- 22 Sep, 2021 8 commits
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Mark Brown authored
As part of moving to remove the old style defines for the bus clocks update the fsl_ssi driver to use more modern terminology for clocking. Signed-off-by:
Mark Brown <broonie@kernel.org> Reviewed-by:
Fabio Estevam <festevam@gmail.com> Link: https://lore.kernel.org/r/20210921213542.31688-7-broonie@kernel.orgSigned-off-by:
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Mark Brown authored
As part of moving to remove the old style defines for the bus clocks update the fsl_sai driver to use more modern terminology for clocking. Signed-off-by:
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Mark Brown authored
As part of moving to remove the old style defines for the bus clocks update the fsl-mqs driver to use more modern terminology for clocking. Signed-off-by:
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Mark Brown authored
As part of moving to remove the old style defines for the bus clocks update the fsl-esai driver to use more modern terminology for clocking. Signed-off-by:
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Mark Brown authored
As part of moving to remove the old style defines for the bus clocks update the fsl-audmix driver to use more modern terminology for clocking. Signed-off-by:
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Mark Brown authored
As part of moving to remove the old style defines for the bus clocks update the fsl-asoc-card driver to use more modern terminology for clocking. Signed-off-by:
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Mark Brown authored
As part of moving to remove the old style defines for the bus clocks update the eureka-tlv320 driver to use more modern terminology for clocking. Signed-off-by:
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Mark Brown authored
As part of moving to remove the old style defines for the bus clocks update the cros_ec_codec driver to use more modern terminology for clocking. Signed-off-by:
Enric Balletbo i Serra <enric.balletbo@collabora.com> Link: https://lore.kernel.org/r/20210920170414.17903-1-broonie@kernel.orgSigned-off-by:
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- 21 Sep, 2021 4 commits
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Sameer Pujar authored
commit 82d3ec1d ("ASoC: Use schema reference for sound-name-prefix") added name-prefix.yaml schema and the same reference was used in couple of other schemas. But this is causing following warning and the same is fixed in current patch. Documentation/devicetree/bindings/sound/nxp,tfa989x.example.dt.yaml: audio-codec@34: 'sound-name-prefix' does not match any of the regexes: 'pinctrl-[0-9]+' From schema: Documentation/devicetree/bindings/sound/nxp,tfa989x.yaml Documentation/devicetree/bindings/sound/nxp,tfa989x.example.dt.yaml: audio-codec@36: 'sound-name-prefix' does not match any of the regexes: 'pinctrl-[0-9]+' From schema: Documentation/devicetree/bindings/sound/nxp,tfa989x.yaml Fixes: 82d3ec1d ("ASoC: Use schema reference for sound-name-prefix") Reported-by:
Rob Herring <robh+dt@kernel.org> Suggested-by:
Sameer Pujar <spujar@nvidia.com> Link: https://lore.kernel.org/r/1632238860-16947-1-git-send-email-spujar@nvidia.comSigned-off-by:
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Colin Ian King authored
There is a spelling mistake in the module description. Fix it. Signed-off-by:
Colin Ian King <colin.king@canonical.com> Reviewed-by:
Srinivas Kandagatla <srinivas.kandagatla@linaro.org> Link: https://lore.kernel.org/r/20210920184152.18109-1-colin.king@canonical.comSigned-off-by:
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Shengjiu Wang authored
On i.MX8ULP the spdif works with EDMA, so add compatible string and soc specific data for i.MX8ULP. Signed-off-by:
Shengjiu Wang <shengjiu.wang@nxp.com> Link: https://lore.kernel.org/r/1631238562-27081-1-git-send-email-shengjiu.wang@nxp.comSigned-off-by:
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Rikard Falkeborn authored
These are only assigned to the ops field in the snd_soc_dai_link struct which is a pointer to const struct snd_soc_ops. Make them const to allow the compiler to put them in read-only memory. Signed-off-by:
Rikard Falkeborn <rikard.falkeborn@gmail.com> Link: https://lore.kernel.org/r/20210920193947.10237-1-rikard.falkeborn@gmail.comSigned-off-by:
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- 20 Sep, 2021 21 commits
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Mark Brown authored
Revert 6e8cc4dd ("spi: tegra20-slink: Declare runtime suspend and resume functions conditionally") which was mistakenly applied to the ASoC tree not the SPI tree (where it was also applied. Signed-off-by:
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Mark Brown authored
Merge series "ASoC: compress: Support module_get on stream open" from Peter Ujfalusi <peter.ujfalusi@linux.intel.com>: Hi, SOF is marking all componet drivers with module_get_upon_open = 1 which works fine with normal PCM streams, however on compressed side the module get upon open is not supported. The module_get works when module_get_upon_open is not set becasue the snd_soc_component_module_get_when_probe() will pass NULL for the substream parameter of snd_soc_component_module_get(). In order to re-use the existing infrastructure for module_get, the proposal is to convert the mark_module to void pointer (like the pm mark) and implement matching code for the compressed open/free to pcm open/close. Regards, Peter --- Peter Ujfalusi (2): ASoC: soc-component: Convert the mark_module to void* ASoC: compress/component: Use module_get_when_open/put_when_close for cstream include/sound/soc-component.h | 14 ++++---- sound/soc/soc-component.c | 61 +++++++++++++++-------------------- sound/soc/soc-compress.c | 43 +++++++++++++++++++++--- 3 files changed, 71 insertions(+), 47 deletions(-) -- 2.33.0 -
Mark Brown authored
Merge series "Extend AHUB audio support for Tegra210 and later" from Sameer Pujar <spujar@nvidia.com>: Earlier as part of series [0], support for ADMAIF and I/O modules (such as I2S, DMIC and DSPK) was added. This series aims at exposing some of the AHUB internal modules (listed below), which can be used for audio pre or post processing. * SFC (Sampling Frequency Converter) * MVC (Master Volume Control) * AMX (Audio Multiplexer) * ADX (Audio Demultiplexer) * Mixer These modules can be plugged into audio paths and relevant processing can be done. The MUX routes are extended to allow add or remove above modules in the path via mixer controls. This is similar to how specific ADMAIF channels are connected to relevant I/O module instances at the moment. Some of these modules can alter PCM parameters. Consider example of resampler (44.1 -> 48 kHz) in the path. aplay(44.1 kHz) -> ADMAIF -> SFC -> (48 kHz) I2S -> (48kHz) Codec The modules following SFC should be using converted sample rate and DAIs need to be configured accordingly. The audio-graph driver provides a mechanism to fixup the new parameters which can be specified in DT for a given DAI. Then core uses these new values via fixup callback and then pass it to respective DAIs hw_param() callback. The "convert-rate", described in [1], property can be used when there is rate conversion in the audio path. Similarly "convert-channels" can be used when there is channel conversion in the path. There is no "convert-xxx" property for sample size conversions. It can be added if necessary. [0] https://www.lkml.org/lkml/2020/7/21/1357 [1] Documentation/devicetree/bindings/sound/audio-graph-port.yaml Changelog ========= v1 -> v2 -------- * Put comments for soft reset application in the drivers. * Split out mute/volume control logic in put() calls of MVC driver and use separate callbacks for the respective kcontrols. * Update kcontrol put() callback in MVC driver to return 1 whenever there is change. Similar change is done in other drivers too. * Use name-prefix.yaml reference for the driver documentation now. * Add sound-name-prefix pattern for MIXER driver and use prefix accordingly in DT. Sameer Pujar (13): ASoC: soc-pcm: Don't reconnect an already active BE ASoC: simple-card-utils: Increase maximum DAI links limit to 512 ASoC: audio-graph: Fixup CPU endpoint hw_params in a BE<->BE link ASoC: dt-bindings: tegra: Few more Tegra210 AHUB modules ASoC: tegra: Add routes for few AHUB modules ASoC: tegra: Add Tegra210 based MVC driver ASoC: tegra: Add Tegra210 based SFC driver ASoC: tegra: Add Tegra210 based AMX driver ASoC: tegra: Add Tegra210 based ADX driver ASoC: tegra: Add Tegra210 based Mixer driver arm64: defconfig: Enable few Tegra210 based AHUB drivers arm64: tegra: Add few AHUB devices for Tegra210 and later arm64: tegra: Extend APE audio support on Jetson platforms .../bindings/sound/nvidia,tegra210-adx.yaml | 76 + .../bindings/sound/nvidia,tegra210-ahub.yaml | 20 + .../bindings/sound/nvidia,tegra210-amx.yaml | 76 + .../bindings/sound/nvidia,tegra210-mixer.yaml | 74 + .../bindings/sound/nvidia,tegra210-mvc.yaml | 76 + .../bindings/sound/nvidia,tegra210-sfc.yaml | 73 + arch/arm64/boot/dts/nvidia/tegra186-p2771-0000.dts | 1554 ++++++++- arch/arm64/boot/dts/nvidia/tegra186.dtsi | 120 + arch/arm64/boot/dts/nvidia/tegra194-p2972-0000.dts | 1493 +++++++- .../arm64/boot/dts/nvidia/tegra194-p3509-0000.dtsi | 1520 ++++++++- arch/arm64/boot/dts/nvidia/tegra194.dtsi | 116 + arch/arm64/boot/dts/nvidia/tegra210-p2371-2180.dts | 876 +++++ arch/arm64/boot/dts/nvidia/tegra210-p3450-0000.dts | 876 +++++ arch/arm64/boot/dts/nvidia/tegra210.dtsi | 77 + arch/arm64/configs/defconfig | 5 + include/sound/simple_card_utils.h | 2 +- sound/soc/generic/audio-graph-card.c | 4 +- sound/soc/soc-pcm.c | 4 + sound/soc/tegra/Kconfig | 48 + sound/soc/tegra/Makefile | 10 + sound/soc/tegra/tegra210_adx.c | 531 +++ sound/soc/tegra/tegra210_adx.h | 72 + sound/soc/tegra/tegra210_ahub.c | 511 ++- sound/soc/tegra/tegra210_amx.c | 600 ++++ sound/soc/tegra/tegra210_amx.h | 93 + sound/soc/tegra/tegra210_mixer.c | 674 ++++ sound/soc/tegra/tegra210_mixer.h | 100 + sound/soc/tegra/tegra210_mvc.c | 645 ++++ sound/soc/tegra/tegra210_mvc.h | 117 + sound/soc/tegra/tegra210_sfc.c | 3549 ++++++++++++++++++++ sound/soc/tegra/tegra210_sfc.h | 78 + 31 files changed, 13647 insertions(+), 423 deletions(-) create mode 100644 Documentation/devicetree/bindings/sound/nvidia,tegra210-adx.yaml create mode 100644 Documentation/devicetree/bindings/sound/nvidia,tegra210-amx.yaml create mode 100644 Documentation/devicetree/bindings/sound/nvidia,tegra210-mixer.yaml create mode 100644 Documentation/devicetree/bindings/sound/nvidia,tegra210-mvc.yaml create mode 100644 Documentation/devicetree/bindings/sound/nvidia,tegra210-sfc.yaml create mode 100644 sound/soc/tegra/tegra210_adx.c create mode 100644 sound/soc/tegra/tegra210_adx.h create mode 100644 sound/soc/tegra/tegra210_amx.c create mode 100644 sound/soc/tegra/tegra210_amx.h create mode 100644 sound/soc/tegra/tegra210_mixer.c create mode 100644 sound/soc/tegra/tegra210_mixer.h create mode 100644 sound/soc/tegra/tegra210_mvc.c create mode 100644 sound/soc/tegra/tegra210_mvc.h create mode 100644 sound/soc/tegra/tegra210_sfc.c create mode 100644 sound/soc/tegra/tegra210_sfc.h -- 2.7.4
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Wolfram Sang authored
We have a dedicated pointer for that, so use it. Much easier to read and less computation involved. Signed-off-by:
Wolfram Sang <wsa+renesas@sang-engineering.com> Link: https://lore.kernel.org/r/20210918213553.14514-2-wsa+renesas@sang-engineering.comSigned-off-by:
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Peter Ujfalusi authored
As part of the effort to remove our old APIs based on outdated terminology update the Intel board drivers to use modern terminology. Signed-off-by:
Peter Ujfalusi <peter.ujfalusi@linux.intel.com> Reviewed-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com Link: https://lore.kernel.org/r/20210920065508.7854-1-peter.ujfalusi@linux.intel.comSigned-off-by:
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Krzysztof Kozlowski authored
Correct several errors in rt5682s dtschema: 1. The examples should be under "examples": 'example' is not one of ['$id', '$schema', 'title', 'description', 'examples', ... 2. Missing type for vendor properties 3. clock-names should be an array: properties:clock-names:items: {'const': 'mclk'} is not of type 'array' 4. Example DTS should include headers: [scripts/Makefile.lib:386: Documentation/devicetree/bindings/sound/realtek,rt5682s.example.dt.yaml] Error 1 5. Node name in example DTS misses unit address and does not match DT convention (generic name): Warning (reg_format): /example-0/rt5682s:reg: property has invalid length (4 bytes) (#address-cells == 1, #size-cells == 1) 6. Node address should be in size-cells:0 block in example DTS: Warning (reg_format): /example-0/codec@1a:reg: property has invalid length (4 bytes) (#address-cells == 1, #size-cells == 1) Fixes: 50159fdb ("ASoC: dt-bindings: rt5682s: add bindings for rt5682s") Signed-off-by:Krzysztof Kozlowski <krzysztof.kozlowski@canonical.com> Link: https://lore.kernel.org/r/20210920112106.140918-1-krzysztof.kozlowski@canonical.comSigned-off-by:
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Mark Brown authored
As part of moving to remove the old style defines for the bus clocks update the ab8500 driver to use more modern terminology for clocking. Signed-off-by:
Linus Walleij <linus.walleij@linaro.org> Link: https://lore.kernel.org/r/20210916141335.43818-1-broonie@kernel.orgSigned-off-by:
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Peter Ujfalusi authored
sof_get_ops() is not used and the struct sof_ops_table is only used by that macro. Signed-off-by:
Guennadi Liakhovetski <guennadi.liakhovetski@linux.intel.com> Reviewed-by:
Kai Vehmanen <kai.vehmanen@linux.intel.com> Reviewed-by:
Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com> Link: https://lore.kernel.org/r/20210920064156.4763-1-peter.ujfalusi@linux.intel.comSigned-off-by:
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Jiapeng Chong authored
This symbol is not used outside of rt5682s.c, so marks it static. Fix the following sparse warning: sound/soc/codecs/rt5682s.c:2848:39: warning: symbol 'rt5682s_soc_component_dev' was not declared. Should it be static? sound/soc/codecs/rt5682s.c:2842:30: warning: symbol 'rt5682s_aif2_dai_ops' was not declared. Should it be static? Reported-by:
Abaci Robot <abaci@linux.alibaba.com> Signed-off-by:
Jiapeng Chong <jiapeng.chong@linux.alibaba.com> Link: https://lore.kernel.org/r/1631955726-77693-1-git-send-email-jiapeng.chong@linux.alibaba.comSigned-off-by:
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Sameer Pujar authored
The Mixer supports mixing of up to ten 7.1 audio input streams and generate five outputs (each of which can be any combination of the ten input streams) This patch registers Mixer driver with ASoC framework. The component driver exposes DAPM widgets, routes and kcontrols for the device. The DAI driver exposes Mixer interfaces, which can be used to connect different components in the ASoC layer. Makefile and Kconfig support is added to allow build the driver. It can be enabled in the DT via "nvidia,tegra210-amixer" compatible binding. Signed-off-by:
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Sameer Pujar authored
The Audio Demultiplexer (ADX) block takes an input stream with up to 16 channels and demultiplexes it into four output streams of up to 16 channels each. A byte RAM helps to form output frames by any combination of bytes from the input frame. Its design is identical to that of byte RAM in the AMX except that the data flow direction is reversed. This patch registers ADX driver with ASoC framework. The component driver exposes DAPM widgets, routes and kcontrols for the device. The DAI driver exposes ADX interfaces, which can be used to connect different components in the ASoC layer. Makefile and Kconfig support is added to allow build the driver. It can be enabled in the DT via "nvidia,tegra210-adx" compatible binding. Signed-off-by:
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Sameer Pujar authored
The Audio Multiplexer (AMX) block can multiplex up to four input streams each of which can have maximum 16 channels and generate an output stream with maximum 16 channels. A byte RAM helps to form an output frame by any combination of bytes from the input frames. This patch registers AMX driver with ASoC framework. The component driver exposes DAPM widgets, routes and kcontrols for the device. The DAI driver exposes AMX interfaces, which can be used to connect different components in the ASoC layer. Makefile and Kconfig support is added to allow build the driver. It can be enabled in the DT via "nvidia,tegra210-amx" for Tegra210 and Tegra186. For Tegra194 and later, "nvidia,tegra194-amx" can be used. Signed-off-by:
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Sameer Pujar authored
The Sampling Frequency Converter (SFC) converts the sampling frequency of the input signal from one frequency to another. It supports sampling frequency conversions of streams of up to two channels (stereo). This patch registers SFC driver with ASoC framework. The component driver exposes DAPM widgets, routes and kcontrols for the device. The DAI driver exposes SFC interfaces, which can be used to connect different components in the ASoC layer. Makefile and Kconfig support is added to allow build the driver. It can be enabled in the DT via "nvidia,tegra210-sfc" compatible binding. Signed-off-by:
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Sameer Pujar authored
The Master Volume Control (MVC) provides gain or attenuation to a digital signal path. It can be used in input or output signal path for per-stream volume control or it can be used as master volume control. The MVC block has one input and one output. The input digital stream can be mono or multi-channel (up to 7.1 channels) stream. An independent mute control is also included in the MVC block. This patch registers MVC driver with ASoC framework. The component driver exposes DAPM widgets, routes and kcontrols for the device. The DAI driver exposes MVC interfaces, which can be used to connect different components in the ASoC layer. Makefile and Kconfig support is added to allow build the driver. It can be enabled in the DT via "nvidia,tegra210-mvc" compatible binding. Signed-off-by:
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Sameer Pujar authored
Add routing support for following modules of AHUB: * SFC (Sampling Frequency Converter) * MVC (Master Volume Control) * AMX (Audio Multiplexer) * ADX (Audio Demultiplexer) * Mixer These modules can be plugged into audio path as per the need using routing controls similar to the already existing routes to I/O modules such as I2S, DMIC and DSPK. Signed-off-by:
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Sameer Pujar authored
This patch adds YAML schema for DT bindings of few AHUB modules. These devices will be registered as ASoC components and bindings will be used on Tegra210 and later chips. The bindings for below mentioned modules are added: * SFC (Sampling Frequency Converter) * MVC (Master Volume Control) * AMX (Audio Multiplexer) * ADX (Audio Demultiplexer) * Mixer Signed-off-by:
Rob Herring <robh@kernel.org> Link: https://lore.kernel.org/r/1631551342-25469-5-git-send-email-spujar@nvidia.comSigned-off-by:
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Sameer Pujar authored
When multiple components are connected back to back in an audio path, hw_param fixup may be required for CPU or Codec endpoint of BE<->BE DAI links. Currently fixup support is available for Codec and this commit adds similar feature for CPU endpoint of a BE<->BE link. For example a resampler component can be plugged into an audio path. [ FE -> BE1 -> ... -> resampler -> ... BEn ] The resampler DAI links can be: BEx (CPU) -> resampler input (Codec) resampler output (CPU) -> BEy (Codec) Thus input and output sample rate parameters for resampler can be fixed up as per the resample requirement. Signed-off-by:
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Sameer Pujar authored
The current limit of 128 is not sufficient when more components are added to the audio map on Tegra210 and later platforms. Thus it is resulting in probe failure. The requirement is of nearly ~200 DAI links. To give sufficient room for future additions the maximum limit is increased to 512 DAI links. This is a preparatory patch to add more components like resampler, mixer, multiplexers, demultiplexers and volume controllers to Tegra210 and later platforms. Signed-off-by:
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Sameer Pujar authored
In some cases, multiple FE components have the same BE component in their respective DPCM paths. One such example would be a mixer component, which can receive two or more inputs and sends a mixed output. In such cases, to avoid reconfiguration of already active DAI (mixer output DAI in this case), check the BE stream state to filter out the redundancy. In summary, allow connection of BE if the respective current stream state is either NEW or CLOSED. Signed-off-by:
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Peter Ujfalusi authored
Currently the try_module_get() and module_put() is not possible for compressed streams if the module_get_upon_open is set to 1 which means that\ the components are not protected in a same way as components when normal audio is used. SOF is setting module_get_upon_open to 1 for component drivers which works correctly for audio stream but when compressed stream is used then the module is not protected. Convert the compress open and free operation to mimic the steps of it's pcm counterpart to fix this issue. Signed-off-by:
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Peter Ujfalusi authored
The mark_module of the snd_soc_component is strict snd_pcm_substream type which prevents it to be used by compressed streams. Change the type to void* along with the snd_soc_component_module_get() and snd_soc_component_module_put() to allow the same mark to be used by compressed when it's module_get_upon_open is set to 1. Signed-off-by:
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- 17 Sep, 2021 6 commits
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Mark Brown authored
Merge series "ASoC: SOF: ipc: Small cleanups for message handler functions" from Peter Ujfalusi <peter.ujfalusi@linux.intel.com>: Hi, Rename the parameter for ipc_trace_message() to match it's content and use %#x" for hexadecimal prints in remaining places. Regards, Peter --- Peter Ujfalusi (2): ASoC: SOF: ipc: Clarify the parameter name for ipc_trace_message() ASoC: SOF: ipc: Print 0x prefix for errors in ipc_trace/stream_message() sound/soc/sof/ipc.c | 11 +++++------ 1 file changed, 5 insertions(+), 6 deletions(-) -- 2.33.0 -
Mark Brown authored
As part of moving to remove the old style defines for the bus clocks update the 88pm860x driver to use more modern terminology for clocking. Signed-off-by:
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Peter Ujfalusi authored
From the name sof_arch_ops one can not decipher that these ops are DSP architecture ops. Rename it to dsp_arch_ops and change also the macro to retrieve the DSP architecture specific ops as well. Signed-off-by:
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Peter Ujfalusi authored
If the snd_sof_dsp_send_msg() failed then we have already returned from sof_ipc_tx_message_unlocked() with the error message. There is no need to check if ret is really 0 after this and we can return directly the return value from tx_wait_done() At the same time make the remaining checks for error (ret) to match. Signed-off-by:
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Peter Ujfalusi authored
The dev_err() in ipc_trace_message() and ipc_stream_message() is missing the 0x prefix for the hexadecimal number when printed. Signed-off-by:
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Peter Ujfalusi authored
ipc_trace_message() receives the type not the ID. Use the same naming as the ipc_stream_message() function: msg_type to help the reader to follow the code. Signed-off-by:
Bard Liao <bard.liao@intel.com> Link: https://lore.kernel.org/r/20210917085823.27222-2-peter.ujfalusi@linux.intel.comSigned-off-by:
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- 16 Sep, 2021 1 commit
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Mark Brown authored
Merge series "ASoC: cs42l42: Implement Manual Type detection as fallback" from Vitaly Rodionov <vitalyr@opensource.cirrus.com>: For some headsets CS42L42 autodetect mode is not working correctly. They will be detected as unknown types or as headphones. According to the CS42L42 datasheet, if the headset autodetect failed, then the driver should switch to manual mode and perform a manual steps sequence. These steps were missing in the current driver code. This patch will add manual mode fallback steps in case autodetect failed. The default behavior is not affected, manual mode runs only when autodetect failed. Tested for regression with autodetect with all known headsets - no regression. Tested with all headsets customers reported as false detected: Gumdrop DropTech B1 - detected as headset OK HUAWEI AM115 - detected as headset OK UGREEN EP103 - detected as headset OK HONOR AM116 - detected as headset OK Stefan Binding (1): ASoC: cs42l42: Implement Manual Type detection as fallback sound/soc/codecs/cs42l42.c | 104 ++++++++++++++++++++++++++++++++----- sound/soc/codecs/cs42l42.h | 54 +++++++++++++++++++ 2 files changed, 146 insertions(+), 12 deletions(-) -- 2.25.1
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