September 2012 - Alsa-devel - mailman.alsa-project.org

[alsa-devel] [PATCH v2 1/3] ASoC: fsl: add PPC_MPC52xx dependency to SND_POWERPC_SOC
by Eric Millbrandt 22 Sep '12

22 Sep '12

mpc52xx socs do not define FSL_SOC but need SND_POWERPC_SOC defined to build ASoC drivers. Signed-off-by: Eric Millbrandt <emillbrandt(a)dekaresearch.com> --- Changes since v1: Patch was "powerpc/52xx: define FSL_SOC" Changed from defining FSL_SOC for PPC_MPC52xx to adding PPC_MPC52xx as a dependency of SND_POWERPC_SOC as per Anatolij Gustschin's comment. sound/soc/fsl/Kconfig | 2 +- 1 files changed, 1 insertions(+), 1 deletions(-) diff --git a/sound/soc/fsl/Kconfig b/sound/soc/fsl/Kconfig index d701330..4563b28 100644 --- a/sound/soc/fsl/Kconfig +++ b/sound/soc/fsl/Kconfig @@ -6,7 +6,7 @@ config SND_SOC_FSL_UTILS menuconfig SND_POWERPC_SOC tristate "SoC Audio for Freescale PowerPC CPUs" - depends on FSL_SOC + depends on FSL_SOC || PPC_MPC52xx help Say Y or M if you want to add support for codecs attached to the PowerPC CPUs. -- 1.7.2.5

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[alsa-devel] [PATCH] ASoC: twl6040: Convert to use DAI DAPM widgets
by Peter Ujfalusi 22 Sep '12

22 Sep '12

Use DAPM mapping for stream events and give unique names for the streams. This change also fixes the following warning: twl6040-codec twl6040-codec: Failed to create Capture debugfs file Signed-off-by: Peter Ujfalusi <peter.ujfalusi(a)ti.com> --- sound/soc/codecs/twl6040.c | 43 +++++++++++++++++++++++++++++-------------- 1 file changed, 29 insertions(+), 14 deletions(-) diff --git a/sound/soc/codecs/twl6040.c b/sound/soc/codecs/twl6040.c index c084c54..e8f97af 100644 --- a/sound/soc/codecs/twl6040.c +++ b/sound/soc/codecs/twl6040.c @@ -727,10 +727,8 @@ static const struct snd_soc_dapm_widget twl6040_dapm_widgets[] = { TWL6040_REG_MICRCTL, 1, 0, NULL, 0), /* ADCs */ - SND_SOC_DAPM_ADC("ADC Left", "Left Front Capture", - TWL6040_REG_MICLCTL, 2, 0), - SND_SOC_DAPM_ADC("ADC Right", "Right Front Capture", - TWL6040_REG_MICRCTL, 2, 0), + SND_SOC_DAPM_ADC("ADC Left", NULL, TWL6040_REG_MICLCTL, 2, 0), + SND_SOC_DAPM_ADC("ADC Right", NULL, TWL6040_REG_MICRCTL, 2, 0), /* Microphone bias */ SND_SOC_DAPM_SUPPLY("Headset Mic Bias", @@ -743,15 +741,12 @@ static const struct snd_soc_dapm_widget twl6040_dapm_widgets[] = { TWL6040_REG_DMICBCTL, 4, 0, NULL, 0), /* DACs */ - SND_SOC_DAPM_DAC("HSDAC Left", "Headset Playback", SND_SOC_NOPM, 0, 0), - SND_SOC_DAPM_DAC("HSDAC Right", "Headset Playback", SND_SOC_NOPM, 0, 0), - SND_SOC_DAPM_DAC("HFDAC Left", "Handsfree Playback", - TWL6040_REG_HFLCTL, 0, 0), - SND_SOC_DAPM_DAC("HFDAC Right", "Handsfree Playback", - TWL6040_REG_HFRCTL, 0, 0), + SND_SOC_DAPM_DAC("HSDAC Left", NULL, SND_SOC_NOPM, 0, 0), + SND_SOC_DAPM_DAC("HSDAC Right", NULL, SND_SOC_NOPM, 0, 0), + SND_SOC_DAPM_DAC("HFDAC Left", NULL, TWL6040_REG_HFLCTL, 0, 0), + SND_SOC_DAPM_DAC("HFDAC Right", NULL, TWL6040_REG_HFRCTL, 0, 0), /* Virtual DAC for vibra path (DL4 channel) */ - SND_SOC_DAPM_DAC("VIBRA DAC", "Vibra Playback", - SND_SOC_NOPM, 0, 0), + SND_SOC_DAPM_DAC("VIBRA DAC", NULL, SND_SOC_NOPM, 0, 0), SND_SOC_DAPM_MUX("Handsfree Left Playback", SND_SOC_NOPM, 0, 0, &hfl_mux_controls), @@ -810,6 +805,26 @@ static const struct snd_soc_dapm_widget twl6040_dapm_widgets[] = { }; static const struct snd_soc_dapm_route intercon[] = { + /* Stream -> DAC mapping */ + {"HSDAC Left", NULL, "Legacy Playback"}, + {"HSDAC Left", NULL, "Headset Playback"}, + {"HSDAC Right", NULL, "Legacy Playback"}, + {"HSDAC Right", NULL, "Headset Playback"}, + + {"HFDAC Left", NULL, "Legacy Playback"}, + {"HFDAC Left", NULL, "Handsfree Playback"}, + {"HFDAC Right", NULL, "Legacy Playback"}, + {"HFDAC Right", NULL, "Handsfree Playback"}, + + {"VIBRA DAC", NULL, "Legacy Playback"}, + {"VIBRA DAC", NULL, "Vibra Playback"}, + + /* ADC -> Stream mapping */ + {"ADC Left", NULL, "Legacy Capture"}, + {"ADC Left", NULL, "Capture"}, + {"ADC Right", NULL, "Legacy Capture"}, + {"ADC Right", NULL, "Capture"}, + /* Capture path */ {"Analog Left Capture Route", "Headset Mic", "HSMIC"}, {"Analog Left Capture Route", "Main Mic", "MAINMIC"}, @@ -1028,14 +1043,14 @@ static struct snd_soc_dai_driver twl6040_dai[] = { { .name = "twl6040-legacy", .playback = { - .stream_name = "Playback", + .stream_name = "Legacy Playback", .channels_min = 1, .channels_max = 5, .rates = TWL6040_RATES, .formats = TWL6040_FORMATS, }, .capture = { - .stream_name = "Capture", + .stream_name = "Legacy Capture", .channels_min = 1, .channels_max = 2, .rates = TWL6040_RATES, -- 1.7.12

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[alsa-devel] [PATCH] ASoC: twl4030: Convert to use DAI DAPM widgets
by Peter Ujfalusi 22 Sep '12

22 Sep '12

Use DAPM mapping for stream events and give unique names for the streams. This change also fixes the following warning: twl4030-codec twl4030-codec: Failed to create Capture debugfs file Signed-off-by: Peter Ujfalusi <peter.ujfalusi(a)ti.com> --- sound/soc/codecs/twl4030.c | 48 +++++++++++++++++++++++++++------------------- 1 file changed, 28 insertions(+), 20 deletions(-) diff --git a/sound/soc/codecs/twl4030.c b/sound/soc/codecs/twl4030.c index 2548f5c..962341d 100644 --- a/sound/soc/codecs/twl4030.c +++ b/sound/soc/codecs/twl4030.c @@ -1237,16 +1237,11 @@ static const struct snd_soc_dapm_widget twl4030_dapm_widgets[] = { SND_SOC_DAPM_OUTPUT("Virtual Voice OUT"), /* DACs */ - SND_SOC_DAPM_DAC("DAC Right1", "Right Front HiFi Playback", - SND_SOC_NOPM, 0, 0), - SND_SOC_DAPM_DAC("DAC Left1", "Left Front HiFi Playback", - SND_SOC_NOPM, 0, 0), - SND_SOC_DAPM_DAC("DAC Right2", "Right Rear HiFi Playback", - SND_SOC_NOPM, 0, 0), - SND_SOC_DAPM_DAC("DAC Left2", "Left Rear HiFi Playback", - SND_SOC_NOPM, 0, 0), - SND_SOC_DAPM_DAC("DAC Voice", "Voice Playback", - SND_SOC_NOPM, 0, 0), + SND_SOC_DAPM_DAC("DAC Right1", NULL, SND_SOC_NOPM, 0, 0), + SND_SOC_DAPM_DAC("DAC Left1", NULL, SND_SOC_NOPM, 0, 0), + SND_SOC_DAPM_DAC("DAC Right2", NULL, SND_SOC_NOPM, 0, 0), + SND_SOC_DAPM_DAC("DAC Left2", NULL, SND_SOC_NOPM, 0, 0), + SND_SOC_DAPM_DAC("DAC Voice", NULL, SND_SOC_NOPM, 0, 0), /* Analog bypasses */ SND_SOC_DAPM_SWITCH("Right1 Analog Loopback", SND_SOC_NOPM, 0, 0, @@ -1375,14 +1370,10 @@ static const struct snd_soc_dapm_widget twl4030_dapm_widgets[] = { /* Introducing four virtual ADC, since TWL4030 have four channel for capture */ - SND_SOC_DAPM_ADC("ADC Virtual Left1", "Left Front Capture", - SND_SOC_NOPM, 0, 0), - SND_SOC_DAPM_ADC("ADC Virtual Right1", "Right Front Capture", - SND_SOC_NOPM, 0, 0), - SND_SOC_DAPM_ADC("ADC Virtual Left2", "Left Rear Capture", - SND_SOC_NOPM, 0, 0), - SND_SOC_DAPM_ADC("ADC Virtual Right2", "Right Rear Capture", - SND_SOC_NOPM, 0, 0), + SND_SOC_DAPM_ADC("ADC Virtual Left1", NULL, SND_SOC_NOPM, 0, 0), + SND_SOC_DAPM_ADC("ADC Virtual Right1", NULL, SND_SOC_NOPM, 0, 0), + SND_SOC_DAPM_ADC("ADC Virtual Left2", NULL, SND_SOC_NOPM, 0, 0), + SND_SOC_DAPM_ADC("ADC Virtual Right2", NULL, SND_SOC_NOPM, 0, 0), /* Analog/Digital mic path selection. TX1 Left/Right: either analog Left/Right or Digimic0 @@ -1426,6 +1417,23 @@ static const struct snd_soc_dapm_widget twl4030_dapm_widgets[] = { }; static const struct snd_soc_dapm_route intercon[] = { + /* Stream -> DAC mapping */ + {"DAC Right1", NULL, "HiFi Playback"}, + {"DAC Left1", NULL, "HiFi Playback"}, + {"DAC Right2", NULL, "HiFi Playback"}, + {"DAC Left2", NULL, "HiFi Playback"}, + {"DAC Voice", NULL, "Voice Playback"}, + + /* ADC -> Stream mapping */ + {"HiFi Capture", NULL, "ADC Virtual Left1"}, + {"HiFi Capture", NULL, "ADC Virtual Right1"}, + {"HiFi Capture", NULL, "ADC Virtual Left2"}, + {"HiFi Capture", NULL, "ADC Virtual Right2"}, + {"Voice Capture", NULL, "ADC Virtual Left1"}, + {"Voice Capture", NULL, "ADC Virtual Right1"}, + {"Voice Capture", NULL, "ADC Virtual Left2"}, + {"Voice Capture", NULL, "ADC Virtual Right2"}, + {"Digital L1 Playback Mixer", NULL, "DAC Left1"}, {"Digital R1 Playback Mixer", NULL, "DAC Right1"}, {"Digital L2 Playback Mixer", NULL, "DAC Left2"}, @@ -2170,7 +2178,7 @@ static struct snd_soc_dai_driver twl4030_dai[] = { .formats = TWL4030_FORMATS, .sig_bits = 24,}, .capture = { - .stream_name = "Capture", + .stream_name = "HiFi Capture", .channels_min = 2, .channels_max = 4, .rates = TWL4030_RATES, @@ -2187,7 +2195,7 @@ static struct snd_soc_dai_driver twl4030_dai[] = { .rates = SNDRV_PCM_RATE_8000 | SNDRV_PCM_RATE_16000, .formats = SNDRV_PCM_FMTBIT_S16_LE,}, .capture = { - .stream_name = "Capture", + .stream_name = "Voice Capture", .channels_min = 1, .channels_max = 2, .rates = SNDRV_PCM_RATE_8000 | SNDRV_PCM_RATE_16000, -- 1.7.12

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Re: [alsa-devel] snd_usb changes broke audio
by Daniel Mack 22 Sep '12

22 Sep '12

(Cc: ALSA list, please don't drop it on replies!) On 22.09.2012 10:21, Nate Garner wrote: > output of lsusb -v (assuming you just want the sound card related part. > if you need more let me know): Ok, thanks - more stuff below. > Bus 009 Device 003: ID 041e:3040 Creative Technology, Ltd SoundBlaster > Live! 24-bit External SB0490 > Device Descriptor: > bLength 18 > bDescriptorType 1 > bcdUSB 1.10 > bDeviceClass 0 (Defined at Interface level) > bDeviceSubClass 0 > bDeviceProtocol 0 > bMaxPacketSize0 8 > idVendor 0x041e Creative Technology, Ltd > idProduct 0x3040 SoundBlaster Live! 24-bit External SB0490 > bcdDevice 1.00 > iManufacturer 1 Creative Technology > iProduct 2 SB Live! 24-bit External? > iSerial 0 > bNumConfigurations 1 > Configuration Descriptor: > bLength 9 > bDescriptorType 2 > wTotalLength 979 > bNumInterfaces 3 > bConfigurationValue 1 > iConfiguration 0 > bmAttributes 0x80 > (Bus Powered) > MaxPower 308mA > Interface Descriptor: > bLength 9 > bDescriptorType 4 > bInterfaceNumber 0 > bAlternateSetting 0 > bNumEndpoints 1 > bInterfaceClass 1 Audio > bInterfaceSubClass 1 Control Device > bInterfaceProtocol 0 > iInterface 0 > AudioControl Interface Descriptor: > bLength 10 > bDescriptorType 36 > bDescriptorSubtype 1 (HEADER) > bcdADC 1.00 > wTotalLength 76 > bInCollection 2 > baInterfaceNr( 0) 1 > baInterfaceNr( 1) 2 > AudioControl Interface Descriptor: > bLength 12 > bDescriptorType 36 > bDescriptorSubtype 2 (INPUT_TERMINAL) > bTerminalID 1 > wTerminalType 0x0101 USB Streaming > bAssocTerminal 0 > bNrChannels 6 > wChannelConfig 0x003f > Left Front (L) > Right Front (R) > Center Front (C) > Low Freqency Enhancement (LFE) > Left Surround (LS) > Right Surround (RS) > iChannelNames 0 > iTerminal 0 > AudioControl Interface Descriptor: > bLength 14 > bDescriptorType 36 > bDescriptorSubtype 6 (FEATURE_UNIT) > bUnitID 2 > bSourceID 1 > bControlSize 1 > bmaControls( 0) 0x03 > Mute Control > Volume Control > bmaControls( 1) 0x00 > bmaControls( 2) 0x00 > bmaControls( 3) 0x00 > bmaControls( 4) 0x00 > bmaControls( 5) 0x00 > bmaControls( 6) 0x00 > iFeature 0 > AudioControl Interface Descriptor: > bLength 9 > bDescriptorType 36 > bDescriptorSubtype 3 (OUTPUT_TERMINAL) > bTerminalID 3 > wTerminalType 0x0301 Speaker > bAssocTerminal 0 > bSourceID 2 > iTerminal 0 > AudioControl Interface Descriptor: > bLength 12 > bDescriptorType 36 > bDescriptorSubtype 2 (INPUT_TERMINAL) > bTerminalID 4 > wTerminalType 0x0603 Line Connector > bAssocTerminal 0 > bNrChannels 2 > wChannelConfig 0x0003 > Left Front (L) > Right Front (R) > iChannelNames 0 > iTerminal 0 > AudioControl Interface Descriptor: > bLength 10 > bDescriptorType 36 > bDescriptorSubtype 6 (FEATURE_UNIT) > bUnitID 5 > bSourceID 4 > bControlSize 1 > bmaControls( 0) 0x02 > Volume Control > bmaControls( 1) 0x00 > bmaControls( 2) 0x00 > iFeature 0 > AudioControl Interface Descriptor: > bLength 9 > bDescriptorType 36 > bDescriptorSubtype 3 (OUTPUT_TERMINAL) > bTerminalID 6 > wTerminalType 0x0101 USB Streaming > bAssocTerminal 0 > bSourceID 5 > iTerminal 0 > Endpoint Descriptor: > bLength 7 > bDescriptorType 5 > bEndpointAddress 0x83 EP 3 IN > bmAttributes 3 > Transfer Type Interrupt > Synch Type None > Usage Type Data > wMaxPacketSize 0x0008 1x 8 bytes > bInterval 10 > 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 1 > 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 1 PCM > AudioStreaming Interface Descriptor: > bLength 11 > bDescriptorType 36 > bDescriptorSubtype 2 (FORMAT_TYPE) > bFormatType 1 (FORMAT_TYPE_I) > bNrChannels 2 > bSubframeSize 2 > bBitResolution 16 > bSamFreqType 1 Discrete > tSamFreq[ 0] 44100 > Endpoint Descriptor: > bLength 9 > bDescriptorType 5 > bEndpointAddress 0x01 EP 1 OUT > bmAttributes 9 > Transfer Type Isochronous > Synch Type Adaptive > Usage Type Data > wMaxPacketSize 0x00b4 1x 180 bytes > bInterval 1 > bRefresh 0 > bSynchAddress 0 > AudioControl Endpoint Descriptor: > bLength 7 > bDescriptorType 37 > bDescriptorSubtype 1 (EP_GENERAL) > bmAttributes 0x01 > Sampling Frequency > bLockDelayUnits 0 Undefined > wLockDelay 0 Undefined > Interface Descriptor: > bLength 9 > bDescriptorType 4 > bInterfaceNumber 1 > bAlternateSetting 2 > bNumEndpoints 1 > 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 1 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] 44100 > Endpoint Descriptor: > bLength 9 > bDescriptorType 5 > bEndpointAddress 0x01 EP 1 OUT > bmAttributes 9 > Transfer Type Isochronous > Synch Type Adaptive > Usage Type Data > wMaxPacketSize 0x010e 1x 270 bytes > bInterval 1 > bRefresh 0 > bSynchAddress 0 > AudioControl Endpoint Descriptor: > bLength 7 > bDescriptorType 37 > bDescriptorSubtype 1 (EP_GENERAL) > bmAttributes 0x01 > Sampling Frequency > bLockDelayUnits 0 Undefined > wLockDelay 0 Undefined > Interface Descriptor: > bLength 9 > bDescriptorType 4 > bInterfaceNumber 1 > bAlternateSetting 3 > bNumEndpoints 1 > 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 1 PCM > AudioStreaming Interface Descriptor: > bLength 11 > bDescriptorType 36 > bDescriptorSubtype 2 (FORMAT_TYPE) > bFormatType 1 (FORMAT_TYPE_I) > bNrChannels 4 > bSubframeSize 2 > bBitResolution 16 > bSamFreqType 1 Discrete > tSamFreq[ 0] 44100 > Endpoint Descriptor: > bLength 9 > bDescriptorType 5 > bEndpointAddress 0x01 EP 1 OUT > bmAttributes 9 > Transfer Type Isochronous > Synch Type Adaptive > Usage Type Data > wMaxPacketSize 0x0168 1x 360 bytes > bInterval 1 > bRefresh 0 > bSynchAddress 0 > AudioControl Endpoint Descriptor: > bLength 7 > bDescriptorType 37 > bDescriptorSubtype 1 (EP_GENERAL) > bmAttributes 0x01 > Sampling Frequency > bLockDelayUnits 0 Undefined > wLockDelay 0 Undefined > Interface Descriptor: > bLength 9 > bDescriptorType 4 > bInterfaceNumber 1 > bAlternateSetting 4 > bNumEndpoints 1 > 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 1 PCM > AudioStreaming Interface Descriptor: > bLength 11 > bDescriptorType 36 > bDescriptorSubtype 2 (FORMAT_TYPE) > bFormatType 1 (FORMAT_TYPE_I) > bNrChannels 4 > bSubframeSize 3 > bBitResolution 24 > bSamFreqType 1 Discrete > tSamFreq[ 0] 44100 > Endpoint Descriptor: > bLength 9 > bDescriptorType 5 > bEndpointAddress 0x01 EP 1 OUT > bmAttributes 9 > Transfer Type Isochronous > Synch Type Adaptive > Usage Type Data > wMaxPacketSize 0x021c 1x 540 bytes > bInterval 1 > bRefresh 0 > bSynchAddress 0 > AudioControl Endpoint Descriptor: > bLength 7 > bDescriptorType 37 > bDescriptorSubtype 1 (EP_GENERAL) > bmAttributes 0x01 > Sampling Frequency > bLockDelayUnits 0 Undefined > wLockDelay 0 Undefined > Interface Descriptor: > bLength 9 > bDescriptorType 4 > bInterfaceNumber 1 > bAlternateSetting 5 > bNumEndpoints 1 > 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 1 PCM > AudioStreaming Interface Descriptor: > bLength 11 > bDescriptorType 36 > bDescriptorSubtype 2 (FORMAT_TYPE) > bFormatType 1 (FORMAT_TYPE_I) > bNrChannels 6 > bSubframeSize 2 > bBitResolution 16 > bSamFreqType 1 Discrete > tSamFreq[ 0] 44100 > Endpoint Descriptor: > bLength 9 > bDescriptorType 5 > bEndpointAddress 0x01 EP 1 OUT > bmAttributes 9 > Transfer Type Isochronous > Synch Type Adaptive > Usage Type Data > wMaxPacketSize 0x021c 1x 540 bytes > bInterval 1 > bRefresh 0 > bSynchAddress 0 > AudioControl Endpoint Descriptor: > bLength 7 > bDescriptorType 37 > bDescriptorSubtype 1 (EP_GENERAL) > bmAttributes 0x01 > Sampling Frequency > bLockDelayUnits 0 Undefined > wLockDelay 0 Undefined > Interface Descriptor: > bLength 9 > bDescriptorType 4 > bInterfaceNumber 1 > bAlternateSetting 6 > bNumEndpoints 1 > 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 1 PCM > AudioStreaming Interface Descriptor: > bLength 11 > bDescriptorType 36 > bDescriptorSubtype 2 (FORMAT_TYPE) > bFormatType 1 (FORMAT_TYPE_I) > bNrChannels 6 > bSubframeSize 3 > bBitResolution 24 > bSamFreqType 1 Discrete > tSamFreq[ 0] 44100 > Endpoint Descriptor: > bLength 9 > bDescriptorType 5 > bEndpointAddress 0x01 EP 1 OUT > bmAttributes 9 > Transfer Type Isochronous > Synch Type Adaptive > Usage Type Data > wMaxPacketSize 0x032a 1x 810 bytes > bInterval 1 > bRefresh 0 > bSynchAddress 0 > AudioControl Endpoint Descriptor: > bLength 7 > bDescriptorType 37 > bDescriptorSubtype 1 (EP_GENERAL) > bmAttributes 0x01 > Sampling Frequency > bLockDelayUnits 0 Undefined > wLockDelay 0 Undefined > Interface Descriptor: > bLength 9 > bDescriptorType 4 > bInterfaceNumber 1 > bAlternateSetting 7 > bNumEndpoints 1 > 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 1 PCM > AudioStreaming Interface Descriptor: > bLength 11 > bDescriptorType 36 > bDescriptorSubtype 2 (FORMAT_TYPE) > bFormatType 1 (FORMAT_TYPE_I) > bNrChannels 2 > bSubframeSize 2 > bBitResolution 16 > bSamFreqType 1 Discrete > tSamFreq[ 0] 48000 > Endpoint Descriptor: > bLength 9 > bDescriptorType 5 > bEndpointAddress 0x01 EP 1 OUT > bmAttributes 9 > Transfer Type Isochronous > Synch Type Adaptive > Usage Type Data > wMaxPacketSize 0x00c0 1x 192 bytes > bInterval 1 > bRefresh 0 > bSynchAddress 0 > AudioControl Endpoint Descriptor: > bLength 7 > bDescriptorType 37 > bDescriptorSubtype 1 (EP_GENERAL) > bmAttributes 0x01 > Sampling Frequency > bLockDelayUnits 0 Undefined > wLockDelay 0 Undefined > Interface Descriptor: > bLength 9 > bDescriptorType 4 > bInterfaceNumber 1 > bAlternateSetting 8 > bNumEndpoints 1 > 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 1 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 0x01 EP 1 OUT > bmAttributes 9 > Transfer Type Isochronous > Synch Type Adaptive > Usage Type Data > wMaxPacketSize 0x0120 1x 288 bytes > bInterval 1 > bRefresh 0 > bSynchAddress 0 > AudioControl Endpoint Descriptor: > bLength 7 > bDescriptorType 37 > bDescriptorSubtype 1 (EP_GENERAL) > bmAttributes 0x01 > Sampling Frequency > bLockDelayUnits 0 Undefined > wLockDelay 0 Undefined > Interface Descriptor: > bLength 9 > bDescriptorType 4 > bInterfaceNumber 1 > bAlternateSetting 9 > bNumEndpoints 1 > 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 1 PCM > AudioStreaming Interface Descriptor: > bLength 11 > bDescriptorType 36 > bDescriptorSubtype 2 (FORMAT_TYPE) > bFormatType 1 (FORMAT_TYPE_I) > bNrChannels 4 > bSubframeSize 2 > bBitResolution 16 > bSamFreqType 1 Discrete > tSamFreq[ 0] 48000 > Endpoint Descriptor: > bLength 9 > bDescriptorType 5 > bEndpointAddress 0x01 EP 1 OUT > bmAttributes 9 > Transfer Type Isochronous > Synch Type Adaptive > Usage Type Data > wMaxPacketSize 0x0180 1x 384 bytes > bInterval 1 > bRefresh 0 > bSynchAddress 0 > AudioControl Endpoint Descriptor: > bLength 7 > bDescriptorType 37 > bDescriptorSubtype 1 (EP_GENERAL) > bmAttributes 0x01 > Sampling Frequency > bLockDelayUnits 0 Undefined > wLockDelay 0 Undefined > Interface Descriptor: > bLength 9 > bDescriptorType 4 > bInterfaceNumber 1 > bAlternateSetting 10 > bNumEndpoints 1 > 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 1 PCM > AudioStreaming Interface Descriptor: > bLength 11 > bDescriptorType 36 > bDescriptorSubtype 2 (FORMAT_TYPE) > bFormatType 1 (FORMAT_TYPE_I) > bNrChannels 4 > bSubframeSize 3 > bBitResolution 24 > bSamFreqType 1 Discrete > tSamFreq[ 0] 48000 > Endpoint Descriptor: > bLength 9 > bDescriptorType 5 > bEndpointAddress 0x01 EP 1 OUT > bmAttributes 9 > Transfer Type Isochronous > Synch Type Adaptive > Usage Type Data > wMaxPacketSize 0x0240 1x 576 bytes > bInterval 1 > bRefresh 0 > bSynchAddress 0 > AudioControl Endpoint Descriptor: > bLength 7 > bDescriptorType 37 > bDescriptorSubtype 1 (EP_GENERAL) > bmAttributes 0x01 > Sampling Frequency > bLockDelayUnits 0 Undefined > wLockDelay 0 Undefined > Interface Descriptor: > bLength 9 > bDescriptorType 4 > bInterfaceNumber 1 > bAlternateSetting 11 > bNumEndpoints 1 > 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 1 PCM > AudioStreaming Interface Descriptor: > bLength 11 > bDescriptorType 36 > bDescriptorSubtype 2 (FORMAT_TYPE) > bFormatType 1 (FORMAT_TYPE_I) > bNrChannels 6 > bSubframeSize 2 > bBitResolution 16 > bSamFreqType 1 Discrete > tSamFreq[ 0] 48000 > Endpoint Descriptor: > bLength 9 > bDescriptorType 5 > bEndpointAddress 0x01 EP 1 OUT > bmAttributes 9 > Transfer Type Isochronous > Synch Type Adaptive > Usage Type Data > wMaxPacketSize 0x0240 1x 576 bytes > bInterval 1 > bRefresh 0 > bSynchAddress 0 > AudioControl Endpoint Descriptor: > bLength 7 > bDescriptorType 37 > bDescriptorSubtype 1 (EP_GENERAL) > bmAttributes 0x01 > Sampling Frequency > bLockDelayUnits 0 Undefined > wLockDelay 0 Undefined > Interface Descriptor: > bLength 9 > bDescriptorType 4 > bInterfaceNumber 1 > bAlternateSetting 12 > bNumEndpoints 1 > 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 1 PCM > AudioStreaming Interface Descriptor: > bLength 11 > bDescriptorType 36 > bDescriptorSubtype 2 (FORMAT_TYPE) > bFormatType 1 (FORMAT_TYPE_I) > bNrChannels 6 > bSubframeSize 3 > bBitResolution 24 > bSamFreqType 1 Discrete > tSamFreq[ 0] 48000 > Endpoint Descriptor: > bLength 9 > bDescriptorType 5 > bEndpointAddress 0x01 EP 1 OUT > bmAttributes 9 > Transfer Type Isochronous > Synch Type Adaptive > Usage Type Data > wMaxPacketSize 0x0360 1x 864 bytes > bInterval 1 > bRefresh 0 > bSynchAddress 0 > AudioControl Endpoint Descriptor: > bLength 7 > bDescriptorType 37 > bDescriptorSubtype 1 (EP_GENERAL) > bmAttributes 0x01 > Sampling Frequency > bLockDelayUnits 0 Undefined > wLockDelay 0 Undefined > Interface Descriptor: > bLength 9 > bDescriptorType 4 > bInterfaceNumber 1 > bAlternateSetting 13 > bNumEndpoints 1 > 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 1 PCM > AudioStreaming Interface Descriptor: > bLength 11 > bDescriptorType 36 > bDescriptorSubtype 2 (FORMAT_TYPE) > bFormatType 1 (FORMAT_TYPE_I) > bNrChannels 2 > bSubframeSize 2 > bBitResolution 16 > bSamFreqType 1 Discrete > tSamFreq[ 0] 96000 > Endpoint Descriptor: > bLength 9 > bDescriptorType 5 > bEndpointAddress 0x01 EP 1 OUT > bmAttributes 9 > Transfer Type Isochronous > Synch Type Adaptive > Usage Type Data > wMaxPacketSize 0x0180 1x 384 bytes > bInterval 1 > bRefresh 0 > bSynchAddress 0 > AudioControl Endpoint Descriptor: > bLength 7 > bDescriptorType 37 > bDescriptorSubtype 1 (EP_GENERAL) > bmAttributes 0x01 > Sampling Frequency > bLockDelayUnits 0 Undefined > wLockDelay 0 Undefined > Interface Descriptor: > bLength 9 > bDescriptorType 4 > bInterfaceNumber 1 > bAlternateSetting 14 > bNumEndpoints 1 > 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 1 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] 96000 > Endpoint Descriptor: > bLength 9 > bDescriptorType 5 > bEndpointAddress 0x01 EP 1 OUT > bmAttributes 9 > Transfer Type Isochronous > Synch Type Adaptive > Usage Type Data > wMaxPacketSize 0x0240 1x 576 bytes > bInterval 1 > bRefresh 0 > bSynchAddress 0 > AudioControl Endpoint Descriptor: > bLength 7 > bDescriptorType 37 > bDescriptorSubtype 1 (EP_GENERAL) > bmAttributes 0x01 > Sampling Frequency > bLockDelayUnits 0 Undefined > wLockDelay 0 Undefined > 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 6 > bDelay 1 frames > wFormatTag 1 PCM > AudioStreaming Interface Descriptor: > bLength 11 > bDescriptorType 36 > bDescriptorSubtype 2 (FORMAT_TYPE) > bFormatType 1 (FORMAT_TYPE_I) > bNrChannels 2 > bSubframeSize 2 > bBitResolution 16 > bSamFreqType 1 Discrete > tSamFreq[ 0] 44100 > Endpoint Descriptor: > bLength 9 > bDescriptorType 5 > bEndpointAddress 0x82 EP 2 IN > bmAttributes 13 > Transfer Type Isochronous > Synch Type Synchronous > Usage Type Data > wMaxPacketSize 0x00b4 1x 180 bytes > bInterval 1 > bRefresh 0 > bSynchAddress 0 > AudioControl Endpoint Descriptor: > bLength 7 > bDescriptorType 37 > bDescriptorSubtype 1 (EP_GENERAL) > bmAttributes 0x01 > Sampling Frequency > bLockDelayUnits 0 Undefined > wLockDelay 0 Undefined > Interface Descriptor: > bLength 9 > bDescriptorType 4 > bInterfaceNumber 2 > bAlternateSetting 2 > bNumEndpoints 1 > bInterfaceClass 1 Audio > bInterfaceSubClass 2 Streaming > bInterfaceProtocol 0 > iInterface 0 > AudioStreaming Interface Descriptor: > bLength 7 > bDescriptorType 36 > bDescriptorSubtype 1 (AS_GENERAL) > bTerminalLink 6 > bDelay 1 frames > wFormatTag 1 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] 44100 > Endpoint Descriptor: > bLength 9 > bDescriptorType 5 > bEndpointAddress 0x82 EP 2 IN > bmAttributes 13 > Transfer Type Isochronous > Synch Type Synchronous > Usage Type Data > wMaxPacketSize 0x010e 1x 270 bytes > bInterval 1 > bRefresh 0 > bSynchAddress 0 > AudioControl Endpoint Descriptor: > bLength 7 > bDescriptorType 37 > bDescriptorSubtype 1 (EP_GENERAL) > bmAttributes 0x01 > Sampling Frequency > bLockDelayUnits 0 Undefined > wLockDelay 0 Undefined > Interface Descriptor: > bLength 9 > bDescriptorType 4 > bInterfaceNumber 2 > bAlternateSetting 3 > bNumEndpoints 1 > bInterfaceClass 1 Audio > bInterfaceSubClass 2 Streaming > bInterfaceProtocol 0 > iInterface 0 > AudioStreaming Interface Descriptor: > bLength 7 > bDescriptorType 36 > bDescriptorSubtype 1 (AS_GENERAL) > bTerminalLink 6 > bDelay 1 frames > wFormatTag 1 PCM > AudioStreaming Interface Descriptor: > bLength 11 > bDescriptorType 36 > bDescriptorSubtype 2 (FORMAT_TYPE) > bFormatType 1 (FORMAT_TYPE_I) > bNrChannels 2 > bSubframeSize 2 > bBitResolution 16 > bSamFreqType 1 Discrete > tSamFreq[ 0] 48000 > Endpoint Descriptor: > bLength 9 > bDescriptorType 5 > bEndpointAddress 0x82 EP 2 IN > bmAttributes 13 > Transfer Type Isochronous > Synch Type Synchronous > Usage Type Data > wMaxPacketSize 0x00c4 1x 196 bytes > bInterval 1 > bRefresh 0 > bSynchAddress 0 > AudioControl Endpoint Descriptor: > bLength 7 > bDescriptorType 37 > bDescriptorSubtype 1 (EP_GENERAL) > bmAttributes 0x01 > Sampling Frequency > bLockDelayUnits 0 Undefined > wLockDelay 0 Undefined > Interface Descriptor: > bLength 9 > bDescriptorType 4 > bInterfaceNumber 2 > bAlternateSetting 4 > bNumEndpoints 1 > bInterfaceClass 1 Audio > bInterfaceSubClass 2 Streaming > bInterfaceProtocol 0 > iInterface 0 > AudioStreaming Interface Descriptor: > bLength 7 > bDescriptorType 36 > bDescriptorSubtype 1 (AS_GENERAL) > bTerminalLink 6 > bDelay 1 frames > wFormatTag 1 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 0x82 EP 2 IN > bmAttributes 13 > Transfer Type Isochronous > Synch Type Synchronous > Usage Type Data > wMaxPacketSize 0x0126 1x 294 bytes > bInterval 1 > bRefresh 0 > bSynchAddress 0 > AudioControl Endpoint Descriptor: > bLength 7 > bDescriptorType 37 > bDescriptorSubtype 1 (EP_GENERAL) > bmAttributes 0x01 > Sampling Frequency > bLockDelayUnits 0 Undefined > wLockDelay 0 Undefined > Interface Descriptor: > bLength 9 > bDescriptorType 4 > bInterfaceNumber 2 > bAlternateSetting 5 > bNumEndpoints 1 > bInterfaceClass 1 Audio > bInterfaceSubClass 2 Streaming > bInterfaceProtocol 0 > iInterface 0 > AudioStreaming Interface Descriptor: > bLength 7 > bDescriptorType 36 > bDescriptorSubtype 1 (AS_GENERAL) > bTerminalLink 6 > bDelay 1 frames > wFormatTag 1 PCM > AudioStreaming Interface Descriptor: > bLength 11 > bDescriptorType 36 > bDescriptorSubtype 2 (FORMAT_TYPE) > bFormatType 1 (FORMAT_TYPE_I) > bNrChannels 2 > bSubframeSize 2 > bBitResolution 16 > bSamFreqType 1 Discrete > tSamFreq[ 0] 96000 > Endpoint Descriptor: > bLength 9 > bDescriptorType 5 > bEndpointAddress 0x82 EP 2 IN > bmAttributes 13 > Transfer Type Isochronous > Synch Type Synchronous > Usage Type Data > wMaxPacketSize 0x0184 1x 388 bytes > bInterval 1 > bRefresh 0 > bSynchAddress 0 > AudioControl Endpoint Descriptor: > bLength 7 > bDescriptorType 37 > bDescriptorSubtype 1 (EP_GENERAL) > bmAttributes 0x01 > Sampling Frequency > bLockDelayUnits 0 Undefined > wLockDelay 0 Undefined > Interface Descriptor: > bLength 9 > bDescriptorType 4 > bInterfaceNumber 2 > bAlternateSetting 6 > bNumEndpoints 1 > bInterfaceClass 1 Audio > bInterfaceSubClass 2 Streaming > bInterfaceProtocol 0 > iInterface 0 > AudioStreaming Interface Descriptor: > bLength 7 > bDescriptorType 36 > bDescriptorSubtype 1 (AS_GENERAL) > bTerminalLink 6 > bDelay 1 frames > wFormatTag 1 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] 96000 > Endpoint Descriptor: > bLength 9 > bDescriptorType 5 > bEndpointAddress 0x82 EP 2 IN > bmAttributes 13 > Transfer Type Isochronous > Synch Type Synchronous > Usage Type Data > wMaxPacketSize 0x0246 1x 582 bytes > bInterval 1 > bRefresh 0 > bSynchAddress 0 > AudioControl Endpoint Descriptor: > bLength 7 > bDescriptorType 37 > bDescriptorSubtype 1 (EP_GENERAL) > bmAttributes 0x01 > Sampling Frequency > bLockDelayUnits 0 Undefined > wLockDelay 0 Undefined > Device Status: 0x0000 > (Bus Powered) > > Nothing in dmesg looks relevent. Errors.log contains: > localhost pulseaudio[900]: [pulseaudio] bluetooth-util.c: > org.bluez.Manager.ListAdapters() failed: > org.freedesktop.DBus.Error.Spawn.ChildExited: Launch helper exited with > unknown return code 1 > > nothing else obvious I can see. Are you sure you there's no "not enough bandwidth" message anywhere? And is your device connected to the host directly or via a hub? Thanks, Daniel > > On Wed, Sep 19, 2012 at 11:41 PM, Daniel Mack <zonque(a)gmail.com > <mailto:zonque@gmail.com>> wrote: > > (Cc the ALSA list - please don't drop it in replies) > > On 20.09.2012 02:16, Nate Garner wrote: > > I recently updated my kernel from 3.5.3 to 3.5.4. Upon rebooting, I no > > longer got any audio through my external sound card, but my headphones > > worked. I noticed in the kernel change log that you made some > changes to > > snd-usb, and thought that might be the culprit. I hope you're the > right > > person to report this too, if not, feel free to ignore. > > Relevant info: > > Distro: arch linux, x86_64 > > sound card: creative labs model SB0490 (soundblaser Live! 24-bit > > external (link here > > > <http://support.creative.com/Products/ProductDetails.aspx?catID=1&subCatID=2…>) > > alsa version 1.0.26 > > pulseaudio version 2.1 > > Argh! I believed we finally fixed all the regressions. > > Are there any messages in the logs or in dmesg that look releated? Can > you send the output of "lsusb -v", please? > > > Daniel > >

1 0

[alsa-devel] [PATCH] test: add audio_time
by Pierre-Louis Bossart 22 Sep '12

22 Sep '12

Simple test to create playback and capture streams, and check elapsed time vs. sample counts reported by driver. This should be helpful for driver developers and anyone interested in system/audio time drift. tested only on HDAudio TODO: - make period configurable - better output messages - support for wall clock when it's in the mainline Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> --- test/audio_time.c | 237 +++++++++++++++++++++++++++++++++++++++++++++++++++++ 1 file changed, 237 insertions(+) create mode 100644 test/audio_time.c diff --git a/test/audio_time.c b/test/audio_time.c new file mode 100644 index 0000000..a910783 --- /dev/null +++ b/test/audio_time.c @@ -0,0 +1,237 @@ +/* + * This program only tracks the difference between system time + * and audio time, as reported in snd_pcm_status(). It should be + * helpful to verify the information reported by drivers. + */ + +#include "../include/asoundlib.h" +#include <math.h> + +static char *device = "hw:0,0"; + +snd_output_t *output = NULL; + +long long timestamp2ns(snd_htimestamp_t t) +{ + long long nsec; + + nsec = t.tv_sec * 1000000000; + nsec += t.tv_nsec; + + return nsec; +} + +long long timediff(snd_htimestamp_t t1, snd_htimestamp_t t2) +{ + long long nsec1, nsec2; + + nsec1 = timestamp2ns(t1); + nsec2 = timestamp2ns(t2); + + return nsec1 - nsec2; +} + +void gettimestamp(snd_pcm_t *handle, snd_htimestamp_t *timestamp, + snd_htimestamp_t *trigger_timestamp, + snd_pcm_uframes_t *avail, snd_pcm_sframes_t *delay) +{ + int err; + snd_pcm_status_t *status; + + snd_pcm_status_alloca(&status); + if ((err = snd_pcm_status(handle, status)) < 0) { + printf("Stream status error: %s\n", snd_strerror(err)); + exit(0); + } + snd_pcm_status_get_trigger_htstamp(status, trigger_timestamp); + snd_pcm_status_get_htstamp(status, timestamp); + *avail = snd_pcm_status_get_avail(status); + *delay = snd_pcm_status_get_delay(status); +} + +#define PERIOD 6000 +#define PCM_LINK /* sync start for playback and capture */ +#define TRACK_CAPTURE /* dump capture timing info */ +#define TRACK_PLAYBACK /* dump playback timing info */ +#define PLAYBACK_BUFFERS 4 + + +int main(void) +{ + int err; + unsigned int i; + snd_pcm_t *handle_p = NULL; + snd_pcm_t *handle_c = NULL; + snd_pcm_sframes_t frames; + snd_htimestamp_t tstamp_c, tstamp_p; + snd_htimestamp_t trigger_tstamp_c, trigger_tstamp_p; + unsigned char buffer_p[PERIOD*4*4]; + unsigned char buffer_c[PERIOD*4*4]; + + snd_pcm_sw_params_t *swparams_p; + snd_pcm_sw_params_t *swparams_c; + + snd_pcm_uframes_t curr_count_c; + snd_pcm_uframes_t frame_count_c = 0; + snd_pcm_uframes_t curr_count_p; + snd_pcm_uframes_t frame_count_p = 0; + + snd_pcm_sframes_t delay_p, delay_c; + snd_pcm_uframes_t avail_p, avail_c; + + if ((err = snd_pcm_open(&handle_p, device, SND_PCM_STREAM_PLAYBACK, 0)) < 0) { + printf("Playback open error: %s\n", snd_strerror(err)); + goto _exit; + } + if ((err = snd_pcm_set_params(handle_p, + SND_PCM_FORMAT_S16, + SND_PCM_ACCESS_RW_INTERLEAVED, + 2, + 48000, + 0, + 500000)) < 0) { /* 0.5sec */ + printf("Playback open error: %s\n", snd_strerror(err)); + goto _exit; + } + + snd_pcm_sw_params_alloca(&swparams_p); + /* get the current swparams */ + err = snd_pcm_sw_params_current(handle_p, swparams_p); + if (err < 0) { + printf("Unable to determine current swparams_p: %s\n", snd_strerror(err)); + goto _exit; + } + + /* enable tstamp */ + err = snd_pcm_sw_params_set_tstamp_mode(handle_p, swparams_p, SND_PCM_TSTAMP_ENABLE); + if (err < 0) { + printf("Unable to set tstamp mode : %s\n", snd_strerror(err)); + goto _exit; + } + + /* write the sw parameters */ + err = snd_pcm_sw_params(handle_p, swparams_p); + if (err < 0) { + printf("Unable to set swparams_p : %s\n", snd_strerror(err)); + goto _exit; + } + + if ((err = snd_pcm_open(&handle_c, device, SND_PCM_STREAM_CAPTURE, SND_PCM_NONBLOCK)) < 0) { + printf("Capture open error: %s\n", snd_strerror(err)); + goto _exit; + } + if ((err = snd_pcm_set_params(handle_c, + SND_PCM_FORMAT_S16, + SND_PCM_ACCESS_RW_INTERLEAVED, + 2, + 48000, + 0, + 500000)) < 0) { /* 0.5sec */ + printf("Capture open error: %s\n", snd_strerror(err)); + goto _exit; + } + + snd_pcm_sw_params_alloca(&swparams_c); + /* get the current swparams */ + err = snd_pcm_sw_params_current(handle_c, swparams_c); + if (err < 0) { + printf("Unable to determine current swparams_c: %s\n", snd_strerror(err)); + goto _exit; + } + + /* enable tstamp */ + err = snd_pcm_sw_params_set_tstamp_mode(handle_c, swparams_c, SND_PCM_TSTAMP_ENABLE); + if (err < 0) { + printf("Unable to set tstamp mode : %s\n", snd_strerror(err)); + goto _exit; + } + + /* write the sw parameters */ + err = snd_pcm_sw_params(handle_c, swparams_c); + if (err < 0) { + printf("Unable to set swparams_c : %s\n", snd_strerror(err)); + goto _exit; + } + +#ifdef PCM_LINK + if ((err = snd_pcm_link(handle_c, handle_p)) < 0) { + printf("Streams link error: %s\n", snd_strerror(err)); + exit(0); + } +#endif + + i = PLAYBACK_BUFFERS; + while (i--) { + frames = snd_pcm_writei(handle_p, buffer_p, PERIOD); + if (frames < 0) { + printf("snd_pcm_writei failed: %s\n", snd_strerror(frames)); + goto _exit; + } + frame_count_p += frames; + } + + if (PLAYBACK_BUFFERS != 4) + snd_pcm_start(handle_p); + +#ifndef PCM_LINK + /* need to start capture explicitly */ + snd_pcm_start(handle_c); +#endif + + while (1) { + + frames = snd_pcm_wait(handle_c, -1); + if (frames < 0) { + printf("snd_pcm_wait failed: %s\n", snd_strerror(frames)); + goto _exit; + } + + frames = snd_pcm_readi(handle_c, buffer_c, PERIOD); + if (frames < 0) { + printf("snd_pcm_readi failed: %s\n", snd_strerror(frames)); + goto _exit; + } + frame_count_c += frames; + + frames = snd_pcm_writei(handle_p, buffer_p, PERIOD); + if (frames < 0) { + printf("snd_pcm_writei failed: %s\n", snd_strerror(frames)); + goto _exit; + } + + frame_count_p += frames; + +#if defined(TRACK_PLAYBACK) + gettimestamp(handle_p, &tstamp_p, &trigger_tstamp_p, &avail_p, &delay_p); + + curr_count_p = frame_count_p - delay_p; /* written minus queued */ + + printf("playback: systime: %lli nsec, sample time %lli nsec \tsystime delta %lli \n", + timediff(tstamp_p,trigger_tstamp_p), + (long long)round(((float)curr_count_p * 1000000000.0 / 48000.0)), + timediff(tstamp_p, trigger_tstamp_p) - (long long)round((double)curr_count_p * 1000000000.0 / 48000.0) + ); +#endif + +#if defined(TRACK_CAPTURE) + gettimestamp(handle_c, &tstamp_c, &trigger_tstamp_c, &avail_c, &delay_c); + + curr_count_c = frame_count_c + delay_c; /* read plus queued */ + + printf("\t capture: systime: %lli nsec, sample time %lli nsec \tsystime delta %lli \n", + timediff(tstamp_c,trigger_tstamp_c), + (long long)round(((float)curr_count_c * 1000000000.0 / 48000.0)), + timediff(tstamp_c, trigger_tstamp_c) - (long long)round((double)curr_count_c * 1000000000.0 / 48000.0) + ); +#endif + + } + +_exit: + if (handle_p) + snd_pcm_close(handle_p); + if (handle_c) + snd_pcm_close(handle_c); + + return 0; +} -- 1.7.9.5

2 1

[alsa-devel] [PATCH 1/3] ALSA: hda - force use of SSYNC bits
by Pierre-Louis Bossart 22 Sep '12

22 Sep '12

SSYNC bits are typically used to start multiple streams synchronously. It makes sense to use them for a single stream for a more predictable startup sequence. The transfers only start once the DMA and FIFOs are ready. This results in a better correlation between timestamps and number of samples played. Credits to Kar Leong Wang for suggesting this improvement. Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> --- sound/pci/hda/hda_intel.c | 36 ++++++++++++++++-------------------- 1 file changed, 16 insertions(+), 20 deletions(-) diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c index 60882c6..7ba187e 100644 --- a/sound/pci/hda/hda_intel.c +++ b/sound/pci/hda/hda_intel.c @@ -1959,14 +1959,14 @@ static int azx_pcm_trigger(struct snd_pcm_substream *substream, int cmd) } spin_lock(&chip->reg_lock); - if (nsync > 1) { - /* first, set SYNC bits of corresponding streams */ - if (chip->driver_caps & AZX_DCAPS_OLD_SSYNC) - azx_writel(chip, OLD_SSYNC, - azx_readl(chip, OLD_SSYNC) | sbits); - else - azx_writel(chip, SSYNC, azx_readl(chip, SSYNC) | sbits); - } + + /* first, set SYNC bits of corresponding streams */ + if (chip->driver_caps & AZX_DCAPS_OLD_SSYNC) + azx_writel(chip, OLD_SSYNC, + azx_readl(chip, OLD_SSYNC) | sbits); + else + azx_writel(chip, SSYNC, azx_readl(chip, SSYNC) | sbits); + snd_pcm_group_for_each_entry(s, substream) { if (s->pcm->card != substream->pcm->card) continue; @@ -1984,8 +1984,6 @@ static int azx_pcm_trigger(struct snd_pcm_substream *substream, int cmd) } spin_unlock(&chip->reg_lock); if (start) { - if (nsync == 1) - return 0; /* wait until all FIFOs get ready */ for (timeout = 5000; timeout; timeout--) { nwait = 0; @@ -2018,16 +2016,14 @@ static int azx_pcm_trigger(struct snd_pcm_substream *substream, int cmd) cpu_relax(); } } - if (nsync > 1) { - spin_lock(&chip->reg_lock); - /* reset SYNC bits */ - if (chip->driver_caps & AZX_DCAPS_OLD_SSYNC) - azx_writel(chip, OLD_SSYNC, - azx_readl(chip, OLD_SSYNC) & ~sbits); - else - azx_writel(chip, SSYNC, azx_readl(chip, SSYNC) & ~sbits); - spin_unlock(&chip->reg_lock); - } + spin_lock(&chip->reg_lock); + /* reset SYNC bits */ + if (chip->driver_caps & AZX_DCAPS_OLD_SSYNC) + azx_writel(chip, OLD_SSYNC, + azx_readl(chip, OLD_SSYNC) & ~sbits); + else + azx_writel(chip, SSYNC, azx_readl(chip, SSYNC) & ~sbits); + spin_unlock(&chip->reg_lock); return 0; } -- 1.7.9.5

2 3

[alsa-devel] line-out mute on "phantom" HP jack (Via codec)
by Herton Ronaldo Krzesinski 22 Sep '12

22 Sep '12

Hi, On Via codecs reported on https://bugs.launchpad.net/bugs/1052499, while investigating the report, it can be seen that on Headphone pin the plug is not detectable (NO_PRESENCE). It could be that front panel is AC97 wiring or something else, I don't know. But it makes the jack detection handling code to add the headphone jack as "phantom", which makes snd_hda_jack_detect return headphone jack as always present (commit 80c8bfbe768 is applied/cherry-picked on the reported kernel). Thus after jacks are added, the automute code will always detect HP as present, and keep muted all line-outs. The proposed solution following this is using is_jack_detectable inside via_hp_automute to handle this, other codecs seem to use also is_jack_detectable at the moment in similar situations. Other approach could be also to not report presence on phantom jacks with HP or Line Outs. -- []'s Herton

3 3

Re: [alsa-devel] question about realtek ALC269VB
by Takashi Iwai 21 Sep '12

21 Sep '12

At Fri, 21 Sep 2012 19:07:28 +0200, Oleksij Rempel wrote: > > Hallo all, Takashi, > > in attachment is a patch to enable external mic on Asus Zenbook UX32A. > It works fine for me, except there created some mixer/controls which > make no sense on this laptop. For example "Digital" input and IEC958. > How are they detected? By pin descriptor, or some code insight of > patch_realtek.c It must come from the default pin setup. I guess your machine has either a HDMI or a DP output, no? If it has, IEC958 is a mandatory control for the audio output over such a digital interface. "Digital" is likely the volume control created by alsa-lib, so it's no driver issue, and you can safely ignore it. Takashi > amixer and codec dump are attached too. > > > -- > Regards, > Oleksij > [2 0001-ALSA-hda-Add-external-mic-quirk-for-Asus-Zenbook-UX3.patch <text/x-patch (7bit)>] > >From 46472c1fbd9b82ea95cf22f2b49c74d4a7e82360 Mon Sep 17 00:00:00 2001 > From: Oleksij Rempel <bug-track(a)fisher-privat.net> > Date: Fri, 21 Sep 2012 17:44:58 +0200 > Subject: [PATCH] ALSA: hda - Add external mic quirk for Asus Zenbook UX31A > > Signed-off-by: Oleksij Rempel <bug-track(a)fisher-privat.net> > --- > sound/pci/hda/patch_realtek.c | 1 + > 1 file changed, 1 insertion(+) > > diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c > index 1907dda..3d440fc 100644 > --- a/sound/pci/hda/patch_realtek.c > +++ b/sound/pci/hda/patch_realtek.c > @@ -6189,6 +6189,7 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = { > SND_PCI_QUIRK(0x1025, 0x0349, "Acer AOD260", ALC269_FIXUP_INV_DMIC), > SND_PCI_QUIRK(0x103c, 0x1586, "HP", ALC269_FIXUP_MIC2_MUTE_LED), > SND_PCI_QUIRK(0x1043, 0x1427, "Asus Zenbook UX31E", ALC269VB_FIXUP_DMIC), > + SND_PCI_QUIRK(0x1043, 0x1517, "Asus Zenbook UX31A", ALC269VB_FIXUP_DMIC), > SND_PCI_QUIRK(0x1043, 0x1a13, "Asus G73Jw", ALC269_FIXUP_ASUS_G73JW), > SND_PCI_QUIRK(0x1043, 0x1b13, "Asus U41SV", ALC269_FIXUP_INV_DMIC), > SND_PCI_QUIRK(0x1043, 0x16e3, "ASUS UX50", ALC269_FIXUP_STEREO_DMIC), > -- > 1.7.10.4 > > [3 amixer <text/plain; UTF-8 (base64)>] > Simple mixer control 'Master',0 > Capabilities: pvolume pvolume-joined pswitch pswitch-joined penum > Playback channels: Mono > Limits: Playback 0 - 87 > Mono: Playback 47 [54%] [-30.00dB] [on] > Simple mixer control 'Headphone',0 > Capabilities: pvolume pswitch penum > Playback channels: Front Left - Front Right > Limits: Playback 0 - 87 > Mono: > Front Left: Playback 87 [100%] [0.00dB] [on] > Front Right: Playback 87 [100%] [0.00dB] [on] > Simple mixer control 'Speaker',0 > Capabilities: pvolume pswitch penum > Playback channels: Front Left - Front Right > Limits: Playback 0 - 87 > Mono: > Front Left: Playback 87 [100%] [0.00dB] [on] > Front Right: Playback 87 [100%] [0.00dB] [on] > Simple mixer control 'PCM',0 > Capabilities: pvolume penum > Playback channels: Front Left - Front Right > Limits: Playback 0 - 255 > Mono: > Front Left: Playback 255 [100%] [0.00dB] > Front Right: Playback 255 [100%] [0.00dB] > Simple mixer control 'Mic',0 > Capabilities: pvolume pswitch penum > Playback channels: Front Left - Front Right > Limits: Playback 0 - 31 > Mono: > Front Left: Playback 0 [0%] [-34.50dB] [off] > Front Right: Playback 0 [0%] [-34.50dB] [off] > Simple mixer control 'Mic Boost',0 > Capabilities: volume penum > Playback channels: Front Left - Front Right > Capture channels: Front Left - Front Right > Limits: 0 - 3 > Front Left: 0 [0%] [0.00dB] > Front Right: 0 [0%] [0.00dB] > Simple mixer control 'IEC958',0 > Capabilities: pswitch pswitch-joined penum > Playback channels: Mono > Mono: Playback [off] > Simple mixer control 'Capture',0 > Capabilities: cvolume cswitch penum > Capture channels: Front Left - Front Right > Limits: Capture 0 - 31 > Front Left: Capture 14 [45%] [4.50dB] [on] > Front Right: Capture 14 [45%] [4.50dB] [on] > Simple mixer control 'Auto-Mute Mode',0 > Capabilities: enum > Items: 'Disabled' 'Enabled' > Item0: 'Enabled' > Simple mixer control 'Digital',0 > Capabilities: cvolume penum > Capture channels: Front Left - Front Right > Limits: Capture 0 - 120 > Front Left: Capture 120 [100%] [30.00dB] > Front Right: Capture 120 [100%] [30.00dB] > Simple mixer control 'Internal Mic Boost',0 > Capabilities: volume penum > Playback channels: Front Left - Front Right > Capture channels: Front Left - Front Right > Limits: 0 - 3 > Front Left: 2 [67%] [24.00dB] > Front Right: 2 [67%] [24.00dB] > [4 hda_1 <text/plain; UTF-8 (base64)>] > Codec: Realtek ALC269VB > Address: 0 > AFG Function Id: 0x1 (unsol 1) > Vendor Id: 0x10ec0269 > Subsystem Id: 0x10431517 > Revision Id: 0x100100 > No Modem Function Group found > Default PCM: > rates [0x560]: 44100 48000 96000 192000 > bits [0xe]: 16 20 24 > formats [0x1]: PCM > Default Amp-In caps: N/A > Default Amp-Out caps: N/A > GPIO: io=2, o=0, i=0, unsolicited=1, wake=0 > IO[0]: enable=0, dir=0, wake=0, sticky=0, data=0, unsol=0 > IO[1]: enable=0, dir=0, wake=0, sticky=0, data=0, unsol=0 > Node 0x02 [Audio Output] wcaps 0x1d: Stereo Amp-Out > Control: name="Headphone Playback Volume", index=0, device=0 > ControlAmp: chs=3, dir=Out, idx=0, ofs=0 > Device: name="ALC269VB Analog", type="Audio", device=0 > Amp-Out caps: ofs=0x57, nsteps=0x57, stepsize=0x02, mute=0 > Amp-Out vals: [0x2f 0x2f] > Converter: stream=8, channel=0 > PCM: > rates [0x560]: 44100 48000 96000 192000 > bits [0xe]: 16 20 24 > formats [0x1]: PCM > Node 0x03 [Audio Output] wcaps 0x1d: Stereo Amp-Out > Control: name="Speaker Playback Volume", index=0, device=0 > ControlAmp: chs=3, dir=Out, idx=0, ofs=0 > Amp-Out caps: ofs=0x57, nsteps=0x57, stepsize=0x02, mute=0 > Amp-Out vals: [0x2f 0x2f] > Converter: stream=8, channel=0 > PCM: > rates [0x560]: 44100 48000 96000 192000 > bits [0xe]: 16 20 24 > formats [0x1]: PCM > Node 0x04 [Vendor Defined Widget] wcaps 0xf00000: Mono > Node 0x05 [Vendor Defined Widget] wcaps 0xf00000: Mono > Node 0x06 [Audio Output] wcaps 0x211: Stereo Digital > Converter: stream=0, channel=0 > Digital: > Digital category: 0x0 > PCM: > rates [0x5e0]: 44100 48000 88200 96000 192000 > bits [0xe]: 16 20 24 > formats [0x1]: PCM > Node 0x07 [Vendor Defined Widget] wcaps 0xf00000: Mono > Node 0x08 [Audio Input] wcaps 0x10011b: Stereo Amp-In > Amp-In caps: ofs=0x0b, nsteps=0x1f, stepsize=0x05, mute=1 > Amp-In vals: [0x8b 0x8b] > Converter: stream=0, channel=0 > SDI-Select: 0 > PCM: > rates [0x560]: 44100 48000 96000 192000 > bits [0xe]: 16 20 24 > formats [0x1]: PCM > Connection: 1 > 0x23 > Node 0x09 [Audio Input] wcaps 0x10011b: Stereo Amp-In > Control: name="Capture Switch", index=0, device=0 > Control: name="Capture Volume", index=0, device=0 > Device: name="ALC269VB Analog", type="Audio", device=0 > Amp-In caps: ofs=0x0b, nsteps=0x1f, stepsize=0x05, mute=1 > Amp-In vals: [0x0e 0x0e] > Converter: stream=4, channel=0 > SDI-Select: 0 > PCM: > rates [0x560]: 44100 48000 96000 192000 > bits [0xe]: 16 20 24 > formats [0x1]: PCM > Connection: 1 > 0x22 > Node 0x0a [Vendor Defined Widget] wcaps 0xf00000: Mono > Node 0x0b [Audio Mixer] wcaps 0x20010b: Stereo Amp-In > Control: name="Mic Playback Volume", index=0, device=0 > ControlAmp: chs=3, dir=In, idx=0, ofs=0 > Control: name="Mic Playback Switch", index=0, device=0 > ControlAmp: chs=3, dir=In, idx=0, ofs=0 > Amp-In caps: ofs=0x17, nsteps=0x1f, stepsize=0x05, mute=1 > Amp-In vals: [0x80 0x80] [0x80 0x80] [0x80 0x80] [0x80 0x80] [0x80 0x80] > Connection: 5 > 0x18 0x19 0x1a 0x1b 0x1d > Node 0x0c [Audio Mixer] wcaps 0x20010b: Stereo Amp-In > Amp-In caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 > Amp-In vals: [0x00 0x00] [0x00 0x00] > Connection: 2 > 0x02 0x0b > Node 0x0d [Audio Mixer] wcaps 0x20010b: Stereo Amp-In > Amp-In caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 > Amp-In vals: [0x00 0x00] [0x00 0x00] > Connection: 2 > 0x03 0x0b > Node 0x0e [Vendor Defined Widget] wcaps 0xf00000: Mono > Node 0x0f [Audio Mixer] wcaps 0x20010a: Mono Amp-In > Amp-In caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 > Amp-In vals: [0x00] [0x80] > Connection: 2 > 0x02 0x0b > Node 0x10 [Vendor Defined Widget] wcaps 0xf00000: Mono > Node 0x11 [Vendor Defined Widget] wcaps 0xf00000: Mono > Node 0x12 [Pin Complex] wcaps 0x40000b: Stereo Amp-In > Control: name="Internal Mic Boost Volume", index=0, device=0 > ControlAmp: chs=3, dir=In, idx=0, ofs=0 > Control: name="Internal Mic Phantom Jack", index=0, device=0 > Amp-In caps: ofs=0x00, nsteps=0x03, stepsize=0x2f, mute=0 > Amp-In vals: [0x01 0x01] > Pincap 0x00000020: IN > Pin Default 0x99a3092f: [Fixed] Mic at Int ATAPI > Conn = ATAPI, Color = Unknown > DefAssociation = 0x2, Sequence = 0xf > Misc = NO_PRESENCE > Pin-ctls: 0x20: IN > Node 0x13 [Vendor Defined Widget] wcaps 0xf00000: Mono > Node 0x14 [Pin Complex] wcaps 0x40018d: Stereo Amp-Out > Control: name="Speaker Playback Switch", index=0, device=0 > ControlAmp: chs=3, dir=Out, idx=0, ofs=0 > Control: name="Speaker Phantom Jack", index=0, device=0 > Amp-Out caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 > Amp-Out vals: [0x00 0x00] > Pincap 0x00010014: OUT EAPD Detect > EAPD 0x2: EAPD > Pin Default 0x99130110: [Fixed] Speaker at Int ATAPI > Conn = ATAPI, Color = Unknown > DefAssociation = 0x1, Sequence = 0x0 > Misc = NO_PRESENCE > Pin-ctls: 0x40: OUT > Unsolicited: tag=00, enabled=0 > Connection: 2 > 0x0c 0x0d* > Node 0x15 [Vendor Defined Widget] wcaps 0xf00000: Mono > Node 0x16 [Vendor Defined Widget] wcaps 0xf00000: Mono > Node 0x17 [Pin Complex] wcaps 0x40010c: Mono Amp-Out > Amp-Out caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 > Amp-Out vals: [0x80] > Pincap 0x00000010: OUT > Pin Default 0x411111f0: [N/A] Speaker at Ext Rear > Conn = 1/8, Color = Black > DefAssociation = 0xf, Sequence = 0x0 > Misc = NO_PRESENCE > Pin-ctls: 0x00: > Connection: 1 > 0x0f > Node 0x18 [Pin Complex] wcaps 0x40018f: Stereo Amp-In Amp-Out > Control: name="Mic Boost Volume", index=0, device=0 > ControlAmp: chs=3, dir=In, idx=0, ofs=0 > Control: name="Mic Jack", index=0, device=0 > Amp-In caps: ofs=0x00, nsteps=0x03, stepsize=0x2f, mute=0 > Amp-In vals: [0x00 0x00] > Amp-Out caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 > Amp-Out vals: [0x80 0x80] > Pincap 0x00001734: IN OUT Detect > Vref caps: HIZ 50 GRD 80 > Pin Default 0x01a19c20: [Jack] Mic at Ext Rear > Conn = 1/8, Color = Pink > DefAssociation = 0x2, Sequence = 0x0 > Pin-ctls: 0x24: IN VREF_80 > Unsolicited: tag=02, enabled=1 > Connection: 1 > 0x0d > Node 0x19 [Pin Complex] wcaps 0x40008b: Stereo Amp-In > Amp-In caps: ofs=0x00, nsteps=0x03, stepsize=0x2f, mute=0 > Amp-In vals: [0x00 0x00] > Pincap 0x00001724: IN Detect > Vref caps: HIZ 50 GRD 80 > Pin Default 0x411111f0: [N/A] Speaker at Ext Rear > Conn = 1/8, Color = Black > DefAssociation = 0xf, Sequence = 0x0 > Misc = NO_PRESENCE > Pin-ctls: 0x20: IN VREF_HIZ > Unsolicited: tag=00, enabled=0 > Node 0x1a [Pin Complex] wcaps 0x40018f: Stereo Amp-In Amp-Out > Amp-In caps: ofs=0x00, nsteps=0x03, stepsize=0x2f, mute=0 > Amp-In vals: [0x00 0x00] > Amp-Out caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 > Amp-Out vals: [0x80 0x80] > Pincap 0x0000003c: IN OUT HP Detect > Pin Default 0x411111f0: [N/A] Speaker at Ext Rear > Conn = 1/8, Color = Black > DefAssociation = 0xf, Sequence = 0x0 > Misc = NO_PRESENCE > Pin-ctls: 0x20: IN > Unsolicited: tag=00, enabled=0 > Connection: 2 > 0x0c* 0x0d > Node 0x1b [Pin Complex] wcaps 0x40018f: Stereo Amp-In Amp-Out > Amp-In caps: ofs=0x00, nsteps=0x03, stepsize=0x2f, mute=0 > Amp-In vals: [0x00 0x00] > Amp-Out caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 > Amp-Out vals: [0x80 0x80] > Pincap 0x00000034: IN OUT Detect > Pin Default 0x411111f0: [N/A] Speaker at Ext Rear > Conn = 1/8, Color = Black > DefAssociation = 0xf, Sequence = 0x0 > Misc = NO_PRESENCE > Pin-ctls: 0x20: IN > Unsolicited: tag=00, enabled=0 > Connection: 2 > 0x0c* 0x0d > Node 0x1c [Vendor Defined Widget] wcaps 0xf00000: Mono > Node 0x1d [Pin Complex] wcaps 0x400000: Mono > Pincap 0x00000020: IN > Pin Default 0x40079a2d: [N/A] Line Out at Ext N/A > Conn = Analog, Color = Pink > DefAssociation = 0x2, Sequence = 0xd > Pin-ctls: 0x20: IN > Node 0x1e [Pin Complex] wcaps 0x400381: Stereo Digital > Pincap 0x00000014: OUT Detect > Pin Default 0x411111f0: [N/A] Speaker at Ext Rear > Conn = 1/8, Color = Black > DefAssociation = 0xf, Sequence = 0x0 > Misc = NO_PRESENCE > Pin-ctls: 0x40: OUT > Unsolicited: tag=00, enabled=0 > Connection: 1 > 0x06 > Node 0x1f [Vendor Defined Widget] wcaps 0xf00000: Mono > Node 0x20 [Vendor Defined Widget] wcaps 0xf00040: Mono > Processing caps: benign=0, ncoeff=25 > Node 0x21 [Pin Complex] wcaps 0x40018d: Stereo Amp-Out > Control: name="Headphone Playback Switch", index=0, device=0 > ControlAmp: chs=3, dir=Out, idx=0, ofs=0 > Control: name="Headphone Jack", index=0, device=0 > Amp-Out caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 > Amp-Out vals: [0x00 0x00] > Pincap 0x0000001c: OUT HP Detect > Pin Default 0x0121401f: [Jack] HP Out at Ext Rear > Conn = 1/8, Color = Green > DefAssociation = 0x1, Sequence = 0xf > Pin-ctls: 0xc0: OUT HP > Unsolicited: tag=01, enabled=1 > Connection: 2 > 0x0c* 0x0d > Node 0x22 [Audio Selector] wcaps 0x30010b: Stereo Amp-In > Amp-In caps: N/A > Amp-In vals: [0x00 0x00] [0x00 0x00] [0x00 0x00] [0x00 0x00] [0x00 0x00] [0x00 0x00] [0x00 0x00] > Connection: 7 > 0x18 0x19 0x1a 0x1b 0x1d 0x0b 0x12* > Node 0x23 [Audio Mixer] wcaps 0x20010b: Stereo Amp-In > Amp-In caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 > Amp-In vals: [0x80 0x80] [0x80 0x80] [0x80 0x80] [0x80 0x80] [0x80 0x80] [0x80 0x80] > Connection: 6 > 0x18 0x19 0x1a 0x1b 0x1d 0x0b

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[alsa-devel] Infinite loop in snd_pcm_hw_htimestamp() for capture PCMs?
by Andrew Eikum 21 Sep '12

21 Sep '12

Hi folks, I've found some unexpected behavior in alsa-lib. The attached program uses 100% CPU in an infinite loop, when I'd expect it to return either an error or a valid htimestamp. I believe the loop occurs in <src/pcm/pcm_hw.c:snd_pcm_hw_htimestamp()>. Am I not supposed to call htimestamp() on this type of PCM? Is the PCM still in an invalid state when htimestamp() is called? Card: HDA ATI SB Chip: Realtek ALC889 Happy for any insight. Andrew

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[alsa-devel] [PATCH] ASoC: wm8960: remove 'dres' field from platform data structure
by Timur Tabi 21 Sep '12

21 Sep '12

The 'dres' field (discharge resistance for headphone outputs) is no longer used in the driver, so remove it. It was used in the original version of the driver when entering standby from off, but we stopped using it when we switched from having a single startup sequence to having separate cap and capless sequences. Signed-off-by: Timur Tabi <timur(a)freescale.com> --- include/sound/wm8960.h | 2 -- sound/soc/codecs/wm8960.c | 5 ----- 2 files changed, 0 insertions(+), 7 deletions(-) diff --git a/include/sound/wm8960.h b/include/sound/wm8960.h index b5a1ab9e..e8ce8ee 100644 --- a/include/sound/wm8960.h +++ b/include/sound/wm8960.h @@ -18,8 +18,6 @@ struct wm8960_data { bool capless; /* Headphone outputs configured in capless mode */ - int dres; /* Discharge resistance for headphone outputs */ - bool shared_lrclk; /* DAC and ADC LRCLKs are wired together */ }; diff --git a/sound/soc/codecs/wm8960.c b/sound/soc/codecs/wm8960.c index 782faa0..f0f6f660 100644 --- a/sound/soc/codecs/wm8960.c +++ b/sound/soc/codecs/wm8960.c @@ -962,11 +962,6 @@ static int wm8960_probe(struct snd_soc_codec *codec) if (!pdata) { dev_warn(codec->dev, "No platform data supplied\n"); } else { - if (pdata->dres > WM8960_DRES_MAX) { - dev_err(codec->dev, "Invalid DRES: %d\n", pdata->dres); - pdata->dres = 0; - } - if (pdata->capless) wm8960->set_bias_level = wm8960_set_bias_level_capless; } -- 1.7.3.4

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