Now, snd_soc_pcm_runtime supports multi cpu_dai/codec_dai.
It still has cpu_dai/codec_dai for single DAI,
and has cpu_dais/codec_dais for multi DAIs.
dais = [][][][][][][][][][][][][][][][][][]
^cpu_dais ^codec_dais
|--- num_cpus ---|--- num_codecs --|
/* for multi DAIs */
rtd->cpu_dais = &rtd->dais[0];
rtd->codec_dais = &rtd->dais[dai_link->num_cpus];
/* for single DAI */
rtd->cpu_dai = rtd->cpu_dais[0];
rtd->codec_dai = rtd->codec_dais[0];
But, these can be replaced by dais.
This patch adds asoc_rtd_to_cpu() / asoc_rtd_to_codec() macro for it.
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Tested-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Link: https://lore.kernel.org/r/875zevk5va.wl-kuninori.morimoto.gx@renesas.com
Signed-off-by: Mark Brown <broonie@kernel.org>
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
The AC'97 based PXA machines currently don't build reliably as they don't
ensure that an AC'97 bus is built, causing at least eseries_pxa_defconfig
to fail to build. Add selects to fix this.
Reported-by: KernelCI <bot@kernelci.org>
Signed-off-by: Mark Brown <broonie@kernel.org>
Link: https://lore.kernel.org/r/20200326180116.21375-1-broonie@kernel.org
Signed-off-by: Mark Brown <broonie@kernel.org>
regmap needs to be selected by users which for machine drivers that select
AC'97 CODEC drivers means that we need to also select regmap to ensure that
the CODEC driver will build if nothing else enables regmap as is likely for
such systems.
Signed-off-by: Mark Brown <broonie@kernel.org>
Link: https://lore.kernel.org/r/20200326151053.40806-1-broonie@kernel.org
Signed-off-by: Mark Brown <broonie@kernel.org>
If the DAI was configured with a set_tdm_slots() call, use the information.
A platform or machine driver may configure each amplifier to extract
different bitSlots from the frame, or extract the same data and use
processing to generate the relevant output. The latter case is easier
to handle in case of orientation changes.
Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Link: https://lore.kernel.org/r/20200325212905.28145-3-pierre-louis.bossart@linux.intel.com
Signed-off-by: Mark Brown <broonie@kernel.org>
Further align HDA init sequence to the legacy non-DSP HDA driver by
calling snd_hdac_set_codec_wakeup() during the chip init sequence.
Signed-off-by: Kai Vehmanen <kai.vehmanen@linux.intel.com>
Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Reviewed-by: Ranjani Sridharan <ranjani.sridharan@linux.intel.com>
Link: https://lore.kernel.org/r/20200325211233.27394-6-pierre-louis.bossart@linux.intel.com
Signed-off-by: Mark Brown <broonie@kernel.org>
The misc clock gating (MISCBDCGE) is disabled for controller reset and
reenabled once reset is complete.
Fix the case when error happens during reset, and clock gating is
left disabled. The clock gating should be reenabled also in this case.
Signed-off-by: Kai Vehmanen <kai.vehmanen@linux.intel.com>
Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Reviewed-by: Ranjani Sridharan <ranjani.sridharan@linux.intel.com>
Link: https://lore.kernel.org/r/20200325211233.27394-5-pierre-louis.bossart@linux.intel.com
Signed-off-by: Mark Brown <broonie@kernel.org>
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@linux.intel.com>
Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Reviewed-by: Kai Vehmanen <kai.vehmanen@linux.intel.com>
Reviewed-by: Ranjani Sridharan <ranjani.sridharan@linux.intel.com>
Link: https://lore.kernel.org/r/20200325211233.27394-4-pierre-louis.bossart@linux.intel.com
Signed-off-by: Mark Brown <broonie@kernel.org>
Use for_each_pcm_streams() to enumerate streams in sof_dai_load()
instead of doing that manually.
Signed-off-by: Guennadi Liakhovetski <guennadi.liakhovetski@linux.intel.com>
Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Reviewed-by: Kai Vehmanen <kai.vehmanen@linux.intel.com>
Reviewed-by: Ranjani Sridharan <ranjani.sridharan@linux.intel.com>
Link: https://lore.kernel.org/r/20200325211233.27394-3-pierre-louis.bossart@linux.intel.com
Signed-off-by: Mark Brown <broonie@kernel.org>
Improve the DSP power state logs with the state names
instead of values.
Signed-off-by: Ranjani Sridharan <ranjani.sridharan@linux.intel.com>
Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Link: https://lore.kernel.org/r/20200325211233.27394-2-pierre-louis.bossart@linux.intel.com
Signed-off-by: Mark Brown <broonie@kernel.org>
Update tgl mach table with: Maxim98373 Amp and ALC5682 hp codec.
Both of the codecs are on I2S bus.
Signed-off-by: Jairaj Arava <jairaj.arava@intel.com>
Signed-off-by: Sathyanarayana Nujella <sathyanarayana.nujella@intel.com>
Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Link: https://lore.kernel.org/r/20200325213245.28247-5-pierre-louis.bossart@linux.intel.com
Signed-off-by: Mark Brown <broonie@kernel.org>
This patch does the below:
1. Adds the driver data and updates quirk info for TGL
with Max98373 speaker amp and ALC5682 headset codec.
2. Added max98373 speaker related code to common file for re-use.
Signed-off-by: Jairaj Arava <jairaj.arava@intel.com>
Signed-off-by: Sathyanarayana Nujella <sathyanarayana.nujella@intel.com>
Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Link: https://lore.kernel.org/r/20200325213245.28247-4-pierre-louis.bossart@linux.intel.com
Signed-off-by: Mark Brown <broonie@kernel.org>
Add "Spk Switch" and associated widget, route to max98360a
speaker amp for power saving, also remove the speaker_amp_init()
callback with complete separated tables for max98373 and max98360a.
Signed-off-by: Bhat, Uday M <uday.m.bhat@intel.com>
Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Signed-off-by: Yong Zhi <yong.zhi@intel.com>
Link: https://lore.kernel.org/r/20200325213245.28247-3-pierre-louis.bossart@linux.intel.com
Signed-off-by: Mark Brown <broonie@kernel.org>
Without the dynamic flag to allow runtime routing, the card cannot
probe on chromebooks because SOF is constantly waiting for the link.
Adding flag back to allow upstream kernels to work on rt5682 based
chromebooks since SOF can now ignore the hard coded front end.
Signed-off-by: Curtis Malainey <cujomalainey@chromium.org>
Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Link: https://lore.kernel.org/r/20200325213245.28247-2-pierre-louis.bossart@linux.intel.com
Signed-off-by: Mark Brown <broonie@kernel.org>
The signed 1 bit bitfields should be unsigned, so make them unsigned.
Signed-off-by: Colin Ian King <colin.king@canonical.com>
Reviewed-by: Codrin Ciubotariu <codrin.ciubotariu@microchip.com>
Link: https://lore.kernel.org/r/20200325132913.110115-1-colin.king@canonical.com
Signed-off-by: Mark Brown <broonie@kernel.org>
Convert the textual binding documentation for the AIC (AC97/I2S
Controller) of Ingenic SoCs to a YAML schema, and add the new compatible
strings in the process.
Signed-off-by: Paul Cercueil <paul@crapouillou.net>
Reviewed-by: Rob Herring <robh@kernel.org>
Link: https://lore.kernel.org/r/20200306222931.39664-1-paul@crapouillou.net
Signed-off-by: Mark Brown <broonie@kernel.org>
Before the JZ4770, the playback and capture sampling rates had to match.
The JZ4770 supports independent sampling rates for both.
Signed-off-by: Paul Cercueil <paul@crapouillou.net>
Link: https://lore.kernel.org/r/20200306222931.39664-6-paul@crapouillou.net
Signed-off-by: Mark Brown <broonie@kernel.org>
The change of offset for the {rx,tx}_threshold fields in the conf
register predates the JZ4780, and was first introduced in the JZ4760.
Signed-off-by: Paul Cercueil <paul@crapouillou.net>
Link: https://lore.kernel.org/r/20200306222931.39664-5-paul@crapouillou.net
Signed-off-by: Mark Brown <broonie@kernel.org>
'#sound-dai-cells' is required to properly interpret
the list of DAI specified in the 'sound-dai' property,
so add them to 'rockchip-i2s.yaml'
Signed-off-by: Johan Jonker <jbx6244@gmail.com>
Reviewed-by: Rob Herring <robh@kernel.org>
Link: https://lore.kernel.org/r/20200324094149.6904-2-jbx6244@gmail.com
Signed-off-by: Mark Brown <broonie@kernel.org>
Current dts files with 'i2s' nodes are manually verified.
In order to automate this process rockchip-i2s.txt
has to be converted to yaml.
Signed-off-by: Johan Jonker <jbx6244@gmail.com>
Reviewed-by: Rob Herring <robh@kernel.org>
Link: https://lore.kernel.org/r/20200324094149.6904-1-jbx6244@gmail.com
Signed-off-by: Mark Brown <broonie@kernel.org>
sound/soc/codecs/wm8974.c:200:38: warning:
wm8974_aux_boost_controls defined but not used [-Wunused-const-variable=]
sound/soc/codecs/wm8974.c:204:38: warning:
wm8974_mic_boost_controls defined but not used [-Wunused-const-variable=]
commit 8a123ee2a4 ("ASoC: WM8974 DAPM cleanups")
left behind this, remove them.
Signed-off-by: YueHaibing <yuehaibing@huawei.com>
Link: https://lore.kernel.org/r/20200324070615.16248-1-yuehaibing@huawei.com
Signed-off-by: Mark Brown <broonie@kernel.org>
Hello,
This small series adds audio route for built-in microphone on NVIDIA Tegra
boards that use WM8903 CODEC. In particular this is needed in order to unmute
internal microphone on Acer A500 tablet device. I'm planning to send out the
device tree for the A500 for 5.8, so will be nice to get the microphone
sorted out. Please review and apply, thanks in advance.
Dmitry Osipenko (2):
dt-bindings: sound: tegra-wm8903: Document built-in microphone audio
source
ASoC: tegra: tegra_wm8903: Support DAPM events for built-in microphone
.../sound/nvidia,tegra-audio-wm8903.txt | 1 +
sound/soc/tegra/tegra_wm8903.c | 18 ++++++++++++++++++
2 files changed, 19 insertions(+)
--
2.25.1
The patch adds a property for DMIC clock rate (hz) and changes the
default to the common optimize DMIC clock rate.
Signed-off-by: Oder Chiou <oder_chiou@realtek.com>
Link: https://lore.kernel.org/r/20200323082547.7898-1-oder_chiou@realtek.com
Signed-off-by: Mark Brown <broonie@kernel.org>
The SPDX-License-Identifier shall not be suffixed with anything further.
This makes ./scripts/spdxcheck.py complain:
sound/soc/codecs/mt6660.c: 1:36 Invalid token: //
Clean up SPDX-License-Identifier line to make spdxcheck.py happy.
Signed-off-by: Lukas Bulwahn <lukas.bulwahn@gmail.com>
Link: https://lore.kernel.org/r/20200321114022.8545-1-lukas.bulwahn@gmail.com
Signed-off-by: Mark Brown <broonie@kernel.org>
The internal microphone source is needed in order to be able to describe
the hardware audio routing for devices that have the built-in microphone
in addition to the external Mic Jack.
Signed-off-by: Dmitry Osipenko <digetx@gmail.com>
Acked-by: Stephen Warren <swarren@nvidia.com>
Acked-by: Jon Hunter <jonathanh@nvidia.com>
Link: https://lore.kernel.org/r/20200320205504.30466-2-digetx@gmail.com
Signed-off-by: Mark Brown <broonie@kernel.org>
The enable-GPIO needs to be toggled on a DAPM event in order to turn
microphone ON/OFF, otherwise microphone won't work.
Signed-off-by: Dmitry Osipenko <digetx@gmail.com>
Acked-by: Stephen Warren <swarren@nvidia.com>
Acked-by: Jon Hunter <jonathanh@nvidia.com>
Link: https://lore.kernel.org/r/20200320205504.30466-3-digetx@gmail.com
Signed-off-by: Mark Brown <broonie@kernel.org>
Now CPU/Codec DAIs are alias for dais.
Thus, we can directly use for_each_rtd_dais() macro
for soc_dai_pcm_new().
This patch merge CPU/Codec for it.
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Reviewed-by: Ranjani Sridharan <ranjani.sridharan@linux.intel.com>
Reviewed-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Link: https://lore.kernel.org/r/87r1xsolen.wl-kuninori.morimoto.gx@renesas.com
Signed-off-by: Mark Brown <broonie@kernel.org>
Now we can use for_each_rtd_dais().
Let's use it instead of for_each_rtd_cpu/codec_dais().
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Reviewed-by: Ranjani Sridharan <ranjani.sridharan@linux.intel.com>
Reviewed-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Link: https://lore.kernel.org/r/87sgi8olet.wl-kuninori.morimoto.gx@renesas.com
Signed-off-by: Mark Brown <broonie@kernel.org>
Now we can use for_each_rtd_dais().
Let's use it instead of for_each_rtd_cpu/codec_dais().
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Reviewed-by: Ranjani Sridharan <ranjani.sridharan@linux.intel.com>
Reviewed-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Link: https://lore.kernel.org/r/87tv2ooley.wl-kuninori.morimoto.gx@renesas.com
Signed-off-by: Mark Brown <broonie@kernel.org>
Now we can use for_each_rtd_dais().
Let's use it instead of for_each_rtd_cpu/codec_dais().
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Reviewed-by: Ranjani Sridharan <ranjani.sridharan@linux.intel.com>
Reviewed-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Link: https://lore.kernel.org/r/87v9n4olf4.wl-kuninori.morimoto.gx@renesas.com
Signed-off-by: Mark Brown <broonie@kernel.org>
ALSA SoC is currently categorizing CPU/Codec DAIs,
and it works well.
But modern devices require more complex connections,
for example Codec to Codec, etc, and future devices will
enable to more complex connections.
Because of these background, CPU/Codec DAIs categorizing is
no longer good much to modern device.
Currently, rtd has both CPU/Codec DAIs pointer.
rtd->cpu_dais = [][][][][][][][][]
rtd->codec_dais = [][][][][][][][][]
This patch merges these into DAIs pointer.
rtd->dais = [][][][][][][][][][][][][][][][][][]
^cpu_dais ^codec_dais
|--- num_cpus ---|--- num_codecs --|
Then, we can merge for_each_rtd_cpu/codec_dais() from this patch.
- for_each_rtd_cpu_dais() {
- ...
- }
- for_each_rtd_codec_dais() {
- ...
- }
+ for_each_rtd_dais() {
+ ...
+ }
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Reviewed-by: Ranjani Sridharan <ranjani.sridharan@linux.intel.com>
Reviewed-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Link: https://lore.kernel.org/r/87wo7kolfa.wl-kuninori.morimoto.gx@renesas.com
Signed-off-by: Mark Brown <broonie@kernel.org>
According to SoundWire Specification Version 1.2.
"A Data Port number X (in the range 0-14) which supports only one
value of WordLength may implement the WordLength field in the
DPX_BlockCtrl1 Register as Read-Only, returning the fixed value of
WordLength in response to reads."
As WSA881x interfaces in PDM mode making the only field "WordLength"
in DPX_BlockCtrl1" fixed and read-only. Behaviour of writing to this
register on WSA881x soundwire slave with Qualcomm Soundwire Controller
is throwing up an error. Not sure how other controllers deal with
writing to readonly registers, but this patch provides a way to avoid
writes to DPN_BlockCtrl1 register by providing a read_only_wordlength
flag in struct sdw_dpn_prop
Signed-off-by: Srinivas Kandagatla <srinivas.kandagatla@linaro.org>
Link: https://lore.kernel.org/r/20200311113545.23773-2-srinivas.kandagatla@linaro.org
Signed-off-by: Vinod Koul <vkoul@kernel.org>
This patchset corrects a rebind issue on STM32 SPDIFRX and I2S drivers.
The same correction has already been applied for SAI driver:
0d6defc7e0 ("ASoC: stm32: sai: manage rebind issue")
The commit e894efef9a ("ASoC: core: add support to card rebind")
allows to rebind the sound card after a rebind of one of its component.
With this commit, the sound card is actually rebound,
but may be no more functional.
The following problems have been seen on STM32 drivers.
1) DMA channel is not requested:
With the sound card rebind the simplified call sequence is:
probe
snd_soc_register_component
snd_soc_try_rebind_card
snd_soc_instantiate_card
devm_snd_dmaengine_pcm_register
The problem occurs because the pcm must be registered,
before snd_soc_instantiate_card() is called.
Modify the driver, to change the call sequence as follows:
probe
devm_snd_dmaengine_pcm_register
snd_soc_register_component
snd_soc_try_rebind_card
2) DMA channel is not released:
dma_release_channel() is not called when
devm_dmaengine_pcm_release() is executed.
This occurs because SND_DMAENGINE_PCM_DRV_NAME component,
has already been released through devm_component_release().
devm_dmaengine_pcm_release() should be called before
devm_component_release() to avoid this problem.
Call snd_dmaengine_pcm_unregister() and snd_soc_unregister_component()
explicitly from the driver, to have the right sequence.
Olivier Moysan (3):
ASoC: stm32: spdifrx: fix regmap status check
ASoC: stm32: spdifrx: manage rebind issue
ASoC: stm32: i2s: manage rebind issue
sound/soc/stm/stm32_i2s.c | 40 ++++++++++++++++------
sound/soc/stm/stm32_spdifrx.c | 64 +++++++++++++++++++----------------
2 files changed, 63 insertions(+), 41 deletions(-)
--
2.17.1
Recent addition of SoundWire stream state-machine checks in linux-next
have shown an existing issue with handling soundwire streams in codec drivers.
In general soundwire stream prepare/enable/disable can be called from either
codec/machine/controller driver. However calling it in codec driver means
that if multiple instances(Left/Right speakers) of the same codec is
connected to the same stream then it will endup calling stream
prepare/enable/disable more than once. This will mess up the stream
state-machine checks in the soundwire core.
Moving this stream handling to machine driver would fix this issue
and also allow board/platform specfic power sequencing.
Changes since v1:
- removed false error check while setting sruntime.
Srinivas Kandagatla (2):
ASoC: qcom: sdm845: handle soundwire stream
ASoC: codecs: wsa881x: remove soundwire stream handling
sound/soc/codecs/wsa881x.c | 44 +------------------------
sound/soc/qcom/Kconfig | 2 +-
sound/soc/qcom/sdm845.c | 67 ++++++++++++++++++++++++++++++++++++++
3 files changed, 69 insertions(+), 44 deletions(-)
--
2.21.0
In existing setup WSA881x codec handles soundwire stream,
however DB845c and other machines based on SDM845c have 2
instances for WSA881x codec. This will force soundwire stream
to be prepared/enabled twice or multiple times.
Handling SoundWire Stream in machine driver would fix this issue.
Signed-off-by: Srinivas Kandagatla <srinivas.kandagatla@linaro.org>
Reviewed-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Link: https://lore.kernel.org/r/20200317151233.8763-2-srinivas.kandagatla@linaro.org
Signed-off-by: Mark Brown <broonie@kernel.org>
There could be multiple instances of this codec on any platform,
so handling stream directly in this codec driver can lead to
multiple calls to prepare/enable/disable on the same SoundWire stream.
Move this stream handling to machine driver to fix this issue.
Signed-off-by: Srinivas Kandagatla <srinivas.kandagatla@linaro.org>
Link: https://lore.kernel.org/r/20200317151233.8763-3-srinivas.kandagatla@linaro.org
Signed-off-by: Mark Brown <broonie@kernel.org>
The commit e894efef9a ("ASoC: core: add support to card rebind")
allows to rebind the sound card after a rebind of one of its component.
With this commit, the sound card is actually rebound,
but may be no more functional.
Corrections:
- Call snd_dmaengine_pcm_register() before snd_soc_register_component().
- Call snd_dmaengine_pcm_unregister() and snd_soc_unregister_component()
explicitly from I2S driver.
Signed-off-by: Olivier Moysan <olivier.moysan@st.com>
Link: https://lore.kernel.org/r/20200318144125.9163-4-olivier.moysan@st.com
Signed-off-by: Mark Brown <broonie@kernel.org>
The commit e894efef9a ("ASoC: core: add support to card rebind")
allows to rebind the sound card after a rebind of one of its component.
With this commit, the sound card is actually rebound,
but may be no more functional.
Corrections:
- Call snd_dmaengine_pcm_register() before snd_soc_register_component().
- Call snd_dmaengine_pcm_unregister() and snd_soc_unregister_component()
explicitly from SPDFIRX driver.
Signed-off-by: Olivier Moysan <olivier.moysan@st.com>
Link: https://lore.kernel.org/r/20200318144125.9163-3-olivier.moysan@st.com
Signed-off-by: Mark Brown <broonie@kernel.org>
This series adds more WMA profiles and WMA decoder parameters to UAPI and
then support for these in qcom driver. It also adds FLAC and APE IDs and
decoder parameters to UAPI and then support in qcom driver
This was tested on Dragon board RB3.
Last, bump up the compressed version so that userspace can check for the
support.
Since the series touches compress uapi and asoc, it would make sense to go
thru asoc tree with acks.
Changes in v3:
- add r-b from Srini
- use macros for FLAC channel layout tags
Changes in v2:
- use bitflags for wma profiles
Vinod Koul (9):
ALSA: compress: add wma codec profiles
ALSA: compress: Add wma decoder params
ASoC: qcom: q6asm: pass codec profile to q6asm_open_write
ASoC: qcom: q6asm: add support to wma config
ASoC: qcom: q6asm-dai: add support to wma decoder
ALSA: compress: add alac & ape decoder params
ASoC: qcom: q6asm: add support for alac and ape configs
ASoC: qcom: q6asm-dai: add support for ALAC and APE decoders
ALSA: compress: bump the version
include/uapi/sound/compress_offload.h | 2 +-
include/uapi/sound/compress_params.h | 37 +++-
sound/soc/qcom/qdsp6/q6asm-dai.c | 139 ++++++++++++++-
sound/soc/qcom/qdsp6/q6asm.c | 243 +++++++++++++++++++++++++-
sound/soc/qcom/qdsp6/q6asm.h | 51 +++++-
5 files changed, 465 insertions(+), 7 deletions(-)
--
2.24.1
snd_soc_dai_get_sdw_stream() returns null if dai does not support
this callback, this is no very useful for the caller to
differentiate if this is an error or unsupported call for the dai.
return -ENOTSUPP in cases where this callback is not supported.
Signed-off-by: Srinivas Kandagatla <srinivas.kandagatla@linaro.org>
Link: https://lore.kernel.org/r/20200316151110.2580-1-srinivas.kandagatla@linaro.org
Signed-off-by: Mark Brown <broonie@kernel.org>
We have added support for bunch of new decoders and parameters for
decoders. To help users find the support bump the version up to 0,2,0.
Signed-off-by: Vinod Koul <vkoul@kernel.org>
Reviewed-by: Takashi Iwai <tiwai@suse.de>
Link: https://lore.kernel.org/r/20200316055221.1944464-10-vkoul@kernel.org
Signed-off-by: Mark Brown <broonie@kernel.org>
