October 2020 - Alsa-devel - mailman.alsa-project.org

[PATCH] ASoC: jz4740-i2s: Remove manual DMA peripheral ID assignment
by Lars-Peter Clausen 27 Oct '20

27 Oct '20

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(a)metafoo.de> --- sound/soc/jz4740/jz4740-i2s.c | 5 ----- 1 file changed, 5 deletions(-) diff --git a/sound/soc/jz4740/jz4740-i2s.c b/sound/soc/jz4740/jz4740-i2s.c index c7bd20104b20..e8bc7ca5ee5e 100644 --- a/sound/soc/jz4740/jz4740-i2s.c +++ b/sound/soc/jz4740/jz4740-i2s.c @@ -26,9 +26,6 @@ #include "jz4740-i2s.h" -#define JZ4740_DMA_TYPE_AIC_TRANSMIT 24 -#define JZ4740_DMA_TYPE_AIC_RECEIVE 25 - #define JZ_REG_AIC_CONF 0x00 #define JZ_REG_AIC_CTRL 0x04 #define JZ_REG_AIC_I2S_FMT 0x10 @@ -377,13 +374,11 @@ static void jz4740_i2c_init_pcm_config(struct jz4740_i2s *i2s) /* Playback */ dma_data = &i2s->playback_dma_data; dma_data->maxburst = 16; - dma_data->slave_id = JZ4740_DMA_TYPE_AIC_TRANSMIT; dma_data->addr = i2s->phys_base + JZ_REG_AIC_FIFO; /* Capture */ dma_data = &i2s->capture_dma_data; dma_data->maxburst = 16; - dma_data->slave_id = JZ4740_DMA_TYPE_AIC_RECEIVE; dma_data->addr = i2s->phys_base + JZ_REG_AIC_FIFO; } -- 2.20.1

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[PATCH] ASoC: wm5102: Use get_unaligned_be16() for dac_comp_coeff
by Richard Fitzgerald 27 Oct '20

27 Oct '20

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(a)opensource.cirrus.com> --- sound/soc/codecs/wm5102.c | 4 +--- 1 file changed, 1 insertion(+), 3 deletions(-) diff --git a/sound/soc/codecs/wm5102.c b/sound/soc/codecs/wm5102.c index 2ed3fa67027d..70d353b63fe0 100644 --- a/sound/soc/codecs/wm5102.c +++ b/sound/soc/codecs/wm5102.c @@ -682,9 +682,7 @@ static int wm5102_out_comp_coeff_put(struct snd_kcontrol *kcontrol, struct arizona *arizona = dev_get_drvdata(component->dev->parent); mutex_lock(&arizona->dac_comp_lock); - memcpy(&arizona->dac_comp_coeff, ucontrol->value.bytes.data, - sizeof(arizona->dac_comp_coeff)); - arizona->dac_comp_coeff = be16_to_cpu(arizona->dac_comp_coeff); + arizona->dac_comp_coeff = get_unaligned_be16(ucontrol->value.bytes.data); mutex_unlock(&arizona->dac_comp_lock); return 0; base-commit: 2ebefde9de62a0828936e47b62975501993b5ee9 -- 2.20.1

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[PATCH 1/2] ASoC: dt-bindings: fsl_spdif: Add new compatible string for i.MX8QM
by Shengjiu Wang 27 Oct '20

27 Oct '20

Add new compatible string "fsl,imx8qm-spdif" for supporting spdif module on i.MX8QM. Signed-off-by: Shengjiu Wang <shengjiu.wang(a)nxp.com> --- Documentation/devicetree/bindings/sound/fsl,spdif.yaml | 1 + 1 file changed, 1 insertion(+) diff --git a/Documentation/devicetree/bindings/sound/fsl,spdif.yaml b/Documentation/devicetree/bindings/sound/fsl,spdif.yaml index 2ac671f5cb9b..50449b6d1048 100644 --- a/Documentation/devicetree/bindings/sound/fsl,spdif.yaml +++ b/Documentation/devicetree/bindings/sound/fsl,spdif.yaml @@ -20,6 +20,7 @@ properties: - fsl,imx35-spdif - fsl,vf610-spdif - fsl,imx6sx-spdif + - fsl,imx8qm-spdif reg: maxItems: 1 -- 2.27.0

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[PATCH 0/6] ASoC: merge soc_pcm_hw_param() rollback and soc_pcm_hw_free()
by Kuninori Morimoto 27 Oct '20

27 Oct '20

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

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[PATCH v2 00/17] ASoC: sun8i-codec: support for AIF2 and AIF3
by Samuel Holland 27 Oct '20

27 Oct '20

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

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[PATCH] ASoC: adau1977: remove platform data and move micbias bindings include
by Alexandru Ardelean 27 Oct '20

27 Oct '20

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(a)analog.com> --- include/dt-bindings/sound/adi,adau1977.h | 15 ++++++++ include/linux/platform_data/adau1977.h | 44 ------------------------ sound/soc/codecs/adau1977.c | 9 ++--- 3 files changed, 18 insertions(+), 50 deletions(-) create mode 100644 include/dt-bindings/sound/adi,adau1977.h delete mode 100644 include/linux/platform_data/adau1977.h diff --git a/include/dt-bindings/sound/adi,adau1977.h b/include/dt-bindings/sound/adi,adau1977.h new file mode 100644 index 000000000000..8eebec6570f2 --- /dev/null +++ b/include/dt-bindings/sound/adi,adau1977.h @@ -0,0 +1,15 @@ +/* SPDX-License-Identifier: GPL-2.0-only */ +#ifndef __DT_BINDINGS_ADI_ADAU1977_H__ +#define __DT_BINDINGS_ADI_ADAU1977_H__ + +#define ADAU1977_MICBIAS_5V0 0x0 +#define ADAU1977_MICBIAS_5V5 0x1 +#define ADAU1977_MICBIAS_6V0 0x2 +#define ADAU1977_MICBIAS_6V5 0x3 +#define ADAU1977_MICBIAS_7V0 0x4 +#define ADAU1977_MICBIAS_7V5 0x5 +#define ADAU1977_MICBIAS_8V0 0x6 +#define ADAU1977_MICBIAS_8V5 0x7 +#define ADAU1977_MICBIAS_9V0 0x8 + +#endif /* __DT_BINDINGS_ADI_ADAU1977_H__ */ diff --git a/include/linux/platform_data/adau1977.h b/include/linux/platform_data/adau1977.h deleted file mode 100644 index 86667235077a..000000000000 --- a/include/linux/platform_data/adau1977.h +++ /dev/null @@ -1,44 +0,0 @@ -/* SPDX-License-Identifier: GPL-2.0-only */ -/* - * ADAU1977/ADAU1978/ADAU1979 driver - * - * Copyright 2014 Analog Devices Inc. - * Author: Lars-Peter Clausen <lars(a)metafoo.de> - */ - -#ifndef __LINUX_PLATFORM_DATA_ADAU1977_H__ -#define __LINUX_PLATFORM_DATA_ADAU1977_H__ - -/** - * enum adau1977_micbias - ADAU1977 MICBIAS pin voltage setting - * @ADAU1977_MICBIAS_5V0: MICBIAS is set to 5.0 V - * @ADAU1977_MICBIAS_5V5: MICBIAS is set to 5.5 V - * @ADAU1977_MICBIAS_6V0: MICBIAS is set to 6.0 V - * @ADAU1977_MICBIAS_6V5: MICBIAS is set to 6.5 V - * @ADAU1977_MICBIAS_7V0: MICBIAS is set to 7.0 V - * @ADAU1977_MICBIAS_7V5: MICBIAS is set to 7.5 V - * @ADAU1977_MICBIAS_8V0: MICBIAS is set to 8.0 V - * @ADAU1977_MICBIAS_8V5: MICBIAS is set to 8.5 V - * @ADAU1977_MICBIAS_9V0: MICBIAS is set to 9.0 V - */ -enum adau1977_micbias { - ADAU1977_MICBIAS_5V0 = 0x0, - ADAU1977_MICBIAS_5V5 = 0x1, - ADAU1977_MICBIAS_6V0 = 0x2, - ADAU1977_MICBIAS_6V5 = 0x3, - ADAU1977_MICBIAS_7V0 = 0x4, - ADAU1977_MICBIAS_7V5 = 0x5, - ADAU1977_MICBIAS_8V0 = 0x6, - ADAU1977_MICBIAS_8V5 = 0x7, - ADAU1977_MICBIAS_9V0 = 0x8, -}; - -/** - * struct adau1977_platform_data - Platform configuration data for the ADAU1977 - * @micbias: Specifies the voltage for the MICBIAS pin - */ -struct adau1977_platform_data { - enum adau1977_micbias micbias; -}; - -#endif diff --git a/sound/soc/codecs/adau1977.c b/sound/soc/codecs/adau1977.c index 0a36e523584c..8260f49caa24 100644 --- a/sound/soc/codecs/adau1977.c +++ b/sound/soc/codecs/adau1977.c @@ -12,7 +12,6 @@ #include <linux/i2c.h> #include <linux/init.h> #include <linux/module.h> -#include <linux/platform_data/adau1977.h> #include <linux/regmap.h> #include <linux/regulator/consumer.h> #include <linux/slab.h> @@ -24,6 +23,8 @@ #include <sound/soc.h> #include <sound/tlv.h> +#include <dt-bindings/sound/adi,adau1977.h> + #include "adau1977.h" #define ADAU1977_REG_POWER 0x00 @@ -881,13 +882,9 @@ static const struct snd_soc_component_driver adau1977_component_driver = { static int adau1977_setup_micbias(struct adau1977 *adau1977) { - struct adau1977_platform_data *pdata = adau1977->dev->platform_data; unsigned int micbias; - if (pdata) - micbias = pdata->micbias; - else if (device_property_read_u32(adau1977->dev, "adi,micbias", - &micbias)) + if (device_property_read_u32(adau1977->dev, "adi,micbias", &micbias)) micbias = ADAU1977_MICBIAS_8V5; if (micbias > ADAU1977_MICBIAS_9V0) { -- 2.17.1

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[PATCH v4 0/2] DAI driver for new XCVR IP
by Viorel Suman (OSS) 27 Oct '20

27 Oct '20

From: Viorel Suman <viorel.suman(a)nxp.com> DAI driver for new XCVR IP found in i.MX8MP. Viorel Suman (2): ASoC: fsl_xcvr: Add XCVR ASoC CPU DAI driver ASoC: dt-bindings: fsl_xcvr: Add document for XCVR Changes since v1: - improved 6- and 12-ch layout comment - used regmap polling function, improved clocks handling in runtime_resume - added FW size check in FW load function, improved IRQ handler, removed dummy IRQ handlers - fixed yaml file Changes since v2: - used devm_reset_control_get_exclusive instead of of_reset_control_get - moved reset_control_assert into runtime_suspend Changes since v3: - removed "firmware-name" DTS property from both documentation and source code by porting it into SoC specific 'compatible' data structure. .../devicetree/bindings/sound/fsl,xcvr.yaml | 104 ++ sound/soc/fsl/Kconfig | 10 + sound/soc/fsl/Makefile | 2 + sound/soc/fsl/fsl_xcvr.c | 1359 +++++++++++++++++ sound/soc/fsl/fsl_xcvr.h | 266 ++++ 5 files changed, 1741 insertions(+) create mode 100644 Documentation/devicetree/bindings/sound/fsl,xcvr.yaml create mode 100644 sound/soc/fsl/fsl_xcvr.c create mode 100644 sound/soc/fsl/fsl_xcvr.h -- 2.26.2

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[PATCH 0/8] use semicolons rather than commas to separate statements
by Julia Lawall 27 Oct '20

27 Oct '20

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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[PATCH v2 0/2] dt-bindings: stm32: convert audio dfsdm to json-schema
by Olivier Moysan 27 Oct '20

27 Oct '20

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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[PATCH v4] ASoC: dt-bindings: stm32: convert sai to json-schema
by Olivier Moysan 27 Oct '20

27 Oct '20

Convert the STM32 SAI bindings to DT schema format using json-schema. Signed-off-by: Olivier Moysan <olivier.moysan(a)st.com> --- Changes in v2: - use pattern for compatible of child nodes - rework dmas and clocks properties - add "additionalProperties" Changes in v3: - move clocks properties for st,stm32h7-sai compatible, to 'else' clause Changes in v4: - fix dtbs_check errors --- .../bindings/sound/st,stm32-sai.txt | 107 ---------- .../bindings/sound/st,stm32-sai.yaml | 200 ++++++++++++++++++ 2 files changed, 200 insertions(+), 107 deletions(-) delete mode 100644 Documentation/devicetree/bindings/sound/st,stm32-sai.txt create mode 100644 Documentation/devicetree/bindings/sound/st,stm32-sai.yaml diff --git a/Documentation/devicetree/bindings/sound/st,stm32-sai.txt b/Documentation/devicetree/bindings/sound/st,stm32-sai.txt deleted file mode 100644 index c42b91e525fa..000000000000 --- a/Documentation/devicetree/bindings/sound/st,stm32-sai.txt +++ /dev/null @@ -1,107 +0,0 @@ -STMicroelectronics STM32 Serial Audio Interface (SAI). - -The SAI interface (Serial Audio Interface) offers a wide set of audio protocols -as I2S standards, LSB or MSB-justified, PCM/DSP, TDM, and AC'97. -The SAI contains two independent audio sub-blocks. Each sub-block has -its own clock generator and I/O lines controller. - -Required properties: - - compatible: Should be "st,stm32f4-sai" or "st,stm32h7-sai" - - reg: Base address and size of SAI common register set. - - clocks: Must contain phandle and clock specifier pairs for each entry - in clock-names. - - clock-names: Must contain "pclk" "x8k" and "x11k" - "pclk": Clock which feeds the peripheral bus interface. - Mandatory for "st,stm32h7-sai" compatible. - Not used for "st,stm32f4-sai" compatible. - "x8k": SAI parent clock for sampling rates multiple of 8kHz. - "x11k": SAI parent clock for sampling rates multiple of 11.025kHz. - - interrupts: cpu DAI interrupt line shared by SAI sub-blocks - -Optional properties: - - resets: Reference to a reset controller asserting the SAI - -SAI subnodes: -Two subnodes corresponding to SAI sub-block instances A et B can be defined. -Subnode can be omitted for unsused sub-block. - -SAI subnodes required properties: - - compatible: Should be "st,stm32-sai-sub-a" or "st,stm32-sai-sub-b" - for SAI sub-block A or B respectively. - - reg: Base address and size of SAI sub-block register set. - - clocks: Must contain one phandle and clock specifier pair - for sai_ck which feeds the internal clock generator. - If the SAI shares a master clock, with another SAI set as MCLK - clock provider, SAI provider phandle must be specified here. - - clock-names: Must contain "sai_ck". - Must also contain "MCLK", if SAI shares a master clock, - with a SAI set as MCLK clock provider. - - dmas: see Documentation/devicetree/bindings/dma/st,stm32-dma.yaml - - dma-names: identifier string for each DMA request line - "tx": if sai sub-block is configured as playback DAI - "rx": if sai sub-block is configured as capture DAI - - pinctrl-names: should contain only value "default" - - pinctrl-0: see Documentation/devicetree/bindings/pinctrl/st,stm32-pinctrl.yaml - -SAI subnodes Optional properties: - - st,sync: specify synchronization mode. - By default SAI sub-block is in asynchronous mode. - This property sets SAI sub-block as slave of another SAI sub-block. - Must contain the phandle and index of the sai sub-block providing - the synchronization. - - st,iec60958: support S/PDIF IEC6958 protocol for playback - IEC60958 protocol is not available for capture. - By default, custom protocol is assumed, meaning that protocol is - configured according to protocol defined in related DAI link node, - such as i2s, left justified, right justified, dsp and pdm protocols. - Note: ac97 protocol is not supported by SAI driver - - #clock-cells: should be 0. This property must be present if the SAI device - is a master clock provider, according to clocks bindings, described in - Documentation/devicetree/bindings/clock/clock-bindings.txt. - -The device node should contain one 'port' child node with one child 'endpoint' -node, according to the bindings defined in Documentation/devicetree/bindings/ -graph.txt. - -Example: -sound_card { - compatible = "audio-graph-card"; - dais = <&sai1b_port>; -}; - -sai1: sai1@40015800 { - compatible = "st,stm32h7-sai"; - #address-cells = <1>; - #size-cells = <1>; - ranges = <0 0x40015800 0x400>; - reg = <0x40015800 0x4>; - clocks = <&rcc SAI1_CK>, <&rcc PLL1_Q>, <&rcc PLL2_P>; - clock-names = "pclk", "x8k", "x11k"; - interrupts = <87>; - - sai1a: audio-controller@40015804 { - compatible = "st,stm32-sai-sub-a"; - reg = <0x4 0x1C>; - clocks = <&rcc SAI1_CK>; - clock-names = "sai_ck"; - dmas = <&dmamux1 1 87 0x400 0x0>; - dma-names = "tx"; - pinctrl-names = "default"; - pinctrl-0 = <&pinctrl_sai1a>; - - sai1b_port: port { - cpu_endpoint: endpoint { - remote-endpoint = <&codec_endpoint>; - format = "i2s"; - }; - }; - }; -}; - -audio-codec { - codec_port: port { - codec_endpoint: endpoint { - remote-endpoint = <&cpu_endpoint>; - }; - }; -}; diff --git a/Documentation/devicetree/bindings/sound/st,stm32-sai.yaml b/Documentation/devicetree/bindings/sound/st,stm32-sai.yaml new file mode 100644 index 000000000000..6ad48c7681c1 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/st,stm32-sai.yaml @@ -0,0 +1,200 @@ +# SPDX-License-Identifier: (GPL-2.0-only OR BSD-2-Clause) +%YAML 1.2 +--- +$id: http://devicetree.org/schemas/sound/st,stm32-sai.yaml# +$schema: http://devicetree.org/meta-schemas/core.yaml# + +title: STMicroelectronics STM32 Serial Audio Interface (SAI) + +maintainers: + - Olivier Moysan <olivier.moysan(a)st.com> + +description: + The SAI interface (Serial Audio Interface) offers a wide set of audio + protocols as I2S standards, LSB or MSB-justified, PCM/DSP, TDM, and AC'97. + The SAI contains two independent audio sub-blocks. Each sub-block has + its own clock generator and I/O lines controller. + +properties: + compatible: + enum: + - st,stm32f4-sai + - st,stm32h7-sai + + reg: + items: + - description: Base address and size of SAI common register set. + - description: Base address and size of SAI identification register set. + minItems: 1 + maxItems: 2 + + ranges: + maxItems: 1 + + interrupts: + maxItems: 1 + + resets: + maxItems: 1 + + "#address-cells": + const: 1 + + "#size-cells": + const: 1 + + clocks: + maxItems: 3 + + clock-names: + maxItems: 3 + +required: + - compatible + - reg + - ranges + - "#address-cells" + - "#size-cells" + - clocks + - clock-names + +patternProperties: + "^audio-controller@[0-9a-f]+$": + type: object + description: + Two subnodes corresponding to SAI sub-block instances A et B + can be defined. Subnode can be omitted for unsused sub-block. + + properties: + compatible: + description: Compatible for SAI sub-block A or B. + pattern: "st,stm32-sai-sub-[ab]" + + "#sound-dai-cells": + const: 0 + + reg: + maxItems: 1 + + clocks: + items: + - description: sai_ck clock feeding the internal clock generator. + - description: MCLK clock from a SAI set as master clock provider. + minItems: 1 + maxItems: 2 + + clock-names: + items: + - const: sai_ck + - const: MCLK + minItems: 1 + maxItems: 2 + + dmas: + maxItems: 1 + + dma-names: + description: | + rx: SAI sub-block is configured as a capture DAI. + tx: SAI sub-block is configured as a playback DAI. + enum: [ rx, tx ] + + st,sync: + description: + Configure the SAI sub-block as slave of another SAI sub-block. + By default SAI sub-block is in asynchronous mode. + Must contain the phandle and index of the SAI sub-block providing + the synchronization. + allOf: + - $ref: /schemas/types.yaml#definitions/phandle-array + - maxItems: 1 + + st,iec60958: + description: + If set, support S/PDIF IEC6958 protocol for playback. + IEC60958 protocol is not available for capture. + By default, custom protocol is assumed, meaning that protocol is + configured according to protocol defined in related DAI link node, + such as i2s, left justified, right justified, dsp and pdm protocols. + allOf: + - $ref: /schemas/types.yaml#definitions/flag + + "#clock-cells": + description: Configure the SAI device as master clock provider. + const: 0 + + required: + - compatible + - "#sound-dai-cells" + - reg + - clocks + - clock-names + - dmas + - dma-names + +allOf: + - if: + properties: + compatible: + contains: + const: st,stm32f4-sai + + - then: + properties: + clocks: + items: + - description: x8k, SAI parent clock for sampling rates multiple of 8kHz. + - description: x11k, SAI parent clock for sampling rates multiple of 11.025kHz. + + clock-names: + items: + - const: x8k + - const: x11k + + - else: + properties: + clocks: + items: + - description: pclk feeds the peripheral bus interface. + - description: x8k, SAI parent clock for sampling rates multiple of 8kHz. + - description: x11k, SAI parent clock for sampling rates multiple of 11.025kHz. + + clock-names: + items: + - const: pclk + - const: x8k + - const: x11k + +additionalProperties: false + +examples: + - | + #include <dt-bindings/interrupt-controller/arm-gic.h> + #include <dt-bindings/clock/stm32mp1-clks.h> + #include <dt-bindings/reset/stm32mp1-resets.h> + sai2: sai@4400b000 { + compatible = "st,stm32h7-sai"; + #address-cells = <1>; + #size-cells = <1>; + ranges = <0 0x4400b000 0x400>; + reg = <0x4400b000 0x4>, <0x4400b3f0 0x10>; + clocks = <&rcc SAI2>, <&rcc PLL3_Q>, <&rcc PLL3_R>; + clock-names = "pclk", "x8k", "x11k"; + pinctrl-names = "default", "sleep"; + pinctrl-0 = <&sai2a_pins_a>, <&sai2b_pins_b>; + pinctrl-1 = <&sai2a_sleep_pins_a>, <&sai2b_sleep_pins_b>; + status = "okay"; + + sai2a: audio-controller@4400b004 { + #sound-dai-cells = <0>; + compatible = "st,stm32-sai-sub-a"; + reg = <0x4 0x1c>; + dmas = <&dmamux1 89 0x400 0x01>; + dma-names = "tx"; + clocks = <&rcc SAI2_K>; + clock-names = "sai_ck"; + status = "okay"; + }; + }; + +... -- 2.17.1

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