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

[PATCH AUTOSEL 5.8 035/132] ASoC: SOF: fix a runtime pm issue in SOF when HDMI codec doesn't work
by Sasha Levin 27 Oct '20

27 Oct '20

From: Rander Wang <rander.wang(a)intel.com> [ Upstream commit 6c63c954e1c52f1262f986f36d95f557c6f8fa94 ] When hda_codec_probe() doesn't initialize audio component, we disable the codec and keep going. However,the resources are not released. The child_count of SOF device is increased in snd_hdac_ext_bus_device_init but is not decrease in error case, so SOF can't get suspended. snd_hdac_ext_bus_device_exit will be invoked in HDA framework if it gets a error. Now copy this behavior to release resources and decrease SOF device child_count to release SOF device. Signed-off-by: Rander Wang <rander.wang(a)intel.com> Reviewed-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> Reviewed-by: Bard Liao <yung-chuan.liao(a)linux.intel.com> Reviewed-by: Guennadi Liakhovetski <guennadi.liakhovetski(a)linux.intel.com> Signed-off-by: Ranjani Sridharan <ranjani.sridharan(a)linux.intel.com> Link: https://lore.kernel.org/r/20200825235040.1586478-3-ranjani.sridharan@linux.… Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/sof/intel/hda-codec.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/soc/sof/intel/hda-codec.c b/sound/soc/sof/intel/hda-codec.c index 2c5c451fa19d7..c475955c6eeba 100644 --- a/sound/soc/sof/intel/hda-codec.c +++ b/sound/soc/sof/intel/hda-codec.c @@ -151,7 +151,7 @@ static int hda_codec_probe(struct snd_sof_dev *sdev, int address, if (!hdev->bus->audio_component) { dev_dbg(sdev->dev, "iDisp hw present but no driver\n"); - return -ENOENT; + goto error; } hda_priv->need_display_power = true; } @@ -174,7 +174,7 @@ static int hda_codec_probe(struct snd_sof_dev *sdev, int address, * other return codes without modification */ if (ret == 0) - ret = -ENOENT; + goto error; } return ret; -- 2.25.1

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[PATCH AUTOSEL 5.9 043/147] ASoC: AMD: Clean kernel log from deferred probe error messages
by Sasha Levin 27 Oct '20

27 Oct '20

From: Akshu Agrawal <akshu.agrawal(a)amd.com> [ Upstream commit f7660445c8e7fda91e8b944128554249d886b1d4 ] While the driver waits for DAIs to be probed and retries probing, have the error messages at debug level instead of error. Signed-off-by: Akshu Agrawal <akshu.agrawal(a)amd.com> Link: https://lore.kernel.org/r/20200826185454.5545-1-akshu.agrawal@amd.com Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/amd/acp3x-rt5682-max9836.c | 11 ++++++++--- 1 file changed, 8 insertions(+), 3 deletions(-) diff --git a/sound/soc/amd/acp3x-rt5682-max9836.c b/sound/soc/amd/acp3x-rt5682-max9836.c index 406526e79af34..1a4e8ca0f99c2 100644 --- a/sound/soc/amd/acp3x-rt5682-max9836.c +++ b/sound/soc/amd/acp3x-rt5682-max9836.c @@ -472,12 +472,17 @@ static int acp3x_probe(struct platform_device *pdev) ret = devm_snd_soc_register_card(&pdev->dev, card); if (ret) { - dev_err(&pdev->dev, + if (ret != -EPROBE_DEFER) + dev_err(&pdev->dev, "devm_snd_soc_register_card(%s) failed: %d\n", card->name, ret); - return ret; + else + dev_dbg(&pdev->dev, + "devm_snd_soc_register_card(%s) probe deferred: %d\n", + card->name, ret); } - return 0; + + return ret; } static const struct acpi_device_id acp3x_audio_acpi_match[] = { -- 2.25.1

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[PATCH] ASoC: amd: support other audio modes for raven
by Vijendar Mukunda 27 Oct '20

27 Oct '20

ACP supports different audio configurations other than I2S. This patch will fix acp driver probe failure for other audio configurations. Tested-by: Prike Liang <Prike.Liang(a)amd.com> Reviewed-by: Prike Liang <Prike.Liang(a)amd.com> Acked-by: Alex Deucher <alexander.deucher(a)amd.com> Signed-off-by: Vijendar Mukunda <Vijendar.Mukunda(a)amd.com> --- sound/soc/amd/raven/pci-acp3x.c | 4 +--- 1 file changed, 1 insertion(+), 3 deletions(-) diff --git a/sound/soc/amd/raven/pci-acp3x.c b/sound/soc/amd/raven/pci-acp3x.c index 31b797c..cd5b097 100644 --- a/sound/soc/amd/raven/pci-acp3x.c +++ b/sound/soc/amd/raven/pci-acp3x.c @@ -231,9 +231,7 @@ static int snd_acp3x_probe(struct pci_dev *pci, } break; default: - dev_err(&pci->dev, "Invalid ACP audio mode : %d\n", val); - ret = -ENODEV; - goto disable_msi; + dev_info(&pci->dev, "ACP audio mode : %d\n", val); } pm_runtime_set_autosuspend_delay(&pci->dev, 2000); pm_runtime_use_autosuspend(&pci->dev); -- 2.7.4

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

27 Oct '20

All platforms that use the jz4740-i2s driver have been switched to devicetree for a while now and the assignment of the DMA peripheral ID is done in the devicetree. It is no longer necessary to manually assign the peripheral ID in the driver, so remove that. The DMA driver does not even look at the value assigned in the driver anymore and always uses the value provided by the devicetree. Signed-off-by: Lars-Peter Clausen <lars(a)metafoo.de> --- sound/soc/jz4740/jz4740-i2s.c | 5 ----- 1 file changed, 5 deletions(-) diff --git a/sound/soc/jz4740/jz4740-i2s.c b/sound/soc/jz4740/jz4740-i2s.c index c7bd20104b20..e8bc7ca5ee5e 100644 --- a/sound/soc/jz4740/jz4740-i2s.c +++ b/sound/soc/jz4740/jz4740-i2s.c @@ -26,9 +26,6 @@ #include "jz4740-i2s.h" -#define JZ4740_DMA_TYPE_AIC_TRANSMIT 24 -#define JZ4740_DMA_TYPE_AIC_RECEIVE 25 - #define JZ_REG_AIC_CONF 0x00 #define JZ_REG_AIC_CTRL 0x04 #define JZ_REG_AIC_I2S_FMT 0x10 @@ -377,13 +374,11 @@ static void jz4740_i2c_init_pcm_config(struct jz4740_i2s *i2s) /* Playback */ dma_data = &i2s->playback_dma_data; dma_data->maxburst = 16; - dma_data->slave_id = JZ4740_DMA_TYPE_AIC_TRANSMIT; dma_data->addr = i2s->phys_base + JZ_REG_AIC_FIFO; /* Capture */ dma_data = &i2s->capture_dma_data; dma_data->maxburst = 16; - dma_data->slave_id = JZ4740_DMA_TYPE_AIC_RECEIVE; dma_data->addr = i2s->phys_base + JZ_REG_AIC_FIFO; } -- 2.20.1

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

27 Oct '20

Replace the two-step copy-and-convert in wm5102_out_comp_coeff_put() with get_unaligned_be16(). Apart from looking nicer, it avoids this sparse warning: wm5102.c:687:35: sparse: sparse: cast to restricted __be16 Signed-off-by: Richard Fitzgerald <rf(a)opensource.cirrus.com> --- sound/soc/codecs/wm5102.c | 4 +--- 1 file changed, 1 insertion(+), 3 deletions(-) diff --git a/sound/soc/codecs/wm5102.c b/sound/soc/codecs/wm5102.c index 2ed3fa67027d..70d353b63fe0 100644 --- a/sound/soc/codecs/wm5102.c +++ b/sound/soc/codecs/wm5102.c @@ -682,9 +682,7 @@ static int wm5102_out_comp_coeff_put(struct snd_kcontrol *kcontrol, struct arizona *arizona = dev_get_drvdata(component->dev->parent); mutex_lock(&arizona->dac_comp_lock); - memcpy(&arizona->dac_comp_coeff, ucontrol->value.bytes.data, - sizeof(arizona->dac_comp_coeff)); - arizona->dac_comp_coeff = be16_to_cpu(arizona->dac_comp_coeff); + arizona->dac_comp_coeff = get_unaligned_be16(ucontrol->value.bytes.data); mutex_unlock(&arizona->dac_comp_lock); return 0; base-commit: 2ebefde9de62a0828936e47b62975501993b5ee9 -- 2.20.1

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

27 Oct '20

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

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

27 Oct '20

Hi Mark soc_pcm_hw_params() does rollback when failed (A), but, it is almost same as soc_pcm_hw_free(). static int soc_pcm_hw_params(xxx) { ... if (ret < 0) goto xxx_err; ... return ret; ^ component_err: | ... | interface_err: (A) ... | codec_err: | ... v return ret; } This kind of duplicated code can be a hotbed of bugs, thus, this patch-set share soc_pcm_hw_free() and rollback. Kuninori Morimoto (6): ASoC: soc.h: remove for_each_rtd_dais_rollback() ASoC: soc-pcm: move soc_pcm_hw_free() next to soc_pcm_hw_params() ASoC: soc-link: add mark for snd_soc_link_hw_params/free() ASoC: soc-component: add mark for snd_soc_pcm_component_hw_params/free() ASoC: soc-dai: add mark for snd_soc_dai_hw_params/free() ASoC: soc-pcm: add soc_pcm_hw_clean() and call it from soc_pcm_hw_params/free() include/sound/soc-component.h | 6 +- include/sound/soc-dai.h | 4 +- include/sound/soc-link.h | 3 +- include/sound/soc.h | 7 +- sound/soc/soc-component.c | 19 ++--- sound/soc/soc-dai.c | 13 +++- sound/soc/soc-dapm.c | 4 +- sound/soc/soc-link.c | 12 +++- sound/soc/soc-pcm.c | 131 ++++++++++++++-------------------- 9 files changed, 97 insertions(+), 102 deletions(-) -- 2.25.1

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

27 Oct '20

This series adds support the other two AIFs present in the sun8i codec, which can be used for codec2codec DAI links. This series first fills out the DAI driver, removing assumptions that were made for AIF1 (16 bits, 2 channels, certain clock inversions). Some new logic is required to handle 3 DAIs and the ADC/DAC sharing the same clock. Finally, it adds the new DAIs, and hooks them up with DAPM widgets and routes per the hardware topology. To minimize the number of patches in this series, related device tree patches (increasing #sound-dai-cells, adding new DAI links) will be sent separately. Changes from v1: - Patches 1-8 from v1 (DAPM changes) were merged - Prefixed AIF constants with "SUN8I_CODEC_" [1, 7, 10, 16, 17] - Renamed variables in sun8i_codec_set_fmt for clarity [3] - Update sysclk->sysclk_rate if later calls to hw_params change the sample rate (thanks Chen-Yu for reminding me of this) [11] - Select COMMON_CLK for clk_set_rate_exclusive [12] - Add comments and hopefully clarify the clock protection logic [12] - Make the error message more concise and put it on one line [12] - Drop the "reg" variable holding SUN8I_AIF_CLK_CTRL(dai->id) [15] - Rename "div_reg" to "clk_reg" and adjust comments for clarity [17] - Improve the AIF2/AIF3 rate mismatch error message [17] - Minor grammar/wording fixes in commit messages [2, 4, 7, 8, 16, 17] - Added Maxime's Acked-by: [1-2, 4-9, 11, 13-14, 16] Samuel Holland (17): ASoC: sun8i-codec: Prepare to extend the DAI driver ASoC: sun8i-codec: Program DAI format before clock inversion ASoC: sun8i-codec: Enable all supported clock inversions ASoC: sun8i-codec: Use the provided word size ASoC: sun8i-codec: Round up the LRCK divisor ASoC: sun8i-codec: Correct the BCLK divisor calculation ASoC: sun8i-codec: Support the TDM slot binding ASoC: sun8i-codec: Enforce symmetric DAI parameters ASoC: sun8i-codec: Enable all supported sample rates ASoC: sun8i-codec: Automatically set the system sample rate ASoC: sun8i-codec: Constrain to compatible sample rates ASoC: sun8i-codec: Protect the clock rate while streams are open ASoC: sun8i-codec: Require an exact BCLK divisor match ASoC: sun8i-codec: Enable all supported PCM formats ASoC: sun8i-codec: Generalize AIF clock control ASoC: sun8i-codec: Add the AIF2 DAI, widgets, and routes ASoC: sun8i-codec: Add the AIF3 DAI, widgets, and routes sound/soc/sunxi/Kconfig | 1 + sound/soc/sunxi/sun8i-codec.c | 834 ++++++++++++++++++++++++++++------ 2 files changed, 699 insertions(+), 136 deletions(-) -- 2.26.2

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

27 Oct '20

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

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

27 Oct '20

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

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