August 2024 - Alsa-devel - mailman.alsa-project.org

discussion: why is the mono profile in ALSA not autogenerated by default?
by xeroxslayer＠proton.me 20 Aug '24

20 Aug '24

Sorry if this is the wrong place to discuss this, but from what I managed to find out, this is primarily the responsibility of ALSA. This was pointed out by a reddit user, and to be frank, I was kinda surprised that this is not done by default. The user in question is hearing impaired (can listen on only one ear), so the user has no benefit of actually listening to stereo or any other multichannel audio, thus wanted to just set everything to mono (dual mono I presume) and was puzzled when there was no such option in pulseaudio (which pulls the profiles from ALSA). I did some more investigation and as it turns out, this profile is autogenerated, but only if the hardware has such capability (as is the case with any other profile). You can enable it manually by editing config files (in which case I presume the channel downconversion is done in software), but it seemed odd that one has to go through such trouble just to listen to mono sound. Thus, I think the mono profile should be taken as a special case scenario and be autogenerated/configured by default, regardless if it's supported in hardware or not. Once again, I'm sorry if this is not the place to ask/discuss this, but would appreciate if someone pointed me to where I can actually do this.

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[RFC PATCH v2 0/6] ASoC: fsl: add memory to memory function for ASRC
by Shengjiu Wang 20 Aug '24

20 Aug '24

This function is base on the accelerator implementation for compress API: https://patchwork.kernel.org/project/alsa-devel/patch/20240731083843.59911-… Audio signal processing also has the requirement for memory to memory similar as Video. This asrc memory to memory (memory ->asrc->memory) case is a non real time use case. User fills the input buffer to the asrc module, after conversion, then asrc sends back the output buffer to user. So it is not a traditional ALSA playback and capture case. Because we had implemented the "memory -> asrc ->i2s device-> codec" use case in ALSA. Now the "memory->asrc->memory" needs to reuse the code in asrc driver, so the patch 1 and patch 2 is for refining the code to make it can be shared by the "memory->asrc->memory" driver. Other change is to add memory to memory support for two kinds of i.MX ASRC modules. changes in v2: - Remove the changes in compress API - drop the SNDRV_COMPRESS_SRC_RATIO_MOD - drop the SND_AUDIOCODEC_SRC and struct snd_dec_src - define private metadata key value ASRC_OUTPUT_FORMAT/ASRC_OUTPUT_RATE/ASRC_RATIO_MOD Shengjiu Wang (6): ALSA: compress: reserve space in snd_compr_metadata.key for private usage ASoC: fsl_asrc: define functions for memory to memory usage ASoC: fsl_easrc: define functions for memory to memory usage ASoC: fsl_asrc_m2m: Add memory to memory function ASoC: fsl_asrc: register m2m platform device ASoC: fsl_easrc: register m2m platform device include/uapi/sound/compress_offload.h | 2 +- sound/soc/fsl/Kconfig | 1 + sound/soc/fsl/Makefile | 2 +- sound/soc/fsl/fsl_asrc.c | 176 +++++- sound/soc/fsl/fsl_asrc.h | 2 + sound/soc/fsl/fsl_asrc_common.h | 68 +++ sound/soc/fsl/fsl_asrc_m2m.c | 791 ++++++++++++++++++++++++++ sound/soc/fsl/fsl_easrc.c | 259 ++++++++- sound/soc/fsl/fsl_easrc.h | 4 + 9 files changed, 1297 insertions(+), 8 deletions(-) create mode 100644 sound/soc/fsl/fsl_asrc_m2m.c -- 2.34.1

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[PATCH alsa-utils] aseqdump: Add missing dump of UMP Set Key Signature Message
by Takashi Iwai 20 Aug '24

20 Aug '24

The handling of Set Key Signature message of the Flex Data type was missing by some reason. Now the definition was added in alsa-lib ump_msg.h, so let's show the contents. Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- seq/aseqdump/aseqdump.c | 8 ++++++++ 1 file changed, 8 insertions(+) diff --git a/seq/aseqdump/aseqdump.c b/seq/aseqdump/aseqdump.c index 1d9c01e01dd3..e37c87b45fc1 100644 --- a/seq/aseqdump/aseqdump.c +++ b/seq/aseqdump/aseqdump.c @@ -964,6 +964,14 @@ static void dump_ump_flex_data_event(const unsigned int *ump) return; } + if (fh->meta.status_bank == SND_UMP_FLEX_DATA_MSG_BANK_SETUP && + fh->meta.status == SND_UMP_FLEX_DATA_MSG_STATUS_SET_KEY_SIGNATURE) { + printf("UMP Set Key Signature sharps/flats %d, tonic %d\n", + fh->set_key_sig.sharps_flats, + fh->set_key_sig.tonic_note); + return; + } + if (fh->meta.status_bank == SND_UMP_FLEX_DATA_MSG_BANK_SETUP && fh->meta.status == SND_UMP_FLEX_DATA_MSG_STATUS_SET_CHORD_NAME) { printf("UMP Set Chord Name tonic %d %d %d, alt1 %d/%d, alt2 %d/%d, alt3 %d/%d, alt4 %d/%d, bass %d %d %d, alt1 %d/%d alt2 %d/%d\n", -- 2.43.0

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[PATCH alsa-lib] ump_msg: Add missing definition for Set Key Signature Message
by Takashi Iwai 20 Aug '24

20 Aug '24

The Set Key Signature message was missing at the time we defined for Flex Data types. Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- include/ump_msg.h | 34 ++++++++++++++++++++++++++++++++++ 1 file changed, 34 insertions(+) diff --git a/include/ump_msg.h b/include/ump_msg.h index a64cb317889e..5f0a45acd44c 100644 --- a/include/ump_msg.h +++ b/include/ump_msg.h @@ -606,6 +606,39 @@ typedef struct _snd_ump_msg_set_metronome { #endif } __attribute((packed)) snd_ump_msg_set_metronome_t; +/** Set Key Signature Message (128bit) */ +typedef struct _snd_ump_msg_set_key_sig { +#ifdef SNDRV_BIG_ENDIAN_BITFIELD + uint8_t type:4; /**< UMP packet type */ + uint8_t group:4; /**< UMP Group */ + uint8_t format:2; /**< Format */ + uint8_t addrs:2; /**< Address */ + uint8_t channel:4; /**< Channel */ + uint8_t status_bank; /**< Status Bank */ + uint8_t status; /**< Status */ + + uint8_t sharps_flats:4; /**< Sharps/Flats */ + uint8_t tonic_note:4; /**< Tonic Note 1 */ + uint8_t reserved1[3]; /**< Unused */ + + uint32_t reserved2[2]; /**< Unused */ +#else + uint8_t status; /**< Status */ + uint8_t status_bank; /**< Status Bank */ + uint8_t channel:4; /**< Channel */ + uint8_t addrs:2; /**< Address */ + uint8_t format:2; /**< Format */ + uint8_t group:4; /**< UMP Group */ + uint8_t type:4; /**< UMP packet type */ + + uint8_t reserved1[3]; /**< Unused */ + uint8_t tonic_note:4; /**< Tonic Note */ + uint8_t sharps_flats:4; /**< Sharps/Flats */ + + uint32_t reserved2[2]; /**< Unused */ +#endif +} __attribute((packed)) snd_ump_msg_set_key_sig_t; + /** Set Chord Name Message (128bit) */ typedef struct _snd_ump_msg_set_chord_name { #ifdef SNDRV_BIG_ENDIAN_BITFIELD @@ -678,6 +711,7 @@ typedef union _snd_ump_msg_flex_data { snd_ump_msg_set_tempo_t set_tempo; /**< Set Tempo */ snd_ump_msg_set_time_sig_t set_time_sig; /**< Set Time Signature */ snd_ump_msg_set_metronome_t set_metronome; /**< Set Metronome */ + snd_ump_msg_set_key_sig_t set_key_sig; /**< Set Key Signature */ snd_ump_msg_set_chord_name_t set_chord_name; /**< Set Chord Name */ snd_ump_msg_hdr_t hdr; /**< UMP header */ uint32_t raw[4]; /**< raw UMP packet */ -- 2.43.0

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[PATCH 1/1] ASoC: dt-bindings: convert tlv320aic31xx.txt to yaml
by Frank Li 19 Aug '24

19 Aug '24

Convert binding doc tlv320aic31xx.txt to yaml format. Additional change: - add i2c node in example. - replace MICBIAS_OFF with MICBIAS_2_0v in example because MICBIAS_OFF have been defined in header file. - add ref to dai-common.yaml. - add #sound-dai-cells. Fix below warning: arch/arm64/boot/dts/freescale/imx8mq-zii-ultra-rmb3.dtb: /soc@0/bus@30800000/i2c@30a30000/codec@18: failed to match any schema with compatible: ['ti,tlv320dac3100'] Signed-off-by: Frank Li <Frank.Li(a)nxp.com> --- .../bindings/sound/ti,tlv320dac3100.yaml | 127 ++++++++++++++++++ .../bindings/sound/tlv320aic31xx.txt | 77 ----------- 2 files changed, 127 insertions(+), 77 deletions(-) create mode 100644 Documentation/devicetree/bindings/sound/ti,tlv320dac3100.yaml delete mode 100644 Documentation/devicetree/bindings/sound/tlv320aic31xx.txt diff --git a/Documentation/devicetree/bindings/sound/ti,tlv320dac3100.yaml b/Documentation/devicetree/bindings/sound/ti,tlv320dac3100.yaml new file mode 100644 index 0000000000000..85e937e34962d --- /dev/null +++ b/Documentation/devicetree/bindings/sound/ti,tlv320dac3100.yaml @@ -0,0 +1,127 @@ +# SPDX-License-Identifier: (GPL-2.0-only OR BSD-2-Clause) +%YAML 1.2 +--- +$id: http://devicetree.org/schemas/sound/ti,tlv320dac3100.yaml# +$schema: http://devicetree.org/meta-schemas/core.yaml# + +title: Texas Instruments - tlv320aic31xx Codec module + +maintainers: + - Shenghao Ding <shenghao-ding(a)ti.com> + +description: | + CODEC output pins: + * HPL + * HPR + * SPL, devices with stereo speaker amp + * SPR, devices with stereo speaker amp + * SPK, devices with mono speaker amp + * MICBIAS + + CODEC input pins: + * MIC1LP, devices with ADC + * MIC1RP, devices with ADC + * MIC1LM, devices with ADC + * AIN1, devices without ADC + * AIN2, devices without ADC + + The pins can be used in referring sound node's audio-routing property. + +properties: + compatible: + enum: + - ti,tlv320aic310x # - Generic TLV320AIC31xx with mono speaker amp + - ti,tlv320aic311x # - Generic TLV320AIC31xx with stereo speaker amp + - ti,tlv320aic3100 # - TLV320AIC3100 (mono speaker amp, no MiniDSP) + - ti,tlv320aic3110 # - TLV320AIC3110 (stereo speaker amp, no MiniDSP) + - ti,tlv320aic3120 # - TLV320AIC3120 (mono speaker amp, MiniDSP) + - ti,tlv320aic3111 # - TLV320AIC3111 (stereo speaker amp, MiniDSP) + - ti,tlv320dac3100 # - TLV320DAC3100 (no ADC, mono speaker amp, no MiniDSP) + - ti,tlv320dac3101 # - TLV320DAC3101 (no ADC, stereo speaker amp, no MiniDSP) + + reg: + maxItems: 1 + + '#sound-dai-cells': + const: 0 + + HPVDD-supply: true + + SPRVDD-supply: true + + SPLVDD-supply: true + + AVDD-supply: true + + IOVDD-supply: true + + DVDD-supply: true + + reset-gpios: + description: GPIO specification for the active low RESET input. + + ai31xx-micbias-vg: + $ref: /schemas/types.yaml#/definitions/uint32 + default: 1 + enum: [1, 2, 3] + description: | + MicBias Voltage setting + 1 or MICBIAS_2_0V - MICBIAS output is powered to 2.0V + 2 or MICBIAS_2_5V - MICBIAS output is powered to 2.5V + 3 or MICBIAS_AVDD - MICBIAS output is connected to AVDD + + ai31xx-ocmv: + $ref: /schemas/types.yaml#/definitions/uint32 + enum: [0, 1, 2, 3] + description: | + output common-mode voltage setting + 0 - 1.35V, + 1 - 1.5V, + 2 - 1.65V, + 3 - 1.8V + + gpio-reset: + description: gpio pin number used for codec reset + deprecated: true + + +required: + - compatible + - reg + - HPVDD-supply + - SPRVDD-supply + - SPLVDD-supply + - AVDD-supply + - IOVDD-supply + - DVDD-supply + +allOf: + - $ref: dai-common.yaml# + +unevaluatedProperties: false + +examples: + - | + #include <dt-bindings/gpio/gpio.h> + #include <dt-bindings/sound/tlv320aic31xx.h> + + i2c { + #address-cells = <1>; + #size-cells = <0>; + + sound@18 { + compatible = "ti,tlv320aic311x"; + reg = <0x18>; + + ai31xx-micbias-vg = <MICBIAS_2_0V>; + reset-gpios = <&gpio1 17 GPIO_ACTIVE_LOW>; + + HPVDD-supply = <&regulator>; + SPRVDD-supply = <&regulator>; + SPLVDD-supply = <&regulator>; + AVDD-supply = <&regulator>; + IOVDD-supply = <&regulator>; + DVDD-supply = <&regulator>; + }; + }; + diff --git a/Documentation/devicetree/bindings/sound/tlv320aic31xx.txt b/Documentation/devicetree/bindings/sound/tlv320aic31xx.txt deleted file mode 100644 index bbad98d5b9862..0000000000000 --- a/Documentation/devicetree/bindings/sound/tlv320aic31xx.txt +++ /dev/null @@ -1,77 +0,0 @@ -Texas Instruments - tlv320aic31xx Codec module - -The tlv320aic31xx serial control bus communicates through I2C protocols - -Required properties: - -- compatible - "string" - One of: - "ti,tlv320aic310x" - Generic TLV320AIC31xx with mono speaker amp - "ti,tlv320aic311x" - Generic TLV320AIC31xx with stereo speaker amp - "ti,tlv320aic3100" - TLV320AIC3100 (mono speaker amp, no MiniDSP) - "ti,tlv320aic3110" - TLV320AIC3110 (stereo speaker amp, no MiniDSP) - "ti,tlv320aic3120" - TLV320AIC3120 (mono speaker amp, MiniDSP) - "ti,tlv320aic3111" - TLV320AIC3111 (stereo speaker amp, MiniDSP) - "ti,tlv320dac3100" - TLV320DAC3100 (no ADC, mono speaker amp, no MiniDSP) - "ti,tlv320dac3101" - TLV320DAC3101 (no ADC, stereo speaker amp, no MiniDSP) - -- reg - <int> - I2C slave address -- HPVDD-supply, SPRVDD-supply, SPLVDD-supply, AVDD-supply, IOVDD-supply, - DVDD-supply : power supplies for the device as covered in - Documentation/devicetree/bindings/regulator/regulator.txt - - -Optional properties: - -- reset-gpios - GPIO specification for the active low RESET input. -- ai31xx-micbias-vg - MicBias Voltage setting - 1 or MICBIAS_2_0V - MICBIAS output is powered to 2.0V - 2 or MICBIAS_2_5V - MICBIAS output is powered to 2.5V - 3 or MICBIAS_AVDD - MICBIAS output is connected to AVDD - If this node is not mentioned or if the value is unknown, then - micbias is set to 2.0V. -- ai31xx-ocmv - output common-mode voltage setting - 0 - 1.35V, - 1 - 1.5V, - 2 - 1.65V, - 3 - 1.8V - -Deprecated properties: - -- gpio-reset - gpio pin number used for codec reset - -CODEC output pins: - * HPL - * HPR - * SPL, devices with stereo speaker amp - * SPR, devices with stereo speaker amp - * SPK, devices with mono speaker amp - * MICBIAS - -CODEC input pins: - * MIC1LP, devices with ADC - * MIC1RP, devices with ADC - * MIC1LM, devices with ADC - * AIN1, devices without ADC - * AIN2, devices without ADC - -The pins can be used in referring sound node's audio-routing property. - -Example: -#include <dt-bindings/gpio/gpio.h> -#include <dt-bindings/sound/tlv320aic31xx.h> - -tlv320aic31xx: tlv320aic31xx@18 { - compatible = "ti,tlv320aic311x"; - reg = <0x18>; - - ai31xx-micbias-vg = <MICBIAS_OFF>; - - reset-gpios = <&gpio1 17 GPIO_ACTIVE_LOW>; - - HPVDD-supply = <&regulator>; - SPRVDD-supply = <&regulator>; - SPLVDD-supply = <&regulator>; - AVDD-supply = <&regulator>; - IOVDD-supply = <&regulator>; - DVDD-supply = <&regulator>; -}; -- 2.34.1

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Re: [PATCH] ALSA: hda/realtek: Convert existing CS35L56 products to use autodetect fixup function
by Takashi Iwai 19 Aug '24

19 Aug '24

On Mon, 19 Aug 2024 14:37:36 +0200, Simon Trimmer wrote: > > The existing CS35L56 products can make use of the fixup function that > works out the component binding details so we can remove the fixed > configuration fixup functions. > > Signed-off-by: Simon Trimmer <simont(a)opensource.cirrus.com> Looks like a nice cleanup. Applied to for-next branch now. Thanks. Takashi

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[PATCH v6 6/7] arm64: dts: imx8m: update spdif sound card node properties
by Elinor Montmasson 19 Aug '24

19 Aug '24

The merge of imx-spdif driver into fsl-asoc-card brought new DT properties that can be used with the "fsl,imx-audio-spdif" compatible: * The "spdif-controller" property from imx-spdif is named "audio-cpu" in fsl-asoc-card. * fsl-asoc-card uses codecs explicitly declared in DT with "audio-codec". With an S/PDIF, codec drivers spdif_transmitter and spdif_receiver should be used. Driver imx-spdif used instead the dummy codec and a pair of boolean properties, "spdif-in" and "spdif-out". While backward compatibility is kept to support properties "spdif-controller", "spdif-in" and "spdif-out", using new properties has several benefits: * "audio-cpu" and "audio-codec" are more generic names reflecting that the fsl-asoc-card driver supports multiple hardware. They are properties already used by devices using the fsl-asoc-card driver. They are also similar to properties of simple-card: "cpu" and "codec". * "spdif-in" and "spdif-out" imply the use of the dummy codec in the driver. However, there are already two codec drivers for the S/PDIF, spdif_transmitter and spdif_receiver. It is better to declare S/PDIF Tx and Rx devices in a DT, and then reference them with "audio-codec" than using the dummy codec. For those reasons, this commit updates in-tree DTs to use the new properties: * Rename "spdif-controller" property to "audio-cpu". * Declare S/PDIF transmitter and/or receiver devices, and use them with the "audio-codec" property instead of "spdif-out" and/or "spdif-in". These modifications were tested only on an imx8mn-evk board. Note that out-of-tree and old DTs are still supported. Signed-off-by: Elinor Montmasson <elinor.montmasson(a)savoirfairelinux.com> --- arch/arm64/boot/dts/freescale/imx8mm-evk.dtsi | 15 +++++++++--- arch/arm64/boot/dts/freescale/imx8mn-evk.dtsi | 15 +++++++++--- arch/arm64/boot/dts/freescale/imx8mq-evk.dts | 24 +++++++++++++++---- 3 files changed, 43 insertions(+), 11 deletions(-) diff --git a/arch/arm64/boot/dts/freescale/imx8mm-evk.dtsi b/arch/arm64/boot/dts/freescale/imx8mm-evk.dtsi index 90d1901df2b1..348855a41852 100644 --- a/arch/arm64/boot/dts/freescale/imx8mm-evk.dtsi +++ b/arch/arm64/boot/dts/freescale/imx8mm-evk.dtsi @@ -180,12 +180,21 @@ cpu { }; }; + spdif_out: spdif-out { + #sound-dai-cells = <0>; + compatible = "linux,spdif-dit"; + }; + + spdif_in: spdif-in { + #sound-dai-cells = <0>; + compatible = "linux,spdif-dir"; + }; + sound-spdif { compatible = "fsl,imx-audio-spdif"; model = "imx-spdif"; - spdif-controller = <&spdif1>; - spdif-out; - spdif-in; + audio-cpu = <&spdif1>; + audio-codec = <&spdif_out>, <&spdif_in>; }; }; diff --git a/arch/arm64/boot/dts/freescale/imx8mn-evk.dtsi b/arch/arm64/boot/dts/freescale/imx8mn-evk.dtsi index 9e0259ddf4bc..6a47e09703a7 100644 --- a/arch/arm64/boot/dts/freescale/imx8mn-evk.dtsi +++ b/arch/arm64/boot/dts/freescale/imx8mn-evk.dtsi @@ -124,12 +124,21 @@ sound-wm8524 { "Line Out Jack", "LINEVOUTR"; }; + spdif_out: spdif-out { + #sound-dai-cells = <0>; + compatible = "linux,spdif-dit"; + }; + + spdif_in: spdif-in { + #sound-dai-cells = <0>; + compatible = "linux,spdif-dir"; + }; + sound-spdif { compatible = "fsl,imx-audio-spdif"; model = "imx-spdif"; - spdif-controller = <&spdif1>; - spdif-out; - spdif-in; + audio-cpu = <&spdif1>; + audio-codec = <&spdif_out>, <&spdif_in>; }; sound-micfil { diff --git a/arch/arm64/boot/dts/freescale/imx8mq-evk.dts b/arch/arm64/boot/dts/freescale/imx8mq-evk.dts index 7507548cdb16..b953865f0b46 100644 --- a/arch/arm64/boot/dts/freescale/imx8mq-evk.dts +++ b/arch/arm64/boot/dts/freescale/imx8mq-evk.dts @@ -125,19 +125,33 @@ link_codec: simple-audio-card,codec { }; }; + spdif_out: spdif-out { + #sound-dai-cells = <0>; + compatible = "linux,spdif-dit"; + }; + + spdif_in: spdif-in { + #sound-dai-cells = <0>; + compatible = "linux,spdif-dir"; + }; + sound-spdif { compatible = "fsl,imx-audio-spdif"; model = "imx-spdif"; - spdif-controller = <&spdif1>; - spdif-out; - spdif-in; + audio-cpu = <&spdif1>; + audio-codec = <&spdif_out>, <&spdif_in>; + }; + + hdmi_arc_in: hdmi-arc-in { + #sound-dai-cells = <0>; + compatible = "linux,spdif-dir"; }; sound-hdmi-arc { compatible = "fsl,imx-audio-spdif"; model = "imx-hdmi-arc"; - spdif-controller = <&spdif2>; - spdif-in; + audio-cpu = <&spdif2>; + audio-codec = <&hdmi_arc_in>; }; }; -- 2.34.1

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[PATCH v11 00/10] spi: Add support for stacked/parallel memories
by Amit Kumar Mahapatra 19 Aug '24

19 Aug '24

This patch series updated the spi-nor, spi core and the AMD-Xilinx GQSPI driver to add stacked and parallel memories support. The first nine patches https://lore.kernel.org/all/20230119185342.2093323-1-amit.kumar-mahapatra@a… https://lore.kernel.org/all/20230310173217.3429788-2-amit.kumar-mahapatra@a… https://lore.kernel.org/all/20230310173217.3429788-3-amit.kumar-mahapatra@a… https://lore.kernel.org/all/20230310173217.3429788-4-amit.kumar-mahapatra@a… https://lore.kernel.org/all/20230310173217.3429788-5-amit.kumar-mahapatra@a… https://lore.kernel.org/all/20230310173217.3429788-6-amit.kumar-mahapatra@a… https://lore.kernel.org/all/20230310173217.3429788-7-amit.kumar-mahapatra@a… https://lore.kernel.org/all/20230310173217.3429788-8-amit.kumar-mahapatra@a… https://lore.kernel.org/all/20230310173217.3429788-9-amit.kumar-mahapatra@a… of the previous series got applied to https://git.kernel.org/pub/scm/linux/kernel/git/broonie/spi.git for-next But the rest of the patches in the series did not get applied as failure was reported for spi driver with GPIO CS, so send the remaining patches in the series after rebasing it on top of for-next branch and fixing the issue. CS GPIO is not tested on our hardware, but it has been tested by @Stefan https://lore.kernel.org/all/005001da1efc$619ad5a0$24d080e0$@opensource.cirr… --- BRANCH: for-next Changes in v11: - Rebased patch series on top of latest for-next branch. - Added a new patch(1/10) to replace spi->chip_select with spi_get_chipselect() call in tps6594-spi.c. - Added a new patch(2/10) to replace spi->chip_select with spi_get_chipseletc() call in cs35l56_hda_spi.c. - In spi.c initialized unused CS[] to 0xff and spi->cs_index_mask to 0x01 in all flows. - Updated spi_dev_check() to compare the CS of old spi device with the new spi device CS. - Updated cover letter description to add information regarding GPIO CS testing and added Stefen's Tested-by tag in 3/10 patch. Changes in v10: - Rebased patch series on top of latest for-next branch and fixed merge conflicts. Changes in v9: - Updated 1/8 patch description to add an high-level overview of parallel(multi-cs) & stacked design. - Initialized all unused CS to 0xFF. - Moved CS check from spi_add_device() to __spi_add_device(). - Updated __spi_add_device() to check to make sure that multiple logical CS don't map to the same physical CS and same physical CS doesn't map to different logical CS. - Updated 1/8, 5/8 & 7/8 to support arbitrary number of flash devices connected in parallel or stacked mode. - Updated documentation for chip_select. - Added a new spi-nor structure member nor->num_flash to keep track of the number of flashes connected. - Added a new patch in the series 4/8 to move write_enable call just before spi_mem ops call in SPI-NOR. - Added comments in SPI core & SPI-NOR. - Rebased the patch series on top of the latest for-next branch. Changes in v8: - Updated __spi_add_device() and spi_set_cs() to fix spi driver failure with GPIO CS. - Rebased the patch series on top of latest for-next branch and fixed merge conflicts. - Updated cover letter description to add information regarding GPIO CS testing and request Stefan to provide his Tested-by tag for 1/7 patch. - Updated 1/7 patch description. Changes in v7: - Updated spi_dev_check() to avoid failures for spi driver GPIO CS and moved the error message from __spi_add_device() to spi_dev_check(). - Resolved code indentation issue in spi_set_cs(). - In spi_set_cs() call spi_delay_exec( ) once if the controller supports multi cs with both the CS backed by GPIO. - Updated __spi_validate()to add checks for both the GPIO CS. - Replaced cs_index_mask bit mask with SPI_CS_CNT_MAX. - Updated struct spi_controller to represent multi CS capability of the spi controller through a flag bit SPI_CONTROLLER_MULTI_CS instead of a boolen structure member "multi_cs_cap". - Updated 1/7 patch description . Changes in v6: - Rebased on top of latest v6.3-rc1 and fixed merge conflicts in spi-mpc512x-psc.c, sfdp.c, spansion.c files and removed spi-omap-100k.c. - Updated spi_dev_check( ) to reject new devices if any one of the chipselect is used by another device. Changes in v5: - Rebased the patches on top of v6.3-rc1 and fixed the merge conflicts. - Fixed compilation warnings in spi-sh-msiof.c with shmobile_defconfig Changes in v4: - Fixed build error in spi-pl022.c file - reported by Mark. - Fixed build error in spi-sn-f-ospi.c file. - Added Reviewed-by: Serge Semin <fancer.lancer(a)gmail.com> tag. - Added two more patches to replace spi->chip_select with API calls in mpc832x_rdb.c & cs35l41_hda_spi.c files. Changes in v3: - Rebased the patches on top of https://git.kernel.org/pub/scm/linux/kernel/git/broonie/spi.git for-next - Added a patch to convert spi_nor_otp_region_len(nor) & spi_nor_otp_n_regions(nor) macros into inline functions - Added Reviewed-by & Acked-by tags Changes in v2: - Rebased the patches on top of v6.2-rc1 - Created separate patch to add get & set APIs for spi->chip_select & spi->cs_gpiod, and replaced all spi->chip_select and spi->cs_gpiod references with the API calls. - Created separate patch to add get & set APIs for nor->params. --- Amit Kumar Mahapatra (10): mfd: tps6594: Use set/get APIs to access spi->chip_select ALSA: hda/cs35l56: Use set/get APIs to access spi->chip_select spi: Add multi-cs memories support in SPI core mtd: spi-nor: Convert macros with inline functions mtd: spi-nor: Add APIs to set/get nor->params mtd: spi-nor: Move write enable inside specific write & erase APIs mtd: spi-nor: Add stacked memories support in spi-nor spi: spi-zynqmp-gqspi: Add stacked memories support in GQSPI driver mtd: spi-nor: Add parallel memories support in spi-nor spi: spi-zynqmp-gqspi: Add parallel memories support in GQSPI driver drivers/mfd/tps6594-spi.c | 2 +- drivers/mtd/spi-nor/atmel.c | 17 +- drivers/mtd/spi-nor/core.c | 683 +++++++++++++++++++++++++------ drivers/mtd/spi-nor/core.h | 8 + drivers/mtd/spi-nor/debugfs.c | 4 +- drivers/mtd/spi-nor/gigadevice.c | 4 +- drivers/mtd/spi-nor/issi.c | 11 +- drivers/mtd/spi-nor/macronix.c | 10 +- drivers/mtd/spi-nor/micron-st.c | 35 +- drivers/mtd/spi-nor/otp.c | 48 ++- drivers/mtd/spi-nor/sfdp.c | 33 +- drivers/mtd/spi-nor/spansion.c | 57 +-- drivers/mtd/spi-nor/sst.c | 7 +- drivers/mtd/spi-nor/swp.c | 25 +- drivers/mtd/spi-nor/winbond.c | 2 +- drivers/mtd/spi-nor/xilinx.c | 18 +- drivers/spi/spi-zynqmp-gqspi.c | 58 ++- drivers/spi/spi.c | 259 ++++++++++-- include/linux/mtd/spi-nor.h | 21 +- include/linux/spi/spi.h | 51 ++- sound/pci/hda/cs35l56_hda_spi.c | 2 +- 21 files changed, 1085 insertions(+), 270 deletions(-) -- 2.17.1

10 61

[PATCH v3] MAINTAINERS: update entries in TEXAS INSTRUMENTS LOW/MIDDLE-POWER AUDIO AMPLIFIER (ASoC/HDA) DRIVERS and add entries for haptic driver
by Shenghao Ding 19 Aug '24

19 Aug '24

Due to internal re-org, Kevin is no longer mnaintaining audio driver. Due to job change, drop entries for the audio converter and add entries for both haptics drivers and middle-power audio amplifier drivers. Add audio converter, set the Status as Supported. So far, the Software maintainer has not been confirmed. Once the maintainer was confimred, the guy will update his mail into audio converter section. Signed-off-by: Shenghao Ding <shenghao-ding(a)ti.com> --- v3: - Add Audio converter section - Set the section of LOW/MIDDLE-POWER AUDIO AMPLIFIER as Supported v2: - Add the detailed information of the maintained drivers. --- MAINTAINERS | 33 ++++++++++++++++++++++++--------- 1 file changed, 24 insertions(+), 9 deletions(-) diff --git a/MAINTAINERS b/MAINTAINERS index f328373463b0..32954c5a63c3 100644 --- a/MAINTAINERS +++ b/MAINTAINERS @@ -22569,26 +22569,42 @@ S: Maintained F: Documentation/devicetree/bindings/sound/davinci-mcasp-audio.yaml F: sound/soc/ti/ -TEXAS INSTRUMENTS AUDIO (ASoC/HDA) DRIVERS +TEXAS INSTRUMENTS LOW/MIDDLE-POWER AUDIO AMPLIFIER (ASoC/HDA) & HAPTICS DRIVERS M: Shenghao Ding <shenghao-ding(a)ti.com> -M: Kevin Lu <kevin-lu(a)ti.com> M: Baojun Xu <baojun.xu(a)ti.com> L: alsa-devel(a)alsa-project.org (moderated for non-subscribers) -S: Maintained +S: Supported +F: Documentation/devicetree/bindings/input/ti,drv260x.yaml +F: Documentation/devicetree/bindings/input/ti,drv266x.yaml F: Documentation/devicetree/bindings/sound/tas2552.txt +F: Documentation/devicetree/bindings/sound/tas5720.txt F: Documentation/devicetree/bindings/sound/ti,tas2562.yaml F: Documentation/devicetree/bindings/sound/ti,tas2770.yaml +F: Documentation/devicetree/bindings/sound/ti,tas2781.yaml F: Documentation/devicetree/bindings/sound/ti,tas27xx.yaml +F: Documentation/devicetree/bindings/sound/ti,tas5086.txt +F: Documentation/devicetree/bindings/sound/ti,tas57xx.yaml +F: Documentation/devicetree/bindings/sound/ti,tas5805m.yaml +F: Documentation/devicetree/bindings/sound/tpa6130a2.txt +F: drivers/input/misc/drv2*.c +F: include/sound/tas2*.h +F: include/sound/tas5086.h +F: include/sound/tpa6130a2-plat.h +F: sound/pci/hda/tas2781_hda_i2c.c +F: sound/soc/codecs/tas2*.* +F: sound/soc/codecs/tas5*.* +F: sound/soc/codecs/tpa6130a2.* + +TEXAS INSTRUMENTS AUDIO Converter (ASoC) DRIVERS +L: alsa-devel(a)alsa-project.org (moderated for non-subscribers) +S: Maintained F: Documentation/devicetree/bindings/sound/ti,pcm1681.yaml F: Documentation/devicetree/bindings/sound/ti,pcm3168a.yaml +F: Documentation/devicetree/bindings/sound/ti,pcm6240.yaml F: Documentation/devicetree/bindings/sound/ti,tlv320*.yaml F: Documentation/devicetree/bindings/sound/ti,tlv320adcx140.yaml F: Documentation/devicetree/bindings/sound/tlv320aic31xx.txt -F: Documentation/devicetree/bindings/sound/tpa6130a2.txt -F: include/sound/tas2*.h F: include/sound/tlv320*.h -F: include/sound/tpa6130a2-plat.h -F: sound/pci/hda/tas2781_hda_i2c.c F: sound/soc/codecs/pcm1681.c F: sound/soc/codecs/pcm1789*.* F: sound/soc/codecs/pcm179x*.* @@ -22598,9 +22614,8 @@ F: sound/soc/codecs/pcm3060*.* F: sound/soc/codecs/pcm3168a*.* F: sound/soc/codecs/pcm5102a.c F: sound/soc/codecs/pcm512x*.* -F: sound/soc/codecs/tas2*.* +F: sound/soc/codecs/pcm6240.* F: sound/soc/codecs/tlv320*.* -F: sound/soc/codecs/tpa6130a2.* TEXAS INSTRUMENTS DMA DRIVERS M: Peter Ujfalusi <peter.ujfalusi(a)gmail.com> -- 2.34.1

1 0

Port API coverage reporting to Python 3 and adapt to current doxygen output
by GitHub pull_request - opened 18 Aug '24

18 Aug '24

alsa-project/alsa-python pull request #15 was opened from tormodvolden: Tested by comparing the new output from alsa-python-coverage.py with the old COVERAGE file. For transparency and easier review there are multiple commits for the Python 3 fixups, but this can be squashed if desired. Request URL : https://github.com/alsa-project/alsa-python/pull/15 Patch URL : https://github.com/alsa-project/alsa-python/pull/15.patch Repository URL: https://github.com/alsa-project/alsa-python

1 0

Add persistent ID's for soundcards and midi devices (like coreaudio)
by GitHub issues - opened 18 Aug '24

18 Aug '24

alsa-project/alsa-lib issue #407 was opened from sjomae: "This is something where macOS' Coreaudio/MIDI shines. Unlike macOS Linux/ALSA has no persistent unique IDs for soundcards or MIDI devices. ALSA supports hotplug, and first come first server sequential numeric IDs. The best you^Wpipewire can do is keep track of cards by name. So this is not something pipewire can reliably address, until ALSA get support to identify cards by vendor and serial-number, and provide a UUID." https://lists.linuxaudio.org/hyperkitty/list/linux-audio-dev@lists.linuxaud… Issue URL : https://github.com/alsa-project/alsa-lib/issues/407 Repository URL: https://github.com/alsa-project/alsa-lib

1 0

[PATCH 1/1] ASoC: dt-bindings: Convert tpa6130a2.txt to yaml
by Frank Li 16 Aug '24

16 Aug '24

Convert binding doc tpa6130a2.txt to yaml format. Additional change: - add ref to dai-common.yaml - add i2c node in example Fix below warning: arch/arm64/boot/dts/freescale/imx8mq-zii-ultra-rmb3.dtb: /soc@0/bus@30800000/i2c@30a20000/amp@60: failed to match any schema with compatible: ['ti,tpa6130a2'] Signed-off-by: Frank Li <Frank.Li(a)nxp.com> --- .../bindings/sound/ti,tpa6130a2.yaml | 55 +++++++++++++++++++ .../devicetree/bindings/sound/tpa6130a2.txt | 27 --------- 2 files changed, 55 insertions(+), 27 deletions(-) create mode 100644 Documentation/devicetree/bindings/sound/ti,tpa6130a2.yaml delete mode 100644 Documentation/devicetree/bindings/sound/tpa6130a2.txt diff --git a/Documentation/devicetree/bindings/sound/ti,tpa6130a2.yaml b/Documentation/devicetree/bindings/sound/ti,tpa6130a2.yaml new file mode 100644 index 0000000000000..a42bf9bde6940 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/ti,tpa6130a2.yaml @@ -0,0 +1,55 @@ +# SPDX-License-Identifier: (GPL-2.0-only OR BSD-2-Clause) +%YAML 1.2 +--- +$id: http://devicetree.org/schemas/sound/ti,tpa6130a2.yaml# +$schema: http://devicetree.org/meta-schemas/core.yaml# + +title: Texas Instruments - tpa6130a2 Codec module + +maintainers: + - Sebastian Reichel <sre(a)kernel.org> + +description: + Stereo, analog input headphone amplifier + +properties: + compatible: + enum: + - ti,tpa6130a2 + - ti,tpa6140a2 + + reg: + maxItems: 1 + + Vdd-supply: + description: power supply regulator + + power-gpio: + description: gpio pin to power the device + +required: + - compatible + - reg + - Vdd-supply + +allOf: + - $ref: dai-common.yaml# + +unevaluatedProperties: false + +examples: + - | + #include <dt-bindings/gpio/gpio.h> + + i2c { + #address-cells = <1>; + #size-cells = <0>; + + amplifier@60 { + compatible = "ti,tpa6130a2"; + reg = <0x60>; + Vdd-supply = <&vmmc2>; + power-gpio = <&gpio4 2 GPIO_ACTIVE_HIGH>; + }; + }; + diff --git a/Documentation/devicetree/bindings/sound/tpa6130a2.txt b/Documentation/devicetree/bindings/sound/tpa6130a2.txt deleted file mode 100644 index 6dfa740e4b2d8..0000000000000 --- a/Documentation/devicetree/bindings/sound/tpa6130a2.txt +++ /dev/null @@ -1,27 +0,0 @@ -Texas Instruments - tpa6130a2 Codec module - -The tpa6130a2 serial control bus communicates through I2C protocols - -Required properties: - -- compatible - "string" - One of: - "ti,tpa6130a2" - TPA6130A2 - "ti,tpa6140a2" - TPA6140A2 - - -- reg - <int> - I2C slave address - -- Vdd-supply - <phandle> - power supply regulator - -Optional properties: - -- power-gpio - gpio pin to power the device - -Example: - -tpa6130a2: tpa6130a2@60 { - compatible = "ti,tpa6130a2"; - reg = <0x60>; - Vdd-supply = <&vmmc2>; - power-gpio = <&gpio4 2 GPIO_ACTIVE_HIGH>; -}; -- 2.34.1

3 2

ASUS Maximus Z790 Hero (ALC4082) - USB Audio 5.1 speakers assigned incorrectly and subwoofer not working
by GitHub issues - opened 16 Aug '24

16 Aug '24

alsa-project/alsa-ucm-conf issue #437 was opened from Gess79: I hope I'm correct to post this here. Still new to things Linux. **Situation** OS: Kubuntu 24.04 Mainboard = ASUS ROG Maximus Z790 Hero USB Audio = 0b05:1a53 ASUSTek Computer, Inc. USB Audio Speakers = Logitech Z906 Surround Kernel = 6.8.0-40-generic Alsa version = 1.2.9 **Issue** The speakers seem to be assigned incorrectly and the subwoofer isn't working. I'm testing the speakers through pulseaudio as follows: ![image](https://github.com/user-attachments/assets/8bf65600-63ad-46cc-af68-d51ab6d4a7e6) I'm listing the results below in three categories: - The icon clicked within pulseaudio - The soundfile played - The speaker the sound is actually output to _Icon clicked - Soundfile played - Physical speaker output_ Front Left - Front Left - Front Left Front Center - Front Center - Rear Left Front Right - Front Right - Front Right Rear Left - Rear Left - Front Center Subwoofer - Rear Center - Rear Right Rear Right - (nothing) - (nothing) I've had a lengthy discussion on Reddit but I couldn't find a fix for it so far: https://www.reddit.com/r/linux4noobs/comments/1980uv1/comment/kieqsor/ I'm happy to provide all information necessary to pinpoint the issue and fix it. Issue URL : https://github.com/alsa-project/alsa-ucm-conf/issues/437 Repository URL: https://github.com/alsa-project/alsa-ucm-conf

1 0

Use PyBytes and not Unicode strings for IEC958 values
by GitHub pull_request - edited 16 Aug '24

16 Aug '24

alsa-project/alsa-python pull request #14 was edited from tormodvolden: Suggested fix for #13. - the first commit is not strictly related but also needed to avoid hctltest1.py crashing with my audio devices, where some have info.count = 0. Let me know if more information is needed. I don't know if this is expected or if a warning should be issued. - I went for PyBytes instead of PyByteArray - for setting values, if the provided PyBytes value is shorter than the value, it will be zero-padded. - if it is too long, it will be silently truncated - following the existing code style, no empty lines were inserted, but this could help readability - the third commit is not for fixing the crash, but for fixing set_tuple() to actually do anything for IEC958 - I haven't changed the reference counting, which seems wrong to me in the original code as well. This would be a different issue. Request URL : https://github.com/alsa-project/alsa-python/pull/14 Patch URL : https://github.com/alsa-project/alsa-python/pull/14.patch Repository URL: https://github.com/alsa-project/alsa-python

1 0

Use PyBytes and not Unicode strings for IEC958 values
by GitHub pull_request - edited 16 Aug '24

16 Aug '24

alsa-project/alsa-python pull request #14 was edited from tormodvolden: Suggested fix for #13. - the first commit is not strictly related but also needed to avoid hctltest1.py crashing with my audio devices, where some have info.count = 0. Let me know if more information is needed. I don't know if this is expected or if a warning should be issued. - I went for PyBytes instead of PyByteArray - for setting values, if the provided PyBytes value is shorter than the value, it will be zero-padded. - if it is too long, it will be silently truncated - following the existing code style, no empty lines were inserted, but this could help readability - the third commit is not for fixing the crash, but for fixing set_tuple() to actually do anything for IEC958 Request URL : https://github.com/alsa-project/alsa-python/pull/14 Patch URL : https://github.com/alsa-project/alsa-python/pull/14.patch Repository URL: https://github.com/alsa-project/alsa-python

1 0

[PATCH] ASoC: dt-bindings: qcom,lpass-wsa-macro: correct clocks on SM8250
by srinivas.kandagatla＠linaro.org 16 Aug '24

16 Aug '24

From: Srinivas Kandagatla <srinivas.kandagatla(a)linaro.org> we seems to have ended up with duplicate clocks for frame-sync on sm8250, it has both va and fsgen which are exactly same things. Remove the redundant va clock and make it align with other SoCs. Codec driver does not even handle va clock, so remove this from the bindings and examples to avoid any confusion. Signed-off-by: Srinivas Kandagatla <srinivas.kandagatla(a)linaro.org> --- .../bindings/sound/qcom,lpass-wsa-macro.yaml | 22 ++----------------- 1 file changed, 2 insertions(+), 20 deletions(-) diff --git a/Documentation/devicetree/bindings/sound/qcom,lpass-wsa-macro.yaml b/Documentation/devicetree/bindings/sound/qcom,lpass-wsa-macro.yaml index 06b5f7be3608..6f5644a89feb 100644 --- a/Documentation/devicetree/bindings/sound/qcom,lpass-wsa-macro.yaml +++ b/Documentation/devicetree/bindings/sound/qcom,lpass-wsa-macro.yaml @@ -64,6 +64,7 @@ allOf: compatible: enum: - qcom,sc7280-lpass-wsa-macro + - qcom,sm8250-lpass-wsa-macro - qcom,sm8450-lpass-wsa-macro - qcom,sc8280xp-lpass-wsa-macro then: @@ -79,24 +80,6 @@ allOf: - const: dcodec - const: fsgen - - if: - properties: - compatible: - enum: - - qcom,sm8250-lpass-wsa-macro - then: - properties: - clocks: - minItems: 6 - clock-names: - items: - - const: mclk - - const: npl - - const: macro - - const: dcodec - - const: va - - const: fsgen - - if: properties: compatible: @@ -130,8 +113,7 @@ examples: <&audiocc 0>, <&q6afecc LPASS_HW_MACRO_VOTE LPASS_CLK_ATTRIBUTE_COUPLE_NO>, <&q6afecc LPASS_HW_DCODEC_VOTE LPASS_CLK_ATTRIBUTE_COUPLE_NO>, - <&aoncc LPASS_CDC_VA_MCLK>, <&vamacro>; - clock-names = "mclk", "npl", "macro", "dcodec", "va", "fsgen"; + clock-names = "mclk", "npl", "macro", "dcodec", "fsgen"; clock-output-names = "mclk"; }; -- 2.25.1

3 2

[PATCH v1] ASoc: tas2781: Rename dai_driver name to unify the name between TAS2563 and TAS2781
by Shenghao Ding 16 Aug '24

16 Aug '24

Rename dai_driver name to unify the name between TAS2563 and TAS2781, remove unnecessary line feed, strscpy replace scnprintf. Signed-off-by: Shenghao Ding <shenghao-ding(a)ti.com> --- v1: - Use strscpy to replace scnprintf. - rename dai_driver name from "tas2781_codec" to "tasdev_codec", in case of misunderstanding whether tas2781 or tas2563. - Remove unnecessary line feed for tasdevice_dsp_create_ctrls --- sound/soc/codecs/tas2781-i2c.c | 14 ++++++-------- 1 file changed, 6 insertions(+), 8 deletions(-) diff --git a/sound/soc/codecs/tas2781-i2c.c b/sound/soc/codecs/tas2781-i2c.c index 8a57c045afdd..6631254d4985 100644 --- a/sound/soc/codecs/tas2781-i2c.c +++ b/sound/soc/codecs/tas2781-i2c.c @@ -353,7 +353,7 @@ static int tasdevice_create_control(struct tasdevice_priv *tas_priv) ret = -ENOMEM; goto out; } - scnprintf(name, SNDRV_CTL_ELEM_ID_NAME_MAXLEN, "Speaker Profile Id"); + strscpy(name, "Speaker Profile Id", SNDRV_CTL_ELEM_ID_NAME_MAXLEN); prof_ctrls[mix_index].name = name; prof_ctrls[mix_index].iface = SNDRV_CTL_ELEM_IFACE_MIXER; prof_ctrls[mix_index].info = tasdevice_info_profile; @@ -424,8 +424,7 @@ static int tasdevice_configuration_put( return ret; } -static int tasdevice_dsp_create_ctrls( - struct tasdevice_priv *tas_priv) +static int tasdevice_dsp_create_ctrls(struct tasdevice_priv *tas_priv) { struct snd_kcontrol_new *dsp_ctrls; char *prog_name, *conf_name; @@ -453,8 +452,8 @@ static int tasdevice_dsp_create_ctrls( goto out; } - scnprintf(prog_name, SNDRV_CTL_ELEM_ID_NAME_MAXLEN, - "Speaker Program Id"); + strscpy(prog_name, "Speaker Program Id", + SNDRV_CTL_ELEM_ID_NAME_MAXLEN); dsp_ctrls[mix_index].name = prog_name; dsp_ctrls[mix_index].iface = SNDRV_CTL_ELEM_IFACE_MIXER; dsp_ctrls[mix_index].info = tasdevice_info_programs; @@ -462,8 +461,7 @@ static int tasdevice_dsp_create_ctrls( dsp_ctrls[mix_index].put = tasdevice_program_put; mix_index++; - scnprintf(conf_name, SNDRV_CTL_ELEM_ID_NAME_MAXLEN, - "Speaker Config Id"); + strscpy(conf_name, "Speaker Config Id", SNDRV_CTL_ELEM_ID_NAME_MAXLEN); dsp_ctrls[mix_index].name = conf_name; dsp_ctrls[mix_index].iface = SNDRV_CTL_ELEM_IFACE_MIXER; dsp_ctrls[mix_index].info = tasdevice_info_configurations; @@ -673,7 +671,7 @@ static const struct snd_soc_dai_ops tasdevice_dai_ops = { static struct snd_soc_dai_driver tasdevice_dai_driver[] = { { - .name = "tas2781_codec", + .name = "tasdev_codec", .id = 0, .playback = { .stream_name = "Playback", -- 2.34.1

4 6

[PATCH] ASoC: SOF: amd: Fix for acp init sequence
by Vijendar Mukunda 16 Aug '24

16 Aug '24

When ACP is not powered on by default, acp power on sequence explicitly invoked by programming pgfsm control mask. The existing implementation checks the same PGFSM status mask and programs the same PGFSM control mask in all ACP variants which breaks acp power on sequence for ACP6.0 and ACP6.3 variants. So to fix this issue, update ACP pgfsm control mask and status mask based on acp descriptor rev field, which will vary based on acp variant. Fixes: 846aef1d7cc0 ("ASoC: SOF: amd: Add Renoir ACP HW support") Signed-off-by: Vijendar Mukunda <Vijendar.Mukunda(a)amd.com> --- sound/soc/sof/amd/acp.c | 19 +++++++++++++++++-- sound/soc/sof/amd/acp.h | 7 +++++-- 2 files changed, 22 insertions(+), 4 deletions(-) diff --git a/sound/soc/sof/amd/acp.c b/sound/soc/sof/amd/acp.c index 35e56d22930f..0f6115c8b005 100644 --- a/sound/soc/sof/amd/acp.c +++ b/sound/soc/sof/amd/acp.c @@ -433,6 +433,7 @@ static int acp_power_on(struct snd_sof_dev *sdev) const struct sof_amd_acp_desc *desc = get_chip_info(sdev->pdata); unsigned int base = desc->pgfsm_base; unsigned int val; + unsigned int acp_pgfsm_status_mask, acp_pgfsm_cntl_mask; int ret; val = snd_sof_dsp_read(sdev, ACP_DSP_BAR, base + PGFSM_STATUS_OFFSET); @@ -440,9 +441,23 @@ static int acp_power_on(struct snd_sof_dev *sdev) if (val == ACP_POWERED_ON) return 0; - if (val & ACP_PGFSM_STATUS_MASK) + switch (desc->rev) { + case 3: + case 5: + acp_pgfsm_status_mask = ACP3X_PGFSM_STATUS_MASK; + acp_pgfsm_cntl_mask = ACP3X_PGFSM_CNTL_POWER_ON_MASK; + break; + case 6: + acp_pgfsm_status_mask = ACP6X_PGFSM_STATUS_MASK; + acp_pgfsm_cntl_mask = ACP6X_PGFSM_CNTL_POWER_ON_MASK; + break; + default: + return -EINVAL; + } + + if (val & acp_pgfsm_status_mask) snd_sof_dsp_write(sdev, ACP_DSP_BAR, base + PGFSM_CONTROL_OFFSET, - ACP_PGFSM_CNTL_POWER_ON_MASK); + acp_pgfsm_cntl_mask); ret = snd_sof_dsp_read_poll_timeout(sdev, ACP_DSP_BAR, base + PGFSM_STATUS_OFFSET, val, !val, ACP_REG_POLL_INTERVAL, ACP_REG_POLL_TIMEOUT_US); diff --git a/sound/soc/sof/amd/acp.h b/sound/soc/sof/amd/acp.h index f6f0fcfeb691..321c40cc6d50 100644 --- a/sound/soc/sof/amd/acp.h +++ b/sound/soc/sof/amd/acp.h @@ -25,8 +25,11 @@ #define ACP_REG_POLL_TIMEOUT_US 2000 #define ACP_DMA_COMPLETE_TIMEOUT_US 5000 -#define ACP_PGFSM_CNTL_POWER_ON_MASK 0x01 -#define ACP_PGFSM_STATUS_MASK 0x03 +#define ACP3X_PGFSM_CNTL_POWER_ON_MASK 0x01 +#define ACP3X_PGFSM_STATUS_MASK 0x03 +#define ACP6X_PGFSM_CNTL_POWER_ON_MASK 0x07 +#define ACP6X_PGFSM_STATUS_MASK 0x0F + #define ACP_POWERED_ON 0x00 #define ACP_ASSERT_RESET 0x01 #define ACP_RELEASE_RESET 0x00 -- 2.34.1

2 1

Use PyBytes and not Unicode strings for IEC958 values
by GitHub pull_request - edited 16 Aug '24

16 Aug '24

alsa-project/alsa-python pull request #14 was edited from tormodvolden: Suggested fix for #13. - the first commit is not strictly related but also needed to avoid hctltest1.py crashing with my audio devices, where some have info.count = 0. Let me know if more information is needed. I don't know if this is expected or if a warning should be issued. - I went for PyBytes instead of PyByteArray - for setting values, if the provided PyBytes value is shorter than the value, it will be zero-padded. - if it is too long, it will be silently truncated - following the existing code style, no empty lines were inserted, but this could help readability Request URL : https://github.com/alsa-project/alsa-python/pull/14 Patch URL : https://github.com/alsa-project/alsa-python/pull/14.patch Repository URL: https://github.com/alsa-project/alsa-python

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Use PyBytes and not Unicode strings for IEC958 values
by GitHub pull_request - opened 16 Aug '24

16 Aug '24

alsa-project/alsa-python pull request #14 was opened from tormodvolden: Suggested fix for #13. - the first commit is not strictly related but also needed to avoid hctltest1.py crashing with my audio devices, where some have info.count = 0. Let me know if more information is needed. - I went for PyBytes instead of PyByteArray - for setting values, if the provided PyBytes value is shorter than the value, it will be zero-padded. - if it is too long, it will be silently truncated - following the existing code style, no empty lines were inserted, but this could help readability Request URL : https://github.com/alsa-project/alsa-python/pull/14 Patch URL : https://github.com/alsa-project/alsa-python/pull/14.patch Repository URL: https://github.com/alsa-project/alsa-python

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[PATCH] ASoC: codecs: lpass-va-macro: set the default codec version for sm8250
by srinivas.kandagatla＠linaro.org 16 Aug '24

16 Aug '24

From: Srinivas Kandagatla <srinivas.kandagatla(a)linaro.org> sm8250 and sc7280 have lpass codec version 1.0, as these are very old platforms, they do not have a reliable way to get the codec version from core_id registers. Add the version info into of_data, so that it does not need to use core_id registers to get version number. Fixes: 378918d59181 ("ASoC: codecs: lpass-macro: add helpers to get codec version") Fixes: dbacef05898d ("ASoC: codec: lpass-rx-macro: prepare driver to accomdate new codec versions") Fixes: 727de4fbc546 ("ASoC: codecs: lpass-wsa-macro: Correct support for newer v2.5 version") Signed-off-by: Srinivas Kandagatla <srinivas.kandagatla(a)linaro.org> --- sound/soc/codecs/lpass-va-macro.c | 11 ++++++++++- 1 file changed, 10 insertions(+), 1 deletion(-) diff --git a/sound/soc/codecs/lpass-va-macro.c b/sound/soc/codecs/lpass-va-macro.c index 8454193ed22a..e95d1f29ef18 100644 --- a/sound/soc/codecs/lpass-va-macro.c +++ b/sound/soc/codecs/lpass-va-macro.c @@ -228,11 +228,13 @@ struct va_macro { struct va_macro_data { bool has_swr_master; bool has_npl_clk; + int version; }; static const struct va_macro_data sm8250_va_data = { .has_swr_master = false, .has_npl_clk = false, + .version = LPASS_CODEC_VERSION_1_0, }; static const struct va_macro_data sm8450_va_data = { @@ -1587,7 +1589,14 @@ static int va_macro_probe(struct platform_device *pdev) goto err_npl; } - va_macro_set_lpass_codec_version(va); + /** + * old version of codecs do not have a reliable way to determine the + * version from registers, get them from soc specific data + */ + if (data->version) + lpass_macro_set_codec_version(data->version); + else /* read version from register */ + va_macro_set_lpass_codec_version(va); if (va->has_swr_master) { /* Set default CLK div to 1 */ -- 2.25.1

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[PATCH v1] ASoC: codecs: ES8326: Adjust buttons
by zhangyi＠everest-semi.com 16 Aug '24

16 Aug '24

The content of this message was lost. It was probably cross-posted to multiple lists and previously handled on another list.

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[PATCH] ALSA: aoa: Use helper function for_each_child_of_node()
by Zhang Zekun 16 Aug '24

16 Aug '24

for_each_child_of_node can help to iterate through the device_node, and we don't need to use while loop. No functional change with this conversion. Signed-off-by: Zhang Zekun <zhangzekun11(a)huawei.com> --- sound/aoa/soundbus/i2sbus/core.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/aoa/soundbus/i2sbus/core.c b/sound/aoa/soundbus/i2sbus/core.c index 5431d2c49421..ce84288168e4 100644 --- a/sound/aoa/soundbus/i2sbus/core.c +++ b/sound/aoa/soundbus/i2sbus/core.c @@ -335,7 +335,7 @@ static int i2sbus_add_dev(struct macio_dev *macio, static int i2sbus_probe(struct macio_dev* dev, const struct of_device_id *match) { - struct device_node *np = NULL; + struct device_node *np; int got = 0, err; struct i2sbus_control *control = NULL; @@ -347,7 +347,7 @@ static int i2sbus_probe(struct macio_dev* dev, const struct of_device_id *match) return -ENODEV; } - while ((np = of_get_next_child(dev->ofdev.dev.of_node, np))) { + for_each_child_of_node(dev->ofdev.dev.of_node, np) { if (of_device_is_compatible(np, "i2sbus") || of_device_is_compatible(np, "i2s-modem")) { got += i2sbus_add_dev(dev, control, np); -- 2.17.1

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[PATCH v2 0/6] ASoC: codecs: wsa88xx: add support for static port mapping.
by srinivas.kandagatla＠linaro.org 15 Aug '24

15 Aug '24

Existing way of allocating soundwire master ports on Qualcommm platforms is dynamic, and in linear order starting from 1 to MAX_PORTS. This will work as long as soundwire device ports are 1:1 mapped linearly. However on most Qcom SoCs like SM8550, SM8650, x1e80100, these are NOT mapped in that order. The result of this is that only one speaker among the pair of speakers is always silent, With recent changes for WSA codec to support codec versions and along with these patches we are able to get all speakers working on these SoCs. Signed-off-by: Srinivas Kandagatla <srinivas.kandagatla(a)linaro.org> --- Changes in v2: - used dev_dbg instead of dev_info - Link to v1: https://lore.kernel.org/r/20240626-port-map-v1-0-bd8987d2b332@linaro.org --- Manikantan R (1): ASoC: dt-bindings: wsa883x: Document port mapping property Srinivas Kandagatla (5): ASoC: codecs: wsa883x: parse port-mapping information ASoC: dt-bindings: wsa8840: Document port mapping property ASoC: codecs: wsa884x: parse port-mapping information arm64: dts: x1e80100-crd: fix wsa soundwire port mapping arm64: dts: x1e80100-qcp: fix wsa soundwire port mapping Documentation/devicetree/bindings/sound/qcom,wsa883x.yaml | 8 ++++++++ Documentation/devicetree/bindings/sound/qcom,wsa8840.yaml | 8 ++++++++ arch/arm64/boot/dts/qcom/x1e80100-crd.dts | 4 ++++ arch/arm64/boot/dts/qcom/x1e80100-qcp.dts | 2 ++ sound/soc/codecs/wsa883x.c | 8 ++++++++ sound/soc/codecs/wsa884x.c | 8 ++++++++ 6 files changed, 38 insertions(+) --- base-commit: 9935be184a55dd84fc3275094f2df095491f6ea1 change-id: 20240626-port-map-ef50c3304d4a Best regards, -- Srinivas Kandagatla <srinivas.kandagatla(a)linaro.org>

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[PATCH 0/6] soc: qcom: use scoped device node handling to simplify error paths
by Krzysztof Kozlowski 15 Aug '24

15 Aug '24

Use cleanup.h/scope on device nodes to get rid of of_node_put(). Code is a bit simpler. Best regards, Krzysztof --- Krzysztof Kozlowski (6): soc: qcom: apr: simplify with scoped for each OF child loop soc: qcom: aoss: simplify with scoped for each OF child loop soc: qcom: ice: use scoped device node handling to simplify error paths soc: qcom: ocmem: use scoped device node handling to simplify error paths soc: qcom: pbs: use scoped device node handling to simplify error paths soc: qcom: smp2p: use scoped device node handling to simplify error paths drivers/soc/qcom/apr.c | 5 +---- drivers/soc/qcom/ice.c | 14 +++++--------- drivers/soc/qcom/ocmem.c | 7 ++----- drivers/soc/qcom/qcom-pbs.c | 16 ++++++---------- drivers/soc/qcom/qcom_aoss.c | 8 +++----- drivers/soc/qcom/smp2p.c | 16 ++++------------ 6 files changed, 21 insertions(+), 45 deletions(-) --- base-commit: d54f33dafd9e93367ffaf95318ed371138a6c8c0 change-id: 20240813-b4-cleanup-h-of-node-put-other-7e9c7128abe4 Best regards, -- Krzysztof Kozlowski <krzysztof.kozlowski(a)linaro.org>

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