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

Nux Mighty Plug USB - no clock source
by Peter Allen 02 Feb '21

02 Feb '21

I have a NUX Mighty Plug guitar effects which has a USB Audio interface. Unfortunately it doesn't work in Linux, and I'd love to fix it. It gets registered with Alsa, except it ends up without any valid sample rates (by the look of it due to a buggy clock source descriptor). This is on Ubuntu-Studio 20.04 LTS _Focal Fossa with kernel 5.4.0-62-lowlatency on a HP Omen 15 with i5-7300HQ I don't know what I'm doing (but did a little module development rather a long time ago in the 2.2 days :-)) Help is greatly appreciated. When plugged in I get: [12476.413792] usb 1-1: New USB device found, idVendor=1fc9, idProduct=8260, bcdDevice= 1.00 [12476.413797] usb 1-1: New USB device strings: Mfr=1, Product=2, SerialNumber=3 [12476.413800] usb 1-1: Product: MIGHTY PLUG USB [12476.413804] usb 1-1: Manufacturer: NUX [12476.413807] usb 1-1: SerialNumber: 2007151403 [12476.444988] usb 1-1: __uac_clock_find_source(): selector reported illegal value, id 40, ret 0 [12476.444992] usb 1-1: parse_audio_format_rates_v2v3(): unable to find clock source (clock -22) [12476.445413] usb 1-1: __uac_clock_find_source(): selector reported illegal value, id 40, ret 0 [12476.445479] usb 1-1: __uac_clock_find_source(): selector reported illegal value, id 40, ret 0 [12476.447095] usb 1-1: __uac_clock_find_source(): selector reported illegal value, id 40, ret 0 [12476.447098] usb 1-1: parse_audio_format_rates_v2v3(): unable to find clock source (clock -22) [12476.447459] usb 1-1: __uac_clock_find_source(): selector reported illegal value, id 40, ret 0 [12476.447684] usb 1-1: __uac_clock_find_source(): selector reported illegal value, id 40, ret 0 [12476.450397] usbcore: registered new interface driver snd-usb-audio ===== /proc/asound/cards give ====== 0 [PCH ]: HDA-Intel - HDA Intel PCH HDA Intel PCH at 0xb4428000 irq 134 1 [NVidia ]: HDA-Intel - HDA NVidia HDA NVidia at 0xb4000000 irq 17 2 [USB ]: USB-Audio - MIGHTY PLUG USB NUX MIGHTY PLUG USB at usb-0000:00:14.0-1, high speed ==== /proc/asound/stream======== NUX MIGHTY PLUG USB at usb-0000:00:14.0-1, high speed : USB Audio Playback: Status: Stop Interface 1 Altset 1 Format: S32_LE Channels: 2 Endpoint: 1 OUT (ASYNC) Rates: Data packet interval: 1000 us Bits: 24 Capture: Status: Stop Interface 2 Altset 1 Format: S32_LE Channels: 2 Endpoint: 2 IN (ASYNC) Rates: Data packet interval: 1000 us Bits: 24 ======== arecord =========== arecord -D "hw:2" -c 2 -f s24 -r 44100 -d 2 -v test.wav Recording WAVE 'xxxc' : Signed 24 bit Little Endian, Rate 44100 Hz, Stereo arecord: set_params:1314: Broken configuration for this PCM: no configuration available ========lsusb -v ========== iSerial 2 200901010001 bNumConfigurations 1 Bus 001 Device 005: ID 1fc9:8260 NXP Semiconductors MIGHTY PLUG USB Device Descriptor: bLength 18 bDescriptorType 1 bcdUSB 2.00 bDeviceClass 239 Miscellaneous Device bDeviceSubClass 2 bDeviceProtocol 1 Interface Association bMaxPacketSize0 64 idVendor 0x1fc9 NXP Semiconductors idProduct 0x8260 bcdDevice 1.00 iManufacturer 1 NUX iProduct 2 MIGHTY PLUG USB iSerial 3 2007151403 bNumConfigurations 1 Configuration Descriptor: bLength 9 bDescriptorType 2 wTotalLength 0x013e bNumInterfaces 4 bConfigurationValue 1 iConfiguration 0 bmAttributes 0xc0 Self Powered MaxPower 2mA Interface Association: bLength 8 bDescriptorType 11 bFirstInterface 0 bInterfaceCount 3 bFunctionClass 1 Audio bFunctionSubClass 0 bFunctionProtocol 32 iFunction 0 Interface Descriptor: bLength 9 bDescriptorType 4 bInterfaceNumber 0 bAlternateSetting 0 bNumEndpoints 0 bInterfaceClass 1 Audio bInterfaceSubClass 1 Control Device bInterfaceProtocol 32 iInterface 0 AudioControl Interface Descriptor: bLength 9 bDescriptorType 36 bDescriptorSubtype 1 (HEADER) bcdADC 2.00 bCategory 8 wTotalLength 0x0065 bmControls 0x00 AudioControl Interface Descriptor: bLength 8 bDescriptorType 36 bDescriptorSubtype 10 (CLOCK_SOURCE) bClockID 41 bmAttributes 3 Internal programmable clock bmControls 0x07 Clock Frequency Control (read/write) Clock Validity Control (read-only) bAssocTerminal 0 iClockSource 0 AudioControl Interface Descriptor: bLength 8 bDescriptorType 36 bDescriptorSubtype 11 (CLOCK_SELECTOR) bClockID 40 bNrInPins 1 baCSourceID(0) 41 bmControls 0x03 Clock Selector Control (read/write) iClockSelector 0 AudioControl Interface Descriptor: bLength 17 bDescriptorType 36 bDescriptorSubtype 2 (INPUT_TERMINAL) bTerminalID 2 wTerminalType 0x0101 USB Streaming bAssocTerminal 0 bCSourceID 40 bNrChannels 2 bmChannelConfig 0x00000000 iChannelNames 0 bmControls 0x0000 iTerminal 0 AudioControl Interface Descriptor: bLength 18 bDescriptorType 36 bDescriptorSubtype 6 (FEATURE_UNIT) bUnitID 10 bSourceID 2 bmaControls(0) 0x0000000f Mute Control (read/write) Volume Control (read/write) bmaControls(1) 0x0000000f Mute Control (read/write) Volume Control (read/write) bmaControls(2) 0x0000000f Mute Control (read/write) Volume Control (read/write) iFeature 0 AudioControl Interface Descriptor: bLength 12 bDescriptorType 36 bDescriptorSubtype 3 (OUTPUT_TERMINAL) bTerminalID 20 wTerminalType 0x0301 Speaker bAssocTerminal 0 bSourceID 10 bCSourceID 40 bmControls 0x0000 iTerminal 0 AudioControl Interface Descriptor: bLength 17 bDescriptorType 36 bDescriptorSubtype 2 (INPUT_TERMINAL) bTerminalID 1 wTerminalType 0x0201 Microphone bAssocTerminal 0 bCSourceID 40 bNrChannels 2 bmChannelConfig 0x00000000 iChannelNames 0 bmControls 0x0000 iTerminal 0 AudioControl Interface Descriptor: bLength 12 bDescriptorType 36 bDescriptorSubtype 3 (OUTPUT_TERMINAL) bTerminalID 22 wTerminalType 0x0101 USB Streaming bAssocTerminal 0 bSourceID 2 bCSourceID 40 bmControls 0x0000 iTerminal 0 Interface Descriptor: bLength 9 bDescriptorType 4 bInterfaceNumber 1 bAlternateSetting 0 bNumEndpoints 0 bInterfaceClass 1 Audio bInterfaceSubClass 2 Streaming bInterfaceProtocol 32 iInterface 0 Interface Descriptor: bLength 9 bDescriptorType 4 bInterfaceNumber 1 bAlternateSetting 1 bNumEndpoints 2 bInterfaceClass 1 Audio bInterfaceSubClass 2 Streaming bInterfaceProtocol 32 iInterface 0 AudioStreaming Interface Descriptor: bLength 16 bDescriptorType 36 bDescriptorSubtype 1 (AS_GENERAL) bTerminalLink 2 bmControls 0x00 bFormatType 1 bmFormats 0x00000001 PCM bNrChannels 2 bmChannelConfig 0x00000000 iChannelNames 0 AudioStreaming Interface Descriptor: bLength 6 bDescriptorType 36 bDescriptorSubtype 2 (FORMAT_TYPE) bFormatType 1 (FORMAT_TYPE_I) bSubslotSize 4 bBitResolution 24 Endpoint Descriptor: bLength 7 bDescriptorType 5 bEndpointAddress 0x01 EP 1 OUT bmAttributes 5 Transfer Type Isochronous Synch Type Asynchronous Usage Type Data wMaxPacketSize 0x0400 1x 1024 bytes bInterval 4 AudioStreaming Endpoint Descriptor: bLength 8 bDescriptorType 37 bDescriptorSubtype 1 (EP_GENERAL) bmAttributes 0x00 bmControls 0x00 bLockDelayUnits 0 Undefined wLockDelay 0x0000 Endpoint Descriptor: bLength 7 bDescriptorType 5 bEndpointAddress 0x81 EP 1 IN bmAttributes 17 Transfer Type Isochronous Synch Type None Usage Type Feedback wMaxPacketSize 0x0004 1x 4 bytes bInterval 4 Interface Descriptor: bLength 9 bDescriptorType 4 bInterfaceNumber 2 bAlternateSetting 0 bNumEndpoints 0 bInterfaceClass 1 Audio bInterfaceSubClass 2 Streaming bInterfaceProtocol 32 iInterface 0 Interface Descriptor: bLength 9 bDescriptorType 4 bInterfaceNumber 2 bAlternateSetting 1 bNumEndpoints 1 bInterfaceClass 1 Audio bInterfaceSubClass 2 Streaming bInterfaceProtocol 32 iInterface 0 AudioStreaming Interface Descriptor: bLength 16 bDescriptorType 36 bDescriptorSubtype 1 (AS_GENERAL) bTerminalLink 22 bmControls 0x00 bFormatType 1 bmFormats 0x00000001 PCM bNrChannels 2 bmChannelConfig 0x00000000 iChannelNames 0 AudioStreaming Interface Descriptor: bLength 6 bDescriptorType 36 bDescriptorSubtype 2 (FORMAT_TYPE) bFormatType 1 (FORMAT_TYPE_I) bSubslotSize 4 bBitResolution 24 Endpoint Descriptor: bLength 7 bDescriptorType 5 bEndpointAddress 0x82 EP 2 IN bmAttributes 5 Transfer Type Isochronous Synch Type Asynchronous Usage Type Data wMaxPacketSize 0x0400 1x 1024 bytes bInterval 4 AudioStreaming Endpoint Descriptor: bLength 8 bDescriptorType 37 bDescriptorSubtype 1 (EP_GENERAL) bmAttributes 0x00 bmControls 0x00 bLockDelayUnits 0 Undefined wLockDelay 0x0000 Interface Descriptor: bLength 9 bDescriptorType 4 bInterfaceNumber 3 bAlternateSetting 0 bNumEndpoints 2 bInterfaceClass 1 Audio bInterfaceSubClass 3 MIDI Streaming bInterfaceProtocol 0 iInterface 4 MIGHTY PLUG USB MIDI 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 0x04 EP 4 OUT bmAttributes 2 Transfer Type Bulk Synch Type None Usage Type Data wMaxPacketSize 0x0100 1x 256 bytes bInterval 0 bRefresh 0 bSynchAddress 0 MIDIStreaming Endpoint Descriptor: bLength 5 bDescriptorType 37 bDescriptorSubtype 1 (GENERAL) bNumEmbMIDIJack 1 baAssocJackID( 0) 1 Endpoint Descriptor: bLength 9 bDescriptorType 5 bEndpointAddress 0x83 EP 3 IN bmAttributes 2 Transfer Type Bulk Synch Type None Usage Type Data wMaxPacketSize 0x0100 1x 256 bytes bInterval 0 bRefresh 0 bSynchAddress 0 MIDIStreaming Endpoint Descriptor: bLength 5 bDescriptorType 37 bDescriptorSubtype 1 (GENERAL) bNumEmbMIDIJack 1 baAssocJackID( 0) 3 Device Qualifier (for other device speed): bLength 10 bDescriptorType 6 bcdUSB 2.00 bDeviceClass 0 bDeviceSubClass 0 bDeviceProtocol 0 bMaxPacketSize0 64 bNumConfigurations 1 can't get debug descriptor: Resource temporarily unavailable Device Status: 0x0001 Self Powered Bus 001 Device 001: ID 1d6b:0002 Linux Foundation 2.0 root hub Device Descriptor: bLength 18 bDescriptorType 1 bcdUSB 2.00 bDeviceClass 9 Hub bDeviceSubClass 0 bDeviceProtocol 1 Single TT bMaxPacketSize0 64 idVendor 0x1d6b Linux Foundation idProduct 0x0002 2.0 root hub bcdDevice 5.04 iManufacturer 3 Linux 5.4.0-58-lowlatency xhci-hcd iProduct 2 xHCI Host Controller iSerial 1 0000:00:14.0 bNumConfigurations 1 Configuration Descriptor: bLength 9 bDescriptorType 2 wTotalLength 0x0019 bNumInterfaces 1 bConfigurationValue 1 iConfiguration 0 bmAttributes 0xe0 Self Powered Remote Wakeup MaxPower 0mA Interface Descriptor: bLength 9 bDescriptorType 4 bInterfaceNumber 0 bAlternateSetting 0 bNumEndpoints 1 bInterfaceClass 9 Hub bInterfaceSubClass 0 bInterfaceProtocol 0 Full speed (or root) hub iInterface 0 Endpoint Descriptor: bLength 7 bDescriptorType 5 bEndpointAddress 0x81 EP 1 IN bmAttributes 3 Transfer Type Interrupt Synch Type None Usage Type Data wMaxPacketSize 0x0004 1x 4 bytes bInterval 12 Hub Descriptor: bLength 13 bDescriptorType 41 nNbrPorts 16 wHubCharacteristic 0x000a No power switching (usb 1.0) Per-port overcurrent protection TT think time 8 FS bits bPwrOn2PwrGood 10 * 2 milli seconds bHubContrCurrent 0 milli Ampere DeviceRemovable 0x90 0x01 0x00 PortPwrCtrlMask 0xff 0xff 0xff Hub Port Status: Port 1: 0000.0503 highspeed power enable connect Port 2: 0000.0100 power Port 3: 0000.0100 power Port 4: 0000.0503 highspeed power enable connect Port 5: 0000.0100 power Port 6: 0000.0100 power Port 7: 0000.0103 power enable connect Port 8: 0000.0100 power Port 9: 0000.0100 power Port 10: 0000.0100 power Port 11: 0000.0100 power Port 12: 0000.0100 power Port 13: 0000.0100 power Port 14: 0000.0100 power Port 15: 0000.0100 power Port 16: 0000.0100 power can't get device qualifier: Resource temporarily unavailable can't get debug descriptor: Resource temporarily unavailable Device Status: 0x0001 Self Powered

2 3

[PATCH v2 0/2] ALSA: usb-audio: Add Pioneer DJM-450 support
by Olivia Mackintosh 02 Feb '21

02 Feb '21

Re-roll of this patchset to fix the following: 1. Missing 'Signed-of-by:' line 2. Spaces around operators 3. Test that I'm threading submissions correctly Apologies for the clumsiness on my part. Olivia Olivia Mackintosh (2): ALSA: usb-audio: Add DJM450 to Pioneer format quirk ALSA: usb-audio: Add DJM-450 to the quirks table sound/usb/quirks-table.h | 57 ++++++++++++++++++++++++++++++++++++++++ sound/usb/quirks.c | 20 ++++++++++++++ 2 files changed, 77 insertions(+) -- 2.30.0

2 3

[PATCH] ASoC: topology: KUnit: Convert from cpu to data format
by Amadeusz Sławiński 02 Feb '21

02 Feb '21

When creating topology templates and overriding data in specific test cases it should be done with cpu_to_le32 macro, so we operate on correct data on all architectures, as topology parser use le32_to_cpu to parse data from structures. Reported-by: kernel test robot <lkp(a)intel.com> Signed-off-by: Amadeusz Sławiński <amadeuszx.slawinski(a)linux.intel.com> --- sound/soc/soc-topology-test.c | 66 +++++++++++++++++------------------ 1 file changed, 33 insertions(+), 33 deletions(-) diff --git a/sound/soc/soc-topology-test.c b/sound/soc/soc-topology-test.c index 36e2a3486dbf..ae3968161509 100644 --- a/sound/soc/soc-topology-test.c +++ b/sound/soc/soc-topology-test.c @@ -121,19 +121,19 @@ struct tplg_tmpl_001 { static struct tplg_tmpl_001 tplg_tmpl_empty = { .header = { - .magic = SND_SOC_TPLG_MAGIC, - .abi = 5, + .magic = cpu_to_le32(SND_SOC_TPLG_MAGIC), + .abi = cpu_to_le32(5), .version = 0, - .type = SND_SOC_TPLG_TYPE_MANIFEST, - .size = sizeof(struct snd_soc_tplg_hdr), + .type = cpu_to_le32(SND_SOC_TPLG_TYPE_MANIFEST), + .size = cpu_to_le32(sizeof(struct snd_soc_tplg_hdr)), .vendor_type = 0, - .payload_size = sizeof(struct snd_soc_tplg_manifest), + .payload_size = cpu_to_le32(sizeof(struct snd_soc_tplg_manifest)), .index = 0, - .count = 1, + .count = cpu_to_le32(1), }, .manifest = { - .size = sizeof(struct snd_soc_tplg_manifest), + .size = cpu_to_le32(sizeof(struct snd_soc_tplg_manifest)), /* rest of fields is 0 */ }, }; @@ -149,60 +149,60 @@ struct tplg_tmpl_002 { static struct tplg_tmpl_002 tplg_tmpl_with_pcm = { .header = { - .magic = SND_SOC_TPLG_MAGIC, - .abi = 5, + .magic = cpu_to_le32(SND_SOC_TPLG_MAGIC), + .abi = cpu_to_le32(5), .version = 0, - .type = SND_SOC_TPLG_TYPE_MANIFEST, - .size = sizeof(struct snd_soc_tplg_hdr), + .type = cpu_to_le32(SND_SOC_TPLG_TYPE_MANIFEST), + .size = cpu_to_le32(sizeof(struct snd_soc_tplg_hdr)), .vendor_type = 0, - .payload_size = sizeof(struct snd_soc_tplg_manifest), + .payload_size = cpu_to_le32(sizeof(struct snd_soc_tplg_manifest)), .index = 0, - .count = 1, + .count = cpu_to_le32(1), }, .manifest = { - .size = sizeof(struct snd_soc_tplg_manifest), - .pcm_elems = 1, + .size = cpu_to_le32(sizeof(struct snd_soc_tplg_manifest)), + .pcm_elems = cpu_to_le32(1), /* rest of fields is 0 */ }, .pcm_header = { - .magic = SND_SOC_TPLG_MAGIC, - .abi = 5, + .magic = cpu_to_le32(SND_SOC_TPLG_MAGIC), + .abi = cpu_to_le32(5), .version = 0, - .type = SND_SOC_TPLG_TYPE_PCM, - .size = sizeof(struct snd_soc_tplg_hdr), + .type = cpu_to_le32(SND_SOC_TPLG_TYPE_PCM), + .size = cpu_to_le32(sizeof(struct snd_soc_tplg_hdr)), .vendor_type = 0, - .payload_size = sizeof(struct snd_soc_tplg_pcm), + .payload_size = cpu_to_le32(sizeof(struct snd_soc_tplg_pcm)), .index = 0, - .count = 1, + .count = cpu_to_le32(1), }, .pcm = { - .size = sizeof(struct snd_soc_tplg_pcm), + .size = cpu_to_le32(sizeof(struct snd_soc_tplg_pcm)), .pcm_name = "KUNIT Audio", .dai_name = "kunit-audio-dai", .pcm_id = 0, .dai_id = 0, - .playback = 1, - .capture = 1, + .playback = cpu_to_le32(1), + .capture = cpu_to_le32(1), .compress = 0, .stream = { [0] = { - .channels = 2, + .channels = cpu_to_le32(2), }, [1] = { - .channels = 2, + .channels = cpu_to_le32(2), }, }, .num_streams = 0, .caps = { [0] = { .name = "kunit-audio-playback", - .channels_min = 2, - .channels_max = 2, + .channels_min = cpu_to_le32(2), + .channels_max = cpu_to_le32(2), }, [1] = { .name = "kunit-audio-capture", - .channels_min = 2, - .channels_max = 2, + .channels_min = cpu_to_le32(2), + .channels_max = cpu_to_le32(2), }, }, .flag_mask = 0, @@ -460,7 +460,7 @@ static void snd_soc_tplg_test_load_empty_tplg_bad_magic(struct kunit *test) * override abi * any value != magic number is wrong */ - data->header.magic = SND_SOC_TPLG_MAGIC + 1; + data->header.magic = cpu_to_le32(SND_SOC_TPLG_MAGIC + 1); kunit_comp->fw.data = (u8 *)data; kunit_comp->fw.size = size; @@ -516,7 +516,7 @@ static void snd_soc_tplg_test_load_empty_tplg_bad_abi(struct kunit *test) * override abi * any value != accepted range is wrong */ - data->header.abi = SND_SOC_TPLG_ABI_VERSION + 1; + data->header.abi = cpu_to_le32(SND_SOC_TPLG_ABI_VERSION + 1); kunit_comp->fw.data = (u8 *)data; kunit_comp->fw.size = size; @@ -572,7 +572,7 @@ static void snd_soc_tplg_test_load_empty_tplg_bad_size(struct kunit *test) * override size * any value != struct size is wrong */ - data->header.size = sizeof(struct snd_soc_tplg_hdr) + 1; + data->header.size = cpu_to_le32(sizeof(struct snd_soc_tplg_hdr) + 1); kunit_comp->fw.data = (u8 *)data; kunit_comp->fw.size = size; -- 2.25.1

2 1

[PATCH] ASoC: fsl_spdif: Utilize the defined parameter to clear code
by Tang Bin 02 Feb '21

02 Feb '21

Utilize the defined parameter 'dev' to make the code cleaner. Signed-off-by: Tang Bin <tangbin(a)cmss.chinamobile.com> --- sound/soc/fsl/fsl_spdif.c | 10 +++++----- 1 file changed, 5 insertions(+), 5 deletions(-) diff --git a/sound/soc/fsl/fsl_spdif.c b/sound/soc/fsl/fsl_spdif.c index 455f96908..b6d5563df 100644 --- a/sound/soc/fsl/fsl_spdif.c +++ b/sound/soc/fsl/fsl_spdif.c @@ -1215,7 +1215,7 @@ static int fsl_spdif_probe_txclk(struct fsl_spdif_priv *spdif_priv, for (i = 0; i < STC_TXCLK_SRC_MAX; i++) { sprintf(tmp, "rxtx%d", i); - clk = devm_clk_get(&pdev->dev, tmp); + clk = devm_clk_get(dev, tmp); if (IS_ERR(clk)) { dev_err(dev, "no rxtx%d clock in devicetree\n", i); return PTR_ERR(clk); @@ -1237,14 +1237,14 @@ static int fsl_spdif_probe_txclk(struct fsl_spdif_priv *spdif_priv, break; } - dev_dbg(&pdev->dev, "use rxtx%d as tx clock source for %dHz sample rate\n", + dev_dbg(dev, "use rxtx%d as tx clock source for %dHz sample rate\n", spdif_priv->txclk_src[index], rate[index]); - dev_dbg(&pdev->dev, "use txclk df %d for %dHz sample rate\n", + dev_dbg(dev, "use txclk df %d for %dHz sample rate\n", spdif_priv->txclk_df[index], rate[index]); if (clk_is_match(spdif_priv->txclk[index], spdif_priv->sysclk)) - dev_dbg(&pdev->dev, "use sysclk df %d for %dHz sample rate\n", + dev_dbg(dev, "use sysclk df %d for %dHz sample rate\n", spdif_priv->sysclk_df[index], rate[index]); - dev_dbg(&pdev->dev, "the best rate for %dHz sample rate is %dHz\n", + dev_dbg(dev, "the best rate for %dHz sample rate is %dHz\n", rate[index], spdif_priv->txrate[index]); return 0; -- 2.20.1.windows.1

3 2

[PATCH] ASoC: qcom: Fix typo error in HDMI regmap config callbacks
by Srinivasa Rao Mandadapu 02 Feb '21

02 Feb '21

Had a typo in lpass platform driver that resulted in crash during suspend/resume with an HDMI dongle connected. The regmap read/write/volatile regesters validation callbacks in lpass-cpu were using MI2S rdma_channels count instead of hdmi_rdma_channels. This typo error causing to read registers from the regmap beyond the length of the mapping created by ioremap(). This fix avoids the need for reducing number hdmi_rdma_channels, which is done in commit 7dfe20ee92f6 ("ASoC: qcom: Fix number of HDMI RDMA channels on sc7180"). So reverting the same. Fixes: 7cb37b7bd0d3c ("ASoC: qcom: Add support for lpass hdmi driver") Signed-off-by: Srinivasa Rao Mandadapu <srivasam(a)codeaurora.org> --- sound/soc/qcom/lpass-cpu.c | 8 ++++---- sound/soc/qcom/lpass-sc7180.c | 2 +- 2 files changed, 5 insertions(+), 5 deletions(-) diff --git a/sound/soc/qcom/lpass-cpu.c b/sound/soc/qcom/lpass-cpu.c index a669202e0001..c642e5f8f28c 100644 --- a/sound/soc/qcom/lpass-cpu.c +++ b/sound/soc/qcom/lpass-cpu.c @@ -595,7 +595,7 @@ static bool lpass_hdmi_regmap_writeable(struct device *dev, unsigned int reg) return true; } - for (i = 0; i < v->rdma_channels; ++i) { + for (i = 0; i < v->hdmi_rdma_channels; ++i) { if (reg == LPAIF_HDMI_RDMACTL_REG(v, i)) return true; if (reg == LPAIF_HDMI_RDMABASE_REG(v, i)) @@ -641,7 +641,7 @@ static bool lpass_hdmi_regmap_readable(struct device *dev, unsigned int reg) if (reg == LPASS_HDMITX_APP_IRQSTAT_REG(v)) return true; - for (i = 0; i < v->rdma_channels; ++i) { + for (i = 0; i < v->hdmi_rdma_channels; ++i) { if (reg == LPAIF_HDMI_RDMACTL_REG(v, i)) return true; if (reg == LPAIF_HDMI_RDMABASE_REG(v, i)) @@ -668,7 +668,7 @@ static bool lpass_hdmi_regmap_volatile(struct device *dev, unsigned int reg) if (reg == LPASS_HDMI_TX_LEGACY_ADDR(v)) return true; - for (i = 0; i < v->rdma_channels; ++i) { + for (i = 0; i < v->hdmi_rdma_channels; ++i) { if (reg == LPAIF_HDMI_RDMACURR_REG(v, i)) return true; } @@ -812,7 +812,7 @@ int asoc_qcom_lpass_cpu_platform_probe(struct platform_device *pdev) return PTR_ERR(drvdata->hdmiif); lpass_hdmi_regmap_config.max_register = LPAIF_HDMI_RDMAPER_REG(variant, - variant->hdmi_rdma_channels); + variant->hdmi_rdma_channels - 1); drvdata->hdmiif_map = devm_regmap_init_mmio(dev, drvdata->hdmiif, &lpass_hdmi_regmap_config); if (IS_ERR(drvdata->hdmiif_map)) { diff --git a/sound/soc/qcom/lpass-sc7180.c b/sound/soc/qcom/lpass-sc7180.c index 735c9dac28f2..8c168d3c589e 100644 --- a/sound/soc/qcom/lpass-sc7180.c +++ b/sound/soc/qcom/lpass-sc7180.c @@ -171,7 +171,7 @@ static struct lpass_variant sc7180_data = { .rdma_channels = 5, .hdmi_rdma_reg_base = 0x64000, .hdmi_rdma_reg_stride = 0x1000, - .hdmi_rdma_channels = 3, + .hdmi_rdma_channels = 4, .dmactl_audif_start = 1, .wrdma_reg_base = 0x18000, .wrdma_reg_stride = 0x1000, -- Qualcomm India Private Limited, on behalf of Qualcomm Innovation Center, Inc., is a member of Code Aurora Forum, a Linux Foundation Collaborative Project.

3 2

[PATCH 0/6] soundwire: qcom: various improvements
by Srinivas Kandagatla 02 Feb '21

02 Feb '21

During testing SoundWire controller on SM8250 MTP, we found few issues like all the interrupts are not handled, all transport parameters are not read from device tree. Other major issue was register read/writes which was interrupt based was an overhead and puts lot of limitation on context it can be used from. So this patchset add various improvements to the existing driver to address above issues. Tested it on SM8250 MTP with 2x WSA881x speakers, HeadPhones on WCD938x via lpass-rx-macro and Analog MICs via lpass-tx-macro. Also tested on DragonBoard DB845c with 2xWSA881x speakers. Srinivas Kandagatla (6): soundwire: qcom: add support to missing transport params soundwire: qcom: extract version field soundwire: qcom: set continue execution flag for ignored commands soundwire: qcom: start the clock during initialization soundwire: qcom: update register read/write routine soundwire: qcom: add support to new interrupts drivers/soundwire/qcom.c | 471 ++++++++++++++++++++++++++++++--------- 1 file changed, 366 insertions(+), 105 deletions(-) -- 2.21.0

3 29

hdspmixer: Hardware output loopback toggle buttons
by GitHub pull_request - edited 02 Feb '21

02 Feb '21

alsa-project/alsa-tools pull request #4 was edited from pfandl: Hi, as promised in the [sound/pci/rme9652 patch](https://github.com/tiwai/sound/commit/da2a040ee7cfe1dd57d5bec7906cb9… here are the user land changes which provide the user the ability to toggle the hardware output loopback functionality per output channel. I tried to hide the buttons for devices which do not expose the snd controls yet, but due to probably unnecessary struggle i cancelled that. Please let me know though should you want them removed for not yet enabled devices. Request URL : https://github.com/alsa-project/alsa-tools/pull/4 Patch URL : https://github.com/alsa-project/alsa-tools/pull/4.patch Repository URL: https://github.com/alsa-project/alsa-tools

1 0

[PATCH] ASoC: mt6359: reduce log verbosity for optional DT properties
by Tzung-Bi Shih 02 Feb '21

02 Feb '21

DT properties "dmic-mode" and "mic-type-X" are optional. Reduces the log verbosity and changes the message a bit to avoid misleading. Signed-off-by: Tzung-Bi Shih <tzungbi(a)google.com> --- Inspired by https://www.spinics.net/lists/arm-kernel/msg872744.html. sound/soc/codecs/mt6359.c | 18 +++++++++++------- 1 file changed, 11 insertions(+), 7 deletions(-) diff --git a/sound/soc/codecs/mt6359.c b/sound/soc/codecs/mt6359.c index 6de0d744fa9e..6f4b1da52082 100644 --- a/sound/soc/codecs/mt6359.c +++ b/sound/soc/codecs/mt6359.c @@ -2754,7 +2754,8 @@ static int mt6359_parse_dt(struct mt6359_priv *priv) ret = of_property_read_u32(np, "mediatek,dmic-mode", &priv->dmic_one_wire_mode); if (ret) { - dev_warn(priv->dev, "%s() failed to read dmic-mode\n", + dev_info(priv->dev, + "%s() failed to read dmic-mode, use default (0)\n", __func__); priv->dmic_one_wire_mode = 0; } @@ -2762,24 +2763,27 @@ static int mt6359_parse_dt(struct mt6359_priv *priv) ret = of_property_read_u32(np, "mediatek,mic-type-0", &priv->mux_select[MUX_MIC_TYPE_0]); if (ret) { - dev_warn(priv->dev, "%s() failed to read mic-type-0\n", - __func__); + dev_info(priv->dev, + "%s() failed to read mic-type-0, use default (%d)\n", + __func__, MIC_TYPE_MUX_IDLE); priv->mux_select[MUX_MIC_TYPE_0] = MIC_TYPE_MUX_IDLE; } ret = of_property_read_u32(np, "mediatek,mic-type-1", &priv->mux_select[MUX_MIC_TYPE_1]); if (ret) { - dev_warn(priv->dev, "%s() failed to read mic-type-1\n", - __func__); + dev_info(priv->dev, + "%s() failed to read mic-type-1, use default (%d)\n", + __func__, MIC_TYPE_MUX_IDLE); priv->mux_select[MUX_MIC_TYPE_1] = MIC_TYPE_MUX_IDLE; } ret = of_property_read_u32(np, "mediatek,mic-type-2", &priv->mux_select[MUX_MIC_TYPE_2]); if (ret) { - dev_warn(priv->dev, "%s() failed to read mic-type-2\n", - __func__); + dev_info(priv->dev, + "%s() failed to read mic-type-2, use default (%d)\n", + __func__, MIC_TYPE_MUX_IDLE); priv->mux_select[MUX_MIC_TYPE_2] = MIC_TYPE_MUX_IDLE; } -- 2.30.0.365.g02bc693789-goog

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[PATCH] ASoC: SOF: add be_hw_params_fixup() for ALH
by Kai Vehmanen 02 Feb '21

02 Feb '21

From: Rander Wang <rander.wang(a)intel.com> Fixup BE DAI links channel count to match topology settings. Normally the channel count of BE is equal to FE's so we don't have any issue. For some cases like DSM with 2-channel FE and 4-channel BE the mismatch of BE and topology will result in audio issues. Signed-off-by: Rander Wang <rander.wang(a)intel.com> Reviewed-by: Keyon Jie <yang.jie(a)intel.com> Signed-off-by: Kai Vehmanen <kai.vehmanen(a)linux.intel.com> --- sound/soc/sof/pcm.c | 7 ++++++- 1 file changed, 6 insertions(+), 1 deletion(-) diff --git a/sound/soc/sof/pcm.c b/sound/soc/sof/pcm.c index 0dc39fbcd81d..61c3fe17342d 100644 --- a/sound/soc/sof/pcm.c +++ b/sound/soc/sof/pcm.c @@ -707,7 +707,12 @@ int sof_pcm_dai_link_fixup(struct snd_soc_pcm_runtime *rtd, struct snd_pcm_hw_pa } break; case SOF_DAI_INTEL_ALH: - /* do nothing for ALH dai_link */ + /* + * Dai could run with different channel count compared with + * front end, so get dai channel count from topology + */ + channels->min = dai->dai_config->alh.channels; + channels->max = dai->dai_config->alh.channels; break; case SOF_DAI_IMX_ESAI: rate->min = dai->dai_config->esai.fsync_rate; base-commit: 8ac9e476b86851c94e0f33bea758e0a00e3f875e -- 2.29.2

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[PATCH AUTOSEL 5.4 07/17] ASoC: ak4458: correct reset polarity
by Sasha Levin 02 Feb '21

02 Feb '21

From: Eliot Blennerhassett <eliot(a)blennerhassett.gen.nz> [ Upstream commit e953daeb68b1abd8a7d44902786349fdeef5c297 ] Reset (aka power off) happens when the reset gpio is made active. Change function name to ak4458_reset to match devicetree property "reset-gpios" Signed-off-by: Eliot Blennerhassett <eliot(a)blennerhassett.gen.nz> Reviewed-by: Linus Walleij <linus.walleij(a)linaro.org> Link: https://lore.kernel.org/r/ce650f47-4ff6-e486-7846-cc3d033f3601@blennerhasse… Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/codecs/ak4458.c | 22 +++++++--------------- 1 file changed, 7 insertions(+), 15 deletions(-) diff --git a/sound/soc/codecs/ak4458.c b/sound/soc/codecs/ak4458.c index 71562154c0b1e..217e8ce9a4ba4 100644 --- a/sound/soc/codecs/ak4458.c +++ b/sound/soc/codecs/ak4458.c @@ -523,18 +523,10 @@ static struct snd_soc_dai_driver ak4497_dai = { .ops = &ak4458_dai_ops, }; -static void ak4458_power_off(struct ak4458_priv *ak4458) +static void ak4458_reset(struct ak4458_priv *ak4458, bool active) { if (ak4458->reset_gpiod) { - gpiod_set_value_cansleep(ak4458->reset_gpiod, 0); - usleep_range(1000, 2000); - } -} - -static void ak4458_power_on(struct ak4458_priv *ak4458) -{ - if (ak4458->reset_gpiod) { - gpiod_set_value_cansleep(ak4458->reset_gpiod, 1); + gpiod_set_value_cansleep(ak4458->reset_gpiod, active); usleep_range(1000, 2000); } } @@ -548,7 +540,7 @@ static int ak4458_init(struct snd_soc_component *component) if (ak4458->mute_gpiod) gpiod_set_value_cansleep(ak4458->mute_gpiod, 1); - ak4458_power_on(ak4458); + ak4458_reset(ak4458, false); ret = snd_soc_component_update_bits(component, AK4458_00_CONTROL1, 0x80, 0x80); /* ACKS bit = 1; 10000000 */ @@ -571,7 +563,7 @@ static void ak4458_remove(struct snd_soc_component *component) { struct ak4458_priv *ak4458 = snd_soc_component_get_drvdata(component); - ak4458_power_off(ak4458); + ak4458_reset(ak4458, true); } #ifdef CONFIG_PM @@ -581,7 +573,7 @@ static int __maybe_unused ak4458_runtime_suspend(struct device *dev) regcache_cache_only(ak4458->regmap, true); - ak4458_power_off(ak4458); + ak4458_reset(ak4458, true); if (ak4458->mute_gpiod) gpiod_set_value_cansleep(ak4458->mute_gpiod, 0); @@ -596,8 +588,8 @@ static int __maybe_unused ak4458_runtime_resume(struct device *dev) if (ak4458->mute_gpiod) gpiod_set_value_cansleep(ak4458->mute_gpiod, 1); - ak4458_power_off(ak4458); - ak4458_power_on(ak4458); + ak4458_reset(ak4458, true); + ak4458_reset(ak4458, false); regcache_cache_only(ak4458->regmap, false); regcache_mark_dirty(ak4458->regmap); -- 2.27.0

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