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2013-04-19vm: convert snd_pcm_lib_mmap_iomem() to vm_iomap_memory() helperLinus Torvalds1-10/+2
This is my example conversion of a few existing mmap users. The pcm mmap case is one of the more straightforward ones. Acked-by: Takashi Iwai <tiwai@suse.de> Signed-off-by: Linus Torvalds <torvalds@linux-foundation.org>
2013-04-12Merge tag 'asoc-v3.9-rc6' of ↵Takashi Iwai6-33/+28
git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound into for-linus ASoC: Updates for v3.9 A few updates, more than I'd like, fixing some relatively small issues but mostly driver specific ones. Nothing wildly exciting so if it doesn't make v3.9 it won't be the end of the world but it'd be nice.
2013-04-11Merge remote-tracking branch 'asoc/fix/wm8903' into tmpMark Brown1-0/+2
2013-04-11Merge remote-tracking branch 'asoc/fix/tegra' into tmpMark Brown1-23/+1
2013-04-11Merge remote-tracking branch 'asoc/fix/samsung' into tmpMark Brown1-5/+12
2013-04-11Merge remote-tracking branch 'asoc/fix/core' into tmpMark Brown1-1/+1
2013-04-11Merge remote-tracking branch 'asoc/fix/compress' into tmpMark Brown1-3/+11
2013-04-09ASoC: wm5102: Correct lookup of arizona struct in SYSCLK eventMark Brown1-1/+1
Reported-by: Ryo Tsutsui <Ryo.Tsutsui@wolfsonmicro.com> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com> Cc: stable@vger.kernel.org
2013-04-09ASoC: wm8903: Fix the bypass to HP/LINEOUT when no DAC or ADC is runningAlban Bedel1-0/+2
The Charge Pump needs the DSP clock to work properly, without it the bypass to HP/LINEOUT is not working properly. This requirement is not mentioned in the datasheet but has been confirmed by Mark Brown from Wolfson. Signed-off-by: Alban Bedel <alban.bedel@avionic-design.de> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com> Cc: stable@vger.kernel.org
2013-04-07ALSA: usb-audio: fix endianness bug in snd_nativeinstruments_*Eldad Zack2-3/+3
The usb_control_msg() function expects __u16 types and performs the endianness conversions by itself. However, in three places, a conversion is performed before it is handed over to usb_control_msg(), which leads to a double conversion (= no conversion): * snd_usb_nativeinstruments_boot_quirk() * snd_nativeinstruments_control_get() * snd_nativeinstruments_control_put() Caught by sparse: sound/usb/mixer_quirks.c:512:38: warning: incorrect type in argument 6 (different base types) sound/usb/mixer_quirks.c:512:38: expected unsigned short [unsigned] [usertype] index sound/usb/mixer_quirks.c:512:38: got restricted __le16 [usertype] <noident> sound/usb/mixer_quirks.c:543:35: warning: incorrect type in argument 5 (different base types) sound/usb/mixer_quirks.c:543:35: expected unsigned short [unsigned] [usertype] value sound/usb/mixer_quirks.c:543:35: got restricted __le16 [usertype] <noident> sound/usb/mixer_quirks.c:543:56: warning: incorrect type in argument 6 (different base types) sound/usb/mixer_quirks.c:543:56: expected unsigned short [unsigned] [usertype] index sound/usb/mixer_quirks.c:543:56: got restricted __le16 [usertype] <noident> sound/usb/quirks.c:502:35: warning: incorrect type in argument 5 (different base types) sound/usb/quirks.c:502:35: expected unsigned short [unsigned] [usertype] value sound/usb/quirks.c:502:35: got restricted __le16 [usertype] <noident> Signed-off-by: Eldad Zack <eldad@fogrefinery.com> Acked-by: Daniel Mack <zonque@gmail.com> Cc: <stable@vger.kernel.org> Signed-off-by: Takashi Iwai <tiwai@suse.de>
2013-04-05ALSA: hda/generic - fix uninitialized variableJiri Slaby1-1/+1
changed is not initialized in path_power_down_sync, but it is expected to be false in case no change happened in the loop. So set it to false. Signed-off-by: Jiri Slaby <jslaby@suse.cz> Signed-off-by: Takashi Iwai <tiwai@suse.de>
2013-04-04Revert "ALSA: hda - Allow power_save_controller option override DCAPS"Takashi Iwai1-4/+2
This reverts commit 6ab317419c62850a71e2adfd1573e5ee87d8774f. The commit [6ab317419c: ALSA: hda - Allow power_save_controller option override DCAPS] changed the behavior of power_save_controller so that it can override the driver capability. This assumed that this option is rarely changed dynamically unlike power_save option. Too naive. It turned out that the user-space power-management tool tries to set power_save_controller option to 1 together with power_save option without knowing what's actually doing. This enabled forcibly the runtime PM of the controller, which is known to be broken om many chips thus disabled as default. So, the only sane fix is to revert this commit again. It was intended to ease debugging/testing for runtime PM enablement, but obviously we need another way for it. Bugzilla: https://bugzilla.kernel.org/show_bug.cgi?id=56171 Reported-and-tested-by: Nikita Tsukanov <keks9n@gmail.com> Signed-off-by: Takashi Iwai <tiwai@suse.de>
2013-04-04ALSA: hda - fix typo in proc outputDavid Henningsson1-1/+1
Rename "Digitial In" to "Digital In". This function is only used for proc output, so should not cause any problems to change. Signed-off-by: David Henningsson <david.henningsson@canonical.com> Signed-off-by: Takashi Iwai <tiwai@suse.de>
2013-04-04ALSA: hda - Enabling Realtek ALC 671 codecRainer Koenig1-1/+3
* Added the device ID to the modalias list and assinged ALC662 patches for it * Added 4 port support for the device ID 0671 in alc662_parse_auto_config Signed-off-by: Rainer Koenig <Rainer.Koenig@ts.fujitsu.com> Signed-off-by: Takashi Iwai <tiwai@suse.de>
2013-04-03ASoC: tegra: Don't claim to support PCM pause and resumeLars-Peter Clausen1-23/+1
The tegra dmaengine driver does not support pausing and resuming a DMA stream. The tegra PCM driver still claims to support pause and resume though and implements them by stopping and restarting the stream. This is not what an application using pause/resume would expect. Usually applications have support for working around PCMs which do not support suspend and resume, so don't set the SNDRV_PCM_INFO_PAUSE and SNDRV_PCM_INFO_RESUME flags for the tegra PCM and use the default snd_dmaengine_pcm_trigger callback. Signed-off-by: Lars-Peter Clausen <lars@metafoo.de> Reviewed-by: Stephen Warren <swarren@nvidia.com> Tested-by: Stephen Warren <swarren@nvidia.com> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2013-04-03ASoC: Samsung: set drvdata before adding secondary devicePrathyush K1-5/+8
Currently, a new platform device is created for secondary device by calling platform_device_register_resndata and then the drvdata is set for this device. The following patch has been added to driver core: "driver core: fix possible missing of device probe". This results in the added device getting probed immediately but the drvdata for the secondary device is not yet set. This patch removes the platform_device_register_resndata call and instead calls platform_device_alloc, platform_set_drvdata and platform_device_add which fixes the above issue. Signed-off-by: Prathyush K <prathyush.k@samsung.com> Signed-off-by: Padmavathi Venna <padma.v@samsung.com> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2013-04-03ASoC: Samsung: return error if drvdata is not setPrathyush K1-0/+4
This patch fixes a possible crash in case drvdata for the secondary device is not set. Signed-off-by: Prathyush K <prathyush.k@samsung.com> Signed-off-by: Padmavathi Venna <padma.v@samsung.com> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2013-04-03ALSA: usb: Work around CM6631 sample rate change bugTorstein Hegge1-10/+35
The C-Media CM6631 USB receiver doesn't respond to changes in sample rate while the interface is active. The same behavior is observed in other UAC2 hardware like the VIA VT1731. Reset the interface after setting the sampling frequency on sample rate changes, to ensure that the sample rate set by snd_usb_init_sample_rate() is used. Otherwise, the device will try to use the sample rate of the previous stream, causing distorted sound on sample rate changes. The reset is performed for all UAC2 devices, as it should not affect a standards compliant device, but it is only necessary for C-Media CM6631, VIA VT1731 and possibly others. Failure to read sample rate from the device is not handled as an error in set_sample_rate_v2(), as (permanent or intermittent) failure to read sample rate isn't essential for a successful sample rate set. Signed-off-by: Torstein Hegge <hegge@resisty.net> Acked-by: Clemens Ladisch <clemens@ladisch.de> Signed-off-by: Takashi Iwai <tiwai@suse.de>
2013-04-02ALSA: hda - bug fix on HDMI ELD debug messageMengdong Lin1-1/+1
This patch let ELD debug message show 'pin_eld->monitor_present' which reflects the real pin response to verb GET_PIN_SENSE. 'eld->monitor_present' should not be used here because 'eld' is a temp structure now and so its "monitor_present" is not set. Signed-off-by: Mengdong Lin <mengdong.lin@intel.com> Acked-by: David Henningsson <david.henningsson@canonical.com> Signed-off-by: Takashi Iwai <tiwai@suse.de>
2013-04-02ALSA: hda - bug fix on return value when getting HDMI ELD infoMengdong Lin1-1/+1
In function snd_hdmi_get_eld(), the variable 'ret' should be initialized to 0. Otherwise it will be returned uninitialized as non-zero after ELD info is got successfully. Thus hdmi_present_sense() will always assume ELD info is invalid by mistake, and /proc file system cannot show the proper ELD info. Signed-off-by: Mengdong Lin <mengdong.lin@intel.com> Cc: stable@vger.kernel.org Acked-by: David Henningsson <david.henningsson@canonical.com> Signed-off-by: Takashi Iwai <tiwai@suse.de>
2013-03-28ASoC: compress: Cancel delayed power down if neededCharles Keepax1-3/+11
When a new stream is being opened it is necessary to cancel any delayed power down of the audio. [Fixed unused variable -- broonie] Signed-off-by: Charles Keepax <ckeepax@opensource.wolfsonmicro.com> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2013-03-26ASoC: core: Fix to check return value of snd_soc_update_bits_locked()Joonyoung Shim1-1/+1
It can be 0 or 1 return value of snd_soc_update_bits_locked() when it is success. So just check return value is negative. Signed-off-by: Joonyoung Shim <jy0922.shim@samsung.com> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com> Cc: stable@vger.kernel.org
2013-03-26Merge remote-tracking branch 'asoc/fix/spear' into asoc-nextMark Brown1-6/+6
2013-03-26Merge remote-tracking branch 'asoc/fix/si476x' into asoc-nextMark Brown1-0/+1
2013-03-26Merge remote-tracking branch 'asoc/fix/sh' into asoc-nextMark Brown1-2/+2
2013-03-26Merge remote-tracking branch 'asoc/fix/max98090' into asoc-nextMark Brown2-0/+0
2013-03-26Merge remote-tracking branch 'asoc/fix/fsl' into asoc-nextMark Brown2-1/+6
2013-03-26Merge remote-tracking branch 'asoc/fix/dapm' into asoc-nextMark Brown1-0/+14
2013-03-26Merge remote-tracking branch 'asoc/fix/core' into asoc-nextMark Brown1-4/+4
2013-03-26Merge remote-tracking branch 'asoc/fix/adsp' into asoc-nextMark Brown1-2/+3
2013-03-22ASoC: dma-sh7760: Fix compile errorLars-Peter Clausen1-2/+2
The dma-sh7760 currently fails with the following compile error: sound/soc/sh/dma-sh7760.c:346:2: error: unknown field 'pcm_ops' specified in initializer sound/soc/sh/dma-sh7760.c:346:2: warning: initialization from incompatible pointer type sound/soc/sh/dma-sh7760.c:347:2: error: unknown field 'pcm_new' specified in initializer sound/soc/sh/dma-sh7760.c:347:2: warning: initialization makes integer from pointer without a cast sound/soc/sh/dma-sh7760.c:348:2: error: unknown field 'pcm_free' specified in initializer sound/soc/sh/dma-sh7760.c:348:2: warning: initialization from incompatible pointer type sound/soc/sh/dma-sh7760.c: In function 'sh7760_soc_platform_probe': sound/soc/sh/dma-sh7760.c:353:2: warning: passing argument 2 of 'snd_soc_register_platform' from incompatible pointer type include/sound/soc.h:368:5: note: expected 'struct snd_soc_platform_driver *' but argument is of type 'struct snd_soc_platform *' This is due the misnaming of the snd_soc_platform_driver type name and 'ops' field. The issue was introduced in commit f0fba2a("ASoC: multi-component - ASoC Multi-Component Support"). Signed-off-by: Lars-Peter Clausen <lars@metafoo.de> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com> Cc: stable@vger.kernel.org
2013-03-21ALSA: hda - Fix DAC assignment for independent HPTakashi Iwai1-0/+46
The generic parser should evaluate the availability of the independent HP when specified. Otherwise a DAC without the direct connection to the corresponding pin may be assigned for the HP, but the driver doesn't check it at all. The problem was actually seen on some machines with VT1708s or equivalent codec, where DAC0 is assigned to HP although it can be connected only via aamix. This patch adds the badness evaluation for the independent HP to make it working properly. Reported-by: Lydia Wang <LydiaWang@viatech.com.cn> Signed-off-by: Takashi Iwai <tiwai@suse.de>
2013-03-20ALSA: hda - Fix abuse of snd_hda_lock_devices() for DSP loaderTakashi Iwai1-23/+109
The current DSP loader code abuses snd_hda_lock_devices() for ensuring the DSP loader not conflicting with the other normal operations. But this trick obviously doesn't work for the PM resume since the streams are kept opened there where snd_hda_lock_devices() returns -EBUSY. That means we need another lock mechanism instead of abuse. This patch provides the new lock state to azx_dev. Theoretically it's possible that the DSP loader conflicts with the stream that has been already assigned for another PCM. If it's running, the DSP loader should simply fail. If not -- it's the case for PM resume --, we should assign this stream temporarily to the DSP loader, and take it back to the PCM after finishing DSP loading. If the PCM is operated during the DSP loading, it should get an error, too. Reported-and-tested-by: Dylan Reid <dgreid@chromium.org> Signed-off-by: Takashi Iwai <tiwai@suse.de>
2013-03-20ALSA: hda - Fix typo in checking IEC958 emphasis bitTakashi Iwai1-1/+1
There is a typo in convert_to_spdif_status() about checking the emphasis IEC958 status bit. It should check the given value instead of the resultant value. Reported-by: Martin Weishart <martin.weishart@telosalliance.com> Cc: <stable@vger.kernel.org> Signed-off-by: Takashi Iwai <tiwai@suse.de>
2013-03-20ASoC: core: fix invalid free of devm_ allocated dataSilviu-Mihai Popescu1-2/+0
The objects allocated by devm_* APIs are managed by devres and are freed when the device is detached. Hence there is no need to use kfree() explicitly. Signed-off-by: Silviu-Mihai Popescu <silviupopescu1990@gmail.com> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2013-03-20ASoC: spear_pcm: Update to new pcm_new() APILars-Peter Clausen1-6/+6
Commit 552d1ef6 ("ASoC: core - Optimise and refactor pcm_new() to pass only rtd") updated the pcm_new() callback to take the rtd as the only parameter. The spear PCM driver (which was merged much later) still uses the old API. This patch updates the driver to the new API. Signed-off-by: Lars-Peter Clausen <lars@metafoo.de> Acked-by: Rajeev Kumar <rajeev-dlh.kumar@st.com> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com> Cc: stable@vger.kernel.org
2013-03-20ASoC:: max98090: Remove executable bitJoe Perches2-0/+0
Source files shouldn't have the executable bit set. Signed-off-by: Joe Perches <joe@perches.com> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2013-03-20ALSA: snd-usb: mixer: ignore -EINVAL in snd_usb_mixer_controls()Daniel Mack1-3/+3
Creation of individual mixer controls may fail, but that shouldn't cause the entire mixer creation to fail. Even worse, if the mixer creation fails, that will error out the entire device probing. All the functions called by parse_audio_unit() should return -EINVAL if they find descriptors that are unsupported or believed to be malformed, so we can safely handle this error code as a non-fatal condition in snd_usb_mixer_controls(). That fixes a long standing bug which is commonly worked around by adding quirks which make the driver ignore entire interfaces. Some of them might now be unnecessary. Signed-off-by: Daniel Mack <zonque@gmail.com> Reported-and-tested-by: Rodolfo Thomazelli <pe.soberbo@gmail.com> Cc: <stable@vger.kernel.org> Signed-off-by: Takashi Iwai <tiwai@suse.de>
2013-03-20ALSA: snd-usb: mixer: propagate errors up the call chainDaniel Mack1-4/+6
In check_input_term() and parse_audio_feature_unit(), propagate the error value that has been returned by a failing function instead of -EINVAL. That helps cleaning up the error pathes in the mixer. Signed-off-by: Daniel Mack <zonque@gmail.com> Cc: <stable@vger.kernel.org> Signed-off-by: Takashi Iwai <tiwai@suse.de>
2013-03-20ALSA: usb: Parse UAC2 extension unit like for UAC1Torstein Hegge1-1/+4
UAC2_EXTENSION_UNIT_V2 differs from UAC1_EXTENSION_UNIT, but can be handled in the same way when parsing the unit. Otherwise parse_audio_unit() fails when it sees an extension unit on a UAC2 device. UAC2_EXTENSION_UNIT_V2 is outside the range allocated by UAC1. Signed-off-by: Torstein Hegge <hegge@resisty.net> Acked-by: Daniel Mack <zonque@gmail.com> Cc: <stable@vger.kernel.org> Signed-off-by: Takashi Iwai <tiwai@suse.de>
2013-03-18ALSA: hda - Fix yet missing GPIO/EAPD setup in cirrus driverTakashi Iwai1-2/+2
I forgot to update spec->gpio_data in the automute hook, so it will be overridden at the init sequence, thus the machine is still silent when no headphone jack is plugged at boot time. Signed-off-by: Takashi Iwai <tiwai@suse.de>
2013-03-18ALSA: hda/cirrus - Fix the digital beep registrationTakashi Iwai1-4/+4
The argument passed to snd_hda_attach_beep_device() is a widget NID while spec->beep_amp holds the composed value for amp controls. Cc: <stable@vger.kernel.org> Signed-off-by: Takashi Iwai <tiwai@suse.de>
2013-03-18ALSA: hda - Fix missing beep detach in patch_conexant.cTakashi Iwai1-1/+7
This leaks the beep input device after module unload, which leads to Oops. Bugzilla: https://bugzilla.kernel.org/show_bug.cgi?id=55321 Signed-off-by: Takashi Iwai <tiwai@suse.de>
2013-03-15ALSA: hda - Fix missing EAPD/GPIO setup for Cirrus codecsTakashi Iwai1-0/+4
During the transition to the generic parser, the hook to the codec specific automute function was forgotten. This resulted in the silent output on some MacBooks. Signed-off-by: Takashi Iwai <tiwai@suse.de>
2013-03-15sound: sequencer: cap array index in seq_chn_common_event()Dan Carpenter1-0/+6
"chn" here is a number between 0 and 255, but ->chn_info[] only has 16 elements so there is a potential write beyond the end of the array. If the seq_mode isn't SEQ_2 then we let the individual drivers (either opl3.c or midi_synth.c) handle it. Those functions all do a bounds check on "chn" so I haven't changed anything here. The opl3.c driver has up to 18 channels and not 16. Signed-off-by: Dan Carpenter <dan.carpenter@oracle.com> Signed-off-by: Takashi Iwai <tiwai@suse.de>
2013-03-15ALSA: hda/ca0132 - Remove extra setting of dsp_state.Dylan Reid1-9/+6
spec->dsp_state is initialized to DSP_DOWNLOAD_INIT, no need to reset and check it in ca0132_download_dsp(). Signed-off-by: Dylan Reid <dgreid@chromium.org> Signed-off-by: Takashi Iwai <tiwai@suse.de>
2013-03-15ALSA: hda/ca0132 - Check download state of DSP.Dylan Reid1-2/+3
Instead of using the dspload_is_loaded() function, check the dsp_state that is kept in the spec. The dspload_is_loaded() function returns true if the DSP transfer was never started. This false-positive leads to multiple second delays when ca0132_setup_efaults() times out on each write. Signed-off-by: Dylan Reid <dgreid@chromium.org> Signed-off-by: Takashi Iwai <tiwai@suse.de>
2013-03-15ALSA: hda/ca0132 - Check if dspload_image succeeded.Dylan Reid1-2/+6
If dspload_image() fails, it was ignored and dspload_wait_loaded() was still called. dsp_loaded should never be set to true in this case, skip it. The check in dspload_wait_loaded() return true if the DSP is loaded or if it never started. Signed-off-by: Dylan Reid <dgreid@chromium.org> Signed-off-by: Takashi Iwai <tiwai@suse.de>
2013-03-15ASoC: dapm: Fix pointer dereference in is_connected_output_ep()Peter Ujfalusi1-1/+0
*path is not yet initialized when we check if the widget is connected. The compiler also warns about this: sound/soc/soc-dapm.c: In function 'is_connected_output_ep': sound/soc/soc-dapm.c:824:18: warning: 'path' may be used uninitialized in this function Signed-off-by: Peter Ujfalusi <peter.ujfalusi@ti.com> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2013-03-14ALSA: hda - Disable IDT eapd_switch if there are no internal speakersDavid Henningsson1-0/+29
If there are no internal speakers, we should not turn the eapd switch off, because it might be necessary to keep high for Headphone. BugLink: https://bugs.launchpad.net/bugs/1155016 Signed-off-by: David Henningsson <david.henningsson@canonical.com> Signed-off-by: Takashi Iwai <tiwai@suse.de>