June 2010 - Alsa-devel - mailman.alsa-project.org

[alsa-devel] newie, adding a file to the project
by javi perez 23 Jun '10

23 Jun '10

Hello, I am trying to add a file (aplay_miversion.c) to the project to make tests. I have modified the Makefile.am file of the aplay directory in the project. I have run ./configure and make and I don't get the file compiled. How shold I change the lines? LIBRT = @LIBRT@ INCLUDES = -I$(top_srcdir)/include LDADD = $(LIBINTL) $(LIBRT) # debug flags #LDFLAGS = -static #LDADD += -ldl bin_PROGRAMS = aplay aplay_miversion aplay_SOURCES = aplay.c aplay_miversion_SOURCES = aplay_miversion.c man_MANS = aplay.1 arecord.1 noinst_HEADERS = formats.h EXTRA_DIST = aplay.1 arecord.1 EXTRA_CLEAN = arecord arecord: aplay rm -f arecord $(LN_S) $< $@ arecord.1: aplay.1 rm -f arecord.1 $(LN_S) $< $@ install-exec-hook: arecord rm -f $(DESTDIR)$(bindir)/arecord (cd $(DESTDIR)$(bindir) && $(LN_S) aplay arecord) install-data-hook: rm -f $(DESTDIR)$(mandir)/man1/arecord.1 (cd $(DESTDIR)$(mandir)/man1 && $(LN_S) aplay.1 arecord.1)

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[alsa-devel] IEC switch issues
by pl bossart 23 Jun '10

23 Jun '10

If I play my AC3 data on my HDAudio/SPDIF output using the hw:0,1 device, I can use alsamixer/amixer to mute/unmute. [ume@plb PassThough]$ amixer cset numid=12 on numid=12,iface=MIXER,name='IEC958 Playback Switch' ; type=BOOLEAN,access=rw------,values=1 : values=on [ume@plb PassThough]$ amixer cset numid=12 off numid=12,iface=MIXER,name='IEC958 Playback Switch' ; type=BOOLEAN,access=rw------,values=1 : values=off Now if I use the iec958: plugin, I can't control the mute switch any longer: [ume@plb PassThough]$ amixer cset numid=12 off amixer: Control default element write error: Operation not permitted [ume@plb PassThough]$ amixer cset numid=12 on amixer: Control default element write error: Operation not permitted What could possibly cause this issue? the IEC plugin relies on the device 1, am I missing something here? Thanks for your help - Pierre

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[alsa-devel] AndreaElectronics PureAudio USB-SA
by rmfc 23 Jun '10

23 Jun '10

Hello, I'm looking for a sound card that can be used to speech recognition in noisy environments. The best device that I found so far is AndreaElectronics PureAudio USB-SA but unfortunately it isn't properly supported in Linux. The device is recognized as an USB Audio device (module used is snd_usb_audio) and I can record audio in Audacity. The problem is that none of advanced DSP features is enabled. I tried then Asus Xonar U1 but the problem is the same, no configuration is done in devices, so they record audio without quality to be used in a speech recognizer (not enough energy, audio DC components when software gain is applied in Sound Preferences...). Is there any way known way to configure a USB Audio device? If I have to implement some code, where should I start? I send you output of "lsusb" of AndreaElectronics USB-SA sound card. ------------------------------------------------------------ Bus 003 Device 002: ID 08a8:000c Andrea Electronics Device Descriptor: bLength 18 bDescriptorType 1 bcdUSB 1.10 bDeviceClass 0 (Defined at Interface level) bDeviceSubClass 0 bDeviceProtocol 0 bMaxPacketSize0 64 idVendor 0x08a8 Andrea Electronics idProduct 0x000c bcdDevice 1.00 iManufacturer 3 Andrea Electronics iProduct 1 Andrea PureAudio USB-SA iSerial 0 bNumConfigurations 1 Configuration Descriptor: bLength 9 bDescriptorType 2 wTotalLength 293 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 128 bInCollection 2 baInterfaceNr( 0) 1 baInterfaceNr( 1) 2 AudioControl Interface Descriptor: bLength 12 bDescriptorType 36 bDescriptorSubtype 2 (INPUT_TERMINAL) bTerminalID 1 wTerminalType 0x0101 USB Streaming bAssocTerminal 0 bNrChannels 2 wChannelConfig 0x0003 Left Front (L) Right Front (R) iChannelNames 0 iTerminal 0 AudioControl Interface Descriptor: bLength 12 bDescriptorType 36 bDescriptorSubtype 2 (INPUT_TERMINAL) bTerminalID 2 wTerminalType 0x0201 Microphone bAssocTerminal 0 bNrChannels 2 wChannelConfig 0x0003 Left Front (L) Right Front (R) iChannelNames 0 iTerminal 0 AudioControl Interface Descriptor: bLength 12 bDescriptorType 36 bDescriptorSubtype 2 (INPUT_TERMINAL) bTerminalID 3 wTerminalType 0x0603 Line Connector bAssocTerminal 0 bNrChannels 2 wChannelConfig 0x0003 Left Front (L) Right Front (R) iChannelNames 0 iTerminal 0 AudioControl Interface Descriptor: bLength 12 bDescriptorType 36 bDescriptorSubtype 2 (INPUT_TERMINAL) bTerminalID 4 wTerminalType 0x0605 SPDIF interface bAssocTerminal 0 bNrChannels 2 wChannelConfig 0x0003 Left Front (L) Right Front (R) iChannelNames 0 iTerminal 0 AudioControl Interface Descriptor: bLength 9 bDescriptorType 36 bDescriptorSubtype 3 (OUTPUT_TERMINAL) bTerminalID 6 wTerminalType 0x0301 Speaker bAssocTerminal 0 bSourceID 9 iTerminal 0 AudioControl Interface Descriptor: bLength 9 bDescriptorType 36 bDescriptorSubtype 3 (OUTPUT_TERMINAL) bTerminalID 7 wTerminalType 0x0101 USB Streaming bAssocTerminal 0 bSourceID 8 iTerminal 0 AudioControl Interface Descriptor: bLength 9 bDescriptorType 36 bDescriptorSubtype 5 (SELECTOR_UNIT) bUnitID 8 bNrInPins 3 baSource( 0) 10 baSource( 1) 11 baSource( 2) 12 iSelector 0 AudioControl Interface Descriptor: bLength 13 bDescriptorType 36 bDescriptorSubtype 6 (FEATURE_UNIT) bUnitID 9 bSourceID 1 bControlSize 2 bmaControls( 0) 0x01 bmaControls( 0) 0x02 Mute Loudness bmaControls( 1) 0x02 bmaControls( 1) 0x00 Volume bmaControls( 2) 0x02 bmaControls( 2) 0x00 Volume iFeature 0 AudioControl Interface Descriptor: bLength 10 bDescriptorType 36 bDescriptorSubtype 6 (FEATURE_UNIT) bUnitID 10 bSourceID 2 bControlSize 1 bmaControls( 0) 0x01 Mute bmaControls( 1) 0x02 Volume bmaControls( 2) 0x02 Volume iFeature 0 AudioControl Interface Descriptor: bLength 10 bDescriptorType 36 bDescriptorSubtype 6 (FEATURE_UNIT) bUnitID 11 bSourceID 3 bControlSize 1 bmaControls( 0) 0x01 Mute bmaControls( 1) 0x02 Volume bmaControls( 2) 0x02 Volume iFeature 0 AudioControl Interface Descriptor: bLength 10 bDescriptorType 36 bDescriptorSubtype 6 (FEATURE_UNIT) bUnitID 12 bSourceID 4 bControlSize 1 bmaControls( 0) 0x01 Mute bmaControls( 1) 0x00 bmaControls( 2) 0x00 iFeature 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 1 bDelay 1 frames wFormatTag 1 PCM AudioStreaming Interface Descriptor: bLength 20 bDescriptorType 36 bDescriptorSubtype 2 (FORMAT_TYPE) bFormatType 1 (FORMAT_TYPE_I) bNrChannels 2 bSubframeSize 2 bBitResolution 16 bSamFreqType 4 Discrete tSamFreq[ 0] 8000 tSamFreq[ 1] 16000 tSamFreq[ 2] 44100 tSamFreq[ 3] 48000 Endpoint Descriptor: bLength 9 bDescriptorType 5 bEndpointAddress 0x01 EP 1 OUT bmAttributes 9 Transfer Type Isochronous Synch Type Adaptive Usage Type Data wMaxPacketSize 0x00c8 1x 200 bytes bInterval 1 bRefresh 0 bSynchAddress 0 AudioControl Endpoint Descriptor: bLength 7 bDescriptorType 37 bDescriptorSubtype 1 (EP_GENERAL) bmAttributes 0x01 Sampling Frequency bLockDelayUnits 1 Milliseconds wLockDelay 1 Milliseconds 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 7 bDelay 1 frames wFormatTag 1 PCM AudioStreaming Interface Descriptor: bLength 20 bDescriptorType 36 bDescriptorSubtype 2 (FORMAT_TYPE) bFormatType 1 (FORMAT_TYPE_I) bNrChannels 2 bSubframeSize 2 bBitResolution 16 bSamFreqType 4 Discrete tSamFreq[ 0] 8000 tSamFreq[ 1] 16000 tSamFreq[ 2] 44100 tSamFreq[ 3] 48000 Endpoint Descriptor: bLength 9 bDescriptorType 5 bEndpointAddress 0x82 EP 2 IN bmAttributes 5 Transfer Type Isochronous Synch Type Asynchronous Usage Type Data wMaxPacketSize 0x00c8 1x 200 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 50 Report Descriptors: ** UNAVAILABLE ** Endpoint Descriptor: bLength 7 bDescriptorType 5 bEndpointAddress 0x87 EP 7 IN bmAttributes 3 Transfer Type Interrupt Synch Type None Usage Type Data wMaxPacketSize 0x0010 1x 16 bytes bInterval 1 Device Status: 0x0000 (Bus Powered) ------------------------------------------------------------ Thanks for any help that you can give me. Renato

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[alsa-devel] Pause functionality
by dv 23 Jun '10

23 Jun '10

Hi, it is stated in the documentation that I need to check with snd_pcm_hw_params_can_pause() to see if the device can pause or not. If it can, I can use snd_pcm_pause() to pause/resume. But what if snd_pcm_hw_params_can_pause() tells me it can't pause? What does one usually do then? Continue writing to pcm, with a buffer containing zeroed samples?

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[alsa-devel] usb-audio: some cleanup patches for 2.6.36
by Daniel Mack 23 Jun '10

23 Jun '10

Here are five patches that mostly do some basic cleanups. They're unurgent and should be queued for 2.6.36. I'm still pondering about the clock topology and how to fully support all corner cases. That might take awhile to sort out. Daniel [PATCH 1/5] ALSA: usb-audio: clean up includes in clock.c [PATCH 2/5] ALSA: usb-audio: unify UAC macros and struct names [PATCH 3/5] ALSA: usb-midi: whitespace fixes [PATCH 4/5] ALSA: usb-audio: move and add some comments [PATCH 5/5] ALSA: usb-audio: simplify control interface access

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[alsa-devel] [PATCH] Add volume resolution quirk for some Logitech webcams.
by Alexey Fisher 23 Jun '10

23 Jun '10

Some programs like Skype trying to set capture volume automatically. Normally it will tray, carefully step by step lover or higher, set the volume. In real word it work not really well, because devises and vendors lie about real audio settings. For example most Logitech webcams have 6400 or 3500 steps for capture volume. They do not tell that actual resolution is 384. So we have only 7 or 18 real steps. In this patch I set real resolution only for tested devices. Signed-off-by: Alexey Fisher <bug-track(a)fisher-privat.net> --- sound/usb/mixer.c | 13 +++++++++++++ 1 files changed, 13 insertions(+), 0 deletions(-) diff --git a/sound/usb/mixer.c b/sound/usb/mixer.c index 6939d0f..736d134 100644 --- a/sound/usb/mixer.c +++ b/sound/usb/mixer.c @@ -1107,6 +1107,19 @@ static void build_feature_ctl(struct mixer_build *state, void *raw_desc, } break; + case USB_ID(0x046d, 0x0809): + case USB_ID(0x046d, 0x0991): + /* Most audio usb devices lie about volume resolution. + * Most Logitech webcams have res = 384. + * Proboly there is some logitech magic behind this number --fishor + */ + if (!strcmp(kctl->id.name, "Mic Capture Volume")) { + snd_printk(KERN_INFO + "set resolution quirk: cval->res = 384\n"); + cval->res = 384; + } + break; + } snd_printdd(KERN_INFO "[%d] FU [%s] ch = %d, val = %d/%d/%d\n", -- 1.7.0.4

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[alsa-devel] [PATCH] Add volume resolution quirk for some logitech webcams
by Alexey Fisher 23 Jun '10

23 Jun '10

Some programs like Skype trying to set capture volume automatically. Normally it will tray carefully, step by step lover or higher the volume. In real word it work not really well, because devises and vendors lie about real audio settings. For example most Logitech webcams have 6400 or 3500 steps for capture volume. They do not tell that actual resolution is 384. So we have only 7 or 18 real steps. In this patch I set real resolution only for tested devices. Signed-off-by: Alexey Fisher <bug-track(a)fisher-privat.net> --- sound/usb/mixer.c | 12 ++++++++++++ 1 files changed, 12 insertions(+), 0 deletions(-) diff --git a/sound/usb/mixer.c b/sound/usb/mixer.c index 6939d0f..fd9f377 100644 --- a/sound/usb/mixer.c +++ b/sound/usb/mixer.c @@ -1107,6 +1107,18 @@ static void build_feature_ctl(struct mixer_build *state, void *raw_desc, } break; + case USB_ID(0x046d, 0x0809): + case USB_ID(0x046d, 0x0991): + /* most audio usb divise lighe about volume resolution. + * most of them have res = 384. Proboly there is some magick behind + * this number. */ + if (!strcmp(kctl->id.name, "Mic Capture Volume")) { + snd_printk(KERN_INFO + "set volume quirk: cval->res = 384\n"); + cval->res = 384; + } + break; + } snd_printdd(KERN_INFO "[%d] FU [%s] ch = %d, val = %d/%d/%d\n", -- 1.7.0.4

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[alsa-devel] [PATCH 1/2] ASoC: RX-51: Add Jack Function kcontrol
by Jarkko Nikula 23 Jun '10

23 Jun '10

Nokia RX-51/N900 has multifunction 4-pole audio-video jack that can be used as headphone, headset or audio-video connector. This patch implements the control 'Jack Function' which is used to select the desired function. At the moment only TV-out without audio is supported. Signed-off-by: Jarkko Nikula <jhnikula(a)gmail.com> --- sound/soc/omap/rx51.c | 44 ++++++++++++++++++++++++++++++++++++++++++++ 1 files changed, 44 insertions(+), 0 deletions(-) diff --git a/sound/soc/omap/rx51.c b/sound/soc/omap/rx51.c index 47d831e..1a2de34 100644 --- a/sound/soc/omap/rx51.c +++ b/sound/soc/omap/rx51.c @@ -37,14 +37,21 @@ #include "omap-pcm.h" #include "../codecs/tlv320aic3x.h" +#define RX51_TVOUT_SEL_GPIO 40 /* * REVISIT: TWL4030 GPIO base in RX-51. Now statically defined to 192. This * gpio is reserved in arch/arm/mach-omap2/board-rx51-peripherals.c */ #define RX51_SPEAKER_AMP_TWL_GPIO (192 + 7) +enum { + RX51_JACK_DISABLED, + RX51_JACK_TVOUT, /* tv-out */ +}; + static int rx51_spk_func; static int rx51_dmic_func; +static int rx51_jack_func; static void rx51_ext_control(struct snd_soc_codec *codec) { @@ -57,6 +64,9 @@ static void rx51_ext_control(struct snd_soc_codec *codec) else snd_soc_dapm_disable_pin(codec, "DMic"); + gpio_set_value(RX51_TVOUT_SEL_GPIO, + rx51_jack_func == RX51_JACK_TVOUT); + snd_soc_dapm_sync(codec); } @@ -162,6 +172,28 @@ static int rx51_set_input(struct snd_kcontrol *kcontrol, return 1; } +static int rx51_get_jack(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + ucontrol->value.integer.value[0] = rx51_jack_func; + + return 0; +} + +static int rx51_set_jack(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol); + + if (rx51_jack_func == ucontrol->value.integer.value[0]) + return 0; + + rx51_jack_func = ucontrol->value.integer.value[0]; + rx51_ext_control(codec); + + return 1; +} + static const struct snd_soc_dapm_widget aic34_dapm_widgets[] = { SND_SOC_DAPM_SPK("Ext Spk", rx51_spk_event), SND_SOC_DAPM_MIC("DMic", NULL), @@ -177,10 +209,12 @@ static const struct snd_soc_dapm_route audio_map[] = { static const char *spk_function[] = {"Off", "On"}; static const char *input_function[] = {"ADC", "Digital Mic"}; +static const char *jack_function[] = {"Off", "TV-OUT"}; static const struct soc_enum rx51_enum[] = { SOC_ENUM_SINGLE_EXT(ARRAY_SIZE(spk_function), spk_function), SOC_ENUM_SINGLE_EXT(ARRAY_SIZE(input_function), input_function), + SOC_ENUM_SINGLE_EXT(ARRAY_SIZE(jack_function), jack_function), }; static const struct snd_kcontrol_new aic34_rx51_controls[] = { @@ -188,6 +222,8 @@ static const struct snd_kcontrol_new aic34_rx51_controls[] = { rx51_get_spk, rx51_set_spk), SOC_ENUM_EXT("Input Select", rx51_enum[1], rx51_get_input, rx51_set_input), + SOC_ENUM_EXT("Jack Function", rx51_enum[2], + rx51_get_jack, rx51_set_jack), }; static int rx51_aic34_init(struct snd_soc_codec *codec) @@ -259,6 +295,11 @@ static int __init rx51_soc_init(void) if (!machine_is_nokia_rx51()) return -ENODEV; + err = gpio_request(RX51_TVOUT_SEL_GPIO, "tvout_sel"); + if (err) + goto err_gpio_tvout_sel; + gpio_direction_output(RX51_TVOUT_SEL_GPIO, 0); + rx51_snd_device = platform_device_alloc("soc-audio", -1); if (!rx51_snd_device) { err = -ENOMEM; @@ -277,6 +318,8 @@ static int __init rx51_soc_init(void) err2: platform_device_put(rx51_snd_device); err1: + gpio_free(RX51_TVOUT_SEL_GPIO); +err_gpio_tvout_sel: return err; } @@ -284,6 +327,7 @@ err1: static void __exit rx51_soc_exit(void) { platform_device_unregister(rx51_snd_device); + gpio_free(RX51_TVOUT_SEL_GPIO); } module_init(rx51_soc_init); -- 1.7.1

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[alsa-devel] [PATCH] codecs/tlv320aic23: fix suspend/resume
by Eric Bénard 23 Jun '10

23 Jun '10

The PWR register actually can't be restored at resume because its cached value was set to 0xFFFF when suspending and thus the initial state won't be restored. Thus, sound doesn't work after resume (tested on an i.MX27). To workaround this, this patch saves the PWR value before turning the codec's power off and store this value in the cached registers. Signed-off-by: Eric Bénard <eric(a)eukrea.com> Cc: broonie(a)opensource.wolfsonmicro.com Cc: anuj.aggarwal(a)ti.com Cc: lrg(a)slimlogic.co.uk --- sound/soc/codecs/tlv320aic23.c | 8 ++++++-- 1 files changed, 6 insertions(+), 2 deletions(-) diff --git a/sound/soc/codecs/tlv320aic23.c b/sound/soc/codecs/tlv320aic23.c index b0bae35..75e320b 100644 --- a/sound/soc/codecs/tlv320aic23.c +++ b/sound/soc/codecs/tlv320aic23.c @@ -614,9 +614,15 @@ static int tlv320aic23_suspend(struct platform_device *pdev, { struct snd_soc_device *socdev = platform_get_drvdata(pdev); struct snd_soc_codec *codec = socdev->card->codec; + u16 tmp; tlv320aic23_write(codec, TLV320AIC23_ACTIVE, 0x0); + /* backup PWR register */ + tmp = tlv320aic23_read_reg_cache(codec, TLV320AIC23_PWR); + /* turn off the codec */ tlv320aic23_set_bias_level(codec, SND_SOC_BIAS_OFF); + /* restore PWR register in the cache */ + tlv320aic23_write_reg_cache(codec, TLV320AIC23_PWR, tmp); return 0; } @@ -633,8 +639,6 @@ static int tlv320aic23_resume(struct platform_device *pdev) tlv320aic23_write(codec, reg, val); } - tlv320aic23_set_bias_level(codec, SND_SOC_BIAS_STANDBY); - return 0; } -- 1.6.3.3

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[alsa-devel] imx-ssi.c & SND_SOC_DAIFMT_I2S
by Eric Bénard 23 Jun '10

23 Jun '10

Hi Sascha, while trying to get a TI TLV320AIC23B codec configured as I2S master to work with an i.MX27 using the latest git kernel, I noticed there may be a problem in the slave I2S setting in imx-ssi.c Please find two captures of the I2S bus signals : - http://eukrea.com/tmp/initial_i2s.png : this is not an I2S signal, both left & right are sent while Frame Sync is high (so I only get mono output) - http://eukrea.com/tmp/i2s_slave.png : when I set I2S_MODE to 0x2 in SCR register (scr |= SSI_SCR_NET | 0x40; at line 101 of imx-ssi.c), thus setting the SSI in "I2S slave mode", I get a real I2S signal as shown on this capture. I understand the actual I2S configuration may work with the Freescale's PMIC but this chipset may not be fully I2S compliant and I think that the actual setting prevent real I2S codecs from being used with the i.MX. Am I wrong here ? Shouldn't we use the I2S slave mode of the SSI instead of what is done actually ? Thanks and best regards, Eric

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