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* vm: convert snd_pcm_lib_mmap_iomem() to vm_iomap_memory() helperLinus Torvalds2013-04-19
| | | | | | | | 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>
* Merge tag 'asoc-v3.9-rc6' of ↵Takashi Iwai2013-04-12
|\ | | | | | | | | | | | | | | | | | | 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.
| * Merge remote-tracking branch 'asoc/fix/wm8903' into tmpMark Brown2013-04-11
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| | * ASoC: wm8903: Fix the bypass to HP/LINEOUT when no DAC or ADC is runningAlban Bedel2013-04-09
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | 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
| * | Merge remote-tracking branch 'asoc/fix/tegra' into tmpMark Brown2013-04-11
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| | * | ASoC: tegra: Don't claim to support PCM pause and resumeLars-Peter Clausen2013-04-03
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | 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>
| * | | Merge remote-tracking branch 'asoc/fix/samsung' into tmpMark Brown2013-04-11
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| | * | | ASoC: Samsung: set drvdata before adding secondary devicePrathyush K2013-04-03
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | 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>
| | * | | ASoC: Samsung: return error if drvdata is not setPrathyush K2013-04-03
| | |/ / | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | 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>
| * | | Merge remote-tracking branch 'asoc/fix/core' into tmpMark Brown2013-04-11
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| | * | | ASoC: core: Fix to check return value of snd_soc_update_bits_locked()Joonyoung Shim2013-03-26
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | 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
| * | | | Merge remote-tracking branch 'asoc/fix/compress' into tmpMark Brown2013-04-11
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| | * | | | ASoC: compress: Cancel delayed power down if neededCharles Keepax2013-03-27
| | | |/ / | | |/| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | 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>
| * | | | ASoC: wm5102: Correct lookup of arizona struct in SYSCLK eventMark Brown2013-04-09
| | |_|/ | |/| | | | | | | | | | | | | | | | | | Reported-by: Ryo Tsutsui <Ryo.Tsutsui@wolfsonmicro.com> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com> Cc: stable@vger.kernel.org
* / | | ALSA: usb-audio: fix endianness bug in snd_nativeinstruments_*Eldad Zack2013-04-07
|/ / / | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | 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>
* | | ALSA: hda/generic - fix uninitialized variableJiri Slaby2013-04-05
| | | | | | | | | | | | | | | | | | | | | | | | | | | 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>
* | | Revert "ALSA: hda - Allow power_save_controller option override DCAPS"Takashi Iwai2013-04-04
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | 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>
* | | ALSA: hda - fix typo in proc outputDavid Henningsson2013-04-04
| | | | | | | | | | | | | | | | | | | | | | | | 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>
* | | ALSA: hda - Enabling Realtek ALC 671 codecRainer Koenig2013-04-04
| | | | | | | | | | | | | | | | | | | | | | | | | | | * 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>
* | | ALSA: usb: Work around CM6631 sample rate change bugTorstein Hegge2013-04-03
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | 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>
* | | ALSA: hda - bug fix on HDMI ELD debug messageMengdong Lin2013-04-02
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | 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>
* | | ALSA: hda - bug fix on return value when getting HDMI ELD infoMengdong Lin2013-04-02
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | 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>
* | | Merge remote-tracking branch 'asoc/fix/spear' into asoc-nextMark Brown2013-03-26
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| * | | ASoC: spear_pcm: Update to new pcm_new() APILars-Peter Clausen2013-03-20
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | 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
* | | | Merge remote-tracking branch 'asoc/fix/si476x' into asoc-nextMark Brown2013-03-26
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| * | | | ASoC: si476x: Add missing break for SNDRV_PCM_FORMAT_S8 switch caseAxel Lin2013-02-20
| | | | | | | | | | | | | | | | | | | | | | | | | Signed-off-by: Axel Lin <axel.lin@ingics.com> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
* | | | | Merge remote-tracking branch 'asoc/fix/sh' into asoc-nextMark Brown2013-03-26
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| * | | | | ASoC: dma-sh7760: Fix compile errorLars-Peter Clausen2013-03-22
| | |/ / / | |/| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | 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
* | | | | Merge remote-tracking branch 'asoc/fix/max98090' into asoc-nextMark Brown2013-03-26
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| * | | | | ASoC:: max98090: Remove executable bitJoe Perches2013-03-20
| |/ / / / | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | 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>
* | | | | Merge remote-tracking branch 'asoc/fix/fsl' into asoc-nextMark Brown2013-03-26
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| * | | | | ASoC: pcm030 audio fabric: remove __init from probeMarkus Pargmann2013-03-12
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | Remove probe function from the init section. Signed-off-by: Markus Pargmann <mpa@pengutronix.de> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
| * | | | | ASoC: imx-ssi: Fix occasional AC97 reset failureSascha Hauer2013-03-12
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | Signed-off-by: Sascha Hauer <s.hauer@pengutronix.de> Signed-off-by: Markus Pargmann <mpa@pengutronix.de> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com> Cc: stable@vger.kernel.org
* | | | | | Merge remote-tracking branch 'asoc/fix/dapm' into asoc-nextMark Brown2013-03-26
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| * | | | | | ASoC: dapm: Fix pointer dereference in is_connected_output_ep()Peter Ujfalusi2013-03-14
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | *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>
| * | | | | | ASoC: dapm: Fix handling of loopsMark Brown2013-02-25
| | |_|/ / / | |/| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | Currently if a path loops back on itself we correctly skip over it to avoid going into an infinite loop but this causes us to ignore the need to power up the path as we don't count the loop for the purposes of counting inputs and outputs. This means that internal loopbacks within a device that have powered devices on them won't be powered up. Fix this by treating any path that is currently in the process of being recursed as having a single input or output so that it is counted for the purposes of power decisions. Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com> Acked-by: Liam Girdwood <liam.r.girdwood@linux.intel.com>
* | | | | | Merge remote-tracking branch 'asoc/fix/core' into asoc-nextMark Brown2013-03-26
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| * | | | | ASoC: core: fix invalid free of devm_ allocated dataSilviu-Mihai Popescu2013-03-20
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | 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>
| * | | | | ASoC: core: fix possible memory leak in snd_soc_bytes_put()Wei Yongjun2013-03-12
| | |/ / / | |/| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | 'data' is malloced in snd_soc_bytes_put() and should be freed before leaving from the error handling cases, otherwise it will cause memory leak. Signed-off-by: Wei Yongjun <yongjun_wei@trendmicro.com.cn> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
* | | | | Merge remote-tracking branch 'asoc/fix/adsp' into asoc-nextMark Brown2013-03-26
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| * | | | ASoC: wm_adsp: fix possible memory leak in wm_adsp_load_coeff()Wei Yongjun2013-03-12
| |/ / / | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | 'file' is malloced in wm_adsp_load_coeff() and should be freed before leaving from the error handling cases, otherwise it will cause memory leak. Signed-off-by: Wei Yongjun <yongjun_wei@trendmicro.com.cn> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
* | | | ALSA: hda - Fix DAC assignment for independent HPTakashi Iwai2013-03-21
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | 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>
* | | | ALSA: hda - Fix abuse of snd_hda_lock_devices() for DSP loaderTakashi Iwai2013-03-20
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | 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>
* | | | ALSA: hda - Fix typo in checking IEC958 emphasis bitTakashi Iwai2013-03-20
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | 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>
* | | | ALSA: snd-usb: mixer: ignore -EINVAL in snd_usb_mixer_controls()Daniel Mack2013-03-20
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | 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>
* | | | ALSA: snd-usb: mixer: propagate errors up the call chainDaniel Mack2013-03-20
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | 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>
* | | | ALSA: usb: Parse UAC2 extension unit like for UAC1Torstein Hegge2013-03-20
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | 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>
* | | | ALSA: hda - Fix yet missing GPIO/EAPD setup in cirrus driverTakashi Iwai2013-03-18
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | 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>
* | | | ALSA: hda/cirrus - Fix the digital beep registrationTakashi Iwai2013-03-18
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | 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>
* | | | ALSA: hda - Fix missing beep detach in patch_conexant.cTakashi Iwai2013-03-18
| |_|/ |/| | | | | | | | | | | | | | | | | | | | 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>
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