Daniel Baluta [Mon, 20 Jul 2020 07:20:40 +0000 (10:20 +0300)]
ASoC: SOF: define INFO_ flags in dsp_ops for imx8
In the past, the INFO_ flags such as PAUSE/NO_PERIOD_WAKEUP were
defined in the SOF PCM core, but that was changed since
commit 27e322fabd50 ("ASoC: SOF: define INFO_ flags in dsp_ops")
Now these flags must be set in DSP ops.
Signed-off-by: Daniel Baluta <daniel.baluta@nxp.com> Reviewed-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> Reviewed-by: Guennadi Liakhovetski <guennadi.liakhovetski@linux.intel.com> Link: https://lore.kernel.org/r/20200720072046.8152-2-daniel.baluta@oss.nxp.com Signed-off-by: Mark Brown <broonie@kernel.org>
Mark Brown [Tue, 21 Jul 2020 22:44:59 +0000 (23:44 +0100)]
Merge series "Add ASoC AHUB components for Tegra210 and later" from Sameer Pujar <spujar@nvidia.com>:
Overview
========
Audio Processing Engine (APE) comprises of Audio DMA (ADMA) and Audio
Hub (AHUB) unit. AHUB is a collection of hardware accelerators for audio
pre-processing and post-processing. It also includes a programmable full
crossbar for routing audio data across these accelerators.
This series exposes some of these below mentioned HW devices as ASoC
components for Tegra platforms from Tegra210 onwards.
* ADMAIF : The interface between ADMA and AHUB
* XBAR : Crossbar for routing audio samples across various modules
* I2S : Inter-IC Sound Controller
* DMIC : Digital Microphone
* DSPK : Digital Speaker
Following is the summary of current series.
* Add YAML DT binding documentation for above mentioned modules.
* Helper function for ACIF programming is exposed for Tegra210 and later.
* Add ASoC driver components for each of the above modules.
* Build ACONNECT and ADMA drivers which are essential to realize audio
use case.
* Add DT entries for above components for Tegra210, Tegra186 and
Tegra194.
As per the suggestion in [0] audio graph based sound card support
is pushed in a separate series.
[0] https://lkml.org/lkml/2020/6/27/4
Changelog
=========
v4 -> v5
--------
* Common changes
- simple-card driver changes are dropped. Changes are migrated to audio
graph card and are moved to a separate series as suggested.
- '#sound-dai-cells' property is not needed for planned audio graph card
Hence dropped from documentation and related DT binding of component
drivers.
- CIF and DAP DAIs are added for I/O drivers (DMIC, DSPK, I2S) to
represent DAI links using audio graph card. Similary DAIs are added in
AHUB driver to describe endpoints in audio crossbar. Routing is updated
to reflect the same in drivers.
v3 -> v4
--------
* [1/23] "ASoC: dt-bindings: tegra: Add DT bindings for Tegra210"
- Removed multiple examples and retained one example per doc
- Fixed as per inputs on the previous series
- Tested bindings with 'make dt_binding_check/dtbs_check'
* [2/23] "ASoC: tegra: Add support for CIF programming"
- No change
* Common changes (for patch [3/10] to [7/10])
- Mixer control overrides, for PCM parameters (rate, channel, bits),
in each driver are dropped.
- Updated routing as per DPCM usage
- Minor changes related to formatting
* New changes (patch [8/23] to [18/23] and patch [23/23])
- Based on discussions in following threads DPCM is used for Tegra Audio.
https://lkml.org/lkml/2020/2/20/91
https://lkml.org/lkml/2020/4/30/519
- The simple-card driver is used for Tegra Audio and accordingly
some enhancements are made in simple-card and core drivers.
- Patch [8/23] to [18/23] are related to simple-card and core changes.
- Patch [23/23] adds sound card support to realize complete audio path.
This is based on simple-card driver with proposed enhancements.
- Re-ordered patches depending on above
* [2/10] "ASoC: tegra: add support for CIF programming"
- Removed tegra_cif.c
- Instead added inline helper function in tegra_cif.h
* common changes (for patch [3/10] to [7/10])
- Replace LATE system calls with Normal sleep
- Remove explicit RPM suspend in driver remove() call
- Use devm_kzalloc() instead of devm_kcalloc() for single element
- Replace 'ret' with 'err' for better reading
- Consistent error printing style across drivers
- Minor formating fixes
* [8/10] "arm64: tegra: add AHUB components for few Tegra chips"
- no change
* [9/10] "arm64: tegra: enable AHUB modules for few Tegra chips"
- no change
* [10/10] "arm64: defconfig: enable AHUB components for Tegra210 and later"
(New patch)
- Enables ACONNECT and AHUB components. With this AHUB and components are
registered with ASoC core.
v1 -> v2
--------
* [1/9] "dt-bindings: sound: tegra: add DT binding for AHUB"
- no changes
* [2/9] "ASoC: tegra: add support for CIF programming"
- removed CIF programming changes for legacy chips.
- this patch now exposes helper function for CIF programming,
which can be used on Tegra210 later.
- later tegra_cif.c can be extended for legacy chips as well.
- updated commit message accordingly
* [3/9] "ASoC: tegra: add Tegra210 based DMIC driver"
- removed unnecessary initialization of 'ret' in probe()
* [4/9] "ASoC: tegra: add Tegra210 based I2S driver"
- removed unnecessary initialization of 'ret' in probe()
- fixed indentation
- added consistent bracing for if-else clauses
- updated 'rx_fifo_th' type to 'unsigned int'
- used BIT() macro for defines like '1 << {x}' in tegra210_i2s.h
* [5/9] "ASoC: tegra: add Tegra210 based AHUB driver"
- used of_device_get_match_data() to get 'soc_data' and removed
explicit of_match_device()
- used devm_platform_ioremap_resource() and removed explicit
platform_get_resource()
- fixed indentation for devm_snd_soc_register_component()
- updated commit message
- updated commit message to reflect compatible binding for Tegra186 and
Tegra194.
* [6/9] "ASoC: tegra: add Tegra186 based DSPK driver"
- removed unnecessary initialization of 'ret' in probe()
- updated 'max_th' to 'unsigned int'
- shortened lengthy macro names to avoid wrapping in
tegra186_dspk_wr_reg() and to be consistent
* [7/9] "ASoC: tegra: add Tegra210 based ADMAIF driver"
- used of_device_get_match_data() and removed explicit of_match_device()
- used BIT() macro for defines like '1 << {x}' in tegra210_admaif.h
- updated commit message to reflect compatible binding for Tegra186 and
Tegra194.
* [8/9] "arm64: tegra: add AHUB components for few Tegra chips"
- no change
* [9/9] "arm64: tegra: enable AHUB modules for few Tegra chips"
- no change
* common changes for patch [3/9] to [7/9]
- sorted headers in alphabetical order
- moved MODULE_DEVICE_TABLE() right below *_of_match table
- removed macro DRV_NAME
- removed explicit 'owner' field from platform_driver structure
- added 'const' to snd_soc_dai_ops structure
Sameer Pujar (11):
ASoC: dt-bindings: tegra: Add DT bindings for Tegra210
ASoC: tegra: Add support for CIF programming
ASoC: tegra: Add Tegra210 based DMIC driver
ASoC: tegra: Add Tegra210 based I2S driver
ASoC: tegra: Add Tegra210 based AHUB driver
ASoC: tegra: Add Tegra186 based DSPK driver
ASoC: tegra: Add Tegra210 based ADMAIF driver
arm64: defconfig: Build AHUB component drivers
arm64: defconfig: Build ADMA and ACONNECT driver
arm64: tegra: Enable ACONNECT, ADMA and AGIC on Jetson Nano
arm64: tegra: Add DT binding for AHUB components
ADMAIF is the interface between ADMA and AHUB. Each ADMA channel that
sends/receives data to/from AHUB must intreface through an ADMAIF channel.
ADMA channel sending data to AHUB pairs with an ADMAIF Tx channel and
similarly ADMA channel receiving data from AHUB pairs with an ADMAIF Rx
channel. Buffer size is configurable for each ADMAIF channel, but currently
SW uses default values.
This patch registers ADMAIF driver with ASoC framework. The component
driver exposes DAPM widgets, routes and kcontrols for the device. The DAI
driver exposes ADMAIF interfaces, which can be used to connect different
components in the ASoC layer. Makefile and Kconfig support is added to
allow to build the driver. The ADMAIF device can be enabled in the DT via
"nvidia,tegra210-admaif" compatible binding.
Tegra PCM driver is updated to expose required PCM interfaces and
snd_pcm_ops callbacks.
Mark Brown [Mon, 20 Jul 2020 15:08:24 +0000 (16:08 +0100)]
Merge series "ASoC: Intel: machine driver updates for 5.9" from Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>:
Small patchset to harden the SoundWire machine driver, change bad
HIDs, update PLL settings and avoid memory leaks. Given that the
SoundWire core parts are not upstream it's probably not necessary to
provide the patches to stable branches.
Bard Liao (1):
ASoC: Intel: sof_sdw_rt711: remove hard-coded codec name
Kai Vehmanen (2):
ASoC: Intel: sof_sdw: add support for systems without i915 audio
ASoC: Intel: sof_sdw: avoid crash if invalid DSP topology loaded
Libin Yang (1):
ASoC: Intel: common: change match table ehl-rt5660
In commit d696a61413b4 ("ASoC: rt1015: Add condition to prevent SoC
providing bclk in ratio of 50 times of sample rate."), PLL input at 50fs
is no longer supported, the new recommended settings at 48Khz rate are:
Kai Vehmanen [Fri, 17 Jul 2020 21:13:35 +0000 (16:13 -0500)]
ASoC: Intel: sof_sdw: avoid crash if invalid DSP topology loaded
The mc_private->hdmi_pcm_list is populated by elements loaded during
DSP topology load. Valid topologies for this machine driver will always
have PCM nodes for HDMI, but driver should fail gracefully even in the case
this is not true. Add a sanity check to sof_sdw_hdmi_card_late_probe()
for this case. Without the fix, a null pcm handle gets dereferenced.
Signed-off-by: Kai Vehmanen <kai.vehmanen@linux.intel.com> Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com> Reviewed-by: Guennadi Liakhovetski <guennadi.liakhovetski@linux.intel.com> Reviewed-by: Rander Wang <rander.wang@linux.intel.com> Link: https://lore.kernel.org/r/20200717211337.31956-5-pierre-louis.bossart@linux.intel.com Signed-off-by: Mark Brown <broonie@kernel.org>
Kai Vehmanen [Fri, 17 Jul 2020 21:13:34 +0000 (16:13 -0500)]
ASoC: Intel: sof_sdw: add support for systems without i915 audio
Extend the generic SOF Soundwire machine driver to support systems where
iDisp HDMI/DP audio codec is disabled for some reason (i915 driver
disabled, HDMI/DP implemented with a discrete GPU, etc). Switch codecs
to SoC dummy in the affected DAI links. This allows to reuse existing
topologies for 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: Guennadi Liakhovetski <guennadi.liakhovetski@linux.intel.com> Reviewed-by: Rander Wang <rander.wang@linux.intel.com> Link: https://lore.kernel.org/r/20200717211337.31956-4-pierre-louis.bossart@linux.intel.com Signed-off-by: Mark Brown <broonie@kernel.org>
ASoC: Intel: sof_sdw_rt711: remove properties in card remove
The rt711 jack detection properties are set from the machine drivers
during the card probe, as done in other ASoC examples.
KASAN reports a use-after-free error when unbinding drivers due to a
confusing sequence between the ACPI core, the device core and the
SoundWire device cleanups.
Rather than fixing this sequence, follow the recommendation to have
the same caller add and remove properties, add an explicit
device_remove_properties() in the card .remove() callback.
In future patches the use of device_add/remove_properties will be
replaced by a direct handling of a swnode, but the sequence will
remain the same.
Mark Brown [Mon, 20 Jul 2020 14:34:33 +0000 (15:34 +0100)]
Merge series "Add ASoC AHUB components for Tegra210 and later" from Sameer Pujar <spujar@nvidia.com>:
Overview
========
Audio Processing Engine (APE) comprises of Audio DMA (ADMA) and Audio
Hub (AHUB) unit. AHUB is a collection of hardware accelerators for audio
pre-processing and post-processing. It also includes a programmable full
crossbar for routing audio data across these accelerators.
This series exposes some of these below mentioned HW devices as ASoC
components for Tegra platforms from Tegra210 onwards.
* ADMAIF : The interface between ADMA and AHUB
* XBAR : Crossbar for routing audio samples across various modules
* I2S : Inter-IC Sound Controller
* DMIC : Digital Microphone
* DSPK : Digital Speaker
Following is the summary of current series.
* Add YAML DT binding documentation for above mentioned modules.
* Helper function for ACIF programming is exposed for Tegra210 and later.
* Add ASoC driver components for each of the above modules.
* Build ACONNECT and ADMA drivers which are essential to realize audio
use case.
* Add DT entries for above components for Tegra210, Tegra186 and
Tegra194.
As per the suggestion in [0] audio graph based sound card support
is pushed in a separate series.
[0] https://lkml.org/lkml/2020/6/27/4
Changelog
=========
v4 -> v5
--------
* Common changes
- simple-card driver changes are dropped. Changes are migrated to audio
graph card and are moved to a separate series as suggested.
- '#sound-dai-cells' property is not needed for planned audio graph card
Hence dropped from documentation and related DT binding of component
drivers.
- CIF and DAP DAIs are added for I/O drivers (DMIC, DSPK, I2S) to
represent DAI links using audio graph card. Similary DAIs are added in
AHUB driver to describe endpoints in audio crossbar. Routing is updated
to reflect the same in drivers.
v3 -> v4
--------
* [1/23] "ASoC: dt-bindings: tegra: Add DT bindings for Tegra210"
- Removed multiple examples and retained one example per doc
- Fixed as per inputs on the previous series
- Tested bindings with 'make dt_binding_check/dtbs_check'
* [2/23] "ASoC: tegra: Add support for CIF programming"
- No change
* Common changes (for patch [3/10] to [7/10])
- Mixer control overrides, for PCM parameters (rate, channel, bits),
in each driver are dropped.
- Updated routing as per DPCM usage
- Minor changes related to formatting
* New changes (patch [8/23] to [18/23] and patch [23/23])
- Based on discussions in following threads DPCM is used for Tegra Audio.
https://lkml.org/lkml/2020/2/20/91
https://lkml.org/lkml/2020/4/30/519
- The simple-card driver is used for Tegra Audio and accordingly
some enhancements are made in simple-card and core drivers.
- Patch [8/23] to [18/23] are related to simple-card and core changes.
- Patch [23/23] adds sound card support to realize complete audio path.
This is based on simple-card driver with proposed enhancements.
- Re-ordered patches depending on above
* [2/10] "ASoC: tegra: add support for CIF programming"
- Removed tegra_cif.c
- Instead added inline helper function in tegra_cif.h
* common changes (for patch [3/10] to [7/10])
- Replace LATE system calls with Normal sleep
- Remove explicit RPM suspend in driver remove() call
- Use devm_kzalloc() instead of devm_kcalloc() for single element
- Replace 'ret' with 'err' for better reading
- Consistent error printing style across drivers
- Minor formating fixes
* [8/10] "arm64: tegra: add AHUB components for few Tegra chips"
- no change
* [9/10] "arm64: tegra: enable AHUB modules for few Tegra chips"
- no change
* [10/10] "arm64: defconfig: enable AHUB components for Tegra210 and later"
(New patch)
- Enables ACONNECT and AHUB components. With this AHUB and components are
registered with ASoC core.
v1 -> v2
--------
* [1/9] "dt-bindings: sound: tegra: add DT binding for AHUB"
- no changes
* [2/9] "ASoC: tegra: add support for CIF programming"
- removed CIF programming changes for legacy chips.
- this patch now exposes helper function for CIF programming,
which can be used on Tegra210 later.
- later tegra_cif.c can be extended for legacy chips as well.
- updated commit message accordingly
* [3/9] "ASoC: tegra: add Tegra210 based DMIC driver"
- removed unnecessary initialization of 'ret' in probe()
* [4/9] "ASoC: tegra: add Tegra210 based I2S driver"
- removed unnecessary initialization of 'ret' in probe()
- fixed indentation
- added consistent bracing for if-else clauses
- updated 'rx_fifo_th' type to 'unsigned int'
- used BIT() macro for defines like '1 << {x}' in tegra210_i2s.h
* [5/9] "ASoC: tegra: add Tegra210 based AHUB driver"
- used of_device_get_match_data() to get 'soc_data' and removed
explicit of_match_device()
- used devm_platform_ioremap_resource() and removed explicit
platform_get_resource()
- fixed indentation for devm_snd_soc_register_component()
- updated commit message
- updated commit message to reflect compatible binding for Tegra186 and
Tegra194.
* [6/9] "ASoC: tegra: add Tegra186 based DSPK driver"
- removed unnecessary initialization of 'ret' in probe()
- updated 'max_th' to 'unsigned int'
- shortened lengthy macro names to avoid wrapping in
tegra186_dspk_wr_reg() and to be consistent
* [7/9] "ASoC: tegra: add Tegra210 based ADMAIF driver"
- used of_device_get_match_data() and removed explicit of_match_device()
- used BIT() macro for defines like '1 << {x}' in tegra210_admaif.h
- updated commit message to reflect compatible binding for Tegra186 and
Tegra194.
* [8/9] "arm64: tegra: add AHUB components for few Tegra chips"
- no change
* [9/9] "arm64: tegra: enable AHUB modules for few Tegra chips"
- no change
* common changes for patch [3/9] to [7/9]
- sorted headers in alphabetical order
- moved MODULE_DEVICE_TABLE() right below *_of_match table
- removed macro DRV_NAME
- removed explicit 'owner' field from platform_driver structure
- added 'const' to snd_soc_dai_ops structure
Sameer Pujar (11):
ASoC: dt-bindings: tegra: Add DT bindings for Tegra210
ASoC: tegra: Add support for CIF programming
ASoC: tegra: Add Tegra210 based DMIC driver
ASoC: tegra: Add Tegra210 based I2S driver
ASoC: tegra: Add Tegra210 based AHUB driver
ASoC: tegra: Add Tegra186 based DSPK driver
ASoC: tegra: Add Tegra210 based ADMAIF driver
arm64: defconfig: Build AHUB component drivers
arm64: defconfig: Build ADMA and ACONNECT driver
arm64: tegra: Enable ACONNECT, ADMA and AGIC on Jetson Nano
arm64: tegra: Add DT binding for AHUB components
Mark Brown [Mon, 20 Jul 2020 14:34:31 +0000 (15:34 +0100)]
Merge series "ASoC: Intel: machine driver updates for 5.9" from Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>:
Small patchset to harden the SoundWire machine driver, change bad
HIDs, update PLL settings and avoid memory leaks. Given that the
SoundWire core parts are not upstream it's probably not necessary to
provide the patches to stable branches.
Bard Liao (1):
ASoC: Intel: sof_sdw_rt711: remove hard-coded codec name
Kai Vehmanen (2):
ASoC: Intel: sof_sdw: add support for systems without i915 audio
ASoC: Intel: sof_sdw: avoid crash if invalid DSP topology loaded
Libin Yang (1):
ASoC: Intel: common: change match table ehl-rt5660
Randy Dunlap [Sun, 19 Jul 2020 00:33:07 +0000 (17:33 -0700)]
ASoC: soc-dai.h: drop a duplicated word
Drop the repeated word "be" in a comment.
Signed-off-by: Randy Dunlap <rdunlap@infradead.org> Cc: Liam Girdwood <lgirdwood@gmail.com> Cc: Mark Brown <broonie@kernel.org> Cc: alsa-devel@alsa-project.org Link: https://lore.kernel.org/r/20200719003307.21403-1-rdunlap@infradead.org Signed-off-by: Mark Brown <broonie@kernel.org>
Randy Dunlap [Sun, 19 Jul 2020 18:09:12 +0000 (11:09 -0700)]
ASoC: tegra20_das.h: delete duplicated words
Delete the doubled word "to" in two comments.
Signed-off-by: Randy Dunlap <rdunlap@infradead.org> Cc: Stephen Warren <swarren@nvidia.com> Cc: Mark Brown <broonie@kernel.org> Cc: alsa-devel@alsa-project.org Link: https://lore.kernel.org/r/20200719180912.30770-1-rdunlap@infradead.org Signed-off-by: Mark Brown <broonie@kernel.org>
ASoC: Intel: Skylake: Avoid the use of one-element array
One-element arrays are being deprecated[1]. Replace the one-element
array with a simple value type 'u8 reserved'[2], once it seems this
is just a placeholder for alignment.
The Digital Speaker Controller (DSPK) converts the multi-bit Pulse Code
Modulation (PCM) audio input to oversampled 1-bit Pulse Density Modulation
(PDM) output. From the signal flow perpsective, the DSPK can be viewed as
a PDM transmitter that up-samples the input to the desired sampling rate
by interpolation then converts the oversampled PCM input to the desired
1-bit output via Delta Sigma Modulation (DSM).
This patch registers DSPK component with ASoC framework. The component
driver exposes DAPM widgets, routes and kcontrols for the device. The DAI
driver exposes DSPK interfaces, which can be used to connect different
components in the ASoC layer. Makefile and Kconfig support is added to
allow to build the driver. The DSPK devices can be enabled in the DT via
"nvidia,tegra186-dspk" compatible binding. This driver can be used
on Tegra194 chip as well.
The Audio Hub (AHUB) comprises a collection of hardware accelerators for
audio pre/post-processing and a programmable full crossbar (XBAR) for
routing audio data across these accelerators in time and in parallel.
AHUB supports multiple interfaces to I2S, DSPK, DMIC etc., XBAR is a
switch used to configure or modify audio routing between HW accelerators
present inside AHUB.
This patch registers AHUB component with ASoC framework. The component
driver exposes DAPM widgets, routes and kcontrols for the device. The DAI
driver exposes AHUB interfaces, which can be used to connect different
components in the ASoC layer. Currently the driver takes care of XBAR
programming to allow audio data flow through various clients of the AHUB.
Makefile and Kconfig support is added to allow to build the driver. The
AHUB component can be enabled in the DT via below compatible bindings.
- "nvidia,tegra210-ahub" for Tegra210
- "nvidia,tegra186-ahub" for Tegra186 and Tegra194
The Inter-IC Sound (I2S) controller implements full-duplex, bi-directional
and single direction point to point serial interface. It can interface
with I2S compatible devices. Tegra I2S controller can operate as both
master and slave.
This patch registers I2S controller with ASoC framework. The component
driver exposes DAPM widgets, routes and kcontrols for the device. The DAI
driver exposes I2S interfaces, which can be used to connect different
components in the ASoC layer. Makefile and Kconfig support is added to
allow to build the driver. The I2S devices can be enabled in the DT via
"nvidia,tegra210-i2s" compatible binding.
The Digital MIC (DMIC) Controller is used to interface with Pulse Density
Modulation (PDM) input devices. The DMIC controller implements a converter
to convert PDM signals to Pulse Code Modulation (PCM) signals. From signal
flow perspective, the DMIC can be viewed as a PDM receiver.
This patch registers DMIC component with ASoC framework. The component
driver exposes DAPM widgets, routes and kcontrols for the device. The DAI
driver exposes DMIC interfaces, which can be used to connect different
components in the ASoC layer. Makefile and Kconfig support is added to
allow to build the driver. The DMIC devices can be enabled in the DT via
"nvidia,tegra210-dmic" compatible string. This driver can be used for
Tegra186 and Tegra194 chips as well.
Audio Client Interface (CIF) is a proprietary interface employed to route
audio samples through Audio Hub (AHUB) components by inter connecting the
various modules.
This patch exports an inline function tegra_set_cif() which can be used,
for now, to program CIF on Tegra210 and later Tegra generations. Later it
can be extended to include helpers for legacy chips as well.
ASoC: dt-bindings: tegra: Add DT bindings for Tegra210
This patch adds YAML schema for DT binding of AHUB and few of its
following components. These devices will be registered as ASoC
components and binding will be used on Tegra210 and later chips.
* ADMAIF
* I2S
* DMIC
* DSPK
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Shuming Fan [Fri, 17 Jul 2020 07:02:28 +0000 (15:02 +0800)]
ASoC: rt5682: optimize the power consumption
Some settings should set to default value after the calibration.
This patch also disables the 25MHz and 1MHz clock power when the jack unplugged.
The JD is triggered by JDH, therefore this patch removes JDL setting.
Tang Bin [Tue, 14 Jul 2020 11:27:44 +0000 (19:27 +0800)]
ASoC: qcom: qdsp6: Use IS_ERR() instead of IS_ERR_OR_NULL()
In the function q6adm_open(), q6adm_alloc_copp() doesn't return
NULL. Thus use IS_ERR() to validate the returned value instead
of IS_ERR_OR_NULL(). And delete the extra line.
Shuming Fan [Fri, 17 Jul 2020 07:02:56 +0000 (15:02 +0800)]
ASoC: rt5682: disable MICBIAS and Vref2 widget in default
The pin status of the widget was connected after the sound card registered.
The rt5682_headset_detect function will use the pin status of these two widgets
to decide the certain register setting on/off.
Therefore this patch disables the pin of these two widgets in the codec probe.
This patch could avoid the misjudgment.
ASoC: atmel: merge .digital_mute() into .mute_stream()
snd_soc_dai_digital_mute() is internally using both
mute_stream() (1) or digital_mute() (2), but the difference between
these 2 are only handling direction.
We can merge digital_mute() into mute_stream
int snd_soc_dai_digital_mute(xxx, int direction)
{
...
else if (dai->driver->ops->mute_stream)
(1) return dai->driver->ops->mute_stream(xxx, direction);
else if (direction == SNDRV_PCM_STREAM_PLAYBACK &&
dai->driver->ops->digital_mute)
(2) return dai->driver->ops->digital_mute(xxx);
...
}
-
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com> Reviewed-by: Peter Ujfalusi <peter.ujfalusi@ti.com> Reviewed-by: Alexandre Belloni <alexandre.belloni@bootlin.com> Link: https://lore.kernel.org/r/87eepb2dnq.wl-kuninori.morimoto.gx@renesas.com Signed-off-by: Mark Brown <broonie@kernel.org>
ASoC: meson: fixes the missed kfree() for axg_card_add_tdm_loopback
axg_card_add_tdm_loopback() misses to call kfree() in an error path. We
can use devm_kasprintf() to fix the issue, also improve maintainability.
So use it instead.
Fixes: c84836d7f650 ("ASoC: meson: axg-card: use modern dai_link style") Signed-off-by: Jing Xiangfeng <jingxiangfeng@huawei.com> Reviewed-by: Jerome Brunet <jbrunet@baylibre.com> Link: https://lore.kernel.org/r/20200717082242.130627-1-jingxiangfeng@huawei.com Signed-off-by: Mark Brown <broonie@kernel.org>
Mark Brown [Thu, 16 Jul 2020 22:51:54 +0000 (23:51 +0100)]
Merge series "ALSA: hda: export snd_hda_codec_cleanup()" from Kai Vehmanen <kai.vehmanen@linux.intel.com>:
Hi,
this small series is preparation for a set of bugfix ASoC patches
addressing a memleak at module unload for the HDA codec wrapper.
Instead of duplicating HDA code in ASoC tree, I chose to export
more functionality from hda_codec.c so it can be (re)used in ASoC's
hdac_hda.c.
Full series:
https://github.com/thesofproject/linux/pull/2252
Takashi and Mark, feedback is welcome on how to best handle this
kind of series where I have dependent patches both in sound/pci/hda
and in ASoC. For this series, I'm sending the patches separately
and when/if first set is merged by Takashi, I'll route the ASoC
patches via our usually SOF set to Mark.
Mark Brown [Thu, 16 Jul 2020 22:51:52 +0000 (23:51 +0100)]
Merge series "ASoC: fsl-asoc-card: Support hp and mic detection" from Shengjiu Wang <shengjiu.wang@nxp.com>:
Support hp and mic detection.
Add a parameter for asoc_simple_init_jack.
Shengjiu Wang (3):
ASoC: simple-card-utils: Support configure pin_name for
asoc_simple_init_jack
ASoC: bindings: fsl-asoc-card: Support hp-det-gpio and mic-det-gpio
ASoC: fsl-asoc-card: Support Headphone and Microphone Jack detection
changes in v2:
- Add more comments in third commit
- Add Acked-by Nicolin.
Mark Brown [Thu, 16 Jul 2020 22:51:51 +0000 (23:51 +0100)]
Merge series "ASoC: merge .digital_mute() into .mute_stream()" from Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>:
Hi Mark
These are v3 patch-set.
ALSA SoC has 2 mute callbacks (= .digital_mute(), .mute_stream()).
But the difference between these 2 are very small.
.digital_mute() is for Playback
.mute_stream() is for Playback/Capture
This patch-set adds new .no_capture_mute flag and emulate
.digital_mute() by .mute_stream().
v2 -> v3
- uses "xxx_mute_stream" for .mute_stream naming
if it was better
- removed verbose Cc email address
v1 -> v2
- return -ENOTSUPP at hdmi-codec
- add new .no_capture_mute flag and emulate .digital_mute()
by .mute_stream()
Link: https://lore.kernel.org/r/874kqy2y5t.wl-kuninori.morimoto.gx@renesas.com Link: https://lore.kernel.org/r/87ftam37ko.wl-kuninori.morimoto.gx@renesas.com
Kuninori Morimoto (21):
ASoC: hdmi-codec: return -ENOTSUPP for digital_mute
ASoC: soc-dai.c: add .no_capture_mute support
ASoC: hdmi-codec: merge .digital_mute() into .mute_stream()
ASoC: ti: merge .digital_mute() into .mute_stream()
ASoC: spear: merge .digital_mute() into .mute_stream()
ASoC: meson: merge .digital_mute() into .mute_stream()
ASoC: atmel: merge .digital_mute() into .mute_stream()
ASoC: codecs: merge .digital_mute() into .mute_stream()
ASoC: codecs: tlv*: merge .digital_mute() into .mute_stream()
ASoC: codecs: tas*: merge .digital_mute() into .mute_stream()
ASoC: codecs: ssm*: merge .digital_mute() into .mute_stream()
ASoC: codecs: pcm*: merge .digital_mute() into .mute_stream()
ASoC: codecs: max*: merge .digital_mute() into .mute_stream()
ASoC: codecs: alc*: merge .digital_mute() into .mute_stream()
ASoC: codecs: wm*: merge .digital_mute() into .mute_stream()
ASoC: codecs: es*: merge .digital_mute() into .mute_stream()
ASoC: codecs: da*: merge .digital_mute() into .mute_stream()
ASoC: codecs: cs*: merge .digital_mute() into .mute_stream()
ASoC: codecs: ak*: merge .digital_mute() into .mute_stream()
ASoC: soc-dai: remove .digital_mute
ASoC: soc-core: snd_soc_dai_digital_mute() for both CPU/Codec
Lee Jones [Wed, 15 Jul 2020 15:00:09 +0000 (16:00 +0100)]
ASoC: fsl: fsl-asoc-card: Trivial: Fix misspelling of 'exists'
Signed-off-by: Lee Jones <lee.jones@linaro.org> Cc: Timur Tabi <timur@kernel.org> Cc: Nicolin Chen <nicoleotsuka@gmail.com> Cc: Xiubo Li <Xiubo.Lee@gmail.com> Cc: Fabio Estevam <festevam@gmail.com> Cc: linuxppc-dev@lists.ozlabs.org Link: https://lore.kernel.org/r/20200715150009.407442-1-lee.jones@linaro.org Signed-off-by: Mark Brown <broonie@kernel.org>
Shengjiu Wang [Wed, 15 Jul 2020 14:09:39 +0000 (22:09 +0800)]
ASoC: fsl-asoc-card: Support Headphone and Microphone Jack detection
Use asoc_simple_init_jack function from simple card to implement
the Headphone and Microphone detection.
Register notifier to disable Speaker when Headphone is plugged in
and enable Speaker when Headphone is unplugged.
Register notifier to disable Digital Microphone when Analog Microphone
is plugged in and enable DMIC when Analog Microphone is unplugged.
Shengjiu Wang [Wed, 15 Jul 2020 14:09:37 +0000 (22:09 +0800)]
ASoC: simple-card-utils: Support configure pin_name for asoc_simple_init_jack
Currently the pin_name is fixed in asoc_simple_init_jack, but some driver
may use a different pin_name. So add a new parameter in
asoc_simple_init_jack for configuring pin_name.
If this parameter is NULL, then the default pin_name is used.
ASoC: codecs: ak*: merge .digital_mute() into .mute_stream()
snd_soc_dai_digital_mute() is internally using both
mute_stream() (1) or digital_mute() (2), but the difference between
these 2 are only handling direction.
We can merge digital_mute() into mute_stream
int snd_soc_dai_digital_mute(xxx, int direction)
{
...
else if (dai->driver->ops->mute_stream)
(1) return dai->driver->ops->mute_stream(xxx, direction);
else if (direction == SNDRV_PCM_STREAM_PLAYBACK &&
dai->driver->ops->digital_mute)
(2) return dai->driver->ops->digital_mute(xxx);
...
}
ASoC: codecs: cs*: merge .digital_mute() into .mute_stream()
snd_soc_dai_digital_mute() is internally using both
mute_stream() (1) or digital_mute() (2), but the difference between
these 2 are only handling direction.
We can merge digital_mute() into mute_stream
int snd_soc_dai_digital_mute(xxx, int direction)
{
...
else if (dai->driver->ops->mute_stream)
(1) return dai->driver->ops->mute_stream(xxx, direction);
else if (direction == SNDRV_PCM_STREAM_PLAYBACK &&
dai->driver->ops->digital_mute)
(2) return dai->driver->ops->digital_mute(xxx);
...
}
ASoC: codecs: da*: merge .digital_mute() into .mute_stream()
snd_soc_dai_digital_mute() is internally using both
mute_stream() (1) or digital_mute() (2), but the difference between
these 2 are only handling direction.
We can merge digital_mute() into mute_stream
int snd_soc_dai_digital_mute(xxx, int direction)
{
...
else if (dai->driver->ops->mute_stream)
(1) return dai->driver->ops->mute_stream(xxx, direction);
else if (direction == SNDRV_PCM_STREAM_PLAYBACK &&
dai->driver->ops->digital_mute)
(2) return dai->driver->ops->digital_mute(xxx);
...
}
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com> Reviewed-by: Peter Ujfalusi <peter.ujfalusi@ti.com> Reviewed-by: Adam Thomson <Adam.Thomson.Opensource@diasemi.com> Link: https://lore.kernel.org/r/87sge1wiwi.wl-kuninori.morimoto.gx@renesas.com Signed-off-by: Mark Brown <broonie@kernel.org>
ASoC: codecs: es*: merge .digital_mute() into .mute_stream()
snd_soc_dai_digital_mute() is internally using both
mute_stream() (1) or digital_mute() (2), but the difference between
these 2 are only handling direction.
We can merge digital_mute() into mute_stream
int snd_soc_dai_digital_mute(xxx, int direction)
{
...
else if (dai->driver->ops->mute_stream)
(1) return dai->driver->ops->mute_stream(xxx, direction);
else if (direction == SNDRV_PCM_STREAM_PLAYBACK &&
dai->driver->ops->digital_mute)
(2) return dai->driver->ops->digital_mute(xxx);
...
}
ASoC: codecs: wm*: merge .digital_mute() into .mute_stream()
snd_soc_dai_digital_mute() is internally using both
mute_stream() (1) or digital_mute() (2), but the difference between
these 2 are only handling direction.
We can merge digital_mute() into mute_stream
int snd_soc_dai_digital_mute(xxx, int direction)
{
...
else if (dai->driver->ops->mute_stream)
(1) return dai->driver->ops->mute_stream(xxx, direction);
else if (direction == SNDRV_PCM_STREAM_PLAYBACK &&
dai->driver->ops->digital_mute)
(2) return dai->driver->ops->digital_mute(xxx);
...
}
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com> Reviewed-by: Peter Ujfalusi <peter.ujfalusi@ti.com> Acked-by: Charles Keepax <ckeepax@opensource.cirrus.com> Link: https://lore.kernel.org/r/87v9ixwiwr.wl-kuninori.morimoto.gx@renesas.com Signed-off-by: Mark Brown <broonie@kernel.org>
ASoC: codecs: alc*: merge .digital_mute() into .mute_stream()
snd_soc_dai_digital_mute() is internally using both
mute_stream() (1) or digital_mute() (2), but the difference between
these 2 are only handling direction.
We can merge digital_mute() into mute_stream
int snd_soc_dai_digital_mute(xxx, int direction)
{
...
else if (dai->driver->ops->mute_stream)
(1) return dai->driver->ops->mute_stream(xxx, direction);
else if (direction == SNDRV_PCM_STREAM_PLAYBACK &&
dai->driver->ops->digital_mute)
(2) return dai->driver->ops->digital_mute(xxx);
...
}
ASoC: codecs: max*: merge .digital_mute() into .mute_stream()
snd_soc_dai_digital_mute() is internally using both
mute_stream() (1) or digital_mute() (2), but the difference between
these 2 are only handling direction.
We can merge digital_mute() into mute_stream
int snd_soc_dai_digital_mute(xxx, int direction)
{
...
else if (dai->driver->ops->mute_stream)
(1) return dai->driver->ops->mute_stream(xxx, direction);
else if (direction == SNDRV_PCM_STREAM_PLAYBACK &&
dai->driver->ops->digital_mute)
(2) return dai->driver->ops->digital_mute(xxx);
...
}
ASoC: codecs: pcm*: merge .digital_mute() into .mute_stream()
snd_soc_dai_digital_mute() is internally using both
mute_stream() (1) or digital_mute() (2), but the difference between
these 2 are only handling direction.
We can merge digital_mute() into mute_stream
int snd_soc_dai_digital_mute(xxx, int direction)
{
...
else if (dai->driver->ops->mute_stream)
(1) return dai->driver->ops->mute_stream(xxx, direction);
else if (direction == SNDRV_PCM_STREAM_PLAYBACK &&
dai->driver->ops->digital_mute)
(2) return dai->driver->ops->digital_mute(xxx);
...
}
ASoC: codecs: ssm*: merge .digital_mute() into .mute_stream()
snd_soc_dai_digital_mute() is internally using both
mute_stream() (1) or digital_mute() (2), but the difference between
these 2 are only handling direction.
We can merge digital_mute() into mute_stream
int snd_soc_dai_digital_mute(xxx, int direction)
{
...
else if (dai->driver->ops->mute_stream)
(1) return dai->driver->ops->mute_stream(xxx, direction);
else if (direction == SNDRV_PCM_STREAM_PLAYBACK &&
dai->driver->ops->digital_mute)
(2) return dai->driver->ops->digital_mute(xxx);
...
}
ASoC: codecs: tas*: merge .digital_mute() into .mute_stream()
snd_soc_dai_digital_mute() is internally using both
mute_stream() (1) or digital_mute() (2), but the difference between
these 2 are only handling direction.
We can merge digital_mute() into mute_stream
int snd_soc_dai_digital_mute(xxx, int direction)
{
...
else if (dai->driver->ops->mute_stream)
(1) return dai->driver->ops->mute_stream(xxx, direction);
else if (direction == SNDRV_PCM_STREAM_PLAYBACK &&
dai->driver->ops->digital_mute)
(2) return dai->driver->ops->digital_mute(xxx);
...
}
ASoC: codecs: tlv*: merge .digital_mute() into .mute_stream()
snd_soc_dai_digital_mute() is internally using both
mute_stream() (1) or digital_mute() (2), but the difference between
these 2 are only handling direction.
We can merge digital_mute() into mute_stream
int snd_soc_dai_digital_mute(xxx, int direction)
{
...
else if (dai->driver->ops->mute_stream)
(1) return dai->driver->ops->mute_stream(xxx, direction);
else if (direction == SNDRV_PCM_STREAM_PLAYBACK &&
dai->driver->ops->digital_mute)
(2) return dai->driver->ops->digital_mute(xxx);
...
}
ASoC: codecs: merge .digital_mute() into .mute_stream()
snd_soc_dai_digital_mute() is internally using both
mute_stream() (1) or digital_mute() (2), but the difference between
these 2 are only handling direction.
We can merge digital_mute() into mute_stream
int snd_soc_dai_digital_mute(xxx, int direction)
{
...
else if (dai->driver->ops->mute_stream)
(1) return dai->driver->ops->mute_stream(xxx, direction);
else if (direction == SNDRV_PCM_STREAM_PLAYBACK &&
dai->driver->ops->digital_mute)
(2) return dai->driver->ops->digital_mute(xxx);
...
}
ASoC: meson: merge .digital_mute() into .mute_stream()
snd_soc_dai_digital_mute() is internally using both
mute_stream() (1) or digital_mute() (2), but the difference between
these 2 are only handling direction.
We can merge digital_mute() into mute_stream
int snd_soc_dai_digital_mute(xxx, int direction)
{
...
else if (dai->driver->ops->mute_stream)
(1) return dai->driver->ops->mute_stream(xxx, direction);
else if (direction == SNDRV_PCM_STREAM_PLAYBACK &&
dai->driver->ops->digital_mute)
(2) return dai->driver->ops->digital_mute(xxx);
...
}
ASoC: spear: merge .digital_mute() into .mute_stream()
snd_soc_dai_digital_mute() is internally using both
mute_stream() (1) or digital_mute() (2), but the difference between
these 2 are only handling direction.
We can merge digital_mute() into mute_stream
int snd_soc_dai_digital_mute(xxx, int direction)
{
...
else if (dai->driver->ops->mute_stream)
(1) return dai->driver->ops->mute_stream(xxx, direction);
else if (direction == SNDRV_PCM_STREAM_PLAYBACK &&
dai->driver->ops->digital_mute)
(2) return dai->driver->ops->digital_mute(xxx);
...
}
ASoC: ti: merge .digital_mute() into .mute_stream()
snd_soc_dai_digital_mute() is internally using both
mute_stream() (1) or digital_mute() (2), but the difference between
these 2 are only handling direction.
We can merge digital_mute() into mute_stream
int snd_soc_dai_digital_mute(xxx, int direction)
{
...
else if (dai->driver->ops->mute_stream)
(1) return dai->driver->ops->mute_stream(xxx, direction);
else if (direction == SNDRV_PCM_STREAM_PLAYBACK &&
dai->driver->ops->digital_mute)
(2) return dai->driver->ops->digital_mute(xxx);
...
}
ASoC: hdmi-codec: merge .digital_mute() into .mute_stream()
snd_soc_dai_digital_mute() is internally using both
mute_stream() (1) or digital_mute() (2), but the difference between
these 2 are only handling direction.
We can merge digital_mute() into mute_stream
int snd_soc_dai_digital_mute(xxx, int direction)
{
...
else if (dai->driver->ops->mute_stream)
(1) return dai->driver->ops->mute_stream(xxx, direction);
else if (direction == SNDRV_PCM_STREAM_PLAYBACK &&
dai->driver->ops->digital_mute)
(2) return dai->driver->ops->digital_mute(xxx);
...
}
For hdmi-codec, we need to update struct hdmi_codec_ops,
and all its users in the same time.
snd_soc_dai_digital_mute() is internally using both
mute_stream() (1) or digital_mute() (2), but the difference between
these 2 are only handling "direction".
We can merge digital_mute() into mute_stream
int snd_soc_dai_digital_mute(xxx, int direction)
{
...
else if (dai->driver->ops->mute_stream)
(1) return dai->driver->ops->mute_stream(xxx, direction);
else if (direction == SNDRV_PCM_STREAM_PLAYBACK &&
dai->driver->ops->digital_mute)
(2) return dai->driver->ops->digital_mute(xxx);
...
}
To prepare merging mute_stream()/digital_mute(),
this patch adds .no_capture_mute support to emulate .digital_mute().
ASoC: hdmi-codec: return -ENOTSUPP for digital_mute
snd_soc_dai_digital_mute() will return -ENOTSUPP if driver doesn't
support mute.
In hdmi-codec case, hdmi_codec_digital_mute() will be used for it,
and each driver has .digital_mute() callback.
hdmi_codec_digital_mute() want to return -ENOTSUPP to follow it.
snd_byt_cht_es8316_mc_probe() misses to call put_device() in an error
path. Add the missed function call to fix it.
Fixes: ba49cf6f8e4a ("ASoC: Intel: bytcht_es8316: Add quirk for inverted jack detect") Signed-off-by: Jing Xiangfeng <jingxiangfeng@huawei.com> Reviewed-by: Hans de Goede <hdegoede@redhat.com> Acked-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com> Link: https://lore.kernel.org/r/20200714080918.148196-1-jingxiangfeng@huawei.com Signed-off-by: Mark Brown <broonie@kernel.org>
Mark Brown [Tue, 14 Jul 2020 15:55:17 +0000 (16:55 +0100)]
Merge series "ASoC: sh: remove discriminatory terms" from Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>:
Renesas SH drivers are using discriminatory terms.
This patch-set removes or changes it as much as possible.
But, because DMA related API function name, it still exists.
I hope all these are removed someday.
v1 -> v2
- use "secondary" instead of "follower"
- care siu/ssi drivers
- tidyup git-log
Link: https://lore.kernel.org/r/87r1tg3swv.wl-kuninori.morimoto.gx@renesas.com
Kuninori Morimoto (5):
ASoC: rsnd: don't use discriminatory terms for function names
ASoC: rsnd: don't use discriminatory terms for comment
ASoC: fsi: don't use discriminatory terms for comment
ASoC: siu: don't use discriminatory terms for parameter
ASoC: ssi: don't use discriminatory terms for debug log
This patch converts Rockchip rk3328 audio codec binding to DT schema.
And adds description about "mclk" clock and fixes some errors in
original example.
Mark Brown [Fri, 10 Jul 2020 15:06:46 +0000 (16:06 +0100)]
Merge series "ASoC: mediatek: mt8183: support DP audio" from Tzung-Bi Shih <tzungbi@google.com>:
This series is a follow up for a long time ago series
(https://patchwork.kernel.org/cover/11204303/).
The old series bound too much on the patches of DRM bridge and ASoC
machine driver. And unluckily, the dependencies
(https://lore.kernel.org/patchwork/patch/1126819/) have not applied.
Revewing the ASoC patches in the old series, I found that they could be
decoupled from the DRM bridge patches. And they are harmless as it is
an optional attribute ("hdmi-codec") in DTS.
This series arranges and rebases the harmless ASoC patches for
mt8183-mt6358-ts3a227-max98357 and mt8183-da7219-max98357.
The 1st and 4th patch add an optional DT property. The 1st patch was
acked long time ago (https://patchwork.kernel.org/patch/11204321/).
The 2nd and 5th patch add DAI link for using hdmi-codec.
The 3rd and 6th patch support the HDMI jack reporting.
Tzung-Bi Shih (6):
ASoC: dt-bindings: mt8183: add a property "mediatek,hdmi-codec"
ASoC: mediatek: mt8183: use hdmi-codec
ASoC: mediatek: mt8183: support HDMI jack reporting
ASoC: dt-bindings: mt8183-da7219: add a property "mediatek,hdmi-codec"
ASoC: mediatek: mt8183-da7219: use hdmi-codec
ASoC: mediatek: mt8183-da7219: support HDMI jack reporting
Mark Brown [Fri, 10 Jul 2020 15:06:45 +0000 (16:06 +0100)]
Merge series "ASoC: Clean-up W=1 build warnings - part2" from Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>:
Both Lee Jones and I submitted separate series, this is the second
part of the merged result, for which no feedback was provided.
I picked Lee's patches for rt5659 and ak4458 and added the pxa and
ux500 that I didn't fix. The rest is largely identical between our
respective series, with the exception of the sunxi which I documented
and Lee removed. I don't have any specific preference and will go with
the flow on this.
Changes since v3:
Improved commit subjects from 'fix kernel-doc' as suggested by Lee
Jones. In a couple of cases I just reverted to Lee's patches when the
code was identical.
Added a couple of CC: tags from Lee's patches.
Added Arnaud Pouliquen's Acked-by tag in first patch.
Lee Jones (6):
ASoC: sunxi: sun4i-spdif: Fix misspelling of 'reg_dac_txdata' in
kernel-doc
ASoC: pxa: pxa-ssp: Demote seemingly unintentional kerneldoc header
ASoC: ux500: ux500_msp_i2s: Remove unused variables 'reg_val_DR' and
'reg_val_TSTDR'
ASoC: codecs: rt5659: Remove many unused const variables
ASoC: codecs: tlv320aic26: Demote seemingly unintentional kerneldoc
header
ASoC: codecs: ak4458: Remove set but never checked variable 'ret'
Pierre-Louis Bossart (4):
ASoC: sti: uniperif: fix 'defined by not used' warning
ASoC: qcom: qdsp6: q6asm: Provide documentation for 'codec_profile'
ASoC: sunxi: sun4i-i2s: add missing clock and format arguments in
kernel-doc
ASoC: codecs: rt5631: fix spurious kernel-doc start and missing
arguments
Hans de Goede [Fri, 3 Jul 2020 10:38:40 +0000 (12:38 +0200)]
ASoC: Intel: cht_bsw_rt5672: Improve dai-set-fmt comment in cht_codec_fixup()
As Pierre-Louis Bossart pointed out, saying that the default mode for the
SSP is TDM 4 slot is not entirely accurate.
There really are 2 default modes:
The default mode for the SSP configuration is TDM 4 slot for the
cpu-dai (hard-coded in DSP firmware),
The default mode for the SSP configuration is I2S for the codec-dai
(hard-coded in the 'SSP2-Codec" .dai_fmt masks, so far unused).
This commit updates the comment in cht_codec_fixup() to properly reflect
this.
Shengjiu Wang [Tue, 7 Jul 2020 08:54:25 +0000 (16:54 +0800)]
ASoC: fsl_spdif: Clear the validity bit for TX
In IEC958 spec, "The validity bit is logical "0" if the
information in the main data field is reliable, and it
is logical "1" if it is not".
The default value of "ValCtrl" is zero, which means
"Outgoing Validity always set", then all the data is not
reliable, then some spdif sink device will drop the data.
So set "ValCtrl" to 1, that is to clear "Outgoing Validity"
in default.
Lee Jones [Thu, 9 Jul 2020 16:23:27 +0000 (11:23 -0500)]
ASoC: codecs: ak4458: Remove set but never checked variable 'ret'
Looks as though the result of snd_soc_update_bits() has never been checked.
Fixes the following W=1 kernel build warning(s):
sound/soc/codecs/ak4458.c: In function ‘ak4458_set_dai_mute’:
sound/soc/codecs/ak4458.c:408:16: warning: variable ‘ret’ set but not
used [-Wunused-but-set-variable]
Signed-off-by: Lee Jones <lee.jones@linaro.org> Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com> Cc: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com> Cc: Junichi Wakasugi <wakasugi.jb@om.asahi-kasei.co.jp> Cc: Mihai Serban <mihai.serban@nxp.com> Link: https://lore.kernel.org/r/20200709162328.259586-11-pierre-louis.bossart@linux.intel.com Signed-off-by: Mark Brown <broonie@kernel.org>
This is the only use of kerneldoc in the sourcefile and no
descriptions are provided.
Fixes the following W=1 kernel build warning(s):
sound/soc/codecs/tlv320aic26.c:138: warning: Function parameter or
member 'dai' not described in 'aic26_mute'
sound/soc/codecs/tlv320aic26.c:138: warning: Function parameter or
member 'mute' not described in 'aic26_mute'
Signed-off-by: Lee Jones <lee.jones@linaro.org> Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com> Cc: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com> Cc: Grant Likely <grant.likely@secretlab.ca> Link: https://lore.kernel.org/r/20200709162328.259586-10-pierre-louis.bossart@linux.intel.com Signed-off-by: Mark Brown <broonie@kernel.org>
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