commit fa31a2c787aeaf61d02b2a57bd9765ca5e67d949
Author: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Date: Fri Jan 15 13:56:30 2021 +0900
ASoC: soc-dai.h: remove symmetric_rates/samplebits
All drivers are using new name.
Let's remove old one.
commit f14654ddf2e982537ab476d392a87fcbf90374c3
Author: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Date: Fri Jan 15 13:52:54 2021 +0900
ASoC: sync parameter naming : rate / sample_bits
snd_pcm_runtime / snd_soc_dai / snd_soc_dai_driver / snd_soc_dai_link
have related parameter which is similar but not same naming.
The idea is the same as the previous attempt: calls ac101_trigger() just
before set_clock(1).
https://github.com/respeaker/seeed-voicecard/issues/290
6mics / linear 4 mics: fails to record against v5.10 kernel
https://github.com/raspberrypi/linux/issues/4279
[regression] alsa system call blocks on record between 5.4.83 and 5.5.19
In v5.5,
commit 4378f1fbe924054a09ff0d4e39e1a581b9245252
Author: Peter Ujfalusi <peter.ujfalusi@ti.com>
Date: Fri Sep 27 10:16:46 2019 +0300
ASoC: soc-pcm: Use different sequence for start/stop trigger
On stream stop currently we stop the DMA first followed by the CPU DAI.
This can cause underflow (playback) or overflow (capture) on the DAI side
as the DMA is no longer feeding data while the DAI is still active.
It can be observed easily if the DAI side does not have FIFO (or it is
disabled) to survive the time while the DMA is stopped, but still can
happen on relatively slow CPUs when relatively high sampling rate is used:
the FIFO is drained between the time the DMA is stopped and the DAI is
stopped.
It can only fixed by using different sequence within trigger for 'stop' and
'start':
case SNDRV_PCM_TRIGGER_START:
case SNDRV_PCM_TRIGGER_RESUME:
case SNDRV_PCM_TRIGGER_PAUSE_RELEASE:
Trigger order: dai_link, DMA, CPU DAI then the codec
case SNDRV_PCM_TRIGGER_STOP:
case SNDRV_PCM_TRIGGER_SUSPEND:
case SNDRV_PCM_TRIGGER_PAUSE_PUSH:
Trigger order: codec, CPU DAI, DMA then dai_link
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@ti.com>
Reviewed-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Link: https://lore.kernel.org/r/20190927071646.22319-1-peter.ujfalusi@ti.com
Signed-off-by: Mark Brown <broonie@kernel.org>
Need to revisit, and do in a different way.
https://github.com/respeaker/seeed-voicecard/issues/290
6mics / linear 4 mics: fails to record against v5.10 kernel
https://github.com/raspberrypi/linux/issues/4279
[regression] alsa system call blocks on record between 5.4.83 and 5.5.19
In v5.5,
commit 4378f1fbe924054a09ff0d4e39e1a581b9245252
Author: Peter Ujfalusi <peter.ujfalusi@ti.com>
Date: Fri Sep 27 10:16:46 2019 +0300
ASoC: soc-pcm: Use different sequence for start/stop trigger
On stream stop currently we stop the DMA first followed by the CPU DAI.
This can cause underflow (playback) or overflow (capture) on the DAI side
as the DMA is no longer feeding data while the DAI is still active.
It can be observed easily if the DAI side does not have FIFO (or it is
disabled) to survive the time while the DMA is stopped, but still can
happen on relatively slow CPUs when relatively high sampling rate is used:
the FIFO is drained between the time the DMA is stopped and the DAI is
stopped.
It can only fixed by using different sequence within trigger for 'stop' and
'start':
case SNDRV_PCM_TRIGGER_START:
case SNDRV_PCM_TRIGGER_RESUME:
case SNDRV_PCM_TRIGGER_PAUSE_RELEASE:
Trigger order: dai_link, DMA, CPU DAI then the codec
case SNDRV_PCM_TRIGGER_STOP:
case SNDRV_PCM_TRIGGER_SUSPEND:
case SNDRV_PCM_TRIGGER_PAUSE_PUSH:
Trigger order: codec, CPU DAI, DMA then dai_link
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@ti.com>
Reviewed-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Link: https://lore.kernel.org/r/20190927071646.22319-1-peter.ujfalusi@ti.com
Signed-off-by: Mark Brown <broonie@kernel.org>
commit fe7ed4dec2e6289eab81dd18c0d613c0851d85a1
Author: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Date: Mon Jan 21 16:40:59 2019 +0900
ASoC: simple-card: rename to asoc_simple_card_canonicalize_platform()
Current simple-card is using asoc_simple_card_canonicalize_dailink().
Its naming is "dailink", but is for "platform".
We already have asoc_simple_card_canonicalize_cpu() for "cpu",
let's follow same naming rule.
It never return error, so, void function is better idea.
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
commit 95cfc0a0aaf575207152dd7601e782702565a6f1
Author: Samuel Holland <samuel@sholland.org>
Date: Wed Mar 4 23:11:43 2020 -0600
ASoC: simple-card: Add support for codec2codec DAI links
Following the example in cb2cf0de1174 ("ASoC: soc-core: care Codec <->
Codec case by non_legacy_dai_naming"), determine if a DAI link contains
only codec DAIs by examining the non_legacy_dai_naming flag in each
DAI's component.
For now, we assume there is only one or a small set of valid PCM stream
parameters, so num_params == 1 is good enough. We also assume that the
same params are valid for all supported streams. params is set to the
subset of parameters common among all DAIs, and then the existing code
automatically chooses the highest quality of the remaining values when
the link is brought up.
commit 0580dde59438686d60762b6da9229ebec693b94f
Author: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Date: Wed Mar 20 13:54:42 2019 +0900
ASoC: simple-card-utils: add asoc_simple_debug_info()
Current simple-card-utils has dev_dbg(), but people want to
add #define DEBUG at simple-card/audio-graph, not simple-card-utils.
And, people want to get all information.
This patch adds new asoc_simple_debug_info() to indicates information.
commit 39853b1438bf9b07349c8c44b48f6c2eda6f8840
Merge: d6fea46e086b 5b554b0a29ce
Author: Mark Brown <broonie@kernel.org>
Date: Mon Jun 22 15:36:06 2020 +0100
Merge series "ASoC: merge snd_soc_component_read() and snd_soc_component_read32()" from Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>:
$ git diff d6fea46e086b...5b554b0a29ce
commit 5b554b0a29ce9610e3c237c77a1f76db87454b72
Author: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Date: Tue Jun 16 14:21:55 2020 +0900
ASoC: remove snd_soc_component_read32()
No driver is using snd_soc_component_read32() anymore.
This patch removes it.
Reference:
commit e2978c45e5ed3bab7f69477b882ef588185b30cc
Author: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Date: Fri Jul 17 09:21:54 2020 +0900
ASoC: soc-dai: remove .digital_mute
All drivers are now using .mute_stream.
Let's remove .digital_mute.
commit 22e9b54307987787efa0ee534aa9e31982ec1161
Merge: dc9584c5a3b8 a0234d0e6014
Author: Mark Brown <broonie@kernel.org>
Date: Fri Jul 17 14:47:04 2020 +0100
Merge series "ASoC: merge .digital_mute() into .mute_stream()" from Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>:
These are v4 digital_mute() patch which adjusts
to atmel which had conflict on v3.
v3 -> v4
- tidyup for atmel which had conflict
$ git diff dc9584c5a3b8...a0234d0e6014
commit bdd0c277d9846977ec3f175341d4e7475ed26ef7
Merge: d235b2823698 50891431aaad
Author: Mark Brown <broonie@kernel.org>
Date: Thu Jul 16 23:51:51 2020 +0100
Merge series "ASoC: merge .digital_mute() into .mute_stream()" from Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>:
These are v3 patch-set.
ALSA SoC has 2 mute callbacks (= .digital_mute(), .mute_stream()).
But the difference between these 2 are very small.
.digital_mute() is for Playback
.mute_stream() is for Playback/Capture
This patch-set adds new .no_capture_mute flag and emulate
.digital_mute() by .mute_stream().
v2 -> v3
- uses "xxx_mute_stream" for .mute_stream naming
if it was better
- removed verbose Cc email address
$ git diff d235b2823698...50891431aaad
commit 350d993510115e3d9e78f1b3359bff7b68e88418
Author: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Date: Thu Jul 9 10:55:41 2020 +0900
ASoC: soc-dai.c: add .no_capture_mute support
