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>
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>
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
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
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
