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

[PATCH alsa-ucm-conf v3 1/2] sof-soundwire: Add basic support for cs42l43
by Charles Keepax 20 Dec '23

20 Dec '23

cs42l43 is a codec device, add basic support for it. Including a dual channel DMIC input, stereo headphones, and a mono headset microphone. Signed-off-by: Charles Keepax <ckeepax(a)opensource.cirrus.com> --- No changes since v2. Thanks, Charles ucm2/sof-soundwire/cs42l43-dmic.conf | 28 +++++++++++++++++ ucm2/sof-soundwire/cs42l43.conf | 47 ++++++++++++++++++++++++++++ 2 files changed, 75 insertions(+) create mode 100644 ucm2/sof-soundwire/cs42l43-dmic.conf create mode 100644 ucm2/sof-soundwire/cs42l43.conf diff --git a/ucm2/sof-soundwire/cs42l43-dmic.conf b/ucm2/sof-soundwire/cs42l43-dmic.conf new file mode 100644 index 0000000..bf59eca --- /dev/null +++ b/ucm2/sof-soundwire/cs42l43-dmic.conf @@ -0,0 +1,28 @@ +# Use case Configuration for sof-soundwire card + +SectionDevice."Mic" { + Comment "Microphones" + + ConflictingDevice [ + "Headset" + ] + + EnableSequence [ + cset "name='cs42l43 DP1TX1 Input' 'Decimator 3'" + cset "name='cs42l43 DP1TX2 Input' 'Decimator 4'" + cset "name='cs42l43 Decimator 3 Switch' on" + cset "name='cs42l43 Decimator 4 Switch' on" + ] + + DisableSequence [ + cset "name='cs42l43 Decimator 3 Switch' off" + cset "name='cs42l43 Decimator 4 Switch' off" + cset "name='cs42l43 DP1TX1 Input' 'None'" + cset "name='cs42l43 DP1TX2 Input' 'None'" + ] + + Value { + CapturePriority 100 + CapturePCM "hw:${CardId},4" + } +} diff --git a/ucm2/sof-soundwire/cs42l43.conf b/ucm2/sof-soundwire/cs42l43.conf new file mode 100644 index 0000000..378dbb2 --- /dev/null +++ b/ucm2/sof-soundwire/cs42l43.conf @@ -0,0 +1,47 @@ +# Use case Configuration for sof-soundwire card + +SectionDevice."Headphones" { + Comment "Headphones" + + EnableSequence [ + cset "name='cs42l43 Headphone L Input 1' 'DP5RX1'" + cset "name='cs42l43 Headphone R Input 1' 'DP5RX2'" + ] + + DisableSequence [ + cset "name='cs42l43 Headphone L Input 1' 'None'" + cset "name='cs42l43 Headphone R Input 1' 'None'" + ] + + Value { + PlaybackPriority 200 + PlaybackPCM "hw:${CardId},0" + PlaybackVolume "cs42l43 Headphone Digital Volume" + JackControl "Headphone Jack" + } +} + +SectionDevice."Headset" { + Comment "Headset Microphone" + + EnableSequence [ + cset "name='cs42l43 ADC1 Input' 'IN1'" + cset "name='cs42l43 Decimator 1 Mode' 'ADC'" + + cset "name='cs42l43 DP1TX1 Input' 'Decimator 1'" + cset "name='cs42l43 DP1TX2 Input' 'Decimator 1'" + ] + + DisableSequence [ + cset "name='cs42l43 DP1TX1 Input' 'None'" + cset "name='cs42l43 DP1TX2 Input' 'None'" + ] + + Value { + CapturePriority 200 + CapturePCM "hw:${CardId},4" + CaptureVolume "cs42l43 Decimator 1 Volume" + CaptureSwitch "cs42l43 Decimator 1 Switch" + JackControl "Headset Mic Jack" + } +} -- 2.39.2

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[PATCH] ALSA: hda: cs35l41: Add HP override
by Lorenz Brun 20 Dec '23

20 Dec '23

This adds an override for a series of notebooks using a common config taken from HP's proprietary Windows driver (csaudioext). This has been tested on a HP 15-ey0xxxx device (subsystem 103C8A31) together with another Realtek quirk and the calibration files from the proprietary driver. Signed-off-by: Lorenz Brun <lorenz(a)brun.one> --- sound/pci/hda/cs35l41_hda_property.c | 59 ++++++++++++++++++++++++++++ 1 file changed, 59 insertions(+) diff --git a/sound/pci/hda/cs35l41_hda_property.c b/sound/pci/hda/cs35l41_hda_property.c index c83328971728..8135ea532a94 100644 --- a/sound/pci/hda/cs35l41_hda_property.c +++ b/sound/pci/hda/cs35l41_hda_property.c @@ -6,6 +6,7 @@ // // Author: Stefan Binding <sbinding(a)opensource.cirrus.com> +#include <linux/acpi.h> #include <linux/gpio/consumer.h> #include <linux/string.h> #include "cs35l41_hda_property.h" @@ -81,6 +82,42 @@ static int hp_vision_acpi_fix(struct cs35l41_hda *cs35l41, struct device *physde return 0; } +/* + * HP 2-channel I2C configuration with internal boost (4.1A inductor current) with no _DSD, + * reset GPIO can still be extracted from ACPI by index. Covers HP configurations 251, 252, + * 253, 254, 351, 352 and 353 in the proprietary driver (csaudioext). + */ +static int hp_i2c_2ch_vbst_ipk41(struct cs35l41_hda *cs35l41, struct device *physdev, int id, + const char *hid) +{ + // In case a valid _DSD exists, use that instead of the override. This stops applying + // the override in case HP ever fixes their firmware. + if (device_property_count_u32(physdev, "cirrus,dev-index") > 0) + return -ENOENT; + + struct cs35l41_hw_cfg *hw_cfg = &cs35l41->hw_cfg; + + cs35l41->index = id == 0x40 ? 0 : 1; + cs35l41->channel_index = 0; + // Get reset GPIO (shared for both instances) from ACPI GpioIo at index 0. + cs35l41->reset_gpio = gpiod_get_index(physdev, NULL, 0, GPIOD_OUT_HIGH); + // Speaker ID GPIO is ACPI GpioIo index 1. + cs35l41->speaker_id = cs35l41_get_speaker_id(physdev, 0, 0, 1); + + hw_cfg->spk_pos = cs35l41->index ? 1 : 0; // left:right + hw_cfg->gpio1.func = CS35L41_NOT_USED; + hw_cfg->gpio1.valid = true; + hw_cfg->gpio2.func = CS35L41_INTERRUPT; + hw_cfg->gpio2.valid = true; + hw_cfg->bst_type = CS35L41_INT_BOOST; + hw_cfg->bst_ind = 1000; + hw_cfg->bst_ipk = 4100; + hw_cfg->bst_cap = 10; // Exact value unknown, maps into correct range + hw_cfg->valid = true; + + return 0; +} + struct cs35l41_prop_model { const char *hid; const char *ssid; @@ -92,6 +129,28 @@ static const struct cs35l41_prop_model cs35l41_prop_model_table[] = { { "CLSA0100", NULL, lenovo_legion_no_acpi }, { "CLSA0101", NULL, lenovo_legion_no_acpi }, { "CSC3551", "103C89C6", hp_vision_acpi_fix }, + { "CSC3551", "103C8A28", hp_i2c_2ch_vbst_ipk41 }, + { "CSC3551", "103C8A29", hp_i2c_2ch_vbst_ipk41 }, + { "CSC3551", "103C8A2A", hp_i2c_2ch_vbst_ipk41 }, + { "CSC3551", "103C8A2B", hp_i2c_2ch_vbst_ipk41 }, + { "CSC3551", "103C8A2C", hp_i2c_2ch_vbst_ipk41 }, + { "CSC3551", "103C8A2D", hp_i2c_2ch_vbst_ipk41 }, + { "CSC3551", "103C8A2E", hp_i2c_2ch_vbst_ipk41 }, + { "CSC3551", "103C8A30", hp_i2c_2ch_vbst_ipk41 }, + { "CSC3551", "103C8A31", hp_i2c_2ch_vbst_ipk41 }, + { "CSC3551", "103C8BB3", hp_i2c_2ch_vbst_ipk41 }, + { "CSC3551", "103C8BB4", hp_i2c_2ch_vbst_ipk41 }, + { "CSC3551", "103C8BDF", hp_i2c_2ch_vbst_ipk41 }, + { "CSC3551", "103C8BE0", hp_i2c_2ch_vbst_ipk41 }, + { "CSC3551", "103C8BE1", hp_i2c_2ch_vbst_ipk41 }, + { "CSC3551", "103C8BE2", hp_i2c_2ch_vbst_ipk41 }, + { "CSC3551", "103C8BE9", hp_i2c_2ch_vbst_ipk41 }, + { "CSC3551", "103C8BDD", hp_i2c_2ch_vbst_ipk41 }, + { "CSC3551", "103C8BDE", hp_i2c_2ch_vbst_ipk41 }, + { "CSC3551", "103C8BE3", hp_i2c_2ch_vbst_ipk41 }, + { "CSC3551", "103C8BE5", hp_i2c_2ch_vbst_ipk41 }, + { "CSC3551", "103C8BE6", hp_i2c_2ch_vbst_ipk41 }, + { "CSC3551", "103C8B3A", hp_i2c_2ch_vbst_ipk41 }, {} }; -- 2.42.0

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Re: linux-next: Tree for Dec 19 [sound/pci/hda/snd-hda-scodec-cs35l41.ko]
by Randy Dunlap 20 Dec '23

20 Dec '23

On 12/18/23 20:41, Stephen Rothwell wrote: > Hi all, > > Changes since 20231218: > on riscv64: # CONFIG_SPI is not set ERROR: modpost: "spi_setup" [sound/pci/hda/snd-hda-scodec-cs35l41.ko] undefined! from sound/pci/hda/cs35l41_hda_property.c Full randconfig file is attached. -- #Randy https://people.kernel.org/tglx/notes-about-netiquette https://subspace.kernel.org/etiquette.html

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[PATCH v2 0/5] ASoC: don't use original dummy dlc
by Kuninori Morimoto 19 Dec '23

19 Dec '23

Hi Mark, and each driver owner This v2 patch-set try to not use original dummy dlc. "Empty" dlc might be used on Platform, but "dummy" dlc is not needed for it. [PATCH 1/5][PATCH 2/5] removes "dummy" dlc from Platform. Now ASoC have common dummy dlc (= snd_soc_dummy_dlc). [PATCH 3/5][PATCH 4/5] will use it instead of original dummy dlc. Many drivers are using below macro SND_SOC_DAILINK_DEFS(link, DAILINK_COMP_ARRAY(COMP_CPU(...)), (X) DAILINK_COMP_ARRAY(COMP_DUMMY()), DAILINK_COMP_ARRAY(COMP_EMPTY())); But (X) part will create original dummy dlc. [PATCH 5/5] will try not to create original dummy dlc, and replace it to common dummy dlc. v1 -> v2 - Do nothing for Platform on snd_soc_fill_dummy_dai() Kuninori Morimoto (5): ASoC: fsl: fsl-asoc-card: don't need DUMMY Platform ASoC: samsung: odroid: don't need DUMMY Platform ASoC: intel: hdaudio.c: use snd_soc_dummy_dlc ASoC: sof: use snd_soc_dummy_dlc ASoC: soc.h: don't create dummy Component via COMP_DUMMY() include/sound/soc.h | 2 +- sound/soc/fsl/fsl-asoc-card.c | 3 +-- sound/soc/intel/avs/boards/hdaudio.c | 6 ++---- sound/soc/samsung/odroid.c | 3 +-- sound/soc/soc-core.c | 24 ++++++++++++++++++++++++ sound/soc/sof/sof-client-probes.c | 4 +--- 6 files changed, 30 insertions(+), 12 deletions(-) -- 2.25.1

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[PATCH 0/4] GPIO inclusion fixes to misc sound drivers
by Linus Walleij 19 Dec '23

19 Dec '23

Mostly dropping unused headers, and a single driver rewrite. Signed-off-by: Linus Walleij <linus.walleij(a)linaro.org> --- Linus Walleij (4): ASoC: hisilicon: Drop GPIO include ASoC: qcom: sc7180: Drop GPIO include ASoC: simple-card-utils: Drop GPIO include ASoC: tegra: tegra20_ac97: Convert to use GPIO descriptors arch/arm/boot/dts/nvidia/tegra20-colibri.dtsi | 2 +- sound/soc/generic/audio-graph-card.c | 1 - sound/soc/generic/audio-graph-card2.c | 1 - sound/soc/generic/simple-card-utils.c | 1 - sound/soc/hisilicon/hi6210-i2s.c | 1 - sound/soc/qcom/sc7180.c | 1 - sound/soc/tegra/tegra20_ac97.c | 55 +++++++++++++-------------- sound/soc/tegra/tegra20_ac97.h | 4 +- 8 files changed, 29 insertions(+), 37 deletions(-) --- base-commit: b85ea95d086471afb4ad062012a4d73cd328fa86 change-id: 20231212-gpio-descriptors-sound-misc-8bc161358ef1 Best regards, -- Linus Walleij <linus.walleij(a)linaro.org>

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[PATCH 0/5] ASoC: don't use original dummy dlc
by Kuninori Morimoto 19 Dec '23

19 Dec '23

Hi Mark, and each driver owner This patch-set try to not use original dummy dlc. "Empty" dlc might be used on Platform, but "dummy" dlc is not needed for it. [PATCH 1/5][PATCH 2/5] removes "dummy" dlc from Platform. Now ASoC have common dummy dlc (= snd_soc_dummy_dlc). [PATCH 3/5][PATCH 4/5] will use it instead of original dummy dlc. Many drivers are using below macro SND_SOC_DAILINK_DEFS(link, DAILINK_COMP_ARRAY(COMP_CPU(...)), (X) DAILINK_COMP_ARRAY(COMP_DUMMY()), DAILINK_COMP_ARRAY(COMP_EMPTY())); But (X) part will create original dummy dlc. [PATCH 5/5] will try not to create original dummy dlc, and replace it to common dummy dlc. Kuninori Morimoto (5): ASoC: fsl: fsl-asoc-card: don't need DUMMY Platform ASoC: samsung: odroid: don't need DUMMY Platform ASoC: intel: hdaudio.c: use snd_soc_dummy_dlc ASoC: sof: use snd_soc_dummy_dlc ASoC: soc.h: don't create dummy Component via COMP_DUMMY() include/sound/soc.h | 2 +- sound/soc/fsl/fsl-asoc-card.c | 3 +-- sound/soc/intel/avs/boards/hdaudio.c | 6 ++---- sound/soc/samsung/odroid.c | 3 +-- sound/soc/soc-core.c | 30 ++++++++++++++++++++++++++++ sound/soc/sof/sof-client-probes.c | 4 +--- 6 files changed, 36 insertions(+), 12 deletions(-) -- 2.25.1

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[PATCH 0/2] ASoC: qcom: add sound card support for SM8650
by Neil Armstrong 19 Dec '23

19 Dec '23

Document the SM8650 sound card using the SM8450 fallback and add the SM8650 compatible to the sc8280xp sound card driver to use the sm8650 card driver_name like SM8450 & SM8550. Signed-off-by: Neil Armstrong <neil.armstrong(a)linaro.org> --- Neil Armstrong (2): ASoC: dt-bindings: qcom,sm8250: document SM8650 sound card ASoC: qcom: sc8280xp: Add support for SM8650 Documentation/devicetree/bindings/sound/qcom,sm8250.yaml | 1 + sound/soc/qcom/sc8280xp.c | 1 + 2 files changed, 2 insertions(+) --- base-commit: bbd220ce4e29ed55ab079007cff0b550895258eb change-id: 20231212-topic-sm8650-upstream-snd-card-15eb7afa27f9 Best regards, -- Neil Armstrong <neil.armstrong(a)linaro.org>

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[PATCH] ASoC: dt-bindings: qcom,lpass-va-macro: remove spurious contains in if statement
by Neil Armstrong 19 Dec '23

19 Dec '23

Remove this spurious "contains" which causes the bindings check of qcom,sm8450-lpass-va-macro compatible to fail with: codec@33f0000: clocks: [[156, 57, 1], [156, 102, 1], [156, 103, 1], [156, 70, 1]] is too long from schema $id: http://devicetree.org/schemas/sound/qcom,lpass-va-macro.yaml# codec@33f0000: clock-names: ['mclk', 'macro', 'dcodec', 'npl'] is too long from schema $id: http://devicetree.org/schemas/sound/qcom,lpass-va-macro.yaml# Seems the double "contains" was considered as valid by the tool but broke the entire if statements. Fixes: f243ef746d0a ("ASoC: dt-bindings: qcom,lpass-va-macro: Add SM8650 LPASS VA") Signed-off-by: Neil Armstrong <neil.armstrong(a)linaro.org> --- Documentation/devicetree/bindings/sound/qcom,lpass-va-macro.yaml | 5 ++--- 1 file changed, 2 insertions(+), 3 deletions(-) diff --git a/Documentation/devicetree/bindings/sound/qcom,lpass-va-macro.yaml b/Documentation/devicetree/bindings/sound/qcom,lpass-va-macro.yaml index c03ff9472a85..6b483fa3c428 100644 --- a/Documentation/devicetree/bindings/sound/qcom,lpass-va-macro.yaml +++ b/Documentation/devicetree/bindings/sound/qcom,lpass-va-macro.yaml @@ -121,9 +121,8 @@ allOf: properties: compatible: contains: - contains: - enum: - - qcom,sm8550-lpass-va-macro + enum: + - qcom,sm8550-lpass-va-macro then: properties: clocks: --- base-commit: aa4db8324c4d0e67aa4670356df4e9fae14b4d37 change-id: 20231219-topic-sm8x50-upstream-va-macro-bindings-fix-924ffff6ae31 Best regards, -- Neil Armstrong <neil.armstrong(a)linaro.org>

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[PATCH 1/5] ASoC: SOF: amd: Refactor spinlock_irq(&sdev->ipc_lock) sequence in irq_handler
by Venkata Prasad Potturu 19 Dec '23

19 Dec '23

Refactor spinlock_irq(&sdev->ipc_lock) sequence in irq_handler to avoid race conditions for acquiring hw_semaphore. Signed-off-by: Venkata Prasad Potturu <venkataprasad.potturu(a)amd.com> --- sound/soc/sof/amd/acp-ipc.c | 4 +--- sound/soc/sof/amd/acp.c | 3 +++ 2 files changed, 4 insertions(+), 3 deletions(-) diff --git a/sound/soc/sof/amd/acp-ipc.c b/sound/soc/sof/amd/acp-ipc.c index fcb54f545fea..2743f07a5e08 100644 --- a/sound/soc/sof/amd/acp-ipc.c +++ b/sound/soc/sof/amd/acp-ipc.c @@ -3,7 +3,7 @@ // This file is provided under a dual BSD/GPLv2 license. When using or // redistributing this file, you may do so under either license. // -// Copyright(c) 2021 Advanced Micro Devices, Inc. +// Copyright(c) 2021, 2023 Advanced Micro Devices, Inc. // // Authors: Balakishore Pati <Balakishore.pati(a)amd.com> // Ajit Kumar Pandey <AjitKumar.Pandey(a)amd.com> @@ -188,13 +188,11 @@ irqreturn_t acp_sof_ipc_irq_thread(int irq, void *context) dsp_ack = snd_sof_dsp_read(sdev, ACP_DSP_BAR, ACP_SCRATCH_REG_0 + dsp_ack_write); if (dsp_ack) { - spin_lock_irq(&sdev->ipc_lock); /* handle immediate reply from DSP core */ acp_dsp_ipc_get_reply(sdev); snd_sof_ipc_reply(sdev, 0); /* set the done bit */ acp_dsp_ipc_dsp_done(sdev); - spin_unlock_irq(&sdev->ipc_lock); ipc_irq = true; } diff --git a/sound/soc/sof/amd/acp.c b/sound/soc/sof/amd/acp.c index 603ea5fc0d0d..7860724c4d2d 100644 --- a/sound/soc/sof/amd/acp.c +++ b/sound/soc/sof/amd/acp.c @@ -343,11 +343,13 @@ static irqreturn_t acp_irq_thread(int irq, void *context) const struct sof_amd_acp_desc *desc = get_chip_info(sdev->pdata); unsigned int count = ACP_HW_SEM_RETRY_COUNT; + spin_lock_irq(&sdev->ipc_lock); while (snd_sof_dsp_read(sdev, ACP_DSP_BAR, desc->hw_semaphore_offset)) { /* Wait until acquired HW Semaphore lock or timeout */ count--; if (!count) { dev_err(sdev->dev, "%s: Failed to acquire HW lock\n", __func__); + spin_unlock_irq(&sdev->ipc_lock); return IRQ_NONE; } } @@ -356,6 +358,7 @@ static irqreturn_t acp_irq_thread(int irq, void *context) /* Unlock or Release HW Semaphore */ snd_sof_dsp_write(sdev, ACP_DSP_BAR, desc->hw_semaphore_offset, 0x0); + spin_unlock_irq(&sdev->ipc_lock); return IRQ_HANDLED; }; -- 2.25.1

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[PATCH v2] ASoC: fsl_sai: Fix channel swap issue on i.MX8MP
by Shengjiu Wang 19 Dec '23

19 Dec '23

When flag mclk_with_tere and mclk_direction_output enabled, The SAI transmitter or receiver will be enabled in very early stage, that if FSL_SAI_xMR is set by previous case, for example previous case is one channel, current case is two channels, then current case started with wrong xMR in the beginning, then channel swap happen. The patch is to clear xMR in hw_free() to avoid such channel swap issue. Fixes: 3e4a82612998 ("ASoC: fsl_sai: MCLK bind with TX/RX enable bit") Signed-off-by: Shengjiu Wang <shengjiu.wang(a)nxp.com> --- changes in v2: - add fixes tag sound/soc/fsl/fsl_sai.c | 3 +++ 1 file changed, 3 insertions(+) diff --git a/sound/soc/fsl/fsl_sai.c b/sound/soc/fsl/fsl_sai.c index 32bbe5056a63..546bd4e333b5 100644 --- a/sound/soc/fsl/fsl_sai.c +++ b/sound/soc/fsl/fsl_sai.c @@ -714,6 +714,9 @@ static int fsl_sai_hw_free(struct snd_pcm_substream *substream, bool tx = substream->stream == SNDRV_PCM_STREAM_PLAYBACK; unsigned int ofs = sai->soc_data->reg_offset; + /* Clear xMR to avoid channel swap with mclk_with_tere enabled case */ + regmap_write(sai->regmap, FSL_SAI_xMR(tx), 0); + regmap_update_bits(sai->regmap, FSL_SAI_xCR3(tx, ofs), FSL_SAI_CR3_TRCE_MASK, 0); -- 2.34.1

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