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* ALSA: hda/realtek: Enable audio jacks of ASUS UX433FN/UX333FA with ALC294Jian-Hong Pan2018-12-101-0/+4
| | | | | | | | | | | The ASUS UX433FN and UX333FA with ALC294 cannot detect the headset MIC and output through the internal speaker and the headphone until ALC294_FIXUP_ASUS_SPK and ALC294_FIXUP_ASUS_HEADSET_MIC quirk applied. Signed-off-by: Daniel Drake <drake@endlessm.com> Signed-off-by: Jian-Hong Pan <jian-hong@endlessm.com> Cc: <stable@vger.kernel.org> Signed-off-by: Takashi Iwai <tiwai@suse.de>
* ALSA: hda/realtek: Enable audio jacks of ASUS UX533FD with ALC294Jian-Hong Pan2018-12-101-0/+23
| | | | | | | | | | | The ASUS UX533FD with ALC294 cannot detect the headset MIC and outputs through the internal speaker and the headphone until ALC294_FIXUP_ASUS_SPK and ALC294_FIXUP_ASUS_HEADSET_MIC quirk applied. Signed-off-by: Daniel Drake <drake@endlessm.com> Signed-off-by: Jian-Hong Pan <jian-hong@endlessm.com> Cc: <stable@vger.kernel.org> Signed-off-by: Takashi Iwai <tiwai@suse.de>
* ALSA: hda/realtek: ALC294 mic and headset-mode fixups for ASUS X542UNChris Chiu2018-12-101-0/+15
| | | | | | | | | | | | | The known ALC256_FIXUP_ASUS_MIC fixup can fix the headphone jack sensing and enable use of the internal microphone on this laptop X542UN. However, it's ALC294 so create a new fixup named ALC294_FIXUP_ASUS_MIC to avoid confusion. Signed-off-by: Jian-Hong Pan <jian-hong@endlessm.com> Signed-off-by: Daniel Drake <drake@endlessm.com> Signed-off-by: Chris Chiu <chiu@endlessm.com> Cc: <stable@vger.kernel.org> Signed-off-by: Takashi Iwai <tiwai@suse.de>
* ALSA: fireface: fix reference to wrong register for clock configurationTakashi Sakamoto2018-12-091-1/+1
| | | | | | | | | | | | | | | In an initial commit, 'SYNC_STATUS' register is referred to get clock configuration, however this is wrong, according to my local note at hand for reverse-engineering about packet dump. It should be 'CLOCK_CONFIG' register. Actually, ff400_dump_clock_config() is correctly programmed. This commit fixes the bug. Cc: <stable@vger.kernel.org> # v4.12+ Fixes: 76fdb3a9e13a ('ALSA: fireface: add support for Fireface 400') Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp> Signed-off-by: Takashi Iwai <tiwai@suse.de>
* ALSA: hda/realtek - Fix the mute LED regresion on Lenovo X1 CarbonHui Wang2018-12-091-0/+2
| | | | | | | | | | | | | Users reported a mute LED regression on Lenovo X1 Carbon, the root cause is we applied the fixup of ALC285_FIXUP_LENOVO_HEADPHONE_NOISE to this machine, then the machine can't apply the fixup of ALC269_FIXUP_THINKPAD_ACPI anymore. To fix it, we chain two fixup together. Fixes: c4cfcf6f4297 ("ALSA: hda/realtek - fix the pop noise on headphone for lenovo laptops") Cc: <stable@vger.kernel.org> Signed-off-by: Hui Wang <hui.wang@canonical.com> Signed-off-by: Takashi Iwai <tiwai@suse.de>
* ALSA: hda/realtek - Fixed headphone issue for ALC700Kailang Yang2018-12-071-0/+33
| | | | | | | | | | | | If it plugged headphone or headset into the jack, then do the reboot, it will have a chance to cause headphone no sound. It just need to run the headphone mode procedure after boot time. The issue will be fixed. It also suitable for ALC234 ALC274 and ALC294. Signed-off-by: Kailang Yang <kailang@realtek.com> Cc: <stable@vger.kernel.org> Signed-off-by: Takashi Iwai <tiwai@suse.de>
* ALSA: hda/realtek: Fix mic issue on Acer AIO Veriton Z4860G/Z6860GChris Chiu2018-12-051-0/+2
| | | | | | | | | | | | Acer AIO Veriton Z4860G/Z6860G with the same ALC286 codec has issues with the input from external microphone. The issue can be fixed by the fixup ALC286_FIXUP_ACER_AIO_MIC_NO_PRESENCE for Veriton Z4660G. Signed-off-by: Jian-Hong Pan <jian-hong@endlessm.com> Signed-off-by: Daniel Drake <drake@endlessm.com> Signed-off-by: Chris Chiu <chiu@endlessm.com> Cc: <stable@vger.kernel.org> Signed-off-by: Takashi Iwai <tiwai@suse.de>
* ALSA: hda/realtek: Fix mic issue on Acer AIO Veriton Z4660GChris Chiu2018-12-051-0/+1
| | | | | | | | | | | | | Acer AIO Veriton Z4660G with ALC286 codec has issue with the input from external microphones connecting via 'Front Mic' jack. The fixup ALC286_FIXUP_ACER_AIO_MIC_NO_PRESENCE enables the jack sensing of the headset and fix the audio input issue of external microphone. Signed-off-by: Jian-Hong Pan <jian-hong@endlessm.com> Signed-off-by: Daniel Drake <drake@endlessm.com> Signed-off-by: Chris Chiu <chiu@endlessm.com> Cc: <stable@vger.kernel.org> Signed-off-by: Takashi Iwai <tiwai@suse.de>
* ALSA: hda/realtek - Add support for Acer Aspire C24-860 headset micChris Chiu2018-12-051-0/+1
| | | | | | | | | | | | The Acer AIO Aspire C24-860 with ALC286 can't detect the headset microphone. Just like another Acer AIO U27-880, it needs a different pin value for 0x18 and the headset fixup to make headset mic work. Signed-off-by: Jian-Hong Pan <jian-hong@endlessm.com> Signed-off-by: Daniel Drake <drake@endlessm.com> Signed-off-by: Chris Chiu <chiu@endlessm.com> Cc: <stable@vger.kernel.org> Signed-off-by: Takashi Iwai <tiwai@suse.de>
* ALSA: hda/realtek: ALC286 mic and headset-mode fixups for Acer Aspire U27-880Chris Chiu2018-12-051-0/+14
| | | | | | | | | | | | | | | Acer Aspire U27-880(AIO) with ALC286 codec can not detect headset mic and internal mic not working either. It needs the similar quirk like Sony laptops to fix headphone jack sensing and enables use of the internal microphone. Unfortunately jack sensing for the headset mic is still not working. Signed-off-by: Jian-Hong Pan <jian-hong@endlessm.com> Signed-off-by: Daniel Drake <drake@endlessm.com> Signed-off-by: Chris Chiu <chiu@endlessm.com> Cc: <stable@vger.kernel.org> Signed-off-by: Takashi Iwai <tiwai@suse.de>
* ALSA: usb-audio: Fix UAF decrement if card has no live interfaces in card.cHui Peng2018-12-031-1/+4
| | | | | | | | | | | | | | | | | | | | | If a USB sound card reports 0 interfaces, an error condition is triggered and the function usb_audio_probe errors out. In the error path, there was a use-after-free vulnerability where the memory object of the card was first freed, followed by a decrement of the number of active chips. Moving the decrement above the atomic_dec fixes the UAF. [ The original problem was introduced in 3.1 kernel, while it was developed in a different form. The Fixes tag below indicates the original commit but it doesn't mean that the patch is applicable cleanly. -- tiwai ] Fixes: 362e4e49abe5 ("ALSA: usb-audio - clear chip->probing on error exit") Reported-by: Hui Peng <benquike@gmail.com> Reported-by: Mathias Payer <mathias.payer@nebelwelt.net> Signed-off-by: Hui Peng <benquike@gmail.com> Signed-off-by: Mathias Payer <mathias.payer@nebelwelt.net> Cc: <stable@vger.kernel.org> Signed-off-by: Takashi Iwai <tiwai@suse.de>
* ALSA: hda/realtek - Fix speaker output regression on Thinkpad T570Takashi Iwai2018-12-031-0/+9
| | | | | | | | | | | | | | | | | | | | | | | | | We've got a regression report for some Thinkpad models (at least T570s) which shows the too low speaker output volume. The bisection leaded to the commit 61fcf8ece9b6 ("ALSA: hda/realtek - Enable Thinkpad Dock device for ALC298 platform"), and it's basically adding the two pin configurations for the dock, and looks harmless. The real culprit seems, though, that the DAC assignment for the speaker pin is implicitly assumed on these devices, i.e. pin NID 0x14 to be coupled with DAC NID 0x03. When more pins are configured by the commit above, the auto-parser changes the DAC assignment, and this resulted in the regression. As a workaround, just provide the fixed pin / DAC mapping table for this Thinkpad fixup function. It's no generic solution, but the problem itself is pretty much device-specific, so must be good enough. Bugzilla: https://bugzilla.redhat.com/show_bug.cgi?id=1554304 Fixes: 61fcf8ece9b6 ("ALSA: hda/realtek - Enable Thinkpad Dock device for ALC298 platform") Cc: <stable@vger.kernel.org> Reported-and-tested-by: Jeremy Cline <jcline@redhat.com> Signed-off-by: Takashi Iwai <tiwai@suse.de>
* ALSA: hda: Add support for AMD Stoney RidgeKai-Heng Feng2018-11-291-0/+4
| | | | | | | | | It's similar to other AMD audio devices, it also supports D3, which can save some power drain. Signed-off-by: Kai-Heng Feng <kai.heng.feng@canonical.com> Cc: <stable@vger.kernel.org> Signed-off-by: Takashi Iwai <tiwai@suse.de>
* ALSA: usb-audio: Add SMSL D1 to quirks for native DSD supportTony Das2018-11-291-0/+1
| | | | | | | | | | | This patch adds quirk VID/PID IDs for the SMSL D1 in order to enable Native DSD support. [ Moved the added entry in numerical order -- tiwai ] Signed-off-by: Tony Das <tdas444@gmail.com> Cc: <stable@vger.kernel.org> Signed-off-by: Takashi Iwai <tiwai@suse.de>
* ALSA: pcm: Fix starvation on down_write_nonblock()Chanho Min2018-11-291-5/+6
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | Commit 67ec1072b053 ("ALSA: pcm: Fix rwsem deadlock for non-atomic PCM stream") fixes deadlock for non-atomic PCM stream. But, This patch causes antother stuck. If writer is RT thread and reader is a normal thread, the reader thread will be difficult to get scheduled. It may not give chance to release readlocks and writer gets stuck for a long time if they are pinned to single cpu. The deadlock described in the previous commit is because the linux rwsem queues like a FIFO. So, we might need non-FIFO writelock, not non-block one. My suggestion is that the writer gives reader a chance to be scheduled by using the minimum msleep() instaed of spinning without blocking by writer. Also, The *_nonblock may be changed to *_nonfifo appropriately to this concept. In terms of performance, when trylock is failed, this minimum periodic msleep will have the same performance as the tick-based schedule()/wake_up_q(). [ Although this has a fairly high performance penalty, the relevant code path became already rare due to the previous commit ("ALSA: pcm: Call snd_pcm_unlink() conditionally at closing"). That is, now this unconditional msleep appears only when using linked streams, and this must be a rare case. So we accept this as a quick workaround until finding a more suitable one -- tiwai ] Fixes: 67ec1072b053 ("ALSA: pcm: Fix rwsem deadlock for non-atomic PCM stream") Suggested-by: Wonmin Jung <wonmin.jung@lge.com> Signed-off-by: Chanho Min <chanho.min@lge.com> Cc: <stable@vger.kernel.org> Signed-off-by: Takashi Iwai <tiwai@suse.de>
* ALSA: pcm: Call snd_pcm_unlink() conditionally at closingTakashi Iwai2018-11-291-1/+2
| | | | | | | | | | | | | | | | | | | Currently the PCM core calls snd_pcm_unlink() always unconditionally at closing a stream. However, since snd_pcm_unlink() invokes the global rwsem down, the lock can be easily contended. More badly, when a thread runs in a high priority RT-FIFO, it may stall at spinning. Basically the call of snd_pcm_unlink() is required only for the linked streams that are already rare occasion. For normal use cases, this code path is fairly superfluous. As an optimization (and also as a workaround for the RT problem above in normal situations without linked streams), this patch adds a check before calling snd_pcm_unlink() and calls it only when needed. Reported-by: Chanho Min <chanho.min@lge.com> Cc: <stable@vger.kernel.org> Signed-off-by: Takashi Iwai <tiwai@suse.de>
* ALSA: usb-audio: Add vendor and product name for Dell WD19 DockHui Wang2018-11-281-0/+10
| | | | | | | | | Like the Dell WD15 Dock, the WD19 Dock (0bda:402e) doens't provide useful string for the vendor and product names too. In order to share the UCM with WD15, here we keep the profile_name same as the WD15. Signed-off-by: Hui Wang <hui.wang@canonical.com> Signed-off-by: Takashi Iwai <tiwai@suse.de>
* Merge tag 'asoc-v4.20-rc4' of ↵Takashi Iwai2018-11-2724-258/+358
|\ | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound into for-linus ASoC: Fixes for v4.20 Lots of fixes here, the majority of which are driver specific but there's a couple of core things and one notable driver specific one: - A core fix for a DAPM regression introduced during the component refactoring, we'd lost the code that forced a reevaluation of the DAPM graph after probe (which we suppress during init to save lots of recalcuation) and have now restored it. - A core fix for error handling using the newly added for_each_rtd_codec_dai_rollback() macro. - A fix for the names of widgets in the newly introduced pcm3060 driver, merged as a fix so we don't have a release with legacy names.
| * ASoC: omap-dmic: Add pm_qos handling to avoid overruns with CPU_IDLEPeter Ujfalusi2018-11-231-0/+9
| | | | | | | | | | | | | | | | | | | | We need to block sleep states which would require longer time to leave than the time the DMA must react to the DMA request in order to keep the FIFO serviced without overrun. Signed-off-by: Peter Ujfalusi <peter.ujfalusi@ti.com> Acked-by: Jarkko Nikula <jarkko.nikula@bitmer.com> Signed-off-by: Mark Brown <broonie@kernel.org>
| * ASoC: omap-mcpdm: Add pm_qos handling to avoid under/overruns with CPU_IDLEPeter Ujfalusi2018-11-231-1/+42
| | | | | | | | | | | | | | | | | | | | We need to block sleep states which would require longer time to leave than the time the DMA must react to the DMA request in order to keep the FIFO serviced without under of overrun. Signed-off-by: Peter Ujfalusi <peter.ujfalusi@ti.com> Acked-by: Jarkko Nikula <jarkko.nikula@bitmer.com> Signed-off-by: Mark Brown <broonie@kernel.org>
| * ASoC: omap-mcbsp: Fix latency value calculation for pm_qosPeter Ujfalusi2018-11-231-3/+3
| | | | | | | | | | | | | | | | | | The latency number is in usec for the pm_qos. Correct the calculation to give us the time in usec Signed-off-by: Peter Ujfalusi <peter.ujfalusi@ti.com> Acked-by: Jarkko Nikula <jarkko.nikula@bitmer.com> Signed-off-by: Mark Brown <broonie@kernel.org>
| * ASoC: acpi: fix: continue searching when machine is ignoredKeyon Jie2018-11-201-2/+8
| | | | | | | | | | | | | | | | | | | | The machine_quirk may return NULL which means the acpi entries should be skipped and search for next matched entry is needed, here add return check here and continue for NULL case. Signed-off-by: Keyon Jie <yang.jie@linux.intel.com> Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com> Signed-off-by: Mark Brown <broonie@kernel.org>
| * ASoC: Intel: Skylake: fix Kconfigs, make HDaudio codec optionalPierre-Louis Bossart2018-11-203-15/+55
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | The Skylake driver currently has a set of problems supporting load/unload modules. We need to make the HDaudio codec support optional to help narrow down the issues. Support for HDaudio codecs also leads to a Kconfig issue. We want the hdac_hda codec to be compilable independently of Skylake (e.g. with ALL_CODECS) but when Skylake is selected as built-in the hdac_hda codec needs to use the same option due a a code dependency Solve both problems by adding a user-selectable boolean Kconfig, select HDAC_HDA as needed and make the HDaudio codec support in the Skylake driver optional. Tests on a Chell Chromebook device without HDaudio show no regression for speaker and HDMI playback. This is submitted as an RFC to allow for comments and more validation. Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com> Signed-off-by: Mark Brown <broonie@kernel.org>
| * ASoC: pcm186x: Fix device reset-registers trigger valueAndreas Dannenberg2018-11-151-1/+1
| | | | | | | | | | | | | | | | | | | | | | | | | | | | According to the current device datasheet (TI Lit # SLAS831D, revised March 2018) the value written to the device's PAGE register to trigger a complete register reset should be 0xfe, not 0xff. So go ahead and update to the correct value. Reported-by: Stephane Le Provost <stephane.leprovost@mediatek.com> Tested-by: Stephane Le Provost <stephane.leprovost@mediatek.com> Signed-off-by: Andreas Dannenberg <dannenberg@ti.com> Acked-by: Andrew F. Davis <afd@ti.com> Signed-off-by: Mark Brown <broonie@kernel.org> Cc: stable@vger.kernel.org
| * ASoC: dapm: Recalculate audio map forcely when card instantiatedTzung-Bi Shih2018-11-141-0/+1
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | Audio map are possible in wrong state before card->instantiated has been set to true. Imaging the following examples: time 1: at the beginning in:-1 in:-1 in:-1 in:-1 out:-1 out:-1 out:-1 out:-1 SIGGEN A B Spk time 2: after someone called snd_soc_dapm_new_widgets() (e.g. create_fill_widget_route_map() in sound/soc/codecs/hdac_hdmi.c) in:1 in:0 in:0 in:0 out:0 out:0 out:0 out:1 SIGGEN A B Spk time 3: routes added in:1 in:0 in:0 in:0 out:0 out:0 out:0 out:1 SIGGEN -----> A -----> B ---> Spk In the end, the path should be powered on but it did not. At time 3, "in" of SIGGEN and "out" of Spk did not propagate to their neighbors because snd_soc_dapm_add_path() will not invalidate the paths if the card has not instantiated (i.e. card->instantiated is false). To correct the state of audio map, recalculate the whole map forcely. Signed-off-by: Tzung-Bi Shih <tzungbi@google.com> Signed-off-by: Mark Brown <broonie@kernel.org>
| * ASoC: omap-abe-twl6040: Fix missing audio card caused by deferred probingPeter Ujfalusi2018-11-141-38/+29
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | The machine driver fails to probe in next-20181113 with: [ 2.539093] omap-abe-twl6040 sound: ASoC: CODEC DAI twl6040-legacy not registered [ 2.546630] omap-abe-twl6040 sound: devm_snd_soc_register_card() failed: -517 ... [ 3.693206] omap-abe-twl6040 sound: ASoC: Both platform name/of_node are set for TWL6040 [ 3.701446] omap-abe-twl6040 sound: ASoC: failed to init link TWL6040 [ 3.708007] omap-abe-twl6040 sound: devm_snd_soc_register_card() failed: -22 [ 3.715148] omap-abe-twl6040: probe of sound failed with error -22 Bisect pointed to a merge commit: first bad commit: [0f688ab20a540aafa984c5dbd68a71debebf4d7f] Merge remote-tracking branch 'net-next/master' and a diff between a working kernel does not reveal anything which would explain the change in behavior. Further investigation showed that on the second try of loading fails because the dai_link->platform is no longer NULL and it might be pointing to uninitialized memory. The fix is to move the snd_soc_dai_link and snd_soc_card inside of the abe_twl6040 struct, which is dynamically allocated every time the driver probes. Signed-off-by: Peter Ujfalusi <peter.ujfalusi@ti.com> Signed-off-by: Mark Brown <broonie@kernel.org>
| * ASoC: pcm3060: Rename output widgetsKirill Marinushkin2018-11-141-8/+4
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | In the initial commit [1], I added differential output of the codec as separate `+` and `-` widgets: OUTL+ OUTR+ OUTL- OUTR- Later, in the commit [2], I added a device tree property to configure the output as single-ended or differential. Having this property, the `+` and `-` separation in widgets seems for me confusing. There are no functional benefits in such separation, so I find reasonable to get rid of it: OUTL OUTR The new naming is more friendly for sound cards, and is better aligned with other codec drivers in kernel. Renaming the output widgets now should not be a problem from the backwards- compatibility perspective, as the driver for PCM3060 is added into the mainline very recently, and did not yet appear in any releases. [1] commit 6ee47d4a8dac ("ASoC: pcm3060: Add codec driver") [2] commit a78c62de00d5 ("ASoC: pcm3060: Add DT property for single-ended output") Signed-off-by: Kirill Marinushkin <kmarinushkin@birdec.tech> Signed-off-by: Mark Brown <broonie@kernel.org>
| * ASoC: qcom: Set dai_link id to each dai_linkRohit kumar2018-11-141-3/+6
| | | | | | | | | | | | | | | | | | | | | | | | Frontend dai_link id is used for closing ADM sessions. During concurrent usecase when one session is closed, it closes other ADM session associated with other usecase too. Dai_link->id should always point to Frontend dai id. Set cpu_dai id as dai_link id to fix the issue. Signed-off-by: Rohit kumar <rohitkr@codeaurora.org> Acked-by: Srinivas Kandagatla <srinivas.kandagatla@linaro.org> Signed-off-by: Mark Brown <broonie@kernel.org>
| * ASoC: sun8i-codec: add missing route for ADCVasily Khoruzhick2018-11-141-1/+5
| | | | | | | | | | | | | | | | | | sun8i-codec misses a route from ADC to AIF1 Slot 0 ADC. Add it to the driver to avoid adding it to every dts. Fixes: eda85d1fee05d ("ASoC: sun8i-codec: Add ADC support for a33") Signed-off-by: Vasily Khoruzhick <anarsoul@gmail.com> Signed-off-by: Mark Brown <broonie@kernel.org>
| * ASoC: Intel: Power down links before turning off display audio powerPierre-Louis Bossart2018-11-132-12/+11
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | On certain platforms, Display HDMI HDA codec was not going to sleep state after the use when links are powered down after turning off the display power. As per the HW recommendation, links are powered down before turning off the display power to ensure that the codec goes to sleep state. This patch was updated from an earlier version submitted upstream [1] which conflicted with the changes merged for HDaudio codec support with the Intel DSP. [1] https://patchwork.kernel.org/patch/10540213/ Signed-off-by: Sriram Periyasamy <sriramx.periyasamy@intel.com> Signed-off-by: Sanyog Kale <sanyog.r.kale@intel.com> Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com> Signed-off-by: Mark Brown <broonie@kernel.org>
| * ASoC: wm_adsp: Fix dma-unsafe read of scratch registersRichard Fitzgerald2018-11-131-17/+20
| | | | | | | | | | | | | | | | | | | | | | | | | | Stack memory isn't DMA-safe so it isn't safe to use either regmap_raw_read or regmap_bulk_read to read into stack memory. The two functions to read the scratch registers were using stack memory and regmap_raw_read. It's not worth allocating memory just for this trivial read, and it isn't time-critical. A simple regmap_read for each register is sufficient. Signed-off-by: Richard Fitzgerald <rf@opensource.cirrus.com> Signed-off-by: Mark Brown <broonie@kernel.org>
| * ASoC: rockchip: add missing slave_config setting for I2SKatsuhiro Suzuki2018-11-131-0/+1
| | | | | | | | | | | | | | | | This patch adds missing prepare_sleve_config that is needed for setup the DMA slave channel for I2S. Signed-off-by: Katsuhiro Suzuki <katsuhiro@katsuster.net> Signed-off-by: Mark Brown <broonie@kernel.org>
| * ASoC: sun8i-codec: fix crash on module removalVasily Khoruzhick2018-11-091-6/+0
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | drvdata is actually sun8i_codec, not snd_soc_card, so it crashes when calling snd_soc_card_get_drvdata(). Drop card and scodec vars anyway since we don't need to disable/unprepare clocks - it's already done by calling runtime_suspend() Drop clk_disable_unprepare() calls for the same reason. Fixes: 36c684936fae7 ("ASoC: Add sun8i digital audio codec") Signed-off-by: Vasily Khoruzhick <anarsoul@gmail.com> Acked-by: Maxime Ripard <maxime.ripard@bootlin.com> Signed-off-by: Mark Brown <broonie@kernel.org>
| * ASoC: qdsp6: q6afe-dai: Fix the dai widgetsSrinivas Kandagatla2018-11-061-104/+104
| | | | | | | | | | | | | | | | | | | | | | | | | | For some reason the dapm widgets are incorrectly defined from the start, Not sure how we ended up with such thing. Fix them now! Without this fix the backend dais are always powered up even if there is no active stream. Reported-by: Jimmy Cheng-Yi Chiang <cychiang@google.com> Reported-by: Rohit kumar <rohitkr@codeaurora.org> Signed-off-by: Srinivas Kandagatla <srinivas.kandagatla@linaro.org> Signed-off-by: Mark Brown <broonie@kernel.org>
| * ASoC: qdsp6: q6asm-dai: Only add routing once.Srinivas Kandagatla2018-11-062-33/+19
| | | | | | | | | | | | | | | | | | q6asm routing gets added multiple times as part of dai probe. Move this to q6routing routes which has those widgets defined, this also fixes the issue where these are added each time at dai probe. Signed-off-by: Srinivas Kandagatla <srinivas.kandagatla@linaro.org> Signed-off-by: Mark Brown <broonie@kernel.org>
| * ASoC: qdsp6: q6afe: Fix wrong MI2S SD line maskRohit kumar2018-11-051-8/+8
| | | | | | | | | | | | | | | | SD line mask for MI2S starts from BIT 0 instead of BIT 1. Fix all bit mask for MI2S SD lines. Signed-off-by: Rohit kumar <rohitkr@codeaurora.org> Signed-off-by: Mark Brown <broonie@kernel.org>
| * ASoC: intel: cht_bsw_max98090_ti: Add quirk for boards using pmc_plt_clk_0Hans de Goede2018-11-051-3/+29
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | Some boards such as the Swanky model Chromebooks use pmc_plt_clk_0 for the mclk instead of pmc_plt_clk_3. This commit adds a DMI based quirk for this. This fixing audio no longer working on these devices after commit 648e921888ad ("clk: x86: Stop marking clocks as CLK_IS_CRITICAL") that commit fixes us unnecessary keeping unused clocks on, but in case of the Swanky that was breaking audio support since we were not using the right clock in the cht_bsw_max98090_ti machine driver. Cc: stable@vger.kernel.org Fixes: 648e921888ad ("clk: x86: Stop marking clocks as CLK_IS_CRITICAL") Reported-and-tested-by: Dean Wallace <duffydack73@gmail.com> Signed-off-by: Hans de Goede <hdegoede@redhat.com> Signed-off-by: Mark Brown <broonie@kernel.org>
| * ASoC: sunxi: rename SND_SUNXI_ADDA_PR_REGMAP to SND_SUN8I_ADDA_PR_REGMAPVasily Khoruzhick2018-11-051-1/+1
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | SND_SUN50I_CODEC_ANALOG selects SND_SUNXI_ADDA_PR_REGMAP which is leftover of renaming SND_SUNXI_ADDA_PR_REGMAP to SND_SUN8I_ADDA_PR_REGMAP. Replace it with SND_SUN8I_ADDA_PR_REGMAP to fix possible link errors for some configurations: sound/soc/sunxi/sun50i-codec-analog.o: In function `sun50i_codec_analog_probe': sun50i-codec-analog.c:(.text+0x62): undefined reference to `sun8i_adda_pr_regmap_init' Fixes: 42371f327df0 ("ASoC: sunxi: Add new driver for Allwinner A64 codec's analog path controls") Signed-off-by: Vasily Khoruzhick <anarsoul@gmail.com> Reviewed-by: Andre Przywara <andre.przywara@arm.com> Signed-off-by: Mark Brown <broonie@kernel.org>
| * ASoC: rsnd: fixup clock start checkerKuninori Morimoto2018-10-311-1/+1
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | commit 4d230d12710646 ("ASoC: rsnd: fixup not to call clk_get/set under non-atomic") fixuped clock start timing. But it exchanged clock start checker from ssi->usrcnt to ssi->rate. Current rsnd_ssi_master_clk_start() is called from .prepare, but some player (for example GStreamer) might calls it many times. In such case, the checker might returns error even though it was not error. It should check ssi->usrcnt instead of ssi->rate. This patch fixup it. Without this patch, GStreamer can't switch 48kHz / 44.1kHz. Reported-by: Yusuke Goda <yusuke.goda.sx@renesas.com> Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com> Tested-by: Yusuke Goda <yusuke.goda.sx@renesas.com> Signed-off-by: Mark Brown <broonie@kernel.org>
| * ASoC: stm32: sai: fix noderef.cocci warningskbuild test robot2018-10-241-1/+1
| | | | | | | | | | | | | | | | | | | | | | | | | | | | sound/soc/stm/stm32_sai_sub.c:393:26-32: ERROR: application of sizeof to pointer sizeof when applied to a pointer typed expression gives the size of the pointer Generated by: scripts/coccinelle/misc/noderef.cocci Fixes: 8307b2afd386 ("ASoC: stm32: sai: set sai as mclk clock provider") CC: Olivier Moysan <olivier.moysan@st.com> Signed-off-by: kbuild test robot <fengguang.wu@intel.com> Signed-off-by: Mark Brown <broonie@kernel.org>
* | ALSA: hda/realtek - Support ALC300Kailang Yang2018-11-271-0/+8
| | | | | | | | | | | | | | | | | | | | | | This patch will enable ALC300. [ It's almost equivalent with other ALC269-compatible ones, and apparently has no loopback mixer -- tiwai ] Signed-off-by: Kailang Yang <kailang@realtek.com> Cc: <stable@vger.kernel.org> Signed-off-by: Takashi Iwai <tiwai@suse.de>
* | ALSA: hda/realtek - Add auto-mute quirk for HP Spectre x360 laptopGirija Kumar Kasinadhuni2018-11-271-0/+7
| | | | | | | | | | | | | | | | | | | | This device makes a loud buzzing sound when a headphone is inserted while playing audio at full volume through the speaker. Fixes: bbf8ff6b1d2a ("ALSA: hda/realtek - Fixup for HP x360 laptops with B&O speakers") Signed-off-by: Girija Kumar Kasinadhuni <gkumar@neverware.com> Cc: <stable@vger.kernel.org> Signed-off-by: Takashi Iwai <tiwai@suse.de>
* | ALSA: hda/realtek - fix the pop noise on headphone for lenovo laptopsHui Wang2018-11-261-0/+20
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | We have several Lenovo laptops with the codec alc285, when playing sound via headphone, we can hear click/pop noise in the headphone, if we let the headphone share the DAC of NID 0x2 with the speaker, the noise disappears. The Lenovo laptops here include P52, P72, X1 yoda2 and X1 carbon. I have tried to set preferred_dacs and override_conn, but neither of them worked. Thanks for Kailang, he told me to invalidate the NID 0x3 through override_wcaps. BugLink: https://bugs.launchpad.net/bugs/1805079 Cc: <stable@vger.kernel.org> Signed-off-by: Kailang Yang <kailang@realtek.com> Signed-off-by: Hui Wang <hui.wang@canonical.com> Signed-off-by: Takashi Iwai <tiwai@suse.de>
* | ALSA: control: Fix race between adding and removing a user elementTakashi Iwai2018-11-241-35/+45
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | The procedure for adding a user control element has some window opened for race against the concurrent removal of a user element. This was caught by syzkaller, hitting a KASAN use-after-free error. This patch addresses the bug by wrapping the whole procedure to add a user control element with the card->controls_rwsem, instead of only around the increment of card->user_ctl_count. This required a slight code refactoring, too. The function snd_ctl_add() is split to two parts: a core function to add the control element and a part calling it. The former is called from the function for adding a user control element inside the controls_rwsem. One change to be noted is that snd_ctl_notify() for adding a control element gets called inside the controls_rwsem as well while it was called outside the rwsem. But this should be OK, as snd_ctl_notify() takes another (finer) rwlock instead of rwsem, and the call of snd_ctl_notify() inside rwsem is already done in another code path. Reported-by: syzbot+dc09047bce3820621ba2@syzkaller.appspotmail.com Cc: <stable@vger.kernel.org> Signed-off-by: Takashi Iwai <tiwai@suse.de>
* | ALSA: sparc: Fix invalid snd_free_pages() at error pathTakashi Iwai2018-11-241-6/+2
| | | | | | | | | | | | | | | | | | | | | | | | | | Some spurious calls of snd_free_pages() have been overlooked and remain in the error paths of sparc cs4231 driver code. Since runtime->dma_area is managed by the PCM core helper, we shouldn't release manually. Drop the superfluous calls. Reviewed-by: Takashi Sakamoto <o-takashi@sakamocchi.jp> Cc: <stable@vger.kernel.org> Signed-off-by: Takashi Iwai <tiwai@suse.de>
* | ALSA: wss: Fix invalid snd_free_pages() at error pathTakashi Iwai2018-11-241-2/+0
| | | | | | | | | | | | | | | | | | | | | | | | Some spurious calls of snd_free_pages() have been overlooked and remain in the error paths of wss driver code. Since runtime->dma_area is managed by the PCM core helper, we shouldn't release manually. Drop the superfluous calls. Reviewed-by: Takashi Sakamoto <o-takashi@sakamocchi.jp> Cc: <stable@vger.kernel.org> Signed-off-by: Takashi Iwai <tiwai@suse.de>
* | ALSA: hda/realtek - fix headset mic detection for MSI MS-B171Anisse Astier2018-11-231-0/+1
| | | | | | | | | | | | | | | | | | | | | | | | MSI Cubi N 8GL (MS-B171) needs the same fixup as its older model, the MS-B120, in order for the headset mic to be properly detected. They both use a single 3-way jack for both mic and headset with an ALC283 codec, with the same pins used. Cc: stable@vger.kernel.org Signed-off-by: Anisse Astier <anisse@astier.eu> Signed-off-by: Takashi Iwai <tiwai@suse.de>
* | ALSA: hda: Add ASRock N68C-S UCC the power_save blacklistHans de Goede2018-11-231-0/+2
| | | | | | | | | | | | | | | | | | | | | | Power-saving is causing plops on audio start/stop on the built-in audio of the nForce 430 based ASRock N68C-S UCC motherboard, add this model to the power_save blacklist. BugLink: https://bugzilla.redhat.com/show_bug.cgi?id=1525104 Cc: <stable@vger.kernel.org> Signed-off-by: Hans de Goede <hdegoede@redhat.com> Signed-off-by: Takashi Iwai <tiwai@suse.de>
* | ALSA: ac97: Fix incorrect bit shift at AC97-SPSA control writeTakashi Iwai2018-11-231-1/+1
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | The function snd_ac97_put_spsa() gets the bit shift value from the associated private_value, but it extracts too much; the current code extracts 8 bit values in bits 8-15, but this is a combination of two nibbles (bits 8-11 and bits 12-15) for left and right shifts. Due to the incorrect bits extraction, the actual shift may go beyond the 32bit value, as spotted recently by UBSAN check: UBSAN: Undefined behaviour in sound/pci/ac97/ac97_codec.c:836:7 shift exponent 68 is too large for 32-bit type 'int' This patch fixes the shift value extraction by masking the properly with 0x0f instead of 0xff. Reported-and-tested-by: Meelis Roos <mroos@linux.ee> Cc: <stable@vger.kernel.org> Signed-off-by: Takashi Iwai <tiwai@suse.de>
* | ALSA: hda/ca0132 - fix AE-5 pincfgConnor McAdams2018-11-191-1/+1
| | | | | | | | | | | | | | | | | | This patch fixes the pincfg assignment for the AE-5, which was previously using the Recon3D pincfg's by mistake. Fixes: d06feaf02fe6 ("ALSA: hda/ca0132 - Add pincfg for AE-5") Signed-off-by: Connor McAdams <conmanx360@gmail.com> Signed-off-by: Takashi Iwai <tiwai@suse.de>
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