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[alsa-devel] abdul, please add me to your LinkedIn network
by Maen Suleiman via LinkedIn 27 Sep '13

27 Sep '13

LinkedIn ------------ Maen Suleiman has indicated you are a Friend ------------------------------------------ Since you are a person I trust, I wanted to invite you to join my network on LinkedIn. Accept invitation from Maen Suleiman http://www.linkedin.com/e/jpxkk3-hm3ttpo1-7/uii-rtptP1H4ViCg9YiBUOvto1r4_5G… View profile of Maen Suleiman http://www.linkedin.com/e/jpxkk3-hm3ttpo1-7/rso/2181468/NG8u/name/138036396… ------------------------------------------ You are receiving Invitation emails. This email was intended for abdul moiz. Learn why this is included: http://www.linkedin.com/e/jpxkk3-hm3ttpo1-7/plh/http%3A%2F%2Fhelp%2Elinkedi… (c) 2012, LinkedIn Corporation. 2029 Stierlin Ct, Mountain View, CA 94043, USA.

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[alsa-devel] [PATCH] ASoc: kirkwood: Extend the min and max number of bytes per period
by Jean-Francois Moine 27 Sep '13

27 Sep '13

This patch extends the min and max number of bytes per period. It mainly permits to reduce the sound delay in MIDI real-time playing. Signed-off-by: Jean-Francois Moine <moinejf(a)free.fr> --- sound/soc/kirkwood/kirkwood.h | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/soc/kirkwood/kirkwood.h b/sound/soc/kirkwood/kirkwood.h index f8e1ccc..bf23afb 100644 --- a/sound/soc/kirkwood/kirkwood.h +++ b/sound/soc/kirkwood/kirkwood.h @@ -123,8 +123,8 @@ /* need to find where they come from */ #define KIRKWOOD_SND_MIN_PERIODS 8 #define KIRKWOOD_SND_MAX_PERIODS 16 -#define KIRKWOOD_SND_MIN_PERIOD_BYTES 0x4000 -#define KIRKWOOD_SND_MAX_PERIOD_BYTES 0x4000 +#define KIRKWOOD_SND_MIN_PERIOD_BYTES 0x800 +#define KIRKWOOD_SND_MAX_PERIOD_BYTES 0x8000 #define KIRKWOOD_SND_MAX_BUFFER_BYTES (KIRKWOOD_SND_MAX_PERIOD_BYTES \ * KIRKWOOD_SND_MAX_PERIODS) -- Ken ar c'hentañ | ** Breizh ha Linux atav! ** Jef | http://moinejf.free.fr/

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[alsa-devel] usb-audio mic problem with Polycom C100S
by Meelis Roos 27 Sep '13

27 Sep '13

I tried to use Skype (4.1) with Polycom Communicator C100S. Audio output works fine but mic seems to be dead, although it is set to capture and volume is even max. Dmesg tells the following two kinds of "cannot get freq at ep ..." messages while trying to use it. Linux is 3.12-rc2 but I saw the same behaviour a couple releases ago when I first tried it. I am not subscribed to alsa-devel, please CC in replies. dmesg [117273.437015] usb 1-1.2.3: new full-speed USB device number 9 using ehci-pci [117273.901033] usb 1-1.2.3: New USB device found, idVendor=095d, idProduct=0005 [117273.901036] usb 1-1.2.3: New USB device strings: Mfr=1, Product=2, SerialNumber=3 [117273.901037] usb 1-1.2.3: Product: Polycom Communicator [117273.901038] usb 1-1.2.3: Manufacturer: Polycom,Inc. [117273.901039] usb 1-1.2.3: SerialNumber: L80646021E21 [117273.962134] hid-generic 0003:095D:0005.0005: hiddev0,hidraw3: USB HID v1.00 Device [Polycom,Inc. Polycom Communicator] on usb-0000:00:1a.0-1.2.3/input3 [117273.974525] ALSA sound/usb/clock.c:309 9:2:1: cannot get freq at ep 0x1 [117273.978540] ALSA sound/usb/clock.c:309 9:1:1: cannot get freq at ep 0x82 [117274.019749] usbcore: registered new interface driver snd-usb-audio [117380.633919] ALSA sound/usb/clock.c:309 9:2:1: cannot get freq at ep 0x1 [117380.636540] ALSA sound/usb/clock.c:309 9:2:1: cannot get freq at ep 0x1 [117383.587846] ALSA sound/usb/clock.c:309 9:2:1: cannot get freq at ep 0x1 [117383.619463] ALSA sound/usb/clock.c:309 9:1:1: cannot get freq at ep 0x82 [117383.639484] ALSA sound/usb/clock.c:309 9:2:1: cannot get freq at ep 0x1 [117383.671451] ALSA sound/usb/clock.c:309 9:1:1: cannot get freq at ep 0x82 [117383.697219] ALSA sound/usb/clock.c:309 9:2:1: cannot get freq at ep 0x1 [117383.699840] ALSA sound/usb/clock.c:309 9:2:1: cannot get freq at ep 0x1 [117385.663752] ALSA sound/usb/clock.c:309 9:1:1: cannot get freq at ep 0x82 [117385.666495] ALSA sound/usb/clock.c:309 9:1:1: cannot get freq at ep 0x82 [117386.090400] ALSA sound/usb/clock.c:309 9:2:1: cannot get freq at ep 0x1 [117386.122894] ALSA sound/usb/clock.c:309 9:1:1: cannot get freq at ep 0x82 [117386.143015] ALSA sound/usb/clock.c:309 9:2:1: cannot get freq at ep 0x1 [117386.174880] ALSA sound/usb/clock.c:309 9:1:1: cannot get freq at ep 0x82 [117386.200626] ALSA sound/usb/clock.c:309 9:2:1: cannot get freq at ep 0x1 [117386.203252] ALSA sound/usb/clock.c:309 9:2:1: cannot get freq at ep 0x1 [117392.696024] ALSA sound/usb/clock.c:309 9:2:1: cannot get freq at ep 0x1 [117392.698650] ALSA sound/usb/clock.c:309 9:2:1: cannot get freq at ep 0x1 [117395.407411] ALSA sound/usb/clock.c:309 9:1:1: cannot get freq at ep 0x82 [117395.410160] ALSA sound/usb/clock.c:309 9:1:1: cannot get freq at ep 0x82 [117449.635703] ALSA sound/usb/clock.c:309 9:2:1: cannot get freq at ep 0x1 [117449.638451] ALSA sound/usb/clock.c:309 9:2:1: cannot get freq at ep 0x1 lsusb -v Bus 001 Device 008: ID 095d:0005 Polycom, Inc. Device Descriptor: bLength 18 bDescriptorType 1 bcdUSB 1.10 bDeviceClass 0 (Defined at Interface level) bDeviceSubClass 0 bDeviceProtocol 0 bMaxPacketSize0 8 idVendor 0x095d Polycom, Inc. idProduct 0x0005 bcdDevice 1.00 iManufacturer 1 iProduct 2 iSerial 3 bNumConfigurations 1 Configuration Descriptor: bLength 9 bDescriptorType 2 wTotalLength 231 bNumInterfaces 4 bConfigurationValue 1 iConfiguration 0 bmAttributes 0x80 (Bus Powered) MaxPower 500mA Interface Descriptor: bLength 9 bDescriptorType 4 bInterfaceNumber 0 bAlternateSetting 0 bNumEndpoints 0 bInterfaceClass 1 Audio bInterfaceSubClass 1 Control Device bInterfaceProtocol 0 iInterface 0 AudioControl Interface Descriptor: bLength 10 bDescriptorType 36 bDescriptorSubtype 1 (HEADER) bcdADC 1.00 wTotalLength 83 bInCollection 2 baInterfaceNr( 0) 2 baInterfaceNr( 1) 1 AudioControl Interface Descriptor: bLength 12 bDescriptorType 36 bDescriptorSubtype 2 (INPUT_TERMINAL) bTerminalID 1 wTerminalType 0x0201 Microphone 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 2 bSourceID 1 bControlSize 1 bmaControls( 0) 0x03 Mute Control Volume Control bmaControls( 1) 0x00 bmaControls( 2) 0x00 iFeature 0 AudioControl Interface Descriptor: bLength 12 bDescriptorType 36 bDescriptorSubtype 4 (MIXER_UNIT) bUnitID 3 bNrInPins 1 baSourceID( 0) 2 bNrChannels 2 wChannelConfig 0x0003 Left Front (L) Right Front (R) iChannelNames 0 bmControls 0x00 iMixer 0 AudioControl Interface Descriptor: bLength 9 bDescriptorType 36 bDescriptorSubtype 3 (OUTPUT_TERMINAL) bTerminalID 4 wTerminalType 0x0101 USB Streaming bAssocTerminal 0 bSourceID 3 iTerminal 0 AudioControl Interface Descriptor: bLength 12 bDescriptorType 36 bDescriptorSubtype 2 (INPUT_TERMINAL) bTerminalID 5 wTerminalType 0x0101 USB Streaming 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 6 bSourceID 5 bControlSize 1 bmaControls( 0) 0x03 Mute Control Volume Control bmaControls( 1) 0x00 bmaControls( 2) 0x00 iFeature 0 AudioControl Interface Descriptor: bLength 9 bDescriptorType 36 bDescriptorSubtype 3 (OUTPUT_TERMINAL) bTerminalID 7 wTerminalType 0x0301 Speaker bAssocTerminal 0 bSourceID 6 iTerminal 0 Interface Descriptor: bLength 9 bDescriptorType 4 bInterfaceNumber 1 bAlternateSetting 0 bNumEndpoints 0 bInterfaceClass 1 Audio bInterfaceSubClass 2 Streaming bInterfaceProtocol 0 iInterface 0 Interface Descriptor: bLength 9 bDescriptorType 4 bInterfaceNumber 1 bAlternateSetting 1 bNumEndpoints 1 bInterfaceClass 1 Audio bInterfaceSubClass 2 Streaming bInterfaceProtocol 0 iInterface 0 AudioStreaming Interface Descriptor: bLength 7 bDescriptorType 36 bDescriptorSubtype 1 (AS_GENERAL) bTerminalLink 4 bDelay 1 frames wFormatTag 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 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 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 5 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 13 Transfer Type Isochronous Synch Type Synchronous 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 3 bAlternateSetting 0 bNumEndpoints 1 bInterfaceClass 3 Human Interface Device bInterfaceSubClass 0 No Subclass bInterfaceProtocol 0 None iInterface 0 HID Device Descriptor: bLength 9 bDescriptorType 33 bcdHID 1.00 bCountryCode 0 Not supported bNumDescriptors 1 bDescriptorType 34 Report wDescriptorLength 52 Report Descriptors: ** UNAVAILABLE ** 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 -- Meelis Roos (mroos(a)linux.ee)

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[alsa-devel] [PATCH] ASoC: adav80x: Convert to direct regmap usage
by Lars-Peter Clausen 27 Sep '13

27 Sep '13

Signed-off-by: Lars-Peter Clausen <lars(a)metafoo.de> --- sound/soc/codecs/adav80x.c | 147 +++++++++++++++++++++++++++++++-------------- 1 file changed, 102 insertions(+), 45 deletions(-) diff --git a/sound/soc/codecs/adav80x.c b/sound/soc/codecs/adav80x.c index 15b012d0..14a7c16 100644 --- a/sound/soc/codecs/adav80x.c +++ b/sound/soc/codecs/adav80x.c @@ -115,22 +115,34 @@ #define ADAV80X_PLL_OUTE_SYSCLKPD(x) BIT(2 - (x)) -static u8 adav80x_default_regs[] = { - 0x00, 0x00, 0x00, 0x00, 0x01, 0x01, 0x02, 0x01, 0x80, 0x26, 0x00, 0x00, - 0x02, 0x40, 0x20, 0x00, 0x09, 0x08, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, - 0x04, 0x00, 0x01, 0x00, 0x00, 0x00, 0x00, 0x00, 0xd1, 0x92, 0xb1, 0x37, - 0x48, 0xd2, 0xfb, 0xca, 0xd2, 0x15, 0xe8, 0x29, 0xb9, 0x6a, 0xda, 0x2b, - 0xb7, 0xc0, 0x11, 0x65, 0x5c, 0xf6, 0xff, 0x8d, 0x00, 0x00, 0x00, 0x00, - 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, - 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0xa5, 0x00, 0x00, - 0x00, 0xe8, 0x46, 0xe1, 0x5b, 0xd3, 0x43, 0x77, 0x93, 0xa7, 0x44, 0xee, - 0x32, 0x12, 0xc0, 0x11, 0x00, 0x00, 0x00, 0x00, 0xff, 0xff, 0x3f, 0x3f, - 0x00, 0x00, 0x00, 0x00, 0xff, 0xff, 0x00, 0x1d, 0x00, 0x00, 0x00, 0x00, - 0x00, 0x00, 0x00, 0x00, 0x52, 0x00, +static struct reg_default adav80x_reg_defaults[] = { + { ADAV80X_PLAYBACK_CTRL, 0x01 }, + { ADAV80X_AUX_IN_CTRL, 0x01 }, + { ADAV80X_REC_CTRL, 0x02 }, + { ADAV80X_AUX_OUT_CTRL, 0x01 }, + { ADAV80X_DPATH_CTRL1, 0xc0 }, + { ADAV80X_DPATH_CTRL2, 0x11 }, + { ADAV80X_DAC_CTRL1, 0x00 }, + { ADAV80X_DAC_CTRL2, 0x00 }, + { ADAV80X_DAC_CTRL3, 0x00 }, + { ADAV80X_DAC_L_VOL, 0xff }, + { ADAV80X_DAC_R_VOL, 0xff }, + { ADAV80X_PGA_L_VOL, 0x00 }, + { ADAV80X_PGA_R_VOL, 0x00 }, + { ADAV80X_ADC_CTRL1, 0x00 }, + { ADAV80X_ADC_CTRL2, 0x00 }, + { ADAV80X_ADC_L_VOL, 0xff }, + { ADAV80X_ADC_R_VOL, 0xff }, + { ADAV80X_PLL_CTRL1, 0x00 }, + { ADAV80X_PLL_CTRL2, 0x00 }, + { ADAV80X_ICLK_CTRL1, 0x00 }, + { ADAV80X_ICLK_CTRL2, 0x00 }, + { ADAV80X_PLL_CLK_SRC, 0x00 }, + { ADAV80X_PLL_OUTE, 0x00 }, }; struct adav80x { - enum snd_soc_control_type control_type; + struct regmap *regmap; enum adav80x_clk_src clk_src; unsigned int sysclk; @@ -298,7 +310,7 @@ static int adav80x_set_deemph(struct snd_soc_codec *codec) val = ADAV80X_DAC_CTRL2_DEEMPH_NONE; } - return snd_soc_update_bits(codec, ADAV80X_DAC_CTRL2, + return regmap_update_bits(adav80x->regmap, ADAV80X_DAC_CTRL2, ADAV80X_DAC_CTRL2_DEEMPH_MASK, val); } @@ -394,10 +406,11 @@ static int adav80x_set_dai_fmt(struct snd_soc_dai *dai, unsigned int fmt) return -EINVAL; } - snd_soc_update_bits(codec, adav80x_port_ctrl_regs[dai->id][0], + regmap_update_bits(adav80x->regmap, adav80x_port_ctrl_regs[dai->id][0], ADAV80X_CAPTURE_MODE_MASK | ADAV80X_CAPTURE_MODE_MASTER, capture); - snd_soc_write(codec, adav80x_port_ctrl_regs[dai->id][1], playback); + regmap_write(adav80x->regmap, adav80x_port_ctrl_regs[dai->id][1], + playback); adav80x->dai_fmt[dai->id] = fmt & SND_SOC_DAIFMT_FORMAT_MASK; @@ -407,6 +420,7 @@ static int adav80x_set_dai_fmt(struct snd_soc_dai *dai, unsigned int fmt) static int adav80x_set_adc_clock(struct snd_soc_codec *codec, unsigned int sample_rate) { + struct adav80x *adav80x = snd_soc_codec_get_drvdata(codec); unsigned int val; if (sample_rate <= 48000) @@ -414,7 +428,7 @@ static int adav80x_set_adc_clock(struct snd_soc_codec *codec, else val = ADAV80X_ADC_CTRL1_MODULATOR_64FS; - snd_soc_update_bits(codec, ADAV80X_ADC_CTRL1, + regmap_update_bits(adav80x->regmap, ADAV80X_ADC_CTRL1, ADAV80X_ADC_CTRL1_MODULATOR_MASK, val); return 0; @@ -423,6 +437,7 @@ static int adav80x_set_adc_clock(struct snd_soc_codec *codec, static int adav80x_set_dac_clock(struct snd_soc_codec *codec, unsigned int sample_rate) { + struct adav80x *adav80x = snd_soc_codec_get_drvdata(codec); unsigned int val; if (sample_rate <= 48000) @@ -430,7 +445,7 @@ static int adav80x_set_dac_clock(struct snd_soc_codec *codec, else val = ADAV80X_DAC_CTRL2_DIV2 | ADAV80X_DAC_CTRL2_INTERPOL_128FS; - snd_soc_update_bits(codec, ADAV80X_DAC_CTRL2, + regmap_update_bits(adav80x->regmap, ADAV80X_DAC_CTRL2, ADAV80X_DAC_CTRL2_DIV_MASK | ADAV80X_DAC_CTRL2_INTERPOL_MASK, val); @@ -440,6 +455,7 @@ static int adav80x_set_dac_clock(struct snd_soc_codec *codec, static int adav80x_set_capture_pcm_format(struct snd_soc_codec *codec, struct snd_soc_dai *dai, snd_pcm_format_t format) { + struct adav80x *adav80x = snd_soc_codec_get_drvdata(codec); unsigned int val; switch (format) { @@ -459,7 +475,7 @@ static int adav80x_set_capture_pcm_format(struct snd_soc_codec *codec, return -EINVAL; } - snd_soc_update_bits(codec, adav80x_port_ctrl_regs[dai->id][0], + regmap_update_bits(adav80x->regmap, adav80x_port_ctrl_regs[dai->id][0], ADAV80X_CAPTURE_WORD_LEN_MASK, val); return 0; @@ -491,7 +507,7 @@ static int adav80x_set_playback_pcm_format(struct snd_soc_codec *codec, return -EINVAL; } - snd_soc_update_bits(codec, adav80x_port_ctrl_regs[dai->id][1], + regmap_update_bits(adav80x->regmap, adav80x_port_ctrl_regs[dai->id][1], ADAV80X_PLAYBACK_MODE_MASK, val); return 0; @@ -554,8 +570,10 @@ static int adav80x_set_sysclk(struct snd_soc_codec *codec, ADAV80X_ICLK_CTRL1_ICLK2_SRC(clk_id); iclk_ctrl2 = ADAV80X_ICLK_CTRL2_ICLK1_SRC(clk_id); - snd_soc_write(codec, ADAV80X_ICLK_CTRL1, iclk_ctrl1); - snd_soc_write(codec, ADAV80X_ICLK_CTRL2, iclk_ctrl2); + regmap_write(adav80x->regmap, ADAV80X_ICLK_CTRL1, + iclk_ctrl1); + regmap_write(adav80x->regmap, ADAV80X_ICLK_CTRL2, + iclk_ctrl2); snd_soc_dapm_sync(&codec->dapm); } @@ -575,10 +593,12 @@ static int adav80x_set_sysclk(struct snd_soc_codec *codec, mask = ADAV80X_PLL_OUTE_SYSCLKPD(clk_id); if (freq == 0) { - snd_soc_update_bits(codec, ADAV80X_PLL_OUTE, mask, mask); + regmap_update_bits(adav80x->regmap, ADAV80X_PLL_OUTE, + mask, mask); adav80x->sysclk_pd[clk_id] = true; } else { - snd_soc_update_bits(codec, ADAV80X_PLL_OUTE, mask, 0); + regmap_update_bits(adav80x->regmap, ADAV80X_PLL_OUTE, + mask, 0); adav80x->sysclk_pd[clk_id] = false; } @@ -650,9 +670,9 @@ static int adav80x_set_pll(struct snd_soc_codec *codec, int pll_id, return -EINVAL; } - snd_soc_update_bits(codec, ADAV80X_PLL_CTRL1, ADAV80X_PLL_CTRL1_PLLDIV, - pll_ctrl1); - snd_soc_update_bits(codec, ADAV80X_PLL_CTRL2, + regmap_update_bits(adav80x->regmap, ADAV80X_PLL_CTRL1, + ADAV80X_PLL_CTRL1_PLLDIV, pll_ctrl1); + regmap_update_bits(adav80x->regmap, ADAV80X_PLL_CTRL2, ADAV80X_PLL_CTRL2_PLL_MASK(pll_id), pll_ctrl2); if (source != adav80x->pll_src) { @@ -661,7 +681,7 @@ static int adav80x_set_pll(struct snd_soc_codec *codec, int pll_id, else pll_src = ADAV80X_PLL_CLK_SRC_PLL_XIN(pll_id); - snd_soc_update_bits(codec, ADAV80X_PLL_CLK_SRC, + regmap_update_bits(adav80x->regmap, ADAV80X_PLL_CLK_SRC, ADAV80X_PLL_CLK_SRC_PLL_MASK(pll_id), pll_src); adav80x->pll_src = source; @@ -675,6 +695,7 @@ static int adav80x_set_pll(struct snd_soc_codec *codec, int pll_id, static int adav80x_set_bias_level(struct snd_soc_codec *codec, enum snd_soc_bias_level level) { + struct adav80x *adav80x = snd_soc_codec_get_drvdata(codec); unsigned int mask = ADAV80X_DAC_CTRL1_PD; switch (level) { @@ -683,10 +704,12 @@ static int adav80x_set_bias_level(struct snd_soc_codec *codec, case SND_SOC_BIAS_PREPARE: break; case SND_SOC_BIAS_STANDBY: - snd_soc_update_bits(codec, ADAV80X_DAC_CTRL1, mask, 0x00); + regmap_update_bits(adav80x->regmap, ADAV80X_DAC_CTRL1, mask, + 0x00); break; case SND_SOC_BIAS_OFF: - snd_soc_update_bits(codec, ADAV80X_DAC_CTRL1, mask, mask); + regmap_update_bits(adav80x->regmap, ADAV80X_DAC_CTRL1, mask, + mask); break; } @@ -780,7 +803,7 @@ static int adav80x_probe(struct snd_soc_codec *codec) int ret; struct adav80x *adav80x = snd_soc_codec_get_drvdata(codec); - ret = snd_soc_codec_set_cache_io(codec, 7, 9, adav80x->control_type); + ret = snd_soc_codec_set_cache_io(codec, 0, 0, SND_SOC_REGMAP); if (ret) { dev_err(codec->dev, "failed to set cache I/O: %d\n", ret); return ret; @@ -791,23 +814,31 @@ static int adav80x_probe(struct snd_soc_codec *codec) snd_soc_dapm_force_enable_pin(&codec->dapm, "PLL2"); /* Power down S/PDIF receiver, since it is currently not supported */ - snd_soc_write(codec, ADAV80X_PLL_OUTE, 0x20); + regmap_write(adav80x->regmap, ADAV80X_PLL_OUTE, 0x20); /* Disable DAC zero flag */ - snd_soc_write(codec, ADAV80X_DAC_CTRL3, 0x6); + regmap_write(adav80x->regmap, ADAV80X_DAC_CTRL3, 0x6); return adav80x_set_bias_level(codec, SND_SOC_BIAS_STANDBY); } static int adav80x_suspend(struct snd_soc_codec *codec) { - return adav80x_set_bias_level(codec, SND_SOC_BIAS_OFF); + struct adav80x *adav80x = snd_soc_codec_get_drvdata(codec); + int ret; + + ret = adav80x_set_bias_level(codec, SND_SOC_BIAS_OFF); + regcache_cache_only(adav80x->regmap, true); + + return ret; } static int adav80x_resume(struct snd_soc_codec *codec) { + struct adav80x *adav80x = snd_soc_codec_get_drvdata(codec); + + regcache_cache_only(adav80x->regmap, false); adav80x_set_bias_level(codec, SND_SOC_BIAS_STANDBY); - codec->cache_sync = 1; - snd_soc_cache_sync(codec); + regcache_sync(adav80x->regmap); return 0; } @@ -827,10 +858,6 @@ static struct snd_soc_codec_driver adav80x_codec_driver = { .set_pll = adav80x_set_pll, .set_sysclk = adav80x_set_sysclk, - .reg_word_size = sizeof(u8), - .reg_cache_size = ARRAY_SIZE(adav80x_default_regs), - .reg_cache_default = adav80x_default_regs, - .controls = adav80x_controls, .num_controls = ARRAY_SIZE(adav80x_controls), .dapm_widgets = adav80x_dapm_widgets, @@ -839,18 +866,21 @@ static struct snd_soc_codec_driver adav80x_codec_driver = { .num_dapm_routes = ARRAY_SIZE(adav80x_dapm_routes), }; -static int adav80x_bus_probe(struct device *dev, - enum snd_soc_control_type control_type) +static int adav80x_bus_probe(struct device *dev, struct regmap *regmap) { struct adav80x *adav80x; int ret; + if (IS_ERR(regmap)) + return PTR_ERR(regmap); + adav80x = kzalloc(sizeof(*adav80x), GFP_KERNEL); if (!adav80x) return -ENOMEM; + dev_set_drvdata(dev, adav80x); - adav80x->control_type = control_type; + adav80x->regmap = regmap; ret = snd_soc_register_codec(dev, &adav80x_codec_driver, adav80x_dais, ARRAY_SIZE(adav80x_dais)); @@ -868,6 +898,19 @@ static int adav80x_bus_remove(struct device *dev) } #if defined(CONFIG_SPI_MASTER) +static const struct regmap_config adav80x_spi_regmap_config = { + .val_bits = 8, + .pad_bits = 1, + .reg_bits = 7, + .read_flag_mask = 0x01, + + .max_register = ADAV80X_PLL_OUTE, + + .cache_type = REGCACHE_RBTREE, + .reg_defaults = adav80x_reg_defaults, + .num_reg_defaults = ARRAY_SIZE(adav80x_reg_defaults), +}; + static const struct spi_device_id adav80x_spi_id[] = { { "adav801", 0 }, { } @@ -876,7 +919,8 @@ MODULE_DEVICE_TABLE(spi, adav80x_spi_id); static int adav80x_spi_probe(struct spi_device *spi) { - return adav80x_bus_probe(&spi->dev, SND_SOC_SPI); + return adav80x_bus_probe(&spi->dev, + devm_regmap_init_spi(spi, &adav80x_spi_regmap_config)); } static int adav80x_spi_remove(struct spi_device *spi) @@ -896,6 +940,18 @@ static struct spi_driver adav80x_spi_driver = { #endif #if defined(CONFIG_I2C) || defined(CONFIG_I2C_MODULE) +static const struct regmap_config adav80x_i2c_regmap_config = { + .val_bits = 8, + .pad_bits = 1, + .reg_bits = 7, + + .max_register = ADAV80X_PLL_OUTE, + + .cache_type = REGCACHE_RBTREE, + .reg_defaults = adav80x_reg_defaults, + .num_reg_defaults = ARRAY_SIZE(adav80x_reg_defaults), +}; + static const struct i2c_device_id adav80x_i2c_id[] = { { "adav803", 0 }, { } @@ -905,7 +961,8 @@ MODULE_DEVICE_TABLE(i2c, adav80x_i2c_id); static int adav80x_i2c_probe(struct i2c_client *client, const struct i2c_device_id *id) { - return adav80x_bus_probe(&client->dev, SND_SOC_I2C); + return adav80x_bus_probe(&client->dev, + devm_regmap_init_i2c(client, &adav80x_i2c_regmap_config)); } static int adav80x_i2c_remove(struct i2c_client *client) -- 1.8.0

2 1

[alsa-devel] [PATCH 1/3] ASoC: adau1373: Convert to direct regmap usage
by Lars-Peter Clausen 27 Sep '13

27 Sep '13

Signed-off-by: Lars-Peter Clausen <lars(a)metafoo.de> --- sound/soc/codecs/adau1373.c | 295 +++++++++++++++++++++++++++++++++++--------- 1 file changed, 235 insertions(+), 60 deletions(-) diff --git a/sound/soc/codecs/adau1373.c b/sound/soc/codecs/adau1373.c index 1aa10dd..c57c1f8 100644 --- a/sound/soc/codecs/adau1373.c +++ b/sound/soc/codecs/adau1373.c @@ -32,6 +32,7 @@ struct adau1373_dai { }; struct adau1373 { + struct regmap *regmap; struct adau1373_dai dais[3]; }; @@ -152,37 +153,172 @@ struct adau1373 { #define ADAU1373_EP_CTRL_MICBIAS1_OFFSET 4 #define ADAU1373_EP_CTRL_MICBIAS2_OFFSET 2 -static const uint8_t adau1373_default_regs[] = { - 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, /* 0x00 */ - 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, - 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, /* 0x10 */ - 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, - 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, /* 0x20 */ - 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x02, 0x00, - 0x00, 0x00, 0x00, 0x00, 0x00, 0x02, 0x00, 0x00, /* 0x30 */ - 0x00, 0x00, 0x00, 0x80, 0x00, 0x01, 0x00, 0x00, - 0x00, 0x00, 0x00, 0x00, 0x0a, 0x0a, 0x0a, 0x00, /* 0x40 */ - 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, - 0x00, 0x08, 0x08, 0x08, 0x00, 0x00, 0x00, 0x00, /* 0x50 */ - 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, - 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, /* 0x60 */ - 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, - 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, /* 0x70 */ - 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, - 0x78, 0x18, 0x00, 0x00, 0x00, 0xc0, 0x00, 0x00, /* 0x80 */ - 0x00, 0xc0, 0x88, 0x7a, 0xdf, 0x20, 0x00, 0x00, - 0x78, 0x18, 0x00, 0x00, 0x00, 0xc0, 0x00, 0x00, /* 0x90 */ - 0x00, 0xc0, 0x88, 0x7a, 0xdf, 0x20, 0x00, 0x00, - 0x78, 0x18, 0x00, 0x00, 0x00, 0xc0, 0x00, 0x00, /* 0xa0 */ - 0x00, 0xc0, 0x88, 0x7a, 0xdf, 0x20, 0x00, 0x00, - 0x00, 0x00, 0x00, 0xff, 0xff, 0xff, 0xff, 0xff, /* 0xb0 */ - 0xff, 0xff, 0xff, 0xff, 0xff, 0x1f, 0x00, 0x00, - 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, /* 0xc0 */ - 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, - 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, /* 0xd0 */ - 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, - 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x02, 0x00, /* 0xe0 */ - 0x00, 0x1f, 0x0f, 0x00, 0x00, +static const struct reg_default adau1373_reg_defaults[] = { + { ADAU1373_INPUT_MODE, 0x00 }, + { ADAU1373_AINL_CTRL(0), 0x00 }, + { ADAU1373_AINR_CTRL(0), 0x00 }, + { ADAU1373_AINL_CTRL(1), 0x00 }, + { ADAU1373_AINR_CTRL(1), 0x00 }, + { ADAU1373_AINL_CTRL(2), 0x00 }, + { ADAU1373_AINR_CTRL(2), 0x00 }, + { ADAU1373_AINL_CTRL(3), 0x00 }, + { ADAU1373_AINR_CTRL(3), 0x00 }, + { ADAU1373_LLINE_OUT(0), 0x00 }, + { ADAU1373_RLINE_OUT(0), 0x00 }, + { ADAU1373_LLINE_OUT(1), 0x00 }, + { ADAU1373_RLINE_OUT(1), 0x00 }, + { ADAU1373_LSPK_OUT, 0x00 }, + { ADAU1373_RSPK_OUT, 0x00 }, + { ADAU1373_LHP_OUT, 0x00 }, + { ADAU1373_RHP_OUT, 0x00 }, + { ADAU1373_ADC_GAIN, 0x00 }, + { ADAU1373_LADC_MIXER, 0x00 }, + { ADAU1373_RADC_MIXER, 0x00 }, + { ADAU1373_LLINE1_MIX, 0x00 }, + { ADAU1373_RLINE1_MIX, 0x00 }, + { ADAU1373_LLINE2_MIX, 0x00 }, + { ADAU1373_RLINE2_MIX, 0x00 }, + { ADAU1373_LSPK_MIX, 0x00 }, + { ADAU1373_RSPK_MIX, 0x00 }, + { ADAU1373_LHP_MIX, 0x00 }, + { ADAU1373_RHP_MIX, 0x00 }, + { ADAU1373_EP_MIX, 0x00 }, + { ADAU1373_HP_CTRL, 0x00 }, + { ADAU1373_HP_CTRL2, 0x00 }, + { ADAU1373_LS_CTRL, 0x00 }, + { ADAU1373_EP_CTRL, 0x00 }, + { ADAU1373_MICBIAS_CTRL1, 0x00 }, + { ADAU1373_MICBIAS_CTRL2, 0x00 }, + { ADAU1373_OUTPUT_CTRL, 0x00 }, + { ADAU1373_PWDN_CTRL1, 0x00 }, + { ADAU1373_PWDN_CTRL2, 0x00 }, + { ADAU1373_PWDN_CTRL3, 0x00 }, + { ADAU1373_DPLL_CTRL(0), 0x00 }, + { ADAU1373_PLL_CTRL1(0), 0x00 }, + { ADAU1373_PLL_CTRL2(0), 0x00 }, + { ADAU1373_PLL_CTRL3(0), 0x00 }, + { ADAU1373_PLL_CTRL4(0), 0x00 }, + { ADAU1373_PLL_CTRL5(0), 0x00 }, + { ADAU1373_PLL_CTRL6(0), 0x02 }, + { ADAU1373_DPLL_CTRL(1), 0x00 }, + { ADAU1373_PLL_CTRL1(1), 0x00 }, + { ADAU1373_PLL_CTRL2(1), 0x00 }, + { ADAU1373_PLL_CTRL3(1), 0x00 }, + { ADAU1373_PLL_CTRL4(1), 0x00 }, + { ADAU1373_PLL_CTRL5(1), 0x00 }, + { ADAU1373_PLL_CTRL6(1), 0x02 }, + { ADAU1373_HEADDECT, 0x00 }, + { ADAU1373_ADC_CTRL, 0x00 }, + { ADAU1373_CLK_SRC_DIV(0), 0x00 }, + { ADAU1373_CLK_SRC_DIV(1), 0x00 }, + { ADAU1373_DAI(0), 0x0a }, + { ADAU1373_DAI(1), 0x0a }, + { ADAU1373_DAI(2), 0x0a }, + { ADAU1373_BCLKDIV(0), 0x00 }, + { ADAU1373_BCLKDIV(1), 0x00 }, + { ADAU1373_BCLKDIV(2), 0x00 }, + { ADAU1373_SRC_RATIOA(0), 0x00 }, + { ADAU1373_SRC_RATIOB(0), 0x00 }, + { ADAU1373_SRC_RATIOA(1), 0x00 }, + { ADAU1373_SRC_RATIOB(1), 0x00 }, + { ADAU1373_SRC_RATIOA(2), 0x00 }, + { ADAU1373_SRC_RATIOB(2), 0x00 }, + { ADAU1373_DEEMP_CTRL, 0x00 }, + { ADAU1373_SRC_DAI_CTRL(0), 0x08 }, + { ADAU1373_SRC_DAI_CTRL(1), 0x08 }, + { ADAU1373_SRC_DAI_CTRL(2), 0x08 }, + { ADAU1373_DIN_MIX_CTRL(0), 0x00 }, + { ADAU1373_DIN_MIX_CTRL(1), 0x00 }, + { ADAU1373_DIN_MIX_CTRL(2), 0x00 }, + { ADAU1373_DIN_MIX_CTRL(3), 0x00 }, + { ADAU1373_DIN_MIX_CTRL(4), 0x00 }, + { ADAU1373_DOUT_MIX_CTRL(0), 0x00 }, + { ADAU1373_DOUT_MIX_CTRL(1), 0x00 }, + { ADAU1373_DOUT_MIX_CTRL(2), 0x00 }, + { ADAU1373_DOUT_MIX_CTRL(3), 0x00 }, + { ADAU1373_DOUT_MIX_CTRL(4), 0x00 }, + { ADAU1373_DAI_PBL_VOL(0), 0x00 }, + { ADAU1373_DAI_PBR_VOL(0), 0x00 }, + { ADAU1373_DAI_PBL_VOL(1), 0x00 }, + { ADAU1373_DAI_PBR_VOL(1), 0x00 }, + { ADAU1373_DAI_PBL_VOL(2), 0x00 }, + { ADAU1373_DAI_PBR_VOL(2), 0x00 }, + { ADAU1373_DAI_RECL_VOL(0), 0x00 }, + { ADAU1373_DAI_RECR_VOL(0), 0x00 }, + { ADAU1373_DAI_RECL_VOL(1), 0x00 }, + { ADAU1373_DAI_RECR_VOL(1), 0x00 }, + { ADAU1373_DAI_RECL_VOL(2), 0x00 }, + { ADAU1373_DAI_RECR_VOL(2), 0x00 }, + { ADAU1373_DAC1_PBL_VOL, 0x00 }, + { ADAU1373_DAC1_PBR_VOL, 0x00 }, + { ADAU1373_DAC2_PBL_VOL, 0x00 }, + { ADAU1373_DAC2_PBR_VOL, 0x00 }, + { ADAU1373_ADC_RECL_VOL, 0x00 }, + { ADAU1373_ADC_RECR_VOL, 0x00 }, + { ADAU1373_DMIC_RECL_VOL, 0x00 }, + { ADAU1373_DMIC_RECR_VOL, 0x00 }, + { ADAU1373_VOL_GAIN1, 0x00 }, + { ADAU1373_VOL_GAIN2, 0x00 }, + { ADAU1373_VOL_GAIN3, 0x00 }, + { ADAU1373_HPF_CTRL, 0x00 }, + { ADAU1373_BASS1, 0x00 }, + { ADAU1373_BASS2, 0x00 }, + { ADAU1373_DRC(0) + 0x0, 0x78 }, + { ADAU1373_DRC(0) + 0x1, 0x18 }, + { ADAU1373_DRC(0) + 0x2, 0x00 }, + { ADAU1373_DRC(0) + 0x3, 0x00 }, + { ADAU1373_DRC(0) + 0x4, 0x00 }, + { ADAU1373_DRC(0) + 0x5, 0xc0 }, + { ADAU1373_DRC(0) + 0x6, 0x00 }, + { ADAU1373_DRC(0) + 0x7, 0x00 }, + { ADAU1373_DRC(0) + 0x8, 0x00 }, + { ADAU1373_DRC(0) + 0x9, 0xc0 }, + { ADAU1373_DRC(0) + 0xa, 0x88 }, + { ADAU1373_DRC(0) + 0xb, 0x7a }, + { ADAU1373_DRC(0) + 0xc, 0xdf }, + { ADAU1373_DRC(0) + 0xd, 0x20 }, + { ADAU1373_DRC(0) + 0xe, 0x00 }, + { ADAU1373_DRC(0) + 0xf, 0x00 }, + { ADAU1373_DRC(1) + 0x0, 0x78 }, + { ADAU1373_DRC(1) + 0x1, 0x18 }, + { ADAU1373_DRC(1) + 0x2, 0x00 }, + { ADAU1373_DRC(1) + 0x3, 0x00 }, + { ADAU1373_DRC(1) + 0x4, 0x00 }, + { ADAU1373_DRC(1) + 0x5, 0xc0 }, + { ADAU1373_DRC(1) + 0x6, 0x00 }, + { ADAU1373_DRC(1) + 0x7, 0x00 }, + { ADAU1373_DRC(1) + 0x8, 0x00 }, + { ADAU1373_DRC(1) + 0x9, 0xc0 }, + { ADAU1373_DRC(1) + 0xa, 0x88 }, + { ADAU1373_DRC(1) + 0xb, 0x7a }, + { ADAU1373_DRC(1) + 0xc, 0xdf }, + { ADAU1373_DRC(1) + 0xd, 0x20 }, + { ADAU1373_DRC(1) + 0xe, 0x00 }, + { ADAU1373_DRC(1) + 0xf, 0x00 }, + { ADAU1373_DRC(2) + 0x0, 0x78 }, + { ADAU1373_DRC(2) + 0x1, 0x18 }, + { ADAU1373_DRC(2) + 0x2, 0x00 }, + { ADAU1373_DRC(2) + 0x3, 0x00 }, + { ADAU1373_DRC(2) + 0x4, 0x00 }, + { ADAU1373_DRC(2) + 0x5, 0xc0 }, + { ADAU1373_DRC(2) + 0x6, 0x00 }, + { ADAU1373_DRC(2) + 0x7, 0x00 }, + { ADAU1373_DRC(2) + 0x8, 0x00 }, + { ADAU1373_DRC(2) + 0x9, 0xc0 }, + { ADAU1373_DRC(2) + 0xa, 0x88 }, + { ADAU1373_DRC(2) + 0xb, 0x7a }, + { ADAU1373_DRC(2) + 0xc, 0xdf }, + { ADAU1373_DRC(2) + 0xd, 0x20 }, + { ADAU1373_DRC(2) + 0xe, 0x00 }, + { ADAU1373_DRC(2) + 0xf, 0x00 }, + { ADAU1373_3D_CTRL1, 0x00 }, + { ADAU1373_3D_CTRL2, 0x00 }, + { ADAU1373_FDSP_SEL1, 0x00 }, + { ADAU1373_FDSP_SEL2, 0x00 }, + { ADAU1373_FDSP_SEL2, 0x00 }, + { ADAU1373_FDSP_SEL4, 0x00 }, + { ADAU1373_DIGMICCTRL, 0x00 }, + { ADAU1373_DIGEN, 0x00 }, }; static const unsigned int adau1373_out_tlv[] = { @@ -418,6 +554,7 @@ static int adau1373_pll_event(struct snd_soc_dapm_widget *w, struct snd_kcontrol *kcontrol, int event) { struct snd_soc_codec *codec = w->codec; + struct adau1373 *adau1373 = snd_soc_codec_get_drvdata(codec); unsigned int pll_id = w->name[3] - '1'; unsigned int val; @@ -426,7 +563,7 @@ static int adau1373_pll_event(struct snd_soc_dapm_widget *w, else val = 0; - snd_soc_update_bits(codec, ADAU1373_PLL_CTRL6(pll_id), + regmap_update_bits(adau1373->regmap, ADAU1373_PLL_CTRL6(pll_id), ADAU1373_PLL_CTRL6_PLL_EN, val); if (SND_SOC_DAPM_EVENT_ON(event)) @@ -938,7 +1075,7 @@ static int adau1373_hw_params(struct snd_pcm_substream *substream, adau1373_dai->enable_src = (div != 0); - snd_soc_update_bits(codec, ADAU1373_BCLKDIV(dai->id), + regmap_update_bits(adau1373->regmap, ADAU1373_BCLKDIV(dai->id), ADAU1373_BCLKDIV_SR_MASK | ADAU1373_BCLKDIV_BCLK_MASK, (div << 2) | ADAU1373_BCLKDIV_64); @@ -959,7 +1096,7 @@ static int adau1373_hw_params(struct snd_pcm_substream *substream, return -EINVAL; } - return snd_soc_update_bits(codec, ADAU1373_DAI(dai->id), + return regmap_update_bits(adau1373->regmap, ADAU1373_DAI(dai->id), ADAU1373_DAI_WLEN_MASK, ctrl); } @@ -1016,7 +1153,7 @@ static int adau1373_set_dai_fmt(struct snd_soc_dai *dai, unsigned int fmt) return -EINVAL; } - snd_soc_update_bits(codec, ADAU1373_DAI(dai->id), + regmap_update_bits(adau1373->regmap, ADAU1373_DAI(dai->id), ~ADAU1373_DAI_WLEN_MASK, ctrl); return 0; @@ -1039,7 +1176,7 @@ static int adau1373_set_dai_sysclk(struct snd_soc_dai *dai, adau1373_dai->sysclk = freq; adau1373_dai->clk_src = clk_id; - snd_soc_update_bits(dai->codec, ADAU1373_BCLKDIV(dai->id), + regmap_update_bits(adau1373->regmap, ADAU1373_BCLKDIV(dai->id), ADAU1373_BCLKDIV_SOURCE, clk_id << 5); return 0; @@ -1120,6 +1257,7 @@ static struct snd_soc_dai_driver adau1373_dai_driver[] = { static int adau1373_set_pll(struct snd_soc_codec *codec, int pll_id, int source, unsigned int freq_in, unsigned int freq_out) { + struct adau1373 *adau1373 = snd_soc_codec_get_drvdata(codec); unsigned int dpll_div = 0; unsigned int x, r, n, m, i, j, mode; @@ -1187,36 +1325,36 @@ static int adau1373_set_pll(struct snd_soc_codec *codec, int pll_id, if (dpll_div) { dpll_div = 11 - dpll_div; - snd_soc_update_bits(codec, ADAU1373_PLL_CTRL6(pll_id), + regmap_update_bits(adau1373->regmap, ADAU1373_PLL_CTRL6(pll_id), ADAU1373_PLL_CTRL6_DPLL_BYPASS, 0); } else { - snd_soc_update_bits(codec, ADAU1373_PLL_CTRL6(pll_id), + regmap_update_bits(adau1373->regmap, ADAU1373_PLL_CTRL6(pll_id), ADAU1373_PLL_CTRL6_DPLL_BYPASS, ADAU1373_PLL_CTRL6_DPLL_BYPASS); } - snd_soc_write(codec, ADAU1373_DPLL_CTRL(pll_id), + regmap_write(adau1373->regmap, ADAU1373_DPLL_CTRL(pll_id), (source << 4) | dpll_div); - snd_soc_write(codec, ADAU1373_PLL_CTRL1(pll_id), (m >> 8) & 0xff); - snd_soc_write(codec, ADAU1373_PLL_CTRL2(pll_id), m & 0xff); - snd_soc_write(codec, ADAU1373_PLL_CTRL3(pll_id), (n >> 8) & 0xff); - snd_soc_write(codec, ADAU1373_PLL_CTRL4(pll_id), n & 0xff); - snd_soc_write(codec, ADAU1373_PLL_CTRL5(pll_id), + regmap_write(adau1373->regmap, ADAU1373_PLL_CTRL1(pll_id), (m >> 8) & 0xff); + regmap_write(adau1373->regmap, ADAU1373_PLL_CTRL2(pll_id), m & 0xff); + regmap_write(adau1373->regmap, ADAU1373_PLL_CTRL3(pll_id), (n >> 8) & 0xff); + regmap_write(adau1373->regmap, ADAU1373_PLL_CTRL4(pll_id), n & 0xff); + regmap_write(adau1373->regmap, ADAU1373_PLL_CTRL5(pll_id), (r << 3) | (x << 1) | mode); /* Set sysclk to pll_rate / 4 */ - snd_soc_update_bits(codec, ADAU1373_CLK_SRC_DIV(pll_id), 0x3f, 0x09); + regmap_update_bits(adau1373->regmap, ADAU1373_CLK_SRC_DIV(pll_id), 0x3f, 0x09); return 0; } -static void adau1373_load_drc_settings(struct snd_soc_codec *codec, +static void adau1373_load_drc_settings(struct adau1373 *adau1373, unsigned int nr, uint8_t *drc) { unsigned int i; for (i = 0; i < ADAU1373_DRC_SIZE; ++i) - snd_soc_write(codec, ADAU1373_DRC(nr) + i, drc[i]); + regmap_write(adau1373->regmap, ADAU1373_DRC(nr) + i, drc[i]); } static bool adau1373_valid_micbias(enum adau1373_micbias_voltage micbias) @@ -1235,13 +1373,14 @@ static bool adau1373_valid_micbias(enum adau1373_micbias_voltage micbias) static int adau1373_probe(struct snd_soc_codec *codec) { + struct adau1373 *adau1373 = snd_soc_codec_get_drvdata(codec); struct adau1373_platform_data *pdata = codec->dev->platform_data; bool lineout_differential = false; unsigned int val; int ret; int i; - ret = snd_soc_codec_set_cache_io(codec, 8, 8, SND_SOC_I2C); + ret = snd_soc_codec_set_cache_io(codec, 0, 0, SND_SOC_REGMAP); if (ret) { dev_err(codec->dev, "failed to set cache I/O: %d\n", ret); return ret; @@ -1256,7 +1395,7 @@ static int adau1373_probe(struct snd_soc_codec *codec) return -EINVAL; for (i = 0; i < pdata->num_drc; ++i) { - adau1373_load_drc_settings(codec, i, + adau1373_load_drc_settings(adau1373, i, pdata->drc_setting[i]); } @@ -1268,18 +1407,18 @@ static int adau1373_probe(struct snd_soc_codec *codec) if (pdata->input_differential[i]) val |= BIT(i); } - snd_soc_write(codec, ADAU1373_INPUT_MODE, val); + regmap_write(adau1373->regmap, ADAU1373_INPUT_MODE, val); val = 0; if (pdata->lineout_differential) val |= ADAU1373_OUTPUT_CTRL_LDIFF; if (pdata->lineout_ground_sense) val |= ADAU1373_OUTPUT_CTRL_LNFBEN; - snd_soc_write(codec, ADAU1373_OUTPUT_CTRL, val); + regmap_write(adau1373->regmap, ADAU1373_OUTPUT_CTRL, val); lineout_differential = pdata->lineout_differential; - snd_soc_write(codec, ADAU1373_EP_CTRL, + regmap_write(adau1373->regmap, ADAU1373_EP_CTRL, (pdata->micbias1 << ADAU1373_EP_CTRL_MICBIAS1_OFFSET) | (pdata->micbias2 << ADAU1373_EP_CTRL_MICBIAS2_OFFSET)); } @@ -1289,7 +1428,7 @@ static int adau1373_probe(struct snd_soc_codec *codec) ARRAY_SIZE(adau1373_lineout2_controls)); } - snd_soc_write(codec, ADAU1373_ADC_CTRL, + regmap_write(adau1373->regmap, ADAU1373_ADC_CTRL, ADAU1373_ADC_CTRL_RESET_FORCE | ADAU1373_ADC_CTRL_PEAK_DETECT); return 0; @@ -1298,17 +1437,19 @@ static int adau1373_probe(struct snd_soc_codec *codec) static int adau1373_set_bias_level(struct snd_soc_codec *codec, enum snd_soc_bias_level level) { + struct adau1373 *adau1373 = snd_soc_codec_get_drvdata(codec); + switch (level) { case SND_SOC_BIAS_ON: break; case SND_SOC_BIAS_PREPARE: break; case SND_SOC_BIAS_STANDBY: - snd_soc_update_bits(codec, ADAU1373_PWDN_CTRL3, + regmap_update_bits(adau1373->regmap, ADAU1373_PWDN_CTRL3, ADAU1373_PWDN_CTRL3_PWR_EN, ADAU1373_PWDN_CTRL3_PWR_EN); break; case SND_SOC_BIAS_OFF: - snd_soc_update_bits(codec, ADAU1373_PWDN_CTRL3, + regmap_update_bits(adau1373->regmap, ADAU1373_PWDN_CTRL3, ADAU1373_PWDN_CTRL3_PWR_EN, 0); break; } @@ -1324,17 +1465,49 @@ static int adau1373_remove(struct snd_soc_codec *codec) static int adau1373_suspend(struct snd_soc_codec *codec) { - return adau1373_set_bias_level(codec, SND_SOC_BIAS_OFF); + struct adau1373 *adau1373 = snd_soc_codec_get_drvdata(codec); + int ret; + + ret = adau1373_set_bias_level(codec, SND_SOC_BIAS_OFF); + regcache_cache_only(adau1373->regmap, true); + + return ret; } static int adau1373_resume(struct snd_soc_codec *codec) { + struct adau1373 *adau1373 = snd_soc_codec_get_drvdata(codec); + + regcache_cache_only(adau1373->regmap, false); adau1373_set_bias_level(codec, SND_SOC_BIAS_STANDBY); - snd_soc_cache_sync(codec); + regcache_sync(adau1373->regmap); return 0; } +static bool adau1373_register_volatile(struct device *dev, unsigned int reg) +{ + switch (reg) { + case ADAU1373_SOFT_RESET: + case ADAU1373_ADC_DAC_STATUS: + return true; + default: + return false; + } +} + +static const struct regmap_config adau1373_regmap_config = { + .val_bits = 8, + .reg_bits = 8, + + .volatile_reg = adau1373_register_volatile, + .max_register = ADAU1373_SOFT_RESET, + + .cache_type = REGCACHE_RBTREE, + .reg_defaults = adau1373_reg_defaults, + .num_reg_defaults = ARRAY_SIZE(adau1373_reg_defaults), +}; + static struct snd_soc_codec_driver adau1373_codec_driver = { .probe = adau1373_probe, .remove = adau1373_remove, @@ -1342,9 +1515,6 @@ static struct snd_soc_codec_driver adau1373_codec_driver = { .resume = adau1373_resume, .set_bias_level = adau1373_set_bias_level, .idle_bias_off = true, - .reg_cache_size = ARRAY_SIZE(adau1373_default_regs), - .reg_cache_default = adau1373_default_regs, - .reg_word_size = sizeof(uint8_t), .set_pll = adau1373_set_pll, @@ -1366,6 +1536,11 @@ static int adau1373_i2c_probe(struct i2c_client *client, if (!adau1373) return -ENOMEM; + adau1373->regmap = devm_regmap_init_i2c(client, + &adau1373_regmap_config); + if (IS_ERR(adau1373->regmap)) + return PTR_ERR(adau1373->regmap); + dev_set_drvdata(&client->dev, adau1373); ret = snd_soc_register_codec(&client->dev, &adau1373_codec_driver, -- 1.8.0

2 3

[alsa-devel] [RFC/RFT PATCH] ALSA: hda - hdmi: Add ATI/AMD multi-channel audio support
by Anssi Hannula 27 Sep '13

27 Sep '13

Request for comments and testing. AMD just released the documentation needed for multi-channel HDMI support, so here is the initial patch adding such support. This has been tested on codec 0x1002aa01, revision 0x100200 with the exception of custom channel maps, which I intend to test later. I'm especially interested in testers with: - Older codecs other than 0x1002aa01. My best guess is that the new code should work on them as well, but I'm not really 100% sure. - Codec 0x1002aa01 revisions 0x100300 (Rev ID 3) or later (see /proc/asound/cardX/codec#Y). These support individual channel remapping and that code is entirely untested (even for regular 5.1/7.1). I decided it was best to get this out there early even though major parts of it are still untested, for comments and for other people to test :) Note that this recent commit is probably required for multi-channel to work with the open-source radeon driver: ALSA: hda - hdmi: Fallback to ALSA allocation when selecting CA https://git.kernel.org/cgit/linux/kernel/git/torvalds/linux.git/commit/soun… Easiest way to test basic multichannel is by speaker-test, e.g. speaker-test -D hdmi:CARD=Generic,DEV=0 -c8 -r192000 -F S16_LE for 7.1 16-bit 192kHz Detailed discussion below: ATI/AMD codecs do not support all the standard HDA HDMI/DP functions. In particular, these are missing: - ELD support - standard channel mapping support - standard infoframe configuration support In place of them, custom HDA pin verbs are included, that allow: - getting speaker mask from EDID - getting SADs from EDID - getting connection type from EDID (HDMI, DisplayPort) - setting CA for infoframe - setting down-mix information for infoframe - channel pair remapping - individual channel remapping (revision ID 3+) The documentation for the verbs has now been released by AMD: http://www.x.org/docs/AMD/AMD_HDA_verbs.pdf Also, converted nodes do not have all their supported rates, formats and channel counts listed. Add support for the ATI/AMD specific verbs and use them instead of the generic methods on ATI/AMD codecs. This allows multi-channel audio to work. Is the way it is done OK, or should something more generic be put into place (custom callbacks and/or flags in struct hdmi_spec maybe)? On the older ATI/AMD codecs that do not support individual remapping, channel remapping is restricted to pair remapping. This does not affect the standard multi-channel configurations. Since there is no ELD support (unless out-of-band communication with the graphics driver is added so that they can provide it to us directly), ELD is currently emulated with only the basic information populated (speaker mask, SADs, connection type). There are some bits missing on the radeon driver side (the SADs etc are not populated) so currently the ELD emulation only works on fglrx. ELD emulation is not needed for multi-channel playback. Rafał, are you interested in implementing the missing bits in the radeon driver, as I see you have looked into this before? The ELD emulation was quite quickly hacked together and hasn't been cleaned up yet, since I'm not sure how we want to ultimately handle this. Is ELD emulation the way to go? If so, where does the code belong (the ATI specific code is in patch_hdmi.c, but the ELD parsing/defines etc are in hda_eld.c)? Codec 0x1002aa01 is switched back to patch_atihdmi to support the ATI/AMD specific features. HBR mode (needed for DTS-HD/TrueHD passthrough) is not working yet, and the released documentation does not mention it. Signed-off-by: Anssi Hannula <anssi.hannula(a)iki.fi> Tested-by: Peter Frühberger <fritsch(a)xbmc.org> Cc: Rafał Miłecki <zajec5(a)gmail.com> --- sound/pci/hda/patch_hdmi.c | 418 ++++++++++++++++++++++++++++++++++++++++----- 1 file changed, 374 insertions(+), 44 deletions(-) diff --git a/sound/pci/hda/patch_hdmi.c b/sound/pci/hda/patch_hdmi.c index 3d8cd044..741d75e 100644 --- a/sound/pci/hda/patch_hdmi.c +++ b/sound/pci/hda/patch_hdmi.c @@ -6,6 +6,7 @@ * Copyright (c) 2006 ATI Technologies Inc. * Copyright (c) 2008 NVIDIA Corp. All rights reserved. * Copyright (c) 2008 Wei Ni <wni(a)nvidia.com> + * Copyright (c) 2013 Anssi Hannula <anssi.hannula(a)iki.fi> * * Authors: * Wu Fengguang <wfg(a)linux.intel.com> @@ -45,6 +46,9 @@ module_param(static_hdmi_pcm, bool, 0644); MODULE_PARM_DESC(static_hdmi_pcm, "Don't restrict PCM parameters per ELD info"); #define is_haswell(codec) ((codec)->vendor_id == 0x80862807) +#define is_atihdmi(codec) (((codec)->vendor_id & 0xffff0000) == 0x10020000) +#define has_amd_full_remap_support(codec) \ + ((codec)->vendor_id == 0x1002791a && ((codec)->revision_id & 0xff00) >= 0x0300) struct hdmi_spec_per_cvt { hda_nid_t cvt_nid; @@ -89,7 +93,7 @@ struct hdmi_spec { struct hdmi_eld temp_eld; /* - * Non-generic ATI/NVIDIA specific + * Non-generic VIA/NVIDIA specific */ struct hda_multi_out multiout; struct hda_pcm_stream pcm_playback; @@ -573,6 +577,20 @@ static int hdmi_channel_allocation(struct hdmi_eld *eld, int channels) return ca; } +#ifdef CONFIG_SND_DEBUG_VERBOSE +static int atihdmi_get_chan_slot(struct hda_codec *codec, hda_nid_t pin_nid, int asp_slot); + +static int hdmi_get_chan_slot(struct hda_codec *codec, hda_nid_t pin_nid, int asp_slot) +{ + if (is_atihdmi(codec)) + return atihdmi_get_chan_slot(codec, pin_nid, asp_slot); + + return snd_hda_codec_read(codec, pin_nid, 0, + AC_VERB_GET_HDMI_CHAN_SLOT, + asp_slot); +} +#endif + static void hdmi_debug_channel_mapping(struct hda_codec *codec, hda_nid_t pin_nid) { @@ -581,14 +599,26 @@ static void hdmi_debug_channel_mapping(struct hda_codec *codec, int slot; for (i = 0; i < 8; i++) { - slot = snd_hda_codec_read(codec, pin_nid, 0, - AC_VERB_GET_HDMI_CHAN_SLOT, i); + slot = hdmi_get_chan_slot(codec, pin_nid, i); printk(KERN_DEBUG "HDMI: ASP channel %d => slot %d\n", slot >> 4, slot & 0xf); } #endif } +static int atihdmi_set_chan_slot(struct hda_codec *codec, hda_nid_t pin_nid, + int chanslot_setup); + +static int hdmi_set_chan_slot(struct hda_codec *codec, hda_nid_t pin_nid, + int chanslot_setup) +{ + if (is_atihdmi(codec)) + return atihdmi_set_chan_slot(codec, pin_nid, chanslot_setup); + + return snd_hda_codec_write(codec, pin_nid, 0, + AC_VERB_SET_HDMI_CHAN_SLOT, + chanslot_setup); +} static void hdmi_std_setup_channel_mapping(struct hda_codec *codec, hda_nid_t pin_nid, @@ -617,9 +647,8 @@ static void hdmi_std_setup_channel_mapping(struct hda_codec *codec, } for (i = 0; i < 8; i++) { - err = snd_hda_codec_write(codec, pin_nid, 0, - AC_VERB_SET_HDMI_CHAN_SLOT, - non_pcm ? non_pcm_mapping[i] : hdmi_channel_mapping[ca][i]); + err = hdmi_set_chan_slot(codec, pin_nid, + non_pcm ? non_pcm_mapping[i] : hdmi_channel_mapping[ca][i]); if (err) { snd_printdd(KERN_NOTICE "HDMI: channel mapping failed\n"); @@ -728,8 +757,7 @@ static int hdmi_manual_setup_channel_mapping(struct hda_codec *codec, else val = 0xf; val |= (i << 4); - err = snd_hda_codec_write(codec, pin_nid, 0, - AC_VERB_SET_HDMI_CHAN_SLOT, val); + err = hdmi_set_chan_slot(codec, pin_nid, val); if (err) return -EINVAL; } @@ -883,6 +911,8 @@ static bool hdmi_infoframe_uptodate(struct hda_codec *codec, hda_nid_t pin_nid, return true; } +static void atihdmi_set_ca(struct hda_codec *codec, hda_nid_t pin_nid, int ca); + static void hdmi_setup_audio_infoframe(struct hda_codec *codec, struct hdmi_spec_per_pin *per_pin, bool non_pcm) @@ -912,6 +942,16 @@ static void hdmi_setup_audio_infoframe(struct hda_codec *codec, if (ca < 0) ca = 0; + if (is_atihdmi(codec)) { + /* for ATI/AMD we just want to map channels and set ca */ + hdmi_setup_channel_mapping(codec, pin_nid, non_pcm, ca, + channels, per_pin->chmap, + per_pin->chmap_set); + atihdmi_set_ca(codec, pin_nid, ca); + per_pin->non_pcm = non_pcm; + return; + } + memset(&ai, 0, sizeof(ai)); if (eld->info.conn_type == 0) { /* HDMI */ struct hdmi_audio_infoframe *hdmi_ai = &ai.hdmi; @@ -1274,6 +1314,9 @@ static int hdmi_read_pin_conn(struct hda_codec *codec, int pin_idx) return 0; } +static int atihdmi_get_eld(struct hda_codec *codec, hda_nid_t nid, + unsigned char *buf, int *eld_size); + static void hdmi_present_sense(struct hdmi_spec_per_pin *per_pin, int repoll) { struct hda_codec *codec = per_pin->codec; @@ -1303,8 +1346,12 @@ static void hdmi_present_sense(struct hdmi_spec_per_pin *per_pin, int repoll) "HDMI status: Codec=%d Pin=%d Presence_Detect=%d ELD_Valid=%d\n", codec->addr, pin_nid, pin_eld->monitor_present, eld->eld_valid); + if (pin_eld->monitor_present && is_atihdmi(codec)) + eld->eld_valid = (atihdmi_get_eld(codec, pin_nid, eld->eld_buffer, &eld->eld_size) == 0); + if (eld->eld_valid) { - if (snd_hdmi_get_eld(codec, pin_nid, eld->eld_buffer, + if (!is_atihdmi(codec) && + snd_hdmi_get_eld(codec, pin_nid, eld->eld_buffer, &eld->eld_size) < 0) eld->eld_valid = false; else { @@ -1603,6 +1650,8 @@ static int hdmi_chmap_ctl_info(struct snd_kcontrol *kcontrol, return 0; } +static int atihdmi_swap_fc_lfe(int pos); + static int hdmi_chmap_ctl_tlv(struct snd_kcontrol *kcontrol, int op_flag, unsigned int size, unsigned int __user *tlv) { @@ -1613,6 +1662,10 @@ static int hdmi_chmap_ctl_tlv(struct snd_kcontrol *kcontrol, int op_flag, FL | FR | RL | RR | LFE | FC | RLC | RRC; unsigned int __user *dst; int chs, count = 0; + int tlv_type = SNDRV_CTL_TLVT_CHMAP_VAR; + + if (is_atihdmi(codec) && !has_amd_full_remap_support(codec)) + tlv_type = SNDRV_CTL_TLVT_CHMAP_PAIRED; if (size < 8) return -ENOMEM; @@ -1620,19 +1673,33 @@ static int hdmi_chmap_ctl_tlv(struct snd_kcontrol *kcontrol, int op_flag, return -EFAULT; size -= 8; dst = tlv + 2; - for (chs = 2; chs <= spec->channels_max; chs++) { + for (chs = 2; chs <= spec->channels_max; + chs += (tlv_type == SNDRV_CTL_TLVT_CHMAP_PAIRED) ? 2 : 1) { int i, c; struct cea_channel_speaker_allocation *cap; cap = channel_allocations; for (i = 0; i < ARRAY_SIZE(channel_allocations); i++, cap++) { int chs_bytes = chs * 4; - if (cap->channels != chs) + if (tlv_type == SNDRV_CTL_TLVT_CHMAP_PAIRED) { + int chanpairs = 0; + /* in paired mode we need to take into account + * the occupied channel pairs instead of just the + * channel count */ + for (c = 0; c < 8; c++) { + if (cap->speakers[c] || cap->speakers[c+1]) + chanpairs++; + } + if (chanpairs * 2 != chs) + continue; + + } else if (cap->channels != chs) continue; + if (cap->spk_mask & ~valid_mask) continue; if (size < 8) return -ENOMEM; - if (put_user(SNDRV_CTL_TLVT_CHMAP_VAR, dst) || + if (put_user(tlv_type, dst) || put_user(chs_bytes, dst + 1)) return -EFAULT; dst += 2; @@ -1643,10 +1710,25 @@ static int hdmi_chmap_ctl_tlv(struct snd_kcontrol *kcontrol, int op_flag, size -= chs_bytes; count += chs_bytes; for (c = 7; c >= 0; c--) { - int spk = cap->speakers[c]; - if (!spk) - continue; - if (put_user(spk_to_chmap(spk), dst)) + int spk; + int chan = c; + int chpos; + + if (tlv_type == SNDRV_CTL_TLVT_CHMAP_PAIRED) + chan = 7 - atihdmi_swap_fc_lfe(7 - chan); + + spk = cap->speakers[chan]; + if (spk) + chpos = spk_to_chmap(spk); + else { + /* We need to reserve an N/A channel in paired mode */ + if (tlv_type == SNDRV_CTL_TLVT_CHMAP_PAIRED) + chpos = SNDRV_CHMAP_NA; + else + continue; + } + + if (put_user(chpos, dst)) return -EFAULT; dst++; } @@ -1672,6 +1754,18 @@ static int hdmi_chmap_ctl_get(struct snd_kcontrol *kcontrol, return 0; } +static int atihdmi_pairwise_chmap_check_order(struct hda_codec *codec, int ca, + int chs, unsigned char *map); + +static int hdmi_chmap_check_order(struct hda_codec *codec, int ca, + int chs, unsigned char *map) +{ + if (is_atihdmi(codec) && !has_amd_full_remap_support(codec)) + return atihdmi_pairwise_chmap_check_order(codec, ca, chs, map); + + return 0; /* anything can be remapped as needed */ +} + static int hdmi_chmap_ctl_put(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) { @@ -1683,7 +1777,7 @@ static int hdmi_chmap_ctl_put(struct snd_kcontrol *kcontrol, unsigned int ctl_idx; struct snd_pcm_substream *substream; unsigned char chmap[8]; - int i, ca, prepared = 0; + int i, err, ca, prepared = 0; ctl_idx = snd_ctl_get_ioffidx(kcontrol, &ucontrol->id); substream = snd_pcm_chmap_substream(info, ctl_idx); @@ -1707,6 +1801,9 @@ static int hdmi_chmap_ctl_put(struct snd_kcontrol *kcontrol, ca = hdmi_manual_channel_allocation(ARRAY_SIZE(chmap), chmap); if (ca < 0) return -EINVAL; + err = hdmi_chmap_check_order(codec, ca, ARRAY_SIZE(chmap), chmap); + if (err < 0) + return -EINVAL; per_pin->chmap_set = true; memcpy(per_pin->chmap, chmap, sizeof(chmap)); if (prepared) @@ -2551,48 +2648,281 @@ static int patch_nvhdmi_8ch_7x(struct hda_codec *codec) /* * ATI-specific implementations - * - * FIXME: we may omit the whole this and use the generic code once after - * it's confirmed to work. */ -#define ATIHDMI_CVT_NID 0x02 /* audio converter */ -#define ATIHDMI_PIN_NID 0x03 /* HDMI output pin */ +/* ATI/AMD specific HDA pin verbs, see the AMD HDA Verbs specification */ +#define ATI_VERB_SET_CHANNEL_ALLOCATION 0x771 +#define ATI_VERB_SET_DOWNMIX_INFO 0x772 +#define ATI_VERB_SET_AUDIO_DESCRIPTOR 0x776 +#define ATI_VERB_SET_MULTICHANNEL_01 0x777 +#define ATI_VERB_SET_MULTICHANNEL_23 0x778 +#define ATI_VERB_SET_MULTICHANNEL_45 0x779 +#define ATI_VERB_SET_MULTICHANNEL_67 0x77a +#define ATI_VERB_SET_MULTICHANNEL_1 0x785 +#define ATI_VERB_SET_MULTICHANNEL_3 0x786 +#define ATI_VERB_SET_MULTICHANNEL_5 0x787 +#define ATI_VERB_SET_MULTICHANNEL_7 0x788 +#define ATI_VERB_SET_MULTICHANNEL_MODE 0x789 +#define ATI_VERB_GET_SPEAKER_ALLOCATION 0xf70 +#define ATI_VERB_GET_CHANNEL_ALLOCATION 0xf71 +#define ATI_VERB_GET_DOWNMIX_INFO 0xf72 +#define ATI_VERB_GET_AUDIO_DESCRIPTOR 0xf76 +#define ATI_VERB_GET_MULTICHANNEL_01 0xf77 +#define ATI_VERB_GET_MULTICHANNEL_23 0xf78 +#define ATI_VERB_GET_MULTICHANNEL_45 0xf79 +#define ATI_VERB_GET_MULTICHANNEL_67 0xf7a +#define ATI_VERB_GET_MULTICHANNEL_1 0xf85 +#define ATI_VERB_GET_MULTICHANNEL_3 0xf86 +#define ATI_VERB_GET_MULTICHANNEL_5 0xf87 +#define ATI_VERB_GET_MULTICHANNEL_7 0xf88 +#define ATI_VERB_GET_MULTICHANNEL_MODE 0xf89 + +#define ATI_OUT_ENABLE 0x1 + +/* TODO: cleanup this function */ +static int atihdmi_get_eld(struct hda_codec *codec, hda_nid_t nid, + unsigned char *buf, int *eld_size) +{ + int spkalloc, ati_sad; + int pos, i; + + /* ATI/AMD does not have ELD, emulate it */ + + spkalloc = snd_hda_codec_read(codec, nid, 0, ATI_VERB_GET_SPEAKER_ALLOCATION, 0); + + if (!spkalloc) + return -EINVAL; + + memset(buf, 0, 84); + + /* speaker allocation from EDID */ + buf[7] = spkalloc & 0x00007f; + + /* is DisplayPort? */ + if (spkalloc & 0x000200) + buf[5] |= 0x04; + + pos = 20; + for (i = 1; i <= 14; i++) { + if (i == 13) /* not handled by ATI/AMD */ + continue; + + snd_hda_codec_write(codec, nid, 0, ATI_VERB_SET_AUDIO_DESCRIPTOR, i << 3); + ati_sad = snd_hda_codec_read(codec, nid, 0, ATI_VERB_GET_AUDIO_DESCRIPTOR, 0); + + if (ati_sad & 0xff00) { + memcpy(buf + pos, &ati_sad, 3); + pos += 3; + } + + if (i == 1 && ati_sad & 0xff000000 && (ati_sad & 0xff00) != (ati_sad & 0xff000000) >> 16) { + /* for PCM there is a separate stereo rate mask */ + memcpy(buf + pos, &ati_sad, 3); + /* max channels to 2 */ + buf[pos+0] = (buf[pos+0] & ~0x7) | 0x1; + /* rates from extra byte */ + buf[pos+1] = (ati_sad & 0xff000000) >> 24; + pos += 3; + } + } + + if (pos == 20) + return -EINVAL; + + /* version */ + buf[0] = 0x10; + + /* Baseline length */ + buf[2] = pos - 4; + + /* SAD count */ + buf[5] |= ((pos - 20) / 3) << 4; + + *eld_size = pos; + + return 0; +} + +static void atihdmi_set_ca(struct hda_codec *codec, hda_nid_t pin_nid, int ca) +{ + printk("ATI: setting ca %d\n", ca); + snd_hda_codec_write(codec, pin_nid, 0, ATI_VERB_SET_CHANNEL_ALLOCATION, ca); +} + +static int atihdmi_swap_fc_lfe(int pos) +{ + /* + * Older ATI/AMD without channel-wise mapping + * have automatic FC/LFE swap built-in. + */ + + switch (pos) { + /* see channel_allocations[].speakers[] */ + case 2: return 3; + case 3: return 2; + default: break; + } + + return pos; +} + +static int atihdmi_pairwise_chmap_check_order(struct hda_codec *codec, int ca, + int chs, unsigned char *map) +{ + struct cea_channel_speaker_allocation *cap; + int i, j; + + /* check that only channel pairs need to be remapped on old ATI/AMD */ + + cap = &channel_allocations[get_channel_allocation_order(ca)]; + for (i = 0; i < chs; ++i) { + int mask = to_spk_mask(map[i]); + bool ok = false; + bool companion_ok = false; + + if (!mask) + continue; + + for (j = 0 + i % 2; j < 8; j += 2) { + int chan_idx = 7 - atihdmi_swap_fc_lfe(j); + if (cap->speakers[chan_idx] == mask) { + /* channel is in a supported position */ + ok = true; + + if (i % 2 == 0 && i + 1 < chs) { + /* even channel, check the odd companion */ + int comp_chan_idx = 7 - atihdmi_swap_fc_lfe(j + 1); + int comp_mask_req = to_spk_mask(map[i+1]); + int comp_mask_act = cap->speakers[comp_chan_idx]; + + if (comp_mask_req == comp_mask_act) + companion_ok = true; + else + return -EINVAL; + } + break; + } + } + + if (!ok) + return -EINVAL; + + if (companion_ok) + i++; /* companion channel already checked */ + } + + return 0; +} + +static int atihdmi_set_chan_slot(struct hda_codec *codec, hda_nid_t pin_nid, + int chanslot_setup) +{ + int hdmi_slot = chanslot_setup & 0xf; + int stream_channel = chanslot_setup >> 4; + int verb; + int ati_channel_setup = 0; + + if (!has_amd_full_remap_support(codec)) { + hdmi_slot = atihdmi_swap_fc_lfe(hdmi_slot); + + /* In case this is an odd slot but without stream channel, do not + * disable the slot since the corresponding even slot could have a + * channel. In case neither have a channel, the slot pair will be + * disabled when this function is called for the even slot. */ + if (hdmi_slot % 2 != 0 && stream_channel == 0xf) + return 0; + + hdmi_slot -= hdmi_slot % 2; + + if (stream_channel != 0xf) + stream_channel -= stream_channel % 2; + } + + verb = ATI_VERB_SET_MULTICHANNEL_01 + hdmi_slot/2 + (hdmi_slot % 2) * 0x00e; + + /* ati_channel_setup format: [7..4] = stream_channel_id, [1] = mute, [0] = enable */ + + if (stream_channel != 0xf) + ati_channel_setup = (stream_channel << 4) | ATI_OUT_ENABLE; + + return snd_hda_codec_write(codec, pin_nid, 0, verb, ati_channel_setup); +} + +#ifdef CONFIG_SND_DEBUG_VERBOSE +static int atihdmi_get_chan_slot(struct hda_codec *codec, hda_nid_t pin_nid, int asp_slot) +{ + bool was_odd = false; + int ati_asp_slot = asp_slot; + int verb; + int ati_channel_setup; + + /* emulate AC_VERB_GET_HDMI_CHAN_SLOT */ + + if (!has_amd_full_remap_support(codec)) { + ati_asp_slot = atihdmi_swap_fc_lfe(asp_slot); + if (ati_asp_slot % 2 != 0) { + ati_asp_slot -= 1; + was_odd = true; + } + } -static int atihdmi_playback_pcm_prepare(struct hda_pcm_stream *hinfo, - struct hda_codec *codec, - unsigned int stream_tag, - unsigned int format, - struct snd_pcm_substream *substream) + verb = ATI_VERB_GET_MULTICHANNEL_01 + ati_asp_slot/2 + (ati_asp_slot % 2) * 0x00e; + + ati_channel_setup = snd_hda_codec_read(codec, pin_nid, 0, verb, 0); + + if (!(ati_channel_setup & ATI_OUT_ENABLE)) + return (0xf << 4) | asp_slot; + + return ((ati_channel_setup & 0xf0) + ((!!was_odd) << 4)) | asp_slot; +} +#endif + +static int atihdmi_init(struct hda_codec *codec) { struct hdmi_spec *spec = codec->spec; - struct hdmi_spec_per_cvt *per_cvt = get_cvt(spec, 0); - int chans = substream->runtime->channels; - int i, err; + int pin_idx, err; - err = simple_playback_pcm_prepare(hinfo, codec, stream_tag, format, - substream); - if (err < 0) + err = generic_hdmi_init(codec); + + if (err) return err; - snd_hda_codec_write(codec, per_cvt->cvt_nid, 0, - AC_VERB_SET_CVT_CHAN_COUNT, chans - 1); - /* FIXME: XXX */ - for (i = 0; i < chans; i++) { - snd_hda_codec_write(codec, per_cvt->cvt_nid, 0, - AC_VERB_SET_HDMI_CHAN_SLOT, - (i << 4) | i); + + for (pin_idx = 0; pin_idx < spec->num_pins; pin_idx++) { + struct hdmi_spec_per_pin *per_pin = get_pin(spec, pin_idx); + + snd_hda_codec_write(codec, per_pin->pin_nid, 0, ATI_VERB_SET_DOWNMIX_INFO, 0); + + /* enable channel-wise remap mode if supported */ + if (has_amd_full_remap_support(codec)) + snd_hda_codec_write(codec, per_pin->pin_nid, 0, ATI_VERB_SET_MULTICHANNEL_MODE, 1); } + return 0; } static int patch_atihdmi(struct hda_codec *codec) { struct hdmi_spec *spec; - int err = patch_simple_hdmi(codec, ATIHDMI_CVT_NID, ATIHDMI_PIN_NID); - if (err < 0) + struct hdmi_spec_per_cvt *per_cvt; + int err, cvt_idx; + + err = patch_generic_hdmi(codec); + + if (err) return err; + + codec->patch_ops.init = atihdmi_init; + + /* ATI/AMD converters do not advertise all of their capabilities */ spec = codec->spec; - spec->pcm_playback.ops.prepare = atihdmi_playback_pcm_prepare; + for (cvt_idx = 0; cvt_idx < spec->num_cvts; cvt_idx++) { + per_cvt = get_cvt(spec, cvt_idx); + per_cvt->channels_max = max(per_cvt->channels_max, 8u); + per_cvt->rates |= SUPPORTED_RATES; + per_cvt->formats |= SUPPORTED_FORMATS; + per_cvt->maxbps = max(per_cvt->maxbps, 24u); + } + return 0; } @@ -2612,7 +2942,7 @@ static const struct hda_codec_preset snd_hda_preset_hdmi[] = { { .id = 0x1002793c, .name = "RS600 HDMI", .patch = patch_atihdmi }, { .id = 0x10027919, .name = "RS600 HDMI", .patch = patch_atihdmi }, { .id = 0x1002791a, .name = "RS690/780 HDMI", .patch = patch_atihdmi }, -{ .id = 0x1002aa01, .name = "R6xx HDMI", .patch = patch_generic_hdmi }, +{ .id = 0x1002aa01, .name = "R6xx HDMI", .patch = patch_atihdmi }, { .id = 0x10951390, .name = "SiI1390 HDMI", .patch = patch_generic_hdmi }, { .id = 0x10951392, .name = "SiI1392 HDMI", .patch = patch_generic_hdmi }, { .id = 0x17e80047, .name = "Chrontel HDMI", .patch = patch_generic_hdmi }, -- 1.8.1.5

5 10

[alsa-devel] pcm_rate API question
by Rob Sykes 27 Sep '13

27 Sep '13

Hello alsa devs, There's an ongoing attempt to see if libsoxr-lsr can be used as a plug-in replacement for libsamplerate in pulseaudio and in alsa (libsoxr-lsr is designed with this usage in mind). With pulseaudio, it seems to 'just work', but with alsa, it looks like there is a buffering problem. Progress reports for this are here: http://blog.ivitera.com/pavel/linux-audio/pulseaudio-with-ld_preloading-lib… http://blog.ivitera.com/pavel/linux-audio/simple-use-of-libsoxr-lsr-in-alsa… AFAICT, the problem with alsa relates to the use of this function in pcm_rate.h: typedef struct snd_pcm_rate_ops { ... void (*convert_s16)(void *obj, int16_t *dst, unsigned int dst_frames, const int16_t *src, unsigned int src_frames); Is there documentation available regarding the use of this function? For example, is the plugin obliged always to consume src_frames and generate dst_frames samples? If so, how is the plugin expected to comply (say, if it's not been given enough input to generate that amount of output)? TIA, Rob

2 1

[alsa-devel] [PATCH] ASoC: wm8993: drop regulator_bulk_free of devm_ allocated data
by Wei Yongjun 27 Sep '13

27 Sep '13

From: Wei Yongjun <yongjun_wei(a)trendmicro.com.cn> It's not necessary to free regulator consumers allocated with devm_regulator_bulk_get. Signed-off-by: Wei Yongjun <yongjun_wei(a)trendmicro.com.cn> --- sound/soc/codecs/wm8993.c | 1 - 1 file changed, 1 deletion(-) diff --git a/sound/soc/codecs/wm8993.c b/sound/soc/codecs/wm8993.c index 433d59a..2ee23a3 100644 --- a/sound/soc/codecs/wm8993.c +++ b/sound/soc/codecs/wm8993.c @@ -1562,7 +1562,6 @@ static int wm8993_remove(struct snd_soc_codec *codec) struct wm8993_priv *wm8993 = snd_soc_codec_get_drvdata(codec); wm8993_set_bias_level(codec, SND_SOC_BIAS_OFF); - regulator_bulk_free(ARRAY_SIZE(wm8993->supplies), wm8993->supplies); return 0; }

2 1

[alsa-devel] [PATCH] ASoC: wm8993: drop regulator_bulk_free of devm_ allocated data
by Wei Yongjun 27 Sep '13

27 Sep '13

From: Wei Yongjun <yongjun_wei(a)trendmicro.com.cn> It's not necessary to free regulator consumers allocated with devm_regulator_bulk_get. Signed-off-by: Wei Yongjun <yongjun_wei(a)trendmicro.com.cn> --- sound/soc/codecs/wm8993.c | 1 - 1 file changed, 1 deletion(-) diff --git a/sound/soc/codecs/wm8993.c b/sound/soc/codecs/wm8993.c index 433d59a..2ee23a3 100644 --- a/sound/soc/codecs/wm8993.c +++ b/sound/soc/codecs/wm8993.c @@ -1562,7 +1562,6 @@ static int wm8993_remove(struct snd_soc_codec *codec) struct wm8993_priv *wm8993 = snd_soc_codec_get_drvdata(codec); wm8993_set_bias_level(codec, SND_SOC_BIAS_OFF); - regulator_bulk_free(ARRAY_SIZE(wm8993->supplies), wm8993->supplies); return 0; }

1 0

[alsa-devel] [PATCH] ASoC: rcar: remove unnecessary mach/clock.h
by Kuninori Morimoto 27 Sep '13

27 Sep '13

${LINUX}/sound/soc/sh driver can be compiled from SuperH and ARM. but, ${LINUX}/sound/soc/sh/rcar driver included SH-ARM specific header. This patch removes it Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> --- sound/soc/sh/rcar/adg.c | 1 - 1 file changed, 1 deletion(-) diff --git a/sound/soc/sh/rcar/adg.c b/sound/soc/sh/rcar/adg.c index ad9ceb9..9430097 100644 --- a/sound/soc/sh/rcar/adg.c +++ b/sound/soc/sh/rcar/adg.c @@ -8,7 +8,6 @@ * for more details. */ #include <linux/sh_clk.h> -#include <mach/clock.h> #include "rsnd.h" #define CLKA 0 -- 1.7.9.5

3 2

[alsa-devel] [PATCH] ASoC: fsl/imx-sgtl5000: Fix uninitialized pointer use in error path
by Philipp Zabel 26 Sep '13

26 Sep '13

This patch avoids to dereference the uninitialized data pointer if the error path is entered before devm_kzalloc is called (or if the allocation fails). It fixes the following warning: sound/soc/fsl/imx-sgtl5000.c: In function 'imx_sgtl5000_probe': sound/soc/fsl/imx-sgtl5000.c:175:18: warning: 'data' may be used uninitialized in this function [-Wmaybe-uninitialized] Signed-off-by: Philipp Zabel <p.zabel(a)pengutronix.de> --- sound/soc/fsl/imx-sgtl5000.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/soc/fsl/imx-sgtl5000.c b/sound/soc/fsl/imx-sgtl5000.c index c49e4cb..6f4bdc8 100644 --- a/sound/soc/fsl/imx-sgtl5000.c +++ b/sound/soc/fsl/imx-sgtl5000.c @@ -62,7 +62,7 @@ static int imx_sgtl5000_probe(struct platform_device *pdev) struct device_node *ssi_np, *codec_np; struct platform_device *ssi_pdev; struct i2c_client *codec_dev; - struct imx_sgtl5000_data *data; + struct imx_sgtl5000_data *data = NULL; int int_port, ext_port; int ret; @@ -172,7 +172,7 @@ static int imx_sgtl5000_probe(struct platform_device *pdev) return 0; fail: - if (!IS_ERR(data->codec_clk)) + if (data && !IS_ERR(data->codec_clk)) clk_put(data->codec_clk); if (ssi_np) of_node_put(ssi_np); -- 1.8.4.rc3

2 1

Re: [alsa-devel] [PATCH 39/51] DMA-API: others: use dma_set_coherent_mask()
by Archit Taneja 26 Sep '13

26 Sep '13

Hi, On Friday 20 September 2013 04:41 AM, Russell King wrote: > The correct way for a driver to specify the coherent DMA mask is > not to directly access the field in the struct device, but to use > dma_set_coherent_mask(). Only arch and bus code should access this > member directly. > > Convert all direct write accesses to using the correct API. > > Signed-off-by: Russell King <rmk+kernel(a)arm.linux.org.uk> > --- > drivers/ata/pata_ixp4xx_cf.c | 5 ++++- > drivers/gpu/drm/exynos/exynos_drm_drv.c | 6 +++++- > drivers/gpu/drm/omapdrm/omap_dmm_tiler.c | 5 +++-- > 3 files changed, 12 insertions(+), 4 deletions(-) > > diff --git a/drivers/ata/pata_ixp4xx_cf.c b/drivers/ata/pata_ixp4xx_cf.c > index 1ec53f8..ddf470c 100644 > --- a/drivers/ata/pata_ixp4xx_cf.c > +++ b/drivers/ata/pata_ixp4xx_cf.c > @@ -144,6 +144,7 @@ static int ixp4xx_pata_probe(struct platform_device *pdev) > struct ata_host *host; > struct ata_port *ap; > struct ixp4xx_pata_data *data = dev_get_platdata(&pdev->dev); > + int ret; > > cs0 = platform_get_resource(pdev, IORESOURCE_MEM, 0); > cs1 = platform_get_resource(pdev, IORESOURCE_MEM, 1); > @@ -157,7 +158,9 @@ static int ixp4xx_pata_probe(struct platform_device *pdev) > return -ENOMEM; > > /* acquire resources and fill host */ > - pdev->dev.coherent_dma_mask = DMA_BIT_MASK(32); > + ret = dma_set_coherent_mask(&pdev->dev, DMA_BIT_MASK(32)); > + if (ret) > + return ret; > > data->cs0 = devm_ioremap(&pdev->dev, cs0->start, 0x1000); > data->cs1 = devm_ioremap(&pdev->dev, cs1->start, 0x1000); > diff --git a/drivers/gpu/drm/exynos/exynos_drm_drv.c b/drivers/gpu/drm/exynos/exynos_drm_drv.c > index bb82ef7..81192d0 100644 > --- a/drivers/gpu/drm/exynos/exynos_drm_drv.c > +++ b/drivers/gpu/drm/exynos/exynos_drm_drv.c > @@ -286,7 +286,11 @@ static struct drm_driver exynos_drm_driver = { > > static int exynos_drm_platform_probe(struct platform_device *pdev) > { > - pdev->dev.coherent_dma_mask = DMA_BIT_MASK(32); > + int ret; > + > + ret = dma_set_coherent_mask(&pdev->dev, DMA_BIT_MASK(32)); > + if (ret) > + return ret; > > return drm_platform_init(&exynos_drm_driver, pdev); > } > diff --git a/drivers/gpu/drm/omapdrm/omap_dmm_tiler.c b/drivers/gpu/drm/omapdrm/omap_dmm_tiler.c > index acf6678..701c4c1 100644 > --- a/drivers/gpu/drm/omapdrm/omap_dmm_tiler.c > +++ b/drivers/gpu/drm/omapdrm/omap_dmm_tiler.c > @@ -664,8 +664,9 @@ static int omap_dmm_probe(struct platform_device *dev) > } > > /* set dma mask for device */ > - /* NOTE: this is a workaround for the hwmod not initializing properly */ > - dev->dev.coherent_dma_mask = DMA_BIT_MASK(32); > + ret = dma_set_coherent_mask(&dev->dev, DMA_BIT_MASK(32)); > + if (ret) > + goto fail; Tested with omapdrm on omap4 panda es board. Thanks, Archit

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Re: [alsa-devel] [PATCH v2] sound/soc/fsl/fsl_ssi.c: let check zero instead of check NO_IRQ
by Mark Brown 26 Sep '13

26 Sep '13

On Mon, Sep 23, 2013 at 11:36:21AM +0800, Chen Gang wrote: > NO_IRQ may be defined as '(unsigned int) -1' in some architectures > (arm, sh ...), and either may not be defined in some architectures > which can enable SND_SOC_FSL_SSI (e.g. allmodconfig for arc). Applied, thanks.

2 1

[alsa-devel] [PATCH] HDA-Intel: present up to 8 HDMI/DP outputs via "hdmi" device
by Anssi Hannula 26 Sep '13

26 Sep '13

Some new AMD cards have HDA codecs presenting 6 connected HDMI/DP pin nodes (plus 1 unconnected pin node) according to the ALSA card database. Example: http://www.alsa-project.org/db/?f=de3ced7af41de0ed54d218650e5e2f16c511787b Bump the maximum number of presented HDMI outputs per card via the "hdmi" PCM from 4 to 8 (so that the last possible device is DEV=7). Note that HDMI PCM devices DEV=4..7 use shared PCM device numbers, so HDA cards that have over 4 audio PCM devices or multiple S/PDIF or modem devices will have their remaining PCM devices misrepresented as HDMI devices. Signed-off-by: Anssi Hannula <anssi.hannula(a)iki.fi> --- Another example: http://www.alsa-project.org/db/?f=cd12a9d38f51c47eeedd380e4c5713f0d9a6c9de I do not have confirmed that all of these pins are actually usable (and therefore this patch is untested on affected hardware), but I guess that is probable. Maybe the large amount of them has something to do with DisplayPort supporting multiple displays via a single connector, but that is just a wild guess on my part... But do we want some sort of a solution for the possible conflicts between non-HDMI devices in the 10+ physical PCM device numbering range? src/conf/cards/HDA-Intel.conf | 92 +++++++++++++++++++++++++++++++++++++++++++ 1 file changed, 92 insertions(+) diff --git a/src/conf/cards/HDA-Intel.conf b/src/conf/cards/HDA-Intel.conf index 3957c12..7976b6c 100644 --- a/src/conf/cards/HDA-Intel.conf +++ b/src/conf/cards/HDA-Intel.conf @@ -305,6 +305,98 @@ HDA-Intel.pcm.hdmi.3 { } } +HDA-Intel.pcm.hdmi.4 { + @args [ CARD AES0 AES1 AES2 AES3 ] + @args.CARD { type string } + @args.AES0 { type integer } + @args.AES1 { type integer } + @args.AES2 { type integer } + @args.AES3 { type integer } + @func refer + name { + @func concat + strings [ + "cards.HDA-Intel.pcm.hdmi.common:" + "CARD=" $CARD "," + "DEVICE=10," + "CTLINDEX=4," + "AES0=" $AES0 "," + "AES1=" $AES1 "," + "AES2=" $AES2 "," + "AES3=" $AES3 + ] + } +} + +HDA-Intel.pcm.hdmi.5 { + @args [ CARD AES0 AES1 AES2 AES3 ] + @args.CARD { type string } + @args.AES0 { type integer } + @args.AES1 { type integer } + @args.AES2 { type integer } + @args.AES3 { type integer } + @func refer + name { + @func concat + strings [ + "cards.HDA-Intel.pcm.hdmi.common:" + "CARD=" $CARD "," + "DEVICE=11," + "CTLINDEX=5," + "AES0=" $AES0 "," + "AES1=" $AES1 "," + "AES2=" $AES2 "," + "AES3=" $AES3 + ] + } +} + +HDA-Intel.pcm.hdmi.6 { + @args [ CARD AES0 AES1 AES2 AES3 ] + @args.CARD { type string } + @args.AES0 { type integer } + @args.AES1 { type integer } + @args.AES2 { type integer } + @args.AES3 { type integer } + @func refer + name { + @func concat + strings [ + "cards.HDA-Intel.pcm.hdmi.common:" + "CARD=" $CARD "," + "DEVICE=12," + "CTLINDEX=6," + "AES0=" $AES0 "," + "AES1=" $AES1 "," + "AES2=" $AES2 "," + "AES3=" $AES3 + ] + } +} + +HDA-Intel.pcm.hdmi.7 { + @args [ CARD AES0 AES1 AES2 AES3 ] + @args.CARD { type string } + @args.AES0 { type integer } + @args.AES1 { type integer } + @args.AES2 { type integer } + @args.AES3 { type integer } + @func refer + name { + @func concat + strings [ + "cards.HDA-Intel.pcm.hdmi.common:" + "CARD=" $CARD "," + "DEVICE=13," + "CTLINDEX=7," + "AES0=" $AES0 "," + "AES1=" $AES1 "," + "AES2=" $AES2 "," + "AES3=" $AES3 + ] + } +} + <confdir:pcm/modem.conf> HDA-Intel.pcm.modem.0 { -- 1.8.1.5

2 2

[alsa-devel] [PATCH 1/1] ALSA: firewire-lib: use inlune function to calculate frame bytes
by Takashi Sakamoto 26 Sep '13

26 Sep '13

Calculating frame bytes can be replaced with inline function in include/sound/pcm.h. Signed-off-by: Takashi Sakamoto <o-takashi(a)sakamocchi.jp> --- sound/firewire/amdtp.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/firewire/amdtp.c b/sound/firewire/amdtp.c index ea995af..4b08b25 100644 --- a/sound/firewire/amdtp.c +++ b/sound/firewire/amdtp.c @@ -268,7 +268,7 @@ static void amdtp_write_s32(struct amdtp_out_stream *s, channels = s->pcm_channels; src = (void *)runtime->dma_area + - s->pcm_buffer_pointer * (runtime->frame_bits / 8); + frames_to_bytes(runtime, s->pcm_buffer_pointer); remaining_frames = runtime->buffer_size - s->pcm_buffer_pointer; frame_step = s->data_block_quadlets - channels; @@ -294,7 +294,7 @@ static void amdtp_write_s16(struct amdtp_out_stream *s, channels = s->pcm_channels; src = (void *)runtime->dma_area + - s->pcm_buffer_pointer * (runtime->frame_bits / 8); + frames_to_bytes(runtime, s->pcm_buffer_pointer); remaining_frames = runtime->buffer_size - s->pcm_buffer_pointer; frame_step = s->data_block_quadlets - channels; -- 1.8.1.2

2 1

Re: [alsa-devel] [PATCH 28/51] DMA-API: sound: fix dma mask handling in a lot of drivers
by Takashi Iwai 26 Sep '13

26 Sep '13

Hi, sorry for the lat response, as I've been traveling in the last weeks. At Thu, 19 Sep 2013 22:53:02 +0100, Russell King wrote: > > This code sequence is unsafe in modules: > > static u64 mask = DMA_BIT_MASK(something); > ... > if (!dev->dma_mask) > dev->dma_mask = &mask; > > as if a module is reloaded, the mask will be pointing at the original > module's mask address, and this can lead to oopses. Moreover, they > all follow this with: > > if (!dev->coherent_dma_mask) > dev->coherent_dma_mask = mask; > > where 'mask' is the same value as the statically defined mask, and this > bypasses the architecture's check on whether the DMA mask is possible. > > Fix these issues by using the new dma_coerce_coherent_and_mask() > function. > > Signed-off-by: Russell King <rmk+kernel(a)arm.linux.org.uk> Applied with Mark's ack now. BTW, sound/soc/fsl/imx-pcm-fiq.c wasn't covered by this patch, so I fixed it, too. Thanks! Takashi > --- > sound/arm/pxa2xx-pcm.c | 9 +++------ > sound/soc/atmel/atmel-pcm.c | 11 ++++------- > sound/soc/blackfin/bf5xx-ac97-pcm.c | 11 ++++------- > sound/soc/blackfin/bf5xx-i2s-pcm.c | 10 ++++------ > sound/soc/davinci/davinci-pcm.c | 9 +++------ > sound/soc/fsl/fsl_dma.c | 9 +++------ > sound/soc/fsl/mpc5200_dma.c | 10 ++++------ > sound/soc/jz4740/jz4740-pcm.c | 12 ++++-------- > sound/soc/kirkwood/kirkwood-dma.c | 9 +++------ > sound/soc/nuc900/nuc900-pcm.c | 9 ++++----- > sound/soc/omap/omap-pcm.c | 11 ++++------- > sound/soc/pxa/pxa2xx-pcm.c | 11 ++++------- > sound/soc/s6000/s6000-pcm.c | 9 +++------ > sound/soc/samsung/dma.c | 11 ++++------- > sound/soc/samsung/idma.c | 11 ++++------- > 15 files changed, 55 insertions(+), 97 deletions(-) > > diff --git a/sound/arm/pxa2xx-pcm.c b/sound/arm/pxa2xx-pcm.c > index 69a2455..fb3b76f 100644 > --- a/sound/arm/pxa2xx-pcm.c > +++ b/sound/arm/pxa2xx-pcm.c > @@ -83,8 +83,6 @@ static struct snd_pcm_ops pxa2xx_pcm_ops = { > .mmap = pxa2xx_pcm_mmap, > }; > > -static u64 pxa2xx_pcm_dmamask = 0xffffffff; > - > int pxa2xx_pcm_new(struct snd_card *card, struct pxa2xx_pcm_client *client, > struct snd_pcm **rpcm) > { > @@ -100,10 +98,9 @@ int pxa2xx_pcm_new(struct snd_card *card, struct pxa2xx_pcm_client *client, > pcm->private_data = client; > pcm->private_free = pxa2xx_pcm_free_dma_buffers; > > - if (!card->dev->dma_mask) > - card->dev->dma_mask = &pxa2xx_pcm_dmamask; > - if (!card->dev->coherent_dma_mask) > - card->dev->coherent_dma_mask = 0xffffffff; > + ret = dma_coerce_mask_and_coherent_mask(card->dev, DMA_BIT_MASK(32)); > + if (ret) > + goto out; > > if (play) { > int stream = SNDRV_PCM_STREAM_PLAYBACK; > diff --git a/sound/soc/atmel/atmel-pcm.c b/sound/soc/atmel/atmel-pcm.c > index 3109db7..fbb87e3 100644 > --- a/sound/soc/atmel/atmel-pcm.c > +++ b/sound/soc/atmel/atmel-pcm.c > @@ -68,18 +68,15 @@ int atmel_pcm_mmap(struct snd_pcm_substream *substream, > } > EXPORT_SYMBOL_GPL(atmel_pcm_mmap); > > -static u64 atmel_pcm_dmamask = DMA_BIT_MASK(32); > - > int atmel_pcm_new(struct snd_soc_pcm_runtime *rtd) > { > struct snd_card *card = rtd->card->snd_card; > struct snd_pcm *pcm = rtd->pcm; > - int ret = 0; > + int ret; > > - if (!card->dev->dma_mask) > - card->dev->dma_mask = &atmel_pcm_dmamask; > - if (!card->dev->coherent_dma_mask) > - card->dev->coherent_dma_mask = DMA_BIT_MASK(32); > + ret = dma_coerce_mask_and_coherent(card->dev, DMA_BIT_MASK(32)); > + if (ret) > + return ret; > > if (pcm->streams[SNDRV_PCM_STREAM_PLAYBACK].substream) { > pr_debug("atmel-pcm: allocating PCM playback DMA buffer\n"); > diff --git a/sound/soc/blackfin/bf5xx-ac97-pcm.c b/sound/soc/blackfin/bf5xx-ac97-pcm.c > index 53f8408..1d4c676 100644 > --- a/sound/soc/blackfin/bf5xx-ac97-pcm.c > +++ b/sound/soc/blackfin/bf5xx-ac97-pcm.c > @@ -415,19 +415,16 @@ static void bf5xx_pcm_free_dma_buffers(struct snd_pcm *pcm) > } > } > > -static u64 bf5xx_pcm_dmamask = DMA_BIT_MASK(32); > - > static int bf5xx_pcm_ac97_new(struct snd_soc_pcm_runtime *rtd) > { > struct snd_card *card = rtd->card->snd_card; > struct snd_pcm *pcm = rtd->pcm; > - int ret = 0; > + int ret; > > pr_debug("%s enter\n", __func__); > - if (!card->dev->dma_mask) > - card->dev->dma_mask = &bf5xx_pcm_dmamask; > - if (!card->dev->coherent_dma_mask) > - card->dev->coherent_dma_mask = DMA_BIT_MASK(32); > + ret = dma_coerce_mask_and_coherent(card->dev, DMA_BIT_MASK(32)); > + if (ret) > + return ret; > > if (pcm->streams[SNDRV_PCM_STREAM_PLAYBACK].substream) { > ret = bf5xx_pcm_preallocate_dma_buffer(pcm, > diff --git a/sound/soc/blackfin/bf5xx-i2s-pcm.c b/sound/soc/blackfin/bf5xx-i2s-pcm.c > index 9cb4a80..2a5b434 100644 > --- a/sound/soc/blackfin/bf5xx-i2s-pcm.c > +++ b/sound/soc/blackfin/bf5xx-i2s-pcm.c > @@ -323,18 +323,16 @@ static struct snd_pcm_ops bf5xx_pcm_i2s_ops = { > .silence = bf5xx_pcm_silence, > }; > > -static u64 bf5xx_pcm_dmamask = DMA_BIT_MASK(32); > - > static int bf5xx_pcm_i2s_new(struct snd_soc_pcm_runtime *rtd) > { > struct snd_card *card = rtd->card->snd_card; > size_t size = bf5xx_pcm_hardware.buffer_bytes_max; > + int ret; > > pr_debug("%s enter\n", __func__); > - if (!card->dev->dma_mask) > - card->dev->dma_mask = &bf5xx_pcm_dmamask; > - if (!card->dev->coherent_dma_mask) > - card->dev->coherent_dma_mask = DMA_BIT_MASK(32); > + ret = dma_coerce_mask_and_coherent(card->dev, DMA_BIT_MASK(32)); > + if (ret) > + return ret; > > return snd_pcm_lib_preallocate_pages_for_all(rtd->pcm, > SNDRV_DMA_TYPE_DEV, card->dev, size, size); > diff --git a/sound/soc/davinci/davinci-pcm.c b/sound/soc/davinci/davinci-pcm.c > index 8460edc..84a63c6 100644 > --- a/sound/soc/davinci/davinci-pcm.c > +++ b/sound/soc/davinci/davinci-pcm.c > @@ -844,18 +844,15 @@ static void davinci_pcm_free(struct snd_pcm *pcm) > } > } > > -static u64 davinci_pcm_dmamask = DMA_BIT_MASK(32); > - > static int davinci_pcm_new(struct snd_soc_pcm_runtime *rtd) > { > struct snd_card *card = rtd->card->snd_card; > struct snd_pcm *pcm = rtd->pcm; > int ret; > > - if (!card->dev->dma_mask) > - card->dev->dma_mask = &davinci_pcm_dmamask; > - if (!card->dev->coherent_dma_mask) > - card->dev->coherent_dma_mask = DMA_BIT_MASK(32); > + ret = dma_coerce_mask_and_coherent(card->dev, DMA_BIT_MASK(32)); > + if (ret) > + return ret; > > if (pcm->streams[SNDRV_PCM_STREAM_PLAYBACK].substream) { > ret = davinci_pcm_preallocate_dma_buffer(pcm, > diff --git a/sound/soc/fsl/fsl_dma.c b/sound/soc/fsl/fsl_dma.c > index 9cc5c1f..f73c7ef 100644 > --- a/sound/soc/fsl/fsl_dma.c > +++ b/sound/soc/fsl/fsl_dma.c > @@ -298,14 +298,11 @@ static int fsl_dma_new(struct snd_soc_pcm_runtime *rtd) > { > struct snd_card *card = rtd->card->snd_card; > struct snd_pcm *pcm = rtd->pcm; > - static u64 fsl_dma_dmamask = DMA_BIT_MASK(36); > int ret; > > - if (!card->dev->dma_mask) > - card->dev->dma_mask = &fsl_dma_dmamask; > - > - if (!card->dev->coherent_dma_mask) > - card->dev->coherent_dma_mask = fsl_dma_dmamask; > + ret = dma_coerce_mask_and_coherent(card->dev, DMA_BIT_MASK(36)); > + if (ret) > + return ret; > > /* Some codecs have separate DAIs for playback and capture, so we > * should allocate a DMA buffer only for the streams that are valid. > diff --git a/sound/soc/fsl/mpc5200_dma.c b/sound/soc/fsl/mpc5200_dma.c > index 2a847ca..8fcf224 100644 > --- a/sound/soc/fsl/mpc5200_dma.c > +++ b/sound/soc/fsl/mpc5200_dma.c > @@ -299,7 +299,6 @@ static struct snd_pcm_ops psc_dma_ops = { > .hw_params = psc_dma_hw_params, > }; > > -static u64 psc_dma_dmamask = DMA_BIT_MASK(32); > static int psc_dma_new(struct snd_soc_pcm_runtime *rtd) > { > struct snd_card *card = rtd->card->snd_card; > @@ -307,15 +306,14 @@ static int psc_dma_new(struct snd_soc_pcm_runtime *rtd) > struct snd_pcm *pcm = rtd->pcm; > struct psc_dma *psc_dma = snd_soc_dai_get_drvdata(rtd->cpu_dai); > size_t size = psc_dma_hardware.buffer_bytes_max; > - int rc = 0; > + int rc; > > dev_dbg(rtd->platform->dev, "psc_dma_new(card=%p, dai=%p, pcm=%p)\n", > card, dai, pcm); > > - if (!card->dev->dma_mask) > - card->dev->dma_mask = &psc_dma_dmamask; > - if (!card->dev->coherent_dma_mask) > - card->dev->coherent_dma_mask = DMA_BIT_MASK(32); > + rc = dma_coerce_mask_and_coherent(card->dev, DMA_BIT_MASK(32)); > + if (rc) > + return rc; > > if (pcm->streams[SNDRV_PCM_STREAM_PLAYBACK].substream) { > rc = snd_dma_alloc_pages(SNDRV_DMA_TYPE_DEV, pcm->card->dev, > diff --git a/sound/soc/jz4740/jz4740-pcm.c b/sound/soc/jz4740/jz4740-pcm.c > index 7100592..1d7ef28 100644 > --- a/sound/soc/jz4740/jz4740-pcm.c > +++ b/sound/soc/jz4740/jz4740-pcm.c > @@ -297,19 +297,15 @@ static void jz4740_pcm_free(struct snd_pcm *pcm) > } > } > > -static u64 jz4740_pcm_dmamask = DMA_BIT_MASK(32); > - > static int jz4740_pcm_new(struct snd_soc_pcm_runtime *rtd) > { > struct snd_card *card = rtd->card->snd_card; > struct snd_pcm *pcm = rtd->pcm; > - int ret = 0; > - > - if (!card->dev->dma_mask) > - card->dev->dma_mask = &jz4740_pcm_dmamask; > + int ret; > > - if (!card->dev->coherent_dma_mask) > - card->dev->coherent_dma_mask = DMA_BIT_MASK(32); > + ret = dma_coerce_mask_and_coherent(card->dev, DMA_BIT_MASK(32)); > + if (ret) > + return ret; > > if (pcm->streams[SNDRV_PCM_STREAM_PLAYBACK].substream) { > ret = jz4740_pcm_preallocate_dma_buffer(pcm, > diff --git a/sound/soc/kirkwood/kirkwood-dma.c b/sound/soc/kirkwood/kirkwood-dma.c > index b238434..3814bb0 100644 > --- a/sound/soc/kirkwood/kirkwood-dma.c > +++ b/sound/soc/kirkwood/kirkwood-dma.c > @@ -59,8 +59,6 @@ static struct snd_pcm_hardware kirkwood_dma_snd_hw = { > .fifo_size = 0, > }; > > -static u64 kirkwood_dma_dmamask = DMA_BIT_MASK(32); > - > static irqreturn_t kirkwood_dma_irq(int irq, void *dev_id) > { > struct kirkwood_dma_data *priv = dev_id; > @@ -292,10 +290,9 @@ static int kirkwood_dma_new(struct snd_soc_pcm_runtime *rtd) > struct snd_pcm *pcm = rtd->pcm; > int ret; > > - if (!card->dev->dma_mask) > - card->dev->dma_mask = &kirkwood_dma_dmamask; > - if (!card->dev->coherent_dma_mask) > - card->dev->coherent_dma_mask = DMA_BIT_MASK(32); > + ret = dma_coerce_mask_and_coherent(card->dev, DMA_BIT_MASK(32)); > + if (ret) > + return ret; > > if (pcm->streams[SNDRV_PCM_STREAM_PLAYBACK].substream) { > ret = kirkwood_dma_preallocate_dma_buffer(pcm, > diff --git a/sound/soc/nuc900/nuc900-pcm.c b/sound/soc/nuc900/nuc900-pcm.c > index c894ff0..f588ee4 100644 > --- a/sound/soc/nuc900/nuc900-pcm.c > +++ b/sound/soc/nuc900/nuc900-pcm.c > @@ -314,16 +314,15 @@ static void nuc900_dma_free_dma_buffers(struct snd_pcm *pcm) > snd_pcm_lib_preallocate_free_for_all(pcm); > } > > -static u64 nuc900_pcm_dmamask = DMA_BIT_MASK(32); > static int nuc900_dma_new(struct snd_soc_pcm_runtime *rtd) > { > struct snd_card *card = rtd->card->snd_card; > struct snd_pcm *pcm = rtd->pcm; > + int ret; > > - if (!card->dev->dma_mask) > - card->dev->dma_mask = &nuc900_pcm_dmamask; > - if (!card->dev->coherent_dma_mask) > - card->dev->coherent_dma_mask = DMA_BIT_MASK(32); > + ret = dma_coerce_mask_and_coherent(card->dev, DMA_BIT_MASK(32)); > + if (ret) > + return ret; > > snd_pcm_lib_preallocate_pages_for_all(pcm, SNDRV_DMA_TYPE_DEV, > card->dev, 4 * 1024, (4 * 1024) - 1); > diff --git a/sound/soc/omap/omap-pcm.c b/sound/soc/omap/omap-pcm.c > index a11405d..b8fa986 100644 > --- a/sound/soc/omap/omap-pcm.c > +++ b/sound/soc/omap/omap-pcm.c > @@ -156,8 +156,6 @@ static struct snd_pcm_ops omap_pcm_ops = { > .mmap = omap_pcm_mmap, > }; > > -static u64 omap_pcm_dmamask = DMA_BIT_MASK(64); > - > static int omap_pcm_preallocate_dma_buffer(struct snd_pcm *pcm, > int stream) > { > @@ -202,12 +200,11 @@ static int omap_pcm_new(struct snd_soc_pcm_runtime *rtd) > { > struct snd_card *card = rtd->card->snd_card; > struct snd_pcm *pcm = rtd->pcm; > - int ret = 0; > + int ret; > > - if (!card->dev->dma_mask) > - card->dev->dma_mask = &omap_pcm_dmamask; > - if (!card->dev->coherent_dma_mask) > - card->dev->coherent_dma_mask = DMA_BIT_MASK(64); > + ret = dma_coerce_mask_and_coherent(card->dev, DMA_BIT_MASK(64)); > + if (ret) > + return ret; > > if (pcm->streams[SNDRV_PCM_STREAM_PLAYBACK].substream) { > ret = omap_pcm_preallocate_dma_buffer(pcm, > diff --git a/sound/soc/pxa/pxa2xx-pcm.c b/sound/soc/pxa/pxa2xx-pcm.c > index 806da27..d58b09f 100644 > --- a/sound/soc/pxa/pxa2xx-pcm.c > +++ b/sound/soc/pxa/pxa2xx-pcm.c > @@ -87,18 +87,15 @@ static struct snd_pcm_ops pxa2xx_pcm_ops = { > .mmap = pxa2xx_pcm_mmap, > }; > > -static u64 pxa2xx_pcm_dmamask = DMA_BIT_MASK(32); > - > static int pxa2xx_soc_pcm_new(struct snd_soc_pcm_runtime *rtd) > { > struct snd_card *card = rtd->card->snd_card; > struct snd_pcm *pcm = rtd->pcm; > - int ret = 0; > + int ret; > > - if (!card->dev->dma_mask) > - card->dev->dma_mask = &pxa2xx_pcm_dmamask; > - if (!card->dev->coherent_dma_mask) > - card->dev->coherent_dma_mask = DMA_BIT_MASK(32); > + ret = dma_coerce_mask_and_coherent(card->dev, DMA_BIT_MASK(32)); > + if (ret) > + return ret; > > if (pcm->streams[SNDRV_PCM_STREAM_PLAYBACK].substream) { > ret = pxa2xx_pcm_preallocate_dma_buffer(pcm, > diff --git a/sound/soc/s6000/s6000-pcm.c b/sound/soc/s6000/s6000-pcm.c > index d0740a7..283620a 100644 > --- a/sound/soc/s6000/s6000-pcm.c > +++ b/sound/soc/s6000/s6000-pcm.c > @@ -444,8 +444,6 @@ static void s6000_pcm_free(struct snd_pcm *pcm) > snd_pcm_lib_preallocate_free_for_all(pcm); > } > > -static u64 s6000_pcm_dmamask = DMA_BIT_MASK(32); > - > static int s6000_pcm_new(struct snd_soc_pcm_runtime *runtime) > { > struct snd_card *card = runtime->card->snd_card; > @@ -456,10 +454,9 @@ static int s6000_pcm_new(struct snd_soc_pcm_runtime *runtime) > params = snd_soc_dai_get_dma_data(runtime->cpu_dai, > pcm->streams[SNDRV_PCM_STREAM_PLAYBACK].substream); > > - if (!card->dev->dma_mask) > - card->dev->dma_mask = &s6000_pcm_dmamask; > - if (!card->dev->coherent_dma_mask) > - card->dev->coherent_dma_mask = DMA_BIT_MASK(32); > + res = dma_coerce_mask_and_coherent(card->dev, DMA_BIT_MASK(32)); > + if (res) > + return res; > > if (params->dma_in) { > s6dmac_disable_chan(DMA_MASK_DMAC(params->dma_in), > diff --git a/sound/soc/samsung/dma.c b/sound/soc/samsung/dma.c > index 9338d11..fe2748b 100644 > --- a/sound/soc/samsung/dma.c > +++ b/sound/soc/samsung/dma.c > @@ -406,20 +406,17 @@ static void dma_free_dma_buffers(struct snd_pcm *pcm) > } > } > > -static u64 dma_mask = DMA_BIT_MASK(32); > - > static int dma_new(struct snd_soc_pcm_runtime *rtd) > { > struct snd_card *card = rtd->card->snd_card; > struct snd_pcm *pcm = rtd->pcm; > - int ret = 0; > + int ret; > > pr_debug("Entered %s\n", __func__); > > - if (!card->dev->dma_mask) > - card->dev->dma_mask = &dma_mask; > - if (!card->dev->coherent_dma_mask) > - card->dev->coherent_dma_mask = DMA_BIT_MASK(32); > + ret = dma_coerce_mask_and_coherent(card->dev, DMA_BIT_MASK(32)); > + if (ret) > + return ret; > > if (pcm->streams[SNDRV_PCM_STREAM_PLAYBACK].substream) { > ret = preallocate_dma_buffer(pcm, > diff --git a/sound/soc/samsung/idma.c b/sound/soc/samsung/idma.c > index ce1e1e1..e4f318f 100644 > --- a/sound/soc/samsung/idma.c > +++ b/sound/soc/samsung/idma.c > @@ -383,18 +383,15 @@ static int preallocate_idma_buffer(struct snd_pcm *pcm, int stream) > return 0; > } > > -static u64 idma_mask = DMA_BIT_MASK(32); > - > static int idma_new(struct snd_soc_pcm_runtime *rtd) > { > struct snd_card *card = rtd->card->snd_card; > struct snd_pcm *pcm = rtd->pcm; > - int ret = 0; > + int ret; > > - if (!card->dev->dma_mask) > - card->dev->dma_mask = &idma_mask; > - if (!card->dev->coherent_dma_mask) > - card->dev->coherent_dma_mask = DMA_BIT_MASK(32); > + ret = dma_coerce_mask_and_coherent(card->dev, DMA_BIT_MASK(32)); > + if (ret) > + return ret; > > if (pcm->streams[SNDRV_PCM_STREAM_PLAYBACK].substream) { > ret = preallocate_idma_buffer(pcm, > -- > 1.7.4.4 >

2 3

Re: [alsa-devel] Cannot load snd-usb-audio on 3.12-rc2
by Daniel Mack 26 Sep '13

26 Sep '13

Hi Sarah, On 24.09.2013 01:41, Sarah Sharp wrote: > The snd-usb-audio driver no longer loads properly on 3.12-rc2 when I > plug in my USB headset. It worked fine on 3.11-rc4 (not sure about > vanilla 3.11). > > Trying to manually load the driver fails: > > sarah@xanatos:~$ sudo modprobe snd-usb-audio > FATAL: Error inserting snd_usb_audio > (/lib/modules/3.12.0-rc2/kernel/sound/usb/snd-usb-audio.ko): Invalid argument > > dmesg shows: > > [ 44.231668] snd_usb_audio: `-2' invalid for parameter `index' That's really strange. I just tested on an ARM board where I currently run 3.12-rc2 as well, and I can load the module without problems. Also, sound/usb/card.c (where the params are declared) hasn't seen an update in 3.12. I suspect your OS is passing "index=-2" from something like /etc/modprobe.d/* or cmdline settings. Could you investigate in that direction? HTH, Daniel

2 1

[alsa-devel] [PATCH] ALSA: improve buffer size computations for USB PCM audio
by Alan Stern 26 Sep '13

26 Sep '13

This patch changes the way URBs are allocated and their sizes are determined for PCM playback in the snd-usb-audio driver. Currently the driver allocates too few URBs for endpoints that don't use implicit sync, making underruns more likely to occur. This may be a holdover from before I/O delays could be measured accurately; in any case, it is no longer necessary. The patch allocates as many URBs as possible, subject to four limitations: The total number of URBs for the endpoint is not allowed to exceed MAX_URBS (which the patch increases from 8 to 12). The total number of packets per URB is not allowed to exceed MAX_PACKS (or MAX_PACKS_HS for high-speed devices), which is decreased from 20 to 6. The total duration of queued data is not allowed to exceed MAX_QUEUE, which is decreased from 24 ms to 18 ms. The total number of ALSA frames in the output queue is not allowed to exceed the ALSA buffer size. The last requirement is the hardest to implement. Currently the number of URBs needed to fill a buffer cannot be determined in advance, because a buffer contains a fixed number of frames whereas the number of frames in an URB varies to match shifts in the device's clock rate. To solve this problem, the patch changes the logic for deciding how many packets an URB should contain. Rather than using as many as possible without exceeding an ALSA period boundary, now the driver uses only as many packets as needed to transfer a predetermined number of frames. As a result, unless the device's clock has an exceedingly variable rate, the number of URBs making up each period (and hence each buffer) will remain constant. The overall effect of the patch is that playback works better in low-latency settings. The user can still specify values for frames/period and periods/buffer that exceed the capabilities of the hardware, of course. But for values that are within those capabilities, the performance will be improved. For example, testing shows that a high-speed device can handle 32 frames/period and 3 periods/buffer at 48 KHz, whereas the current driver starts to get glitchy at 64 frames/period and 2 periods/buffer. A side effect of these changes is that the "nrpacks" module parameter is no longer used. The patch removes it. Signed-off-by: Alan Stern <stern(a)rowland.harvard.edu> CC: Clemens Ladisch <clemens(a)ladisch.de> Tested-by: Daniel Mack <zonque(a)gmail.com> Tested-by: Eldad Zack <eldad(a)fogrefinery.com> --- [as1712] sound/usb/card.c | 8 --- sound/usb/card.h | 10 +++- sound/usb/endpoint.c | 108 +++++++++++++++++++++++++++++---------------------- sound/usb/endpoint.h | 2 sound/usb/pcm.c | 14 +++++- sound/usb/usbaudio.h | 1 6 files changed, 83 insertions(+), 60 deletions(-) Index: usb-3.11/sound/usb/usbaudio.h =================================================================== --- usb-3.11.orig/sound/usb/usbaudio.h +++ usb-3.11/sound/usb/usbaudio.h @@ -55,7 +55,6 @@ struct snd_usb_audio { struct list_head mixer_list; /* list of mixer interfaces */ int setup; /* from the 'device_setup' module param */ - int nrpacks; /* from the 'nrpacks' module param */ bool autoclock; /* from the 'autoclock' module param */ struct usb_host_interface *ctrl_intf; /* the audio control interface */ Index: usb-3.11/sound/usb/card.c =================================================================== --- usb-3.11.orig/sound/usb/card.c +++ usb-3.11/sound/usb/card.c @@ -79,7 +79,6 @@ static bool enable[SNDRV_CARDS] = SNDRV_ /* Vendor/product IDs for this card */ static int vid[SNDRV_CARDS] = { [0 ... (SNDRV_CARDS-1)] = -1 }; static int pid[SNDRV_CARDS] = { [0 ... (SNDRV_CARDS-1)] = -1 }; -static int nrpacks = 8; /* max. number of packets per urb */ static int device_setup[SNDRV_CARDS]; /* device parameter for this card */ static bool ignore_ctl_error; static bool autoclock = true; @@ -94,8 +93,6 @@ module_param_array(vid, int, NULL, 0444) MODULE_PARM_DESC(vid, "Vendor ID for the USB audio device."); module_param_array(pid, int, NULL, 0444); MODULE_PARM_DESC(pid, "Product ID for the USB audio device."); -module_param(nrpacks, int, 0644); -MODULE_PARM_DESC(nrpacks, "Max. number of packets per URB."); module_param_array(device_setup, int, NULL, 0444); MODULE_PARM_DESC(device_setup, "Specific device setup (if needed)."); module_param(ignore_ctl_error, bool, 0444); @@ -374,7 +371,6 @@ static int snd_usb_audio_create(struct u chip->dev = dev; chip->card = card; chip->setup = device_setup[idx]; - chip->nrpacks = nrpacks; chip->autoclock = autoclock; chip->probing = 1; @@ -756,10 +752,6 @@ static struct usb_driver usb_audio_drive static int __init snd_usb_audio_init(void) { - if (nrpacks < 1 || nrpacks > MAX_PACKS) { - printk(KERN_WARNING "invalid nrpacks value.\n"); - return -EINVAL; - } return usb_register(&usb_audio_driver); } Index: usb-3.11/sound/usb/endpoint.h =================================================================== --- usb-3.11.orig/sound/usb/endpoint.h +++ usb-3.11/sound/usb/endpoint.h @@ -12,6 +12,8 @@ int snd_usb_endpoint_set_params(struct s snd_pcm_format_t pcm_format, unsigned int channels, unsigned int period_bytes, + unsigned int period_frames, + unsigned int buffer_periods, unsigned int rate, struct audioformat *fmt, struct snd_usb_endpoint *sync_ep); Index: usb-3.11/sound/usb/card.h =================================================================== --- usb-3.11.orig/sound/usb/card.h +++ usb-3.11/sound/usb/card.h @@ -2,11 +2,11 @@ #define __USBAUDIO_CARD_H #define MAX_NR_RATES 1024 -#define MAX_PACKS 20 +#define MAX_PACKS 6 /* per URB */ #define MAX_PACKS_HS (MAX_PACKS * 8) /* in high speed mode */ -#define MAX_URBS 8 +#define MAX_URBS 12 #define SYNC_URBS 4 /* always four urbs for sync */ -#define MAX_QUEUE 24 /* try not to exceed this queue length, in ms */ +#define MAX_QUEUE 18 /* try not to exceed this queue length, in ms */ struct audioformat { struct list_head list; @@ -87,6 +87,7 @@ struct snd_usb_endpoint { unsigned int phase; /* phase accumulator */ unsigned int maxpacksize; /* max packet size in bytes */ unsigned int maxframesize; /* max packet size in frames */ + unsigned int max_urb_frames; /* max URB size in frames */ unsigned int curpacksize; /* current packet size in bytes (for capture) */ unsigned int curframesize; /* current packet size in frames (for capture) */ unsigned int syncmaxsize; /* sync endpoint packet size */ @@ -116,6 +117,8 @@ struct snd_usb_substream { unsigned int channels_max; /* max channels in the all audiofmts */ unsigned int cur_rate; /* current rate (for hw_params callback) */ unsigned int period_bytes; /* current period bytes (for hw_params callback) */ + unsigned int period_frames; /* current frames per period */ + unsigned int buffer_periods; /* current periods per buffer */ unsigned int altset_idx; /* USB data format: index of alternate setting */ unsigned int txfr_quirk:1; /* allow sub-frame alignment */ unsigned int fmt_type; /* USB audio format type (1-3) */ @@ -125,6 +128,7 @@ struct snd_usb_substream { unsigned int hwptr_done; /* processed byte position in the buffer */ unsigned int transfer_done; /* processed frames since last period update */ + unsigned int frame_limit; /* limits number of packets in URB */ /* data and sync endpoints for this stream */ unsigned int ep_num; /* the endpoint number */ Index: usb-3.11/sound/usb/pcm.c =================================================================== --- usb-3.11.orig/sound/usb/pcm.c +++ usb-3.11/sound/usb/pcm.c @@ -562,6 +562,7 @@ static int configure_sync_endpoint(struc subs->pcm_format, subs->channels, subs->period_bytes, + 0, 0, subs->cur_rate, subs->cur_audiofmt, NULL); @@ -598,6 +599,7 @@ static int configure_sync_endpoint(struc subs->pcm_format, sync_fp->channels, sync_period_bytes, + 0, 0, subs->cur_rate, sync_fp, NULL); @@ -620,6 +622,8 @@ static int configure_endpoint(struct snd subs->pcm_format, subs->channels, subs->period_bytes, + subs->period_frames, + subs->buffer_periods, subs->cur_rate, subs->cur_audiofmt, subs->sync_endpoint); @@ -656,6 +660,8 @@ static int snd_usb_hw_params(struct snd_ subs->pcm_format = params_format(hw_params); subs->period_bytes = params_period_bytes(hw_params); + subs->period_frames = params_period_size(hw_params); + subs->buffer_periods = params_periods(hw_params); subs->channels = params_channels(hw_params); subs->cur_rate = params_rate(hw_params); @@ -1330,6 +1336,7 @@ static void prepare_playback_urb(struct frames = 0; urb->number_of_packets = 0; spin_lock_irqsave(&subs->lock, flags); + subs->frame_limit += ep->max_urb_frames; for (i = 0; i < ctx->packets; i++) { if (ctx->packet_size[i]) counts = ctx->packet_size[i]; @@ -1344,6 +1351,7 @@ static void prepare_playback_urb(struct subs->transfer_done += counts; if (subs->transfer_done >= runtime->period_size) { subs->transfer_done -= runtime->period_size; + subs->frame_limit = 0; period_elapsed = 1; if (subs->fmt_type == UAC_FORMAT_TYPE_II) { if (subs->transfer_done > 0) { @@ -1366,8 +1374,10 @@ static void prepare_playback_urb(struct break; } } - if (period_elapsed && - !snd_usb_endpoint_implicit_feedback_sink(subs->data_endpoint)) /* finish at the period boundary */ + /* finish at the period boundary or after enough frames */ + if ((period_elapsed || + subs->transfer_done >= subs->frame_limit) && + !snd_usb_endpoint_implicit_feedback_sink(ep)) break; } bytes = frames * ep->stride; Index: usb-3.11/sound/usb/endpoint.c =================================================================== --- usb-3.11.orig/sound/usb/endpoint.c +++ usb-3.11/sound/usb/endpoint.c @@ -571,11 +571,14 @@ static int data_ep_set_params(struct snd snd_pcm_format_t pcm_format, unsigned int channels, unsigned int period_bytes, + unsigned int frames_per_period, + unsigned int periods_per_buffer, struct audioformat *fmt, struct snd_usb_endpoint *sync_ep) { - unsigned int maxsize, i, urb_packs, total_packs, packs_per_ms; - int is_playback = usb_pipeout(ep->pipe); + unsigned int maxsize, minsize, packs_per_ms, max_packs_per_urb; + unsigned int max_packs_per_period, urbs_per_period, urb_packs; + unsigned int max_urbs, i; int frame_bits = snd_pcm_format_physical_width(pcm_format) * channels; if (pcm_format == SNDRV_PCM_FORMAT_DSD_U16_LE && fmt->dsd_dop) { @@ -608,58 +611,67 @@ static int data_ep_set_params(struct snd else ep->curpacksize = maxsize; - if (snd_usb_get_speed(ep->chip->dev) != USB_SPEED_FULL) + if (snd_usb_get_speed(ep->chip->dev) != USB_SPEED_FULL) { packs_per_ms = 8 >> ep->datainterval; - else - packs_per_ms = 1; - - if (is_playback && !snd_usb_endpoint_implicit_feedback_sink(ep)) { - urb_packs = max(ep->chip->nrpacks, 1); - urb_packs = min(urb_packs, (unsigned int) MAX_PACKS); + max_packs_per_urb = MAX_PACKS_HS; } else { - urb_packs = 1; + packs_per_ms = 1; + max_packs_per_urb = MAX_PACKS; } - - urb_packs *= packs_per_ms; - if (sync_ep && !snd_usb_endpoint_implicit_feedback_sink(ep)) - urb_packs = min(urb_packs, 1U << sync_ep->syncinterval); + max_packs_per_urb = min(max_packs_per_urb, + 1U << sync_ep->syncinterval); + max_packs_per_urb = max(1u, max_packs_per_urb >> ep->datainterval); + + /* + * Capture endpoints need to use small URBs because there's no way + * to tell in advance where the next period will end, and we don't + * want the next URB to complete much after the period ends. + * + * Playback endpoints with implicit sync much use the same parameters + * as their corresponding capture endpoint. + */ + if (usb_pipein(ep->pipe) || + snd_usb_endpoint_implicit_feedback_sink(ep)) { + + /* make capture URBs <= 1 ms and smaller than a period */ + urb_packs = min(max_packs_per_urb, packs_per_ms); + while (urb_packs > 1 && urb_packs * maxsize >= period_bytes) + urb_packs >>= 1; + ep->nurbs = MAX_URBS; - /* decide how many packets to be used */ - if (is_playback && !snd_usb_endpoint_implicit_feedback_sink(ep)) { - unsigned int minsize, maxpacks; + /* + * Playback endpoints without implicit sync are adjusted so that + * a period fits as evenly as possible in the smallest number of + * URBs. The total number of URBs is adjusted to the size of the + * ALSA buffer, subject to the MAX_URBS and MAX_QUEUE limits. + */ + } else { /* determine how small a packet can be */ - minsize = (ep->freqn >> (16 - ep->datainterval)) - * (frame_bits >> 3); + minsize = (ep->freqn >> (16 - ep->datainterval)) * + (frame_bits >> 3); /* with sync from device, assume it can be 12% lower */ if (sync_ep) minsize -= minsize >> 3; minsize = max(minsize, 1u); - total_packs = (period_bytes + minsize - 1) / minsize; - /* we need at least two URBs for queueing */ - if (total_packs < 2) { - total_packs = 2; - } else { - /* and we don't want too long a queue either */ - maxpacks = max(MAX_QUEUE * packs_per_ms, urb_packs * 2); - total_packs = min(total_packs, maxpacks); - } - } else { - while (urb_packs > 1 && urb_packs * maxsize >= period_bytes) - urb_packs >>= 1; - total_packs = MAX_URBS * urb_packs; - } - ep->nurbs = (total_packs + urb_packs - 1) / urb_packs; - if (ep->nurbs > MAX_URBS) { - /* too much... */ - ep->nurbs = MAX_URBS; - total_packs = MAX_URBS * urb_packs; - } else if (ep->nurbs < 2) { - /* too little - we need at least two packets - * to ensure contiguous playback/capture - */ - ep->nurbs = 2; + /* how many packets will contain an entire ALSA period? */ + max_packs_per_period = DIV_ROUND_UP(period_bytes, minsize); + + /* how many URBs will contain a period? */ + urbs_per_period = DIV_ROUND_UP(max_packs_per_period, + max_packs_per_urb); + /* how many packets are needed in each URB? */ + urb_packs = DIV_ROUND_UP(max_packs_per_period, urbs_per_period); + + /* limit the number of frames in a single URB */ + ep->max_urb_frames = DIV_ROUND_UP(frames_per_period, + urbs_per_period); + + /* try to use enough URBs to contain an entire ALSA buffer */ + max_urbs = min((unsigned) MAX_URBS, + MAX_QUEUE * packs_per_ms / urb_packs); + ep->nurbs = min(max_urbs, urbs_per_period * periods_per_buffer); } /* allocate and initialize data urbs */ @@ -667,8 +679,7 @@ static int data_ep_set_params(struct snd struct snd_urb_ctx *u = &ep->urb[i]; u->index = i; u->ep = ep; - u->packets = (i + 1) * total_packs / ep->nurbs - - i * total_packs / ep->nurbs; + u->packets = urb_packs; u->buffer_size = maxsize * u->packets; if (fmt->fmt_type == UAC_FORMAT_TYPE_II) @@ -745,6 +756,8 @@ out_of_memory: * @pcm_format: the audio fomat. * @channels: the number of audio channels. * @period_bytes: the number of bytes in one alsa period. + * @period_frames: the number of frames in one alsa period. + * @buffer_periods: the number of periods in one alsa buffer. * @rate: the frame rate. * @fmt: the USB audio format information * @sync_ep: the sync endpoint to use, if any @@ -757,6 +770,8 @@ int snd_usb_endpoint_set_params(struct s snd_pcm_format_t pcm_format, unsigned int channels, unsigned int period_bytes, + unsigned int period_frames, + unsigned int buffer_periods, unsigned int rate, struct audioformat *fmt, struct snd_usb_endpoint *sync_ep) @@ -790,7 +805,8 @@ int snd_usb_endpoint_set_params(struct s switch (ep->type) { case SND_USB_ENDPOINT_TYPE_DATA: err = data_ep_set_params(ep, pcm_format, channels, - period_bytes, fmt, sync_ep); + period_bytes, period_frames, + buffer_periods, fmt, sync_ep); break; case SND_USB_ENDPOINT_TYPE_SYNC: err = sync_ep_set_params(ep, fmt);

2 1

[alsa-devel] How to deal with snooping in firmware?
by Paul Menzel 26 Sep '13

26 Sep '13

Dear ALSA folks, with Ubuntu 13.04 and AMD Kabini boards, there are audio distortions, which are not present under Microsoft Windows [1][2]. Disabling snooping fixes this, which is in line with the information from an earlier discussion [3]. As it works under Microsoft Windows, is this indeed a driver problem and should be fixed there? Is it better to disabling snooping by default in the firmware and let the driver enable it? What do you propose how to fix this? Thanks, Paul [1] http://review.coreboot.org/#/c/3934/ [2] http://review.coreboot.org/#/c/3934/1/src/mainboard/amd/olivehill/get_bus_c… [3] http://lists.freedesktop.org/archives/pulseaudio-discuss/2012-May/013568.ht… [pulseaudio-discuss] [alsa-devel] Adjust volume may cause audio playback corruption

2 1

Re: [alsa-devel] Bug: Conexant CX20588, Kernel 3.11, Win8 changes GPIO settings
by Takashi Iwai 26 Sep '13

26 Sep '13

At Sun, 22 Sep 2013 21:33:59 +0200, Christopher wrote: > > Hi, > > I recently installed Windows 8 parallel to linux and there is no sound in > linux after rebooting. I guess the win8 conexant driver messes something up. > > A quick fix is to change this in HDAAnalyzer: > > GPIO: io=4, o=0, i=0, unsolicited=1, wake=0 > IO[0]: enable=1, dir=1, wake=0, sticky=0, data=0, unsol=0 > IO[1]: enable=0, dir=0, wake=0, sticky=0, data=0, unsol=0 > IO[2]: enable=0, dir=0, wake=0, sticky=0, data=0, unsol=0 > IO[3]: enable=0, dir=0, wake=0, sticky=0, data=0, unsol=0 > > Either set enable=0 or data=1 -> Internal Audio + Headphones work again. > This is persistent between reboots. Booting Win8 resets this again. > > I would be very happy to test a patch. So the GPIO should have been adjusted by the driver from the beginning. Could you give your hardware details? At best, give the output of alsa-info.sh. Run it with --no-upload option and attach the output. thanks, Takashi

1 0

[alsa-devel] [PATCH] ALSA: compress: Make sure we trigger STOP before closing the stream.
by Liam Girdwood 26 Sep '13

26 Sep '13

Currently we assume that userspace will shut down the compressed stream correctly. However, if userspcae dies (e.g. cplay & ctrl-C) we dont stop the stream before freeing it. This now checks that the stream is stopped before freeing. Signed-off-by: Liam Girdwood <liam.r.girdwood(a)linux.intel.com> --- sound/core/compress_offload.c | 12 ++++++++++++ 1 file changed, 12 insertions(+) diff --git a/sound/core/compress_offload.c b/sound/core/compress_offload.c index 99db892..a9ae4f3 100644 --- a/sound/core/compress_offload.c +++ b/sound/core/compress_offload.c @@ -139,6 +139,18 @@ static int snd_compr_open(struct inode *inode, struct file *f) static int snd_compr_free(struct inode *inode, struct file *f) { struct snd_compr_file *data = f->private_data; + struct snd_compr_runtime *runtime = data->stream.runtime; + + switch (runtime->state) { + case SNDRV_PCM_STATE_RUNNING: + case SNDRV_PCM_STATE_DRAINING: + case SNDRV_PCM_STATE_PAUSED: + data->stream.ops->trigger(&data->stream, SNDRV_PCM_TRIGGER_STOP); + break; + default: + break; + } + data->stream.ops->free(&data->stream); kfree(data->stream.runtime->buffer); kfree(data->stream.runtime); -- 1.8.1.2

4 10

Re: [alsa-devel] [PATCH 4/5] ALSA: Fix assignment of 0/1 to bool variables
by Takashi Iwai 26 Sep '13

26 Sep '13

At Sun, 22 Sep 2013 20:44:12 +0200, Peter Senna Tschudin wrote: > > Convert 0 to false and 1 to true when assigning values to bool > variables. Inspired by commit 3db1cd5c05f35fb43eb134df6f321de4e63141f2. > > The simplified semantic patch that find this problem is as > follows (http://coccinelle.lip6.fr/): > > @@ > bool b; > @@ > ( > -b = 0 > +b = false > | > -b = 1 > +b = true > ) > > Signed-off-by: Peter Senna Tschudin <peter.senna(a)gmail.com> Thanks, applied now. Takashi > > --- > sound/pci/azt3328.c | 14 +++++++------- > sound/usb/mixer.c | 4 ++-- > 2 files changed, 9 insertions(+), 9 deletions(-) > > diff -u -p a/sound/usb/mixer.c b/sound/usb/mixer.c > --- a/sound/usb/mixer.c > +++ b/sound/usb/mixer.c > @@ -1151,14 +1151,14 @@ static void check_no_speaker_on_headset( > const char *names_to_check[] = { > "Headset", "headset", "Headphone", "headphone", NULL}; > const char **s; > - bool found = 0; > + bool found = false; > > if (strcmp("Speaker", kctl->id.name)) > return; > > for (s = names_to_check; *s; s++) > if (strstr(card->shortname, *s)) { > - found = 1; > + found = true; > break; > } > > diff -u -p a/sound/pci/azt3328.c b/sound/pci/azt3328.c > --- a/sound/pci/azt3328.c > +++ b/sound/pci/azt3328.c > @@ -715,14 +715,14 @@ snd_azf3328_mixer_ac97_read(struct snd_a > const struct snd_azf3328 *chip = ac97->private_data; > unsigned short reg_azf = snd_azf3328_mixer_ac97_map_reg_idx(reg_ac97); > unsigned short reg_val = 0; > - bool unsupported = 0; > + bool unsupported = false; > > snd_azf3328_dbgmixer( > "snd_azf3328_mixer_ac97_read reg_ac97 %u\n", > reg_ac97 > ); > if (reg_azf & AZF_AC97_REG_UNSUPPORTED) > - unsupported = 1; > + unsupported = true; > else { > if (reg_azf & AZF_AC97_REG_REAL_IO_READ) > reg_val = snd_azf3328_mixer_inw(chip, > @@ -759,7 +759,7 @@ snd_azf3328_mixer_ac97_read(struct snd_a > reg_val = azf_emulated_ac97_vendor_id & 0xffff; > break; > default: > - unsupported = 1; > + unsupported = true; > break; > } > } > @@ -776,14 +776,14 @@ snd_azf3328_mixer_ac97_write(struct snd_ > { > const struct snd_azf3328 *chip = ac97->private_data; > unsigned short reg_azf = snd_azf3328_mixer_ac97_map_reg_idx(reg_ac97); > - bool unsupported = 0; > + bool unsupported = false; > > snd_azf3328_dbgmixer( > "snd_azf3328_mixer_ac97_write reg_ac97 %u val %u\n", > reg_ac97, val > ); > if (reg_azf & AZF_AC97_REG_UNSUPPORTED) > - unsupported = 1; > + unsupported = true; > else { > if (reg_azf & AZF_AC97_REG_REAL_IO_WRITE) > snd_azf3328_mixer_outw( > @@ -808,7 +808,7 @@ snd_azf3328_mixer_ac97_write(struct snd_ > */ > break; > default: > - unsupported = 1; > + unsupported = true; > break; > } > } > @@ -1559,7 +1559,7 @@ snd_azf3328_pcm_trigger(struct snd_pcm_s > struct snd_azf3328_codec_data *codec = runtime->private_data; > int result = 0; > u16 flags1; > - bool previously_muted = 0; > + bool previously_muted = false; > bool is_main_mixer_playback_codec = (AZF_CODEC_PLAYBACK == codec->type); > > snd_azf3328_dbgcalls("snd_azf3328_pcm_trigger cmd %d\n", cmd); >

1 0

[alsa-devel] [PATCH v2] ALSA : hda - not use assigned converters for all unused pins
by mengdong.lin＠intel.com 26 Sep '13

26 Sep '13

From: Mengdong Lin <mengdong.lin(a)intel.com> BIOS can mark a pin as "no physical connection" if the port is used by an integrated display which is not audio capable. And audio driver will overlook such pins. On Haswell, such a disconneted pin will keep muted and connected to the 1st converter by default. But if the 1st convertor is assigned to a connected pin for audio streaming. The muted disconnected pin can make the connected pin no sound output. So this patch avoids using assigned converters for all unused pins for Haswell, including the disconected pins. Signed-off-by: Mengdong Lin <mengdong.lin(a)intel.com> diff --git a/sound/pci/hda/patch_hdmi.c b/sound/pci/hda/patch_hdmi.c index 3d8cd044..7ea0245 100644 --- a/sound/pci/hda/patch_hdmi.c +++ b/sound/pci/hda/patch_hdmi.c @@ -1149,32 +1149,43 @@ static int hdmi_choose_cvt(struct hda_codec *codec, } static void haswell_config_cvts(struct hda_codec *codec, - int pin_id, int mux_id) + hda_nid_t pin_nid, int mux_idx) { struct hdmi_spec *spec = codec->spec; - struct hdmi_spec_per_pin *per_pin; - int pin_idx, mux_idx; - int curr; - int err; + hda_nid_t nid, end_nid; + int cvt_idx, curr; + struct hdmi_spec_per_cvt *per_cvt; - for (pin_idx = 0; pin_idx < spec->num_pins; pin_idx++) { - per_pin = get_pin(spec, pin_idx); + /* configure all pins, including "no physical connection" ones */ + end_nid = codec->start_nid + codec->num_nodes; + for (nid = codec->start_nid; nid < end_nid; nid++) { + unsigned int wid_caps = get_wcaps(codec, nid); + unsigned int wid_type = get_wcaps_type(wid_caps); - if (pin_idx == pin_id) + if (wid_type != AC_WID_PIN) continue; - curr = snd_hda_codec_read(codec, per_pin->pin_nid, 0, + if (nid == pin_nid) + continue; + + curr = snd_hda_codec_read(codec, nid, 0, AC_VERB_GET_CONNECT_SEL, 0); + if (curr != mux_idx) + continue; - /* Choose another unused converter */ - if (curr == mux_id) { - err = hdmi_choose_cvt(codec, pin_idx, NULL, &mux_idx); - if (err < 0) - return; - snd_printdd("HDMI: choose converter %d for pin %d\n", mux_idx, pin_idx); - snd_hda_codec_write_cache(codec, per_pin->pin_nid, 0, + /* choose an unassigned converter. The conveters in the + * connection list are in the same order as in the codec. + */ + for (cvt_idx = 0; cvt_idx < spec->num_cvts; cvt_idx++) { + per_cvt = get_cvt(spec, cvt_idx); + if (!per_cvt->assigned) { + snd_printdd("choose cvt %d for pin nid %d\n", + cvt_idx, nid); + snd_hda_codec_write_cache(codec, nid, 0, AC_VERB_SET_CONNECT_SEL, - mux_idx); + cvt_idx); + break; + } } } } @@ -1216,7 +1227,7 @@ static int hdmi_pcm_open(struct hda_pcm_stream *hinfo, /* configure unused pins to choose other converters */ if (is_haswell(codec)) - haswell_config_cvts(codec, pin_idx, mux_idx); + haswell_config_cvts(codec, per_pin->pin_nid, mux_idx); snd_hda_spdif_ctls_assign(codec, pin_idx, per_cvt->cvt_nid); -- 1.8.1.2

3 2

[alsa-devel] [PATCH] ASoC: sn95031: Convert to regmap
by Mark Brown 25 Sep '13

25 Sep '13

From: Mark Brown <broonie(a)linaro.org> This moves us towards being able to remove the duplicated register I/O functionality in ASoC. Signed-off-by: Mark Brown <broonie(a)linaro.org> --- sound/soc/codecs/sn95031.c | 35 ++++++++++++++++++++--------------- 1 file changed, 20 insertions(+), 15 deletions(-) diff --git a/sound/soc/codecs/sn95031.c b/sound/soc/codecs/sn95031.c index dba26e63..13045f2 100644 --- a/sound/soc/codecs/sn95031.c +++ b/sound/soc/codecs/sn95031.c @@ -164,30 +164,28 @@ static unsigned int sn95031_get_mic_bias(struct snd_soc_codec *codec) } /*end - adc helper functions */ -static inline unsigned int sn95031_read(struct snd_soc_codec *codec, - unsigned int reg) +static int sn95031_read(void *ctx, unsigned int reg, unsigned int *val) { u8 value = 0; int ret; ret = intel_scu_ipc_ioread8(reg, &value); - if (ret) - pr_err("read of %x failed, err %d\n", reg, ret); - return value; + if (ret == 0) + *val = value; + return ret; } -static inline int sn95031_write(struct snd_soc_codec *codec, - unsigned int reg, unsigned int value) +static int sn95031_write(void *ctx, unsigned int reg, unsigned int value) { - int ret; - - ret = intel_scu_ipc_iowrite8(reg, value); - if (ret) - pr_err("write of %x failed, err %d\n", reg, ret); - return ret; + return intel_scu_ipc_iowrite8(reg, value); } +static const struct regmap_config sn95031_regmap = { + .reg_read = sn95031_read, + .reg_write = sn95031_write, +}; + static int sn95031_set_vaud_bias(struct snd_soc_codec *codec, enum snd_soc_bias_level level) { @@ -827,6 +825,8 @@ static int sn95031_codec_probe(struct snd_soc_codec *codec) { pr_debug("codec_probe called\n"); + snd_soc_codec_set_cache_io(codec, 0, 0, SND_SOC_REGMAP); + /* PCM interface config * This sets the pcm rx slot conguration to max 6 slots * for max 4 dais (2 stereo and 2 mono) @@ -886,8 +886,6 @@ static int sn95031_codec_remove(struct snd_soc_codec *codec) static struct snd_soc_codec_driver sn95031_codec = { .probe = sn95031_codec_probe, .remove = sn95031_codec_remove, - .read = sn95031_read, - .write = sn95031_write, .set_bias_level = sn95031_set_vaud_bias, .idle_bias_off = true, .dapm_widgets = sn95031_dapm_widgets, @@ -898,7 +896,14 @@ static struct snd_soc_codec_driver sn95031_codec = { static int sn95031_device_probe(struct platform_device *pdev) { + struct regmap *regmap; + pr_debug("codec device probe called for %s\n", dev_name(&pdev->dev)); + + regmap = devm_regmap_init(&pdev->dev, NULL, NULL, &sn95031_regmap); + if (IS_ERR(regmap)) + return PTR_ERR(regmap); + return snd_soc_register_codec(&pdev->dev, &sn95031_codec, sn95031_dais, ARRAY_SIZE(sn95031_dais)); } -- 1.8.4.rc3

1 0

Re: [alsa-devel] IMX WM8740 I2S support
by Fabio Estevam 25 Sep '13

25 Sep '13

[Please always Cc the mailing list] On Wed, Sep 25, 2013 at 2:05 PM, stéphane cerveau <scerveau(a)gmail.com> wrote: > I'm using an old kernel from a BSP provided by a retailer. This list is about mainline kernel. If you would like to use mainline kernel we could help you here. If you decide to stay with the vendor BSP, then this is not the appropriate place for support.

2 1

[alsa-devel] IMX WM8740 I2S support
by stéphane cerveau 25 Sep '13

25 Sep '13

Dear all, I'm currently working on supporting WM8740 device software configured with imx6q system. For now my driver seems to be configured correctly. The playback is launched but i get: ALSA sound/core/pcm_lib.c:1822 playback write error (DMA or IRQ trouble? write error: Input/output error The imx ssi is not sending me any data on I2S bus( nothing can be seen on the oscillo). I would like to know how do I have to configure my master clock in order to launch the I2S. My master clock is for now 24MHz. Sorry to be confused a bit but i'm quite newbie with this device. Best regards. Stéphane

2 8

[alsa-devel] [PATCH] ASoC: ab8500: Explicitly set I/O up
by Mark Brown 25 Sep '13

25 Sep '13

From: Mark Brown <broonie(a)linaro.org> We do some I/O in probe so we need to ensure the I/O operations are fully set up then. Reported-by: Olof Johansson <olof(a)lixom.net> Signed-off-by: Mark Brown <broonie(a)linaro.org> --- sound/soc/codecs/ab8500-codec.c | 2 ++ 1 file changed, 2 insertions(+) diff --git a/sound/soc/codecs/ab8500-codec.c b/sound/soc/codecs/ab8500-codec.c index ca500fd..a0394a8 100644 --- a/sound/soc/codecs/ab8500-codec.c +++ b/sound/soc/codecs/ab8500-codec.c @@ -2473,6 +2473,8 @@ static int ab8500_codec_probe(struct snd_soc_codec *codec) dev_dbg(dev, "%s: Enter.\n", __func__); + snd_soc_codec_set_cache_io(codec, 0, 0, SND_SOC_REGMAP); + /* Setup AB8500 according to board-settings */ pdata = dev_get_platdata(dev->parent); -- 1.8.4.rc3

3 7

[alsa-devel] [PATCH] ASoC: imx-sgtl5000: do not use devres on a foreign device
by Philipp Zabel 25 Sep '13

25 Sep '13

Calling devm_clk_get with any device pointer other than our own confuses devres. Use clk_get instead. This avoids hitting the following warning in the imx-sgtl5000 error path: imx-sgtl5000 sound.12: snd_soc_register_card failed (-517) platform sound.12: Driver imx-sgtl5000 requests probe deferral ------------[ cut here ]------------ WARNING: CPU: 0 PID: 75 at drivers/base/dd.c:272 driver_probe_device+0x194/0x218() Modules linked in: snd_soc_sgtl5000(+) snd_soc_imx_sgtl5000 coda snd_soc_imx_audmux imx_sdma snd_soc_fsl_spdif snd_soc_fsl_ssi CPU: 0 PID: 75 Comm: udevd Not tainted 3.11.0-rc6+ #4682 Backtrace: [<80010bc4>] (dump_backtrace+0x0/0x10c) from [<80010d60>] (show_stack+0x18/0x1c) r6:00000110 r5:00000009 r4:00000000 r3:00000000 [<80010d48>] (show_stack+0x0/0x1c) from [<804f0764>] (dump_stack+0x20/0x28) [<804f0744>] (dump_stack+0x0/0x28) from [<8001a4a4>] (warn_slowpath_common+0x6c/0x8c) [<8001a438>] (warn_slowpath_common+0x0/0x8c) from [<8001a4e8>] (warn_slowpath_null+0x24/0x2c) r8:7f032000 r7:7f02f93c r6:cf8eaa54 r5:cf8eaa20 r4:80728a0c [<8001a4c4>] (warn_slowpath_null+0x0/0x2c) from [<80286bdc>] (driver_probe_device+0x194/0x218) [<80286a48>] (driver_probe_device+0x0/0x218) from [<80286cf4>] (__driver_attach+0x94/0x98) r7:00000000 r6:cf8eaa54 r5:7f02f93c r4:cf8eaa20 [<80286c60>] (__driver_attach+0x0/0x98) from [<802851c8>] (bus_for_each_dev+0x5c/0x90) r6:80286c60 r5:7f02f93c r4:00000000 r3:cf8ef03c [<8028516c>] (bus_for_each_dev+0x0/0x90) from [<80286654>] (driver_attach+0x24/0x28) r6:806d0424 r5:cf16a580 r4:7f02f93c [<80286630>] (driver_attach+0x0/0x28) from [<802861e4>] (bus_add_driver+0xdc/0x234) [<80286108>] (bus_add_driver+0x0/0x234) from [<802871d4>] (driver_register+0x80/0x154) r8:7f032000 r7:00000001 r6:7f02fa68 r5:7f02fa74 r4:7f02f93c [<80287154>] (driver_register+0x0/0x154) from [<8033c278>] (i2c_register_driver+0x34/0xbc) [<8033c244>] (i2c_register_driver+0x0/0xbc) from [<7f032018>] (sgtl5000_i2c_driver_init+0x18/0x24 [snd_soc_sgtl5000]) r5:7f02fa74 r4:cfb7ff48 [<7f032000>] (sgtl5000_i2c_driver_init+0x0/0x24 [snd_soc_sgtl5000]) from [<80008738>] (do_one_initcall+0xf4/0x150) [<80008644>] (do_one_initcall+0x0/0x150) from [<80053f64>] (load_module+0x174c/0x1db4) [<80052818>] (load_module+0x0/0x1db4) from [<800546ac>] (SyS_init_module+0xe0/0xf4) [<800545cc>] (SyS_init_module+0x0/0xf4) from [<8000e540>] (ret_fast_syscall+0x0/0x30) r6:00005b22 r5:00afed68 r4:00000000 ---[ end trace b24c5c3bb145dbdd ]--- Signed-off-by: Philipp Zabel <p.zabel(a)pengutronix.de> --- sound/soc/fsl/imx-sgtl5000.c | 5 ++++- 1 file changed, 4 insertions(+), 1 deletion(-) diff --git a/sound/soc/fsl/imx-sgtl5000.c b/sound/soc/fsl/imx-sgtl5000.c index 46c5b4f..52df7d5 100644 --- a/sound/soc/fsl/imx-sgtl5000.c +++ b/sound/soc/fsl/imx-sgtl5000.c @@ -128,7 +128,7 @@ static int imx_sgtl5000_probe(struct platform_device *pdev) goto fail; } - data->codec_clk = devm_clk_get(&codec_dev->dev, NULL); + data->codec_clk = clk_get(&codec_dev->dev, NULL); if (IS_ERR(data->codec_clk)) { ret = PTR_ERR(data->codec_clk); goto fail; @@ -172,6 +172,8 @@ static int imx_sgtl5000_probe(struct platform_device *pdev) return 0; fail: + if (!IS_ERR(data->codec_clk)) + clk_put(data->codec_clk); if (ssi_np) of_node_put(ssi_np); if (codec_np) @@ -185,6 +187,7 @@ static int imx_sgtl5000_remove(struct platform_device *pdev) struct imx_sgtl5000_data *data = platform_get_drvdata(pdev); snd_soc_unregister_card(&data->card); + clk_put(data->codec_clk); return 0; } -- 1.8.4.rc3

3 5

[alsa-devel] [PATCH 1/3] mfd: mc13xxx: Don't require lock for simple register I/O
by Mark Brown 25 Sep '13

25 Sep '13

From: Mark Brown <broonie(a)linaro.org> Since the conversion to regmap there has been no need for device level locking for I/O as regmap provides locking so remove the locks. Signed-off-by: Mark Brown <broonie(a)linaro.org> --- drivers/mfd/mc13xxx-core.c | 5 ----- 1 file changed, 5 deletions(-) diff --git a/drivers/mfd/mc13xxx-core.c b/drivers/mfd/mc13xxx-core.c index 2a9b100..dbbf8ee 100644 --- a/drivers/mfd/mc13xxx-core.c +++ b/drivers/mfd/mc13xxx-core.c @@ -158,8 +158,6 @@ int mc13xxx_reg_read(struct mc13xxx *mc13xxx, unsigned int offset, u32 *val) { int ret; - BUG_ON(!mutex_is_locked(&mc13xxx->lock)); - if (offset > MC13XXX_NUMREGS) return -EINVAL; @@ -172,8 +170,6 @@ EXPORT_SYMBOL(mc13xxx_reg_read); int mc13xxx_reg_write(struct mc13xxx *mc13xxx, unsigned int offset, u32 val) { - BUG_ON(!mutex_is_locked(&mc13xxx->lock)); - dev_vdbg(mc13xxx->dev, "[0x%02x] <- 0x%06x\n", offset, val); if (offset > MC13XXX_NUMREGS || val > 0xffffff) @@ -186,7 +182,6 @@ EXPORT_SYMBOL(mc13xxx_reg_write); int mc13xxx_reg_rmw(struct mc13xxx *mc13xxx, unsigned int offset, u32 mask, u32 val) { - BUG_ON(!mutex_is_locked(&mc13xxx->lock)); BUG_ON(val & ~mask); dev_vdbg(mc13xxx->dev, "[0x%02x] <- 0x%06x (mask: 0x%06x)\n", offset, val, mask); -- 1.8.4.rc3

2 11

[alsa-devel] [PATCH 1/3] ASoC: tlv320aic26: Convert to table based control init
by Mark Brown 25 Sep '13

25 Sep '13

From: Mark Brown <broonie(a)linaro.org> Signed-off-by: Mark Brown <broonie(a)linaro.org> --- sound/soc/codecs/tlv320aic26.c | 10 +++------- 1 file changed, 3 insertions(+), 7 deletions(-) diff --git a/sound/soc/codecs/tlv320aic26.c b/sound/soc/codecs/tlv320aic26.c index 7b8f3d9..32c6b07 100644 --- a/sound/soc/codecs/tlv320aic26.c +++ b/sound/soc/codecs/tlv320aic26.c @@ -377,7 +377,7 @@ static DEVICE_ATTR(keyclick, 0644, aic26_keyclick_show, aic26_keyclick_set); static int aic26_probe(struct snd_soc_codec *codec) { struct aic26 *aic26 = dev_get_drvdata(codec->dev); - int ret, err, i, reg; + int ret, i, reg; aic26->codec = codec; @@ -403,12 +403,6 @@ static int aic26_probe(struct snd_soc_codec *codec) if (ret) dev_info(codec->dev, "error creating sysfs files\n"); - /* register controls */ - dev_dbg(codec->dev, "Registering controls\n"); - err = snd_soc_add_codec_controls(codec, aic26_snd_controls, - ARRAY_SIZE(aic26_snd_controls)); - WARN_ON(err < 0); - return 0; } @@ -418,6 +412,8 @@ static struct snd_soc_codec_driver aic26_soc_codec_dev = { .write = aic26_reg_write, .reg_cache_size = AIC26_NUM_REGS, .reg_word_size = sizeof(u16), + .controls = aic26_snd_controls, + .num_controls = ARRAY_SIZE(aic26_snd_controls), .dapm_widgets = tlv320aic26_dapm_widgets, .num_dapm_widgets = ARRAY_SIZE(tlv320aic26_dapm_widgets), .dapm_routes = tlv320aic26_dapm_routes, -- 1.8.4.rc3

1 2

[alsa-devel] [PATCH 1/2] ASoC: tlv320aic32x4: Move GPIO acquisition to I2C probe
by Mark Brown 25 Sep '13

25 Sep '13

From: Mark Brown <broonie(a)linaro.org> This is more idiomatic and interacts better with deferred probe. Signed-off-by: Mark Brown <broonie(a)linaro.org> --- sound/soc/codecs/tlv320aic32x4.c | 12 +++++++----- 1 file changed, 7 insertions(+), 5 deletions(-) diff --git a/sound/soc/codecs/tlv320aic32x4.c b/sound/soc/codecs/tlv320aic32x4.c index 2ed57d4..cf70bf8 100644 --- a/sound/soc/codecs/tlv320aic32x4.c +++ b/sound/soc/codecs/tlv320aic32x4.c @@ -617,16 +617,11 @@ static int aic32x4_probe(struct snd_soc_codec *codec) { struct aic32x4_priv *aic32x4 = snd_soc_codec_get_drvdata(codec); u32 tmp_reg; - int ret; codec->hw_write = (hw_write_t) i2c_master_send; codec->control_data = aic32x4->control_data; if (aic32x4->rstn_gpio >= 0) { - ret = devm_gpio_request_one(codec->dev, aic32x4->rstn_gpio, - GPIOF_OUT_INIT_LOW, "tlv320aic32x4 rstn"); - if (ret != 0) - return ret; ndelay(10); gpio_set_value(aic32x4->rstn_gpio, 1); } @@ -735,6 +730,13 @@ static int aic32x4_i2c_probe(struct i2c_client *i2c, aic32x4->rstn_gpio = -1; } + if (aic32x4->rstn_gpio >= 0) { + ret = devm_gpio_request_one(&i2c->dev, aic32x4->rstn_gpio, + GPIOF_OUT_INIT_LOW, "tlv320aic32x4 rstn"); + if (ret != 0) + return ret; + } + ret = snd_soc_register_codec(&i2c->dev, &soc_codec_dev_aic32x4, &aic32x4_dai, 1); return ret; -- 1.8.4.rc3

1 1

[alsa-devel] [TINYCOMPRESS][PATCH 1/1] compress: Must check for POLLERR before POLLOUT/POLLIN
by Richard Fitzgerald 25 Sep '13

25 Sep '13

In the case of error the ALSA compressed driver sets revents as (POLLOUT | POLLWRNORM | POLLERR) or (POLLIN | POLLWRNORM | POLLERR). So we can't assume that POLLOUT or POLLIN indicate success, we must check for POLLERR first. Signed-off-by: Richard Fitzgerald <rf(a)opensource.wolfsonmicro.com> --- compress.c | 18 +++++++++--------- 1 files changed, 9 insertions(+), 9 deletions(-) diff --git a/compress.c b/compress.c index 0e71c28..5cd0966 100644 --- a/compress.c +++ b/compress.c @@ -378,6 +378,9 @@ int compress_write(struct compress *compress, const void *buf, unsigned int size return total; ret = poll(&fds, 1, compress->max_poll_wait_ms); + if (fds.revents & POLLERR) { + return oops(compress, EIO, "poll returned error!"); + } /* A pause will cause -EBADFD or zero. * This is not an error, just stop writing */ if ((ret == 0) || (ret == -EBADFD)) @@ -387,9 +390,6 @@ int compress_write(struct compress *compress, const void *buf, unsigned int size if (fds.revents & POLLOUT) { continue; } - if (fds.revents & POLLERR) { - return oops(compress, EIO, "poll returned error!"); - } } /* write avail bytes */ if (size > avail.avail) @@ -438,6 +438,9 @@ int compress_read(struct compress *compress, void *buf, unsigned int size) return total; ret = poll(&fds, 1, compress->max_poll_wait_ms); + if (fds.revents & POLLERR) { + return oops(compress, EIO, "poll returned error!"); + } /* A pause will cause -EBADFD or zero. * This is not an error, just stop reading */ if ((ret == 0) || (ret == -EBADFD)) @@ -447,9 +450,6 @@ int compress_read(struct compress *compress, void *buf, unsigned int size) if (fds.revents & POLLIN) { continue; } - if (fds.revents & POLLERR) { - return oops(compress, EIO, "poll returned error!"); - } } /* read avail bytes */ if (size > avail.avail) @@ -616,15 +616,15 @@ int compress_wait(struct compress *compress, int timeout_ms) fds.events = POLLOUT | POLLIN; ret = poll(&fds, 1, timeout_ms); + if (fds.revents & POLLERR) { + return oops(compress, EIO, "poll returned error!"); + } /* A pause will cause -EBADFD or zero. */ if ((ret < 0) && (ret != -EBADFD)) return oops(compress, errno, "poll error"); if (fds.revents & (POLLOUT | POLLIN)) { return 0; } - if (fds.revents & POLLERR) { - return oops(compress, EIO, "poll returned error!"); - } return ret; } -- 1.7.2.5

2 1

[alsa-devel] [PATCH] ASoC: kirkwood: simplify clock handling
by Uwe Kleine-König 24 Sep '13

24 Sep '13

There is no need to not use extclk if it is identical to the main clk. The main motivation for this patch is dropping devm_clk_put which is used in a wrong way by all other users. While at it also extend the comments. Signed-off-by: Uwe Kleine-König <u.kleine-koenig(a)pengutronix.de> --- Hello, there might be some further optimisations possible. I only note them here because I don't have the hardware to test: - only enable extclk if it a clock rate used that makes use of the external clock. Not sure if that works; hardware docs reading necessary. - only provide extclk if it's != the internal clock. - The code uses: priv->clk = devm_clk_get(&pdev->dev, np ? "internal" : NULL); i.e. provides a con_id in the dt-case. I think that using NULL unconditionally should also work, i.e. return the first clk associated to the device. OTOH the current code might make things clearer because it's more explicit. Best regards Uwe --- sound/soc/kirkwood/kirkwood-i2s.c | 24 +++++++++++++++--------- 1 file changed, 15 insertions(+), 9 deletions(-) diff --git a/sound/soc/kirkwood/kirkwood-i2s.c b/sound/soc/kirkwood/kirkwood-i2s.c index 0f3d73d..8224f93 100644 --- a/sound/soc/kirkwood/kirkwood-i2s.c +++ b/sound/soc/kirkwood/kirkwood-i2s.c @@ -104,16 +104,25 @@ static void kirkwood_set_rate(struct snd_soc_dai *dai, uint32_t clks_ctrl; if (rate == 44100 || rate == 48000 || rate == 96000) { - /* use internal dco for the supported rates - * defined in kirkwood_i2s_dai */ + /* + * use internal dco for the supported rates + * defined in kirkwood_i2s_dai + * Note: For these specific rates the dco is also used if + * kirkwood_i2s_dai_extclk is in use. + */ dev_dbg(dai->dev, "%s: dco set rate = %lu\n", __func__, rate); kirkwood_set_dco(priv->io, rate); clks_ctrl = KIRKWOOD_MCLK_SOURCE_DCO; } else { - /* use the external clock for the other rates - * defined in kirkwood_i2s_dai_extclk */ + /* + * use the external clock for the other rates + * defined in kirkwood_i2s_dai_extclk + * This case isn't reached if kirkwood_i2s_dai is in use (and so + * extclk isn't valid) because it only supports the three rates + * above. + */ dev_dbg(dai->dev, "%s: extclk set rate = %lu -> %lu\n", __func__, rate, 256 * rate); clk_set_rate(priv->extclk, 256 * rate); @@ -497,12 +506,9 @@ static int kirkwood_i2s_dev_probe(struct platform_device *pdev) priv->extclk = devm_clk_get(&pdev->dev, "extclk"); if (!IS_ERR(priv->extclk)) { - if (priv->extclk == priv->clk) { - devm_clk_put(&pdev->dev, priv->extclk); - priv->extclk = ERR_PTR(-EINVAL); - } else { + clk_prepare_enable(priv->extclk); + if (priv->extclk != priv->clk) { dev_info(&pdev->dev, "found external clock\n"); - clk_prepare_enable(priv->extclk); soc_dai = &kirkwood_i2s_dai_extclk; } } -- 1.8.4.rc3

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Re: [alsa-devel] [PATCH v2 4/4] ASoC: rsnd: use regmap instead of original register mapping method
by Mark Brown 24 Sep '13

24 Sep '13

On Tue, Sep 17, 2013 at 10:23:58PM +0900, Kuninori Morimoto wrote: > For example, SSI device have many ports (0-8), > but ADG is only 1 device. > On this driver, all devices are using rsnd_write/read() to > access register. OK, this sounds like the module should have this information - ie check something in the module structure rather than asking regmap.

2 4

[alsa-devel] [PATCH 1/3] ASoC: tlv320aic23: Remove #defines for I2C
by Mark Brown 24 Sep '13

24 Sep '13

From: Mark Brown <broonie(a)linaro.org> The only control interface supported by this driver is I2C so there is no need for conditional compilation around the control interface. Signed-off-by: Mark Brown <broonie(a)linaro.org> --- sound/soc/codecs/tlv320aic23.c | 25 +------------------------ 1 file changed, 1 insertion(+), 24 deletions(-) diff --git a/sound/soc/codecs/tlv320aic23.c b/sound/soc/codecs/tlv320aic23.c index 31762eb..3299459 100644 --- a/sound/soc/codecs/tlv320aic23.c +++ b/sound/soc/codecs/tlv320aic23.c @@ -613,7 +613,6 @@ static struct snd_soc_codec_driver soc_codec_dev_tlv320aic23 = { .num_dapm_routes = ARRAY_SIZE(tlv320aic23_intercon), }; -#if defined(CONFIG_I2C) || defined(CONFIG_I2C_MODULE) /* * If the i2c layer weren't so broken, we could pass this kind of data * around @@ -660,29 +659,7 @@ static struct i2c_driver tlv320aic23_i2c_driver = { .id_table = tlv320aic23_id, }; -#endif - -static int __init tlv320aic23_modinit(void) -{ - int ret; -#if defined(CONFIG_I2C) || defined(CONFIG_I2C_MODULE) - ret = i2c_add_driver(&tlv320aic23_i2c_driver); - if (ret != 0) { - printk(KERN_ERR "Failed to register TLV320AIC23 I2C driver: %d\n", - ret); - } -#endif - return ret; -} -module_init(tlv320aic23_modinit); - -static void __exit tlv320aic23_exit(void) -{ -#if defined(CONFIG_I2C) || defined(CONFIG_I2C_MODULE) - i2c_del_driver(&tlv320aic23_i2c_driver); -#endif -} -module_exit(tlv320aic23_exit); +module_i2c_driver(tlv320aic23_i2c_driver); MODULE_DESCRIPTION("ASoC TLV320AIC23 codec driver"); MODULE_AUTHOR("Arun KS <arunks(a)mistralsolutions.com>"); -- 1.8.4.rc3

1 2

[alsa-devel] [PATCH v3 1/2] ASoC: kirkwood: clk: probe defer when clock not yet ready
by Jean-Francois Moine 24 Sep '13

24 Sep '13

At probe time, a clock device may not be ready when some other device wants to use it. This patch lets the functions clk_get/devm_clk_get return a probe defer when the clock is defined in the DT but not yet available. It also fixes an erroneous call to clk_get_sys() when __clk_get() fails. Signed-off-by: Jean-Francois Moine <moinejf(a)free.fr> --- v2: fix __clk_get() failure (from Russell King) Note: the associated patch v2 2/2 is unchanged and not resent --- drivers/clk/clk.c | 2 +- drivers/clk/clkdev.c | 7 ++++++- 2 files changed, 7 insertions(+), 2 deletions(-) diff --git a/drivers/clk/clk.c b/drivers/clk/clk.c index a004769..582abc8 100644 --- a/drivers/clk/clk.c +++ b/drivers/clk/clk.c @@ -2181,7 +2181,7 @@ EXPORT_SYMBOL_GPL(of_clk_del_provider); struct clk *of_clk_get_from_provider(struct of_phandle_args *clkspec) { struct of_clk_provider *provider; - struct clk *clk = ERR_PTR(-ENOENT); + struct clk *clk = ERR_PTR(-EPROBE_DEFER); /* Check if we have such a provider in our array */ mutex_lock(&of_clk_lock); diff --git a/drivers/clk/clkdev.c b/drivers/clk/clkdev.c index 442a313..1f402b9 100644 --- a/drivers/clk/clkdev.c +++ b/drivers/clk/clkdev.c @@ -157,7 +157,12 @@ struct clk *clk_get(struct device *dev, const char *con_id) if (dev) { clk = of_clk_get_by_name(dev->of_node, con_id); - if (!IS_ERR(clk) && __clk_get(clk)) + if (!IS_ERR(clk)) { + if (!__clk_get(clk)) + clk = ERR_PTR(-ENOENT); + return clk; + } + if (PTR_ERR(clk) == -EPROBE_DEFER) return clk; } -- Ken ar c'hentañ | ** Breizh ha Linux atav! ** Jef | http://moinejf.free.fr/

1 0

[alsa-devel] "ASoC: ab8500: Convert register I/O to regmap" breaks snowball
by Olof Johansson 24 Sep '13

24 Sep '13

Mark, Lee, [I wasn't subscribed to alsa-devel when the patch was posted, so I'm sending this as a standalone email instead of as a reply to that] linux-next 20130924 fails to boot on snowball (u8500_defconfig, non-DT boot): Unable to handle kernel NULL pointer dereference at virtual address 00000000 pgd = c0004000 [00000000] *pgd=00000000 Internal error: Oops: 80000005 [#1] PREEMPT SMP ARM Modules linked in: CPU: 0 PID: 1 Comm: swapper/0 Not tainted 3.12.0-rc1-00003-gff795d6 #135 task: ef060000 ti: ef064000 task.ti: ef064000 PC is at 0x0 LR is at snd_soc_update_bits+0x4c/0x88 pc : [<00000000>] lr : [<c02d4f98>] psr: 60000013 sp : ef065d48 ip : 00000020 fp : ef0d054c r10: 00000000 r9 : ef3c0810 r8 : 00000000 r7 : 00000014 r6 : 00000000 r5 : 00000007 r4 : ef36ea00 r3 : 00000000 r2 : 00000007 r1 : 00000014 r0 : ef36ea00 Flags: nZCv IRQs on FIQs on Mode SVC_32 ISA ARM Segment kernel Control: 10c5787d Table: 0000404a DAC: 00000015 Process swapper/0 (pid: 1, stack limit = 0xef064240) Stack: (0xef065d48 to 0xef066000) 5d40: ef153410 c0550bb8 ef153410 ef36ea00 c053dc64 ef36ea00 5d60: c053dc64 c03d5e0c ef36eaac c02e4144 00000000 02000000 ef36ea00 00000000 5d80: c055a688 ef17bc10 ef0a7b00 c02e4358 00000000 00000000 00000000 ef36ea00 5da0: c055a5dc c055a688 c055daa4 c055aa0c ef36eaac c02d74fc c03d592c c055a678 5dc0: ee8115c8 c055daa4 ef0d00c0 00000000 00000000 c055a668 c03d5bf4 c02d8258 5de0: c055dab4 00000000 00000004 eec2d990 c055db7c ef0d0780 c055dbac c055db9c 5e00: c055a6a8 ef36ea30 ef36ea00 ef0d087c ef0d084c ef36ea30 c04a98f4 c055dac4 5e20: c021ce54 c055daa4 000005b8 c053f7f0 c055da64 c051dde8 00000069 ef064028 5e40: 00000000 c02d8ef0 00000000 c055da50 00000000 c02e684c c05924fc c053f7f0 5e60: 00000000 c055da64 c051dde8 c021bc88 c021bc70 c021aa94 00000000 c053f7f0 5e80: c055da64 c053f824 00000000 c021ac80 00000000 c055da64 c021abf4 c0218f6c 5ea0: ef03265c ef1b6734 c055da64 ef192500 c054fea8 c021a220 c0475770 c055da64 5ec0: 00000006 c055da64 00000006 c0563c80 c0563c80 c021b25c 00000000 c052c5f4 5ee0: 00000006 c0008798 ef010900 c04787e4 ef09c100 c03ac5e0 00000000 00000001 5f00: 000014dc c0540b9c 60000013 00000001 00000003 00000000 c0da19bd c03bf6b0 5f20: 00000069 c0036aa4 c04de34c 00000006 c0da19c8 00000006 00000000 c052c5f4 5f40: 00000006 c0563c80 c0563c80 c05014dc 00000069 c0523fec c0523fe0 c0501bc4 5f60: 00000006 00000006 c05014dc 00000000 88000000 00000000 00000000 00000000 5f80: c03a14ec 00000000 00000000 00000000 00000000 00000000 00000000 c03a14f4 5fa0: 00000000 00000000 c03a14ec c000e438 00000000 00000000 00000000 00000000 5fc0: 00000000 00000000 00000000 00000000 00000000 00000000 00000000 00000000 5fe0: 00000000 00000000 00000000 00000000 00000013 00000000 00000000 20000000 [<c02d4f98>] (snd_soc_update_bits+0x4c/0x88) from [<c02e4144>] (ab8500_audio_setup_mics+0xc8/0x118) [<c02e4144>] (ab8500_audio_setup_mics+0xc8/0x118) from [<c02e4358>] (ab8500_codec_probe+0x180/0x374) [<c02e4358>] (ab8500_codec_probe+0x180/0x374) from [<c02d74fc>] (soc_probe_codec+0x178/0x2f4) [<c02d74fc>] (soc_probe_codec+0x178/0x2f4) from [<c02d8258>] (snd_soc_instantiate_card+0x794/0x11c0) [<c02d8258>] (snd_soc_instantiate_card+0x794/0x11c0) from [<c02d8ef0>] (snd_soc_register_card+0x26c/0x324) [<c02d8ef0>] (snd_soc_register_card+0x26c/0x324) from [<c02e684c>] (mop500_probe+0xd0/0x118) [<c02e684c>] (mop500_probe+0xd0/0x118) from [<c021bc88>] (platform_drv_probe+0x18/0x1c) [<c021bc88>] (platform_drv_probe+0x18/0x1c) from [<c021aa94>] (driver_probe_device+0x10c/0x228) ... I bisected it down to: commit ff795d614bfa62a3c6fc0bcb75cb8842e5a87892 Author: Mark Brown <broonie(a)linaro.org> Date: Fri Sep 20 10:38:16 2013 +0100 ASoC: ab8500: Convert register I/O to regmap As part of a general push to eliminate the duplicated register I/O support in ASoC convert ab8500 to use regmap. Signed-off-by: Mark Brown <broonie(a)linaro.org> Acked-by: Lee Jones <lee.jones(a)linaro.org> ... but I didn't look closer at what was actually broken. -Olof

2 1

[alsa-devel] [PATCH 1/3] ASoC: max98095: Remove custom hw_write() implementation
by Mark Brown 24 Sep '13

24 Sep '13

From: Mark Brown <broonie(a)linaro.org> The registers that are being kept uncached are marked as volatile anyway so the call has no practical impact. Signed-off-by: Mark Brown <broonie(a)linaro.org> --- sound/soc/codecs/max98095.c | 25 ++++--------------------- 1 file changed, 4 insertions(+), 21 deletions(-) diff --git a/sound/soc/codecs/max98095.c b/sound/soc/codecs/max98095.c index 8dbcacd..29c266b 100644 --- a/sound/soc/codecs/max98095.c +++ b/sound/soc/codecs/max98095.c @@ -612,23 +612,6 @@ static int max98095_volatile(struct snd_soc_codec *codec, unsigned int reg) } /* - * Filter coefficients are in a separate register segment - * and they share the address space of the normal registers. - * The coefficient registers do not need or share the cache. - */ -static int max98095_hw_write(struct snd_soc_codec *codec, unsigned int reg, - unsigned int value) -{ - int ret; - - codec->cache_bypass = 1; - ret = snd_soc_write(codec, reg, value); - codec->cache_bypass = 0; - - return ret ? -EIO : 0; -} - -/* * Load equalizer DSP coefficient configurations registers */ static void m98095_eq_band(struct snd_soc_codec *codec, unsigned int dai, @@ -648,8 +631,8 @@ static void m98095_eq_band(struct snd_soc_codec *codec, unsigned int dai, /* Step through the registers and coefs */ for (i = 0; i < M98095_COEFS_PER_BAND; i++) { - max98095_hw_write(codec, eq_reg++, M98095_BYTE1(coefs[i])); - max98095_hw_write(codec, eq_reg++, M98095_BYTE0(coefs[i])); + snd_soc_write(codec, eq_reg++, M98095_BYTE1(coefs[i])); + snd_soc_write(codec, eq_reg++, M98095_BYTE0(coefs[i])); } } @@ -673,8 +656,8 @@ static void m98095_biquad_band(struct snd_soc_codec *codec, unsigned int dai, /* Step through the registers and coefs */ for (i = 0; i < M98095_COEFS_PER_BAND; i++) { - max98095_hw_write(codec, bq_reg++, M98095_BYTE1(coefs[i])); - max98095_hw_write(codec, bq_reg++, M98095_BYTE0(coefs[i])); + snd_soc_write(codec, bq_reg++, M98095_BYTE1(coefs[i])); + snd_soc_write(codec, bq_reg++, M98095_BYTE0(coefs[i])); } } -- 1.8.4.rc3

2 4

[alsa-devel] [PATCH 1/5] ASoC: Docs: Update codec documentation
by Liam Girdwood 24 Sep '13

24 Sep '13

Update the codec class driver documentation and bring it up to date with the current code base. This includes API changes, regmap and multi component. Signed-off-by: Liam Girdwood <liam.r.girdwood(a)linux.intel.com> --- Documentation/sound/alsa/soc/codec.txt | 46 +++++++++++----------------------- 1 file changed, 15 insertions(+), 31 deletions(-) diff --git a/Documentation/sound/alsa/soc/codec.txt b/Documentation/sound/alsa/soc/codec.txt index bce23a4..db5f9c9 100644 --- a/Documentation/sound/alsa/soc/codec.txt +++ b/Documentation/sound/alsa/soc/codec.txt @@ -1,22 +1,23 @@ -ASoC Codec Driver -================= +ASoC Codec Class Driver +======================= -The codec driver is generic and hardware independent code that configures the -codec to provide audio capture and playback. It should contain no code that is -specific to the target platform or machine. All platform and machine specific -code should be added to the platform and machine drivers respectively. +The codec class driver is generic and hardware independent code that configures +the codec, FM, MODEM, BT or external DSP to provide audio capture and playback. +It should contain no code that is specific to the target platform or machine. +All platform and machine specific code should be added to the platform and +machine drivers respectively. -Each codec driver *must* provide the following features:- +Each codec class driver *must* provide the following features:- 1) Codec DAI and PCM configuration - 2) Codec control IO - using I2C, 3 Wire(SPI) or both APIs + 2) Codec control IO - using RegMap API 3) Mixers and audio controls 4) Codec audio operations + 5) DAPM description. + 6) DAPM event handler. Optionally, codec drivers can also provide:- - 5) DAPM description. - 6) DAPM event handler. 7) DAC Digital mute control. Its probably best to use this guide in conjunction with the existing codec @@ -64,26 +65,9 @@ struct snd_soc_dai_driver wm8731_dai = { 2 - Codec control IO -------------------- The codec can usually be controlled via an I2C or SPI style interface -(AC97 combines control with data in the DAI). The codec drivers provide -functions to read and write the codec registers along with supplying a -register cache:- - - /* IO control data and register cache */ - void *control_data; /* codec control (i2c/3wire) data */ - void *reg_cache; - -Codec read/write should do any data formatting and call the hardware -read write below to perform the IO. These functions are called by the -core and ALSA when performing DAPM or changing the mixer:- - - unsigned int (*read)(struct snd_soc_codec *, unsigned int); - int (*write)(struct snd_soc_codec *, unsigned int, unsigned int); - -Codec hardware IO functions - usually points to either the I2C, SPI or AC97 -read/write:- - - hw_write_t hw_write; - hw_read_t hw_read; +(AC97 combines control with data in the DAI). The codec driver should use the +Regmap API for all codec IO. Please see include/linux/regmap.h and existing +codec drivers for example regmap usage. 3 - Mixers and audio controls @@ -127,7 +111,7 @@ Defines a stereo enumerated control 4 - Codec Audio Operations -------------------------- -The codec driver also supports the following ALSA operations:- +The codec driver also supports the following ALSA PCM operations:- /* SoC audio ops */ struct snd_soc_ops { -- 1.8.1.2

3 6

[alsa-devel] [PATCH] ASoC: fsl: fsl_ssi: Fix simultaneous capture and playback
by Fabio Estevam 24 Sep '13

24 Sep '13

From: Fabio Estevam <fabio.estevam(a)freescale.com> When doing simultaneous capture and playback on a mx6 board we get the following error: $ arecord -f cd | aplay -f cd imx-sgtl5000 sound.13: set sample size in capture stream first fsl-ssi-dai 2028000.ssi: ASoC: can't open interface 2028000.ssi: -11 ALSA lib pcm_dmix.c:1018:(snd_pcm_dmix_ open) unable to open slave aplay: main:660: audio open error: Device or resource busy Recording WAVE 'stdin' : Signed 16 bit Little Endian, Rate 44100 Hz, Stereo The 'arecord -f cd | aplay -f cd' always trigger cause the 'if (!first_runtime->sample_bits)' block to be true which returns an error. Adjust the logic inside fsl_ssi_startup(), so that we do not always hit the error when playing 'arecord | aplay' line for the first time. Reported-by: Chris Clepper <cgclepper(a)gmail.com> Suggested-by: Nicolin Chen <b42378(a)freescale.com> Signed-off-by: Fabio Estevam <fabio.estevam(a)freescale.com> --- sound/soc/fsl/fsl_ssi.c | 15 ++++----------- 1 file changed, 4 insertions(+), 11 deletions(-) diff --git a/sound/soc/fsl/fsl_ssi.c b/sound/soc/fsl/fsl_ssi.c index 6ac8730..cdbb641 100644 --- a/sound/soc/fsl/fsl_ssi.c +++ b/sound/soc/fsl/fsl_ssi.c @@ -469,19 +469,12 @@ static int fsl_ssi_startup(struct snd_pcm_substream *substream, * parameters, then the second stream may be * constrained to the wrong sample rate or size. */ - if (!first_runtime->sample_bits) { - dev_err(substream->pcm->card->dev, - "set sample size in %s stream first\n", - substream->stream == - SNDRV_PCM_STREAM_PLAYBACK - ? "capture" : "playback"); - return -EAGAIN; - } - - snd_pcm_hw_constraint_minmax(substream->runtime, - SNDRV_PCM_HW_PARAM_SAMPLE_BITS, + if (first_runtime->sample_bits) { + snd_pcm_hw_constraint_minmax(substream->runtime, + SNDRV_PCM_HW_PARAM_SAMPLE_BITS, first_runtime->sample_bits, first_runtime->sample_bits); + } } ssi_private->second_stream = substream; -- 1.8.1.2

2 1

[alsa-devel] Audio Miniconference 2013 - Edinburgh 21st October
by Mark Brown 24 Sep '13

24 Sep '13

As in previous years we're planning to have an audio mini-summit to bring everyone together to discuss the current state of audio in Linux, try to resolve outstanding design issues and do some hacking. This year we will be meeting in Edinburgh on 21st October as part of the Linux Foundation conferences being held that week. What topics would people like to discuss? There are a number of ongoing issues like DSP integration and related topics such as compressed audio and I know that there are some issues which have come up while people have been working on deploying UCM especially with PulseAudio integration. What other areas are of interest to people? To help with planning could you please fill out the form at: https://docs.google.com/forms/d/105uav9tkwwkDU4P1T3IOWjfsa3kgtDbFAJOlHgBQR5… if you're planning to attend, ideally by 21st September. Since the facilities are being provided by the Linux Foundation attendees will need to be registered for at least one of the other conferences that week, if that's an issue please let us know. There's a space for topics of interest on the form but please also bring things up on the list so other people can get a feel for what's going on and ideally so we can resolve them all via e-mail!

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[alsa-devel] [PATCH 0/2][RFC] snd_soc_codec include snd_soc_component
by Kuninori Morimoto 24 Sep '13

24 Sep '13

Hi Mark, Lars These are very rough patch for merging snd_soc_codec <-> snd_soc_component. As Lars idea, "codec" includes "component" on this patch, but it doesn't care about "platform" here, since "platform" doesn't have "dai". Current component is caring "dai", but... ? (and what about "card" ?) Anyway, this patch focus only codec <-> component. The "driver code" was not big difference, since current component is very small at this point. Attached 1st patch is rough version of ASoC core code. the main topic on this patch is that snd_soc_register_dai[s] will be called via component. But, we will need many "codec dai name exchange", becouse 1) the dai name rule on 1-dai / multi-dai are different, it was depend on snd_soc_register_dai[s]. 1) component is using both snd_soc_register_dai[s] inside (this was for "cpu" side limit) 2) codec have been used snd_soc_register_dais even though it was not multi-dai 2nd patch is rough version of "code dai name exchange" for it. Of course we need more and more codec driver patches, but it is wm8978 only as trial now. I guess "codec driver" should have "component driver" pointer too ? (it has .name only at this point, and 2nd patch includes it) We can move some feature from codec to component after this step ? In my quick check, below codec have 0 dai (= we need care it on component ?) sound/soc/codecs/lm4857.c sound/soc/codecs/max9768.c sound/soc/codecs/max9877.c sound/soc/codecs/wm2000.c sound/soc/codecs/wm9090.c below codecs have 1 dai (= we need name exchange on platform) sound/soc/codecs/ac97.c sound/soc/codecs/ad1836.c sound/soc/codecs/ad193x.c sound/soc/codecs/ad193x.c sound/soc/codecs/ad1980.c sound/soc/codecs/ad73311.c sound/soc/codecs/adau1701.c sound/soc/codecs/ads117x.c sound/soc/codecs/ak4104.c sound/soc/codecs/ak4535.c sound/soc/codecs/ak4554.c sound/soc/codecs/ak4642.c sound/soc/codecs/ak4671.c sound/soc/codecs/ak5386.c sound/soc/codecs/alc5623.c sound/soc/codecs/alc5632.c sound/soc/codecs/bt-sco.c sound/soc/codecs/cq93vc.c sound/soc/codecs/cs4270.c sound/soc/codecs/cs4271.c sound/soc/codecs/cs4271.c sound/soc/codecs/cs42l51.c sound/soc/codecs/cs42l52.c sound/soc/codecs/cx20442.c sound/soc/codecs/da7210.c sound/soc/codecs/da7210.c sound/soc/codecs/da7213.c sound/soc/codecs/da9055.c sound/soc/codecs/dmic.c sound/soc/codecs/hdmi.c sound/soc/codecs/jz4740.c sound/soc/codecs/max9850.c sound/soc/codecs/ml26124.c sound/soc/codecs/pcm1681.c sound/soc/codecs/pcm1792a.c sound/soc/codecs/pcm3008.c sound/soc/codecs/sgtl5000.c sound/soc/codecs/si476x.c sound/soc/codecs/spdif_receiver.c sound/soc/codecs/spdif_transmitter.c sound/soc/codecs/ssm2518.c sound/soc/codecs/ssm2602.c sound/soc/codecs/ssm2602.c sound/soc/codecs/sta32x.c sound/soc/codecs/sta529.c sound/soc/codecs/tas5086.c sound/soc/codecs/tlv320aic23.c sound/soc/codecs/tlv320aic26.c sound/soc/codecs/tlv320aic32x4.c sound/soc/codecs/tlv320aic3x.c sound/soc/codecs/tlv320dac33.c sound/soc/codecs/uda134x.c sound/soc/codecs/wl1273.c sound/soc/codecs/wm1250-ev1.c sound/soc/codecs/wm2200.c sound/soc/codecs/wm8350.c sound/soc/codecs/wm8400.c sound/soc/codecs/wm8510.c sound/soc/codecs/wm8510.c sound/soc/codecs/wm8523.c sound/soc/codecs/wm8711.c sound/soc/codecs/wm8711.c sound/soc/codecs/wm8727.c sound/soc/codecs/wm8728.c sound/soc/codecs/wm8728.c sound/soc/codecs/wm8731.c sound/soc/codecs/wm8731.c sound/soc/codecs/wm8737.c sound/soc/codecs/wm8737.c sound/soc/codecs/wm8741.c sound/soc/codecs/wm8741.c sound/soc/codecs/wm8750.c sound/soc/codecs/wm8750.c sound/soc/codecs/wm8770.c sound/soc/codecs/wm8782.c sound/soc/codecs/wm8804.c sound/soc/codecs/wm8804.c sound/soc/codecs/wm8900.c sound/soc/codecs/wm8900.c sound/soc/codecs/wm8903.c sound/soc/codecs/wm8904.c sound/soc/codecs/wm8940.c sound/soc/codecs/wm8955.c sound/soc/codecs/wm8960.c sound/soc/codecs/wm8961.c sound/soc/codecs/wm8962.c sound/soc/codecs/wm8971.c sound/soc/codecs/wm8974.c sound/soc/codecs/wm8978.c sound/soc/codecs/wm8983.c sound/soc/codecs/wm8983.c sound/soc/codecs/wm8985.c sound/soc/codecs/wm8985.c sound/soc/codecs/wm8988.c sound/soc/codecs/wm8988.c sound/soc/codecs/wm8990.c sound/soc/codecs/wm8991.c sound/soc/codecs/wm8993.c sound/soc/codecs/wm9081.c sound/soc/codecs/max98090.c sound/soc/codecs/rt5631.c below codecs have multi dai sound/soc/codecs/88pm860x-codec.c sound/soc/codecs/ab8500-codec.c sound/soc/codecs/adau1373.c sound/soc/codecs/adav80x.c sound/soc/codecs/ak4641.c sound/soc/codecs/cs42l73.c sound/soc/codecs/da732x.c sound/soc/codecs/isabelle.c sound/soc/codecs/lm49453.c sound/soc/codecs/max98088.c sound/soc/codecs/max98095.c sound/soc/codecs/rt5640.c sound/soc/codecs/sn95031.c sound/soc/codecs/stac9766.c sound/soc/codecs/twl4030.c sound/soc/codecs/twl6040.c sound/soc/codecs/uda1380.c sound/soc/codecs/wm0010.c sound/soc/codecs/wm5100.c sound/soc/codecs/wm5102.c sound/soc/codecs/wm5110.c sound/soc/codecs/wm8580.c sound/soc/codecs/wm8753.c sound/soc/codecs/wm8776.c sound/soc/codecs/wm8994.c sound/soc/codecs/wm8995.c sound/soc/codecs/wm8996.c sound/soc/codecs/wm8997.c sound/soc/codecs/wm9705.c sound/soc/codecs/wm9712.c sound/soc/codecs/wm9713.c and, below codec has both 1-dai, and multi-dai sound/soc/codecs/mc13783.c Kuninori Morimoto (2): [RFC] ASoC: snd_soc_codec include snd_soc_component [RFC] ASoC: wm8978: use component arch/arm/mach-shmobile/board-armadillo800eva.c | 2 +- include/sound/soc.h | 33 ++++++++++-------- sound/soc/codecs/wm8978.c | 6 ++++ sound/soc/soc-core.c | 44 ++++++++++++++++++------ 4 files changed, 59 insertions(+), 26 deletions(-)

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Re: [alsa-devel] [PATCH 3/3] ASoC: max98088: Convert to direct regmap API usage
by Dylan Reid 24 Sep '13

24 Sep '13

On Mon, Sep 23, 2013 at 9:58 AM, Mark Brown <broonie(a)kernel.org> wrote: > From: Mark Brown <broonie(a)linaro.org> > > This saves code and moves us towards removing the redundant register I/O > implementation in ASoC. > > Signed-off-by: Mark Brown <broonie(a)linaro.org> all three successfully tested. > --- > sound/soc/codecs/max98088.c | 580 +++++++++++++++++++------------------------- > 1 file changed, 251 insertions(+), 329 deletions(-) > > diff --git a/sound/soc/codecs/max98088.c b/sound/soc/codecs/max98088.c > index 8896d5e..31912d5 100644 > --- a/sound/soc/codecs/max98088.c > +++ b/sound/soc/codecs/max98088.c > @@ -15,6 +15,7 @@ > #include <linux/delay.h> > #include <linux/pm.h> > #include <linux/i2c.h> > +#include <linux/regmap.h> > #include <sound/core.h> > #include <sound/pcm.h> > #include <sound/pcm_params.h> > @@ -38,6 +39,7 @@ struct max98088_cdata { > }; > > struct max98088_priv { > + struct regmap *regmap; > enum max98088_type devtype; > struct max98088_pdata *pdata; > unsigned int sysclk; > @@ -54,278 +56,206 @@ struct max98088_priv { > unsigned int extmic_mode; > }; > > -static const u8 max98088_reg[M98088_REG_CNT] = { > - 0x00, /* 00 IRQ status */ > - 0x00, /* 01 MIC status */ > - 0x00, /* 02 jack status */ > - 0x00, /* 03 battery voltage */ > - 0x00, /* 04 */ > - 0x00, /* 05 */ > - 0x00, /* 06 */ > - 0x00, /* 07 */ > - 0x00, /* 08 */ > - 0x00, /* 09 */ > - 0x00, /* 0A */ > - 0x00, /* 0B */ > - 0x00, /* 0C */ > - 0x00, /* 0D */ > - 0x00, /* 0E */ > - 0x00, /* 0F interrupt enable */ > - > - 0x00, /* 10 master clock */ > - 0x00, /* 11 DAI1 clock mode */ > - 0x00, /* 12 DAI1 clock control */ > - 0x00, /* 13 DAI1 clock control */ > - 0x00, /* 14 DAI1 format */ > - 0x00, /* 15 DAI1 clock */ > - 0x00, /* 16 DAI1 config */ > - 0x00, /* 17 DAI1 TDM */ > - 0x00, /* 18 DAI1 filters */ > - 0x00, /* 19 DAI2 clock mode */ > - 0x00, /* 1A DAI2 clock control */ > - 0x00, /* 1B DAI2 clock control */ > - 0x00, /* 1C DAI2 format */ > - 0x00, /* 1D DAI2 clock */ > - 0x00, /* 1E DAI2 config */ > - 0x00, /* 1F DAI2 TDM */ > - > - 0x00, /* 20 DAI2 filters */ > - 0x00, /* 21 data config */ > - 0x00, /* 22 DAC mixer */ > - 0x00, /* 23 left ADC mixer */ > - 0x00, /* 24 right ADC mixer */ > - 0x00, /* 25 left HP mixer */ > - 0x00, /* 26 right HP mixer */ > - 0x00, /* 27 HP control */ > - 0x00, /* 28 left REC mixer */ > - 0x00, /* 29 right REC mixer */ > - 0x00, /* 2A REC control */ > - 0x00, /* 2B left SPK mixer */ > - 0x00, /* 2C right SPK mixer */ > - 0x00, /* 2D SPK control */ > - 0x00, /* 2E sidetone */ > - 0x00, /* 2F DAI1 playback level */ > - > - 0x00, /* 30 DAI1 playback level */ > - 0x00, /* 31 DAI2 playback level */ > - 0x00, /* 32 DAI2 playbakc level */ > - 0x00, /* 33 left ADC level */ > - 0x00, /* 34 right ADC level */ > - 0x00, /* 35 MIC1 level */ > - 0x00, /* 36 MIC2 level */ > - 0x00, /* 37 INA level */ > - 0x00, /* 38 INB level */ > - 0x00, /* 39 left HP volume */ > - 0x00, /* 3A right HP volume */ > - 0x00, /* 3B left REC volume */ > - 0x00, /* 3C right REC volume */ > - 0x00, /* 3D left SPK volume */ > - 0x00, /* 3E right SPK volume */ > - 0x00, /* 3F MIC config */ > - > - 0x00, /* 40 MIC threshold */ > - 0x00, /* 41 excursion limiter filter */ > - 0x00, /* 42 excursion limiter threshold */ > - 0x00, /* 43 ALC */ > - 0x00, /* 44 power limiter threshold */ > - 0x00, /* 45 power limiter config */ > - 0x00, /* 46 distortion limiter config */ > - 0x00, /* 47 audio input */ > - 0x00, /* 48 microphone */ > - 0x00, /* 49 level control */ > - 0x00, /* 4A bypass switches */ > - 0x00, /* 4B jack detect */ > - 0x00, /* 4C input enable */ > - 0x00, /* 4D output enable */ > - 0xF0, /* 4E bias control */ > - 0x00, /* 4F DAC power */ > - > - 0x0F, /* 50 DAC power */ > - 0x00, /* 51 system */ > - 0x00, /* 52 DAI1 EQ1 */ > - 0x00, /* 53 DAI1 EQ1 */ > - 0x00, /* 54 DAI1 EQ1 */ > - 0x00, /* 55 DAI1 EQ1 */ > - 0x00, /* 56 DAI1 EQ1 */ > - 0x00, /* 57 DAI1 EQ1 */ > - 0x00, /* 58 DAI1 EQ1 */ > - 0x00, /* 59 DAI1 EQ1 */ > - 0x00, /* 5A DAI1 EQ1 */ > - 0x00, /* 5B DAI1 EQ1 */ > - 0x00, /* 5C DAI1 EQ2 */ > - 0x00, /* 5D DAI1 EQ2 */ > - 0x00, /* 5E DAI1 EQ2 */ > - 0x00, /* 5F DAI1 EQ2 */ > - > - 0x00, /* 60 DAI1 EQ2 */ > - 0x00, /* 61 DAI1 EQ2 */ > - 0x00, /* 62 DAI1 EQ2 */ > - 0x00, /* 63 DAI1 EQ2 */ > - 0x00, /* 64 DAI1 EQ2 */ > - 0x00, /* 65 DAI1 EQ2 */ > - 0x00, /* 66 DAI1 EQ3 */ > - 0x00, /* 67 DAI1 EQ3 */ > - 0x00, /* 68 DAI1 EQ3 */ > - 0x00, /* 69 DAI1 EQ3 */ > - 0x00, /* 6A DAI1 EQ3 */ > - 0x00, /* 6B DAI1 EQ3 */ > - 0x00, /* 6C DAI1 EQ3 */ > - 0x00, /* 6D DAI1 EQ3 */ > - 0x00, /* 6E DAI1 EQ3 */ > - 0x00, /* 6F DAI1 EQ3 */ > - > - 0x00, /* 70 DAI1 EQ4 */ > - 0x00, /* 71 DAI1 EQ4 */ > - 0x00, /* 72 DAI1 EQ4 */ > - 0x00, /* 73 DAI1 EQ4 */ > - 0x00, /* 74 DAI1 EQ4 */ > - 0x00, /* 75 DAI1 EQ4 */ > - 0x00, /* 76 DAI1 EQ4 */ > - 0x00, /* 77 DAI1 EQ4 */ > - 0x00, /* 78 DAI1 EQ4 */ > - 0x00, /* 79 DAI1 EQ4 */ > - 0x00, /* 7A DAI1 EQ5 */ > - 0x00, /* 7B DAI1 EQ5 */ > - 0x00, /* 7C DAI1 EQ5 */ > - 0x00, /* 7D DAI1 EQ5 */ > - 0x00, /* 7E DAI1 EQ5 */ > - 0x00, /* 7F DAI1 EQ5 */ > - > - 0x00, /* 80 DAI1 EQ5 */ > - 0x00, /* 81 DAI1 EQ5 */ > - 0x00, /* 82 DAI1 EQ5 */ > - 0x00, /* 83 DAI1 EQ5 */ > - 0x00, /* 84 DAI2 EQ1 */ > - 0x00, /* 85 DAI2 EQ1 */ > - 0x00, /* 86 DAI2 EQ1 */ > - 0x00, /* 87 DAI2 EQ1 */ > - 0x00, /* 88 DAI2 EQ1 */ > - 0x00, /* 89 DAI2 EQ1 */ > - 0x00, /* 8A DAI2 EQ1 */ > - 0x00, /* 8B DAI2 EQ1 */ > - 0x00, /* 8C DAI2 EQ1 */ > - 0x00, /* 8D DAI2 EQ1 */ > - 0x00, /* 8E DAI2 EQ2 */ > - 0x00, /* 8F DAI2 EQ2 */ > - > - 0x00, /* 90 DAI2 EQ2 */ > - 0x00, /* 91 DAI2 EQ2 */ > - 0x00, /* 92 DAI2 EQ2 */ > - 0x00, /* 93 DAI2 EQ2 */ > - 0x00, /* 94 DAI2 EQ2 */ > - 0x00, /* 95 DAI2 EQ2 */ > - 0x00, /* 96 DAI2 EQ2 */ > - 0x00, /* 97 DAI2 EQ2 */ > - 0x00, /* 98 DAI2 EQ3 */ > - 0x00, /* 99 DAI2 EQ3 */ > - 0x00, /* 9A DAI2 EQ3 */ > - 0x00, /* 9B DAI2 EQ3 */ > - 0x00, /* 9C DAI2 EQ3 */ > - 0x00, /* 9D DAI2 EQ3 */ > - 0x00, /* 9E DAI2 EQ3 */ > - 0x00, /* 9F DAI2 EQ3 */ > - > - 0x00, /* A0 DAI2 EQ3 */ > - 0x00, /* A1 DAI2 EQ3 */ > - 0x00, /* A2 DAI2 EQ4 */ > - 0x00, /* A3 DAI2 EQ4 */ > - 0x00, /* A4 DAI2 EQ4 */ > - 0x00, /* A5 DAI2 EQ4 */ > - 0x00, /* A6 DAI2 EQ4 */ > - 0x00, /* A7 DAI2 EQ4 */ > - 0x00, /* A8 DAI2 EQ4 */ > - 0x00, /* A9 DAI2 EQ4 */ > - 0x00, /* AA DAI2 EQ4 */ > - 0x00, /* AB DAI2 EQ4 */ > - 0x00, /* AC DAI2 EQ5 */ > - 0x00, /* AD DAI2 EQ5 */ > - 0x00, /* AE DAI2 EQ5 */ > - 0x00, /* AF DAI2 EQ5 */ > - > - 0x00, /* B0 DAI2 EQ5 */ > - 0x00, /* B1 DAI2 EQ5 */ > - 0x00, /* B2 DAI2 EQ5 */ > - 0x00, /* B3 DAI2 EQ5 */ > - 0x00, /* B4 DAI2 EQ5 */ > - 0x00, /* B5 DAI2 EQ5 */ > - 0x00, /* B6 DAI1 biquad */ > - 0x00, /* B7 DAI1 biquad */ > - 0x00, /* B8 DAI1 biquad */ > - 0x00, /* B9 DAI1 biquad */ > - 0x00, /* BA DAI1 biquad */ > - 0x00, /* BB DAI1 biquad */ > - 0x00, /* BC DAI1 biquad */ > - 0x00, /* BD DAI1 biquad */ > - 0x00, /* BE DAI1 biquad */ > - 0x00, /* BF DAI1 biquad */ > - > - 0x00, /* C0 DAI2 biquad */ > - 0x00, /* C1 DAI2 biquad */ > - 0x00, /* C2 DAI2 biquad */ > - 0x00, /* C3 DAI2 biquad */ > - 0x00, /* C4 DAI2 biquad */ > - 0x00, /* C5 DAI2 biquad */ > - 0x00, /* C6 DAI2 biquad */ > - 0x00, /* C7 DAI2 biquad */ > - 0x00, /* C8 DAI2 biquad */ > - 0x00, /* C9 DAI2 biquad */ > - 0x00, /* CA */ > - 0x00, /* CB */ > - 0x00, /* CC */ > - 0x00, /* CD */ > - 0x00, /* CE */ > - 0x00, /* CF */ > - > - 0x00, /* D0 */ > - 0x00, /* D1 */ > - 0x00, /* D2 */ > - 0x00, /* D3 */ > - 0x00, /* D4 */ > - 0x00, /* D5 */ > - 0x00, /* D6 */ > - 0x00, /* D7 */ > - 0x00, /* D8 */ > - 0x00, /* D9 */ > - 0x00, /* DA */ > - 0x70, /* DB */ > - 0x00, /* DC */ > - 0x00, /* DD */ > - 0x00, /* DE */ > - 0x00, /* DF */ > - > - 0x00, /* E0 */ > - 0x00, /* E1 */ > - 0x00, /* E2 */ > - 0x00, /* E3 */ > - 0x00, /* E4 */ > - 0x00, /* E5 */ > - 0x00, /* E6 */ > - 0x00, /* E7 */ > - 0x00, /* E8 */ > - 0x00, /* E9 */ > - 0x00, /* EA */ > - 0x00, /* EB */ > - 0x00, /* EC */ > - 0x00, /* ED */ > - 0x00, /* EE */ > - 0x00, /* EF */ > - > - 0x00, /* F0 */ > - 0x00, /* F1 */ > - 0x00, /* F2 */ > - 0x00, /* F3 */ > - 0x00, /* F4 */ > - 0x00, /* F5 */ > - 0x00, /* F6 */ > - 0x00, /* F7 */ > - 0x00, /* F8 */ > - 0x00, /* F9 */ > - 0x00, /* FA */ > - 0x00, /* FB */ > - 0x00, /* FC */ > - 0x00, /* FD */ > - 0x00, /* FE */ > - 0x00, /* FF */ > +static const struct reg_default max98088_reg[] = { > + { 0xf, 0x00 }, /* 0F interrupt enable */ > + > + { 0x10, 0x00 }, /* 10 master clock */ > + { 0x11, 0x00 }, /* 11 DAI1 clock mode */ > + { 0x12, 0x00 }, /* 12 DAI1 clock control */ > + { 0x13, 0x00 }, /* 13 DAI1 clock control */ > + { 0x14, 0x00 }, /* 14 DAI1 format */ > + { 0x15, 0x00 }, /* 15 DAI1 clock */ > + { 0x16, 0x00 }, /* 16 DAI1 config */ > + { 0x17, 0x00 }, /* 17 DAI1 TDM */ > + { 0x18, 0x00 }, /* 18 DAI1 filters */ > + { 0x19, 0x00 }, /* 19 DAI2 clock mode */ > + { 0x1a, 0x00 }, /* 1A DAI2 clock control */ > + { 0x1b, 0x00 }, /* 1B DAI2 clock control */ > + { 0x1c, 0x00 }, /* 1C DAI2 format */ > + { 0x1d, 0x00 }, /* 1D DAI2 clock */ > + { 0x1e, 0x00 }, /* 1E DAI2 config */ > + { 0x1f, 0x00 }, /* 1F DAI2 TDM */ > + > + { 0x20, 0x00 }, /* 20 DAI2 filters */ > + { 0x21, 0x00 }, /* 21 data config */ > + { 0x22, 0x00 }, /* 22 DAC mixer */ > + { 0x23, 0x00 }, /* 23 left ADC mixer */ > + { 0x24, 0x00 }, /* 24 right ADC mixer */ > + { 0x25, 0x00 }, /* 25 left HP mixer */ > + { 0x26, 0x00 }, /* 26 right HP mixer */ > + { 0x27, 0x00 }, /* 27 HP control */ > + { 0x28, 0x00 }, /* 28 left REC mixer */ > + { 0x29, 0x00 }, /* 29 right REC mixer */ > + { 0x2a, 0x00 }, /* 2A REC control */ > + { 0x2b, 0x00 }, /* 2B left SPK mixer */ > + { 0x2c, 0x00 }, /* 2C right SPK mixer */ > + { 0x2d, 0x00 }, /* 2D SPK control */ > + { 0x2e, 0x00 }, /* 2E sidetone */ > + { 0x2f, 0x00 }, /* 2F DAI1 playback level */ > + > + { 0x30, 0x00 }, /* 30 DAI1 playback level */ > + { 0x31, 0x00 }, /* 31 DAI2 playback level */ > + { 0x32, 0x00 }, /* 32 DAI2 playbakc level */ > + { 0x33, 0x00 }, /* 33 left ADC level */ > + { 0x34, 0x00 }, /* 34 right ADC level */ > + { 0x35, 0x00 }, /* 35 MIC1 level */ > + { 0x36, 0x00 }, /* 36 MIC2 level */ > + { 0x37, 0x00 }, /* 37 INA level */ > + { 0x38, 0x00 }, /* 38 INB level */ > + { 0x39, 0x00 }, /* 39 left HP volume */ > + { 0x3a, 0x00 }, /* 3A right HP volume */ > + { 0x3b, 0x00 }, /* 3B left REC volume */ > + { 0x3c, 0x00 }, /* 3C right REC volume */ > + { 0x3d, 0x00 }, /* 3D left SPK volume */ > + { 0x3e, 0x00 }, /* 3E right SPK volume */ > + { 0x3f, 0x00 }, /* 3F MIC config */ > + > + { 0x40, 0x00 }, /* 40 MIC threshold */ > + { 0x41, 0x00 }, /* 41 excursion limiter filter */ > + { 0x42, 0x00 }, /* 42 excursion limiter threshold */ > + { 0x43, 0x00 }, /* 43 ALC */ > + { 0x44, 0x00 }, /* 44 power limiter threshold */ > + { 0x45, 0x00 }, /* 45 power limiter config */ > + { 0x46, 0x00 }, /* 46 distortion limiter config */ > + { 0x47, 0x00 }, /* 47 audio input */ > + { 0x48, 0x00 }, /* 48 microphone */ > + { 0x49, 0x00 }, /* 49 level control */ > + { 0x4a, 0x00 }, /* 4A bypass switches */ > + { 0x4b, 0x00 }, /* 4B jack detect */ > + { 0x4c, 0x00 }, /* 4C input enable */ > + { 0x4d, 0x00 }, /* 4D output enable */ > + { 0x4e, 0xF0 }, /* 4E bias control */ > + { 0x4f, 0x00 }, /* 4F DAC power */ > + > + { 0x50, 0x0F }, /* 50 DAC power */ > + { 0x51, 0x00 }, /* 51 system */ > + { 0x52, 0x00 }, /* 52 DAI1 EQ1 */ > + { 0x53, 0x00 }, /* 53 DAI1 EQ1 */ > + { 0x54, 0x00 }, /* 54 DAI1 EQ1 */ > + { 0x55, 0x00 }, /* 55 DAI1 EQ1 */ > + { 0x56, 0x00 }, /* 56 DAI1 EQ1 */ > + { 0x57, 0x00 }, /* 57 DAI1 EQ1 */ > + { 0x58, 0x00 }, /* 58 DAI1 EQ1 */ > + { 0x59, 0x00 }, /* 59 DAI1 EQ1 */ > + { 0x5a, 0x00 }, /* 5A DAI1 EQ1 */ > + { 0x5b, 0x00 }, /* 5B DAI1 EQ1 */ > + { 0x5c, 0x00 }, /* 5C DAI1 EQ2 */ > + { 0x5d, 0x00 }, /* 5D DAI1 EQ2 */ > + { 0x5e, 0x00 }, /* 5E DAI1 EQ2 */ > + { 0x5f, 0x00 }, /* 5F DAI1 EQ2 */ > + > + { 0x60, 0x00 }, /* 60 DAI1 EQ2 */ > + { 0x61, 0x00 }, /* 61 DAI1 EQ2 */ > + { 0x62, 0x00 }, /* 62 DAI1 EQ2 */ > + { 0x63, 0x00 }, /* 63 DAI1 EQ2 */ > + { 0x64, 0x00 }, /* 64 DAI1 EQ2 */ > + { 0x65, 0x00 }, /* 65 DAI1 EQ2 */ > + { 0x66, 0x00 }, /* 66 DAI1 EQ3 */ > + { 0x67, 0x00 }, /* 67 DAI1 EQ3 */ > + { 0x68, 0x00 }, /* 68 DAI1 EQ3 */ > + { 0x69, 0x00 }, /* 69 DAI1 EQ3 */ > + { 0x6a, 0x00 }, /* 6A DAI1 EQ3 */ > + { 0x6b, 0x00 }, /* 6B DAI1 EQ3 */ > + { 0x6c, 0x00 }, /* 6C DAI1 EQ3 */ > + { 0x6d, 0x00 }, /* 6D DAI1 EQ3 */ > + { 0x6e, 0x00 }, /* 6E DAI1 EQ3 */ > + { 0x6f, 0x00 }, /* 6F DAI1 EQ3 */ > + > + { 0x70, 0x00 }, /* 70 DAI1 EQ4 */ > + { 0x71, 0x00 }, /* 71 DAI1 EQ4 */ > + { 0x72, 0x00 }, /* 72 DAI1 EQ4 */ > + { 0x73, 0x00 }, /* 73 DAI1 EQ4 */ > + { 0x74, 0x00 }, /* 74 DAI1 EQ4 */ > + { 0x75, 0x00 }, /* 75 DAI1 EQ4 */ > + { 0x76, 0x00 }, /* 76 DAI1 EQ4 */ > + { 0x77, 0x00 }, /* 77 DAI1 EQ4 */ > + { 0x78, 0x00 }, /* 78 DAI1 EQ4 */ > + { 0x79, 0x00 }, /* 79 DAI1 EQ4 */ > + { 0x7a, 0x00 }, /* 7A DAI1 EQ5 */ > + { 0x7b, 0x00 }, /* 7B DAI1 EQ5 */ > + { 0x7c, 0x00 }, /* 7C DAI1 EQ5 */ > + { 0x7d, 0x00 }, /* 7D DAI1 EQ5 */ > + { 0x7e, 0x00 }, /* 7E DAI1 EQ5 */ > + { 0x7f, 0x00 }, /* 7F DAI1 EQ5 */ > + > + { 0x80, 0x00 }, /* 80 DAI1 EQ5 */ > + { 0x81, 0x00 }, /* 81 DAI1 EQ5 */ > + { 0x82, 0x00 }, /* 82 DAI1 EQ5 */ > + { 0x83, 0x00 }, /* 83 DAI1 EQ5 */ > + { 0x84, 0x00 }, /* 84 DAI2 EQ1 */ > + { 0x85, 0x00 }, /* 85 DAI2 EQ1 */ > + { 0x86, 0x00 }, /* 86 DAI2 EQ1 */ > + { 0x87, 0x00 }, /* 87 DAI2 EQ1 */ > + { 0x88, 0x00 }, /* 88 DAI2 EQ1 */ > + { 0x89, 0x00 }, /* 89 DAI2 EQ1 */ > + { 0x8a, 0x00 }, /* 8A DAI2 EQ1 */ > + { 0x8b, 0x00 }, /* 8B DAI2 EQ1 */ > + { 0x8c, 0x00 }, /* 8C DAI2 EQ1 */ > + { 0x8d, 0x00 }, /* 8D DAI2 EQ1 */ > + { 0x8e, 0x00 }, /* 8E DAI2 EQ2 */ > + { 0x8f, 0x00 }, /* 8F DAI2 EQ2 */ > + > + { 0x90, 0x00 }, /* 90 DAI2 EQ2 */ > + { 0x91, 0x00 }, /* 91 DAI2 EQ2 */ > + { 0x92, 0x00 }, /* 92 DAI2 EQ2 */ > + { 0x93, 0x00 }, /* 93 DAI2 EQ2 */ > + { 0x94, 0x00 }, /* 94 DAI2 EQ2 */ > + { 0x95, 0x00 }, /* 95 DAI2 EQ2 */ > + { 0x96, 0x00 }, /* 96 DAI2 EQ2 */ > + { 0x97, 0x00 }, /* 97 DAI2 EQ2 */ > + { 0x98, 0x00 }, /* 98 DAI2 EQ3 */ > + { 0x99, 0x00 }, /* 99 DAI2 EQ3 */ > + { 0x9a, 0x00 }, /* 9A DAI2 EQ3 */ > + { 0x9b, 0x00 }, /* 9B DAI2 EQ3 */ > + { 0x9c, 0x00 }, /* 9C DAI2 EQ3 */ > + { 0x9d, 0x00 }, /* 9D DAI2 EQ3 */ > + { 0x9e, 0x00 }, /* 9E DAI2 EQ3 */ > + { 0x9f, 0x00 }, /* 9F DAI2 EQ3 */ > + > + { 0xa0, 0x00 }, /* A0 DAI2 EQ3 */ > + { 0xa1, 0x00 }, /* A1 DAI2 EQ3 */ > + { 0xa2, 0x00 }, /* A2 DAI2 EQ4 */ > + { 0xa3, 0x00 }, /* A3 DAI2 EQ4 */ > + { 0xa4, 0x00 }, /* A4 DAI2 EQ4 */ > + { 0xa5, 0x00 }, /* A5 DAI2 EQ4 */ > + { 0xa6, 0x00 }, /* A6 DAI2 EQ4 */ > + { 0xa7, 0x00 }, /* A7 DAI2 EQ4 */ > + { 0xa8, 0x00 }, /* A8 DAI2 EQ4 */ > + { 0xa9, 0x00 }, /* A9 DAI2 EQ4 */ > + { 0xaa, 0x00 }, /* AA DAI2 EQ4 */ > + { 0xab, 0x00 }, /* AB DAI2 EQ4 */ > + { 0xac, 0x00 }, /* AC DAI2 EQ5 */ > + { 0xad, 0x00 }, /* AD DAI2 EQ5 */ > + { 0xae, 0x00 }, /* AE DAI2 EQ5 */ > + { 0xaf, 0x00 }, /* AF DAI2 EQ5 */ > + > + { 0xb0, 0x00 }, /* B0 DAI2 EQ5 */ > + { 0xb1, 0x00 }, /* B1 DAI2 EQ5 */ > + { 0xb2, 0x00 }, /* B2 DAI2 EQ5 */ > + { 0xb3, 0x00 }, /* B3 DAI2 EQ5 */ > + { 0xb4, 0x00 }, /* B4 DAI2 EQ5 */ > + { 0xb5, 0x00 }, /* B5 DAI2 EQ5 */ > + { 0xb6, 0x00 }, /* B6 DAI1 biquad */ > + { 0xb7, 0x00 }, /* B7 DAI1 biquad */ > + { 0xb8 ,0x00 }, /* B8 DAI1 biquad */ > + { 0xb9, 0x00 }, /* B9 DAI1 biquad */ > + { 0xba, 0x00 }, /* BA DAI1 biquad */ > + { 0xbb, 0x00 }, /* BB DAI1 biquad */ > + { 0xbc, 0x00 }, /* BC DAI1 biquad */ > + { 0xbd, 0x00 }, /* BD DAI1 biquad */ > + { 0xbe, 0x00 }, /* BE DAI1 biquad */ > + { 0xbf, 0x00 }, /* BF DAI1 biquad */ > + > + { 0xc0, 0x00 }, /* C0 DAI2 biquad */ > + { 0xc1, 0x00 }, /* C1 DAI2 biquad */ > + { 0xc2, 0x00 }, /* C2 DAI2 biquad */ > + { 0xc3, 0x00 }, /* C3 DAI2 biquad */ > + { 0xc4, 0x00 }, /* C4 DAI2 biquad */ > + { 0xc5, 0x00 }, /* C5 DAI2 biquad */ > + { 0xc6, 0x00 }, /* C6 DAI2 biquad */ > + { 0xc7, 0x00 }, /* C7 DAI2 biquad */ > + { 0xc8, 0x00 }, /* C8 DAI2 biquad */ > + { 0xc9, 0x00 }, /* C9 DAI2 biquad */ > }; > > static struct { > @@ -606,11 +536,27 @@ static struct { > { 0xFF, 0x00, 1 }, /* FF */ > }; > > -static int max98088_volatile_register(struct snd_soc_codec *codec, unsigned int reg) > +static bool max98088_readable_register(struct device *dev, unsigned int reg) > +{ > + return max98088_access[reg].readable; > +} > + > +static bool max98088_volatile_register(struct device *dev, unsigned int reg) > { > return max98088_access[reg].vol; > } > > +static const struct regmap_config max98088_regmap = { > + .reg_bits = 8, > + .val_bits = 8, > + > + .readable_reg = max98088_readable_register, > + .volatile_reg = max98088_volatile_register, > + > + .reg_defaults = max98088_reg, > + .num_reg_defaults = ARRAY_SIZE(max98088_reg), > + .cache_type = REGCACHE_RBTREE, > +}; > > /* > * Load equalizer DSP coefficient configurations registers > @@ -1610,58 +1556,34 @@ static int max98088_dai2_digital_mute(struct snd_soc_dai *codec_dai, int mute) > return 0; > } > > -static void max98088_sync_cache(struct snd_soc_codec *codec) > -{ > - u8 *reg_cache = codec->reg_cache; > - int i; > - > - if (!codec->cache_sync) > - return; > - > - codec->cache_only = 0; > - > - /* write back cached values if they're writeable and > - * different from the hardware default. > - */ > - for (i = 1; i < codec->driver->reg_cache_size; i++) { > - if (!max98088_access[i].writable) > - continue; > - > - if (reg_cache[i] == max98088_reg[i]) > - continue; > - > - snd_soc_write(codec, i, reg_cache[i]); > - } > - > - codec->cache_sync = 0; > -} > - > static int max98088_set_bias_level(struct snd_soc_codec *codec, > enum snd_soc_bias_level level) > { > - switch (level) { > - case SND_SOC_BIAS_ON: > - break; > - > - case SND_SOC_BIAS_PREPARE: > - break; > - > - case SND_SOC_BIAS_STANDBY: > - if (codec->dapm.bias_level == SND_SOC_BIAS_OFF) > - max98088_sync_cache(codec); > - > - snd_soc_update_bits(codec, M98088_REG_4C_PWR_EN_IN, > - M98088_MBEN, M98088_MBEN); > - break; > - > - case SND_SOC_BIAS_OFF: > - snd_soc_update_bits(codec, M98088_REG_4C_PWR_EN_IN, > - M98088_MBEN, 0); > - codec->cache_sync = 1; > - break; > - } > - codec->dapm.bias_level = level; > - return 0; > + struct max98088_priv *max98088 = snd_soc_codec_get_drvdata(codec); > + > + switch (level) { > + case SND_SOC_BIAS_ON: > + break; > + > + case SND_SOC_BIAS_PREPARE: > + break; > + > + case SND_SOC_BIAS_STANDBY: > + if (codec->dapm.bias_level == SND_SOC_BIAS_OFF) > + regcache_sync(max98088->regmap); > + > + snd_soc_update_bits(codec, M98088_REG_4C_PWR_EN_IN, > + M98088_MBEN, M98088_MBEN); > + break; > + > + case SND_SOC_BIAS_OFF: > + snd_soc_update_bits(codec, M98088_REG_4C_PWR_EN_IN, > + M98088_MBEN, 0); > + regcache_mark_dirty(max98088->regmap); > + break; > + } > + codec->dapm.bias_level = level; > + return 0; > } > > #define MAX98088_RATES SNDRV_PCM_RATE_8000_96000 > @@ -1988,9 +1910,9 @@ static int max98088_probe(struct snd_soc_codec *codec) > struct max98088_cdata *cdata; > int ret = 0; > > - codec->cache_sync = 1; > + regcache_mark_dirty(max98088->regmap); > > - ret = snd_soc_codec_set_cache_io(codec, 8, 8, SND_SOC_I2C); > + ret = snd_soc_codec_set_cache_io(codec, 8, 8, SND_SOC_REGMAP); > if (ret != 0) { > dev_err(codec->dev, "Failed to set cache I/O: %d\n", ret); > return ret; > @@ -2070,10 +1992,6 @@ static struct snd_soc_codec_driver soc_codec_dev_max98088 = { > .set_bias_level = max98088_set_bias_level, > .controls = max98088_snd_controls, > .num_controls = ARRAY_SIZE(max98088_snd_controls), > - .reg_cache_size = ARRAY_SIZE(max98088_reg), > - .reg_word_size = sizeof(u8), > - .reg_cache_default = max98088_reg, > - .volatile_register = max98088_volatile_register, > .dapm_widgets = max98088_dapm_widgets, > .num_dapm_widgets = ARRAY_SIZE(max98088_dapm_widgets), > .dapm_routes = max98088_audio_map, > @@ -2081,7 +1999,7 @@ static struct snd_soc_codec_driver soc_codec_dev_max98088 = { > }; > > static int max98088_i2c_probe(struct i2c_client *i2c, > - const struct i2c_device_id *id) > + const struct i2c_device_id *id) > { > struct max98088_priv *max98088; > int ret; > @@ -2091,6 +2009,10 @@ static int max98088_i2c_probe(struct i2c_client *i2c, > if (max98088 == NULL) > return -ENOMEM; > > + max98088->regmap = devm_regmap_init_i2c(i2c, &max98088_regmap); > + if (IS_ERR(max98088->regmap)) > + return PTR_ERR(max98088->regmap); > + > max98088->devtype = id->driver_data; > > i2c_set_clientdata(i2c, max98088); > -- > 1.8.4.rc3 >

1 0

[alsa-devel] [PATCH 1/6] ASoC: tlv320aic3x: Remove nonsense comment for register cache
by Mark Brown 24 Sep '13

24 Sep '13

From: Mark Brown <broonie(a)linaro.org> Every statement in this comment is incorrect either through bitrot or (mostly) through never having corresponded to reality in the first place. Signed-off-by: Mark Brown <broonie(a)linaro.org> --- sound/soc/codecs/tlv320aic3x.c | 6 ------ 1 file changed, 6 deletions(-) diff --git a/sound/soc/codecs/tlv320aic3x.c b/sound/soc/codecs/tlv320aic3x.c index 6e3f269..3abbff3 100644 --- a/sound/soc/codecs/tlv320aic3x.c +++ b/sound/soc/codecs/tlv320aic3x.c @@ -90,12 +90,6 @@ struct aic3x_priv { enum aic3x_micbias_voltage micbias_vg; }; -/* - * AIC3X register cache - * We can't read the AIC3X register space when we are - * using 2 wire for device control, so we cache them instead. - * There is no point in caching the reset register - */ static const u8 aic3x_reg[AIC3X_CACHEREGNUM] = { 0x00, 0x00, 0x00, 0x10, /* 0 */ 0x04, 0x00, 0x00, 0x00, /* 4 */ -- 1.8.4.rc3

1 5

[alsa-devel] [PATCH] ASoC: max9850: Convert to direct regmap API usage
by Mark Brown 23 Sep '13

23 Sep '13

From: Mark Brown <broonie(a)linaro.org> This prepares for removal of the duplicated register I/O functionality in ASoC. Signed-off-by: Mark Brown <broonie(a)linaro.org> --- sound/soc/codecs/max9850.c | 39 +++++++++++++++++++++++++++++---------- 1 file changed, 29 insertions(+), 10 deletions(-) diff --git a/sound/soc/codecs/max9850.c b/sound/soc/codecs/max9850.c index 58c38a5..c5dd617 100644 --- a/sound/soc/codecs/max9850.c +++ b/sound/soc/codecs/max9850.c @@ -18,6 +18,7 @@ #include <linux/module.h> #include <linux/init.h> #include <linux/i2c.h> +#include <linux/regmap.h> #include <linux/slab.h> #include <sound/pcm.h> #include <sound/pcm_params.h> @@ -27,18 +28,26 @@ #include "max9850.h" struct max9850_priv { + struct regmap *regmap; unsigned int sysclk; }; /* max9850 register cache */ -static const u8 max9850_reg[MAX9850_CACHEREGNUM] = { - 0x00, 0x00, 0x0c, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00 +static const struct reg_default max9850_reg[] = { + { 2, 0x0c }, + { 3, 0x00 }, + { 4, 0x00 }, + { 5, 0x00 }, + { 6, 0x00 }, + { 7, 0x00 }, + { 8, 0x00 }, + { 9, 0x00 }, + { 10, 0x00 }, }; /* these registers are not used at the moment but provided for the sake of * completeness */ -static int max9850_volatile_register(struct snd_soc_codec *codec, - unsigned int reg) +static bool max9850_volatile_register(struct device *dev, unsigned int reg) { switch (reg) { case MAX9850_STATUSA: @@ -49,6 +58,15 @@ static int max9850_volatile_register(struct snd_soc_codec *codec, } } +static const struct regmap_config max9850_regmap = { + .reg_bits = 8, + .val_bits = 8, + + .max_register = MAX9850_DIGITAL_AUDIO, + .volatile_reg = max9850_volatile_register, + .cache_type = REGCACHE_RBTREE, +}; + static const unsigned int max9850_tlv[] = { TLV_DB_RANGE_HEAD(4), 0x18, 0x1f, TLV_DB_SCALE_ITEM(-7450, 400, 0), @@ -225,6 +243,7 @@ static int max9850_set_dai_fmt(struct snd_soc_dai *codec_dai, unsigned int fmt) static int max9850_set_bias_level(struct snd_soc_codec *codec, enum snd_soc_bias_level level) { + struct max9850_priv *max9850 = snd_soc_codec_get_drvdata(codec); int ret; switch (level) { @@ -234,7 +253,7 @@ static int max9850_set_bias_level(struct snd_soc_codec *codec, break; case SND_SOC_BIAS_STANDBY: if (codec->dapm.bias_level == SND_SOC_BIAS_OFF) { - ret = snd_soc_cache_sync(codec); + ret = regcache_sync(max9850->regmap); if (ret) { dev_err(codec->dev, "Failed to sync cache: %d\n", ret); @@ -295,7 +314,7 @@ static int max9850_probe(struct snd_soc_codec *codec) { int ret; - ret = snd_soc_codec_set_cache_io(codec, 8, 8, SND_SOC_I2C); + ret = snd_soc_codec_set_cache_io(codec, 8, 8, SND_SOC_REGMAP); if (ret < 0) { dev_err(codec->dev, "Failed to set cache I/O: %d\n", ret); return ret; @@ -316,10 +335,6 @@ static struct snd_soc_codec_driver soc_codec_dev_max9850 = { .suspend = max9850_suspend, .resume = max9850_resume, .set_bias_level = max9850_set_bias_level, - .reg_cache_size = ARRAY_SIZE(max9850_reg), - .reg_word_size = sizeof(u8), - .reg_cache_default = max9850_reg, - .volatile_register = max9850_volatile_register, .controls = max9850_controls, .num_controls = ARRAY_SIZE(max9850_controls), @@ -340,6 +355,10 @@ static int max9850_i2c_probe(struct i2c_client *i2c, if (max9850 == NULL) return -ENOMEM; + max9850->regmap = devm_regmap_init_i2c(i2c, &max9850_regmap); + if (IS_ERR(max9850->regmap)) + return PTR_ERR(max9850->regmap); + i2c_set_clientdata(i2c, max9850); ret = snd_soc_register_codec(&i2c->dev, -- 1.8.4.rc3

1 0

[alsa-devel] [PATCH 1/3] ASoC: max98088: Fix indentation
by Mark Brown 23 Sep '13

23 Sep '13

From: Mark Brown <broonie(a)linaro.org> Signed-off-by: Mark Brown <broonie(a)linaro.org> --- sound/soc/codecs/max98088.c | 46 ++++++++++++++++++++++----------------------- 1 file changed, 23 insertions(+), 23 deletions(-) diff --git a/sound/soc/codecs/max98088.c b/sound/soc/codecs/max98088.c index 566a367..391f669 100644 --- a/sound/soc/codecs/max98088.c +++ b/sound/soc/codecs/max98088.c @@ -38,20 +38,20 @@ struct max98088_cdata { }; struct max98088_priv { - enum max98088_type devtype; - struct max98088_pdata *pdata; - unsigned int sysclk; - struct max98088_cdata dai[2]; - int eq_textcnt; - const char **eq_texts; - struct soc_enum eq_enum; - u8 ina_state; - u8 inb_state; - unsigned int ex_mode; - unsigned int digmic; - unsigned int mic1pre; - unsigned int mic2pre; - unsigned int extmic_mode; + enum max98088_type devtype; + struct max98088_pdata *pdata; + unsigned int sysclk; + struct max98088_cdata dai[2]; + int eq_textcnt; + const char **eq_texts; + struct soc_enum eq_enum; + u8 ina_state; + u8 inb_state; + unsigned int ex_mode; + unsigned int digmic; + unsigned int mic1pre; + unsigned int mic2pre; + unsigned int extmic_mode; }; static const u8 max98088_reg[M98088_REG_CNT] = { @@ -2066,15 +2066,15 @@ static int max98088_remove(struct snd_soc_codec *codec) } static struct snd_soc_codec_driver soc_codec_dev_max98088 = { - .probe = max98088_probe, - .remove = max98088_remove, - .suspend = max98088_suspend, - .resume = max98088_resume, - .set_bias_level = max98088_set_bias_level, - .reg_cache_size = ARRAY_SIZE(max98088_reg), - .reg_word_size = sizeof(u8), - .reg_cache_default = max98088_reg, - .volatile_register = max98088_volatile_register, + .probe = max98088_probe, + .remove = max98088_remove, + .suspend = max98088_suspend, + .resume = max98088_resume, + .set_bias_level = max98088_set_bias_level, + .reg_cache_size = ARRAY_SIZE(max98088_reg), + .reg_word_size = sizeof(u8), + .reg_cache_default = max98088_reg, + .volatile_register = max98088_volatile_register, .dapm_widgets = max98088_dapm_widgets, .num_dapm_widgets = ARRAY_SIZE(max98088_dapm_widgets), .dapm_routes = max98088_audio_map, -- 1.8.4.rc3

1 2

Re: [alsa-devel] [PATCH 24/51] DMA-API: dma: pl330: add dma_set_mask_and_coherent() call
by Russell King - ARM Linux 23 Sep '13

23 Sep '13

On Fri, Sep 20, 2013 at 07:26:27PM +0200, Heiko Stübner wrote: > Am Donnerstag, 19. September 2013, 23:49:01 schrieb Russell King: > > The DMA API requires drivers to call the appropriate dma_set_mask() > > functions before doing any DMA mapping. Add this required call to > > the AMBA PL08x driver. > ^--- copy and paste error - should of course be PL330 Fixed, thanks.

2 1

[alsa-devel] fsl_ssi: Cannot do simultaneous capture and playback
by Fabio Estevam 23 Sep '13

23 Sep '13

Hi, I am trying to do simultaneous capture and playback on mx6 wandboard and I noticed that it only works if I run 'aplay first'. Here is the sequence that works: 1) aplay file.wav 2) Run 'alsamixer' and select line-in in the capture tab 3) Connect audio via line-in input 4) arecord -f cd -c 2 | aplay -f cd Then I am able to hear the playback. However, if I do not do step 1, and try to do the arecord/aplay directly I get: $ arecord -f cd | aplay -f cd imx-sgtl5000 sound.13: set sample size in capture stream first fsl-ssi-dai 2028000.ssi: ASoC: can't open interface 2028000.ssi: -11 ALSA lib pcm_dmix.c:1018:(snd_pcm_dmix_ open) unable to open slave aplay: main:660: audio open error: Device or resource busy Recording WAVE 'stdin' : Signed 16 bit Little Endian, Rate 44100 Hz, Stereo I will try to debug this, but if anyone has any suggestions, please let me know. Thanks, Fabio Estevam

2 5

[alsa-devel] Buffer size for ALSA USB PCM audio
by Alan Stern 22 Sep '13

22 Sep '13

I have been studying the data_ep_set_params() function in sound/usb/endpoint.c. This is the routine that calculates the number of samples and I/O requests to keep on the USB hardware queue for PCM audio, based on the ALSA parameters. It uses the PERIOD_BYTES parameter but not BUFFER_BYTES. In simplified terms (ignoring rounding, boundary cases, and other things), the number of periods per buffer is fixed at 24 for recording and 1 for playback, completely ignoring the user's setting. If you look at the parameters copied in snd_usb_hw_params() in sound/usb/pcm.c, you'll see what I mean. Is this really the intended behavior? It doesn't seem right at all. Alan Stern

7 44

[alsa-devel] [PATCH] ASoC: bells: Specify REFCLK for FLLs
by Charles Keepax 22 Sep '13

22 Sep '13

Wolfson recommends using a higher frequency REFCLK for best performance rather than the default 32kHz reference clock source. We should set a good example and do so. Signed-off-by: Charles Keepax <ckeepax(a)opensource.wolfsonmicro.com> --- sound/soc/samsung/bells.c | 18 ++++++++++++++++++ 1 files changed, 18 insertions(+), 0 deletions(-) diff --git a/sound/soc/samsung/bells.c b/sound/soc/samsung/bells.c index 84f5d8b..89ad1d7 100644 --- a/sound/soc/samsung/bells.c +++ b/sound/soc/samsung/bells.c @@ -181,6 +181,24 @@ static int bells_late_probe(struct snd_soc_card *card) return ret; } + ret = snd_soc_codec_set_pll(codec, WM5102_FLL1_REFCLK, + ARIZONA_FLL_SRC_MCLK1, + MCLK_RATE, + bells->sysclk_rate); + if (ret != 0) { + dev_err(codec->dev, "Failed to set REFCLK for FLL1: %d\n", ret); + return ret; + } + + ret = snd_soc_codec_set_pll(codec, WM5102_FLL2_REFCLK, + ARIZONA_FLL_SRC_MCLK1, + MCLK_RATE, + bells->asyncclk_rate); + if (ret != 0) { + dev_err(codec->dev, "Failed to set REFCLK for FLL2: %d\n", ret); + return ret; + } + aif2_dai = card->rtd[DAI_CODEC_CP].cpu_dai; ret = snd_soc_dai_set_sysclk(aif2_dai, ARIZONA_CLK_ASYNCCLK, 0, 0); -- 1.7.2.5

2 3

[alsa-devel] [PATCH v2 1/2] ASoC: kirkwood: clk: probe defer when clock not yet ready
by Jean-Francois Moine 21 Sep '13

21 Sep '13

At probe time, a clock device may not be ready when some other device wants to use it. This patch lets the functions clk_get/devm_clk_get return a probe defer when the clock is defined in the DT but not yet available. Signed-off-by: Jean-Francois Moine <moinejf(a)free.fr> --- drivers/clk/clk.c | 2 +- drivers/clk/clkdev.c | 9 +++++++-- 2 files changed, 8 insertions(+), 3 deletions(-) diff --git a/drivers/clk/clk.c b/drivers/clk/clk.c index a004769..582abc8 100644 --- a/drivers/clk/clk.c +++ b/drivers/clk/clk.c @@ -2181,7 +2181,7 @@ EXPORT_SYMBOL_GPL(of_clk_del_provider); struct clk *of_clk_get_from_provider(struct of_phandle_args *clkspec) { struct of_clk_provider *provider; - struct clk *clk = ERR_PTR(-ENOENT); + struct clk *clk = ERR_PTR(-EPROBE_DEFER); /* Check if we have such a provider in our array */ mutex_lock(&of_clk_lock); diff --git a/drivers/clk/clkdev.c b/drivers/clk/clkdev.c index 442a313..f8b28d9 100644 --- a/drivers/clk/clkdev.c +++ b/drivers/clk/clkdev.c @@ -157,8 +157,13 @@ struct clk *clk_get(struct device *dev, const char *con_id) if (dev) { clk = of_clk_get_by_name(dev->of_node, con_id); - if (!IS_ERR(clk) && __clk_get(clk)) - return clk; + if (!IS_ERR(clk)) { + if (__clk_get(clk)) + return clk; + } else { + if (PTR_ERR(clk) == -EPROBE_DEFER) + return clk; + } } return clk_get_sys(dev_id, con_id); -- Ken ar c'hentañ | ** Breizh ha Linux atav! ** Jef | http://moinejf.free.fr/

2 4

[alsa-devel] [PATCH v2 2/2] ASoC: kirkwood: fix loss of external clock at probe time
by Jean-Francois Moine 21 Sep '13

21 Sep '13

At probe time, when the clock driver is not yet initialized, the external clock of the kirkwood sound device will not be usable. This patch fixes this problem defering the device probe. Signed-off-by: Jean-Francois Moine <moinejf(a)free.fr> --- sound/soc/kirkwood/kirkwood-i2s.c | 5 ++++- 1 file changed, 4 insertions(+), 1 deletion(-) diff --git a/sound/soc/kirkwood/kirkwood-i2s.c b/sound/soc/kirkwood/kirkwood-i2s.c index 0f3d73d..14f32d9 100644 --- a/sound/soc/kirkwood/kirkwood-i2s.c +++ b/sound/soc/kirkwood/kirkwood-i2s.c @@ -496,7 +496,10 @@ static int kirkwood_i2s_dev_probe(struct platform_device *pdev) return err; priv->extclk = devm_clk_get(&pdev->dev, "extclk"); - if (!IS_ERR(priv->extclk)) { + if (IS_ERR(priv->extclk)) { + if (PTR_ERR(priv->extclk) == -EPROBE_DEFER) + return -EPROBE_DEFER; + } else { if (priv->extclk == priv->clk) { devm_clk_put(&pdev->dev, priv->extclk); priv->extclk = ERR_PTR(-EINVAL); -- Ken ar c'hentañ | ** Breizh ha Linux atav! ** Jef | http://moinejf.free.fr/

2 1

[alsa-devel] [PATCH] ASoC: adav80x: Convert to direct regmap API usage
by Mark Brown 21 Sep '13

21 Sep '13

From: Mark Brown <broonie(a)linaro.org> As part of a move towards eliminating the ASoC level I/O code convert to use regmap directly. Signed-off-by: Mark Brown <broonie(a)linaro.org> --- sound/soc/codecs/adav80x.c | 244 ++++++++++++++++++++++++++++++++++----------- 1 file changed, 186 insertions(+), 58 deletions(-) diff --git a/sound/soc/codecs/adav80x.c b/sound/soc/codecs/adav80x.c index 15b012d0..5d9e913 100644 --- a/sound/soc/codecs/adav80x.c +++ b/sound/soc/codecs/adav80x.c @@ -115,22 +115,146 @@ #define ADAV80X_PLL_OUTE_SYSCLKPD(x) BIT(2 - (x)) -static u8 adav80x_default_regs[] = { - 0x00, 0x00, 0x00, 0x00, 0x01, 0x01, 0x02, 0x01, 0x80, 0x26, 0x00, 0x00, - 0x02, 0x40, 0x20, 0x00, 0x09, 0x08, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, - 0x04, 0x00, 0x01, 0x00, 0x00, 0x00, 0x00, 0x00, 0xd1, 0x92, 0xb1, 0x37, - 0x48, 0xd2, 0xfb, 0xca, 0xd2, 0x15, 0xe8, 0x29, 0xb9, 0x6a, 0xda, 0x2b, - 0xb7, 0xc0, 0x11, 0x65, 0x5c, 0xf6, 0xff, 0x8d, 0x00, 0x00, 0x00, 0x00, - 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, - 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0xa5, 0x00, 0x00, - 0x00, 0xe8, 0x46, 0xe1, 0x5b, 0xd3, 0x43, 0x77, 0x93, 0xa7, 0x44, 0xee, - 0x32, 0x12, 0xc0, 0x11, 0x00, 0x00, 0x00, 0x00, 0xff, 0xff, 0x3f, 0x3f, - 0x00, 0x00, 0x00, 0x00, 0xff, 0xff, 0x00, 0x1d, 0x00, 0x00, 0x00, 0x00, - 0x00, 0x00, 0x00, 0x00, 0x52, 0x00, +static const struct reg_default adav80x_default_regs[] = { + { 0, 0x00 }, + { 1, 0x00 }, + { 2, 0x00 }, + { 3, 0x00 }, + { 4, 0x01 }, + { 5, 0x01 }, + { 6, 0x02 }, + { 7, 0x01 }, + { 8, 0x80 }, + { 9, 0x26 }, + { 10, 0x00 }, + { 11, 0x00 }, + { 12, 0x02 }, + { 13, 0x40 }, + { 14, 0x20 }, + { 15, 0x00 }, + { 16, 0x09 }, + { 17, 0x08 }, + { 18, 0x00 }, + { 19, 0x00 }, + { 20, 0x00 }, + { 21, 0x00 }, + { 22, 0x00 }, + { 23, 0x00 }, + { 24, 0x04 }, + { 25, 0x00 }, + { 26, 0x01 }, + { 27, 0x00 }, + { 28, 0x00 }, + { 29, 0x00 }, + { 30, 0x00 }, + { 31, 0x00 }, + { 32, 0xd1 }, + { 33, 0x92 }, + { 34, 0xb1 }, + { 35, 0x37 }, + { 36, 0x48 }, + { 37, 0xd2 }, + { 38, 0xfb }, + { 39, 0xca }, + { 40, 0xd2 }, + { 41, 0x15 }, + { 42, 0xe8 }, + { 43, 0x29 }, + { 44, 0xb9 }, + { 45, 0x6a }, + { 46, 0xda }, + { 47, 0x2b }, + { 48, 0xb7 }, + { 49, 0xc0 }, + { 50, 0x11 }, + { 51, 0x65 }, + { 52, 0x5c }, + { 53, 0xf6 }, + { 54, 0xff }, + { 55, 0x8d }, + { 56, 0x00 }, + { 57, 0x00 }, + { 58, 0x00 }, + { 59, 0x00 }, + { 60, 0x00 }, + { 61, 0x00 }, + { 62, 0x00 }, + { 63, 0x00 }, + { 64, 0x00 }, + { 65, 0x00 }, + { 66, 0x00 }, + { 67, 0x00 }, + { 68, 0x00 }, + { 69, 0x00 }, + { 70, 0x00 }, + { 71, 0x00 }, + { 72, 0x00 }, + { 73, 0x00 }, + { 74, 0x00 }, + { 75, 0x00 }, + { 76, 0x00 }, + { 77, 0x00 }, + { 78, 0x00 }, + { 79, 0x00 }, + { 80, 0x00 }, + { 81, 0xa5 }, + { 82, 0x00 }, + { 83, 0x00 }, + { 84, 0x00 }, + { 85, 0xe8 }, + { 86, 0x46 }, + { 87, 0xe1 }, + { 88, 0x5b }, + { 89, 0xd3 }, + { 90, 0x43 }, + { 91, 0x77 }, + { 92, 0x93 }, + { 93, 0xa7 }, + { 94, 0x44 }, + { 95, 0xee }, + { 96, 0x32 }, + { 97, 0x12 }, + { 98, 0xc0 }, + { 99, 0x11 }, + { 100, 0x00 }, + { 101, 0x00 }, + { 102, 0x00 }, + { 103, 0x00 }, + { 104, 0xff }, + { 105, 0xff }, + { 106, 0x3f }, + { 107, 0x3f }, + { 108, 0x00 }, + { 109, 0x00 }, + { 110, 0x00 }, + { 111, 0x00 }, + { 112, 0xff }, + { 113, 0xff }, + { 114, 0x00 }, + { 115, 0x1d }, + { 116, 0x00 }, + { 117, 0x00 }, + { 118, 0x00 }, + { 119, 0x00 }, + { 120, 0x00 }, + { 121, 0x00 }, + { 122, 0x00 }, + { 123, 0x00 }, + { 124, 0x52 }, + { 125, 0x00 }, +}; + +static const struct regmap_config adav80x_regmap = { + .reg_bits = 7, + .val_bits = 9, + + .reg_defaults = adav80x_default_regs, + .num_reg_defaults = ARRAY_SIZE(adav80x_default_regs), + .max_register = ARRAY_SIZE(adav80x_default_regs) - 1, }; struct adav80x { - enum snd_soc_control_type control_type; + struct regmap *regmap; enum adav80x_clk_src clk_src; unsigned int sysclk; @@ -777,14 +901,9 @@ static struct snd_soc_dai_driver adav80x_dais[] = { static int adav80x_probe(struct snd_soc_codec *codec) { - int ret; struct adav80x *adav80x = snd_soc_codec_get_drvdata(codec); - ret = snd_soc_codec_set_cache_io(codec, 7, 9, adav80x->control_type); - if (ret) { - dev_err(codec->dev, "failed to set cache I/O: %d\n", ret); - return ret; - } + codec->control_data = adav80x->regmap; /* Force PLLs on for SYSCLK output */ snd_soc_dapm_force_enable_pin(&codec->dapm, "PLL1"); @@ -800,14 +919,25 @@ static int adav80x_probe(struct snd_soc_codec *codec) static int adav80x_suspend(struct snd_soc_codec *codec) { - return adav80x_set_bias_level(codec, SND_SOC_BIAS_OFF); + struct adav80x *adav80x = snd_soc_codec_get_drvdata(codec); + int ret; + + ret = adav80x_set_bias_level(codec, SND_SOC_BIAS_OFF); + + regcache_cache_only(adav80x->regmap, true); + + return ret; } static int adav80x_resume(struct snd_soc_codec *codec) { + struct adav80x *adav80x = snd_soc_codec_get_drvdata(codec); + + regcache_cache_only(adav80x->regmap, false); + adav80x_set_bias_level(codec, SND_SOC_BIAS_STANDBY); - codec->cache_sync = 1; - snd_soc_cache_sync(codec); + + regcache_sync(adav80x->regmap); return 0; } @@ -827,10 +957,6 @@ static struct snd_soc_codec_driver adav80x_codec_driver = { .set_pll = adav80x_set_pll, .set_sysclk = adav80x_set_sysclk, - .reg_word_size = sizeof(u8), - .reg_cache_size = ARRAY_SIZE(adav80x_default_regs), - .reg_cache_default = adav80x_default_regs, - .controls = adav80x_controls, .num_controls = ARRAY_SIZE(adav80x_controls), .dapm_widgets = adav80x_dapm_widgets, @@ -839,34 +965,6 @@ static struct snd_soc_codec_driver adav80x_codec_driver = { .num_dapm_routes = ARRAY_SIZE(adav80x_dapm_routes), }; -static int adav80x_bus_probe(struct device *dev, - enum snd_soc_control_type control_type) -{ - struct adav80x *adav80x; - int ret; - - adav80x = kzalloc(sizeof(*adav80x), GFP_KERNEL); - if (!adav80x) - return -ENOMEM; - - dev_set_drvdata(dev, adav80x); - adav80x->control_type = control_type; - - ret = snd_soc_register_codec(dev, &adav80x_codec_driver, - adav80x_dais, ARRAY_SIZE(adav80x_dais)); - if (ret) - kfree(adav80x); - - return ret; -} - -static int adav80x_bus_remove(struct device *dev) -{ - snd_soc_unregister_codec(dev); - kfree(dev_get_drvdata(dev)); - return 0; -} - #if defined(CONFIG_SPI_MASTER) static const struct spi_device_id adav80x_spi_id[] = { { "adav801", 0 }, @@ -876,12 +974,27 @@ MODULE_DEVICE_TABLE(spi, adav80x_spi_id); static int adav80x_spi_probe(struct spi_device *spi) { - return adav80x_bus_probe(&spi->dev, SND_SOC_SPI); + struct adav80x *adav80x; + + adav80x = devm_kzalloc(&spi->dev, sizeof(*adav80x), GFP_KERNEL); + if (!adav80x) + return -ENOMEM; + + adav80x->regmap = devm_regmap_init_spi(spi, &adav80x_regmap); + if (IS_ERR(adav80x->regmap)) + return PTR_ERR(adav80x->regmap); + + dev_set_drvdata(&spi->dev, adav80x); + + return snd_soc_register_codec(&spi->dev, &adav80x_codec_driver, + adav80x_dais, ARRAY_SIZE(adav80x_dais)); } static int adav80x_spi_remove(struct spi_device *spi) { - return adav80x_bus_remove(&spi->dev); + snd_soc_unregister_codec(&spi->dev); + + return 0; } static struct spi_driver adav80x_spi_driver = { @@ -905,12 +1018,27 @@ MODULE_DEVICE_TABLE(i2c, adav80x_i2c_id); static int adav80x_i2c_probe(struct i2c_client *client, const struct i2c_device_id *id) { - return adav80x_bus_probe(&client->dev, SND_SOC_I2C); + struct adav80x *adav80x; + + adav80x = devm_kzalloc(&client->dev, sizeof(*adav80x), GFP_KERNEL); + if (!adav80x) + return -ENOMEM; + + adav80x->regmap = devm_regmap_init_i2c(client, &adav80x_regmap); + if (IS_ERR(adav80x->regmap)) + return PTR_ERR(adav80x->regmap); + + dev_set_drvdata(&client->dev, adav80x); + + return snd_soc_register_codec(&client->dev, &adav80x_codec_driver, + adav80x_dais, ARRAY_SIZE(adav80x_dais)); } static int adav80x_i2c_remove(struct i2c_client *client) { - return adav80x_bus_remove(&client->dev); + snd_soc_unregister_codec(&client->dev); + + return 0; } static struct i2c_driver adav80x_i2c_driver = { -- 1.8.4.rc3

2 4

[alsa-devel] [PATCH v2 0/2] ASoC: kirkwood: fix loss of external clock at probe time
by Jean-Francois Moine 21 Sep '13

21 Sep '13

This patch series fixes the loss of clock when the requested clock device is not yet initialized at probe time. The first patch adds a -EPROBE_DEFER when the clock referenced by a phandle is not available. The second patch uses this feature to fix such a problem in the kirkwood audio driver. v2: - return the probe defer from the clock subsystem (Marc Brown remark) -- Ken ar c'hentañ | ** Breizh ha Linux atav! ** Jef | http://moinejf.free.fr/

1 0

[alsa-devel] hdmi audio on Shuttle XS35GS V3L
by Olaf Hering 20 Sep '13

20 Sep '13

Does anyone happen to know if HDMI audio is supposed to work on the Shuttle XS35GS V3L? I have tried an AOC E2260Pwhu monitor and also a Philips 22PFL3805H/12 TV, both show the picture but no sound is played via HDMI. OS is openSuSE 12.3 and 13.1beta1. Below is dmesg and /proc/asound output. Olaf ==> /proc/asound/card0/codec#0 <== Codec: IDT 92HD87B1/3 Address: 0 AFG Function Id: 0x1 (unsol 1) Vendor Id: 0x111d76d1 Subsystem Id: 0x12974012 Revision Id: 0x100107 No Modem Function Group found Default PCM: rates [0x5e0]: 44100 48000 88200 96000 192000 bits [0xe]: 16 20 24 ==> /proc/asound/card0/codec#1 <== Codec: Intel CedarTrail HDMI Address: 1 AFG Function Id: 0x1 (unsol 0) Vendor Id: 0x80862880 Subsystem Id: 0x80860101 Revision Id: 0x100000 No Modem Function Group found Default PCM: rates [0x0]: bits [0x0]: ==> /proc/asound/card0/eld#1.0 <== monitor_present 0 eld_valid 0 ==> /proc/asound/card0/id <== Intel ==> /proc/asound/card0/pcm0c/info <== card: 0 device: 0 subdevice: 0 stream: CAPTURE id: 92HD87B1/3 Analog name: 92HD87B1/3 Analog subname: subdevice #0 class: 0 subclass: 0 subdevices_count: 1 ==> /proc/asound/card0/pcm0c/sub0/hw_params <== closed ==> /proc/asound/card0/pcm0c/sub0/info <== card: 0 device: 0 subdevice: 0 stream: CAPTURE id: 92HD87B1/3 Analog name: 92HD87B1/3 Analog subname: subdevice #0 class: 0 subclass: 0 subdevices_count: 1 ==> /proc/asound/card0/pcm0c/sub0/prealloc <== 1024 ==> /proc/asound/card0/pcm0c/sub0/prealloc_max <== 32768 ==> /proc/asound/card0/pcm0c/sub0/status <== closed ==> /proc/asound/card0/pcm0c/sub0/sw_params <== closed ==> /proc/asound/card0/pcm0c/xrun_debug <== 0 ==> /proc/asound/card0/pcm0p/info <== card: 0 device: 0 subdevice: 0 stream: PLAYBACK id: 92HD87B1/3 Analog name: 92HD87B1/3 Analog subname: subdevice #0 class: 0 subclass: 0 subdevices_count: 1 ==> /proc/asound/card0/pcm0p/sub0/hw_params <== closed ==> /proc/asound/card0/pcm0p/sub0/info <== card: 0 device: 0 subdevice: 0 stream: PLAYBACK id: 92HD87B1/3 Analog name: 92HD87B1/3 Analog subname: subdevice #0 class: 0 subclass: 0 subdevices_count: 1 ==> /proc/asound/card0/pcm0p/sub0/prealloc <== 1024 ==> /proc/asound/card0/pcm0p/sub0/prealloc_max <== 32768 ==> /proc/asound/card0/pcm0p/sub0/status <== closed ==> /proc/asound/card0/pcm0p/sub0/sw_params <== closed ==> /proc/asound/card0/pcm0p/xrun_debug <== 0 ==> /proc/asound/card0/pcm3p/info <== card: 0 device: 3 subdevice: 0 stream: PLAYBACK id: HDMI 0 name: HDMI 0 subname: subdevice #0 class: 0 subclass: 0 subdevices_count: 1 ==> /proc/asound/card0/pcm3p/sub0/hw_params <== closed ==> /proc/asound/card0/pcm3p/sub0/info <== card: 0 device: 3 subdevice: 0 stream: PLAYBACK id: HDMI 0 name: HDMI 0 subname: subdevice #0 class: 0 subclass: 0 subdevices_count: 1 ==> /proc/asound/card0/pcm3p/sub0/prealloc <== 1024 ==> /proc/asound/card0/pcm3p/sub0/prealloc_max <== 32768 ==> /proc/asound/card0/pcm3p/sub0/status <== closed ==> /proc/asound/card0/pcm3p/sub0/sw_params <== closed ==> /proc/asound/card0/pcm3p/xrun_debug <== 0 ==> /proc/asound/card1/codec#0 <== Codec: ATI R6xx HDMI Address: 0 AFG Function Id: 0x1 (unsol 0) Vendor Id: 0x1002aa01 Subsystem Id: 0x00aa0100 Revision Id: 0x100200 No Modem Function Group found Default PCM: rates [0x70]: 32000 44100 48000 bits [0x2]: 16 ==> /proc/asound/card1/eld#0.0 <== monitor_present 0 eld_valid 0 ==> /proc/asound/card1/id <== Generic ==> /proc/asound/card1/pcm3p/info <== card: 1 device: 3 subdevice: 0 stream: PLAYBACK id: HDMI 0 name: HDMI 0 subname: subdevice #0 class: 0 subclass: 0 subdevices_count: 1 ==> /proc/asound/card1/pcm3p/sub0/hw_params <== closed ==> /proc/asound/card1/pcm3p/sub0/info <== card: 1 device: 3 subdevice: 0 stream: PLAYBACK id: HDMI 0 name: HDMI 0 subname: subdevice #0 class: 0 subclass: 0 subdevices_count: 1 ==> /proc/asound/card1/pcm3p/sub0/prealloc <== 1024 ==> /proc/asound/card1/pcm3p/sub0/prealloc_max <== 32768 ==> /proc/asound/card1/pcm3p/sub0/status <== closed ==> /proc/asound/card1/pcm3p/sub0/sw_params <== closed ==> /proc/asound/card1/pcm3p/xrun_debug <== 0 ==> /proc/asound/cards <== 0 [Intel ]: HDA-Intel - HDA Intel HDA Intel at 0xd0300000 irq 47 1 [Generic ]: HDA-Intel - HD-Audio Generic HD-Audio Generic at 0xd0040000 irq 48 ==> /proc/asound/devices <== 1: : sequencer 2: [ 0- 3]: digital audio playback 3: [ 0- 0]: digital audio playback 4: [ 0- 0]: digital audio capture 5: [ 0- 1]: hardware dependent 6: [ 0- 0]: hardware dependent 7: [ 0] : control 8: [ 1- 3]: digital audio playback 9: [ 1- 0]: hardware dependent 10: [ 1] : control ==> /proc/asound/hwdep <== 00-01: HDA Codec 1 00-00: HDA Codec 0 01-00: HDA Codec 0 ==> /proc/asound/modules <== 0 snd_hda_intel 1 snd_hda_intel ==> /proc/asound/oss/devices <== ==> /proc/asound/oss/sndstat <== Sound Driver:3.8.1a-980706 (ALSA emulation code) Kernel: Linux shuttle 3.11.1-1.g1383321-desktop #1 SMP PREEMPT Sat Sep 14 18:49:04 UTC 2013 (1383321) x86_64 Config options: 0 Installed drivers: Type 10: ALSA emulation Card config: HDA Intel at 0xd0300000 irq 47 HD-Audio Generic at 0xd0040000 irq 48 ==> /proc/asound/pcm <== 00-00: 92HD87B1/3 Analog : 92HD87B1/3 Analog : playback 1 : capture 1 00-03: HDMI 0 : HDMI 0 : playback 1 01-03: HDMI 0 : HDMI 0 : playback 1 ==> /proc/asound/seq/clients <== Client info cur clients : 2 peak clients : 2 max clients : 192 Client 0 : "System" [Kernel] Port 0 : "Timer" (Rwe-) Port 1 : "Announce" (R-e-) Client 14 : "Midi Through" [Kernel] Port 0 : "Midi Through Port-0" (RWe-) ==> /proc/asound/seq/drivers <== ==> /proc/asound/seq/queues <== ==> /proc/asound/seq/timer <== ==> /proc/asound/timers <== G0: system timer : 1000.000us (10000000 ticks) P0-0-0: PCM playback 0-0-0 : SLAVE P0-0-1: PCM capture 0-0-1 : SLAVE P0-3-0: PCM playback 0-3-0 : SLAVE P1-3-0: PCM playback 1-3-0 : SLAVE ==> /proc/asound/version <== Advanced Linux Sound Architecture Driver Version k3.11.1-1.g1383321-desktop. shuttle:~ # dmesg [ 0.000000] Initializing cgroup subsys cpuset [ 0.000000] Initializing cgroup subsys cpu [ 0.000000] Initializing cgroup subsys cpuacct [ 0.000000] Linux version 3.11.1-1.g1383321-desktop (geeko@buildhost) (gcc version 4.8.1 20130909 [gcc-4_8-branch revision 202388] (SUSE Linux) ) #1 SMP PREEMPT Sat Sep 14 18:49:04 UTC 2013 (1383321) [ 0.000000] Command line: BOOT_IMAGE=/boot/vmlinuz-3.11.1-1.g1383321-desktop root=UUID=b8a2a174-296a-4da2-82e3-9680fadcff42 quiet panic=9 sysrq=yes dhcp=1 resume=/dev/disk/by-id/ata-KINGSTON_SV300S37A60G_50026B723106E9DA-part1 splash=silent quiet showopts [ 0.000000] Disabled fast string operations [ 0.000000] e820: BIOS-provided physical RAM map: [ 0.000000] BIOS-e820: [mem 0x0000000000000000-0x000000000009fbff] usable [ 0.000000] BIOS-e820: [mem 0x000000000009fc00-0x000000000009ffff] reserved [ 0.000000] BIOS-e820: [mem 0x00000000000e0000-0x00000000000fffff] reserved [ 0.000000] BIOS-e820: [mem 0x0000000000100000-0x00000000bf7f3fff] usable [ 0.000000] BIOS-e820: [mem 0x00000000bf7f4000-0x00000000bf877fff] ACPI NVS [ 0.000000] BIOS-e820: [mem 0x00000000bf878000-0x00000000bf888fff] reserved [ 0.000000] BIOS-e820: [mem 0x00000000bf889000-0x00000000bf88ffff] ACPI data [ 0.000000] BIOS-e820: [mem 0x00000000bf890000-0x00000000bf8abfff] reserved [ 0.000000] BIOS-e820: [mem 0x00000000bf8ac000-0x00000000bf8acfff] usable [ 0.000000] BIOS-e820: [mem 0x00000000bf8ad000-0x00000000bf8cafff] reserved [ 0.000000] BIOS-e820: [mem 0x00000000bf8cb000-0x00000000bf8ccfff] usable [ 0.000000] BIOS-e820: [mem 0x00000000bf8cd000-0x00000000bf8dbfff] ACPI NVS [ 0.000000] BIOS-e820: [mem 0x00000000bf8dc000-0x00000000bf901fff] reserved [ 0.000000] BIOS-e820: [mem 0x00000000bf902000-0x00000000bf944fff] ACPI NVS [ 0.000000] BIOS-e820: [mem 0x00000000bf945000-0x00000000bfd1afff] usable [ 0.000000] BIOS-e820: [mem 0x00000000bfd1b000-0x00000000bfee4fff] reserved [ 0.000000] BIOS-e820: [mem 0x00000000bfee5000-0x00000000bfeeffff] usable [ 0.000000] BIOS-e820: [mem 0x00000000bfef0000-0x00000000bfffffff] reserved [ 0.000000] BIOS-e820: [mem 0x00000000e0000000-0x00000000efffffff] reserved [ 0.000000] BIOS-e820: [mem 0x00000000fec00000-0x00000000fec00fff] reserved [ 0.000000] BIOS-e820: [mem 0x00000000fed00000-0x00000000fed00fff] reserved [ 0.000000] BIOS-e820: [mem 0x00000000fed1c000-0x00000000fed8ffff] reserved [ 0.000000] BIOS-e820: [mem 0x00000000fee00000-0x00000000fee00fff] reserved [ 0.000000] BIOS-e820: [mem 0x00000000ffe00000-0x00000000ffffffff] reserved [ 0.000000] BIOS-e820: [mem 0x0000000100000000-0x000000013fffffff] usable [ 0.000000] NX (Execute Disable) protection: active [ 0.000000] SMBIOS 2.7 present. [ 0.000000] DMI: Shuttle Inc. XS35V3/XS35V3, BIOS 1.11 12/13/2012 [ 0.000000] e820: update [mem 0x00000000-0x00000fff] usable ==> reserved [ 0.000000] e820: remove [mem 0x000a0000-0x000fffff] usable [ 0.000000] No AGP bridge found [ 0.000000] e820: last_pfn = 0x140000 max_arch_pfn = 0x400000000 [ 0.000000] MTRR default type: uncachable [ 0.000000] MTRR fixed ranges enabled: [ 0.000000] 00000-9FFFF write-back [ 0.000000] A0000-BFFFF uncachable [ 0.000000] C0000-E7FFF write-through [ 0.000000] E8000-FFFFF write-protect [ 0.000000] MTRR variable ranges enabled: [ 0.000000] 0 base 000000000 mask F00000000 write-back [ 0.000000] 1 base 100000000 mask FC0000000 write-back [ 0.000000] 2 base 0BFF00000 mask FFFF00000 write-through [ 0.000000] 3 base 0C0000000 mask FC0000000 uncachable [ 0.000000] 4 disabled [ 0.000000] 5 disabled [ 0.000000] 6 disabled [ 0.000000] x86 PAT enabled: cpu 0, old 0x7040600070406, new 0x7010600070106 [ 0.000000] e820: last_pfn = 0xbfef0 max_arch_pfn = 0x400000000 [ 0.000000] found SMP MP-table at [mem 0x000fc9c0-0x000fc9cf] mapped at [ffff8800000fc9c0] [ 0.000000] Scanning 1 areas for low memory corruption [ 0.000000] Base memory trampoline at [ffff880000099000] 99000 size 24576 [ 0.000000] init_memory_mapping: [mem 0x00000000-0x000fffff] [ 0.000000] [mem 0x00000000-0x000fffff] page 4k [ 0.000000] BRK [0x01fb3000, 0x01fb3fff] PGTABLE [ 0.000000] BRK [0x01fb4000, 0x01fb4fff] PGTABLE [ 0.000000] BRK [0x01fb5000, 0x01fb5fff] PGTABLE [ 0.000000] init_memory_mapping: [mem 0x13fe00000-0x13fffffff] [ 0.000000] [mem 0x13fe00000-0x13fffffff] page 2M [ 0.000000] BRK [0x01fb6000, 0x01fb6fff] PGTABLE [ 0.000000] init_memory_mapping: [mem 0x13c000000-0x13fdfffff] [ 0.000000] [mem 0x13c000000-0x13fdfffff] page 2M [ 0.000000] init_memory_mapping: [mem 0x100000000-0x13bffffff] [ 0.000000] [mem 0x100000000-0x13bffffff] page 2M [ 0.000000] init_memory_mapping: [mem 0x00100000-0xbf7f3fff] [ 0.000000] [mem 0x00100000-0x001fffff] page 4k [ 0.000000] [mem 0x00200000-0xbf5fffff] page 2M [ 0.000000] [mem 0xbf600000-0xbf7f3fff] page 4k [ 0.000000] init_memory_mapping: [mem 0xbf8ac000-0xbf8acfff] [ 0.000000] [mem 0xbf8ac000-0xbf8acfff] page 4k [ 0.000000] BRK [0x01fb7000, 0x01fb7fff] PGTABLE [ 0.000000] init_memory_mapping: [mem 0xbf8cb000-0xbf8ccfff] [ 0.000000] [mem 0xbf8cb000-0xbf8ccfff] page 4k [ 0.000000] init_memory_mapping: [mem 0xbf945000-0xbfd1afff] [ 0.000000] [mem 0xbf945000-0xbf9fffff] page 4k [ 0.000000] [mem 0xbfa00000-0xbfbfffff] page 2M [ 0.000000] [mem 0xbfc00000-0xbfd1afff] page 4k [ 0.000000] BRK [0x01fb8000, 0x01fb8fff] PGTABLE [ 0.000000] init_memory_mapping: [mem 0xbfee5000-0xbfeeffff] [ 0.000000] [mem 0xbfee5000-0xbfeeffff] page 4k [ 0.000000] RAMDISK: [mem 0x34eee000-0x3676efff] [ 0.000000] ACPI: RSDP 00000000000f0450 00024 (v02 Shuttl) [ 0.000000] ACPI: XSDT 00000000bf889078 0006C (v01 Shuttl Shuttle 01072009 AMI 00010013) [ 0.000000] ACPI: FACP 00000000bf88ea08 000F4 (v04 Shuttl Shuttle 01072009 AMI 00010013) [ 0.000000] ACPI: DSDT 00000000bf889170 05894 (v02 Shuttl XS35V300 00000000 INTL 20051117) [ 0.000000] ACPI: FACS 00000000bf8d9f80 00040 [ 0.000000] ACPI: APIC 00000000bf88eb00 00072 (v03 Shuttl Shuttle 01072009 AMI 00010013) [ 0.000000] ACPI: MCFG 00000000bf88eb78 0003C (v01 Shuttl Shuttle 01072009 MSFT 00000097) [ 0.000000] ACPI: SLIC 00000000bf88ebb8 00176 (v01 Shuttl Shuttle 01072009 AMI 00010013) [ 0.000000] ACPI: HPET 00000000bf88ed30 00038 (v01 Shuttl Shuttle 01072009 AMI. 00000005) [ 0.000000] ACPI: SSDT 00000000bf88ed68 00655 (v01 PmRef CpuPm 00003000 INTL 20051117) [ 0.000000] ACPI: SSDT 00000000bf88f3c0 00259 (v01 PmRef Cpu0Tst 00003000 INTL 20051117) [ 0.000000] ACPI: SSDT 00000000bf88f620 0020F (v01 PmRef ApTst 00003000 INTL 20051117) [ 0.000000] ACPI: IFEU 00000000bf88f830 00042 (v01 Shuttl Shuttle 01072009 00000000) [ 0.000000] ACPI: Local APIC address 0xfee00000 [ 0.000000] No NUMA configuration found [ 0.000000] Faking a node at [mem 0x0000000000000000-0x000000013fffffff] [ 0.000000] Initmem setup node 0 [mem 0x00000000-0x13fffffff] [ 0.000000] NODE_DATA [mem 0x13ffe8000-0x13fffbfff] [ 0.000000] [ffffea0000000000-ffffea00045fffff] PMD -> [ffff88013b600000-ffff88013edfffff] on node 0 [ 0.000000] Zone ranges: [ 0.000000] DMA [mem 0x00001000-0x00ffffff] [ 0.000000] DMA32 [mem 0x01000000-0xffffffff] [ 0.000000] Normal [mem 0x100000000-0x13fffffff] [ 0.000000] Movable zone start for each node [ 0.000000] Early memory node ranges [ 0.000000] node 0: [mem 0x00001000-0x0009efff] [ 0.000000] node 0: [mem 0x00100000-0xbf7f3fff] [ 0.000000] node 0: [mem 0xbf8ac000-0xbf8acfff] [ 0.000000] node 0: [mem 0xbf8cb000-0xbf8ccfff] [ 0.000000] node 0: [mem 0xbf945000-0xbfd1afff] [ 0.000000] node 0: [mem 0xbfee5000-0xbfeeffff] [ 0.000000] node 0: [mem 0x100000000-0x13fffffff] [ 0.000000] On node 0 totalpages: 1047414 [ 0.000000] DMA zone: 56 pages used for memmap [ 0.000000] DMA zone: 21 pages reserved [ 0.000000] DMA zone: 3998 pages, LIFO batch:0 [ 0.000000] DMA32 zone: 10682 pages used for memmap [ 0.000000] DMA32 zone: 781272 pages, LIFO batch:31 [ 0.000000] Normal zone: 3584 pages used for memmap [ 0.000000] Normal zone: 262144 pages, LIFO batch:31 [ 0.000000] ACPI: PM-Timer IO Port: 0x408 [ 0.000000] ACPI: Local APIC address 0xfee00000 [ 0.000000] ACPI: LAPIC (acpi_id[0x01] lapic_id[0x00] enabled) [ 0.000000] ACPI: LAPIC (acpi_id[0x02] lapic_id[0x01] enabled) [ 0.000000] ACPI: LAPIC (acpi_id[0x03] lapic_id[0x02] enabled) [ 0.000000] ACPI: LAPIC (acpi_id[0x04] lapic_id[0x03] enabled) [ 0.000000] ACPI: LAPIC_NMI (acpi_id[0xff] high edge lint[0x1]) [ 0.000000] ACPI: IOAPIC (id[0x04] address[0xfec00000] gsi_base[0]) [ 0.000000] IOAPIC[0]: apic_id 4, version 32, address 0xfec00000, GSI 0-23 [ 0.000000] ACPI: INT_SRC_OVR (bus 0 bus_irq 0 global_irq 2 dfl dfl) [ 0.000000] ACPI: INT_SRC_OVR (bus 0 bus_irq 9 global_irq 9 high level) [ 0.000000] ACPI: IRQ0 used by override. [ 0.000000] ACPI: IRQ2 used by override. [ 0.000000] ACPI: IRQ9 used by override. [ 0.000000] Using ACPI (MADT) for SMP configuration information [ 0.000000] ACPI: HPET id: 0x8086a201 base: 0xfed00000 [ 0.000000] smpboot: Allowing 4 CPUs, 0 hotplug CPUs [ 0.000000] nr_irqs_gsi: 40 [ 0.000000] PM: Registered nosave memory: [mem 0x0009f000-0x0009ffff] [ 0.000000] PM: Registered nosave memory: [mem 0x000a0000-0x000dffff] [ 0.000000] PM: Registered nosave memory: [mem 0x000e0000-0x000fffff] [ 0.000000] PM: Registered nosave memory: [mem 0xbf7f4000-0xbf877fff] [ 0.000000] PM: Registered nosave memory: [mem 0xbf878000-0xbf888fff] [ 0.000000] PM: Registered nosave memory: [mem 0xbf889000-0xbf88ffff] [ 0.000000] PM: Registered nosave memory: [mem 0xbf890000-0xbf8abfff] [ 0.000000] PM: Registered nosave memory: [mem 0xbf8ad000-0xbf8cafff] [ 0.000000] PM: Registered nosave memory: [mem 0xbf8cd000-0xbf8dbfff] [ 0.000000] PM: Registered nosave memory: [mem 0xbf8dc000-0xbf901fff] [ 0.000000] PM: Registered nosave memory: [mem 0xbf902000-0xbf944fff] [ 0.000000] PM: Registered nosave memory: [mem 0xbfd1b000-0xbfee4fff] [ 0.000000] PM: Registered nosave memory: [mem 0xbfef0000-0xbfffffff] [ 0.000000] PM: Registered nosave memory: [mem 0xc0000000-0xdfffffff] [ 0.000000] PM: Registered nosave memory: [mem 0xe0000000-0xefffffff] [ 0.000000] PM: Registered nosave memory: [mem 0xf0000000-0xfebfffff] [ 0.000000] PM: Registered nosave memory: [mem 0xfec00000-0xfec00fff] [ 0.000000] PM: Registered nosave memory: [mem 0xfec01000-0xfecfffff] [ 0.000000] PM: Registered nosave memory: [mem 0xfed00000-0xfed00fff] [ 0.000000] PM: Registered nosave memory: [mem 0xfed01000-0xfed1bfff] [ 0.000000] PM: Registered nosave memory: [mem 0xfed1c000-0xfed8ffff] [ 0.000000] PM: Registered nosave memory: [mem 0xfed90000-0xfedfffff] [ 0.000000] PM: Registered nosave memory: [mem 0xfee00000-0xfee00fff] [ 0.000000] PM: Registered nosave memory: [mem 0xfee01000-0xffdfffff] [ 0.000000] PM: Registered nosave memory: [mem 0xffe00000-0xffffffff] [ 0.000000] e820: [mem 0xc0000000-0xdfffffff] available for PCI devices [ 0.000000] Booting paravirtualized kernel on bare hardware [ 0.000000] setup_percpu: NR_CPUS:512 nr_cpumask_bits:512 nr_cpu_ids:4 nr_node_ids:1 [ 0.000000] PERCPU: Embedded 28 pages/cpu @ffff88013fc00000 s83584 r8192 d22912 u524288 [ 0.000000] pcpu-alloc: s83584 r8192 d22912 u524288 alloc=1*2097152 [ 0.000000] pcpu-alloc: [0] 0 1 2 3 [ 0.000000] Built 1 zonelists in Node order, mobility grouping on. Total pages: 1033071 [ 0.000000] Policy zone: Normal [ 0.000000] Kernel command line: BOOT_IMAGE=/boot/vmlinuz-3.11.1-1.g1383321-desktop root=UUID=b8a2a174-296a-4da2-82e3-9680fadcff42 quiet panic=9 sysrq=yes dhcp=1 resume=/dev/disk/by-id/ata-KINGSTON_SV300S37A60G_50026B723106E9DA-part1 splash=silent quiet showopts [ 0.000000] PID hash table entries: 4096 (order: 3, 32768 bytes) [ 0.000000] Checking aperture... [ 0.000000] No AGP bridge found [ 0.000000] Memory: 4024188K/4189656K available (5868K kernel code, 778K rwdata, 4368K rodata, 1332K init, 1640K bss, 165468K reserved) [ 0.000000] Preemptible hierarchical RCU implementation. [ 0.000000] RCU dyntick-idle grace-period acceleration is enabled. [ 0.000000] RCU restricting CPUs from NR_CPUS=512 to nr_cpu_ids=4. [ 0.000000] Offload RCU callbacks from all CPUs [ 0.000000] Offload RCU callbacks from CPUs: 0-511. [ 0.000000] NR_IRQS:33024 nr_irqs:712 16 [ 0.000000] spurious 8259A interrupt: IRQ7. [ 0.000000] Console: colour dummy device 80x25 [ 0.000000] console [tty0] enabled [ 0.000000] allocated 16777216 bytes of page_cgroup [ 0.000000] please try 'cgroup_disable=memory' option if you don't want memory cgroups [ 0.000000] hpet clockevent registered [ 0.000000] tsc: Fast TSC calibration using PIT [ 0.001000] tsc: Detected 1861.931 MHz processor [ 0.000006] Calibrating delay loop (skipped), value calculated using timer frequency.. 3723.86 BogoMIPS (lpj=1861931) [ 0.000012] pid_max: default: 32768 minimum: 301 [ 0.000105] Security Framework initialized [ 0.000129] AppArmor: AppArmor initialized [ 0.000790] Dentry cache hash table entries: 524288 (order: 10, 4194304 bytes) [ 0.004142] Inode-cache hash table entries: 262144 (order: 9, 2097152 bytes) [ 0.005662] Mount-cache hash table entries: 256 [ 0.006164] Initializing cgroup subsys memory [ 0.006197] Initializing cgroup subsys devices [ 0.006203] Initializing cgroup subsys freezer [ 0.006207] Initializing cgroup subsys net_cls [ 0.006211] Initializing cgroup subsys blkio [ 0.006215] Initializing cgroup subsys perf_event [ 0.006222] Initializing cgroup subsys hugetlb [ 0.006263] Disabled fast string operations [ 0.006271] CPU: Physical Processor ID: 0 [ 0.006274] CPU: Processor Core ID: 0 [ 0.006278] mce: CPU supports 5 MCE banks [ 0.006289] CPU0: Thermal monitoring enabled (TM1) [ 0.006301] Last level iTLB entries: 4KB 32, 2MB 0, 4MB 0 Last level dTLB entries: 4KB 64, 2MB 0, 4MB 8 tlb_flushall_shift: 6 [ 0.006426] Freeing SMP alternatives memory: 24K (ffffffff81e11000 - ffffffff81e17000) [ 0.007539] ACPI: Core revision 20130517 [ 0.020130] ACPI: All ACPI Tables successfully acquired [ 0.021223] ..TIMER: vector=0x30 apic1=0 pin1=2 apic2=0 pin2=0 [ 0.031231] smpboot: CPU0: Intel(R) Atom(TM) CPU D2550 @ 1.86GHz (fam: 06, model: 36, stepping: 01) [ 0.132916] Performance Events: PEBS fmt0+, 8-deep LBR, Atom events, Intel PMU driver. [ 0.132929] ... version: 3 [ 0.132932] ... bit width: 40 [ 0.132935] ... generic registers: 2 [ 0.132938] ... value mask: 000000ffffffffff [ 0.132941] ... max period: 000000007fffffff [ 0.132943] ... fixed-purpose events: 3 [ 0.132946] ... event mask: 0000000700000003 [ 0.155179] Disabled fast string operations [ 0.157493] NMI watchdog: enabled on all CPUs, permanently consumes one hw-PMU counter. [ 0.170159] Disabled fast string operations [ 0.144138] smpboot: Booting Node 0, Processors #1 #2 #3 OK [ 0.185160] Disabled fast string operations [ 0.187336] Brought up 4 CPUs [ 0.187345] smpboot: Total of 4 processors activated (14895.44 BogoMIPS) [ 0.188501] devtmpfs: initialized [ 0.192382] PM: Registering ACPI NVS region [mem 0xbf7f4000-0xbf877fff] (540672 bytes) [ 0.192426] PM: Registering ACPI NVS region [mem 0xbf8cd000-0xbf8dbfff] (61440 bytes) [ 0.192434] PM: Registering ACPI NVS region [mem 0xbf902000-0xbf944fff] (274432 bytes) [ 0.192811] RTC time: 14:10:10, date: 09/19/13 [ 0.192959] NET: Registered protocol family 16 [ 0.193372] ACPI: bus type PCI registered [ 0.193379] acpiphp: ACPI Hot Plug PCI Controller Driver version: 0.5 [ 0.193516] PCI: MMCONFIG for domain 0000 [bus 00-ff] at [mem 0xe0000000-0xefffffff] (base 0xe0000000) [ 0.193523] PCI: MMCONFIG at [mem 0xe0000000-0xefffffff] reserved in E820 [ 0.229808] PCI: Using configuration type 1 for base access [ 0.232752] bio: create slab <bio-0> at 0 [ 0.233147] ACPI: Added _OSI(Module Device) [ 0.233154] ACPI: Added _OSI(Processor Device) [ 0.233158] ACPI: Added _OSI(3.0 _SCP Extensions) [ 0.233163] ACPI: Added _OSI(Processor Aggregator Device) [ 0.235600] ACPI: EC: Look up EC in DSDT [ 0.238399] ACPI: Executed 1 blocks of module-level executable AML code [ 0.248370] [Firmware Bug]: ACPI: BIOS _OSI(Linux) query ignored [ 0.252986] ACPI: Interpreter enabled [ 0.253014] ACPI Exception: AE_NOT_FOUND, While evaluating Sleep State [\_S2_] (20130517/hwxface-571) [ 0.253053] ACPI: (supports S0 S1 S3 S4 S5) [ 0.253057] ACPI: Using IOAPIC for interrupt routing [ 0.253158] PCI: Using host bridge windows from ACPI; if necessary, use "pci=nocrs" and report a bug [ 0.253496] ACPI: No dock devices found. [ 0.273701] ACPI: Power Resource [FN00] (off) [ 0.275547] ACPI: PCI Root Bridge [PCI0] (domain 0000 [bus 00-ff]) [ 0.275734] \_SB_.PCI0:_OSC invalid UUID [ 0.275739] _OSC request data:1 8 0 [ 0.276882] PCI host bridge to bus 0000:00 [ 0.276891] pci_bus 0000:00: root bus resource [bus 00-ff] [ 0.276898] pci_bus 0000:00: root bus resource [io 0x0000-0x0cf7] [ 0.276903] pci_bus 0000:00: root bus resource [io 0x0d00-0xffff] [ 0.276920] pci_bus 0000:00: root bus resource [mem 0x000a0000-0x000bffff] [ 0.276925] pci_bus 0000:00: root bus resource [mem 0x000c0000-0x000dffff] [ 0.276931] pci_bus 0000:00: root bus resource [mem 0x000e0000-0x000effff] [ 0.276936] pci_bus 0000:00: root bus resource [mem 0x000f0000-0x000fffff] [ 0.276941] pci_bus 0000:00: root bus resource [mem 0xc0000000-0xfebfffff] [ 0.276958] pci 0000:00:00.0: [8086:0bf3] type 00 class 0x060000 [ 0.277240] pci 0000:00:1b.0: [8086:27d8] type 00 class 0x040300 [ 0.277274] pci 0000:00:1b.0: reg 0x10: [mem 0xd0300000-0xd0303fff 64bit] [ 0.277404] pci 0000:00:1b.0: PME# supported from D0 D3hot D3cold [ 0.277619] pci 0000:00:1c.0: [8086:27d0] type 01 class 0x060400 [ 0.277757] pci 0000:00:1c.0: PME# supported from D0 D3hot D3cold [ 0.277901] pci 0000:00:1c.0: System wakeup disabled by ACPI [ 0.278003] pci 0000:00:1c.1: [8086:27d2] type 01 class 0x060400 [ 0.278139] pci 0000:00:1c.1: PME# supported from D0 D3hot D3cold [ 0.278286] pci 0000:00:1c.1: System wakeup disabled by ACPI [ 0.278383] pci 0000:00:1c.3: [8086:27d6] type 01 class 0x060400 [ 0.278518] pci 0000:00:1c.3: PME# supported from D0 D3hot D3cold [ 0.278660] pci 0000:00:1c.3: System wakeup disabled by ACPI [ 0.278754] pci 0000:00:1d.0: [8086:27c8] type 00 class 0x0c0300 [ 0.278827] pci 0000:00:1d.0: reg 0x20: [io 0xf0a0-0xf0bf] [ 0.279052] pci 0000:00:1d.0: System wakeup disabled by ACPI [ 0.279145] pci 0000:00:1d.1: [8086:27c9] type 00 class 0x0c0300 [ 0.279216] pci 0000:00:1d.1: reg 0x20: [io 0xf080-0xf09f] [ 0.279434] pci 0000:00:1d.1: System wakeup disabled by ACPI [ 0.279522] pci 0000:00:1d.2: [8086:27ca] type 00 class 0x0c0300 [ 0.279593] pci 0000:00:1d.2: reg 0x20: [io 0xf060-0xf07f] [ 0.279801] pci 0000:00:1d.2: System wakeup disabled by ACPI [ 0.279888] pci 0000:00:1d.3: [8086:27cb] type 00 class 0x0c0300 [ 0.279969] pci 0000:00:1d.3: reg 0x20: [io 0xf040-0xf05f] [ 0.280177] pci 0000:00:1d.3: System wakeup disabled by ACPI [ 0.280279] pci 0000:00:1d.7: [8086:27cc] type 00 class 0x0c0320 [ 0.280314] pci 0000:00:1d.7: reg 0x10: [mem 0xd0305000-0xd03053ff] [ 0.280444] pci 0000:00:1d.7: PME# supported from D0 D3hot D3cold [ 0.280579] pci 0000:00:1d.7: System wakeup disabled by ACPI [ 0.280673] pci 0000:00:1e.0: [8086:2448] type 01 class 0x060401 [ 0.280893] pci 0000:00:1e.0: System wakeup disabled by ACPI [ 0.280993] pci 0000:00:1f.0: [8086:27bc] type 00 class 0x060100 [ 0.281327] pci 0000:00:1f.2: [8086:27c1] type 00 class 0x010601 [ 0.281357] pci 0000:00:1f.2: reg 0x10: [io 0xf0f0-0xf0f7] [ 0.281373] pci 0000:00:1f.2: reg 0x14: [io 0xf0e0-0xf0e3] [ 0.281391] pci 0000:00:1f.2: reg 0x18: [io 0xf0d0-0xf0d7] [ 0.281407] pci 0000:00:1f.2: reg 0x1c: [io 0xf0c0-0xf0c3] [ 0.281425] pci 0000:00:1f.2: reg 0x20: [io 0xf020-0xf02f] [ 0.281442] pci 0000:00:1f.2: reg 0x24: [mem 0xd0304000-0xd03043ff] [ 0.281516] pci 0000:00:1f.2: PME# supported from D3hot [ 0.281716] pci 0000:00:1f.3: [8086:27da] type 00 class 0x0c0500 [ 0.281788] pci 0000:00:1f.3: reg 0x20: [io 0xf000-0xf01f] [ 0.282172] pci 0000:01:00.0: [1002:6761] type 00 class 0x030000 [ 0.282208] pci 0000:01:00.0: reg 0x10: [mem 0xc0000000-0xcfffffff 64bit pref] [ 0.282235] pci 0000:01:00.0: reg 0x18: [mem 0xd0020000-0xd003ffff 64bit] [ 0.282254] pci 0000:01:00.0: reg 0x20: [io 0xe000-0xe0ff] [ 0.282286] pci 0000:01:00.0: reg 0x30: [mem 0xd0000000-0xd001ffff pref] [ 0.282367] pci 0000:01:00.0: supports D1 D2 [ 0.282425] pci 0000:01:00.0: System wakeup disabled by ACPI [ 0.282527] pci 0000:01:00.1: [1002:aa98] type 00 class 0x040300 [ 0.282562] pci 0000:01:00.1: reg 0x10: [mem 0xd0040000-0xd0043fff 64bit] [ 0.282707] pci 0000:01:00.1: supports D1 D2 [ 0.283939] pci 0000:00:1c.0: PCI bridge to [bus 01] [ 0.283949] pci 0000:00:1c.0: bridge window [io 0xe000-0xefff] [ 0.283958] pci 0000:00:1c.0: bridge window [mem 0xc0000000-0xd00fffff] [ 0.284221] acpiphp: Slot [1] registered [ 0.284298] pci 0000:02:00.0: [10ec:5289] type 00 class 0xff0000 [ 0.284326] pci 0000:02:00.0: reg 0x10: [mem 0xd0200000-0xd020ffff] [ 0.284496] pci 0000:02:00.0: supports D1 D2 [ 0.284501] pci 0000:02:00.0: PME# supported from D1 D2 D3hot D3cold [ 0.284583] pci 0000:02:00.0: System wakeup disabled by ACPI [ 0.284695] pci 0000:02:00.2: [10ec:8168] type 00 class 0x020000 [ 0.284721] pci 0000:02:00.2: reg 0x10: [io 0xd000-0xd0ff] [ 0.284761] pci 0000:02:00.2: reg 0x18: [mem 0xd0214000-0xd0214fff 64bit pref] [ 0.284788] pci 0000:02:00.2: reg 0x20: [mem 0xd0210000-0xd0213fff 64bit pref] [ 0.284891] pci 0000:02:00.2: supports D1 D2 [ 0.284896] pci 0000:02:00.2: PME# supported from D0 D1 D2 D3hot D3cold [ 0.286942] pci 0000:00:1c.1: PCI bridge to [bus 02] [ 0.286952] pci 0000:00:1c.1: bridge window [io 0xd000-0xdfff] [ 0.286960] pci 0000:00:1c.1: bridge window [mem 0xd0200000-0xd02fffff] [ 0.287110] pci 0000:00:1c.3: PCI bridge to [bus 03] [ 0.287120] pci 0000:00:1c.3: bridge window [io 0xc000-0xcfff] [ 0.287128] pci 0000:00:1c.3: bridge window [mem 0xd0100000-0xd01fffff] [ 0.287280] pci 0000:00:1e.0: PCI bridge to [bus 04] (subtractive decode) [ 0.287302] pci 0000:00:1e.0: bridge window [io 0x0000-0x0cf7] (subtractive decode) [ 0.287308] pci 0000:00:1e.0: bridge window [io 0x0d00-0xffff] (subtractive decode) [ 0.287314] pci 0000:00:1e.0: bridge window [mem 0x000a0000-0x000bffff] (subtractive decode) [ 0.287319] pci 0000:00:1e.0: bridge window [mem 0x000c0000-0x000dffff] (subtractive decode) [ 0.287325] pci 0000:00:1e.0: bridge window [mem 0x000e0000-0x000effff] (subtractive decode) [ 0.287330] pci 0000:00:1e.0: bridge window [mem 0x000f0000-0x000fffff] (subtractive decode) [ 0.287335] pci 0000:00:1e.0: bridge window [mem 0xc0000000-0xfebfffff] (subtractive decode) [ 0.287573] \_SB_.PCI0:_OSC invalid UUID [ 0.287577] _OSC request data:1 1f 0 [ 0.287587] acpi PNP0A08:00: ACPI _OSC support notification failed, disabling PCIe ASPM [ 0.287592] acpi PNP0A08:00: Unable to request _OSC control (_OSC support mask: 0x08) [ 0.288565] ACPI: PCI Interrupt Link [LNKA] (IRQs 1 3 4 5 6 7 *10 12 14 15) [ 0.288824] ACPI: PCI Interrupt Link [LNKB] (IRQs 1 3 4 *5 6 7 11 12 14 15) [ 0.289089] ACPI: PCI Interrupt Link [LNKC] (IRQs 1 3 *4 5 6 7 10 12 14 15) [ 0.289350] ACPI: PCI Interrupt Link [LNKD] (IRQs 1 *3 4 5 6 7 11 12 14 15) [ 0.289606] ACPI: PCI Interrupt Link [LNKE] (IRQs 1 3 4 5 6 7 10 12 14 15) *0, disabled. [ 0.289861] ACPI: PCI Interrupt Link [LNKF] (IRQs 1 3 4 5 6 7 11 12 14 15) *0, disabled. [ 0.290128] ACPI: PCI Interrupt Link [LNKG] (IRQs 1 3 4 5 6 7 10 12 14 15) *11 [ 0.290389] ACPI: PCI Interrupt Link [LNKH] (IRQs 1 3 4 5 6 *7 11 12 14 15) [ 0.290937] ACPI: Enabled 3 GPEs in block 00 to 1F [ 0.290957] ACPI: \_SB_.PCI0: notify handler is installed [ 0.291065] Found 1 acpi root devices [ 0.291266] vgaarb: device added: PCI:0000:01:00.0,decodes=io+mem,owns=io+mem,locks=none [ 0.291272] vgaarb: loaded [ 0.291275] vgaarb: bridge control possible 0000:01:00.0 [ 0.291535] SCSI subsystem initialized [ 0.291541] ACPI: bus type ATA registered [ 0.291682] libata version 3.00 loaded. [ 0.291701] ACPI: bus type USB registered [ 0.291762] usbcore: registered new interface driver usbfs [ 0.291797] usbcore: registered new interface driver hub [ 0.291875] usbcore: registered new device driver usb [ 0.292087] PCI: Using ACPI for IRQ routing [ 0.304952] PCI: pci_cache_line_size set to 64 bytes [ 0.305047] e820: reserve RAM buffer [mem 0x0009fc00-0x0009ffff] [ 0.305053] e820: reserve RAM buffer [mem 0xbf7f4000-0xbfffffff] [ 0.305059] e820: reserve RAM buffer [mem 0xbf8ad000-0xbfffffff] [ 0.305063] e820: reserve RAM buffer [mem 0xbf8cd000-0xbfffffff] [ 0.305068] e820: reserve RAM buffer [mem 0xbfd1b000-0xbfffffff] [ 0.305072] e820: reserve RAM buffer [mem 0xbfef0000-0xbfffffff] [ 0.305297] NetLabel: Initializing [ 0.305302] NetLabel: domain hash size = 128 [ 0.305304] NetLabel: protocols = UNLABELED CIPSOv4 [ 0.305332] NetLabel: unlabeled traffic allowed by default [ 0.305430] hpet0: at MMIO 0xfed00000, IRQs 2, 8, 0 [ 0.305439] hpet0: 3 comparators, 64-bit 14.318180 MHz counter [ 0.307505] Switched to clocksource hpet [ 0.313499] AppArmor: AppArmor Filesystem Enabled [ 0.313595] pnp: PnP ACPI init [ 0.313626] ACPI: bus type PNP registered [ 0.313714] pnp 00:00: [dma 4] [ 0.313775] pnp 00:00: Plug and Play ACPI device, IDs PNP0200 (active) [ 0.313852] pnp 00:01: Plug and Play ACPI device, IDs INT0800 (active) [ 0.314234] system 00:02: [mem 0xfed00000-0xfed003ff] has been reserved [ 0.314243] system 00:02: Plug and Play ACPI device, IDs PNP0103 PNP0c01 (active) [ 0.314344] pnp 00:03: Plug and Play ACPI device, IDs PNP0c04 (active) [ 0.314465] system 00:04: [io 0x0680-0x069f] has been reserved [ 0.314472] system 00:04: [io 0x1000-0x100f] has been reserved [ 0.314478] system 00:04: [io 0xffff] has been reserved [ 0.314484] system 00:04: [io 0xffff] has been reserved [ 0.314491] system 00:04: [io 0x0400-0x047f] could not be reserved [ 0.314497] system 00:04: [io 0x0500-0x057f] has been reserved [ 0.314531] system 00:04: [io 0x0600-0x061f] has been reserved [ 0.314539] system 00:04: Plug and Play ACPI device, IDs PNP0c02 (active) [ 0.314712] system 00:05: [io 0x06a0-0x06af] has been reserved [ 0.314719] system 00:05: [io 0x06b0-0x06ff] has been reserved [ 0.314726] system 00:05: Plug and Play ACPI device, IDs PNP0c02 (active) [ 0.314813] pnp 00:06: Plug and Play ACPI device, IDs PNP0b00 (active) [ 0.315318] system 00:07: [io 0x0a00-0x0a0f] has been reserved [ 0.315326] system 00:07: [io 0x0a10-0x0a1f] has been reserved [ 0.315333] system 00:07: Plug and Play ACPI device, IDs PNP0c02 (active) [ 0.315528] system 00:08: [io 0x04d0-0x04d1] has been reserved [ 0.315536] system 00:08: Plug and Play ACPI device, IDs PNP0c02 (active) [ 0.316026] system 00:09: [mem 0xfed1c000-0xfed1ffff] has been reserved [ 0.316034] system 00:09: [mem 0x00000000-0x00003fff] could not be reserved [ 0.316040] system 00:09: [mem 0x00000000-0x00000fff] could not be reserved [ 0.316046] system 00:09: [mem 0x00000000-0x00000fff] could not be reserved [ 0.316052] system 00:09: [mem 0xfed45000-0xfed8ffff] has been reserved [ 0.316059] system 00:09: Plug and Play ACPI device, IDs PNP0c02 (active) [ 0.316847] system 00:0a: [mem 0xfed14000-0xfed19fff] has been reserved [ 0.316855] system 00:0a: [mem 0xe0000000-0xefffffff] has been reserved [ 0.316862] system 00:0a: Plug and Play ACPI device, IDs PNP0c01 (active) [ 0.316948] pnp 00:0b: Plug and Play ACPI device, IDs INT0800 (active) [ 0.317650] system 00:0c: [io 0x0400-0x047f] could not be reserved [ 0.317658] system 00:0c: [io 0x0500-0x053f] has been reserved [ 0.317665] system 00:0c: [mem 0xfec00000-0xfec00fff] could not be reserved [ 0.317672] system 00:0c: [mem 0xfee00000-0xfee00fff] has been reserved [ 0.317678] system 00:0c: [mem 0xfed1c000-0xfed1ffff] has been reserved [ 0.317684] system 00:0c: [mem 0xfed20000-0xfed8ffff] could not be reserved [ 0.317690] system 00:0c: [mem 0xffe00000-0xffffffff] has been reserved [ 0.317697] system 00:0c: Plug and Play ACPI device, IDs PNP0c02 (active) [ 0.317707] pnp: PnP ACPI: found 13 devices [ 0.317710] ACPI: bus type PNP unregistered [ 0.328034] pci 0000:00:1c.0: bridge window [mem 0x00100000-0x000fffff 64bit pref] to [bus 01] add_size 200000 [ 0.328057] pci 0000:00:1c.1: bridge window [mem 0x00100000-0x000fffff 64bit pref] to [bus 02] add_size 200000 [ 0.328075] pci 0000:00:1c.3: bridge window [mem 0x00100000-0x000fffff 64bit pref] to [bus 03] add_size 200000 [ 0.328105] pci 0000:00:1c.0: res[15]=[mem 0x00100000-0x000fffff 64bit pref] get_res_add_size add_size 200000 [ 0.328112] pci 0000:00:1c.1: res[15]=[mem 0x00100000-0x000fffff 64bit pref] get_res_add_size add_size 200000 [ 0.328117] pci 0000:00:1c.3: res[15]=[mem 0x00100000-0x000fffff 64bit pref] get_res_add_size add_size 200000 [ 0.328130] pci 0000:00:1c.0: BAR 15: assigned [mem 0xd0400000-0xd05fffff 64bit pref] [ 0.328139] pci 0000:00:1c.1: BAR 15: assigned [mem 0xd0600000-0xd07fffff 64bit pref] [ 0.328147] pci 0000:00:1c.3: BAR 15: assigned [mem 0xd0800000-0xd09fffff 64bit pref] [ 0.328154] pci 0000:00:1c.0: PCI bridge to [bus 01] [ 0.328162] pci 0000:00:1c.0: bridge window [io 0xe000-0xefff] [ 0.328173] pci 0000:00:1c.0: bridge window [mem 0xc0000000-0xd00fffff] [ 0.328182] pci 0000:00:1c.0: bridge window [mem 0xd0400000-0xd05fffff 64bit pref] [ 0.328194] pci 0000:00:1c.1: PCI bridge to [bus 02] [ 0.328201] pci 0000:00:1c.1: bridge window [io 0xd000-0xdfff] [ 0.328211] pci 0000:00:1c.1: bridge window [mem 0xd0200000-0xd02fffff] [ 0.328219] pci 0000:00:1c.1: bridge window [mem 0xd0600000-0xd07fffff 64bit pref] [ 0.328231] pci 0000:00:1c.3: PCI bridge to [bus 03] [ 0.328238] pci 0000:00:1c.3: bridge window [io 0xc000-0xcfff] [ 0.328248] pci 0000:00:1c.3: bridge window [mem 0xd0100000-0xd01fffff] [ 0.328256] pci 0000:00:1c.3: bridge window [mem 0xd0800000-0xd09fffff 64bit pref] [ 0.328269] pci 0000:00:1e.0: PCI bridge to [bus 04] [ 0.328746] pci 0000:00:1e.0: setting latency timer to 64 [ 0.328756] pci_bus 0000:00: resource 4 [io 0x0000-0x0cf7] [ 0.328761] pci_bus 0000:00: resource 5 [io 0x0d00-0xffff] [ 0.328766] pci_bus 0000:00: resource 6 [mem 0x000a0000-0x000bffff] [ 0.328771] pci_bus 0000:00: resource 7 [mem 0x000c0000-0x000dffff] [ 0.328776] pci_bus 0000:00: resource 8 [mem 0x000e0000-0x000effff] [ 0.328781] pci_bus 0000:00: resource 9 [mem 0x000f0000-0x000fffff] [ 0.328786] pci_bus 0000:00: resource 10 [mem 0xc0000000-0xfebfffff] [ 0.328791] pci_bus 0000:01: resource 0 [io 0xe000-0xefff] [ 0.328796] pci_bus 0000:01: resource 1 [mem 0xc0000000-0xd00fffff] [ 0.328801] pci_bus 0000:01: resource 2 [mem 0xd0400000-0xd05fffff 64bit pref] [ 0.328806] pci_bus 0000:02: resource 0 [io 0xd000-0xdfff] [ 0.328811] pci_bus 0000:02: resource 1 [mem 0xd0200000-0xd02fffff] [ 0.328816] pci_bus 0000:02: resource 2 [mem 0xd0600000-0xd07fffff 64bit pref] [ 0.328822] pci_bus 0000:03: resource 0 [io 0xc000-0xcfff] [ 0.328826] pci_bus 0000:03: resource 1 [mem 0xd0100000-0xd01fffff] [ 0.328832] pci_bus 0000:03: resource 2 [mem 0xd0800000-0xd09fffff 64bit pref] [ 0.328837] pci_bus 0000:04: resource 4 [io 0x0000-0x0cf7] [ 0.328842] pci_bus 0000:04: resource 5 [io 0x0d00-0xffff] [ 0.328847] pci_bus 0000:04: resource 6 [mem 0x000a0000-0x000bffff] [ 0.328851] pci_bus 0000:04: resource 7 [mem 0x000c0000-0x000dffff] [ 0.328856] pci_bus 0000:04: resource 8 [mem 0x000e0000-0x000effff] [ 0.328861] pci_bus 0000:04: resource 9 [mem 0x000f0000-0x000fffff] [ 0.328866] pci_bus 0000:04: resource 10 [mem 0xc0000000-0xfebfffff] [ 0.329013] NET: Registered protocol family 2 [ 0.329579] TCP established hash table entries: 32768 (order: 7, 524288 bytes) [ 0.329949] TCP bind hash table entries: 32768 (order: 7, 524288 bytes) [ 0.330267] TCP: Hash tables configured (established 32768 bind 32768) [ 0.330390] TCP: reno registered [ 0.330420] UDP hash table entries: 2048 (order: 4, 65536 bytes) [ 0.330483] UDP-Lite hash table entries: 2048 (order: 4, 65536 bytes) [ 0.330773] NET: Registered protocol family 1 [ 0.332266] pci 0000:01:00.0: Boot video device [ 0.332287] PCI: CLS 64 bytes, default 64 [ 0.332425] Unpacking initramfs... [ 0.907296] Freeing initrd memory: 25092K (ffff880034eee000 - ffff88003676f000) [ 0.907307] PCI-DMA: Using software bounce buffering for IO (SWIOTLB) [ 0.907313] software IO TLB [mem 0xbb7f4000-0xbf7f4000] (64MB) mapped at [ffff8800bb7f4000-ffff8800bf7f3fff] [ 0.908255] microcode: CPU0 sig=0x30661, pf=0x4, revision=0x10d [ 0.908271] microcode: CPU1 sig=0x30661, pf=0x4, revision=0x10d [ 0.908284] microcode: CPU2 sig=0x30661, pf=0x4, revision=0x10d [ 0.908301] microcode: CPU3 sig=0x30661, pf=0x4, revision=0x10d [ 0.908391] microcode: Microcode Update Driver: v2.00 <tigran(a)aivazian.fsnet.co.uk>, Peter Oruba [ 0.908396] Scanning for low memory corruption every 60 seconds [ 0.909133] audit: initializing netlink socket (disabled) [ 0.909169] type=2000 audit(1379599809.803:1): initialized [ 0.948091] HugeTLB registered 2 MB page size, pre-allocated 0 pages [ 0.948801] zbud: loaded [ 0.949199] VFS: Disk quotas dquot_6.5.2 [ 0.949263] Dquot-cache hash table entries: 512 (order 0, 4096 bytes) [ 0.949786] msgmni has been set to 7908 [ 0.950392] Block layer SCSI generic (bsg) driver version 0.4 loaded (major 252) [ 0.950484] io scheduler noop registered [ 0.950490] io scheduler deadline registered [ 0.950533] io scheduler cfq registered (default) [ 0.950742] pcieport 0000:00:1c.0: irq 40 for MSI/MSI-X [ 0.950900] pcieport 0000:00:1c.1: irq 41 for MSI/MSI-X [ 0.951046] pcieport 0000:00:1c.3: irq 42 for MSI/MSI-X [ 0.951216] pci_hotplug: PCI Hot Plug PCI Core version: 0.5 [ 0.951253] pciehp: PCI Express Hot Plug Controller Driver version: 0.4 [ 0.951366] vesafb: mode is 1400x1050x32, linelength=5632, pages=0 [ 0.951369] vesafb: scrolling: redraw [ 0.951374] vesafb: Truecolor: size=0:8:8:8, shift=0:16:8:0 [ 0.952335] vesafb: framebuffer at 0xc0000000, mapped to 0xffffc90010800000, using 5824k, total 5824k [ 1.742876] Console: switching to colour frame buffer device 175x65 [ 1.910348] tsc: Refined TSC clocksource calibration: 1861.999 MHz [ 2.529563] fb0: VESA VGA frame buffer device [ 2.529598] intel_idle: does not run on family 6 model 54 [ 2.529650] GHES: HEST is not enabled! [ 2.529840] Serial: 8250/16550 driver, 32 ports, IRQ sharing disabled [ 2.534053] Non-volatile memory driver v1.3 [ 2.534059] Linux agpgart interface v0.103 [ 2.534365] ahci 0000:00:1f.2: version 3.0 [ 2.534586] ahci 0000:00:1f.2: irq 43 for MSI/MSI-X [ 2.534656] ahci: SSS flag set, parallel bus scan disabled [ 2.534708] ahci 0000:00:1f.2: AHCI 0001.0100 32 slots 4 ports 3 Gbps 0x3 impl SATA mode [ 2.534715] ahci 0000:00:1f.2: flags: 64bit ncq stag pm led clo pio slum part [ 2.534753] ahci 0000:00:1f.2: setting latency timer to 64 [ 2.535805] scsi0 : ahci [ 2.536050] scsi1 : ahci [ 2.536220] scsi2 : ahci [ 2.536396] scsi3 : ahci [ 2.536506] ata1: SATA max UDMA/133 abar m1024@0xd0304000 port 0xd0304100 irq 43 [ 2.536513] ata2: SATA max UDMA/133 abar m1024@0xd0304000 port 0xd0304180 irq 43 [ 2.536517] ata3: DUMMY [ 2.536520] ata4: DUMMY [ 2.536591] ehci_hcd: USB 2.0 'Enhanced' Host Controller (EHCI) Driver [ 2.536597] ehci-pci: EHCI PCI platform driver [ 2.536889] ehci-pci 0000:00:1d.7: setting latency timer to 64 [ 2.536912] ehci-pci 0000:00:1d.7: EHCI Host Controller [ 2.536933] ehci-pci 0000:00:1d.7: new USB bus registered, assigned bus number 1 [ 2.536972] ehci-pci 0000:00:1d.7: debug port 1 [ 2.540913] ehci-pci 0000:00:1d.7: cache line size of 64 is not supported [ 2.540959] ehci-pci 0000:00:1d.7: irq 23, io mem 0xd0305000 [ 2.546747] ehci-pci 0000:00:1d.7: USB 2.0 started, EHCI 1.00 [ 2.546799] usb usb1: New USB device found, idVendor=1d6b, idProduct=0002 [ 2.546805] usb usb1: New USB device strings: Mfr=3, Product=2, SerialNumber=1 [ 2.546809] usb usb1: Product: EHCI Host Controller [ 2.546814] usb usb1: Manufacturer: Linux 3.11.1-1.g1383321-desktop ehci_hcd [ 2.546818] usb usb1: SerialNumber: 0000:00:1d.7 [ 2.547119] hub 1-0:1.0: USB hub found [ 2.547131] hub 1-0:1.0: 8 ports detected [ 2.547892] ohci_hcd: USB 1.1 'Open' Host Controller (OHCI) Driver [ 2.547896] uhci_hcd: USB Universal Host Controller Interface driver [ 2.548104] uhci_hcd 0000:00:1d.0: setting latency timer to 64 [ 2.548112] uhci_hcd 0000:00:1d.0: UHCI Host Controller [ 2.548129] uhci_hcd 0000:00:1d.0: new USB bus registered, assigned bus number 2 [ 2.548176] uhci_hcd 0000:00:1d.0: irq 23, io base 0x0000f0a0 [ 2.548262] usb usb2: New USB device found, idVendor=1d6b, idProduct=0001 [ 2.548267] usb usb2: New USB device strings: Mfr=3, Product=2, SerialNumber=1 [ 2.548272] usb usb2: Product: UHCI Host Controller [ 2.548277] usb usb2: Manufacturer: Linux 3.11.1-1.g1383321-desktop uhci_hcd [ 2.548281] usb usb2: SerialNumber: 0000:00:1d.0 [ 2.548575] hub 2-0:1.0: USB hub found [ 2.548591] hub 2-0:1.0: 2 ports detected [ 2.549013] uhci_hcd 0000:00:1d.1: setting latency timer to 64 [ 2.549021] uhci_hcd 0000:00:1d.1: UHCI Host Controller [ 2.549033] uhci_hcd 0000:00:1d.1: new USB bus registered, assigned bus number 3 [ 2.549105] uhci_hcd 0000:00:1d.1: irq 19, io base 0x0000f080 [ 2.549191] usb usb3: New USB device found, idVendor=1d6b, idProduct=0001 [ 2.549196] usb usb3: New USB device strings: Mfr=3, Product=2, SerialNumber=1 [ 2.549201] usb usb3: Product: UHCI Host Controller [ 2.549206] usb usb3: Manufacturer: Linux 3.11.1-1.g1383321-desktop uhci_hcd [ 2.549210] usb usb3: SerialNumber: 0000:00:1d.1 [ 2.549483] hub 3-0:1.0: USB hub found [ 2.549497] hub 3-0:1.0: 2 ports detected [ 2.549903] uhci_hcd 0000:00:1d.2: setting latency timer to 64 [ 2.549911] uhci_hcd 0000:00:1d.2: UHCI Host Controller [ 2.549923] uhci_hcd 0000:00:1d.2: new USB bus registered, assigned bus number 4 [ 2.549997] uhci_hcd 0000:00:1d.2: irq 18, io base 0x0000f060 [ 2.550081] usb usb4: New USB device found, idVendor=1d6b, idProduct=0001 [ 2.550086] usb usb4: New USB device strings: Mfr=3, Product=2, SerialNumber=1 [ 2.550091] usb usb4: Product: UHCI Host Controller [ 2.550096] usb usb4: Manufacturer: Linux 3.11.1-1.g1383321-desktop uhci_hcd [ 2.550100] usb usb4: SerialNumber: 0000:00:1d.2 [ 2.550378] hub 4-0:1.0: USB hub found [ 2.550392] hub 4-0:1.0: 2 ports detected [ 2.550796] uhci_hcd 0000:00:1d.3: setting latency timer to 64 [ 2.550803] uhci_hcd 0000:00:1d.3: UHCI Host Controller [ 2.550824] uhci_hcd 0000:00:1d.3: new USB bus registered, assigned bus number 5 [ 2.550895] uhci_hcd 0000:00:1d.3: irq 16, io base 0x0000f040 [ 2.550979] usb usb5: New USB device found, idVendor=1d6b, idProduct=0001 [ 2.550985] usb usb5: New USB device strings: Mfr=3, Product=2, SerialNumber=1 [ 2.550989] usb usb5: Product: UHCI Host Controller [ 2.550994] usb usb5: Manufacturer: Linux 3.11.1-1.g1383321-desktop uhci_hcd [ 2.550998] usb usb5: SerialNumber: 0000:00:1d.3 [ 2.551277] hub 5-0:1.0: USB hub found [ 2.551291] hub 5-0:1.0: 2 ports detected [ 2.551629] i8042: PNP: No PS/2 controller found. Probing ports directly. [ 2.840811] ata1: SATA link up 3.0 Gbps (SStatus 123 SControl 300) [ 3.077348] Switched to clocksource tsc [ 3.602621] ata1.00: ATA-8: KINGSTON SV300S37A60G, 505ABBF0, max UDMA/133 [ 3.602627] ata1.00: 117231408 sectors, multi 16: LBA48 NCQ (depth 31/32), AA [ 3.603675] i8042: No controller found [ 3.606784] mousedev: PS/2 mouse device common for all mice [ 3.607032] rtc_cmos 00:06: RTC can wake from S4 [ 3.607277] rtc_cmos 00:06: rtc core: registered rtc_cmos as rtc0 [ 3.607328] rtc_cmos 00:06: alarms up to one month, y3k, 242 bytes nvram, hpet irqs [ 3.607349] cpuidle: using governor ladder [ 3.607353] cpuidle: using governor menu [ 3.607361] ledtrig-cpu: registered to indicate activity on CPUs [ 3.607399] hidraw: raw HID events driver (C) Jiri Kosina [ 3.607611] usbcore: registered new interface driver usbhid [ 3.607615] usbhid: USB HID core driver [ 3.607799] TCP: cubic registered [ 3.607907] NET: Registered protocol family 10 [ 3.608294] Key type dns_resolver registered [ 3.608976] PM: Checking hibernation image partition /dev/disk/by-id/ata-KINGSTON_SV300S37A60G_50026B723106E9DA-part1 [ 3.614537] ata1.00: configured for UDMA/133 [ 3.614794] scsi 0:0:0:0: Direct-Access ATA KINGSTON SV300S3 505A PQ: 0 ANSI: 5 [ 3.615223] sd 0:0:0:0: [sda] 117231408 512-byte logical blocks: (60.0 GB/55.8 GiB) [ 3.615484] sd 0:0:0:0: [sda] Write Protect is off [ 3.615492] sd 0:0:0:0: [sda] Mode Sense: 00 3a 00 00 [ 3.615566] sd 0:0:0:0: [sda] Write cache: enabled, read cache: enabled, doesn't support DPO or FUA [ 3.616644] sda: sda1 sda2 sda3 [ 3.617504] sd 0:0:0:0: [sda] Attached SCSI disk [ 3.919889] ata2: SATA link down (SStatus 0 SControl 300) [ 3.919995] PM: Hibernation image not present or could not be loaded. [ 3.920049] registered taskstats version 1 [ 3.921162] Magic number: 13:546:182 [ 3.921351] rtc_cmos 00:06: setting system clock to 2013-09-19 14:10:13 UTC (1379599813) [ 3.924713] Freeing unused kernel memory: 1332K (ffffffff81cc4000 - ffffffff81e11000) [ 3.924721] Write protecting the kernel read-only data: 12288k [ 3.926920] Freeing unused kernel memory: 264K (ffff8800015be000 - ffff880001600000) [ 3.938318] Freeing unused kernel memory: 1776K (ffff880001a44000 - ffff880001c00000) [ 4.048266] alua: device handler registered [ 4.051640] emc: device handler registered [ 4.054978] hp_sw: device handler registered [ 4.058430] rdac: device handler registered [ 4.059898] usb 2-1: new full-speed USB device number 2 using uhci_hcd [ 4.070181] systemd-udevd[131]: starting version 207 [ 4.134183] thermal LNXTHERM:00: registered as thermal_zone0 [ 4.134196] ACPI: Thermal Zone [TZ01] (27 C) [ 4.134482] ACPI: Fan [FAN0] (off) [ 4.141405] [drm] Initialized drm 1.1.0 20060810 [ 4.156991] [drm] radeon kernel modesetting enabled. [ 4.157158] checking generic (c0000000 5b0000) vs hw (c0000000 10000000) [ 4.157166] fb: conflicting fb hw usage radeondrmfb vs VESA VGA - removing generic driver [ 4.157211] Console: switching to colour dummy device 80x25 [ 4.158551] [drm] initializing kernel modesetting (CAICOS 0x1002:0x6761 0x1297:0x4012). [ 4.158615] [drm] register mmio base: 0xD0020000 [ 4.158621] [drm] register mmio size: 131072 [ 4.158976] ATOM BIOS: Shuttle [ 4.159598] radeon 0000:01:00.0: VRAM: 512M 0x0000000000000000 - 0x000000001FFFFFFF (512M used) [ 4.159609] radeon 0000:01:00.0: GTT: 512M 0x0000000020000000 - 0x000000003FFFFFFF [ 4.159614] [drm] Detected VRAM RAM=512M, BAR=256M [ 4.159618] [drm] RAM width 64bits DDR [ 4.159801] [TTM] Zone kernel: Available graphics memory: 2026338 kiB [ 4.159809] [TTM] Initializing pool allocator [ 4.159823] [TTM] Initializing DMA pool allocator [ 4.159954] [drm] radeon: 512M of VRAM memory ready [ 4.159960] [drm] radeon: 512M of GTT memory ready. [ 4.160922] [drm] GART: num cpu pages 131072, num gpu pages 131072 [ 4.171257] [drm] Loading CAICOS Microcode [ 4.181153] [drm] PCIE GART of 512M enabled (table at 0x0000000000273000). [ 4.181584] radeon 0000:01:00.0: WB enabled [ 4.181597] radeon 0000:01:00.0: fence driver on ring 0 use gpu addr 0x0000000020000c00 and cpu addr 0xffff880135e03c00 [ 4.181607] radeon 0000:01:00.0: fence driver on ring 3 use gpu addr 0x0000000020000c0c and cpu addr 0xffff880135e03c0c [ 4.200521] radeon 0000:01:00.0: fence driver on ring 5 use gpu addr 0x0000000000072118 and cpu addr 0xffffc90010eb2118 [ 4.200534] [drm] Supports vblank timestamp caching Rev 1 (10.10.2010). [ 4.200539] [drm] Driver supports precise vblank timestamp query. [ 4.200611] radeon 0000:01:00.0: irq 44 for MSI/MSI-X [ 4.200650] radeon 0000:01:00.0: radeon: using MSI. [ 4.200746] [drm] radeon: irq initialized. [ 4.214436] usb 2-1: New USB device found, idVendor=045e, idProduct=0745 [ 4.214446] usb 2-1: New USB device strings: Mfr=1, Product=2, SerialNumber=0 [ 4.214454] usb 2-1: Product: Microsoft® 2.4GHz Transceiver v7.0 [ 4.214461] usb 2-1: Manufacturer: Microsoft [ 4.222120] input: Microsoft Microsoft® 2.4GHz Transceiver v7.0 as /devices/pci0000:00/0000:00:1d.0/usb2/2-1/2-1:1.0/input/input0 [ 4.222490] hid-generic 0003:045E:0745.0001: input,hidraw0: USB HID v1.11 Keyboard [Microsoft Microsoft® 2.4GHz Transceiver v7.0] on usb-0000:00:1d.0-1/input0 [ 4.222815] [drm] ring test on 0 succeeded in 1 usecs [ 4.222930] [drm] ring test on 3 succeeded in 1 usecs [ 4.229641] input: Microsoft Microsoft® 2.4GHz Transceiver v7.0 as /devices/pci0000:00/0000:00:1d.0/usb2/2-1/2-1:1.1/input/input1 [ 4.230095] hid-generic 0003:045E:0745.0002: input,hidraw1: USB HID v1.11 Mouse [Microsoft Microsoft® 2.4GHz Transceiver v7.0] on usb-0000:00:1d.0-1/input1 [ 4.255991] input: Microsoft Microsoft® 2.4GHz Transceiver v7.0 as /devices/pci0000:00/0000:00:1d.0/usb2/2-1/2-1:1.2/input/input2 [ 4.256486] hid-generic 0003:045E:0745.0003: input,hiddev0,hidraw2: USB HID v1.11 Device [Microsoft Microsoft® 2.4GHz Transceiver v7.0] on usb-0000:00:1d.0-1/input2 [ 4.280722] xor: measuring software checksum speed [ 4.289905] prefetch64-sse: 4540.000 MB/sec [ 4.301956] generic_sse: 4224.000 MB/sec [ 4.301964] xor: using function: prefetch64-sse (4540.000 MB/sec) [ 4.320256] raid6: sse2x1 156 MB/s [ 4.336997] raid6: sse2x2 257 MB/s [ 4.353993] raid6: sse2x4 523 MB/s [ 4.353999] raid6: using algorithm sse2x4 (523 MB/s) [ 4.354004] raid6: using ssse3x2 recovery algorithm [ 4.361309] bio: create slab <bio-1> at 1 [ 4.361660] Btrfs loaded [ 4.412706] [drm] ring test on 5 succeeded in 1 usecs [ 4.412719] [drm] UVD initialized successfully. [ 4.413723] [drm] ib test on ring 0 succeeded in 0 usecs [ 4.413845] [drm] ib test on ring 3 succeeded in 0 usecs [ 4.566733] [drm] ib test on ring 5 succeeded [ 4.567346] [drm] Radeon Display Connectors [ 4.567350] [drm] Connector 0: [ 4.567353] [drm] HDMI-A-1 [ 4.567356] [drm] HPD1 [ 4.567361] [drm] DDC: 0x6430 0x6430 0x6434 0x6434 0x6438 0x6438 0x643c 0x643c [ 4.567364] [drm] Encoders: [ 4.567367] [drm] DFP1: INTERNAL_UNIPHY1 [ 4.567370] [drm] Connector 1: [ 4.567373] [drm] VGA-1 [ 4.567377] [drm] DDC: 0x64d8 0x64d8 0x64dc 0x64dc 0x64e0 0x64e0 0x64e4 0x64e4 [ 4.567379] [drm] Encoders: [ 4.567382] [drm] CRT1: INTERNAL_KLDSCP_DAC1 [ 4.567541] [drm] Internal thermal controller with fan control [ 4.568762] [drm] radeon: power management initialized [ 4.710870] [drm] fb mappable at 0xC0375000 [ 4.710874] [drm] vram apper at 0xC0000000 [ 4.710877] [drm] size 8294400 [ 4.710880] [drm] fb depth is 24 [ 4.710883] [drm] pitch is 7680 [ 4.711041] fbcon: radeondrmfb (fb0) is primary device [ 4.932297] Console: switching to colour frame buffer device 240x67 [ 4.999732] radeon 0000:01:00.0: fb0: radeondrmfb frame buffer device [ 4.999737] radeon 0000:01:00.0: registered panic notifier [ 4.999768] [drm] Initialized radeon 2.34.0 20080528 for 0000:01:00.0 on minor 0 [ 7.713893] NET: Registered protocol family 17 [ 7.759016] PM: Marking nosave pages: [mem 0x0009f000-0x000fffff] [ 7.759029] PM: Marking nosave pages: [mem 0xbf7f4000-0xbf8abfff] [ 7.759040] PM: Marking nosave pages: [mem 0xbf8ad000-0xbf8cafff] [ 7.759044] PM: Marking nosave pages: [mem 0xbf8cd000-0xbf944fff] [ 7.759052] PM: Marking nosave pages: [mem 0xbfd1b000-0xbfee4fff] [ 7.759071] PM: Marking nosave pages: [mem 0xbfef0000-0xffffffff] [ 7.761713] PM: Basic memory bitmaps created [ 7.816156] PM: Basic memory bitmaps freed [ 7.876380] PM: Starting manual resume from disk [ 7.876389] PM: Hibernation image partition 8:1 present [ 7.876393] PM: Looking for hibernation image. [ 7.877975] PM: Image not found (code -22) [ 7.877980] PM: Hibernation image not present or could not be loaded. [ 7.931129] btrfs: disk space caching is enabled [ 7.938125] Btrfs detected SSD devices, enabling SSD mode [ 8.194664] device-mapper: uevent: version 1.0.3 [ 8.194857] device-mapper: ioctl: 4.25.0-ioctl (2013-06-26) initialised: dm-devel(a)redhat.com [ 8.197390] LVM: Activation generator successfully completed. [ 8.581758] sd 0:0:0:0: Attached scsi generic sg0 type 0 [ 8.594831] systemd-udevd[330]: starting version 207 [ 8.722650] btrfs: disk space caching is enabled [ 8.849759] input: Power Button as /devices/LNXSYSTM:00/device:00/PNP0C0C:00/input/input3 [ 8.849917] ACPI: Power Button [PWRB] [ 8.850086] input: Sleep Button as /devices/LNXSYSTM:00/device:00/PNP0C0E:00/input/input4 [ 8.850235] ACPI: Sleep Button [SLPB] [ 8.850447] input: Power Button as /devices/LNXSYSTM:00/LNXPWRBN:00/input/input5 [ 8.850633] ACPI: Power Button [PWRF] [ 8.917055] shpchp: Standard Hot Plug PCI Controller Driver version: 0.4 [ 8.987771] r8169 Gigabit Ethernet driver 2.3LK-NAPI loaded [ 8.987813] r8169 0000:02:00.2: can't disable ASPM; OS doesn't have ASPM control [ 8.988222] r8169 0000:02:00.2: irq 45 for MSI/MSI-X [ 8.988914] r8169 0000:02:00.2 eth0: RTL8411 at 0xffffc90000616000, 80:ee:73:5d:59:d5, XID 08800800 IRQ 45 [ 8.988926] r8169 0000:02:00.2 eth0: jumbo features [frames: 9200 bytes, tx checksumming: ko] [ 8.989473] rtsx_pci 0000:02:00.0: irq 46 for MSI/MSI-X [ 8.989519] rtsx_pci 0000:02:00.0: rtsx_pci_acquire_irq: pcr->msi_en = 1, pci->irq = 46 [ 9.042648] i801_smbus 0000:00:1f.3: SMBus using PCI Interrupt [ 9.045703] ACPI Warning: 0x0000000000000428-0x000000000000042f SystemIO conflicts with Region \SWC1 1 (20130517/utaddress-251) [ 9.045723] ACPI Warning: 0x0000000000000428-0x000000000000042f SystemIO conflicts with Region \PMIO 2 (20130517/utaddress-251) [ 9.045735] ACPI: If an ACPI driver is available for this device, you should use it instead of the native driver [ 9.045744] ACPI Warning: 0x0000000000000530-0x000000000000053f SystemIO conflicts with Region \GPIO 1 (20130517/utaddress-251) [ 9.045755] ACPI: If an ACPI driver is available for this device, you should use it instead of the native driver [ 9.045761] ACPI Warning: 0x0000000000000500-0x000000000000052f SystemIO conflicts with Region \GPIO 1 (20130517/utaddress-251) [ 9.045771] ACPI: If an ACPI driver is available for this device, you should use it instead of the native driver [ 9.045775] lpc_ich: Resource conflict(s) found affecting gpio_ich [ 9.069532] input: PC Speaker as /devices/platform/pcspkr/input/input6 [ 9.161452] snd_hda_intel 0000:00:1b.0: irq 47 for MSI/MSI-X [ 9.173414] coretemp coretemp.0: Unable to read TjMax from CPU 0 [ 9.173454] coretemp coretemp.0: Unable to read TjMax from CPU 2 [ 9.183151] ALSA hda_auto_parser.c:393 autoconfig: line_outs=1 (0xd/0x0/0x0/0x0/0x0) type:speaker [ 9.183163] ALSA hda_auto_parser.c:397 speaker_outs=0 (0x0/0x0/0x0/0x0/0x0) [ 9.183169] ALSA hda_auto_parser.c:401 hp_outs=1 (0xb/0x0/0x0/0x0/0x0) [ 9.183175] ALSA hda_auto_parser.c:402 mono: mono_out=0x0 [ 9.183179] ALSA hda_auto_parser.c:406 inputs: [ 9.183186] ALSA hda_auto_parser.c:410 Mic=0xa [ 9.185481] input: HDA Digital PCBeep as /devices/pci0000:00/0000:00:1b.0/input/input7 [ 9.212908] input: HDA Intel HDMI/DP,pcm=3 as /devices/pci0000:00/0000:00:1b.0/sound/card0/input8 [ 9.213158] input: HDA Intel Front Headphone as /devices/pci0000:00/0000:00:1b.0/sound/card0/input9 [ 9.213947] input: HDA Intel Mic as /devices/pci0000:00/0000:00:1b.0/sound/card0/input10 [ 9.215296] ALSA hda_intel.c:3116 0000:01:00.1: Handle VGA-switcheroo audio client [ 9.215308] ALSA hda_intel.c:3317 0000:01:00.1: Using LPIB position fix [ 9.215809] snd_hda_intel 0000:01:00.1: irq 48 for MSI/MSI-X [ 9.219577] ALSA hda_intel.c:1787 0000:01:00.1: Enable sync_write for stable communication [ 9.221420] input: HD-Audio Generic HDMI/DP,pcm=3 as /devices/pci0000:00/0000:00:1c.0/0000:01:00.1/sound/card1/input11 [ 9.482902] iTCO_vendor_support: vendor-support=0 [ 9.484128] iTCO_wdt: Intel TCO WatchDog Timer Driver v1.10 [ 9.484199] iTCO_wdt: Found a NM10 TCO device (Version=2, TCOBASE=0x0460) [ 9.485279] iTCO_wdt: initialized. heartbeat=30 sec (nowayout=0) [ 13.532853] r8169 0000:02:00.2 ens1f2: link down [ 13.533034] IPv6: ADDRCONF(NETDEV_UP): ens1f2: link is not ready [ 15.100806] r8169 0000:02:00.2 ens1f2: link up [ 15.100824] IPv6: ADDRCONF(NETDEV_CHANGE): ens1f2: link becomes ready [ 37.401917] fuse init (API version 7.22)

2 4

[alsa-devel] [RFC PATCH] ALSA : hda - not use assigned converters for all unused pins
by mengdong.lin＠intel.com 20 Sep '13

20 Sep '13

From: Mengdong Lin <mengdong.lin(a)intel.com> BIOS can mark a pin as "no physical connection" if the port is used by an integrated display which is not audio capable. And audio driver will overlook such pins. On Haswell, such a disconneted pin will keep muted and connected to the 1st converter by default. But if the 1st convertor is assigned to a connected pin for audio streaming. The muted disconnected pin can make the connected pin no sound output. So this patch avoids using assigned converters for all unused pins for Haswell, including the disconected pins. Signed-off-by: Mengdong Lin <mengdong.lin(a)intel.com> diff --git a/sound/pci/hda/patch_hdmi.c b/sound/pci/hda/patch_hdmi.c index 3d8cd044..4960695 100644 --- a/sound/pci/hda/patch_hdmi.c +++ b/sound/pci/hda/patch_hdmi.c @@ -1152,29 +1152,34 @@ static void haswell_config_cvts(struct hda_codec *codec, int pin_id, int mux_id) { struct hdmi_spec *spec = codec->spec; - struct hdmi_spec_per_pin *per_pin; - int pin_idx, mux_idx; + hda_nid_t pin_nid; + int pin_idx, cvt_idx; int curr; - int err; - - for (pin_idx = 0; pin_idx < spec->num_pins; pin_idx++) { - per_pin = get_pin(spec, pin_idx); + struct hdmi_spec_per_cvt *per_cvt; + /* configure all 3 pins, including "no physical connection" ones */ + for (pin_nid = 5; pin_nid < 8; pin_nid++) { + pin_idx = pin_nid_to_pin_index(spec, pin_nid); if (pin_idx == pin_id) continue; - curr = snd_hda_codec_read(codec, per_pin->pin_nid, 0, + curr = snd_hda_codec_read(codec, pin_nid, 0, AC_VERB_GET_CONNECT_SEL, 0); + if (curr != mux_id) + continue; + + for (cvt_idx = 0; cvt_idx < spec->num_cvts; cvt_idx++) { + per_cvt = get_cvt(spec, cvt_idx); - /* Choose another unused converter */ - if (curr == mux_id) { - err = hdmi_choose_cvt(codec, pin_idx, NULL, &mux_idx); - if (err < 0) - return; - snd_printdd("HDMI: choose converter %d for pin %d\n", mux_idx, pin_idx); - snd_hda_codec_write_cache(codec, per_pin->pin_nid, 0, + /* Must not already be assigned */ + if (!per_cvt->assigned) { + snd_printdd("choose converter %d for pin %d\n", + cvt_idx, pin_idx); + snd_hda_codec_write_cache(codec, pin_nid, 0, AC_VERB_SET_CONNECT_SEL, - mux_idx); + cvt_idx); + break; + } } } } -- 1.8.1.2

3 4

Re: [alsa-devel] [PATCH 01/51] DMA-API: provide a helper to set both DMA and coherent DMA masks
by Russell King - ARM Linux 20 Sep '13

20 Sep '13

On Fri, Sep 20, 2013 at 02:21:37AM +0100, Ben Hutchings wrote: > On Thu, 2013-09-19 at 22:25 +0100, Russell King wrote: > [...] > > -dma_set_coherent_mask() will always be able to set the same or a > > -smaller mask as dma_set_mask(). However for the rare case that a > > +The coherent coherent mask will always be able to set the same or a > > +smaller mask as the streaming mask. However for the rare case that a > [...] > > The new wording doesn't make sense; a mask doesn't set itself. I would > suggest: > > "The coherent mask can always be set to the same or a smaller mask than > the streaming mask." Yes, the original sentence is not particularly good, but I think even your modified version can be interpreted as "a mask setting itself" for all the same reasons that the original can be (which mask does "same" refer to?) Even so, I prefer your version. Thanks. :)

1 0

[alsa-devel] [PATCH 1/3] ASoC: ab8500: Downgrade noisy log message
by Mark Brown 20 Sep '13

20 Sep '13

From: Mark Brown <broonie(a)linaro.org> Signed-off-by: Mark Brown <broonie(a)linaro.org> --- sound/soc/codecs/ab8500-codec.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/codecs/ab8500-codec.c b/sound/soc/codecs/ab8500-codec.c index 80555d7..2726e5a 100644 --- a/sound/soc/codecs/ab8500-codec.c +++ b/sound/soc/codecs/ab8500-codec.c @@ -2606,7 +2606,7 @@ static int ab8500_codec_driver_probe(struct platform_device *pdev) static int ab8500_codec_driver_remove(struct platform_device *pdev) { - dev_info(&pdev->dev, "%s Enter.\n", __func__); + dev_dbg(&pdev->dev, "%s Enter.\n", __func__); snd_soc_unregister_codec(&pdev->dev); -- 1.8.4.rc3

2 3

[alsa-devel] Alsa mmap and asynchronous callbacks
by Bruce 20 Sep '13

20 Sep '13

Hi, I tried posting a query related to async callbacks on the alsa-user list, and the Hammerfall forum, but noone appeared to know what I was talking about. I'm hoping someone here can at least give me some clues. *Working:* I wrote an application, based around the alsa example code alsa-lib/test/pcm.c using mmap and asynchronous callback notifications. This application runs fine with 32 channels in and 2 out on a Hammerfall MADI system and under alsa 1.0.15. It now works on 2 different machines setup with SUSE 10, then with alsa (and card driver) upgraded to 1.0.15. alsamixer is used to setup the soundcard in 96kHz mode (double speed). /Note:/ For some reason (I can't recall) I never managed to get the playback async callback to work at the same time as capture callback, so I simply called the playback "top-up" routine from the capture async routine. That works fine. The application was used for some years ! *Problem:* SUSE 11 with alsa 1.0.18, same machine, same sound card, application hangs after a few callbacks. No more asynchronous callbacks occur after approx 1s of playback/record. However, the callbacks are working for a short time. alsamixer looks quite different for the Hammerfall with this newer alsa, and it does appear to be the change of alsa driver that causes the problem. SUSE 11/Ubuntu with alsa 1.0.27. I tried this configuration in order to try a different Xonar sound card. The application was modified for only 2 channels, and interleaved instead of non-interleaved sample format (as required by the hardware if not using plughw). Both Xonar and Hammerfall hang in the same way. aplay and alsamixer both work with both soundcards. The alsa calls I'm using do not appear to be in the deprecated list (except "snd_pcm_sw_params_set_xfer_align" in 1.0.27). Would you like me to post code fragments ? /Note: /As a workaround, I attempted to install the working 1.0.15 under the later kernal in Suse 11. Unfortunately ./configure fails in alsa-driver due to differing kernel hooks. It is not realistic to stay with the very old SUSE 10; much of the machine hardware is not recognised. -- Cheers, Bruce

1 0

[alsa-devel] No sound with Sony VAIO VPCZ1 (ALC889)
by Tormen 20 Sep '13

20 Sep '13

Hi, alsa-driver: 1.0.25.3453.g60df0 (the latest I guess -- I used alsa-compile.sh) Kernel: 3.9.6-1~bpo70+1 (debian 7.1 backport) Problem: Sound works fine, except that no speaker gets detected. The internal speakers in my Sony VAIO VPCZ1 notebook did get detected in the past though. If I am not mistaken, this is the problem: [ 5.864681] ALSA hda_auto_parser.c:397 speaker_outs=0 (0x0/0x0/0x0/0x0/0x0) alsa-info.sh: !!################################ !!ALSA Information Script v 0.4.62 !!################################ !!Script ran on: Wed Jul 3 17:38:37 UTC 2013 !!Linux Distribution !!------------------ Debian GNU/Linux 7 \n \l PRETTY_NAME="Debian GNU/Linux 7 (wheezy)" NAME="Debian GNU/Linux" ID=debian HOME_URL="http://www.debian.org/" SUPPORT_URL="http://www.debian.org/support/" BUG_REPORT_URL="http://bugs.debian.org/" !!DMI Information !!--------------- Manufacturer: Sony Corporation Product Name: VPCZ1290S Product Version: J0049ZZ5 Firmware Version: R3030C3 !!Kernel Information !!------------------ Kernel release: 3.9-0.bpo.1-amd64 Operating System: GNU/Linux Architecture: x86_64 Processor: unknown SMP Enabled: Yes !!ALSA Version !!------------ Driver version: 1.0.25.3453.g60df0 Library version: 1.0.25 Utilities version: 1.0.25 !!Loaded ALSA modules !!------------------- snd_hda_intel !!Sound Servers on this system !!---------------------------- Pulseaudio: Installed - Yes (/usr/bin/pulseaudio) Running - Yes !!Soundcards recognised by ALSA !!----------------------------- 0 [MID ]: HDA-Intel - HDA Intel MID HDA Intel MID at 0xb6420000 irq 45 !!PCI Soundcards installed in the system !!-------------------------------------- 00:1b.0 Audio device: Intel Corporation 5 Series/3400 Series Chipset High Definition Audio (rev 05) !!Advanced information - PCI Vendor/Device/Subsystem ID's !!------------------------------------------------------- 00:1b.0 0403: 8086:3b56 (rev 05) Subsystem: 104d:905a !!Modprobe options (Sound related) !!-------------------------------- snd_pcsp: index=-2 snd_usb_audio: index=-2 snd_atiixp_modem: index=-2 snd_intel8x0m: index=-2 snd_via82xx_modem: index=-2 snd_hda_intel: model=no-primary-hp !!Loaded sound module options !!--------------------------- !!Module: snd_hda_intel align_buffer_size : -1 bdl_pos_adj : 32,-1,-1,-1,-1,-1,-1,-1 beep_mode : Y,Y,Y,Y,Y,Y,Y,Y enable : Y,Y,Y,Y,Y,Y,Y,Y enable_msi : -1 id : (null),(null),(null),(null),(null),(null),(null),(null) index : -1,-1,-1,-1,-1,-1,-1,-1 jackpoll_ms : 0,0,0,0,0,0,0,0 model : no-primary-hp,(null),(null),(null),(null),(null),(null),(null) patch : (null),(null),(null),(null),(null),(null),(null),(null) position_fix : -1,-1,-1,-1,-1,-1,-1,-1 power_save : 0 power_save_controller : Y probe_mask : -1,-1,-1,-1,-1,-1,-1,-1 probe_only : 0,0,0,0,0,0,0,0 single_cmd : N snoop : Y !!HDA-Intel Codec information !!--------------------------- --startcollapse-- Codec: Realtek ALC889 Address: 0 AFG Function Id: 0x1 (unsol 1) Vendor Id: 0x10ec0889 Subsystem Id: 0x104d4400 Revision Id: 0x100005 No Modem Function Group found Default PCM: rates [0x5f0]: 32000 44100 48000 88200 96000 192000 bits [0xe]: 16 20 24 formats [0x1]: PCM Default Amp-In caps: N/A Default Amp-Out caps: N/A State of AFG node 0x01: Power states: D0 D1 D2 D3 CLKSTOP EPSS Power: setting=D0, actual=D0 GPIO: io=2, o=0, i=0, unsolicited=1, wake=1 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 0x11: Stereo Device: name="ALC889 Analog", type="Audio", device=0 Converter: stream=8, channel=0 PCM: rates [0x5e0]: 44100 48000 88200 96000 192000 bits [0xe]: 16 20 24 formats [0x1]: PCM Node 0x03 [Audio Output] wcaps 0x11: Stereo Converter: stream=8, channel=0 PCM: rates [0x5e0]: 44100 48000 88200 96000 192000 bits [0xe]: 16 20 24 formats [0x1]: PCM Node 0x04 [Audio Output] wcaps 0x11: Stereo Converter: stream=8, channel=0 PCM: rates [0x5e0]: 44100 48000 88200 96000 192000 bits [0xe]: 16 20 24 formats [0x1]: PCM Node 0x05 [Audio Output] wcaps 0x11: Stereo Converter: stream=8, channel=0 PCM: rates [0x5e0]: 44100 48000 88200 96000 192000 bits [0xe]: 16 20 24 formats [0x1]: PCM Node 0x06 [Audio Output] wcaps 0x211: Stereo Digital Converter: stream=0, channel=0 Digital: Digital category: 0x0 IEC Coding Type: 0x0 PCM: rates [0x5f0]: 32000 44100 48000 88200 96000 192000 bits [0xe]: 16 20 24 formats [0x1]: PCM Node 0x07 [Audio Input] wcaps 0x10011b: Stereo Amp-In Control: name="Capture Volume", index=0, device=0 ControlAmp: chs=3, dir=In, idx=0, ofs=0 Control: name="Capture Switch", index=0, device=0 ControlAmp: chs=3, dir=In, idx=0, ofs=0 Device: name="ALC889 Analog", type="Audio", device=0 Amp-In caps: ofs=0x10, nsteps=0x2e, stepsize=0x03, mute=1 Amp-In vals: [0x28 0x28] Converter: stream=4, channel=0 SDI-Select: 0 PCM: rates [0x5e0]: 44100 48000 88200 96000 192000 bits [0xe]: 16 20 24 formats [0x1]: PCM Connection: 1 0x24 Node 0x08 [Audio Input] wcaps 0x10011b: Stereo Amp-In Amp-In caps: ofs=0x10, nsteps=0x2e, stepsize=0x03, mute=1 Amp-In vals: [0x90 0x90] Converter: stream=0, channel=0 SDI-Select: 0 PCM: rates [0x5e0]: 44100 48000 88200 96000 192000 bits [0xe]: 16 20 24 formats [0x1]: PCM Connection: 1 0x23 Node 0x09 [Audio Input] wcaps 0x10011b: Stereo Amp-In Amp-In caps: ofs=0x10, nsteps=0x2e, stepsize=0x03, mute=1 Amp-In vals: [0x90 0x90] Converter: stream=0, channel=0 SDI-Select: 0 PCM: rates [0x5e0]: 44100 48000 88200 96000 192000 bits [0xe]: 16 20 24 formats [0x1]: PCM Connection: 1 0x22 Node 0x0a [Audio Input] wcaps 0x100391: Stereo Digital Converter: stream=0, channel=0 SDI-Select: 0 Digital: Digital category: 0x0 IEC Coding Type: 0x0 PCM: rates [0x570]: 32000 44100 48000 96000 192000 bits [0xe]: 16 20 24 formats [0x1]: PCM Unsolicited: tag=00, enabled=0 Connection: 1 0x1f 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 Control: name="Mic Playback Volume", index=1, device=0 ControlAmp: chs=3, dir=In, idx=1, ofs=0 Control: name="Mic Playback Switch", index=1, device=0 ControlAmp: chs=3, dir=In, idx=1, ofs=0 Control: name="Beep Playback Volume", index=0, device=0 ControlAmp: chs=3, dir=In, idx=5, ofs=0 Control: name="Beep Playback Switch", index=0, device=0 ControlAmp: chs=3, dir=In, idx=5, ofs=0 Amp-In caps: ofs=0x17, nsteps=0x1f, stepsize=0x05, mute=1 Amp-In vals: [0x8f 0x8f] [0x80 0x80] [0x80 0x80] [0x80 0x80] [0x80 0x80] [0x80 0x80] [0x80 0x80] [0x80 0x80] [0x80 0x80] [0x80 0x80] Connection: 10 0x18 0x19 0x1a 0x1b 0x1c 0x1d 0x14 0x15 0x16 0x17 Node 0x0c [Audio Mixer] wcaps 0x20010f: Stereo Amp-In Amp-Out Control: name="Headphone Playback Volume", index=0, device=0 ControlAmp: chs=3, dir=Out, idx=0, ofs=0 Amp-In caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 Amp-In vals: [0x00 0x00] [0x00 0x00] Amp-Out caps: ofs=0x3e, nsteps=0x40, stepsize=0x03, mute=0 Amp-Out vals: [0x3e 0x3e] Connection: 2 0x02 0x0b Node 0x0d [Audio Mixer] wcaps 0x20010f: Stereo Amp-In Amp-Out Control: name="Surround Playback Volume", index=0, device=0 ControlAmp: chs=3, dir=Out, idx=0, ofs=0 Amp-In caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 Amp-In vals: [0x80 0x80] [0x80 0x80] Amp-Out caps: ofs=0x3e, nsteps=0x40, stepsize=0x03, mute=0 Amp-Out vals: [0x3e 0x3e] Connection: 2 0x03 0x0b Node 0x0e [Audio Mixer] wcaps 0x20010f: Stereo Amp-In Amp-Out Control: name="Center Playback Volume", index=0, device=0 ControlAmp: chs=1, dir=Out, idx=0, ofs=0 Control: name="LFE Playback Volume", index=0, device=0 ControlAmp: chs=2, dir=Out, idx=0, ofs=0 Amp-In caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 Amp-In vals: [0x80 0x80] [0x80 0x80] Amp-Out caps: ofs=0x3e, nsteps=0x40, stepsize=0x03, mute=0 Amp-Out vals: [0x3e 0x3e] Connection: 2 0x04 0x0b Node 0x0f [Audio Mixer] wcaps 0x20010f: Stereo Amp-In Amp-Out Control: name="Speaker Playback Volume", index=0, device=0 ControlAmp: chs=3, dir=Out, idx=0, ofs=0 Amp-In caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 Amp-In vals: [0x00 0x00] [0x00 0x00] Amp-Out caps: ofs=0x3e, nsteps=0x40, stepsize=0x03, mute=0 Amp-Out vals: [0x40 0x40] Connection: 2 0x05 0x0b Node 0x10 [Audio Output] wcaps 0x211: Stereo Digital Converter: stream=0, channel=0 Digital: Digital category: 0x0 IEC Coding Type: 0x0 PCM: rates [0x5f0]: 32000 44100 48000 88200 96000 192000 bits [0xe]: 16 20 24 formats [0x1]: PCM Node 0x11 [Pin Complex] wcaps 0x400300: Mono Digital 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: 0x40: OUT Connection: 1 0x10 Node 0x12 [Pin Complex] wcaps 0x400001: Stereo Control: name="Internal Mic Phantom Jack", index=0, device=0 Pincap 0x00000020: IN Pin Default 0x90a60120: [Fixed] Mic at Int N/A Conn = Digital, Color = Unknown DefAssociation = 0x2, Sequence = 0x0 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 0x0001003c: IN OUT HP EAPD Detect EAPD 0x2: EAPD Pin Default 0x90170110: [Fixed] Speaker at Int N/A Conn = Analog, Color = Unknown DefAssociation = 0x1, Sequence = 0x0 Misc = NO_PRESENCE Pin-ctls: 0x40: OUT Unsolicited: tag=00, enabled=0 Connection: 5 0x0c 0x0d 0x0e 0x0f* 0x26 Node 0x15 [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 0x0001003c: IN OUT HP EAPD Detect EAPD 0x2: EAPD Pin Default 0x0421101f: [Jack] HP Out at Ext Right Conn = 1/8, Color = Black DefAssociation = 0x1, Sequence = 0xf Pin-ctls: 0xc0: OUT HP Unsolicited: tag=01, enabled=1 Connection: 5 0x0c* 0x0d 0x0e 0x0f 0x26 Node 0x16 [Pin Complex] wcaps 0x40018d: Stereo Amp-Out 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: 0x00: Unsolicited: tag=00, enabled=0 Connection: 5 0x0c 0x0d 0x0e* 0x0f 0x26 Node 0x17 [Pin Complex] wcaps 0x40018d: Stereo Amp-Out 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: 0x00: Unsolicited: tag=00, enabled=0 Connection: 5 0x0c 0x0d 0x0e 0x0f* 0x26 Node 0x18 [Pin Complex] wcaps 0x40018f: Stereo Amp-In Amp-Out Control: name="Surround Playback Switch", index=0, device=0 ControlAmp: chs=3, dir=Out, idx=0, ofs=0 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=0x27, mute=0 Amp-In vals: [0x00 0x00] Amp-Out caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 Amp-Out vals: [0x00 0x00] Pincap 0x0000373c: IN OUT HP Detect Vref caps: HIZ 50 GRD 80 100 Pin Default 0x04a1102e: [Jack] Mic at Ext Right Conn = 1/8, Color = Black DefAssociation = 0x2, Sequence = 0xe Pin-ctls: 0x24: IN VREF_80 Unsolicited: tag=02, enabled=1 Connection: 5 0x0c 0x0d 0x0e 0x0f 0x26* Node 0x19 [Pin Complex] wcaps 0x40018f: Stereo Amp-In Amp-Out Control: name="Center Playback Switch", index=0, device=0 ControlAmp: chs=1, dir=Out, idx=0, ofs=0 Control: name="LFE Playback Switch", index=0, device=0 ControlAmp: chs=2, dir=Out, idx=0, ofs=0 Control: name="Mic Boost Volume", index=1, device=0 ControlAmp: chs=3, dir=In, idx=0, ofs=0 Control: name="Mic Jack", index=1, device=0 Amp-In caps: ofs=0x00, nsteps=0x03, stepsize=0x27, mute=0 Amp-In vals: [0x00 0x00] Amp-Out caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 Amp-Out vals: [0x00 0x00] Pincap 0x0000373c: IN OUT HP Detect Vref caps: HIZ 50 GRD 80 100 Pin Default 0x04a11030: [Jack] Mic at Ext Right Conn = 1/8, Color = Black DefAssociation = 0x3, Sequence = 0x0 Pin-ctls: 0x24: IN VREF_80 Unsolicited: tag=03, enabled=1 Connection: 5 0x0c 0x0d 0x0e 0x0f 0x26* Node 0x1a [Pin Complex] wcaps 0x40018f: Stereo Amp-In Amp-Out Amp-In caps: ofs=0x00, nsteps=0x03, stepsize=0x27, mute=0 Amp-In vals: [0x00 0x00] Amp-Out caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 Amp-Out vals: [0x80 0x80] Pincap 0x0000373c: IN OUT HP Detect Vref caps: HIZ 50 GRD 80 100 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 Connection: 5 0x0c 0x0d 0x0e 0x0f 0x26* Node 0x1b [Pin Complex] wcaps 0x40018f: Stereo Amp-In Amp-Out Amp-In caps: ofs=0x00, nsteps=0x03, stepsize=0x27, mute=0 Amp-In vals: [0x00 0x00] Amp-Out caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 Amp-Out vals: [0x80 0x80] Pincap 0x0000373c: IN OUT HP Detect Vref caps: HIZ 50 GRD 80 100 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 Connection: 5 0x0c 0x0d 0x0e 0x0f 0x26* Node 0x1c [Pin Complex] wcaps 0x400081: Stereo Pincap 0x00000024: IN 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 Node 0x1d [Pin Complex] wcaps 0x400000: Mono Pincap 0x00000020: IN 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 Node 0x1e [Pin Complex] wcaps 0x400300: Mono Digital 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: 0x40: OUT Connection: 1 0x06 Node 0x1f [Pin Complex] wcaps 0x400280: Mono Digital Pincap 0x00000020: IN 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 Node 0x20 [Vendor Defined Widget] wcaps 0xf00040: Mono Processing caps: benign=0, ncoeff=28 Node 0x21 [Vendor Defined Widget] wcaps 0xf00000: Mono Node 0x22 [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] [0x80 0x80] [0x80 0x80] [0x80 0x80] [0x80 0x80] [0x80 0x80] Connection: 11 0x18 0x19 0x1a 0x1b 0x1c 0x1d 0x14 0x15 0x16 0x17 0x0b 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] [0x80 0x80] [0x80 0x80] [0x80 0x80] [0x80 0x80] [0x80 0x80] Connection: 11 0x18 0x19 0x1a 0x1b 0x1c 0x1d 0x14 0x15 0x16 0x17 0x0b Node 0x24 [Audio Selector] wcaps 0x300101: Stereo Connection: 12 0x18 0x19 0x1a 0x1b 0x1c 0x1d 0x14 0x15 0x16 0x17 0x0b 0x12* Node 0x25 [Audio Output] wcaps 0x11: Stereo Converter: stream=0, channel=0 PCM: rates [0x5e0]: 44100 48000 88200 96000 192000 bits [0xe]: 16 20 24 formats [0x1]: PCM Node 0x26 [Audio Mixer] wcaps 0x20010f: Stereo Amp-In Amp-Out Amp-In caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 Amp-In vals: [0x00 0x00] [0x80 0x80] Amp-Out caps: ofs=0x3e, nsteps=0x40, stepsize=0x03, mute=0 Amp-Out vals: [0x3e 0x3e] Connection: 2 0x25 0x0b --endcollapse-- !!ALSA Device nodes !!----------------- crw-rw---T+ 1 root audio 116, 0 Jul 3 19:35 /dev/snd/controlC0 crw-rw---T+ 1 root audio 116, 4 Jul 3 19:35 /dev/snd/hwC0D0 crw-rw---T+ 1 root audio 116, 24 Jul 3 19:36 /dev/snd/pcmC0D0c crw-rw---T+ 1 root audio 116, 16 Jul 3 19:36 /dev/snd/pcmC0D0p crw-rw---T+ 1 root audio 116, 1 Jul 3 19:35 /dev/snd/seq crw-rw---T+ 1 root audio 116, 33 Jul 3 19:35 /dev/snd/timer /dev/snd/by-path: total 0 drwxr-xr-x 2 root root 60 Jul 3 19:35 . drwxr-xr-x 3 root root 180 Jul 3 19:35 .. lrwxrwxrwx 1 root root 12 Jul 3 19:35 pci-0000:00:1b.0 -> ../controlC0 !!ALSA configuration files !!------------------------ !!System wide config file (/etc/asound.conf) pcm.pulse { type pulse } ctl.pulse { type pulse } pcm.!default { type pulse } ctl.!default { type pulse } !!Aplay/Arecord output !!-------------------- APLAY **** List of PLAYBACK Hardware Devices **** card 0: MID [HDA Intel MID], device 0: ALC889 Analog [ALC889 Analog] Subdevices: 1/1 Subdevice #0: subdevice #0 ARECORD **** List of CAPTURE Hardware Devices **** card 0: MID [HDA Intel MID], device 0: ALC889 Analog [ALC889 Analog] Subdevices: 1/1 Subdevice #0: subdevice #0 !!Amixer output !!------------- !!-------Mixer controls for card 0 [MID] Card hw:0 'MID'/'HDA Intel MID at 0xb6420000 irq 45' Mixer name : 'Realtek ALC889' Components : 'HDA:10ec0889,104d4400,00100005' Controls : 34 Simple ctrls : 18 Simple mixer control 'Master',0 Capabilities: pvolume pvolume-joined pswitch pswitch-joined penum Playback channels: Mono Limits: Playback 0 - 64 Mono: Playback 64 [100%] [0.00dB] [on] Simple mixer control 'Headphone',0 Capabilities: pvolume pswitch penum Playback channels: Front Left - Front Right Limits: Playback 0 - 64 Mono: Front Left: Playback 62 [97%] [0.00dB] [on] Front Right: Playback 62 [97%] [0.00dB] [on] Simple mixer control 'Speaker',0 Capabilities: pvolume pswitch penum Playback channels: Front Left - Front Right Limits: Playback 0 - 64 Mono: Front Left: Playback 64 [100%] [2.00dB] [on] Front Right: Playback 64 [100%] [2.00dB] [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 'Surround',0 Capabilities: pvolume pswitch penum Playback channels: Front Left - Front Right Limits: Playback 0 - 64 Mono: Front Left: Playback 62 [97%] [0.00dB] [on] Front Right: Playback 62 [97%] [0.00dB] [on] Simple mixer control 'Center',0 Capabilities: pvolume pvolume-joined pswitch pswitch-joined penum Playback channels: Mono Limits: Playback 0 - 64 Mono: Playback 62 [97%] [0.00dB] [on] Simple mixer control 'LFE',0 Capabilities: pvolume pvolume-joined pswitch pswitch-joined penum Playback channels: Mono Limits: Playback 0 - 64 Mono: Playback 62 [97%] [0.00dB] [on] Simple mixer control 'Mic',0 Capabilities: pvolume pswitch cswitch cswitch-joined cswitch-exclusive penum Capture exclusive group: 0 Playback channels: Front Left - Front Right Capture channels: Mono Limits: Playback 0 - 31 Mono: Capture [off] Front Left: Playback 15 [48%] [-12.00dB] [off] Front Right: Playback 15 [48%] [-12.00dB] [off] Simple mixer control 'Mic 1',0 Capabilities: cswitch cswitch-joined cswitch-exclusive penum Capture exclusive group: 0 Capture channels: Mono Mono: Capture [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 'Mic',1 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',1 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 'Beep',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 'Capture',0 Capabilities: cvolume cswitch penum Capture channels: Front Left - Front Right Limits: Capture 0 - 46 Front Left: Capture 40 [87%] [24.00dB] [on] Front Right: Capture 40 [87%] [24.00dB] [on] Simple mixer control 'Auto-Mute Mode',0 Capabilities: enum Items: 'Disabled' 'Enabled' Item0: 'Enabled' Simple mixer control 'Channel Mode',0 Capabilities: enum Items: '2ch' '4ch' '6ch' Item0: '2ch' Simple mixer control 'Digital',0 Capabilities: cvolume penum Capture channels: Front Left - Front Right Limits: Capture 0 - 120 Front Left: Capture 0 [0%] [-30.00dB] Front Right: Capture 0 [0%] [-30.00dB] Simple mixer control 'Internal Mic',0 Capabilities: cswitch cswitch-joined cswitch-exclusive penum Capture exclusive group: 0 Capture channels: Mono Mono: Capture [on] !!Alsactl output !!-------------- --startcollapse-- state.MID { control.1 { iface MIXER name 'Channel Mode' value '2ch' comment { access 'read write' type ENUMERATED count 1 item.0 '2ch' item.1 '4ch' item.2 '6ch' } } control.2 { iface MIXER name 'Headphone Playback Volume' value.0 62 value.1 62 comment { access 'read write' type INTEGER count 2 range '0 - 64' dbmin -6200 dbmax 200 dbvalue.0 0 dbvalue.1 0 } } control.3 { iface MIXER name 'Headphone Playback Switch' value.0 true value.1 true comment { access 'read write' type BOOLEAN count 2 } } control.4 { iface MIXER name 'Surround Playback Volume' value.0 62 value.1 62 comment { access 'read write' type INTEGER count 2 range '0 - 64' dbmin -6200 dbmax 200 dbvalue.0 0 dbvalue.1 0 } } control.5 { iface MIXER name 'Surround Playback Switch' value.0 true value.1 true comment { access 'read write' type BOOLEAN count 2 } } control.6 { iface MIXER name 'Center Playback Volume' value 62 comment { access 'read write' type INTEGER count 1 range '0 - 64' dbmin -6200 dbmax 200 dbvalue.0 0 } } control.7 { iface MIXER name 'LFE Playback Volume' value 62 comment { access 'read write' type INTEGER count 1 range '0 - 64' dbmin -6200 dbmax 200 dbvalue.0 0 } } control.8 { iface MIXER name 'Center Playback Switch' value true comment { access 'read write' type BOOLEAN count 1 } } control.9 { iface MIXER name 'LFE Playback Switch' value true comment { access 'read write' type BOOLEAN count 1 } } control.10 { iface MIXER name 'Speaker Playback Volume' value.0 64 value.1 64 comment { access 'read write' type INTEGER count 2 range '0 - 64' dbmin -6200 dbmax 200 dbvalue.0 200 dbvalue.1 200 } } control.11 { iface MIXER name 'Speaker Playback Switch' value.0 true value.1 true comment { access 'read write' type BOOLEAN count 2 } } control.12 { iface MIXER name 'Mic Playback Volume' value.0 15 value.1 15 comment { access 'read write' type INTEGER count 2 range '0 - 31' dbmin -3450 dbmax 1200 dbvalue.0 -1200 dbvalue.1 -1200 } } control.13 { iface MIXER name 'Mic Playback Switch' value.0 false value.1 false comment { access 'read write' type BOOLEAN count 2 } } control.14 { iface MIXER name 'Mic Playback Volume' index 1 value.0 0 value.1 0 comment { access 'read write' type INTEGER count 2 range '0 - 31' dbmin -3450 dbmax 1200 dbvalue.0 -3450 dbvalue.1 -3450 } } control.15 { iface MIXER name 'Mic Playback Switch' index 1 value.0 false value.1 false comment { access 'read write' type BOOLEAN count 2 } } control.16 { iface MIXER name 'Auto-Mute Mode' value Enabled comment { access 'read write' type ENUMERATED count 1 item.0 Disabled item.1 Enabled } } control.17 { iface MIXER name 'Capture Source' value 'Internal Mic' comment { access 'read write' type ENUMERATED count 1 item.0 Mic item.1 'Mic 1' item.2 'Internal Mic' } } control.18 { iface MIXER name 'Capture Volume' value.0 40 value.1 40 comment { access 'read write' type INTEGER count 2 range '0 - 46' dbmin -1600 dbmax 3000 dbvalue.0 2400 dbvalue.1 2400 } } control.19 { iface MIXER name 'Capture Switch' value.0 true value.1 true comment { access 'read write' type BOOLEAN count 2 } } control.20 { iface MIXER name 'Mic Boost Volume' value.0 0 value.1 0 comment { access 'read write' type INTEGER count 2 range '0 - 3' dbmin 0 dbmax 3000 dbvalue.0 0 dbvalue.1 0 } } control.21 { iface MIXER name 'Mic Boost Volume' index 1 value.0 0 value.1 0 comment { access 'read write' type INTEGER count 2 range '0 - 3' dbmin 0 dbmax 3000 dbvalue.0 0 dbvalue.1 0 } } control.22 { iface MIXER name 'Master Playback Volume' value 64 comment { access 'read write' type INTEGER count 1 range '0 - 64' dbmin -6400 dbmax 0 dbvalue.0 0 } } control.23 { iface MIXER name 'Master Playback Switch' value true comment { access 'read write' type BOOLEAN count 1 } } control.24 { iface CARD name 'Mic Jack' value false comment { access read type BOOLEAN count 1 } } control.25 { iface CARD name 'Mic Jack' index 1 value false comment { access read type BOOLEAN count 1 } } control.26 { iface CARD name 'Internal Mic Phantom Jack' value true comment { access read type BOOLEAN count 1 } } control.27 { iface CARD name 'Headphone Jack' value false comment { access read type BOOLEAN count 1 } } control.28 { iface CARD name 'Speaker Phantom Jack' value true comment { access read type BOOLEAN count 1 } } control.29 { iface MIXER name 'Beep Playback Volume' value.0 0 value.1 0 comment { access 'read write' type INTEGER count 2 range '0 - 31' dbmin -3450 dbmax 1200 dbvalue.0 -3450 dbvalue.1 -3450 } } control.30 { iface MIXER name 'Beep Playback Switch' value.0 false value.1 false comment { access 'read write' type BOOLEAN count 2 } } control.31 { iface PCM name 'Playback Channel Map' value.0 0 value.1 0 comment { access read type INTEGER count 2 range '0 - 36' } } control.32 { iface PCM name 'Capture Channel Map' value.0 0 value.1 0 comment { access read type INTEGER count 2 range '0 - 36' } } control.33 { iface MIXER name 'PCM Playback Volume' value.0 255 value.1 255 comment { access 'read write user' type INTEGER count 2 range '0 - 255' tlv '0000000100000008ffffec1400000014' dbmin -5100 dbmax 0 dbvalue.0 0 dbvalue.1 0 } } control.34 { iface MIXER name 'Digital Capture Volume' value.0 0 value.1 0 comment { access 'read write user' type INTEGER count 2 range '0 - 120' tlv '0000000100000008fffff44800000032' dbmin -3000 dbmax 3000 dbvalue.0 -3000 dbvalue.1 -3000 } } } --endcollapse-- !!All Loaded Modules !!------------------ Module parport_pc ppdev lp parport binfmt_misc fuse ext2 mbcache dm_crypt loop snd_hda_codec_realtek videobuf2_vmalloc videobuf2_memops videobuf2_core videodev media joydev snd_hda_intel snd_hda_codec snd_hwdep snd_pcm_oss snd_mixer_oss snd_pcm snd_page_alloc snd_seq_dummy snd_seq_oss snd_seq_midi snd_seq_midi_event snd_rawmidi snd_seq arc4 coretemp iwldvm kvm_intel mac80211 snd_seq_device kvm snd_timer crc32c_intel ghash_clmulni_intel snd iTCO_wdt iTCO_vendor_support aesni_intel iwlwifi i915 aes_x86_64 ablk_helper mxm_wmi cryptd xts cfg80211 sg lrw gf128mul sr_mod cdrom psmouse drm_kms_helper serio_raw sony_laptop evdev drm rfkill pcspkr video wmi i2c_i801 i2c_algo_bit battery button i2c_core ehci_pci ehci_hcd e1000e intel_ips soundcore acpi_cpufreq mperf ac lpc_ich usbcore mfd_core mei sdhci_pci sdhci ptp mmc_core pps_core usb_common processor btrfs crc32c libcrc32c xor zlib_deflate raid6_pq dm_mod raid0 md_mod sd_mod crc_t10dif ahci libahci microcode thermal thermal_sys libata scsi_mod !!Sysfs Files !!----------- /sys/class/sound/hwC0D0/init_pin_configs: 0x11 0x411111f0 0x12 0x90a60120 0x14 0x90170110 0x15 0x0421101f 0x16 0x411111f0 0x17 0x411111f0 0x18 0x04a1102e 0x19 0x04a11030 0x1a 0x411111f0 0x1b 0x411111f0 0x1c 0x411111f0 0x1d 0x411111f0 0x1e 0x411111f0 0x1f 0x411111f0 /sys/class/sound/hwC0D0/driver_pin_configs: /sys/class/sound/hwC0D0/user_pin_configs: /sys/class/sound/hwC0D0/init_verbs: /sys/class/sound/hwC0D0/hints: !!ALSA/HDA dmesg !!-------------- [ 5.121872] pipe_off wait timed out [ 5.121921] Modules linked in: uvcvideo videobuf2_vmalloc videobuf2_memops videobuf2_core videodev media joydev snd_hda_intel(O+) snd_hda_codec(O) snd_hwdep(O) snd_pcm_oss(O) snd_mixer_oss(O) snd_pcm(O) snd_page_alloc(O) snd_seq_dummy(O) snd_seq_oss(O) snd_seq_midi(O) snd_seq_midi_event(O) snd_rawmidi(O) snd_seq(O) arc4 coretemp iwldvm kvm_intel mac80211 snd_seq_device(O) kvm snd_timer(O) crc32c_intel ghash_clmulni_intel snd(O) iTCO_wdt iTCO_vendor_support aesni_intel iwlwifi i915(+) aes_x86_64 ablk_helper mxm_wmi cryptd xts cfg80211 sg lrw gf128mul sr_mod cdrom psmouse drm_kms_helper serio_raw sony_laptop evdev drm rfkill pcspkr video wmi i2c_i801 i2c_algo_bit battery button i2c_core ehci_pci ehci_hcd e1000e intel_ips soundcore acpi_cpufreq mperf ac lpc_ich usbcore mfd_core mei sdhci_pci sdhci ptp mmc_core pps_core usb_common processor btrfs crc32c libcrc32c xor zlib_deflate raid6_pq dm_mod raid0 md_mod sd_mod crc_t10dif ahci libahci microcode thermal thermal_sys libata scsi_mod [ 5.121941] Pid: 635, comm: modprobe Tainted: G O 3.9-0.bpo.1-amd64 #1 Debian 3.9.6-1~bpo70+1 -- [ 5.835404] [drm] Initialized i915 1.6.0 20080730 for 0000:00:02.0 on minor 0 [ 5.835752] snd_hda_intel 0000:00:1b.0: irq 45 for MSI/MSI-X [ 5.864320] hda_codec: ALC889: SKU not ready 0x411111f0 [ 5.864675] ALSA hda_auto_parser.c:393 autoconfig: line_outs=1 (0x14/0x0/0x0/0x0/0x0) type:speaker [ 5.864681] ALSA hda_auto_parser.c:397 speaker_outs=0 (0x0/0x0/0x0/0x0/0x0) [ 5.864686] ALSA hda_auto_parser.c:401 hp_outs=1 (0x15/0x0/0x0/0x0/0x0) [ 5.864689] ALSA hda_auto_parser.c:402 mono: mono_out=0x0 [ 5.864693] ALSA hda_auto_parser.c:406 inputs: [ 5.864697] ALSA hda_auto_parser.c:410 Mic=0x18 [ 5.864701] ALSA hda_auto_parser.c:410 Mic=0x19 [ 5.864704] ALSA hda_auto_parser.c:410 Internal Mic=0x12 [ 5.864709] ALSA patch_realtek.c:486 realtek: No valid SSID, checking pincfg 0x411111f0 for NID 0x1d [ 5.864713] ALSA patch_realtek.c:569 realtek: Enable default setup for auto mode as fallback [ 5.868644] input: HDA Digital PCBeep as /devices/pci0000:00/0000:00:1b.0/input/input9 [ 5.874177] input: HDA Intel MID Headphone as /devices/pci0000:00/0000:00:1b.0/sound/card0/input10 [ 5.874232] input: HDA Intel MID Mic as /devices/pci0000:00/0000:00:1b.0/sound/card0/input11 [ 5.874284] input: HDA Intel MID Mic as /devices/pci0000:00/0000:00:1b.0/sound/card0/input12 [ 6.760494] btrfs: use ssd allocation scheme -- [ 26.166424] pipe_off wait timed out [ 26.166426] Modules linked in: parport_pc ppdev lp parport binfmt_misc fuse ext2 mbcache dm_crypt loop snd_hda_codec_realtek(O) uvcvideo videobuf2_vmalloc videobuf2_memops videobuf2_core videodev media joydev snd_hda_intel(O) snd_hda_codec(O) snd_hwdep(O) snd_pcm_oss(O) snd_mixer_oss(O) snd_pcm(O) snd_page_alloc(O) snd_seq_dummy(O) snd_seq_oss(O) snd_seq_midi(O) snd_seq_midi_event(O) snd_rawmidi(O) snd_seq(O) arc4 coretemp iwldvm kvm_intel mac80211 snd_seq_device(O) kvm snd_timer(O) crc32c_intel ghash_clmulni_intel snd(O) iTCO_wdt iTCO_vendor_support aesni_intel iwlwifi i915 aes_x86_64 ablk_helper mxm_wmi cryptd xts cfg80211 sg lrw gf128mul sr_mod cdrom psmouse drm_kms_helper serio_raw sony_laptop evdev drm rfkill pcspkr video wmi i2c_i801 i2c_algo_bit battery button i2c_core ehci_pci ehci_hcd e1000e intel_ips soundcore acpi_cpufreq mperf ac lpc_ich usbcore mfd_core mei sdhci_pci sdhci ptp mmc_core pps_core usb_common processor btrfs crc32c libcrc32c xor zlib_deflate raid6_pq dm_mod raid0 md_mod sd_mod crc_t10dif ahci libahci microcode thermal thermal_sys libata scsi_mod [ 26.166527] Pid: 4157, comm: Xorg Tainted: G W O 3.9-0.bpo.1-amd64 #1 Debian 3.9.6-1~bpo70+1

4 38

[alsa-devel] [PATCH 1/2] ASoC: 88pm60x: Don't use control data for i2c
by Mark Brown 19 Sep '13

19 Sep '13

From: Mark Brown <broonie(a)linaro.org> In preparation for using the regmap directly in the CODEC driver replace references to the I2C client using control_data with references to the driver private data. Signed-off-by: Mark Brown <broonie(a)linaro.org> --- sound/soc/codecs/88pm860x-codec.c | 17 +++++++++-------- 1 file changed, 9 insertions(+), 8 deletions(-) diff --git a/sound/soc/codecs/88pm860x-codec.c b/sound/soc/codecs/88pm860x-codec.c index 259d1ac..ca5f9f0 100644 --- a/sound/soc/codecs/88pm860x-codec.c +++ b/sound/soc/codecs/88pm860x-codec.c @@ -1169,6 +1169,7 @@ static int pm860x_i2s_set_dai_fmt(struct snd_soc_dai *codec_dai, static int pm860x_set_bias_level(struct snd_soc_codec *codec, enum snd_soc_bias_level level) { + struct pm860x_priv *pm860x = snd_soc_codec_get_drvdata(codec); int data; switch (level) { @@ -1182,17 +1183,17 @@ static int pm860x_set_bias_level(struct snd_soc_codec *codec, if (codec->dapm.bias_level == SND_SOC_BIAS_OFF) { /* Enable Audio PLL & Audio section */ data = AUDIO_PLL | AUDIO_SECTION_ON; - pm860x_reg_write(codec->control_data, REG_MISC2, data); + pm860x_reg_write(pm860x->i2c, REG_MISC2, data); udelay(300); data = AUDIO_PLL | AUDIO_SECTION_RESET | AUDIO_SECTION_ON; - pm860x_reg_write(codec->control_data, REG_MISC2, data); + pm860x_reg_write(pm860x->i2c, REG_MISC2, data); } break; case SND_SOC_BIAS_OFF: data = AUDIO_PLL | AUDIO_SECTION_RESET | AUDIO_SECTION_ON; - pm860x_set_bits(codec->control_data, REG_MISC2, data, 0); + pm860x_set_bits(pm860x->i2c, REG_MISC2, data, 0); break; } codec->dapm.bias_level = level; @@ -1322,17 +1323,17 @@ int pm860x_hs_jack_detect(struct snd_soc_codec *codec, pm860x->det.lo_shrt = lo_shrt; if (det & SND_JACK_HEADPHONE) - pm860x_set_bits(codec->control_data, REG_HS_DET, + pm860x_set_bits(pm860x->i2c, REG_HS_DET, EN_HS_DET, EN_HS_DET); /* headset short detect */ if (hs_shrt) { data = CLR_SHORT_HS2 | CLR_SHORT_HS1; - pm860x_set_bits(codec->control_data, REG_SHORTS, data, data); + pm860x_set_bits(pm860x->i2c, REG_SHORTS, data, data); } /* Lineout short detect */ if (lo_shrt) { data = CLR_SHORT_LO2 | CLR_SHORT_LO1; - pm860x_set_bits(codec->control_data, REG_SHORTS, data, data); + pm860x_set_bits(pm860x->i2c, REG_SHORTS, data, data); } /* sync status */ @@ -1350,7 +1351,7 @@ int pm860x_mic_jack_detect(struct snd_soc_codec *codec, pm860x->det.mic_det = det; if (det & SND_JACK_MICROPHONE) - pm860x_set_bits(codec->control_data, REG_MIC_DET, + pm860x_set_bits(pm860x->i2c, REG_MIC_DET, MICDET_MASK, MICDET_MASK); /* sync status */ @@ -1380,7 +1381,7 @@ static int pm860x_probe(struct snd_soc_codec *codec) pm860x_set_bias_level(codec, SND_SOC_BIAS_STANDBY); - ret = pm860x_bulk_read(codec->control_data, REG_CACHE_BASE, + ret = pm860x_bulk_read(pm860x->i2c, REG_CACHE_BASE, REG_CACHE_SIZE, codec->reg_cache); if (ret < 0) { dev_err(codec->dev, "Failed to fill register cache: %d\n", -- 1.8.4.rc3

1 1

[alsa-devel] [RFC PATCH] ES938 support for ES18xx driver
by Ondrej Zary 19 Sep '13

19 Sep '13

Hello, this is an attempt to support ES938 3-D Audio Effects Processor found on some ES18xx (and possibly other) sound cards, doing bass/treble and 3D control. ES938 is controlled by MIDI SysEx commands sent through card's MPU401 port. The following patch works but has a problem: the midi port cannot be used by userspace applications. Opening and closing the rawmidi device on each control change would probably work (as long as the port is not used by an application) but that just does not seem right. Is there a way to cleanly fix this? (There are two problems: 3D level does not have any effect and alsamixer does not display the TLV dB values - but don't know why). --- sound/isa/Kconfig | 4 + sound/isa/Makefile | 2 + sound/isa/es18xx.c | 12 +++- sound/isa/es938.c | 181 ++++++++++++++++++++++++++++++++++++++++++++++++++++ sound/isa/es938.h | 24 +++++++ 5 files changed, 222 insertions(+), 1 deletions(-) create mode 100644 sound/isa/es938.c create mode 100644 sound/isa/es938.h diff --git a/sound/isa/Kconfig b/sound/isa/Kconfig index affa134..a8bf3e3 100644 --- a/sound/isa/Kconfig +++ b/sound/isa/Kconfig @@ -17,6 +17,9 @@ config SND_SB16_DSP select SND_PCM select SND_SB_COMMON +config SND_ES938 + tristate + menuconfig SND_ISA bool "ISA sound devices" depends on ISA && ISA_DMA_API @@ -183,6 +186,7 @@ config SND_ES18XX select SND_OPL3_LIB select SND_MPU401_UART select SND_PCM + select SND_ES938 help Say Y here to include support for ESS AudioDrive ES18xx chips. diff --git a/sound/isa/Makefile b/sound/isa/Makefile index 9a15f14..d59e0bf 100644 --- a/sound/isa/Makefile +++ b/sound/isa/Makefile @@ -8,6 +8,7 @@ snd-als100-objs := als100.o snd-azt2320-objs := azt2320.o snd-cmi8328-objs := cmi8328.o snd-cmi8330-objs := cmi8330.o +snd-es938-objs := es938.o snd-es18xx-objs := es18xx.o snd-opl3sa2-objs := opl3sa2.o snd-sc6000-objs := sc6000.o @@ -19,6 +20,7 @@ obj-$(CONFIG_SND_ALS100) += snd-als100.o obj-$(CONFIG_SND_AZT2320) += snd-azt2320.o obj-$(CONFIG_SND_CMI8328) += snd-cmi8328.o obj-$(CONFIG_SND_CMI8330) += snd-cmi8330.o +obj-$(CONFIG_SND_ES938) += snd-es938.o obj-$(CONFIG_SND_ES18XX) += snd-es18xx.o obj-$(CONFIG_SND_OPL3SA2) += snd-opl3sa2.o obj-$(CONFIG_SND_SC6000) += snd-sc6000.o diff --git a/sound/isa/es18xx.c b/sound/isa/es18xx.c index 12978b8..c502aa0 100644 --- a/sound/isa/es18xx.c +++ b/sound/isa/es18xx.c @@ -96,6 +96,7 @@ #define SNDRV_LEGACY_FIND_FREE_IRQ #define SNDRV_LEGACY_FIND_FREE_DMA #include <sound/initval.h> +#include "es938.h" #define PFX "es18xx: " @@ -122,6 +123,7 @@ struct snd_es18xx { struct snd_pcm_substream *playback_b_substream; struct snd_rawmidi *rmidi; + struct snd_es938 es938; struct snd_kcontrol *hw_volume; struct snd_kcontrol *hw_switch; @@ -2166,7 +2168,15 @@ static int snd_audiodrive_probe(struct snd_card *card, int dev) return err; } - return snd_card_register(card); + err = snd_card_register(card); + if (err < 0) + return err; + + if (mpu_port[dev] > 0 && mpu_port[dev] != SNDRV_AUTO_PORT) + if (snd_es938_init(&chip->es938, card, 0, 0) == 0) + printk("es938 found!\n"); + + return 0; } static int snd_es18xx_isa_match(struct device *pdev, unsigned int dev) diff --git a/sound/isa/es938.c b/sound/isa/es938.c new file mode 100644 index 0000000..cfe8ae7 --- /dev/null +++ b/sound/isa/es938.c @@ -0,0 +1,181 @@ +/* + * Driver for ESS ES938 3-D Audio Effects Processor + * Copyright (c) 2013 Ondrej Zary + * + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License as published by + * the Free Software Foundation; either version 2 of the License, or + * (at your option) any later version. + * + * This program is distributed in the hope that it will be useful, + * but WITHOUT ANY WARRANTY; without even the implied warranty of + * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the + * GNU General Public License for more details. + * + * You should have received a copy of the GNU General Public License + * along with this program; if not, write to the Free Software + * Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA + * + */ + +//#include <linux/init.h> +#include <linux/module.h> +#include <sound/asoundef.h> +#include <sound/control.h> +#include <sound/core.h> +#include <sound/rawmidi.h> +#include "es938.h" + +#define PFX "es938: " + +MODULE_AUTHOR("Ondrej Zary"); +MODULE_DESCRIPTION("ESS ES938"); +MODULE_LICENSE("GPL"); + +static int snd_es938_read_reg(struct snd_es938 *chip, u8 reg, u8 *out) +{ + u8 buf[8]; + int i = 0, res; + u8 sysex[] = { MIDI_CMD_COMMON_SYSEX, ES938_ID, ES938_CMD_REG_R, reg, + MIDI_CMD_COMMON_SYSEX_END }; + + snd_rawmidi_kernel_write(chip->rfile.output, sysex, sizeof(sysex)); + + memset(buf, 0, sizeof(buf)); + while (i < sizeof(buf)) { + res = snd_rawmidi_kernel_read(chip->rfile.input, buf + i, sizeof(buf) - i); + if (res > 0) + i+= res; + } + + /* check reply */ + if (memcmp(buf, sysex, 6) || buf[7] != MIDI_CMD_COMMON_SYSEX_END) + return -1; + + if (out) + *out = buf[6]; + + return 0; +} + +static void snd_es938_write_reg(struct snd_es938 *chip, u8 reg, u8 val) +{ + u8 sysex[] = { MIDI_CMD_COMMON_SYSEX, ES938_ID, ES938_CMD_REG_W, reg, + val, MIDI_CMD_COMMON_SYSEX_END }; + + snd_rawmidi_kernel_write(chip->rfile.output, sysex, sizeof(sysex)); + chip->regs[reg] = val; +} + +static void snd_es938_write_reg_mask(struct snd_es938 *chip, u8 reg, u8 val, u8 mask) +{ + u8 oldval = chip->regs[reg]; + oldval &= ~mask; + snd_es938_write_reg(chip, reg, oldval | (val & mask)); +} + +#define ES938_MIXER(xname, xindex, reg, shift, mask) \ +{ .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = xname, .index = xindex, \ + .info = snd_es938_info_mixer, \ + .get = snd_es938_get_mixer, .put = snd_es938_put_mixer, \ + .private_value = reg | (shift << 8) | (mask << 16) } + +#define ES938_MIXER_TLV(xname, xindex, reg, shift, mask, xtlv) \ +{ .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = xname, .index = xindex, \ + .access = SNDRV_CTL_ELEM_ACCESS_READWRITE | SNDRV_CTL_ELEM_ACCESS_TLV_READ, \ + .info = snd_es938_info_mixer, \ + .get = snd_es938_get_mixer, .put = snd_es938_put_mixer, \ + .private_value = reg | (shift << 8) | (mask << 16), \ + .tlv = { .p = (xtlv) } } + +static const DECLARE_TLV_DB_SCALE(db_scale_tone, -900, 300, 0); + +static struct snd_kcontrol_new snd_es938_controls[] = { +ES938_MIXER_TLV("Tone Control - Bass", 0, ES938_REG_TONE, 0, 7, db_scale_tone), +ES938_MIXER_TLV("Tone Control - Treble", 0, ES938_REG_TONE, 4, 7, db_scale_tone), +ES938_MIXER("3D Control - Level", 0, ES938_REG_SPATIAL, 1, 63), +ES938_MIXER("3D Control - Switch", 0, ES938_REG_SPATIAL_EN, 0, 1), +}; + +int snd_es938_init(struct snd_es938 *chip, struct snd_card *card, int device, + int subdevice) +{ + int err, i; + + if ((err = snd_rawmidi_kernel_open(card, device, subdevice, + SNDRV_RAWMIDI_LFLG_OPEN | SNDRV_RAWMIDI_LFLG_APPEND, + &chip->rfile)) < 0) { + snd_printk(KERN_WARNING PFX "unable to open MIDI device\n"); + return err; + } + + /* try to read a register to detect chip presence */ + if (snd_es938_read_reg(chip, ES938_REG_MISC, NULL) < 0) + return -ENODEV; + + /* write default values (there's no reset) */ + snd_es938_write_reg(chip, ES938_REG_MISC, 0x49); + snd_es938_write_reg(chip, ES938_REG_TONE, 0x33); + snd_es938_write_reg(chip, ES938_REG_SPATIAL, 0x00); + /* datasheet specifies invalid value 0x00 as default */ + snd_es938_write_reg(chip, ES938_REG_SPATIAL_EN, 0x02); + snd_es938_write_reg(chip, ES938_REG_POWER, 0x0e); + + strlcat(card->mixername, " + ES938", sizeof(card->mixername)); + for (i = 0; i < ARRAY_SIZE(snd_es938_controls); i++) + if ((err = snd_ctl_add(card, snd_ctl_new1(&snd_es938_controls[i], chip))) < 0) + return err; + + return 0; +} +EXPORT_SYMBOL(snd_es938_init); + +void snd_es938_release(struct snd_es938 *chip) +{ + snd_rawmidi_kernel_release(&chip->rfile); +} +EXPORT_SYMBOL(snd_es938_release); + + +int snd_es938_info_mixer(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_info *uinfo) +{ + int mask = (kcontrol->private_value >> 16) & 0xff; + + uinfo->type = mask == 1 ? SNDRV_CTL_ELEM_TYPE_BOOLEAN : SNDRV_CTL_ELEM_TYPE_INTEGER; + uinfo->count = 1; + uinfo->value.integer.min = 0; + uinfo->value.integer.max = mask; + return 0; +} +EXPORT_SYMBOL(snd_es938_info_mixer); + +int snd_es938_get_mixer(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) +{ + struct snd_es938 *chip = snd_kcontrol_chip(kcontrol); + int reg = kcontrol->private_value & 0xff; + int shift = (kcontrol->private_value >> 8) & 0xff; + int mask = (kcontrol->private_value >> 16) & 0xff; + u8 val = chip->regs[reg]; + + ucontrol->value.integer.value[0] = (val >> shift) & mask; + return 0; +} +EXPORT_SYMBOL(snd_es938_get_mixer); + +int snd_es938_put_mixer(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) +{ + struct snd_es938 *chip = snd_kcontrol_chip(kcontrol); + int reg = kcontrol->private_value & 0xff; + int shift = (kcontrol->private_value >> 8) & 0xff; + int mask = (kcontrol->private_value >> 16) & 0xff; + u8 val = ucontrol->value.integer.value[0] & mask; + u8 oldval = chip->regs[reg]; + + mask <<= shift; + val <<= shift; + + snd_es938_write_reg_mask(chip, reg, val, mask); + return chip->regs[reg] != oldval; +} +EXPORT_SYMBOL(snd_es938_put_mixer); diff --git a/sound/isa/es938.h b/sound/isa/es938.h new file mode 100644 index 0000000..1f8595f --- /dev/null +++ b/sound/isa/es938.h @@ -0,0 +1,24 @@ +#include <sound/tlv.h> + +#define ES938_ID 0x00, 0x00, 0x7b + +#define ES938_CMD_REG_R 0x7e +#define ES938_CMD_REG_W 0x7f + +#define ES938_REG_MISC 0 +#define ES938_REG_TONE 1 +#define ES938_REG_SPATIAL 5 +#define ES938_REG_SPATIAL_EN 6 +#define ES938_REG_POWER 7 + +struct snd_es938 { + u8 regs[8]; + struct snd_rawmidi_file rfile; +}; + +int snd_es938_init(struct snd_es938 *chip, struct snd_card *card, int device, + int subdevice); +void snd_es938_release(struct snd_es938 *chip); +int snd_es938_info_mixer(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_info *uinfo); +int snd_es938_get_mixer(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol); +int snd_es938_put_mixer(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol); -- Ondrej Zary

2 8

[alsa-devel] [PATCH] Fix access of freed memory in namehints
by David Henningsson 19 Sep '13

19 Sep '13

Sometimes a hook manipulates the config tree, which makes currently running iterators point to freed memory. As a workaround, make two copies, one for the iterators and another for the hooks. BugLink: https://bugs.launchpad.net/bugs/1008600 Signed-off-by: David Henningsson <david.henningsson(a)canonical.com> --- src/control/namehint.c | 22 +++++++++++++--------- 1 file changed, 13 insertions(+), 9 deletions(-) diff --git a/src/control/namehint.c b/src/control/namehint.c index 8d5e925..28975a4 100644 --- a/src/control/namehint.c +++ b/src/control/namehint.c @@ -406,7 +406,7 @@ static const next_devices_t next_devices[] = { }; #endif -static int add_card(snd_config_t *config, struct hint_list *list, int card) +static int add_card(snd_config_t *config, snd_config_t *rw_config, struct hint_list *list, int card) { int err, ok; snd_config_t *conf, *n; @@ -449,7 +449,7 @@ static int add_card(snd_config_t *config, struct hint_list *list, int card) ok = 0; for (device = 0; err >= 0 && device <= max_device; device++) { list->device = device; - err = try_config(config, list, list->siface, str); + err = try_config(rw_config, list, list->siface, str); if (err < 0) break; ok++; @@ -464,7 +464,7 @@ static int add_card(snd_config_t *config, struct hint_list *list, int card) if (err < 0) { list->card = card; list->device = -1; - err = try_config(config, list, list->siface, str); + err = try_config(rw_config, list, list->siface, str); } if (err == -ENOMEM) goto __error; @@ -493,7 +493,8 @@ static int get_card_name(struct hint_list *list, int card) return 0; } -static int add_software_devices(snd_config_t *config, struct hint_list *list) +static int add_software_devices(snd_config_t *config, snd_config_t *rw_config, + struct hint_list *list) { int err; snd_config_t *conf, *n; @@ -509,7 +510,7 @@ static int add_software_devices(snd_config_t *config, struct hint_list *list) continue; list->card = -1; list->device = -1; - err = try_config(config, list, list->siface, str); + err = try_config(rw_config, list, list->siface, str); if (err == -ENOMEM) return -ENOMEM; } @@ -547,7 +548,7 @@ int snd_device_name_hint(int card, const char *iface, void ***hints) struct hint_list list; char ehints[24]; const char *str; - snd_config_t *conf, *local_config = NULL; + snd_config_t *conf, *local_config = NULL, *local_config_rw = NULL; snd_config_update_t *local_config_update = NULL; snd_config_iterator_t i, next; int err; @@ -557,6 +558,7 @@ int snd_device_name_hint(int card, const char *iface, void ***hints) err = snd_config_update_r(&local_config, &local_config_update, NULL); if (err < 0) return err; + err = snd_config_copy(&local_config_rw, local_config); list.list = NULL; list.count = list.allocated = 0; list.siface = iface; @@ -586,9 +588,9 @@ int snd_device_name_hint(int card, const char *iface, void ***hints) if (card >= 0) { err = get_card_name(&list, card); if (err >= 0) - err = add_card(local_config, &list, card); + err = add_card(local_config, local_config_rw, &list, card); } else { - add_software_devices(local_config, &list); + add_software_devices(local_config, local_config_rw, &list); err = snd_card_next(&card); if (err < 0) goto __error; @@ -596,7 +598,7 @@ int snd_device_name_hint(int card, const char *iface, void ***hints) err = get_card_name(&list, card); if (err < 0) goto __error; - err = add_card(local_config, &list, card); + err = add_card(local_config, local_config_rw, &list, card); if (err < 0) goto __error; err = snd_card_next(&card); @@ -630,6 +632,8 @@ int snd_device_name_hint(int card, const char *iface, void ***hints) if (list.cardname) free(list.cardname); } + if (local_config_rw) + snd_config_delete(local_config_rw); if (local_config) snd_config_delete(local_config); if (local_config_update) -- 1.8.3.2

2 1

[alsa-devel] question about webcam audio capture
by vichy 19 Sep '13

19 Sep '13

hi all: I try to do webcam audio capture on my arm platform. when I execute audio capture I got below message "retire_capture_urb: 108 callbacks suppressed" my questions: 1. where I can get the message "retire_capture_urb: 108 callbacks suppressed" show in sound/usb driver? I try to grep -rwn '"retire_capture_urb.*' sound/usb/*, but get nothing. 2. from usb mon log, it seems HW keep Active status high even until ehci_io_watchdog happen, because the ep status is -EXDEV. if my observation is correct, is it possible to modify usb sound driver to purposely send fix number urbs. Because from the usb mon log, the first 8 urb will get time out already. thanks for your help in advance,

2 7

[alsa-devel] [PATCH 1/1] ASoC: mmp-sspa: Use devm_snd_soc_register_component
by Sachin Kamat 19 Sep '13

19 Sep '13

devm_snd_soc_register_component makes code simpler. Signed-off-by: Sachin Kamat <sachin.kamat(a)linaro.org> --- sound/soc/pxa/mmp-sspa.c | 5 ++--- 1 file changed, 2 insertions(+), 3 deletions(-) diff --git a/sound/soc/pxa/mmp-sspa.c b/sound/soc/pxa/mmp-sspa.c index 41752a5..5bf5f1f 100644 --- a/sound/soc/pxa/mmp-sspa.c +++ b/sound/soc/pxa/mmp-sspa.c @@ -455,8 +455,8 @@ static int asoc_mmp_sspa_probe(struct platform_device *pdev) priv->dai_fmt = (unsigned int) -1; platform_set_drvdata(pdev, priv); - return snd_soc_register_component(&pdev->dev, &mmp_sspa_component, - &mmp_sspa_dai, 1); + return devm_snd_soc_register_component(&pdev->dev, &mmp_sspa_component, + &mmp_sspa_dai, 1); } static int asoc_mmp_sspa_remove(struct platform_device *pdev) @@ -466,7 +466,6 @@ static int asoc_mmp_sspa_remove(struct platform_device *pdev) clk_disable(priv->audio_clk); clk_put(priv->audio_clk); clk_put(priv->sysclk); - snd_soc_unregister_component(&pdev->dev); return 0; } -- 1.7.9.5

2 1

[alsa-devel] [PATCH 1/1] ASoC: omap-mcpdm: Use devm_snd_soc_register_component
by Sachin Kamat 19 Sep '13

19 Sep '13

devm_snd_soc_register_component makes code simpler. Signed-off-by: Sachin Kamat <sachin.kamat(a)linaro.org> --- Compile tested. --- sound/soc/omap/omap-mcpdm.c | 12 +++--------- 1 file changed, 3 insertions(+), 9 deletions(-) diff --git a/sound/soc/omap/omap-mcpdm.c b/sound/soc/omap/omap-mcpdm.c index 90d2a7c..cd9ee16 100644 --- a/sound/soc/omap/omap-mcpdm.c +++ b/sound/soc/omap/omap-mcpdm.c @@ -490,14 +490,9 @@ static int asoc_mcpdm_probe(struct platform_device *pdev) mcpdm->dev = &pdev->dev; - return snd_soc_register_component(&pdev->dev, &omap_mcpdm_component, - &omap_mcpdm_dai, 1); -} - -static int asoc_mcpdm_remove(struct platform_device *pdev) -{ - snd_soc_unregister_component(&pdev->dev); - return 0; + return devm_snd_soc_register_component(&pdev->dev, + &omap_mcpdm_component, + &omap_mcpdm_dai, 1); } static const struct of_device_id omap_mcpdm_of_match[] = { @@ -514,7 +509,6 @@ static struct platform_driver asoc_mcpdm_driver = { }, .probe = asoc_mcpdm_probe, - .remove = asoc_mcpdm_remove, }; module_platform_driver(asoc_mcpdm_driver); -- 1.7.9.5

2 1

[alsa-devel] [PATCH 1/1] ASoC: mxs-saif: Use devm_snd_soc_register_component
by Sachin Kamat 19 Sep '13

19 Sep '13

devm_snd_soc_register_component makes code simpler. Signed-off-by: Sachin Kamat <sachin.kamat(a)linaro.org> --- sound/soc/mxs/mxs-saif.c | 12 +++--------- 1 file changed, 3 insertions(+), 9 deletions(-) diff --git a/sound/soc/mxs/mxs-saif.c b/sound/soc/mxs/mxs-saif.c index b56b8a0..14152f6 100644 --- a/sound/soc/mxs/mxs-saif.c +++ b/sound/soc/mxs/mxs-saif.c @@ -768,8 +768,8 @@ static int mxs_saif_probe(struct platform_device *pdev) dev_warn(&pdev->dev, "failed to init clocks\n"); } - ret = snd_soc_register_component(&pdev->dev, &mxs_saif_component, - &mxs_saif_dai, 1); + ret = devm_snd_soc_register_component(&pdev->dev, &mxs_saif_component, + &mxs_saif_dai, 1); if (ret) { dev_err(&pdev->dev, "register DAI failed\n"); return ret; @@ -778,21 +778,15 @@ static int mxs_saif_probe(struct platform_device *pdev) ret = mxs_pcm_platform_register(&pdev->dev); if (ret) { dev_err(&pdev->dev, "register PCM failed: %d\n", ret); - goto failed_pdev_alloc; + return ret; } return 0; - -failed_pdev_alloc: - snd_soc_unregister_component(&pdev->dev); - - return ret; } static int mxs_saif_remove(struct platform_device *pdev) { mxs_pcm_platform_unregister(&pdev->dev); - snd_soc_unregister_component(&pdev->dev); return 0; } -- 1.7.9.5

2 1

[alsa-devel] [PATCH 1/1] ASoC: fsl_spdif: Use devm_snd_soc_register_component
by Sachin Kamat 19 Sep '13

19 Sep '13

devm_snd_soc_register_component makes code simpler. Signed-off-by: Sachin Kamat <sachin.kamat(a)linaro.org> --- sound/soc/fsl/fsl_spdif.c | 14 +++----------- 1 file changed, 3 insertions(+), 11 deletions(-) diff --git a/sound/soc/fsl/fsl_spdif.c b/sound/soc/fsl/fsl_spdif.c index 3920c3e..44378e6 100644 --- a/sound/soc/fsl/fsl_spdif.c +++ b/sound/soc/fsl/fsl_spdif.c @@ -1172,23 +1172,16 @@ static int fsl_spdif_probe(struct platform_device *pdev) /* Register with ASoC */ dev_set_drvdata(&pdev->dev, spdif_priv); - ret = snd_soc_register_component(&pdev->dev, &fsl_spdif_component, - &spdif_priv->cpu_dai_drv, 1); + ret = devm_snd_soc_register_component(&pdev->dev, &fsl_spdif_component, + &spdif_priv->cpu_dai_drv, 1); if (ret) { dev_err(&pdev->dev, "failed to register DAI: %d\n", ret); return ret; } ret = imx_pcm_dma_init(pdev); - if (ret) { + if (ret) dev_err(&pdev->dev, "imx_pcm_dma_init failed: %d\n", ret); - goto error_component; - } - - return ret; - -error_component: - snd_soc_unregister_component(&pdev->dev); return ret; } @@ -1196,7 +1189,6 @@ error_component: static int fsl_spdif_remove(struct platform_device *pdev) { imx_pcm_dma_exit(pdev); - snd_soc_unregister_component(&pdev->dev); return 0; } -- 1.7.9.5

2 1

[alsa-devel] [PATCH 1/1] ASoC: omap-twl4030: Use devm_snd_soc_register_card
by Sachin Kamat 19 Sep '13

19 Sep '13

devm_snd_soc_register_card makes code simpler. Signed-off-by: Sachin Kamat <sachin.kamat(a)linaro.org> --- sound/soc/omap/omap-twl4030.c | 5 ++--- 1 file changed, 2 insertions(+), 3 deletions(-) diff --git a/sound/soc/omap/omap-twl4030.c b/sound/soc/omap/omap-twl4030.c index 2a9324f..6a8d6b5 100644 --- a/sound/soc/omap/omap-twl4030.c +++ b/sound/soc/omap/omap-twl4030.c @@ -338,9 +338,9 @@ static int omap_twl4030_probe(struct platform_device *pdev) } snd_soc_card_set_drvdata(card, priv); - ret = snd_soc_register_card(card); + ret = devm_snd_soc_register_card(&pdev->dev, card); if (ret) { - dev_err(&pdev->dev, "snd_soc_register_card() failed: %d\n", + dev_err(&pdev->dev, "devm_snd_soc_register_card() failed: %d\n", ret); return ret; } @@ -357,7 +357,6 @@ static int omap_twl4030_remove(struct platform_device *pdev) snd_soc_jack_free_gpios(&priv->hs_jack, ARRAY_SIZE(hs_jack_gpios), hs_jack_gpios); - snd_soc_unregister_card(card); return 0; } -- 1.7.9.5

2 1

[alsa-devel] [PATCH 1/1] ASoC: mfld: Use devm_snd_soc_register_card
by Sachin Kamat 19 Sep '13

19 Sep '13

devm_snd_soc_register_card makes code simpler. Signed-off-by: Sachin Kamat <sachin.kamat(a)linaro.org> --- sound/soc/mid-x86/mfld_machine.c | 10 +--------- 1 file changed, 1 insertion(+), 9 deletions(-) diff --git a/sound/soc/mid-x86/mfld_machine.c b/sound/soc/mid-x86/mfld_machine.c index ee36384..d3d4c32 100644 --- a/sound/soc/mid-x86/mfld_machine.c +++ b/sound/soc/mid-x86/mfld_machine.c @@ -400,7 +400,7 @@ static int snd_mfld_mc_probe(struct platform_device *pdev) } /* register the soc card */ snd_soc_card_mfld.dev = &pdev->dev; - ret_val = snd_soc_register_card(&snd_soc_card_mfld); + ret_val = devm_snd_soc_register_card(&pdev->dev, &snd_soc_card_mfld); if (ret_val) { pr_debug("snd_soc_register_card failed %d\n", ret_val); return ret_val; @@ -410,20 +410,12 @@ static int snd_mfld_mc_probe(struct platform_device *pdev) return 0; } -static int snd_mfld_mc_remove(struct platform_device *pdev) -{ - pr_debug("snd_mfld_mc_remove called\n"); - snd_soc_unregister_card(&snd_soc_card_mfld); - return 0; -} - static struct platform_driver snd_mfld_mc_driver = { .driver = { .owner = THIS_MODULE, .name = "msic_audio", }, .probe = snd_mfld_mc_probe, - .remove = snd_mfld_mc_remove, }; module_platform_driver(snd_mfld_mc_driver); -- 1.7.9.5

2 1

[alsa-devel] [PATCH 1/3] ASoC: imx-sgtl5000: Use devm_snd_soc_register_card
by Sachin Kamat 19 Sep '13

19 Sep '13

devm_snd_soc_register_card makes code simpler. Signed-off-by: Sachin Kamat <sachin.kamat(a)linaro.org> --- This series compile tested. --- sound/soc/fsl/imx-sgtl5000.c | 12 +----------- 1 file changed, 1 insertion(+), 11 deletions(-) diff --git a/sound/soc/fsl/imx-sgtl5000.c b/sound/soc/fsl/imx-sgtl5000.c index 46c5b4f..78f86d8 100644 --- a/sound/soc/fsl/imx-sgtl5000.c +++ b/sound/soc/fsl/imx-sgtl5000.c @@ -159,7 +159,7 @@ static int imx_sgtl5000_probe(struct platform_device *pdev) data->card.dapm_widgets = imx_sgtl5000_dapm_widgets; data->card.num_dapm_widgets = ARRAY_SIZE(imx_sgtl5000_dapm_widgets); - ret = snd_soc_register_card(&data->card); + ret = devm_snd_soc_register_card(&pdev->dev, &data->card); if (ret) { dev_err(&pdev->dev, "snd_soc_register_card failed (%d)\n", ret); goto fail; @@ -180,15 +180,6 @@ fail: return ret; } -static int imx_sgtl5000_remove(struct platform_device *pdev) -{ - struct imx_sgtl5000_data *data = platform_get_drvdata(pdev); - - snd_soc_unregister_card(&data->card); - - return 0; -} - static const struct of_device_id imx_sgtl5000_dt_ids[] = { { .compatible = "fsl,imx-audio-sgtl5000", }, { /* sentinel */ } @@ -202,7 +193,6 @@ static struct platform_driver imx_sgtl5000_driver = { .of_match_table = imx_sgtl5000_dt_ids, }, .probe = imx_sgtl5000_probe, - .remove = imx_sgtl5000_remove, }; module_platform_driver(imx_sgtl5000_driver); -- 1.7.9.5

2 3

Re: [alsa-devel] [PATCH] ASoC: rt5640: Omit ACPI match table only if !ACPI
by Mark Brown 19 Sep '13

19 Sep '13

On Thu, Sep 19, 2013 at 01:21:04PM +0200, Thierry Reding wrote: > Or perhaps we can annotate the table so that the compiler will stop > complaining but remove the table nonetheless if it isn't used. Similarly > to what we can do with > > if (IS_ENABLED(CONFIG_FOO)) { > ... > } > > constructs. I can't really think of a way to do that, though. Perhaps > gcc will throw away the table anyway, so we'll just need a way to make > it shut up? There's an annotation which can stop the unreferenced variable warning and then modern compilers are able to eliminate unreferenced static symbols. Not sure if it's actually worth the effort of doing all this though, but it is nice to be able to avoid needing multiple builds.

1 0

[alsa-devel] [PATCH] ASoC: fsl: imx-sgtl5000: fix error path if codec driver not found
by Andrew Gabbasov 19 Sep '13

19 Sep '13

Handling of this error condition should make some cleanup on function exit instead of immediate return. Signed-off-by: Andrew Gabbasov <andrew_gabbasov(a)mentor.com> --- sound/soc/fsl/imx-sgtl5000.c | 3 ++- 1 file changed, 2 insertions(+), 1 deletion(-) diff --git a/sound/soc/fsl/imx-sgtl5000.c b/sound/soc/fsl/imx-sgtl5000.c index 46c5b4f..7ea5658 100644 --- a/sound/soc/fsl/imx-sgtl5000.c +++ b/sound/soc/fsl/imx-sgtl5000.c @@ -119,7 +119,8 @@ static int imx_sgtl5000_probe(struct platform_device *pdev) codec_dev = of_find_i2c_device_by_node(codec_np); if (!codec_dev) { dev_err(&pdev->dev, "failed to find codec platform device\n"); - return -EPROBE_DEFER; + ret = -EPROBE_DEFER; + goto fail; } data = devm_kzalloc(&pdev->dev, sizeof(*data), GFP_KERNEL); -- 1.7.10.4

1 0

[alsa-devel] [PATCH] ASoC: fsl: imx-wm8962: defer probing if ssi/codec not found
by Andrew Gabbasov 19 Sep '13

19 Sep '13

If the ssi or codec driver is not yet loaded by the time of imx-wm8962 driver probing, this probe should be deferred until the dependent drivers loading. Return -EPROBE_DEFER in this case. Signed-off-by: Andrew Gabbasov <andrew_gabbasov(a)mentor.com> --- sound/soc/fsl/imx-wm8962.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/soc/fsl/imx-wm8962.c b/sound/soc/fsl/imx-wm8962.c index 722afe6..9e60aa8 100644 --- a/sound/soc/fsl/imx-wm8962.c +++ b/sound/soc/fsl/imx-wm8962.c @@ -211,13 +211,13 @@ static int imx_wm8962_probe(struct platform_device *pdev) ssi_pdev = of_find_device_by_node(ssi_np); if (!ssi_pdev) { dev_err(&pdev->dev, "failed to find SSI platform device\n"); - ret = -EINVAL; + ret = -EPROBE_DEFER; goto fail; } codec_dev = of_find_i2c_device_by_node(codec_np); if (!codec_dev || !codec_dev->driver) { dev_err(&pdev->dev, "failed to find codec platform device\n"); - ret = -EINVAL; + ret = -EPROBE_DEFER; goto fail; } -- 1.7.10.4

1 0

[alsa-devel] [PATCH 1/3] mfd: mc13xxx: Don't require lock for simple register I/O
by Mark Brown 19 Sep '13

19 Sep '13

From: Mark Brown <broonie(a)linaro.org> Since the conversion to regmap there has been no need for device level locking for I/O as regmap provides locking so remove the locks. Signed-off-by: Mark Brown <broonie(a)linaro.org> --- drivers/mfd/mc13xxx-core.c | 5 ----- 1 file changed, 5 deletions(-) diff --git a/drivers/mfd/mc13xxx-core.c b/drivers/mfd/mc13xxx-core.c index 2a9b100..dbbf8ee 100644 --- a/drivers/mfd/mc13xxx-core.c +++ b/drivers/mfd/mc13xxx-core.c @@ -158,8 +158,6 @@ int mc13xxx_reg_read(struct mc13xxx *mc13xxx, unsigned int offset, u32 *val) { int ret; - BUG_ON(!mutex_is_locked(&mc13xxx->lock)); - if (offset > MC13XXX_NUMREGS) return -EINVAL; @@ -172,8 +170,6 @@ EXPORT_SYMBOL(mc13xxx_reg_read); int mc13xxx_reg_write(struct mc13xxx *mc13xxx, unsigned int offset, u32 val) { - BUG_ON(!mutex_is_locked(&mc13xxx->lock)); - dev_vdbg(mc13xxx->dev, "[0x%02x] <- 0x%06x\n", offset, val); if (offset > MC13XXX_NUMREGS || val > 0xffffff) @@ -186,7 +182,6 @@ EXPORT_SYMBOL(mc13xxx_reg_write); int mc13xxx_reg_rmw(struct mc13xxx *mc13xxx, unsigned int offset, u32 mask, u32 val) { - BUG_ON(!mutex_is_locked(&mc13xxx->lock)); BUG_ON(val & ~mask); dev_vdbg(mc13xxx->dev, "[0x%02x] <- 0x%06x (mask: 0x%06x)\n", offset, val, mask); -- 1.8.4.rc3

2 8

[alsa-devel] [PATCH] ASoC: rt5640: ifdef for ACPI module table
by Olof Johansson 19 Sep '13

19 Sep '13

Commit 02b80773de37 (ASoC: rt5640: Add ACPI probing support.) causes a warning due to unreferenced variable on non-ACPI configs such as tegra_defconfig on ARM: sound/soc/codecs/rt5640.c:2085:30: warning: 'rt5640_acpi_match' defined but not used [-Wunused-variable] Signed-off-by: Olof Johansson <olof(a)lixom.net> --- sound/soc/codecs/rt5640.c | 2 ++ 1 file changed, 2 insertions(+) diff --git a/sound/soc/codecs/rt5640.c b/sound/soc/codecs/rt5640.c index 0bfb960..a856706 100644 --- a/sound/soc/codecs/rt5640.c +++ b/sound/soc/codecs/rt5640.c @@ -2082,11 +2082,13 @@ static const struct i2c_device_id rt5640_i2c_id[] = { }; MODULE_DEVICE_TABLE(i2c, rt5640_i2c_id); +#if CONFIG_ACPI static struct acpi_device_id rt5640_acpi_match[] = { { "INT33CA", 0 }, { }, }; MODULE_DEVICE_TABLE(acpi, rt5640_acpi_match); +#endif static int rt5640_parse_dt(struct rt5640_priv *rt5640, struct device_node *np) { -- 1.7.10.4

3 11

[alsa-devel] [PATCH] ASoC: kirkwood: fix loss of external clock at probe time
by Jean-Francois Moine 19 Sep '13

19 Sep '13

At probe time, when the clock driver is not yet initialized, the external clock of the kirkwood sound device will not be usable. This patch fixes this problem defering the device probe. Signed-off-by: Jean-Francois Moine <moinejf(a)free.fr> --- sound/soc/kirkwood/kirkwood-i2s.c | 5 ++++- 1 file changed, 4 insertions(+), 1 deletion(-) diff --git a/sound/soc/kirkwood/kirkwood-i2s.c b/sound/soc/kirkwood/kirkwood-i2s.c index 0f3d73d..14f32d9 100644 --- a/sound/soc/kirkwood/kirkwood-i2s.c +++ b/sound/soc/kirkwood/kirkwood-i2s.c @@ -496,7 +496,10 @@ static int kirkwood_i2s_dev_probe(struct platform_device *pdev) return err; priv->extclk = devm_clk_get(&pdev->dev, "extclk"); - if (!IS_ERR(priv->extclk)) { + if (IS_ERR(priv->extclk)) { + if (np && of_property_match_string(np, "clock-names", "extclk") >= 0) + return -EPROBE_DEFER; + } else { if (priv->extclk == priv->clk) { devm_clk_put(&pdev->dev, priv->extclk); priv->extclk = ERR_PTR(-EINVAL); -- Ken ar c'hentañ | ** Breizh ha Linux atav! ** Jef | http://moinejf.free.fr/

2 1

Re: [alsa-devel] [PATCH] ASoC: rt5640: Omit ACPI match table only if !ACPI
by Mark Brown 19 Sep '13

19 Sep '13

On Thu, Sep 19, 2013 at 11:18:06AM +0200, Thierry Reding wrote: > The ACPI_PTR() macro evaluates to NULL if ACPI is disabled and hence the > ACPI match table won't be used, causing the compiler to complain. Avoid > this by protecting the table using an #ifdef CONFIG_ACPI. Applied, thanks. Like I was saying to Olof we really should be able to do better than needing the ifdefs though.

1 0

[alsa-devel] [PATCH 1/1] ASoC: SPEAr spdif_out: Remove redundant variable
by Sachin Kamat 19 Sep '13

19 Sep '13

Return directly and remove the intermediate local variable. Signed-off-by: Sachin Kamat <sachin.kamat(a)linaro.org> --- sound/soc/spear/spdif_out.c | 6 ++---- 1 file changed, 2 insertions(+), 4 deletions(-) diff --git a/sound/soc/spear/spdif_out.c b/sound/soc/spear/spdif_out.c index 70fc4d6..b6ef6f7 100644 --- a/sound/soc/spear/spdif_out.c +++ b/sound/soc/spear/spdif_out.c @@ -280,7 +280,6 @@ static int spdif_out_probe(struct platform_device *pdev) struct spdif_out_dev *host; struct spear_spdif_platform_data *pdata; struct resource *res; - int ret; host = devm_kzalloc(&pdev->dev, sizeof(*host), GFP_KERNEL); if (!host) { @@ -307,9 +306,8 @@ static int spdif_out_probe(struct platform_device *pdev) dev_set_drvdata(&pdev->dev, host); - ret = devm_snd_soc_register_component(&pdev->dev, &spdif_out_component, - &spdif_out_dai, 1); - return ret; + return devm_snd_soc_register_component(&pdev->dev, &spdif_out_component, + &spdif_out_dai, 1); } #ifdef CONFIG_PM -- 1.7.9.5

2 1

[alsa-devel] Fwd: Re: Realtek ALC889: HDA Intel and kernel 3.1 gives choppy sound (again)
by Colomban Wendling 18 Sep '13

18 Sep '13

Here's my initial mail, but with gzipped outputs. Regards, Colomban -------- Message original -------- Sujet: Re: [alsa-devel] Realtek ALC889: HDA Intel and kernel 3.1 gives choppy sound (again) Date : Thu, 10 Nov 2011 02:11:17 +0100 De : Colomban Wendling <lists.ban(a)herbesfolles.org> Pour : Takashi Iwai <tiwai(a)suse.de> Copie à : alsa-devel(a)alsa-project.org Hi again, and sorry for the delay. Le 08/11/2011 07:32, Takashi Iwai a écrit : > At Mon, 07 Nov 2011 23:45:40 +0100, > Colomban Wendling wrote: >> >> Hi, >> >> Back to 3.0, the sound on my soundcard [1] started to be choppy, I >> reported it [2] and it got fixed (thanks to Takashi Iwai!). >> >> However, the story repeated with 3.1 (and probably 3.0.8 or before [3]): >> I've got similar choppy sound again. >> >> I bisected the few commits that happened on >> sound/pci/hda/patch_realtek.c, and finally found the villain: >> >> "8974bd51 ALSA: hda/realtek - Fix auto-mute with HP+LO configuration" >> >> Reverting it from v3.1 fixes the problem. > > Does it? Very weird. This patch has nothing to do with the HD-audio > controller side but purely a codec change. Yep, I just check once again and it really fixes the problem here. >> master (94956ee) also suffers of the problem, but I couldn't test >> without that commit because reverting it fails, too much changes happened. > > Did you try to pass position_fix=1 (or 2) option? > Also, passing enable_msi=0 (or 1) may change anything? I just tried with the 3 kernels: * vanilla v3.1 * vanilla 3.2-rc1 (master at 1ea6b8f) * v3.1 with 8974bd51 reverted and none of the option changed anything: both vanilla always failed, the third always succeeded. Though, I haven't combined the options, just tried each at once. > In anyway, please give alsa-info.sh output again with the fresh > kernel. Run it with --no-upload option and attach the output. Here it is, for the 3 kernels cited above. Regards, Colomban

4 14

Re: [alsa-devel] snd_pcm_delay issue - How to debug?
by Sander Jansen 18 Sep '13

18 Sep '13

Never mind. Issue not related to alsa. On Tue, Jul 16, 2013 at 6:27 PM, Sander Jansen <s.jansen(a)gmail.com> wrote: > Here the two alsa-info dumps for the hardware I'm having the problem with. > Also a log of my music player. It reports: > > The 'position' the number of frames it has written to the device (also > reported in seconds) and the delay at that time (after writing). > Also the 'timestamp' which is the reported playback progress in seconds > reported to the UI. When the second track is played (line 590), the > problems start to become noticeable eg. playback progress is a couple of > seconds out of sync. > > Any ideas? > > Thanks, > > Sander > > > > On Tue, Jul 16, 2013 at 9:28 AM, Sander Jansen <s.jansen(a)gmail.com> wrote: > >> Hi, >> >> I'm seeing an issue where snd_pcm_delay is returning bogus values >> (usually a lot larger than in actuality) in that it ultimately affects the >> progress timer in my audio player. Both are HDA based audio cards, though >> one connects to the HDMI output and while the other connects to analog out. >> >> I'm suspecting I'm seeing a driver bug since I only start noticing the >> issue since a few kernel releases ago (though not sure which one started >> it). >> >> How can i best debug this issue and find out the cause. >> >> Thanks, >> >> Sander >> > >

1 0

Re: [alsa-devel] [PATCH v2 05/11] ASoC: davinci-mcasp: Remove interrupt property from DT bindings doc
by Mark Brown 18 Sep '13

18 Sep '13

On Wed, Sep 18, 2013 at 09:48:34AM -0500, Nishanth Menon wrote: > On 09/18/2013 09:46 AM, Mark Brown wrote: > > How about moving these to optional instead of removing them completely > > (or just leaving them alone)? That way if the driver does want to start > > using them people will have hooked them up already so no updates will be > > required. > the definitions as they stand now make no sense as the ordering of the > interrupts are undefined - example is the first index RX or TX? the > alternative will be PIO McASP driver, which might be nice, but having > a vague binding with 0 code behind is of no use for anyone right? So the sensible fix seems to be to specify the order rather than remove them then...

1 0

[alsa-devel] [PATCH 1/2] ASoC: SPEAr spdif_in: Use devm_snd_soc_register_component
by Sachin Kamat 18 Sep '13

18 Sep '13

devm_snd_soc_register_component makes code simpler. Signed-off-by: Sachin Kamat <sachin.kamat(a)linaro.org> --- sound/soc/spear/spdif_in.c | 12 ++---------- 1 file changed, 2 insertions(+), 10 deletions(-) diff --git a/sound/soc/spear/spdif_in.c b/sound/soc/spear/spdif_in.c index 63acfeb..21a8c95 100644 --- a/sound/soc/spear/spdif_in.c +++ b/sound/soc/spear/spdif_in.c @@ -257,20 +257,12 @@ static int spdif_in_probe(struct platform_device *pdev) return ret; } - return snd_soc_register_component(&pdev->dev, &spdif_in_component, - &spdif_in_dai, 1); -} - -static int spdif_in_remove(struct platform_device *pdev) -{ - snd_soc_unregister_component(&pdev->dev); - - return 0; + return devm_snd_soc_register_component(&pdev->dev, &spdif_in_component, + &spdif_in_dai, 1); } static struct platform_driver spdif_in_driver = { .probe = spdif_in_probe, - .remove = spdif_in_remove, .driver = { .name = "spdif-in", .owner = THIS_MODULE, -- 1.7.9.5

3 5

[alsa-devel] Incorrect sbits for snd-usb-audio
by Sander Jansen 17 Sep '13

17 Sep '13

Hi, I have a Focusrite Scarlett 2i4 USB Audio Interface. The audio format exposed by ALSA for this device is S32_LE. From the hardware specs and lsusb. it supposed to only have a 24 bit resolution (with the LSB bits set to 0): AudioStreaming Interface Descriptor: bLength 6 bDescriptorType 36 bDescriptorSubtype 2 (FORMAT_TYPE) bFormatType 1 (FORMAT_TYPE_I) bSubslotSize 4 bBitResolution 24 When I query the sbits for this device, I expect it to return 24, but instead it returns 32. Looking through the usb audio driver code, it doesn't look like this information is actually exposed to userspace. The 'parse_audio_format_i_type' in usb/format.c merely returns the supported formats, but doesn't include any msbits info. (and it doesn't look like it can be easily changed to return the msbits info as well). Was this intentionally left out, or is this a missing feature in the driver? Thanks, Sander

1 0

[alsa-devel] [PATCH 0/4] add regmap_filelds and use it on Renesas Sound driver
by Kuninori Morimoto 17 Sep '13

17 Sep '13

Hi Mark These patches add new regmap_filelds API on kernel. ~ and use it on Renesas sound driver instead of original method. It can care about multi port register offset via regmap. 0xAAAA + 0x40 -- port 0 -- regX regY regZ 0xAAAA + 0x80 -- port 1 -- regX regY regZ This case, current API needs 2 (= port) x 3 (= regX/Y/Z) regmap_fileld, but this new API can care about all port via 3 regmap_filelds with port ID. I'm not sure that regmap_filelds is good naming or not. Please let me know if you have good naming idea. These are based on below branchs regmap/for-next + asoc/for-next Kuninori Morimoto (4): regmap: add regmap_field_update_bits() regmap: Add regmap_fields APIs ASoC: rsnd: gen: rsnd_gen_ops cares .probe and .remove ASoC: rsnd: use regmap instead of original register mapping method drivers/base/regmap/internal.h | 3 + drivers/base/regmap/regmap.c | 117 +++++++++++++++++ include/linux/regmap.h | 14 +++ sound/soc/sh/rcar/core.c | 45 ------- sound/soc/sh/rcar/gen.c | 270 +++++++++++++++++++++++++--------------- 5 files changed, 307 insertions(+), 142 deletions(-) Best regards --- Kuninori Morimoto

2 14

[alsa-devel] [PATCH 1/1] ASoC: fsl_ssi: Remove redundant dev_set_drvdata
by Sachin Kamat 17 Sep '13

17 Sep '13

Driver core sets the driver data to NULL on detach. Signed-off-by: Sachin Kamat <sachin.kamat(a)linaro.org> --- sound/soc/fsl/fsl_ssi.c | 1 - 1 file changed, 1 deletion(-) diff --git a/sound/soc/fsl/fsl_ssi.c b/sound/soc/fsl/fsl_ssi.c index 4973be7..6ac8730 100644 --- a/sound/soc/fsl/fsl_ssi.c +++ b/sound/soc/fsl/fsl_ssi.c @@ -1135,7 +1135,6 @@ static int fsl_ssi_remove(struct platform_device *pdev) if (ssi_private->ssi_on_imx) imx_pcm_dma_exit(pdev); snd_soc_unregister_component(&pdev->dev); - dev_set_drvdata(&pdev->dev, NULL); device_remove_file(&pdev->dev, &ssi_private->dev_attr); if (ssi_private->ssi_on_imx) clk_disable_unprepare(ssi_private->clk); -- 1.7.9.5

2 1

[alsa-devel] [PATCH 1/1] ASoC: fsl_spdif: Remove redundant semicolon
by Sachin Kamat 17 Sep '13

17 Sep '13

Redundant semicolon at the end of brace is removed. Signed-off-by: Sachin Kamat <sachin.kamat(a)linaro.org> --- sound/soc/fsl/fsl_spdif.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/soc/fsl/fsl_spdif.c b/sound/soc/fsl/fsl_spdif.c index 3920c3e..c0fea02 100644 --- a/sound/soc/fsl/fsl_spdif.c +++ b/sound/soc/fsl/fsl_spdif.c @@ -963,7 +963,7 @@ static bool fsl_spdif_readable_reg(struct device *dev, unsigned int reg) return true; default: return false; - }; + } } static bool fsl_spdif_writeable_reg(struct device *dev, unsigned int reg) @@ -982,7 +982,7 @@ static bool fsl_spdif_writeable_reg(struct device *dev, unsigned int reg) return true; default: return false; - }; + } } static const struct regmap_config fsl_spdif_regmap_config = { -- 1.7.9.5

2 1

[alsa-devel] [PATCH 1/1] ASoC: ak4642: Remove redundant break
by Sachin Kamat 17 Sep '13

17 Sep '13

'break' after return statement is redundant. Remove it. Signed-off-by: Sachin Kamat <sachin.kamat(a)linaro.org> --- sound/soc/codecs/ak4642.c | 2 -- 1 file changed, 2 deletions(-) diff --git a/sound/soc/codecs/ak4642.c b/sound/soc/codecs/ak4642.c index 2d03787..21c35ed 100644 --- a/sound/soc/codecs/ak4642.c +++ b/sound/soc/codecs/ak4642.c @@ -352,7 +352,6 @@ static int ak4642_dai_set_fmt(struct snd_soc_dai *dai, unsigned int fmt) */ default: return -EINVAL; - break; } snd_soc_update_bits(codec, MD_CTL1, DIF_MASK, data); @@ -405,7 +404,6 @@ static int ak4642_dai_hw_params(struct snd_pcm_substream *substream, break; default: return -EINVAL; - break; } snd_soc_update_bits(codec, MD_CTL2, FS_MASK, rate); -- 1.7.9.5

2 1

[alsa-devel] [PATCH 1/1] ASoC: twl6040: Remove redundant semicolon
by Sachin Kamat 17 Sep '13

17 Sep '13

Redundant semicolon removed. Signed-off-by: Sachin Kamat <sachin.kamat(a)linaro.org> --- sound/soc/codecs/twl6040.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/soc/codecs/twl6040.c b/sound/soc/codecs/twl6040.c index 3c79dbb..35059a2 100644 --- a/sound/soc/codecs/twl6040.c +++ b/sound/soc/codecs/twl6040.c @@ -246,7 +246,7 @@ static bool twl6040_is_path_unmuted(struct snd_soc_codec *codec, return priv->dl2_unmuted; default: return 1; - }; + } } /* @@ -1100,7 +1100,7 @@ static void twl6040_mute_path(struct snd_soc_codec *codec, enum twl6040_dai_id i break; default: break; - }; + } } static int twl6040_digital_mute(struct snd_soc_dai *dai, int mute) -- 1.7.9.5

3 2

Re: [alsa-devel] [PATCH] ASoC: blackfin: Add missing break statement to bf6xx
by Mark Brown 17 Sep '13

17 Sep '13

On Sat, Sep 14, 2013 at 02:20:37AM +0300, Valentin Ilie wrote: > SNDRV_PCM_FORMAT_S8 isn't supposed to fall through to SNDRV_PCM_FORMAT_S16_LE > > Signed-off-by: Valentin Ilie <valentin.ilie(a)gmail.com> Applied, thanks.

1 0

[alsa-devel] [PATCH] ASoC: ak4104: provide a module device table
by Daniel Mack 17 Sep '13

17 Sep '13

Provide a module device table for the SPI subsystem, so the driver can be autoloaded by the SPI core. While at it, get rid of an unnecessary #define. Signed-off-by: Daniel Mack <zonque(a)gmail.com> --- sound/soc/codecs/ak4104.c | 11 ++++++++--- 1 file changed, 8 insertions(+), 3 deletions(-) diff --git a/sound/soc/codecs/ak4104.c b/sound/soc/codecs/ak4104.c index 71059c0..b4819dc 100644 --- a/sound/soc/codecs/ak4104.c +++ b/sound/soc/codecs/ak4104.c @@ -45,8 +45,6 @@ #define AK4104_TX_TXE (1 << 0) #define AK4104_TX_V (1 << 1) -#define DRV_NAME "ak4104-codec" - struct ak4104_private { struct regmap *regmap; }; @@ -291,12 +289,19 @@ static const struct of_device_id ak4104_of_match[] = { }; MODULE_DEVICE_TABLE(of, ak4104_of_match); +static const struct spi_device_id ak4104_id_table[] = { + { "ak4104", 0 }, + { } +}; +MODULE_DEVICE_TABLE(spi, ak4104_id_table); + static struct spi_driver ak4104_spi_driver = { .driver = { - .name = DRV_NAME, + .name = "ak4104", .owner = THIS_MODULE, .of_match_table = ak4104_of_match, }, + .id_table = ak4104_id_table, .probe = ak4104_spi_probe, .remove = ak4104_spi_remove, }; -- 1.8.3.1

2 1

[alsa-devel] [PATCH] ASoC: arizona: Improve handling of setting REFCLK to 0
by Charles Keepax 17 Sep '13

17 Sep '13

This patch suppresses calculation of REFCLK parameters when the REFCLK source frequency is set to zero, additionally it will consider a source frequency of zero as the REFCLK being disabled and switch to using the SYNCCLK. Reported-by: Kyung Kwee Ryu <Kyung-Kwee.Ryu(a)wolfsonmicro.com> Signed-off-by: Charles Keepax <ckeepax(a)opensource.wolfsonmicro.com> --- This patch missed getting upstreamed at the time it was written, sorry about that. Thanks, Charles sound/soc/codecs/arizona.c | 23 ++++++++++++++--------- 1 files changed, 14 insertions(+), 9 deletions(-) diff --git a/sound/soc/codecs/arizona.c b/sound/soc/codecs/arizona.c index 657808b..6f05b17 100644 --- a/sound/soc/codecs/arizona.c +++ b/sound/soc/codecs/arizona.c @@ -1477,21 +1477,25 @@ static void arizona_enable_fll(struct arizona_fll *fll, { struct arizona *arizona = fll->arizona; int ret; + bool use_sync = false; /* * If we have both REFCLK and SYNCCLK then enable both, * otherwise apply the SYNCCLK settings to REFCLK. */ - if (fll->ref_src >= 0 && fll->ref_src != fll->sync_src) { + if (fll->ref_src >= 0 && fll->ref_freq && + fll->ref_src != fll->sync_src) { regmap_update_bits(arizona->regmap, fll->base + 5, ARIZONA_FLL1_OUTDIV_MASK, ref->outdiv << ARIZONA_FLL1_OUTDIV_SHIFT); arizona_apply_fll(arizona, fll->base, ref, fll->ref_src, false); - if (fll->sync_src >= 0) + if (fll->sync_src >= 0) { arizona_apply_fll(arizona, fll->base + 0x10, sync, fll->sync_src, true); + use_sync = true; + } } else if (fll->sync_src >= 0) { regmap_update_bits(arizona->regmap, fll->base + 5, ARIZONA_FLL1_OUTDIV_MASK, @@ -1511,7 +1515,7 @@ static void arizona_enable_fll(struct arizona_fll *fll, * Increase the bandwidth if we're not using a low frequency * sync source. */ - if (fll->sync_src >= 0 && fll->sync_freq > 100000) + if (use_sync && fll->sync_freq > 100000) regmap_update_bits(arizona->regmap, fll->base + 0x17, ARIZONA_FLL1_SYNC_BW, 0); else @@ -1526,8 +1530,7 @@ static void arizona_enable_fll(struct arizona_fll *fll, regmap_update_bits(arizona->regmap, fll->base + 1, ARIZONA_FLL1_ENA, ARIZONA_FLL1_ENA); - if (fll->ref_src >= 0 && fll->sync_src >= 0 && - fll->ref_src != fll->sync_src) + if (use_sync) regmap_update_bits(arizona->regmap, fll->base + 0x11, ARIZONA_FLL1_SYNC_ENA, ARIZONA_FLL1_SYNC_ENA); @@ -1561,10 +1564,12 @@ int arizona_set_fll_refclk(struct arizona_fll *fll, int source, if (fll->ref_src == source && fll->ref_freq == Fref) return 0; - if (fll->fout && Fref > 0) { - ret = arizona_calc_fll(fll, &ref, Fref, fll->fout); - if (ret != 0) - return ret; + if (fll->fout) { + if (Fref > 0) { + ret = arizona_calc_fll(fll, &ref, Fref, fll->fout); + if (ret != 0) + return ret; + } if (fll->sync_src >= 0) { ret = arizona_calc_fll(fll, &sync, fll->sync_freq, -- 1.7.2.5

2 1

[alsa-devel] [PATCH] ASoC: bells: Add missing route to power up DSP clock
by Charles Keepax 17 Sep '13

17 Sep '13

Signed-off-by: Charles Keepax <ckeepax(a)opensource.wolfsonmicro.com> --- sound/soc/samsung/bells.c | 1 + 1 files changed, 1 insertions(+), 0 deletions(-) diff --git a/sound/soc/samsung/bells.c b/sound/soc/samsung/bells.c index 29e2468..84f5d8b 100644 --- a/sound/soc/samsung/bells.c +++ b/sound/soc/samsung/bells.c @@ -356,6 +356,7 @@ static struct snd_soc_dapm_widget bells_widgets[] = { static struct snd_soc_dapm_route bells_routes[] = { { "Sub CLK_SYS", NULL, "OPCLK" }, + { "CLKIN", NULL, "OPCLK" }, { "DMIC", NULL, "MICBIAS2" }, { "IN2L", NULL, "DMIC" }, -- 1.7.2.5

2 1

[alsa-devel] [PATCH] ASoC: core: Add API for configuration of DAI BCLK ratio
by Liam Girdwood 17 Sep '13

17 Sep '13

Some codec drivers when running in slave mode require that BCLK to sample rate ratio is explicitly set by the machine driver as it may not be exactly rate * frame size. Extend the DAI API by adding :- int snd_soc_dai_set_bclk_ratio(struct snd_soc_dai *dai, unsigned int ratio); Signed-off-by: Liam Girdwood <liam.r.girdwood(a)linux.intel.com> --- include/sound/soc-dai.h | 3 +++ sound/soc/soc-core.c | 16 ++++++++++++++++ 2 files changed, 19 insertions(+) diff --git a/include/sound/soc-dai.h b/include/sound/soc-dai.h index ae9a227..d8acf0c 100644 --- a/include/sound/soc-dai.h +++ b/include/sound/soc-dai.h @@ -105,6 +105,8 @@ int snd_soc_dai_set_clkdiv(struct snd_soc_dai *dai, int snd_soc_dai_set_pll(struct snd_soc_dai *dai, int pll_id, int source, unsigned int freq_in, unsigned int freq_out); +int snd_soc_dai_set_bclk_ratio(struct snd_soc_dai *dai, unsigned int ratio); + /* Digital Audio interface formatting */ int snd_soc_dai_set_fmt(struct snd_soc_dai *dai, unsigned int fmt); @@ -131,6 +133,7 @@ struct snd_soc_dai_ops { int (*set_pll)(struct snd_soc_dai *dai, int pll_id, int source, unsigned int freq_in, unsigned int freq_out); int (*set_clkdiv)(struct snd_soc_dai *dai, int div_id, int div); + int (*set_bclk_ratio)(struct snd_soc_dai *dai, unsigned int ratio); /* * DAI format configuration diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index 09e341d..8014839 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -3380,6 +3380,22 @@ int snd_soc_codec_set_pll(struct snd_soc_codec *codec, int pll_id, int source, EXPORT_SYMBOL_GPL(snd_soc_codec_set_pll); /** + * snd_soc_dai_set_bclk_ratio - configure BCLK to sample rate ratio. + * @dai: DAI + * @ratio Ratio of BCLK to Sample rate. + * + * Configures the DAI for a preset BCLK to sample rate ratio. + */ +int snd_soc_dai_set_bclk_ratio(struct snd_soc_dai *dai, unsigned int ratio) +{ + if (dai->driver && dai->driver->ops->set_bclk_ratio) + return dai->driver->ops->set_bclk_ratio(dai, ratio); + else + return -EINVAL; +} +EXPORT_SYMBOL_GPL(snd_soc_dai_set_bclk_ratio); + +/** * snd_soc_dai_set_fmt - configure DAI hardware audio format. * @dai: DAI * @fmt: SND_SOC_DAIFMT_ format value. -- 1.8.1.2

2 1

[alsa-devel] USB Audio initialization race
by David Henningsson 17 Sep '13

17 Sep '13

I'm sometimes seeing that USB audio devices are not initialized correctly by PulseAudio, but I can't reproduce this myself. I have, however, spotted what I think is a race condition in the driver code, if a device has more than one interface. snd_card_register is called once for every interface, which means that it will be shown to userspace after the first interface has been probed. So the race could look like: 1) The first interface is probed, which might contain the mixer controls 2) snd_card_register is called, which sends a signal to userspace 3) PulseAudio probes the device and notices that it has no PCMs, so it ignores the device 4) The second interface is probed, which adds the PCMs 5) snd_card_register is again called, which is a no-op since the card already exist 6) User is unhappy because his plugged in sound card did not show up in PulseAudio. So, assuming all this is right, it seems like we need some type of callback from the usb driver when all the interfaces for the sound card has been probed, so we can call snd_card_register at that point instead. Thoughts? -- David Henningsson, Canonical Ltd. https://launchpad.net/~diwic

3 10

[alsa-devel] [PATCH 1/2] ASoC: core: Add devm_snd_soc_register_card()
by Mark Brown 16 Sep '13

16 Sep '13

From: Mark Brown <broonie(a)linaro.org> Simplify error handling and remove repetitive (and rarely executed) code for unregistration by providing a devm_snd_soc_register() card. Signed-off-by: Mark Brown <broonie(a)linaro.org> --- include/sound/soc.h | 1 + sound/soc/soc-devres.c | 34 ++++++++++++++++++++++++++++++++++ 2 files changed, 35 insertions(+) diff --git a/include/sound/soc.h b/include/sound/soc.h index f126676..7ab5d2b 100644 --- a/include/sound/soc.h +++ b/include/sound/soc.h @@ -364,6 +364,7 @@ int snd_soc_codec_set_pll(struct snd_soc_codec *codec, int pll_id, int source, int snd_soc_register_card(struct snd_soc_card *card); int snd_soc_unregister_card(struct snd_soc_card *card); +int devm_snd_soc_register_card(struct device *dev, struct snd_soc_card *card); int snd_soc_suspend(struct device *dev); int snd_soc_resume(struct device *dev); int snd_soc_poweroff(struct device *dev); diff --git a/sound/soc/soc-devres.c b/sound/soc/soc-devres.c index 13fe86f..b1d7322 100644 --- a/sound/soc/soc-devres.c +++ b/sound/soc/soc-devres.c @@ -50,3 +50,37 @@ int devm_snd_soc_register_component(struct device *dev, return ret; } EXPORT_SYMBOL_GPL(devm_snd_soc_register_component); + +static void devm_card_release(struct device *dev, void *res) +{ + snd_soc_unregister_card(*(struct snd_soc_card **)res); +} + +/** + * devm_snd_soc_register_card - resource managed card registration + * @dev: Device used to manage card + * @card: Card to register + * + * Register a card with automatic unregistration when the device is + * unregistered. + */ +int devm_snd_soc_register_card(struct device *dev, struct snd_soc_card *card) +{ + struct device **ptr; + int ret; + + ptr = devres_alloc(devm_card_release, sizeof(*ptr), GFP_KERNEL); + if (!ptr) + return -ENOMEM; + + ret = snd_soc_register_card(card); + if (ret == 0) { + *ptr = dev; + devres_add(dev, ptr); + } else { + devres_free(ptr); + } + + return ret; +} +EXPORT_SYMBOL_GPL(devm_snd_soc_register_card); -- 1.8.4.rc3

2 2

[alsa-devel] [PATCH] ASoC: core: Only add platform DAI widgets once.
by Liam Girdwood 16 Sep '13

16 Sep '13

Currently platform CPU DAI widgets are created in soc_probe_platform and soc_probe_link_dais. Remove the extra call in soc_probe_link_dais(). Signed-off-by: Liam Girdwood <liam.r.girdwood(a)linux.intel.com> --- sound/soc/soc-core.c | 1 - 1 file changed, 1 deletion(-) diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index 8e86aba..09e341d 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -1365,7 +1365,6 @@ static int soc_probe_link_dais(struct snd_soc_card *card, int num, int order) return -ENODEV; list_add(&cpu_dai->dapm.list, &card->dapm_list); - snd_soc_dapm_new_dai_widgets(&cpu_dai->dapm, cpu_dai); } if (cpu_dai->driver->probe) { -- 1.8.1.2

2 10

Re: [alsa-devel] No sound from internal speaker on 2013 Macbook Air 6, 1
by Ben Whitten 16 Sep '13

16 Sep '13

> > > OK, then the normal jack detection doesn't work with this machine, as > > > it seems. You can try the above to all pins for checking if any of > > > other pins reacts to the headphone jack. > > > > No luck I'm afraid - none of the other pins seem to react to the > > headphones. Is there anything else you can think of that I can try > > tomorrow? > > Well, it might be notified via GPIO. Try to read GPIO pins except for > GPIO0. That is, set the all unmasked, the direction to READ. We have found that jack sense comes in on the headphone pin in a windows bootcamp install. However this bit isn't set in linux using either the default chip, bootloader, or windows pin configs. No other GPIO's are enabled in windows. Ben

1 0

[alsa-devel] Roland UA-55 (snd-usb-audio)
by Michael Findlater 15 Sep '13

15 Sep '13

Hello there! I am attempting to use the Roland UA-55, which apparently should work after recent changes were made for support for Roland devices by Clemens Ladisch. I am using the 3.11 kernel upon Debian sid. The device shows in /proc/asound/cards but does not seem to work. The GNOME volume mixer it shows the device as a "QUAD-CAPTURE Analog Surround 4.0", but I am not sure whether this is correct. When I select the card as the output device and try to play to it, nothing will play, 'aplay' does not work either (I hear nothing). If I turn the headphone output knob on the device up I hear a very soft static in my right ear. 'alsamixer' reports that 'This sound device does not have any controls.' I do not know where to go from here, but have ran the "Alsa Information Script", an output of this is available at: http://www.alsa-project.org/db/?f=8f62ce460ecc5cac394c0b014b2e2aa4d28ae2dc I am stuck with how to proceed to debug the issue. If at all possible, I would be grateful if someone is able to point me in the right direction. Best regards, Michael Findlater

1 0

[alsa-devel] [PATCH RFC 0/8] Fix AM335x-evm analog audio support
by oku＠iki.fi 13 Sep '13

13 Sep '13

From: Jyri Sarha <jsarha(a)ti.com> This set of patches fixes the basic audio support for am335x-evm. It should also be relatively simple to add the necessary nodes to relevant dts files to get BeagleBone + AudioCape and am335x-evmsk working too. The patch set depends on following patches: [PATCH v2] ARM: dts: add AM33XX EDMA support https://lkml.org/lkml/2013/8/26/57 [PATCH v11 5/8] ARM: dts: add AM33XX SPI DMA support https://lkml.org/lkml/2013/6/18/55 [PATCH v2] ARM: EDMA: Fix clearing of unused list for DT DMA resources https://lkml.org/lkml/2013/7/22/441 I have tried my best not to break the existing support for older davinci boards, but since I do not have those boards I can not be sure. Some commit comments refer to a dmaengine based davinci audio implementation which is planned for but nothing has been done yet. Best regards, Jyri Darren Etheridge (1): ARM/dts: am335x-evm: Add audio support for am335x-evm.dts Hebbar, Gururaja (2): ASoC: davinci-evm: Add device tree binding ASoC: davinci: Add support for AM33xx SoC Audio Joel A Fernandes (1): ASoC: davinci-mcasp: Add pinctrl support Jyri Sarha (3): ASoC: davinci-mcasp: Get DMA related properties from DT ASoC: davinci-mcasp: Extract DMA channels directly from DT ARM/dts: am33xx: mcasp: Add new dma related properties Pantelis Antoniou (1): ARM/dts: am33xx: Add mcasp0 and mcasp1 device tree entries .../bindings/sound/davinci-evm-audio.txt | 53 ++++++ .../bindings/sound/davinci-mcasp-audio.txt | 13 +- arch/arm/boot/dts/am335x-evm.dts | 56 ++++++ arch/arm/boot/dts/am33xx.dtsi | 35 ++++ include/linux/platform_data/davinci_asp.h | 2 + sound/soc/davinci/Kconfig | 8 + sound/soc/davinci/Makefile | 3 + sound/soc/davinci/davinci-evm.c | 181 +++++++++++++++++--- sound/soc/davinci/davinci-mcasp.c | 82 +++++++-- 9 files changed, 391 insertions(+), 42 deletions(-) create mode 100644 Documentation/devicetree/bindings/sound/davinci-evm-audio.txt -- 1.7.9.5

4 25

[alsa-devel] [patch] ASoC: ab8500-codec: info leak in anc_status_control_put()
by Dan Carpenter 13 Sep '13

13 Sep '13

If the user passes an invalid value it leads to an info leak when we print the error message or it could oops. This is called with user supplied data from snd_ctl_elem_write(). Signed-off-by: Dan Carpenter <dan.carpenter(a)oracle.com> diff --git a/sound/soc/codecs/ab8500-codec.c b/sound/soc/codecs/ab8500-codec.c index b8ba0ad..80555d7 100644 --- a/sound/soc/codecs/ab8500-codec.c +++ b/sound/soc/codecs/ab8500-codec.c @@ -1225,13 +1225,18 @@ static int anc_status_control_put(struct snd_kcontrol *kcontrol, struct ab8500_codec_drvdata *drvdata = dev_get_drvdata(codec->dev); struct device *dev = codec->dev; bool apply_fir, apply_iir; - int req, status; + unsigned int req; + int status; dev_dbg(dev, "%s: Enter.\n", __func__); mutex_lock(&drvdata->anc_lock); req = ucontrol->value.integer.value[0]; + if (req >= ARRAY_SIZE(enum_anc_state)) { + status = -EINVAL; + goto cleanup; + } if (req != ANC_APPLY_FIR_IIR && req != ANC_APPLY_FIR && req != ANC_APPLY_IIR) { dev_err(dev, "%s: ERROR: Unsupported status to set '%s'!\n",

2 1

[alsa-devel] [patch] ASoC: 88pm860x: array overflow in snd_soc_put_volsw_2r_st()
by Dan Carpenter 13 Sep '13

13 Sep '13

This is called from snd_ctl_elem_write() with user supplied data so we need to add some bounds checking. Signed-off-by: Dan Carpenter <dan.carpenter(a)oracle.com> diff --git a/sound/soc/codecs/88pm860x-codec.c b/sound/soc/codecs/88pm860x-codec.c index 8af0434..259d1ac 100644 --- a/sound/soc/codecs/88pm860x-codec.c +++ b/sound/soc/codecs/88pm860x-codec.c @@ -349,6 +349,9 @@ static int snd_soc_put_volsw_2r_st(struct snd_kcontrol *kcontrol, val = ucontrol->value.integer.value[0]; val2 = ucontrol->value.integer.value[1]; + if (val >= ARRAY_SIZE(st_table) || val2 >= ARRAY_SIZE(st_table)) + return -EINVAL; + err = snd_soc_update_bits(codec, reg, 0x3f, st_table[val].m); if (err < 0) return err;

2 1

[alsa-devel] [patch] ASoC: max98095: a couple array underflows
by Dan Carpenter 13 Sep '13

13 Sep '13

The ->put() function are called from snd_ctl_elem_write() with user supplied data. The limit checks here could underflow leading to a crash. Signed-off-by: Dan Carpenter <dan.carpenter(a)oracle.com> diff --git a/sound/soc/codecs/max98095.c b/sound/soc/codecs/max98095.c index 41cdd16..8dbcacd 100644 --- a/sound/soc/codecs/max98095.c +++ b/sound/soc/codecs/max98095.c @@ -1863,7 +1863,7 @@ static int max98095_put_eq_enum(struct snd_kcontrol *kcontrol, struct max98095_pdata *pdata = max98095->pdata; int channel = max98095_get_eq_channel(kcontrol->id.name); struct max98095_cdata *cdata; - int sel = ucontrol->value.integer.value[0]; + unsigned int sel = ucontrol->value.integer.value[0]; struct max98095_eq_cfg *coef_set; int fs, best, best_val, i; int regmask, regsave; @@ -2016,7 +2016,7 @@ static int max98095_put_bq_enum(struct snd_kcontrol *kcontrol, struct max98095_pdata *pdata = max98095->pdata; int channel = max98095_get_bq_channel(codec, kcontrol->id.name); struct max98095_cdata *cdata; - int sel = ucontrol->value.integer.value[0]; + unsigned int sel = ucontrol->value.integer.value[0]; struct max98095_biquad_cfg *coef_set; int fs, best, best_val, i; int regmask, regsave;

2 1

[alsa-devel] [PATCH] ASoC: rt5640: Provide more useful hw_params error reasons.
by Liam Girdwood 13 Sep '13

13 Sep '13

Signed-off-by: Liam Girdwood <liam.r.girdwood(a)linux.intel.com> --- sound/soc/codecs/rt5640.c | 3 ++- 1 file changed, 2 insertions(+), 1 deletion(-) diff --git a/sound/soc/codecs/rt5640.c b/sound/soc/codecs/rt5640.c index 2f6bb16..de40bd9 100644 --- a/sound/soc/codecs/rt5640.c +++ b/sound/soc/codecs/rt5640.c @@ -1609,7 +1609,8 @@ static int rt5640_hw_params(struct snd_pcm_substream *substream, rt5640->lrck[dai->id] = params_rate(params); pre_div = get_clk_info(rt5640->sysclk, rt5640->lrck[dai->id]); if (pre_div < 0) { - dev_err(codec->dev, "Unsupported clock setting\n"); + dev_err(codec->dev, "Unsupported clock setting %d for DAI %d\n", + rt5640->lrck[dai->id], dai->id); return -EINVAL; } frame_size = snd_soc_params_to_frame_size(params); -- 1.8.1.2

2 2

[alsa-devel] virtual alsa mixer device
by André Roth 13 Sep '13

13 Sep '13

hello, I'd like to implement a mixer control device for an external amplifier, which can be controlled via rs232. would be nice to adjust volume with any alsa mixer program.. is there a way to do this in user space ? any hints for the right api / examples ? -- andré

1 0

[alsa-devel] [PATCH] ASoC: codecs: da9055: Update driver name to fix breakage due to pmic driver with same name
by Ashish Chavan 13 Sep '13

13 Sep '13

This patch updates i2c driver name and device id of da9055 codec driver. DA9055 is a PMIC + CODEC and currently, the corresponding PMIC driver also registers itself with the same name as codec, i.e. "da9055". Because of this the codec driver was broken. Now codec driver uses "da9055-codec" as driver name instead of "da9055". Signed-off-by: Ashish Chavan <ashish.chavan(a)kpitcummins.com> Signed-off-by: David Dajun Chen <david.chen(a)diasemi.com> --- sound/soc/codecs/da9055.c | 4 ++-- 1 files changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/soc/codecs/da9055.c b/sound/soc/codecs/da9055.c index 2e1bf5d..e146f68 100644 --- a/sound/soc/codecs/da9055.c +++ b/sound/soc/codecs/da9055.c @@ -1544,7 +1544,7 @@ static int da9055_remove(struct i2c_client *client) } static const struct i2c_device_id da9055_i2c_id[] = { - { "da9055", 0 }, + { "da9055-codec", 0 }, { } }; MODULE_DEVICE_TABLE(i2c, da9055_i2c_id); @@ -1552,7 +1552,7 @@ MODULE_DEVICE_TABLE(i2c, da9055_i2c_id); /* I2C codec control layer */ static struct i2c_driver da9055_i2c_driver = { .driver = { - .name = "da9055", + .name = "da9055-codec", .owner = THIS_MODULE, }, .probe = da9055_i2c_probe, -- 1.7.1

3 29

[alsa-devel] No sound from internal speaker on 2013 Macbook Air 6, 1
by Ian Munsie 13 Sep '13

13 Sep '13

Hey all, I'm trying to get sound working on my new mid 2013 Macbook Air 6,1. Sound works via the headphone jack, but I haven't managed to get the internal speaker to work. I've checked all the controls in the mixer, tried a few different mac related model= options to snd-hda-intel and even tried tweaking things with hda_analyzer to no avail. I've uninstalled pulseaudio (for now) to get rid of it's ALSA hook, but had the same result both with and without it. I'm running on a 3.11-rc6+ built today (had a few other fixes to make for this machine) - I didn't bother including the hda i915 driver as I don't need sound via the display port, but the hda driver does print a warning about that missing module. The speaker wasn't working on the 3.10 kernel from Debian either which included that module. Any help would be appreciated - I've attached my alsa-info output taken after a reboot. Cheers, -Ian -- http://sites.google.com/site/DarkStarJunkSpace -- http://darkstarshout.blogspot.com/ -- Please avoid sending me Word or PowerPoint attachments. See http://www.gnu.org/philosophy/no-word-attachments.html

3 15

[alsa-devel] [PATCH 1/1] ALSA: rme9652: Remove redundant break
by Sachin Kamat 13 Sep '13

13 Sep '13

'break' after return statement is not necessary. Signed-off-by: Sachin Kamat <sachin.kamat(a)linaro.org> --- sound/pci/rme9652/hdspm.c | 5 ----- 1 file changed, 5 deletions(-) diff --git a/sound/pci/rme9652/hdspm.c b/sound/pci/rme9652/hdspm.c index 3cde55b..2907e68 100644 --- a/sound/pci/rme9652/hdspm.c +++ b/sound/pci/rme9652/hdspm.c @@ -3996,7 +3996,6 @@ static int hdspm_tco_sync_check(struct hdspm *hdspm) return 1; } return 0; - break; case AES32: status = hdspm_read(hdspm, HDSPM_statusRegister); if (status & HDSPM_tcoLockAes) { @@ -4006,9 +4005,6 @@ static int hdspm_tco_sync_check(struct hdspm *hdspm) return 1; } return 0; - - break; - case RayDAT: case AIO: status = hdspm_read(hdspm, HDSPM_RD_STATUS_1); @@ -4018,7 +4014,6 @@ static int hdspm_tco_sync_check(struct hdspm *hdspm) if (status & 0x4000000) return 1; /* Lock */ return 0; /* No signal */ - break; default: break; -- 1.7.9.5

2 1

[alsa-devel] [PATCH 1/1] ALSA: au88x0: Remove redundant break
by Sachin Kamat 13 Sep '13

13 Sep '13

'break' after a return statement is redundant. Remove it. Signed-off-by: Sachin Kamat <sachin.kamat(a)linaro.org> --- sound/pci/au88x0/au88x0_synth.c | 29 ++++++++++------------------- 1 file changed, 10 insertions(+), 19 deletions(-) diff --git a/sound/pci/au88x0/au88x0_synth.c b/sound/pci/au88x0/au88x0_synth.c index 8bef473..922a84b 100644 --- a/sound/pci/au88x0/au88x0_synth.c +++ b/sound/pci/au88x0/au88x0_synth.c @@ -219,7 +219,6 @@ vortex_wt_SetReg(vortex_t * vortex, unsigned char reg, int wt, */ hwwrite(vortex->mmio, WT_RUN(wt), val); return 0xc; - break; case 1: /* param 0 */ /* printk(KERN_DEBUG "vortex: WT SetReg(0x%x) = 0x%08x\n", @@ -227,7 +226,6 @@ vortex_wt_SetReg(vortex_t * vortex, unsigned char reg, int wt, */ hwwrite(vortex->mmio, WT_PARM(wt, 0), val); return 0xc; - break; case 2: /* param 1 */ /* printk(KERN_DEBUG "vortex: WT SetReg(0x%x) = 0x%08x\n", @@ -235,7 +233,6 @@ vortex_wt_SetReg(vortex_t * vortex, unsigned char reg, int wt, */ hwwrite(vortex->mmio, WT_PARM(wt, 1), val); return 0xc; - break; case 3: /* param 2 */ /* printk(KERN_DEBUG "vortex: WT SetReg(0x%x) = 0x%08x\n", @@ -243,7 +240,6 @@ vortex_wt_SetReg(vortex_t * vortex, unsigned char reg, int wt, */ hwwrite(vortex->mmio, WT_PARM(wt, 2), val); return 0xc; - break; case 4: /* param 3 */ /* printk(KERN_DEBUG "vortex: WT SetReg(0x%x) = 0x%08x\n", @@ -251,7 +247,6 @@ vortex_wt_SetReg(vortex_t * vortex, unsigned char reg, int wt, */ hwwrite(vortex->mmio, WT_PARM(wt, 3), val); return 0xc; - break; case 6: /* mute */ /* printk(KERN_DEBUG "vortex: WT SetReg(0x%x) = 0x%08x\n", @@ -259,20 +254,17 @@ vortex_wt_SetReg(vortex_t * vortex, unsigned char reg, int wt, */ hwwrite(vortex->mmio, WT_MUTE(wt), val); return 0xc; - break; case 0xb: - { /* delay */ - /* - printk(KERN_DEBUG "vortex: WT SetReg(0x%x) = 0x%08x\n", - WT_DELAY(wt,0), (int)val); - */ - hwwrite(vortex->mmio, WT_DELAY(wt, 3), val); - hwwrite(vortex->mmio, WT_DELAY(wt, 2), val); - hwwrite(vortex->mmio, WT_DELAY(wt, 1), val); - hwwrite(vortex->mmio, WT_DELAY(wt, 0), val); - return 0xc; - } - break; + /* delay */ + /* + printk(KERN_DEBUG "vortex: WT SetReg(0x%x) = 0x%08x\n", + WT_DELAY(wt,0), (int)val); + */ + hwwrite(vortex->mmio, WT_DELAY(wt, 3), val); + hwwrite(vortex->mmio, WT_DELAY(wt, 2), val); + hwwrite(vortex->mmio, WT_DELAY(wt, 1), val); + hwwrite(vortex->mmio, WT_DELAY(wt, 0), val); + return 0xc; /* Global WT block parameters */ case 5: /* sramp */ ecx = WT_SRAMP(wt); @@ -291,7 +283,6 @@ vortex_wt_SetReg(vortex_t * vortex, unsigned char reg, int wt, break; default: return 0; - break; } /* printk(KERN_DEBUG "vortex: WT SetReg(0x%x) = 0x%08x\n", ecx, (int)val); -- 1.7.9.5

2 1

[alsa-devel] [PATCH 1/2] ALSA: ctxfi: Staticize local symbols
by Sachin Kamat 13 Sep '13

13 Sep '13

Local symbols used only in this file are made static. Signed-off-by: Sachin Kamat <sachin.kamat(a)linaro.org> --- sound/pci/ctxfi/ctdaio.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/pci/ctxfi/ctdaio.c b/sound/pci/ctxfi/ctdaio.c index 0c00eb4..84f86bf 100644 --- a/sound/pci/ctxfi/ctdaio.c +++ b/sound/pci/ctxfi/ctdaio.c @@ -33,7 +33,7 @@ struct daio_rsc_idx { unsigned short right; }; -struct daio_rsc_idx idx_20k1[NUM_DAIOTYP] = { +static struct daio_rsc_idx idx_20k1[NUM_DAIOTYP] = { [LINEO1] = {.left = 0x00, .right = 0x01}, [LINEO2] = {.left = 0x18, .right = 0x19}, [LINEO3] = {.left = 0x08, .right = 0x09}, @@ -44,7 +44,7 @@ struct daio_rsc_idx idx_20k1[NUM_DAIOTYP] = { [SPDIFI1] = {.left = 0x95, .right = 0x9d}, }; -struct daio_rsc_idx idx_20k2[NUM_DAIOTYP] = { +static struct daio_rsc_idx idx_20k2[NUM_DAIOTYP] = { [LINEO1] = {.left = 0x40, .right = 0x41}, [LINEO2] = {.left = 0x60, .right = 0x61}, [LINEO3] = {.left = 0x50, .right = 0x51}, -- 1.7.9.5

2 2

[alsa-devel] [patch] ALSA: asihpi: a couple array out of bounds issues
by Dan Carpenter 13 Sep '13

13 Sep '13

These ->put() functions are called from snd_ctl_elem_write() with user supplied data. snd_asihpi_tuner_band_put() is missing a limit check and the check in snd_asihpi_clksrc_put() can underflow. Signed-off-by: Dan Carpenter <dan.carpenter(a)oracle.com> diff --git a/sound/pci/asihpi/asihpi.c b/sound/pci/asihpi/asihpi.c index dc632cd..5f2acd3 100644 --- a/sound/pci/asihpi/asihpi.c +++ b/sound/pci/asihpi/asihpi.c @@ -1913,6 +1913,7 @@ static int snd_asihpi_tuner_band_put(struct snd_kcontrol *kcontrol, struct snd_card_asihpi *asihpi = snd_kcontrol_chip(kcontrol); */ u32 h_control = kcontrol->private_value; + unsigned int idx; u16 band; u16 tuner_bands[HPI_TUNER_BAND_LAST]; u32 num_bands = 0; @@ -1920,7 +1921,10 @@ static int snd_asihpi_tuner_band_put(struct snd_kcontrol *kcontrol, num_bands = asihpi_tuner_band_query(kcontrol, tuner_bands, HPI_TUNER_BAND_LAST); - band = tuner_bands[ucontrol->value.enumerated.item[0]]; + idx = ucontrol->value.enumerated.item[0]; + if (idx >= ARRAY_SIZE(tuner_bands)) + idx = ARRAY_SIZE(tuner_bands) - 1; + band = tuner_bands[idx]; hpi_handle_error(hpi_tuner_set_band(h_control, band)); return 1; @@ -2383,7 +2387,8 @@ static int snd_asihpi_clksrc_put(struct snd_kcontrol *kcontrol, struct snd_card_asihpi *asihpi = (struct snd_card_asihpi *)(kcontrol->private_data); struct clk_cache *clkcache = &asihpi->cc; - int change, item; + unsigned int item; + int change; u32 h_control = kcontrol->private_value; change = 1;

2 1

[alsa-devel] [PATCH 1/2] ASoC: fsl_ssi: Staticize local symbols
by Sachin Kamat 13 Sep '13

13 Sep '13

Local symbols used only in this file are made static. Signed-off-by: Sachin Kamat <sachin.kamat(a)linaro.org> --- sound/soc/fsl/fsl_ssi.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/soc/fsl/fsl_ssi.c b/sound/soc/fsl/fsl_ssi.c index c6b7439..4973be7 100644 --- a/sound/soc/fsl/fsl_ssi.c +++ b/sound/soc/fsl/fsl_ssi.c @@ -748,7 +748,7 @@ static void fsl_ssi_ac97_init(void) fsl_ssi_setup(fsl_ac97_data); } -void fsl_ssi_ac97_write(struct snd_ac97 *ac97, unsigned short reg, +static void fsl_ssi_ac97_write(struct snd_ac97 *ac97, unsigned short reg, unsigned short val) { struct ccsr_ssi *ssi = fsl_ac97_data->ssi; @@ -770,7 +770,7 @@ void fsl_ssi_ac97_write(struct snd_ac97 *ac97, unsigned short reg, udelay(100); } -unsigned short fsl_ssi_ac97_read(struct snd_ac97 *ac97, +static unsigned short fsl_ssi_ac97_read(struct snd_ac97 *ac97, unsigned short reg) { struct ccsr_ssi *ssi = fsl_ac97_data->ssi; -- 1.7.9.5

2 3

[alsa-devel] [PATCH] ASoC: SAMSUNG: Allow mono in i2s driver
by Charles Keepax 13 Sep '13

13 Sep '13

Signed-off-by: Charles Keepax <ckeepax(a)opensource.wolfsonmicro.com> --- I can't see any reason to not allow mono playback in the i2s driver and indeed it seems to work well on our test system here. However, I am fairly unfamilar with the code in this part of the tree so apologies if I am missing some constraint. Thanks, Charles sound/soc/samsung/i2s.c | 2 +- 1 files changed, 1 insertions(+), 1 deletions(-) diff --git a/sound/soc/samsung/i2s.c b/sound/soc/samsung/i2s.c index 959c702..70d08d7 100644 --- a/sound/soc/samsung/i2s.c +++ b/sound/soc/samsung/i2s.c @@ -991,7 +991,7 @@ static struct i2s_dai *i2s_alloc_dai(struct platform_device *pdev, bool sec) i2s->i2s_dai_drv.ops = &samsung_i2s_dai_ops; i2s->i2s_dai_drv.suspend = i2s_suspend; i2s->i2s_dai_drv.resume = i2s_resume; - i2s->i2s_dai_drv.playback.channels_min = 2; + i2s->i2s_dai_drv.playback.channels_min = 1; i2s->i2s_dai_drv.playback.channels_max = 2; i2s->i2s_dai_drv.playback.rates = SAMSUNG_I2S_RATES; i2s->i2s_dai_drv.playback.formats = SAMSUNG_I2S_FMTS; -- 1.7.2.5

2 1

[alsa-devel] USB Audio/ALSA
by Rajaram R 13 Sep '13

13 Sep '13

Hi I am trying to bring up Android Accessory Audio in my Samsung Grand mobile.. I am using a Linux PC for the same. The mobile re-enumerates in AOA mode with an Audio interface. But when I play song, I see nothing on the bus. I can see an AOA audio device entry in /proc/asound/cards. The mobile runs Android JB The error i see when I play is "Sorry this player does not support this type of audio file android" . I also tried a Note 2 & tab 2 with JB, i dont see the error , but still i dont see anything on the bus.. Nexus is working fine in the same setup. Also logcat shows information as below. When i search net i understand some problem with ALSA/Android handshake AudioTrack: obtainBuffer timed out (is the CPU pegged?).......some address............ AudioTrack: obtainBuffer timed out (is the CPU pegged?).......some address............ AudioTrack: obtainBuffer timed out (is the CPU pegged?).......some address............ AudioTrack: obtainBuffer timed out (is the CPU pegged?).......some address............ Looking forward for some support

4 5

[alsa-devel] [PATCH 1/2] snd-ac97-codec: Add ID for TI TLV320AIC27 codec
by Ondrej Zary 12 Sep '13

12 Sep '13

Add 0x54584e03 ID for TI TLV320AIC27 AC'97 codec according to datasheet: http://www.ti.com/lit/ds/slas253a/slas253a.pdf The weird thing is that the chip is physically marked 320AD91. Signed-off-by: Ondrej Zary <linux(a)rainbow-software.org> --- sound/pci/ac97/ac97_codec.c | 1 + 1 files changed, 1 insertions(+), 0 deletions(-) diff --git a/sound/pci/ac97/ac97_codec.c b/sound/pci/ac97/ac97_codec.c index 445ca48..bf578ba2 100644 --- a/sound/pci/ac97/ac97_codec.c +++ b/sound/pci/ac97/ac97_codec.c @@ -175,6 +175,7 @@ static const struct ac97_codec_id snd_ac97_codec_ids[] = { { 0x54524106, 0xffffffff, "TR28026", NULL, NULL }, { 0x54524108, 0xffffffff, "TR28028", patch_tritech_tr28028, NULL }, // added by xin jin [07/09/99] { 0x54524123, 0xffffffff, "TR28602", NULL, NULL }, // only guess --jk [TR28023 = eMicro EM28023 (new CT1297)] +{ 0x54584e03, 0xffffffff, "TLV320AIC27", NULL, NULL }, { 0x54584e20, 0xffffffff, "TLC320AD9xC", NULL, NULL }, { 0x56494161, 0xffffffff, "VIA1612A", NULL, NULL }, // modified ICE1232 with S/PDIF { 0x56494170, 0xffffffff, "VIA1617A", patch_vt1617a, NULL }, // modified VT1616 with S/PDIF -- Ondrej Zary

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Re: [alsa-devel] linux-next: Tree for Sep 12 (sound/soc/)
by Randy Dunlap 12 Sep '13

12 Sep '13

On 09/11/13 21:34, Stephen Rothwell wrote: > Hi all, > > Please do not add any code for v3.13 to your linux-next included branches > until after v3.12-rc1 is released. > > Changes since 20130911: > on i386: when CONFIG_REGMAP_I2C=m and CONFIG_SND_SOC=y sound/built-in.o: In function `snd_soc_codec_set_cache_io': (.text+0x2dbb5): undefined reference to `regmap_init_i2c' Full randconfig file is attached. -- ~Randy

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[alsa-devel] Hello
by Michael Jordan 12 Sep '13

12 Sep '13

Hello Gururaja, If I am trying to ask some questions about the alsa system, do I need to copy somebody directly to get a response? I have posted about 4 questions on the group in the last few months, and I have not even gotten a slap on the hand. The reason I am contacting you is you have mentioned working on the TLV320AIC3X which is the chip used on the OMAPL138 board that we have. I am attempting to write a driver for a PCM3168A which we have already constructed a board for, and plugs into the 100 pin audio expander connecter. I have the basic draft of a driver, but everything in the code seems to be fairly strongly coded for the DA850 boards to use the TLV320AIC. I can prevent it loading, and my module will load, but things are not "connecting" internally. I assumed that I would need to make some changes in the arch/arm/mach-davinci/board-da850-evm.c. However what I am running into is chasing me up a trail that eventually has me modifying mach-types.h and creating a whole new board configuration. I said ok, so I copied board-da850-evm.c to board-da850-evm-mod.c which made it necessary for me to begin modifying some Kconfig's, etc., and I still cannot get the kernel to compile with the new kernel driver. I just need to know, am I heading down the wrong path with this? I have a pcm3168a.c and pcm3168a.h which have a basic structure for the new codec, but I cannot figure out how to get the system to actually use it because it basically appears that the TLV320AIC3X is "hard-wired" into the system. Help please, Michael

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[alsa-devel] Fwd: Lynx HiLo USB DAC issues
by Radoslaw Szkodzinski 12 Sep '13

12 Sep '13

Hello, I'm now an owner of a Lynx Hilo USB DAC/ADC. It is supposedly USB Audio Class 2 compatible, but it does require an advanced driver on both Windows and Mac OS X. The device has a complex UAC2 mixer and extra features. It enumerates fine, but there's some problem with the clock selector and sink parameters. I've attached lsusb -vvv and dmesg output. Apparently bNrInPins is invalid for the clock selector for some reason. After adding an extra log of that variable, I get: [49574.136478] __uac_clock_find_source(): selector reported illegal value, id 40, ret 6, pins 1 This can actually be correct if the current clock source is not supported via USB. HiLo has a separate clock source variant for USB. I think that in this case, the clock shouldn't be rejected as invalid, instead switched to the available source if possible. During initialization, there can be any kind of garbage present in the selector. Thanks for any help, -- Radosław Szkodziński

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[alsa-devel] [PATCH] ASoC: wm8903: disable IRQs during BIAS_OFF
by Stephen Warren 11 Sep '13

11 Sep '13

From: Stephen Warren <swarren(a)nvidia.com> If the WM8903 IRQ is triggered after the device is suspended, or before it is resumed, the underlying I2C bus that the WM8903 is attached to may not be available. This may cause the IRQ handler to emit the following error: wm8903 0-001a: Failed to read IRQ status: -16 ... or perhaps even cause a failure inside the I2C driver. To prevent this, explicitly mask all interrupt sources when entering BIAS_OFF. Also, modify the IRQ handler to immediately return if the device is in BIAS_OFF, so that shared interrupt lines don't cause a similar problem. This requires that WM8903 interrupts be disabled to avoid interrupt storms. During resume (actually, the transition back to BIAS_STANDBY), restore the desired IRQ mask. Signed-off-by: Stephen Warren <swarren(a)nvidia.com> --- sound/soc/codecs/wm8903.c | 38 ++++++++++++++++++++++++++++---------- 1 file changed, 28 insertions(+), 10 deletions(-) diff --git a/sound/soc/codecs/wm8903.c b/sound/soc/codecs/wm8903.c index eebcb1d..e847bbe 100644 --- a/sound/soc/codecs/wm8903.c +++ b/sound/soc/codecs/wm8903.c @@ -1091,6 +1091,22 @@ static const struct snd_soc_dapm_route wm8903_intercon[] = { { "Right Line Output PGA", NULL, "Charge Pump" }, }; +static void wm8903_setup_mic_det_irqs(struct snd_soc_codec *codec) +{ + struct wm8903_priv *wm8903 = snd_soc_codec_get_drvdata(codec); + int irq_mask = WM8903_MICDET_EINT | WM8903_MICSHRT_EINT; + + /* Enable interrupts we've got a report configured for */ + if (wm8903->mic_det) + irq_mask &= ~WM8903_MICDET_EINT; + if (wm8903->mic_short) + irq_mask &= ~WM8903_MICSHRT_EINT; + + snd_soc_update_bits(codec, WM8903_INTERRUPT_STATUS_1_MASK, + WM8903_MICDET_EINT | WM8903_MICSHRT_EINT, + irq_mask); +} + static int wm8903_set_bias_level(struct snd_soc_codec *codec, enum snd_soc_bias_level level) { @@ -1169,6 +1185,8 @@ static int wm8903_set_bias_level(struct snd_soc_codec *codec, WM8903_CP_DYN_V, WM8903_CP_DYN_FREQ | WM8903_CP_DYN_V); + + wm8903_setup_mic_det_irqs(codec); } snd_soc_update_bits(codec, WM8903_VMID_CONTROL_0, @@ -1177,6 +1195,8 @@ static int wm8903_set_bias_level(struct snd_soc_codec *codec, break; case SND_SOC_BIAS_OFF: + snd_soc_write(codec, WM8903_INTERRUPT_STATUS_1_MASK, 0xffff); + snd_soc_update_bits(codec, WM8903_BIAS_CONTROL_0, WM8903_BIAS_ENA, 0); @@ -1600,7 +1620,6 @@ int wm8903_mic_detect(struct snd_soc_codec *codec, struct snd_soc_jack *jack, int det, int shrt) { struct wm8903_priv *wm8903 = snd_soc_codec_get_drvdata(codec); - int irq_mask = WM8903_MICDET_EINT | WM8903_MICSHRT_EINT; dev_dbg(codec->dev, "Enabling microphone detection: %x %x\n", det, shrt); @@ -1610,15 +1629,7 @@ int wm8903_mic_detect(struct snd_soc_codec *codec, struct snd_soc_jack *jack, wm8903->mic_det = det; wm8903->mic_short = shrt; - /* Enable interrupts we've got a report configured for */ - if (det) - irq_mask &= ~WM8903_MICDET_EINT; - if (shrt) - irq_mask &= ~WM8903_MICSHRT_EINT; - - snd_soc_update_bits(codec, WM8903_INTERRUPT_STATUS_1_MASK, - WM8903_MICDET_EINT | WM8903_MICSHRT_EINT, - irq_mask); + wm8903_setup_mic_det_irqs(codec); if (det || shrt) { /* Enable mic detection, this may not have been set through @@ -1642,6 +1653,13 @@ static irqreturn_t wm8903_irq(int irq, void *data) int mic_report, ret; unsigned int int_val, mask, int_pol; + /* + * During suspend, a shared IRQ may be triggered by some other device. + * If so, just ignore it; WM8903 shouldn't be generating interrupts. + */ + if (wm8903->codec->dapm.bias_level == SND_SOC_BIAS_OFF) + return IRQ_NONE; + ret = regmap_read(wm8903->regmap, WM8903_INTERRUPT_STATUS_1_MASK, &mask); if (ret != 0) { -- 1.8.1.5

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Re: [alsa-devel] alsa crash with “ ALSA sound/usb/urb.c:179 cannot submit urb (err = -27)” on imx6q sabrelite
by Daniel Mack 11 Sep '13

11 Sep '13

On 11.09.2013 14:53, Ankit Agarwal wrote: > > On Sun, Sep 8, 2013 at 8:27 PM, Daniel Mack <zonque(a)gmail.com > <mailto:zonque@gmail.com>> wrote: > > On 02.09.2013 12:51, Ankit Agarwal wrote: > > I am using alsa pcm player to listen to audio. The alsa pcm > crashes with > > the error: ALSA sound/usb/urb.c:179 cannot submit urb (err = -27). > I am > > able to hear the data initially but when other transaction begins > on HID > > report endpoint, alsa crashes with the mentioned error. > > -27 == -EFBIG, which seems to relate to a scheduling problem on the bus, > and hence probably rather to a host controller bug than an issue with > snd-usb. > > > I am working on imx6q board with linux kernel-3.0.35. I am using Apple > > device to get the digital audio. > > Please try with a more recent kernel, and see if you can find the > location where EFBIG is initially reported. > > > I have found that EFBIG is being reported by ehci-host controller. It is > coming from file ehci-sched.c under iso_stream_schedule() function. The > error is thrown when the controller tries to schedule the urb too far > into the future, ie, there is overflow of the scheduled URBs. I might be > happening because the URBs are not scheduled properly. Ok. What about trying a newer kernel? The issue might be fixed already. In any case, this does not seem related to snd-usb, so I copied the usb list. Daniel

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[alsa-devel] Scarlett Mixer 8i6
by Abhayadev S 11 Sep '13

11 Sep '13

just checked the ALSA tree and found that the support for 18i6 mixer is enabled but not able to find the same in the kernel mainline tree at kernel.org. When is this planned to be patched in the mainline kernel? Also, i have a 8i6 for which i would like to extend the support for the internal mixer. Can i use the code for 18i6 here? or does it needs any changes? in anyway i could help it to add the support for 8i6 device too? Regards, Abhayadev S

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[alsa-devel] [GIT PULL] ASoC updates for v3.12
by Mark Brown 11 Sep '13

11 Sep '13

The following changes since commit 6e4664525b1db28f8c4e1130957f70a94c19213e: Linux 3.11 (2013-09-02 13:46:10 -0700) are available in the git repository at: git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git tags/asoc-v3.12-4 for you to fetch changes up to 2ae2caff8348378c9e464353e9860ee97583c00b: Merge remote-tracking branch 'asoc/fix/rsnd' into asoc-linus (2013-09-11 11:17:17 +0100) No idea what git thinks it's doing with the diffstat here... ---------------------------------------------------------------- ASoC: Fixes for v3.12 A few small fixes, nothing with any broad impact but all useful for the affected systems. The Kirkwood compatible string change is fixing up a string just added in the merge window so that we don't get any changes in released kernels. ---------------------------------------------------------------- Bo Shen (1): ASoC: atmel: disable error interrupt Fabio Estevam (1): ASoC: fsl: imx-audmux: Do not call imx_audmux_parse_dt_defaults() on non-dt kernel Kuninori Morimoto (1): ASoC: rsnd: fixup flag name of rsnd_scu_platform_info Mark Brown (6): ASoC: fsl_spdif: Select regmap-mmio Merge remote-tracking branch 'asoc/fix/atmel' into asoc-linus Merge remote-tracking branch 'asoc/fix/fsl' into asoc-linus Merge remote-tracking branch 'asoc/fix/kirkwood' into asoc-linus Merge remote-tracking branch 'asoc/fix/mc13783' into asoc-linus Merge remote-tracking branch 'asoc/fix/rsnd' into asoc-linus Steffen Trumtrar (1): ASoC: mc13783: add spi errata fix Thomas Petazzoni (1): ASoC: kirkwood: change the compatible string of the kirkwood-i2s driver .../devicetree/bindings/misc/atmel-ssc.txt | 23 +- .../devicetree/bindings/serial/mrvl,pxa-ssp.txt | 65 + Documentation/devicetree/bindings/sound/ak4554.c | 11 + .../devicetree/bindings/sound/alc5632.txt | 19 + .../bindings/sound/atmel-sam9x5-wm8731-audio.txt | 35 + .../devicetree/bindings/sound/atmel-wm8904.txt | 55 + .../devicetree/bindings/sound/fsl,spdif.txt | 54 + .../{powerpc/fsl/ssi.txt => sound/fsl,ssi.txt} | 12 + .../devicetree/bindings/sound/imx-audio-spdif.txt | 34 + .../devicetree/bindings/sound/imx-audmux.txt | 9 + .../devicetree/bindings/sound/mrvl,pxa-ssp.txt | 28 + .../devicetree/bindings/sound/mrvl,pxa2xx-pcm.txt | 15 + .../devicetree/bindings/sound/mvebu-audio.txt | 33 + .../bindings/sound/nvidia,tegra-audio-alc5632.txt | 24 +- .../bindings/sound/nvidia,tegra-audio-rt5640.txt | 26 +- .../bindings/sound/nvidia,tegra-audio-wm8753.txt | 27 +- .../bindings/sound/nvidia,tegra-audio-wm8903.txt | 24 +- .../devicetree/bindings/sound/pcm1792a.txt | 18 + Documentation/devicetree/bindings/sound/rt5640.txt | 20 + .../devicetree/bindings/sound/samsung-i2s.txt | 22 +- .../devicetree/bindings/sound/soc-ac97link.txt | 28 + .../devicetree/bindings/sound/ti,pcm1681.txt | 15 + .../devicetree/bindings/sound/tlv320aic3x.txt | 9 +- Documentation/devicetree/bindings/sound/wm8731.txt | 9 + Documentation/devicetree/bindings/sound/wm8753.txt | 24 +- Documentation/devicetree/bindings/sound/wm8903.txt | 19 + Documentation/devicetree/bindings/sound/wm8994.txt | 4 + Documentation/sound/alsa/HD-Audio-Models.txt | 1 + Documentation/sound/alsa/HD-Audio.txt | 2 + MAINTAINERS | 13 + arch/arm/boot/dts/exynos5250.dtsi | 9 +- arch/arm/mach-dove/common.c | 4 +- arch/arm/mach-kirkwood/common.c | 24 +- arch/arm/plat-pxa/ssp.c | 171 +- arch/arm/plat-samsung/s3c-dma-ops.c | 13 +- include/dt-bindings/sound/fsl-imx-audmux.h | 56 + include/linux/atmel-ssc.h | 2 +- include/linux/mfd/arizona/gpio.h | 96 + include/linux/platform_data/asoc-s3c.h | 1 + include/linux/platform_data/omap-abe-twl6040.h | 49 - include/linux/pxa2xx_ssp.h | 11 + include/sound/core.h | 8 + include/sound/pxa2xx-lib.h | 7 - include/sound/rcar_snd.h | 84 + include/sound/soc-dapm.h | 201 +- include/sound/soc-dpcm.h | 2 +- include/sound/soc.h | 47 +- include/uapi/sound/hdspm.h | 2 +- sound/arm/pxa2xx-ac97.c | 26 +- sound/arm/pxa2xx-pcm-lib.c | 52 +- sound/arm/pxa2xx-pcm.c | 5 +- sound/arm/pxa2xx-pcm.h | 6 +- sound/core/Kconfig | 3 + sound/core/Makefile | 3 + .../soc-dmaengine-pcm.c => core/pcm_dmaengine.c} | 0 sound/core/pcm_lib.c | 4 +- sound/drivers/dummy.c | 2 +- sound/firewire/speakers.c | 4 +- sound/isa/gus/interwave.c | 3 +- sound/oss/dmabuf.c | 3 +- sound/pci/hda/Kconfig | 9 +- sound/pci/hda/hda_codec.c | 64 +- sound/pci/hda/hda_codec.h | 21 + sound/pci/hda/hda_generic.c | 79 +- sound/pci/hda/hda_generic.h | 1 + sound/pci/hda/hda_hwdep.c | 6 +- sound/pci/hda/hda_intel.c | 34 +- sound/pci/hda/hda_jack.c | 22 +- sound/pci/hda/hda_jack.h | 13 +- sound/pci/hda/hda_proc.c | 33 + sound/pci/hda/patch_analog.c | 4662 +------------------- sound/pci/hda/patch_conexant.c | 79 +- sound/pci/hda/patch_hdmi.c | 9 +- sound/pci/hda/patch_realtek.c | 190 +- sound/pci/hda/patch_sigmatel.c | 14 +- sound/pci/hda/patch_via.c | 2 +- sound/pci/rme96.c | 307 +- sound/pci/rme9652/hdspm.c | 779 +++- sound/soc/Kconfig | 5 +- sound/soc/Makefile | 4 - sound/soc/atmel/Kconfig | 21 + sound/soc/atmel/Makefile | 4 + sound/soc/atmel/atmel-pcm-dma.c | 118 +- sound/soc/atmel/atmel_ssc_dai.c | 36 +- sound/soc/atmel/atmel_wm8904.c | 254 ++ sound/soc/atmel/sam9x5_wm8731.c | 208 + sound/soc/au1x/db1200.c | 4 + sound/soc/au1x/psc-ac97.c | 3 - sound/soc/blackfin/bf5xx-ac97.h | 1 - sound/soc/cirrus/ep93xx-ac97.c | 3 - sound/soc/cirrus/ep93xx-i2s.c | 5 - sound/soc/codecs/Kconfig | 19 + sound/soc/codecs/Makefile | 8 + sound/soc/codecs/ac97.c | 15 + sound/soc/codecs/ad1980.c | 43 + sound/soc/codecs/ad73311.c | 22 +- sound/soc/codecs/adau1701.c | 25 +- sound/soc/codecs/adav80x.c | 13 +- sound/soc/codecs/ads117x.c | 29 +- sound/soc/codecs/ak4104.c | 34 +- sound/soc/codecs/ak4554.c | 106 + sound/soc/codecs/ak5386.c | 17 +- sound/soc/codecs/arizona.c | 69 +- sound/soc/codecs/arizona.h | 5 +- sound/soc/codecs/bt-sco.c | 22 +- sound/soc/codecs/cs4270.c | 20 + sound/soc/codecs/cs4271.c | 30 +- sound/soc/codecs/dmic.c | 17 +- sound/soc/codecs/hdmi.c | 30 +- sound/soc/codecs/lm4857.c | 107 +- sound/soc/codecs/max9768.c | 16 + sound/soc/codecs/max98090.c | 10 +- sound/soc/codecs/max9877.c | 294 +- sound/soc/codecs/mc13783.c | 5 +- sound/soc/codecs/pcm1681.c | 339 ++ sound/soc/codecs/pcm1792a.c | 257 ++ sound/soc/codecs/pcm1792a.h | 26 + sound/soc/codecs/pcm3008.c | 150 +- sound/soc/codecs/rt5640.c | 235 +- sound/soc/codecs/rt5640.h | 12 + sound/soc/codecs/sgtl5000.c | 10 +- sound/soc/codecs/si476x.c | 14 + sound/soc/codecs/spdif_receiver.c | 17 +- sound/soc/codecs/spdif_transmitter.c | 18 +- sound/soc/codecs/ssm2602.c | 3 +- sound/soc/codecs/sta32x.c | 10 +- sound/soc/codecs/tlv320aic26.c | 51 +- sound/soc/codecs/tlv320aic32x4.c | 22 +- sound/soc/codecs/tlv320aic3x.c | 56 +- sound/soc/codecs/twl4030.c | 2 - sound/soc/codecs/twl6040.c | 7 +- sound/soc/codecs/uda134x.c | 88 +- sound/soc/codecs/wl1273.c | 17 + sound/soc/codecs/wm0010.c | 12 +- sound/soc/codecs/wm5102.c | 53 +- sound/soc/codecs/wm5110.c | 35 +- sound/soc/codecs/wm8350.c | 6 +- sound/soc/codecs/wm8727.c | 17 +- sound/soc/codecs/wm8731.c | 60 +- sound/soc/codecs/wm8753.c | 5 +- sound/soc/codecs/wm8782.c | 17 +- sound/soc/codecs/wm8903.c | 4 +- sound/soc/codecs/wm8904.c | 3 +- sound/soc/codecs/wm8960.c | 10 +- sound/soc/codecs/wm8962.c | 9 +- sound/soc/codecs/wm8994.c | 39 +- sound/soc/codecs/wm8995.c | 5 +- sound/soc/codecs/wm8997.c | 1175 +++++ sound/soc/codecs/wm8997.h | 23 + sound/soc/codecs/wm_adsp.c | 124 +- sound/soc/codecs/wm_adsp.h | 3 +- sound/soc/codecs/wm_hubs.c | 8 +- sound/soc/dwc/designware_i2s.c | 5 +- sound/soc/fsl/Kconfig | 24 +- sound/soc/fsl/Makefile | 4 + sound/soc/fsl/fsl_spdif.c | 1225 +++++ sound/soc/fsl/fsl_spdif.h | 191 + sound/soc/fsl/fsl_ssi.c | 501 ++- sound/soc/fsl/imx-audmux.c | 79 +- sound/soc/fsl/imx-audmux.h | 52 +- sound/soc/fsl/imx-mc13783.c | 1 + sound/soc/fsl/imx-pcm-dma.c | 4 +- sound/soc/fsl/imx-pcm-fiq.c | 20 +- sound/soc/fsl/imx-pcm.h | 26 +- sound/soc/fsl/imx-sgtl5000.c | 4 +- sound/soc/fsl/imx-spdif.c | 148 + sound/soc/fsl/imx-ssi.c | 11 +- sound/soc/fsl/imx-ssi.h | 1 + sound/soc/fsl/imx-wm8962.c | 3 +- sound/soc/generic/simple-card.c | 2 + sound/soc/kirkwood/Kconfig | 13 +- sound/soc/kirkwood/Makefile | 4 +- sound/soc/kirkwood/kirkwood-dma.c | 108 +- sound/soc/kirkwood/kirkwood-i2s.c | 94 +- sound/soc/kirkwood/kirkwood-openrd.c | 6 +- sound/soc/kirkwood/kirkwood-t5325.c | 6 +- sound/soc/kirkwood/kirkwood.h | 11 +- sound/soc/mxs/Kconfig | 3 +- sound/soc/mxs/mxs-saif.c | 1 - sound/soc/mxs/mxs-sgtl5000.c | 32 +- sound/soc/nuc900/nuc900-ac97.c | 3 - sound/soc/omap/Kconfig | 8 +- sound/soc/omap/mcbsp.c | 2 +- sound/soc/omap/omap-abe-twl6040.c | 133 +- sound/soc/omap/omap-dmic.c | 9 +- sound/soc/omap/omap-mcbsp.c | 5 + sound/soc/omap/omap-mcpdm.c | 3 - sound/soc/pxa/Kconfig | 2 +- sound/soc/pxa/brownstone.c | 1 + sound/soc/pxa/mioa701_wm9713.c | 7 +- sound/soc/pxa/mmp-pcm.c | 7 +- sound/soc/pxa/mmp-sspa.c | 15 +- sound/soc/pxa/pxa-ssp.c | 76 +- sound/soc/pxa/pxa2xx-ac97.c | 67 +- sound/soc/pxa/pxa2xx-i2s.c | 28 +- sound/soc/pxa/pxa2xx-pcm.c | 21 +- sound/soc/pxa/ttc-dkb.c | 1 + sound/soc/s6000/s6105-ipcam.c | 2 +- sound/soc/samsung/ac97.c | 11 +- sound/soc/samsung/dma.c | 19 +- sound/soc/samsung/dma.h | 4 +- sound/soc/samsung/i2s-regs.h | 51 +- sound/soc/samsung/i2s.c | 193 +- sound/soc/samsung/pcm.c | 4 +- sound/soc/samsung/s3c2412-i2s.c | 4 +- sound/soc/samsung/s3c24xx-i2s.c | 4 +- sound/soc/samsung/smdk_wm8994.c | 58 +- sound/soc/samsung/spdif.c | 12 +- sound/soc/sh/Kconfig | 7 + sound/soc/sh/Makefile | 3 + sound/soc/sh/fsi.c | 51 +- sound/soc/sh/rcar/Makefile | 2 + sound/soc/sh/rcar/adg.c | 234 + sound/soc/sh/rcar/core.c | 861 ++++ sound/soc/sh/rcar/gen.c | 280 ++ sound/soc/sh/rcar/rsnd.h | 302 ++ sound/soc/sh/rcar/scu.c | 236 + sound/soc/sh/rcar/ssi.c | 728 +++ sound/soc/soc-compress.c | 13 +- sound/soc/soc-core.c | 253 +- sound/soc/soc-dapm.c | 819 ++-- sound/soc/soc-jack.c | 4 +- sound/soc/soc-pcm.c | 25 +- sound/soc/spear/Kconfig | 2 +- sound/soc/tegra/Kconfig | 14 +- sound/soc/tegra/tegra20_ac97.c | 8 - sound/soc/tegra/tegra_alc5632.c | 2 - sound/soc/tegra/tegra_rt5640.c | 1 + sound/soc/tegra/tegra_wm8753.c | 2 - sound/soc/tegra/trimslice.c | 2 - sound/soc/txx9/txx9aclc-ac97.c | 3 - sound/soc/ux500/mop500.c | 1 + sound/usb/6fire/firmware.c | 4 +- sound/usb/endpoint.c | 3 + sound/usb/pcm.c | 243 +- sound/usb/usx2y/usbusx2y.c | 8 +- 236 files changed, 12494 insertions(+), 7410 deletions(-) create mode 100644 Documentation/devicetree/bindings/serial/mrvl,pxa-ssp.txt create mode 100644 Documentation/devicetree/bindings/sound/ak4554.c create mode 100644 Documentation/devicetree/bindings/sound/atmel-sam9x5-wm8731-audio.txt create mode 100644 Documentation/devicetree/bindings/sound/atmel-wm8904.txt create mode 100644 Documentation/devicetree/bindings/sound/fsl,spdif.txt rename Documentation/devicetree/bindings/{powerpc/fsl/ssi.txt => sound/fsl,ssi.txt} (87%) create mode 100644 Documentation/devicetree/bindings/sound/imx-audio-spdif.txt create mode 100644 Documentation/devicetree/bindings/sound/mrvl,pxa-ssp.txt create mode 100644 Documentation/devicetree/bindings/sound/mrvl,pxa2xx-pcm.txt create mode 100644 Documentation/devicetree/bindings/sound/mvebu-audio.txt create mode 100644 Documentation/devicetree/bindings/sound/pcm1792a.txt create mode 100644 Documentation/devicetree/bindings/sound/soc-ac97link.txt create mode 100644 Documentation/devicetree/bindings/sound/ti,pcm1681.txt create mode 100644 include/dt-bindings/sound/fsl-imx-audmux.h create mode 100644 include/linux/mfd/arizona/gpio.h delete mode 100644 include/linux/platform_data/omap-abe-twl6040.h create mode 100644 include/sound/rcar_snd.h rename sound/{soc/soc-dmaengine-pcm.c => core/pcm_dmaengine.c} (100%) create mode 100644 sound/soc/atmel/atmel_wm8904.c create mode 100644 sound/soc/atmel/sam9x5_wm8731.c create mode 100644 sound/soc/codecs/ak4554.c create mode 100644 sound/soc/codecs/pcm1681.c create mode 100644 sound/soc/codecs/pcm1792a.c create mode 100644 sound/soc/codecs/pcm1792a.h create mode 100644 sound/soc/codecs/wm8997.c create mode 100644 sound/soc/codecs/wm8997.h create mode 100644 sound/soc/fsl/fsl_spdif.c create mode 100644 sound/soc/fsl/fsl_spdif.h create mode 100644 sound/soc/fsl/imx-spdif.c create mode 100644 sound/soc/sh/rcar/Makefile create mode 100644 sound/soc/sh/rcar/adg.c create mode 100644 sound/soc/sh/rcar/core.c create mode 100644 sound/soc/sh/rcar/gen.c create mode 100644 sound/soc/sh/rcar/rsnd.h create mode 100644 sound/soc/sh/rcar/scu.c create mode 100644 sound/soc/sh/rcar/ssi.c

2 1

[alsa-devel] [PATCH v2] codecs: mc13783: add spi errata fix
by Steffen Trumtrar 10 Sep '13

10 Sep '13

The MC13783 Chip Errata, Rev. 4 says, that depending on SPI clock and main audio clock speed, the Audio Codec or Stereo DAC do sometimes not start when programmed to do so. This is due to an internal clock timing issue related to the loading of the SPI bits into the audio block. On an i.MX27 based system, this issue lead to switched audio channels under certain circumstances: RTC + Touch + Audio are used and loaded at startup. The mentioned workaround of writing registers 40 and 41 two times is implemented here. Signed-off-by: Steffen Trumtrar <s.trumtrar(a)pengutronix.de> Cc: stable(a)vger.kernel.org --- Changes in v2: - use correct register defines - fix spelling error in commit description sound/soc/codecs/mc13783.c | 4 ++++ 1 file changed, 4 insertions(+) diff --git a/sound/soc/codecs/mc13783.c b/sound/soc/codecs/mc13783.c index 4d3c8fd..ea141e1 100644 --- a/sound/soc/codecs/mc13783.c +++ b/sound/soc/codecs/mc13783.c @@ -125,6 +125,10 @@ static int mc13783_write(struct snd_soc_codec *codec, ret = mc13xxx_reg_write(priv->mc13xxx, reg, value); + /* include errata fix for spi audio problems */ + if (reg == MC13783_AUDIO_CODEC || reg == MC13783_AUDIO_DAC) + ret = mc13xxx_reg_write(priv->mc13xxx, reg, value); + mc13xxx_unlock(priv->mc13xxx); return ret; -- 1.8.4.rc3

2 1

[alsa-devel] Support for Scarlett Mixer
by Abhayadev S 10 Sep '13

10 Sep '13

just check the ALSA tree and foudn that the support for 18i6 mixer is enabled but not able to find the same in the kernel mainline tree at kernel.org. When is this planned to be patched in the mainline kernel? Also, i have a 8i6 for which i would like to extend the support for the internal mixer. Can i use the same code for 8i6 also? or does it needs any changes? in anyway i could help it to add the support for 8i6 device too? Regards, Abhayadev S

1 0

[alsa-devel] Sound distorted on Intel HDA (Realtek ALC889) (Kernel 3.x)
by Vincent Thiele 09 Sep '13

09 Sep '13

Using Audio device: Intel Corporation 6 Series/C200 Series Chipset Family High Definition Audio Controller (rev 05) Mainboard is: http://www.asus.com/Motherboards/MAXIMUS_IV_GENEZ/ Only fix at the moment is changing "load-module module-udev-detect use_ucm=0" in "/etc/pulse/default.pa" to "load-module module-udev-detect tsched=0". Maybe you should add a specific rule for this device which adds this fix automatically. Best regards alsa-info.sh output: upload=true&script=true&cardinfo= !!################################ !!ALSA Information Script v 0.4.62 !!################################ !!Script ran on: Mon Jul 22 18:15:08 UTC 2013 !!Linux Distribution !!------------------ Ubuntu 13.04 \n \l DISTRIB_ID=Ubuntu DISTRIB_DESCRIPTION="Ubuntu 13.04" NAME="Ubuntu" ID=ubuntu ID_LIKE=debian PRETTY_NAME="Ubuntu 13.04" HOME_URL="http://www.ubuntu.com/" SUPPORT_URL="http://help.ubuntu.com/" BUG_REPORT_URL="http://bugs.launchpad.net/ubuntu/" !!DMI Information !!--------------- Manufacturer: System manufacturer Product Name: System Product Name Product Version: System Version Firmware Version: 3603 !!Kernel Information !!------------------ Kernel release: 3.11.0-031100rc2-generic Operating System: GNU/Linux Architecture: x86_64 Processor: x86_64 SMP Enabled: Yes !!ALSA Version !!------------ Driver version: k3.11.0-031100rc2-generic Library version: 1.0.25 Utilities version: 1.0.25 !!Loaded ALSA modules !!------------------- snd_hda_intel snd_hda_intel !!Sound Servers on this system !!---------------------------- Pulseaudio: Installed - Yes (/usr/bin/pulseaudio) Running - Yes !!Soundcards recognised by ALSA !!----------------------------- 0 [PCH ]: HDA-Intel - HDA Intel PCH HDA Intel PCH at 0xf7420000 irq 54 1 [NVidia ]: HDA-Intel - HDA NVidia HDA NVidia at 0xf7080000 irq 17 !!PCI Soundcards installed in the system !!-------------------------------------- 00:1b.0 Audio device: Intel Corporation 6 Series/C200 Series Chipset Family High Definition Audio Controller (rev 05) 01:00.1 Audio device: NVIDIA Corporation Device 0e0b (rev a1) !!Advanced information - PCI Vendor/Device/Subsystem ID's !!------------------------------------------------------- 00:1b.0 0403: 8086:1c20 (rev 05) Subsystem: 1043:84dc -- 01:00.1 0403: 10de:0e0b (rev a1) Subsystem: 1043:8422 !!Modprobe options (Sound related) !!-------------------------------- snd_atiixp_modem: index=-2 snd_intel8x0m: index=-2 snd_via82xx_modem: index=-2 snd_usb_audio: index=-2 snd_usb_caiaq: index=-2 snd_usb_ua101: index=-2 snd_usb_us122l: index=-2 snd_usb_usx2y: index=-2 snd_cmipci: mpu_port=0x330 fm_port=0x388 snd_pcsp: index=-2 snd_usb_audio: index=-2 !!Loaded sound module options !!--------------------------- !!Module: snd_hda_intel align_buffer_size : -1 bdl_pos_adj : 1,32,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1 beep_mode : N,N,N,N,N,N,N,N,N,N,N,N,N,N,N,N,N,N,N,N,N,N,N,N,N,N,N,N,N,N,N,N enable : Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y enable_msi : -1 id : (null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null) index : -1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1 jackpoll_ms : 0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0 model : (null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null) patch : (null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null) position_fix : -1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1 power_save : 0 power_save_controller : Y probe_mask : -1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1 probe_only : 0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0 single_cmd : N snoop : Y !!Module: snd_hda_intel align_buffer_size : -1 bdl_pos_adj : 1,32,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1 beep_mode : N,N,N,N,N,N,N,N,N,N,N,N,N,N,N,N,N,N,N,N,N,N,N,N,N,N,N,N,N,N,N,N enable : Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y enable_msi : -1 id : (null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null) index : -1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1 jackpoll_ms : 0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0 model : (null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null) patch : (null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null) position_fix : -1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1 power_save : 0 power_save_controller : Y probe_mask : -1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1 probe_only : 0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0 single_cmd : N snoop : Y !!HDA-Intel Codec information !!--------------------------- --startcollapse-- Codec: Realtek ALC889 Address: 0 AFG Function Id: 0x1 (unsol 1) Vendor Id: 0x10ec0889 Subsystem Id: 0x1043846b Revision Id: 0x100004 No Modem Function Group found Default PCM: rates [0x5f0]: 32000 44100 48000 88200 96000 192000 bits [0xe]: 16 20 24 formats [0x1]: PCM Default Amp-In caps: N/A Default Amp-Out caps: N/A State of AFG node 0x01: Power states: D0 D1 D2 D3 CLKSTOP EPSS Power: setting=D0, actual=D0 GPIO: io=2, o=0, i=0, unsolicited=1, wake=1 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 0x11: Stereo Device: name="ALC889 Analog", type="Audio", device=0 Converter: stream=8, channel=0 PCM: rates [0x5e0]: 44100 48000 88200 96000 192000 bits [0xe]: 16 20 24 formats [0x1]: PCM Node 0x03 [Audio Output] wcaps 0x11: Stereo Converter: stream=8, channel=0 PCM: rates [0x5e0]: 44100 48000 88200 96000 192000 bits [0xe]: 16 20 24 formats [0x1]: PCM Node 0x04 [Audio Output] wcaps 0x11: Stereo Converter: stream=8, channel=0 PCM: rates [0x5e0]: 44100 48000 88200 96000 192000 bits [0xe]: 16 20 24 formats [0x1]: PCM Node 0x05 [Audio Output] wcaps 0x11: Stereo Converter: stream=8, channel=0 PCM: rates [0x5e0]: 44100 48000 88200 96000 192000 bits [0xe]: 16 20 24 formats [0x1]: PCM Node 0x06 [Audio Output] wcaps 0x211: Stereo Digital Converter: stream=8, channel=0 Digital: Digital category: 0x0 IEC Coding Type: 0x0 PCM: rates [0x5f0]: 32000 44100 48000 88200 96000 192000 bits [0xe]: 16 20 24 formats [0x1]: PCM Node 0x07 [Audio Input] wcaps 0x10011b: Stereo Amp-In Control: name="Capture Volume", index=0, device=0 ControlAmp: chs=3, dir=In, idx=0, ofs=0 Control: name="Capture Switch", index=0, device=0 ControlAmp: chs=3, dir=In, idx=0, ofs=0 Device: name="ALC889 Analog", type="Audio", device=0 Amp-In caps: ofs=0x10, nsteps=0x2e, stepsize=0x03, mute=1 Amp-In vals: [0x00 0x00] Converter: stream=4, channel=0 SDI-Select: 0 PCM: rates [0x5e0]: 44100 48000 88200 96000 192000 bits [0xe]: 16 20 24 formats [0x1]: PCM Connection: 1 0x24 Node 0x08 [Audio Input] wcaps 0x10011b: Stereo Amp-In Control: name="Capture Volume", index=1, device=0 ControlAmp: chs=3, dir=In, idx=0, ofs=0 Control: name="Capture Switch", index=1, device=0 ControlAmp: chs=3, dir=In, idx=0, ofs=0 Device: name="ALC889 Alt Analog", type="Audio", device=2 Amp-In caps: ofs=0x10, nsteps=0x2e, stepsize=0x03, mute=1 Amp-In vals: [0x80 0x80] Converter: stream=0, channel=0 SDI-Select: 0 PCM: rates [0x5e0]: 44100 48000 88200 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 Volume", index=2, device=0 ControlAmp: chs=3, dir=In, idx=0, ofs=0 Control: name="Capture Switch", index=2, device=0 ControlAmp: chs=3, dir=In, idx=0, ofs=0 Amp-In caps: ofs=0x10, nsteps=0x2e, stepsize=0x03, mute=1 Amp-In vals: [0x80 0x80] Converter: stream=0, channel=0 SDI-Select: 0 PCM: rates [0x5e0]: 44100 48000 88200 96000 192000 bits [0xe]: 16 20 24 formats [0x1]: PCM Connection: 1 0x22 Node 0x0a [Audio Input] wcaps 0x100391: Stereo Digital Converter: stream=0, channel=0 SDI-Select: 0 Digital: Digital category: 0x0 IEC Coding Type: 0x0 PCM: rates [0x570]: 32000 44100 48000 96000 192000 bits [0xe]: 16 20 24 formats [0x1]: PCM Unsolicited: tag=00, enabled=0 Connection: 1 0x1f Node 0x0b [Audio Mixer] wcaps 0x20010b: Stereo Amp-In Control: name="Front Mic Playback Volume", index=0, device=0 ControlAmp: chs=3, dir=In, idx=1, ofs=0 Control: name="Front Mic Playback Switch", index=0, device=0 ControlAmp: chs=3, dir=In, idx=1, ofs=0 Control: name="Rear Mic Playback Volume", index=0, device=0 ControlAmp: chs=3, dir=In, idx=0, ofs=0 Control: name="Rear Mic Playback Switch", index=0, device=0 ControlAmp: chs=3, dir=In, idx=0, ofs=0 Control: name="Line Playback Volume", index=0, device=0 ControlAmp: chs=3, dir=In, idx=2, ofs=0 Control: name="Line Playback Switch", index=0, device=0 ControlAmp: chs=3, dir=In, idx=2, ofs=0 Control: name="Beep Playback Volume", index=0, device=0 ControlAmp: chs=3, dir=In, idx=5, ofs=0 Control: name="Beep Playback Switch", index=0, device=0 ControlAmp: chs=3, dir=In, idx=5, 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] [0x80 0x80] [0x80 0x80] [0x80 0x80] [0x80 0x80] [0x80 0x80] Connection: 10 0x18 0x19 0x1a 0x1b 0x1c 0x1d 0x14 0x15 0x16 0x17 Node 0x0c [Audio Mixer] wcaps 0x20010f: Stereo Amp-In Amp-Out Control: name="Front Playback Volume", index=0, device=0 ControlAmp: chs=3, dir=Out, idx=0, ofs=0 Amp-In caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 Amp-In vals: [0x00 0x00] [0x00 0x00] Amp-Out caps: ofs=0x3e, nsteps=0x40, stepsize=0x03, mute=0 Amp-Out vals: [0x39 0x39] Connection: 2 0x02 0x0b Node 0x0d [Audio Mixer] wcaps 0x20010f: Stereo Amp-In Amp-Out Control: name="Surround Playback Volume", index=0, device=0 ControlAmp: chs=3, dir=Out, idx=0, ofs=0 Amp-In caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 Amp-In vals: [0x00 0x00] [0x00 0x00] Amp-Out caps: ofs=0x3e, nsteps=0x40, stepsize=0x03, mute=0 Amp-Out vals: [0x39 0x39] Connection: 2 0x03 0x0b Node 0x0e [Audio Mixer] wcaps 0x20010f: Stereo Amp-In Amp-Out Control: name="Center Playback Volume", index=0, device=0 ControlAmp: chs=1, dir=Out, idx=0, ofs=0 Control: name="LFE Playback Volume", index=0, device=0 ControlAmp: chs=2, dir=Out, idx=0, ofs=0 Amp-In caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 Amp-In vals: [0x00 0x00] [0x00 0x00] Amp-Out caps: ofs=0x3e, nsteps=0x40, stepsize=0x03, mute=0 Amp-Out vals: [0x39 0x39] Connection: 2 0x04 0x0b Node 0x0f [Audio Mixer] wcaps 0x20010f: Stereo Amp-In Amp-Out Control: name="Side Playback Volume", index=0, device=0 ControlAmp: chs=3, dir=Out, idx=0, ofs=0 Amp-In caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 Amp-In vals: [0x00 0x00] [0x00 0x00] Amp-Out caps: ofs=0x3e, nsteps=0x40, stepsize=0x03, mute=0 Amp-Out vals: [0x39 0x39] Connection: 2 0x05 0x0b Node 0x10 [Audio Output] wcaps 0x211: Stereo Digital Control: name="IEC958 Playback Con Mask", index=0, device=0 Control: name="IEC958 Playback Pro Mask", index=0, device=0 Control: name="IEC958 Playback Default", index=0, device=0 Control: name="IEC958 Playback Switch", index=0, device=0 Control: name="IEC958 Default PCM Playback Switch", index=0, device=0 Device: name="ALC889 Digital", type="SPDIF", device=1 Converter: stream=8, channel=0 Digital: Digital category: 0x0 IEC Coding Type: 0x0 PCM: rates [0x5f0]: 32000 44100 48000 88200 96000 192000 bits [0xe]: 16 20 24 formats [0x1]: PCM Node 0x11 [Pin Complex] wcaps 0x400300: Mono Digital Control: name="SPDIF Phantom Jack", index=0, device=0 Pincap 0x00000010: OUT Pin Default 0x99430140: [Fixed] SPDIF Out at Int ATAPI Conn = ATAPI, Color = Unknown DefAssociation = 0x4, Sequence = 0x0 Misc = NO_PRESENCE Pin-ctls: 0x40: OUT Connection: 1 0x10 Node 0x12 [Pin Complex] wcaps 0x400001: Stereo Pincap 0x00000020: IN Pin Default 0x411111f0: [N/A] Speaker at Ext Rear Conn = 1/8, Color = Black DefAssociation = 0xf, Sequence = 0x0 Misc = NO_PRESENCE Pin-ctls: 0x00: Node 0x13 [Vendor Defined Widget] wcaps 0xf00000: Mono Node 0x14 [Pin Complex] wcaps 0x40018d: Stereo Amp-Out Control: name="Front Playback Switch", index=0, device=0 ControlAmp: chs=3, dir=Out, idx=0, ofs=0 Control: name="Line Out Front Jack", index=0, device=0 Amp-Out caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 Amp-Out vals: [0x00 0x00] Pincap 0x0001003c: IN OUT HP EAPD Detect EAPD 0x2: EAPD Pin Default 0x01014010: [Jack] Line Out at Ext Rear Conn = 1/8, Color = Green DefAssociation = 0x1, Sequence = 0x0 Pin-ctls: 0x40: OUT Unsolicited: tag=05, enabled=1 Connection: 5 0x0c* 0x0d 0x0e 0x0f 0x26 Node 0x15 [Pin Complex] wcaps 0x40018d: Stereo Amp-Out Control: name="Surround Playback Switch", index=0, device=0 ControlAmp: chs=3, dir=Out, idx=0, ofs=0 Control: name="Line Out Surround Jack", index=0, device=0 Amp-Out caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 Amp-Out vals: [0x00 0x00] Pincap 0x0001003c: IN OUT HP EAPD Detect EAPD 0x2: EAPD Pin Default 0x01011012: [Jack] Line Out at Ext Rear Conn = 1/8, Color = Black DefAssociation = 0x1, Sequence = 0x2 Pin-ctls: 0x40: OUT Unsolicited: tag=06, enabled=1 Connection: 5 0x0c 0x0d* 0x0e 0x0f 0x26 Node 0x16 [Pin Complex] wcaps 0x40018d: Stereo Amp-Out Control: name="Center Playback Switch", index=0, device=0 ControlAmp: chs=1, dir=Out, idx=0, ofs=0 Control: name="LFE Playback Switch", index=0, device=0 ControlAmp: chs=2, dir=Out, idx=0, ofs=0 Control: name="Line Out CLFE Jack", index=0, device=0 Amp-Out caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 Amp-Out vals: [0x00 0x00] Pincap 0x00000034: IN OUT Detect Pin Default 0x01016011: [Jack] Line Out at Ext Rear Conn = 1/8, Color = Orange DefAssociation = 0x1, Sequence = 0x1 Pin-ctls: 0x40: OUT Unsolicited: tag=07, enabled=1 Connection: 5 0x0c 0x0d 0x0e* 0x0f 0x26 Node 0x17 [Pin Complex] wcaps 0x40018d: Stereo Amp-Out Control: name="Side Playback Switch", index=0, device=0 ControlAmp: chs=3, dir=Out, idx=0, ofs=0 Control: name="Line Out Side Jack", index=0, device=0 Amp-Out caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 Amp-Out vals: [0x00 0x00] Pincap 0x00000034: IN OUT Detect Pin Default 0x01012014: [Jack] Line Out at Ext Rear Conn = 1/8, Color = Grey DefAssociation = 0x1, Sequence = 0x4 Pin-ctls: 0x40: OUT Unsolicited: tag=08, enabled=1 Connection: 5 0x0c 0x0d 0x0e 0x0f* 0x26 Node 0x18 [Pin Complex] wcaps 0x40018f: Stereo Amp-In Amp-Out Control: name="Rear Mic Boost Volume", index=0, device=0 ControlAmp: chs=3, dir=In, idx=0, ofs=0 Control: name="Rear Mic Jack", index=0, device=0 Amp-In caps: ofs=0x00, nsteps=0x03, stepsize=0x27, mute=0 Amp-In vals: [0x00 0x00] Amp-Out caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 Amp-Out vals: [0x80 0x80] Pincap 0x0000373c: IN OUT HP Detect Vref caps: HIZ 50 GRD 80 100 Pin Default 0x01a19850: [Jack] Mic at Ext Rear Conn = 1/8, Color = Pink DefAssociation = 0x5, Sequence = 0x0 Pin-ctls: 0x24: IN VREF_80 Unsolicited: tag=03, enabled=1 Connection: 5 0x0c 0x0d 0x0e 0x0f 0x26* Node 0x19 [Pin Complex] wcaps 0x40018f: Stereo Amp-In Amp-Out Control: name="Front Mic Boost Volume", index=0, device=0 ControlAmp: chs=3, dir=In, idx=0, ofs=0 Control: name="Front Mic Jack", index=0, device=0 Amp-In caps: ofs=0x00, nsteps=0x03, stepsize=0x27, mute=0 Amp-In vals: [0x00 0x00] Amp-Out caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 Amp-Out vals: [0x80 0x80] Pincap 0x0000373c: IN OUT HP Detect Vref caps: HIZ 50 GRD 80 100 Pin Default 0x02a19c60: [Jack] Mic at Ext Front Conn = 1/8, Color = Pink DefAssociation = 0x6, Sequence = 0x0 Pin-ctls: 0x24: IN VREF_80 Unsolicited: tag=02, enabled=1 Connection: 5 0x0c 0x0d 0x0e 0x0f 0x26* Node 0x1a [Pin Complex] wcaps 0x40018f: Stereo Amp-In Amp-Out Control: name="Line Boost Volume", index=0, device=0 ControlAmp: chs=3, dir=In, idx=0, ofs=0 Control: name="Line Jack", index=0, device=0 Amp-In caps: ofs=0x00, nsteps=0x03, stepsize=0x27, mute=0 Amp-In vals: [0x00 0x00] Amp-Out caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 Amp-Out vals: [0x80 0x80] Pincap 0x0000373c: IN OUT HP Detect Vref caps: HIZ 50 GRD 80 100 Pin Default 0x0181305f: [Jack] Line In at Ext Rear Conn = 1/8, Color = Blue DefAssociation = 0x5, Sequence = 0xf Pin-ctls: 0x20: IN VREF_HIZ Unsolicited: tag=04, enabled=1 Connection: 5 0x0c 0x0d 0x0e 0x0f 0x26* Node 0x1b [Pin Complex] wcaps 0x40018f: Stereo Amp-In Amp-Out Control: name="Headphone Playback Switch", index=0, device=0 ControlAmp: chs=3, dir=Out, idx=0, ofs=0 Control: name="Front Headphone Jack", index=0, device=0 Amp-In caps: ofs=0x00, nsteps=0x03, stepsize=0x27, mute=0 Amp-In vals: [0x00 0x00] Amp-Out caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 Amp-Out vals: [0x00 0x00] Pincap 0x0000373c: IN OUT HP Detect Vref caps: HIZ 50 GRD 80 100 Pin Default 0x02214c20: [Jack] HP Out at Ext Front Conn = 1/8, Color = Green DefAssociation = 0x2, Sequence = 0x0 Pin-ctls: 0xc0: OUT HP VREF_HIZ Unsolicited: tag=01, enabled=1 Connection: 5 0x0c 0x0d 0x0e 0x0f 0x26* Node 0x1c [Pin Complex] wcaps 0x400081: Stereo Pincap 0x00000024: IN 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 Node 0x1d [Pin Complex] wcaps 0x400000: Mono Pincap 0x00000020: IN Pin Default 0x4007e629: [N/A] Line Out at Ext N/A Conn = Analog, Color = White DefAssociation = 0x2, Sequence = 0x9 Pin-ctls: 0x20: IN Node 0x1e [Pin Complex] wcaps 0x400300: Mono Digital Control: name="SPDIF Phantom Jack", index=1, device=0 Pincap 0x00000010: OUT Pin Default 0x01451130: [Jack] SPDIF Out at Ext Rear Conn = Optical, Color = Black DefAssociation = 0x3, Sequence = 0x0 Misc = NO_PRESENCE Pin-ctls: 0x40: OUT Connection: 1 0x06 Node 0x1f [Pin Complex] wcaps 0x400280: Mono Digital Pincap 0x00000020: IN 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 Node 0x20 [Vendor Defined Widget] wcaps 0xf00040: Mono Processing caps: benign=0, ncoeff=28 Node 0x21 [Vendor Defined Widget] wcaps 0xf00000: Mono Node 0x22 [Audio Mixer] wcaps 0x20010b: Stereo Amp-In Amp-In caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 Amp-In vals: [0x80 0x80] [0x00 0x00] [0x80 0x80] [0x80 0x80] [0x80 0x80] [0x80 0x80] [0x80 0x80] [0x80 0x80] [0x80 0x80] [0x80 0x80] [0x80 0x80] Connection: 11 0x18 0x19 0x1a 0x1b 0x1c 0x1d 0x14 0x15 0x16 0x17 0x0b 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] [0x00 0x00] [0x80 0x80] [0x80 0x80] [0x80 0x80] [0x80 0x80] [0x80 0x80] [0x80 0x80] [0x80 0x80] [0x80 0x80] [0x80 0x80] Connection: 11 0x18 0x19 0x1a 0x1b 0x1c 0x1d 0x14 0x15 0x16 0x17 0x0b Node 0x24 [Audio Selector] wcaps 0x300101: Stereo Connection: 12 0x18 0x19* 0x1a 0x1b 0x1c 0x1d 0x14 0x15 0x16 0x17 0x0b 0x12 Node 0x25 [Audio Output] wcaps 0x11: Stereo Converter: stream=8, channel=0 PCM: rates [0x5e0]: 44100 48000 88200 96000 192000 bits [0xe]: 16 20 24 formats [0x1]: PCM Node 0x26 [Audio Mixer] wcaps 0x20010f: Stereo Amp-In Amp-Out Control: name="Headphone Playback Volume", index=0, device=0 ControlAmp: chs=3, dir=Out, idx=0, ofs=0 Amp-In caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 Amp-In vals: [0x00 0x00] [0x00 0x00] Amp-Out caps: ofs=0x3e, nsteps=0x40, stepsize=0x03, mute=0 Amp-Out vals: [0x39 0x39] Connection: 2 0x25 0x0b Codec: Nvidia GPU 41 HDMI/DP Address: 0 AFG Function Id: 0x1 (unsol 0) Vendor Id: 0x10de0041 Subsystem Id: 0x10438422 Revision Id: 0x100100 No Modem Function Group found Default PCM: rates [0x0]: bits [0x0]: formats [0x0]: Default Amp-In caps: N/A Default Amp-Out caps: N/A State of AFG node 0x01: Power states: D0 D1 D2 D3 CLKSTOP EPSS Power: setting=D0, actual=D0 GPIO: io=0, o=0, i=0, unsolicited=0, wake=0 Node 0x04 [Pin Complex] wcaps 0x407381: 8-Channels Digital CP Control: name="HDMI/DP,pcm=3 Jack", index=0, device=0 Control: name="IEC958 Playback Con Mask", index=0, device=0 Control: name="IEC958 Playback Pro Mask", index=0, device=0 Control: name="IEC958 Playback Default", index=0, device=0 Control: name="IEC958 Playback Switch", index=0, device=0 Control: name="ELD", index=0, device=3 Pincap 0x09000094: OUT Detect HBR HDMI DP Pin Default 0x185600f0: [Jack] Digital Out at Int HDMI Conn = Digital, Color = Unknown DefAssociation = 0xf, Sequence = 0x0 Pin-ctls: 0x40: OUT Unsolicited: tag=01, enabled=1 Connection: 4 0x08* 0x09 0x0a 0x0b Node 0x05 [Pin Complex] wcaps 0x407381: 8-Channels Digital CP Control: name="HDMI/DP,pcm=7 Jack", index=0, device=0 Control: name="IEC958 Playback Con Mask", index=1, device=0 Control: name="IEC958 Playback Pro Mask", index=1, device=0 Control: name="IEC958 Playback Default", index=1, device=0 Control: name="IEC958 Playback Switch", index=1, device=0 Control: name="ELD", index=0, device=7 Pincap 0x09000094: OUT Detect HBR HDMI DP Pin Default 0x185600f0: [Jack] Digital Out at Int HDMI Conn = Digital, Color = Unknown DefAssociation = 0xf, Sequence = 0x0 Pin-ctls: 0x40: OUT Unsolicited: tag=02, enabled=1 Connection: 4 0x08* 0x09 0x0a 0x0b Node 0x06 [Pin Complex] wcaps 0x407381: 8-Channels Digital CP Control: name="HDMI/DP,pcm=8 Jack", index=0, device=0 Control: name="IEC958 Playback Con Mask", index=2, device=0 Control: name="IEC958 Playback Pro Mask", index=2, device=0 Control: name="IEC958 Playback Default", index=2, device=0 Control: name="IEC958 Playback Switch", index=2, device=0 Control: name="ELD", index=0, device=8 Pincap 0x09000094: OUT Detect HBR HDMI DP Pin Default 0x185600f0: [Jack] Digital Out at Int HDMI Conn = Digital, Color = Unknown DefAssociation = 0xf, Sequence = 0x0 Pin-ctls: 0x40: OUT Unsolicited: tag=03, enabled=1 Connection: 4 0x08* 0x09 0x0a 0x0b Node 0x07 [Pin Complex] wcaps 0x407381: 8-Channels Digital CP Control: name="HDMI/DP,pcm=9 Jack", index=0, device=0 Control: name="IEC958 Playback Con Mask", index=3, device=0 Control: name="IEC958 Playback Pro Mask", index=3, device=0 Control: name="IEC958 Playback Default", index=3, device=0 Control: name="IEC958 Playback Switch", index=3, device=0 Control: name="ELD", index=0, device=9 Pincap 0x09000094: OUT Detect HBR HDMI DP Pin Default 0x185600f0: [Jack] Digital Out at Int HDMI Conn = Digital, Color = Unknown DefAssociation = 0xf, Sequence = 0x0 Pin-ctls: 0x40: OUT Unsolicited: tag=04, enabled=1 Connection: 4 0x08* 0x09 0x0a 0x0b Node 0x08 [Audio Output] wcaps 0x62b1: 8-Channels Digital Stripe Converter: stream=8, channel=0 Digital: Enabled Digital category: 0x0 IEC Coding Type: 0x0 PCM: rates [0x7f0]: 32000 44100 48000 88200 96000 176400 192000 bits [0xe]: 16 20 24 formats [0x5]: PCM AC3 Unsolicited: tag=00, enabled=0 Node 0x09 [Audio Output] wcaps 0x62b1: 8-Channels Digital Stripe Converter: stream=0, channel=0 Digital: Digital category: 0x0 IEC Coding Type: 0x0 PCM: rates [0x7f0]: 32000 44100 48000 88200 96000 176400 192000 bits [0xe]: 16 20 24 formats [0x5]: PCM AC3 Unsolicited: tag=00, enabled=0 Node 0x0a [Audio Output] wcaps 0x62b1: 8-Channels Digital Stripe Converter: stream=0, channel=0 Digital: Digital category: 0x0 IEC Coding Type: 0x0 PCM: rates [0x7f0]: 32000 44100 48000 88200 96000 176400 192000 bits [0xe]: 16 20 24 formats [0x5]: PCM AC3 Unsolicited: tag=00, enabled=0 Node 0x0b [Audio Output] wcaps 0x62b1: 8-Channels Digital Stripe Converter: stream=0, channel=0 Digital: Digital category: 0x0 IEC Coding Type: 0x0 PCM: rates [0x7f0]: 32000 44100 48000 88200 96000 176400 192000 bits [0xe]: 16 20 24 formats [0x5]: PCM AC3 Unsolicited: tag=00, enabled=0 --endcollapse-- !!ALSA Device nodes !!----------------- crw-rw---T+ 1 root audio 116, 7 Jul 22 20:08 /dev/snd/controlC0 crw-rw---T+ 1 root audio 116, 13 Jul 22 20:08 /dev/snd/controlC1 crw-rw---T+ 1 root audio 116, 6 Jul 22 20:08 /dev/snd/hwC0D0 crw-rw---T+ 1 root audio 116, 12 Jul 22 20:08 /dev/snd/hwC1D0 crw-rw---T+ 1 root audio 116, 5 Jul 22 20:09 /dev/snd/pcmC0D0c crw-rw---T+ 1 root audio 116, 4 Jul 22 20:10 /dev/snd/pcmC0D0p crw-rw---T+ 1 root audio 116, 3 Jul 22 20:09 /dev/snd/pcmC0D1p crw-rw---T+ 1 root audio 116, 2 Jul 22 20:08 /dev/snd/pcmC0D2c crw-rw---T+ 1 root audio 116, 11 Jul 22 20:09 /dev/snd/pcmC1D3p crw-rw---T+ 1 root audio 116, 10 Jul 22 20:09 /dev/snd/pcmC1D7p crw-rw---T+ 1 root audio 116, 9 Jul 22 20:09 /dev/snd/pcmC1D8p crw-rw---T+ 1 root audio 116, 8 Jul 22 20:09 /dev/snd/pcmC1D9p crw-rw---T+ 1 root audio 116, 1 Jul 22 20:08 /dev/snd/seq crw-rw---T+ 1 root audio 116, 33 Jul 22 20:08 /dev/snd/timer /dev/snd/by-path: total 0 drwxr-xr-x 2 root root 80 Jul 22 20:08 . drwxr-xr-x 3 root root 340 Jul 22 20:08 .. lrwxrwxrwx 1 root root 12 Jul 22 20:08 pci-0000:00:1b.0 -> ../controlC0 lrwxrwxrwx 1 root root 12 Jul 22 20:08 pci-0000:01:00.1 -> ../controlC1 !!Aplay/Arecord output !!-------------------- APLAY **** List of PLAYBACK Hardware Devices **** card 0: PCH [HDA Intel PCH], device 0: ALC889 Analog [ALC889 Analog] Subdevices: 1/1 Subdevice #0: subdevice #0 card 0: PCH [HDA Intel PCH], device 1: ALC889 Digital [ALC889 Digital] Subdevices: 1/1 Subdevice #0: subdevice #0 card 1: NVidia [HDA NVidia], device 3: HDMI 0 [HDMI 0] Subdevices: 1/1 Subdevice #0: subdevice #0 card 1: NVidia [HDA NVidia], device 7: HDMI 1 [HDMI 1] Subdevices: 1/1 Subdevice #0: subdevice #0 card 1: NVidia [HDA NVidia], device 8: HDMI 2 [HDMI 2] Subdevices: 1/1 Subdevice #0: subdevice #0 card 1: NVidia [HDA NVidia], device 9: HDMI 3 [HDMI 3] Subdevices: 1/1 Subdevice #0: subdevice #0 ARECORD **** List of CAPTURE Hardware Devices **** card 0: PCH [HDA Intel PCH], device 0: ALC889 Analog [ALC889 Analog] Subdevices: 1/1 Subdevice #0: subdevice #0 card 0: PCH [HDA Intel PCH], device 2: ALC889 Alt Analog [ALC889 Alt Analog] Subdevices: 2/2 Subdevice #0: subdevice #0 Subdevice #1: subdevice #1 !!Amixer output !!------------- !!-------Mixer controls for card 0 [PCH] Card hw:0 'PCH'/'HDA Intel PCH at 0xf7420000 irq 54' Mixer name : 'Realtek ALC889' Components : 'HDA:10ec0889,1043846b,00100004' Controls : 57 Simple ctrls : 25 Simple mixer control 'Master',0 Capabilities: pvolume pvolume-joined pswitch pswitch-joined penum Playback channels: Mono Limits: Playback 0 - 64 Mono: Playback 59 [92%] [-5.00dB] [on] Simple mixer control 'Headphone',0 Capabilities: pvolume pswitch penum Playback channels: Front Left - Front Right Limits: Playback 0 - 64 Mono: Front Left: Playback 62 [97%] [0.00dB] [on] Front Right: Playback 62 [97%] [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 254 [100%] [0.20dB] Front Right: Playback 254 [100%] [0.20dB] Simple mixer control 'Front',0 Capabilities: pvolume pswitch penum Playback channels: Front Left - Front Right Limits: Playback 0 - 64 Mono: Front Left: Playback 62 [97%] [0.00dB] [on] Front Right: Playback 62 [97%] [0.00dB] [on] Simple mixer control 'Front 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 'Front 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 'Surround',0 Capabilities: pvolume pswitch penum Playback channels: Front Left - Front Right Limits: Playback 0 - 64 Mono: Front Left: Playback 62 [97%] [0.00dB] [on] Front Right: Playback 62 [97%] [0.00dB] [on] Simple mixer control 'Center',0 Capabilities: pvolume pvolume-joined pswitch pswitch-joined penum Playback channels: Mono Limits: Playback 0 - 64 Mono: Playback 62 [97%] [0.00dB] [on] Simple mixer control 'LFE',0 Capabilities: pvolume pvolume-joined pswitch pswitch-joined penum Playback channels: Mono Limits: Playback 0 - 64 Mono: Playback 62 [97%] [0.00dB] [on] Simple mixer control 'Side',0 Capabilities: pvolume pswitch penum Playback channels: Front Left - Front Right Limits: Playback 0 - 64 Mono: Front Left: Playback 62 [97%] [0.00dB] [on] Front Right: Playback 62 [97%] [0.00dB] [on] Simple mixer control 'Line',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 'Line 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 'IEC958 Default PCM',0 Capabilities: pswitch pswitch-joined penum Playback channels: Mono Mono: Playback [on] Simple mixer control 'Beep',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 'Capture',0 Capabilities: cvolume cswitch penum Capture channels: Front Left - Front Right Limits: Capture 0 - 46 Front Left: Capture 0 [0%] [-16.00dB] [on] Front Right: Capture 0 [0%] [-16.00dB] [on] Simple mixer control 'Capture',1 Capabilities: cvolume cswitch penum Capture channels: Front Left - Front Right Limits: Capture 0 - 46 Front Left: Capture 0 [0%] [-16.00dB] [off] Front Right: Capture 0 [0%] [-16.00dB] [off] Simple mixer control 'Capture',2 Capabilities: cvolume cswitch penum Capture channels: Front Left - Front Right Limits: Capture 0 - 46 Front Left: Capture 0 [0%] [-16.00dB] [off] Front Right: Capture 0 [0%] [-16.00dB] [off] 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 60 [50%] [0.00dB] Front Right: Capture 60 [50%] [0.00dB] Simple mixer control 'Input Source',0 Capabilities: cenum Items: 'Front Mic' 'Rear Mic' 'Line' Item0: 'Front Mic' Simple mixer control 'Input Source',1 Capabilities: cenum Items: 'Front Mic' 'Rear Mic' 'Line' Item0: 'Front Mic' Simple mixer control 'Input Source',2 Capabilities: cenum Items: 'Front Mic' 'Rear Mic' 'Line' Item0: 'Front Mic' Simple mixer control 'Rear 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 'Rear 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] !!-------Mixer controls for card 1 [NVidia] Card hw:1 'NVidia'/'HDA NVidia at 0xf7080000 irq 17' Mixer name : 'Nvidia GPU 41 HDMI/DP' Components : 'HDA:10de0041,10438422,00100100' Controls : 28 Simple ctrls : 4 Simple mixer control 'IEC958',0 Capabilities: pswitch pswitch-joined penum Playback channels: Mono Mono: Playback [on] Simple mixer control 'IEC958',1 Capabilities: pswitch pswitch-joined penum Playback channels: Mono Mono: Playback [on] Simple mixer control 'IEC958',2 Capabilities: pswitch pswitch-joined penum Playback channels: Mono Mono: Playback [on] Simple mixer control 'IEC958',3 Capabilities: pswitch pswitch-joined penum Playback channels: Mono Mono: Playback [on] !!Alsactl output !!-------------- --startcollapse-- state.PCH { control.1 { iface MIXER name 'Front Playback Volume' value.0 62 value.1 62 comment { access 'read write' type INTEGER count 2 range '0 - 64' dbmin -6200 dbmax 200 dbvalue.0 0 dbvalue.1 0 } } control.2 { iface MIXER name 'Front Playback Switch' value.0 true value.1 true comment { access 'read write' type BOOLEAN count 2 } } control.3 { iface MIXER name 'Surround Playback Volume' value.0 62 value.1 62 comment { access 'read write' type INTEGER count 2 range '0 - 64' dbmin -6200 dbmax 200 dbvalue.0 0 dbvalue.1 0 } } control.4 { iface MIXER name 'Surround Playback Switch' value.0 true value.1 true comment { access 'read write' type BOOLEAN count 2 } } control.5 { iface MIXER name 'Center Playback Volume' value 62 comment { access 'read write' type INTEGER count 1 range '0 - 64' dbmin -6200 dbmax 200 dbvalue.0 0 } } control.6 { iface MIXER name 'LFE Playback Volume' value 62 comment { access 'read write' type INTEGER count 1 range '0 - 64' dbmin -6200 dbmax 200 dbvalue.0 0 } } control.7 { iface MIXER name 'Center Playback Switch' value true comment { access 'read write' type BOOLEAN count 1 } } control.8 { iface MIXER name 'LFE Playback Switch' value true comment { access 'read write' type BOOLEAN count 1 } } control.9 { iface MIXER name 'Side Playback Volume' value.0 62 value.1 62 comment { access 'read write' type INTEGER count 2 range '0 - 64' dbmin -6200 dbmax 200 dbvalue.0 0 dbvalue.1 0 } } control.10 { iface MIXER name 'Side Playback Switch' value.0 true value.1 true comment { access 'read write' type BOOLEAN count 2 } } control.11 { iface MIXER name 'Headphone Playback Volume' value.0 62 value.1 62 comment { access 'read write' type INTEGER count 2 range '0 - 64' dbmin -6200 dbmax 200 dbvalue.0 0 dbvalue.1 0 } } control.12 { iface MIXER name 'Headphone Playback Switch' value.0 true value.1 true comment { access 'read write' type BOOLEAN count 2 } } control.13 { iface MIXER name 'Front Mic Playback Volume' value.0 0 value.1 0 comment { access 'read write' type INTEGER count 2 range '0 - 31' dbmin -3450 dbmax 1200 dbvalue.0 -3450 dbvalue.1 -3450 } } control.14 { iface MIXER name 'Front Mic Playback Switch' value.0 false value.1 false comment { access 'read write' type BOOLEAN count 2 } } control.15 { iface MIXER name 'Rear Mic Playback Volume' value.0 0 value.1 0 comment { access 'read write' type INTEGER count 2 range '0 - 31' dbmin -3450 dbmax 1200 dbvalue.0 -3450 dbvalue.1 -3450 } } control.16 { iface MIXER name 'Rear Mic Playback Switch' value.0 false value.1 false comment { access 'read write' type BOOLEAN count 2 } } control.17 { iface MIXER name 'Line Playback Volume' value.0 0 value.1 0 comment { access 'read write' type INTEGER count 2 range '0 - 31' dbmin -3450 dbmax 1200 dbvalue.0 -3450 dbvalue.1 -3450 } } control.18 { iface MIXER name 'Line Playback Switch' value.0 false value.1 false comment { access 'read write' type BOOLEAN count 2 } } control.19 { iface MIXER name 'Auto-Mute Mode' value Enabled comment { access 'read write' type ENUMERATED count 1 item.0 Disabled item.1 Enabled } } control.20 { iface MIXER name 'Input Source' value 'Front Mic' comment { access 'read write' type ENUMERATED count 1 item.0 'Front Mic' item.1 'Rear Mic' item.2 Line } } control.21 { iface MIXER name 'Input Source' index 1 value 'Front Mic' comment { access 'read write' type ENUMERATED count 1 item.0 'Front Mic' item.1 'Rear Mic' item.2 Line } } control.22 { iface MIXER name 'Input Source' index 2 value 'Front Mic' comment { access 'read write' type ENUMERATED count 1 item.0 'Front Mic' item.1 'Rear Mic' item.2 Line } } control.23 { iface MIXER name 'Capture Volume' value.0 0 value.1 0 comment { access 'read write' type INTEGER count 2 range '0 - 46' dbmin -1600 dbmax 3000 dbvalue.0 -1600 dbvalue.1 -1600 } } control.24 { iface MIXER name 'Capture Switch' value.0 true value.1 true comment { access 'read write' type BOOLEAN count 2 } } control.25 { iface MIXER name 'Capture Volume' index 1 value.0 0 value.1 0 comment { access 'read write' type INTEGER count 2 range '0 - 46' dbmin -1600 dbmax 3000 dbvalue.0 -1600 dbvalue.1 -1600 } } control.26 { iface MIXER name 'Capture Switch' index 1 value.0 false value.1 false comment { access 'read write' type BOOLEAN count 2 } } control.27 { iface MIXER name 'Capture Volume' index 2 value.0 0 value.1 0 comment { access 'read write' type INTEGER count 2 range '0 - 46' dbmin -1600 dbmax 3000 dbvalue.0 -1600 dbvalue.1 -1600 } } control.28 { iface MIXER name 'Capture Switch' index 2 value.0 false value.1 false comment { access 'read write' type BOOLEAN count 2 } } control.29 { iface MIXER name 'Front Mic Boost Volume' value.0 0 value.1 0 comment { access 'read write' type INTEGER count 2 range '0 - 3' dbmin 0 dbmax 3000 dbvalue.0 0 dbvalue.1 0 } } control.30 { iface MIXER name 'Rear Mic Boost Volume' value.0 0 value.1 0 comment { access 'read write' type INTEGER count 2 range '0 - 3' dbmin 0 dbmax 3000 dbvalue.0 0 dbvalue.1 0 } } control.31 { iface MIXER name 'Line Boost Volume' value.0 0 value.1 0 comment { access 'read write' type INTEGER count 2 range '0 - 3' dbmin 0 dbmax 3000 dbvalue.0 0 dbvalue.1 0 } } control.32 { iface MIXER name 'IEC958 Playback Con Mask' value '0fff000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000' comment { access read type IEC958 count 1 } } control.33 { iface MIXER name 'IEC958 Playback Pro Mask' value '0f00000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000' comment { access read type IEC958 count 1 } } control.34 { iface MIXER name 'IEC958 Playback Default' value '0400000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000' comment { access 'read write' type IEC958 count 1 } } control.35 { iface MIXER name 'IEC958 Playback Switch' value false comment { access 'read write' type BOOLEAN count 1 } } control.36 { iface MIXER name 'IEC958 Default PCM Playback Switch' value true comment { access 'read write' type BOOLEAN count 1 } } control.37 { iface MIXER name 'Master Playback Volume' value 59 comment { access 'read write' type INTEGER count 1 range '0 - 64' dbmin -6400 dbmax 0 dbvalue.0 -500 } } control.38 { iface MIXER name 'Master Playback Switch' value true comment { access 'read write' type BOOLEAN count 1 } } control.39 { iface CARD name 'Front Mic Jack' value false comment { access read type BOOLEAN count 1 } } control.40 { iface CARD name 'Rear Mic Jack' value false comment { access read type BOOLEAN count 1 } } control.41 { iface CARD name 'Line Jack' value false comment { access read type BOOLEAN count 1 } } control.42 { iface CARD name 'Line Out Front Jack' value true comment { access read type BOOLEAN count 1 } } control.43 { iface CARD name 'Line Out Surround Jack' value false comment { access read type BOOLEAN count 1 } } control.44 { iface CARD name 'Line Out CLFE Jack' value false comment { access read type BOOLEAN count 1 } } control.45 { iface CARD name 'Line Out Side Jack' value false comment { access read type BOOLEAN count 1 } } control.46 { iface CARD name 'Front Headphone Jack' value false comment { access read type BOOLEAN count 1 } } control.47 { iface CARD name 'SPDIF Phantom Jack' value true comment { access read type BOOLEAN count 1 } } control.48 { iface CARD name 'SPDIF Phantom Jack' index 1 value true comment { access read type BOOLEAN count 1 } } control.49 { iface MIXER name 'Beep Playback Volume' value.0 0 value.1 0 comment { access 'read write' type INTEGER count 2 range '0 - 31' dbmin -3450 dbmax 1200 dbvalue.0 -3450 dbvalue.1 -3450 } } control.50 { iface MIXER name 'Beep Playback Switch' value.0 false value.1 false comment { access 'read write' type BOOLEAN count 2 } } control.51 { iface PCM name 'Playback Channel Map' value.0 0 value.1 0 value.2 0 value.3 0 value.4 0 value.5 0 value.6 0 value.7 0 comment { access read type INTEGER count 8 range '0 - 36' } } control.52 { iface PCM name 'Capture Channel Map' value.0 0 value.1 0 comment { access read type INTEGER count 2 range '0 - 36' } } control.53 { iface PCM device 1 name 'Playback Channel Map' value.0 0 value.1 0 comment { access read type INTEGER count 2 range '0 - 36' } } control.54 { iface PCM device 2 name 'Capture Channel Map' value.0 0 value.1 0 comment { access read type INTEGER count 2 range '0 - 36' } } control.55 { iface PCM device 2 name 'Capture Channel Map' index 1 value.0 0 value.1 0 comment { access read type INTEGER count 2 range '0 - 36' } } control.56 { iface MIXER name 'PCM Playback Volume' value.0 254 value.1 254 comment { access 'read write user' type INTEGER count 2 range '0 - 255' tlv '0000000100000008ffffec1400000014' dbmin -5100 dbmax 0 dbvalue.0 -20 dbvalue.1 -20 } } control.57 { iface MIXER name 'Digital Capture Volume' value.0 60 value.1 60 comment { access 'read write user' type INTEGER count 2 range '0 - 120' tlv '0000000100000008fffff44800000032' dbmin -3000 dbmax 3000 dbvalue.0 0 dbvalue.1 0 } } } state.NVidia { control.1 { iface CARD name 'HDMI/DP,pcm=3 Jack' value false comment { access read type BOOLEAN count 1 } } control.2 { iface MIXER name 'IEC958 Playback Con Mask' value '0fff000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000' comment { access read type IEC958 count 1 } } control.3 { iface MIXER name 'IEC958 Playback Pro Mask' value '0f00000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000' comment { access read type IEC958 count 1 } } control.4 { iface MIXER name 'IEC958 Playback Default' value '0482000200000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000' comment { access 'read write' type IEC958 count 1 } } control.5 { iface MIXER name 'IEC958 Playback Switch' value true comment { access 'read write' type BOOLEAN count 1 } } control.6 { iface PCM device 3 name ELD value '' comment { access 'read volatile' type BYTES count 0 } } control.7 { iface CARD name 'HDMI/DP,pcm=7 Jack' value false comment { access read type BOOLEAN count 1 } } control.8 { iface MIXER name 'IEC958 Playback Con Mask' index 1 value '0fff000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000' comment { access read type IEC958 count 1 } } control.9 { iface MIXER name 'IEC958 Playback Pro Mask' index 1 value '0f00000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000' comment { access read type IEC958 count 1 } } control.10 { iface MIXER name 'IEC958 Playback Default' index 1 value '0400000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000' comment { access 'read write' type IEC958 count 1 } } control.11 { iface MIXER name 'IEC958 Playback Switch' index 1 value true comment { access 'read write' type BOOLEAN count 1 } } control.12 { iface PCM device 7 name ELD value '' comment { access 'read volatile' type BYTES count 0 } } control.13 { iface CARD name 'HDMI/DP,pcm=8 Jack' value false comment { access read type BOOLEAN count 1 } } control.14 { iface MIXER name 'IEC958 Playback Con Mask' index 2 value '0fff000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000' comment { access read type IEC958 count 1 } } control.15 { iface MIXER name 'IEC958 Playback Pro Mask' index 2 value '0f00000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000' comment { access read type IEC958 count 1 } } control.16 { iface MIXER name 'IEC958 Playback Default' index 2 value '0400000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000' comment { access 'read write' type IEC958 count 1 } } control.17 { iface MIXER name 'IEC958 Playback Switch' index 2 value true comment { access 'read write' type BOOLEAN count 1 } } control.18 { iface PCM device 8 name ELD value '' comment { access 'read volatile' type BYTES count 0 } } control.19 { iface CARD name 'HDMI/DP,pcm=9 Jack' value false comment { access read type BOOLEAN count 1 } } control.20 { iface MIXER name 'IEC958 Playback Con Mask' index 3 value '0fff000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000' comment { access read type IEC958 count 1 } } control.21 { iface MIXER name 'IEC958 Playback Pro Mask' index 3 value '0f00000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000' comment { access read type IEC958 count 1 } } control.22 { iface MIXER name 'IEC958 Playback Default' index 3 value '0400000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000' comment { access 'read write' type IEC958 count 1 } } control.23 { iface MIXER name 'IEC958 Playback Switch' index 3 value true comment { access 'read write' type BOOLEAN count 1 } } control.24 { iface PCM device 9 name ELD value '' comment { access 'read volatile' type BYTES count 0 } } control.25 { iface PCM device 3 name 'Playback Channel Map' value.0 0 value.1 0 value.2 0 value.3 0 value.4 0 value.5 0 value.6 0 value.7 0 comment { access 'read write' type INTEGER count 8 range '0 - 36' } } control.26 { iface PCM device 7 name 'Playback Channel Map' value.0 0 value.1 0 value.2 0 value.3 0 value.4 0 value.5 0 value.6 0 value.7 0 comment { access 'read write' type INTEGER count 8 range '0 - 36' } } control.27 { iface PCM device 8 name 'Playback Channel Map' value.0 0 value.1 0 value.2 0 value.3 0 value.4 0 value.5 0 value.6 0 value.7 0 comment { access 'read write' type INTEGER count 8 range '0 - 36' } } control.28 { iface PCM device 9 name 'Playback Channel Map' value.0 0 value.1 0 value.2 0 value.3 0 value.4 0 value.5 0 value.6 0 value.7 0 comment { access 'read write' type INTEGER count 8 range '0 - 36' } } } --endcollapse-- !!All Loaded Modules !!------------------ Module pci_stub vboxpci vboxnetadp vboxnetflt vboxdrv snd_hda_codec_hdmi hid_generic usbhid hid rfcomm bnep parport_pc bluetooth ppdev joydev snd_hda_codec_realtek xpad ff_memless snd_hda_intel snd_hda_codec snd_hwdep snd_pcm snd_page_alloc nls_iso8859_1 snd_seq_midi eeepc_wmi asus_wmi sparse_keymap snd_seq_midi_event snd_rawmidi snd_seq snd_seq_device nouveau psmouse snd_timer serio_raw snd mxm_wmi ttm drm_kms_helper drm lpc_ich i2c_algo_bit wmi video mac_hid soundcore mei_me mei lp parport usb_storage ahci libahci e1000e ptp pps_core !!Sysfs Files !!----------- /sys/class/sound/hwC0D0/init_pin_configs: 0x11 0x99430140 0x12 0x411111f0 0x14 0x01014010 0x15 0x01011012 0x16 0x01016011 0x17 0x01012014 0x18 0x01a19850 0x19 0x02a19c60 0x1a 0x0181305f 0x1b 0x02214c20 0x1c 0x411111f0 0x1d 0x4007e629 0x1e 0x01451130 0x1f 0x411111f0 /sys/class/sound/hwC0D0/driver_pin_configs: /sys/class/sound/hwC0D0/user_pin_configs: /sys/class/sound/hwC0D0/init_verbs: /sys/class/sound/hwC0D0/hints: /sys/class/sound/hwC1D0/init_pin_configs: 0x04 0x185600f0 0x05 0x185600f0 0x06 0x185600f0 0x07 0x185600f0 /sys/class/sound/hwC1D0/driver_pin_configs: /sys/class/sound/hwC1D0/user_pin_configs: /sys/class/sound/hwC1D0/init_verbs: /sys/class/sound/hwC1D0/hints: !!ALSA/HDA dmesg !!-------------- [ 2.420876] usb 2-1.3: new full-speed USB device number 3 using ehci-pci [ 2.428149] snd_hda_intel 0000:00:1b.0: enabling device (0000 -> 0002) [ 2.428271] snd_hda_intel 0000:00:1b.0: irq 54 for MSI/MSI-X [ 2.441009] input: Microsoft X-Box 360 pad as /devices/pci0000:00/0000:00:1c.4/0000:03:00.0/usb3/3-2/3-2:1.0/input/input3 -- [ 2.445106] realtek: Enabling init ASM_ID=0x846b CODEC_ID=10ec0889 [ 2.468021] input: HDA Intel PCH Front Headphone as /devices/pci0000:00/0000:00:1b.0/sound/card0/input4 [ 2.468197] input: HDA Intel PCH Line Out Side as /devices/pci0000:00/0000:00:1b.0/sound/card0/input5 [ 2.468305] input: HDA Intel PCH Line Out CLFE as /devices/pci0000:00/0000:00:1b.0/sound/card0/input6 [ 2.468419] input: HDA Intel PCH Line Out Surround as /devices/pci0000:00/0000:00:1b.0/sound/card0/input7 [ 2.468529] input: HDA Intel PCH Line Out Front as /devices/pci0000:00/0000:00:1b.0/sound/card0/input8 [ 2.468644] input: HDA Intel PCH Line as /devices/pci0000:00/0000:00:1b.0/sound/card0/input9 [ 2.468720] input: HDA Intel PCH Rear Mic as /devices/pci0000:00/0000:00:1b.0/sound/card0/input10 [ 2.468788] input: HDA Intel PCH Front Mic as /devices/pci0000:00/0000:00:1b.0/sound/card0/input11 [ 2.502287] init: avahi-cups-reload main process (1122) terminated with status 1 -- [ 2.592465] [drm] Initialized nouveau 1.1.1 20120801 for 0000:01:00.0 on minor 0 [ 2.592525] snd_hda_intel 0000:01:00.1: enabling device (0000 -> 0002) [ 2.592581] hda_intel: Disabling MSI [ 2.592587] hda-intel 0000:01:00.1: Handle VGA-switcheroo audio client [ 2.592621] hda-intel 0000:01:00.1: Disabling 64bit DMA [ 2.593469] usb 2-1.4: new low-speed USB device number 4 using ehci-pci [ 2.599949] hda-intel 0000:01:00.1: Enable delay in RIRB handling [ 2.613131] usbcore: registered new interface driver usbhid -- [ 3.060378] sd 9:0:0:0: [sdd] Attached SCSI disk [ 3.093719] input: HDA NVidia HDMI/DP,pcm=9 as /devices/pci0000:00/0000:00:01.0/0000:01:00.1/sound/card1/input15 [ 3.093814] input: HDA NVidia HDMI/DP,pcm=8 as /devices/pci0000:00/0000:00:01.0/0000:01:00.1/sound/card1/input16 [ 3.093882] input: HDA NVidia HDMI/DP,pcm=7 as /devices/pci0000:00/0000:00:01.0/0000:01:00.1/sound/card1/input17 [ 3.093948] input: HDA NVidia HDMI/DP,pcm=3 as /devices/pci0000:00/0000:00:01.0/0000:01:00.1/sound/card1/input18 [ 3.190738] scsi 10:0:0:0: Direct-Access WD Ext HDD 1021 2002 PQ: 0 ANSI: 4 -- [ 4.633575] IPv6: ADDRCONF(NETDEV_CHANGE): eth0: link becomes ready [ 4.694338] hda-intel: IRQ timing workaround is activated for card #1. Suggest a bigger bdl_pos_adj. pulseverbose.log: ( 0.000| 0.000) I: [pulseaudio] main.c: setrlimit(RLIMIT_NICE, (31, 31)) failed: Operation not permitted ( 0.000| 0.000) I: [pulseaudio] main.c: setrlimit(RLIMIT_RTPRIO, (9, 9)) failed: Operation not permitted ( 0.000| 0.000) D: [pulseaudio] core-rtclock.c: Timer slack is set to 50 us. ( 0.003| 0.003) D: [pulseaudio] core-util.c: RealtimeKit worked. ( 0.003| 0.000) I: [pulseaudio] core-util.c: Successfully gained nice level -11. ( 0.003| 0.000) I: [pulseaudio] main.c: This is PulseAudio 3.0 ( 0.003| 0.000) D: [pulseaudio] main.c: Compilation host: x86_64-pc-linux-gnu ( 0.003| 0.000) D: [pulseaudio] main.c: Compilation CFLAGS: -g -O2 -fstack-protector --param=ssp-buffer-size=4 -Wformat -Werror=format-security -Wall -W -Wextra -pipe -Wno-long-long -Wno-overlength-strings -Wunsafe-loop-optimizations -Wundef -Wformat=2 -Wlogical-op -Wsign-compare -Wformat-security -Wmissing-include-dirs -Wformat-nonliteral -Wpointer-arith -Winit-self -Wdeclaration-after-statement -Wfloat-equal -Wmissing-prototypes -Wredundant-decls -Wmissing-declarations -Wmissing-noreturn -Wshadow -Wendif-labels -Wcast-align -Wstrict-aliasing -Wwrite-strings -Wno-unused-parameter -ffast-math -Wp,-D_FORTIFY_SOURCE=2 -fno-common -fdiagnostics-show-option ( 0.003| 0.000) D: [pulseaudio] main.c: Running on host: Linux x86_64 3.11.0-031100rc2-generic #201307211535 SMP Sun Jul 21 19:36:42 UTC 2013 ( 0.003| 0.000) D: [pulseaudio] main.c: Found 4 CPUs. ( 0.003| 0.000) I: [pulseaudio] main.c: Page size is 4096 bytes ( 0.003| 0.000) D: [pulseaudio] main.c: Compiled with Valgrind support: no ( 0.003| 0.000) D: [pulseaudio] main.c: Running in valgrind mode: no ( 0.003| 0.000) D: [pulseaudio] main.c: Running in VM: no ( 0.003| 0.000) D: [pulseaudio] main.c: Optimized build: yes ( 0.003| 0.000) D: [pulseaudio] main.c: FASTPATH defined, only fast path asserts disabled. ( 0.003| 0.000) I: [pulseaudio] main.c: Machine ID is 7bd695a64b57f140878196e351d1dc20. ( 0.003| 0.000) I: [pulseaudio] main.c: Using runtime directory /run/user/vincent/pulse. ( 0.003| 0.000) I: [pulseaudio] main.c: Using state directory /home/vincent/.config/pulse. ( 0.003| 0.000) I: [pulseaudio] main.c: Using modules directory /usr/lib/pulse-3.0/modules. ( 0.003| 0.000) I: [pulseaudio] main.c: Running in system mode: no ( 0.003| 0.000) I: [pulseaudio] main.c: Fresh high-resolution timers available! Bon appetit! ( 0.004| 0.000) D: [pulseaudio] memblock.c: Using shared memory pool with 1024 slots of size 64.0 KiB each, total size is 64.0 MiB, maximum usable slot size is 65472 ( 0.004| 0.000) I: [pulseaudio] cpu-x86.c: CPU flags: CMOV MMX SSE SSE2 SSE3 SSSE3 SSE4_1 SSE4_2 ( 0.004| 0.000) I: [pulseaudio] svolume_mmx.c: Initialising MMX optimized volume functions. ( 0.004| 0.000) I: [pulseaudio] remap_mmx.c: Initialising MMX optimized remappers. ( 0.004| 0.000) I: [pulseaudio] svolume_sse.c: Initialising SSE2 optimized volume functions. ( 0.004| 0.000) I: [pulseaudio] remap_sse.c: Initialising SSE2 optimized remappers. ( 0.004| 0.000) I: [pulseaudio] sconv_sse.c: Initialising SSE2 optimized conversions. ( 0.004| 0.000) I: [pulseaudio] svolume_orc.c: Initialising ORC optimized volume functions. ( 0.004| 0.000) D: [pulseaudio] database-tdb.c: Opened TDB database '/home/vincent/.config/pulse/7bd695a64b57f140878196e351d1dc20-device-volumes.tdb' ( 0.004| 0.000) I: [pulseaudio] module-device-restore.c: Successfully opened database file '/home/vincent/.config/pulse/7bd695a64b57f140878196e351d1dc20-device-volumes'. ( 0.004| 0.000) I: [pulseaudio] module.c: Loaded "module-device-restore" (index: #0; argument: ""). ( 0.004| 0.000) D: [pulseaudio] database-tdb.c: Opened TDB database '/home/vincent/.config/pulse/7bd695a64b57f140878196e351d1dc20-stream-volumes.tdb' ( 0.004| 0.000) I: [pulseaudio] module-stream-restore.c: Successfully opened database file '/home/vincent/.config/pulse/7bd695a64b57f140878196e351d1dc20-stream-volumes'. ( 0.005| 0.000) D: [pulseaudio] protocol-dbus.c: Interface org.PulseAudio.Ext.StreamRestore1 added for object /org/pulseaudio/stream_restore1 ( 0.005| 0.000) D: [pulseaudio] protocol-dbus.c: Interface org.PulseAudio.Ext.StreamRestore1.RestoreEntry added for object /org/pulseaudio/stream_restore1/entry0 ( 0.005| 0.000) D: [pulseaudio] protocol-dbus.c: Interface org.PulseAudio.Ext.StreamRestore1.RestoreEntry added for object /org/pulseaudio/stream_restore1/entry1 ( 0.005| 0.000) D: [pulseaudio] protocol-dbus.c: Interface org.PulseAudio.Ext.StreamRestore1.RestoreEntry added for object /org/pulseaudio/stream_restore1/entry2 ( 0.005| 0.000) D: [pulseaudio] protocol-dbus.c: Interface org.PulseAudio.Ext.StreamRestore1.RestoreEntry added for object /org/pulseaudio/stream_restore1/entry3 ( 0.005| 0.000) D: [pulseaudio] protocol-dbus.c: Interface org.PulseAudio.Ext.StreamRestore1.RestoreEntry added for object /org/pulseaudio/stream_restore1/entry4 ( 0.005| 0.000) D: [pulseaudio] protocol-dbus.c: Interface org.PulseAudio.Ext.StreamRestore1.RestoreEntry added for object /org/pulseaudio/stream_restore1/entry5 ( 0.005| 0.000) D: [pulseaudio] protocol-dbus.c: Interface org.PulseAudio.Ext.StreamRestore1.RestoreEntry added for object /org/pulseaudio/stream_restore1/entry6 ( 0.005| 0.000) D: [pulseaudio] protocol-dbus.c: Interface org.PulseAudio.Ext.StreamRestore1.RestoreEntry added for object /org/pulseaudio/stream_restore1/entry7 ( 0.005| 0.000) D: [pulseaudio] protocol-dbus.c: Interface org.PulseAudio.Ext.StreamRestore1.RestoreEntry added for object /org/pulseaudio/stream_restore1/entry8 ( 0.005| 0.000) D: [pulseaudio] protocol-dbus.c: Interface org.PulseAudio.Ext.StreamRestore1.RestoreEntry added for object /org/pulseaudio/stream_restore1/entry9 ( 0.005| 0.000) D: [pulseaudio] protocol-dbus.c: Interface org.PulseAudio.Ext.StreamRestore1.RestoreEntry added for object /org/pulseaudio/stream_restore1/entry10 ( 0.005| 0.000) D: [pulseaudio] protocol-dbus.c: Interface org.PulseAudio.Ext.StreamRestore1.RestoreEntry added for object /org/pulseaudio/stream_restore1/entry11 ( 0.005| 0.000) D: [pulseaudio] protocol-dbus.c: Interface org.PulseAudio.Ext.StreamRestore1.RestoreEntry added for object /org/pulseaudio/stream_restore1/entry12 ( 0.005| 0.000) D: [pulseaudio] protocol-dbus.c: Interface org.PulseAudio.Ext.StreamRestore1.RestoreEntry added for object /org/pulseaudio/stream_restore1/entry13 ( 0.005| 0.000) D: [pulseaudio] protocol-dbus.c: Interface org.PulseAudio.Ext.StreamRestore1.RestoreEntry added for object /org/pulseaudio/stream_restore1/entry14 ( 0.005| 0.000) D: [pulseaudio] protocol-dbus.c: Interface org.PulseAudio.Ext.StreamRestore1.RestoreEntry added for object /org/pulseaudio/stream_restore1/entry15 ( 0.005| 0.000) D: [pulseaudio] protocol-dbus.c: Interface org.PulseAudio.Ext.StreamRestore1.RestoreEntry added for object /org/pulseaudio/stream_restore1/entry16 ( 0.005| 0.000) D: [pulseaudio] protocol-dbus.c: Interface org.PulseAudio.Ext.StreamRestore1.RestoreEntry added for object /org/pulseaudio/stream_restore1/entry17 ( 0.005| 0.000) D: [pulseaudio] protocol-dbus.c: Interface org.PulseAudio.Ext.StreamRestore1.RestoreEntry added for object /org/pulseaudio/stream_restore1/entry18 ( 0.005| 0.000) D: [pulseaudio] protocol-dbus.c: Interface org.PulseAudio.Ext.StreamRestore1.RestoreEntry added for object /org/pulseaudio/stream_restore1/entry19 ( 0.005| 0.000) D: [pulseaudio] protocol-dbus.c: Interface org.PulseAudio.Ext.StreamRestore1.RestoreEntry added for object /org/pulseaudio/stream_restore1/entry20 ( 0.005| 0.000) D: [pulseaudio] protocol-dbus.c: Interface org.PulseAudio.Ext.StreamRestore1.RestoreEntry added for object /org/pulseaudio/stream_restore1/entry21 ( 0.005| 0.000) D: [pulseaudio] protocol-dbus.c: Interface org.PulseAudio.Ext.StreamRestore1.RestoreEntry added for object /org/pulseaudio/stream_restore1/entry22 ( 0.005| 0.000) D: [pulseaudio] protocol-dbus.c: Interface org.PulseAudio.Ext.StreamRestore1.RestoreEntry added for object /org/pulseaudio/stream_restore1/entry23 ( 0.005| 0.000) D: [pulseaudio] protocol-dbus.c: Interface org.PulseAudio.Ext.StreamRestore1.RestoreEntry added for object /org/pulseaudio/stream_restore1/entry24 ( 0.005| 0.000) D: [pulseaudio] protocol-dbus.c: Interface org.PulseAudio.Ext.StreamRestore1.RestoreEntry added for object /org/pulseaudio/stream_restore1/entry25 ( 0.005| 0.000) D: [pulseaudio] protocol-dbus.c: Interface org.PulseAudio.Ext.StreamRestore1.RestoreEntry added for object /org/pulseaudio/stream_restore1/entry26 ( 0.005| 0.000) D: [pulseaudio] protocol-dbus.c: Interface org.PulseAudio.Ext.StreamRestore1.RestoreEntry added for object /org/pulseaudio/stream_restore1/entry27 ( 0.005| 0.000) I: [pulseaudio] module.c: Loaded "module-stream-restore" (index: #1; argument: ""). ( 0.005| 0.000) D: [pulseaudio] database-tdb.c: Opened TDB database '/home/vincent/.config/pulse/7bd695a64b57f140878196e351d1dc20-card-database.tdb' ( 0.005| 0.000) I: [pulseaudio] module-card-restore.c: Successfully opened database file '/home/vincent/.config/pulse/7bd695a64b57f140878196e351d1dc20-card-database'. ( 0.005| 0.000) I: [pulseaudio] module.c: Loaded "module-card-restore" (index: #2; argument: ""). ( 0.006| 0.000) I: [pulseaudio] module.c: Loaded "module-augment-properties" (index: #3; argument: ""). ( 0.006| 0.000) D: [pulseaudio] cli-command.c: Checking for existence of '/usr/lib/pulse-3.0/modules/module-udev-detect.so': success ( 0.006| 0.000) D: [pulseaudio] module-udev-detect.c: /dev/snd/controlC1 is accessible: yes ( 0.006| 0.000) D: [pulseaudio] module-udev-detect.c: /devices/pci0000:00/0000:00:01.0/0000:01:00.1/sound/card1 is busy: no ( 0.006| 0.000) D: [pulseaudio] module-udev-detect.c: Loading module-alsa-card with arguments 'device_id="1" name="pci-0000_01_00.1" card_name="alsa_card.pci-0000_01_00.1" namereg_fail=false tsched=no fixed_latency_range=no ignore_dB=no deferred_volume=yes use_ucm=yes card_properties="module-udev-detect.discovered=1"' ( 0.008| 0.001) D: [pulseaudio] dbus-util.c: Successfully connected to D-Bus session bus 8923dc8a05563d0ffe2f214c51ed7550 as :1.110 ( 0.008| 0.000) D: [pulseaudio] reserve-wrap.c: Successfully acquired reservation lock on device 'Audio1' ( 0.008| 0.000) I: [pulseaudio] (alsa-lib)utils.c: could not open configuration file /usr/share/alsa/ucm/HDA NVidia/HDA NVidia.conf ( 0.008| 0.000) I: [pulseaudio] (alsa-lib)parser.c: error: could not parse configuration for card HDA NVidia ( 0.008| 0.000) I: [pulseaudio] (alsa-lib)main.c: error: failed to import HDA NVidia use case configuration -2 ( 0.008| 0.000) I: [pulseaudio] alsa-ucm.c: UCM not available for card HDA NVidia ( 0.009| 0.000) D: [pulseaudio] alsa-mixer.c: Looking at profile input:analog-mono ( 0.009| 0.000) D: [pulseaudio] alsa-mixer.c: Checking for recording on Analog Mono (analog-mono) ( 0.009| 0.000) D: [pulseaudio] alsa-util.c: Trying hw:1 with SND_PCM_NO_AUTO_FORMAT ... ( 0.009| 0.000) I: [pulseaudio] (alsa-lib)pcm_hw.c: open '/dev/snd/pcmC1D0c' failed (-2) ( 0.009| 0.000) I: [pulseaudio] alsa-util.c: Error opening PCM device hw:1: No such file or directory ( 0.010| 0.000) D: [pulseaudio] alsa-mixer.c: Caching failure to open input:analog-mono ( 0.010| 0.000) D: [pulseaudio] alsa-mixer.c: Looking at profile input:analog-stereo ( 0.010| 0.000) D: [pulseaudio] alsa-mixer.c: Checking for recording on Analog Stereo (analog-stereo) ( 0.010| 0.000) D: [pulseaudio] alsa-util.c: Trying front:1 with SND_PCM_NO_AUTO_FORMAT ... ( 0.011| 0.001) I: [pulseaudio] (alsa-lib)pcm_hw.c: open '/dev/snd/pcmC1D0c' failed (-2) ( 0.011| 0.000) I: [pulseaudio] alsa-util.c: Error opening PCM device front:1: No such file or directory ( 0.011| 0.000) D: [pulseaudio] alsa-util.c: Trying hw:1 with SND_PCM_NO_AUTO_FORMAT ... ( 0.011| 0.000) I: [pulseaudio] (alsa-lib)pcm_hw.c: open '/dev/snd/pcmC1D0c' failed (-2) ( 0.011| 0.000) I: [pulseaudio] alsa-util.c: Error opening PCM device hw:1: No such file or directory ( 0.011| 0.000) D: [pulseaudio] alsa-mixer.c: Caching failure to open input:analog-stereo ( 0.011| 0.000) D: [pulseaudio] alsa-mixer.c: Looking at profile input:iec958-stereo ( 0.011| 0.000) D: [pulseaudio] alsa-mixer.c: Checking for recording on Digital Stereo (IEC958) (iec958-stereo) ( 0.011| 0.000) D: [pulseaudio] alsa-util.c: Trying iec958:1 with SND_PCM_NO_AUTO_FORMAT ... ( 0.011| 0.000) I: [pulseaudio] (alsa-lib)pcm_hw.c: open '/dev/snd/pcmC1D1c' failed (-2) ( 0.011| 0.000) I: [pulseaudio] alsa-util.c: Error opening PCM device iec958:1: No such file or directory ( 0.011| 0.000) D: [pulseaudio] alsa-mixer.c: Caching failure to open input:iec958-stereo ( 0.011| 0.000) D: [pulseaudio] alsa-mixer.c: Looking at profile output:analog-mono ( 0.011| 0.000) D: [pulseaudio] alsa-mixer.c: Checking for playback on Analog Mono (analog-mono) ( 0.011| 0.000) D: [pulseaudio] alsa-util.c: Trying hw:1 with SND_PCM_NO_AUTO_FORMAT ... ( 0.011| 0.000) I: [pulseaudio] (alsa-lib)pcm_hw.c: open '/dev/snd/pcmC1D0p' failed (-2) ( 0.011| 0.000) I: [pulseaudio] alsa-util.c: Error opening PCM device hw:1: No such file or directory ( 0.011| 0.000) D: [pulseaudio] alsa-mixer.c: Caching failure to open output:analog-mono ( 0.011| 0.000) D: [pulseaudio] alsa-mixer.c: Skipping profile output:analog-mono+input:analog-mono - will not be able to open output:analog-mono ( 0.011| 0.000) D: [pulseaudio] alsa-mixer.c: Skipping profile output:analog-mono+input:analog-stereo - will not be able to open output:analog-mono ( 0.011| 0.000) D: [pulseaudio] alsa-mixer.c: Skipping profile output:analog-mono+input:iec958-stereo - will not be able to open output:analog-mono ( 0.011| 0.000) D: [pulseaudio] alsa-mixer.c: Looking at profile output:analog-stereo ( 0.011| 0.000) D: [pulseaudio] alsa-mixer.c: Checking for playback on Analog Stereo (analog-stereo) ( 0.011| 0.000) D: [pulseaudio] alsa-util.c: Trying front:1 with SND_PCM_NO_AUTO_FORMAT ... ( 0.011| 0.000) I: [pulseaudio] (alsa-lib)pcm_hw.c: open '/dev/snd/pcmC1D0p' failed (-2) ( 0.011| 0.000) I: [pulseaudio] alsa-util.c: Error opening PCM device front:1: No such file or directory ( 0.011| 0.000) D: [pulseaudio] alsa-util.c: Trying hw:1 with SND_PCM_NO_AUTO_FORMAT ... ( 0.011| 0.000) I: [pulseaudio] (alsa-lib)pcm_hw.c: open '/dev/snd/pcmC1D0p' failed (-2) ( 0.011| 0.000) I: [pulseaudio] alsa-util.c: Error opening PCM device hw:1: No such file or directory ( 0.012| 0.000) D: [pulseaudio] alsa-mixer.c: Caching failure to open output:analog-stereo ( 0.012| 0.000) D: [pulseaudio] alsa-mixer.c: Skipping profile output:analog-stereo+input:analog-mono - will not be able to open output:analog-stereo ( 0.012| 0.000) D: [pulseaudio] alsa-mixer.c: Skipping profile output:analog-stereo+input:analog-stereo - will not be able to open output:analog-stereo ( 0.012| 0.000) D: [pulseaudio] alsa-mixer.c: Skipping profile output:analog-stereo+input:iec958-stereo - will not be able to open output:analog-stereo ( 0.012| 0.000) D: [pulseaudio] alsa-mixer.c: Looking at profile output:analog-surround-40 ( 0.012| 0.000) D: [pulseaudio] alsa-mixer.c: Checking for playback on Analog Surround 4.0 (analog-surround-40) ( 0.012| 0.000) D: [pulseaudio] alsa-util.c: Trying surround40:1 with SND_PCM_NO_AUTO_FORMAT ... ( 0.012| 0.000) I: [pulseaudio] (alsa-lib)pcm_hw.c: open '/dev/snd/pcmC1D0p' failed (-2) ( 0.012| 0.000) I: [pulseaudio] alsa-util.c: Error opening PCM device surround40:1: No such file or directory ( 0.012| 0.000) D: [pulseaudio] alsa-mixer.c: Caching failure to open output:analog-surround-40 ( 0.012| 0.000) D: [pulseaudio] alsa-mixer.c: Skipping profile output:analog-surround-40+input:analog-mono - will not be able to open output:analog-surround-40 ( 0.012| 0.000) D: [pulseaudio] alsa-mixer.c: Skipping profile output:analog-surround-40+input:analog-stereo - will not be able to open output:analog-surround-40 ( 0.012| 0.000) D: [pulseaudio] alsa-mixer.c: Skipping profile output:analog-surround-40+input:iec958-stereo - will not be able to open output:analog-surround-40 ( 0.012| 0.000) D: [pulseaudio] alsa-mixer.c: Looking at profile output:analog-surround-41 ( 0.012| 0.000) D: [pulseaudio] alsa-mixer.c: Checking for playback on Analog Surround 4.1 (analog-surround-41) ( 0.012| 0.000) D: [pulseaudio] alsa-util.c: Trying surround41:1 with SND_PCM_NO_AUTO_FORMAT ... ( 0.012| 0.000) I: [pulseaudio] (alsa-lib)pcm_hw.c: open '/dev/snd/pcmC1D0p' failed (-2) ( 0.012| 0.000) I: [pulseaudio] alsa-util.c: Error opening PCM device surround41:1: No such file or directory ( 0.012| 0.000) D: [pulseaudio] alsa-mixer.c: Caching failure to open output:analog-surround-41 ( 0.012| 0.000) D: [pulseaudio] alsa-mixer.c: Skipping profile output:analog-surround-41+input:analog-mono - will not be able to open output:analog-surround-41 ( 0.012| 0.000) D: [pulseaudio] alsa-mixer.c: Skipping profile output:analog-surround-41+input:analog-stereo - will not be able to open output:analog-surround-41 ( 0.012| 0.000) D: [pulseaudio] alsa-mixer.c: Skipping profile output:analog-surround-41+input:iec958-stereo - will not be able to open output:analog-surround-41 ( 0.012| 0.000) D: [pulseaudio] alsa-mixer.c: Looking at profile output:analog-surround-50 ( 0.012| 0.000) D: [pulseaudio] alsa-mixer.c: Checking for playback on Analog Surround 5.0 (analog-surround-50) ( 0.012| 0.000) D: [pulseaudio] alsa-util.c: Trying surround50:1 with SND_PCM_NO_AUTO_FORMAT ... ( 0.012| 0.000) I: [pulseaudio] (alsa-lib)pcm_hw.c: open '/dev/snd/pcmC1D0p' failed (-2) ( 0.012| 0.000) I: [pulseaudio] alsa-util.c: Error opening PCM device surround50:1: No such file or directory ( 0.012| 0.000) D: [pulseaudio] alsa-mixer.c: Caching failure to open output:analog-surround-50 ( 0.012| 0.000) D: [pulseaudio] alsa-mixer.c: Skipping profile output:analog-surround-50+input:analog-mono - will not be able to open output:analog-surround-50 ( 0.012| 0.000) D: [pulseaudio] alsa-mixer.c: Skipping profile output:analog-surround-50+input:analog-stereo - will not be able to open output:analog-surround-50 ( 0.012| 0.000) D: [pulseaudio] alsa-mixer.c: Skipping profile output:analog-surround-50+input:iec958-stereo - will not be able to open output:analog-surround-50 ( 0.012| 0.000) D: [pulseaudio] alsa-mixer.c: Looking at profile output:analog-surround-51 ( 0.012| 0.000) D: [pulseaudio] alsa-mixer.c: Checking for playback on Analog Surround 5.1 (analog-surround-51) ( 0.012| 0.000) D: [pulseaudio] alsa-util.c: Trying surround51:1 with SND_PCM_NO_AUTO_FORMAT ... ( 0.012| 0.000) I: [pulseaudio] (alsa-lib)pcm_hw.c: open '/dev/snd/pcmC1D0p' failed (-2) ( 0.012| 0.000) I: [pulseaudio] alsa-util.c: Error opening PCM device surround51:1: No such file or directory ( 0.012| 0.000) D: [pulseaudio] alsa-mixer.c: Caching failure to open output:analog-surround-51 ( 0.012| 0.000) D: [pulseaudio] alsa-mixer.c: Skipping profile output:analog-surround-51+input:analog-mono - will not be able to open output:analog-surround-51 ( 0.012| 0.000) D: [pulseaudio] alsa-mixer.c: Skipping profile output:analog-surround-51+input:analog-stereo - will not be able to open output:analog-surround-51 ( 0.012| 0.000) D: [pulseaudio] alsa-mixer.c: Skipping profile output:analog-surround-51+input:iec958-stereo - will not be able to open output:analog-surround-51 ( 0.012| 0.000) D: [pulseaudio] alsa-mixer.c: Looking at profile output:analog-surround-71 ( 0.012| 0.000) D: [pulseaudio] alsa-mixer.c: Checking for playback on Analog Surround 7.1 (analog-surround-71) ( 0.012| 0.000) D: [pulseaudio] alsa-util.c: Trying surround71:1 with SND_PCM_NO_AUTO_FORMAT ... ( 0.012| 0.000) I: [pulseaudio] (alsa-lib)pcm_hw.c: open '/dev/snd/pcmC1D0p' failed (-2) ( 0.012| 0.000) I: [pulseaudio] alsa-util.c: Error opening PCM device surround71:1: No such file or directory ( 0.012| 0.000) D: [pulseaudio] alsa-mixer.c: Caching failure to open output:analog-surround-71 ( 0.012| 0.000) D: [pulseaudio] alsa-mixer.c: Skipping profile output:analog-surround-71+input:analog-mono - will not be able to open output:analog-surround-71 ( 0.012| 0.000) D: [pulseaudio] alsa-mixer.c: Skipping profile output:analog-surround-71+input:analog-stereo - will not be able to open output:analog-surround-71 ( 0.012| 0.000) D: [pulseaudio] alsa-mixer.c: Skipping profile output:analog-surround-71+input:iec958-stereo - will not be able to open output:analog-surround-71 ( 0.012| 0.000) D: [pulseaudio] alsa-mixer.c: Looking at profile output:iec958-stereo ( 0.012| 0.000) D: [pulseaudio] alsa-mixer.c: Checking for playback on Digital Stereo (IEC958) (iec958-stereo) ( 0.012| 0.000) D: [pulseaudio] alsa-util.c: Trying iec958:1 with SND_PCM_NO_AUTO_FORMAT ... ( 0.013| 0.000) I: [pulseaudio] (alsa-lib)pcm_hw.c: open '/dev/snd/pcmC1D1p' failed (-2) ( 0.013| 0.000) I: [pulseaudio] alsa-util.c: Error opening PCM device iec958:1: No such file or directory ( 0.013| 0.000) D: [pulseaudio] alsa-mixer.c: Caching failure to open output:iec958-stereo ( 0.013| 0.000) D: [pulseaudio] alsa-mixer.c: Skipping profile output:iec958-stereo+input:analog-mono - will not be able to open output:iec958-stereo ( 0.013| 0.000) D: [pulseaudio] alsa-mixer.c: Skipping profile output:iec958-stereo+input:analog-stereo - will not be able to open output:iec958-stereo ( 0.013| 0.000) D: [pulseaudio] alsa-mixer.c: Skipping profile output:iec958-stereo+input:iec958-stereo - will not be able to open output:iec958-stereo ( 0.013| 0.000) D: [pulseaudio] alsa-mixer.c: Looking at profile output:iec958-ac3-surround-40 ( 0.013| 0.000) D: [pulseaudio] alsa-mixer.c: Checking for playback on Digital Surround 4.0 (IEC958/AC3) (iec958-ac3-surround-40) ( 0.013| 0.000) D: [pulseaudio] alsa-util.c: Trying a52:1 with SND_PCM_NO_AUTO_FORMAT ... ( 0.013| 0.000) I: [pulseaudio] (alsa-lib)pcm.c: Unknown PCM a52:1 ( 0.013| 0.000) I: [pulseaudio] alsa-util.c: Error opening PCM device a52:1: No such file or directory ( 0.013| 0.000) D: [pulseaudio] alsa-mixer.c: Caching failure to open output:iec958-ac3-surround-40 ( 0.013| 0.000) D: [pulseaudio] alsa-mixer.c: Skipping profile output:iec958-ac3-surround-40+input:analog-mono - will not be able to open output:iec958-ac3-surround-40 ( 0.013| 0.000) D: [pulseaudio] alsa-mixer.c: Skipping profile output:iec958-ac3-surround-40+input:analog-stereo - will not be able to open output:iec958-ac3-surround-40 ( 0.013| 0.000) D: [pulseaudio] alsa-mixer.c: Skipping profile output:iec958-ac3-surround-40+input:iec958-stereo - will not be able to open output:iec958-ac3-surround-40 ( 0.013| 0.000) D: [pulseaudio] alsa-mixer.c: Looking at profile output:iec958-ac3-surround-51 ( 0.013| 0.000) D: [pulseaudio] alsa-mixer.c: Checking for playback on Digital Surround 5.1 (IEC958/AC3) (iec958-ac3-surround-51) ( 0.013| 0.000) D: [pulseaudio] alsa-util.c: Trying a52:1 with SND_PCM_NO_AUTO_FORMAT ... ( 0.013| 0.000) I: [pulseaudio] (alsa-lib)pcm.c: Unknown PCM a52:1 ( 0.013| 0.000) I: [pulseaudio] alsa-util.c: Error opening PCM device a52:1: No such file or directory ( 0.013| 0.000) D: [pulseaudio] alsa-mixer.c: Caching failure to open output:iec958-ac3-surround-51 ( 0.013| 0.000) D: [pulseaudio] alsa-mixer.c: Skipping profile output:iec958-ac3-surround-51+input:analog-mono - will not be able to open output:iec958-ac3-surround-51 ( 0.013| 0.000) D: [pulseaudio] alsa-mixer.c: Skipping profile output:iec958-ac3-surround-51+input:analog-stereo - will not be able to open output:iec958-ac3-surround-51 ( 0.013| 0.000) D: [pulseaudio] alsa-mixer.c: Skipping profile output:iec958-ac3-surround-51+input:iec958-stereo - will not be able to open output:iec958-ac3-surround-51 ( 0.013| 0.000) D: [pulseaudio] alsa-mixer.c: Looking at profile output:iec958-dts-surround-51 ( 0.013| 0.000) D: [pulseaudio] alsa-mixer.c: Checking for playback on Digital Surround 5.1 (IEC958/DTS) (iec958-dts-surround-51) ( 0.013| 0.000) D: [pulseaudio] alsa-util.c: Trying dca:1 with SND_PCM_NO_AUTO_FORMAT ... ( 0.013| 0.000) I: [pulseaudio] (alsa-lib)pcm.c: Unknown PCM dca:1 ( 0.013| 0.000) I: [pulseaudio] alsa-util.c: Error opening PCM device dca:1: No such file or directory ( 0.013| 0.000) D: [pulseaudio] alsa-mixer.c: Caching failure to open output:iec958-dts-surround-51 ( 0.013| 0.000) D: [pulseaudio] alsa-mixer.c: Skipping profile output:iec958-dts-surround-51+input:analog-mono - will not be able to open output:iec958-dts-surround-51 ( 0.013| 0.000) D: [pulseaudio] alsa-mixer.c: Skipping profile output:iec958-dts-surround-51+input:analog-stereo - will not be able to open output:iec958-dts-surround-51 ( 0.013| 0.000) D: [pulseaudio] alsa-mixer.c: Skipping profile output:iec958-dts-surround-51+input:iec958-stereo - will not be able to open output:iec958-dts-surround-51 ( 0.013| 0.000) D: [pulseaudio] alsa-mixer.c: Looking at profile output:hdmi-stereo ( 0.013| 0.000) D: [pulseaudio] alsa-mixer.c: Checking for playback on Digital Stereo (HDMI) (hdmi-stereo) ( 0.013| 0.000) D: [pulseaudio] alsa-util.c: Trying hdmi:1 with SND_PCM_NO_AUTO_FORMAT ... ( 0.013| 0.000) D: [pulseaudio] alsa-util.c: Managed to open hdmi:1 ( 0.013| 0.000) D: [pulseaudio] alsa-util.c: Maximum hw buffer size is 371 ms ( 0.029| 0.016) D: [pulseaudio] alsa-util.c: Set buffer size first (to 2028 samples), period size second (to 1014 samples). ( 0.045| 0.016) D: [pulseaudio] alsa-mixer.c: Profile output:hdmi-stereo supported. ( 0.045| 0.000) I: [pulseaudio] (alsa-lib)control.c: Invalid CTL hdmi:1 ( 0.045| 0.000) I: [pulseaudio] alsa-util.c: Unable to attach to mixer hdmi:1: No such file or directory ( 0.046| 0.000) I: [pulseaudio] alsa-util.c: Successfully attached to mixer 'hw:1' ( 0.046| 0.000) D: [pulseaudio] alsa-mixer.c: Probing path 'hdmi-output-0' ( 0.046| 0.000) D: [pulseaudio] alsa-mixer.c: Probe of jack 'HDMI/DP,pcm=3 Jack' succeeded (found!) ( 0.046| 0.000) D: [pulseaudio] alsa-mixer.c: Available mixer paths (after tidying): ( 0.046| 0.000) D: [pulseaudio] alsa-mixer.c: Path Set 0xc7ab50, direction=1 ( 0.046| 0.000) D: [pulseaudio] alsa-mixer.c: Path hdmi-output-0 (HDMI / DisplayPort), direction=1, priority=59, probed=yes, supported=yes, has_mute=no, has_volume=no, has_dB=no, min_volume=0, max_volume=0, min_dB=inf, max_dB=-inf ( 0.046| 0.000) D: [pulseaudio] alsa-mixer.c: Jack HDMI/DP,pcm=3, alsa_name='HDMI/DP,pcm=3 Jack', detection possible ( 0.046| 0.000) D: [pulseaudio] alsa-mixer.c: Skipping profile output:hdmi-stereo+input:analog-mono - will not be able to open input:analog-mono ( 0.046| 0.000) D: [pulseaudio] alsa-mixer.c: Skipping profile output:hdmi-stereo+input:analog-stereo - will not be able to open input:analog-stereo ( 0.046| 0.000) D: [pulseaudio] alsa-mixer.c: Skipping profile output:hdmi-stereo+input:iec958-stereo - will not be able to open input:iec958-stereo ( 0.046| 0.000) D: [pulseaudio] alsa-mixer.c: Looking at profile output:hdmi-surround ( 0.046| 0.000) D: [pulseaudio] alsa-mixer.c: Checking for playback on Digital Surround 5.1 (HDMI) (hdmi-surround) ( 0.046| 0.000) D: [pulseaudio] alsa-util.c: Trying hdmi:1 with SND_PCM_NO_AUTO_FORMAT ... ( 0.046| 0.000) D: [pulseaudio] alsa-util.c: Managed to open hdmi:1 ( 0.046| 0.000) D: [pulseaudio] alsa-util.c: Maximum hw buffer size is 123 ms ( 0.077| 0.031) D: [pulseaudio] alsa-util.c: Set buffer size first (to 2028 samples), period size second (to 1014 samples). ( 0.093| 0.016) D: [pulseaudio] alsa-mixer.c: Profile output:hdmi-surround supported. ( 0.093| 0.000) I: [pulseaudio] (alsa-lib)control.c: Invalid CTL hdmi:1 ( 0.093| 0.000) I: [pulseaudio] alsa-util.c: Unable to attach to mixer hdmi:1: No such file or directory ( 0.093| 0.000) I: [pulseaudio] alsa-util.c: Successfully attached to mixer 'hw:1' ( 0.094| 0.000) D: [pulseaudio] alsa-mixer.c: Available mixer paths (after tidying): ( 0.094| 0.000) D: [pulseaudio] alsa-mixer.c: Path Set 0xc8cce0, direction=1 ( 0.094| 0.000) D: [pulseaudio] alsa-mixer.c: Path hdmi-output-0 (HDMI / DisplayPort), direction=1, priority=59, probed=yes, supported=yes, has_mute=no, has_volume=no, has_dB=no, min_volume=0, max_volume=0, min_dB=inf, max_dB=-inf ( 0.094| 0.000) D: [pulseaudio] alsa-mixer.c: Jack HDMI/DP,pcm=3, alsa_name='HDMI/DP,pcm=3 Jack', detection possible ( 0.094| 0.000) D: [pulseaudio] alsa-mixer.c: Skipping profile output:hdmi-surround+input:analog-mono - will not be able to open input:analog-mono ( 0.094| 0.000) D: [pulseaudio] alsa-mixer.c: Skipping profile output:hdmi-surround+input:analog-stereo - will not be able to open input:analog-stereo ( 0.094| 0.000) D: [pulseaudio] alsa-mixer.c: Skipping profile output:hdmi-surround+input:iec958-stereo - will not be able to open input:iec958-stereo ( 0.094| 0.000) D: [pulseaudio] alsa-mixer.c: Looking at profile output:hdmi-stereo-extra1 ( 0.094| 0.000) D: [pulseaudio] alsa-mixer.c: Checking for playback on Digital Stereo (HDMI) (hdmi-stereo-extra1) ( 0.094| 0.000) D: [pulseaudio] alsa-util.c: Trying hdmi:1,1 with SND_PCM_NO_AUTO_FORMAT ... ( 0.094| 0.000) D: [pulseaudio] alsa-util.c: Managed to open hdmi:1,1 ( 0.094| 0.000) D: [pulseaudio] alsa-util.c: Maximum hw buffer size is 371 ms ( 0.125| 0.031) D: [pulseaudio] alsa-util.c: Set buffer size first (to 2028 samples), period size second (to 1014 samples). ( 0.141| 0.015) D: [pulseaudio] alsa-mixer.c: Profile output:hdmi-stereo-extra1 supported. ( 0.141| 0.000) I: [pulseaudio] (alsa-lib)control.c: Invalid CTL hdmi:1,1 ( 0.141| 0.000) I: [pulseaudio] alsa-util.c: Unable to attach to mixer hdmi:1,1: No such file or directory ( 0.142| 0.000) I: [pulseaudio] alsa-util.c: Successfully attached to mixer 'hw:1' ( 0.142| 0.000) D: [pulseaudio] alsa-mixer.c: Probing path 'hdmi-output-1' ( 0.142| 0.000) D: [pulseaudio] alsa-mixer.c: Probe of jack 'HDMI/DP,pcm=7 Jack' succeeded (found!) ( 0.142| 0.000) D: [pulseaudio] alsa-mixer.c: Available mixer paths (after tidying): ( 0.142| 0.000) D: [pulseaudio] alsa-mixer.c: Path Set 0xc8ef10, direction=1 ( 0.142| 0.000) D: [pulseaudio] alsa-mixer.c: Path hdmi-output-1 (HDMI / DisplayPort 2), direction=1, priority=58, probed=yes, supported=yes, has_mute=no, has_volume=no, has_dB=no, min_volume=0, max_volume=0, min_dB=inf, max_dB=-inf ( 0.142| 0.000) D: [pulseaudio] alsa-mixer.c: Jack HDMI/DP,pcm=7, alsa_name='HDMI/DP,pcm=7 Jack', detection possible ( 0.142| 0.000) D: [pulseaudio] alsa-mixer.c: Skipping profile output:hdmi-stereo-extra1+input:analog-mono - will not be able to open input:analog-mono ( 0.142| 0.000) D: [pulseaudio] alsa-mixer.c: Skipping profile output:hdmi-stereo-extra1+input:analog-stereo - will not be able to open input:analog-stereo ( 0.142| 0.000) D: [pulseaudio] alsa-mixer.c: Skipping profile output:hdmi-stereo-extra1+input:iec958-stereo - will not be able to open input:iec958-stereo ( 0.142| 0.000) D: [pulseaudio] alsa-mixer.c: Looking at profile output:hdmi-surround-extra1 ( 0.142| 0.000) D: [pulseaudio] alsa-mixer.c: Checking for playback on Digital Surround 5.1 (HDMI) (hdmi-surround-extra1) ( 0.142| 0.000) D: [pulseaudio] alsa-util.c: Trying hdmi:1,1 with SND_PCM_NO_AUTO_FORMAT ... ( 0.142| 0.000) D: [pulseaudio] alsa-util.c: Managed to open hdmi:1,1 ( 0.142| 0.000) D: [pulseaudio] alsa-util.c: Maximum hw buffer size is 123 ms ( 0.173| 0.031) D: [pulseaudio] alsa-util.c: Set buffer size first (to 2028 samples), period size second (to 1014 samples). ( 0.189| 0.015) D: [pulseaudio] alsa-mixer.c: Profile output:hdmi-surround-extra1 supported. ( 0.189| 0.000) I: [pulseaudio] (alsa-lib)control.c: Invalid CTL hdmi:1,1 ( 0.189| 0.000) I: [pulseaudio] alsa-util.c: Unable to attach to mixer hdmi:1,1: No such file or directory ( 0.189| 0.000) I: [pulseaudio] alsa-util.c: Successfully attached to mixer 'hw:1' ( 0.190| 0.000) D: [pulseaudio] alsa-mixer.c: Available mixer paths (after tidying): ( 0.190| 0.000) D: [pulseaudio] alsa-mixer.c: Path Set 0xcb2960, direction=1 ( 0.190| 0.000) D: [pulseaudio] alsa-mixer.c: Path hdmi-output-1 (HDMI / DisplayPort 2), direction=1, priority=58, probed=yes, supported=yes, has_mute=no, has_volume=no, has_dB=no, min_volume=0, max_volume=0, min_dB=inf, max_dB=-inf ( 0.190| 0.000) D: [pulseaudio] alsa-mixer.c: Jack HDMI/DP,pcm=7, alsa_name='HDMI/DP,pcm=7 Jack', detection possible ( 0.190| 0.000) D: [pulseaudio] alsa-mixer.c: Skipping profile output:hdmi-surround-extra1+input:analog-mono - will not be able to open input:analog-mono ( 0.190| 0.000) D: [pulseaudio] alsa-mixer.c: Skipping profile output:hdmi-surround-extra1+input:analog-stereo - will not be able to open input:analog-stereo ( 0.190| 0.000) D: [pulseaudio] alsa-mixer.c: Skipping profile output:hdmi-surround-extra1+input:iec958-stereo - will not be able to open input:iec958-stereo ( 0.190| 0.000) D: [pulseaudio] alsa-mixer.c: Looking at profile output:hdmi-stereo-extra2 ( 0.190| 0.000) D: [pulseaudio] alsa-mixer.c: Checking for playback on Digital Stereo (HDMI) (hdmi-stereo-extra2) ( 0.190| 0.000) D: [pulseaudio] alsa-util.c: Trying hdmi:1,2 with SND_PCM_NO_AUTO_FORMAT ... ( 0.190| 0.000) D: [pulseaudio] alsa-util.c: Managed to open hdmi:1,2 ( 0.190| 0.000) D: [pulseaudio] alsa-util.c: Maximum hw buffer size is 371 ms ( 0.221| 0.031) D: [pulseaudio] alsa-util.c: Set buffer size first (to 2028 samples), period size second (to 1014 samples). ( 0.237| 0.015) D: [pulseaudio] alsa-mixer.c: Profile output:hdmi-stereo-extra2 supported. ( 0.237| 0.000) I: [pulseaudio] (alsa-lib)control.c: Invalid CTL hdmi:1,2 ( 0.237| 0.000) I: [pulseaudio] alsa-util.c: Unable to attach to mixer hdmi:1,2: No such file or directory ( 0.238| 0.000) I: [pulseaudio] alsa-util.c: Successfully attached to mixer 'hw:1' ( 0.238| 0.000) D: [pulseaudio] alsa-mixer.c: Probing path 'hdmi-output-2' ( 0.238| 0.000) D: [pulseaudio] alsa-mixer.c: Probe of jack 'HDMI/DP,pcm=8 Jack' succeeded (found!) ( 0.238| 0.000) D: [pulseaudio] alsa-mixer.c: Available mixer paths (after tidying): ( 0.238| 0.000) D: [pulseaudio] alsa-mixer.c: Path Set 0xcb2d80, direction=1 ( 0.238| 0.000) D: [pulseaudio] alsa-mixer.c: Path hdmi-output-2 (HDMI / DisplayPort 3), direction=1, priority=57, probed=yes, supported=yes, has_mute=no, has_volume=no, has_dB=no, min_volume=0, max_volume=0, min_dB=inf, max_dB=-inf ( 0.238| 0.000) D: [pulseaudio] alsa-mixer.c: Jack HDMI/DP,pcm=8, alsa_name='HDMI/DP,pcm=8 Jack', detection possible ( 0.238| 0.000) D: [pulseaudio] alsa-mixer.c: Skipping profile output:hdmi-stereo-extra2+input:analog-mono - will not be able to open input:analog-mono ( 0.238| 0.000) D: [pulseaudio] alsa-mixer.c: Skipping profile output:hdmi-stereo-extra2+input:analog-stereo - will not be able to open input:analog-stereo ( 0.238| 0.000) D: [pulseaudio] alsa-mixer.c: Skipping profile output:hdmi-stereo-extra2+input:iec958-stereo - will not be able to open input:iec958-stereo ( 0.238| 0.000) D: [pulseaudio] alsa-mixer.c: Looking at profile output:hdmi-surround-extra2 ( 0.238| 0.000) D: [pulseaudio] alsa-mixer.c: Checking for playback on Digital Surround 5.1 (HDMI) (hdmi-surround-extra2) ( 0.238| 0.000) D: [pulseaudio] alsa-util.c: Trying hdmi:1,2 with SND_PCM_NO_AUTO_FORMAT ... ( 0.238| 0.000) D: [pulseaudio] alsa-util.c: Managed to open hdmi:1,2 ( 0.238| 0.000) D: [pulseaudio] alsa-util.c: Maximum hw buffer size is 123 ms ( 0.269| 0.031) D: [pulseaudio] alsa-util.c: Set buffer size first (to 2028 samples), period size second (to 1014 samples). ( 0.285| 0.016) D: [pulseaudio] alsa-mixer.c: Profile output:hdmi-surround-extra2 supported. ( 0.285| 0.000) I: [pulseaudio] (alsa-lib)control.c: Invalid CTL hdmi:1,2 ( 0.285| 0.000) I: [pulseaudio] alsa-util.c: Unable to attach to mixer hdmi:1,2: No such file or directory ( 0.285| 0.000) I: [pulseaudio] alsa-util.c: Successfully attached to mixer 'hw:1' ( 0.285| 0.000) D: [pulseaudio] alsa-mixer.c: Available mixer paths (after tidying): ( 0.286| 0.000) D: [pulseaudio] alsa-mixer.c: Path Set 0xcb4f20, direction=1 ( 0.286| 0.000) D: [pulseaudio] alsa-mixer.c: Path hdmi-output-2 (HDMI / DisplayPort 3), direction=1, priority=57, probed=yes, supported=yes, has_mute=no, has_volume=no, has_dB=no, min_volume=0, max_volume=0, min_dB=inf, max_dB=-inf ( 0.286| 0.000) D: [pulseaudio] alsa-mixer.c: Jack HDMI/DP,pcm=8, alsa_name='HDMI/DP,pcm=8 Jack', detection possible ( 0.286| 0.000) D: [pulseaudio] alsa-mixer.c: Skipping profile output:hdmi-surround-extra2+input:analog-mono - will not be able to open input:analog-mono ( 0.286| 0.000) D: [pulseaudio] alsa-mixer.c: Skipping profile output:hdmi-surround-extra2+input:analog-stereo - will not be able to open input:analog-stereo ( 0.286| 0.000) D: [pulseaudio] alsa-mixer.c: Skipping profile output:hdmi-surround-extra2+input:iec958-stereo - will not be able to open input:iec958-stereo ( 0.286| 0.000) D: [pulseaudio] alsa-mixer.c: Looking at profile output:hdmi-stereo-extra3 ( 0.286| 0.000) D: [pulseaudio] alsa-mixer.c: Checking for playback on Digital Stereo (HDMI) (hdmi-stereo-extra3) ( 0.286| 0.000) D: [pulseaudio] alsa-util.c: Trying hdmi:1,3 with SND_PCM_NO_AUTO_FORMAT ... ( 0.286| 0.000) D: [pulseaudio] alsa-util.c: Managed to open hdmi:1,3 ( 0.286| 0.000) D: [pulseaudio] alsa-util.c: Maximum hw buffer size is 371 ms ( 0.317| 0.031) D: [pulseaudio] alsa-util.c: Set buffer size first (to 2028 samples), period size second (to 1014 samples). ( 0.333| 0.016) D: [pulseaudio] alsa-mixer.c: Profile output:hdmi-stereo-extra3 supported. ( 0.333| 0.000) I: [pulseaudio] (alsa-lib)control.c: Invalid CTL hdmi:1,3 ( 0.333| 0.000) I: [pulseaudio] alsa-util.c: Unable to attach to mixer hdmi:1,3: No such file or directory ( 0.334| 0.000) I: [pulseaudio] alsa-util.c: Successfully attached to mixer 'hw:1' ( 0.334| 0.000) D: [pulseaudio] alsa-mixer.c: Probing path 'hdmi-output-3' ( 0.334| 0.000) D: [pulseaudio] alsa-mixer.c: Probe of jack 'HDMI/DP,pcm=9 Jack' succeeded (found!) ( 0.334| 0.000) D: [pulseaudio] alsa-mixer.c: Available mixer paths (after tidying): ( 0.334| 0.000) D: [pulseaudio] alsa-mixer.c: Path Set 0xc190e0, direction=1 ( 0.334| 0.000) D: [pulseaudio] alsa-mixer.c: Path hdmi-output-3 (HDMI / DisplayPort 4), direction=1, priority=56, probed=yes, supported=yes, has_mute=no, has_volume=no, has_dB=no, min_volume=0, max_volume=0, min_dB=inf, max_dB=-inf ( 0.334| 0.000) D: [pulseaudio] alsa-mixer.c: Jack HDMI/DP,pcm=9, alsa_name='HDMI/DP,pcm=9 Jack', detection possible ( 0.334| 0.000) D: [pulseaudio] alsa-mixer.c: Skipping profile output:hdmi-stereo-extra3+input:analog-mono - will not be able to open input:analog-mono ( 0.334| 0.000) D: [pulseaudio] alsa-mixer.c: Skipping profile output:hdmi-stereo-extra3+input:analog-stereo - will not be able to open input:analog-stereo ( 0.334| 0.000) D: [pulseaudio] alsa-mixer.c: Skipping profile output:hdmi-stereo-extra3+input:iec958-stereo - will not be able to open input:iec958-stereo ( 0.334| 0.000) D: [pulseaudio] alsa-mixer.c: Looking at profile output:hdmi-surround-extra3 ( 0.334| 0.000) D: [pulseaudio] alsa-mixer.c: Checking for playback on Digital Surround 5.1 (HDMI) (hdmi-surround-extra3) ( 0.334| 0.000) D: [pulseaudio] alsa-util.c: Trying hdmi:1,3 with SND_PCM_NO_AUTO_FORMAT ... ( 0.334| 0.000) D: [pulseaudio] alsa-util.c: Managed to open hdmi:1,3 ( 0.334| 0.000) D: [pulseaudio] alsa-util.c: Maximum hw buffer size is 123 ms ( 0.365| 0.031) D: [pulseaudio] alsa-util.c: Set buffer size first (to 2028 samples), period size second (to 1014 samples). ( 0.381| 0.016) D: [pulseaudio] alsa-mixer.c: Profile output:hdmi-surround-extra3 supported. ( 0.381| 0.000) I: [pulseaudio] (alsa-lib)control.c: Invalid CTL hdmi:1,3 ( 0.381| 0.000) I: [pulseaudio] alsa-util.c: Unable to attach to mixer hdmi:1,3: No such file or directory ( 0.381| 0.000) I: [pulseaudio] alsa-util.c: Successfully attached to mixer 'hw:1' ( 0.382| 0.000) D: [pulseaudio] alsa-mixer.c: Available mixer paths (after tidying): ( 0.382| 0.000) D: [pulseaudio] alsa-mixer.c: Path Set 0xcb53c0, direction=1 ( 0.382| 0.000) D: [pulseaudio] alsa-mixer.c: Path hdmi-output-3 (HDMI / DisplayPort 4), direction=1, priority=56, probed=yes, supported=yes, has_mute=no, has_volume=no, has_dB=no, min_volume=0, max_volume=0, min_dB=inf, max_dB=-inf ( 0.382| 0.000) D: [pulseaudio] alsa-mixer.c: Jack HDMI/DP,pcm=9, alsa_name='HDMI/DP,pcm=9 Jack', detection possible ( 0.382| 0.000) D: [pulseaudio] alsa-mixer.c: Skipping profile output:hdmi-surround-extra3+input:analog-mono - will not be able to open input:analog-mono ( 0.382| 0.000) D: [pulseaudio] alsa-mixer.c: Skipping profile output:hdmi-surround-extra3+input:analog-stereo - will not be able to open input:analog-stereo ( 0.382| 0.000) D: [pulseaudio] alsa-mixer.c: Skipping profile output:hdmi-surround-extra3+input:iec958-stereo - will not be able to open input:iec958-stereo ( 0.382| 0.000) D: [pulseaudio] alsa-mixer.c: Profile set 0xc26dc0, auto_profiles=yes, probed=yes, n_mappings=8, n_profiles=8, n_decibel_fixes=0 ( 0.382| 0.000) D: [pulseaudio] alsa-mixer.c: Mapping hdmi-stereo (Digital Stereo (HDMI)), priority=54, channel_map=front-left,front-right, supported=yes, direction=1 ( 0.382| 0.000) D: [pulseaudio] alsa-mixer.c: Mapping hdmi-surround (Digital Surround 5.1 (HDMI)), priority=3, channel_map=front-left,front-right,rear-left,rear-right,front-center,lfe, supported=yes, direction=1 ( 0.382| 0.000) D: [pulseaudio] alsa-mixer.c: Mapping hdmi-stereo-extra1 (Digital Stereo (HDMI)), priority=52, channel_map=front-left,front-right, supported=yes, direction=1 ( 0.382| 0.000) D: [pulseaudio] alsa-mixer.c: Mapping hdmi-surround-extra1 (Digital Surround 5.1 (HDMI)), priority=1, channel_map=front-left,front-right,rear-left,rear-right,front-center,lfe, supported=yes, direction=1 ( 0.382| 0.000) D: [pulseaudio] alsa-mixer.c: Mapping hdmi-stereo-extra2 (Digital Stereo (HDMI)), priority=52, channel_map=front-left,front-right, supported=yes, direction=1 ( 0.382| 0.000) D: [pulseaudio] alsa-mixer.c: Mapping hdmi-surround-extra2 (Digital Surround 5.1 (HDMI)), priority=1, channel_map=front-left,front-right,rear-left,rear-right,front-center,lfe, supported=yes, direction=1 ( 0.382| 0.000) D: [pulseaudio] alsa-mixer.c: Mapping hdmi-stereo-extra3 (Digital Stereo (HDMI)), priority=52, channel_map=front-left,front-right, supported=yes, direction=1 ( 0.382| 0.000) D: [pulseaudio] alsa-mixer.c: Mapping hdmi-surround-extra3 (Digital Surround 5.1 (HDMI)), priority=1, channel_map=front-left,front-right,rear-left,rear-right,front-center,lfe, supported=yes, direction=1 ( 0.382| 0.000) D: [pulseaudio] alsa-mixer.c: Profile output:hdmi-stereo (Digital Stereo (HDMI) Output), priority=5400, supported=yes n_input_mappings=0, n_output_mappings=1 ( 0.382| 0.000) D: [pulseaudio] alsa-mixer.c: Output hdmi-stereo ( 0.382| 0.000) D: [pulseaudio] alsa-mixer.c: Profile output:hdmi-surround (Digital Surround 5.1 (HDMI) Output), priority=300, supported=yes n_input_mappings=0, n_output_mappings=1 ( 0.382| 0.000) D: [pulseaudio] alsa-mixer.c: Output hdmi-surround ( 0.382| 0.000) D: [pulseaudio] alsa-mixer.c: Profile output:hdmi-stereo-extra1 (Digital Stereo (HDMI) Output), priority=5200, supported=yes n_input_mappings=0, n_output_mappings=1 ( 0.382| 0.000) D: [pulseaudio] alsa-mixer.c: Output hdmi-stereo-extra1 ( 0.382| 0.000) D: [pulseaudio] alsa-mixer.c: Profile output:hdmi-surround-extra1 (Digital Surround 5.1 (HDMI) Output), priority=100, supported=yes n_input_mappings=0, n_output_mappings=1 ( 0.382| 0.000) D: [pulseaudio] alsa-mixer.c: Output hdmi-surround-extra1 ( 0.382| 0.000) D: [pulseaudio] alsa-mixer.c: Profile output:hdmi-stereo-extra2 (Digital Stereo (HDMI) Output), priority=5200, supported=yes n_input_mappings=0, n_output_mappings=1 ( 0.382| 0.000) D: [pulseaudio] alsa-mixer.c: Output hdmi-stereo-extra2 ( 0.382| 0.000) D: [pulseaudio] alsa-mixer.c: Profile output:hdmi-surround-extra2 (Digital Surround 5.1 (HDMI) Output), priority=100, supported=yes n_input_mappings=0, n_output_mappings=1 ( 0.382| 0.000) D: [pulseaudio] alsa-mixer.c: Output hdmi-surround-extra2 ( 0.382| 0.000) D: [pulseaudio] alsa-mixer.c: Profile output:hdmi-stereo-extra3 (Digital Stereo (HDMI) Output), priority=5200, supported=yes n_input_mappings=0, n_output_mappings=1 ( 0.382| 0.000) D: [pulseaudio] alsa-mixer.c: Output hdmi-stereo-extra3 ( 0.382| 0.000) D: [pulseaudio] alsa-mixer.c: Profile output:hdmi-surround-extra3 (Digital Surround 5.1 (HDMI) Output), priority=100, supported=yes n_input_mappings=0, n_output_mappings=1 ( 0.382| 0.000) D: [pulseaudio] alsa-mixer.c: Output hdmi-surround-extra3 ( 0.382| 0.000) I: [pulseaudio] module-card-restore.c: Restoring profile for card alsa_card.pci-0000_01_00.1. ( 0.382| 0.000) I: [pulseaudio] module-card-restore.c: Restoring port latency offsets for card alsa_card.pci-0000_01_00.1. ( 0.382| 0.000) I: [pulseaudio] card.c: Created 0 "alsa_card.pci-0000_01_00.1" ( 0.382| 0.000) D: [pulseaudio] reserve-wrap.c: Successfully create reservation lock monitor for device 'Audio1' ( 0.383| 0.000) D: [pulseaudio] alsa-util.c: Trying hdmi:1,1 with SND_PCM_NO_AUTO_FORMAT ... ( 0.383| 0.000) D: [pulseaudio] alsa-util.c: Managed to open hdmi:1,1 ( 0.383| 0.000) D: [pulseaudio] alsa-util.c: Maximum hw buffer size is 371 ms ( 0.413| 0.030) D: [pulseaudio] alsa-util.c: Set buffer size first (to 2028 samples), period size second (to 1014 samples). ( 0.429| 0.015) I: [pulseaudio] alsa-sink.c: Successfully opened device hdmi:1,1. ( 0.429| 0.000) I: [pulseaudio] alsa-sink.c: Selected mapping 'Digital Stereo (HDMI)' (hdmi-stereo-extra1). ( 0.429| 0.000) I: [pulseaudio] alsa-sink.c: Successfully enabled mmap() mode. ( 0.429| 0.000) I: [pulseaudio] (alsa-lib)control.c: Invalid CTL hdmi:1,1 ( 0.429| 0.000) I: [pulseaudio] alsa-util.c: Unable to attach to mixer hdmi:1,1: No such file or directory ( 0.430| 0.000) I: [pulseaudio] alsa-util.c: Successfully attached to mixer 'hw:1' ( 0.430| 0.000) D: [pulseaudio] alsa-mixer.c: Added 1 ports ( 0.430| 0.000) D: [pulseaudio] module-device-restore.c: Database contains invalid data for key: sink:alsa_output.pci-0000_01_00.1.hdmi-stereo-extra1:null ( 0.430| 0.000) I: [pulseaudio] sink.c: Created sink 0 "alsa_output.pci-0000_01_00.1.hdmi-stereo-extra1" with sample spec s16le 2ch 44100Hz and channel map front-left,front-right ( 0.430| 0.000) I: [pulseaudio] sink.c: alsa.resolution_bits = "16" ( 0.430| 0.000) I: [pulseaudio] sink.c: device.api = "alsa" ( 0.430| 0.000) I: [pulseaudio] sink.c: device.class = "sound" ( 0.430| 0.000) I: [pulseaudio] sink.c: alsa.class = "generic" ( 0.430| 0.000) I: [pulseaudio] sink.c: alsa.subclass = "generic-mix" ( 0.430| 0.000) I: [pulseaudio] sink.c: alsa.name = "HDMI 1" ( 0.430| 0.000) I: [pulseaudio] sink.c: alsa.id = "HDMI 1" ( 0.430| 0.000) I: [pulseaudio] sink.c: alsa.subdevice = "0" ( 0.430| 0.000) I: [pulseaudio] sink.c: alsa.subdevice_name = "subdevice #0" ( 0.430| 0.000) I: [pulseaudio] sink.c: alsa.device = "7" ( 0.430| 0.000) I: [pulseaudio] sink.c: alsa.card = "1" ( 0.430| 0.000) I: [pulseaudio] sink.c: alsa.card_name = "HDA NVidia" ( 0.430| 0.000) I: [pulseaudio] sink.c: alsa.long_card_name = "HDA NVidia at 0xf7080000 irq 17" ( 0.430| 0.000) I: [pulseaudio] sink.c: alsa.driver_name = "snd_hda_intel" ( 0.430| 0.000) I: [pulseaudio] sink.c: device.bus_path = "pci-0000:01:00.1" ( 0.430| 0.000) I: [pulseaudio] sink.c: sysfs.path = "/devices/pci0000:00/0000:00:01.0/0000:01:00.1/sound/card1" ( 0.430| 0.000) I: [pulseaudio] sink.c: device.bus = "pci" ( 0.430| 0.000) I: [pulseaudio] sink.c: device.vendor.id = "10de" ( 0.430| 0.000) I: [pulseaudio] sink.c: device.vendor.name = "NVIDIA Corporation" ( 0.430| 0.000) I: [pulseaudio] sink.c: device.string = "hdmi:1,1" ( 0.430| 0.000) I: [pulseaudio] sink.c: device.buffering.buffer_size = "8064" ( 0.430| 0.000) I: [pulseaudio] sink.c: device.buffering.fragment_size = "2688" ( 0.430| 0.000) I: [pulseaudio] sink.c: device.access_mode = "mmap" ( 0.430| 0.000) I: [pulseaudio] sink.c: device.profile.name = "hdmi-stereo-extra1" ( 0.430| 0.000) I: [pulseaudio] sink.c: device.profile.description = "Digital Stereo (HDMI)" ( 0.430| 0.000) I: [pulseaudio] sink.c: device.description = "HDA NVidia Digital Stereo (HDMI)" ( 0.430| 0.000) I: [pulseaudio] sink.c: alsa.mixer_name = "Nvidia GPU 41 HDMI/DP" ( 0.430| 0.000) I: [pulseaudio] sink.c: alsa.components = "HDA:10de0041,10438422,00100100" ( 0.430| 0.000) I: [pulseaudio] sink.c: module-udev-detect.discovered = "1" ( 0.430| 0.000) I: [pulseaudio] sink.c: device.icon_name = "audio-card-pci" ( 0.430| 0.000) D: [pulseaudio] core-subscribe.c: Dropped redundant event due to change event. ( 0.430| 0.000) I: [pulseaudio] source.c: Created source 0 "alsa_output.pci-0000_01_00.1.hdmi-stereo-extra1.monitor" with sample spec s16le 2ch 44100Hz and channel map front-left,front-right ( 0.430| 0.000) I: [pulseaudio] source.c: device.description = "Monitor of HDA NVidia Digital Stereo (HDMI)" ( 0.430| 0.000) I: [pulseaudio] source.c: device.class = "monitor" ( 0.430| 0.000) I: [pulseaudio] source.c: alsa.card = "1" ( 0.430| 0.000) I: [pulseaudio] source.c: alsa.card_name = "HDA NVidia" ( 0.430| 0.000) I: [pulseaudio] source.c: alsa.long_card_name = "HDA NVidia at 0xf7080000 irq 17" ( 0.430| 0.000) I: [pulseaudio] source.c: alsa.driver_name = "snd_hda_intel" ( 0.430| 0.000) I: [pulseaudio] source.c: device.bus_path = "pci-0000:01:00.1" ( 0.430| 0.000) I: [pulseaudio] source.c: sysfs.path = "/devices/pci0000:00/0000:00:01.0/0000:01:00.1/sound/card1" ( 0.430| 0.000) I: [pulseaudio] source.c: device.bus = "pci" ( 0.430| 0.000) I: [pulseaudio] source.c: device.vendor.id = "10de" ( 0.430| 0.000) I: [pulseaudio] source.c: device.vendor.name = "NVIDIA Corporation" ( 0.430| 0.000) I: [pulseaudio] source.c: device.string = "1" ( 0.430| 0.000) I: [pulseaudio] source.c: module-udev-detect.discovered = "1" ( 0.430| 0.000) I: [pulseaudio] source.c: device.icon_name = "audio-card-pci" ( 0.430| 0.000) I: [pulseaudio] alsa-sink.c: Using 3.0 fragments of size 2688 bytes (15.24ms), buffer size is 8064 bytes (45.71ms) ( 0.430| 0.000) D: [pulseaudio] alsa-sink.c: hwbuf_unused=0 ( 0.430| 0.000) D: [pulseaudio] alsa-sink.c: setting avail_min=1 ( 0.430| 0.000) D: [pulseaudio] alsa-mixer.c: Activating path hdmi-output-1 ( 0.430| 0.000) D: [pulseaudio] alsa-mixer.c: Path hdmi-output-1 (HDMI / DisplayPort 2), direction=1, priority=58, probed=yes, supported=yes, has_mute=no, has_volume=no, has_dB=no, min_volume=0, max_volume=0, min_dB=inf, max_dB=-inf ( 0.430| 0.000) D: [pulseaudio] alsa-mixer.c: Jack HDMI/DP,pcm=7, alsa_name='HDMI/DP,pcm=7 Jack', detection possible ( 0.430| 0.000) I: [pulseaudio] alsa-sink.c: Driver does not support hardware volume control, falling back to software volume control. ( 0.430| 0.000) I: [pulseaudio] alsa-sink.c: Driver does not support hardware mute control, falling back to software mute control. ( 0.430| 0.000) D: [pulseaudio] alsa-util.c: snd_pcm_dump(): ( 0.430| 0.000) D: [pulseaudio] alsa-util.c: Hooks PCM ( 0.430| 0.000) D: [pulseaudio] alsa-util.c: Its setup is: ( 0.430| 0.000) D: [pulseaudio] alsa-util.c: stream : PLAYBACK ( 0.430| 0.000) D: [pulseaudio] alsa-util.c: access : MMAP_INTERLEAVED ( 0.430| 0.000) D: [pulseaudio] alsa-util.c: format : S16_LE ( 0.430| 0.000) D: [pulseaudio] alsa-util.c: subformat : STD ( 0.430| 0.000) D: [pulseaudio] alsa-util.c: channels : 2 ( 0.430| 0.000) D: [pulseaudio] alsa-util.c: rate : 44100 ( 0.430| 0.000) D: [pulseaudio] alsa-util.c: exact rate : 44100 (44100/1) ( 0.430| 0.000) D: [pulseaudio] alsa-util.c: msbits : 16 ( 0.430| 0.000) D: [pulseaudio] alsa-util.c: buffer_size : 2016 ( 0.430| 0.000) D: [pulseaudio] alsa-util.c: period_size : 672 ( 0.430| 0.000) D: [pulseaudio] alsa-util.c: period_time : 15238 ( 0.430| 0.000) D: [pulseaudio] alsa-util.c: tstamp_mode : ENABLE ( 0.430| 0.000) D: [pulseaudio] alsa-util.c: period_step : 1 ( 0.430| 0.000) D: [pulseaudio] alsa-util.c: avail_min : 672 ( 0.430| 0.000) D: [pulseaudio] alsa-util.c: period_event : 1 ( 0.430| 0.000) D: [pulseaudio] alsa-util.c: start_threshold : -1 ( 0.430| 0.000) D: [pulseaudio] alsa-util.c: stop_threshold : 9079256848778919936 ( 0.430| 0.000) D: [pulseaudio] alsa-util.c: silence_threshold: 0 ( 0.430| 0.000) D: [pulseaudio] alsa-util.c: silence_size : 0 ( 0.430| 0.000) D: [pulseaudio] alsa-util.c: boundary : 9079256848778919936 ( 0.430| 0.000) D: [pulseaudio] alsa-util.c: Slave: Hardware PCM card 1 'HDA NVidia' device 7 subdevice 0 ( 0.430| 0.000) D: [pulseaudio] alsa-util.c: Its setup is: ( 0.430| 0.000) D: [pulseaudio] alsa-util.c: stream : PLAYBACK ( 0.430| 0.000) D: [pulseaudio] alsa-util.c: access : MMAP_INTERLEAVED ( 0.430| 0.000) D: [pulseaudio] alsa-util.c: format : S16_LE ( 0.430| 0.000) D: [pulseaudio] alsa-util.c: subformat : STD ( 0.430| 0.000) D: [pulseaudio] alsa-util.c: channels : 2 ( 0.430| 0.000) D: [pulseaudio] alsa-util.c: rate : 44100 ( 0.430| 0.000) D: [pulseaudio] alsa-util.c: exact rate : 44100 (44100/1) ( 0.430| 0.000) D: [pulseaudio] alsa-util.c: msbits : 16 ( 0.430| 0.000) D: [pulseaudio] alsa-util.c: buffer_size : 2016 ( 0.430| 0.000) D: [pulseaudio] alsa-util.c: period_size : 672 ( 0.430| 0.000) D: [pulseaudio] alsa-util.c: period_time : 15238 ( 0.430| 0.000) D: [pulseaudio] alsa-util.c: tstamp_mode : ENABLE ( 0.430| 0.000) D: [pulseaudio] alsa-util.c: period_step : 1 ( 0.430| 0.000) D: [pulseaudio] alsa-util.c: avail_min : 672 ( 0.430| 0.000) D: [pulseaudio] alsa-util.c: period_event : 1 ( 0.430| 0.000) D: [pulseaudio] alsa-util.c: start_threshold : -1 ( 0.430| 0.000) D: [pulseaudio] alsa-util.c: stop_threshold : 9079256848778919936 ( 0.430| 0.000) D: [pulseaudio] alsa-util.c: silence_threshold: 0 ( 0.430| 0.000) D: [pulseaudio] alsa-util.c: silence_size : 0 ( 0.430| 0.000) D: [pulseaudio] alsa-util.c: boundary : 9079256848778919936 ( 0.430| 0.000) D: [pulseaudio] alsa-util.c: appl_ptr : 0 ( 0.430| 0.000) D: [pulseaudio] alsa-util.c: hw_ptr : 0 ( 0.431| 0.000) D: [alsa-sink] alsa-sink.c: Thread starting up ( 0.434| 0.003) D: [alsa-sink] core-util.c: RealtimeKit worked. ( 0.434| 0.000) I: [alsa-sink] core-util.c: Successfully enabled SCHED_RR scheduling for thread, with priority 5. ( 0.434| 0.000) I: [alsa-sink] alsa-sink.c: Starting playback. ( 0.434| 0.000) D: [pulseaudio] module-alsa-card.c: Found 4 jacks. ( 0.434| 0.000) I: [pulseaudio] alsa-util.c: Successfully attached to mixer 'hw:1' ( 0.434| 0.000) D: [pulseaudio] module-alsa-card.c: Jack 'HDMI/DP,pcm=3 Jack' is now unplugged ( 0.434| 0.000) D: [pulseaudio] device-port.c: Setting port hdmi-output-0 to status no ( 0.434| 0.000) D: [pulseaudio] core-subscribe.c: Dropped redundant event due to change event. ( 0.434| 0.000) D: [pulseaudio] module-alsa-card.c: Jack 'HDMI/DP,pcm=7 Jack' is now unplugged ( 0.434| 0.000) D: [pulseaudio] device-port.c: Setting port hdmi-output-1 to status no ( 0.434| 0.000) D: [pulseaudio] core-subscribe.c: Dropped redundant event due to change event. ( 0.434| 0.000) D: [pulseaudio] module-alsa-card.c: Jack 'HDMI/DP,pcm=8 Jack' is now unplugged ( 0.434| 0.000) D: [pulseaudio] device-port.c: Setting port hdmi-output-2 to status no ( 0.434| 0.000) D: [pulseaudio] core-subscribe.c: Dropped redundant event due to change event. ( 0.434| 0.000) D: [pulseaudio] module-alsa-card.c: Jack 'HDMI/DP,pcm=9 Jack' is now unplugged ( 0.434| 0.000) D: [pulseaudio] device-port.c: Setting port hdmi-output-3 to status no ( 0.434| 0.000) D: [pulseaudio] core-subscribe.c: Dropped redundant event due to change event. ( 0.434| 0.000) I: [pulseaudio] module.c: Loaded "module-alsa-card" (index: #4; argument: "device_id="1" name="pci-0000_01_00.1" card_name="alsa_card.pci-0000_01_00.1" namereg_fail=false tsched=no fixed_latency_range=no ignore_dB=no deferred_volume=yes use_ucm=yes card_properties="module-udev-detect.discovered=1""). ( 0.435| 0.000) I: [pulseaudio] module-udev-detect.c: Card /devices/pci0000:00/0000:00:01.0/0000:01:00.1/sound/card1 (alsa_card.pci-0000_01_00.1) module loaded. ( 0.435| 0.000) D: [pulseaudio] module-udev-detect.c: /dev/snd/controlC0 is accessible: yes ( 0.435| 0.000) D: [pulseaudio] module-udev-detect.c: /devices/pci0000:00/0000:00:1b.0/sound/card0 is busy: no ( 0.435| 0.000) D: [pulseaudio] module-udev-detect.c: Loading module-alsa-card with arguments 'device_id="0" name="pci-0000_00_1b.0" card_name="alsa_card.pci-0000_00_1b.0" namereg_fail=false tsched=no fixed_latency_range=no ignore_dB=no deferred_volume=yes use_ucm=yes card_properties="module-udev-detect.discovered=1"' ( 0.435| 0.000) D: [pulseaudio] reserve-wrap.c: Successfully acquired reservation lock on device 'Audio0' ( 0.435| 0.000) I: [pulseaudio] (alsa-lib)utils.c: could not open configuration file /usr/share/alsa/ucm/HDA Intel PCH/HDA Intel PCH.conf ( 0.435| 0.000) I: [pulseaudio] (alsa-lib)parser.c: error: could not parse configuration for card HDA Intel PCH ( 0.435| 0.000) I: [pulseaudio] (alsa-lib)main.c: error: failed to import HDA Intel PCH use case configuration -2 ( 0.435| 0.000) I: [pulseaudio] alsa-ucm.c: UCM not available for card HDA Intel PCH ( 0.436| 0.000) D: [pulseaudio] alsa-mixer.c: Looking at profile input:analog-mono ( 0.436| 0.000) D: [pulseaudio] alsa-mixer.c: Checking for recording on Analog Mono (analog-mono) ( 0.436| 0.000) D: [pulseaudio] alsa-util.c: Trying hw:0 with SND_PCM_NO_AUTO_FORMAT ... ( 0.436| 0.000) D: [pulseaudio] alsa-util.c: Managed to open hw:0 ( 0.436| 0.000) D: [pulseaudio] alsa-util.c: snd_pcm_hw_params_set_channels(1) failed: Invalid argument ( 0.436| 0.000) D: [pulseaudio] alsa-util.c: Trying hw:0 without SND_PCM_NO_AUTO_FORMAT ... ( 0.436| 0.000) D: [pulseaudio] alsa-util.c: Managed to open hw:0 ( 0.436| 0.000) D: [pulseaudio] alsa-util.c: snd_pcm_hw_params_set_channels(1) failed: Invalid argument ( 0.436| 0.000) D: [pulseaudio] alsa-util.c: Trying plug:hw:0 with SND_PCM_NO_AUTO_FORMAT ... ( 0.436| 0.000) D: [pulseaudio] alsa-util.c: Managed to open plug:hw:0 ( 0.436| 0.000) D: [pulseaudio] alsa-util.c: snd_pcm_hw_params_set_channels(1) failed: Invalid argument ( 0.436| 0.000) D: [pulseaudio] alsa-util.c: Trying plug:hw:0 without SND_PCM_NO_AUTO_FORMAT ... ( 0.436| 0.000) D: [pulseaudio] alsa-util.c: Managed to open plug:hw:0 ( 0.436| 0.000) D: [pulseaudio] alsa-util.c: snd_pcm_hw_params_set_channels(1) failed: Invalid argument ( 0.436| 0.000) I: [pulseaudio] alsa-util.c: Failed to set hardware parameters on plug:hw:0: Invalid argument ( 0.436| 0.000) D: [pulseaudio] alsa-mixer.c: Caching failure to open input:analog-mono ( 0.436| 0.000) D: [pulseaudio] alsa-mixer.c: Looking at profile input:analog-stereo ( 0.436| 0.000) D: [pulseaudio] alsa-mixer.c: Checking for recording on Analog Stereo (analog-stereo) ( 0.436| 0.000) D: [pulseaudio] alsa-util.c: Trying front:0 with SND_PCM_NO_AUTO_FORMAT ... ( 0.437| 0.000) D: [pulseaudio] alsa-util.c: Managed to open front:0 ( 0.437| 0.000) D: [pulseaudio] alsa-util.c: Maximum hw buffer size is 371 ms ( 0.437| 0.000) D: [pulseaudio] alsa-util.c: Set buffer size first (to 2028 samples), period size second (to 1014 samples). ( 0.437| 0.000) D: [pulseaudio] alsa-mixer.c: Profile input:analog-stereo supported. ( 0.438| 0.001) I: [pulseaudio] (alsa-lib)control.c: Invalid CTL front:0 ( 0.438| 0.000) I: [pulseaudio] alsa-util.c: Unable to attach to mixer front:0: No such file or directory ( 0.438| 0.000) I: [pulseaudio] alsa-util.c: Successfully attached to mixer 'hw:0' ( 0.438| 0.000) D: [pulseaudio] alsa-mixer.c: Probing path 'analog-input-microphone-front' ( 0.439| 0.000) D: [pulseaudio] alsa-mixer.c: Probe of jack 'Front Mic Jack' succeeded (found!) ( 0.439| 0.000) D: [pulseaudio] alsa-mixer.c: Probe of element 'Front Mic Boost' succeeded (volume=1, switch=0, enumeration=0). ( 0.439| 0.000) D: [pulseaudio] alsa-mixer.c: Probe of element 'Front Mic' succeeded (volume=0, switch=0, enumeration=0). ( 0.439| 0.000) D: [pulseaudio] alsa-mixer.c: Probe of element 'Capture' succeeded (volume=1, switch=1, enumeration=0). ( 0.439| 0.000) D: [pulseaudio] alsa-mixer.c: Probe of element 'Input Source' succeeded (volume=0, switch=0, enumeration=1). ( 0.439| 0.000) D: [pulseaudio] alsa-mixer.c: Probe of element 'Capture Source' succeeded (volume=0, switch=0, enumeration=0). ( 0.439| 0.000) D: [pulseaudio] alsa-mixer.c: Probe of element 'Mic' succeeded (volume=0, switch=0, enumeration=0). ( 0.439| 0.000) D: [pulseaudio] alsa-mixer.c: Probe of element 'Internal Mic' succeeded (volume=0, switch=0, enumeration=0). ( 0.439| 0.000) D: [pulseaudio] alsa-mixer.c: Probe of element 'Rear Mic' succeeded (volume=0, switch=0, enumeration=0). ( 0.439| 0.000) D: [pulseaudio] alsa-mixer.c: Probe of element 'Dock Mic' succeeded (volume=0, switch=0, enumeration=0). ( 0.439| 0.000) D: [pulseaudio] alsa-mixer.c: Probe of element 'Mic Boost' succeeded (volume=0, switch=0, enumeration=0). ( 0.439| 0.000) D: [pulseaudio] alsa-mixer.c: Probe of element 'Dock Mic Boost' succeeded (volume=0, switch=0, enumeration=0). ( 0.439| 0.000) D: [pulseaudio] alsa-mixer.c: Probe of element 'Internal Mic Boost' succeeded (volume=0, switch=0, enumeration=0). ( 0.439| 0.000) D: [pulseaudio] alsa-mixer.c: Probe of element 'Rear Mic Boost' succeeded (volume=2, switch=0, enumeration=0). ( 0.439| 0.000) D: [pulseaudio] alsa-mixer.c: Probe of element 'Line' succeeded (volume=0, switch=0, enumeration=0). ( 0.439| 0.000) D: [pulseaudio] alsa-mixer.c: Probe of element 'Line Boost' succeeded (volume=2, switch=0, enumeration=0). ( 0.439| 0.000) D: [pulseaudio] alsa-mixer.c: Probe of element 'Aux' succeeded (volume=0, switch=0, enumeration=0). ( 0.439| 0.000) D: [pulseaudio] alsa-mixer.c: Probe of element 'Video' succeeded (volume=0, switch=0, enumeration=0). ( 0.439| 0.000) D: [pulseaudio] alsa-mixer.c: Probe of element 'Mic/Line' succeeded (volume=0, switch=0, enumeration=0). ( 0.439| 0.000) D: [pulseaudio] alsa-mixer.c: Probe of element 'TV Tuner' succeeded (volume=0, switch=0, enumeration=0). ( 0.439| 0.000) D: [pulseaudio] alsa-mixer.c: Probe of element 'FM' succeeded (volume=0, switch=0, enumeration=0). ( 0.439| 0.000) D: [pulseaudio] alsa-mixer.c: Probe of element 'Inverted Internal Mic' succeeded (volume=0, switch=0, enumeration=0). ( 0.439| 0.000) D: [pulseaudio] alsa-mixer.c: Probe of element 'Mic Jack Mode' succeeded (volume=0, switch=0, enumeration=0). ( 0.439| 0.000) D: [pulseaudio] alsa-mixer.c: Probing path 'analog-input-microphone-rear' ( 0.439| 0.000) D: [pulseaudio] alsa-mixer.c: Probe of jack 'Rear Mic Jack' succeeded (found!) ( 0.439| 0.000) D: [pulseaudio] alsa-mixer.c: Probe of element 'Rear Mic Boost' succeeded (volume=1, switch=0, enumeration=0). ( 0.439| 0.000) D: [pulseaudio] alsa-mixer.c: Probe of element 'Rear Mic' succeeded (volume=0, switch=0, enumeration=0). ( 0.439| 0.000) D: [pulseaudio] alsa-mixer.c: Probe of element 'Capture' succeeded (volume=1, switch=1, enumeration=0). ( 0.439| 0.000) D: [pulseaudio] alsa-mixer.c: Probe of element 'Input Source' succeeded (volume=0, switch=0, enumeration=1). ( 0.439| 0.000) D: [pulseaudio] alsa-mixer.c: Probe of element 'Capture Source' succeeded (volume=0, switch=0, enumeration=0). ( 0.439| 0.000) D: [pulseaudio] alsa-mixer.c: Probe of element 'Mic' succeeded (volume=0, switch=0, enumeration=0). ( 0.439| 0.000) D: [pulseaudio] alsa-mixer.c: Probe of element 'Internal Mic' succeeded (volume=0, switch=0, enumeration=0). ( 0.439| 0.000) D: [pulseaudio] alsa-mixer.c: Probe of element 'Front Mic' succeeded (volume=0, switch=0, enumeration=0). ( 0.439| 0.000) D: [pulseaudio] alsa-mixer.c: Probe of element 'Dock Mic' succeeded (volume=0, switch=0, enumeration=0). ( 0.439| 0.000) D: [pulseaudio] alsa-mixer.c: Probe of element 'Mic Boost' succeeded (volume=0, switch=0, enumeration=0). ( 0.439| 0.000) D: [pulseaudio] alsa-mixer.c: Probe of element 'Dock Mic Boost' succeeded (volume=0, switch=0, enumeration=0). ( 0.439| 0.000) D: [pulseaudio] alsa-mixer.c: Probe of element 'Internal Mic Boost' succeeded (volume=0, switch=0, enumeration=0). ( 0.439| 0.000) D: [pulseaudio] alsa-mixer.c: Probe of element 'Front Mic Boost' succeeded (volume=2, switch=0, enumeration=0). ( 0.439| 0.000) D: [pulseaudio] alsa-mixer.c: Probe of element 'Line' succeeded (volume=0, switch=0, enumeration=0). ( 0.439| 0.000) D: [pulseaudio] alsa-mixer.c: Probe of element 'Line Boost' succeeded (volume=2, switch=0, enumeration=0). ( 0.439| 0.000) D: [pulseaudio] alsa-mixer.c: Probe of element 'Aux' succeeded (volume=0, switch=0, enumeration=0). ( 0.439| 0.000) D: [pulseaudio] alsa-mixer.c: Probe of element 'Video' succeeded (volume=0, switch=0, enumeration=0). ( 0.439| 0.000) D: [pulseaudio] alsa-mixer.c: Probe of element 'Mic/Line' succeeded (volume=0, switch=0, enumeration=0). ( 0.439| 0.000) D: [pulseaudio] alsa-mixer.c: Probe of element 'TV Tuner' succeeded (volume=0, switch=0, enumeration=0). ( 0.439| 0.000) D: [pulseaudio] alsa-mixer.c: Probe of element 'FM' succeeded (volume=0, switch=0, enumeration=0). ( 0.439| 0.000) D: [pulseaudio] alsa-mixer.c: Probe of element 'Inverted Internal Mic' succeeded (volume=0, switch=0, enumeration=0). ( 0.439| 0.000) D: [pulseaudio] alsa-mixer.c: Probe of element 'Mic Jack Mode' succeeded (volume=0, switch=0, enumeration=0). ( 0.439| 0.000) D: [pulseaudio] alsa-mixer.c: Probing path 'analog-input-microphone-internal' ( 0.439| 0.000) D: [pulseaudio] alsa-mixer.c: Probe of jack 'Mic Jack' succeeded (not found) ( 0.439| 0.000) D: [pulseaudio] alsa-mixer.c: Probe of jack 'Dock Mic Jack' succeeded (not found) ( 0.439| 0.000) D: [pulseaudio] alsa-mixer.c: Probe of jack 'Front Mic Jack' succeeded (found!) ( 0.439| 0.000) D: [pulseaudio] alsa-mixer.c: Probe of jack 'Rear Mic Jack' succeeded (found!) ( 0.439| 0.000) D: [pulseaudio] alsa-mixer.c: Probe of jack 'Internal Mic Phantom Jack' succeeded (not found) ( 0.439| 0.000) D: [pulseaudio] alsa-mixer.c: Probe of element 'Internal Mic Boost' succeeded (volume=0, switch=0, enumeration=0). ( 0.439| 0.000) D: [pulseaudio] alsa-mixer.c: Probe of element 'Int Mic Boost' succeeded (volume=0, switch=0, enumeration=0). ( 0.439| 0.000) D: [pulseaudio] alsa-mixer.c: Probe of element 'Internal Mic' succeeded (volume=0, switch=0, enumeration=0). ( 0.439| 0.000) D: [pulseaudio] alsa-mixer.c: Probe of element 'Int Mic' succeeded (volume=0, switch=0, enumeration=0). ( 0.439| 0.000) D: [pulseaudio] alsa-mixer.c: Probe of element 'Capture' succeeded (volume=1, switch=1, enumeration=0). ( 0.439| 0.000) D: [pulseaudio] alsa-mixer.c: Probe of element 'Input Source' succeeded (volume=0, switch=0, enumeration=1). ( 0.439| 0.000) D: [pulseaudio] alsa-mixer.c: Probe of element 'Capture Source' succeeded (volume=0, switch=0, enumeration=0). ( 0.439| 0.000) D: [pulseaudio] alsa-mixer.c: Probe of element 'Mic' succeeded (volume=0, switch=0, enumeration=0). ( 0.439| 0.000) D: [pulseaudio] alsa-mixer.c: Probe of element 'Dock Mic' succeeded (volume=0, switch=0, enumeration=0). ( 0.439| 0.000) D: [pulseaudio] alsa-mixer.c: Probe of element 'Front Mic' succeeded (volume=0, switch=0, enumeration=0). ( 0.439| 0.000) D: [pulseaudio] alsa-mixer.c: Probe of element 'Rear Mic' succeeded (volume=0, switch=0, enumeration=0). ( 0.439| 0.000) D: [pulseaudio] alsa-mixer.c: Probe of element 'Headphone Mic' succeeded (volume=0, switch=0, enumeration=0). ( 0.439| 0.000) D: [pulseaudio] alsa-mixer.c: Probe of element 'Headphone Mic Boost' succeeded (volume=0, switch=0, enumeration=0). ( 0.439| 0.000) D: [pulseaudio] alsa-mixer.c: Probe of element 'Mic Boost' succeeded (volume=0, switch=0, enumeration=0). ( 0.439| 0.000) D: [pulseaudio] alsa-mixer.c: Probe of element 'Dock Mic Boost' succeeded (volume=0, switch=0, enumeration=0). ( 0.439| 0.000) D: [pulseaudio] alsa-mixer.c: Probe of element 'Front Mic Boost' succeeded (volume=2, switch=0, enumeration=0). ( 0.439| 0.000) D: [pulseaudio] alsa-mixer.c: Probe of element 'Rear Mic Boost' succeeded (volume=2, switch=0, enumeration=0). ( 0.439| 0.000) D: [pulseaudio] alsa-mixer.c: Probe of element 'Line' succeeded (volume=0, switch=0, enumeration=0). ( 0.439| 0.000) D: [pulseaudio] alsa-mixer.c: Probe of element 'Line Boost' succeeded (volume=2, switch=0, enumeration=0). ( 0.439| 0.000) D: [pulseaudio] alsa-mixer.c: Probe of element 'Aux' succeeded (volume=0, switch=0, enumeration=0). ( 0.439| 0.000) D: [pulseaudio] alsa-mixer.c: Probe of element 'Video' succeeded (volume=0, switch=0, enumeration=0). ( 0.439| 0.000) D: [pulseaudio] alsa-mixer.c: Probe of element 'Mic/Line' succeeded (volume=0, switch=0, enumeration=0). ( 0.439| 0.000) D: [pulseaudio] alsa-mixer.c: Probe of element 'TV Tuner' succeeded (volume=0, switch=0, enumeration=0). ( 0.439| 0.000) D: [pulseaudio] alsa-mixer.c: Probe of element 'FM' succeeded (volume=0, switch=0, enumeration=0). ( 0.439| 0.000) D: [pulseaudio] alsa-mixer.c: Probe of element 'Inverted Internal Mic' succeeded (volume=0, switch=0, enumeration=0). ( 0.439| 0.000) D: [pulseaudio] alsa-mixer.c: Probe of element 'Mic Jack Mode' succeeded (volume=0, switch=0, enumeration=0). ( 0.439| 0.000) D: [pulseaudio] alsa-mixer.c: Skipping path 'analog-input-microphone-internal', none of required-any elements preset. ( 0.439| 0.000) D: [pulseaudio] alsa-mixer.c: Probing path 'analog-input-microphone-dock' ( 0.439| 0.000) D: [pulseaudio] alsa-mixer.c: Probe of jack 'Dock Mic Jack' succeeded (not found) ( 0.439| 0.000) D: [pulseaudio] alsa-mixer.c: Probe of element 'Dock Mic Boost' succeeded (volume=0, switch=0, enumeration=0). ( 0.439| 0.000) D: [pulseaudio] alsa-mixer.c: Probe of element 'Dock Mic' succeeded (volume=0, switch=0, enumeration=0). ( 0.439| 0.000) D: [pulseaudio] alsa-mixer.c: Probe of element 'Capture' succeeded (volume=1, switch=1, enumeration=0). ( 0.439| 0.000) D: [pulseaudio] alsa-mixer.c: Probe of element 'Input Source' succeeded (volume=0, switch=0, enumeration=1). ( 0.439| 0.000) D: [pulseaudio] alsa-mixer.c: Probe of element 'Capture Source' succeeded (volume=0, switch=0, enumeration=0). ( 0.439| 0.000) D: [pulseaudio] alsa-mixer.c: Probe of element 'Mic' succeeded (volume=0, switch=0, enumeration=0). ( 0.439| 0.000) D: [pulseaudio] alsa-mixer.c: Probe of element 'Internal Mic' succeeded (volume=0, switch=0, enumeration=0). ( 0.439| 0.000) D: [pulseaudio] alsa-mixer.c: Probe of element 'Front Mic' succeeded (volume=0, switch=0, enumeration=0). ( 0.439| 0.000) D: [pulseaudio] alsa-mixer.c: Probe of element 'Rear Mic' succeeded (volume=0, switch=0, enumeration=0). ( 0.439| 0.000) D: [pulseaudio] alsa-mixer.c: Probe of element 'Mic Boost' succeeded (volume=0, switch=0, enumeration=0). ( 0.439| 0.000) D: [pulseaudio] alsa-mixer.c: Probe of element 'Internal Mic Boost' succeeded (volume=0, switch=0, enumeration=0). ( 0.439| 0.000) D: [pulseaudio] alsa-mixer.c: Probe of element 'Front Mic Boost' succeeded (volume=2, switch=0, enumeration=0). ( 0.439| 0.000) D: [pulseaudio] alsa-mixer.c: Probe of element 'Rear Mic Boost' succeeded (volume=2, switch=0, enumeration=0). ( 0.439| 0.000) D: [pulseaudio] alsa-mixer.c: Probe of element 'Line' succeeded (volume=0, switch=0, enumeration=0). ( 0.439| 0.000) D: [pulseaudio] alsa-mixer.c: Probe of element 'Line Boost' succeeded (volume=2, switch=0, enumeration=0). ( 0.439| 0.000) D: [pulseaudio] alsa-mixer.c: Probe of element 'Aux' succeeded (volume=0, switch=0, enumeration=0). ( 0.439| 0.000) D: [pulseaudio] alsa-mixer.c: Probe of element 'Video' succeeded (volume=0, switch=0, enumeration=0). ( 0.439| 0.000) D: [pulseaudio] alsa-mixer.c: Probe of element 'Mic/Line' succeeded (volume=0, switch=0, enumeration=0). ( 0.439| 0.000) D: [pulseaudio] alsa-mixer.c: Probe of element 'TV Tuner' succeeded (volume=0, switch=0, enumeration=0). ( 0.439| 0.000) D: [pulseaudio] alsa-mixer.c: Probe of element 'FM' succeeded (volume=0, switch=0, enumeration=0). ( 0.439| 0.000) D: [pulseaudio] alsa-mixer.c: Probe of element 'Inverted Internal Mic' succeeded (volume=0, switch=0, enumeration=0). ( 0.439| 0.000) D: [pulseaudio] alsa-mixer.c: Probe of element 'Mic Jack Mode' succeeded (volume=0, switch=0, enumeration=0). ( 0.439| 0.000) D: [pulseaudio] alsa-mixer.c: Skipping path 'analog-input-microphone-dock', none of required-any elements preset. ( 0.439| 0.000) D: [pulseaudio] alsa-mixer.c: Probing path 'analog-input' ( 0.439| 0.000) D: [pulseaudio] alsa-mixer.c: Probe of element 'Capture' succeeded (volume=1, switch=1, enumeration=0). ( 0.439| 0.000) D: [pulseaudio] alsa-mixer.c: Probe of element 'Mic' succeeded (volume=0, switch=0, enumeration=0). ( 0.439| 0.000) D: [pulseaudio] alsa-mixer.c: Probe of element 'Dock Mic' succeeded (volume=0, switch=0, enumeration=0). ( 0.439| 0.000) D: [pulseaudio] alsa-mixer.c: Probe of element 'Dock Mic Boost' succeeded (volume=0, switch=0, enumeration=0). ( 0.439| 0.000) D: [pulseaudio] alsa-mixer.c: Probe of element 'Front Mic' succeeded (volume=0, switch=0, enumeration=0). ( 0.439| 0.000) D: [pulseaudio] alsa-mixer.c: Probe of element 'Front Mic Boost' failed. ( 0.439| 0.000) D: [pulseaudio] alsa-mixer.c: Probing path 'analog-input-microphone' ( 0.439| 0.000) D: [pulseaudio] alsa-mixer.c: Probe of jack 'Mic Jack' succeeded (not found) ( 0.439| 0.000) D: [pulseaudio] alsa-mixer.c: Probe of element 'Mic Boost' succeeded (volume=0, switch=0, enumeration=0). ( 0.439| 0.000) D: [pulseaudio] alsa-mixer.c: Probe of element 'Mic' succeeded (volume=0, switch=0, enumeration=0). ( 0.440| 0.000) D: [pulseaudio] alsa-mixer.c: Probe of element 'Capture' succeeded (volume=1, switch=1, enumeration=0). ( 0.440| 0.000) D: [pulseaudio] alsa-mixer.c: Probe of element 'Input Source' succeeded (volume=0, switch=0, enumeration=1). ( 0.440| 0.000) D: [pulseaudio] alsa-mixer.c: Probe of element 'Capture Source' succeeded (volume=0, switch=0, enumeration=0). ( 0.440| 0.000) D: [pulseaudio] alsa-mixer.c: Probe of element 'Mic Select' succeeded (volume=0, switch=0, enumeration=0). ( 0.440| 0.000) D: [pulseaudio] alsa-mixer.c: Probe of element 'Mic Boost (+20dB)' succeeded (volume=0, switch=0, enumeration=0). ( 0.440| 0.000) D: [pulseaudio] alsa-mixer.c: Probe of element 'Front Mic' succeeded (volume=0, switch=0, enumeration=0). ( 0.440| 0.000) D: [pulseaudio] alsa-mixer.c: Probe of element 'Internal Mic' succeeded (volume=0, switch=0, enumeration=0). ( 0.440| 0.000) D: [pulseaudio] alsa-mixer.c: Probe of element 'Rear Mic' succeeded (volume=0, switch=0, enumeration=0). ( 0.440| 0.000) D: [pulseaudio] alsa-mixer.c: Probe of element 'Dock Mic' succeeded (volume=0, switch=0, enumeration=0). ( 0.440| 0.000) D: [pulseaudio] alsa-mixer.c: Probe of element 'Dock Mic Boost' succeeded (volume=0, switch=0, enumeration=0). ( 0.440| 0.000) D: [pulseaudio] alsa-mixer.c: Probe of element 'Internal Mic Boost' succeeded (volume=0, switch=0, enumeration=0). ( 0.440| 0.000) D: [pulseaudio] alsa-mixer.c: Probe of element 'Front Mic Boost' succeeded (volume=2, switch=0, enumeration=0). ( 0.440| 0.000) D: [pulseaudio] alsa-mixer.c: Probe of element 'Rear Mic Boost' succeeded (volume=2, switch=0, enumeration=0). ( 0.440| 0.000) D: [pulseaudio] alsa-mixer.c: Probe of element 'Line' succeeded (volume=0, switch=0, enumeration=0). ( 0.440| 0.000) D: [pulseaudio] alsa-mixer.c: Probe of element 'Line Boost' succeeded (volume=2, switch=0, enumeration=0). ( 0.440| 0.000) D: [pulseaudio] alsa-mixer.c: Probe of element 'Aux' succeeded (volume=0, switch=0, enumeration=0). ( 0.440| 0.000) D: [pulseaudio] alsa-mixer.c: Probe of element 'Video' succeeded (volume=0, switch=0, enumeration=0). ( 0.440| 0.000) D: [pulseaudio] alsa-mixer.c: Probe of element 'Mic/Line' succeeded (volume=0, switch=0, enumeration=0). ( 0.440| 0.000) D: [pulseaudio] alsa-mixer.c: Probe of element 'TV Tuner' succeeded (volume=0, switch=0, enumeration=0). ( 0.440| 0.000) D: [pulseaudio] alsa-mixer.c: Probe of element 'FM' succeeded (volume=0, switch=0, enumeration=0). ( 0.440| 0.000) D: [pulseaudio] alsa-mixer.c: Probe of element 'Inverted Internal Mic' succeeded (volume=0, switch=0, enumeration=0). ( 0.440| 0.000) D: [pulseaudio] alsa-mixer.c: Probe of element 'Mic Jack Mode' succeeded (volume=0, switch=0, enumeration=0). ( 0.440| 0.000) D: [pulseaudio] alsa-mixer.c: Skipping path 'analog-input-microphone', none of required-any elements preset. ( 0.440| 0.000) D: [pulseaudio] alsa-mixer.c: Probing path 'analog-input-linein' ( 0.440| 0.000) D: [pulseaudio] alsa-mixer.c: Probe of jack 'Line Jack' succeeded (found!) ( 0.440| 0.000) D: [pulseaudio] alsa-mixer.c: Probe of element 'Capture' succeeded (volume=1, switch=1, enumeration=0). ( 0.440| 0.000) D: [pulseaudio] alsa-mixer.c: Probe of element 'Line Boost' succeeded (volume=1, switch=0, enumeration=0). ( 0.440| 0.000) D: [pulseaudio] alsa-mixer.c: Probe of element 'Line' succeeded (volume=0, switch=0, enumeration=0). ( 0.440| 0.000) D: [pulseaudio] alsa-mixer.c: Probe of element 'Input Source' succeeded (volume=0, switch=0, enumeration=1). ( 0.440| 0.000) D: [pulseaudio] alsa-mixer.c: Probe of element 'Capture Source' succeeded (volume=0, switch=0, enumeration=0). ( 0.440| 0.000) D: [pulseaudio] alsa-mixer.c: Probe of element 'Mic' succeeded (volume=0, switch=0, enumeration=0). ( 0.440| 0.000) D: [pulseaudio] alsa-mixer.c: Probe of element 'Dock Mic' succeeded (volume=0, switch=0, enumeration=0). ( 0.440| 0.000) D: [pulseaudio] alsa-mixer.c: Probe of element 'Internal Mic' succeeded (volume=0, switch=0, enumeration=0). ( 0.440| 0.000) D: [pulseaudio] alsa-mixer.c: Probe of element 'Front Mic' succeeded (volume=0, switch=0, enumeration=0). ( 0.440| 0.000) D: [pulseaudio] alsa-mixer.c: Probe of element 'Rear Mic' succeeded (volume=0, switch=0, enumeration=0). ( 0.440| 0.000) D: [pulseaudio] alsa-mixer.c: Probe of element 'Mic Boost' succeeded (volume=0, switch=0, enumeration=0). ( 0.440| 0.000) D: [pulseaudio] alsa-mixer.c: Probe of element 'Dock Mic Boost' succeeded (volume=0, switch=0, enumeration=0). ( 0.440| 0.000) D: [pulseaudio] alsa-mixer.c: Probe of element 'Internal Mic Boost' succeeded (volume=0, switch=0, enumeration=0). ( 0.440| 0.000) D: [pulseaudio] alsa-mixer.c: Probe of element 'Front Mic Boost' succeeded (volume=2, switch=0, enumeration=0). ( 0.440| 0.000) D: [pulseaudio] alsa-mixer.c: Probe of element 'Rear Mic Boost' succeeded (volume=2, switch=0, enumeration=0). ( 0.440| 0.000) D: [pulseaudio] alsa-mixer.c: Probe of element 'Aux' succeeded (volume=0, switch=0, enumeration=0). ( 0.440| 0.000) D: [pulseaudio] alsa-mixer.c: Probe of element 'Video' succeeded (volume=0, switch=0, enumeration=0). ( 0.440| 0.000) D: [pulseaudio] alsa-mixer.c: Probe of element 'Mic/Line' succeeded (volume=0, switch=0, enumeration=0). ( 0.440| 0.000) D: [pulseaudio] alsa-mixer.c: Probe of element 'TV Tuner' succeeded (volume=0, switch=0, enumeration=0). ( 0.440| 0.000) D: [pulseaudio] alsa-mixer.c: Probe of element 'FM' succeeded (volume=0, switch=0, enumeration=0). ( 0.440| 0.000) D: [pulseaudio] alsa-mixer.c: Probe of element 'Mic Jack Mode' succeeded (volume=0, switch=0, enumeration=0). ( 0.440| 0.000) D: [pulseaudio] alsa-mixer.c: Probing path 'analog-input' ( 0.440| 0.000) D: [pulseaudio] alsa-mixer.c: Probe of element 'Capture' succeeded (volume=1, switch=1, enumeration=0). ( 0.440| 0.000) D: [pulseaudio] alsa-mixer.c: Probe of element 'Mic' succeeded (volume=0, switch=0, enumeration=0). ( 0.440| 0.000) D: [pulseaudio] alsa-mixer.c: Probe of element 'Internal Mic' succeeded (volume=0, switch=0, enumeration=0). ( 0.440| 0.000) D: [pulseaudio] alsa-mixer.c: Probe of element 'Line' succeeded (volume=0, switch=0, enumeration=0). ( 0.440| 0.000) D: [pulseaudio] alsa-mixer.c: Probe of element 'Aux' failed. ( 0.440| 0.000) D: [pulseaudio] alsa-mixer.c: Probing path 'analog-input-video' ( 0.440| 0.000) D: [pulseaudio] alsa-mixer.c: Probe of element 'Capture' succeeded (volume=1, switch=1, enumeration=0). ( 0.440| 0.000) D: [pulseaudio] alsa-mixer.c: Probe of element 'Mic' succeeded (volume=0, switch=0, enumeration=0). ( 0.440| 0.000) D: [pulseaudio] alsa-mixer.c: Probe of element 'Internal Mic' succeeded (volume=0, switch=0, enumeration=0). ( 0.440| 0.000) D: [pulseaudio] alsa-mixer.c: Probe of element 'Line' succeeded (volume=0, switch=0, enumeration=0). ( 0.440| 0.000) D: [pulseaudio] alsa-mixer.c: Probe of element 'Aux' succeeded (volume=0, switch=0, enumeration=0). ( 0.440| 0.000) D: [pulseaudio] alsa-mixer.c: Probe of element 'Video' failed. ( 0.440| 0.000) D: [pulseaudio] alsa-mixer.c: Probing path 'analog-input-video' ( 0.440| 0.000) D: [pulseaudio] alsa-mixer.c: Probe of element 'Capture' succeeded (volume=1, switch=1, enumeration=0). ( 0.440| 0.000) D: [pulseaudio] alsa-mixer.c: Probe of element 'Mic' succeeded (volume=0, switch=0, enumeration=0). ( 0.440| 0.000) D: [pulseaudio] alsa-mixer.c: Probe of element 'Internal Mic' succeeded (volume=0, switch=0, enumeration=0). ( 0.440| 0.000) D: [pulseaudio] alsa-mixer.c: Probe of element 'Line' succeeded (volume=0, switch=0, enumeration=0). ( 0.440| 0.000) D: [pulseaudio] alsa-mixer.c: Probe of element 'Aux' succeeded (volume=0, switch=0, enumeration=0). ( 0.440| 0.000) D: [pulseaudio] alsa-mixer.c: Probe of element 'Video' succeeded (volume=0, switch=0, enumeration=0). ( 0.440| 0.000) D: [pulseaudio] alsa-mixer.c: Probe of element 'Mic/Line' succeeded (volume=0, switch=0, enumeration=0). ( 0.440| 0.000) D: [pulseaudio] alsa-mixer.c: Probe of element 'TV Tuner' failed. ( 0.440| 0.000) D: [pulseaudio] alsa-mixer.c: Probing path 'analog-input-radio' ( 0.440| 0.000) D: [pulseaudio] alsa-mixer.c: Probe of element 'Capture' succeeded (volume=1, switch=1, enumeration=0). ( 0.440| 0.000) D: [pulseaudio] alsa-mixer.c: Probe of element 'Mic' succeeded (volume=0, switch=0, enumeration=0). ( 0.440| 0.000) D: [pulseaudio] alsa-mixer.c: Probe of element 'Internal Mic' succeeded (volume=0, switch=0, enumeration=0). ( 0.440| 0.000) D: [pulseaudio] alsa-mixer.c: Probe of element 'Line' succeeded (volume=0, switch=0, enumeration=0). ( 0.440| 0.000) D: [pulseaudio] alsa-mixer.c: Probe of element 'Aux' succeeded (volume=0, switch=0, enumeration=0). ( 0.440| 0.000) D: [pulseaudio] alsa-mixer.c: Probe of element 'Video' succeeded (volume=0, switch=0, enumeration=0). ( 0.440| 0.000) D: [pulseaudio] alsa-mixer.c: Probe of element 'Mic/Line' succeeded (volume=0, switch=0, enumeration=0). ( 0.440| 0.000) D: [pulseaudio] alsa-mixer.c: Probe of element 'TV Tuner' succeeded (volume=0, switch=0, enumeration=0). ( 0.440| 0.000) D: [pulseaudio] alsa-mixer.c: Probe of element 'FM' failed. ( 0.440| 0.000) D: [pulseaudio] alsa-mixer.c: Probing path 'analog-input' ( 0.440| 0.000) D: [pulseaudio] alsa-mixer.c: Probe of element 'Capture' succeeded (volume=1, switch=1, enumeration=0). ( 0.440| 0.000) D: [pulseaudio] alsa-mixer.c: Probe of element 'Mic' succeeded (volume=0, switch=0, enumeration=0). ( 0.440| 0.000) D: [pulseaudio] alsa-mixer.c: Probe of element 'Internal Mic' succeeded (volume=0, switch=0, enumeration=0). ( 0.440| 0.000) D: [pulseaudio] alsa-mixer.c: Probe of element 'Line' succeeded (volume=0, switch=0, enumeration=0). ( 0.440| 0.000) D: [pulseaudio] alsa-mixer.c: Probe of element 'Line Boost' succeeded (volume=2, switch=0, enumeration=0). ( 0.440| 0.000) D: [pulseaudio] alsa-mixer.c: Probe of element 'Aux' succeeded (volume=0, switch=0, enumeration=0). ( 0.440| 0.000) D: [pulseaudio] alsa-mixer.c: Probe of element 'Video' succeeded (volume=0, switch=0, enumeration=0). ( 0.440| 0.000) D: [pulseaudio] alsa-mixer.c: Probe of element 'Mic/Line' failed. ( 0.440| 0.000) D: [pulseaudio] alsa-mixer.c: Probing path 'analog-input-microphone' ( 0.440| 0.000) D: [pulseaudio] alsa-mixer.c: Probe of jack 'Headphone Mic Jack' succeeded (not found) ( 0.440| 0.000) D: [pulseaudio] alsa-mixer.c: Probe of element 'Headphone Mic Boost' succeeded (volume=0, switch=0, enumeration=0). ( 0.440| 0.000) D: [pulseaudio] alsa-mixer.c: Probe of element 'Headphone Mic' succeeded (volume=0, switch=0, enumeration=0). ( 0.440| 0.000) D: [pulseaudio] alsa-mixer.c: Probe of element 'Capture' succeeded (volume=1, switch=1, enumeration=0). ( 0.440| 0.000) D: [pulseaudio] alsa-mixer.c: Probe of element 'Input Source' succeeded (volume=0, switch=0, enumeration=1). ( 0.440| 0.000) D: [pulseaudio] alsa-mixer.c: Probe of element 'Capture Source' succeeded (volume=0, switch=0, enumeration=0). ( 0.440| 0.000) D: [pulseaudio] alsa-mixer.c: Probe of element 'Auto-Mute Mode' succeeded (volume=0, switch=0, enumeration=1). ( 0.440| 0.000) D: [pulseaudio] alsa-mixer.c: Probe of element 'Front Mic' succeeded (volume=0, switch=0, enumeration=0). ( 0.440| 0.000) D: [pulseaudio] alsa-mixer.c: Probe of element 'Internal Mic' succeeded (volume=0, switch=0, enumeration=0). ( 0.440| 0.000) D: [pulseaudio] alsa-mixer.c: Probe of element 'Rear Mic' succeeded (volume=0, switch=0, enumeration=0). ( 0.440| 0.000) D: [pulseaudio] alsa-mixer.c: Probe of element 'Dock Mic' succeeded (volume=0, switch=0, enumeration=0). ( 0.440| 0.000) D: [pulseaudio] alsa-mixer.c: Probe of element 'Dock Mic Boost' succeeded (volume=0, switch=0, enumeration=0). ( 0.440| 0.000) D: [pulseaudio] alsa-mixer.c: Probe of element 'Internal Mic Boost' succeeded (volume=0, switch=0, enumeration=0). ( 0.440| 0.000) D: [pulseaudio] alsa-mixer.c: Probe of element 'Front Mic Boost' succeeded (volume=2, switch=0, enumeration=0). ( 0.440| 0.000) D: [pulseaudio] alsa-mixer.c: Probe of element 'Rear Mic Boost' succeeded (volume=2, switch=0, enumeration=0). ( 0.440| 0.000) D: [pulseaudio] alsa-mixer.c: Probe of element 'Line' succeeded (volume=0, switch=0, enumeration=0). ( 0.440| 0.000) D: [pulseaudio] alsa-mixer.c: Probe of element 'Line Boost' succeeded (volume=2, switch=0, enumeration=0). ( 0.440| 0.000) D: [pulseaudio] alsa-mixer.c: Probe of element 'Aux' succeeded (volume=0, switch=0, enumeration=0). ( 0.440| 0.000) D: [pulseaudio] alsa-mixer.c: Probe of element 'Video' succeeded (volume=0, switch=0, enumeration=0). ( 0.440| 0.000) D: [pulseaudio] alsa-mixer.c: Probe of element 'Mic/Line' succeeded (volume=0, switch=0, enumeration=0). ( 0.440| 0.000) D: [pulseaudio] alsa-mixer.c: Probe of element 'TV Tuner' succeeded (volume=0, switch=0, enumeration=0). ( 0.440| 0.000) D: [pulseaudio] alsa-mixer.c: Probe of element 'FM' succeeded (volume=0, switch=0, enumeration=0). ( 0.440| 0.000) D: [pulseaudio] alsa-mixer.c: Probe of element 'Inverted Internal Mic' succeeded (volume=0, switch=0, enumeration=0). ( 0.440| 0.000) D: [pulseaudio] alsa-mixer.c: Probe of element 'Mic Jack Mode' succeeded (volume=0, switch=0, enumeration=0). ( 0.440| 0.000) D: [pulseaudio] alsa-mixer.c: Skipping path 'analog-input-microphone', none of required-any elements preset. ( 0.440| 0.000) D: [pulseaudio] alsa-mixer.c: Available mixer paths (after tidying): ( 0.440| 0.000) D: [pulseaudio] alsa-mixer.c: Path Set 0xc76970, direction=2 ( 0.440| 0.000) D: [pulseaudio] alsa-mixer.c: Path analog-input-microphone-front (Front Microphone), direction=2, priority=85, probed=yes, supported=yes, has_mute=yes, has_volume=yes, has_dB=yes, min_volume=0, max_volume=3, min_dB=-16, max_dB=60 ( 0.440| 0.000) D: [pulseaudio] alsa-mixer.c: Element Front Mic Boost, direction=2, switch=0, volume=1, volume_limit=-1, enumeration=0, required=0, required_any=4, required_absent=0, mask=0x3600000000f66, n_channels=2, override_map=yes ( 0.440| 0.000) D: [pulseaudio] alsa-mixer.c: Element Capture, direction=2, switch=1, volume=1, volume_limit=-1, enumeration=0, required=0, required_any=0, required_absent=0, mask=0x3600000000f66, n_channels=2, override_map=yes ( 0.440| 0.000) D: [pulseaudio] alsa-mixer.c: Element Input Source, direction=2, switch=0, volume=0, volume_limit=-1, enumeration=1, required=0, required_any=0, required_absent=0, mask=0x0, n_channels=0, override_map=no ( 0.440| 0.000) D: [pulseaudio] alsa-mixer.c: Option Front Mic (analog-input-microphone-front/analog-input-microphone-front) index=0, priority=0 ( 0.440| 0.000) D: [pulseaudio] alsa-mixer.c: Element Rear Mic Boost, direction=2, switch=0, volume=2, volume_limit=-1, enumeration=0, required=0, required_any=0, required_absent=0, mask=0x6, n_channels=2, override_map=no ( 0.441| 0.000) D: [pulseaudio] alsa-mixer.c: Element Line Boost, direction=2, switch=0, volume=2, volume_limit=-1, enumeration=0, required=0, required_any=0, required_absent=0, mask=0x6, n_channels=2, override_map=no ( 0.441| 0.000) D: [pulseaudio] alsa-mixer.c: Jack Front Mic, alsa_name='Front Mic Jack', detection possible ( 0.441| 0.000) D: [pulseaudio] alsa-mixer.c: Setting analog-input-microphone-front (analog-input-microphone-front) priority=0 ( 0.441| 0.000) D: [pulseaudio] alsa-mixer.c: Path analog-input-microphone-rear (Rear Microphone), direction=2, priority=82, probed=yes, supported=yes, has_mute=yes, has_volume=yes, has_dB=yes, min_volume=0, max_volume=3, min_dB=-16, max_dB=60 ( 0.441| 0.000) D: [pulseaudio] alsa-mixer.c: Element Rear Mic Boost, direction=2, switch=0, volume=1, volume_limit=-1, enumeration=0, required=0, required_any=4, required_absent=0, mask=0x3600000000f66, n_channels=2, override_map=yes ( 0.441| 0.000) D: [pulseaudio] alsa-mixer.c: Element Capture, direction=2, switch=1, volume=1, volume_limit=-1, enumeration=0, required=0, required_any=0, required_absent=0, mask=0x3600000000f66, n_channels=2, override_map=yes ( 0.441| 0.000) D: [pulseaudio] alsa-mixer.c: Element Input Source, direction=2, switch=0, volume=0, volume_limit=-1, enumeration=1, required=0, required_any=0, required_absent=0, mask=0x0, n_channels=0, override_map=no ( 0.441| 0.000) D: [pulseaudio] alsa-mixer.c: Option Rear Mic (analog-input-microphone-rear/analog-input-microphone-rear) index=1, priority=0 ( 0.441| 0.000) D: [pulseaudio] alsa-mixer.c: Element Front Mic Boost, direction=2, switch=0, volume=2, volume_limit=-1, enumeration=0, required=0, required_any=0, required_absent=0, mask=0x6, n_channels=2, override_map=no ( 0.441| 0.000) D: [pulseaudio] alsa-mixer.c: Element Line Boost, direction=2, switch=0, volume=2, volume_limit=-1, enumeration=0, required=0, required_any=0, required_absent=0, mask=0x6, n_channels=2, override_map=no ( 0.441| 0.000) D: [pulseaudio] alsa-mixer.c: Jack Rear Mic, alsa_name='Rear Mic Jack', detection possible ( 0.441| 0.000) D: [pulseaudio] alsa-mixer.c: Setting analog-input-microphone-rear (analog-input-microphone-rear) priority=0 ( 0.441| 0.000) D: [pulseaudio] alsa-mixer.c: Path analog-input-linein (Line In), direction=2, priority=81, probed=yes, supported=yes, has_mute=yes, has_volume=yes, has_dB=yes, min_volume=0, max_volume=46, min_dB=-16, max_dB=60 ( 0.441| 0.000) D: [pulseaudio] alsa-mixer.c: Element Capture, direction=2, switch=1, volume=1, volume_limit=-1, enumeration=0, required=0, required_any=0, required_absent=0, mask=0x3600000000f66, n_channels=2, override_map=yes ( 0.441| 0.000) D: [pulseaudio] alsa-mixer.c: Element Line Boost, direction=2, switch=0, volume=1, volume_limit=-1, enumeration=0, required=0, required_any=4, required_absent=0, mask=0x3600000000f66, n_channels=2, override_map=yes ( 0.441| 0.000) D: [pulseaudio] alsa-mixer.c: Element Input Source, direction=2, switch=0, volume=0, volume_limit=-1, enumeration=1, required=0, required_any=0, required_absent=0, mask=0x0, n_channels=0, override_map=no ( 0.441| 0.000) D: [pulseaudio] alsa-mixer.c: Option Line (analog-input-linein/analog-input-linein) index=2, priority=0 ( 0.441| 0.000) D: [pulseaudio] alsa-mixer.c: Element Front Mic Boost, direction=2, switch=0, volume=2, volume_limit=-1, enumeration=0, required=0, required_any=0, required_absent=0, mask=0x6, n_channels=2, override_map=no ( 0.441| 0.000) D: [pulseaudio] alsa-mixer.c: Element Rear Mic Boost, direction=2, switch=0, volume=2, volume_limit=-1, enumeration=0, required=0, required_any=0, required_absent=0, mask=0x6, n_channels=2, override_map=no ( 0.441| 0.000) D: [pulseaudio] alsa-mixer.c: Jack Line, alsa_name='Line Jack', detection possible ( 0.441| 0.000) D: [pulseaudio] alsa-mixer.c: Setting analog-input-linein (analog-input-linein) priority=0 ( 0.441| 0.000) D: [pulseaudio] alsa-mixer.c: Looking at profile input:iec958-stereo ( 0.441| 0.000) D: [pulseaudio] alsa-mixer.c: Checking for recording on Digital Stereo (IEC958) (iec958-stereo) ( 0.441| 0.000) D: [pulseaudio] alsa-util.c: Trying iec958:0 with SND_PCM_NO_AUTO_FORMAT ... ( 0.441| 0.000) I: [pulseaudio] (alsa-lib)pcm_hw.c: open '/dev/snd/pcmC0D1c' failed (-2) ( 0.441| 0.000) I: [pulseaudio] alsa-util.c: Error opening PCM device iec958:0: No such file or directory ( 0.441| 0.000) D: [pulseaudio] alsa-mixer.c: Caching failure to open input:iec958-stereo ( 0.441| 0.000) D: [pulseaudio] alsa-mixer.c: Looking at profile output:analog-mono ( 0.441| 0.000) D: [pulseaudio] alsa-mixer.c: Checking for playback on Analog Mono (analog-mono) ( 0.441| 0.000) D: [pulseaudio] alsa-util.c: Trying hw:0 with SND_PCM_NO_AUTO_FORMAT ... ( 0.441| 0.000) D: [pulseaudio] alsa-util.c: Managed to open hw:0 ( 0.441| 0.000) D: [pulseaudio] alsa-util.c: snd_pcm_hw_params_set_channels(1) failed: Invalid argument ( 0.441| 0.000) D: [pulseaudio] alsa-util.c: Trying hw:0 without SND_PCM_NO_AUTO_FORMAT ... ( 0.441| 0.000) D: [pulseaudio] alsa-util.c: Managed to open hw:0 ( 0.441| 0.000) D: [pulseaudio] alsa-util.c: snd_pcm_hw_params_set_channels(1) failed: Invalid argument ( 0.441| 0.000) D: [pulseaudio] alsa-util.c: Trying plug:hw:0 with SND_PCM_NO_AUTO_FORMAT ... ( 0.441| 0.000) D: [pulseaudio] alsa-util.c: Managed to open plug:hw:0 ( 0.441| 0.000) D: [pulseaudio] alsa-util.c: snd_pcm_hw_params_set_channels(1) failed: Invalid argument ( 0.441| 0.000) D: [pulseaudio] alsa-util.c: Trying plug:hw:0 without SND_PCM_NO_AUTO_FORMAT ... ( 0.441| 0.000) D: [pulseaudio] alsa-util.c: Managed to open plug:hw:0 ( 0.442| 0.000) D: [pulseaudio] alsa-util.c: snd_pcm_hw_params_set_channels(1) failed: Invalid argument ( 0.442| 0.000) I: [pulseaudio] alsa-util.c: Failed to set hardware parameters on plug:hw:0: Invalid argument ( 0.442| 0.000) D: [pulseaudio] alsa-mixer.c: Caching failure to open output:analog-mono ( 0.442| 0.000) D: [pulseaudio] alsa-mixer.c: Skipping profile output:analog-mono+input:analog-mono - will not be able to open output:analog-mono ( 0.442| 0.000) D: [pulseaudio] alsa-mixer.c: Skipping profile output:analog-mono+input:analog-stereo - will not be able to open output:analog-mono ( 0.442| 0.000) D: [pulseaudio] alsa-mixer.c: Skipping profile output:analog-mono+input:iec958-stereo - will not be able to open output:analog-mono ( 0.442| 0.000) D: [pulseaudio] alsa-mixer.c: Looking at profile output:analog-stereo ( 0.442| 0.000) D: [pulseaudio] alsa-mixer.c: Checking for playback on Analog Stereo (analog-stereo) ( 0.442| 0.000) D: [pulseaudio] alsa-util.c: Trying front:0 with SND_PCM_NO_AUTO_FORMAT ... ( 0.442| 0.000) D: [pulseaudio] alsa-util.c: Managed to open front:0 ( 0.442| 0.000) D: [pulseaudio] alsa-util.c: Maximum hw buffer size is 371 ms ( 0.442| 0.000) D: [pulseaudio] alsa-util.c: Set buffer size first (to 2028 samples), period size second (to 1014 samples). ( 0.442| 0.000) D: [pulseaudio] alsa-mixer.c: Profile output:analog-stereo supported. ( 0.443| 0.000) I: [pulseaudio] (alsa-lib)control.c: Invalid CTL front:0 ( 0.443| 0.000) I: [pulseaudio] alsa-util.c: Unable to attach to mixer front:0: No such file or directory ( 0.443| 0.000) I: [pulseaudio] alsa-util.c: Successfully attached to mixer 'hw:0' ( 0.443| 0.000) D: [pulseaudio] alsa-mixer.c: Probing path 'analog-output' ( 0.443| 0.000) D: [pulseaudio] alsa-mixer.c: Probe of element 'Hardware Master' succeeded (volume=0, switch=0, enumeration=0). ( 0.443| 0.000) D: [pulseaudio] alsa-mixer.c: Probe of element 'Master' succeeded (volume=1, switch=1, enumeration=0). ( 0.443| 0.000) D: [pulseaudio] alsa-mixer.c: Probe of element 'Master Front' succeeded (volume=0, switch=0, enumeration=0). ( 0.443| 0.000) D: [pulseaudio] alsa-mixer.c: Probe of element 'Master Mono' succeeded (volume=0, switch=0, enumeration=0). ( 0.443| 0.000) D: [pulseaudio] alsa-mixer.c: Probe of element 'Line HP Swap' succeeded (volume=0, switch=0, enumeration=0). ( 0.443| 0.000) D: [pulseaudio] alsa-mixer.c: Probe of element 'Headphone' succeeded (volume=3, switch=1, enumeration=0). ( 0.443| 0.000) D: [pulseaudio] alsa-mixer.c: Probe of element 'Headphone2' succeeded (volume=0, switch=0, enumeration=0). ( 0.443| 0.000) D: [pulseaudio] alsa-mixer.c: Probe of element 'Speaker' succeeded (volume=0, switch=0, enumeration=0). ( 0.443| 0.000) D: [pulseaudio] alsa-mixer.c: Probe of element 'Desktop Speaker' succeeded (volume=0, switch=0, enumeration=0). ( 0.443| 0.000) D: [pulseaudio] alsa-mixer.c: Probe of element 'Front' succeeded (volume=1, switch=1, enumeration=0). ( 0.443| 0.000) D: [pulseaudio] alsa-mixer.c: Probe of element 'Rear' succeeded (volume=0, switch=0, enumeration=0). ( 0.443| 0.000) D: [pulseaudio] alsa-mixer.c: Probe of element 'Surround' succeeded (volume=1, switch=1, enumeration=0). ( 0.443| 0.000) D: [pulseaudio] alsa-mixer.c: Probe of element 'Side' succeeded (volume=1, switch=1, enumeration=0). ( 0.443| 0.000) D: [pulseaudio] alsa-mixer.c: Probe of element 'Center' succeeded (volume=1, switch=1, enumeration=0). ( 0.443| 0.000) D: [pulseaudio] alsa-mixer.c: Probe of element 'LFE' succeeded (volume=1, switch=1, enumeration=0). ( 0.443| 0.000) D: [pulseaudio] alsa-mixer.c: Probe of element 'CLFE' succeeded (volume=0, switch=0, enumeration=0). ( 0.443| 0.000) D: [pulseaudio] alsa-mixer.c: Probe of element 'PCM' succeeded (volume=1, switch=0, enumeration=0). ( 0.443| 0.000) D: [pulseaudio] alsa-mixer.c: Probe of element 'External Amplifier' succeeded (volume=0, switch=0, enumeration=0). ( 0.443| 0.000) D: [pulseaudio] alsa-mixer.c: Probe of element 'Bass Boost' succeeded (volume=0, switch=0, enumeration=0). ( 0.443| 0.000) D: [pulseaudio] alsa-mixer.c: Probe of element 'IEC958' succeeded (volume=0, switch=2, enumeration=0). ( 0.443| 0.000) D: [pulseaudio] alsa-mixer.c: Probe of element 'IEC958 Optical Raw' succeeded (volume=0, switch=0, enumeration=0). ( 0.443| 0.000) D: [pulseaudio] alsa-mixer.c: Probe of element 'Analog Output' succeeded (volume=0, switch=0, enumeration=0). ( 0.443| 0.000) D: [pulseaudio] alsa-mixer.c: Probing path 'analog-output-speaker' ( 0.443| 0.000) D: [pulseaudio] alsa-mixer.c: Probe of jack 'Headphone Jack' succeeded (not found) ( 0.443| 0.000) D: [pulseaudio] alsa-mixer.c: Probe of jack 'Front Headphone Jack' succeeded (found!) ( 0.443| 0.000) D: [pulseaudio] alsa-mixer.c: Probe of jack 'Speaker Phantom Jack' succeeded (not found) ( 0.443| 0.000) D: [pulseaudio] alsa-mixer.c: Probe of element 'Hardware Master' succeeded (volume=0, switch=0, enumeration=0). ( 0.443| 0.000) D: [pulseaudio] alsa-mixer.c: Probe of element 'Master' succeeded (volume=1, switch=1, enumeration=0). ( 0.443| 0.000) D: [pulseaudio] alsa-mixer.c: Probe of element 'Master Front' succeeded (volume=0, switch=0, enumeration=0). ( 0.443| 0.000) D: [pulseaudio] alsa-mixer.c: Probe of element 'Master Mono' succeeded (volume=0, switch=0, enumeration=0). ( 0.443| 0.000) D: [pulseaudio] alsa-mixer.c: Probe of element 'Headphone' succeeded (volume=3, switch=1, enumeration=0). ( 0.443| 0.000) D: [pulseaudio] alsa-mixer.c: Probe of element 'Headphone2' succeeded (volume=0, switch=0, enumeration=0). ( 0.443| 0.000) D: [pulseaudio] alsa-mixer.c: Probe of element 'Speaker' succeeded (volume=0, switch=0, enumeration=0). ( 0.443| 0.000) D: [pulseaudio] alsa-mixer.c: Probe of element 'Desktop Speaker' succeeded (volume=0, switch=0, enumeration=0). ( 0.443| 0.000) D: [pulseaudio] alsa-mixer.c: Probe of element 'Front' succeeded (volume=1, switch=1, enumeration=0). ( 0.443| 0.000) D: [pulseaudio] alsa-mixer.c: Probe of element 'Front Speaker' succeeded (volume=0, switch=0, enumeration=0). ( 0.443| 0.000) D: [pulseaudio] alsa-mixer.c: Probe of element 'Rear' succeeded (volume=0, switch=0, enumeration=0). ( 0.443| 0.000) D: [pulseaudio] alsa-mixer.c: Probe of element 'Surround' succeeded (volume=1, switch=1, enumeration=0). ( 0.443| 0.000) D: [pulseaudio] alsa-mixer.c: Probe of element 'Surround Speaker' succeeded (volume=0, switch=0, enumeration=0). ( 0.443| 0.000) D: [pulseaudio] alsa-mixer.c: Probe of element 'Side' succeeded (volume=1, switch=1, enumeration=0). ( 0.443| 0.000) D: [pulseaudio] alsa-mixer.c: Probe of element 'Center' succeeded (volume=1, switch=1, enumeration=0). ( 0.443| 0.000) D: [pulseaudio] alsa-mixer.c: Probe of element 'Center Speaker' succeeded (volume=0, switch=0, enumeration=0). ( 0.443| 0.000) D: [pulseaudio] alsa-mixer.c: Probe of element 'LFE' succeeded (volume=1, switch=1, enumeration=0). ( 0.443| 0.000) D: [pulseaudio] alsa-mixer.c: Probe of element 'LFE Speaker' succeeded (volume=0, switch=0, enumeration=0). ( 0.443| 0.000) D: [pulseaudio] alsa-mixer.c: Probe of element 'Bass Speaker' succeeded (volume=0, switch=0, enumeration=0). ( 0.443| 0.000) D: [pulseaudio] alsa-mixer.c: Probe of element 'CLFE' succeeded (volume=0, switch=0, enumeration=0). ( 0.443| 0.000) D: [pulseaudio] alsa-mixer.c: Probe of element 'PCM' succeeded (volume=1, switch=0, enumeration=0). ( 0.443| 0.000) D: [pulseaudio] alsa-mixer.c: Probe of element 'External Amplifier' succeeded (volume=0, switch=0, enumeration=0). ( 0.443| 0.000) D: [pulseaudio] alsa-mixer.c: Probe of element 'Bass Boost' succeeded (volume=0, switch=0, enumeration=0). ( 0.443| 0.000) D: [pulseaudio] alsa-mixer.c: Probe of element 'IEC958' succeeded (volume=0, switch=2, enumeration=0). ( 0.443| 0.000) D: [pulseaudio] alsa-mixer.c: Probe of element 'IEC958 Optical Raw' succeeded (volume=0, switch=0, enumeration=0). ( 0.443| 0.000) D: [pulseaudio] alsa-mixer.c: Probe of element 'Analog Output' succeeded (volume=0, switch=0, enumeration=0). ( 0.443| 0.000) D: [pulseaudio] alsa-mixer.c: Skipping path 'analog-output-speaker', none of required-any elements preset. ( 0.443| 0.000) D: [pulseaudio] alsa-mixer.c: Probing path 'analog-output-speaker' ( 0.443| 0.000) D: [pulseaudio] alsa-mixer.c: Probe of element 'Hardware Master' succeeded (volume=0, switch=0, enumeration=0). ( 0.443| 0.000) D: [pulseaudio] alsa-mixer.c: Probe of element 'Master' succeeded (volume=1, switch=1, enumeration=0). ( 0.443| 0.000) D: [pulseaudio] alsa-mixer.c: Probe of element 'Master Mono' succeeded (volume=0, switch=0, enumeration=0). ( 0.443| 0.000) D: [pulseaudio] alsa-mixer.c: Probe of element 'Headphone' succeeded (volume=3, switch=1, enumeration=0). ( 0.443| 0.000) D: [pulseaudio] alsa-mixer.c: Probe of element 'Headphone2' succeeded (volume=0, switch=0, enumeration=0). ( 0.443| 0.000) D: [pulseaudio] alsa-mixer.c: Probe of element 'Speaker' succeeded (volume=0, switch=0, enumeration=0). ( 0.444| 0.000) D: [pulseaudio] alsa-mixer.c: Probe of element 'Desktop Speaker' failed. ( 0.444| 0.000) D: [pulseaudio] alsa-mixer.c: Probing path 'analog-output-headphones' ( 0.444| 0.000) D: [pulseaudio] alsa-mixer.c: Probe of jack 'Front Headphone Jack' succeeded (found!) ( 0.444| 0.000) D: [pulseaudio] alsa-mixer.c: Probe of jack 'Headphone Jack' succeeded (not found) ( 0.444| 0.000) D: [pulseaudio] alsa-mixer.c: Probe of jack 'Headphone Mic Jack' succeeded (not found) ( 0.444| 0.000) D: [pulseaudio] alsa-mixer.c: Probe of element 'Hardware Master' succeeded (volume=0, switch=0, enumeration=0). ( 0.444| 0.000) D: [pulseaudio] alsa-mixer.c: Probe of element 'Master' succeeded (volume=1, switch=1, enumeration=0). ( 0.444| 0.000) D: [pulseaudio] alsa-mixer.c: Probe of element 'Master Front' succeeded (volume=0, switch=0, enumeration=0). ( 0.444| 0.000) D: [pulseaudio] alsa-mixer.c: Probe of element 'Master Mono' succeeded (volume=0, switch=0, enumeration=0). ( 0.444| 0.000) D: [pulseaudio] alsa-mixer.c: Probe of element 'Headphone' succeeded (volume=1, switch=1, enumeration=0). ( 0.444| 0.000) D: [pulseaudio] alsa-mixer.c: Probe of element 'Line HP Swap' succeeded (volume=0, switch=0, enumeration=0). ( 0.444| 0.000) D: [pulseaudio] alsa-mixer.c: Probe of element 'Headphone2' succeeded (volume=0, switch=0, enumeration=0). ( 0.444| 0.000) D: [pulseaudio] alsa-mixer.c: Probe of element 'Speaker' succeeded (volume=0, switch=0, enumeration=0). ( 0.444| 0.000) D: [pulseaudio] alsa-mixer.c: Probe of element 'Desktop Speaker' succeeded (volume=0, switch=0, enumeration=0). ( 0.444| 0.000) D: [pulseaudio] alsa-mixer.c: Probe of element 'Front' succeeded (volume=3, switch=1, enumeration=0). ( 0.444| 0.000) D: [pulseaudio] alsa-mixer.c: Probe of element 'Rear' succeeded (volume=0, switch=0, enumeration=0). ( 0.444| 0.000) D: [pulseaudio] alsa-mixer.c: Probe of element 'Surround' succeeded (volume=2, switch=2, enumeration=0). ( 0.444| 0.000) D: [pulseaudio] alsa-mixer.c: Probe of element 'Side' succeeded (volume=2, switch=2, enumeration=0). ( 0.444| 0.000) D: [pulseaudio] alsa-mixer.c: Probe of element 'Center' succeeded (volume=2, switch=2, enumeration=0). ( 0.444| 0.000) D: [pulseaudio] alsa-mixer.c: Probe of element 'LFE' succeeded (volume=2, switch=2, enumeration=0). ( 0.444| 0.000) D: [pulseaudio] alsa-mixer.c: Probe of element 'PCM' succeeded (volume=1, switch=0, enumeration=0). ( 0.444| 0.000) D: [pulseaudio] alsa-mixer.c: Probe of element 'External Amplifier' succeeded (volume=0, switch=0, enumeration=0). ( 0.444| 0.000) D: [pulseaudio] alsa-mixer.c: Probe of element 'Bass Boost' succeeded (volume=0, switch=0, enumeration=0). ( 0.444| 0.000) D: [pulseaudio] alsa-mixer.c: Probe of element 'IEC958' succeeded (volume=0, switch=2, enumeration=0). ( 0.444| 0.000) D: [pulseaudio] alsa-mixer.c: Probe of element 'IEC958 Optical Raw' succeeded (volume=0, switch=0, enumeration=0). ( 0.444| 0.000) D: [pulseaudio] alsa-mixer.c: Probe of element 'Analog Output' succeeded (volume=0, switch=0, enumeration=0). ( 0.444| 0.000) D: [pulseaudio] alsa-mixer.c: Probing path 'analog-output-headphones' ( 0.444| 0.000) D: [pulseaudio] alsa-mixer.c: Probe of element 'Hardware Master' succeeded (volume=0, switch=0, enumeration=0). ( 0.444| 0.000) D: [pulseaudio] alsa-mixer.c: Probe of element 'Master' succeeded (volume=1, switch=1, enumeration=0). ( 0.444| 0.000) D: [pulseaudio] alsa-mixer.c: Probe of element 'Master Front' succeeded (volume=0, switch=0, enumeration=0). ( 0.444| 0.000) D: [pulseaudio] alsa-mixer.c: Probe of element 'Master Mono' succeeded (volume=0, switch=0, enumeration=0). ( 0.444| 0.000) D: [pulseaudio] alsa-mixer.c: Probe of element 'Headphone' succeeded (volume=3, switch=1, enumeration=0). ( 0.444| 0.000) D: [pulseaudio] alsa-mixer.c: Probe of element 'Headphone2' failed. ( 0.444| 0.000) D: [pulseaudio] alsa-mixer.c: Probing path 'analog-output-mono' ( 0.444| 0.000) D: [pulseaudio] alsa-mixer.c: Probe of element 'Hardware Master' succeeded (volume=0, switch=0, enumeration=0). ( 0.444| 0.000) D: [pulseaudio] alsa-mixer.c: Probe of element 'Master' succeeded (volume=2, switch=2, enumeration=0). ( 0.444| 0.000) D: [pulseaudio] alsa-mixer.c: Probe of element 'Master Front' succeeded (volume=0, switch=0, enumeration=0). ( 0.444| 0.000) D: [pulseaudio] alsa-mixer.c: Probe of element 'Master Mono' failed. ( 0.444| 0.000) D: [pulseaudio] alsa-mixer.c: Available mixer paths (after tidying): ( 0.444| 0.000) D: [pulseaudio] alsa-mixer.c: Path Set 0xc72640, direction=1 ( 0.444| 0.000) D: [pulseaudio] alsa-mixer.c: Path analog-output (Analog Output), direction=1, priority=99, probed=yes, supported=yes, has_mute=yes, has_volume=yes, has_dB=yes, min_volume=0, max_volume=64, min_dB=-177, max_dB=2 ( 0.444| 0.000) D: [pulseaudio] alsa-mixer.c: Element Master, direction=1, switch=1, volume=1, volume_limit=-1, enumeration=0, required=0, required_any=0, required_absent=0, mask=0x7ffffffffffff, n_channels=1, override_map=yes ( 0.444| 0.000) D: [pulseaudio] alsa-mixer.c: Element Headphone, direction=1, switch=1, volume=3, volume_limit=-1, enumeration=0, required=0, required_any=0, required_absent=0, mask=0x6, n_channels=2, override_map=no ( 0.444| 0.000) D: [pulseaudio] alsa-mixer.c: Element Front, direction=1, switch=1, volume=1, volume_limit=-1, enumeration=0, required=0, required_any=0, required_absent=0, mask=0x6, n_channels=2, override_map=yes ( 0.444| 0.000) D: [pulseaudio] alsa-mixer.c: Element Surround, direction=1, switch=1, volume=1, volume_limit=-1, enumeration=0, required=0, required_any=0, required_absent=0, mask=0x60, n_channels=2, override_map=yes ( 0.444| 0.000) D: [pulseaudio] alsa-mixer.c: Element Side, direction=1, switch=1, volume=1, volume_limit=-1, enumeration=0, required=0, required_any=0, required_absent=0, mask=0xc00, n_channels=2, override_map=yes ( 0.444| 0.000) D: [pulseaudio] alsa-mixer.c: Element Center, direction=1, switch=1, volume=1, volume_limit=-1, enumeration=0, required=0, required_any=0, required_absent=0, mask=0x4900000000018, n_channels=1, override_map=yes ( 0.444| 0.000) D: [pulseaudio] alsa-mixer.c: Element LFE, direction=1, switch=1, volume=1, volume_limit=-1, enumeration=0, required=0, required_any=0, required_absent=0, mask=0x80, n_channels=1, override_map=yes ( 0.444| 0.000) D: [pulseaudio] alsa-mixer.c: Element PCM, direction=1, switch=0, volume=1, volume_limit=-1, enumeration=0, required=0, required_any=0, required_absent=0, mask=0x3600000000f66, n_channels=2, override_map=yes ( 0.444| 0.000) D: [pulseaudio] alsa-mixer.c: Element IEC958, direction=1, switch=2, volume=0, volume_limit=-1, enumeration=0, required=0, required_any=0, required_absent=0, mask=0x0, n_channels=0, override_map=no ( 0.444| 0.000) D: [pulseaudio] alsa-mixer.c: Path analog-output-headphones (Headphones), direction=1, priority=90, probed=yes, supported=yes, has_mute=yes, has_volume=yes, has_dB=yes, min_volume=0, max_volume=64, min_dB=-177, max_dB=2 ( 0.444| 0.000) D: [pulseaudio] alsa-mixer.c: Element Master, direction=1, switch=1, volume=1, volume_limit=-1, enumeration=0, required=0, required_any=0, required_absent=0, mask=0x7ffffffffffff, n_channels=1, override_map=yes ( 0.444| 0.000) D: [pulseaudio] alsa-mixer.c: Element Headphone, direction=1, switch=1, volume=1, volume_limit=-1, enumeration=0, required=0, required_any=4, required_absent=0, mask=0x3600000000f66, n_channels=2, override_map=yes ( 0.444| 0.000) D: [pulseaudio] alsa-mixer.c: Element Front, direction=1, switch=1, volume=3, volume_limit=-1, enumeration=0, required=0, required_any=0, required_absent=0, mask=0x6, n_channels=2, override_map=no ( 0.444| 0.000) D: [pulseaudio] alsa-mixer.c: Element Surround, direction=1, switch=2, volume=2, volume_limit=-1, enumeration=0, required=0, required_any=0, required_absent=0, mask=0x6, n_channels=2, override_map=no ( 0.444| 0.000) D: [pulseaudio] alsa-mixer.c: Element Side, direction=1, switch=2, volume=2, volume_limit=-1, enumeration=0, required=0, required_any=0, required_absent=0, mask=0x6, n_channels=2, override_map=no ( 0.444| 0.000) D: [pulseaudio] alsa-mixer.c: Element Center, direction=1, switch=2, volume=2, volume_limit=-1, enumeration=0, required=0, required_any=0, required_absent=0, mask=0x7ffffffffffff, n_channels=1, override_map=no ( 0.444| 0.000) D: [pulseaudio] alsa-mixer.c: Element LFE, direction=1, switch=2, volume=2, volume_limit=-1, enumeration=0, required=0, required_any=0, required_absent=0, mask=0x7ffffffffffff, n_channels=1, override_map=no ( 0.444| 0.000) D: [pulseaudio] alsa-mixer.c: Element PCM, direction=1, switch=0, volume=1, volume_limit=-1, enumeration=0, required=0, required_any=0, required_absent=0, mask=0x3600000000f66, n_channels=2, override_map=yes ( 0.444| 0.000) D: [pulseaudio] alsa-mixer.c: Element IEC958, direction=1, switch=2, volume=0, volume_limit=-1, enumeration=0, required=0, required_any=0, required_absent=0, mask=0x0, n_channels=0, override_map=no ( 0.444| 0.000) D: [pulseaudio] alsa-mixer.c: Jack Front Headphone, alsa_name='Front Headphone Jack', detection possible ( 0.444| 0.000) D: [pulseaudio] alsa-mixer.c: Jack Headphone, alsa_name='Headphone Jack', detection unavailable ( 0.444| 0.000) D: [pulseaudio] alsa-mixer.c: Jack Headphone Mic, alsa_name='Headphone Mic Jack', detection unavailable ( 0.444| 0.000) D: [pulseaudio] alsa-mixer.c: Skipping profile output:analog-stereo+input:analog-mono - will not be able to open input:analog-mono ( 0.444| 0.000) D: [pulseaudio] alsa-mixer.c: Looking at profile output:analog-stereo+input:analog-stereo ( 0.444| 0.000) D: [pulseaudio] alsa-mixer.c: Checking for recording on Analog Stereo (analog-stereo) ( 0.444| 0.000) D: [pulseaudio] alsa-util.c: Trying front:0 with SND_PCM_NO_AUTO_FORMAT ... ( 0.444| 0.000) D: [pulseaudio] alsa-util.c: Managed to open front:0 ( 0.444| 0.000) D: [pulseaudio] alsa-util.c: Maximum hw buffer size is 371 ms ( 0.444| 0.000) D: [pulseaudio] alsa-util.c: Set buffer size first (to 2028 samples), period size second (to 1014 samples). ( 0.444| 0.000) D: [pulseaudio] alsa-mixer.c: Profile output:analog-stereo+input:analog-stereo supported. ( 0.444| 0.000) D: [pulseaudio] alsa-mixer.c: Skipping profile output:analog-stereo+input:iec958-stereo - will not be able to open input:iec958-stereo ( 0.444| 0.000) D: [pulseaudio] alsa-mixer.c: Looking at profile output:analog-surround-40 ( 0.444| 0.000) D: [pulseaudio] alsa-mixer.c: Checking for playback on Analog Surround 4.0 (analog-surround-40) ( 0.444| 0.000) D: [pulseaudio] alsa-util.c: Trying surround40:0 with SND_PCM_NO_AUTO_FORMAT ... ( 0.445| 0.000) D: [pulseaudio] alsa-util.c: Managed to open surround40:0 ( 0.445| 0.000) D: [pulseaudio] alsa-util.c: Maximum hw buffer size is 185 ms ( 2.493| 2.048) D: [pulseaudio] alsa-util.c: Set buffer size first (to 2028 samples), period size second (to 1014 samples). ( 2.493| 0.000) D: [pulseaudio] alsa-mixer.c: Profile output:analog-surround-40 supported. ( 2.493| 0.000) I: [pulseaudio] (alsa-lib)control.c: Invalid CTL surround40:0 ( 2.493| 0.000) I: [pulseaudio] alsa-util.c: Unable to attach to mixer surround40:0: No such file or directory ( 2.494| 0.000) I: [pulseaudio] alsa-util.c: Successfully attached to mixer 'hw:0' ( 2.494| 0.000) D: [pulseaudio] alsa-mixer.c: Available mixer paths (after tidying): ( 2.494| 0.000) D: [pulseaudio] alsa-mixer.c: Path Set 0xc30dd0, direction=1 ( 2.494| 0.000) D: [pulseaudio] alsa-mixer.c: Path analog-output (Analog Output), direction=1, priority=99, probed=yes, supported=yes, has_mute=yes, has_volume=yes, has_dB=yes, min_volume=0, max_volume=64, min_dB=-177, max_dB=2 ( 2.494| 0.000) D: [pulseaudio] alsa-mixer.c: Element Master, direction=1, switch=1, volume=1, volume_limit=-1, enumeration=0, required=0, required_any=0, required_absent=0, mask=0x7ffffffffffff, n_channels=1, override_map=yes ( 2.494| 0.000) D: [pulseaudio] alsa-mixer.c: Element Headphone, direction=1, switch=1, volume=3, volume_limit=-1, enumeration=0, required=0, required_any=0, required_absent=0, mask=0x6, n_channels=2, override_map=no ( 2.494| 0.000) D: [pulseaudio] alsa-mixer.c: Element Front, direction=1, switch=1, volume=1, volume_limit=-1, enumeration=0, required=0, required_any=0, required_absent=0, mask=0x6, n_channels=2, override_map=yes ( 2.494| 0.000) D: [pulseaudio] alsa-mixer.c: Element Surround, direction=1, switch=1, volume=1, volume_limit=-1, enumeration=0, required=0, required_any=0, required_absent=0, mask=0x60, n_channels=2, override_map=yes ( 2.494| 0.000) D: [pulseaudio] alsa-mixer.c: Element Side, direction=1, switch=1, volume=1, volume_limit=-1, enumeration=0, required=0, required_any=0, required_absent=0, mask=0xc00, n_channels=2, override_map=yes ( 2.494| 0.000) D: [pulseaudio] alsa-mixer.c: Element Center, direction=1, switch=1, volume=1, volume_limit=-1, enumeration=0, required=0, required_any=0, required_absent=0, mask=0x4900000000018, n_channels=1, override_map=yes ( 2.494| 0.000) D: [pulseaudio] alsa-mixer.c: Element LFE, direction=1, switch=1, volume=1, volume_limit=-1, enumeration=0, required=0, required_any=0, required_absent=0, mask=0x80, n_channels=1, override_map=yes ( 2.494| 0.000) D: [pulseaudio] alsa-mixer.c: Element PCM, direction=1, switch=0, volume=1, volume_limit=-1, enumeration=0, required=0, required_any=0, required_absent=0, mask=0x3600000000f66, n_channels=2, override_map=yes ( 2.494| 0.000) D: [pulseaudio] alsa-mixer.c: Element IEC958, direction=1, switch=2, volume=0, volume_limit=-1, enumeration=0, required=0, required_any=0, required_absent=0, mask=0x0, n_channels=0, override_map=no ( 2.494| 0.000) D: [pulseaudio] alsa-mixer.c: Skipping profile output:analog-surround-40+input:analog-mono - will not be able to open input:analog-mono ( 2.494| 0.000) D: [pulseaudio] alsa-mixer.c: Looking at profile output:analog-surround-40+input:analog-stereo ( 2.494| 0.000) D: [pulseaudio] alsa-mixer.c: Checking for recording on Analog Stereo (analog-stereo) ( 2.494| 0.000) D: [pulseaudio] alsa-util.c: Trying front:0 with SND_PCM_NO_AUTO_FORMAT ... ( 2.494| 0.000) D: [pulseaudio] alsa-util.c: Managed to open front:0 ( 2.494| 0.000) D: [pulseaudio] alsa-util.c: Maximum hw buffer size is 371 ms ( 2.494| 0.000) D: [pulseaudio] alsa-util.c: Set buffer size first (to 2028 samples), period size second (to 1014 samples). ( 2.494| 0.000) D: [pulseaudio] alsa-mixer.c: Profile output:analog-surround-40+input:analog-stereo supported. ( 2.494| 0.000) D: [pulseaudio] alsa-mixer.c: Skipping profile output:analog-surround-40+input:iec958-stereo - will not be able to open input:iec958-stereo ( 2.494| 0.000) D: [pulseaudio] alsa-mixer.c: Looking at profile output:analog-surround-41 ( 2.494| 0.000) D: [pulseaudio] alsa-mixer.c: Checking for playback on Analog Surround 4.1 (analog-surround-41) ( 2.494| 0.000) D: [pulseaudio] alsa-util.c: Trying surround41:0 with SND_PCM_NO_AUTO_FORMAT ... ( 2.494| 0.000) D: [pulseaudio] alsa-util.c: Managed to open surround41:0 ( 2.495| 0.000) D: [pulseaudio] alsa-util.c: Maximum hw buffer size is 123 ms ( 6.549| 4.054) D: [pulseaudio] alsa-util.c: Set buffer size first (to 2028 samples), period size second (to 1014 samples). ( 6.549| 0.000) D: [pulseaudio] alsa-mixer.c: Profile output:analog-surround-41 supported. ( 6.549| 0.000) I: [pulseaudio] (alsa-lib)control.c: Invalid CTL surround41:0 ( 6.549| 0.000) I: [pulseaudio] alsa-util.c: Unable to attach to mixer surround41:0: No such file or directory ( 6.550| 0.000) I: [pulseaudio] alsa-util.c: Successfully attached to mixer 'hw:0' ( 6.550| 0.000) D: [pulseaudio] alsa-mixer.c: Available mixer paths (after tidying): ( 6.550| 0.000) D: [pulseaudio] alsa-mixer.c: Path Set 0xd28070, direction=1 ( 6.550| 0.000) D: [pulseaudio] alsa-mixer.c: Path analog-output (Analog Output), direction=1, priority=99, probed=yes, supported=yes, has_mute=yes, has_volume=yes, has_dB=yes, min_volume=0, max_volume=64, min_dB=-177, max_dB=2 ( 6.550| 0.000) D: [pulseaudio] alsa-mixer.c: Element Master, direction=1, switch=1, volume=1, volume_limit=-1, enumeration=0, required=0, required_any=0, required_absent=0, mask=0x7ffffffffffff, n_channels=1, override_map=yes ( 6.550| 0.000) D: [pulseaudio] alsa-mixer.c: Element Headphone, direction=1, switch=1, volume=3, volume_limit=-1, enumeration=0, required=0, required_any=0, required_absent=0, mask=0x6, n_channels=2, override_map=no ( 6.550| 0.000) D: [pulseaudio] alsa-mixer.c: Element Front, direction=1, switch=1, volume=1, volume_limit=-1, enumeration=0, required=0, required_any=0, required_absent=0, mask=0x6, n_channels=2, override_map=yes ( 6.550| 0.000) D: [pulseaudio] alsa-mixer.c: Element Surround, direction=1, switch=1, volume=1, volume_limit=-1, enumeration=0, required=0, required_any=0, required_absent=0, mask=0x60, n_channels=2, override_map=yes ( 6.550| 0.000) D: [pulseaudio] alsa-mixer.c: Element Side, direction=1, switch=1, volume=1, volume_limit=-1, enumeration=0, required=0, required_any=0, required_absent=0, mask=0xc00, n_channels=2, override_map=yes ( 6.550| 0.000) D: [pulseaudio] alsa-mixer.c: Element Center, direction=1, switch=1, volume=1, volume_limit=-1, enumeration=0, required=0, required_any=0, required_absent=0, mask=0x4900000000018, n_channels=1, override_map=yes ( 6.550| 0.000) D: [pulseaudio] alsa-mixer.c: Element LFE, direction=1, switch=1, volume=1, volume_limit=-1, enumeration=0, required=0, required_any=0, required_absent=0, mask=0x80, n_channels=1, override_map=yes ( 6.550| 0.000) D: [pulseaudio] alsa-mixer.c: Element PCM, direction=1, switch=0, volume=1, volume_limit=-1, enumeration=0, required=0, required_any=0, required_absent=0, mask=0x3600000000f66, n_channels=2, override_map=yes ( 6.550| 0.000) D: [pulseaudio] alsa-mixer.c: Element IEC958, direction=1, switch=2, volume=0, volume_limit=-1, enumeration=0, required=0, required_any=0, required_absent=0, mask=0x0, n_channels=0, override_map=no ( 6.550| 0.000) D: [pulseaudio] alsa-mixer.c: Skipping profile output:analog-surround-41+input:analog-mono - will not be able to open input:analog-mono ( 6.550| 0.000) D: [pulseaudio] alsa-mixer.c: Looking at profile output:analog-surround-41+input:analog-stereo ( 6.550| 0.000) D: [pulseaudio] alsa-mixer.c: Checking for recording on Analog Stereo (analog-stereo) ( 6.550| 0.000) D: [pulseaudio] alsa-util.c: Trying front:0 with SND_PCM_NO_AUTO_FORMAT ... ( 6.550| 0.000) D: [pulseaudio] alsa-util.c: Managed to open front:0 ( 6.550| 0.000) D: [pulseaudio] alsa-util.c: Maximum hw buffer size is 371 ms ( 6.550| 0.000) D: [pulseaudio] alsa-util.c: Set buffer size first (to 2028 samples), period size second (to 1014 samples). ( 6.550| 0.000) D: [pulseaudio] alsa-mixer.c: Profile output:analog-surround-41+input:analog-stereo supported. ( 6.550| 0.000) D: [pulseaudio] alsa-mixer.c: Skipping profile output:analog-surround-41+input:iec958-stereo - will not be able to open input:iec958-stereo ( 6.550| 0.000) D: [pulseaudio] alsa-mixer.c: Looking at profile output:analog-surround-50 ( 6.550| 0.000) D: [pulseaudio] alsa-mixer.c: Checking for playback on Analog Surround 5.0 (analog-surround-50) ( 6.550| 0.000) D: [pulseaudio] alsa-util.c: Trying surround50:0 with SND_PCM_NO_AUTO_FORMAT ... ( 6.550| 0.000) D: [pulseaudio] alsa-util.c: Managed to open surround50:0 ( 6.551| 0.000) D: [pulseaudio] alsa-util.c: Maximum hw buffer size is 123 ms ( 6.551| 0.000) D: [pulseaudio] alsa-util.c: Set buffer size first (to 2028 samples), period size second (to 1014 samples). ( 6.551| 0.000) D: [pulseaudio] alsa-mixer.c: Profile output:analog-surround-50 supported. ( 6.551| 0.000) I: [pulseaudio] (alsa-lib)control.c: Invalid CTL surround50:0 ( 6.551| 0.000) I: [pulseaudio] alsa-util.c: Unable to attach to mixer surround50:0: No such file or directory ( 6.551| 0.000) I: [pulseaudio] alsa-util.c: Successfully attached to mixer 'hw:0' ( 6.551| 0.000) D: [pulseaudio] alsa-mixer.c: Available mixer paths (after tidying): ( 6.551| 0.000) D: [pulseaudio] alsa-mixer.c: Path Set 0xcf8270, direction=1 ( 6.551| 0.000) D: [pulseaudio] alsa-mixer.c: Path analog-output (Analog Output), direction=1, priority=99, probed=yes, supported=yes, has_mute=yes, has_volume=yes, has_dB=yes, min_volume=0, max_volume=64, min_dB=-177, max_dB=2 ( 6.551| 0.000) D: [pulseaudio] alsa-mixer.c: Element Master, direction=1, switch=1, volume=1, volume_limit=-1, enumeration=0, required=0, required_any=0, required_absent=0, mask=0x7ffffffffffff, n_channels=1, override_map=yes ( 6.551| 0.000) D: [pulseaudio] alsa-mixer.c: Element Headphone, direction=1, switch=1, volume=3, volume_limit=-1, enumeration=0, required=0, required_any=0, required_absent=0, mask=0x6, n_channels=2, override_map=no ( 6.551| 0.000) D: [pulseaudio] alsa-mixer.c: Element Front, direction=1, switch=1, volume=1, volume_limit=-1, enumeration=0, required=0, required_any=0, required_absent=0, mask=0x6, n_channels=2, override_map=yes ( 6.551| 0.000) D: [pulseaudio] alsa-mixer.c: Element Surround, direction=1, switch=1, volume=1, volume_limit=-1, enumeration=0, required=0, required_any=0, required_absent=0, mask=0x60, n_channels=2, override_map=yes ( 6.551| 0.000) D: [pulseaudio] alsa-mixer.c: Element Side, direction=1, switch=1, volume=1, volume_limit=-1, enumeration=0, required=0, required_any=0, required_absent=0, mask=0xc00, n_channels=2, override_map=yes ( 6.551| 0.000) D: [pulseaudio] alsa-mixer.c: Element Center, direction=1, switch=1, volume=1, volume_limit=-1, enumeration=0, required=0, required_any=0, required_absent=0, mask=0x4900000000018, n_channels=1, override_map=yes ( 6.551| 0.000) D: [pulseaudio] alsa-mixer.c: Element LFE, direction=1, switch=1, volume=1, volume_limit=-1, enumeration=0, required=0, required_any=0, required_absent=0, mask=0x80, n_channels=1, override_map=yes ( 6.551| 0.000) D: [pulseaudio] alsa-mixer.c: Element PCM, direction=1, switch=0, volume=1, volume_limit=-1, enumeration=0, required=0, required_any=0, required_absent=0, mask=0x3600000000f66, n_channels=2, override_map=yes ( 6.551| 0.000) D: [pulseaudio] alsa-mixer.c: Element IEC958, direction=1, switch=2, volume=0, volume_limit=-1, enumeration=0, required=0, required_any=0, required_absent=0, mask=0x0, n_channels=0, override_map=no ( 6.551| 0.000) D: [pulseaudio] alsa-mixer.c: Skipping profile output:analog-surround-50+input:analog-mono - will not be able to open input:analog-mono ( 6.551| 0.000) D: [pulseaudio] alsa-mixer.c: Looking at profile output:analog-surround-50+input:analog-stereo ( 6.551| 0.000) D: [pulseaudio] alsa-mixer.c: Checking for recording on Analog Stereo (analog-stereo) ( 6.551| 0.000) D: [pulseaudio] alsa-util.c: Trying front:0 with SND_PCM_NO_AUTO_FORMAT ... ( 6.551| 0.000) D: [pulseaudio] alsa-util.c: Managed to open front:0 ( 6.551| 0.000) D: [pulseaudio] alsa-util.c: Maximum hw buffer size is 371 ms ( 6.551| 0.000) D: [pulseaudio] alsa-util.c: Set buffer size first (to 2028 samples), period size second (to 1014 samples). ( 6.551| 0.000) D: [pulseaudio] alsa-mixer.c: Profile output:analog-surround-50+input:analog-stereo supported. ( 6.551| 0.000) D: [pulseaudio] alsa-mixer.c: Skipping profile output:analog-surround-50+input:iec958-stereo - will not be able to open input:iec958-stereo ( 6.551| 0.000) D: [pulseaudio] alsa-mixer.c: Looking at profile output:analog-surround-51 ( 6.551| 0.000) D: [pulseaudio] alsa-mixer.c: Checking for playback on Analog Surround 5.1 (analog-surround-51) ( 6.551| 0.000) D: [pulseaudio] alsa-util.c: Trying surround51:0 with SND_PCM_NO_AUTO_FORMAT ... ( 6.552| 0.000) D: [pulseaudio] alsa-util.c: Managed to open surround51:0 ( 6.552| 0.000) D: [pulseaudio] alsa-util.c: Maximum hw buffer size is 123 ms ( 6.552| 0.000) D: [pulseaudio] alsa-util.c: Set buffer size first (to 2028 samples), period size second (to 1014 samples). ( 6.552| 0.000) D: [pulseaudio] alsa-mixer.c: Profile output:analog-surround-51 supported. ( 6.552| 0.000) I: [pulseaudio] (alsa-lib)control.c: Invalid CTL surround51:0 ( 6.552| 0.000) I: [pulseaudio] alsa-util.c: Unable to attach to mixer surround51:0: No such file or directory ( 6.552| 0.000) I: [pulseaudio] alsa-util.c: Successfully attached to mixer 'hw:0' ( 6.552| 0.000) D: [pulseaudio] alsa-mixer.c: Available mixer paths (after tidying): ( 6.552| 0.000) D: [pulseaudio] alsa-mixer.c: Path Set 0xd20420, direction=1 ( 6.552| 0.000) D: [pulseaudio] alsa-mixer.c: Path analog-output (Analog Output), direction=1, priority=99, probed=yes, supported=yes, has_mute=yes, has_volume=yes, has_dB=yes, min_volume=0, max_volume=64, min_dB=-177, max_dB=2 ( 6.552| 0.000) D: [pulseaudio] alsa-mixer.c: Element Master, direction=1, switch=1, volume=1, volume_limit=-1, enumeration=0, required=0, required_any=0, required_absent=0, mask=0x7ffffffffffff, n_channels=1, override_map=yes ( 6.552| 0.000) D: [pulseaudio] alsa-mixer.c: Element Headphone, direction=1, switch=1, volume=3, volume_limit=-1, enumeration=0, required=0, required_any=0, required_absent=0, mask=0x6, n_channels=2, override_map=no ( 6.552| 0.000) D: [pulseaudio] alsa-mixer.c: Element Front, direction=1, switch=1, volume=1, volume_limit=-1, enumeration=0, required=0, required_any=0, required_absent=0, mask=0x6, n_channels=2, override_map=yes ( 6.552| 0.000) D: [pulseaudio] alsa-mixer.c: Element Surround, direction=1, switch=1, volume=1, volume_limit=-1, enumeration=0, required=0, required_any=0, required_absent=0, mask=0x60, n_channels=2, override_map=yes ( 6.552| 0.000) D: [pulseaudio] alsa-mixer.c: Element Side, direction=1, switch=1, volume=1, volume_limit=-1, enumeration=0, required=0, required_any=0, required_absent=0, mask=0xc00, n_channels=2, override_map=yes ( 6.552| 0.000) D: [pulseaudio] alsa-mixer.c: Element Center, direction=1, switch=1, volume=1, volume_limit=-1, enumeration=0, required=0, required_any=0, required_absent=0, mask=0x4900000000018, n_channels=1, override_map=yes ( 6.552| 0.000) D: [pulseaudio] alsa-mixer.c: Element LFE, direction=1, switch=1, volume=1, volume_limit=-1, enumeration=0, required=0, required_any=0, required_absent=0, mask=0x80, n_channels=1, override_map=yes ( 6.552| 0.000) D: [pulseaudio] alsa-mixer.c: Element PCM, direction=1, switch=0, volume=1, volume_limit=-1, enumeration=0, required=0, required_any=0, required_absent=0, mask=0x3600000000f66, n_channels=2, override_map=yes ( 6.552| 0.000) D: [pulseaudio] alsa-mixer.c: Element IEC958, direction=1, switch=2, volume=0, volume_limit=-1, enumeration=0, required=0, required_any=0, required_absent=0, mask=0x0, n_channels=0, override_map=no ( 6.552| 0.000) D: [pulseaudio] alsa-mixer.c: Skipping profile output:analog-surround-51+input:analog-mono - will not be able to open input:analog-mono ( 6.552| 0.000) D: [pulseaudio] alsa-mixer.c: Looking at profile output:analog-surround-51+input:analog-stereo ( 6.552| 0.000) D: [pulseaudio] alsa-mixer.c: Checking for recording on Analog Stereo (analog-stereo) ( 6.552| 0.000) D: [pulseaudio] alsa-util.c: Trying front:0 with SND_PCM_NO_AUTO_FORMAT ... ( 6.552| 0.000) D: [pulseaudio] alsa-util.c: Managed to open front:0 ( 6.552| 0.000) D: [pulseaudio] alsa-util.c: Maximum hw buffer size is 371 ms ( 6.552| 0.000) D: [pulseaudio] alsa-util.c: Set buffer size first (to 2028 samples), period size second (to 1014 samples). ( 6.552| 0.000) D: [pulseaudio] alsa-mixer.c: Profile output:analog-surround-51+input:analog-stereo supported. ( 6.552| 0.000) D: [pulseaudio] alsa-mixer.c: Skipping profile output:analog-surround-51+input:iec958-stereo - will not be able to open input:iec958-stereo ( 6.552| 0.000) D: [pulseaudio] alsa-mixer.c: Looking at profile output:analog-surround-71 ( 6.552| 0.000) D: [pulseaudio] alsa-mixer.c: Checking for playback on Analog Surround 7.1 (analog-surround-71) ( 6.552| 0.000) D: [pulseaudio] alsa-util.c: Trying surround71:0 with SND_PCM_NO_AUTO_FORMAT ... ( 6.553| 0.000) D: [pulseaudio] alsa-util.c: Managed to open surround71:0 ( 6.553| 0.000) D: [pulseaudio] alsa-util.c: Maximum hw buffer size is 92 ms ( 8.597| 2.044) D: [pulseaudio] alsa-util.c: Set buffer size first (to 2028 samples), period size second (to 1014 samples). ( 8.597| 0.000) D: [pulseaudio] alsa-mixer.c: Profile output:analog-surround-71 supported. ( 8.597| 0.000) I: [pulseaudio] (alsa-lib)control.c: Invalid CTL surround71:0 ( 8.597| 0.000) I: [pulseaudio] alsa-util.c: Unable to attach to mixer surround71:0: No such file or directory ( 8.598| 0.000) I: [pulseaudio] alsa-util.c: Successfully attached to mixer 'hw:0' ( 8.598| 0.000) D: [pulseaudio] alsa-mixer.c: Available mixer paths (after tidying): ( 8.598| 0.000) D: [pulseaudio] alsa-mixer.c: Path Set 0xc619a0, direction=1 ( 8.598| 0.000) D: [pulseaudio] alsa-mixer.c: Path analog-output (Analog Output), direction=1, priority=99, probed=yes, supported=yes, has_mute=yes, has_volume=yes, has_dB=yes, min_volume=0, max_volume=64, min_dB=-177, max_dB=2 ( 8.598| 0.000) D: [pulseaudio] alsa-mixer.c: Element Master, direction=1, switch=1, volume=1, volume_limit=-1, enumeration=0, required=0, required_any=0, required_absent=0, mask=0x7ffffffffffff, n_channels=1, override_map=yes ( 8.598| 0.000) D: [pulseaudio] alsa-mixer.c: Element Headphone, direction=1, switch=1, volume=3, volume_limit=-1, enumeration=0, required=0, required_any=0, required_absent=0, mask=0x6, n_channels=2, override_map=no ( 8.598| 0.000) D: [pulseaudio] alsa-mixer.c: Element Front, direction=1, switch=1, volume=1, volume_limit=-1, enumeration=0, required=0, required_any=0, required_absent=0, mask=0x6, n_channels=2, override_map=yes ( 8.598| 0.000) D: [pulseaudio] alsa-mixer.c: Element Surround, direction=1, switch=1, volume=1, volume_limit=-1, enumeration=0, required=0, required_any=0, required_absent=0, mask=0x60, n_channels=2, override_map=yes ( 8.598| 0.000) D: [pulseaudio] alsa-mixer.c: Element Side, direction=1, switch=1, volume=1, volume_limit=-1, enumeration=0, required=0, required_any=0, required_absent=0, mask=0xc00, n_channels=2, override_map=yes ( 8.598| 0.000) D: [pulseaudio] alsa-mixer.c: Element Center, direction=1, switch=1, volume=1, volume_limit=-1, enumeration=0, required=0, required_any=0, required_absent=0, mask=0x4900000000018, n_channels=1, override_map=yes ( 8.598| 0.000) D: [pulseaudio] alsa-mixer.c: Element LFE, direction=1, switch=1, volume=1, volume_limit=-1, enumeration=0, required=0, required_any=0, required_absent=0, mask=0x80, n_channels=1, override_map=yes ( 8.598| 0.000) D: [pulseaudio] alsa-mixer.c: Element PCM, direction=1, switch=0, volume=1, volume_limit=-1, enumeration=0, required=0, required_any=0, required_absent=0, mask=0x3600000000f66, n_channels=2, override_map=yes ( 8.598| 0.000) D: [pulseaudio] alsa-mixer.c: Element IEC958, direction=1, switch=2, volume=0, volume_limit=-1, enumeration=0, required=0, required_any=0, required_absent=0, mask=0x0, n_channels=0, override_map=no ( 8.598| 0.000) D: [pulseaudio] alsa-mixer.c: Skipping profile output:analog-surround-71+input:analog-mono - will not be able to open input:analog-mono ( 8.598| 0.000) D: [pulseaudio] alsa-mixer.c: Looking at profile output:analog-surround-71+input:analog-stereo ( 8.598| 0.000) D: [pulseaudio] alsa-mixer.c: Checking for recording on Analog Stereo (analog-stereo) ( 8.598| 0.000) D: [pulseaudio] alsa-util.c: Trying front:0 with SND_PCM_NO_AUTO_FORMAT ... ( 8.598| 0.000) D: [pulseaudio] alsa-util.c: Managed to open front:0 ( 8.598| 0.000) D: [pulseaudio] alsa-util.c: Maximum hw buffer size is 371 ms ( 8.598| 0.000) D: [pulseaudio] alsa-util.c: Set buffer size first (to 2028 samples), period size second (to 1014 samples). ( 8.598| 0.000) D: [pulseaudio] alsa-mixer.c: Profile output:analog-surround-71+input:analog-stereo supported. ( 8.598| 0.000) D: [pulseaudio] alsa-mixer.c: Skipping profile output:analog-surround-71+input:iec958-stereo - will not be able to open input:iec958-stereo ( 8.598| 0.000) D: [pulseaudio] alsa-mixer.c: Looking at profile output:iec958-stereo ( 8.598| 0.000) D: [pulseaudio] alsa-mixer.c: Checking for playback on Digital Stereo (IEC958) (iec958-stereo) ( 8.598| 0.000) D: [pulseaudio] alsa-util.c: Trying iec958:0 with SND_PCM_NO_AUTO_FORMAT ... ( 8.598| 0.000) D: [pulseaudio] alsa-util.c: Managed to open iec958:0 ( 8.599| 0.000) D: [pulseaudio] alsa-util.c: Maximum hw buffer size is 371 ms ( 9.617| 1.018) D: [pulseaudio] alsa-util.c: Set buffer size first (to 2028 samples), period size second (to 1014 samples). ( 9.618| 0.000) D: [pulseaudio] alsa-mixer.c: Profile output:iec958-stereo supported. ( 9.618| 0.000) I: [pulseaudio] (alsa-lib)control.c: Invalid CTL iec958:0 ( 9.618| 0.000) I: [pulseaudio] alsa-util.c: Unable to attach to mixer iec958:0: No such file or directory ( 9.618| 0.000) I: [pulseaudio] alsa-util.c: Successfully attached to mixer 'hw:0' ( 9.618| 0.000) D: [pulseaudio] alsa-mixer.c: Probing path 'iec958-stereo-output' ( 9.618| 0.000) D: [pulseaudio] alsa-mixer.c: Probe of element 'IEC958' succeeded (volume=0, switch=1, enumeration=0). ( 9.618| 0.000) D: [pulseaudio] alsa-mixer.c: Available mixer paths (after tidying): ( 9.618| 0.000) D: [pulseaudio] alsa-mixer.c: Path Set 0xce30e0, direction=1 ( 9.618| 0.000) D: [pulseaudio] alsa-mixer.c: Path iec958-stereo-output (Digital Output (S/PDIF)), direction=1, priority=0, probed=yes, supported=yes, has_mute=yes, has_volume=no, has_dB=no, min_volume=0, max_volume=0, min_dB=inf, max_dB=-inf ( 9.618| 0.000) D: [pulseaudio] alsa-mixer.c: Element IEC958, direction=1, switch=1, volume=0, volume_limit=-1, enumeration=0, required=0, required_any=0, required_absent=0, mask=0x0, n_channels=0, override_map=no ( 9.618| 0.000) D: [pulseaudio] alsa-mixer.c: Skipping profile output:iec958-stereo+input:analog-mono - will not be able to open input:analog-mono ( 9.618| 0.000) D: [pulseaudio] alsa-mixer.c: Looking at profile output:iec958-stereo+input:analog-stereo ( 9.618| 0.000) D: [pulseaudio] alsa-mixer.c: Checking for recording on Analog Stereo (analog-stereo) ( 9.618| 0.000) D: [pulseaudio] alsa-util.c: Trying front:0 with SND_PCM_NO_AUTO_FORMAT ... ( 9.618| 0.000) D: [pulseaudio] alsa-util.c: Managed to open front:0 ( 9.618| 0.000) D: [pulseaudio] alsa-util.c: Maximum hw buffer size is 371 ms ( 9.618| 0.000) D: [pulseaudio] alsa-util.c: Set buffer size first (to 2028 samples), period size second (to 1014 samples). ( 9.618| 0.000) D: [pulseaudio] alsa-mixer.c: Profile output:iec958-stereo+input:analog-stereo supported. ( 9.618| 0.000) D: [pulseaudio] alsa-mixer.c: Skipping profile output:iec958-stereo+input:iec958-stereo - will not be able to open input:iec958-stereo ( 9.618| 0.000) D: [pulseaudio] alsa-mixer.c: Looking at profile output:iec958-ac3-surround-40 ( 9.618| 0.000) D: [pulseaudio] alsa-mixer.c: Checking for playback on Digital Surround 4.0 (IEC958/AC3) (iec958-ac3-surround-40) ( 9.618| 0.000) D: [pulseaudio] alsa-util.c: Trying a52:0 with SND_PCM_NO_AUTO_FORMAT ... ( 9.618| 0.000) I: [pulseaudio] (alsa-lib)pcm.c: Unknown PCM a52:0 ( 9.619| 0.000) I: [pulseaudio] alsa-util.c: Error opening PCM device a52:0: No such file or directory ( 9.619| 0.000) D: [pulseaudio] alsa-mixer.c: Caching failure to open output:iec958-ac3-surround-40 ( 9.619| 0.000) D: [pulseaudio] alsa-mixer.c: Skipping profile output:iec958-ac3-surround-40+input:analog-mono - will not be able to open output:iec958-ac3-surround-40 ( 9.619| 0.000) D: [pulseaudio] alsa-mixer.c: Skipping profile output:iec958-ac3-surround-40+input:analog-stereo - will not be able to open output:iec958-ac3-surround-40 ( 9.619| 0.000) D: [pulseaudio] alsa-mixer.c: Skipping profile output:iec958-ac3-surround-40+input:iec958-stereo - will not be able to open output:iec958-ac3-surround-40 ( 9.619| 0.000) D: [pulseaudio] alsa-mixer.c: Looking at profile output:iec958-ac3-surround-51 ( 9.619| 0.000) D: [pulseaudio] alsa-mixer.c: Checking for playback on Digital Surround 5.1 (IEC958/AC3) (iec958-ac3-surround-51) ( 9.619| 0.000) D: [pulseaudio] alsa-util.c: Trying a52:0 with SND_PCM_NO_AUTO_FORMAT ... ( 9.619| 0.000) I: [pulseaudio] (alsa-lib)pcm.c: Unknown PCM a52:0 ( 9.619| 0.000) I: [pulseaudio] alsa-util.c: Error opening PCM device a52:0: No such file or directory ( 9.619| 0.000) D: [pulseaudio] alsa-mixer.c: Caching failure to open output:iec958-ac3-surround-51 ( 9.619| 0.000) D: [pulseaudio] alsa-mixer.c: Skipping profile output:iec958-ac3-surround-51+input:analog-mono - will not be able to open output:iec958-ac3-surround-51 ( 9.619| 0.000) D: [pulseaudio] alsa-mixer.c: Skipping profile output:iec958-ac3-surround-51+input:analog-stereo - will not be able to open output:iec958-ac3-surround-51 ( 9.619| 0.000) D: [pulseaudio] alsa-mixer.c: Skipping profile output:iec958-ac3-surround-51+input:iec958-stereo - will not be able to open output:iec958-ac3-surround-51 ( 9.619| 0.000) D: [pulseaudio] alsa-mixer.c: Looking at profile output:iec958-dts-surround-51 ( 9.619| 0.000) D: [pulseaudio] alsa-mixer.c: Checking for playback on Digital Surround 5.1 (IEC958/DTS) (iec958-dts-surround-51) ( 9.619| 0.000) D: [pulseaudio] alsa-util.c: Trying dca:0 with SND_PCM_NO_AUTO_FORMAT ... ( 9.619| 0.000) I: [pulseaudio] (alsa-lib)pcm.c: Unknown PCM dca:0 ( 9.619| 0.000) I: [pulseaudio] alsa-util.c: Error opening PCM device dca:0: No such file or directory ( 9.619| 0.000) D: [pulseaudio] alsa-mixer.c: Caching failure to open output:iec958-dts-surround-51 ( 9.619| 0.000) D: [pulseaudio] alsa-mixer.c: Skipping profile output:iec958-dts-surround-51+input:analog-mono - will not be able to open output:iec958-dts-surround-51 ( 9.619| 0.000) D: [pulseaudio] alsa-mixer.c: Skipping profile output:iec958-dts-surround-51+input:analog-stereo - will not be able to open output:iec958-dts-surround-51 ( 9.619| 0.000) D: [pulseaudio] alsa-mixer.c: Skipping profile output:iec958-dts-surround-51+input:iec958-stereo - will not be able to open output:iec958-dts-surround-51 ( 9.619| 0.000) D: [pulseaudio] alsa-mixer.c: Looking at profile output:hdmi-stereo ( 9.619| 0.000) D: [pulseaudio] alsa-mixer.c: Checking for playback on Digital Stereo (HDMI) (hdmi-stereo) ( 9.619| 0.000) D: [pulseaudio] alsa-util.c: Trying hdmi:0 with SND_PCM_NO_AUTO_FORMAT ... ( 9.619| 0.000) I: [pulseaudio] (alsa-lib)pcm_hw.c: open '/dev/snd/pcmC0D3p' failed (-2) ( 9.619| 0.000) I: [pulseaudio] alsa-util.c: Error opening PCM device hdmi:0: No such file or directory ( 9.619| 0.000) D: [pulseaudio] alsa-mixer.c: Caching failure to open output:hdmi-stereo ( 9.619| 0.000) D: [pulseaudio] alsa-mixer.c: Skipping profile output:hdmi-stereo+input:analog-mono - will not be able to open output:hdmi-stereo ( 9.619| 0.000) D: [pulseaudio] alsa-mixer.c: Skipping profile output:hdmi-stereo+input:analog-stereo - will not be able to open output:hdmi-stereo ( 9.619| 0.000) D: [pulseaudio] alsa-mixer.c: Skipping profile output:hdmi-stereo+input:iec958-stereo - will not be able to open output:hdmi-stereo ( 9.619| 0.000) D: [pulseaudio] alsa-mixer.c: Looking at profile output:hdmi-surround ( 9.619| 0.000) D: [pulseaudio] alsa-mixer.c: Checking for playback on Digital Surround 5.1 (HDMI) (hdmi-surround) ( 9.619| 0.000) D: [pulseaudio] alsa-util.c: Trying hdmi:0 with SND_PCM_NO_AUTO_FORMAT ... ( 9.619| 0.000) I: [pulseaudio] (alsa-lib)pcm_hw.c: open '/dev/snd/pcmC0D3p' failed (-2) ( 9.619| 0.000) I: [pulseaudio] alsa-util.c: Error opening PCM device hdmi:0: No such file or directory ( 9.619| 0.000) D: [pulseaudio] alsa-mixer.c: Caching failure to open output:hdmi-surround ( 9.619| 0.000) D: [pulseaudio] alsa-mixer.c: Skipping profile output:hdmi-surround+input:analog-mono - will not be able to open output:hdmi-surround ( 9.619| 0.000) D: [pulseaudio] alsa-mixer.c: Skipping profile output:hdmi-surround+input:analog-stereo - will not be able to open output:hdmi-surround ( 9.619| 0.000) D: [pulseaudio] alsa-mixer.c: Skipping profile output:hdmi-surround+input:iec958-stereo - will not be able to open output:hdmi-surround ( 9.619| 0.000) D: [pulseaudio] alsa-mixer.c: Looking at profile output:hdmi-stereo-extra1 ( 9.619| 0.000) D: [pulseaudio] alsa-mixer.c: Checking for playback on Digital Stereo (HDMI) (hdmi-stereo-extra1) ( 9.619| 0.000) D: [pulseaudio] alsa-util.c: Trying hdmi:0,1 with SND_PCM_NO_AUTO_FORMAT ... ( 9.619| 0.000) I: [pulseaudio] (alsa-lib)pcm_hw.c: open '/dev/snd/pcmC0D7p' failed (-2) ( 9.619| 0.000) I: [pulseaudio] alsa-util.c: Error opening PCM device hdmi:0,1: No such file or directory ( 9.619| 0.000) D: [pulseaudio] alsa-mixer.c: Caching failure to open output:hdmi-stereo-extra1 ( 9.619| 0.000) D: [pulseaudio] alsa-mixer.c: Skipping profile output:hdmi-stereo-extra1+input:analog-mono - will not be able to open output:hdmi-stereo-extra1 ( 9.619| 0.000) D: [pulseaudio] alsa-mixer.c: Skipping profile output:hdmi-stereo-extra1+input:analog-stereo - will not be able to open output:hdmi-stereo-extra1 ( 9.619| 0.000) D: [pulseaudio] alsa-mixer.c: Skipping profile output:hdmi-stereo-extra1+input:iec958-stereo - will not be able to open output:hdmi-stereo-extra1 ( 9.619| 0.000) D: [pulseaudio] alsa-mixer.c: Looking at profile output:hdmi-surround-extra1 ( 9.619| 0.000) D: [pulseaudio] alsa-mixer.c: Checking for playback on Digital Surround 5.1 (HDMI) (hdmi-surround-extra1) ( 9.619| 0.000) D: [pulseaudio] alsa-util.c: Trying hdmi:0,1 with SND_PCM_NO_AUTO_FORMAT ... ( 9.620| 0.000) I: [pulseaudio] (alsa-lib)pcm_hw.c: open '/dev/snd/pcmC0D7p' failed (-2) ( 9.620| 0.000) I: [pulseaudio] alsa-util.c: Error opening PCM device hdmi:0,1: No such file or directory ( 9.620| 0.000) D: [pulseaudio] alsa-mixer.c: Caching failure to open output:hdmi-surround-extra1 ( 9.620| 0.000) D: [pulseaudio] alsa-mixer.c: Skipping profile output:hdmi-surround-extra1+input:analog-mono - will not be able to open output:hdmi-surround-extra1 ( 9.620| 0.000) D: [pulseaudio] alsa-mixer.c: Skipping profile output:hdmi-surround-extra1+input:analog-stereo - will not be able to open output:hdmi-surround-extra1 ( 9.620| 0.000) D: [pulseaudio] alsa-mixer.c: Skipping profile output:hdmi-surround-extra1+input:iec958-stereo - will not be able to open output:hdmi-surround-extra1 ( 9.620| 0.000) D: [pulseaudio] alsa-mixer.c: Looking at profile output:hdmi-stereo-extra2 ( 9.620| 0.000) D: [pulseaudio] alsa-mixer.c: Checking for playback on Digital Stereo (HDMI) (hdmi-stereo-extra2) ( 9.620| 0.000) D: [pulseaudio] alsa-util.c: Trying hdmi:0,2 with SND_PCM_NO_AUTO_FORMAT ... ( 9.620| 0.000) I: [pulseaudio] (alsa-lib)pcm_hw.c: open '/dev/snd/pcmC0D8p' failed (-2) ( 9.620| 0.000) I: [pulseaudio] alsa-util.c: Error opening PCM device hdmi:0,2: No such file or directory ( 9.620| 0.000) D: [pulseaudio] alsa-mixer.c: Caching failure to open output:hdmi-stereo-extra2 ( 9.620| 0.000) D: [pulseaudio] alsa-mixer.c: Skipping profile output:hdmi-stereo-extra2+input:analog-mono - will not be able to open output:hdmi-stereo-extra2 ( 9.620| 0.000) D: [pulseaudio] alsa-mixer.c: Skipping profile output:hdmi-stereo-extra2+input:analog-stereo - will not be able to open output:hdmi-stereo-extra2 ( 9.620| 0.000) D: [pulseaudio] alsa-mixer.c: Skipping profile output:hdmi-stereo-extra2+input:iec958-stereo - will not be able to open output:hdmi-stereo-extra2 ( 9.620| 0.000) D: [pulseaudio] alsa-mixer.c: Looking at profile output:hdmi-surround-extra2 ( 9.620| 0.000) D: [pulseaudio] alsa-mixer.c: Checking for playback on Digital Surround 5.1 (HDMI) (hdmi-surround-extra2) ( 9.620| 0.000) D: [pulseaudio] alsa-util.c: Trying hdmi:0,2 with SND_PCM_NO_AUTO_FORMAT ... ( 9.620| 0.000) I: [pulseaudio] (alsa-lib)pcm_hw.c: open '/dev/snd/pcmC0D8p' failed (-2) ( 9.620| 0.000) I: [pulseaudio] alsa-util.c: Error opening PCM device hdmi:0,2: No such file or directory ( 9.620| 0.000) D: [pulseaudio] alsa-mixer.c: Caching failure to open output:hdmi-surround-extra2 ( 9.620| 0.000) D: [pulseaudio] alsa-mixer.c: Skipping profile output:hdmi-surround-extra2+input:analog-mono - will not be able to open output:hdmi-surround-extra2 ( 9.620| 0.000) D: [pulseaudio] alsa-mixer.c: Skipping profile output:hdmi-surround-extra2+input:analog-stereo - will not be able to open output:hdmi-surround-extra2 ( 9.620| 0.000) D: [pulseaudio] alsa-mixer.c: Skipping profile output:hdmi-surround-extra2+input:iec958-stereo - will not be able to open output:hdmi-surround-extra2 ( 9.620| 0.000) D: [pulseaudio] alsa-mixer.c: Looking at profile output:hdmi-stereo-extra3 ( 9.620| 0.000) D: [pulseaudio] alsa-mixer.c: Checking for playback on Digital Stereo (HDMI) (hdmi-stereo-extra3) ( 9.620| 0.000) D: [pulseaudio] alsa-util.c: Trying hdmi:0,3 with SND_PCM_NO_AUTO_FORMAT ... ( 9.620| 0.000) I: [pulseaudio] (alsa-lib)pcm_hw.c: open '/dev/snd/pcmC0D9p' failed (-2) ( 9.620| 0.000) I: [pulseaudio] alsa-util.c: Error opening PCM device hdmi:0,3: No such file or directory ( 9.620| 0.000) D: [pulseaudio] alsa-mixer.c: Caching failure to open output:hdmi-stereo-extra3 ( 9.620| 0.000) D: [pulseaudio] alsa-mixer.c: Skipping profile output:hdmi-stereo-extra3+input:analog-mono - will not be able to open output:hdmi-stereo-extra3 ( 9.620| 0.000) D: [pulseaudio] alsa-mixer.c: Skipping profile output:hdmi-stereo-extra3+input:analog-stereo - will not be able to open output:hdmi-stereo-extra3 ( 9.620| 0.000) D: [pulseaudio] alsa-mixer.c: Skipping profile output:hdmi-stereo-extra3+input:iec958-stereo - will not be able to open output:hdmi-stereo-extra3 ( 9.620| 0.000) D: [pulseaudio] alsa-mixer.c: Looking at profile output:hdmi-surround-extra3 ( 9.620| 0.000) D: [pulseaudio] alsa-mixer.c: Checking for playback on Digital Surround 5.1 (HDMI) (hdmi-surround-extra3) ( 9.620| 0.000) D: [pulseaudio] alsa-util.c: Trying hdmi:0,3 with SND_PCM_NO_AUTO_FORMAT ... ( 9.621| 0.000) I: [pulseaudio] (alsa-lib)pcm_hw.c: open '/dev/snd/pcmC0D9p' failed (-2) ( 9.621| 0.000) I: [pulseaudio] alsa-util.c: Error opening PCM device hdmi:0,3: No such file or directory ( 9.621| 0.000) D: [pulseaudio] alsa-mixer.c: Caching failure to open output:hdmi-surround-extra3 ( 9.621| 0.000) D: [pulseaudio] alsa-mixer.c: Skipping profile output:hdmi-surround-extra3+input:analog-mono - will not be able to open output:hdmi-surround-extra3 ( 9.621| 0.000) D: [pulseaudio] alsa-mixer.c: Skipping profile output:hdmi-surround-extra3+input:analog-stereo - will not be able to open output:hdmi-surround-extra3 ( 9.621| 0.000) D: [pulseaudio] alsa-mixer.c: Skipping profile output:hdmi-surround-extra3+input:iec958-stereo - will not be able to open output:hdmi-surround-extra3 ( 9.621| 0.000) D: [pulseaudio] alsa-mixer.c: Profile set 0xc5d8d0, auto_profiles=yes, probed=yes, n_mappings=7, n_profiles=15, n_decibel_fixes=0 ( 9.621| 0.000) D: [pulseaudio] alsa-mixer.c: Mapping analog-stereo (Analog Stereo), priority=60, channel_map=front-left,front-right, supported=yes, direction=0 ( 9.621| 0.000) D: [pulseaudio] alsa-mixer.c: Mapping analog-surround-40 (Analog Surround 4.0), priority=7, channel_map=front-left,front-right,rear-left,rear-right, supported=yes, direction=1 ( 9.621| 0.000) D: [pulseaudio] alsa-mixer.c: Mapping analog-surround-41 (Analog Surround 4.1), priority=8, channel_map=front-left,front-right,rear-left,rear-right,lfe, supported=yes, direction=1 ( 9.621| 0.000) D: [pulseaudio] alsa-mixer.c: Mapping analog-surround-50 (Analog Surround 5.0), priority=7, channel_map=front-left,front-right,rear-left,rear-right,front-center, supported=yes, direction=1 ( 9.621| 0.000) D: [pulseaudio] alsa-mixer.c: Mapping analog-surround-51 (Analog Surround 5.1), priority=8, channel_map=front-left,front-right,rear-left,rear-right,front-center,lfe, supported=yes, direction=1 ( 9.621| 0.000) D: [pulseaudio] alsa-mixer.c: Mapping analog-surround-71 (Analog Surround 7.1), priority=7, channel_map=front-left,front-right,rear-left,rear-right,front-center,lfe,side-left,side-right, supported=yes, direction=1 ( 9.621| 0.000) D: [pulseaudio] alsa-mixer.c: Mapping iec958-stereo (Digital Stereo (IEC958)), priority=55, channel_map=front-left,front-right, supported=yes, direction=0 ( 9.621| 0.000) D: [pulseaudio] alsa-mixer.c: Profile input:analog-stereo (Analog Stereo Input), priority=60, supported=yes n_input_mappings=1, n_output_mappings=0 ( 9.621| 0.000) D: [pulseaudio] alsa-mixer.c: Input analog-stereo ( 9.621| 0.000) D: [pulseaudio] alsa-mixer.c: Profile output:analog-stereo (Analog Stereo Output), priority=6000, supported=yes n_input_mappings=0, n_output_mappings=1 ( 9.621| 0.000) D: [pulseaudio] alsa-mixer.c: Output analog-stereo ( 9.621| 0.000) D: [pulseaudio] alsa-mixer.c: Profile output:analog-stereo+input:analog-stereo (Analog Stereo Duplex), priority=6060, supported=yes n_input_mappings=1, n_output_mappings=1 ( 9.621| 0.000) D: [pulseaudio] alsa-mixer.c: Input analog-stereo ( 9.621| 0.000) D: [pulseaudio] alsa-mixer.c: Output analog-stereo ( 9.621| 0.000) D: [pulseaudio] alsa-mixer.c: Profile output:analog-surround-40 (Analog Surround 4.0 Output), priority=700, supported=yes n_input_mappings=0, n_output_mappings=1 ( 9.621| 0.000) D: [pulseaudio] alsa-mixer.c: Output analog-surround-40 ( 9.621| 0.000) D: [pulseaudio] alsa-mixer.c: Profile output:analog-surround-40+input:analog-stereo (Analog Surround 4.0 Output + Analog Stereo Input), priority=760, supported=yes n_input_mappings=1, n_output_mappings=1 ( 9.621| 0.000) D: [pulseaudio] alsa-mixer.c: Input analog-stereo ( 9.621| 0.000) D: [pulseaudio] alsa-mixer.c: Output analog-surround-40 ( 9.621| 0.000) D: [pulseaudio] alsa-mixer.c: Profile output:analog-surround-41 (Analog Surround 4.1 Output), priority=800, supported=yes n_input_mappings=0, n_output_mappings=1 ( 9.621| 0.000) D: [pulseaudio] alsa-mixer.c: Output analog-surround-41 ( 9.621| 0.000) D: [pulseaudio] alsa-mixer.c: Profile output:analog-surround-41+input:analog-stereo (Analog Surround 4.1 Output + Analog Stereo Input), priority=860, supported=yes n_input_mappings=1, n_output_mappings=1 ( 9.621| 0.000) D: [pulseaudio] alsa-mixer.c: Input analog-stereo ( 9.621| 0.000) D: [pulseaudio] alsa-mixer.c: Output analog-surround-41 ( 9.621| 0.000) D: [pulseaudio] alsa-mixer.c: Profile output:analog-surround-50 (Analog Surround 5.0 Output), priority=700, supported=yes n_input_mappings=0, n_output_mappings=1 ( 9.621| 0.000) D: [pulseaudio] alsa-mixer.c: Output analog-surround-50 ( 9.621| 0.000) D: [pulseaudio] alsa-mixer.c: Profile output:analog-surround-50+input:analog-stereo (Analog Surround 5.0 Output + Analog Stereo Input), priority=760, supported=yes n_input_mappings=1, n_output_mappings=1 ( 9.621| 0.000) D: [pulseaudio] alsa-mixer.c: Input analog-stereo ( 9.621| 0.000) D: [pulseaudio] alsa-mixer.c: Output analog-surround-50 ( 9.621| 0.000) D: [pulseaudio] alsa-mixer.c: Profile output:analog-surround-51 (Analog Surround 5.1 Output), priority=800, supported=yes n_input_mappings=0, n_output_mappings=1 ( 9.621| 0.000) D: [pulseaudio] alsa-mixer.c: Output analog-surround-51 ( 9.621| 0.000) D: [pulseaudio] alsa-mixer.c: Profile output:analog-surround-51+input:analog-stereo (Analog Surround 5.1 Output + Analog Stereo Input), priority=860, supported=yes n_input_mappings=1, n_output_mappings=1 ( 9.621| 0.000) D: [pulseaudio] alsa-mixer.c: Input analog-stereo ( 9.621| 0.000) D: [pulseaudio] alsa-mixer.c: Output analog-surround-51 ( 9.621| 0.000) D: [pulseaudio] alsa-mixer.c: Profile output:analog-surround-71 (Analog Surround 7.1 Output), priority=700, supported=yes n_input_mappings=0, n_output_mappings=1 ( 9.621| 0.000) D: [pulseaudio] alsa-mixer.c: Output analog-surround-71 ( 9.621| 0.000) D: [pulseaudio] alsa-mixer.c: Profile output:analog-surround-71+input:analog-stereo (Analog Surround 7.1 Output + Analog Stereo Input), priority=760, supported=yes n_input_mappings=1, n_output_mappings=1 ( 9.621| 0.000) D: [pulseaudio] alsa-mixer.c: Input analog-stereo ( 9.621| 0.000) D: [pulseaudio] alsa-mixer.c: Output analog-surround-71 ( 9.621| 0.000) D: [pulseaudio] alsa-mixer.c: Profile output:iec958-stereo (Digital Stereo (IEC958) Output), priority=5500, supported=yes n_input_mappings=0, n_output_mappings=1 ( 9.621| 0.000) D: [pulseaudio] alsa-mixer.c: Output iec958-stereo ( 9.621| 0.000) D: [pulseaudio] alsa-mixer.c: Profile output:iec958-stereo+input:analog-stereo (Digital Stereo (IEC958) Output + Analog Stereo Input), priority=5560, supported=yes n_input_mappings=1, n_output_mappings=1 ( 9.621| 0.000) D: [pulseaudio] alsa-mixer.c: Input analog-stereo ( 9.621| 0.000) D: [pulseaudio] alsa-mixer.c: Output iec958-stereo ( 9.621| 0.000) I: [pulseaudio] module-card-restore.c: Restoring profile for card alsa_card.pci-0000_00_1b.0. ( 9.621| 0.000) I: [pulseaudio] module-card-restore.c: Restoring port latency offsets for card alsa_card.pci-0000_00_1b.0. ( 9.621| 0.000) I: [pulseaudio] card.c: Created 1 "alsa_card.pci-0000_00_1b.0" ( 9.622| 0.000) D: [pulseaudio] reserve-wrap.c: Successfully create reservation lock monitor for device 'Audio0' ( 9.622| 0.000) D: [pulseaudio] alsa-util.c: Trying front:0 with SND_PCM_NO_AUTO_FORMAT ... ( 9.622| 0.000) D: [pulseaudio] alsa-util.c: Managed to open front:0 ( 9.622| 0.000) D: [pulseaudio] alsa-util.c: Maximum hw buffer size is 371 ms ( 12.681| 3.059) D: [pulseaudio] alsa-util.c: Set buffer size first (to 2028 samples), period size second (to 1014 samples). ( 12.681| 0.000) I: [pulseaudio] alsa-sink.c: Successfully opened device front:0. ( 12.681| 0.000) I: [pulseaudio] alsa-sink.c: Selected mapping 'Analog Stereo' (analog-stereo). ( 12.681| 0.000) I: [pulseaudio] alsa-sink.c: Successfully enabled mmap() mode. ( 12.682| 0.000) I: [pulseaudio] (alsa-lib)control.c: Invalid CTL front:0 ( 12.682| 0.000) I: [pulseaudio] alsa-util.c: Unable to attach to mixer front:0: No such file or directory ( 12.682| 0.000) I: [pulseaudio] alsa-util.c: Successfully attached to mixer 'hw:0' ( 12.682| 0.000) D: [pulseaudio] alsa-mixer.c: Added 2 ports ( 12.682| 0.000) D: [pulseaudio] core-subscribe.c: Dropped redundant event due to change event. ( 12.682| 0.000) D: [pulseaudio] module-device-restore.c: Database contains invalid data for key: sink:alsa_output.pci-0000_00_1b.0.analog-stereo:null ( 12.682| 0.000) I: [pulseaudio] sink.c: Created sink 1 "alsa_output.pci-0000_00_1b.0.analog-stereo" with sample spec s16le 2ch 44100Hz and channel map front-left,front-right ( 12.682| 0.000) I: [pulseaudio] sink.c: alsa.resolution_bits = "16" ( 12.682| 0.000) I: [pulseaudio] sink.c: device.api = "alsa" ( 12.682| 0.000) I: [pulseaudio] sink.c: device.class = "sound" ( 12.682| 0.000) I: [pulseaudio] sink.c: alsa.class = "generic" ( 12.682| 0.000) I: [pulseaudio] sink.c: alsa.subclass = "generic-mix" ( 12.682| 0.000) I: [pulseaudio] sink.c: alsa.name = "ALC889 Analog" ( 12.682| 0.000) I: [pulseaudio] sink.c: alsa.id = "ALC889 Analog" ( 12.682| 0.000) I: [pulseaudio] sink.c: alsa.subdevice = "0" ( 12.682| 0.000) I: [pulseaudio] sink.c: alsa.subdevice_name = "subdevice #0" ( 12.682| 0.000) I: [pulseaudio] sink.c: alsa.device = "0" ( 12.682| 0.000) I: [pulseaudio] sink.c: alsa.card = "0" ( 12.682| 0.000) I: [pulseaudio] sink.c: alsa.card_name = "HDA Intel PCH" ( 12.682| 0.000) I: [pulseaudio] sink.c: alsa.long_card_name = "HDA Intel PCH at 0xf7420000 irq 54" ( 12.682| 0.000) I: [pulseaudio] sink.c: alsa.driver_name = "snd_hda_intel" ( 12.682| 0.000) I: [pulseaudio] sink.c: device.bus_path = "pci-0000:00:1b.0" ( 12.682| 0.000) I: [pulseaudio] sink.c: sysfs.path = "/devices/pci0000:00/0000:00:1b.0/sound/card0" ( 12.682| 0.000) I: [pulseaudio] sink.c: device.bus = "pci" ( 12.682| 0.000) I: [pulseaudio] sink.c: device.vendor.id = "8086" ( 12.682| 0.000) I: [pulseaudio] sink.c: device.vendor.name = "Intel Corporation" ( 12.682| 0.000) I: [pulseaudio] sink.c: device.product.name = "6 Series/C200 Series Chipset Family High Definition Audio Controller" ( 12.682| 0.000) I: [pulseaudio] sink.c: device.form_factor = "internal" ( 12.682| 0.000) I: [pulseaudio] sink.c: device.string = "front:0" ( 12.682| 0.000) I: [pulseaudio] sink.c: device.buffering.buffer_size = "8112" ( 12.682| 0.000) I: [pulseaudio] sink.c: device.buffering.fragment_size = "4056" ( 12.682| 0.000) I: [pulseaudio] sink.c: device.access_mode = "mmap" ( 12.682| 0.000) I: [pulseaudio] sink.c: device.profile.name = "analog-stereo" ( 12.682| 0.000) I: [pulseaudio] sink.c: device.profile.description = "Analog Stereo" ( 12.682| 0.000) I: [pulseaudio] sink.c: device.description = "Built-in Audio Analog Stereo" ( 12.682| 0.000) I: [pulseaudio] sink.c: alsa.mixer_name = "Realtek ALC889" ( 12.682| 0.000) I: [pulseaudio] sink.c: alsa.components = "HDA:10ec0889,1043846b,00100004" ( 12.682| 0.000) I: [pulseaudio] sink.c: module-udev-detect.discovered = "1" ( 12.682| 0.000) I: [pulseaudio] sink.c: device.icon_name = "audio-card-pci" ( 12.682| 0.000) D: [pulseaudio] core-subscribe.c: Dropped redundant event due to change event. ( 12.682| 0.000) I: [pulseaudio] source.c: Created source 1 "alsa_output.pci-0000_00_1b.0.analog-stereo.monitor" with sample spec s16le 2ch 44100Hz and channel map front-left,front-right ( 12.682| 0.000) I: [pulseaudio] source.c: device.description = "Monitor of Built-in Audio Analog Stereo" ( 12.682| 0.000) I: [pulseaudio] source.c: device.class = "monitor" ( 12.682| 0.000) I: [pulseaudio] source.c: alsa.card = "0" ( 12.682| 0.000) I: [pulseaudio] source.c: alsa.card_name = "HDA Intel PCH" ( 12.682| 0.000) I: [pulseaudio] source.c: alsa.long_card_name = "HDA Intel PCH at 0xf7420000 irq 54" ( 12.682| 0.000) I: [pulseaudio] source.c: alsa.driver_name = "snd_hda_intel" ( 12.682| 0.000) I: [pulseaudio] source.c: device.bus_path = "pci-0000:00:1b.0" ( 12.682| 0.000) I: [pulseaudio] source.c: sysfs.path = "/devices/pci0000:00/0000:00:1b.0/sound/card0" ( 12.682| 0.000) I: [pulseaudio] source.c: device.bus = "pci" ( 12.682| 0.000) I: [pulseaudio] source.c: device.vendor.id = "8086" ( 12.682| 0.000) I: [pulseaudio] source.c: device.vendor.name = "Intel Corporation" ( 12.682| 0.000) I: [pulseaudio] source.c: device.product.name = "6 Series/C200 Series Chipset Family High Definition Audio Controller" ( 12.682| 0.000) I: [pulseaudio] source.c: device.form_factor = "internal" ( 12.682| 0.000) I: [pulseaudio] source.c: device.string = "0" ( 12.682| 0.000) I: [pulseaudio] source.c: module-udev-detect.discovered = "1" ( 12.682| 0.000) I: [pulseaudio] source.c: device.icon_name = "audio-card-pci" ( 12.682| 0.000) I: [pulseaudio] alsa-sink.c: Using 2.0 fragments of size 4056 bytes (22.99ms), buffer size is 8112 bytes (45.99ms) ( 12.683| 0.000) D: [pulseaudio] alsa-sink.c: hwbuf_unused=0 ( 12.683| 0.000) D: [pulseaudio] alsa-sink.c: setting avail_min=1 ( 12.683| 0.000) D: [pulseaudio] alsa-mixer.c: Activating path analog-output ( 12.683| 0.000) D: [pulseaudio] alsa-mixer.c: Path analog-output (Analog Output), direction=1, priority=99, probed=yes, supported=yes, has_mute=yes, has_volume=yes, has_dB=yes, min_volume=0, max_volume=64, min_dB=-177, max_dB=2 ( 12.683| 0.000) D: [pulseaudio] alsa-mixer.c: Element Master, direction=1, switch=1, volume=1, volume_limit=-1, enumeration=0, required=0, required_any=0, required_absent=0, mask=0x7ffffffffffff, n_channels=1, override_map=yes ( 12.683| 0.000) D: [pulseaudio] alsa-mixer.c: Element Headphone, direction=1, switch=1, volume=3, volume_limit=-1, enumeration=0, required=0, required_any=0, required_absent=0, mask=0x6, n_channels=2, override_map=no ( 12.683| 0.000) D: [pulseaudio] alsa-mixer.c: Element Front, direction=1, switch=1, volume=1, volume_limit=-1, enumeration=0, required=0, required_any=0, required_absent=0, mask=0x6, n_channels=2, override_map=yes ( 12.683| 0.000) D: [pulseaudio] alsa-mixer.c: Element Surround, direction=1, switch=1, volume=1, volume_limit=-1, enumeration=0, required=0, required_any=0, required_absent=0, mask=0x60, n_channels=2, override_map=yes ( 12.683| 0.000) D: [pulseaudio] alsa-mixer.c: Element Side, direction=1, switch=1, volume=1, volume_limit=-1, enumeration=0, required=0, required_any=0, required_absent=0, mask=0xc00, n_channels=2, override_map=yes ( 12.683| 0.000) D: [pulseaudio] alsa-mixer.c: Element Center, direction=1, switch=1, volume=1, volume_limit=-1, enumeration=0, required=0, required_any=0, required_absent=0, mask=0x4900000000018, n_channels=1, override_map=yes ( 12.683| 0.000) D: [pulseaudio] alsa-mixer.c: Element LFE, direction=1, switch=1, volume=1, volume_limit=-1, enumeration=0, required=0, required_any=0, required_absent=0, mask=0x80, n_channels=1, override_map=yes ( 12.683| 0.000) D: [pulseaudio] alsa-mixer.c: Element PCM, direction=1, switch=0, volume=1, volume_limit=-1, enumeration=0, required=0, required_any=0, required_absent=0, mask=0x3600000000f66, n_channels=2, override_map=yes ( 12.683| 0.000) D: [pulseaudio] alsa-mixer.c: Element IEC958, direction=1, switch=2, volume=0, volume_limit=-1, enumeration=0, required=0, required_any=0, required_absent=0, mask=0x0, n_channels=0, override_map=no ( 12.683| 0.000) I: [pulseaudio] alsa-sink.c: Successfully enabled deferred volume. ( 12.683| 0.000) I: [pulseaudio] alsa-sink.c: Hardware volume ranges from -177.00 dB to 2.00 dB. ( 12.683| 0.000) I: [pulseaudio] alsa-sink.c: Fixing base volume to -2.00 dB ( 12.683| 0.000) I: [pulseaudio] alsa-sink.c: Using hardware volume control. Hardware dB scale supported. ( 12.683| 0.000) I: [pulseaudio] alsa-sink.c: Using hardware mute control. ( 12.683| 0.000) D: [pulseaudio] alsa-util.c: snd_pcm_dump(): ( 12.683| 0.000) D: [pulseaudio] alsa-util.c: Soft volume PCM ( 12.683| 0.000) D: [pulseaudio] alsa-util.c: Control: PCM Playback Volume ( 12.683| 0.000) D: [pulseaudio] alsa-util.c: min_dB: -51 ( 12.683| 0.000) D: [pulseaudio] alsa-util.c: max_dB: 0 ( 12.683| 0.000) D: [pulseaudio] alsa-util.c: resolution: 256 ( 12.683| 0.000) D: [pulseaudio] alsa-util.c: Its setup is: ( 12.683| 0.000) D: [pulseaudio] alsa-util.c: stream : PLAYBACK ( 12.683| 0.000) D: [pulseaudio] alsa-util.c: access : MMAP_INTERLEAVED ( 12.683| 0.000) D: [pulseaudio] alsa-util.c: format : S16_LE ( 12.683| 0.000) D: [pulseaudio] alsa-util.c: subformat : STD ( 12.683| 0.000) D: [pulseaudio] alsa-util.c: channels : 2 ( 12.683| 0.000) D: [pulseaudio] alsa-util.c: rate : 44100 ( 12.683| 0.000) D: [pulseaudio] alsa-util.c: exact rate : 44100 (44100/1) ( 12.683| 0.000) D: [pulseaudio] alsa-util.c: msbits : 16 ( 12.683| 0.000) D: [pulseaudio] alsa-util.c: buffer_size : 2028 ( 12.683| 0.000) D: [pulseaudio] alsa-util.c: period_size : 1014 ( 12.683| 0.000) D: [pulseaudio] alsa-util.c: period_time : 22993 ( 12.683| 0.000) D: [pulseaudio] alsa-util.c: tstamp_mode : ENABLE ( 12.683| 0.000) D: [pulseaudio] alsa-util.c: period_step : 1 ( 12.683| 0.000) D: [pulseaudio] alsa-util.c: avail_min : 1014 ( 12.683| 0.000) D: [pulseaudio] alsa-util.c: period_event : 1 ( 12.683| 0.000) D: [pulseaudio] alsa-util.c: start_threshold : -1 ( 12.683| 0.000) D: [pulseaudio] alsa-util.c: stop_threshold : 9133300044307365888 ( 12.683| 0.000) D: [pulseaudio] alsa-util.c: silence_threshold: 0 ( 12.683| 0.000) D: [pulseaudio] alsa-util.c: silence_size : 0 ( 12.683| 0.000) D: [pulseaudio] alsa-util.c: boundary : 9133300044307365888 ( 12.683| 0.000) D: [pulseaudio] alsa-util.c: Slave: Hardware PCM card 0 'HDA Intel PCH' device 0 subdevice 0 ( 12.683| 0.000) D: [pulseaudio] alsa-util.c: Its setup is: ( 12.683| 0.000) D: [pulseaudio] alsa-util.c: stream : PLAYBACK ( 12.683| 0.000) D: [pulseaudio] alsa-util.c: access : MMAP_INTERLEAVED ( 12.683| 0.000) D: [pulseaudio] alsa-util.c: format : S16_LE ( 12.683| 0.000) D: [pulseaudio] alsa-util.c: subformat : STD ( 12.683| 0.000) D: [pulseaudio] alsa-util.c: channels : 2 ( 12.683| 0.000) D: [pulseaudio] alsa-util.c: rate : 44100 ( 12.683| 0.000) D: [pulseaudio] alsa-util.c: exact rate : 44100 (44100/1) ( 12.683| 0.000) D: [pulseaudio] alsa-util.c: msbits : 16 ( 12.683| 0.000) D: [pulseaudio] alsa-util.c: buffer_size : 2028 ( 12.683| 0.000) D: [pulseaudio] alsa-util.c: period_size : 1014 ( 12.683| 0.000) D: [pulseaudio] alsa-util.c: period_time : 22993 ( 12.683| 0.000) D: [pulseaudio] alsa-util.c: tstamp_mode : ENABLE ( 12.683| 0.000) D: [pulseaudio] alsa-util.c: period_step : 1 ( 12.683| 0.000) D: [pulseaudio] alsa-util.c: avail_min : 1014 ( 12.683| 0.000) D: [pulseaudio] alsa-util.c: period_event : 1 ( 12.683| 0.000) D: [pulseaudio] alsa-util.c: start_threshold : -1 ( 12.683| 0.000) D: [pulseaudio] alsa-util.c: stop_threshold : 9133300044307365888 ( 12.683| 0.000) D: [pulseaudio] alsa-util.c: silence_threshold: 0 ( 12.683| 0.000) D: [pulseaudio] alsa-util.c: silence_size : 0 ( 12.683| 0.000) D: [pulseaudio] alsa-util.c: boundary : 9133300044307365888 ( 12.683| 0.000) D: [pulseaudio] alsa-util.c: appl_ptr : 0 ( 12.683| 0.000) D: [pulseaudio] alsa-util.c: hw_ptr : 0 ( 12.683| 0.000) D: [pulseaudio] alsa-sink.c: Read hardware volume: 0: 76% 1: 76% ( 12.683| 0.000) D: [alsa-sink] alsa-sink.c: Thread starting up ( 12.683| 0.000) D: [pulseaudio] alsa-sink.c: in dB: 0: -7.20 dB 1: -7.20 dB ( 12.687| 0.003) D: [alsa-sink] core-util.c: RealtimeKit worked. ( 12.687| 0.000) I: [alsa-sink] core-util.c: Successfully enabled SCHED_RR scheduling for thread, with priority 5. ( 12.687| 0.000) I: [alsa-sink] alsa-sink.c: Starting playback. ( 12.687| 0.000) D: [pulseaudio] module-device-restore.c: Could not set format on sink alsa_output.pci-0000_00_1b.0.analog-stereo ( 12.687| 0.000) D: [pulseaudio] alsa-util.c: Trying front:0 with SND_PCM_NO_AUTO_FORMAT ... ( 12.687| 0.000) D: [pulseaudio] alsa-util.c: Managed to open front:0 ( 12.687| 0.000) D: [pulseaudio] alsa-util.c: Maximum hw buffer size is 371 ms ( 12.687| 0.000) D: [pulseaudio] alsa-util.c: Set buffer size first (to 2028 samples), period size second (to 1014 samples). ( 12.687| 0.000) I: [pulseaudio] alsa-source.c: Successfully opened device front:0. ( 12.687| 0.000) I: [pulseaudio] alsa-source.c: Selected mapping 'Analog Stereo' (analog-stereo). ( 12.687| 0.000) I: [pulseaudio] alsa-source.c: Successfully enabled mmap() mode. ( 12.687| 0.000) I: [pulseaudio] (alsa-lib)control.c: Invalid CTL front:0 ( 12.687| 0.000) I: [pulseaudio] alsa-util.c: Unable to attach to mixer front:0: No such file or directory ( 12.687| 0.000) I: [pulseaudio] alsa-util.c: Successfully attached to mixer 'hw:0' ( 12.688| 0.000) D: [pulseaudio] alsa-mixer.c: Added 3 ports ( 12.688| 0.000) D: [pulseaudio] core-subscribe.c: Dropped redundant event due to change event. ( 12.688| 0.000) D: [pulseaudio] module-device-restore.c: Database contains invalid data for key: source:alsa_input.pci-0000_00_1b.0.analog-stereo:null ( 12.688| 0.000) I: [pulseaudio] source.c: Created source 2 "alsa_input.pci-0000_00_1b.0.analog-stereo" with sample spec s16le 2ch 44100Hz and channel map front-left,front-right ( 12.688| 0.000) I: [pulseaudio] source.c: alsa.resolution_bits = "16" ( 12.688| 0.000) I: [pulseaudio] source.c: device.api = "alsa" ( 12.688| 0.000) I: [pulseaudio] source.c: device.class = "sound" ( 12.688| 0.000) I: [pulseaudio] source.c: alsa.class = "generic" ( 12.688| 0.000) I: [pulseaudio] source.c: alsa.subclass = "generic-mix" ( 12.688| 0.000) I: [pulseaudio] source.c: alsa.name = "ALC889 Analog" ( 12.688| 0.000) I: [pulseaudio] source.c: alsa.id = "ALC889 Analog" ( 12.688| 0.000) I: [pulseaudio] source.c: alsa.subdevice = "0" ( 12.688| 0.000) I: [pulseaudio] source.c: alsa.subdevice_name = "subdevice #0" ( 12.688| 0.000) I: [pulseaudio] source.c: alsa.device = "0" ( 12.688| 0.000) I: [pulseaudio] source.c: alsa.card = "0" ( 12.688| 0.000) I: [pulseaudio] source.c: alsa.card_name = "HDA Intel PCH" ( 12.688| 0.000) I: [pulseaudio] source.c: alsa.long_card_name = "HDA Intel PCH at 0xf7420000 irq 54" ( 12.688| 0.000) I: [pulseaudio] source.c: alsa.driver_name = "snd_hda_intel" ( 12.688| 0.000) I: [pulseaudio] source.c: device.bus_path = "pci-0000:00:1b.0" ( 12.688| 0.000) I: [pulseaudio] source.c: sysfs.path = "/devices/pci0000:00/0000:00:1b.0/sound/card0" ( 12.688| 0.000) I: [pulseaudio] source.c: device.bus = "pci" ( 12.688| 0.000) I: [pulseaudio] source.c: device.vendor.id = "8086" ( 12.688| 0.000) I: [pulseaudio] source.c: device.vendor.name = "Intel Corporation" ( 12.688| 0.000) I: [pulseaudio] source.c: device.product.name = "6 Series/C200 Series Chipset Family High Definition Audio Controller" ( 12.688| 0.000) I: [pulseaudio] source.c: device.form_factor = "internal" ( 12.688| 0.000) I: [pulseaudio] source.c: device.string = "front:0" ( 12.688| 0.000) I: [pulseaudio] source.c: device.buffering.buffer_size = "8112" ( 12.688| 0.000) I: [pulseaudio] source.c: device.buffering.fragment_size = "4056" ( 12.688| 0.000) I: [pulseaudio] source.c: device.access_mode = "mmap" ( 12.688| 0.000) I: [pulseaudio] source.c: device.profile.name = "analog-stereo" ( 12.688| 0.000) I: [pulseaudio] source.c: device.profile.description = "Analog Stereo" ( 12.688| 0.000) I: [pulseaudio] source.c: device.description = "Built-in Audio Analog Stereo" ( 12.688| 0.000) I: [pulseaudio] source.c: alsa.mixer_name = "Realtek ALC889" ( 12.688| 0.000) I: [pulseaudio] source.c: alsa.components = "HDA:10ec0889,1043846b,00100004" ( 12.688| 0.000) I: [pulseaudio] source.c: module-udev-detect.discovered = "1" ( 12.688| 0.000) I: [pulseaudio] source.c: device.icon_name = "audio-card-pci" ( 12.688| 0.000) I: [pulseaudio] alsa-source.c: Using 2.0 fragments of size 4056 bytes (22.99ms), buffer size is 8112 bytes (45.99ms) ( 12.688| 0.000) D: [pulseaudio] alsa-source.c: hwbuf_unused=0 ( 12.688| 0.000) D: [pulseaudio] alsa-source.c: setting avail_min=1 ( 12.688| 0.000) D: [pulseaudio] alsa-mixer.c: Activating path analog-input-microphone-front ( 12.688| 0.000) D: [pulseaudio] alsa-mixer.c: Path analog-input-microphone-front (Front Microphone), direction=2, priority=85, probed=yes, supported=yes, has_mute=yes, has_volume=yes, has_dB=yes, min_volume=0, max_volume=3, min_dB=-16, max_dB=60 ( 12.688| 0.000) D: [pulseaudio] alsa-mixer.c: Element Front Mic Boost, direction=2, switch=0, volume=1, volume_limit=-1, enumeration=0, required=0, required_any=4, required_absent=0, mask=0x3600000000f66, n_channels=2, override_map=yes ( 12.688| 0.000) D: [pulseaudio] alsa-mixer.c: Element Capture, direction=2, switch=1, volume=1, volume_limit=-1, enumeration=0, required=0, required_any=0, required_absent=0, mask=0x3600000000f66, n_channels=2, override_map=yes ( 12.688| 0.000) D: [pulseaudio] alsa-mixer.c: Element Input Source, direction=2, switch=0, volume=0, volume_limit=-1, enumeration=1, required=0, required_any=0, required_absent=0, mask=0x0, n_channels=0, override_map=no ( 12.688| 0.000) D: [pulseaudio] alsa-mixer.c: Option Front Mic (analog-input-microphone-front/analog-input-microphone-front) index=0, priority=0 ( 12.688| 0.000) D: [pulseaudio] alsa-mixer.c: Element Rear Mic Boost, direction=2, switch=0, volume=2, volume_limit=-1, enumeration=0, required=0, required_any=0, required_absent=0, mask=0x6, n_channels=2, override_map=no ( 12.688| 0.000) D: [pulseaudio] alsa-mixer.c: Element Line Boost, direction=2, switch=0, volume=2, volume_limit=-1, enumeration=0, required=0, required_any=0, required_absent=0, mask=0x6, n_channels=2, override_map=no ( 12.688| 0.000) D: [pulseaudio] alsa-mixer.c: Jack Front Mic, alsa_name='Front Mic Jack', detection possible ( 12.688| 0.000) D: [pulseaudio] alsa-mixer.c: Setting analog-input-microphone-front (analog-input-microphone-front) priority=0 ( 12.688| 0.000) I: [pulseaudio] alsa-source.c: Successfully enabled deferred volume. ( 12.688| 0.000) I: [pulseaudio] alsa-source.c: Hardware volume ranges from -16.00 dB to 60.00 dB. ( 12.688| 0.000) I: [pulseaudio] alsa-source.c: Fixing base volume to -60.00 dB ( 12.688| 0.000) I: [pulseaudio] alsa-source.c: Using hardware volume control. Hardware dB scale supported. ( 12.688| 0.000) I: [pulseaudio] alsa-source.c: Using hardware mute control. ( 12.688| 0.000) D: [pulseaudio] alsa-util.c: snd_pcm_dump(): ( 12.688| 0.000) D: [pulseaudio] alsa-util.c: Hardware PCM card 0 'HDA Intel PCH' device 0 subdevice 0 ( 12.688| 0.000) D: [pulseaudio] alsa-util.c: Its setup is: ( 12.688| 0.000) D: [pulseaudio] alsa-util.c: stream : CAPTURE ( 12.688| 0.000) D: [pulseaudio] alsa-util.c: access : MMAP_INTERLEAVED ( 12.688| 0.000) D: [pulseaudio] alsa-util.c: format : S16_LE ( 12.688| 0.000) D: [pulseaudio] alsa-util.c: subformat : STD ( 12.688| 0.000) D: [pulseaudio] alsa-util.c: channels : 2 ( 12.688| 0.000) D: [pulseaudio] alsa-util.c: rate : 44100 ( 12.688| 0.000) D: [pulseaudio] alsa-util.c: exact rate : 44100 (44100/1) ( 12.688| 0.000) D: [pulseaudio] alsa-util.c: msbits : 16 ( 12.688| 0.000) D: [pulseaudio] alsa-util.c: buffer_size : 2028 ( 12.688| 0.000) D: [pulseaudio] alsa-util.c: period_size : 1014 ( 12.688| 0.000) D: [pulseaudio] alsa-util.c: period_time : 22993 ( 12.688| 0.000) D: [pulseaudio] alsa-util.c: tstamp_mode : ENABLE ( 12.688| 0.000) D: [pulseaudio] alsa-util.c: period_step : 1 ( 12.688| 0.000) D: [pulseaudio] alsa-util.c: avail_min : 1014 ( 12.688| 0.000) D: [pulseaudio] alsa-util.c: period_event : 1 ( 12.688| 0.000) D: [pulseaudio] alsa-util.c: start_threshold : -1 ( 12.688| 0.000) D: [pulseaudio] alsa-util.c: stop_threshold : 9133300044307365888 ( 12.688| 0.000) D: [pulseaudio] alsa-util.c: silence_threshold: 0 ( 12.688| 0.000) D: [pulseaudio] alsa-util.c: silence_size : 0 ( 12.688| 0.000) D: [pulseaudio] alsa-util.c: boundary : 9133300044307365888 ( 12.688| 0.000) D: [pulseaudio] alsa-util.c: appl_ptr : 0 ( 12.688| 0.000) D: [pulseaudio] alsa-util.c: hw_ptr : 0 ( 12.688| 0.000) D: [pulseaudio] alsa-source.c: Read hardware volume: 0: 5% 1: 5% ( 12.688| 0.000) D: [alsa-source] alsa-source.c: Thread starting up ( 12.688| 0.000) D: [pulseaudio] alsa-source.c: in dB: 0: -76.00 dB 1: -76.00 dB ( 12.691| 0.002) D: [alsa-source] core-util.c: RealtimeKit worked. ( 12.691| 0.000) I: [alsa-source] core-util.c: Successfully enabled SCHED_RR scheduling for thread, with priority 5. ( 12.691| 0.000) I: [alsa-source] alsa-source.c: Starting capture. ( 12.691| 0.000) D: [pulseaudio] module-alsa-card.c: Found 4 jacks. ( 12.691| 0.000) I: [pulseaudio] alsa-util.c: Successfully attached to mixer 'hw:0' ( 12.691| 0.000) D: [pulseaudio] module-alsa-card.c: Jack 'Front Headphone Jack' is now unplugged ( 12.691| 0.000) D: [pulseaudio] device-port.c: Setting port analog-output-headphones to status no ( 12.691| 0.000) D: [pulseaudio] core-subscribe.c: Dropped redundant event due to change event. ( 12.691| 0.000) D: [pulseaudio] module-alsa-card.c: Jack 'Front Mic Jack' is now unplugged ( 12.691| 0.000) D: [pulseaudio] device-port.c: Setting port analog-input-microphone-front to status no ( 12.691| 0.000) D: [pulseaudio] core-subscribe.c: Dropped redundant event due to change event. ( 12.691| 0.000) D: [pulseaudio] module-alsa-card.c: Jack 'Rear Mic Jack' is now unplugged ( 12.691| 0.000) D: [pulseaudio] device-port.c: Setting port analog-input-microphone-rear to status no ( 12.691| 0.000) D: [pulseaudio] core-subscribe.c: Dropped redundant event due to change event. ( 12.691| 0.000) D: [pulseaudio] module-alsa-card.c: Jack 'Line Jack' is now unplugged ( 12.691| 0.000) D: [pulseaudio] device-port.c: Setting port analog-input-linein to status no ( 12.691| 0.000) D: [pulseaudio] core-subscribe.c: Dropped redundant event due to change event. ( 12.691| 0.000) I: [pulseaudio] module.c: Loaded "module-alsa-card" (index: #5; argument: "device_id="0" name="pci-0000_00_1b.0" card_name="alsa_card.pci-0000_00_1b.0" namereg_fail=false tsched=no fixed_latency_range=no ignore_dB=no deferred_volume=yes use_ucm=yes card_properties="module-udev-detect.discovered=1""). ( 12.691| 0.000) I: [pulseaudio] module-udev-detect.c: Card /devices/pci0000:00/0000:00:1b.0/sound/card0 (alsa_card.pci-0000_00_1b.0) module loaded. ( 12.691| 0.000) I: [pulseaudio] module-udev-detect.c: Found 2 cards. ( 12.691| 0.000) I: [pulseaudio] module.c: Loaded "module-udev-detect" (index: #6; argument: "tsched=0"). ( 12.691| 0.000) D: [pulseaudio] cli-command.c: Checking for existence of '/usr/lib/pulse-3.0/modules/module-jackdbus-detect.so': failure ( 12.691| 0.000) D: [pulseaudio] cli-command.c: Checking for existence of '/usr/lib/pulse-3.0/modules/module-bluetooth-policy.so': success ( 12.692| 0.000) I: [pulseaudio] module.c: Loaded "module-bluetooth-policy" (index: #7; argument: ""). ( 12.692| 0.000) D: [pulseaudio] cli-command.c: Checking for existence of '/usr/lib/pulse-3.0/modules/module-bluetooth-discover.so': success ( 12.692| 0.000) D: [pulseaudio] dbus-util.c: Successfully connected to D-Bus system bus 200a58f6723ca27a0d9b04b351ed7524 as :1.89 ( 12.693| 0.000) D: [pulseaudio] bluetooth-util.c: dbus: interface=org.freedesktop.DBus, path=/org/freedesktop/DBus, member=NameAcquired ( 12.693| 0.000) I: [pulseaudio] module.c: Loaded "module-bluetooth-discover" (index: #8; argument: ""). ( 12.693| 0.000) D: [pulseaudio] cli-command.c: Checking for existence of '/usr/lib/pulse-3.0/modules/module-esound-protocol-unix.so': failure ( 12.694| 0.000) I: [pulseaudio] module.c: Loaded "module-native-protocol-unix" (index: #9; argument: ""). ( 12.694| 0.000) D: [pulseaudio] cli-command.c: Checking for existence of '/usr/lib/pulse-3.0/modules/module-gconf.so': failure ( 12.694| 0.000) D: [pulseaudio] core-subscribe.c: Dropped redundant event due to change event. ( 12.694| 0.000) I: [pulseaudio] module-default-device-restore.c: Restored default sink 'alsa_output.pci-0000_00_1b.0.analog-stereo'. ( 12.694| 0.000) D: [pulseaudio] core-subscribe.c: Dropped redundant event due to change event. ( 12.694| 0.000) I: [pulseaudio] module-default-device-restore.c: Restored default source 'alsa_input.pci-0000_00_1b.0.analog-stereo'. ( 12.694| 0.000) I: [pulseaudio] module.c: Loaded "module-default-device-restore" (index: #10; argument: ""). ( 12.694| 0.000) I: [pulseaudio] module.c: Loaded "module-rescue-streams" (index: #11; argument: ""). ( 12.694| 0.000) I: [pulseaudio] module.c: Loaded "module-always-sink" (index: #12; argument: ""). ( 12.694| 0.000) I: [pulseaudio] module.c: Loaded "module-intended-roles" (index: #13; argument: ""). ( 12.695| 0.000) D: [pulseaudio] module-suspend-on-idle.c: Sink alsa_output.pci-0000_01_00.1.hdmi-stereo-extra1 becomes idle, timeout in 5 seconds. ( 12.695| 0.000) D: [pulseaudio] module-suspend-on-idle.c: Sink alsa_output.pci-0000_00_1b.0.analog-stereo becomes idle, timeout in 5 seconds. ( 12.695| 0.000) D: [pulseaudio] module-suspend-on-idle.c: Source alsa_input.pci-0000_00_1b.0.analog-stereo becomes idle, timeout in 5 seconds. ( 12.695| 0.000) I: [pulseaudio] module.c: Loaded "module-suspend-on-idle" (index: #14; argument: ""). ( 12.695| 0.000) D: [pulseaudio] cli-command.c: Checking for existence of '/usr/lib/pulse-3.0/modules/module-console-kit.so': success ( 12.695| 0.000) I: [pulseaudio] client.c: Created 0 "ConsoleKit Session /org/freedesktop/ConsoleKit/Session2" ( 12.695| 0.000) D: [pulseaudio] module-console-kit.c: Added new session /org/freedesktop/ConsoleKit/Session2 ( 12.695| 0.000) I: [pulseaudio] module.c: Loaded "module-console-kit" (index: #15; argument: ""). ( 12.695| 0.000) D: [pulseaudio] cli-command.c: Checking for existence of '/usr/lib/pulse-3.0/modules/module-systemd-login.so': failure ( 12.695| 0.000) I: [pulseaudio] module.c: Loaded "module-position-event-sounds" (index: #16; argument: ""). ( 12.696| 0.000) I: [pulseaudio] module.c: Loaded "module-filter-heuristics" (index: #17; argument: ""). ( 12.696| 0.000) I: [pulseaudio] module.c: Loaded "module-filter-apply" (index: #18; argument: ""). ( 12.696| 0.000) D: [pulseaudio] module-switch-on-port-available.c: finding port hdmi-output-0 ( 12.696| 0.000) D: [pulseaudio] module-switch-on-port-available.c: finding port hdmi-output-1 ( 12.696| 0.000) D: [pulseaudio] module-switch-on-port-available.c: finding port hdmi-output-2 ( 12.696| 0.000) D: [pulseaudio] module-switch-on-port-available.c: finding port hdmi-output-3 ( 12.696| 0.000) D: [pulseaudio] module-switch-on-port-available.c: finding port analog-input-microphone-front ( 12.696| 0.000) D: [pulseaudio] module-switch-on-port-available.c: finding port analog-input-microphone-rear ( 12.696| 0.000) D: [pulseaudio] module-switch-on-port-available.c: finding port analog-input-linein ( 12.696| 0.000) D: [pulseaudio] module-switch-on-port-available.c: finding port analog-output-headphones ( 12.696| 0.000) I: [pulseaudio] module.c: Loaded "module-switch-on-port-available" (index: #19; argument: ""). ( 12.696| 0.000) D: [pulseaudio] main.c: Got org.PulseAudio1! ( 12.697| 0.000) D: [pulseaudio] main.c: Got org.pulseaudio.Server! ( 12.697| 0.000) I: [pulseaudio] main.c: Daemon startup complete. ( 12.697| 0.000) I: [pulseaudio] client.c: Created 1 "Native client (UNIX socket client)" ( 12.697| 0.000) D: [pulseaudio] module-udev-detect.c: /dev/snd/controlC1 is accessible: yes ( 12.697| 0.000) D: [pulseaudio] module-udev-detect.c: /dev/snd/controlC0 is accessible: yes ( 12.697| 0.000) I: [pulseaudio] client.c: Created 2 "Native client (UNIX socket client)" ( 12.697| 0.000) D: [pulseaudio] protocol-native.c: Protocol version: remote 27, local 27 ( 12.697| 0.000) I: [pulseaudio] protocol-native.c: Got credentials: uid=1000 gid=1000 success=1 ( 12.697| 0.000) D: [pulseaudio] protocol-native.c: SHM possible: yes ( 12.697| 0.000) D: [pulseaudio] protocol-native.c: Negotiated SHM: yes ( 12.697| 0.000) D: [pulseaudio] protocol-native.c: Protocol version: remote 27, local 27 ( 12.697| 0.000) I: [pulseaudio] protocol-native.c: Got credentials: uid=1000 gid=1000 success=1 ( 12.697| 0.000) D: [pulseaudio] protocol-native.c: SHM possible: yes ( 12.697| 0.000) D: [pulseaudio] protocol-native.c: Negotiated SHM: yes ( 12.697| 0.000) D: [pulseaudio] module-augment-properties.c: Looking for .desktop file for indicator-sound-service ( 12.697| 0.000) D: [pulseaudio] module-augment-properties.c: Looking for .desktop file for gnome-settings-daemon ( 17.697| 5.000) I: [pulseaudio] module-suspend-on-idle.c: Source alsa_input.pci-0000_00_1b.0.analog-stereo idle for too long, suspending ... ( 17.698| 0.000) D: [pulseaudio] source.c: Suspend cause of source alsa_input.pci-0000_00_1b.0.analog-stereo is 0x0004, suspending ( 17.698| 0.000) I: [alsa-source] alsa-source.c: Device suspended... ( 17.698| 0.000) D: [pulseaudio] core.c: Hmm, no streams around, trying to vacuum. ( 17.698| 0.000) I: [pulseaudio] module-suspend-on-idle.c: Sink alsa_output.pci-0000_00_1b.0.analog-stereo idle for too long, suspending ... ( 17.698| 0.000) D: [pulseaudio] sink.c: Suspend cause of sink alsa_output.pci-0000_00_1b.0.analog-stereo is 0x0004, suspending ( 17.699| 0.001) I: [alsa-sink] alsa-sink.c: Device suspended... ( 17.699| 0.000) D: [pulseaudio] core.c: Hmm, no streams around, trying to vacuum. ( 17.699| 0.000) I: [pulseaudio] module-suspend-on-idle.c: Sink alsa_output.pci-0000_01_00.1.hdmi-stereo-extra1 idle for too long, suspending ... ( 17.699| 0.000) D: [pulseaudio] sink.c: Suspend cause of sink alsa_output.pci-0000_01_00.1.hdmi-stereo-extra1 is 0x0004, suspending ( 17.700| 0.000) I: [alsa-sink] alsa-sink.c: Device suspended... ( 17.700| 0.000) D: [pulseaudio] core.c: Hmm, no streams around, trying to vacuum. ( 17.700| 0.000) D: [pulseaudio] module-udev-detect.c: /dev/snd/controlC1 is accessible: yes ( 17.700| 0.000) D: [pulseaudio] module-udev-detect.c: /dev/snd/controlC0 is accessible: yes ( 19.383| 1.683) I: [pulseaudio] client.c: Created 3 "Native client (UNIX socket client)" ( 19.537| 0.153) D: [pulseaudio] protocol-native.c: Protocol version: remote 27, local 27 ( 19.537| 0.000) I: [pulseaudio] protocol-native.c: Got credentials: uid=1000 gid=1000 success=1 ( 19.537| 0.000) D: [pulseaudio] protocol-native.c: SHM possible: yes ( 19.537| 0.000) D: [pulseaudio] protocol-native.c: Negotiated SHM: yes ( 19.565| 0.028) D: [pulseaudio] module-augment-properties.c: Looking for .desktop file for gnome-control-center ( 19.565| 0.000) D: [pulseaudio] module-augment-properties.c: Found /usr/share/applications/gnome-control-center.desktop. ( 30.916| 11.350) I: [pulseaudio] client.c: Freed 3 "GNOME Volume Control Dialog" ( 30.916| 0.000) I: [pulseaudio] protocol-native.c: Connection died. ( 71.860| 40.944) I: [pulseaudio] main.c: Got signal SIGINT. ( 71.860| 0.000) I: [pulseaudio] main.c: Exiting. ( 71.860| 0.000) I: [pulseaudio] main.c: Daemon shutdown initiated. ( 71.860| 0.000) I: [pulseaudio] module.c: Unloading "module-device-restore" (index: #0). ( 71.860| 0.000) I: [pulseaudio] module.c: Unloaded "module-device-restore" (index: #0). ( 71.860| 0.000) I: [pulseaudio] module.c: Unloading "module-stream-restore" (index: #1). ( 71.860| 0.000) D: [pulseaudio] protocol-dbus.c: Interface org.PulseAudio.Ext.StreamRestore1 removed from object /org/pulseaudio/stream_restore1 ( 71.860| 0.000) D: [pulseaudio] protocol-dbus.c: Interface org.PulseAudio.Ext.StreamRestore1.RestoreEntry removed from object /org/pulseaudio/stream_restore1/entry0 ( 71.860| 0.000) D: [pulseaudio] protocol-dbus.c: Interface org.PulseAudio.Ext.StreamRestore1.RestoreEntry removed from object /org/pulseaudio/stream_restore1/entry1 ( 71.860| 0.000) D: [pulseaudio] protocol-dbus.c: Interface org.PulseAudio.Ext.StreamRestore1.RestoreEntry removed from object /org/pulseaudio/stream_restore1/entry2 ( 71.860| 0.000) D: [pulseaudio] protocol-dbus.c: Interface org.PulseAudio.Ext.StreamRestore1.RestoreEntry removed from object /org/pulseaudio/stream_restore1/entry3 ( 71.860| 0.000) D: [pulseaudio] protocol-dbus.c: Interface org.PulseAudio.Ext.StreamRestore1.RestoreEntry removed from object /org/pulseaudio/stream_restore1/entry4 ( 71.860| 0.000) D: [pulseaudio] protocol-dbus.c: Interface org.PulseAudio.Ext.StreamRestore1.RestoreEntry removed from object /org/pulseaudio/stream_restore1/entry5 ( 71.860| 0.000) D: [pulseaudio] protocol-dbus.c: Interface org.PulseAudio.Ext.StreamRestore1.RestoreEntry removed from object /org/pulseaudio/stream_restore1/entry6 ( 71.860| 0.000) D: [pulseaudio] protocol-dbus.c: Interface org.PulseAudio.Ext.StreamRestore1.RestoreEntry removed from object /org/pulseaudio/stream_restore1/entry7 ( 71.860| 0.000) D: [pulseaudio] protocol-dbus.c: Interface org.PulseAudio.Ext.StreamRestore1.RestoreEntry removed from object /org/pulseaudio/stream_restore1/entry8 ( 71.860| 0.000) D: [pulseaudio] protocol-dbus.c: Interface org.PulseAudio.Ext.StreamRestore1.RestoreEntry removed from object /org/pulseaudio/stream_restore1/entry9 ( 71.860| 0.000) D: [pulseaudio] protocol-dbus.c: Interface org.PulseAudio.Ext.StreamRestore1.RestoreEntry removed from object /org/pulseaudio/stream_restore1/entry10 ( 71.860| 0.000) D: [pulseaudio] protocol-dbus.c: Interface org.PulseAudio.Ext.StreamRestore1.RestoreEntry removed from object /org/pulseaudio/stream_restore1/entry11 ( 71.860| 0.000) D: [pulseaudio] protocol-dbus.c: Interface org.PulseAudio.Ext.StreamRestore1.RestoreEntry removed from object /org/pulseaudio/stream_restore1/entry12 ( 71.860| 0.000) D: [pulseaudio] protocol-dbus.c: Interface org.PulseAudio.Ext.StreamRestore1.RestoreEntry removed from object /org/pulseaudio/stream_restore1/entry13 ( 71.860| 0.000) D: [pulseaudio] protocol-dbus.c: Interface org.PulseAudio.Ext.StreamRestore1.RestoreEntry removed from object /org/pulseaudio/stream_restore1/entry14 ( 71.860| 0.000) D: [pulseaudio] protocol-dbus.c: Interface org.PulseAudio.Ext.StreamRestore1.RestoreEntry removed from object /org/pulseaudio/stream_restore1/entry15 ( 71.860| 0.000) D: [pulseaudio] protocol-dbus.c: Interface org.PulseAudio.Ext.StreamRestore1.RestoreEntry removed from object /org/pulseaudio/stream_restore1/entry16 ( 71.860| 0.000) D: [pulseaudio] protocol-dbus.c: Interface org.PulseAudio.Ext.StreamRestore1.RestoreEntry removed from object /org/pulseaudio/stream_restore1/entry17 ( 71.860| 0.000) D: [pulseaudio] protocol-dbus.c: Interface org.PulseAudio.Ext.StreamRestore1.RestoreEntry removed from object /org/pulseaudio/stream_restore1/entry18 ( 71.860| 0.000) D: [pulseaudio] protocol-dbus.c: Interface org.PulseAudio.Ext.StreamRestore1.RestoreEntry removed from object /org/pulseaudio/stream_restore1/entry19 ( 71.860| 0.000) D: [pulseaudio] protocol-dbus.c: Interface org.PulseAudio.Ext.StreamRestore1.RestoreEntry removed from object /org/pulseaudio/stream_restore1/entry20 ( 71.860| 0.000) D: [pulseaudio] protocol-dbus.c: Interface org.PulseAudio.Ext.StreamRestore1.RestoreEntry removed from object /org/pulseaudio/stream_restore1/entry21 ( 71.860| 0.000) D: [pulseaudio] flist.c: pulsecore/hashmap.c: entries flist is full (don't worry) ( 71.861| 0.000) D: [pulseaudio] flist.c: pulsecore/hashmap.c: entries flist is full (don't worry) ( 71.861| 0.000) D: [pulseaudio] protocol-dbus.c: Interface org.PulseAudio.Ext.StreamRestore1.RestoreEntry removed from object /org/pulseaudio/stream_restore1/entry22 ( 71.861| 0.000) D: [pulseaudio] flist.c: pulsecore/hashmap.c: entries flist is full (don't worry) ( 71.861| 0.000) D: [pulseaudio] flist.c: pulsecore/hashmap.c: entries flist is full (don't worry) ( 71.861| 0.000) D: [pulseaudio] flist.c: pulsecore/hashmap.c: entries flist is full (don't worry) ( 71.861| 0.000) D: [pulseaudio] flist.c: pulsecore/hashmap.c: entries flist is full (don't worry) ( 71.861| 0.000) D: [pulseaudio] flist.c: pulsecore/hashmap.c: entries flist is full (don't worry) ( 71.861| 0.000) D: [pulseaudio] flist.c: pulsecore/hashmap.c: entries flist is full (don't worry) ( 71.861| 0.000) D: [pulseaudio] flist.c: pulsecore/hashmap.c: entries flist is full (don't worry) ( 71.861| 0.000) D: [pulseaudio] flist.c: pulsecore/hashmap.c: entries flist is full (don't worry) ( 71.861| 0.000) D: [pulseaudio] flist.c: pulsecore/hashmap.c: entries flist is full (don't worry) ( 71.861| 0.000) D: [pulseaudio] protocol-dbus.c: Interface org.PulseAudio.Ext.StreamRestore1.RestoreEntry removed from object /org/pulseaudio/stream_restore1/entry23 ( 71.861| 0.000) D: [pulseaudio] protocol-dbus.c: Interface org.PulseAudio.Ext.StreamRestore1.RestoreEntry removed from object /org/pulseaudio/stream_restore1/entry24 ( 71.861| 0.000) D: [pulseaudio] protocol-dbus.c: Interface org.PulseAudio.Ext.StreamRestore1.RestoreEntry removed from object /org/pulseaudio/stream_restore1/entry25 ( 71.861| 0.000) D: [pulseaudio] protocol-dbus.c: Interface org.PulseAudio.Ext.StreamRestore1.RestoreEntry removed from object /org/pulseaudio/stream_restore1/entry26 ( 71.861| 0.000) D: [pulseaudio] protocol-dbus.c: Interface org.PulseAudio.Ext.StreamRestore1.RestoreEntry removed from object /org/pulseaudio/stream_restore1/entry27 ( 71.861| 0.000) I: [pulseaudio] module.c: Unloaded "module-stream-restore" (index: #1). ( 71.861| 0.000) I: [pulseaudio] module.c: Unloading "module-card-restore" (index: #2). ( 71.861| 0.000) I: [pulseaudio] module.c: Unloaded "module-card-restore" (index: #2). ( 71.861| 0.000) I: [pulseaudio] module.c: Unloading "module-augment-properties" (index: #3). ( 71.861| 0.000) I: [pulseaudio] module.c: Unloaded "module-augment-properties" (index: #3). ( 71.861| 0.000) I: [pulseaudio] module.c: Unloading "module-alsa-card" (index: #4). ( 71.861| 0.000) D: [pulseaudio] module-rescue-streams.c: No sink inputs to move away. ( 71.861| 0.000) D: [pulseaudio] module-rescue-streams.c: No source outputs to move away. ( 71.861| 0.000) D: [alsa-sink] alsa-sink.c: Thread shutting down ( 71.861| 0.000) I: [pulseaudio] sink.c: Freeing sink 0 "alsa_output.pci-0000_01_00.1.hdmi-stereo-extra1" ( 71.861| 0.000) I: [pulseaudio] source.c: Freeing source 0 "alsa_output.pci-0000_01_00.1.hdmi-stereo-extra1.monitor" ( 71.861| 0.000) I: [pulseaudio] card.c: Freed 0 "alsa_card.pci-0000_01_00.1" ( 71.861| 0.000) I: [pulseaudio] module.c: Unloaded "module-alsa-card" (index: #4). ( 71.861| 0.000) I: [pulseaudio] module.c: Unloading "module-alsa-card" (index: #5). ( 71.861| 0.000) D: [pulseaudio] module-always-sink.c: Autoloading null-sink as no other sinks detected. ( 71.861| 0.000) I: [pulseaudio] sink.c: Created sink 2 "auto_null" with sample spec s16le 2ch 44100Hz and channel map front-left,front-right ( 71.861| 0.000) I: [pulseaudio] sink.c: device.description = "Dummy Output" ( 71.861| 0.000) I: [pulseaudio] sink.c: device.class = "abstract" ( 71.861| 0.000) I: [pulseaudio] sink.c: device.icon_name = "audio-card" ( 71.862| 0.000) I: [pulseaudio] source.c: Created source 3 "auto_null.monitor" with sample spec s16le 2ch 44100Hz and channel map front-left,front-right ( 71.862| 0.000) I: [pulseaudio] source.c: device.description = "Monitor of Dummy Output" ( 71.862| 0.000) I: [pulseaudio] source.c: device.class = "monitor" ( 71.862| 0.000) I: [pulseaudio] source.c: device.icon_name = "audio-input-microphone" ( 71.862| 0.000) D: [null-sink] module-null-sink.c: Thread starting up ( 71.862| 0.000) D: [pulseaudio] module-suspend-on-idle.c: Sink auto_null becomes idle, timeout in 5 seconds. ( 71.862| 0.000) I: [pulseaudio] module.c: Loaded "module-null-sink" (index: #20; argument: "sink_name=auto_null sink_properties='device.description="Dummy Output"'"). ( 71.862| 0.000) D: [pulseaudio] module-rescue-streams.c: No sink inputs to move away. ( 71.862| 0.000) D: [pulseaudio] module-rescue-streams.c: No source outputs to move away. ( 71.862| 0.000) D: [alsa-sink] alsa-sink.c: Thread shutting down ( 71.862| 0.000) I: [pulseaudio] sink.c: Freeing sink 1 "alsa_output.pci-0000_00_1b.0.analog-stereo" ( 71.862| 0.000) I: [pulseaudio] source.c: Freeing source 1 "alsa_output.pci-0000_00_1b.0.analog-stereo.monitor" ( 71.862| 0.000) D: [pulseaudio] module-rescue-streams.c: No source outputs to move away. ( 71.862| 0.000) D: [pulseaudio] core-subscribe.c: Dropped redundant event due to change event. ( 71.862| 0.000) D: [alsa-source] alsa-source.c: Thread shutting down ( 71.862| 0.000) I: [pulseaudio] source.c: Freeing source 2 "alsa_input.pci-0000_00_1b.0.analog-stereo" ( 71.862| 0.000) I: [pulseaudio] card.c: Freed 1 "alsa_card.pci-0000_00_1b.0" ( 71.862| 0.000) I: [pulseaudio] module.c: Unloaded "module-alsa-card" (index: #5). ( 71.862| 0.000) I: [pulseaudio] module.c: Unloading "module-udev-detect" (index: #6). ( 71.862| 0.000) I: [pulseaudio] module.c: Unloaded "module-udev-detect" (index: #6). ( 71.862| 0.000) I: [pulseaudio] module.c: Unloading "module-bluetooth-policy" (index: #7). ( 71.862| 0.000) I: [pulseaudio] module.c: Unloaded "module-bluetooth-policy" (index: #7). ( 71.862| 0.000) I: [pulseaudio] module.c: Unloading "module-bluetooth-discover" (index: #8). ( 71.863| 0.000) I: [pulseaudio] module.c: Unloaded "module-bluetooth-discover" (index: #8). ( 71.863| 0.000) I: [pulseaudio] module.c: Unloading "module-native-protocol-unix" (index: #9). ( 71.863| 0.000) I: [pulseaudio] client.c: Freed 1 "Indicator Sound" ( 71.863| 0.000) I: [pulseaudio] client.c: Freed 2 "GNOME Volume Control Media Keys" ( 71.863| 0.000) I: [pulseaudio] module.c: Unloaded "module-native-protocol-unix" (index: #9). ( 71.863| 0.000) I: [pulseaudio] module.c: Unloading "module-default-device-restore" (index: #10). ( 71.863| 0.000) I: [pulseaudio] module.c: Unloaded "module-default-device-restore" (index: #10). ( 71.863| 0.000) I: [pulseaudio] module.c: Unloading "module-rescue-streams" (index: #11). ( 71.863| 0.000) I: [pulseaudio] module.c: Unloaded "module-rescue-streams" (index: #11). ( 71.863| 0.000) I: [pulseaudio] module.c: Unloading "module-always-sink" (index: #12). ( 71.863| 0.000) I: [pulseaudio] module.c: Unloaded "module-always-sink" (index: #12). ( 71.863| 0.000) I: [pulseaudio] module.c: Unloading "module-intended-roles" (index: #13). ( 71.863| 0.000) I: [pulseaudio] module.c: Unloaded "module-intended-roles" (index: #13). ( 71.863| 0.000) I: [pulseaudio] module.c: Unloading "module-suspend-on-idle" (index: #14). ( 71.863| 0.000) I: [pulseaudio] module.c: Unloaded "module-suspend-on-idle" (index: #14). ( 71.863| 0.000) I: [pulseaudio] module.c: Unloading "module-console-kit" (index: #15). ( 71.863| 0.000) D: [pulseaudio] module-console-kit.c: Removing session /org/freedesktop/ConsoleKit/Session2 ( 71.863| 0.000) I: [pulseaudio] client.c: Freed 0 "ConsoleKit Session /org/freedesktop/ConsoleKit/Session2" ( 71.863| 0.000) I: [pulseaudio] module.c: Unloaded "module-console-kit" (index: #15). ( 71.863| 0.000) I: [pulseaudio] module.c: Unloading "module-position-event-sounds" (index: #16). ( 71.863| 0.000) I: [pulseaudio] module.c: Unloaded "module-position-event-sounds" (index: #16). ( 71.863| 0.000) I: [pulseaudio] module.c: Unloading "module-filter-heuristics" (index: #17). ( 71.863| 0.000) I: [pulseaudio] module.c: Unloaded "module-filter-heuristics" (index: #17). ( 71.863| 0.000) I: [pulseaudio] module.c: Unloading "module-filter-apply" (index: #18). ( 71.863| 0.000) I: [pulseaudio] module.c: Unloaded "module-filter-apply" (index: #18). ( 71.863| 0.000) I: [pulseaudio] module.c: Unloading "module-switch-on-port-available" (index: #19). ( 71.863| 0.000) I: [pulseaudio] module.c: Unloaded "module-switch-on-port-available" (index: #19). ( 71.863| 0.000) I: [pulseaudio] module.c: Unloading "module-null-sink" (index: #20). ( 71.863| 0.000) D: [pulseaudio] core-subscribe.c: Dropped redundant event due to remove event. ( 71.863| 0.000) D: [pulseaudio] core-subscribe.c: Dropped redundant event due to remove event. ( 71.863| 0.000) D: [pulseaudio] core-subscribe.c: Dropped redundant event due to remove event. ( 71.863| 0.000) D: [pulseaudio] core-subscribe.c: Dropped redundant event due to remove event. ( 71.864| 0.000) D: [null-sink] module-null-sink.c: Thread shutting down ( 71.864| 0.000) I: [pulseaudio] sink.c: Freeing sink 2 "auto_null" ( 71.864| 0.000) I: [pulseaudio] source.c: Freeing source 3 "auto_null.monitor" ( 71.864| 0.000) I: [pulseaudio] module.c: Unloaded "module-null-sink" (index: #20). ( 71.864| 0.000) D: [pulseaudio] core-subscribe.c: Dropped redundant event due to remove event. ( 71.864| 0.000) I: [pulseaudio] main.c: Daemon terminated.

1 0

[alsa-devel] [PATCH] ALSA: hda - hdmi: Fallback to ALSA allocation when selecting CA
by Anssi Hannula 09 Sep '13

09 Sep '13

hdmi_channel_allocation() tries to find a HDMI channel allocation that matches the number channels in the playback stream and contains only speakers that the HDMI sink has reported as available via EDID. If no such allocation is found, 0 (stereo audio) is used. Using CA 0 causes the audio causes the sink to discard everything except the first two channels (front left and front right). However, the sink may be capable of receiving more channels than it has speakers (and then perform downmix or discard the extra channels), in which case it is preferable to use a CA that contains extra channels than to use CA 0 which discards all the non-stereo channels. Additionally, it seems that HBR (HD) passthrough output does not work on Intel HDMI codecs when CA is set to 0 (possibly the codec zeroes channels not present in CA). This happens with all receivers that report a 5.1 speaker mask since a HBR stream is carried on 8 channels to the codec. Add a fallback in the CA selection so that the CA channel count at least matches the stream channel count, even if the stream contains channels not present in the sink speaker descriptor. Thanks to GrimGriefer at OpenELEC forums for discovering that changing the sink speaker mask allowed HBR output. Reported-by: GrimGriefer Reported-by: Ashecrow Reported-by: Frank Zafka <kafkaesque1978(a)gmail.com> Reported-by: Peter Frühberger <fritsch(a)xbmc.org> Signed-off-by: Anssi Hannula <anssi.hannula(a)iki.fi> Cc: <stable(a)vger.kernel.org> --- Hopefully this fixes HBR (HD passthrough) for the remaining Intel users who were still experiencing problems. BTW, the hdmi_channel_allocation() logic seems also otherwise somewhat suspect to me. Shouldn't we always select an allocation matching ALSA channel mapping, instead of re-assigning channels received from userspace "randomly" to sink speakers (in case of unusual sink speaker mask)? Anyway, I let it be for now at least. sound/pci/hda/patch_hdmi.c | 11 +++++++++++ 1 file changed, 11 insertions(+) diff --git a/sound/pci/hda/patch_hdmi.c b/sound/pci/hda/patch_hdmi.c index 030ca86..354fc55 100644 --- a/sound/pci/hda/patch_hdmi.c +++ b/sound/pci/hda/patch_hdmi.c @@ -551,6 +551,17 @@ static int hdmi_channel_allocation(struct hdmi_eld *eld, int channels) } } + if (!ca) { + /* if there was no match, select the regular ALSA channel + * allocation with the matching number of channels */ + for (i = 0; i < ARRAY_SIZE(channel_allocations); i++) { + if (channels == channel_allocations[i].channels) { + ca = channel_allocations[i].ca_index; + break; + } + } + } + snd_print_channel_allocation(eld->info.spk_alloc, buf, sizeof(buf)); snd_printdd("HDMI: select CA 0x%x for %d-channel allocation: %s\n", ca, channels, buf); -- 1.8.1.5

3 6

[alsa-devel] [PATCH] codecs: mc13783: add spi errate fix
by Steffen Trumtrar 09 Sep '13

09 Sep '13

The MC13783 Chip Errata, Rev. 4 says, that depending on SPI clock and main audio clock speed, the Audio Codec or Stereo DAC do sometimes not start when programmed to do so. This is due to an internal clock timing issue related to the loading of the SPI bits into the audio block. On an i.MX27 based system, this issue lead to switched audio channels under certain circumstances: RTC + Touch + Audio are used and loaded at startup. The mentioned workaround of writing registers 40 and 41 two times is implemented here. Signed-off-by: Steffen Trumtrar <s.trumtrar(a)pengutronix.de> Cc: stable(a)vger.kernel.org --- sound/soc/codecs/mc13783.c | 4 ++++ 1 file changed, 4 insertions(+) diff --git a/sound/soc/codecs/mc13783.c b/sound/soc/codecs/mc13783.c index 4d3c8fd..e78353d 100644 --- a/sound/soc/codecs/mc13783.c +++ b/sound/soc/codecs/mc13783.c @@ -125,6 +125,10 @@ static int mc13783_write(struct snd_soc_codec *codec, ret = mc13xxx_reg_write(priv->mc13xxx, reg, value); + /* include errata fix for spi audio problems */ + if (reg == PMIC_AUDIO_CODEC || reg == PMIC_AUDIO_DAC) + ret = mc13xxx_reg_write(priv->mc13xxx, reg, value); + mc13xxx_unlock(priv->mc13xxx); return ret; -- 1.8.4.rc3

2 3

[alsa-devel] [PATCH] ASoC: rsnd: fixup flag name of rsnd_scu_platform_info
by Kuninori Morimoto 09 Sep '13

09 Sep '13

it should be *USE*, not *USB* Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> --- include/sound/rcar_snd.h | 2 +- sound/soc/sh/rcar/scu.c | 4 ++-- 2 files changed, 3 insertions(+), 3 deletions(-) diff --git a/include/sound/rcar_snd.h b/include/sound/rcar_snd.h index d35412a..fe66533 100644 --- a/include/sound/rcar_snd.h +++ b/include/sound/rcar_snd.h @@ -55,7 +55,7 @@ struct rsnd_ssi_platform_info { /* * flags */ -#define RSND_SCU_USB_HPBIF (1 << 31) /* it needs RSND_SSI_DEPENDENT */ +#define RSND_SCU_USE_HPBIF (1 << 31) /* it needs RSND_SSI_DEPENDENT */ struct rsnd_scu_platform_info { u32 flags; diff --git a/sound/soc/sh/rcar/scu.c b/sound/soc/sh/rcar/scu.c index 184d900..2df2e91 100644 --- a/sound/soc/sh/rcar/scu.c +++ b/sound/soc/sh/rcar/scu.c @@ -157,9 +157,9 @@ static int rsnd_scu_start(struct rsnd_mod *mod, int ret; /* - * SCU will be used if it has RSND_SCU_USB_HPBIF flags + * SCU will be used if it has RSND_SCU_USE_HPBIF flags */ - if (!(flags & RSND_SCU_USB_HPBIF)) { + if (!(flags & RSND_SCU_USE_HPBIF)) { /* it use PIO transter */ dev_dbg(dev, "%s%d is not used\n", rsnd_mod_name(mod), rsnd_mod_id(mod)); -- 1.7.9.5

2 1

[alsa-devel] [PATCH] ASoC: ad1980: fix the indentation
by Barry Song 09 Sep '13

09 Sep '13

Signed-off-by: Barry Song <Baohua.Song(a)csr.com> --- sound/soc/codecs/ad1980.c | 44 ++++++++++++++++++++++---------------------- 1 file changed, 22 insertions(+), 22 deletions(-) diff --git a/sound/soc/codecs/ad1980.c b/sound/soc/codecs/ad1980.c index 7257a88..be5032c 100644 --- a/sound/soc/codecs/ad1980.c +++ b/sound/soc/codecs/ad1980.c @@ -61,39 +61,39 @@ static const struct soc_enum ad1980_cap_src = SOC_ENUM_DOUBLE(AC97_REC_SEL, 8, 0, 7, ad1980_rec_sel); static const struct snd_kcontrol_new ad1980_snd_ac97_controls[] = { -SOC_DOUBLE("Master Playback Volume", AC97_MASTER, 8, 0, 31, 1), -SOC_SINGLE("Master Playback Switch", AC97_MASTER, 15, 1, 1), + SOC_DOUBLE("Master Playback Volume", AC97_MASTER, 8, 0, 31, 1), + SOC_SINGLE("Master Playback Switch", AC97_MASTER, 15, 1, 1), -SOC_DOUBLE("Headphone Playback Volume", AC97_HEADPHONE, 8, 0, 31, 1), -SOC_SINGLE("Headphone Playback Switch", AC97_HEADPHONE, 15, 1, 1), + SOC_DOUBLE("Headphone Playback Volume", AC97_HEADPHONE, 8, 0, 31, 1), + SOC_SINGLE("Headphone Playback Switch", AC97_HEADPHONE, 15, 1, 1), -SOC_DOUBLE("PCM Playback Volume", AC97_PCM, 8, 0, 31, 1), -SOC_SINGLE("PCM Playback Switch", AC97_PCM, 15, 1, 1), + SOC_DOUBLE("PCM Playback Volume", AC97_PCM, 8, 0, 31, 1), + SOC_SINGLE("PCM Playback Switch", AC97_PCM, 15, 1, 1), -SOC_DOUBLE("PCM Capture Volume", AC97_REC_GAIN, 8, 0, 31, 0), -SOC_SINGLE("PCM Capture Switch", AC97_REC_GAIN, 15, 1, 1), + SOC_DOUBLE("PCM Capture Volume", AC97_REC_GAIN, 8, 0, 31, 0), + SOC_SINGLE("PCM Capture Switch", AC97_REC_GAIN, 15, 1, 1), -SOC_SINGLE("Mono Playback Volume", AC97_MASTER_MONO, 0, 31, 1), -SOC_SINGLE("Mono Playback Switch", AC97_MASTER_MONO, 15, 1, 1), + SOC_SINGLE("Mono Playback Volume", AC97_MASTER_MONO, 0, 31, 1), + SOC_SINGLE("Mono Playback Switch", AC97_MASTER_MONO, 15, 1, 1), -SOC_SINGLE("Phone Capture Volume", AC97_PHONE, 0, 31, 1), -SOC_SINGLE("Phone Capture Switch", AC97_PHONE, 15, 1, 1), + SOC_SINGLE("Phone Capture Volume", AC97_PHONE, 0, 31, 1), + SOC_SINGLE("Phone Capture Switch", AC97_PHONE, 15, 1, 1), -SOC_SINGLE("Mic Volume", AC97_MIC, 0, 31, 1), -SOC_SINGLE("Mic Switch", AC97_MIC, 15, 1, 1), + SOC_SINGLE("Mic Volume", AC97_MIC, 0, 31, 1), + SOC_SINGLE("Mic Switch", AC97_MIC, 15, 1, 1), -SOC_SINGLE("Stereo Mic Switch", AC97_AD_MISC, 6, 1, 0), -SOC_DOUBLE("Line HP Swap Switch", AC97_AD_MISC, 10, 5, 1, 0), + SOC_SINGLE("Stereo Mic Switch", AC97_AD_MISC, 6, 1, 0), + SOC_DOUBLE("Line HP Swap Switch", AC97_AD_MISC, 10, 5, 1, 0), -SOC_DOUBLE("Surround Playback Volume", AC97_SURROUND_MASTER, 8, 0, 31, 1), -SOC_DOUBLE("Surround Playback Switch", AC97_SURROUND_MASTER, 15, 7, 1, 1), + SOC_DOUBLE("Surround Playback Volume", AC97_SURROUND_MASTER, 8, 0, 31, 1), + SOC_DOUBLE("Surround Playback Switch", AC97_SURROUND_MASTER, 15, 7, 1, 1), -SOC_DOUBLE("Center/LFE Playback Volume", AC97_CENTER_LFE_MASTER, 8, 0, 31, 1), -SOC_DOUBLE("Center/LFE Playback Switch", AC97_CENTER_LFE_MASTER, 15, 7, 1, 1), + SOC_DOUBLE("Center/LFE Playback Volume", AC97_CENTER_LFE_MASTER, 8, 0, 31, 1), + SOC_DOUBLE("Center/LFE Playback Switch", AC97_CENTER_LFE_MASTER, 15, 7, 1, 1), -SOC_ENUM("Capture Source", ad1980_cap_src), + SOC_ENUM("Capture Source", ad1980_cap_src), -SOC_SINGLE("Mic Boost Switch", AC97_MIC, 6, 1, 0), + SOC_SINGLE("Mic Boost Switch", AC97_MIC, 6, 1, 0), }; static const struct snd_soc_dapm_widget ad1980_dapm_widgets[] = { -- 1.8.2.3

2 1

[alsa-devel] [PATCH] aplay - fix two off by one errors
by Jurgen Kramer 09 Sep '13

09 Sep '13

Attached patch for aplay fixes two off-by-one errors preventing usage of the last available sample format. diff -uNrp alsa-utils-1.0.27.2/aplay/aplay.c alsa-utils-1.0.27.2-jk/aplay/aplay.c --- alsa-utils-1.0.27.2/aplay/aplay.c 2013-07-09 17:28:37.000000000 +0200 +++ alsa-utils-1.0.27.2-jk/aplay/aplay.c 2013-09-08 13:46:23.746825429 +0200 @@ -242,7 +242,7 @@ _("Usage: %s [OPTION]... [FILE]...\n" ) , command); printf(_("Recognized sample formats are:")); - for (k = 0; k < SND_PCM_FORMAT_LAST; ++k) { + for (k = 0; k <= SND_PCM_FORMAT_LAST; ++k) { const char *s = snd_pcm_format_name(k); if (s) printf(" %s", s); @@ -1114,7 +1114,7 @@ static void show_available_sample_format snd_pcm_format_t format; fprintf(stderr, "Available formats:\n"); - for (format = 0; format < SND_PCM_FORMAT_LAST; format++) { + for (format = 0; format <= SND_PCM_FORMAT_LAST; format++) { if (snd_pcm_hw_params_test_format(handle, params, format) == 0) fprintf(stderr, "- %s\n", snd_pcm_format_name(format)); } Jurgen

2 1

[alsa-devel] [PATCH] ALSA: hda - Add Toshiba Satellite C870 to MSI blacklist
by Takashi Iwai 09 Sep '13

09 Sep '13

Toshiba Satellite C870 shows interrupt problems occasionally when certain mixer controls like "Mic Switch" is toggled. This seems worked around by not using MSI. Bugzilla: https://bugzilla.novell.com/show_bug.cgi?id=833585 Cc: <stable(a)vger.kernel.org> Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/pci/hda/hda_intel.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c index c6c9829..7f88dd6 100644 --- a/sound/pci/hda/hda_intel.c +++ b/sound/pci/hda/hda_intel.c @@ -3400,6 +3400,7 @@ static struct snd_pci_quirk msi_black_list[] = { SND_PCI_QUIRK(0x1043, 0x81f2, "ASUS", 0), /* Athlon64 X2 + nvidia */ SND_PCI_QUIRK(0x1043, 0x81f6, "ASUS", 0), /* nvidia */ SND_PCI_QUIRK(0x1043, 0x822d, "ASUS", 0), /* Athlon64 X2 + nvidia MCP55 */ + SND_PCI_QUIRK(0x1179, 0xfb44, "Toshiba Satellite C870", 0), /* AMD Hudson */ SND_PCI_QUIRK(0x1849, 0x0888, "ASRock", 0), /* Athlon64 X2 + nvidia */ SND_PCI_QUIRK(0xa0a0, 0x0575, "Aopen MZ915-M", 0), /* ICH6 */ {} -- 1.8.4

1 0

[alsa-devel] Help with editing the Raw Midi data with ALSA?
by Ember Autumn Rose Leona 09 Sep '13

09 Sep '13

Dear ALSA team, I just joined the mailing list on after being referred to it. Is it possible for me to edit the Raw Midi data with ALSA... I looked a RtMidi and that opens a port but doesn't help me Midi Poison or Midi Message Map. What I want to do is edit the Midi Messages, re-mapping the notes. I simply want to flip the progression of the keys on a midi piano from high notes to low notes. On the fly. Basically I want to take the note on message byte and use the note number as an index for an array of remapped keys. I want to flip the keyboard on a D note for now... then later allow users to turn this remapping off and on (on the fly.) Also it would be nice to have certain note ranges mapped to different programs/patches. -Ember PS> I wish I could help test the sound drivers... but I only have an inexpensive intel integrated sound chip and I am on a dell which needs to be replaced with something par.

3 5

[alsa-devel] where to get snd_pcm_hardware info when usb audio device plug in?
by loody 09 Sep '13

09 Sep '13

hi all: is it possible to get snd_pcm_hardware information in /proc, /sys or anywhere when usb audio device plug in? if no, is there api I can take in usermode to get snd_pcm_hardware of this usb audio device? -- Thanks for your great help,

2 6

[alsa-devel] [PATCH] ALSA: snd-usb: sample rate detection workaround for Axe-Fx II
by Daniel Mack 08 Sep '13

08 Sep '13

Fractal Audio Systems Axe-Fx II and maybe other devices fail to return their sample rate range descriptor when only asked for the first two bytes. Add a workaround and assume exactly one sample rate in this case, and continue with the full descriptor fetch. Signed-off-by: Daniel Mack <zonque(a)gmail.com> Reported-and-tested-by: Joachim Gahl <joachim.gahl(a)web.de> --- sound/usb/format.c | 21 +++++++++++++-------- 1 file changed, 13 insertions(+), 8 deletions(-) diff --git a/sound/usb/format.c b/sound/usb/format.c index 3525231..d9de43d 100644 --- a/sound/usb/format.c +++ b/sound/usb/format.c @@ -297,13 +297,17 @@ static int parse_audio_format_rates_v2(struct snd_usb_audio *chip, snd_usb_ctrl_intf(chip) | (clock << 8), tmp, sizeof(tmp)); - if (ret < 0) { - snd_printk(KERN_ERR "%s(): unable to retrieve number of sample rates (clock %d)\n", - __func__, clock); - goto err; - } + /* + * Some devices, including Fractal Audio Systems Axe-Fx II, fail to + * return only a fraction of this descriptor. If that happens, assume + * we have only one sample rate and continue. In case of any other + * error, we will bail on the full descriptor fetch below. + */ + if (ret < 0) + nr_triplets = 1; + else + nr_triplets = combine_word(tmp); - nr_triplets = (tmp[1] << 8) | tmp[0]; data_size = 2 + 12 * nr_triplets; data = kzalloc(data_size, GFP_KERNEL); if (!data) { @@ -319,8 +323,9 @@ static int parse_audio_format_rates_v2(struct snd_usb_audio *chip, data, data_size); if (ret < 0) { - snd_printk(KERN_ERR "%s(): unable to retrieve sample rate range (clock %d)\n", - __func__, clock); + snd_printk(KERN_ERR "%s(): unable to retrieve sample rate range " + "(clock %d, nr_triplets %d)\n", + __func__, clock, nr_triplets); ret = -EINVAL; goto err_free; } -- 1.8.3.1

1 1

[alsa-devel] [PATCH] ASoC: fsl_spdif: Select regmap-mmio
by Mark Brown 08 Sep '13

08 Sep '13

From: Mark Brown <broonie(a)linaro.org> The S/PDIF driver needs regmap so select it to make sure it gets included in the build. Reported-by: Fengguang Wu <fengguang.wu(a)intel.com> Signed-off-by: Mark Brown <broonie(a)linaro.org> --- sound/soc/fsl/Kconfig | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/soc/fsl/Kconfig b/sound/soc/fsl/Kconfig index 704e246..b7ab71f 100644 --- a/sound/soc/fsl/Kconfig +++ b/sound/soc/fsl/Kconfig @@ -198,6 +198,7 @@ config SND_SOC_IMX_SPDIF select SND_SOC_IMX_PCM_DMA select SND_SOC_FSL_SPDIF select SND_SOC_SPDIF + select REGMAP_MMIO help SoC Audio support for i.MX boards with S/PDIF Say Y if you want to add support for SoC audio on an i.MX board with -- 1.8.4.rc3

2 1

[alsa-devel] alsa crash with “ ALSA sound/usb/urb.c:179 cannot submit urb (err = -27)” on imx6q sabrelite
by Ankit Agarwal 08 Sep '13

08 Sep '13

I am using alsa pcm player to listen to audio. The alsa pcm crashes with the error: ALSA sound/usb/urb.c:179 cannot submit urb (err = -27). I am able to hear the data initially but when other transaction begins on HID report endpoint, alsa crashes with the mentioned error. I am working on imx6q board with linux kernel-3.0.35. I am using Apple device to get the digital audio. -- Best Regards, Ankit Agarwal

2 1

[alsa-devel] US-1641 drivers involvment
by Luca Novarini 08 Sep '13

08 Sep '13

Hello, I would participate, if possible, in the ALSA development to add support to the USB card Tascam US-1641. Any reply or help from who tried in this task before is really really appreciated. Thank you! Bye. LL

3 4

[alsa-devel] Request for information
by Michael Jordan 07 Sep '13

07 Sep '13

Hello, I am developing a driver for the PCM3168A. We have created an expansion board with the codec, op-amp buffers, etc. which is designed to connect to the audio expansion connector on an OMAPL138 EXPKIT from LogicPD. I have the rough draft of the pcm3168a.c and pcm3168a.h. Looking through the code, I see that the DA850 is "hard-wired" in the code to expect the TLV320AIC3106, which is the on-board codec on the kit. My question is, would it be proper for me to create entries in davinci-evm.c, mach-types.h, etc. in order to provide a way to register this new driver? Or is there a different way I should proceed? I'll be modifying my code base for testing purposes, but when it comes time for submission, I would like to know the proper way to handle this. Thank you, Michael

1 0

[alsa-devel] [PATCH v2 2/2] ALSA: hda - unmute pin amplifier in infoframe setup for Haswell
by mengdong.lin＠intel.com 06 Sep '13

06 Sep '13

From: Mengdong Lin <mengdong.lin(a)intel.com> When Gfx driver reconnects a port and transcoder, the pin amplifier will be muted. To enable sound, the pin amp need to be unmuted. This patch - moves pin amp unmuting from stream preparing to hdmi_setup_audio_infoframe(). So if port:transcoder reconnection happens during stream playback, the ELDV unsol event can stil trigger pin's amp unmuting when re-setting up audio info frame. - remove reading pin amp status before unmuting for speed-up, since pin amp should always be unmuted. - rename haswell_verify_pin_D0() to haswell_verify_D0(), since the convertor power state is also fixed here. This patch is mostly based on suggestion of David Henningsson. Cc: David Henningsson <david.henningsson(a)canonical.com> Signed-off-by: Mengdong Lin <mengdong.lin(a)intel.com> diff --git a/sound/pci/hda/patch_hdmi.c b/sound/pci/hda/patch_hdmi.c index 974b55c..3d8cd044 100644 --- a/sound/pci/hda/patch_hdmi.c +++ b/sound/pci/hda/patch_hdmi.c @@ -896,6 +896,11 @@ static void hdmi_setup_audio_infoframe(struct hda_codec *codec, if (!channels) return; + if (is_haswell(codec)) + snd_hda_codec_write(codec, pin_nid, 0, + AC_VERB_SET_AMP_GAIN_MUTE, + AMP_OUT_UNMUTE); + eld = &per_pin->sink_eld; if (!eld->monitor_present) return; @@ -1035,10 +1040,10 @@ static void hdmi_unsol_event(struct hda_codec *codec, unsigned int res) hdmi_non_intrinsic_event(codec, res); } -static void haswell_verify_pin_D0(struct hda_codec *codec, +static void haswell_verify_D0(struct hda_codec *codec, hda_nid_t cvt_nid, hda_nid_t nid) { - int pwr, lamp, ramp; + int pwr; /* For Haswell, the converter 1/2 may keep in D3 state after bootup, * thus pins could only choose converter 0 for use. Make sure the @@ -1054,25 +1059,6 @@ static void haswell_verify_pin_D0(struct hda_codec *codec, pwr = (pwr & AC_PWRST_ACTUAL) >> AC_PWRST_ACTUAL_SHIFT; snd_printd("Haswell HDMI audio: Power for pin 0x%x is now D%d\n", nid, pwr); } - - lamp = snd_hda_codec_read(codec, nid, 0, - AC_VERB_GET_AMP_GAIN_MUTE, - AC_AMP_GET_LEFT | AC_AMP_GET_OUTPUT); - ramp = snd_hda_codec_read(codec, nid, 0, - AC_VERB_GET_AMP_GAIN_MUTE, - AC_AMP_GET_RIGHT | AC_AMP_GET_OUTPUT); - if (lamp != ramp) { - snd_hda_codec_write(codec, nid, 0, AC_VERB_SET_AMP_GAIN_MUTE, - AC_AMP_SET_RIGHT | AC_AMP_SET_OUTPUT | lamp); - - lamp = snd_hda_codec_read(codec, nid, 0, - AC_VERB_GET_AMP_GAIN_MUTE, - AC_AMP_GET_LEFT | AC_AMP_GET_OUTPUT); - ramp = snd_hda_codec_read(codec, nid, 0, - AC_VERB_GET_AMP_GAIN_MUTE, - AC_AMP_GET_RIGHT | AC_AMP_GET_OUTPUT); - snd_printd("Haswell HDMI audio: Mute after set on pin 0x%x: [0x%x 0x%x]\n", nid, lamp, ramp); - } } /* @@ -1090,7 +1076,7 @@ static int hdmi_setup_stream(struct hda_codec *codec, hda_nid_t cvt_nid, int new_pinctl = 0; if (is_haswell(codec)) - haswell_verify_pin_D0(codec, cvt_nid, pin_nid); + haswell_verify_D0(codec, cvt_nid, pin_nid); if (snd_hda_query_pin_caps(codec, pin_nid) & AC_PINCAP_HBR) { pinctl = snd_hda_codec_read(codec, pin_nid, 0, @@ -1361,13 +1347,9 @@ static void hdmi_present_sense(struct hdmi_spec_per_pin *per_pin, int repoll) * changed during the stream playback */ if (is_haswell(codec) && - eld->eld_valid && !old_eld_valid && per_pin->setup) { - snd_hda_codec_write(codec, pin_nid, 0, - AC_VERB_SET_AMP_GAIN_MUTE, - AMP_OUT_UNMUTE); + eld->eld_valid && !old_eld_valid && per_pin->setup) hdmi_setup_audio_infoframe(codec, per_pin, per_pin->non_pcm); - } } mutex_unlock(&pin_eld->lock); -- 1.8.1.2

2 1

[alsa-devel] [PATCH] ALSA: hda - Add dock speaker support for ASUS TX300
by Takashi Iwai 06 Sep '13

06 Sep '13

ASUS TX300 has a built-in speaker in the tablet part and in the dock part, and the tablet speaker is supposed to be unused while the machine is docked. The current HD-audio driver, however, doesn't support the dock speaker, partly because BIOS doesn't set up the pin for the corresponding output. But, not only the missing pin config, also the missing unsol event handling is another issue. Otherwise the automatic switching via dock/undock won't work. Through debugging sessions, we found out that the dock speaker pin is NID 0x1b, and it generates an unsol event at docking/undocking, the docking state can be inquired via the normal pin detection verb. Also, it's turned out that GPIO 2 is needed as an amp. So, all materials are ready to cook. This patch provides the basic dock speaker support with TX300: - The dock speaker is turned on/off via "Dock Speaker" mixer mute. - The dock speaker is automatically muted when docked. This is independently from the mixer mute switch, just like the headphone auto-mute function. The implementation is a bit tricky. Since we want to handle it as a secondary speaker, we set it up a pin as a speaker with a jack detection. Then, the fixup function registers the own unsol callback for this pin because the standard automute can't handle the thing like a "speaker jack". In the own automute hook, we apply the mute of the tablet speaker in addition by checking the dock state. Also, the speaker control names are slightly shuffled because the generic parser doesn't give good names but blindly assumes a bass speaker as a secondary speaker. Bugzilla: https://bugzilla.kernel.org/show_bug.cgi?id=59791 Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/pci/hda/patch_realtek.c | 56 +++++++++++++++++++++++++++++++++++++++++++ 1 file changed, 56 insertions(+) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 4a90917..d38d6a0 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -3443,6 +3443,56 @@ static void alc283_fixup_chromebook(struct hda_codec *codec, } } +/* mute tablet speaker pin (0x14) via dock plugging in addition */ +static void asus_tx300_automute(struct hda_codec *codec) +{ + struct alc_spec *spec = codec->spec; + snd_hda_gen_update_outputs(codec); + if (snd_hda_jack_detect(codec, 0x1b)) + spec->gen.mute_bits |= (1ULL << 0x14); +} + +static void alc282_fixup_asus_tx300(struct hda_codec *codec, + const struct hda_fixup *fix, int action) +{ + struct alc_spec *spec = codec->spec; + /* TX300 needs to set up GPIO2 for the speaker amp */ + static const struct hda_verb gpio2_verbs[] = { + { 0x01, AC_VERB_SET_GPIO_MASK, 0x04 }, + { 0x01, AC_VERB_SET_GPIO_DIRECTION, 0x04 }, + { 0x01, AC_VERB_SET_GPIO_DATA, 0x04 }, + {} + }; + static const struct hda_pintbl dock_pins[] = { + { 0x1b, 0x21114000 }, /* dock speaker pin */ + {} + }; + struct snd_kcontrol *kctl; + + switch (action) { + case HDA_FIXUP_ACT_PRE_PROBE: + snd_hda_add_verbs(codec, gpio2_verbs); + snd_hda_apply_pincfgs(codec, dock_pins); + spec->gen.auto_mute_via_amp = 1; + spec->gen.automute_hook = asus_tx300_automute; + snd_hda_jack_detect_enable_callback(codec, 0x1b, + HDA_GEN_HP_EVENT, + snd_hda_gen_hp_automute); + break; + case HDA_FIXUP_ACT_BUILD: + /* this is a bit tricky; give more sane names for the main + * (tablet) speaker and the dock speaker, respectively + */ + kctl = snd_hda_find_mixer_ctl(codec, "Speaker Playback Switch"); + if (kctl) + strcpy(kctl->id.name, "Dock Speaker Playback Switch"); + kctl = snd_hda_find_mixer_ctl(codec, "Bass Speaker Playback Switch"); + if (kctl) + strcpy(kctl->id.name, "Speaker Playback Switch"); + break; + } +} + enum { ALC269_FIXUP_SONY_VAIO, ALC275_FIXUP_SONY_VAIO_GPIO2, @@ -3480,6 +3530,7 @@ enum { ALC269_FIXUP_LIMIT_INT_MIC_BOOST, ALC269VB_FIXUP_ORDISSIMO_EVE2, ALC283_FIXUP_CHROME_BOOK, + ALC282_FIXUP_ASUS_TX300, }; static const struct hda_fixup alc269_fixups[] = { @@ -3735,6 +3786,10 @@ static const struct hda_fixup alc269_fixups[] = { .type = HDA_FIXUP_FUNC, .v.func = alc283_fixup_chromebook, }, + [ALC282_FIXUP_ASUS_TX300] = { + .type = HDA_FIXUP_FUNC, + .v.func = alc282_fixup_asus_tx300, + }, }; static const struct snd_pci_quirk alc269_fixup_tbl[] = { @@ -3784,6 +3839,7 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK(0x103c, 0x1983, "HP Pavilion", ALC269_FIXUP_HP_MUTE_LED_MIC1), SND_PCI_QUIRK(0x103c, 0x21ed, "HP Falco Chromebook", ALC283_FIXUP_CHROME_BOOK), SND_PCI_QUIRK_VENDOR(0x103c, "HP", ALC269_FIXUP_HP_MUTE_LED), + SND_PCI_QUIRK(0x1043, 0x103f, "ASUS TX300", ALC282_FIXUP_ASUS_TX300), SND_PCI_QUIRK(0x1043, 0x106d, "Asus K53BE", ALC269_FIXUP_LIMIT_INT_MIC_BOOST), SND_PCI_QUIRK(0x1043, 0x115d, "Asus 1015E", ALC269_FIXUP_LIMIT_INT_MIC_BOOST), SND_PCI_QUIRK(0x1043, 0x1427, "Asus Zenbook UX31E", ALC269VB_FIXUP_DMIC), -- 1.8.4

1 0

[alsa-devel] [PATCH] ASoC: core: Remove dependencies on GENERIC_HARDIRQS
by Mark Brown 06 Sep '13

06 Sep '13

From: Mark Brown <broonie(a)linaro.org> All architectures now use the generic IRQ subsystem so this is redundant. Signed-off-by: Mark Brown <broonie(a)linaro.org> --- sound/soc/codecs/Kconfig | 2 +- sound/soc/samsung/Kconfig | 2 +- 2 files changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig index 15106c0..b33b45d 100644 --- a/sound/soc/codecs/Kconfig +++ b/sound/soc/codecs/Kconfig @@ -107,7 +107,7 @@ config SND_SOC_ALL_CODECS select SND_SOC_WM8782 select SND_SOC_WM8804 if SND_SOC_I2C_AND_SPI select SND_SOC_WM8900 if I2C - select SND_SOC_WM8903 if I2C && GENERIC_HARDIRQS + select SND_SOC_WM8903 if I2C select SND_SOC_WM8904 if I2C select SND_SOC_WM8940 if I2C select SND_SOC_WM8955 if I2C diff --git a/sound/soc/samsung/Kconfig b/sound/soc/samsung/Kconfig index 1b0cccd..d446c6b 100644 --- a/sound/soc/samsung/Kconfig +++ b/sound/soc/samsung/Kconfig @@ -63,7 +63,7 @@ config SND_SOC_SAMSUNG_SMDK_WM8580 config SND_SOC_SAMSUNG_SMDK_WM8994 tristate "SoC I2S Audio support for WM8994 on SMDK" depends on SND_SOC_SAMSUNG - depends on I2C=y && GENERIC_HARDIRQS + depends on I2C=y select MFD_WM8994 select SND_SOC_WM8994 select SND_SAMSUNG_I2S -- 1.8.4.rc3

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[alsa-devel] [PATCH] ASoC: kirkwood: change the compatible string of the kirkwood-i2s driver
by Thomas Petazzoni 06 Sep '13

06 Sep '13

The compatible string of the kirkwood-i2s driver was chosen as "marvell,mvebu-audio". Using such a compatible string is not a good idea, since "mvebu" is the name of a large family of SOCs, in which new, unknown SOCs will be coming in the future. It is therefore impossible to know what will be evolutions of this hardware block in the next generations of the SOCs. For this reason, the recommandation for compatible strings of on-SOCs devices has always been to use the name of the oldest SOC that has the hardware block. New SOCs that have an exactly compatible hardware block can reference it using the same compatible string. See [1], [2] and [3] for various cases were this suggestion was made, including from Rob Herring, a Device Tree binding maintainer. As an example, there are already small differences between current generations: * On Kirkwood, only one interrupt is used for audio. * On Dove, two interrupts are used, one for audio data and one for error reporting. In the near future, I'll be adding audio support to Armada 370, which allows has the same hardware block (but maybe with minor variants). Therefore, this patch changes the driver to accept "marvell,kirkwood-audio" and "marvell,dove-audio" as compatible strings instead of the too-generic "marvell,mvebu-audio". The reason for the two different compatible strings is the difference in the number of interrupts used by the two SOCs for audio. This Device Tree binding has never been part of a Linux kernel stable release so far, so it can be changed now without breaking backward compatibility. [1] http://lists.infradead.org/pipermail/linux-mtd/2012-March/040417.html [2] http://lists.infradead.org/pipermail/linux-arm-kernel/2013-April/161065.html [3] http://lists.infradead.org/pipermail/linux-arm-kernel/2012-March/087702.html Signed-off-by: Thomas Petazzoni <thomas.petazzoni(a)free-electrons.com> --- Documentation/devicetree/bindings/sound/mvebu-audio.txt | 12 ++++++++---- sound/soc/kirkwood/kirkwood-i2s.c | 3 ++- 2 files changed, 10 insertions(+), 5 deletions(-) diff --git a/Documentation/devicetree/bindings/sound/mvebu-audio.txt b/Documentation/devicetree/bindings/sound/mvebu-audio.txt index 7e5fd37..f0062c5 100644 --- a/Documentation/devicetree/bindings/sound/mvebu-audio.txt +++ b/Documentation/devicetree/bindings/sound/mvebu-audio.txt @@ -2,13 +2,17 @@ Required properties: -- compatible: "marvell,mvebu-audio" +- compatible: + "marvell,kirkwood-audio" for Kirkwood platforms + "marvell,dove-audio" for Dove platforms - reg: physical base address of the controller and length of memory mapped region. -- interrupts: list of two irq numbers. - The first irq is used for data flow and the second one is used for errors. +- interrupts: + with "marvell,kirkwood-audio", the audio interrupt + with "marvell,dove-audio", a list of two interrupts, the first for + the data flow, and the second for errors. - clocks: one or two phandles. The first one is mandatory and defines the internal clock. @@ -21,7 +25,7 @@ Required properties: Example: i2s1: audio-controller@b4000 { - compatible = "marvell,mvebu-audio"; + compatible = "marvell,dove-audio"; reg = <0xb4000 0x2210>; interrupts = <21>, <22>; clocks = <&gate_clk 13>; diff --git a/sound/soc/kirkwood/kirkwood-i2s.c b/sound/soc/kirkwood/kirkwood-i2s.c index 7fce340..0f3d73d 100644 --- a/sound/soc/kirkwood/kirkwood-i2s.c +++ b/sound/soc/kirkwood/kirkwood-i2s.c @@ -559,7 +559,8 @@ static int kirkwood_i2s_dev_remove(struct platform_device *pdev) #ifdef CONFIG_OF static struct of_device_id mvebu_audio_of_match[] = { - { .compatible = "marvell,mvebu-audio" }, + { .compatible = "marvell,kirkwood-audio" }, + { .compatible = "marvell,dove-audio" }, { } }; MODULE_DEVICE_TABLE(of, mvebu_audio_of_match); -- 1.8.1.2

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Re: [alsa-devel] [PATCH 4/5 v2] ARM: shmobile: bockw: add R-Car sound support (PIO)
by Simon Horman 06 Sep '13

06 Sep '13

[ CC Mark Brown, alsa-devel ] On Thu, Sep 05, 2013 at 02:57:51PM +0900, Simon Horman wrote: > On Mon, Aug 26, 2013 at 01:52:35AM -0700, Kuninori Morimoto wrote: > > This patch enables R-Car sound, > > AK4643 (CN19) and AK4554 (CN20/CN21) codec chip > > on Bock-W. > > > > But, it supports PIO transfer only at this point. > > User can check sound settings (Dip-switch/PFC etc) > > via this patch, but will get under/over flow error > > when playback/capture. > > Because PIO transfer via SSI will be interrupted > > "sampling rate" times per 1 second. > > > > DMA transfer will be supported when HPB-DMAC was > > enabled on r8a7778. > > > > You will notice strange ALSA sound card HW > > numbering on Bock-W board. > > This came from AK4554 strange format on playback/capture. > > The format on playback/capture is same on "normal" codec chip, > > but AK4554 was different. > > Because of that, AK4554 playback/capture are > > registered as a different sound card. > > > > Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> > > --- > > v1 -> v2 > > Thanks, I have queued this up for v3.13. > > There was a bit of diff-noise, please make sure I have applied it > correctly. The patch I applied is below: Unfortunately I didn't read your cover email carefully before queuing this up. I now see that it has dependencies that aren't present in v3.11-rc and thus I have temporarily dropped it. It seems to me that the dependencies could be provided by the topic/car branch of git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git. Morimoto-san, is this correct? If so, Mark, would it be possible for you to provide topic/car or similar as stable branch for me to use as a base? My intention is to queue these up for v3.13.

3 4

[alsa-devel] Roland Sound Canvas SD-50
by Keith A. Milner 05 Sep '13

05 Sep '13

I saw a recent discussion on Alsa-users about the Roland SD-50 which seemed to tail off inconclusively. Does anyone know if this was found to work or not? I ask because I have the option of picking one up (relatively) cheaply. It looks like an interesting little box. But if there isn't a working ALSA solution yet, I might be tempted to pick it up anyway and try to get it working, although I may need some help with that. Cheers, Keith

2 2

[alsa-devel] Documentation Request
by Carl Canuck 05 Sep '13

05 Sep '13

Hello, I've recently begun the work of porting a Windows application to LInux/ALSA. I think ALSA and it's drivers are quite good, but the documentation and example code are terribly lacking, especially for one of the most important use-cases of all: the full on digital audio workstation. I humbly request that a full_duplex_audio_with_midi.c file be added to the example code with the following parameters: 1. Full duplex audio, just simply memcpy'ing the input to the output would suffice 2. Accepts MIDI input; it doesn't have to do anything useful with the data, just printing the event parameters to stdout would suffice. 3. At run time, the application lists all available audio devices and allows the user to select the device, then make settings such as buffer size, sample rate, bit depth and whether to use mmap based on the capabilities of the chosen device 4. At run time, allows optionally selecting a MIDI device. 5. Copious code comments that makes no assumption that the person reading knows anything about the ALSA API or it's constructs, invocations and methodologies. This could either be an interactive CLI application written in C, or even a full GUI application using Qt or GTK. This could then be used as a shell for writing more complex applications, instead of each developer having to reinvent the wheel by trying to figure out how to combine about 4 or so of the other example applications into a single app. /Carl

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[alsa-devel] Constraints and hw_params with highly configurable HW
by Uli Franke 05 Sep '13

05 Sep '13

Hi everybody We're currently working on some neat enhancements for the still experimental snd_dice driver originally initiated by Clemens Ladisch. To list some of the new and already working features: * firmware loading, backed by a user-space command line utility * generic ALSA control support for mostly all device specific settings for all our (i.e. Weiss Engineering) DICE based products and last but not least * capture support * slave mode (external clock source support). I'm currently debugging and fine-tuning things and that's where some questions arose. The initial implementation of Clemens tried to fetch all possible configurations of a device during probe in order to have all combinations of rates and channels at hand when it comes to formulate the constraints set within the ALSA pcm substream open callback. That's a nice approach but will fail with a lot of devices as their configurations can (and mostly will) be volatile. Our devices for instance feature several switches, options and alike which have heavy influence on the channel layout and sample rates and combination of both. I consulted Takashi's documentation on constraints which suggests to "use the source" but I couldn't find any similar situation within other drivers (in case I should have missed something, just let me know of course). Therefore I had the idea to define some maximum constraints and let the ALSA hw_params callback fail in case some settings or combinations are determined to be not supported during its invocation. Is this viable? Could this be/is this consistent with the specs? If yes, would EINVAL an appropriate error code to return? Regards Uli

2 3

[alsa-devel] Recall: [PATCH] ALSA: hda - unmute pin amplifier in infoframe setup for Haswell
by Lin, Mengdong 05 Sep '13

05 Sep '13

Lin, Mengdong would like to recall the message, "[PATCH] ALSA: hda - unmute pin amplifier in infoframe setup for Haswell".

1 0

[alsa-devel] [PATCH] ALSA: hda - unmute pin amplifier in infoframe setup for Haswell
by mengdong.lin＠intel.com 05 Sep '13

05 Sep '13

From: Mengdong Lin <mengdong.lin(a)intel.com> When Gfx driver reconnects a port and transcoder, the pin amplifier will be muted. To enable sound, the pin amp need to be unmuted. This patch - moves pin amp unmuting from stream preparing to hdmi_setup_audio_infoframe(). So if port:transcoder reconnection happens during stream playback, the ELDV unsol event can stil trigger pin's amp unmuting when re-setting up audio info frame. - remove reading pin amp status before unmuting for speed-up, since pin amp should always be unmuted. - rename haswell_verify_pin_D0() to haswell_verify_pin_cvt_D0(), since the convertor power state is also fixed here. - define is_haswell() to replace checking Haswell vendor ID everywhere. This patch is mostly based on suggestion of David Henningsson. Cc: David Henningsson <david.henningsson(a)canonical.com> Signed-off-by: Mengdong Lin <mengdong.lin(a)intel.com> diff --git a/sound/pci/hda/patch_hdmi.c b/sound/pci/hda/patch_hdmi.c index 9a58893..471c158 100644 --- a/sound/pci/hda/patch_hdmi.c +++ b/sound/pci/hda/patch_hdmi.c @@ -44,6 +44,8 @@ static bool static_hdmi_pcm; module_param(static_hdmi_pcm, bool, 0644); MODULE_PARM_DESC(static_hdmi_pcm, "Don't restrict PCM parameters per ELD info"); +#define is_haswell(codec) ((codec)->vendor_id == 0x80862807) + struct hdmi_spec_per_cvt { hda_nid_t cvt_nid; int assigned; @@ -894,6 +896,11 @@ static void hdmi_setup_audio_infoframe(struct hda_codec *codec, if (!channels) return; + if (is_haswell(codec)) + snd_hda_codec_write(codec, pin_nid, 0, + AC_VERB_SET_AMP_GAIN_MUTE, + AMP_OUT_UNMUTE); + eld = &per_pin->sink_eld; if (!eld->monitor_present) return; @@ -1033,10 +1040,10 @@ static void hdmi_unsol_event(struct hda_codec *codec, unsigned int res) hdmi_non_intrinsic_event(codec, res); } -static void haswell_verify_pin_D0(struct hda_codec *codec, +static void haswell_verify_pin_cvt_D0(struct hda_codec *codec, hda_nid_t cvt_nid, hda_nid_t nid) { - int pwr, lamp, ramp; + int pwr; /* For Haswell, the converter 1/2 may keep in D3 state after bootup, * thus pins could only choose converter 0 for use. Make sure the @@ -1052,25 +1059,6 @@ static void haswell_verify_pin_D0(struct hda_codec *codec, pwr = (pwr & AC_PWRST_ACTUAL) >> AC_PWRST_ACTUAL_SHIFT; snd_printd("Haswell HDMI audio: Power for pin 0x%x is now D%d\n", nid, pwr); } - - lamp = snd_hda_codec_read(codec, nid, 0, - AC_VERB_GET_AMP_GAIN_MUTE, - AC_AMP_GET_LEFT | AC_AMP_GET_OUTPUT); - ramp = snd_hda_codec_read(codec, nid, 0, - AC_VERB_GET_AMP_GAIN_MUTE, - AC_AMP_GET_RIGHT | AC_AMP_GET_OUTPUT); - if (lamp != ramp) { - snd_hda_codec_write(codec, nid, 0, AC_VERB_SET_AMP_GAIN_MUTE, - AC_AMP_SET_RIGHT | AC_AMP_SET_OUTPUT | lamp); - - lamp = snd_hda_codec_read(codec, nid, 0, - AC_VERB_GET_AMP_GAIN_MUTE, - AC_AMP_GET_LEFT | AC_AMP_GET_OUTPUT); - ramp = snd_hda_codec_read(codec, nid, 0, - AC_VERB_GET_AMP_GAIN_MUTE, - AC_AMP_GET_RIGHT | AC_AMP_GET_OUTPUT); - snd_printd("Haswell HDMI audio: Mute after set on pin 0x%x: [0x%x 0x%x]\n", nid, lamp, ramp); - } } /* @@ -1087,8 +1075,8 @@ static int hdmi_setup_stream(struct hda_codec *codec, hda_nid_t cvt_nid, int pinctl; int new_pinctl = 0; - if (codec->vendor_id == 0x80862807) - haswell_verify_pin_D0(codec, cvt_nid, pin_nid); + if (is_haswell(codec)) + haswell_verify_pin_cvt_D0(codec, cvt_nid, pin_nid); if (snd_hda_query_pin_caps(codec, pin_nid) & AC_PINCAP_HBR) { pinctl = snd_hda_codec_read(codec, pin_nid, 0, @@ -1227,7 +1215,7 @@ static int hdmi_pcm_open(struct hda_pcm_stream *hinfo, mux_idx); /* configure unused pins to choose other converters */ - if (codec->vendor_id == 0x80862807) + if (is_haswell(codec)) haswell_config_cvts(codec, pin_idx, mux_idx); snd_hda_spdif_ctls_assign(codec, pin_idx, per_cvt->cvt_nid); @@ -1358,14 +1346,10 @@ static void hdmi_present_sense(struct hdmi_spec_per_pin *per_pin, int repoll) /* Haswell-specific workaround: re-setup when the transcoder is * changed during the stream playback */ - if (codec->vendor_id == 0x80862807 && - eld->eld_valid && !old_eld_valid && per_pin->setup) { - snd_hda_codec_write(codec, pin_nid, 0, - AC_VERB_SET_AMP_GAIN_MUTE, - AMP_OUT_UNMUTE); + if (is_haswell(codec) && + eld->eld_valid && !old_eld_valid && per_pin->setup) hdmi_setup_audio_infoframe(codec, per_pin, per_pin->non_pcm); - } } mutex_unlock(&pin_eld->lock); @@ -1405,7 +1389,7 @@ static int hdmi_add_pin(struct hda_codec *codec, hda_nid_t pin_nid) if (get_defcfg_connect(config) == AC_JACK_PORT_NONE) return 0; - if (codec->vendor_id == 0x80862807) + if (is_haswell(codec)) intel_haswell_fixup_connect_list(codec, pin_nid); pin_idx = spec->num_pins; @@ -2014,7 +1998,7 @@ static int patch_generic_hdmi(struct hda_codec *codec) codec->spec = spec; hdmi_array_init(spec, 4); - if (codec->vendor_id == 0x80862807) { + if (is_haswell(codec)) { intel_haswell_enable_all_pins(codec, true); intel_haswell_fixup_enable_dp12(codec); } @@ -2025,7 +2009,7 @@ static int patch_generic_hdmi(struct hda_codec *codec) return -EINVAL; } codec->patch_ops = generic_hdmi_patch_ops; - if (codec->vendor_id == 0x80862807) { + if (is_haswell(codec)) { codec->patch_ops.set_power_state = haswell_set_power_state; codec->dp_mst = true; } -- 1.8.1.2

4 3

[alsa-devel] [PATCH v2 1/2] ALSA: hda - define is_haswell() to check if a display audio codec is Haswell
by mengdong.lin＠intel.com 04 Sep '13

04 Sep '13

From: Mengdong Lin <mengdong.lin(a)intel.com> To apply Haswell specific fixings, this patch defines is_haswell() to check whether a display audio codec is Haswell, to avoid explicitly checking Haswell vendor ID everywhere. Signed-off-by: Mengdong Lin <mengdong.lin(a)intel.com> diff --git a/sound/pci/hda/patch_hdmi.c b/sound/pci/hda/patch_hdmi.c index 9a58893..974b55c 100644 --- a/sound/pci/hda/patch_hdmi.c +++ b/sound/pci/hda/patch_hdmi.c @@ -44,6 +44,8 @@ static bool static_hdmi_pcm; module_param(static_hdmi_pcm, bool, 0644); MODULE_PARM_DESC(static_hdmi_pcm, "Don't restrict PCM parameters per ELD info"); +#define is_haswell(codec) ((codec)->vendor_id == 0x80862807) + struct hdmi_spec_per_cvt { hda_nid_t cvt_nid; int assigned; @@ -1087,7 +1089,7 @@ static int hdmi_setup_stream(struct hda_codec *codec, hda_nid_t cvt_nid, int pinctl; int new_pinctl = 0; - if (codec->vendor_id == 0x80862807) + if (is_haswell(codec)) haswell_verify_pin_D0(codec, cvt_nid, pin_nid); if (snd_hda_query_pin_caps(codec, pin_nid) & AC_PINCAP_HBR) { @@ -1227,7 +1229,7 @@ static int hdmi_pcm_open(struct hda_pcm_stream *hinfo, mux_idx); /* configure unused pins to choose other converters */ - if (codec->vendor_id == 0x80862807) + if (is_haswell(codec)) haswell_config_cvts(codec, pin_idx, mux_idx); snd_hda_spdif_ctls_assign(codec, pin_idx, per_cvt->cvt_nid); @@ -1358,7 +1360,7 @@ static void hdmi_present_sense(struct hdmi_spec_per_pin *per_pin, int repoll) /* Haswell-specific workaround: re-setup when the transcoder is * changed during the stream playback */ - if (codec->vendor_id == 0x80862807 && + if (is_haswell(codec) && eld->eld_valid && !old_eld_valid && per_pin->setup) { snd_hda_codec_write(codec, pin_nid, 0, AC_VERB_SET_AMP_GAIN_MUTE, @@ -1405,7 +1407,7 @@ static int hdmi_add_pin(struct hda_codec *codec, hda_nid_t pin_nid) if (get_defcfg_connect(config) == AC_JACK_PORT_NONE) return 0; - if (codec->vendor_id == 0x80862807) + if (is_haswell(codec)) intel_haswell_fixup_connect_list(codec, pin_nid); pin_idx = spec->num_pins; @@ -2014,7 +2016,7 @@ static int patch_generic_hdmi(struct hda_codec *codec) codec->spec = spec; hdmi_array_init(spec, 4); - if (codec->vendor_id == 0x80862807) { + if (is_haswell(codec)) { intel_haswell_enable_all_pins(codec, true); intel_haswell_fixup_enable_dp12(codec); } @@ -2025,7 +2027,7 @@ static int patch_generic_hdmi(struct hda_codec *codec) return -EINVAL; } codec->patch_ops = generic_hdmi_patch_ops; - if (codec->vendor_id == 0x80862807) { + if (is_haswell(codec)) { codec->patch_ops.set_power_state = haswell_set_power_state; codec->dp_mst = true; } -- 1.8.1.2

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[alsa-devel] [PATCH 1/2] ASoC: core: Implement devm_snd_soc_register_component()
by Mark Brown 04 Sep '13

04 Sep '13

From: Mark Brown <broonie(a)linaro.org> Since with the wider use of devres many drivers are now only calling snd_soc_unregister_component() in their remove functions providing a managed version will save a reasonable amount of code. Signed-off-by: Mark Brown <broonie(a)linaro.org> --- include/sound/soc.h | 3 +++ sound/soc/Makefile | 2 +- sound/soc/soc-devres.c | 52 ++++++++++++++++++++++++++++++++++++++++++++++++++ 3 files changed, 56 insertions(+), 1 deletion(-) create mode 100644 sound/soc/soc-devres.c diff --git a/include/sound/soc.h b/include/sound/soc.h index a72af63..f126676 100644 --- a/include/sound/soc.h +++ b/include/sound/soc.h @@ -381,6 +381,9 @@ void snd_soc_unregister_codec(struct device *dev); int snd_soc_register_component(struct device *dev, const struct snd_soc_component_driver *cmpnt_drv, struct snd_soc_dai_driver *dai_drv, int num_dai); +int devm_snd_soc_register_component(struct device *dev, + const struct snd_soc_component_driver *cmpnt_drv, + struct snd_soc_dai_driver *dai_drv, int num_dai); void snd_soc_unregister_component(struct device *dev); int snd_soc_codec_volatile_register(struct snd_soc_codec *codec, unsigned int reg); diff --git a/sound/soc/Makefile b/sound/soc/Makefile index 61a64d2..8b9e701 100644 --- a/sound/soc/Makefile +++ b/sound/soc/Makefile @@ -1,5 +1,5 @@ snd-soc-core-objs := soc-core.o soc-dapm.o soc-jack.o soc-cache.o soc-utils.o -snd-soc-core-objs += soc-pcm.o soc-compress.o soc-io.o +snd-soc-core-objs += soc-pcm.o soc-compress.o soc-io.o soc-devres.o ifneq ($(CONFIG_SND_SOC_GENERIC_DMAENGINE_PCM),) snd-soc-core-objs += soc-generic-dmaengine-pcm.o diff --git a/sound/soc/soc-devres.c b/sound/soc/soc-devres.c new file mode 100644 index 0000000..13fe86f --- /dev/null +++ b/sound/soc/soc-devres.c @@ -0,0 +1,52 @@ +/* + * soc-devres.c -- ALSA SoC Audio Layer devres functions + * + * Copyright (C) 2013 Linaro Ltd + * + * This program is free software; you can redistribute it and/or modify it + * under the terms of the GNU General Public License as published by the + * Free Software Foundation; either version 2 of the License, or (at your + * option) any later version. + */ + +#include <linux/module.h> +#include <linux/moduleparam.h> +#include <sound/soc.h> + +static void devm_component_release(struct device *dev, void *res) +{ + snd_soc_unregister_component(*(struct device **)res); +} + +/** + * devm_snd_soc_register_component - resource managed component registration + * @dev: Device used to manage component + * @cmpnt_drv: Component driver + * @dai_drv: DAI driver + * @num_dai: Number of DAIs to register + * + * Register a component with automatic unregistration when the device is + * unregistered. + */ +int devm_snd_soc_register_component(struct device *dev, + const struct snd_soc_component_driver *cmpnt_drv, + struct snd_soc_dai_driver *dai_drv, int num_dai) +{ + struct device **ptr; + int ret; + + ptr = devres_alloc(devm_component_release, sizeof(*ptr), GFP_KERNEL); + if (!ptr) + return -ENOMEM; + + ret = snd_soc_register_component(dev, cmpnt_drv, dai_drv, num_dai); + if (ret == 0) { + *ptr = dev; + devres_add(dev, ptr); + } else { + devres_free(ptr); + } + + return ret; +} +EXPORT_SYMBOL_GPL(devm_snd_soc_register_component); -- 1.8.4.rc3

1 1

[alsa-devel] [PATCH 00/14] SPDIF support
by Russell King - ARM Linux 04 Sep '13

04 Sep '13

This is my latest set of patches to get SPDIF working on the Dove Cubox platform. As per the last time, I've left out the "card" part of this, because that is only useful for non-DT setups, which aren't supported in mainline. The first seven patches have already been applied by Mark, and are included here for completeness by anyone who wishes to test this series. There is very little change between this and the previous set of patches. The HACK patch is gone, as I've found out through much digging and experimentation with DAPM (inspite of the complete lack of assistance from ASoC developers) how to use the stream widgets without the widget overwriting bug getting in the way. The CPU DAI widgets are attached to the CPU DAI context, and their access to the driver private data updated accordingly. Why attaching them to the CPU DAI context previously didn't work, I'm not entirely sure, but that certainly previously failed in testing (it may have been due to the duplicated DAI routes and/or the lack of output pin on the SPDIF tranceiver.) I've added three patches: one to add IEC958 channel status setting, another to prefer the external clock over the internal clock, and removal of the IEC958 subframe support (see the commit message for that patch why this can not be supported.) This is a non-DPCM driver, and is the absolute minimum solution which can provide correct hardware behaviour while still being conformant with single zImage kernels and not causing any known regressions. The same conditions apply as per my previous posting - the DAI link needs to be setup and the associated DAPM routes to tell the CPU DAI which outputs are in use, like this: DAI link: .name = "S/PDIF1", .stream_name = "IEC958 Playback", .platform_name = "mvebu-audio.1", .cpu_dai_name = "mvebu-audio.1", .codec_dai_name = "dit-hifi", .codec_name = "spdif-dit", static const struct snd_soc_dapm_route routes[] = { { "Playback", NULL, "spdifdo" }, }; When DPCM is eventually implemented, the DAPM route remains, the DAI link becomes two, as previously illustrated in other discussions. Full diffstat for this series: arch/arm/mach-dove/common.c | 4 +- arch/arm/mach-kirkwood/common.c | 24 ++--- sound/soc/codecs/spdif_transciever.c | 15 +++- sound/soc/kirkwood/Kconfig | 5 - sound/soc/kirkwood/Makefile | 4 +- sound/soc/kirkwood/kirkwood-dma.c | 114 ++++-------------- sound/soc/kirkwood/kirkwood-i2s.c | 213 +++++++++++++++++++++++++++++----- sound/soc/kirkwood/kirkwood-openrd.c | 12 ++- sound/soc/kirkwood/kirkwood-t5325.c | 8 +- sound/soc/kirkwood/kirkwood.h | 45 +++++++- sound/soc/soc-dapm.c | 2 +- 11 files changed, 295 insertions(+), 151 deletions(-) Diffstat for the patches remaining for mainline: sound/soc/kirkwood/kirkwood-dma.c | 4 +- sound/soc/kirkwood/kirkwood-i2s.c | 170 +++++++++++++++++++++++++++++++--- sound/soc/kirkwood/kirkwood-openrd.c | 8 ++ sound/soc/kirkwood/kirkwood-t5325.c | 4 + sound/soc/kirkwood/kirkwood.h | 34 +++++++ 5 files changed, 205 insertions(+), 15 deletions(-)

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[alsa-devel] [PATCH 0/3] ASoC: generic simple sound card DT bindings
by Markus Pargmann 04 Sep '13

04 Sep '13

Hi, The patches were already in the fsl-ssi ac97 series. I moved the seperated simple sound card DT driver into the existing simple card driver. Regards, Markus Markus Pargmann (3): ASoC: generic: simple card, use private data ASoC: generic simple sound card DT bindings ASoC: fsl: Kconfig, visible config items for simple card .../bindings/sound/generic,simple-dt-card.txt | 28 +++ include/sound/simple_card.h | 4 - sound/soc/fsl/Kconfig | 23 ++- sound/soc/generic/Kconfig | 1 + sound/soc/generic/simple-card.c | 225 ++++++++++++++++++--- 5 files changed, 246 insertions(+), 35 deletions(-) create mode 100644 Documentation/devicetree/bindings/sound/generic,simple-dt-card.txt

4 8

[alsa-devel] [PATCH] ASoC: atmel: disable error interrupt
by Bo Shen 04 Sep '13

04 Sep '13

As once the error interrupt is triggered, it can not be cleared. So, disable it. No side effect found while testing on sama5d3xek and at91sam9x5ek boards. Signed-off-by: Bo Shen <voice.shen(a)atmel.com> Acked-by: Nicolas Ferre <nicolas.ferre(a)atmel.com> --- sound/soc/atmel/atmel_ssc_dai.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/atmel/atmel_ssc_dai.c b/sound/soc/atmel/atmel_ssc_dai.c index 0ecf356..bb53dea 100644 --- a/sound/soc/atmel/atmel_ssc_dai.c +++ b/sound/soc/atmel/atmel_ssc_dai.c @@ -649,7 +649,7 @@ static int atmel_ssc_prepare(struct snd_pcm_substream *substream, dma_params = ssc_p->dma_params[dir]; ssc_writel(ssc_p->ssc->regs, CR, dma_params->mask->ssc_enable); - ssc_writel(ssc_p->ssc->regs, IER, dma_params->mask->ssc_error); + ssc_writel(ssc_p->ssc->regs, IDR, dma_params->mask->ssc_error); pr_debug("%s enabled SSC_SR=0x%08x\n", dir ? "receive" : "transmit", -- 1.7.9.5

2 1

Re: [alsa-devel] Documentation Request
by Carl Canuck 04 Sep '13

04 Sep '13

Sorry, I only sent that last reply to Grant, the reply-to-anybody-but-sender options are a little counter-intuitive in Gmail. On Tue, Sep 3, 2013 at 11:10 PM, Carl Canuck <carl.canuck.official(a)gmail.com > wrote: > Hi Grant, > > I have in fact spent a little time with Jack and even conjured up a > working prototype using Jack audio with ALSA MIDI (that will never see the > light of day). However, you're ignoring the fact that: > > 1. Some people don't like having 20 separate audio applications running > and having to save all of their states individually > 2. If you believe that any existing Jack session manager is a stable, > reliable application that adequately solves problem #1, then it's obvious > that you've not spent much time with Jack session managers > 3. Jack is something like 15 years old now, and they still haven't even > managed to solve #1 and #2 yet, among other long-standing issues? > 4. With points #1, #2 and #3 all being common knowledge, why should I put > my faith in Jack competently wrapping ALSA for me instead of taking matters > into my own hands and going straight to ALSA? > 5. If you visit Windows/Mac audio forums, there are legions of people who > will tell you they tried Linux once, found Jack to be buggy and terrible in > general, never intend to try Linux again. In fact, you can register a user > name on any non-Linux music forum, and ask "i'm thinking of trying Linux > for audio, thoughts?", and people will say exactly the things I described > to you, not because they have any unfounded bias against Linux or Jack, but > because that was truly their experience. > 6. Jack's configuration settings and error logging are atrocious, it is > far too often excessively difficult to understand why Jack won't start and > how to fix it. > 7. Is it really so unreasonable to ask for a basic example of a common > use-case for ALSA without being told that I should just abandon the quest > for that knowledge and put my faith in the Jack developers? Should > deciphering the intended use of any API be an exercise left to the would-be > developer? > > If you like Jack, then by all means use it, but it's not ideal for my > use-case, and I won't be spending any more time evaluating it until the > Jack developers address these long-standing issues. > > Carl > > <http://users.suse.com/~mana/alsa090_howto.html> Also, I have in fact seen the link you provided Grant, but that once again is only bits and pieces of the use-case. I'd like to know that I'm using ALSA as it's creators intended, rather than stringing together bits and pieces and hoping that I'm doing it right. Regards, Carl

2 1

[alsa-devel] [PATCH] ALSA: hda - Re-setup HDMI pin and audio infoframe on stream switches
by Takashi Iwai 04 Sep '13

04 Sep '13

When the transcoder:port mapping on Haswell HDMI/DP audio is changed during the stream playback, the sound gets lost. Typically this problem is seen when the user switches the graphics mode from eDP+DP to DP-only configuration, where CRTC 1 is used for DP in the former while CRTC 0 is used for the latter. The graphics controller notifies the change via the normal ELD update procedure, so we get the intrinsic event. For enabling the sound again, the HDMI audio driver needs to reset the pin and set up the audio infoframe again. This patch achieves it by: - keep the current status of channels and info frame setup in per_pin struct, - check the reconnection in the intrinsic event handler, - reset the pin and the re-invoke hdmi_setup_audio_infoframe() accordingly. The hdmi_setup_audio_infoframe() function has been changed, too, so that it can be invoked without passing the substream instance. The patch is mostly based on the work by Mengdong Lin. Cc: Mengdong Lin <mengdong.lin(a)intel.com> Cc: <stable(a)vger.kernel.org> Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/pci/hda/patch_hdmi.c | 41 +++++++++++++++++++++++++++++++---------- 1 file changed, 31 insertions(+), 10 deletions(-) diff --git a/sound/pci/hda/patch_hdmi.c b/sound/pci/hda/patch_hdmi.c index b83b14f..22b5089 100644 --- a/sound/pci/hda/patch_hdmi.c +++ b/sound/pci/hda/patch_hdmi.c @@ -67,6 +67,8 @@ struct hdmi_spec_per_pin { struct delayed_work work; struct snd_kcontrol *eld_ctl; int repoll_count; + bool setup; /* the stream has been set up by prepare callback */ + int channels; /* current number of channels */ bool non_pcm; bool chmap_set; /* channel-map override by ALSA API? */ unsigned char chmap[8]; /* ALSA API channel-map */ @@ -879,18 +881,19 @@ static bool hdmi_infoframe_uptodate(struct hda_codec *codec, hda_nid_t pin_nid, return true; } -static void hdmi_setup_audio_infoframe(struct hda_codec *codec, int pin_idx, - bool non_pcm, - struct snd_pcm_substream *substream) +static void hdmi_setup_audio_infoframe(struct hda_codec *codec, + struct hdmi_spec_per_pin *per_pin, + bool non_pcm) { - struct hdmi_spec *spec = codec->spec; - struct hdmi_spec_per_pin *per_pin = get_pin(spec, pin_idx); hda_nid_t pin_nid = per_pin->pin_nid; - int channels = substream->runtime->channels; + int channels = per_pin->channels; struct hdmi_eld *eld; int ca; union audio_infoframe ai; + if (!channels) + return; + eld = &per_pin->sink_eld; if (!eld->monitor_present) return; @@ -1341,6 +1344,7 @@ static void hdmi_present_sense(struct hdmi_spec_per_pin *per_pin, int repoll) eld_changed = true; } if (update_eld) { + bool old_eld_valid = pin_eld->eld_valid; pin_eld->eld_valid = eld->eld_valid; eld_changed = pin_eld->eld_size != eld->eld_size || memcmp(pin_eld->eld_buffer, eld->eld_buffer, @@ -1350,6 +1354,18 @@ static void hdmi_present_sense(struct hdmi_spec_per_pin *per_pin, int repoll) eld->eld_size); pin_eld->eld_size = eld->eld_size; pin_eld->info = eld->info; + + /* Haswell-specific workaround: re-setup when the transcoder is + * changed during the stream playback + */ + if (codec->vendor_id == 0x80862807 && + eld->eld_valid && !old_eld_valid && per_pin->setup) { + snd_hda_codec_write(codec, pin_nid, 0, + AC_VERB_SET_AMP_GAIN_MUTE, + AMP_OUT_UNMUTE); + hdmi_setup_audio_infoframe(codec, per_pin, + per_pin->non_pcm); + } } mutex_unlock(&pin_eld->lock); @@ -1522,14 +1538,17 @@ static int generic_hdmi_playback_pcm_prepare(struct hda_pcm_stream *hinfo, hda_nid_t cvt_nid = hinfo->nid; struct hdmi_spec *spec = codec->spec; int pin_idx = hinfo_to_pin_index(spec, hinfo); - hda_nid_t pin_nid = get_pin(spec, pin_idx)->pin_nid; + struct hdmi_spec_per_pin *per_pin = get_pin(spec, pin_idx); + hda_nid_t pin_nid = per_pin->pin_nid; bool non_pcm; non_pcm = check_non_pcm_per_cvt(codec, cvt_nid); + per_pin->channels = substream->runtime->channels; + per_pin->setup = true; hdmi_set_channel_count(codec, cvt_nid, substream->runtime->channels); - hdmi_setup_audio_infoframe(codec, pin_idx, non_pcm, substream); + hdmi_setup_audio_infoframe(codec, per_pin, non_pcm); return hdmi_setup_stream(codec, cvt_nid, pin_nid, stream_tag, format); } @@ -1569,6 +1588,9 @@ static int hdmi_pcm_close(struct hda_pcm_stream *hinfo, snd_hda_spdif_ctls_unassign(codec, pin_idx); per_pin->chmap_set = false; memset(per_pin->chmap, 0, sizeof(per_pin->chmap)); + + per_pin->setup = false; + per_pin->channels = 0; } return 0; @@ -1704,8 +1726,7 @@ static int hdmi_chmap_ctl_put(struct snd_kcontrol *kcontrol, per_pin->chmap_set = true; memcpy(per_pin->chmap, chmap, sizeof(chmap)); if (prepared) - hdmi_setup_audio_infoframe(codec, pin_idx, per_pin->non_pcm, - substream); + hdmi_setup_audio_infoframe(codec, per_pin, per_pin->non_pcm); return 0; } -- 1.8.3.4

3 7

[alsa-devel] [PATCH v2] ASoC: fsl: imx-audmux: Do not call imx_audmux_parse_dt_defaults() on non-dt kernel
by Fabio Estevam 04 Sep '13

04 Sep '13

Booting a mx51babbage board with a non-dt kernel leads to the following crash: Unable to handle kernel NULL pointer dereference at virtual address 0000001c pgd = 80004000 [0000001c] *pgd=00000000 Internal error: Oops: 5 [#1] SMP ARM Modules linked in: CPU: 0 PID: 1 Comm: swapper/0 Not tainted 3.11.0-next-20130903 #287 task: 9f860000 ti: 9f862000 task.ti: 9f862000 PC is at of_get_next_available_child+0x5c/0x68 LR is at of_get_next_available_child+0x1c/0x68 pc : [<8043ea58>] lr : [<8043ea18>] psr: 60000193 sp : 9f863d58 ip : 00000000 fp : 9f863d74 r10: 9f89a010 r9 : 9f862000 r8 : 807bb26c r7 : 80615d5c r6 : 00000000 r5 : 60000113 r4 : 00000000 r3 : 00000000 r2 : 808770a4 r1 : 00000011 r0 : 60000113 Flags: nZCv IRQs off FIQs on Mode SVC_32 ISA ARM Segment kernel Control: 10c5387d Table: 90004019 DAC: 00000017 Process swapper/0 (pid: 1, stack limit = 0x9f862240) Stack: (0x9f863d58 to 0x9f864000) This is caused by commit 8548a464b9 (ASoC: imx-audmux: Read default configuration from devicetree). In order to fix this, add a check for 'of_id' so that imx_audmux_parse_dt_defaults() only gets called when a dt kernel is running. Signed-off-by: Fabio Estevam <fabio.estevam(a)freescale.com> --- Changes since v1: - Only call imx_audmux_parse_dt_defaults for dt kernel. sound/soc/fsl/imx-audmux.c | 3 ++- 1 file changed, 2 insertions(+), 1 deletion(-) diff --git a/sound/soc/fsl/imx-audmux.c b/sound/soc/fsl/imx-audmux.c index ab17381..d3bf71a 100644 --- a/sound/soc/fsl/imx-audmux.c +++ b/sound/soc/fsl/imx-audmux.c @@ -335,7 +335,8 @@ static int imx_audmux_probe(struct platform_device *pdev) if (audmux_type == IMX31_AUDMUX) audmux_debugfs_init(); - imx_audmux_parse_dt_defaults(pdev, pdev->dev.of_node); + if (of_id) + imx_audmux_parse_dt_defaults(pdev, pdev->dev.of_node); return 0; } -- 1.8.1.2

2 1

[alsa-devel] [PATCH] ASoC: fsl: imx-audmux: Check if we are running a dt kernel
by Fabio Estevam 03 Sep '13

03 Sep '13

Booting a mx51babbage board with a non-dt kernel leads to the following crash: Unable to handle kernel NULL pointer dereference at virtual address 0000001c pgd = 80004000 [0000001c] *pgd=00000000 Internal error: Oops: 5 [#1] SMP ARM Modules linked in: CPU: 0 PID: 1 Comm: swapper/0 Not tainted 3.11.0-next-20130903 #287 task: 9f860000 ti: 9f862000 task.ti: 9f862000 PC is at of_get_next_available_child+0x5c/0x68 LR is at of_get_next_available_child+0x1c/0x68 pc : [<8043ea58>] lr : [<8043ea18>] psr: 60000193 sp : 9f863d58 ip : 00000000 fp : 9f863d74 r10: 9f89a010 r9 : 9f862000 r8 : 807bb26c r7 : 80615d5c r6 : 00000000 r5 : 60000113 r4 : 00000000 r3 : 00000000 r2 : 808770a4 r1 : 00000011 r0 : 60000113 Flags: nZCv IRQs off FIQs on Mode SVC_32 ISA ARM Segment kernel Control: 10c5387d Table: 90004019 DAC: 00000017 Process swapper/0 (pid: 1, stack limit = 0x9f862240) Stack: (0x9f863d58 to 0x9f864000) In order to fix it, return an error when 'of_id' is NULL, which happens for the non-dt case. Also checked that audio is still functional when running the dt kernel for this board after applying this patch. Signed-off-by: Fabio Estevam <fabio.estevam(a)freescale.com> --- sound/soc/fsl/imx-audmux.c | 3 +++ 1 file changed, 3 insertions(+) diff --git a/sound/soc/fsl/imx-audmux.c b/sound/soc/fsl/imx-audmux.c index ab17381..2b29b48 100644 --- a/sound/soc/fsl/imx-audmux.c +++ b/sound/soc/fsl/imx-audmux.c @@ -317,6 +317,9 @@ static int imx_audmux_probe(struct platform_device *pdev) const struct of_device_id *of_id = of_match_device(imx_audmux_dt_ids, &pdev->dev); + if (!of_id) + return -EINVAL; + res = platform_get_resource(pdev, IORESOURCE_MEM, 0); audmux_base = devm_ioremap_resource(&pdev->dev, res); if (IS_ERR(audmux_base)) -- 1.8.1.2

2 1

[alsa-devel] [PATCH] ASoC: mxs: mxs-sgtl5000: Simplify probe function
by Fabio Estevam 03 Sep '13

03 Sep '13

From: Fabio Estevam <fabio.estevam(a)freescale.com> mxs is a device tree only platform, which allows us to simplify a bit mxs_sgtl5000_probe(), because there is no need to check whether device tree is supported or not. Remove mxs_sgtl5000_probe_dt() and place its content inside mxs_sgtl5000_probe() for making the code simpler. Signed-off-by: Fabio Estevam <fabio.estevam(a)freescale.com> --- sound/soc/mxs/mxs-sgtl5000.c | 20 +++----------------- 1 file changed, 3 insertions(+), 17 deletions(-) diff --git a/sound/soc/mxs/mxs-sgtl5000.c b/sound/soc/mxs/mxs-sgtl5000.c index 4bb2737..61822cc 100644 --- a/sound/soc/mxs/mxs-sgtl5000.c +++ b/sound/soc/mxs/mxs-sgtl5000.c @@ -122,14 +122,12 @@ static struct snd_soc_card mxs_sgtl5000 = { .num_links = ARRAY_SIZE(mxs_sgtl5000_dai), }; -static int mxs_sgtl5000_probe_dt(struct platform_device *pdev) +static int mxs_sgtl5000_probe(struct platform_device *pdev) { + struct snd_soc_card *card = &mxs_sgtl5000; + int ret, i; struct device_node *np = pdev->dev.of_node; struct device_node *saif_np[2], *codec_np; - int i; - - if (!np) - return 1; /* no device tree */ saif_np[0] = of_parse_phandle(np, "saif-controllers", 0); saif_np[1] = of_parse_phandle(np, "saif-controllers", 1); @@ -152,18 +150,6 @@ static int mxs_sgtl5000_probe_dt(struct platform_device *pdev) of_node_put(saif_np[0]); of_node_put(saif_np[1]); - return 0; -} - -static int mxs_sgtl5000_probe(struct platform_device *pdev) -{ - struct snd_soc_card *card = &mxs_sgtl5000; - int ret; - - ret = mxs_sgtl5000_probe_dt(pdev); - if (ret < 0) - return ret; - /* * Set an init clock(11.28Mhz) for sgtl5000 initialization(i2c r/w). * The Sgtl5000 sysclk is derived from saif0 mclk and it's range -- 1.8.1.2

2 1

[alsa-devel] ASoC updates for v3.12
by Mark Brown 02 Sep '13

02 Sep '13

The following changes since commit 3fd8a8041b3775840c8f85b6338b61beb5057160: Merge tag 'asoc-v3.12-2' of git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound into for-next (2013-08-30 12:35:49 +0200) are available in the git repository at: git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git tags/asoc-v3.12-3 for you to fetch changes up to 5df498a27ab385294fdb7cad19c623d4da94a780: Merge remote-tracking branch 'asoc/topic/ssm2602' into tmp (2013-09-01 21:15:59 +0100) These haven't all been in a -next build yet, they work for me though. ---------------------------------------------------------------- ASoC: Final updates for v3.12 A few final updates for v3.12 - some cleanups, a bug fix for ssm2602, pop removal for rt5640 and fixes for the reporting of unidirectional links in the MXS SGTL5000 driver. ---------------------------------------------------------------- Bard Liao (1): ASoC: rt5640: change widget sequence for depop Fabio Estevam (2): ASoC: soc-pcm: Allow to specify unidirectional dai_link ASoC: mxs-sgtl5000: Configure the dai_links as unidirectional Lars-Peter Clausen (5): ASoC: dapm: Fix auto-disable for inverted controls ASoC: Remove unused control_type field from snd_soc_codec struct ASoC: Remove unused debugfs_dapm field from snd_soc_{platform,codec} struct ASoC: Remove unused sysfs_registered field from snd_soc_codec struct ASoC: ssm2602: Fix cache sync Mark Brown (6): Merge remote-tracking branch 'asoc/topic/rt5640' into asoc-dapm ASoC: fsl_spdif: Staticse non-exported symbols Merge remote-tracking branch 'asoc/topic/core' into tmp Merge remote-tracking branch 'asoc/topic/dapm' into tmp Merge remote-tracking branch 'asoc/topic/fsl' into tmp Merge remote-tracking branch 'asoc/topic/ssm2602' into tmp Nicolin Chen (2): ASoC: fsl: Add wrapping for dev_dbg() in fsl_spdif.c ASoC: fsl: Add one blank space after ':=' in Makefile include/sound/soc.h | 8 +- sound/soc/codecs/rt5640.c | 217 ++++++++++++++++++++++++++++++++----------- sound/soc/codecs/rt5640.h | 12 +++ sound/soc/codecs/ssm2602.c | 3 +- sound/soc/fsl/Makefile | 2 +- sound/soc/fsl/fsl_spdif.c | 12 +-- sound/soc/mxs/mxs-sgtl5000.c | 2 + sound/soc/soc-dapm.c | 2 + sound/soc/soc-pcm.c | 10 ++ 9 files changed, 202 insertions(+), 66 deletions(-)

2 1

[alsa-devel] [PATCH 1/5] ASoC: Remove support for reg_access_defaults
by Lars-Peter Clausen 02 Sep '13

02 Sep '13

No users of reg_access_defaults are left and new drivers are going to use regmap for this, so support for it can be removed. Signed-off-by: Lars-Peter Clausen <lars(a)metafoo.de> --- include/sound/soc.h | 24 -------------------- sound/soc/soc-cache.c | 63 --------------------------------------------------- sound/soc/soc-core.c | 9 -------- 3 files changed, 96 deletions(-) diff --git a/include/sound/soc.h b/include/sound/soc.h index 7184805..ae6efc6d 100644 --- a/include/sound/soc.h +++ b/include/sound/soc.h @@ -403,12 +403,6 @@ int snd_soc_cache_write(struct snd_soc_codec *codec, unsigned int reg, unsigned int value); int snd_soc_cache_read(struct snd_soc_codec *codec, unsigned int reg, unsigned int *value); -int snd_soc_default_volatile_register(struct snd_soc_codec *codec, - unsigned int reg); -int snd_soc_default_readable_register(struct snd_soc_codec *codec, - unsigned int reg); -int snd_soc_default_writable_register(struct snd_soc_codec *codec, - unsigned int reg); int snd_soc_platform_read(struct snd_soc_platform *platform, unsigned int reg); int snd_soc_platform_write(struct snd_soc_platform *platform, @@ -542,22 +536,6 @@ int snd_soc_put_strobe(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol); /** - * struct snd_soc_reg_access - Describes whether a given register is - * readable, writable or volatile. - * - * @reg: the register number - * @read: whether this register is readable - * @write: whether this register is writable - * @vol: whether this register is volatile - */ -struct snd_soc_reg_access { - u16 reg; - u16 read; - u16 write; - u16 vol; -}; - -/** * struct snd_soc_jack_pin - Describes a pin to update based on jack detection * * @pin: name of the pin to update @@ -760,8 +738,6 @@ struct snd_soc_codec_driver { short reg_cache_step; short reg_word_size; const void *reg_cache_default; - short reg_access_size; - const struct snd_soc_reg_access *reg_access_default; enum snd_soc_compress_type compress_type; /* codec bias level */ diff --git a/sound/soc/soc-cache.c b/sound/soc/soc-cache.c index e72f554..eaa898f 100644 --- a/sound/soc/soc-cache.c +++ b/sound/soc/soc-cache.c @@ -275,66 +275,3 @@ int snd_soc_cache_sync(struct snd_soc_codec *codec) return ret; } EXPORT_SYMBOL_GPL(snd_soc_cache_sync); - -static int snd_soc_get_reg_access_index(struct snd_soc_codec *codec, - unsigned int reg) -{ - const struct snd_soc_codec_driver *codec_drv; - unsigned int min, max, index; - - codec_drv = codec->driver; - min = 0; - max = codec_drv->reg_access_size - 1; - do { - index = (min + max) / 2; - if (codec_drv->reg_access_default[index].reg == reg) - return index; - if (codec_drv->reg_access_default[index].reg < reg) - min = index + 1; - else - max = index; - } while (min <= max); - return -1; -} - -int snd_soc_default_volatile_register(struct snd_soc_codec *codec, - unsigned int reg) -{ - int index; - - if (reg >= codec->driver->reg_cache_size) - return 1; - index = snd_soc_get_reg_access_index(codec, reg); - if (index < 0) - return 0; - return codec->driver->reg_access_default[index].vol; -} -EXPORT_SYMBOL_GPL(snd_soc_default_volatile_register); - -int snd_soc_default_readable_register(struct snd_soc_codec *codec, - unsigned int reg) -{ - int index; - - if (reg >= codec->driver->reg_cache_size) - return 1; - index = snd_soc_get_reg_access_index(codec, reg); - if (index < 0) - return 0; - return codec->driver->reg_access_default[index].read; -} -EXPORT_SYMBOL_GPL(snd_soc_default_readable_register); - -int snd_soc_default_writable_register(struct snd_soc_codec *codec, - unsigned int reg) -{ - int index; - - if (reg >= codec->driver->reg_cache_size) - return 1; - index = snd_soc_get_reg_access_index(codec, reg); - if (index < 0) - return 0; - return codec->driver->reg_access_default[index].write; -} -EXPORT_SYMBOL_GPL(snd_soc_default_writable_register); diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index 7d13c4f..b78fa93 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -4222,15 +4222,6 @@ int snd_soc_register_codec(struct device *dev, } } - if (codec_drv->reg_access_size && codec_drv->reg_access_default) { - if (!codec->volatile_register) - codec->volatile_register = snd_soc_default_volatile_register; - if (!codec->readable_register) - codec->readable_register = snd_soc_default_readable_register; - if (!codec->writable_register) - codec->writable_register = snd_soc_default_writable_register; - } - for (i = 0; i < num_dai; i++) { fixup_codec_formats(&dai_drv[i].playback); fixup_codec_formats(&dai_drv[i].capture); -- 1.8.0

2 5

[alsa-devel] [PATCH 0/5] Remove ASoC-level I/O functions from cq93vc
by Mark Brown 02 Sep '13

02 Sep '13

This patch series removes the use of the ASoC-level read and write functions from the cq93vc driver as part of a wider push to remove them completely and just use regmap for all register I/O. Since the driver is essentially doing what regmap-mmio is doing this is done by adding a MMIO regmap to the core device and using that. This is compile tested only, I don't have any hardware to run on. Since I anticipate doing more cleanups over the CODEC drivers during the next release cycle it'd be good to merge via ASoC, though there shouldn't be any overlap with the first patch. Mark Brown (5): mfd: davinci_voicecodec: Remove unused read and write functions mfd: davinci_voicecodec: Provide a regmap for register I/O ASoC: cq93vc: Use core I/O functions ASoC: cq93vc: Don't use control data for core driver data ASoC: cq93vc: Use regmap for I/O drivers/mfd/davinci_voicecodec.c | 23 +++++++++++-------- include/linux/mfd/davinci_voicecodec.h | 3 +++ sound/soc/codecs/cq93vc.c | 42 +++++++++++----------------------- 3 files changed, 29 insertions(+), 39 deletions(-)

2 9

[alsa-devel] [PATCH 00/18] ARM: s3c64xx: Let amba-pl08x driver handle DMA
by Tomasz Figa 02 Sep '13

02 Sep '13

This is first non-RFC version of my patches extending support of amba-pl08x DMA engine driver to PL080S DMA engine (PL080 modified by Samsung) found in Samsung S3C64xx SoCs. Due to changes scattered across different areas of kernel, patches are based on merged 3 branches: - for-next of Kgene's Samsung tree, - clk-next of Mike's clock tree, - next of Vinod's slave DMA tree. To ease testing I have prepared a branch in my private tree for anyone willing to check the patches out: git://github.com/tom3q/linux.git v3.12-pl080 Dependencies (already applied in my branch): - for patches 14 and 16 - CCF-based clock driver for s3c64xx. Some of the patches not related to the amba-pl08x driver itself can be likely applied into appropriate trees separately, namely: - 09/18 - ASoC: Samsung: Do not queue cyclic buffers multiple times, - 14/18 - clk: samsung: s3c64xx: Add aliases for DMA clocks. After patch 14/18, both old and new DMA drivers can be supported on S3C64xx, depending on Kconfig options. Patches 15-18 remove the old driver leaving support only for the generic pl08x driver. Feel free to drop those patches for now if we want more testing, but I don't suspect any problems. On S3C64xx-based Mini6410 and SMDK6410 boards, with I2S audio playback and capture (including full duplex operation) and also SPI using spidev: Tested-by: Tomasz Figa <tomasz.figa(a)gmail.com> Changes since RFC v2: - Added clkdev lookups to old clock driver. - Added patches removing the old DMA driver and any remaining code needed by it. - Fixed DMA support for SPI. - Added a word about PL080S to amba-pl08x.c file header. - Changed definition of LLI words from enums to macros. - Extended debugging messages to handle PL080S variant as well. - Little cleanup of LLI dumping code. - Added check for peripheral flow control, which is unsupported by PL080S to dma_set_runtime_config. - Corrected transfer size mask of PL080S. Changes since RFC v1: - Returned to original way of storing quirks as booleans, as suggested by Russell, Linus and Arnd. - Added reg_config field to pl08x_phy_chan struct, which stores variant-specific address of channel config register, as suggested by Russell. - Simplified handling of extended maximum transfer size of PL080S (no more conditional passing of 0 as length to pl08x_cctl_bits()). - Reworked LLI handling in the driver to stop casting arbitrary memory to a struct and allow different word count of LLI entry, as suggested by Linus. - Removed AMBA ID override from S3C64xx PL080 initialization code. - Fixed brokenness of Samsung DMA wrapper API, which caused cyclic buffers to be queued multiple times when DMA engine is used. - Included patch adding clock aliases for DMA engines (depends on Common Clock Framework driver for S3C64xx). - Fixed several minor stylistic issues. For reference, here is the original description of the series: One of the biggest roadblocks on the way of S3C64xx to DeviceTree support is its DMA driver, which is completely platform-specific and provides private API (s3c-dma), not even saying that its design is completely against multiplatform-awareness. The DMA controller present on this SoC series is a custom variant of ARM PrimeCell PL080 modified by Samsung to add some extra features. It is mostly compatible with original PL080, except: - CH_CONTROL2 register is added between CH_CONTROL and CH_CONFIG, - offset of CH_CONFIG register is different, - transfer size field is moved from CH_CONTROL to CH_CONTROL2, - transfer size field is extended to 24 bits, allowing much bigger single transfer, - LLI consists of one more word, to account for CH_CONTROL2 register. Since all the rest is fully compatible with standard PL080 there is no point in having separate driver just for this single variant, so I decided to look into adding support for it to the amba-pl08x driver. There was already some attempt to achieve this before, but this was before Russel's big rework of the driver to use virtual channels, making the old patches being not much of use. This RFC series is a proof of concept that I managed to make during last days of hacking. Except one patch adding clkdev lookup to clock driver (which is being replaced with a CCF-compliant driver ATM), this is enough to get memcpy and slave transfers to work on S3C64xx. I have tested this on Mini6410 and SMDK6410 boards using dmatest for memcpy and Samsung I2S with madplay/aplay for slave transfers. Unfortunately I do not have access to other platforms with PL08x so I could not test for any regressions introduced on them. Credits for two patches go to Alban Bedel, who made a series fixing this driver to make it usable with audio drivers. I rebased his patches on top of mine and corrected coding style a bit. OK, that's all. Any comments are welcome. Feel free to start throwing eggs and tomatoes if you find this awful, but I won't be upset if I get some Tested-by or Acked-by as well. ;) Alban Bedel (2): dmaengine: PL08x: Fix reading the byte count in cctl dmaengine: PL08x: Add cyclic transfer support Tomasz Figa (16): dmaengine: PL08x: Refactor pl08x_getbytes_chan() to lower indentation dmaengine: PL08x: Add support for different offset of CONFIG register dmaengine: PL08x: Rework LLI handling to be less fragile dmaengine: PL08x: Move LLI dumping code into separate function dmaengine: PL08x: Add support for PL080S variant dmaengine: PL08x: Add support for different maximum transfer size ASoC: Samsung: Do not queue cyclic buffers multiple times spi: s3c64xx: Do not require legacy DMA API in case of S3C64XX ASoC: Samsung: Do not require legacy DMA API in case of S3C64XX ARM: s3c64xx: Add support for DMA using generic amba-pl08x driver ARM: s3c64xx: clock: Add clkdev lookup for DMA clocks clk: samsung: s3c64xx: Add aliases for DMA clocks ARM: s3c64xx: Remove legacy DMA driver clk: samsung: s3c64xx: Remove clock aliases of old DMA driver spi: s3c64xx: Always select S3C64XX_PL080 when ARCH_S3C64XX is enabled ASoC: Samsung: Always select S3C64XX_PL080 when ARCH_S3C64XX is enabled arch/arm/Kconfig | 1 + arch/arm/mach-s3c64xx/Kconfig | 7 +- arch/arm/mach-s3c64xx/Makefile | 2 +- arch/arm/mach-s3c64xx/clock.c | 28 +- arch/arm/mach-s3c64xx/common.h | 5 + arch/arm/mach-s3c64xx/dma.c | 753 ------------------------------- arch/arm/mach-s3c64xx/include/mach/dma.h | 144 ++---- arch/arm/mach-s3c64xx/pl080.c | 244 ++++++++++ arch/arm/plat-samsung/devs.c | 6 +- arch/arm/plat-samsung/s3c-dma-ops.c | 13 +- drivers/clk/samsung/clk-s3c64xx.c | 4 +- drivers/dma/amba-pl08x.c | 501 ++++++++++++++------ drivers/spi/Kconfig | 2 +- include/linux/amba/pl080.h | 1 + sound/soc/samsung/Kconfig | 2 +- sound/soc/samsung/dma.c | 7 + 16 files changed, 705 insertions(+), 1015 deletions(-) delete mode 100644 arch/arm/mach-s3c64xx/dma.c create mode 100644 arch/arm/mach-s3c64xx/pl080.c -- 1.8.3.2

5 44

[alsa-devel] [PATCH 1/2] ASoC: soc-pcm: Allow to specify unidirectional dai_link
by Fabio Estevam 01 Sep '13

01 Sep '13

Add 'playback_only' and 'capture_only' fields that can be used for specifying that a dai_link has a unidirectional capability. The motivation for this is for the cases of systems, such as Freescale MX28, that has two unidirectional DAIs. Signed-off-by: Fabio Estevam <fabio.estevam(a)freescale.com> --- include/sound/soc.h | 4 ++++ sound/soc/soc-pcm.c | 10 ++++++++++ 2 files changed, 14 insertions(+) diff --git a/include/sound/soc.h b/include/sound/soc.h index 8e2ad52..774d3be 100644 --- a/include/sound/soc.h +++ b/include/sound/soc.h @@ -934,6 +934,10 @@ struct snd_soc_dai_link { /* machine stream operations */ const struct snd_soc_ops *ops; const struct snd_soc_compr_ops *compr_ops; + + /* For unidirectional dai links */ + bool playback_only; + bool capture_only; }; struct snd_soc_codec_conf { diff --git a/sound/soc/soc-pcm.c b/sound/soc/soc-pcm.c index fb70fbe..5e44863 100644 --- a/sound/soc/soc-pcm.c +++ b/sound/soc/soc-pcm.c @@ -2020,6 +2020,16 @@ int soc_new_pcm(struct snd_soc_pcm_runtime *rtd, int num) capture = 1; } + if (rtd->dai_link->playback_only) { + playback = 1; + capture = 0; + } + + if (rtd->dai_link->capture_only) { + playback = 0; + capture = 1; + } + /* create the PCM */ if (rtd->dai_link->no_pcm) { snprintf(new_name, sizeof(new_name), "(%s)", -- 1.8.1.2

2 2

[alsa-devel] [PATCH v3 00/11] ALSA: usb-audio: fix playback/capture concurrent usage
by Eldad Zack 01 Sep '13

01 Sep '13

Hi, This patch series attempts to fix a long standing problem with the concurrent usage of playback and capture on implicit feedback devices. Applies against current for-next (HEAD 68538bf2bce557c3b5fe8c59b034d45352500db1 ). v2: http://mailman.alsa-project.org/pipermail/alsa-devel/2013-August/065649.html v1: http://mailman.alsa-project.org/pipermail/alsa-devel/2013-August/065517.html * Changed "altset_idx" to "altsetting" thanks to Clemens' clarification. #3: Thanks to Daniel's test, fixed a NULL dereference caused by userspace calling snd_pcm_open() and then snd_pcm_close() immediately. #8: subs->altset_idx was left intact, since it corresponds to the array index. #10: this patch now removed subs->altset_idx, but also changes the check to be performed against fmt->altsetting. #11: New. This changes the name of snd_usb_endpoint member alt_idx to altsetting for the same reason Clemens described. Clemens and Daniel - thank you very much for your help! Cheers, Eldad Eldad Zack (11): ALSA: usb-audio: remove unused parameter from sync_ep_set_params ALSA: usb-audio: remove deactivate_endpoints() ALSA: usb-audio: prevent NULL dereference on stop trigger ALSA: usb-audio: don't deactivate URBs on in-use EP ALSA: usb-audio: void return type of snd_usb_endpoint_deactivate() ALSA: usb-audio: clear SUBSTREAM_FLAG_SYNC_EP_STARTED on error ALSA: usb-audio: correct ep use_count semantics (add set_param flag) ALSA: usb-audio: conditional interface altsetting ALSA: usb-audio: conditional concurrent usage of endpoint ALSA: usb-audio: remove altset_idx from snd_usb_substream ALSA: usb-audio: rename alt_idx to altsetting sound/usb/card.h | 4 +- sound/usb/endpoint.c | 82 ++++++++++++++++++------- sound/usb/endpoint.h | 11 +++- sound/usb/pcm.c | 168 ++++++++++++++++++++++++++++++++++++++------------- sound/usb/pcm.h | 2 + sound/usb/proc.c | 4 +- 6 files changed, 201 insertions(+), 70 deletions(-) -- 1.8.1.5

2 14

Re: [alsa-devel] [PATCH 2/2] ARM: Dove: Add the audio device to the Cubox DT
by Jean-Francois Moine 01 Sep '13

01 Sep '13

On Sat, 31 Aug 2013 12:24:31 +0100 Russell King - ARM Linux <linux(a)arm.linux.org.uk> wrote: > > According to the Dove specification, the audio controller works with > > the samples rates 44.1, 48 and 96 kHz, so, I don't see the usage of the > > external clock, except when using the two audio controllers with > > different sample rates. > > I don't see what the Dove specification has to do with that statement: > what the Dove spec says is that if you use just the internal DCO, then > only 44.1kHz, 48kHz and 96kHz are supported (with some trimming of that.) > However, the use of an external clock allows further rates to be supported. > If you have an external clock, there is no requirement to use the DCO for > those sample rates - you can if you wish, or you can use the external clock. > > The mainline driver implements the use of the DCO for the standard 44.1, > 48 and 96kHz rates, otherwise it uses the external clock if present. This > is entirely conformant with the Dove spec. Russell, I looked again at the Armada-510-Functional-Spec, and I found: - section 12.1 ... I2S Supported Features An audio sample rate (Fs) of 44.1/48/96 kHz. ... S/PDIF Supported Features The IEC60958-1, IEC60958-3, and IEC61937 specifications. An audio sample rate (Fs) of 44.1/48/96 kHz. ... - Table 436: Clocks Control Register Offset: Unit0: 0x000B1230, Unit1: 0x000B5230 ... 1:0 MCLK source RW MCLK = 256 Fs, where Fs = 44.1 kHz, 48 kHz, 96 kHz. 0x0 Playback should not be active when selecting an external MCLK, or moving from one external MCLK to another MCLK source. 0 = DCO: MCLK is from DCO. 1 = Reserved 2 = Reserved 3 = External: MCLK is from an external source. As they don't work with my TV set, may you confirm that these webradios http://kerne.online.stalig.net/live-ori.ogg http://www.tv-radio.com/station/france_bleu_breiz_izel_mp3/france_bleu_brei… http://plumfm.online.stalig.net/live-ori.ogg work for you with the external clock? (sample rates resp. 33.075, 24 and 22.05 kHz) -- Ken ar c'hentañ | ** Breizh ha Linux atav! ** Jef | http://moinejf.free.fr/

1 0

[alsa-devel] [PATCH] ASoC: fsl_spdif: Staticse non-exported symbols
by Mark Brown 01 Sep '13

01 Sep '13

From: Mark Brown <broonie(a)linaro.org> Signed-off-by: Mark Brown <broonie(a)linaro.org> --- sound/soc/fsl/fsl_spdif.c | 8 ++++---- 1 file changed, 4 insertions(+), 4 deletions(-) diff --git a/sound/soc/fsl/fsl_spdif.c b/sound/soc/fsl/fsl_spdif.c index e93dc0d..c5c228a 100644 --- a/sound/soc/fsl/fsl_spdif.c +++ b/sound/soc/fsl/fsl_spdif.c @@ -411,8 +411,8 @@ static int spdif_set_sample_rate(struct snd_pcm_substream *substream, return 0; } -int fsl_spdif_startup(struct snd_pcm_substream *substream, - struct snd_soc_dai *cpu_dai) +static int fsl_spdif_startup(struct snd_pcm_substream *substream, + struct snd_soc_dai *cpu_dai) { struct snd_soc_pcm_runtime *rtd = substream->private_data; struct fsl_spdif_priv *spdif_priv = snd_soc_dai_get_drvdata(rtd->cpu_dai); @@ -546,7 +546,7 @@ static int fsl_spdif_trigger(struct snd_pcm_substream *substream, return 0; } -struct snd_soc_dai_ops fsl_spdif_dai_ops = { +static struct snd_soc_dai_ops fsl_spdif_dai_ops = { .startup = fsl_spdif_startup, .hw_params = fsl_spdif_hw_params, .trigger = fsl_spdif_trigger, @@ -919,7 +919,7 @@ static int fsl_spdif_dai_probe(struct snd_soc_dai *dai) return 0; } -struct snd_soc_dai_driver fsl_spdif_dai = { +static struct snd_soc_dai_driver fsl_spdif_dai = { .probe = &fsl_spdif_dai_probe, .playback = { .channels_min = 2, -- 1.8.4.rc3

2 1

[alsa-devel] [PATCH] ASoC: ssm2602: Fix cache sync
by Lars-Peter Clausen 31 Aug '13

31 Aug '13

The ssm2602 uses regmap for caching not soc-cache, so we need to use regcache_sync() instead of snd_soc_cache_sync(). Signed-off-by: Lars-Peter Clausen <lars(a)metafoo.de> --- sound/soc/codecs/ssm2602.c | 3 ++- 1 file changed, 2 insertions(+), 1 deletion(-) diff --git a/sound/soc/codecs/ssm2602.c b/sound/soc/codecs/ssm2602.c index f8d30e5..492644e 100644 --- a/sound/soc/codecs/ssm2602.c +++ b/sound/soc/codecs/ssm2602.c @@ -561,8 +561,9 @@ static int ssm2602_suspend(struct snd_soc_codec *codec) static int ssm2602_resume(struct snd_soc_codec *codec) { - snd_soc_cache_sync(codec); + struct ssm2602_priv *ssm2602 = snd_soc_codec_get_drvdata(codec); + regcache_sync(ssm2602->regmap); ssm2602_set_bias_level(codec, SND_SOC_BIAS_STANDBY); return 0; -- 1.8.0

2 1

[alsa-devel] [PATCH] ASoC: cq93vc: Use table based control registration
by Mark Brown 31 Aug '13

31 Aug '13

From: Mark Brown <broonie(a)linaro.org> Saves a little code. Signed-off-by: Mark Brown <broonie(a)linaro.org> --- sound/soc/codecs/cq93vc.c | 6 ++---- 1 file changed, 2 insertions(+), 4 deletions(-) diff --git a/sound/soc/codecs/cq93vc.c b/sound/soc/codecs/cq93vc.c index 3ce535b..43737a27 100644 --- a/sound/soc/codecs/cq93vc.c +++ b/sound/soc/codecs/cq93vc.c @@ -142,10 +142,6 @@ static int cq93vc_probe(struct snd_soc_codec *codec) snd_soc_codec_set_cache_io(codec, 32, 32, SND_SOC_REGMAP); - /* Set controls */ - snd_soc_add_codec_controls(codec, cq93vc_snd_controls, - ARRAY_SIZE(cq93vc_snd_controls)); - /* Off, with power on */ cq93vc_set_bias_level(codec, SND_SOC_BIAS_STANDBY); @@ -164,6 +160,8 @@ static struct snd_soc_codec_driver soc_codec_dev_cq93vc = { .probe = cq93vc_probe, .remove = cq93vc_remove, .resume = cq93vc_resume, + .controls = cq93vc_snd_controls, + .num_controls = ARRAY_SIZE(cq93vc_snd_controls), }; static int cq93vc_platform_probe(struct platform_device *pdev) -- 1.8.4.rc3

1 0

[alsa-devel] [PATCH 1/2] ASoC: wm8400: Use supplies to manage input power
by Mark Brown 31 Aug '13

31 Aug '13

From: Mark Brown <broonie(a)linaro.org> Rather than using a fake register to manage input power create some supply widgets and use those. Signed-off-by: Mark Brown <broonie(a)linaro.org> --- sound/soc/codecs/wm8400.c | 73 ++++++++++++----------------------------------- 1 file changed, 18 insertions(+), 55 deletions(-) diff --git a/sound/soc/codecs/wm8400.c b/sound/soc/codecs/wm8400.c index d2a0928..95c33d1 100644 --- a/sound/soc/codecs/wm8400.c +++ b/sound/soc/codecs/wm8400.c @@ -32,13 +32,6 @@ #include "wm8400.h" -/* Fake register for internal state */ -#define WM8400_INTDRIVBITS (WM8400_REGISTER_COUNT + 1) -#define WM8400_INMIXL_PWR 0 -#define WM8400_AINLMUX_PWR 1 -#define WM8400_INMIXR_PWR 2 -#define WM8400_AINRMUX_PWR 3 - static struct regulator_bulk_data power[] = { { .supply = "I2S1VDD", @@ -79,10 +72,7 @@ static inline unsigned int wm8400_read(struct snd_soc_codec *codec, { struct wm8400_priv *wm8400 = snd_soc_codec_get_drvdata(codec); - if (reg == WM8400_INTDRIVBITS) - return wm8400->fake_register; - else - return wm8400_reg_read(wm8400->wm8400, reg); + return wm8400_reg_read(wm8400->wm8400, reg); } /* @@ -93,11 +83,7 @@ static int wm8400_write(struct snd_soc_codec *codec, unsigned int reg, { struct wm8400_priv *wm8400 = snd_soc_codec_get_drvdata(codec); - if (reg == WM8400_INTDRIVBITS) { - wm8400->fake_register = value; - return 0; - } else - return wm8400_set_bits(wm8400->wm8400, reg, 0xffff, value); + return wm8400_set_bits(wm8400->wm8400, reg, 0xffff, value); } static void wm8400_codec_reset(struct snd_soc_codec *codec) @@ -352,32 +338,6 @@ SOC_SINGLE("RIN34 Mute Switch", WM8400_RIGHT_LINE_INPUT_3_4_VOLUME, * _DAPM_ Controls */ -static int inmixer_event (struct snd_soc_dapm_widget *w, - struct snd_kcontrol *kcontrol, int event) -{ - u16 reg, fakepower; - - reg = snd_soc_read(w->codec, WM8400_POWER_MANAGEMENT_2); - fakepower = snd_soc_read(w->codec, WM8400_INTDRIVBITS); - - if (fakepower & ((1 << WM8400_INMIXL_PWR) | - (1 << WM8400_AINLMUX_PWR))) { - reg |= WM8400_AINL_ENA; - } else { - reg &= ~WM8400_AINL_ENA; - } - - if (fakepower & ((1 << WM8400_INMIXR_PWR) | - (1 << WM8400_AINRMUX_PWR))) { - reg |= WM8400_AINR_ENA; - } else { - reg &= ~WM8400_AINR_ENA; - } - snd_soc_write(w->codec, WM8400_POWER_MANAGEMENT_2, reg); - - return 0; -} - static int outmixer_event (struct snd_soc_dapm_widget *w, struct snd_kcontrol * kcontrol, int event) { @@ -658,27 +618,26 @@ SND_SOC_DAPM_MIXER("RIN34 PGA", WM8400_POWER_MANAGEMENT_2, 0, &wm8400_dapm_rin34_pga_controls[0], ARRAY_SIZE(wm8400_dapm_rin34_pga_controls)), +SND_SOC_DAPM_SUPPLY("INL", WM8400_POWER_MANAGEMENT_2, WM8400_AINL_ENA_SHIFT, + 0, NULL, 0), +SND_SOC_DAPM_SUPPLY("INR", WM8400_POWER_MANAGEMENT_2, WM8400_AINR_ENA_SHIFT, + 0, NULL, 0), + /* INMIXL */ -SND_SOC_DAPM_MIXER_E("INMIXL", WM8400_INTDRIVBITS, WM8400_INMIXL_PWR, 0, +SND_SOC_DAPM_MIXER("INMIXL", SND_SOC_NOPM, 0, 0, &wm8400_dapm_inmixl_controls[0], - ARRAY_SIZE(wm8400_dapm_inmixl_controls), - inmixer_event, SND_SOC_DAPM_POST_PMU | SND_SOC_DAPM_POST_PMD), + ARRAY_SIZE(wm8400_dapm_inmixl_controls)), /* AINLMUX */ -SND_SOC_DAPM_MUX_E("AILNMUX", WM8400_INTDRIVBITS, WM8400_AINLMUX_PWR, 0, - &wm8400_dapm_ainlmux_controls, inmixer_event, - SND_SOC_DAPM_POST_PMU | SND_SOC_DAPM_POST_PMD), +SND_SOC_DAPM_MUX("AILNMUX", SND_SOC_NOPM, 0, 0, &wm8400_dapm_ainlmux_controls), /* INMIXR */ -SND_SOC_DAPM_MIXER_E("INMIXR", WM8400_INTDRIVBITS, WM8400_INMIXR_PWR, 0, +SND_SOC_DAPM_MIXER("INMIXR", SND_SOC_NOPM, 0, 0, &wm8400_dapm_inmixr_controls[0], - ARRAY_SIZE(wm8400_dapm_inmixr_controls), - inmixer_event, SND_SOC_DAPM_POST_PMU | SND_SOC_DAPM_POST_PMD), + ARRAY_SIZE(wm8400_dapm_inmixr_controls)), /* AINRMUX */ -SND_SOC_DAPM_MUX_E("AIRNMUX", WM8400_INTDRIVBITS, WM8400_AINRMUX_PWR, 0, - &wm8400_dapm_ainrmux_controls, inmixer_event, - SND_SOC_DAPM_POST_PMU | SND_SOC_DAPM_POST_PMD), +SND_SOC_DAPM_MUX("AIRNMUX", SND_SOC_NOPM, 0, 0, &wm8400_dapm_ainrmux_controls), /* Output Side */ /* DACs */ @@ -789,11 +748,13 @@ static const struct snd_soc_dapm_route wm8400_dapm_routes[] = { {"LIN34 PGA", "LIN3 Switch", "LIN3"}, {"LIN34 PGA", "LIN4 Switch", "LIN4/RXN"}, /* INMIXL */ + {"INMIXL", NULL, "INL"}, {"INMIXL", "Record Left Volume", "LOMIX"}, {"INMIXL", "LIN2 Volume", "LIN2"}, {"INMIXL", "LINPGA12 Switch", "LIN12 PGA"}, {"INMIXL", "LINPGA34 Switch", "LIN34 PGA"}, /* AILNMUX */ + {"AILNMUX", NULL, "INL"}, {"AILNMUX", "INMIXL Mix", "INMIXL"}, {"AILNMUX", "DIFFINL Mix", "LIN12 PGA"}, {"AILNMUX", "DIFFINL Mix", "LIN34 PGA"}, @@ -808,12 +769,14 @@ static const struct snd_soc_dapm_route wm8400_dapm_routes[] = { /* RIN34 PGA */ {"RIN34 PGA", "RIN3 Switch", "RIN3"}, {"RIN34 PGA", "RIN4 Switch", "RIN4/RXP"}, - /* INMIXL */ + /* INMIXR */ + {"INMIXR", NULL, "INR"}, {"INMIXR", "Record Right Volume", "ROMIX"}, {"INMIXR", "RIN2 Volume", "RIN2"}, {"INMIXR", "RINPGA12 Switch", "RIN12 PGA"}, {"INMIXR", "RINPGA34 Switch", "RIN34 PGA"}, /* AIRNMUX */ + {"AIRNMUX", NULL, "INR"}, {"AIRNMUX", "INMIXR Mix", "INMIXR"}, {"AIRNMUX", "DIFFINR Mix", "RIN12 PGA"}, {"AIRNMUX", "DIFFINR Mix", "RIN34 PGA"}, -- 1.8.4.rc3

1 1

[alsa-devel] [PATCH 1/3] ASoC: Remove unused control_type field from snd_soc_codec struct
by Lars-Peter Clausen 31 Aug '13

31 Aug '13

The control_type field was used by the core to track which raw IO methods to use, but when switching to regmap this was no longer necessary and so the last user of the field was removed in commit be3ea3b9 ("ASoC: Use new register map API for ASoC generic physical I/O"). The field is now completely unused and can be removed. Signed-off-by: Lars-Peter Clausen <lars(a)metafoo.de> --- include/sound/soc.h | 1 - 1 file changed, 1 deletion(-) diff --git a/include/sound/soc.h b/include/sound/soc.h index 8e2ad52..9d4c2ab 100644 --- a/include/sound/soc.h +++ b/include/sound/soc.h @@ -705,7 +705,6 @@ struct snd_soc_codec { /* codec IO */ void *control_data; /* codec control (i2c/3wire) data */ - enum snd_soc_control_type control_type; hw_write_t hw_write; unsigned int (*hw_read)(struct snd_soc_codec *, unsigned int); unsigned int (*read)(struct snd_soc_codec *, unsigned int); -- 1.8.0

2 3

[alsa-devel] [PATCH v11 0/7] ASoC: fsl-ssi: ac97-slave support
by Markus Pargmann 31 Aug '13

31 Aug '13

Hi, this series implements ac97-slave support for fsl-ssi and adds a simple generic DT sound card. Regards, Markus Changes in v11: - Dropped phycard specific sound patches - Added a generic simple DT sound card with support for ac97 using ac97-codec - Some new bugfix patches Markus Pargmann (5): ASoC: imx-pcm-dma: Add missing include ASoC: core: Generic ac97 link reset functions ASoC: generic: simple DT sound card ASoC: fsl: Kconfig, visible config items for simple card ASoC: fsl-ssi: ac97-slave support Michael Grzeschik (2): ASoC: fsl-ssi: imx-pcm-fiq bugfix ASoC: fsl: disable ssi irq for imx Steffen Trumtrar (1): ASoC: fsl-ssi: add SSIEN errata work around .../devicetree/bindings/sound/fsl,ssi.txt | 4 + .../bindings/sound/generic,simple-dt-card.txt | 28 ++ .../devicetree/bindings/sound/soc-ac97link.txt | 28 ++ include/sound/soc.h | 2 + sound/soc/fsl/Kconfig | 23 +- sound/soc/fsl/fsl_ssi.c | 369 ++++++++++++++++----- sound/soc/fsl/imx-pcm-dma.c | 1 + sound/soc/generic/Kconfig | 1 + sound/soc/generic/Makefile | 3 +- sound/soc/generic/simple-dt-card.c | 186 +++++++++++ sound/soc/soc-core.c | 153 +++++++++ 11 files changed, 711 insertions(+), 87 deletions(-) create mode 100644 Documentation/devicetree/bindings/sound/generic,simple-dt-card.txt create mode 100644 Documentation/devicetree/bindings/sound/soc-ac97link.txt create mode 100644 sound/soc/generic/simple-dt-card.c

2 16

[alsa-devel] [PATCH 1/2] ASoC: fsl: Add wrapping for dev_dbg() in fsl_spdif.c
by Nicolin Chen 31 Aug '13

31 Aug '13

Add wrapping '\n' for dev_dbg() in fsl_spdif.c Signed-off-by: Nicolin Chen <b42378(a)freescale.com> --- sound/soc/fsl/fsl_spdif.c | 4 ++-- 1 files changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/soc/fsl/fsl_spdif.c b/sound/soc/fsl/fsl_spdif.c index e93dc0d..98741e9 100644 --- a/sound/soc/fsl/fsl_spdif.c +++ b/sound/soc/fsl/fsl_spdif.c @@ -1071,9 +1071,9 @@ 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", + dev_dbg(&pdev->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 divisor %d for %dHz sample rate", + dev_dbg(&pdev->dev, "use divisor %d for %dHz sample rate\n", spdif_priv->txclk_div[index], rate[index]); return 0; -- 1.7.1

2 2

[alsa-devel] [PATCH RFC 00/13] Adding SPDIF support to kirkwood-i2s
by Russell King - ARM Linux 30 Aug '13

30 Aug '13

This is only a RFC at present due to the need to test on Kirkwood and fix at least one bug in ASoC core code. I can't say that this stuff is fully tested, because I don't have any Kirkwood platforms, but I do have one Dove platform, which as I've said in the past doesn't run mainline. I don't run DT on it, so these patches also don't contain anything DT related. However, this patch set is aimed at sorting out the SPDIF on Dove and Kirkwood, which the Dove-based Cubox uses. The first patch in the series I've already sent, and I think Mark may have already picked up. The second is my implementation of what I think Mark committed to use the devm_* APIs for the external clock. Patches 3 and 4 fix problems which I've found with the ASoC core code, which prevent these patches from working correctly. Patch 3 I've already discussed with Mark over the weekend, and I think that's pretty much settled, but patch 4 is a definite hack. Patch 4 addresses an issue where we end up registering the CPU DAIs stream widgets twice - once against the codec DAPM context and again against the platform DAPM context. When these are registered, pointers to these widgets are saved into the CPU DAI structure, so one set of these widgets gets overwritten by the other set. The set which wins (normally the ones in the CPU DAPM context) ends up being the set which are made active. However, the platform ones remain on the DAI list, and these can end up being bound to by routes. ASoC people need to fix this problem as a pre-requisit to the following patches working. Patches 5 and 6 are just cleanups - I decided it was easier to do my own version of merging the I2S and DMA files rather than using Jean-F's patch (sorry, I just wanted something that worked...) Patch 7 is based on some changes in Rabeeh's kernel - I will have a following patch at some point to adjust these parameters to something a little saner. This just gets mainline closer to the tree which I can test. Patch 8 adds the I2S widgets to kirkwood-i2s and uses them. With this patch applied, kirkwood-i2s will be silent unless at least one of the output widgets is "powered up" - in the case of this patch, that's the "i2sdo" widget - naming following what's in the functional spec. Patch 9 and 10 add the DAPM routes to connect the I2S inputs/outputs to the codec streams, restoring audio output on these platforms. As I don't have these platforms, this should be tested by someone who has. Patch 11, again, discussed with Mark, this adds an always enabled output pin for the SPDIF transceiver, so that the output is powered up by DAPM. Without this, the SPDIF output stream goes into "active" mode during playback, but is never powered up. This results in any connected source widgets also not being powered up. Patch 12 adds the SPDIF output widget to kirkwood-i2s, called "spdifdo", again a name derived from the functional spec. The actual pin is called "AU_SPDIFDO". Patch 13 adds the IEC958 channel status data to patch 12, allowing the channel status to be set by applications using an appropriate ALSA library configuration file via the standard methods. What I haven't included is the board file (like the openrd and t5325 ones already present) for SPDIF support on Dove, as that's based on the pre-DT setups, and I'm not sure anyone in mainline is interested in that. So the above provides the components necessary, hopefully someone with ASoC & DT knowledge can bind these components together via appropriate DT bindings. For reference, the DAI link looks like this: .name = "S/PDIF1", .stream_name = "IEC958 Playback", .platform_name = "mvebu-audio.1", .cpu_dai_name = "mvebu-audio.1", .codec_dai_name = "dit-hifi", .codec_name = "spdif-dit", and the required route: static const struct snd_soc_dapm_route routes[] = { { "Playback", NULL, "spdifdo" }, };

11 71

[alsa-devel] [GIT PULL] ASoC updates for v3.12
by Mark Brown 30 Aug '13

30 Aug '13

The following changes since commit 1c9a341bbdc14051a4d8c74ea67269786c7d3736: ALSA: hda - Simplify CONFIG_SND_HDA_I915 condition (2013-08-27 14:49:59 +0200) are available in the git repository at: git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git tags/asoc-v3.12-2 for you to fetch changes up to 6d52825fd4e78b251f4844f6662a3bc4e31671c7: Merge remote-tracking branch 'asoc/topic/wm8904' into tmp (2013-08-30 11:04:35 +0100) The diffstat is large as this includes a merge up of v3.11-rc7. ---------------------------------------------------------------- ASoC: Updates for v3.12 A few more updates for v3.12, mostly small cleanups plus the addition of the DT bindings for kirkwood and the new i.MX S/PDIF driver. ---------------------------------------------------------------- Aaro Koskinen (1): ARM: OMAP: rx51: change musb mode to OTG Al Viro (1): cope with potentially long ->d_dname() output for shmem/hugetlb Alan Stern (1): USB: OHCI: add missing PCI PM callbacks to ohci-pci.c Alex Deucher (1): drm/radeon/r7xx: fix copy paste typo in golden register setup Alexey Khoroshilov (1): via-ircc: don't return zero if via_ircc_open() failed Anatolij Gustschin (1): usb: phy: fix build breakage Andrey Vagin (1): tcp: set timestamps for restored skb-s Anthony Foiani (1): sata_fsl: save irqs while coalescing Anton Blanchard (1): [SCSI] lpfc: Don't force CONFIG_GENERIC_CSUM on Ariel Elior (2): bnx2x: Protect against VFs' ndos when SR-IOV is disabled bnx2x: set VF DMAE when first function has 0 supported VFs Baruch Siach (1): sched_clock: Fix integer overflow Ben Hutchings (1): sfc: Fix lookup of default RX MAC filters when steered using ethtool Ben Skeggs (2): drm/nouveau/mc: fix race condition between constructor and request_irq() drm/nv04/disp: fix framebuffer pin refcounting Benjamin Marzinski (1): GFS2: don't overrun reserved revokes Benoit Cousson (1): MAINTAINERS: Update email address for Benoit Cousson Boris BREZILLON (1): ARM: at91: add missing uart clocks DT entries Chen Gang (1): arm64: KVM: use 'int' instead of 'u32' for variable 'target' in kvm_host.h. Chris Wilson (2): drm/i915: Don't deref pipe->cpu_transcoder in the hangcheck code drm/i915: Invalidate TLBs for the rings after a reset Christian König (1): drm/radeon: fix UVD message buffer validation Christoffer Dall (3): ARM: KVM: Fix 64-bit coprocessor handling ARM: KVM: Fix unaligned unmap_range leak KVM: ARM: Squash len warning Chuck Anderson (1): xen/smp: initialize IPI vectors before marking CPU online Dan Carpenter (6): Hostap: copying wrong data prism2_ioctl_giwaplist() nl80211: nl80211hdr_put() doesn't return an ERR_PTR GFS2: alloc_workqueue() doesn't return an ERR_PTR efs: iget_locked() doesn't return an ERR_PTR() bfs: iget_locked() doesn't return an ERR_PTR VFS: collect_mounts() should return an ERR_PTR Daniel Drake (2): x86: Don't clear olpc_ofw_header when sentinel is detected drivers/platform/olpc/olpc-ec.c: initialise earlier Daniel Gimpelevich (2): hso: Earlier catch of error condition hso: Fix stack corruption on some architectures Daniel Mack (1): ARM: OMAP2: fix musb usage for n8x0 Daniel Serpell (1): sony-laptop: Fix reporting of gfx_switch_status Daniel Vetter (1): drm/i915: unpin backing storage in dmabuf_unmap Dave Airlie (4): Merge branch 'drm-fixes-3.11' of git://people.freedesktop.org/~agd5f/linux Merge tag 'drm-intel-fixes-2013-08-15' of git://people.freedesktop.org/~danvet/drm-intel Merge branch 'drm-nouveau-next' of git://anongit.freedesktop.org/git/nouveau/linux-2.6 into drm-fixes Merge tag 'drm-intel-fixes-2013-08-23' of git://people.freedesktop.org/~danvet/drm-intel into drm-fixes David Daney (1): MIPS: Handle OCTEON BBIT instructions in FPU emulator. David S. Miller (3): Merge tag 'batman-adv-fix-for-davem' of git://git.open-mesh.org/linux-merge Merge branch 'for-davem' of git://git.kernel.org/.../linville/wireless Merge branch 'sfc-3.11' of git://git.kernel.org/.../bwh/sfc David Vrabel (3): x86/xen: do not identity map UNUSABLE regions in the machine E820 xen/events: initialize local per-cpu mask for all possible events xen/events: mask events when changing their VCPU binding Dmitry Kravkov (3): bnx2x: dropless flow control not always functional bnx2x: Change DCB context handling bnx2x: Consider DCBX remote error Duan Jiong (1): ipv6: handle Redirect ICMP Message with no Redirected Header option Emmanuel Grumbach (1): iwlwifi: pcie: disable L1 Active after pci_enable_device Fabio Estevam (8): ARM: 7819/1: fiq: Cast the first argument of flush_icache_range() ASoC: simple-card: Provide owner and MODULE_ALIAS() ASoC: fsl_spdif: Remove unnecessary dev_set_drvdata() ASoC: fsl_spdif: Reduce the noise on comments ASoC: fsl_ssi: Remove unnecessary dev_set_drvdata() ASoC: designware_i2s: Remove unnecessary dev_set_drvdata() ASoC: ep93xx-i2s: Remove unnecessary dev_set_drvdata() ASoC: fsl: Drop SND_SOC_FSL_UTILS from SND_SOC_IMX_SPDIF Geert Uytterhoeven (1): dm cache: avoid conflicting remove_mapping() in mq policy Giuseppe CAVALLARO (1): net: phy: rtl8211: fix interrupt on status link change Greg Kroah-Hartman (1): Merge tag 'iio-fixes-for-3.11c' of git://git.kernel.org/.../jic23/iio into staging-linus Guillaume Clement (1): gma500: Fix SDVO turning off randomly Guy Cohen (1): Revert "iwlwifi: pcie: clear RFKILL interrupt in AMPG" Hannes Frederic Sowa (2): ipv6: remove max_addresses check from ipv6_create_tempaddr ipv6: drop packets with multiple fragmentation headers Ian Abbott (1): staging: comedi: bug-fix NULL pointer dereference on failed attach Ilia Mirkin (1): drm/nouveau/fb: fix null derefs in nv49 and nv4e init Ingo Molnar (3): Merge branch 'timers/urgent' of git://git.kernel.org/.../frederic/linux-dynticks into timers/urgent Merge branch 'fortglx/3.11/time' of git://git.linaro.org/people/jstultz/linux into timers/urgent Merge tag 'amd_ucode_fixes' of git://git.kernel.org/.../bp/bp into x86/urgent Jean-Francois Moine (2): ASoc: kirkwood: add DT support to the mvebu audio subsystem ASoc: kirkwood: Use the Kirkwood audio driver in Dove boards Joern Rennecke (1): ARC: [lib] strchr breakage in Big-endian configuration Johannes Berg (3): iwlwifi: mvm: disconnect if time event scheduling fails cfg80211: don't request disconnect if not connected Revert "genetlink: fix family dump race" John W. Linville (3): Merge branch 'for-john' of git://git.kernel.org/.../iwlwifi/iwlwifi-fixes Merge branch 'for-john' of git://git.kernel.org/.../jberg/mac80211 Merge branch 'master' of git://git.kernel.org/.../linville/wireless into for-davem Julien Grall (1): xen/arm: missing put_cpu in xen_percpu_init Jussi Kivilinna (1): zd1201: do not use stack as URB transfer_buffer Kukjin Kim (1): ARM: SAMSUNG: fix to support for missing cpu specific map_io Kuninori Morimoto (1): ASoC: fsi: reserve prefetch period on DMA transferring Lars-Peter Clausen (8): ASoC: dmic: Convert table based DAPM setup ASoC: dapm: Fix marking widgets dirty when a route is added ASoC: tlv320aic32x4: Convert table based control and DAPM setup ASoC: wm8904: Remove unnecessary call to snd_soc_dapm_new_widgets() ASoC: jack: Remove unnecessary call to snd_soc_dapm_new_widgets() ASoC: Call snd_soc_dapm_new_widgets() only once during card initialization ASoC: Move call to snd_soc_dapm_new_widgets() after snd_soc_dapm_auto_nc_codec_pins() ASoC: Pass card instead of dapm context to snd_soc_dapm_new_widgets() Li Zefan (1): cpuset: fix a regression in validating config change Li Zhong (1): nohz: fix compile warning in tick_nohz_init() Linus Lüssing (1): batman-adv: check return type of unicast packet preparations Linus Torvalds (27): Merge branch 'drm-fixes' of git://people.freedesktop.org/~airlied/linux Merge branch 'timers-urgent-for-linus' of git://git.kernel.org/.../tip/tip Merge branch 'x86-urgent-for-linus' of git://git.kernel.org/.../tip/tip Merge git://git.kernel.org/.../steve/gfs2-3.0-fixes proc: more readdir conversion bug-fixes Merge tag 'pinctrl-for-v3.11-3' of git://git.kernel.org/.../linusw/linux-pinctrl Merge tag 'for-linus' of git://git.kernel.org/pub/scm/virt/kvm/kvm Merge tag 'arm64-stable' of git://git.kernel.org/.../cmarinas/linux-aarch64 Merge branch 'upstream' of git://git.linux-mips.org/pub/scm/ralf/upstream-linus Merge tag 'stable/for-linus-3.11-rc6-tag' of git://git.kernel.org/.../xen/tip Revert "x86 get_unmapped_area(): use proper mmap base for bottom-up direction" Merge tag 'dm-3.11-fixes' of git://git.kernel.org/.../device-mapper/linux-dm Merge tag 'devicetree-fixes-for-3.11' of git://sources.calxeda.com/kernel/linux Merge tag 'fixes-for-linus' of git://git.kernel.org/.../arm/arm-soc Merge branch 'linux-next' of git://cavan.codon.org.uk/platform-drivers-x86 Merge branch 'akpm' (patches from Andrew Morton) Merge git://git.kernel.org/.../davem/net Merge branch 'drm-fixes' of git://people.freedesktop.org/~airlied/linux Merge branch 'for-3.11-fixes' of git://git.kernel.org/.../tj/cgroup Merge branch 'for-3.11-fixes' of git://git.kernel.org/.../tj/libata Merge tag 'scsi-fixes' of git://git.kernel.org/.../jejb/scsi Merge tag 'acpi-3.11-rc7' of git://git.kernel.org/.../rafael/linux-pm Merge branch 'for-linus' of git://git.kernel.org/.../viro/vfs Merge branch 'fixes' of git://git.linaro.org/people/rmk/linux-arm Merge tag 'usb-3.11-rc7' of git://git.kernel.org/.../gregkh/usb Merge tag 'staging-3.11-rc7' of git://git.kernel.org/.../gregkh/staging Linux 3.11-rc7 Maarten Lankhorst (2): drm/nouveau/ltcg: fix ltcg memory initialization after suspend drm/nouveau/ltcg: fix allocating memory as free Marc Zyngier (3): arm64: KVM: fix 2-level page tables unmapping arm64: KVM: perform save/restore of PAR_EL1 arm64: KVM: add missing dsb before invalidating Stage-2 TLBs Mark Brown (11): Merge remote-tracking branch 'asoc/topic/dmic' into asoc-core Merge remote-tracking branch 'asoc/topic/tlv320aic32x4' into asoc-core Merge remote-tracking branch 'asoc/topic/dapm' into asoc-core Merge remote-tracking branch 'asoc/topic/core' into tmp Merge remote-tracking branch 'asoc/topic/designware' into tmp Merge remote-tracking branch 'asoc/topic/ep93xx' into tmp Merge remote-tracking branch 'asoc/topic/fsi' into tmp Merge remote-tracking branch 'asoc/topic/fsl' into tmp Merge remote-tracking branch 'asoc/topic/kirkwood' into tmp Merge remote-tracking branch 'asoc/topic/samsung' into tmp Merge remote-tracking branch 'asoc/topic/wm8904' into tmp Mark Langsdorf (1): sata, highbank: fix ordering of SGPIO signals Martin Peschke (3): [SCSI] zfcp: fix lock imbalance by reworking request queue locking [SCSI] zfcp: fix schedule-inside-lock in scsi_device list loops [SCSI] zfcp: remove access control tables interface (keep sysfs files) Matthew Garrett (1): Revert "hp-wmi: Enable hotkeys on some systems" Maxime Ripard (3): pinctrl: sunxi: Read register before writing to it in irq_set_type pinctrl: sunxi: Fix gpio_set behaviour pinctrl: sunxi: Add spinlocks Michal Hocko (1): memcg: get rid of swapaccount leftovers Nicolas Ferre (2): ARM: at91/DT: at91sam9x5ek: fix USB host property to enable port C ARM: at91/DT: fix at91sam9n12ek memory node Nicolas Pitre (1): ARM: 7816/1: CONFIG_KUSER_HELPERS: fix help text Nicolin Chen (1): ASoC: fsl: Add S/PDIF machine driver Oleg Nesterov (1): proc: kill the extra proc_readfd_common()->dir_emit_dots() Olof Johansson (2): Merge tag 'samsung-fixes-2' of git://git.kernel.org/.../kgene/linux-samsung into fixes Merge tag 'at91-fixes' of git://github.com/at91linux/linux-at91 into fixes Pali Rohár (1): drm/nouveau: fix reclocking on nv40 Paolo Bonzini (2): Merge branch 'kvm-arm64/fixes-3.11-rc4' of git://git.kernel.org/.../maz/arm-platforms into kvm-master Merge tag 'kvm-arm-fixes-3.11' of git://git.linaro.org/people/cdall/linux-kvm-arm into kvm-master Peter Meerwald (1): iio: adjd_s311: Fix non-scan mode data read Peter Wu (1): r8169: remember WOL preferences on driver load Radu Caragea (1): x86 get_unmapped_area: Access mmap_legacy_base through mm_struct member Rafael J. Wysocki (1): Revert "ACPI / video: Always call acpi_video_init_brightness() on init" Rafał Miłecki (1): drm/radeon: fix WREG32_OR macro setting bits in a register Randy Dunlap (1): kernel: fix new kernel-doc warning in wait.c Richard Genoud (1): proc: return on proc_readdir error Richard Laager (1): lib/lz4: correct the LZ4 license Roland Dreier (1): [SCSI] sg: Fix user memory corruption when SG_IO is interrupted by a signal Russell King (1): Merge branch 'security-fixes' into fixes Sathya Perla (1): be2net: fix disabling TX in be_close() Sekhar Nori (1): ARM: davinci: nand: specify ecc strength Stanislaw Gruszka (1): iwlwifi: dvm: fix calling ieee80211_chswitch_done() with NULL Stefano Stabellini (1): Merging v3.10-rc2 as I need to apply a fix for Stephen Warren (1): ARM: tegra: always enable USB VBUS regulators Steven Rostedt (1): nohz: Do not warn about unstable tsc unless user uses nohz_full Steven Whitehouse (2): GFS2: Fix typo in gfs2_create_inode() GFS2: Check for glock already held in gfs2_getxattr Tejun Heo (1): GFS2: WQ_NON_REENTRANT is meaningless and going away Terry Suereth (1): libata: apply behavioral quirks to sil3826 PMP Tomasz Figa (1): ASoC: Samsung: Do not queue cyclic buffers multiple times Torsten Kaiser (2): x86, microcode, AMD: Make cpu_has_amd_erratum() use the correct struct cpuinfo_x86 x86, microcode, AMD: Fix early microcode loading Toshiaki Makita (1): bridge: Use the correct bit length for bitmap functions in the VLAN code Vijaya Kumar K (1): ARM: 7815/1: kexec: offline non panic CPUs on Kdump panic Vlad Yasevich (3): macvtap: simplify usage of tap_features macvtap: Correctly set tap features when IFF_VNET_HDR is disabled. macvtap: Ignore tap features when VNET_HDR is off Vyacheslav Dubeyko (2): nilfs2: remove double bio_put() in nilfs_end_bio_write() for BIO_EOPNOTSUPP error nilfs2: fix issue with counting number of bio requests for BIO_EOPNOTSUPP error detection Wei Yongjun (2): sony-laptop: return a negative error code in sonypi_compat_init() ASoC: fsl_spdif: remove redundant dev_err call in fsl_spdif_probe() Will Deacon (2): arm64: perf: fix array out of bounds access in armpmu_map_hw_event() arm64: perf: fix event validation for software group leaders Willem de Bruijn (1): packet: restore packet statistics tp_packets to include drops Wladislav Wiebe (1): of: fdt: fix memory initialization for expanded DT Yuval Mintz (1): bnx2x: prevent VF benign attentions stephen hemminger (1): ipv4: expose IPV4_DEVCONF .../devicetree/bindings/sound/imx-audio-spdif.txt | 34 +++++ .../devicetree/bindings/sound/mvebu-audio.txt | 29 ++++ Documentation/kernel-parameters.txt | 2 +- MAINTAINERS | 6 +- Makefile | 2 +- arch/arc/lib/strchr-700.S | 10 ++ arch/arm/boot/dts/at91sam9n12ek.dts | 4 +- arch/arm/boot/dts/at91sam9x5ek.dtsi | 5 +- arch/arm/boot/dts/tegra20-seaboard.dts | 2 + arch/arm/boot/dts/tegra20-trimslice.dts | 2 + arch/arm/boot/dts/tegra20-whistler.dts | 4 + arch/arm/kernel/fiq.c | 3 +- arch/arm/kernel/machine_kexec.c | 1 + arch/arm/kvm/coproc.c | 26 +++- arch/arm/kvm/coproc.h | 3 + arch/arm/kvm/coproc_a15.c | 6 +- arch/arm/kvm/mmio.c | 3 +- arch/arm/kvm/mmu.c | 36 +++-- arch/arm/mach-at91/at91sam9x5.c | 2 + arch/arm/mach-davinci/board-dm355-leopard.c | 1 + arch/arm/mach-davinci/board-dm644x-evm.c | 1 + arch/arm/mach-davinci/board-dm646x-evm.c | 1 + arch/arm/mach-davinci/board-neuros-osd2.c | 1 + arch/arm/mach-omap2/board-n8x0.c | 4 - arch/arm/mach-omap2/board-rx51.c | 2 +- arch/arm/mach-omap2/usb-musb.c | 5 +- arch/arm/mm/Kconfig | 9 +- arch/arm/plat-samsung/init.c | 5 +- arch/arm/plat-samsung/s3c-dma-ops.c | 13 +- arch/arm/xen/enlighten.c | 1 + arch/arm64/include/asm/kvm_asm.h | 17 ++- arch/arm64/include/asm/kvm_host.h | 2 +- arch/arm64/kernel/perf_event.c | 10 +- arch/arm64/kvm/hyp.S | 13 ++ arch/arm64/kvm/sys_regs.c | 3 + arch/mips/math-emu/cp1emu.c | 26 ++++ arch/x86/include/asm/bootparam_utils.h | 4 +- arch/x86/include/asm/microcode_amd.h | 2 +- arch/x86/kernel/cpu/amd.c | 20 +-- arch/x86/kernel/microcode_amd.c | 27 ++-- arch/x86/kernel/microcode_amd_early.c | 27 ++-- arch/x86/kernel/sys_x86_64.c | 2 +- arch/x86/mm/mmap.c | 8 +- arch/x86/xen/setup.c | 22 +++ arch/x86/xen/smp.c | 11 +- drivers/acpi/video.c | 11 +- drivers/ata/libata-pmp.c | 12 +- drivers/ata/sata_fsl.c | 5 +- drivers/ata/sata_highbank.c | 4 +- drivers/gpu/drm/gma500/psb_intel_sdvo.c | 3 +- drivers/gpu/drm/i915/i915_gem_dmabuf.c | 8 ++ drivers/gpu/drm/i915/i915_reg.h | 2 + drivers/gpu/drm/i915/intel_display.c | 86 ++++++++---- drivers/gpu/drm/i915/intel_ringbuffer.c | 12 ++ drivers/gpu/drm/nouveau/core/core/mm.c | 4 + drivers/gpu/drm/nouveau/core/include/subdev/mc.h | 7 +- drivers/gpu/drm/nouveau/core/subdev/fb/ramnv49.c | 12 +- drivers/gpu/drm/nouveau/core/subdev/fb/ramnv4e.c | 4 +- drivers/gpu/drm/nouveau/core/subdev/ltcg/nvc0.c | 34 +++-- drivers/gpu/drm/nouveau/core/subdev/mc/base.c | 6 +- drivers/gpu/drm/nouveau/core/subdev/mc/nv04.c | 3 +- drivers/gpu/drm/nouveau/core/subdev/mc/nv44.c | 3 +- drivers/gpu/drm/nouveau/core/subdev/mc/nv50.c | 3 +- drivers/gpu/drm/nouveau/core/subdev/mc/nv98.c | 3 +- drivers/gpu/drm/nouveau/core/subdev/mc/nvc0.c | 3 +- drivers/gpu/drm/nouveau/dispnv04/crtc.c | 58 +++++--- drivers/gpu/drm/nouveau/dispnv04/disp.h | 1 + drivers/gpu/drm/nouveau/nouveau_display.c | 3 + drivers/gpu/drm/nouveau/nv40_pm.c | 2 +- drivers/gpu/drm/radeon/radeon.h | 2 +- drivers/gpu/drm/radeon/radeon_uvd.c | 8 ++ drivers/gpu/drm/radeon/rv770.c | 12 +- drivers/iio/light/adjd_s311.c | 3 +- drivers/md/dm-cache-policy-mq.c | 16 +-- drivers/net/ethernet/broadcom/bnx2x/bnx2x.h | 2 + drivers/net/ethernet/broadcom/bnx2x/bnx2x_dcb.c | 49 +++++-- drivers/net/ethernet/broadcom/bnx2x/bnx2x_dcb.h | 3 + drivers/net/ethernet/broadcom/bnx2x/bnx2x_main.c | 45 +++++-- drivers/net/ethernet/broadcom/bnx2x/bnx2x_sriov.c | 56 ++++---- drivers/net/ethernet/emulex/benet/be_main.c | 2 +- drivers/net/ethernet/realtek/r8169.c | 2 +- drivers/net/ethernet/sfc/filter.c | 2 +- drivers/net/irda/via-ircc.c | 6 +- drivers/net/macvtap.c | 18 +-- drivers/net/phy/realtek.c | 4 +- drivers/net/usb/hso.c | 15 ++- drivers/net/wireless/hostap/hostap_ioctl.c | 4 +- drivers/net/wireless/iwlwifi/dvm/mac80211.c | 5 +- drivers/net/wireless/iwlwifi/iwl-prph.h | 2 - drivers/net/wireless/iwlwifi/mvm/time-event.c | 33 +++-- drivers/net/wireless/iwlwifi/pcie/rx.c | 8 -- drivers/net/wireless/iwlwifi/pcie/trans.c | 10 +- drivers/net/wireless/zd1201.c | 4 +- drivers/of/fdt.c | 2 + drivers/pinctrl/pinctrl-sunxi.c | 66 ++++++++- drivers/pinctrl/pinctrl-sunxi.h | 2 + drivers/platform/olpc/olpc-ec.c | 2 +- drivers/platform/x86/hp-wmi.c | 16 --- drivers/platform/x86/sony-laptop.c | 8 +- drivers/s390/scsi/zfcp_erp.c | 29 +++- drivers/s390/scsi/zfcp_qdio.c | 8 +- drivers/s390/scsi/zfcp_sysfs.c | 14 ++ drivers/scsi/Kconfig | 1 - drivers/staging/comedi/drivers.c | 2 +- drivers/usb/host/ohci-pci.c | 5 + drivers/usb/phy/phy-fsl-usb.h | 2 +- drivers/usb/phy/phy-fsm-usb.c | 2 +- drivers/xen/events.c | 13 +- fs/bfs/inode.c | 2 +- fs/bio.c | 20 ++- fs/dcache.c | 11 ++ fs/efs/inode.c | 2 +- fs/gfs2/glock.c | 8 +- fs/gfs2/glops.c | 18 ++- fs/gfs2/inode.c | 6 +- fs/gfs2/main.c | 2 +- fs/hugetlbfs/inode.c | 8 +- fs/namespace.c | 2 +- fs/nilfs2/segbuf.c | 5 +- fs/proc/fd.c | 2 - fs/proc/generic.c | 2 +- fs/proc/root.c | 4 +- include/linux/dcache.h | 1 + include/linux/inetdevice.h | 34 +---- include/linux/ipv6.h | 1 + include/linux/mm_types.h | 1 + include/linux/sched.h | 1 - include/linux/wait.h | 57 ++++++++ include/net/ip6_route.h | 2 + include/sound/soc-dapm.h | 2 +- include/uapi/linux/ip.h | 34 +++++ init/Kconfig | 2 +- kernel/cpuset.c | 14 +- kernel/time/sched_clock.c | 2 +- kernel/time/tick-sched.c | 5 +- kernel/wait.c | 3 +- lib/lz4/lz4_compress.c | 4 +- lib/lz4/lz4_decompress.c | 6 +- lib/lz4/lz4hc_compress.c | 4 +- mm/memcontrol.c | 1 - mm/shmem.c | 8 +- net/batman-adv/unicast.c | 10 +- net/bridge/br_fdb.c | 10 +- net/bridge/br_netlink.c | 4 +- net/bridge/br_vlan.c | 4 +- net/ipv4/tcp.c | 7 + net/ipv6/addrconf.c | 10 +- net/ipv6/ndisc.c | 4 +- net/ipv6/reassembly.c | 5 + net/ipv6/route.c | 21 +++ net/netlink/genetlink.c | 7 - net/packet/af_packet.c | 2 + net/wireless/nl80211.c | 22 +-- net/wireless/sme.c | 10 +- sound/soc/cirrus/ep93xx-i2s.c | 2 - sound/soc/codecs/dmic.c | 17 +-- sound/soc/codecs/tlv320aic32x4.c | 22 +-- sound/soc/codecs/wm8904.c | 1 - sound/soc/dwc/designware_i2s.c | 5 +- sound/soc/fsl/Kconfig | 10 ++ sound/soc/fsl/Makefile | 2 + sound/soc/fsl/fsl_spdif.c | 17 +-- sound/soc/fsl/fsl_ssi.c | 1 - sound/soc/fsl/imx-spdif.c | 148 +++++++++++++++++++++ sound/soc/generic/simple-card.c | 2 + sound/soc/kirkwood/Kconfig | 4 +- sound/soc/kirkwood/kirkwood-i2s.c | 26 +++- sound/soc/samsung/dma.c | 7 + sound/soc/sh/fsi.c | 51 ++++--- sound/soc/soc-core.c | 9 +- sound/soc/soc-dapm.c | 9 +- sound/soc/soc-jack.c | 2 - 172 files changed, 1316 insertions(+), 598 deletions(-) create mode 100644 Documentation/devicetree/bindings/sound/imx-audio-spdif.txt create mode 100644 Documentation/devicetree/bindings/sound/mvebu-audio.txt create mode 100644 sound/soc/fsl/imx-spdif.c

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[alsa-devel] [PATCH] ASoC: fsl: Drop SND_SOC_FSL_UTILS from SND_SOC_IMX_SPDIF
by Fabio Estevam 30 Aug '13

30 Aug '13

From: Fabio Estevam <fabio.estevam(a)freescale.com> SND_SOC_FSL_UTILS is only used by PowerPC machines, so let's drop it in the i.mx case. Signed-off-by: Fabio Estevam <fabio.estevam(a)freescale.com> --- sound/soc/fsl/Kconfig | 1 - 1 file changed, 1 deletion(-) diff --git a/sound/soc/fsl/Kconfig b/sound/soc/fsl/Kconfig index a708380..704e246 100644 --- a/sound/soc/fsl/Kconfig +++ b/sound/soc/fsl/Kconfig @@ -197,7 +197,6 @@ config SND_SOC_IMX_SPDIF tristate "SoC Audio support for i.MX boards with S/PDIF" select SND_SOC_IMX_PCM_DMA select SND_SOC_FSL_SPDIF - select SND_SOC_FSL_UTILS select SND_SOC_SPDIF help SoC Audio support for i.MX boards with S/PDIF -- 1.8.1.2

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[alsa-devel] [PATCH v12] ASoC: fsl: Add S/PDIF machine driver
by Nicolin Chen 29 Aug '13

29 Aug '13

This patch implements a device-tree-only machine driver for Freescale i.MX series Soc. It works with spdif_transmitter/spdif_receiver and fsl_spdif.c drivers. Signed-off-by: Nicolin Chen <b42378(a)freescale.com> --- Changelog v11->v12: * Dropped unused spdif_pdev. * Register spdif-dit/dir depending on DT bool property. v10->v11: * Use boolean properties for spdif-out/in switch instead of codec phandles. * Accordingly create dummy codec driver in probe() instead of DT nodes. .../devicetree/bindings/sound/imx-audio-spdif.txt | 34 +++++ sound/soc/fsl/Kconfig | 11 ++ sound/soc/fsl/Makefile | 2 + sound/soc/fsl/imx-spdif.c | 148 ++++++++++++++++++++ 4 files changed, 195 insertions(+), 0 deletions(-) create mode 100644 Documentation/devicetree/bindings/sound/imx-audio-spdif.txt create mode 100644 sound/soc/fsl/imx-spdif.c diff --git a/Documentation/devicetree/bindings/sound/imx-audio-spdif.txt b/Documentation/devicetree/bindings/sound/imx-audio-spdif.txt new file mode 100644 index 0000000..7d13479 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/imx-audio-spdif.txt @@ -0,0 +1,34 @@ +Freescale i.MX audio complex with S/PDIF transceiver + +Required properties: + + - compatible : "fsl,imx-audio-spdif" + + - model : The user-visible name of this sound complex + + - spdif-controller : The phandle of the i.MX S/PDIF controller + + +Optional properties: + + - spdif-out : This is a boolean property. If present, the transmitting + function of S/PDIF will be enabled, indicating there's a physical + S/PDIF out connector/jack on the board or it's connecting to some + other IP block, such as an HDMI encoder/display-controller. + + - spdif-in : This is a boolean property. If present, the receiving + function of S/PDIF will be enabled, indicating there's a physical + S/PDIF in connector/jack on the board. + +* Note: At least one of these two properties should be set in the DT binding. + + +Example: + +sound-spdif { + compatible = "fsl,imx-audio-spdif"; + model = "imx-spdif"; + spdif-controller = <&spdif>; + spdif-out; + spdif-in; +}; diff --git a/sound/soc/fsl/Kconfig b/sound/soc/fsl/Kconfig index cd088cc..a708380 100644 --- a/sound/soc/fsl/Kconfig +++ b/sound/soc/fsl/Kconfig @@ -193,6 +193,17 @@ config SND_SOC_IMX_SGTL5000 Say Y if you want to add support for SoC audio on an i.MX board with a sgtl5000 codec. +config SND_SOC_IMX_SPDIF + tristate "SoC Audio support for i.MX boards with S/PDIF" + select SND_SOC_IMX_PCM_DMA + select SND_SOC_FSL_SPDIF + select SND_SOC_FSL_UTILS + select SND_SOC_SPDIF + help + SoC Audio support for i.MX boards with S/PDIF + Say Y if you want to add support for SoC audio on an i.MX board with + a S/DPDIF. + config SND_SOC_IMX_MC13783 tristate "SoC Audio support for I.MX boards with mc13783" depends on MFD_MC13783 && ARM diff --git a/sound/soc/fsl/Makefile b/sound/soc/fsl/Makefile index 4b5970e..e2aaff7 100644 --- a/sound/soc/fsl/Makefile +++ b/sound/soc/fsl/Makefile @@ -45,6 +45,7 @@ snd-soc-mx27vis-aic32x4-objs := mx27vis-aic32x4.o snd-soc-wm1133-ev1-objs := wm1133-ev1.o snd-soc-imx-sgtl5000-objs := imx-sgtl5000.o snd-soc-imx-wm8962-objs := imx-wm8962.o +snd-soc-imx-spdif-objs :=imx-spdif.o snd-soc-imx-mc13783-objs := imx-mc13783.o obj-$(CONFIG_SND_SOC_EUKREA_TLV320) += snd-soc-eukrea-tlv320.o @@ -53,4 +54,5 @@ obj-$(CONFIG_SND_SOC_MX27VIS_AIC32X4) += snd-soc-mx27vis-aic32x4.o obj-$(CONFIG_SND_MXC_SOC_WM1133_EV1) += snd-soc-wm1133-ev1.o obj-$(CONFIG_SND_SOC_IMX_SGTL5000) += snd-soc-imx-sgtl5000.o obj-$(CONFIG_SND_SOC_IMX_WM8962) += snd-soc-imx-wm8962.o +obj-$(CONFIG_SND_SOC_IMX_SPDIF) += snd-soc-imx-spdif.o obj-$(CONFIG_SND_SOC_IMX_MC13783) += snd-soc-imx-mc13783.o diff --git a/sound/soc/fsl/imx-spdif.c b/sound/soc/fsl/imx-spdif.c new file mode 100644 index 0000000..816013b --- /dev/null +++ b/sound/soc/fsl/imx-spdif.c @@ -0,0 +1,148 @@ +/* + * Copyright (C) 2013 Freescale Semiconductor, Inc. + * + * The code contained herein is licensed under the GNU General Public + * License. You may obtain a copy of the GNU General Public License + * Version 2 or later at the following locations: + * + * http://www.opensource.org/licenses/gpl-license.html + * http://www.gnu.org/copyleft/gpl.html + */ + +#include <linux/module.h> +#include <linux/of_platform.h> +#include <sound/soc.h> + +struct imx_spdif_data { + struct snd_soc_dai_link dai[2]; + struct snd_soc_card card; + struct platform_device *txdev; + struct platform_device *rxdev; +}; + +static int imx_spdif_audio_probe(struct platform_device *pdev) +{ + struct device_node *spdif_np, *np = pdev->dev.of_node; + struct imx_spdif_data *data; + int ret = 0, num_links = 0; + + spdif_np = of_parse_phandle(np, "spdif-controller", 0); + if (!spdif_np) { + dev_err(&pdev->dev, "failed to find spdif-controller\n"); + ret = -EINVAL; + goto end; + } + + data = devm_kzalloc(&pdev->dev, sizeof(*data), GFP_KERNEL); + if (!data) { + dev_err(&pdev->dev, "failed to allocate memory\n"); + ret = -ENOMEM; + goto end; + } + + if (of_property_read_bool(np, "spdif-out")) { + data->dai[num_links].name = "S/PDIF TX"; + data->dai[num_links].stream_name = "S/PDIF PCM Playback"; + data->dai[num_links].codec_dai_name = "dit-hifi"; + data->dai[num_links].codec_name = "spdif-dit"; + data->dai[num_links].cpu_of_node = spdif_np; + data->dai[num_links].platform_of_node = spdif_np; + num_links++; + + data->txdev = platform_device_register_simple("spdif-dit", -1, NULL, 0); + if (IS_ERR(data->txdev)) { + ret = PTR_ERR(data->txdev); + dev_err(&pdev->dev, "register dit failed: %d\n", ret); + goto end; + } + } + + if (of_property_read_bool(np, "spdif-in")) { + data->dai[num_links].name = "S/PDIF RX"; + data->dai[num_links].stream_name = "S/PDIF PCM Capture"; + data->dai[num_links].codec_dai_name = "dir-hifi"; + data->dai[num_links].codec_name = "spdif-dir"; + data->dai[num_links].cpu_of_node = spdif_np; + data->dai[num_links].platform_of_node = spdif_np; + num_links++; + + data->rxdev = platform_device_register_simple("spdif-dir", -1, NULL, 0); + if (IS_ERR(data->rxdev)) { + ret = PTR_ERR(data->rxdev); + dev_err(&pdev->dev, "register dir failed: %d\n", ret); + goto error_dit; + } + } + + if (!num_links) { + dev_err(&pdev->dev, "no enabled S/PDIF DAI link\n"); + goto error_dir; + } + + data->card.dev = &pdev->dev; + data->card.num_links = num_links; + data->card.dai_link = data->dai; + + ret = snd_soc_of_parse_card_name(&data->card, "model"); + if (ret) + goto error_dir; + + ret = snd_soc_register_card(&data->card); + if (ret) { + dev_err(&pdev->dev, "snd_soc_register_card failed: %d\n", ret); + goto error_dir; + } + + platform_set_drvdata(pdev, data); + + goto end; + +error_dir: + if (data->rxdev) + platform_device_unregister(data->rxdev); +error_dit: + if (data->txdev) + platform_device_unregister(data->txdev); +end: + if (spdif_np) + of_node_put(spdif_np); + + return ret; +} + +static int imx_spdif_audio_remove(struct platform_device *pdev) +{ + struct imx_spdif_data *data = platform_get_drvdata(pdev); + + if (data->rxdev) + platform_device_unregister(data->rxdev); + if (data->txdev) + platform_device_unregister(data->txdev); + + snd_soc_unregister_card(&data->card); + + return 0; +} + +static const struct of_device_id imx_spdif_dt_ids[] = { + { .compatible = "fsl,imx-audio-spdif", }, + { /* sentinel */ } +}; +MODULE_DEVICE_TABLE(of, imx_spdif_dt_ids); + +static struct platform_driver imx_spdif_driver = { + .driver = { + .name = "imx-spdif", + .owner = THIS_MODULE, + .of_match_table = imx_spdif_dt_ids, + }, + .probe = imx_spdif_audio_probe, + .remove = imx_spdif_audio_remove, +}; + +module_platform_driver(imx_spdif_driver); + +MODULE_AUTHOR("Freescale Semiconductor, Inc."); +MODULE_DESCRIPTION("Freescale i.MX S/PDIF machine driver"); +MODULE_LICENSE("GPL v2"); +MODULE_ALIAS("platform:imx-spdif"); -- 1.7.1

3 2

[alsa-devel] [PATCH] alsa-info: Run without lspci (if there is no pci bus)
by David Henningsson 29 Aug '13

29 Aug '13

On some embedded ARM hardware, there is no pci bus. In that case, do not require lspci so we can still get as much info as we can. Signed-off-by: David Henningsson <david.henningsson(a)canonical.com> --- alsa/utils/alsa-info.sh | 13 +++++++++---- 1 file changed, 9 insertions(+), 4 deletions(-) diff --git a/alsa/utils/alsa-info.sh b/alsa/utils/alsa-info.sh index 9d996aa..2280af5 100755 --- a/alsa/utils/alsa-info.sh +++ b/alsa/utils/alsa-info.sh @@ -393,8 +393,10 @@ trap cleanup 0 if [ "$PROCEED" = "yes" ]; then if [ -z "$LSPCI" ]; then - echo "This script requires lspci. Please install it, and re-run this script." - exit 0 + if [ -d /sys/bus/pci ]; then + echo "This script requires lspci. Please install it, and re-run this script." + exit 0 + fi fi #Fetch the info and store in temp files/variables @@ -407,8 +409,6 @@ KERNEL_OS=`uname -o` ALSA_DRIVER_VERSION=`cat /proc/asound/version |head -n1|awk {'print $7'} |sed 's/\.$//'` get_alsa_library_version ALSA_UTILS_VERSION=`amixer -v |awk {'print $3'}` -VENDOR_ID=`lspci -vn | grep 040[1-3] | awk -F':' '{print $3}'| awk {'print substr($0, 2);}' >$TEMPDIR/vendor_id.tmp` -DEVICE_ID=`lspci -vn | grep 040[1-3] | awk -F':' '{print $4}'| awk {'print $1'} >$TEMPDIR/device_id.tmp` LAST_CARD=$((`grep "]: " /proc/asound/cards | wc -l` - 1 )) ESDINST=$(which esd 2>/dev/null| sed 's|^[^/]*||' 2>/dev/null) @@ -434,7 +434,9 @@ fi cat /proc/asound/modules 2>/dev/null|awk {'print $2'}>$TEMPDIR/alsamodules.tmp cat /proc/asound/cards >$TEMPDIR/alsacards.tmp +if [[ ! -z "$LSPCI" ]]; then lspci |grep -i "multi\|audio">$TEMPDIR/lspci.tmp +fi #Check for HDA-Intel cards codec#* cat /proc/asound/card*/codec\#* > $TEMPDIR/alsa-hda-intel.tmp 2> /dev/null @@ -547,6 +549,8 @@ echo "" >> $FILE cat $TEMPDIR/alsacards.tmp >> $FILE echo "" >> $FILE echo "" >> $FILE + +if [[ ! -z "$LSPCI" ]]; then echo "!!PCI Soundcards installed in the system" >> $FILE echo "!!--------------------------------------" >> $FILE echo "" >> $FILE @@ -559,6 +563,7 @@ echo "" >> $FILE lspci -vvn |grep -A1 040[1-3] >> $FILE echo "" >> $FILE echo "" >> $FILE +fi if [ "$SNDOPTIONS" ] then -- 1.7.9.5

2 3

[alsa-devel] Compiling pyalsa with the latest alsalib
by Proyecto MultiK 29 Aug '13

29 Aug '13

Hello, Together with a team, I am working on a multiseat project for which we need to access more than 32 sound cards on a Linux system (not all used simultaneously). We are running Ubuntu with a modified version of the 3.6.3 kernel. We used the code from Takashi Iwai's sound repository to patch the original 3.6.3 source to allow ALSA to recognize and name more than 32 devices (since there was a bug introduced in the 3.8 kernel which threw a kernel panic when connecting more than 28 cards, we could not use the latest kernel). Currently, we can detect more than 36 sound cards, they have ids and names given in /proc/asound/cards. Before applying Takashi's patches, they would stop being named at id = 31. So this seems to be working all right. The problem comes when trying to access the cards using pyalsa. We downloaded the pyalsa 1.0.26 tar from ftp://ftp.alsa-project.org/pub/pyalsa/pyalsa-1.0.26.tar.bz2. We need to know how to compile pyalsa with the latest alsalib that supports more than 32 cards. If we run pyalsa.alsacard.asoundlib_version() from the python interpreter, we get '1.0.25'. We know the latest version is 1.0.27 and we have put the alsa headers in pyalsa-1.0.26/pyalsa/alsa/ dir before compiling, but it doesn't seem to compile using these files and keeps saying version is 1.0.25. Guidance on this would be greatly appreciated. Thanks, Esteban Serrano MultiK team

2 1

[alsa-devel] [PATCH 1/3] ASoC: fsl_ssi: Remove unnecessary dev_set_drvdata()
by Fabio Estevam 29 Aug '13

29 Aug '13

Driver core clears the driver data to NULL after device_release or on probe failure, so just remove it from here. Signed-off-by: Fabio Estevam <fabio.estevam(a)freescale.com> --- sound/soc/fsl/fsl_ssi.c | 2 -- 1 file changed, 2 deletions(-) diff --git a/sound/soc/fsl/fsl_ssi.c b/sound/soc/fsl/fsl_ssi.c index 0c072ff..6f4a4c5 100644 --- a/sound/soc/fsl/fsl_ssi.c +++ b/sound/soc/fsl/fsl_ssi.c @@ -909,7 +909,6 @@ error_dai: snd_soc_unregister_component(&pdev->dev); error_dev: - dev_set_drvdata(&pdev->dev, NULL); device_remove_file(&pdev->dev, dev_attr); error_clk: @@ -935,7 +934,6 @@ static int fsl_ssi_remove(struct platform_device *pdev) snd_soc_unregister_component(&pdev->dev); device_remove_file(&pdev->dev, &ssi_private->dev_attr); irq_dispose_mapping(ssi_private->irq); - dev_set_drvdata(&pdev->dev, NULL); return 0; } -- 1.8.1.2

2 5

Re: [alsa-devel] [PATCH -next] ASoC: fsl_spdif: remove redundant dev_err call in fsl_spdif_probe()
by Mark Brown 29 Aug '13

29 Aug '13

On Thu, Aug 29, 2013 at 08:00:05AM +0800, Wei Yongjun wrote: > From: Wei Yongjun <yongjun_wei(a)trendmicro.com.cn> > > There is a error message within devm_ioremap_resource > already, so remove the dev_err call to avoid redundant > error message. Applied, thanks.

1 0

[alsa-devel] [RFC] ASoC: soc-pcm: Add support for unidirectional dai links
by Fabio Estevam 29 Aug '13

29 Aug '13

From: Fabio Estevam <fabio.estevam(a)freescale.com> On a mx28 board, running "aplay -l" and "arecord -l" results in the following: $ aplay -l **** List of PLAYBACK Hardware Devices **** card 0: mxssgtl5000 [mxs_sgtl5000], device 0: Playback sgtl5000-0 [] Subdevices: 1/1 Subdevice #0: subdevice #0 card 0: mxssgtl5000 [mxs_sgtl5000], device 1: Capture sgtl5000-1 [] Subdevices: 1/1 Subdevice #0: subdevice #0 $ arecord -l **** List of CAPTURE Hardware Devices **** card 0: mxssgtl5000 [mxs_sgtl5000], device 0: Playback sgtl5000-0 [] Subdevices: 1/1 Subdevice #0: subdevice #0 card 0: mxssgtl5000 [mxs_sgtl5000], device 1: Capture sgtl5000-1 [] Subdevices: 1/1 Subdevice #0: subdevice #0 ,which is not correct because we got a capture device listed in aplay and a playback device listed in arecord. On mx28 there are two serial audio interface ports (SAIF0 and SAIF1) and each one of them are unidirectional. Allow to specify a dai link as 'playback_only' or 'capture_only', which suits well for this case. After this change we can correctly report the capabilities as follows: $ aplay -l **** List of PLAYBACK Hardware Devices **** card 0: mxssgtl5000 [mxs_sgtl5000], device 0: HiFi Playback sgtl5000-0 [] Subdevices: 1/1 Subdevice #0: subdevice #0 $ arecord -l **** List of CAPTURE Hardware Devices **** card 0: mxssgtl5000 [mxs_sgtl5000], device 1: HiFi Capture sgtl5000-1 [] Subdevices: 1/1 Subdevice #0: subdevice #0 Also tested playback and capture on the mx28evk board. Signed-off-by: Fabio Estevam <fabio.estevam(a)freescale.com> --- After I get some feedback, I can split the patch in two parts (one for soc-pcm and another one for mxs-sgtl5000). include/sound/soc.h | 4 ++++ sound/soc/mxs/mxs-sgtl5000.c | 2 ++ sound/soc/soc-pcm.c | 10 ++++++++++ 3 files changed, 16 insertions(+) diff --git a/include/sound/soc.h b/include/sound/soc.h index 8e2ad52..774d3be 100644 --- a/include/sound/soc.h +++ b/include/sound/soc.h @@ -934,6 +934,10 @@ struct snd_soc_dai_link { /* machine stream operations */ const struct snd_soc_ops *ops; const struct snd_soc_compr_ops *compr_ops; + + /* For unidirectional dai links */ + bool playback_only; + bool capture_only; }; struct snd_soc_codec_conf { diff --git a/sound/soc/mxs/mxs-sgtl5000.c b/sound/soc/mxs/mxs-sgtl5000.c index ce084eb..4bb2737 100644 --- a/sound/soc/mxs/mxs-sgtl5000.c +++ b/sound/soc/mxs/mxs-sgtl5000.c @@ -105,11 +105,13 @@ static struct snd_soc_dai_link mxs_sgtl5000_dai[] = { .stream_name = "HiFi Playback", .codec_dai_name = "sgtl5000", .ops = &mxs_sgtl5000_hifi_ops, + .playback_only = true, }, { .name = "HiFi Rx", .stream_name = "HiFi Capture", .codec_dai_name = "sgtl5000", .ops = &mxs_sgtl5000_hifi_ops, + .capture_only = true, }, }; diff --git a/sound/soc/soc-pcm.c b/sound/soc/soc-pcm.c index fb70fbe..5e44863 100644 --- a/sound/soc/soc-pcm.c +++ b/sound/soc/soc-pcm.c @@ -2020,6 +2020,16 @@ int soc_new_pcm(struct snd_soc_pcm_runtime *rtd, int num) capture = 1; } + if (rtd->dai_link->playback_only) { + playback = 1; + capture = 0; + } + + if (rtd->dai_link->capture_only) { + playback = 0; + capture = 1; + } + /* create the PCM */ if (rtd->dai_link->no_pcm) { snprintf(new_name, sizeof(new_name), "(%s)", -- 1.8.1.2

3 3

[alsa-devel] [PATCH 1/5] ASoC: Add ADAU1X61 and ADAU1X81 CODECs common code
by Lars-Peter Clausen 29 Aug '13

29 Aug '13

The ADAU1X61 and ADAU1X81 are very similar in the digital domain, but are quite different in the analog domain. This patch adds support for the common parts of the ADAU1X61 and ADAU1X81 CODECs. Signed-off-by: Lars-Peter Clausen <lars(a)metafoo.de> --- include/linux/platform_data/adau17x1.h | 23 + sound/soc/codecs/Kconfig | 4 + sound/soc/codecs/Makefile | 2 + sound/soc/codecs/adau17x1.c | 895 +++++++++++++++++++++++++++++++++ sound/soc/codecs/adau17x1.h | 148 ++++++ 5 files changed, 1072 insertions(+) create mode 100644 include/linux/platform_data/adau17x1.h create mode 100644 sound/soc/codecs/adau17x1.c create mode 100644 sound/soc/codecs/adau17x1.h diff --git a/include/linux/platform_data/adau17x1.h b/include/linux/platform_data/adau17x1.h new file mode 100644 index 0000000..fdf419a --- /dev/null +++ b/include/linux/platform_data/adau17x1.h @@ -0,0 +1,23 @@ +/* + * Driver for ADAU1761/ADAU1461/ADAU1761/ADAU1961/ADAU1781/ADAU1781 codecs + * + * Copyright 2011-2013 Analog Devices Inc. + * Author: Lars-Peter Clausen <lars(a)metafoo.de> + * + * Licensed under the GPL-2 or later. + */ + +#ifndef __LINUX_PLATFORM_DATA_ADAU17X1_H__ +#define __LINUX_PLATFORM_DATA_ADAU17X1_H__ + +/** + * enum adau17x1_micbias_voltage - Microphone bias voltage + * @ADAU17X1_MICBIAS_0_90_AVDD: 0.9 * AVDD + * @ADAU17X1_MICBIAS_0_65_AVDD: 0.65 * AVDD + */ +enum adau17x1_micbias_voltage { + ADAU17X1_MICBIAS_0_90_AVDD = 0, + ADAU17X1_MICBIAS_0_65_AVDD = 1, +}; + +#endif diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig index 15106c0..56137d4 100644 --- a/sound/soc/codecs/Kconfig +++ b/sound/soc/codecs/Kconfig @@ -193,6 +193,10 @@ config SND_SOC_ADAU1701 config SND_SOC_ADAU1373 tristate +config SND_SOC_ADAU17X1 + select SND_SOC_SIGMADSP + tristate + config SND_SOC_ADAV80X tristate diff --git a/sound/soc/codecs/Makefile b/sound/soc/codecs/Makefile index bc12676..b4ecf6c 100644 --- a/sound/soc/codecs/Makefile +++ b/sound/soc/codecs/Makefile @@ -7,6 +7,7 @@ snd-soc-ad1980-objs := ad1980.o snd-soc-ad73311-objs := ad73311.o snd-soc-adau1701-objs := adau1701.o snd-soc-adau1373-objs := adau1373.o +snd-soc-adau17x1-objs := adau17x1.o snd-soc-adav80x-objs := adav80x.o snd-soc-ads117x-objs := ads117x.o snd-soc-ak4104-objs := ak4104.o @@ -138,6 +139,7 @@ obj-$(CONFIG_SND_SOC_AD1980) += snd-soc-ad1980.o obj-$(CONFIG_SND_SOC_AD73311) += snd-soc-ad73311.o obj-$(CONFIG_SND_SOC_ADAU1373) += snd-soc-adau1373.o obj-$(CONFIG_SND_SOC_ADAU1701) += snd-soc-adau1701.o +obj-$(CONFIG_SND_SOC_ADAU17X1) += snd-soc-adau17x1.o obj-$(CONFIG_SND_SOC_ADAV80X) += snd-soc-adav80x.o obj-$(CONFIG_SND_SOC_ADS117X) += snd-soc-ads117x.o obj-$(CONFIG_SND_SOC_AK4104) += snd-soc-ak4104.o diff --git a/sound/soc/codecs/adau17x1.c b/sound/soc/codecs/adau17x1.c new file mode 100644 index 0000000..b7d678d --- /dev/null +++ b/sound/soc/codecs/adau17x1.c @@ -0,0 +1,895 @@ +/* + * Common code for ADAU1X61 and ADAU1X81 codecs + * + * Copyright 2011-2013 Analog Devices Inc. + * Author: Lars-Peter Clausen <lars(a)metafoo.de> + * + * Licensed under the GPL-2 or later. + */ + +#include <linux/module.h> +#include <linux/init.h> +#include <linux/delay.h> +#include <linux/slab.h> +#include <sound/core.h> +#include <sound/pcm.h> +#include <sound/pcm_params.h> +#include <sound/soc.h> +#include <sound/tlv.h> +#include <linux/gcd.h> +#include <linux/i2c.h> +#include <linux/spi/spi.h> +#include <linux/regmap.h> + +#include "sigmadsp.h" +#include "adau17x1.h" + +static const char * const adau17x1_capture_mixer_boost_text[] = { + "Normal operation", "Boost Level 1", "Boost Level 2", "Boost Level 3", +}; + +static const SOC_ENUM_SINGLE_DECL(adau17x1_capture_boost_enum, + ADAU17X1_REC_POWER_MGMT, 5, adau17x1_capture_mixer_boost_text); + +static const char * const adau17x1_mic_bias_mode_text[] = { + "Normal operation", "High performance", +}; + +static SOC_ENUM_SINGLE_DECL(adau17x1_mic_bias_mode_enum, + ADAU17X1_MICBIAS, 3, adau17x1_mic_bias_mode_text); + +static const char * const adau17x1_mono_stereo_text[] = { + "Stereo", + "Mono Left Channel (L+R)", + "Mono Right Channel (L+R)", + "Mono (L+R)", +}; + +static const SOC_ENUM_SINGLE_DECL(adau17x1_dac_mode_enum, + ADAU17X1_DAC_CONTROL0, 6, adau17x1_mono_stereo_text); + +static const DECLARE_TLV_DB_MINMAX(adau17x1_digital_tlv, -9563, 0); + +static const struct snd_kcontrol_new adau17x1_controls[] = { + SOC_DOUBLE_R_TLV("Digital Capture Volume", + ADAU17X1_LEFT_INPUT_DIGITAL_VOL, + ADAU17X1_RIGHT_INPUT_DIGITAL_VOL, + 0, 0xff, 1, adau17x1_digital_tlv), + SOC_DOUBLE_R_TLV("Digital Playback Volume", ADAU17X1_DAC_CONTROL1, + ADAU17X1_DAC_CONTROL2, 0, 0xff, 1, adau17x1_digital_tlv), + + SOC_SINGLE("ADC High Pass Filter Switch", ADAU17X1_ADC_CONTROL, + 5, 1, 0), + SOC_SINGLE("Playback De-emphasis Switch", ADAU17X1_DAC_CONTROL0, + 2, 1, 0), + + SOC_ENUM("Capture Boost", adau17x1_capture_boost_enum), + + SOC_ENUM("Mic Bias Mode", adau17x1_mic_bias_mode_enum), + + SOC_ENUM("DAC Mono Stereo", adau17x1_dac_mode_enum), +}; + +static int adau17x1_pll_event(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *kcontrol, int event) +{ + struct adau *adau = snd_soc_codec_get_drvdata(w->codec); + int ret; + + if (SND_SOC_DAPM_EVENT_ON(event)) { + adau->pll_regs[5] = 1; + } else { + adau->pll_regs[5] = 0; + /* Bypass the PLL when disabled, otherwise registers will become + * inaccessible. */ + regmap_update_bits(adau->regmap, ADAU17X1_CLOCK_CONTROL, + ADAU17X1_CLOCK_CONTROL_CORECLK_SRC_PLL, 0); + } + + /* The PLL register is 6 bytes long and can only be written at once. */ + ret = regmap_raw_write(adau->regmap, ADAU17X1_PLL_CONTROL, + adau->pll_regs, ARRAY_SIZE(adau->pll_regs)); + + if (SND_SOC_DAPM_EVENT_ON(event)) { + mdelay(5); + regmap_update_bits(adau->regmap, ADAU17X1_CLOCK_CONTROL, + ADAU17X1_CLOCK_CONTROL_CORECLK_SRC_PLL, + ADAU17X1_CLOCK_CONTROL_CORECLK_SRC_PLL); + } + + return 0; +} + +static const struct snd_soc_dapm_widget adau17x1_dapm_widgets[] = { + SND_SOC_DAPM_SUPPLY_S("PLL", 3, SND_SOC_NOPM, 0, 0, adau17x1_pll_event, + SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMD), + + SND_SOC_DAPM_SUPPLY("AIFCLK", SND_SOC_NOPM, 0, 0, NULL, 0), + + SND_SOC_DAPM_SUPPLY("MICBIAS", ADAU17X1_MICBIAS, 0, 0, NULL, 0), + + SND_SOC_DAPM_SUPPLY("Left Playback Enable", ADAU17X1_PLAY_POWER_MGMT, + 0, 0, NULL, 0), + SND_SOC_DAPM_SUPPLY("Right Playback Enable", ADAU17X1_PLAY_POWER_MGMT, + 1, 0, NULL, 0), + + SND_SOC_DAPM_SUPPLY("Left Dec Filter Enable", ADAU17X1_ADC_CONTROL, + 0, 0, NULL, 0), + SND_SOC_DAPM_SUPPLY("Right Dec Filter Enable", ADAU17X1_ADC_CONTROL, + 1, 0, NULL, 0), + + SND_SOC_DAPM_DAC("Left DAC", NULL, ADAU17X1_DAC_CONTROL0, 0, 0), + SND_SOC_DAPM_DAC("Right DAC", NULL, ADAU17X1_DAC_CONTROL0, 1, 0), + SND_SOC_DAPM_ADC("Left ADC", NULL, SND_SOC_NOPM, 0, 0), + SND_SOC_DAPM_ADC("Right ADC", NULL, SND_SOC_NOPM, 0, 0), + + SND_SOC_DAPM_AIF_OUT("AIFOUT", "Capture", 0, SND_SOC_NOPM, 0, 0), + SND_SOC_DAPM_AIF_IN("AIFIN", "Playback", 0, SND_SOC_NOPM, 0, 0), +}; + +static const struct snd_soc_dapm_route adau17x1_dapm_routes[] = { + { "Left ADC", NULL, "SYSCLK" }, + { "Right ADC", NULL, "SYSCLK" }, + { "Left DAC", NULL, "SYSCLK" }, + { "Right DAC", NULL, "SYSCLK" }, + { "AIFOUT", NULL, "SYSCLK" }, + { "AIFIN", NULL, "SYSCLK" }, + + { "Left ADC", NULL, "Left Dec Filter Enable" }, + { "Right ADC", NULL, "Right Dec Filter Enable" }, + + { "AIFOUT", NULL, "AIFCLK" }, + { "AIFIN", NULL, "AIFCLK" }, +}; + +static const struct snd_soc_dapm_route adau17x1_dapm_pll_route = { + "SYSCLK", NULL, "PLL", +}; + +static const char * const adau17x1_dac_mux_text[] = { + "AIFIN", + "DSP", +}; + +int adau17x1_dsp_mux_enum_put(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_codec *codec = snd_soc_dapm_kcontrol_codec(kcontrol); + struct adau *adau = snd_soc_codec_get_drvdata(codec); + struct soc_enum *e = (struct soc_enum *)kcontrol->private_value; + struct snd_soc_dapm_update update; + unsigned int val, change; + + if (ucontrol->value.enumerated.item[0] >= e->max) + return -EINVAL; + + switch (ucontrol->value.enumerated.item[0]) { + case 0: + switch (e->reg) { + case ADAU17X1_SERIAL_INPUT_ROUTE: + val = (adau->tdm_dac_slot * 2) + 1; + adau->dsp_playback_bypass = true; + break; + case ADAU17X1_SERIAL_OUTPUT_ROUTE: + val = (adau->tdm_adc_slot * 2) + 1; + adau->dsp_capture_bypass = true; + break; + default: + return -EINVAL; + } + break; + default: + val = 0; + + switch (e->reg) { + case ADAU17X1_SERIAL_INPUT_ROUTE: + adau->dsp_playback_bypass = false; + break; + case ADAU17X1_SERIAL_OUTPUT_ROUTE: + adau->dsp_capture_bypass = false; + break; + default: + return -EINVAL; + } + break; + } + + change = snd_soc_test_bits(codec, e->reg, 0xff, val); + if (change) { + update.kcontrol = kcontrol; + update.reg = e->reg; + update.mask = 0xff; + update.val = val; + + snd_soc_dapm_mux_update_power(&codec->dapm, kcontrol, + ucontrol->value.enumerated.item[0], e, &update); + } + + return change; +} +EXPORT_SYMBOL_GPL(adau17x1_dsp_mux_enum_put); + +int adau17x1_dsp_mux_enum_get(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_codec *codec = snd_soc_dapm_kcontrol_codec(kcontrol); + struct soc_enum *e = (struct soc_enum *)kcontrol->private_value; + unsigned int val; + + val = snd_soc_read(codec, e->reg); + ucontrol->value.enumerated.item[0] = val == 0; + + return 0; +} +EXPORT_SYMBOL_GPL(adau17x1_dsp_mux_enum_get); + +static const SOC_ENUM_SINGLE_DECL(adau17x1_dac_mux_enum, + ADAU17X1_SERIAL_INPUT_ROUTE, 0, + adau17x1_dac_mux_text); + +static const struct snd_kcontrol_new adau17x1_dac_mux = + ADAU17X1_DSP_MUX_ENUM("DAC Playback Mux", adau17x1_dac_mux_enum); + +static const struct snd_soc_dapm_widget adau17x1_dsp_dapm_widgets[] = { + SND_SOC_DAPM_PGA("DSP", ADAU17X1_DSP_RUN, 0, 0, NULL, 0), + SND_SOC_DAPM_SIGGEN("DSP Siggen"), + + SND_SOC_DAPM_VIRT_MUX("DAC Playback Mux", SND_SOC_NOPM, 0, 0, + &adau17x1_dac_mux), + +}; + +static const struct snd_soc_dapm_route adau17x1_dsp_dapm_routes[] = { + { "DAC Playback Mux", "DSP", "DSP" }, + { "DAC Playback Mux", "AIFIN", "AIFIN" }, + + { "Left DAC", NULL, "DAC Playback Mux" }, + { "Right DAC", NULL, "DAC Playback Mux" }, + + { "AIFOUT Capture Mux", "DSP", "DSP" }, + + { "AIFOUT", NULL, "AIFOUT Capture Mux" }, + + { "DSP", NULL, "DSP Siggen" }, +}; + +static const struct snd_soc_dapm_route adau17x1_no_dsp_dapm_routes[] = { + { "Left DAC", NULL, "AIFIN" }, + { "Right DAC", NULL, "AIFIN" }, +}; + +static void adau17x1_check_aifclk(struct snd_soc_codec *codec) +{ + struct adau *adau = snd_soc_codec_get_drvdata(codec); + + /* If we are in master mode we need to generate bit- and frameclock, + * regardless of whether there is an active path or not */ + if (codec->active && adau->master) + snd_soc_dapm_force_enable_pin(&codec->dapm, "AIFCLK"); + else + snd_soc_dapm_disable_pin(&codec->dapm, "AIFCLK"); + snd_soc_dapm_sync(&codec->dapm); +} + +bool adau17x1_has_dsp(struct adau *adau) +{ + switch (adau->type) { + case ADAU1761: + case ADAU1381: + case ADAU1781: + return true; + default: + return false; + } +} +EXPORT_SYMBOL_GPL(adau17x1_has_dsp); + +static int adau17x1_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params, struct snd_soc_dai *dai) +{ + struct snd_soc_codec *codec = dai->codec; + struct adau *adau = snd_soc_codec_get_drvdata(codec); + unsigned int val, div, dsp_div; + unsigned int freq; + + if (adau->clk_src == ADAU17X1_CLK_SRC_PLL) + freq = adau->pll_freq; + else + freq = adau->sysclk / adau->sysclk_div; + + if (freq % params_rate(params) != 0) + return -EINVAL; + + switch (freq / params_rate(params)) { + case 1024: /* fs */ + div = 0; + dsp_div = 1; + break; + case 6144: /* fs / 6 */ + div = 1; + dsp_div = 6; + break; + case 4096: /* fs / 4 */ + div = 2; + dsp_div = 5; + break; + case 3072: /* fs / 3 */ + div = 3; + dsp_div = 4; + break; + case 2048: /* fs / 2 */ + div = 4; + dsp_div = 3; + break; + case 1536: /* fs / 1.5 */ + div = 5; + dsp_div = 2; + break; + case 512: /* fs / 0.5 */ + div = 6; + dsp_div = 0; + break; + default: + return -EINVAL; + } + + regmap_update_bits(adau->regmap, ADAU17X1_CONVERTER0, 7, div); + if (adau17x1_has_dsp(adau)) { + regmap_write(adau->regmap, ADAU17X1_SERIAL_SAMPLING_RATE, div); + regmap_write(adau->regmap, ADAU17X1_DSP_SAMPLING_RATE, dsp_div); + } + + adau17x1_check_aifclk(codec); + + if (adau->dai_fmt != SND_SOC_DAIFMT_RIGHT_J) + return 0; + + switch (params_format(params)) { + case SNDRV_PCM_FORMAT_S16_LE: + val = ADAU17X1_SERIAL_PORT1_DELAY16; + break; + case SNDRV_PCM_FORMAT_S24_LE: + val = ADAU17X1_SERIAL_PORT1_DELAY8; + break; + case SNDRV_PCM_FORMAT_S32_LE: + val = ADAU17X1_SERIAL_PORT1_DELAY0; + break; + default: + return -EINVAL; + } + + return regmap_update_bits(adau->regmap, ADAU17X1_SERIAL_PORT1, + ADAU17X1_SERIAL_PORT1_DELAY_MASK, val); +} + +static int adau17x1_dai_startup(struct snd_pcm_substream *substream, + struct snd_soc_dai *dai) +{ + adau17x1_check_aifclk(dai->codec); + + return 0; +} + +static void adau17x1_dai_shutdown(struct snd_pcm_substream *substream, + struct snd_soc_dai *dai) +{ + adau17x1_check_aifclk(dai->codec); +} + +static int adau17x1_set_dai_pll(struct snd_soc_dai *dai, int pll_id, + int source, unsigned int freq_in, unsigned int freq_out) +{ + struct snd_soc_codec *codec = dai->codec; + struct adau *adau = snd_soc_codec_get_drvdata(codec); + unsigned int div; + unsigned int r, n, m, i, j; + + if (freq_in < 8000000 || freq_in > 27000000) + return -EINVAL; + + if (!freq_out) { + r = 0; + n = 0; + m = 0; + div = 0; + } else { + if (freq_out % freq_in != 0) { + div = DIV_ROUND_UP(freq_in, 13500000); + freq_in /= div; + r = freq_out / freq_in; + i = freq_out % freq_in; + j = gcd(i, freq_in); + n = i / j; + m = freq_in / j; + div--; + } else { + r = freq_out / freq_in; + n = 0; + m = 0; + div = 0; + } + if (n > 0xffff || m > 0xffff || div > 3 || r > 8 || r < 2) + return -EINVAL; + } + + adau->pll_regs[0] = m >> 8; + adau->pll_regs[1] = m & 0xff; + adau->pll_regs[2] = n >> 8; + adau->pll_regs[3] = n & 0xff; + adau->pll_regs[4] = (r << 3) | (div << 1); + if (m != 0) + adau->pll_regs[4] |= 1; /* Fractional mode */ + adau->pll_regs[5] = 0; + + adau->pll_freq = freq_out; + + return 0; +} + +static int adau17x1_set_dai_sysclk(struct snd_soc_dai *dai, + int clk_id, unsigned int freq, int dir) +{ + struct adau *adau = snd_soc_codec_get_drvdata(dai->codec); + struct snd_soc_dapm_context *dapm = &dai->codec->dapm; + + switch (clk_id) { + case ADAU17X1_CLK_SRC_MCLK: + case ADAU17X1_CLK_SRC_PLL: + break; + default: + return -EINVAL; + } + + adau->sysclk = freq; + + if (adau->clk_src != clk_id) { + if (clk_id == ADAU17X1_CLK_SRC_PLL) { + snd_soc_dapm_add_routes(dapm, + &adau17x1_dapm_pll_route, 1); + } else { + snd_soc_dapm_del_routes(dapm, + &adau17x1_dapm_pll_route, 1); + } + } + + adau->clk_src = clk_id; + + return 0; +} + +static int adau17x1_set_dai_clkdiv(struct snd_soc_dai *dai, int div_id, int div) +{ + struct adau *adau = snd_soc_codec_get_drvdata(dai->codec); + + if (div < 0 || div > 4) + return -EINVAL; + + adau->sysclk_div = div; + + return regmap_update_bits(adau->regmap, ADAU17X1_CLOCK_CONTROL, + ADAU17X1_CLOCK_CONTROL_INFREQ_MASK, (div - 1) << 1); +} + +static int adau17x1_set_dai_fmt(struct snd_soc_dai *dai, + unsigned int fmt) +{ + struct adau *adau = snd_soc_codec_get_drvdata(dai->codec); + unsigned int ctrl0, ctrl1; + int lrclk_pol; + + switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) { + case SND_SOC_DAIFMT_CBM_CFM: + ctrl0 = ADAU17X1_SERIAL_PORT0_MASTER; + adau->master = true; + break; + case SND_SOC_DAIFMT_CBS_CFS: + ctrl0 = 0; + adau->master = false; + break; + default: + return -EINVAL; + } + + switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) { + case SND_SOC_DAIFMT_I2S: + lrclk_pol = 0; + ctrl1 = ADAU17X1_SERIAL_PORT1_DELAY1; + break; + case SND_SOC_DAIFMT_LEFT_J: + case SND_SOC_DAIFMT_RIGHT_J: + lrclk_pol = 1; + ctrl1 = ADAU17X1_SERIAL_PORT1_DELAY0; + break; + case SND_SOC_DAIFMT_DSP_A: + lrclk_pol = 1; + ctrl0 |= ADAU17X1_SERIAL_PORT0_PULSE_MODE; + ctrl1 = ADAU17X1_SERIAL_PORT1_DELAY1; + break; + case SND_SOC_DAIFMT_DSP_B: + lrclk_pol = 1; + ctrl0 |= ADAU17X1_SERIAL_PORT0_PULSE_MODE; + ctrl1 = ADAU17X1_SERIAL_PORT1_DELAY0; + break; + default: + return -EINVAL; + } + + switch (fmt & SND_SOC_DAIFMT_INV_MASK) { + case SND_SOC_DAIFMT_NB_NF: + break; + case SND_SOC_DAIFMT_IB_NF: + ctrl0 |= ADAU17X1_SERIAL_PORT0_BCLK_POL; + break; + case SND_SOC_DAIFMT_NB_IF: + lrclk_pol = !lrclk_pol; + break; + case SND_SOC_DAIFMT_IB_IF: + ctrl0 |= ADAU17X1_SERIAL_PORT0_BCLK_POL; + lrclk_pol = !lrclk_pol; + break; + default: + return -EINVAL; + } + + if (lrclk_pol) + ctrl0 |= ADAU17X1_SERIAL_PORT0_LRCLK_POL; + + regmap_write(adau->regmap, ADAU17X1_SERIAL_PORT0, ctrl0); + regmap_write(adau->regmap, ADAU17X1_SERIAL_PORT1, ctrl1); + + adau->dai_fmt = fmt & SND_SOC_DAIFMT_FORMAT_MASK; + + return 0; +} + +static int adau17x1_set_dai_tdm_slot(struct snd_soc_dai *dai, + unsigned int tx_mask, unsigned int rx_mask, int slots, int slot_width) +{ + struct adau *adau = snd_soc_codec_get_drvdata(dai->codec); + unsigned int ser_ctrl0, ser_ctrl1; + unsigned int conv_ctrl0, conv_ctrl1; + + /* I2S mode */ + if (slots == 0) { + slots = 2; + rx_mask = 3; + tx_mask = 3; + slot_width = 32; + } + + switch (slots) { + case 2: + ser_ctrl0 = ADUA_SERIAL_PORT0_STEREO; + break; + case 4: + ser_ctrl0 = ADUA_SERIAL_PORT0_TDM4; + break; + case 8: + if (adau->type == ADAU1361) + return -EINVAL; + + ser_ctrl0 = ADUA_SERIAL_PORT0_TDM8; + break; + default: + return -EINVAL; + } + + switch (slot_width * slots) { + case 32: + if (adau->type == ADAU1761) + return -EINVAL; + + ser_ctrl1 = ADUA_SERIAL_PORT1_BCLK32; + break; + case 64: + ser_ctrl1 = ADUA_SERIAL_PORT1_BCLK64; + break; + case 48: + ser_ctrl1 = ADUA_SERIAL_PORT1_BCLK48; + break; + case 128: + ser_ctrl1 = ADUA_SERIAL_PORT1_BCLK128; + break; + case 256: + if (adau->type == ADAU1361) + return -EINVAL; + + ser_ctrl1 = ADUA_SERIAL_PORT1_BCLK256; + break; + default: + return -EINVAL; + } + + switch (rx_mask) { + case 0x03: + conv_ctrl1 = ADAU17X1_CONVERTER1_ADC_PAIR(1); + adau->tdm_adc_slot = 0; + break; + case 0x0c: + conv_ctrl1 = ADAU17X1_CONVERTER1_ADC_PAIR(2); + adau->tdm_adc_slot = 1; + break; + case 0x30: + conv_ctrl1 = ADAU17X1_CONVERTER1_ADC_PAIR(3); + adau->tdm_adc_slot = 2; + break; + case 0xc0: + conv_ctrl1 = ADAU17X1_CONVERTER1_ADC_PAIR(4); + adau->tdm_adc_slot = 3; + break; + default: + return -EINVAL; + } + + switch (tx_mask) { + case 0x03: + conv_ctrl0 = ADAU17X1_CONVERTER0_DAC_PAIR(1); + adau->tdm_dac_slot = 0; + break; + case 0x0c: + conv_ctrl0 = ADAU17X1_CONVERTER0_DAC_PAIR(2); + adau->tdm_dac_slot = 1; + break; + case 0x30: + conv_ctrl0 = ADAU17X1_CONVERTER0_DAC_PAIR(3); + adau->tdm_dac_slot = 2; + break; + case 0xc0: + conv_ctrl0 = ADAU17X1_CONVERTER0_DAC_PAIR(4); + adau->tdm_dac_slot = 3; + break; + default: + return -EINVAL; + } + + regmap_update_bits(adau->regmap, ADAU17X1_CONVERTER0, + ADAU17X1_CONVERTER0_DAC_PAIR_MASK, conv_ctrl0); + regmap_update_bits(adau->regmap, ADAU17X1_CONVERTER1, + ADAU17X1_CONVERTER1_ADC_PAIR_MASK, conv_ctrl1); + regmap_update_bits(adau->regmap, ADAU17X1_SERIAL_PORT0, + ADAU17X1_SERIAL_PORT0_TDM_MASK, ser_ctrl0); + regmap_update_bits(adau->regmap, ADAU17X1_SERIAL_PORT1, + ADAU17X1_SERIAL_PORT1_BCLK_MASK, ser_ctrl1); + + if (adau17x1_has_dsp(adau)) { + if (adau->dsp_playback_bypass) { + regmap_write(adau->regmap, ADAU17X1_SERIAL_INPUT_ROUTE, + (adau->tdm_dac_slot * 2) + 1); + } + if (adau->dsp_capture_bypass) { + regmap_write(adau->regmap, ADAU17X1_SERIAL_OUTPUT_ROUTE, + (adau->tdm_adc_slot * 2) + 1); + } + } + + return 0; +} + +const struct snd_soc_dai_ops adau17x1_dai_ops = { + .hw_params = adau17x1_hw_params, + .set_sysclk = adau17x1_set_dai_sysclk, + .set_fmt = adau17x1_set_dai_fmt, + .set_pll = adau17x1_set_dai_pll, + .set_clkdiv = adau17x1_set_dai_clkdiv, + .set_tdm_slot = adau17x1_set_dai_tdm_slot, + .startup = adau17x1_dai_startup, + .shutdown = adau17x1_dai_shutdown, +}; +EXPORT_SYMBOL_GPL(adau17x1_dai_ops); + +int adau17x1_set_micbias_voltage(struct snd_soc_codec *codec, + enum adau17x1_micbias_voltage micbias) +{ + struct adau *adau = snd_soc_codec_get_drvdata(codec); + + switch (micbias) { + case ADAU17X1_MICBIAS_0_90_AVDD: + case ADAU17X1_MICBIAS_0_65_AVDD: + break; + default: + return -EINVAL; + } + + regmap_write(adau->regmap, ADAU17X1_MICBIAS, micbias << 2); + + return 0; +} +EXPORT_SYMBOL_GPL(adau17x1_set_micbias_voltage); + +bool adau17x1_readable_register(struct device *dev, unsigned int reg) +{ + switch (reg) { + case ADAU17X1_CLOCK_CONTROL: + case ADAU17X1_PLL_CONTROL: + case ADAU17X1_REC_POWER_MGMT: + case ADAU17X1_MICBIAS: + case ADAU17X1_SERIAL_PORT0: + case ADAU17X1_SERIAL_PORT1: + case ADAU17X1_CONVERTER0: + case ADAU17X1_CONVERTER1: + case ADAU17X1_LEFT_INPUT_DIGITAL_VOL: + case ADAU17X1_RIGHT_INPUT_DIGITAL_VOL: + case ADAU17X1_ADC_CONTROL: + case ADAU17X1_PLAY_POWER_MGMT: + case ADAU17X1_DAC_CONTROL0: + case ADAU17X1_DAC_CONTROL1: + case ADAU17X1_DAC_CONTROL2: + case ADAU17X1_SERIAL_PORT_PAD: + case ADAU17X1_CONTROL_PORT_PAD0: + case ADAU17X1_CONTROL_PORT_PAD1: + case ADAU17X1_DSP_SAMPLING_RATE: + case ADAU17X1_SERIAL_INPUT_ROUTE: + case ADAU17X1_SERIAL_OUTPUT_ROUTE: + case ADAU17X1_DSP_ENABLE: + case ADAU17X1_DSP_RUN: + case ADAU17X1_SERIAL_SAMPLING_RATE: + return true; + default: + break; + } + return false; +} +EXPORT_SYMBOL_GPL(adau17x1_readable_register); + +bool adau17x1_volatile_register(struct device *dev, unsigned int reg) +{ + /* SigmaDSP parameter and program memory */ + if (reg < 0x4000) + return true; + + switch (reg) { + /* The PLL register is 6 bytes long */ + case ADAU17X1_PLL_CONTROL: + case ADAU17X1_PLL_CONTROL + 1: + case ADAU17X1_PLL_CONTROL + 2: + case ADAU17X1_PLL_CONTROL + 3: + case ADAU17X1_PLL_CONTROL + 4: + case ADAU17X1_PLL_CONTROL + 5: + return true; + default: + break; + } + + return false; +} +EXPORT_SYMBOL_GPL(adau17x1_volatile_register); + +int adau17x1_load_firmware(struct adau *adau, struct device *dev, + const char *firmware) +{ + int ret; + int dspsr; + + ret = regmap_read(adau->regmap, ADAU17X1_DSP_SAMPLING_RATE, &dspsr); + if (ret) + return ret; + + regmap_write(adau->regmap, ADAU17X1_DSP_ENABLE, 1); + regmap_write(adau->regmap, ADAU17X1_DSP_SAMPLING_RATE, 0xf); + + ret = process_sigma_firmware_regmap(dev, adau->regmap, firmware); + if (ret) { + regmap_write(adau->regmap, ADAU17X1_DSP_ENABLE, 0); + return ret; + } + regmap_write(adau->regmap, ADAU17X1_DSP_SAMPLING_RATE, dspsr); + + return 0; +} +EXPORT_SYMBOL_GPL(adau17x1_load_firmware); + +int adau17x1_probe(struct snd_soc_codec *codec) +{ + struct adau *adau = snd_soc_codec_get_drvdata(codec); + int ret; + + ret = snd_soc_codec_set_cache_io(codec, 0, 0, SND_SOC_REGMAP); + if (ret) + return ret; + + codec->driver->set_bias_level(codec, SND_SOC_BIAS_STANDBY); + + ret = snd_soc_add_codec_controls(codec, adau17x1_controls, + ARRAY_SIZE(adau17x1_controls)); + if (ret) + return ret; + ret = snd_soc_dapm_new_controls(&codec->dapm, adau17x1_dapm_widgets, + ARRAY_SIZE(adau17x1_dapm_widgets)); + if (ret) + return ret; + + if (adau17x1_has_dsp(adau)) { + ret = snd_soc_dapm_new_controls(&codec->dapm, + adau17x1_dsp_dapm_widgets, + ARRAY_SIZE(adau17x1_dsp_dapm_widgets)); + } + return ret; +} +EXPORT_SYMBOL_GPL(adau17x1_probe); + +int adau17x1_add_routes(struct snd_soc_codec *codec) +{ + struct adau *adau = snd_soc_codec_get_drvdata(codec); + int ret; + + ret = snd_soc_dapm_add_routes(&codec->dapm, adau17x1_dapm_routes, + ARRAY_SIZE(adau17x1_dapm_routes)); + if (ret) + return ret; + + if (adau17x1_has_dsp(adau)) { + ret = snd_soc_dapm_add_routes(&codec->dapm, + adau17x1_dsp_dapm_routes, + ARRAY_SIZE(adau17x1_dsp_dapm_routes)); + } else { + ret = snd_soc_dapm_add_routes(&codec->dapm, + adau17x1_no_dsp_dapm_routes, + ARRAY_SIZE(adau17x1_no_dsp_dapm_routes)); + } + return ret; +} +EXPORT_SYMBOL_GPL(adau17x1_add_routes); + +#if IS_ENABLED(CONFIG_SPI_MASTER) +static void adau17x1_spi_mode(struct device *dev) +{ + /* To get the device into SPI mode CLATCH has to be pulled low three + * times. Do this by issuing three dummy reads. */ + spi_w8r8(to_spi_device(dev), 0x00); + spi_w8r8(to_spi_device(dev), 0x00); + spi_w8r8(to_spi_device(dev), 0x00); +} +#else +static inline void adau17x1_spi_mode(struct device *dev) {} +#endif + +int adau17x1_suspend(struct snd_soc_codec *codec) +{ + codec->driver->set_bias_level(codec, SND_SOC_BIAS_OFF); + return 0; +} +EXPORT_SYMBOL_GPL(adau17x1_suspend); + +int adau17x1_resume(struct snd_soc_codec *codec) +{ + struct adau *adau = snd_soc_codec_get_drvdata(codec); + + if (adau->spi_mode) + adau17x1_spi_mode(codec->dev); + + codec->driver->set_bias_level(codec, SND_SOC_BIAS_STANDBY); + regcache_sync(adau->regmap); + + return 0; +} +EXPORT_SYMBOL_GPL(adau17x1_resume); + +int adau17x1_bus_probe(struct device *dev, struct regmap *regmap, + enum adau17x1_type type, bool spi_mode) +{ + struct adau *adau; + + if (IS_ERR(regmap)) + return PTR_ERR(regmap); + + adau = devm_kzalloc(dev, sizeof(*adau), GFP_KERNEL); + if (!adau) + return -ENOMEM; + + adau->regmap = regmap; + adau->spi_mode = spi_mode; + adau->type = type; + adau->sysclk_div = 1; + + dev_set_drvdata(dev, adau); + + if (spi_mode) + adau17x1_spi_mode(dev); + + return 0; +} +EXPORT_SYMBOL_GPL(adau17x1_bus_probe); + +MODULE_DESCRIPTION("ASoC ADAU1X61/ADAU1X81 common code"); +MODULE_AUTHOR("Lars-Peter Clausen <lars(a)metafoo.de>"); +MODULE_LICENSE("GPL"); diff --git a/sound/soc/codecs/adau17x1.h b/sound/soc/codecs/adau17x1.h new file mode 100644 index 0000000..d49442e --- /dev/null +++ b/sound/soc/codecs/adau17x1.h @@ -0,0 +1,148 @@ +#ifndef __ADAU17X1_H__ +#define __ADAU17X1_H__ + +#include <linux/regmap.h> +#include <linux/platform_data/adau17x1.h> + +enum adau17x1_type { + ADAU1361, + ADAU1761, + ADAU1381, + ADAU1781, +}; + +enum adau17x1_pll { + ADAU17X1_PLL, +}; + +enum adau17x1_pll_src { + ADAU17X1_PLL_SRC_MCLK, +}; + +enum adau17x1_clk_src { + ADAU17X1_CLK_SRC_MCLK, + ADAU17X1_CLK_SRC_PLL, +}; + +struct adau { + unsigned int sysclk; + unsigned int sysclk_div; + unsigned int pll_freq; + + enum adau17x1_clk_src clk_src; + enum adau17x1_type type; + bool spi_mode; + + unsigned int dai_fmt; + + uint8_t pll_regs[6]; + + bool master; + + unsigned int tdm_dac_slot; + unsigned int tdm_adc_slot; + + bool dsp_playback_bypass; + bool dsp_capture_bypass; + + struct regmap *regmap; +}; + +int adau17x1_probe(struct snd_soc_codec *codec); +int adau17x1_add_routes(struct snd_soc_codec *codec); +int adau17x1_bus_probe(struct device *dev, + struct regmap *regmap, enum adau17x1_type type, + bool spi_mode); +int adau17x1_set_micbias_voltage(struct snd_soc_codec *codec, + enum adau17x1_micbias_voltage micbias); +bool adau17x1_readable_register(struct device *dev, unsigned int reg); +bool adau17x1_volatile_register(struct device *dev, unsigned int reg); +int adau17x1_suspend(struct snd_soc_codec *codec); +int adau17x1_resume(struct snd_soc_codec *codec); + +extern const struct snd_soc_dai_ops adau17x1_dai_ops; + +int adau17x1_load_firmware(struct adau *adau, struct device *dev, + const char *firmware); +bool adau17x1_has_dsp(struct adau *adau); +int adau17x1_dsp_mux_enum_put(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol); +int adau17x1_dsp_mux_enum_get(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol); + +#define ADAU17X1_DSP_MUX_ENUM(xname, xenum) \ + SOC_DAPM_ENUM_EXT(xname, xenum, \ + adau17x1_dsp_mux_enum_get, adau17x1_dsp_mux_enum_put) + +#define DECLARE_ADAU17X1_AIFOUT_MUX(name, text) \ +static const char * const adau17x1_ ## name ## _aifout_text[] = { \ + text, \ + "DSP", \ +}; \ +static const SOC_ENUM_SINGLE_DECL(adau17x1_ ## name ## _aifout_mux_enum, \ + ADAU17X1_SERIAL_OUTPUT_ROUTE, 0, adau17x1_ ## name ## _aifout_text); \ +static const struct snd_kcontrol_new adau17x1_ ## name ## _aifout_mux = \ + ADAU17X1_DSP_MUX_ENUM("AIFOUT Capture Mux", \ + adau17x1_ ## name ## _aifout_mux_enum); \ +static const struct snd_soc_dapm_widget adau17x1_ ## name ## _dsp_widget = \ + SND_SOC_DAPM_VIRT_MUX("AIFOUT Capture Mux", SND_SOC_NOPM, 0, 0, \ + &adau17x1_ ## name ## _aifout_mux) \ + +#define ADAU17X1_CLOCK_CONTROL 0x4000 +#define ADAU17X1_PLL_CONTROL 0x4002 +#define ADAU17X1_REC_POWER_MGMT 0x4009 +#define ADAU17X1_MICBIAS 0x4010 +#define ADAU17X1_SERIAL_PORT0 0x4015 +#define ADAU17X1_SERIAL_PORT1 0x4016 +#define ADAU17X1_CONVERTER0 0x4017 +#define ADAU17X1_CONVERTER1 0x4018 +#define ADAU17X1_LEFT_INPUT_DIGITAL_VOL 0x401a +#define ADAU17X1_RIGHT_INPUT_DIGITAL_VOL 0x401b +#define ADAU17X1_ADC_CONTROL 0x4019 +#define ADAU17X1_PLAY_POWER_MGMT 0x4029 +#define ADAU17X1_DAC_CONTROL0 0x402a +#define ADAU17X1_DAC_CONTROL1 0x402b +#define ADAU17X1_DAC_CONTROL2 0x402c +#define ADAU17X1_SERIAL_PORT_PAD 0x402d +#define ADAU17X1_CONTROL_PORT_PAD0 0x402f +#define ADAU17X1_CONTROL_PORT_PAD1 0x4030 +#define ADAU17X1_DSP_SAMPLING_RATE 0x40eb +#define ADAU17X1_SERIAL_INPUT_ROUTE 0x40f2 +#define ADAU17X1_SERIAL_OUTPUT_ROUTE 0x40f3 +#define ADAU17X1_DSP_ENABLE 0x40f5 +#define ADAU17X1_DSP_RUN 0x40f6 +#define ADAU17X1_SERIAL_SAMPLING_RATE 0x40f8 + +#define ADAU17X1_SERIAL_PORT0_BCLK_POL BIT(4) +#define ADAU17X1_SERIAL_PORT0_LRCLK_POL BIT(3) +#define ADAU17X1_SERIAL_PORT0_MASTER BIT(0) + +#define ADAU17X1_SERIAL_PORT1_DELAY1 0x00 +#define ADAU17X1_SERIAL_PORT1_DELAY0 0x01 +#define ADAU17X1_SERIAL_PORT1_DELAY8 0x02 +#define ADAU17X1_SERIAL_PORT1_DELAY16 0x03 +#define ADAU17X1_SERIAL_PORT1_DELAY_MASK 0x03 + +#define ADAU17X1_CLOCK_CONTROL_INFREQ_MASK 0x6 +#define ADAU17X1_CLOCK_CONTROL_CORECLK_SRC_PLL BIT(3) +#define ADAU17X1_CLOCK_CONTROL_SYSCLK_EN BIT(0) + +#define ADUA_SERIAL_PORT1_BCLK32 (0x0 << 5) +#define ADUA_SERIAL_PORT1_BCLK48 (0x1 << 5) +#define ADUA_SERIAL_PORT1_BCLK64 (0x2 << 5) +#define ADUA_SERIAL_PORT1_BCLK128 (0x3 << 5) +#define ADUA_SERIAL_PORT1_BCLK256 (0x4 << 5) +#define ADAU17X1_SERIAL_PORT1_BCLK_MASK (0x7 << 5) + +#define ADUA_SERIAL_PORT0_STEREO (0x0 << 1) +#define ADUA_SERIAL_PORT0_TDM4 (0x1 << 1) +#define ADUA_SERIAL_PORT0_TDM8 (0x2 << 1) +#define ADAU17X1_SERIAL_PORT0_TDM_MASK (0x3 << 1) +#define ADAU17X1_SERIAL_PORT0_PULSE_MODE BIT(5) + +#define ADAU17X1_CONVERTER0_DAC_PAIR(x) (((x) - 1) << 5) +#define ADAU17X1_CONVERTER0_DAC_PAIR_MASK (0x3 << 5) +#define ADAU17X1_CONVERTER1_ADC_PAIR(x) ((x) - 1) +#define ADAU17X1_CONVERTER1_ADC_PAIR_MASK 0x3 + +#endif -- 1.8.0

2 11

[alsa-devel] 32337ea "ASoC: warn if we will overwrite the DAI stream widgets"
by Stephen Warren 29 Aug '13

29 Aug '13

Russell, if the warning in commit 32337ea "ASoC: warn if we will overwrite the DAI stream widgets" fires, what do I need to do to the ASoC driver to fix it? (I'm deliberately ignoring any other issues with this patch, and looking only to discover what is apparently wrong with my driver that's triggering this...). Steev Klimaszewski reported this warning to me, and I've repro'd it running next-20130827 on a couple of Tegra systems (TrimSlice and Seaboard at least), using sound/soc/tegra/tegra_{trimslice,wm8903}.c, and associated ASoC components. For the record, the commit description is: > ASoC: warn if we will overwrite the DAI stream widgets > > The DAI stream widgets can end up being created multiple times if the > CPU DAI and DMA/platform components share the same struct device. This > bug has been reported in various forms to the ASoC maintainer(s) several > times, yet they refuse to acknowledge this: > > > Essentially all the dmaengine based platforms in mainline use a shared > > device for DMA and DAI; I'm fairly sure someone would have mentioned if > > there were problems. > > > > As you have been repeatedly told the Kirkwood drivers are the first > > drivers submitted to mainline which use DPCM and therefore it is not > > surprising that there are a few issues which need to be worked through, > > there were a few revisions to the framework which went in as a result of > > review during the mainline merge. The problem you are seeing here is > > due to this being the first platform with a *shared* device to use DPCM. > > This is inspite of stating multiple times that this has nothing to do > with DPCM what so ever. So, having tried to convince them over the > last month that this is the case, and them being obstructive and/or > obtuse, this is the only way left open to convince them that there is > a problem here. > > It is only through complete frustration over their inability to recognise > that there could be a problem that I am putting this into linux-next; > this is not something I want to do, but they leave me no other choice. > > If they are correct, then this patch will have no effect what so ever. > If I am correct, there will be warnings produced where a CPU DAI and > DMA/platform component shares the same struct device. > > Signed-off-by: Russell King <rmk+kernel(a)arm.linux.org.uk>

2 1

Re: [alsa-devel] [PATCH V3] i2c: move of helpers into the core
by Grant Likely 28 Aug '13

28 Aug '13

On Thu, 22 Aug 2013 18:00:14 +0200, Wolfram Sang <wsa(a)the-dreams.de> wrote: > I2C of helpers used to live in of_i2c.c but experience (from SPI) shows > that it is much cleaner to have this in the core. This also removes a > circular dependency between the helpers and the core, and so we can > finally register child nodes in the core instead of doing this manually > in each driver. So, fix the drivers and documentation, too. > > Acked-by: Rob Herring <rob.herring(a)calxeda.com> > Reviewed-by: Felipe Balbi <balbi(a)ti.com> > Acked-by: Rafael J. Wysocki <rafael.j.wysocki(a)intel.com> > Tested-by: Sylwester Nawrocki <s.nawrocki(a)samsung.com> > Signed-off-by: Wolfram Sang <wsa(a)the-dreams.de> Acked-by: Grant Likely <grant.likely(a)linaro.org> > --- > > V2->V3: Was trying to be too smart by only fixing includes needed. > Took a more general approach this time, converting of_i2c.h > to i2c.h in case i2c.h was not already there. Otherwise > remove it. Improved my build scripts and no build failures, > no complaints from buildbot as well. > > > Documentation/acpi/enumeration.txt | 1 - > arch/powerpc/platforms/44x/warp.c | 1 - > drivers/gpu/drm/tilcdc/tilcdc_slave.c | 1 - > drivers/gpu/drm/tilcdc/tilcdc_tfp410.c | 1 - > drivers/gpu/host1x/drm/output.c | 2 +- > drivers/i2c/busses/i2c-at91.c | 3 - > drivers/i2c/busses/i2c-cpm.c | 6 -- > drivers/i2c/busses/i2c-davinci.c | 2 - > drivers/i2c/busses/i2c-designware-platdrv.c | 2 - > drivers/i2c/busses/i2c-gpio.c | 3 - > drivers/i2c/busses/i2c-i801.c | 2 - > drivers/i2c/busses/i2c-ibm_iic.c | 4 - > drivers/i2c/busses/i2c-imx.c | 3 - > drivers/i2c/busses/i2c-mpc.c | 2 - > drivers/i2c/busses/i2c-mv64xxx.c | 3 - > drivers/i2c/busses/i2c-mxs.c | 3 - > drivers/i2c/busses/i2c-nomadik.c | 3 - > drivers/i2c/busses/i2c-ocores.c | 3 - > drivers/i2c/busses/i2c-octeon.c | 3 - > drivers/i2c/busses/i2c-omap.c | 3 - > drivers/i2c/busses/i2c-pnx.c | 3 - > drivers/i2c/busses/i2c-powermac.c | 9 +- > drivers/i2c/busses/i2c-pxa.c | 2 - > drivers/i2c/busses/i2c-s3c2410.c | 2 - > drivers/i2c/busses/i2c-sh_mobile.c | 2 - > drivers/i2c/busses/i2c-sirf.c | 3 - > drivers/i2c/busses/i2c-stu300.c | 2 - > drivers/i2c/busses/i2c-tegra.c | 3 - > drivers/i2c/busses/i2c-versatile.c | 2 - > drivers/i2c/busses/i2c-wmt.c | 3 - > drivers/i2c/busses/i2c-xiic.c | 3 - > drivers/i2c/i2c-core.c | 109 +++++++++++++++++++++- > drivers/i2c/i2c-mux.c | 3 - > drivers/i2c/muxes/i2c-arb-gpio-challenge.c | 1 - > drivers/i2c/muxes/i2c-mux-gpio.c | 1 - > drivers/i2c/muxes/i2c-mux-pinctrl.c | 1 - > drivers/media/platform/exynos4-is/fimc-is-i2c.c | 4 +- > drivers/media/platform/exynos4-is/fimc-is.c | 2 +- > drivers/media/platform/exynos4-is/media-dev.c | 1 - > drivers/of/Kconfig | 6 -- > drivers/of/Makefile | 1 - > drivers/of/of_i2c.c | 114 ----------------------- > drivers/staging/imx-drm/imx-tve.c | 2 +- > include/linux/i2c.h | 20 ++++ > include/linux/of_i2c.h | 46 --------- > sound/soc/fsl/imx-sgtl5000.c | 2 +- > sound/soc/fsl/imx-wm8962.c | 2 +- > 47 files changed, 138 insertions(+), 262 deletions(-) > delete mode 100644 drivers/of/of_i2c.c > delete mode 100644 include/linux/of_i2c.h > > diff --git a/Documentation/acpi/enumeration.txt b/Documentation/acpi/enumeration.txt > index d9be7a9..958266e 100644 > --- a/Documentation/acpi/enumeration.txt > +++ b/Documentation/acpi/enumeration.txt > @@ -238,7 +238,6 @@ An I2C bus (controller) driver does: > if (ret) > /* handle error */ > > - of_i2c_register_devices(adapter); > /* Enumerate the slave devices behind this bus via ACPI */ > acpi_i2c_register_devices(adapter); > > diff --git a/arch/powerpc/platforms/44x/warp.c b/arch/powerpc/platforms/44x/warp.c > index 4cfa499..534574a 100644 > --- a/arch/powerpc/platforms/44x/warp.c > +++ b/arch/powerpc/platforms/44x/warp.c > @@ -16,7 +16,6 @@ > #include <linux/interrupt.h> > #include <linux/delay.h> > #include <linux/of_gpio.h> > -#include <linux/of_i2c.h> > #include <linux/slab.h> > #include <linux/export.h> > > diff --git a/drivers/gpu/drm/tilcdc/tilcdc_slave.c b/drivers/gpu/drm/tilcdc/tilcdc_slave.c > index dfffaf0..a19f657 100644 > --- a/drivers/gpu/drm/tilcdc/tilcdc_slave.c > +++ b/drivers/gpu/drm/tilcdc/tilcdc_slave.c > @@ -16,7 +16,6 @@ > */ > > #include <linux/i2c.h> > -#include <linux/of_i2c.h> > #include <linux/pinctrl/pinmux.h> > #include <linux/pinctrl/consumer.h> > #include <drm/drm_encoder_slave.h> > diff --git a/drivers/gpu/drm/tilcdc/tilcdc_tfp410.c b/drivers/gpu/drm/tilcdc/tilcdc_tfp410.c > index 925c7cd..c38b56b 100644 > --- a/drivers/gpu/drm/tilcdc/tilcdc_tfp410.c > +++ b/drivers/gpu/drm/tilcdc/tilcdc_tfp410.c > @@ -16,7 +16,6 @@ > */ > > #include <linux/i2c.h> > -#include <linux/of_i2c.h> > #include <linux/gpio.h> > #include <linux/of_gpio.h> > #include <linux/pinctrl/pinmux.h> > diff --git a/drivers/gpu/host1x/drm/output.c b/drivers/gpu/host1x/drm/output.c > index 8140fc6..137ae81 100644 > --- a/drivers/gpu/host1x/drm/output.c > +++ b/drivers/gpu/host1x/drm/output.c > @@ -9,7 +9,7 @@ > > #include <linux/module.h> > #include <linux/of_gpio.h> > -#include <linux/of_i2c.h> > +#include <linux/i2c.h> > > #include "drm.h" > > diff --git a/drivers/i2c/busses/i2c-at91.c b/drivers/i2c/busses/i2c-at91.c > index 6bb839b..fd05930 100644 > --- a/drivers/i2c/busses/i2c-at91.c > +++ b/drivers/i2c/busses/i2c-at91.c > @@ -28,7 +28,6 @@ > #include <linux/module.h> > #include <linux/of.h> > #include <linux/of_device.h> > -#include <linux/of_i2c.h> > #include <linux/platform_device.h> > #include <linux/slab.h> > #include <linux/platform_data/dma-atmel.h> > @@ -775,8 +774,6 @@ static int at91_twi_probe(struct platform_device *pdev) > return rc; > } > > - of_i2c_register_devices(&dev->adapter); > - > dev_info(dev->dev, "AT91 i2c bus driver.\n"); > return 0; > } > diff --git a/drivers/i2c/busses/i2c-cpm.c b/drivers/i2c/busses/i2c-cpm.c > index 2e1f7eb..b2b8aa9 100644 > --- a/drivers/i2c/busses/i2c-cpm.c > +++ b/drivers/i2c/busses/i2c-cpm.c > @@ -42,7 +42,6 @@ > #include <linux/dma-mapping.h> > #include <linux/of_device.h> > #include <linux/of_platform.h> > -#include <linux/of_i2c.h> > #include <sysdev/fsl_soc.h> > #include <asm/cpm.h> > > @@ -681,11 +680,6 @@ static int cpm_i2c_probe(struct platform_device *ofdev) > dev_dbg(&ofdev->dev, "hw routines for %s registered.\n", > cpm->adap.name); > > - /* > - * register OF I2C devices > - */ > - of_i2c_register_devices(&cpm->adap); > - > return 0; > out_shut: > cpm_i2c_shutdown(cpm); > diff --git a/drivers/i2c/busses/i2c-davinci.c b/drivers/i2c/busses/i2c-davinci.c > index fa55605..62be3b3 100644 > --- a/drivers/i2c/busses/i2c-davinci.c > +++ b/drivers/i2c/busses/i2c-davinci.c > @@ -38,7 +38,6 @@ > #include <linux/slab.h> > #include <linux/cpufreq.h> > #include <linux/gpio.h> > -#include <linux/of_i2c.h> > #include <linux/of_device.h> > > #include <mach/hardware.h> > @@ -728,7 +727,6 @@ static int davinci_i2c_probe(struct platform_device *pdev) > dev_err(&pdev->dev, "failure adding adapter\n"); > goto err_unuse_clocks; > } > - of_i2c_register_devices(adap); > > return 0; > > diff --git a/drivers/i2c/busses/i2c-designware-platdrv.c b/drivers/i2c/busses/i2c-designware-platdrv.c > index 4c5fada..27ea436 100644 > --- a/drivers/i2c/busses/i2c-designware-platdrv.c > +++ b/drivers/i2c/busses/i2c-designware-platdrv.c > @@ -35,7 +35,6 @@ > #include <linux/err.h> > #include <linux/interrupt.h> > #include <linux/of.h> > -#include <linux/of_i2c.h> > #include <linux/platform_device.h> > #include <linux/pm.h> > #include <linux/pm_runtime.h> > @@ -172,7 +171,6 @@ static int dw_i2c_probe(struct platform_device *pdev) > dev_err(&pdev->dev, "failure adding adapter\n"); > return r; > } > - of_i2c_register_devices(adap); > acpi_i2c_register_devices(adap); > > pm_runtime_set_autosuspend_delay(&pdev->dev, 1000); > diff --git a/drivers/i2c/busses/i2c-gpio.c b/drivers/i2c/busses/i2c-gpio.c > index bc6e139..e5da9fe 100644 > --- a/drivers/i2c/busses/i2c-gpio.c > +++ b/drivers/i2c/busses/i2c-gpio.c > @@ -16,7 +16,6 @@ > #include <linux/platform_device.h> > #include <linux/gpio.h> > #include <linux/of_gpio.h> > -#include <linux/of_i2c.h> > > struct i2c_gpio_private_data { > struct i2c_adapter adap; > @@ -224,8 +223,6 @@ static int i2c_gpio_probe(struct platform_device *pdev) > if (ret) > goto err_add_bus; > > - of_i2c_register_devices(adap); > - > platform_set_drvdata(pdev, priv); > > dev_info(&pdev->dev, "using pins %u (SDA) and %u (SCL%s)\n", > diff --git a/drivers/i2c/busses/i2c-i801.c b/drivers/i2c/busses/i2c-i801.c > index 4ebceed..4296d17 100644 > --- a/drivers/i2c/busses/i2c-i801.c > +++ b/drivers/i2c/busses/i2c-i801.c > @@ -87,7 +87,6 @@ > #include <linux/slab.h> > #include <linux/wait.h> > #include <linux/err.h> > -#include <linux/of_i2c.h> > > #if (defined CONFIG_I2C_MUX_GPIO || defined CONFIG_I2C_MUX_GPIO_MODULE) && \ > defined CONFIG_DMI > @@ -1230,7 +1229,6 @@ static int i801_probe(struct pci_dev *dev, const struct pci_device_id *id) > goto exit_free_irq; > } > > - of_i2c_register_devices(&priv->adapter); > i801_probe_optional_slaves(priv); > /* We ignore errors - multiplexing is optional */ > i801_add_mux(priv); > diff --git a/drivers/i2c/busses/i2c-ibm_iic.c b/drivers/i2c/busses/i2c-ibm_iic.c > index 973f516..ff3caa0 100644 > --- a/drivers/i2c/busses/i2c-ibm_iic.c > +++ b/drivers/i2c/busses/i2c-ibm_iic.c > @@ -42,7 +42,6 @@ > #include <linux/io.h> > #include <linux/i2c.h> > #include <linux/of_platform.h> > -#include <linux/of_i2c.h> > > #include "i2c-ibm_iic.h" > > @@ -759,9 +758,6 @@ static int iic_probe(struct platform_device *ofdev) > dev_info(&ofdev->dev, "using %s mode\n", > dev->fast_mode ? "fast (400 kHz)" : "standard (100 kHz)"); > > - /* Now register all the child nodes */ > - of_i2c_register_devices(adap); > - > return 0; > > error_cleanup: > diff --git a/drivers/i2c/busses/i2c-imx.c b/drivers/i2c/busses/i2c-imx.c > index e242797..bbbea6b 100644 > --- a/drivers/i2c/busses/i2c-imx.c > +++ b/drivers/i2c/busses/i2c-imx.c > @@ -50,7 +50,6 @@ > #include <linux/slab.h> > #include <linux/of.h> > #include <linux/of_device.h> > -#include <linux/of_i2c.h> > #include <linux/platform_data/i2c-imx.h> > > /** Defines ******************************************************************** > @@ -570,8 +569,6 @@ static int __init i2c_imx_probe(struct platform_device *pdev) > return ret; > } > > - of_i2c_register_devices(&i2c_imx->adapter); > - > /* Set up platform driver data */ > platform_set_drvdata(pdev, i2c_imx); > > diff --git a/drivers/i2c/busses/i2c-mpc.c b/drivers/i2c/busses/i2c-mpc.c > index 7607dc0..9f2513d 100644 > --- a/drivers/i2c/busses/i2c-mpc.c > +++ b/drivers/i2c/busses/i2c-mpc.c > @@ -18,7 +18,6 @@ > #include <linux/sched.h> > #include <linux/init.h> > #include <linux/of_platform.h> > -#include <linux/of_i2c.h> > #include <linux/slab.h> > > #include <linux/io.h> > @@ -691,7 +690,6 @@ static int fsl_i2c_probe(struct platform_device *op) > dev_err(i2c->dev, "failed to add adapter\n"); > goto fail_add; > } > - of_i2c_register_devices(&i2c->adap); > > return result; > > diff --git a/drivers/i2c/busses/i2c-mv64xxx.c b/drivers/i2c/busses/i2c-mv64xxx.c > index b1f42bf..8220322 100644 > --- a/drivers/i2c/busses/i2c-mv64xxx.c > +++ b/drivers/i2c/busses/i2c-mv64xxx.c > @@ -21,7 +21,6 @@ > #include <linux/of.h> > #include <linux/of_device.h> > #include <linux/of_irq.h> > -#include <linux/of_i2c.h> > #include <linux/clk.h> > #include <linux/err.h> > > @@ -689,8 +688,6 @@ mv64xxx_i2c_probe(struct platform_device *pd) > goto exit_free_irq; > } > > - of_i2c_register_devices(&drv_data->adapter); > - > return 0; > > exit_free_irq: > diff --git a/drivers/i2c/busses/i2c-mxs.c b/drivers/i2c/busses/i2c-mxs.c > index df8ff5a..62ed07d 100644 > --- a/drivers/i2c/busses/i2c-mxs.c > +++ b/drivers/i2c/busses/i2c-mxs.c > @@ -27,7 +27,6 @@ > #include <linux/stmp_device.h> > #include <linux/of.h> > #include <linux/of_device.h> > -#include <linux/of_i2c.h> > #include <linux/dma-mapping.h> > #include <linux/dmaengine.h> > > @@ -701,8 +700,6 @@ static int mxs_i2c_probe(struct platform_device *pdev) > return err; > } > > - of_i2c_register_devices(adap); > - > return 0; > } > > diff --git a/drivers/i2c/busses/i2c-nomadik.c b/drivers/i2c/busses/i2c-nomadik.c > index 512dfe6..519df17 100644 > --- a/drivers/i2c/busses/i2c-nomadik.c > +++ b/drivers/i2c/busses/i2c-nomadik.c > @@ -24,7 +24,6 @@ > #include <linux/pm_runtime.h> > #include <linux/platform_data/i2c-nomadik.h> > #include <linux/of.h> > -#include <linux/of_i2c.h> > #include <linux/pinctrl/consumer.h> > > #define DRIVER_NAME "nmk-i2c" > @@ -1045,8 +1044,6 @@ static int nmk_i2c_probe(struct amba_device *adev, const struct amba_id *id) > goto err_add_adap; > } > > - of_i2c_register_devices(adap); > - > pm_runtime_put(&adev->dev); > > return 0; > diff --git a/drivers/i2c/busses/i2c-ocores.c b/drivers/i2c/busses/i2c-ocores.c > index 0e1f824..0a52b78 100644 > --- a/drivers/i2c/busses/i2c-ocores.c > +++ b/drivers/i2c/busses/i2c-ocores.c > @@ -24,7 +24,6 @@ > #include <linux/i2c-ocores.h> > #include <linux/slab.h> > #include <linux/io.h> > -#include <linux/of_i2c.h> > #include <linux/log2.h> > > struct ocores_i2c { > @@ -435,8 +434,6 @@ static int ocores_i2c_probe(struct platform_device *pdev) > if (pdata) { > for (i = 0; i < pdata->num_devices; i++) > i2c_new_device(&i2c->adap, pdata->devices + i); > - } else { > - of_i2c_register_devices(&i2c->adap); > } > > return 0; > diff --git a/drivers/i2c/busses/i2c-octeon.c b/drivers/i2c/busses/i2c-octeon.c > index 956fe32..b929ba2 100644 > --- a/drivers/i2c/busses/i2c-octeon.c > +++ b/drivers/i2c/busses/i2c-octeon.c > @@ -15,7 +15,6 @@ > #include <linux/interrupt.h> > #include <linux/kernel.h> > #include <linux/module.h> > -#include <linux/of_i2c.h> > #include <linux/delay.h> > #include <linux/sched.h> > #include <linux/slab.h> > @@ -599,8 +598,6 @@ static int octeon_i2c_probe(struct platform_device *pdev) > } > dev_info(i2c->dev, "version %s\n", DRV_VERSION); > > - of_i2c_register_devices(&i2c->adap); > - > return 0; > > out: > diff --git a/drivers/i2c/busses/i2c-omap.c b/drivers/i2c/busses/i2c-omap.c > index 142b694d..a9f0f80 100644 > --- a/drivers/i2c/busses/i2c-omap.c > +++ b/drivers/i2c/busses/i2c-omap.c > @@ -38,7 +38,6 @@ > #include <linux/clk.h> > #include <linux/io.h> > #include <linux/of.h> > -#include <linux/of_i2c.h> > #include <linux/of_device.h> > #include <linux/slab.h> > #include <linux/i2c-omap.h> > @@ -1245,8 +1244,6 @@ omap_i2c_probe(struct platform_device *pdev) > dev_info(dev->dev, "bus %d rev%d.%d at %d kHz\n", adap->nr, > major, minor, dev->speed); > > - of_i2c_register_devices(adap); > - > pm_runtime_mark_last_busy(dev->dev); > pm_runtime_put_autosuspend(dev->dev); > > diff --git a/drivers/i2c/busses/i2c-pnx.c b/drivers/i2c/busses/i2c-pnx.c > index 5f39c6d..7b57d67 100644 > --- a/drivers/i2c/busses/i2c-pnx.c > +++ b/drivers/i2c/busses/i2c-pnx.c > @@ -23,7 +23,6 @@ > #include <linux/err.h> > #include <linux/clk.h> > #include <linux/slab.h> > -#include <linux/of_i2c.h> > > #define I2C_PNX_TIMEOUT_DEFAULT 10 /* msec */ > #define I2C_PNX_SPEED_KHZ_DEFAULT 100 > @@ -741,8 +740,6 @@ static int i2c_pnx_probe(struct platform_device *pdev) > goto out_irq; > } > > - of_i2c_register_devices(&alg_data->adapter); > - > dev_dbg(&pdev->dev, "%s: Master at %#8x, irq %d.\n", > alg_data->adapter.name, res->start, alg_data->irq); > > diff --git a/drivers/i2c/busses/i2c-powermac.c b/drivers/i2c/busses/i2c-powermac.c > index 8dc90da..1010f2e 100644 > --- a/drivers/i2c/busses/i2c-powermac.c > +++ b/drivers/i2c/busses/i2c-powermac.c > @@ -440,7 +440,9 @@ static int i2c_powermac_probe(struct platform_device *dev) > adapter->algo = &i2c_powermac_algorithm; > i2c_set_adapdata(adapter, bus); > adapter->dev.parent = &dev->dev; > - adapter->dev.of_node = dev->dev.of_node; > + > + /* Clear of_node to skip automatic registration of i2c child nodes */ > + adapter->dev.of_node = NULL; > rc = i2c_add_adapter(adapter); > if (rc) { > printk(KERN_ERR "i2c-powermac: Adapter %s registration " > @@ -450,9 +452,8 @@ static int i2c_powermac_probe(struct platform_device *dev) > > printk(KERN_INFO "PowerMac i2c bus %s registered\n", adapter->name); > > - /* Cannot use of_i2c_register_devices() due to Apple device-tree > - * funkyness > - */ > + /* Use custom child registration due to Apple device-tree funkyness */ > + adapter->dev.of_node = dev->dev.of_node; > i2c_powermac_register_devices(adapter, bus); > > return rc; > diff --git a/drivers/i2c/busses/i2c-pxa.c b/drivers/i2c/busses/i2c-pxa.c > index fbafed2..bc65014 100644 > --- a/drivers/i2c/busses/i2c-pxa.c > +++ b/drivers/i2c/busses/i2c-pxa.c > @@ -31,7 +31,6 @@ > #include <linux/i2c-pxa.h> > #include <linux/of.h> > #include <linux/of_device.h> > -#include <linux/of_i2c.h> > #include <linux/platform_device.h> > #include <linux/err.h> > #include <linux/clk.h> > @@ -1185,7 +1184,6 @@ static int i2c_pxa_probe(struct platform_device *dev) > printk(KERN_INFO "I2C: Failed to add bus\n"); > goto eadapt; > } > - of_i2c_register_devices(&i2c->adap); > > platform_set_drvdata(dev, i2c); > > diff --git a/drivers/i2c/busses/i2c-s3c2410.c b/drivers/i2c/busses/i2c-s3c2410.c > index cab1c91..643426e 100644 > --- a/drivers/i2c/busses/i2c-s3c2410.c > +++ b/drivers/i2c/busses/i2c-s3c2410.c > @@ -36,7 +36,6 @@ > #include <linux/cpufreq.h> > #include <linux/slab.h> > #include <linux/io.h> > -#include <linux/of_i2c.h> > #include <linux/of_gpio.h> > #include <linux/pinctrl/consumer.h> > > @@ -1154,7 +1153,6 @@ static int s3c24xx_i2c_probe(struct platform_device *pdev) > return ret; > } > > - of_i2c_register_devices(&i2c->adap); > platform_set_drvdata(pdev, i2c); > > pm_runtime_enable(&pdev->dev); > diff --git a/drivers/i2c/busses/i2c-sh_mobile.c b/drivers/i2c/busses/i2c-sh_mobile.c > index debf745..aa1268f 100644 > --- a/drivers/i2c/busses/i2c-sh_mobile.c > +++ b/drivers/i2c/busses/i2c-sh_mobile.c > @@ -27,7 +27,6 @@ > #include <linux/platform_device.h> > #include <linux/interrupt.h> > #include <linux/i2c.h> > -#include <linux/of_i2c.h> > #include <linux/err.h> > #include <linux/pm_runtime.h> > #include <linux/clk.h> > @@ -758,7 +757,6 @@ static int sh_mobile_i2c_probe(struct platform_device *dev) > "I2C adapter %d with bus speed %lu Hz (L/H=%x/%x)\n", > adap->nr, pd->bus_speed, pd->iccl, pd->icch); > > - of_i2c_register_devices(adap); > return 0; > > err_all: > diff --git a/drivers/i2c/busses/i2c-sirf.c b/drivers/i2c/busses/i2c-sirf.c > index a63c7d5..0ff22e2 100644 > --- a/drivers/i2c/busses/i2c-sirf.c > +++ b/drivers/i2c/busses/i2c-sirf.c > @@ -12,7 +12,6 @@ > #include <linux/slab.h> > #include <linux/platform_device.h> > #include <linux/i2c.h> > -#include <linux/of_i2c.h> > #include <linux/clk.h> > #include <linux/err.h> > #include <linux/io.h> > @@ -366,8 +365,6 @@ static int i2c_sirfsoc_probe(struct platform_device *pdev) > > clk_disable(clk); > > - of_i2c_register_devices(adap); > - > dev_info(&pdev->dev, " I2C adapter ready to operate\n"); > > return 0; > diff --git a/drivers/i2c/busses/i2c-stu300.c b/drivers/i2c/busses/i2c-stu300.c > index d1a6b20..047546c 100644 > --- a/drivers/i2c/busses/i2c-stu300.c > +++ b/drivers/i2c/busses/i2c-stu300.c > @@ -17,7 +17,6 @@ > #include <linux/clk.h> > #include <linux/io.h> > #include <linux/slab.h> > -#include <linux/of_i2c.h> > > /* the name of this kernel module */ > #define NAME "stu300" > @@ -936,7 +935,6 @@ stu300_probe(struct platform_device *pdev) > platform_set_drvdata(pdev, dev); > dev_info(&pdev->dev, "ST DDC I2C @ %p, irq %d\n", > dev->virtbase, dev->irq); > - of_i2c_register_devices(adap); > > return 0; > } > diff --git a/drivers/i2c/busses/i2c-tegra.c b/drivers/i2c/busses/i2c-tegra.c > index 9aa1b60..c457cb4 100644 > --- a/drivers/i2c/busses/i2c-tegra.c > +++ b/drivers/i2c/busses/i2c-tegra.c > @@ -25,7 +25,6 @@ > #include <linux/interrupt.h> > #include <linux/delay.h> > #include <linux/slab.h> > -#include <linux/of_i2c.h> > #include <linux/of_device.h> > #include <linux/module.h> > #include <linux/clk/tegra.h> > @@ -802,8 +801,6 @@ static int tegra_i2c_probe(struct platform_device *pdev) > return ret; > } > > - of_i2c_register_devices(&i2c_dev->adapter); > - > return 0; > } > > diff --git a/drivers/i2c/busses/i2c-versatile.c b/drivers/i2c/busses/i2c-versatile.c > index f3a8790..6bb3a89 100644 > --- a/drivers/i2c/busses/i2c-versatile.c > +++ b/drivers/i2c/busses/i2c-versatile.c > @@ -16,7 +16,6 @@ > #include <linux/platform_device.h> > #include <linux/slab.h> > #include <linux/io.h> > -#include <linux/of_i2c.h> > > #define I2C_CONTROL 0x00 > #define I2C_CONTROLS 0x00 > @@ -108,7 +107,6 @@ static int i2c_versatile_probe(struct platform_device *dev) > ret = i2c_bit_add_numbered_bus(&i2c->adap); > if (ret >= 0) { > platform_set_drvdata(dev, i2c); > - of_i2c_register_devices(&i2c->adap); > return 0; > } > > diff --git a/drivers/i2c/busses/i2c-wmt.c b/drivers/i2c/busses/i2c-wmt.c > index baaa7d1..c65da3d 100644 > --- a/drivers/i2c/busses/i2c-wmt.c > +++ b/drivers/i2c/busses/i2c-wmt.c > @@ -21,7 +21,6 @@ > #include <linux/module.h> > #include <linux/of.h> > #include <linux/of_address.h> > -#include <linux/of_i2c.h> > #include <linux/of_irq.h> > #include <linux/platform_device.h> > > @@ -439,8 +438,6 @@ static int wmt_i2c_probe(struct platform_device *pdev) > > platform_set_drvdata(pdev, i2c_dev); > > - of_i2c_register_devices(adap); > - > return 0; > } > > diff --git a/drivers/i2c/busses/i2c-xiic.c b/drivers/i2c/busses/i2c-xiic.c > index 3d0f052..8823db7 100644 > --- a/drivers/i2c/busses/i2c-xiic.c > +++ b/drivers/i2c/busses/i2c-xiic.c > @@ -40,7 +40,6 @@ > #include <linux/i2c-xiic.h> > #include <linux/io.h> > #include <linux/slab.h> > -#include <linux/of_i2c.h> > > #define DRIVER_NAME "xiic-i2c" > > @@ -752,8 +751,6 @@ static int xiic_i2c_probe(struct platform_device *pdev) > i2c_new_device(&i2c->adap, pdata->devices + i); > } > > - of_i2c_register_devices(&i2c->adap); > - > return 0; > > add_adapter_failed: > diff --git a/drivers/i2c/i2c-core.c b/drivers/i2c/i2c-core.c > index f32ca29..08ebd78 100644 > --- a/drivers/i2c/i2c-core.c > +++ b/drivers/i2c/i2c-core.c > @@ -23,7 +23,11 @@ > SMBus 2.0 support by Mark Studebaker <mdsxyz123(a)yahoo.com> and > Jean Delvare <khali(a)linux-fr.org> > Mux support by Rodolfo Giometti <giometti(a)enneenne.com> and > - Michael Lawnick <michael.lawnick.ext(a)nsn.com> */ > + Michael Lawnick <michael.lawnick.ext(a)nsn.com> > + OF support is copyright (c) 2008 Jochen Friedrich <jochen(a)scram.de> > + (based on a previous patch from Jon Smirl <jonsmirl(a)gmail.com>) and > + (c) 2013 Wolfram Sang <wsa(a)the-dreams.de> > + */ > > #include <linux/module.h> > #include <linux/kernel.h> > @@ -35,7 +39,9 @@ > #include <linux/init.h> > #include <linux/idr.h> > #include <linux/mutex.h> > +#include <linux/of.h> > #include <linux/of_device.h> > +#include <linux/of_irq.h> > #include <linux/completion.h> > #include <linux/hardirq.h> > #include <linux/irqflags.h> > @@ -954,6 +960,104 @@ static void i2c_scan_static_board_info(struct i2c_adapter *adapter) > up_read(&__i2c_board_lock); > } > > +/* OF support code */ > + > +#if IS_ENABLED(CONFIG_OF) > +static void of_i2c_register_devices(struct i2c_adapter *adap) > +{ > + void *result; > + struct device_node *node; > + > + /* Only register child devices if the adapter has a node pointer set */ > + if (!adap->dev.of_node) > + return; > + > + dev_dbg(&adap->dev, "of_i2c: walking child nodes\n"); > + > + for_each_available_child_of_node(adap->dev.of_node, node) { > + struct i2c_board_info info = {}; > + struct dev_archdata dev_ad = {}; > + const __be32 *addr; > + int len; > + > + dev_dbg(&adap->dev, "of_i2c: register %s\n", node->full_name); > + > + if (of_modalias_node(node, info.type, sizeof(info.type)) < 0) { > + dev_err(&adap->dev, "of_i2c: modalias failure on %s\n", > + node->full_name); > + continue; > + } > + > + addr = of_get_property(node, "reg", &len); > + if (!addr || (len < sizeof(int))) { > + dev_err(&adap->dev, "of_i2c: invalid reg on %s\n", > + node->full_name); > + continue; > + } > + > + info.addr = be32_to_cpup(addr); > + if (info.addr > (1 << 10) - 1) { > + dev_err(&adap->dev, "of_i2c: invalid addr=%x on %s\n", > + info.addr, node->full_name); > + continue; > + } > + > + info.irq = irq_of_parse_and_map(node, 0); > + info.of_node = of_node_get(node); > + info.archdata = &dev_ad; > + > + if (of_get_property(node, "wakeup-source", NULL)) > + info.flags |= I2C_CLIENT_WAKE; > + > + request_module("%s%s", I2C_MODULE_PREFIX, info.type); > + > + result = i2c_new_device(adap, &info); > + if (result == NULL) { > + dev_err(&adap->dev, "of_i2c: Failure registering %s\n", > + node->full_name); > + of_node_put(node); > + irq_dispose_mapping(info.irq); > + continue; > + } > + } > +} > + > +static int of_dev_node_match(struct device *dev, void *data) > +{ > + return dev->of_node == data; > +} > + > +/* must call put_device() when done with returned i2c_client device */ > +struct i2c_client *of_find_i2c_device_by_node(struct device_node *node) > +{ > + struct device *dev; > + > + dev = bus_find_device(&i2c_bus_type, NULL, node, > + of_dev_node_match); > + if (!dev) > + return NULL; > + > + return i2c_verify_client(dev); > +} > +EXPORT_SYMBOL(of_find_i2c_device_by_node); > + > +/* must call put_device() when done with returned i2c_adapter device */ > +struct i2c_adapter *of_find_i2c_adapter_by_node(struct device_node *node) > +{ > + struct device *dev; > + > + dev = bus_find_device(&i2c_bus_type, NULL, node, > + of_dev_node_match); > + if (!dev) > + return NULL; > + > + return i2c_verify_adapter(dev); > +} > +EXPORT_SYMBOL(of_find_i2c_adapter_by_node); > +#else > +static void of_i2c_register_devices(struct i2c_adapter *adap) { } > +#endif /* CONFIG_OF */ > + > static int i2c_do_add_adapter(struct i2c_driver *driver, > struct i2c_adapter *adap) > { > @@ -1058,6 +1162,8 @@ static int i2c_register_adapter(struct i2c_adapter *adap) > > exit_recovery: > /* create pre-declared device nodes */ > + of_i2c_register_devices(adap); > + > if (adap->nr < __i2c_first_dynamic_bus_num) > i2c_scan_static_board_info(adap); > > @@ -1282,7 +1388,6 @@ void i2c_del_adapter(struct i2c_adapter *adap) > } > EXPORT_SYMBOL(i2c_del_adapter); > > - > /* ------------------------------------------------------------------------- */ > > int i2c_for_each_dev(void *data, int (*fn)(struct device *, void *)) > diff --git a/drivers/i2c/i2c-mux.c b/drivers/i2c/i2c-mux.c > index 7409ebb..797e311 100644 > --- a/drivers/i2c/i2c-mux.c > +++ b/drivers/i2c/i2c-mux.c > @@ -25,7 +25,6 @@ > #include <linux/i2c.h> > #include <linux/i2c-mux.h> > #include <linux/of.h> > -#include <linux/of_i2c.h> > > /* multiplexer per channel data */ > struct i2c_mux_priv { > @@ -185,8 +184,6 @@ struct i2c_adapter *i2c_add_mux_adapter(struct i2c_adapter *parent, > dev_info(&parent->dev, "Added multiplexed i2c bus %d\n", > i2c_adapter_id(&priv->adap)); > > - of_i2c_register_devices(&priv->adap); > - > return &priv->adap; > } > EXPORT_SYMBOL_GPL(i2c_add_mux_adapter); > diff --git a/drivers/i2c/muxes/i2c-arb-gpio-challenge.c b/drivers/i2c/muxes/i2c-arb-gpio-challenge.c > index 210b6f7..b901638 100644 > --- a/drivers/i2c/muxes/i2c-arb-gpio-challenge.c > +++ b/drivers/i2c/muxes/i2c-arb-gpio-challenge.c > @@ -21,7 +21,6 @@ > #include <linux/i2c-mux.h> > #include <linux/init.h> > #include <linux/module.h> > -#include <linux/of_i2c.h> > #include <linux/of_gpio.h> > #include <linux/platform_device.h> > #include <linux/slab.h> > diff --git a/drivers/i2c/muxes/i2c-mux-gpio.c b/drivers/i2c/muxes/i2c-mux-gpio.c > index 5a0ce00..128a981 100644 > --- a/drivers/i2c/muxes/i2c-mux-gpio.c > +++ b/drivers/i2c/muxes/i2c-mux-gpio.c > @@ -16,7 +16,6 @@ > #include <linux/module.h> > #include <linux/slab.h> > #include <linux/gpio.h> > -#include <linux/of_i2c.h> > #include <linux/of_gpio.h> > > struct gpiomux { > diff --git a/drivers/i2c/muxes/i2c-mux-pinctrl.c b/drivers/i2c/muxes/i2c-mux-pinctrl.c > index a43c0ce..859a6d2 100644 > --- a/drivers/i2c/muxes/i2c-mux-pinctrl.c > +++ b/drivers/i2c/muxes/i2c-mux-pinctrl.c > @@ -20,7 +20,6 @@ > #include <linux/i2c-mux.h> > #include <linux/init.h> > #include <linux/module.h> > -#include <linux/of_i2c.h> > #include <linux/pinctrl/consumer.h> > #include <linux/i2c-mux-pinctrl.h> > #include <linux/platform_device.h> > diff --git a/drivers/media/platform/exynos4-is/fimc-is-i2c.c b/drivers/media/platform/exynos4-is/fimc-is-i2c.c > index 617a798..9930556 100644 > --- a/drivers/media/platform/exynos4-is/fimc-is-i2c.c > +++ b/drivers/media/platform/exynos4-is/fimc-is-i2c.c > @@ -12,7 +12,7 @@ > > #include <linux/clk.h> > #include <linux/module.h> > -#include <linux/of_i2c.h> > +#include <linux/i2c.h> > #include <linux/platform_device.h> > #include <linux/pm_runtime.h> > #include <linux/slab.h> > @@ -67,8 +67,6 @@ static int fimc_is_i2c_probe(struct platform_device *pdev) > pm_runtime_enable(&pdev->dev); > pm_runtime_enable(&i2c_adap->dev); > > - of_i2c_register_devices(i2c_adap); > - > return 0; > } > > diff --git a/drivers/media/platform/exynos4-is/fimc-is.c b/drivers/media/platform/exynos4-is/fimc-is.c > index 967f6a9..2276fdc 100644 > --- a/drivers/media/platform/exynos4-is/fimc-is.c > +++ b/drivers/media/platform/exynos4-is/fimc-is.c > @@ -21,7 +21,7 @@ > #include <linux/interrupt.h> > #include <linux/kernel.h> > #include <linux/module.h> > -#include <linux/of_i2c.h> > +#include <linux/i2c.h> > #include <linux/of_irq.h> > #include <linux/of_address.h> > #include <linux/of_platform.h> > diff --git a/drivers/media/platform/exynos4-is/media-dev.c b/drivers/media/platform/exynos4-is/media-dev.c > index 19f556c..f8c66b4 100644 > --- a/drivers/media/platform/exynos4-is/media-dev.c > +++ b/drivers/media/platform/exynos4-is/media-dev.c > @@ -20,7 +20,6 @@ > #include <linux/of.h> > #include <linux/of_platform.h> > #include <linux/of_device.h> > -#include <linux/of_i2c.h> > #include <linux/platform_device.h> > #include <linux/pm_runtime.h> > #include <linux/types.h> > diff --git a/drivers/of/Kconfig b/drivers/of/Kconfig > index 80e5c13..78cc760 100644 > --- a/drivers/of/Kconfig > +++ b/drivers/of/Kconfig > @@ -48,12 +48,6 @@ config OF_IRQ > def_bool y > depends on !SPARC > > -config OF_I2C > - def_tristate I2C > - depends on I2C > - help > - OpenFirmware I2C accessors > - > config OF_NET > depends on NETDEVICES > def_bool y > diff --git a/drivers/of/Makefile b/drivers/of/Makefile > index 1f9c0c4..efd0510 100644 > --- a/drivers/of/Makefile > +++ b/drivers/of/Makefile > @@ -3,7 +3,6 @@ obj-$(CONFIG_OF_FLATTREE) += fdt.o > obj-$(CONFIG_OF_PROMTREE) += pdt.o > obj-$(CONFIG_OF_ADDRESS) += address.o > obj-$(CONFIG_OF_IRQ) += irq.o > -obj-$(CONFIG_OF_I2C) += of_i2c.o > obj-$(CONFIG_OF_NET) += of_net.o > obj-$(CONFIG_OF_SELFTEST) += selftest.o > obj-$(CONFIG_OF_MDIO) += of_mdio.o > diff --git a/drivers/of/of_i2c.c b/drivers/of/of_i2c.c > deleted file mode 100644 > index b667264..0000000 > --- a/drivers/of/of_i2c.c > +++ /dev/null > @@ -1,114 +0,0 @@ > -/* > - * OF helpers for the I2C API > - * > - * Copyright (c) 2008 Jochen Friedrich <jochen(a)scram.de> > - * > - * Based on a previous patch from Jon Smirl <jonsmirl(a)gmail.com> > - * > - * This program is free software; you can redistribute it and/or modify > - * it under the terms of the GNU General Public License as published by > - * the Free Software Foundation; either version 2 of the License, or > - * (at your option) any later version. > - */ > - > -#include <linux/i2c.h> > -#include <linux/irq.h> > -#include <linux/of.h> > -#include <linux/of_i2c.h> > -#include <linux/of_irq.h> > -#include <linux/module.h> > - > -void of_i2c_register_devices(struct i2c_adapter *adap) > -{ > - void *result; > - struct device_node *node; > - > - /* Only register child devices if the adapter has a node pointer set */ > - if (!adap->dev.of_node) > - return; > - > - dev_dbg(&adap->dev, "of_i2c: walking child nodes\n"); > - > - for_each_available_child_of_node(adap->dev.of_node, node) { > - struct i2c_board_info info = {}; > - struct dev_archdata dev_ad = {}; > - const __be32 *addr; > - int len; > - > - dev_dbg(&adap->dev, "of_i2c: register %s\n", node->full_name); > - > - if (of_modalias_node(node, info.type, sizeof(info.type)) < 0) { > - dev_err(&adap->dev, "of_i2c: modalias failure on %s\n", > - node->full_name); > - continue; > - } > - > - addr = of_get_property(node, "reg", &len); > - if (!addr || (len < sizeof(int))) { > - dev_err(&adap->dev, "of_i2c: invalid reg on %s\n", > - node->full_name); > - continue; > - } > - > - info.addr = be32_to_cpup(addr); > - if (info.addr > (1 << 10) - 1) { > - dev_err(&adap->dev, "of_i2c: invalid addr=%x on %s\n", > - info.addr, node->full_name); > - continue; > - } > - > - info.irq = irq_of_parse_and_map(node, 0); > - info.of_node = of_node_get(node); > - info.archdata = &dev_ad; > - > - if (of_get_property(node, "wakeup-source", NULL)) > - info.flags |= I2C_CLIENT_WAKE; > - > - request_module("%s%s", I2C_MODULE_PREFIX, info.type); > - > - result = i2c_new_device(adap, &info); > - if (result == NULL) { > - dev_err(&adap->dev, "of_i2c: Failure registering %s\n", > - node->full_name); > - of_node_put(node); > - irq_dispose_mapping(info.irq); > - continue; > - } > - } > -} > -EXPORT_SYMBOL(of_i2c_register_devices); > - > -static int of_dev_node_match(struct device *dev, void *data) > -{ > - return dev->of_node == data; > -} > - > -/* must call put_device() when done with returned i2c_client device */ > -struct i2c_client *of_find_i2c_device_by_node(struct device_node *node) > -{ > - struct device *dev; > - > - dev = bus_find_device(&i2c_bus_type, NULL, node, > - of_dev_node_match); > - if (!dev) > - return NULL; > - > - return i2c_verify_client(dev); > -} > -EXPORT_SYMBOL(of_find_i2c_device_by_node); > - > -/* must call put_device() when done with returned i2c_adapter device */ > -struct i2c_adapter *of_find_i2c_adapter_by_node(struct device_node *node) > -{ > - struct device *dev; > - > - dev = bus_find_device(&i2c_bus_type, NULL, node, > - of_dev_node_match); > - if (!dev) > - return NULL; > - > - return i2c_verify_adapter(dev); > -} > -EXPORT_SYMBOL(of_find_i2c_adapter_by_node); > - > -MODULE_LICENSE("GPL"); > diff --git a/drivers/staging/imx-drm/imx-tve.c b/drivers/staging/imx-drm/imx-tve.c > index a56797d..2d76fd4 100644 > --- a/drivers/staging/imx-drm/imx-tve.c > +++ b/drivers/staging/imx-drm/imx-tve.c > @@ -21,7 +21,7 @@ > #include <linux/clk.h> > #include <linux/clk-provider.h> > #include <linux/module.h> > -#include <linux/of_i2c.h> > +#include <linux/i2c.h> > #include <linux/regmap.h> > #include <linux/regulator/consumer.h> > #include <linux/spinlock.h> > diff --git a/include/linux/i2c.h b/include/linux/i2c.h > index e988fa9..2189189 100644 > --- a/include/linux/i2c.h > +++ b/include/linux/i2c.h > @@ -542,6 +542,26 @@ static inline int i2c_adapter_id(struct i2c_adapter *adap) > > #endif /* I2C */ > > +#if IS_ENABLED(CONFIG_OF) > +/* must call put_device() when done with returned i2c_client device */ > +extern struct i2c_client *of_find_i2c_device_by_node(struct device_node *node); > + > +/* must call put_device() when done with returned i2c_adapter device */ > +extern struct i2c_adapter *of_find_i2c_adapter_by_node(struct device_node *node); > + > +#else > + > +static inline struct i2c_client *of_find_i2c_device_by_node(struct device_node *node) > +{ > + return NULL; > +} > + > +static inline struct i2c_adapter *of_find_i2c_adapter_by_node(struct device_node *node) > +{ > + return NULL; > +} > +#endif /* CONFIG_OF */ > + > #if IS_ENABLED(CONFIG_ACPI_I2C) > extern void acpi_i2c_register_devices(struct i2c_adapter *adap); > #else > diff --git a/include/linux/of_i2c.h b/include/linux/of_i2c.h > deleted file mode 100644 > index cfb545c..0000000 > --- a/include/linux/of_i2c.h > +++ /dev/null > @@ -1,46 +0,0 @@ > -/* > - * Generic I2C API implementation for PowerPC. > - * > - * Copyright (c) 2008 Jochen Friedrich <jochen(a)scram.de> > - * > - * This program is free software; you can redistribute it and/or modify > - * it under the terms of the GNU General Public License as published by > - * the Free Software Foundation; either version 2 of the License, or > - * (at your option) any later version. > - */ > - > -#ifndef __LINUX_OF_I2C_H > -#define __LINUX_OF_I2C_H > - > -#if defined(CONFIG_OF_I2C) || defined(CONFIG_OF_I2C_MODULE) > -#include <linux/i2c.h> > - > -extern void of_i2c_register_devices(struct i2c_adapter *adap); > - > -/* must call put_device() when done with returned i2c_client device */ > -extern struct i2c_client *of_find_i2c_device_by_node(struct device_node *node); > - > -/* must call put_device() when done with returned i2c_adapter device */ > -extern struct i2c_adapter *of_find_i2c_adapter_by_node( > - struct device_node *node); > - > -#else > -static inline void of_i2c_register_devices(struct i2c_adapter *adap) > -{ > - return; > -} > - > -static inline struct i2c_client *of_find_i2c_device_by_node(struct device_node *node) > -{ > - return NULL; > -} > - > -/* must call put_device() when done with returned i2c_adapter device */ > -static inline struct i2c_adapter *of_find_i2c_adapter_by_node( > - struct device_node *node) > -{ > - return NULL; > -} > -#endif /* CONFIG_OF_I2C */ > - > -#endif /* __LINUX_OF_I2C_H */ > diff --git a/sound/soc/fsl/imx-sgtl5000.c b/sound/soc/fsl/imx-sgtl5000.c > index 3f726e4..f2fbde9 100644 > --- a/sound/soc/fsl/imx-sgtl5000.c > +++ b/sound/soc/fsl/imx-sgtl5000.c > @@ -13,7 +13,7 @@ > #include <linux/module.h> > #include <linux/of.h> > #include <linux/of_platform.h> > -#include <linux/of_i2c.h> > +#include <linux/i2c.h> > #include <linux/clk.h> > #include <sound/soc.h> > > diff --git a/sound/soc/fsl/imx-wm8962.c b/sound/soc/fsl/imx-wm8962.c > index 52a36a9..9fd7a65 100644 > --- a/sound/soc/fsl/imx-wm8962.c > +++ b/sound/soc/fsl/imx-wm8962.c > @@ -15,7 +15,7 @@ > > #include <linux/module.h> > #include <linux/of_platform.h> > -#include <linux/of_i2c.h> > +#include <linux/i2c.h> > #include <linux/slab.h> > #include <linux/clk.h> > #include <sound/soc.h> > -- > 1.7.10.4 > > -- > To unsubscribe from this list: send the line "unsubscribe linux-kernel" in > the body of a message to majordomo(a)vger.kernel.org > More majordomo info at http://vger.kernel.org/majordomo-info.html > Please read the FAQ at http://www.tux.org/lkml/

1 0

[alsa-devel] [PATCH] ASoc: kirkwood: Use the Kirkwood audio driver in Dove boards
by Jean-Francois Moine 28 Aug '13

28 Aug '13

This patch permits the generation of the Kirkwood audio driver which may be used in the Dove boards. Signed-off-by: Jean-Francois Moine <moinejf(a)free.fr> --- sound/soc/kirkwood/Kconfig | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/soc/kirkwood/Kconfig b/sound/soc/kirkwood/Kconfig index 9e1970c..78ed4a4 100644 --- a/sound/soc/kirkwood/Kconfig +++ b/sound/soc/kirkwood/Kconfig @@ -1,6 +1,6 @@ config SND_KIRKWOOD_SOC - tristate "SoC Audio for the Marvell Kirkwood chip" - depends on ARCH_KIRKWOOD || COMPILE_TEST + tristate "SoC Audio for the Marvell Kirkwood and Dove chips" + depends on ARCH_KIRKWOOD || ARCH_DOVE || COMPILE_TEST help Say Y or M if you want to add support for codecs attached to the Kirkwood I2S interface. You will also need to select the -- Ken ar c'hentañ | ** Breizh ha Linux atav! ** Jef | http://moinejf.free.fr/

2 1

Re: [alsa-devel] Buffer size for ALSA USB PCM audio
by Alan Stern 27 Aug '13

27 Aug '13

On Tue, 27 Aug 2013, James Stone wrote: > This does not help at all with running jackd at lower latencies. > > With this patch, I get xruns at 128 frames/period or less with jackd > (accompanied by iso underrun messages), whereas with vanilla 3.10.9 > (which now has both yours and Clemen's patches applied), performance > is extremely good (64 frames per period possible). Was this was both recording and playback, or playback only? If you were doing both, what happens with just playback? Assuming you see the same problem with just playback, please collect a usbmon trace showing one or two of those underruns at say 64 frames/period. And for comparison, it would be nice to have a usbmon trace showing a couple of seconds of normal operation under vanilla 3.10.9. Alan Stern

1 0

Re: [alsa-devel] [PATCH] ASoC: codecs/sgtl5000.c: specify correct reg_stride in regmap_config.
by David Jander 27 Aug '13

27 Aug '13

On Sun, 25 Aug 2013 16:44:26 +0800 zengzhaoming <zengzm.kernel(a)gmail.com> wrote: > On 08-22/08:23, David Jander wrote: > > This makes the RB-Tree more compact and efficient. > > > > Signed-off-by: David Jander <david(a)protonic.nl> > > --- > > sound/soc/codecs/sgtl5000.c | 1 + > > 1 file changed, 1 insertion(+) > > > > diff --git a/sound/soc/codecs/sgtl5000.c b/sound/soc/codecs/sgtl5000.c > > index 760e8bf..ba5ca67 100644 > > --- a/sound/soc/codecs/sgtl5000.c > > +++ b/sound/soc/codecs/sgtl5000.c > > @@ -1480,6 +1480,7 @@ static struct snd_soc_codec_driver sgtl5000_driver = > > { static const struct regmap_config sgtl5000_regmap = { > > .reg_bits = 16, > > .val_bits = 16, > > + .reg_stride = 2, > > > > .max_register = SGTL5000_MAX_REG_OFFSET, > > .volatile_reg = sgtl5000_volatile, > > -- > > 1.8.1.2 > > Fabio has post same patch for this. > http://comments.gmane.org/gmane.linux.alsa.devel/109903 Oh. I didn't see that one, sorry. Fabio's patch is from July, 5th. It takes quite some time for this to hit mainline then.... Best regards, -- David Jander Protonic Holland.

2 1

[alsa-devel] [PATCH v11] ASoC: fsl: Add S/PDIF machine driver
by Nicolin Chen 27 Aug '13

27 Aug '13

This patch implements a device-tree-only machine driver for Freescale i.MX series Soc. It works with spdif_transmitter/spdif_receiver and fsl_spdif.c drivers. Signed-off-by: Nicolin Chen <b42378(a)freescale.com> --- Changelog v10->v11: * Use boolean properties for spdif-out/in switch instead of codec phandles. * Accordingly create dummy codec driver in probe() instead of DT nodes. .../devicetree/bindings/sound/imx-audio-spdif.txt | 34 +++++ sound/soc/fsl/Kconfig | 11 ++ sound/soc/fsl/Makefile | 2 + sound/soc/fsl/imx-spdif.c | 152 ++++++++++++++++++++ 4 files changed, 199 insertions(+), 0 deletions(-) create mode 100644 Documentation/devicetree/bindings/sound/imx-audio-spdif.txt create mode 100644 sound/soc/fsl/imx-spdif.c diff --git a/Documentation/devicetree/bindings/sound/imx-audio-spdif.txt b/Documentation/devicetree/bindings/sound/imx-audio-spdif.txt new file mode 100644 index 0000000..7d13479 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/imx-audio-spdif.txt @@ -0,0 +1,34 @@ +Freescale i.MX audio complex with S/PDIF transceiver + +Required properties: + + - compatible : "fsl,imx-audio-spdif" + + - model : The user-visible name of this sound complex + + - spdif-controller : The phandle of the i.MX S/PDIF controller + + +Optional properties: + + - spdif-out : This is a boolean property. If present, the transmitting + function of S/PDIF will be enabled, indicating there's a physical + S/PDIF out connector/jack on the board or it's connecting to some + other IP block, such as an HDMI encoder/display-controller. + + - spdif-in : This is a boolean property. If present, the receiving + function of S/PDIF will be enabled, indicating there's a physical + S/PDIF in connector/jack on the board. + +* Note: At least one of these two properties should be set in the DT binding. + + +Example: + +sound-spdif { + compatible = "fsl,imx-audio-spdif"; + model = "imx-spdif"; + spdif-controller = <&spdif>; + spdif-out; + spdif-in; +}; diff --git a/sound/soc/fsl/Kconfig b/sound/soc/fsl/Kconfig index cd088cc..a708380 100644 --- a/sound/soc/fsl/Kconfig +++ b/sound/soc/fsl/Kconfig @@ -193,6 +193,17 @@ config SND_SOC_IMX_SGTL5000 Say Y if you want to add support for SoC audio on an i.MX board with a sgtl5000 codec. +config SND_SOC_IMX_SPDIF + tristate "SoC Audio support for i.MX boards with S/PDIF" + select SND_SOC_IMX_PCM_DMA + select SND_SOC_FSL_SPDIF + select SND_SOC_FSL_UTILS + select SND_SOC_SPDIF + help + SoC Audio support for i.MX boards with S/PDIF + Say Y if you want to add support for SoC audio on an i.MX board with + a S/DPDIF. + config SND_SOC_IMX_MC13783 tristate "SoC Audio support for I.MX boards with mc13783" depends on MFD_MC13783 && ARM diff --git a/sound/soc/fsl/Makefile b/sound/soc/fsl/Makefile index 4b5970e..e2aaff7 100644 --- a/sound/soc/fsl/Makefile +++ b/sound/soc/fsl/Makefile @@ -45,6 +45,7 @@ snd-soc-mx27vis-aic32x4-objs := mx27vis-aic32x4.o snd-soc-wm1133-ev1-objs := wm1133-ev1.o snd-soc-imx-sgtl5000-objs := imx-sgtl5000.o snd-soc-imx-wm8962-objs := imx-wm8962.o +snd-soc-imx-spdif-objs :=imx-spdif.o snd-soc-imx-mc13783-objs := imx-mc13783.o obj-$(CONFIG_SND_SOC_EUKREA_TLV320) += snd-soc-eukrea-tlv320.o @@ -53,4 +54,5 @@ obj-$(CONFIG_SND_SOC_MX27VIS_AIC32X4) += snd-soc-mx27vis-aic32x4.o obj-$(CONFIG_SND_MXC_SOC_WM1133_EV1) += snd-soc-wm1133-ev1.o obj-$(CONFIG_SND_SOC_IMX_SGTL5000) += snd-soc-imx-sgtl5000.o obj-$(CONFIG_SND_SOC_IMX_WM8962) += snd-soc-imx-wm8962.o +obj-$(CONFIG_SND_SOC_IMX_SPDIF) += snd-soc-imx-spdif.o obj-$(CONFIG_SND_SOC_IMX_MC13783) += snd-soc-imx-mc13783.o diff --git a/sound/soc/fsl/imx-spdif.c b/sound/soc/fsl/imx-spdif.c new file mode 100644 index 0000000..af5b553 --- /dev/null +++ b/sound/soc/fsl/imx-spdif.c @@ -0,0 +1,152 @@ +/* + * Copyright (C) 2013 Freescale Semiconductor, Inc. + * + * The code contained herein is licensed under the GNU General Public + * License. You may obtain a copy of the GNU General Public License + * Version 2 or later at the following locations: + * + * http://www.opensource.org/licenses/gpl-license.html + * http://www.gnu.org/copyleft/gpl.html + */ + +#include <linux/module.h> +#include <linux/of_platform.h> +#include <sound/soc.h> + +struct imx_spdif_data { + struct snd_soc_dai_link dai[2]; + struct snd_soc_card card; + struct platform_device *txdev; + struct platform_device *rxdev; +}; + +static int imx_spdif_audio_probe(struct platform_device *pdev) +{ + struct device_node *spdif_np, *np = pdev->dev.of_node; + struct platform_device *spdif_pdev; + struct imx_spdif_data *data; + int ret = 0, num_links = 0; + + spdif_np = of_parse_phandle(np, "spdif-controller", 0); + if (!spdif_np) { + dev_err(&pdev->dev, "failed to find spdif-controller\n"); + ret = -EINVAL; + goto end; + } + + spdif_pdev = of_find_device_by_node(spdif_np); + if (!spdif_pdev) { + dev_err(&pdev->dev, "failed to find S/PDIF device\n"); + ret = -EINVAL; + goto end; + } + + data = devm_kzalloc(&pdev->dev, sizeof(*data), GFP_KERNEL); + if (!data) { + dev_err(&pdev->dev, "failed to allocate memory\n"); + ret = -ENOMEM; + goto end; + } + + if (of_property_read_bool(np, "spdif-out")) { + data->dai[num_links].name = "S/PDIF TX"; + data->dai[num_links].stream_name = "S/PDIF PCM Playback"; + data->dai[num_links].codec_dai_name = "dit-hifi"; + data->dai[num_links].codec_name = "spdif-dit"; + data->dai[num_links].cpu_of_node = spdif_np; + data->dai[num_links].platform_of_node = spdif_np; + num_links++; + } + + if (of_property_read_bool(np, "spdif-in")) { + data->dai[num_links].name = "S/PDIF RX"; + data->dai[num_links].stream_name = "S/PDIF PCM Capture"; + data->dai[num_links].codec_dai_name = "dir-hifi"; + data->dai[num_links].codec_name = "spdif-dir"; + data->dai[num_links].cpu_of_node = spdif_np; + data->dai[num_links].platform_of_node = spdif_np; + num_links++; + } + + if (!num_links) { + dev_err(&pdev->dev, "no enabled S/PDIF DAI link\n"); + goto end; + } + + data->card.dev = &pdev->dev; + data->card.num_links = num_links; + data->card.dai_link = data->dai; + + ret = snd_soc_of_parse_card_name(&data->card, "model"); + if (ret) + goto end; + + data->txdev = platform_device_register_simple("spdif-dit", -1, NULL, 0); + if (IS_ERR(data->txdev)) { + ret = PTR_ERR(data->txdev); + dev_err(&pdev->dev, "register spdif-dit failed (%d)\n", ret); + goto end; + } + + data->rxdev = platform_device_register_simple("spdif-dir", -1, NULL, 0); + if (IS_ERR(data->rxdev)) { + ret = PTR_ERR(data->rxdev); + dev_err(&pdev->dev, "register spdif-dir failed (%d)\n", ret); + goto error_dit; + } + + ret = snd_soc_register_card(&data->card); + if (ret) { + dev_err(&pdev->dev, "snd_soc_register_card failed (%d)\n", ret); + goto error_dir; + } + + platform_set_drvdata(pdev, data); + + goto end; + +error_dir: + platform_device_unregister(data->rxdev); +error_dit: + platform_device_unregister(data->txdev); +end: + if (spdif_np) + of_node_put(spdif_np); + + return ret; +} + +static int imx_spdif_audio_remove(struct platform_device *pdev) +{ + struct imx_spdif_data *data = platform_get_drvdata(pdev); + + platform_device_unregister(data->rxdev); + platform_device_unregister(data->txdev); + + snd_soc_unregister_card(&data->card); + + return 0; +} + +static const struct of_device_id imx_spdif_dt_ids[] = { + { .compatible = "fsl,imx-audio-spdif", }, + { /* sentinel */ } +}; +MODULE_DEVICE_TABLE(of, imx_spdif_dt_ids); + +static struct platform_driver imx_spdif_driver = { + .driver = { + .name = "imx-spdif", + .owner = THIS_MODULE, + .of_match_table = imx_spdif_dt_ids, + }, + .probe = imx_spdif_audio_probe, + .remove = imx_spdif_audio_remove, +}; + +module_platform_driver(imx_spdif_driver); + +MODULE_AUTHOR("Freescale Semiconductor, Inc."); +MODULE_DESCRIPTION("Freescale i.MX S/PDIF machine driver"); +MODULE_LICENSE("GPL v2"); +MODULE_ALIAS("platform:imx-spdif"); -- 1.7.1

4 6

[alsa-devel] [PATCH 1/8] ASoC: dapm: Fix marking widgets dirty when a route is added
by Lars-Peter Clausen 27 Aug '13

27 Aug '13

The current calls to dapm_mark_dirty() in snd_soc_dapm_add_path() are on a path that is only reached if the sink widget is either a mixer or a mux. Move the calls further up so they are called for all widget types. Signed-off-by: Lars-Peter Clausen <lars(a)metafoo.de> --- sound/soc/soc-dapm.c | 6 +++--- 1 file changed, 3 insertions(+), 3 deletions(-) diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c index d84bd0f..7e9afbc4 100644 --- a/sound/soc/soc-dapm.c +++ b/sound/soc/soc-dapm.c @@ -2374,6 +2374,9 @@ static int snd_soc_dapm_add_path(struct snd_soc_dapm_context *dapm, wsource->ext = 1; } + dapm_mark_dirty(wsource, "Route added"); + dapm_mark_dirty(wsink, "Route added"); + /* connect static paths */ if (control == NULL) { list_add(&path->list, &dapm->card->paths); @@ -2436,9 +2439,6 @@ static int snd_soc_dapm_add_path(struct snd_soc_dapm_context *dapm, return 0; } - dapm_mark_dirty(wsource, "Route added"); - dapm_mark_dirty(wsink, "Route added"); - return 0; err: kfree(path); -- 1.8.0

2 17

[alsa-devel] [PATCH 0/9] ALSA: compress offfload fixes
by Vinod Koul 27 Aug '13

27 Aug '13

Hi Takashi, Here is the patch series which fixes various issues being reported by users (out of tree sadly) The first three and and last one are marked to stable as would like these to be fixed in older kernels as well. It would be good if you can send them as fixes to linus for 3.11. Rest can go in the merge window Fixes: - using lock for all operation was a very bad idead. This is bad as some of the ioctls like drain, partial drain can be time consuming and thus prevent any other operation while these are ongoing like Pause, Stop or timestamp query, so fix this be removing bunch of ioctls not to use device lock. - Now we dont have lock for pointer updates so this maybe racy, so use lock for doing lowest level calculation. - As disscused on our sample rate problem, lets move to use rate values and I will fix the lib too. Since the driver are not upstream the impact of this change wont be huge. - Plus few fix like use snprintf, state chacks for pause, write etc.. Vinod Koul (9): ALSA: Compress - dont use lock for all ioctls ALSA: compress: use mutex in drain ASoC: compress: dont aquire lock for draining states ALSA: compress: use snprint instread of sprintf ALSA: compres: wakeup the poll thread on pause ALSA: compress: dont write when stream is paused ALSA: compress: allow write when stream is setup ALSA: compress: call pointer callback and updates under a lock ALSA: compress: use rate values for passing sampling rates include/sound/compress_driver.h | 2 + include/uapi/sound/compress_offload.h | 2 +- sound/core/compress_offload.c | 140 +++++++++++++++++++++++++------- sound/soc/soc-compress.c | 10 +++ 4 files changed, 122 insertions(+), 32 deletions(-) Thanks ~Vinod

3 39

[alsa-devel] hda: Is it ok to enable SND_HDA_I915 to support Haswell Gfx power well by default?
by Lin, Mengdong 27 Aug '13

27 Aug '13

Hi Takashi and David, Is it okay to enable SND_HDA_I915 by default? This would make it easy to upgrade kernel on Haswell. A user only needs to apply default configurations for the new kernel, and display audio can work automatically. The user need not have any idea about the display power well. Now the user needs to manually enable SND_HDA_I915, otherwise display audio won't work since Gfx driver may turn off the power well which make both the controller and codec lose power. Thanks Mengdong

3 4

[alsa-devel] [PATCH v2 1/4] ALSA: hda - add flags and routines to get devices selection info for DP1.2 MST
by mengdong.lin＠intel.com 27 Aug '13

27 Aug '13

From: Mengdong Lin <mengdong.lin(a)intel.com> This patch adds flags and routines to get device list & selection info on a pin. To support Display Port 1.2 multi-stream transport (MST) over single DP port, a pin can support multiple devices. Please refer to HD-A spec Document Change Notificaton HDA040-A. A display audio codec can set flag "dp_mst" in its patch, indicating its pins can support MST. But at runtime, a pin may not be multi-streaming capable and report the device list is empty, depending on Gfx driver configuration. Signed-off-by: Mengdong Lin <mengdong.lin(a)intel.com> diff --git a/sound/pci/hda/hda_codec.c b/sound/pci/hda/hda_codec.c index fdbb09a..2596e36 100644 --- a/sound/pci/hda/hda_codec.c +++ b/sound/pci/hda/hda_codec.c @@ -666,6 +666,66 @@ int snd_hda_get_conn_index(struct hda_codec *codec, hda_nid_t mux, } EXPORT_SYMBOL_HDA(snd_hda_get_conn_index); + +/* return DEVLIST_LEN parameter of the given widget */ +static unsigned int get_num_devices(struct hda_codec *codec, hda_nid_t nid) +{ + unsigned int wcaps = get_wcaps(codec, nid); + unsigned int parm; + + if (!codec->dp_mst || !(wcaps & AC_WCAP_DIGITAL) || + get_wcaps_type(wcaps) != AC_WID_PIN) + return 0; + + parm = snd_hda_param_read(codec, nid, AC_PAR_DEVLIST_LEN); + if (parm == -1 && codec->bus->rirb_error) + parm = 0; + return parm & AC_DEVLIST_LEN; +} + +/** + * snd_hda_get_devices - copy device list without cache + * @codec: the HDA codec + * @nid: NID of the pin to parse + * @dev_list: device list array + * @max_devices: max. number of devices to store + * + * Copy the device list. This info is dynamic and so not cached. + * Currently called only from hda_proc.c, so not exported. + */ +int snd_hda_get_devices(struct hda_codec *codec, hda_nid_t nid, + struct hda_devlist_entry *dev_list, int max_devices) +{ + unsigned int parm; + int i, dev_len, devices; + + parm = get_num_devices(codec, nid); + if (!parm) /* not multi-stream capable */ + return 0; + + dev_len = parm + 1; + dev_len = dev_len < max_devices ? dev_len : max_devices; + + devices = 0; + while (devices < dev_len) { + parm = snd_hda_codec_read(codec, nid, 0, + AC_VERB_GET_DEVICE_LIST, devices); + if (parm == -1 && codec->bus->rirb_error) + break; + + for (i = 0; i < 8; i++) { + dev_list[devices].present_detect = !!(parm & AC_DE_PD); + dev_list[devices].eld_valid = !!(parm & AC_DE_ELDV); + dev_list[devices].inactive = !!(parm & AC_DE_IA); + parm >>= 4; + devices++; + if (devices >= dev_len) + break; + } + } + return devices; +} + /** * snd_hda_queue_unsol_event - add an unsolicited event to queue * @bus: the BUS diff --git a/sound/pci/hda/hda_codec.h b/sound/pci/hda/hda_codec.h index 701c2e0..316076e 100644 --- a/sound/pci/hda/hda_codec.h +++ b/sound/pci/hda/hda_codec.h @@ -94,6 +94,8 @@ enum { #define AC_VERB_GET_HDMI_DIP_XMIT 0x0f32 #define AC_VERB_GET_HDMI_CP_CTRL 0x0f33 #define AC_VERB_GET_HDMI_CHAN_SLOT 0x0f34 +#define AC_VERB_GET_DEVICE_SEL 0xf35 +#define AC_VERB_GET_DEVICE_LIST 0xf36 /* * SET verbs @@ -133,6 +135,7 @@ enum { #define AC_VERB_SET_HDMI_DIP_XMIT 0x732 #define AC_VERB_SET_HDMI_CP_CTRL 0x733 #define AC_VERB_SET_HDMI_CHAN_SLOT 0x734 +#define AC_VERB_SET_DEVICE_SEL 0x735 /* * Parameter IDs @@ -154,6 +157,7 @@ enum { #define AC_PAR_GPIO_CAP 0x11 #define AC_PAR_AMP_OUT_CAP 0x12 #define AC_PAR_VOL_KNB_CAP 0x13 +#define AC_PAR_DEVLIST_LEN 0x15 #define AC_PAR_HDMI_LPCM_CAP 0x20 /* @@ -460,6 +464,11 @@ enum { #define AC_DEFCFG_PORT_CONN (0x3<<30) #define AC_DEFCFG_PORT_CONN_SHIFT 30 +/* Display pin's device list entry */ +#define AC_DE_PD (1<<0) +#define AC_DE_ELDV (1<<1) +#define AC_DE_IA (1<<2) + /* device device types (0x0-0xf) */ enum { AC_JACK_LINE_OUT, @@ -795,6 +804,16 @@ struct hda_pcm { bool own_chmap; /* codec driver provides own channel maps */ }; +/* display pin's device list entry */ +struct hda_devlist_entry { + unsigned int present_detect:1; + unsigned int eld_valid:1; + unsigned int inactive:1; +}; + +#define AC_MAX_DEVLIST_LEN 64 +#define AC_DEVLIST_LEN 0x3f + /* codec information */ struct hda_codec { struct hda_bus *bus; @@ -885,6 +904,7 @@ struct hda_codec { unsigned int pcm_format_first:1; /* PCM format must be set first */ unsigned int epss:1; /* supporting EPSS? */ unsigned int cached_write:1; /* write only to caches */ + unsigned int dp_mst:1; /* support DP1.2 Multi-stream transport */ #ifdef CONFIG_PM unsigned int power_on :1; /* current (global) power-state */ unsigned int d3_stop_clk:1; /* support D3 operation without BCLK */ @@ -972,6 +992,8 @@ int snd_hda_override_conn_list(struct hda_codec *codec, hda_nid_t nid, int nums, const hda_nid_t *list); int snd_hda_get_conn_index(struct hda_codec *codec, hda_nid_t mux, hda_nid_t nid, int recursive); +int snd_hda_get_devices(struct hda_codec *codec, hda_nid_t nid, + struct hda_devlist_entry *dev_list, int max_devices); int snd_hda_query_supported_pcm(struct hda_codec *codec, hda_nid_t nid, u32 *ratesp, u64 *formatsp, unsigned int *bpsp); -- 1.8.1.2

3 5

[alsa-devel] Fw: trouble with AMD Hudson Azalia sound card, no sound, alsamixer installed but not found, no /etc/asound.conf file
by Julie Brunias 27 Aug '13

27 Aug '13

hello i contact the Alsa Development Team because I face a problem with sound under Ubuntu. My sound card AMD Hudson Azalia is not seen by phonon and alsa cannot play any sound. But I do see my sound card in lspci. I tried many stuff found on forums but still not working... Would you be so kind and help me to figure out what's wrong? My ubuntu is 12.04 with alsa-base and alsa-utils. I cannot access to alsamixer (even with root) "not found", and cannot access to cat /etc/asound.conf "not found". Everything in my Bios is enabled. Here is an output I made. http://www.alsa-project.org/db/?f=82c2830fb9e05e618677523a179726de01569667 Until now (since 10 years) I never had such problem. My kernel is ok, i already uninstalled and purged alsa-utils and reinstall the whole thing as described on a ubuntu forum, not solving the issue. That would be very cool from you if you could help. I saw many users on the net with this s**** sound card who face problems. Apparently only on Ubuntu. If I am not at the right place to ask for some help, would you mind to orient me to the right people? I receive no answer to my post on Ubuntu Forums and I hope you could help. Best regards and thx for all the good work! Tomski

1 0

[alsa-devel] [trivial PATCH] treewide: Fix printks with 0x%#
by Joe Perches 27 Aug '13

27 Aug '13

Using 0x%# emits 0x0x. Only one is necessary. Signed-off-by: Joe Perches <joe(a)perches.com> --- arch/parisc/kernel/signal.c | 2 +- drivers/clocksource/acpi_pm.c | 4 ++-- drivers/net/ethernet/sis/sis900.c | 2 +- mm/memory-failure.c | 2 +- sound/pci/ens1370.c | 2 +- sound/pci/via82xx.c | 2 +- 6 files changed, 7 insertions(+), 7 deletions(-) diff --git a/arch/parisc/kernel/signal.c b/arch/parisc/kernel/signal.c index 940188d..35c5bf1 100644 --- a/arch/parisc/kernel/signal.c +++ b/arch/parisc/kernel/signal.c @@ -85,7 +85,7 @@ restore_sigcontext(struct sigcontext __user *sc, struct pt_regs *regs) err |= __copy_from_user(regs->iaoq, sc->sc_iaoq, sizeof(regs->iaoq)); err |= __copy_from_user(regs->iasq, sc->sc_iasq, sizeof(regs->iasq)); err |= __get_user(regs->sar, &sc->sc_sar); - DBG(2,"restore_sigcontext: iaoq is 0x%#lx / 0x%#lx\n", + DBG(2,"restore_sigcontext: iaoq is %#lx / %#lx\n", regs->iaoq[0],regs->iaoq[1]); DBG(2,"restore_sigcontext: r28 is %ld\n", regs->gr[28]); return err; diff --git a/drivers/clocksource/acpi_pm.c b/drivers/clocksource/acpi_pm.c index 6efe4d1..6eab889 100644 --- a/drivers/clocksource/acpi_pm.c +++ b/drivers/clocksource/acpi_pm.c @@ -200,14 +200,14 @@ static int __init init_acpi_pm_clocksource(void) if ((value2 < value1) && ((value2) < 0xFFF)) break; printk(KERN_INFO "PM-Timer had inconsistent results:" - " 0x%#llx, 0x%#llx - aborting.\n", + " %#llx, %#llx - aborting.\n", value1, value2); pmtmr_ioport = 0; return -EINVAL; } if (i == ACPI_PM_READ_CHECKS) { printk(KERN_INFO "PM-Timer failed consistency check " - " (0x%#llx) - aborting.\n", value1); + " (%#llx) - aborting.\n", value1); pmtmr_ioport = 0; return -ENODEV; } diff --git a/drivers/net/ethernet/sis/sis900.c b/drivers/net/ethernet/sis/sis900.c index eb4aea3..6c1e34c 100644 --- a/drivers/net/ethernet/sis/sis900.c +++ b/drivers/net/ethernet/sis/sis900.c @@ -1723,7 +1723,7 @@ static irqreturn_t sis900_interrupt(int irq, void *dev_instance) if(netif_msg_intr(sis_priv)) printk(KERN_DEBUG "%s: exiting interrupt, " - "interrupt status = 0x%#8.8x.\n", + "interrupt status = %#8.8x\n", net_dev->name, sr32(isr)); spin_unlock (&sis_priv->lock); diff --git a/mm/memory-failure.c b/mm/memory-failure.c index 09ae111..29d3f38 100644 --- a/mm/memory-failure.c +++ b/mm/memory-failure.c @@ -1267,7 +1267,7 @@ void memory_failure_queue(unsigned long pfn, int trapno, int flags) if (kfifo_put(&mf_cpu->fifo, &entry)) schedule_work_on(smp_processor_id(), &mf_cpu->work); else - pr_err("Memory failure: buffer overflow when queuing memory failure at 0x%#lx\n", + pr_err("Memory failure: buffer overflow when queuing memory failure at %#lx\n", pfn); spin_unlock_irqrestore(&mf_cpu->lock, proc_flags); put_cpu_var(memory_failure_cpu); diff --git a/sound/pci/ens1370.c b/sound/pci/ens1370.c index ca8929b..61262f3 100644 --- a/sound/pci/ens1370.c +++ b/sound/pci/ens1370.c @@ -1842,7 +1842,7 @@ static int snd_ensoniq_create_gameport(struct ensoniq *ensoniq, int dev) default: if (!request_region(io_port, 8, "ens137x: gameport")) { - printk(KERN_WARNING "ens137x: gameport io port 0x%#x in use\n", + printk(KERN_WARNING "ens137x: gameport io port %#x in use\n", io_port); return -EBUSY; } diff --git a/sound/pci/via82xx.c b/sound/pci/via82xx.c index 3c511d0..5ae6f04 100644 --- a/sound/pci/via82xx.c +++ b/sound/pci/via82xx.c @@ -1940,7 +1940,7 @@ static int snd_via686_create_gameport(struct via82xx *chip, unsigned char *legac r = request_region(JOYSTICK_ADDR, 8, "VIA686 gameport"); if (!r) { - printk(KERN_WARNING "via82xx: cannot reserve joystick port 0x%#x\n", + printk(KERN_WARNING "via82xx: cannot reserve joystick port %#x\n", JOYSTICK_ADDR); return -EBUSY; }

3 3

[alsa-devel] [PATCH v2 1/2] ASoC: fsl_spdif: Remove unnecessary dev_set_drvdata()
by Fabio Estevam 27 Aug '13

27 Aug '13

From: Fabio Estevam <fabio.estevam(a)freescale.com> Driver core clears the driver data to NULL after device_release or on probe failure, so just remove it from here. Signed-off-by: Fabio Estevam <fabio.estevam(a)freescale.com> --- Changes since v1: - Also remove dev_set_drvdata from the remove function sound/soc/fsl/fsl_spdif.c | 5 +---- 1 file changed, 1 insertion(+), 4 deletions(-) diff --git a/sound/soc/fsl/fsl_spdif.c b/sound/soc/fsl/fsl_spdif.c index 42a4382..a8ef46a 100644 --- a/sound/soc/fsl/fsl_spdif.c +++ b/sound/soc/fsl/fsl_spdif.c @@ -1184,7 +1184,7 @@ static int fsl_spdif_probe(struct platform_device *pdev) &spdif_priv->cpu_dai_drv, 1); if (ret) { dev_err(&pdev->dev, "failed to register DAI: %d\n", ret); - goto error_dev; + return ret; } ret = imx_pcm_dma_init(pdev); @@ -1197,8 +1197,6 @@ static int fsl_spdif_probe(struct platform_device *pdev) error_component: snd_soc_unregister_component(&pdev->dev); -error_dev: - dev_set_drvdata(&pdev->dev, NULL); return ret; } @@ -1207,7 +1205,6 @@ static int fsl_spdif_remove(struct platform_device *pdev) { imx_pcm_dma_exit(pdev); snd_soc_unregister_component(&pdev->dev); - dev_set_drvdata(&pdev->dev, NULL); return 0; } -- 1.8.1.2

3 6

[alsa-devel] [PATCH v3 4/4] ALSA: hda - add device entry and inactive flag to unsolicited response
by mengdong.lin＠intel.com 27 Aug '13

27 Aug '13

From: Mengdong Lin <mengdong.lin(a)intel.com> This patch adds two fields to unsolicited response, according to spec HDA040-A: - Device Entry (bit 20:15) - Inactive (bit 2) and show the info in debug message. Signed-off-by: Mengdong Lin <mengdong.lin(a)intel.com> diff --git a/sound/pci/hda/hda_codec.h b/sound/pci/hda/hda_codec.h index b838c70..7aa9870 100644 --- a/sound/pci/hda/hda_codec.h +++ b/sound/pci/hda/hda_codec.h @@ -255,6 +255,11 @@ enum { #define AC_UNSOL_RES_TAG_SHIFT 26 #define AC_UNSOL_RES_SUBTAG (0x1f<<21) #define AC_UNSOL_RES_SUBTAG_SHIFT 21 +#define AC_UNSOL_RES_DE (0x3f<<15) /* Device Entry + * (for DP1.2 MST) + */ +#define AC_UNSOL_RES_DE_SHIFT 15 +#define AC_UNSOL_RES_IA (1<<2) /* Inactive (for DP1.2 MST) */ #define AC_UNSOL_RES_ELDV (1<<1) /* ELD Data valid (for HDMI) */ #define AC_UNSOL_RES_PD (1<<0) /* pinsense detect */ #define AC_UNSOL_RES_CP_STATE (1<<1) /* content protection */ diff --git a/sound/pci/hda/patch_hdmi.c b/sound/pci/hda/patch_hdmi.c index 87ca984..895a0d3 100644 --- a/sound/pci/hda/patch_hdmi.c +++ b/sound/pci/hda/patch_hdmi.c @@ -959,6 +959,7 @@ static void hdmi_intrinsic_event(struct hda_codec *codec, unsigned int res) int pin_nid; int pin_idx; struct hda_jack_tbl *jack; + int dev_entry = (res & AC_UNSOL_RES_DE) >> AC_UNSOL_RES_DE_SHIFT; jack = snd_hda_jack_tbl_get_from_tag(codec, tag); if (!jack) @@ -967,8 +968,8 @@ static void hdmi_intrinsic_event(struct hda_codec *codec, unsigned int res) jack->jack_dirty = 1; _snd_printd(SND_PR_VERBOSE, - "HDMI hot plug event: Codec=%d Pin=%d Presence_Detect=%d ELD_Valid=%d\n", - codec->addr, pin_nid, + "HDMI hot plug event: Codec=%d Pin=%d Device=%d Inactive=%d Presence_Detect=%d ELD_Valid=%d\n", + codec->addr, pin_nid, dev_entry, !!(res & AC_UNSOL_RES_IA), !!(res & AC_UNSOL_RES_PD), !!(res & AC_UNSOL_RES_ELDV)); pin_idx = pin_nid_to_pin_index(spec, pin_nid); -- 1.8.1.2

1 0

[alsa-devel] [PATCH v3 3/4] ALSA: hda - Haswell codec exposes device list/select info on pins
by mengdong.lin＠intel.com 27 Aug '13

27 Aug '13

From: Mengdong Lin <mengdong.lin(a)intel.com> This patch is only to allow codec proc file to expose devices list/select info for Haswell codec pins. Since Haswell Gfx driver cannot support DP1.2 MST now, so all pins' device list is empty, meaning no pin is multi-streaming capaple. Signed-off-by: Mengdong Lin <mengdong.lin(a)intel.com> diff --git a/sound/pci/hda/patch_hdmi.c b/sound/pci/hda/patch_hdmi.c index 030ca86..87ca984 100644 --- a/sound/pci/hda/patch_hdmi.c +++ b/sound/pci/hda/patch_hdmi.c @@ -1989,8 +1989,10 @@ static int patch_generic_hdmi(struct hda_codec *codec) return -EINVAL; } codec->patch_ops = generic_hdmi_patch_ops; - if (codec->vendor_id == 0x80862807) + if (codec->vendor_id == 0x80862807) { codec->patch_ops.set_power_state = haswell_set_power_state; + codec->dp_mst = true; + } generic_hdmi_init_per_pins(codec); -- 1.8.1.2

1 0

[alsa-devel] [PATCH v3 2/4] ALSA: hda - add device list & select info of display pins to codec proc file
by mengdong.lin＠intel.com 27 Aug '13

27 Aug '13

From: Mengdong Lin <mengdong.lin(a)intel.com> If a display codec supports multi-stream transport on the pins, the pin's device list length and device entries will be exposed to codec proc file. Signed-off-by: Mengdong Lin <mengdong.lin(a)intel.com> diff --git a/sound/pci/hda/hda_proc.c b/sound/pci/hda/hda_proc.c index 9760f00..a8cb22e 100644 --- a/sound/pci/hda/hda_proc.c +++ b/sound/pci/hda/hda_proc.c @@ -582,6 +582,36 @@ static void print_gpio(struct snd_info_buffer *buffer, print_nid_array(buffer, codec, nid, &codec->nids); } +static void print_device_list(struct snd_info_buffer *buffer, + struct hda_codec *codec, hda_nid_t nid) +{ + int i, curr = -1; + u8 dev_list[AC_MAX_DEV_LIST_LEN]; + int devlist_len; + + devlist_len = snd_hda_get_devices(codec, nid, dev_list, + AC_MAX_DEV_LIST_LEN); + snd_iprintf(buffer, " Devices: %d\n", devlist_len); + if (devlist_len <= 0) + return; + + curr = snd_hda_codec_read(codec, nid, 0, + AC_VERB_GET_DEVICE_SEL, 0); + + for (i = 0; i < devlist_len; i++) { + if (i == curr) + snd_iprintf(buffer, " *"); + else + snd_iprintf(buffer, " "); + + snd_iprintf(buffer, + "Dev %02d: PD = %d, ELDV = %d, IA = %d\n", i, + !!(dev_list[i] & AC_DE_PD), + !!(dev_list[i] & AC_DE_ELDV), + !!(dev_list[i] & AC_DE_IA)); + } +} + static void print_codec_info(struct snd_info_entry *entry, struct snd_info_buffer *buffer) { @@ -751,6 +781,9 @@ static void print_codec_info(struct snd_info_entry *entry, (wid_caps & AC_WCAP_DELAY) >> AC_WCAP_DELAY_SHIFT); + if (wid_type == AC_WID_PIN && codec->dp_mst) + print_device_list(buffer, codec, nid); + if (wid_caps & AC_WCAP_CONN_LIST) print_conn_list(buffer, codec, nid, wid_type, conn, conn_len); -- 1.8.1.2

1 0

[alsa-devel] [PATCH v3 1/4] ALSA: hda - add flags and routines to get devices selection info for DP1.2 MST
by mengdong.lin＠intel.com 27 Aug '13

27 Aug '13

From: Mengdong Lin <mengdong.lin(a)intel.com> This patch adds flags and routines to get device list & selection info on a pin. To support Display Port 1.2 multi-stream transport (MST) over single DP port, a pin can support multiple devices. Please refer to HD-A spec Document Change Notificaton HDA040-A. A display audio codec can set flag "dp_mst" in its patch, indicating its pins can support MST. But at runtime, a pin may not be multi-streaming capable and report the device list is empty, depending on Gfx driver configuration. Signed-off-by: Mengdong Lin <mengdong.lin(a)intel.com> diff --git a/sound/pci/hda/hda_codec.c b/sound/pci/hda/hda_codec.c index fdbb09a..5b6c4e3 100644 --- a/sound/pci/hda/hda_codec.c +++ b/sound/pci/hda/hda_codec.c @@ -666,6 +666,64 @@ int snd_hda_get_conn_index(struct hda_codec *codec, hda_nid_t mux, } EXPORT_SYMBOL_HDA(snd_hda_get_conn_index); + +/* return DEVLIST_LEN parameter of the given widget */ +static unsigned int get_num_devices(struct hda_codec *codec, hda_nid_t nid) +{ + unsigned int wcaps = get_wcaps(codec, nid); + unsigned int parm; + + if (!codec->dp_mst || !(wcaps & AC_WCAP_DIGITAL) || + get_wcaps_type(wcaps) != AC_WID_PIN) + return 0; + + parm = snd_hda_param_read(codec, nid, AC_PAR_DEVLIST_LEN); + if (parm == -1 && codec->bus->rirb_error) + parm = 0; + return parm & AC_DEV_LIST_LEN_MASK; +} + +/** + * snd_hda_get_devices - copy device list without cache + * @codec: the HDA codec + * @nid: NID of the pin to parse + * @dev_list: device list array + * @max_devices: max. number of devices to store + * + * Copy the device list. This info is dynamic and so not cached. + * Currently called only from hda_proc.c, so not exported. + */ +int snd_hda_get_devices(struct hda_codec *codec, hda_nid_t nid, + u8 *dev_list, int max_devices) +{ + unsigned int parm; + int i, dev_len, devices; + + parm = get_num_devices(codec, nid); + if (!parm) /* not multi-stream capable */ + return 0; + + dev_len = parm + 1; + dev_len = dev_len < max_devices ? dev_len : max_devices; + + devices = 0; + while (devices < dev_len) { + parm = snd_hda_codec_read(codec, nid, 0, + AC_VERB_GET_DEVICE_LIST, devices); + if (parm == -1 && codec->bus->rirb_error) + break; + + for (i = 0; i < 8; i++) { + dev_list[devices] = (u8)parm; + parm >>= 4; + devices++; + if (devices >= dev_len) + break; + } + } + return devices; +} + /** * snd_hda_queue_unsol_event - add an unsolicited event to queue * @bus: the BUS diff --git a/sound/pci/hda/hda_codec.h b/sound/pci/hda/hda_codec.h index 701c2e0..b838c70 100644 --- a/sound/pci/hda/hda_codec.h +++ b/sound/pci/hda/hda_codec.h @@ -94,6 +94,8 @@ enum { #define AC_VERB_GET_HDMI_DIP_XMIT 0x0f32 #define AC_VERB_GET_HDMI_CP_CTRL 0x0f33 #define AC_VERB_GET_HDMI_CHAN_SLOT 0x0f34 +#define AC_VERB_GET_DEVICE_SEL 0xf35 +#define AC_VERB_GET_DEVICE_LIST 0xf36 /* * SET verbs @@ -133,6 +135,7 @@ enum { #define AC_VERB_SET_HDMI_DIP_XMIT 0x732 #define AC_VERB_SET_HDMI_CP_CTRL 0x733 #define AC_VERB_SET_HDMI_CHAN_SLOT 0x734 +#define AC_VERB_SET_DEVICE_SEL 0x735 /* * Parameter IDs @@ -154,6 +157,7 @@ enum { #define AC_PAR_GPIO_CAP 0x11 #define AC_PAR_AMP_OUT_CAP 0x12 #define AC_PAR_VOL_KNB_CAP 0x13 +#define AC_PAR_DEVLIST_LEN 0x15 #define AC_PAR_HDMI_LPCM_CAP 0x20 /* @@ -352,6 +356,10 @@ enum { #define AC_LPCMCAP_44K (1<<30) /* 44.1kHz support */ #define AC_LPCMCAP_44K_MS (1<<31) /* 44.1kHz-multiplies support */ +/* Display pin's device list length */ +#define AC_DEV_LIST_LEN_MASK 0x3f +#define AC_MAX_DEV_LIST_LEN 64 + /* * Control Parameters */ @@ -460,6 +468,11 @@ enum { #define AC_DEFCFG_PORT_CONN (0x3<<30) #define AC_DEFCFG_PORT_CONN_SHIFT 30 +/* Display pin's device list entry */ +#define AC_DE_PD (1<<0) +#define AC_DE_ELDV (1<<1) +#define AC_DE_IA (1<<2) + /* device device types (0x0-0xf) */ enum { AC_JACK_LINE_OUT, @@ -885,6 +898,7 @@ struct hda_codec { unsigned int pcm_format_first:1; /* PCM format must be set first */ unsigned int epss:1; /* supporting EPSS? */ unsigned int cached_write:1; /* write only to caches */ + unsigned int dp_mst:1; /* support DP1.2 Multi-stream transport */ #ifdef CONFIG_PM unsigned int power_on :1; /* current (global) power-state */ unsigned int d3_stop_clk:1; /* support D3 operation without BCLK */ @@ -972,6 +986,8 @@ int snd_hda_override_conn_list(struct hda_codec *codec, hda_nid_t nid, int nums, const hda_nid_t *list); int snd_hda_get_conn_index(struct hda_codec *codec, hda_nid_t mux, hda_nid_t nid, int recursive); +int snd_hda_get_devices(struct hda_codec *codec, hda_nid_t nid, + u8 *dev_list, int max_devices); int snd_hda_query_supported_pcm(struct hda_codec *codec, hda_nid_t nid, u32 *ratesp, u64 *formatsp, unsigned int *bpsp); -- 1.8.1.2

1 0

[alsa-devel] Support new HDA codec
by Jorge Fernandez Monteagudo 26 Aug '13

26 Aug '13

Hi all, This is my first email to the devel mailing list. I'm asking the same question in the user mailing list but I suspect that I'll have to add some code in order to get this hda codec from Wolfson to work. Attached is the log I get from alsa-info.sh. This codec has three output paths: 1) Node 0x06, DAC1, and Node 0x11, Port A, the stereo headphone playback, which is the default one and it seems it works because I can see the stream set to 5 when a sound is played. 2) Node 0x04, SPDIF Tx1, and Node 0x0f, SPDIF OUT, the SPDIF transmit functionality provides an IEC-60958-3 compatible S/PIDF output which I would like to enable but without luck yet. I can enable the node 0x04, with hda-verb, and the spdif amplifier receives spdif frames but without audio info. 3) Node 0x07, DAC2, and Node 0x12, Port G, the stereo line playback with a differential stereo output wich I would like to enable but without luck again. I just can mute/unmute the node but without effect in the outputs. I would like to know if I have to develop a patch for this device or with the attached log info I can expect to use it with the current code in kernel. Now, with alsamixer and other config tools I can't see any control related to the outputs for 2) and 3)... Any hint is welcome! Thanks Jorge --- upload=true&script=true&cardinfo= !!################################ !!ALSA Information Script v 0.4.60 !!################################ !!Script ran on: Tue Feb 21 09:44:41 UTC 2012 !!Linux Distribution !!------------------ Ubuntu 11.10 \n \l DISTRIB_ID=Ubuntu DISTRIB_DESCRIPTION="Ubuntu 11.10" !!DMI Information !!--------------- Manufacturer: nVidia Product Name: MMP9-ION Product Version: To Be Filled By O.E.M. !!Kernel Information !!------------------ Kernel release: 3.0.0-12-generic Operating System: GNU/Linux Architecture: i686 Processor: i686 SMP Enabled: Yes !!ALSA Version !!------------ Driver version: 1.0.24 Library version: 1.0.24.1 Utilities version: 1.0.24.2 !!Loaded ALSA modules !!------------------- snd_hda_intel !!Sound Servers on this system !!---------------------------- Pulseaudio: Installed - Yes (/usr/bin/pulseaudio) Running - Yes ESound Daemon: Installed - Yes (/usr/bin/esd) Running - No !!Soundcards recognised by ALSA !!----------------------------- 0 [NVidia ]: HDA-Intel - HDA NVidia HDA NVidia at 0xfae78000 irq 20 !!PCI Soundcards installed in the system !!-------------------------------------- 00:08.0 Audio device: nVidia Corporation MCP79 High Definition Audio (rev b1) !!Advanced information - PCI Vendor/Device/Subsystem ID's !!-------------------------------------------------------- 00:08.0 0403: 10de:0ac0 (rev b1) Subsystem: 10de:cb84 !!Modprobe options (Sound related) !!-------------------------------- snd-atiixp-modem: index=-2 snd-intel8x0m: index=-2 snd-via82xx-modem: index=-2 snd-usb-audio: index=-2 snd-usb-caiaq: index=-2 snd-usb-ua101: index=-2 snd-usb-us122l: index=-2 snd-usb-usx2y: index=-2 snd-cmipci: mpu_port=0x330 fm_port=0x388 snd-pcsp: index=-2 snd-usb-audio: index=-2 !!Loaded sound module options !!-------------------------- !!Module: snd_hda_intel bdl_pos_adj : 32,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1 beep_mode : 0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0 enable : Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y enable_msi : -1 id : (null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null) index : -1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1 model : (null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null) patch : (null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null) position_fix : 0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0 power_save : 0 power_save_controller : Y probe_mask : -1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1 probe_only : 0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0 single_cmd : N !!HDA-Intel Codec information !!--------------------------- --startcollapse-- Codec: Wolfson Microelectronics ID 8800 Address: 0 AFG Function Id: 0x1 (unsol 1) Vendor Id: 0x1aec8800 Subsystem Id: 0x00000100 Revision Id: 0x100301 No Modem Function Group found Default PCM: rates [0x0]: bits [0x0]: formats [0x0]: Default Amp-In caps: ofs=0x00, nsteps=0x03, stepsize=0x27, mute=0 Default Amp-Out caps: ofs=0x7f, nsteps=0x7f, stepsize=0x01, mute=1 GPIO: io=2, o=0, i=0, unsolicited=1, wake=1 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=1, unsol=0 Node 0x02 [Audio Input] wcaps 0x1005d1: Stereo Device: name="HDA Generic", type="Audio", device=0 Converter: stream=1, channel=0 SDI-Select: 0 PCM: rates [0x1ff]: 8000 11025 16000 22050 32000 44100 48000 88200 96000 bits [0x1e]: 16 20 24 32 formats [0x3]: PCM FLOAT Unsolicited: tag=02, enabled=0 Power states: D0 D1 D2 D3 Power: setting=D0, actual=D0 Connection: 1 0x0a Processing caps: benign=0, ncoeff=0 Node 0x03 [Audio Input] wcaps 0x1005d1: Stereo Converter: stream=0, channel=0 SDI-Select: 0 PCM: rates [0x7f]: 8000 11025 16000 22050 32000 44100 48000 bits [0x1e]: 16 20 24 32 formats [0x3]: PCM FLOAT Unsolicited: tag=03, enabled=0 Power states: D0 D1 D2 D3 Power: setting=D0, actual=D0 Connection: 1 0x09 Processing caps: benign=0, ncoeff=0 Node 0x04 [Audio Output] wcaps 0x2d1: Stereo Digital Converter: stream=0, channel=0 Digital: Digital category: 0x0 PCM: rates [0x7f0]: 32000 44100 48000 88200 96000 176400 192000 bits [0x1e]: 16 20 24 32 formats [0x3]: PCM FLOAT Unsolicited: tag=04, enabled=0 Processing caps: benign=0, ncoeff=0 Node 0x05 [Audio Input] wcaps 0x1003d1: Stereo Digital Converter: stream=0, channel=0 SDI-Select: 0 Digital: Copyright Digital category: 0x0 PCM: rates [0x1f0]: 32000 44100 48000 88200 96000 bits [0x1e]: 16 20 24 32 formats [0x3]: PCM FLOAT Unsolicited: tag=05, enabled=0 Connection: 1 0x10 Processing caps: benign=0, ncoeff=0 Node 0x06 [Audio Output] wcaps 0x495: Stereo Amp-Out Control: name="Master Playback Switch", index=0, device=0 ControlAmp: chs=3, dir=Out, idx=0, ofs=0 Control: name="Master Playback Volume", index=0, device=0 ControlAmp: chs=3, dir=Out, idx=0, ofs=0 Device: name="HDA Generic", type="Audio", device=0 Amp-Out caps: N/A Amp-Out vals: [0x57 0x57] Converter: stream=5, channel=0 PCM: rates [0x7ff]: 8000 11025 16000 22050 32000 44100 48000 88200 96000 176400 192000 bits [0x1e]: 16 20 24 32 formats [0x3]: PCM FLOAT Unsolicited: tag=06, enabled=0 Power states: D0 D1 D2 D3 Power: setting=D0, actual=D0 Node 0x07 [Audio Output] wcaps 0x495: Stereo Amp-Out Amp-Out caps: N/A Amp-Out vals: [0xff 0xff] Converter: stream=0, channel=0 PCM: rates [0x7ff]: 8000 11025 16000 22050 32000 44100 48000 88200 96000 176400 192000 bits [0x1e]: 16 20 24 32 formats [0x3]: PCM FLOAT Unsolicited: tag=07, enabled=0 Power states: D0 D1 D2 D3 Power: setting=D0, actual=D0 Node 0x08 [Audio Output] wcaps 0x495: Stereo Amp-Out Amp-Out caps: N/A Amp-Out vals: [0xff 0xff] Converter: stream=0, channel=0 PCM: rates [0x7ff]: 8000 11025 16000 22050 32000 44100 48000 88200 96000 176400 192000 bits [0x1e]: 16 20 24 32 formats [0x3]: PCM FLOAT Unsolicited: tag=08, enabled=0 Power states: D0 D1 D2 D3 Power: setting=D0, actual=D0 Node 0x09 [Audio Selector] wcaps 0x300101: Stereo Connection: 2 0x0b* 0x0e Node 0x0a [Audio Selector] wcaps 0x30010d: Stereo Amp-Out Amp-Out caps: ofs=0x18, nsteps=0x30, stepsize=0x01, mute=1 Amp-Out vals: [0x18 0x18] Connection: 1 0x0c Node 0x0b [Audio Selector] wcaps 0x30010d: Stereo Amp-Out Amp-Out caps: ofs=0x18, nsteps=0x30, stepsize=0x01, mute=1 Amp-Out vals: [0x98 0x98] Connection: 1 0x0d Node 0x0c [Pin Complex] wcaps 0x400081: Stereo Pincap 0x00000064: IN Detect Balanced Pin Default 0x018130f0: [Jack] Line In at Ext Rear Conn = 1/8, Color = Blue DefAssociation = 0xf, Sequence = 0x0 Pin-ctls: 0x20: IN Unsolicited: tag=0c, enabled=0 Node 0x0d [Pin Complex] wcaps 0x400083: Stereo Amp-In Amp-In caps: N/A Amp-In vals: [0x00 0x00] Pincap 0x00001767: IN Detect Balanced Trigger ImpSense Vref caps: HIZ 50 GRD 80 Pin Default 0x02a190f0: [Jack] Mic at Ext Front Conn = 1/8, Color = Pink DefAssociation = 0xf, Sequence = 0x0 Pin-ctls: 0x20: IN VREF_HIZ Unsolicited: tag=0d, enabled=0 Node 0x0e [Pin Complex] wcaps 0x40020b: Stereo Digital Amp-In Amp-In caps: ofs=0x18, nsteps=0x58, stepsize=0x01, mute=1 Amp-In vals: [0x18 0x18] Pincap 0x00000020: IN Pin Default 0x10a600f0: [Jack] Mic at Int N/A Conn = Digital, Color = Unknown DefAssociation = 0xf, Sequence = 0x0 Pin-ctls: 0x20: IN Node 0x0f [Pin Complex] wcaps 0x400381: Stereo Digital Pincap 0x00000014: OUT Detect Pin Default 0x014190f0: [Jack] SPDIF Out at Ext Rear Conn = 1/8, Color = Pink DefAssociation = 0xf, Sequence = 0x0 Pin-ctls: 0x40: OUT Unsolicited: tag=0f, enabled=0 Connection: 1 0x04 Node 0x10 [Pin Complex] wcaps 0x400281: Stereo Digital Pincap 0x00000020: IN Pin Default 0x01c520f0: [Jack] SPDIF In at Ext Rear Conn = Optical, Color = Grey DefAssociation = 0xf, Sequence = 0x0 Pin-ctls: 0x20: IN Unsolicited: tag=10, enabled=0 Node 0x11 [Pin Complex] wcaps 0x400181: Stereo Pincap 0x0000001c: OUT HP Detect Pin Default 0x01014020: [Jack] Line Out at Ext Rear Conn = 1/8, Color = Green DefAssociation = 0x2, Sequence = 0x0 Pin-ctls: 0xc0: OUT HP Unsolicited: tag=11, enabled=0 Connection: 1 0x06 Node 0x12 [Pin Complex] wcaps 0x400181: Stereo Pincap 0x00000014: OUT Detect Pin Default 0x01016121: [Jack] Line Out at Ext Rear Conn = 1/8, Color = Orange DefAssociation = 0x2, Sequence = 0x1 Misc = NO_PRESENCE Pin-ctls: 0x40: OUT Unsolicited: tag=12, enabled=0 Connection: 1 0x07 Node 0x13 [Pin Complex] wcaps 0x400181: Stereo Pincap 0x00000014: OUT Detect Pin Default 0x01011022: [Jack] Line Out at Ext Rear Conn = 1/8, Color = Black DefAssociation = 0x2, Sequence = 0x2 Pin-ctls: 0x40: OUT Unsolicited: tag=13, enabled=0 Connection: 1 0x08 Node 0x14 [Beep Generator Widget] wcaps 0x70000c: Mono Amp-Out Amp-Out caps: ofs=0x04, nsteps=0x04, stepsize=0x17, mute=0 Amp-Out vals: [0x04] Node 0x15 [Audio Input] wcaps 0x1003d1: Stereo Digital Converter: stream=0, channel=2 Digital: Digital category: 0x0 PCM: rates [0x7f]: 8000 11025 16000 22050 32000 44100 48000 bits [0x1e]: 16 20 24 32 formats [0x3]: PCM FLOAT Unsolicited: tag=15, enabled=0 Connection: 1 0x16 Processing caps: benign=0, ncoeff=0 Node 0x16 [Pin Complex] wcaps 0x40030b: Stereo Digital Amp-In Amp-In caps: ofs=0x18, nsteps=0x58, stepsize=0x01, mute=1 Amp-In vals: [0x18 0x18] Pincap 0x00000030: IN OUT Pin Default 0x10a601f0: [Jack] Mic at Int N/A Conn = Digital, Color = Unknown DefAssociation = 0xf, Sequence = 0x0 Misc = NO_PRESENCE Pin-ctls: 0x20: IN Connection: 1 0x17 Node 0x17 [Audio Output] wcaps 0x291: Stereo Digital Converter: stream=0, channel=0 Digital: Digital category: 0x0 PCM: rates [0x7f0]: 32000 44100 48000 88200 96000 176400 192000 bits [0x1e]: 16 20 24 32 formats [0x3]: PCM FLOAT Unsolicited: tag=17, enabled=0 --endcollapse-- !!Aplay/Arecord output !!------------ APLAY **** List of PLAYBACK Hardware Devices **** card 0: NVidia [HDA NVidia], device 0: HDA Generic [HDA Generic] Subdevices: 1/1 Subdevice #0: subdevice #0 ARECORD **** List of CAPTURE Hardware Devices **** card 0: NVidia [HDA NVidia], device 0: HDA Generic [HDA Generic] Subdevices: 1/1 Subdevice #0: subdevice #0 !!Amixer output !!------------- !!-------Mixer controls for card 0 [NVidia] Card hw:0 'NVidia'/'HDA NVidia at 0xfae78000 irq 20' Mixer name : 'Wolfson Microelectronics ID 8800' Components : 'HDA:1aec8800,00000100,00100301' Controls : 3 Simple ctrls : 2 Simple mixer control 'Master',0 Capabilities: pvolume pswitch penum Playback channels: Front Left - Front Right Limits: Playback 0 - 127 Mono: Front Left: Playback 87 [69%] [-20.00dB] [on] Front Right: Playback 87 [69%] [-20.00dB] [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] !!Alsactl output !!------------- --startcollapse-- state.NVidia { control.1 { iface MIXER name 'Master Playback Switch' value.0 true value.1 true comment { access 'read write' type BOOLEAN count 2 } } control.2 { iface MIXER name 'Master Playback Volume' value.0 87 value.1 87 comment { access 'read write' type INTEGER count 2 range '0 - 127' dbmin -6350 dbmax 0 dbvalue.0 -2000 dbvalue.1 -2000 } } control.3 { iface MIXER name 'PCM Playback Volume' value.0 255 value.1 255 comment { access 'read write user' type INTEGER count 2 range '0 - 255' tlv '0000000100000008ffffec1400000014' dbmin -5100 dbmax 0 dbvalue.0 0 dbvalue.1 0 } } } --endcollapse-- !!ALSA Device nodes !!----------------- crw-rw----+ 1 root audio 116, 5 Feb 21 09:38 /dev/snd/controlC0 crw-rw----+ 1 root audio 116, 4 Feb 21 09:38 /dev/snd/hwC0D0 crw-rw----+ 1 root audio 116, 3 Feb 21 09:38 /dev/snd/pcmC0D0c crw-rw----+ 1 root audio 116, 2 Feb 21 09:39 /dev/snd/pcmC0D0p crw-rw----+ 1 root audio 116, 1 Feb 21 09:37 /dev/snd/seq crw-rw----+ 1 root audio 116, 33 Feb 21 09:37 /dev/snd/timer /dev/snd/by-path: total 0 drwxr-xr-x 2 root root 60 Feb 21 09:38 . drwxr-xr-x 3 root root 180 Feb 21 09:38 .. lrwxrwxrwx 1 root root 12 Feb 21 09:38 pci-0000:00:08.0 -> ../controlC0 Este mensaje se dirige exclusivamente a su destinatario y puede contener información privilegiada o CONFIDENCIAL. 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