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[PATCH AUTOSEL 5.12 01/43] ASoC: max98088: fix ni clock divider calculation
by Sasha Levin 03 Jun '21

03 Jun '21

From: Marco Felsch <m.felsch(a)pengutronix.de> [ Upstream commit 6c9762a78c325107dc37d20ee21002b841679209 ] The ni1/ni2 ratio formula [1] uses the pclk which is the prescaled mclk. The max98088 datasheet [2] has no such formula but table-12 equals so we can assume that it is the same for both devices. While on it make use of DIV_ROUND_CLOSEST_ULL(). [1] https://datasheets.maximintegrated.com/en/ds/MAX98089.pdf; page 86 [2] https://datasheets.maximintegrated.com/en/ds/MAX98088.pdf; page 82 Signed-off-by: Marco Felsch <m.felsch(a)pengutronix.de> Link: https://lore.kernel.org/r/20210423135402.32105-1-m.felsch@pengutronix.de Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/codecs/max98088.c | 13 +++++++++++-- 1 file changed, 11 insertions(+), 2 deletions(-) diff --git a/sound/soc/codecs/max98088.c b/sound/soc/codecs/max98088.c index 4be24e7f51c8..f8e49e45ce33 100644 --- a/sound/soc/codecs/max98088.c +++ b/sound/soc/codecs/max98088.c @@ -41,6 +41,7 @@ struct max98088_priv { enum max98088_type devtype; struct max98088_pdata *pdata; struct clk *mclk; + unsigned char mclk_prescaler; unsigned int sysclk; struct max98088_cdata dai[2]; int eq_textcnt; @@ -998,13 +999,16 @@ static int max98088_dai1_hw_params(struct snd_pcm_substream *substream, /* Configure NI when operating as master */ if (snd_soc_component_read(component, M98088_REG_14_DAI1_FORMAT) & M98088_DAI_MAS) { + unsigned long pclk; + if (max98088->sysclk == 0) { dev_err(component->dev, "Invalid system clock frequency\n"); return -EINVAL; } ni = 65536ULL * (rate < 50000 ? 96ULL : 48ULL) * (unsigned long long int)rate; - do_div(ni, (unsigned long long int)max98088->sysclk); + pclk = DIV_ROUND_CLOSEST(max98088->sysclk, max98088->mclk_prescaler); + ni = DIV_ROUND_CLOSEST_ULL(ni, pclk); snd_soc_component_write(component, M98088_REG_12_DAI1_CLKCFG_HI, (ni >> 8) & 0x7F); snd_soc_component_write(component, M98088_REG_13_DAI1_CLKCFG_LO, @@ -1065,13 +1069,16 @@ static int max98088_dai2_hw_params(struct snd_pcm_substream *substream, /* Configure NI when operating as master */ if (snd_soc_component_read(component, M98088_REG_1C_DAI2_FORMAT) & M98088_DAI_MAS) { + unsigned long pclk; + if (max98088->sysclk == 0) { dev_err(component->dev, "Invalid system clock frequency\n"); return -EINVAL; } ni = 65536ULL * (rate < 50000 ? 96ULL : 48ULL) * (unsigned long long int)rate; - do_div(ni, (unsigned long long int)max98088->sysclk); + pclk = DIV_ROUND_CLOSEST(max98088->sysclk, max98088->mclk_prescaler); + ni = DIV_ROUND_CLOSEST_ULL(ni, pclk); snd_soc_component_write(component, M98088_REG_1A_DAI2_CLKCFG_HI, (ni >> 8) & 0x7F); snd_soc_component_write(component, M98088_REG_1B_DAI2_CLKCFG_LO, @@ -1113,8 +1120,10 @@ static int max98088_dai_set_sysclk(struct snd_soc_dai *dai, */ if ((freq >= 10000000) && (freq < 20000000)) { snd_soc_component_write(component, M98088_REG_10_SYS_CLK, 0x10); + max98088->mclk_prescaler = 1; } else if ((freq >= 20000000) && (freq < 30000000)) { snd_soc_component_write(component, M98088_REG_10_SYS_CLK, 0x20); + max98088->mclk_prescaler = 2; } else { dev_err(component->dev, "Invalid master clock frequency\n"); return -EINVAL; -- 2.30.2

1 6

Allen&Heath Xone43C: USB communication
by Roman 03 Jun '21

03 Jun '21

Hello all, hello list, I am not sure whether this is the correct list, since it's not strictly an ALSA affair (fw to kernel-devel). Recent progress on driver support for Pioneer devices encouraged me to dissect USB communication with an Allen&Heath Xone43C, a mixer with USB interface. It does feature 16 channels (8 in, 8 out), but only 4 (2 in, 2 out) show up, 2 of which remain automagically silent. There is no linux driver yet, and maybe I can help? Addresses are: 0x80 Endpoint 0 (in) 0x00 Endpoint 0 (out) 0x81 Endpoint 1 (in) Interface 2.1 and Interface 2.2 (vendor spec) 0x02 Endpoint 2 (out) Interface 1.1 and Interface 1.2 (vendor spec) 0x85 Endpoint 5 (in) Interface 1.1 0x83 Endpoint 3 (in) Interface 3 MIDI 0x04 Endpoint 4 (out) Interface 3 MIDI ---------------------------------- I have a sniff here: https://drive.google.com/file/d/1UPXCN-r7jc9hfm1d-Hip8vckSs2RQNtC/view? usp=sharing --------------------------------- USB info: Bus 001 Device 006: ID 22f0:000f Allen&Heath Xone:43C Device Descriptor: bLength 18 bDescriptorType 1 bcdUSB 2.00 bDeviceClass 239 Miscellaneous Device bDeviceSubClass 2 bDeviceProtocol 1 Interface Association bMaxPacketSize0 64 idVendor 0x22f0 idProduct 0x000f bcdDevice 2.00 iManufacturer 1 Allen&Heath iProduct 2 Xone:43C iSerial 3 no serial number bNumConfigurations 1 Configuration Descriptor: bLength 9 bDescriptorType 2 wTotalLength 0x0126 bNumInterfaces 4 bConfigurationValue 1 iConfiguration 0 bmAttributes 0xc0 Self Powered MaxPower 0mA Interface Descriptor: bLength 9 bDescriptorType 4 bInterfaceNumber 0 bAlternateSetting 0 bNumEndpoints 0 bInterfaceClass 1 Audio bInterfaceSubClass 1 Control Device bInterfaceProtocol 0 iInterface 0 AudioControl Interface Descriptor: bLength 11 bDescriptorType 36 bDescriptorSubtype 1 (HEADER) bcdADC 1.00 wTotalLength 0x0035 bInCollection 3 baInterfaceNr(0) 1 baInterfaceNr(1) 2 baInterfaceNr(2) 3 AudioControl Interface Descriptor: bLength 12 bDescriptorType 36 bDescriptorSubtype 2 (INPUT_TERMINAL) bTerminalID 1 wTerminalType 0x0101 USB Streaming bAssocTerminal 0 bNrChannels 2 wChannelConfig 0x0003 Left Front (L) Right Front (R) iChannelNames 0 iTerminal 0 AudioControl Interface Descriptor: bLength 9 bDescriptorType 36 bDescriptorSubtype 3 (OUTPUT_TERMINAL) bTerminalID 2 wTerminalType 0x0602 Digital Audio Interface bAssocTerminal 0 bSourceID 1 iTerminal 0 AudioControl Interface Descriptor: bLength 12 bDescriptorType 36 bDescriptorSubtype 2 (INPUT_TERMINAL) bTerminalID 3 wTerminalType 0x0602 Digital Audio Interface bAssocTerminal 0 bNrChannels 2 wChannelConfig 0x0003 Left Front (L) Right Front (R) iChannelNames 0 iTerminal 0 AudioControl Interface Descriptor: bLength 9 bDescriptorType 36 bDescriptorSubtype 3 (OUTPUT_TERMINAL) bTerminalID 4 wTerminalType 0x0101 USB Streaming bAssocTerminal 0 bSourceID 3 iTerminal 0 Interface Descriptor: bLength 9 bDescriptorType 4 bInterfaceNumber 1 bAlternateSetting 0 bNumEndpoints 0 bInterfaceClass 1 Audio bInterfaceSubClass 2 Streaming bInterfaceProtocol 0 iInterface 0 Interface Descriptor: bLength 9 bDescriptorType 4 bInterfaceNumber 1 bAlternateSetting 1 bNumEndpoints 2 bInterfaceClass 1 Audio bInterfaceSubClass 2 Streaming bInterfaceProtocol 0 iInterface 0 AudioStreaming Interface Descriptor: bLength 7 bDescriptorType 36 bDescriptorSubtype 1 (AS_GENERAL) bTerminalLink 1 bDelay 1 frames wFormatTag 0x0001 PCM AudioStreaming Interface Descriptor: bLength 11 bDescriptorType 36 bDescriptorSubtype 2 (FORMAT_TYPE) bFormatType 1 (FORMAT_TYPE_I) bNrChannels 2 bSubframeSize 3 bBitResolution 24 bSamFreqType 1 Discrete tSamFreq[ 0] 48000 Endpoint Descriptor: bLength 9 bDescriptorType 5 bEndpointAddress 0x02 EP 2 OUT bmAttributes 5 Transfer Type Isochronous Synch Type Asynchronous Usage Type Data wMaxPacketSize 0x0126 1x 294 bytes bInterval 4 bRefresh 0 bSynchAddress 133 AudioStreaming Endpoint Descriptor: bLength 7 bDescriptorType 37 bDescriptorSubtype 1 (EP_GENERAL) bmAttributes 0x01 Sampling Frequency bLockDelayUnits 2 Decoded PCM samples wLockDelay 0x0000 Endpoint Descriptor: bLength 9 bDescriptorType 5 bEndpointAddress 0x85 EP 5 IN bmAttributes 1 Transfer Type Isochronous Synch Type None Usage Type Data wMaxPacketSize 0x0003 1x 3 bytes bInterval 4 bRefresh 5 bSynchAddress 0 Interface Descriptor: bLength 9 bDescriptorType 4 bInterfaceNumber 1 bAlternateSetting 2 bNumEndpoints 1 bInterfaceClass 255 Vendor Specific Class bInterfaceSubClass 0 bInterfaceProtocol 0 iInterface 0 Endpoint Descriptor: bLength 9 bDescriptorType 5 bEndpointAddress 0x02 EP 2 OUT bmAttributes 5 Transfer Type Isochronous Synch Type Asynchronous Usage Type Data wMaxPacketSize 0x0200 1x 512 bytes bInterval 1 bRefresh 0 bSynchAddress 0 Interface Descriptor: bLength 9 bDescriptorType 4 bInterfaceNumber 2 bAlternateSetting 0 bNumEndpoints 0 bInterfaceClass 1 Audio bInterfaceSubClass 2 Streaming bInterfaceProtocol 0 iInterface 0 Interface Descriptor: bLength 9 bDescriptorType 4 bInterfaceNumber 2 bAlternateSetting 1 bNumEndpoints 1 bInterfaceClass 1 Audio bInterfaceSubClass 2 Streaming bInterfaceProtocol 0 iInterface 0 AudioStreaming Interface Descriptor: bLength 7 bDescriptorType 36 bDescriptorSubtype 1 (AS_GENERAL) bTerminalLink 4 bDelay 1 frames wFormatTag 0x0001 PCM AudioStreaming Interface Descriptor: bLength 11 bDescriptorType 36 bDescriptorSubtype 2 (FORMAT_TYPE) bFormatType 1 (FORMAT_TYPE_I) bNrChannels 2 bSubframeSize 3 bBitResolution 24 bSamFreqType 1 Discrete tSamFreq[ 0] 48000 Endpoint Descriptor: bLength 9 bDescriptorType 5 bEndpointAddress 0x81 EP 1 IN bmAttributes 5 Transfer Type Isochronous Synch Type Asynchronous Usage Type Data wMaxPacketSize 0x0126 1x 294 bytes bInterval 4 bRefresh 0 bSynchAddress 0 AudioStreaming Endpoint Descriptor: bLength 7 bDescriptorType 37 bDescriptorSubtype 1 (EP_GENERAL) bmAttributes 0x01 Sampling Frequency bLockDelayUnits 2 Decoded PCM samples wLockDelay 0x0000 Interface Descriptor: bLength 9 bDescriptorType 4 bInterfaceNumber 2 bAlternateSetting 2 bNumEndpoints 1 bInterfaceClass 255 Vendor Specific Class bInterfaceSubClass 0 bInterfaceProtocol 0 iInterface 0 Endpoint Descriptor: bLength 9 bDescriptorType 5 bEndpointAddress 0x81 EP 1 IN bmAttributes 5 Transfer Type Isochronous Synch Type Asynchronous Usage Type Data wMaxPacketSize 0x0200 1x 512 bytes bInterval 1 bRefresh 0 bSynchAddress 0 Interface Descriptor: bLength 9 bDescriptorType 4 bInterfaceNumber 3 bAlternateSetting 0 bNumEndpoints 2 bInterfaceClass 1 Audio bInterfaceSubClass 3 MIDI Streaming bInterfaceProtocol 0 iInterface 0 MIDIStreaming Interface Descriptor: bLength 7 bDescriptorType 36 bDescriptorSubtype 1 (HEADER) bcdADC 1.00 wTotalLength 0x0041 MIDIStreaming Interface Descriptor: bLength 6 bDescriptorType 36 bDescriptorSubtype 2 (MIDI_IN_JACK) bJackType 1 Embedded bJackID 1 iJack 0 MIDIStreaming Interface Descriptor: bLength 6 bDescriptorType 36 bDescriptorSubtype 2 (MIDI_IN_JACK) bJackType 2 External bJackID 2 iJack 0 MIDIStreaming Interface Descriptor: bLength 9 bDescriptorType 36 bDescriptorSubtype 3 (MIDI_OUT_JACK) bJackType 1 Embedded bJackID 3 bNrInputPins 1 baSourceID( 0) 2 BaSourcePin( 0) 1 iJack 0 MIDIStreaming Interface Descriptor: bLength 9 bDescriptorType 36 bDescriptorSubtype 3 (MIDI_OUT_JACK) bJackType 2 External bJackID 4 bNrInputPins 1 baSourceID( 0) 1 BaSourcePin( 0) 1 iJack 0 Endpoint Descriptor: bLength 9 bDescriptorType 5 bEndpointAddress 0x83 EP 3 IN bmAttributes 2 Transfer Type Bulk Synch Type None Usage Type Data wMaxPacketSize 0x0200 1x 512 bytes bInterval 1 bRefresh 0 bSynchAddress 0 MIDIStreaming Endpoint Descriptor: bLength 5 bDescriptorType 37 bDescriptorSubtype 1 (GENERAL) bNumEmbMIDIJack 1 baAssocJackID( 0) 3 Endpoint Descriptor: bLength 9 bDescriptorType 5 bEndpointAddress 0x04 EP 4 OUT bmAttributes 3 Transfer Type Interrupt Synch Type None Usage Type Data wMaxPacketSize 0x0010 1x 16 bytes bInterval 4 bRefresh 0 bSynchAddress 0 MIDIStreaming Endpoint Descriptor: bLength 5 bDescriptorType 37 bDescriptorSubtype 1 (GENERAL) bNumEmbMIDIJack 1 baAssocJackID( 0) 1 Device Qualifier (for other device speed): bLength 10 bDescriptorType 6 bcdUSB 2.00 bDeviceClass 0 bDeviceSubClass 0 bDeviceProtocol 0 bMaxPacketSize0 64 bNumConfigurations 1 Device Status: 0x0001 Self Powered --- Best wishes, Roman

2 1

[PATCH] ASoC: SOF: Intel: Remove duplicate include of shim.h
by Wan Jiabing 03 Jun '21

03 Jun '21

shim.h is included twice. As shim.h is not related to later header file, keep one which has comment and remove the former one. Signed-off-by: Wan Jiabing <wanjiabing(a)vivo.com> --- sound/soc/sof/intel/pci-tng.c | 1 - 1 file changed, 1 deletion(-) diff --git a/sound/soc/sof/intel/pci-tng.c b/sound/soc/sof/intel/pci-tng.c index 4ee1da3..4bded66 100644 --- a/sound/soc/sof/intel/pci-tng.c +++ b/sound/soc/sof/intel/pci-tng.c @@ -15,7 +15,6 @@ #include <sound/sof.h> #include "../ops.h" #include "atom.h" -#include "shim.h" #include "../sof-pci-dev.h" #include "../sof-audio.h" -- 2.7.4

2 1

[PATCH v2 0/4] ASoC: codecs: wcd934x: add Headset and button detection support
by Srinivas Kandagatla 03 Jun '21

03 Jun '21

This patchset adds support to MBHC(Multi Button Headset Control) block found in Qualcomm WCD codecs. MBHC support headset type detection, both Mechanical and electrical insert/removal detection along with 8 buttons detection, Over current interrupts on HPHL/R, Impedance Measurements on HPHL/R. Eventhough MBHC block supports things like OverCurrent detection, Currently its reported as a kernel debug message. Should this be reported as an uevent to userspace? like the way USB reports? Any suggestions? First patch adds a common mbhc driver and the second one wcd934x specific driver changes along with sdm845 soundcard related changes. Common wcd-mbhc-v2 driver should be reusable across multiple codecs like WCD9335, WCD934x, WCD937x and WCD938x. Most of the work is derived from downstream Qualcomm kernels. Credits to various Qualcomm authors from Patrick Lai's team who have contributed to this code. This patchset thanks, srini Changes since v1: - updated buttons-vthreshold bindings from review comments for wcd938x Srinivas Kandagatla (4): ASoC: dt-bindings: wcd934x: add bindings for Headset Button detection ASoC: codecs: wcd: add multi button Headset detection support ASoC: codecs: wcd934x: add mbhc support ASoC: qcom: sdm845: add jack support for WCD934x .../bindings/sound/qcom,wcd934x.yaml | 30 + include/linux/mfd/wcd934x/registers.h | 57 + sound/soc/codecs/Kconfig | 4 + sound/soc/codecs/Makefile | 2 + sound/soc/codecs/wcd-mbhc-v2.c | 1468 +++++++++++++++++ sound/soc/codecs/wcd-mbhc-v2.h | 340 ++++ sound/soc/codecs/wcd934x.c | 884 +++++++++- sound/soc/qcom/sdm845.c | 8 + 8 files changed, 2778 insertions(+), 15 deletions(-) create mode 100644 sound/soc/codecs/wcd-mbhc-v2.c create mode 100644 sound/soc/codecs/wcd-mbhc-v2.h -- 2.21.0

3 7

[PATCH][next] ASoC: rsnd: check for zero node count
by Colin King 03 Jun '21

03 Jun '21

From: Colin Ian King <colin.king(a)canonical.com> The call to rsnd_node_count can potentially return a zero node count so add a check for this corner case. (Note that the two other calls to rsnd_node_count in the kernel perform this check, so I think it justifies adding this). This avoids using a zero nr in a devm_kcalloc call. Addresses-Coverity: ("Unchecked return value") Fixes: c413983eb66a ("ASoC: rsnd: adjust disabled module") Signed-off-by: Colin Ian King <colin.king(a)canonical.com> --- sound/soc/sh/rcar/ssiu.c | 3 +++ 1 file changed, 3 insertions(+) diff --git a/sound/soc/sh/rcar/ssiu.c b/sound/soc/sh/rcar/ssiu.c index 5682c74bb7ff..0d8f97633dd2 100644 --- a/sound/soc/sh/rcar/ssiu.c +++ b/sound/soc/sh/rcar/ssiu.c @@ -515,6 +515,9 @@ int rsnd_ssiu_probe(struct rsnd_priv *priv) else nr = priv->ssi_nr; + if (!nr) + return -EINVAL; + ssiu = devm_kcalloc(dev, nr, sizeof(*ssiu), GFP_KERNEL); if (!ssiu) return -ENOMEM; -- 2.31.1

4 3

[PATCH v12 0/4] ASoC: codecs: add rk817 support
by Chris Morgan 03 Jun '21

03 Jun '21

From: Chris Morgan <macromorgan(a)hotmail.com> This series is to add support for the Rockchip rk817 audio codec which is present in all Rockchip rk817 MFD chips. Changes in v12: - Updated codec cell in MFD driver to a single line. Changes in v11: - Stopped checking for presence of parent node in codec driver, as driver should never be called if parent device doesn't exist to call it. - Made codec subnode optional and stopped returning errors if it is missing. - Downgraded missing codec clock message from dev_err to dev_dbg. Users who choose to use the rk817 PMIC but do not choose to use the codec will not get an error message unless they have debugging enabled, whereas users who wish to use the codec but don't have the clock defined will not get an error message unless they have debug messages enabled. This seems like a fair compromise between mututally exclusive users. Note that errors with the clock or codec will still result in a dev_err. - Updated DT documentation to note that codec subnode is optional if no properties are to exist beneath it. Changes in v10: - Correct order of test/ack/signed-off to chronological order. - Removed ifdef from mfd driver keep cell from being added when the codec was not to be built. - Changed codec DT parsing messages from error to debug to prevent errors in dmesg log in the rare but valid case a user has the rk817 PMIC but does not use the codec. Changes in v9: - Add cover letter. - Remove documentation for interrupt parent per Rob Herring's request. - Remove unused MODULE_DEVICE_TABLE to fix a bug identified by kernel test robot. Changes in v8: - Added additional documentation for missing properties of #sound-dai-cells, interrupt-parent, and wakeup-source for mfd documentation. - Corrected order of elements descriptions in device tree documentation. - Changed name of "mic-in-differential" to "rockchip,mic-in-differential". - Changed name of sound card from "rockchip,rk817-codec" to "Analog". - Removed unused resets and reset-names from the i2s1_2ch node. Changes in v7: - Removed ifdef around register definitions for MFD. - Replaced codec documentation with updates to MFD documentation. - Reordered elements in example to comply with upstream rules. - Added binding update back for Odroid Go Advance as requested. - Submitting patches from gmail now. Changes in v6: - Included additional project maintainers for correct subsystems. - Removed unneeded compatible from DT documentation. - Removed binding update for Odroid Go Advance (will do in separate series). Changes in v5: - Move register definitions from rk817_codec.h to main rk808.h register definitions. - Add volatile register for codec bits. - Add default values for codec bits. - Removed of_compatible from mtd driver (not necessary). - Switched to using parent regmap instead of private regmap for codec. Changes in v4: - Created set_pll() call. - Created user visible gain control in mic. - Check for return value of clk_prepare_enable(). - Removed duplicate clk_prepare_enable(). - Split DT documentation to separate commit. Changes in v3: - Use DAPM macros to set audio path. - Updated devicetree binding (as every rk817 has this codec chip). - Changed documentation to yaml format. - Split MFD changes to separate commit. Changes in v2: - Fixed audio path registers to solve some bugs. Chris Morgan (4): mfd: Add Rockchip rk817 audio CODEC support ASoC: Add Rockchip rk817 audio CODEC support dt-bindings: Add Rockchip rk817 audio CODEC support arm64: dts: rockchip: add rk817 codec to Odroid Go .../devicetree/bindings/mfd/rk808.txt | 188 ++++++ .../boot/dts/rockchip/rk3326-odroid-go2.dts | 36 +- drivers/mfd/rk808.c | 81 +++ include/linux/mfd/rk808.h | 81 +++ sound/soc/codecs/Kconfig | 6 + sound/soc/codecs/Makefile | 2 + sound/soc/codecs/rk817_codec.c | 539 ++++++++++++++++++ 7 files changed, 931 insertions(+), 2 deletions(-) create mode 100644 sound/soc/codecs/rk817_codec.c -- 2.25.1

4 13

Speakers + Microphone Volume Reset
by GitHub issues - edited 03 Jun '21

03 Jun '21

alsa-project/alsa-lib issue #144 was edited from rosenbergd: Every time I start up, connect or disconnect headphones, my volume resets to 74%, I've also noticed the microphone is doing the same thing. I'm not sure if this is a pulseaudio bug, but here are the logs from `pulseaudio -vv` which seems to show alsa increasing the volume to 74% after initially setting it to 25% which is the correct volume that it should be at: ``` D: [pulseaudio] module-alsa-card.c: Jack 'Headphone Jack' is now plugged in D: [pulseaudio] device-port.c: Setting port analog-output-headphones to status yes D: [pulseaudio] module-switch-on-port-available.c: Trying to switch to port analog-output-headphones I: [pulseaudio] alsa-sink.c: Successfully enabled deferred volume. I: [pulseaudio] alsa-sink.c: Hardware volume ranges from -179.00 dB to 0.00 dB. I: [pulseaudio] alsa-sink.c: Fixing base volume to 0.00 dB I: [pulseaudio] alsa-sink.c: Using hardware volume control. Hardware dB scale supported. I: [pulseaudio] alsa-sink.c: Using hardware mute control. D: [alsa-sink-ALC892 Analog] alsa-mixer.c: Activating path analog-output-headphones D: [alsa-sink-ALC892 Analog] alsa-mixer.c: Path analog-output-headphones (Headphones), direction=1, priority=99, probed=yes, supported=yes, has_mute=yes, has_volume=yes, has_dB=yes, min_volume=0, max_volume=64, min_dB=-179, max_dB=0 D: [alsa-sink-ALC892 Analog] alsa-mixer.c: Element 'Master', direction=1, switch=1, volume=1, volume_limit=-1, enumeration=0, required=0, required_any=0, required_absent=0, mask=0x7ffffffffffff, n_channels=1, override_map=yes D: [alsa-sink-ALC892 Analog] alsa-mixer.c: Element 'Headphone', direction=1, switch=1, volume=1, volume_limit=-1, enumeration=0, required=0, required_any=4, required_absent=0, mask=0x3600000000f66, n_channels=2, override_map=yes D: [alsa-sink-ALC892 Analog] alsa-mixer.c: Element 'Speaker', direction=1, switch=2, volume=2, volume_limit=-1, enumeration=0, required=0, required_any=0, required_absent=0, mask=0x6, n_channels=2, override_map=no D: [alsa-sink-ALC892 Analog] alsa-mixer.c: Element 'Bass Speaker', direction=1, switch=2, volume=2, volume_limit=-1, enumeration=0, required=0, required_any=0, required_absent=0, mask=0x6, n_channels=2, override_map=no D: [alsa-sink-ALC892 Analog] alsa-mixer.c: Element 'PCM', direction=1, switch=0, volume=1, volume_limit=-1, enumeration=0, required=0, required_any=0, required_absent=0, mask=0x3600000000f66, n_channels=2, override_map=yes D: [alsa-sink-ALC892 Analog] alsa-mixer.c: Element 'IEC958', direction=1, switch=2, volume=0, volume_limit=-1, enumeration=0, required=0, required_any=0, required_absent=0, mask=0x0, n_channels=0, override_map=no D: [alsa-sink-ALC892 Analog] alsa-mixer.c: Jack Dock Headphone, alsa_name='Dock Headphone Jack', index='0', detection unavailable D: [alsa-sink-ALC892 Analog] alsa-mixer.c: Jack Dock Headphone Phantom, alsa_name='Dock Headphone Phantom Jack', index='0', detection unavailable D: [alsa-sink-ALC892 Analog] alsa-mixer.c: Jack Front Headphone, alsa_name='Front Headphone Jack', index='0', detection unavailable D: [alsa-sink-ALC892 Analog] alsa-mixer.c: Jack Front Headphone, alsa_name='Front Headphone Jack', index='1', detection unavailable D: [alsa-sink-ALC892 Analog] alsa-mixer.c: Jack Front Headphone Front, alsa_name='Front Headphone Front Jack', index='0', detection unavailable D: [alsa-sink-ALC892 Analog] alsa-mixer.c: Jack Front Headphone Surround, alsa_name='Front Headphone Surround Jack', index='0', detection unavailable D: [alsa-sink-ALC892 Analog] alsa-mixer.c: Jack Front Headphone Phantom, alsa_name='Front Headphone Phantom Jack', index='0', detection unavailable D: [alsa-sink-ALC892 Analog] alsa-mixer.c: Jack Headphone, alsa_name='Headphone Jack', index='0', detection possible D: [alsa-sink-ALC892 Analog] alsa-mixer.c: Jack Headphone Phantom, alsa_name='Headphone Phantom Jack', index='0', detection unavailable D: [alsa-sink-ALC892 Analog] alsa-mixer.c: Jack Headphone Mic, alsa_name='Headphone Mic Jack', index='0', detection unavailable D: [alsa-sink-ALC892 Analog] alsa-mixer.c: Jack Headphone - Output, alsa_name='Headphone - Output Jack', index='0', detection unavailable I: [pulseaudio] sink.c: Changed port of sink 0 "alsa_output.pci-0000_00_1f.3.analog-stereo" to analog-output-headphones I: [pulseaudio] module-device-restore.c: Restoring volume for sink alsa_output.pci-0000_00_1f.3.analog-stereo. D: [pulseaudio] sink.c: The reference volume of sink alsa_output.pci-0000_00_1f.3.analog-stereo changed from front-left: 1369 / 2% / -100.80 dB, front-right: 1369 / 2% / -100.80 dB to front-left: 16380 / 25% / -36.13 dB, front-right: 16380 / 25% / -36.13 dB. D: [pulseaudio] core-subscribe.c: Dropped redundant event due to change event. D: [alsa-sink-ALC892 Analog] alsa-sink.c: Requested volume: front-left: 16380 / 25% / -36.13 dB, front-right: 16380 / 25% / -36.13 dB D: [alsa-sink-ALC892 Analog] alsa-sink.c: Got hardware volume: front-left: 16462 / 25% / -36.00 dB, front-right: 16462 / 25% / -36.00 dB D: [alsa-sink-ALC892 Analog] alsa-sink.c: Calculated software volume: front-left: 65210 / 100% / -0.13 dB, front-right: 65210 / 100% / -0.13 dB (accurate-enough=yes) D: [alsa-sink-ALC892 Analog] sink.c: Volume going up to 16380 at 263091695 D: [alsa-sink-ALC892 Analog] sink.c: Volume change to 16380 at 263091695 was written 5 usec late I: [pulseaudio] module-device-restore.c: Restoring mute state for sink alsa_output.pci-0000_00_1f.3.analog-stereo. D: [pulseaudio] sink.c: The mute of sink alsa_output.pci-0000_00_1f.3.analog-stereo changed from yes to no. D: [pulseaudio] core-subscribe.c: Dropped redundant event due to change event. D: [pulseaudio] device-port.c: Setting port analog-output-speaker to status no D: [pulseaudio] core-subscribe.c: Dropped redundant event due to change event. D: [pulseaudio] card.c: Setting card alsa_card.pci-0000_00_1f.3 profile output:analog-surround-21 to availability status no D: [pulseaudio] core-subscribe.c: Dropped redundant event due to change event. D: [pulseaudio] card.c: Setting card alsa_card.pci-0000_00_1f.3 profile output:analog-surround-21+input:analog-stereo to availability status no D: [pulseaudio] core-subscribe.c: Dropped redundant event due to change event. D: [pulseaudio] card.c: Setting card alsa_card.pci-0000_00_1f.3 profile output:analog-surround-40 to availability status no D: [pulseaudio] core-subscribe.c: Dropped redundant event due to change event. D: [pulseaudio] card.c: Setting card alsa_card.pci-0000_00_1f.3 profile output:analog-surround-40+input:analog-stereo to availability status no D: [pulseaudio] core-subscribe.c: Dropped redundant event due to change event. D: [alsa-sink-ALC892 Analog] alsa-sink.c: Read hardware volume: front-left: 5578 / 9% / -64.20 dB, front-right: 5578 / 9% / -64.20 dB D: [pulseaudio] sink.c: The reference volume of sink alsa_output.pci-0000_00_1f.3.analog-stereo changed from front-left: 16380 / 25% / -36.13 dB, front-right: 16380 / 25% / -36.13 dB to front-left: 5578 / 9% / -64.20 dB, front-right: 5578 / 9% / -64.20 dB. D: [pulseaudio] sink.c: The mute of sink alsa_output.pci-0000_00_1f.3.analog-stereo changed from no to yes. D: [pulseaudio] core-subscribe.c: Dropped redundant event due to change event. D: [alsa-sink-ALC892 Analog] alsa-sink.c: Read hardware volume: front-left: 49714 / 76% / -7.20 dB, front-right: 49714 / 76% / -7.20 dB D: [pulseaudio] sink.c: The reference volume of sink alsa_output.pci-0000_00_1f.3.analog-stereo changed from front-left: 5578 / 9% / -64.20 dB, front-right: 5578 / 9% / -64.20 dB to front-left: 49714 / 76% / -7.20 dB, front-right: 49714 / 76% / -7.20 dB. D: [pulseaudio] core-subscribe.c: Dropped redundant event due to change event. D: [pulseaudio] sink.c: The mute of sink alsa_output.pci-0000_00_1f.3.analog-stereo changed from yes to no. D: [pulseaudio] core-subscribe.c: Dropped redundant event due to change event. D: [alsa-sink-ALC892 Analog] alsa-sink.c: Read hardware volume: front-left: 48583 / 74% / -7.80 dB, front-right: 48583 / 74% / -7.80 dB D: [pulseaudio] sink.c: The reference volume of sink alsa_output.pci-0000_00_1f.3.analog-stereo changed from front-left: 49714 / 76% / -7.20 dB, front-right: 49714 / 76% / -7.20 dB to front-left: 48583 / 74% / -7.80 dB, front-right: 48583 / 74% / -7.80 dB. D: [pulseaudio] core-subscribe.c: Dropped redundant event due to change event. D: [alsa-sink-ALC892 Analog] alsa-sink.c: Read hardware volume: front-left: 48583 / 74% / -7.80 dB, front-right: 48583 / 74% / -7.80 dB D: [alsa-sink-ALC892 Analog] alsa-sink.c: Read hardware volume: front-left: 48583 / 74% / -7.80 dB, front-right: 48583 / 74% / -7.80 dB D: [alsa-sink-ALC892 Analog] alsa-sink.c: Read hardware volume: front-left: 48583 / 74% / -7.80 dB, front-right: 48583 / 74% / -7.80 dB D: [alsa-sink-ALC892 Analog] alsa-sink.c: Read hardware volume: front-left: 48583 / 74% / -7.80 dB, front-right: 48583 / 74% / -7.80 dB D: [alsa-sink-ALC892 Analog] alsa-sink.c: Read hardware volume: front-left: 48583 / 74% / -7.80 dB, front-right: 48583 / 74% / -7.80 dB D: [alsa-sink-ALC892 Analog] alsa-sink.c: Read hardware volume: front-left: 48583 / 74% / -7.80 dB, front-right: 48583 / 74% / -7.80 dB D: [alsa-sink-ALC892 Analog] alsa-sink.c: Read hardware volume: front-left: 48583 / 74% / -7.80 dB, front-right: 48583 / 74% / -7.80 dB I: [pulseaudio] module-device-restore.c: Storing volume/mute for device+port sink:alsa_output.pci-0000_00_1f.3.analog-stereo:analog-output-headphones. I: [pulseaudio] module-device-restore.c: Synced. ``` I'm using Linux Mint 20.1 Cinnamon v4.8.6 Kernel `v5.4.0-74` (but this was happening on `-73` as well) Issue URL : https://github.com/alsa-project/alsa-lib/issues/144 Repository URL: https://github.com/alsa-project/alsa-lib

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Speakers + Microphone Volume Reset
by GitHub issues - opened 03 Jun '21

03 Jun '21

alsa-project/alsa-lib issue #144 was opened from rosenbergd: Every time I start up, connect or disconnect headphones, my volume resets to 74%, I've also noticed the microphone is doing the same thing. I'm not sure if this is a pulseaudio bug, but here are the logs from `pulseaudio -vv` which seems to show alsa increasing the volume to 74% after initially setting it to 25% which is the correct volume that it should be at: ``` D: [pulseaudio] module-alsa-card.c: Jack 'Headphone Jack' is now plugged in D: [pulseaudio] device-port.c: Setting port analog-output-headphones to status yes D: [pulseaudio] module-switch-on-port-available.c: Trying to switch to port analog-output-headphones I: [pulseaudio] alsa-sink.c: Successfully enabled deferred volume. I: [pulseaudio] alsa-sink.c: Hardware volume ranges from -179.00 dB to 0.00 dB. I: [pulseaudio] alsa-sink.c: Fixing base volume to 0.00 dB I: [pulseaudio] alsa-sink.c: Using hardware volume control. Hardware dB scale supported. I: [pulseaudio] alsa-sink.c: Using hardware mute control. D: [alsa-sink-ALC892 Analog] alsa-mixer.c: Activating path analog-output-headphones D: [alsa-sink-ALC892 Analog] alsa-mixer.c: Path analog-output-headphones (Headphones), direction=1, priority=99, probed=yes, supported=yes, has_mute=yes, has_volume=yes, has_dB=yes, min_volume=0, max_volume=64, min_dB=-179, max_dB=0 D: [alsa-sink-ALC892 Analog] alsa-mixer.c: Element 'Master', direction=1, switch=1, volume=1, volume_limit=-1, enumeration=0, required=0, required_any=0, required_absent=0, mask=0x7ffffffffffff, n_channels=1, override_map=yes D: [alsa-sink-ALC892 Analog] alsa-mixer.c: Element 'Headphone', direction=1, switch=1, volume=1, volume_limit=-1, enumeration=0, required=0, required_any=4, required_absent=0, mask=0x3600000000f66, n_channels=2, override_map=yes D: [alsa-sink-ALC892 Analog] alsa-mixer.c: Element 'Speaker', direction=1, switch=2, volume=2, volume_limit=-1, enumeration=0, required=0, required_any=0, required_absent=0, mask=0x6, n_channels=2, override_map=no D: [alsa-sink-ALC892 Analog] alsa-mixer.c: Element 'Bass Speaker', direction=1, switch=2, volume=2, volume_limit=-1, enumeration=0, required=0, required_any=0, required_absent=0, mask=0x6, n_channels=2, override_map=no D: [alsa-sink-ALC892 Analog] alsa-mixer.c: Element 'PCM', direction=1, switch=0, volume=1, volume_limit=-1, enumeration=0, required=0, required_any=0, required_absent=0, mask=0x3600000000f66, n_channels=2, override_map=yes D: [alsa-sink-ALC892 Analog] alsa-mixer.c: Element 'IEC958', direction=1, switch=2, volume=0, volume_limit=-1, enumeration=0, required=0, required_any=0, required_absent=0, mask=0x0, n_channels=0, override_map=no D: [alsa-sink-ALC892 Analog] alsa-mixer.c: Jack Dock Headphone, alsa_name='Dock Headphone Jack', index='0', detection unavailable D: [alsa-sink-ALC892 Analog] alsa-mixer.c: Jack Dock Headphone Phantom, alsa_name='Dock Headphone Phantom Jack', index='0', detection unavailable D: [alsa-sink-ALC892 Analog] alsa-mixer.c: Jack Front Headphone, alsa_name='Front Headphone Jack', index='0', detection unavailable D: [alsa-sink-ALC892 Analog] alsa-mixer.c: Jack Front Headphone, alsa_name='Front Headphone Jack', index='1', detection unavailable D: [alsa-sink-ALC892 Analog] alsa-mixer.c: Jack Front Headphone Front, alsa_name='Front Headphone Front Jack', index='0', detection unavailable D: [alsa-sink-ALC892 Analog] alsa-mixer.c: Jack Front Headphone Surround, alsa_name='Front Headphone Surround Jack', index='0', detection unavailable D: [alsa-sink-ALC892 Analog] alsa-mixer.c: Jack Front Headphone Phantom, alsa_name='Front Headphone Phantom Jack', index='0', detection unavailable D: [alsa-sink-ALC892 Analog] alsa-mixer.c: Jack Headphone, alsa_name='Headphone Jack', index='0', detection possible D: [alsa-sink-ALC892 Analog] alsa-mixer.c: Jack Headphone Phantom, alsa_name='Headphone Phantom Jack', index='0', detection unavailable D: [alsa-sink-ALC892 Analog] alsa-mixer.c: Jack Headphone Mic, alsa_name='Headphone Mic Jack', index='0', detection unavailable D: [alsa-sink-ALC892 Analog] alsa-mixer.c: Jack Headphone - Output, alsa_name='Headphone - Output Jack', index='0', detection unavailable I: [pulseaudio] sink.c: Changed port of sink 0 "alsa_output.pci-0000_00_1f.3.analog-stereo" to analog-output-headphones I: [pulseaudio] module-device-restore.c: Restoring volume for sink alsa_output.pci-0000_00_1f.3.analog-stereo. D: [pulseaudio] sink.c: The reference volume of sink alsa_output.pci-0000_00_1f.3.analog-stereo changed from front-left: 1369 / 2% / -100.80 dB, front-right: 1369 / 2% / -100.80 dB to front-left: 16380 / 25% / -36.13 dB, front-right: 16380 / 25% / -36.13 dB. D: [pulseaudio] core-subscribe.c: Dropped redundant event due to change event. D: [alsa-sink-ALC892 Analog] alsa-sink.c: Requested volume: front-left: 16380 / 25% / -36.13 dB, front-right: 16380 / 25% / -36.13 dB D: [alsa-sink-ALC892 Analog] alsa-sink.c: Got hardware volume: front-left: 16462 / 25% / -36.00 dB, front-right: 16462 / 25% / -36.00 dB D: [alsa-sink-ALC892 Analog] alsa-sink.c: Calculated software volume: front-left: 65210 / 100% / -0.13 dB, front-right: 65210 / 100% / -0.13 dB (accurate-enough=yes) D: [alsa-sink-ALC892 Analog] sink.c: Volume going up to 16380 at 263091695 D: [alsa-sink-ALC892 Analog] sink.c: Volume change to 16380 at 263091695 was written 5 usec late I: [pulseaudio] module-device-restore.c: Restoring mute state for sink alsa_output.pci-0000_00_1f.3.analog-stereo. D: [pulseaudio] sink.c: The mute of sink alsa_output.pci-0000_00_1f.3.analog-stereo changed from yes to no. D: [pulseaudio] core-subscribe.c: Dropped redundant event due to change event. D: [pulseaudio] device-port.c: Setting port analog-output-speaker to status no D: [pulseaudio] core-subscribe.c: Dropped redundant event due to change event. D: [pulseaudio] card.c: Setting card alsa_card.pci-0000_00_1f.3 profile output:analog-surround-21 to availability status no D: [pulseaudio] core-subscribe.c: Dropped redundant event due to change event. D: [pulseaudio] card.c: Setting card alsa_card.pci-0000_00_1f.3 profile output:analog-surround-21+input:analog-stereo to availability status no D: [pulseaudio] core-subscribe.c: Dropped redundant event due to change event. D: [pulseaudio] card.c: Setting card alsa_card.pci-0000_00_1f.3 profile output:analog-surround-40 to availability status no D: [pulseaudio] core-subscribe.c: Dropped redundant event due to change event. D: [pulseaudio] card.c: Setting card alsa_card.pci-0000_00_1f.3 profile output:analog-surround-40+input:analog-stereo to availability status no D: [pulseaudio] core-subscribe.c: Dropped redundant event due to change event. D: [alsa-sink-ALC892 Analog] alsa-sink.c: Read hardware volume: front-left: 5578 / 9% / -64.20 dB, front-right: 5578 / 9% / -64.20 dB D: [pulseaudio] sink.c: The reference volume of sink alsa_output.pci-0000_00_1f.3.analog-stereo changed from front-left: 16380 / 25% / -36.13 dB, front-right: 16380 / 25% / -36.13 dB to front-left: 5578 / 9% / -64.20 dB, front-right: 5578 / 9% / -64.20 dB. D: [pulseaudio] sink.c: The mute of sink alsa_output.pci-0000_00_1f.3.analog-stereo changed from no to yes. D: [pulseaudio] core-subscribe.c: Dropped redundant event due to change event. D: [alsa-sink-ALC892 Analog] alsa-sink.c: Read hardware volume: front-left: 49714 / 76% / -7.20 dB, front-right: 49714 / 76% / -7.20 dB D: [pulseaudio] sink.c: The reference volume of sink alsa_output.pci-0000_00_1f.3.analog-stereo changed from front-left: 5578 / 9% / -64.20 dB, front-right: 5578 / 9% / -64.20 dB to front-left: 49714 / 76% / -7.20 dB, front-right: 49714 / 76% / -7.20 dB. D: [pulseaudio] core-subscribe.c: Dropped redundant event due to change event. D: [pulseaudio] sink.c: The mute of sink alsa_output.pci-0000_00_1f.3.analog-stereo changed from yes to no. D: [pulseaudio] core-subscribe.c: Dropped redundant event due to change event. D: [alsa-sink-ALC892 Analog] alsa-sink.c: Read hardware volume: front-left: 48583 / 74% / -7.80 dB, front-right: 48583 / 74% / -7.80 dB D: [pulseaudio] sink.c: The reference volume of sink alsa_output.pci-0000_00_1f.3.analog-stereo changed from front-left: 49714 / 76% / -7.20 dB, front-right: 49714 / 76% / -7.20 dB to front-left: 48583 / 74% / -7.80 dB, front-right: 48583 / 74% / -7.80 dB. D: [pulseaudio] core-subscribe.c: Dropped redundant event due to change event. D: [alsa-sink-ALC892 Analog] alsa-sink.c: Read hardware volume: front-left: 48583 / 74% / -7.80 dB, front-right: 48583 / 74% / -7.80 dB D: [alsa-sink-ALC892 Analog] alsa-sink.c: Read hardware volume: front-left: 48583 / 74% / -7.80 dB, front-right: 48583 / 74% / -7.80 dB D: [alsa-sink-ALC892 Analog] alsa-sink.c: Read hardware volume: front-left: 48583 / 74% / -7.80 dB, front-right: 48583 / 74% / -7.80 dB D: [alsa-sink-ALC892 Analog] alsa-sink.c: Read hardware volume: front-left: 48583 / 74% / -7.80 dB, front-right: 48583 / 74% / -7.80 dB D: [alsa-sink-ALC892 Analog] alsa-sink.c: Read hardware volume: front-left: 48583 / 74% / -7.80 dB, front-right: 48583 / 74% / -7.80 dB D: [alsa-sink-ALC892 Analog] alsa-sink.c: Read hardware volume: front-left: 48583 / 74% / -7.80 dB, front-right: 48583 / 74% / -7.80 dB D: [alsa-sink-ALC892 Analog] alsa-sink.c: Read hardware volume: front-left: 48583 / 74% / -7.80 dB, front-right: 48583 / 74% / -7.80 dB I: [pulseaudio] module-device-restore.c: Storing volume/mute for device+port sink:alsa_output.pci-0000_00_1f.3.analog-stereo:analog-output-headphones. I: [pulseaudio] module-device-restore.c: Synced. ``` Issue URL : https://github.com/alsa-project/alsa-lib/issues/144 Repository URL: https://github.com/alsa-project/alsa-lib

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[PATCH v2] ASoC: fsl-asoc-card: change dev_err to dev_err_probe for defer probe
by Shengjiu Wang 03 Jun '21

03 Jun '21

Don't need to print error message for defer probe Signed-off-by: Shengjiu Wang <shengjiu.wang(a)nxp.com> --- changes in v2: - use dev_err_probe instead of dev_dbg sound/soc/fsl/fsl-asoc-card.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/fsl/fsl-asoc-card.c b/sound/soc/fsl/fsl-asoc-card.c index 4f55b316cf0f..2fa371ec6b1c 100644 --- a/sound/soc/fsl/fsl-asoc-card.c +++ b/sound/soc/fsl/fsl-asoc-card.c @@ -709,8 +709,8 @@ static int fsl_asoc_card_probe(struct platform_device *pdev) of_node_put(framemaster); if (!fsl_asoc_card_is_ac97(priv) && !codec_dev) { - dev_err(&pdev->dev, "failed to find codec device\n"); ret = -EPROBE_DEFER; + dev_err_probe(&pdev->dev, ret, "failed to find codec device\n"); goto asrc_fail; } -- 2.27.0

1 0

trivial patch
by josef radinger 02 Jun '21

02 Jun '21

just fix some trivial typo in comments, sorry for the third try. now the direction between diffs is correct. diff --git a/sound/soc/qcom/qdsp6/q6adm.c b/sound/soc/qcom/qdsp6/q6adm.c index 1855b805eba2..3c68df82c95c 100644 --- a/sound/soc/qcom/qdsp6/q6adm.c +++ b/sound/soc/qcom/qdsp6/q6adm.c @@ -373,7 +373,7 @@ static int q6adm_device_open(struct q6adm *adm, struct q6copp *copp, * @rate: rate at which copp is required. * @channel_mode: channel mode * @topology: adm topology id - * @perf_mode: performace mode. + * @perf_mode: performance mode. * @bit_width: audio sample bit width * @app_type: Application type. * @acdb_id: ACDB id @@ -452,7 +452,7 @@ EXPORT_SYMBOL_GPL(q6adm_get_copp_id); * @dev: Pointer to adm child device. * @path: playback or capture path. * @payload_map: map between session id and afe ports. - * @perf_mode: Performace mode. + * @perf_mode: Performance mode. * * Return: Will be an negative on error or a zero on success. */ yours josef

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