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

[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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[PATCH] ASoC: pcm5102a: Make codec selectable
by Codrin Ciubotariu 27 Oct '20

27 Oct '20

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(a)microchip.com> --- sound/soc/codecs/Kconfig | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig index 34c6dd04b85a..5791b7056af6 100644 --- a/sound/soc/codecs/Kconfig +++ b/sound/soc/codecs/Kconfig @@ -1003,7 +1003,7 @@ config SND_SOC_PCM3168A_SPI select REGMAP_SPI config SND_SOC_PCM5102A - tristate + tristate "Texas Instruments PCM5102A CODEC" config SND_SOC_PCM512x tristate -- 2.25.1

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[PATCH v2] ASoC: fsl_spdif: Add support for higher sample rates
by Shengjiu Wang 27 Oct '20

27 Oct '20

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(a)nxp.com> Signed-off-by: Shengjiu Wang <shengjiu.wang(a)nxp.com> Acked-by: Nicolin Chen <nicoleotsuka(a)gmail.com> --- changes in v2 - reorder the signed-off - add acked-by Nicolin sound/soc/fsl/fsl_spdif.c | 16 +++++++++++++--- sound/soc/fsl/fsl_spdif.h | 9 ++++++++- 2 files changed, 21 insertions(+), 4 deletions(-) diff --git a/sound/soc/fsl/fsl_spdif.c b/sound/soc/fsl/fsl_spdif.c index 4d14f4076ead..1c030142556a 100644 --- a/sound/soc/fsl/fsl_spdif.c +++ b/sound/soc/fsl/fsl_spdif.c @@ -459,10 +459,18 @@ static int spdif_set_sample_rate(struct snd_pcm_substream *substream, rate = SPDIF_TXRATE_48000; csfs = IEC958_AES3_CON_FS_48000; break; + case 88200: + rate = SPDIF_TXRATE_88200; + csfs = IEC958_AES3_CON_FS_88200; + break; case 96000: rate = SPDIF_TXRATE_96000; csfs = IEC958_AES3_CON_FS_96000; break; + case 176400: + rate = SPDIF_TXRATE_176400; + csfs = IEC958_AES3_CON_FS_176400; + break; case 192000: rate = SPDIF_TXRATE_192000; csfs = IEC958_AES3_CON_FS_192000; @@ -857,7 +865,7 @@ static int fsl_spdif_rxrate_info(struct snd_kcontrol *kcontrol, uinfo->type = SNDRV_CTL_ELEM_TYPE_INTEGER; uinfo->count = 1; uinfo->value.integer.min = 16000; - uinfo->value.integer.max = 96000; + uinfo->value.integer.max = 192000; return 0; } @@ -1175,7 +1183,8 @@ static u32 fsl_spdif_txclk_caldiv(struct fsl_spdif_priv *spdif_priv, struct clk *clk, u64 savesub, enum spdif_txrate index, bool round) { - static const u32 rate[] = { 32000, 44100, 48000, 96000, 192000 }; + static const u32 rate[] = { 32000, 44100, 48000, 88200, 96000, 176400, + 192000, }; bool is_sysclk = clk_is_match(clk, spdif_priv->sysclk); u64 rate_ideal, rate_actual, sub; u32 arate; @@ -1235,7 +1244,8 @@ static u32 fsl_spdif_txclk_caldiv(struct fsl_spdif_priv *spdif_priv, static int fsl_spdif_probe_txclk(struct fsl_spdif_priv *spdif_priv, enum spdif_txrate index) { - static const u32 rate[] = { 32000, 44100, 48000, 96000, 192000 }; + static const u32 rate[] = { 32000, 44100, 48000, 88200, 96000, 176400, + 192000, }; struct platform_device *pdev = spdif_priv->pdev; struct device *dev = &pdev->dev; u64 savesub = 100000, ret; diff --git a/sound/soc/fsl/fsl_spdif.h b/sound/soc/fsl/fsl_spdif.h index e6c61e07bc1a..d5f1dfd58740 100644 --- a/sound/soc/fsl/fsl_spdif.h +++ b/sound/soc/fsl/fsl_spdif.h @@ -163,7 +163,9 @@ enum spdif_txrate { SPDIF_TXRATE_32000 = 0, SPDIF_TXRATE_44100, SPDIF_TXRATE_48000, + SPDIF_TXRATE_88200, SPDIF_TXRATE_96000, + SPDIF_TXRATE_176400, SPDIF_TXRATE_192000, }; #define SPDIF_TXRATE_MAX (SPDIF_TXRATE_192000 + 1) @@ -177,15 +179,20 @@ enum spdif_txrate { #define FSL_SPDIF_RATES_PLAYBACK (SNDRV_PCM_RATE_32000 | \ SNDRV_PCM_RATE_44100 | \ SNDRV_PCM_RATE_48000 | \ + SNDRV_PCM_RATE_88200 | \ SNDRV_PCM_RATE_96000 | \ + SNDRV_PCM_RATE_176400 | \ SNDRV_PCM_RATE_192000) #define FSL_SPDIF_RATES_CAPTURE (SNDRV_PCM_RATE_16000 | \ SNDRV_PCM_RATE_32000 | \ SNDRV_PCM_RATE_44100 | \ SNDRV_PCM_RATE_48000 | \ + SNDRV_PCM_RATE_88200 | \ SNDRV_PCM_RATE_64000 | \ - SNDRV_PCM_RATE_96000) + SNDRV_PCM_RATE_96000 | \ + SNDRV_PCM_RATE_176400 | \ + SNDRV_PCM_RATE_192000) #define FSL_SPDIF_FORMATS_PLAYBACK (SNDRV_PCM_FMTBIT_S16_LE | \ SNDRV_PCM_FMTBIT_S20_3LE | \ -- 2.27.0

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[PATCH] ASoC: Intel: kbl_rt5663_max98927: Fix kabylake_ssp_fixup function
by Tomasz Figa 27 Oct '20

27 Oct '20

This is a copy of commit 5c5f1baee85a ("ASoC: Intel: kbl_rt5663_rt5514_max98927: Fix kabylake_ssp_fixup function") applied to the kbl_rt5663_max98927 board file. Original explanation of the change: kabylake_ssp_fixup function uses snd_soc_dpcm to identify the codecs DAIs. The HW parameters are changed based on the codec DAI of the stream. The earlier approach to get snd_soc_dpcm was using container_of() macro on snd_pcm_hw_params. The structures have been modified over time and snd_soc_dpcm does not have snd_pcm_hw_params as a reference but as a copy. This causes the current driver to crash when used. This patch changes the way snd_soc_dpcm is extracted. snd_soc_pcm_runtime holds 2 dpcm instances (one for playback and one for capture). 2 codecs on the SSP are dmic (capture) and speakers (playback). Based on the stream direction, snd_soc_dpcm is extracted from snd_soc_pcm_runtime. Fixes a boot crash on a HP Chromebook x2: [ 16.582225] BUG: kernel NULL pointer dereference, address: 0000000000000050 [ 16.582231] #PF: supervisor read access in kernel mode [ 16.582233] #PF: error_code(0x0000) - not-present page [ 16.582234] PGD 0 P4D 0 [ 16.582238] Oops: 0000 [#1] PREEMPT SMP PTI [ 16.582241] CPU: 0 PID: 1980 Comm: cras Tainted: G C 5.4.58 #1 [ 16.582243] Hardware name: HP Soraka/Soraka, BIOS Google_Soraka.10431.75.0 08/30/2018 [ 16.582247] RIP: 0010:kabylake_ssp_fixup+0x19/0xbb [snd_soc_kbl_rt5663_max98927] [ 16.582250] Code: c6 6f c5 80 c0 44 89 f2 31 c0 e8 3e c9 4c d6 eb de 0f 1f 44 00 00 55 48 89 e5 41 57 41 56 53 48 89 f3 48 8b 46 c8 48 8b 4e d0 <48> 8b 49 10 4c 8b 78 10 4c 8b 31 4c 89 f7 48 c7 c6 4b c2 80 c0 e8 [ 16.582252] RSP: 0000:ffffaf7e81e0b958 EFLAGS: 00010282 [ 16.582254] RAX: ffffffff96f13e0d RBX: ffffaf7e81e0ba00 RCX: 0000000000000040 [ 16.582256] RDX: ffffaf7e81e0ba00 RSI: ffffaf7e81e0ba00 RDI: ffffa3b208558028 [ 16.582258] RBP: ffffaf7e81e0b970 R08: ffffa3b203b54160 R09: ffffaf7e81e0ba00 [ 16.582259] R10: 0000000000000000 R11: ffffffffc080b345 R12: ffffa3b209fb6e00 [ 16.582261] R13: ffffa3b1b1a47838 R14: ffffa3b1e6197f28 R15: ffffaf7e81e0ba00 [ 16.582263] FS: 00007eb3f25aaf80(0000) GS:ffffa3b236a00000(0000) knlGS:0000000000000000 [ 16.582265] CS: 0010 DS: 0000 ES: 0000 CR0: 0000000080050033 [ 16.582267] CR2: 0000000000000050 CR3: 0000000246bc8006 CR4: 00000000003606f0 [ 16.582269] Call Trace: [ 16.582275] snd_soc_link_be_hw_params_fixup+0x21/0x68 [ 16.582278] snd_soc_dai_hw_params+0x25/0x94 [ 16.582282] soc_pcm_hw_params+0x2d8/0x583 [ 16.582288] dpcm_be_dai_hw_params+0x172/0x29e [ 16.582291] dpcm_fe_dai_hw_params+0x9f/0x12f [ 16.582295] snd_pcm_hw_params+0x137/0x41c [ 16.582298] snd_pcm_hw_params_user+0x3c/0x71 [ 16.582301] snd_pcm_common_ioctl+0x2c6/0x565 [ 16.582304] snd_pcm_ioctl+0x32/0x36 [ 16.582307] do_vfs_ioctl+0x506/0x783 [ 16.582311] ksys_ioctl+0x58/0x83 [ 16.582313] __x64_sys_ioctl+0x1a/0x1e [ 16.582316] do_syscall_64+0x54/0x7e [ 16.582319] entry_SYSCALL_64_after_hwframe+0x44/0xa9 [ 16.582322] RIP: 0033:0x7eb3f1886157 [ 16.582324] Code: 8a 66 90 48 8b 05 11 dd 2b 00 64 c7 00 26 00 00 00 48 c7 c0 ff ff ff ff c3 66 2e 0f 1f 84 00 00 00 00 00 b8 10 00 00 00 0f 05 <48> 3d 01 f0 ff ff 73 01 c3 48 8b 0d e1 dc 2b 00 f7 d8 64 89 01 48 [ 16.582326] RSP: 002b:00007ffff7559818 EFLAGS: 00000246 ORIG_RAX: 0000000000000010 [ 16.582329] RAX: ffffffffffffffda RBX: 00005acc9188b140 RCX: 00007eb3f1886157 [ 16.582330] RDX: 00007ffff7559940 RSI: 00000000c2604111 RDI: 000000000000001e [ 16.582332] RBP: 00007ffff7559840 R08: 0000000000000004 R09: 0000000000000000 [ 16.582333] R10: 0000000000000000 R11: 0000000000000246 R12: 000000000000bb80 [ 16.582335] R13: 00005acc91702e80 R14: 00007ffff7559940 R15: 00005acc91702e80 [ 16.582337] Modules linked in: rfcomm cmac algif_hash algif_skcipher af_alg uinput hid_google_hammer snd_soc_kbl_rt5663_max98927 snd_soc_hdac_hdmi snd_soc_dmic snd_soc_skl_ssp_clk snd_soc_skl snd_soc_sst_ipc snd_soc_sst_dsp snd_soc_hdac_hda snd_soc_acpi_intel_match snd_soc_acpi snd_hda_ext_core snd_intel_dspcfg snd_hda_codec snd_hwdep snd_hda_core ipu3_cio2 ipu3_imgu(C) videobuf2_v4l2 videobuf2_common videobuf2_dma_sg videobuf2_memops snd_soc_rt5663 snd_soc_max98927 snd_soc_rl6231 ov5670 ov13858 acpi_als v4l2_fwnode dw9714 fuse xt_MASQUERADE iio_trig_sysfs cros_ec_light_prox cros_ec_sensors cros_ec_sensors_core cros_ec_sensors_ring industrialio_triggered_buffer kfifo_buf industrialio cros_ec_sensorhub cdc_ether usbnet btusb btrtl btintel btbcm bluetooth ecdh_generic ecc lzo_rle lzo_compress iwlmvm zram iwl7000_mac80211 r8152 mii iwlwifi cfg80211 joydev [ 16.584243] gsmi: Log Shutdown Reason 0x03 [ 16.584246] CR2: 0000000000000050 [ 16.584248] ---[ end trace c8511d090c11edff ]--- Suggested-by: Łukasz Majczak <lmajczak(a)google.com> Fixes: 2e5894d73789e ("ASoC: pcm: Add support for DAI multicodec") Signed-off-by: Tomasz Figa <tfiga(a)chromium.org> --- sound/soc/intel/boards/kbl_rt5663_max98927.c | 39 ++++++++++++++++---- 1 file changed, 31 insertions(+), 8 deletions(-) diff --git a/sound/soc/intel/boards/kbl_rt5663_max98927.c b/sound/soc/intel/boards/kbl_rt5663_max98927.c index 3ea4602dfb3e..9a4b3d0973f6 100644 --- a/sound/soc/intel/boards/kbl_rt5663_max98927.c +++ b/sound/soc/intel/boards/kbl_rt5663_max98927.c @@ -401,17 +401,40 @@ static int kabylake_ssp_fixup(struct snd_soc_pcm_runtime *rtd, struct snd_interval *chan = hw_param_interval(params, SNDRV_PCM_HW_PARAM_CHANNELS); struct snd_mask *fmt = hw_param_mask(params, SNDRV_PCM_HW_PARAM_FORMAT); - struct snd_soc_dpcm *dpcm = container_of( - params, struct snd_soc_dpcm, hw_params); - struct snd_soc_dai_link *fe_dai_link = dpcm->fe->dai_link; - struct snd_soc_dai_link *be_dai_link = dpcm->be->dai_link; + struct snd_soc_dpcm *dpcm, *rtd_dpcm = NULL; + + /* + * The following loop will be called only for playback stream + * In this platform, there is only one playback device on every SSP + */ + for_each_dpcm_fe(rtd, SNDRV_PCM_STREAM_PLAYBACK, dpcm) { + rtd_dpcm = dpcm; + break; + } + + /* + * This following loop will be called only for capture stream + * In this platform, there is only one capture device on every SSP + */ + for_each_dpcm_fe(rtd, SNDRV_PCM_STREAM_CAPTURE, dpcm) { + rtd_dpcm = dpcm; + break; + } + + if (!rtd_dpcm) + return -EINVAL; + + /* + * The above 2 loops are mutually exclusive based on the stream direction, + * thus rtd_dpcm variable will never be overwritten + */ /* * The ADSP will convert the FE rate to 48k, stereo, 24 bit */ - if (!strcmp(fe_dai_link->name, "Kbl Audio Port") || - !strcmp(fe_dai_link->name, "Kbl Audio Headset Playback") || - !strcmp(fe_dai_link->name, "Kbl Audio Capture Port")) { + if (!strcmp(rtd_dpcm->fe->dai_link->name, "Kbl Audio Port") || + !strcmp(rtd_dpcm->fe->dai_link->name, "Kbl Audio Headset Playback") || + !strcmp(rtd_dpcm->fe->dai_link->name, "Kbl Audio Capture Port")) { rate->min = rate->max = 48000; chan->min = chan->max = 2; snd_mask_none(fmt); @@ -421,7 +444,7 @@ static int kabylake_ssp_fixup(struct snd_soc_pcm_runtime *rtd, * The speaker on the SSP0 supports S16_LE and not S24_LE. * thus changing the mask here */ - if (!strcmp(be_dai_link->name, "SSP0-Codec")) + if (!strcmp(rtd_dpcm->be->dai_link->name, "SSP0-Codec")) snd_mask_set_format(fmt, SNDRV_PCM_FORMAT_S16_LE); return 0; -- 2.28.0.1011.ga647a8990f-goog

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