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-rw-r--r--sound/firewire/amdtp.c15
-rw-r--r--sound/pci/hda/hda_codec.h1
-rw-r--r--sound/pci/hda/hda_generic.c79
-rw-r--r--sound/pci/hda/hda_intel.c3
-rw-r--r--sound/pci/hda/patch_conexant.c23
-rw-r--r--sound/pci/hda/patch_realtek.c38
-rw-r--r--sound/pci/hda/patch_sigmatel.c3
-rw-r--r--sound/soc/atmel/sam9x5_wm8731.c2
-rw-r--r--sound/soc/codecs/wm5110.c25
-rw-r--r--sound/soc/codecs/wm8731.c4
-rw-r--r--sound/soc/codecs/wm8990.c2
-rw-r--r--sound/soc/fsl/pcm030-audio-fabric.c3
-rw-r--r--sound/soc/omap/n810.c4
-rw-r--r--sound/soc/sh/Kconfig1
-rw-r--r--sound/soc/soc-core.c4
-rw-r--r--sound/soc/soc-devres.c4
-rw-r--r--sound/soc/soc-pcm.c18
-rw-r--r--sound/usb/endpoint.c16
18 files changed, 165 insertions, 80 deletions
diff --git a/sound/firewire/amdtp.c b/sound/firewire/amdtp.c
index d3226892ad6b..9048777228e2 100644
--- a/sound/firewire/amdtp.c
+++ b/sound/firewire/amdtp.c
@@ -434,17 +434,14 @@ static void queue_out_packet(struct amdtp_out_stream *s, unsigned int cycle)
return;
index = s->packet_index;
+ /* this module generate empty packet for 'no data' */
syt = calculate_syt(s, cycle);
- if (!(s->flags & CIP_BLOCKING)) {
+ if (!(s->flags & CIP_BLOCKING))
data_blocks = calculate_data_blocks(s);
- } else {
- if (syt != 0xffff) {
- data_blocks = s->syt_interval;
- } else {
- data_blocks = 0;
- syt = 0xffffff;
- }
- }
+ else if (syt != 0xffff)
+ data_blocks = s->syt_interval;
+ else
+ data_blocks = 0;
buffer = s->buffer.packets[index].buffer;
buffer[0] = cpu_to_be32(ACCESS_ONCE(s->source_node_id_field) |
diff --git a/sound/pci/hda/hda_codec.h b/sound/pci/hda/hda_codec.h
index 77db69480c19..7aa9870040c1 100644
--- a/sound/pci/hda/hda_codec.h
+++ b/sound/pci/hda/hda_codec.h
@@ -698,7 +698,6 @@ struct hda_bus {
unsigned int in_reset:1; /* during reset operation */
unsigned int power_keep_link_on:1; /* don't power off HDA link */
unsigned int no_response_fallback:1; /* don't fallback at RIRB error */
- unsigned int avoid_link_reset:1; /* don't reset link at runtime PM */
int primary_dig_out_type; /* primary digital out PCM type */
};
diff --git a/sound/pci/hda/hda_generic.c b/sound/pci/hda/hda_generic.c
index 3067ed4fe3b2..c4671d00babd 100644
--- a/sound/pci/hda/hda_generic.c
+++ b/sound/pci/hda/hda_generic.c
@@ -2506,12 +2506,8 @@ static int create_out_jack_modes(struct hda_codec *codec, int num_pins,
for (i = 0; i < num_pins; i++) {
hda_nid_t pin = pins[i];
- if (pin == spec->hp_mic_pin) {
- int ret = create_hp_mic_jack_mode(codec, pin);
- if (ret < 0)
- return ret;
+ if (pin == spec->hp_mic_pin)
continue;
- }
if (get_out_jack_num_items(codec, pin) > 1) {
struct snd_kcontrol_new *knew;
char name[SNDRV_CTL_ELEM_ID_NAME_MAXLEN];
@@ -2764,7 +2760,7 @@ static int hp_mic_jack_mode_put(struct snd_kcontrol *kcontrol,
val &= ~(AC_PINCTL_VREFEN | PIN_HP);
val |= get_vref_idx(vref_caps, idx) | PIN_IN;
} else
- val = snd_hda_get_default_vref(codec, nid);
+ val = snd_hda_get_default_vref(codec, nid) | PIN_IN;
}
snd_hda_set_pin_ctl_cache(codec, nid, val);
call_hp_automute(codec, NULL);
@@ -2784,9 +2780,6 @@ static int create_hp_mic_jack_mode(struct hda_codec *codec, hda_nid_t pin)
struct hda_gen_spec *spec = codec->spec;
struct snd_kcontrol_new *knew;
- if (get_out_jack_num_items(codec, pin) <= 1 &&
- get_in_jack_num_items(codec, pin) <= 1)
- return 0; /* no need */
knew = snd_hda_gen_add_kctl(spec, "Headphone Mic Jack Mode",
&hp_mic_jack_mode_enum);
if (!knew)
@@ -2815,6 +2808,42 @@ static int add_loopback_list(struct hda_gen_spec *spec, hda_nid_t mix, int idx)
return 0;
}
+/* return true if either a volume or a mute amp is found for the given
+ * aamix path; the amp has to be either in the mixer node or its direct leaf
+ */
+static bool look_for_mix_leaf_ctls(struct hda_codec *codec, hda_nid_t mix_nid,
+ hda_nid_t pin, unsigned int *mix_val,
+ unsigned int *mute_val)
+{
+ int idx, num_conns;
+ const hda_nid_t *list;
+ hda_nid_t nid;
+
+ idx = snd_hda_get_conn_index(codec, mix_nid, pin, true);
+ if (idx < 0)
+ return false;
+
+ *mix_val = *mute_val = 0;
+ if (nid_has_volume(codec, mix_nid, HDA_INPUT))
+ *mix_val = HDA_COMPOSE_AMP_VAL(mix_nid, 3, idx, HDA_INPUT);
+ if (nid_has_mute(codec, mix_nid, HDA_INPUT))
+ *mute_val = HDA_COMPOSE_AMP_VAL(mix_nid, 3, idx, HDA_INPUT);
+ if (*mix_val && *mute_val)
+ return true;
+
+ /* check leaf node */
+ num_conns = snd_hda_get_conn_list(codec, mix_nid, &list);
+ if (num_conns < idx)
+ return false;
+ nid = list[idx];
+ if (!*mix_val && nid_has_volume(codec, nid, HDA_OUTPUT))
+ *mix_val = HDA_COMPOSE_AMP_VAL(nid, 3, 0, HDA_OUTPUT);
+ if (!*mute_val && nid_has_mute(codec, nid, HDA_OUTPUT))
+ *mute_val = HDA_COMPOSE_AMP_VAL(nid, 3, 0, HDA_OUTPUT);
+
+ return *mix_val || *mute_val;
+}
+
/* create input playback/capture controls for the given pin */
static int new_analog_input(struct hda_codec *codec, int input_idx,
hda_nid_t pin, const char *ctlname, int ctlidx,
@@ -2822,12 +2851,11 @@ static int new_analog_input(struct hda_codec *codec, int input_idx,
{
struct hda_gen_spec *spec = codec->spec;
struct nid_path *path;
- unsigned int val;
+ unsigned int mix_val, mute_val;
int err, idx;
- if (!nid_has_volume(codec, mix_nid, HDA_INPUT) &&
- !nid_has_mute(codec, mix_nid, HDA_INPUT))
- return 0; /* no need for analog loopback */
+ if (!look_for_mix_leaf_ctls(codec, mix_nid, pin, &mix_val, &mute_val))
+ return 0;
path = snd_hda_add_new_path(codec, pin, mix_nid, 0);
if (!path)
@@ -2836,20 +2864,18 @@ static int new_analog_input(struct hda_codec *codec, int input_idx,
spec->loopback_paths[input_idx] = snd_hda_get_path_idx(codec, path);
idx = path->idx[path->depth - 1];
- if (nid_has_volume(codec, mix_nid, HDA_INPUT)) {
- val = HDA_COMPOSE_AMP_VAL(mix_nid, 3, idx, HDA_INPUT);
- err = __add_pb_vol_ctrl(spec, HDA_CTL_WIDGET_VOL, ctlname, ctlidx, val);
+ if (mix_val) {
+ err = __add_pb_vol_ctrl(spec, HDA_CTL_WIDGET_VOL, ctlname, ctlidx, mix_val);
if (err < 0)
return err;
- path->ctls[NID_PATH_VOL_CTL] = val;
+ path->ctls[NID_PATH_VOL_CTL] = mix_val;
}
- if (nid_has_mute(codec, mix_nid, HDA_INPUT)) {
- val = HDA_COMPOSE_AMP_VAL(mix_nid, 3, idx, HDA_INPUT);
- err = __add_pb_sw_ctrl(spec, HDA_CTL_WIDGET_MUTE, ctlname, ctlidx, val);
+ if (mute_val) {
+ err = __add_pb_sw_ctrl(spec, HDA_CTL_WIDGET_MUTE, ctlname, ctlidx, mute_val);
if (err < 0)
return err;
- path->ctls[NID_PATH_MUTE_CTL] = val;
+ path->ctls[NID_PATH_MUTE_CTL] = mute_val;
}
path->active = true;
@@ -4383,6 +4409,17 @@ int snd_hda_gen_parse_auto_config(struct hda_codec *codec,
if (err < 0)
return err;
+ /* create "Headphone Mic Jack Mode" if no input selection is
+ * available (or user specifies add_jack_modes hint)
+ */
+ if (spec->hp_mic_pin &&
+ (spec->auto_mic || spec->input_mux.num_items == 1 ||
+ spec->add_jack_modes)) {
+ err = create_hp_mic_jack_mode(codec, spec->hp_mic_pin);
+ if (err < 0)
+ return err;
+ }
+
if (spec->add_jack_modes) {
if (cfg->line_out_type != AUTO_PIN_SPEAKER_OUT) {
err = create_out_jack_modes(codec, cfg->line_outs,
diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c
index 7a09404579a7..c6d230193da6 100644
--- a/sound/pci/hda/hda_intel.c
+++ b/sound/pci/hda/hda_intel.c
@@ -2994,8 +2994,7 @@ static int azx_runtime_suspend(struct device *dev)
STATESTS_INT_MASK);
azx_stop_chip(chip);
- if (!chip->bus->avoid_link_reset)
- azx_enter_link_reset(chip);
+ azx_enter_link_reset(chip);
azx_clear_irq_pending(chip);
if (chip->driver_caps & AZX_DCAPS_I915_POWERWELL)
hda_display_power(false);
diff --git a/sound/pci/hda/patch_conexant.c b/sound/pci/hda/patch_conexant.c
index c205bb1747fd..1f2717f817a0 100644
--- a/sound/pci/hda/patch_conexant.c
+++ b/sound/pci/hda/patch_conexant.c
@@ -3244,9 +3244,29 @@ enum {
#if IS_ENABLED(CONFIG_THINKPAD_ACPI)
#include <linux/thinkpad_acpi.h>
+#include <acpi/acpi.h>
static int (*led_set_func)(int, bool);
+static acpi_status acpi_check_cb(acpi_handle handle, u32 lvl, void *context,
+ void **rv)
+{
+ bool *found = context;
+ *found = true;
+ return AE_OK;
+}
+
+static bool is_thinkpad(struct hda_codec *codec)
+{
+ bool found = false;
+ if (codec->subsystem_id >> 16 != 0x17aa)
+ return false;
+ if (ACPI_SUCCESS(acpi_get_devices("LEN0068", acpi_check_cb, &found, NULL)) && found)
+ return true;
+ found = false;
+ return ACPI_SUCCESS(acpi_get_devices("IBM0068", acpi_check_cb, &found, NULL)) && found;
+}
+
static void update_tpacpi_mute_led(void *private_data, int enabled)
{
struct hda_codec *codec = private_data;
@@ -3279,6 +3299,8 @@ static void cxt_fixup_thinkpad_acpi(struct hda_codec *codec,
bool removefunc = false;
if (action == HDA_FIXUP_ACT_PROBE) {
+ if (!is_thinkpad(codec))
+ return;
if (!led_set_func)
led_set_func = symbol_request(tpacpi_led_set);
if (!led_set_func) {
@@ -3494,6 +3516,7 @@ static const struct snd_pci_quirk cxt5066_fixups[] = {
SND_PCI_QUIRK(0x17aa, 0x3975, "Lenovo U300s", CXT_FIXUP_STEREO_DMIC),
SND_PCI_QUIRK(0x17aa, 0x3977, "Lenovo IdeaPad U310", CXT_FIXUP_STEREO_DMIC),
SND_PCI_QUIRK(0x17aa, 0x397b, "Lenovo S205", CXT_FIXUP_STEREO_DMIC),
+ SND_PCI_QUIRK_VENDOR(0x17aa, "Thinkpad", CXT_FIXUP_THINKPAD_ACPI),
SND_PCI_QUIRK(0x1c06, 0x2011, "Lemote A1004", CXT_PINCFG_LEMOTE_A1004),
SND_PCI_QUIRK(0x1c06, 0x2012, "Lemote A1205", CXT_PINCFG_LEMOTE_A1205),
{}
diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c
index 5e42059f10a1..c770bdba6531 100644
--- a/sound/pci/hda/patch_realtek.c
+++ b/sound/pci/hda/patch_realtek.c
@@ -1782,6 +1782,8 @@ enum {
ALC889_FIXUP_IMAC91_VREF,
ALC882_FIXUP_INV_DMIC,
ALC882_FIXUP_NO_PRIMARY_HP,
+ ALC887_FIXUP_ASUS_BASS,
+ ALC887_FIXUP_BASS_CHMAP,
};
static void alc889_fixup_coef(struct hda_codec *codec,
@@ -1915,6 +1917,9 @@ static void alc882_fixup_no_primary_hp(struct hda_codec *codec,
}
}
+static void alc_fixup_bass_chmap(struct hda_codec *codec,
+ const struct hda_fixup *fix, int action);
+
static const struct hda_fixup alc882_fixups[] = {
[ALC882_FIXUP_ABIT_AW9D_MAX] = {
.type = HDA_FIXUP_PINS,
@@ -2105,6 +2110,19 @@ static const struct hda_fixup alc882_fixups[] = {
.type = HDA_FIXUP_FUNC,
.v.func = alc882_fixup_no_primary_hp,
},
+ [ALC887_FIXUP_ASUS_BASS] = {
+ .type = HDA_FIXUP_PINS,
+ .v.pins = (const struct hda_pintbl[]) {
+ {0x16, 0x99130130}, /* bass speaker */
+ {}
+ },
+ .chained = true,
+ .chain_id = ALC887_FIXUP_BASS_CHMAP,
+ },
+ [ALC887_FIXUP_BASS_CHMAP] = {
+ .type = HDA_FIXUP_FUNC,
+ .v.func = alc_fixup_bass_chmap,
+ },
};
static const struct snd_pci_quirk alc882_fixup_tbl[] = {
@@ -2138,6 +2156,7 @@ static const struct snd_pci_quirk alc882_fixup_tbl[] = {
SND_PCI_QUIRK(0x1043, 0x1873, "ASUS W90V", ALC882_FIXUP_ASUS_W90V),
SND_PCI_QUIRK(0x1043, 0x1971, "Asus W2JC", ALC882_FIXUP_ASUS_W2JC),
SND_PCI_QUIRK(0x1043, 0x835f, "Asus Eee 1601", ALC888_FIXUP_EEE1601),
+ SND_PCI_QUIRK(0x1043, 0x84bc, "ASUS ET2700", ALC887_FIXUP_ASUS_BASS),
SND_PCI_QUIRK(0x104d, 0x9047, "Sony Vaio TT", ALC889_FIXUP_VAIO_TT),
SND_PCI_QUIRK(0x104d, 0x905a, "Sony Vaio Z", ALC882_FIXUP_NO_PRIMARY_HP),
SND_PCI_QUIRK(0x104d, 0x9043, "Sony Vaio VGC-LN51JGB", ALC882_FIXUP_NO_PRIMARY_HP),
@@ -3798,6 +3817,7 @@ enum {
ALC271_FIXUP_HP_GATE_MIC_JACK,
ALC269_FIXUP_ACER_AC700,
ALC269_FIXUP_LIMIT_INT_MIC_BOOST,
+ ALC269VB_FIXUP_ASUS_ZENBOOK,
ALC269_FIXUP_LIMIT_INT_MIC_BOOST_MUTE_LED,
ALC269VB_FIXUP_ORDISSIMO_EVE2,
ALC283_FIXUP_CHROME_BOOK,
@@ -4075,6 +4095,12 @@ static const struct hda_fixup alc269_fixups[] = {
.chained = true,
.chain_id = ALC269_FIXUP_THINKPAD_ACPI,
},
+ [ALC269VB_FIXUP_ASUS_ZENBOOK] = {
+ .type = HDA_FIXUP_FUNC,
+ .v.func = alc269_fixup_limit_int_mic_boost,
+ .chained = true,
+ .chain_id = ALC269VB_FIXUP_DMIC,
+ },
[ALC269_FIXUP_LIMIT_INT_MIC_BOOST_MUTE_LED] = {
.type = HDA_FIXUP_FUNC,
.v.func = alc269_fixup_limit_int_mic_boost,
@@ -4189,8 +4215,8 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = {
SND_PCI_QUIRK(0x1043, 0x103f, "ASUS TX300", ALC282_FIXUP_ASUS_TX300),
SND_PCI_QUIRK(0x1043, 0x106d, "Asus K53BE", ALC269_FIXUP_LIMIT_INT_MIC_BOOST),
SND_PCI_QUIRK(0x1043, 0x115d, "Asus 1015E", ALC269_FIXUP_LIMIT_INT_MIC_BOOST),
- SND_PCI_QUIRK(0x1043, 0x1427, "Asus Zenbook UX31E", ALC269VB_FIXUP_DMIC),
- SND_PCI_QUIRK(0x1043, 0x1517, "Asus Zenbook UX31A", ALC269VB_FIXUP_DMIC),
+ SND_PCI_QUIRK(0x1043, 0x1427, "Asus Zenbook UX31E", ALC269VB_FIXUP_ASUS_ZENBOOK),
+ SND_PCI_QUIRK(0x1043, 0x1517, "Asus Zenbook UX31A", ALC269VB_FIXUP_ASUS_ZENBOOK),
SND_PCI_QUIRK(0x1043, 0x16e3, "ASUS UX50", ALC269_FIXUP_STEREO_DMIC),
SND_PCI_QUIRK(0x1043, 0x1a13, "Asus G73Jw", ALC269_FIXUP_ASUS_G73JW),
SND_PCI_QUIRK(0x1043, 0x1b13, "Asus U41SV", ALC269_FIXUP_INV_DMIC),
@@ -4715,7 +4741,7 @@ static const struct snd_pcm_chmap_elem asus_pcm_2_1_chmaps[] = {
};
/* override the 2.1 chmap */
-static void alc662_fixup_bass_chmap(struct hda_codec *codec,
+static void alc_fixup_bass_chmap(struct hda_codec *codec,
const struct hda_fixup *fix, int action)
{
if (action == HDA_FIXUP_ACT_BUILD) {
@@ -4923,7 +4949,7 @@ static const struct hda_fixup alc662_fixups[] = {
},
[ALC662_FIXUP_BASS_CHMAP] = {
.type = HDA_FIXUP_FUNC,
- .v.func = alc662_fixup_bass_chmap,
+ .v.func = alc_fixup_bass_chmap,
.chained = true,
.chain_id = ALC662_FIXUP_ASUS_MODE4
},
@@ -4936,7 +4962,7 @@ static const struct hda_fixup alc662_fixups[] = {
},
[ALC662_FIXUP_BASS_1A_CHMAP] = {
.type = HDA_FIXUP_FUNC,
- .v.func = alc662_fixup_bass_chmap,
+ .v.func = alc_fixup_bass_chmap,
.chained = true,
.chain_id = ALC662_FIXUP_BASS_1A,
},
@@ -5118,6 +5144,7 @@ static int patch_alc662(struct hda_codec *codec)
case 0x10ec0272:
case 0x10ec0663:
case 0x10ec0665:
+ case 0x10ec0668:
set_beep_amp(spec, 0x0b, 0x04, HDA_INPUT);
break;
case 0x10ec0273:
@@ -5175,6 +5202,7 @@ static int patch_alc680(struct hda_codec *codec)
*/
static const struct hda_codec_preset snd_hda_preset_realtek[] = {
{ .id = 0x10ec0221, .name = "ALC221", .patch = patch_alc269 },
+ { .id = 0x10ec0231, .name = "ALC231", .patch = patch_alc269 },
{ .id = 0x10ec0233, .name = "ALC233", .patch = patch_alc269 },
{ .id = 0x10ec0255, .name = "ALC255", .patch = patch_alc269 },
{ .id = 0x10ec0260, .name = "ALC260", .patch = patch_alc260 },
diff --git a/sound/pci/hda/patch_sigmatel.c b/sound/pci/hda/patch_sigmatel.c
index d2cc0041d9d3..088a5afbd1b9 100644
--- a/sound/pci/hda/patch_sigmatel.c
+++ b/sound/pci/hda/patch_sigmatel.c
@@ -2094,7 +2094,8 @@ static void stac92hd83xxx_fixup_hp_mic_led(struct hda_codec *codec,
if (action == HDA_FIXUP_ACT_PRE_PROBE) {
spec->mic_mute_led_gpio = 0x08; /* GPIO3 */
- codec->bus->avoid_link_reset = 1;
+ /* resetting controller clears GPIO, so we need to keep on */
+ codec->bus->power_keep_link_on = 1;
}
}
diff --git a/sound/soc/atmel/sam9x5_wm8731.c b/sound/soc/atmel/sam9x5_wm8731.c
index 992ae38d5a15..1b372283bd01 100644
--- a/sound/soc/atmel/sam9x5_wm8731.c
+++ b/sound/soc/atmel/sam9x5_wm8731.c
@@ -97,6 +97,8 @@ static int sam9x5_wm8731_driver_probe(struct platform_device *pdev)
goto out;
}
+ snd_soc_card_set_drvdata(card, priv);
+
card->dev = &pdev->dev;
card->owner = THIS_MODULE;
card->dai_link = dai;
diff --git a/sound/soc/codecs/wm5110.c b/sound/soc/codecs/wm5110.c
index c3c7396a6181..99b359e19d35 100644
--- a/sound/soc/codecs/wm5110.c
+++ b/sound/soc/codecs/wm5110.c
@@ -248,19 +248,6 @@ ARIZONA_MIXER_CONTROLS("SPKDAT1R", ARIZONA_OUT5RMIX_INPUT_1_SOURCE),
ARIZONA_MIXER_CONTROLS("SPKDAT2L", ARIZONA_OUT6LMIX_INPUT_1_SOURCE),
ARIZONA_MIXER_CONTROLS("SPKDAT2R", ARIZONA_OUT6RMIX_INPUT_1_SOURCE),
-SOC_SINGLE("HPOUT1 High Performance Switch", ARIZONA_OUTPUT_PATH_CONFIG_1L,
- ARIZONA_OUT1_OSR_SHIFT, 1, 0),
-SOC_SINGLE("HPOUT2 High Performance Switch", ARIZONA_OUTPUT_PATH_CONFIG_2L,
- ARIZONA_OUT2_OSR_SHIFT, 1, 0),
-SOC_SINGLE("HPOUT3 High Performance Switch", ARIZONA_OUTPUT_PATH_CONFIG_3L,
- ARIZONA_OUT3_OSR_SHIFT, 1, 0),
-SOC_SINGLE("Speaker High Performance Switch", ARIZONA_OUTPUT_PATH_CONFIG_4L,
- ARIZONA_OUT4_OSR_SHIFT, 1, 0),
-SOC_SINGLE("SPKDAT1 High Performance Switch", ARIZONA_OUTPUT_PATH_CONFIG_5L,
- ARIZONA_OUT5_OSR_SHIFT, 1, 0),
-SOC_SINGLE("SPKDAT2 High Performance Switch", ARIZONA_OUTPUT_PATH_CONFIG_6L,
- ARIZONA_OUT6_OSR_SHIFT, 1, 0),
-
SOC_DOUBLE_R("HPOUT1 Digital Switch", ARIZONA_DAC_DIGITAL_VOLUME_1L,
ARIZONA_DAC_DIGITAL_VOLUME_1R, ARIZONA_OUT1L_MUTE_SHIFT, 1, 1),
SOC_DOUBLE_R("HPOUT2 Digital Switch", ARIZONA_DAC_DIGITAL_VOLUME_2L,
@@ -293,18 +280,6 @@ SOC_DOUBLE_R_TLV("SPKDAT2 Digital Volume", ARIZONA_DAC_DIGITAL_VOLUME_6L,
ARIZONA_DAC_DIGITAL_VOLUME_6R, ARIZONA_OUT6L_VOL_SHIFT,
0xbf, 0, digital_tlv),
-SOC_DOUBLE_R_RANGE_TLV("HPOUT1 Volume", ARIZONA_OUTPUT_PATH_CONFIG_1L,
- ARIZONA_OUTPUT_PATH_CONFIG_1R,
- ARIZONA_OUT1L_PGA_VOL_SHIFT,
- 0x34, 0x40, 0, ana_tlv),
-SOC_DOUBLE_R_RANGE_TLV("HPOUT2 Volume", ARIZONA_OUTPUT_PATH_CONFIG_2L,
- ARIZONA_OUTPUT_PATH_CONFIG_2R,
- ARIZONA_OUT2L_PGA_VOL_SHIFT,
- 0x34, 0x40, 0, ana_tlv),
-SOC_DOUBLE_R_RANGE_TLV("HPOUT3 Volume", ARIZONA_OUTPUT_PATH_CONFIG_3L,
- ARIZONA_OUTPUT_PATH_CONFIG_3R,
- ARIZONA_OUT3L_PGA_VOL_SHIFT, 0x34, 0x40, 0, ana_tlv),
-
SOC_DOUBLE("SPKDAT1 Switch", ARIZONA_PDM_SPK1_CTRL_1, ARIZONA_SPK1L_MUTE_SHIFT,
ARIZONA_SPK1R_MUTE_SHIFT, 1, 1),
SOC_DOUBLE("SPKDAT2 Switch", ARIZONA_PDM_SPK2_CTRL_1, ARIZONA_SPK2L_MUTE_SHIFT,
diff --git a/sound/soc/codecs/wm8731.c b/sound/soc/codecs/wm8731.c
index 456bb8c6d759..bc7472c968e3 100644
--- a/sound/soc/codecs/wm8731.c
+++ b/sound/soc/codecs/wm8731.c
@@ -447,10 +447,10 @@ static int wm8731_set_dai_fmt(struct snd_soc_dai *codec_dai,
iface |= 0x0001;
break;
case SND_SOC_DAIFMT_DSP_A:
- iface |= 0x0003;
+ iface |= 0x0013;
break;
case SND_SOC_DAIFMT_DSP_B:
- iface |= 0x0013;
+ iface |= 0x0003;
break;
default:
return -EINVAL;
diff --git a/sound/soc/codecs/wm8990.c b/sound/soc/codecs/wm8990.c
index 253c88bb7a4c..4f05fb88bddf 100644
--- a/sound/soc/codecs/wm8990.c
+++ b/sound/soc/codecs/wm8990.c
@@ -1259,6 +1259,8 @@ static int wm8990_set_bias_level(struct snd_soc_codec *codec,
/* disable POBCTRL, SOFT_ST and BUFDCOPEN */
snd_soc_write(codec, WM8990_ANTIPOP2, 0x0);
+
+ codec->cache_sync = 1;
break;
}
diff --git a/sound/soc/fsl/pcm030-audio-fabric.c b/sound/soc/fsl/pcm030-audio-fabric.c
index eb4373840bb6..3665f612819d 100644
--- a/sound/soc/fsl/pcm030-audio-fabric.c
+++ b/sound/soc/fsl/pcm030-audio-fabric.c
@@ -69,7 +69,6 @@ static int pcm030_fabric_probe(struct platform_device *op)
return -ENOMEM;
card->dev = &op->dev;
- platform_set_drvdata(op, pdata);
pdata->card = card;
@@ -98,6 +97,8 @@ static int pcm030_fabric_probe(struct platform_device *op)
if (ret)
dev_err(&op->dev, "snd_soc_register_card() failed: %d\n", ret);
+ platform_set_drvdata(op, pdata);
+
return ret;
}
diff --git a/sound/soc/omap/n810.c b/sound/soc/omap/n810.c
index 6d216cb6c19b..3fde9e402710 100644
--- a/sound/soc/omap/n810.c
+++ b/sound/soc/omap/n810.c
@@ -100,12 +100,12 @@ static int n810_startup(struct snd_pcm_substream *substream)
SNDRV_PCM_HW_PARAM_CHANNELS, 2, 2);
n810_ext_control(&codec->dapm);
- return clk_enable(sys_clkout2);
+ return clk_prepare_enable(sys_clkout2);
}
static void n810_shutdown(struct snd_pcm_substream *substream)
{
- clk_disable(sys_clkout2);
+ clk_disable_unprepare(sys_clkout2);
}
static int n810_hw_params(struct snd_pcm_substream *substream,
diff --git a/sound/soc/sh/Kconfig b/sound/soc/sh/Kconfig
index 14011d90d70a..ff60e11ecb56 100644
--- a/sound/soc/sh/Kconfig
+++ b/sound/soc/sh/Kconfig
@@ -37,6 +37,7 @@ config SND_SOC_SH4_SIU
config SND_SOC_RCAR
tristate "R-Car series SRU/SCU/SSIU/SSI support"
select SND_SIMPLE_CARD
+ select REGMAP
help
This option enables R-Car SUR/SCU/SSIU/SSI sound support
diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c
index 4e53d87e881d..a66783e13a9c 100644
--- a/sound/soc/soc-core.c
+++ b/sound/soc/soc-core.c
@@ -3212,11 +3212,11 @@ int snd_soc_bytes_get(struct snd_kcontrol *kcontrol,
break;
case 2:
((u16 *)(&ucontrol->value.bytes.data))[0]
- &= ~params->mask;
+ &= cpu_to_be16(~params->mask);
break;
case 4:
((u32 *)(&ucontrol->value.bytes.data))[0]
- &= ~params->mask;
+ &= cpu_to_be32(~params->mask);
break;
default:
return -EINVAL;
diff --git a/sound/soc/soc-devres.c b/sound/soc/soc-devres.c
index b1d732255c02..3449c1e909ae 100644
--- a/sound/soc/soc-devres.c
+++ b/sound/soc/soc-devres.c
@@ -66,7 +66,7 @@ static void devm_card_release(struct device *dev, void *res)
*/
int devm_snd_soc_register_card(struct device *dev, struct snd_soc_card *card)
{
- struct device **ptr;
+ struct snd_soc_card **ptr;
int ret;
ptr = devres_alloc(devm_card_release, sizeof(*ptr), GFP_KERNEL);
@@ -75,7 +75,7 @@ int devm_snd_soc_register_card(struct device *dev, struct snd_soc_card *card)
ret = snd_soc_register_card(card);
if (ret == 0) {
- *ptr = dev;
+ *ptr = card;
devres_add(dev, ptr);
} else {
devres_free(ptr);
diff --git a/sound/soc/soc-pcm.c b/sound/soc/soc-pcm.c
index 42782c01e413..11a90cd027fa 100644
--- a/sound/soc/soc-pcm.c
+++ b/sound/soc/soc-pcm.c
@@ -148,12 +148,12 @@ static void soc_pcm_apply_msb(struct snd_pcm_substream *substream,
}
}
-static void soc_pcm_init_runtime_hw(struct snd_pcm_hardware *hw,
+static void soc_pcm_init_runtime_hw(struct snd_pcm_runtime *runtime,
struct snd_soc_pcm_stream *codec_stream,
struct snd_soc_pcm_stream *cpu_stream)
{
- hw->rate_min = max(codec_stream->rate_min, cpu_stream->rate_min);
- hw->rate_max = max(codec_stream->rate_max, cpu_stream->rate_max);
+ struct snd_pcm_hardware *hw = &runtime->hw;
+
hw->channels_min = max(codec_stream->channels_min,
cpu_stream->channels_min);
hw->channels_max = min(codec_stream->channels_max,
@@ -166,6 +166,13 @@ static void soc_pcm_init_runtime_hw(struct snd_pcm_hardware *hw,
if (cpu_stream->rates
& (SNDRV_PCM_RATE_KNOT | SNDRV_PCM_RATE_CONTINUOUS))
hw->rates |= codec_stream->rates;
+
+ snd_pcm_limit_hw_rates(runtime);
+
+ hw->rate_min = max(hw->rate_min, cpu_stream->rate_min);
+ hw->rate_min = max(hw->rate_min, codec_stream->rate_min);
+ hw->rate_max = min_not_zero(hw->rate_max, cpu_stream->rate_max);
+ hw->rate_max = min_not_zero(hw->rate_max, codec_stream->rate_max);
}
/*
@@ -235,15 +242,14 @@ static int soc_pcm_open(struct snd_pcm_substream *substream)
/* Check that the codec and cpu DAIs are compatible */
if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) {
- soc_pcm_init_runtime_hw(&runtime->hw, &codec_dai_drv->playback,
+ soc_pcm_init_runtime_hw(runtime, &codec_dai_drv->playback,
&cpu_dai_drv->playback);
} else {
- soc_pcm_init_runtime_hw(&runtime->hw, &codec_dai_drv->capture,
+ soc_pcm_init_runtime_hw(runtime, &codec_dai_drv->capture,
&cpu_dai_drv->capture);
}
ret = -EINVAL;
- snd_pcm_limit_hw_rates(runtime);
if (!runtime->hw.rates) {
printk(KERN_ERR "ASoC: %s <-> %s No matching rates\n",
codec_dai->name, cpu_dai->name);
diff --git a/sound/usb/endpoint.c b/sound/usb/endpoint.c
index b9ba0fcc45df..83aabea259d7 100644
--- a/sound/usb/endpoint.c
+++ b/sound/usb/endpoint.c
@@ -636,8 +636,22 @@ static int data_ep_set_params(struct snd_usb_endpoint *ep,
if (usb_pipein(ep->pipe) ||
snd_usb_endpoint_implicit_feedback_sink(ep)) {
+ urb_packs = packs_per_ms;
+ /*
+ * Wireless devices can poll at a max rate of once per 4ms.
+ * For dataintervals less than 5, increase the packet count to
+ * allow the host controller to use bursting to fill in the
+ * gaps.
+ */
+ if (snd_usb_get_speed(ep->chip->dev) == USB_SPEED_WIRELESS) {
+ int interval = ep->datainterval;
+ while (interval < 5) {
+ urb_packs <<= 1;
+ ++interval;
+ }
+ }
/* make capture URBs <= 1 ms and smaller than a period */
- urb_packs = min(max_packs_per_urb, packs_per_ms);
+ urb_packs = min(max_packs_per_urb, urb_packs);
while (urb_packs > 1 && urb_packs * maxsize >= period_bytes)
urb_packs >>= 1;
ep->nurbs = MAX_URBS;
generated by cgit v1.2.3 (git 2.25.1) at 2024-11-23 07:30:25 +0000