May 2023 - Alsa-devel - mailman.alsa-project.org

Can't play 44100hz, 88200hz files Natively.
by GitHub issues - opened 11 May '23

11 May '23

alsa-project/alsa-ucm-conf issue #313 was opened from RenEH01: Can't play audio files at native sample rate. Everything using the Hifi 2.0 output configuration forces a sample rate of 48000hz. When I switch to the Pro Audio driver under pavucontrol, 44100 hz can play natively. Is there a forced resampling occuring in the UCM2 driver? Issue URL : https://github.com/alsa-project/alsa-ucm-conf/issues/313 Repository URL: https://github.com/alsa-project/alsa-ucm-conf

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[PATCH 0/4] ASoC: qcom: add display port module support.
by Srinivas Kandagatla 11 May '23

11 May '23

This patchset adds support to displayport on AudioReach. Patches are tested on X13s with two display ports. Srinivas Kandagatla (4): ASoC: qcom: q6dsp-common: move channel allocation to common ASoC: qcom: audioreach: add support for DISPLAY PORT SINK module ASoC: qcom: q6dsp: add support to more display ports ASoC: qcom: q6apm: add support to display ports in lpass dais .../sound/qcom,q6dsp-lpass-ports.h | 8 ++ sound/soc/qcom/qdsp6/audioreach.c | 75 +++++++++++++++++++ sound/soc/qcom/qdsp6/audioreach.h | 12 +++ sound/soc/qcom/qdsp6/q6afe-dai.c | 34 ++------- sound/soc/qcom/qdsp6/q6apm-lpass-dais.c | 39 ++++++++++ sound/soc/qcom/qdsp6/q6dsp-common.c | 35 +++++++++ sound/soc/qcom/qdsp6/q6dsp-common.h | 1 + sound/soc/qcom/qdsp6/q6dsp-lpass-ports.c | 43 +++++++---- 8 files changed, 204 insertions(+), 43 deletions(-) -- 2.21.0

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[alsa-devel@alsa-project.org: [PATCH 3/5] ASoC: mediatek: mt8195-afe-pcm: Simplify runtime PM during probe]
by Mark Brown 11 May '23

11 May '23

Hi, Here's another mail (one of several in this series) that got completely mangled by the alsa-project.org mailman to the point of unusability. I didn't see any response to my last mail about this, is there any news on fixing mailman to not do this? It's extremely disruptive to working with lore. Thanks, Mark

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[PATCH v1] ASoC: jz4740-i2s: Make I2S divider calculations more robust
by Aidan MacDonald 11 May '23

11 May '23

When the CPU supplies bit/frame clocks, the system clock (clk_i2s) is divided to produce the bit clock. This is a simple 1/N divider with a fairly limited range, so for a given system clock frequency only a few sample rates can be produced. Usually a wider range of sample rates is supported by varying the system clock frequency. The old calculation method was not very robust and could easily produce the wrong clock rate, especially with non-standard rates. For example, if the system clock is 1.99x the target bit clock rate, the divider would be calculated as 1 instead of the more accurate 2. Instead, use a more accurate method that considers two adjacent divider settings and selects the one that produces the least error versus the requested rate. If the error is 5% or higher then the rate setting is rejected to prevent garbled audio. Skip divider calculation when the codec is supplying both the bit and frame clock; in that case, the divider outputs are unused and we don't want to constrain the sample rate. Signed-off-by: Aidan MacDonald <aidanmacdonald.0x0(a)gmail.com> --- sound/soc/jz4740/jz4740-i2s.c | 54 ++++++++++++++++++++++++++++++++--- 1 file changed, 50 insertions(+), 4 deletions(-) diff --git a/sound/soc/jz4740/jz4740-i2s.c b/sound/soc/jz4740/jz4740-i2s.c index 6d9cfe0a5041..d0f6c945d9ae 100644 --- a/sound/soc/jz4740/jz4740-i2s.c +++ b/sound/soc/jz4740/jz4740-i2s.c @@ -218,18 +218,48 @@ static int jz4740_i2s_set_fmt(struct snd_soc_dai *dai, unsigned int fmt) return 0; } +static int jz4740_i2s_get_i2sdiv(unsigned long mclk, unsigned long rate, + unsigned long i2sdiv_max) +{ + unsigned long div, rate1, rate2, err1, err2; + + div = mclk / (64 * rate); + if (div == 0) + div = 1; + + rate1 = mclk / (64 * div); + rate2 = mclk / (64 * (div + 1)); + + err1 = abs(rate1 - rate); + err2 = abs(rate2 - rate); + + /* + * Choose the divider that produces the smallest error in the + * output rate and reject dividers with a 5% or higher error. + * In the event that both dividers are outside the acceptable + * error margin, reject the rate to prevent distorted audio. + * (The number 5% is arbitrary.) + */ + if (div <= i2sdiv_max && err1 <= err2 && err1 < rate/20) + return div; + if (div < i2sdiv_max && err2 < rate/20) + return div + 1; + + return -EINVAL; +} + static int jz4740_i2s_hw_params(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *params, struct snd_soc_dai *dai) { struct jz4740_i2s *i2s = snd_soc_dai_get_drvdata(dai); struct regmap_field *div_field; + unsigned long i2sdiv_max; unsigned int sample_size; - uint32_t ctrl; - int div; + uint32_t ctrl, conf; + int div = 1; regmap_read(i2s->regmap, JZ_REG_AIC_CTRL, &ctrl); - - div = clk_get_rate(i2s->clk_i2s) / (64 * params_rate(params)); + regmap_read(i2s->regmap, JZ_REG_AIC_CONF, &conf); switch (params_format(params)) { case SNDRV_PCM_FORMAT_S8: @@ -258,11 +288,27 @@ static int jz4740_i2s_hw_params(struct snd_pcm_substream *substream, ctrl &= ~JZ_AIC_CTRL_MONO_TO_STEREO; div_field = i2s->field_i2sdiv_playback; + i2sdiv_max = GENMASK(i2s->soc_info->field_i2sdiv_playback.msb, + i2s->soc_info->field_i2sdiv_playback.lsb); } else { ctrl &= ~JZ_AIC_CTRL_INPUT_SAMPLE_SIZE; ctrl |= FIELD_PREP(JZ_AIC_CTRL_INPUT_SAMPLE_SIZE, sample_size); div_field = i2s->field_i2sdiv_capture; + i2sdiv_max = GENMASK(i2s->soc_info->field_i2sdiv_capture.msb, + i2s->soc_info->field_i2sdiv_capture.lsb); + } + + /* + * Only calculate I2SDIV if we're supplying the bit or frame clock. + * If the codec is supplying both clocks then the divider output is + * unused, and we don't want it to limit the allowed sample rates. + */ + if (conf & (JZ_AIC_CONF_BIT_CLK_MASTER | JZ_AIC_CONF_SYNC_CLK_MASTER)) { + div = jz4740_i2s_get_i2sdiv(clk_get_rate(i2s->clk_i2s), + params_rate(params), i2sdiv_max); + if (div < 0) + return div; } regmap_write(i2s->regmap, JZ_REG_AIC_CTRL, ctrl); -- 2.39.2

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[PATCH] ASoC: dt-bindings: Adjust #sound-dai-cells on TI's single-DAI codecs
by Martin Povišer 11 May '23

11 May '23

A bunch of TI's codecs have binding schemas which force #sound-dai-cells to one despite those codecs only having a single DAI. Allow for bindings with zero DAI cells and deprecate the former non-zero value. Signed-off-by: Martin Povišer <povik+lin(a)cutebit.org> --- Documentation/devicetree/bindings/sound/tas2562.yaml | 6 ++++-- Documentation/devicetree/bindings/sound/tas2770.yaml | 6 ++++-- Documentation/devicetree/bindings/sound/tas27xx.yaml | 6 ++++-- 3 files changed, 12 insertions(+), 6 deletions(-) diff --git a/Documentation/devicetree/bindings/sound/tas2562.yaml b/Documentation/devicetree/bindings/sound/tas2562.yaml index a5bb561bfcfb..31a3024ea789 100644 --- a/Documentation/devicetree/bindings/sound/tas2562.yaml +++ b/Documentation/devicetree/bindings/sound/tas2562.yaml @@ -55,7 +55,9 @@ properties: description: TDM TX current sense time slot. '#sound-dai-cells': - const: 1 + # The codec has a single DAI, the #sound-dai-cells=<1>; case is left in for backward + # compatibility but is deprecated. + enum: [0, 1] required: - compatible @@ -72,7 +74,7 @@ examples: codec: codec@4c { compatible = "ti,tas2562"; reg = <0x4c>; - #sound-dai-cells = <1>; + #sound-dai-cells = <0>; interrupt-parent = <&gpio1>; interrupts = <14>; shutdown-gpios = <&gpio1 15 0>; diff --git a/Documentation/devicetree/bindings/sound/tas2770.yaml b/Documentation/devicetree/bindings/sound/tas2770.yaml index 26088adb9dc2..8908bf1122e9 100644 --- a/Documentation/devicetree/bindings/sound/tas2770.yaml +++ b/Documentation/devicetree/bindings/sound/tas2770.yaml @@ -57,7 +57,9 @@ properties: - 1 # Falling edge '#sound-dai-cells': - const: 1 + # The codec has a single DAI, the #sound-dai-cells=<1>; case is left in for backward + # compatibility but is deprecated. + enum: [0, 1] required: - compatible @@ -74,7 +76,7 @@ examples: codec: codec@41 { compatible = "ti,tas2770"; reg = <0x41>; - #sound-dai-cells = <1>; + #sound-dai-cells = <0>; interrupt-parent = <&gpio1>; interrupts = <14>; reset-gpio = <&gpio1 15 0>; diff --git a/Documentation/devicetree/bindings/sound/tas27xx.yaml b/Documentation/devicetree/bindings/sound/tas27xx.yaml index 8cba01316855..a876545ec87d 100644 --- a/Documentation/devicetree/bindings/sound/tas27xx.yaml +++ b/Documentation/devicetree/bindings/sound/tas27xx.yaml @@ -50,7 +50,9 @@ properties: description: TDM TX voltage sense time slot. '#sound-dai-cells': - const: 1 + # The codec has a single DAI, the #sound-dai-cells=<1>; case is left in for backward + # compatibility but is deprecated. + enum: [0, 1] required: - compatible @@ -67,7 +69,7 @@ examples: codec: codec@38 { compatible = "ti,tas2764"; reg = <0x38>; - #sound-dai-cells = <1>; + #sound-dai-cells = <0>; interrupt-parent = <&gpio1>; interrupts = <14>; reset-gpios = <&gpio1 15 0>; -- 2.38.3

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[PATCH v3 0/3] Add TDM audio on StarFive JH7110
by Walker Chen 11 May '23

11 May '23

This patchset adds TDM audio driver for the StarFive JH7110 SoC. The first patch adds device tree binding for TDM module. The second patch adds tdm driver support for JH7110 SoC. The last patch adds device node of tdm and sound card to JH7110 dts. The series has been tested on the VisionFive 2 board by plugging an audio expansion board. For more information of audio expansion board, you can take a look at the following webpage: https://wiki.seeedstudio.com/ReSpeaker_2_Mics_Pi_HAT/ Changes since v2: - Use dt-overlay to describe sound card because need to plug the audio expansion board into the VisionFive2 board. - Modified the coding style for driver. - Moved assignment of stop_dma_first to startup function of dai_driver. - Dropped some useless macro definition. - Use loops to get/enable/disable clocks. Changes since v1: - Rebased on Linux 6.3-rc4. - Added the dts file dedicated to describe audio card. - Added the item for JH7110 audio board to the dt-binding of StarFive SoC-based boards. --- v2: https://lore.kernel.org/all/20230420024118.22677-1-walker.chen@starfivetech… v1: https://lore.kernel.org/all/20230329153320.31390-1-walker.chen@starfivetech… Walker Chen (3): dt-bindings: sound: Add TDM for StarFive JH7110 ASoC: starfive: Add JH7110 TDM driver riscv: dts: starfive: add tdm node and sound card .../bindings/sound/starfive,jh7110-tdm.yaml | 98 +++ MAINTAINERS | 6 + arch/riscv/boot/dts/starfive/Makefile | 3 + .../jh7110-starfive-visionfive-2-wm8960.dtso | 70 +++ .../jh7110-starfive-visionfive-2.dtsi | 40 ++ arch/riscv/boot/dts/starfive/jh7110.dtsi | 21 + sound/soc/Kconfig | 1 + sound/soc/Makefile | 1 + sound/soc/starfive/Kconfig | 15 + sound/soc/starfive/Makefile | 2 + sound/soc/starfive/jh7110_tdm.c | 573 ++++++++++++++++++ sound/soc/starfive/jh7110_tdm.h | 147 +++++ 12 files changed, 977 insertions(+) create mode 100644 Documentation/devicetree/bindings/sound/starfive,jh7110-tdm.yaml create mode 100644 arch/riscv/boot/dts/starfive/jh7110-starfive-visionfive-2-wm8960.dtso create mode 100644 sound/soc/starfive/Kconfig create mode 100644 sound/soc/starfive/Makefile create mode 100644 sound/soc/starfive/jh7110_tdm.c create mode 100644 sound/soc/starfive/jh7110_tdm.h base-commit: 197b6b60ae7bc51dd0814953c562833143b292aa prerequisite-patch-id: 30bec4dba6f250a6edd0c2cbab2ce09442e50e8a prerequisite-patch-id: bb939c0c7c26b08addfccd890f9d3974b6eaec53 prerequisite-patch-id: 8a6f135bcabdad4a4bfb21f0c6a0ffd2bb57efe7 prerequisite-patch-id: c2366f993a9d85e28c06d8d09f064dd5e8b29a61 prerequisite-patch-id: 50d53a21f91f4087fc80b6f1f72864adfb0002b9 prerequisite-patch-id: 0df3703af91c30f1ca2c47f5609012f2d7200028 prerequisite-patch-id: 89f049f951e5acf75aab92541992f816fd0acc0d prerequisite-patch-id: 551fae54377090044c3612fca9740a9b359abdd2 prerequisite-patch-id: c7fdf904f398d478f0ed6d57eb878982bc73329d prerequisite-patch-id: 1b2d0982b18da060c82134f05bf3ce16425bac8d prerequisite-patch-id: 090ba4b78d47bc19204916e76fdbc70021785388 prerequisite-patch-id: a5d9e0f7d4f8163f566678894cf693015119f2d9 prerequisite-patch-id: 4637a8fa2334a45fa6b64351f4e9e28d3e2d60d3 prerequisite-patch-id: 32647ec60a3b614e1c59ec8e54cb511ae832c22f prerequisite-patch-id: aa06658ecf89c92d0dfdd6a4ba6d9e6e67532971 prerequisite-patch-id: 1387a7e87b446329dfc21f3e575ceae7ebcf954c prerequisite-patch-id: 258ea5f9b8bf41b6981345dcc81795f25865d38f prerequisite-patch-id: 8b6f2c9660c0ac0ee4e73e4c21aca8e6b75e81b9 prerequisite-patch-id: dbb0c0151b8bdf093e6ce79fd2fe3f60791a6e0b prerequisite-patch-id: 9007c8610fdcd387592475949864edde874c20a2 prerequisite-patch-id: d57e95d31686772abc4c4d5aa1cadc344dc293cd prerequisite-patch-id: 9f911969d0a550648493952c99096d26e05d4d83 prerequisite-patch-id: 2ddada18ab6ea5cd1da14212aaf59632f5203d40 prerequisite-patch-id: 80042661ff6156ce577a72e9eb8c0b218b624829 prerequisite-patch-id: 398744c61913c76a35754de867c4f820ca7a8d99 prerequisite-patch-id: f59269382164b5d642a5e10443ca447f5caa595c prerequisite-patch-id: 1babe83d6bf999bad17584dc595480f9070a5369 prerequisite-patch-id: 77be3d122d66df813f13088141ce27b21107a341 prerequisite-patch-id: 9fbb7ad1dd258bb8ff5946c4a0e59de4bfd82a04 prerequisite-patch-id: 6f6984916dffd0cc66aa733c9b6bd3a55495a50c prerequisite-patch-id: 39e1be2a3d1593577ab997f55f59367cba665aa7 prerequisite-patch-id: 584c256c9acb52ee2773d0c81c3f4977fc18155a prerequisite-patch-id: b37ac15032973e1fcd918f157c82a0606775c9e9 prerequisite-patch-id: 32deea16304859842af5c2151bc41d91cf6dfc9b prerequisite-patch-id: 20ac2450fb93b3f69f83fc720fd4800a95e618a6 prerequisite-patch-id: 6abf359fa445f4104432ddee27044dfbfb128417 -- 2.17.1

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[PATCH 1/2] ASoC: dt-bindings: Add adi,ssm3515 amp schema
by Martin Povišer 11 May '23

11 May '23

Add a DT schema for the SSM3515 amp by Analog Devices. It's a simple mono amplifier with digital input. Signed-off-by: Martin Povišer <povik+lin(a)cutebit.org> --- .../bindings/sound/adi,ssm3515.yaml | 66 +++++++++++++++++++ 1 file changed, 66 insertions(+) create mode 100644 Documentation/devicetree/bindings/sound/adi,ssm3515.yaml diff --git a/Documentation/devicetree/bindings/sound/adi,ssm3515.yaml b/Documentation/devicetree/bindings/sound/adi,ssm3515.yaml new file mode 100644 index 000000000000..19b7185ae8e2 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/adi,ssm3515.yaml @@ -0,0 +1,66 @@ +# SPDX-License-Identifier: (GPL-2.0-only OR BSD-2-Clause) +# Copyright (C) 2019-20 Texas Instruments Incorporated +%YAML 1.2 +--- +$id: "http://devicetree.org/schemas/sound/adi,ssm3515.yaml#" +$schema: "http://devicetree.org/meta-schemas/core.yaml#" + +title: Analog Devices SSM3515 Audio Amplifier + +maintainers: + - Martin Povišer <povik+lin(a)cutebit.org> + +description: | + SSM3515 is a mono Class-D audio amplifier with digital input. + + https://www.analog.com/media/en/technical-documentation/data-sheets/SSM3515… + +allOf: + - $ref: dai-common.yaml# + +properties: + compatible: + enum: + - adi,ssm3515 + + reg: + maxItems: 1 + + adi,ana-gain: + $ref: /schemas/types.yaml#/definitions/uint32 + enum: [0, 1, 2, 3] + description: | + The value to be set in the ANA_GAIN register field on the codec. This determines + the full voltage span of the codec's analog output. + + To quote the datasheet on the available options: + + 00: 8.4 V Full-Scale Gain Mapping + 01: 12.6 V Full-Scale Gain Mapping + 10: 14 V Full-Scale Gain Mapping + 11: 15 V Full-Scale Gain Mapping + + '#sound-dai-cells': + const: 0 + +required: + - compatible + - reg + - adi,ana-gain + +additionalProperties: true + +examples: + - | + i2c { + #address-cells = <1>; + #size-cells = <0>; + + codec@14 { + compatible = "adi,ssm3515"; + reg = <0x14>; + #sound-dai-cells = <0>; + adi,ana-gain = <0>; + sound-name-prefix = "Left Tweeter"; + }; + }; -- 2.38.3

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[PATCH AUTOSEL 5.15 2/4] soundwire: qcom: gracefully handle too many ports in DT
by Sasha Levin 10 May '23

10 May '23

From: Krzysztof Kozlowski <krzysztof.kozlowski(a)linaro.org> [ Upstream commit 2367e0ecb498764e95cfda691ff0828f7d25f9a4 ] There are two issues related to the number of ports coming from Devicetree when exceeding in total QCOM_SDW_MAX_PORTS. Both lead to incorrect memory accesses: 1. With DTS having too big value of input or output ports, the driver, when copying port parameters from local/stack arrays into 'pconfig' array in 'struct qcom_swrm_ctrl', will iterate over their sizes. 2. If DTS also has too many parameters for these ports (e.g. qcom,ports-sinterval-low), the driver will overflow buffers on the stack when reading these properties from DTS. Add a sanity check so incorrect DTS will not cause kernel memory corruption. Signed-off-by: Krzysztof Kozlowski <krzysztof.kozlowski(a)linaro.org> Reviewed-by: Srinivas Kandagatla <srinivas.kandagatla(a)linaro.org> Reviewed-by: Konrad Dybcio <konrad.dybcio(a)linaro.org> Link: https://lore.kernel.org/r/20230222144412.237832-2-krzysztof.kozlowski@linar… Signed-off-by: Vinod Koul <vkoul(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- drivers/soundwire/qcom.c | 3 +++ 1 file changed, 3 insertions(+) diff --git a/drivers/soundwire/qcom.c b/drivers/soundwire/qcom.c index 500035a1fd460..e76d4cee5e7f4 100644 --- a/drivers/soundwire/qcom.c +++ b/drivers/soundwire/qcom.c @@ -1142,6 +1142,9 @@ static int qcom_swrm_get_port_config(struct qcom_swrm_ctrl *ctrl) ctrl->num_dout_ports = val; nports = ctrl->num_dout_ports + ctrl->num_din_ports; + if (nports > QCOM_SDW_MAX_PORTS) + return -EINVAL; + /* Valid port numbers are from 1-14, so mask out port 0 explicitly */ set_bit(0, &ctrl->dout_port_mask); set_bit(0, &ctrl->din_port_mask); -- 2.39.2

1 0

[PATCH AUTOSEL 6.1 1/8] soundwire: dmi-quirks: add remapping for Intel 'Rooks County' NUC M15
by Sasha Levin 10 May '23

10 May '23

From: Eugene Huang <eugene.huang99(a)gmail.com> [ Upstream commit 01b33e284ca28cc977bdcfb23be2c719f2139175 ] Same DSDT problem as the HP Omen 16-k0005TX, except rt1316 amp is on link2. Link: https://github.com/thesofproject/linux/issues/4088 Signed-off-by: Eugene Huang <eugene.huang99(a)gmail.com> Reviewed-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> Reviewed-by: Péter Ujfalusi <peter.ujfalusi(a)linux.intel.com> Signed-off-by: Bard Liao <yung-chuan.liao(a)linux.intel.com> Link: https://lore.kernel.org/r/20230314090618.498716-1-yung-chuan.liao@linux.int… Signed-off-by: Vinod Koul <vkoul(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- drivers/soundwire/dmi-quirks.c | 25 +++++++++++++++++++++++++ 1 file changed, 25 insertions(+) diff --git a/drivers/soundwire/dmi-quirks.c b/drivers/soundwire/dmi-quirks.c index 7969881f126dc..58ea013fa918a 100644 --- a/drivers/soundwire/dmi-quirks.c +++ b/drivers/soundwire/dmi-quirks.c @@ -73,6 +73,23 @@ static const struct adr_remap hp_omen_16[] = { {} }; +/* + * Intel NUC M15 LAPRC510 and LAPRC710 + */ +static const struct adr_remap intel_rooks_county[] = { + /* rt711-sdca on link0 */ + { + 0x000020025d071100ull, + 0x000030025d071101ull + }, + /* rt1316-sdca on link2 */ + { + 0x000120025d071100ull, + 0x000230025d131601ull + }, + {} +}; + static const struct dmi_system_id adr_remap_quirk_table[] = { /* TGL devices */ { @@ -98,6 +115,14 @@ static const struct dmi_system_id adr_remap_quirk_table[] = { }, .driver_data = (void *)intel_tgl_bios, }, + { + /* quirk used for NUC15 'Rooks County' LAPRC510 and LAPRC710 skews */ + .matches = { + DMI_MATCH(DMI_SYS_VENDOR, "Intel(R) Client Systems"), + DMI_MATCH(DMI_PRODUCT_NAME, "LAPRC"), + }, + .driver_data = (void *)intel_rooks_county, + }, { .matches = { DMI_MATCH(DMI_SYS_VENDOR, "Dell Inc"), -- 2.39.2

1 2

[PATCH AUTOSEL 6.2 5/9] soundwire: bus: Fix unbalanced pm_runtime_put() causing usage count underflow
by Sasha Levin 10 May '23

10 May '23

From: Richard Fitzgerald <rf(a)opensource.cirrus.com> [ Upstream commit e9537962519e88969f5f69cd0571eb4f6984403c ] This reverts commit 443a98e649b4 ("soundwire: bus: use pm_runtime_resume_and_get()") Change calls to pm_runtime_resume_and_get() back to pm_runtime_get_sync(). This fixes a usage count underrun caused by doing a pm_runtime_put() even though pm_runtime_resume_and_get() returned an error. The three affected functions ignore -EACCES error from trying to get pm_runtime, and carry on, including a put at the end of the function. But pm_runtime_resume_and_get() does not increment the usage count if it returns an error. So in the -EACCES case you must not call pm_runtime_put(). The documentation for pm_runtime_get_sync() says: "Consider using pm_runtime_resume_and_get() ... as this is likely to result in cleaner code." In this case I don't think it results in cleaner code because the pm_runtime_put() at the end of the function would have to be conditional on the return value from pm_runtime_resume_and_get() at the top of the function. pm_runtime_get_sync() doesn't have this problem because it always increments the count, so always needs a put. The code can just flow through and do the pm_runtime_put() unconditionally. Signed-off-by: Richard Fitzgerald <rf(a)opensource.cirrus.com> Reviewed-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> Link: https://lore.kernel.org/r/20230406134640.8582-1-rf@opensource.cirrus.com Signed-off-by: Vinod Koul <vkoul(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- drivers/soundwire/bus.c | 15 ++++++++++----- 1 file changed, 10 insertions(+), 5 deletions(-) diff --git a/drivers/soundwire/bus.c b/drivers/soundwire/bus.c index 76515c33e639e..4fd221d0cc818 100644 --- a/drivers/soundwire/bus.c +++ b/drivers/soundwire/bus.c @@ -571,9 +571,11 @@ int sdw_nread(struct sdw_slave *slave, u32 addr, size_t count, u8 *val) { int ret; - ret = pm_runtime_resume_and_get(&slave->dev); - if (ret < 0 && ret != -EACCES) + ret = pm_runtime_get_sync(&slave->dev); + if (ret < 0 && ret != -EACCES) { + pm_runtime_put_noidle(&slave->dev); return ret; + } ret = sdw_nread_no_pm(slave, addr, count, val); @@ -595,9 +597,11 @@ int sdw_nwrite(struct sdw_slave *slave, u32 addr, size_t count, const u8 *val) { int ret; - ret = pm_runtime_resume_and_get(&slave->dev); - if (ret < 0 && ret != -EACCES) + ret = pm_runtime_get_sync(&slave->dev); + if (ret < 0 && ret != -EACCES) { + pm_runtime_put_noidle(&slave->dev); return ret; + } ret = sdw_nwrite_no_pm(slave, addr, count, val); @@ -1565,9 +1569,10 @@ static int sdw_handle_slave_alerts(struct sdw_slave *slave) sdw_modify_slave_status(slave, SDW_SLAVE_ALERT); - ret = pm_runtime_resume_and_get(&slave->dev); + ret = pm_runtime_get_sync(&slave->dev); if (ret < 0 && ret != -EACCES) { dev_err(&slave->dev, "Failed to resume device: %d\n", ret); + pm_runtime_put_noidle(&slave->dev); return ret; } -- 2.39.2

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