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

[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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[PATCH] ASoC: codecs: aw88395: Uninitialized variable bug in aw_dev_parse_dev_type_v1()
by Dan Carpenter 06 Feb '23

06 Feb '23

The "cur_scene_id" variable is never initialized. It needs to be set to zero for the code to work. Fixes: 4345865b003b ("ASoC: codecs: ACF bin parsing and check library file for aw88395") Signed-off-by: Dan Carpenter <error27(a)gmail.com> --- Presumably this code was tested with CONFIG_CC_HAS_AUTO_VAR_INIT_ZERO=y or whatever to automatically initialize it to zero or it would have died specatularly. ;) 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; -- 2.35.1

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[PATCH v5] ASoC: dt-bindings: Add tas2781 amplifier
by Shenghao Ding 05 Feb '23

05 Feb '23

Create tas2781.yaml for tas2781 driver. Signed-off-by: Shenghao Ding <13916275206(a)139.com> --- Changes in v5: - Drop the full stop in the Subject. - Drop the reset-gpios reference to gpio.txt. - Add Changelog. Changes to be committed: new file: Documentation/devicetree/bindings/sound/ti,tas2781.yaml --- .../devicetree/bindings/sound/ti,tas2781.yaml | 87 +++++++++++++++++++ 1 file changed, 87 insertions(+) create mode 100644 Documentation/devicetree/bindings/sound/ti,tas2781.yaml diff --git a/Documentation/devicetree/bindings/sound/ti,tas2781.yaml b/Documentation/devicetree/bindings/sound/ti,tas2781.yaml new file mode 100644 index 000000000000..8af44792a904 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/ti,tas2781.yaml @@ -0,0 +1,87 @@ +# SPDX-License-Identifier: (GPL-2.0-only OR BSD-2-Clause) +# Copyright (C) 2022 - 2023 Texas Instruments Incorporated +%YAML 1.2 +--- +$id: http://devicetree.org/schemas/sound/ti,tas2781.yaml# +$schema: http://devicetree.org/meta-schemas/core.yaml# + +title: Texas Instruments TAS2781 SmartAMP + +maintainers: + - Shenghao Ding <shenghao-ding(a)ti.com> + - Kevin Lu <kevin-lu(a)ti.com> + +description: | + The TAS2781 is a mono, digital input Class-D audio amplifier + optimized for efficiently driving high peak power into small + loudspeakers. Integrated an on-chip DSP supports Texas Instruments + Smart Amp speaker protection algorithm. The integrated speaker + voltage and current sense provides for real time + monitoring of loudspeaker behavior. + +properties: + compatible: + enum: + - ti,tas2781 + + reg: + maxItems: 1 + description: | + I2C address of the device can be in range from 0x38 to 0x40. + + reset-gpios: + maxItems: 1 + description: | + A GPIO line handling reset of the chip. As the line is active high, + it should be marked GPIO_ACTIVE_HIGH. + + interrupts: + maxItems: 1 + + ti,audio-slots: + $ref: /schemas/types.yaml#/definitions/uint32-array + minItems: 1 + maxItems: 4 + description: | + I2c address of the device for different audio slots, + useless in mono case. + + ti,broadcast-addr: + $ref: /schemas/types.yaml#/definitions/uint32 + description: | + Generic i2c address for all the tas2781 devices in + purpose of I2C broadcast during the multi-device + writes, useless in mono case. + + '#sound-dai-cells': + const: 1 + +required: + - compatible + - reg + +additionalProperties: false + +examples: + - | + #include <dt-bindings/gpio/gpio.h> + i2c { + /* example with quad support, such as tablet or pad device */ + #address-cells = <1>; + #size-cells = <0>; + quad: codec@38 { + compatible = "ti,tas2781"; + reg = <0x38>; + #sound-dai-cells = <1>; + reset-gpios = < &gpio1 10 GPIO_ACTIVE_HIGH >; + interrupt-parent = <&gpio1>; + interrupts = <15>; + ti,audio-slots = < 0x38 /* topleft-channel */ + 0x39 /* topright-channel */ + 0x3a /* bottomleft-channel */ + 0x3b /* bottomright-channel */ + >; + ti,broadcast-addr = <0x40>; + }; + }; +... -- 2.34.1

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Make ALC4080 front panel audio work on ASUS ROG STRIX X670E-E GAMING WIFI
by Jesper Juhl 04 Feb '23

04 Feb '23

Hi In order to have working front panel audio for my computer based on a ASUS ROG STRIX X670E-E GAMING WIFI motherboard with ALC4080 audio, I need to apply the following patch. It would be great if it could be applied up-stream so I (and other users with the same hardware) don't have to do this manually in the future. Kind regards, Jesper Juhl --- /tmp/USB-Audio.conf.original 2023-02-04 12:50:27.691689435 +0100 +++ /usr/share/alsa/ucm2/USB-Audio/USB-Audio.conf 2023-02-04 12:53:43.307961781 +0100 @@ -40,6 +40,7 @@ # 0b05:1996 ASUS on multiple boards (including ASUS ROG Maximus XIII) # 0b05:1a16 ASUS ROG Strix B660-F Gaming WiFi # 0b05:1a20 ASUS ROG STRIX Z690-I Gaming Wifi + # 0b05:1a52 ASUS ROG STRIX X670E-E GAMING WIFI # 0b05:1a27 ALC4082 on ASUS ROG Maximus Z690 Hero # 0db0:005a MSI MPG Z690 CARBON WIFI # 0db0:151f MSI X570S EDGE MAX WIFI @@ -50,7 +51,7 @@ # 0db0:a47c MSI MEG X570S Ace Max # 0db0:b202 MSI MAG Z690 Tomahawk Wifi # 0db0:d6e7 MSI MPG X670E Carbon Wifi - Regex "USB((0414:a00e)|(0b05:(1996|1a(16|2[07])))|(0db0:(005a|151f|1feb|419c|82c7|a073|a47c|b202|d6e7)))" + Regex "USB((0414:a00e)|(0b05:(1996|1a(52|16|2[07])))|(0db0:(005a|151f|1feb|419c|82c7|a073|a47c|b202|d6e7)))" } True.Define.ProfileName "Realtek/ALC4080" }

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