January 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

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[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 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

[PATCH v9 0/1] audio jack software injection
by Hui Wang 02 Feb '21

02 Feb '21

the changes in the v9: - move the "Phantom" jack comparison into the snd_jack_debugfs_add_inject_node(), then the comparison is only executed when SND_JACK_INJECTION_DEBUG is enabled. the changes in the v8: - change "for (i = 0; i < strlen(tname); i++)" to "for (i = 0; tname[i]; i++)" - use CONFIG_SND_DEBUG to wrap some debugfs variable and struct members. the changes in the v7: - change the format of the last part in jack-injection.rst - add dependence SND_DEBUG in the Kconfig - create debugfs_mount_dir/sound and debugfs_mount_dir/sound/cardN only SND_DEBUG is enabled - change simple_write_to_buffer(buf, count, ppos, from, count) to simple_write_to_buffer(buf, sizeof(buf) - 1, ppos, from, count) - rewrite the function parse_mask_bits() according to v6's comment the changes in the v6: - use the sizeof(buf) to replace the digits in the scnprintf - squash the document patch into the 0001-xxx.patch. the changes in the v5: - add a Kconfig to enable/disable the jack injection feature - replace all kzalloc with local char[] in the debugfs fops - replace the chars of !isalnum() to '_' for the jack folder's name - squash all .c files patches into one patch - add a document to explain jack injection, for easier review, put the document in a standalone patch. If needed, will squash this patch to the previous one. the changes in the v4: - change the sound-core to sound and change the sound_core_debugfs_root to sound_debugfs_root in the 0001-xxx.patch - change kzalloc/kfree to kvzalloc/kvfree in the debugfs fops for 0001-xxx.patch and 0003-xxx.patch - And if needed, I could squash 4 patches into 1 patch before merging. the changes in the v3 (for easy to review, divide change into 4 patches): - address the comment about the snd_jack_report() and _snd_jack_report(), the v2 design is hard to understand and is hard to review, in the v3, separate the jack_report to snd_jack_report() and snd_jack_inject_report(), hw jack events call snd_jack_report() as before, if a jack contains multi jack_kctl and the jack_kctl's sw_inject is enabled, the status and the related input-dev's events will not be reproted. The injection events call snd_jack_inject_report(). This change is squashed to 0001-xxx.patch, the rest part of 0001-xxx.patch is same as v2. - address the comment about folders'name in the 0002-xxx.patch, so far, drop the '/', ',', '=' and ' ' from the folders' name. - address the comment about adding more debugfs nodes in the 0003-xxx.patch, it adds kctl_id, mask_bits, status and type. - address the comment about save-n-restore jack's hw status in the 0004-xxx.patch, adding a hw_status_cache and save the last reported jack hw event, once the sw_inject is disabled, will restore all jack_kctl's state under the same snd_jack with hw_status_cache. [snip] the changes in the V2: - using debugfs instead of sysfs - using jack_ctrl to create a folder instead of snd_jack, since ASoC drivers could create multi jack_ctrls within a snd_jack - create a folder for each jack_ctrl instead for all jack_ctrls [ snip ] Hui Wang (1): alsa: jack: implement software jack injection via debugfs Documentation/sound/designs/index.rst | 1 + .../sound/designs/jack-injection.rst | 166 ++++++++++ include/sound/core.h | 6 + include/sound/jack.h | 1 + sound/core/Kconfig | 9 + sound/core/init.c | 16 + sound/core/jack.c | 304 +++++++++++++++++- sound/core/sound.c | 13 + 8 files changed, 512 insertions(+), 4 deletions(-) create mode 100644 Documentation/sound/designs/jack-injection.rst -- 2.25.1

2 3

[PATCH] sound: pci/rme9652 - implement and expose controls for output
by fassl 02 Feb '21

02 Feb '21

>From 980aa253c17d233966e05a43f3611693e2e02040 Mon Sep 17 00:00:00 2001 From: Jasmin Fazlic <superfassl(a)gmail.com> Date: Sun, 31 Jan 2021 22:17:22 +0100 Subject: [PATCH] sound: pci/rme9652 - implement and expose controls for output loopback - so far only tested and enabled for RME HDSP9632 Signed-off-by: Jasmin Fazlic <superfassl(a)gmail.com> --- sound/pci/rme9652/hdsp.c | 85 +++++++++++++++++++++++++++++++++++++++- 1 file changed, 84 insertions(+), 1 deletion(-) diff --git a/sound/pci/rme9652/hdsp.c b/sound/pci/rme9652/hdsp.c index cea53a878c36..7e832a502cea 100644 --- a/sound/pci/rme9652/hdsp.c +++ b/sound/pci/rme9652/hdsp.c @@ -469,6 +469,7 @@ struct hdsp { unsigned char qs_out_channels; unsigned char ds_out_channels; unsigned char ss_out_channels; + u32 io_loopback; /* output loopback channel states*/ struct snd_dma_buffer capture_dma_buf; struct snd_dma_buffer playback_dma_buf; @@ -3253,6 +3254,71 @@ static const struct snd_kcontrol_new snd_hdsp_96xx_aeb = HDSP_AnalogExtensionBoard); static struct snd_kcontrol_new snd_hdsp_adat_sync_check = HDSP_ADAT_SYNC_CHECK; + +static bool hdsp_loopback_get(struct hdsp *const hdsp, const u8 channel) +{ + return hdsp->io_loopback & (1 << channel); +} + +static int hdsp_loopback_set(struct hdsp *const hdsp, const u8 channel, const bool enable) +{ + if (hdsp_loopback_get(hdsp, channel) == enable) + return 0; + + hdsp->io_loopback ^= (1 << channel); + + hdsp_write(hdsp, HDSP_inputEnable + (4 * (hdsp->max_channels + channel)), enable); + + return 1; +} + +static int snd_hdsp_loopback_info(struct snd_kcontrol *const kcontrol, + struct snd_ctl_elem_info *const uinfo) +{ + uinfo->type = SNDRV_CTL_ELEM_TYPE_BOOLEAN; + uinfo->count = 1; + uinfo->value.integer.min = 0; + uinfo->value.integer.max = 1; + return 0; +} + +static int snd_hdsp_loopback_get(struct snd_kcontrol *const kcontrol, + struct snd_ctl_elem_value *const ucontrol) +{ + struct hdsp *const hdsp = snd_kcontrol_chip(kcontrol); + const u8 channel = kcontrol->id.index; + + if (channel >= hdsp->max_channels) + return -ENOENT; + + ucontrol->value.integer.value[0] = hdsp_loopback_get(hdsp, channel); + + return 0; +} + +static int snd_hdsp_loopback_put(struct snd_kcontrol *const kcontrol, + struct snd_ctl_elem_value *const ucontrol) +{ + struct hdsp *const hdsp = snd_kcontrol_chip(kcontrol); + const u8 channel = ucontrol->id.index; + const bool enable = ucontrol->value.integer.value[0] & 1; + + if (channel >= hdsp->max_channels) + return -ENOENT; + + return hdsp_loopback_set(hdsp, channel, enable); +} + +static struct snd_kcontrol_new snd_hdsp_loopback_control = { + .iface = SNDRV_CTL_ELEM_IFACE_HWDEP, + .name = "Output Loopback", + .index = 0, + .access = SNDRV_CTL_ELEM_ACCESS_READWRITE, + .info = snd_hdsp_loopback_info, + .get = snd_hdsp_loopback_get, + .put = snd_hdsp_loopback_put +}; + static int snd_hdsp_create_controls(struct snd_card *card, struct hdsp *hdsp) { unsigned int idx; @@ -3297,6 +3363,17 @@ static int snd_hdsp_create_controls(struct snd_card *card, struct hdsp *hdsp) } } + /* Output loopback controls for H9632 cards */ + if (hdsp->io_type == H9632) { + for (idx = 0; idx < hdsp->max_channels; idx++) { + snd_hdsp_loopback_control.index = idx; + kctl = snd_ctl_new1(&snd_hdsp_loopback_control, hdsp); + err = snd_ctl_add(card, kctl); + if (err < 0) + return err; + } + } + /* AEB control for H96xx card */ if (hdsp->io_type == H9632 || hdsp->io_type == H9652) { if ((err = snd_ctl_add(card, kctl = snd_ctl_new1(&snd_hdsp_96xx_aeb, hdsp))) < 0) @@ -4956,7 +5033,7 @@ static int snd_hdsp_enable_io (struct hdsp *hdsp) static void snd_hdsp_initialize_channels(struct hdsp *hdsp) { - int status, aebi_channels, aebo_channels; + int status, aebi_channels, aebo_channels, i; switch (hdsp->io_type) { case Digiface: @@ -4983,6 +5060,12 @@ static void snd_hdsp_initialize_channels(struct hdsp *hdsp) hdsp->ss_out_channels = H9632_SS_CHANNELS+aebo_channels; hdsp->ds_out_channels = H9632_DS_CHANNELS+aebo_channels; hdsp->qs_out_channels = H9632_QS_CHANNELS+aebo_channels; + /* Disable loopback of output channels, as the set function + * only sets on a change we fake all bits (channels) as enabled. + */ + hdsp->io_loopback = 0xffffffff; + for (i = 0; i < hdsp->max_channels; ++i) + hdsp_loopback_set(hdsp, i, false); break; case Multiface: -- 2.27.0

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[PATCH] ASoC: tegra: SND_SOC_TEGRA_AUDIO_GRAPH_CARD should depend on SND_SOC_TEGRA
by Geert Uytterhoeven 01 Feb '21

01 Feb '21

Audio Graph Card based Tegra driver is only useful on NVIDIA Tegra SoCs. Hence add a dependency on SND_SOC_TEGRA, to prevent asking the user about this driver when configuring a kernel without Tegra sound support. Wrap all Tegra sound config options inside a big if/endif block, instead of just adding the dependency to the single config option that does not have it yet, to preventing similar future mistakes. Fixes: 202e2f7745437aa5 ("ASoC: tegra: Add audio graph based card driver") Signed-off-by: Geert Uytterhoeven <geert+renesas(a)glider.be> --- sound/soc/tegra/Kconfig | 33 +++++++++++++-------------------- 1 file changed, 13 insertions(+), 20 deletions(-) diff --git a/sound/soc/tegra/Kconfig b/sound/soc/tegra/Kconfig index 6dc83ad8607a1de2..a4e6760944d028cf 100644 --- a/sound/soc/tegra/Kconfig +++ b/sound/soc/tegra/Kconfig @@ -9,9 +9,10 @@ config SND_SOC_TEGRA help Say Y or M here if you want support for SoC audio on Tegra. +if SND_SOC_TEGRA + config SND_SOC_TEGRA20_AC97 tristate "Tegra20 AC97 interface" - depends on SND_SOC_TEGRA select SND_SOC_AC97_BUS select SND_SOC_TEGRA20_DAS help @@ -21,7 +22,6 @@ config SND_SOC_TEGRA20_AC97 config SND_SOC_TEGRA20_DAS tristate "Tegra20 DAS module" - depends on SND_SOC_TEGRA help Say Y or M if you want to add support for the Tegra20 DAS module. You will also need to select the individual machine drivers to @@ -29,7 +29,6 @@ config SND_SOC_TEGRA20_DAS config SND_SOC_TEGRA20_I2S tristate "Tegra20 I2S interface" - depends on SND_SOC_TEGRA select SND_SOC_TEGRA20_DAS help Say Y or M if you want to add support for codecs attached to the @@ -38,7 +37,6 @@ config SND_SOC_TEGRA20_I2S config SND_SOC_TEGRA20_SPDIF tristate "Tegra20 SPDIF interface" - depends on SND_SOC_TEGRA help Say Y or M if you want to add support for the Tegra20 SPDIF interface. You will also need to select the individual machine drivers to support @@ -46,7 +44,6 @@ config SND_SOC_TEGRA20_SPDIF config SND_SOC_TEGRA30_AHUB tristate "Tegra30 AHUB module" - depends on SND_SOC_TEGRA help Say Y or M if you want to add support for the Tegra30 AHUB module. You will also need to select the individual machine drivers to @@ -54,7 +51,6 @@ config SND_SOC_TEGRA30_AHUB config SND_SOC_TEGRA30_I2S tristate "Tegra30 I2S interface" - depends on SND_SOC_TEGRA select SND_SOC_TEGRA30_AHUB help Say Y or M if you want to add support for codecs attached to the @@ -63,7 +59,6 @@ config SND_SOC_TEGRA30_I2S config SND_SOC_TEGRA210_AHUB tristate "Tegra210 AHUB module" - depends on SND_SOC_TEGRA help Config to enable Audio Hub (AHUB) module, which comprises of a switch called Audio Crossbar (AXBAR) used to configure or modify @@ -73,7 +68,6 @@ config SND_SOC_TEGRA210_AHUB config SND_SOC_TEGRA210_DMIC tristate "Tegra210 DMIC module" - depends on SND_SOC_TEGRA help Config to enable the Digital MIC (DMIC) controller which is used to interface with Pulse Density Modulation (PDM) input devices. @@ -84,7 +78,6 @@ config SND_SOC_TEGRA210_DMIC config SND_SOC_TEGRA210_I2S tristate "Tegra210 I2S module" - depends on SND_SOC_TEGRA help Config to enable the Inter-IC Sound (I2S) Controller which implements full-duplex and bidirectional and single direction @@ -94,7 +87,6 @@ config SND_SOC_TEGRA210_I2S config SND_SOC_TEGRA186_DSPK tristate "Tegra186 DSPK module" - depends on SND_SOC_TEGRA help Config to enable the Digital Speaker Controller (DSPK) which converts the multi-bit Pulse Code Modulation (PCM) audio input to @@ -107,7 +99,6 @@ config SND_SOC_TEGRA186_DSPK config SND_SOC_TEGRA210_ADMAIF tristate "Tegra210 ADMAIF module" - depends on SND_SOC_TEGRA help Config to enable ADMAIF which is the interface between ADMA and Audio Hub (AHUB). Each ADMA channel that sends/receives data to/ @@ -128,7 +119,7 @@ config SND_SOC_TEGRA_AUDIO_GRAPH_CARD config SND_SOC_TEGRA_RT5640 tristate "SoC Audio support for Tegra boards using an RT5640 codec" - depends on SND_SOC_TEGRA && I2C && GPIOLIB + depends on I2C && GPIOLIB select SND_SOC_RT5640 help Say Y or M here if you want to add support for SoC audio on Tegra @@ -136,7 +127,7 @@ config SND_SOC_TEGRA_RT5640 config SND_SOC_TEGRA_WM8753 tristate "SoC Audio support for Tegra boards using a WM8753 codec" - depends on SND_SOC_TEGRA && I2C && GPIOLIB + depends on I2C && GPIOLIB select SND_SOC_WM8753 help Say Y or M here if you want to add support for SoC audio on Tegra @@ -144,7 +135,7 @@ config SND_SOC_TEGRA_WM8753 config SND_SOC_TEGRA_WM8903 tristate "SoC Audio support for Tegra boards using a WM8903 codec" - depends on SND_SOC_TEGRA && I2C && GPIOLIB + depends on I2C && GPIOLIB select SND_SOC_WM8903 help Say Y or M here if you want to add support for SoC audio on Tegra @@ -153,7 +144,7 @@ config SND_SOC_TEGRA_WM8903 config SND_SOC_TEGRA_WM9712 tristate "SoC Audio support for Tegra boards using a WM9712 codec" - depends on SND_SOC_TEGRA && GPIOLIB + depends on GPIOLIB select SND_SOC_TEGRA20_AC97 select SND_SOC_WM9712 help @@ -162,7 +153,7 @@ config SND_SOC_TEGRA_WM9712 config SND_SOC_TEGRA_TRIMSLICE tristate "SoC Audio support for TrimSlice board" - depends on SND_SOC_TEGRA && I2C + depends on I2C select SND_SOC_TLV320AIC23_I2C help Say Y or M here if you want to add support for SoC audio on the @@ -170,7 +161,7 @@ config SND_SOC_TEGRA_TRIMSLICE config SND_SOC_TEGRA_ALC5632 tristate "SoC Audio support for Tegra boards using an ALC5632 codec" - depends on SND_SOC_TEGRA && I2C && GPIOLIB + depends on I2C && GPIOLIB select SND_SOC_ALC5632 help Say Y or M here if you want to add support for SoC audio on the @@ -178,7 +169,7 @@ config SND_SOC_TEGRA_ALC5632 config SND_SOC_TEGRA_MAX98090 tristate "SoC Audio support for Tegra boards using a MAX98090 codec" - depends on SND_SOC_TEGRA && I2C && GPIOLIB + depends on I2C && GPIOLIB select SND_SOC_MAX98090 help Say Y or M here if you want to add support for SoC audio on Tegra @@ -186,7 +177,7 @@ config SND_SOC_TEGRA_MAX98090 config SND_SOC_TEGRA_RT5677 tristate "SoC Audio support for Tegra boards using a RT5677 codec" - depends on SND_SOC_TEGRA && I2C && GPIOLIB + depends on I2C && GPIOLIB select SND_SOC_RT5677 help Say Y or M here if you want to add support for SoC audio on Tegra @@ -194,9 +185,11 @@ config SND_SOC_TEGRA_RT5677 config SND_SOC_TEGRA_SGTL5000 tristate "SoC Audio support for Tegra boards using a SGTL5000 codec" - depends on SND_SOC_TEGRA && I2C && GPIOLIB + depends on I2C && GPIOLIB select SND_SOC_SGTL5000 help Say Y or M here if you want to add support for SoC audio on Tegra boards using the SGTL5000 codec, such as Apalis T30, Apalis TK1 or Colibri T30. + +endif -- 2.25.1

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[bug report] ASoC: cpcap: new codec
by Dan Carpenter 01 Feb '21

01 Feb '21

Hello Sebastian Reichel, The patch f6cdf2d3445d: "ASoC: cpcap: new codec" from Feb 23, 2018, leads to the following static checker warning: sound/soc/codecs/cpcap.c:1276 cpcap_voice_hw_params() warn: 'CPCAP_BIT_MIC1_RX_TIMESLOT0' is a shifter (not for '|='). sound/soc/codecs/cpcap.c:1279 cpcap_voice_hw_params() warn: 'CPCAP_BIT_MIC2_TIMESLOT0' is a shifter (not for '|='). sound/soc/codecs/cpcap.c 1253 static int cpcap_voice_hw_params(struct snd_pcm_substream *substream, 1254 struct snd_pcm_hw_params *params, 1255 struct snd_soc_dai *dai) 1256 { 1257 struct snd_soc_component *component = dai->component; 1258 struct device *dev = component->dev; 1259 struct cpcap_audio *cpcap = snd_soc_component_get_drvdata(component); 1260 static const u16 reg_cdi = CPCAP_REG_CDI; 1261 int rate = params_rate(params); 1262 int channels = params_channels(params); 1263 int direction = substream->stream; 1264 u16 val, mask; 1265 int err; 1266 1267 dev_dbg(dev, "Voice setup HW params: rate=%d, direction=%d, chan=%d", 1268 rate, direction, channels); 1269 1270 err = cpcap_set_samprate(cpcap, CPCAP_DAI_VOICE, rate); 1271 if (err) 1272 return err; 1273 1274 if (direction == SNDRV_PCM_STREAM_CAPTURE) { 1275 mask = 0x0000; 1276 mask |= CPCAP_BIT_MIC1_RX_TIMESLOT0; ^^^^^^^^^^^^^^^^^^^^^^^^^^^ This should probably be BIT(CPCAP_BIT_MIC1_RX_TIMESLOT0). 1277 mask |= CPCAP_BIT_MIC1_RX_TIMESLOT1; 1278 mask |= CPCAP_BIT_MIC1_RX_TIMESLOT2; 1279 mask |= CPCAP_BIT_MIC2_TIMESLOT0; ^^^^^^^^^^^^^^^^^^^^^^^^ Same for all the others as well I guess. 1280 mask |= CPCAP_BIT_MIC2_TIMESLOT1; 1281 mask |= CPCAP_BIT_MIC2_TIMESLOT2; 1282 val = 0x0000; 1283 if (channels >= 2) 1284 val = BIT(CPCAP_BIT_MIC1_RX_TIMESLOT0); ^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^ Here it's BIT(CPCAP_BIT_MIC1_RX_TIMESLOT0). 1285 err = regmap_update_bits(cpcap->regmap, reg_cdi, mask, val); 1286 if (err) 1287 return err; 1288 } 1289 1290 return 0; 1291 } regards, dan carpenter

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[PATCH v2] ASoC: qcom: lpass: Fix i2s ctl register bit map
by Jun Nie 01 Feb '21

01 Feb '21

Fix bitwidth mapping in i2s ctl register per APQ8016 document. Fixes: b5022a36d28f ("ASoC: qcom: lpass: Use regmap_field for i2sctl and dmactl registers") Signed-off-by: Jun Nie <jun.nie(a)linaro.org> --- sound/soc/qcom/lpass-apq8016.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/qcom/lpass-apq8016.c b/sound/soc/qcom/lpass-apq8016.c index 8507ef8f6679..3efa133d1c64 100644 --- a/sound/soc/qcom/lpass-apq8016.c +++ b/sound/soc/qcom/lpass-apq8016.c @@ -250,7 +250,7 @@ static struct lpass_variant apq8016_data = { .micmode = REG_FIELD_ID(0x1000, 4, 7, 4, 0x1000), .micmono = REG_FIELD_ID(0x1000, 3, 3, 4, 0x1000), .wssrc = REG_FIELD_ID(0x1000, 2, 2, 4, 0x1000), - .bitwidth = REG_FIELD_ID(0x1000, 0, 0, 4, 0x1000), + .bitwidth = REG_FIELD_ID(0x1000, 0, 1, 4, 0x1000), .rdma_dyncclk = REG_FIELD_ID(0x8400, 12, 12, 2, 0x1000), .rdma_bursten = REG_FIELD_ID(0x8400, 11, 11, 2, 0x1000), -- 2.25.1

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[PATCH] soundwire: return earlier if no slave is attached
by Bard Liao 01 Feb '21

01 Feb '21

From: Chao Song <chao.song(a)linux.intel.com> If there is no slave attached to soundwire bus, we can return earlier from sdw_bus_prep_clk_stop() and sdw_bus_exit_clk_stop(), this saves a redundant value check. Signed-off-by: Chao Song <chao.song(a)linux.intel.com> Reviewed-by: Rander Wang <rander.wang(a)intel.com> Reviewed-by: Guennadi Liakhovetski <guennadi.liakhovetski(a)linux.intel.com> Reviewed-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> Signed-off-by: Bard Liao <yung-chuan.liao(a)linux.intel.com> --- drivers/soundwire/bus.c | 23 ++++++++++------------- 1 file changed, 10 insertions(+), 13 deletions(-) diff --git a/drivers/soundwire/bus.c b/drivers/soundwire/bus.c index 82df088c9333..d9deafdcf495 100644 --- a/drivers/soundwire/bus.c +++ b/drivers/soundwire/bus.c @@ -951,17 +951,17 @@ int sdw_bus_prep_clk_stop(struct sdw_bus *bus) simple_clk_stop = false; } - if (is_slave && !simple_clk_stop) { + /* Skip remaining clock stop preparation if no Slave is attached */ + if (!is_slave) + return ret; + + if (!simple_clk_stop) { ret = sdw_bus_wait_for_clk_prep_deprep(bus, SDW_BROADCAST_DEV_NUM); if (ret < 0) return ret; } - /* Don't need to inform slaves if there is no slave attached */ - if (!is_slave) - return ret; - /* Inform slaves that prep is done */ list_for_each_entry(slave, &bus->slaves, node) { if (!slave->dev_num) @@ -1075,16 +1075,13 @@ int sdw_bus_exit_clk_stop(struct sdw_bus *bus) "clk stop deprep failed:%d", ret); } - if (is_slave && !simple_clk_stop) + /* Skip remaining clock stop de-preparation if no Slave is attached */ + if (!is_slave) + return 0; + + if (!simple_clk_stop) sdw_bus_wait_for_clk_prep_deprep(bus, SDW_BROADCAST_DEV_NUM); - /* - * Don't need to call slave callback function if there is no slave - * attached - */ - if (!is_slave) - return 0; - list_for_each_entry(slave, &bus->slaves, node) { if (!slave->dev_num) continue; -- 2.17.1

2 1

[PATCH] soundwire: bus: add better dev_dbg to track complete() calls
by Bard Liao 01 Feb '21

01 Feb '21

From: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> Add a dev_dbg() log for both enumeration and initialization completion to better track suspend-resume issues. Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> Reviewed-by: Rander Wang <rander.wang(a)intel.com> Reviewed-by: Chao Song <chao.song(a)intel.com> Signed-off-by: Bard Liao <yung-chuan.liao(a)linux.intel.com> --- drivers/soundwire/bus.c | 11 ++++++++--- 1 file changed, 8 insertions(+), 3 deletions(-) diff --git a/drivers/soundwire/bus.c b/drivers/soundwire/bus.c index 6e1c988f3845..82df088c9333 100644 --- a/drivers/soundwire/bus.c +++ b/drivers/soundwire/bus.c @@ -784,7 +784,7 @@ static void sdw_modify_slave_status(struct sdw_slave *slave, if (status == SDW_SLAVE_UNATTACHED) { dev_dbg(&slave->dev, - "%s: initializing completion for Slave %d\n", + "%s: initializing enumeration and init completion for Slave %d\n", __func__, slave->dev_num); init_completion(&slave->enumeration_complete); @@ -793,7 +793,7 @@ static void sdw_modify_slave_status(struct sdw_slave *slave, } else if ((status == SDW_SLAVE_ATTACHED) && (slave->status == SDW_SLAVE_UNATTACHED)) { dev_dbg(&slave->dev, - "%s: signaling completion for Slave %d\n", + "%s: signaling enumeration completion for Slave %d\n", __func__, slave->dev_num); complete(&slave->enumeration_complete); @@ -1758,8 +1758,13 @@ int sdw_handle_slave_status(struct sdw_bus *bus, if (ret) dev_err(slave->bus->dev, "Update Slave status failed:%d\n", ret); - if (attached_initializing) + if (attached_initializing) { + dev_dbg(&slave->dev, + "%s: signaling initialization completion for Slave %d\n", + __func__, slave->dev_num); + complete(&slave->initialization_complete); + } } return ret; -- 2.17.1

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