If pci device is in D0, wakeen interrupt will be
aggregated at cAVS level as interrupt. This commit
check the wakeen status and process it in irq thread
Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Signed-off-by: Rander Wang <rander.wang@intel.com>
Link: https://lore.kernel.org/r/20200325215027.28716-11-pierre-louis.bossart@linux.intel.com
Signed-off-by: Mark Brown <broonie@kernel.org>
When a SoundWire link is in clock stop state, a Slave device may wake
up the Master for some events such as jack detection. The WAKEEN
interrupt will be triggered and processed by the audio pci device.
If audio device is in D3, the interrupt will be routed to PME, or
aggregated at cAVS level as interrupt when audio device is in D0. This
patch only supports D3 case, where the audio pci device will be
resumed by a PME event and the WAKEEN interrupt will be processed
after audio pci device is powered up and ROM is initialized
successfully.
The WAKEEN handling is only enabled after the first boot due to
dependencies on a shim_lock mutex being initialized.
Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Signed-off-by: Rander Wang <rander.wang@intel.com>
Link: https://lore.kernel.org/r/20200325215027.28716-10-pierre-louis.bossart@linux.intel.com
Signed-off-by: Mark Brown <broonie@kernel.org>
We have a single irq handler for SOF interrupts. We can further merge
SoundWire ones to completely remove MSI interrupts handling issues
leading to timeouts.
Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Signed-off-by: Bard Liao <yung-chuan.liao@linux.intel.com>
Link: https://lore.kernel.org/r/20200325215027.28716-8-pierre-louis.bossart@linux.intel.com
Signed-off-by: Mark Brown <broonie@kernel.org>
Doing this avoid conflicts and errors reported on the bus.
The interrupts are only re-enabled on resume after the firmware is
downloaded, so the behavior is not fully symmetric
Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Link: https://lore.kernel.org/r/20200325215027.28716-7-pierre-louis.bossart@linux.intel.com
Signed-off-by: Mark Brown <broonie@kernel.org>
For now we have a limited number of machine driver configurations, and
we can detect them based on the link configuration returned after
checking hardware and firmware (BIOS) configurations.
The link configuration is checked with a link_mask as well as a list
of _ADR descriptors for each link.
There is a chance that in extreme cases where the BIOS contains too
much information we would need to detect which Slave devices actually
report as 'attached'. This would be more accurate than static
table-based solutions, but it also introduces timing dependencies
since we don't know when those devices might become attached, so will
only be only be looked at if we see limitations with static methods
and the usual quirks based e.g. on DMI information.
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: Rander Wang <rander.wang@intel.com>
Link: https://lore.kernel.org/r/20200325215027.28716-6-pierre-louis.bossart@linux.intel.com
Signed-off-by: Mark Brown <broonie@kernel.org>
These callbacks are invoked when a matching hw_params/hw_free() DAI
operation takes place, and will result in IPC operations with the SOF
firmware.
Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Link: https://lore.kernel.org/r/20200325215027.28716-5-pierre-louis.bossart@linux.intel.com
Signed-off-by: Mark Brown <broonie@kernel.org>
Now that the SoundWire core supports the multi-step initialization,
call the relevant APIs.
The actual hardware enablement can be done in two places, ideally we'd
want to startup the SoundWire IP as soon as possible (while still
taking power rail dependencies into account)
However when suspend/resume is implemented, the DSP device will be
resumed first, and only when the DSP firmware is downloaded/booted
would the SoundWire child devices be resumed, so there are only
marginal benefits in starting the IP earlier for the first probe.
Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Link: https://lore.kernel.org/r/20200325215027.28716-3-pierre-louis.bossart@linux.intel.com
Signed-off-by: Mark Brown <broonie@kernel.org>
For SoundWire, we need to know if endpoints needs to be 'aggregated'
(MIPI parlance, meaning logically grouped), e.g. when two speaker
amplifiers need to be handled as a single logical output.
We don't necessarily have the information at the firmware (BIOS)
level, so add a notion of endpoints and specify if a device/endpoint
is part of a group, with a position.
This may be expanded in future solutions, for now only provide a group
and position information.
Since we modify the header file, change all existing upstream tables
as well to avoid breaking compilation/bisect.
Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Link: https://lore.kernel.org/r/20200325215027.28716-2-pierre-louis.bossart@linux.intel.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>
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/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 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>
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>
Qualcomm DSPs also supports the ALAC and APE decoders, so add support
for these and convert the snd_codec_params to qdsp format.
Signed-off-by: Vinod Koul <vkoul@kernel.org>
Reviewed-by: Srinivas Kandagatla <srinivas.kandagatla@linaro.org>
Reviewed-by: Takashi Iwai <tiwai@suse.de>
Link: https://lore.kernel.org/r/20200316055221.1944464-9-vkoul@kernel.org
Signed-off-by: Mark Brown <broonie@kernel.org>
Qualcomm DSPs expect ALAC and APE configs to be send for decoders,
so add the API to program the respective config to the DSP.
Signed-off-by: Vinod Koul <vkoul@kernel.org>
Reviewed-by: Srinivas Kandagatla <srinivas.kandagatla@linaro.org>
Reviewed-by: Takashi Iwai <tiwai@suse.de>
Link: https://lore.kernel.org/r/20200316055221.1944464-8-vkoul@kernel.org
Signed-off-by: Mark Brown <broonie@kernel.org>
Qualcomm DSPs also supports the wma decoder, so add support for wma
decoder and convert the snd_codec_params to qdsp format.
Signed-off-by: Vinod Koul <vkoul@kernel.org>
Reviewed-by: Srinivas Kandagatla <srinivas.kandagatla@linaro.org>
Reviewed-by: Takashi Iwai <tiwai@suse.de>
Link: https://lore.kernel.org/r/20200316055221.1944464-6-vkoul@kernel.org
Signed-off-by: Mark Brown <broonie@kernel.org>
