December 2021 - Alsa-devel - mailman.alsa-project.org

Re: [PATCH] ASoC: tas2770: Fix setting of high sample rates
by Mark Brown 15 Dec '21

15 Dec '21

On Mon, 06 Dec 2021 22:45:43 +0000, Martin Povišer wrote: > Although the codec advertises support for 176.4 and 192 ksps, without > this fix setting those sample rates fails with EINVAL at hw_params time. > > Applied to https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git for-linus Thanks! [1/1] ASoC: tas2770: Fix setting of high sample rates commit: 80d5be1a057e05f01d66e986cfd34d71845e5190 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

1 0

[PATCH v2 0/2] ASoC: meson: aiu: two fixes (for 5.16)
by Martin Blumenstingl 15 Dec '21

15 Dec '21

In this series I am proposing two fixes for the "aiu" driver, used on Amlogic Meson8, Meson8b, Meson8m2, GXBB, GXL and GXM SoCs. The first patch is the result of me trying to understand the way how we get the DMA buffer and address for the audio data. I'm not an expert in terms of DMA. From what I understand we need to inform DMA core about the limitations of the hardware. In case of AIU it's DMA address registers only take 32 bits, so DMA core should be aware of this. The second patch is what I could come up with to fix the infamous I2S buffer underrun issue, also called the "machine gun noise" (MGN) bug. After a lot of testing, debugging and comparing vendor code with the upstream "aiu" driver I have come up with this fix. I have written down my thoughts in the description of that patch. To be clear: these are my thoughts, unfortunately I have no way of proving this other than asking other people to test this patch (off-list I have already received positive feedback along with confirmation that both 2-ch and 6-ch audio are still working fine. Even with Kodi's menu - which is an easy way to reproduce the MGN bug - sound output is fine with this patch). @Mark, @Liam: It would be great if at least the second patch could be queued as a fix for "for-5.16" as this solves a long standing buffer underrun. This issue is nasty because it can occur at any time and it can even loop forever. Christian provided an example with the speakers on low volume: [1]. Imagine the same sound on "medium" volume after playing a movie. Changes since v1 at [0]: - Dropped a paragraph about FIFO IRQs from the second patch because Jerome has a valid point that this behavior can either mean that AIU_I2S_MISC is related to the FIFO or the FIFO consumer (= encoder) - Add a bit of documentation explaining why we set the AIU_I2S_MISC_FORCE_LEFT_RIGHT bit (affects patch #2) - Collect Christian's and Geraldo's Tested-by as well as Jerome's Acked-by for patch #2 (thanks to all three of you!) - Cc linux-stable for the second patch - Dropped RFC prefix [0] https://chewitt.libreelec.tv/testing/wp2_audio_noise.mov [1] https://patchwork.kernel.org/project/linux-amlogic/cover/20211205180816.208… Martin Blumenstingl (2): ASoC: meson: aiu: fifo: Add missing dma_coerce_mask_and_coherent() ASoC: meson: aiu: Move AIU_I2S_MISC hold setting to aiu-fifo-i2s sound/soc/meson/aiu-encoder-i2s.c | 33 ------------------------------- sound/soc/meson/aiu-fifo-i2s.c | 19 ++++++++++++++++++ sound/soc/meson/aiu-fifo.c | 6 ++++++ 3 files changed, 25 insertions(+), 33 deletions(-) -- 2.34.1

2 3

[PATCH v7 0/2] Codec driver for TLV320ADC3001 and 3101
by Ricard Wanderlof 15 Dec '21

15 Dec '21

This is a codec driver for the TLV320ADC3001 and TLV320ADC3101 2 channel audio ADC chips from Texas Instruments. Based on an old driver for these chips that I was referred to by Texas Instruments, but which apparently was never upstreamed, I've taken patches from various incarnations, ported them and augmented them with various new features in order to make a working driver. There is currently no support for the on-chip "miniDSP". Although the TLV320ADC3001 and TLV320ADC3101 have similarities with other TI codec chips for which there are existing drivers, upon closer inspection I decided the differences were too great to warrant attempting to modify an existing driver, especially without access to all supported chips and resources for testing them all. There are certainly things that could be improved, among others in the area of PLL register value calculations, but it has been tested on the Axis ARTPEC-8 platform and has been found to work satisfactorily at least with a TLV320ADC3101 operating as an I2S slave at 48 kHz with 32 bit bit depth, 1 or 2 channels, both with the on-chip PLL enabled and disabled. I have not tested this on a TLV320ADC3001 chip; in the original driver the only differences were related to the now-removed initialization code, so I've just retained the compatible strings for both chips. Limitations: - GPIO pins on chip can only be set up as outputs. - Using BCLK as an input to clock chain is supported by the hw but not by driver. Changes v6->v7: - Minor review fixes in devicetree binding file and related include file. Changes v5->v6: - Added missing "depends on I2C" in the Kconfig (reported by lkp(a)intel.com). - Fold include file into source as it is not needed elsewhere. - Fix SPDX-License-Identifier which was using a deprecated license. - Clean up authorship information in file header, removing history. - Remove useless GPIOLIB ifdefs. - Use devm_gpiod_get for reset gpio rather than devm_gpio_request etc. - Handle EPROBE_DEFER properly for reset gpio request. - Use dev_err_probe rather than explicit EPROBE_DEFER checking. - Properly check value from DT parsing for mic bias and codec gpio pins. No changes to the devicetree bindings. Changes v4->v5: Yet another fix based on kernel test robot (lkp(a)intel.com), this time a legitimate issue: the return value from of_get_named_gpio_flags() was crammed into an unsigned int so any potential negative error codes were lost. No changes to the devicetree bindings. Changes v3->v4: More fixes reported by kernel test robot (lkp(a)intel.com). Don't know why these weren't reported the first time. Changes v2->v3: Fix a couple of issues reported by the kernel test robot (lkp(a)intel.com). Changes v1->v2: Large portions of the driver have been reworked from v1 based on remarks from Mark. More specifically: - Remove an unneeded #ifdef and EXPORT_SYMBOL_GPL. - Add ADC3XXX_ prefix to all macro names to avoid namespace contention. - Clean up certain macro names with mixed lower case - upper case names. - Do away with the D0..D7 macros representing bit values. - Fix general formatting: spaces before and after {, }, replace multi spaces with tabs, maximum 2 blank lines anywhere, and other minor issues. - Page register does not need to be volatile. - Rather than use 0 / -6dB enum for input attenuator, use standard TLV. - The dither offset control contains reserved values so use VALUE_ENUM instead of straight ENUM. - Replace vector of SOC_ENUMs with individual variables. - Fix ALSA control names: - Use Volume at the end of all gain controls, and remove "Gain" where applicable. - Use "Switch" at the end of all switch controls, either adding it or changing "switch" into "Switch" as applicable. - Switches and Volumes related to capture should be set up as such, so add "Capture" to widget names where appropriate. This is mostly important for the PGA Capture and Switch widgets, not just for consistency, but also because amixer simple Switch controls expect different names (cap/nocap instead of on/off) for control values for capture. - Make widget names have capital first letters for all consitutent words. - When resuming, the registers will not have changed, so no need to call snd_soc_component_cache_sync(). - Move reset gpio init to I2C probe - Remove needless _priv at end of private struct name. - Make hex constants lower case - _CBM_CFM/_CBS_CFS -> _CBP_CFP/_CBC_CFC - Instead of using device specific ALSA controls to control the two GPIO pins, use GPIOLIB. GPIO pin functions still can be set up in the devicetree as the pins can be used for other functions which are not GPIO related. - Limitation: The GPIO pins can only be used as outputs, even though the hardware alternatively supports using them as inputs. - Handle GPIO configs in a vector rather than separately. - Use more appropriate module_i2c_driver instead of generic module_init/module_exit. - Remove useless initialization data. - As the ADC mute function had no control, add it. - Clean up GPIO initialization which was done both in the I2C probe and in the component probe. - Remove probe, remove, suspend and resume callbacks, which either do not do anything at all anymore, or uselessly just do set_bias_level. - The PLL should not be configured in the devicetree, so move the corresponding definitions to tlv320adc3xxx.h where they can be picked up by a machine driver if needed. - Default to PLL AUTO mode. - Use "reset-gpios" instead of too-generic "gpios" for dt property for configuring codec reset pin. - Use function for printing PLL mode rather than macro. - dev_info is too verbose for most of the printouts, change them to dev_dbg. - Add support for configuring micbias voltage in DT. - Remove ALSA control for mic bias. - Move PLL and NADC, MADC divisor enable to DAPM, including 10 ms post PLL start up delay via POST_PMU event. - ADC is powered on using DAPM, so needless to do it in set_bias_level, thus remove it. - Let DAPM handle BCLK divisor enable. With this, we can remove set_bias_level as it doesn't do anything anymore. - Fix incorrect argument order to a snd_soc_component_update_bits call. - Although the chip itself can operate using BCLK as the clock source, either directly to the divider chain or via the PLL, this is not yet supported in the driver, so require the clocks property for the MCLK for now. - Move PLL rate structure out of .h file where it does not belong. /Ricard Ricard Wanderlof (2): dt-bindings: sound: tlv320adc3xxx: New codec driver ASoC: codec: tlv320adc3xxx: New codec driver .../bindings/sound/ti,tlv320adc3xxx.yaml | 137 ++ include/dt-bindings/sound/tlv320adc3xxx.h | 28 + sound/soc/codecs/Kconfig | 8 + sound/soc/codecs/Makefile | 2 + sound/soc/codecs/tlv320adc3xxx.c | 1311 +++++++++++++++++ 5 files changed, 1486 insertions(+) create mode 100644 Documentation/devicetree/bindings/sound/ti,tlv320adc3xxx.yaml create mode 100644 include/dt-bindings/sound/tlv320adc3xxx.h create mode 100644 sound/soc/codecs/tlv320adc3xxx.c -- 2.20.1 -- Ricard Wolf Wanderlof ricardw(at)axis.com Axis Communications AB, Lund, Sweden www.axis.com Phone +46 46 272 2016 Fax +46 46 13 61 30

1 0

[PATCH v4 00/10] Add support for CS35L41 in HDA systems
by Lucas Tanure 15 Dec '21

15 Dec '21

Add support for laptops that have CS35L41 connected to an HDA codec by I2S and direct I2C connection to the CPU. Laptops that use CS35L41 and are SPI will be added in the future, after the support for it is resolved at i2c-multi-instantiate driver. i2c-multi-instantiate thread: https://lkml.org/lkml/2021/12/10/557 Hardware: - Some laptops have two CS35L41 amplifiers, connected to Realtek ALC287 by an I2S bus and by and direct I2C to the CPU. - The ALC287 codec is connected to the CPU by an HDA bus. - The CS35L41 has a DSP which will require firmware to be loaded. Architecture: - Creation of a library of shared functions for CS35L41 ASoC and HDA - HDA codec driver (RealTek) and CS35L41 HDA driver are combined by using component binding, where it uses device names to find the components and bind to the master - The HDA CS35L41 driver applies pre-defined registers sequences for each action in playback for HDA Sound card Changes from V3: - SPI bus driver starter added - Use separate modules approach instead of liking library - Add CSC3551 ACPI _HID for more I2C laptops - Removed regulators from HDA driver - Add note about Non-conforming _HID V3: https://lkml.org/lkml/2021/11/23/723 Changes from V2: - Not an RFC - Create a new HDA driver for CS35L41 instead of using the ASoC one - Use component binding and device names to find the CS35L41 driver - Create a shared library for ASoC and HDA CS35L41 drivers v2: https://lkml.org/lkml/2021/10/8/344 Lucas Tanure (9): ASoC: cs35l41: Convert tables to shared source code ASoC: cs35l41: Move cs35l41_otp_unpack to shared code ASoC: cs35l41: Move power initializations to reg_sequence ASoC: cs35l41: Create shared function for errata patches ASoC: cs35l41: Create shared function for setting channels ASoC: cs35l41: Create shared function for boost configuration hda: cs35l41: Add support for CS35L41 in HDA systems ACPI / scan: Create platform device for CLSA0100 and CSC3551 ACPI nodes ALSA: hda/realtek: Add support for Legion 7 16ACHg6 laptop Stefan Binding (1): ALSA: hda/realtek: Add CS35L41 support for Thinkpad laptops MAINTAINERS | 2 + drivers/acpi/scan.c | 3 + drivers/platform/x86/i2c-multi-instantiate.c | 11 + include/sound/cs35l41.h | 739 ++++++++++++++++++ sound/pci/hda/Kconfig | 27 + sound/pci/hda/Makefile | 10 + sound/pci/hda/cs35l41_hda.c | 525 +++++++++++++ sound/pci/hda/cs35l41_hda.h | 68 ++ sound/pci/hda/cs35l41_hda_i2c.c | 66 ++ sound/pci/hda/cs35l41_hda_spi.c | 63 ++ sound/pci/hda/hda_component.h | 20 + sound/pci/hda/patch_realtek.c | 147 ++++ sound/soc/codecs/Kconfig | 7 + sound/soc/codecs/Makefile | 4 +- sound/soc/codecs/cs35l41-i2c.c | 1 - .../{cs35l41-tables.c => cs35l41-lib.c} | 353 ++++++++- sound/soc/codecs/cs35l41-spi.c | 1 - sound/soc/codecs/cs35l41.c | 360 +-------- sound/soc/codecs/cs35l41.h | 734 ----------------- 19 files changed, 2049 insertions(+), 1092 deletions(-) create mode 100644 sound/pci/hda/cs35l41_hda.c create mode 100644 sound/pci/hda/cs35l41_hda.h create mode 100644 sound/pci/hda/cs35l41_hda_i2c.c create mode 100644 sound/pci/hda/cs35l41_hda_spi.c create mode 100644 sound/pci/hda/hda_component.h rename sound/soc/codecs/{cs35l41-tables.c => cs35l41-lib.c} (71%) -- 2.34.1

4 13

[PATCH] ALSA: hda/realtek: fix mute/micmute LEDs for a HP ProBook
by Jeremy Szu 15 Dec '21

15 Dec '21

There is a HP ProBook which using ALC236 codec and need the ALC236_FIXUP_HP_MUTE_LED_MICMUTE_VREF quirk to make mute LED and micmute LED work. Signed-off-by: Jeremy Szu <jeremy.szu(a)canonical.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 3599f4c85ebf..e97b3cd1577b 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -8705,6 +8705,7 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK(0x103c, 0x8896, "HP EliteBook 855 G8 Notebook PC", ALC285_FIXUP_HP_MUTE_LED), SND_PCI_QUIRK(0x103c, 0x8898, "HP EliteBook 845 G8 Notebook PC", ALC285_FIXUP_HP_LIMIT_INT_MIC_BOOST), SND_PCI_QUIRK(0x103c, 0x88d0, "HP Pavilion 15-eh1xxx (mainboard 88D0)", ALC287_FIXUP_HP_GPIO_LED), + SND_PCI_QUIRK(0x103c, 0x89ca, "HP", ALC236_FIXUP_HP_MUTE_LED_MICMUTE_VREF), SND_PCI_QUIRK(0x1043, 0x103e, "ASUS X540SA", ALC256_FIXUP_ASUS_MIC), SND_PCI_QUIRK(0x1043, 0x103f, "ASUS TX300", ALC282_FIXUP_ASUS_TX300), SND_PCI_QUIRK(0x1043, 0x106d, "Asus K53BE", ALC269_FIXUP_LIMIT_INT_MIC_BOOST), -- 2.33.0

2 1

Cant't open mixer and no sound , helem (MIXER, 'Front Mic Boots Valume', 0, 1, 0) appears twice or more
by GitHub issues - edited 15 Dec '21

15 Dec '21

alsa-project/alsa-lib issue #205 was edited from WolsYang: **I had try the way in this issue, but not work for me : (** **helem (MIXER,'Master Playback Volume',0,1,0) appears twice or more cannot load mixer controls: Invalid argument**#47 **In default I only can play sound with Back port** **Here is my computer audio port look like.** |codec | ALC888-VD | ALC888-VD | |:-------------|:----------------:|:----------------:| |position | Back | Front | |Function | 0:Line Out, 1:MIC | 0:Line Out, 1:MIC| |Port Num | 2 | 2 | |HW:dev|0:0|0:2| |/sys/class/cound|hwC0D0|hwC0D1| **OS:Ubuntu 20.04.2LTS** **$ uname -r** ``` 5.13.0-051300-generic ``` **$ amixer** ``` ALSA lib simple_none.c:1544:(simple_add1) helem (MIXER,'Front Mic Boots Valume',0,1,0) appears twice or more amixer: Mixer default load error: Invalid aggument ``` **$ aplay -l** ``` **** List of PLAYBACK Hardware Devices **** card 0: PCH [HDA Intel PCH], device 0: ALC888-VD Analog [ALC888-VD Analog] Subdevices: 1/1 Subdevice #0: subdevice #0 card 0: PCH [HDA Intel PCH], device 2: ALC888-VD Analog [ALC888-VD Analog] Subdevices: 1/1 Subdevice #0: subdevice #0 card 0: PCH [HDA Intel PCH], device 3: HDMI 0 [HDMI 0] Subdevices: 1/1 Subdevice #0: subdevice #0 card 0: PCH [HDA Intel PCH], device 7: HDMI 1 [HDMI 1] Subdevices: 1/1 Subdevice #0: subdevice #0 card 0: PCH [HDA Intel PCH], device 8: HDMI 2 [HDMI 2] Subdevices: 1/1 Subdevice #0: subdevice #0 ``` **$ amixer -c 0 controls** ``` numid=18,iface=CARD,name='Front Line Out Front Phantom Jack' numid=16,iface=CARD,name='Front Mic Jack' numid=76,iface=CARD,name='Front Mic Jack',index=1 numid=41,iface=CARD,name='HDMI/DP,pcm=3 Jack' numid=47,iface=CARD,name='HDMI/DP,pcm=7 Jack' numid=53,iface=CARD,name='HDMI/DP,pcm=8 Jack' numid=78,iface=CARD,name='Headphone Front Jack' numid=79,iface=CARD,name='Headphone Surround Jack' numid=19,iface=CARD,name='Line Out Surround Phantom Jack' numid=17,iface=CARD,name='Rear Mic Jack' numid=77,iface=CARD,name='Rear Mic Jack',index=1 numid=15,iface=MIXER,name='Master Playback Switch' numid=14,iface=MIXER,name='Master Playback Volume' numid=64,iface=MIXER,name='Headphone Playback Switch' numid=66,iface=MIXER,name='Headphone Playback Switch',index=1 numid=63,iface=MIXER,name='Headphone Playback Volume' numid=65,iface=MIXER,name='Headphone Playback Volume',index=1 numid=62,iface=MIXER,name='PCM Playback Volume' numid=12,iface=MIXER,name='Front Mic Boost Volume' numid=7,iface=MIXER,name='Front Mic Playback Switch' numid=6,iface=MIXER,name='Front Mic Playback Volume' numid=74,iface=MIXER,name='Front Mic Boost Volume',device=1 numid=69,iface=MIXER,name='Front Mic Playback Switch',device=1 numid=68,iface=MIXER,name='Front Mic Playback Volume',device=1 numid=2,iface=MIXER,name='Front Playback Switch' numid=1,iface=MIXER,name='Front Playback Volume' numid=4,iface=MIXER,name='Surround Playback Switch' numid=3,iface=MIXER,name='Surround Playback Volume' numid=11,iface=MIXER,name='Capture Switch' numid=73,iface=MIXER,name='Capture Switch',device=1 numid=10,iface=MIXER,name='Capture Volume' numid=72,iface=MIXER,name='Capture Volume',device=1 numid=5,iface=MIXER,name='Loopback Mixing' numid=67,iface=MIXER,name='Loopback Mixing',device=1 numid=42,iface=MIXER,name='IEC958 Playback Con Mask' numid=48,iface=MIXER,name='IEC958 Playback Con Mask',index=1 numid=54,iface=MIXER,name='IEC958 Playback Con Mask',index=2 numid=43,iface=MIXER,name='IEC958 Playback Pro Mask' numid=49,iface=MIXER,name='IEC958 Playback Pro Mask',index=1 numid=55,iface=MIXER,name='IEC958 Playback Pro Mask',index=2 numid=44,iface=MIXER,name='IEC958 Playback Default' numid=50,iface=MIXER,name='IEC958 Playback Default',index=1 numid=56,iface=MIXER,name='IEC958 Playback Default',index=2 numid=45,iface=MIXER,name='IEC958 Playback Switch' numid=51,iface=MIXER,name='IEC958 Playback Switch',index=1 numid=57,iface=MIXER,name='IEC958 Playback Switch',index=2 numid=13,iface=MIXER,name='Rear Mic Boost Volume' numid=9,iface=MIXER,name='Rear Mic Playback Switch' numid=8,iface=MIXER,name='Rear Mic Playback Volume' numid=75,iface=MIXER,name='Rear Mic Boost Volume',device=1 numid=71,iface=MIXER,name='Rear Mic Playback Switch',device=1 numid=70,iface=MIXER,name='Rear Mic Playback Volume',device=1 numid=21,iface=PCM,name='Capture Channel Map' numid=20,iface=PCM,name='Playback Channel Map' numid=81,iface=PCM,name='Capture Channel Map',device=2 numid=80,iface=PCM,name='Playback Channel Map',device=2 numid=46,iface=PCM,name='ELD',device=3 numid=59,iface=PCM,name='Playback Channel Map',device=3 numid=52,iface=PCM,name='ELD',device=7 numid=60,iface=PCM,name='Playback Channel Map',device=7 numid=58,iface=PCM,name='ELD',device=8 numid=61,iface=PCM,name='Playback Channel Map',device=8 ``` **$ pactl list modules** ``` Card #0 Name: alsa_card.pci-0000_00_1b.0 Driver: module-alsa-card.c Owner Module: 7 Properties: alsa.card = "0" alsa.card_name = "HDA Intel PCH" alsa.long_card_name = "HDA Intel PCH at 0x81410000 irq 132" alsa.driver_name = "snd_hda_intel" device.bus_path = "pci-0000:00:1b.0" sysfs.path = "/devices/pci0000:00/0000:00:1b.0/sound/card0" device.bus = "pci" device.vendor.id = "8086" device.vendor.name = "Intel Corporation" device.product.id = "2284" device.product.name = "Atom/Celeron/Pentium Processor x5-E8000/J3xxx/N3xxx Series High Definition Audio Controller" device.form_factor = "internal" device.string = "0" device.description = "Built-in Audio" module-udev-detect.discovered = "1" device.icon_name = "audio-card-pci" Profiles: input:analog-stereo: Analog Stereo Input (sinks: 0, sources: 1, priority: 65, available: yes) output:analog-stereo: Analog Stereo Output (sinks: 1, sources: 0, priority: 6500, available: yes) output:analog-stereo+input:analog-stereo: Analog Stereo Duplex (sinks: 1, sources: 1, priority: 6565, available: yes) output:analog-surround-40: Analog Surround 4.0 Output (sinks: 1, sources: 0, priority: 1200, available: yes) output:analog-surround-40+input:analog-stereo: Analog Surround 4.0 Output + Analog Stereo Input (sinks: 1, sources: 1, priority: 1265, available: yes) output:hdmi-stereo: Digital Stereo (HDMI) Output (sinks: 1, sources: 0, priority: 5900, available: yes) output:hdmi-stereo+input:analog-stereo: Digital Stereo (HDMI) Output + Analog Stereo Input (sinks: 1, sources: 1, priority: 5965, available: yes) output:hdmi-surround: Digital Surround 5.1 (HDMI) Output (sinks: 1, sources: 0, priority: 800, available: yes) output:hdmi-surround+input:analog-stereo: Digital Surround 5.1 (HDMI) Output + Analog Stereo Input (sinks: 1, sources: 1, priority: 865, available: yes) output:hdmi-surround71: Digital Surround 7.1 (HDMI) Output (sinks: 1, sources: 0, priority: 800, available: yes) output:hdmi-surround71+input:analog-stereo: Digital Surround 7.1 (HDMI) Output + Analog Stereo Input (sinks: 1, sources: 1, priority: 865, available: yes) output:hdmi-stereo-extra1: Digital Stereo (HDMI 2) Output (sinks: 1, sources: 0, priority: 5700, available: yes) output:hdmi-stereo-extra1+input:analog-stereo: Digital Stereo (HDMI 2) Output + Analog Stereo Input (sinks: 1, sources: 1, priority: 5765, available: yes) output:hdmi-surround-extra1: Digital Surround 5.1 (HDMI 2) Output (sinks: 1, sources: 0, priority: 600, available: yes) output:hdmi-surround-extra1+input:analog-stereo: Digital Surround 5.1 (HDMI 2) Output + Analog Stereo Input (sinks: 1, sources: 1, priority: 665, available: yes) output:hdmi-surround71-extra1: Digital Surround 7.1 (HDMI 2) Output (sinks: 1, sources: 0, priority: 600, available: yes) output:hdmi-surround71-extra1+input:analog-stereo: Digital Surround 7.1 (HDMI 2) Output + Analog Stereo Input (sinks: 1, sources: 1, priority: 665, available: yes) output:hdmi-stereo-extra2: Digital Stereo (HDMI 3) Output (sinks: 1, sources: 0, priority: 5700, available: yes) output:hdmi-stereo-extra2+input:analog-stereo: Digital Stereo (HDMI 3) Output + Analog Stereo Input (sinks: 1, sources: 1, priority: 5765, available: yes) off: Off (sinks: 0, sources: 0, priority: 0, available: yes) Active Profile: output:analog-stereo ``` Issue URL : https://github.com/alsa-project/alsa-lib/issues/205 Repository URL: https://github.com/alsa-project/alsa-lib

1 0

Cant't open mixer and no sound , helem (MIXER, 'Front Mic Boots Valume', 0, 1, 0) appears twice or more
by GitHub issues - opened 15 Dec '21

15 Dec '21

alsa-project/alsa-lib issue #205 was opened from WolsYang: **I had try the way in this issue, but not work for me : (** **helem (MIXER,'Master Playback Volume',0,1,0) appears twice or more cannot load mixer controls: Invalid argument**#47 **In default I only can play sound with Back port** **Here is my computer audio port look like.** |codec | ALC888-VD | ALC888-VD | |:-------------|:----------------:|:----------------:| |position | Back | Front | |Function | 0:Line Out, 1:MIC | 0:Line Out, 1:MIC| |Port Num | 2 | 2 | |HW:dev|0:0|0:2| |/sys/class/cound|hwC0D0|hwC0D1| **OS:Ubuntu 20.04.2LTS** **$ uname -r** ``` 5.13.0-051300-generic ``` **$ amixer** ``` ALSA lib simple_none.c:1544:(simple_add1) helem (MIXER,'Front Mic Boots Valume',0,1,0) appears twice or more amixer: Mixer default load error: Invalid aggument ``` **$ aplay -l** ``` **** List of PLAYBACK Hardware Devices **** card 0: PCH [HDA Intel PCH], device 0: ALC888-VD Analog [ALC888-VD Analog] Subdevices: 1/1 Subdevice #0: subdevice #0 card 0: PCH [HDA Intel PCH], device 2: ALC888-VD Analog [ALC888-VD Analog] Subdevices: 1/1 Subdevice #0: subdevice #0 card 0: PCH [HDA Intel PCH], device 3: HDMI 0 [HDMI 0] Subdevices: 1/1 Subdevice #0: subdevice #0 card 0: PCH [HDA Intel PCH], device 7: HDMI 1 [HDMI 1] Subdevices: 1/1 Subdevice #0: subdevice #0 card 0: PCH [HDA Intel PCH], device 8: HDMI 2 [HDMI 2] Subdevices: 1/1 Subdevice #0: subdevice #0 ``` **$ pactl list modules** ``` Card #0 Name: alsa_card.pci-0000_00_1b.0 Driver: module-alsa-card.c Owner Module: 7 Properties: alsa.card = "0" alsa.card_name = "HDA Intel PCH" alsa.long_card_name = "HDA Intel PCH at 0x81410000 irq 132" alsa.driver_name = "snd_hda_intel" device.bus_path = "pci-0000:00:1b.0" sysfs.path = "/devices/pci0000:00/0000:00:1b.0/sound/card0" device.bus = "pci" device.vendor.id = "8086" device.vendor.name = "Intel Corporation" device.product.id = "2284" device.product.name = "Atom/Celeron/Pentium Processor x5-E8000/J3xxx/N3xxx Series High Definition Audio Controller" device.form_factor = "internal" device.string = "0" device.description = "Built-in Audio" module-udev-detect.discovered = "1" device.icon_name = "audio-card-pci" Profiles: input:analog-stereo: Analog Stereo Input (sinks: 0, sources: 1, priority: 65, available: yes) output:analog-stereo: Analog Stereo Output (sinks: 1, sources: 0, priority: 6500, available: yes) output:analog-stereo+input:analog-stereo: Analog Stereo Duplex (sinks: 1, sources: 1, priority: 6565, available: yes) output:analog-surround-40: Analog Surround 4.0 Output (sinks: 1, sources: 0, priority: 1200, available: yes) output:analog-surround-40+input:analog-stereo: Analog Surround 4.0 Output + Analog Stereo Input (sinks: 1, sources: 1, priority: 1265, available: yes) output:hdmi-stereo: Digital Stereo (HDMI) Output (sinks: 1, sources: 0, priority: 5900, available: yes) output:hdmi-stereo+input:analog-stereo: Digital Stereo (HDMI) Output + Analog Stereo Input (sinks: 1, sources: 1, priority: 5965, available: yes) output:hdmi-surround: Digital Surround 5.1 (HDMI) Output (sinks: 1, sources: 0, priority: 800, available: yes) output:hdmi-surround+input:analog-stereo: Digital Surround 5.1 (HDMI) Output + Analog Stereo Input (sinks: 1, sources: 1, priority: 865, available: yes) output:hdmi-surround71: Digital Surround 7.1 (HDMI) Output (sinks: 1, sources: 0, priority: 800, available: yes) output:hdmi-surround71+input:analog-stereo: Digital Surround 7.1 (HDMI) Output + Analog Stereo Input (sinks: 1, sources: 1, priority: 865, available: yes) output:hdmi-stereo-extra1: Digital Stereo (HDMI 2) Output (sinks: 1, sources: 0, priority: 5700, available: yes) output:hdmi-stereo-extra1+input:analog-stereo: Digital Stereo (HDMI 2) Output + Analog Stereo Input (sinks: 1, sources: 1, priority: 5765, available: yes) output:hdmi-surround-extra1: Digital Surround 5.1 (HDMI 2) Output (sinks: 1, sources: 0, priority: 600, available: yes) output:hdmi-surround-extra1+input:analog-stereo: Digital Surround 5.1 (HDMI 2) Output + Analog Stereo Input (sinks: 1, sources: 1, priority: 665, available: yes) output:hdmi-surround71-extra1: Digital Surround 7.1 (HDMI 2) Output (sinks: 1, sources: 0, priority: 600, available: yes) output:hdmi-surround71-extra1+input:analog-stereo: Digital Surround 7.1 (HDMI 2) Output + Analog Stereo Input (sinks: 1, sources: 1, priority: 665, available: yes) output:hdmi-stereo-extra2: Digital Stereo (HDMI 3) Output (sinks: 1, sources: 0, priority: 5700, available: yes) output:hdmi-stereo-extra2+input:analog-stereo: Digital Stereo (HDMI 3) Output + Analog Stereo Input (sinks: 1, sources: 1, priority: 5765, available: yes) off: Off (sinks: 0, sources: 0, priority: 0, available: yes) Active Profile: output:analog-stereo ``` Issue URL : https://github.com/alsa-project/alsa-lib/issues/205 Repository URL: https://github.com/alsa-project/alsa-lib

1 0

[PATCH] ASoC: qdsp6: Fix an IS_ERR() vs NULL bug
by Miaoqian Lin 15 Dec '21

15 Dec '21

The function gpr_alloc_port return ERR_PTR on errors, it doesn't return null. Signed-off-by: Miaoqian Lin <linmq006(a)gmail.com> --- sound/soc/qcom/qdsp6/q6apm.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/soc/qcom/qdsp6/q6apm.c b/sound/soc/qcom/qdsp6/q6apm.c index 13598ef5bacb..3e007d609a9b 100644 --- a/sound/soc/qcom/qdsp6/q6apm.c +++ b/sound/soc/qcom/qdsp6/q6apm.c @@ -630,9 +630,9 @@ struct q6apm_graph *q6apm_graph_open(struct device *dev, q6apm_cb cb, init_waitqueue_head(&graph->cmd_wait); graph->port = gpr_alloc_port(apm->gdev, dev, graph_callback, graph); - if (!graph->port) { + if (IS_ERR(graph->port)) { kfree(graph); - ret = -ENOMEM; + ret = PTR_ERR(graph->port); goto err; } -- 2.17.1

2 1

[v3] SoC: qcom: Distinguish headset codec by codec_dai->name
by Judy Hsiao 15 Dec '21

15 Dec '21

Distinguish which headset codec is on the board by codec_dai->name instead of card->name. It fixes the crash of being unable to handle kernel paging requests at virtual address ADDR by initializing the correct audio codec on the board. Call stack of the crash: ``` Unable to handle kernel paging request at virtual address ... ... Call trace: rt5682_set_component_pll+0xcc/0xb78 [snd_soc_rt5682] snd_soc_component_set_pll+0x90/0x154 snd_soc_dai_set_pll+0xf4/0x1ac sc7180_snd_startup+0x268/0x3c0 [snd_soc_sc7180] snd_soc_link_startup+0xa4/0x180 soc_pcm_open+0x35c/0x15c8 snd_pcm_open_substream+0xa90/0x13b0 snd_pcm_open+0x1a4/0x55c snd_pcm_capture_open+0x7c/0xe8 snd_open+0x2b8/0x2e4 chrdev_open+0x364/0x3d4 do_dentry_open+0x66c/0xc58 vfs_open+0x7c/0x8c path_openat+0x108c/0x2bbc do_filp_open+0x15c/0x258 do_sys_open+0x278/0x62c __arm64_compat_sys_openat+0x9c/0xb0 ... ``` Fixes: 425c5fce8a03 ("ASoC: qcom: Add support for ALC5682I-VS codec") Signed-off-by: Judy Hsiao <judyhsiao(a)chromium.org> --- sound/soc/qcom/sc7180.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/soc/qcom/sc7180.c b/sound/soc/qcom/sc7180.c index 2fff764a00a7..37225ef2563a 100644 --- a/sound/soc/qcom/sc7180.c +++ b/sound/soc/qcom/sc7180.c @@ -131,13 +131,13 @@ static int sc7180_snd_startup(struct snd_pcm_substream *substream) struct snd_soc_dai *codec_dai = asoc_rtd_to_codec(rtd, 0); int pll_id, pll_source, pll_in, pll_out, clk_id, ret; - if (!(strcmp(card->name, "sc7180-rt5682-max98357a-1mic"))) { + if (!strcmp(codec_dai->name, "rt5682-aif1")) { pll_source = RT5682_PLL1_S_MCLK; pll_id = 0; clk_id = RT5682_SCLK_S_PLL1; pll_out = RT5682_PLL1_FREQ; pll_in = DEFAULT_MCLK_RATE; - } else if (!(strcmp(card->name, "sc7180-rt5682s-max98357a-1mic"))) { + } else if (!strcmp(codec_dai->name, "rt5682s-aif1")) { pll_source = RT5682S_PLL_S_MCLK; pll_id = RT5682S_PLL2; clk_id = RT5682S_SCLK_S_PLL2; -- 2.31.0

2 2

[PATCH] ASoC: rt5682: fix the wrong jack type detected
by derek.fang＠realtek.com 15 Dec '21

15 Dec '21

From: Derek Fang <derek.fang(a)realtek.com> Some powers were changed during the jack insert detection and clk's enable/disable in CCF. If in parallel, the influence has a chance to detect the wrong jack type, so add a lock. Signed-off-by: Derek Fang <derek.fang(a)realtek.com> --- sound/soc/codecs/rt5682.c | 4 ++++ 1 file changed, 4 insertions(+) diff --git a/sound/soc/codecs/rt5682.c b/sound/soc/codecs/rt5682.c index 5224123d0d3b..b34a8542077d 100644 --- a/sound/soc/codecs/rt5682.c +++ b/sound/soc/codecs/rt5682.c @@ -929,6 +929,8 @@ int rt5682_headset_detect(struct snd_soc_component *component, int jack_insert) unsigned int val, count; if (jack_insert) { + snd_soc_dapm_mutex_lock(dapm); + snd_soc_component_update_bits(component, RT5682_PWR_ANLG_1, RT5682_PWR_VREF2 | RT5682_PWR_MB, RT5682_PWR_VREF2 | RT5682_PWR_MB); @@ -979,6 +981,8 @@ int rt5682_headset_detect(struct snd_soc_component *component, int jack_insert) snd_soc_component_update_bits(component, RT5682_MICBIAS_2, RT5682_PWR_CLK25M_MASK | RT5682_PWR_CLK1M_MASK, RT5682_PWR_CLK25M_PU | RT5682_PWR_CLK1M_PU); + + snd_soc_dapm_mutex_unlock(dapm); } else { rt5682_enable_push_button_irq(component, false); snd_soc_component_update_bits(component, RT5682_CBJ_CTRL_1, -- 2.17.1

2 1