- 26 Oct, 2020 40 commits
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Mark Brown authored
Merge series "ASoC: merge soc_pcm_hw_param() rollback and soc_pcm_hw_free()" from Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>: Hi Mark soc_pcm_hw_params() does rollback when failed (A), but, it is almost same as soc_pcm_hw_free(). static int soc_pcm_hw_params(xxx) { ... if (ret < 0) goto xxx_err; ... return ret; ^ component_err: | ... | interface_err: (A) ... | codec_err: | ... v return ret; } This kind of duplicated code can be a hotbed of bugs, thus, this patch-set share soc_pcm_hw_free() and rollback. Kuninori Morimoto (6): ASoC: soc.h: remove for_each_rtd_dais_rollback() ASoC: soc-pcm: move soc_pcm_hw_free() next to soc_pcm_hw_params() ASoC: soc-link: add mark for snd_soc_link_hw_params/free() ASoC: soc-component: add mark for snd_soc_pcm_component_hw_params/free() ASoC: soc-dai: add mark for snd_soc_dai_hw_params/free() ASoC: soc-pcm: add soc_pcm_hw_clean() and call it from soc_pcm_hw_params/free() include/sound/soc-component.h | 6 +- include/sound/soc-dai.h | 4 +- include/sound/soc-link.h | 3 +- include/sound/soc.h | 7 +- sound/soc/soc-component.c | 19 ++--- sound/soc/soc-dai.c | 13 +++- sound/soc/soc-dapm.c | 4 +- sound/soc/soc-link.c | 12 +++- sound/soc/soc-pcm.c | 131 ++++++++++++++-------------------- 9 files changed, 97 insertions(+), 102 deletions(-) -- 2.25.1 -
Mark Brown authored
Merge series "ASoC: qcom: add support for QRB5165 RB5 machine" from Srinivas Kandagatla <srinivas.kandagatla@linaro.org>: This patchset adds support to Qualcomm Robotics RB5 Development Kit based on QRB5165 Robotics SoC. This board has 2 WSA881X smart speakers with onboard DMIC connected to internal LPASS codec via WSA and VA macros respectively. Srinivas Kandagatla (2): ASoC: qcom: dt-bindings: Add SM8250 sound card bindings ASoC: qcom: sm8250: add sound card qrb5165-rb5 support .../bindings/sound/qcom,sm8250.yaml | 161 +++++++++++++ sound/soc/qcom/Kconfig | 11 + sound/soc/qcom/Makefile | 2 + sound/soc/qcom/sm8250.c | 228 ++++++++++++++++++ 4 files changed, 402 insertions(+) create mode 100644 Documentation/devicetree/bindings/sound/qcom,sm8250.yaml create mode 100644 sound/soc/qcom/sm8250.c -- 2.21.0
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Mark Brown authored
Merge series "use semicolons rather than commas to separate statements" from Julia Lawall <Julia.Lawall@inria.fr>: These patches replace commas by semicolons. This was done using the Coccinelle semantic patch (http://coccinelle.lip6.fr/) shown below. This semantic patch ensures that commas inside for loop headers will not be transformed. It also doesn't touch macro definitions. Coccinelle ensures that braces are added as needed when a single-statement branch turns into a multi-statement one. This semantic patch has a few false positives, for variable delcarations such as: LIST_HEAD(x), *y; The semantic patch could be improved to avoid these, but for the moment they have been removed manually (2 occurrences). // <smpl> @initialize:ocaml@ @@ let infunction p = (* avoid macros *) (List.hd p).current_element <> "something_else" let combined p1 p2 = (List.hd p1).line_end = (List.hd p2).line || (((List.hd p1).line_end < (List.hd p2).line) && ((List.hd p1).col < (List.hd p2).col)) @bad@ statement S; declaration d; position p; @@ S@p d // special cases where newlines are needed (hope for no more than 5) @@ expression e1,e2; statement S; position p != bad.p; position p1; position p2 : script:ocaml(p1) { infunction p1 && combined p1 p2 }; @@ - e1@p1,@S@p e2@p2; + e1; e2; @@ expression e1,e2; statement S; position p != bad.p; position p1; position p2 : script:ocaml(p1) { infunction p1 && combined p1 p2 }; @@ - e1@p1,@S@p e2@p2; + e1; e2; @@ expression e1,e2; statement S; position p != bad.p; position p1; position p2 : script:ocaml(p1) { infunction p1 && combined p1 p2 }; @@ - e1@p1,@S@p e2@p2; + e1; e2; @@ expression e1,e2; statement S; position p != bad.p; position p1; position p2 : script:ocaml(p1) { infunction p1 && combined p1 p2 }; @@ - e1@p1,@S@p e2@p2; + e1; e2; @@ expression e1,e2; statement S; position p != bad.p; position p1; position p2 : script:ocaml(p1) { infunction p1 && combined p1 p2 }; @@ - e1@p1,@S@p e2@p2; + e1; e2; @r@ expression e1,e2; statement S; position p != bad.p; @@ e1 ,@S@p e2; @@ expression e1,e2; position p1; position p2 : script:ocaml(p1) { infunction p1 && not(combined p1 p2) }; statement S; position r.p; @@ e1@p1 -,@S@p +; e2@p2 ... when any // </smpl> --- sound/firewire/fireworks/fireworks_pcm.c | 2 +- sound/pci/hda/patch_ca0132.c | 2 +- sound/pci/hda/patch_hdmi.c | 2 +- sound/soc/codecs/madera.c | 4 ++-- sound/soc/codecs/wm8350.c | 3 ++- sound/soc/intel/boards/bytcr_rt5651.c | 2 +- sound/soc/samsung/snow.c | 2 +- sound/soc/soc-dapm.c | 2 +- sound/soc/sof/intel/hda-dsp.c | 2 +- 9 files changed, 11 insertions(+), 10 deletions(-)
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Mark Brown authored
Merge series "ASoC: sun8i-codec: support for AIF2 and AIF3" from Samuel Holland <samuel@sholland.org>: This series adds support the other two AIFs present in the sun8i codec, which can be used for codec2codec DAI links. This series first fills out the DAI driver, removing assumptions that were made for AIF1 (16 bits, 2 channels, certain clock inversions). Some new logic is required to handle 3 DAIs and the ADC/DAC sharing the same clock. Finally, it adds the new DAIs, and hooks them up with DAPM widgets and routes per the hardware topology. To minimize the number of patches in this series, related device tree patches (increasing #sound-dai-cells, adding new DAI links) will be sent separately. Changes from v1: - Patches 1-8 from v1 (DAPM changes) were merged - Prefixed AIF constants with "SUN8I_CODEC_" [1, 7, 10, 16, 17] - Renamed variables in sun8i_codec_set_fmt for clarity [3] - Update sysclk->sysclk_rate if later calls to hw_params change the sample rate (thanks Chen-Yu for reminding me of this) [11] - Select COMMON_CLK for clk_set_rate_exclusive [12] - Add comments and hopefully clarify the clock protection logic [12] - Make the error message more concise and put it on one line [12] - Drop the "reg" variable holding SUN8I_AIF_CLK_CTRL(dai->id) [15] - Rename "div_reg" to "clk_reg" and adjust comments for clarity [17] - Improve the AIF2/AIF3 rate mismatch error message [17] - Minor grammar/wording fixes in commit messages [2, 4, 7, 8, 16, 17] - Added Maxime's Acked-by: [1-2, 4-9, 11, 13-14, 16] Samuel Holland (17): ASoC: sun8i-codec: Prepare to extend the DAI driver ASoC: sun8i-codec: Program DAI format before clock inversion ASoC: sun8i-codec: Enable all supported clock inversions ASoC: sun8i-codec: Use the provided word size ASoC: sun8i-codec: Round up the LRCK divisor ASoC: sun8i-codec: Correct the BCLK divisor calculation ASoC: sun8i-codec: Support the TDM slot binding ASoC: sun8i-codec: Enforce symmetric DAI parameters ASoC: sun8i-codec: Enable all supported sample rates ASoC: sun8i-codec: Automatically set the system sample rate ASoC: sun8i-codec: Constrain to compatible sample rates ASoC: sun8i-codec: Protect the clock rate while streams are open ASoC: sun8i-codec: Require an exact BCLK divisor match ASoC: sun8i-codec: Enable all supported PCM formats ASoC: sun8i-codec: Generalize AIF clock control ASoC: sun8i-codec: Add the AIF2 DAI, widgets, and routes ASoC: sun8i-codec: Add the AIF3 DAI, widgets, and routes sound/soc/sunxi/Kconfig | 1 + sound/soc/sunxi/sun8i-codec.c | 834 ++++++++++++++++++++++++++++------ 2 files changed, 699 insertions(+), 136 deletions(-) -- 2.26.2 -
Mark Brown authored
Merge series "dt-bindings: stm32: convert audio dfsdm to json-schema" from Olivier Moysan <olivier.moysan@st.com>: Some audio properties documented in st,stm32-adfsdm.txt are already documented in Documentation/devicetree/bindings/iio/adc/st,stm32-dfsdm-adc.yaml bindings. Move remaining properties from st,stm32-adfsdm.txt to st,stm32-dfsdm-adc.yaml, and remove st,stm32-adfsdm.txt. Changes in v2: - Complete st,stm32-dfsdm-adc.yaml rather than converting st,stm32-adfsdm.txt Olivier Moysan (2): dt-bindings: stm32: dfsdm: update audio properties dt-bindings: stm32: dfsdm: remove stm32-adfsdm.txt binding .../bindings/iio/adc/st,stm32-dfsdm-adc.yaml | 7 ++- .../bindings/sound/st,stm32-adfsdm.txt | 63 ------------------- 2 files changed, 5 insertions(+), 65 deletions(-) delete mode 100644 Documentation/devicetree/bindings/sound/st,stm32-adfsdm.txt -- 2.17.1
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Shengjiu Wang authored
Add 88200Hz and 176400Hz sample rates support for TX. Add 88200Hz, 176400Hz, 192000Hz sample rates support for RX. Signed-off-by:
Viorel Suman <viorel.suman@nxp.com> Signed-off-by:
Shengjiu Wang <shengjiu.wang@nxp.com> Acked-by:
Nicolin Chen <nicoleotsuka@gmail.com> Reviewed-by:
Daniel Baluta <daniel.baluta@nxp.com> Link: https://lore.kernel.org/r/1602557360-18795-1-git-send-email-shengjiu.wang@nxp.comSigned-off-by:
Mark Brown <broonie@kernel.org>
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Shengjiu Wang authored
On i.MX8QM, there are separate interrupts for TX and RX. As the EDMA can't be configured to swing back to first FIFO after writing the second FIFO, so we need to force the burst size to be 2 on i.MX8QM. And EDMA don't support to shift the data from S24_LE to S16_LE, so the supported TX format is also different on i.MX8QM. Signed-off-by:
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Shengjiu Wang authored
Add new compatible string "fsl,imx8qm-spdif" for supporting spdif module on i.MX8QM. Signed-off-by:
Rob Herring <robh@kernel.org> Link: https://lore.kernel.org/r/1602739728-4433-1-git-send-email-shengjiu.wang@nxp.comSigned-off-by:
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Vijendar Mukunda authored
ACP supports different audio configurations other than I2S. This patch will fix acp driver probe failure for other audio configurations. Tested-by:
Prike Liang <Prike.Liang@amd.com> Reviewed-by:
Alex Deucher <alexander.deucher@amd.com> Signed-off-by:
Vijendar Mukunda <Vijendar.Mukunda@amd.com> Link: https://lore.kernel.org/r/1603476441-3506-1-git-send-email-Vijendar.Mukunda@amd.comSigned-off-by:
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Richard Fitzgerald authored
Replace the two-step copy-and-convert in wm5102_out_comp_coeff_put() with get_unaligned_be16(). Apart from looking nicer, it avoids this sparse warning: wm5102.c:687:35: sparse: sparse: cast to restricted __be16 Signed-off-by:
Richard Fitzgerald <rf@opensource.cirrus.com> Link: https://lore.kernel.org/r/20201015102703.24622-1-rf@opensource.cirrus.comSigned-off-by:
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Olivier Moysan authored
Convert the STM32 SAI bindings to DT schema format using json-schema. Signed-off-by:
Olivier Moysan <olivier.moysan@st.com> Reviewed-by:
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Alexandru Ardelean authored
The change removes the platform_data include/definition. It only contains some values for the MICBIAS. These are moved into 'dt-bindings/sound/adi,adau1977.h' so that they can be used inside device-trees. When moving then, they need to be converted to pre-compiler defines, so that the DT compiler can understand them. The driver then, also needs to include the new 'dt-bindings/sound/adi,adau1977.h' file. Signed-off-by:
Alexandru Ardelean <alexandru.ardelean@analog.com> Link: https://lore.kernel.org/r/20201019105313.24862-1-alexandru.ardelean@analog.comSigned-off-by:
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Codrin Ciubotariu authored
The TI PCM5102A codec driver can be used with the generic sound card drivers, so it should be selectable. For example, with the addition of #sound-dai-cells = <0> property in DT, it can be used with simple/graph card drivers. Signed-off-by:
Codrin Ciubotariu <codrin.ciubotariu@microchip.com> Link: https://lore.kernel.org/r/20201012141911.3150996-1-codrin.ciubotariu@microchip.comSigned-off-by:
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Lars-Peter Clausen authored
All platforms that use the jz4740-i2s driver have been switched to devicetree for a while now and the assignment of the DMA peripheral ID is done in the devicetree. It is no longer necessary to manually assign the peripheral ID in the driver, so remove that. The DMA driver does not even look at the value assigned in the driver anymore and always uses the value provided by the devicetree. Signed-off-by:
Lars-Peter Clausen <lars@metafoo.de> Reviewed-by:
Paul Cercueil <paul@crapouillou.net> Link: https://lore.kernel.org/r/20201023092346.5777-1-lars@metafoo.deSigned-off-by:
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Srinivas Kandagatla authored
Add support to Qualcomm Robotics RB5 Development Kit based on QRB5165 Robotics SoC. This board has 2 WSA881X smart speakers with onboard DMIC connected to internal LPASS codec via WSA and VA macros respectively. Signed-off-by:
Srinivas Kandagatla <srinivas.kandagatla@linaro.org> Link: https://lore.kernel.org/r/20201026170947.10567-3-srinivas.kandagatla@linaro.orgSigned-off-by:
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Srinivas Kandagatla authored
This patch adds bindings required for SM8250 based soundcards for example Qualcomm Robotics RB5 Development Kit which makes use of ADSP and Internal LPASS codec. Signed-off-by:
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Olivier Moysan authored
Device tree audio configuration for STM32 DFSDM is already covered in the following binding: Documentation/devicetree/bindings/iio/adc/st,stm32-dfsdm-adc.yaml Remove stm32-adfsdm.txt obsolete binding. Signed-off-by:
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Olivier Moysan authored
- Add missing compatible property in audio node. - Remove obsolete "st,stm32-dfsdm-pdm" compatible. - Remove useless comment in adc io-channels description. Signed-off-by:
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Kuninori Morimoto authored
soc_pcm_hw_params() does rollback when failed (A), but, it is almost same as soc_pcm_hw_free(). static int soc_pcm_hw_params(xxx) { ... if (ret < 0) goto xxx_err; ... return ret; ^ component_err: | ... | interface_err: (A) ... | codec_err: | ... v return ret; } The difference is soc_pcm_hw_free() is for all dai/component/substream, rollback is for succeeded part only. This kind of duplicated code can be a hotbed of bugs, thus, we want to share soc_pcm_hw_free() and rollback. Now, soc_pcm_hw_params/free() are handling 1) snd_soc_link_hw_params/free() 2) snd_soc_pcm_component_hw_params/free() 3) snd_soc_dai_hw_params/free() Now, 1) to 3) are handled. This patch adds new soc_pcm_hw_clean() and call it from soc_pcm_hw_params() as rollback, and from soc_pcm_hw_free() as normal close handler. Other difference is that soc_pcm_hw_free() handles digital mute if it was last user. Rollback also handles it by this patch. Signed-off-by:Kuninori Morimoto <kuninori.morimoto.gx@renesas.com> Link: https://lore.kernel.org/r/87h7rhgqab.wl-kuninori.morimoto.gx@renesas.comSigned-off-by:
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Kuninori Morimoto authored
soc_pcm_hw_params() does rollback when failed (A), but, it is almost same as soc_pcm_hw_free(). static int soc_pcm_hw_params(xxx) { ... if (ret < 0) goto xxx_err; ... return ret; ^ component_err: | ... | interface_err: (A) ... | codec_err: | ... v return ret; } The difference is soc_pcm_hw_free() is for all dai/component/substream, rollback is for succeeded part only. This kind of duplicated code can be a hotbed of bugs, thus, we want to share soc_pcm_hw_free() and rollback. Now, soc_pcm_hw_params/free() are handling 1) snd_soc_link_hw_params/free() 2) snd_soc_pcm_component_hw_params/free() => 3) snd_soc_dai_hw_params/free() This patch is for 3) snd_soc_dai_hw_params/free(). The idea of having bit-flag or counter is not enough for this purpose. For example if one DAI is used for 2xPlaybacks for some reasons, and if 1st Playback was succeeded but 2nd Playback was failed, 2nd Playback rollback doesn't need to call shutdown. But it has succeeded bit-flag or counter via 1st Playback, thus, 2nd Playback rollback will call unneeded shutdown. And 1st Playback's necessary shutdown will not be called, because bit-flag or counter was cleared by wrong 2nd Playback rollback. To avoid such case, this patch marks substream pointer when hw_params() was succeeded. If rollback needed, it will check rollback flag and marked substream pointer. One note here is that it cares *previous* hw_params() only now, but we might want to check *whole* marked substream in the future. This patch is using macro named "push/pop", so that it can be easily update. Signed-off-by: -
Kuninori Morimoto authored
soc_pcm_hw_params() does rollback when failed (A), but, it is almost same as soc_pcm_hw_free(). static int soc_pcm_hw_params(xxx) { ... if (ret < 0) goto xxx_err; ... return ret; ^ component_err: | ... | interface_err: (A) ... | codec_err: | ... v return ret; } The difference is soc_pcm_hw_free() is for all dai/component/substream, rollback is for succeeded part only. This kind of duplicated code can be a hotbed of bugs, thus, we want to share soc_pcm_hw_free() and rollback. Now, soc_pcm_hw_params/free() are handling 1) snd_soc_link_hw_params/free() => 2) snd_soc_pcm_component_hw_params/free() 3) snd_soc_dai_hw_params/free() This patch is for 2) snd_soc_pcm_component_hw_params/free(). The idea of having bit-flag or counter is not enough for this purpose. For example if one DAI is used for 2xPlaybacks for some reasons, and if 1st Playback was succeeded but 2nd Playback was failed, 2nd Playback rollback doesn't need to call shutdown. But it has succeeded bit-flag or counter via 1st Playback, thus, 2nd Playback rollback will call unneeded shutdown. And 1st Playback's necessary shutdown will not be called, because bit-flag or counter was cleared by wrong 2nd Playback rollback. To avoid such case, this patch marks substream pointer when hw_params() was succeeded. If rollback needed, it will check rollback flag and marked substream pointer. One note here is that it cares *previous* hw_params() only now, but we might want to check *whole* marked substream in the future. This patch is using macro named "push/pop", so that it can be easily update. Signed-off-by: -
Kuninori Morimoto authored
soc_pcm_hw_params() does rollback when failed (A), but, it is almost same as soc_pcm_hw_free(). static int soc_pcm_hw_params(xxx) { ... if (ret < 0) goto xxx_err; ... return ret; ^ component_err: | ... | interface_err: (A) ... | codec_err: | ... v return ret; } The difference is soc_pcm_hw_free() is for all dai/component/substream, rollback is for succeeded part only. This kind of duplicated code can be a hotbed of bugs, thus, we want to share soc_pcm_hw_free() and rollback. Now, soc_pcm_hw_params/free() are handling => 1) snd_soc_link_hw_params/free() 2) snd_soc_pcm_component_hw_params/free() 3) snd_soc_dai_hw_params/free() This patch is for 1) snd_soc_link_hw_params/free(). The idea of having bit-flag or counter is not enough for this purpose. For example if one DAI is used for 2xPlaybacks for some reasons, and if 1st Playback was succeeded but 2nd Playback was failed, 2nd Playback rollback doesn't need to call shutdown. But it has succeeded bit-flag or counter via 1st Playback, thus, 2nd Playback rollback will call unneeded shutdown. And 1st Playback's necessary shutdown will not be called, because bit-flag or counter was cleared by wrong 2nd Playback rollback. To avoid such case, this patch marks substream pointer when hw_params() was succeeded. If rollback needed, it will check rollback flag and marked substream pointer. One note here ist that it cares *previous* hw_params() only now, but we might want to check *whole* marked substream in the future. This patch is using macro named "push/pop", so that it can be easily update. Signed-off-by: -
Kuninori Morimoto authored
This patch moves soc_pcm_hw_free() next to soc_pcm_hw_params(). This is prepare for soc_pcm_hw_params() cleanup Signed-off-by:
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Kuninori Morimoto authored
commit 140a4532 ("ASoC: soc-pcm: add soc_pcm_clean() and call it from soc_pcm_open/close()") uses soc_pcm_clean() and then for_each_rtd_dais_rollback() is no longer used. This patch removes it. Signed-off-by:
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Julia Lawall authored
Replace commas with semicolons. What is done is essentially described by the following Coccinelle semantic patch (http://coccinelle.lip6.fr/): // <smpl> @@ expression e1,e2; @@ e1 -, +; e2 ... when any // </smpl> Signed-off-by:
Julia Lawall <Julia.Lawall@inria.fr> Link: https://lore.kernel.org/r/1602407979-29038-9-git-send-email-Julia.Lawall@inria.frSigned-off-by:
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Julia Lawall authored
Replace commas with semicolons. What is done is essentially described by the following Coccinelle semantic patch (http://coccinelle.lip6.fr/): // <smpl> @@ expression e1,e2; @@ e1 -, +; e2 ... when any // </smpl> Signed-off-by:
Charles Keepax <ckeepax@opensource.cirrus.com> Link: https://lore.kernel.org/r/1602407979-29038-8-git-send-email-Julia.Lawall@inria.frSigned-off-by:
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Julia Lawall authored
Replace commas with semicolons. What is done is essentially described by the following Coccinelle semantic patch (http://coccinelle.lip6.fr/): // <smpl> @@ expression e1,e2; @@ e1 -, +; e2 ... when any // </smpl> Signed-off-by:
Krzysztof Kozlowski <krzk@kernel.org> Link: https://lore.kernel.org/r/1602407979-29038-7-git-send-email-Julia.Lawall@inria.frSigned-off-by:
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Julia Lawall authored
Replace commas with semicolons. What is done is essentially described by the following Coccinelle semantic patch (http://coccinelle.lip6.fr/): // <smpl> @@ expression e1,e2; @@ e1 -, +; e2 ... when any // </smpl> Signed-off-by:
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Julia Lawall authored
Replace commas with semicolons. What is done is essentially described by the following Coccinelle semantic patch (http://coccinelle.lip6.fr/): // <smpl> @@ expression e1,e2; @@ e1 -, +; e2 ... when any // </smpl> Signed-off-by:
Cezary Rojewski <cezary.rojewski@intel.com> Link: https://lore.kernel.org/r/1602407979-29038-4-git-send-email-Julia.Lawall@inria.frSigned-off-by:
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Julia Lawall authored
Replace commas with semicolons. What is done is essentially described by the following Coccinelle semantic patch (http://coccinelle.lip6.fr/): // <smpl> @@ expression e1,e2; @@ e1 -, +; e2 ... when any // </smpl> Signed-off-by:
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Samuel Holland authored
AIF3 has some differences from AIF1 and AIF2: - It supports one channel only - It supports master mode only - It is not directly connected to any of the mixers; instead all audio goes through a mux with AIF2. - It does not have its own clock dividers; instead it reuses AIF2 BCLK and LRCK. This means that when both AIF2 and AIF3 are active, they must use the same sample rate and total frame width. Since AIF2 and AIF3 are only used for codec2codec DAI links, constraints are not applicable here; the only thing we can do when the rates don't match is report an error. Make the necessary adjustments to support this AIF. Signed-off-by:
Samuel Holland <samuel@sholland.org> Link: https://lore.kernel.org/r/20201014061941.4306-18-samuel@sholland.orgSigned-off-by:
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Samuel Holland authored
This adds support for AIF2, which is stereo and has fullly independent clocking capability, making it very similar to AIF1. Acked-by:
Maxime Ripard <mripard@kernel.org> Signed-off-by:
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Samuel Holland authored
The AIF clock control register has the same layout for all three AIFs. The only difference between them is that AIF3 is missing some fields. We can reuse the same register field definitions for all three registers, and use the DAI ID to select the correct register address. Signed-off-by:
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Samuel Holland authored
Now that the DAI clock setup is correct for all hardware-supported PCM formats, we can enable them in the driver. With the appropriate support in the CPU DAI driver, this allows userspace to access the additional formats. Since this codec is connected to the CPU via a DAI, not directly, we do not care if the CPU DAI is using 3-byte or 4-byte formats, so we can support them both. Acked-by:
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Samuel Holland authored
Now that we guarantee that SYSCLK is running at the optimal rate when hw_params succeeds, and that it will continue running at that rate, SYSCLK will always be an integer multiple of BCLK. So we can always pick the exact divider, not just the closest divider. Acked-by:
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Samuel Holland authored
The codec's clock input is shared among all AIFs, and shared with other audio-related hardware in the SoC, including I2S and SPDIF controllers. To ensure sample rates selected by userspace or by codec2codec DAI links are maintained, the clock rate must be protected while it is in use. Signed-off-by:
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Samuel Holland authored
While another stream is active, only allow userspace to use sample rates that are compatible with the current SYSCLK frequency. This ensures the actual sample rate will always match what is given in hw_params. Acked-by:
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Samuel Holland authored
The sun8i codec has three clock/sample rate domains: - The AIF1 domain, with a sample rate equal to AIF1 LRCK - The AIF2 domain, with a sample rate equal to AIF2 LRCK - The SYSCLK domain, containing the ADC, DAC, and effects (AGC/DRC), with a sample rate given by a divisor from SYSCLK. The divisor is controlled by the AIF1_FS or AIF2_FS field in SYS_SR_CTRL, depending on if SYSCLK's source is AIF1CLK or AIF2CLK, respectively. The exact sample rate depends on if SYSCLK is running at 22.6 MHz or 24.6 MHz. When an AIF (currently only AIF1) is active, the ADC and DAC should run at that sample rate to avoid artifacting. Sample rate conversion is only available when multiple AIFs are active and are routed to each other; this means the sample rate conversion hardware usually cannot be used. Only attach the event hook to the channel 0 AIF widgets, since we only need one event when a DAI stream starts or stops. Channel 0 is always brought up with a DAI stream, regardless of the number of channels in the stream. The ADC and DAC (along with their effects blocks) can be used even if no AIFs are in use. In that case, we should select an appropriate sample rate divisor, instead of keeping the last-used AIF sample rate. 44.1/48 kHz was chosen to balance audio quality and power consumption. Since the sample rate is tied to active AIF paths, disabling pmdown_time allows switching to the optimal sample rate immediately, instead of after a 5 second delay. Signed-off-by:
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Samuel Holland authored
The system sample rate programmed into the hardware is really a clock divider from SYSCLK to the ADC and DAC. Since we support two SYSCLK frequencies, we can use all sample rates corresponding to one of those frequencies divided by any available divisor. This commit enables support for those sample rates. It also stops advertising support for a 64 kHz sample rate, which is not supported. Acked-by:
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Samuel Holland authored
The AIFs have a single register controlling DAI parameters in both directions, including BCLK/LRCK divisor and word size. The DAIs produce only noise or silence if any of these parameters is wrong. Therefore, we need to enforce symmetry for these parameters, so starting a new substream will not break an existing substream. Acked-by:
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