We can get codec name from dai link.
Suggested-by: Rander Wang <rander.wang@intel.com>
Signed-off-by: Bard Liao <yung-chuan.liao@linux.intel.com>
Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Reviewed-by: Kai Vehmanen <kai.vehmanen@linux.intel.com>
Link: https://lore.kernel.org/r/20200717211337.31956-2-pierre-louis.bossart@linux.intel.com
Signed-off-by: Mark Brown <broonie@kernel.org>
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/2020/2/20/91https://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
v2 -> v3
--------
* [1/10] "dt-bindings: sound: tegra: add DT binding for AHUB
- Updated licence
- Removed redundancy w.r.t items/const/enum
- Added constraints wherever needed with "pattern" property
* [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
.../bindings/sound/nvidia,tegra186-dspk.yaml | 83 +++
.../bindings/sound/nvidia,tegra210-admaif.yaml | 111 +++
.../bindings/sound/nvidia,tegra210-ahub.yaml | 136 ++++
.../bindings/sound/nvidia,tegra210-dmic.yaml | 83 +++
.../bindings/sound/nvidia,tegra210-i2s.yaml | 101 +++
arch/arm64/boot/dts/nvidia/tegra186.dtsi | 217 +++++-
arch/arm64/boot/dts/nvidia/tegra194.dtsi | 225 +++++-
arch/arm64/boot/dts/nvidia/tegra210-p3450-0000.dts | 12 +
arch/arm64/boot/dts/nvidia/tegra210.dtsi | 140 ++++
arch/arm64/configs/defconfig | 8 +
sound/soc/tegra/Kconfig | 56 ++
sound/soc/tegra/Makefile | 10 +
sound/soc/tegra/tegra186_dspk.c | 442 +++++++++++
sound/soc/tegra/tegra186_dspk.h | 70 ++
sound/soc/tegra/tegra210_admaif.c | 800 ++++++++++++++++++++
sound/soc/tegra/tegra210_admaif.h | 162 ++++
sound/soc/tegra/tegra210_ahub.c | 676 +++++++++++++++++
sound/soc/tegra/tegra210_ahub.h | 127 ++++
sound/soc/tegra/tegra210_dmic.c | 455 ++++++++++++
sound/soc/tegra/tegra210_dmic.h | 82 +++
sound/soc/tegra/tegra210_i2s.c | 812 +++++++++++++++++++++
sound/soc/tegra/tegra210_i2s.h | 126 ++++
sound/soc/tegra/tegra_cif.h | 65 ++
sound/soc/tegra/tegra_pcm.c | 235 +++++-
sound/soc/tegra/tegra_pcm.h | 21 +-
25 files changed, 5251 insertions(+), 4 deletions(-)
create mode 100644 Documentation/devicetree/bindings/sound/nvidia,tegra186-dspk.yaml
create mode 100644 Documentation/devicetree/bindings/sound/nvidia,tegra210-admaif.yaml
create mode 100644 Documentation/devicetree/bindings/sound/nvidia,tegra210-ahub.yaml
create mode 100644 Documentation/devicetree/bindings/sound/nvidia,tegra210-dmic.yaml
create mode 100644 Documentation/devicetree/bindings/sound/nvidia,tegra210-i2s.yaml
create mode 100644 sound/soc/tegra/tegra186_dspk.c
create mode 100644 sound/soc/tegra/tegra186_dspk.h
create mode 100644 sound/soc/tegra/tegra210_admaif.c
create mode 100644 sound/soc/tegra/tegra210_admaif.h
create mode 100644 sound/soc/tegra/tegra210_ahub.c
create mode 100644 sound/soc/tegra/tegra210_ahub.h
create mode 100644 sound/soc/tegra/tegra210_dmic.c
create mode 100644 sound/soc/tegra/tegra210_dmic.h
create mode 100644 sound/soc/tegra/tegra210_i2s.c
create mode 100644 sound/soc/tegra/tegra210_i2s.h
create mode 100644 sound/soc/tegra/tegra_cif.h
--
2.7.4
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.
Signed-off-by: Sameer Pujar <spujar@nvidia.com>
Link: https://lore.kernel.org/r/1595134890-16470-7-git-send-email-spujar@nvidia.com
Signed-off-by: Mark Brown <broonie@kernel.org>
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
Signed-off-by: Sameer Pujar <spujar@nvidia.com>
Link: https://lore.kernel.org/r/1595134890-16470-6-git-send-email-spujar@nvidia.com
Signed-off-by: Mark Brown <broonie@kernel.org>
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.
Signed-off-by: Sameer Pujar <spujar@nvidia.com>
Link: https://lore.kernel.org/r/1595134890-16470-5-git-send-email-spujar@nvidia.com
Signed-off-by: Mark Brown <broonie@kernel.org>
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.
Signed-off-by: Sameer Pujar <spujar@nvidia.com>
Link: https://lore.kernel.org/r/1595134890-16470-4-git-send-email-spujar@nvidia.com
Signed-off-by: Mark Brown <broonie@kernel.org>
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.
Signed-off-by: Sameer Pujar <spujar@nvidia.com>
Reviewed-by: Jon Hunter <jonathanh@nvidia.com>
Reviewed-by: Dmitry Osipenko <digetx@gmail.com>
Link: https://lore.kernel.org/r/1595134890-16470-3-git-send-email-spujar@nvidia.com
Signed-off-by: Mark Brown <broonie@kernel.org>
This configuration is for EHL with the RT5660 codec. RT5660
should use "10EC5660" ID instead of "INTC1027".
Signed-off-by: Libin Yang <libin.yang@intel.com>
Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Reviewed-by: Bard Liao <yung-chuan.liao@linux.intel.com>
Link: https://lore.kernel.org/r/20200717211337.31956-7-pierre-louis.bossart@linux.intel.com
Signed-off-by: Mark Brown <broonie@kernel.org>
All drivers are now using .mute_stream.
Let's remove .digital_mute.
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Reviewed-by: Peter Ujfalusi <peter.ujfalusi@ti.com>
Link: https://lore.kernel.org/r/87h7u72dqz.wl-kuninori.morimoto.gx@renesas.com
Signed-off-by: Mark Brown <broonie@kernel.org>
An awful lot of mostly small fixes here, mainly for x86 based platforms
and the CODEC drivers mainly used on them. For the most part this is
either minor device specific stuff which seems to come from detailed
testing or robustness against errors which comes from people having done
some fuzzing runs aginst the topology code.
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Merge tag 'asoc-fix-v5.8-rc5' of https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound into for-linus
ASoC: Fixes for v5.8
An awful lot of mostly small fixes here, mainly for x86 based platforms
and the CODEC drivers mainly used on them. For the most part this is
either minor device specific stuff which seems to come from detailed
testing or robustness against errors which comes from people having done
some fuzzing runs aginst the topology code.
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.
Signed-off-by: Shuming Fan <shumingf@realtek.com>
Link: https://lore.kernel.org/r/20200717070228.28660-1-shumingf@realtek.com
Signed-off-by: Mark Brown <broonie@kernel.org>
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.
Signed-off-by: Zhang Shengju <zhangshengju@cmss.chinamobile.com>
Signed-off-by: Tang Bin <tangbin@cmss.chinamobile.com>
Link: https://lore.kernel.org/r/20200714112744.20560-1-tangbin@cmss.chinamobile.com
Signed-off-by: Mark Brown <broonie@kernel.org>
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.
Signed-off-by: Shuming Fan <shumingf@realtek.com>
Link: https://lore.kernel.org/r/20200717070256.28712-1-shumingf@realtek.com
Signed-off-by: Mark Brown <broonie@kernel.org>
snd_soc_dai_digital_mute() is used for both CPU and Codec.
For example, soc_pcm_prepare() / soc_pcm_hw_free() are caring
both CPU and Codec.
But soc_resume_deferred() / snd_soc_suspend() are not.
This patch cares it.
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Reviewed-by: Peter Ujfalusi <peter.ujfalusi@ti.com>
Link: https://lore.kernel.org/r/87ft9r2dqr.wl-kuninori.morimoto.gx@renesas.com
Signed-off-by: Mark Brown <broonie@kernel.org>
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>
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: c84836d7f6 ("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>
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.
.../bindings/sound/fsl-asoc-card.txt | 3 +
include/sound/simple_card_utils.h | 6 +-
sound/soc/fsl/Kconfig | 1 +
sound/soc/fsl/fsl-asoc-card.c | 77 ++++++++++++++++++-
sound/soc/generic/simple-card-utils.c | 7 +-
5 files changed, 86 insertions(+), 8 deletions(-)
--
2.27.0
Add missed return for calling soc_component_ret, otherwise the return
value is wrong.
Fixes: e2329eeba4 ("ASoC: soc-component: add soc_component_err()")
Signed-off-by: Shengjiu Wang <shengjiu.wang@nxp.com>
Acked-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Link: https://lore.kernel.org/r/1594876028-1845-1-git-send-email-shengjiu.wang@nxp.com
Signed-off-by: Mark Brown <broonie@kernel.org>
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.
Signed-off-by: Shengjiu Wang <shengjiu.wang@nxp.com>
Acked-by: Nicolin Chen <nicoleotsuka@gmail.com>
Link: https://lore.kernel.org/r/1594822179-1849-4-git-send-email-shengjiu.wang@nxp.com
Signed-off-by: Mark Brown <broonie@kernel.org>
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.
Signed-off-by: Shengjiu Wang <shengjiu.wang@nxp.com>
Acked-by: Nicolin Chen <nicoleotsuka@gmail.com>
Link: https://lore.kernel.org/r/1594822179-1849-2-git-send-email-shengjiu.wang@nxp.com
Signed-off-by: Mark Brown <broonie@kernel.org>
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>
Link: https://lore.kernel.org/r/87pn95wiwa.wl-kuninori.morimoto.gx@renesas.com
Signed-off-by: Mark Brown <broonie@kernel.org>
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>
Link: https://lore.kernel.org/r/87r1tlwiwe.wl-kuninori.morimoto.gx@renesas.com
Signed-off-by: Mark Brown <broonie@kernel.org>
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>
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>
Link: https://lore.kernel.org/r/87tuyhwiwm.wl-kuninori.morimoto.gx@renesas.com
Signed-off-by: Mark Brown <broonie@kernel.org>
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>
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>
Link: https://lore.kernel.org/r/87wo3dwiwv.wl-kuninori.morimoto.gx@renesas.com
Signed-off-by: Mark Brown <broonie@kernel.org>
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>
Link: https://lore.kernel.org/r/87y2ntwix0.wl-kuninori.morimoto.gx@renesas.com
Signed-off-by: Mark Brown <broonie@kernel.org>
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>
Link: https://lore.kernel.org/r/87zh89wix5.wl-kuninori.morimoto.gx@renesas.com
Signed-off-by: Mark Brown <broonie@kernel.org>
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>
Link: https://lore.kernel.org/r/871rllxxhp.wl-kuninori.morimoto.gx@renesas.com
Signed-off-by: Mark Brown <broonie@kernel.org>
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>
Link: https://lore.kernel.org/r/873661xxhu.wl-kuninori.morimoto.gx@renesas.com
Signed-off-by: Mark Brown <broonie@kernel.org>
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>
Link: https://lore.kernel.org/r/874kqhxxhz.wl-kuninori.morimoto.gx@renesas.com
Signed-off-by: Mark Brown <broonie@kernel.org>
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>
Link: https://lore.kernel.org/r/875zaxxxi4.wl-kuninori.morimoto.gx@renesas.com
Signed-off-by: Mark Brown <broonie@kernel.org>
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>
Link: https://lore.kernel.org/r/878sftxxie.wl-kuninori.morimoto.gx@renesas.com
Signed-off-by: Mark Brown <broonie@kernel.org>
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>
Link: https://lore.kernel.org/r/87a709xxij.wl-kuninori.morimoto.gx@renesas.com
Signed-off-by: Mark Brown <broonie@kernel.org>
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>
Link: https://lore.kernel.org/r/87blkpxxip.wl-kuninori.morimoto.gx@renesas.com
Signed-off-by: Mark Brown <broonie@kernel.org>
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.
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Reviewed-by: Peter Ujfalusi <peter.ujfalusi@ti.com>
Link: https://lore.kernel.org/r/87d055xxj2.wl-kuninori.morimoto.gx@renesas.com
Signed-off-by: Mark Brown <broonie@kernel.org>
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().
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Link: https://lore.kernel.org/r/87eeplxxj7.wl-kuninori.morimoto.gx@renesas.com
Signed-off-by: Mark Brown <broonie@kernel.org>
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.
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Link: https://lore.kernel.org/r/87fta1xxjc.wl-kuninori.morimoto.gx@renesas.com
Signed-off-by: Mark Brown <broonie@kernel.org>
The irq work will be manipulated by resume function, and it will report
the wrong jack type while the jack type is headphone in the button event.
Signed-off-by: Oder Chiou <oder_chiou@realtek.com>
Link: https://lore.kernel.org/r/20200716030123.27122-1-oder_chiou@realtek.com
Signed-off-by: Mark Brown <broonie@kernel.org>
snd_byt_cht_es8316_mc_probe() misses to call put_device() in an error
path. Add the missed function call to fix it.
Fixes: ba49cf6f8e ("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>
Now that there's a function that calculates the SHA-256 digest of a
buffer in one step, use it instead of sha256_init() + sha256_update() +
sha256_final().
Also simplify the code by inlining calculate_sha256() into its caller
and switching a debug log statement to use %*phN instead of bin2hex().
Acked-by: Tzung-Bi Shih <tzungbi@google.com>
Reviewed-by: Ard Biesheuvel <ardb@kernel.org>
Cc: alsa-devel@alsa-project.org
Cc: Ard Biesheuvel <ardb@kernel.org>
Cc: Cheng-Yi Chiang <cychiang@chromium.org>
Cc: Enric Balletbo i Serra <enric.balletbo@collabora.com>
Cc: Guenter Roeck <groeck@chromium.org>
Cc: Tzung-Bi Shih <tzungbi@google.com>
Signed-off-by: Eric Biggers <ebiggers@google.com>
Signed-off-by: Herbert Xu <herbert@gondor.apana.org.au>
siu is using discriminatory terms for function parameter.
This patch changes it to "secondary"
One note here is that it do nothing to DMA related naming
for now, because it needs framework level modification.
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Link: https://lore.kernel.org/r/87d04z3qqg.wl-kuninori.morimoto.gx@renesas.com
Signed-off-by: Mark Brown <broonie@kernel.org>
rsnd is using discriminatory terms for function names.
This patch changes it to "secondary"
One note here is that it do nothing to DMA related naming
for now, because it needs framework level modification.
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Link: https://lore.kernel.org/r/87h7ub3qra.wl-kuninori.morimoto.gx@renesas.com
Signed-off-by: Mark Brown <broonie@kernel.org>
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
.../bindings/sound/mt8183-da7219-max98357.txt | 4 +++
.../sound/mt8183-mt6358-ts3a227-max98357.txt | 2 ++
sound/soc/mediatek/Kconfig | 2 ++
.../mediatek/mt8183/mt8183-da7219-max98357.c | 29 +++++++++++++++++--
.../mt8183/mt8183-mt6358-ts3a227-max98357.c | 29 +++++++++++++++++--
5 files changed, 60 insertions(+), 6 deletions(-)
--
2.27.0.383.g050319c2ae-goog
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
sound/soc/codecs/ak4458.c | 6 +++---
sound/soc/codecs/rt5631.c | 8 +++++--
sound/soc/codecs/rt5659.c | 37 ---------------------------------
sound/soc/codecs/tlv320aic26.c | 2 +-
sound/soc/pxa/pxa-ssp.c | 2 +-
sound/soc/qcom/qdsp6/q6asm.c | 2 +-
sound/soc/sti/uniperif.h | 2 +-
sound/soc/sunxi/sun4i-i2s.c | 10 ++++++++-
sound/soc/sunxi/sun4i-spdif.c | 2 +-
sound/soc/ux500/ux500_msp_i2s.c | 8 +++----
10 files changed, 27 insertions(+), 52 deletions(-)
base-commit: 6940701c71
--
2.25.1
Clear the validity bit for TX
Add kctl for configuring TX validity bit
Shengjiu Wang (2):
ASoC: fsl_spdif: Clear the validity bit for TX
ASoC: fsl_spdif: Add kctl for configuring TX validity bit
sound/soc/fsl/fsl_spdif.c | 51 ++++++++++++++++++++++++++++++++++++---
1 file changed, 47 insertions(+), 4 deletions(-)
--
2.21.0
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.
Suggested-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Signed-off-by: Hans de Goede <hdegoede@redhat.com>
Link: https://lore.kernel.org/r/20200703103840.333732-1-hdegoede@redhat.com
Signed-off-by: Mark Brown <broonie@kernel.org>
Add one kctl for configuring TX validity bit from user
space.
The type of this kctl is boolean:
on - Outgoing validity always set
off - Outgoing validity always clear
Signed-off-by: Shengjiu Wang <shengjiu.wang@nxp.com>
Link: https://lore.kernel.org/r/1594112066-31297-3-git-send-email-shengjiu.wang@nxp.com
Signed-off-by: Mark Brown <broonie@kernel.org>
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.
Signed-off-by: Shengjiu Wang <shengjiu.wang@nxp.com>
Link: https://lore.kernel.org/r/1594112066-31297-2-git-send-email-shengjiu.wang@nxp.com
Signed-off-by: Mark Brown <broonie@kernel.org>
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>
Looks as though they've never been used.
Fixes the following W=1 kernel build warning(s):
In file included from sound/soc/codecs/rt5659.c:25:
In file included from sound/soc/codecs/rt5659.c:25:
sound/soc/codecs/rt5659.c:1232:2: warning: ‘rt5659_ad_monor_asrc_enum’ defined but not used [-Wunused-const-variable=]
1232 | rt5659_ad_monor_asrc_enum, RT5659_ASRC_3, RT5659_AD_MONO_R_T_SFT, 0x7,
| ^~~~~~~~~~~~~~~~~~~~~~~~~
include/sound/soc.h:359:24: note: in definition of macro ‘SOC_VALUE_ENUM_DOUBLE_DECL’
359 | const struct soc_enum name = SOC_VALUE_ENUM_DOUBLE(xreg, xshift_l, xshift_r, xmask, | ^~~~
sound/soc/codecs/rt5659.c:1231:8: note: in expansion of macro ‘SOC_VALUE_ENUM_SINGLE_DECL’
1231 | static SOC_VALUE_ENUM_SINGLE_DECL(
| ^~~~~~~~~~~~~~~~~~~~~~~~~~
sound/soc/codecs/rt5659.c:1228:2: warning: ‘rt5659_ad_monol_asrc_enum’ defined but not used [-Wunused-const-variable=]
1228 | rt5659_ad_monol_asrc_enum, RT5659_ASRC_3, RT5659_AD_MONO_L_T_SFT, 0x7,
| ^~~~~~~~~~~~~~~~~~~~~~~~~
include/sound/soc.h:359:24: note: in definition of macro ‘SOC_VALUE_ENUM_DOUBLE_DECL’
359 | const struct soc_enum name = SOC_VALUE_ENUM_DOUBLE(xreg, xshift_l, xshift_r, xmask, | ^~~~
sound/soc/codecs/rt5659.c:1227:8: note: in expansion of macro ‘SOC_VALUE_ENUM_SINGLE_DECL’
1227 | static SOC_VALUE_ENUM_SINGLE_DECL(
| ^~~~~~~~~~~~~~~~~~~~~~~~~~
sound/soc/codecs/rt5659.c:1224:2: warning: ‘rt5659_ad_sto2_asrc_enum’ defined but not used [-Wunused-const-variable=]
1224 | rt5659_ad_sto2_asrc_enum, RT5659_ASRC_3, RT5659_AD_STO2_T_SFT, 0x7,
| ^~~~~~~~~~~~~~~~~~~~~~~~
include/sound/soc.h:359:24: note: in definition of macro ‘SOC_VALUE_ENUM_DOUBLE_DECL’
359 | const struct soc_enum name = SOC_VALUE_ENUM_DOUBLE(xreg, xshift_l, xshift_r, xmask, | ^~~~
sound/soc/codecs/rt5659.c:1223:8: note: in expansion of macro ‘SOC_VALUE_ENUM_SINGLE_DECL’
1223 | static SOC_VALUE_ENUM_SINGLE_DECL(
| ^~~~~~~~~~~~~~~~~~~~~~~~~~
sound/soc/codecs/rt5659.c:1220:2: warning: ‘rt5659_ad_sto1_asrc_enum’ defined but not used [-Wunused-const-variable=]
1220 | rt5659_ad_sto1_asrc_enum, RT5659_ASRC_2, RT5659_AD_STO1_T_SFT, 0x7,
| ^~~~~~~~~~~~~~~~~~~~~~~~
include/sound/soc.h:359:24: note: in definition of macro ‘SOC_VALUE_ENUM_DOUBLE_DECL’
359 | const struct soc_enum name = SOC_VALUE_ENUM_DOUBLE(xreg, xshift_l, xshift_r, xmask, | ^~~~
sound/soc/codecs/rt5659.c:1219:8: note: in expansion of macro ‘SOC_VALUE_ENUM_SINGLE_DECL’
1219 | static SOC_VALUE_ENUM_SINGLE_DECL(
| ^~~~~~~~~~~~~~~~~~~~~~~~~~
sound/soc/codecs/rt5659.c:1216:2: warning: ‘rt5659_da_monor_asrc_enum’ defined but not used [-Wunused-const-variable=]
1216 | rt5659_da_monor_asrc_enum, RT5659_ASRC_2, RT5659_DA_MONO_R_T_SFT, 0x7,
| ^~~~~~~~~~~~~~~~~~~~~~~~~
include/sound/soc.h:359:24: note: in definition of macro ‘SOC_VALUE_ENUM_DOUBLE_DECL’
359 | const struct soc_enum name = SOC_VALUE_ENUM_DOUBLE(xreg, xshift_l, xshift_r, xmask, | ^~~~
sound/soc/codecs/rt5659.c:1215:8: note: in expansion of macro ‘SOC_VALUE_ENUM_SINGLE_DECL’
1215 | static SOC_VALUE_ENUM_SINGLE_DECL(
| ^~~~~~~~~~~~~~~~~~~~~~~~~~
sound/soc/codecs/rt5659.c:1212:2: warning: ‘rt5659_da_monol_asrc_enum’ defined but not used [-Wunused-const-variable=]
1212 | rt5659_da_monol_asrc_enum, RT5659_ASRC_2, RT5659_DA_MONO_L_T_SFT, 0x7,
| ^~~~~~~~~~~~~~~~~~~~~~~~~
include/sound/soc.h:359:24: note: in definition of macro ‘SOC_VALUE_ENUM_DOUBLE_DECL’
359 | const struct soc_enum name = SOC_VALUE_ENUM_DOUBLE(xreg, xshift_l, xshift_r, xmask, | ^~~~
sound/soc/codecs/rt5659.c:1211:8: note: in expansion of macro ‘SOC_VALUE_ENUM_SINGLE_DECL’
1211 | static SOC_VALUE_ENUM_SINGLE_DECL(
| ^~~~~~~~~~~~~~~~~~~~~~~~~~
sound/soc/codecs/rt5659.c:1208:2: warning: ‘rt5659_da_sto_asrc_enum’ defined but not used [-Wunused-const-variable=]
1208 | rt5659_da_sto_asrc_enum, RT5659_ASRC_2, RT5659_DA_STO_T_SFT, 0x7,
| ^~~~~~~~~~~~~~~~~~~~~~~
include/sound/soc.h:359:24: note: in definition of macro ‘SOC_VALUE_ENUM_DOUBLE_DECL’
359 | const struct soc_enum name = SOC_VALUE_ENUM_DOUBLE(xreg, xshift_l, xshift_r, xmask, | ^~~~
sound/soc/codecs/rt5659.c:1207:8: note: in expansion of macro ‘SOC_VALUE_ENUM_SINGLE_DECL’
1207 | static SOC_VALUE_ENUM_SINGLE_DECL(
| ^~~~~~~~~~~~~~~~~~~~~~~~~~
Signed-off-by: Lee Jones <lee.jones@linaro.org>
Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Cc: Oder Chiou <oder_chiou@realtek.com>
Link: https://lore.kernel.org/r/20200709162328.259586-9-pierre-louis.bossart@linux.intel.com
Signed-off-by: Mark Brown <broonie@kernel.org>
Fixes the following W=1 kernel build warning(s):
sound/soc/codecs/rt5631.c:72: warning: Function parameter or member
'component' not described in 'rt5631_write_index'
sound/soc/codecs/rt5631.c:72: warning: Function parameter or member
'reg' not described in 'rt5631_write_index'
sound/soc/codecs/rt5631.c:72: warning: Function parameter or member
'value' not described in 'rt5631_write_index'
sound/soc/codecs/rt5631.c:82: warning: Function parameter or member
'component' not described in 'rt5631_read_index'
sound/soc/codecs/rt5631.c:82: warning: Function parameter or member
'reg' not described in 'rt5631_read_index'
sound/soc/codecs/rt5631.c:367: warning: Function parameter or member
'component' not described in 'onebit_depop_power_stage'
sound/soc/codecs/rt5631.c:405: warning: Function parameter or member
'component' not described in 'onebit_depop_mute_stage'
sound/soc/codecs/rt5631.c:443: warning: Function parameter or member
'component' not described in 'depop_seq_power_stage'
sound/soc/codecs/rt5631.c:515: warning: Function parameter or member
'component' not described in 'depop_seq_mute_stage'
Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Link: https://lore.kernel.org/r/20200709162328.259586-8-pierre-louis.bossart@linux.intel.com
Signed-off-by: Mark Brown <broonie@kernel.org>
Looks like these have been unchecked since the driver's inception in 2012.
Fixes the following W=1 kernel build warning(s):
sound/soc/ux500/ux500_msp_i2s.c: In function ‘flush_fifo_rx’:
sound/soc/ux500/ux500_msp_i2s.c:398:6: warning: variable ‘reg_val_DR’
set but not used [-Wunused-but-set-variable]
sound/soc/ux500/ux500_msp_i2s.c: In function ‘flush_fifo_tx’:
sound/soc/ux500/ux500_msp_i2s.c:415:6: warning: variable
‘reg_val_TSTDR’ 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: zhong jiang <zhongjiang@huawei.com>
Cc: Ola Lilja <ola.o.lilja@stericsson.com>
Cc: Roger Nilsson <roger.xr.nilsson@stericsson.com>
Cc: Sandeep Kaushik <sandeep.kaushik@st.com>
Link: https://lore.kernel.org/r/20200709162328.259586-7-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 full
descriptions are not provided.
Fixes the following W=1 kernel build warning(s):
sound/soc/pxa/pxa-ssp.c:186: warning: Function parameter or member
'ssp' not described in 'pxa_ssp_set_scr'
Signed-off-by: Lee Jones <lee.jones@linaro.org>
Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Cc: Daniel Mack <daniel@zonque.org>
Cc: Haojian Zhuang <haojian.zhuang@gmail.com>
Cc: Robert Jarzmik <robert.jarzmik@free.fr>
Link: https://lore.kernel.org/r/20200709162328.259586-6-pierre-louis.bossart@linux.intel.com
Signed-off-by: Mark Brown <broonie@kernel.org>
Property name descriptions need to match exactly.
Fixes the following W=1 kernel build warning(s):
sound/soc/sunxi/sun4i-spdif.c:178: warning: Function parameter or
member 'reg_dac_txdata' not described in 'sun4i_spdif_quirks'
Signed-off-by: Lee Jones <lee.jones@linaro.org>
Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Cc: Maxime Ripard <mripard@kernel.org>
Cc: Chen-Yu Tsai <wens@csie.org>
Cc: Philipp Zabel <p.zabel@pengutronix.de>
Cc: Andrea Venturi <be17068@iperbole.bo.it>
Cc: Marcus Cooper <codekipper@gmail.com>
Link: https://lore.kernel.org/r/20200709162328.259586-5-pierre-louis.bossart@linux.intel.com
Signed-off-by: Mark Brown <broonie@kernel.org>
Fix W=1 warnings - missing fields in description
sound/soc/sunxi/sun4i-i2s.c:160: warning: Function parameter or
member 'bclk_dividers' not described in 'sun4i_i2s_quirks'
sound/soc/sunxi/sun4i-i2s.c:160: warning: Function parameter or member
'num_bclk_dividers' not described in 'sun4i_i2s_quirks'
sound/soc/sunxi/sun4i-i2s.c:160: warning: Function parameter or member
'mclk_dividers' not described in 'sun4i_i2s_quirks'
sound/soc/sunxi/sun4i-i2s.c:160: warning: Function parameter or member
'num_mclk_dividers' not described in 'sun4i_i2s_quirks'
sound/soc/sunxi/sun4i-i2s.c:160: warning: Function parameter or member
'get_bclk_parent_rate' not described in 'sun4i_i2s_quirks'
sound/soc/sunxi/sun4i-i2s.c:160: warning: Function parameter or member
'get_sr' not described in 'sun4i_i2s_quirks'
sound/soc/sunxi/sun4i-i2s.c:160: warning: Function parameter or member
'get_wss' not described in 'sun4i_i2s_quirks'
sound/soc/sunxi/sun4i-i2s.c:160: warning: Function parameter or member
'set_chan_cfg' not described in 'sun4i_i2s_quirks'
sound/soc/sunxi/sun4i-i2s.c:160: warning: Function parameter or member
'set_fmt' not described in 'sun4i_i2s_quirks'
Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Cc: Maxime Ripard <mripard@kernel.org>
Cc: Chen-Yu Tsai <wens@csie.org>
Cc: Philipp Zabel <p.zabel@pengutronix.de>
Cc: Andrea Venturi <be17068@iperbole.bo.it>
Link: https://lore.kernel.org/r/20200709162328.259586-4-pierre-louis.bossart@linux.intel.com
Signed-off-by: Mark Brown <broonie@kernel.org>
Fixes the following W=1 kernel build warning(s):
sound/soc/qcom/qdsp6/q6asm.c:924: warning: Function parameter or
member 'codec_profile' not described in 'q6asm_open_write'
Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Cc: Patrick Lai <plai@codeaurora.org>
Cc: Banajit Goswami <bgoswami@codeaurora.org>
Cc: Srinivas Kandagatla <srinivas.kandagatla@linaro.org>
Cc: Vinod Koul <vkoul@kernel.org>
Link: https://lore.kernel.org/r/20200709162328.259586-3-pierre-louis.bossart@linux.intel.com
Signed-off-by: Mark Brown <broonie@kernel.org>
Fix W=1 warning. The table uni_tdm_hw is declared in a header included
by multiple C file. This isn't really a good practice but for now
using __maybe_unused makes the following warning go away.
sound/soc/sti/sti_uniperif.c:12:
sound/soc/sti/uniperif.h:1351:38: warning: ‘uni_tdm_hw’ defined but
not used [-Wunused-const-variable=]
1351 | static const struct snd_pcm_hardware uni_tdm_hw = {
| ^~~~~~~~~~
Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Acked-by: Arnaud Pouliquen <arnaud.pouliquen@st.com>
Link: https://lore.kernel.org/r/20200709162328.259586-2-pierre-louis.bossart@linux.intel.com
Signed-off-by: Mark Brown <broonie@kernel.org>
This series tries to reuse mt8183-da7219-max98357.c for supporting machine
driver with rt1015 speaker amplifier.
The first 3 patches refactor the code for easier to change for subsequent
patches.
The 4th patch adds document for the new proposed compatible string.
The 5th patch changes the machine driver to support either "MAX98357A" or
"RT1015" codecs.
Tzung-Bi Shih (5):
ASoC: mediatek: mt8183-da7219: sort header inclusions in alphabetical
ASoC: mediatek: mt8183-da7219: remove forward declaration of
headset_init
ASoC: mediatek: mt8183-da7219: extract codec and DAI names
ASoC: mediatek: mt8183-da7219: add compatible string for using rt1015
ASoC: mediatek: mt8183-da7219: support machine driver with rt1015
.../bindings/sound/mt8183-da7219-max98357.txt | 5 +-
sound/soc/mediatek/Kconfig | 5 +-
.../mediatek/mt8183/mt8183-da7219-max98357.c | 244 ++++++++++++++----
3 files changed, 197 insertions(+), 57 deletions(-)
--
2.27.0.383.g050319c2ae-goog
Add the TX offset slot programming. There is no RX offset slot
register.
Since there is no RX offset the check for slot symmetry can be removed.
Signed-off-by: Dan Murphy <dmurphy@ti.com>
Link: https://lore.kernel.org/r/20200709185129.10505-2-dmurphy@ti.com
Signed-off-by: Mark Brown <broonie@kernel.org>
The CPU and the codec both are represented now as components, so for
PDMIC we are registering two componenets with the same name. Since
there is no actual codec, we will merge the codec component into the
CPU one and use a dummy codec instead, for the DAI link.
As a bonus, debugfs will no longer report an error when will try to
create entries for both componenets with the same name.
Signed-off-by: Codrin Ciubotariu <codrin.ciubotariu@microchip.com>
Link: https://lore.kernel.org/r/20200708163359.2698696-1-codrin.ciubotariu@microchip.com
Signed-off-by: Mark Brown <broonie@kernel.org>
The CPU and the codec both are represented now as components, so for
CLASS-D we are registering two componenets with the same name. Since
there is no actual codec, we will merge the codec component into the
CPU one and use a dummy codec instead, for the DAI link.
As a bonus, debugfs will no longer report an error when will try to
create entries for both componenets with the same name.
Signed-off-by: Codrin Ciubotariu <codrin.ciubotariu@microchip.com>
Link: https://lore.kernel.org/r/20200708101249.2626560-1-codrin.ciubotariu@microchip.com
Signed-off-by: Mark Brown <broonie@kernel.org>
Fix W=1 warning and removed unused table. In this case this a
duplicate of
static const struct of_device_id max98390_of_match[] = {
{ .compatible = "maxim,max98390", },
{}
};
MODULE_DEVICE_TABLE(of, max98390_of_match);
already used in the rest of the code.
Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Link: https://lore.kernel.org/r/20200707190612.97799-13-pierre-louis.bossart@linux.intel.com
Signed-off-by: Mark Brown <broonie@kernel.org>
Looks like 'w' has remained unchecked since the driver's inception.
Fixes the following W=1 kernel build warning(s):
sound/soc/ti/omap-mcbsp-st.c: In function ‘omap_mcbsp_st_chgain’:
sound/soc/ti/omap-mcbsp-st.c:145:6: warning: variable ‘w’ set but not used [-Wunused-but-set-variable]
Peter suggested that the whole read can be removed, so that's
been done too.
Signed-off-by: Lee Jones <lee.jones@linaro.org>
Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Acked-by: Peter Ujfalusi <peter.ujfalusi@ti.com>
Cc: Peter Ujfalusi <peter.ujfalusi@ti.com>
Cc: Jarkko Nikula <jarkko.nikula@bitmer.com>
Cc: Samuel Ortiz <samuel.ortiz@nokia.com>
Cc: linux-omap@vger.kernel.org
Link: https://lore.kernel.org/r/20200707190612.97799-10-pierre-louis.bossart@linux.intel.com
Signed-off-by: Mark Brown <broonie@kernel.org>
Fixes the following W=1 kernel build warning(s):
In file included from include/sound/tlv.h:10,
from sound/soc/codecs/jz4770.c:19:
sound/soc/codecs/jz4770.c:306:35: warning: ‘mic_boost_tlv’ defined but not used [-Wunused-const-variable=]
306 | static const DECLARE_TLV_DB_SCALE(mic_boost_tlv, 0, 400, 0);
| ^~~~~~~~~~~~~
include/uapi/sound/tlv.h:64:15: note: in definition of macro ‘SNDRV_CTL_TLVD_DECLARE_DB_SCALE’
64 | unsigned int name[] = { | ^~~~
sound/soc/codecs/jz4770.c:306:14: note: in expansion of macro ‘DECLARE_TLV_DB_SCALE’
306 | static const DECLARE_TLV_DB_SCALE(mic_boost_tlv, 0, 400, 0);
| ^~~~~~~~~~~~~~~~~~~~
Signed-off-by: Lee Jones <lee.jones@linaro.org>
Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Reviewed-by: Paul Cercueil <paul@crapouillou.net>
Cc: ter Huurne <maarten@treewalker.org>
Link: https://lore.kernel.org/r/20200707190612.97799-9-pierre-louis.bossart@linux.intel.com
Signed-off-by: Mark Brown <broonie@kernel.org>
Fix W=1 warning, the kernel-doc syntax was probably from Doxygen?
Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Acked-by: Adam Thomson <Adam.Thomson.Opensource@diasemi.com>
Link: https://lore.kernel.org/r/20200707190612.97799-7-pierre-louis.bossart@linux.intel.com
Signed-off-by: Mark Brown <broonie@kernel.org>
Fix W=1 warnings - missing fields in structure
Credits to Sylwester Nawrocki for the pclk and cclk descriptions.
Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Reviewed-by: Sylwester Nawrocki <s.nawrocki@samsung.com>
Link: https://lore.kernel.org/r/20200707190612.97799-4-pierre-louis.bossart@linux.intel.com
Signed-off-by: Mark Brown <broonie@kernel.org>
Fix W=1 warning
Kernel-doc is not used in one file and missing argument in the second.
Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Acked-by: Alexandre Belloni <alexandre.belloni@bootlin.com>
Link: https://lore.kernel.org/r/20200707190612.97799-2-pierre-louis.bossart@linux.intel.com
Signed-off-by: Mark Brown <broonie@kernel.org>
Disable Left and Right Spk pin after boot so that sof can get
suspended.
This follows the same logic added to another machine driver with
commit 94d2d08974 ("ASoC: Intel: Boards: tgl_max98373: add dai_trigger function")
Signed-off-by: randerwang <rander.wang@linux.intel.com>
Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Reviewed-by: Bard Liao <yung-chuan.liao@linux.intel.com>
Link: https://lore.kernel.org/r/20200708203215.231776-5-pierre-louis.bossart@linux.intel.com
Signed-off-by: Mark Brown <broonie@kernel.org>
Reflect Kconfig changes and add both SoundWire and I2C modes
Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Reviewed-by: Rander Wang <rander.wang@linux.intel.com>
Reviewed-by: Kai Vehmanen <kai.vehmanen@linux.intel.com>
Link: https://lore.kernel.org/r/20200708203215.231776-4-pierre-louis.bossart@linux.intel.com
Signed-off-by: Mark Brown <broonie@kernel.org>
Add SoundWire specific parts and extend common ones already split from
I2C.
Signed-off-by: Ryan Lee <ryans.lee@maximintegrated.com>
Signed-off-by: Naveen Manohar <naveen.m@intel.com>
Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Reviewed-by: Rander Wang <rander.wang@linux.intel.com>
Reviewed-by: Kai Vehmanen <kai.vehmanen@linux.intel.com>
Link: https://lore.kernel.org/r/20200708203215.231776-3-pierre-louis.bossart@linux.intel.com
Signed-off-by: Mark Brown <broonie@kernel.org>
To prepare support for SoundWire, let's first split the I2C and common
parts. No new functionality, just indents and formatting to make
checkpatch happy.
Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Reviewed-by: Rander Wang <rander.wang@linux.intel.com>
Reviewed-by: Kai Vehmanen <kai.vehmanen@linux.intel.com>
Link: https://lore.kernel.org/r/20200708203215.231776-2-pierre-louis.bossart@linux.intel.com
Signed-off-by: Mark Brown <broonie@kernel.org>
Supports machine driver with rt1015 ("mt8183-da7219-rt1015"). Embeds in
existing mt8183-da7219-max98357.c because they share most of the code.
Signed-off-by: Tzung-Bi Shih <tzungbi@google.com>
Link: https://lore.kernel.org/r/20200709122445.1584497-6-tzungbi@google.com
Signed-off-by: Mark Brown <broonie@kernel.org>
The HV/VREF should not turn off if the headphone jack plug-in.
This patch could solve the unexpected interrupt issue in some devices.
Signed-off-by: Shuming Fan <shumingf@realtek.com>
Tested-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Link: https://lore.kernel.org/r/20200709101345.11449-1-shumingf@realtek.com
Signed-off-by: Mark Brown <broonie@kernel.org>
In DSP_A mode, BIT7 of IFACE should bit 0 according to datasheet (ie.
inverted frame clock is not support in this mode).
Signed-off-by: Puyou Lu <puyou.lu@gmail.com>
Acked-by: Charles Keepax <ckeepax@opensource.cirrus.com>
Link: https://lore.kernel.org/r/1593657056-4989-1-git-send-email-puyou.lu@gmail.com
Signed-off-by: Mark Brown <broonie@kernel.org>
A collection of small, mostly device-specific fixes.
The significant one is the regression fix for USB-audio implicit
feedback devices due to the incorrect frame size calculation, which
landed in 5.8 and stable trees. In addition, a few usual HD-audio
and USB-audio quirks, Intel HDMI fixes, ASoC fsl and rt5682 fixes,
as well as the fix in compress-offload partial drain operation.
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Merge tag 'sound-5.8-rc5' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound
Pull sound fixes from Takashi Iwai:
"A collection of small, mostly device-specific fixes.
The significant one is the regression fix for USB-audio implicit
feedback devices due to the incorrect frame size calculation, which
landed in 5.8 and stable trees.
In addition, a few usual HD-audio and USB-audio quirks, Intel HDMI
fixes, ASoC fsl and rt5682 fixes, as well as the fix in
compress-offload partial drain operation"
* tag 'sound-5.8-rc5' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound:
ALSA: compress: fix partial_drain completion state
ALSA: usb-audio: Add implicit feedback quirk for RTX6001
ALSA: usb-audio: add quirk for MacroSilicon MS2109
ALSA: hda/realtek: Enable headset mic of Acer Veriton N4660G with ALC269VC
ALSA: hda/realtek: Enable headset mic of Acer C20-820 with ALC269VC
ALSA: hda/realtek - Enable audio jacks of Acer vCopperbox with ALC269VC
ALSA: hda/realtek - Fix Lenovo Thinkpad X1 Carbon 7th quirk subdevice id
ALSA: hda/hdmi: improve debug traces for stream lookups
ALSA: hda/hdmi: fix failures at PCM open on Intel ICL and later
ALSA: opl3: fix infoleak in opl3
ALSA: usb-audio: Replace s/frame/packet/ where appropriate
ALSA: usb-audio: Fix packet size calculation
AsoC: amd: add missing snd- module prefix to the acp3x-rn driver kernel module
ALSA: hda - let hs_mic be picked ahead of hp_mic
ASoC: rt5682: fix the pop noise while OMTP type headset plugin
ASoC: fsl_mqs: Fix unchecked return value for clk_prepare_enable
ASoC: fsl_mqs: Don't check clock is NULL before calling clk API
This series tries to reuse mt8183-mt6358-ts3a227-max98357.c for supporting
machine driver with rt1015 speaker amplifier.
The 1st patch is straightforward: re-order the header inclusions.
The 2nd patch adds document for the new proposed compatible string.
The 3rd patch changes the machine driver to support either "MAX98357A" or
"RT1015" codecs.
Tzung-Bi Shih (3):
ASoC: mediatek: mt8183: sort header inclusions in alphabetical
dt-bindings: mt8183: add compatible string for using rt1015
ASoC: mediatek: mt8183: support machine driver with rt1015
.../sound/mt8183-mt6358-ts3a227-max98357.txt | 5 +-
sound/soc/mediatek/Kconfig | 5 +-
.../mt8183/mt8183-mt6358-ts3a227-max98357.c | 171 +++++++++++++++---
3 files changed, 153 insertions(+), 28 deletions(-)
--
2.27.0.383.g050319c2ae-goog
While experimenting and introducing errors in Baytrail topology files
until I got them right, I encountered multiple kernel oopses and
memory leaks. This is a first batch to harden the code, but we should
probably think of a tool to fuzz the topology...
Pierre-Louis Bossart (5):
ASoC: topology: fix kernel oops on route addition error
ASoC: topology: fix tlvs in error handling for widget_dmixer
ASoC: topology: use break on errors, not continue
ASoC: topology: factor kfree(se) in error handling
ASoC: topology: add more logs when topology load fails.
sound/soc/soc-topology.c | 97 ++++++++++++++++++++++++----------------
1 file changed, 58 insertions(+), 39 deletions(-)
base-commit: a5911ac579
--
2.25.1
This patchset adds gapless compressed audio support on q6asm.
Gapless on q6asm is implemented using 2 streams in a single asm session.
First few patches are enhacements done to q6asm interface to allow
stream id per each command, gapless flags and silence meta data.
Along with this there are few trivial changes which I thought are necessary!
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.
Thanks,
srini
Srinivas Kandagatla (11):
ASoC: q6asm: add command opcode to timeout error report
ASoC: q6asm: rename misleading session id variable
ASoC: q6asm: make commands specific to streams
ASoC: q6asm: use flags directly from asm-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 asm open
ASoC: q6asm-dai: add next track metadata support
ASoC: qdsp6: use dev_err instead of pr_err
ASoC: qdsp6-dai: add gapless support
ASoC: q6asm-dai: add support to copy callback
sound/soc/qcom/qdsp6/q6asm-dai.c | 397 +++++++++++++++++++++++--------
sound/soc/qcom/qdsp6/q6asm.c | 173 +++++++++-----
sound/soc/qcom/qdsp6/q6asm.h | 48 ++--
3 files changed, 458 insertions(+), 160 deletions(-)
--
2.21.0
Supports machine driver with rt1015 ("mt8183-mt6358-ts3a227-rt1015").
Embeds in existing mt8183-mt6358-ts3a227-max98357.c because they share
most of the code.
Signed-off-by: Tzung-Bi Shih <tzungbi@google.com>
Link: https://lore.kernel.org/r/20200708113233.3994206-4-tzungbi@google.com
Signed-off-by: Mark Brown <broonie@kernel.org>
Make the error reporting more useful by adding opcode to it.
Without this its almost impossible to say which command actually
timed out.
Signed-off-by: Srinivas Kandagatla <srinivas.kandagatla@linaro.org>
Link: https://lore.kernel.org/r/20200707163641.17113-2-srinivas.kandagatla@linaro.org
Signed-off-by: Mark Brown <broonie@kernel.org>
This is hopefully the last set of fixes to avoid probe errors due to
stricter checks of DAI capabilities introduced late in the 5.8 cycle.
Daniel Baluta (1):
ASoC: SOF: imx: add min/max channels for SAI/ESAI on i.MX8/i.MX8M
Pierre-Louis Bossart (2):
ASoC: soc-dai: set dai_link dpcm_ flags with a helper
ASoC: Intel: bdw-rt5677: fix non BE conversion
include/sound/soc-dai.h | 1 +
sound/soc/generic/audio-graph-card.c | 4 +--
sound/soc/generic/simple-card.c | 4 +--
sound/soc/intel/boards/bdw-rt5677.c | 1 +
sound/soc/soc-dai.c | 38 ++++++++++++++++++++++++++++
sound/soc/sof/imx/imx8.c | 8 ++++++
sound/soc/sof/imx/imx8m.c | 8 ++++++
7 files changed, 60 insertions(+), 4 deletions(-)
base-commit: a5911ac579
--
2.25.1
While experimenting and introducing errors in Baytrail topology files
until I got them right, I encountered multiple kernel oopses and
memory leaks. This is a first batch to harden the code, but we should
probably think of a tool to fuzz the topology...
Pierre-Louis Bossart (5):
ASoC: topology: fix kernel oops on route addition error
ASoC: topology: fix tlvs in error handling for widget_dmixer
ASoC: topology: use break on errors, not continue
ASoC: topology: factor kfree(se) in error handling
ASoC: topology: add more logs when topology load fails.
sound/soc/soc-topology.c | 97 ++++++++++++++++++++++++----------------
1 file changed, 58 insertions(+), 39 deletions(-)
base-commit: a5911ac579
--
2.25.1
The DSP should be notified for device removal only if the
probe was successful. Fixes the following KASAN bug:
BUG: KASAN: null-ptr-deref in sof_ipc_tx_message+0x80/0x160 [snd_sof]
Signed-off-by: Ranjani Sridharan <ranjani.sridharan@linux.intel.com>
Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Reviewed-by: Kai Vehmanen <kai.vehmanen@linux.intel.com>
Link: https://lore.kernel.org/r/20200707204027.114169-1-pierre-louis.bossart@linux.intel.com
Signed-off-by: Mark Brown <broonie@kernel.org>
Idle_bias_on is used to decide bias on/off in standby state by dapm.
When Idle_bias_on is set to one, dapm will keep max98373 active at
idle time. Max98373 is doing nothing in this state, so remove
idle_bias_on setting to let max98373 get suspended when it is idle.
Signed-off-by: randerwang <rander.wang@linux.intel.com>
Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Reviewed-by: Ryan Lee <ryans.lee@maximintegrated.com>
Reviewed-by: Kai Vehmanen <kai.vehmanen@linux.intel.com>
Link: https://lore.kernel.org/r/20200707205740.114927-4-pierre-louis.bossart@linux.intel.com
Signed-off-by: Mark Brown <broonie@kernel.org>
Volume control in probe function is not necessary.
Signed-off-by: Ryan Lee <ryans.lee@maximintegrated.com>
Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Reviewed-by: Rander Wang <rander.wang@linux.intel.com>
Reviewed-by: Kai Vehmanen <kai.vehmanen@linux.intel.com>
Link: https://lore.kernel.org/r/20200707205740.114927-3-pierre-louis.bossart@linux.intel.com
Signed-off-by: Mark Brown <broonie@kernel.org>
Add more dev_err() logs to help trace topology load failures, since we
have multiple error causes (e.g. invalid header or header that could
not be loaded).
Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Reviewed-by: Ranjani Sridharan <ranjani.sridharan@linux.intel.com>
Reviewed-by: Kai Vehmanen <kai.vehmanen@linux.intel.com>
Reviewed-by: Rander Wang <rander.wang@linux.intel.com>
Link: https://lore.kernel.org/r/20200707203749.113883-6-pierre-louis.bossart@linux.intel.com
Signed-off-by: Mark Brown <broonie@kernel.org>
No need to repeat the same thing multiple times when it can be done in
one location.
Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Reviewed-by: Ranjani Sridharan <ranjani.sridharan@linux.intel.com>
Reviewed-by: Kai Vehmanen <kai.vehmanen@linux.intel.com>
Link: https://lore.kernel.org/r/20200707203749.113883-5-pierre-louis.bossart@linux.intel.com
Signed-off-by: Mark Brown <broonie@kernel.org>
Since the beginning of the topology, the code continues to the next
object even when an error is detected.
The topology should be handled with an all-or-nothing design, loading
a partially valid topology is a sure way to get bug reports that are
difficult to deal with.
Changing the behavior may break previous solutions and expose problems
in topology files delivered in the past, so it's probably not wise to
add this patch to stable branches without revalidation.
Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Reviewed-by: Ranjani Sridharan <ranjani.sridharan@linux.intel.com>
Reviewed-by: Kai Vehmanen <kai.vehmanen@linux.intel.com>
Link: https://lore.kernel.org/r/20200707203749.113883-4-pierre-louis.bossart@linux.intel.com
Signed-off-by: Mark Brown <broonie@kernel.org>
we need to free all allocated tlvs, not just the one allocated in
the loop before releasing kcontrols - other the tlvs references will
leak.
Fixes: 9f90af3a99 ('ASoC: topology: Consolidate and fix asoc_tplg_dapm_widget_*_create flow')
Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Reviewed-by: Ranjani Sridharan <ranjani.sridharan@linux.intel.com>
Reviewed-by: Kai Vehmanen <kai.vehmanen@linux.intel.com>
Link: https://lore.kernel.org/r/20200707203749.113883-3-pierre-louis.bossart@linux.intel.com
Signed-off-by: Mark Brown <broonie@kernel.org>
When errors happens while loading graph components, the kernel oopses
while trying to remove all topology components. This can be
root-caused to a list pointing to memory that was already freed on
error.
remove_route() is already called on errors and will perform the
required cleanups so there's no need to free the route memory in
soc_tplg_dapm_graph_elems_load() if the route was added to the
list. We do however want to free the routes allocated but not added to
the list.
Fixes: 7df04ea7a3 ('ASoC: topology: modify dapm route loading routine and add dapm route unloading')
Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Reviewed-by: Ranjani Sridharan <ranjani.sridharan@linux.intel.com>
Reviewed-by: Kai Vehmanen <kai.vehmanen@linux.intel.com>
Link: https://lore.kernel.org/r/20200707203749.113883-2-pierre-louis.bossart@linux.intel.com
Signed-off-by: Mark Brown <broonie@kernel.org>
This is identical with change for Intel platforms done with
commit 8c05246c0b ("ASoC: SOF: Intel: add min/max channels for SSP on Baytrail/Broadwell")
and fixes a regression on i.MX8/i.MX8M:
[ 25.705750] esai-Codec: ASoC: no backend playback stream
[ 27.923378] esai-Codec: ASoC: no users playback at close - state
This is root-caused to the introduction of the DAI capability checks
with snd_soc_dai_stream_valid(). Its use in soc-pcm.c makes it a
requirement for all DAIs to report at least a non-zero min_channels
field.
Fixes: 9b5db05936 ("ASoC: soc-pcm: dpcm: Only allow playback/capture if supported")
Signed-off-by: Daniel Baluta <daniel.baluta@nxp.com>
Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Reviewed-by: Ranjani Sridharan <ranjani.sridharan@linux.intel.com>
Link: https://lore.kernel.org/r/20200707210439.115300-4-pierre-louis.bossart@linux.intel.com
Signed-off-by: Mark Brown <broonie@kernel.org>
When SOF is used, the normal links are converted into DPCM ones. This
generates an error
[ 58.276668] bdw-rt5677 bdw-rt5677: CPU DAI spi-RT5677AA:00 for rtd
Wake on Voice does not support playback
[ 58.276676] bdw-rt5677 bdw-rt5677: ASoC: can't create pcm Wake on
Voice :-22
Fix by forcing the capture direction.
Fixes: b73287f0b0 ('ASoC: soc-pcm: dpcm: fix playback/capture checks')
Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Reviewed-by: Guennadi Liakhovetski <guennadi.liakhovetski@linux.intel.com>
Reviewed-by: Curtis Malainey <curtis@malainey.com>
Link: https://lore.kernel.org/r/20200707210439.115300-3-pierre-louis.bossart@linux.intel.com
Signed-off-by: Mark Brown <broonie@kernel.org>
Add a helper to walk through all the DAIs and set dpcm_playback and
dpcm_capture flags based on the DAIs capabilities, and use this helper
to avoid setting these flags arbitrarily in generic cards.
The commit referenced in the Fixes tag did not introduce the
configuration issue but will prevent the card from probing when
detecting invalid configurations.
Fixes: b73287f0b0 ('ASoC: soc-pcm: dpcm: fix playback/capture checks')
Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Reviewed-by: Kai Vehmanen <kai.vehmanen@linux.intel.com>
Reviewed-by: Guennadi Liakhovetski <guennadi.liakhovetski@linux.intel.com>
Link: https://lore.kernel.org/r/20200707210439.115300-2-pierre-louis.bossart@linux.intel.com
Signed-off-by: Mark Brown <broonie@kernel.org>
This adds default clock/PLL configuration to the driver
for usage with generic drivers like simple-card for usage
with a fixed rate clock.
Signed-off-by: Sebastian Reichel <sebastian.reichel@collabora.com>
Reviewed-by: Adam Thomson <Adam.Thomson.Opensource@diasemi.com>
Link: https://lore.kernel.org/r/20200626164623.87894-1-sebastian.reichel@collabora.com
Signed-off-by: Mark Brown <broonie@kernel.org>
The ASoC devm_ functions that register a component
(devm_snd_soc_register_component and devm_snd_dmaengine_pcm_register) will
clean their component by running snd_soc_unregister_component.
snd_soc_unregister_component will then remove all the components for the
device that was used to register the component in the first place.
However, some drivers register several components (such as a DAI and a
dmaengine PCM) on the same device, and if the dmaengine PCM is registered
first, then the DAI will be cleaned up first and
snd_dmaengine_pcm_unregister will be called next.
snd_dmaengine_pcm_unregister will then lookup the dmaengine PCM component
on the device, and if there's one unregister that component and release its
dmaengine channels. That doesn't happen in practice though since the first
call to snd_soc_unregister_component removed all the components, so we
never get the chance to release the dmaengine channels.
In order to fix this, instead of removing all the components for a given
device, we can simply remove the component that was registered in the first
place. We should have the same number of component registration than we
have components, so it should work just fine.
Signed-off-by: Maxime Ripard <maxime@cerno.tech>
Link: https://lore.kernel.org/r/20200707074237.287171-1-maxime@cerno.tech
Signed-off-by: Mark Brown <broonie@kernel.org>
Hi All,
Here is in essence a resend of my 2 cleanup patches for the rt5670 ASoC
codec code, rebased on top of broonie/sound/for-5.9 with
broonie/sound/for-5.8 merged in.
Regards,
Hans
p.s.
I'll also send out the patch improving the comment in
cht_bsw_rt5672.c cht_codec_fixup() which Pierre-Louis requested soon.
Fix W=1 warnings. The kernel-doc support is partial, add more
descriptions and follow proper syntax
Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Acked-by: Nicolin Chen <nicoleotsuka@gmail.com>
Link: https://lore.kernel.org/r/20200702192141.168018-3-pierre-louis.bossart@linux.intel.com
Signed-off-by: Mark Brown <broonie@kernel.org>
Add logic to check DMIC hardware exists or not on
the platform at runtime.
Add module param for overriding DMIC hardware check
at runtime.
Signed-off-by: Vijendar Mukunda <Vijendar.Mukunda@amd.com>
Link: https://lore.kernel.org/r/1593710826-1106-1-git-send-email-Vijendar.Mukunda@amd.com
Signed-off-by: Mark Brown <broonie@kernel.org>
Add the missing unlock before return from function j721e_audio_hw_params()
in the error handling case.
Fixes: 6748d05590 ("ASoC: ti: Add custom machine driver for j721e EVM (CPB and IVI)")
Reported-by: Hulk Robot <hulkci@huawei.com>
Signed-off-by: Wei Yongjun <weiyongjun1@huawei.com>
Acked-by: Peter Ujfalusi <peter.ujfalusi@ti.com>
Link: https://lore.kernel.org/r/20200703030910.75047-1-weiyongjun1@huawei.com
Signed-off-by: Mark Brown <broonie@kernel.org>
Fix W=1 warnings. The kernel-doc support is partial, add more
descriptions and follow proper syntax
Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Acked-by: Nicolin Chen <nicoleotsuka@gmail.com>
Link: https://lore.kernel.org/r/20200702192141.168018-3-pierre-louis.bossart@linux.intel.com
Signed-off-by: Mark Brown <broonie@kernel.org>
Rename the not really descriptive dev_gpio quirk / setting to
gpio1_is_irq, which describes what it actually does.
Signed-off-by: Hans de Goede <hdegoede@redhat.com>
Reviewed-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Link: https://lore.kernel.org/r/20200703100823.258033-3-hdegoede@redhat.com
Signed-off-by: Mark Brown <broonie@kernel.org>
platform_data is an obsolete concept, instead device_properties,
set through e.g. device-tree, should be used.
struct rt5670_platform_data is only used internally by the rt5670 codec
driver, so lets remove it before someone starts relying on it.
Signed-off-by: Hans de Goede <hdegoede@redhat.com>
Reviewed-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Link: https://lore.kernel.org/r/20200703100823.258033-2-hdegoede@redhat.com
Signed-off-by: Mark Brown <broonie@kernel.org>
Currently if the ctx->spkamp is not recognized an error message is
reported but the code continues to set up the device with uninitialized
variables such as the number of widgets. Fix this by returning -EINVAL
for unrecognized speaker amplifier types.
Fixes: e1435a1feb ("ASoC: Intel: bxt-da7219-max98357a: support MAX98390 speaker amp")
Signed-off-by: Colin Ian King <colin.king@canonical.com>
Acked-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Addresses-Coverity: ("Uninitialized scalar variable")
Link: https://lore.kernel.org/r/20200702114835.37889-1-colin.king@canonical.com
Signed-off-by: Mark Brown <broonie@kernel.org>
Fix W=1 warnings
cs4270.c:508: warning: Function parameter or member 'component' not
described in 'cs4270_probe'
cs4270.c:508: warning: Excess function parameter 'pdev' description in
'cs4270_probe'
cs4270.c:548: warning: Function parameter or member 'component' not
described in 'cs4270_remove'
cs4270.c:548: warning: Excess function parameter 'pdev' description in
'cs4270_remove'
Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Link: https://lore.kernel.org/r/20200701181320.80848-2-pierre-louis.bossart@linux.intel.com
Signed-off-by: Mark Brown <broonie@kernel.org>
Fix W=1 warning. The VOIP controls were not used in the mainline but
in special versions of Android. Keep and use __maybe_used to make
warning go away.
Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Link: https://lore.kernel.org/r/20200701183716.83314-3-pierre-louis.bossart@linux.intel.com
Signed-off-by: Mark Brown <broonie@kernel.org>
Support MAX98390 speaker amplifier on cometlake platform. Driver now
detects amplifier type in the probe function and installs corresponding
controls and DAPM widgets/routes in the late_probe function.
Signed-off-by: Brent Lu <brent.lu@intel.com>
Acked-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Link: https://lore.kernel.org/r/1593596211-28344-1-git-send-email-brent.lu@intel.com
Signed-off-by: Mark Brown <broonie@kernel.org>
Hi,
Changes since v3:
- Fix the single clock source handling and typo
Changes since v2:
- DT binding:
- use proper (?) patch subject for the binding docuemtn patch
- drop pll4 and pll15 from DT - driver should check the rate via
clk_get_parent. If it is not available (as it is not currently) then use the
match_data provided rates.
- add simple explanation for the clocking setup
- Use descriptive names for clocks: cpb/ivi-mcasp-auxclk and cpb/ivi-codec-scki
- dt_binding_check shows no errors/warnings
- ASoC machine driver:
- Try to read the PLL4/15 rate with clk API (parent of the two clock divider)
if it is not available then use the match_data provided numbers.
- Support for single PLL setup
Changes since v1:
- Fixed DT binding documentation errors
- Rebased on ASoC head and updated the driver to compile and work
This series adds support for the analog audio setup on the j721e EVM.
The audio setup of the EVM is:
Common Processor Board (CPB): McASP10 <-> pcm3168a
Infotainment Expansion Board (IVI): McASP0 <-> 2x pcm3168a
Both CPB and IVI wired in parallel serializer setup.
The first patch adds the stream_name for McASP driver as it is needed in
multicodec (and would be needed in DPCM) setup for proper DAPM handling.
The second patch adds two DT schema, one for the cpb and one for the cpb+ivi
card.
Regards,
Peter
---
Peter Ujfalusi (3):
ASoC: ti: davinci-mcasp: Specify stream_name for playback/capture
ASoC: dt-bindings: Add documentation for TI j721e EVM (CPB and IVI)
ASoC: ti: Add custom machine driver for j721e EVM (CPB and IVI)
.../bindings/sound/ti,j721e-cpb-audio.yaml | 95 ++
.../sound/ti,j721e-cpb-ivi-audio.yaml | 150 +++
sound/soc/ti/Kconfig | 8 +
sound/soc/ti/Makefile | 2 +
sound/soc/ti/davinci-mcasp.c | 3 +
sound/soc/ti/j721e-evm.c | 896 ++++++++++++++++++
6 files changed, 1154 insertions(+)
create mode 100644 Documentation/devicetree/bindings/sound/ti,j721e-cpb-audio.yaml
create mode 100644 Documentation/devicetree/bindings/sound/ti,j721e-cpb-ivi-audio.yaml
create mode 100644 sound/soc/ti/j721e-evm.c
--
Peter
Texas Instruments Finland Oy, Porkkalankatu 22, 00180 Helsinki.
Y-tunnus/Business ID: 0615521-4. Kotipaikka/Domicile: Helsinki
The ASRC not only supports ideal ratio mode, but also supports
internal ratio mode.
For internal rato mode, the rate of clock source should be divided
with no remainder by sample rate, otherwise there is sound
distortion.
Add function fsl_asrc_select_clk() to find proper clock source for
internal ratio mode, if the clock source is available then internal
ratio mode will be selected.
With change, the ideal ratio mode is not the only option for user.
Signed-off-by: Shengjiu Wang <shengjiu.wang@nxp.com>
Reviewed-by: Nicolin Chen <nicoleotsuka@gmail.com>
Reviewed-by: Fabio Estevam <festevam@gmail.com>
Link: https://lore.kernel.org/r/1593525367-23221-1-git-send-email-shengjiu.wang@nxp.com
Signed-off-by: Mark Brown <broonie@kernel.org>
Fixes interrupt enable condition check with which now
interrupt gets enabled in dma_open.
Prior to this patch it was getting enabled in runtime_resume only.
Signed-off-by: Akshu Agrawal <akshu.agrawal@amd.com>
Link: https://lore.kernel.org/r/20200630183754.20641-1-akshu.agrawal@amd.com
Signed-off-by: Mark Brown <broonie@kernel.org>
Clang warns:
In file included from sound/soc/intel/keembay/kmb_platform.c:14:
sound/soc/intel/keembay/kmb_platform.h:9:9: warning: 'KMB_PLATFORM_H_'
is used as a header guard here, followed by #define of a different
macro [-Wheader-guard]
#ifndef KMB_PLATFORM_H_
^~~~~~~~~~~~~~~
sound/soc/intel/keembay/kmb_platform.h:10:9: note: 'KMB_PLATFORMP_H_'
is defined here; did you mean 'KMB_PLATFORM_H_'?
#define KMB_PLATFORMP_H_
^~~~~~~~~~~~~~~~
KMB_PLATFORM_H_
1 warning generated.
Fix the typo so that the header guard works as intended.
Fixes: c5477e9667 ("ASoC: Intel: Add KeemBay platform driver")
Signed-off-by: Nathan Chancellor <natechancellor@gmail.com>
Reviewed-by: Nick Desaulniers <ndesaulniers@google.com>
Cc: Sia, Jee Heng <jee.heng.sia@intel.com>; alsa-devel@alsa-project.org; linux-kernel@vger.kernel.org; clang-built-linux@googlegroups.com; Nathan Chancellor <natechancellor@gmail.com>
Link: https://github.com/ClangBuiltLinux/linux/issues/1053
To: Rojewski, Cezary <cezary.rojewski@intel.com>; Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>; Liam Girdwood <liam.r.girdwood@linux.intel.com>; Jie Yang <yang.jie@linux.intel.com>; Mark Brown <broonie@kernel.org>
Link: https://lore.kernel.org/r/20200617010232.23222-1-natechancellor@gmail.com
Signed-off-by: Mark Brown <broonie@kernel.org>
When there is dedicated power domain bound with device, after probing
the power will be disabled, then registers are not accessible in
fsl_sai_dai_probe(), so regcache only need to be enabled in end of
probe() and regcache_mark_dirty should be moved to pm runtime resume
callback function.
Signed-off-by: Shengjiu Wang <shengjiu.wang@nxp.com>
Acked-by: Nicolin Chen <nicoleotsuka@gmail.com>
Link: https://lore.kernel.org/r/1593412953-10897-1-git-send-email-shengjiu.wang@nxp.com
Signed-off-by: Mark Brown <broonie@kernel.org>
The audio support on the board is using pcm3168a codec connected to McASP10
serializers in parallel setup.
The pcm3168a SCKI clock is coming via the j721e AUDIO_REFCLK2 pin.
In order to support 48KHz and 44.1KHz family of sampling rates the parent clock
for AUDIO_REFCLK2 needs to be changed between PLL4 (for 48KHz) and PLL15 (for
44.1KHz). The same PLLs are used for McASP10's AUXCLK clock via different
HSDIVIDER.
Generic card can not be used for the board as we need to switch between
clock paths for different sampling rate families and also need to change
the slot_width between 16 and 24 bit audio.
The audio support on the Infotainment Expansion Board consists of McASP0
connected to two pcm3168a codecs with dedicated set of serializers to each.
The SCKI for pcm3168a is sourced from j721e AUDIO_REFCLK0 pin.
It is extending the audio support on the CPB.
Due to the fact that the same PLL4/15 is used by both domains (CPB/IVI)
there are cross restriction on sampling rates.
The IVI side is represented as multicodec setup.
PCMs available on a plain CPB (no IVI addon):
hw:0,0 - cpb playback (8 channels)
hw:0,1 - cpb capture (6 channels)
When the IVI addon is present, additional two PCMs will be present:
hw:0,2 - ivi multicodec playback (16 channels)
hw:0,3 - ivi multicodec capture (12 channels)
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@ti.com>
Link: https://lore.kernel.org/r/20200630125843.11561-4-peter.ujfalusi@ti.com
Signed-off-by: Mark Brown <broonie@kernel.org>
S8 and S24 formats does not work on this machine driver so force to use
S16_LE instead.
In addition, add constraint to limit the max value of rate because the
rate higher than 96000(172000, 192000) is not stable either.
Signed-off-by: Yu-Hsuan Hsu <yuhsuan@chromium.org>
Link: https://lore.kernel.org/r/20200630091615.4020059-1-yuhsuan@chromium.org
Signed-off-by: Mark Brown <broonie@kernel.org>
When snd_pci_acp3x driver loads we see:
WARNING kernel:snd_pci_acp3x 0000:04:00.5: Unbalanced pm_runtime_enable!
at boot time.
same can be observed in /var/log/messages/.
Modifying pm runtime sequence for fixing unbalanced pm issue.
Signed-off-by: Ravulapati Vishnu vardhan rao <Vishnuvardhanrao.Ravulapati@amd.com>
Link: https://lore.kernel.org/r/20200630092242.7799-1-Vishnuvardhanrao.Ravulapati@amd.com
Signed-off-by: Mark Brown <broonie@kernel.org>
A few small driver specific fixes, nothing particularly dramatic.
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Merge tag 'asoc-fix-v5.8-rc3' of https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound into for-linus
ASoC: Fixes for v5.8
A few small driver specific fixes, nothing particularly dramatic.
The adc_vol_tlv volume-control has a range from -17.625 dB to +30 dB,
not -176.25 dB to + 300 dB. This wrong scale is esp. a problem in userspace
apps which translate the dB scale to a linear scale. With the logarithmic
dB scale being of by a factor of 10 we loose all precision in the lower
area of the range when apps translate things to a linear scale.
E.g. the 0 dB default, which corresponds with a value of 47 of the
0 - 127 range for the control, would be shown as 0/100 in alsa-mixer.
Since the centi-dB values used in the TLV struct cannot represent the
0.375 dB step size used by these controls, change the TLV definition
for them to specify a min and max value instead of min + stepsize.
Signed-off-by: Hans de Goede <hdegoede@redhat.com>
Link: https://lore.kernel.org/r/20200628155231.71089-5-hdegoede@redhat.com
Signed-off-by: Mark Brown <broonie@kernel.org>
The Lenovo Miix 2 10 has a keyboard dock with extra speakers in the dock.
Rather then the ACL5672's GPIO1 pin being used as IRQ to the CPU, it is
actually used to enable the amplifier for these speakers
(the IRQ to the CPU comes directly from the jack-detect switch).
Add a quirk for having an ext speaker-amplifier enable pin on GPIO1
and replace the Lenovo Miix 2 10's dmi_system_id table entry's wrong
GPIO_DEV quirk (which needs to be renamed to GPIO1_IS_IRQ) with the
new RT5670_GPIO1_IS_EXT_SPK_EN quirk, so that we enable the external
speaker-amplifier as necessary.
Also update the ident field for the dmi_system_id table entry, the
Miix models are not Thinkpads.
Fixes: 67e03ff3f3 ("ASoC: codecs: rt5670: add Thinkpad Tablet 10 quirk")
Signed-off-by: Hans de Goede <hdegoede@redhat.com>
BugLink: https://bugzilla.redhat.com/show_bug.cgi?id=1786723
Link: https://lore.kernel.org/r/20200628155231.71089-4-hdegoede@redhat.com
Signed-off-by: Mark Brown <broonie@kernel.org>
The RT5670_PWR_ANLG1 register has 3 bits to select the LDO voltage,
so the correct mask is 0x7 not 0x3.
Because of this wrong mask we were programming the ldo bits
to a setting of binary 001 (0x05 & 0x03) instead of binary 101
when moving to SND_SOC_BIAS_PREPARE.
According to the datasheet 001 is a reserved value, so no idea
what it did, since the driver was working fine before I guess we
got lucky and it does something which is ok.
Fixes: 5e8351de74 ("ASoC: add RT5670 CODEC driver")
Signed-off-by: Hans de Goede <hdegoede@redhat.com>
Cc: stable@vger.kernel.org
Link: https://lore.kernel.org/r/20200628155231.71089-3-hdegoede@redhat.com
Signed-off-by: Mark Brown <broonie@kernel.org>
The default mode for SSP configuration is TDM 4 slot and so far we were
using this for the bus format on cht-bsw-rt56732 boards.
One board, the Lenovo Miix 2 10 uses not 1 but 2 codecs connected to SSP2.
The second piggy-backed, output-only codec is inside the keyboard-dock
(which has extra speakers). Unlike the main rt5672 codec, we cannot
configure this codec, it is hard coded to use 2 channel 24 bit I2S.
Using 4 channel TDM leads to the dock speakers codec (which listens in on
the data send from the SSP to the rt5672 codec) emiting horribly distorted
sound.
Since we only support 2 channels anyways, there is no need for TDM on any
cht-bsw-rt5672 designs. So we can simply use I2S 2ch everywhere.
This commit fixes the Lenovo Miix 2 10 dock speakers issue by changing
the bus format set in cht_codec_fixup() to I2S 2 channel.
This change has been tested on the following devices with a rt5672 codec:
Lenovo Miix 2 10
Lenovo Thinkpad 8
Lenovo Thinkpad 10 (gen 1)
Signed-off-by: Hans de Goede <hdegoede@redhat.com>
Cc: stable@vger.kernel.org
BugLink: https://bugzilla.redhat.com/show_bug.cgi?id=1786723
Link: https://lore.kernel.org/r/20200628155231.71089-2-hdegoede@redhat.com
Signed-off-by: Mark Brown <broonie@kernel.org>
Add Vsense slot configuration based on the device tree. Adding this
property enables the slot programming to be moved to the tdm_set_slot
callback. This in affect sets the slots for the Isense and Vsense and
enabling this these modes are now based on whether these features were
powered on or not.
Signed-off-by: Dan Murphy <dmurphy@ti.com>
Link: https://lore.kernel.org/r/20200626154143.20351-3-dmurphy@ti.com
Signed-off-by: Mark Brown <broonie@kernel.org>
Add programming for the tdm slots for the right and left. This also
requires configuring the RX/TX offsets for the DAI format type.
Signed-off-by: Dan Murphy <dmurphy@ti.com>
Link: https://lore.kernel.org/r/20200626154143.20351-1-dmurphy@ti.com
Signed-off-by: Mark Brown <broonie@kernel.org>
When building on allyesconfig kernel for a NO_DMA=y platform (e.g.
Sun-3), CONFIG_SND_SOC_QCOM_COMMON=y, but CONFIG_SND_SOC_QDSP6_AFE=n,
leading to a link failure:
sound/soc/qcom/common.o: In function `qcom_snd_parse_of':
common.c:(.text+0x2e2): undefined reference to `q6afe_is_rx_port'
While SND_SOC_QDSP6 depends on HAS_DMA, SND_SOC_MSM8996 and SND_SOC_SDM845
don't, so the following warning is seen:
WARNING: unmet direct dependencies detected for SND_SOC_QDSP6
Depends on [n]: SOUND [=y] && !UML && SND [=y] && SND_SOC [=y] && QCOM_APR [=y] && HAS_DMA [=n]
Selected by [y]:
- SND_SOC_MSM8996 [=y] && SOUND [=y] && !UML && SND [=y] && SND_SOC [=y] && QCOM_APR [=y]
- SND_SOC_SDM845 [=y] && SOUND [=y] && !UML && SND [=y] && SND_SOC [=y] && QCOM_APR [=y] && CROS_EC [=y] && I2C [=y] && SOUNDWIRE [=y]
Until recently, this warning was harmless (from a compile-testing
point-of-view), but the new user of q6afe_is_rx_port() turned this into
a hard failure.
As the QDSP6 driver itself builds fine if NO_DMA=y, and it depends on
QCOM_APR (which in turns depends on ARCH_QCOM || COMPILE_TEST), it is
safe to increase compile testing coverage. Hence fix the link failure
by dropping the HAS_DMA dependency of SND_SOC_QDSP6.
Fixes: a212008925 ("ASoC: qcom: common: set correct directions for dailinks")
Fixes: 6b1687bf76 ("ASoC: qcom: add sdm845 sound card support")
Fixes: a6f933f63f ("ASoC: qcom: apq8096: Add db820c machine driver")
Signed-off-by: Geert Uytterhoeven <geert@linux-m68k.org>
Link: https://lore.kernel.org/r/20200629122443.21736-1-geert@linux-m68k.org
Signed-off-by: Mark Brown <broonie@kernel.org>
Add max98373-sdw helper function, which configures 2x MAX98373 codecs to
Link1. This patch shares code between the I2S and SoundWire modes of
MAX98373 and adds the trigger already added for I2S.
Signed-off-by: Rander Wang <rander.wang@linux.intel.com>
Signed-off-by: Naveen Manohar <naveen.m@intel.com>
Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Reviewed-by: Guennadi Liakhovetski <guennadi.liakhovetski@linux.intel.com>
Link: https://lore.kernel.org/r/20200625192620.4312-1-pierre-louis.bossart@linux.intel.com
Signed-off-by: Mark Brown <broonie@kernel.org>
RT5682 is in Soundwire mode on Link0 & 2x MAX98373 on link1.
Signed-off-by: Naveen Manohar <naveen.m@intel.com>
Reviewed-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/20200625192708.4416-1-pierre-louis.bossart@linux.intel.com
Signed-off-by: Mark Brown <broonie@kernel.org>
Silence documentation build warning by correcting kernel-doc comment
for snd_soc_runtime_action.
./sound/soc/soc-pcm.c:220: warning: Function parameter or member 'action' not described in 'snd_soc_runtime_action'
Signed-off-by: Colton Lewis <colton.w.lewis@protonmail.com>
Link: https://lore.kernel.org/r/20200626053953.68797-1-colton.w.lewis@protonmail.com
Signed-off-by: Mark Brown <broonie@kernel.org>
This patch updates tx_mask, so that (0-3)slots are reserved for
Maxim amps to feedback data.
V0->slot0,
I0->slot1,
V1->slot2,
I1->slot3.
also update slot_width in tdm configuration to 24 as the BE
configuration is 24 in topology.
Signed-off-by: Dharageswari R <dharageswari.r@intel.com>
Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Reviewed-by: Ranjani Sridharan <ranjani.sridharan@linux.intel.com>
Link: https://lore.kernel.org/r/20200625191308.3322-14-pierre-louis.bossart@linux.intel.com
Signed-off-by: Mark Brown <broonie@kernel.org>
for_each_card_prelinks() is a common API to walk through each prelink
in the card.
Signed-off-by: Bard Liao <yung-chuan.liao@linux.intel.com>
Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Reviewed-by: Ranjani Sridharan <ranjani.sridharan@linux.intel.com>
Link: https://lore.kernel.org/r/20200625191308.3322-13-pierre-louis.bossart@linux.intel.com
Signed-off-by: Mark Brown <broonie@kernel.org>
When the cml_rt1011_rt5682_dailink[].codecs pointer is overridden by
a quirk with a devm allocated structure and the probe is deferred,
in the next probe we will see an use-after-free condition
(verified with KASAN). This can be avoided by using statically allocated
configurations - which simplifies the code quite a bit as well.
KASAN issue fixed.
[ 23.301373] cml_rt1011_rt5682 cml_rt1011_rt5682: sof_rt1011_quirk = f
[ 23.301875] ==================================================================
[ 23.302018] BUG: KASAN: use-after-free in snd_cml_rt1011_probe+0x23a/0x3d0 [snd_soc_cml_rt1011_rt5682]
[ 23.302178] Read of size 8 at addr ffff8881ec6acae0 by task kworker/0:2/105
[ 23.302320] CPU: 0 PID: 105 Comm: kworker/0:2 Not tainted 5.7.0-rc7-test+ #3
[ 23.302322] Hardware name: Google Helios/Helios, BIOS 01/21/2020
[ 23.302329] Workqueue: events deferred_probe_work_func
[ 23.302331] Call Trace:
[ 23.302339] dump_stack+0x76/0xa0
[ 23.302345] print_address_description.constprop.0.cold+0xd3/0x43e
[ 23.302351] ? _raw_spin_lock_irqsave+0x7b/0xd0
[ 23.302355] ? _raw_spin_trylock_bh+0xf0/0xf0
[ 23.302362] ? snd_cml_rt1011_probe+0x23a/0x3d0 [snd_soc_cml_rt1011_rt5682]
[ 23.302365] __kasan_report.cold+0x37/0x86
[ 23.302371] ? snd_cml_rt1011_probe+0x23a/0x3d0 [snd_soc_cml_rt1011_rt5682]
[ 23.302375] kasan_report+0x38/0x50
[ 23.302382] snd_cml_rt1011_probe+0x23a/0x3d0 [snd_soc_cml_rt1011_rt5682]
[ 23.302389] platform_drv_probe+0x66/0xc0
Fixes: 629ba12e99 ("ASoC: Intel: boards: split woofer and tweeter support")
Suggested-by: Ranjani Sridharan <ranjani.sridharan@linux.intel.com>
Signed-off-by: Fred Oh <fred.oh@linux.intel.com>
Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Reviewed-by: Ranjani Sridharan <ranjani.sridharan@linux.intel.com>
Reviewed-by: Bard Liao <yung-chuan.liao@linux.intel.com>
Link: https://lore.kernel.org/r/20200625191308.3322-12-pierre-louis.bossart@linux.intel.com
Signed-off-by: Mark Brown <broonie@kernel.org>
Bard Liao