June 2022 - Alsa-devel - mailman.alsa-project.org

[PATCH] ASoC: fsl_asrc_dma: fully initialize structs
by Sascha Hauer 05 Aug '22

05 Aug '22

The driver uses two statically ininitialized struct dma_slave_config, but only one of them is initialized to zero. Initialize config_be to zero as well to make sure that no fields are filled with random values. Let the compiler do this instead of explicitly calling memset() which makes it easier to read. Signed-off-by: Sascha Hauer <s.hauer(a)pengutronix.de> --- sound/soc/fsl/fsl_asrc_dma.c | 3 +-- 1 file changed, 1 insertion(+), 2 deletions(-) diff --git a/sound/soc/fsl/fsl_asrc_dma.c b/sound/soc/fsl/fsl_asrc_dma.c index 5038faf035cba..fec56dbb64e60 100644 --- a/sound/soc/fsl/fsl_asrc_dma.c +++ b/sound/soc/fsl/fsl_asrc_dma.c @@ -138,7 +138,7 @@ static int fsl_asrc_dma_hw_params(struct snd_soc_component *component, struct dma_chan *tmp_chan = NULL, *be_chan = NULL; struct snd_soc_component *component_be = NULL; struct fsl_asrc *asrc = pair->asrc; - struct dma_slave_config config_fe, config_be; + struct dma_slave_config config_fe = {}, config_be = {}; enum asrc_pair_index index = pair->index; struct device *dev = component->dev; struct device_node *of_dma_node; @@ -181,7 +181,6 @@ static int fsl_asrc_dma_hw_params(struct snd_soc_component *component, return -EINVAL; } - memset(&config_fe, 0, sizeof(config_fe)); ret = snd_dmaengine_pcm_prepare_slave_config(substream, params, &config_fe); if (ret) { dev_err(dev, "failed to prepare DMA config for Front-End\n"); -- 2.30.2

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[PATCH][next] ASoC: q6asm: use kcalloc() instead of kzalloc()
by Gustavo A. R. Silva 28 Jul '22

28 Jul '22

Use 2-factor multiplication argument form kcalloc() instead of kzalloc(). Link: https://github.com/KSPP/linux/issues/162 Signed-off-by: Gustavo A. R. Silva <gustavoars(a)kernel.org> --- sound/soc/qcom/qdsp6/q6asm.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/qcom/qdsp6/q6asm.c b/sound/soc/qcom/qdsp6/q6asm.c index 9251d8548965..195780f75d05 100644 --- a/sound/soc/qcom/qdsp6/q6asm.c +++ b/sound/soc/qcom/qdsp6/q6asm.c @@ -513,7 +513,7 @@ int q6asm_map_memory_regions(unsigned int dir, struct audio_client *ac, return 0; } - buf = kzalloc(((sizeof(struct audio_buffer)) * periods), GFP_ATOMIC); + buf = kcalloc(periods, sizeof(*buf), GFP_ATOMIC); if (!buf) { spin_unlock_irqrestore(&ac->lock, flags); return -ENOMEM; -- 2.27.0

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Question about DPCM locking
by Kuninori Morimoto 26 Jul '22

26 Jul '22

Hi Takashi May I ask about this patch ? ASoC: soc-pcm: Fix and cleanup DPCM locking b7898396f4bbe160f546d0c5e9fa17cca9a7d153 Q1. It exchanges many function parameters - func(substream, xxx) + func(rtd, substream, xxx) ^^^^ I guess the purpose is it want to call snd_soc_dpcm_mutex_assert_held() inside the func ? But it looks very verbose. we can do like this, but am I misunderstanding ? func(substream, xxx) { struct snd_soc_pcm_runtime *rtd = asoc_substream_to_rtd(substream); snd_soc_dpcm_mutex_assert_held(rtd); ... } Q2. It added new __soc_pcm_close(). But soc_pcm_close() is using soc_pcm_clean() instead of __soc_pcm_close(). Is this just a mistake ? static int soc_pcm_close(struct snd_pcm_substream *substream) { struct snd_soc_pcm_runtime *rtd = asoc_substream_to_rtd(substream); snd_soc_dpcm_mutex_lock(rtd); - soc_pcm_clean(substream, 0); + __soc_pcm_close(substream); snd_soc_dpcm_mutex_unlock(rtd); return 0; } Thank you for your help !! Best regards --- Kuninori Morimoto

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[PATCH 0/4] ASoC: Intel: Mark BE DAIs as nonatomic for hsw and
by Cezary Rojewski 15 Jul '22

15 Jul '22

Address the warning: "Codec: dpcm_be_connect: FE is nonatomic but BE is not, forcing BE as nonatomic" by marking BE DAI as nonatomic. Aligns with what is already done for FE DAIs. This patchset iterates the change over all HSW and BDW related machine board drivers. Cezary Rojewski (4): ASoC: Intel: hsw_rt5640: Mark BE DAI as nonatomic ASoC: Intel: bdw_rt286: Mark BE DAI as nonatomic ASoC: Intel: bdw_rt5650: Mark BE DAI as nonatomic ASoC: Intel: bdw_rt5677: Mark BE DAI as nonatomic sound/soc/intel/boards/bdw-rt5650.c | 1 + sound/soc/intel/boards/bdw-rt5677.c | 1 + sound/soc/intel/boards/bdw_rt286.c | 1 + sound/soc/intel/boards/hsw_rt5640.c | 1 + 4 files changed, 4 insertions(+) -- 2.25.1

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[PATCH v8 00/14] ALSA: hda: cirrus: Add initial DSP support and firmware loading
by Vitaly Rodionov 15 Jul '22

15 Jul '22

The CS35L41 Amplifier contains a DSP, capable of running firmware. The firmware can run algorithms such as Speaker Protection, to ensure that playback at high gains do not harm the speakers. Adding support for CS35L41 firmware into the CS35L41 HDA driver also allows us to support several extra features, such as hiberation and interrupts. The chain adds support in stages: - General fixes to improve generalization and code re-use inside the CS35L41 HDA driver. - Add support for interrupts into the driver, which is required for complete support of the firmware. - Refactor ASoC CS35L41 code which deals with firmware to allow for code re-use inside the CS35L41 HDA driver. - Add support for loading firmware and tuning files from file system, and creating alsa controls to control it. - Support firmware load paths for different hardware systems. - Support suspend/resume in the driver when using firmware. The firmware supports hibernation, which allows the CS35L41 to drop into a low power mode during suspend. - Support the ability to unload firmware, swap and reload the firmware. This is to allow different firmware to run during calibration. The intended use-case is to load the firmware once on boot, and the driver autmatically tries to load the firmware after it binds to the HDA driver. This behaviour can be switched off using a kconfig, if desired. changes since v7: - Use private_data rather than private_value to save control info - Clean up alsa control memory allocation/deallocation - Remove unnecessary whitespace - Get subsystem id from codec, rather than saving it separately changes since v6: - Fix warning by kernel test robot <lkp(a)intel.com> changes since v5: - Fix warning by kernel test robot <lkp(a)intel.com> changes since v4: - Fully remove tlv remnants from control add apis - Remove unnecessary debug - Rename variable to be more generic - Remove redundent length check from read/write control apis - Use SNDRV_CTL_ELEM_IFACE_CARD for firmware load controls - Make kcontrol add/remove synchronous - Load firmware asynchronous when loading via control - Used cached controls when reloading firmware; only delete controls when removing the driver itself - Improve kcontrol remove - Fix control write + notify - Cleanup of unnecessary code - Fix race condition when loading firmware before playback - Ensure errors are properly propogated - Fix include for Module parameters Stefan Binding (13): ALSA: hda: hda_cs_dsp_ctl: Add Library to support CS_DSP ALSA controls ALSA: hda: hda_cs_dsp_ctl: Add apis to write the controls directly ALSA: hda: cs35l41: Save codec object inside component struct ALSA: hda: cs35l41: Save Subsystem ID inside CS35L41 Driver ALSA: hda: cs35l41: Support reading subsystem id from ACPI ALSA: hda: cs35l41: Support multiple load paths for firmware ALSA: hda: cs35l41: Support Speaker ID for laptops ALSA: hda: cs35l41: Support Hibernation during Suspend ALSA: hda: cs35l41: Read Speaker Calibration data from UEFI variables ALSA: hda: hda_cs_dsp_ctl: Add fw id strings ALSA: hda: cs35l41: Add defaulted values into dsp bypass config sequence ALSA: hda: cs35l41: Support Firmware switching and reloading ALSA: hda: cs35l41: Add module parameter to control firmware load Vitaly Rodionov (1): ALSA: hda: cs35l41: Add initial DSP support and firmware loading MAINTAINERS | 1 + include/sound/cs35l41.h | 4 + sound/pci/hda/Kconfig | 8 + sound/pci/hda/Makefile | 2 + sound/pci/hda/cs35l41_hda.c | 883 +++++++++++++++++++++++++++++++- sound/pci/hda/cs35l41_hda.h | 39 ++ sound/pci/hda/cs35l41_hda_i2c.c | 1 + sound/pci/hda/cs35l41_hda_spi.c | 1 + sound/pci/hda/hda_component.h | 3 + sound/pci/hda/hda_cs_dsp_ctl.c | 240 +++++++++ sound/pci/hda/hda_cs_dsp_ctl.h | 39 ++ sound/pci/hda/patch_realtek.c | 26 +- 12 files changed, 1242 insertions(+), 5 deletions(-) create mode 100644 sound/pci/hda/hda_cs_dsp_ctl.c create mode 100644 sound/pci/hda/hda_cs_dsp_ctl.h -- 2.34.1

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[PATCH] ASoC: nau8821: Add headset button detection
by Seven Lee 11 Jul '22

11 Jul '22

This patch adds the function of headphone button detection, Button detection will be enabled if the device tree has a key_enable property. Signed-off-by: Seven Lee <wtli(a)nuvoton.com> --- .../devicetree/bindings/sound/nau8821.txt | 2 +- sound/soc/codecs/nau8821.c | 35 +++++++++++++++++++ sound/soc/codecs/nau8821.h | 1 + 3 files changed, 37 insertions(+), 1 deletion(-) diff --git a/Documentation/devicetree/bindings/sound/nau8821.txt b/Documentation/devicetree/bindings/sound/nau8821.txt index 6c3baf7a5f21..7c84e7c7327a 100644 --- a/Documentation/devicetree/bindings/sound/nau8821.txt +++ b/Documentation/devicetree/bindings/sound/nau8821.txt @@ -34,7 +34,7 @@ Optional properties: - nuvoton,jack-eject-debounce: number from 0 to 7 that sets debounce time to 2^(n+2) ms - nuvoton,dmic-clk-threshold: the ADC threshold of DMIC clock. - + - nuvoton,key_enable: Headset button detection switch. Example: diff --git a/sound/soc/codecs/nau8821.c b/sound/soc/codecs/nau8821.c index ce4e7f46bb06..7b0a9c8db40a 100644 --- a/sound/soc/codecs/nau8821.c +++ b/sound/soc/codecs/nau8821.c @@ -29,6 +29,8 @@ #define NAU_FVCO_MAX 100000000 #define NAU_FVCO_MIN 90000000 +#define NAU8821_BUTTON SND_JACK_BTN_0 + /* the maximum frequency of CLK_ADC and CLK_DAC */ #define CLK_DA_AD_MAX 6144000 @@ -911,6 +913,20 @@ static void nau8821_eject_jack(struct nau8821 *nau8821) /* Recover to normal channel input */ regmap_update_bits(regmap, NAU8821_R2B_ADC_RATE, NAU8821_ADC_R_SRC_EN, 0); + if (nau8821->key_enable) { + regmap_update_bits(regmap, NAU8821_R0F_INTERRUPT_MASK, + NAU8821_IRQ_KEY_RELEASE_EN | + NAU8821_IRQ_KEY_PRESS_EN, + NAU8821_IRQ_KEY_RELEASE_EN | + NAU8821_IRQ_KEY_PRESS_EN); + regmap_update_bits(regmap, + NAU8821_R12_INTERRUPT_DIS_CTRL, + NAU8821_IRQ_KEY_RELEASE_DIS | + NAU8821_IRQ_KEY_PRESS_DIS, + NAU8821_IRQ_KEY_RELEASE_DIS | + NAU8821_IRQ_KEY_PRESS_DIS); + } + } static void nau8821_jdet_work(struct work_struct *work) @@ -940,6 +956,15 @@ static void nau8821_jdet_work(struct work_struct *work) */ regmap_update_bits(regmap, NAU8821_R2B_ADC_RATE, NAU8821_ADC_R_SRC_EN, NAU8821_ADC_R_SRC_EN); + if (nau8821->key_enable) { + regmap_update_bits(regmap, NAU8821_R0F_INTERRUPT_MASK, + NAU8821_IRQ_KEY_RELEASE_EN | + NAU8821_IRQ_KEY_PRESS_EN, 0); + regmap_update_bits(regmap, + NAU8821_R12_INTERRUPT_DIS_CTRL, + NAU8821_IRQ_KEY_RELEASE_DIS | + NAU8821_IRQ_KEY_PRESS_DIS, 0); + } } else { dev_dbg(nau8821->dev, "Headphone connected\n"); event |= SND_JACK_HEADPHONE; @@ -999,6 +1024,13 @@ static irqreturn_t nau8821_interrupt(int irq, void *data) nau8821_eject_jack(nau8821); event_mask |= SND_JACK_HEADSET; clear_irq = NAU8821_JACK_EJECT_IRQ_MASK; + } else if (active_irq & NAU8821_KEY_SHORT_PRESS_IRQ) { + event |= NAU8821_BUTTON; + event_mask |= NAU8821_BUTTON; + clear_irq = NAU8821_KEY_SHORT_PRESS_IRQ; + } else if (active_irq & NAU8821_KEY_RELEASE_IRQ) { + event_mask = NAU8821_BUTTON; + clear_irq = NAU8821_KEY_RELEASE_IRQ; } else if ((active_irq & NAU8821_JACK_INSERT_IRQ_MASK) == NAU8821_JACK_INSERT_DETECTED) { regmap_update_bits(regmap, NAU8821_R71_ANALOG_ADC_1, @@ -1490,6 +1522,7 @@ static void nau8821_print_device_properties(struct nau8821 *nau8821) nau8821->jack_eject_debounce); dev_dbg(dev, "dmic-clk-threshold: %d\n", nau8821->dmic_clk_threshold); + dev_dbg(dev, "key_enable: %d\n", nau8821->key_enable); } static int nau8821_read_device_properties(struct device *dev, @@ -1503,6 +1536,8 @@ static int nau8821_read_device_properties(struct device *dev, "nuvoton,jkdet-pull-enable"); nau8821->jkdet_pull_up = device_property_read_bool(dev, "nuvoton,jkdet-pull-up"); + nau8821->key_enable = device_property_read_bool(dev, + "nuvoton,key-enable"); ret = device_property_read_u32(dev, "nuvoton,jkdet-polarity", &nau8821->jkdet_polarity); if (ret) diff --git a/sound/soc/codecs/nau8821.h b/sound/soc/codecs/nau8821.h index a92edfeb9d3a..c44251f54d48 100644 --- a/sound/soc/codecs/nau8821.h +++ b/sound/soc/codecs/nau8821.h @@ -525,6 +525,7 @@ struct nau8821 { int jack_eject_debounce; int fs; int dmic_clk_threshold; + int key_enable; }; int nau8821_enable_jack_detect(struct snd_soc_component *component, -- 2.25.1

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[PATCH 00/12] Clang -Wformat warning fixes
by Bill Wendling 11 Jul '22

11 Jul '22

This patch set fixes some clang warnings when -Wformat is enabled. Bill Wendling (12): x86/mce: use correct format characters x86/CPU/AMD: use correct format characters x86/e820: use correct format characters blk-cgroup: use correct format characters fs: quota: use correct format characters PNP: use correct format characters driver/char: use correct format characters cdrom: use correct format characters ALSA: seq: use correct format characters ALSA: seq: use correct format characters ALSA: control: use correct format characters netfilter: conntrack: use correct format characters arch/x86/kernel/cpu/mce/amd.c | 9 +++++---- arch/x86/kernel/cpu/mce/core.c | 2 +- arch/x86/kernel/e820.c | 4 ++-- drivers/cdrom/cdrom.c | 2 +- drivers/char/mem.c | 2 +- drivers/pnp/interface.c | 2 +- fs/quota/dquot.c | 2 +- mm/backing-dev.c | 2 +- net/netfilter/nf_conntrack_helper.c | 2 +- scripts/Makefile.extrawarn | 4 ++-- sound/core/control.c | 2 +- sound/core/seq/seq_clientmgr.c | 2 +- sound/core/sound.c | 2 +- 13 files changed, 19 insertions(+), 18 deletions(-) -- 2.36.1.255.ge46751e96f-goog

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[PATCH v2 0/3] ASoC: sun50i-codec-analog: Internal bias support
by Samuel Holland 08 Jul '22

08 Jul '22

This series adds support for enabling the codec's internal microphone bias, which is needed on at least some versions of the PinePhone. Changes in v2: - Move register update from component probe to device probe Arnaud Ferraris (2): ASoC: dt-bindings: sun50i-codec: Add binding for internal bias ASoC: sun50i-codec-analog: Add support for internal bias Samuel Holland (1): arm64: dts: allwinner: pinephone: Enable internal HMIC bias .../bindings/sound/allwinner,sun50i-a64-codec-analog.yaml | 5 +++++ .../arm64/boot/dts/allwinner/sun50i-a64-pinephone-1.0.dts | 4 ++++ .../arm64/boot/dts/allwinner/sun50i-a64-pinephone-1.1.dts | 4 ++++ sound/soc/sunxi/sun50i-codec-analog.c | 8 ++++++++ 4 files changed, 21 insertions(+) -- 2.35.1

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[PATCH] ASoC: core: Exit all links before removing their components
by Cezary Rojewski 08 Jul '22

08 Jul '22

Flows leading to link->init() and link->exit() are not symmetric. Currently the relevant part of card probe sequence goes as: for_each_card_rtds(card, rtd) for_each_rtd_components(rtd, i, component) component->probe() for_each_card_rtds(card, rtd) for_each_rtd_dais(rtd, i, dai) dai->probe() for_each_card_rtds(card, rtd) rtd->init() On the other side, equivalent remove sequence goes as: for_each_card_rtds(card, rtd) for_each_rtd_dais(rtd, i, dai) dai->remove() for_each_card_rtds(card, rtd) for_each_rtd_components(rtd, i, component) component->remove() for_each_card_rtds(card, rtd) rtd->exit() what can lead to errors as link->exit() may still operate on resources owned by its components despite the probability of them being freed during the component->remove(). This change modifies the remove sequence to: for_each_card_rtds(card, rtd) rtd->exit() for_each_card_rtds(card, rtd) for_each_rtd_dais(rtd, i, dai) dai->remove() for_each_card_rtds(card, rtd) for_each_rtd_components(rtd, i, component) component->remove() so code found in link->exit() is safe to touch any component stuff as component->remove() has not been called yet. Signed-off-by: Cezary Rojewski <cezary.rojewski(a)intel.com> Reviewed-by: Amadeusz Sławiński <amadeuszx.slawinski(a)linux.intel.com> --- sound/soc/soc-core.c | 6 +++--- 1 file changed, 3 insertions(+), 3 deletions(-) diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index 4a3fca50a536..638e781df3b0 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -935,9 +935,6 @@ void snd_soc_remove_pcm_runtime(struct snd_soc_card *card, { lockdep_assert_held(&client_mutex); - /* release machine specific resources */ - snd_soc_link_exit(rtd); - /* * Notify the machine driver for extra destruction */ @@ -1888,6 +1885,9 @@ static void soc_cleanup_card_resources(struct snd_soc_card *card) snd_soc_dapm_shutdown(card); + /* release machine specific resources */ + for_each_card_rtds(card, rtd) + snd_soc_link_exit(rtd); /* remove and free each DAI */ soc_remove_link_dais(card); soc_remove_link_components(card); -- 2.25.1

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[PATCH 0/3] ADD legacy audio driver support for rembrandt
by V sujith kumar Reddy 07 Jul '22

07 Jul '22

Add Generic driver to support multiple platform, ADD HS control instance for Rembrandt platform. Add nau8825,max98560 and rt5682s,rt1019 combination support for legacy platform. V sujith kumar Reddy (3): ASoC: amd: acp: Remove rt1019_1 codec conf from machine driver ASoC: amd: acp: ACP code generic to support newer platforms ASoC: amd: acp: Add legacy audio driver support for Rembrandt platform sound/soc/amd/acp/Kconfig | 11 + sound/soc/amd/acp/Makefile | 2 + sound/soc/amd/acp/acp-i2s.c | 169 +++++++++++- sound/soc/amd/acp/acp-legacy-mach.c | 32 +++ sound/soc/amd/acp/acp-mach-common.c | 104 +++++-- sound/soc/amd/acp/acp-mach.h | 6 + sound/soc/amd/acp/acp-pci.c | 6 + sound/soc/amd/acp/acp-pdm.c | 8 +- sound/soc/amd/acp/acp-platform.c | 32 ++- sound/soc/amd/acp/acp-rembrandt.c | 397 +++++++++++++++++++++++++++ sound/soc/amd/acp/acp-renoir.c | 38 ++- sound/soc/amd/acp/amd.h | 86 +++++- sound/soc/amd/acp/chip_offset_byte.h | 40 ++- 13 files changed, 862 insertions(+), 69 deletions(-) create mode 100644 sound/soc/amd/acp/acp-rembrandt.c -- 2.25.1

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