March 2020 - Alsa-devel - mailman.alsa-project.org

[PATCH AUTOSEL 5.5 26/28] ALSA: pcm: oss: Remove WARNING from snd_pcm_plug_alloc() checks
by Sasha Levin 27 Mar '20

27 Mar '20

From: Takashi Iwai <tiwai(a)suse.de> [ Upstream commit 5461e0530c222129dfc941058be114b5cbc00837 ] The return value checks in snd_pcm_plug_alloc() are covered with snd_BUG_ON() macro that may trigger a kernel WARNING depending on the kconfig. But since the error condition can be triggered by a weird user space parameter passed to OSS layer, we shouldn't give the kernel stack trace just for that. As it's a normal error condition, let's remove snd_BUG_ON() macro usage there. Reported-by: syzbot+2a59ee7a9831b264f45e(a)syzkaller.appspotmail.com Cc: <stable(a)vger.kernel.org> Link: https://lore.kernel.org/r/20200312155730.7520-1-tiwai@suse.de Signed-off-by: Takashi Iwai <tiwai(a)suse.de> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/core/oss/pcm_plugin.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/core/oss/pcm_plugin.c b/sound/core/oss/pcm_plugin.c index 9b588c6a6f099..732bbede7ebfd 100644 --- a/sound/core/oss/pcm_plugin.c +++ b/sound/core/oss/pcm_plugin.c @@ -111,7 +111,7 @@ int snd_pcm_plug_alloc(struct snd_pcm_substream *plug, snd_pcm_uframes_t frames) while (plugin->next) { if (plugin->dst_frames) frames = plugin->dst_frames(plugin, frames); - if (snd_BUG_ON((snd_pcm_sframes_t)frames <= 0)) + if ((snd_pcm_sframes_t)frames <= 0) return -ENXIO; plugin = plugin->next; err = snd_pcm_plugin_alloc(plugin, frames); @@ -123,7 +123,7 @@ int snd_pcm_plug_alloc(struct snd_pcm_substream *plug, snd_pcm_uframes_t frames) while (plugin->prev) { if (plugin->src_frames) frames = plugin->src_frames(plugin, frames); - if (snd_BUG_ON((snd_pcm_sframes_t)frames <= 0)) + if ((snd_pcm_sframes_t)frames <= 0) return -ENXIO; plugin = plugin->prev; err = snd_pcm_plugin_alloc(plugin, frames); -- 2.20.1

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[PATCH AUTOSEL 5.5 25/28] ALSA: line6: Fix endless MIDI read loop
by Sasha Levin 27 Mar '20

27 Mar '20

From: Takashi Iwai <tiwai(a)suse.de> [ Upstream commit d683469b3c93d7e2afd39e6e1970f24700eb7a68 ] The MIDI input event parser of the LINE6 driver may enter into an endless loop when the unexpected data sequence is given, as it tries to continue the secondary bytes without termination. Also, when the input data is too short, the parser returns a negative error, while the caller doesn't handle it properly. This would lead to the unexpected behavior as well. This patch addresses those issues by checking the return value correctly and handling the one-byte event in the parser properly. The bug was reported by syzkaller. Reported-by: syzbot+cce32521ee0a824c21f7(a)syzkaller.appspotmail.com Cc: <stable(a)vger.kernel.org> Link: https://lore.kernel.org/r/000000000000033087059f8f8fa3@google.com Link: https://lore.kernel.org/r/20200309095922.30269-1-tiwai@suse.de Signed-off-by: Takashi Iwai <tiwai(a)suse.de> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/usb/line6/driver.c | 2 +- sound/usb/line6/midibuf.c | 2 +- 2 files changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/usb/line6/driver.c b/sound/usb/line6/driver.c index b5a3f754a4f13..4f096685ed654 100644 --- a/sound/usb/line6/driver.c +++ b/sound/usb/line6/driver.c @@ -305,7 +305,7 @@ static void line6_data_received(struct urb *urb) line6_midibuf_read(mb, line6->buffer_message, LINE6_MIDI_MESSAGE_MAXLEN); - if (done == 0) + if (done <= 0) break; line6->message_length = done; diff --git a/sound/usb/line6/midibuf.c b/sound/usb/line6/midibuf.c index 8d6eefa0d9362..6a70463f82c4e 100644 --- a/sound/usb/line6/midibuf.c +++ b/sound/usb/line6/midibuf.c @@ -159,7 +159,7 @@ int line6_midibuf_read(struct midi_buffer *this, unsigned char *data, int midi_length_prev = midibuf_message_length(this->command_prev); - if (midi_length_prev > 0) { + if (midi_length_prev > 1) { midi_length = midi_length_prev - 1; repeat = 1; } else -- 2.20.1

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[PATCH AUTOSEL 5.5 24/28] ALSA: pcm: oss: Avoid plugin buffer overflow
by Sasha Levin 27 Mar '20

27 Mar '20

From: Takashi Iwai <tiwai(a)suse.de> [ Upstream commit f2ecf903ef06eb1bbbfa969db9889643d487e73a ] Each OSS PCM plugins allocate its internal buffer per pre-calculation of the max buffer size through the chain of plugins (calling src_frames and dst_frames callbacks). This works for most plugins, but the rate plugin might behave incorrectly. The calculation in the rate plugin involves with the fractional position, i.e. it may vary depending on the input position. Since the buffer size pre-calculation is always done with the offset zero, it may return a shorter size than it might be; this may result in the out-of-bound access as spotted by fuzzer. This patch addresses those possible buffer overflow accesses by simply setting the upper limit per the given buffer size for each plugin before src_frames() and after dst_frames() calls. Reported-by: syzbot+e1fe9f44fb8ecf4fb5dd(a)syzkaller.appspotmail.com Cc: <stable(a)vger.kernel.org> Link: https://lore.kernel.org/r/000000000000b25ea005a02bcf21@google.com Link: https://lore.kernel.org/r/20200309082148.19855-1-tiwai@suse.de Signed-off-by: Takashi Iwai <tiwai(a)suse.de> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/core/oss/pcm_plugin.c | 8 ++++++++ 1 file changed, 8 insertions(+) diff --git a/sound/core/oss/pcm_plugin.c b/sound/core/oss/pcm_plugin.c index 31cb2acf8afcc..9b588c6a6f099 100644 --- a/sound/core/oss/pcm_plugin.c +++ b/sound/core/oss/pcm_plugin.c @@ -209,6 +209,8 @@ snd_pcm_sframes_t snd_pcm_plug_client_size(struct snd_pcm_substream *plug, snd_p if (stream == SNDRV_PCM_STREAM_PLAYBACK) { plugin = snd_pcm_plug_last(plug); while (plugin && drv_frames > 0) { + if (drv_frames > plugin->buf_frames) + drv_frames = plugin->buf_frames; plugin_prev = plugin->prev; if (plugin->src_frames) drv_frames = plugin->src_frames(plugin, drv_frames); @@ -220,6 +222,8 @@ snd_pcm_sframes_t snd_pcm_plug_client_size(struct snd_pcm_substream *plug, snd_p plugin_next = plugin->next; if (plugin->dst_frames) drv_frames = plugin->dst_frames(plugin, drv_frames); + if (drv_frames > plugin->buf_frames) + drv_frames = plugin->buf_frames; plugin = plugin_next; } } else @@ -248,11 +252,15 @@ snd_pcm_sframes_t snd_pcm_plug_slave_size(struct snd_pcm_substream *plug, snd_pc if (frames < 0) return frames; } + if (frames > plugin->buf_frames) + frames = plugin->buf_frames; plugin = plugin_next; } } else if (stream == SNDRV_PCM_STREAM_CAPTURE) { plugin = snd_pcm_plug_last(plug); while (plugin) { + if (frames > plugin->buf_frames) + frames = plugin->buf_frames; plugin_prev = plugin->prev; if (plugin->src_frames) { frames = plugin->src_frames(plugin, frames); -- 2.20.1

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[PATCH AUTOSEL 5.5 04/28] ALSA: hda/realtek: Fix pop noise on ALC225
by Sasha Levin 27 Mar '20

27 Mar '20

From: Kai-Heng Feng <kai.heng.feng(a)canonical.com> [ Upstream commit 3b36b13d5e69d6f51ff1c55d1b404a74646c9757 ] Commit 317d9313925c ("ALSA: hda/realtek - Set default power save node to 0") makes the ALC225 have pop noise on S3 resume and cold boot. So partially revert this commit for ALC225 to fix the regression. Fixes: 317d9313925c ("ALSA: hda/realtek - Set default power save node to 0") BugLink: https://bugs.launchpad.net/bugs/1866357 Signed-off-by: Kai-Heng Feng <kai.heng.feng(a)canonical.com> Link: https://lore.kernel.org/r/20200311061328.17614-1-kai.heng.feng@canonical.com Signed-off-by: Takashi Iwai <tiwai(a)suse.de> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/pci/hda/patch_realtek.c | 2 ++ 1 file changed, 2 insertions(+) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 835af7d2bbd4d..6afa8e195044a 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -8051,6 +8051,8 @@ static int patch_alc269(struct hda_codec *codec) spec->gen.mixer_nid = 0; break; case 0x10ec0225: + codec->power_save_node = 1; + /* fall through */ case 0x10ec0295: case 0x10ec0299: spec->codec_variant = ALC269_TYPE_ALC225; -- 2.20.1

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[RFC][PATCH] sound: qcom: Kconfig: Tweak dependencies on SND_SOC_SDM845
by John Stultz 26 Mar '20

26 Mar '20

CROS_EC isn't strictly required for audio to work on other SDM845 platforms (like the Dragonboard 845c). So lets remove the dependency and select the related CROS_EC options if CROS_EC is already enabled. Cc: Srinivas Kandagatla <srinivas.kandagatla(a)linaro.org> Cc: Rohit kumar <rohitkr(a)codeaurora.org> Cc: Patrick Lai <plai(a)codeaurora.org> Cc: Banajit Goswami <bgoswami(a)codeaurora.org> Cc: Liam Girdwood <lgirdwood(a)gmail.com> Cc: Mark Brown <broonie(a)kernel.org> Cc: Jaroslav Kysela <perex(a)perex.cz> Cc: Takashi Iwai <tiwai(a)suse.com> Cc: alsa-devel(a)alsa-project.org Signed-off-by: John Stultz <john.stultz(a)linaro.org> --- sound/soc/qcom/Kconfig | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/soc/qcom/Kconfig b/sound/soc/qcom/Kconfig index 6530d2462a9e..6f1bc6ea130e 100644 --- a/sound/soc/qcom/Kconfig +++ b/sound/soc/qcom/Kconfig @@ -99,12 +99,12 @@ config SND_SOC_MSM8996 config SND_SOC_SDM845 tristate "SoC Machine driver for SDM845 boards" - depends on QCOM_APR && CROS_EC && I2C + depends on QCOM_APR && I2C select SND_SOC_QDSP6 select SND_SOC_QCOM_COMMON select SND_SOC_RT5663 select SND_SOC_MAX98927 - select SND_SOC_CROS_EC_CODEC + select SND_SOC_CROS_EC_CODEC if CROS_EC help To add support for audio on Qualcomm Technologies Inc. SDM845 SoC-based systems. -- 2.17.1

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[PATCH 0/4] ASoC: Intel: machine driver updates
by Pierre-Louis Bossart 26 Mar '20

26 Mar '20

One correction for GeminiLake and 2 additional machine drivers for Chromebooks. Curtis Malainey (1): ASoC: Intel: Make glk+rt5682 echo ref dynamic Sathyanarayana Nujella (2): ASoC: Intel: sof_rt5682: Add support for tgl-max98373-rt5682 ASoC: Intel: common: Add mach table for tgl-max98373-rt5682 Yong Zhi (1): ASoC: intel: sof_da7219_max98373: Add speaker switch sound/soc/intel/boards/Kconfig | 1 + sound/soc/intel/boards/Makefile | 2 +- sound/soc/intel/boards/glk_rt5682_max98357a.c | 1 + sound/soc/intel/boards/sof_da7219_max98373.c | 67 +++++++++------- sound/soc/intel/boards/sof_maxim_common.c | 80 +++++++++++++++++++ sound/soc/intel/boards/sof_maxim_common.h | 24 ++++++ sound/soc/intel/boards/sof_rt5682.c | 21 +++++ .../intel/common/soc-acpi-intel-tgl-match.c | 13 +++ 8 files changed, 178 insertions(+), 31 deletions(-) create mode 100644 sound/soc/intel/boards/sof_maxim_common.c create mode 100644 sound/soc/intel/boards/sof_maxim_common.h -- 2.20.1

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[PATCH 0/5] ASoC: SOF: cleanups and improvements
by Pierre-Louis Bossart 26 Mar '20

26 Mar '20

Couple of small patches to improve error handling, inits, logs. Hope we can have this for 5.7? Guennadi Liakhovetski (2): ASoC: SOF: (cosmetic) use for_each_pcm_streams() in sof_dai_load() ASoC: SOF: fix uninitialised "work" with VirtIO Kai Vehmanen (2): ASoC: SOF: Intel: hda: do not leave clock gating off upon error ASoC: SOF: Intel: hda: call codec wake at chip init Ranjani Sridharan (1): ASoC: SOF: Intel: hda: Improve DSP state logging sound/soc/sof/intel/hda-ctrl.c | 15 +++++++++--- sound/soc/sof/intel/hda-dsp.c | 43 ++++++++++++++++++++++++++++++++-- sound/soc/sof/pcm.c | 4 +--- sound/soc/sof/sof-audio.h | 3 +++ sound/soc/sof/topology.c | 17 ++++++++++---- 5 files changed, 69 insertions(+), 13 deletions(-) -- 2.20.1

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Applied "ASoC: SOF: (cosmetic) use for_each_pcm_streams() in sof_dai_load()" to the asoc tree
by Mark Brown 26 Mar '20

26 Mar '20

The patch ASoC: SOF: (cosmetic) use for_each_pcm_streams() in sof_dai_load() has been applied to the asoc tree at https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git All being well this means that it will be integrated into the linux-next tree (usually sometime in the next 24 hours) and sent to Linus during the next merge window (or sooner if it is a bug fix), however if problems are discovered then the patch may be dropped or reverted. You may get further e-mails resulting from automated or manual testing and review of the tree, please engage with people reporting problems and send followup patches addressing any issues that are reported if needed. If any updates are required or you are submitting further changes they should be sent as incremental updates against current git, existing patches will not be replaced. Please add any relevant lists and maintainers to the CCs when replying to this mail. Thanks, Mark >From c688cf1d3a2cc1bca5737e7849325b3ac8e69a41 Mon Sep 17 00:00:00 2001 From: Guennadi Liakhovetski <guennadi.liakhovetski(a)linux.intel.com> Date: Wed, 25 Mar 2020 16:12:30 -0500 Subject: [PATCH] ASoC: SOF: (cosmetic) use for_each_pcm_streams() in sof_dai_load() Use for_each_pcm_streams() to enumerate streams in sof_dai_load() instead of doing that manually. Signed-off-by: Guennadi Liakhovetski <guennadi.liakhovetski(a)linux.intel.com> Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> Reviewed-by: Kai Vehmanen <kai.vehmanen(a)linux.intel.com> Reviewed-by: Ranjani Sridharan <ranjani.sridharan(a)linux.intel.com> Link: https://lore.kernel.org/r/20200325211233.27394-3-pierre-louis.bossart@linux… Signed-off-by: Mark Brown <broonie(a)kernel.org> --- sound/soc/sof/topology.c | 13 ++++++++----- 1 file changed, 8 insertions(+), 5 deletions(-) diff --git a/sound/soc/sof/topology.c b/sound/soc/sof/topology.c index 058de94fb8cf..54437caf9488 100644 --- a/sound/soc/sof/topology.c +++ b/sound/soc/sof/topology.c @@ -2448,7 +2448,7 @@ static int sof_dai_load(struct snd_soc_component *scomp, int index, struct snd_soc_tplg_stream_caps *caps; struct snd_soc_tplg_private *private = &pcm->priv; struct snd_sof_pcm *spcm; - int stream = SNDRV_PCM_STREAM_PLAYBACK; + int stream; int ret = 0; /* nothing to do for BEs atm */ @@ -2460,8 +2460,9 @@ static int sof_dai_load(struct snd_soc_component *scomp, int index, return -ENOMEM; spcm->scomp = scomp; - spcm->stream[SNDRV_PCM_STREAM_PLAYBACK].comp_id = COMP_ID_UNASSIGNED; - spcm->stream[SNDRV_PCM_STREAM_CAPTURE].comp_id = COMP_ID_UNASSIGNED; + + for_each_pcm_streams(stream) + spcm->stream[stream].comp_id = COMP_ID_UNASSIGNED; spcm->pcm = *pcm; dev_dbg(scomp->dev, "tplg: load pcm %s\n", pcm->dai_name); @@ -2482,8 +2483,10 @@ static int sof_dai_load(struct snd_soc_component *scomp, int index, if (!spcm->pcm.playback) goto capture; + stream = SNDRV_PCM_STREAM_PLAYBACK; + dev_vdbg(scomp->dev, "tplg: pcm %s stream tokens: playback d0i3:%d\n", - spcm->pcm.pcm_name, spcm->stream[0].d0i3_compatible); + spcm->pcm.pcm_name, spcm->stream[stream].d0i3_compatible); caps = &spcm->pcm.caps[stream]; @@ -2513,7 +2516,7 @@ static int sof_dai_load(struct snd_soc_component *scomp, int index, return ret; dev_vdbg(scomp->dev, "tplg: pcm %s stream tokens: capture d0i3:%d\n", - spcm->pcm.pcm_name, spcm->stream[1].d0i3_compatible); + spcm->pcm.pcm_name, spcm->stream[stream].d0i3_compatible); caps = &spcm->pcm.caps[stream]; -- 2.20.1

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Applied "ASoC: SOF: fix uninitialised "work" with VirtIO" to the asoc tree
by Mark Brown 26 Mar '20

26 Mar '20

The patch ASoC: SOF: fix uninitialised "work" with VirtIO has been applied to the asoc tree at https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git All being well this means that it will be integrated into the linux-next tree (usually sometime in the next 24 hours) and sent to Linus during the next merge window (or sooner if it is a bug fix), however if problems are discovered then the patch may be dropped or reverted. You may get further e-mails resulting from automated or manual testing and review of the tree, please engage with people reporting problems and send followup patches addressing any issues that are reported if needed. If any updates are required or you are submitting further changes they should be sent as incremental updates against current git, existing patches will not be replaced. Please add any relevant lists and maintainers to the CCs when replying to this mail. Thanks, Mark >From 9ef91cad92ba75d17d5a5203230746c9d9009705 Mon Sep 17 00:00:00 2001 From: Guennadi Liakhovetski <guennadi.liakhovetski(a)linux.intel.com> Date: Wed, 25 Mar 2020 16:12:31 -0500 Subject: [PATCH] ASoC: SOF: fix uninitialised "work" with VirtIO In the VirtIO case the sof_pcm_open() function isn't called on the host during guest streaming, which then leaves "work" structures uninitialised. However it is then used to handle position update messages from the DSP. Move their initialisation to immediately after allocation of the containing structure. Signed-off-by: Guennadi Liakhovetski <guennadi.liakhovetski(a)linux.intel.com> Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> Reviewed-by: Kai Vehmanen <kai.vehmanen(a)linux.intel.com> Reviewed-by: Ranjani Sridharan <ranjani.sridharan(a)linux.intel.com> Link: https://lore.kernel.org/r/20200325211233.27394-4-pierre-louis.bossart@linux… Signed-off-by: Mark Brown <broonie(a)kernel.org> --- sound/soc/sof/pcm.c | 4 +--- sound/soc/sof/sof-audio.h | 3 +++ sound/soc/sof/topology.c | 6 +++++- 3 files changed, 9 insertions(+), 4 deletions(-) diff --git a/sound/soc/sof/pcm.c b/sound/soc/sof/pcm.c index f4769e19965a..47cd741f2a8c 100644 --- a/sound/soc/sof/pcm.c +++ b/sound/soc/sof/pcm.c @@ -57,7 +57,7 @@ static int sof_pcm_dsp_params(struct snd_sof_pcm *spcm, struct snd_pcm_substream /* * sof pcm period elapse work */ -static void sof_pcm_period_elapsed_work(struct work_struct *work) +void snd_sof_pcm_period_elapsed_work(struct work_struct *work) { struct snd_sof_pcm_stream *sps = container_of(work, struct snd_sof_pcm_stream, @@ -475,8 +475,6 @@ static int sof_pcm_open(struct snd_soc_component *component, dev_dbg(component->dev, "pcm: open stream %d dir %d\n", spcm->pcm.pcm_id, substream->stream); - INIT_WORK(&spcm->stream[substream->stream].period_elapsed_work, - sof_pcm_period_elapsed_work); caps = &spcm->pcm.caps[substream->stream]; diff --git a/sound/soc/sof/sof-audio.h b/sound/soc/sof/sof-audio.h index eacd10e4da11..bf65f31af858 100644 --- a/sound/soc/sof/sof-audio.h +++ b/sound/soc/sof/sof-audio.h @@ -11,6 +11,8 @@ #ifndef __SOUND_SOC_SOF_AUDIO_H #define __SOUND_SOC_SOF_AUDIO_H +#include <linux/workqueue.h> + #include <sound/soc.h> #include <sound/control.h> #include <sound/sof/stream.h> /* needs to be included before control.h */ @@ -189,6 +191,7 @@ struct snd_sof_pcm *snd_sof_find_spcm_comp(struct snd_soc_component *scomp, struct snd_sof_pcm *snd_sof_find_spcm_pcm_id(struct snd_soc_component *scomp, unsigned int pcm_id); void snd_sof_pcm_period_elapsed(struct snd_pcm_substream *substream); +void snd_sof_pcm_period_elapsed_work(struct work_struct *work); /* * Mixer IPC diff --git a/sound/soc/sof/topology.c b/sound/soc/sof/topology.c index 54437caf9488..fe8ba3e05e08 100644 --- a/sound/soc/sof/topology.c +++ b/sound/soc/sof/topology.c @@ -9,6 +9,7 @@ // #include <linux/firmware.h> +#include <linux/workqueue.h> #include <sound/tlv.h> #include <sound/pcm_params.h> #include <uapi/sound/sof/tokens.h> @@ -2461,8 +2462,11 @@ static int sof_dai_load(struct snd_soc_component *scomp, int index, spcm->scomp = scomp; - for_each_pcm_streams(stream) + for_each_pcm_streams(stream) { spcm->stream[stream].comp_id = COMP_ID_UNASSIGNED; + INIT_WORK(&spcm->stream[stream].period_elapsed_work, + snd_sof_pcm_period_elapsed_work); + } spcm->pcm = *pcm; dev_dbg(scomp->dev, "tplg: load pcm %s\n", pcm->dai_name); -- 2.20.1

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[PATCH 0/2] ASoC: rt1308-sdw: configure amplifier with set_tdm_slot()
by Pierre-Louis Bossart 26 Mar '20

26 Mar '20

When two (or more) amplifiers are on the same link, the integrator may: a) assign dedicated slots for each of the amplifiers. b) provide the same configuration to all amplifiers, and rely on additional controls/processing in the amplifier to generate different outputs. case a) was the initial direction for SoundWire and is required for amplifiers with limited capabilities, but to deal with orientation or 'posture' changes it's easier to implement case b) when the amplifier can deal with multiple channels. This patchset suggest the use of the set_tdm_slot() API to define which of the channels will be consumed by what amplifiers. This maps well with SoundWire's 'ChannelEnable' registers. The notion of slot_width is however irrelevant here and ignored, and SoundWire ports are typically single direction, so only one of the two masks shall be used. Pierre-Louis Bossart (2): ASoC: rt1308-sdw: add set_tdm_slot() support ASoC: rt1308-sdw: use slot and rx_mask to configure stream sound/soc/codecs/rt1308-sdw.c | 38 +++++++++++++++++++++++++++++++---- sound/soc/codecs/rt1308-sdw.h | 2 ++ 2 files changed, 36 insertions(+), 4 deletions(-) -- 2.20.1

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