Merge series from Bard Liao <yung-chuan.liao@linux.intel.com>:
As discussed in [1], ASoC core supports multi codec DAIs
on a DAI link. However it does not do so for CPU DAIs.
So, add support for multi CPU DAIs on a DAI Link by adding
multi CPU DAI in Card instantiation, suspend and resume
functions, PCM ops, stream handling functions and DAPM.
[1]: https://www.spinics.net/lists/alsa-devel/msg71369.html
changes in v5:
- rebase to latest kernel base
Bard Liao (2):
ASoC: Return error if the function does not support multi-cpu
ASoC: pcm: check if cpu-dai supports a given stream
Shreyas NC (4):
ASoC: Add initial support for multiple CPU DAIs
ASoC: Add multiple CPU DAI support for PCM ops
ASoC: Add dapm_add_valid_dai_widget helper
ASoC: Add multiple CPU DAI support in DAPM
include/sound/soc.h | 15 +
sound/soc/soc-compress.c | 5 +-
sound/soc/soc-core.c | 168 +++++-----
sound/soc/soc-dapm.c | 133 ++++----
sound/soc/soc-generic-dmaengine-pcm.c | 18 +
sound/soc/soc-pcm.c | 463 ++++++++++++++++++--------
6 files changed, 531 insertions(+), 271 deletions(-)
--
2.17.1
Merge series from Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>:
The first two patches prepare the support of multi-cpu dais for
synchronized playback and capture. We remove an unused set of
prototypes and add a get_sdw_stream() callback prototype currently
missing (the implementation will come later as part of the
synchronized playback)
The last exposes macros used internally, so that they can be reused to
extract information from the _ADR 64-bit values in SOF platform
drivers and related machine drivers.
I think it's simpler if all these simple patches are merged through
the SoundWire tree. With the additional changes to remove the platform
drivers and the merge of interrupt handling, that will result in a
single immutable tag provided to Mark Brown.
Pierre-Louis Bossart (3):
soundwire: cadence: remove useless prototypes
ASoC: soc-dai: add get_sdw_stream() callback
soundwire: add helper macros for devID fields
drivers/soundwire/bus.c | 21 +++++----------------
drivers/soundwire/cadence_master.h | 8 --------
include/linux/soundwire/sdw.h | 23 +++++++++++++++++++++++
include/sound/soc-dai.h | 21 +++++++++++++++++++++
4 files changed, 49 insertions(+), 24 deletions(-)
--
2.20.1
ASoC core supports multiple codec DAIs but supports only a CPU DAI.
To support multiple cpu DAIs, add cpu_dai and num_cpu_dai in
snd_soc_dai_link and snd_soc_pcm_runtime structures similar to
support for codec_dai. This is intended as a preparatory patch to
eventually support the unification of the Codec and CPU DAI.
Inline with multiple codec DAI approach, add support to allocate,
init, bind and probe multiple cpu_dai on init if driver specifies
that. Also add support to loop over multiple cpu_dai during
suspend and resume.
This is intended as a preparatory patch to eventually unify the CPU
and Codec DAI into DAI components.
Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Signed-off-by: Bard Liao <yung-chuan.liao@linux.intel.com>
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Signed-off-by: Shreyas NC <shreyas.nc@intel.com>
Link: https://lore.kernel.org/r/20200225133917.21314-2-yung-chuan.liao@linux.intel.com
Signed-off-by: Mark Brown <broonie@kernel.org>
ASoC component open/close and snd_soc_component_module_get/put are called
independently for each component-substream pair, so the logic added in
commit dd03907bf1 ("ASoC: soc-pcm: call snd_soc_component_open/close()
once") was not sufficient and led to PCM playback and module unload errors.
Implement handling of failures directly in soc_pcm_components_open(),
so that any successfully opened components are closed upon error with
other components. This allows to clean up error handling in
soc_pcm_open() without adding more state tracking.
Fixes: dd03907bf1 ("ASoC: soc-pcm: call snd_soc_component_open/close() once")
Signed-off-by: Kai Vehmanen <kai.vehmanen@linux.intel.com>
Tested-by: Dmitry Osipenko <digetx@gmail.com>
Link: https://lore.kernel.org/r/20200220094955.16968-1-kai.vehmanen@linux.intel.com
Signed-off-by: Mark Brown <broonie@kernel.org>
dpcm_path_put() (A) is calling kfree(*list).
The freed list is created by dapm_widget_list_create() (B) which is called
from snd_soc_dapm_dai_get_connected_widgets() (C) which is called from
dpcm_path_get() (D).
(B) dapm_widget_list_create(**list, ...)
{
...
=> *list = kzalloc();
...
}
(C) snd_soc_dapm_dai_get_connected_widgets(..., **list, ...)
{
...
dapm_widget_list_create(list, ...);
...
}
(D) dpcm_path_get(..., **list)
{
...
snd_soc_dapm_dai_get_connected_widgets(..., list, ...);
...
}
(A) dpcm_path_put(**list)
{
=> kfree(*list);
}
This kind of unbalance code is very difficult to read/understand.
To avoid this issue, this patch adds each missing paired function
dapm_widget_list_free() for dapm_widget_list_create() (B), and
snd_soc_dapm_dai_free_widgets() for snd_soc_dapm_dai_get_connected_widgets() (C).
This patch uses these, and moves dpcm_path_put() next to dpcm_path_get().
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Link: https://lore.kernel.org/r/87a75fjc9q.wl-kuninori.morimoto.gx@renesas.com
Signed-off-by: Mark Brown <broonie@kernel.org>
DAI driver has playback/capture stream.
OTOH, we have SNDRV_PCM_STREAM_PLAYBACK/CAPTURE.
Because of this kind of implementation,
ALSA SoC needs to have many verbose code.
To solve this issue, this patch adds snd_soc_dai_get_pcm_stream() macro
to get playback/capture stream pointer from stream.
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Link: https://lore.kernel.org/r/87ftf7jcab.wl-kuninori.morimoto.gx@renesas.com
Signed-off-by: Mark Brown <broonie@kernel.org>
commit b0edff4236 ("ASoC: soc-pcm/soc-compress: use
snd_soc_dapm_stream_stop() for SND_SOC_DAPM_STREAM_STOP")
uses snd_soc_dapm_stream_stop() for soc_compr_free_fe()
and dpcm_fe_dai_shutdown() because it didn't care about pmdown_time.
But, it didn't need to care.
This patch rollback to original code.
Some system will wait unneeded timed-out without this patch.
Special Thanks for reporting to Chris Gorman.
...
intel_sst_acpi 808622A8:00: Wait timed-out condition:0x0, msg_id:0x1 fw_state 0x3
intel_sst_acpi 808622A8:00: fw returned err -16
sst-mfld-platform sst-mfld-platform: ASoC: PRE_PMD: pcm0_in event failed: -16
...
Fixes: commit b0edff4236 ("ASoC: soc-pcm/soc-compress: use snd_soc_dapm_stream_stop() for SND_SOC_DAPM_STREAM_STOP")
Reported-by: Chris Gorman <chrisjohgorman@gmail.com>
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Link: https://lore.kernel.org/r/87lfowspeb.wl-kuninori.morimoto.gx@renesas.com
Signed-off-by: Mark Brown <broonie@kernel.org>
On i386 randconfig:
sound/soc/codecs/wm9705.o: In function `wm9705_soc_resume':
wm9705.c:(.text+0x128): undefined reference to `snd_ac97_reset'
sound/soc/codecs/wm9712.o: In function `wm9712_soc_resume':
wm9712.c:(.text+0x2d1): undefined reference to `snd_ac97_reset'
sound/soc/codecs/wm9713.o: In function `wm9713_soc_resume':
wm9713.c:(.text+0x820): undefined reference to `snd_ac97_reset'
Fix this by adding the missing dependencies on SND_SOC_AC97_BUS.
Reported-by: Randy Dunlap <rdunlap@infradead.org>
Signed-off-by: Geert Uytterhoeven <geert@linux-m68k.org>
Link: https://lore.kernel.org/r/20200224112537.14483-1-geert@linux-m68k.org
Signed-off-by: Mark Brown <broonie@kernel.org>
The TLV320ADCx140 parts support Dynamic Range Enhancer (DRE) as defined
in Section 8.3.2 of the data sheets.
The DRE achieves a complete-channel dynamic range as high as 120 dB.
At a system level, the DRE scheme enables far-field, high-fidelity recording
of audio signals in very quiet environments and low-distortion recording in
loud environments.
There are 2 enables for DRE. The first is a global setting that enables
the DRE engine in the device and the other enable is per channel. If
the DRE is enabled globally then either DRE or AGC can be used per each
configured channel. If global DRE is disabled then even setting the DRE
enable bit in the channel config register will have no effect.
Signed-off-by: Dan Murphy <dmurphy@ti.com>
Link: https://lore.kernel.org/r/20200221181358.22526-1-dmurphy@ti.com
Signed-off-by: Mark Brown <broonie@kernel.org>
Add support for digital volume control. There is no dedicated register
for volume control but instead there are 4. The values of the registers
are determined with exponential floating point math.
So a table was created with register values for 2dB step increments
from -110dB to 0dB.
Signed-off-by: Dan Murphy <dmurphy@ti.com>
Link: https://lore.kernel.org/r/20200221124151.8774-1-dmurphy@ti.com
Signed-off-by: Mark Brown <broonie@kernel.org>
The AIU audio controller on the Meson8 and Meson8b SoC families is
compatible with the one found in the later GXBB family. Add compatible
strings for these two older SoC families so the driver can be loaded for
them.
Instead of using the I2S divider from the AIU_CLK_CTRL_MORE register we
need to use the I2S divider from the AIU_CLK_CTRL register. This older
register is less flexible because it only supports four divider settings
(1, 2, 4, 8) compared to the AIU_CLK_CTRL_MORE register (which supports
dividers in the range 0..64).
Signed-off-by: Martin Blumenstingl <martin.blumenstingl@googlemail.com>
Reviewed-by: Jerome Brunet <jbrunet@baylibre.com>
Link: https://lore.kernel.org/r/20200220205711.77953-4-martin.blumenstingl@googlemail.com
Signed-off-by: Mark Brown <broonie@kernel.org>
Introduce a struct aiu_platform_data to make the driver aware of
platform specific information. Convert the existing check for the
internal stereo audio codec (only available on GXL) to this new struct.
Support for the 32-bit SoCs will need this as well because the
AIU_CLK_CTRL_MORE register doesn't have the I2S divider bits (and we
need to use the I2S divider from AIU_CLK_CTRL instead).
Signed-off-by: Martin Blumenstingl <martin.blumenstingl@googlemail.com>
Reviewed-by: Jerome Brunet <jbrunet@baylibre.com>
Link: https://lore.kernel.org/r/20200220205711.77953-3-martin.blumenstingl@googlemail.com
Signed-off-by: Mark Brown <broonie@kernel.org>
Add the tlv320adcx140 codec driver family.
The TLV320ADCx140 is a Burr-Brown™ highperformance, audio analog-to-digital
converter (ADC) that supports simultaneous sampling of up to four analog
channels or eight digital channels for the pulse density modulation (PDM)
microphone input. The device supports line and microphone inputs, and
allows for both single-ended and differential input configurations.
Signed-off-by: Dan Murphy <dmurphy@ti.com>
Link: https://lore.kernel.org/r/20200220210759.31466-3-dmurphy@ti.com
Signed-off-by: Mark Brown <broonie@kernel.org>
Currently it is possible to get the following bad config:
WARNING: unmet direct dependencies detected for SND_SOC_WM5110
Depends on [n]: SOUND [=y] && !UML && SND [=y] && SND_SOC [=y] && MFD_WM5110 [=n]
commit ea00d95200 ("ASoC: Use imply for SND_SOC_ALL_CODECS")
commit d8dd3f92a6 ("ASoC: Fix SND_SOC_ALL_CODECS imply misc fallout")
After these two patches the machine drivers still selects the
SND_SOC_WM5110 symbol which doesn't take account of the dependency
added on the MFD_WM5110 symbol, fix this by also adding a dependency on
MFD_WM5110 itself.
Reported-by: Randy Dunlap <rdunlap@infradead.org>
Signed-off-by: Charles Keepax <ckeepax@opensource.cirrus.com>
Link: https://lore.kernel.org/r/20200220125654.7064-1-ckeepax@opensource.cirrus.com
Signed-off-by: Mark Brown <broonie@kernel.org>
