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

Rockchip I2S commit possibly ignored
by Geraldo Nascimento 16 Aug '22

16 Aug '22

Hi Mark, I was looking at Rockchip I2S commits and it seems "ASoC: rockchip: i2s: Reset the controller if soft reset failed" was supposed to have been merged to your sound.git but never was. I don't know if this was intentional or a real miss but in any case I'm letting you know. According to 515b436be291ff197c52198282bbb19e79c9d197 'Merge series "Patches to update for rockchip i2s" from Sugar Zhang <sugar.zhang(a)rock-chips.com>:' it should have been merged to your tree. In rockchip-i2s.yaml there's even documentation that references the missing commit. However in the alsa-devel archives, https://mailman.alsa-project.org/pipermail/alsa-devel/2021-August/189050.ht… I see there was no commit info for the unmerged patch. Perhaps this caused it to be black-holed? Thanks, Geraldo Nascimento

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Re: [PATCH v6 1/2] ASoC: dt-bindings: Add Everest ES8326 audio CODEC
by Mark Brown 16 Aug '22

16 Aug '22

On Tue, 16 Aug 2022 10:44:55 +0800, Zhu Ning wrote: > Add device tree binding documentation for Everest ES8326 > > ---- > v5 tested by dtschema > > Applied to https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git for-next Thanks! [1/2] ASoC: dt-bindings: Add Everest ES8326 audio CODEC commit: 8c6789f4e2d4ee7d6c8c60daa88ea7a4c4cf6779 [2/2] ASoC: codecs: add support for ES8326 commit: 5c439937775d77a334696a98fb2a25dee72ffa2d All being well this means that it will be integrated into the linux-next tree (usually sometime in the next 24 hours) and sent to Linus during the next merge window (or sooner if it is a bug fix), however if problems are discovered then the patch may be dropped or reverted. You may get further e-mails resulting from automated or manual testing and review of the tree, please engage with people reporting problems and send followup patches addressing any issues that are reported if needed. If any updates are required or you are submitting further changes they should be sent as incremental updates against current git, existing patches will not be replaced. Please add any relevant lists and maintainers to the CCs when replying to this mail. Thanks, Mark

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[RESEND v4 0/2] Add support for linked list to store acp_stream and tdm support.
by Venkata Prasad Potturu 16 Aug '22

16 Aug '22

This patch is to add support for linked list to store acp_stream instead static array and add tdm support for acp I2S stream. Ajit Kumar Pandey (1): ASoC: amd: acp: Initialize list to store acp_stream during pcm_open Venkata Prasad Potturu (1): ASoC: amd: acp: Add TDM support for acp i2s stream Changes since v3: -- Modify to spin_lock() in irq handler and spin_lock_irq() in dma open and close functions instead of spin_lock_irqsave(). -- Add tdm support patch to this series. Changes since v2: -- Add locking mechanism in irq handler, linked list updatation and deletion. Changes since v1: -- Fix compile error and remove unused variable. sound/soc/amd/acp/acp-i2s.c | 80 +++++++++++++++++++++++++++++++- sound/soc/amd/acp/acp-platform.c | 40 ++++++++-------- sound/soc/amd/acp/amd.h | 16 ++++++- 3 files changed, 113 insertions(+), 23 deletions(-) -- 2.25.1

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[PATCH v5 1/2] ASoC: dt-bindings: Add Everest ES8326 audio CODEC
by Zhu Ning 16 Aug '22

16 Aug '22

Add device tree binding documentation for Everest ES8326 Signed-off-by: David Yang <yangxiaohua(a)everest-semi.com> Signed-off-by: Zhu Ning <zhuning(a)everest-semi.com> ---- v5 tested by dtschema --- .../bindings/sound/everest,es8326.yaml | 116 ++++++++++++++++++ 1 file changed, 116 insertions(+) create mode 100755 Documentation/devicetree/bindings/sound/everest,es8326.yaml diff --git a/Documentation/devicetree/bindings/sound/everest,es8326.yaml b/Documentation/devicetree/bindings/sound/everest,es8326.yaml new file mode 100755 index 000000000000..07781408e788 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/everest,es8326.yaml @@ -0,0 +1,116 @@ +# SPDX-License-Identifier: (GPL-2.0-only OR BSD-2-Clause) +%YAML 1.2 +--- +$id: http://devicetree.org/schemas/sound/everest,es8326.yaml# +$schema: http://devicetree.org/meta-schemas/core.yaml# + +title: Everest ES8326 audio CODEC + +maintainers: + - David Yang <yangxiaohua(a)everest-semi.com> + +properties: + compatible: + const: everest,es8326 + + reg: + maxItems: 1 + + clocks: + items: + - description: clock for master clock (MCLK) + + clock-names: + items: + - const: mclk + + "#sound-dai-cells": + const: 0 + + everest,jack-pol: + $ref: /schemas/types.yaml#/definitions/uint8 + description: | + just the value of reg 57. Bit(3) decides whether the jack polarity is inverted. + Bit(2) decides whether the button on the headset is inverted. + Bit(1)/(0) decides the mic properity to be OMTP/CTIA or auto. + minimum: 0x00 + maximum: 0x0f + default: 0x0f + + everest,mic1-src: + $ref: /schemas/types.yaml#/definitions/uint8 + description: + the value of reg 2A when headset plugged. + minimum: 0x00 + maximum: 0x77 + default: 0x22 + + everest,mic2-src: + $ref: /schemas/types.yaml#/definitions/uint8 + description: + the value of reg 2A when headset unplugged. + minimum: 0x00 + maximum: 0x77 + default: 0x44 + + everest,jack-detect-inverted: + $ref: /schemas/types.yaml#/definitions/flag + description: + Defined to invert the jack detection. + + everest,interrupt-src: + $ref: /schemas/types.yaml#/definitions/uint8 + description: | + value of reg 0x58, Defines the interrupt source. + Bit(2) 1 means button press triggers irq, 0 means not. + Bit(3) 1 means PIN9 is the irq source for jack detection. When set to 0, + bias change on PIN9 do not triggers irq. + Bit(4) 1 means PIN27 is the irq source for jack detection. + Bit(5) 1 means PIN9 is the irq source after MIC detect. + Bit(6) 1 means PIN27 is the irq source after MIC detect. + minimum: 0 + maximum: 0x3c + default: 0x08 + + everest,interrupt-clk: + $ref: /schemas/types.yaml#/definitions/uint8 + description: | + value of reg 0x59, Defines the interrupt output behavior. + Bit(0-3) 0 means irq pulse equals 512*internal clock + 1 means irq pulse equals 1024*internal clock + 2 means ... + 7 means irq pulse equals 65536*internal clock + 8 means irq mutes PA + 9 means irq mutes PA and DAC output + Bit(4) 1 means we invert the interrupt output. + Bit(6) 1 means the chip do not detect jack type after button released. + 0 means the chip detect jack type again after button released. + minimum: 0 + maximum: 0x7f + default: 0x45 + +required: + - compatible + - reg + - "#sound-dai-cells" + +additionalProperties: false + +examples: + - | + i2c { + #address-cells = <1>; + #size-cells = <0>; + es8326: codec@19 { + compatible = "everest,es8326"; + reg = <0x19>; + clocks = <&clks 10>; + clock-names = "mclk"; + #sound-dai-cells = <0>; + everest,mic1-src = [22]; + everest,mic2-src = [44]; + everest,jack-pol = [0e]; + everest,interrupt-src = [08]; + everest,interrupt-clk = [45]; + }; + }; -- 2.36.1

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[PATCH v2] ASoC: fsl_sai: fix incorrect mclk number in error message
by Pieterjan Camerlynck 16 Aug '22

16 Aug '22

In commit c3ecef21c3f26 ("ASoC: fsl_sai: add sai master mode support") the loop was changed to start iterating from 1 instead of 0. The error message however was not updated, reporting the wrong clock to the user. Signed-off-by: Pieterjan Camerlynck <pieterjan.camerlynck(a)gmail.com> --- V2: rebase against latest version --- sound/soc/fsl/fsl_sai.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/fsl/fsl_sai.c b/sound/soc/fsl/fsl_sai.c index 7523bb944b21..d430eece1d6b 100644 --- a/sound/soc/fsl/fsl_sai.c +++ b/sound/soc/fsl/fsl_sai.c @@ -1306,7 +1306,7 @@ static int fsl_sai_probe(struct platform_device *pdev) sai->mclk_clk[i] = devm_clk_get(dev, tmp); if (IS_ERR(sai->mclk_clk[i])) { dev_err(dev, "failed to get mclk%d clock: %ld\n", - i + 1, PTR_ERR(sai->mclk_clk[i])); + i, PTR_ERR(sai->mclk_clk[i])); sai->mclk_clk[i] = NULL; } } -- 2.25.1

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[PATCH] ALSA: hda/realtek: Add quirk for Lenovo Yoga7 14IAL7
by Takashi Iwai 16 Aug '22

16 Aug '22

Lenovo Yoga7 14IAL7 requires the same quirk as Lenovo Yoga9 14IAP7 for fixing the bass speaker problems. Reported-by: Pascal Gross <baipush(a)tutanota.com> Cc: <stable(a)vger.kernel.org> Link: https://lore.kernel.org/r/N9_CjBz--3-2@tutanota.com Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/pci/hda/patch_realtek.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index f544761eb11b..b42496c01c43 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -9492,6 +9492,7 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK(0x17aa, 0x3852, "Lenovo Yoga 7 14ITL5", ALC287_FIXUP_YOGA7_14ITL_SPEAKERS), SND_PCI_QUIRK(0x17aa, 0x3853, "Lenovo Yoga 7 15ITL5", ALC287_FIXUP_YOGA7_14ITL_SPEAKERS), SND_PCI_QUIRK(0x17aa, 0x3855, "Legion 7 16ITHG6", ALC287_FIXUP_LEGION_16ITHG6), + SND_PCI_QUIRK(0x17aa, 0x3869, "Lenovo Yoga7 14IAL7", ALC287_FIXUP_YOGA9_14IAP7_BASS_SPK_PIN), SND_PCI_QUIRK(0x17aa, 0x3902, "Lenovo E50-80", ALC269_FIXUP_DMIC_THINKPAD_ACPI), SND_PCI_QUIRK(0x17aa, 0x3977, "IdeaPad S210", ALC283_FIXUP_INT_MIC), SND_PCI_QUIRK(0x17aa, 0x3978, "Lenovo B50-70", ALC269_FIXUP_DMIC_THINKPAD_ACPI), -- 2.35.3

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[GIT PULL] ASoC fixes for v6.0-rc1
by Mark Brown 16 Aug '22

16 Aug '22

The following changes since commit 40d060b8158e26438398bf1132925f666e3b6480: ASoC: q6asm: use kcalloc() instead of kzalloc() (2022-07-28 11:59:10 +0100) are available in the Git repository at: https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git tags/asoc-fix-v6.0-rc1 for you to fetch changes up to b4b5f29a076e52181f63e45a2ad1bc88593072e3: ASoC: codec: tlv320aic32x4: fix mono playback via I2S (2022-08-10 14:52:05 +0100) ---------------------------------------------------------------- ASoC: Fixes for v6.0 A relatively large batch of fixes that came in since my pull request, none of them too major and mostly device specific apart from a series of security/robustness improvements from Takashi. ---------------------------------------------------------------- Biju Das (1): ASoC: sh: rz-ssi: Improve error handling in rz_ssi_probe() error path Justin Stitt (1): ASoC: SOF: ipc3-topology: Fix clang -Wformat warning Mark Brown (2): ASoC: Fix theoretical buffer overflow by snprintf() TAS2770 fixes Martin Povišer (4): ASoC: tas2770: Set correct FSYNC polarity ASoC: tas2770: Allow mono streams ASoC: tas2770: Drop conflicting set_bias_level power setting ASoC: tas2770: Fix handling of mute/unmute Oder Chiou (1): ASoC: rt5640: Fix the JD voltage dropping issue Philipp Zabel (1): ASoC: codec: tlv320aic32x4: fix mono playback via I2S Pierre-Louis Bossart (1): ASoC: Intel: fix sof_es8336 probe Takashi Iwai (4): ASoC: Intel: avs: Fix potential buffer overflow by snprintf() ASoC: SOF: debug: Fix potential buffer overflow by snprintf() ASoC: SOF: Intel: hda: Fix potential buffer overflow by snprintf() ASoC: DPCM: Don't pick up BE without substream syed sabakareem (1): ASoC: amd: yc: Update DMI table entries for AMD platforms sound/soc/amd/yc/acp6x-mach.c | 28 +++++++++++ sound/soc/codecs/rt5640.c | 5 +- sound/soc/codecs/tas2770.c | 98 ++++++++++++++++--------------------- sound/soc/codecs/tas2770.h | 5 ++ sound/soc/codecs/tlv320aic32x4.c | 9 ++++ sound/soc/intel/avs/pcm.c | 4 +- sound/soc/intel/boards/sof_es8336.c | 4 +- sound/soc/sh/rz-ssi.c | 26 +++++----- sound/soc/soc-pcm.c | 3 ++ sound/soc/sof/debug.c | 6 +-- sound/soc/sof/intel/hda.c | 2 +- sound/soc/sof/ipc3-topology.c | 2 +- 12 files changed, 116 insertions(+), 76 deletions(-)

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[PATCH v2] ALSA: hda: cs35l41: Clarify support for CSC3551 without _DSD Properties
by Stefan Binding 16 Aug '22

16 Aug '22

For devices which use HID CSC3551, correct ACPI _DSD properties are required to be able support those systems. Add error message to clarify this. Signed-off-by: Stefan Binding <sbinding(a)opensource.cirrus.com> --- sound/pci/hda/cs35l41_hda.c | 5 +++++ 1 file changed, 5 insertions(+) diff --git a/sound/pci/hda/cs35l41_hda.c b/sound/pci/hda/cs35l41_hda.c index 129bffb431c2..15e2a0009080 100644 --- a/sound/pci/hda/cs35l41_hda.c +++ b/sound/pci/hda/cs35l41_hda.c @@ -1163,6 +1163,11 @@ static int cs35l41_no_acpi_dsd(struct cs35l41_hda *cs35l41, struct device *physd hw_cfg->gpio1.func = CS35l41_VSPK_SWITCH; hw_cfg->gpio1.valid = true; } else { + /* + * Note: CLSA010(0/1) are special cases which use a slightly different design. + * All other HIDs e.g. CSC3551 require valid ACPI _DSD properties to be supported. + */ + dev_err(cs35l41->dev, "Error: ACPI _DSD Properties are missing for HID %s.\n", hid); hw_cfg->valid = false; hw_cfg->gpio1.valid = false; hw_cfg->gpio2.valid = false; -- 2.34.1

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[PATCH v2 0/6] ALSA: hda: Unify codec construction
by Cezary Rojewski 16 Aug '22

16 Aug '22

A follow up to the recent HDAudio fixes series [1]. Given the recently reported regression [2], before the page fault occurring on codec shutdown can be fixed, codec construction procedure needs to be updated for skylake and sof-intel drivers. Drivers: pci-hda and avs need no changes - already making use of snd_hda_codec_device_init(). As subject touches code used by the sof-driver, additional review has been conducted on thesofproject/linux [3]. Changes in v2: - dropped snd_hda_ext_core <-> snd_hda_codec dependency by calling snd_hda_codec_device_init() directly in skylake and sof drivers probe enumeration routines, as suggested by Takashi - skylake/sof portion of the change has been split into two separate patches - new functions that aim to replace hdac_ext codec init & exit functionality are added first - for skylake and sof drivers both - third patch in the series now combines the "field -> pointer" change for hdac_hda_priv->codec plus the codec-enumeration adjustments for skylake and sof drivers Both above are here to keep git bisect happy, as suggested by Pierre [1]: https://lore.kernel.org/alsa-devel/20220706120230.427296-7-cezary.rojewski@… [2]: https://lore.kernel.org/alsa-devel/3c40df55-3aee-1e08-493b-7b30cd84dc00@lin… [3]: https://github.com/thesofproject/linux/pull/3775 Cezary Rojewski (6): ASoC: Intel: Skylake: Introduce HDA codec init and exit routines ASoC: SOF: Intel: Introduce HDA codec init and exit routines ASoC: Intel: Drop hdac_ext usage for codec device creation ALSA: hda: Always free codec on the device release ALSA: hda: Remove codec init and exit routines ALSA: hda: Fix page fault in snd_hda_codec_shutdown() include/sound/hda_codec.h | 2 - include/sound/hdaudio_ext.h | 3 -- sound/hda/ext/hdac_ext_bus.c | 53 ------------------- sound/pci/hda/hda_codec.c | 49 ++++++++--------- sound/soc/codecs/hdac_hda.c | 26 ++++----- sound/soc/codecs/hdac_hda.h | 2 +- sound/soc/intel/boards/hda_dsp_common.c | 2 +- sound/soc/intel/boards/skl_hda_dsp_generic.c | 2 +- sound/soc/intel/skylake/skl.c | 53 ++++++++++++++----- sound/soc/sof/intel/hda-codec.c | 55 ++++++++++++++------ 10 files changed, 113 insertions(+), 134 deletions(-) -- 2.25.1

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[PATCH] ASoC: Fix the include guard used for include/sound/wm8904.h
by Christophe JAILLET 16 Aug '22

16 Aug '22

__MFD_WM8994_PDATA_H__ is already used for: include/linux/mfd/wm8994/pdata.h Based on file names, use __MFD_WM8904_PDATA_H__ instead here. Signed-off-by: Christophe JAILLET <christophe.jaillet(a)wanadoo.fr> --- Not sure of if a Fixes: tag is needed. If yes, I would use: 9e50108668a7 ("mfd: Add initial WM8994 support") which introduced the conflict, even if: a91eb199e4dc8 ("ASoC: Initial WM8904 CODEC driver") which introduced the typo is more logical. No strong opinion on it. --- include/sound/wm8904.h | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/include/sound/wm8904.h b/include/sound/wm8904.h index 88ac1870510e..f74293a83a17 100644 --- a/include/sound/wm8904.h +++ b/include/sound/wm8904.h @@ -7,8 +7,8 @@ * Author: Mark Brown <broonie(a)opensource.wolfsonmicro.com> */ -#ifndef __MFD_WM8994_PDATA_H__ -#define __MFD_WM8994_PDATA_H__ +#ifndef __MFD_WM8904_PDATA_H__ +#define __MFD_WM8904_PDATA_H__ /* Used to enable configuration of a GPIO to all zeros */ #define WM8904_GPIO_NO_CONFIG 0x8000 -- 2.34.1

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