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

[PATCH] ALSA: usb-audio: Add implicit feedback quirk for Zoom UAC-2
by Keith Winstein 27 Oct '20

27 Oct '20

Hi folks, We're using a Zoom UAC-2 USB audio interface, which provides an async playback endpoint ("1 OUT (ASYNC)") and an async capture endpoint ("2 IN (ASYNC)"), both with 2-channel S32_LE in 44100, 48000, 88200, 96000, 176400, or 192000 samples per second. For whatever reason, the explicit feedback on the playback endpoint seems a bit unstable and also biased a little high. alsaloop -t 1000 gets repeated playback underruns and starts trying to resample the audio to adjust the rate (even though it's just looping from capture back to playback on the same device!). I've found that using implicit feedback produces much more stable results and doesn't seem to hurt anything else (patch below). I wasn't sure if the other "implicit feedback quirk" devices in sound/usb/pcm.c are also ASYNC devices that happen to perform better using implicit feedback instead (in which case adding the Zoom UAC-2 to the list seems pretty routine), or if the others are SYNC devices that only need help identifying the right endpoint and ifnum -- i.e., how unorthodox is it to use implicit feedback on an ASYNC device? Hopefully not very. Please let me know if you'd like any additional information or testing. Best regards, Keith >From 2c8d691d39ab2824c344ac2562a1e8605b3384ba Mon Sep 17 00:00:00 2001 From: Keith Winstein <keithw(a)cs.stanford.edu> Date: Sun, 25 Oct 2020 22:05:47 -0700 Subject: [PATCH] ALSA: usb-audio: Add implicit feedback quirk for Zoom UAC-2 Signed-off-by: Keith Winstein <keithw(a)cs.stanford.edu> Tested-by: Keith Winstein <keithw(a)cs.stanford.edu> --- sound/usb/pcm.c | 4 ++++ 1 file changed, 4 insertions(+) diff --git a/sound/usb/pcm.c b/sound/usb/pcm.c index b401ee894e1b..5723e8e69cf1 100644 --- a/sound/usb/pcm.c +++ b/sound/usb/pcm.c @@ -352,6 +352,10 @@ static int set_sync_ep_implicit_fb_quirk(struct snd_usb_substream *subs, ep = 0x81; ifnum = 2; goto add_sync_ep_from_ifnum; + case USB_ID(0x1686, 0xf029): /* ZOOM UAC-2 */ + ep = 0x82; + ifnum = 2; + goto add_sync_ep_from_ifnum; case USB_ID(0x1397, 0x0001): /* Behringer UFX1604 */ case USB_ID(0x1397, 0x0002): /* Behringer UFX1204 */ ep = 0x81; -- 2.25.1

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RE: [PATCH] ALSA: compress: allow pause and resume during draining
by Gyeongtaek Lee 27 Oct '20

27 Oct '20

On 9/29/20 04:13 PM, Takashi Iwai wrote: >On Tue, 29 Sep 2020 03:51:35 +0200, >Gyeongtaek Lee wrote: >> >> On 9/28/20 11:35 PM, Pierre-Louis Bossart wrote: >> >On 9/28/20 6:13 AM, Jaroslav Kysela wrote: >> >> Dne 28. 09. 20 v 12:50 Gyeongtaek Lee napsal(a): >> >>> With a stream with low bitrate, user can't pause or resume the stream >> >>> near the end of the stream because current ALSA doesn't allow it. >> >>> If the stream has very low bitrate enough to store whole stream into >> >>> the buffer, user can't do anything except stop the stream and then >> >>> restart it from the first. >> >>> If pause and resume is allowed during draining, user experience can be >> >>> enhanced. >> >> >> >> It seems that we need a new state to handle the pause + drain condition for >> >> this case. >> >> >> >> With this proposed change, the pause state in drain is invisible. >> > >> >Indeed it's be much nicer to have a new state, e..g >> >SNDRV_PCM_STATE_DRAINING_PAUSED. >> Ok. I will add the new state. >> > >> >One concern is that states are defined in uapi/sound/asoc.h, so wouldn't >> >this have impacts on userspace as well? We'd change the value of >> >SNDRV_PCM_STATE_LAST. >> > >> I also agree that adding new state and increase LAST value in the header of uapi >> could be dangerous. So, I added it to comress_offload.h for now. >> It could be merged into snd_pcm_state_t in someday with big changes. >> Could you review the fixed patch below? > >Hrm, this resulted in rather more complex changes than the original >patch. It shows that introducing yet another state is no good idea >for this particular case. > >Since the possible application's behavior after this pause is as same >as the normal pause (i.e. either resuming pause or dropping), I find >it OK to take the original approach. Thank you for the review. I think that I should resend the original patch. Am I right? > > >thanks, > >Takashi > >> With a stream with low bitrate, user can't pause or resume the stream >> near the end of the stream because current ALSA doesn't allow it. >> If the stream has very low bitrate enough to store whole stream into >> the buffer, user can't do anything except stop the stream and then >> restart it from first. >> If pause, resume are allowed during draining, user experience can be >> enhanced. >> >> New state for pause during draining is defined in compress_offload.h for >> now. If PCM_STATEs in uapi/sound/asound.h is changed, pcm libraries and >> userspace application will be affected. >> >> Signed-off-by: Gyeongtaek Lee <gt82.lee(a)samsung.com> >> Cc: stable(a)vger.kernel.org >> --- >> include/uapi/sound/compress_offload.h | 3 ++ >> sound/core/compress_offload.c | 47 ++++++++++++++++++++++----- >> 2 files changed, 41 insertions(+), 9 deletions(-) >> >> diff --git a/include/uapi/sound/compress_offload.h b/include/uapi/sound/compress_offload.h >> index 7184265c0b0d..f30b9851d1d7 100644 >> --- a/include/uapi/sound/compress_offload.h >> +++ b/include/uapi/sound/compress_offload.h >> @@ -189,4 +189,7 @@ struct snd_compr_metadata { >> #define SND_COMPR_TRIGGER_DRAIN 7 /*FIXME move this to pcm.h */ >> #define SND_COMPR_TRIGGER_NEXT_TRACK 8 >> #define SND_COMPR_TRIGGER_PARTIAL_DRAIN 9 >> + >> +/* FIXME move this to asound.h */ >> +#define SNDRV_PCM_STATE_DRAINING_PAUSED (SNDRV_PCM_STATE_LAST + 1) >> #endif >> diff --git a/sound/core/compress_offload.c b/sound/core/compress_offload.c >> index 0e53f6f31916..58fbe0d99431 100644 >> --- a/sound/core/compress_offload.c >> +++ b/sound/core/compress_offload.c >> @@ -151,6 +151,7 @@ static int snd_compr_free(struct inode *inode, struct file *f) >> case SNDRV_PCM_STATE_RUNNING: >> case SNDRV_PCM_STATE_DRAINING: >> case SNDRV_PCM_STATE_PAUSED: >> + case SNDRV_PCM_STATE_DRAINING_PAUSED: >> data->stream.ops->trigger(&data->stream, SNDRV_PCM_TRIGGER_STOP); >> break; >> default: >> @@ -431,6 +432,7 @@ static __poll_t snd_compr_poll(struct file *f, poll_table *wait) >> case SNDRV_PCM_STATE_RUNNING: >> case SNDRV_PCM_STATE_PREPARED: >> case SNDRV_PCM_STATE_PAUSED: >> + case SNDRV_PCM_STATE_DRAINING_PAUSED: >> if (avail >= stream->runtime->fragment_size) >> retval = snd_compr_get_poll(stream); >> break; >> @@ -708,11 +710,23 @@ static int snd_compr_pause(struct snd_compr_stream *stream) >> { >> int retval; >> >> - if (stream->runtime->state != SNDRV_PCM_STATE_RUNNING) >> + switch (stream->runtime->state) { >> + case SNDRV_PCM_STATE_RUNNING: >> + retval = stream->ops->trigger(stream, >> + SNDRV_PCM_TRIGGER_PAUSE_PUSH); >> + if (!retval) >> + stream->runtime->state = SNDRV_PCM_STATE_PAUSED; >> + break; >> + case SNDRV_PCM_STATE_DRAINING: >> + retval = stream->ops->trigger(stream, >> + SNDRV_PCM_TRIGGER_PAUSE_PUSH); >> + if (!retval) >> + stream->runtime->state = >> + SNDRV_PCM_STATE_DRAINING_PAUSED; >> + break; >> + default: >> return -EPERM; >> - retval = stream->ops->trigger(stream, SNDRV_PCM_TRIGGER_PAUSE_PUSH); >> - if (!retval) >> - stream->runtime->state = SNDRV_PCM_STATE_PAUSED; >> + } >> return retval; >> } >> >> @@ -720,11 +734,22 @@ static int snd_compr_resume(struct snd_compr_stream *stream) >> { >> int retval; >> >> - if (stream->runtime->state != SNDRV_PCM_STATE_PAUSED) >> + switch (stream->runtime->state) { >> + case SNDRV_PCM_STATE_PAUSED: >> + retval = stream->ops->trigger(stream, >> + SNDRV_PCM_TRIGGER_PAUSE_RELEASE); >> + if (!retval) >> + stream->runtime->state = SNDRV_PCM_STATE_RUNNING; >> + break; >> + case SNDRV_PCM_STATE_DRAINING_PAUSED: >> + retval = stream->ops->trigger(stream, >> + SNDRV_PCM_TRIGGER_PAUSE_RELEASE); >> + if (!retval) >> + stream->runtime->state = SNDRV_PCM_STATE_DRAINING; >> + break; >> + default: >> return -EPERM; >> - retval = stream->ops->trigger(stream, SNDRV_PCM_TRIGGER_PAUSE_RELEASE); >> - if (!retval) >> - stream->runtime->state = SNDRV_PCM_STATE_RUNNING; >> + } >> return retval; >> } >> >> @@ -835,7 +860,9 @@ static int snd_compress_wait_for_drain(struct snd_compr_stream *stream) >> */ >> >> ret = wait_event_interruptible(stream->runtime->sleep, >> - (stream->runtime->state != SNDRV_PCM_STATE_DRAINING)); >> + (stream->runtime->state != SNDRV_PCM_STATE_DRAINING) && >> + (stream->runtime->state != >> + SNDRV_PCM_STATE_DRAINING_PAUSED)); >> if (ret == -ERESTARTSYS) >> pr_debug("wait aborted by a signal\n"); >> else if (ret) >> @@ -857,6 +884,7 @@ static int snd_compr_drain(struct snd_compr_stream *stream) >> case SNDRV_PCM_STATE_SETUP: >> case SNDRV_PCM_STATE_PREPARED: >> case SNDRV_PCM_STATE_PAUSED: >> + case SNDRV_PCM_STATE_DRAINING_PAUSED: >> return -EPERM; >> case SNDRV_PCM_STATE_XRUN: >> return -EPIPE; >> @@ -909,6 +937,7 @@ static int snd_compr_partial_drain(struct snd_compr_stream *stream) >> case SNDRV_PCM_STATE_SETUP: >> case SNDRV_PCM_STATE_PREPARED: >> case SNDRV_PCM_STATE_PAUSED: >> + case SNDRV_PCM_STATE_DRAINING_PAUSED: >> return -EPERM; >> case SNDRV_PCM_STATE_XRUN: >> return -EPIPE; >> >> base-commit: a1b8638ba1320e6684aa98233c15255eb803fac7 >> -- >> 2.21.0 >> >> >

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[PATCH AUTOSEL 5.8 035/132] ASoC: SOF: fix a runtime pm issue in SOF when HDMI codec doesn't work
by Sasha Levin 27 Oct '20

27 Oct '20

From: Rander Wang <rander.wang(a)intel.com> [ Upstream commit 6c63c954e1c52f1262f986f36d95f557c6f8fa94 ] When hda_codec_probe() doesn't initialize audio component, we disable the codec and keep going. However,the resources are not released. The child_count of SOF device is increased in snd_hdac_ext_bus_device_init but is not decrease in error case, so SOF can't get suspended. snd_hdac_ext_bus_device_exit will be invoked in HDA framework if it gets a error. Now copy this behavior to release resources and decrease SOF device child_count to release SOF device. Signed-off-by: Rander Wang <rander.wang(a)intel.com> Reviewed-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> Reviewed-by: Bard Liao <yung-chuan.liao(a)linux.intel.com> Reviewed-by: Guennadi Liakhovetski <guennadi.liakhovetski(a)linux.intel.com> Signed-off-by: Ranjani Sridharan <ranjani.sridharan(a)linux.intel.com> Link: https://lore.kernel.org/r/20200825235040.1586478-3-ranjani.sridharan@linux.… Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/sof/intel/hda-codec.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/soc/sof/intel/hda-codec.c b/sound/soc/sof/intel/hda-codec.c index 2c5c451fa19d7..c475955c6eeba 100644 --- a/sound/soc/sof/intel/hda-codec.c +++ b/sound/soc/sof/intel/hda-codec.c @@ -151,7 +151,7 @@ static int hda_codec_probe(struct snd_sof_dev *sdev, int address, if (!hdev->bus->audio_component) { dev_dbg(sdev->dev, "iDisp hw present but no driver\n"); - return -ENOENT; + goto error; } hda_priv->need_display_power = true; } @@ -174,7 +174,7 @@ static int hda_codec_probe(struct snd_sof_dev *sdev, int address, * other return codes without modification */ if (ret == 0) - ret = -ENOENT; + goto error; } return ret; -- 2.25.1

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[PATCH AUTOSEL 5.9 043/147] ASoC: AMD: Clean kernel log from deferred probe error messages
by Sasha Levin 27 Oct '20

27 Oct '20

From: Akshu Agrawal <akshu.agrawal(a)amd.com> [ Upstream commit f7660445c8e7fda91e8b944128554249d886b1d4 ] While the driver waits for DAIs to be probed and retries probing, have the error messages at debug level instead of error. Signed-off-by: Akshu Agrawal <akshu.agrawal(a)amd.com> Link: https://lore.kernel.org/r/20200826185454.5545-1-akshu.agrawal@amd.com Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/amd/acp3x-rt5682-max9836.c | 11 ++++++++--- 1 file changed, 8 insertions(+), 3 deletions(-) diff --git a/sound/soc/amd/acp3x-rt5682-max9836.c b/sound/soc/amd/acp3x-rt5682-max9836.c index 406526e79af34..1a4e8ca0f99c2 100644 --- a/sound/soc/amd/acp3x-rt5682-max9836.c +++ b/sound/soc/amd/acp3x-rt5682-max9836.c @@ -472,12 +472,17 @@ static int acp3x_probe(struct platform_device *pdev) ret = devm_snd_soc_register_card(&pdev->dev, card); if (ret) { - dev_err(&pdev->dev, + if (ret != -EPROBE_DEFER) + dev_err(&pdev->dev, "devm_snd_soc_register_card(%s) failed: %d\n", card->name, ret); - return ret; + else + dev_dbg(&pdev->dev, + "devm_snd_soc_register_card(%s) probe deferred: %d\n", + card->name, ret); } - return 0; + + return ret; } static const struct acpi_device_id acp3x_audio_acpi_match[] = { -- 2.25.1

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[PATCH] ASoC: amd: support other audio modes for raven
by Vijendar Mukunda 27 Oct '20

27 Oct '20

ACP supports different audio configurations other than I2S. This patch will fix acp driver probe failure for other audio configurations. Tested-by: Prike Liang <Prike.Liang(a)amd.com> Reviewed-by: Prike Liang <Prike.Liang(a)amd.com> Acked-by: Alex Deucher <alexander.deucher(a)amd.com> Signed-off-by: Vijendar Mukunda <Vijendar.Mukunda(a)amd.com> --- sound/soc/amd/raven/pci-acp3x.c | 4 +--- 1 file changed, 1 insertion(+), 3 deletions(-) diff --git a/sound/soc/amd/raven/pci-acp3x.c b/sound/soc/amd/raven/pci-acp3x.c index 31b797c..cd5b097 100644 --- a/sound/soc/amd/raven/pci-acp3x.c +++ b/sound/soc/amd/raven/pci-acp3x.c @@ -231,9 +231,7 @@ static int snd_acp3x_probe(struct pci_dev *pci, } break; default: - dev_err(&pci->dev, "Invalid ACP audio mode : %d\n", val); - ret = -ENODEV; - goto disable_msi; + dev_info(&pci->dev, "ACP audio mode : %d\n", val); } pm_runtime_set_autosuspend_delay(&pci->dev, 2000); pm_runtime_use_autosuspend(&pci->dev); -- 2.7.4

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[PATCH] ASoC: jz4740-i2s: Remove manual DMA peripheral ID assignment
by Lars-Peter Clausen 27 Oct '20

27 Oct '20

All platforms that use the jz4740-i2s driver have been switched to devicetree for a while now and the assignment of the DMA peripheral ID is done in the devicetree. It is no longer necessary to manually assign the peripheral ID in the driver, so remove that. The DMA driver does not even look at the value assigned in the driver anymore and always uses the value provided by the devicetree. Signed-off-by: Lars-Peter Clausen <lars(a)metafoo.de> --- sound/soc/jz4740/jz4740-i2s.c | 5 ----- 1 file changed, 5 deletions(-) diff --git a/sound/soc/jz4740/jz4740-i2s.c b/sound/soc/jz4740/jz4740-i2s.c index c7bd20104b20..e8bc7ca5ee5e 100644 --- a/sound/soc/jz4740/jz4740-i2s.c +++ b/sound/soc/jz4740/jz4740-i2s.c @@ -26,9 +26,6 @@ #include "jz4740-i2s.h" -#define JZ4740_DMA_TYPE_AIC_TRANSMIT 24 -#define JZ4740_DMA_TYPE_AIC_RECEIVE 25 - #define JZ_REG_AIC_CONF 0x00 #define JZ_REG_AIC_CTRL 0x04 #define JZ_REG_AIC_I2S_FMT 0x10 @@ -377,13 +374,11 @@ static void jz4740_i2c_init_pcm_config(struct jz4740_i2s *i2s) /* Playback */ dma_data = &i2s->playback_dma_data; dma_data->maxburst = 16; - dma_data->slave_id = JZ4740_DMA_TYPE_AIC_TRANSMIT; dma_data->addr = i2s->phys_base + JZ_REG_AIC_FIFO; /* Capture */ dma_data = &i2s->capture_dma_data; dma_data->maxburst = 16; - dma_data->slave_id = JZ4740_DMA_TYPE_AIC_RECEIVE; dma_data->addr = i2s->phys_base + JZ_REG_AIC_FIFO; } -- 2.20.1

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[PATCH] ASoC: wm5102: Use get_unaligned_be16() for dac_comp_coeff
by Richard Fitzgerald 27 Oct '20

27 Oct '20

Replace the two-step copy-and-convert in wm5102_out_comp_coeff_put() with get_unaligned_be16(). Apart from looking nicer, it avoids this sparse warning: wm5102.c:687:35: sparse: sparse: cast to restricted __be16 Signed-off-by: Richard Fitzgerald <rf(a)opensource.cirrus.com> --- sound/soc/codecs/wm5102.c | 4 +--- 1 file changed, 1 insertion(+), 3 deletions(-) diff --git a/sound/soc/codecs/wm5102.c b/sound/soc/codecs/wm5102.c index 2ed3fa67027d..70d353b63fe0 100644 --- a/sound/soc/codecs/wm5102.c +++ b/sound/soc/codecs/wm5102.c @@ -682,9 +682,7 @@ static int wm5102_out_comp_coeff_put(struct snd_kcontrol *kcontrol, struct arizona *arizona = dev_get_drvdata(component->dev->parent); mutex_lock(&arizona->dac_comp_lock); - memcpy(&arizona->dac_comp_coeff, ucontrol->value.bytes.data, - sizeof(arizona->dac_comp_coeff)); - arizona->dac_comp_coeff = be16_to_cpu(arizona->dac_comp_coeff); + arizona->dac_comp_coeff = get_unaligned_be16(ucontrol->value.bytes.data); mutex_unlock(&arizona->dac_comp_lock); return 0; base-commit: 2ebefde9de62a0828936e47b62975501993b5ee9 -- 2.20.1

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[PATCH 1/2] ASoC: dt-bindings: fsl_spdif: Add new compatible string for i.MX8QM
by Shengjiu Wang 27 Oct '20

27 Oct '20

Add new compatible string "fsl,imx8qm-spdif" for supporting spdif module on i.MX8QM. Signed-off-by: Shengjiu Wang <shengjiu.wang(a)nxp.com> --- Documentation/devicetree/bindings/sound/fsl,spdif.yaml | 1 + 1 file changed, 1 insertion(+) diff --git a/Documentation/devicetree/bindings/sound/fsl,spdif.yaml b/Documentation/devicetree/bindings/sound/fsl,spdif.yaml index 2ac671f5cb9b..50449b6d1048 100644 --- a/Documentation/devicetree/bindings/sound/fsl,spdif.yaml +++ b/Documentation/devicetree/bindings/sound/fsl,spdif.yaml @@ -20,6 +20,7 @@ properties: - fsl,imx35-spdif - fsl,vf610-spdif - fsl,imx6sx-spdif + - fsl,imx8qm-spdif reg: maxItems: 1 -- 2.27.0

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[PATCH 0/6] ASoC: merge soc_pcm_hw_param() rollback and soc_pcm_hw_free()
by Kuninori Morimoto 27 Oct '20

27 Oct '20

Hi Mark soc_pcm_hw_params() does rollback when failed (A), but, it is almost same as soc_pcm_hw_free(). static int soc_pcm_hw_params(xxx) { ... if (ret < 0) goto xxx_err; ... return ret; ^ component_err: | ... | interface_err: (A) ... | codec_err: | ... v return ret; } This kind of duplicated code can be a hotbed of bugs, thus, this patch-set share soc_pcm_hw_free() and rollback. Kuninori Morimoto (6): ASoC: soc.h: remove for_each_rtd_dais_rollback() ASoC: soc-pcm: move soc_pcm_hw_free() next to soc_pcm_hw_params() ASoC: soc-link: add mark for snd_soc_link_hw_params/free() ASoC: soc-component: add mark for snd_soc_pcm_component_hw_params/free() ASoC: soc-dai: add mark for snd_soc_dai_hw_params/free() ASoC: soc-pcm: add soc_pcm_hw_clean() and call it from soc_pcm_hw_params/free() include/sound/soc-component.h | 6 +- include/sound/soc-dai.h | 4 +- include/sound/soc-link.h | 3 +- include/sound/soc.h | 7 +- sound/soc/soc-component.c | 19 ++--- sound/soc/soc-dai.c | 13 +++- sound/soc/soc-dapm.c | 4 +- sound/soc/soc-link.c | 12 +++- sound/soc/soc-pcm.c | 131 ++++++++++++++-------------------- 9 files changed, 97 insertions(+), 102 deletions(-) -- 2.25.1

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[PATCH v2 00/17] ASoC: sun8i-codec: support for AIF2 and AIF3
by Samuel Holland 27 Oct '20

27 Oct '20

This series adds support the other two AIFs present in the sun8i codec, which can be used for codec2codec DAI links. This series first fills out the DAI driver, removing assumptions that were made for AIF1 (16 bits, 2 channels, certain clock inversions). Some new logic is required to handle 3 DAIs and the ADC/DAC sharing the same clock. Finally, it adds the new DAIs, and hooks them up with DAPM widgets and routes per the hardware topology. To minimize the number of patches in this series, related device tree patches (increasing #sound-dai-cells, adding new DAI links) will be sent separately. Changes from v1: - Patches 1-8 from v1 (DAPM changes) were merged - Prefixed AIF constants with "SUN8I_CODEC_" [1, 7, 10, 16, 17] - Renamed variables in sun8i_codec_set_fmt for clarity [3] - Update sysclk->sysclk_rate if later calls to hw_params change the sample rate (thanks Chen-Yu for reminding me of this) [11] - Select COMMON_CLK for clk_set_rate_exclusive [12] - Add comments and hopefully clarify the clock protection logic [12] - Make the error message more concise and put it on one line [12] - Drop the "reg" variable holding SUN8I_AIF_CLK_CTRL(dai->id) [15] - Rename "div_reg" to "clk_reg" and adjust comments for clarity [17] - Improve the AIF2/AIF3 rate mismatch error message [17] - Minor grammar/wording fixes in commit messages [2, 4, 7, 8, 16, 17] - Added Maxime's Acked-by: [1-2, 4-9, 11, 13-14, 16] Samuel Holland (17): ASoC: sun8i-codec: Prepare to extend the DAI driver ASoC: sun8i-codec: Program DAI format before clock inversion ASoC: sun8i-codec: Enable all supported clock inversions ASoC: sun8i-codec: Use the provided word size ASoC: sun8i-codec: Round up the LRCK divisor ASoC: sun8i-codec: Correct the BCLK divisor calculation ASoC: sun8i-codec: Support the TDM slot binding ASoC: sun8i-codec: Enforce symmetric DAI parameters ASoC: sun8i-codec: Enable all supported sample rates ASoC: sun8i-codec: Automatically set the system sample rate ASoC: sun8i-codec: Constrain to compatible sample rates ASoC: sun8i-codec: Protect the clock rate while streams are open ASoC: sun8i-codec: Require an exact BCLK divisor match ASoC: sun8i-codec: Enable all supported PCM formats ASoC: sun8i-codec: Generalize AIF clock control ASoC: sun8i-codec: Add the AIF2 DAI, widgets, and routes ASoC: sun8i-codec: Add the AIF3 DAI, widgets, and routes sound/soc/sunxi/Kconfig | 1 + sound/soc/sunxi/sun8i-codec.c | 834 ++++++++++++++++++++++++++++------ 2 files changed, 699 insertions(+), 136 deletions(-) -- 2.26.2

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