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

[PATCH v1] ALSA: hda: cs35l41: Override the _DSD for HP Zbook Fury 17 G9 to correct boost type
by Stefan Binding 24 Aug '23

24 Aug '23

CS35L41 HDA driver requires ACPI to contain correct _DSD properties to correctly configure the device. Whilst the HP Zbook Fury 17 G9 contains valid _DSD properties, the boost type has been configured incorrectly in the _DSD for this laptop. We can override these properties to fix the boost type. Signed-off-by: Stefan Binding <sbinding(a)opensource.cirrus.com> --- sound/pci/hda/cs35l41_hda_property.c | 32 ++++++++++++++++++++++++++++ 1 file changed, 32 insertions(+) diff --git a/sound/pci/hda/cs35l41_hda_property.c b/sound/pci/hda/cs35l41_hda_property.c index 9b5a1d61575e..b62a4e6968e2 100644 --- a/sound/pci/hda/cs35l41_hda_property.c +++ b/sound/pci/hda/cs35l41_hda_property.c @@ -43,6 +43,37 @@ static int lenovo_legion_no_acpi(struct cs35l41_hda *cs35l41, struct device *phy return 0; } +/* + * Device 103C89C6 does have _DSD, however it is setup to use the wrong boost type. + * We can override the _DSD to correct the boost type here. + * Since this laptop has valid ACPI, we do not need to handle cs-gpios, since that already exists + * in the ACPI. + */ +static int hp_vision_acpi_fix(struct cs35l41_hda *cs35l41, struct device *physdev, int id, + const char *hid) +{ + struct cs35l41_hw_cfg *hw_cfg = &cs35l41->hw_cfg; + + dev_info(cs35l41->dev, "Adding DSD properties for %s\n", cs35l41->acpi_subsystem_id); + + cs35l41->index = id; + cs35l41->channel_index = 0; + cs35l41->reset_gpio = gpiod_get_index(physdev, NULL, 1, GPIOD_OUT_HIGH); + cs35l41->speaker_id = -ENOENT; + hw_cfg->spk_pos = cs35l41->index ? 1 : 0; // right:left + 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 = 4500; + hw_cfg->bst_cap = 24; + hw_cfg->valid = true; + + return 0; +} + struct cs35l41_prop_model { const char *hid; const char *ssid; @@ -53,6 +84,7 @@ struct cs35l41_prop_model { 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 }, {} }; -- 2.34.1

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[PATCH 0/4] ALSA: ump: Improve legacy rawmidi & co
by Takashi Iwai 24 Aug '23

24 Aug '23

Hi, this is a patch set to fix / enhance the legacy rawmidi handling in UMP core and USB-audio driver, as well as the update of MIDI 2.0 documentation. Takashi === Takashi Iwai (4): ALSA: usb-audio: Attach legacy rawmidi after probing all UMP EPs ALSA: ump: Fill group names for legacy rawmidi substreams ALSA: ump: Don't create unused substreams for static blocks ALSA: documentation: Add description for USB MIDI 2.0 gadget driver Documentation/sound/designs/midi-2.0.rst | 188 +++++++++++++++++++++++ include/sound/ump.h | 1 + sound/core/ump.c | 58 ++++++- sound/usb/midi2.c | 15 +- 4 files changed, 250 insertions(+), 12 deletions(-) -- 2.35.3

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[GIT PULL] ASoC fixes for v6.5-rc7
by Mark Brown 24 Aug '23

24 Aug '23

The following changes since commit 37aba3190891d4de189bd5192ee95220e295f34d: ASoC: rt1308-sdw: fix random louder sound (2023-08-13 18:16:32 +0100) are available in the Git repository at: https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git tags/asoc-fix-v6.5-rc7 for you to fetch changes up to 1613781d7e8a93618ff3a6b37f81f06769b53717: ASoC: cs35l41: Correct amp_gain_tlv values (2023-08-23 13:27:06 +0100) ---------------------------------------------------------------- ASoC: Fixes for v6.5 A relatively large but generally not super urgent set of fixes for ASoC, including some quirks and a MAINTAINERS update. There's also an update to cs35l56 to change the firmware ABI, there are no current shipping systems which use the current interface and the sooner we get the new interface in the less likely it is that something will start. It'd be nice if these landed for v6.5 but not the end of the world if they wait till v6.6. ---------------------------------------------------------------- BrenoRCBrito (1): ASoC: amd: yc: Add VivoBook Pro 15 to quirks list for acp6x Chao Song (1): ASoC: SOF: ipc4-pcm: fix possible null pointer deference Charles Keepax (1): ASoC: cs35l41: Correct amp_gain_tlv values Kevin-Lu (1): MAINTAINERS: Add entries for TEXAS INSTRUMENTS ASoC DRIVERS Maciej Strozek (1): ASoC: cs35l56: Read firmware uuid from a device property instead of _SUB Mark Brown (1): ASoC: cs35l56: Update ACPI HID and property Shenghao Ding (1): ASoC: tas2781: fixed register access error when switching to other chips Simon Trimmer (1): ASoC: cs35l56: Add an ACPI match table MAINTAINERS | 33 +++++++++++++++++++++++++++++++++ sound/soc/amd/yc/acp6x-mach.c | 7 +++++++ sound/soc/codecs/cs35l41.c | 2 +- sound/soc/codecs/cs35l56-i2c.c | 9 +++++++++ sound/soc/codecs/cs35l56-spi.c | 9 +++++++++ sound/soc/codecs/cs35l56.c | 31 ++++++++++++------------------- sound/soc/codecs/tas2781-comlib.c | 19 ++++++++++--------- sound/soc/sof/ipc4-pcm.c | 3 +++ 8 files changed, 84 insertions(+), 29 deletions(-)

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[PATCH v1 0/5] ASoC: sunxi: Add support for D1/T113s internal audio codec
by Maksim Kiselev 24 Aug '23

24 Aug '23

Hi, This is the series adding support for internal audio codec on Allwinner D1/T113s SoCs family. My work based on Icenowy Zheng's patches from sipeed repo that adds support for the R329 audio codec. But, unfortunately, these patches were not sent to upstream. So, I adapted them to fit the T113s. In the future, support for R329 could easily be added on top of this series. Because D1/T113s and R329 SoCs have almost the same audio codec IP, with slight differences in the analog part, plus the R329 has two additional ADCs. The series includes modifications for the existing sun4i-codec driver that brings support for the digital part of the D1 codec. And adds the new sun20i-d1-codec-analog driver for the analog part. I would be glad if someone have a look :) Cheers, Maksim Maksim Kiselev (5): ASoC: dt-bindings: sun4i-a10-codec: Add binding for Allwinner D1 SoC ASoC: dt-bindings: Add schema for "allwinner,sun20i-d1-codec-analog" ASoC: sunxi: sun4i-codec: add basic support for D1 audio codec ASoC: sunxi: Add new driver for Allwinner D1/T113s codec's analog path controls riscv: dts: allwinner: d1: Add device nodes for internal audio codec .../allwinner,sun20i-d1-codec-analog.yaml | 33 ++ .../sound/allwinner,sun4i-a10-codec.yaml | 64 ++- .../boot/dts/allwinner/sunxi-d1s-t113.dtsi | 22 ++ sound/soc/sunxi/Kconfig | 11 + sound/soc/sunxi/Makefile | 1 + sound/soc/sunxi/sun20i-d1-codec-analog.c | 220 +++++++++++ sound/soc/sunxi/sun4i-codec.c | 364 +++++++++++++++--- 7 files changed, 643 insertions(+), 72 deletions(-) create mode 100644 Documentation/devicetree/bindings/sound/allwinner,sun20i-d1-codec-analog.yaml create mode 100644 sound/soc/sunxi/sun20i-d1-codec-analog.c -- 2.39.2

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[PATCH] ASoC: cs42l43: Initialize ret in default case in cs42l43_pll_ev()
by Nathan Chancellor 24 Aug '23

24 Aug '23

clang warns (or errors with CONFIG_WERROR=y): sound/soc/codecs/cs42l43.c:1371:2: error: variable 'ret' is used uninitialized whenever switch default is taken [-Werror,-Wsometimes-uninitialized] 1371 | default: | ^~~~~~~ sound/soc/codecs/cs42l43.c:1377:9: note: uninitialized use occurs here 1377 | return ret; | ^~~ sound/soc/codecs/cs42l43.c:1349:9: note: initialize the variable 'ret' to silence this warning 1349 | int ret; | ^ | = 0 1 error generated. Initialize ret to 0 in the default case, as there was nothing to do for other event types. Closes: https://github.com/ClangBuiltLinux/linux/issues/1922 Fixes: fc918cbe874e ("ASoC: cs42l43: Add support for the cs42l43") Signed-off-by: Nathan Chancellor <nathan(a)kernel.org> --- sound/soc/codecs/cs42l43.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/soc/codecs/cs42l43.c b/sound/soc/codecs/cs42l43.c index 55a79219af35..18178ffa82f2 100644 --- a/sound/soc/codecs/cs42l43.c +++ b/sound/soc/codecs/cs42l43.c @@ -1369,6 +1369,7 @@ static int cs42l43_pll_ev(struct snd_soc_dapm_widget *w, clk_disable_unprepare(priv->mclk); break; default: + ret = 0; break; } --- base-commit: 014ee0692f29da8b08fed5da0fa14e04698a50f7 change-id: 20230823-cs42l43_pll_ev-init-ret-fb1e47ea0605 Best regards, -- Nathan Chancellor <nathan(a)kernel.org>

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[PATCH] ASoC: audio-graph-card2: add comment for format property
by Kuninori Morimoto 23 Aug '23

23 Aug '23

We don't need to have "format" property on DT any more if CPU/Codec driver has .auto_selectable_formats settings on snd_soc_dai_ops. The sample dtsi doesn't have it. To avoid user confusion, this patch indicates it on comment. Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> --- .../soc/generic/audio-graph-card2-custom-sample.dtsi | 7 +++++++ sound/soc/generic/audio-graph-card2.c | 12 ++++++++++++ 2 files changed, 19 insertions(+) diff --git a/sound/soc/generic/audio-graph-card2-custom-sample.dtsi b/sound/soc/generic/audio-graph-card2-custom-sample.dtsi index 2ac0de3c21da..8acaa2ddb335 100644 --- a/sound/soc/generic/audio-graph-card2-custom-sample.dtsi +++ b/sound/soc/generic/audio-graph-card2-custom-sample.dtsi @@ -34,6 +34,13 @@ * ... * }; * + * + * Below sample doesn't use "format" property, + * because test-component driver (test-cpu/test-codec) is supporting + * snd_soc_dai_ops :: .auto_selectable_formats. + * see + * snd_soc_runtime_get_dai_fmt() + * linux/sound/soc/generic/test-component.c :: test_dai_formats */ / { /* diff --git a/sound/soc/generic/audio-graph-card2.c b/sound/soc/generic/audio-graph-card2.c index 542c4a114940..1fae810a2833 100644 --- a/sound/soc/generic/audio-graph-card2.c +++ b/sound/soc/generic/audio-graph-card2.c @@ -47,6 +47,18 @@ see graph_parse_daifmt(). + "format" property is no longer needed on DT if both CPU/Codec drivers are + supporting snd_soc_dai_ops :: .auto_selectable_formats. + see + snd_soc_runtime_get_dai_fmt() + + sample driver + linux/sound/soc/sh/rcar/core.c + linux/sound/soc/codecs/ak4613.c + linux/sound/soc/codecs/pcm3168a.c + linux/sound/soc/soc-utils.c + linux/sound/soc/generic/test-component.c + ************************************ Normal Audio-Graph ************************************ -- 2.25.1

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[PATCH 0/5] ASoC: rsnd: tidyup ADG
by Kuninori Morimoto 23 Aug '23

23 Aug '23

Hi Mark ------ for git-log ------ 8< ------ 8< ------ 8< ------ Renesas Sound has ADG for clock control. Basically it needs accurately divisible external input clock. But sometimes sometimes it doesn't have to be accurate for some reason. We can use ADG clk_i for such case. It came from CPG as very high rate clock, but is not accurately divisible for 48kHz/44.1kHz rate, but enough for approximate rate. This patch set support such use case. ------ extra info ------ 8< ------ 8< ------ 8< ------ 8< ------ Kuninori Morimoto (5): ASoC: rsnd: enable clk_i approximate rate usage ASoC: rsnd: setup clock-out only when all conditions are right ASoC: rsnd: tidyup brga/brgb default value ASoC: rsnd: remove default division of clock out ASoC: rsnd: setup BRGCKR/BRRA/BRRB on rsnd_adg_clk_control() sound/soc/sh/rcar/adg.c | 68 +++++++++++++++++++++++++++++++++-------- 1 file changed, 55 insertions(+), 13 deletions(-) -- 2.25.1

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[PATCH] ALSA: ac97: Fix possible NULL dereference in snd_ac97_mixer
by Su Hui 23 Aug '23

23 Aug '23

smatch error: sound/pci/ac97/ac97_codec.c:2354 snd_ac97_mixer() error: we previously assumed 'rac97' could be null (see line 2072) remove redundant assignment, return error if rac97 is NULL. Fixes: da3cec35dd3c ("ALSA: Kill snd_assert() in sound/pci/*") Signed-off-by: Su Hui <suhui(a)nfschina.com> --- sound/pci/ac97/ac97_codec.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/pci/ac97/ac97_codec.c b/sound/pci/ac97/ac97_codec.c index 9afc5906d662..80a65b8ad7b9 100644 --- a/sound/pci/ac97/ac97_codec.c +++ b/sound/pci/ac97/ac97_codec.c @@ -2069,8 +2069,8 @@ int snd_ac97_mixer(struct snd_ac97_bus *bus, struct snd_ac97_template *template, .dev_disconnect = snd_ac97_dev_disconnect, }; - if (rac97) - *rac97 = NULL; + if (!rac97) + return -EINVAL; if (snd_BUG_ON(!bus || !template)) return -EINVAL; if (snd_BUG_ON(template->num >= 4)) -- 2.30.2

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[PATCH 1/7] ASoC: SOF: amd: remove unused sha dma interrupt code
by Vijendar Mukunda 23 Aug '23

23 Aug '23

During initial development time for RN platform, when SHA dma gets completed, SHA DMA engine used to raise the ACP interrupt. In ACP interrupt handler, SHA DMA interrupt got handled. Currently SHA DMA compleition is verified by checking transfer count using read poll time out logic. Remove unused SHA dma interrupt handling code. Signed-off-by: Vijendar Mukunda <Vijendar.Mukunda(a)amd.com> --- sound/soc/sof/amd/acp.c | 9 +-------- 1 file changed, 1 insertion(+), 8 deletions(-) diff --git a/sound/soc/sof/amd/acp.c b/sound/soc/sof/amd/acp.c index b2e00a10a03e..630c2c5fe4c7 100644 --- a/sound/soc/sof/amd/acp.c +++ b/sound/soc/sof/amd/acp.c @@ -337,14 +337,7 @@ static irqreturn_t acp_irq_thread(int irq, void *context) { struct snd_sof_dev *sdev = context; const struct sof_amd_acp_desc *desc = get_chip_info(sdev->pdata); - unsigned int val, count = ACP_HW_SEM_RETRY_COUNT; - - val = snd_sof_dsp_read(sdev, ACP_DSP_BAR, desc->ext_intr_stat); - if (val & ACP_SHA_STAT) { - /* Clear SHA interrupt raised by PSP */ - snd_sof_dsp_write(sdev, ACP_DSP_BAR, desc->ext_intr_stat, val); - return IRQ_HANDLED; - } + unsigned int count = ACP_HW_SEM_RETRY_COUNT; while (snd_sof_dsp_read(sdev, ACP_DSP_BAR, desc->hw_semaphore_offset)) { /* Wait until acquired HW Semaphore lock or timeout */ -- 2.34.1

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[PATCH 1/2] ASoC: dt-bindings: nau8821: Add single-ended input feature
by Seven Lee 23 Aug '23

23 Aug '23

Add input with single-ended control. Signed-off-by: Seven Lee <wtli(a)nuvoton.com> --- .../devicetree/bindings/sound/nuvoton,nau8821.yaml | 7 +++++++ 1 file changed, 7 insertions(+) diff --git a/Documentation/devicetree/bindings/sound/nuvoton,nau8821.yaml b/Documentation/devicetree/bindings/sound/nuvoton,nau8821.yaml index fc2f4ce4db88..3e54abd4ca74 100644 --- a/Documentation/devicetree/bindings/sound/nuvoton,nau8821.yaml +++ b/Documentation/devicetree/bindings/sound/nuvoton,nau8821.yaml @@ -89,6 +89,12 @@ properties: $ref: /schemas/types.yaml#/definitions/uint32 default: 3072000 + nuvoton,left-input-single-end: + description: Enable left input with single-ended settings if set. + For the headset mic application, the single-ended control is + just limited to the left adc for design demand. + type: boolean + '#sound-dai-cells': const: 0 @@ -114,6 +120,7 @@ examples: nuvoton,jkdet-pull-enable; nuvoton,jkdet-pull-up; nuvoton,key-enable; + nuvoton,left-input-single-end; nuvoton,jkdet-polarity = <GPIO_ACTIVE_LOW>; nuvoton,micbias-voltage = <6>; nuvoton,vref-impedance = <2>; -- 2.25.1

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