Hans de Goede [Thu, 2 Apr 2020 18:53:58 +0000 (20:53 +0200)]
ASoC: Intel: atom: Check drv->lock is locked in sst_fill_and_send_cmd_unlocked
sst_fill_and_send_cmd_unlocked must be called with the drv->lock mutex
locked already. In the past there have been cases where this was not the
case, add a WARN_ON to check for drv->lock being locked.
sound/soc/codecs/rt5682.c:3716:1: warning: data definition has no type or storage class
module_i2c_driver(rt5682_i2c_driver);
^~~~~~~~~~~~~~~~~
sound/soc/codecs/rt5682.c:3716:1: error: type defaults to 'int' in declaration of 'module_i2c_driver' [-Werror=implicit-int]
sound/soc/codecs/rt5682.c:3716:1: warning: parameter names (without types) in function declaration
ASoC: dpcm: allow start or stop during pause for backend
soc_compr_trigger_fe() allows start or stop after pause_push.
In dpcm_be_dai_trigger(), however, only pause_release is allowed
command after pause_push.
So, start or stop after pause in compress offload is always
returned as error if the compress offload is used with dpcm.
To fix the problem, SND_SOC_DPCM_STATE_PAUSED should be allowed
for start or stop command.
이경택 [Tue, 31 Mar 2020 07:55:16 +0000 (16:55 +0900)]
ASoC: dapm: connect virtual mux with default value
Since a virtual mixer has no backing registers
to decide which path to connect,
it will try to match with initial state.
This is to ensure that the default mixer choice will be
correctly powered up during initialization.
Invert flag is used to select initial state of the virtual switch.
Since actual hardware can't be disconnected by virtual switch,
connected is better choice as initial state in many cases.
Stephan Gerhold [Mon, 30 Mar 2020 17:52:10 +0000 (19:52 +0200)]
ASoC: qcom: q6asm-dai: Add SNDRV_PCM_INFO_BATCH flag
At the moment, playing audio with PulseAudio with the qdsp6 driver
results in distorted sound. It seems like its timer-based scheduling
does not work properly with qdsp6 since setting tsched=0 in
the PulseAudio configuration avoids the issue.
Apparently this happens when the pointer() callback is not accurate
enough. There is a SNDRV_PCM_INFO_BATCH flag that can be used to stop
PulseAudio from using timer-based scheduling by default.
According to https://www.alsa-project.org/pipermail/alsa-devel/2014-March/073816.html:
The flag is being used in the sense explained in the previous audio
meeting -- the data transfer granularity isn't fine enough but aligned
to the period size (or less).
Mark Brown [Mon, 30 Mar 2020 17:22:38 +0000 (18:22 +0100)]
Merge series "ASoC: Intel: boards: Remove ignore_suspend flag from SSP0 dai link" from Cezary Rojewski <cezary.rojewski@intel.com>:
As of commit:
ASoC: soc-core: care .ignore_suspend for Component suspend
function soc-core::snd_soc_suspend no longer ignores 'ignore_suspend'
flag for dai links. While BE dai link for System Pin is
supposed to follow standard suspend-resume flow, appended
'ignore_suspend' flag disturbs that flow and causes audio to break
right after resume. Remove the flag to address this.
Link to first message in conversation:
https://lkml.org/lkml/2020/3/18/54
Cezary Rojewski (4):
ASoC: Intel: broadwell: Remove ignore_suspend flag from SSP0 dai link
ASoC: Intel: haswell: Remove ignore_suspend flag from SSP0 dai link
ASoC: Intel: bdw-rt5677: Remove ignore_suspend flag from SSP0 dai link
ASoC: Intel: bdw-rt5650: Remove ignore_suspend flag from SSP0 dai link
The addition of a single flag to track the DAI status prevents the DAI
startup sequence from being called on capture if the DAI is already
used for playback.
Fix by extending the existing code with one flag per direction.
이경택 [Mon, 30 Mar 2020 07:35:59 +0000 (16:35 +0900)]
ASoC: fix regwmask
If regwshift is 32 and the selected architecture compiles '<<' operator
for signed int literal into rotating shift, '1<<regwshift' became 1 and
it makes regwmask to 0x0.
The literal is set to unsigned long to get intended regwmask.
Cezary Rojewski [Thu, 19 Mar 2020 20:49:47 +0000 (21:49 +0100)]
ASoC: Intel: bdw-rt5650: Remove ignore_suspend flag from SSP0 dai link
As of commit:
ASoC: soc-core: care .ignore_suspend for Component suspend
function soc-core::snd_soc_suspend no longer ignores 'ignore_suspend'
flag for dai links. While BE dai link for System Pin is
supposed to follow standard suspend-resume flow, appended
'ignore_suspend' flag disturbs that flow and causes audio to break
right after resume. Remove the flag to address this.
Signed-off-by: Cezary Rojewski <cezary.rojewski@intel.com> Acked-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com> Cc: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com> Cc: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com> Cc: Dominik Brodowski <linux@dominikbrodowski.net> Cc: Mark Brown <broonie@kernel.org> Link: https://lore.kernel.org/r/20200319204947.18963-5-cezary.rojewski@intel.com Signed-off-by: Mark Brown <broonie@kernel.org>
Cezary Rojewski [Thu, 19 Mar 2020 20:49:46 +0000 (21:49 +0100)]
ASoC: Intel: bdw-rt5677: Remove ignore_suspend flag from SSP0 dai link
As of commit:
ASoC: soc-core: care .ignore_suspend for Component suspend
function soc-core::snd_soc_suspend no longer ignores 'ignore_suspend'
flag for dai links. While BE dai link for System Pin is
supposed to follow standard suspend-resume flow, appended
'ignore_suspend' flag disturbs that flow and causes audio to break
right after resume. Remove the flag to address this.
Signed-off-by: Cezary Rojewski <cezary.rojewski@intel.com> Acked-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com> Cc: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com> Cc: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com> Cc: Dominik Brodowski <linux@dominikbrodowski.net> Cc: Mark Brown <broonie@kernel.org> Link: https://lore.kernel.org/r/20200319204947.18963-4-cezary.rojewski@intel.com Signed-off-by: Mark Brown <broonie@kernel.org>
Cezary Rojewski [Thu, 19 Mar 2020 20:49:45 +0000 (21:49 +0100)]
ASoC: Intel: haswell: Remove ignore_suspend flag from SSP0 dai link
As of commit:
ASoC: soc-core: care .ignore_suspend for Component suspend
function soc-core::snd_soc_suspend no longer ignores 'ignore_suspend'
flag for dai links. While BE dai link for System Pin is
supposed to follow standard suspend-resume flow, appended
'ignore_suspend' flag disturbs that flow and causes audio to break
right after resume. Remove the flag to address this.
Signed-off-by: Cezary Rojewski <cezary.rojewski@intel.com> Acked-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com> Cc: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com> Cc: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com> Cc: Dominik Brodowski <linux@dominikbrodowski.net> Cc: Mark Brown <broonie@kernel.org> Link: https://lore.kernel.org/r/20200319204947.18963-3-cezary.rojewski@intel.com Signed-off-by: Mark Brown <broonie@kernel.org>
Cezary Rojewski [Thu, 19 Mar 2020 20:49:44 +0000 (21:49 +0100)]
ASoC: Intel: broadwell: Remove ignore_suspend flag from SSP0 dai link
As of commit:
ASoC: soc-core: care .ignore_suspend for Component suspend
function soc-core::snd_soc_suspend no longer ignores 'ignore_suspend'
flag for dai links. While BE dai link for System Pin is
supposed to follow standard suspend-resume flow, appended
'ignore_suspend' flag disturbs that flow and causes audio to break
right after resume. Remove the flag to address this.
Link to first message in conversation:
https://lkml.org/lkml/2020/3/18/54
Reported-by: Dominik Brodowski <linux@dominikbrodowski.net> Suggested-by: Mark Brown <broonie@kernel.org> Signed-off-by: Cezary Rojewski <cezary.rojewski@intel.com> Acked-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com> Cc: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com> Cc: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com> Link: https://lore.kernel.org/r/20200319204947.18963-2-cezary.rojewski@intel.com Signed-off-by: Mark Brown <broonie@kernel.org>
Takashi Iwai [Mon, 30 Mar 2020 13:56:45 +0000 (15:56 +0200)]
ASoC: bcm: Fix pointer cast warning
The NULL check can be done gracefully without cast. It fixes a
compile warning like:
sound/soc/bcm/bcm63xx-pcm-whistler.c:184:6: warning: cast to pointer from integer of different size [-Wint-to-pointer-cast]
Fabio Estevam [Fri, 27 Mar 2020 15:57:21 +0000 (12:57 -0300)]
dt-bindings: sound: cs42l51: Remove unneeded I2C unit name
The following warning is seen with 'make dt_binding_check':
Documentation/devicetree/bindings/sound/cirrus,cs42l51.example.dts:18.15-34.11: Warning (unit_address_vs_reg): /example-0/i2c@0: node has a unit name, but no reg or ranges property
Mark Brown [Fri, 27 Mar 2020 17:28:36 +0000 (17:28 +0000)]
Merge series "ASoC: Intel: add SoundWire machine driver" from Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>:
To handle multiple hardware combinations, this patchset suggests a
single machine driver which will create and initialize dailinks
dynamically. This allows us to support new configurations easily, as
shown with the TigerLake rt5682 example.
Each configuration updates the card component string, and UCM can test
for the presence of components to configure them as needed.
Since we use a single the machine driver name, all previous ACPI
tables need to be updated. That should have no impact since the
machine drivers listed at the time were not upstreamed and are no
longer maintained.
Naveen Manohar (2):
ASoC: Intel: common: add match table for TGL RT5682 SoundWire driver
ASoC: Intel: sof_sdw: Add Volteer support with RT5682 SNDW helper
function
Dan Murphy [Fri, 27 Mar 2020 16:24:32 +0000 (11:24 -0500)]
ASoC: tlv320adcx140: Remove undocumented property
Remove undocumented and unneeded ti,use-internal-reg from the example as
it was an artifact from initial development. The code does not query
for this property and as the document indicates if areg-supply is
undefined then the internal regulator is used.
Fixes: 302c0b7490cd ("dt-bindings: sound: Add TLV320ADCx140 dt
bindings") Signed-off-by: Dan Murphy <dmurphy@ti.com> CC: Rob Herring <robh@kernel.org> Link: https://lore.kernel.org/r/20200327162432.17067-1-dmurphy@ti.com Signed-off-by: Mark Brown <broonie@kernel.org>
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: c079d427cc3c ("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: 822cc0095db3 ("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: 3120da671a6f ("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=]
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