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[PATCH] ASoC: dt-bindings: renesas: adjust to R-Car Gen4
by Kuninori Morimoto 07 Feb '23

07 Feb '23

From: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> R-Car Gen4 is not compatible with Gen3, this patch adjusts to R-Car Gen4. Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> --- The "required" with if - then - else on "rcar_sound,ssi" is always match to "then" even though it is checking "renesas,rcar_sound-gen4" or not. Why ?? Is it my fault ?? .../bindings/sound/renesas,rsnd.yaml | 62 ++++++++++++++----- 1 file changed, 46 insertions(+), 16 deletions(-) diff --git a/Documentation/devicetree/bindings/sound/renesas,rsnd.yaml b/Documentation/devicetree/bindings/sound/renesas,rsnd.yaml index d106de00c6b2..9a88b1c34e72 100644 --- a/Documentation/devicetree/bindings/sound/renesas,rsnd.yaml +++ b/Documentation/devicetree/bindings/sound/renesas,rsnd.yaml @@ -106,7 +106,9 @@ properties: items: oneOf: - const: ssi-all + - const: clkin - pattern: '^ssi\.[0-9]$' + - pattern: '^ssiu\.[0-9]$' - pattern: '^src\.[0-9]$' - pattern: '^mix\.[0-1]$' - pattern: '^ctu\.[0-1]$' @@ -254,10 +256,20 @@ properties: no-busif: description: BUSIF is not used when [mem -> SSI] via DMA case $ref: /schemas/types.yaml#/definitions/flag - required: - - interrupts - - dmas - - dma-names + allOf: + - if: + properties: + compatible: + contains: + const: renesas,rcar_sound-gen4 + then: + required: + - interrupts + else: + required: + - interrupts + - dmas + - dma-names additionalProperties: false # For DAI base @@ -307,18 +319,36 @@ allOf: - ssi - adg else: - properties: - reg: - maxItems: 5 - reg-names: - maxItems: 5 - items: - enum: - - scu - - adg - - ssiu - - ssi - - audmapp + if: + properties: + compatible: + contains: + const: renesas,rcar_sound-gen4 + then: + properties: + reg: + maxItems: 4 + reg-names: + maxItems: 4 + items: + enum: + - adg + - ssiu + - ssi + - sdmc + else: + properties: + reg: + maxItems: 5 + reg-names: + maxItems: 5 + items: + enum: + - scu + - adg + - ssiu + - ssi + - audmapp unevaluatedProperties: false -- 2.25.1

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[PATCH] ASoC: codecs: es8326: Fix DTS properties reading
by Alexey Firago 06 Feb '23

06 Feb '23

Seems like properties parsing and reading was copy-pasted, so "everest,interrupt-src" and "everest,interrupt-clk" are saved into the es8326->jack_pol variable. This might lead to wrong settings being saved into the reg 57 (ES8326_HP_DET). Fix this by using proper variables while reading properties. Signed-off-by: Alexey Firago <a.firago(a)yadro.com> --- sound/soc/codecs/es8326.c | 6 ++++-- 1 file changed, 4 insertions(+), 2 deletions(-) diff --git a/sound/soc/codecs/es8326.c b/sound/soc/codecs/es8326.c index 9ddf6a35e91c..28a0565c2a95 100644 --- a/sound/soc/codecs/es8326.c +++ b/sound/soc/codecs/es8326.c @@ -729,14 +729,16 @@ static int es8326_probe(struct snd_soc_component *component) } dev_dbg(component->dev, "jack-pol %x", es8326->jack_pol); - ret = device_property_read_u8(component->dev, "everest,interrupt-src", &es8326->jack_pol); + ret = device_property_read_u8(component->dev, "everest,interrupt-src", + &es8326->interrupt_src); if (ret != 0) { dev_dbg(component->dev, "interrupt-src return %d", ret); es8326->interrupt_src = ES8326_HP_DET_SRC_PIN9; } dev_dbg(component->dev, "interrupt-src %x", es8326->interrupt_src); - ret = device_property_read_u8(component->dev, "everest,interrupt-clk", &es8326->jack_pol); + ret = device_property_read_u8(component->dev, "everest,interrupt-clk", + &es8326->interrupt_clk); if (ret != 0) { dev_dbg(component->dev, "interrupt-clk return %d", ret); es8326->interrupt_clk = 0x45; -- 2.34.1

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[PATCH v3 0/2] Two bug fixes for tas5805m codec driver
by Daniel Beer 06 Feb '23

06 Feb '23

This pair of patches fixes two issues which crept in while revising the original submission, at a time when I no longer had access to test hardware. The fixes here have been tested and verified on hardware. Changes v2 -> v3: - Resubmit with cover letter Changes v1 -> v2: - Fix issue reported by kernel test robot Daniel Beer (2): ASoC: tas5805m: rework to avoid scheduling while atomic. ASoC: tas5805m: add missing page switch. sound/soc/codecs/tas5805m.c | 131 +++++++++++++++++++++++++----------- 1 file changed, 90 insertions(+), 41 deletions(-) -- 2.38.1

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[PATCH] ALSA: hda/realtek: Enable mute/micmute LEDs on HP Elitebook, 645 G9
by Elvis Angelaccio 06 Feb '23

06 Feb '23

The HP Elitebook 645 G9 laptop (with motherboard model 89D2) uses the ALC236 codec and requires the alc236_fixup_hp_mute_led_micmute_vref fixup in order to enable mute/micmute LEDs. Note: the alc236_fixup_hp_gpio_led fixup, which is used by the Elitebook 640 G9, does not work with the 645 G9. Signed-off-by: Elvis Angelaccio <elvis.angelaccio(a)kde.org> --- 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 4055a8f5880a..11bc71b152ff 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -9427,6 +9427,7 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK(0x103c, 0x8aa0, "HP ProBook 440 G9 (MB 8A9E)", ALC236_FIXUP_HP_GPIO_LED), SND_PCI_QUIRK(0x103c, 0x8aa3, "HP ProBook 450 G9 (MB 8AA1)", ALC236_FIXUP_HP_GPIO_LED), SND_PCI_QUIRK(0x103c, 0x8aa8, "HP EliteBook 640 G9 (MB 8AA6)", ALC236_FIXUP_HP_GPIO_LED), + SND_PCI_QUIRK(0x103c, 0x89d3, "HP EliteBook 645 G9 (MB 89D2)", ALC236_FIXUP_HP_MUTE_LED_MICMUTE_VREF), SND_PCI_QUIRK(0x103c, 0x8aab, "HP EliteBook 650 G9 (MB 8AA9)", ALC236_FIXUP_HP_GPIO_LED), SND_PCI_QUIRK(0x103c, 0x8abb, "HP ZBook Firefly 14 G9", ALC245_FIXUP_CS35L41_SPI_2_HP_GPIO_LED), SND_PCI_QUIRK(0x103c, 0x8ad1, "HP EliteBook 840 14 inch G9 Notebook PC", ALC245_FIXUP_CS35L41_SPI_2_HP_GPIO_LED), -- 2.39.1

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[PATCH 00/19] Add soundwire support for Pink Sardine platform
by Vijendar Mukunda 06 Feb '23

06 Feb '23

Pink Sardine(ps) platform is based on ACP6.3 Architecture. ACP6.3 IP has two soundwire controller instance support. This patchset add support for Soundwire controller on Pink Sardine platform. Signed-off-by: Vijendar Mukunda <Vijendar.Mukunda(a)amd.com> Vijendar Mukunda (19): ASoC: amd: ps: create platform devices based on acp config soundwire: amd: Add support for AMD Master driver soundwire: amd: register sdw controller dai ops soundwire: amd: enable build for AMD soundwire master driver soundwire: amd: add soundwire interrupt handling ASoC: amd: ps: add support for soundwire interrupts in acp pci driver ASoC: amd: ps: add soundwire dma driver for pink sardine platform ASoC: amd: ps: add soundwire dma driver dma ops ASoC: amd: ps: add support for Soundwire DMA interrupts ASoC: amd: ps: enable Soundwire DMA driver build ASoC: amd: update comments in Kconfig file ASoC: amd: ps: Add soundwire specific checks in pci driver in pm ops. ASoC: amd: ps: add support for runtime pm ops for soundwire dma driver soundwire: amd: add runtime pm ops for AMD master driver soundwire: amd: add startup and shutdown dai ops soundwire: amd: handle wake enable interrupt soundwire: amd: add pm_prepare callback and pm ops support ASoC: amd: ps: implement system level pm ops for soundwire dma driver ASoC: amd: ps: increase runtime suspend delay drivers/soundwire/Kconfig | 9 + drivers/soundwire/Makefile | 4 + drivers/soundwire/amd_master.c | 1734 +++++++++++++++++++++++++++++ drivers/soundwire/amd_master.h | 284 +++++ include/linux/soundwire/sdw_amd.h | 65 ++ sound/soc/amd/Kconfig | 3 +- sound/soc/amd/ps/Makefile | 2 + sound/soc/amd/ps/acp63.h | 98 +- sound/soc/amd/ps/pci-ps.c | 383 ++++++- sound/soc/amd/ps/ps-sdw-dma.c | 728 ++++++++++++ 10 files changed, 3287 insertions(+), 23 deletions(-) create mode 100644 drivers/soundwire/amd_master.c create mode 100644 drivers/soundwire/amd_master.h create mode 100644 include/linux/soundwire/sdw_amd.h create mode 100644 sound/soc/amd/ps/ps-sdw-dma.c -- 2.34.1

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[PATCH v1] ALSA: hda/realtek: Add quirk for ASUS UM3402 using CS35L41
by Stefan Binding 06 Feb '23

06 Feb '23

This Asus Zenbook laptop use Realtek HDA codec combined with 2xCS35L41 Amplifiers using I2C with External Boost. Signed-off-by: Stefan Binding <sbinding(a)opensource.cirrus.com> --- 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 4055a8f5880a..251b58069d6c 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -9480,6 +9480,7 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK(0x1043, 0x1d4e, "ASUS TM420", ALC256_FIXUP_ASUS_HPE), SND_PCI_QUIRK(0x1043, 0x1e02, "ASUS UX3402", ALC245_FIXUP_CS35L41_SPI_2), SND_PCI_QUIRK(0x1043, 0x1e11, "ASUS Zephyrus G15", ALC289_FIXUP_ASUS_GA502), + SND_PCI_QUIRK(0x1043, 0x1e12, "ASUS UM3402", ALC287_FIXUP_CS35L41_I2C_2), SND_PCI_QUIRK(0x1043, 0x1e51, "ASUS Zephyrus M15", ALC294_FIXUP_ASUS_GU502_PINS), SND_PCI_QUIRK(0x1043, 0x1e5e, "ASUS ROG Strix G513", ALC294_FIXUP_ASUS_G513_PINS), SND_PCI_QUIRK(0x1043, 0x1e8e, "ASUS Zephyrus G15", ALC289_FIXUP_ASUS_GA401), -- 2.34.1

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Overflow in calculating audio timestamp
by Alan Young 06 Feb '23

06 Feb '23

sound/core/pcm_lib.c:update_audio_tstamp() contains the following calculation: audio_nsecs = div_u64(audio_frames * 1000000000LL, runtime->rate); This will result in a 64-bit overflow after 4.4 days at 48000 Hz, or 1.1 days at 192000. Are you interested in a patch to improve this? The same calculation occurs in a couple of other places. Alan.

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sample_size field in snd_dec_flac structure
by Carlos Rafael Giani 06 Feb '23

06 Feb '23

In snd_dec_flac (see: https://github.com/torvalds/linux/blob/master/include/uapi/sound/compress_p… ), there is the sample_size field. It seems to me that this is meant to contain the same value as the sample size as the one in the frame header: https://xiph.org/flac/format.html#frame_header Does this mean that if I always send a full FLAC stream into the device, with its complete STREAMINFO header, I can just always set sample_size to 0? The spec does mention that 0 mean "get [the sample size] from STREAMINFO metadata block".

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[PATCH] ASoC: codecs: Fix unsigned comparison with less than zero
by Jiapeng Chong 06 Feb '23

06 Feb '23

The val is defined as unsigned int type, if(val<0) is redundant, so delete it. sound/soc/codecs/idt821034.c:449 idt821034_kctrl_gain_put() warn: unsigned 'val' is never less than zero. Reported-by: Abaci Robot <abaci(a)linux.alibaba.com> Link: https://bugzilla.openanolis.cn/show_bug.cgi?id=3947 Signed-off-by: Jiapeng Chong <jiapeng.chong(a)linux.alibaba.com> --- sound/soc/codecs/idt821034.c | 2 -- 1 file changed, 2 deletions(-) diff --git a/sound/soc/codecs/idt821034.c b/sound/soc/codecs/idt821034.c index 5d01787b1c1f..2cc7b9166e69 100644 --- a/sound/soc/codecs/idt821034.c +++ b/sound/soc/codecs/idt821034.c @@ -446,8 +446,6 @@ static int idt821034_kctrl_gain_put(struct snd_kcontrol *kcontrol, u8 ch; val = ucontrol->value.integer.value[0]; - if (val < 0) - return -EINVAL; if (val > max - min) return -EINVAL; -- 2.20.1.7.g153144c

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[PATCH] ASoC: codecs: aw88395: initialize cur_scene_id to 0
by Tom Rix 06 Feb '23

06 Feb '23

cppcheck reports sound/soc/codecs/aw88395/aw88395_lib.c:789:6: error: Uninitialized variable: cur_scene_id [uninitvar] if (cur_scene_id == 0) { ^ Passing a garbage value to aw_dev_parse_data_by_sec_type_v1() will cause a crash when the value is used as an array index. This check assumes cur_scene_id is initialized to 0, so initialize it to 0. Fixes: 4345865b003b ("ASoC: codecs: ACF bin parsing and check library file for aw88395") Signed-off-by: Tom Rix <trix(a)redhat.com> --- sound/soc/codecs/aw88395/aw88395_lib.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/codecs/aw88395/aw88395_lib.c b/sound/soc/codecs/aw88395/aw88395_lib.c index 64dde972f3f0..d7c31a202adc 100644 --- a/sound/soc/codecs/aw88395/aw88395_lib.c +++ b/sound/soc/codecs/aw88395/aw88395_lib.c @@ -769,7 +769,7 @@ static int aw_dev_parse_dev_type_v1(struct aw_device *aw_dev, { struct aw_cfg_dde_v1 *cfg_dde = (struct aw_cfg_dde_v1 *)((char *)prof_hdr + prof_hdr->hdr_offset); - int cur_scene_id; + int cur_scene_id = 0; unsigned int i; int ret; -- 2.26.3

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