| Age | Commit message (Collapse) | Author |
|---|
|
With this change we no longer need to update num_drv when adding
new DAI driver.
Signed-off-by: Daniel Baluta <daniel.baluta@nxp.com>
Reviewed-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Reviewed-by: Kai Vehmanen <kai.vehmanen@linux.intel.com>
Reviewed-by: Ranjani Sridharan <ranjani.sridharan@linux.intel.com>
Reviewed-by: Guennadi Liakhovetski <guennadi.liakhovetski@linux.intel.com>
Link: https://lore.kernel.org/r/20200720072046.8152-3-daniel.baluta@oss.nxp.com
Signed-off-by: Mark Brown <broonie@kernel.org>
|
|
In the past, the INFO_ flags such as PAUSE/NO_PERIOD_WAKEUP were
defined in the SOF PCM core, but that was changed since
commit 27e322fabd50 ("ASoC: SOF: define INFO_ flags in dsp_ops")
Now these flags must be set in DSP ops.
Signed-off-by: Daniel Baluta <daniel.baluta@nxp.com>
Reviewed-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Reviewed-by: Kai Vehmanen <kai.vehmanen@linux.intel.com>
Reviewed-by: Ranjani Sridharan <ranjani.sridharan@linux.intel.com>
Reviewed-by: Guennadi Liakhovetski <guennadi.liakhovetski@linux.intel.com>
Link: https://lore.kernel.org/r/20200720072046.8152-2-daniel.baluta@oss.nxp.com
Signed-off-by: Mark Brown <broonie@kernel.org>
|
|
Pujar <spujar@nvidia.com>:
Overview
========
Audio Processing Engine (APE) comprises of Audio DMA (ADMA) and Audio
Hub (AHUB) unit. AHUB is a collection of hardware accelerators for audio
pre-processing and post-processing. It also includes a programmable full
crossbar for routing audio data across these accelerators.
This series exposes some of these below mentioned HW devices as ASoC
components for Tegra platforms from Tegra210 onwards.
* ADMAIF : The interface between ADMA and AHUB
* XBAR : Crossbar for routing audio samples across various modules
* I2S : Inter-IC Sound Controller
* DMIC : Digital Microphone
* DSPK : Digital Speaker
Following is the summary of current series.
* Add YAML DT binding documentation for above mentioned modules.
* Helper function for ACIF programming is exposed for Tegra210 and later.
* Add ASoC driver components for each of the above modules.
* Build ACONNECT and ADMA drivers which are essential to realize audio
use case.
* Add DT entries for above components for Tegra210, Tegra186 and
Tegra194.
As per the suggestion in [0] audio graph based sound card support
is pushed in a separate series.
[0] https://lkml.org/lkml/2020/6/27/4
Changelog
=========
v4 -> v5
--------
* Common changes
- simple-card driver changes are dropped. Changes are migrated to audio
graph card and are moved to a separate series as suggested.
- '#sound-dai-cells' property is not needed for planned audio graph card
Hence dropped from documentation and related DT binding of component
drivers.
- CIF and DAP DAIs are added for I/O drivers (DMIC, DSPK, I2S) to
represent DAI links using audio graph card. Similary DAIs are added in
AHUB driver to describe endpoints in audio crossbar. Routing is updated
to reflect the same in drivers.
v3 -> v4
--------
* [1/23] "ASoC: dt-bindings: tegra: Add DT bindings for Tegra210"
- Removed multiple examples and retained one example per doc
- Fixed as per inputs on the previous series
- Tested bindings with 'make dt_binding_check/dtbs_check'
* [2/23] "ASoC: tegra: Add support for CIF programming"
- No change
* Common changes (for patch [3/10] to [7/10])
- Mixer control overrides, for PCM parameters (rate, channel, bits),
in each driver are dropped.
- Updated routing as per DPCM usage
- Minor changes related to formatting
* New changes (patch [8/23] to [18/23] and patch [23/23])
- Based on discussions in following threads DPCM is used for Tegra Audio.
https://lkml.org/lkml/2020/2/20/91
https://lkml.org/lkml/2020/4/30/519
- The simple-card driver is used for Tegra Audio and accordingly
some enhancements are made in simple-card and core drivers.
- Patch [8/23] to [18/23] are related to simple-card and core changes.
- Patch [23/23] adds sound card support to realize complete audio path.
This is based on simple-card driver with proposed enhancements.
- Re-ordered patches depending on above
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
|
|
ADMAIF is the interface between ADMA and AHUB. Each ADMA channel that
sends/receives data to/from AHUB must intreface through an ADMAIF channel.
ADMA channel sending data to AHUB pairs with an ADMAIF Tx channel and
similarly ADMA channel receiving data from AHUB pairs with an ADMAIF Rx
channel. Buffer size is configurable for each ADMAIF channel, but currently
SW uses default values.
This patch registers ADMAIF driver with ASoC framework. The component
driver exposes DAPM widgets, routes and kcontrols for the device. The DAI
driver exposes ADMAIF interfaces, which can be used to connect different
components in the ASoC layer. Makefile and Kconfig support is added to
allow to build the driver. The ADMAIF device can be enabled in the DT via
"nvidia,tegra210-admaif" compatible binding.
Tegra PCM driver is updated to expose required PCM interfaces and
snd_pcm_ops callbacks.
Signed-off-by: Sameer Pujar <spujar@nvidia.com>
Link: https://lore.kernel.org/r/1595134890-16470-8-git-send-email-spujar@nvidia.com
Signed-off-by: Mark Brown <broonie@kernel.org>
|
|
Fix the reset property name when allocating the GPIO descriptor.
The gpiod_get_optional appends either the -gpio or -gpios suffix to the
name.
Fixes: 1a476abc723e6 ("tas2770: add tas2770 smart PA kernel driver")
Signed-off-by: Dan Murphy <dmurphy@ti.com>
Link: https://lore.kernel.org/r/20200720181202.31000-2-dmurphy@ti.com
Signed-off-by: Mark Brown <broonie@kernel.org>
|
|
Bossart <pierre-louis.bossart@linux.intel.com>:
Small patchset to harden the SoundWire machine driver, change bad
HIDs, update PLL settings and avoid memory leaks. Given that the
SoundWire core parts are not upstream it's probably not necessary to
provide the patches to stable branches.
Bard Liao (1):
ASoC: Intel: sof_sdw_rt711: remove hard-coded codec name
Kai Vehmanen (2):
ASoC: Intel: sof_sdw: add support for systems without i915 audio
ASoC: Intel: sof_sdw: avoid crash if invalid DSP topology loaded
Libin Yang (1):
ASoC: Intel: common: change match table ehl-rt5660
Pierre-Louis Bossart (1):
ASoC: Intel: sof_sdw_rt711: remove properties in card remove
Yong Zhi (1):
ASoC: intel: board: sof_rt5682: Update rt1015 pll input clk freq
sound/soc/intel/boards/sof_rt5682.c | 9 +++++-
sound/soc/intel/boards/sof_sdw.c | 31 +++++++++++++------
sound/soc/intel/boards/sof_sdw_common.h | 2 ++
sound/soc/intel/boards/sof_sdw_hdmi.c | 6 ++++
sound/soc/intel/boards/sof_sdw_rt711.c | 17 +++++++++-
.../intel/common/soc-acpi-intel-ehl-match.c | 2 +-
6 files changed, 54 insertions(+), 13 deletions(-)
base-commit: 22e9b54307987787efa0ee534aa9e31982ec1161
--
2.25.1
|
|
In commit d696a61413b4 ("ASoC: rt1015: Add condition to prevent SoC
providing bclk in ratio of 50 times of sample rate."), PLL input at 50fs
is no longer supported, the new recommended settings at 48Khz rate are:
PLL input SSP bclk
------------------------
64fs 3.073Mhz
100fs 4.8Mhz
(bclk update is reflected in topoplogy.)
Signed-off-by: Yong Zhi <yong.zhi@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/20200717211337.31956-6-pierre-louis.bossart@linux.intel.com
Signed-off-by: Mark Brown <broonie@kernel.org>
|
|
The mc_private->hdmi_pcm_list is populated by elements loaded during
DSP topology load. Valid topologies for this machine driver will always
have PCM nodes for HDMI, but driver should fail gracefully even in the case
this is not true. Add a sanity check to sof_sdw_hdmi_card_late_probe()
for this case. Without the fix, a null pcm handle gets dereferenced.
Signed-off-by: Kai Vehmanen <kai.vehmanen@linux.intel.com>
Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Reviewed-by: Guennadi Liakhovetski <guennadi.liakhovetski@linux.intel.com>
Reviewed-by: Rander Wang <rander.wang@linux.intel.com>
Link: https://lore.kernel.org/r/20200717211337.31956-5-pierre-louis.bossart@linux.intel.com
Signed-off-by: Mark Brown <broonie@kernel.org>
|
|
Extend the generic SOF Soundwire machine driver to support systems where
iDisp HDMI/DP audio codec is disabled for some reason (i915 driver
disabled, HDMI/DP implemented with a discrete GPU, etc). Switch codecs
to SoC dummy in the affected DAI links. This allows to reuse existing
topologies for this case.
Signed-off-by: Kai Vehmanen <kai.vehmanen@linux.intel.com>
Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Reviewed-by: Guennadi Liakhovetski <guennadi.liakhovetski@linux.intel.com>
Reviewed-by: Rander Wang <rander.wang@linux.intel.com>
Link: https://lore.kernel.org/r/20200717211337.31956-4-pierre-louis.bossart@linux.intel.com
Signed-off-by: Mark Brown <broonie@kernel.org>
|
|
The rt711 jack detection properties are set from the machine drivers
during the card probe, as done in other ASoC examples.
KASAN reports a use-after-free error when unbinding drivers due to a
confusing sequence between the ACPI core, the device core and the
SoundWire device cleanups.
Rather than fixing this sequence, follow the recommendation to have
the same caller add and remove properties, add an explicit
device_remove_properties() in the card .remove() callback.
In future patches the use of device_add/remove_properties will be
replaced by a direct handling of a swnode, but the sequence will
remain the same.
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/20200717211337.31956-3-pierre-louis.bossart@linux.intel.com
Signed-off-by: Mark Brown <broonie@kernel.org>
|
|
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>
|
|
Pujar <spujar@nvidia.com>:
Overview
========
Audio Processing Engine (APE) comprises of Audio DMA (ADMA) and Audio
Hub (AHUB) unit. AHUB is a collection of hardware accelerators for audio
pre-processing and post-processing. It also includes a programmable full
crossbar for routing audio data across these accelerators.
This series exposes some of these below mentioned HW devices as ASoC
components for Tegra platforms from Tegra210 onwards.
* ADMAIF : The interface between ADMA and AHUB
* XBAR : Crossbar for routing audio samples across various modules
* I2S : Inter-IC Sound Controller
* DMIC : Digital Microphone
* DSPK : Digital Speaker
Following is the summary of current series.
* Add YAML DT binding documentation for above mentioned modules.
* Helper function for ACIF programming is exposed for Tegra210 and later.
* Add ASoC driver components for each of the above modules.
* Build ACONNECT and ADMA drivers which are essential to realize audio
use case.
* Add DT entries for above components for Tegra210, Tegra186 and
Tegra194.
As per the suggestion in [0] audio graph based sound card support
is pushed in a separate series.
[0] https://lkml.org/lkml/2020/6/27/4
Changelog
=========
v4 -> v5
--------
* Common changes
- simple-card driver changes are dropped. Changes are migrated to audio
graph card and are moved to a separate series as suggested.
- '#sound-dai-cells' property is not needed for planned audio graph card
Hence dropped from documentation and related DT binding of component
drivers.
- CIF and DAP DAIs are added for I/O drivers (DMIC, DSPK, I2S) to
represent DAI links using audio graph card. Similary DAIs are added in
AHUB driver to describe endpoints in audio crossbar. Routing is updated
to reflect the same in drivers.
v3 -> v4
--------
* [1/23] "ASoC: dt-bindings: tegra: Add DT bindings for Tegra210"
- Removed multiple examples and retained one example per doc
- Fixed as per inputs on the previous series
- Tested bindings with 'make dt_binding_check/dtbs_check'
* [2/23] "ASoC: tegra: Add support for CIF programming"
- No change
* Common changes (for patch [3/10] to [7/10])
- Mixer control overrides, for PCM parameters (rate, channel, bits),
in each driver are dropped.
- Updated routing as per DPCM usage
- Minor changes related to formatting
* New changes (patch [8/23] to [18/23] and patch [23/23])
- Based on discussions in following threads DPCM is used for Tegra Audio.
https://lkml.org/lkml/2020/2/20/91
https://lkml.org/lkml/2020/4/30/519
- The simple-card driver is used for Tegra Audio and accordingly
some enhancements are made in simple-card and core drivers.
- Patch [8/23] to [18/23] are related to simple-card and core changes.
- Patch [23/23] adds sound card support to realize complete audio path.
This is based on simple-card driver with proposed enhancements.
- Re-ordered patches depending on above
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
|
|
Bossart <pierre-louis.bossart@linux.intel.com>:
Small patchset to harden the SoundWire machine driver, change bad
HIDs, update PLL settings and avoid memory leaks. Given that the
SoundWire core parts are not upstream it's probably not necessary to
provide the patches to stable branches.
Bard Liao (1):
ASoC: Intel: sof_sdw_rt711: remove hard-coded codec name
Kai Vehmanen (2):
ASoC: Intel: sof_sdw: add support for systems without i915 audio
ASoC: Intel: sof_sdw: avoid crash if invalid DSP topology loaded
Libin Yang (1):
ASoC: Intel: common: change match table ehl-rt5660
Pierre-Louis Bossart (1):
ASoC: Intel: sof_sdw_rt711: remove properties in card remove
Yong Zhi (1):
ASoC: intel: board: sof_rt5682: Update rt1015 pll input clk freq
sound/soc/intel/boards/sof_rt5682.c | 9 +++++-
sound/soc/intel/boards/sof_sdw.c | 31 +++++++++++++------
sound/soc/intel/boards/sof_sdw_common.h | 2 ++
sound/soc/intel/boards/sof_sdw_hdmi.c | 6 ++++
sound/soc/intel/boards/sof_sdw_rt711.c | 17 +++++++++-
.../intel/common/soc-acpi-intel-ehl-match.c | 2 +-
6 files changed, 54 insertions(+), 13 deletions(-)
base-commit: 22e9b54307987787efa0ee534aa9e31982ec1161
--
2.25.1
|
|
Fix the doubled word "in" in a comment by adding punctuation
in 3 places and capitalization.
Signed-off-by: Randy Dunlap <rdunlap@infradead.org>
Cc: Mark Brown <broonie@kernel.org>
Cc: alsa-devel@alsa-project.org
Link: https://lore.kernel.org/r/20200719180901.30720-1-rdunlap@infradead.org
Signed-off-by: Mark Brown <broonie@kernel.org>
|
|
Delete the doubled word "to" in two comments.
Signed-off-by: Randy Dunlap <rdunlap@infradead.org>
Cc: Stephen Warren <swarren@nvidia.com>
Cc: Mark Brown <broonie@kernel.org>
Cc: alsa-devel@alsa-project.org
Link: https://lore.kernel.org/r/20200719180912.30770-1-rdunlap@infradead.org
Signed-off-by: Mark Brown <broonie@kernel.org>
|
|
One-element arrays are being deprecated[1]. Replace the one-element
array with a simple value type 'u8 reserved'[2], once it seems this
is just a placeholder for alignment.
[1] https://github.com/KSPP/linux/issues/79
[2] https://github.com/KSPP/linux/issues/86
Signed-off-by: Gustavo A. R. Silva <gustavoars@kernel.org>
Tested-by: kernel test robot <lkp@intel.com>
Reviewed-by: Amadeusz Sławiński <amadeuszx.slawinski@linux.intel.com>
Link: https://github.com/GustavoARSilva/linux-hardening/blob/master/cii/0-day/skylake-20200717.md
Link: https://lore.kernel.org/r/20200717215500.GA13910@embeddedor
Signed-off-by: Mark Brown <broonie@kernel.org>
|
|
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>
|
|
Morimoto <kuninori.morimoto.gx@renesas.com>:
Hi Mark
These are v4 digital_mute() patch which adjusts
to atmel which had conflict on v3.
v3 -> v4
- tidyup for atmel which had conflict
v2 -> v3
- uses "xxx_mute_stream" for .mute_stream naming
if it was better
- removed verbose Cc email address
v1 -> v2
- return -ENOTSUPP at hdmi-codec
- add new .no_capture_mute flag and emulate .digital_mute()
by .mute_stream()
Link: https://lore.kernel.org/r/87h7uhxxk6.wl-kuninori.morimoto.gx@renesas.com
Link: https://lore.kernel.org/r/874kqy2y5t.wl-kuninori.morimoto.gx@renesas.com
Link: https://lore.kernel.org/r/87ftam37ko.wl-kuninori.morimoto.gx@renesas.com
Kuninori Morimoto (3):
ASoC: soc-dai: remove .digital_mute
ASoC: soc-core: snd_soc_dai_digital_mute() for both CPU/Codec
ASoC: soc-xxx: add asoc_substream_to_rtd()
include/sound/soc-dai.h | 1 -
include/sound/soc.h | 2 +
sound/soc/soc-component.c | 20 ++++-----
sound/soc/soc-core.c | 4 +-
sound/soc/soc-dai.c | 12 ++----
sound/soc/soc-dapm.c | 6 +--
sound/soc/soc-generic-dmaengine-pcm.c | 4 +-
sound/soc/soc-link.c | 12 +++---
sound/soc/soc-pcm.c | 62 +++++++++++++--------------
sound/soc/soc-utils.c | 2 +-
10 files changed, 61 insertions(+), 64 deletions(-)
--
2.25.1
|
|
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>
|
|
Use the defined variable "dev" to make the code cleaner.
Signed-off-by: Zhang Shengju <zhangshengju@cmss.chinamobile.com>
Signed-off-by: Tang Bin <tangbin@cmss.chinamobile.com>
Link: https://lore.kernel.org/r/20200714112923.14944-1-tangbin@cmss.chinamobile.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: c84836d7f650 ("ASoC: meson: axg-card: use modern dai_link style")
Signed-off-by: Jing Xiangfeng <jingxiangfeng@huawei.com>
Reviewed-by: Jerome Brunet <jbrunet@baylibre.com>
Link: https://lore.kernel.org/r/20200717082242.130627-1-jingxiangfeng@huawei.com
Signed-off-by: Mark Brown <broonie@kernel.org>
|
|
<kai.vehmanen@linux.intel.com>:
Hi,
this small series is preparation for a set of bugfix ASoC patches
addressing a memleak at module unload for the HDA codec wrapper.
Instead of duplicating HDA code in ASoC tree, I chose to export
more functionality from hda_codec.c so it can be (re)used in ASoC's
hdac_hda.c.
Full series:
https://github.com/thesofproject/linux/pull/2252
Takashi and Mark, feedback is welcome on how to best handle this
kind of series where I have dependent patches both in sound/pci/hda
and in ASoC. For this series, I'm sending the patches separately
and when/if first set is merged by Takashi, I'll route the ASoC
patches via our usually SOF set to Mark.
Kai Vehmanen (2):
ALSA: hda: export snd_hda_codec_cleanup_for_unbind()
ALSA: hda: fix snd_hda_codec_cleanup() documentation
include/sound/hda_codec.h | 2 ++
sound/pci/hda/hda_codec.c | 3 ++-
2 files changed, 4 insertions(+), 1 deletion(-)
--
2.27.0
|
|
Shengjiu Wang <shengjiu.wang@nxp.com>:
Support hp and mic detection.
Add a parameter for asoc_simple_init_jack.
Shengjiu Wang (3):
ASoC: simple-card-utils: Support configure pin_name for
asoc_simple_init_jack
ASoC: bindings: fsl-asoc-card: Support hp-det-gpio and mic-det-gpio
ASoC: fsl-asoc-card: Support Headphone and Microphone Jack detection
changes in v2:
- Add more comments in third commit
- Add Acked-by Nicolin.
.../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
|
|
Morimoto <kuninori.morimoto.gx@renesas.com>:
Hi Mark
These are v3 patch-set.
ALSA SoC has 2 mute callbacks (= .digital_mute(), .mute_stream()).
But the difference between these 2 are very small.
.digital_mute() is for Playback
.mute_stream() is for Playback/Capture
This patch-set adds new .no_capture_mute flag and emulate
.digital_mute() by .mute_stream().
v2 -> v3
- uses "xxx_mute_stream" for .mute_stream naming
if it was better
- removed verbose Cc email address
v1 -> v2
- return -ENOTSUPP at hdmi-codec
- add new .no_capture_mute flag and emulate .digital_mute()
by .mute_stream()
Link: https://lore.kernel.org/r/874kqy2y5t.wl-kuninori.morimoto.gx@renesas.com
Link: https://lore.kernel.org/r/87ftam37ko.wl-kuninori.morimoto.gx@renesas.com
Kuninori Morimoto (21):
ASoC: hdmi-codec: return -ENOTSUPP for digital_mute
ASoC: soc-dai.c: add .no_capture_mute support
ASoC: hdmi-codec: merge .digital_mute() into .mute_stream()
ASoC: ti: merge .digital_mute() into .mute_stream()
ASoC: spear: merge .digital_mute() into .mute_stream()
ASoC: meson: merge .digital_mute() into .mute_stream()
ASoC: atmel: merge .digital_mute() into .mute_stream()
ASoC: codecs: merge .digital_mute() into .mute_stream()
ASoC: codecs: tlv*: merge .digital_mute() into .mute_stream()
ASoC: codecs: tas*: merge .digital_mute() into .mute_stream()
ASoC: codecs: ssm*: merge .digital_mute() into .mute_stream()
ASoC: codecs: pcm*: merge .digital_mute() into .mute_stream()
ASoC: codecs: max*: merge .digital_mute() into .mute_stream()
ASoC: codecs: alc*: merge .digital_mute() into .mute_stream()
ASoC: codecs: wm*: merge .digital_mute() into .mute_stream()
ASoC: codecs: es*: merge .digital_mute() into .mute_stream()
ASoC: codecs: da*: merge .digital_mute() into .mute_stream()
ASoC: codecs: cs*: merge .digital_mute() into .mute_stream()
ASoC: codecs: ak*: merge .digital_mute() into .mute_stream()
ASoC: soc-dai: remove .digital_mute
ASoC: soc-core: snd_soc_dai_digital_mute() for both CPU/Codec
drivers/gpu/drm/bridge/sii902x.c | 7 ++++---
drivers/gpu/drm/exynos/exynos_hdmi.c | 6 ++++--
drivers/gpu/drm/i2c/tda998x_drv.c | 7 ++++---
drivers/gpu/drm/mediatek/mtk_hdmi.c | 6 ++++--
drivers/gpu/drm/rockchip/cdn-dp-core.c | 7 ++++---
drivers/gpu/drm/sti/sti_hdmi.c | 6 ++++--
drivers/gpu/drm/zte/zx_hdmi.c | 7 ++++---
include/sound/hdmi-codec.h | 6 +++++-
include/sound/soc-dai.h | 4 +++-
sound/soc/atmel/atmel-classd.c | 7 ++++---
sound/soc/codecs/88pm860x-codec.c | 8 +++++---
sound/soc/codecs/ad193x.c | 5 +++--
sound/soc/codecs/adau1701.c | 5 +++--
sound/soc/codecs/ak4458.c | 5 +++--
sound/soc/codecs/ak4535.c | 6 ++++--
sound/soc/codecs/ak4641.c | 8 +++++---
sound/soc/codecs/alc5623.c | 5 +++--
sound/soc/codecs/alc5632.c | 5 +++--
sound/soc/codecs/cpcap.c | 11 +++++++----
sound/soc/codecs/cq93vc.c | 5 +++--
sound/soc/codecs/cs4265.c | 5 +++--
sound/soc/codecs/cs4270.c | 5 +++--
sound/soc/codecs/cs42l42.c | 5 +++--
sound/soc/codecs/cs42l51.c | 5 +++--
sound/soc/codecs/cs42l52.c | 5 +++--
sound/soc/codecs/cs42l56.c | 5 +++--
sound/soc/codecs/cs42xx8.c | 5 +++--
sound/soc/codecs/cs4341.c | 5 +++--
sound/soc/codecs/cs4349.c | 5 +++--
sound/soc/codecs/da7210.c | 5 +++--
sound/soc/codecs/da7213.c | 5 +++--
sound/soc/codecs/da9055.c | 5 +++--
sound/soc/codecs/es8316.c | 5 +++--
sound/soc/codecs/es8328.c | 5 +++--
sound/soc/codecs/hdmi-codec.c | 25 +++++++++++++++++--------
sound/soc/codecs/isabelle.c | 15 +++++++++------
sound/soc/codecs/jz4770.c | 5 +++--
sound/soc/codecs/lm49453.c | 25 +++++++++++++++----------
sound/soc/codecs/max98088.c | 12 ++++++++----
sound/soc/codecs/max98090.c | 6 ++++--
sound/soc/codecs/max9867.c | 5 +++--
sound/soc/codecs/ml26124.c | 5 +++--
sound/soc/codecs/nau8822.c | 5 +++--
sound/soc/codecs/pcm1681.c | 5 +++--
sound/soc/codecs/pcm1789.c | 5 +++--
sound/soc/codecs/pcm179x.c | 5 +++--
sound/soc/codecs/pcm3168a.c | 5 +++--
sound/soc/codecs/pcm512x.c | 5 +++--
sound/soc/codecs/rk3328_codec.c | 5 +++--
sound/soc/codecs/sgtl5000.c | 5 +++--
sound/soc/codecs/ssm2518.c | 5 +++--
sound/soc/codecs/ssm2602.c | 5 +++--
sound/soc/codecs/ssm4567.c | 5 +++--
sound/soc/codecs/sta529.c | 5 +++--
sound/soc/codecs/tas2552.c | 5 +++--
sound/soc/codecs/tas2562.c | 5 +++--
sound/soc/codecs/tas2770.c | 5 +++--
sound/soc/codecs/tas571x.c | 5 +++--
sound/soc/codecs/tas5720.c | 5 +++--
sound/soc/codecs/tas6424.c | 5 +++--
sound/soc/codecs/tfa9879.c | 5 +++--
sound/soc/codecs/tlv320aic23.c | 5 +++--
sound/soc/codecs/tlv320aic26.c | 5 +++--
sound/soc/codecs/tlv320aic31xx.c | 6 ++++--
sound/soc/codecs/tlv320aic32x4.c | 5 +++--
sound/soc/codecs/tlv320aic3x.c | 5 +++--
sound/soc/codecs/twl6040.c | 5 +++--
sound/soc/codecs/uda134x.c | 5 +++--
sound/soc/codecs/wm8350.c | 5 +++--
sound/soc/codecs/wm8400.c | 5 +++--
sound/soc/codecs/wm8510.c | 5 +++--
sound/soc/codecs/wm8580.c | 5 +++--
sound/soc/codecs/wm8711.c | 5 +++--
sound/soc/codecs/wm8728.c | 5 +++--
sound/soc/codecs/wm8731.c | 5 +++--
sound/soc/codecs/wm8741.c | 5 +++--
sound/soc/codecs/wm8750.c | 5 +++--
sound/soc/codecs/wm8753.c | 8 +++++---
sound/soc/codecs/wm8770.c | 5 +++--
sound/soc/codecs/wm8776.c | 5 +++--
sound/soc/codecs/wm8900.c | 5 +++--
sound/soc/codecs/wm8903.c | 5 +++--
sound/soc/codecs/wm8904.c | 5 +++--
sound/soc/codecs/wm8940.c | 5 +++--
sound/soc/codecs/wm8955.c | 5 +++--
sound/soc/codecs/wm8960.c | 5 +++--
sound/soc/codecs/wm8961.c | 5 +++--
sound/soc/codecs/wm8962.c | 5 +++--
sound/soc/codecs/wm8971.c | 5 +++--
sound/soc/codecs/wm8974.c | 5 +++--
sound/soc/codecs/wm8978.c | 5 +++--
sound/soc/codecs/wm8983.c | 7 ++++---
sound/soc/codecs/wm8985.c | 7 ++++---
sound/soc/codecs/wm8988.c | 5 +++--
sound/soc/codecs/wm8990.c | 5 +++--
sound/soc/codecs/wm8991.c | 7 ++++---
sound/soc/codecs/wm8993.c | 5 +++--
sound/soc/codecs/wm8994.c | 9 ++++++---
sound/soc/codecs/wm8995.c | 8 +++++---
sound/soc/codecs/wm9081.c | 5 +++--
sound/soc/meson/axg-spdifout.c | 5 +++--
sound/soc/soc-core.c | 4 ++--
sound/soc/soc-dai.c | 12 +++++++-----
sound/soc/spear/spdif_out.c | 8 +++++---
sound/soc/ti/ams-delta.c | 9 +++++----
105 files changed, 390 insertions(+), 249 deletions(-)
--
2.25.1
|
|
Add missed return for calling soc_component_ret, otherwise the return
value is wrong.
Fixes: e2329eeba45f ("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>
|
|
Signed-off-by: Lee Jones <lee.jones@linaro.org>
Cc: Timur Tabi <timur@kernel.org>
Cc: Nicolin Chen <nicoleotsuka@gmail.com>
Cc: Xiubo Li <Xiubo.Lee@gmail.com>
Cc: Fabio Estevam <festevam@gmail.com>
Cc: linuxppc-dev@lists.ozlabs.org
Link: https://lore.kernel.org/r/20200715150009.407442-1-lee.jones@linaro.org
Signed-off-by: Mark Brown <broonie@kernel.org>
|
|
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>
|
