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[PATCH 00/11] ASoC: use pm_runtime_resume_and_get() when possible
by Pierre-Louis Bossart 28 Jun '22

28 Jun '22

After a set of SOF-specific changes, this patchset correct problematic uses of pm_runtime_get_sync() in ASoC, or simplifies the flow with no functional changes. Two patches for Intel platforms also add a test on resume success. Additional changes were initially suggested to completely remove the use of pm_runtime_get_sync(). These changes were dropped since they are way too invasive, specifically in cases where the return values were not tested, which would lead to duplicate pm_runtime_put(). The remaining uses of pm_runtime_get_sync() cannot really be blindly modified without context and knowledge of each driver. Pierre-Louis Bossart (11): ASoC: Intel: catpt: use pm_runtime_resume_and_get() ASoC: Intel: skylake: skl-pcm: use pm_runtime_resume_and_get() ASoC: soc-component: use pm_runtime_resume_and_get() ASoC: wcd-mbhc-v2: use pm_runtime_resume_and_get() ASoC: wsa881x: use pm_runtime_resume_and_get() ASoC: rockchip: i2s_tdm: use pm_runtime_resume_and_get() ASoC: fsl: fsl_sai: use pm_runtime_resume_and_get() ASoC: img: img-i2s-out: use pm_runtime_resume_and_get() ASoC: rockchip: pdm: use pm_runtime_resume_and_get() ASoC: tas2552: use pm_runtime_resume_and_get() ASoC: ti: davinci-mcasp: use pm_runtime_resume_and_get() sound/soc/codecs/tas2552.c | 2 +- sound/soc/codecs/wcd-mbhc-v2.c | 10 ++++------ sound/soc/codecs/wsa881x.c | 6 ++---- sound/soc/fsl/fsl_sai.c | 6 ++---- sound/soc/img/img-i2s-out.c | 12 ++++-------- sound/soc/intel/catpt/pcm.c | 26 ++++++++++++++++++++------ sound/soc/intel/catpt/sysfs.c | 4 +++- sound/soc/intel/skylake/skl-pcm.c | 5 ++++- sound/soc/rockchip/rockchip_i2s_tdm.c | 6 ++---- sound/soc/rockchip/rockchip_pdm.c | 6 ++---- sound/soc/soc-component.c | 8 ++++---- sound/soc/ti/davinci-mcasp.c | 3 +-- 12 files changed, 49 insertions(+), 45 deletions(-) -- 2.34.1

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[PATCH v2 0/7] Add PDM/DSD/dataline configuration support
by Shengjiu Wang 28 Jun '22

28 Jun '22

Support PDM format and DSD format. Add new dts property to configure dataline. The SAI has multiple successive FIFO registers, but in some use case the required dataline/FIFOs are not successive. Changes in v2: - refine the commit subject of patch 5/7 Shengjiu Wang (7): ASoC: fsl_sai: Add PDM daifmt support ASoC: fsl_sai: Add DSD bit format support ASoC: fsl_sai: Add support for more sample rates ASoc: fsl_sai: Add pinctrl operation for PDM and DSD ASoC: fsl_sai: Make res a member of struct fsl_sai ASoC: dt-bindings: fsl-sai: Add new property to configure dataline ASoC: fsl_sai: Configure dataline/FIFO information from dts property .../devicetree/bindings/sound/fsl-sai.txt | 8 + sound/soc/fsl/fsl_sai.c | 260 ++++++++++++++++-- sound/soc/fsl/fsl_sai.h | 26 +- 3 files changed, 272 insertions(+), 22 deletions(-) -- 2.17.1

4 10

[PATCH 0/7] Add PDM/DSD/dataline configuration support
by Shengjiu Wang 28 Jun '22

28 Jun '22

Support PDM format and DSD format. Add new dts property to configure dataline. The SAI has multiple successive FIFO registers, but in some use case the required dataline/FIFOs are not successive. Shengjiu Wang (7): ASoC: fsl_sai: Add PDM daifmt support ASoC: fsl_sai: Add DSD bit format support ASoC: fsl_sai: Add support for more sample rates ASoc: fsl_sai: Add pinctrl operation for PDM and DSD ASoC: fsl_sai: Move res variable to be global ASoC: dt-bindings: fsl-sai: Add new property to configure dataline ASoC: fsl_sai: Configure dataline/FIFO information from dts property .../devicetree/bindings/sound/fsl-sai.txt | 8 + sound/soc/fsl/fsl_sai.c | 260 ++++++++++++++++-- sound/soc/fsl/fsl_sai.h | 26 +- 3 files changed, 272 insertions(+), 22 deletions(-) -- 2.17.1

4 10

[PATCH v2] ASoC: samsung: h1940_uda1380: Include proper GPIO consumer header
by Krzysztof Kozlowski 28 Jun '22

28 Jun '22

h1940_uda1380 uses gpiod*/GPIOD* so it should include GPIO consumer header. Fixes: 9666e27f90b9 ("ASoC: samsung: h1940: turn into platform driver") Signed-off-by: Krzysztof Kozlowski <krzysztof.kozlowski(a)linaro.org> --- Changes since v1: 1. Fix typo in commit subject. --- sound/soc/samsung/h1940_uda1380.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/samsung/h1940_uda1380.c b/sound/soc/samsung/h1940_uda1380.c index 907266aee839..fa45a54ab18f 100644 --- a/sound/soc/samsung/h1940_uda1380.c +++ b/sound/soc/samsung/h1940_uda1380.c @@ -8,7 +8,7 @@ // Based on version from Arnaud Patard <arnaud.patard(a)rtp-net.org> #include <linux/types.h> -#include <linux/gpio.h> +#include <linux/gpio/consumer.h> #include <linux/module.h> #include <sound/soc.h> -- 2.34.1

1 0

[PATCH 0/4] seq: remove-filter: rewrite as object
by GitHub pull_request - opened 28 Jun '22

28 Jun '22

alsa-project/alsa-gobject pull request #81 was opened from takaswie: Current implementation of ALSASeq.RemoveFilter doesn't support multiple condition to query events for removal. It's not necessarily convenient. This commit rewrites ALSASeq.RemoveFilter. ``` Takashi Sakamoto (4): seq: event-cntr: remove obsoleted internal function seq: enums: add rest of entries for remove filter flags seq: remove-filter: rewrite as GObject-derived object update README README.rst | 9 +- src/seq/alsaseq-enum-types.h | 26 ++- src/seq/alsaseq.map | 17 +- src/seq/privates.h | 7 +- src/seq/remove-filter.c | 388 +++++++++++++++++++++-------------- src/seq/remove-filter.h | 35 +--- src/seq/user-client.c | 7 +- tests/alsaseq-enums | 8 + tests/alsaseq-remove-filter | 25 ++- 9 files changed, 305 insertions(+), 217 deletions(-) ``` Request URL : https://github.com/alsa-project/alsa-gobject/pull/81 Patch URL : https://github.com/alsa-project/alsa-gobject/pull/81.patch Repository URL: https://github.com/alsa-project/alsa-gobject

1 0

[PATCH v2] ASoC: dt-bindings: fsl, micfil: Convert format to json-schema
by Shengjiu Wang 28 Jun '22

28 Jun '22

Convert the NXP MICFIL binding to DT schema format using json-schema. Signed-off-by: Shengjiu Wang <shengjiu.wang(a)nxp.com> --- changes in v2 - add decription of interrupts. .../devicetree/bindings/sound/fsl,micfil.txt | 33 -------- .../devicetree/bindings/sound/fsl,micfil.yaml | 77 +++++++++++++++++++ 2 files changed, 77 insertions(+), 33 deletions(-) delete mode 100644 Documentation/devicetree/bindings/sound/fsl,micfil.txt create mode 100644 Documentation/devicetree/bindings/sound/fsl,micfil.yaml diff --git a/Documentation/devicetree/bindings/sound/fsl,micfil.txt b/Documentation/devicetree/bindings/sound/fsl,micfil.txt deleted file mode 100644 index 1ea05d4996c7..000000000000 --- a/Documentation/devicetree/bindings/sound/fsl,micfil.txt +++ /dev/null @@ -1,33 +0,0 @@ -NXP MICFIL Digital Audio Interface (MICFIL). - -The MICFIL digital interface provides a 16-bit audio signal from a PDM -microphone bitstream in a configurable output sampling rate. - -Required properties: - - - compatible : Compatible list, contains "fsl,imx8mm-micfil" - or "fsl,imx8mp-micfil" - - - reg : Offset and length of the register set for the device. - - - interrupts : Contains the micfil interrupts. - - - clocks : Must contain an entry for each entry in clock-names. - - - clock-names : Must include the "ipg_clk" for register access and - "ipg_clk_app" for internal micfil clock. - - - dmas : Generic dma devicetree binding as described in - Documentation/devicetree/bindings/dma/dma.txt. - -Example: -micfil: micfil@30080000 { - compatible = "fsl,imx8mm-micfil"; - reg = <0x0 0x30080000 0x0 0x10000>; - interrupts = <GIC_SPI 109 IRQ_TYPE_LEVEL_HIGH>, - <GIC_SPI 110 IRQ_TYPE_LEVEL_HIGH>; - clocks = <&clk IMX8MM_CLK_PDM_IPG>, - <&clk IMX8MM_CLK_PDM_ROOT>; - clock-names = "ipg_clk", "ipg_clk_app"; - dmas = <&sdma2 24 26 0x80000000>; -}; diff --git a/Documentation/devicetree/bindings/sound/fsl,micfil.yaml b/Documentation/devicetree/bindings/sound/fsl,micfil.yaml new file mode 100644 index 000000000000..bf24faed7a4d --- /dev/null +++ b/Documentation/devicetree/bindings/sound/fsl,micfil.yaml @@ -0,0 +1,77 @@ +# SPDX-License-Identifier: (GPL-2.0-only OR BSD-2-Clause) +%YAML 1.2 +--- +$id: http://devicetree.org/schemas/sound/fsl,micfil.yaml# +$schema: http://devicetree.org/meta-schemas/core.yaml# + +title: NXP MICFIL Digital Audio Interface (MICFIL) + +maintainers: + - Shengjiu Wang <shengjiu.wang(a)nxp.com> + +description: | + The MICFIL digital interface provides a 16-bit or 24-bit audio signal + from a PDM microphone bitstream in a configurable output sampling rate. + +properties: + compatible: + enum: + - fsl,imx8mm-micfil + - fsl,imx8mp-micfil + + reg: + maxItems: 1 + + interrupts: + items: + - description: Digital Microphone interface interrupt + - description: Digital Microphone interface error interrupt + - description: voice activity detector event interrupt + - description: voice activity detector error interrupt + + dmas: + items: + - description: DMA controller phandle and request line for RX + + dma-names: + items: + - const: rx + + clocks: + items: + - description: The ipg clock for register access + - description: internal micfil clock + + clock-names: + items: + - const: ipg_clk + - const: ipg_clk_app + +required: + - compatible + - reg + - interrupts + - dmas + - dma-names + - clocks + - clock-names + +additionalProperties: false + +examples: + - | + #include <dt-bindings/interrupt-controller/arm-gic.h> + #include <dt-bindings/clock/imx8mm-clock.h> + micfil: audio-controller@30080000 { + compatible = "fsl,imx8mm-micfil"; + reg = <0x30080000 0x10000>; + interrupts = <GIC_SPI 109 IRQ_TYPE_LEVEL_HIGH>, + <GIC_SPI 110 IRQ_TYPE_LEVEL_HIGH>, + <GIC_SPI 44 IRQ_TYPE_LEVEL_HIGH>, + <GIC_SPI 45 IRQ_TYPE_LEVEL_HIGH>; + clocks = <&clk IMX8MM_CLK_PDM_IPG>, + <&clk IMX8MM_CLK_PDM_ROOT>; + clock-names = "ipg_clk", "ipg_clk_app"; + dmas = <&sdma2 24 25 0>; + dma-names = "rx"; + }; -- 2.17.1

3 2

[PATCH AUTOSEL 4.14 05/17] ALSA: x86: intel_hdmi_audio: enable pm_runtime and set autosuspend delay
by Sasha Levin 28 Jun '22

28 Jun '22

From: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> [ Upstream commit e87c65aeb46ca4f5b7dc08531200bcb8a426c62e ] The existing code uses pm_runtime_get_sync/put_autosuspend, but pm_runtime was not explicitly enabled. The autosuspend delay was not set either, the value is set to 5s since HDMI is rather painful to resume. Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> Reviewed-by: Bard Liao <yung-chuan.liao(a)linux.intel.com> Reviewed-by: Kai Vehmanen <kai.vehmanen(a)linux.intel.com> Reviewed-by: Ranjani Sridharan <ranjani.sridharan(a)linux.intel.com> Link: https://lore.kernel.org/r/20220616222910.136854-2-pierre-louis.bossart@linu… Signed-off-by: Takashi Iwai <tiwai(a)suse.de> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/x86/intel_hdmi_audio.c | 5 +++++ 1 file changed, 5 insertions(+) diff --git a/sound/x86/intel_hdmi_audio.c b/sound/x86/intel_hdmi_audio.c index bf894087e5fa..84ac01b6e93d 100644 --- a/sound/x86/intel_hdmi_audio.c +++ b/sound/x86/intel_hdmi_audio.c @@ -42,6 +42,8 @@ #include <drm/intel_lpe_audio.h> #include "intel_hdmi_audio.h" +#define INTEL_HDMI_AUDIO_SUSPEND_DELAY_MS 5000 + #define for_each_pipe(card_ctx, pipe) \ for ((pipe) = 0; (pipe) < (card_ctx)->num_pipes; (pipe)++) #define for_each_port(card_ctx, port) \ @@ -1885,8 +1887,11 @@ static int hdmi_lpe_audio_probe(struct platform_device *pdev) pdata->notify_audio_lpe = notify_audio_lpe; spin_unlock_irq(&pdata->lpe_audio_slock); + pm_runtime_set_autosuspend_delay(&pdev->dev, INTEL_HDMI_AUDIO_SUSPEND_DELAY_MS); pm_runtime_use_autosuspend(&pdev->dev); + pm_runtime_enable(&pdev->dev); pm_runtime_mark_last_busy(&pdev->dev); + pm_runtime_idle(&pdev->dev); dev_dbg(&pdev->dev, "%s: handle pending notification\n", __func__); for_each_port(card_ctx, port) { -- 2.35.1

1 0

[PATCH AUTOSEL 4.14 03/17] ALSA: usb-audio: US16x08: Move overflow check before array access
by Sasha Levin 28 Jun '22

28 Jun '22

From: Daniil Dementev <d.dementev(a)ispras.ru> [ Upstream commit 3ddbe35d9a2ebd4924d458e0246b4ba6c13bb456 ] Buffer overflow could occur in the loop "while", due to accessing an array element before checking the index. Found by Linux Verification Center (linuxtesting.org) with SVACE. Signed-off-by: Daniil Dementev <d.dementev(a)ispras.ru> Reviewed-by: Alexey Khoroshilov <khoroshilov(a)ispras.ru> Link: https://lore.kernel.org/r/20220610165732.2904-1-d.dementev@ispras.ru Signed-off-by: Takashi Iwai <tiwai(a)suse.de> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/usb/mixer_us16x08.c | 6 +++--- 1 file changed, 3 insertions(+), 3 deletions(-) diff --git a/sound/usb/mixer_us16x08.c b/sound/usb/mixer_us16x08.c index 7db3032e723a..5e3b7fd43fc2 100644 --- a/sound/usb/mixer_us16x08.c +++ b/sound/usb/mixer_us16x08.c @@ -647,10 +647,10 @@ static int snd_get_meter_comp_index(struct snd_us16x08_meter_store *store) } } else { /* skip channels with no compressor active */ - while (!store->comp_store->val[ + while (store->comp_index <= SND_US16X08_MAX_CHANNELS + && !store->comp_store->val[ COMP_STORE_IDX(SND_US16X08_ID_COMP_SWITCH)] - [store->comp_index - 1] - && store->comp_index <= SND_US16X08_MAX_CHANNELS) { + [store->comp_index - 1]) { store->comp_index++; } ret = store->comp_index++; -- 2.35.1

1 0

[PATCH AUTOSEL 4.19 06/22] ALSA: x86: intel_hdmi_audio: enable pm_runtime and set autosuspend delay
by Sasha Levin 28 Jun '22

28 Jun '22

From: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> [ Upstream commit e87c65aeb46ca4f5b7dc08531200bcb8a426c62e ] The existing code uses pm_runtime_get_sync/put_autosuspend, but pm_runtime was not explicitly enabled. The autosuspend delay was not set either, the value is set to 5s since HDMI is rather painful to resume. Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> Reviewed-by: Bard Liao <yung-chuan.liao(a)linux.intel.com> Reviewed-by: Kai Vehmanen <kai.vehmanen(a)linux.intel.com> Reviewed-by: Ranjani Sridharan <ranjani.sridharan(a)linux.intel.com> Link: https://lore.kernel.org/r/20220616222910.136854-2-pierre-louis.bossart@linu… Signed-off-by: Takashi Iwai <tiwai(a)suse.de> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/x86/intel_hdmi_audio.c | 5 +++++ 1 file changed, 5 insertions(+) diff --git a/sound/x86/intel_hdmi_audio.c b/sound/x86/intel_hdmi_audio.c index 3841336dc9cd..0c39ab628267 100644 --- a/sound/x86/intel_hdmi_audio.c +++ b/sound/x86/intel_hdmi_audio.c @@ -42,6 +42,8 @@ #include <drm/intel_lpe_audio.h> #include "intel_hdmi_audio.h" +#define INTEL_HDMI_AUDIO_SUSPEND_DELAY_MS 5000 + #define for_each_pipe(card_ctx, pipe) \ for ((pipe) = 0; (pipe) < (card_ctx)->num_pipes; (pipe)++) #define for_each_port(card_ctx, port) \ @@ -1898,8 +1900,11 @@ static int hdmi_lpe_audio_probe(struct platform_device *pdev) pdata->notify_audio_lpe = notify_audio_lpe; spin_unlock_irq(&pdata->lpe_audio_slock); + pm_runtime_set_autosuspend_delay(&pdev->dev, INTEL_HDMI_AUDIO_SUSPEND_DELAY_MS); pm_runtime_use_autosuspend(&pdev->dev); + pm_runtime_enable(&pdev->dev); pm_runtime_mark_last_busy(&pdev->dev); + pm_runtime_idle(&pdev->dev); dev_dbg(&pdev->dev, "%s: handle pending notification\n", __func__); for_each_port(card_ctx, port) { -- 2.35.1

1 0

[PATCH AUTOSEL 4.19 03/22] ALSA: usb-audio: US16x08: Move overflow check before array access
by Sasha Levin 28 Jun '22

28 Jun '22

From: Daniil Dementev <d.dementev(a)ispras.ru> [ Upstream commit 3ddbe35d9a2ebd4924d458e0246b4ba6c13bb456 ] Buffer overflow could occur in the loop "while", due to accessing an array element before checking the index. Found by Linux Verification Center (linuxtesting.org) with SVACE. Signed-off-by: Daniil Dementev <d.dementev(a)ispras.ru> Reviewed-by: Alexey Khoroshilov <khoroshilov(a)ispras.ru> Link: https://lore.kernel.org/r/20220610165732.2904-1-d.dementev@ispras.ru Signed-off-by: Takashi Iwai <tiwai(a)suse.de> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/usb/mixer_us16x08.c | 6 +++--- 1 file changed, 3 insertions(+), 3 deletions(-) diff --git a/sound/usb/mixer_us16x08.c b/sound/usb/mixer_us16x08.c index 7db3032e723a..5e3b7fd43fc2 100644 --- a/sound/usb/mixer_us16x08.c +++ b/sound/usb/mixer_us16x08.c @@ -647,10 +647,10 @@ static int snd_get_meter_comp_index(struct snd_us16x08_meter_store *store) } } else { /* skip channels with no compressor active */ - while (!store->comp_store->val[ + while (store->comp_index <= SND_US16X08_MAX_CHANNELS + && !store->comp_store->val[ COMP_STORE_IDX(SND_US16X08_ID_COMP_SWITCH)] - [store->comp_index - 1] - && store->comp_index <= SND_US16X08_MAX_CHANNELS) { + [store->comp_index - 1]) { store->comp_index++; } ret = store->comp_index++; -- 2.35.1

1 0