This machine driver provides support for different configurations:
RT700, RT711, RT1308 (1x and 2x, I2S or SoundWire mode), and RT715
CometLake, Icelake, TigerLake.
PDM digital microphones
HDMI
To avoid introducing one driver per configuration, this common machine
driver relies on platform-specific information, tables and quirks to
dynamically create the relevant dailinks.
Unlike a lot of machine drivers, we use different DAI links for
SoundWire capture and playback since the Cadence PDIs can do capture
OR playback, not both simultaneously.
For each configuration, the card component string is updated so that UCM
can select the relevant parts.
Mark Brown [Fri, 27 Mar 2020 15:33:10 +0000 (15:33 +0000)]
Merge series "ASoC: remove rtd->cpu/codec_dai{s}" from Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>:
Hi Mark
Now, CPU/Codec DAI(s) were replaced by rtd->dais.
Thus, We don't need rtd->cpu/codec_dai{s} anymore.
This pathset replaces it by new macro.
Kuninori Morimoto (36):
ASoC: soc-core: add asoc_rtd_to_cpu/codec() macro
ASoC: amd: use asoc_rtd_to_cpu() / asoc_rtd_to_codec() macro for DAI pointer
ASoC: atmel: use asoc_rtd_to_cpu() / asoc_rtd_to_codec() macro for DAI pointer
ASoC: au1x: use asoc_rtd_to_cpu() / asoc_rtd_to_codec() macro for DAI pointer
ASoC: bcm: use asoc_rtd_to_cpu() / asoc_rtd_to_codec() macro for DAI pointer
ASoC: cirrus: use asoc_rtd_to_cpu() / asoc_rtd_to_codec() macro for DAI pointer
ASoC: dwc: use asoc_rtd_to_cpu() / asoc_rtd_to_codec() macro for DAI pointer
ASoC: fsl: use asoc_rtd_to_cpu() / asoc_rtd_to_codec() macro for DAI pointer
ASoC: generic: use asoc_rtd_to_cpu() / asoc_rtd_to_codec() macro for DAI pointer
ASoC: img: use asoc_rtd_to_cpu() / asoc_rtd_to_codec() macro for DAI pointer
ASoC: intel: use asoc_rtd_to_cpu() / asoc_rtd_to_codec() macro for DAI pointer
ASoC: kirkwood: use asoc_rtd_to_cpu() / asoc_rtd_to_codec() macro for DAI pointer
ASoC: mediatek: use asoc_rtd_to_cpu() / asoc_rtd_to_codec() macro for DAI pointer
ASoC: meson: use asoc_rtd_to_cpu() / asoc_rtd_to_codec() macro for DAI pointer
ASoC: mxs: use asoc_rtd_to_cpu() / asoc_rtd_to_codec() macro for DAI pointer
ASoC: pxa: use asoc_rtd_to_cpu() / asoc_rtd_to_codec() macro for DAI pointer
ASoC: qcom: use asoc_rtd_to_cpu() / asoc_rtd_to_codec() macro for DAI pointer
ASoC: rockchip: use asoc_rtd_to_cpu() / asoc_rtd_to_codec() macro for DAI pointer
ASoC: samsung: use asoc_rtd_to_cpu() / asoc_rtd_to_codec() macro for DAI pointer
ASoC: sh: use asoc_rtd_to_cpu() / asoc_rtd_to_codec() macro for DAI pointer
ASoC: sof: use asoc_rtd_to_cpu() / asoc_rtd_to_codec() macro for DAI pointer
ASoC: sprd: use asoc_rtd_to_cpu() / asoc_rtd_to_codec() macro for DAI pointer
ASoC: stm: use asoc_rtd_to_cpu() / asoc_rtd_to_codec() macro for DAI pointer
ASoC: sunxi: use asoc_rtd_to_cpu() / asoc_rtd_to_codec() macro for DAI pointer
ASoC: tegra: use asoc_rtd_to_cpu() / asoc_rtd_to_codec() macro for DAI pointer
ASoC: ti: use asoc_rtd_to_cpu() / asoc_rtd_to_codec() macro for DAI pointer
ASoC: txx9: use asoc_rtd_to_cpu() / asoc_rtd_to_codec() macro for DAI pointer
ASoC: uniphier: use asoc_rtd_to_cpu() / asoc_rtd_to_codec() macro for DAI pointer
ASoC: ux500: use asoc_rtd_to_cpu() / asoc_rtd_to_codec() macro for DAI pointer
ASoC: xtensa: use asoc_rtd_to_cpu() / asoc_rtd_to_codec() macro for DAI pointer
ASoC: arm: use asoc_rtd_to_cpu() / asoc_rtd_to_codec() macro for DAI pointer
ASoC: codecs: use asoc_rtd_to_cpu() / asoc_rtd_to_codec() macro for DAI pointer
ASoC: soc: use asoc_rtd_to_cpu() / asoc_rtd_to_codec() macro for DAI pointer
ASoC: soc-core: set rtd->num_cpu/codec at soc_new_pcm_runtime()
ASoC: soc-core: tidyup soc_new_pcm_runtime() rtd setups
ASoC: soc-core: remove cpu_dai/codec_dai/cpu_dais/codec_dais
Mark Brown [Fri, 27 Mar 2020 15:33:09 +0000 (15:33 +0000)]
Merge series "ASoC: SOF: Intel: add SoundWire support" from Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>:
This patchset provides the support for SoundWire support on Intel
CometLake, IcelLake and TigerLake RVP platforms and form-factor
devices to be released 'soon'.
The bulk of the code is about detecting a valid SoundWire
configuration from ACPI, and implementing the interfaces suggested in
'[PATCH 0/8] soundwire: remove platform devices, add SOF interfaces'
for interrupts, PCI wakes and clock-stop configurations.
Since that SoundWire series will not be in 5.7, the build support for
SOF w/ SoundWire is not provided for now, and fall-back functions will
be used. This code is tested on a daily basis in the SOF tree and is
not expected to change in significant ways.
Changes since v2:
Corrected error in ACPI table (thanks Amadeusz)
Added patch 11 to add reset cycle required on some SoundWire platforms
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
Rander Wang [Wed, 25 Mar 2020 21:50:25 +0000 (16:50 -0500)]
ASoC: SOF: Intel: hda: add WAKEEN interrupt support for SoundWire
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.
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.
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.
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.
ASoC: soc-acpi: expand description of _ADR-based devices
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.
Mark Brown [Thu, 26 Mar 2020 19:04:33 +0000 (19:04 +0000)]
Merge series "ASoC: rt1308-sdw: configure amplifier with set_tdm_slot()" from Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>:
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
Mark Brown [Thu, 26 Mar 2020 18:01:16 +0000 (18:01 +0000)]
ASoC: pxa: Enable AC'97 bus support for PXA machines
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.
Mark Brown [Thu, 26 Mar 2020 15:10:53 +0000 (15:10 +0000)]
ASoC: pxa: Select regmap from AC'97 machines
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.
ASoC: rt1308-sdw: use slot and rx_mask to configure stream
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.
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>
ASoC: Intel: sof_rt5682: Add support for tgl-max98373-rt5682
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.
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.
Curtis Malainey [Wed, 25 Mar 2020 21:32:42 +0000 (16:32 -0500)]
ASoC: Intel: Make glk+rt5682 echo ref dynamic
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.
Cezary Rojewski [Wed, 25 Mar 2020 13:16:11 +0000 (14:16 +0100)]
ASoC: Intel: bdw-rt5650: Revert SSP0 link to use dummy components
Recent series of patches targeting broadwell boards, while enabling
SOF, changed behavior for non-SOF solutions. In essence replacing
platform 'dummy' with actual 'platform' causes redundant stream
initialization to occur during audio start. hw_params for haswell-pcm
destroys initial stream right after its creation - only to recreate it
again from proceed from there.
While harmless so far, this flow isn't right and should be corrected.
The actual need for dummy components for SSP0 link is questionable but
that issue is subject for another series.
Fixes: a40acc6bfceb ("ASoC: Intel: bdw-rt5650: change cpu_dai and platform components for SOF") Signed-off-by: Cezary Rojewski <cezary.rojewski@intel.com> Acked-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com> Link: https://lore.kernel.org/r/20200325131611.545-4-cezary.rojewski@intel.com Signed-off-by: Mark Brown <broonie@kernel.org>
Cezary Rojewski [Wed, 25 Mar 2020 13:16:10 +0000 (14:16 +0100)]
ASoC: Intel: bdw-rt5677: Revert SSP0 link to use dummy components
Recent series of patches targeting broadwell boards, while enabling
SOF, changed behavior for non-SOF solutions. In essence replacing
platform 'dummy' with actual 'platform' causes redundant stream
initialization to occur during audio start. hw_params for haswell-pcm
destroys initial stream right after its creation - only to recreate it
again from proceed from there.
While harmless so far, this flow isn't right and should be corrected.
The actual need for dummy components for SSP0 link is questionable but
that issue is subject for another series.
Fixes: 4865bde187b2 ("ASoC: Intel: bdw-rt5677: change cpu_dai and platform components for SOF") Signed-off-by: Cezary Rojewski <cezary.rojewski@intel.com> Acked-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com> Link: https://lore.kernel.org/r/20200325131611.545-3-cezary.rojewski@intel.com Signed-off-by: Mark Brown <broonie@kernel.org>
Cezary Rojewski [Wed, 25 Mar 2020 13:16:09 +0000 (14:16 +0100)]
ASoC: Intel: broadwell: Revert back SSP0 link to use dummy components
Recent series of patches targeting broadwell boards, while enabling
SOF, changed behavior for non-SOF solutions. In essence replacing
platform 'dummy' with actual 'platform' causes redundant stream
initialization to occur during audio start. hw_params for haswell-pcm
destroys initial stream right after its creation - only to recreate it
again from proceed from there.
While harmless so far, this flow isn't correct and should be corrected.
The actual need for dummy components for SSP0 link is questionable but
that issue is subject for another series.
Link to first message in conversation:
https://lkml.org/lkml/2020/3/18/54
Fixes: 64df6afa0dab ("ASoC: Intel: broadwell: change cpu_dai and platform components for SOF") Reported-by: Dominik Brodowski <linux@dominikbrodowski.net> Signed-off-by: Cezary Rojewski <cezary.rojewski@intel.com> Acked-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com> Link: https://lore.kernel.org/r/20200325131611.545-2-cezary.rojewski@intel.com Signed-off-by: Mark Brown <broonie@kernel.org>
Paul Cercueil [Fri, 6 Mar 2020 22:29:26 +0000 (23:29 +0100)]
ASoC: Convert jz4740-i2s doc to YAML
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.
YueHaibing [Tue, 24 Mar 2020 07:06:15 +0000 (15:06 +0800)]
ASoC: wm8974: remove unused variables
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=]
Jonghwan Choi [Thu, 19 Mar 2020 14:00:44 +0000 (23:00 +0900)]
ASoC: tas2562: Fixed incorrect amp_level setting.
According to the tas2562 datasheet,the bits[5:1] represents the amp_level value.
So to set the amp_level value correctly,the shift value should be set to 1.
Mark Brown [Mon, 23 Mar 2020 18:17:26 +0000 (18:17 +0000)]
Merge series "Support built-in Mic on Tegra boards that use WM8903" from Dmitry Osipenko <digetx@gmail.com>:
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
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.
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.
soundwire: stream: Add read_only_wordlength flag to port properties
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
Mark Brown [Wed, 18 Mar 2020 21:41:27 +0000 (21:41 +0000)]
Merge series "ASoC: stm32: manage rebind issue" from Olivier Moysan <olivier.moysan@st.com>:
This patchset corrects a rebind issue on STM32 SPDIFRX and I2S drivers.
The same correction has already been applied for SAI driver: 0d6defc7e0e4 ("ASoC: stm32: sai: manage rebind issue")
The commit e894efef9ac7 ("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.
Mark Brown [Wed, 18 Mar 2020 21:41:26 +0000 (21:41 +0000)]
Merge series "ASoC: sdm845: fix soundwire stream handling" from Srinivas Kandagatla <srinivas.kandagatla@linaro.org>:
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.
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.
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.
Olivier Moysan [Wed, 18 Mar 2020 14:41:25 +0000 (15:41 +0100)]
ASoC: stm32: i2s: manage rebind issue
The commit e894efef9ac7 ("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.
Olivier Moysan [Wed, 18 Mar 2020 14:41:24 +0000 (15:41 +0100)]
ASoC: stm32: spdifrx: manage rebind issue
The commit e894efef9ac7 ("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.
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