- 23 Jun, 2020 1 commit
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Christoffer Nielsen authored
Similar to the Kingston HyperX AMP, the Kingston HyperX Cloud Alpha S (0951:0x16ea) uses two interfaces, but only the second interface contains the capture stream. This patch delays the registration until the second interface appears. Signed-off-by:
Christoffer Nielsen <cn@obviux.dk> Cc: <stable@vger.kernel.org> Link: https://lore.kernel.org/r/CAOtG2YHOM3zy+ed9KS-J4HkZo_QGzcUG9MigSp4e4_-13r6B=Q@mail.gmail.comSigned-off-by:
Takashi Iwai <tiwai@suse.de>
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- 22 Jun, 2020 1 commit
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Takashi Iwai authored
Merge tag 'asoc-fix-v5.8-rc2' of https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound into for-linus ASoC: Fixes for v5.8 This is a collection of mostly small fixes, mostly fixing fallout from some of the DPCM changes that went in last time around which shook out some issues on i.MX and Qualcomm platforms. The addition of a managed version of snd_soc_register_dai() is to fix resource leaks. There's also a few new device IDs for x86 systems.
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- 18 Jun, 2020 3 commits
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Qiushi Wu authored
Calling pm_runtime_get_sync increments the counter even in case of failure, causing incorrect ref count if pm_runtime_put is not called in error handling paths. Call pm_runtime_put if pm_runtime_get_sync fails. Fixes: fc05a5b2 ("ASoC: rockchip: add support for pdm controller") Signed-off-by:
Qiushi Wu <wu000273@umn.edu> Reviewed-by:
Heiko Stuebner <heiko@sntech.de> Link: https://lore.kernel.org/r/20200613205158.27296-1-wu000273@umn.eduSigned-off-by:
Mark Brown <broonie@kernel.org>
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Ravulapati Vishnu vardhan rao authored
The steps to reproduce: Record from the internal mic : (arecord -D hw:1,2 -f dat /dev/null -V stereos) Record from the headphone mic: (arecord -D hw:1,0 -f dat /dev/null -V stereos) Kill the recording from internal mic. We can see the recording from the headphone mic is broken. This patch rectifies the issue reported. Signed-off-by:
Ravulapati Vishnu vardhan rao <Vishnuvardhanrao.Ravulapati@amd.com> Link: https://lore.kernel.org/r/20200618072653.27103-1-Vishnuvardhanrao.Ravulapati@amd.comSigned-off-by:
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Pierre-Louis Bossart authored
Mirror PCI ids used for SOF. Signed-off-by:
Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com> Reviewed-by:
Guennadi Liakhovetski <guennadi.liakhovetski@linux.intel.com> Reviewed-by:
Kai Vehmanen <kai.vehmanen@linux.intel.com> Link: https://lore.kernel.org/r/20200617164909.18225-1-pierre-louis.bossart@linux.intel.comSigned-off-by:
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- 17 Jun, 2020 7 commits
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Mark Brown authored
Merge series "ASoC: SOF: Intel: update PCI IDs" from Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>: Clean-up CometLake and add missing PCI IDs. Changes for the legacy driver are sent separately. Pierre-Louis Bossart (3): ASoC: Intel: SOF: merge COMETLAKE_LP and COMETLAKE_H ASoC: SOF: Intel: add PCI ID for CometLake-S ASoC: SOF: Intel: add PCI IDs for ICL-H and TGL-H sound/hda/intel-dsp-config.c | 4 +--- sound/soc/intel/boards/Kconfig | 4 ++-- sound/soc/sof/intel/Kconfig | 29 ++++++++--------------------- sound/soc/sof/sof-pci-dev.c | 24 ++++++++++++++---------- 4 files changed, 25 insertions(+), 36 deletions(-) -- 2.20.1
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Pierre-Louis Bossart authored
kmemleak throws error reports on module load/unload tests, add snd_hdac_regmap_exit() in .remove(). While we are at it, also fix the error handling flow in .probe() to use snd_hdac_regmap_exit() if needed. Signed-off-by:
Daniel Baluta <daniel.baluta@gmail.com> Reviewed-by:
Rander Wang <rander.wang@linux.intel.com> Reviewed-by:
Bard Liao <yung-chuan.liao@linux.intel.com> Link: https://lore.kernel.org/r/20200617164144.17859-1-pierre-louis.bossart@linux.intel.comSigned-off-by:
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Pierre-Louis Bossart authored
Usually the DSP is not traditionally enabled on H skews but this might be used moving forward. Signed-off-by:
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Pierre-Louis Bossart authored
Mirror ID added for legacy HDaudio Signed-off-by:
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Pierre-Louis Bossart authored
We already have two configurations for CometLake, and a third one coming. On other platforms, we used a single Kconfig option, so we should follow the same trend by merging the two cases in a backwards compatible way. The backwards compatibility is handled by overloading the COMETLAKE_LP kconfig as COMETLAKE. In practice we've never seen a case where COMETLAKE_H is not selected along with COMETLAKE_LP, so keeping one of the two is enough. Signed-off-by:
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Kai-Heng Feng authored
There are two more HP systems control mute LED from HDA codec and need to expose micmute led class so SoF can control micmute LED. Add quirks to support them. Signed-off-by:
Kai-Heng Feng <kai.heng.feng@canonical.com> Cc: <stable@vger.kernel.org> Link: https://lore.kernel.org/r/20200617102906.16156-2-kai.heng.feng@canonical.comSigned-off-by:
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Takashi Iwai authored
With the recent full-duplex support of implicit feedback streams, an endpoint can be still running after closing the capture stream as long as the playback stream with the sync-endpoint is running. In such a state, the URBs are still be handled and they may call retire_data_urb callback, which tries to transfer the data from the PCM buffer. Since the PCM stream gets closed, this may lead to use-after-free. This patch adds the proper clearance of the callback at stopping the capture stream for addressing the possible UAF above. Fixes: 10ce77e4 ("ALSA: usb-audio: Add duplex sound support for USB devices using implicit feedback") Link: https://lore.kernel.org/r/20200616120921.12249-1-tiwai@suse.deSigned-off-by:
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- 16 Jun, 2020 2 commits
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Takashi Iwai authored
MSI GE63 laptop with ALC1220 codec requires the very same quirk (ALC1220_FIXUP_CLEVO_P950) as other MSI devices for the proper sound output. BugLink: https://bugzilla.kernel.org/show_bug.cgi?id=208057 Cc: <stable@vger.kernel.org> Link: https://lore.kernel.org/r/20200616132150.8778-1-tiwai@suse.deSigned-off-by:
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Shengjiu Wang authored
For mono channel, SSI will switch to Normal mode. In Normal mode and Network mode, the Word Length Control bits control the word length divider in clock generator, which is different with I2S Master mode (the word length is fixed to 32bit), it should be the value of params_width(hw_params). The condition "slots == 2" is not good for I2S Master mode, because for Network mode and Normal mode, the slots can also be 2. Then we need to use (ssi->i2s_net & SSI_SCR_I2S_MODE_MASK) to check if it is I2S Master mode. So we refine the formula for mono channel, otherwise there will be sound issue for S24_LE. Fixes: b0a7043d ("ASoC: fsl_ssi: Caculate bit clock rate using slot number and width") Signed-off-by:
Shengjiu Wang <shengjiu.wang@nxp.com> Reviewed-by:
Nicolin Chen <nicoleotsuka@gmail.com> Link: https://lore.kernel.org/r/034eff1435ff6ce300b6c781130cefd9db22ab9a.1592276147.git.shengjiu.wang@nxp.comSigned-off-by:
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- 15 Jun, 2020 11 commits
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Mark Brown authored
Merge series "ASoC: topology: fix use-after-free when removing components" from Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>: This patchset fixes a memory allocation issue and removes a 100% reproducible use-after-free report thrown by KASAN in automated module removal tests across multiple platforms. All the credit goes to Bard Liao for root-causing the issue. DAIs may be registered at the same time as a component, or when the topology is loaded. This two-step registration causes the memory for topology-based DAIs to allocated last, and conversely to be released first by devres, before the component is released and the DAIs removed from the component DAI list with snd_soc_unregister_dais(). When we remove a component, by the time we walk through its dai list to unregister all dais, the dais allocated by the topology have been freed already by devres and the list is corrupted with pointers that are no longer valid. The suggestion is to add an explicit devm_ based registration for topology-based dais, so that each dai is cleanly removed from the component dai list in the release operation before devres releases the allocated memory. Pierre-Louis Bossart (2): ASoC: soc-devres: add devm_snd_soc_register_dai() ASoC: soc-topology: use devm_snd_soc_register_dai() include/sound/soc.h | 4 ++++ sound/soc/soc-devres.c | 37 +++++++++++++++++++++++++++++++++++++ sound/soc/soc-topology.c | 3 +-- 3 files changed, 42 insertions(+), 2 deletions(-) -- 2.20.1
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Brent Lu authored
Port commit 6d011d50 ("ALSA: hda: Clear RIRB status before reading WP") from legacy HDA driver to fix the get response timeout issue. Current SOF driver does not suffer from this issue because sync write is enabled in hda_init. The issue will come back if the sync write is disabled for some reason. Signed-off-by:
Brent Lu <brent.lu@intel.com> Reviewed-by:
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Jack Yu authored
Update rt1015 default register value according to spec modification. Signed-off-by:
Jack Yu <jack.yu@realtek.com> Link: https://lore.kernel.org/r/20200615032433.31061-1-jack.yu@realtek.comSigned-off-by:
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Srinivas Kandagatla authored
Currently both FE and BE dai-links are configured bi-directional, However the DSP BE dais are only single directional, so set the directions as supported by the BE dais. Fixes: c25e295c (ASoC: qcom: Add support to parse common audio device nodes) Reported-by:
John Stultz <john.stultz@linaro.org> Signed-off-by:
Srinivas Kandagatla <srinivas.kandagatla@linaro.org> Tested-by:
Vinod Koul <vkoul@kernel.org> Link: https://lore.kernel.org/r/20200612123711.29130-2-srinivas.kandagatla@linaro.orgSigned-off-by:
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Srinivas Kandagatla authored
This patch adds support to q6afe_is_rx_port() to get direction of DSP BE dai port, this is useful for setting dailink directions correctly. Fixes: c25e295c (ASoC: qcom: Add support to parse common audio device nodes) Reported-by:
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Pierre-Louis Bossart authored
soc_dpcm_fe_runtime_update() is called for all dailinks, and we want to first discard all back-ends, then deal with front-ends. The existing code first reports an error with multi-cpu front-ends, and that check needs to be moved after we know that we are dealing with a front-end. Fixes: 6e1276a5 ('ASoC: Return error if the function does not support multi-cpu') Signed-off-by:
Ranjani Sridharan <ranjani.sridharan@linux.intel.com> Reviewed-by:
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derek.fang authored
According to ideal rt5682 CCF, the root clk is mclk. But in some platforms, mclk is not exported to CCF. In this condition, rt5682_register_dai_clks will not be called. This patch lets dai clks could be registered whether mclk exists or not. Signed-off-by:
derek.fang <derek.fang@realtek.com> Link: https://lore.kernel.org/r/1591938925-1070-5-git-send-email-derek.fang@realtek.comSigned-off-by:
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Pierre-Louis Bossart authored
Use devm_ to avoid use-after-free KASAN reports and simplify error handling. Signed-off-by:
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Pierre-Louis Bossart authored
The registration of DAIs may be done at two distinct times, once during a component registration and later when loading a topology. Since devm_ managed resources are freed in the reverse order they were allocated, when a component starts unregistering DAIs by walking through the DAI list, the memory allocated for the topology-registered DAIs was freed already, which leads to 100% reproducible KASAN use-after-free reports. This patch suggests a new devm_ function to force the DAI list to be updated prior to freeing the memory chunks referenced by the list pointers. Suggested-by:
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Christopher Swenson authored
Like the Line6 devices, the Rode Rodecaster Pro does not support UAC2_CS_RANGE and only supports a sample rate of 48 kHz. Tested against a Rode Rodecaster Pro. Tested-by:
Christopher Swenson <swenson@swenson.io> Signed-off-by:
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Yick W. Tse authored
fix error "clock source 41 is not valid, cannot use" [] New USB device found, idVendor=154e, idProduct=1002, bcdDevice= 1.00 [] New USB device strings: Mfr=1, Product=2, SerialNumber=0 [] Product: DCD-1500RE [] Manufacturer: D & M Holdings Inc. [] [] clock source 41 is not valid, cannot use [] usbcore: registered new interface driver snd-usb-audio Signed-off-by:
Yick W. Tse <y_w_tse@yahoo.com.hk> Cc: <stable@vger.kernel.org> Link: https://lore.kernel.org/r/1373857985.210365.1592048406997@mail.yahoo.comSigned-off-by:
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- 12 Jun, 2020 5 commits
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Shengjiu Wang authored
With EDMA, there is two dma channels can be used for dev_to_dev, one is from ASRC, one is from another peripheral (ESAI or SAI). If we select the dma channel of ASRC, there is an issue for ideal ratio case, the speed of copy data is faster than sample frequency, because ASRC output data is very fast in ideal ratio mode. So it is reasonable to use the dma channel of Back-End peripheral. then copying speed of DMA is controlled by data consumption speed in the peripheral FIFO, Signed-off-by:
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Shengjiu Wang authored
The dma channel has been requested by Back-End cpu dai driver already. If fsl_asrc_dma requests dma chan with same dma:tx symlink, then there will be below warning with SDMA. [ 48.174236] fsl-esai-dai 2024000.esai: Cannot create DMA dma:tx symlink So if we can reuse the dma channel of Back-End, then the issue can be fixed. In order to get the dma channel which is already requested in Back-End. we use the exported two functions (snd_soc_lookup_component_nolocked and soc_component_to_pcm). If we can get the dma channel, then reuse it, if can't, then request a new one. Signed-off-by:
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Shengjiu Wang authored
In DPCM case, Front-End needs to get the dma chan which has been requested by Back-End and reuse it. Signed-off-by:
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Shengjiu Wang authored
snd_soc_lookup_component_nolocked can be used for the DPCM case that Front-End needs to get the unused platform component but added by Back-End cpu dai driver. If the component is gotten, then we can get the dma chan created by Back-End component and reused it in Front-End. Signed-off-by:
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Laurence Tratt authored
This uses the same quirk as the Motu M2 and M4 to ensure the driver uses the audio interface's clock. Tested on an SSL2+. Signed-off-by:
Laurence Tratt <laurie@tratt.net> Cc: <stable@vger.kernel.org> Link: https://lore.kernel.org/r/20200612111807.dgnig6rwhmsl2bod@overdrive.tratt.netSigned-off-by:
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- 11 Jun, 2020 3 commits
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Aaron Plattner authored
These IDs are for upcoming NVIDIA chips with audio functions that are largely similar to the existing ones. Signed-off-by:
Aaron Plattner <aplattner@nvidia.com> Cc: <stable@vger.kernel.org> Link: https://lore.kernel.org/r/20200611180845.39942-1-aplattner@nvidia.comSigned-off-by:
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Srinivas Kandagatla authored
Successful send of EOS command does not indicate that EOS is actually finished, correct event to wait EOS is finished is EOS_RENDERED event. EOS_RENDERED means that the DSP has finished processing all the buffers for that particular session and stream. This patch fixes EOS handling! Fixes: 68fd8480 ("ASoC: qdsp6: q6asm: Add support to audio stream apis") Signed-off-by:
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Steve Lee authored
Update max98390_readable_register and max98390_volatile_reg Signed-off-by:
Steve Lee <steves.lee@maximintegrated.com> Link: https://lore.kernel.org/r/20200611094800.18422-1-steves.lee@maximintegrated.comSigned-off-by:
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- 10 Jun, 2020 1 commit
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Takashi Iwai authored
Merge tag 'asoc-fix-v5.8' of https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound into for-linus ASoC: Fixes for v5.8 A small pile of fixes that came in during the merge window, the DPCM fixes from Pierre are the most notable thing here.
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- 09 Jun, 2020 6 commits
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Mark Brown authored
Merge series "ASoC: Fix dailink checks for DPCM" from Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>: We've had a couple of changes that introduce regressions with the multi-cpu DAI solutions, and while trying to fix them we found additional inconsistencies that should also go to stable branches. Bard Liao (1): ASoC: core: only convert non DPCM link to DPCM link Pierre-Louis Bossart (3): ASoC: soc-pcm: dpcm: fix playback/capture checks ASoC: Intel: boards: replace capture_only by dpcm_capture ASoC: SOF: nocodec: conditionally set dpcm_capture/dpcm_playback flags sound/soc/intel/boards/glk_rt5682_max98357a.c | 2 +- sound/soc/intel/boards/kbl_da7219_max98927.c | 4 +- sound/soc/intel/boards/kbl_rt5663_max98927.c | 2 +- .../intel/boards/kbl_rt5663_rt5514_max98927.c | 2 +- sound/soc/soc-core.c | 22 ++++++++-- sound/soc/soc-pcm.c | 44 ++++++++++++++----- sound/soc/sof/nocodec.c | 6 ++- 7 files changed, 62 insertions(+), 20 deletions(-) base-commit: 8a9144c1 -- 2.20.1
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Hans de Goede authored
The Asus T101HA uses the default jack-detect mode 3, but instead of using an analog microphone it is using a DMIC on dmic-data-pin 1, like the Asus T100HA. Note unlike the T100HA its jack-detect is not inverted. Add a DMI quirk with the correct settings for this model. Signed-off-by:
Hans de Goede <hdegoede@redhat.com> Acked-by:
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Hans de Goede authored
The Toshiba Encore WT10-A tablet almost fully works with the default settings for Bay Trail CR devices. The only issue is that it uses a digital mic. connected the the DMIC1 input instead of an analog mic. Add a quirk for this model using the default settings with the input-map replaced with BYT_RT5640_DMIC1_MAP. Signed-off-by:
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Pierre-Louis Bossart authored
With additional checks on dailinks, we see errors such as [ 3.000418] sof-nocodec sof-nocodec: CPU DAI DMIC01 Pin for rtd NoCodec-6 does not support playback It's not clear why we set the dpcm_playback and dpcm_capture flags unconditionally, add a check on number of channels for each direction to avoid invalid configurations. Fixes: 8017b8fd ('ASoC: SOF: Add Nocodec machine driver support') Signed-off-by:
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Pierre-Louis Bossart authored
It's not clear why specific FE dailinks use capture_only flags, likely blind copy/paste from Chromebook driver to the other. Replace by dpcm_capture, this will make future alignment and removal of flags easier. Signed-off-by:
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Bard Liao authored
Additional checks for valid DAIs expose a corner case, where existing BE dailinks get modified, e.g. HDMI links are tagged with dpcm_capture=1 even if the DAIs are for playback. This patch makes those changes conditional and flags configuration issues when a BE dailink is has no_pcm=0 but dpcm_playback or dpcm_capture=1 (which makes no sense). As discussed on the alsa-devel mailing list, there are redundant flags for dpcm_playback, dpcm_capture, playback_only, capture_only. This will have to be cleaned-up in a future update. For now only correct and flag problematic configurations. Fixes: 218fe9b7 ("ASoC: soc-core: Set dpcm_playback / dpcm_capture") Suggested-by:
Daniel Baluta <daniel.baluta@nxp.com> Signed-off-by:
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