ASoC: Intel: soc-acpi: mirror CML and TGL configurations
Some TGL devices use the same audio hardware as on CML platforms, with
RT711 on link0, RT1308 on link1 and optionally link2, and RT715 on
link 3.
To clarify configurations, the rt1308 configurations are split between
single amp on link1 and dual amps on link1. The case with two amps on
different links is already identified with the group1 attribute.
Bard Liao [Fri, 21 Aug 2020 19:55:48 +0000 (14:55 -0500)]
ASoC: Intel: modify SoundWire version id in acpi match table
The SoundWire version id of the existing RT1308, RT711, and RT715
codecs should be 2 (index for SoundWire 1.1), it was mistakenly set as
1 which pointed to the wrong version (SoundWire 1.0).
This off-by-one error had no functional impact so far since the
version number was not used, however in future patches this version
will be required.
Adam Thomson [Fri, 21 Aug 2020 14:22:59 +0000 (15:22 +0100)]
ASoC: da7219: Fix I/O voltage range configuration during probe
Previous improvements around handling device and codec level
probe functionality added the possibility of the voltage level
being undefined for the scenario where the IO voltage retrieved
from the regulator supply was below 1.2V, whereas previously the
code defaulted to the 2.5V to 3.6V range in that case. This
commit restores the default value to avoid this happening.
Fixes: aa5b18d1c290 ("ASoC: da7219: Move soft reset handling to codec level probe") Reported-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com> Signed-off-by: Adam Thomson <Adam.Thomson.Opensource@diasemi.com> Link: https://lore.kernel.org/r/20200821142259.C2ECE3FB96@swsrvapps-01.diasemi.com Signed-off-by: Mark Brown <broonie@kernel.org>
Select SoundWire capabilities on newer Intel platforms, starting with
CannonLake/CoffeeLake/CometLake.
As done for HDaudio, the SoundWire link is an opt-in capability. We
explicitly test for ACPI to avoid warnings on unmet dependencies on
the SoundWire side.
Mark Brown [Thu, 20 Aug 2020 19:30:34 +0000 (20:30 +0100)]
Merge series "Add mediatek codec mt6359 driver" from Jiaxin Yu <jiaxin.yu@mediatek.com>:
Add mediatek codec (MT6359) driver
MT6359 support playback and capture feature.
On downlink path, it includes three DACs for handset, headset,
and lineout path. On unlink path, it includeds three ADCs for
main mic, second mic, 3rd mic, and headset mic.
By scenario, select *_MUX widget to create damp path.
And by select mic_type_mux to choose DMIC/AMIC/....
For example, select these MUX widget to create headset path
(1) DAC In Mux --> "Normal Path"
(2) HP Mux --> "Audio Playback"
v6 changes:
1. Remove the compatible string in mt6359codec because MFD should be registering the platform device.
v5 changes:
1. Don't need to unregister the component whic is already relegated to devm.
2. patchwork link:
https://patchwork.kernel.org/cover/11716387/
https://patchwork.kernel.org/patch/11717757/
https://patchwork.kernel.org/patch/11716491/
v4 changes:
1. Add a remove() function to undo regulator_enable().
2. Removed unnecessary logs.
3. patchwork link:
https://patchwork.kernel.org/cover/11715553/
https://patchwork.kernel.org/patch/11716015/
https://patchwork.kernel.org/patch/11715557/
Colin Ian King [Wed, 19 Aug 2020 16:01:03 +0000 (17:01 +0100)]
ASoC: qcom: add missing out of memory check on drvdata->clks allocation
Currently drvdata->clks is not being checked for an allocation failure,
leading to potential null pointer dereferencing. Fix this by adding a
check and returning -ENOMEM if an error occurred.
Fixes: 1220f6a76e77 ("ASoC: qcom: Add common array to initialize soc based core clocks") Signed-off-by: Colin Ian King <colin.king@canonical.com> Reviewed-by: Rohit kumar <rohitkr@codeaurora.org>
Addresses-Coverity: ("Dereference null return value") Link: https://lore.kernel.org/r/20200819160103.164893-1-colin.king@canonical.com Signed-off-by: Mark Brown <broonie@kernel.org>
Samuel Holland [Wed, 19 Aug 2020 03:40:38 +0000 (22:40 -0500)]
ASoC: sun8i-codec: Hook up component probe function
Due to a mistake made while reordering patches, commit 90cac932976e
("ASoC: sun8i-codec: Fix DAPM to match the hardware topology") added
the sun8i_codec_component_probe function without referencing it from
the component definition. Add the reference so the probe function gets
called as expected.
Fixes: 90cac932976e ("ASoC: sun8i-codec: Fix DAPM to match the hardware topology") Reported-by: Stephen Rothwell <sfr@canb.auug.org.au> Signed-off-by: Samuel Holland <samuel@sholland.org> Link: https://lore.kernel.org/r/20200819034038.46418-1-samuel@sholland.org Signed-off-by: Mark Brown <broonie@kernel.org>
The constant requires indirectly including a machine header file,
but it's not actually used any more since commit 87b132bc0315 ("ASoC:
samsung: s3c24{xx,12}-i2s: port to use generic dmaengine API"), so
remove it completely.
Signed-off-by: Arnd Bergmann <arnd@arndb.de> Signed-off-by: Krzysztof Kozlowski <krzk@kernel.org> Acked-by: Mark Brown <broonie@kernel.org> Link: https://lore.kernel.org/r/20200806182059.2431-27-krzk@kernel.org Signed-off-by: Mark Brown <broonie@kernel.org>
Johan Jonker [Tue, 18 Aug 2020 14:37:26 +0000 (16:37 +0200)]
ASoC: rockchip-spdif: add description for rk3308
A test with the command below shows that the compatible string
"rockchip,rk3308-spdif", "rockchip,rk3328-spdif"
is already in use, but is not added to a document.
The current fallback string "rockchip,rk3328-spdif" points to a data
set enum RK_SPDIF_RK3366 in rockchip_spdif.c that is not used both
in the mainline as in the manufacturer kernel.
(Of the enum only RK_SPDIF_RK3288 is used.)
So if the properties don't change we might as well use the first SoC
in line as fallback string and add the description for rk3308 as:
"rockchip,rk3308-spdif", "rockchip,rk3066-spdif"
make ARCH=arm64 dtbs_check
DT_SCHEMA_FILES=Documentation/devicetree/bindings/sound/rockchip-spdif.yaml
Charles Keepax [Tue, 18 Aug 2020 16:01:26 +0000 (17:01 +0100)]
ASoC: wm_adsp: Update naming in error handling
It seems the datasheet has never used the word slave for this error
status bit and has always used the term address error. So update the
driver to match the datasheets and also in the process align a bit
better with avoiding the use of such words where possible.
Mark Brown [Tue, 18 Aug 2020 13:53:03 +0000 (14:53 +0100)]
Merge series "ASoC: qdsp6: add gapless compressed audio support" from Srinivas Kandagatla <srinivas.kandagatla@linaro.org>:
This patchset adds gapless compressed audio support on q6asm.
Gapless on q6asm is implemented using 2 streams in a single q6asm session.
First few patches such as stream id per each command, gapless flags
and silence meta data are for preparedness for adding gapless support.
Last patch implements copy callback to allow finer control over buffer offsets,
specially in partial drain cases.
This patchset is tested on RB3 aka DB845c platform.
This patchset as it is will support gapless however QDSP can also
support switching decoders on a single stream. Patches to support such feature
are send in different patchset which involves adding generic interfaces.
Thanks,
srini
Changes since v2:(mostly suggested by Pierre)
- removed unnessary kernel style comments,
- moved TIMESTAMP flag to respective patch.
- move preparatory code from gapless support patch to new one.
- fix subject prefix of one patch.
- add comments to clarify valid stream_ids
Srinivas Kandagatla (10):
ASoC: q6asm: rename misleading session id variable
ASoC: q6asm: make commands specific to streams
ASoC: q6asm: use flags directly from q6asm-dai
ASoC: q6asm: add length to write command token
ASoC: q6asm: add support to remove intial and trailing silence
ASoC: q6asm: add support to gapless flag in q6asm open
ASoC: q6asm-dai: add next track metadata support
ASoC: q6asm-dai: prepare set params to accept profile change
ASoC: q6asm-dai: add gapless support
ASoC: q6asm-dai: add support to copy callback
Mark Brown [Tue, 18 Aug 2020 13:53:02 +0000 (14:53 +0100)]
Merge series "ASoC: Intel: fix cppcheck warnings" from Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>:
This patchset tries to reduce the number of warnings on those drivers,
so that cppcheck can become a viable tool to detect issues (currently
hundreds of reports).
Most of the problems are related to unnecessary/redundant variable
assignments, prototypes and one nice logical mistake resulting in an
always-true condition.
Mark Brown [Tue, 18 Aug 2020 13:53:01 +0000 (14:53 +0100)]
Merge series "ASoC: da7219: Reorganise device/codec level probe/remove" from Adam Thomson <Adam.Thomson.Opensource@diasemi.com>:
This patch set reorganises and fixes device and codec level probe/remove
handling within the driver, to allow clean probe and remove at the codec level.
This set relates to an issue raised by Yong Zhi where a codec level re-probe
would fail due to clks still being registered from the previous instantiation.
In addition some improvements around regulator handling and soft reset have
also been included.
Adam Thomson (3):
ASoC: da7219: Move required devm_* allocations to device level code
ASoC: da7219: Move soft reset handling to codec level probe
ASoC: da7219: Fix clock handling around codec level probe
Mark Brown [Tue, 18 Aug 2020 13:52:57 +0000 (14:52 +0100)]
Merge series "ASoC: qcom: Add support for SC7180 lpass variant" from Rohit kumar <rohitkr@codeaurora.org>:
This patch chain add audio support for SC7180 soc by doing the required
modification in existing common lpass-cpu/lpass-platform driver.
This also fixes some concurrency issue.
This patch series is already tested by Srinivas on Dragon Board 410c.
Changes since v5:
- Fixed remove api in lpass-sc7180.c
- Addressed comments by Rob in yaml Documentation.
Ajit Pandey (4):
ASoC: qcom: Add common array to initialize soc based core clocks
ASoC: qcom: lpass-platform: Replace card->dev with component->dev
include: dt-bindings: sound: Add sc7180-lpass bindings header
ASoC: qcom: lpass-sc7180: Add platform driver for lpass audio
Rohit kumar (8):
ASoC: qcom: lpass-cpu: Move ahbix clk to platform specific function
ASoC: qcom: lpass-platform: fix memory leak
ASoC: qcom: lpass: Use regmap_field for i2sctl and dmactl registers
ASoC: qcom: lpass-cpu: fix concurrency issue
dt-bindings: sound: lpass-cpu: Add sc7180 lpass cpu node
ASoC: qcom: lpass-cpu: Use platform_get_resource
ASoC: qcom: lpass-platform: Use platform_get_irq
dt-bindings: sound: lpass-cpu: Move to yaml format
--
Qualcomm India Private Limited, on behalf of Qualcomm Innovation Center, Inc.,
is a member of Code Aurora Forum, a Linux Foundation Collaborative Project.
Mark Brown [Tue, 18 Aug 2020 13:52:56 +0000 (14:52 +0100)]
Merge series "Codec workaround" from Michael Sit Wei Hong <michael.wei.hong.sit@intel.com>:
This patch series enables some features on the tlv3204 codec and also fixes some issues faced while testing
v2: Fixed the build error from snd_soc_component_read32
v1: initial ASoC: codec: tlv3204: Codec workaround series
Michael Sit Wei Hong (3):
ASoC: codec: tlv3204: Enable 24 bit audio support
ASoC: codec: tlv3204: Increased maximum supported channels
ASoC: codec: tlv3204: Moving GPIO reset and add ADC reset
Mark Brown [Tue, 18 Aug 2020 13:52:55 +0000 (14:52 +0100)]
Merge series "ASoC: sun50i-codec-analog: Cleanup and power management" from Samuel Holland <samuel@sholland.org>:
This series performs some minor cleanup on the driver for the analog
codec in the Allwinner A64, and hooks up the existing mute switches to
DAPM widgets, in order to provide improved power management.
Changes since v1:
- Collected Acked-by/Reviewed-by tags
- Used SOC_MIXER_NAMED_CTL_ARRAY to avoid naming a widget "Earpiece"
Samuel Holland (8):
ASoC: sun50i-codec-analog: Fix duplicate use of ADC enable bits
ASoC: sun50i-codec-analog: Gate the amplifier clock during suspend
ASoC: sun50i-codec-analog: Group and sort mixer routes
ASoC: sun50i-codec-analog: Make headphone routes stereo
ASoC: sun50i-codec-analog: Enable DAPM for headphone switch
ASoC: sun50i-codec-analog: Make line out routes stereo
ASoC: sun50i-codec-analog: Enable DAPM for line out switch
ASoC: sun50i-codec-analog: Enable DAPM for earpiece switch
Mark Brown [Tue, 18 Aug 2020 13:52:52 +0000 (14:52 +0100)]
Merge series "Allwinner A64 digital audio codec fixes" from Samuel Holland <samuel@sholland.org>:
This series fixes a couple of issues with the digital audio codec in the
Allwinner A64 SoC:
1) Left/right channels were swapped when playing/recording audio
2) DAPM topology was wrong, breaking some kcontrols
This is the minimum set of changes necessary to fix these issues in a
backward-compatible way. For that reason, some DAPM widgets still have
incorrect or confusing names; those and other issues will be fixed in
later patch sets.
Samuel Holland (7):
ASoC: dt-bindings: Add a new compatible for the A64 codec
ASoC: sun8i-codec: Fix DAPM to match the hardware topology
ASoC: sun8i-codec: Add missing mixer routes
ASoC: sun8i-codec: Add a quirk for LRCK inversion
ARM: dts: sun8i: a33: Update codec widget names
arm64: dts: allwinner: a64: Update codec widget names
arm64: dts: allwinner: a64: Update the audio codec compatible
Mark Brown [Tue, 18 Aug 2020 13:52:50 +0000 (14:52 +0100)]
Merge series "ASoC: rt5682: Use clk APIs better" from Stephen Boyd <swboyd@chromium.org>:
This patch series drops a printk message down to dev_dbg() because it
was noisy and then migrates this driver to use clk_hw based APIs instead
of clk based APIs because this device is a clk provider, not a clk
consumer. I've only lightly tested the last two patches but I don't have
all combinations of clks for this device.
Cc: Cheng-Yi Chiang <cychiang@chromium.org> Cc: Shuming Fan <shumingf@realtek.com>
Stephen Boyd (3):
ASoC: rt5682: Use dev_dbg() in rt5682_clk_check()
ASoC: rt5682: Drop usage of __clk_get_name()
ASoC: rt5682: Use clk_hw based APIs for registration
Shengjiu Wang [Mon, 10 Aug 2020 08:11:43 +0000 (16:11 +0800)]
ASoC: fsl-asoc-card: Get "extal" clock rate by clk_get_rate
On some platform(.e.g. i.MX8QM MEK), the "extal" clock is different
with the mclk of codec, then the clock rate is also different.
So it is better to get clock rate of "extal" rate by clk_get_rate,
don't reuse the clock rate of mclk.
This allows solutions like ALSA UCM to utilize hardware mono downmix
for cases where mono output to a single speaker is desired only in
specific situations (like on a mobile phone).
Signed-off-by: Sebastian Krzyszkowiak <sebastian.krzyszkowiak@puri.sm> Acked-by: Charles Keepax <ckeepax@opensource.cirrus.com> Link: https://lore.kernel.org/r/3662154.EqNIRYjrc8@pliszka Signed-off-by: Mark Brown <broonie@kernel.org>
Randy Dunlap [Sat, 8 Aug 2020 01:21:56 +0000 (18:21 -0700)]
ASoC: SOF: delete repeated words in comments
Drop the repeated words {that, the} in comments.
Signed-off-by: Randy Dunlap <rdunlap@infradead.org> Acked-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com> Cc: Liam Girdwood <lgirdwood@gmail.com> Cc: Mark Brown <broonie@kernel.org> Cc: alsa-devel@alsa-project.org Link: https://lore.kernel.org/r/20200808012156.10827-1-rdunlap@infradead.org Signed-off-by: Mark Brown <broonie@kernel.org>
Stephan Gerhold [Sat, 1 Aug 2020 10:02:57 +0000 (12:02 +0200)]
ASoC: meson: Use snd_soc_of_parse_aux_devs()
Use the new common snd_soc_of_parse_aux_devs() helper function
to parse auxiliary devices from the device tree. The new helper
is just a copy of meson_card_add_aux_devices() so there is no
functional change.
Stephan Gerhold [Sat, 1 Aug 2020 10:02:56 +0000 (12:02 +0200)]
ASoC: simple-card: Use snd_soc_of_parse_aux_devs()
Use the new common snd_soc_of_parse_aux_devs() helper function
to parse auxiliary devices from the device tree. The code is slightly
different but the binding that is parsed is exactly the same.
Shengjiu Wang [Fri, 14 Aug 2020 09:32:41 +0000 (17:32 +0800)]
ASoC: ak4458: Add regulator support
"AVDD" is for analog power supply, "DVDD" is for digital power
supply, they can improve the power management.
As the regulator is enabled in pm runtime resume, which is
behind the component driver probe, so accessing registers in
component driver probe will fail. Fix this issue by enabling
regcache_cache_only after pm_runtime_enable.
Randy Dunlap [Sat, 8 Aug 2020 01:21:43 +0000 (18:21 -0700)]
ASoC: codecs: delete repeated words in comments
Drop the repeated words {start, it, the} in comments.
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/20200808012143.10777-1-rdunlap@infradead.org Signed-off-by: Mark Brown <broonie@kernel.org>
Randy Dunlap [Sat, 8 Aug 2020 01:22:09 +0000 (18:22 -0700)]
ASoC: various vendors: delete repeated words in comments
Drop the repeated words {related, we, is, the} in comments.
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/20200808012209.10880-1-rdunlap@infradead.org Signed-off-by: Mark Brown <broonie@kernel.org>
return ret_val;
^
sound/soc/intel/atom/sst-mfld-platform-pcm.c:384:6: note: If condition 'ret_val' is true, the function will return/exit
if (ret_val)
^
sound/soc/intel/atom/sst-mfld-platform-pcm.c:387:9: note: Returning identical expression 'ret_val'
return ret_val;
^
sound/soc/intel/atom/sst/sst.c:373:2: warning: Assignment of function
parameter has no effect outside the function. Did you forget
dereferencing it? [uselessAssignmentPtrArg]
ctx = NULL;
^
sound/soc/intel/atom/sst-mfld-platform-compress.c:46:14: style:
Variable 'ret_val' is assigned a value that is never
used. [unreadVariable]
int ret_val = 0;
^
Adam Thomson [Tue, 11 Aug 2020 16:57:25 +0000 (17:57 +0100)]
ASoC: da7219: Fix clock handling around codec level probe
Previously the driver would use devm_* related functions at
the codec level probe() to allocate clock resources for MCLK
and the DAI clocks exposed by the device. This caused issues
when registering clocks on a re-probe (no device level
remove/prove involved) as the devm_* resources were never
freed up so the clocks were still registered from the previous
codec level probe().
This commit updates the clock handling for MCLK usage and DAI
clock provision to fix this discrepancy and allow the codec level
probe/remove functionality to operate as intended.
Adam Thomson [Tue, 11 Aug 2020 16:57:24 +0000 (17:57 +0100)]
ASoC: da7219: Move soft reset handling to codec level probe
As part of the reorganisation of the device level and codec
level probe functionlity, the soft reset handling should really
reside at the codec level and after the instantiation of supplies.
This commit makes the relevant changes to support this change of
scope including the remove of devm_* functions being called for
regulator instantiation at the codec level.
Adam Thomson [Tue, 11 Aug 2020 16:57:23 +0000 (17:57 +0100)]
ASoC: da7219: Move required devm_* allocations to device level code
In preparation for cleanup of device level and codec level probe
funcitonality, all necessary devm_* allocations and fw retrieval
functions are moved to the I2C probe level code.
During gapless playback, its possible for previous track to
end at unaligned boundary, starting next track on the same
boundary can lead to unaligned address exception in dsp.
So implement copy callback for finer control on the buffer offsets.
ASoC: q6asm-dai: prepare set params to accept profile change
rearrange code so that it will be easy to change the codec
profile at runtime. This means moving exiting set_params
to an internal wrapper which can be called when codec
profile changes.
This is also preparing the code for easy to use in gapless cases.
ASoC: q6asm: add support to remove intial and trailing silence
This patch adds support to ASM_DATA_CMD_REMOVE_INITIAL_SILENCE
and ASM_DATA_CMD_REMOVE_TRAILING_SILENCE q6asm command to support
compressed metadata for gapless playback.
Each ASM session can have multiple streams attached to it,
current design was to allow only one static stream id 1 per each session.
However for use-case like gapless, we would need 2 streams to open per session.
This patch converts all the q6asm apis to take stream id as argument
to allow multiple streams to open on a single session, This is useful
for gapless playback cases.
Now the dai driver can specify which stream id for each command.
dt-bindings: sound: intel, keembay-i2s: Add new compatible string
Add a new compatible string that configures the interface to the
desired format.
Signed-off-by: Michael Sit Wei Hong <michael.wei.hong.sit@intel.com> Reviewed-by: Sia Jee Heng <jee.heng.sia@intel.com> Reviewed-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com> Link: https://lore.kernel.org/r/20200811041836.999-3-michael.wei.hong.sit@intel.com Signed-off-by: Mark Brown <broonie@kernel.org>
Enable I2S TDM audio capture for Intel Keem Bay platform.
The I2S TDM will support 4 channel and 8 channel audio capture only.
4 channel and 8 channel audio capture operates only in slave mode.
Signed-off-by: Michael Sit Wei Hong <michael.wei.hong.sit@intel.com> Reviewed-by: Sia Jee Heng <jee.heng.sia@intel.com> Reviewed-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com> Link: https://lore.kernel.org/r/20200811041836.999-2-michael.wei.hong.sit@intel.com Signed-off-by: Mark Brown <broonie@kernel.org>
ASoC: codec: tlv3204: Moving GPIO reset and add ADC reset
Moving GPIO reset to a later stage and before clock registration to
ensure that the host system and codec clocks are in sync. If the host
register clock values prior to gpio reset, the last configured codec clock
is registered to the host. The codec then gets gpio resetted setting the
codec clocks to their default value, causing a mismatch. Host system will
skip clock setting thinking the codec clocks are already at the requested
rate.
ADC reset is added to ensure the next audio capture does not have
undesired artifacts. It is probably related to the original code
where the probe function resets the ADC prior to 1st record.
Signed-off-by: Michael Sit Wei Hong <michael.wei.hong.sit@intel.com> Reviewed-by: Sia Jee Heng <jee.heng.sia@intel.com> Reviewed-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com> Link: https://lore.kernel.org/r/20200812094631.4698-4-michael.wei.hong.sit@intel.com Signed-off-by: Mark Brown <broonie@kernel.org>
ASoC: codec: tlv3204: Increased maximum supported channels
Increased maximum supported channel to 8 channels for audio capture
running in TDM mode.
Signed-off-by: Michael Sit Wei Hong <michael.wei.hong.sit@intel.com> Reviewed-by: Sia Jee Heng <jee.heng.sia@intel.com> Reviewed-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com> Link: https://lore.kernel.org/r/20200812094631.4698-3-michael.wei.hong.sit@intel.com Signed-off-by: Mark Brown <broonie@kernel.org>
Enable 24 bit in 32 bit container audio support.
Using the params_physical_width to differentiate
24 bit in 32 bit container and 24 bit in 24 bit container modes.
Use the sample rate, bit depth and channel parameters to
calculate the bit clock needed.
Signed-off-by: Michael Sit Wei Hong <michael.wei.hong.sit@intel.com> Reviewed-by: Sia Jee Heng <jee.heng.sia@intel.com> Reviewed-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com> Link: https://lore.kernel.org/r/20200812094631.4698-2-michael.wei.hong.sit@intel.com Signed-off-by: Mark Brown <broonie@kernel.org>
Samuel Holland [Sun, 26 Jul 2020 02:53:34 +0000 (21:53 -0500)]
ASoC: sun50i-codec-analog: Enable DAPM for earpiece switch
By including the earpiece mute switch in the DAPM graph, both the
earpiece amplifier and the Mixer/DAC inputs can be powered off when
the earpiece is muted.
While the widget is really just a simple switch, it is represented
as a "mixer with named controls" to avoid including the widget name
in the kcontrol name. Otherwise, it is not possible to give the widget
an accurate, descriptive name without changing the kcontrol name
seen by userspace (which should be stable).
The mute switch is between the source selection and the amplifier,
as per the diagram in the SoC manual.
Samuel Holland [Sun, 26 Jul 2020 02:53:33 +0000 (21:53 -0500)]
ASoC: sun50i-codec-analog: Enable DAPM for line out switch
By including the line out mute switch in the DAPM graph, the
Mixer/DAC inputs can be powered off when the line output is muted.
The line outputs have an unusual routing scheme. The left side mute
switch is between the source selection and the amplifier, as usual.
The right side source selection comes *after* its amplifier (and
after the left side amplifier), and its mute switch controls
whichever source is currently selected. This matches the diagram in
the SoC manual.
Samuel Holland [Sun, 26 Jul 2020 02:53:31 +0000 (21:53 -0500)]
ASoC: sun50i-codec-analog: Enable DAPM for headphone switch
By including the headphone mute switch to the DAPM graph, both the
headphone amplifier and the Mixer/DAC inputs can be powered off when
the headphones are muted.
The mute switch is between the source selection and the amplifier,
as per the diagram in the SoC manual.
Samuel Holland [Sun, 26 Jul 2020 02:53:29 +0000 (21:53 -0500)]
ASoC: sun50i-codec-analog: Group and sort mixer routes
Sort the controls in the same order as the bits in the register. Then
group the routes by sink, and sort them in the same order as the
controls. This makes it much easier to verify that all mixer inputs are
accounted for.
Samuel Holland [Sun, 26 Jul 2020 02:53:28 +0000 (21:53 -0500)]
ASoC: sun50i-codec-analog: Gate the amplifier clock during suspend
The clock must be running for the zero-crossing mute functionality.
However, it must be gated for VDD-SYS to be turned off during system
suspend. Disable it in the suspend callback, after everything has
already been muted, to avoid pops when muting/unmuting outputs.
Samuel Holland [Sun, 26 Jul 2020 02:53:27 +0000 (21:53 -0500)]
ASoC: sun50i-codec-analog: Fix duplicate use of ADC enable bits
The same enable bits are currently used for both the "Left/Right ADC"
and the "Left/Right ADC Mixer" widgets. This happens to work in practice
because the widgets are always enabled/disabled at the same time, but
each register bit should only be associated with a single widget.
To keep symmetry with the DAC widgets, keep the bits on the ADC widgets,
and remove them from the ADC Mixer widgets.
Fixes: 42371f327df0 ("ASoC: sunxi: Add new driver for Allwinner A64 codec's analog path controls") Reported-by: Ondrej Jirman <megous@megous.com> Signed-off-by: Samuel Holland <samuel@sholland.org> Acked-by: Chen-Yu Tsai <wens@csie.org> Link: https://lore.kernel.org/r/20200726025334.59931-2-samuel@sholland.org Signed-off-by: Mark Brown <broonie@kernel.org>
sound/soc/intel/boards/bytcht_cx2072x.c:102:9: warning: Identical
condition and return expression 'ret', return value is always 0
[identicalConditionAfterEarlyExit]