMA_PRIO_HIGH; dma_data->dma_request = dma_params->dma; diff --git a/sound/soc/imx/imx-pcm-fiq.c b/sound/soc/fsl/imx-pcm-fiq.c index 456b7d723d6..456b7d723d6 100644 --- a/sound/soc/imx/imx-pcm-fiq.c +++ b/sound/soc/fsl/imx-pcm-fiq.c diff --git a/sound/soc/imx/imx-pcm.c b/sound/soc/fsl/imx-pcm.c index 93dc360b177..93dc360b177 100644 --- a/sound/soc/imx/imx-pcm.c +++ b/sound/soc/fsl/imx-pcm.c diff --git a/sound/soc/imx/imx-pcm.h b/sound/soc/fsl/imx-pcm.h index b5f5c3acf34..83c0ed7d55c 100644 --- a/sound/soc/imx/imx-pcm.h +++ b/sound/soc/fsl/imx-pcm.h @@ -22,6 +22,7 @@ struct imx_pcm_dma_params { int dma; unsigned long dma_addr; int burstsize; + bool shared_peripheral; /* The peripheral is on SPBA bus */ }; int snd_imx_pcm_mmap(struct snd_pcm_substream *substream, diff --git a/sound/soc/fsl/imx-sgtl5000.c b/sound/soc/fsl/imx-sgtl5000.c new file mode 100644 index 00000000000..3a729caeb8c --- /dev/null +++ b/sound/soc/fsl/imx-sgtl5000.c @@ -0,0 +1,221 @@ +/* + * Copyright 2012 Freescale Semiconductor, Inc. + * Copyright 2012 Linaro Ltd. + * + * The code contained herein is licensed under the GNU General Public + * License. You may obtain a copy of the GNU General Public License + * Version 2 or later at the following locations: + * + * http://www.opensource.org/licenses/gpl-license.html + * http://www.gnu.org/copyleft/gpl.html + */ + +#include <linux/module.h> +#include <linux/of.h> +#include <linux/of_platform.h> +#include <linux/of_i2c.h> +#include <linux/clk.h> +#include <sound/soc.h> + +#include "../codecs/sgtl5000.h" +#include "imx-audmux.h" + +#define DAI_NAME_SIZE 32 + +struct imx_sgtl5000_data { + struct snd_soc_dai_link dai; + struct snd_soc_card card; + char codec_dai_name[DAI_NAME_SIZE]; + char platform_name[DAI_NAME_SIZE]; + struct clk *codec_clk; + unsigned int clk_frequency; +}; + +static int imx_sgtl5000_dai_init(struct snd_soc_pcm_runtime *rtd) +{ + struct imx_sgtl5000_data *data = container_of(rtd->card, + struct imx_sgtl5000_data, card); + struct device *dev = rtd->card->dev; + int ret; + + ret = snd_soc_dai_set_sysclk(rtd->codec_dai, SGTL5000_SYSCLK, + data->clk_frequency, SND_SOC_CLOCK_IN); + if (ret) { + dev_err(dev, "could not set codec driver clock params\n"); + return ret; + } + + return 0; +} + +static const struct snd_soc_dapm_widget imx_sgtl5000_dapm_widgets[] = { + SND_SOC_DAPM_MIC("Mic Jack", NULL), + SND_SOC_DAPM_LINE("Line In Jack", NULL), + SND_SOC_DAPM_HP("Headphone Jack", NULL), + SND_SOC_DAPM_SPK("Line Out Jack", NULL), + SND_SOC_DAPM_SPK("Ext Spk", NULL), +}; + +static int __devinit imx_sgtl5000_probe(struct platform_device *pdev) +{ + struct device_node *np = pdev->dev.of_node; + struct device_node *ssi_np, *codec_np; + struct platform_device *ssi_pdev; + struct i2c_client *codec_dev; + struct imx_sgtl5000_data *data; + int int_port, ext_port; + int ret; + + ret = of_property_read_u32(np, "mux-int-port", &int_port); + if (ret) { + dev_err(&pdev->dev, "mux-int-port missing or invalid\n"); + return ret; + } + ret = of_property_read_u32(np, "mux-ext-port", &ext_port); + if (ret) { + dev_err(&pdev->dev, "mux-ext-port missing or invalid\n"); + return ret; + } + + /* + * The port numbering in the hardware manual starts at 1, while + * the audmux API expects it starts at 0. + */ + int_port--; + ext_port--; + ret = imx_audmux_v2_configure_port(int_port, + IMX_AUDMUX_V2_PTCR_SYN | + IMX_AUDMUX_V2_PTCR_TFSEL(ext_port) | + IMX_AUDMUX_V2_PTCR_TCSEL(ext_port) | + IMX_AUDMUX_V2_PTCR_TFSDIR | + IMX_AUDMUX_V2_PTCR_TCLKDIR, + IMX_AUDMUX_V2_PDCR_RXDSEL(ext_port)); + if (ret) { + dev_err(&pdev->dev, "audmux internal port setup failed\n"); + return ret; + } + imx_audmux_v2_configure_port(ext_port, + IMX_AUDMUX_V2_PTCR_SYN | + IMX_AUDMUX_V2_PTCR_TCSEL(int_port), + IMX_AUDMUX_V2_PDCR_RXDSEL(int_port)); + if (ret) { + dev_err(&pdev->dev, "audmux external port setup failed\n"); + return ret; + } + + ssi_np = of_parse_phandle(pdev->dev.of_node, "ssi-controller", 0); + codec_np = of_parse_phandle(pdev->dev.of_node, "audio-codec", 0); + if (!ssi_np || !codec_np) { + dev_err(&pdev->dev, "phandle missing or invalid\n"); + ret = -EINVAL; + goto fail; + } + + ssi_pdev = of_find_device_by_node(ssi_np); + if (!ssi_pdev) { + dev_err(&pdev->dev, "failed to find SSI platform device\n"); + ret = -EINVAL; + goto fail; + } + codec_dev = of_find_i2c_device_by_node(codec_np); + if (!codec_dev) { + dev_err(&pdev->dev, "failed to find codec platform device\n"); + return -EINVAL; + } + + data = devm_kzalloc(&pdev->dev, sizeof(*data), GFP_KERNEL); + if (!data) { + ret = -ENOMEM; + goto fail; + } + + data->codec_clk = clk_get(&codec_dev->dev, NULL); + if (IS_ERR(data->codec_clk)) { + /* assuming clock enabled by default */ + data->codec_clk = NULL; + ret = of_property_read_u32(codec_np, "clock-frequency", + &data->clk_frequency); + if (ret) { + dev_err(&codec_dev->dev, + "clock-frequency missing or invalid\n"); + goto fail; + } + } else { + data->clk_frequency = clk_get_rate(data->codec_clk); + clk_prepare_enable(data->codec_clk); + } + + data->dai.name = "HiFi"; + data->dai.stream_name = "HiFi"; + data->dai.codec_dai_name = "sgtl5000"; + data->dai.codec_of_node = codec_np; + data->dai.cpu_dai_name = dev_name(&ssi_pdev->dev); + data->dai.platform_name = "imx-pcm-audio"; + data->dai.init = &imx_sgtl5000_dai_init; + data->dai.dai_fmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF | + SND_SOC_DAIFMT_CBM_CFM; + + data->card.dev = &pdev->dev; + ret = snd_soc_of_parse_card_name(&data->card, "model"); + if (ret) + goto clk_fail; + ret = snd_soc_of_parse_audio_routing(&data->card, "audio-routing"); + if (ret) + goto clk_fail; + data->card.num_links = 1; + data->card.dai_link = &data->dai; + data->card.dapm_widgets = imx_sgtl5000_dapm_widgets; + data->card.num_dapm_widgets = ARRAY_SIZE(imx_sgtl5000_dapm_widgets); + + ret = snd_soc_register_card(&data->card); + if (ret) { + dev_err(&pdev->dev, "snd_soc_register_card failed (%d)\n", ret); + goto clk_fail; + } + + platform_set_drvdata(pdev, data); +clk_fail: + clk_put(data->codec_clk); +fail: + if (ssi_np) + of_node_put(ssi_np); + if (codec_np) + of_node_put(codec_np); + + return ret; +} + +static int __devexit imx_sgtl5000_remove(struct platform_device *pdev) +{ + struct imx_sgtl5000_data *data = platform_get_drvdata(pdev); + + if (data->codec_clk) { + clk_disable_unprepare(data->codec_clk); + clk_put(data->codec_clk); + } + snd_soc_unregister_card(&data->card); + + return 0; +} + +static const struct of_device_id imx_sgtl5000_dt_ids[] = { + { .compatible = "fsl,imx-audio-sgtl5000", }, + { /* sentinel */ } +}; +MODULE_DEVICE_TABLE(of, imx_sgtl5000_dt_ids); + +static struct platform_driver imx_sgtl5000_driver = { + .driver = { + .name = "imx-sgtl5000", + .owner = THIS_MODULE, + .of_match_table = imx_sgtl5000_dt_ids, + }, + .probe = imx_sgtl5000_probe, + .remove = __devexit_p(imx_sgtl5000_remove), +}; +module_platform_driver(imx_sgtl5000_driver); + +MODULE_AUTHOR("Shawn Guo <shawn.guo@linaro.org>"); +MODULE_DESCRIPTION("Freescale i.MX SGTL5000 ASoC machine driver"); +MODULE_LICENSE("GPL v2"); +MODULE_ALIAS("platform:imx-sgtl5000"); diff --git a/sound/soc/imx/imx-ssi.c b/sound/soc/fsl/imx-ssi.c index 4f81ed45632..cf3ed0362c9 100644 --- a/sound/soc/imx/imx-ssi.c +++ b/sound/soc/fsl/imx-ssi.c @@ -28,7 +28,7 @@ * value. When we read the same register two times (and the register still * contains the same value) these status bits are not set. We work * around this by not polling these bits but only wait a fixed delay. - * + * */ #include <linux/clk.h> diff --git a/sound/soc/imx/imx-ssi.h b/sound/soc/fsl/imx-ssi.h index 5744e86ca87..5744e86ca87 100644 --- a/sound/soc/imx/imx-ssi.h +++ b/sound/soc/fsl/imx-ssi.h diff --git a/sound/soc/fsl/mpc8610_hpcd.c b/sound/soc/fsl/mpc8610_hpcd.c index 3fea5a15ffe..60bcba1bc30 100644 --- a/sound/soc/fsl/mpc8610_hpcd.c +++ b/sound/soc/fsl/mpc8610_hpcd.c @@ -14,18 +14,16 @@ #include <linux/interrupt.h> #include <linux/of_device.h> #include <linux/slab.h> -#include <linux/of_i2c.h> #include <sound/soc.h> #include <asm/fsl_guts.h> #include "fsl_dma.h" #include "fsl_ssi.h" +#include "fsl_utils.h" /* There's only one global utilities register */ static phys_addr_t guts_phys; -#define DAI_NAME_SIZE 32 - /** * mpc8610_hpcd_data: machine-specific ASoC device data * @@ -43,7 +41,6 @@ struct mpc8610_hpcd_data { unsigned int dma_id[2]; /* 0 = DMA1, 1 = DMA2, etc */ unsigned int dma_channel_id[2]; /* 0 = ch 0, 1 = ch 1, etc*/ char codec_dai_name[DAI_NAME_SIZE]; - char codec_name[DAI_NAME_SIZE]; char platform_name[2][DAI_NAME_SIZE]; /* One for each DMA channel */ }; @@ -181,141 +178,6 @@ static struct snd_soc_ops mpc8610_hpcd_ops = { }; /** - * get_node_by_phandle_name - get a node by its phandle name - * - * This function takes a node, the name of a property in that node, and a - * compatible string. Assuming the property is a phandle to another node, - * it returns that node, (optionally) if that node is compatible. - * - * If the property is not a phandle, or the node it points to is not compatible - * with the specific string, then NULL is returned. - */ -static struct device_node *get_node_by_phandle_name(struct device_node *np, - const char *name, - const char *compatible) -{ - const phandle *ph; - int len; - - ph = of_get_property(np, name, &len); - if (!ph || (len != sizeof(phandle))) - return NULL; - - np = of_find_node_by_phandle(*ph); - if (!np) - return NULL; - - if (compatible && !of_device_is_compatible(np, compatible)) { - of_node_put(np); - return NULL; - } - - return np; -} - -/** - * get_parent_cell_index -- return the cell-index of the parent of a node - * - * Return the value of the cell-index property of the parent of the given - * node. This is used for DMA channel nodes that need to know the DMA ID - * of the controller they are on. - */ -static int get_parent_cell_index(struct device_node *np) -{ - struct device_node *parent = of_get_parent(np); - const u32 *iprop; - - if (!parent) - return -1; - - iprop = of_get_property(parent, "cell-index", NULL); - of_node_put(parent); - - if (!iprop) - return -1; - - return be32_to_cpup(iprop); -} - -/** - * codec_node_dev_name - determine the dev_name for a codec node - * - * This function determines the dev_name for an I2C node. This is the name - * that would be returned by dev_name() if this device_node were part of a - * 'struct device' It's ugly and hackish, but it works. - * - * The dev_name for such devices include the bus number and I2C address. For - * example, "cs4270.0-004f". - */ -static int codec_node_dev_name(struct device_node *np, char *buf, size_t len) -{ - const u32 *iprop; - int addr; - char temp[DAI_NAME_SIZE]; - struct i2c_client *i2c; - - of_modalias_node(np, temp, DAI_NAME_SIZE); - - iprop = of_get_property(np, "reg", NULL); - if (!iprop) - return -EINVAL; - - addr = be32_to_cpup(iprop); - - /* We need the adapter number */ - i2c = of_find_i2c_device_by_node(np); - if (!i2c) - return -ENODEV; - - snprintf(buf, len, "%s.%u-%04x", temp, i2c->adapter->nr, addr); - - return 0; -} - -static int get_dma_channel(struct device_node *ssi_np, - const char *name, - struct snd_soc_dai_link *dai, - unsigned int *dma_channel_id, - unsigned int *dma_id) -{ - struct resource res; - struct device_node *dma_channel_np; - const u32 *iprop; - int ret; - - dma_channel_np = get_node_by_phandle_name(ssi_np, name, - "fsl,ssi-dma-channel"); - if (!dma_channel_np) - return -EINVAL; - - /* Determine the dev_name for the device_node. This code mimics the - * behavior of of_device_make_bus_id(). We need this because ASoC uses - * the dev_name() of the device to match the platform (DMA) device with - * the CPU (SSI) device. It's all ugly and hackish, but it works (for - * now). - * - * dai->platform name should already point to an allocated buffer. - */ - ret = of_address_to_resource(dma_channel_np, 0, &res); - if (ret) - return ret; - snprintf((char *)dai->platform_name, DAI_NAME_SIZE, "%llx.%s", - (unsigned long long) res.start, dma_channel_np->name); - - iprop = of_get_property(dma_channel_np, "cell-index", NULL); - if (!iprop) { - of_node_put(dma_channel_np); - return -EINVAL; - } - - *dma_channel_id = be32_to_cpup(iprop); - *dma_id = get_parent_cell_index(dma_channel_np); - of_node_put(dma_channel_np); - - return 0; -} - -/** * mpc8610_hpcd_probe: platform probe function for the machine driver * * Although this is a machine driver, the SSI node is the "master" node with @@ -352,16 +214,8 @@ static int mpc8610_hpcd_probe(struct platform_device *pdev) machine_data->dai[0].cpu_dai_name = dev_name(&ssi_pdev->dev); machine_data->dai[0].ops = &mpc8610_hpcd_ops; - /* Determine the codec name, it will be used as the codec DAI name */ - ret = codec_node_dev_name(codec_np, machine_data->codec_name, - DAI_NAME_SIZE); - if (ret) { - dev_err(&pdev->dev, "invalid codec node %s\n", - codec_np->full_name); - ret = -EINVAL; - goto error; - } - machine_data->dai[0].codec_name = machine_data->codec_name; + /* ASoC core can match codec with device node */ + machine_data->dai[0].codec_of_node = codec_np; /* The DAI name from the codec (snd_soc_dai_driver.name) */ machine_data->dai[0].codec_dai_name = "cs4270-hifi"; @@ -458,9 +312,10 @@ static int mpc8610_hpcd_probe(struct platform_device *pdev) /* Find the playback DMA channel to use. */ machine_data->dai[0].platform_name = machine_data->platform_name[0]; - ret = get_dma_channel(np, "fsl,playback-dma", &machine_data->dai[0], - &machine_data->dma_channel_id[0], - &machine_data->dma_id[0]); + ret = fsl_asoc_get_dma_channel(np, "fsl,playback-dma", + &machine_data->dai[0], + &machine_data->dma_channel_id[0], + &machine_data->dma_id[0]); if (ret) { dev_err(&pdev->dev, "missing/invalid playback DMA phandle\n"); goto error; @@ -468,9 +323,10 @@ static int mpc8610_hpcd_probe(struct platform_device *pdev) /* Find the capture DMA channel to use. */ machine_data->dai[1].platform_name = machine_data->platform_name[1]; - ret = get_dma_channel(np, "fsl,capture-dma", &machine_data->dai[1], - &machine_data->dma_channel_id[1], - &machine_data->dma_id[1]); + ret = fsl_asoc_get_dma_channel(np, "fsl,capture-dma", + &machine_data->dai[1], + &machine_data->dma_channel_id[1], + &machine_data->dma_id[1]); if (ret) { dev_err(&pdev->dev, "missing/invalid capture DMA phandle\n"); goto error; diff --git a/sound/soc/imx/mx27vis-aic32x4.c b/sound/soc/fsl/mx27vis-aic32x4.c index f6d04ad4bb3..f6d04ad4bb3 100644 --- a/sound/soc/imx/mx27vis-aic32x4.c +++ b/sound/soc/fsl/mx27vis-aic32x4.c diff --git a/sound/soc/fsl/p1022_ds.c b/sound/soc/fsl/p1022_ds.c index 982a1c94498..50adf4032bc 100644 --- a/sound/soc/fsl/p1022_ds.c +++ b/sound/soc/fsl/p1022_ds.c @@ -14,12 +14,12 @@ #include <linux/interrupt.h> #include <linux/of_device.h> #include <linux/slab.h> -#include <linux/of_i2c.h> #include <sound/soc.h> #include <asm/fsl_guts.h> #include "fsl_dma.h" #include "fsl_ssi.h" +#include "fsl_utils.h" /* P1022-specific PMUXCR and DMUXCR bit definitions */ @@ -57,8 +57,6 @@ static inline void guts_set_dmuxcr(struct ccsr_guts __iomem *guts, /* There's only one global utilities register */ static phys_addr_t guts_phys; -#define DAI_NAME_SIZE 32 - /** * machine_data: machine-specific ASoC device data * @@ -75,7 +73,6 @@ struct machine_data { unsigned int ssi_id; /* 0 = SSI1, 1 = SSI2, etc */ unsigned int dma_id[2]; /* 0 = DMA1, 1 = DMA2, etc */ unsigned int dma_channel_id[2]; /* 0 = ch 0, 1 = ch 1, etc*/ - char codec_name[DAI_NAME_SIZE]; char platform_name[2][DAI_NAME_SIZE]; /* One for each DMA channel */ }; @@ -191,136 +188,6 @@ static struct snd_soc_ops p1022_ds_ops = { }; /** - * get_node_by_phandle_name - get a node by its phandle name - * - * This function takes a node, the name of a property in that node, and a - * compatible string. Assuming the property is a phandle to another node, - * it returns that node, (optionally) if that node is compatible. - * - * If the property is not a phandle, or the node it points to is not compatible - * with the specific string, then NULL is returned. - */ -static struct device_node *get_node_by_phandle_name(struct device_node *np, - const char *name, const char *compatible) -{ - np = of_parse_phandle(np, name, 0); - if (!np) - return NULL; - - if (!of_device_is_compatible(np, compatible)) { - of_node_put(np); - return NULL; - } - - return np; -} - -/** - * get_parent_cell_index -- return the cell-index of the parent of a node - * - * Return the value of the cell-index property of the parent of the given - * node. This is used for DMA channel nodes that need to know the DMA ID - * of the controller they are on. - */ -static int get_parent_cell_index(struct device_node *np) -{ - struct device_node *parent = of_get_parent(np); - const u32 *iprop; - int ret = -1; - - if (!parent) - return -1; - - iprop = of_get_property(parent, "cell-index", NULL); - if (iprop) - ret = be32_to_cpup(iprop); - - of_node_put(parent); - - return ret; -} - -/** - * codec_node_dev_name - determine the dev_name for a codec node - * - * This function determines the dev_name for an I2C node. This is the name - * that would be returned by dev_name() if this device_node were part of a - * 'struct device' It's ugly and hackish, but it works. - * - * The dev_name for such devices include the bus number and I2C address. For - * example, "cs4270-codec.0-004f". - */ -static int codec_node_dev_name(struct device_node *np, char *buf, size_t len) -{ - const u32 *iprop; - int addr; - char temp[DAI_NAME_SIZE]; - struct i2c_client *i2c; - - of_modalias_node(np, temp, DAI_NAME_SIZE); - - iprop = of_get_property(np, "reg", NULL); - if (!iprop) - return -EINVAL; - - addr = be32_to_cpup(iprop); - - /* We need the adapter number */ - i2c = of_find_i2c_device_by_node(np); - if (!i2c) - return -ENODEV; - - snprintf(buf, len, "%s.%u-%04x", temp, i2c->adapter->nr, addr); - - return 0; -} - -static int get_dma_channel(struct device_node *ssi_np, - const char *name, - struct snd_soc_dai_link *dai, - unsigned int *dma_channel_id, - unsigned int *dma_id) -{ - struct resource res; - struct device_node *dma_channel_np; - const u32 *iprop; - int ret; - - dma_channel_np = get_node_by_phandle_name(ssi_np, name, - "fsl,ssi-dma-channel"); - if (!dma_channel_np) - return -EINVAL; - - /* Determine the dev_name for the device_node. This code mimics the - * behavior of of_device_make_bus_id(). We need this because ASoC uses - * the dev_name() of the device to match the platform (DMA) device with - * the CPU (SSI) device. It's all ugly and hackish, but it works (for - * now). - * - * dai->platform name should already point to an allocated buffer. - */ - ret = of_address_to_resource(dma_channel_np, 0, &res); - if (ret) { - of_node_put(dma_channel_np); - return ret; - } - snprintf((char *)dai->platform_name, DAI_NAME_SIZE, "%llx.%s", - (unsigned long long) res.start, dma_channel_np->name); - - iprop = of_get_property(dma_channel_np, "cell-index", NULL); - if (!iprop) { - of_node_put(dma_channel_np); - return -EINVAL; - } - - *dma_channel_id = be32_to_cpup(iprop); - *dma_id = get_parent_cell_index(dma_channel_np); - of_node_put(dma_channel_np); - - return 0; -} - -/** * p1022_ds_probe: platform probe function for the machine driver * * Although this is a machine driver, the SSI node is the "master" node with @@ -357,15 +224,8 @@ static int p1022_ds_probe(struct platform_device *pdev) mdata->dai[0].cpu_dai_name = dev_name(&ssi_pdev->dev); mdata->dai[0].ops = &p1022_ds_ops; - /* Determine the codec name, it will be used as the codec DAI name */ - ret = codec_node_dev_name(codec_np, mdata->codec_name, DAI_NAME_SIZE); - if (ret) { - dev_err(&pdev->dev, "invalid codec node %s\n", - codec_np->full_name); - ret = -EINVAL; - goto error; - } - mdata->dai[0].codec_name = mdata->codec_name; + /* ASoC core can match codec with device node */ + mdata->dai[0].codec_of_node = codec_np; /* We register two DAIs per SSI, one for playback and the other for * capture. We support codecs that have separate DAIs for both playback @@ -462,9 +322,9 @@ static int p1022_ds_probe(struct platform_device *pdev) /* Find the playback DMA channel to use. */ mdata->dai[0].platform_name = mdata->platform_name[0]; - ret = get_dma_channel(np, "fsl,playback-dma", &mdata->dai[0], - &mdata->dma_channel_id[0], - &mdata->dma_id[0]); + ret = fsl_asoc_get_dma_channel(np, "fsl,playback-dma", &mdata->dai[0], + &mdata->dma_channel_id[0], + &mdata->dma_id[0]); if (ret) { dev_err(&pdev->dev, "missing/invalid playback DMA phandle\n"); goto error; @@ -472,9 +332,9 @@ static int p1022_ds_probe(struct platform_device *pdev) /* Find the capture DMA channel to use. */ mdata->dai[1].platform_name = mdata->platform_name[1]; - ret = get_dma_channel(np, "fsl,capture-dma", &mdata->dai[1], - &mdata->dma_channel_id[1], - &mdata->dma_id[1]); + ret = fsl_asoc_get_dma_channel(np, "fsl,capture-dma", &mdata->dai[1], + &mdata->dma_channel_id[1], + &mdata->dma_id[1]); if (ret) { dev_err(&pdev->dev, "missing/invalid capture DMA phandle\n"); goto error; diff --git a/sound/soc/imx/phycore-ac97.c b/sound/soc/fsl/phycore-ac97.c index f8da6dd115e..f8da6dd115e 100644 --- a/sound/soc/imx/phycore-ac97.c +++ b/sound/soc/fsl/phycore-ac97.c diff --git a/sound/soc/imx/wm1133-ev1.c b/sound/soc/fsl/wm1133-ev1.c index fe54a69073e..fe54a69073e 100644 --- a/sound/soc/imx/wm1133-ev1.c +++ b/sound/soc/fsl/wm1133-ev1.c diff --git a/sound/soc/generic/Kconfig b/sound/soc/generic/Kconfig new file mode 100644 index 00000000000..610f6125164 --- /dev/null +++ b/sound/soc/generic/Kconfig @@ -0,0 +1,4 @@ +config SND_SIMPLE_CARD + tristate "ASoC Simple sound card support" + help + This option enables generic simple sound card support diff --git a/sound/soc/generic/Makefile b/sound/soc/generic/Makefile new file mode 100644 index 00000000000..9c3b246792b --- /dev/null +++ b/sound/soc/generic/Makefile @@ -0,0 +1,3 @@ +snd-soc-simple-card-objs := simple-card.o + +obj-$(CONFIG_SND_SIMPLE_CARD) += snd-soc-simple-card.o diff --git a/sound/soc/generic/simple-card.c b/sound/soc/generic/simple-card.c new file mode 100644 index 00000000000..b4b4cab3023 --- /dev/null +++ b/sound/soc/generic/simple-card.c @@ -0,0 +1,114 @@ +/* + * ASoC simple sound card support + * + * Copyright (C) 2012 Renesas Solutions Corp. + * Kuninori Morimoto <kuninori.morimoto.gx@renesas.com> + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License version 2 as + * published by the Free Software Foundation. + */ + +#include <linux/platform_device.h> +#include <linux/module.h> +#include <sound/simple_card.h> + +#define asoc_simple_get_card_info(p) \ + container_of(p->dai_link, struct asoc_simple_card_info, snd_link) + +static int asoc_simple_card_dai_init(struct snd_soc_pcm_runtime *rtd) +{ + struct asoc_simple_card_info *cinfo = asoc_simple_get_card_info(rtd); + struct asoc_simple_dai_init_info *iinfo = cinfo->init; + struct snd_soc_dai *codec = rtd->codec_dai; + struct snd_soc_dai *cpu = rtd->cpu_dai; + unsigned int cpu_daifmt = iinfo->fmt | iinfo->cpu_daifmt; + unsigned int codec_daifmt = iinfo->fmt | iinfo->codec_daifmt; + int ret; + + if (codec_daifmt) { + ret = snd_soc_dai_set_fmt(codec, codec_daifmt); + if (ret < 0) + return ret; + } + + if (iinfo->sysclk) { + ret = snd_soc_dai_set_sysclk(codec, 0, iinfo->sysclk, 0); + if (ret < 0) + return ret; + } + + if (cpu_daifmt) { + ret = snd_soc_dai_set_fmt(cpu, cpu_daifmt); + if (ret < 0) + return ret; + } + + return 0; +} + +static int asoc_simple_card_probe(struct platform_device *pdev) +{ + struct asoc_simple_card_info *cinfo = pdev->dev.platform_data; + + if (!cinfo) { + dev_err(&pdev->dev, "no info for asoc-simple-card\n"); + return -EINVAL; + } + + if (!cinfo->name || + !cinfo->card || + !cinfo->cpu_dai || + !cinfo->codec || + !cinfo->platform || + !cinfo->codec_dai) { + dev_err(&pdev->dev, "insufficient asoc_simple_card_info settings\n"); + return -EINVAL; + } + + /* + * init snd_soc_dai_link + */ + cinfo->snd_link.name = cinfo->name; + cinfo->snd_link.stream_name = cinfo->name; + cinfo->snd_link.cpu_dai_name = cinfo->cpu_dai; + cinfo->snd_link.platform_name = cinfo->platform; + cinfo->snd_link.codec_name = cinfo->codec; + cinfo->snd_link.codec_dai_name = cinfo->codec_dai; + + /* enable snd_link.init if cinfo has settings */ + if (cinfo->init) + cinfo->snd_link.init = asoc_simple_card_dai_init; + + /* + * init snd_soc_card + */ + cinfo->snd_card.name = cinfo->card; + cinfo->snd_card.owner = THIS_MODULE; + cinfo->snd_card.dai_link = &cinfo->snd_link; + cinfo->snd_card.num_links = 1; + cinfo->snd_card.dev = &pdev->dev; + + return snd_soc_register_card(&cinfo->snd_card); +} + +static int asoc_simple_card_remove(struct platform_device *pdev) +{ + struct asoc_simple_card_info *cinfo = pdev->dev.platform_data; + + return snd_soc_unregister_card(&cinfo->snd_card); +} + +static struct platform_driver asoc_simple_card = { + .driver = { + .name = "asoc-simple-card", + }, + .probe = asoc_simple_card_probe, + .remove = asoc_simple_card_remove, +}; + +module_platform_driver(asoc_simple_card); + +MODULE_LICENSE("GPL"); +MODULE_DESCRIPTION("ASoC Simple Sound Card"); +MODULE_AUTHOR("Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>"); diff --git a/sound/soc/imx/Kconfig b/sound/soc/imx/Kconfig deleted file mode 100644 index d83e5d0b5d5..00000000000 --- a/sound/soc/imx/Kconfig +++ /dev/null @@ -1,79 +0,0 @@ -menuconfig SND_IMX_SOC - tristate "SoC Audio for Freescale i.MX CPUs" - depends on ARCH_MXC - help - Say Y or M if you want to add support for codecs attached to - the i.MX SSI interface. - - -if SND_IMX_SOC - -config SND_SOC_IMX_SSI - tristate - -config SND_SOC_IMX_PCM - tristate - -config SND_MXC_SOC_FIQ - tristate - select FIQ - select SND_SOC_IMX_PCM - -config SND_MXC_SOC_MX2 - select SND_SOC_DMAENGINE_PCM - tristate - select SND_SOC_IMX_PCM - -config SND_SOC_IMX_AUDMUX - tristate - -config SND_MXC_SOC_WM1133_EV1 - tristate "Audio on the i.MX31ADS with WM1133-EV1 fitted" - depends on MACH_MX31ADS_WM1133_EV1 && EXPERIMENTAL - select SND_SOC_WM8350 - select SND_MXC_SOC_FIQ - select SND_SOC_IMX_AUDMUX - select SND_SOC_IMX_SSI - help - Enable support for audio on the i.MX31ADS with the WM1133-EV1 - PMIC board with WM8835x fitted. - -config SND_SOC_MX27VIS_AIC32X4 - tristate "SoC audio support for Visstrim M10 boards" - depends on MACH_IMX27_VISSTRIM_M10 && I2C - select SND_SOC_TLV320AIC32X4 - select SND_MXC_SOC_MX2 - select SND_SOC_IMX_AUDMUX - select SND_SOC_IMX_SSI - help - Say Y if you want to add support for SoC audio on Visstrim SM10 - board with TLV320AIC32X4 codec. - -config SND_SOC_PHYCORE_AC97 - tristate "SoC Audio support for Phytec phyCORE (and phyCARD) boards" - depends on MACH_PCM043 || MACH_PCA100 - select SND_SOC_AC97_BUS - select SND_SOC_WM9712 - select SND_MXC_SOC_FIQ - select SND_SOC_IMX_AUDMUX - select SND_SOC_IMX_SSI - help - Say Y if you want to add support for SoC audio on Phytec phyCORE - and phyCARD boards in AC97 mode - -config SND_SOC_EUKREA_TLV320 - tristate "Eukrea TLV320" - depends on MACH_EUKREA_MBIMX27_BASEBOARD \ - || MACH_EUKREA_MBIMXSD25_BASEBOARD \ - || MACH_EUKREA_MBIMXSD35_BASEBOARD \ - || MACH_EUKREA_MBIMXSD51_BASEBOARD - depends on I2C - select SND_SOC_TLV320AIC23 - select SND_MXC_SOC_FIQ - select SND_SOC_IMX_AUDMUX - select SND_SOC_IMX_SSI - help - Enable I2S based access to the TLV320AIC23B codec attached - to the SSI interface - -endif # SND_IMX_SOC diff --git a/sound/soc/imx/Makefile b/sound/soc/imx/Makefile deleted file mode 100644 index f5db3e92d0d..00000000000 --- a/sound/soc/imx/Makefile +++ /dev/null @@ -1,22 +0,0 @@ -# i.MX Platform Support -snd-soc-imx-ssi-objs := imx-ssi.o -snd-soc-imx-audmux-objs := imx-audmux.o - -obj-$(CONFIG_SND_SOC_IMX_SSI) += snd-soc-imx-ssi.o -obj-$(CONFIG_SND_SOC_IMX_AUDMUX) += snd-soc-imx-audmux.o - -obj-$(CONFIG_SND_SOC_IMX_PCM) += snd-soc-imx-pcm.o -snd-soc-imx-pcm-y := imx-pcm.o -snd-soc-imx-pcm-$(CONFIG_SND_MXC_SOC_FIQ) += imx-pcm-fiq.o -snd-soc-imx-pcm-$(CONFIG_SND_MXC_SOC_MX2) += imx-pcm-dma-mx2.o - -# i.MX Machine Support -snd-soc-eukrea-tlv320-objs := eukrea-tlv320.o -snd-soc-phycore-ac97-objs := phycore-ac97.o -snd-soc-mx27vis-aic32x4-objs := mx27vis-aic32x4.o -snd-soc-wm1133-ev1-objs := wm1133-ev1.o - -obj-$(CONFIG_SND_SOC_EUKREA_TLV320) += snd-soc-eukrea-tlv320.o -obj-$(CONFIG_SND_SOC_PHYCORE_AC97) += snd-soc-phycore-ac97.o -obj-$(CONFIG_SND_SOC_MX27VIS_AIC32X4) += snd-soc-mx27vis-aic32x4.o -obj-$(CONFIG_SND_MXC_SOC_WM1133_EV1) += snd-soc-wm1133-ev1.o diff --git a/sound/soc/jz4740/jz4740-i2s.c b/sound/soc/jz4740/jz4740-i2s.c index a5af7c42e62..41349670ada 100644 --- a/sound/soc/jz4740/jz4740-i2s.c +++ b/sound/soc/jz4740/jz4740-i2s.c @@ -346,7 +346,7 @@ static void jz4740_i2c_init_pcm_config(struct jz4740_i2s *i2s) /* Playback */ dma_config = &i2s->pcm_config_playback.dma_config; - dma_config->src_width = JZ4740_DMA_WIDTH_32BIT, + dma_config->src_width = JZ4740_DMA_WIDTH_32BIT; dma_config->transfer_size = JZ4740_DMA_TRANSFER_SIZE_16BYTE; dma_config->request_type = JZ4740_DMA_TYPE_AIC_TRANSMIT; dma_config->flags = JZ4740_DMA_SRC_AUTOINC; @@ -355,7 +355,7 @@ static void jz4740_i2c_init_pcm_config(struct jz4740_i2s *i2s) /* Capture */ dma_config = &i2s->pcm_config_capture.dma_config; - dma_config->dst_width = JZ4740_DMA_WIDTH_32BIT, + dma_config->dst_width = JZ4740_DMA_WIDTH_32BIT; dma_config->transfer_size = JZ4740_DMA_TRANSFER_SIZE_16BYTE; dma_config->request_type = JZ4740_DMA_TYPE_AIC_RECEIVE; dma_config->flags = JZ4740_DMA_DST_AUTOINC; diff --git a/sound/soc/mxs/mxs-pcm.c b/sound/soc/mxs/mxs-pcm.c index e373fbbc97a..373dec90579 100644 --- a/sound/soc/mxs/mxs-pcm.c +++ b/sound/soc/mxs/mxs-pcm.c @@ -220,28 +220,16 @@ static struct snd_soc_platform_driver mxs_soc_platform = { .pcm_free = mxs_pcm_free, }; -static int __devinit mxs_soc_platform_probe(struct platform_device *pdev) +int __devinit mxs_pcm_platform_register(struct device *dev) { - return snd_soc_register_platform(&pdev->dev, &mxs_soc_platform); + return snd_soc_register_platform(dev, &mxs_soc_platform); } +EXPORT_SYMBOL_GPL(mxs_pcm_platform_register); -static int __devexit mxs_soc_platform_remove(struct platform_device *pdev) +void __devexit mxs_pcm_platform_unregister(struct device *dev) { - snd_soc_unregister_platform(&pdev->dev); - - return 0; + snd_soc_unregister_platform(dev); } - -static struct platform_driver mxs_pcm_driver = { - .driver = { - .name = "mxs-pcm-audio", - .owner = THIS_MODULE, - }, - .probe = mxs_soc_platform_probe, - .remove = __devexit_p(mxs_soc_platform_remove), -}; - -module_platform_driver(mxs_pcm_driver); +EXPORT_SYMBOL_GPL(mxs_pcm_platform_unregister); MODULE_LICENSE("GPL"); -MODULE_ALIAS("platform:mxs-pcm-audio"); diff --git a/sound/soc/mxs/mxs-pcm.h b/sound/soc/mxs/mxs-pcm.h index 5f01a9124b3..35ba2ca4238 100644 --- a/sound/soc/mxs/mxs-pcm.h +++ b/sound/soc/mxs/mxs-pcm.h @@ -24,4 +24,7 @@ struct mxs_pcm_dma_params { int chan_num; }; +int mxs_pcm_platform_register(struct device *dev); +void mxs_pcm_platform_unregister(struct device *dev); + #endif diff --git a/sound/soc/mxs/mxs-saif.c b/sound/soc/mxs/mxs-saif.c index 7fd224bb732..aba71bfa33b 100644 --- a/sound/soc/mxs/mxs-saif.c +++ b/sound/soc/mxs/mxs-saif.c @@ -18,6 +18,8 @@ #include <linux/module.h> #include <linux/init.h> +#include <linux/of.h> +#include <linux/of_device.h> #include <linux/platform_device.h> #include <linux/slab.h> #include <linux/dma-mapping.h> @@ -621,37 +623,57 @@ static irqreturn_t mxs_saif_irq(int irq, void *dev_id) return IRQ_HANDLED; } -static int mxs_saif_probe(struct platform_device *pdev) +static int __devinit mxs_saif_probe(struct platform_device *pdev) { + struct device_node *np = pdev->dev.of_node; struct resource *iores, *dmares; struct mxs_saif *saif; struct mxs_saif_platform_data *pdata; struct pinctrl *pinctrl; int ret = 0; - if (pdev->id >= ARRAY_SIZE(mxs_saif)) + + if (!np && pdev->id >= ARRAY_SIZE(mxs_saif)) return -EINVAL; saif = devm_kzalloc(&pdev->dev, sizeof(*saif), GFP_KERNEL); if (!saif) return -ENOMEM; - mxs_saif[pdev->id] = saif; - saif->id = pdev->id; - - pdata = pdev->dev.platform_data; - if (pdata && !pdata->master_mode) { - saif->master_id = pdata->master_id; - if (saif->master_id < 0 || - saif->master_id >= ARRAY_SIZE(mxs_saif) || - saif->master_id == saif->id) { - dev_err(&pdev->dev, "get wrong master id\n"); - return -EINVAL; + if (np) { + struct device_node *master; + saif->id = of_alias_get_id(np, "saif"); + if (saif->id < 0) + return saif->id; + /* + * If there is no "fsl,saif-master" phandle, it's a saif + * master. Otherwise, it's a slave and its phandle points + * to the master. + */ + master = of_parse_phandle(np, "fsl,saif-master", 0); + if (!master) { + saif->master_id = saif->id; + } else { + saif->master_id = of_alias_get_id(master, "saif"); + if (saif->master_id < 0) + return saif->master_id; } } else { - saif->master_id = saif->id; + saif->id = pdev->id; + pdata = pdev->dev.platform_data; + if (pdata && !pdata->master_mode) + saif->master_id = pdata->master_id; + else + saif->master_id = saif->id; + } + + if (saif->master_id < 0 || saif->master_id >= ARRAY_SIZE(mxs_saif)) { + dev_err(&pdev->dev, "get wrong master id\n"); + return -EINVAL; } + mxs_saif[saif->id] = saif; + pinctrl = devm_pinctrl_get_select_default(&pdev->dev); if (IS_ERR(pinctrl)) { ret = PTR_ERR(pinctrl); @@ -677,12 +699,19 @@ static int mxs_saif_probe(struct platform_device *pdev) dmares = platform_get_resource(pdev, IORESOURCE_DMA, 0); if (!dmares) { - ret = -ENODEV; - dev_err(&pdev->dev, "failed to get dma resource: %d\n", - ret); - goto failed_get_resource; + /* + * TODO: This is a temporary solution and should be changed + * to use generic DMA binding later when the helplers get in. + */ + ret = of_property_read_u32(np, "fsl,saif-dma-channel", + &saif->dma_param.chan_num); + if (ret) { + dev_err(&pdev->dev, "failed to get dma channel\n"); + goto failed_get_resource; + } + } else { + saif->dma_param.chan_num = dmares->start; } - saif->dma_param.chan_num = dmares->start; saif->irq = platform_get_irq(pdev, 0); if (saif->irq < 0) { @@ -716,24 +745,14 @@ static int mxs_saif_probe(struct platform_device *pdev) goto failed_get_resource; } - saif->soc_platform_pdev = platform_device_alloc( - "mxs-pcm-audio", pdev->id); - if (!saif->soc_platform_pdev) { - ret = -ENOMEM; - goto failed_pdev_alloc; - } - - platform_set_drvdata(saif->soc_platform_pdev, saif); - ret = platform_device_add(saif->soc_platform_pdev); + ret = mxs_pcm_platform_register(&pdev->dev); if (ret) { - dev_err(&pdev->dev, "failed to add soc platform device\n"); - goto failed_pdev_add; + dev_err(&pdev->dev, "register PCM failed: %d\n", ret); + goto failed_pdev_alloc; } return 0; -failed_pdev_add: - platform_device_put(saif->soc_platform_pdev); failed_pdev_alloc: snd_soc_unregister_dai(&pdev->dev); failed_get_resource: @@ -746,13 +765,19 @@ static int __devexit mxs_saif_remove(struct platform_device *pdev) { struct mxs_saif *saif = platform_get_drvdata(pdev); - platform_device_unregister(saif->soc_platform_pdev); + mxs_pcm_platform_unregister(&pdev->dev); snd_soc_unregister_dai(&pdev->dev); clk_put(saif->clk); return 0; } +static const struct of_device_id mxs_saif_dt_ids[] = { + { .compatible = "fsl,imx28-saif", }, + { /* sentinel */ } +}; +MODULE_DEVICE_TABLE(of, mxs_saif_dt_ids); + static struct platform_driver mxs_saif_driver = { .probe = mxs_saif_probe, .remove = __devexit_p(mxs_saif_remove), @@ -760,6 +785,7 @@ static struct platform_driver mxs_saif_driver = { .driver = { .name = "mxs-saif", .owner = THIS_MODULE, + .of_match_table = mxs_saif_dt_ids, }, }; diff --git a/sound/soc/mxs/mxs-saif.h b/sound/soc/mxs/mxs-saif.h index 12c91e4eb94..3cb342e5bc9 100644 --- a/sound/soc/mxs/mxs-saif.h +++ b/sound/soc/mxs/mxs-saif.h @@ -123,7 +123,6 @@ struct mxs_saif { unsigned int cur_rate; unsigned int ongoing; - struct platform_device *soc_platform_pdev; u32 fifo_underrun; u32 fifo_overrun; }; diff --git a/sound/soc/mxs/mxs-sgtl5000.c b/sound/soc/mxs/mxs-sgtl5000.c index 60f052b7cf2..3e6e8764b2e 100644 --- a/sound/soc/mxs/mxs-sgtl5000.c +++ b/sound/soc/mxs/mxs-sgtl5000.c @@ -18,6 +18,8 @@ #include <linux/module.h> #include <linux/device.h> +#include <linux/of.h> +#include <linux/of_device.h> #include <sound/core.h> #include <sound/pcm.h> #include <sound/soc.h> @@ -90,7 +92,7 @@ static struct snd_soc_dai_link mxs_sgtl5000_dai[] = { .codec_dai_name = "sgtl5000", .codec_name = "sgtl5000.0-000a", .cpu_dai_name = "mxs-saif.0", - .platform_name = "mxs-pcm-audio.0", + .platform_name = "mxs-saif.0", .ops = &mxs_sgtl5000_hifi_ops, }, { .name = "HiFi Rx", @@ -98,7 +100,7 @@ static struct snd_soc_dai_link mxs_sgtl5000_dai[] = { .codec_dai_name = "sgtl5000", .codec_name = "sgtl5000.0-000a", .cpu_dai_name = "mxs-saif.1", - .platform_name = "mxs-pcm-audio.1", + .platform_name = "mxs-saif.1", .ops = &mxs_sgtl5000_hifi_ops, }, }; @@ -110,11 +112,48 @@ static struct snd_soc_card mxs_sgtl5000 = { .num_links = ARRAY_SIZE(mxs_sgtl5000_dai), }; +static int __devinit mxs_sgtl5000_probe_dt(struct platform_device *pdev) +{ + struct device_node *np = pdev->dev.of_node; + struct device_node *saif_np[2], *codec_np; + int i, ret = 0; + + if (!np) + return 1; /* no device tree */ + + saif_np[0] = of_parse_phandle(np, "saif-controllers", 0); + saif_np[1] = of_parse_phandle(np, "saif-controllers", 1); + codec_np = of_parse_phandle(np, "audio-codec", 0); + if (!saif_np[0] || !saif_np[1] || !codec_np) { + dev_err(&pdev->dev, "phandle missing or invalid\n"); + return -EINVAL; + } + + for (i = 0; i < 2; i++) { + mxs_sgtl5000_dai[i].codec_name = NULL; + mxs_sgtl5000_dai[i].codec_of_node = codec_np; + mxs_sgtl5000_dai[i].cpu_dai_name = NULL; + mxs_sgtl5000_dai[i].cpu_dai_of_node = saif_np[i]; + mxs_sgtl5000_dai[i].platform_name = NULL; + mxs_sgtl5000_dai[i].platform_of_node = saif_np[i]; + } + + of_node_put(codec_np); + of_node_put(saif_np[0]); + of_node_put(saif_np[1]); + + return ret; +} + static int __devinit mxs_sgtl5000_probe(struct platform_device *pdev) { struct snd_soc_card *card = &mxs_sgtl5000; int ret; + ret = mxs_sgtl5000_probe_dt(pdev); + if (ret < 0) + return ret; + /* * Set an init clock(11.28Mhz) for sgtl5000 initialization(i2c r/w). * The Sgtl5000 sysclk is derived from saif0 mclk and it's range @@ -148,10 +187,17 @@ static int __devexit mxs_sgtl5000_remove(struct platform_device *pdev) return 0; } +static const struct of_device_id mxs_sgtl5000_dt_ids[] = { + { .compatible = "fsl,mxs-audio-sgtl5000", }, + { /* sentinel */ } +}; +MODULE_DEVICE_TABLE(of, mxs_sgtl5000_dt_ids); + static struct platform_driver mxs_sgtl5000_audio_driver = { .driver = { .name = "mxs-sgtl5000", .owner = THIS_MODULE, + .of_match_table = mxs_sgtl5000_dt_ids, }, .probe = mxs_sgtl5000_probe, .remove = __devexit_p(mxs_sgtl5000_remove), diff --git a/sound/soc/omap/Kconfig b/sound/soc/omap/Kconfig index deafbfaacdb..9ccfa5e1c11 100644 --- a/sound/soc/omap/Kconfig +++ b/sound/soc/omap/Kconfig @@ -113,6 +113,7 @@ config SND_OMAP_SOC_OMAP4_HDMI tristate "SoC Audio support for Texas Instruments OMAP4 HDMI" depends on SND_OMAP_SOC && OMAP4_DSS_HDMI && OMAP2_DSS && ARCH_OMAP4 select SND_OMAP_SOC_HDMI + select SND_SOC_OMAP_HDMI_CODEC help Say Y if you want to add support for SoC HDMI audio on Texas Instruments OMAP4 chips diff --git a/sound/soc/pxa/pxa-ssp.c b/sound/soc/pxa/pxa-ssp.c index fd04ce13903..1c2aa7fab3f 100644 --- a/sound/soc/pxa/pxa-ssp.c +++ b/sound/soc/pxa/pxa-ssp.c @@ -85,14 +85,12 @@ struct pxa2xx_pcm_dma_data { char name[20]; }; -static struct pxa2xx_pcm_dma_params * -pxa_ssp_get_dma_params(struct ssp_device *ssp, int width4, int out) +static void pxa_ssp_set_dma_params(struct ssp_device *ssp, int width4, + int out, struct pxa2xx_pcm_dma_params *dma_data) { struct pxa2xx_pcm_dma_data *dma; - dma = kzalloc(sizeof(struct pxa2xx_pcm_dma_data), GFP_KERNEL); - if (dma == NULL) - return NULL; + dma = container_of(dma_data, struct pxa2xx_pcm_dma_data, params); snprintf(dma->name, 20, "SSP%d PCM %s %s", ssp->port_id, width4 ? "32-bit" : "16-bit", out ? "out" : "in"); @@ -103,8 +101,6 @@ pxa_ssp_get_dma_params(struct ssp_device *ssp, int width4, int out) (DCMD_INCTRGADDR | DCMD_FLOWSRC)) | (width4 ? DCMD_WIDTH4 : DCMD_WIDTH2) | DCMD_BURST16; dma->params.dev_addr = ssp->phys_base + SSDR; - - return &dma->params; } static int pxa_ssp_startup(struct snd_pcm_substream *substream, @@ -112,6 +108,7 @@ static int pxa_ssp_startup(struct snd_pcm_substream *substream, { struct ssp_priv *priv = snd_soc_dai_get_drvdata(cpu_dai); struct ssp_device *ssp = priv->ssp; + struct pxa2xx_pcm_dma_data *dma; int ret = 0; if (!cpu_dai->active) { @@ -119,8 +116,10 @@ static int pxa_ssp_startup(struct snd_pcm_substream *substream, pxa_ssp_disable(ssp); } - kfree(snd_soc_dai_get_dma_data(cpu_dai, substream)); - snd_soc_dai_set_dma_data(cpu_dai, substream, NULL); + dma = kzalloc(sizeof(struct pxa2xx_pcm_dma_data), GFP_KERNEL); + if (!dma) + return -ENOMEM; + snd_soc_dai_set_dma_data(cpu_dai, substream, &dma->params); return ret; } @@ -573,18 +572,13 @@ static int pxa_ssp_hw_params(struct snd_pcm_substream *substream, dma_data = snd_soc_dai_get_dma_data(cpu_dai, substream); - /* generate correct DMA params */ - kfree(dma_data); - /* Network mode with one active slot (ttsa == 1) can be used * to force 16-bit frame width on the wire (for S16_LE), even * with two channels. Use 16-bit DMA transfers for this case. */ - dma_data = pxa_ssp_get_dma_params(ssp, - ((chn == 2) && (ttsa != 1)) || (width == 32), - substream->stream == SNDRV_PCM_STREAM_PLAYBACK); - - snd_soc_dai_set_dma_data(cpu_dai, substream, dma_data); + pxa_ssp_set_dma_params(ssp, + ((chn == 2) && (ttsa != 1)) || (width == 32), + substream->stream == SNDRV_PCM_STREAM_PLAYBACK, dma_data); /* we can only change the settings if the port is not in use */ if (pxa_ssp_read_reg(ssp, SSCR0) & SSCR0_SSE) diff --git a/sound/soc/pxa/pxa2xx-i2s.c b/sound/soc/pxa/pxa2xx-i2s.c index d08583790d2..3075a426124 100644 --- a/sound/soc/pxa/pxa2xx-i2s.c +++ b/sound/soc/pxa/pxa2xx-i2s.c @@ -166,7 +166,7 @@ static int pxa2xx_i2s_hw_params(struct snd_pcm_substream *substream, struct pxa2xx_pcm_dma_params *dma_data; BUG_ON(IS_ERR(clk_i2s)); - clk_enable(clk_i2s); + clk_prepare_enable(clk_i2s); clk_ena = 1; pxa_i2s_wait(); @@ -259,7 +259,7 @@ static void pxa2xx_i2s_shutdown(struct snd_pcm_substream *substream, SACR0 &= ~SACR0_ENB; pxa_i2s_wait(); if (clk_ena) { - clk_disable(clk_i2s); + clk_disable_unprepare(clk_i2s); clk_ena = 0; } } diff --git a/sound/soc/samsung/littlemill.c b/sound/soc/samsung/littlemill.c index e7416851bf7..c82c646b8a0 100644 --- a/sound/soc/samsung/littlemill.c +++ b/sound/soc/samsung/littlemill.c @@ -23,10 +23,10 @@ static int littlemill_set_bias_level(struct snd_soc_card *card, struct snd_soc_dapm_context *dapm, enum snd_soc_bias_level level) { - struct snd_soc_dai *codec_dai = card->rtd[0].codec_dai; + struct snd_soc_dai *aif1_dai = card->rtd[0].codec_dai; int ret; - if (dapm->dev != codec_dai->dev) + if (dapm->dev != aif1_dai->dev) return 0; switch (level) { @@ -36,7 +36,7 @@ static int littlemill_set_bias_level(struct snd_soc_card *card, * then do so now, otherwise these are noops. */ if (dapm->bias_level == SND_SOC_BIAS_STANDBY) { - ret = snd_soc_dai_set_pll(codec_dai, WM8994_FLL1, + ret = snd_soc_dai_set_pll(aif1_dai, WM8994_FLL1, WM8994_FLL_SRC_MCLK2, 32768, sample_rate * 512); if (ret < 0) { @@ -44,7 +44,7 @@ static int littlemill_set_bias_level(struct snd_soc_card *card, return ret; } - ret = snd_soc_dai_set_sysclk(codec_dai, + ret = snd_soc_dai_set_sysclk(aif1_dai, WM8994_SYSCLK_FLL1, sample_rate * 512, SND_SOC_CLOCK_IN); @@ -66,25 +66,25 @@ static int littlemill_set_bias_level_post(struct snd_soc_card *card, struct snd_soc_dapm_context *dapm, enum snd_soc_bias_level level) { - struct snd_soc_dai *codec_dai = card->rtd[0].codec_dai; + struct snd_soc_dai *aif1_dai = card->rtd[0].codec_dai; int ret; - if (dapm->dev != codec_dai->dev) + if (dapm->dev != aif1_dai->dev) return 0; switch (level) { case SND_SOC_BIAS_STANDBY: - ret = snd_soc_dai_set_sysclk(codec_dai, WM8994_SYSCLK_MCLK2, + ret = snd_soc_dai_set_sysclk(aif1_dai, WM8994_SYSCLK_MCLK2, 32768, SND_SOC_CLOCK_IN); if (ret < 0) { - pr_err("Failed to switch away from FLL: %d\n", ret); + pr_err("Failed to switch away from FLL1: %d\n", ret); return ret; } - ret = snd_soc_dai_set_pll(codec_dai, WM8994_FLL1, + ret = snd_soc_dai_set_pll(aif1_dai, WM8994_FLL1, 0, 0, 0); if (ret < 0) { - pr_err("Failed to stop FLL: %d\n", ret); + pr_err("Failed to stop FLL1: %d\n", ret); return ret; } break; @@ -131,6 +131,14 @@ static struct snd_soc_ops littlemill_ops = { .hw_params = littlemill_hw_params, }; +static const struct snd_soc_pcm_stream baseband_params = { + .formats = SNDRV_PCM_FMTBIT_S32_LE, + .rate_min = 8000, + .rate_max = 8000, + .channels_min = 2, + .channels_max = 2, +}; + static struct snd_soc_dai_link littlemill_dai[] = { { .name = "CPU", @@ -143,13 +151,75 @@ static struct snd_soc_dai_link littlemill_dai[] = { | SND_SOC_DAIFMT_CBM_CFM, .ops = &littlemill_ops, }, + { + .name = "Baseband", + .stream_name = "Baseband", + .cpu_dai_name = "wm8994-aif2", + .codec_dai_name = "wm1250-ev1", + .codec_name = "wm1250-ev1.1-0027", + .dai_fmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF + | SND_SOC_DAIFMT_CBM_CFM, + .ignore_suspend = 1, + .params = &baseband_params, + }, }; +static int bbclk_ev(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *kcontrol, int event) +{ + struct snd_soc_card *card = w->dapm->card; + struct snd_soc_dai *aif2_dai = card->rtd[1].cpu_dai; + int ret; + + switch (event) { + case SND_SOC_DAPM_PRE_PMU: + ret = snd_soc_dai_set_pll(aif2_dai, WM8994_FLL2, + WM8994_FLL_SRC_BCLK, 64 * 8000, + 8000 * 256); + if (ret < 0) { + pr_err("Failed to start FLL: %d\n", ret); + return ret; + } + + ret = snd_soc_dai_set_sysclk(aif2_dai, WM8994_SYSCLK_FLL2, + 8000 * 256, + SND_SOC_CLOCK_IN); + if (ret < 0) { + pr_err("Failed to set SYSCLK: %d\n", ret); + return ret; + } + break; + case SND_SOC_DAPM_POST_PMD: + ret = snd_soc_dai_set_sysclk(aif2_dai, WM8994_SYSCLK_MCLK2, + 32768, SND_SOC_CLOCK_IN); + if (ret < 0) { + pr_err("Failed to switch away from FLL2: %d\n", ret); + return ret; + } + + ret = snd_soc_dai_set_pll(aif2_dai, WM8994_FLL2, + 0, 0, 0); + if (ret < 0) { + pr_err("Failed to stop FLL2: %d\n", ret); + return ret; + } + break; + default: + return -EINVAL; + } + + return 0; +} + static struct snd_soc_dapm_widget widgets[] = { SND_SOC_DAPM_HP("Headphone", NULL), SND_SOC_DAPM_MIC("AMIC", NULL), SND_SOC_DAPM_MIC("DMIC", NULL), + + SND_SOC_DAPM_SUPPLY_S("Baseband Clock", -1, SND_SOC_NOPM, 0, 0, + bbclk_ev, + SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMD), }; static struct snd_soc_dapm_route audio_paths[] = { @@ -162,6 +232,8 @@ static struct snd_soc_dapm_route audio_paths[] = { { "DMIC", NULL, "MICBIAS2" }, /* Default for DMICBIAS jumper */ { "DMIC1DAT", NULL, "DMIC" }, { "DMIC2DAT", NULL, "DMIC" }, + + { "AIF2CLK", NULL, "Baseband Clock" }, }; static struct snd_soc_jack littlemill_headset; @@ -169,10 +241,16 @@ static struct snd_soc_jack littlemill_headset; static int littlemill_late_probe(struct snd_soc_card *card) { struct snd_soc_codec *codec = card->rtd[0].codec; - struct snd_soc_dai *codec_dai = card->rtd[0].codec_dai; + struct snd_soc_dai *aif1_dai = card->rtd[0].codec_dai; + struct snd_soc_dai *aif2_dai = card->rtd[1].cpu_dai; int ret; - ret = snd_soc_dai_set_sysclk(codec_dai, WM8994_SYSCLK_MCLK2, + ret = snd_soc_dai_set_sysclk(aif1_dai, WM8994_SYSCLK_MCLK2, + 32768, SND_SOC_CLOCK_IN); + if (ret < 0) + return ret; + + ret = snd_soc_dai_set_sysclk(aif2_dai, WM8994_SYSCLK_MCLK2, 32768, SND_SOC_CLOCK_IN); if (ret < 0) return ret; diff --git a/sound/soc/samsung/lowland.c b/sound/soc/samsung/lowland.c index 4adff934f77..6abf341c4a2 100644 --- a/sound/soc/samsung/lowland.c +++ b/sound/soc/samsung/lowland.c @@ -21,33 +21,6 @@ #define MCLK1_RATE (44100 * 512) #define CLKOUT_RATE (44100 * 256) -static int lowland_hw_params(struct snd_pcm_substream *substream, - struct snd_pcm_hw_params *params) -{ - struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct snd_soc_dai *cpu_dai = rtd->cpu_dai; - struct snd_soc_dai *codec_dai = rtd->codec_dai; - int ret; - - ret = snd_soc_dai_set_fmt(codec_dai, SND_SOC_DAIFMT_I2S - | SND_SOC_DAIFMT_NB_NF - | SND_SOC_DAIFMT_CBM_CFM); - if (ret < 0) - return ret; - - ret = snd_soc_dai_set_fmt(cpu_dai, SND_SOC_DAIFMT_I2S - | SND_SOC_DAIFMT_NB_NF - | SND_SOC_DAIFMT_CBM_CFM); - if (ret < 0) - return ret; - - return 0; -} - -static struct snd_soc_ops lowland_ops = { - .hw_params = lowland_hw_params, -}; - static struct snd_soc_jack lowland_headset; /* Headset jack detection DAPM pins */ @@ -101,6 +74,25 @@ static int lowland_wm5100_init(struct snd_soc_pcm_runtime *rtd) return 0; } +static int lowland_wm9081_init(struct snd_soc_pcm_runtime *rtd) +{ + struct snd_soc_codec *codec = rtd->codec; + + snd_soc_dapm_nc_pin(&codec->dapm, "LINEOUT"); + + /* At any time the WM9081 is active it will have this clock */ + return snd_soc_codec_set_sysclk(codec, WM9081_SYSCLK_MCLK, 0, + CLKOUT_RATE, 0); +} + +static const struct snd_soc_pcm_stream sub_params = { + .formats = SNDRV_PCM_FMTBIT_S32_LE, + .rate_min = 44100, + .rate_max = 44100, + .channels_min = 2, + .channels_max = 2, +}; + static struct snd_soc_dai_link lowland_dai[] = { { .name = "CPU", @@ -109,7 +101,8 @@ static struct snd_soc_dai_link lowland_dai[] = { .codec_dai_name = "wm5100-aif1", .platform_name = "samsung-audio", .codec_name = "wm5100.1-001a", - .ops = &lowland_ops, + .dai_fmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF | + SND_SOC_DAIFMT_CBM_CFM, .init = lowland_wm5100_init, }, { @@ -118,24 +111,20 @@ static struct snd_soc_dai_link lowland_dai[] = { .cpu_dai_name = "wm5100-aif2", .codec_dai_name = "wm1250-ev1", .codec_name = "wm1250-ev1.1-0027", - .ops = &lowland_ops, + .dai_fmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF | + SND_SOC_DAIFMT_CBM_CFM, .ignore_suspend = 1, }, -}; - -static int lowland_wm9081_init(struct snd_soc_dapm_context *dapm) -{ - snd_soc_dapm_nc_pin(dapm, "LINEOUT"); - - /* At any time the WM9081 is active it will have this clock */ - return snd_soc_codec_set_sysclk(dapm->codec, WM9081_SYSCLK_MCLK, 0, - CLKOUT_RATE, 0); -} - -static struct snd_soc_aux_dev lowland_aux_dev[] = { { - .name = "wm9081", + .name = "Sub Speaker", + .stream_name = "Sub Speaker", + .cpu_dai_name = "wm5100-aif3", + .codec_dai_name = "wm9081-hifi", .codec_name = "wm9081.1-006c", + .dai_fmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF | + SND_SOC_DAIFMT_CBM_CFM, + .ignore_suspend = 1, + .params = &sub_params, .init = lowland_wm9081_init, }, }; @@ -180,8 +169,6 @@ static struct snd_soc_card lowland = { .owner = THIS_MODULE, .dai_link = lowland_dai, .num_links = ARRAY_SIZE(lowland_dai), - .aux_dev = lowland_aux_dev, - .num_aux_devs = ARRAY_SIZE(lowland_aux_dev), .codec_conf = lowland_codec_conf, .num_configs = ARRAY_SIZE(lowland_codec_conf), diff --git a/sound/soc/samsung/speyside.c b/sound/soc/samsung/speyside.c index f9ab7707a3e..a4a9fc7e8c7 100644 --- a/sound/soc/samsung/speyside.c +++ b/sound/soc/samsung/speyside.c @@ -92,33 +92,6 @@ static int speyside_set_bias_level_post(struct snd_soc_card *card, return 0; } -static int speyside_hw_params(struct snd_pcm_substream *substream, - struct snd_pcm_hw_params *params) -{ - struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct snd_soc_dai *cpu_dai = rtd->cpu_dai; - struct snd_soc_dai *codec_dai = rtd->codec_dai; - int ret; - - ret = snd_soc_dai_set_fmt(codec_dai, SND_SOC_DAIFMT_I2S - | SND_SOC_DAIFMT_NB_NF - | SND_SOC_DAIFMT_CBM_CFM); - if (ret < 0) - return ret; - - ret = snd_soc_dai_set_fmt(cpu_dai, SND_SOC_DAIFMT_I2S - | SND_SOC_DAIFMT_NB_NF - | SND_SOC_DAIFMT_CBM_CFM); - if (ret < 0) - return ret; - - return 0; -} - -static struct snd_soc_ops speyside_ops = { - .hw_params = speyside_hw_params, -}; - static struct snd_soc_jack speyside_headset; /* Headset jack detection DAPM pins */ @@ -208,7 +181,8 @@ static struct snd_soc_dai_link speyside_dai[] = { .platform_name = "samsung-audio", .codec_name = "wm8996.1-001a", .init = speyside_wm8996_init, - .ops = &speyside_ops, + .dai_fmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF + | SND_SOC_DAIFMT_CBM_CFM, }, { .name = "Baseband", @@ -216,7 +190,8 @@ static struct snd_soc_dai_link speyside_dai[] = { .cpu_dai_name = "wm8996-aif2", .codec_dai_name = "wm1250-ev1", .codec_name = "wm1250-ev1.1-0027", - .ops = &speyside_ops, + .dai_fmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF + | SND_SOC_DAIFMT_CBM_CFM, .ignore_suspend = 1, }, }; diff --git a/sound/soc/sh/Kconfig b/sound/soc/sh/Kconfig index d8e06a607a2..6bcb1164d59 100644 --- a/sound/soc/sh/Kconfig +++ b/sound/soc/sh/Kconfig @@ -22,6 +22,7 @@ config SND_SOC_SH4_SSI config SND_SOC_SH4_FSI tristate "SH4 FSI support" + select SND_SIMPLE_CARD help This option enables FSI sound support @@ -46,29 +47,6 @@ config SND_SH7760_AC97 This option enables generic sound support for the first AC97 unit of the SH7760. -config SND_FSI_AK4642 - tristate "FSI-AK4642 sound support" - depends on SND_SOC_SH4_FSI && I2C - select SND_SOC_AK4642 - help - This option enables generic sound support for the - FSI - AK4642 unit - -config SND_FSI_DA7210 - tristate "FSI-DA7210 sound support" - depends on SND_SOC_SH4_FSI && I2C - select SND_SOC_DA7210 - help - This option enables generic sound support for the - FSI - DA7210 unit - -config SND_FSI_HDMI - tristate "FSI-HDMI sound support" - depends on SND_SOC_SH4_FSI && FB_SH_MOBILE_HDMI - help - This option enables generic sound support for the - FSI - HDMI unit - config SND_SIU_MIGOR tristate "SIU sound support on Migo-R" depends on SH_MIGOR diff --git a/sound/soc/sh/Makefile b/sound/soc/sh/Makefile index 94476d4c0fd..849b387d17d 100644 --- a/sound/soc/sh/Makefile +++ b/sound/soc/sh/Makefile @@ -14,13 +14,7 @@ obj-$(CONFIG_SND_SOC_SH4_SIU) += snd-soc-siu.o ## boards snd-soc-sh7760-ac97-objs := sh7760-ac97.o -snd-soc-fsi-ak4642-objs := fsi-ak4642.o -snd-soc-fsi-da7210-objs := fsi-da7210.o -snd-soc-fsi-hdmi-objs := fsi-hdmi.o snd-soc-migor-objs := migor.o obj-$(CONFIG_SND_SH7760_AC97) += snd-soc-sh7760-ac97.o -obj-$(CONFIG_SND_FSI_AK4642) += snd-soc-fsi-ak4642.o -obj-$(CONFIG_SND_FSI_DA7210) += snd-soc-fsi-da7210.o -obj-$(CONFIG_SND_FSI_HDMI) += snd-soc-fsi-hdmi.o obj-$(CONFIG_SND_SIU_MIGOR) += snd-soc-migor.o diff --git a/sound/soc/sh/fsi-ak4642.c b/sound/soc/sh/fsi-ak4642.c deleted file mode 100644 index 97f540aabbd..00000000000 --- a/sound/soc/sh/fsi-ak4642.c +++ /dev/null @@ -1,108 +0,0 @@ -/* - * FSI-AK464x sound support for ms7724se - * - * Copyright (C) 2009 Renesas Solutions Corp. - * Kuninori Morimoto <morimoto.kuninori@renesas.com> - * - * This file is subject to the terms and conditions of the GNU General Public - * License. See the file "COPYING" in the main directory of this archive - * for more details. - */ - -#include <linux/platform_device.h> -#include <linux/module.h> -#include <sound/sh_fsi.h> - -struct fsi_ak4642_data { - const char *name; - const char *card; - const char *cpu_dai; - const char *codec; - const char *platform; - int id; -}; - -static int fsi_ak4642_dai_init(struct snd_soc_pcm_runtime *rtd) -{ - struct snd_soc_dai *codec = rtd->codec_dai; - struct snd_soc_dai *cpu = rtd->cpu_dai; - int ret; - - ret = snd_soc_dai_set_fmt(codec, SND_SOC_DAIFMT_LEFT_J | - SND_SOC_DAIFMT_CBM_CFM); - if (ret < 0) - return ret; - - ret = snd_soc_dai_set_sysclk(codec, 0, 11289600, 0); - if (ret < 0) - return ret; - - ret = snd_soc_dai_set_fmt(cpu, SND_SOC_DAIFMT_LEFT_J | - SND_SOC_DAIFMT_CBS_CFS); - - return ret; -} - -static struct snd_soc_dai_link fsi_dai_link = { - .codec_dai_name = "ak4642-hifi", - .init = fsi_ak4642_dai_init, -}; - -static struct snd_soc_card fsi_soc_card = { - .owner = THIS_MODULE, - .dai_link = &fsi_dai_link, - .num_links = 1, -}; - -static struct platform_device *fsi_snd_device; - -static int fsi_ak4642_probe(struct platform_device *pdev) -{ - int ret = -ENOMEM; - struct fsi_ak4642_info *pinfo = pdev->dev.platform_data; - - if (!pinfo) { - dev_err(&pdev->dev, "no info for fsi ak4642\n"); - goto out; - } - - fsi_snd_device = platform_device_alloc("soc-audio", pinfo->id); - if (!fsi_snd_device) - goto out; - - fsi_dai_link.name = pinfo->name; - fsi_dai_link.stream_name = pinfo->name; - fsi_dai_link.cpu_dai_name = pinfo->cpu_dai; - fsi_dai_link.platform_name = pinfo->platform; - fsi_dai_link.codec_name = pinfo->codec; - fsi_soc_card.name = pinfo->card; - - platform_set_drvdata(fsi_snd_device, &fsi_soc_card); - ret = platform_device_add(fsi_snd_device); - - if (ret) - platform_device_put(fsi_snd_device); - -out: - return ret; -} - -static int fsi_ak4642_remove(struct platform_device *pdev) -{ - platform_device_unregister(fsi_snd_device); - return 0; -} - -static struct platform_driver fsi_ak4642 = { - .driver = { - .name = "fsi-ak4642-audio", - }, - .probe = fsi_ak4642_probe, - .remove = fsi_ak4642_remove, -}; - -module_platform_driver(fsi_ak4642); - -MODULE_LICENSE("GPL"); -MODULE_DESCRIPTION("Generic SH4 FSI-AK4642 sound card"); -MODULE_AUTHOR("Kuninori Morimoto <morimoto.kuninori@renesas.com>"); diff --git a/sound/soc/sh/fsi-da7210.c b/sound/soc/sh/fsi-da7210.c deleted file mode 100644 index 1dd3354c741..00000000000 --- a/sound/soc/sh/fsi-da7210.c +++ /dev/null @@ -1,81 +0,0 @@ -/* - * fsi-da7210.c - * - * Copyright (C) 2009 Renesas Solutions Corp. - * Kuninori Morimoto <morimoto.kuninori@renesas.com> - * - * This program is free software; you can redistribute it and/or modify it - * under the terms of the GNU General Public License as published by the - * Free Software Foundation; either version 2 of the License, or (at your - * option) any later version. - */ - -#include <linux/platform_device.h> -#include <linux/module.h> -#include <sound/sh_fsi.h> - -static int fsi_da7210_init(struct snd_soc_pcm_runtime *rtd) -{ - struct snd_soc_dai *codec = rtd->codec_dai; - struct snd_soc_dai *cpu = rtd->cpu_dai; - int ret; - - ret = snd_soc_dai_set_fmt(codec, - SND_SOC_DAIFMT_I2S | - SND_SOC_DAIFMT_CBM_CFM); - if (ret < 0) - return ret; - - ret = snd_soc_dai_set_fmt(cpu, SND_SOC_DAIFMT_I2S | - SND_SOC_DAIFMT_CBS_CFS); - - return ret; -} - -static struct snd_soc_dai_link fsi_da7210_dai = { - .name = "DA7210", - .stream_name = "DA7210", - .cpu_dai_name = "fsib-dai", /* FSI B */ - .codec_dai_name = "da7210-hifi", - .platform_name = "sh_fsi.0", - .codec_name = "da7210-codec.0-001a", - .init = fsi_da7210_init, -}; - -static struct snd_soc_card fsi_soc_card = { - .name = "FSI-DA7210", - .owner = THIS_MODULE, - .dai_link = &fsi_da7210_dai, - .num_links = 1, -}; - -static struct platform_device *fsi_da7210_snd_device; - -static int __init fsi_da7210_sound_init(void) -{ - int ret; - - fsi_da7210_snd_device = platform_device_alloc("soc-audio", FSI_PORT_B); - if (!fsi_da7210_snd_device) - return -ENOMEM; - - platform_set_drvdata(fsi_da7210_snd_device, &fsi_soc_card); - ret = platform_device_add(fsi_da7210_snd_device); - if (ret) - platform_device_put(fsi_da7210_snd_device); - - return ret; -} - -static void __exit fsi_da7210_sound_exit(void) -{ - platform_device_unregister(fsi_da7210_snd_device); -} - -module_init(fsi_da7210_sound_init); -module_exit(fsi_da7210_sound_exit); - -/* Module information */ -MODULE_DESCRIPTION("ALSA SoC FSI DA2710"); -MODULE_AUTHOR("Kuninori Morimoto <morimoto.kuninori@renesas.com>"); -MODULE_LICENSE("GPL"); diff --git a/sound/soc/sh/fsi-hdmi.c b/sound/soc/sh/fsi-hdmi.c deleted file mode 100644 index 6e41908323e..00000000000 --- a/sound/soc/sh/fsi-hdmi.c +++ /dev/null @@ -1,118 +0,0 @@ -/* - * FSI - HDMI sound support - * - * Copyright (C) 2010 Renesas Solutions Corp. - * Kuninori Morimoto <kuninori.morimoto.gx@renesas.com> - * - * This file is subject to the terms and conditions of the GNU General Public - * License. See the file "COPYING" in the main directory of this archive - * for more details. - */ - -#include <linux/platform_device.h> -#include <linux/module.h> -#include <sound/sh_fsi.h> - -struct fsi_hdmi_data { - const char *cpu_dai; - const char *card; - int id; -}; - -static int fsi_hdmi_dai_init(struct snd_soc_pcm_runtime *rtd) -{ - struct snd_soc_dai *cpu = rtd->cpu_dai; - int ret; - - ret = snd_soc_dai_set_fmt(cpu, SND_SOC_DAIFMT_CBM_CFM); - - return ret; -} - -static struct snd_soc_dai_link fsi_dai_link = { - .name = "HDMI", - .stream_name = "HDMI", - .codec_dai_name = "sh_mobile_hdmi-hifi", - .platform_name = "sh_fsi2", - .codec_name = "sh-mobile-hdmi", - .init = fsi_hdmi_dai_init, -}; - -static struct snd_soc_card fsi_soc_card = { - .owner = THIS_MODULE, - .dai_link = &fsi_dai_link, - .num_links = 1, -}; - -static struct platform_device *fsi_snd_device; - -static int fsi_hdmi_probe(struct platform_device *pdev) -{ - int ret = -ENOMEM; - const struct platform_device_id *id_entry; - struct fsi_hdmi_data *pdata; - - id_entry = pdev->id_entry; - if (!id_entry) { - dev_err(&pdev->dev, "unknown fsi hdmi\n"); - return -ENODEV; - } - - pdata = (struct fsi_hdmi_data *)id_entry->driver_data; - - fsi_snd_device = platform_device_alloc("soc-audio", pdata->id); - if (!fsi_snd_device) - goto out; - - fsi_dai_link.cpu_dai_name = pdata->cpu_dai; - fsi_soc_card.name = pdata->card; - - platform_set_drvdata(fsi_snd_device, &fsi_soc_card); - ret = platform_device_add(fsi_snd_device); - - if (ret) - platform_device_put(fsi_snd_device); - -out: - return ret; -} - -static int fsi_hdmi_remove(struct platform_device *pdev) -{ - platform_device_unregister(fsi_snd_device); - return 0; -} - -static struct fsi_hdmi_data fsi2_a_hdmi = { - .cpu_dai = "fsia-dai", - .card = "FSI2A-HDMI", - .id = FSI_PORT_A, -}; - -static struct fsi_hdmi_data fsi2_b_hdmi = { - .cpu_dai = "fsib-dai", - .card = "FSI2B-HDMI", - .id = FSI_PORT_B, -}; - -static struct platform_device_id fsi_id_table[] = { - /* FSI 2 */ - { "sh_fsi2_a_hdmi", (kernel_ulong_t)&fsi2_a_hdmi }, - { "sh_fsi2_b_hdmi", (kernel_ulong_t)&fsi2_b_hdmi }, - {}, -}; - -static struct platform_driver fsi_hdmi = { - .driver = { - .name = "fsi-hdmi-audio", - }, - .probe = fsi_hdmi_probe, - .remove = fsi_hdmi_remove, - .id_table = fsi_id_table, -}; - -module_platform_driver(fsi_hdmi); - -MODULE_LICENSE("GPL"); -MODULE_DESCRIPTION("Generic SH4 FSI-HDMI sound card"); -MODULE_AUTHOR("Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>"); diff --git a/sound/soc/sh/fsi.c b/sound/soc/sh/fsi.c index 74ed2dffbff..7cee22515d9 100644 --- a/sound/soc/sh/fsi.c +++ b/sound/soc/sh/fsi.c @@ -132,6 +132,25 @@ typedef int (*set_rate_func)(struct device *dev, int rate, int enable); /* + * bus options + * + * 0x000000BA + * + * A : sample widtht 16bit setting + * B : sample widtht 24bit setting + */ + +#define SHIFT_16DATA 0 +#define SHIFT_24DATA 4 + +#define PACKAGE_24BITBUS_BACK 0 +#define PACKAGE_24BITBUS_FRONT 1 +#define PACKAGE_16BITBUS_STREAM 2 + +#define BUSOP_SET(s, a) ((a) << SHIFT_ ## s ## DATA) +#define BUSOP_GET(s, a) (((a) >> SHIFT_ ## s ## DATA) & 0xF) + +/* * FSI driver use below type name for variable * * xxx_num : number of data @@ -189,6 +208,11 @@ struct fsi_stream { int oerr_num; /* + * bus options + */ + u32 bus_option; + + /* * thse are initialized by fsi_handler_init() */ struct fsi_stream_handler *handler; @@ -211,8 +235,7 @@ struct fsi_priv { struct fsi_stream playback; struct fsi_stream capture; - u32 do_fmt; - u32 di_fmt; + u32 fmt; int chan_num:16; int clk_master:1; @@ -321,6 +344,10 @@ static void _fsi_master_mask_set(struct fsi_master *master, /* * basic function */ +static int fsi_version(struct fsi_master *master) +{ + return master->core->ver; +} static struct fsi_master *fsi_get_master(struct fsi_priv *fsi) { @@ -495,6 +522,7 @@ static void fsi_stream_init(struct fsi_priv *fsi, io->period_samples = fsi_frame2sample(fsi, runtime->period_size); io->period_pos = 0; io->sample_width = samples_to_bytes(runtime, 1); + io->bus_option = 0; io->oerr_num = -1; /* ignore 1st err */ io->uerr_num = -1; /* ignore 1st err */ fsi_stream_handler_call(io, init, fsi, io); @@ -522,6 +550,7 @@ static void fsi_stream_quit(struct fsi_priv *fsi, struct fsi_stream *io) io->period_samples = 0; io->period_pos = 0; io->sample_width = 0; + io->bus_option = 0; io->oerr_num = 0; io->uerr_num = 0; spin_unlock_irqrestore(&master->lock, flags); @@ -581,6 +610,53 @@ static int fsi_stream_remove(struct fsi_priv *fsi) } /* + * format/bus/dma setting + */ +static void fsi_format_bus_setup(struct fsi_priv *fsi, struct fsi_stream *io, + u32 bus, struct device *dev) +{ + struct fsi_master *master = fsi_get_master(fsi); + int is_play = fsi_stream_is_play(fsi, io); + u32 fmt = fsi->fmt; + + if (fsi_version(master) >= 2) { + u32 dma = 0; + + /* + * FSI2 needs DMA/Bus setting + */ + switch (bus) { + case PACKAGE_24BITBUS_FRONT: + fmt |= CR_BWS_24; + dma |= VDMD_FRONT; + dev_dbg(dev, "24bit bus / package in front\n"); + break; + case PACKAGE_16BITBUS_STREAM: + fmt |= CR_BWS_16; + dma |= VDMD_STREAM; + dev_dbg(dev, "16bit bus / stream mode\n"); + break; + case PACKAGE_24BITBUS_BACK: + default: + fmt |= CR_BWS_24; + dma |= VDMD_BACK; + dev_dbg(dev, "24bit bus / package in back\n"); + break; + } + + if (is_play) + fsi_reg_write(fsi, OUT_DMAC, dma); + else + fsi_reg_write(fsi, IN_DMAC, dma); + } + + if (is_play) + fsi_reg_write(fsi, DO_FMT, fmt); + else + fsi_reg_write(fsi, DI_FMT, fmt); +} + +/* * irq function */ @@ -629,11 +705,6 @@ static void fsi_spdif_clk_ctrl(struct fsi_priv *fsi, int enable) struct fsi_master *master = fsi_get_master(fsi); u32 mask, val; - if (master->core->ver < 2) { - pr_err("fsi: register access err (%s)\n", __func__); - return; - } - mask = BP | SE; val = enable ? mask : 0; @@ -648,9 +719,7 @@ static void fsi_spdif_clk_ctrl(struct fsi_priv *fsi, int enable) static int fsi_set_master_clk(struct device *dev, struct fsi_priv *fsi, long rate, int enable) { - struct fsi_master *master = fsi_get_master(fsi); set_rate_func set_rate = fsi_get_info_set_rate(fsi); - int fsi_ver = master->core->ver; int ret; if (!set_rate) @@ -682,10 +751,7 @@ static int fsi_set_master_clk(struct device *dev, struct fsi_priv *fsi, data |= (0x3 << 12); break; case SH_FSI_ACKMD_32: - if (fsi_ver < 2) - dev_err(dev, "unsupported ACKMD\n"); - else - data |= (0x4 << 12); + data |= (0x4 << 12); break; } @@ -708,10 +774,7 @@ static int fsi_set_master_clk(struct device *dev, struct fsi_priv *fsi, data |= (0x4 << 8); break; case SH_FSI_BPFMD_16: - if (fsi_ver < 2) - dev_err(dev, "unsupported ACKMD\n"); - else - data |= (0x7 << 8); + data |= (0x7 << 8); break; } @@ -728,11 +791,26 @@ static int fsi_set_master_clk(struct device *dev, struct fsi_priv *fsi, */ static void fsi_pio_push16(struct fsi_priv *fsi, u8 *_buf, int samples) { - u16 *buf = (u16 *)_buf; + u32 enable_stream = fsi_get_info_flags(fsi) & SH_FSI_ENABLE_STREAM_MODE; int i; - for (i = 0; i < samples; i++) - fsi_reg_write(fsi, DODT, ((u32)*(buf + i) << 8)); + if (enable_stream) { + /* + * stream mode + * see + * fsi_pio_push_init() + */ + u32 *buf = (u32 *)_buf; + + for (i = 0; i < samples / 2; i++) + fsi_reg_write(fsi, DODT, buf[i]); + } else { + /* normal mode */ + u16 *buf = (u16 *)_buf; + + for (i = 0; i < samples; i++) + fsi_reg_write(fsi, DODT, ((u32)*(buf + i) << 8)); + } } static void fsi_pio_pop16(struct fsi_priv *fsi, u8 *_buf, int samples) @@ -872,12 +950,44 @@ static void fsi_pio_start_stop(struct fsi_priv *fsi, struct fsi_stream *io, fsi_master_mask_set(master, CLK_RST, clk, (enable) ? clk : 0); } +static int fsi_pio_push_init(struct fsi_priv *fsi, struct fsi_stream *io) +{ + u32 enable_stream = fsi_get_info_flags(fsi) & SH_FSI_ENABLE_STREAM_MODE; + + /* + * we can use 16bit stream mode + * when "playback" and "16bit data" + * and platform allows "stream mode" + * see + * fsi_pio_push16() + */ + if (enable_stream) + io->bus_option = BUSOP_SET(24, PACKAGE_24BITBUS_BACK) | + BUSOP_SET(16, PACKAGE_16BITBUS_STREAM); + else + io->bus_option = BUSOP_SET(24, PACKAGE_24BITBUS_BACK) | + BUSOP_SET(16, PACKAGE_24BITBUS_BACK); + return 0; +} + +static int fsi_pio_pop_init(struct fsi_priv *fsi, struct fsi_stream *io) +{ + /* + * always 24bit bus, package back when "capture" + */ + io->bus_option = BUSOP_SET(24, PACKAGE_24BITBUS_BACK) | + BUSOP_SET(16, PACKAGE_24BITBUS_BACK); + return 0; +} + static struct fsi_stream_handler fsi_pio_push_handler = { + .init = fsi_pio_push_init, .transfer = fsi_pio_push, .start_stop = fsi_pio_start_stop, }; static struct fsi_stream_handler fsi_pio_pop_handler = { + .init = fsi_pio_pop_init, .transfer = fsi_pio_pop, .start_stop = fsi_pio_start_stop, }; @@ -919,6 +1029,13 @@ static int fsi_dma_init(struct fsi_priv *fsi, struct fsi_stream *io) enum dma_data_direction dir = fsi_stream_is_play(fsi, io) ? DMA_TO_DEVICE : DMA_FROM_DEVICE; + /* + * 24bit data : 24bit bus / package in back + * 16bit data : 16bit bus / stream mode + */ + io->bus_option = BUSOP_SET(24, PACKAGE_24BITBUS_BACK) | + BUSOP_SET(16, PACKAGE_16BITBUS_STREAM); + io->dma = dma_map_single(dai->dev, runtime->dma_area, snd_pcm_lib_buffer_bytes(io->substream), dir); return 0; @@ -1055,25 +1172,9 @@ static int fsi_dma_transfer(struct fsi_priv *fsi, struct fsi_stream *io) static void fsi_dma_push_start_stop(struct fsi_priv *fsi, struct fsi_stream *io, int start) { - u32 bws; - u32 dma; + u32 enable = start ? DMA_ON : 0; - switch (io->sample_width * start) { - case 2: - bws = CR_BWS_16; - dma = VDMD_STREAM | DMA_ON; - break; - case 4: - bws = CR_BWS_24; - dma = VDMD_BACK | DMA_ON; - break; - default: - bws = 0; - dma = 0; - } - - fsi_reg_mask_set(fsi, DO_FMT, CR_BWS_MASK, bws); - fsi_reg_write(fsi, OUT_DMAC, dma); + fsi_reg_mask_set(fsi, OUT_DMAC, DMA_ON, enable); } static int fsi_dma_probe(struct fsi_priv *fsi, struct fsi_stream *io) @@ -1176,8 +1277,6 @@ static int fsi_hw_startup(struct fsi_priv *fsi, struct fsi_stream *io, struct device *dev) { - struct fsi_master *master = fsi_get_master(fsi); - int fsi_ver = master->core->ver; u32 flags = fsi_get_info_flags(fsi); u32 data = 0; @@ -1200,10 +1299,6 @@ static int fsi_hw_startup(struct fsi_priv *fsi, fsi_reg_write(fsi, CKG2, data); - /* set format */ - fsi_reg_write(fsi, DO_FMT, fsi->do_fmt); - fsi_reg_write(fsi, DI_FMT, fsi->di_fmt); - /* spdif ? */ if (fsi_is_spdif(fsi)) { fsi_spdif_clk_ctrl(fsi, 1); @@ -1211,15 +1306,18 @@ static int fsi_hw_startup(struct fsi_priv *fsi, } /* - * FIXME - * - * FSI driver assumed that data package is in-back. - * FSI2 chip can select it. + * get bus settings */ - if (fsi_ver >= 2) { - fsi_reg_write(fsi, OUT_DMAC, (1 << 4)); - fsi_reg_write(fsi, IN_DMAC, (1 << 4)); + data = 0; + switch (io->sample_width) { + case 2: + data = BUSOP_GET(16, io->bus_option); + break; + case 4: + data = BUSOP_GET(24, io->bus_option); + break; } + fsi_format_bus_setup(fsi, io, data, dev); /* irq clear */ fsi_irq_disable(fsi, io); @@ -1243,7 +1341,9 @@ static int fsi_dai_startup(struct snd_pcm_substream *substream, { struct fsi_priv *fsi = fsi_get_priv(substream); - return fsi_hw_startup(fsi, fsi_stream_get(fsi, substream), dai->dev); + fsi->rate = 0; + + return 0; } static void fsi_dai_shutdown(struct snd_pcm_substream *substream, @@ -1251,7 +1351,6 @@ static void fsi_dai_shutdown(struct snd_pcm_substream *substream, { struct fsi_priv *fsi = fsi_get_priv(substream); - fsi_hw_shutdown(fsi, dai->dev); fsi->rate = 0; } @@ -1265,11 +1364,13 @@ static int fsi_dai_trigger(struct snd_pcm_substream *substream, int cmd, switch (cmd) { case SNDRV_PCM_TRIGGER_START: fsi_stream_init(fsi, io, substream); + fsi_hw_startup(fsi, io, dai->dev); ret = fsi_stream_transfer(io); if (0 == ret) fsi_stream_start(fsi, io); break; case SNDRV_PCM_TRIGGER_STOP: + fsi_hw_shutdown(fsi, dai->dev); fsi_stream_stop(fsi, io); fsi_stream_quit(fsi, io); break; @@ -1280,42 +1381,33 @@ static int fsi_dai_trigger(struct snd_pcm_substream *substream, int cmd, static int fsi_set_fmt_dai(struct fsi_priv *fsi, unsigned int fmt) { - u32 data = 0; - switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) { case SND_SOC_DAIFMT_I2S: - data = CR_I2S; + fsi->fmt = CR_I2S; fsi->chan_num = 2; break; case SND_SOC_DAIFMT_LEFT_J: - data = CR_PCM; + fsi->fmt = CR_PCM; fsi->chan_num = 2; break; default: return -EINVAL; } - fsi->do_fmt = data; - fsi->di_fmt = data; - return 0; } static int fsi_set_fmt_spdif(struct fsi_priv *fsi) { struct fsi_master *master = fsi_get_master(fsi); - u32 data = 0; - if (master->core->ver < 2) + if (fsi_version(master) < 2) return -EINVAL; - data = CR_BWS_16 | CR_DTMD_SPDIF_PCM | CR_PCM; + fsi->fmt = CR_DTMD_SPDIF_PCM | CR_PCM; fsi->chan_num = 2; fsi->spdif = 1; - fsi->do_fmt = data; - fsi->di_fmt = data; - return 0; } diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index c88d9741b9e..b37ee8077ed 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -39,6 +39,7 @@ #include <sound/pcm.h> #include <sound/pcm_params.h> #include <sound/soc.h> +#include <sound/soc-dpcm.h> #include <sound/initval.h> #define CREATE_TRACE_POINTS @@ -54,7 +55,6 @@ EXPORT_SYMBOL_GPL(snd_soc_debugfs_root); #endif static DEFINE_MUTEX(client_mutex); -static LIST_HEAD(card_list); static LIST_HEAD(dai_list); static LIST_HEAD(platform_list); static LIST_HEAD(codec_list); @@ -465,6 +465,35 @@ static inline void soc_cleanup_card_debugfs(struct snd_soc_card *card) } #endif +struct snd_pcm_substream *snd_soc_get_dai_substream(struct snd_soc_card *card, + const char *dai_link, int stream) +{ + int i; + + for (i = 0; i < card->num_links; i++) { + if (card->rtd[i].dai_link->no_pcm && + !strcmp(card->rtd[i].dai_link->name, dai_link)) + return card->rtd[i].pcm->streams[stream].substream; + } + dev_dbg(card->dev, "failed to find dai link %s\n", dai_link); + return NULL; +} +EXPORT_SYMBOL_GPL(snd_soc_get_dai_substream); + +struct snd_soc_pcm_runtime *snd_soc_get_pcm_runtime(struct snd_soc_card *card, + const char *dai_link) +{ + int i; + + for (i = 0; i < card->num_links; i++) { + if (!strcmp(card->rtd[i].dai_link->name, dai_link)) + return &card->rtd[i]; + } + dev_dbg(card->dev, "failed to find rtd %s\n", dai_link); + return NULL; +} +EXPORT_SYMBOL_GPL(snd_soc_get_pcm_runtime); + #ifdef CONFIG_SND_SOC_AC97_BUS /* unregister ac97 codec */ static int soc_ac97_dev_unregister(struct snd_soc_codec *codec) @@ -567,19 +596,16 @@ int snd_soc_suspend(struct device *dev) } for (i = 0; i < card->num_rtd; i++) { - struct snd_soc_dai *codec_dai = card->rtd[i].codec_dai; if (card->rtd[i].dai_link->ignore_suspend) continue; snd_soc_dapm_stream_event(&card->rtd[i], SNDRV_PCM_STREAM_PLAYBACK, - codec_dai, SND_SOC_DAPM_STREAM_SUSPEND); snd_soc_dapm_stream_event(&card->rtd[i], SNDRV_PCM_STREAM_CAPTURE, - codec_dai, SND_SOC_DAPM_STREAM_SUSPEND); } @@ -683,17 +709,16 @@ static void soc_resume_deferred(struct work_struct *work) } for (i = 0; i < card->num_rtd; i++) { - struct snd_soc_dai *codec_dai = card->rtd[i].codec_dai; if (card->rtd[i].dai_link->ignore_suspend) continue; snd_soc_dapm_stream_event(&card->rtd[i], - SNDRV_PCM_STREAM_PLAYBACK, codec_dai, + SNDRV_PCM_STREAM_PLAYBACK, SND_SOC_DAPM_STREAM_RESUME); snd_soc_dapm_stream_event(&card->rtd[i], - SNDRV_PCM_STREAM_CAPTURE, codec_dai, + SNDRV_PCM_STREAM_CAPTURE, SND_SOC_DAPM_STREAM_RESUME); } @@ -783,15 +808,9 @@ static int soc_bind_dai_link(struct snd_soc_card *card, int num) struct snd_soc_dai *codec_dai, *cpu_dai; const char *platform_name; - if (rtd->complete) - return 1; dev_dbg(card->dev, "binding %s at idx %d\n", dai_link->name, num); - /* do we already have the CPU DAI for this link ? */ - if (rtd->cpu_dai) { - goto find_codec; - } - /* no, then find CPU DAI from registered DAIs*/ + /* Find CPU DAI from registered DAIs*/ list_for_each_entry(cpu_dai, &dai_list, list) { if (dai_link->cpu_dai_of_node) { if (cpu_dai->dev->of_node != dai_link->cpu_dai_of_node) @@ -802,18 +821,15 @@ static int soc_bind_dai_link(struct snd_soc_card *card, int num) } rtd->cpu_dai = cpu_dai; - goto find_codec; } - dev_dbg(card->dev, "CPU DAI %s not registered\n", - dai_link->cpu_dai_name); -find_codec: - /* do we already have the CODEC for this link ? */ - if (rtd->codec) { - goto find_platform; + if (!rtd->cpu_dai) { + dev_dbg(card->dev, "CPU DAI %s not registered\n", + dai_link->cpu_dai_name); + return -EPROBE_DEFER; } - /* no, then find CODEC from registered CODECs*/ + /* Find CODEC from registered CODECs */ list_for_each_entry(codec, &codec_list, list) { if (dai_link->codec_of_node) { if (codec->dev->of_node != dai_link->codec_of_node) @@ -835,28 +851,28 @@ find_codec: dai_link->codec_dai_name)) { rtd->codec_dai = codec_dai; - goto find_platform; } } - dev_dbg(card->dev, "CODEC DAI %s not registered\n", - dai_link->codec_dai_name); - goto find_platform; + if (!rtd->codec_dai) { + dev_dbg(card->dev, "CODEC DAI %s not registered\n", + dai_link->codec_dai_name); + return -EPROBE_DEFER; + } } - dev_dbg(card->dev, "CODEC %s not registered\n", - dai_link->codec_name); -find_platform: - /* do we need a platform? */ - if (rtd->platform) - goto out; + if (!rtd->codec) { + dev_dbg(card->dev, "CODEC %s not registered\n", + dai_link->codec_name); + return -EPROBE_DEFER; + } /* if there's no platform we match on the empty platform */ platform_name = dai_link->platform_name; if (!platform_name && !dai_link->platform_of_node) platform_name = "snd-soc-dummy"; - /* no, then find one from the set of registered platforms */ + /* find one from the set of registered platforms */ list_for_each_entry(platform, &platform_list, list) { if (dai_link->platform_of_node) { if (platform->dev->of_node != @@ -868,20 +884,16 @@ find_platform: } rtd->platform = platform; - goto out; } - - dev_dbg(card->dev, "platform %s not registered\n", + if (!rtd->platform) { + dev_dbg(card->dev, "platform %s not registered\n", dai_link->platform_name); - return 0; - -out: - /* mark rtd as complete if we found all 4 of our client devices */ - if (rtd->codec && rtd->codec_dai && rtd->platform && rtd->cpu_dai) { - rtd->complete = 1; - card->num_rtd++; + return -EPROBE_DEFER; } - return 1; + + card->num_rtd++; + + return 0; } static void soc_remove_codec(struct snd_soc_codec *codec) @@ -1068,6 +1080,7 @@ static int soc_probe_platform(struct snd_soc_card *card, { int ret = 0; const struct snd_soc_platform_driver *driver = platform->driver; + struct snd_soc_dai *dai; platform->card = card; platform->dapm.card = card; @@ -1081,6 +1094,14 @@ static int soc_probe_platform(struct snd_soc_card *card, snd_soc_dapm_new_controls(&platform->dapm, driver->dapm_widgets, driver->num_dapm_widgets); + /* Create DAPM widgets for each DAI stream */ + list_for_each_entry(dai, &dai_list, list) { + if (dai->dev != platform->dev) + continue; + + snd_soc_dapm_new_dai_widgets(&platform->dapm, dai); + } + platform->dapm.idle_bias_off = 1; if (driver->probe) { @@ -1170,6 +1191,10 @@ static int soc_post_component_init(struct snd_soc_card *card, rtd->dev->init_name = name; dev_set_drvdata(rtd->dev, rtd); mutex_init(&rtd->pcm_mutex); + INIT_LIST_HEAD(&rtd->dpcm[SNDRV_PCM_STREAM_PLAYBACK].be_clients); + INIT_LIST_HEAD(&rtd->dpcm[SNDRV_PCM_STREAM_CAPTURE].be_clients); + INIT_LIST_HEAD(&rtd->dpcm[SNDRV_PCM_STREAM_PLAYBACK].fe_clients); + INIT_LIST_HEAD(&rtd->dpcm[SNDRV_PCM_STREAM_CAPTURE].fe_clients); ret = device_add(rtd->dev); if (ret < 0) { dev_err(card->dev, @@ -1191,6 +1216,17 @@ static int soc_post_component_init(struct snd_soc_card *card, dev_err(codec->dev, "asoc: failed to add codec sysfs files: %d\n", ret); +#ifdef CONFIG_DEBUG_FS + /* add DPCM sysfs entries */ + if (!dailess && !dai_link->dynamic) + goto out; + + ret = soc_dpcm_debugfs_add(rtd); + if (ret < 0) + dev_err(rtd->dev, "asoc: failed to add dpcm sysfs entries: %d\n", ret); + +out: +#endif return 0; } @@ -1200,14 +1236,15 @@ static int soc_probe_dai_link(struct snd_soc_card *card, int num, int order) struct snd_soc_pcm_runtime *rtd = &card->rtd[num]; struct snd_soc_codec *codec = rtd->codec; struct snd_soc_platform *platform = rtd->platform; - struct snd_soc_dai *codec_dai = rtd->codec_dai, *cpu_dai = rtd->cpu_dai; + struct snd_soc_dai *codec_dai = rtd->codec_dai; + struct snd_soc_dai *cpu_dai = rtd->cpu_dai; + struct snd_soc_dapm_widget *play_w, *capture_w; int ret; dev_dbg(card->dev, "probe %s dai link %d late %d\n", card->name, num, order); /* config components */ - codec_dai->codec = codec; cpu_dai->platform = platform; codec_dai->card = card; cpu_dai->card = card; @@ -1218,9 +1255,12 @@ static int soc_probe_dai_link(struct snd_soc_card *card, int num, int order) /* probe the cpu_dai */ if (!cpu_dai->probed && cpu_dai->driver->probe_order == order) { + cpu_dai->dapm.card = card; if (!try_module_get(cpu_dai->dev->driver->owner)) return -ENODEV; + snd_soc_dapm_new_dai_widgets(&cpu_dai->dapm, cpu_dai); + if (cpu_dai->driver->probe) { ret = cpu_dai->driver->probe(cpu_dai); if (ret < 0) { @@ -1279,12 +1319,39 @@ static int soc_probe_dai_link(struct snd_soc_card *card, int num, int order) if (ret < 0) pr_warn("asoc: failed to add pmdown_time sysfs:%d\n", ret); - /* create the pcm */ - ret = soc_new_pcm(rtd, num); - if (ret < 0) { - pr_err("asoc: can't create pcm %s :%d\n", - dai_link->stream_name, ret); - return ret; + if (!dai_link->params) { + /* create the pcm */ + ret = soc_new_pcm(rtd, num); + if (ret < 0) { + pr_err("asoc: can't create pcm %s :%d\n", + dai_link->stream_name, ret); + return ret; + } + } else { + /* link the DAI widgets */ + play_w = codec_dai->playback_widget; + capture_w = cpu_dai->capture_widget; + if (play_w && capture_w) { + ret = snd_soc_dapm_new_pcm(card, dai_link->params, + capture_w, play_w); + if (ret != 0) { + dev_err(card->dev, "Can't link %s to %s: %d\n", + play_w->name, capture_w->name, ret); + return ret; + } + } + + play_w = cpu_dai->playback_widget; + capture_w = codec_dai->capture_widget; + if (play_w && capture_w) { + ret = snd_soc_dapm_new_pcm(card, dai_link->params, + capture_w, play_w); + if (ret != 0) { + dev_err(card->dev, "Can't link %s to %s: %d\n", + play_w->name, capture_w->name, ret); + return ret; + } + } } /* add platform data for AC97 devices */ @@ -1334,6 +1401,20 @@ static void soc_unregister_ac97_dai_link(struct snd_soc_codec *codec) } #endif +static int soc_check_aux_dev(struct snd_soc_card *card, int num) +{ + struct snd_soc_aux_dev *aux_dev = &card->aux_dev[num]; + struct snd_soc_codec *codec; + + /* find CODEC from registered CODECs*/ + list_for_each_entry(codec, &codec_list, list) { + if (!strcmp(codec->name, aux_dev->codec_name)) + return 0; + } + + return -EPROBE_DEFER; +} + static int soc_probe_aux_dev(struct snd_soc_card *card, int num) { struct snd_soc_aux_dev *aux_dev = &card->aux_dev[num]; @@ -1354,7 +1435,7 @@ static int soc_probe_aux_dev(struct snd_soc_card *card, int num) } /* codec not found */ dev_err(card->dev, "asoc: codec %s not found", aux_dev->codec_name); - goto out; + return -EPROBE_DEFER; found: ret = soc_probe_codec(card, codec); @@ -1404,29 +1485,28 @@ static int snd_soc_init_codec_cache(struct snd_soc_codec *codec, return 0; } -static void snd_soc_instantiate_card(struct snd_soc_card *card) +static int snd_soc_instantiate_card(struct snd_soc_card *card) { struct snd_soc_codec *codec; struct snd_soc_codec_conf *codec_conf; enum snd_soc_compress_type compress_type; struct snd_soc_dai_link *dai_link; - int ret, i, order; + int ret, i, order, dai_fmt; - mutex_lock(&card->mutex); - - if (card->instantiated) { - mutex_unlock(&card->mutex); - return; - } + mutex_lock_nested(&card->mutex, SND_SOC_CARD_CLASS_INIT); /* bind DAIs */ - for (i = 0; i < card->num_links; i++) - soc_bind_dai_link(card, i); + for (i = 0; i < card->num_links; i++) { + ret = soc_bind_dai_link(card, i); + if (ret != 0) + goto base_error; + } - /* bind completed ? */ - if (card->num_rtd != card->num_links) { - mutex_unlock(&card->mutex); - return; + /* check aux_devs too */ + for (i = 0; i < card->num_aux_devs; i++) { + ret = soc_check_aux_dev(card, i); + if (ret != 0) + goto base_error; } /* initialize the register cache for each available codec */ @@ -1446,10 +1526,8 @@ static void snd_soc_instantiate_card(struct snd_soc_card *card) } } ret = snd_soc_init_codec_cache(codec, compress_type); - if (ret < 0) { - mutex_unlock(&card->mutex); - return; - } + if (ret < 0) + goto base_error; } /* card bind complete so register a sound card */ @@ -1458,8 +1536,7 @@ static void snd_soc_instantiate_card(struct snd_soc_card *card) if (ret < 0) { pr_err("asoc: can't create sound card for card %s: %d\n", card->name, ret); - mutex_unlock(&card->mutex); - return; + goto base_error; } card->snd_card->dev = card->dev; @@ -1523,17 +1600,47 @@ static void snd_soc_instantiate_card(struct snd_soc_card *card) for (i = 0; i < card->num_links; i++) { dai_link = &card->dai_link[i]; + dai_fmt = dai_link->dai_fmt; - if (dai_link->dai_fmt) { + if (dai_fmt) { ret = snd_soc_dai_set_fmt(card->rtd[i].codec_dai, - dai_link->dai_fmt); + dai_fmt); if (ret != 0 && ret != -ENOTSUPP) dev_warn(card->rtd[i].codec_dai->dev, "Failed to set DAI format: %d\n", ret); + } + /* If this is a regular CPU link there will be a platform */ + if (dai_fmt && + (dai_link->platform_name || dai_link->platform_of_node)) { ret = snd_soc_dai_set_fmt(card->rtd[i].cpu_dai, - dai_link->dai_fmt); + dai_fmt); + if (ret != 0 && ret != -ENOTSUPP) + dev_warn(card->rtd[i].cpu_dai->dev, + "Failed to set DAI format: %d\n", + ret); + } else if (dai_fmt) { + /* Flip the polarity for the "CPU" end */ + dai_fmt &= ~SND_SOC_DAIFMT_MASTER_MASK; + switch (dai_link->dai_fmt & + SND_SOC_DAIFMT_MASTER_MASK) { + case SND_SOC_DAIFMT_CBM_CFM: + dai_fmt |= SND_SOC_DAIFMT_CBS_CFS; + break; + case SND_SOC_DAIFMT_CBM_CFS: + dai_fmt |= SND_SOC_DAIFMT_CBS_CFM; + break; + case SND_SOC_DAIFMT_CBS_CFM: + dai_fmt |= SND_SOC_DAIFMT_CBM_CFS; + break; + case SND_SOC_DAIFMT_CBS_CFS: + dai_fmt |= SND_SOC_DAIFMT_CBM_CFM; + break; + } + + ret = snd_soc_dai_set_fmt(card->rtd[i].cpu_dai, + dai_fmt); if (ret != 0 && ret != -ENOTSUPP) dev_warn(card->rtd[i].cpu_dai->dev, "Failed to set DAI format: %d\n", @@ -1599,7 +1706,8 @@ static void snd_soc_instantiate_card(struct snd_soc_card *card) card->instantiated = 1; snd_soc_dapm_sync(&card->dapm); mutex_unlock(&card->mutex); - return; + + return 0; probe_aux_dev_err: for (i = 0; i < card->num_aux_devs; i++) @@ -1614,18 +1722,10 @@ card_probe_error: snd_card_free(card->snd_card); +base_error: mutex_unlock(&card->mutex); -} -/* - * Attempt to initialise any uninitialised cards. Must be called with - * client_mutex. - */ -static void snd_soc_instantiate_cards(void) -{ - struct snd_soc_card *card; - list_for_each_entry(card, &card_list, list) - snd_soc_instantiate_card(card); + return ret; } /* probes a new socdev */ @@ -2527,6 +2627,87 @@ int snd_soc_put_volsw(struct snd_kcontrol *kcontrol, EXPORT_SYMBOL_GPL(snd_soc_put_volsw); /** + * snd_soc_get_volsw_sx - single mixer get callback + * @kcontrol: mixer control + * @ucontrol: control element information + * + * Callback to get the value of a single mixer control, or a double mixer + * control that spans 2 registers. + * + * Returns 0 for success. + */ +int snd_soc_get_volsw_sx(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol); + struct soc_mixer_control *mc = + (struct soc_mixer_control *)kcontrol->private_value; + + unsigned int reg = mc->reg; + unsigned int reg2 = mc->rreg; + unsigned int shift = mc->shift; + unsigned int rshift = mc->rshift; + int max = mc->max; + int min = mc->min; + int mask = (1 << (fls(min + max) - 1)) - 1; + + ucontrol->value.integer.value[0] = + ((snd_soc_read(codec, reg) >> shift) - min) & mask; + + if (snd_soc_volsw_is_stereo(mc)) + ucontrol->value.integer.value[1] = + ((snd_soc_read(codec, reg2) >> rshift) - min) & mask; + + return 0; +} +EXPORT_SYMBOL_GPL(snd_soc_get_volsw_sx); + +/** + * snd_soc_put_volsw_sx - double mixer set callback + * @kcontrol: mixer control + * @uinfo: control element information + * + * Callback to set the value of a double mixer control that spans 2 registers. + * + * Returns 0 for success. + */ +int snd_soc_put_volsw_sx(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol); + struct soc_mixer_control *mc = + (struct soc_mixer_control *)kcontrol->private_value; + + unsigned int reg = mc->reg; + unsigned int reg2 = mc->rreg; + unsigned int shift = mc->shift; + unsigned int rshift = mc->rshift; + int max = mc->max; + int min = mc->min; + int mask = (1 << (fls(min + max) - 1)) - 1; + int err = 0; + unsigned short val, val_mask, val2 = 0; + + val_mask = mask << shift; + val = (ucontrol->value.integer.value[0] + min) & mask; + val = val << shift; + + if (snd_soc_update_bits_locked(codec, reg, val_mask, val)) + return err; + + if (snd_soc_volsw_is_stereo(mc)) { + val_mask = mask << rshift; + val2 = (ucontrol->value.integer.value[1] + min) & mask; + val2 = val2 << rshift; + + if (snd_soc_update_bits_locked(codec, reg2, val_mask, val2)) + return err; + } + return 0; +} +EXPORT_SYMBOL_GPL(snd_soc_put_volsw_sx); + +/** * snd_soc_info_volsw_s8 - signed mixer info callback * @kcontrol: mixer control * @uinfo: control element information @@ -2647,99 +2828,6 @@ int snd_soc_limit_volume(struct snd_soc_codec *codec, } EXPORT_SYMBOL_GPL(snd_soc_limit_volume); -/** - * snd_soc_info_volsw_2r_sx - double with tlv and variable data size - * mixer info callback - * @kcontrol: mixer control - * @uinfo: control element information - * - * Returns 0 for success. - */ -int snd_soc_info_volsw_2r_sx(struct snd_kcontrol *kcontrol, - struct snd_ctl_elem_info *uinfo) -{ - struct soc_mixer_control *mc = - (struct soc_mixer_control *)kcontrol->private_value; - int max = mc->max; - int min = mc->min; - - uinfo->type = SNDRV_CTL_ELEM_TYPE_INTEGER; - uinfo->count = 2; - uinfo->value.integer.min = 0; - uinfo->value.integer.max = max-min; - - return 0; -} -EXPORT_SYMBOL_GPL(snd_soc_info_volsw_2r_sx); - -/** - * snd_soc_get_volsw_2r_sx - double with tlv and variable data size - * mixer get callback - * @kcontrol: mixer control - * @uinfo: control element information - * - * Returns 0 for success. - */ -int snd_soc_get_volsw_2r_sx(struct snd_kcontrol *kcontrol, - struct snd_ctl_elem_value *ucontrol) -{ - struct soc_mixer_control *mc = - (struct soc_mixer_control *)kcontrol->private_value; - struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol); - unsigned int mask = (1<<mc->shift)-1; - int min = mc->min; - int val = snd_soc_read(codec, mc->reg) & mask; - int valr = snd_soc_read(codec, mc->rreg) & mask; - - ucontrol->value.integer.value[0] = ((val & 0xff)-min) & mask; - ucontrol->value.integer.value[1] = ((valr & 0xff)-min) & mask; - return 0; -} -EXPORT_SYMBOL_GPL(snd_soc_get_volsw_2r_sx); - -/** - * snd_soc_put_volsw_2r_sx - double with tlv and variable data size - * mixer put callback - * @kcontrol: mixer control - * @uinfo: control element information - * - * Returns 0 for success. - */ -int snd_soc_put_volsw_2r_sx(struct snd_kcontrol *kcontrol, - struct snd_ctl_elem_value *ucontrol) -{ - struct soc_mixer_control *mc = - (struct soc_mixer_control *)kcontrol->private_value; - struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol); - unsigned int mask = (1<<mc->shift)-1; - int min = mc->min; - int ret; - unsigned int val, valr, oval, ovalr; - - val = ((ucontrol->value.integer.value[0]+min) & 0xff); - val &= mask; - valr = ((ucontrol->value.integer.value[1]+min) & 0xff); - valr &= mask; - - oval = snd_soc_read(codec, mc->reg) & mask; - ovalr = snd_soc_read(codec, mc->rreg) & mask; - - ret = 0; - if (oval != val) { - ret = snd_soc_write(codec, mc->reg, val); - if (ret < 0) - return ret; - } - if (ovalr != valr) { - ret = snd_soc_write(codec, mc->rreg, valr); - if (ret < 0) - return ret; - } - - return 0; -} -EXPORT_SYMBOL_GPL(snd_soc_put_volsw_2r_sx); - int snd_soc_bytes_info(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_info *uinfo) { @@ -2850,6 +2938,186 @@ int snd_soc_bytes_put(struct snd_kcontrol *kcontrol, EXPORT_SYMBOL_GPL(snd_soc_bytes_put); /** + * snd_soc_info_xr_sx - signed multi register info callback + * @kcontrol: mreg control + * @uinfo: control element information + * + * Callback to provide information of a control that can + * span multiple codec registers which together + * forms a single signed value in a MSB/LSB manner. + * + * Returns 0 for success. + */ +int snd_soc_info_xr_sx(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_info *uinfo) +{ + struct soc_mreg_control *mc = + (struct soc_mreg_control *)kcontrol->private_value; + uinfo->type = SNDRV_CTL_ELEM_TYPE_INTEGER; + uinfo->count = 1; + uinfo->value.integer.min = mc->min; + uinfo->value.integer.max = mc->max; + + return 0; +} +EXPORT_SYMBOL_GPL(snd_soc_info_xr_sx); + +/** + * snd_soc_get_xr_sx - signed multi register get callback + * @kcontrol: mreg control + * @ucontrol: control element information + * + * Callback to get the value of a control that can span + * multiple codec registers which together forms a single + * signed value in a MSB/LSB manner. The control supports + * specifying total no of bits used to allow for bitfields + * across the multiple codec registers. + * + * Returns 0 for success. + */ +int snd_soc_get_xr_sx(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct soc_mreg_control *mc = + (struct soc_mreg_control *)kcontrol->private_value; + struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol); + unsigned int regbase = mc->regbase; + unsigned int regcount = mc->regcount; + unsigned int regwshift = codec->driver->reg_word_size * BITS_PER_BYTE; + unsigned int regwmask = (1<<regwshift)-1; + unsigned int invert = mc->invert; + unsigned long mask = (1UL<<mc->nbits)-1; + long min = mc->min; + long max = mc->max; + long val = 0; + unsigned long regval; + unsigned int i; + + for (i = 0; i < regcount; i++) { + regval = snd_soc_read(codec, regbase+i) & regwmask; + val |= regval << (regwshift*(regcount-i-1)); + } + val &= mask; + if (min < 0 && val > max) + val |= ~mask; + if (invert) + val = max - val; + ucontrol->value.integer.value[0] = val; + + return 0; +} +EXPORT_SYMBOL_GPL(snd_soc_get_xr_sx); + +/** + * snd_soc_put_xr_sx - signed multi register get callback + * @kcontrol: mreg control + * @ucontrol: control element information + * + * Callback to set the value of a control that can span + * multiple codec registers which together forms a single + * signed value in a MSB/LSB manner. The control supports + * specifying total no of bits used to allow for bitfields + * across the multiple codec registers. + * + * Returns 0 for success. + */ +int snd_soc_put_xr_sx(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct soc_mreg_control *mc = + (struct soc_mreg_control *)kcontrol->private_value; + struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol); + unsigned int regbase = mc->regbase; + unsigned int regcount = mc->regcount; + unsigned int regwshift = codec->driver->reg_word_size * BITS_PER_BYTE; + unsigned int regwmask = (1<<regwshift)-1; + unsigned int invert = mc->invert; + unsigned long mask = (1UL<<mc->nbits)-1; + long max = mc->max; + long val = ucontrol->value.integer.value[0]; + unsigned int i, regval, regmask; + int err; + + if (invert) + val = max - val; + val &= mask; + for (i = 0; i < regcount; i++) { + regval = (val >> (regwshift*(regcount-i-1))) & regwmask; + regmask = (mask >> (regwshift*(regcount-i-1))) & regwmask; + err = snd_soc_update_bits_locked(codec, regbase+i, + regmask, regval); + if (err < 0) + return err; + } + + return 0; +} +EXPORT_SYMBOL_GPL(snd_soc_put_xr_sx); + +/** + * snd_soc_get_strobe - strobe get callback + * @kcontrol: mixer control + * @ucontrol: control element information + * + * Callback get the value of a strobe mixer control. + * + * Returns 0 for success. + */ +int snd_soc_get_strobe(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct soc_mixer_control *mc = + (struct soc_mixer_control *)kcontrol->private_value; + struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol); + unsigned int reg = mc->reg; + unsigned int shift = mc->shift; + unsigned int mask = 1 << shift; + unsigned int invert = mc->invert != 0; + unsigned int val = snd_soc_read(codec, reg) & mask; + + if (shift != 0 && val != 0) + val = val >> shift; + ucontrol->value.enumerated.item[0] = val ^ invert; + + return 0; +} +EXPORT_SYMBOL_GPL(snd_soc_get_strobe); + +/** + * snd_soc_put_strobe - strobe put callback + * @kcontrol: mixer control + * @ucontrol: control element information + * + * Callback strobe a register bit to high then low (or the inverse) + * in one pass of a single mixer enum control. + * + * Returns 1 for success. + */ +int snd_soc_put_strobe(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct soc_mixer_control *mc = + (struct soc_mixer_control *)kcontrol->private_value; + struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol); + unsigned int reg = mc->reg; + unsigned int shift = mc->shift; + unsigned int mask = 1 << shift; + unsigned int invert = mc->invert != 0; + unsigned int strobe = ucontrol->value.enumerated.item[0] != 0; + unsigned int val1 = (strobe ^ invert) ? mask : 0; + unsigned int val2 = (strobe ^ invert) ? 0 : mask; + int err; + + err = snd_soc_update_bits_locked(codec, reg, mask, val1); + if (err < 0) + return err; + + err = snd_soc_update_bits_locked(codec, reg, mask, val2); + return err; +} +EXPORT_SYMBOL_GPL(snd_soc_put_strobe); + +/** * snd_soc_dai_set_sysclk - configure DAI system or master clock. * @dai: DAI * @clk_id: DAI specific clock ID @@ -3048,7 +3316,7 @@ int snd_soc_dai_digital_mute(struct snd_soc_dai *dai, int mute) if (dai->driver && dai->driver->ops->digital_mute) return dai->driver->ops->digital_mute(dai, mute); else - return -EINVAL; + return -ENOTSUPP; } EXPORT_SYMBOL_GPL(snd_soc_dai_digital_mute); @@ -3060,7 +3328,7 @@ EXPORT_SYMBOL_GPL(snd_soc_dai_digital_mute); */ int snd_soc_register_card(struct snd_soc_card *card) { - int i; + int i, ret; if (!card->name || !card->dev) return -EINVAL; @@ -3123,15 +3391,13 @@ int snd_soc_register_card(struct snd_soc_card *card) INIT_LIST_HEAD(&card->dapm_dirty); card->instantiated = 0; mutex_init(&card->mutex); + mutex_init(&card->dapm_mutex); - mutex_lock(&client_mutex); - list_add(&card->list, &card_list); - snd_soc_instantiate_cards(); - mutex_unlock(&client_mutex); - - dev_dbg(card->dev, "Registered card '%s'\n", card->name); + ret = snd_soc_instantiate_card(card); + if (ret != 0) + soc_cleanup_card_debugfs(card); - return 0; + return ret; } EXPORT_SYMBOL_GPL(snd_soc_register_card); @@ -3145,9 +3411,6 @@ int snd_soc_unregister_card(struct snd_soc_card *card) { if (card->instantiated) soc_cleanup_card_resources(card); - mutex_lock(&client_mutex); - list_del(&card->list); - mutex_unlock(&client_mutex); dev_dbg(card->dev, "Unregistered card '%s'\n", card->name); return 0; @@ -3221,6 +3484,7 @@ static inline char *fmt_multiple_name(struct device *dev, int snd_soc_register_dai(struct device *dev, struct snd_soc_dai_driver *dai_drv) { + struct snd_soc_codec *codec; struct snd_soc_dai *dai; dev_dbg(dev, "dai register %s\n", dev_name(dev)); @@ -3238,12 +3502,23 @@ int snd_soc_register_dai(struct device *dev, dai->dev = dev; dai->driver = dai_drv; + dai->dapm.dev = dev; if (!dai->driver->ops) dai->driver->ops = &null_dai_ops; mutex_lock(&client_mutex); + + list_for_each_entry(codec, &codec_list, list) { + if (codec->dev == dev) { + dev_dbg(dev, "Mapped DAI %s to CODEC %s\n", + dai->name, codec->name); + dai->codec = codec; + break; + } + } + list_add(&dai->list, &dai_list); - snd_soc_instantiate_cards(); + mutex_unlock(&client_mutex); pr_debug("Registered DAI '%s'\n", dai->name); @@ -3287,6 +3562,7 @@ EXPORT_SYMBOL_GPL(snd_soc_unregister_dai); int snd_soc_register_dais(struct device *dev, struct snd_soc_dai_driver *dai_drv, size_t count) { + struct snd_soc_codec *codec; struct snd_soc_dai *dai; int i, ret = 0; @@ -3314,19 +3590,28 @@ int snd_soc_register_dais(struct device *dev, dai->id = dai->driver->id; else dai->id = i; + dai->dapm.dev = dev; if (!dai->driver->ops) dai->driver->ops = &null_dai_ops; mutex_lock(&client_mutex); + + list_for_each_entry(codec, &codec_list, list) { + if (codec->dev == dev) { + dev_dbg(dev, "Mapped DAI %s to CODEC %s\n", + dai->name, codec->name); + dai->codec = codec; + break; + } + } + list_add(&dai->list, &dai_list); + mutex_unlock(&client_mutex); pr_debug("Registered DAI '%s'\n", dai->name); } - mutex_lock(&client_mutex); - snd_soc_instantiate_cards(); - mutex_unlock(&client_mutex); return 0; err: @@ -3384,7 +3669,6 @@ int snd_soc_register_platform(struct device *dev, mutex_lock(&client_mutex); list_add(&platform->list, &platform_list); - snd_soc_instantiate_cards(); mutex_unlock(&client_mutex); pr_debug("Registered platform '%s'\n", platform->name); @@ -3534,18 +3818,18 @@ int snd_soc_register_codec(struct device *dev, fixup_codec_formats(&dai_drv[i].capture); } + mutex_lock(&client_mutex); + list_add(&codec->list, &codec_list); + mutex_unlock(&client_mutex); + /* register any DAIs */ if (num_dai) { ret = snd_soc_register_dais(dev, dai_drv, num_dai); if (ret < 0) - goto fail; + dev_err(codec->dev, "Failed to regster DAIs: %d\n", + ret); } - mutex_lock(&client_mutex); - list_add(&codec->list, &codec_list); - snd_soc_instantiate_cards(); - mutex_unlock(&client_mutex); - pr_debug("Registered codec '%s'\n", codec->name); return 0; diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c index 1bb6d4a63cd..90ee77d2409 100644 --- a/sound/soc/soc-dapm.c +++ b/sound/soc/soc-dapm.c @@ -52,6 +52,7 @@ static int dapm_up_seq[] = { [snd_soc_dapm_supply] = 1, [snd_soc_dapm_regulator_supply] = 1, [snd_soc_dapm_micbias] = 2, + [snd_soc_dapm_dai_link] = 2, [snd_soc_dapm_dai] = 3, [snd_soc_dapm_aif_in] = 3, [snd_soc_dapm_aif_out] = 3, @@ -90,9 +91,10 @@ static int dapm_down_seq[] = { [snd_soc_dapm_aif_in] = 10, [snd_soc_dapm_aif_out] = 10, [snd_soc_dapm_dai] = 10, - [snd_soc_dapm_regulator_supply] = 11, - [snd_soc_dapm_supply] = 11, - [snd_soc_dapm_post] = 12, + [snd_soc_dapm_dai_link] = 11, + [snd_soc_dapm_regulator_supply] = 12, + [snd_soc_dapm_supply] = 12, + [snd_soc_dapm_post] = 13, }; static void pop_wait(u32 pop_time) @@ -208,7 +210,23 @@ static int soc_widget_write(struct snd_soc_dapm_widget *w, int reg, int val) return -1; } -static int soc_widget_update_bits(struct snd_soc_dapm_widget *w, +static inline void soc_widget_lock(struct snd_soc_dapm_widget *w) +{ + if (w->codec && !w->codec->using_regmap) + mutex_lock(&w->codec->mutex); + else if (w->platform) + mutex_lock(&w->platform->mutex); +} + +static inline void soc_widget_unlock(struct snd_soc_dapm_widget *w) +{ + if (w->codec && !w->codec->using_regmap) + mutex_unlock(&w->codec->mutex); + else if (w->platform) + mutex_unlock(&w->platform->mutex); +} + +static int soc_widget_update_bits_locked(struct snd_soc_dapm_widget *w, unsigned short reg, unsigned int mask, unsigned int value) { bool change; @@ -221,18 +239,24 @@ static int soc_widget_update_bits(struct snd_soc_dapm_widget *w, if (ret != 0) return ret; } else { + soc_widget_lock(w); ret = soc_widget_read(w, reg); - if (ret < 0) + if (ret < 0) { + soc_widget_unlock(w); return ret; + } old = ret; new = (old & ~mask) | (value & mask); change = old != new; if (change) { ret = soc_widget_write(w, reg, new); - if (ret < 0) + if (ret < 0) { + soc_widget_unlock(w); return ret; + } } + soc_widget_unlock(w); } return change; @@ -374,6 +398,7 @@ static void dapm_set_path_status(struct snd_soc_dapm_widget *w, case snd_soc_dapm_mic: case snd_soc_dapm_spk: case snd_soc_dapm_line: + case snd_soc_dapm_dai_link: p->connect = 1; break; /* does affect routing - dynamically connected */ @@ -682,11 +707,51 @@ static int snd_soc_dapm_suspend_check(struct snd_soc_dapm_widget *widget) } } +/* add widget to list if it's not already in the list */ +static int dapm_list_add_widget(struct snd_soc_dapm_widget_list **list, + struct snd_soc_dapm_widget *w) +{ + struct snd_soc_dapm_widget_list *wlist; + int wlistsize, wlistentries, i; + + if (*list == NULL) + return -EINVAL; + + wlist = *list; + + /* is this widget already in the list */ + for (i = 0; i < wlist->num_widgets; i++) { + if (wlist->widgets[i] == w) + return 0; + } + + /* allocate some new space */ + wlistentries = wlist->num_widgets + 1; + wlistsize = sizeof(struct snd_soc_dapm_widget_list) + + wlistentries * sizeof(struct snd_soc_dapm_widget *); + *list = krealloc(wlist, wlistsize, GFP_KERNEL); + if (*list == NULL) { + dev_err(w->dapm->dev, "can't allocate widget list for %s\n", + w->name); + return -ENOMEM; + } + wlist = *list; + + /* insert the widget */ + dev_dbg(w->dapm->dev, "added %s in widget list pos %d\n", + w->name, wlist->num_widgets); + + wlist->widgets[wlist->num_widgets] = w; + wlist->num_widgets++; + return 1; +} + /* * Recursively check for a completed path to an active or physically connected * output widget. Returns number of complete paths. */ -static int is_connected_output_ep(struct snd_soc_dapm_widget *widget) +static int is_connected_output_ep(struct snd_soc_dapm_widget *widget, + struct snd_soc_dapm_widget_list **list) { struct snd_soc_dapm_path *path; int con = 0; @@ -742,9 +807,23 @@ static int is_connected_output_ep(struct snd_soc_dapm_widget *widget) if (path->walked) continue; + trace_snd_soc_dapm_output_path(widget, path); + if (path->sink && path->connect) { path->walked = 1; - con += is_connected_output_ep(path->sink); + + /* do we need to add this widget to the list ? */ + if (list) { + int err; + err = dapm_list_add_widget(list, path->sink); + if (err < 0) { + dev_err(widget->dapm->dev, "could not add widget %s\n", + widget->name); + return con; + } + } + + con += is_connected_output_ep(path->sink, list); } } @@ -757,7 +836,8 @@ static int is_connected_output_ep(struct snd_soc_dapm_widget *widget) * Recursively check for a completed path to an active or physically connected * input widget. Returns number of complete paths. */ -static int is_connected_input_ep(struct snd_soc_dapm_widget *widget) +static int is_connected_input_ep(struct snd_soc_dapm_widget *widget, + struct snd_soc_dapm_widget_list **list) { struct snd_soc_dapm_path *path; int con = 0; @@ -825,9 +905,23 @@ static int is_connected_input_ep(struct snd_soc_dapm_widget *widget) if (path->walked) continue; + trace_snd_soc_dapm_input_path(widget, path); + if (path->source && path->connect) { path->walked = 1; - con += is_connected_input_ep(path->source); + + /* do we need to add this widget to the list ? */ + if (list) { + int err; + err = dapm_list_add_widget(list, path->sink); + if (err < 0) { + dev_err(widget->dapm->dev, "could not add widget %s\n", + widget->name); + return con; + } + } + + con += is_connected_input_ep(path->source, list); } } @@ -836,6 +930,39 @@ static int is_connected_input_ep(struct snd_soc_dapm_widget *widget) return con; } +/** + * snd_soc_dapm_get_connected_widgets - query audio path and it's widgets. + * @dai: the soc DAI. + * @stream: stream direction. + * @list: list of active widgets for this stream. + * + * Queries DAPM graph as to whether an valid audio stream path exists for + * the initial stream specified by name. This takes into account + * current mixer and mux kcontrol settings. Creates list of valid widgets. + * + * Returns the number of valid paths or negative error. + */ +int snd_soc_dapm_dai_get_connected_widgets(struct snd_soc_dai *dai, int stream, + struct snd_soc_dapm_widget_list **list) +{ + struct snd_soc_card *card = dai->card; + int paths; + + mutex_lock_nested(&card->dapm_mutex, SND_SOC_DAPM_CLASS_RUNTIME); + dapm_reset(card); + + if (stream == SNDRV_PCM_STREAM_PLAYBACK) + paths = is_connected_output_ep(dai->playback_widget, list); + else + paths = is_connected_input_ep(dai->playback_widget, list); + + trace_snd_soc_dapm_connected(paths, stream); + dapm_clear_walk(&card->dapm); + mutex_unlock(&card->dapm_mutex); + + return paths; +} + /* * Handler for generic register modifier widget. */ @@ -849,7 +976,7 @@ int dapm_reg_event(struct snd_soc_dapm_widget *w, else val = w->off_val; - soc_widget_update_bits(w, -(w->reg + 1), + soc_widget_update_bits_locked(w, -(w->reg + 1), w->mask << w->shift, val << w->shift); return 0; @@ -863,9 +990,9 @@ int dapm_regulator_event(struct snd_soc_dapm_widget *w, struct snd_kcontrol *kcontrol, int event) { if (SND_SOC_DAPM_EVENT_ON(event)) - return regulator_enable(w->priv); + return regulator_enable(w->regulator); else - return regulator_disable_deferred(w->priv, w->shift); + return regulator_disable_deferred(w->regulator, w->shift); } EXPORT_SYMBOL_GPL(dapm_regulator_event); @@ -892,9 +1019,9 @@ static int dapm_generic_check_power(struct snd_soc_dapm_widget *w) DAPM_UPDATE_STAT(w, power_checks); - in = is_connected_input_ep(w); + in = is_connected_input_ep(w, NULL); dapm_clear_walk(w->dapm); - out = is_connected_output_ep(w); + out = is_connected_output_ep(w, NULL); dapm_clear_walk(w->dapm); return out != 0 && in != 0; } @@ -903,7 +1030,10 @@ static int dapm_dai_check_power(struct snd_soc_dapm_widget *w) { DAPM_UPDATE_STAT(w, power_checks); - return w->active; + if (w->active) + return w->active; + + return dapm_generic_check_power(w); } /* Check to see if an ADC has power */ @@ -914,7 +1044,7 @@ static int dapm_adc_check_power(struct snd_soc_dapm_widget *w) DAPM_UPDATE_STAT(w, power_checks); if (w->active) { - in = is_connected_input_ep(w); + in = is_connected_input_ep(w, NULL); dapm_clear_walk(w->dapm); return in != 0; } else { @@ -930,7 +1060,7 @@ static int dapm_dac_check_power(struct snd_soc_dapm_widget *w) DAPM_UPDATE_STAT(w, power_checks); if (w->active) { - out = is_connected_output_ep(w); + out = is_connected_output_ep(w, NULL); dapm_clear_walk(w->dapm); return out != 0; } else { @@ -1107,7 +1237,7 @@ static void dapm_seq_run_coalesced(struct snd_soc_dapm_context *dapm, "pop test : Applying 0x%x/0x%x to %x in %dms\n", value, mask, reg, card->pop_time); pop_wait(card->pop_time); - soc_widget_update_bits(w, reg, mask, value); + soc_widget_update_bits_locked(w, reg, mask, value); } list_for_each_entry(w, pending, power_list) { @@ -1237,7 +1367,7 @@ static void dapm_widget_update(struct snd_soc_dapm_context *dapm) w->name, ret); } - ret = snd_soc_update_bits(w->codec, update->reg, update->mask, + ret = soc_widget_update_bits_locked(w, update->reg, update->mask, update->val); if (ret < 0) pr_err("%s DAPM update failed: %d\n", w->name, ret); @@ -1421,12 +1551,10 @@ static int dapm_power_widgets(struct snd_soc_dapm_context *dapm, int event) trace_snd_soc_dapm_start(card); list_for_each_entry(d, &card->dapm_list, list) { - if (d->n_widgets || d->codec == NULL) { - if (d->idle_bias_off) - d->target_bias_level = SND_SOC_BIAS_OFF; - else - d->target_bias_level = SND_SOC_BIAS_STANDBY; - } + if (d->idle_bias_off) + d->target_bias_level = SND_SOC_BIAS_OFF; + else + d->target_bias_level = SND_SOC_BIAS_STANDBY; } dapm_reset(card); @@ -1471,32 +1599,6 @@ static int dapm_power_widgets(struct snd_soc_dapm_context *dapm, int event) } - /* If there are no DAPM widgets then try to figure out power from the - * event type. - */ - if (!dapm->n_widgets) { - switch (event) { - case SND_SOC_DAPM_STREAM_START: - case SND_SOC_DAPM_STREAM_RESUME: - dapm->target_bias_level = SND_SOC_BIAS_ON; - break; - case SND_SOC_DAPM_STREAM_STOP: - if (dapm->codec && dapm->codec->active) - dapm->target_bias_level = SND_SOC_BIAS_ON; - else - dapm->target_bias_level = SND_SOC_BIAS_STANDBY; - break; - case SND_SOC_DAPM_STREAM_SUSPEND: - dapm->target_bias_level = SND_SOC_BIAS_STANDBY; - break; - case SND_SOC_DAPM_STREAM_NOP: - dapm->target_bias_level = dapm->bias_level; - break; - default: - break; - } - } - /* Force all contexts in the card to the same bias state if * they're not ground referenced. */ @@ -1560,9 +1662,9 @@ static ssize_t dapm_widget_power_read_file(struct file *file, if (!buf) return -ENOMEM; - in = is_connected_input_ep(w); + in = is_connected_input_ep(w, NULL); dapm_clear_walk(w->dapm); - out = is_connected_output_ep(w); + out = is_connected_output_ep(w, NULL); dapm_clear_walk(w->dapm); ret = snprintf(buf, PAGE_SIZE, "%s: %s%s in %d out %d", @@ -1709,7 +1811,7 @@ static inline void dapm_debugfs_cleanup(struct snd_soc_dapm_context *dapm) #endif /* test and update the power status of a mux widget */ -int snd_soc_dapm_mux_update_power(struct snd_soc_dapm_widget *widget, +static int soc_dapm_mux_update_power(struct snd_soc_dapm_widget *widget, struct snd_kcontrol *kcontrol, int mux, struct soc_enum *e) { struct snd_soc_dapm_path *path; @@ -1746,12 +1848,26 @@ int snd_soc_dapm_mux_update_power(struct snd_soc_dapm_widget *widget, dapm_power_widgets(widget->dapm, SND_SOC_DAPM_STREAM_NOP); } - return 0; + return found; +} + +int snd_soc_dapm_mux_update_power(struct snd_soc_dapm_widget *widget, + struct snd_kcontrol *kcontrol, int mux, struct soc_enum *e) +{ + struct snd_soc_card *card = widget->dapm->card; + int ret; + + mutex_lock_nested(&card->dapm_mutex, SND_SOC_DAPM_CLASS_RUNTIME); + ret = soc_dapm_mux_update_power(widget, kcontrol, mux, e); + mutex_unlock(&card->dapm_mutex); + if (ret > 0) + soc_dpcm_runtime_update(widget); + return ret; } EXPORT_SYMBOL_GPL(snd_soc_dapm_mux_update_power); /* test and update the power status of a mixer or switch widget */ -int snd_soc_dapm_mixer_update_power(struct snd_soc_dapm_widget *widget, +static int soc_dapm_mixer_update_power(struct snd_soc_dapm_widget *widget, struct snd_kcontrol *kcontrol, int connect) { struct snd_soc_dapm_path *path; @@ -1778,7 +1894,21 @@ int snd_soc_dapm_mixer_update_power(struct snd_soc_dapm_widget *widget, dapm_power_widgets(widget->dapm, SND_SOC_DAPM_STREAM_NOP); } - return 0; + return found; +} + +int snd_soc_dapm_mixer_update_power(struct snd_soc_dapm_widget *widget, + struct snd_kcontrol *kcontrol, int connect) +{ + struct snd_soc_card *card = widget->dapm->card; + int ret; + + mutex_lock_nested(&card->dapm_mutex, SND_SOC_DAPM_CLASS_RUNTIME); + ret = soc_dapm_mixer_update_power(widget, kcontrol, connect); + mutex_unlock(&card->dapm_mutex); + if (ret > 0) + soc_dpcm_runtime_update(widget); + return ret; } EXPORT_SYMBOL_GPL(snd_soc_dapm_mixer_update_power); @@ -1939,6 +2069,8 @@ static int snd_soc_dapm_set_pin(struct snd_soc_dapm_context *dapm, */ int snd_soc_dapm_sync(struct snd_soc_dapm_context *dapm) { + int ret; + /* * Suppress early reports (eg, jacks syncing their state) to avoid * silly DAPM runs during card startup. @@ -1946,7 +2078,10 @@ int snd_soc_dapm_sync(struct snd_soc_dapm_context *dapm) if (!dapm->card || !dapm->card->instantiated) return 0; - return dapm_power_widgets(dapm, SND_SOC_DAPM_STREAM_NOP); + mutex_lock_nested(&dapm->card->dapm_mutex, SND_SOC_DAPM_CLASS_RUNTIME); + ret = dapm_power_widgets(dapm, SND_SOC_DAPM_STREAM_NOP); + mutex_unlock(&dapm->card->dapm_mutex); + return ret; } EXPORT_SYMBOL_GPL(snd_soc_dapm_sync); @@ -2055,6 +2190,7 @@ static int snd_soc_dapm_add_route(struct snd_soc_dapm_context *dapm, case snd_soc_dapm_aif_in: case snd_soc_dapm_aif_out: case snd_soc_dapm_dai: + case snd_soc_dapm_dai_link: list_add(&path->list, &dapm->card->paths); list_add(&path->list_sink, &wsink->sources); list_add(&path->list_source, &wsource->sinks); @@ -2110,19 +2246,21 @@ err: int snd_soc_dapm_add_routes(struct snd_soc_dapm_context *dapm, const struct snd_soc_dapm_route *route, int num) { - int i, ret; + int i, ret = 0; + mutex_lock_nested(&dapm->card->dapm_mutex, SND_SOC_DAPM_CLASS_INIT); for (i = 0; i < num; i++) { ret = snd_soc_dapm_add_route(dapm, route); if (ret < 0) { dev_err(dapm->dev, "Failed to add route %s->%s\n", route->source, route->sink); - return ret; + break; } route++; } + mutex_unlock(&dapm->card->dapm_mutex); - return 0; + return ret; } EXPORT_SYMBOL_GPL(snd_soc_dapm_add_routes); @@ -2193,12 +2331,14 @@ int snd_soc_dapm_weak_routes(struct snd_soc_dapm_context *dapm, int i, err; int ret = 0; + mutex_lock_nested(&dapm->card->dapm_mutex, SND_SOC_DAPM_CLASS_INIT); for (i = 0; i < num; i++) { err = snd_soc_dapm_weak_route(dapm, route); if (err) ret = err; route++; } + mutex_unlock(&dapm->card->dapm_mutex); return ret; } @@ -2217,6 +2357,8 @@ int snd_soc_dapm_new_widgets(struct snd_soc_dapm_context *dapm) struct snd_soc_dapm_widget *w; unsigned int val; + mutex_lock_nested(&dapm->card->dapm_mutex, SND_SOC_DAPM_CLASS_INIT); + list_for_each_entry(w, &dapm->card->widgets, list) { if (w->new) @@ -2226,8 +2368,10 @@ int snd_soc_dapm_new_widgets(struct snd_soc_dapm_context *dapm) w->kcontrols = kzalloc(w->num_kcontrols * sizeof(struct snd_kcontrol *), GFP_KERNEL); - if (!w->kcontrols) + if (!w->kcontrols) { + mutex_unlock(&dapm->card->dapm_mutex); return -ENOMEM; + } } switch(w->id) { @@ -2267,6 +2411,7 @@ int snd_soc_dapm_new_widgets(struct snd_soc_dapm_context *dapm) } dapm_power_widgets(dapm, SND_SOC_DAPM_STREAM_NOP); + mutex_unlock(&dapm->card->dapm_mutex); return 0; } EXPORT_SYMBOL_GPL(snd_soc_dapm_new_widgets); @@ -2326,6 +2471,7 @@ int snd_soc_dapm_put_volsw(struct snd_kcontrol *kcontrol, struct snd_soc_dapm_widget_list *wlist = snd_kcontrol_chip(kcontrol); struct snd_soc_dapm_widget *widget = wlist->widgets[0]; struct snd_soc_codec *codec = widget->codec; + struct snd_soc_card *card = codec->card; struct soc_mixer_control *mc = (struct soc_mixer_control *)kcontrol->private_value; unsigned int reg = mc->reg; @@ -2352,7 +2498,7 @@ int snd_soc_dapm_put_volsw(struct snd_kcontrol *kcontrol, /* old connection must be powered down */ connect = invert ? 1 : 0; - mutex_lock(&codec->mutex); + mutex_lock_nested(&card->dapm_mutex, SND_SOC_DAPM_CLASS_RUNTIME); change = snd_soc_test_bits(widget->codec, reg, mask, val); if (change) { @@ -2368,13 +2514,13 @@ int snd_soc_dapm_put_volsw(struct snd_kcontrol *kcontrol, update.val = val; widget->dapm->update = &update; - snd_soc_dapm_mixer_update_power(widget, kcontrol, connect); + soc_dapm_mixer_update_power(widget, kcontrol, connect); widget->dapm->update = NULL; } } - mutex_unlock(&codec->mutex); + mutex_unlock(&card->dapm_mutex); return 0; } EXPORT_SYMBOL_GPL(snd_soc_dapm_put_volsw); @@ -2423,6 +2569,7 @@ int snd_soc_dapm_put_enum_double(struct snd_kcontrol *kcontrol, struct snd_soc_dapm_widget_list *wlist = snd_kcontrol_chip(kcontrol); struct snd_soc_dapm_widget *widget = wlist->widgets[0]; struct snd_soc_codec *codec = widget->codec; + struct snd_soc_card *card = codec->card; struct soc_enum *e = (struct soc_enum *)kcontrol->private_value; unsigned int val, mux, change; unsigned int mask, bitmask; @@ -2443,7 +2590,7 @@ int snd_soc_dapm_put_enum_double(struct snd_kcontrol *kcontrol, mask |= (bitmask - 1) << e->shift_r; } - mutex_lock(&codec->mutex); + mutex_lock_nested(&card->dapm_mutex, SND_SOC_DAPM_CLASS_RUNTIME); change = snd_soc_test_bits(widget->codec, e->reg, mask, val); if (change) { @@ -2459,13 +2606,13 @@ int snd_soc_dapm_put_enum_double(struct snd_kcontrol *kcontrol, update.val = val; widget->dapm->update = &update; - snd_soc_dapm_mux_update_power(widget, kcontrol, mux, e); + soc_dapm_mux_update_power(widget, kcontrol, mux, e); widget->dapm->update = NULL; } } - mutex_unlock(&codec->mutex); + mutex_unlock(&card->dapm_mutex); return change; } EXPORT_SYMBOL_GPL(snd_soc_dapm_put_enum_double); @@ -2502,6 +2649,7 @@ int snd_soc_dapm_put_enum_virt(struct snd_kcontrol *kcontrol, struct snd_soc_dapm_widget_list *wlist = snd_kcontrol_chip(kcontrol); struct snd_soc_dapm_widget *widget = wlist->widgets[0]; struct snd_soc_codec *codec = widget->codec; + struct snd_soc_card *card = codec->card; struct soc_enum *e = (struct soc_enum *)kcontrol->private_value; int change; @@ -2511,7 +2659,7 @@ int snd_soc_dapm_put_enum_virt(struct snd_kcontrol *kcontrol, if (ucontrol->value.enumerated.item[0] >= e->max) return -EINVAL; - mutex_lock(&codec->mutex); + mutex_lock_nested(&card->dapm_mutex, SND_SOC_DAPM_CLASS_RUNTIME); change = widget->value != ucontrol->value.enumerated.item[0]; if (change) { @@ -2520,11 +2668,11 @@ int snd_soc_dapm_put_enum_virt(struct snd_kcontrol *kcontrol, widget->value = ucontrol->value.enumerated.item[0]; - snd_soc_dapm_mux_update_power(widget, kcontrol, widget->value, e); + soc_dapm_mux_update_power(widget, kcontrol, widget->value, e); } } - mutex_unlock(&codec->mutex); + mutex_unlock(&card->dapm_mutex); return ret; } EXPORT_SYMBOL_GPL(snd_soc_dapm_put_enum_virt); @@ -2589,6 +2737,7 @@ int snd_soc_dapm_put_value_enum_double(struct snd_kcontrol *kcontrol, struct snd_soc_dapm_widget_list *wlist = snd_kcontrol_chip(kcontrol); struct snd_soc_dapm_widget *widget = wlist->widgets[0]; struct snd_soc_codec *codec = widget->codec; + struct snd_soc_card *card = codec->card; struct soc_enum *e = (struct soc_enum *)kcontrol->private_value; unsigned int val, mux, change; unsigned int mask; @@ -2607,7 +2756,7 @@ int snd_soc_dapm_put_value_enum_double(struct snd_kcontrol *kcontrol, mask |= e->mask << e->shift_r; } - mutex_lock(&codec->mutex); + mutex_lock_nested(&card->dapm_mutex, SND_SOC_DAPM_CLASS_RUNTIME); change = snd_soc_test_bits(widget->codec, e->reg, mask, val); if (change) { @@ -2623,13 +2772,13 @@ int snd_soc_dapm_put_value_enum_double(struct snd_kcontrol *kcontrol, update.val = val; widget->dapm->update = &update; - snd_soc_dapm_mux_update_power(widget, kcontrol, mux, e); + soc_dapm_mux_update_power(widget, kcontrol, mux, e); widget->dapm->update = NULL; } } - mutex_unlock(&codec->mutex); + mutex_unlock(&card->dapm_mutex); return change; } EXPORT_SYMBOL_GPL(snd_soc_dapm_put_value_enum_double); @@ -2666,12 +2815,12 @@ int snd_soc_dapm_get_pin_switch(struct snd_kcontrol *kcontrol, struct snd_soc_card *card = snd_kcontrol_chip(kcontrol); const char *pin = (const char *)kcontrol->private_value; - mutex_lock(&card->mutex); + mutex_lock_nested(&card->dapm_mutex, SND_SOC_DAPM_CLASS_RUNTIME); ucontrol->value.integer.value[0] = snd_soc_dapm_get_pin_status(&card->dapm, pin); - mutex_unlock(&card->mutex); + mutex_unlock(&card->dapm_mutex); return 0; } @@ -2689,17 +2838,16 @@ int snd_soc_dapm_put_pin_switch(struct snd_kcontrol *kcontrol, struct snd_soc_card *card = snd_kcontrol_chip(kcontrol); const char *pin = (const char *)kcontrol->private_value; - mutex_lock(&card->mutex); + mutex_lock_nested(&card->dapm_mutex, SND_SOC_DAPM_CLASS_RUNTIME); if (ucontrol->value.integer.value[0]) snd_soc_dapm_enable_pin(&card->dapm, pin); else snd_soc_dapm_disable_pin(&card->dapm, pin); - snd_soc_dapm_sync(&card->dapm); - - mutex_unlock(&card->mutex); + mutex_unlock(&card->dapm_mutex); + snd_soc_dapm_sync(&card->dapm); return 0; } EXPORT_SYMBOL_GPL(snd_soc_dapm_put_pin_switch); @@ -2717,9 +2865,9 @@ snd_soc_dapm_new_control(struct snd_soc_dapm_context *dapm, switch (w->id) { case snd_soc_dapm_regulator_supply: - w->priv = devm_regulator_get(dapm->dev, w->name); - if (IS_ERR(w->priv)) { - ret = PTR_ERR(w->priv); + w->regulator = devm_regulator_get(dapm->dev, w->name); + if (IS_ERR(w->regulator)) { + ret = PTR_ERR(w->regulator); dev_err(dapm->dev, "Failed to request %s: %d\n", w->name, ret); return NULL; @@ -2771,6 +2919,7 @@ snd_soc_dapm_new_control(struct snd_soc_dapm_context *dapm, case snd_soc_dapm_hp: case snd_soc_dapm_mic: case snd_soc_dapm_line: + case snd_soc_dapm_dai_link: w->power_check = dapm_generic_check_power; break; case snd_soc_dapm_supply: @@ -2816,21 +2965,177 @@ int snd_soc_dapm_new_controls(struct snd_soc_dapm_context *dapm, { struct snd_soc_dapm_widget *w; int i; + int ret = 0; + mutex_lock_nested(&dapm->card->dapm_mutex, SND_SOC_DAPM_CLASS_INIT); for (i = 0; i < num; i++) { w = snd_soc_dapm_new_control(dapm, widget); if (!w) { dev_err(dapm->dev, "ASoC: Failed to create DAPM control %s\n", widget->name); - return -ENOMEM; + ret = -ENOMEM; + break; } widget++; } - return 0; + mutex_unlock(&dapm->card->dapm_mutex); + return ret; } EXPORT_SYMBOL_GPL(snd_soc_dapm_new_controls); +static int snd_soc_dai_link_event(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *kcontrol, int event) +{ + struct snd_soc_dapm_path *source_p, *sink_p; + struct snd_soc_dai *source, *sink; + const struct snd_soc_pcm_stream *config = w->params; + struct snd_pcm_substream substream; + struct snd_pcm_hw_params *params = NULL; + u64 fmt; + int ret; + + BUG_ON(!config); + BUG_ON(list_empty(&w->sources) || list_empty(&w->sinks)); + + /* We only support a single source and sink, pick the first */ + source_p = list_first_entry(&w->sources, struct snd_soc_dapm_path, + list_sink); + sink_p = list_first_entry(&w->sinks, struct snd_soc_dapm_path, + list_source); + + BUG_ON(!source_p || !sink_p); + BUG_ON(!sink_p->source || !source_p->sink); + BUG_ON(!source_p->source || !sink_p->sink); + + source = source_p->source->priv; + sink = sink_p->sink->priv; + + /* Be a little careful as we don't want to overflow the mask array */ + if (config->formats) { + fmt = ffs(config->formats) - 1; + } else { + dev_warn(w->dapm->dev, "Invalid format %llx specified\n", + config->formats); + fmt = 0; + } + + /* Currently very limited parameter selection */ + params = kzalloc(sizeof(*params), GFP_KERNEL); + if (!params) { + ret = -ENOMEM; + goto out; + } + snd_mask_set(hw_param_mask(params, SNDRV_PCM_HW_PARAM_FORMAT), fmt); + + hw_param_interval(params, SNDRV_PCM_HW_PARAM_RATE)->min = + config->rate_min; + hw_param_interval(params, SNDRV_PCM_HW_PARAM_RATE)->max = + config->rate_max; + + hw_param_interval(params, SNDRV_PCM_HW_PARAM_CHANNELS)->min + = config->channels_min; + hw_param_interval(params, SNDRV_PCM_HW_PARAM_CHANNELS)->max + = config->channels_max; + + memset(&substream, 0, sizeof(substream)); + + switch (event) { + case SND_SOC_DAPM_PRE_PMU: + if (source->driver->ops && source->driver->ops->hw_params) { + substream.stream = SNDRV_PCM_STREAM_CAPTURE; + ret = source->driver->ops->hw_params(&substream, + params, source); + if (ret != 0) { + dev_err(source->dev, + "hw_params() failed: %d\n", ret); + goto out; + } + } + + if (sink->driver->ops && sink->driver->ops->hw_params) { + substream.stream = SNDRV_PCM_STREAM_PLAYBACK; + ret = sink->driver->ops->hw_params(&substream, params, + sink); + if (ret != 0) { + dev_err(sink->dev, + "hw_params() failed: %d\n", ret); + goto out; + } + } + break; + + case SND_SOC_DAPM_POST_PMU: + ret = snd_soc_dai_digital_mute(sink, 0); + if (ret != 0 && ret != -ENOTSUPP) + dev_warn(sink->dev, "Failed to unmute: %d\n", ret); + ret = 0; + break; + + case SND_SOC_DAPM_PRE_PMD: + ret = snd_soc_dai_digital_mute(sink, 1); + if (ret != 0 && ret != -ENOTSUPP) + dev_warn(sink->dev, "Failed to mute: %d\n", ret); + ret = 0; + break; + + default: + BUG(); + return -EINVAL; + } + +out: + kfree(params); + return ret; +} + +int snd_soc_dapm_new_pcm(struct snd_soc_card *card, + const struct snd_soc_pcm_stream *params, + struct snd_soc_dapm_widget *source, + struct snd_soc_dapm_widget *sink) +{ + struct snd_soc_dapm_route routes[2]; + struct snd_soc_dapm_widget template; + struct snd_soc_dapm_widget *w; + size_t len; + char *link_name; + + len = strlen(source->name) + strlen(sink->name) + 2; + link_name = devm_kzalloc(card->dev, len, GFP_KERNEL); + if (!link_name) + return -ENOMEM; + snprintf(link_name, len, "%s-%s", source->name, sink->name); + + memset(&template, 0, sizeof(template)); + template.reg = SND_SOC_NOPM; + template.id = snd_soc_dapm_dai_link; + template.name = link_name; + template.event = snd_soc_dai_link_event; + template.event_flags = SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMU | + SND_SOC_DAPM_PRE_PMD; + + dev_dbg(card->dev, "adding %s widget\n", link_name); + + w = snd_soc_dapm_new_control(&card->dapm, &template); + if (!w) { + dev_err(card->dev, "Failed to create %s widget\n", + link_name); + return -ENOMEM; + } + + w->params = params; + + memset(&routes, 0, sizeof(routes)); + + routes[0].source = source->name; + routes[0].sink = link_name; + routes[1].source = link_name; + routes[1].sink = sink->name; + + return snd_soc_dapm_add_routes(&card->dapm, routes, + ARRAY_SIZE(routes)); +} + int snd_soc_dapm_new_dai_widgets(struct snd_soc_dapm_context *dapm, struct snd_soc_dai *dai) { @@ -2934,37 +3239,61 @@ int snd_soc_dapm_link_dai_widgets(struct snd_soc_card *card) return 0; } -static void soc_dapm_stream_event(struct snd_soc_dapm_context *dapm, - int stream, struct snd_soc_dai *dai, - int event) +static void soc_dapm_stream_event(struct snd_soc_pcm_runtime *rtd, int stream, + int event) { - struct snd_soc_dapm_widget *w; - if (stream == SNDRV_PCM_STREAM_PLAYBACK) - w = dai->playback_widget; - else - w = dai->capture_widget; + struct snd_soc_dapm_widget *w_cpu, *w_codec; + struct snd_soc_dai *cpu_dai = rtd->cpu_dai; + struct snd_soc_dai *codec_dai = rtd->codec_dai; - if (!w) - return; + if (stream == SNDRV_PCM_STREAM_PLAYBACK) { + w_cpu = cpu_dai->playback_widget; + w_codec = codec_dai->playback_widget; + } else { + w_cpu = cpu_dai->capture_widget; + w_codec = codec_dai->capture_widget; + } - dapm_mark_dirty(w, "stream event"); + if (w_cpu) { - switch (event) { - case SND_SOC_DAPM_STREAM_START: - w->active = 1; - break; - case SND_SOC_DAPM_STREAM_STOP: - w->active = 0; - break; - case SND_SOC_DAPM_STREAM_SUSPEND: - case SND_SOC_DAPM_STREAM_RESUME: - case SND_SOC_DAPM_STREAM_PAUSE_PUSH: - case SND_SOC_DAPM_STREAM_PAUSE_RELEASE: - break; + dapm_mark_dirty(w_cpu, "stream event"); + + switch (event) { + case SND_SOC_DAPM_STREAM_START: + w_cpu->active = 1; + break; + case SND_SOC_DAPM_STREAM_STOP: + w_cpu->active = 0; + break; + case SND_SOC_DAPM_STREAM_SUSPEND: + case SND_SOC_DAPM_STREAM_RESUME: + case SND_SOC_DAPM_STREAM_PAUSE_PUSH: + case SND_SOC_DAPM_STREAM_PAUSE_RELEASE: + break; + } + } + + if (w_codec) { + + dapm_mark_dirty(w_codec, "stream event"); + + switch (event) { + case SND_SOC_DAPM_STREAM_START: + w_codec->active = 1; + break; + case SND_SOC_DAPM_STREAM_STOP: + w_codec->active = 0; + break; + case SND_SOC_DAPM_STREAM_SUSPEND: + case SND_SOC_DAPM_STREAM_RESUME: + case SND_SOC_DAPM_STREAM_PAUSE_PUSH: + case SND_SOC_DAPM_STREAM_PAUSE_RELEASE: + break; + } } - dapm_power_widgets(dapm, event); + dapm_power_widgets(&rtd->card->dapm, event); } /** @@ -2978,15 +3307,14 @@ static void soc_dapm_stream_event(struct snd_soc_dapm_context *dapm, * * Returns 0 for success else error. */ -int snd_soc_dapm_stream_event(struct snd_soc_pcm_runtime *rtd, int stream, - struct snd_soc_dai *dai, int event) +void snd_soc_dapm_stream_event(struct snd_soc_pcm_runtime *rtd, int stream, + int event) { - struct snd_soc_codec *codec = rtd->codec; + struct snd_soc_card *card = rtd->card; - mutex_lock(&codec->mutex); - soc_dapm_stream_event(&codec->dapm, stream, dai, event); - mutex_unlock(&codec->mutex); - return 0; + mutex_lock_nested(&card->dapm_mutex, SND_SOC_DAPM_CLASS_RUNTIME); + soc_dapm_stream_event(rtd, stream, event); + mutex_unlock(&card->dapm_mutex); } /** diff --git a/sound/soc/soc-jack.c b/sound/soc/soc-jack.c index ee4353f843e..7f8b3b7428b 100644 --- a/sound/soc/soc-jack.c +++ b/sound/soc/soc-jack.c @@ -36,6 +36,7 @@ int snd_soc_jack_new(struct snd_soc_codec *codec, const char *id, int type, struct snd_soc_jack *jack) { + mutex_init(&jack->mutex); jack->codec = codec; INIT_LIST_HEAD(&jack->pins); INIT_LIST_HEAD(&jack->jack_zones); @@ -75,7 +76,7 @@ void snd_soc_jack_report(struct snd_soc_jack *jack, int status, int mask) codec = jack->codec; dapm = &codec->dapm; - mutex_lock(&codec->mutex); + mutex_lock(&jack->mutex); oldstatus = jack->status; @@ -109,7 +110,7 @@ void snd_soc_jack_report(struct snd_soc_jack *jack, int status, int mask) snd_jack_report(jack->jack, jack->status); out: - mutex_unlock(&codec->mutex); + mutex_unlock(&jack->mutex); } EXPORT_SYMBOL_GPL(snd_soc_jack_report); diff --git a/sound/soc/soc-pcm.c b/sound/soc/soc-pcm.c index 0ad8dcacd2f..bedd1717a37 100644 --- a/sound/soc/soc-pcm.c +++ b/sound/soc/soc-pcm.c @@ -22,12 +22,38 @@ #include <linux/pm_runtime.h> #include <linux/slab.h> #include <linux/workqueue.h> +#include <linux/export.h> +#include <linux/debugfs.h> #include <sound/core.h> #include <sound/pcm.h> #include <sound/pcm_params.h> #include <sound/soc.h> +#include <sound/soc-dpcm.h> #include <sound/initval.h> +#define DPCM_MAX_BE_USERS 8 + +/* DPCM stream event, send event to FE and all active BEs. */ +static int dpcm_dapm_stream_event(struct snd_soc_pcm_runtime *fe, int dir, + int event) +{ + struct snd_soc_dpcm *dpcm; + + list_for_each_entry(dpcm, &fe->dpcm[dir].be_clients, list_be) { + + struct snd_soc_pcm_runtime *be = dpcm->be; + + dev_dbg(be->dev, "pm: BE %s event %d dir %d\n", + be->dai_link->name, event, dir); + + snd_soc_dapm_stream_event(be, dir, event); + } + + snd_soc_dapm_stream_event(fe, dir, event); + + return 0; +} + static int soc_pcm_apply_symmetry(struct snd_pcm_substream *substream, struct snd_soc_dai *soc_dai) { @@ -156,6 +182,10 @@ static int soc_pcm_open(struct snd_pcm_substream *substream) } } + /* Dynamic PCM DAI links compat checks use dynamic capabilities */ + if (rtd->dai_link->dynamic || rtd->dai_link->no_pcm) + goto dynamic; + /* Check that the codec and cpu DAIs are compatible */ if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { runtime->hw.rate_min = @@ -248,6 +278,7 @@ static int soc_pcm_open(struct snd_pcm_substream *substream) pr_debug("asoc: min rate %d max rate %d\n", runtime->hw.rate_min, runtime->hw.rate_max); +dynamic: if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { cpu_dai->playback_active++; codec_dai->playback_active++; @@ -308,7 +339,7 @@ static void close_delayed_work(struct work_struct *work) if (codec_dai->pop_wait == 1) { codec_dai->pop_wait = 0; snd_soc_dapm_stream_event(rtd, SNDRV_PCM_STREAM_PLAYBACK, - codec_dai, SND_SOC_DAPM_STREAM_STOP); + SND_SOC_DAPM_STREAM_STOP); } mutex_unlock(&rtd->pcm_mutex); @@ -373,7 +404,6 @@ static int soc_pcm_close(struct snd_pcm_substream *substream) /* powered down playback stream now */ snd_soc_dapm_stream_event(rtd, SNDRV_PCM_STREAM_PLAYBACK, - codec_dai, SND_SOC_DAPM_STREAM_STOP); } else { /* start delayed pop wq here for playback streams */ @@ -384,7 +414,7 @@ static int soc_pcm_close(struct snd_pcm_substream *substream) } else { /* capture streams can be powered down now */ snd_soc_dapm_stream_event(rtd, SNDRV_PCM_STREAM_CAPTURE, - codec_dai, SND_SOC_DAPM_STREAM_STOP); + SND_SOC_DAPM_STREAM_STOP); } mutex_unlock(&rtd->pcm_mutex); @@ -453,8 +483,8 @@ static int soc_pcm_prepare(struct snd_pcm_substream *substream) cancel_delayed_work(&rtd->delayed_work); } - snd_soc_dapm_stream_event(rtd, substream->stream, codec_dai, - SND_SOC_DAPM_STREAM_START); + snd_soc_dapm_stream_event(rtd, substream->stream, + SND_SOC_DAPM_STREAM_START); snd_soc_dai_digital_mute(codec_dai, 0); @@ -602,6 +632,34 @@ static int soc_pcm_trigger(struct snd_pcm_substream *substream, int cmd) return 0; } +static int soc_pcm_bespoke_trigger(struct snd_pcm_substream *substream, + int cmd) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_soc_platform *platform = rtd->platform; + struct snd_soc_dai *cpu_dai = rtd->cpu_dai; + struct snd_soc_dai *codec_dai = rtd->codec_dai; + int ret; + + if (codec_dai->driver->ops->bespoke_trigger) { + ret = codec_dai->driver->ops->bespoke_trigger(substream, cmd, codec_dai); + if (ret < 0) + return ret; + } + + if (platform->driver->bespoke_trigger) { + ret = platform->driver->bespoke_trigger(substream, cmd); + if (ret < 0) + return ret; + } + + if (cpu_dai->driver->ops->bespoke_trigger) { + ret = cpu_dai->driver->ops->bespoke_trigger(substream, cmd, cpu_dai); + if (ret < 0) + return ret; + } + return 0; +} /* * soc level wrapper for pointer callback * If cpu_dai, codec_dai, platform driver has the delay callback, than @@ -634,6 +692,1308 @@ static snd_pcm_uframes_t soc_pcm_pointer(struct snd_pcm_substream *substream) return offset; } +/* connect a FE and BE */ +static int dpcm_be_connect(struct snd_soc_pcm_runtime *fe, + struct snd_soc_pcm_runtime *be, int stream) +{ + struct snd_soc_dpcm *dpcm; + + /* only add new dpcms */ + list_for_each_entry(dpcm, &fe->dpcm[stream].be_clients, list_be) { + if (dpcm->be == be && dpcm->fe == fe) + return 0; + } + + dpcm = kzalloc(sizeof(struct snd_soc_dpcm), GFP_KERNEL); + if (!dpcm) + return -ENOMEM; + + dpcm->be = be; + dpcm->fe = fe; + be->dpcm[stream].runtime = fe->dpcm[stream].runtime; + dpcm->state = SND_SOC_DPCM_LINK_STATE_NEW; + list_add(&dpcm->list_be, &fe->dpcm[stream].be_clients); + list_add(&dpcm->list_fe, &be->dpcm[stream].fe_clients); + + dev_dbg(fe->dev, " connected new DPCM %s path %s %s %s\n", + stream ? "capture" : "playback", fe->dai_link->name, + stream ? "<-" : "->", be->dai_link->name); + +#ifdef CONFIG_DEBUG_FS + dpcm->debugfs_state = debugfs_create_u32(be->dai_link->name, 0644, + fe->debugfs_dpcm_root, &dpcm->state); +#endif + return 1; +} + +/* reparent a BE onto another FE */ +static void dpcm_be_reparent(struct snd_soc_pcm_runtime *fe, + struct snd_soc_pcm_runtime *be, int stream) +{ + struct snd_soc_dpcm *dpcm; + struct snd_pcm_substream *fe_substream, *be_substream; + + /* reparent if BE is connected to other FEs */ + if (!be->dpcm[stream].users) + return; + + be_substream = snd_soc_dpcm_get_substream(be, stream); + + list_for_each_entry(dpcm, &be->dpcm[stream].fe_clients, list_fe) { + if (dpcm->fe == fe) + continue; + + dev_dbg(fe->dev, " reparent %s path %s %s %s\n", + stream ? "capture" : "playback", + dpcm->fe->dai_link->name, + stream ? "<-" : "->", dpcm->be->dai_link->name); + + fe_substream = snd_soc_dpcm_get_substream(dpcm->fe, stream); + be_substream->runtime = fe_substream->runtime; + break; + } +} + +/* disconnect a BE and FE */ +static void dpcm_be_disconnect(struct snd_soc_pcm_runtime *fe, int stream) +{ + struct snd_soc_dpcm *dpcm, *d; + + list_for_each_entry_safe(dpcm, d, &fe->dpcm[stream].be_clients, list_be) { + dev_dbg(fe->dev, "BE %s disconnect check for %s\n", + stream ? "capture" : "playback", + dpcm->be->dai_link->name); + + if (dpcm->state != SND_SOC_DPCM_LINK_STATE_FREE) + continue; + + dev_dbg(fe->dev, " freed DSP %s path %s %s %s\n", + stream ? "capture" : "playback", fe->dai_link->name, + stream ? "<-" : "->", dpcm->be->dai_link->name); + + /* BEs still alive need new FE */ + dpcm_be_reparent(fe, dpcm->be, stream); + +#ifdef CONFIG_DEBUG_FS + debugfs_remove(dpcm->debugfs_state); +#endif + list_del(&dpcm->list_be); + list_del(&dpcm->list_fe); + kfree(dpcm); + } +} + +/* get BE for DAI widget and stream */ +static struct snd_soc_pcm_runtime *dpcm_get_be(struct snd_soc_card *card, + struct snd_soc_dapm_widget *widget, int stream) +{ + struct snd_soc_pcm_runtime *be; + int i; + + if (stream == SNDRV_PCM_STREAM_PLAYBACK) { + for (i = 0; i < card->num_links; i++) { + be = &card->rtd[i]; + + if (be->cpu_dai->playback_widget == widget || + be->codec_dai->playback_widget == widget) + return be; + } + } else { + + for (i = 0; i < card->num_links; i++) { + be = &card->rtd[i]; + + if (be->cpu_dai->capture_widget == widget || + be->codec_dai->capture_widget == widget) + return be; + } + } + + dev_err(card->dev, "can't get %s BE for %s\n", + stream ? "capture" : "playback", widget->name); + return NULL; +} + +static inline struct snd_soc_dapm_widget * + rtd_get_cpu_widget(struct snd_soc_pcm_runtime *rtd, int stream) +{ + if (stream == SNDRV_PCM_STREAM_PLAYBACK) + return rtd->cpu_dai->playback_widget; + else + return rtd->cpu_dai->capture_widget; +} + +static inline struct snd_soc_dapm_widget * + rtd_get_codec_widget(struct snd_soc_pcm_runtime *rtd, int stream) +{ + if (stream == SNDRV_PCM_STREAM_PLAYBACK) + return rtd->codec_dai->playback_widget; + else + return rtd->codec_dai->capture_widget; +} + +static int widget_in_list(struct snd_soc_dapm_widget_list *list, + struct snd_soc_dapm_widget *widget) +{ + int i; + + for (i = 0; i < list->num_widgets; i++) { + if (widget == list->widgets[i]) + return 1; + } + + return 0; +} + +static int dpcm_path_get(struct snd_soc_pcm_runtime *fe, + int stream, struct snd_soc_dapm_widget_list **list_) +{ + struct snd_soc_dai *cpu_dai = fe->cpu_dai; + struct snd_soc_dapm_widget_list *list; + int paths; + + list = kzalloc(sizeof(struct snd_soc_dapm_widget_list) + + sizeof(struct snd_soc_dapm_widget *), GFP_KERNEL); + if (list == NULL) + return -ENOMEM; + + /* get number of valid DAI paths and their widgets */ + paths = snd_soc_dapm_dai_get_connected_widgets(cpu_dai, stream, &list); + + dev_dbg(fe->dev, "found %d audio %s paths\n", paths, + stream ? "capture" : "playback"); + + *list_ = list; + return paths; +} + +static inline void dpcm_path_put(struct snd_soc_dapm_widget_list **list) +{ + kfree(*list); +} + +static int dpcm_prune_paths(struct snd_soc_pcm_runtime *fe, int stream, + struct snd_soc_dapm_widget_list **list_) +{ + struct snd_soc_dpcm *dpcm; + struct snd_soc_dapm_widget_list *list = *list_; + struct snd_soc_dapm_widget *widget; + int prune = 0; + + /* Destroy any old FE <--> BE connections */ + list_for_each_entry(dpcm, &fe->dpcm[stream].be_clients, list_be) { + + /* is there a valid CPU DAI widget for this BE */ + widget = rtd_get_cpu_widget(dpcm->be, stream); + + /* prune the BE if it's no longer in our active list */ + if (widget && widget_in_list(list, widget)) + continue; + + /* is there a valid CODEC DAI widget for this BE */ + widget = rtd_get_codec_widget(dpcm->be, stream); + + /* prune the BE if it's no longer in our active list */ + if (widget && widget_in_list(list, widget)) + continue; + + dev_dbg(fe->dev, "pruning %s BE %s for %s\n", + stream ? "capture" : "playback", + dpcm->be->dai_link->name, fe->dai_link->name); + dpcm->state = SND_SOC_DPCM_LINK_STATE_FREE; + dpcm->be->dpcm[stream].runtime_update = SND_SOC_DPCM_UPDATE_BE; + prune++; + } + + dev_dbg(fe->dev, "found %d old BE paths for pruning\n", prune); + return prune; +} + +static int dpcm_add_paths(struct snd_soc_pcm_runtime *fe, int stream, + struct snd_soc_dapm_widget_list **list_) +{ + struct snd_soc_card *card = fe->card; + struct snd_soc_dapm_widget_list *list = *list_; + struct snd_soc_pcm_runtime *be; + int i, new = 0, err; + + /* Create any new FE <--> BE connections */ + for (i = 0; i < list->num_widgets; i++) { + + if (list->widgets[i]->id != snd_soc_dapm_dai) + continue; + + /* is there a valid BE rtd for this widget */ + be = dpcm_get_be(card, list->widgets[i], stream); + if (!be) { + dev_err(fe->dev, "no BE found for %s\n", + list->widgets[i]->name); + continue; + } + + /* make sure BE is a real BE */ + if (!be->dai_link->no_pcm) + continue; + + /* don't connect if FE is not running */ + if (!fe->dpcm[stream].runtime) + continue; + + /* newly connected FE and BE */ + err = dpcm_be_connect(fe, be, stream); + if (err < 0) { + dev_err(fe->dev, "can't connect %s\n", + list->widgets[i]->name); + break; + } else if (err == 0) /* already connected */ + continue; + + /* new */ + be->dpcm[stream].runtime_update = SND_SOC_DPCM_UPDATE_BE; + new++; + } + + dev_dbg(fe->dev, "found %d new BE paths\n", new); + return new; +} + +/* + * Find the corresponding BE DAIs that source or sink audio to this + * FE substream. + */ +static int dpcm_process_paths(struct snd_soc_pcm_runtime *fe, + int stream, struct snd_soc_dapm_widget_list **list, int new) +{ + if (new) + return dpcm_add_paths(fe, stream, list); + else + return dpcm_prune_paths(fe, stream, list); +} + +static void dpcm_clear_pending_state(struct snd_soc_pcm_runtime *fe, int stream) +{ + struct snd_soc_dpcm *dpcm; + + list_for_each_entry(dpcm, &fe->dpcm[stream].be_clients, list_be) + dpcm->be->dpcm[stream].runtime_update = + SND_SOC_DPCM_UPDATE_NO; +} + +static void dpcm_be_dai_startup_unwind(struct snd_soc_pcm_runtime *fe, + int stream) +{ + struct snd_soc_dpcm *dpcm; + + /* disable any enabled and non active backends */ + list_for_each_entry(dpcm, &fe->dpcm[stream].be_clients, list_be) { + + struct snd_soc_pcm_runtime *be = dpcm->be; + struct snd_pcm_substream *be_substream = + snd_soc_dpcm_get_substream(be, stream); + + if (be->dpcm[stream].users == 0) + dev_err(be->dev, "no users %s at close - state %d\n", + stream ? "capture" : "playback", + be->dpcm[stream].state); + + if (--be->dpcm[stream].users != 0) + continue; + + if (be->dpcm[stream].state != SND_SOC_DPCM_STATE_OPEN) + continue; + + soc_pcm_close(be_substream); + be_substream->runtime = NULL; + be->dpcm[stream].state = SND_SOC_DPCM_STATE_CLOSE; + } +} + +static int dpcm_be_dai_startup(struct snd_soc_pcm_runtime *fe, int stream) +{ + struct snd_soc_dpcm *dpcm; + int err, count = 0; + + /* only startup BE DAIs that are either sinks or sources to this FE DAI */ + list_for_each_entry(dpcm, &fe->dpcm[stream].be_clients, list_be) { + + struct snd_soc_pcm_runtime *be = dpcm->be; + struct snd_pcm_substream *be_substream = + snd_soc_dpcm_get_substream(be, stream); + + /* is this op for this BE ? */ + if (!snd_soc_dpcm_be_can_update(fe, be, stream)) + continue; + + /* first time the dpcm is open ? */ + if (be->dpcm[stream].users == DPCM_MAX_BE_USERS) + dev_err(be->dev, "too many users %s at open %d\n", + stream ? "capture" : "playback", + be->dpcm[stream].state); + + if (be->dpcm[stream].users++ != 0) + continue; + + if ((be->dpcm[stream].state != SND_SOC_DPCM_STATE_NEW) && + (be->dpcm[stream].state != SND_SOC_DPCM_STATE_CLOSE)) + continue; + + dev_dbg(be->dev, "dpcm: open BE %s\n", be->dai_link->name); + + be_substream->runtime = be->dpcm[stream].runtime; + err = soc_pcm_open(be_substream); + if (err < 0) { + dev_err(be->dev, "BE open failed %d\n", err); + be->dpcm[stream].users--; + if (be->dpcm[stream].users < 0) + dev_err(be->dev, "no users %s at unwind %d\n", + stream ? "capture" : "playback", + be->dpcm[stream].state); + + be->dpcm[stream].state = SND_SOC_DPCM_STATE_CLOSE; + goto unwind; + } + + be->dpcm[stream].state = SND_SOC_DPCM_STATE_OPEN; + count++; + } + + return count; + +unwind: + /* disable any enabled and non active backends */ + list_for_each_entry_continue_reverse(dpcm, &fe->dpcm[stream].be_clients, list_be) { + struct snd_soc_pcm_runtime *be = dpcm->be; + struct snd_pcm_substream *be_substream = + snd_soc_dpcm_get_substream(be, stream); + + if (!snd_soc_dpcm_be_can_update(fe, be, stream)) + continue; + + if (be->dpcm[stream].users == 0) + dev_err(be->dev, "no users %s at close %d\n", + stream ? "capture" : "playback", + be->dpcm[stream].state); + + if (--be->dpcm[stream].users != 0) + continue; + + if (be->dpcm[stream].state != SND_SOC_DPCM_STATE_OPEN) + continue; + + soc_pcm_close(be_substream); + be_substream->runtime = NULL; + be->dpcm[stream].state = SND_SOC_DPCM_STATE_CLOSE; + } + + return err; +} + +static void dpcm_set_fe_runtime(struct snd_pcm_substream *substream) +{ + struct snd_pcm_runtime *runtime = substream->runtime; + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_soc_dai *cpu_dai = rtd->cpu_dai; + struct snd_soc_dai_driver *cpu_dai_drv = cpu_dai->driver; + + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { + runtime->hw.rate_min = cpu_dai_drv->playback.rate_min; + runtime->hw.rate_max = cpu_dai_drv->playback.rate_max; + runtime->hw.channels_min = cpu_dai_drv->playback.channels_min; + runtime->hw.channels_max = cpu_dai_drv->playback.channels_max; + runtime->hw.formats &= cpu_dai_drv->playback.formats; + runtime->hw.rates = cpu_dai_drv->playback.rates; + } else { + runtime->hw.rate_min = cpu_dai_drv->capture.rate_min; + runtime->hw.rate_max = cpu_dai_drv->capture.rate_max; + runtime->hw.channels_min = cpu_dai_drv->capture.channels_min; + runtime->hw.channels_max = cpu_dai_drv->capture.channels_max; + runtime->hw.formats &= cpu_dai_drv->capture.formats; + runtime->hw.rates = cpu_dai_drv->capture.rates; + } +} + +static int dpcm_fe_dai_startup(struct snd_pcm_substream *fe_substream) +{ + struct snd_soc_pcm_runtime *fe = fe_substream->private_data; + struct snd_pcm_runtime *runtime = fe_substream->runtime; + int stream = fe_substream->stream, ret = 0; + + fe->dpcm[stream].runtime_update = SND_SOC_DPCM_UPDATE_FE; + + ret = dpcm_be_dai_startup(fe, fe_substream->stream); + if (ret < 0) { + dev_err(fe->dev,"dpcm: failed to start some BEs %d\n", ret); + goto be_err; + } + + dev_dbg(fe->dev, "dpcm: open FE %s\n", fe->dai_link->name); + + /* start the DAI frontend */ + ret = soc_pcm_open(fe_substream); + if (ret < 0) { + dev_err(fe->dev,"dpcm: failed to start FE %d\n", ret); + goto unwind; + } + + fe->dpcm[stream].state = SND_SOC_DPCM_STATE_OPEN; + + dpcm_set_fe_runtime(fe_substream); + snd_pcm_limit_hw_rates(runtime); + + fe->dpcm[stream].runtime_update = SND_SOC_DPCM_UPDATE_NO; + return 0; + +unwind: + dpcm_be_dai_startup_unwind(fe, fe_substream->stream); +be_err: + fe->dpcm[stream].runtime_update = SND_SOC_DPCM_UPDATE_NO; + return ret; +} + +static int dpcm_be_dai_shutdown(struct snd_soc_pcm_runtime *fe, int stream) +{ + struct snd_soc_dpcm *dpcm; + + /* only shutdown BEs that are either sinks or sources to this FE DAI */ + list_for_each_entry(dpcm, &fe->dpcm[stream].be_clients, list_be) { + + struct snd_soc_pcm_runtime *be = dpcm->be; + struct snd_pcm_substream *be_substream = + snd_soc_dpcm_get_substream(be, stream); + + /* is this op for this BE ? */ + if (!snd_soc_dpcm_be_can_update(fe, be, stream)) + continue; + + if (be->dpcm[stream].users == 0) + dev_err(be->dev, "no users %s at close - state %d\n", + stream ? "capture" : "playback", + be->dpcm[stream].state); + + if (--be->dpcm[stream].users != 0) + continue; + + if ((be->dpcm[stream].state != SND_SOC_DPCM_STATE_HW_FREE) && + (be->dpcm[stream].state != SND_SOC_DPCM_STATE_OPEN)) + continue; + + dev_dbg(be->dev, "dpcm: close BE %s\n", + dpcm->fe->dai_link->name); + + soc_pcm_close(be_substream); + be_substream->runtime = NULL; + + be->dpcm[stream].state = SND_SOC_DPCM_STATE_CLOSE; + } + return 0; +} + +static int dpcm_fe_dai_shutdown(struct snd_pcm_substream *substream) +{ + struct snd_soc_pcm_runtime *fe = substream->private_data; + int stream = substream->stream; + + fe->dpcm[stream].runtime_update = SND_SOC_DPCM_UPDATE_FE; + + /* shutdown the BEs */ + dpcm_be_dai_shutdown(fe, substream->stream); + + dev_dbg(fe->dev, "dpcm: close FE %s\n", fe->dai_link->name); + + /* now shutdown the frontend */ + soc_pcm_close(substream); + + /* run the stream event for each BE */ + dpcm_dapm_stream_event(fe, stream, SND_SOC_DAPM_STREAM_STOP); + + fe->dpcm[stream].state = SND_SOC_DPCM_STATE_CLOSE; + fe->dpcm[stream].runtime_update = SND_SOC_DPCM_UPDATE_NO; + return 0; +} + +static int dpcm_be_dai_hw_free(struct snd_soc_pcm_runtime *fe, int stream) +{ + struct snd_soc_dpcm *dpcm; + + /* only hw_params backends that are either sinks or sources + * to this frontend DAI */ + list_for_each_entry(dpcm, &fe->dpcm[stream].be_clients, list_be) { + + struct snd_soc_pcm_runtime *be = dpcm->be; + struct snd_pcm_substream *be_substream = + snd_soc_dpcm_get_substream(be, stream); + + /* is this op for this BE ? */ + if (!snd_soc_dpcm_be_can_update(fe, be, stream)) + continue; + + /* only free hw when no longer used - check all FEs */ + if (!snd_soc_dpcm_can_be_free_stop(fe, be, stream)) + continue; + + if ((be->dpcm[stream].state != SND_SOC_DPCM_STATE_HW_PARAMS) && + (be->dpcm[stream].state != SND_SOC_DPCM_STATE_PREPARE) && + (be->dpcm[stream].state != SND_SOC_DPCM_STATE_HW_FREE) && + (be->dpcm[stream].state != SND_SOC_DPCM_STATE_STOP)) + continue; + + dev_dbg(be->dev, "dpcm: hw_free BE %s\n", + dpcm->fe->dai_link->name); + + soc_pcm_hw_free(be_substream); + + be->dpcm[stream].state = SND_SOC_DPCM_STATE_HW_FREE; + } + + return 0; +} + +static int dpcm_fe_dai_hw_free(struct snd_pcm_substream *substream) +{ + struct snd_soc_pcm_runtime *fe = substream->private_data; + int err, stream = substream->stream; + + mutex_lock_nested(&fe->card->mutex, SND_SOC_CARD_CLASS_RUNTIME); + fe->dpcm[stream].runtime_update = SND_SOC_DPCM_UPDATE_FE; + + dev_dbg(fe->dev, "dpcm: hw_free FE %s\n", fe->dai_link->name); + + /* call hw_free on the frontend */ + err = soc_pcm_hw_free(substream); + if (err < 0) + dev_err(fe->dev,"dpcm: hw_free FE %s failed\n", + fe->dai_link->name); + + /* only hw_params backends that are either sinks or sources + * to this frontend DAI */ + err = dpcm_be_dai_hw_free(fe, stream); + + fe->dpcm[stream].state = SND_SOC_DPCM_STATE_HW_FREE; + fe->dpcm[stream].runtime_update = SND_SOC_DPCM_UPDATE_NO; + + mutex_unlock(&fe->card->mutex); + return 0; +} + +static int dpcm_be_dai_hw_params(struct snd_soc_pcm_runtime *fe, int stream) +{ + struct snd_soc_dpcm *dpcm; + int ret; + + list_for_each_entry(dpcm, &fe->dpcm[stream].be_clients, list_be) { + + struct snd_soc_pcm_runtime *be = dpcm->be; + struct snd_pcm_substream *be_substream = + snd_soc_dpcm_get_substream(be, stream); + + /* is this op for this BE ? */ + if (!snd_soc_dpcm_be_can_update(fe, be, stream)) + continue; + + /* only allow hw_params() if no connected FEs are running */ + if (!snd_soc_dpcm_can_be_params(fe, be, stream)) + continue; + + if ((be->dpcm[stream].state != SND_SOC_DPCM_STATE_OPEN) && + (be->dpcm[stream].state != SND_SOC_DPCM_STATE_HW_PARAMS) && + (be->dpcm[stream].state != SND_SOC_DPCM_STATE_HW_FREE)) + continue; + + dev_dbg(be->dev, "dpcm: hw_params BE %s\n", + dpcm->fe->dai_link->name); + + /* copy params for each dpcm */ + memcpy(&dpcm->hw_params, &fe->dpcm[stream].hw_params, + sizeof(struct snd_pcm_hw_params)); + + /* perform any hw_params fixups */ + if (be->dai_link->be_hw_params_fixup) { + ret = be->dai_link->be_hw_params_fixup(be, + &dpcm->hw_params); + if (ret < 0) { + dev_err(be->dev, + "dpcm: hw_params BE fixup failed %d\n", + ret); + goto unwind; + } + } + + ret = soc_pcm_hw_params(be_substream, &dpcm->hw_params); + if (ret < 0) { + dev_err(dpcm->be->dev, + "dpcm: hw_params BE failed %d\n", ret); + goto unwind; + } + + be->dpcm[stream].state = SND_SOC_DPCM_STATE_HW_PARAMS; + } + return 0; + +unwind: + /* disable any enabled and non active backends */ + list_for_each_entry_continue_reverse(dpcm, &fe->dpcm[stream].be_clients, list_be) { + struct snd_soc_pcm_runtime *be = dpcm->be; + struct snd_pcm_substream *be_substream = + snd_soc_dpcm_get_substream(be, stream); + + if (!snd_soc_dpcm_be_can_update(fe, be, stream)) + continue; + + /* only allow hw_free() if no connected FEs are running */ + if (!snd_soc_dpcm_can_be_free_stop(fe, be, stream)) + continue; + + if ((be->dpcm[stream].state != SND_SOC_DPCM_STATE_OPEN) && + (be->dpcm[stream].state != SND_SOC_DPCM_STATE_HW_PARAMS) && + (be->dpcm[stream].state != SND_SOC_DPCM_STATE_HW_FREE) && + (be->dpcm[stream].state != SND_SOC_DPCM_STATE_STOP)) + continue; + + soc_pcm_hw_free(be_substream); + } + + return ret; +} + +static int dpcm_fe_dai_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params) +{ + struct snd_soc_pcm_runtime *fe = substream->private_data; + int ret, stream = substream->stream; + + mutex_lock_nested(&fe->card->mutex, SND_SOC_CARD_CLASS_RUNTIME); + fe->dpcm[stream].runtime_update = SND_SOC_DPCM_UPDATE_FE; + + memcpy(&fe->dpcm[substream->stream].hw_params, params, + sizeof(struct snd_pcm_hw_params)); + ret = dpcm_be_dai_hw_params(fe, substream->stream); + if (ret < 0) { + dev_err(fe->dev,"dpcm: hw_params BE failed %d\n", ret); + goto out; + } + + dev_dbg(fe->dev, "dpcm: hw_params FE %s rate %d chan %x fmt %d\n", + fe->dai_link->name, params_rate(params), + params_channels(params), params_format(params)); + + /* call hw_params on the frontend */ + ret = soc_pcm_hw_params(substream, params); + if (ret < 0) { + dev_err(fe->dev,"dpcm: hw_params FE failed %d\n", ret); + dpcm_be_dai_hw_free(fe, stream); + } else + fe->dpcm[stream].state = SND_SOC_DPCM_STATE_HW_PARAMS; + +out: + fe->dpcm[stream].runtime_update = SND_SOC_DPCM_UPDATE_NO; + mutex_unlock(&fe->card->mutex); + return ret; +} + +static int dpcm_do_trigger(struct snd_soc_dpcm *dpcm, + struct snd_pcm_substream *substream, int cmd) +{ + int ret; + + dev_dbg(dpcm->be->dev, "dpcm: trigger BE %s cmd %d\n", + dpcm->fe->dai_link->name, cmd); + + ret = soc_pcm_trigger(substream, cmd); + if (ret < 0) + dev_err(dpcm->be->dev,"dpcm: trigger BE failed %d\n", ret); + + return ret; +} + +static int dpcm_be_dai_trigger(struct snd_soc_pcm_runtime *fe, int stream, + int cmd) +{ + struct snd_soc_dpcm *dpcm; + int ret = 0; + + list_for_each_entry(dpcm, &fe->dpcm[stream].be_clients, list_be) { + + struct snd_soc_pcm_runtime *be = dpcm->be; + struct snd_pcm_substream *be_substream = + snd_soc_dpcm_get_substream(be, stream); + + /* is this op for this BE ? */ + if (!snd_soc_dpcm_be_can_update(fe, be, stream)) + continue; + + switch (cmd) { + case SNDRV_PCM_TRIGGER_START: + if ((be->dpcm[stream].state != SND_SOC_DPCM_STATE_PREPARE) && + (be->dpcm[stream].state != SND_SOC_DPCM_STATE_STOP)) + continue; + + ret = dpcm_do_trigger(dpcm, be_substream, cmd); + if (ret) + return ret; + + be->dpcm[stream].state = SND_SOC_DPCM_STATE_START; + break; + case SNDRV_PCM_TRIGGER_RESUME: + if ((be->dpcm[stream].state != SND_SOC_DPCM_STATE_SUSPEND)) + continue; + + ret = dpcm_do_trigger(dpcm, be_substream, cmd); + if (ret) + return ret; + + be->dpcm[stream].state = SND_SOC_DPCM_STATE_START; + break; + case SNDRV_PCM_TRIGGER_PAUSE_RELEASE: + if ((be->dpcm[stream].state != SND_SOC_DPCM_STATE_PAUSED)) + continue; + + ret = dpcm_do_trigger(dpcm, be_substream, cmd); + if (ret) + return ret; + + be->dpcm[stream].state = SND_SOC_DPCM_STATE_START; + break; + case SNDRV_PCM_TRIGGER_STOP: + if (be->dpcm[stream].state != SND_SOC_DPCM_STATE_START) + continue; + + if (!snd_soc_dpcm_can_be_free_stop(fe, be, stream)) + continue; + + ret = dpcm_do_trigger(dpcm, be_substream, cmd); + if (ret) + return ret; + + be->dpcm[stream].state = SND_SOC_DPCM_STATE_STOP; + break; + case SNDRV_PCM_TRIGGER_SUSPEND: + if (be->dpcm[stream].state != SND_SOC_DPCM_STATE_STOP) + continue; + + if (!snd_soc_dpcm_can_be_free_stop(fe, be, stream)) + continue; + + ret = dpcm_do_trigger(dpcm, be_substream, cmd); + if (ret) + return ret; + + be->dpcm[stream].state = SND_SOC_DPCM_STATE_SUSPEND; + break; + case SNDRV_PCM_TRIGGER_PAUSE_PUSH: + if (be->dpcm[stream].state != SND_SOC_DPCM_STATE_START) + continue; + + if (!snd_soc_dpcm_can_be_free_stop(fe, be, stream)) + continue; + + ret = dpcm_do_trigger(dpcm, be_substream, cmd); + if (ret) + return ret; + + be->dpcm[stream].state = SND_SOC_DPCM_STATE_PAUSED; + break; + } + } + + return ret; +} +EXPORT_SYMBOL_GPL(dpcm_be_dai_trigger); + +static int dpcm_fe_dai_trigger(struct snd_pcm_substream *substream, int cmd) +{ + struct snd_soc_pcm_runtime *fe = substream->private_data; + int stream = substream->stream, ret; + enum snd_soc_dpcm_trigger trigger = fe->dai_link->trigger[stream]; + + fe->dpcm[stream].runtime_update = SND_SOC_DPCM_UPDATE_FE; + + switch (trigger) { + case SND_SOC_DPCM_TRIGGER_PRE: + /* call trigger on the frontend before the backend. */ + + dev_dbg(fe->dev, "dpcm: pre trigger FE %s cmd %d\n", + fe->dai_link->name, cmd); + + ret = soc_pcm_trigger(substream, cmd); + if (ret < 0) { + dev_err(fe->dev,"dpcm: trigger FE failed %d\n", ret); + goto out; + } + + ret = dpcm_be_dai_trigger(fe, substream->stream, cmd); + break; + case SND_SOC_DPCM_TRIGGER_POST: + /* call trigger on the frontend after the backend. */ + + ret = dpcm_be_dai_trigger(fe, substream->stream, cmd); + if (ret < 0) { + dev_err(fe->dev,"dpcm: trigger FE failed %d\n", ret); + goto out; + } + + dev_dbg(fe->dev, "dpcm: post trigger FE %s cmd %d\n", + fe->dai_link->name, cmd); + + ret = soc_pcm_trigger(substream, cmd); + break; + case SND_SOC_DPCM_TRIGGER_BESPOKE: + /* bespoke trigger() - handles both FE and BEs */ + + dev_dbg(fe->dev, "dpcm: bespoke trigger FE %s cmd %d\n", + fe->dai_link->name, cmd); + + ret = soc_pcm_bespoke_trigger(substream, cmd); + if (ret < 0) { + dev_err(fe->dev,"dpcm: trigger FE failed %d\n", ret); + goto out; + } + break; + default: + dev_err(fe->dev, "dpcm: invalid trigger cmd %d for %s\n", cmd, + fe->dai_link->name); + ret = -EINVAL; + goto out; + } + + switch (cmd) { + case SNDRV_PCM_TRIGGER_START: + case SNDRV_PCM_TRIGGER_RESUME: + case SNDRV_PCM_TRIGGER_PAUSE_RELEASE: + fe->dpcm[stream].state = SND_SOC_DPCM_STATE_START; + break; + case SNDRV_PCM_TRIGGER_STOP: + case SNDRV_PCM_TRIGGER_SUSPEND: + case SNDRV_PCM_TRIGGER_PAUSE_PUSH: + fe->dpcm[stream].state = SND_SOC_DPCM_STATE_STOP; + break; + } + +out: + fe->dpcm[stream].runtime_update = SND_SOC_DPCM_UPDATE_NO; + return ret; +} + +static int dpcm_be_dai_prepare(struct snd_soc_pcm_runtime *fe, int stream) +{ + struct snd_soc_dpcm *dpcm; + int ret = 0; + + list_for_each_entry(dpcm, &fe->dpcm[stream].be_clients, list_be) { + + struct snd_soc_pcm_runtime *be = dpcm->be; + struct snd_pcm_substream *be_substream = + snd_soc_dpcm_get_substream(be, stream); + + /* is this op for this BE ? */ + if (!snd_soc_dpcm_be_can_update(fe, be, stream)) + continue; + + if ((be->dpcm[stream].state != SND_SOC_DPCM_STATE_HW_PARAMS) && + (be->dpcm[stream].state != SND_SOC_DPCM_STATE_STOP)) + continue; + + dev_dbg(be->dev, "dpcm: prepare BE %s\n", + dpcm->fe->dai_link->name); + + ret = soc_pcm_prepare(be_substream); + if (ret < 0) { + dev_err(be->dev, "dpcm: backend prepare failed %d\n", + ret); + break; + } + + be->dpcm[stream].state = SND_SOC_DPCM_STATE_PREPARE; + } + return ret; +} + +static int dpcm_fe_dai_prepare(struct snd_pcm_substream *substream) +{ + struct snd_soc_pcm_runtime *fe = substream->private_data; + int stream = substream->stream, ret = 0; + + mutex_lock_nested(&fe->card->mutex, SND_SOC_CARD_CLASS_RUNTIME); + + dev_dbg(fe->dev, "dpcm: prepare FE %s\n", fe->dai_link->name); + + fe->dpcm[stream].runtime_update = SND_SOC_DPCM_UPDATE_FE; + + /* there is no point preparing this FE if there are no BEs */ + if (list_empty(&fe->dpcm[stream].be_clients)) { + dev_err(fe->dev, "dpcm: no backend DAIs enabled for %s\n", + fe->dai_link->name); + ret = -EINVAL; + goto out; + } + + ret = dpcm_be_dai_prepare(fe, substream->stream); + if (ret < 0) + goto out; + + /* call prepare on the frontend */ + ret = soc_pcm_prepare(substream); + if (ret < 0) { + dev_err(fe->dev,"dpcm: prepare FE %s failed\n", + fe->dai_link->name); + goto out; + } + + /* run the stream event for each BE */ + dpcm_dapm_stream_event(fe, stream, SND_SOC_DAPM_STREAM_START); + fe->dpcm[stream].state = SND_SOC_DPCM_STATE_PREPARE; + +out: + fe->dpcm[stream].runtime_update = SND_SOC_DPCM_UPDATE_NO; + mutex_unlock(&fe->card->mutex); + + return ret; +} + +static int soc_pcm_ioctl(struct snd_pcm_substream *substream, + unsigned int cmd, void *arg) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_soc_platform *platform = rtd->platform; + + if (platform->driver->ops->ioctl) + return platform->driver->ops->ioctl(substream, cmd, arg); + return snd_pcm_lib_ioctl(substream, cmd, arg); +} + +static int dpcm_run_update_shutdown(struct snd_soc_pcm_runtime *fe, int stream) +{ + struct snd_pcm_substream *substream = + snd_soc_dpcm_get_substream(fe, stream); + enum snd_soc_dpcm_trigger trigger = fe->dai_link->trigger[stream]; + int err; + + dev_dbg(fe->dev, "runtime %s close on FE %s\n", + stream ? "capture" : "playback", fe->dai_link->name); + + if (trigger == SND_SOC_DPCM_TRIGGER_BESPOKE) { + /* call bespoke trigger - FE takes care of all BE triggers */ + dev_dbg(fe->dev, "dpcm: bespoke trigger FE %s cmd stop\n", + fe->dai_link->name); + + err = soc_pcm_bespoke_trigger(substream, SNDRV_PCM_TRIGGER_STOP); + if (err < 0) + dev_err(fe->dev,"dpcm: trigger FE failed %d\n", err); + } else { + dev_dbg(fe->dev, "dpcm: trigger FE %s cmd stop\n", + fe->dai_link->name); + + err = dpcm_be_dai_trigger(fe, stream, SNDRV_PCM_TRIGGER_STOP); + if (err < 0) + dev_err(fe->dev,"dpcm: trigger FE failed %d\n", err); + } + + err = dpcm_be_dai_hw_free(fe, stream); + if (err < 0) + dev_err(fe->dev,"dpcm: hw_free FE failed %d\n", err); + + err = dpcm_be_dai_shutdown(fe, stream); + if (err < 0) + dev_err(fe->dev,"dpcm: shutdown FE failed %d\n", err); + + /* run the stream event for each BE */ + dpcm_dapm_stream_event(fe, stream, SND_SOC_DAPM_STREAM_NOP); + + return 0; +} + +static int dpcm_run_update_startup(struct snd_soc_pcm_runtime *fe, int stream) +{ + struct snd_pcm_substream *substream = + snd_soc_dpcm_get_substream(fe, stream); + struct snd_soc_dpcm *dpcm; + enum snd_soc_dpcm_trigger trigger = fe->dai_link->trigger[stream]; + int ret; + + dev_dbg(fe->dev, "runtime %s open on FE %s\n", + stream ? "capture" : "playback", fe->dai_link->name); + + /* Only start the BE if the FE is ready */ + if (fe->dpcm[stream].state == SND_SOC_DPCM_STATE_HW_FREE || + fe->dpcm[stream].state == SND_SOC_DPCM_STATE_CLOSE) + return -EINVAL; + + /* startup must always be called for new BEs */ + ret = dpcm_be_dai_startup(fe, stream); + if (ret < 0) { + goto disconnect; + return ret; + } + + /* keep going if FE state is > open */ + if (fe->dpcm[stream].state == SND_SOC_DPCM_STATE_OPEN) + return 0; + + ret = dpcm_be_dai_hw_params(fe, stream); + if (ret < 0) { + goto close; + return ret; + } + + /* keep going if FE state is > hw_params */ + if (fe->dpcm[stream].state == SND_SOC_DPCM_STATE_HW_PARAMS) + return 0; + + + ret = dpcm_be_dai_prepare(fe, stream); + if (ret < 0) { + goto hw_free; + return ret; + } + + /* run the stream event for each BE */ + dpcm_dapm_stream_event(fe, stream, SND_SOC_DAPM_STREAM_NOP); + + /* keep going if FE state is > prepare */ + if (fe->dpcm[stream].state == SND_SOC_DPCM_STATE_PREPARE || + fe->dpcm[stream].state == SND_SOC_DPCM_STATE_STOP) + return 0; + + if (trigger == SND_SOC_DPCM_TRIGGER_BESPOKE) { + /* call trigger on the frontend - FE takes care of all BE triggers */ + dev_dbg(fe->dev, "dpcm: bespoke trigger FE %s cmd start\n", + fe->dai_link->name); + + ret = soc_pcm_bespoke_trigger(substream, SNDRV_PCM_TRIGGER_START); + if (ret < 0) { + dev_err(fe->dev,"dpcm: bespoke trigger FE failed %d\n", ret); + goto hw_free; + } + } else { + dev_dbg(fe->dev, "dpcm: trigger FE %s cmd start\n", + fe->dai_link->name); + + ret = dpcm_be_dai_trigger(fe, stream, + SNDRV_PCM_TRIGGER_START); + if (ret < 0) { + dev_err(fe->dev,"dpcm: trigger FE failed %d\n", ret); + goto hw_free; + } + } + + return 0; + +hw_free: + dpcm_be_dai_hw_free(fe, stream); +close: + dpcm_be_dai_shutdown(fe, stream); +disconnect: + /* disconnect any non started BEs */ + list_for_each_entry(dpcm, &fe->dpcm[stream].be_clients, list_be) { + struct snd_soc_pcm_runtime *be = dpcm->be; + if (be->dpcm[stream].state != SND_SOC_DPCM_STATE_START) + dpcm->state = SND_SOC_DPCM_LINK_STATE_FREE; + } + + return ret; +} + +static int dpcm_run_new_update(struct snd_soc_pcm_runtime *fe, int stream) +{ + int ret; + + fe->dpcm[stream].runtime_update = SND_SOC_DPCM_UPDATE_BE; + ret = dpcm_run_update_startup(fe, stream); + if (ret < 0) + dev_err(fe->dev, "failed to startup some BEs\n"); + fe->dpcm[stream].runtime_update = SND_SOC_DPCM_UPDATE_NO; + + return ret; +} + +static int dpcm_run_old_update(struct snd_soc_pcm_runtime *fe, int stream) +{ + int ret; + + fe->dpcm[stream].runtime_update = SND_SOC_DPCM_UPDATE_BE; + ret = dpcm_run_update_shutdown(fe, stream); + if (ret < 0) + dev_err(fe->dev, "failed to shutdown some BEs\n"); + fe->dpcm[stream].runtime_update = SND_SOC_DPCM_UPDATE_NO; + + return ret; +} + +/* Called by DAPM mixer/mux changes to update audio routing between PCMs and + * any DAI links. + */ +int soc_dpcm_runtime_update(struct snd_soc_dapm_widget *widget) +{ + struct snd_soc_card *card; + int i, old, new, paths; + + if (widget->codec) + card = widget->codec->card; + else if (widget->platform) + card = widget->platform->card; + else + return -EINVAL; + + mutex_lock_nested(&card->mutex, SND_SOC_CARD_CLASS_RUNTIME); + for (i = 0; i < card->num_rtd; i++) { + struct snd_soc_dapm_widget_list *list; + struct snd_soc_pcm_runtime *fe = &card->rtd[i]; + + /* make sure link is FE */ + if (!fe->dai_link->dynamic) + continue; + + /* only check active links */ + if (!fe->cpu_dai->active) + continue; + + /* DAPM sync will call this to update DSP paths */ + dev_dbg(fe->dev, "DPCM runtime update for FE %s\n", + fe->dai_link->name); + + /* skip if FE doesn't have playback capability */ + if (!fe->cpu_dai->driver->playback.channels_min) + goto capture; + + paths = dpcm_path_get(fe, SNDRV_PCM_STREAM_PLAYBACK, &list); + if (paths < 0) { + dev_warn(fe->dev, "%s no valid %s path\n", + fe->dai_link->name, "playback"); + mutex_unlock(&card->mutex); + return paths; + } + + /* update any new playback paths */ + new = dpcm_process_paths(fe, SNDRV_PCM_STREAM_PLAYBACK, &list, 1); + if (new) { + dpcm_run_new_update(fe, SNDRV_PCM_STREAM_PLAYBACK); + dpcm_clear_pending_state(fe, SNDRV_PCM_STREAM_PLAYBACK); + dpcm_be_disconnect(fe, SNDRV_PCM_STREAM_PLAYBACK); + } + + /* update any old playback paths */ + old = dpcm_process_paths(fe, SNDRV_PCM_STREAM_PLAYBACK, &list, 0); + if (old) { + dpcm_run_old_update(fe, SNDRV_PCM_STREAM_PLAYBACK); + dpcm_clear_pending_state(fe, SNDRV_PCM_STREAM_PLAYBACK); + dpcm_be_disconnect(fe, SNDRV_PCM_STREAM_PLAYBACK); + } + +capture: + /* skip if FE doesn't have capture capability */ + if (!fe->cpu_dai->driver->capture.channels_min) + continue; + + paths = dpcm_path_get(fe, SNDRV_PCM_STREAM_CAPTURE, &list); + if (paths < 0) { + dev_warn(fe->dev, "%s no valid %s path\n", + fe->dai_link->name, "capture"); + mutex_unlock(&card->mutex); + return paths; + } + + /* update any new capture paths */ + new = dpcm_process_paths(fe, SNDRV_PCM_STREAM_CAPTURE, &list, 1); + if (new) { + dpcm_run_new_update(fe, SNDRV_PCM_STREAM_CAPTURE); + dpcm_clear_pending_state(fe, SNDRV_PCM_STREAM_CAPTURE); + dpcm_be_disconnect(fe, SNDRV_PCM_STREAM_CAPTURE); + } + + /* update any old capture paths */ + old = dpcm_process_paths(fe, SNDRV_PCM_STREAM_CAPTURE, &list, 0); + if (old) { + dpcm_run_old_update(fe, SNDRV_PCM_STREAM_CAPTURE); + dpcm_clear_pending_state(fe, SNDRV_PCM_STREAM_CAPTURE); + dpcm_be_disconnect(fe, SNDRV_PCM_STREAM_CAPTURE); + } + + dpcm_path_put(&list); + } + + mutex_unlock(&card->mutex); + return 0; +} +int soc_dpcm_be_digital_mute(struct snd_soc_pcm_runtime *fe, int mute) +{ + struct snd_soc_dpcm *dpcm; + struct list_head *clients = + &fe->dpcm[SNDRV_PCM_STREAM_PLAYBACK].be_clients; + + list_for_each_entry(dpcm, clients, list_be) { + + struct snd_soc_pcm_runtime *be = dpcm->be; + struct snd_soc_dai *dai = be->codec_dai; + struct snd_soc_dai_driver *drv = dai->driver; + + if (be->dai_link->ignore_suspend) + continue; + + dev_dbg(be->dev, "BE digital mute %s\n", be->dai_link->name); + + if (drv->ops->digital_mute && dai->playback_active) + drv->ops->digital_mute(dai, mute); + } + + return 0; +} + +static int dpcm_fe_dai_open(struct snd_pcm_substream *fe_substream) +{ + struct snd_soc_pcm_runtime *fe = fe_substream->private_data; + struct snd_soc_dpcm *dpcm; + struct snd_soc_dapm_widget_list *list; + int ret; + int stream = fe_substream->stream; + + mutex_lock_nested(&fe->card->mutex, SND_SOC_CARD_CLASS_RUNTIME); + fe->dpcm[stream].runtime = fe_substream->runtime; + + if (dpcm_path_get(fe, stream, &list) <= 0) { + dev_warn(fe->dev, "asoc: %s no valid %s route\n", + fe->dai_link->name, stream ? "capture" : "playback"); + mutex_unlock(&fe->card->mutex); + return -EINVAL; + } + + /* calculate valid and active FE <-> BE dpcms */ + dpcm_process_paths(fe, stream, &list, 1); + + ret = dpcm_fe_dai_startup(fe_substream); + if (ret < 0) { + /* clean up all links */ + list_for_each_entry(dpcm, &fe->dpcm[stream].be_clients, list_be) + dpcm->state = SND_SOC_DPCM_LINK_STATE_FREE; + + dpcm_be_disconnect(fe, stream); + fe->dpcm[stream].runtime = NULL; + } + + dpcm_clear_pending_state(fe, stream); + dpcm_path_put(&list); + mutex_unlock(&fe->card->mutex); + return ret; +} + +static int dpcm_fe_dai_close(struct snd_pcm_substream *fe_substream) +{ + struct snd_soc_pcm_runtime *fe = fe_substream->private_data; + struct snd_soc_dpcm *dpcm; + int stream = fe_substream->stream, ret; + + mutex_lock_nested(&fe->card->mutex, SND_SOC_CARD_CLASS_RUNTIME); + ret = dpcm_fe_dai_shutdown(fe_substream); + + /* mark FE's links ready to prune */ + list_for_each_entry(dpcm, &fe->dpcm[stream].be_clients, list_be) + dpcm->state = SND_SOC_DPCM_LINK_STATE_FREE; + + dpcm_be_disconnect(fe, stream); + + fe->dpcm[stream].runtime = NULL; + mutex_unlock(&fe->card->mutex); + return ret; +} + /* create a new pcm */ int soc_new_pcm(struct snd_soc_pcm_runtime *rtd, int num) { @@ -641,56 +2001,94 @@ int soc_new_pcm(struct snd_soc_pcm_runtime *rtd, int num) struct snd_soc_platform *platform = rtd->platform; struct snd_soc_dai *codec_dai = rtd->codec_dai; struct snd_soc_dai *cpu_dai = rtd->cpu_dai; - struct snd_pcm_ops *soc_pcm_ops = &rtd->ops; struct snd_pcm *pcm; char new_name[64]; int ret = 0, playback = 0, capture = 0; - soc_pcm_ops->open = soc_pcm_open; - soc_pcm_ops->close = soc_pcm_close; - soc_pcm_ops->hw_params = soc_pcm_hw_params; - soc_pcm_ops->hw_free = soc_pcm_hw_free; - soc_pcm_ops->prepare = soc_pcm_prepare; - soc_pcm_ops->trigger = soc_pcm_trigger; - soc_pcm_ops->pointer = soc_pcm_pointer; - - /* check client and interface hw capabilities */ - snprintf(new_name, sizeof(new_name), "%s %s-%d", - rtd->dai_link->stream_name, codec_dai->name, num); - - if (codec_dai->driver->playback.channels_min) - playback = 1; - if (codec_dai->driver->capture.channels_min) - capture = 1; - - dev_dbg(rtd->card->dev, "registered pcm #%d %s\n",num,new_name); - ret = snd_pcm_new(rtd->card->snd_card, new_name, - num, playback, capture, &pcm); + if (rtd->dai_link->dynamic || rtd->dai_link->no_pcm) { + if (cpu_dai->driver->playback.channels_min) + playback = 1; + if (cpu_dai->driver->capture.channels_min) + capture = 1; + } else { + if (codec_dai->driver->playback.channels_min) + playback = 1; + if (codec_dai->driver->capture.channels_min) + capture = 1; + } + + /* create the PCM */ + if (rtd->dai_link->no_pcm) { + snprintf(new_name, sizeof(new_name), "(%s)", + rtd->dai_link->stream_name); + + ret = snd_pcm_new_internal(rtd->card->snd_card, new_name, num, + playback, capture, &pcm); + } else { + if (rtd->dai_link->dynamic) + snprintf(new_name, sizeof(new_name), "%s (*)", + rtd->dai_link->stream_name); + else + snprintf(new_name, sizeof(new_name), "%s %s-%d", + rtd->dai_link->stream_name, codec_dai->name, num); + + ret = snd_pcm_new(rtd->card->snd_card, new_name, num, playback, + capture, &pcm); + } if (ret < 0) { printk(KERN_ERR "asoc: can't create pcm for codec %s\n", codec->name); return ret; } + dev_dbg(rtd->card->dev, "registered pcm #%d %s\n",num, new_name); /* DAPM dai link stream work */ INIT_DELAYED_WORK(&rtd->delayed_work, close_delayed_work); rtd->pcm = pcm; pcm->private_data = rtd; + + if (rtd->dai_link->no_pcm) { + if (playback) + pcm->streams[SNDRV_PCM_STREAM_PLAYBACK].substream->private_data = rtd; + if (capture) + pcm->streams[SNDRV_PCM_STREAM_CAPTURE].substream->private_data = rtd; + goto out; + } + + /* ASoC PCM operations */ + if (rtd->dai_link->dynamic) { + rtd->ops.open = dpcm_fe_dai_open; + rtd->ops.hw_params = dpcm_fe_dai_hw_params; + rtd->ops.prepare = dpcm_fe_dai_prepare; + rtd->ops.trigger = dpcm_fe_dai_trigger; + rtd->ops.hw_free = dpcm_fe_dai_hw_free; + rtd->ops.close = dpcm_fe_dai_close; + rtd->ops.pointer = soc_pcm_pointer; + rtd->ops.ioctl = soc_pcm_ioctl; + } else { + rtd->ops.open = soc_pcm_open; + rtd->ops.hw_params = soc_pcm_hw_params; + rtd->ops.prepare = soc_pcm_prepare; + rtd->ops.trigger = soc_pcm_trigger; + rtd->ops.hw_free = soc_pcm_hw_free; + rtd->ops.close = soc_pcm_close; + rtd->ops.pointer = soc_pcm_pointer; + rtd->ops.ioctl = soc_pcm_ioctl; + } + if (platform->driver->ops) { - soc_pcm_ops->mmap = platform->driver->ops->mmap; - soc_pcm_ops->pointer = platform->driver->ops->pointer; - soc_pcm_ops->ioctl = platform->driver->ops->ioctl; - soc_pcm_ops->copy = platform->driver->ops->copy; - soc_pcm_ops->silence = platform->driver->ops->silence; - soc_pcm_ops->ack = platform->driver->ops->ack; - soc_pcm_ops->page = platform->driver->ops->page; + rtd->ops.ack = platform->driver->ops->ack; + rtd->ops.copy = platform->driver->ops->copy; + rtd->ops.silence = platform->driver->ops->silence; + rtd->ops.page = platform->driver->ops->page; + rtd->ops.mmap = platform->driver->ops->mmap; } if (playback) - snd_pcm_set_ops(pcm, SNDRV_PCM_STREAM_PLAYBACK, soc_pcm_ops); + snd_pcm_set_ops(pcm, SNDRV_PCM_STREAM_PLAYBACK, &rtd->ops); if (capture) - snd_pcm_set_ops(pcm, SNDRV_PCM_STREAM_CAPTURE, soc_pcm_ops); + snd_pcm_set_ops(pcm, SNDRV_PCM_STREAM_CAPTURE, &rtd->ops); if (platform->driver->pcm_new) { ret = platform->driver->pcm_new(rtd); @@ -701,7 +2099,257 @@ int soc_new_pcm(struct snd_soc_pcm_runtime *rtd, int num) } pcm->private_free = platform->driver->pcm_free; +out: printk(KERN_INFO "asoc: %s <-> %s mapping ok\n", codec_dai->name, cpu_dai->name); return ret; } + +/* is the current PCM operation for this FE ? */ +int snd_soc_dpcm_fe_can_update(struct snd_soc_pcm_runtime *fe, int stream) +{ + if (fe->dpcm[stream].runtime_update == SND_SOC_DPCM_UPDATE_FE) + return 1; + return 0; +} +EXPORT_SYMBOL_GPL(snd_soc_dpcm_fe_can_update); + +/* is the current PCM operation for this BE ? */ +int snd_soc_dpcm_be_can_update(struct snd_soc_pcm_runtime *fe, + struct snd_soc_pcm_runtime *be, int stream) +{ + if ((fe->dpcm[stream].runtime_update == SND_SOC_DPCM_UPDATE_FE) || + ((fe->dpcm[stream].runtime_update == SND_SOC_DPCM_UPDATE_BE) && + be->dpcm[stream].runtime_update)) + return 1; + return 0; +} +EXPORT_SYMBOL_GPL(snd_soc_dpcm_be_can_update); + +/* get the substream for this BE */ +struct snd_pcm_substream * + snd_soc_dpcm_get_substream(struct snd_soc_pcm_runtime *be, int stream) +{ + return be->pcm->streams[stream].substream; +} +EXPORT_SYMBOL_GPL(snd_soc_dpcm_get_substream); + +/* get the BE runtime state */ +enum snd_soc_dpcm_state + snd_soc_dpcm_be_get_state(struct snd_soc_pcm_runtime *be, int stream) +{ + return be->dpcm[stream].state; +} +EXPORT_SYMBOL_GPL(snd_soc_dpcm_be_get_state); + +/* set the BE runtime state */ +void snd_soc_dpcm_be_set_state(struct snd_soc_pcm_runtime *be, + int stream, enum snd_soc_dpcm_state state) +{ + be->dpcm[stream].state = state; +} +EXPORT_SYMBOL_GPL(snd_soc_dpcm_be_set_state); + +/* + * We can only hw_free, stop, pause or suspend a BE DAI if any of it's FE + * are not running, paused or suspended for the specified stream direction. + */ +int snd_soc_dpcm_can_be_free_stop(struct snd_soc_pcm_runtime *fe, + struct snd_soc_pcm_runtime *be, int stream) +{ + struct snd_soc_dpcm *dpcm; + int state; + + list_for_each_entry(dpcm, &be->dpcm[stream].fe_clients, list_fe) { + + if (dpcm->fe == fe) + continue; + + state = dpcm->fe->dpcm[stream].state; + if (state == SND_SOC_DPCM_STATE_START || + state == SND_SOC_DPCM_STATE_PAUSED || + state == SND_SOC_DPCM_STATE_SUSPEND) + return 0; + } + + /* it's safe to free/stop this BE DAI */ + return 1; +} +EXPORT_SYMBOL_GPL(snd_soc_dpcm_can_be_free_stop); + +/* + * We can only change hw params a BE DAI if any of it's FE are not prepared, + * running, paused or suspended for the specified stream direction. + */ +int snd_soc_dpcm_can_be_params(struct snd_soc_pcm_runtime *fe, + struct snd_soc_pcm_runtime *be, int stream) +{ + struct snd_soc_dpcm *dpcm; + int state; + + list_for_each_entry(dpcm, &be->dpcm[stream].fe_clients, list_fe) { + + if (dpcm->fe == fe) + continue; + + state = dpcm->fe->dpcm[stream].state; + if (state == SND_SOC_DPCM_STATE_START || + state == SND_SOC_DPCM_STATE_PAUSED || + state == SND_SOC_DPCM_STATE_SUSPEND || + state == SND_SOC_DPCM_STATE_PREPARE) + return 0; + } + + /* it's safe to change hw_params */ + return 1; +} +EXPORT_SYMBOL_GPL(snd_soc_dpcm_can_be_params); + +int snd_soc_platform_trigger(struct snd_pcm_substream *substream, + int cmd, struct snd_soc_platform *platform) +{ + if (platform->driver->ops->trigger) + return platform->driver->ops->trigger(substream, cmd); + return 0; +} +EXPORT_SYMBOL_GPL(snd_soc_platform_trigger); + +#ifdef CONFIG_DEBUG_FS +static char *dpcm_state_string(enum snd_soc_dpcm_state state) +{ + switch (state) { + case SND_SOC_DPCM_STATE_NEW: + return "new"; + case SND_SOC_DPCM_STATE_OPEN: + return "open"; + case SND_SOC_DPCM_STATE_HW_PARAMS: + return "hw_params"; + case SND_SOC_DPCM_STATE_PREPARE: + return "prepare"; + case SND_SOC_DPCM_STATE_START: + return "start"; + case SND_SOC_DPCM_STATE_STOP: + return "stop"; + case SND_SOC_DPCM_STATE_SUSPEND: + return "suspend"; + case SND_SOC_DPCM_STATE_PAUSED: + return "paused"; + case SND_SOC_DPCM_STATE_HW_FREE: + return "hw_free"; + case SND_SOC_DPCM_STATE_CLOSE: + return "close"; + } + + return "unknown"; +} + +static ssize_t dpcm_show_state(struct snd_soc_pcm_runtime *fe, + int stream, char *buf, size_t size) +{ + struct snd_pcm_hw_params *params = &fe->dpcm[stream].hw_params; + struct snd_soc_dpcm *dpcm; + ssize_t offset = 0; + + /* FE state */ + offset += snprintf(buf + offset, size - offset, + "[%s - %s]\n", fe->dai_link->name, + stream ? "Capture" : "Playback"); + + offset += snprintf(buf + offset, size - offset, "State: %s\n", + dpcm_state_string(fe->dpcm[stream].state)); + + if ((fe->dpcm[stream].state >= SND_SOC_DPCM_STATE_HW_PARAMS) && + (fe->dpcm[stream].state <= SND_SOC_DPCM_STATE_STOP)) + offset += snprintf(buf + offset, size - offset, + "Hardware Params: " + "Format = %s, Channels = %d, Rate = %d\n", + snd_pcm_format_name(params_format(params)), + params_channels(params), + params_rate(params)); + + /* BEs state */ + offset += snprintf(buf + offset, size - offset, "Backends:\n"); + + if (list_empty(&fe->dpcm[stream].be_clients)) { + offset += snprintf(buf + offset, size - offset, + " No active DSP links\n"); + goto out; + } + + list_for_each_entry(dpcm, &fe->dpcm[stream].be_clients, list_be) { + struct snd_soc_pcm_runtime *be = dpcm->be; + params = &dpcm->hw_params; + + offset += snprintf(buf + offset, size - offset, + "- %s\n", be->dai_link->name); + + offset += snprintf(buf + offset, size - offset, + " State: %s\n", + dpcm_state_string(be->dpcm[stream].state)); + + if ((be->dpcm[stream].state >= SND_SOC_DPCM_STATE_HW_PARAMS) && + (be->dpcm[stream].state <= SND_SOC_DPCM_STATE_STOP)) + offset += snprintf(buf + offset, size - offset, + " Hardware Params: " + "Format = %s, Channels = %d, Rate = %d\n", + snd_pcm_format_name(params_format(params)), + params_channels(params), + params_rate(params)); + } + +out: + return offset; +} + +static ssize_t dpcm_state_read_file(struct file *file, char __user *user_buf, + size_t count, loff_t *ppos) +{ + struct snd_soc_pcm_runtime *fe = file->private_data; + ssize_t out_count = PAGE_SIZE, offset = 0, ret = 0; + char *buf; + + buf = kmalloc(out_count, GFP_KERNEL); + if (!buf) + return -ENOMEM; + + if (fe->cpu_dai->driver->playback.channels_min) + offset += dpcm_show_state(fe, SNDRV_PCM_STREAM_PLAYBACK, + buf + offset, out_count - offset); + + if (fe->cpu_dai->driver->capture.channels_min) + offset += dpcm_show_state(fe, SNDRV_PCM_STREAM_CAPTURE, + buf + offset, out_count - offset); + + ret = simple_read_from_buffer(user_buf, count, ppos, buf, offset); + + kfree(buf); + return ret; +} + +static const struct file_operations dpcm_state_fops = { + .open = simple_open, + .read = dpcm_state_read_file, + .llseek = default_llseek, +}; + +int soc_dpcm_debugfs_add(struct snd_soc_pcm_runtime *rtd) +{ + if (!rtd->dai_link) + return 0; + + rtd->debugfs_dpcm_root = debugfs_create_dir(rtd->dai_link->name, + rtd->card->debugfs_card_root); + if (!rtd->debugfs_dpcm_root) { + dev_dbg(rtd->dev, + "ASoC: Failed to create dpcm debugfs directory %s\n", + rtd->dai_link->name); + return -EINVAL; + } + + rtd->debugfs_dpcm_state = debugfs_create_file("state", 0444, + rtd->debugfs_dpcm_root, + rtd, &dpcm_state_fops); + + return 0; +} +#endif diff --git a/sound/soc/tegra/Kconfig b/sound/soc/tegra/Kconfig index ce1b773c351..c1c8e955f4d 100644 --- a/sound/soc/tegra/Kconfig +++ b/sound/soc/tegra/Kconfig @@ -1,26 +1,63 @@ config SND_SOC_TEGRA tristate "SoC Audio for the Tegra System-on-Chip" depends on ARCH_TEGRA && TEGRA_SYSTEM_DMA + select REGMAP_MMIO help Say Y or M here if you want support for SoC audio on Tegra. -config SND_SOC_TEGRA_I2S +config SND_SOC_TEGRA20_DAS tristate - depends on SND_SOC_TEGRA + depends on SND_SOC_TEGRA && ARCH_TEGRA_2x_SOC + help + Say Y or M if you want to add support for the Tegra20 DAS module. + You will also need to select the individual machine drivers to + support below. + +config SND_SOC_TEGRA20_I2S + tristate + depends on SND_SOC_TEGRA && ARCH_TEGRA_2x_SOC + select SND_SOC_TEGRA20_DAS help Say Y or M if you want to add support for codecs attached to the - Tegra I2S interface. You will also need to select the individual + Tegra20 I2S interface. You will also need to select the individual machine drivers to support below. -config SND_SOC_TEGRA_SPDIF +config SND_SOC_TEGRA20_SPDIF tristate - depends on SND_SOC_TEGRA + depends on SND_SOC_TEGRA && ARCH_TEGRA_2x_SOC default m help - Say Y or M if you want to add support for the SPDIF interface. + Say Y or M if you want to add support for the Tegra20 SPDIF interface. You will also need to select the individual machine drivers to support below. +config SND_SOC_TEGRA30_AHUB + tristate + depends on SND_SOC_TEGRA && ARCH_TEGRA_3x_SOC + help + Say Y or M if you want to add support for the Tegra20 AHUB module. + You will also need to select the individual machine drivers to + support below. + +config SND_SOC_TEGRA30_I2S + tristate + depends on SND_SOC_TEGRA && ARCH_TEGRA_3x_SOC + select SND_SOC_TEGRA30_AHUB + help + Say Y or M if you want to add support for codecs attached to the + Tegra30 I2S interface. You will also need to select the individual + machine drivers to support below. + +config SND_SOC_TEGRA_WM8753 + tristate "SoC Audio support for Tegra boards using a WM8753 codec" + depends on SND_SOC_TEGRA && I2C + select SND_SOC_TEGRA20_I2S if ARCH_TEGRA_2x_SOC + select SND_SOC_TEGRA30_I2S if ARCH_TEGRA_3x_SOC + select SND_SOC_WM8753 + help + Say Y or M here if you want to add support for SoC audio on Tegra + boards using the WM8753 codec, such as Whistler. + config MACH_HAS_SND_SOC_TEGRA_WM8903 bool help @@ -32,7 +69,8 @@ config SND_SOC_TEGRA_WM8903 tristate "SoC Audio support for Tegra boards using a WM8903 codec" depends on SND_SOC_TEGRA && I2C depends on MACH_HAS_SND_SOC_TEGRA_WM8903 - select SND_SOC_TEGRA_I2S + select SND_SOC_TEGRA20_I2S if ARCH_TEGRA_2x_SOC + select SND_SOC_TEGRA30_I2S if ARCH_TEGRA_3x_SOC select SND_SOC_WM8903 help Say Y or M here if you want to add support for SoC audio on Tegra @@ -42,17 +80,17 @@ config SND_SOC_TEGRA_WM8903 config SND_SOC_TEGRA_TRIMSLICE tristate "SoC Audio support for TrimSlice board" depends on SND_SOC_TEGRA && MACH_TRIMSLICE && I2C - select SND_SOC_TEGRA_I2S + select SND_SOC_TEGRA20_I2S if ARCH_TEGRA_2x_SOC select SND_SOC_TLV320AIC23 help Say Y or M here if you want to add support for SoC audio on the TrimSlice platform. config SND_SOC_TEGRA_ALC5632 - tristate "SoC Audio support for Tegra boards using an ALC5632 codec" - depends on SND_SOC_TEGRA && I2C - select SND_SOC_TEGRA_I2S - select SND_SOC_ALC5632 - help - Say Y or M here if you want to add support for SoC audio on the - Toshiba AC100 netbook. + tristate "SoC Audio support for Tegra boards using an ALC5632 codec" + depends on SND_SOC_TEGRA && I2C + select SND_SOC_TEGRA20_I2S if ARCH_TEGRA_2x_SOC + select SND_SOC_ALC5632 + help + Say Y or M here if you want to add support for SoC audio on the + Toshiba AC100 netbook. diff --git a/sound/soc/tegra/Makefile b/sound/soc/tegra/Makefile index 8e584b8fcfb..391e78a34c0 100644 --- a/sound/soc/tegra/Makefile +++ b/sound/soc/tegra/Makefile @@ -1,21 +1,27 @@ # Tegra platform Support -snd-soc-tegra-das-objs := tegra_das.o snd-soc-tegra-pcm-objs := tegra_pcm.o -snd-soc-tegra-i2s-objs := tegra_i2s.o -snd-soc-tegra-spdif-objs := tegra_spdif.o snd-soc-tegra-utils-objs += tegra_asoc_utils.o +snd-soc-tegra20-das-objs := tegra20_das.o +snd-soc-tegra20-i2s-objs := tegra20_i2s.o +snd-soc-tegra20-spdif-objs := tegra20_spdif.o +snd-soc-tegra30-ahub-objs := tegra30_ahub.o +snd-soc-tegra30-i2s-objs := tegra30_i2s.o -obj-$(CONFIG_SND_SOC_TEGRA) += snd-soc-tegra-utils.o -obj-$(CONFIG_SND_SOC_TEGRA) += snd-soc-tegra-das.o obj-$(CONFIG_SND_SOC_TEGRA) += snd-soc-tegra-pcm.o -obj-$(CONFIG_SND_SOC_TEGRA_I2S) += snd-soc-tegra-i2s.o -obj-$(CONFIG_SND_SOC_TEGRA_SPDIF) += snd-soc-tegra-spdif.o +obj-$(CONFIG_SND_SOC_TEGRA) += snd-soc-tegra-utils.o +obj-$(CONFIG_SND_SOC_TEGRA20_DAS) += snd-soc-tegra20-das.o +obj-$(CONFIG_SND_SOC_TEGRA20_I2S) += snd-soc-tegra20-i2s.o +obj-$(CONFIG_SND_SOC_TEGRA20_SPDIF) += snd-soc-tegra20-spdif.o +obj-$(CONFIG_SND_SOC_TEGRA30_AHUB) += snd-soc-tegra30-ahub.o +obj-$(CONFIG_SND_SOC_TEGRA30_I2S) += snd-soc-tegra30-i2s.o # Tegra machine Support +snd-soc-tegra-wm8753-objs := tegra_wm8753.o snd-soc-tegra-wm8903-objs := tegra_wm8903.o snd-soc-tegra-trimslice-objs := trimslice.o snd-soc-tegra-alc5632-objs := tegra_alc5632.o +obj-$(CONFIG_SND_SOC_TEGRA_WM8753) += snd-soc-tegra-wm8753.o obj-$(CONFIG_SND_SOC_TEGRA_WM8903) += snd-soc-tegra-wm8903.o obj-$(CONFIG_SND_SOC_TEGRA_TRIMSLICE) += snd-soc-tegra-trimslice.o obj-$(CONFIG_SND_SOC_TEGRA_ALC5632) += snd-soc-tegra-alc5632.o diff --git a/sound/soc/tegra/tegra20_das.c b/sound/soc/tegra/tegra20_das.c new file mode 100644 index 00000000000..bf99296bce9 --- /dev/null +++ b/sound/soc/tegra/tegra20_das.c @@ -0,0 +1,233 @@ +/* + * tegra20_das.c - Tegra20 DAS driver + * + * Author: Stephen Warren <swarren@nvidia.com> + * Copyright (C) 2010 - NVIDIA, Inc. + * + * This program is free software; you can redistribute it and/or + * modify it under the terms of the GNU General Public License + * version 2 as published by the Free Software Foundation. + * + * This program is distributed in the hope that it will be useful, but + * WITHOUT ANY WARRANTY; without even the implied warranty of + * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU + * General Public License for more details. + * + * You should have received a copy of the GNU General Public License + * along with this program; if not, write to the Free Software + * Foundation, Inc., 51 Franklin St, Fifth Floor, Boston, MA + * 02110-1301 USA + * + */ + +#include <linux/device.h> +#include <linux/io.h> +#include <linux/module.h> +#include <linux/platform_device.h> +#include <linux/regmap.h> +#include <linux/slab.h> +#include <sound/soc.h> +#include "tegra20_das.h" + +#define DRV_NAME "tegra20-das" + +static struct tegra20_das *das; + +static inline void tegra20_das_write(u32 reg, u32 val) +{ + regmap_write(das->regmap, reg, val); +} + +static inline u32 tegra20_das_read(u32 reg) +{ + u32 val; + regmap_read(das->regmap, reg, &val); + return val; +} + +int tegra20_das_connect_dap_to_dac(int dap, int dac) +{ + u32 addr; + u32 reg; + + if (!das) + return -ENODEV; + + addr = TEGRA20_DAS_DAP_CTRL_SEL + + (dap * TEGRA20_DAS_DAP_CTRL_SEL_STRIDE); + reg = dac << TEGRA20_DAS_DAP_CTRL_SEL_DAP_CTRL_SEL_P; + + tegra20_das_write(addr, reg); + + return 0; +} +EXPORT_SYMBOL_GPL(tegra20_das_connect_dap_to_dac); + +int tegra20_das_connect_dap_to_dap(int dap, int otherdap, int master, + int sdata1rx, int sdata2rx) +{ + u32 addr; + u32 reg; + + if (!das) + return -ENODEV; + + addr = TEGRA20_DAS_DAP_CTRL_SEL + + (dap * TEGRA20_DAS_DAP_CTRL_SEL_STRIDE); + reg = otherdap << TEGRA20_DAS_DAP_CTRL_SEL_DAP_CTRL_SEL_P | + !!sdata2rx << TEGRA20_DAS_DAP_CTRL_SEL_DAP_SDATA2_TX_RX_P | + !!sdata1rx << TEGRA20_DAS_DAP_CTRL_SEL_DAP_SDATA1_TX_RX_P | + !!master << TEGRA20_DAS_DAP_CTRL_SEL_DAP_MS_SEL_P; + + tegra20_das_write(addr, reg); + + return 0; +} +EXPORT_SYMBOL_GPL(tegra20_das_connect_dap_to_dap); + +int tegra20_das_connect_dac_to_dap(int dac, int dap) +{ + u32 addr; + u32 reg; + + if (!das) + return -ENODEV; + + addr = TEGRA20_DAS_DAC_INPUT_DATA_CLK_SEL + + (dac * TEGRA20_DAS_DAC_INPUT_DATA_CLK_SEL_STRIDE); + reg = dap << TEGRA20_DAS_DAC_INPUT_DATA_CLK_SEL_DAC_CLK_SEL_P | + dap << TEGRA20_DAS_DAC_INPUT_DATA_CLK_SEL_DAC_SDATA1_SEL_P | + dap << TEGRA20_DAS_DAC_INPUT_DATA_CLK_SEL_DAC_SDATA2_SEL_P; + + tegra20_das_write(addr, reg); + + return 0; +} +EXPORT_SYMBOL_GPL(tegra20_das_connect_dac_to_dap); + +#define LAST_REG(name) \ + (TEGRA20_DAS_##name + \ + (TEGRA20_DAS_##name##_STRIDE * (TEGRA20_DAS_##name##_COUNT - 1))) + +static bool tegra20_das_wr_rd_reg(struct device *dev, unsigned int reg) +{ + if ((reg >= TEGRA20_DAS_DAP_CTRL_SEL) && + (reg <= LAST_REG(DAP_CTRL_SEL))) + return true; + if ((reg >= TEGRA20_DAS_DAC_INPUT_DATA_CLK_SEL) && + (reg <= LAST_REG(DAC_INPUT_DATA_CLK_SEL))) + return true; + + return false; +} + +static const struct regmap_config tegra20_das_regmap_config = { + .reg_bits = 32, + .reg_stride = 4, + .val_bits = 32, + .max_register = LAST_REG(DAC_INPUT_DATA_CLK_SEL), + .writeable_reg = tegra20_das_wr_rd_reg, + .readable_reg = tegra20_das_wr_rd_reg, + .cache_type = REGCACHE_RBTREE, +}; + +static int __devinit tegra20_das_probe(struct platform_device *pdev) +{ + struct resource *res, *region; + void __iomem *regs; + int ret = 0; + + if (das) + return -ENODEV; + + das = devm_kzalloc(&pdev->dev, sizeof(struct tegra20_das), GFP_KERNEL); + if (!das) { + dev_err(&pdev->dev, "Can't allocate tegra20_das\n"); + ret = -ENOMEM; + goto err; + } + das->dev = &pdev->dev; + + res = platform_get_resource(pdev, IORESOURCE_MEM, 0); + if (!res) { + dev_err(&pdev->dev, "No memory resource\n"); + ret = -ENODEV; + goto err; + } + + region = devm_request_mem_region(&pdev->dev, res->start, + resource_size(res), pdev->name); + if (!region) { + dev_err(&pdev->dev, "Memory region already claimed\n"); + ret = -EBUSY; + goto err; + } + + regs = devm_ioremap(&pdev->dev, res->start, resource_size(res)); + if (!regs) { + dev_err(&pdev->dev, "ioremap failed\n"); + ret = -ENOMEM; + goto err; + } + + das->regmap = devm_regmap_init_mmio(&pdev->dev, regs, + &tegra20_das_regmap_config); + if (IS_ERR(das->regmap)) { + dev_err(&pdev->dev, "regmap init failed\n"); + ret = PTR_ERR(das->regmap); + goto err; + } + + ret = tegra20_das_connect_dap_to_dac(TEGRA20_DAS_DAP_ID_1, + TEGRA20_DAS_DAP_SEL_DAC1); + if (ret) { + dev_err(&pdev->dev, "Can't set up DAS DAP connection\n"); + goto err; + } + ret = tegra20_das_connect_dac_to_dap(TEGRA20_DAS_DAC_ID_1, + TEGRA20_DAS_DAC_SEL_DAP1); + if (ret) { + dev_err(&pdev->dev, "Can't set up DAS DAC connection\n"); + goto err; + } + + platform_set_drvdata(pdev, das); + + return 0; + +err: + das = NULL; + return ret; +} + +static int __devexit tegra20_das_remove(struct platform_device *pdev) +{ + if (!das) + return -ENODEV; + + das = NULL; + + return 0; +} + +static const struct of_device_id tegra20_das_of_match[] __devinitconst = { + { .compatible = "nvidia,tegra20-das", }, + {}, +}; + +static struct platform_driver tegra20_das_driver = { + .probe = tegra20_das_probe, + .remove = __devexit_p(tegra20_das_remove), + .driver = { + .name = DRV_NAME, + .owner = THIS_MODULE, + .of_match_table = tegra20_das_of_match, + }, +}; +module_platform_driver(tegra20_das_driver); + +MODULE_AUTHOR("Stephen Warren <swarren@nvidia.com>"); +MODULE_DESCRIPTION("Tegra20 DAS driver"); +MODULE_LICENSE("GPL"); +MODULE_ALIAS("platform:" DRV_NAME); +MODULE_DEVICE_TABLE(of, tegra20_das_of_match); diff --git a/sound/soc/tegra/tegra20_das.h b/sound/soc/tegra/tegra20_das.h new file mode 100644 index 00000000000..be217f3d3a7 --- /dev/null +++ b/sound/soc/tegra/tegra20_das.h @@ -0,0 +1,134 @@ +/* + * tegra20_das.h - Definitions for Tegra20 DAS driver + * + * Author: Stephen Warren <swarren@nvidia.com> + * Copyright (C) 2010,2012 - NVIDIA, Inc. + * + * This program is free software; you can redistribute it and/or + * modify it under the terms of the GNU General Public License + * version 2 as published by the Free Software Foundation. + * + * This program is distributed in the hope that it will be useful, but + * WITHOUT ANY WARRANTY; without even the implied warranty of + * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU + * General Public License for more details. + * + * You should have received a copy of the GNU General Public License + * along with this program; if not, write to the Free Software + * Foundation, Inc., 51 Franklin St, Fifth Floor, Boston, MA + * 02110-1301 USA + * + */ + +#ifndef __TEGRA20_DAS_H__ +#define __TEGRA20_DAS_H__ + +/* Register TEGRA20_DAS_DAP_CTRL_SEL */ +#define TEGRA20_DAS_DAP_CTRL_SEL 0x00 +#define TEGRA20_DAS_DAP_CTRL_SEL_COUNT 5 +#define TEGRA20_DAS_DAP_CTRL_SEL_STRIDE 4 +#define TEGRA20_DAS_DAP_CTRL_SEL_DAP_MS_SEL_P 31 +#define TEGRA20_DAS_DAP_CTRL_SEL_DAP_MS_SEL_S 1 +#define TEGRA20_DAS_DAP_CTRL_SEL_DAP_SDATA1_TX_RX_P 30 +#define TEGRA20_DAS_DAP_CTRL_SEL_DAP_SDATA1_TX_RX_S 1 +#define TEGRA20_DAS_DAP_CTRL_SEL_DAP_SDATA2_TX_RX_P 29 +#define TEGRA20_DAS_DAP_CTRL_SEL_DAP_SDATA2_TX_RX_S 1 +#define TEGRA20_DAS_DAP_CTRL_SEL_DAP_CTRL_SEL_P 0 +#define TEGRA20_DAS_DAP_CTRL_SEL_DAP_CTRL_SEL_S 5 + +/* Values for field TEGRA20_DAS_DAP_CTRL_SEL_DAP_CTRL_SEL */ +#define TEGRA20_DAS_DAP_SEL_DAC1 0 +#define TEGRA20_DAS_DAP_SEL_DAC2 1 +#define TEGRA20_DAS_DAP_SEL_DAC3 2 +#define TEGRA20_DAS_DAP_SEL_DAP1 16 +#define TEGRA20_DAS_DAP_SEL_DAP2 17 +#define TEGRA20_DAS_DAP_SEL_DAP3 18 +#define TEGRA20_DAS_DAP_SEL_DAP4 19 +#define TEGRA20_DAS_DAP_SEL_DAP5 20 + +/* Register TEGRA20_DAS_DAC_INPUT_DATA_CLK_SEL */ +#define TEGRA20_DAS_DAC_INPUT_DATA_CLK_SEL 0x40 +#define TEGRA20_DAS_DAC_INPUT_DATA_CLK_SEL_COUNT 3 +#define TEGRA20_DAS_DAC_INPUT_DATA_CLK_SEL_STRIDE 4 +#define TEGRA20_DAS_DAC_INPUT_DATA_CLK_SEL_DAC_SDATA2_SEL_P 28 +#define TEGRA20_DAS_DAC_INPUT_DATA_CLK_SEL_DAC_SDATA2_SEL_S 4 +#define TEGRA20_DAS_DAC_INPUT_DATA_CLK_SEL_DAC_SDATA1_SEL_P 24 +#define TEGRA20_DAS_DAC_INPUT_DATA_CLK_SEL_DAC_SDATA1_SEL_S 4 +#define TEGRA20_DAS_DAC_INPUT_DATA_CLK_SEL_DAC_CLK_SEL_P 0 +#define TEGRA20_DAS_DAC_INPUT_DATA_CLK_SEL_DAC_CLK_SEL_S 4 + +/* + * Values for: + * TEGRA20_DAS_DAC_INPUT_DATA_CLK_SEL_DAC_SDATA2_SEL + * TEGRA20_DAS_DAC_INPUT_DATA_CLK_SEL_DAC_SDATA1_SEL + * TEGRA20_DAS_DAC_INPUT_DATA_CLK_SEL_DAC_CLK_SEL + */ +#define TEGRA20_DAS_DAC_SEL_DAP1 0 +#define TEGRA20_DAS_DAC_SEL_DAP2 1 +#define TEGRA20_DAS_DAC_SEL_DAP3 2 +#define TEGRA20_DAS_DAC_SEL_DAP4 3 +#define TEGRA20_DAS_DAC_SEL_DAP5 4 + +/* + * Names/IDs of the DACs/DAPs. + */ + +#define TEGRA20_DAS_DAP_ID_1 0 +#define TEGRA20_DAS_DAP_ID_2 1 +#define TEGRA20_DAS_DAP_ID_3 2 +#define TEGRA20_DAS_DAP_ID_4 3 +#define TEGRA20_DAS_DAP_ID_5 4 + +#define TEGRA20_DAS_DAC_ID_1 0 +#define TEGRA20_DAS_DAC_ID_2 1 +#define TEGRA20_DAS_DAC_ID_3 2 + +struct tegra20_das { + struct device *dev; + struct regmap *regmap; +}; + +/* + * Terminology: + * DAS: Digital audio switch (HW module controlled by this driver) + * DAP: Digital audio port (port/pins on Tegra device) + * DAC: Digital audio controller (e.g. I2S or AC97 controller elsewhere) + * + * The Tegra DAS is a mux/cross-bar which can connect each DAP to a specific + * DAC, or another DAP. When DAPs are connected, one must be the master and + * one the slave. Each DAC allows selection of a specific DAP for input, to + * cater for the case where N DAPs are connected to 1 DAC for broadcast + * output. + * + * This driver is dumb; no attempt is made to ensure that a valid routing + * configuration is programmed. + */ + +/* + * Connect a DAP to to a DAC + * dap_id: DAP to connect: TEGRA20_DAS_DAP_ID_* + * dac_sel: DAC to connect to: TEGRA20_DAS_DAP_SEL_DAC* + */ +extern int tegra20_das_connect_dap_to_dac(int dap_id, int dac_sel); + +/* + * Connect a DAP to to another DAP + * dap_id: DAP to connect: TEGRA20_DAS_DAP_ID_* + * other_dap_sel: DAP to connect to: TEGRA20_DAS_DAP_SEL_DAP* + * master: Is this DAP the master (1) or slave (0) + * sdata1rx: Is this DAP's SDATA1 pin RX (1) or TX (0) + * sdata2rx: Is this DAP's SDATA2 pin RX (1) or TX (0) + */ +extern int tegra20_das_connect_dap_to_dap(int dap_id, int other_dap_sel, + int master, int sdata1rx, + int sdata2rx); + +/* + * Connect a DAC's input to a DAP + * (DAC outputs are selected by the DAP) + * dac_id: DAC ID to connect: TEGRA20_DAS_DAC_ID_* + * dap_sel: DAP to receive input from: TEGRA20_DAS_DAC_SEL_DAP* + */ +extern int tegra20_das_connect_dac_to_dap(int dac_id, int dap_sel); + +#endif diff --git a/sound/soc/tegra/tegra20_i2s.c b/sound/soc/tegra/tegra20_i2s.c new file mode 100644 index 00000000000..0c7af63d444 --- /dev/null +++ b/sound/soc/tegra/tegra20_i2s.c @@ -0,0 +1,494 @@ +/* + * tegra20_i2s.c - Tegra20 I2S driver + * + * Author: Stephen Warren <swarren@nvidia.com> + * Copyright (C) 2010,2012 - NVIDIA, Inc. + * + * Based on code copyright/by: + * + * Copyright (c) 2009-2010, NVIDIA Corporation. + * Scott Peterson <speterson@nvidia.com> + * + * Copyright (C) 2010 Google, Inc. + * Iliyan Malchev <malchev@google.com> + * + * This program is free software; you can redistribute it and/or + * modify it under the terms of the GNU General Public License + * version 2 as published by the Free Software Foundation. + * + * This program is distributed in the hope that it will be useful, but + * WITHOUT ANY WARRANTY; without even the implied warranty of + * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU + * General Public License for more details. + * + * You should have received a copy of the GNU General Public License + * along with this program; if not, write to the Free Software + * Foundation, Inc., 51 Franklin St, Fifth Floor, Boston, MA + * 02110-1301 USA + * + */ + +#include <linux/clk.h> +#include <linux/device.h> +#include <linux/io.h> +#include <linux/module.h> +#include <linux/of.h> +#include <linux/platform_device.h> +#include <linux/pm_runtime.h> +#include <linux/regmap.h> +#include <linux/slab.h> +#include <sound/core.h> +#include <sound/pcm.h> +#include <sound/pcm_params.h> +#include <sound/soc.h> + +#include "tegra20_i2s.h" + +#define DRV_NAME "tegra20-i2s" + +static inline void tegra20_i2s_write(struct tegra20_i2s *i2s, u32 reg, u32 val) +{ + regmap_write(i2s->regmap, reg, val); +} + +static inline u32 tegra20_i2s_read(struct tegra20_i2s *i2s, u32 reg) +{ + u32 val; + regmap_read(i2s->regmap, reg, &val); + return val; +} + +static int tegra20_i2s_runtime_suspend(struct device *dev) +{ + struct tegra20_i2s *i2s = dev_get_drvdata(dev); + + clk_disable(i2s->clk_i2s); + + return 0; +} + +static int tegra20_i2s_runtime_resume(struct device *dev) +{ + struct tegra20_i2s *i2s = dev_get_drvdata(dev); + int ret; + + ret = clk_enable(i2s->clk_i2s); + if (ret) { + dev_err(dev, "clk_enable failed: %d\n", ret); + return ret; + } + + return 0; +} + +static int tegra20_i2s_set_fmt(struct snd_soc_dai *dai, + unsigned int fmt) +{ + struct tegra20_i2s *i2s = snd_soc_dai_get_drvdata(dai); + + switch (fmt & SND_SOC_DAIFMT_INV_MASK) { + case SND_SOC_DAIFMT_NB_NF: + break; + default: + return -EINVAL; + } + + i2s->reg_ctrl &= ~TEGRA20_I2S_CTRL_MASTER_ENABLE; + switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) { + case SND_SOC_DAIFMT_CBS_CFS: + i2s->reg_ctrl |= TEGRA20_I2S_CTRL_MASTER_ENABLE; + break; + case SND_SOC_DAIFMT_CBM_CFM: + break; + default: + return -EINVAL; + } + + i2s->reg_ctrl &= ~(TEGRA20_I2S_CTRL_BIT_FORMAT_MASK | + TEGRA20_I2S_CTRL_LRCK_MASK); + switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) { + case SND_SOC_DAIFMT_DSP_A: + i2s->reg_ctrl |= TEGRA20_I2S_CTRL_BIT_FORMAT_DSP; + i2s->reg_ctrl |= TEGRA20_I2S_CTRL_LRCK_L_LOW; + break; + case SND_SOC_DAIFMT_DSP_B: + i2s->reg_ctrl |= TEGRA20_I2S_CTRL_BIT_FORMAT_DSP; + i2s->reg_ctrl |= TEGRA20_I2S_CTRL_LRCK_R_LOW; + break; + case SND_SOC_DAIFMT_I2S: + i2s->reg_ctrl |= TEGRA20_I2S_CTRL_BIT_FORMAT_I2S; + i2s->reg_ctrl |= TEGRA20_I2S_CTRL_LRCK_L_LOW; + break; + case SND_SOC_DAIFMT_RIGHT_J: + i2s->reg_ctrl |= TEGRA20_I2S_CTRL_BIT_FORMAT_RJM; + i2s->reg_ctrl |= TEGRA20_I2S_CTRL_LRCK_L_LOW; + break; + case SND_SOC_DAIFMT_LEFT_J: + i2s->reg_ctrl |= TEGRA20_I2S_CTRL_BIT_FORMAT_LJM; + i2s->reg_ctrl |= TEGRA20_I2S_CTRL_LRCK_L_LOW; + break; + default: + return -EINVAL; + } + + return 0; +} + +static int tegra20_i2s_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params, + struct snd_soc_dai *dai) +{ + struct device *dev = substream->pcm->card->dev; + struct tegra20_i2s *i2s = snd_soc_dai_get_drvdata(dai); + u32 reg; + int ret, sample_size, srate, i2sclock, bitcnt; + + i2s->reg_ctrl &= ~TEGRA20_I2S_CTRL_BIT_SIZE_MASK; + switch (params_format(params)) { + case SNDRV_PCM_FORMAT_S16_LE: + i2s->reg_ctrl |= TEGRA20_I2S_CTRL_BIT_SIZE_16; + sample_size = 16; + break; + case SNDRV_PCM_FORMAT_S24_LE: + i2s->reg_ctrl |= TEGRA20_I2S_CTRL_BIT_SIZE_24; + sample_size = 24; + break; + case SNDRV_PCM_FORMAT_S32_LE: + i2s->reg_ctrl |= TEGRA20_I2S_CTRL_BIT_SIZE_32; + sample_size = 32; + break; + default: + return -EINVAL; + } + + srate = params_rate(params); + + /* Final "* 2" required by Tegra hardware */ + i2sclock = srate * params_channels(params) * sample_size * 2; + + ret = clk_set_rate(i2s->clk_i2s, i2sclock); + if (ret) { + dev_err(dev, "Can't set I2S clock rate: %d\n", ret); + return ret; + } + + bitcnt = (i2sclock / (2 * srate)) - 1; + if (bitcnt < 0 || bitcnt > TEGRA20_I2S_TIMING_CHANNEL_BIT_COUNT_MASK_US) + return -EINVAL; + reg = bitcnt << TEGRA20_I2S_TIMING_CHANNEL_BIT_COUNT_SHIFT; + + if (i2sclock % (2 * srate)) + reg |= TEGRA20_I2S_TIMING_NON_SYM_ENABLE; + + tegra20_i2s_write(i2s, TEGRA20_I2S_TIMING, reg); + + tegra20_i2s_write(i2s, TEGRA20_I2S_FIFO_SCR, + TEGRA20_I2S_FIFO_SCR_FIFO2_ATN_LVL_FOUR_SLOTS | + TEGRA20_I2S_FIFO_SCR_FIFO1_ATN_LVL_FOUR_SLOTS); + + return 0; +} + +static void tegra20_i2s_start_playback(struct tegra20_i2s *i2s) +{ + i2s->reg_ctrl |= TEGRA20_I2S_CTRL_FIFO1_ENABLE; + tegra20_i2s_write(i2s, TEGRA20_I2S_CTRL, i2s->reg_ctrl); +} + +static void tegra20_i2s_stop_playback(struct tegra20_i2s *i2s) +{ + i2s->reg_ctrl &= ~TEGRA20_I2S_CTRL_FIFO1_ENABLE; + tegra20_i2s_write(i2s, TEGRA20_I2S_CTRL, i2s->reg_ctrl); +} + +static void tegra20_i2s_start_capture(struct tegra20_i2s *i2s) +{ + i2s->reg_ctrl |= TEGRA20_I2S_CTRL_FIFO2_ENABLE; + tegra20_i2s_write(i2s, TEGRA20_I2S_CTRL, i2s->reg_ctrl); +} + +static void tegra20_i2s_stop_capture(struct tegra20_i2s *i2s) +{ + i2s->reg_ctrl &= ~TEGRA20_I2S_CTRL_FIFO2_ENABLE; + tegra20_i2s_write(i2s, TEGRA20_I2S_CTRL, i2s->reg_ctrl); +} + +static int tegra20_i2s_trigger(struct snd_pcm_substream *substream, int cmd, + struct snd_soc_dai *dai) +{ + struct tegra20_i2s *i2s = snd_soc_dai_get_drvdata(dai); + + switch (cmd) { + case SNDRV_PCM_TRIGGER_START: + case SNDRV_PCM_TRIGGER_PAUSE_RELEASE: + case SNDRV_PCM_TRIGGER_RESUME: + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) + tegra20_i2s_start_playback(i2s); + else + tegra20_i2s_start_capture(i2s); + break; + case SNDRV_PCM_TRIGGER_STOP: + case SNDRV_PCM_TRIGGER_PAUSE_PUSH: + case SNDRV_PCM_TRIGGER_SUSPEND: + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) + tegra20_i2s_stop_playback(i2s); + else + tegra20_i2s_stop_capture(i2s); + break; + default: + return -EINVAL; + } + + return 0; +} + +static int tegra20_i2s_probe(struct snd_soc_dai *dai) +{ + struct tegra20_i2s *i2s = snd_soc_dai_get_drvdata(dai); + + dai->capture_dma_data = &i2s->capture_dma_data; + dai->playback_dma_data = &i2s->playback_dma_data; + + return 0; +} + +static const struct snd_soc_dai_ops tegra20_i2s_dai_ops = { + .set_fmt = tegra20_i2s_set_fmt, + .hw_params = tegra20_i2s_hw_params, + .trigger = tegra20_i2s_trigger, +}; + +static const struct snd_soc_dai_driver tegra20_i2s_dai_template = { + .probe = tegra20_i2s_probe, + .playback = { + .channels_min = 2, + .channels_max = 2, + .rates = SNDRV_PCM_RATE_8000_96000, + .formats = SNDRV_PCM_FMTBIT_S16_LE, + }, + .capture = { + .channels_min = 2, + .channels_max = 2, + .rates = SNDRV_PCM_RATE_8000_96000, + .formats = SNDRV_PCM_FMTBIT_S16_LE, + }, + .ops = &tegra20_i2s_dai_ops, + .symmetric_rates = 1, +}; + +static bool tegra20_i2s_wr_rd_reg(struct device *dev, unsigned int reg) +{ + switch (reg) { + case TEGRA20_I2S_CTRL: + case TEGRA20_I2S_STATUS: + case TEGRA20_I2S_TIMING: + case TEGRA20_I2S_FIFO_SCR: + case TEGRA20_I2S_PCM_CTRL: + case TEGRA20_I2S_NW_CTRL: + case TEGRA20_I2S_TDM_CTRL: + case TEGRA20_I2S_TDM_TX_RX_CTRL: + case TEGRA20_I2S_FIFO1: + case TEGRA20_I2S_FIFO2: + return true; + default: + return false; + }; +} + +static bool tegra20_i2s_volatile_reg(struct device *dev, unsigned int reg) +{ + switch (reg) { + case TEGRA20_I2S_STATUS: + case TEGRA20_I2S_FIFO_SCR: + case TEGRA20_I2S_FIFO1: + case TEGRA20_I2S_FIFO2: + return true; + default: + return false; + }; +} + +static bool tegra20_i2s_precious_reg(struct device *dev, unsigned int reg) +{ + switch (reg) { + case TEGRA20_I2S_FIFO1: + case TEGRA20_I2S_FIFO2: + return true; + default: + return false; + }; +} + +static const struct regmap_config tegra20_i2s_regmap_config = { + .reg_bits = 32, + .reg_stride = 4, + .val_bits = 32, + .max_register = TEGRA20_I2S_FIFO2, + .writeable_reg = tegra20_i2s_wr_rd_reg, + .readable_reg = tegra20_i2s_wr_rd_reg, + .volatile_reg = tegra20_i2s_volatile_reg, + .precious_reg = tegra20_i2s_precious_reg, + .cache_type = REGCACHE_RBTREE, +}; + +static __devinit int tegra20_i2s_platform_probe(struct platform_device *pdev) +{ + struct tegra20_i2s *i2s; + struct resource *mem, *memregion, *dmareq; + u32 of_dma[2]; + u32 dma_ch; + void __iomem *regs; + int ret; + + i2s = devm_kzalloc(&pdev->dev, sizeof(struct tegra20_i2s), GFP_KERNEL); + if (!i2s) { + dev_err(&pdev->dev, "Can't allocate tegra20_i2s\n"); + ret = -ENOMEM; + goto err; + } + dev_set_drvdata(&pdev->dev, i2s); + + i2s->dai = tegra20_i2s_dai_template; + i2s->dai.name = dev_name(&pdev->dev); + + i2s->clk_i2s = clk_get(&pdev->dev, NULL); + if (IS_ERR(i2s->clk_i2s)) { + dev_err(&pdev->dev, "Can't retrieve i2s clock\n"); + ret = PTR_ERR(i2s->clk_i2s); + goto err; + } + + mem = platform_get_resource(pdev, IORESOURCE_MEM, 0); + if (!mem) { + dev_err(&pdev->dev, "No memory resource\n"); + ret = -ENODEV; + goto err_clk_put; + } + + dmareq = platform_get_resource(pdev, IORESOURCE_DMA, 0); + if (!dmareq) { + if (of_property_read_u32_array(pdev->dev.of_node, + "nvidia,dma-request-selector", + of_dma, 2) < 0) { + dev_err(&pdev->dev, "No DMA resource\n"); + ret = -ENODEV; + goto err_clk_put; + } + dma_ch = of_dma[1]; + } else { + dma_ch = dmareq->start; + } + + memregion = devm_request_mem_region(&pdev->dev, mem->start, + resource_size(mem), DRV_NAME); + if (!memregion) { + dev_err(&pdev->dev, "Memory region already claimed\n"); + ret = -EBUSY; + goto err_clk_put; + } + + regs = devm_ioremap(&pdev->dev, mem->start, resource_size(mem)); + if (!regs) { + dev_err(&pdev->dev, "ioremap failed\n"); + ret = -ENOMEM; + goto err_clk_put; + } + + i2s->regmap = devm_regmap_init_mmio(&pdev->dev, regs, + &tegra20_i2s_regmap_config); + if (IS_ERR(i2s->regmap)) { + dev_err(&pdev->dev, "regmap init failed\n"); + ret = PTR_ERR(i2s->regmap); + goto err_clk_put; + } + + i2s->capture_dma_data.addr = mem->start + TEGRA20_I2S_FIFO2; + i2s->capture_dma_data.wrap = 4; + i2s->capture_dma_data.width = 32; + i2s->capture_dma_data.req_sel = dma_ch; + + i2s->playback_dma_data.addr = mem->start + TEGRA20_I2S_FIFO1; + i2s->playback_dma_data.wrap = 4; + i2s->playback_dma_data.width = 32; + i2s->playback_dma_data.req_sel = dma_ch; + + i2s->reg_ctrl = TEGRA20_I2S_CTRL_FIFO_FORMAT_PACKED; + + pm_runtime_enable(&pdev->dev); + if (!pm_runtime_enabled(&pdev->dev)) { + ret = tegra20_i2s_runtime_resume(&pdev->dev); + if (ret) + goto err_pm_disable; + } + + ret = snd_soc_register_dai(&pdev->dev, &i2s->dai); + if (ret) { + dev_err(&pdev->dev, "Could not register DAI: %d\n", ret); + ret = -ENOMEM; + goto err_suspend; + } + + ret = tegra_pcm_platform_register(&pdev->dev); + if (ret) { + dev_err(&pdev->dev, "Could not register PCM: %d\n", ret); + goto err_unregister_dai; + } + + return 0; + +err_unregister_dai: + snd_soc_unregister_dai(&pdev->dev); +err_suspend: + if (!pm_runtime_status_suspended(&pdev->dev)) + tegra20_i2s_runtime_suspend(&pdev->dev); +err_pm_disable: + pm_runtime_disable(&pdev->dev); +err_clk_put: + clk_put(i2s->clk_i2s); +err: + return ret; +} + +static int __devexit tegra20_i2s_platform_remove(struct platform_device *pdev) +{ + struct tegra20_i2s *i2s = dev_get_drvdata(&pdev->dev); + + pm_runtime_disable(&pdev->dev); + if (!pm_runtime_status_suspended(&pdev->dev)) + tegra20_i2s_runtime_suspend(&pdev->dev); + + tegra_pcm_platform_unregister(&pdev->dev); + snd_soc_unregister_dai(&pdev->dev); + + clk_put(i2s->clk_i2s); + + return 0; +} + +static const struct of_device_id tegra20_i2s_of_match[] __devinitconst = { + { .compatible = "nvidia,tegra20-i2s", }, + {}, +}; + +static const struct dev_pm_ops tegra20_i2s_pm_ops __devinitconst = { + SET_RUNTIME_PM_OPS(tegra20_i2s_runtime_suspend, + tegra20_i2s_runtime_resume, NULL) +}; + +static struct platform_driver tegra20_i2s_driver = { + .driver = { + .name = DRV_NAME, + .owner = THIS_MODULE, + .of_match_table = tegra20_i2s_of_match, + .pm = &tegra20_i2s_pm_ops, + }, + .probe = tegra20_i2s_platform_probe, + .remove = __devexit_p(tegra20_i2s_platform_remove), +}; +module_platform_driver(tegra20_i2s_driver); + +MODULE_AUTHOR("Stephen Warren <swarren@nvidia.com>"); +MODULE_DESCRIPTION("Tegra20 I2S ASoC driver"); +MODULE_LICENSE("GPL"); +MODULE_ALIAS("platform:" DRV_NAME); +MODULE_DEVICE_TABLE(of, tegra20_i2s_of_match); diff --git a/sound/soc/tegra/tegra20_i2s.h b/sound/soc/tegra/tegra20_i2s.h new file mode 100644 index 00000000000..a57efc6a597 --- /dev/null +++ b/sound/soc/tegra/tegra20_i2s.h @@ -0,0 +1,164 @@ +/* + * tegra20_i2s.h - Definitions for Tegra20 I2S driver + * + * Author: Stephen Warren <swarren@nvidia.com> + * Copyright (C) 2010,2012 - NVIDIA, Inc. + * + * Based on code copyright/by: + * + * Copyright (c) 2009-2010, NVIDIA Corporation. + * Scott Peterson <speterson@nvidia.com> + * + * Copyright (C) 2010 Google, Inc. + * Iliyan Malchev <malchev@google.com> + * + * This program is free software; you can redistribute it and/or + * modify it under the terms of the GNU General Public License + * version 2 as published by the Free Software Foundation. + * + * This program is distributed in the hope that it will be useful, but + * WITHOUT ANY WARRANTY; without even the implied warranty of + * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU + * General Public License for more details. + * + * You should have received a copy of the GNU General Public License + * along with this program; if not, write to the Free Software + * Foundation, Inc., 51 Franklin St, Fifth Floor, Boston, MA + * 02110-1301 USA + * + */ + +#ifndef __TEGRA20_I2S_H__ +#define __TEGRA20_I2S_H__ + +#include "tegra_pcm.h" + +/* Register offsets from TEGRA20_I2S1_BASE and TEGRA20_I2S2_BASE */ + +#define TEGRA20_I2S_CTRL 0x00 +#define TEGRA20_I2S_STATUS 0x04 +#define TEGRA20_I2S_TIMING 0x08 +#define TEGRA20_I2S_FIFO_SCR 0x0c +#define TEGRA20_I2S_PCM_CTRL 0x10 +#define TEGRA20_I2S_NW_CTRL 0x14 +#define TEGRA20_I2S_TDM_CTRL 0x20 +#define TEGRA20_I2S_TDM_TX_RX_CTRL 0x24 +#define TEGRA20_I2S_FIFO1 0x40 +#define TEGRA20_I2S_FIFO2 0x80 + +/* Fields in TEGRA20_I2S_CTRL */ + +#define TEGRA20_I2S_CTRL_FIFO2_TX_ENABLE (1 << 30) +#define TEGRA20_I2S_CTRL_FIFO1_ENABLE (1 << 29) +#define TEGRA20_I2S_CTRL_FIFO2_ENABLE (1 << 28) +#define TEGRA20_I2S_CTRL_FIFO1_RX_ENABLE (1 << 27) +#define TEGRA20_I2S_CTRL_FIFO_LPBK_ENABLE (1 << 26) +#define TEGRA20_I2S_CTRL_MASTER_ENABLE (1 << 25) + +#define TEGRA20_I2S_LRCK_LEFT_LOW 0 +#define TEGRA20_I2S_LRCK_RIGHT_LOW 1 + +#define TEGRA20_I2S_CTRL_LRCK_SHIFT 24 +#define TEGRA20_I2S_CTRL_LRCK_MASK (1 << TEGRA20_I2S_CTRL_LRCK_SHIFT) +#define TEGRA20_I2S_CTRL_LRCK_L_LOW (TEGRA20_I2S_LRCK_LEFT_LOW << TEGRA20_I2S_CTRL_LRCK_SHIFT) +#define TEGRA20_I2S_CTRL_LRCK_R_LOW (TEGRA20_I2S_LRCK_RIGHT_LOW << TEGRA20_I2S_CTRL_LRCK_SHIFT) + +#define TEGRA20_I2S_BIT_FORMAT_I2S 0 +#define TEGRA20_I2S_BIT_FORMAT_RJM 1 +#define TEGRA20_I2S_BIT_FORMAT_LJM 2 +#define TEGRA20_I2S_BIT_FORMAT_DSP 3 + +#define TEGRA20_I2S_CTRL_BIT_FORMAT_SHIFT 10 +#define TEGRA20_I2S_CTRL_BIT_FORMAT_MASK (3 << TEGRA20_I2S_CTRL_BIT_FORMAT_SHIFT) +#define TEGRA20_I2S_CTRL_BIT_FORMAT_I2S (TEGRA20_I2S_BIT_FORMAT_I2S << TEGRA20_I2S_CTRL_BIT_FORMAT_SHIFT) +#define TEGRA20_I2S_CTRL_BIT_FORMAT_RJM (TEGRA20_I2S_BIT_FORMAT_RJM << TEGRA20_I2S_CTRL_BIT_FORMAT_SHIFT) +#define TEGRA20_I2S_CTRL_BIT_FORMAT_LJM (TEGRA20_I2S_BIT_FORMAT_LJM << TEGRA20_I2S_CTRL_BIT_FORMAT_SHIFT) +#define TEGRA20_I2S_CTRL_BIT_FORMAT_DSP (TEGRA20_I2S_BIT_FORMAT_DSP << TEGRA20_I2S_CTRL_BIT_FORMAT_SHIFT) + +#define TEGRA20_I2S_BIT_SIZE_16 0 +#define TEGRA20_I2S_BIT_SIZE_20 1 +#define TEGRA20_I2S_BIT_SIZE_24 2 +#define TEGRA20_I2S_BIT_SIZE_32 3 + +#define TEGRA20_I2S_CTRL_BIT_SIZE_SHIFT 8 +#define TEGRA20_I2S_CTRL_BIT_SIZE_MASK (3 << TEGRA20_I2S_CTRL_BIT_SIZE_SHIFT) +#define TEGRA20_I2S_CTRL_BIT_SIZE_16 (TEGRA20_I2S_BIT_SIZE_16 << TEGRA20_I2S_CTRL_BIT_SIZE_SHIFT) +#define TEGRA20_I2S_CTRL_BIT_SIZE_20 (TEGRA20_I2S_BIT_SIZE_20 << TEGRA20_I2S_CTRL_BIT_SIZE_SHIFT) +#define TEGRA20_I2S_CTRL_BIT_SIZE_24 (TEGRA20_I2S_BIT_SIZE_24 << TEGRA20_I2S_CTRL_BIT_SIZE_SHIFT) +#define TEGRA20_I2S_CTRL_BIT_SIZE_32 (TEGRA20_I2S_BIT_SIZE_32 << TEGRA20_I2S_CTRL_BIT_SIZE_SHIFT) + +#define TEGRA20_I2S_FIFO_16_LSB 0 +#define TEGRA20_I2S_FIFO_20_LSB 1 +#define TEGRA20_I2S_FIFO_24_LSB 2 +#define TEGRA20_I2S_FIFO_32 3 +#define TEGRA20_I2S_FIFO_PACKED 7 + +#define TEGRA20_I2S_CTRL_FIFO_FORMAT_SHIFT 4 +#define TEGRA20_I2S_CTRL_FIFO_FORMAT_MASK (7 << TEGRA20_I2S_CTRL_FIFO_FORMAT_SHIFT) +#define TEGRA20_I2S_CTRL_FIFO_FORMAT_16_LSB (TEGRA20_I2S_FIFO_16_LSB << TEGRA20_I2S_CTRL_FIFO_FORMAT_SHIFT) +#define TEGRA20_I2S_CTRL_FIFO_FORMAT_20_LSB (TEGRA20_I2S_FIFO_20_LSB << TEGRA20_I2S_CTRL_FIFO_FORMAT_SHIFT) +#define TEGRA20_I2S_CTRL_FIFO_FORMAT_24_LSB (TEGRA20_I2S_FIFO_24_LSB << TEGRA20_I2S_CTRL_FIFO_FORMAT_SHIFT) +#define TEGRA20_I2S_CTRL_FIFO_FORMAT_32 (TEGRA20_I2S_FIFO_32 << TEGRA20_I2S_CTRL_FIFO_FORMAT_SHIFT) +#define TEGRA20_I2S_CTRL_FIFO_FORMAT_PACKED (TEGRA20_I2S_FIFO_PACKED << TEGRA20_I2S_CTRL_FIFO_FORMAT_SHIFT) + +#define TEGRA20_I2S_CTRL_IE_FIFO1_ERR (1 << 3) +#define TEGRA20_I2S_CTRL_IE_FIFO2_ERR (1 << 2) +#define TEGRA20_I2S_CTRL_QE_FIFO1 (1 << 1) +#define TEGRA20_I2S_CTRL_QE_FIFO2 (1 << 0) + +/* Fields in TEGRA20_I2S_STATUS */ + +#define TEGRA20_I2S_STATUS_FIFO1_RDY (1 << 31) +#define TEGRA20_I2S_STATUS_FIFO2_RDY (1 << 30) +#define TEGRA20_I2S_STATUS_FIFO1_BSY (1 << 29) +#define TEGRA20_I2S_STATUS_FIFO2_BSY (1 << 28) +#define TEGRA20_I2S_STATUS_FIFO1_ERR (1 << 3) +#define TEGRA20_I2S_STATUS_FIFO2_ERR (1 << 2) +#define TEGRA20_I2S_STATUS_QS_FIFO1 (1 << 1) +#define TEGRA20_I2S_STATUS_QS_FIFO2 (1 << 0) + +/* Fields in TEGRA20_I2S_TIMING */ + +#define TEGRA20_I2S_TIMING_NON_SYM_ENABLE (1 << 12) +#define TEGRA20_I2S_TIMING_CHANNEL_BIT_COUNT_SHIFT 0 +#define TEGRA20_I2S_TIMING_CHANNEL_BIT_COUNT_MASK_US 0x7fff +#define TEGRA20_I2S_TIMING_CHANNEL_BIT_COUNT_MASK (TEGRA20_I2S_TIMING_CHANNEL_BIT_COUNT_MASK_US << TEGRA20_I2S_TIMING_CHANNEL_BIT_COUNT_SHIFT) + +/* Fields in TEGRA20_I2S_FIFO_SCR */ + +#define TEGRA20_I2S_FIFO_SCR_FIFO2_FULL_EMPTY_COUNT_SHIFT 24 +#define TEGRA20_I2S_FIFO_SCR_FIFO1_FULL_EMPTY_COUNT_SHIFT 16 +#define TEGRA20_I2S_FIFO_SCR_FIFO_FULL_EMPTY_COUNT_MASK 0x3f + +#define TEGRA20_I2S_FIFO_SCR_FIFO2_CLR (1 << 12) +#define TEGRA20_I2S_FIFO_SCR_FIFO1_CLR (1 << 8) + +#define TEGRA20_I2S_FIFO_ATN_LVL_ONE_SLOT 0 +#define TEGRA20_I2S_FIFO_ATN_LVL_FOUR_SLOTS 1 +#define TEGRA20_I2S_FIFO_ATN_LVL_EIGHT_SLOTS 2 +#define TEGRA20_I2S_FIFO_ATN_LVL_TWELVE_SLOTS 3 + +#define TEGRA20_I2S_FIFO_SCR_FIFO2_ATN_LVL_SHIFT 4 +#define TEGRA20_I2S_FIFO_SCR_FIFO2_ATN_LVL_MASK (3 << TEGRA20_I2S_FIFO_SCR_FIFO2_ATN_LVL_SHIFT) +#define TEGRA20_I2S_FIFO_SCR_FIFO2_ATN_LVL_ONE_SLOT (TEGRA20_I2S_FIFO_ATN_LVL_ONE_SLOT << TEGRA20_I2S_FIFO_SCR_FIFO2_ATN_LVL_SHIFT) +#define TEGRA20_I2S_FIFO_SCR_FIFO2_ATN_LVL_FOUR_SLOTS (TEGRA20_I2S_FIFO_ATN_LVL_FOUR_SLOTS << TEGRA20_I2S_FIFO_SCR_FIFO2_ATN_LVL_SHIFT) +#define TEGRA20_I2S_FIFO_SCR_FIFO2_ATN_LVL_EIGHT_SLOTS (TEGRA20_I2S_FIFO_ATN_LVL_EIGHT_SLOTS << TEGRA20_I2S_FIFO_SCR_FIFO2_ATN_LVL_SHIFT) +#define TEGRA20_I2S_FIFO_SCR_FIFO2_ATN_LVL_TWELVE_SLOTS (TEGRA20_I2S_FIFO_ATN_LVL_TWELVE_SLOTS << TEGRA20_I2S_FIFO_SCR_FIFO2_ATN_LVL_SHIFT) + +#define TEGRA20_I2S_FIFO_SCR_FIFO1_ATN_LVL_SHIFT 0 +#define TEGRA20_I2S_FIFO_SCR_FIFO1_ATN_LVL_MASK (3 << TEGRA20_I2S_FIFO_SCR_FIFO1_ATN_LVL_SHIFT) +#define TEGRA20_I2S_FIFO_SCR_FIFO1_ATN_LVL_ONE_SLOT (TEGRA20_I2S_FIFO_ATN_LVL_ONE_SLOT << TEGRA20_I2S_FIFO_SCR_FIFO1_ATN_LVL_SHIFT) +#define TEGRA20_I2S_FIFO_SCR_FIFO1_ATN_LVL_FOUR_SLOTS (TEGRA20_I2S_FIFO_ATN_LVL_FOUR_SLOTS << TEGRA20_I2S_FIFO_SCR_FIFO1_ATN_LVL_SHIFT) +#define TEGRA20_I2S_FIFO_SCR_FIFO1_ATN_LVL_EIGHT_SLOTS (TEGRA20_I2S_FIFO_ATN_LVL_EIGHT_SLOTS << TEGRA20_I2S_FIFO_SCR_FIFO1_ATN_LVL_SHIFT) +#define TEGRA20_I2S_FIFO_SCR_FIFO1_ATN_LVL_TWELVE_SLOTS (TEGRA20_I2S_FIFO_ATN_LVL_TWELVE_SLOTS << TEGRA20_I2S_FIFO_SCR_FIFO1_ATN_LVL_SHIFT) + +struct tegra20_i2s { + struct snd_soc_dai_driver dai; + struct clk *clk_i2s; + struct tegra_pcm_dma_params capture_dma_data; + struct tegra_pcm_dma_params playback_dma_data; + struct regmap *regmap; + u32 reg_ctrl; +}; + +#endif diff --git a/sound/soc/tegra/tegra20_spdif.c b/sound/soc/tegra/tegra20_spdif.c new file mode 100644 index 00000000000..f9b57418bd0 --- /dev/null +++ b/sound/soc/tegra/tegra20_spdif.c @@ -0,0 +1,404 @@ +/* + * tegra20_spdif.c - Tegra20 SPDIF driver + * + * Author: Stephen Warren <swarren@nvidia.com> + * Copyright (C) 2011-2012 - NVIDIA, Inc. + * + * This program is free software; you can redistribute it and/or + * modify it under the terms of the GNU General Public License + * version 2 as published by the Free Software Foundation. + * + * This program is distributed in the hope that it will be useful, but + * WITHOUT ANY WARRANTY; without even the implied warranty of + * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU + * General Public License for more details. + * + * You should have received a copy of the GNU General Public License + * along with this program; if not, write to the Free Software + * Foundation, Inc., 51 Franklin St, Fifth Floor, Boston, MA + * 02110-1301 USA + * + */ + +#include <linux/clk.h> +#include <linux/device.h> +#include <linux/io.h> +#include <linux/module.h> +#include <linux/platform_device.h> +#include <linux/pm_runtime.h> +#include <linux/regmap.h> +#include <linux/slab.h> +#include <sound/core.h> +#include <sound/pcm.h> +#include <sound/pcm_params.h> +#include <sound/soc.h> + +#include "tegra20_spdif.h" + +#define DRV_NAME "tegra20-spdif" + +static inline void tegra20_spdif_write(struct tegra20_spdif *spdif, u32 reg, + u32 val) +{ + regmap_write(spdif->regmap, reg, val); +} + +static inline u32 tegra20_spdif_read(struct tegra20_spdif *spdif, u32 reg) +{ + u32 val; + regmap_read(spdif->regmap, reg, &val); + return val; +} + +static int tegra20_spdif_runtime_suspend(struct device *dev) +{ + struct tegra20_spdif *spdif = dev_get_drvdata(dev); + + clk_disable(spdif->clk_spdif_out); + + return 0; +} + +static int tegra20_spdif_runtime_resume(struct device *dev) +{ + struct tegra20_spdif *spdif = dev_get_drvdata(dev); + int ret; + + ret = clk_enable(spdif->clk_spdif_out); + if (ret) { + dev_err(dev, "clk_enable failed: %d\n", ret); + return ret; + } + + return 0; +} + +static int tegra20_spdif_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params, + struct snd_soc_dai *dai) +{ + struct device *dev = substream->pcm->card->dev; + struct tegra20_spdif *spdif = snd_soc_dai_get_drvdata(dai); + int ret, spdifclock; + + spdif->reg_ctrl &= ~TEGRA20_SPDIF_CTRL_PACK; + spdif->reg_ctrl &= ~TEGRA20_SPDIF_CTRL_BIT_MODE_MASK; + switch (params_format(params)) { + case SNDRV_PCM_FORMAT_S16_LE: + spdif->reg_ctrl |= TEGRA20_SPDIF_CTRL_PACK; + spdif->reg_ctrl |= TEGRA20_SPDIF_CTRL_BIT_MODE_16BIT; + break; + default: + return -EINVAL; + } + + switch (params_rate(params)) { + case 32000: + spdifclock = 4096000; + break; + case 44100: + spdifclock = 5644800; + break; + case 48000: + spdifclock = 6144000; + break; + case 88200: + spdifclock = 11289600; + break; + case 96000: + spdifclock = 12288000; + break; + case 176400: + spdifclock = 22579200; + break; + case 192000: + spdifclock = 24576000; + break; + default: + return -EINVAL; + } + + ret = clk_set_rate(spdif->clk_spdif_out, spdifclock); + if (ret) { + dev_err(dev, "Can't set SPDIF clock rate: %d\n", ret); + return ret; + } + + return 0; +} + +static void tegra20_spdif_start_playback(struct tegra20_spdif *spdif) +{ + spdif->reg_ctrl |= TEGRA20_SPDIF_CTRL_TX_EN; + tegra20_spdif_write(spdif, TEGRA20_SPDIF_CTRL, spdif->reg_ctrl); +} + +static void tegra20_spdif_stop_playback(struct tegra20_spdif *spdif) +{ + spdif->reg_ctrl &= ~TEGRA20_SPDIF_CTRL_TX_EN; + tegra20_spdif_write(spdif, TEGRA20_SPDIF_CTRL, spdif->reg_ctrl); +} + +static int tegra20_spdif_trigger(struct snd_pcm_substream *substream, int cmd, + struct snd_soc_dai *dai) +{ + struct tegra20_spdif *spdif = snd_soc_dai_get_drvdata(dai); + + switch (cmd) { + case SNDRV_PCM_TRIGGER_START: + case SNDRV_PCM_TRIGGER_PAUSE_RELEASE: + case SNDRV_PCM_TRIGGER_RESUME: + tegra20_spdif_start_playback(spdif); + break; + case SNDRV_PCM_TRIGGER_STOP: + case SNDRV_PCM_TRIGGER_PAUSE_PUSH: + case SNDRV_PCM_TRIGGER_SUSPEND: + tegra20_spdif_stop_playback(spdif); + break; + default: + return -EINVAL; + } + + return 0; +} + +static int tegra20_spdif_probe(struct snd_soc_dai *dai) +{ + struct tegra20_spdif *spdif = snd_soc_dai_get_drvdata(dai); + + dai->capture_dma_data = NULL; + dai->playback_dma_data = &spdif->playback_dma_data; + + return 0; +} + +static const struct snd_soc_dai_ops tegra20_spdif_dai_ops = { + .hw_params = tegra20_spdif_hw_params, + .trigger = tegra20_spdif_trigger, +}; + +static struct snd_soc_dai_driver tegra20_spdif_dai = { + .name = DRV_NAME, + .probe = tegra20_spdif_probe, + .playback = { + .channels_min = 2, + .channels_max = 2, + .rates = SNDRV_PCM_RATE_32000 | SNDRV_PCM_RATE_44100 | + SNDRV_PCM_RATE_48000, + .formats = SNDRV_PCM_FMTBIT_S16_LE, + }, + .ops = &tegra20_spdif_dai_ops, +}; + +static bool tegra20_spdif_wr_rd_reg(struct device *dev, unsigned int reg) +{ + switch (reg) { + case TEGRA20_SPDIF_CTRL: + case TEGRA20_SPDIF_STATUS: + case TEGRA20_SPDIF_STROBE_CTRL: + case TEGRA20_SPDIF_DATA_FIFO_CSR: + case TEGRA20_SPDIF_DATA_OUT: + case TEGRA20_SPDIF_DATA_IN: + case TEGRA20_SPDIF_CH_STA_RX_A: + case TEGRA20_SPDIF_CH_STA_RX_B: + case TEGRA20_SPDIF_CH_STA_RX_C: + case TEGRA20_SPDIF_CH_STA_RX_D: + case TEGRA20_SPDIF_CH_STA_RX_E: + case TEGRA20_SPDIF_CH_STA_RX_F: + case TEGRA20_SPDIF_CH_STA_TX_A: + case TEGRA20_SPDIF_CH_STA_TX_B: + case TEGRA20_SPDIF_CH_STA_TX_C: + case TEGRA20_SPDIF_CH_STA_TX_D: + case TEGRA20_SPDIF_CH_STA_TX_E: + case TEGRA20_SPDIF_CH_STA_TX_F: + case TEGRA20_SPDIF_USR_STA_RX_A: + case TEGRA20_SPDIF_USR_DAT_TX_A: + return true; + default: + return false; + }; +} + +static bool tegra20_spdif_volatile_reg(struct device *dev, unsigned int reg) +{ + switch (reg) { + case TEGRA20_SPDIF_STATUS: + case TEGRA20_SPDIF_DATA_FIFO_CSR: + case TEGRA20_SPDIF_DATA_OUT: + case TEGRA20_SPDIF_DATA_IN: + case TEGRA20_SPDIF_CH_STA_RX_A: + case TEGRA20_SPDIF_CH_STA_RX_B: + case TEGRA20_SPDIF_CH_STA_RX_C: + case TEGRA20_SPDIF_CH_STA_RX_D: + case TEGRA20_SPDIF_CH_STA_RX_E: + case TEGRA20_SPDIF_CH_STA_RX_F: + case TEGRA20_SPDIF_USR_STA_RX_A: + case TEGRA20_SPDIF_USR_DAT_TX_A: + return true; + default: + return false; + }; +} + +static bool tegra20_spdif_precious_reg(struct device *dev, unsigned int reg) +{ + switch (reg) { + case TEGRA20_SPDIF_DATA_OUT: + case TEGRA20_SPDIF_DATA_IN: + case TEGRA20_SPDIF_USR_STA_RX_A: + case TEGRA20_SPDIF_USR_DAT_TX_A: + return true; + default: + return false; + }; +} + +static const struct regmap_config tegra20_spdif_regmap_config = { + .reg_bits = 32, + .reg_stride = 4, + .val_bits = 32, + .max_register = TEGRA20_SPDIF_USR_DAT_TX_A, + .writeable_reg = tegra20_spdif_wr_rd_reg, + .readable_reg = tegra20_spdif_wr_rd_reg, + .volatile_reg = tegra20_spdif_volatile_reg, + .precious_reg = tegra20_spdif_precious_reg, + .cache_type = REGCACHE_RBTREE, +}; + +static __devinit int tegra20_spdif_platform_probe(struct platform_device *pdev) +{ + struct tegra20_spdif *spdif; + struct resource *mem, *memregion, *dmareq; + void __iomem *regs; + int ret; + + spdif = devm_kzalloc(&pdev->dev, sizeof(struct tegra20_spdif), + GFP_KERNEL); + if (!spdif) { + dev_err(&pdev->dev, "Can't allocate tegra20_spdif\n"); + ret = -ENOMEM; + goto err; + } + dev_set_drvdata(&pdev->dev, spdif); + + spdif->clk_spdif_out = clk_get(&pdev->dev, "spdif_out"); + if (IS_ERR(spdif->clk_spdif_out)) { + pr_err("Can't retrieve spdif clock\n"); + ret = PTR_ERR(spdif->clk_spdif_out); + goto err; + } + + mem = platform_get_resource(pdev, IORESOURCE_MEM, 0); + if (!mem) { + dev_err(&pdev->dev, "No memory resource\n"); + ret = -ENODEV; + goto err_clk_put; + } + + dmareq = platform_get_resource(pdev, IORESOURCE_DMA, 0); + if (!dmareq) { + dev_err(&pdev->dev, "No DMA resource\n"); + ret = -ENODEV; + goto err_clk_put; + } + + memregion = devm_request_mem_region(&pdev->dev, mem->start, + resource_size(mem), DRV_NAME); + if (!memregion) { + dev_err(&pdev->dev, "Memory region already claimed\n"); + ret = -EBUSY; + goto err_clk_put; + } + + regs = devm_ioremap(&pdev->dev, mem->start, resource_size(mem)); + if (!regs) { + dev_err(&pdev->dev, "ioremap failed\n"); + ret = -ENOMEM; + goto err_clk_put; + } + + spdif->regmap = devm_regmap_init_mmio(&pdev->dev, regs, + &tegra20_spdif_regmap_config); + if (IS_ERR(spdif->regmap)) { + dev_err(&pdev->dev, "regmap init failed\n"); + ret = PTR_ERR(spdif->regmap); + goto err_clk_put; + } + + spdif->playback_dma_data.addr = mem->start + TEGRA20_SPDIF_DATA_OUT; + spdif->playback_dma_data.wrap = 4; + spdif->playback_dma_data.width = 32; + spdif->playback_dma_data.req_sel = dmareq->start; + + pm_runtime_enable(&pdev->dev); + if (!pm_runtime_enabled(&pdev->dev)) { + ret = tegra20_spdif_runtime_resume(&pdev->dev); + if (ret) + goto err_pm_disable; + } + + ret = snd_soc_register_dai(&pdev->dev, &tegra20_spdif_dai); + if (ret) { + dev_err(&pdev->dev, "Could not register DAI: %d\n", ret); + ret = -ENOMEM; + goto err_suspend; + } + + ret = tegra_pcm_platform_register(&pdev->dev); + if (ret) { + dev_err(&pdev->dev, "Could not register PCM: %d\n", ret); + goto err_unregister_dai; + } + + return 0; + +err_unregister_dai: + snd_soc_unregister_dai(&pdev->dev); +err_suspend: + if (!pm_runtime_status_suspended(&pdev->dev)) + tegra20_spdif_runtime_suspend(&pdev->dev); +err_pm_disable: + pm_runtime_disable(&pdev->dev); +err_clk_put: + clk_put(spdif->clk_spdif_out); +err: + return ret; +} + +static int __devexit tegra20_spdif_platform_remove(struct platform_device *pdev) +{ + struct tegra20_spdif *spdif = dev_get_drvdata(&pdev->dev); + + pm_runtime_disable(&pdev->dev); + if (!pm_runtime_status_suspended(&pdev->dev)) + tegra20_spdif_runtime_suspend(&pdev->dev); + + tegra_pcm_platform_unregister(&pdev->dev); + snd_soc_unregister_dai(&pdev->dev); + + clk_put(spdif->clk_spdif_out); + + return 0; +} + +static const struct dev_pm_ops tegra20_spdif_pm_ops __devinitconst = { + SET_RUNTIME_PM_OPS(tegra20_spdif_runtime_suspend, + tegra20_spdif_runtime_resume, NULL) +}; + +static struct platform_driver tegra20_spdif_driver = { + .driver = { + .name = DRV_NAME, + .owner = THIS_MODULE, + .pm = &tegra20_spdif_pm_ops, + }, + .probe = tegra20_spdif_platform_probe, + .remove = __devexit_p(tegra20_spdif_platform_remove), +}; + +module_platform_driver(tegra20_spdif_driver); + +MODULE_AUTHOR("Stephen Warren <swarren@nvidia.com>"); +MODULE_DESCRIPTION("Tegra20 SPDIF ASoC driver"); +MODULE_LICENSE("GPL"); +MODULE_ALIAS("platform:" DRV_NAME); diff --git a/sound/soc/tegra/tegra20_spdif.h b/sound/soc/tegra/tegra20_spdif.h new file mode 100644 index 00000000000..ed756527efe --- /dev/null +++ b/sound/soc/tegra/tegra20_spdif.h @@ -0,0 +1,471 @@ +/* + * tegra20_spdif.h - Definitions for Tegra20 SPDIF driver + * + * Author: Stephen Warren <swarren@nvidia.com> + * Copyright (C) 2011 - NVIDIA, Inc. + * + * Based on code copyright/by: + * Copyright (c) 2008-2009, NVIDIA Corporation + * + * This program is free software; you can redistribute it and/or + * modify it under the terms of the GNU General Public License + * version 2 as published by the Free Software Foundation. + * + * This program is distributed in the hope that it will be useful, but + * WITHOUT ANY WARRANTY; without even the implied warranty of + * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU + * General Public License for more details. + * + * You should have received a copy of the GNU General Public License + * along with this program; if not, write to the Free Software + * Foundation, Inc., 51 Franklin St, Fifth Floor, Boston, MA + * 02110-1301 USA + * + */ + +#ifndef __TEGRA20_SPDIF_H__ +#define __TEGRA20_SPDIF_H__ + +#include "tegra_pcm.h" + +/* Offsets from TEGRA20_SPDIF_BASE */ + +#define TEGRA20_SPDIF_CTRL 0x0 +#define TEGRA20_SPDIF_STATUS 0x4 +#define TEGRA20_SPDIF_STROBE_CTRL 0x8 +#define TEGRA20_SPDIF_DATA_FIFO_CSR 0x0C +#define TEGRA20_SPDIF_DATA_OUT 0x40 +#define TEGRA20_SPDIF_DATA_IN 0x80 +#define TEGRA20_SPDIF_CH_STA_RX_A 0x100 +#define TEGRA20_SPDIF_CH_STA_RX_B 0x104 +#define TEGRA20_SPDIF_CH_STA_RX_C 0x108 +#define TEGRA20_SPDIF_CH_STA_RX_D 0x10C +#define TEGRA20_SPDIF_CH_STA_RX_E 0x110 +#define TEGRA20_SPDIF_CH_STA_RX_F 0x114 +#define TEGRA20_SPDIF_CH_STA_TX_A 0x140 +#define TEGRA20_SPDIF_CH_STA_TX_B 0x144 +#define TEGRA20_SPDIF_CH_STA_TX_C 0x148 +#define TEGRA20_SPDIF_CH_STA_TX_D 0x14C +#define TEGRA20_SPDIF_CH_STA_TX_E 0x150 +#define TEGRA20_SPDIF_CH_STA_TX_F 0x154 +#define TEGRA20_SPDIF_USR_STA_RX_A 0x180 +#define TEGRA20_SPDIF_USR_DAT_TX_A 0x1C0 + +/* Fields in TEGRA20_SPDIF_CTRL */ + +/* Start capturing from 0=right, 1=left channel */ +#define TEGRA20_SPDIF_CTRL_CAP_LC (1 << 30) + +/* SPDIF receiver(RX) enable */ +#define TEGRA20_SPDIF_CTRL_RX_EN (1 << 29) + +/* SPDIF Transmitter(TX) enable */ +#define TEGRA20_SPDIF_CTRL_TX_EN (1 << 28) + +/* Transmit Channel status */ +#define TEGRA20_SPDIF_CTRL_TC_EN (1 << 27) + +/* Transmit user Data */ +#define TEGRA20_SPDIF_CTRL_TU_EN (1 << 26) + +/* Interrupt on transmit error */ +#define TEGRA20_SPDIF_CTRL_IE_TXE (1 << 25) + +/* Interrupt on receive error */ +#define TEGRA20_SPDIF_CTRL_IE_RXE (1 << 24) + +/* Interrupt on invalid preamble */ +#define TEGRA20_SPDIF_CTRL_IE_P (1 << 23) + +/* Interrupt on "B" preamble */ +#define TEGRA20_SPDIF_CTRL_IE_B (1 << 22) + +/* Interrupt when block of channel status received */ +#define TEGRA20_SPDIF_CTRL_IE_C (1 << 21) + +/* Interrupt when a valid information unit (IU) is received */ +#define TEGRA20_SPDIF_CTRL_IE_U (1 << 20) + +/* Interrupt when RX user FIFO attention level is reached */ +#define TEGRA20_SPDIF_CTRL_QE_RU (1 << 19) + +/* Interrupt when TX user FIFO attention level is reached */ +#define TEGRA20_SPDIF_CTRL_QE_TU (1 << 18) + +/* Interrupt when RX data FIFO attention level is reached */ +#define TEGRA20_SPDIF_CTRL_QE_RX (1 << 17) + +/* Interrupt when TX data FIFO attention level is reached */ +#define TEGRA20_SPDIF_CTRL_QE_TX (1 << 16) + +/* Loopback test mode enable */ +#define TEGRA20_SPDIF_CTRL_LBK_EN (1 << 15) + +/* + * Pack data mode: + * 0 = Single data (16 bit needs to be padded to match the + * interface data bit size). + * 1 = Packeted left/right channel data into a single word. + */ +#define TEGRA20_SPDIF_CTRL_PACK (1 << 14) + +/* + * 00 = 16bit data + * 01 = 20bit data + * 10 = 24bit data + * 11 = raw data + */ +#define TEGRA20_SPDIF_BIT_MODE_16BIT 0 +#define TEGRA20_SPDIF_BIT_MODE_20BIT 1 +#define TEGRA20_SPDIF_BIT_MODE_24BIT 2 +#define TEGRA20_SPDIF_BIT_MODE_RAW 3 + +#define TEGRA20_SPDIF_CTRL_BIT_MODE_SHIFT 12 +#define TEGRA20_SPDIF_CTRL_BIT_MODE_MASK (3 << TEGRA20_SPDIF_CTRL_BIT_MODE_SHIFT) +#define TEGRA20_SPDIF_CTRL_BIT_MODE_16BIT (TEGRA20_SPDIF_BIT_MODE_16BIT << TEGRA20_SPDIF_CTRL_BIT_MODE_SHIFT) +#define TEGRA20_SPDIF_CTRL_BIT_MODE_20BIT (TEGRA20_SPDIF_BIT_MODE_20BIT << TEGRA20_SPDIF_CTRL_BIT_MODE_SHIFT) +#define TEGRA20_SPDIF_CTRL_BIT_MODE_24BIT (TEGRA20_SPDIF_BIT_MODE_24BIT << TEGRA20_SPDIF_CTRL_BIT_MODE_SHIFT) +#define TEGRA20_SPDIF_CTRL_BIT_MODE_RAW (TEGRA20_SPDIF_BIT_MODE_RAW << TEGRA20_SPDIF_CTRL_BIT_MODE_SHIFT) + +/* Fields in TEGRA20_SPDIF_STATUS */ + +/* + * Note: IS_P, IS_B, IS_C, and IS_U are sticky bits. Software must + * write a 1 to the corresponding bit location to clear the status. + */ + +/* + * Receiver(RX) shifter is busy receiving data. + * This bit is asserted when the receiver first locked onto the + * preamble of the data stream after RX_EN is asserted. This bit is + * deasserted when either, + * (a) the end of a frame is reached after RX_EN is deeasserted, or + * (b) the SPDIF data stream becomes inactive. + */ +#define TEGRA20_SPDIF_STATUS_RX_BSY (1 << 29) + +/* + * Transmitter(TX) shifter is busy transmitting data. + * This bit is asserted when TX_EN is asserted. + * This bit is deasserted when the end of a frame is reached after + * TX_EN is deasserted. + */ +#define TEGRA20_SPDIF_STATUS_TX_BSY (1 << 28) + +/* + * TX is busy shifting out channel status. + * This bit is asserted when both TX_EN and TC_EN are asserted and + * data from CH_STA_TX_A register is loaded into the internal shifter. + * This bit is deasserted when either, + * (a) the end of a frame is reached after TX_EN is deasserted, or + * (b) CH_STA_TX_F register is loaded into the internal shifter. + */ +#define TEGRA20_SPDIF_STATUS_TC_BSY (1 << 27) + +/* + * TX User data FIFO busy. + * This bit is asserted when TX_EN and TXU_EN are asserted and + * there's data in the TX user FIFO. This bit is deassert when either, + * (a) the end of a frame is reached after TX_EN is deasserted, or + * (b) there's no data left in the TX user FIFO. + */ +#define TEGRA20_SPDIF_STATUS_TU_BSY (1 << 26) + +/* TX FIFO Underrun error status */ +#define TEGRA20_SPDIF_STATUS_TX_ERR (1 << 25) + +/* RX FIFO Overrun error status */ +#define TEGRA20_SPDIF_STATUS_RX_ERR (1 << 24) + +/* Preamble status: 0=Preamble OK, 1=bad/missing preamble */ +#define TEGRA20_SPDIF_STATUS_IS_P (1 << 23) + +/* B-preamble detection status: 0=not detected, 1=B-preamble detected */ +#define TEGRA20_SPDIF_STATUS_IS_B (1 << 22) + +/* + * RX channel block data receive status: + * 0=entire block not recieved yet. + * 1=received entire block of channel status, + */ +#define TEGRA20_SPDIF_STATUS_IS_C (1 << 21) + +/* RX User Data Valid flag: 1=valid IU detected, 0 = no IU detected. */ +#define TEGRA20_SPDIF_STATUS_IS_U (1 << 20) + +/* + * RX User FIFO Status: + * 1=attention level reached, 0=attention level not reached. + */ +#define TEGRA20_SPDIF_STATUS_QS_RU (1 << 19) + +/* + * TX User FIFO Status: + * 1=attention level reached, 0=attention level not reached. + */ +#define TEGRA20_SPDIF_STATUS_QS_TU (1 << 18) + +/* + * RX Data FIFO Status: + * 1=attention level reached, 0=attention level not reached. + */ +#define TEGRA20_SPDIF_STATUS_QS_RX (1 << 17) + +/* + * TX Data FIFO Status: + * 1=attention level reached, 0=attention level not reached. + */ +#define TEGRA20_SPDIF_STATUS_QS_TX (1 << 16) + +/* Fields in TEGRA20_SPDIF_STROBE_CTRL */ + +/* + * Indicates the approximate number of detected SPDIFIN clocks within a + * bi-phase period. + */ +#define TEGRA20_SPDIF_STROBE_CTRL_PERIOD_SHIFT 16 +#define TEGRA20_SPDIF_STROBE_CTRL_PERIOD_MASK (0xff << TEGRA20_SPDIF_STROBE_CTRL_PERIOD_SHIFT) + +/* Data strobe mode: 0=Auto-locked 1=Manual locked */ +#define TEGRA20_SPDIF_STROBE_CTRL_STROBE (1 << 15) + +/* + * Manual data strobe time within the bi-phase clock period (in terms of + * the number of over-sampling clocks). + */ +#define TEGRA20_SPDIF_STROBE_CTRL_DATA_STROBES_SHIFT 8 +#define TEGRA20_SPDIF_STROBE_CTRL_DATA_STROBES_MASK (0x1f << TEGRA20_SPDIF_STROBE_CTRL_DATA_STROBES_SHIFT) + +/* + * Manual SPDIFIN bi-phase clock period (in terms of the number of + * over-sampling clocks). + */ +#define TEGRA20_SPDIF_STROBE_CTRL_CLOCK_PERIOD_SHIFT 0 +#define TEGRA20_SPDIF_STROBE_CTRL_CLOCK_PERIOD_MASK (0x3f << TEGRA20_SPDIF_STROBE_CTRL_CLOCK_PERIOD_SHIFT) + +/* Fields in SPDIF_DATA_FIFO_CSR */ + +/* Clear Receiver User FIFO (RX USR.FIFO) */ +#define TEGRA20_SPDIF_DATA_FIFO_CSR_RU_CLR (1 << 31) + +#define TEGRA20_SPDIF_FIFO_ATN_LVL_U_ONE_SLOT 0 +#define TEGRA20_SPDIF_FIFO_ATN_LVL_U_TWO_SLOTS 1 +#define TEGRA20_SPDIF_FIFO_ATN_LVL_U_THREE_SLOTS 2 +#define TEGRA20_SPDIF_FIFO_ATN_LVL_U_FOUR_SLOTS 3 + +/* RU FIFO attention level */ +#define TEGRA20_SPDIF_DATA_FIFO_CSR_RU_ATN_LVL_SHIFT 29 +#define TEGRA20_SPDIF_DATA_FIFO_CSR_RU_ATN_LVL_MASK \ + (0x3 << TEGRA20_SPDIF_DATA_FIFO_CSR_RU_ATN_LVL_SHIFT) +#define TEGRA20_SPDIF_DATA_FIFO_CSR_RU_ATN_LVL_RU1_WORD_FULL \ + (TEGRA20_SPDIF_FIFO_ATN_LVL_U_ONE_SLOT << TEGRA20_SPDIF_DATA_FIFO_CSR_RU_ATN_LVL_SHIFT) +#define TEGRA20_SPDIF_DATA_FIFO_CSR_RU_ATN_LVL_RU2_WORD_FULL \ + (TEGRA20_SPDIF_FIFO_ATN_LVL_U_TWO_SLOTS << TEGRA20_SPDIF_DATA_FIFO_CSR_RU_ATN_LVL_SHIFT) +#define TEGRA20_SPDIF_DATA_FIFO_CSR_RU_ATN_LVL_RU3_WORD_FULL \ + (TEGRA20_SPDIF_FIFO_ATN_LVL_U_THREE_SLOTS << TEGRA20_SPDIF_DATA_FIFO_CSR_RU_ATN_LVL_SHIFT) +#define TEGRA20_SPDIF_DATA_FIFO_CSR_RU_ATN_LVL_RU4_WORD_FULL \ + (TEGRA20_SPDIF_FIFO_ATN_LVL_U_FOUR_SLOTS << TEGRA20_SPDIF_DATA_FIFO_CSR_RU_ATN_LVL_SHIFT) + +/* Number of RX USR.FIFO levels with valid data. */ +#define TEGRA20_SPDIF_DATA_FIFO_CSR_RU_FULL_COUNT_SHIFT 24 +#define TEGRA20_SPDIF_DATA_FIFO_CSR_RU_FULL_COUNT_MASK (0x1f << TEGRA20_SPDIF_DATA_FIFO_CSR_RU_FULL_COUNT_SHIFT) + +/* Clear Transmitter User FIFO (TX USR.FIFO) */ +#define TEGRA20_SPDIF_DATA_FIFO_CSR_TU_CLR (1 << 23) + +/* TU FIFO attention level */ +#define TEGRA20_SPDIF_DATA_FIFO_CSR_TU_ATN_LVL_SHIFT 21 +#define TEGRA20_SPDIF_DATA_FIFO_CSR_TU_ATN_LVL_MASK \ + (0x3 << TEGRA20_SPDIF_DATA_FIFO_CSR_TU_ATN_LVL_SHIFT) +#define TEGRA20_SPDIF_DATA_FIFO_CSR_TU_ATN_LVL_TU1_WORD_FULL \ + (TEGRA20_SPDIF_FIFO_ATN_LVL_U_ONE_SLOT << TEGRA20_SPDIF_DATA_FIFO_CSR_TU_ATN_LVL_SHIFT) +#define TEGRA20_SPDIF_DATA_FIFO_CSR_TU_ATN_LVL_TU2_WORD_FULL \ + (TEGRA20_SPDIF_FIFO_ATN_LVL_U_TWO_SLOTS << TEGRA20_SPDIF_DATA_FIFO_CSR_TU_ATN_LVL_SHIFT) +#define TEGRA20_SPDIF_DATA_FIFO_CSR_TU_ATN_LVL_TU3_WORD_FULL \ + (TEGRA20_SPDIF_FIFO_ATN_LVL_U_THREE_SLOTS << TEGRA20_SPDIF_DATA_FIFO_CSR_TU_ATN_LVL_SHIFT) +#define TEGRA20_SPDIF_DATA_FIFO_CSR_TU_ATN_LVL_TU4_WORD_FULL \ + (TEGRA20_SPDIF_FIFO_ATN_LVL_U_FOUR_SLOTS << TEGRA20_SPDIF_DATA_FIFO_CSR_TU_ATN_LVL_SHIFT) + +/* Number of TX USR.FIFO levels that could be filled. */ +#define TEGRA20_SPDIF_DATA_FIFO_CSR_TU_EMPTY_COUNT_SHIFT 16 +#define TEGRA20_SPDIF_DATA_FIFO_CSR_TU_EMPTY_COUNT_MASK (0x1f << SPDIF_DATA_FIFO_CSR_TU_EMPTY_COUNT_SHIFT) + +/* Clear Receiver Data FIFO (RX DATA.FIFO) */ +#define TEGRA20_SPDIF_DATA_FIFO_CSR_RX_CLR (1 << 15) + +#define TEGRA20_SPDIF_FIFO_ATN_LVL_D_ONE_SLOT 0 +#define TEGRA20_SPDIF_FIFO_ATN_LVL_D_FOUR_SLOTS 1 +#define TEGRA20_SPDIF_FIFO_ATN_LVL_D_EIGHT_SLOTS 2 +#define TEGRA20_SPDIF_FIFO_ATN_LVL_D_TWELVE_SLOTS 3 + +/* RU FIFO attention level */ +#define TEGRA20_SPDIF_DATA_FIFO_CSR_RX_ATN_LVL_SHIFT 13 +#define TEGRA20_SPDIF_DATA_FIFO_CSR_RX_ATN_LVL_MASK \ + (0x3 << TEGRA20_SPDIF_DATA_FIFO_CSR_RX_ATN_LVL_SHIFT) +#define TEGRA20_SPDIF_DATA_FIFO_CSR_RX_ATN_LVL_RU1_WORD_FULL \ + (TEGRA20_SPDIF_FIFO_ATN_LVL_D_ONE_SLOT << TEGRA20_SPDIF_DATA_FIFO_CSR_RX_ATN_LVL_SHIFT) +#define TEGRA20_SPDIF_DATA_FIFO_CSR_RX_ATN_LVL_RU4_WORD_FULL \ + (TEGRA20_SPDIF_FIFO_ATN_LVL_D_FOUR_SLOTS << TEGRA20_SPDIF_DATA_FIFO_CSR_RX_ATN_LVL_SHIFT) +#define TEGRA20_SPDIF_DATA_FIFO_CSR_RX_ATN_LVL_RU8_WORD_FULL \ + (TEGRA20_SPDIF_FIFO_ATN_LVL_D_EIGHT_SLOTS << TEGRA20_SPDIF_DATA_FIFO_CSR_RX_ATN_LVL_SHIFT) +#define TEGRA20_SPDIF_DATA_FIFO_CSR_RX_ATN_LVL_RU12_WORD_FULL \ + (TEGRA20_SPDIF_FIFO_ATN_LVL_D_TWELVE_SLOTS << TEGRA20_SPDIF_DATA_FIFO_CSR_RX_ATN_LVL_SHIFT) + +/* Number of RX DATA.FIFO levels with valid data. */ +#define TEGRA20_SPDIF_DATA_FIFO_CSR_RX_FULL_COUNT_SHIFT 8 +#define TEGRA20_SPDIF_DATA_FIFO_CSR_RX_FULL_COUNT_MASK (0x1f << TEGRA20_SPDIF_DATA_FIFO_CSR_RX_FULL_COUNT_SHIFT) + +/* Clear Transmitter Data FIFO (TX DATA.FIFO) */ +#define TEGRA20_SPDIF_DATA_FIFO_CSR_TX_CLR (1 << 7) + +/* TU FIFO attention level */ +#define TEGRA20_SPDIF_DATA_FIFO_CSR_TX_ATN_LVL_SHIFT 5 +#define TEGRA20_SPDIF_DATA_FIFO_CSR_TX_ATN_LVL_MASK \ + (0x3 << TEGRA20_SPDIF_DATA_FIFO_CSR_TX_ATN_LVL_SHIFT) +#define TEGRA20_SPDIF_DATA_FIFO_CSR_TX_ATN_LVL_TU1_WORD_FULL \ + (TEGRA20_SPDIF_FIFO_ATN_LVL_D_ONE_SLOT << TEGRA20_SPDIF_DATA_FIFO_CSR_TX_ATN_LVL_SHIFT) +#define TEGRA20_SPDIF_DATA_FIFO_CSR_TX_ATN_LVL_TU4_WORD_FULL \ + (TEGRA20_SPDIF_FIFO_ATN_LVL_D_FOUR_SLOTS << TEGRA20_SPDIF_DATA_FIFO_CSR_TX_ATN_LVL_SHIFT) +#define TEGRA20_SPDIF_DATA_FIFO_CSR_TX_ATN_LVL_TU8_WORD_FULL \ + (TEGRA20_SPDIF_FIFO_ATN_LVL_D_EIGHT_SLOTS << TEGRA20_SPDIF_DATA_FIFO_CSR_TX_ATN_LVL_SHIFT) +#define TEGRA20_SPDIF_DATA_FIFO_CSR_TX_ATN_LVL_TU12_WORD_FULL \ + (TEGRA20_SPDIF_FIFO_ATN_LVL_D_TWELVE_SLOTS << TEGRA20_SPDIF_DATA_FIFO_CSR_TX_ATN_LVL_SHIFT) + +/* Number of TX DATA.FIFO levels that could be filled. */ +#define TEGRA20_SPDIF_DATA_FIFO_CSR_TX_EMPTY_COUNT_SHIFT 0 +#define TEGRA20_SPDIF_DATA_FIFO_CSR_TX_EMPTY_COUNT_MASK (0x1f << SPDIF_DATA_FIFO_CSR_TX_EMPTY_COUNT_SHIFT) + +/* Fields in TEGRA20_SPDIF_DATA_OUT */ + +/* + * This register has 5 different formats: + * 16-bit (BIT_MODE=00, PACK=0) + * 20-bit (BIT_MODE=01, PACK=0) + * 24-bit (BIT_MODE=10, PACK=0) + * raw (BIT_MODE=11, PACK=0) + * 16-bit packed (BIT_MODE=00, PACK=1) + */ + +#define TEGRA20_SPDIF_DATA_OUT_DATA_16_SHIFT 0 +#define TEGRA20_SPDIF_DATA_OUT_DATA_16_MASK (0xffff << TEGRA20_SPDIF_DATA_OUT_DATA_16_SHIFT) + +#define TEGRA20_SPDIF_DATA_OUT_DATA_20_SHIFT 0 +#define TEGRA20_SPDIF_DATA_OUT_DATA_20_MASK (0xfffff << TEGRA20_SPDIF_DATA_OUT_DATA_20_SHIFT) + +#define TEGRA20_SPDIF_DATA_OUT_DATA_24_SHIFT 0 +#define TEGRA20_SPDIF_DATA_OUT_DATA_24_MASK (0xffffff << TEGRA20_SPDIF_DATA_OUT_DATA_24_SHIFT) + +#define TEGRA20_SPDIF_DATA_OUT_DATA_RAW_P (1 << 31) +#define TEGRA20_SPDIF_DATA_OUT_DATA_RAW_C (1 << 30) +#define TEGRA20_SPDIF_DATA_OUT_DATA_RAW_U (1 << 29) +#define TEGRA20_SPDIF_DATA_OUT_DATA_RAW_V (1 << 28) + +#define TEGRA20_SPDIF_DATA_OUT_DATA_RAW_DATA_SHIFT 8 +#define TEGRA20_SPDIF_DATA_OUT_DATA_RAW_DATA_MASK (0xfffff << TEGRA20_SPDIF_DATA_OUT_DATA_RAW_DATA_SHIFT) + +#define TEGRA20_SPDIF_DATA_OUT_DATA_RAW_AUX_SHIFT 4 +#define TEGRA20_SPDIF_DATA_OUT_DATA_RAW_AUX_MASK (0xf << TEGRA20_SPDIF_DATA_OUT_DATA_RAW_AUX_SHIFT) + +#define TEGRA20_SPDIF_DATA_OUT_DATA_RAW_PREAMBLE_SHIFT 0 +#define TEGRA20_SPDIF_DATA_OUT_DATA_RAW_PREAMBLE_MASK (0xf << TEGRA20_SPDIF_DATA_OUT_DATA_RAW_PREAMBLE_SHIFT) + +#define TEGRA20_SPDIF_DATA_OUT_DATA_16_PACKED_RIGHT_SHIFT 16 +#define TEGRA20_SPDIF_DATA_OUT_DATA_16_PACKED_RIGHT_MASK (0xffff << TEGRA20_SPDIF_DATA_OUT_DATA_16_PACKED_RIGHT_SHIFT) + +#define TEGRA20_SPDIF_DATA_OUT_DATA_16_PACKED_LEFT_SHIFT 0 +#define TEGRA20_SPDIF_DATA_OUT_DATA_16_PACKED_LEFT_MASK (0xffff << TEGRA20_SPDIF_DATA_OUT_DATA_16_PACKED_LEFT_SHIFT) + +/* Fields in TEGRA20_SPDIF_DATA_IN */ + +/* + * This register has 5 different formats: + * 16-bit (BIT_MODE=00, PACK=0) + * 20-bit (BIT_MODE=01, PACK=0) + * 24-bit (BIT_MODE=10, PACK=0) + * raw (BIT_MODE=11, PACK=0) + * 16-bit packed (BIT_MODE=00, PACK=1) + * + * Bits 31:24 are common to all modes except 16-bit packed + */ + +#define TEGRA20_SPDIF_DATA_IN_DATA_P (1 << 31) +#define TEGRA20_SPDIF_DATA_IN_DATA_C (1 << 30) +#define TEGRA20_SPDIF_DATA_IN_DATA_U (1 << 29) +#define TEGRA20_SPDIF_DATA_IN_DATA_V (1 << 28) + +#define TEGRA20_SPDIF_DATA_IN_DATA_PREAMBLE_SHIFT 24 +#define TEGRA20_SPDIF_DATA_IN_DATA_PREAMBLE_MASK (0xf << TEGRA20_SPDIF_DATA_IN_DATA_PREAMBLE_SHIFT) + +#define TEGRA20_SPDIF_DATA_IN_DATA_16_SHIFT 0 +#define TEGRA20_SPDIF_DATA_IN_DATA_16_MASK (0xffff << TEGRA20_SPDIF_DATA_IN_DATA_16_SHIFT) + +#define TEGRA20_SPDIF_DATA_IN_DATA_20_SHIFT 0 +#define TEGRA20_SPDIF_DATA_IN_DATA_20_MASK (0xfffff << TEGRA20_SPDIF_DATA_IN_DATA_20_SHIFT) + +#define TEGRA20_SPDIF_DATA_IN_DATA_24_SHIFT 0 +#define TEGRA20_SPDIF_DATA_IN_DATA_24_MASK (0xffffff << TEGRA20_SPDIF_DATA_IN_DATA_24_SHIFT) + +#define TEGRA20_SPDIF_DATA_IN_DATA_RAW_DATA_SHIFT 8 +#define TEGRA20_SPDIF_DATA_IN_DATA_RAW_DATA_MASK (0xfffff << TEGRA20_SPDIF_DATA_IN_DATA_RAW_DATA_SHIFT) + +#define TEGRA20_SPDIF_DATA_IN_DATA_RAW_AUX_SHIFT 4 +#define TEGRA20_SPDIF_DATA_IN_DATA_RAW_AUX_MASK (0xf << TEGRA20_SPDIF_DATA_IN_DATA_RAW_AUX_SHIFT) + +#define TEGRA20_SPDIF_DATA_IN_DATA_RAW_PREAMBLE_SHIFT 0 +#define TEGRA20_SPDIF_DATA_IN_DATA_RAW_PREAMBLE_MASK (0xf << TEGRA20_SPDIF_DATA_IN_DATA_RAW_PREAMBLE_SHIFT) + +#define TEGRA20_SPDIF_DATA_IN_DATA_16_PACKED_RIGHT_SHIFT 16 +#define TEGRA20_SPDIF_DATA_IN_DATA_16_PACKED_RIGHT_MASK (0xffff << TEGRA20_SPDIF_DATA_IN_DATA_16_PACKED_RIGHT_SHIFT) + +#define TEGRA20_SPDIF_DATA_IN_DATA_16_PACKED_LEFT_SHIFT 0 +#define TEGRA20_SPDIF_DATA_IN_DATA_16_PACKED_LEFT_MASK (0xffff << TEGRA20_SPDIF_DATA_IN_DATA_16_PACKED_LEFT_SHIFT) + +/* Fields in TEGRA20_SPDIF_CH_STA_RX_A */ +/* Fields in TEGRA20_SPDIF_CH_STA_RX_B */ +/* Fields in TEGRA20_SPDIF_CH_STA_RX_C */ +/* Fields in TEGRA20_SPDIF_CH_STA_RX_D */ +/* Fields in TEGRA20_SPDIF_CH_STA_RX_E */ +/* Fields in TEGRA20_SPDIF_CH_STA_RX_F */ + +/* + * The 6-word receive channel data page buffer holds a block (192 frames) of + * channel status information. The order of receive is from LSB to MSB + * bit, and from CH_STA_RX_A to CH_STA_RX_F then back to CH_STA_RX_A. + */ + +/* Fields in TEGRA20_SPDIF_CH_STA_TX_A */ +/* Fields in TEGRA20_SPDIF_CH_STA_TX_B */ +/* Fields in TEGRA20_SPDIF_CH_STA_TX_C */ +/* Fields in TEGRA20_SPDIF_CH_STA_TX_D */ +/* Fields in TEGRA20_SPDIF_CH_STA_TX_E */ +/* Fields in TEGRA20_SPDIF_CH_STA_TX_F */ + +/* + * The 6-word transmit channel data page buffer holds a block (192 frames) of + * channel status information. The order of transmission is from LSB to MSB + * bit, and from CH_STA_TX_A to CH_STA_TX_F then back to CH_STA_TX_A. + */ + +/* Fields in TEGRA20_SPDIF_USR_STA_RX_A */ + +/* + * This 4-word deep FIFO receives user FIFO field information. The order of + * receive is from LSB to MSB bit. + */ + +/* Fields in TEGRA20_SPDIF_USR_DAT_TX_A */ + +/* + * This 4-word deep FIFO transmits user FIFO field information. The order of + * transmission is from LSB to MSB bit. + */ + +struct tegra20_spdif { + struct clk *clk_spdif_out; + struct tegra_pcm_dma_params capture_dma_data; + struct tegra_pcm_dma_params playback_dma_data; + struct regmap *regmap; + u32 reg_ctrl; +}; + +#endif diff --git a/sound/soc/tegra/tegra30_ahub.c b/sound/soc/tegra/tegra30_ahub.c new file mode 100644 index 00000000000..57cd419f743 --- /dev/null +++ b/sound/soc/tegra/tegra30_ahub.c @@ -0,0 +1,631 @@ +/* + * tegra30_ahub.c - Tegra30 AHUB driver + * + * Copyright (c) 2011,2012, NVIDIA CORPORATION. All rights reserved. + * + * This program is free software; you can redistribute it and/or modify it + * under the terms and conditions of the GNU General Public License, + * version 2, as published by the Free Software Foundation. + * + * This program is distributed in the hope it will be useful, but WITHOUT + * ANY WARRANTY; without even the implied warranty of MERCHANTABILITY or + * FITNESS FOR A PARTICULAR PURPOSE. See the GNU General Public License for + * more details. + * + * You should have received a copy of the GNU General Public License + * along with this program. If not, see <http://www.gnu.org/licenses/>. + */ + +#include <linux/clk.h> +#include <linux/device.h> +#include <linux/io.h> +#include <linux/module.h> +#include <linux/of_platform.h> +#include <linux/platform_device.h> +#include <linux/pm_runtime.h> +#include <linux/regmap.h> +#include <linux/slab.h> +#include <mach/clk.h> +#include <mach/dma.h> +#include <sound/soc.h> +#include "tegra30_ahub.h" + +#define DRV_NAME "tegra30-ahub" + +static struct tegra30_ahub *ahub; + +static inline void tegra30_apbif_write(u32 reg, u32 val) +{ + regmap_write(ahub->regmap_apbif, reg, val); +} + +static inline u32 tegra30_apbif_read(u32 reg) +{ + u32 val; + regmap_read(ahub->regmap_apbif, reg, &val); + return val; +} + +static inline void tegra30_audio_write(u32 reg, u32 val) +{ + regmap_write(ahub->regmap_ahub, reg, val); +} + +static int tegra30_ahub_runtime_suspend(struct device *dev) +{ + regcache_cache_only(ahub->regmap_apbif, true); + regcache_cache_only(ahub->regmap_ahub, true); + + clk_disable(ahub->clk_apbif); + clk_disable(ahub->clk_d_audio); + + return 0; +} + +/* + * clk_apbif isn't required for an I2S<->I2S configuration where no PCM data + * is read from or sent to memory. However, that's not something the rest of + * the driver supports right now, so we'll just treat the two clocks as one + * for now. + * + * These functions should not be a plain ref-count. Instead, each active stream + * contributes some requirement to the minimum clock rate, so starting or + * stopping streams should dynamically adjust the clock as required. However, + * this is not yet implemented. + */ +static int tegra30_ahub_runtime_resume(struct device *dev) +{ + int ret; + + ret = clk_enable(ahub->clk_d_audio); + if (ret) { + dev_err(dev, "clk_enable d_audio failed: %d\n", ret); + return ret; + } + ret = clk_enable(ahub->clk_apbif); + if (ret) { + dev_err(dev, "clk_enable apbif failed: %d\n", ret); + clk_disable(ahub->clk_d_audio); + return ret; + } + + regcache_cache_only(ahub->regmap_apbif, false); + regcache_cache_only(ahub->regmap_ahub, false); + + return 0; +} + +int tegra30_ahub_allocate_rx_fifo(enum tegra30_ahub_rxcif *rxcif, + unsigned long *fiforeg, + unsigned long *reqsel) +{ + int channel; + u32 reg, val; + + channel = find_first_zero_bit(ahub->rx_usage, + TEGRA30_AHUB_CHANNEL_CTRL_COUNT); + if (channel >= TEGRA30_AHUB_CHANNEL_CTRL_COUNT) + return -EBUSY; + + __set_bit(channel, ahub->rx_usage); + + *rxcif = TEGRA30_AHUB_RXCIF_APBIF_RX0 + channel; + *fiforeg = ahub->apbif_addr + TEGRA30_AHUB_CHANNEL_RXFIFO + + (channel * TEGRA30_AHUB_CHANNEL_RXFIFO_STRIDE); + *reqsel = ahub->dma_sel + channel; + + reg = TEGRA30_AHUB_CHANNEL_CTRL + + (channel * TEGRA30_AHUB_CHANNEL_CTRL_STRIDE); + val = tegra30_apbif_read(reg); + val &= ~(TEGRA30_AHUB_CHANNEL_CTRL_RX_THRESHOLD_MASK | + TEGRA30_AHUB_CHANNEL_CTRL_RX_PACK_MASK); + val |= (7 << TEGRA30_AHUB_CHANNEL_CTRL_RX_THRESHOLD_SHIFT) | + TEGRA30_AHUB_CHANNEL_CTRL_RX_PACK_EN | + TEGRA30_AHUB_CHANNEL_CTRL_RX_PACK_16; + tegra30_apbif_write(reg, val); + + reg = TEGRA30_AHUB_CIF_RX_CTRL + + (channel * TEGRA30_AHUB_CIF_RX_CTRL_STRIDE); + val = (0 << TEGRA30_AUDIOCIF_CTRL_FIFO_THRESHOLD_SHIFT) | + (1 << TEGRA30_AUDIOCIF_CTRL_AUDIO_CHANNELS_SHIFT) | + (1 << TEGRA30_AUDIOCIF_CTRL_CLIENT_CHANNELS_SHIFT) | + TEGRA30_AUDIOCIF_CTRL_AUDIO_BITS_16 | + TEGRA30_AUDIOCIF_CTRL_CLIENT_BITS_16 | + TEGRA30_AUDIOCIF_CTRL_DIRECTION_RX; + tegra30_apbif_write(reg, val); + + return 0; +} +EXPORT_SYMBOL_GPL(tegra30_ahub_allocate_rx_fifo); + +int tegra30_ahub_enable_rx_fifo(enum tegra30_ahub_rxcif rxcif) +{ + int channel = rxcif - TEGRA30_AHUB_RXCIF_APBIF_RX0; + int reg, val; + + reg = TEGRA30_AHUB_CHANNEL_CTRL + + (channel * TEGRA30_AHUB_CHANNEL_CTRL_STRIDE); + val = tegra30_apbif_read(reg); + val |= TEGRA30_AHUB_CHANNEL_CTRL_RX_EN; + tegra30_apbif_write(reg, val); + + return 0; +} +EXPORT_SYMBOL_GPL(tegra30_ahub_enable_rx_fifo); + +int tegra30_ahub_disable_rx_fifo(enum tegra30_ahub_rxcif rxcif) +{ + int channel = rxcif - TEGRA30_AHUB_RXCIF_APBIF_RX0; + int reg, val; + + reg = TEGRA30_AHUB_CHANNEL_CTRL + + (channel * TEGRA30_AHUB_CHANNEL_CTRL_STRIDE); + val = tegra30_apbif_read(reg); + val &= ~TEGRA30_AHUB_CHANNEL_CTRL_RX_EN; + tegra30_apbif_write(reg, val); + + return 0; +} +EXPORT_SYMBOL_GPL(tegra30_ahub_disable_rx_fifo); + +int tegra30_ahub_free_rx_fifo(enum tegra30_ahub_rxcif rxcif) +{ + int channel = rxcif - TEGRA30_AHUB_RXCIF_APBIF_RX0; + + __clear_bit(channel, ahub->rx_usage); + + return 0; +} +EXPORT_SYMBOL_GPL(tegra30_ahub_free_rx_fifo); + +int tegra30_ahub_allocate_tx_fifo(enum tegra30_ahub_txcif *txcif, + unsigned long *fiforeg, + unsigned long *reqsel) +{ + int channel; + u32 reg, val; + + channel = find_first_zero_bit(ahub->tx_usage, + TEGRA30_AHUB_CHANNEL_CTRL_COUNT); + if (channel >= TEGRA30_AHUB_CHANNEL_CTRL_COUNT) + return -EBUSY; + + __set_bit(channel, ahub->tx_usage); + + *txcif = TEGRA30_AHUB_TXCIF_APBIF_TX0 + channel; + *fiforeg = ahub->apbif_addr + TEGRA30_AHUB_CHANNEL_TXFIFO + + (channel * TEGRA30_AHUB_CHANNEL_TXFIFO_STRIDE); + *reqsel = ahub->dma_sel + channel; + + reg = TEGRA30_AHUB_CHANNEL_CTRL + + (channel * TEGRA30_AHUB_CHANNEL_CTRL_STRIDE); + val = tegra30_apbif_read(reg); + val &= ~(TEGRA30_AHUB_CHANNEL_CTRL_TX_THRESHOLD_MASK | + TEGRA30_AHUB_CHANNEL_CTRL_TX_PACK_MASK); + val |= (7 << TEGRA30_AHUB_CHANNEL_CTRL_TX_THRESHOLD_SHIFT) | + TEGRA30_AHUB_CHANNEL_CTRL_TX_PACK_EN | + TEGRA30_AHUB_CHANNEL_CTRL_TX_PACK_16; + tegra30_apbif_write(reg, val); + + reg = TEGRA30_AHUB_CIF_TX_CTRL + + (channel * TEGRA30_AHUB_CIF_TX_CTRL_STRIDE); + val = (0 << TEGRA30_AUDIOCIF_CTRL_FIFO_THRESHOLD_SHIFT) | + (1 << TEGRA30_AUDIOCIF_CTRL_AUDIO_CHANNELS_SHIFT) | + (1 << TEGRA30_AUDIOCIF_CTRL_CLIENT_CHANNELS_SHIFT) | + TEGRA30_AUDIOCIF_CTRL_AUDIO_BITS_16 | + TEGRA30_AUDIOCIF_CTRL_CLIENT_BITS_16 | + TEGRA30_AUDIOCIF_CTRL_DIRECTION_TX; + tegra30_apbif_write(reg, val); + + return 0; +} +EXPORT_SYMBOL_GPL(tegra30_ahub_allocate_tx_fifo); + +int tegra30_ahub_enable_tx_fifo(enum tegra30_ahub_txcif txcif) +{ + int channel = txcif - TEGRA30_AHUB_TXCIF_APBIF_TX0; + int reg, val; + + reg = TEGRA30_AHUB_CHANNEL_CTRL + + (channel * TEGRA30_AHUB_CHANNEL_CTRL_STRIDE); + val = tegra30_apbif_read(reg); + val |= TEGRA30_AHUB_CHANNEL_CTRL_TX_EN; + tegra30_apbif_write(reg, val); + + return 0; +} +EXPORT_SYMBOL_GPL(tegra30_ahub_enable_tx_fifo); + +int tegra30_ahub_disable_tx_fifo(enum tegra30_ahub_txcif txcif) +{ + int channel = txcif - TEGRA30_AHUB_TXCIF_APBIF_TX0; + int reg, val; + + reg = TEGRA30_AHUB_CHANNEL_CTRL + + (channel * TEGRA30_AHUB_CHANNEL_CTRL_STRIDE); + val = tegra30_apbif_read(reg); + val &= ~TEGRA30_AHUB_CHANNEL_CTRL_TX_EN; + tegra30_apbif_write(reg, val); + + return 0; +} +EXPORT_SYMBOL_GPL(tegra30_ahub_disable_tx_fifo); + +int tegra30_ahub_free_tx_fifo(enum tegra30_ahub_txcif txcif) +{ + int channel = txcif - TEGRA30_AHUB_TXCIF_APBIF_TX0; + + __clear_bit(channel, ahub->tx_usage); + + return 0; +} +EXPORT_SYMBOL_GPL(tegra30_ahub_free_tx_fifo); + +int tegra30_ahub_set_rx_cif_source(enum tegra30_ahub_rxcif rxcif, + enum tegra30_ahub_txcif txcif) +{ + int channel = rxcif - TEGRA30_AHUB_RXCIF_APBIF_RX0; + int reg; + + reg = TEGRA30_AHUB_AUDIO_RX + + (channel * TEGRA30_AHUB_AUDIO_RX_STRIDE); + tegra30_audio_write(reg, 1 << txcif); + + return 0; +} +EXPORT_SYMBOL_GPL(tegra30_ahub_set_rx_cif_source); + +int tegra30_ahub_unset_rx_cif_source(enum tegra30_ahub_rxcif rxcif) +{ + int channel = rxcif - TEGRA30_AHUB_RXCIF_APBIF_RX0; + int reg; + + reg = TEGRA30_AHUB_AUDIO_RX + + (channel * TEGRA30_AHUB_AUDIO_RX_STRIDE); + tegra30_audio_write(reg, 0); + + return 0; +} +EXPORT_SYMBOL_GPL(tegra30_ahub_unset_rx_cif_source); + +static const char * const configlink_clocks[] __devinitconst = { + "i2s0", + "i2s1", + "i2s2", + "i2s3", + "i2s4", + "dam0", + "dam1", + "dam2", + "spdif_in", +}; + +struct of_dev_auxdata ahub_auxdata[] __devinitdata = { + OF_DEV_AUXDATA("nvidia,tegra30-i2s", 0x70080300, "tegra30-i2s.0", NULL), + OF_DEV_AUXDATA("nvidia,tegra30-i2s", 0x70080400, "tegra30-i2s.1", NULL), + OF_DEV_AUXDATA("nvidia,tegra30-i2s", 0x70080500, "tegra30-i2s.2", NULL), + OF_DEV_AUXDATA("nvidia,tegra30-i2s", 0x70080600, "tegra30-i2s.3", NULL), + OF_DEV_AUXDATA("nvidia,tegra30-i2s", 0x70080700, "tegra30-i2s.4", NULL), + {} +}; + +#define LAST_REG(name) \ + (TEGRA30_AHUB_##name + \ + (TEGRA30_AHUB_##name##_STRIDE * TEGRA30_AHUB_##name##_COUNT) - 4) + +#define REG_IN_ARRAY(reg, name) \ + ((reg >= TEGRA30_AHUB_##name) && \ + (reg <= LAST_REG(name) && \ + (!((reg - TEGRA30_AHUB_##name) % TEGRA30_AHUB_##name##_STRIDE)))) + +static bool tegra30_ahub_apbif_wr_rd_reg(struct device *dev, unsigned int reg) +{ + switch (reg) { + case TEGRA30_AHUB_CONFIG_LINK_CTRL: + case TEGRA30_AHUB_MISC_CTRL: + case TEGRA30_AHUB_APBDMA_LIVE_STATUS: + case TEGRA30_AHUB_I2S_LIVE_STATUS: + case TEGRA30_AHUB_SPDIF_LIVE_STATUS: + case TEGRA30_AHUB_I2S_INT_MASK: + case TEGRA30_AHUB_DAM_INT_MASK: + case TEGRA30_AHUB_SPDIF_INT_MASK: + case TEGRA30_AHUB_APBIF_INT_MASK: + case TEGRA30_AHUB_I2S_INT_STATUS: + case TEGRA30_AHUB_DAM_INT_STATUS: + case TEGRA30_AHUB_SPDIF_INT_STATUS: + case TEGRA30_AHUB_APBIF_INT_STATUS: + case TEGRA30_AHUB_I2S_INT_SOURCE: + case TEGRA30_AHUB_DAM_INT_SOURCE: + case TEGRA30_AHUB_SPDIF_INT_SOURCE: + case TEGRA30_AHUB_APBIF_INT_SOURCE: + case TEGRA30_AHUB_I2S_INT_SET: + case TEGRA30_AHUB_DAM_INT_SET: + case TEGRA30_AHUB_SPDIF_INT_SET: + case TEGRA30_AHUB_APBIF_INT_SET: + return true; + default: + break; + }; + + if (REG_IN_ARRAY(reg, CHANNEL_CTRL) || + REG_IN_ARRAY(reg, CHANNEL_CLEAR) || + REG_IN_ARRAY(reg, CHANNEL_STATUS) || + REG_IN_ARRAY(reg, CHANNEL_TXFIFO) || + REG_IN_ARRAY(reg, CHANNEL_RXFIFO) || + REG_IN_ARRAY(reg, CIF_TX_CTRL) || + REG_IN_ARRAY(reg, CIF_RX_CTRL) || + REG_IN_ARRAY(reg, DAM_LIVE_STATUS)) + return true; + + return false; +} + +static bool tegra30_ahub_apbif_volatile_reg(struct device *dev, + unsigned int reg) +{ + switch (reg) { + case TEGRA30_AHUB_CONFIG_LINK_CTRL: + case TEGRA30_AHUB_MISC_CTRL: + case TEGRA30_AHUB_APBDMA_LIVE_STATUS: + case TEGRA30_AHUB_I2S_LIVE_STATUS: + case TEGRA30_AHUB_SPDIF_LIVE_STATUS: + case TEGRA30_AHUB_I2S_INT_STATUS: + case TEGRA30_AHUB_DAM_INT_STATUS: + case TEGRA30_AHUB_SPDIF_INT_STATUS: + case TEGRA30_AHUB_APBIF_INT_STATUS: + case TEGRA30_AHUB_I2S_INT_SET: + case TEGRA30_AHUB_DAM_INT_SET: + case TEGRA30_AHUB_SPDIF_INT_SET: + case TEGRA30_AHUB_APBIF_INT_SET: + return true; + default: + break; + }; + + if (REG_IN_ARRAY(reg, CHANNEL_CLEAR) || + REG_IN_ARRAY(reg, CHANNEL_STATUS) || + REG_IN_ARRAY(reg, CHANNEL_TXFIFO) || + REG_IN_ARRAY(reg, CHANNEL_RXFIFO) || + REG_IN_ARRAY(reg, DAM_LIVE_STATUS)) + return true; + + return false; +} + +static bool tegra30_ahub_apbif_precious_reg(struct device *dev, + unsigned int reg) +{ + if (REG_IN_ARRAY(reg, CHANNEL_TXFIFO) || + REG_IN_ARRAY(reg, CHANNEL_RXFIFO)) + return true; + + return false; +} + +static const struct regmap_config tegra30_ahub_apbif_regmap_config = { + .name = "apbif", + .reg_bits = 32, + .val_bits = 32, + .reg_stride = 4, + .max_register = TEGRA30_AHUB_APBIF_INT_SET, + .writeable_reg = tegra30_ahub_apbif_wr_rd_reg, + .readable_reg = tegra30_ahub_apbif_wr_rd_reg, + .volatile_reg = tegra30_ahub_apbif_volatile_reg, + .precious_reg = tegra30_ahub_apbif_precious_reg, + .cache_type = REGCACHE_RBTREE, +}; + +static bool tegra30_ahub_ahub_wr_rd_reg(struct device *dev, unsigned int reg) +{ + if (REG_IN_ARRAY(reg, AUDIO_RX)) + return true; + + return false; +} + +static const struct regmap_config tegra30_ahub_ahub_regmap_config = { + .name = "ahub", + .reg_bits = 32, + .val_bits = 32, + .reg_stride = 4, + .max_register = LAST_REG(AUDIO_RX), + .writeable_reg = tegra30_ahub_ahub_wr_rd_reg, + .readable_reg = tegra30_ahub_ahub_wr_rd_reg, + .cache_type = REGCACHE_RBTREE, +}; + +static int __devinit tegra30_ahub_probe(struct platform_device *pdev) +{ + struct clk *clk; + int i; + struct resource *res0, *res1, *region; + u32 of_dma[2]; + void __iomem *regs_apbif, *regs_ahub; + int ret = 0; + + if (ahub) + return -ENODEV; + + /* + * The AHUB hosts a register bus: the "configlink". For this to + * operate correctly, all devices on this bus must be out of reset. + * Ensure that here. + */ + for (i = 0; i < ARRAY_SIZE(configlink_clocks); i++) { + clk = clk_get_sys(NULL, configlink_clocks[i]); + if (IS_ERR(clk)) { + dev_err(&pdev->dev, "Can't get clock %s\n", + configlink_clocks[i]); + ret = PTR_ERR(clk); + goto err; + } + tegra_periph_reset_deassert(clk); + clk_put(clk); + } + + ahub = devm_kzalloc(&pdev->dev, sizeof(struct tegra30_ahub), + GFP_KERNEL); + if (!ahub) { + dev_err(&pdev->dev, "Can't allocate tegra30_ahub\n"); + ret = -ENOMEM; + goto err; + } + dev_set_drvdata(&pdev->dev, ahub); + + ahub->dev = &pdev->dev; + + ahub->clk_d_audio = clk_get(&pdev->dev, "d_audio"); + if (IS_ERR(ahub->clk_d_audio)) { + dev_err(&pdev->dev, "Can't retrieve ahub d_audio clock\n"); + ret = PTR_ERR(ahub->clk_d_audio); + goto err; + } + + ahub->clk_apbif = clk_get(&pdev->dev, "apbif"); + if (IS_ERR(ahub->clk_apbif)) { + dev_err(&pdev->dev, "Can't retrieve ahub apbif clock\n"); + ret = PTR_ERR(ahub->clk_apbif); + goto err_clk_put_d_audio; + } + + if (of_property_read_u32_array(pdev->dev.of_node, + "nvidia,dma-request-selector", + of_dma, 2) < 0) { + dev_err(&pdev->dev, + "Missing property nvidia,dma-request-selector\n"); + ret = -ENODEV; + goto err_clk_put_d_audio; + } + ahub->dma_sel = of_dma[1]; + + res0 = platform_get_resource(pdev, IORESOURCE_MEM, 0); + if (!res0) { + dev_err(&pdev->dev, "No apbif memory resource\n"); + ret = -ENODEV; + goto err_clk_put_apbif; + } + + region = devm_request_mem_region(&pdev->dev, res0->start, + resource_size(res0), DRV_NAME); + if (!region) { + dev_err(&pdev->dev, "request region apbif failed\n"); + ret = -EBUSY; + goto err_clk_put_apbif; + } + ahub->apbif_addr = res0->start; + + regs_apbif = devm_ioremap(&pdev->dev, res0->start, + resource_size(res0)); + if (!regs_apbif) { + dev_err(&pdev->dev, "ioremap apbif failed\n"); + ret = -ENOMEM; + goto err_clk_put_apbif; + } + + ahub->regmap_apbif = devm_regmap_init_mmio(&pdev->dev, regs_apbif, + &tegra30_ahub_apbif_regmap_config); + if (IS_ERR(ahub->regmap_apbif)) { + dev_err(&pdev->dev, "apbif regmap init failed\n"); + ret = PTR_ERR(ahub->regmap_apbif); + goto err_clk_put_apbif; + } + regcache_cache_only(ahub->regmap_apbif, true); + + res1 = platform_get_resource(pdev, IORESOURCE_MEM, 1); + if (!res1) { + dev_err(&pdev->dev, "No ahub memory resource\n"); + ret = -ENODEV; + goto err_clk_put_apbif; + } + + region = devm_request_mem_region(&pdev->dev, res1->start, + resource_size(res1), DRV_NAME); + if (!region) { + dev_err(&pdev->dev, "request region ahub failed\n"); + ret = -EBUSY; + goto err_clk_put_apbif; + } + + regs_ahub = devm_ioremap(&pdev->dev, res1->start, + resource_size(res1)); + if (!regs_ahub) { + dev_err(&pdev->dev, "ioremap ahub failed\n"); + ret = -ENOMEM; + goto err_clk_put_apbif; + } + + ahub->regmap_ahub = devm_regmap_init_mmio(&pdev->dev, regs_ahub, + &tegra30_ahub_ahub_regmap_config); + if (IS_ERR(ahub->regmap_ahub)) { + dev_err(&pdev->dev, "ahub regmap init failed\n"); + ret = PTR_ERR(ahub->regmap_ahub); + goto err_clk_put_apbif; + } + regcache_cache_only(ahub->regmap_ahub, true); + + pm_runtime_enable(&pdev->dev); + if (!pm_runtime_enabled(&pdev->dev)) { + ret = tegra30_ahub_runtime_resume(&pdev->dev); + if (ret) + goto err_pm_disable; + } + + of_platform_populate(pdev->dev.of_node, NULL, ahub_auxdata, + &pdev->dev); + + return 0; + +err_pm_disable: + pm_runtime_disable(&pdev->dev); +err_clk_put_apbif: + clk_put(ahub->clk_apbif); +err_clk_put_d_audio: + clk_put(ahub->clk_d_audio); + ahub = 0; +err: + return ret; +} + +static int __devexit tegra30_ahub_remove(struct platform_device *pdev) +{ + if (!ahub) + return -ENODEV; + + pm_runtime_disable(&pdev->dev); + if (!pm_runtime_status_suspended(&pdev->dev)) + tegra30_ahub_runtime_suspend(&pdev->dev); + + clk_put(ahub->clk_apbif); + clk_put(ahub->clk_d_audio); + + ahub = 0; + + return 0; +} + +static const struct of_device_id tegra30_ahub_of_match[] __devinitconst = { + { .compatible = "nvidia,tegra30-ahub", }, + {}, +}; + +static const struct dev_pm_ops tegra30_ahub_pm_ops __devinitconst = { + SET_RUNTIME_PM_OPS(tegra30_ahub_runtime_suspend, + tegra30_ahub_runtime_resume, NULL) +}; + +static struct platform_driver tegra30_ahub_driver = { + .probe = tegra30_ahub_probe, + .remove = __devexit_p(tegra30_ahub_remove), + .driver = { + .name = DRV_NAME, + .owner = THIS_MODULE, + .of_match_table = tegra30_ahub_of_match, + .pm = &tegra30_ahub_pm_ops, + }, +}; +module_platform_driver(tegra30_ahub_driver); + +MODULE_AUTHOR("Stephen Warren <swarren@nvidia.com>"); +MODULE_DESCRIPTION("Tegra30 AHUB driver"); +MODULE_LICENSE("GPL v2"); +MODULE_ALIAS("platform:" DRV_NAME); diff --git a/sound/soc/tegra/tegra30_ahub.h b/sound/soc/tegra/tegra30_ahub.h new file mode 100644 index 00000000000..e690e2eecc9 --- /dev/null +++ b/sound/soc/tegra/tegra30_ahub.h @@ -0,0 +1,483 @@ +/* + * tegra30_ahub.h - Definitions for Tegra30 AHUB driver + * + * Copyright (c) 2011,2012, NVIDIA CORPORATION. All rights reserved. + * + * This program is free software; you can redistribute it and/or modify it + * under the terms and conditions of the GNU General Public License, + * version 2, as published by the Free Software Foundation. + * + * This program is distributed in the hope it will be useful, but WITHOUT + * ANY WARRANTY; without even the implied warranty of MERCHANTABILITY or + * FITNESS FOR A PARTICULAR PURPOSE. See the GNU General Public License for + * more details. + * + * You should have received a copy of the GNU General Public License + * along with this program. If not, see <http://www.gnu.org/licenses/>. + */ + +#ifndef __TEGRA30_AHUB_H__ +#define __TEGRA30_AHUB_H__ + +/* Fields in *_CIF_RX/TX_CTRL; used by AHUB FIFOs, and all other audio modules */ + +#define TEGRA30_AUDIOCIF_CTRL_FIFO_THRESHOLD_SHIFT 28 +#define TEGRA30_AUDIOCIF_CTRL_FIFO_THRESHOLD_MASK_US 0xf +#define TEGRA30_AUDIOCIF_CTRL_FIFO_THRESHOLD_MASK (TEGRA30_AUDIOCIF_CTRL_FIFO_THRESHOLD_MASK_US << TEGRA30_AUDIOCIF_CTRL_FIFO_THRESHOLD_SHIFT) + +/* Channel count minus 1 */ +#define TEGRA30_AUDIOCIF_CTRL_AUDIO_CHANNELS_SHIFT 24 +#define TEGRA30_AUDIOCIF_CTRL_AUDIO_CHANNELS_MASK_US 7 +#define TEGRA30_AUDIOCIF_CTRL_AUDIO_CHANNELS_MASK (TEGRA30_AUDIOCIF_CTRL_AUDIO_CHANNELS_MASK_US << TEGRA30_AUDIOCIF_CTRL_AUDIO_CHANNELS_SHIFT) + +/* Channel count minus 1 */ +#define TEGRA30_AUDIOCIF_CTRL_CLIENT_CHANNELS_SHIFT 16 +#define TEGRA30_AUDIOCIF_CTRL_CLIENT_CHANNELS_MASK_US 7 +#define TEGRA30_AUDIOCIF_CTRL_CLIENT_CHANNELS_MASK (TEGRA30_AUDIOCIF_CTRL_CLIENT_CHANNELS_MASK_US << TEGRA30_AUDIOCIF_CTRL_CLIENT_CHANNELS_SHIFT) + +#define TEGRA30_AUDIOCIF_BITS_4 0 +#define TEGRA30_AUDIOCIF_BITS_8 1 +#define TEGRA30_AUDIOCIF_BITS_12 2 +#define TEGRA30_AUDIOCIF_BITS_16 3 +#define TEGRA30_AUDIOCIF_BITS_20 4 +#define TEGRA30_AUDIOCIF_BITS_24 5 +#define TEGRA30_AUDIOCIF_BITS_28 6 +#define TEGRA30_AUDIOCIF_BITS_32 7 + +#define TEGRA30_AUDIOCIF_CTRL_AUDIO_BITS_SHIFT 12 +#define TEGRA30_AUDIOCIF_CTRL_AUDIO_BITS_MASK (7 << TEGRA30_AUDIOCIF_CTRL_AUDIO_BITS_SHIFT) +#define TEGRA30_AUDIOCIF_CTRL_AUDIO_BITS_4 (TEGRA30_AUDIOCIF_BITS_4 << TEGRA30_AUDIOCIF_CTRL_AUDIO_BITS_SHIFT) +#define TEGRA30_AUDIOCIF_CTRL_AUDIO_BITS_8 (TEGRA30_AUDIOCIF_BITS_8 << TEGRA30_AUDIOCIF_CTRL_AUDIO_BITS_SHIFT) +#define TEGRA30_AUDIOCIF_CTRL_AUDIO_BITS_12 (TEGRA30_AUDIOCIF_BITS_12 << TEGRA30_AUDIOCIF_CTRL_AUDIO_BITS_SHIFT) +#define TEGRA30_AUDIOCIF_CTRL_AUDIO_BITS_16 (TEGRA30_AUDIOCIF_BITS_16 << TEGRA30_AUDIOCIF_CTRL_AUDIO_BITS_SHIFT) +#define TEGRA30_AUDIOCIF_CTRL_AUDIO_BITS_20 (TEGRA30_AUDIOCIF_BITS_20 << TEGRA30_AUDIOCIF_CTRL_AUDIO_BITS_SHIFT) +#define TEGRA30_AUDIOCIF_CTRL_AUDIO_BITS_24 (TEGRA30_AUDIOCIF_BITS_24 << TEGRA30_AUDIOCIF_CTRL_AUDIO_BITS_SHIFT) +#define TEGRA30_AUDIOCIF_CTRL_AUDIO_BITS_28 (TEGRA30_AUDIOCIF_BITS_28 << TEGRA30_AUDIOCIF_CTRL_AUDIO_BITS_SHIFT) +#define TEGRA30_AUDIOCIF_CTRL_AUDIO_BITS_32 (TEGRA30_AUDIOCIF_BITS_32 << TEGRA30_AUDIOCIF_CTRL_AUDIO_BITS_SHIFT) + +#define TEGRA30_AUDIOCIF_CTRL_CLIENT_BITS_SHIFT 8 +#define TEGRA30_AUDIOCIF_CTRL_CLIENT_BITS_MASK (7 << TEGRA30_AUDIOCIF_CTRL_CLIENT_BITS_SHIFT) +#define TEGRA30_AUDIOCIF_CTRL_CLIENT_BITS_4 (TEGRA30_AUDIOCIF_BITS_4 << TEGRA30_AUDIOCIF_CTRL_CLIENT_BITS_SHIFT) +#define TEGRA30_AUDIOCIF_CTRL_CLIENT_BITS_8 (TEGRA30_AUDIOCIF_BITS_8 << TEGRA30_AUDIOCIF_CTRL_CLIENT_BITS_SHIFT) +#define TEGRA30_AUDIOCIF_CTRL_CLIENT_BITS_12 (TEGRA30_AUDIOCIF_BITS_12 << TEGRA30_AUDIOCIF_CTRL_CLIENT_BITS_SHIFT) +#define TEGRA30_AUDIOCIF_CTRL_CLIENT_BITS_16 (TEGRA30_AUDIOCIF_BITS_16 << TEGRA30_AUDIOCIF_CTRL_CLIENT_BITS_SHIFT) +#define TEGRA30_AUDIOCIF_CTRL_CLIENT_BITS_20 (TEGRA30_AUDIOCIF_BITS_20 << TEGRA30_AUDIOCIF_CTRL_CLIENT_BITS_SHIFT) +#define TEGRA30_AUDIOCIF_CTRL_CLIENT_BITS_24 (TEGRA30_AUDIOCIF_BITS_24 << TEGRA30_AUDIOCIF_CTRL_CLIENT_BITS_SHIFT) +#define TEGRA30_AUDIOCIF_CTRL_CLIENT_BITS_28 (TEGRA30_AUDIOCIF_BITS_28 << TEGRA30_AUDIOCIF_CTRL_CLIENT_BITS_SHIFT) +#define TEGRA30_AUDIOCIF_CTRL_CLIENT_BITS_32 (TEGRA30_AUDIOCIF_BITS_32 << TEGRA30_AUDIOCIF_CTRL_CLIENT_BITS_SHIFT) + +#define TEGRA30_AUDIOCIF_EXPAND_ZERO 0 +#define TEGRA30_AUDIOCIF_EXPAND_ONE 1 +#define TEGRA30_AUDIOCIF_EXPAND_LFSR 2 + +#define TEGRA30_AUDIOCIF_CTRL_EXPAND_SHIFT 6 +#define TEGRA30_AUDIOCIF_CTRL_EXPAND_MASK (3 << TEGRA30_AUDIOCIF_CTRL_EXPAND_SHIFT) +#define TEGRA30_AUDIOCIF_CTRL_EXPAND_ZERO (TEGRA30_AUDIOCIF_EXPAND_ZERO << TEGRA30_AUDIOCIF_CTRL_EXPAND_SHIFT) +#define TEGRA30_AUDIOCIF_CTRL_EXPAND_ONE (TEGRA30_AUDIOCIF_EXPAND_ONE << TEGRA30_AUDIOCIF_CTRL_EXPAND_SHIFT) +#define TEGRA30_AUDIOCIF_CTRL_EXPAND_LFSR (TEGRA30_AUDIOCIF_EXPAND_LFSR << TEGRA30_AUDIOCIF_CTRL_EXPAND_SHIFT) + +#define TEGRA30_AUDIOCIF_STEREO_CONV_CH0 0 +#define TEGRA30_AUDIOCIF_STEREO_CONV_CH1 1 +#define TEGRA30_AUDIOCIF_STEREO_CONV_AVG 2 + +#define TEGRA30_AUDIOCIF_CTRL_STEREO_CONV_SHIFT 4 +#define TEGRA30_AUDIOCIF_CTRL_STEREO_CONV_MASK (3 << TEGRA30_AUDIOCIF_CTRL_STEREO_CONV_SHIFT) +#define TEGRA30_AUDIOCIF_CTRL_STEREO_CONV_CH0 (TEGRA30_AUDIOCIF_STEREO_CONV_CH0 << TEGRA30_AUDIOCIF_CTRL_STEREO_CONV_SHIFT) +#define TEGRA30_AUDIOCIF_CTRL_STEREO_CONV_CH1 (TEGRA30_AUDIOCIF_STEREO_CONV_CH1 << TEGRA30_AUDIOCIF_CTRL_STEREO_CONV_SHIFT) +#define TEGRA30_AUDIOCIF_CTRL_STEREO_CONV_AVG (TEGRA30_AUDIOCIF_STEREO_CONV_AVG << TEGRA30_AUDIOCIF_CTRL_STEREO_CONV_SHIFT) + +#define TEGRA30_AUDIOCIF_CTRL_REPLICATE 3 + +#define TEGRA30_AUDIOCIF_DIRECTION_TX 0 +#define TEGRA30_AUDIOCIF_DIRECTION_RX 1 + +#define TEGRA30_AUDIOCIF_CTRL_DIRECTION_SHIFT 2 +#define TEGRA30_AUDIOCIF_CTRL_DIRECTION_MASK (1 << TEGRA30_AUDIOCIF_CTRL_DIRECTION_SHIFT) +#define TEGRA30_AUDIOCIF_CTRL_DIRECTION_TX (TEGRA30_AUDIOCIF_DIRECTION_TX << TEGRA30_AUDIOCIF_CTRL_DIRECTION_SHIFT) +#define TEGRA30_AUDIOCIF_CTRL_DIRECTION_RX (TEGRA30_AUDIOCIF_DIRECTION_RX << TEGRA30_AUDIOCIF_CTRL_DIRECTION_SHIFT) + +#define TEGRA30_AUDIOCIF_TRUNCATE_ROUND 0 +#define TEGRA30_AUDIOCIF_TRUNCATE_CHOP 1 + +#define TEGRA30_AUDIOCIF_CTRL_TRUNCATE_SHIFT 1 +#define TEGRA30_AUDIOCIF_CTRL_TRUNCATE_MASK (1 << TEGRA30_AUDIOCIF_CTRL_TRUNCATE_SHIFT) +#define TEGRA30_AUDIOCIF_CTRL_TRUNCATE_ROUND (TEGRA30_AUDIOCIF_TRUNCATE_ROUND << TEGRA30_AUDIOCIF_CTRL_TRUNCATE_SHIFT) +#define TEGRA30_AUDIOCIF_CTRL_TRUNCATE_CHOP (TEGRA30_AUDIOCIF_TRUNCATE_CHOP << TEGRA30_AUDIOCIF_CTRL_TRUNCATE_SHIFT) + +#define TEGRA30_AUDIOCIF_MONO_CONV_ZERO 0 +#define TEGRA30_AUDIOCIF_MONO_CONV_COPY 1 + +#define TEGRA30_AUDIOCIF_CTRL_MONO_CONV_SHIFT 0 +#define TEGRA30_AUDIOCIF_CTRL_MONO_CONV_MASK (1 << TEGRA30_AUDIOCIF_CTRL_MONO_CONV_SHIFT) +#define TEGRA30_AUDIOCIF_CTRL_MONO_CONV_ZERO (TEGRA30_AUDIOCIF_MONO_CONV_ZERO << TEGRA30_AUDIOCIF_CTRL_MONO_CONV_SHIFT) +#define TEGRA30_AUDIOCIF_CTRL_MONO_CONV_COPY (TEGRA30_AUDIOCIF_MONO_CONV_COPY << TEGRA30_AUDIOCIF_CTRL_MONO_CONV_SHIFT) + +/* Registers within TEGRA30_AUDIO_CLUSTER_BASE */ + +/* TEGRA30_AHUB_CHANNEL_CTRL */ + +#define TEGRA30_AHUB_CHANNEL_CTRL 0x0 +#define TEGRA30_AHUB_CHANNEL_CTRL_STRIDE 0x20 +#define TEGRA30_AHUB_CHANNEL_CTRL_COUNT 4 +#define TEGRA30_AHUB_CHANNEL_CTRL_TX_EN (1 << 31) +#define TEGRA30_AHUB_CHANNEL_CTRL_RX_EN (1 << 30) +#define TEGRA30_AHUB_CHANNEL_CTRL_LOOPBACK (1 << 29) + +#define TEGRA30_AHUB_CHANNEL_CTRL_TX_THRESHOLD_SHIFT 16 +#define TEGRA30_AHUB_CHANNEL_CTRL_TX_THRESHOLD_MASK_US 0xff +#define TEGRA30_AHUB_CHANNEL_CTRL_TX_THRESHOLD_MASK (TEGRA30_AHUB_CHANNEL_CTRL_TX_THRESHOLD_MASK_US << TEGRA30_AHUB_CHANNEL_CTRL_TX_THRESHOLD_SHIFT) + +#define TEGRA30_AHUB_CHANNEL_CTRL_RX_THRESHOLD_SHIFT 8 +#define TEGRA30_AHUB_CHANNEL_CTRL_RX_THRESHOLD_MASK_US 0xff +#define TEGRA30_AHUB_CHANNEL_CTRL_RX_THRESHOLD_MASK (TEGRA30_AHUB_CHANNEL_CTRL_RX_THRESHOLD_MASK_US << TEGRA30_AHUB_CHANNEL_CTRL_RX_THRESHOLD_SHIFT) + +#define TEGRA30_AHUB_CHANNEL_CTRL_TX_PACK_EN (1 << 6) + +#define TEGRA30_PACK_8_4 2 +#define TEGRA30_PACK_16 3 + +#define TEGRA30_AHUB_CHANNEL_CTRL_TX_PACK_SHIFT 4 +#define TEGRA30_AHUB_CHANNEL_CTRL_TX_PACK_MASK_US 3 +#define TEGRA30_AHUB_CHANNEL_CTRL_TX_PACK_MASK (TEGRA30_AHUB_CHANNEL_CTRL_TX_PACK_MASK_US << TEGRA30_AHUB_CHANNEL_CTRL_TX_PACK_SHIFT) +#define TEGRA30_AHUB_CHANNEL_CTRL_TX_PACK_8_4 (TEGRA30_PACK_8_4 << TEGRA30_AHUB_CHANNEL_CTRL_TX_PACK_SHIFT) +#define TEGRA30_AHUB_CHANNEL_CTRL_TX_PACK_16 (TEGRA30_PACK_16 << TEGRA30_AHUB_CHANNEL_CTRL_TX_PACK_SHIFT) + +#define TEGRA30_AHUB_CHANNEL_CTRL_RX_PACK_EN (1 << 2) + +#define TEGRA30_AHUB_CHANNEL_CTRL_RX_PACK_SHIFT 0 +#define TEGRA30_AHUB_CHANNEL_CTRL_RX_PACK_MASK_US 3 +#define TEGRA30_AHUB_CHANNEL_CTRL_RX_PACK_MASK (TEGRA30_AHUB_CHANNEL_CTRL_RX_PACK_MASK_US << TEGRA30_AHUB_CHANNEL_CTRL_RX_PACK_SHIFT) +#define TEGRA30_AHUB_CHANNEL_CTRL_RX_PACK_8_4 (TEGRA30_PACK_8_4 << TEGRA30_AHUB_CHANNEL_CTRL_RX_PACK_SHIFT) +#define TEGRA30_AHUB_CHANNEL_CTRL_RX_PACK_16 (TEGRA30_PACK_16 << TEGRA30_AHUB_CHANNEL_CTRL_RX_PACK_SHIFT) + +/* TEGRA30_AHUB_CHANNEL_CLEAR */ + +#define TEGRA30_AHUB_CHANNEL_CLEAR 0x4 +#define TEGRA30_AHUB_CHANNEL_CLEAR_STRIDE 0x20 +#define TEGRA30_AHUB_CHANNEL_CLEAR_COUNT 4 +#define TEGRA30_AHUB_CHANNEL_CLEAR_TX_SOFT_RESET (1 << 31) +#define TEGRA30_AHUB_CHANNEL_CLEAR_RX_SOFT_RESET (1 << 30) + +/* TEGRA30_AHUB_CHANNEL_STATUS */ + +#define TEGRA30_AHUB_CHANNEL_STATUS 0x8 +#define TEGRA30_AHUB_CHANNEL_STATUS_STRIDE 0x20 +#define TEGRA30_AHUB_CHANNEL_STATUS_COUNT 4 +#define TEGRA30_AHUB_CHANNEL_STATUS_TX_FREE_SHIFT 24 +#define TEGRA30_AHUB_CHANNEL_STATUS_TX_FREE_MASK_US 0xff +#define TEGRA30_AHUB_CHANNEL_STATUS_TX_FREE_MASK (TEGRA30_AHUB_CHANNEL_STATUS_TX_FREE_MASK_US << TEGRA30_AHUB_CHANNEL_STATUS_TX_FREE_SHIFT) +#define TEGRA30_AHUB_CHANNEL_STATUS_RX_FREE_SHIFT 16 +#define TEGRA30_AHUB_CHANNEL_STATUS_RX_FREE_MASK_US 0xff +#define TEGRA30_AHUB_CHANNEL_STATUS_RX_FREE_MASK (TEGRA30_AHUB_CHANNEL_STATUS_RX_FREE_MASK_US << TEGRA30_AHUB_CHANNEL_STATUS_RX_FREE_SHIFT) +#define TEGRA30_AHUB_CHANNEL_STATUS_TX_TRIG (1 << 1) +#define TEGRA30_AHUB_CHANNEL_STATUS_RX_TRIG (1 << 0) + +/* TEGRA30_AHUB_CHANNEL_TXFIFO */ + +#define TEGRA30_AHUB_CHANNEL_TXFIFO 0xc +#define TEGRA30_AHUB_CHANNEL_TXFIFO_STRIDE 0x20 +#define TEGRA30_AHUB_CHANNEL_TXFIFO_COUNT 4 + +/* TEGRA30_AHUB_CHANNEL_RXFIFO */ + +#define TEGRA30_AHUB_CHANNEL_RXFIFO 0x10 +#define TEGRA30_AHUB_CHANNEL_RXFIFO_STRIDE 0x20 +#define TEGRA30_AHUB_CHANNEL_RXFIFO_COUNT 4 + +/* TEGRA30_AHUB_CIF_TX_CTRL */ + +#define TEGRA30_AHUB_CIF_TX_CTRL 0x14 +#define TEGRA30_AHUB_CIF_TX_CTRL_STRIDE 0x20 +#define TEGRA30_AHUB_CIF_TX_CTRL_COUNT 4 +/* Uses field from TEGRA30_AUDIOCIF_CTRL_* */ + +/* TEGRA30_AHUB_CIF_RX_CTRL */ + +#define TEGRA30_AHUB_CIF_RX_CTRL 0x18 +#define TEGRA30_AHUB_CIF_RX_CTRL_STRIDE 0x20 +#define TEGRA30_AHUB_CIF_RX_CTRL_COUNT 4 +/* Uses field from TEGRA30_AUDIOCIF_CTRL_* */ + +/* TEGRA30_AHUB_CONFIG_LINK_CTRL */ + +#define TEGRA30_AHUB_CONFIG_LINK_CTRL 0x80 +#define TEGRA30_AHUB_CONFIG_LINK_CTRL_MASTER_FIFO_FULL_CNT_SHIFT 28 +#define TEGRA30_AHUB_CONFIG_LINK_CTRL_MASTER_FIFO_FULL_CNT_MASK_US 0xf +#define TEGRA30_AHUB_CONFIG_LINK_CTRL_MASTER_FIFO_FULL_CNT_MASK (TEGRA30_AHUB_CONFIG_LINK_CTRL_MASTER_FIFO_FULL_CNT_MASK_US << TEGRA30_AHUB_CONFIG_LINK_CTRL_MASTER_FIFO_FULL_CNT_SHIFT) +#define TEGRA30_AHUB_CONFIG_LINK_CTRL_TIMEOUT_CNT_SHIFT 16 +#define TEGRA30_AHUB_CONFIG_LINK_CTRL_TIMEOUT_CNT_MASK_US 0xfff +#define TEGRA30_AHUB_CONFIG_LINK_CTRL_TIMEOUT_CNT_MASK (TEGRA30_AHUB_CONFIG_LINK_CTRL_TIMEOUT_CNT_MASK_US << TEGRA30_AHUB_CONFIG_LINK_CTRL_TIMEOUT_CNT_SHIFT) +#define TEGRA30_AHUB_CONFIG_LINK_CTRL_IDLE_CNT_SHIFT 4 +#define TEGRA30_AHUB_CONFIG_LINK_CTRL_IDLE_CNT_MASK_US 0xfff +#define TEGRA30_AHUB_CONFIG_LINK_CTRL_IDLE_CNT_MASK (TEGRA30_AHUB_CONFIG_LINK_CTRL_IDLE_CNT_MASK_US << TEGRA30_AHUB_CONFIG_LINK_CTRL_IDLE_CNT_SHIFT) +#define TEGRA30_AHUB_CONFIG_LINK_CTRL_CG_EN (1 << 2) +#define TEGRA30_AHUB_CONFIG_LINK_CTRL_CLEAR_TIMEOUT_CNTR (1 << 1) +#define TEGRA30_AHUB_CONFIG_LINK_CTRL_SOFT_RESET (1 << 0) + +/* TEGRA30_AHUB_MISC_CTRL */ + +#define TEGRA30_AHUB_MISC_CTRL 0x84 +#define TEGRA30_AHUB_MISC_CTRL_AUDIO_ACTIVE (1 << 31) +#define TEGRA30_AHUB_MISC_CTRL_AUDIO_CG_EN (1 << 8) +#define TEGRA30_AHUB_MISC_CTRL_AUDIO_OBS_SEL_SHIFT 0 +#define TEGRA30_AHUB_MISC_CTRL_AUDIO_OBS_SEL_MASK (0x1f << TEGRA30_AHUB_MISC_CTRL_AUDIO_OBS_SEL_SHIFT) + +/* TEGRA30_AHUB_APBDMA_LIVE_STATUS */ + +#define TEGRA30_AHUB_APBDMA_LIVE_STATUS 0x88 +#define TEGRA30_AHUB_APBDMA_LIVE_STATUS_CH3_RX_CIF_FIFO_FULL (1 << 31) +#define TEGRA30_AHUB_APBDMA_LIVE_STATUS_CH3_TX_CIF_FIFO_FULL (1 << 30) +#define TEGRA30_AHUB_APBDMA_LIVE_STATUS_CH2_RX_CIF_FIFO_FULL (1 << 29) +#define TEGRA30_AHUB_APBDMA_LIVE_STATUS_CH2_TX_CIF_FIFO_FULL (1 << 28) +#define TEGRA30_AHUB_APBDMA_LIVE_STATUS_CH1_RX_CIF_FIFO_FULL (1 << 27) +#define TEGRA30_AHUB_APBDMA_LIVE_STATUS_CH1_TX_CIF_FIFO_FULL (1 << 26) +#define TEGRA30_AHUB_APBDMA_LIVE_STATUS_CH0_RX_CIF_FIFO_FULL (1 << 25) +#define TEGRA30_AHUB_APBDMA_LIVE_STATUS_CH0_TX_CIF_FIFO_FULL (1 << 24) +#define TEGRA30_AHUB_APBDMA_LIVE_STATUS_CH3_RX_CIF_FIFO_EMPTY (1 << 23) +#define TEGRA30_AHUB_APBDMA_LIVE_STATUS_CH3_TX_CIF_FIFO_EMPTY (1 << 22) +#define TEGRA30_AHUB_APBDMA_LIVE_STATUS_CH2_RX_CIF_FIFO_EMPTY (1 << 21) +#define TEGRA30_AHUB_APBDMA_LIVE_STATUS_CH2_TX_CIF_FIFO_EMPTY (1 << 20) +#define TEGRA30_AHUB_APBDMA_LIVE_STATUS_CH1_RX_CIF_FIFO_EMPTY (1 << 19) +#define TEGRA30_AHUB_APBDMA_LIVE_STATUS_CH1_TX_CIF_FIFO_EMPTY (1 << 18) +#define TEGRA30_AHUB_APBDMA_LIVE_STATUS_CH0_RX_CIF_FIFO_EMPTY (1 << 17) +#define TEGRA30_AHUB_APBDMA_LIVE_STATUS_CH0_TX_CIF_FIFO_EMPTY (1 << 16) +#define TEGRA30_AHUB_APBDMA_LIVE_STATUS_CH3_RX_DMA_FIFO_FULL (1 << 15) +#define TEGRA30_AHUB_APBDMA_LIVE_STATUS_CH3_TX_DMA_FIFO_FULL (1 << 14) +#define TEGRA30_AHUB_APBDMA_LIVE_STATUS_CH2_RX_DMA_FIFO_FULL (1 << 13) +#define TEGRA30_AHUB_APBDMA_LIVE_STATUS_CH2_TX_DMA_FIFO_FULL (1 << 12) +#define TEGRA30_AHUB_APBDMA_LIVE_STATUS_CH1_RX_DMA_FIFO_FULL (1 << 11) +#define TEGRA30_AHUB_APBDMA_LIVE_STATUS_CH1_TX_DMA_FIFO_FULL (1 << 10) +#define TEGRA30_AHUB_APBDMA_LIVE_STATUS_CH0_RX_DMA_FIFO_FULL (1 << 9) +#define TEGRA30_AHUB_APBDMA_LIVE_STATUS_CH0_TX_DMA_FIFO_FULL (1 << 8) +#define TEGRA30_AHUB_APBDMA_LIVE_STATUS_CH3_RX_DMA_FIFO_EMPTY (1 << 7) +#define TEGRA30_AHUB_APBDMA_LIVE_STATUS_CH3_TX_DMA_FIFO_EMPTY (1 << 6) +#define TEGRA30_AHUB_APBDMA_LIVE_STATUS_CH2_RX_DMA_FIFO_EMPTY (1 << 5) +#define TEGRA30_AHUB_APBDMA_LIVE_STATUS_CH2_TX_DMA_FIFO_EMPTY (1 << 4) +#define TEGRA30_AHUB_APBDMA_LIVE_STATUS_CH1_RX_DMA_FIFO_EMPTY (1 << 3) +#define TEGRA30_AHUB_APBDMA_LIVE_STATUS_CH1_TX_DMA_FIFO_EMPTY (1 << 2) +#define TEGRA30_AHUB_APBDMA_LIVE_STATUS_CH0_RX_DMA_FIFO_EMPTY (1 << 1) +#define TEGRA30_AHUB_APBDMA_LIVE_STATUS_CH0_TX_DMA_FIFO_EMPTY (1 << 0) + +/* TEGRA30_AHUB_I2S_LIVE_STATUS */ + +#define TEGRA30_AHUB_I2S_LIVE_STATUS 0x8c +#define TEGRA30_AHUB_I2S_LIVE_STATUS_I2S4_RX_FIFO_FULL (1 << 29) +#define TEGRA30_AHUB_I2S_LIVE_STATUS_I2S4_TX_FIFO_FULL (1 << 28) +#define TEGRA30_AHUB_I2S_LIVE_STATUS_I2S3_RX_FIFO_FULL (1 << 27) +#define TEGRA30_AHUB_I2S_LIVE_STATUS_I2S3_TX_FIFO_FULL (1 << 26) +#define TEGRA30_AHUB_I2S_LIVE_STATUS_I2S2_RX_FIFO_FULL (1 << 25) +#define TEGRA30_AHUB_I2S_LIVE_STATUS_I2S2_TX_FIFO_FULL (1 << 24) +#define TEGRA30_AHUB_I2S_LIVE_STATUS_I2S1_RX_FIFO_FULL (1 << 23) +#define TEGRA30_AHUB_I2S_LIVE_STATUS_I2S1_TX_FIFO_FULL (1 << 22) +#define TEGRA30_AHUB_I2S_LIVE_STATUS_I2S0_RX_FIFO_FULL (1 << 21) +#define TEGRA30_AHUB_I2S_LIVE_STATUS_I2S0_TX_FIFO_FULL (1 << 20) +#define TEGRA30_AHUB_I2S_LIVE_STATUS_I2S4_RX_FIFO_ENABLED (1 << 19) +#define TEGRA30_AHUB_I2S_LIVE_STATUS_I2S4_TX_FIFO_ENABLED (1 << 18) +#define TEGRA30_AHUB_I2S_LIVE_STATUS_I2S3_RX_FIFO_ENABLED (1 << 17) +#define TEGRA30_AHUB_I2S_LIVE_STATUS_I2S3_TX_FIFO_ENABLED (1 << 16) +#define TEGRA30_AHUB_I2S_LIVE_STATUS_I2S2_RX_FIFO_ENABLED (1 << 15) +#define TEGRA30_AHUB_I2S_LIVE_STATUS_I2S2_TX_FIFO_ENABLED (1 << 14) +#define TEGRA30_AHUB_I2S_LIVE_STATUS_I2S1_RX_FIFO_ENABLED (1 << 13) +#define TEGRA30_AHUB_I2S_LIVE_STATUS_I2S1_TX_FIFO_ENABLED (1 << 12) +#define TEGRA30_AHUB_I2S_LIVE_STATUS_I2S0_RX_FIFO_ENABLED (1 << 11) +#define TEGRA30_AHUB_I2S_LIVE_STATUS_I2S0_TX_FIFO_ENABLED (1 << 10) +#define TEGRA30_AHUB_I2S_LIVE_STATUS_I2S4_RX_FIFO_EMPTY (1 << 9) +#define TEGRA30_AHUB_I2S_LIVE_STATUS_I2S4_TX_FIFO_EMPTY (1 << 8) +#define TEGRA30_AHUB_I2S_LIVE_STATUS_I2S3_RX_FIFO_EMPTY (1 << 7) +#define TEGRA30_AHUB_I2S_LIVE_STATUS_I2S3_TX_FIFO_EMPTY (1 << 6) +#define TEGRA30_AHUB_I2S_LIVE_STATUS_I2S2_RX_FIFO_EMPTY (1 << 5) +#define TEGRA30_AHUB_I2S_LIVE_STATUS_I2S2_TX_FIFO_EMPTY (1 << 4) +#define TEGRA30_AHUB_I2S_LIVE_STATUS_I2S1_RX_FIFO_EMPTY (1 << 3) +#define TEGRA30_AHUB_I2S_LIVE_STATUS_I2S1_TX_FIFO_EMPTY (1 << 2) +#define TEGRA30_AHUB_I2S_LIVE_STATUS_I2S0_RX_FIFO_EMPTY (1 << 1) +#define TEGRA30_AHUB_I2S_LIVE_STATUS_I2S0_TX_FIFO_EMPTY (1 << 0) + +/* TEGRA30_AHUB_DAM0_LIVE_STATUS */ + +#define TEGRA30_AHUB_DAM_LIVE_STATUS 0x90 +#define TEGRA30_AHUB_DAM_LIVE_STATUS_STRIDE 0x8 +#define TEGRA30_AHUB_DAM_LIVE_STATUS_COUNT 3 +#define TEGRA30_AHUB_DAM_LIVE_STATUS_TX_ENABLED (1 << 26) +#define TEGRA30_AHUB_DAM_LIVE_STATUS_RX1_ENABLED (1 << 25) +#define TEGRA30_AHUB_DAM_LIVE_STATUS_RX0_ENABLED (1 << 24) +#define TEGRA30_AHUB_DAM_LIVE_STATUS_TXFIFO_FULL (1 << 15) +#define TEGRA30_AHUB_DAM_LIVE_STATUS_RX1FIFO_FULL (1 << 9) +#define TEGRA30_AHUB_DAM_LIVE_STATUS_RX0FIFO_FULL (1 << 8) +#define TEGRA30_AHUB_DAM_LIVE_STATUS_TXFIFO_EMPTY (1 << 7) +#define TEGRA30_AHUB_DAM_LIVE_STATUS_RX1FIFO_EMPTY (1 << 1) +#define TEGRA30_AHUB_DAM_LIVE_STATUS_RX0FIFO_EMPTY (1 << 0) + +/* TEGRA30_AHUB_SPDIF_LIVE_STATUS */ + +#define TEGRA30_AHUB_SPDIF_LIVE_STATUS 0xa8 +#define TEGRA30_AHUB_SPDIF_LIVE_STATUS_USER_TX_ENABLED (1 << 11) +#define TEGRA30_AHUB_SPDIF_LIVE_STATUS_USER_RX_ENABLED (1 << 10) +#define TEGRA30_AHUB_SPDIF_LIVE_STATUS_DATA_TX_ENABLED (1 << 9) +#define TEGRA30_AHUB_SPDIF_LIVE_STATUS_DATA_RX_ENABLED (1 << 8) +#define TEGRA30_AHUB_SPDIF_LIVE_STATUS_USER_TXFIFO_FULL (1 << 7) +#define TEGRA30_AHUB_SPDIF_LIVE_STATUS_USER_RXFIFO_FULL (1 << 6) +#define TEGRA30_AHUB_SPDIF_LIVE_STATUS_DATA_TXFIFO_FULL (1 << 5) +#define TEGRA30_AHUB_SPDIF_LIVE_STATUS_DATA_RXFIFO_FULL (1 << 4) +#define TEGRA30_AHUB_SPDIF_LIVE_STATUS_USER_TXFIFO_EMPTY (1 << 3) +#define TEGRA30_AHUB_SPDIF_LIVE_STATUS_USER_RXFIFO_EMPTY (1 << 2) +#define TEGRA30_AHUB_SPDIF_LIVE_STATUS_DATA_TXFIFO_EMPTY (1 << 1) +#define TEGRA30_AHUB_SPDIF_LIVE_STATUS_DATA_RXFIFO_EMPTY (1 << 0) + +/* TEGRA30_AHUB_I2S_INT_MASK */ + +#define TEGRA30_AHUB_I2S_INT_MASK 0xb0 + +/* TEGRA30_AHUB_DAM_INT_MASK */ + +#define TEGRA30_AHUB_DAM_INT_MASK 0xb4 + +/* TEGRA30_AHUB_SPDIF_INT_MASK */ + +#define TEGRA30_AHUB_SPDIF_INT_MASK 0xbc + +/* TEGRA30_AHUB_APBIF_INT_MASK */ + +#define TEGRA30_AHUB_APBIF_INT_MASK 0xc0 + +/* TEGRA30_AHUB_I2S_INT_STATUS */ + +#define TEGRA30_AHUB_I2S_INT_STATUS 0xc8 + +/* TEGRA30_AHUB_DAM_INT_STATUS */ + +#define TEGRA30_AHUB_DAM_INT_STATUS 0xcc + +/* TEGRA30_AHUB_SPDIF_INT_STATUS */ + +#define TEGRA30_AHUB_SPDIF_INT_STATUS 0xd4 + +/* TEGRA30_AHUB_APBIF_INT_STATUS */ + +#define TEGRA30_AHUB_APBIF_INT_STATUS 0xd8 + +/* TEGRA30_AHUB_I2S_INT_SOURCE */ + +#define TEGRA30_AHUB_I2S_INT_SOURCE 0xe0 + +/* TEGRA30_AHUB_DAM_INT_SOURCE */ + +#define TEGRA30_AHUB_DAM_INT_SOURCE 0xe4 + +/* TEGRA30_AHUB_SPDIF_INT_SOURCE */ + +#define TEGRA30_AHUB_SPDIF_INT_SOURCE 0xec + +/* TEGRA30_AHUB_APBIF_INT_SOURCE */ + +#define TEGRA30_AHUB_APBIF_INT_SOURCE 0xf0 + +/* TEGRA30_AHUB_I2S_INT_SET */ + +#define TEGRA30_AHUB_I2S_INT_SET 0xf8 + +/* TEGRA30_AHUB_DAM_INT_SET */ + +#define TEGRA30_AHUB_DAM_INT_SET 0xfc + +/* TEGRA30_AHUB_SPDIF_INT_SET */ + +#define TEGRA30_AHUB_SPDIF_INT_SET 0x100 + +/* TEGRA30_AHUB_APBIF_INT_SET */ + +#define TEGRA30_AHUB_APBIF_INT_SET 0x104 + +/* Registers within TEGRA30_AHUB_BASE */ + +#define TEGRA30_AHUB_AUDIO_RX 0x0 +#define TEGRA30_AHUB_AUDIO_RX_STRIDE 0x4 +#define TEGRA30_AHUB_AUDIO_RX_COUNT 17 +/* This register repeats once for each entry in enum tegra30_ahub_rxcif */ +/* The fields in this register are 1 bit per entry in tegra30_ahub_txcif */ + +/* + * Terminology: + * AHUB: Audio Hub; a cross-bar switch between the audio devices: DMA FIFOs, + * I2S controllers, SPDIF controllers, and DAMs. + * XBAR: The core cross-bar component of the AHUB. + * CIF: Client Interface; the HW module connecting an audio device to the + * XBAR. + * DAM: Digital Audio Mixer: A HW module that mixes multiple audio streams, + * possibly including sample-rate conversion. + * + * Each TX CIF transmits data into the XBAR. Each RX CIF can receive audio + * transmitted by a particular TX CIF. + * + * This driver is currently very simplistic; many HW features are not + * exposed; DAMs are not supported, only 16-bit stereo audio is supported, + * etc. + */ + +enum tegra30_ahub_txcif { + TEGRA30_AHUB_TXCIF_APBIF_TX0, + TEGRA30_AHUB_TXCIF_APBIF_TX1, + TEGRA30_AHUB_TXCIF_APBIF_TX2, + TEGRA30_AHUB_TXCIF_APBIF_TX3, + TEGRA30_AHUB_TXCIF_I2S0_TX0, + TEGRA30_AHUB_TXCIF_I2S1_TX0, + TEGRA30_AHUB_TXCIF_I2S2_TX0, + TEGRA30_AHUB_TXCIF_I2S3_TX0, + TEGRA30_AHUB_TXCIF_I2S4_TX0, + TEGRA30_AHUB_TXCIF_DAM0_TX0, + TEGRA30_AHUB_TXCIF_DAM1_TX0, + TEGRA30_AHUB_TXCIF_DAM2_TX0, + TEGRA30_AHUB_TXCIF_SPDIF_TX0, + TEGRA30_AHUB_TXCIF_SPDIF_TX1, +}; + +enum tegra30_ahub_rxcif { + TEGRA30_AHUB_RXCIF_APBIF_RX0, + TEGRA30_AHUB_RXCIF_APBIF_RX1, + TEGRA30_AHUB_RXcIF_APBIF_RX2, + TEGRA30_AHUB_RXCIF_APBIF_RX3, + TEGRA30_AHUB_RXCIF_I2S0_RX0, + TEGRA30_AHUB_RXCIF_I2S1_RX0, + TEGRA30_AHUB_RXCIF_I2S2_RX0, + TEGRA30_AHUB_RXCIF_I2S3_RX0, + TEGRA30_AHUB_RXCIF_I2S4_RX0, + TEGRA30_AHUB_RXCIF_DAM0_RX0, + TEGRA30_AHUB_RXCIF_DAM0_RX1, + TEGRA30_AHUB_RXCIF_DAM1_RX0, + TEGRA30_AHUB_RXCIF_DAM2_RX1, + TEGRA30_AHUB_RXCIF_DAM3_RX0, + TEGRA30_AHUB_RXCIF_DAM3_RX1, + TEGRA30_AHUB_RXCIF_SPDIF_RX0, + TEGRA30_AHUB_RXCIF_SPDIF_RX1, +}; + +extern int tegra30_ahub_allocate_rx_fifo(enum tegra30_ahub_rxcif *rxcif, + unsigned long *fiforeg, + unsigned long *reqsel); +extern int tegra30_ahub_enable_rx_fifo(enum tegra30_ahub_rxcif rxcif); +extern int tegra30_ahub_disable_rx_fifo(enum tegra30_ahub_rxcif rxcif); +extern int tegra30_ahub_free_rx_fifo(enum tegra30_ahub_rxcif rxcif); + +extern int tegra30_ahub_allocate_tx_fifo(enum tegra30_ahub_txcif *txcif, + unsigned long *fiforeg, + unsigned long *reqsel); +extern int tegra30_ahub_enable_tx_fifo(enum tegra30_ahub_txcif txcif); +extern int tegra30_ahub_disable_tx_fifo(enum tegra30_ahub_txcif txcif); +extern int tegra30_ahub_free_tx_fifo(enum tegra30_ahub_txcif txcif); + +extern int tegra30_ahub_set_rx_cif_source(enum tegra30_ahub_rxcif rxcif, + enum tegra30_ahub_txcif txcif); +extern int tegra30_ahub_unset_rx_cif_source(enum tegra30_ahub_rxcif rxcif); + +struct tegra30_ahub { + struct device *dev; + struct clk *clk_d_audio; + struct clk *clk_apbif; + int dma_sel; + resource_size_t apbif_addr; + struct regmap *regmap_apbif; + struct regmap *regmap_ahub; + DECLARE_BITMAP(rx_usage, TEGRA30_AHUB_CHANNEL_CTRL_COUNT); + DECLARE_BITMAP(tx_usage, TEGRA30_AHUB_CHANNEL_CTRL_COUNT); +}; + +#endif diff --git a/sound/soc/tegra/tegra30_i2s.c b/sound/soc/tegra/tegra30_i2s.c new file mode 100644 index 00000000000..8596032985d --- /dev/null +++ b/sound/soc/tegra/tegra30_i2s.c @@ -0,0 +1,536 @@ +/* + * tegra30_i2s.c - Tegra30 I2S driver + * + * Author: Stephen Warren <swarren@nvidia.com> + * Copyright (c) 2010-2012, NVIDIA CORPORATION. All rights reserved. + * + * Based on code copyright/by: + * + * Copyright (c) 2009-2010, NVIDIA Corporation. + * Scott Peterson <speterson@nvidia.com> + * + * Copyright (C) 2010 Google, Inc. + * Iliyan Malchev <malchev@google.com> + * + * This program is free software; you can redistribute it and/or modify it + * under the terms and conditions of the GNU General Public License, + * version 2, as published by the Free Software Foundation. + * + * This program is distributed in the hope it will be useful, but WITHOUT + * ANY WARRANTY; without even the implied warranty of MERCHANTABILITY or + * FITNESS FOR A PARTICULAR PURPOSE. See the GNU General Public License for + * more details. + * + * You should have received a copy of the GNU General Public License + * along with this program. If not, see <http://www.gnu.org/licenses/>. + */ + +#include <linux/clk.h> +#include <linux/device.h> +#include <linux/io.h> +#include <linux/module.h> +#include <linux/of.h> +#include <linux/platform_device.h> +#include <linux/pm_runtime.h> +#include <linux/regmap.h> +#include <linux/slab.h> +#include <sound/core.h> +#include <sound/pcm.h> +#include <sound/pcm_params.h> +#include <sound/soc.h> + +#include "tegra30_ahub.h" +#include "tegra30_i2s.h" + +#define DRV_NAME "tegra30-i2s" + +static inline void tegra30_i2s_write(struct tegra30_i2s *i2s, u32 reg, u32 val) +{ + regmap_write(i2s->regmap, reg, val); +} + +static inline u32 tegra30_i2s_read(struct tegra30_i2s *i2s, u32 reg) +{ + u32 val; + regmap_read(i2s->regmap, reg, &val); + return val; +} + +static int tegra30_i2s_runtime_suspend(struct device *dev) +{ + struct tegra30_i2s *i2s = dev_get_drvdata(dev); + + regcache_cache_only(i2s->regmap, true); + + clk_disable(i2s->clk_i2s); + + return 0; +} + +static int tegra30_i2s_runtime_resume(struct device *dev) +{ + struct tegra30_i2s *i2s = dev_get_drvdata(dev); + int ret; + + ret = clk_enable(i2s->clk_i2s); + if (ret) { + dev_err(dev, "clk_enable failed: %d\n", ret); + return ret; + } + + regcache_cache_only(i2s->regmap, false); + + return 0; +} + +int tegra30_i2s_startup(struct snd_pcm_substream *substream, + struct snd_soc_dai *dai) +{ + struct tegra30_i2s *i2s = snd_soc_dai_get_drvdata(dai); + int ret; + + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { + ret = tegra30_ahub_allocate_tx_fifo(&i2s->playback_fifo_cif, + &i2s->playback_dma_data.addr, + &i2s->playback_dma_data.req_sel); + i2s->playback_dma_data.wrap = 4; + i2s->playback_dma_data.width = 32; + tegra30_ahub_set_rx_cif_source(i2s->playback_i2s_cif, + i2s->playback_fifo_cif); + } else { + ret = tegra30_ahub_allocate_rx_fifo(&i2s->capture_fifo_cif, + &i2s->capture_dma_data.addr, + &i2s->capture_dma_data.req_sel); + i2s->capture_dma_data.wrap = 4; + i2s->capture_dma_data.width = 32; + tegra30_ahub_set_rx_cif_source(i2s->capture_fifo_cif, + i2s->capture_i2s_cif); + } + + return ret; +} + +void tegra30_i2s_shutdown(struct snd_pcm_substream *substream, + struct snd_soc_dai *dai) +{ + struct tegra30_i2s *i2s = snd_soc_dai_get_drvdata(dai); + + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { + tegra30_ahub_unset_rx_cif_source(i2s->playback_i2s_cif); + tegra30_ahub_free_tx_fifo(i2s->playback_fifo_cif); + } else { + tegra30_ahub_unset_rx_cif_source(i2s->capture_fifo_cif); + tegra30_ahub_free_rx_fifo(i2s->capture_fifo_cif); + } +} + +static int tegra30_i2s_set_fmt(struct snd_soc_dai *dai, + unsigned int fmt) +{ + struct tegra30_i2s *i2s = snd_soc_dai_get_drvdata(dai); + + switch (fmt & SND_SOC_DAIFMT_INV_MASK) { + case SND_SOC_DAIFMT_NB_NF: + break; + default: + return -EINVAL; + } + + i2s->reg_ctrl &= ~TEGRA30_I2S_CTRL_MASTER_ENABLE; + switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) { + case SND_SOC_DAIFMT_CBS_CFS: + i2s->reg_ctrl |= TEGRA30_I2S_CTRL_MASTER_ENABLE; + break; + case SND_SOC_DAIFMT_CBM_CFM: + break; + default: + return -EINVAL; + } + + i2s->reg_ctrl &= ~(TEGRA30_I2S_CTRL_FRAME_FORMAT_MASK | + TEGRA30_I2S_CTRL_LRCK_MASK); + switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) { + case SND_SOC_DAIFMT_DSP_A: + i2s->reg_ctrl |= TEGRA30_I2S_CTRL_FRAME_FORMAT_FSYNC; + i2s->reg_ctrl |= TEGRA30_I2S_CTRL_LRCK_L_LOW; + break; + case SND_SOC_DAIFMT_DSP_B: + i2s->reg_ctrl |= TEGRA30_I2S_CTRL_FRAME_FORMAT_FSYNC; + i2s->reg_ctrl |= TEGRA30_I2S_CTRL_LRCK_R_LOW; + break; + case SND_SOC_DAIFMT_I2S: + i2s->reg_ctrl |= TEGRA30_I2S_CTRL_FRAME_FORMAT_LRCK; + i2s->reg_ctrl |= TEGRA30_I2S_CTRL_LRCK_L_LOW; + break; + case SND_SOC_DAIFMT_RIGHT_J: + i2s->reg_ctrl |= TEGRA30_I2S_CTRL_FRAME_FORMAT_LRCK; + i2s->reg_ctrl |= TEGRA30_I2S_CTRL_LRCK_L_LOW; + break; + case SND_SOC_DAIFMT_LEFT_J: + i2s->reg_ctrl |= TEGRA30_I2S_CTRL_FRAME_FORMAT_LRCK; + i2s->reg_ctrl |= TEGRA30_I2S_CTRL_LRCK_L_LOW; + break; + default: + return -EINVAL; + } + + return 0; +} + +static int tegra30_i2s_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params, + struct snd_soc_dai *dai) +{ + struct device *dev = substream->pcm->card->dev; + struct tegra30_i2s *i2s = snd_soc_dai_get_drvdata(dai); + u32 val; + int ret, sample_size, srate, i2sclock, bitcnt; + + if (params_channels(params) != 2) + return -EINVAL; + + i2s->reg_ctrl &= ~TEGRA30_I2S_CTRL_BIT_SIZE_MASK; + switch (params_format(params)) { + case SNDRV_PCM_FORMAT_S16_LE: + i2s->reg_ctrl |= TEGRA30_I2S_CTRL_BIT_SIZE_16; + sample_size = 16; + break; + default: + return -EINVAL; + } + + srate = params_rate(params); + + /* Final "* 2" required by Tegra hardware */ + i2sclock = srate * params_channels(params) * sample_size * 2; + + bitcnt = (i2sclock / (2 * srate)) - 1; + if (bitcnt < 0 || bitcnt > TEGRA30_I2S_TIMING_CHANNEL_BIT_COUNT_MASK_US) + return -EINVAL; + + ret = clk_set_rate(i2s->clk_i2s, i2sclock); + if (ret) { + dev_err(dev, "Can't set I2S clock rate: %d\n", ret); + return ret; + } + + val = bitcnt << TEGRA30_I2S_TIMING_CHANNEL_BIT_COUNT_SHIFT; + + if (i2sclock % (2 * srate)) + val |= TEGRA30_I2S_TIMING_NON_SYM_ENABLE; + + tegra30_i2s_write(i2s, TEGRA30_I2S_TIMING, val); + + val = (0 << TEGRA30_AUDIOCIF_CTRL_FIFO_THRESHOLD_SHIFT) | + (1 << TEGRA30_AUDIOCIF_CTRL_AUDIO_CHANNELS_SHIFT) | + (1 << TEGRA30_AUDIOCIF_CTRL_CLIENT_CHANNELS_SHIFT) | + TEGRA30_AUDIOCIF_CTRL_AUDIO_BITS_16 | + TEGRA30_AUDIOCIF_CTRL_CLIENT_BITS_16; + + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { + val |= TEGRA30_AUDIOCIF_CTRL_DIRECTION_RX; + tegra30_i2s_write(i2s, TEGRA30_I2S_CIF_RX_CTRL, val); + } else { + val |= TEGRA30_AUDIOCIF_CTRL_DIRECTION_TX; + tegra30_i2s_write(i2s, TEGRA30_I2S_CIF_TX_CTRL, val); + } + + val = (1 << TEGRA30_I2S_OFFSET_RX_DATA_OFFSET_SHIFT) | + (1 << TEGRA30_I2S_OFFSET_TX_DATA_OFFSET_SHIFT); + tegra30_i2s_write(i2s, TEGRA30_I2S_OFFSET, val); + + return 0; +} + +static void tegra30_i2s_start_playback(struct tegra30_i2s *i2s) +{ + tegra30_ahub_enable_tx_fifo(i2s->playback_fifo_cif); + i2s->reg_ctrl |= TEGRA30_I2S_CTRL_XFER_EN_TX; + tegra30_i2s_write(i2s, TEGRA30_I2S_CTRL, i2s->reg_ctrl); +} + +static void tegra30_i2s_stop_playback(struct tegra30_i2s *i2s) +{ + tegra30_ahub_disable_tx_fifo(i2s->playback_fifo_cif); + i2s->reg_ctrl &= ~TEGRA30_I2S_CTRL_XFER_EN_TX; + tegra30_i2s_write(i2s, TEGRA30_I2S_CTRL, i2s->reg_ctrl); +} + +static void tegra30_i2s_start_capture(struct tegra30_i2s *i2s) +{ + tegra30_ahub_enable_rx_fifo(i2s->capture_fifo_cif); + i2s->reg_ctrl |= TEGRA30_I2S_CTRL_XFER_EN_RX; + tegra30_i2s_write(i2s, TEGRA30_I2S_CTRL, i2s->reg_ctrl); +} + +static void tegra30_i2s_stop_capture(struct tegra30_i2s *i2s) +{ + tegra30_ahub_disable_rx_fifo(i2s->capture_fifo_cif); + i2s->reg_ctrl &= ~TEGRA30_I2S_CTRL_XFER_EN_RX; + tegra30_i2s_write(i2s, TEGRA30_I2S_CTRL, i2s->reg_ctrl); +} + +static int tegra30_i2s_trigger(struct snd_pcm_substream *substream, int cmd, + struct snd_soc_dai *dai) +{ + struct tegra30_i2s *i2s = snd_soc_dai_get_drvdata(dai); + + switch (cmd) { + case SNDRV_PCM_TRIGGER_START: + case SNDRV_PCM_TRIGGER_PAUSE_RELEASE: + case SNDRV_PCM_TRIGGER_RESUME: + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) + tegra30_i2s_start_playback(i2s); + else + tegra30_i2s_start_capture(i2s); + break; + case SNDRV_PCM_TRIGGER_STOP: + case SNDRV_PCM_TRIGGER_PAUSE_PUSH: + case SNDRV_PCM_TRIGGER_SUSPEND: + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) + tegra30_i2s_stop_playback(i2s); + else + tegra30_i2s_stop_capture(i2s); + break; + default: + return -EINVAL; + } + + return 0; +} + +static int tegra30_i2s_probe(struct snd_soc_dai *dai) +{ + struct tegra30_i2s *i2s = snd_soc_dai_get_drvdata(dai); + + dai->capture_dma_data = &i2s->capture_dma_data; + dai->playback_dma_data = &i2s->playback_dma_data; + + return 0; +} + +static struct snd_soc_dai_ops tegra30_i2s_dai_ops = { + .startup = tegra30_i2s_startup, + .shutdown = tegra30_i2s_shutdown, + .set_fmt = tegra30_i2s_set_fmt, + .hw_params = tegra30_i2s_hw_params, + .trigger = tegra30_i2s_trigger, +}; + +static const struct snd_soc_dai_driver tegra30_i2s_dai_template = { + .probe = tegra30_i2s_probe, + .playback = { + .channels_min = 2, + .channels_max = 2, + .rates = SNDRV_PCM_RATE_8000_96000, + .formats = SNDRV_PCM_FMTBIT_S16_LE, + }, + .capture = { + .channels_min = 2, + .channels_max = 2, + .rates = SNDRV_PCM_RATE_8000_96000, + .formats = SNDRV_PCM_FMTBIT_S16_LE, + }, + .ops = &tegra30_i2s_dai_ops, + .symmetric_rates = 1, +}; + +static bool tegra30_i2s_wr_rd_reg(struct device *dev, unsigned int reg) +{ + switch (reg) { + case TEGRA30_I2S_CTRL: + case TEGRA30_I2S_TIMING: + case TEGRA30_I2S_OFFSET: + case TEGRA30_I2S_CH_CTRL: + case TEGRA30_I2S_SLOT_CTRL: + case TEGRA30_I2S_CIF_RX_CTRL: + case TEGRA30_I2S_CIF_TX_CTRL: + case TEGRA30_I2S_FLOWCTL: + case TEGRA30_I2S_TX_STEP: + case TEGRA30_I2S_FLOW_STATUS: + case TEGRA30_I2S_FLOW_TOTAL: + case TEGRA30_I2S_FLOW_OVER: + case TEGRA30_I2S_FLOW_UNDER: + case TEGRA30_I2S_LCOEF_1_4_0: + case TEGRA30_I2S_LCOEF_1_4_1: + case TEGRA30_I2S_LCOEF_1_4_2: + case TEGRA30_I2S_LCOEF_1_4_3: + case TEGRA30_I2S_LCOEF_1_4_4: + case TEGRA30_I2S_LCOEF_1_4_5: + case TEGRA30_I2S_LCOEF_2_4_0: + case TEGRA30_I2S_LCOEF_2_4_1: + case TEGRA30_I2S_LCOEF_2_4_2: + return true; + default: + return false; + }; +} + +static bool tegra30_i2s_volatile_reg(struct device *dev, unsigned int reg) +{ + switch (reg) { + case TEGRA30_I2S_FLOW_STATUS: + case TEGRA30_I2S_FLOW_TOTAL: + case TEGRA30_I2S_FLOW_OVER: + case TEGRA30_I2S_FLOW_UNDER: + return true; + default: + return false; + }; +} + +static const struct regmap_config tegra30_i2s_regmap_config = { + .reg_bits = 32, + .reg_stride = 4, + .val_bits = 32, + .max_register = TEGRA30_I2S_LCOEF_2_4_2, + .writeable_reg = tegra30_i2s_wr_rd_reg, + .readable_reg = tegra30_i2s_wr_rd_reg, + .volatile_reg = tegra30_i2s_volatile_reg, + .cache_type = REGCACHE_RBTREE, +}; + +static __devinit int tegra30_i2s_platform_probe(struct platform_device *pdev) +{ + struct tegra30_i2s *i2s; + u32 cif_ids[2]; + struct resource *mem, *memregion; + void __iomem *regs; + int ret; + + i2s = devm_kzalloc(&pdev->dev, sizeof(struct tegra30_i2s), GFP_KERNEL); + if (!i2s) { + dev_err(&pdev->dev, "Can't allocate tegra30_i2s\n"); + ret = -ENOMEM; + goto err; + } + dev_set_drvdata(&pdev->dev, i2s); + + i2s->dai = tegra30_i2s_dai_template; + i2s->dai.name = dev_name(&pdev->dev); + + ret = of_property_read_u32_array(pdev->dev.of_node, + "nvidia,ahub-cif-ids", cif_ids, + ARRAY_SIZE(cif_ids)); + if (ret < 0) + goto err; + + i2s->playback_i2s_cif = cif_ids[0]; + i2s->capture_i2s_cif = cif_ids[1]; + + i2s->clk_i2s = clk_get(&pdev->dev, NULL); + if (IS_ERR(i2s->clk_i2s)) { + dev_err(&pdev->dev, "Can't retrieve i2s clock\n"); + ret = PTR_ERR(i2s->clk_i2s); + goto err; + } + + mem = platform_get_resource(pdev, IORESOURCE_MEM, 0); + if (!mem) { + dev_err(&pdev->dev, "No memory resource\n"); + ret = -ENODEV; + goto err_clk_put; + } + + memregion = devm_request_mem_region(&pdev->dev, mem->start, + resource_size(mem), DRV_NAME); + if (!memregion) { + dev_err(&pdev->dev, "Memory region already claimed\n"); + ret = -EBUSY; + goto err_clk_put; + } + + regs = devm_ioremap(&pdev->dev, mem->start, resource_size(mem)); + if (!regs) { + dev_err(&pdev->dev, "ioremap failed\n"); + ret = -ENOMEM; + goto err_clk_put; + } + + i2s->regmap = devm_regmap_init_mmio(&pdev->dev, regs, + &tegra30_i2s_regmap_config); + if (IS_ERR(i2s->regmap)) { + dev_err(&pdev->dev, "regmap init failed\n"); + ret = PTR_ERR(i2s->regmap); + goto err_clk_put; + } + regcache_cache_only(i2s->regmap, true); + + pm_runtime_enable(&pdev->dev); + if (!pm_runtime_enabled(&pdev->dev)) { + ret = tegra30_i2s_runtime_resume(&pdev->dev); + if (ret) + goto err_pm_disable; + } + + ret = snd_soc_register_dai(&pdev->dev, &i2s->dai); + if (ret) { + dev_err(&pdev->dev, "Could not register DAI: %d\n", ret); + ret = -ENOMEM; + goto err_suspend; + } + + ret = tegra_pcm_platform_register(&pdev->dev); + if (ret) { + dev_err(&pdev->dev, "Could not register PCM: %d\n", ret); + goto err_unregister_dai; + } + + return 0; + +err_unregister_dai: + snd_soc_unregister_dai(&pdev->dev); +err_suspend: + if (!pm_runtime_status_suspended(&pdev->dev)) + tegra30_i2s_runtime_suspend(&pdev->dev); +err_pm_disable: + pm_runtime_disable(&pdev->dev); +err_clk_put: + clk_put(i2s->clk_i2s); +err: + return ret; +} + +static int __devexit tegra30_i2s_platform_remove(struct platform_device *pdev) +{ + struct tegra30_i2s *i2s = dev_get_drvdata(&pdev->dev); + + pm_runtime_disable(&pdev->dev); + if (!pm_runtime_status_suspended(&pdev->dev)) + tegra30_i2s_runtime_suspend(&pdev->dev); + + tegra_pcm_platform_unregister(&pdev->dev); + snd_soc_unregister_dai(&pdev->dev); + + clk_put(i2s->clk_i2s); + + return 0; +} + +static const struct of_device_id tegra30_i2s_of_match[] __devinitconst = { + { .compatible = "nvidia,tegra30-i2s", }, + {}, +}; + +static const struct dev_pm_ops tegra30_i2s_pm_ops __devinitconst = { + SET_RUNTIME_PM_OPS(tegra30_i2s_runtime_suspend, + tegra30_i2s_runtime_resume, NULL) +}; + +static struct platform_driver tegra30_i2s_driver = { + .driver = { + .name = DRV_NAME, + .owner = THIS_MODULE, + .of_match_table = tegra30_i2s_of_match, + .pm = &tegra30_i2s_pm_ops, + }, + .probe = tegra30_i2s_platform_probe, + .remove = __devexit_p(tegra30_i2s_platform_remove), +}; +module_platform_driver(tegra30_i2s_driver); + +MODULE_AUTHOR("Stephen Warren <swarren@nvidia.com>"); +MODULE_DESCRIPTION("Tegra30 I2S ASoC driver"); +MODULE_LICENSE("GPL"); +MODULE_ALIAS("platform:" DRV_NAME); +MODULE_DEVICE_TABLE(of, tegra30_i2s_of_match); diff --git a/sound/soc/tegra/tegra30_i2s.h b/sound/soc/tegra/tegra30_i2s.h new file mode 100644 index 00000000000..91adf29c7a8 --- /dev/null +++ b/sound/soc/tegra/tegra30_i2s.h @@ -0,0 +1,242 @@ +/* + * tegra30_i2s.h - Definitions for Tegra30 I2S driver + * + * Copyright (c) 2011,2012, NVIDIA CORPORATION. All rights reserved. + * + * This program is free software; you can redistribute it and/or modify it + * under the terms and conditions of the GNU General Public License, + * version 2, as published by the Free Software Foundation. + * + * This program is distributed in the hope it will be useful, but WITHOUT + * ANY WARRANTY; without even the implied warranty of MERCHANTABILITY or + * FITNESS FOR A PARTICULAR PURPOSE. See the GNU General Public License for + * more details. + * + * You should have received a copy of the GNU General Public License + * along with this program. If not, see <http://www.gnu.org/licenses/>. + */ + +#ifndef __TEGRA30_I2S_H__ +#define __TEGRA30_I2S_H__ + +#include "tegra_pcm.h" + +/* Register offsets from TEGRA30_I2S*_BASE */ + +#define TEGRA30_I2S_CTRL 0x0 +#define TEGRA30_I2S_TIMING 0x4 +#define TEGRA30_I2S_OFFSET 0x08 +#define TEGRA30_I2S_CH_CTRL 0x0c +#define TEGRA30_I2S_SLOT_CTRL 0x10 +#define TEGRA30_I2S_CIF_RX_CTRL 0x14 +#define TEGRA30_I2S_CIF_TX_CTRL 0x18 +#define TEGRA30_I2S_FLOWCTL 0x1c +#define TEGRA30_I2S_TX_STEP 0x20 +#define TEGRA30_I2S_FLOW_STATUS 0x24 +#define TEGRA30_I2S_FLOW_TOTAL 0x28 +#define TEGRA30_I2S_FLOW_OVER 0x2c +#define TEGRA30_I2S_FLOW_UNDER 0x30 +#define TEGRA30_I2S_LCOEF_1_4_0 0x34 +#define TEGRA30_I2S_LCOEF_1_4_1 0x38 +#define TEGRA30_I2S_LCOEF_1_4_2 0x3c +#define TEGRA30_I2S_LCOEF_1_4_3 0x40 +#define TEGRA30_I2S_LCOEF_1_4_4 0x44 +#define TEGRA30_I2S_LCOEF_1_4_5 0x48 +#define TEGRA30_I2S_LCOEF_2_4_0 0x4c +#define TEGRA30_I2S_LCOEF_2_4_1 0x50 +#define TEGRA30_I2S_LCOEF_2_4_2 0x54 + +/* Fields in TEGRA30_I2S_CTRL */ + +#define TEGRA30_I2S_CTRL_XFER_EN_TX (1 << 31) +#define TEGRA30_I2S_CTRL_XFER_EN_RX (1 << 30) +#define TEGRA30_I2S_CTRL_CG_EN (1 << 29) +#define TEGRA30_I2S_CTRL_SOFT_RESET (1 << 28) +#define TEGRA30_I2S_CTRL_TX_FLOWCTL_EN (1 << 27) + +#define TEGRA30_I2S_CTRL_OBS_SEL_SHIFT 24 +#define TEGRA30_I2S_CTRL_OBS_SEL_MASK (7 << TEGRA30_I2S_CTRL_OBS_SEL_SHIFT) + +#define TEGRA30_I2S_FRAME_FORMAT_LRCK 0 +#define TEGRA30_I2S_FRAME_FORMAT_FSYNC 1 + +#define TEGRA30_I2S_CTRL_FRAME_FORMAT_SHIFT 12 +#define TEGRA30_I2S_CTRL_FRAME_FORMAT_MASK (7 << TEGRA30_I2S_CTRL_FRAME_FORMAT_SHIFT) +#define TEGRA30_I2S_CTRL_FRAME_FORMAT_LRCK (TEGRA30_I2S_FRAME_FORMAT_LRCK << TEGRA30_I2S_CTRL_FRAME_FORMAT_SHIFT) +#define TEGRA30_I2S_CTRL_FRAME_FORMAT_FSYNC (TEGRA30_I2S_FRAME_FORMAT_FSYNC << TEGRA30_I2S_CTRL_FRAME_FORMAT_SHIFT) + +#define TEGRA30_I2S_CTRL_MASTER_ENABLE (1 << 10) + +#define TEGRA30_I2S_LRCK_LEFT_LOW 0 +#define TEGRA30_I2S_LRCK_RIGHT_LOW 1 + +#define TEGRA30_I2S_CTRL_LRCK_SHIFT 9 +#define TEGRA30_I2S_CTRL_LRCK_MASK (1 << TEGRA30_I2S_CTRL_LRCK_SHIFT) +#define TEGRA30_I2S_CTRL_LRCK_L_LOW (TEGRA30_I2S_LRCK_LEFT_LOW << TEGRA30_I2S_CTRL_LRCK_SHIFT) +#define TEGRA30_I2S_CTRL_LRCK_R_LOW (TEGRA30_I2S_LRCK_RIGHT_LOW << TEGRA30_I2S_CTRL_LRCK_SHIFT) + +#define TEGRA30_I2S_CTRL_LPBK_ENABLE (1 << 8) + +#define TEGRA30_I2S_BIT_CODE_LINEAR 0 +#define TEGRA30_I2S_BIT_CODE_ULAW 1 +#define TEGRA30_I2S_BIT_CODE_ALAW 2 + +#define TEGRA30_I2S_CTRL_BIT_CODE_SHIFT 4 +#define TEGRA30_I2S_CTRL_BIT_CODE_MASK (3 << TEGRA30_I2S_CTRL_BIT_CODE_SHIFT) +#define TEGRA30_I2S_CTRL_BIT_CODE_LINEAR (TEGRA30_I2S_BIT_CODE_LINEAR << TEGRA30_I2S_CTRL_BIT_CODE_SHIFT) +#define TEGRA30_I2S_CTRL_BIT_CODE_ULAW (TEGRA30_I2S_BIT_CODE_ULAW << TEGRA30_I2S_CTRL_BIT_CODE_SHIFT) +#define TEGRA30_I2S_CTRL_BIT_CODE_ALAW (TEGRA30_I2S_BIT_CODE_ALAW << TEGRA30_I2S_CTRL_BIT_CODE_SHIFT) + +#define TEGRA30_I2S_BITS_8 1 +#define TEGRA30_I2S_BITS_12 2 +#define TEGRA30_I2S_BITS_16 3 +#define TEGRA30_I2S_BITS_20 4 +#define TEGRA30_I2S_BITS_24 5 +#define TEGRA30_I2S_BITS_28 6 +#define TEGRA30_I2S_BITS_32 7 + +/* Sample container size; see {RX,TX}_MASK field in CH_CTRL below */ +#define TEGRA30_I2S_CTRL_BIT_SIZE_SHIFT 0 +#define TEGRA30_I2S_CTRL_BIT_SIZE_MASK (7 << TEGRA30_I2S_CTRL_BIT_SIZE_SHIFT) +#define TEGRA30_I2S_CTRL_BIT_SIZE_8 (TEGRA30_I2S_BITS_8 << TEGRA30_I2S_CTRL_BIT_SIZE_SHIFT) +#define TEGRA30_I2S_CTRL_BIT_SIZE_12 (TEGRA30_I2S_BITS_12 << TEGRA30_I2S_CTRL_BIT_SIZE_SHIFT) +#define TEGRA30_I2S_CTRL_BIT_SIZE_16 (TEGRA30_I2S_BITS_16 << TEGRA30_I2S_CTRL_BIT_SIZE_SHIFT) +#define TEGRA30_I2S_CTRL_BIT_SIZE_20 (TEGRA30_I2S_BITS_20 << TEGRA30_I2S_CTRL_BIT_SIZE_SHIFT) +#define TEGRA30_I2S_CTRL_BIT_SIZE_24 (TEGRA30_I2S_BITS_24 << TEGRA30_I2S_CTRL_BIT_SIZE_SHIFT) +#define TEGRA30_I2S_CTRL_BIT_SIZE_28 (TEGRA30_I2S_BITS_28 << TEGRA30_I2S_CTRL_BIT_SIZE_SHIFT) +#define TEGRA30_I2S_CTRL_BIT_SIZE_32 (TEGRA30_I2S_BITS_32 << TEGRA30_I2S_CTRL_BIT_SIZE_SHIFT) + +/* Fields in TEGRA30_I2S_TIMING */ + +#define TEGRA30_I2S_TIMING_NON_SYM_ENABLE (1 << 12) +#define TEGRA30_I2S_TIMING_CHANNEL_BIT_COUNT_SHIFT 0 +#define TEGRA30_I2S_TIMING_CHANNEL_BIT_COUNT_MASK_US 0x7fff +#define TEGRA30_I2S_TIMING_CHANNEL_BIT_COUNT_MASK (TEGRA30_I2S_TIMING_CHANNEL_BIT_COUNT_MASK_US << TEGRA30_I2S_TIMING_CHANNEL_BIT_COUNT_SHIFT) + +/* Fields in TEGRA30_I2S_OFFSET */ + +#define TEGRA30_I2S_OFFSET_RX_DATA_OFFSET_SHIFT 16 +#define TEGRA30_I2S_OFFSET_RX_DATA_OFFSET_MASK_US 0x7ff +#define TEGRA30_I2S_OFFSET_RX_DATA_OFFSET_MASK (TEGRA30_I2S_OFFSET_RX_DATA_OFFSET_MASK_US << TEGRA30_I2S_OFFSET_RX_DATA_OFFSET_SHIFT) +#define TEGRA30_I2S_OFFSET_TX_DATA_OFFSET_SHIFT 0 +#define TEGRA30_I2S_OFFSET_TX_DATA_OFFSET_MASK_US 0x7ff +#define TEGRA30_I2S_OFFSET_TX_DATA_OFFSET_MASK (TEGRA30_I2S_OFFSET_TX_DATA_OFFSET_MASK_US << TEGRA30_I2S_OFFSET_TX_DATA_OFFSET_SHIFT) + +/* Fields in TEGRA30_I2S_CH_CTRL */ + +/* (FSYNC width - 1) in bit clocks */ +#define TEGRA30_I2S_CH_CTRL_FSYNC_WIDTH_SHIFT 24 +#define TEGRA30_I2S_CH_CTRL_FSYNC_WIDTH_MASK_US 0xff +#define TEGRA30_I2S_CH_CTRL_FSYNC_WIDTH_MASK (TEGRA30_I2S_CH_CTRL_FSYNC_WIDTH_MASK_US << TEGRA30_I2S_CH_CTRL_FSYNC_WIDTH_SHIFT) + +#define TEGRA30_I2S_HIGHZ_NO 0 +#define TEGRA30_I2S_HIGHZ_YES 1 +#define TEGRA30_I2S_HIGHZ_ON_HALF_BIT_CLK 2 + +#define TEGRA30_I2S_CH_CTRL_HIGHZ_CTRL_SHIFT 12 +#define TEGRA30_I2S_CH_CTRL_HIGHZ_CTRL_MASK (3 << TEGRA30_I2S_CH_CTRL_HIGHZ_CTRL_SHIFT) +#define TEGRA30_I2S_CH_CTRL_HIGHZ_CTRL_NO (TEGRA30_I2S_HIGHZ_NO << TEGRA30_I2S_CH_CTRL_HIGHZ_CTRL_SHIFT) +#define TEGRA30_I2S_CH_CTRL_HIGHZ_CTRL_YES (TEGRA30_I2S_HIGHZ_YES << TEGRA30_I2S_CH_CTRL_HIGHZ_CTRL_SHIFT) +#define TEGRA30_I2S_CH_CTRL_HIGHZ_CTRL_ON_HALF_BIT_CLK (TEGRA30_I2S_HIGHZ_ON_HALF_BIT_CLK << TEGRA30_I2S_CH_CTRL_HIGHZ_CTRL_SHIFT) + +#define TEGRA30_I2S_MSB_FIRST 0 +#define TEGRA30_I2S_LSB_FIRST 1 + +#define TEGRA30_I2S_CH_CTRL_RX_BIT_ORDER_SHIFT 10 +#define TEGRA30_I2S_CH_CTRL_RX_BIT_ORDER_MASK (1 << TEGRA30_I2S_CH_CTRL_RX_BIT_ORDER_SHIFT) +#define TEGRA30_I2S_CH_CTRL_RX_BIT_ORDER_MSB_FIRST (TEGRA30_I2S_MSB_FIRST << TEGRA30_I2S_CH_CTRL_RX_BIT_ORDER_SHIFT) +#define TEGRA30_I2S_CH_CTRL_RX_BIT_ORDER_LSB_FIRST (TEGRA30_I2S_LSB_FIRST << TEGRA30_I2S_CH_CTRL_RX_BIT_ORDER_SHIFT) +#define TEGRA30_I2S_CH_CTRL_TX_BIT_ORDER_SHIFT 9 +#define TEGRA30_I2S_CH_CTRL_TX_BIT_ORDER_MASK (1 << TEGRA30_I2S_CH_CTRL_TX_BIT_ORDER_SHIFT) +#define TEGRA30_I2S_CH_CTRL_TX_BIT_ORDER_MSB_FIRST (TEGRA30_I2S_MSB_FIRST << TEGRA30_I2S_CH_CTRL_TX_BIT_ORDER_SHIFT) +#define TEGRA30_I2S_CH_CTRL_TX_BIT_ORDER_LSB_FIRST (TEGRA30_I2S_LSB_FIRST << TEGRA30_I2S_CH_CTRL_TX_BIT_ORDER_SHIFT) + +#define TEGRA30_I2S_POS_EDGE 0 +#define TEGRA30_I2S_NEG_EDGE 1 + +#define TEGRA30_I2S_CH_CTRL_EGDE_CTRL_SHIFT 8 +#define TEGRA30_I2S_CH_CTRL_EGDE_CTRL_MASK (1 << TEGRA30_I2S_CH_CTRL_EGDE_CTRL_SHIFT) +#define TEGRA30_I2S_CH_CTRL_EGDE_CTRL_POS_EDGE (TEGRA30_I2S_POS_EDGE << TEGRA30_I2S_CH_CTRL_EGDE_CTRL_SHIFT) +#define TEGRA30_I2S_CH_CTRL_EGDE_CTRL_NEG_EDGE (TEGRA30_I2S_NEG_EDGE << TEGRA30_I2S_CH_CTRL_EGDE_CTRL_SHIFT) + +/* Sample size is # bits from BIT_SIZE minus this field */ +#define TEGRA30_I2S_CH_CTRL_RX_MASK_BITS_SHIFT 4 +#define TEGRA30_I2S_CH_CTRL_RX_MASK_BITS_MASK_US 7 +#define TEGRA30_I2S_CH_CTRL_RX_MASK_BITS_MASK (TEGRA30_I2S_CH_CTRL_RX_MASK_BITS_MASK_US << TEGRA30_I2S_CH_CTRL_RX_MASK_BITS_SHIFT) + +#define TEGRA30_I2S_CH_CTRL_TX_MASK_BITS_SHIFT 0 +#define TEGRA30_I2S_CH_CTRL_TX_MASK_BITS_MASK_US 7 +#define TEGRA30_I2S_CH_CTRL_TX_MASK_BITS_MASK (TEGRA30_I2S_CH_CTRL_TX_MASK_BITS_MASK_US << TEGRA30_I2S_CH_CTRL_TX_MASK_BITS_SHIFT) + +/* Fields in TEGRA30_I2S_SLOT_CTRL */ + +/* Number of slots in frame, minus 1 */ +#define TEGRA30_I2S_SLOT_CTRL_TOTAL_SLOTS_SHIFT 16 +#define TEGRA30_I2S_SLOT_CTRL_TOTAL_SLOTS_MASK_US 7 +#define TEGRA30_I2S_SLOT_CTRL_TOTAL_SLOTS_MASK (TEGRA30_I2S_SLOT_CTRL_TOTAL_SLOT_MASK_US << TEGRA30_I2S_SLOT_CTRL_TOTAL_SLOT_SHIFT) + +/* TDM mode slot enable bitmask */ +#define TEGRA30_I2S_SLOT_CTRL_RX_SLOT_ENABLES_SHIFT 8 +#define TEGRA30_I2S_SLOT_CTRL_RX_SLOT_ENABLES_MASK (0xff << TEGRA30_I2S_SLOT_CTRL_RX_SLOT_ENABLES_SHIFT) + +#define TEGRA30_I2S_SLOT_CTRL_TX_SLOT_ENABLES_SHIFT 0 +#define TEGRA30_I2S_SLOT_CTRL_TX_SLOT_ENABLES_MASK (0xff << TEGRA30_I2S_SLOT_CTRL_TX_SLOT_ENABLES_SHIFT) + +/* Fields in TEGRA30_I2S_CIF_RX_CTRL */ +/* Uses field from TEGRA30_AUDIOCIF_CTRL_* in tegra30_ahub.h */ + +/* Fields in TEGRA30_I2S_CIF_TX_CTRL */ +/* Uses field from TEGRA30_AUDIOCIF_CTRL_* in tegra30_ahub.h */ + +/* Fields in TEGRA30_I2S_FLOWCTL */ + +#define TEGRA30_I2S_FILTER_LINEAR 0 +#define TEGRA30_I2S_FILTER_QUAD 1 + +#define TEGRA30_I2S_FLOWCTL_FILTER_SHIFT 31 +#define TEGRA30_I2S_FLOWCTL_FILTER_MASK (1 << TEGRA30_I2S_FLOWCTL_FILTER_SHIFT) +#define TEGRA30_I2S_FLOWCTL_FILTER_LINEAR (TEGRA30_I2S_FILTER_LINEAR << TEGRA30_I2S_FLOWCTL_FILTER_SHIFT) +#define TEGRA30_I2S_FLOWCTL_FILTER_QUAD (TEGRA30_I2S_FILTER_QUAD << TEGRA30_I2S_FLOWCTL_FILTER_SHIFT) + +/* Fields in TEGRA30_I2S_TX_STEP */ + +#define TEGRA30_I2S_TX_STEP_SHIFT 0 +#define TEGRA30_I2S_TX_STEP_MASK_US 0xffff +#define TEGRA30_I2S_TX_STEP_MASK (TEGRA30_I2S_TX_STEP_MASK_US << TEGRA30_I2S_TX_STEP_SHIFT) + +/* Fields in TEGRA30_I2S_FLOW_STATUS */ + +#define TEGRA30_I2S_FLOW_STATUS_UNDERFLOW (1 << 31) +#define TEGRA30_I2S_FLOW_STATUS_OVERFLOW (1 << 30) +#define TEGRA30_I2S_FLOW_STATUS_MONITOR_INT_EN (1 << 4) +#define TEGRA30_I2S_FLOW_STATUS_COUNTER_CLR (1 << 3) +#define TEGRA30_I2S_FLOW_STATUS_MONITOR_CLR (1 << 2) +#define TEGRA30_I2S_FLOW_STATUS_COUNTER_EN (1 << 1) +#define TEGRA30_I2S_FLOW_STATUS_MONITOR_EN (1 << 0) + +/* + * There are no fields in TEGRA30_I2S_FLOW_TOTAL, TEGRA30_I2S_FLOW_OVER, + * TEGRA30_I2S_FLOW_UNDER; they are counters taking the whole register. + */ + +/* Fields in TEGRA30_I2S_LCOEF_* */ + +#define TEGRA30_I2S_LCOEF_COEF_SHIFT 0 +#define TEGRA30_I2S_LCOEF_COEF_MASK_US 0xffff +#define TEGRA30_I2S_LCOEF_COEF_MASK (TEGRA30_I2S_LCOEF_COEF_MASK_US << TEGRA30_I2S_LCOEF_COEF_SHIFT) + +struct tegra30_i2s { + struct snd_soc_dai_driver dai; + int cif_id; + struct clk *clk_i2s; + enum tegra30_ahub_txcif capture_i2s_cif; + enum tegra30_ahub_rxcif capture_fifo_cif; + struct tegra_pcm_dma_params capture_dma_data; + enum tegra30_ahub_rxcif playback_i2s_cif; + enum tegra30_ahub_txcif playback_fifo_cif; + struct tegra_pcm_dma_params playback_dma_data; + struct regmap *regmap; + u32 reg_ctrl; +}; + +#endif diff --git a/sound/soc/tegra/tegra_alc5632.c b/sound/soc/tegra/tegra_alc5632.c index e45ccd851f6..32de7006daf 100644 --- a/sound/soc/tegra/tegra_alc5632.c +++ b/sound/soc/tegra/tegra_alc5632.c @@ -1,16 +1,17 @@ /* -* tegra_alc5632.c -- Toshiba AC100(PAZ00) machine ASoC driver -* -* Copyright (C) 2011 The AC100 Kernel Team <ac100@lists.lauchpad.net> -* -* Authors: Leon Romanovsky <leon@leon.nu> -* Andrey Danin <danindrey@mail.ru> -* Marc Dietrich <marvin24@gmx.de> -* -* This program is free software; you can redistribute it and/or modify -* it under the terms of the GNU General Public License version 2 as -* published by the Free Software Foundation. -*/ + * tegra_alc5632.c -- Toshiba AC100(PAZ00) machine ASoC driver + * + * Copyright (C) 2011 The AC100 Kernel Team <ac100@lists.lauchpad.net> + * Copyright (C) 2012 - NVIDIA, Inc. + * + * Authors: Leon Romanovsky <leon@leon.nu> + * Andrey Danin <danindrey@mail.ru> + * Marc Dietrich <marvin24@gmx.de> + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License version 2 as + * published by the Free Software Foundation. + */ #include <asm/mach-types.h> @@ -28,9 +29,6 @@ #include "../codecs/alc5632.h" -#include "tegra_das.h" -#include "tegra_i2s.h" -#include "tegra_pcm.h" #include "tegra_asoc_utils.h" #define DRV_NAME "tegra-alc5632" @@ -39,7 +37,6 @@ struct tegra_alc5632 { struct tegra_asoc_utils_data util_data; - struct platform_device *pcm_dev; int gpio_requested; int gpio_hp_det; }; @@ -140,7 +137,6 @@ static int tegra_alc5632_asoc_init(struct snd_soc_pcm_runtime *rtd) static struct snd_soc_dai_link tegra_alc5632_dai = { .name = "ALC5632", .stream_name = "ALC5632 PCM", - .platform_name = "tegra-pcm-audio", .codec_dai_name = "alc5632-hifi", .init = tegra_alc5632_asoc_init, .ops = &tegra_alc5632_asoc_ops, @@ -179,8 +175,6 @@ static __devinit int tegra_alc5632_probe(struct platform_device *pdev) platform_set_drvdata(pdev, card); snd_soc_card_set_drvdata(card, alc5632); - alc5632->pcm_dev = ERR_PTR(-EINVAL); - if (!(pdev->dev.of_node)) { dev_err(&pdev->dev, "Must be instantiated using device tree\n"); ret = -EINVAL; @@ -214,18 +208,11 @@ static __devinit int tegra_alc5632_probe(struct platform_device *pdev) goto err; } - alc5632->pcm_dev = platform_device_register_simple( - "tegra-pcm-audio", -1, NULL, 0); - if (IS_ERR(alc5632->pcm_dev)) { - dev_err(&pdev->dev, - "Can't instantiate tegra-pcm-audio\n"); - ret = PTR_ERR(alc5632->pcm_dev); - goto err; - } + tegra_alc5632_dai.platform_of_node = tegra_alc5632_dai.cpu_dai_of_node; ret = tegra_asoc_utils_init(&alc5632->util_data, &pdev->dev); if (ret) - goto err_unregister; + goto err; ret = snd_soc_register_card(card); if (ret) { @@ -238,9 +225,6 @@ static __devinit int tegra_alc5632_probe(struct platform_device *pdev) err_fini_utils: tegra_asoc_utils_fini(&alc5632->util_data); -err_unregister: - if (!IS_ERR(alc5632->pcm_dev)) - platform_device_unregister(alc5632->pcm_dev); err: return ret; } @@ -259,8 +243,6 @@ static int __devexit tegra_alc5632_remove(struct platform_device *pdev) snd_soc_unregister_card(card); tegra_asoc_utils_fini(&machine->util_data); - if (!IS_ERR(machine->pcm_dev)) - platform_device_unregister(machine->pcm_dev); return 0; } diff --git a/sound/soc/tegra/tegra_asoc_utils.c b/sound/soc/tegra/tegra_asoc_utils.c index f8428e410e0..9515ce58ea0 100644 --- a/sound/soc/tegra/tegra_asoc_utils.c +++ b/sound/soc/tegra/tegra_asoc_utils.c @@ -2,7 +2,7 @@ * tegra_asoc_utils.c - Harmony machine ASoC driver * * Author: Stephen Warren <swarren@nvidia.com> - * Copyright (C) 2010 - NVIDIA, Inc. + * Copyright (C) 2010,2012 - NVIDIA, Inc. * * This program is free software; you can redistribute it and/or * modify it under the terms of the GNU General Public License @@ -25,6 +25,7 @@ #include <linux/err.h> #include <linux/kernel.h> #include <linux/module.h> +#include <linux/of.h> #include "tegra_asoc_utils.h" @@ -40,7 +41,10 @@ int tegra_asoc_utils_set_rate(struct tegra_asoc_utils_data *data, int srate, case 22050: case 44100: case 88200: - new_baseclock = 56448000; + if (data->soc == TEGRA_ASOC_UTILS_SOC_TEGRA20) + new_baseclock = 56448000; + else + new_baseclock = 564480000; break; case 8000: case 16000: @@ -48,7 +52,10 @@ int tegra_asoc_utils_set_rate(struct tegra_asoc_utils_data *data, int srate, case 48000: case 64000: case 96000: - new_baseclock = 73728000; + if (data->soc == TEGRA_ASOC_UTILS_SOC_TEGRA20) + new_baseclock = 73728000; + else + new_baseclock = 552960000; break; default: return -EINVAL; @@ -78,7 +85,7 @@ int tegra_asoc_utils_set_rate(struct tegra_asoc_utils_data *data, int srate, return err; } - /* Don't set cdev1 rate; its locked to pll_a_out0 */ + /* Don't set cdev1/extern1 rate; it's locked to pll_a_out0 */ err = clk_enable(data->clk_pll_a); if (err) { @@ -112,6 +119,17 @@ int tegra_asoc_utils_init(struct tegra_asoc_utils_data *data, data->dev = dev; + if (of_machine_is_compatible("nvidia,tegra20")) + data->soc = TEGRA_ASOC_UTILS_SOC_TEGRA20; + else if (of_machine_is_compatible("nvidia,tegra30")) + data->soc = TEGRA_ASOC_UTILS_SOC_TEGRA30; + else if (!dev->of_node) + /* non-DT is always Tegra20 */ + data->soc = TEGRA_ASOC_UTILS_SOC_TEGRA20; + else + /* DT boot, but unknown SoC */ + return -EINVAL; + data->clk_pll_a = clk_get_sys(NULL, "pll_a"); if (IS_ERR(data->clk_pll_a)) { dev_err(data->dev, "Can't retrieve clk pll_a\n"); @@ -126,15 +144,24 @@ int tegra_asoc_utils_init(struct tegra_asoc_utils_data *data, goto err_put_pll_a; } - data->clk_cdev1 = clk_get_sys(NULL, "cdev1"); + if (data->soc == TEGRA_ASOC_UTILS_SOC_TEGRA20) + data->clk_cdev1 = clk_get_sys(NULL, "cdev1"); + else + data->clk_cdev1 = clk_get_sys("extern1", NULL); if (IS_ERR(data->clk_cdev1)) { dev_err(data->dev, "Can't retrieve clk cdev1\n"); ret = PTR_ERR(data->clk_cdev1); goto err_put_pll_a_out0; } + ret = tegra_asoc_utils_set_rate(data, 44100, 256 * 44100); + if (ret) + goto err_put_cdev1; + return 0; +err_put_cdev1: + clk_put(data->clk_cdev1); err_put_pll_a_out0: clk_put(data->clk_pll_a_out0); err_put_pll_a: diff --git a/sound/soc/tegra/tegra_asoc_utils.h b/sound/soc/tegra/tegra_asoc_utils.h index 4818195da25..44db1dbb8f2 100644 --- a/sound/soc/tegra/tegra_asoc_utils.h +++ b/sound/soc/tegra/tegra_asoc_utils.h @@ -2,7 +2,7 @@ * tegra_asoc_utils.h - Definitions for Tegra DAS driver * * Author: Stephen Warren <swarren@nvidia.com> - * Copyright (C) 2010 - NVIDIA, Inc. + * Copyright (C) 2010,2012 - NVIDIA, Inc. * * This program is free software; you can redistribute it and/or * modify it under the terms of the GNU General Public License @@ -26,8 +26,14 @@ struct clk; struct device; +enum tegra_asoc_utils_soc { + TEGRA_ASOC_UTILS_SOC_TEGRA20, + TEGRA_ASOC_UTILS_SOC_TEGRA30, +}; + struct tegra_asoc_utils_data { struct device *dev; + enum tegra_asoc_utils_soc soc; struct clk *clk_pll_a; struct clk *clk_pll_a_out0; struct clk *clk_cdev1; @@ -42,4 +48,3 @@ int tegra_asoc_utils_init(struct tegra_asoc_utils_data *data, void tegra_asoc_utils_fini(struct tegra_asoc_utils_data *data); #endif - diff --git a/sound/soc/tegra/tegra_das.c b/sound/soc/tegra/tegra_das.c deleted file mode 100644 index 3b3c1ba4d23..00000000000 --- a/sound/soc/tegra/tegra_das.c +++ /dev/null @@ -1,261 +0,0 @@ -/* - * tegra_das.c - Tegra DAS driver - * - * Author: Stephen Warren <swarren@nvidia.com> - * Copyright (C) 2010 - NVIDIA, Inc. - * - * This program is free software; you can redistribute it and/or - * modify it under the terms of the GNU General Public License - * version 2 as published by the Free Software Foundation. - * - * This program is distributed in the hope that it will be useful, but - * WITHOUT ANY WARRANTY; without even the implied warranty of - * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU - * General Public License for more details. - * - * You should have received a copy of the GNU General Public License - * along with this program; if not, write to the Free Software - * Foundation, Inc., 51 Franklin St, Fifth Floor, Boston, MA - * 02110-1301 USA - * - */ - -#include <linux/module.h> -#include <linux/debugfs.h> -#include <linux/device.h> -#include <linux/platform_device.h> -#include <linux/seq_file.h> -#include <linux/slab.h> -#include <linux/io.h> -#include <mach/iomap.h> -#include <sound/soc.h> -#include "tegra_das.h" - -#define DRV_NAME "tegra-das" - -static struct tegra_das *das; - -static inline void tegra_das_write(u32 reg, u32 val) -{ - __raw_writel(val, das->regs + reg); -} - -static inline u32 tegra_das_read(u32 reg) -{ - return __raw_readl(das->regs + reg); -} - -int tegra_das_connect_dap_to_dac(int dap, int dac) -{ - u32 addr; - u32 reg; - - if (!das) - return -ENODEV; - - addr = TEGRA_DAS_DAP_CTRL_SEL + - (dap * TEGRA_DAS_DAP_CTRL_SEL_STRIDE); - reg = dac << TEGRA_DAS_DAP_CTRL_SEL_DAP_CTRL_SEL_P; - - tegra_das_write(addr, reg); - - return 0; -} -EXPORT_SYMBOL_GPL(tegra_das_connect_dap_to_dac); - -int tegra_das_connect_dap_to_dap(int dap, int otherdap, int master, - int sdata1rx, int sdata2rx) -{ - u32 addr; - u32 reg; - - if (!das) - return -ENODEV; - - addr = TEGRA_DAS_DAP_CTRL_SEL + - (dap * TEGRA_DAS_DAP_CTRL_SEL_STRIDE); - reg = otherdap << TEGRA_DAS_DAP_CTRL_SEL_DAP_CTRL_SEL_P | - !!sdata2rx << TEGRA_DAS_DAP_CTRL_SEL_DAP_SDATA2_TX_RX_P | - !!sdata1rx << TEGRA_DAS_DAP_CTRL_SEL_DAP_SDATA1_TX_RX_P | - !!master << TEGRA_DAS_DAP_CTRL_SEL_DAP_MS_SEL_P; - - tegra_das_write(addr, reg); - - return 0; -} -EXPORT_SYMBOL_GPL(tegra_das_connect_dap_to_dap); - -int tegra_das_connect_dac_to_dap(int dac, int dap) -{ - u32 addr; - u32 reg; - - if (!das) - return -ENODEV; - - addr = TEGRA_DAS_DAC_INPUT_DATA_CLK_SEL + - (dac * TEGRA_DAS_DAC_INPUT_DATA_CLK_SEL_STRIDE); - reg = dap << TEGRA_DAS_DAC_INPUT_DATA_CLK_SEL_DAC_CLK_SEL_P | - dap << TEGRA_DAS_DAC_INPUT_DATA_CLK_SEL_DAC_SDATA1_SEL_P | - dap << TEGRA_DAS_DAC_INPUT_DATA_CLK_SEL_DAC_SDATA2_SEL_P; - - tegra_das_write(addr, reg); - - return 0; -} -EXPORT_SYMBOL_GPL(tegra_das_connect_dac_to_dap); - -#ifdef CONFIG_DEBUG_FS -static int tegra_das_show(struct seq_file *s, void *unused) -{ - int i; - u32 addr; - u32 reg; - - for (i = 0; i < TEGRA_DAS_DAP_CTRL_SEL_COUNT; i++) { - addr = TEGRA_DAS_DAP_CTRL_SEL + - (i * TEGRA_DAS_DAP_CTRL_SEL_STRIDE); - reg = tegra_das_read(addr); - seq_printf(s, "TEGRA_DAS_DAP_CTRL_SEL[%d] = %08x\n", i, reg); - } - - for (i = 0; i < TEGRA_DAS_DAC_INPUT_DATA_CLK_SEL_COUNT; i++) { - addr = TEGRA_DAS_DAC_INPUT_DATA_CLK_SEL + - (i * TEGRA_DAS_DAC_INPUT_DATA_CLK_SEL_STRIDE); - reg = tegra_das_read(addr); - seq_printf(s, "TEGRA_DAS_DAC_INPUT_DATA_CLK_SEL[%d] = %08x\n", - i, reg); - } - - return 0; -} - -static int tegra_das_debug_open(struct inode *inode, struct file *file) -{ - return single_open(file, tegra_das_show, inode->i_private); -} - -static const struct file_operations tegra_das_debug_fops = { - .open = tegra_das_debug_open, - .read = seq_read, - .llseek = seq_lseek, - .release = single_release, -}; - -static void tegra_das_debug_add(struct tegra_das *das) -{ - das->debug = debugfs_create_file(DRV_NAME, S_IRUGO, - snd_soc_debugfs_root, das, - &tegra_das_debug_fops); -} - -static void tegra_das_debug_remove(struct tegra_das *das) -{ - if (das->debug) - debugfs_remove(das->debug); -} -#else -static inline void tegra_das_debug_add(struct tegra_das *das) -{ -} - -static inline void tegra_das_debug_remove(struct tegra_das *das) -{ -} -#endif - -static int __devinit tegra_das_probe(struct platform_device *pdev) -{ - struct resource *res, *region; - int ret = 0; - - if (das) - return -ENODEV; - - das = devm_kzalloc(&pdev->dev, sizeof(struct tegra_das), GFP_KERNEL); - if (!das) { - dev_err(&pdev->dev, "Can't allocate tegra_das\n"); - ret = -ENOMEM; - goto err; - } - das->dev = &pdev->dev; - - res = platform_get_resource(pdev, IORESOURCE_MEM, 0); - if (!res) { - dev_err(&pdev->dev, "No memory resource\n"); - ret = -ENODEV; - goto err; - } - - region = devm_request_mem_region(&pdev->dev, res->start, - resource_size(res), pdev->name); - if (!region) { - dev_err(&pdev->dev, "Memory region already claimed\n"); - ret = -EBUSY; - goto err; - } - - das->regs = devm_ioremap(&pdev->dev, res->start, resource_size(res)); - if (!das->regs) { - dev_err(&pdev->dev, "ioremap failed\n"); - ret = -ENOMEM; - goto err; - } - - ret = tegra_das_connect_dap_to_dac(TEGRA_DAS_DAP_ID_1, - TEGRA_DAS_DAP_SEL_DAC1); - if (ret) { - dev_err(&pdev->dev, "Can't set up DAS DAP connection\n"); - goto err; - } - ret = tegra_das_connect_dac_to_dap(TEGRA_DAS_DAC_ID_1, - TEGRA_DAS_DAC_SEL_DAP1); - if (ret) { - dev_err(&pdev->dev, "Can't set up DAS DAC connection\n"); - goto err; - } - - tegra_das_debug_add(das); - - platform_set_drvdata(pdev, das); - - return 0; - -err: - das = NULL; - return ret; -} - -static int __devexit tegra_das_remove(struct platform_device *pdev) -{ - if (!das) - return -ENODEV; - - tegra_das_debug_remove(das); - - das = NULL; - - return 0; -} - -static const struct of_device_id tegra_das_of_match[] __devinitconst = { - { .compatible = "nvidia,tegra20-das", }, - {}, -}; - -static struct platform_driver tegra_das_driver = { - .probe = tegra_das_probe, - .remove = __devexit_p(tegra_das_remove), - .driver = { - .name = DRV_NAME, - .owner = THIS_MODULE, - .of_match_table = tegra_das_of_match, - }, -}; -module_platform_driver(tegra_das_driver); - -MODULE_AUTHOR("Stephen Warren <swarren@nvidia.com>"); -MODULE_DESCRIPTION("Tegra DAS driver"); -MODULE_LICENSE("GPL"); -MODULE_ALIAS("platform:" DRV_NAME); -MODULE_DEVICE_TABLE(of, tegra_das_of_match); diff --git a/sound/soc/tegra/tegra_das.h b/sound/soc/tegra/tegra_das.h deleted file mode 100644 index 2c96c7b3c45..00000000000 --- a/sound/soc/tegra/tegra_das.h +++ /dev/null @@ -1,135 +0,0 @@ -/* - * tegra_das.h - Definitions for Tegra DAS driver - * - * Author: Stephen Warren <swarren@nvidia.com> - * Copyright (C) 2010 - NVIDIA, Inc. - * - * This program is free software; you can redistribute it and/or - * modify it under the terms of the GNU General Public License - * version 2 as published by the Free Software Foundation. - * - * This program is distributed in the hope that it will be useful, but - * WITHOUT ANY WARRANTY; without even the implied warranty of - * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU - * General Public License for more details. - * - * You should have received a copy of the GNU General Public License - * along with this program; if not, write to the Free Software - * Foundation, Inc., 51 Franklin St, Fifth Floor, Boston, MA - * 02110-1301 USA - * - */ - -#ifndef __TEGRA_DAS_H__ -#define __TEGRA_DAS_H__ - -/* Register TEGRA_DAS_DAP_CTRL_SEL */ -#define TEGRA_DAS_DAP_CTRL_SEL 0x00 -#define TEGRA_DAS_DAP_CTRL_SEL_COUNT 5 -#define TEGRA_DAS_DAP_CTRL_SEL_STRIDE 4 -#define TEGRA_DAS_DAP_CTRL_SEL_DAP_MS_SEL_P 31 -#define TEGRA_DAS_DAP_CTRL_SEL_DAP_MS_SEL_S 1 -#define TEGRA_DAS_DAP_CTRL_SEL_DAP_SDATA1_TX_RX_P 30 -#define TEGRA_DAS_DAP_CTRL_SEL_DAP_SDATA1_TX_RX_S 1 -#define TEGRA_DAS_DAP_CTRL_SEL_DAP_SDATA2_TX_RX_P 29 -#define TEGRA_DAS_DAP_CTRL_SEL_DAP_SDATA2_TX_RX_S 1 -#define TEGRA_DAS_DAP_CTRL_SEL_DAP_CTRL_SEL_P 0 -#define TEGRA_DAS_DAP_CTRL_SEL_DAP_CTRL_SEL_S 5 - -/* Values for field TEGRA_DAS_DAP_CTRL_SEL_DAP_CTRL_SEL */ -#define TEGRA_DAS_DAP_SEL_DAC1 0 -#define TEGRA_DAS_DAP_SEL_DAC2 1 -#define TEGRA_DAS_DAP_SEL_DAC3 2 -#define TEGRA_DAS_DAP_SEL_DAP1 16 -#define TEGRA_DAS_DAP_SEL_DAP2 17 -#define TEGRA_DAS_DAP_SEL_DAP3 18 -#define TEGRA_DAS_DAP_SEL_DAP4 19 -#define TEGRA_DAS_DAP_SEL_DAP5 20 - -/* Register TEGRA_DAS_DAC_INPUT_DATA_CLK_SEL */ -#define TEGRA_DAS_DAC_INPUT_DATA_CLK_SEL 0x40 -#define TEGRA_DAS_DAC_INPUT_DATA_CLK_SEL_COUNT 3 -#define TEGRA_DAS_DAC_INPUT_DATA_CLK_SEL_STRIDE 4 -#define TEGRA_DAS_DAC_INPUT_DATA_CLK_SEL_DAC_SDATA2_SEL_P 28 -#define TEGRA_DAS_DAC_INPUT_DATA_CLK_SEL_DAC_SDATA2_SEL_S 4 -#define TEGRA_DAS_DAC_INPUT_DATA_CLK_SEL_DAC_SDATA1_SEL_P 24 -#define TEGRA_DAS_DAC_INPUT_DATA_CLK_SEL_DAC_SDATA1_SEL_S 4 -#define TEGRA_DAS_DAC_INPUT_DATA_CLK_SEL_DAC_CLK_SEL_P 0 -#define TEGRA_DAS_DAC_INPUT_DATA_CLK_SEL_DAC_CLK_SEL_S 4 - -/* - * Values for: - * TEGRA_DAS_DAC_INPUT_DATA_CLK_SEL_DAC_SDATA2_SEL - * TEGRA_DAS_DAC_INPUT_DATA_CLK_SEL_DAC_SDATA1_SEL - * TEGRA_DAS_DAC_INPUT_DATA_CLK_SEL_DAC_CLK_SEL - */ -#define TEGRA_DAS_DAC_SEL_DAP1 0 -#define TEGRA_DAS_DAC_SEL_DAP2 1 -#define TEGRA_DAS_DAC_SEL_DAP3 2 -#define TEGRA_DAS_DAC_SEL_DAP4 3 -#define TEGRA_DAS_DAC_SEL_DAP5 4 - -/* - * Names/IDs of the DACs/DAPs. - */ - -#define TEGRA_DAS_DAP_ID_1 0 -#define TEGRA_DAS_DAP_ID_2 1 -#define TEGRA_DAS_DAP_ID_3 2 -#define TEGRA_DAS_DAP_ID_4 3 -#define TEGRA_DAS_DAP_ID_5 4 - -#define TEGRA_DAS_DAC_ID_1 0 -#define TEGRA_DAS_DAC_ID_2 1 -#define TEGRA_DAS_DAC_ID_3 2 - -struct tegra_das { - struct device *dev; - void __iomem *regs; - struct dentry *debug; -}; - -/* - * Terminology: - * DAS: Digital audio switch (HW module controlled by this driver) - * DAP: Digital audio port (port/pins on Tegra device) - * DAC: Digital audio controller (e.g. I2S or AC97 controller elsewhere) - * - * The Tegra DAS is a mux/cross-bar which can connect each DAP to a specific - * DAC, or another DAP. When DAPs are connected, one must be the master and - * one the slave. Each DAC allows selection of a specific DAP for input, to - * cater for the case where N DAPs are connected to 1 DAC for broadcast - * output. - * - * This driver is dumb; no attempt is made to ensure that a valid routing - * configuration is programmed. - */ - -/* - * Connect a DAP to to a DAC - * dap_id: DAP to connect: TEGRA_DAS_DAP_ID_* - * dac_sel: DAC to connect to: TEGRA_DAS_DAP_SEL_DAC* - */ -extern int tegra_das_connect_dap_to_dac(int dap_id, int dac_sel); - -/* - * Connect a DAP to to another DAP - * dap_id: DAP to connect: TEGRA_DAS_DAP_ID_* - * other_dap_sel: DAP to connect to: TEGRA_DAS_DAP_SEL_DAP* - * master: Is this DAP the master (1) or slave (0) - * sdata1rx: Is this DAP's SDATA1 pin RX (1) or TX (0) - * sdata2rx: Is this DAP's SDATA2 pin RX (1) or TX (0) - */ -extern int tegra_das_connect_dap_to_dap(int dap_id, int other_dap_sel, - int master, int sdata1rx, - int sdata2rx); - -/* - * Connect a DAC's input to a DAP - * (DAC outputs are selected by the DAP) - * dac_id: DAC ID to connect: TEGRA_DAS_DAC_ID_* - * dap_sel: DAP to receive input from: TEGRA_DAS_DAC_SEL_DAP* - */ -extern int tegra_das_connect_dac_to_dap(int dac_id, int dap_sel); - -#endif diff --git a/sound/soc/tegra/tegra_i2s.c b/sound/soc/tegra/tegra_i2s.c deleted file mode 100644 index e53349912b2..00000000000 --- a/sound/soc/tegra/tegra_i2s.c +++ /dev/null @@ -1,459 +0,0 @@ -/* - * tegra_i2s.c - Tegra I2S driver - * - * Author: Stephen Warren <swarren@nvidia.com> - * Copyright (C) 2010 - NVIDIA, Inc. - * - * Based on code copyright/by: - * - * Copyright (c) 2009-2010, NVIDIA Corporation. - * Scott Peterson <speterson@nvidia.com> - * - * Copyright (C) 2010 Google, Inc. - * Iliyan Malchev <malchev@google.com> - * - * This program is free software; you can redistribute it and/or - * modify it under the terms of the GNU General Public License - * version 2 as published by the Free Software Foundation. - * - * This program is distributed in the hope that it will be useful, but - * WITHOUT ANY WARRANTY; without even the implied warranty of - * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU - * General Public License for more details. - * - * You should have received a copy of the GNU General Public License - * along with this program; if not, write to the Free Software - * Foundation, Inc., 51 Franklin St, Fifth Floor, Boston, MA - * 02110-1301 USA - * - */ - -#include <linux/clk.h> -#include <linux/module.h> -#include <linux/debugfs.h> -#include <linux/device.h> -#include <linux/platform_device.h> -#include <linux/seq_file.h> -#include <linux/slab.h> -#include <linux/io.h> -#include <linux/of.h> -#include <mach/iomap.h> -#include <sound/core.h> -#include <sound/pcm.h> -#include <sound/pcm_params.h> -#include <sound/soc.h> - -#include "tegra_i2s.h" - -#define DRV_NAME "tegra-i2s" - -static inline void tegra_i2s_write(struct tegra_i2s *i2s, u32 reg, u32 val) -{ - __raw_writel(val, i2s->regs + reg); -} - -static inline u32 tegra_i2s_read(struct tegra_i2s *i2s, u32 reg) -{ - return __raw_readl(i2s->regs + reg); -} - -#ifdef CONFIG_DEBUG_FS -static int tegra_i2s_show(struct seq_file *s, void *unused) -{ -#define REG(r) { r, #r } - static const struct { - int offset; - const char *name; - } regs[] = { - REG(TEGRA_I2S_CTRL), - REG(TEGRA_I2S_STATUS), - REG(TEGRA_I2S_TIMING), - REG(TEGRA_I2S_FIFO_SCR), - REG(TEGRA_I2S_PCM_CTRL), - REG(TEGRA_I2S_NW_CTRL), - REG(TEGRA_I2S_TDM_CTRL), - REG(TEGRA_I2S_TDM_TX_RX_CTRL), - }; -#undef REG - - struct tegra_i2s *i2s = s->private; - int i; - - clk_enable(i2s->clk_i2s); - - for (i = 0; i < ARRAY_SIZE(regs); i++) { - u32 val = tegra_i2s_read(i2s, regs[i].offset); - seq_printf(s, "%s = %08x\n", regs[i].name, val); - } - - clk_disable(i2s->clk_i2s); - - return 0; -} - -static int tegra_i2s_debug_open(struct inode *inode, struct file *file) -{ - return single_open(file, tegra_i2s_show, inode->i_private); -} - -static const struct file_operations tegra_i2s_debug_fops = { - .open = tegra_i2s_debug_open, - .read = seq_read, - .llseek = seq_lseek, - .release = single_release, -}; - -static void tegra_i2s_debug_add(struct tegra_i2s *i2s) -{ - i2s->debug = debugfs_create_file(i2s->dai.name, S_IRUGO, - snd_soc_debugfs_root, i2s, - &tegra_i2s_debug_fops); -} - -static void tegra_i2s_debug_remove(struct tegra_i2s *i2s) -{ - if (i2s->debug) - debugfs_remove(i2s->debug); -} -#else -static inline void tegra_i2s_debug_add(struct tegra_i2s *i2s) -{ -} - -static inline void tegra_i2s_debug_remove(struct tegra_i2s *i2s) -{ -} -#endif - -static int tegra_i2s_set_fmt(struct snd_soc_dai *dai, - unsigned int fmt) -{ - struct tegra_i2s *i2s = snd_soc_dai_get_drvdata(dai); - - switch (fmt & SND_SOC_DAIFMT_INV_MASK) { - case SND_SOC_DAIFMT_NB_NF: - break; - default: - return -EINVAL; - } - - i2s->reg_ctrl &= ~TEGRA_I2S_CTRL_MASTER_ENABLE; - switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) { - case SND_SOC_DAIFMT_CBS_CFS: - i2s->reg_ctrl |= TEGRA_I2S_CTRL_MASTER_ENABLE; - break; - case SND_SOC_DAIFMT_CBM_CFM: - break; - default: - return -EINVAL; - } - - i2s->reg_ctrl &= ~(TEGRA_I2S_CTRL_BIT_FORMAT_MASK | - TEGRA_I2S_CTRL_LRCK_MASK); - switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) { - case SND_SOC_DAIFMT_DSP_A: - i2s->reg_ctrl |= TEGRA_I2S_CTRL_BIT_FORMAT_DSP; - i2s->reg_ctrl |= TEGRA_I2S_CTRL_LRCK_L_LOW; - break; - case SND_SOC_DAIFMT_DSP_B: - i2s->reg_ctrl |= TEGRA_I2S_CTRL_BIT_FORMAT_DSP; - i2s->reg_ctrl |= TEGRA_I2S_CTRL_LRCK_R_LOW; - break; - case SND_SOC_DAIFMT_I2S: - i2s->reg_ctrl |= TEGRA_I2S_CTRL_BIT_FORMAT_I2S; - i2s->reg_ctrl |= TEGRA_I2S_CTRL_LRCK_L_LOW; - break; - case SND_SOC_DAIFMT_RIGHT_J: - i2s->reg_ctrl |= TEGRA_I2S_CTRL_BIT_FORMAT_RJM; - i2s->reg_ctrl |= TEGRA_I2S_CTRL_LRCK_L_LOW; - break; - case SND_SOC_DAIFMT_LEFT_J: - i2s->reg_ctrl |= TEGRA_I2S_CTRL_BIT_FORMAT_LJM; - i2s->reg_ctrl |= TEGRA_I2S_CTRL_LRCK_L_LOW; - break; - default: - return -EINVAL; - } - - return 0; -} - -static int tegra_i2s_hw_params(struct snd_pcm_substream *substream, - struct snd_pcm_hw_params *params, - struct snd_soc_dai *dai) -{ - struct device *dev = substream->pcm->card->dev; - struct tegra_i2s *i2s = snd_soc_dai_get_drvdata(dai); - u32 reg; - int ret, sample_size, srate, i2sclock, bitcnt; - - i2s->reg_ctrl &= ~TEGRA_I2S_CTRL_BIT_SIZE_MASK; - switch (params_format(params)) { - case SNDRV_PCM_FORMAT_S16_LE: - i2s->reg_ctrl |= TEGRA_I2S_CTRL_BIT_SIZE_16; - sample_size = 16; - break; - case SNDRV_PCM_FORMAT_S24_LE: - i2s->reg_ctrl |= TEGRA_I2S_CTRL_BIT_SIZE_24; - sample_size = 24; - break; - case SNDRV_PCM_FORMAT_S32_LE: - i2s->reg_ctrl |= TEGRA_I2S_CTRL_BIT_SIZE_32; - sample_size = 32; - break; - default: - return -EINVAL; - } - - srate = params_rate(params); - - /* Final "* 2" required by Tegra hardware */ - i2sclock = srate * params_channels(params) * sample_size * 2; - - ret = clk_set_rate(i2s->clk_i2s, i2sclock); - if (ret) { - dev_err(dev, "Can't set I2S clock rate: %d\n", ret); - return ret; - } - - bitcnt = (i2sclock / (2 * srate)) - 1; - if (bitcnt < 0 || bitcnt > TEGRA_I2S_TIMING_CHANNEL_BIT_COUNT_MASK_US) - return -EINVAL; - reg = bitcnt << TEGRA_I2S_TIMING_CHANNEL_BIT_COUNT_SHIFT; - - if (i2sclock % (2 * srate)) - reg |= TEGRA_I2S_TIMING_NON_SYM_ENABLE; - - if (!i2s->clk_refs) - clk_enable(i2s->clk_i2s); - - tegra_i2s_write(i2s, TEGRA_I2S_TIMING, reg); - - tegra_i2s_write(i2s, TEGRA_I2S_FIFO_SCR, - TEGRA_I2S_FIFO_SCR_FIFO2_ATN_LVL_FOUR_SLOTS | - TEGRA_I2S_FIFO_SCR_FIFO1_ATN_LVL_FOUR_SLOTS); - - if (!i2s->clk_refs) - clk_disable(i2s->clk_i2s); - - return 0; -} - -static void tegra_i2s_start_playback(struct tegra_i2s *i2s) -{ - i2s->reg_ctrl |= TEGRA_I2S_CTRL_FIFO1_ENABLE; - tegra_i2s_write(i2s, TEGRA_I2S_CTRL, i2s->reg_ctrl); -} - -static void tegra_i2s_stop_playback(struct tegra_i2s *i2s) -{ - i2s->reg_ctrl &= ~TEGRA_I2S_CTRL_FIFO1_ENABLE; - tegra_i2s_write(i2s, TEGRA_I2S_CTRL, i2s->reg_ctrl); -} - -static void tegra_i2s_start_capture(struct tegra_i2s *i2s) -{ - i2s->reg_ctrl |= TEGRA_I2S_CTRL_FIFO2_ENABLE; - tegra_i2s_write(i2s, TEGRA_I2S_CTRL, i2s->reg_ctrl); -} - -static void tegra_i2s_stop_capture(struct tegra_i2s *i2s) -{ - i2s->reg_ctrl &= ~TEGRA_I2S_CTRL_FIFO2_ENABLE; - tegra_i2s_write(i2s, TEGRA_I2S_CTRL, i2s->reg_ctrl); -} - -static int tegra_i2s_trigger(struct snd_pcm_substream *substream, int cmd, - struct snd_soc_dai *dai) -{ - struct tegra_i2s *i2s = snd_soc_dai_get_drvdata(dai); - - switch (cmd) { - case SNDRV_PCM_TRIGGER_START: - case SNDRV_PCM_TRIGGER_PAUSE_RELEASE: - case SNDRV_PCM_TRIGGER_RESUME: - if (!i2s->clk_refs) - clk_enable(i2s->clk_i2s); - i2s->clk_refs++; - if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) - tegra_i2s_start_playback(i2s); - else - tegra_i2s_start_capture(i2s); - break; - case SNDRV_PCM_TRIGGER_STOP: - case SNDRV_PCM_TRIGGER_PAUSE_PUSH: - case SNDRV_PCM_TRIGGER_SUSPEND: - if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) - tegra_i2s_stop_playback(i2s); - else - tegra_i2s_stop_capture(i2s); - i2s->clk_refs--; - if (!i2s->clk_refs) - clk_disable(i2s->clk_i2s); - break; - default: - return -EINVAL; - } - - return 0; -} - -static int tegra_i2s_probe(struct snd_soc_dai *dai) -{ - struct tegra_i2s * i2s = snd_soc_dai_get_drvdata(dai); - - dai->capture_dma_data = &i2s->capture_dma_data; - dai->playback_dma_data = &i2s->playback_dma_data; - - return 0; -} - -static const struct snd_soc_dai_ops tegra_i2s_dai_ops = { - .set_fmt = tegra_i2s_set_fmt, - .hw_params = tegra_i2s_hw_params, - .trigger = tegra_i2s_trigger, -}; - -static const struct snd_soc_dai_driver tegra_i2s_dai_template = { - .probe = tegra_i2s_probe, - .playback = { - .channels_min = 2, - .channels_max = 2, - .rates = SNDRV_PCM_RATE_8000_96000, - .formats = SNDRV_PCM_FMTBIT_S16_LE, - }, - .capture = { - .channels_min = 2, - .channels_max = 2, - .rates = SNDRV_PCM_RATE_8000_96000, - .formats = SNDRV_PCM_FMTBIT_S16_LE, - }, - .ops = &tegra_i2s_dai_ops, - .symmetric_rates = 1, -}; - -static __devinit int tegra_i2s_platform_probe(struct platform_device *pdev) -{ - struct tegra_i2s * i2s; - struct resource *mem, *memregion, *dmareq; - u32 of_dma[2]; - u32 dma_ch; - int ret; - - i2s = devm_kzalloc(&pdev->dev, sizeof(struct tegra_i2s), GFP_KERNEL); - if (!i2s) { - dev_err(&pdev->dev, "Can't allocate tegra_i2s\n"); - ret = -ENOMEM; - goto err; - } - dev_set_drvdata(&pdev->dev, i2s); - - i2s->dai = tegra_i2s_dai_template; - i2s->dai.name = dev_name(&pdev->dev); - - i2s->clk_i2s = clk_get(&pdev->dev, NULL); - if (IS_ERR(i2s->clk_i2s)) { - dev_err(&pdev->dev, "Can't retrieve i2s clock\n"); - ret = PTR_ERR(i2s->clk_i2s); - goto err; - } - - mem = platform_get_resource(pdev, IORESOURCE_MEM, 0); - if (!mem) { - dev_err(&pdev->dev, "No memory resource\n"); - ret = -ENODEV; - goto err_clk_put; - } - - dmareq = platform_get_resource(pdev, IORESOURCE_DMA, 0); - if (!dmareq) { - if (of_property_read_u32_array(pdev->dev.of_node, - "nvidia,dma-request-selector", - of_dma, 2) < 0) { - dev_err(&pdev->dev, "No DMA resource\n"); - ret = -ENODEV; - goto err_clk_put; - } - dma_ch = of_dma[1]; - } else { - dma_ch = dmareq->start; - } - - memregion = devm_request_mem_region(&pdev->dev, mem->start, - resource_size(mem), DRV_NAME); - if (!memregion) { - dev_err(&pdev->dev, "Memory region already claimed\n"); - ret = -EBUSY; - goto err_clk_put; - } - - i2s->regs = devm_ioremap(&pdev->dev, mem->start, resource_size(mem)); - if (!i2s->regs) { - dev_err(&pdev->dev, "ioremap failed\n"); - ret = -ENOMEM; - goto err_clk_put; - } - - i2s->capture_dma_data.addr = mem->start + TEGRA_I2S_FIFO2; - i2s->capture_dma_data.wrap = 4; - i2s->capture_dma_data.width = 32; - i2s->capture_dma_data.req_sel = dma_ch; - - i2s->playback_dma_data.addr = mem->start + TEGRA_I2S_FIFO1; - i2s->playback_dma_data.wrap = 4; - i2s->playback_dma_data.width = 32; - i2s->playback_dma_data.req_sel = dma_ch; - - i2s->reg_ctrl = TEGRA_I2S_CTRL_FIFO_FORMAT_PACKED; - - ret = snd_soc_register_dai(&pdev->dev, &i2s->dai); - if (ret) { - dev_err(&pdev->dev, "Could not register DAI: %d\n", ret); - ret = -ENOMEM; - goto err_clk_put; - } - - tegra_i2s_debug_add(i2s); - - return 0; - -err_clk_put: - clk_put(i2s->clk_i2s); -err: - return ret; -} - -static int __devexit tegra_i2s_platform_remove(struct platform_device *pdev) -{ - struct tegra_i2s *i2s = dev_get_drvdata(&pdev->dev); - - snd_soc_unregister_dai(&pdev->dev); - - tegra_i2s_debug_remove(i2s); - - clk_put(i2s->clk_i2s); - - return 0; -} - -static const struct of_device_id tegra_i2s_of_match[] __devinitconst = { - { .compatible = "nvidia,tegra20-i2s", }, - {}, -}; - -static struct platform_driver tegra_i2s_driver = { - .driver = { - .name = DRV_NAME, - .owner = THIS_MODULE, - .of_match_table = tegra_i2s_of_match, - }, - .probe = tegra_i2s_platform_probe, - .remove = __devexit_p(tegra_i2s_platform_remove), -}; -module_platform_driver(tegra_i2s_driver); - -MODULE_AUTHOR("Stephen Warren <swarren@nvidia.com>"); -MODULE_DESCRIPTION("Tegra I2S ASoC driver"); -MODULE_LICENSE("GPL"); -MODULE_ALIAS("platform:" DRV_NAME); -MODULE_DEVICE_TABLE(of, tegra_i2s_of_match); diff --git a/sound/soc/tegra/tegra_i2s.h b/sound/soc/tegra/tegra_i2s.h deleted file mode 100644 index 15ce1e2e8bd..00000000000 --- a/sound/soc/tegra/tegra_i2s.h +++ /dev/null @@ -1,166 +0,0 @@ -/* - * tegra_i2s.h - Definitions for Tegra I2S driver - * - * Author: Stephen Warren <swarren@nvidia.com> - * Copyright (C) 2010 - NVIDIA, Inc. - * - * Based on code copyright/by: - * - * Copyright (c) 2009-2010, NVIDIA Corporation. - * Scott Peterson <speterson@nvidia.com> - * - * Copyright (C) 2010 Google, Inc. - * Iliyan Malchev <malchev@google.com> - * - * This program is free software; you can redistribute it and/or - * modify it under the terms of the GNU General Public License - * version 2 as published by the Free Software Foundation. - * - * This program is distributed in the hope that it will be useful, but - * WITHOUT ANY WARRANTY; without even the implied warranty of - * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU - * General Public License for more details. - * - * You should have received a copy of the GNU General Public License - * along with this program; if not, write to the Free Software - * Foundation, Inc., 51 Franklin St, Fifth Floor, Boston, MA - * 02110-1301 USA - * - */ - -#ifndef __TEGRA_I2S_H__ -#define __TEGRA_I2S_H__ - -#include "tegra_pcm.h" - -/* Register offsets from TEGRA_I2S1_BASE and TEGRA_I2S2_BASE */ - -#define TEGRA_I2S_CTRL 0x00 -#define TEGRA_I2S_STATUS 0x04 -#define TEGRA_I2S_TIMING 0x08 -#define TEGRA_I2S_FIFO_SCR 0x0c -#define TEGRA_I2S_PCM_CTRL 0x10 -#define TEGRA_I2S_NW_CTRL 0x14 -#define TEGRA_I2S_TDM_CTRL 0x20 -#define TEGRA_I2S_TDM_TX_RX_CTRL 0x24 -#define TEGRA_I2S_FIFO1 0x40 -#define TEGRA_I2S_FIFO2 0x80 - -/* Fields in TEGRA_I2S_CTRL */ - -#define TEGRA_I2S_CTRL_FIFO2_TX_ENABLE (1 << 30) -#define TEGRA_I2S_CTRL_FIFO1_ENABLE (1 << 29) -#define TEGRA_I2S_CTRL_FIFO2_ENABLE (1 << 28) -#define TEGRA_I2S_CTRL_FIFO1_RX_ENABLE (1 << 27) -#define TEGRA_I2S_CTRL_FIFO_LPBK_ENABLE (1 << 26) -#define TEGRA_I2S_CTRL_MASTER_ENABLE (1 << 25) - -#define TEGRA_I2S_LRCK_LEFT_LOW 0 -#define TEGRA_I2S_LRCK_RIGHT_LOW 1 - -#define TEGRA_I2S_CTRL_LRCK_SHIFT 24 -#define TEGRA_I2S_CTRL_LRCK_MASK (1 << TEGRA_I2S_CTRL_LRCK_SHIFT) -#define TEGRA_I2S_CTRL_LRCK_L_LOW (TEGRA_I2S_LRCK_LEFT_LOW << TEGRA_I2S_CTRL_LRCK_SHIFT) -#define TEGRA_I2S_CTRL_LRCK_R_LOW (TEGRA_I2S_LRCK_RIGHT_LOW << TEGRA_I2S_CTRL_LRCK_SHIFT) - -#define TEGRA_I2S_BIT_FORMAT_I2S 0 -#define TEGRA_I2S_BIT_FORMAT_RJM 1 -#define TEGRA_I2S_BIT_FORMAT_LJM 2 -#define TEGRA_I2S_BIT_FORMAT_DSP 3 - -#define TEGRA_I2S_CTRL_BIT_FORMAT_SHIFT 10 -#define TEGRA_I2S_CTRL_BIT_FORMAT_MASK (3 << TEGRA_I2S_CTRL_BIT_FORMAT_SHIFT) -#define TEGRA_I2S_CTRL_BIT_FORMAT_I2S (TEGRA_I2S_BIT_FORMAT_I2S << TEGRA_I2S_CTRL_BIT_FORMAT_SHIFT) -#define TEGRA_I2S_CTRL_BIT_FORMAT_RJM (TEGRA_I2S_BIT_FORMAT_RJM << TEGRA_I2S_CTRL_BIT_FORMAT_SHIFT) -#define TEGRA_I2S_CTRL_BIT_FORMAT_LJM (TEGRA_I2S_BIT_FORMAT_LJM << TEGRA_I2S_CTRL_BIT_FORMAT_SHIFT) -#define TEGRA_I2S_CTRL_BIT_FORMAT_DSP (TEGRA_I2S_BIT_FORMAT_DSP << TEGRA_I2S_CTRL_BIT_FORMAT_SHIFT) - -#define TEGRA_I2S_BIT_SIZE_16 0 -#define TEGRA_I2S_BIT_SIZE_20 1 -#define TEGRA_I2S_BIT_SIZE_24 2 -#define TEGRA_I2S_BIT_SIZE_32 3 - -#define TEGRA_I2S_CTRL_BIT_SIZE_SHIFT 8 -#define TEGRA_I2S_CTRL_BIT_SIZE_MASK (3 << TEGRA_I2S_CTRL_BIT_SIZE_SHIFT) -#define TEGRA_I2S_CTRL_BIT_SIZE_16 (TEGRA_I2S_BIT_SIZE_16 << TEGRA_I2S_CTRL_BIT_SIZE_SHIFT) -#define TEGRA_I2S_CTRL_BIT_SIZE_20 (TEGRA_I2S_BIT_SIZE_20 << TEGRA_I2S_CTRL_BIT_SIZE_SHIFT) -#define TEGRA_I2S_CTRL_BIT_SIZE_24 (TEGRA_I2S_BIT_SIZE_24 << TEGRA_I2S_CTRL_BIT_SIZE_SHIFT) -#define TEGRA_I2S_CTRL_BIT_SIZE_32 (TEGRA_I2S_BIT_SIZE_32 << TEGRA_I2S_CTRL_BIT_SIZE_SHIFT) - -#define TEGRA_I2S_FIFO_16_LSB 0 -#define TEGRA_I2S_FIFO_20_LSB 1 -#define TEGRA_I2S_FIFO_24_LSB 2 -#define TEGRA_I2S_FIFO_32 3 -#define TEGRA_I2S_FIFO_PACKED 7 - -#define TEGRA_I2S_CTRL_FIFO_FORMAT_SHIFT 4 -#define TEGRA_I2S_CTRL_FIFO_FORMAT_MASK (7 << TEGRA_I2S_CTRL_FIFO_FORMAT_SHIFT) -#define TEGRA_I2S_CTRL_FIFO_FORMAT_16_LSB (TEGRA_I2S_FIFO_16_LSB << TEGRA_I2S_CTRL_FIFO_FORMAT_SHIFT) -#define TEGRA_I2S_CTRL_FIFO_FORMAT_20_LSB (TEGRA_I2S_FIFO_20_LSB << TEGRA_I2S_CTRL_FIFO_FORMAT_SHIFT) -#define TEGRA_I2S_CTRL_FIFO_FORMAT_24_LSB (TEGRA_I2S_FIFO_24_LSB << TEGRA_I2S_CTRL_FIFO_FORMAT_SHIFT) -#define TEGRA_I2S_CTRL_FIFO_FORMAT_32 (TEGRA_I2S_FIFO_32 << TEGRA_I2S_CTRL_FIFO_FORMAT_SHIFT) -#define TEGRA_I2S_CTRL_FIFO_FORMAT_PACKED (TEGRA_I2S_FIFO_PACKED << TEGRA_I2S_CTRL_FIFO_FORMAT_SHIFT) - -#define TEGRA_I2S_CTRL_IE_FIFO1_ERR (1 << 3) -#define TEGRA_I2S_CTRL_IE_FIFO2_ERR (1 << 2) -#define TEGRA_I2S_CTRL_QE_FIFO1 (1 << 1) -#define TEGRA_I2S_CTRL_QE_FIFO2 (1 << 0) - -/* Fields in TEGRA_I2S_STATUS */ - -#define TEGRA_I2S_STATUS_FIFO1_RDY (1 << 31) -#define TEGRA_I2S_STATUS_FIFO2_RDY (1 << 30) -#define TEGRA_I2S_STATUS_FIFO1_BSY (1 << 29) -#define TEGRA_I2S_STATUS_FIFO2_BSY (1 << 28) -#define TEGRA_I2S_STATUS_FIFO1_ERR (1 << 3) -#define TEGRA_I2S_STATUS_FIFO2_ERR (1 << 2) -#define TEGRA_I2S_STATUS_QS_FIFO1 (1 << 1) -#define TEGRA_I2S_STATUS_QS_FIFO2 (1 << 0) - -/* Fields in TEGRA_I2S_TIMING */ - -#define TEGRA_I2S_TIMING_NON_SYM_ENABLE (1 << 12) -#define TEGRA_I2S_TIMING_CHANNEL_BIT_COUNT_SHIFT 0 -#define TEGRA_I2S_TIMING_CHANNEL_BIT_COUNT_MASK_US 0x7fff -#define TEGRA_I2S_TIMING_CHANNEL_BIT_COUNT_MASK (TEGRA_I2S_TIMING_CHANNEL_BIT_COUNT_MASK_US << TEGRA_I2S_TIMING_CHANNEL_BIT_COUNT_SHIFT) - -/* Fields in TEGRA_I2S_FIFO_SCR */ - -#define TEGRA_I2S_FIFO_SCR_FIFO2_FULL_EMPTY_COUNT_SHIFT 24 -#define TEGRA_I2S_FIFO_SCR_FIFO1_FULL_EMPTY_COUNT_SHIFT 16 -#define TEGRA_I2S_FIFO_SCR_FIFO_FULL_EMPTY_COUNT_MASK 0x3f - -#define TEGRA_I2S_FIFO_SCR_FIFO2_CLR (1 << 12) -#define TEGRA_I2S_FIFO_SCR_FIFO1_CLR (1 << 8) - -#define TEGRA_I2S_FIFO_ATN_LVL_ONE_SLOT 0 -#define TEGRA_I2S_FIFO_ATN_LVL_FOUR_SLOTS 1 -#define TEGRA_I2S_FIFO_ATN_LVL_EIGHT_SLOTS 2 -#define TEGRA_I2S_FIFO_ATN_LVL_TWELVE_SLOTS 3 - -#define TEGRA_I2S_FIFO_SCR_FIFO2_ATN_LVL_SHIFT 4 -#define TEGRA_I2S_FIFO_SCR_FIFO2_ATN_LVL_MASK (3 << TEGRA_I2S_FIFO_SCR_FIFO2_ATN_LVL_SHIFT) -#define TEGRA_I2S_FIFO_SCR_FIFO2_ATN_LVL_ONE_SLOT (TEGRA_I2S_FIFO_ATN_LVL_ONE_SLOT << TEGRA_I2S_FIFO_SCR_FIFO2_ATN_LVL_SHIFT) -#define TEGRA_I2S_FIFO_SCR_FIFO2_ATN_LVL_FOUR_SLOTS (TEGRA_I2S_FIFO_ATN_LVL_FOUR_SLOTS << TEGRA_I2S_FIFO_SCR_FIFO2_ATN_LVL_SHIFT) -#define TEGRA_I2S_FIFO_SCR_FIFO2_ATN_LVL_EIGHT_SLOTS (TEGRA_I2S_FIFO_ATN_LVL_EIGHT_SLOTS << TEGRA_I2S_FIFO_SCR_FIFO2_ATN_LVL_SHIFT) -#define TEGRA_I2S_FIFO_SCR_FIFO2_ATN_LVL_TWELVE_SLOTS (TEGRA_I2S_FIFO_ATN_LVL_TWELVE_SLOTS << TEGRA_I2S_FIFO_SCR_FIFO2_ATN_LVL_SHIFT) - -#define TEGRA_I2S_FIFO_SCR_FIFO1_ATN_LVL_SHIFT 0 -#define TEGRA_I2S_FIFO_SCR_FIFO1_ATN_LVL_MASK (3 << TEGRA_I2S_FIFO_SCR_FIFO1_ATN_LVL_SHIFT) -#define TEGRA_I2S_FIFO_SCR_FIFO1_ATN_LVL_ONE_SLOT (TEGRA_I2S_FIFO_ATN_LVL_ONE_SLOT << TEGRA_I2S_FIFO_SCR_FIFO1_ATN_LVL_SHIFT) -#define TEGRA_I2S_FIFO_SCR_FIFO1_ATN_LVL_FOUR_SLOTS (TEGRA_I2S_FIFO_ATN_LVL_FOUR_SLOTS << TEGRA_I2S_FIFO_SCR_FIFO1_ATN_LVL_SHIFT) -#define TEGRA_I2S_FIFO_SCR_FIFO1_ATN_LVL_EIGHT_SLOTS (TEGRA_I2S_FIFO_ATN_LVL_EIGHT_SLOTS << TEGRA_I2S_FIFO_SCR_FIFO1_ATN_LVL_SHIFT) -#define TEGRA_I2S_FIFO_SCR_FIFO1_ATN_LVL_TWELVE_SLOTS (TEGRA_I2S_FIFO_ATN_LVL_TWELVE_SLOTS << TEGRA_I2S_FIFO_SCR_FIFO1_ATN_LVL_SHIFT) - -struct tegra_i2s { - struct snd_soc_dai_driver dai; - struct clk *clk_i2s; - int clk_refs; - struct tegra_pcm_dma_params capture_dma_data; - struct tegra_pcm_dma_params playback_dma_data; - void __iomem *regs; - struct dentry *debug; - u32 reg_ctrl; -}; - -#endif diff --git a/sound/soc/tegra/tegra_pcm.c b/sound/soc/tegra/tegra_pcm.c index 8b4457137c7..127348dc09b 100644 --- a/sound/soc/tegra/tegra_pcm.c +++ b/sound/soc/tegra/tegra_pcm.c @@ -2,7 +2,7 @@ * tegra_pcm.c - Tegra PCM driver * * Author: Stephen Warren <swarren@nvidia.com> - * Copyright (C) 2010 - NVIDIA, Inc. + * Copyright (C) 2010,2012 - NVIDIA, Inc. * * Based on code copyright/by: * @@ -29,8 +29,8 @@ * */ -#include <linux/module.h> #include <linux/dma-mapping.h> +#include <linux/module.h> #include <linux/slab.h> #include <sound/core.h> #include <sound/pcm.h> @@ -39,8 +39,6 @@ #include "tegra_pcm.h" -#define DRV_NAME "tegra-pcm-audio" - static const struct snd_pcm_hardware tegra_pcm_hardware = { .info = SNDRV_PCM_INFO_MMAP | SNDRV_PCM_INFO_MMAP_VALID | @@ -372,28 +370,18 @@ static struct snd_soc_platform_driver tegra_pcm_platform = { .pcm_free = tegra_pcm_free, }; -static int __devinit tegra_pcm_platform_probe(struct platform_device *pdev) +int __devinit tegra_pcm_platform_register(struct device *dev) { - return snd_soc_register_platform(&pdev->dev, &tegra_pcm_platform); + return snd_soc_register_platform(dev, &tegra_pcm_platform); } +EXPORT_SYMBOL_GPL(tegra_pcm_platform_register); -static int __devexit tegra_pcm_platform_remove(struct platform_device *pdev) +void __devexit tegra_pcm_platform_unregister(struct device *dev) { - snd_soc_unregister_platform(&pdev->dev); - return 0; + snd_soc_unregister_platform(dev); } - -static struct platform_driver tegra_pcm_driver = { - .driver = { - .name = DRV_NAME, - .owner = THIS_MODULE, - }, - .probe = tegra_pcm_platform_probe, - .remove = __devexit_p(tegra_pcm_platform_remove), -}; -module_platform_driver(tegra_pcm_driver); +EXPORT_SYMBOL_GPL(tegra_pcm_platform_unregister); MODULE_AUTHOR("Stephen Warren <swarren@nvidia.com>"); MODULE_DESCRIPTION("Tegra PCM ASoC driver"); MODULE_LICENSE("GPL"); -MODULE_ALIAS("platform:" DRV_NAME); diff --git a/sound/soc/tegra/tegra_pcm.h b/sound/soc/tegra/tegra_pcm.h index dbb90339fe0..985d418a35e 100644 --- a/sound/soc/tegra/tegra_pcm.h +++ b/sound/soc/tegra/tegra_pcm.h @@ -2,7 +2,7 @@ * tegra_pcm.h - Definitions for Tegra PCM driver * * Author: Stephen Warren <swarren@nvidia.com> - * Copyright (C) 2010 - NVIDIA, Inc. + * Copyright (C) 2010,2012 - NVIDIA, Inc. * * Based on code copyright/by: * @@ -52,4 +52,7 @@ struct tegra_runtime_data { struct tegra_dma_channel *dma_chan; }; +int tegra_pcm_platform_register(struct device *dev); +void tegra_pcm_platform_unregister(struct device *dev); + #endif diff --git a/sound/soc/tegra/tegra_spdif.c b/sound/soc/tegra/tegra_spdif.c deleted file mode 100644 index 9ff2c601445..00000000000 --- a/sound/soc/tegra/tegra_spdif.c +++ /dev/null @@ -1,364 +0,0 @@ -/* - * tegra_spdif.c - Tegra SPDIF driver - * - * Author: Stephen Warren <swarren@nvidia.com> - * Copyright (C) 2011 - NVIDIA, Inc. - * - * This program is free software; you can redistribute it and/or - * modify it under the terms of the GNU General Public License - * version 2 as published by the Free Software Foundation. - * - * This program is distributed in the hope that it will be useful, but - * WITHOUT ANY WARRANTY; without even the implied warranty of - * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU - * General Public License for more details. - * - * You should have received a copy of the GNU General Public License - * along with this program; if not, write to the Free Software - * Foundation, Inc., 51 Franklin St, Fifth Floor, Boston, MA - * 02110-1301 USA - * - */ - -#include <linux/clk.h> -#include <linux/module.h> -#include <linux/debugfs.h> -#include <linux/device.h> -#include <linux/platform_device.h> -#include <linux/seq_file.h> -#include <linux/slab.h> -#include <linux/io.h> -#include <mach/iomap.h> -#include <sound/core.h> -#include <sound/pcm.h> -#include <sound/pcm_params.h> -#include <sound/soc.h> - -#include "tegra_spdif.h" - -#define DRV_NAME "tegra-spdif" - -static inline void tegra_spdif_write(struct tegra_spdif *spdif, u32 reg, - u32 val) -{ - __raw_writel(val, spdif->regs + reg); -} - -static inline u32 tegra_spdif_read(struct tegra_spdif *spdif, u32 reg) -{ - return __raw_readl(spdif->regs + reg); -} - -#ifdef CONFIG_DEBUG_FS -static int tegra_spdif_show(struct seq_file *s, void *unused) -{ -#define REG(r) { r, #r } - static const struct { - int offset; - const char *name; - } regs[] = { - REG(TEGRA_SPDIF_CTRL), - REG(TEGRA_SPDIF_STATUS), - REG(TEGRA_SPDIF_STROBE_CTRL), - REG(TEGRA_SPDIF_DATA_FIFO_CSR), - REG(TEGRA_SPDIF_CH_STA_RX_A), - REG(TEGRA_SPDIF_CH_STA_RX_B), - REG(TEGRA_SPDIF_CH_STA_RX_C), - REG(TEGRA_SPDIF_CH_STA_RX_D), - REG(TEGRA_SPDIF_CH_STA_RX_E), - REG(TEGRA_SPDIF_CH_STA_RX_F), - REG(TEGRA_SPDIF_CH_STA_TX_A), - REG(TEGRA_SPDIF_CH_STA_TX_B), - REG(TEGRA_SPDIF_CH_STA_TX_C), - REG(TEGRA_SPDIF_CH_STA_TX_D), - REG(TEGRA_SPDIF_CH_STA_TX_E), - REG(TEGRA_SPDIF_CH_STA_TX_F), - }; -#undef REG - - struct tegra_spdif *spdif = s->private; - int i; - - clk_enable(spdif->clk_spdif_out); - - for (i = 0; i < ARRAY_SIZE(regs); i++) { - u32 val = tegra_spdif_read(spdif, regs[i].offset); - seq_printf(s, "%s = %08x\n", regs[i].name, val); - } - - clk_disable(spdif->clk_spdif_out); - - return 0; -} - -static int tegra_spdif_debug_open(struct inode *inode, struct file *file) -{ - return single_open(file, tegra_spdif_show, inode->i_private); -} - -static const struct file_operations tegra_spdif_debug_fops = { - .open = tegra_spdif_debug_open, - .read = seq_read, - .llseek = seq_lseek, - .release = single_release, -}; - -static void tegra_spdif_debug_add(struct tegra_spdif *spdif) -{ - spdif->debug = debugfs_create_file(DRV_NAME, S_IRUGO, - snd_soc_debugfs_root, spdif, - &tegra_spdif_debug_fops); -} - -static void tegra_spdif_debug_remove(struct tegra_spdif *spdif) -{ - if (spdif->debug) - debugfs_remove(spdif->debug); -} -#else -static inline void tegra_spdif_debug_add(struct tegra_spdif *spdif) -{ -} - -static inline void tegra_spdif_debug_remove(struct tegra_spdif *spdif) -{ -} -#endif - -static int tegra_spdif_hw_params(struct snd_pcm_substream *substream, - struct snd_pcm_hw_params *params, - struct snd_soc_dai *dai) -{ - struct device *dev = substream->pcm->card->dev; - struct tegra_spdif *spdif = snd_soc_dai_get_drvdata(dai); - int ret, spdifclock; - - spdif->reg_ctrl &= ~TEGRA_SPDIF_CTRL_PACK; - spdif->reg_ctrl &= ~TEGRA_SPDIF_CTRL_BIT_MODE_MASK; - switch (params_format(params)) { - case SNDRV_PCM_FORMAT_S16_LE: - spdif->reg_ctrl |= TEGRA_SPDIF_CTRL_PACK; - spdif->reg_ctrl |= TEGRA_SPDIF_CTRL_BIT_MODE_16BIT; - break; - default: - return -EINVAL; - } - - switch (params_rate(params)) { - case 32000: - spdifclock = 4096000; - break; - case 44100: - spdifclock = 5644800; - break; - case 48000: - spdifclock = 6144000; - break; - case 88200: - spdifclock = 11289600; - break; - case 96000: - spdifclock = 12288000; - break; - case 176400: - spdifclock = 22579200; - break; - case 192000: - spdifclock = 24576000; - break; - default: - return -EINVAL; - } - - ret = clk_set_rate(spdif->clk_spdif_out, spdifclock); - if (ret) { - dev_err(dev, "Can't set SPDIF clock rate: %d\n", ret); - return ret; - } - - return 0; -} - -static void tegra_spdif_start_playback(struct tegra_spdif *spdif) -{ - spdif->reg_ctrl |= TEGRA_SPDIF_CTRL_TX_EN; - tegra_spdif_write(spdif, TEGRA_SPDIF_CTRL, spdif->reg_ctrl); -} - -static void tegra_spdif_stop_playback(struct tegra_spdif *spdif) -{ - spdif->reg_ctrl &= ~TEGRA_SPDIF_CTRL_TX_EN; - tegra_spdif_write(spdif, TEGRA_SPDIF_CTRL, spdif->reg_ctrl); -} - -static int tegra_spdif_trigger(struct snd_pcm_substream *substream, int cmd, - struct snd_soc_dai *dai) -{ - struct tegra_spdif *spdif = snd_soc_dai_get_drvdata(dai); - - switch (cmd) { - case SNDRV_PCM_TRIGGER_START: - case SNDRV_PCM_TRIGGER_PAUSE_RELEASE: - case SNDRV_PCM_TRIGGER_RESUME: - if (!spdif->clk_refs) - clk_enable(spdif->clk_spdif_out); - spdif->clk_refs++; - tegra_spdif_start_playback(spdif); - break; - case SNDRV_PCM_TRIGGER_STOP: - case SNDRV_PCM_TRIGGER_PAUSE_PUSH: - case SNDRV_PCM_TRIGGER_SUSPEND: - tegra_spdif_stop_playback(spdif); - spdif->clk_refs--; - if (!spdif->clk_refs) - clk_disable(spdif->clk_spdif_out); - break; - default: - return -EINVAL; - } - - return 0; -} - -static int tegra_spdif_probe(struct snd_soc_dai *dai) -{ - struct tegra_spdif *spdif = snd_soc_dai_get_drvdata(dai); - - dai->capture_dma_data = NULL; - dai->playback_dma_data = &spdif->playback_dma_data; - - return 0; -} - -static const struct snd_soc_dai_ops tegra_spdif_dai_ops = { - .hw_params = tegra_spdif_hw_params, - .trigger = tegra_spdif_trigger, -}; - -static struct snd_soc_dai_driver tegra_spdif_dai = { - .name = DRV_NAME, - .probe = tegra_spdif_probe, - .playback = { - .channels_min = 2, - .channels_max = 2, - .rates = SNDRV_PCM_RATE_32000 | SNDRV_PCM_RATE_44100 | - SNDRV_PCM_RATE_48000, - .formats = SNDRV_PCM_FMTBIT_S16_LE, - }, - .ops = &tegra_spdif_dai_ops, -}; - -static __devinit int tegra_spdif_platform_probe(struct platform_device *pdev) -{ - struct tegra_spdif *spdif; - struct resource *mem, *memregion, *dmareq; - int ret; - - spdif = kzalloc(sizeof(struct tegra_spdif), GFP_KERNEL); - if (!spdif) { - dev_err(&pdev->dev, "Can't allocate tegra_spdif\n"); - ret = -ENOMEM; - goto exit; - } - dev_set_drvdata(&pdev->dev, spdif); - - spdif->clk_spdif_out = clk_get(&pdev->dev, "spdif_out"); - if (IS_ERR(spdif->clk_spdif_out)) { - pr_err("Can't retrieve spdif clock\n"); - ret = PTR_ERR(spdif->clk_spdif_out); - goto err_free; - } - - mem = platform_get_resource(pdev, IORESOURCE_MEM, 0); - if (!mem) { - dev_err(&pdev->dev, "No memory resource\n"); - ret = -ENODEV; - goto err_clk_put; - } - - dmareq = platform_get_resource(pdev, IORESOURCE_DMA, 0); - if (!dmareq) { - dev_err(&pdev->dev, "No DMA resource\n"); - ret = -ENODEV; - goto err_clk_put; - } - - memregion = request_mem_region(mem->start, resource_size(mem), - DRV_NAME); - if (!memregion) { - dev_err(&pdev->dev, "Memory region already claimed\n"); - ret = -EBUSY; - goto err_clk_put; - } - - spdif->regs = ioremap(mem->start, resource_size(mem)); - if (!spdif->regs) { - dev_err(&pdev->dev, "ioremap failed\n"); - ret = -ENOMEM; - goto err_release; - } - - spdif->playback_dma_data.addr = mem->start + TEGRA_SPDIF_DATA_OUT; - spdif->playback_dma_data.wrap = 4; - spdif->playback_dma_data.width = 32; - spdif->playback_dma_data.req_sel = dmareq->start; - - ret = snd_soc_register_dai(&pdev->dev, &tegra_spdif_dai); - if (ret) { - dev_err(&pdev->dev, "Could not register DAI: %d\n", ret); - ret = -ENOMEM; - goto err_unmap; - } - - tegra_spdif_debug_add(spdif); - - return 0; - -err_unmap: - iounmap(spdif->regs); -err_release: - release_mem_region(mem->start, resource_size(mem)); -err_clk_put: - clk_put(spdif->clk_spdif_out); -err_free: - kfree(spdif); -exit: - return ret; -} - -static int __devexit tegra_spdif_platform_remove(struct platform_device *pdev) -{ - struct tegra_spdif *spdif = dev_get_drvdata(&pdev->dev); - struct resource *res; - - snd_soc_unregister_dai(&pdev->dev); - - tegra_spdif_debug_remove(spdif); - - iounmap(spdif->regs); - - res = platform_get_resource(pdev, IORESOURCE_MEM, 0); - release_mem_region(res->start, resource_size(res)); - - clk_put(spdif->clk_spdif_out); - - kfree(spdif); - - return 0; -} - -static struct platform_driver tegra_spdif_driver = { - .driver = { - .name = DRV_NAME, - .owner = THIS_MODULE, - }, - .probe = tegra_spdif_platform_probe, - .remove = __devexit_p(tegra_spdif_platform_remove), -}; - -module_platform_driver(tegra_spdif_driver); - -MODULE_AUTHOR("Stephen Warren <swarren@nvidia.com>"); -MODULE_DESCRIPTION("Tegra SPDIF ASoC driver"); -MODULE_LICENSE("GPL"); -MODULE_ALIAS("platform:" DRV_NAME); diff --git a/sound/soc/tegra/tegra_spdif.h b/sound/soc/tegra/tegra_spdif.h deleted file mode 100644 index 2e03db43027..00000000000 --- a/sound/soc/tegra/tegra_spdif.h +++ /dev/null @@ -1,473 +0,0 @@ -/* - * tegra_spdif.h - Definitions for Tegra SPDIF driver - * - * Author: Stephen Warren <swarren@nvidia.com> - * Copyright (C) 2011 - NVIDIA, Inc. - * - * Based on code copyright/by: - * Copyright (c) 2008-2009, NVIDIA Corporation - * - * This program is free software; you can redistribute it and/or - * modify it under the terms of the GNU General Public License - * version 2 as published by the Free Software Foundation. - * - * This program is distributed in the hope that it will be useful, but - * WITHOUT ANY WARRANTY; without even the implied warranty of - * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU - * General Public License for more details. - * - * You should have received a copy of the GNU General Public License - * along with this program; if not, write to the Free Software - * Foundation, Inc., 51 Franklin St, Fifth Floor, Boston, MA - * 02110-1301 USA - * - */ - -#ifndef __TEGRA_SPDIF_H__ -#define __TEGRA_SPDIF_H__ - -#include "tegra_pcm.h" - -/* Offsets from TEGRA_SPDIF_BASE */ - -#define TEGRA_SPDIF_CTRL 0x0 -#define TEGRA_SPDIF_STATUS 0x4 -#define TEGRA_SPDIF_STROBE_CTRL 0x8 -#define TEGRA_SPDIF_DATA_FIFO_CSR 0x0C -#define TEGRA_SPDIF_DATA_OUT 0x40 -#define TEGRA_SPDIF_DATA_IN 0x80 -#define TEGRA_SPDIF_CH_STA_RX_A 0x100 -#define TEGRA_SPDIF_CH_STA_RX_B 0x104 -#define TEGRA_SPDIF_CH_STA_RX_C 0x108 -#define TEGRA_SPDIF_CH_STA_RX_D 0x10C -#define TEGRA_SPDIF_CH_STA_RX_E 0x110 -#define TEGRA_SPDIF_CH_STA_RX_F 0x114 -#define TEGRA_SPDIF_CH_STA_TX_A 0x140 -#define TEGRA_SPDIF_CH_STA_TX_B 0x144 -#define TEGRA_SPDIF_CH_STA_TX_C 0x148 -#define TEGRA_SPDIF_CH_STA_TX_D 0x14C -#define TEGRA_SPDIF_CH_STA_TX_E 0x150 -#define TEGRA_SPDIF_CH_STA_TX_F 0x154 -#define TEGRA_SPDIF_USR_STA_RX_A 0x180 -#define TEGRA_SPDIF_USR_DAT_TX_A 0x1C0 - -/* Fields in TEGRA_SPDIF_CTRL */ - -/* Start capturing from 0=right, 1=left channel */ -#define TEGRA_SPDIF_CTRL_CAP_LC (1 << 30) - -/* SPDIF receiver(RX) enable */ -#define TEGRA_SPDIF_CTRL_RX_EN (1 << 29) - -/* SPDIF Transmitter(TX) enable */ -#define TEGRA_SPDIF_CTRL_TX_EN (1 << 28) - -/* Transmit Channel status */ -#define TEGRA_SPDIF_CTRL_TC_EN (1 << 27) - -/* Transmit user Data */ -#define TEGRA_SPDIF_CTRL_TU_EN (1 << 26) - -/* Interrupt on transmit error */ -#define TEGRA_SPDIF_CTRL_IE_TXE (1 << 25) - -/* Interrupt on receive error */ -#define TEGRA_SPDIF_CTRL_IE_RXE (1 << 24) - -/* Interrupt on invalid preamble */ -#define TEGRA_SPDIF_CTRL_IE_P (1 << 23) - -/* Interrupt on "B" preamble */ -#define TEGRA_SPDIF_CTRL_IE_B (1 << 22) - -/* Interrupt when block of channel status received */ -#define TEGRA_SPDIF_CTRL_IE_C (1 << 21) - -/* Interrupt when a valid information unit (IU) is received */ -#define TEGRA_SPDIF_CTRL_IE_U (1 << 20) - -/* Interrupt when RX user FIFO attention level is reached */ -#define TEGRA_SPDIF_CTRL_QE_RU (1 << 19) - -/* Interrupt when TX user FIFO attention level is reached */ -#define TEGRA_SPDIF_CTRL_QE_TU (1 << 18) - -/* Interrupt when RX data FIFO attention level is reached */ -#define TEGRA_SPDIF_CTRL_QE_RX (1 << 17) - -/* Interrupt when TX data FIFO attention level is reached */ -#define TEGRA_SPDIF_CTRL_QE_TX (1 << 16) - -/* Loopback test mode enable */ -#define TEGRA_SPDIF_CTRL_LBK_EN (1 << 15) - -/* - * Pack data mode: - * 0 = Single data (16 bit needs to be padded to match the - * interface data bit size). - * 1 = Packeted left/right channel data into a single word. - */ -#define TEGRA_SPDIF_CTRL_PACK (1 << 14) - -/* - * 00 = 16bit data - * 01 = 20bit data - * 10 = 24bit data - * 11 = raw data - */ -#define TEGRA_SPDIF_BIT_MODE_16BIT 0 -#define TEGRA_SPDIF_BIT_MODE_20BIT 1 -#define TEGRA_SPDIF_BIT_MODE_24BIT 2 -#define TEGRA_SPDIF_BIT_MODE_RAW 3 - -#define TEGRA_SPDIF_CTRL_BIT_MODE_SHIFT 12 -#define TEGRA_SPDIF_CTRL_BIT_MODE_MASK (3 << TEGRA_SPDIF_CTRL_BIT_MODE_SHIFT) -#define TEGRA_SPDIF_CTRL_BIT_MODE_16BIT (TEGRA_SPDIF_BIT_MODE_16BIT << TEGRA_SPDIF_CTRL_BIT_MODE_SHIFT) -#define TEGRA_SPDIF_CTRL_BIT_MODE_20BIT (TEGRA_SPDIF_BIT_MODE_20BIT << TEGRA_SPDIF_CTRL_BIT_MODE_SHIFT) -#define TEGRA_SPDIF_CTRL_BIT_MODE_24BIT (TEGRA_SPDIF_BIT_MODE_24BIT << TEGRA_SPDIF_CTRL_BIT_MODE_SHIFT) -#define TEGRA_SPDIF_CTRL_BIT_MODE_RAW (TEGRA_SPDIF_BIT_MODE_RAW << TEGRA_SPDIF_CTRL_BIT_MODE_SHIFT) - -/* Fields in TEGRA_SPDIF_STATUS */ - -/* - * Note: IS_P, IS_B, IS_C, and IS_U are sticky bits. Software must - * write a 1 to the corresponding bit location to clear the status. - */ - -/* - * Receiver(RX) shifter is busy receiving data. - * This bit is asserted when the receiver first locked onto the - * preamble of the data stream after RX_EN is asserted. This bit is - * deasserted when either, - * (a) the end of a frame is reached after RX_EN is deeasserted, or - * (b) the SPDIF data stream becomes inactive. - */ -#define TEGRA_SPDIF_STATUS_RX_BSY (1 << 29) - -/* - * Transmitter(TX) shifter is busy transmitting data. - * This bit is asserted when TX_EN is asserted. - * This bit is deasserted when the end of a frame is reached after - * TX_EN is deasserted. - */ -#define TEGRA_SPDIF_STATUS_TX_BSY (1 << 28) - -/* - * TX is busy shifting out channel status. - * This bit is asserted when both TX_EN and TC_EN are asserted and - * data from CH_STA_TX_A register is loaded into the internal shifter. - * This bit is deasserted when either, - * (a) the end of a frame is reached after TX_EN is deasserted, or - * (b) CH_STA_TX_F register is loaded into the internal shifter. - */ -#define TEGRA_SPDIF_STATUS_TC_BSY (1 << 27) - -/* - * TX User data FIFO busy. - * This bit is asserted when TX_EN and TXU_EN are asserted and - * there's data in the TX user FIFO. This bit is deassert when either, - * (a) the end of a frame is reached after TX_EN is deasserted, or - * (b) there's no data left in the TX user FIFO. - */ -#define TEGRA_SPDIF_STATUS_TU_BSY (1 << 26) - -/* TX FIFO Underrun error status */ -#define TEGRA_SPDIF_STATUS_TX_ERR (1 << 25) - -/* RX FIFO Overrun error status */ -#define TEGRA_SPDIF_STATUS_RX_ERR (1 << 24) - -/* Preamble status: 0=Preamble OK, 1=bad/missing preamble */ -#define TEGRA_SPDIF_STATUS_IS_P (1 << 23) - -/* B-preamble detection status: 0=not detected, 1=B-preamble detected */ -#define TEGRA_SPDIF_STATUS_IS_B (1 << 22) - -/* - * RX channel block data receive status: - * 0=entire block not recieved yet. - * 1=received entire block of channel status, - */ -#define TEGRA_SPDIF_STATUS_IS_C (1 << 21) - -/* RX User Data Valid flag: 1=valid IU detected, 0 = no IU detected. */ -#define TEGRA_SPDIF_STATUS_IS_U (1 << 20) - -/* - * RX User FIFO Status: - * 1=attention level reached, 0=attention level not reached. - */ -#define TEGRA_SPDIF_STATUS_QS_RU (1 << 19) - -/* - * TX User FIFO Status: - * 1=attention level reached, 0=attention level not reached. - */ -#define TEGRA_SPDIF_STATUS_QS_TU (1 << 18) - -/* - * RX Data FIFO Status: - * 1=attention level reached, 0=attention level not reached. - */ -#define TEGRA_SPDIF_STATUS_QS_RX (1 << 17) - -/* - * TX Data FIFO Status: - * 1=attention level reached, 0=attention level not reached. - */ -#define TEGRA_SPDIF_STATUS_QS_TX (1 << 16) - -/* Fields in TEGRA_SPDIF_STROBE_CTRL */ - -/* - * Indicates the approximate number of detected SPDIFIN clocks within a - * bi-phase period. - */ -#define TEGRA_SPDIF_STROBE_CTRL_PERIOD_SHIFT 16 -#define TEGRA_SPDIF_STROBE_CTRL_PERIOD_MASK (0xff << TEGRA_SPDIF_STROBE_CTRL_PERIOD_SHIFT) - -/* Data strobe mode: 0=Auto-locked 1=Manual locked */ -#define TEGRA_SPDIF_STROBE_CTRL_STROBE (1 << 15) - -/* - * Manual data strobe time within the bi-phase clock period (in terms of - * the number of over-sampling clocks). - */ -#define TEGRA_SPDIF_STROBE_CTRL_DATA_STROBES_SHIFT 8 -#define TEGRA_SPDIF_STROBE_CTRL_DATA_STROBES_MASK (0x1f << TEGRA_SPDIF_STROBE_CTRL_DATA_STROBES_SHIFT) - -/* - * Manual SPDIFIN bi-phase clock period (in terms of the number of - * over-sampling clocks). - */ -#define TEGRA_SPDIF_STROBE_CTRL_CLOCK_PERIOD_SHIFT 0 -#define TEGRA_SPDIF_STROBE_CTRL_CLOCK_PERIOD_MASK (0x3f << TEGRA_SPDIF_STROBE_CTRL_CLOCK_PERIOD_SHIFT) - -/* Fields in SPDIF_DATA_FIFO_CSR */ - -/* Clear Receiver User FIFO (RX USR.FIFO) */ -#define TEGRA_SPDIF_DATA_FIFO_CSR_RU_CLR (1 << 31) - -#define TEGRA_SPDIF_FIFO_ATN_LVL_U_ONE_SLOT 0 -#define TEGRA_SPDIF_FIFO_ATN_LVL_U_TWO_SLOTS 1 -#define TEGRA_SPDIF_FIFO_ATN_LVL_U_THREE_SLOTS 2 -#define TEGRA_SPDIF_FIFO_ATN_LVL_U_FOUR_SLOTS 3 - -/* RU FIFO attention level */ -#define TEGRA_SPDIF_DATA_FIFO_CSR_RU_ATN_LVL_SHIFT 29 -#define TEGRA_SPDIF_DATA_FIFO_CSR_RU_ATN_LVL_MASK \ - (0x3 << TEGRA_SPDIF_DATA_FIFO_CSR_RU_ATN_LVL_SHIFT) -#define TEGRA_SPDIF_DATA_FIFO_CSR_RU_ATN_LVL_RU1_WORD_FULL \ - (TEGRA_SPDIF_FIFO_ATN_LVL_U_ONE_SLOT << TEGRA_SPDIF_DATA_FIFO_CSR_RU_ATN_LVL_SHIFT) -#define TEGRA_SPDIF_DATA_FIFO_CSR_RU_ATN_LVL_RU2_WORD_FULL \ - (TEGRA_SPDIF_FIFO_ATN_LVL_U_TWO_SLOTS << TEGRA_SPDIF_DATA_FIFO_CSR_RU_ATN_LVL_SHIFT) -#define TEGRA_SPDIF_DATA_FIFO_CSR_RU_ATN_LVL_RU3_WORD_FULL \ - (TEGRA_SPDIF_FIFO_ATN_LVL_U_THREE_SLOTS << TEGRA_SPDIF_DATA_FIFO_CSR_RU_ATN_LVL_SHIFT) -#define TEGRA_SPDIF_DATA_FIFO_CSR_RU_ATN_LVL_RU4_WORD_FULL \ - (TEGRA_SPDIF_FIFO_ATN_LVL_U_FOUR_SLOTS << TEGRA_SPDIF_DATA_FIFO_CSR_RU_ATN_LVL_SHIFT) - -/* Number of RX USR.FIFO levels with valid data. */ -#define TEGRA_SPDIF_DATA_FIFO_CSR_RU_FULL_COUNT_SHIFT 24 -#define TEGRA_SPDIF_DATA_FIFO_CSR_RU_FULL_COUNT_MASK (0x1f << TEGRA_SPDIF_DATA_FIFO_CSR_RU_FULL_COUNT_SHIFT) - -/* Clear Transmitter User FIFO (TX USR.FIFO) */ -#define TEGRA_SPDIF_DATA_FIFO_CSR_TU_CLR (1 << 23) - -/* TU FIFO attention level */ -#define TEGRA_SPDIF_DATA_FIFO_CSR_TU_ATN_LVL_SHIFT 21 -#define TEGRA_SPDIF_DATA_FIFO_CSR_TU_ATN_LVL_MASK \ - (0x3 << TEGRA_SPDIF_DATA_FIFO_CSR_TU_ATN_LVL_SHIFT) -#define TEGRA_SPDIF_DATA_FIFO_CSR_TU_ATN_LVL_TU1_WORD_FULL \ - (TEGRA_SPDIF_FIFO_ATN_LVL_U_ONE_SLOT << TEGRA_SPDIF_DATA_FIFO_CSR_TU_ATN_LVL_SHIFT) -#define TEGRA_SPDIF_DATA_FIFO_CSR_TU_ATN_LVL_TU2_WORD_FULL \ - (TEGRA_SPDIF_FIFO_ATN_LVL_U_TWO_SLOTS << TEGRA_SPDIF_DATA_FIFO_CSR_TU_ATN_LVL_SHIFT) -#define TEGRA_SPDIF_DATA_FIFO_CSR_TU_ATN_LVL_TU3_WORD_FULL \ - (TEGRA_SPDIF_FIFO_ATN_LVL_U_THREE_SLOTS << TEGRA_SPDIF_DATA_FIFO_CSR_TU_ATN_LVL_SHIFT) -#define TEGRA_SPDIF_DATA_FIFO_CSR_TU_ATN_LVL_TU4_WORD_FULL \ - (TEGRA_SPDIF_FIFO_ATN_LVL_U_FOUR_SLOTS << TEGRA_SPDIF_DATA_FIFO_CSR_TU_ATN_LVL_SHIFT) - -/* Number of TX USR.FIFO levels that could be filled. */ -#define TEGRA_SPDIF_DATA_FIFO_CSR_TU_EMPTY_COUNT_SHIFT 16 -#define TEGRA_SPDIF_DATA_FIFO_CSR_TU_EMPTY_COUNT_MASK (0x1f << SPDIF_DATA_FIFO_CSR_TU_EMPTY_COUNT_SHIFT) - -/* Clear Receiver Data FIFO (RX DATA.FIFO) */ -#define TEGRA_SPDIF_DATA_FIFO_CSR_RX_CLR (1 << 15) - -#define TEGRA_SPDIF_FIFO_ATN_LVL_D_ONE_SLOT 0 -#define TEGRA_SPDIF_FIFO_ATN_LVL_D_FOUR_SLOTS 1 -#define TEGRA_SPDIF_FIFO_ATN_LVL_D_EIGHT_SLOTS 2 -#define TEGRA_SPDIF_FIFO_ATN_LVL_D_TWELVE_SLOTS 3 - -/* RU FIFO attention level */ -#define TEGRA_SPDIF_DATA_FIFO_CSR_RX_ATN_LVL_SHIFT 13 -#define TEGRA_SPDIF_DATA_FIFO_CSR_RX_ATN_LVL_MASK \ - (0x3 << TEGRA_SPDIF_DATA_FIFO_CSR_RX_ATN_LVL_SHIFT) -#define TEGRA_SPDIF_DATA_FIFO_CSR_RX_ATN_LVL_RU1_WORD_FULL \ - (TEGRA_SPDIF_FIFO_ATN_LVL_D_ONE_SLOT << TEGRA_SPDIF_DATA_FIFO_CSR_RX_ATN_LVL_SHIFT) -#define TEGRA_SPDIF_DATA_FIFO_CSR_RX_ATN_LVL_RU4_WORD_FULL \ - (TEGRA_SPDIF_FIFO_ATN_LVL_D_FOUR_SLOTS << TEGRA_SPDIF_DATA_FIFO_CSR_RX_ATN_LVL_SHIFT) -#define TEGRA_SPDIF_DATA_FIFO_CSR_RX_ATN_LVL_RU8_WORD_FULL \ - (TEGRA_SPDIF_FIFO_ATN_LVL_D_EIGHT_SLOTS << TEGRA_SPDIF_DATA_FIFO_CSR_RX_ATN_LVL_SHIFT) -#define TEGRA_SPDIF_DATA_FIFO_CSR_RX_ATN_LVL_RU12_WORD_FULL \ - (TEGRA_SPDIF_FIFO_ATN_LVL_D_TWELVE_SLOTS << TEGRA_SPDIF_DATA_FIFO_CSR_RX_ATN_LVL_SHIFT) - -/* Number of RX DATA.FIFO levels with valid data. */ -#define TEGRA_SPDIF_DATA_FIFO_CSR_RX_FULL_COUNT_SHIFT 8 -#define TEGRA_SPDIF_DATA_FIFO_CSR_RX_FULL_COUNT_MASK (0x1f << TEGRA_SPDIF_DATA_FIFO_CSR_RX_FULL_COUNT_SHIFT) - -/* Clear Transmitter Data FIFO (TX DATA.FIFO) */ -#define TEGRA_SPDIF_DATA_FIFO_CSR_TX_CLR (1 << 7) - -/* TU FIFO attention level */ -#define TEGRA_SPDIF_DATA_FIFO_CSR_TX_ATN_LVL_SHIFT 5 -#define TEGRA_SPDIF_DATA_FIFO_CSR_TX_ATN_LVL_MASK \ - (0x3 << TEGRA_SPDIF_DATA_FIFO_CSR_TX_ATN_LVL_SHIFT) -#define TEGRA_SPDIF_DATA_FIFO_CSR_TX_ATN_LVL_TU1_WORD_FULL \ - (TEGRA_SPDIF_FIFO_ATN_LVL_D_ONE_SLOT << TEGRA_SPDIF_DATA_FIFO_CSR_TX_ATN_LVL_SHIFT) -#define TEGRA_SPDIF_DATA_FIFO_CSR_TX_ATN_LVL_TU4_WORD_FULL \ - (TEGRA_SPDIF_FIFO_ATN_LVL_D_FOUR_SLOTS << TEGRA_SPDIF_DATA_FIFO_CSR_TX_ATN_LVL_SHIFT) -#define TEGRA_SPDIF_DATA_FIFO_CSR_TX_ATN_LVL_TU8_WORD_FULL \ - (TEGRA_SPDIF_FIFO_ATN_LVL_D_EIGHT_SLOTS << TEGRA_SPDIF_DATA_FIFO_CSR_TX_ATN_LVL_SHIFT) -#define TEGRA_SPDIF_DATA_FIFO_CSR_TX_ATN_LVL_TU12_WORD_FULL \ - (TEGRA_SPDIF_FIFO_ATN_LVL_D_TWELVE_SLOTS << TEGRA_SPDIF_DATA_FIFO_CSR_TX_ATN_LVL_SHIFT) - -/* Number of TX DATA.FIFO levels that could be filled. */ -#define TEGRA_SPDIF_DATA_FIFO_CSR_TX_EMPTY_COUNT_SHIFT 0 -#define TEGRA_SPDIF_DATA_FIFO_CSR_TX_EMPTY_COUNT_MASK (0x1f << SPDIF_DATA_FIFO_CSR_TX_EMPTY_COUNT_SHIFT) - -/* Fields in TEGRA_SPDIF_DATA_OUT */ - -/* - * This register has 5 different formats: - * 16-bit (BIT_MODE=00, PACK=0) - * 20-bit (BIT_MODE=01, PACK=0) - * 24-bit (BIT_MODE=10, PACK=0) - * raw (BIT_MODE=11, PACK=0) - * 16-bit packed (BIT_MODE=00, PACK=1) - */ - -#define TEGRA_SPDIF_DATA_OUT_DATA_16_SHIFT 0 -#define TEGRA_SPDIF_DATA_OUT_DATA_16_MASK (0xffff << TEGRA_SPDIF_DATA_OUT_DATA_16_SHIFT) - -#define TEGRA_SPDIF_DATA_OUT_DATA_20_SHIFT 0 -#define TEGRA_SPDIF_DATA_OUT_DATA_20_MASK (0xfffff << TEGRA_SPDIF_DATA_OUT_DATA_20_SHIFT) - -#define TEGRA_SPDIF_DATA_OUT_DATA_24_SHIFT 0 -#define TEGRA_SPDIF_DATA_OUT_DATA_24_MASK (0xffffff << TEGRA_SPDIF_DATA_OUT_DATA_24_SHIFT) - -#define TEGRA_SPDIF_DATA_OUT_DATA_RAW_P (1 << 31) -#define TEGRA_SPDIF_DATA_OUT_DATA_RAW_C (1 << 30) -#define TEGRA_SPDIF_DATA_OUT_DATA_RAW_U (1 << 29) -#define TEGRA_SPDIF_DATA_OUT_DATA_RAW_V (1 << 28) - -#define TEGRA_SPDIF_DATA_OUT_DATA_RAW_DATA_SHIFT 8 -#define TEGRA_SPDIF_DATA_OUT_DATA_RAW_DATA_MASK (0xfffff << TEGRA_SPDIF_DATA_OUT_DATA_RAW_DATA_SHIFT) - -#define TEGRA_SPDIF_DATA_OUT_DATA_RAW_AUX_SHIFT 4 -#define TEGRA_SPDIF_DATA_OUT_DATA_RAW_AUX_MASK (0xf << TEGRA_SPDIF_DATA_OUT_DATA_RAW_AUX_SHIFT) - -#define TEGRA_SPDIF_DATA_OUT_DATA_RAW_PREAMBLE_SHIFT 0 -#define TEGRA_SPDIF_DATA_OUT_DATA_RAW_PREAMBLE_MASK (0xf << TEGRA_SPDIF_DATA_OUT_DATA_RAW_PREAMBLE_SHIFT) - -#define TEGRA_SPDIF_DATA_OUT_DATA_16_PACKED_RIGHT_SHIFT 16 -#define TEGRA_SPDIF_DATA_OUT_DATA_16_PACKED_RIGHT_MASK (0xffff << TEGRA_SPDIF_DATA_OUT_DATA_16_PACKED_RIGHT_SHIFT) - -#define TEGRA_SPDIF_DATA_OUT_DATA_16_PACKED_LEFT_SHIFT 0 -#define TEGRA_SPDIF_DATA_OUT_DATA_16_PACKED_LEFT_MASK (0xffff << TEGRA_SPDIF_DATA_OUT_DATA_16_PACKED_LEFT_SHIFT) - -/* Fields in TEGRA_SPDIF_DATA_IN */ - -/* - * This register has 5 different formats: - * 16-bit (BIT_MODE=00, PACK=0) - * 20-bit (BIT_MODE=01, PACK=0) - * 24-bit (BIT_MODE=10, PACK=0) - * raw (BIT_MODE=11, PACK=0) - * 16-bit packed (BIT_MODE=00, PACK=1) - * - * Bits 31:24 are common to all modes except 16-bit packed - */ - -#define TEGRA_SPDIF_DATA_IN_DATA_P (1 << 31) -#define TEGRA_SPDIF_DATA_IN_DATA_C (1 << 30) -#define TEGRA_SPDIF_DATA_IN_DATA_U (1 << 29) -#define TEGRA_SPDIF_DATA_IN_DATA_V (1 << 28) - -#define TEGRA_SPDIF_DATA_IN_DATA_PREAMBLE_SHIFT 24 -#define TEGRA_SPDIF_DATA_IN_DATA_PREAMBLE_MASK (0xf << TEGRA_SPDIF_DATA_IN_DATA_PREAMBLE_SHIFT) - -#define TEGRA_SPDIF_DATA_IN_DATA_16_SHIFT 0 -#define TEGRA_SPDIF_DATA_IN_DATA_16_MASK (0xffff << TEGRA_SPDIF_DATA_IN_DATA_16_SHIFT) - -#define TEGRA_SPDIF_DATA_IN_DATA_20_SHIFT 0 -#define TEGRA_SPDIF_DATA_IN_DATA_20_MASK (0xfffff << TEGRA_SPDIF_DATA_IN_DATA_20_SHIFT) - -#define TEGRA_SPDIF_DATA_IN_DATA_24_SHIFT 0 -#define TEGRA_SPDIF_DATA_IN_DATA_24_MASK (0xffffff << TEGRA_SPDIF_DATA_IN_DATA_24_SHIFT) - -#define TEGRA_SPDIF_DATA_IN_DATA_RAW_DATA_SHIFT 8 -#define TEGRA_SPDIF_DATA_IN_DATA_RAW_DATA_MASK (0xfffff << TEGRA_SPDIF_DATA_IN_DATA_RAW_DATA_SHIFT) - -#define TEGRA_SPDIF_DATA_IN_DATA_RAW_AUX_SHIFT 4 -#define TEGRA_SPDIF_DATA_IN_DATA_RAW_AUX_MASK (0xf << TEGRA_SPDIF_DATA_IN_DATA_RAW_AUX_SHIFT) - -#define TEGRA_SPDIF_DATA_IN_DATA_RAW_PREAMBLE_SHIFT 0 -#define TEGRA_SPDIF_DATA_IN_DATA_RAW_PREAMBLE_MASK (0xf << TEGRA_SPDIF_DATA_IN_DATA_RAW_PREAMBLE_SHIFT) - -#define TEGRA_SPDIF_DATA_IN_DATA_16_PACKED_RIGHT_SHIFT 16 -#define TEGRA_SPDIF_DATA_IN_DATA_16_PACKED_RIGHT_MASK (0xffff << TEGRA_SPDIF_DATA_IN_DATA_16_PACKED_RIGHT_SHIFT) - -#define TEGRA_SPDIF_DATA_IN_DATA_16_PACKED_LEFT_SHIFT 0 -#define TEGRA_SPDIF_DATA_IN_DATA_16_PACKED_LEFT_MASK (0xffff << TEGRA_SPDIF_DATA_IN_DATA_16_PACKED_LEFT_SHIFT) - -/* Fields in TEGRA_SPDIF_CH_STA_RX_A */ -/* Fields in TEGRA_SPDIF_CH_STA_RX_B */ -/* Fields in TEGRA_SPDIF_CH_STA_RX_C */ -/* Fields in TEGRA_SPDIF_CH_STA_RX_D */ -/* Fields in TEGRA_SPDIF_CH_STA_RX_E */ -/* Fields in TEGRA_SPDIF_CH_STA_RX_F */ - -/* - * The 6-word receive channel data page buffer holds a block (192 frames) of - * channel status information. The order of receive is from LSB to MSB - * bit, and from CH_STA_RX_A to CH_STA_RX_F then back to CH_STA_RX_A. - */ - -/* Fields in TEGRA_SPDIF_CH_STA_TX_A */ -/* Fields in TEGRA_SPDIF_CH_STA_TX_B */ -/* Fields in TEGRA_SPDIF_CH_STA_TX_C */ -/* Fields in TEGRA_SPDIF_CH_STA_TX_D */ -/* Fields in TEGRA_SPDIF_CH_STA_TX_E */ -/* Fields in TEGRA_SPDIF_CH_STA_TX_F */ - -/* - * The 6-word transmit channel data page buffer holds a block (192 frames) of - * channel status information. The order of transmission is from LSB to MSB - * bit, and from CH_STA_TX_A to CH_STA_TX_F then back to CH_STA_TX_A. - */ - -/* Fields in TEGRA_SPDIF_USR_STA_RX_A */ - -/* - * This 4-word deep FIFO receives user FIFO field information. The order of - * receive is from LSB to MSB bit. - */ - -/* Fields in TEGRA_SPDIF_USR_DAT_TX_A */ - -/* - * This 4-word deep FIFO transmits user FIFO field information. The order of - * transmission is from LSB to MSB bit. - */ - -struct tegra_spdif { - struct clk *clk_spdif_out; - int clk_refs; - struct tegra_pcm_dma_params capture_dma_data; - struct tegra_pcm_dma_params playback_dma_data; - void __iomem *regs; - struct dentry *debug; - u32 reg_ctrl; -}; - -#endif diff --git a/sound/soc/tegra/tegra_wm8753.c b/sound/soc/tegra/tegra_wm8753.c new file mode 100644 index 00000000000..4e77026807a --- /dev/null +++ b/sound/soc/tegra/tegra_wm8753.c @@ -0,0 +1,224 @@ +/* + * tegra_wm8753.c - Tegra machine ASoC driver for boards using WM8753 codec. + * + * Author: Stephen Warren <swarren@nvidia.com> + * Copyright (C) 2010-2012 - NVIDIA, Inc. + * + * Based on code copyright/by: + * + * (c) 2009, 2010 Nvidia Graphics Pvt. Ltd. + * + * Copyright 2007 Wolfson Microelectronics PLC. + * Author: Graeme Gregory + * graeme.gregory@wolfsonmicro.com or linux@wolfsonmicro.com + * + * This program is free software; you can redistribute it and/or + * modify it under the terms of the GNU General Public License + * version 2 as published by the Free Software Foundation. + * + * This program is distributed in the hope that it will be useful, but + * WITHOUT ANY WARRANTY; without even the implied warranty of + * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU + * General Public License for more details. + * + * You should have received a copy of the GNU General Public License + * along with this program; if not, write to the Free Software + * Foundation, Inc., 51 Franklin St, Fifth Floor, Boston, MA + * 02110-1301 USA + * + */ + +#include <asm/mach-types.h> + +#include <linux/module.h> +#include <linux/platform_device.h> +#include <linux/slab.h> +#include <linux/gpio.h> +#include <linux/of_gpio.h> + +#include <sound/core.h> +#include <sound/jack.h> +#include <sound/pcm.h> +#include <sound/pcm_params.h> +#include <sound/soc.h> + +#include "../codecs/wm8753.h" + +#include "tegra_asoc_utils.h" + +#define DRV_NAME "tegra-snd-wm8753" + +struct tegra_wm8753 { + struct tegra_asoc_utils_data util_data; +}; + +static int tegra_wm8753_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_soc_dai *codec_dai = rtd->codec_dai; + struct snd_soc_codec *codec = rtd->codec; + struct snd_soc_card *card = codec->card; + struct tegra_wm8753 *machine = snd_soc_card_get_drvdata(card); + int srate, mclk; + int err; + + srate = params_rate(params); + switch (srate) { + case 11025: + case 22050: + case 44100: + case 88200: + mclk = 11289600; + break; + default: + mclk = 12288000; + break; + } + + err = tegra_asoc_utils_set_rate(&machine->util_data, srate, mclk); + if (err < 0) { + dev_err(card->dev, "Can't configure clocks\n"); + return err; + } + + err = snd_soc_dai_set_sysclk(codec_dai, WM8753_MCLK, mclk, + SND_SOC_CLOCK_IN); + if (err < 0) { + dev_err(card->dev, "codec_dai clock not set\n"); + return err; + } + + return 0; +} + +static struct snd_soc_ops tegra_wm8753_ops = { + .hw_params = tegra_wm8753_hw_params, +}; + +static const struct snd_soc_dapm_widget tegra_wm8753_dapm_widgets[] = { + SND_SOC_DAPM_HP("Headphone Jack", NULL), + SND_SOC_DAPM_MIC("Mic Jack", NULL), +}; + +static struct snd_soc_dai_link tegra_wm8753_dai = { + .name = "WM8753", + .stream_name = "WM8753 PCM", + .codec_dai_name = "wm8753-hifi", + .ops = &tegra_wm8753_ops, + .dai_fmt = SND_SOC_DAIFMT_I2S | + SND_SOC_DAIFMT_NB_NF | + SND_SOC_DAIFMT_CBS_CFS, +}; + +static struct snd_soc_card snd_soc_tegra_wm8753 = { + .name = "tegra-wm8753", + .owner = THIS_MODULE, + .dai_link = &tegra_wm8753_dai, + .num_links = 1, + + .dapm_widgets = tegra_wm8753_dapm_widgets, + .num_dapm_widgets = ARRAY_SIZE(tegra_wm8753_dapm_widgets), + .fully_routed = true, +}; + +static __devinit int tegra_wm8753_driver_probe(struct platform_device *pdev) +{ + struct snd_soc_card *card = &snd_soc_tegra_wm8753; + struct tegra_wm8753 *machine; + int ret; + + machine = devm_kzalloc(&pdev->dev, sizeof(struct tegra_wm8753), + GFP_KERNEL); + if (!machine) { + dev_err(&pdev->dev, "Can't allocate tegra_wm8753 struct\n"); + ret = -ENOMEM; + goto err; + } + + card->dev = &pdev->dev; + platform_set_drvdata(pdev, card); + snd_soc_card_set_drvdata(card, machine); + + ret = snd_soc_of_parse_card_name(card, "nvidia,model"); + if (ret) + goto err; + + ret = snd_soc_of_parse_audio_routing(card, "nvidia,audio-routing"); + if (ret) + goto err; + + tegra_wm8753_dai.codec_of_node = of_parse_phandle( + pdev->dev.of_node, "nvidia,audio-codec", 0); + if (!tegra_wm8753_dai.codec_of_node) { + dev_err(&pdev->dev, + "Property 'nvidia,audio-codec' missing or invalid\n"); + ret = -EINVAL; + goto err; + } + + tegra_wm8753_dai.cpu_dai_of_node = of_parse_phandle( + pdev->dev.of_node, "nvidia,i2s-controller", 0); + if (!tegra_wm8753_dai.cpu_dai_of_node) { + dev_err(&pdev->dev, + "Property 'nvidia,i2s-controller' missing or invalid\n"); + ret = -EINVAL; + goto err; + } + + tegra_wm8753_dai.platform_of_node = + tegra_wm8753_dai.cpu_dai_of_node; + + ret = tegra_asoc_utils_init(&machine->util_data, &pdev->dev); + if (ret) + goto err; + + ret = snd_soc_register_card(card); + if (ret) { + dev_err(&pdev->dev, "snd_soc_register_card failed (%d)\n", + ret); + goto err_fini_utils; + } + + return 0; + +err_fini_utils: + tegra_asoc_utils_fini(&machine->util_data); +err: + return ret; +} + +static int __devexit tegra_wm8753_driver_remove(struct platform_device *pdev) +{ + struct snd_soc_card *card = platform_get_drvdata(pdev); + struct tegra_wm8753 *machine = snd_soc_card_get_drvdata(card); + + snd_soc_unregister_card(card); + + tegra_asoc_utils_fini(&machine->util_data); + + return 0; +} + +static const struct of_device_id tegra_wm8753_of_match[] __devinitconst = { + { .compatible = "nvidia,tegra-audio-wm8753", }, + {}, +}; + +static struct platform_driver tegra_wm8753_driver = { + .driver = { + .name = DRV_NAME, + .owner = THIS_MODULE, + .pm = &snd_soc_pm_ops, + .of_match_table = tegra_wm8753_of_match, + }, + .probe = tegra_wm8753_driver_probe, + .remove = __devexit_p(tegra_wm8753_driver_remove), +}; +module_platform_driver(tegra_wm8753_driver); + +MODULE_AUTHOR("Stephen Warren <swarren@nvidia.com>"); +MODULE_DESCRIPTION("Tegra+WM8753 machine ASoC driver"); +MODULE_LICENSE("GPL"); +MODULE_ALIAS("platform:" DRV_NAME); +MODULE_DEVICE_TABLE(of, tegra_wm8753_of_match); diff --git a/sound/soc/tegra/tegra_wm8903.c b/sound/soc/tegra/tegra_wm8903.c index 566655e23b7..0b0df49d9d3 100644 --- a/sound/soc/tegra/tegra_wm8903.c +++ b/sound/soc/tegra/tegra_wm8903.c @@ -2,7 +2,7 @@ * tegra_wm8903.c - Tegra machine ASoC driver for boards using WM8903 codec. * * Author: Stephen Warren <swarren@nvidia.com> - * Copyright (C) 2010-2011 - NVIDIA, Inc. + * Copyright (C) 2010-2012 - NVIDIA, Inc. * * Based on code copyright/by: * @@ -46,9 +46,6 @@ #include "../codecs/wm8903.h" -#include "tegra_das.h" -#include "tegra_i2s.h" -#include "tegra_pcm.h" #include "tegra_asoc_utils.h" #define DRV_NAME "tegra-snd-wm8903" @@ -61,7 +58,6 @@ struct tegra_wm8903 { struct tegra_wm8903_platform_data pdata; - struct platform_device *pcm_dev; struct tegra_asoc_utils_data util_data; int gpio_requested; }; @@ -354,8 +350,8 @@ static struct snd_soc_dai_link tegra_wm8903_dai = { .name = "WM8903", .stream_name = "WM8903 PCM", .codec_name = "wm8903.0-001a", - .platform_name = "tegra-pcm-audio", - .cpu_dai_name = "tegra-i2s.0", + .platform_name = "tegra20-i2s.0", + .cpu_dai_name = "tegra20-i2s.0", .codec_dai_name = "wm8903-hifi", .init = tegra_wm8903_init, .ops = &tegra_wm8903_ops, @@ -392,7 +388,6 @@ static __devinit int tegra_wm8903_driver_probe(struct platform_device *pdev) ret = -ENOMEM; goto err; } - machine->pcm_dev = ERR_PTR(-EINVAL); card->dev = &pdev->dev; platform_set_drvdata(pdev, card); @@ -428,14 +423,9 @@ static __devinit int tegra_wm8903_driver_probe(struct platform_device *pdev) goto err; } - machine->pcm_dev = platform_device_register_simple( - "tegra-pcm-audio", -1, NULL, 0); - if (IS_ERR(machine->pcm_dev)) { - dev_err(&pdev->dev, - "Can't instantiate tegra-pcm-audio\n"); - ret = PTR_ERR(machine->pcm_dev); - goto err; - } + tegra_wm8903_dai.platform_name = NULL; + tegra_wm8903_dai.platform_of_node = + tegra_wm8903_dai.cpu_dai_of_node; } else { if (machine_is_harmony()) { card->dapm_routes = harmony_audio_map; @@ -454,7 +444,7 @@ static __devinit int tegra_wm8903_driver_probe(struct platform_device *pdev) ret = tegra_asoc_utils_init(&machine->util_data, &pdev->dev); if (ret) - goto err_unregister; + goto err; ret = snd_soc_register_card(card); if (ret) { @@ -467,9 +457,6 @@ static __devinit int tegra_wm8903_driver_probe(struct platform_device *pdev) err_fini_utils: tegra_asoc_utils_fini(&machine->util_data); -err_unregister: - if (!IS_ERR(machine->pcm_dev)) - platform_device_unregister(machine->pcm_dev); err: return ret; } @@ -497,8 +484,6 @@ static int __devexit tegra_wm8903_driver_remove(struct platform_device *pdev) snd_soc_unregister_card(card); tegra_asoc_utils_fini(&machine->util_data); - if (!IS_ERR(machine->pcm_dev)) - platform_device_unregister(machine->pcm_dev); return 0; } diff --git a/sound/soc/tegra/trimslice.c b/sound/soc/tegra/trimslice.c index 2bdfc550cff..4a8d5b672c9 100644 --- a/sound/soc/tegra/trimslice.c +++ b/sound/soc/tegra/trimslice.c @@ -27,6 +27,7 @@ #include <asm/mach-types.h> #include <linux/module.h> +#include <linux/of.h> #include <linux/platform_device.h> #include <linux/slab.h> @@ -38,9 +39,6 @@ #include "../codecs/tlv320aic23.h" -#include "tegra_das.h" -#include "tegra_i2s.h" -#include "tegra_pcm.h" #include "tegra_asoc_utils.h" #define DRV_NAME "tegra-snd-trimslice" @@ -119,8 +117,8 @@ static struct snd_soc_dai_link trimslice_tlv320aic23_dai = { .name = "TLV320AIC23", .stream_name = "AIC23", .codec_name = "tlv320aic23-codec.2-001a", - .platform_name = "tegra-pcm-audio", - .cpu_dai_name = "tegra-i2s.0", + .platform_name = "tegra20-i2s.0", + .cpu_dai_name = "tegra20-i2s.0", .codec_dai_name = "tlv320aic23-hifi", .ops = &trimslice_asoc_ops, }; @@ -152,6 +150,32 @@ static __devinit int tegra_snd_trimslice_probe(struct platform_device *pdev) goto err; } + if (pdev->dev.of_node) { + trimslice_tlv320aic23_dai.codec_name = NULL; + trimslice_tlv320aic23_dai.codec_of_node = of_parse_phandle( + pdev->dev.of_node, "nvidia,audio-codec", 0); + if (!trimslice_tlv320aic23_dai.codec_of_node) { + dev_err(&pdev->dev, + "Property 'nvidia,audio-codec' missing or invalid\n"); + ret = -EINVAL; + goto err; + } + + trimslice_tlv320aic23_dai.cpu_dai_name = NULL; + trimslice_tlv320aic23_dai.cpu_dai_of_node = of_parse_phandle( + pdev->dev.of_node, "nvidia,i2s-controller", 0); + if (!trimslice_tlv320aic23_dai.cpu_dai_of_node) { + dev_err(&pdev->dev, + "Property 'nvidia,i2s-controller' missing or invalid\n"); + ret = -EINVAL; + goto err; + } + + trimslice_tlv320aic23_dai.platform_name = NULL; + trimslice_tlv320aic23_dai.platform_of_node = + trimslice_tlv320aic23_dai.cpu_dai_of_node; + } + ret = tegra_asoc_utils_init(&trimslice->util_data, &pdev->dev); if (ret) goto err; @@ -187,10 +211,17 @@ static int __devexit tegra_snd_trimslice_remove(struct platform_device *pdev) return 0; } +static const struct of_device_id trimslice_of_match[] __devinitconst = { + { .compatible = "nvidia,tegra-audio-trimslice", }, + {}, +}; +MODULE_DEVICE_TABLE(of, trimslice_of_match); + static struct platform_driver tegra_snd_trimslice_driver = { .driver = { .name = DRV_NAME, .owner = THIS_MODULE, + .of_match_table = trimslice_of_match, }, .probe = tegra_snd_trimslice_probe, .remove = __devexit_p(tegra_snd_trimslice_remove), diff --git a/sound/soc/ux500/Kconfig b/sound/soc/ux500/Kconfig new file mode 100644 index 00000000000..44cf43404cd --- /dev/null +++ b/sound/soc/ux500/Kconfig @@ -0,0 +1,14 @@ +# +# Ux500 SoC audio configuration +# +menuconfig SND_SOC_UX500 + tristate "SoC Audio support for Ux500 platform" + depends on SND_SOC + depends on MFD_DB8500_PRCMU + help + Say Y if you want to enable ASoC-support for + any of the Ux500 platforms (e.g. U8500). + +config SND_SOC_UX500_PLAT_MSP_I2S + tristate + depends on SND_SOC_UX500 diff --git a/sound/soc/ux500/Makefile b/sound/soc/ux500/Makefile new file mode 100644 index 00000000000..19974c5a2ea --- /dev/null +++ b/sound/soc/ux500/Makefile @@ -0,0 +1,4 @@ +# Ux500 Platform Support + +snd-soc-ux500-plat-msp-i2s-objs := ux500_msp_dai.o ux500_msp_i2s.o +obj-$(CONFIG_SND_SOC_UX500_PLAT_MSP_I2S) += snd-soc-ux500-plat-msp-i2s.o diff --git a/sound/soc/ux500/ux500_msp_dai.c b/sound/soc/ux500/ux500_msp_dai.c new file mode 100644 index 00000000000..93c6c40e724 --- /dev/null +++ b/sound/soc/ux500/ux500_msp_dai.c @@ -0,0 +1,843 @@ +/* + * Copyright (C) ST-Ericsson SA 2012 + * + * Author: Ola Lilja <ola.o.lilja@stericsson.com>, + * Roger Nilsson <roger.xr.nilsson@stericsson.com> + * for ST-Ericsson. + * + * License terms: + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License version 2 as published + * by the Free Software Foundation. + */ + +#include <linux/module.h> +#include <linux/slab.h> +#include <linux/bitops.h> +#include <linux/platform_device.h> +#include <linux/clk.h> +#include <linux/regulator/consumer.h> +#include <linux/mfd/dbx500-prcmu.h> + +#include <mach/hardware.h> +#include <mach/board-mop500-msp.h> + +#include <sound/soc.h> +#include <sound/soc-dai.h> + +#include "ux500_msp_i2s.h" +#include "ux500_msp_dai.h" + +static int setup_pcm_multichan(struct snd_soc_dai *dai, + struct ux500_msp_config *msp_config) +{ + struct ux500_msp_i2s_drvdata *drvdata = dev_get_drvdata(dai->dev); + struct msp_multichannel_config *multi = + &msp_config->multichannel_config; + + if (drvdata->slots > 1) { + msp_config->multichannel_configured = 1; + + multi->tx_multichannel_enable = true; + multi->rx_multichannel_enable = true; + multi->rx_comparison_enable_mode = MSP_COMPARISON_DISABLED; + + multi->tx_channel_0_enable = drvdata->tx_mask; + multi->tx_channel_1_enable = 0; + multi->tx_channel_2_enable = 0; + multi->tx_channel_3_enable = 0; + + multi->rx_channel_0_enable = drvdata->rx_mask; + multi->rx_channel_1_enable = 0; + multi->rx_channel_2_enable = 0; + multi->rx_channel_3_enable = 0; + + dev_dbg(dai->dev, + "%s: Multichannel enabled. Slots: %d, TX: %u, RX: %u\n", + __func__, drvdata->slots, multi->tx_channel_0_enable, + multi->rx_channel_0_enable); + } + + return 0; +} + +static int setup_frameper(struct snd_soc_dai *dai, unsigned int rate, + struct msp_protdesc *prot_desc) +{ + struct ux500_msp_i2s_drvdata *drvdata = dev_get_drvdata(dai->dev); + + switch (drvdata->slots) { + case 1: + switch (rate) { + case 8000: + prot_desc->frame_period = + FRAME_PER_SINGLE_SLOT_8_KHZ; + break; + + case 16000: + prot_desc->frame_period = + FRAME_PER_SINGLE_SLOT_16_KHZ; + break; + + case 44100: + prot_desc->frame_period = + FRAME_PER_SINGLE_SLOT_44_1_KHZ; + break; + + case 48000: + prot_desc->frame_period = + FRAME_PER_SINGLE_SLOT_48_KHZ; + break; + + default: + dev_err(dai->dev, + "%s: Error: Unsupported sample-rate (freq = %d)!\n", + __func__, rate); + return -EINVAL; + } + break; + + case 2: + prot_desc->frame_period = FRAME_PER_2_SLOTS; + break; + + case 8: + prot_desc->frame_period = FRAME_PER_8_SLOTS; + break; + + case 16: + prot_desc->frame_period = FRAME_PER_16_SLOTS; + break; + default: + dev_err(dai->dev, + "%s: Error: Unsupported slot-count (slots = %d)!\n", + __func__, drvdata->slots); + return -EINVAL; + } + + prot_desc->clocks_per_frame = + prot_desc->frame_period+1; + + dev_dbg(dai->dev, "%s: Clocks per frame: %u\n", + __func__, + prot_desc->clocks_per_frame); + + return 0; +} + +static int setup_pcm_framing(struct snd_soc_dai *dai, unsigned int rate, + struct msp_protdesc *prot_desc) +{ + struct ux500_msp_i2s_drvdata *drvdata = dev_get_drvdata(dai->dev); + + u32 frame_length = MSP_FRAME_LEN_1; + prot_desc->frame_width = 0; + + switch (drvdata->slots) { + case 1: + frame_length = MSP_FRAME_LEN_1; + break; + + case 2: + frame_length = MSP_FRAME_LEN_2; + break; + + case 8: + frame_length = MSP_FRAME_LEN_8; + break; + + case 16: + frame_length = MSP_FRAME_LEN_16; + break; + default: + dev_err(dai->dev, + "%s: Error: Unsupported slot-count (slots = %d)!\n", + __func__, drvdata->slots); + return -EINVAL; + } + + prot_desc->tx_frame_len_1 = frame_length; + prot_desc->rx_frame_len_1 = frame_length; + prot_desc->tx_frame_len_2 = frame_length; + prot_desc->rx_frame_len_2 = frame_length; + + prot_desc->tx_elem_len_1 = MSP_ELEM_LEN_16; + prot_desc->rx_elem_len_1 = MSP_ELEM_LEN_16; + prot_desc->tx_elem_len_2 = MSP_ELEM_LEN_16; + prot_desc->rx_elem_len_2 = MSP_ELEM_LEN_16; + + return setup_frameper(dai, rate, prot_desc); +} + +static int setup_clocking(struct snd_soc_dai *dai, + unsigned int fmt, + struct ux500_msp_config *msp_config) +{ + switch (fmt & SND_SOC_DAIFMT_INV_MASK) { + case SND_SOC_DAIFMT_NB_NF: + break; + + case SND_SOC_DAIFMT_NB_IF: + msp_config->tx_fsync_pol ^= 1 << TFSPOL_SHIFT; + msp_config->rx_fsync_pol ^= 1 << RFSPOL_SHIFT; + + break; + + default: + dev_err(dai->dev, + "%s: Error: Unsopported inversion (fmt = 0x%x)!\n", + __func__, fmt); + + return -EINVAL; + } + + switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) { + case SND_SOC_DAIFMT_CBM_CFM: + dev_dbg(dai->dev, "%s: Codec is master.\n", __func__); + + msp_config->iodelay = 0x20; + msp_config->rx_fsync_sel = 0; + msp_config->tx_fsync_sel = 1 << TFSSEL_SHIFT; + msp_config->tx_clk_sel = 0; + msp_config->rx_clk_sel = 0; + msp_config->srg_clk_sel = 0x2 << SCKSEL_SHIFT; + + break; + + case SND_SOC_DAIFMT_CBS_CFS: + dev_dbg(dai->dev, "%s: Codec is slave.\n", __func__); + + msp_config->tx_clk_sel = TX_CLK_SEL_SRG; + msp_config->tx_fsync_sel = TX_SYNC_SRG_PROG; + msp_config->rx_clk_sel = RX_CLK_SEL_SRG; + msp_config->rx_fsync_sel = RX_SYNC_SRG; + msp_config->srg_clk_sel = 1 << SCKSEL_SHIFT; + + break; + + default: + dev_err(dai->dev, "%s: Error: Unsopported master (fmt = 0x%x)!\n", + __func__, fmt); + + return -EINVAL; + } + + return 0; +} + +static int setup_pcm_protdesc(struct snd_soc_dai *dai, + unsigned int fmt, + struct msp_protdesc *prot_desc) +{ + prot_desc->rx_phase_mode = MSP_SINGLE_PHASE; + prot_desc->tx_phase_mode = MSP_SINGLE_PHASE; + prot_desc->rx_phase2_start_mode = MSP_PHASE2_START_MODE_IMEDIATE; + prot_desc->tx_phase2_start_mode = MSP_PHASE2_START_MODE_IMEDIATE; + prot_desc->rx_byte_order = MSP_BTF_MS_BIT_FIRST; + prot_desc->tx_byte_order = MSP_BTF_MS_BIT_FIRST; + prot_desc->tx_fsync_pol = MSP_FSYNC_POL(MSP_FSYNC_POL_ACT_HI); + prot_desc->rx_fsync_pol = MSP_FSYNC_POL_ACT_HI << RFSPOL_SHIFT; + + if ((fmt & SND_SOC_DAIFMT_FORMAT_MASK) == SND_SOC_DAIFMT_DSP_A) { + dev_dbg(dai->dev, "%s: DSP_A.\n", __func__); + prot_desc->rx_clk_pol = MSP_RISING_EDGE; + prot_desc->tx_clk_pol = MSP_FALLING_EDGE; + + prot_desc->rx_data_delay = MSP_DELAY_1; + prot_desc->tx_data_delay = MSP_DELAY_1; + } else { + dev_dbg(dai->dev, "%s: DSP_B.\n", __func__); + prot_desc->rx_clk_pol = MSP_FALLING_EDGE; + prot_desc->tx_clk_pol = MSP_RISING_EDGE; + + prot_desc->rx_data_delay = MSP_DELAY_0; + prot_desc->tx_data_delay = MSP_DELAY_0; + } + + prot_desc->rx_half_word_swap = MSP_SWAP_NONE; + prot_desc->tx_half_word_swap = MSP_SWAP_NONE; + prot_desc->compression_mode = MSP_COMPRESS_MODE_LINEAR; + prot_desc->expansion_mode = MSP_EXPAND_MODE_LINEAR; + prot_desc->frame_sync_ignore = MSP_FSYNC_IGNORE; + + return 0; +} + +static int setup_i2s_protdesc(struct msp_protdesc *prot_desc) +{ + prot_desc->rx_phase_mode = MSP_DUAL_PHASE; + prot_desc->tx_phase_mode = MSP_DUAL_PHASE; + prot_desc->rx_phase2_start_mode = MSP_PHASE2_START_MODE_FSYNC; + prot_desc->tx_phase2_start_mode = MSP_PHASE2_START_MODE_FSYNC; + prot_desc->rx_byte_order = MSP_BTF_MS_BIT_FIRST; + prot_desc->tx_byte_order = MSP_BTF_MS_BIT_FIRST; + prot_desc->tx_fsync_pol = MSP_FSYNC_POL(MSP_FSYNC_POL_ACT_LO); + prot_desc->rx_fsync_pol = MSP_FSYNC_POL_ACT_LO << RFSPOL_SHIFT; + + prot_desc->rx_frame_len_1 = MSP_FRAME_LEN_1; + prot_desc->rx_frame_len_2 = MSP_FRAME_LEN_1; + prot_desc->tx_frame_len_1 = MSP_FRAME_LEN_1; + prot_desc->tx_frame_len_2 = MSP_FRAME_LEN_1; + prot_desc->rx_elem_len_1 = MSP_ELEM_LEN_16; + prot_desc->rx_elem_len_2 = MSP_ELEM_LEN_16; + prot_desc->tx_elem_len_1 = MSP_ELEM_LEN_16; + prot_desc->tx_elem_len_2 = MSP_ELEM_LEN_16; + + prot_desc->rx_clk_pol = MSP_RISING_EDGE; + prot_desc->tx_clk_pol = MSP_FALLING_EDGE; + + prot_desc->rx_data_delay = MSP_DELAY_0; + prot_desc->tx_data_delay = MSP_DELAY_0; + + prot_desc->tx_half_word_swap = MSP_SWAP_NONE; + prot_desc->rx_half_word_swap = MSP_SWAP_NONE; + prot_desc->compression_mode = MSP_COMPRESS_MODE_LINEAR; + prot_desc->expansion_mode = MSP_EXPAND_MODE_LINEAR; + prot_desc->frame_sync_ignore = MSP_FSYNC_IGNORE; + + return 0; +} + +static int setup_msp_config(struct snd_pcm_substream *substream, + struct snd_soc_dai *dai, + struct ux500_msp_config *msp_config) +{ + struct ux500_msp_i2s_drvdata *drvdata = dev_get_drvdata(dai->dev); + struct msp_protdesc *prot_desc = &msp_config->protdesc; + struct snd_pcm_runtime *runtime = substream->runtime; + unsigned int fmt = drvdata->fmt; + int ret; + + memset(msp_config, 0, sizeof(*msp_config)); + + msp_config->f_inputclk = drvdata->master_clk; + + msp_config->tx_fifo_config = TX_FIFO_ENABLE; + msp_config->rx_fifo_config = RX_FIFO_ENABLE; + msp_config->def_elem_len = 1; + msp_config->direction = substream->stream == SNDRV_PCM_STREAM_PLAYBACK ? + MSP_DIR_TX : MSP_DIR_RX; + msp_config->data_size = MSP_DATA_BITS_32; + msp_config->frame_freq = runtime->rate; + + dev_dbg(dai->dev, "%s: f_inputclk = %u, frame_freq = %u.\n", + __func__, msp_config->f_inputclk, msp_config->frame_freq); + /* To avoid division by zero */ + prot_desc->clocks_per_frame = 1; + + dev_dbg(dai->dev, "%s: rate: %u, channels: %d.\n", __func__, + runtime->rate, runtime->channels); + switch (fmt & + (SND_SOC_DAIFMT_FORMAT_MASK | SND_SOC_DAIFMT_MASTER_MASK)) { + case SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_CBS_CFS: + dev_dbg(dai->dev, "%s: SND_SOC_DAIFMT_I2S.\n", __func__); + + msp_config->default_protdesc = 1; + msp_config->protocol = MSP_I2S_PROTOCOL; + break; + + case SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_CBM_CFM: + dev_dbg(dai->dev, "%s: SND_SOC_DAIFMT_I2S.\n", __func__); + + msp_config->data_size = MSP_DATA_BITS_16; + msp_config->protocol = MSP_I2S_PROTOCOL; + + ret = setup_i2s_protdesc(prot_desc); + if (ret < 0) + return ret; + + break; + + case SND_SOC_DAIFMT_DSP_A | SND_SOC_DAIFMT_CBS_CFS: + case SND_SOC_DAIFMT_DSP_A | SND_SOC_DAIFMT_CBM_CFM: + case SND_SOC_DAIFMT_DSP_B | SND_SOC_DAIFMT_CBS_CFS: + case SND_SOC_DAIFMT_DSP_B | SND_SOC_DAIFMT_CBM_CFM: + dev_dbg(dai->dev, "%s: PCM format.\n", __func__); + + msp_config->data_size = MSP_DATA_BITS_16; + msp_config->protocol = MSP_PCM_PROTOCOL; + + ret = setup_pcm_protdesc(dai, fmt, prot_desc); + if (ret < 0) + return ret; + + ret = setup_pcm_multichan(dai, msp_config); + if (ret < 0) + return ret; + + ret = setup_pcm_framing(dai, runtime->rate, prot_desc); + if (ret < 0) + return ret; + + break; + + default: + dev_err(dai->dev, "%s: Error: Unsopported format (%d)!\n", + __func__, fmt); + return -EINVAL; + } + + return setup_clocking(dai, fmt, msp_config); +} + +static int ux500_msp_dai_startup(struct snd_pcm_substream *substream, + struct snd_soc_dai *dai) +{ + int ret = 0; + struct ux500_msp_i2s_drvdata *drvdata = dev_get_drvdata(dai->dev); + + dev_dbg(dai->dev, "%s: MSP %d (%s): Enter.\n", __func__, dai->id, + snd_pcm_stream_str(substream)); + + /* Enable regulator */ + ret = regulator_enable(drvdata->reg_vape); + if (ret != 0) { + dev_err(drvdata->msp->dev, + "%s: Failed to enable regulator!\n", __func__); + return ret; + } + + /* Enable clock */ + dev_dbg(dai->dev, "%s: Enabling MSP-clock.\n", __func__); + clk_enable(drvdata->clk); + + return 0; +} + +static void ux500_msp_dai_shutdown(struct snd_pcm_substream *substream, + struct snd_soc_dai *dai) +{ + int ret; + struct ux500_msp_i2s_drvdata *drvdata = dev_get_drvdata(dai->dev); + bool is_playback = (substream->stream == SNDRV_PCM_STREAM_PLAYBACK); + + dev_dbg(dai->dev, "%s: MSP %d (%s): Enter.\n", __func__, dai->id, + snd_pcm_stream_str(substream)); + + if (drvdata->vape_opp_constraint == 1) { + prcmu_qos_update_requirement(PRCMU_QOS_APE_OPP, + "ux500_msp_i2s", 50); + drvdata->vape_opp_constraint = 0; + } + + if (ux500_msp_i2s_close(drvdata->msp, + is_playback ? MSP_DIR_TX : MSP_DIR_RX)) { + dev_err(dai->dev, + "%s: Error: MSP %d (%s): Unable to close i2s.\n", + __func__, dai->id, snd_pcm_stream_str(substream)); + } + + /* Disable clock */ + clk_disable(drvdata->clk); + + /* Disable regulator */ + ret = regulator_disable(drvdata->reg_vape); + if (ret < 0) + dev_err(dai->dev, + "%s: ERROR: Failed to disable regulator (%d)!\n", + __func__, ret); +} + +static int ux500_msp_dai_prepare(struct snd_pcm_substream *substream, + struct snd_soc_dai *dai) +{ + int ret = 0; + struct ux500_msp_i2s_drvdata *drvdata = dev_get_drvdata(dai->dev); + struct snd_pcm_runtime *runtime = substream->runtime; + struct ux500_msp_config msp_config; + + dev_dbg(dai->dev, "%s: MSP %d (%s): Enter (rate = %d).\n", __func__, + dai->id, snd_pcm_stream_str(substream), runtime->rate); + + setup_msp_config(substream, dai, &msp_config); + + ret = ux500_msp_i2s_open(drvdata->msp, &msp_config); + if (ret < 0) { + dev_err(dai->dev, "%s: Error: msp_setup failed (ret = %d)!\n", + __func__, ret); + return ret; + } + + /* Set OPP-level */ + if ((drvdata->fmt & SND_SOC_DAIFMT_MASTER_MASK) && + (drvdata->msp->f_bitclk > 19200000)) { + /* If the bit-clock is higher than 19.2MHz, Vape should be + * run in 100% OPP. Only when bit-clock is used (MSP master) */ + prcmu_qos_update_requirement(PRCMU_QOS_APE_OPP, + "ux500-msp-i2s", 100); + drvdata->vape_opp_constraint = 1; + } else { + prcmu_qos_update_requirement(PRCMU_QOS_APE_OPP, + "ux500-msp-i2s", 50); + drvdata->vape_opp_constraint = 0; + } + + return ret; +} + +static int ux500_msp_dai_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params, + struct snd_soc_dai *dai) +{ + unsigned int mask, slots_active; + struct snd_pcm_runtime *runtime = substream->runtime; + struct ux500_msp_i2s_drvdata *drvdata = dev_get_drvdata(dai->dev); + + dev_dbg(dai->dev, "%s: MSP %d (%s): Enter.\n", + __func__, dai->id, snd_pcm_stream_str(substream)); + + switch (drvdata->fmt & SND_SOC_DAIFMT_FORMAT_MASK) { + case SND_SOC_DAIFMT_I2S: + snd_pcm_hw_constraint_minmax(runtime, + SNDRV_PCM_HW_PARAM_CHANNELS, + 1, 2); + break; + + case SND_SOC_DAIFMT_DSP_B: + case SND_SOC_DAIFMT_DSP_A: + mask = substream->stream == SNDRV_PCM_STREAM_PLAYBACK ? + drvdata->tx_mask : + drvdata->rx_mask; + + slots_active = hweight32(mask); + dev_dbg(dai->dev, "TDM-slots active: %d", slots_active); + + snd_pcm_hw_constraint_minmax(runtime, + SNDRV_PCM_HW_PARAM_CHANNELS, + slots_active, slots_active); + break; + + default: + dev_err(dai->dev, + "%s: Error: Unsupported protocol (fmt = 0x%x)!\n", + __func__, drvdata->fmt); + return -EINVAL; + } + + return 0; +} + +static int ux500_msp_dai_set_dai_fmt(struct snd_soc_dai *dai, + unsigned int fmt) +{ + struct ux500_msp_i2s_drvdata *drvdata = dev_get_drvdata(dai->dev); + + dev_dbg(dai->dev, "%s: MSP %d: Enter.\n", __func__, dai->id); + + switch (fmt & (SND_SOC_DAIFMT_FORMAT_MASK | + SND_SOC_DAIFMT_MASTER_MASK)) { + case SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_CBS_CFS: + case SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_CBM_CFM: + case SND_SOC_DAIFMT_DSP_B | SND_SOC_DAIFMT_CBS_CFS: + case SND_SOC_DAIFMT_DSP_B | SND_SOC_DAIFMT_CBM_CFM: + case SND_SOC_DAIFMT_DSP_A | SND_SOC_DAIFMT_CBS_CFS: + case SND_SOC_DAIFMT_DSP_A | SND_SOC_DAIFMT_CBM_CFM: + break; + + default: + dev_err(dai->dev, + "%s: Error: Unsupported protocol/master (fmt = 0x%x)!\n", + __func__, drvdata->fmt); + return -EINVAL; + } + + switch (fmt & SND_SOC_DAIFMT_INV_MASK) { + case SND_SOC_DAIFMT_NB_NF: + case SND_SOC_DAIFMT_NB_IF: + case SND_SOC_DAIFMT_IB_IF: + break; + + default: + dev_err(dai->dev, + "%s: Error: Unsupported inversion (fmt = 0x%x)!\n", + __func__, drvdata->fmt); + return -EINVAL; + } + + drvdata->fmt = fmt; + return 0; +} + +static int ux500_msp_dai_set_tdm_slot(struct snd_soc_dai *dai, + unsigned int tx_mask, + unsigned int rx_mask, + int slots, int slot_width) +{ + struct ux500_msp_i2s_drvdata *drvdata = dev_get_drvdata(dai->dev); + unsigned int cap; + + switch (slots) { + case 1: + cap = 0x01; + break; + case 2: + cap = 0x03; + break; + case 8: + cap = 0xFF; + break; + case 16: + cap = 0xFFFF; + break; + default: + dev_err(dai->dev, "%s: Error: Unsupported slot-count (%d)!\n", + __func__, slots); + return -EINVAL; + } + drvdata->slots = slots; + + if (!(slot_width == 16)) { + dev_err(dai->dev, "%s: Error: Unsupported slot-width (%d)!\n", + __func__, slot_width); + return -EINVAL; + } + drvdata->slot_width = slot_width; + + drvdata->tx_mask = tx_mask & cap; + drvdata->rx_mask = rx_mask & cap; + + return 0; +} + +static int ux500_msp_dai_set_dai_sysclk(struct snd_soc_dai *dai, + int clk_id, unsigned int freq, int dir) +{ + struct ux500_msp_i2s_drvdata *drvdata = dev_get_drvdata(dai->dev); + + dev_dbg(dai->dev, "%s: MSP %d: Enter. clk-id: %d, freq: %u.\n", + __func__, dai->id, clk_id, freq); + + switch (clk_id) { + case UX500_MSP_MASTER_CLOCK: + drvdata->master_clk = freq; + break; + + default: + dev_err(dai->dev, "%s: MSP %d: Invalid clk-id (%d)!\n", + __func__, dai->id, clk_id); + return -EINVAL; + } + + return 0; +} + +static int ux500_msp_dai_trigger(struct snd_pcm_substream *substream, + int cmd, struct snd_soc_dai *dai) +{ + int ret = 0; + struct ux500_msp_i2s_drvdata *drvdata = dev_get_drvdata(dai->dev); + + dev_dbg(dai->dev, "%s: MSP %d (%s): Enter (msp->id = %d, cmd = %d).\n", + __func__, dai->id, snd_pcm_stream_str(substream), + (int)drvdata->msp->id, cmd); + + ret = ux500_msp_i2s_trigger(drvdata->msp, cmd, substream->stream); + + return ret; +} + +static int ux500_msp_dai_probe(struct snd_soc_dai *dai) +{ + struct ux500_msp_i2s_drvdata *drvdata = dev_get_drvdata(dai->dev); + + drvdata->playback_dma_data.dma_cfg = drvdata->msp->dma_cfg_tx; + drvdata->capture_dma_data.dma_cfg = drvdata->msp->dma_cfg_rx; + + dai->playback_dma_data = &drvdata->playback_dma_data; + dai->capture_dma_data = &drvdata->capture_dma_data; + + drvdata->playback_dma_data.data_size = drvdata->slot_width; + drvdata->capture_dma_data.data_size = drvdata->slot_width; + + return 0; +} + +static struct snd_soc_dai_ops ux500_msp_dai_ops[] = { + { + .set_sysclk = ux500_msp_dai_set_dai_sysclk, + .set_fmt = ux500_msp_dai_set_dai_fmt, + .set_tdm_slot = ux500_msp_dai_set_tdm_slot, + .startup = ux500_msp_dai_startup, + .shutdown = ux500_msp_dai_shutdown, + .prepare = ux500_msp_dai_prepare, + .trigger = ux500_msp_dai_trigger, + .hw_params = ux500_msp_dai_hw_params, + } +}; + +static struct snd_soc_dai_driver ux500_msp_dai_drv[UX500_NBR_OF_DAI] = { + { + .name = "ux500-msp-i2s.0", + .probe = ux500_msp_dai_probe, + .id = 0, + .suspend = NULL, + .resume = NULL, + .playback = { + .channels_min = UX500_MSP_MIN_CHANNELS, + .channels_max = UX500_MSP_MAX_CHANNELS, + .rates = UX500_I2S_RATES, + .formats = UX500_I2S_FORMATS, + }, + .capture = { + .channels_min = UX500_MSP_MIN_CHANNELS, + .channels_max = UX500_MSP_MAX_CHANNELS, + .rates = UX500_I2S_RATES, + .formats = UX500_I2S_FORMATS, + }, + .ops = ux500_msp_dai_ops, + }, + { + .name = "ux500-msp-i2s.1", + .probe = ux500_msp_dai_probe, + .id = 1, + .suspend = NULL, + .resume = NULL, + .playback = { + .channels_min = UX500_MSP_MIN_CHANNELS, + .channels_max = UX500_MSP_MAX_CHANNELS, + .rates = UX500_I2S_RATES, + .formats = UX500_I2S_FORMATS, + }, + .capture = { + .channels_min = UX500_MSP_MIN_CHANNELS, + .channels_max = UX500_MSP_MAX_CHANNELS, + .rates = UX500_I2S_RATES, + .formats = UX500_I2S_FORMATS, + }, + .ops = ux500_msp_dai_ops, + }, + { + .name = "ux500-msp-i2s.2", + .id = 2, + .probe = ux500_msp_dai_probe, + .suspend = NULL, + .resume = NULL, + .playback = { + .channels_min = UX500_MSP_MIN_CHANNELS, + .channels_max = UX500_MSP_MAX_CHANNELS, + .rates = UX500_I2S_RATES, + .formats = UX500_I2S_FORMATS, + }, + .capture = { + .channels_min = UX500_MSP_MIN_CHANNELS, + .channels_max = UX500_MSP_MAX_CHANNELS, + .rates = UX500_I2S_RATES, + .formats = UX500_I2S_FORMATS, + }, + .ops = ux500_msp_dai_ops, + }, + { + .name = "ux500-msp-i2s.3", + .probe = ux500_msp_dai_probe, + .id = 3, + .suspend = NULL, + .resume = NULL, + .playback = { + .channels_min = UX500_MSP_MIN_CHANNELS, + .channels_max = UX500_MSP_MAX_CHANNELS, + .rates = UX500_I2S_RATES, + .formats = UX500_I2S_FORMATS, + }, + .capture = { + .channels_min = UX500_MSP_MIN_CHANNELS, + .channels_max = UX500_MSP_MAX_CHANNELS, + .rates = UX500_I2S_RATES, + .formats = UX500_I2S_FORMATS, + }, + .ops = ux500_msp_dai_ops, + }, +}; + +static int __devinit ux500_msp_drv_probe(struct platform_device *pdev) +{ + struct ux500_msp_i2s_drvdata *drvdata; + int ret = 0; + + dev_dbg(&pdev->dev, "%s: Enter (pdev->name = %s).\n", __func__, + pdev->name); + + drvdata = devm_kzalloc(&pdev->dev, + sizeof(struct ux500_msp_i2s_drvdata), + GFP_KERNEL); + drvdata->fmt = 0; + drvdata->slots = 1; + drvdata->tx_mask = 0x01; + drvdata->rx_mask = 0x01; + drvdata->slot_width = 16; + drvdata->master_clk = MSP_INPUT_FREQ_APB; + + drvdata->reg_vape = devm_regulator_get(&pdev->dev, "v-ape"); + if (IS_ERR(drvdata->reg_vape)) { + ret = (int)PTR_ERR(drvdata->reg_vape); + dev_err(&pdev->dev, + "%s: ERROR: Failed to get Vape supply (%d)!\n", + __func__, ret); + return ret; + } + prcmu_qos_add_requirement(PRCMU_QOS_APE_OPP, (char *)pdev->name, 50); + + drvdata->clk = clk_get(&pdev->dev, NULL); + if (IS_ERR(drvdata->clk)) { + ret = (int)PTR_ERR(drvdata->clk); + dev_err(&pdev->dev, "%s: ERROR: clk_get failed (%d)!\n", + __func__, ret); + goto err_clk; + } + + ret = ux500_msp_i2s_init_msp(pdev, &drvdata->msp, + pdev->dev.platform_data); + if (!drvdata->msp) { + dev_err(&pdev->dev, + "%s: ERROR: Failed to init MSP-struct (%d)!", + __func__, ret); + goto err_init_msp; + } + dev_set_drvdata(&pdev->dev, drvdata); + + ret = snd_soc_register_dai(&pdev->dev, + &ux500_msp_dai_drv[drvdata->msp->id]); + if (ret < 0) { + dev_err(&pdev->dev, "Error: %s: Failed to register MSP%d!\n", + __func__, drvdata->msp->id); + goto err_init_msp; + } + + return 0; + +err_init_msp: + clk_put(drvdata->clk); + +err_clk: + devm_regulator_put(drvdata->reg_vape); + + return ret; +} + +static int __devexit ux500_msp_drv_remove(struct platform_device *pdev) +{ + struct ux500_msp_i2s_drvdata *drvdata = dev_get_drvdata(&pdev->dev); + + snd_soc_unregister_dais(&pdev->dev, ARRAY_SIZE(ux500_msp_dai_drv)); + + devm_regulator_put(drvdata->reg_vape); + prcmu_qos_remove_requirement(PRCMU_QOS_APE_OPP, "ux500_msp_i2s"); + + clk_put(drvdata->clk); + + ux500_msp_i2s_cleanup_msp(pdev, drvdata->msp); + + return 0; +} + +static struct platform_driver msp_i2s_driver = { + .driver = { + .name = "ux500-msp-i2s", + .owner = THIS_MODULE, + }, + .probe = ux500_msp_drv_probe, + .remove = ux500_msp_drv_remove, +}; +module_platform_driver(msp_i2s_driver); + +MODULE_LICENSE("GPLv2"); diff --git a/sound/soc/ux500/ux500_msp_dai.h b/sound/soc/ux500/ux500_msp_dai.h new file mode 100644 index 00000000000..98202a34a5d --- /dev/null +++ b/sound/soc/ux500/ux500_msp_dai.h @@ -0,0 +1,79 @@ +/* + * Copyright (C) ST-Ericsson SA 2012 + * + * Author: Ola Lilja <ola.o.lilja@stericsson.com>, + * Roger Nilsson <roger.xr.nilsson@stericsson.com> + * for ST-Ericsson. + * + * License terms: + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License version 2 as published + * by the Free Software Foundation. + */ + +#ifndef UX500_msp_dai_H +#define UX500_msp_dai_H + +#include <linux/types.h> +#include <linux/spinlock.h> + +#include "ux500_msp_i2s.h" + +#define UX500_NBR_OF_DAI 4 + +#define UX500_I2S_RATES (SNDRV_PCM_RATE_8000 | SNDRV_PCM_RATE_16000 | \ + SNDRV_PCM_RATE_44100 | SNDRV_PCM_RATE_48000) + +#define UX500_I2S_FORMATS (SNDRV_PCM_FMTBIT_S16_LE) + +#define FRAME_PER_SINGLE_SLOT_8_KHZ 31 +#define FRAME_PER_SINGLE_SLOT_16_KHZ 124 +#define FRAME_PER_SINGLE_SLOT_44_1_KHZ 63 +#define FRAME_PER_SINGLE_SLOT_48_KHZ 49 +#define FRAME_PER_2_SLOTS 31 +#define FRAME_PER_8_SLOTS 138 +#define FRAME_PER_16_SLOTS 277 + +#ifndef CONFIG_SND_SOC_UX500_AB5500 +#define UX500_MSP_INTERNAL_CLOCK_FREQ 40000000 +#define UX500_MSP1_INTERNAL_CLOCK_FREQ UX500_MSP_INTERNAL_CLOCK_FREQ +#else +#define UX500_MSP_INTERNAL_CLOCK_FREQ 13000000 +#define UX500_MSP1_INTERNAL_CLOCK_FREQ (UX500_MSP_INTERNAL_CLOCK_FREQ * 2) +#endif + +#define UX500_MSP_MIN_CHANNELS 1 +#define UX500_MSP_MAX_CHANNELS 8 + +#define PLAYBACK_CONFIGURED 1 +#define CAPTURE_CONFIGURED 2 + +enum ux500_msp_clock_id { + UX500_MSP_MASTER_CLOCK, +}; + +struct ux500_msp_i2s_drvdata { + struct ux500_msp *msp; + struct regulator *reg_vape; + struct ux500_msp_dma_params playback_dma_data; + struct ux500_msp_dma_params capture_dma_data; + unsigned int fmt; + unsigned int tx_mask; + unsigned int rx_mask; + int slots; + int slot_width; + u8 configured; + int data_delay; + + /* Clocks */ + unsigned int master_clk; + struct clk *clk; + + /* Regulators */ + int vape_opp_constraint; +}; + +int ux500_msp_dai_set_data_delay(struct snd_soc_dai *dai, int delay); + +#endif diff --git a/sound/soc/ux500/ux500_msp_i2s.c b/sound/soc/ux500/ux500_msp_i2s.c new file mode 100644 index 00000000000..496dec10c96 --- /dev/null +++ b/sound/soc/ux500/ux500_msp_i2s.c @@ -0,0 +1,742 @@ +/* + * Copyright (C) ST-Ericsson SA 2012 + * + * Author: Ola Lilja <ola.o.lilja@stericsson.com>, + * Roger Nilsson <roger.xr.nilsson@stericsson.com>, + * Sandeep Kaushik <sandeep.kaushik@st.com> + * for ST-Ericsson. + * + * License terms: + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License version 2 as published + * by the Free Software Foundation. + */ + +#include <linux/module.h> +#include <linux/platform_device.h> +#include <linux/delay.h> +#include <linux/slab.h> + +#include <mach/hardware.h> +#include <mach/board-mop500-msp.h> + +#include <sound/soc.h> + +#include "ux500_msp_i2s.h" + + /* Protocol desciptors */ +static const struct msp_protdesc prot_descs[] = { + { /* I2S */ + MSP_SINGLE_PHASE, + MSP_SINGLE_PHASE, + MSP_PHASE2_START_MODE_IMEDIATE, + MSP_PHASE2_START_MODE_IMEDIATE, + MSP_BTF_MS_BIT_FIRST, + MSP_BTF_MS_BIT_FIRST, + MSP_FRAME_LEN_1, + MSP_FRAME_LEN_1, + MSP_FRAME_LEN_1, + MSP_FRAME_LEN_1, + MSP_ELEM_LEN_32, + MSP_ELEM_LEN_32, + MSP_ELEM_LEN_32, + MSP_ELEM_LEN_32, + MSP_DELAY_1, + MSP_DELAY_1, + MSP_RISING_EDGE, + MSP_FALLING_EDGE, + MSP_FSYNC_POL_ACT_LO, + MSP_FSYNC_POL_ACT_LO, + MSP_SWAP_NONE, + MSP_SWAP_NONE, + MSP_COMPRESS_MODE_LINEAR, + MSP_EXPAND_MODE_LINEAR, + MSP_FSYNC_IGNORE, + 31, + 15, + 32, + }, { /* PCM */ + MSP_DUAL_PHASE, + MSP_DUAL_PHASE, + MSP_PHASE2_START_MODE_FSYNC, + MSP_PHASE2_START_MODE_FSYNC, + MSP_BTF_MS_BIT_FIRST, + MSP_BTF_MS_BIT_FIRST, + MSP_FRAME_LEN_1, + MSP_FRAME_LEN_1, + MSP_FRAME_LEN_1, + MSP_FRAME_LEN_1, + MSP_ELEM_LEN_16, + MSP_ELEM_LEN_16, + MSP_ELEM_LEN_16, + MSP_ELEM_LEN_16, + MSP_DELAY_0, + MSP_DELAY_0, + MSP_RISING_EDGE, + MSP_FALLING_EDGE, + MSP_FSYNC_POL_ACT_HI, + MSP_FSYNC_POL_ACT_HI, + MSP_SWAP_NONE, + MSP_SWAP_NONE, + MSP_COMPRESS_MODE_LINEAR, + MSP_EXPAND_MODE_LINEAR, + MSP_FSYNC_IGNORE, + 255, + 0, + 256, + }, { /* Companded PCM */ + MSP_SINGLE_PHASE, + MSP_SINGLE_PHASE, + MSP_PHASE2_START_MODE_FSYNC, + MSP_PHASE2_START_MODE_FSYNC, + MSP_BTF_MS_BIT_FIRST, + MSP_BTF_MS_BIT_FIRST, + MSP_FRAME_LEN_1, + MSP_FRAME_LEN_1, + MSP_FRAME_LEN_1, + MSP_FRAME_LEN_1, + MSP_ELEM_LEN_8, + MSP_ELEM_LEN_8, + MSP_ELEM_LEN_8, + MSP_ELEM_LEN_8, + MSP_DELAY_0, + MSP_DELAY_0, + MSP_RISING_EDGE, + MSP_RISING_EDGE, + MSP_FSYNC_POL_ACT_HI, + MSP_FSYNC_POL_ACT_HI, + MSP_SWAP_NONE, + MSP_SWAP_NONE, + MSP_COMPRESS_MODE_LINEAR, + MSP_EXPAND_MODE_LINEAR, + MSP_FSYNC_IGNORE, + 255, + 0, + 256, + }, +}; + +static void set_prot_desc_tx(struct ux500_msp *msp, + struct msp_protdesc *protdesc, + enum msp_data_size data_size) +{ + u32 temp_reg = 0; + + temp_reg |= MSP_P2_ENABLE_BIT(protdesc->tx_phase_mode); + temp_reg |= MSP_P2_START_MODE_BIT(protdesc->tx_phase2_start_mode); + temp_reg |= MSP_P1_FRAME_LEN_BITS(protdesc->tx_frame_len_1); + temp_reg |= MSP_P2_FRAME_LEN_BITS(protdesc->tx_frame_len_2); + if (msp->def_elem_len) { + temp_reg |= MSP_P1_ELEM_LEN_BITS(protdesc->tx_elem_len_1); + temp_reg |= MSP_P2_ELEM_LEN_BITS(protdesc->tx_elem_len_2); + } else { + temp_reg |= MSP_P1_ELEM_LEN_BITS(data_size); + temp_reg |= MSP_P2_ELEM_LEN_BITS(data_size); + } + temp_reg |= MSP_DATA_DELAY_BITS(protdesc->tx_data_delay); + temp_reg |= MSP_SET_ENDIANNES_BIT(protdesc->tx_byte_order); + temp_reg |= MSP_FSYNC_POL(protdesc->tx_fsync_pol); + temp_reg |= MSP_DATA_WORD_SWAP(protdesc->tx_half_word_swap); + temp_reg |= MSP_SET_COMPANDING_MODE(protdesc->compression_mode); + temp_reg |= MSP_SET_FSYNC_IGNORE(protdesc->frame_sync_ignore); + + writel(temp_reg, msp->registers + MSP_TCF); +} + +static void set_prot_desc_rx(struct ux500_msp *msp, + struct msp_protdesc *protdesc, + enum msp_data_size data_size) +{ + u32 temp_reg = 0; + + temp_reg |= MSP_P2_ENABLE_BIT(protdesc->rx_phase_mode); + temp_reg |= MSP_P2_START_MODE_BIT(protdesc->rx_phase2_start_mode); + temp_reg |= MSP_P1_FRAME_LEN_BITS(protdesc->rx_frame_len_1); + temp_reg |= MSP_P2_FRAME_LEN_BITS(protdesc->rx_frame_len_2); + if (msp->def_elem_len) { + temp_reg |= MSP_P1_ELEM_LEN_BITS(protdesc->rx_elem_len_1); + temp_reg |= MSP_P2_ELEM_LEN_BITS(protdesc->rx_elem_len_2); + } else { + temp_reg |= MSP_P1_ELEM_LEN_BITS(data_size); + temp_reg |= MSP_P2_ELEM_LEN_BITS(data_size); + } + + temp_reg |= MSP_DATA_DELAY_BITS(protdesc->rx_data_delay); + temp_reg |= MSP_SET_ENDIANNES_BIT(protdesc->rx_byte_order); + temp_reg |= MSP_FSYNC_POL(protdesc->rx_fsync_pol); + temp_reg |= MSP_DATA_WORD_SWAP(protdesc->rx_half_word_swap); + temp_reg |= MSP_SET_COMPANDING_MODE(protdesc->expansion_mode); + temp_reg |= MSP_SET_FSYNC_IGNORE(protdesc->frame_sync_ignore); + + writel(temp_reg, msp->registers + MSP_RCF); +} + +static int configure_protocol(struct ux500_msp *msp, + struct ux500_msp_config *config) +{ + struct msp_protdesc *protdesc; + enum msp_data_size data_size; + u32 temp_reg = 0; + + data_size = config->data_size; + msp->def_elem_len = config->def_elem_len; + if (config->default_protdesc == 1) { + if (config->protocol >= MSP_INVALID_PROTOCOL) { + dev_err(msp->dev, "%s: ERROR: Invalid protocol!\n", + __func__); + return -EINVAL; + } + protdesc = + (struct msp_protdesc *)&prot_descs[config->protocol]; + } else { + protdesc = (struct msp_protdesc *)&config->protdesc; + } + + if (data_size < MSP_DATA_BITS_DEFAULT || data_size > MSP_DATA_BITS_32) { + dev_err(msp->dev, + "%s: ERROR: Invalid data-size requested (data_size = %d)!\n", + __func__, data_size); + return -EINVAL; + } + + if (config->direction & MSP_DIR_TX) + set_prot_desc_tx(msp, protdesc, data_size); + if (config->direction & MSP_DIR_RX) + set_prot_desc_rx(msp, protdesc, data_size); + + /* The code below should not be separated. */ + temp_reg = readl(msp->registers + MSP_GCR) & ~TX_CLK_POL_RISING; + temp_reg |= MSP_TX_CLKPOL_BIT(~protdesc->tx_clk_pol); + writel(temp_reg, msp->registers + MSP_GCR); + temp_reg = readl(msp->registers + MSP_GCR) & ~RX_CLK_POL_RISING; + temp_reg |= MSP_RX_CLKPOL_BIT(protdesc->rx_clk_pol); + writel(temp_reg, msp->registers + MSP_GCR); + + return 0; +} + +static int setup_bitclk(struct ux500_msp *msp, struct ux500_msp_config *config) +{ + u32 reg_val_GCR; + u32 frame_per = 0; + u32 sck_div = 0; + u32 frame_width = 0; + u32 temp_reg = 0; + struct msp_protdesc *protdesc = NULL; + + reg_val_GCR = readl(msp->registers + MSP_GCR); + writel(reg_val_GCR & ~SRG_ENABLE, msp->registers + MSP_GCR); + + if (config->default_protdesc) + protdesc = + (struct msp_protdesc *)&prot_descs[config->protocol]; + else + protdesc = (struct msp_protdesc *)&config->protdesc; + + switch (config->protocol) { + case MSP_PCM_PROTOCOL: + case MSP_PCM_COMPAND_PROTOCOL: + frame_width = protdesc->frame_width; + sck_div = config->f_inputclk / (config->frame_freq * + (protdesc->clocks_per_frame)); + frame_per = protdesc->frame_period; + break; + case MSP_I2S_PROTOCOL: + frame_width = protdesc->frame_width; + sck_div = config->f_inputclk / (config->frame_freq * + (protdesc->clocks_per_frame)); + frame_per = protdesc->frame_period; + break; + default: + dev_err(msp->dev, "%s: ERROR: Unknown protocol (%d)!\n", + __func__, + config->protocol); + return -EINVAL; + } + + temp_reg = (sck_div - 1) & SCK_DIV_MASK; + temp_reg |= FRAME_WIDTH_BITS(frame_width); + temp_reg |= FRAME_PERIOD_BITS(frame_per); + writel(temp_reg, msp->registers + MSP_SRG); + + msp->f_bitclk = (config->f_inputclk)/(sck_div + 1); + + /* Enable bit-clock */ + udelay(100); + reg_val_GCR = readl(msp->registers + MSP_GCR); + writel(reg_val_GCR | SRG_ENABLE, msp->registers + MSP_GCR); + udelay(100); + + return 0; +} + +static int configure_multichannel(struct ux500_msp *msp, + struct ux500_msp_config *config) +{ + struct msp_protdesc *protdesc; + struct msp_multichannel_config *mcfg; + u32 reg_val_MCR; + + if (config->default_protdesc == 1) { + if (config->protocol >= MSP_INVALID_PROTOCOL) { + dev_err(msp->dev, + "%s: ERROR: Invalid protocol (%d)!\n", + __func__, config->protocol); + return -EINVAL; + } + protdesc = (struct msp_protdesc *) + &prot_descs[config->protocol]; + } else { + protdesc = (struct msp_protdesc *)&config->protdesc; + } + + mcfg = &config->multichannel_config; + if (mcfg->tx_multichannel_enable) { + if (protdesc->tx_phase_mode == MSP_SINGLE_PHASE) { + reg_val_MCR = readl(msp->registers + MSP_MCR); + writel(reg_val_MCR | (mcfg->tx_multichannel_enable ? + 1 << TMCEN_BIT : 0), + msp->registers + MSP_MCR); + writel(mcfg->tx_channel_0_enable, + msp->registers + MSP_TCE0); + writel(mcfg->tx_channel_1_enable, + msp->registers + MSP_TCE1); + writel(mcfg->tx_channel_2_enable, + msp->registers + MSP_TCE2); + writel(mcfg->tx_channel_3_enable, + msp->registers + MSP_TCE3); + } else { + dev_err(msp->dev, + "%s: ERROR: Only single-phase supported (TX-mode: %d)!\n", + __func__, protdesc->tx_phase_mode); + return -EINVAL; + } + } + if (mcfg->rx_multichannel_enable) { + if (protdesc->rx_phase_mode == MSP_SINGLE_PHASE) { + reg_val_MCR = readl(msp->registers + MSP_MCR); + writel(reg_val_MCR | (mcfg->rx_multichannel_enable ? + 1 << RMCEN_BIT : 0), + msp->registers + MSP_MCR); + writel(mcfg->rx_channel_0_enable, + msp->registers + MSP_RCE0); + writel(mcfg->rx_channel_1_enable, + msp->registers + MSP_RCE1); + writel(mcfg->rx_channel_2_enable, + msp->registers + MSP_RCE2); + writel(mcfg->rx_channel_3_enable, + msp->registers + MSP_RCE3); + } else { + dev_err(msp->dev, + "%s: ERROR: Only single-phase supported (RX-mode: %d)!\n", + __func__, protdesc->rx_phase_mode); + return -EINVAL; + } + if (mcfg->rx_comparison_enable_mode) { + reg_val_MCR = readl(msp->registers + MSP_MCR); + writel(reg_val_MCR | + (mcfg->rx_comparison_enable_mode << RCMPM_BIT), + msp->registers + MSP_MCR); + + writel(mcfg->comparison_mask, + msp->registers + MSP_RCM); + writel(mcfg->comparison_value, + msp->registers + MSP_RCV); + + } + } + + return 0; +} + +static int enable_msp(struct ux500_msp *msp, struct ux500_msp_config *config) +{ + int status = 0; + u32 reg_val_DMACR, reg_val_GCR; + + /* Check msp state whether in RUN or CONFIGURED Mode */ + if ((msp->msp_state == MSP_STATE_IDLE) && (msp->plat_init)) { + status = msp->plat_init(); + if (status) { + dev_err(msp->dev, "%s: ERROR: Failed to init MSP (%d)!\n", + __func__, status); + return status; + } + } + + /* Configure msp with protocol dependent settings */ + configure_protocol(msp, config); + setup_bitclk(msp, config); + if (config->multichannel_configured == 1) { + status = configure_multichannel(msp, config); + if (status) + dev_warn(msp->dev, + "%s: WARN: configure_multichannel failed (%d)!\n", + __func__, status); + } + + /* Make sure the correct DMA-directions are configured */ + if ((config->direction & MSP_DIR_RX) && (!msp->dma_cfg_rx)) { + dev_err(msp->dev, "%s: ERROR: MSP RX-mode is not configured!", + __func__); + return -EINVAL; + } + if ((config->direction == MSP_DIR_TX) && (!msp->dma_cfg_tx)) { + dev_err(msp->dev, "%s: ERROR: MSP TX-mode is not configured!", + __func__); + return -EINVAL; + } + + reg_val_DMACR = readl(msp->registers + MSP_DMACR); + if (config->direction & MSP_DIR_RX) + reg_val_DMACR |= RX_DMA_ENABLE; + if (config->direction & MSP_DIR_TX) + reg_val_DMACR |= TX_DMA_ENABLE; + writel(reg_val_DMACR, msp->registers + MSP_DMACR); + + writel(config->iodelay, msp->registers + MSP_IODLY); + + /* Enable frame generation logic */ + reg_val_GCR = readl(msp->registers + MSP_GCR); + writel(reg_val_GCR | FRAME_GEN_ENABLE, msp->registers + MSP_GCR); + + return status; +} + +static void flush_fifo_rx(struct ux500_msp *msp) +{ + u32 reg_val_DR, reg_val_GCR, reg_val_FLR; + u32 limit = 32; + + reg_val_GCR = readl(msp->registers + MSP_GCR); + writel(reg_val_GCR | RX_ENABLE, msp->registers + MSP_GCR); + + reg_val_FLR = readl(msp->registers + MSP_FLR); + while (!(reg_val_FLR & RX_FIFO_EMPTY) && limit--) { + reg_val_DR = readl(msp->registers + MSP_DR); + reg_val_FLR = readl(msp->registers + MSP_FLR); + } + + writel(reg_val_GCR, msp->registers + MSP_GCR); +} + +static void flush_fifo_tx(struct ux500_msp *msp) +{ + u32 reg_val_TSTDR, reg_val_GCR, reg_val_FLR; + u32 limit = 32; + + reg_val_GCR = readl(msp->registers + MSP_GCR); + writel(reg_val_GCR | TX_ENABLE, msp->registers + MSP_GCR); + writel(MSP_ITCR_ITEN | MSP_ITCR_TESTFIFO, msp->registers + MSP_ITCR); + + reg_val_FLR = readl(msp->registers + MSP_FLR); + while (!(reg_val_FLR & TX_FIFO_EMPTY) && limit--) { + reg_val_TSTDR = readl(msp->registers + MSP_TSTDR); + reg_val_FLR = readl(msp->registers + MSP_FLR); + } + writel(0x0, msp->registers + MSP_ITCR); + writel(reg_val_GCR, msp->registers + MSP_GCR); +} + +int ux500_msp_i2s_open(struct ux500_msp *msp, + struct ux500_msp_config *config) +{ + u32 old_reg, new_reg, mask; + int res; + unsigned int tx_sel, rx_sel, tx_busy, rx_busy; + + if (in_interrupt()) { + dev_err(msp->dev, + "%s: ERROR: Open called in interrupt context!\n", + __func__); + return -1; + } + + tx_sel = (config->direction & MSP_DIR_TX) > 0; + rx_sel = (config->direction & MSP_DIR_RX) > 0; + if (!tx_sel && !rx_sel) { + dev_err(msp->dev, "%s: Error: No direction selected!\n", + __func__); + return -EINVAL; + } + + tx_busy = (msp->dir_busy & MSP_DIR_TX) > 0; + rx_busy = (msp->dir_busy & MSP_DIR_RX) > 0; + if (tx_busy && tx_sel) { + dev_err(msp->dev, "%s: Error: TX is in use!\n", __func__); + return -EBUSY; + } + if (rx_busy && rx_sel) { + dev_err(msp->dev, "%s: Error: RX is in use!\n", __func__); + return -EBUSY; + } + + msp->dir_busy |= (tx_sel ? MSP_DIR_TX : 0) | (rx_sel ? MSP_DIR_RX : 0); + + /* First do the global config register */ + mask = RX_CLK_SEL_MASK | TX_CLK_SEL_MASK | RX_FSYNC_MASK | + TX_FSYNC_MASK | RX_SYNC_SEL_MASK | TX_SYNC_SEL_MASK | + RX_FIFO_ENABLE_MASK | TX_FIFO_ENABLE_MASK | SRG_CLK_SEL_MASK | + LOOPBACK_MASK | TX_EXTRA_DELAY_MASK; + + new_reg = (config->tx_clk_sel | config->rx_clk_sel | + config->rx_fsync_pol | config->tx_fsync_pol | + config->rx_fsync_sel | config->tx_fsync_sel | + config->rx_fifo_config | config->tx_fifo_config | + config->srg_clk_sel | config->loopback_enable | + config->tx_data_enable); + + old_reg = readl(msp->registers + MSP_GCR); + old_reg &= ~mask; + new_reg |= old_reg; + writel(new_reg, msp->registers + MSP_GCR); + + res = enable_msp(msp, config); + if (res < 0) { + dev_err(msp->dev, "%s: ERROR: enable_msp failed (%d)!\n", + __func__, res); + return -EBUSY; + } + if (config->loopback_enable & 0x80) + msp->loopback_enable = 1; + + /* Flush FIFOs */ + flush_fifo_tx(msp); + flush_fifo_rx(msp); + + msp->msp_state = MSP_STATE_CONFIGURED; + return 0; +} + +static void disable_msp_rx(struct ux500_msp *msp) +{ + u32 reg_val_GCR, reg_val_DMACR, reg_val_IMSC; + + reg_val_GCR = readl(msp->registers + MSP_GCR); + writel(reg_val_GCR & ~RX_ENABLE, msp->registers + MSP_GCR); + reg_val_DMACR = readl(msp->registers + MSP_DMACR); + writel(reg_val_DMACR & ~RX_DMA_ENABLE, msp->registers + MSP_DMACR); + reg_val_IMSC = readl(msp->registers + MSP_IMSC); + writel(reg_val_IMSC & + ~(RX_SERVICE_INT | RX_OVERRUN_ERROR_INT), + msp->registers + MSP_IMSC); + + msp->dir_busy &= ~MSP_DIR_RX; +} + +static void disable_msp_tx(struct ux500_msp *msp) +{ + u32 reg_val_GCR, reg_val_DMACR, reg_val_IMSC; + + reg_val_GCR = readl(msp->registers + MSP_GCR); + writel(reg_val_GCR & ~TX_ENABLE, msp->registers + MSP_GCR); + reg_val_DMACR = readl(msp->registers + MSP_DMACR); + writel(reg_val_DMACR & ~TX_DMA_ENABLE, msp->registers + MSP_DMACR); + reg_val_IMSC = readl(msp->registers + MSP_IMSC); + writel(reg_val_IMSC & + ~(TX_SERVICE_INT | TX_UNDERRUN_ERR_INT), + msp->registers + MSP_IMSC); + + msp->dir_busy &= ~MSP_DIR_TX; +} + +static int disable_msp(struct ux500_msp *msp, unsigned int dir) +{ + u32 reg_val_GCR; + int status = 0; + unsigned int disable_tx, disable_rx; + + reg_val_GCR = readl(msp->registers + MSP_GCR); + disable_tx = dir & MSP_DIR_TX; + disable_rx = dir & MSP_DIR_TX; + if (disable_tx && disable_rx) { + reg_val_GCR = readl(msp->registers + MSP_GCR); + writel(reg_val_GCR | LOOPBACK_MASK, + msp->registers + MSP_GCR); + + /* Flush TX-FIFO */ + flush_fifo_tx(msp); + + /* Disable TX-channel */ + writel((readl(msp->registers + MSP_GCR) & + (~TX_ENABLE)), msp->registers + MSP_GCR); + + /* Flush RX-FIFO */ + flush_fifo_rx(msp); + + /* Disable Loopback and Receive channel */ + writel((readl(msp->registers + MSP_GCR) & + (~(RX_ENABLE | LOOPBACK_MASK))), + msp->registers + MSP_GCR); + + disable_msp_tx(msp); + disable_msp_rx(msp); + } else if (disable_tx) + disable_msp_tx(msp); + else if (disable_rx) + disable_msp_rx(msp); + + return status; +} + +int ux500_msp_i2s_trigger(struct ux500_msp *msp, int cmd, int direction) +{ + u32 reg_val_GCR, enable_bit; + + if (msp->msp_state == MSP_STATE_IDLE) { + dev_err(msp->dev, "%s: ERROR: MSP is not configured!\n", + __func__); + return -EINVAL; + } + + switch (cmd) { + case SNDRV_PCM_TRIGGER_START: + case SNDRV_PCM_TRIGGER_RESUME: + case SNDRV_PCM_TRIGGER_PAUSE_RELEASE: + if (direction == SNDRV_PCM_STREAM_PLAYBACK) + enable_bit = TX_ENABLE; + else + enable_bit = RX_ENABLE; + reg_val_GCR = readl(msp->registers + MSP_GCR); + writel(reg_val_GCR | enable_bit, msp->registers + MSP_GCR); + break; + + case SNDRV_PCM_TRIGGER_STOP: + case SNDRV_PCM_TRIGGER_SUSPEND: + case SNDRV_PCM_TRIGGER_PAUSE_PUSH: + if (direction == SNDRV_PCM_STREAM_PLAYBACK) + disable_msp_tx(msp); + else + disable_msp_rx(msp); + break; + default: + return -EINVAL; + break; + } + + return 0; +} + +int ux500_msp_i2s_close(struct ux500_msp *msp, unsigned int dir) +{ + int status = 0; + + dev_dbg(msp->dev, "%s: Enter (dir = 0x%01x).\n", __func__, dir); + + status = disable_msp(msp, dir); + if (msp->dir_busy == 0) { + /* disable sample rate and frame generators */ + msp->msp_state = MSP_STATE_IDLE; + writel((readl(msp->registers + MSP_GCR) & + (~(FRAME_GEN_ENABLE | SRG_ENABLE))), + msp->registers + MSP_GCR); + if (msp->plat_exit) + status = msp->plat_exit(); + if (status) + dev_warn(msp->dev, + "%s: WARN: ux500_msp_i2s_exit failed (%d)!\n", + __func__, status); + writel(0, msp->registers + MSP_GCR); + writel(0, msp->registers + MSP_TCF); + writel(0, msp->registers + MSP_RCF); + writel(0, msp->registers + MSP_DMACR); + writel(0, msp->registers + MSP_SRG); + writel(0, msp->registers + MSP_MCR); + writel(0, msp->registers + MSP_RCM); + writel(0, msp->registers + MSP_RCV); + writel(0, msp->registers + MSP_TCE0); + writel(0, msp->registers + MSP_TCE1); + writel(0, msp->registers + MSP_TCE2); + writel(0, msp->registers + MSP_TCE3); + writel(0, msp->registers + MSP_RCE0); + writel(0, msp->registers + MSP_RCE1); + writel(0, msp->registers + MSP_RCE2); + writel(0, msp->registers + MSP_RCE3); + } + + return status; + +} + +int ux500_msp_i2s_init_msp(struct platform_device *pdev, + struct ux500_msp **msp_p, + struct msp_i2s_platform_data *platform_data) +{ + int ret = 0; + struct resource *res = NULL; + struct i2s_controller *i2s_cont; + struct ux500_msp *msp; + + dev_dbg(&pdev->dev, "%s: Enter (name: %s, id: %d).\n", __func__, + pdev->name, platform_data->id); + + *msp_p = devm_kzalloc(&pdev->dev, sizeof(struct ux500_msp), GFP_KERNEL); + msp = *msp_p; + + msp->id = platform_data->id; + msp->dev = &pdev->dev; + msp->plat_init = platform_data->msp_i2s_init; + msp->plat_exit = platform_data->msp_i2s_exit; + msp->dma_cfg_rx = platform_data->msp_i2s_dma_rx; + msp->dma_cfg_tx = platform_data->msp_i2s_dma_tx; + + res = platform_get_resource(pdev, IORESOURCE_MEM, 0); + if (res == NULL) { + dev_err(&pdev->dev, "%s: ERROR: Unable to get resource!\n", + __func__); + ret = -ENOMEM; + goto err_res; + } + + msp->registers = ioremap(res->start, (res->end - res->start + 1)); + if (msp->registers == NULL) { + dev_err(&pdev->dev, "%s: ERROR: ioremap failed!\n", __func__); + ret = -ENOMEM; + goto err_res; + } + + msp->msp_state = MSP_STATE_IDLE; + msp->loopback_enable = 0; + + /* I2S-controller is allocated and added in I2S controller class. */ + i2s_cont = devm_kzalloc(&pdev->dev, sizeof(*i2s_cont), GFP_KERNEL); + if (!i2s_cont) { + dev_err(&pdev->dev, + "%s: ERROR: Failed to allocate I2S-controller!\n", + __func__); + goto err_i2s_cont; + } + i2s_cont->dev.parent = &pdev->dev; + i2s_cont->data = (void *)msp; + i2s_cont->id = (s16)msp->id; + snprintf(i2s_cont->name, sizeof(i2s_cont->name), "ux500-msp-i2s.%04x", + msp->id); + dev_dbg(&pdev->dev, "I2S device-name: '%s'\n", i2s_cont->name); + msp->i2s_cont = i2s_cont; + + return 0; + +err_i2s_cont: + iounmap(msp->registers); + +err_res: + devm_kfree(&pdev->dev, msp); + + return ret; +} + +void ux500_msp_i2s_cleanup_msp(struct platform_device *pdev, + struct ux500_msp *msp) +{ + dev_dbg(msp->dev, "%s: Enter (id = %d).\n", __func__, msp->id); + + device_unregister(&msp->i2s_cont->dev); + devm_kfree(&pdev->dev, msp->i2s_cont); + + iounmap(msp->registers); + + devm_kfree(&pdev->dev, msp); +} + +MODULE_LICENSE("GPLv2"); diff --git a/sound/soc/ux500/ux500_msp_i2s.h b/sound/soc/ux500/ux500_msp_i2s.h new file mode 100644 index 00000000000..7f71b4a0d4b --- /dev/null +++ b/sound/soc/ux500/ux500_msp_i2s.h @@ -0,0 +1,553 @@ +/* + * Copyright (C) ST-Ericsson SA 2012 + * + * Author: Ola Lilja <ola.o.lilja@stericsson.com>, + * for ST-Ericsson. + * + * License terms: + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License version 2 as published + * by the Free Software Foundation. + */ + + +#ifndef UX500_MSP_I2S_H +#define UX500_MSP_I2S_H + +#include <linux/platform_device.h> + +#include <mach/board-mop500-msp.h> + +#define MSP_INPUT_FREQ_APB 48000000 + +/*** Stereo mode. Used for APB data accesses as 16 bits accesses (mono), + * 32 bits accesses (stereo). + ***/ +enum msp_stereo_mode { + MSP_MONO, + MSP_STEREO +}; + +/* Direction (Transmit/Receive mode) */ +enum msp_direction { + MSP_TX = 1, + MSP_RX = 2 +}; + +/* Transmit and receive configuration register */ +#define MSP_BIG_ENDIAN 0x00000000 +#define MSP_LITTLE_ENDIAN 0x00001000 +#define MSP_UNEXPECTED_FS_ABORT 0x00000000 +#define MSP_UNEXPECTED_FS_IGNORE 0x00008000 +#define MSP_NON_MODE_BIT_MASK 0x00009000 + +/* Global configuration register */ +#define RX_ENABLE 0x00000001 +#define RX_FIFO_ENABLE 0x00000002 +#define RX_SYNC_SRG 0x00000010 +#define RX_CLK_POL_RISING 0x00000020 +#define RX_CLK_SEL_SRG 0x00000040 +#define TX_ENABLE 0x00000100 +#define TX_FIFO_ENABLE 0x00000200 +#define TX_SYNC_SRG_PROG 0x00001800 +#define TX_SYNC_SRG_AUTO 0x00001000 +#define TX_CLK_POL_RISING 0x00002000 +#define TX_CLK_SEL_SRG 0x00004000 +#define TX_EXTRA_DELAY_ENABLE 0x00008000 +#define SRG_ENABLE 0x00010000 +#define FRAME_GEN_ENABLE 0x00100000 +#define SRG_CLK_SEL_APB 0x00000000 +#define RX_FIFO_SYNC_HI 0x00000000 +#define TX_FIFO_SYNC_HI 0x00000000 +#define SPI_CLK_MODE_NORMAL 0x00000000 + +#define MSP_FRAME_SIZE_AUTO -1 + +#define MSP_DR 0x00 +#define MSP_GCR 0x04 +#define MSP_TCF 0x08 +#define MSP_RCF 0x0c +#define MSP_SRG 0x10 +#define MSP_FLR 0x14 +#define MSP_DMACR 0x18 + +#define MSP_IMSC 0x20 +#define MSP_RIS 0x24 +#define MSP_MIS 0x28 +#define MSP_ICR 0x2c +#define MSP_MCR 0x30 +#define MSP_RCV 0x34 +#define MSP_RCM 0x38 + +#define MSP_TCE0 0x40 +#define MSP_TCE1 0x44 +#define MSP_TCE2 0x48 +#define MSP_TCE3 0x4c + +#define MSP_RCE0 0x60 +#define MSP_RCE1 0x64 +#define MSP_RCE2 0x68 +#define MSP_RCE3 0x6c +#define MSP_IODLY 0x70 + +#define MSP_ITCR 0x80 +#define MSP_ITIP 0x84 +#define MSP_ITOP 0x88 +#define MSP_TSTDR 0x8c + +#define MSP_PID0 0xfe0 +#define MSP_PID1 0xfe4 +#define MSP_PID2 0xfe8 +#define MSP_PID3 0xfec + +#define MSP_CID0 0xff0 +#define MSP_CID1 0xff4 +#define MSP_CID2 0xff8 +#define MSP_CID3 0xffc + +/* Protocol dependant parameters list */ +#define RX_ENABLE_MASK BIT(0) +#define RX_FIFO_ENABLE_MASK BIT(1) +#define RX_FSYNC_MASK BIT(2) +#define DIRECT_COMPANDING_MASK BIT(3) +#define RX_SYNC_SEL_MASK BIT(4) +#define RX_CLK_POL_MASK BIT(5) +#define RX_CLK_SEL_MASK BIT(6) +#define LOOPBACK_MASK BIT(7) +#define TX_ENABLE_MASK BIT(8) +#define TX_FIFO_ENABLE_MASK BIT(9) +#define TX_FSYNC_MASK BIT(10) +#define TX_MSP_TDR_TSR BIT(11) +#define TX_SYNC_SEL_MASK (BIT(12) | BIT(11)) +#define TX_CLK_POL_MASK BIT(13) +#define TX_CLK_SEL_MASK BIT(14) +#define TX_EXTRA_DELAY_MASK BIT(15) +#define SRG_ENABLE_MASK BIT(16) +#define SRG_CLK_POL_MASK BIT(17) +#define SRG_CLK_SEL_MASK (BIT(19) | BIT(18)) +#define FRAME_GEN_EN_MASK BIT(20) +#define SPI_CLK_MODE_MASK (BIT(22) | BIT(21)) +#define SPI_BURST_MODE_MASK BIT(23) + +#define RXEN_SHIFT 0 +#define RFFEN_SHIFT 1 +#define RFSPOL_SHIFT 2 +#define DCM_SHIFT 3 +#define RFSSEL_SHIFT 4 +#define RCKPOL_SHIFT 5 +#define RCKSEL_SHIFT 6 +#define LBM_SHIFT 7 +#define TXEN_SHIFT 8 +#define TFFEN_SHIFT 9 +#define TFSPOL_SHIFT 10 +#define TFSSEL_SHIFT 11 +#define TCKPOL_SHIFT 13 +#define TCKSEL_SHIFT 14 +#define TXDDL_SHIFT 15 +#define SGEN_SHIFT 16 +#define SCKPOL_SHIFT 17 +#define SCKSEL_SHIFT 18 +#define FGEN_SHIFT 20 +#define SPICKM_SHIFT 21 +#define TBSWAP_SHIFT 28 + +#define RCKPOL_MASK BIT(0) +#define TCKPOL_MASK BIT(0) +#define SPICKM_MASK (BIT(1) | BIT(0)) +#define MSP_RX_CLKPOL_BIT(n) ((n & RCKPOL_MASK) << RCKPOL_SHIFT) +#define MSP_TX_CLKPOL_BIT(n) ((n & TCKPOL_MASK) << TCKPOL_SHIFT) + +#define P1ELEN_SHIFT 0 +#define P1FLEN_SHIFT 3 +#define DTYP_SHIFT 10 +#define ENDN_SHIFT 12 +#define DDLY_SHIFT 13 +#define FSIG_SHIFT 15 +#define P2ELEN_SHIFT 16 +#define P2FLEN_SHIFT 19 +#define P2SM_SHIFT 26 +#define P2EN_SHIFT 27 +#define FSYNC_SHIFT 15 + +#define P1ELEN_MASK 0x00000007 +#define P2ELEN_MASK 0x00070000 +#define P1FLEN_MASK 0x00000378 +#define P2FLEN_MASK 0x03780000 +#define DDLY_MASK 0x00003000 +#define DTYP_MASK 0x00000600 +#define P2SM_MASK 0x04000000 +#define P2EN_MASK 0x08000000 +#define ENDN_MASK 0x00001000 +#define TFSPOL_MASK 0x00000400 +#define TBSWAP_MASK 0x30000000 +#define COMPANDING_MODE_MASK 0x00000c00 +#define FSYNC_MASK 0x00008000 + +#define MSP_P1_ELEM_LEN_BITS(n) (n & P1ELEN_MASK) +#define MSP_P2_ELEM_LEN_BITS(n) (((n) << P2ELEN_SHIFT) & P2ELEN_MASK) +#define MSP_P1_FRAME_LEN_BITS(n) (((n) << P1FLEN_SHIFT) & P1FLEN_MASK) +#define MSP_P2_FRAME_LEN_BITS(n) (((n) << P2FLEN_SHIFT) & P2FLEN_MASK) +#define MSP_DATA_DELAY_BITS(n) (((n) << DDLY_SHIFT) & DDLY_MASK) +#define MSP_DATA_TYPE_BITS(n) (((n) << DTYP_SHIFT) & DTYP_MASK) +#define MSP_P2_START_MODE_BIT(n) ((n << P2SM_SHIFT) & P2SM_MASK) +#define MSP_P2_ENABLE_BIT(n) ((n << P2EN_SHIFT) & P2EN_MASK) +#define MSP_SET_ENDIANNES_BIT(n) ((n << ENDN_SHIFT) & ENDN_MASK) +#define MSP_FSYNC_POL(n) ((n << TFSPOL_SHIFT) & TFSPOL_MASK) +#define MSP_DATA_WORD_SWAP(n) ((n << TBSWAP_SHIFT) & TBSWAP_MASK) +#define MSP_SET_COMPANDING_MODE(n) ((n << DTYP_SHIFT) & \ + COMPANDING_MODE_MASK) +#define MSP_SET_FSYNC_IGNORE(n) ((n << FSYNC_SHIFT) & FSYNC_MASK) + +/* Flag register */ +#define RX_BUSY BIT(0) +#define RX_FIFO_EMPTY BIT(1) +#define RX_FIFO_FULL BIT(2) +#define TX_BUSY BIT(3) +#define TX_FIFO_EMPTY BIT(4) +#define TX_FIFO_FULL BIT(5) + +#define RBUSY_SHIFT 0 +#define RFE_SHIFT 1 +#define RFU_SHIFT 2 +#define TBUSY_SHIFT 3 +#define TFE_SHIFT 4 +#define TFU_SHIFT 5 + +/* Multichannel control register */ +#define RMCEN_SHIFT 0 +#define RMCSF_SHIFT 1 +#define RCMPM_SHIFT 3 +#define TMCEN_SHIFT 5 +#define TNCSF_SHIFT 6 + +/* Sample rate generator register */ +#define SCKDIV_SHIFT 0 +#define FRWID_SHIFT 10 +#define FRPER_SHIFT 16 + +#define SCK_DIV_MASK 0x0000003FF +#define FRAME_WIDTH_BITS(n) (((n) << FRWID_SHIFT) & 0x0000FC00) +#define FRAME_PERIOD_BITS(n) (((n) << FRPER_SHIFT) & 0x1FFF0000) + +/* DMA controller register */ +#define RX_DMA_ENABLE BIT(0) +#define TX_DMA_ENABLE BIT(1) + +#define RDMAE_SHIFT 0 +#define TDMAE_SHIFT 1 + +/* Interrupt Register */ +#define RX_SERVICE_INT BIT(0) +#define RX_OVERRUN_ERROR_INT BIT(1) +#define RX_FSYNC_ERR_INT BIT(2) +#define RX_FSYNC_INT BIT(3) +#define TX_SERVICE_INT BIT(4) +#define TX_UNDERRUN_ERR_INT BIT(5) +#define TX_FSYNC_ERR_INT BIT(6) +#define TX_FSYNC_INT BIT(7) +#define ALL_INT 0x000000ff + +/* MSP test control register */ +#define MSP_ITCR_ITEN BIT(0) +#define MSP_ITCR_TESTFIFO BIT(1) + +#define RMCEN_BIT 0 +#define RMCSF_BIT 1 +#define RCMPM_BIT 3 +#define TMCEN_BIT 5 +#define TNCSF_BIT 6 + +/* Single or dual phase mode */ +enum msp_phase_mode { + MSP_SINGLE_PHASE, + MSP_DUAL_PHASE +}; + +/* Frame length */ +enum msp_frame_length { + MSP_FRAME_LEN_1 = 0, + MSP_FRAME_LEN_2 = 1, + MSP_FRAME_LEN_4 = 3, + MSP_FRAME_LEN_8 = 7, + MSP_FRAME_LEN_12 = 11, + MSP_FRAME_LEN_16 = 15, + MSP_FRAME_LEN_20 = 19, + MSP_FRAME_LEN_32 = 31, + MSP_FRAME_LEN_48 = 47, + MSP_FRAME_LEN_64 = 63 +}; + +/* Element length */ +enum msp_elem_length { + MSP_ELEM_LEN_8 = 0, + MSP_ELEM_LEN_10 = 1, + MSP_ELEM_LEN_12 = 2, + MSP_ELEM_LEN_14 = 3, + MSP_ELEM_LEN_16 = 4, + MSP_ELEM_LEN_20 = 5, + MSP_ELEM_LEN_24 = 6, + MSP_ELEM_LEN_32 = 7 +}; + +enum msp_data_xfer_width { + MSP_DATA_TRANSFER_WIDTH_BYTE, + MSP_DATA_TRANSFER_WIDTH_HALFWORD, + MSP_DATA_TRANSFER_WIDTH_WORD +}; + +enum msp_frame_sync { + MSP_FSYNC_UNIGNORE = 0, + MSP_FSYNC_IGNORE = 1, +}; + +enum msp_phase2_start_mode { + MSP_PHASE2_START_MODE_IMEDIATE, + MSP_PHASE2_START_MODE_FSYNC +}; + +enum msp_btf { + MSP_BTF_MS_BIT_FIRST = 0, + MSP_BTF_LS_BIT_FIRST = 1 +}; + +enum msp_fsync_pol { + MSP_FSYNC_POL_ACT_HI = 0, + MSP_FSYNC_POL_ACT_LO = 1 +}; + +/* Data delay (in bit clock cycles) */ +enum msp_delay { + MSP_DELAY_0 = 0, + MSP_DELAY_1 = 1, + MSP_DELAY_2 = 2, + MSP_DELAY_3 = 3 +}; + +/* Configurations of clocks (transmit, receive or sample rate generator) */ +enum msp_edge { + MSP_FALLING_EDGE = 0, + MSP_RISING_EDGE = 1, +}; + +enum msp_hws { + MSP_SWAP_NONE = 0, + MSP_SWAP_BYTE_PER_WORD = 1, + MSP_SWAP_BYTE_PER_HALF_WORD = 2, + MSP_SWAP_HALF_WORD_PER_WORD = 3 +}; + +enum msp_compress_mode { + MSP_COMPRESS_MODE_LINEAR = 0, + MSP_COMPRESS_MODE_MU_LAW = 2, + MSP_COMPRESS_MODE_A_LAW = 3 +}; + +enum msp_spi_burst_mode { + MSP_SPI_BURST_MODE_DISABLE = 0, + MSP_SPI_BURST_MODE_ENABLE = 1 +}; + +enum msp_expand_mode { + MSP_EXPAND_MODE_LINEAR = 0, + MSP_EXPAND_MODE_LINEAR_SIGNED = 1, + MSP_EXPAND_MODE_MU_LAW = 2, + MSP_EXPAND_MODE_A_LAW = 3 +}; + +#define MSP_FRAME_PERIOD_IN_MONO_MODE 256 +#define MSP_FRAME_PERIOD_IN_STEREO_MODE 32 +#define MSP_FRAME_WIDTH_IN_STEREO_MODE 16 + +enum msp_protocol { + MSP_I2S_PROTOCOL, + MSP_PCM_PROTOCOL, + MSP_PCM_COMPAND_PROTOCOL, + MSP_INVALID_PROTOCOL +}; + +/* + * No of registers to backup during + * suspend resume + */ +#define MAX_MSP_BACKUP_REGS 36 + +enum enum_i2s_controller { + MSP_0_I2S_CONTROLLER = 0, + MSP_1_I2S_CONTROLLER, + MSP_2_I2S_CONTROLLER, + MSP_3_I2S_CONTROLLER, +}; + +enum i2s_direction_t { + MSP_DIR_TX = 0x01, + MSP_DIR_RX = 0x02, +}; + +enum msp_data_size { + MSP_DATA_BITS_DEFAULT = -1, + MSP_DATA_BITS_8 = 0x00, + MSP_DATA_BITS_10, + MSP_DATA_BITS_12, + MSP_DATA_BITS_14, + MSP_DATA_BITS_16, + MSP_DATA_BITS_20, + MSP_DATA_BITS_24, + MSP_DATA_BITS_32, +}; + +enum msp_state { + MSP_STATE_IDLE = 0, + MSP_STATE_CONFIGURED = 1, + MSP_STATE_RUNNING = 2, +}; + +enum msp_rx_comparison_enable_mode { + MSP_COMPARISON_DISABLED = 0, + MSP_COMPARISON_NONEQUAL_ENABLED = 2, + MSP_COMPARISON_EQUAL_ENABLED = 3 +}; + +struct msp_multichannel_config { + bool rx_multichannel_enable; + bool tx_multichannel_enable; + enum msp_rx_comparison_enable_mode rx_comparison_enable_mode; + u8 padding; + u32 comparison_value; + u32 comparison_mask; + u32 rx_channel_0_enable; + u32 rx_channel_1_enable; + u32 rx_channel_2_enable; + u32 rx_channel_3_enable; + u32 tx_channel_0_enable; + u32 tx_channel_1_enable; + u32 tx_channel_2_enable; + u32 tx_channel_3_enable; +}; + +struct msp_protdesc { + u32 rx_phase_mode; + u32 tx_phase_mode; + u32 rx_phase2_start_mode; + u32 tx_phase2_start_mode; + u32 rx_byte_order; + u32 tx_byte_order; + u32 rx_frame_len_1; + u32 rx_frame_len_2; + u32 tx_frame_len_1; + u32 tx_frame_len_2; + u32 rx_elem_len_1; + u32 rx_elem_len_2; + u32 tx_elem_len_1; + u32 tx_elem_len_2; + u32 rx_data_delay; + u32 tx_data_delay; + u32 rx_clk_pol; + u32 tx_clk_pol; + u32 rx_fsync_pol; + u32 tx_fsync_pol; + u32 rx_half_word_swap; + u32 tx_half_word_swap; + u32 compression_mode; + u32 expansion_mode; + u32 frame_sync_ignore; + u32 frame_period; + u32 frame_width; + u32 clocks_per_frame; +}; + +struct i2s_message { + enum i2s_direction_t i2s_direction; + void *txdata; + void *rxdata; + size_t txbytes; + size_t rxbytes; + int dma_flag; + int tx_offset; + int rx_offset; + bool cyclic_dma; + dma_addr_t buf_addr; + size_t buf_len; + size_t period_len; +}; + +struct i2s_controller { + struct module *owner; + unsigned int id; + unsigned int class; + const struct i2s_algorithm *algo; /* the algorithm to access the bus */ + void *data; + struct mutex bus_lock; + struct device dev; /* the controller device */ + char name[48]; +}; + +struct ux500_msp_config { + unsigned int f_inputclk; + unsigned int rx_clk_sel; + unsigned int tx_clk_sel; + unsigned int srg_clk_sel; + unsigned int rx_fsync_pol; + unsigned int tx_fsync_pol; + unsigned int rx_fsync_sel; + unsigned int tx_fsync_sel; + unsigned int rx_fifo_config; + unsigned int tx_fifo_config; + unsigned int spi_clk_mode; + unsigned int spi_burst_mode; + unsigned int loopback_enable; + unsigned int tx_data_enable; + unsigned int default_protdesc; + struct msp_protdesc protdesc; + int multichannel_configured; + struct msp_multichannel_config multichannel_config; + unsigned int direction; + unsigned int protocol; + unsigned int frame_freq; + unsigned int frame_size; + enum msp_data_size data_size; + unsigned int def_elem_len; + unsigned int iodelay; + void (*handler) (void *data); + void *tx_callback_data; + void *rx_callback_data; +}; + +struct ux500_msp { + enum enum_i2s_controller id; + void __iomem *registers; + struct device *dev; + struct i2s_controller *i2s_cont; + struct stedma40_chan_cfg *dma_cfg_rx; + struct stedma40_chan_cfg *dma_cfg_tx; + struct dma_chan *tx_pipeid; + struct dma_chan *rx_pipeid; + enum msp_state msp_state; + int (*transfer) (struct ux500_msp *msp, struct i2s_message *message); + int (*plat_init) (void); + int (*plat_exit) (void); + struct timer_list notify_timer; + int def_elem_len; + unsigned int dir_busy; + int loopback_enable; + u32 backup_regs[MAX_MSP_BACKUP_REGS]; + unsigned int f_bitclk; +}; + +struct ux500_msp_dma_params { + unsigned int data_size; + struct stedma40_chan_cfg *dma_cfg; +}; + +int ux500_msp_i2s_init_msp(struct platform_device *pdev, + struct ux500_msp **msp_p, + struct msp_i2s_platform_data *platform_data); +void ux500_msp_i2s_cleanup_msp(struct platform_device *pdev, + struct ux500_msp *msp); +int ux500_msp_i2s_open(struct ux500_msp *msp, struct ux500_msp_config *config); +int ux500_msp_i2s_close(struct ux500_msp *msp, + unsigned int dir); +int ux500_msp_i2s_trigger(struct ux500_msp *msp, int cmd, + int direction); + +#endif diff --git a/sound/sound_core.c b/sound/sound_core.c index c6e81fb928e..fb9255cca21 100644 --- a/sound/sound_core.c +++ b/sound/sound_core.c @@ -361,7 +361,7 @@ int register_sound_special_device(const struct file_operations *fops, int unit, struct device *dev) { const int chain = unit % SOUND_STEP; - int max_unit = 128 + chain; + int max_unit = 256; const char *name; char _name[16]; diff --git a/sound/usb/card.c b/sound/usb/card.c index 4a7be7b9833..d5b5c3388e2 100644 --- a/sound/usb/card.c +++ b/sound/usb/card.c @@ -131,8 +131,9 @@ static void snd_usb_stream_disconnect(struct list_head *head) subs = &as->substream[idx]; if (!subs->num_formats) continue; - snd_usb_release_substream_urbs(subs, 1); subs->interface = -1; + subs->data_endpoint = NULL; + subs->sync_endpoint = NULL; } } @@ -276,6 +277,7 @@ static int snd_usb_create_streams(struct snd_usb_audio *chip, int ctrlif) static int snd_usb_audio_free(struct snd_usb_audio *chip) { + mutex_destroy(&chip->mutex); kfree(chip); return 0; } @@ -336,6 +338,7 @@ static int snd_usb_audio_create(struct usb_device *dev, int idx, return -ENOMEM; } + mutex_init(&chip->mutex); mutex_init(&chip->shutdown_mutex); chip->index = idx; chip->dev = dev; @@ -348,6 +351,7 @@ static int snd_usb_audio_create(struct usb_device *dev, int idx, chip->usb_id = USB_ID(le16_to_cpu(dev->descriptor.idVendor), le16_to_cpu(dev->descriptor.idProduct)); INIT_LIST_HEAD(&chip->pcm_list); + INIT_LIST_HEAD(&chip->ep_list); INIT_LIST_HEAD(&chip->midi_list); INIT_LIST_HEAD(&chip->mixer_list); @@ -565,6 +569,10 @@ static void snd_usb_audio_disconnect(struct usb_device *dev, list_for_each(p, &chip->pcm_list) { snd_usb_stream_disconnect(p); } + /* release the endpoint resources */ + list_for_each(p, &chip->ep_list) { + snd_usb_endpoint_free(p); + } /* release the midi resources */ list_for_each(p, &chip->midi_list) { snd_usbmidi_disconnect(p); diff --git a/sound/usb/card.h b/sound/usb/card.h index da5fa1ac4ed..0d37238b845 100644 --- a/sound/usb/card.h +++ b/sound/usb/card.h @@ -30,20 +30,71 @@ struct audioformat { }; struct snd_usb_substream; +struct snd_usb_endpoint; struct snd_urb_ctx { struct urb *urb; unsigned int buffer_size; /* size of data buffer, if data URB */ struct snd_usb_substream *subs; + struct snd_usb_endpoint *ep; int index; /* index for urb array */ int packets; /* number of packets per urb */ + int packet_size[MAX_PACKS_HS]; /* size of packets for next submission */ + struct list_head ready_list; }; -struct snd_urb_ops { - int (*prepare)(struct snd_usb_substream *subs, struct snd_pcm_runtime *runtime, struct urb *u); - int (*retire)(struct snd_usb_substream *subs, struct snd_pcm_runtime *runtime, struct urb *u); - int (*prepare_sync)(struct snd_usb_substream *subs, struct snd_pcm_runtime *runtime, struct urb *u); - int (*retire_sync)(struct snd_usb_substream *subs, struct snd_pcm_runtime *runtime, struct urb *u); +struct snd_usb_endpoint { + struct snd_usb_audio *chip; + + int use_count; + int ep_num; /* the referenced endpoint number */ + int type; /* SND_USB_ENDPOINT_TYPE_* */ + unsigned long flags; + + void (*prepare_data_urb) (struct snd_usb_substream *subs, + struct urb *urb); + void (*retire_data_urb) (struct snd_usb_substream *subs, + struct urb *urb); + + struct snd_usb_substream *data_subs; + struct snd_usb_endpoint *sync_master; + struct snd_usb_endpoint *sync_slave; + + struct snd_urb_ctx urb[MAX_URBS]; + + struct snd_usb_packet_info { + uint32_t packet_size[MAX_PACKS_HS]; + int packets; + } next_packet[MAX_URBS]; + int next_packet_read_pos, next_packet_write_pos; + struct list_head ready_playback_urbs; + + unsigned int nurbs; /* # urbs */ + unsigned long active_mask; /* bitmask of active urbs */ + unsigned long unlink_mask; /* bitmask of unlinked urbs */ + char *syncbuf; /* sync buffer for all sync URBs */ + dma_addr_t sync_dma; /* DMA address of syncbuf */ + + unsigned int pipe; /* the data i/o pipe */ + unsigned int freqn; /* nominal sampling rate in fs/fps in Q16.16 format */ + unsigned int freqm; /* momentary sampling rate in fs/fps in Q16.16 format */ + int freqshift; /* how much to shift the feedback value to get Q16.16 */ + unsigned int freqmax; /* maximum sampling rate, used for buffer management */ + unsigned int phase; /* phase accumulator */ + unsigned int maxpacksize; /* max packet size in bytes */ + unsigned int maxframesize; /* max packet size in frames */ + unsigned int curpacksize; /* current packet size in bytes (for capture) */ + unsigned int curframesize; /* current packet size in frames (for capture) */ + unsigned int syncmaxsize; /* sync endpoint packet size */ + unsigned int fill_max:1; /* fill max packet size always */ + unsigned int datainterval; /* log_2 of data packet interval */ + unsigned int syncinterval; /* P for adaptive mode, 0 otherwise */ + unsigned char silence_value; + unsigned int stride; + int iface, alt_idx; + + spinlock_t lock; + struct list_head list; }; struct snd_usb_substream { @@ -57,21 +108,6 @@ struct snd_usb_substream { unsigned int cur_rate; /* current rate (for hw_params callback) */ unsigned int period_bytes; /* current period bytes (for hw_params callback) */ unsigned int altset_idx; /* USB data format: index of alternate setting */ - unsigned int datapipe; /* the data i/o pipe */ - unsigned int syncpipe; /* 1 - async out or adaptive in */ - unsigned int datainterval; /* log_2 of data packet interval */ - unsigned int syncinterval; /* P for adaptive mode, 0 otherwise */ - unsigned int freqn; /* nominal sampling rate in fs/fps in Q16.16 format */ - unsigned int freqm; /* momentary sampling rate in fs/fps in Q16.16 format */ - int freqshift; /* how much to shift the feedback value to get Q16.16 */ - unsigned int freqmax; /* maximum sampling rate, used for buffer management */ - unsigned int phase; /* phase accumulator */ - unsigned int maxpacksize; /* max packet size in bytes */ - unsigned int maxframesize; /* max packet size in frames */ - unsigned int curpacksize; /* current packet size in bytes (for capture) */ - unsigned int curframesize; /* current packet size in frames (for capture) */ - unsigned int syncmaxsize; /* sync endpoint packet size */ - unsigned int fill_max: 1; /* fill max packet size always */ unsigned int txfr_quirk:1; /* allow sub-frame alignment */ unsigned int fmt_type; /* USB audio format type (1-3) */ @@ -82,11 +118,10 @@ struct snd_usb_substream { unsigned long active_mask; /* bitmask of active urbs */ unsigned long unlink_mask; /* bitmask of unlinked urbs */ - unsigned int nurbs; /* # urbs */ - struct snd_urb_ctx dataurb[MAX_URBS]; /* data urb table */ - struct snd_urb_ctx syncurb[SYNC_URBS]; /* sync urb table */ - char *syncbuf; /* sync buffer for all sync URBs */ - dma_addr_t sync_dma; /* DMA address of syncbuf */ + /* data and sync endpoints for this stream */ + struct snd_usb_endpoint *data_endpoint; + struct snd_usb_endpoint *sync_endpoint; + unsigned long flags; u64 formats; /* format bitmasks (all or'ed) */ unsigned int num_formats; /* number of supported audio formats (list) */ @@ -94,7 +129,6 @@ struct snd_usb_substream { struct snd_pcm_hw_constraint_list rate_list; /* limited rates */ spinlock_t lock; - struct snd_urb_ops ops; /* callbacks (must be filled at init) */ int last_frame_number; /* stored frame number */ int last_delay; /* stored delay */ }; diff --git a/sound/usb/endpoint.c b/sound/usb/endpoint.c index 08dcce53720..e6906901deb 100644 --- a/sound/usb/endpoint.c +++ b/sound/usb/endpoint.c @@ -20,9 +20,11 @@ #include <linux/ratelimit.h> #include <linux/usb.h> #include <linux/usb/audio.h> +#include <linux/slab.h> #include <sound/core.h> #include <sound/pcm.h> +#include <sound/pcm_params.h> #include "usbaudio.h" #include "helper.h" @@ -30,6 +32,36 @@ #include "endpoint.h" #include "pcm.h" +#define EP_FLAG_ACTIVATED 0 +#define EP_FLAG_RUNNING 1 + +/* + * snd_usb_endpoint is a model that abstracts everything related to an + * USB endpoint and its streaming. + * + * There are functions to activate and deactivate the streaming URBs and + * optional callbacks to let the pcm logic handle the actual content of the + * packets for playback and record. Thus, the bus streaming and the audio + * handlers are fully decoupled. + * + * There are two different types of endpoints in audio applications. + * + * SND_USB_ENDPOINT_TYPE_DATA handles full audio data payload for both + * inbound and outbound traffic. + * + * SND_USB_ENDPOINT_TYPE_SYNC endpoints are for inbound traffic only and + * expect the payload to carry Q10.14 / Q16.16 formatted sync information + * (3 or 4 bytes). + * + * Each endpoint has to be configured prior to being used by calling + * snd_usb_endpoint_set_params(). + * + * The model incorporates a reference counting, so that multiple users + * can call snd_usb_endpoint_start() and snd_usb_endpoint_stop(), and + * only the first user will effectively start the URBs, and only the last + * one to stop it will tear the URBs down again. + */ + /* * convert a sampling rate into our full speed format (fs/1000 in Q16.16) * this will overflow at approx 524 kHz @@ -49,71 +81,415 @@ static inline unsigned get_usb_high_speed_rate(unsigned int rate) } /* - * unlink active urbs. + * release a urb data */ -static int deactivate_urbs(struct snd_usb_substream *subs, int force, int can_sleep) +static void release_urb_ctx(struct snd_urb_ctx *u) { - struct snd_usb_audio *chip = subs->stream->chip; - unsigned int i; - int async; + if (u->buffer_size) + usb_free_coherent(u->ep->chip->dev, u->buffer_size, + u->urb->transfer_buffer, + u->urb->transfer_dma); + usb_free_urb(u->urb); + u->urb = NULL; +} + +static const char *usb_error_string(int err) +{ + switch (err) { + case -ENODEV: + return "no device"; + case -ENOENT: + return "endpoint not enabled"; + case -EPIPE: + return "endpoint stalled"; + case -ENOSPC: + return "not enough bandwidth"; + case -ESHUTDOWN: + return "device disabled"; + case -EHOSTUNREACH: + return "device suspended"; + case -EINVAL: + case -EAGAIN: + case -EFBIG: + case -EMSGSIZE: + return "internal error"; + default: + return "unknown error"; + } +} + +/** + * snd_usb_endpoint_implicit_feedback_sink: Report endpoint usage type + * + * @ep: The snd_usb_endpoint + * + * Determine whether an endpoint is driven by an implicit feedback + * data endpoint source. + */ +int snd_usb_endpoint_implict_feedback_sink(struct snd_usb_endpoint *ep) +{ + return ep->sync_master && + ep->sync_master->type == SND_USB_ENDPOINT_TYPE_DATA && + ep->type == SND_USB_ENDPOINT_TYPE_DATA && + usb_pipeout(ep->pipe); +} - subs->running = 0; +/* + * For streaming based on information derived from sync endpoints, + * prepare_outbound_urb_sizes() will call next_packet_size() to + * determine the number of samples to be sent in the next packet. + * + * For implicit feedback, next_packet_size() is unused. + */ +static int next_packet_size(struct snd_usb_endpoint *ep) +{ + unsigned long flags; + int ret; - if (!force && subs->stream->chip->shutdown) /* to be sure... */ - return -EBADFD; + if (ep->fill_max) + return ep->maxframesize; - async = !can_sleep && chip->async_unlink; + spin_lock_irqsave(&ep->lock, flags); + ep->phase = (ep->phase & 0xffff) + + (ep->freqm << ep->datainterval); + ret = min(ep->phase >> 16, ep->maxframesize); + spin_unlock_irqrestore(&ep->lock, flags); - if (!async && in_interrupt()) - return 0; + return ret; +} - for (i = 0; i < subs->nurbs; i++) { - if (test_bit(i, &subs->active_mask)) { - if (!test_and_set_bit(i, &subs->unlink_mask)) { - struct urb *u = subs->dataurb[i].urb; - if (async) - usb_unlink_urb(u); - else - usb_kill_urb(u); +static void retire_outbound_urb(struct snd_usb_endpoint *ep, + struct snd_urb_ctx *urb_ctx) +{ + if (ep->retire_data_urb) + ep->retire_data_urb(ep->data_subs, urb_ctx->urb); +} + +static void retire_inbound_urb(struct snd_usb_endpoint *ep, + struct snd_urb_ctx *urb_ctx) +{ + struct urb *urb = urb_ctx->urb; + + if (ep->sync_slave) + snd_usb_handle_sync_urb(ep->sync_slave, ep, urb); + + if (ep->retire_data_urb) + ep->retire_data_urb(ep->data_subs, urb); +} + +static void prepare_outbound_urb_sizes(struct snd_usb_endpoint *ep, + struct snd_urb_ctx *ctx) +{ + int i; + + for (i = 0; i < ctx->packets; ++i) + ctx->packet_size[i] = next_packet_size(ep); +} + +/* + * Prepare a PLAYBACK urb for submission to the bus. + */ +static void prepare_outbound_urb(struct snd_usb_endpoint *ep, + struct snd_urb_ctx *ctx) +{ + int i; + struct urb *urb = ctx->urb; + unsigned char *cp = urb->transfer_buffer; + + urb->dev = ep->chip->dev; /* we need to set this at each time */ + + switch (ep->type) { + case SND_USB_ENDPOINT_TYPE_DATA: + if (ep->prepare_data_urb) { + ep->prepare_data_urb(ep->data_subs, urb); + } else { + /* no data provider, so send silence */ + unsigned int offs = 0; + for (i = 0; i < ctx->packets; ++i) { + int counts = ctx->packet_size[i]; + urb->iso_frame_desc[i].offset = offs * ep->stride; + urb->iso_frame_desc[i].length = counts * ep->stride; + offs += counts; } + + urb->number_of_packets = ctx->packets; + urb->transfer_buffer_length = offs * ep->stride; + memset(urb->transfer_buffer, ep->silence_value, + offs * ep->stride); } + break; + + case SND_USB_ENDPOINT_TYPE_SYNC: + if (snd_usb_get_speed(ep->chip->dev) >= USB_SPEED_HIGH) { + /* + * fill the length and offset of each urb descriptor. + * the fixed 12.13 frequency is passed as 16.16 through the pipe. + */ + urb->iso_frame_desc[0].length = 4; + urb->iso_frame_desc[0].offset = 0; + cp[0] = ep->freqn; + cp[1] = ep->freqn >> 8; + cp[2] = ep->freqn >> 16; + cp[3] = ep->freqn >> 24; + } else { + /* + * fill the length and offset of each urb descriptor. + * the fixed 10.14 frequency is passed through the pipe. + */ + urb->iso_frame_desc[0].length = 3; + urb->iso_frame_desc[0].offset = 0; + cp[0] = ep->freqn >> 2; + cp[1] = ep->freqn >> 10; + cp[2] = ep->freqn >> 18; + } + + break; } - if (subs->syncpipe) { - for (i = 0; i < SYNC_URBS; i++) { - if (test_bit(i+16, &subs->active_mask)) { - if (!test_and_set_bit(i+16, &subs->unlink_mask)) { - struct urb *u = subs->syncurb[i].urb; - if (async) - usb_unlink_urb(u); - else - usb_kill_urb(u); - } - } +} + +/* + * Prepare a CAPTURE or SYNC urb for submission to the bus. + */ +static inline void prepare_inbound_urb(struct snd_usb_endpoint *ep, + struct snd_urb_ctx *urb_ctx) +{ + int i, offs; + struct urb *urb = urb_ctx->urb; + + urb->dev = ep->chip->dev; /* we need to set this at each time */ + + switch (ep->type) { + case SND_USB_ENDPOINT_TYPE_DATA: + offs = 0; + for (i = 0; i < urb_ctx->packets; i++) { + urb->iso_frame_desc[i].offset = offs; + urb->iso_frame_desc[i].length = ep->curpacksize; + offs += ep->curpacksize; } + + urb->transfer_buffer_length = offs; + urb->number_of_packets = urb_ctx->packets; + break; + + case SND_USB_ENDPOINT_TYPE_SYNC: + urb->iso_frame_desc[0].length = min(4u, ep->syncmaxsize); + urb->iso_frame_desc[0].offset = 0; + break; } - return 0; } +/* + * Send output urbs that have been prepared previously. URBs are dequeued + * from ep->ready_playback_urbs and in case there there aren't any available + * or there are no packets that have been prepared, this function does + * nothing. + * + * The reason why the functionality of sending and preparing URBs is separated + * is that host controllers don't guarantee the order in which they return + * inbound and outbound packets to their submitters. + * + * This function is only used for implicit feedback endpoints. For endpoints + * driven by dedicated sync endpoints, URBs are immediately re-submitted + * from their completion handler. + */ +static void queue_pending_output_urbs(struct snd_usb_endpoint *ep) +{ + while (test_bit(EP_FLAG_RUNNING, &ep->flags)) { + + unsigned long flags; + struct snd_usb_packet_info *uninitialized_var(packet); + struct snd_urb_ctx *ctx = NULL; + struct urb *urb; + int err, i; + + spin_lock_irqsave(&ep->lock, flags); + if (ep->next_packet_read_pos != ep->next_packet_write_pos) { + packet = ep->next_packet + ep->next_packet_read_pos; + ep->next_packet_read_pos++; + ep->next_packet_read_pos %= MAX_URBS; + + /* take URB out of FIFO */ + if (!list_empty(&ep->ready_playback_urbs)) + ctx = list_first_entry(&ep->ready_playback_urbs, + struct snd_urb_ctx, ready_list); + } + spin_unlock_irqrestore(&ep->lock, flags); + + if (ctx == NULL) + return; + + list_del_init(&ctx->ready_list); + urb = ctx->urb; + + /* copy over the length information */ + for (i = 0; i < packet->packets; i++) + ctx->packet_size[i] = packet->packet_size[i]; + + /* call the data handler to fill in playback data */ + prepare_outbound_urb(ep, ctx); + + err = usb_submit_urb(ctx->urb, GFP_ATOMIC); + if (err < 0) + snd_printk(KERN_ERR "Unable to submit urb #%d: %d (urb %p)\n", + ctx->index, err, ctx->urb); + else + set_bit(ctx->index, &ep->active_mask); + } +} /* - * release a urb data + * complete callback for urbs */ -static void release_urb_ctx(struct snd_urb_ctx *u) +static void snd_complete_urb(struct urb *urb) +{ + struct snd_urb_ctx *ctx = urb->context; + struct snd_usb_endpoint *ep = ctx->ep; + int err; + + if (unlikely(urb->status == -ENOENT || /* unlinked */ + urb->status == -ENODEV || /* device removed */ + urb->status == -ECONNRESET || /* unlinked */ + urb->status == -ESHUTDOWN || /* device disabled */ + ep->chip->shutdown)) /* device disconnected */ + goto exit_clear; + + if (usb_pipeout(ep->pipe)) { + retire_outbound_urb(ep, ctx); + /* can be stopped during retire callback */ + if (unlikely(!test_bit(EP_FLAG_RUNNING, &ep->flags))) + goto exit_clear; + + if (snd_usb_endpoint_implict_feedback_sink(ep)) { + unsigned long flags; + + spin_lock_irqsave(&ep->lock, flags); + list_add_tail(&ctx->ready_list, &ep->ready_playback_urbs); + spin_unlock_irqrestore(&ep->lock, flags); + queue_pending_output_urbs(ep); + + goto exit_clear; + } + + prepare_outbound_urb_sizes(ep, ctx); + prepare_outbound_urb(ep, ctx); + } else { + retire_inbound_urb(ep, ctx); + /* can be stopped during retire callback */ + if (unlikely(!test_bit(EP_FLAG_RUNNING, &ep->flags))) + goto exit_clear; + + prepare_inbound_urb(ep, ctx); + } + + err = usb_submit_urb(urb, GFP_ATOMIC); + if (err == 0) + return; + + snd_printk(KERN_ERR "cannot submit urb (err = %d)\n", err); + //snd_pcm_stop(substream, SNDRV_PCM_STATE_XRUN); + +exit_clear: + clear_bit(ctx->index, &ep->active_mask); +} + +/** + * snd_usb_add_endpoint: Add an endpoint to an USB audio chip + * + * @chip: The chip + * @alts: The USB host interface + * @ep_num: The number of the endpoint to use + * @direction: SNDRV_PCM_STREAM_PLAYBACK or SNDRV_PCM_STREAM_CAPTURE + * @type: SND_USB_ENDPOINT_TYPE_DATA or SND_USB_ENDPOINT_TYPE_SYNC + * + * If the requested endpoint has not been added to the given chip before, + * a new instance is created. Otherwise, a pointer to the previoulsy + * created instance is returned. In case of any error, NULL is returned. + * + * New endpoints will be added to chip->ep_list and must be freed by + * calling snd_usb_endpoint_free(). + */ +struct snd_usb_endpoint *snd_usb_add_endpoint(struct snd_usb_audio *chip, + struct usb_host_interface *alts, + int ep_num, int direction, int type) { - if (u->urb) { - if (u->buffer_size) - usb_free_coherent(u->subs->dev, u->buffer_size, - u->urb->transfer_buffer, - u->urb->transfer_dma); - usb_free_urb(u->urb); - u->urb = NULL; + struct list_head *p; + struct snd_usb_endpoint *ep; + int ret, is_playback = direction == SNDRV_PCM_STREAM_PLAYBACK; + + mutex_lock(&chip->mutex); + + list_for_each(p, &chip->ep_list) { + ep = list_entry(p, struct snd_usb_endpoint, list); + if (ep->ep_num == ep_num && + ep->iface == alts->desc.bInterfaceNumber && + ep->alt_idx == alts->desc.bAlternateSetting) { + snd_printdd(KERN_DEBUG "Re-using EP %x in iface %d,%d @%p\n", + ep_num, ep->iface, ep->alt_idx, ep); + goto __exit_unlock; + } + } + + snd_printdd(KERN_DEBUG "Creating new %s %s endpoint #%x\n", + is_playback ? "playback" : "capture", + type == SND_USB_ENDPOINT_TYPE_DATA ? "data" : "sync", + ep_num); + + /* select the alt setting once so the endpoints become valid */ + ret = usb_set_interface(chip->dev, alts->desc.bInterfaceNumber, + alts->desc.bAlternateSetting); + if (ret < 0) { + snd_printk(KERN_ERR "%s(): usb_set_interface() failed, ret = %d\n", + __func__, ret); + ep = NULL; + goto __exit_unlock; } + + ep = kzalloc(sizeof(*ep), GFP_KERNEL); + if (!ep) + goto __exit_unlock; + + ep->chip = chip; + spin_lock_init(&ep->lock); + ep->type = type; + ep->ep_num = ep_num; + ep->iface = alts->desc.bInterfaceNumber; + ep->alt_idx = alts->desc.bAlternateSetting; + INIT_LIST_HEAD(&ep->ready_playback_urbs); + ep_num &= USB_ENDPOINT_NUMBER_MASK; + + if (is_playback) + ep->pipe = usb_sndisocpipe(chip->dev, ep_num); + else + ep->pipe = usb_rcvisocpipe(chip->dev, ep_num); + + if (type == SND_USB_ENDPOINT_TYPE_SYNC) { + if (get_endpoint(alts, 1)->bLength >= USB_DT_ENDPOINT_AUDIO_SIZE && + get_endpoint(alts, 1)->bRefresh >= 1 && + get_endpoint(alts, 1)->bRefresh <= 9) + ep->syncinterval = get_endpoint(alts, 1)->bRefresh; + else if (snd_usb_get_speed(chip->dev) == USB_SPEED_FULL) + ep->syncinterval = 1; + else if (get_endpoint(alts, 1)->bInterval >= 1 && + get_endpoint(alts, 1)->bInterval <= 16) + ep->syncinterval = get_endpoint(alts, 1)->bInterval - 1; + else + ep->syncinterval = 3; + + ep->syncmaxsize = le16_to_cpu(get_endpoint(alts, 1)->wMaxPacketSize); + } + + list_add_tail(&ep->list, &chip->ep_list); + +__exit_unlock: + mutex_unlock(&chip->mutex); + + return ep; } /* * wait until all urbs are processed. */ -static int wait_clear_urbs(struct snd_usb_substream *subs) +static int wait_clear_urbs(struct snd_usb_endpoint *ep) { unsigned long end_time = jiffies + msecs_to_jiffies(1000); unsigned int i; @@ -121,153 +497,148 @@ static int wait_clear_urbs(struct snd_usb_substream *subs) do { alive = 0; - for (i = 0; i < subs->nurbs; i++) { - if (test_bit(i, &subs->active_mask)) + for (i = 0; i < ep->nurbs; i++) + if (test_bit(i, &ep->active_mask)) alive++; - } - if (subs->syncpipe) { - for (i = 0; i < SYNC_URBS; i++) { - if (test_bit(i + 16, &subs->active_mask)) - alive++; - } - } - if (! alive) + + if (!alive) break; + schedule_timeout_uninterruptible(1); } while (time_before(jiffies, end_time)); + if (alive) - snd_printk(KERN_ERR "timeout: still %d active urbs..\n", alive); + snd_printk(KERN_ERR "timeout: still %d active urbs on EP #%x\n", + alive, ep->ep_num); + return 0; } /* - * release a substream + * unlink active urbs. */ -void snd_usb_release_substream_urbs(struct snd_usb_substream *subs, int force) +static int deactivate_urbs(struct snd_usb_endpoint *ep, int force, int can_sleep) { - int i; + unsigned int i; + int async; - /* stop urbs (to be sure) */ - deactivate_urbs(subs, force, 1); - wait_clear_urbs(subs); - - for (i = 0; i < MAX_URBS; i++) - release_urb_ctx(&subs->dataurb[i]); - for (i = 0; i < SYNC_URBS; i++) - release_urb_ctx(&subs->syncurb[i]); - usb_free_coherent(subs->dev, SYNC_URBS * 4, - subs->syncbuf, subs->sync_dma); - subs->syncbuf = NULL; - subs->nurbs = 0; -} + if (!force && ep->chip->shutdown) /* to be sure... */ + return -EBADFD; -/* - * complete callback from data urb - */ -static void snd_complete_urb(struct urb *urb) -{ - struct snd_urb_ctx *ctx = urb->context; - struct snd_usb_substream *subs = ctx->subs; - struct snd_pcm_substream *substream = ctx->subs->pcm_substream; - int err = 0; - - if ((subs->running && subs->ops.retire(subs, substream->runtime, urb)) || - !subs->running || /* can be stopped during retire callback */ - (err = subs->ops.prepare(subs, substream->runtime, urb)) < 0 || - (err = usb_submit_urb(urb, GFP_ATOMIC)) < 0) { - clear_bit(ctx->index, &subs->active_mask); - if (err < 0) { - snd_printd(KERN_ERR "cannot submit urb (err = %d)\n", err); - snd_pcm_stop(substream, SNDRV_PCM_STATE_XRUN); + async = !can_sleep && ep->chip->async_unlink; + + clear_bit(EP_FLAG_RUNNING, &ep->flags); + + INIT_LIST_HEAD(&ep->ready_playback_urbs); + ep->next_packet_read_pos = 0; + ep->next_packet_write_pos = 0; + + if (!async && in_interrupt()) + return 0; + + for (i = 0; i < ep->nurbs; i++) { + if (test_bit(i, &ep->active_mask)) { + if (!test_and_set_bit(i, &ep->unlink_mask)) { + struct urb *u = ep->urb[i].urb; + if (async) + usb_unlink_urb(u); + else + usb_kill_urb(u); + } } } -} + return 0; +} /* - * complete callback from sync urb + * release an endpoint's urbs */ -static void snd_complete_sync_urb(struct urb *urb) +static void release_urbs(struct snd_usb_endpoint *ep, int force) { - struct snd_urb_ctx *ctx = urb->context; - struct snd_usb_substream *subs = ctx->subs; - struct snd_pcm_substream *substream = ctx->subs->pcm_substream; - int err = 0; - - if ((subs->running && subs->ops.retire_sync(subs, substream->runtime, urb)) || - !subs->running || /* can be stopped during retire callback */ - (err = subs->ops.prepare_sync(subs, substream->runtime, urb)) < 0 || - (err = usb_submit_urb(urb, GFP_ATOMIC)) < 0) { - clear_bit(ctx->index + 16, &subs->active_mask); - if (err < 0) { - snd_printd(KERN_ERR "cannot submit sync urb (err = %d)\n", err); - snd_pcm_stop(substream, SNDRV_PCM_STATE_XRUN); - } - } -} + int i; + /* route incoming urbs to nirvana */ + ep->retire_data_urb = NULL; + ep->prepare_data_urb = NULL; + + /* stop urbs */ + deactivate_urbs(ep, force, 1); + wait_clear_urbs(ep); + + for (i = 0; i < ep->nurbs; i++) + release_urb_ctx(&ep->urb[i]); + + if (ep->syncbuf) + usb_free_coherent(ep->chip->dev, SYNC_URBS * 4, + ep->syncbuf, ep->sync_dma); + + ep->syncbuf = NULL; + ep->nurbs = 0; +} /* - * initialize a substream for plaback/capture + * configure a data endpoint */ -int snd_usb_init_substream_urbs(struct snd_usb_substream *subs, - unsigned int period_bytes, - unsigned int rate, - unsigned int frame_bits) +static int data_ep_set_params(struct snd_usb_endpoint *ep, + struct snd_pcm_hw_params *hw_params, + struct audioformat *fmt, + struct snd_usb_endpoint *sync_ep) { - unsigned int maxsize, i; - int is_playback = subs->direction == SNDRV_PCM_STREAM_PLAYBACK; - unsigned int urb_packs, total_packs, packs_per_ms; - struct snd_usb_audio *chip = subs->stream->chip; + unsigned int maxsize, i, urb_packs, total_packs, packs_per_ms; + int period_bytes = params_period_bytes(hw_params); + int format = params_format(hw_params); + int is_playback = usb_pipeout(ep->pipe); + int frame_bits = snd_pcm_format_physical_width(params_format(hw_params)) * + params_channels(hw_params); + + ep->datainterval = fmt->datainterval; + ep->stride = frame_bits >> 3; + ep->silence_value = format == SNDRV_PCM_FORMAT_U8 ? 0x80 : 0; - /* calculate the frequency in 16.16 format */ - if (snd_usb_get_speed(subs->dev) == USB_SPEED_FULL) - subs->freqn = get_usb_full_speed_rate(rate); - else - subs->freqn = get_usb_high_speed_rate(rate); - subs->freqm = subs->freqn; - subs->freqshift = INT_MIN; /* calculate max. frequency */ - if (subs->maxpacksize) { + if (ep->maxpacksize) { /* whatever fits into a max. size packet */ - maxsize = subs->maxpacksize; - subs->freqmax = (maxsize / (frame_bits >> 3)) - << (16 - subs->datainterval); + maxsize = ep->maxpacksize; + ep->freqmax = (maxsize / (frame_bits >> 3)) + << (16 - ep->datainterval); } else { /* no max. packet size: just take 25% higher than nominal */ - subs->freqmax = subs->freqn + (subs->freqn >> 2); - maxsize = ((subs->freqmax + 0xffff) * (frame_bits >> 3)) - >> (16 - subs->datainterval); + ep->freqmax = ep->freqn + (ep->freqn >> 2); + maxsize = ((ep->freqmax + 0xffff) * (frame_bits >> 3)) + >> (16 - ep->datainterval); } - subs->phase = 0; - if (subs->fill_max) - subs->curpacksize = subs->maxpacksize; + if (ep->fill_max) + ep->curpacksize = ep->maxpacksize; else - subs->curpacksize = maxsize; + ep->curpacksize = maxsize; - if (snd_usb_get_speed(subs->dev) != USB_SPEED_FULL) - packs_per_ms = 8 >> subs->datainterval; + if (snd_usb_get_speed(ep->chip->dev) != USB_SPEED_FULL) + packs_per_ms = 8 >> ep->datainterval; else packs_per_ms = 1; - if (is_playback) { - urb_packs = max(chip->nrpacks, 1); - urb_packs = min(urb_packs, (unsigned int)MAX_PACKS); - } else + if (is_playback && !snd_usb_endpoint_implict_feedback_sink(ep)) { + urb_packs = max(ep->chip->nrpacks, 1); + urb_packs = min(urb_packs, (unsigned int) MAX_PACKS); + } else { urb_packs = 1; + } + urb_packs *= packs_per_ms; - if (subs->syncpipe) - urb_packs = min(urb_packs, 1U << subs->syncinterval); + + if (sync_ep && !snd_usb_endpoint_implict_feedback_sink(ep)) + urb_packs = min(urb_packs, 1U << sync_ep->syncinterval); /* decide how many packets to be used */ - if (is_playback) { + if (is_playback && !snd_usb_endpoint_implict_feedback_sink(ep)) { unsigned int minsize, maxpacks; /* determine how small a packet can be */ - minsize = (subs->freqn >> (16 - subs->datainterval)) + minsize = (ep->freqn >> (16 - ep->datainterval)) * (frame_bits >> 3); /* with sync from device, assume it can be 12% lower */ - if (subs->syncpipe) + if (sync_ep) minsize -= minsize >> 3; minsize = max(minsize, 1u); total_packs = (period_bytes + minsize - 1) / minsize; @@ -284,284 +655,472 @@ int snd_usb_init_substream_urbs(struct snd_usb_substream *subs, urb_packs >>= 1; total_packs = MAX_URBS * urb_packs; } - subs->nurbs = (total_packs + urb_packs - 1) / urb_packs; - if (subs->nurbs > MAX_URBS) { + + ep->nurbs = (total_packs + urb_packs - 1) / urb_packs; + if (ep->nurbs > MAX_URBS) { /* too much... */ - subs->nurbs = MAX_URBS; + ep->nurbs = MAX_URBS; total_packs = MAX_URBS * urb_packs; - } else if (subs->nurbs < 2) { + } else if (ep->nurbs < 2) { /* too little - we need at least two packets * to ensure contiguous playback/capture */ - subs->nurbs = 2; + ep->nurbs = 2; } /* allocate and initialize data urbs */ - for (i = 0; i < subs->nurbs; i++) { - struct snd_urb_ctx *u = &subs->dataurb[i]; + for (i = 0; i < ep->nurbs; i++) { + struct snd_urb_ctx *u = &ep->urb[i]; u->index = i; - u->subs = subs; - u->packets = (i + 1) * total_packs / subs->nurbs - - i * total_packs / subs->nurbs; + u->ep = ep; + u->packets = (i + 1) * total_packs / ep->nurbs + - i * total_packs / ep->nurbs; u->buffer_size = maxsize * u->packets; - if (subs->fmt_type == UAC_FORMAT_TYPE_II) + + if (fmt->fmt_type == UAC_FORMAT_TYPE_II) u->packets++; /* for transfer delimiter */ u->urb = usb_alloc_urb(u->packets, GFP_KERNEL); if (!u->urb) goto out_of_memory; + u->urb->transfer_buffer = - usb_alloc_coherent(subs->dev, u->buffer_size, + usb_alloc_coherent(ep->chip->dev, u->buffer_size, GFP_KERNEL, &u->urb->transfer_dma); if (!u->urb->transfer_buffer) goto out_of_memory; - u->urb->pipe = subs->datapipe; + u->urb->pipe = ep->pipe; u->urb->transfer_flags = URB_ISO_ASAP | URB_NO_TRANSFER_DMA_MAP; - u->urb->interval = 1 << subs->datainterval; + u->urb->interval = 1 << ep->datainterval; u->urb->context = u; u->urb->complete = snd_complete_urb; + INIT_LIST_HEAD(&u->ready_list); } - if (subs->syncpipe) { - /* allocate and initialize sync urbs */ - subs->syncbuf = usb_alloc_coherent(subs->dev, SYNC_URBS * 4, - GFP_KERNEL, &subs->sync_dma); - if (!subs->syncbuf) - goto out_of_memory; - for (i = 0; i < SYNC_URBS; i++) { - struct snd_urb_ctx *u = &subs->syncurb[i]; - u->index = i; - u->subs = subs; - u->packets = 1; - u->urb = usb_alloc_urb(1, GFP_KERNEL); - if (!u->urb) - goto out_of_memory; - u->urb->transfer_buffer = subs->syncbuf + i * 4; - u->urb->transfer_dma = subs->sync_dma + i * 4; - u->urb->transfer_buffer_length = 4; - u->urb->pipe = subs->syncpipe; - u->urb->transfer_flags = URB_ISO_ASAP | - URB_NO_TRANSFER_DMA_MAP; - u->urb->number_of_packets = 1; - u->urb->interval = 1 << subs->syncinterval; - u->urb->context = u; - u->urb->complete = snd_complete_sync_urb; - } - } return 0; out_of_memory: - snd_usb_release_substream_urbs(subs, 0); + release_urbs(ep, 0); return -ENOMEM; } /* - * prepare urb for full speed capture sync pipe - * - * fill the length and offset of each urb descriptor. - * the fixed 10.14 frequency is passed through the pipe. + * configure a sync endpoint */ -static int prepare_capture_sync_urb(struct snd_usb_substream *subs, - struct snd_pcm_runtime *runtime, - struct urb *urb) +static int sync_ep_set_params(struct snd_usb_endpoint *ep, + struct snd_pcm_hw_params *hw_params, + struct audioformat *fmt) { - unsigned char *cp = urb->transfer_buffer; - struct snd_urb_ctx *ctx = urb->context; + int i; + + ep->syncbuf = usb_alloc_coherent(ep->chip->dev, SYNC_URBS * 4, + GFP_KERNEL, &ep->sync_dma); + if (!ep->syncbuf) + return -ENOMEM; + + for (i = 0; i < SYNC_URBS; i++) { + struct snd_urb_ctx *u = &ep->urb[i]; + u->index = i; + u->ep = ep; + u->packets = 1; + u->urb = usb_alloc_urb(1, GFP_KERNEL); + if (!u->urb) + goto out_of_memory; + u->urb->transfer_buffer = ep->syncbuf + i * 4; + u->urb->transfer_dma = ep->sync_dma + i * 4; + u->urb->transfer_buffer_length = 4; + u->urb->pipe = ep->pipe; + u->urb->transfer_flags = URB_ISO_ASAP | + URB_NO_TRANSFER_DMA_MAP; + u->urb->number_of_packets = 1; + u->urb->interval = 1 << ep->syncinterval; + u->urb->context = u; + u->urb->complete = snd_complete_urb; + } + + ep->nurbs = SYNC_URBS; - urb->dev = ctx->subs->dev; /* we need to set this at each time */ - urb->iso_frame_desc[0].length = 3; - urb->iso_frame_desc[0].offset = 0; - cp[0] = subs->freqn >> 2; - cp[1] = subs->freqn >> 10; - cp[2] = subs->freqn >> 18; return 0; + +out_of_memory: + release_urbs(ep, 0); + return -ENOMEM; } -/* - * prepare urb for high speed capture sync pipe +/** + * snd_usb_endpoint_set_params: configure an snd_usb_endpoint + * + * @ep: the snd_usb_endpoint to configure + * @hw_params: the hardware parameters + * @fmt: the USB audio format information + * @sync_ep: the sync endpoint to use, if any * - * fill the length and offset of each urb descriptor. - * the fixed 12.13 frequency is passed as 16.16 through the pipe. + * Determine the number of URBs to be used on this endpoint. + * An endpoint must be configured before it can be started. + * An endpoint that is already running can not be reconfigured. */ -static int prepare_capture_sync_urb_hs(struct snd_usb_substream *subs, - struct snd_pcm_runtime *runtime, - struct urb *urb) +int snd_usb_endpoint_set_params(struct snd_usb_endpoint *ep, + struct snd_pcm_hw_params *hw_params, + struct audioformat *fmt, + struct snd_usb_endpoint *sync_ep) { - unsigned char *cp = urb->transfer_buffer; - struct snd_urb_ctx *ctx = urb->context; + int err; - urb->dev = ctx->subs->dev; /* we need to set this at each time */ - urb->iso_frame_desc[0].length = 4; - urb->iso_frame_desc[0].offset = 0; - cp[0] = subs->freqn; - cp[1] = subs->freqn >> 8; - cp[2] = subs->freqn >> 16; - cp[3] = subs->freqn >> 24; - return 0; + if (ep->use_count != 0) { + snd_printk(KERN_WARNING "Unable to change format on ep #%x: already in use\n", + ep->ep_num); + return -EBUSY; + } + + /* release old buffers, if any */ + release_urbs(ep, 0); + + ep->datainterval = fmt->datainterval; + ep->maxpacksize = fmt->maxpacksize; + ep->fill_max = !!(fmt->attributes & UAC_EP_CS_ATTR_FILL_MAX); + + if (snd_usb_get_speed(ep->chip->dev) == USB_SPEED_FULL) + ep->freqn = get_usb_full_speed_rate(params_rate(hw_params)); + else + ep->freqn = get_usb_high_speed_rate(params_rate(hw_params)); + + /* calculate the frequency in 16.16 format */ + ep->freqm = ep->freqn; + ep->freqshift = INT_MIN; + + ep->phase = 0; + + switch (ep->type) { + case SND_USB_ENDPOINT_TYPE_DATA: + err = data_ep_set_params(ep, hw_params, fmt, sync_ep); + break; + case SND_USB_ENDPOINT_TYPE_SYNC: + err = sync_ep_set_params(ep, hw_params, fmt); + break; + default: + err = -EINVAL; + } + + snd_printdd(KERN_DEBUG "Setting params for ep #%x (type %d, %d urbs), ret=%d\n", + ep->ep_num, ep->type, ep->nurbs, err); + + return err; } -/* - * process after capture sync complete - * - nothing to do +/** + * snd_usb_endpoint_start: start an snd_usb_endpoint + * + * @ep: the endpoint to start + * + * A call to this function will increment the use count of the endpoint. + * In case it is not already running, the URBs for this endpoint will be + * submitted. Otherwise, this function does nothing. + * + * Must be balanced to calls of snd_usb_endpoint_stop(). + * + * Returns an error if the URB submission failed, 0 in all other cases. */ -static int retire_capture_sync_urb(struct snd_usb_substream *subs, - struct snd_pcm_runtime *runtime, - struct urb *urb) +int snd_usb_endpoint_start(struct snd_usb_endpoint *ep) { + int err; + unsigned int i; + + if (ep->chip->shutdown) + return -EBADFD; + + /* already running? */ + if (++ep->use_count != 1) + return 0; + + if (snd_BUG_ON(!test_bit(EP_FLAG_ACTIVATED, &ep->flags))) + return -EINVAL; + + /* just to be sure */ + deactivate_urbs(ep, 0, 1); + wait_clear_urbs(ep); + + ep->active_mask = 0; + ep->unlink_mask = 0; + ep->phase = 0; + + /* + * If this endpoint has a data endpoint as implicit feedback source, + * don't start the urbs here. Instead, mark them all as available, + * wait for the record urbs to return and queue the playback urbs + * from that context. + */ + + set_bit(EP_FLAG_RUNNING, &ep->flags); + + if (snd_usb_endpoint_implict_feedback_sink(ep)) { + for (i = 0; i < ep->nurbs; i++) { + struct snd_urb_ctx *ctx = ep->urb + i; + list_add_tail(&ctx->ready_list, &ep->ready_playback_urbs); + } + + return 0; + } + + for (i = 0; i < ep->nurbs; i++) { + struct urb *urb = ep->urb[i].urb; + + if (snd_BUG_ON(!urb)) + goto __error; + + if (usb_pipeout(ep->pipe)) { + prepare_outbound_urb_sizes(ep, urb->context); + prepare_outbound_urb(ep, urb->context); + } else { + prepare_inbound_urb(ep, urb->context); + } + + err = usb_submit_urb(urb, GFP_ATOMIC); + if (err < 0) { + snd_printk(KERN_ERR "cannot submit urb %d, error %d: %s\n", + i, err, usb_error_string(err)); + goto __error; + } + set_bit(i, &ep->active_mask); + } + return 0; + +__error: + clear_bit(EP_FLAG_RUNNING, &ep->flags); + ep->use_count--; + deactivate_urbs(ep, 0, 0); + return -EPIPE; } -/* - * prepare urb for capture data pipe +/** + * snd_usb_endpoint_stop: stop an snd_usb_endpoint + * + * @ep: the endpoint to stop (may be NULL) * - * fill the offset and length of each descriptor. + * A call to this function will decrement the use count of the endpoint. + * In case the last user has requested the endpoint stop, the URBs will + * actually be deactivated. * - * we use a temporary buffer to write the captured data. - * since the length of written data is determined by host, we cannot - * write onto the pcm buffer directly... the data is thus copied - * later at complete callback to the global buffer. + * Must be balanced to calls of snd_usb_endpoint_start(). */ -static int prepare_capture_urb(struct snd_usb_substream *subs, - struct snd_pcm_runtime *runtime, - struct urb *urb) +void snd_usb_endpoint_stop(struct snd_usb_endpoint *ep, + int force, int can_sleep, int wait) { - int i, offs; - struct snd_urb_ctx *ctx = urb->context; + if (!ep) + return; - offs = 0; - urb->dev = ctx->subs->dev; /* we need to set this at each time */ - for (i = 0; i < ctx->packets; i++) { - urb->iso_frame_desc[i].offset = offs; - urb->iso_frame_desc[i].length = subs->curpacksize; - offs += subs->curpacksize; + if (snd_BUG_ON(ep->use_count == 0)) + return; + + if (snd_BUG_ON(!test_bit(EP_FLAG_ACTIVATED, &ep->flags))) + return; + + if (--ep->use_count == 0) { + deactivate_urbs(ep, force, can_sleep); + ep->data_subs = NULL; + ep->sync_slave = NULL; + ep->retire_data_urb = NULL; + ep->prepare_data_urb = NULL; + + if (wait) + wait_clear_urbs(ep); } - urb->transfer_buffer_length = offs; - urb->number_of_packets = ctx->packets; - return 0; } -/* - * process after capture complete +/** + * snd_usb_endpoint_activate: activate an snd_usb_endpoint + * + * @ep: the endpoint to activate + * + * If the endpoint is not currently in use, this functions will select the + * correct alternate interface setting for the interface of this endpoint. * - * copy the data from each desctiptor to the pcm buffer, and - * update the current position. + * In case of any active users, this functions does nothing. + * + * Returns an error if usb_set_interface() failed, 0 in all other + * cases. */ -static int retire_capture_urb(struct snd_usb_substream *subs, - struct snd_pcm_runtime *runtime, - struct urb *urb) +int snd_usb_endpoint_activate(struct snd_usb_endpoint *ep) { - unsigned long flags; - unsigned char *cp; - int i; - unsigned int stride, frames, bytes, oldptr; - int period_elapsed = 0; + if (ep->use_count != 0) + return 0; - stride = runtime->frame_bits >> 3; + if (!ep->chip->shutdown && + !test_and_set_bit(EP_FLAG_ACTIVATED, &ep->flags)) { + int ret; - for (i = 0; i < urb->number_of_packets; i++) { - cp = (unsigned char *)urb->transfer_buffer + urb->iso_frame_desc[i].offset; - if (urb->iso_frame_desc[i].status && printk_ratelimit()) { - snd_printdd("frame %d active: %d\n", i, urb->iso_frame_desc[i].status); - // continue; - } - bytes = urb->iso_frame_desc[i].actual_length; - frames = bytes / stride; - if (!subs->txfr_quirk) - bytes = frames * stride; - if (bytes % (runtime->sample_bits >> 3) != 0) { -#ifdef CONFIG_SND_DEBUG_VERBOSE - int oldbytes = bytes; -#endif - bytes = frames * stride; - snd_printdd(KERN_ERR "Corrected urb data len. %d->%d\n", - oldbytes, bytes); - } - /* update the current pointer */ - spin_lock_irqsave(&subs->lock, flags); - oldptr = subs->hwptr_done; - subs->hwptr_done += bytes; - if (subs->hwptr_done >= runtime->buffer_size * stride) - subs->hwptr_done -= runtime->buffer_size * stride; - frames = (bytes + (oldptr % stride)) / stride; - subs->transfer_done += frames; - if (subs->transfer_done >= runtime->period_size) { - subs->transfer_done -= runtime->period_size; - period_elapsed = 1; - } - spin_unlock_irqrestore(&subs->lock, flags); - /* copy a data chunk */ - if (oldptr + bytes > runtime->buffer_size * stride) { - unsigned int bytes1 = - runtime->buffer_size * stride - oldptr; - memcpy(runtime->dma_area + oldptr, cp, bytes1); - memcpy(runtime->dma_area, cp + bytes1, bytes - bytes1); - } else { - memcpy(runtime->dma_area + oldptr, cp, bytes); + ret = usb_set_interface(ep->chip->dev, ep->iface, ep->alt_idx); + if (ret < 0) { + snd_printk(KERN_ERR "%s() usb_set_interface() failed, ret = %d\n", + __func__, ret); + clear_bit(EP_FLAG_ACTIVATED, &ep->flags); + return ret; } + + return 0; } - if (period_elapsed) - snd_pcm_period_elapsed(subs->pcm_substream); - return 0; + + return -EBUSY; } -/* - * Process after capture complete when paused. Nothing to do. +/** + * snd_usb_endpoint_deactivate: deactivate an snd_usb_endpoint + * + * @ep: the endpoint to deactivate + * + * If the endpoint is not currently in use, this functions will select the + * alternate interface setting 0 for the interface of this endpoint. + * + * In case of any active users, this functions does nothing. + * + * Returns an error if usb_set_interface() failed, 0 in all other + * cases. */ -static int retire_paused_capture_urb(struct snd_usb_substream *subs, - struct snd_pcm_runtime *runtime, - struct urb *urb) +int snd_usb_endpoint_deactivate(struct snd_usb_endpoint *ep) { - return 0; -} + if (!ep) + return -EINVAL; + if (ep->use_count != 0) + return 0; -/* - * prepare urb for playback sync pipe + if (!ep->chip->shutdown && + test_and_clear_bit(EP_FLAG_ACTIVATED, &ep->flags)) { + int ret; + + ret = usb_set_interface(ep->chip->dev, ep->iface, 0); + if (ret < 0) { + snd_printk(KERN_ERR "%s(): usb_set_interface() failed, ret = %d\n", + __func__, ret); + return ret; + } + + return 0; + } + + return -EBUSY; +} + +/** + * snd_usb_endpoint_free: Free the resources of an snd_usb_endpoint + * + * @ep: the list header of the endpoint to free * - * set up the offset and length to receive the current frequency. + * This function does not care for the endpoint's use count but will tear + * down all the streaming URBs immediately and free all resources. */ -static int prepare_playback_sync_urb(struct snd_usb_substream *subs, - struct snd_pcm_runtime *runtime, - struct urb *urb) +void snd_usb_endpoint_free(struct list_head *head) { - struct snd_urb_ctx *ctx = urb->context; + struct snd_usb_endpoint *ep; - urb->dev = ctx->subs->dev; /* we need to set this at each time */ - urb->iso_frame_desc[0].length = min(4u, ctx->subs->syncmaxsize); - urb->iso_frame_desc[0].offset = 0; - return 0; + ep = list_entry(head, struct snd_usb_endpoint, list); + release_urbs(ep, 1); + kfree(ep); } -/* - * process after playback sync complete - * - * Full speed devices report feedback values in 10.14 format as samples per - * frame, high speed devices in 16.16 format as samples per microframe. - * Because the Audio Class 1 spec was written before USB 2.0, many high speed - * devices use a wrong interpretation, some others use an entirely different - * format. Therefore, we cannot predict what format any particular device uses - * and must detect it automatically. +/** + * snd_usb_handle_sync_urb: parse an USB sync packet + * + * @ep: the endpoint to handle the packet + * @sender: the sending endpoint + * @urb: the received packet + * + * This function is called from the context of an endpoint that received + * the packet and is used to let another endpoint object handle the payload. */ -static int retire_playback_sync_urb(struct snd_usb_substream *subs, - struct snd_pcm_runtime *runtime, - struct urb *urb) +void snd_usb_handle_sync_urb(struct snd_usb_endpoint *ep, + struct snd_usb_endpoint *sender, + const struct urb *urb) { - unsigned int f; int shift; + unsigned int f; unsigned long flags; + snd_BUG_ON(ep == sender); + + /* + * In case the endpoint is operating in implicit feedback mode, prepare + * a new outbound URB that has the same layout as the received packet + * and add it to the list of pending urbs. queue_pending_output_urbs() + * will take care of them later. + */ + if (snd_usb_endpoint_implict_feedback_sink(ep) && + ep->use_count != 0) { + + /* implicit feedback case */ + int i, bytes = 0; + struct snd_urb_ctx *in_ctx; + struct snd_usb_packet_info *out_packet; + + in_ctx = urb->context; + + /* Count overall packet size */ + for (i = 0; i < in_ctx->packets; i++) + if (urb->iso_frame_desc[i].status == 0) + bytes += urb->iso_frame_desc[i].actual_length; + + /* + * skip empty packets. At least M-Audio's Fast Track Ultra stops + * streaming once it received a 0-byte OUT URB + */ + if (bytes == 0) + return; + + spin_lock_irqsave(&ep->lock, flags); + out_packet = ep->next_packet + ep->next_packet_write_pos; + + /* + * Iterate through the inbound packet and prepare the lengths + * for the output packet. The OUT packet we are about to send + * will have the same amount of payload bytes than the IN + * packet we just received. + */ + + out_packet->packets = in_ctx->packets; + for (i = 0; i < in_ctx->packets; i++) { + if (urb->iso_frame_desc[i].status == 0) + out_packet->packet_size[i] = + urb->iso_frame_desc[i].actual_length / ep->stride; + else + out_packet->packet_size[i] = 0; + } + + ep->next_packet_write_pos++; + ep->next_packet_write_pos %= MAX_URBS; + spin_unlock_irqrestore(&ep->lock, flags); + queue_pending_output_urbs(ep); + + return; + } + + /* + * process after playback sync complete + * + * Full speed devices report feedback values in 10.14 format as samples + * per frame, high speed devices in 16.16 format as samples per + * microframe. + * + * Because the Audio Class 1 spec was written before USB 2.0, many high + * speed devices use a wrong interpretation, some others use an + * entirely different format. + * + * Therefore, we cannot predict what format any particular device uses + * and must detect it automatically. + */ + if (urb->iso_frame_desc[0].status != 0 || urb->iso_frame_desc[0].actual_length < 3) - return 0; + return; f = le32_to_cpup(urb->transfer_buffer); if (urb->iso_frame_desc[0].actual_length == 3) f &= 0x00ffffff; else f &= 0x0fffffff; + if (f == 0) - return 0; + return; - if (unlikely(subs->freqshift == INT_MIN)) { + if (unlikely(ep->freqshift == INT_MIN)) { /* * The first time we see a feedback value, determine its format * by shifting it left or right until it matches the nominal @@ -569,398 +1128,34 @@ static int retire_playback_sync_urb(struct snd_usb_substream *subs, * differ from the nominal value more than +50% or -25%. */ shift = 0; - while (f < subs->freqn - subs->freqn / 4) { + while (f < ep->freqn - ep->freqn / 4) { f <<= 1; shift++; } - while (f > subs->freqn + subs->freqn / 2) { + while (f > ep->freqn + ep->freqn / 2) { f >>= 1; shift--; } - subs->freqshift = shift; - } - else if (subs->freqshift >= 0) - f <<= subs->freqshift; + ep->freqshift = shift; + } else if (ep->freqshift >= 0) + f <<= ep->freqshift; else - f >>= -subs->freqshift; + f >>= -ep->freqshift; - if (likely(f >= subs->freqn - subs->freqn / 8 && f <= subs->freqmax)) { + if (likely(f >= ep->freqn - ep->freqn / 8 && f <= ep->freqmax)) { /* * If the frequency looks valid, set it. * This value is referred to in prepare_playback_urb(). */ - spin_lock_irqsave(&subs->lock, flags); - subs->freqm = f; - spin_unlock_irqrestore(&subs->lock, flags); + spin_lock_irqsave(&ep->lock, flags); + ep->freqm = f; + spin_unlock_irqrestore(&ep->lock, flags); } else { /* * Out of range; maybe the shift value is wrong. * Reset it so that we autodetect again the next time. */ - subs->freqshift = INT_MIN; - } - - return 0; -} - -/* determine the number of frames in the next packet */ -static int snd_usb_audio_next_packet_size(struct snd_usb_substream *subs) -{ - if (subs->fill_max) - return subs->maxframesize; - else { - subs->phase = (subs->phase & 0xffff) - + (subs->freqm << subs->datainterval); - return min(subs->phase >> 16, subs->maxframesize); + ep->freqshift = INT_MIN; } } -/* - * Prepare urb for streaming before playback starts or when paused. - * - * We don't have any data, so we send silence. - */ -static int prepare_nodata_playback_urb(struct snd_usb_substream *subs, - struct snd_pcm_runtime *runtime, - struct urb *urb) -{ - unsigned int i, offs, counts; - struct snd_urb_ctx *ctx = urb->context; - int stride = runtime->frame_bits >> 3; - - offs = 0; - urb->dev = ctx->subs->dev; - for (i = 0; i < ctx->packets; ++i) { - counts = snd_usb_audio_next_packet_size(subs); - urb->iso_frame_desc[i].offset = offs * stride; - urb->iso_frame_desc[i].length = counts * stride; - offs += counts; - } - urb->number_of_packets = ctx->packets; - urb->transfer_buffer_length = offs * stride; - memset(urb->transfer_buffer, - runtime->format == SNDRV_PCM_FORMAT_U8 ? 0x80 : 0, - offs * stride); - return 0; -} - -/* - * prepare urb for playback data pipe - * - * Since a URB can handle only a single linear buffer, we must use double - * buffering when the data to be transferred overflows the buffer boundary. - * To avoid inconsistencies when updating hwptr_done, we use double buffering - * for all URBs. - */ -static int prepare_playback_urb(struct snd_usb_substream *subs, - struct snd_pcm_runtime *runtime, - struct urb *urb) -{ - int i, stride; - unsigned int counts, frames, bytes; - unsigned long flags; - int period_elapsed = 0; - struct snd_urb_ctx *ctx = urb->context; - - stride = runtime->frame_bits >> 3; - - frames = 0; - urb->dev = ctx->subs->dev; /* we need to set this at each time */ - urb->number_of_packets = 0; - spin_lock_irqsave(&subs->lock, flags); - for (i = 0; i < ctx->packets; i++) { - counts = snd_usb_audio_next_packet_size(subs); - /* set up descriptor */ - urb->iso_frame_desc[i].offset = frames * stride; - urb->iso_frame_desc[i].length = counts * stride; - frames += counts; - urb->number_of_packets++; - subs->transfer_done += counts; - if (subs->transfer_done >= runtime->period_size) { - subs->transfer_done -= runtime->period_size; - period_elapsed = 1; - if (subs->fmt_type == UAC_FORMAT_TYPE_II) { - if (subs->transfer_done > 0) { - /* FIXME: fill-max mode is not - * supported yet */ - frames -= subs->transfer_done; - counts -= subs->transfer_done; - urb->iso_frame_desc[i].length = - counts * stride; - subs->transfer_done = 0; - } - i++; - if (i < ctx->packets) { - /* add a transfer delimiter */ - urb->iso_frame_desc[i].offset = - frames * stride; - urb->iso_frame_desc[i].length = 0; - urb->number_of_packets++; - } - break; - } - } - if (period_elapsed) /* finish at the period boundary */ - break; - } - bytes = frames * stride; - if (subs->hwptr_done + bytes > runtime->buffer_size * stride) { - /* err, the transferred area goes over buffer boundary. */ - unsigned int bytes1 = - runtime->buffer_size * stride - subs->hwptr_done; - memcpy(urb->transfer_buffer, - runtime->dma_area + subs->hwptr_done, bytes1); - memcpy(urb->transfer_buffer + bytes1, - runtime->dma_area, bytes - bytes1); - } else { - memcpy(urb->transfer_buffer, - runtime->dma_area + subs->hwptr_done, bytes); - } - subs->hwptr_done += bytes; - if (subs->hwptr_done >= runtime->buffer_size * stride) - subs->hwptr_done -= runtime->buffer_size * stride; - - /* update delay with exact number of samples queued */ - runtime->delay = subs->last_delay; - runtime->delay += frames; - subs->last_delay = runtime->delay; - - /* realign last_frame_number */ - subs->last_frame_number = usb_get_current_frame_number(subs->dev); - subs->last_frame_number &= 0xFF; /* keep 8 LSBs */ - - spin_unlock_irqrestore(&subs->lock, flags); - urb->transfer_buffer_length = bytes; - if (period_elapsed) - snd_pcm_period_elapsed(subs->pcm_substream); - return 0; -} - -/* - * process after playback data complete - * - decrease the delay count again - */ -static int retire_playback_urb(struct snd_usb_substream *subs, - struct snd_pcm_runtime *runtime, - struct urb *urb) -{ - unsigned long flags; - int stride = runtime->frame_bits >> 3; - int processed = urb->transfer_buffer_length / stride; - int est_delay; - - spin_lock_irqsave(&subs->lock, flags); - - est_delay = snd_usb_pcm_delay(subs, runtime->rate); - /* update delay with exact number of samples played */ - if (processed > subs->last_delay) - subs->last_delay = 0; - else - subs->last_delay -= processed; - runtime->delay = subs->last_delay; - - /* - * Report when delay estimate is off by more than 2ms. - * The error should be lower than 2ms since the estimate relies - * on two reads of a counter updated every ms. - */ - if (abs(est_delay - subs->last_delay) * 1000 > runtime->rate * 2) - snd_printk(KERN_DEBUG "delay: estimated %d, actual %d\n", - est_delay, subs->last_delay); - - spin_unlock_irqrestore(&subs->lock, flags); - return 0; -} - -static const char *usb_error_string(int err) -{ - switch (err) { - case -ENODEV: - return "no device"; - case -ENOENT: - return "endpoint not enabled"; - case -EPIPE: - return "endpoint stalled"; - case -ENOSPC: - return "not enough bandwidth"; - case -ESHUTDOWN: - return "device disabled"; - case -EHOSTUNREACH: - return "device suspended"; - case -EINVAL: - case -EAGAIN: - case -EFBIG: - case -EMSGSIZE: - return "internal error"; - default: - return "unknown error"; - } -} - -/* - * set up and start data/sync urbs - */ -static int start_urbs(struct snd_usb_substream *subs, struct snd_pcm_runtime *runtime) -{ - unsigned int i; - int err; - - if (subs->stream->chip->shutdown) - return -EBADFD; - - for (i = 0; i < subs->nurbs; i++) { - if (snd_BUG_ON(!subs->dataurb[i].urb)) - return -EINVAL; - if (subs->ops.prepare(subs, runtime, subs->dataurb[i].urb) < 0) { - snd_printk(KERN_ERR "cannot prepare datapipe for urb %d\n", i); - goto __error; - } - } - if (subs->syncpipe) { - for (i = 0; i < SYNC_URBS; i++) { - if (snd_BUG_ON(!subs->syncurb[i].urb)) - return -EINVAL; - if (subs->ops.prepare_sync(subs, runtime, subs->syncurb[i].urb) < 0) { - snd_printk(KERN_ERR "cannot prepare syncpipe for urb %d\n", i); - goto __error; - } - } - } - - subs->active_mask = 0; - subs->unlink_mask = 0; - subs->running = 1; - for (i = 0; i < subs->nurbs; i++) { - err = usb_submit_urb(subs->dataurb[i].urb, GFP_ATOMIC); - if (err < 0) { - snd_printk(KERN_ERR "cannot submit datapipe " - "for urb %d, error %d: %s\n", - i, err, usb_error_string(err)); - goto __error; - } - set_bit(i, &subs->active_mask); - } - if (subs->syncpipe) { - for (i = 0; i < SYNC_URBS; i++) { - err = usb_submit_urb(subs->syncurb[i].urb, GFP_ATOMIC); - if (err < 0) { - snd_printk(KERN_ERR "cannot submit syncpipe " - "for urb %d, error %d: %s\n", - i, err, usb_error_string(err)); - goto __error; - } - set_bit(i + 16, &subs->active_mask); - } - } - return 0; - - __error: - // snd_pcm_stop(subs->pcm_substream, SNDRV_PCM_STATE_XRUN); - deactivate_urbs(subs, 0, 0); - return -EPIPE; -} - - -/* - */ -static struct snd_urb_ops audio_urb_ops[2] = { - { - .prepare = prepare_nodata_playback_urb, - .retire = retire_playback_urb, - .prepare_sync = prepare_playback_sync_urb, - .retire_sync = retire_playback_sync_urb, - }, - { - .prepare = prepare_capture_urb, - .retire = retire_capture_urb, - .prepare_sync = prepare_capture_sync_urb, - .retire_sync = retire_capture_sync_urb, - }, -}; - -/* - * initialize the substream instance. - */ - -void snd_usb_init_substream(struct snd_usb_stream *as, - int stream, struct audioformat *fp) -{ - struct snd_usb_substream *subs = &as->substream[stream]; - - INIT_LIST_HEAD(&subs->fmt_list); - spin_lock_init(&subs->lock); - - subs->stream = as; - subs->direction = stream; - subs->dev = as->chip->dev; - subs->txfr_quirk = as->chip->txfr_quirk; - subs->ops = audio_urb_ops[stream]; - if (snd_usb_get_speed(subs->dev) >= USB_SPEED_HIGH) - subs->ops.prepare_sync = prepare_capture_sync_urb_hs; - - snd_usb_set_pcm_ops(as->pcm, stream); - - list_add_tail(&fp->list, &subs->fmt_list); - subs->formats |= fp->formats; - subs->endpoint = fp->endpoint; - subs->num_formats++; - subs->fmt_type = fp->fmt_type; -} - -int snd_usb_substream_playback_trigger(struct snd_pcm_substream *substream, int cmd) -{ - struct snd_usb_substream *subs = substream->runtime->private_data; - - switch (cmd) { - case SNDRV_PCM_TRIGGER_START: - case SNDRV_PCM_TRIGGER_PAUSE_RELEASE: - subs->ops.prepare = prepare_playback_urb; - return 0; - case SNDRV_PCM_TRIGGER_STOP: - return deactivate_urbs(subs, 0, 0); - case SNDRV_PCM_TRIGGER_PAUSE_PUSH: - subs->ops.prepare = prepare_nodata_playback_urb; - return 0; - } - - return -EINVAL; -} - -int snd_usb_substream_capture_trigger(struct snd_pcm_substream *substream, int cmd) -{ - struct snd_usb_substream *subs = substream->runtime->private_data; - - switch (cmd) { - case SNDRV_PCM_TRIGGER_START: - subs->ops.retire = retire_capture_urb; - return start_urbs(subs, substream->runtime); - case SNDRV_PCM_TRIGGER_STOP: - return deactivate_urbs(subs, 0, 0); - case SNDRV_PCM_TRIGGER_PAUSE_PUSH: - subs->ops.retire = retire_paused_capture_urb; - return 0; - case SNDRV_PCM_TRIGGER_PAUSE_RELEASE: - subs->ops.retire = retire_capture_urb; - return 0; - } - - return -EINVAL; -} - -int snd_usb_substream_prepare(struct snd_usb_substream *subs, - struct snd_pcm_runtime *runtime) -{ - /* clear urbs (to be sure) */ - deactivate_urbs(subs, 0, 1); - wait_clear_urbs(subs); - - /* for playback, submit the URBs now; otherwise, the first hwptr_done - * updates for all URBs would happen at the same time when starting */ - if (subs->direction == SNDRV_PCM_STREAM_PLAYBACK) { - subs->ops.prepare = prepare_nodata_playback_urb; - return start_urbs(subs, runtime); - } - - return 0; -} - diff --git a/sound/usb/endpoint.h b/sound/usb/endpoint.h index 88eb63a636e..ee2723fb174 100644 --- a/sound/usb/endpoint.h +++ b/sound/usb/endpoint.h @@ -1,21 +1,29 @@ #ifndef __USBAUDIO_ENDPOINT_H #define __USBAUDIO_ENDPOINT_H -void snd_usb_init_substream(struct snd_usb_stream *as, - int stream, - struct audioformat *fp); +#define SND_USB_ENDPOINT_TYPE_DATA 0 +#define SND_USB_ENDPOINT_TYPE_SYNC 1 -int snd_usb_init_substream_urbs(struct snd_usb_substream *subs, - unsigned int period_bytes, - unsigned int rate, - unsigned int frame_bits); +struct snd_usb_endpoint *snd_usb_add_endpoint(struct snd_usb_audio *chip, + struct usb_host_interface *alts, + int ep_num, int direction, int type); -void snd_usb_release_substream_urbs(struct snd_usb_substream *subs, int force); +int snd_usb_endpoint_set_params(struct snd_usb_endpoint *ep, + struct snd_pcm_hw_params *hw_params, + struct audioformat *fmt, + struct snd_usb_endpoint *sync_ep); -int snd_usb_substream_prepare(struct snd_usb_substream *subs, - struct snd_pcm_runtime *runtime); +int snd_usb_endpoint_start(struct snd_usb_endpoint *ep); +void snd_usb_endpoint_stop(struct snd_usb_endpoint *ep, + int force, int can_sleep, int wait); +int snd_usb_endpoint_activate(struct snd_usb_endpoint *ep); +int snd_usb_endpoint_deactivate(struct snd_usb_endpoint *ep); +void snd_usb_endpoint_free(struct list_head *head); -int snd_usb_substream_playback_trigger(struct snd_pcm_substream *substream, int cmd); -int snd_usb_substream_capture_trigger(struct snd_pcm_substream *substream, int cmd); +int snd_usb_endpoint_implict_feedback_sink(struct snd_usb_endpoint *ep); + +void snd_usb_handle_sync_urb(struct snd_usb_endpoint *ep, + struct snd_usb_endpoint *sender, + const struct urb *urb); #endif /* __USBAUDIO_ENDPOINT_H */ diff --git a/sound/usb/mixer.c b/sound/usb/mixer.c index ab23869c01b..4f40ba82316 100644 --- a/sound/usb/mixer.c +++ b/sound/usb/mixer.c @@ -486,7 +486,7 @@ static int set_cur_mix_value(struct usb_mixer_elem_info *cval, int channel, /* * TLV callback for mixer volume controls */ -static int mixer_vol_tlv(struct snd_kcontrol *kcontrol, int op_flag, +int snd_usb_mixer_vol_tlv(struct snd_kcontrol *kcontrol, int op_flag, unsigned int size, unsigned int __user *_tlv) { struct usb_mixer_elem_info *cval = kcontrol->private_data; @@ -770,6 +770,26 @@ static void volume_control_quirks(struct usb_mixer_elem_info *cval, struct snd_kcontrol *kctl) { switch (cval->mixer->chip->usb_id) { + case USB_ID(0x0763, 0x2081): /* M-Audio Fast Track Ultra 8R */ + case USB_ID(0x0763, 0x2080): /* M-Audio Fast Track Ultra */ + if (strcmp(kctl->id.name, "Effect Duration") == 0) { + snd_printk(KERN_INFO + "usb-audio: set quirk for FTU Effect Duration\n"); + cval->min = 0x0000; + cval->max = 0x7f00; + cval->res = 0x0100; + break; + } + if (strcmp(kctl->id.name, "Effect Volume") == 0 || + strcmp(kctl->id.name, "Effect Feedback Volume") == 0) { + snd_printk(KERN_INFO + "usb-audio: set quirks for FTU Effect Feedback/Volume\n"); + cval->min = 0x00; + cval->max = 0x7f; + break; + } + break; + case USB_ID(0x0471, 0x0101): case USB_ID(0x0471, 0x0104): case USB_ID(0x0471, 0x0105): @@ -1121,9 +1141,6 @@ static void build_feature_ctl(struct mixer_build *state, void *raw_desc, len = snd_usb_copy_string_desc(state, nameid, kctl->id.name, sizeof(kctl->id.name)); - /* get min/max values */ - get_min_max_with_quirks(cval, 0, kctl); - switch (control) { case UAC_FU_MUTE: case UAC_FU_VOLUME: @@ -1155,17 +1172,7 @@ static void build_feature_ctl(struct mixer_build *state, void *raw_desc, } append_ctl_name(kctl, control == UAC_FU_MUTE ? " Switch" : " Volume"); - if (control == UAC_FU_VOLUME) { - check_mapped_dB(map, cval); - if (cval->dBmin < cval->dBmax || !cval->initialized) { - kctl->tlv.c = mixer_vol_tlv; - kctl->vd[0].access |= - SNDRV_CTL_ELEM_ACCESS_TLV_READ | - SNDRV_CTL_ELEM_ACCESS_TLV_CALLBACK; - } - } break; - default: if (! len) strlcpy(kctl->id.name, audio_feature_info[control-1].name, @@ -1173,6 +1180,19 @@ static void build_feature_ctl(struct mixer_build *state, void *raw_desc, break; } + /* get min/max values */ + get_min_max_with_quirks(cval, 0, kctl); + + if (control == UAC_FU_VOLUME) { + check_mapped_dB(map, cval); + if (cval->dBmin < cval->dBmax || !cval->initialized) { + kctl->tlv.c = snd_usb_mixer_vol_tlv; + kctl->vd[0].access |= + SNDRV_CTL_ELEM_ACCESS_TLV_READ | + SNDRV_CTL_ELEM_ACCESS_TLV_CALLBACK; + } + } + range = (cval->max - cval->min) / cval->res; /* Are there devices with volume range more than 255? I use a bit more * to be sure. 384 is a resolution magic number found on Logitech @@ -1388,7 +1408,7 @@ static int parse_audio_mixer_unit(struct mixer_build *state, int unitid, void *r for (pin = 0; pin < input_pins; pin++) { err = parse_audio_unit(state, desc->baSourceID[pin]); if (err < 0) - return err; + continue; err = check_input_term(state, desc->baSourceID[pin], &iterm); if (err < 0) return err; diff --git a/sound/usb/mixer.h b/sound/usb/mixer.h index 81b2d8a32fb..a7f3d45a8ac 100644 --- a/sound/usb/mixer.h +++ b/sound/usb/mixer.h @@ -68,4 +68,7 @@ int snd_usb_mixer_activate(struct usb_mixer_interface *mixer); int snd_usb_mixer_add_control(struct usb_mixer_interface *mixer, struct snd_kcontrol *kctl); +int snd_usb_mixer_vol_tlv(struct snd_kcontrol *kcontrol, int op_flag, + unsigned int size, unsigned int __user *_tlv); + #endif /* __USBMIXER_H */ diff --git a/sound/usb/mixer_maps.c b/sound/usb/mixer_maps.c index f1324c42383..41daaa24c25 100644 --- a/sound/usb/mixer_maps.c +++ b/sound/usb/mixer_maps.c @@ -288,6 +288,15 @@ static struct usbmix_name_map scratch_live_map[] = { { 0 } /* terminator */ }; +static struct usbmix_name_map ebox44_map[] = { + { 4, NULL }, /* FU */ + { 6, NULL }, /* MU */ + { 7, NULL }, /* FU */ + { 10, NULL }, /* FU */ + { 11, NULL }, /* MU */ + { 0 } +}; + /* "Gamesurround Muse Pocket LT" looks same like "Sound Blaster MP3+" * most importand difference is SU[8], it should be set to "Capture Source" * to make alsamixer and PA working properly. @@ -371,6 +380,10 @@ static struct usbmix_ctl_map usbmix_ctl_maps[] = { .map = scratch_live_map, .ignore_ctl_error = 1, }, + { + .id = USB_ID(0x200c, 0x1018), + .map = ebox44_map, + }, { 0 } /* terminator */ }; diff --git a/sound/usb/mixer_quirks.c b/sound/usb/mixer_quirks.c index ab125ee0b0f..41f4b691192 100644 --- a/sound/usb/mixer_quirks.c +++ b/sound/usb/mixer_quirks.c @@ -42,6 +42,77 @@ extern struct snd_kcontrol_new *snd_usb_feature_unit_ctl; +/* private_free callback */ +static void usb_mixer_elem_free(struct snd_kcontrol *kctl) +{ + kfree(kctl->private_data); + kctl->private_data = NULL; +} + +/* This function allows for the creation of standard UAC controls. + * See the quirks for M-Audio FTUs or Ebox-44. + * If you don't want to set a TLV callback pass NULL. + * + * Since there doesn't seem to be a devices that needs a multichannel + * version, we keep it mono for simplicity. + */ +static int snd_create_std_mono_ctl(struct usb_mixer_interface *mixer, + unsigned int unitid, + unsigned int control, + unsigned int cmask, + int val_type, + const char *name, + snd_kcontrol_tlv_rw_t *tlv_callback) +{ + int err; + struct usb_mixer_elem_info *cval; + struct snd_kcontrol *kctl; + + cval = kzalloc(sizeof(*cval), GFP_KERNEL); + if (!cval) + return -ENOMEM; + + cval->id = unitid; + cval->mixer = mixer; + cval->val_type = val_type; + cval->channels = 1; + cval->control = control; + cval->cmask = cmask; + + /* get_min_max() is called only for integer volumes later, + * so provide a short-cut for booleans */ + cval->min = 0; + cval->max = 1; + cval->res = 0; + cval->dBmin = 0; + cval->dBmax = 0; + + /* Create control */ + kctl = snd_ctl_new1(snd_usb_feature_unit_ctl, cval); + if (!kctl) { + kfree(cval); + return -ENOMEM; + } + + /* Set name */ + snprintf(kctl->id.name, sizeof(kctl->id.name), name); + kctl->private_free = usb_mixer_elem_free; + + /* set TLV */ + if (tlv_callback) { + kctl->tlv.c = tlv_callback; + kctl->vd[0].access |= + SNDRV_CTL_ELEM_ACCESS_TLV_READ | + SNDRV_CTL_ELEM_ACCESS_TLV_CALLBACK; + } + /* Add control to mixer */ + err = snd_usb_mixer_add_control(mixer, kctl); + if (err < 0) + return err; + + return 0; +} + /* * Sound Blaster remote control configuration * @@ -495,60 +566,218 @@ static int snd_nativeinstruments_create_mixer(struct usb_mixer_interface *mixer, } /* M-Audio FastTrack Ultra quirks */ +/* FTU Effect switch */ +struct snd_ftu_eff_switch_priv_val { + struct usb_mixer_interface *mixer; + int cached_value; + int is_cached; +}; -/* private_free callback */ -static void usb_mixer_elem_free(struct snd_kcontrol *kctl) +static int snd_ftu_eff_switch_info(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_info *uinfo) { - kfree(kctl->private_data); - kctl->private_data = NULL; + static const char *texts[8] = {"Room 1", + "Room 2", + "Room 3", + "Hall 1", + "Hall 2", + "Plate", + "Delay", + "Echo" + }; + + uinfo->type = SNDRV_CTL_ELEM_TYPE_ENUMERATED; + uinfo->count = 1; + uinfo->value.enumerated.items = 8; + if (uinfo->value.enumerated.item > 7) + uinfo->value.enumerated.item = 7; + strcpy(uinfo->value.enumerated.name, + texts[uinfo->value.enumerated.item]); + + return 0; } -static int snd_maudio_ftu_create_ctl(struct usb_mixer_interface *mixer, - int in, int out, const char *name) +static int snd_ftu_eff_switch_get(struct snd_kcontrol *kctl, + struct snd_ctl_elem_value *ucontrol) { - struct usb_mixer_elem_info *cval; + struct snd_usb_audio *chip; + struct usb_mixer_interface *mixer; + struct snd_ftu_eff_switch_priv_val *pval; + int err; + unsigned char value[2]; + + const int id = 6; + const int validx = 1; + const int val_len = 2; + + value[0] = 0x00; + value[1] = 0x00; + + pval = (struct snd_ftu_eff_switch_priv_val *) + kctl->private_value; + + if (pval->is_cached) { + ucontrol->value.enumerated.item[0] = pval->cached_value; + return 0; + } + + mixer = (struct usb_mixer_interface *) pval->mixer; + if (snd_BUG_ON(!mixer)) + return -EINVAL; + + chip = (struct snd_usb_audio *) mixer->chip; + if (snd_BUG_ON(!chip)) + return -EINVAL; + + + err = snd_usb_ctl_msg(chip->dev, + usb_rcvctrlpipe(chip->dev, 0), UAC_GET_CUR, + USB_RECIP_INTERFACE | USB_TYPE_CLASS | USB_DIR_IN, + validx << 8, snd_usb_ctrl_intf(chip) | (id << 8), + value, val_len); + if (err < 0) + return err; + + ucontrol->value.enumerated.item[0] = value[0]; + pval->cached_value = value[0]; + pval->is_cached = 1; + + return 0; +} + +static int snd_ftu_eff_switch_put(struct snd_kcontrol *kctl, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_usb_audio *chip; + struct snd_ftu_eff_switch_priv_val *pval; + + struct usb_mixer_interface *mixer; + int changed, cur_val, err, new_val; + unsigned char value[2]; + + + const int id = 6; + const int validx = 1; + const int val_len = 2; + + changed = 0; + + pval = (struct snd_ftu_eff_switch_priv_val *) + kctl->private_value; + cur_val = pval->cached_value; + new_val = ucontrol->value.enumerated.item[0]; + + mixer = (struct usb_mixer_interface *) pval->mixer; + if (snd_BUG_ON(!mixer)) + return -EINVAL; + + chip = (struct snd_usb_audio *) mixer->chip; + if (snd_BUG_ON(!chip)) + return -EINVAL; + + if (!pval->is_cached) { + /* Read current value */ + err = snd_usb_ctl_msg(chip->dev, + usb_rcvctrlpipe(chip->dev, 0), UAC_GET_CUR, + USB_RECIP_INTERFACE | USB_TYPE_CLASS | USB_DIR_IN, + validx << 8, snd_usb_ctrl_intf(chip) | (id << 8), + value, val_len); + if (err < 0) + return err; + + cur_val = value[0]; + pval->cached_value = cur_val; + pval->is_cached = 1; + } + /* update value if needed */ + if (cur_val != new_val) { + value[0] = new_val; + value[1] = 0; + err = snd_usb_ctl_msg(chip->dev, + usb_sndctrlpipe(chip->dev, 0), UAC_SET_CUR, + USB_RECIP_INTERFACE | USB_TYPE_CLASS | USB_DIR_OUT, + validx << 8, snd_usb_ctrl_intf(chip) | (id << 8), + value, val_len); + if (err < 0) + return err; + + pval->cached_value = new_val; + pval->is_cached = 1; + changed = 1; + } + + return changed; +} + +static int snd_ftu_create_effect_switch(struct usb_mixer_interface *mixer) +{ + static struct snd_kcontrol_new template = { + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + .name = "Effect Program Switch", + .index = 0, + .access = SNDRV_CTL_ELEM_ACCESS_READWRITE, + .info = snd_ftu_eff_switch_info, + .get = snd_ftu_eff_switch_get, + .put = snd_ftu_eff_switch_put + }; + + int err; struct snd_kcontrol *kctl; + struct snd_ftu_eff_switch_priv_val *pval; - cval = kzalloc(sizeof(*cval), GFP_KERNEL); - if (!cval) + pval = kzalloc(sizeof(*pval), GFP_KERNEL); + if (!pval) return -ENOMEM; - cval->id = 5; - cval->mixer = mixer; - cval->val_type = USB_MIXER_S16; - cval->channels = 1; - cval->control = out + 1; - cval->cmask = 1 << in; + pval->cached_value = 0; + pval->is_cached = 0; + pval->mixer = mixer; - kctl = snd_ctl_new1(snd_usb_feature_unit_ctl, cval); + template.private_value = (unsigned long) pval; + kctl = snd_ctl_new1(&template, mixer->chip); if (!kctl) { - kfree(cval); + kfree(pval); return -ENOMEM; } - snprintf(kctl->id.name, sizeof(kctl->id.name), name); - kctl->private_free = usb_mixer_elem_free; - return snd_usb_mixer_add_control(mixer, kctl); + err = snd_ctl_add(mixer->chip->card, kctl); + if (err < 0) + return err; + + return 0; } -static int snd_maudio_ftu_create_mixer(struct usb_mixer_interface *mixer) +/* Create volume controls for FTU devices*/ +static int snd_ftu_create_volume_ctls(struct usb_mixer_interface *mixer) { char name[64]; + unsigned int control, cmask; int in, out, err; + const unsigned int id = 5; + const int val_type = USB_MIXER_S16; + for (out = 0; out < 8; out++) { + control = out + 1; for (in = 0; in < 8; in++) { + cmask = 1 << in; snprintf(name, sizeof(name), - "AIn%d - Out%d Capture Volume", in + 1, out + 1); - err = snd_maudio_ftu_create_ctl(mixer, in, out, name); + "AIn%d - Out%d Capture Volume", + in + 1, out + 1); + err = snd_create_std_mono_ctl(mixer, id, control, + cmask, val_type, name, + &snd_usb_mixer_vol_tlv); if (err < 0) return err; } - for (in = 8; in < 16; in++) { + cmask = 1 << in; snprintf(name, sizeof(name), - "DIn%d - Out%d Playback Volume", in - 7, out + 1); - err = snd_maudio_ftu_create_ctl(mixer, in, out, name); + "DIn%d - Out%d Playback Volume", + in - 7, out + 1); + err = snd_create_std_mono_ctl(mixer, id, control, + cmask, val_type, name, + &snd_usb_mixer_vol_tlv); if (err < 0) return err; } @@ -557,6 +786,191 @@ static int snd_maudio_ftu_create_mixer(struct usb_mixer_interface *mixer) return 0; } +/* This control needs a volume quirk, see mixer.c */ +static int snd_ftu_create_effect_volume_ctl(struct usb_mixer_interface *mixer) +{ + static const char name[] = "Effect Volume"; + const unsigned int id = 6; + const int val_type = USB_MIXER_U8; + const unsigned int control = 2; + const unsigned int cmask = 0; + + return snd_create_std_mono_ctl(mixer, id, control, cmask, val_type, + name, snd_usb_mixer_vol_tlv); +} + +/* This control needs a volume quirk, see mixer.c */ +static int snd_ftu_create_effect_duration_ctl(struct usb_mixer_interface *mixer) +{ + static const char name[] = "Effect Duration"; + const unsigned int id = 6; + const int val_type = USB_MIXER_S16; + const unsigned int control = 3; + const unsigned int cmask = 0; + + return snd_create_std_mono_ctl(mixer, id, control, cmask, val_type, + name, snd_usb_mixer_vol_tlv); +} + +/* This control needs a volume quirk, see mixer.c */ +static int snd_ftu_create_effect_feedback_ctl(struct usb_mixer_interface *mixer) +{ + static const char name[] = "Effect Feedback Volume"; + const unsigned int id = 6; + const int val_type = USB_MIXER_U8; + const unsigned int control = 4; + const unsigned int cmask = 0; + + return snd_create_std_mono_ctl(mixer, id, control, cmask, val_type, + name, NULL); +} + +static int snd_ftu_create_effect_return_ctls(struct usb_mixer_interface *mixer) +{ + unsigned int cmask; + int err, ch; + char name[48]; + + const unsigned int id = 7; + const int val_type = USB_MIXER_S16; + const unsigned int control = 7; + + for (ch = 0; ch < 4; ++ch) { + cmask = 1 << ch; + snprintf(name, sizeof(name), + "Effect Return %d Volume", ch + 1); + err = snd_create_std_mono_ctl(mixer, id, control, + cmask, val_type, name, + snd_usb_mixer_vol_tlv); + if (err < 0) + return err; + } + + return 0; +} + +static int snd_ftu_create_effect_send_ctls(struct usb_mixer_interface *mixer) +{ + unsigned int cmask; + int err, ch; + char name[48]; + + const unsigned int id = 5; + const int val_type = USB_MIXER_S16; + const unsigned int control = 9; + + for (ch = 0; ch < 8; ++ch) { + cmask = 1 << ch; + snprintf(name, sizeof(name), + "Effect Send AIn%d Volume", ch + 1); + err = snd_create_std_mono_ctl(mixer, id, control, cmask, + val_type, name, + snd_usb_mixer_vol_tlv); + if (err < 0) + return err; + } + for (ch = 8; ch < 16; ++ch) { + cmask = 1 << ch; + snprintf(name, sizeof(name), + "Effect Send DIn%d Volume", ch - 7); + err = snd_create_std_mono_ctl(mixer, id, control, cmask, + val_type, name, + snd_usb_mixer_vol_tlv); + if (err < 0) + return err; + } + return 0; +} + +static int snd_ftu_create_mixer(struct usb_mixer_interface *mixer) +{ + int err; + + err = snd_ftu_create_volume_ctls(mixer); + if (err < 0) + return err; + + err = snd_ftu_create_effect_switch(mixer); + if (err < 0) + return err; + err = snd_ftu_create_effect_volume_ctl(mixer); + if (err < 0) + return err; + + err = snd_ftu_create_effect_duration_ctl(mixer); + if (err < 0) + return err; + + err = snd_ftu_create_effect_feedback_ctl(mixer); + if (err < 0) + return err; + + err = snd_ftu_create_effect_return_ctls(mixer); + if (err < 0) + return err; + + err = snd_ftu_create_effect_send_ctls(mixer); + if (err < 0) + return err; + + return 0; +} + + +/* + * Create mixer for Electrix Ebox-44 + * + * The mixer units from this device are corrupt, and even where they + * are valid they presents mono controls as L and R channels of + * stereo. So we create a good mixer in code. + */ + +static int snd_ebox44_create_mixer(struct usb_mixer_interface *mixer) +{ + int err; + + err = snd_create_std_mono_ctl(mixer, 4, 1, 0x0, USB_MIXER_INV_BOOLEAN, + "Headphone Playback Switch", NULL); + if (err < 0) + return err; + err = snd_create_std_mono_ctl(mixer, 4, 2, 0x1, USB_MIXER_S16, + "Headphone A Mix Playback Volume", NULL); + if (err < 0) + return err; + err = snd_create_std_mono_ctl(mixer, 4, 2, 0x2, USB_MIXER_S16, + "Headphone B Mix Playback Volume", NULL); + if (err < 0) + return err; + + err = snd_create_std_mono_ctl(mixer, 7, 1, 0x0, USB_MIXER_INV_BOOLEAN, + "Output Playback Switch", NULL); + if (err < 0) + return err; + err = snd_create_std_mono_ctl(mixer, 7, 2, 0x1, USB_MIXER_S16, + "Output A Playback Volume", NULL); + if (err < 0) + return err; + err = snd_create_std_mono_ctl(mixer, 7, 2, 0x2, USB_MIXER_S16, + "Output B Playback Volume", NULL); + if (err < 0) + return err; + + err = snd_create_std_mono_ctl(mixer, 10, 1, 0x0, USB_MIXER_INV_BOOLEAN, + "Input Capture Switch", NULL); + if (err < 0) + return err; + err = snd_create_std_mono_ctl(mixer, 10, 2, 0x1, USB_MIXER_S16, + "Input A Capture Volume", NULL); + if (err < 0) + return err; + err = snd_create_std_mono_ctl(mixer, 10, 2, 0x2, USB_MIXER_S16, + "Input B Capture Volume", NULL); + if (err < 0) + return err; + + return 0; +} + void snd_emuusb_set_samplerate(struct snd_usb_audio *chip, unsigned char samplerate_id) { @@ -600,7 +1014,7 @@ int snd_usb_mixer_apply_create_quirk(struct usb_mixer_interface *mixer) case USB_ID(0x0763, 0x2080): /* M-Audio Fast Track Ultra */ case USB_ID(0x0763, 0x2081): /* M-Audio Fast Track Ultra 8R */ - err = snd_maudio_ftu_create_mixer(mixer); + err = snd_ftu_create_mixer(mixer); break; case USB_ID(0x0b05, 0x1739): @@ -619,6 +1033,10 @@ int snd_usb_mixer_apply_create_quirk(struct usb_mixer_interface *mixer) snd_nativeinstruments_ta10_mixers, ARRAY_SIZE(snd_nativeinstruments_ta10_mixers)); break; + + case USB_ID(0x200c, 0x1018): /* Electrix Ebox-44 */ + err = snd_ebox44_create_mixer(mixer); + break; } return err; diff --git a/sound/usb/pcm.c b/sound/usb/pcm.c index 0eed6115c2d..24839d93264 100644 --- a/sound/usb/pcm.c +++ b/sound/usb/pcm.c @@ -16,6 +16,7 @@ #include <linux/init.h> #include <linux/slab.h> +#include <linux/ratelimit.h> #include <linux/usb.h> #include <linux/usb/audio.h> #include <linux/usb/audio-v2.h> @@ -34,6 +35,9 @@ #include "clock.h" #include "power.h" +#define SUBSTREAM_FLAG_DATA_EP_STARTED 0 +#define SUBSTREAM_FLAG_SYNC_EP_STARTED 1 + /* return the estimated delay based on USB frame counters */ snd_pcm_uframes_t snd_usb_pcm_delay(struct snd_usb_substream *subs, unsigned int rate) @@ -208,6 +212,84 @@ int snd_usb_init_pitch(struct snd_usb_audio *chip, int iface, } } +static int start_endpoints(struct snd_usb_substream *subs) +{ + int err; + + if (!subs->data_endpoint) + return -EINVAL; + + if (!test_and_set_bit(SUBSTREAM_FLAG_DATA_EP_STARTED, &subs->flags)) { + struct snd_usb_endpoint *ep = subs->data_endpoint; + + snd_printdd(KERN_DEBUG "Starting data EP @%p\n", ep); + + ep->data_subs = subs; + err = snd_usb_endpoint_start(ep); + if (err < 0) { + clear_bit(SUBSTREAM_FLAG_DATA_EP_STARTED, &subs->flags); + return err; + } + } + + if (subs->sync_endpoint && + !test_and_set_bit(SUBSTREAM_FLAG_SYNC_EP_STARTED, &subs->flags)) { + struct snd_usb_endpoint *ep = subs->sync_endpoint; + + snd_printdd(KERN_DEBUG "Starting sync EP @%p\n", ep); + + ep->sync_slave = subs->data_endpoint; + err = snd_usb_endpoint_start(ep); + if (err < 0) { + clear_bit(SUBSTREAM_FLAG_SYNC_EP_STARTED, &subs->flags); + return err; + } + } + + return 0; +} + +static void stop_endpoints(struct snd_usb_substream *subs, + int force, int can_sleep, int wait) +{ + if (test_and_clear_bit(SUBSTREAM_FLAG_SYNC_EP_STARTED, &subs->flags)) + snd_usb_endpoint_stop(subs->sync_endpoint, + force, can_sleep, wait); + + if (test_and_clear_bit(SUBSTREAM_FLAG_DATA_EP_STARTED, &subs->flags)) + snd_usb_endpoint_stop(subs->data_endpoint, + force, can_sleep, wait); +} + +static int activate_endpoints(struct snd_usb_substream *subs) +{ + if (subs->sync_endpoint) { + int ret; + + ret = snd_usb_endpoint_activate(subs->sync_endpoint); + if (ret < 0) + return ret; + } + + return snd_usb_endpoint_activate(subs->data_endpoint); +} + +static int deactivate_endpoints(struct snd_usb_substream *subs) +{ + int reta, retb; + + reta = snd_usb_endpoint_deactivate(subs->sync_endpoint); + retb = snd_usb_endpoint_deactivate(subs->data_endpoint); + + if (reta < 0) + return reta; + + if (retb < 0) + return retb; + + return 0; +} + /* * find a matching format and set up the interface */ @@ -219,7 +301,7 @@ static int set_format(struct snd_usb_substream *subs, struct audioformat *fmt) struct usb_interface *iface; unsigned int ep, attr; int is_playback = subs->direction == SNDRV_PCM_STREAM_PLAYBACK; - int err; + int err, implicit_fb = 0; iface = usb_ifnum_to_if(dev, fmt->iface); if (WARN_ON(!iface)) @@ -232,40 +314,11 @@ static int set_format(struct snd_usb_substream *subs, struct audioformat *fmt) if (fmt == subs->cur_audiofmt) return 0; - /* close the old interface */ - if (subs->interface >= 0 && subs->interface != fmt->iface) { - if (usb_set_interface(subs->dev, subs->interface, 0) < 0) { - snd_printk(KERN_ERR "%d:%d:%d: return to setting 0 failed\n", - dev->devnum, fmt->iface, fmt->altsetting); - return -EIO; - } - subs->interface = -1; - subs->altset_idx = 0; - } - - /* set interface */ - if (subs->interface != fmt->iface || subs->altset_idx != fmt->altset_idx) { - if (usb_set_interface(dev, fmt->iface, fmt->altsetting) < 0) { - snd_printk(KERN_ERR "%d:%d:%d: usb_set_interface failed\n", - dev->devnum, fmt->iface, fmt->altsetting); - return -EIO; - } - snd_printdd(KERN_INFO "setting usb interface %d:%d\n", fmt->iface, fmt->altsetting); - subs->interface = fmt->iface; - subs->altset_idx = fmt->altset_idx; - } - - /* create a data pipe */ - ep = fmt->endpoint & USB_ENDPOINT_NUMBER_MASK; - if (is_playback) - subs->datapipe = usb_sndisocpipe(dev, ep); - else - subs->datapipe = usb_rcvisocpipe(dev, ep); - subs->datainterval = fmt->datainterval; - subs->syncpipe = subs->syncinterval = 0; - subs->maxpacksize = fmt->maxpacksize; - subs->syncmaxsize = 0; - subs->fill_max = 0; + subs->data_endpoint = snd_usb_add_endpoint(subs->stream->chip, + alts, fmt->endpoint, subs->direction, + SND_USB_ENDPOINT_TYPE_DATA); + if (!subs->data_endpoint) + return -EINVAL; /* we need a sync pipe in async OUT or adaptive IN mode */ /* check the number of EP, since some devices have broken @@ -273,8 +326,25 @@ static int set_format(struct snd_usb_substream *subs, struct audioformat *fmt) * assume it as adaptive-out or sync-in. */ attr = fmt->ep_attr & USB_ENDPOINT_SYNCTYPE; + + switch (subs->stream->chip->usb_id) { + case USB_ID(0x0763, 0x2080): /* M-Audio FastTrack Ultra */ + case USB_ID(0x0763, 0x2081): + if (is_playback) { + implicit_fb = 1; + ep = 0x81; + iface = usb_ifnum_to_if(dev, 2); + + if (!iface || iface->num_altsetting == 0) + return -EINVAL; + + alts = &iface->altsetting[1]; + goto add_sync_ep; + } + } + if (((is_playback && attr == USB_ENDPOINT_SYNC_ASYNC) || - (! is_playback && attr == USB_ENDPOINT_SYNC_ADAPTIVE)) && + (!is_playback && attr == USB_ENDPOINT_SYNC_ADAPTIVE)) && altsd->bNumEndpoints >= 2) { /* check sync-pipe endpoint */ /* ... and check descriptor size before accessing bSynchAddress @@ -282,7 +352,8 @@ static int set_format(struct snd_usb_substream *subs, struct audioformat *fmt) the audio fields in the endpoint descriptors */ if ((get_endpoint(alts, 1)->bmAttributes & USB_ENDPOINT_XFERTYPE_MASK) != 0x01 || (get_endpoint(alts, 1)->bLength >= USB_DT_ENDPOINT_AUDIO_SIZE && - get_endpoint(alts, 1)->bSynchAddress != 0)) { + get_endpoint(alts, 1)->bSynchAddress != 0 && + !implicit_fb)) { snd_printk(KERN_ERR "%d:%d:%d : invalid synch pipe\n", dev->devnum, fmt->iface, fmt->altsetting); return -EINVAL; @@ -290,33 +361,27 @@ static int set_format(struct snd_usb_substream *subs, struct audioformat *fmt) ep = get_endpoint(alts, 1)->bEndpointAddress; if (get_endpoint(alts, 0)->bLength >= USB_DT_ENDPOINT_AUDIO_SIZE && (( is_playback && ep != (unsigned int)(get_endpoint(alts, 0)->bSynchAddress | USB_DIR_IN)) || - (!is_playback && ep != (unsigned int)(get_endpoint(alts, 0)->bSynchAddress & ~USB_DIR_IN)))) { + (!is_playback && ep != (unsigned int)(get_endpoint(alts, 0)->bSynchAddress & ~USB_DIR_IN)) || + ( is_playback && !implicit_fb))) { snd_printk(KERN_ERR "%d:%d:%d : invalid synch pipe\n", dev->devnum, fmt->iface, fmt->altsetting); return -EINVAL; } - ep &= USB_ENDPOINT_NUMBER_MASK; - if (is_playback) - subs->syncpipe = usb_rcvisocpipe(dev, ep); - else - subs->syncpipe = usb_sndisocpipe(dev, ep); - if (get_endpoint(alts, 1)->bLength >= USB_DT_ENDPOINT_AUDIO_SIZE && - get_endpoint(alts, 1)->bRefresh >= 1 && - get_endpoint(alts, 1)->bRefresh <= 9) - subs->syncinterval = get_endpoint(alts, 1)->bRefresh; - else if (snd_usb_get_speed(subs->dev) == USB_SPEED_FULL) - subs->syncinterval = 1; - else if (get_endpoint(alts, 1)->bInterval >= 1 && - get_endpoint(alts, 1)->bInterval <= 16) - subs->syncinterval = get_endpoint(alts, 1)->bInterval - 1; - else - subs->syncinterval = 3; - subs->syncmaxsize = le16_to_cpu(get_endpoint(alts, 1)->wMaxPacketSize); - } - - /* always fill max packet size */ - if (fmt->attributes & UAC_EP_CS_ATTR_FILL_MAX) - subs->fill_max = 1; + + implicit_fb = (get_endpoint(alts, 1)->bmAttributes & USB_ENDPOINT_USAGE_MASK) + == USB_ENDPOINT_USAGE_IMPLICIT_FB; + +add_sync_ep: + subs->sync_endpoint = snd_usb_add_endpoint(subs->stream->chip, + alts, ep, !subs->direction, + implicit_fb ? + SND_USB_ENDPOINT_TYPE_DATA : + SND_USB_ENDPOINT_TYPE_SYNC); + if (!subs->sync_endpoint) + return -EINVAL; + + subs->data_endpoint->sync_master = subs->sync_endpoint; + } if ((err = snd_usb_init_pitch(subs->stream->chip, subs->interface, alts, fmt)) < 0) return err; @@ -390,15 +455,30 @@ static int snd_usb_hw_params(struct snd_pcm_substream *substream, if (changed) { mutex_lock(&subs->stream->chip->shutdown_mutex); /* format changed */ - snd_usb_release_substream_urbs(subs, 0); - /* influenced: period_bytes, channels, rate, format, */ - ret = snd_usb_init_substream_urbs(subs, params_period_bytes(hw_params), - params_rate(hw_params), - snd_pcm_format_physical_width(params_format(hw_params)) * - params_channels(hw_params)); + stop_endpoints(subs, 0, 0, 0); + deactivate_endpoints(subs); + + ret = activate_endpoints(subs); + if (ret < 0) + goto unlock; + + ret = snd_usb_endpoint_set_params(subs->data_endpoint, hw_params, fmt, + subs->sync_endpoint); + if (ret < 0) + goto unlock; + + if (subs->sync_endpoint) + ret = snd_usb_endpoint_set_params(subs->sync_endpoint, + hw_params, fmt, NULL); +unlock: mutex_unlock(&subs->stream->chip->shutdown_mutex); } + if (ret == 0) { + subs->interface = fmt->iface; + subs->altset_idx = fmt->altset_idx; + } + return ret; } @@ -415,7 +495,7 @@ static int snd_usb_hw_free(struct snd_pcm_substream *substream) subs->cur_rate = 0; subs->period_bytes = 0; mutex_lock(&subs->stream->chip->shutdown_mutex); - snd_usb_release_substream_urbs(subs, 0); + stop_endpoints(subs, 0, 1, 1); mutex_unlock(&subs->stream->chip->shutdown_mutex); return snd_pcm_lib_free_vmalloc_buffer(substream); } @@ -435,19 +515,28 @@ static int snd_usb_pcm_prepare(struct snd_pcm_substream *substream) return -ENXIO; } + if (snd_BUG_ON(!subs->data_endpoint)) + return -EIO; + /* some unit conversions in runtime */ - subs->maxframesize = bytes_to_frames(runtime, subs->maxpacksize); - subs->curframesize = bytes_to_frames(runtime, subs->curpacksize); + subs->data_endpoint->maxframesize = + bytes_to_frames(runtime, subs->data_endpoint->maxpacksize); + subs->data_endpoint->curframesize = + bytes_to_frames(runtime, subs->data_endpoint->curpacksize); /* reset the pointer */ subs->hwptr_done = 0; subs->transfer_done = 0; - subs->phase = 0; subs->last_delay = 0; subs->last_frame_number = 0; runtime->delay = 0; - return snd_usb_substream_prepare(subs, runtime); + /* for playback, submit the URBs now; otherwise, the first hwptr_done + * updates for all URBs would happen at the same time when starting */ + if (subs->direction == SNDRV_PCM_STREAM_PLAYBACK) + return start_endpoints(subs); + + return 0; } static struct snd_pcm_hardware snd_usb_hardware = @@ -842,16 +931,171 @@ static int snd_usb_pcm_open(struct snd_pcm_substream *substream, int direction) static int snd_usb_pcm_close(struct snd_pcm_substream *substream, int direction) { + int ret; struct snd_usb_stream *as = snd_pcm_substream_chip(substream); struct snd_usb_substream *subs = &as->substream[direction]; - if (!as->chip->shutdown && subs->interface >= 0) { - usb_set_interface(subs->dev, subs->interface, 0); - subs->interface = -1; - } + stop_endpoints(subs, 0, 0, 0); + ret = deactivate_endpoints(subs); subs->pcm_substream = NULL; snd_usb_autosuspend(subs->stream->chip); - return 0; + + return ret; +} + +/* Since a URB can handle only a single linear buffer, we must use double + * buffering when the data to be transferred overflows the buffer boundary. + * To avoid inconsistencies when updating hwptr_done, we use double buffering + * for all URBs. + */ +static void retire_capture_urb(struct snd_usb_substream *subs, + struct urb *urb) +{ + struct snd_pcm_runtime *runtime = subs->pcm_substream->runtime; + unsigned int stride, frames, bytes, oldptr; + int i, period_elapsed = 0; + unsigned long flags; + unsigned char *cp; + + stride = runtime->frame_bits >> 3; + + for (i = 0; i < urb->number_of_packets; i++) { + cp = (unsigned char *)urb->transfer_buffer + urb->iso_frame_desc[i].offset; + if (urb->iso_frame_desc[i].status && printk_ratelimit()) { + snd_printdd(KERN_ERR "frame %d active: %d\n", i, urb->iso_frame_desc[i].status); + // continue; + } + bytes = urb->iso_frame_desc[i].actual_length; + frames = bytes / stride; + if (!subs->txfr_quirk) + bytes = frames * stride; + if (bytes % (runtime->sample_bits >> 3) != 0) { +#ifdef CONFIG_SND_DEBUG_VERBOSE + int oldbytes = bytes; +#endif + bytes = frames * stride; + snd_printdd(KERN_ERR "Corrected urb data len. %d->%d\n", + oldbytes, bytes); + } + /* update the current pointer */ + spin_lock_irqsave(&subs->lock, flags); + oldptr = subs->hwptr_done; + subs->hwptr_done += bytes; + if (subs->hwptr_done >= runtime->buffer_size * stride) + subs->hwptr_done -= runtime->buffer_size * stride; + frames = (bytes + (oldptr % stride)) / stride; + subs->transfer_done += frames; + if (subs->transfer_done >= runtime->period_size) { + subs->transfer_done -= runtime->period_size; + period_elapsed = 1; + } + spin_unlock_irqrestore(&subs->lock, flags); + /* copy a data chunk */ + if (oldptr + bytes > runtime->buffer_size * stride) { + unsigned int bytes1 = + runtime->buffer_size * stride - oldptr; + memcpy(runtime->dma_area + oldptr, cp, bytes1); + memcpy(runtime->dma_area, cp + bytes1, bytes - bytes1); + } else { + memcpy(runtime->dma_area + oldptr, cp, bytes); + } + } + + if (period_elapsed) + snd_pcm_period_elapsed(subs->pcm_substream); +} + +static void prepare_playback_urb(struct snd_usb_substream *subs, + struct urb *urb) +{ + struct snd_pcm_runtime *runtime = subs->pcm_substream->runtime; + struct snd_urb_ctx *ctx = urb->context; + unsigned int counts, frames, bytes; + int i, stride, period_elapsed = 0; + unsigned long flags; + + stride = runtime->frame_bits >> 3; + + frames = 0; + urb->number_of_packets = 0; + spin_lock_irqsave(&subs->lock, flags); + for (i = 0; i < ctx->packets; i++) { + counts = ctx->packet_size[i]; + /* set up descriptor */ + urb->iso_frame_desc[i].offset = frames * stride; + urb->iso_frame_desc[i].length = counts * stride; + frames += counts; + urb->number_of_packets++; + subs->transfer_done += counts; + if (subs->transfer_done >= runtime->period_size) { + subs->transfer_done -= runtime->period_size; + period_elapsed = 1; + if (subs->fmt_type == UAC_FORMAT_TYPE_II) { + if (subs->transfer_done > 0) { + /* FIXME: fill-max mode is not + * supported yet */ + frames -= subs->transfer_done; + counts -= subs->transfer_done; + urb->iso_frame_desc[i].length = + counts * stride; + subs->transfer_done = 0; + } + i++; + if (i < ctx->packets) { + /* add a transfer delimiter */ + urb->iso_frame_desc[i].offset = + frames * stride; + urb->iso_frame_desc[i].length = 0; + urb->number_of_packets++; + } + break; + } + } + if (period_elapsed && + !snd_usb_endpoint_implict_feedback_sink(subs->data_endpoint)) /* finish at the period boundary */ + break; + } + bytes = frames * stride; + if (subs->hwptr_done + bytes > runtime->buffer_size * stride) { + /* err, the transferred area goes over buffer boundary. */ + unsigned int bytes1 = + runtime->buffer_size * stride - subs->hwptr_done; + memcpy(urb->transfer_buffer, + runtime->dma_area + subs->hwptr_done, bytes1); + memcpy(urb->transfer_buffer + bytes1, + runtime->dma_area, bytes - bytes1); + } else { + memcpy(urb->transfer_buffer, + runtime->dma_area + subs->hwptr_done, bytes); + } + subs->hwptr_done += bytes; + if (subs->hwptr_done >= runtime->buffer_size * stride) + subs->hwptr_done -= runtime->buffer_size * stride; + runtime->delay += frames; + spin_unlock_irqrestore(&subs->lock, flags); + urb->transfer_buffer_length = bytes; + if (period_elapsed) + snd_pcm_period_elapsed(subs->pcm_substream); +} + +/* + * process after playback data complete + * - decrease the delay count again + */ +static void retire_playback_urb(struct snd_usb_substream *subs, + struct urb *urb) +{ + unsigned long flags; + struct snd_pcm_runtime *runtime = subs->pcm_substream->runtime; + int stride = runtime->frame_bits >> 3; + int processed = urb->transfer_buffer_length / stride; + + spin_lock_irqsave(&subs->lock, flags); + if (processed > runtime->delay) + runtime->delay = 0; + else + runtime->delay -= processed; + spin_unlock_irqrestore(&subs->lock, flags); } static int snd_usb_playback_open(struct snd_pcm_substream *substream) @@ -874,6 +1118,63 @@ static int snd_usb_capture_close(struct snd_pcm_substream *substream) return snd_usb_pcm_close(substream, SNDRV_PCM_STREAM_CAPTURE); } +static int snd_usb_substream_playback_trigger(struct snd_pcm_substream *substream, + int cmd) +{ + struct snd_usb_substream *subs = substream->runtime->private_data; + + switch (cmd) { + case SNDRV_PCM_TRIGGER_START: + case SNDRV_PCM_TRIGGER_PAUSE_RELEASE: + subs->data_endpoint->prepare_data_urb = prepare_playback_urb; + subs->data_endpoint->retire_data_urb = retire_playback_urb; + subs->running = 1; + return 0; + case SNDRV_PCM_TRIGGER_STOP: + stop_endpoints(subs, 0, 0, 0); + subs->running = 0; + return 0; + case SNDRV_PCM_TRIGGER_PAUSE_PUSH: + subs->data_endpoint->prepare_data_urb = NULL; + subs->data_endpoint->retire_data_urb = NULL; + subs->running = 0; + return 0; + } + + return -EINVAL; +} + +int snd_usb_substream_capture_trigger(struct snd_pcm_substream *substream, int cmd) +{ + int err; + struct snd_usb_substream *subs = substream->runtime->private_data; + + switch (cmd) { + case SNDRV_PCM_TRIGGER_START: + err = start_endpoints(subs); + if (err < 0) + return err; + + subs->data_endpoint->retire_data_urb = retire_capture_urb; + subs->running = 1; + return 0; + case SNDRV_PCM_TRIGGER_STOP: + stop_endpoints(subs, 0, 0, 0); + subs->running = 0; + return 0; + case SNDRV_PCM_TRIGGER_PAUSE_PUSH: + subs->data_endpoint->retire_data_urb = NULL; + subs->running = 0; + return 0; + case SNDRV_PCM_TRIGGER_PAUSE_RELEASE: + subs->data_endpoint->retire_data_urb = retire_capture_urb; + subs->running = 1; + return 0; + } + + return -EINVAL; +} + static struct snd_pcm_ops snd_usb_playback_ops = { .open = snd_usb_playback_open, .close = snd_usb_playback_close, diff --git a/sound/usb/proc.c b/sound/usb/proc.c index 961c9a25068..ebc1a5b5b3f 100644 --- a/sound/usb/proc.c +++ b/sound/usb/proc.c @@ -25,6 +25,7 @@ #include "usbaudio.h" #include "helper.h" #include "card.h" +#include "endpoint.h" #include "proc.h" /* convert our full speed USB rate into sampling rate in Hz */ @@ -115,28 +116,33 @@ static void proc_dump_substream_formats(struct snd_usb_substream *subs, struct s } } +static void proc_dump_ep_status(struct snd_usb_substream *subs, + struct snd_usb_endpoint *ep, + struct snd_info_buffer *buffer) +{ + if (!ep) + return; + snd_iprintf(buffer, " Packet Size = %d\n", ep->curpacksize); + snd_iprintf(buffer, " Momentary freq = %u Hz (%#x.%04x)\n", + snd_usb_get_speed(subs->dev) == USB_SPEED_FULL + ? get_full_speed_hz(ep->freqm) + : get_high_speed_hz(ep->freqm), + ep->freqm >> 16, ep->freqm & 0xffff); + if (ep->freqshift != INT_MIN) { + int res = 16 - ep->freqshift; + snd_iprintf(buffer, " Feedback Format = %d.%d\n", + (ep->syncmaxsize > 3 ? 32 : 24) - res, res); + } +} + static void proc_dump_substream_status(struct snd_usb_substream *subs, struct snd_info_buffer *buffer) { if (subs->running) { - unsigned int i; snd_iprintf(buffer, " Status: Running\n"); snd_iprintf(buffer, " Interface = %d\n", subs->interface); snd_iprintf(buffer, " Altset = %d\n", subs->altset_idx); - snd_iprintf(buffer, " URBs = %d [ ", subs->nurbs); - for (i = 0; i < subs->nurbs; i++) - snd_iprintf(buffer, "%d ", subs->dataurb[i].packets); - snd_iprintf(buffer, "]\n"); - snd_iprintf(buffer, " Packet Size = %d\n", subs->curpacksize); - snd_iprintf(buffer, " Momentary freq = %u Hz (%#x.%04x)\n", - snd_usb_get_speed(subs->dev) == USB_SPEED_FULL - ? get_full_speed_hz(subs->freqm) - : get_high_speed_hz(subs->freqm), - subs->freqm >> 16, subs->freqm & 0xffff); - if (subs->freqshift != INT_MIN) - snd_iprintf(buffer, " Feedback Format = %d.%d\n", - (subs->syncmaxsize > 3 ? 32 : 24) - - (16 - subs->freqshift), - 16 - subs->freqshift); + proc_dump_ep_status(subs, subs->data_endpoint, buffer); + proc_dump_ep_status(subs, subs->sync_endpoint, buffer); } else { snd_iprintf(buffer, " Status: Stop\n"); } diff --git a/sound/usb/stream.c b/sound/usb/stream.c index 5ff8010b2d6..6b7d7a2b7ba 100644 --- a/sound/usb/stream.c +++ b/sound/usb/stream.c @@ -73,6 +73,31 @@ static void snd_usb_audio_pcm_free(struct snd_pcm *pcm) } } +/* + * initialize the substream instance. + */ + +static void snd_usb_init_substream(struct snd_usb_stream *as, + int stream, + struct audioformat *fp) +{ + struct snd_usb_substream *subs = &as->substream[stream]; + + INIT_LIST_HEAD(&subs->fmt_list); + spin_lock_init(&subs->lock); + + subs->stream = as; + subs->direction = stream; + subs->dev = as->chip->dev; + subs->txfr_quirk = as->chip->txfr_quirk; + + snd_usb_set_pcm_ops(as->pcm, stream); + + list_add_tail(&fp->list, &subs->fmt_list); + subs->formats |= fp->formats; + subs->num_formats++; + subs->fmt_type = fp->fmt_type; +} /* * add this endpoint to the chip instance. @@ -94,9 +119,9 @@ int snd_usb_add_audio_stream(struct snd_usb_audio *chip, if (as->fmt_type != fp->fmt_type) continue; subs = &as->substream[stream]; - if (!subs->endpoint) + if (!subs->data_endpoint) continue; - if (subs->endpoint == fp->endpoint) { + if (subs->data_endpoint->ep_num == fp->endpoint) { list_add_tail(&fp->list, &subs->fmt_list); subs->num_formats++; subs->formats |= fp->formats; @@ -109,7 +134,7 @@ int snd_usb_add_audio_stream(struct snd_usb_audio *chip, if (as->fmt_type != fp->fmt_type) continue; subs = &as->substream[stream]; - if (subs->endpoint) + if (subs->data_endpoint) continue; err = snd_pcm_new_stream(as->pcm, stream, 1); if (err < 0) diff --git a/sound/usb/usbaudio.h b/sound/usb/usbaudio.h index 3e2b0357793..b8233ebe250 100644 --- a/sound/usb/usbaudio.h +++ b/sound/usb/usbaudio.h @@ -36,6 +36,7 @@ struct snd_usb_audio { struct snd_card *card; struct usb_interface *pm_intf; u32 usb_id; + struct mutex mutex; struct mutex shutdown_mutex; unsigned int shutdown:1; unsigned int probing:1; @@ -46,6 +47,7 @@ struct snd_usb_audio { int num_suspended_intf; struct list_head pcm_list; /* list of pcm streams */ + struct list_head ep_list; /* list of audio-related endpoints */ int pcm_devs; struct list_head midi_list; /* list of midi interfaces */ |