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  • Pulseaudio is no longer working in Debian Squeeze: 'Failed to open module "module-combine-sink": file not found'

    - by mattalexx
    I'm having a problem with pulseaudio. My machine crashed, and when I rebooted and ran pavucontrol, I got a "Connection Failed: Connection refused" dialog. When I run pulseaudio --log-level=info --log-target=stderr from the command line, I get the following output: [...] I: alsa-util.c: Error opening PCM device front:1: No such file or directory I: (alsa-lib)pcm_hw.c: open '/dev/snd/pcmC1D0c' failed (-2) I: alsa-util.c: Error opening PCM device hw:1: No such file or directory I: (alsa-lib)pcm_hw.c: open '/dev/snd/pcmC1D0c' failed (-2) I: alsa-util.c: Error opening PCM device iec958:1: No such file or directory I: (alsa-lib)pcm_hw.c: open '/dev/snd/pcmC1D0c' failed (-2) I: alsa-util.c: Error opening PCM device iec958:1: No such file or directory I: alsa-util.c: Failed to set hardware parameters on plug:iec958:1: Invalid argument I: alsa-util.c: Failed to set hardware parameters on plug:iec958:1: Invalid argument I: alsa-util.c: Failed to set hardware parameters on plug:iec958:1: Invalid argument I: alsa-util.c: Failed to set hardware parameters on plug:iec958:1: Invalid argument I: alsa-util.c: Failed to set hardware parameters on plug:iec958:1: Invalid argument I: (alsa-lib)pcm.c: Unknown PCM a52:1 I: alsa-util.c: Error opening PCM device a52:1: No such file or directory I: (alsa-lib)pcm.c: Unknown PCM a52:1 I: alsa-util.c: Error opening PCM device a52:1: No such file or directory I: (alsa-lib)pcm.c: Unknown PCM a52:1 I: alsa-util.c: Error opening PCM device a52:1: No such file or directory I: (alsa-lib)pcm.c: Unknown PCM a52:1 I: alsa-util.c: Error opening PCM device a52:1: No such file or directory I: (alsa-lib)pcm.c: Unknown PCM a52:1 I: alsa-util.c: Error opening PCM device a52:1: No such file or directory I: (alsa-lib)pcm.c: Unknown PCM a52:1 I: alsa-util.c: Error opening PCM device a52:1: No such file or directory I: (alsa-lib)pcm.c: Unknown PCM a52:1 I: alsa-util.c: Error opening PCM device a52:1: No such file or directory I: (alsa-lib)pcm.c: Unknown PCM a52:1 I: alsa-util.c: Error opening PCM device a52:1: No such file or directory I: (alsa-lib)pcm.c: Unknown PCM a52:1 I: alsa-util.c: Error opening PCM device a52:1: No such file or directory I: (alsa-lib)pcm.c: Unknown PCM a52:1 I: alsa-util.c: Error opening PCM device a52:1: No such file or directory I: (alsa-lib)confmisc.c: Unable to find definition 'cards.USB-Audio.pcm.hdmi.0:CARD=1,AES0=4,AES1=130,AES2=0,AES3=2' I: (alsa-lib)conf.c: function snd_func_refer returned error: No such file or directory I: (alsa-lib)conf.c: Evaluate error: No such file or directory I: (alsa-lib)pcm.c: Unknown PCM hdmi:1 I: alsa-util.c: Error opening PCM device hdmi:1: No such file or directory I: (alsa-lib)confmisc.c: Unable to find definition 'cards.USB-Audio.pcm.hdmi.0:CARD=1,AES0=4,AES1=130,AES2=0,AES3=2' I: (alsa-lib)conf.c: function snd_func_refer returned error: No such file or directory I: (alsa-lib)conf.c: Evaluate error: No such file or directory I: (alsa-lib)pcm.c: Unknown PCM hdmi:1 I: alsa-util.c: Error opening PCM device hdmi:1: No such file or directory I: (alsa-lib)confmisc.c: Unable to find definition 'cards.USB-Audio.pcm.hdmi.0:CARD=1,AES0=4,AES1=130,AES2=0,AES3=2' I: (alsa-lib)conf.c: function snd_func_refer returned error: No such file or directory I: (alsa-lib)conf.c: Evaluate error: No such file or directory I: (alsa-lib)pcm.c: Unknown PCM hdmi:1 I: alsa-util.c: Error opening PCM device hdmi:1: No such file or directory I: (alsa-lib)confmisc.c: Unable to find definition 'cards.USB-Audio.pcm.hdmi.0:CARD=1,AES0=4,AES1=130,AES2=0,AES3=2' I: (alsa-lib)conf.c: function snd_func_refer returned error: No such file or directory I: (alsa-lib)conf.c: Evaluate error: No such file or directory I: (alsa-lib)pcm.c: Unknown PCM hdmi:1 I: alsa-util.c: Error opening PCM device hdmi:1: No such file or directory I: (alsa-lib)confmisc.c: Unable to find definition 'cards.USB-Audio.pcm.hdmi.0:CARD=1,AES0=4,AES1=130,AES2=0,AES3=2' I: (alsa-lib)conf.c: function snd_func_refer returned error: No such file or directory I: (alsa-lib)conf.c: Evaluate error: No such file or directory I: (alsa-lib)pcm.c: Unknown PCM hdmi:1 I: alsa-util.c: Error opening PCM device hdmi:1: No such file or directory I: (alsa-lib)pcm_hw.c: open '/dev/snd/pcmC1D0c' failed (-2) I: alsa-util.c: Error opening PCM device hw:1: No such file or directory I: (alsa-lib)pcm_hw.c: open '/dev/snd/pcmC1D0c' failed (-2) I: alsa-util.c: Error opening PCM device front:1: No such file or directory I: (alsa-lib)pcm_hw.c: open '/dev/snd/pcmC1D0c' failed (-2) I: alsa-util.c: Error opening PCM device hw:1: No such file or directory I: (alsa-lib)pcm_hw.c: open '/dev/snd/pcmC1D0c' failed (-2) I: alsa-util.c: Error opening PCM device iec958:1: No such file or directory I: (alsa-lib)pcm_hw.c: open '/dev/snd/pcmC1D0c' failed (-2) I: alsa-util.c: Error opening PCM device iec958:1: No such file or directory I: card.c: Created 0 "alsa_card.usb-FiiO_DigiHug_USB_Audio-01-Audio" I: alsa-sink.c: Successfully opened device front:1. I: alsa-sink.c: Selected mapping 'Analog Stereo' (analog-stereo). I: alsa-sink.c: Successfully enabled mmap() mode. I: alsa-sink.c: Successfully enabled timer-based scheduling mode. I: (alsa-lib)control.c: Invalid CTL front:1 I: alsa-mixer.c: Unable to attach to mixer front:1: No such file or directory I: alsa-mixer.c: Successfully attached to mixer 'hw:1' W: alsa-mixer.c: Your kernel driver is broken: it reports a volume range from 0.00 dB to 0.00 dB which makes no sense. I: module-device-restore.c: Restoring volume for sink alsa_output.usb-FiiO_DigiHug_USB_Audio-01-Audio.analog-stereo. I: sink.c: Created sink 0 "alsa_output.usb-FiiO_DigiHug_USB_Audio-01-Audio.analog-stereo" with sample spec s16le 2ch 44100Hz and channel map front-left,front-right I: sink.c: alsa.resolution_bits = "16" I: sink.c: device.api = "alsa" I: sink.c: device.class = "sound" I: sink.c: alsa.class = "generic" I: sink.c: alsa.subclass = "generic-mix" I: sink.c: alsa.name = "USB Audio" I: sink.c: alsa.id = "USB Audio" I: sink.c: alsa.subdevice = "0" I: sink.c: alsa.subdevice_name = "subdevice #0" I: sink.c: alsa.device = "0" I: sink.c: alsa.card = "1" I: sink.c: alsa.card_name = "DigiHug USB Audio" I: sink.c: alsa.long_card_name = "FiiO DigiHug USB Audio at usb-0000:00:1a.0-1.2, full speed" I: sink.c: alsa.driver_name = "snd_usb_audio" I: sink.c: device.bus_path = "pci-0000:00:1a.0-usb-0:1.2:1.1" I: sink.c: sysfs.path = "/devices/pci0000:00/0000:00:1a.0/usb1/1-1/1-1.2/1-1.2:1.1/sound/card1" I: sink.c: udev.id = "usb-FiiO_DigiHug_USB_Audio-01-Audio" I: sink.c: device.bus = "usb" I: sink.c: device.vendor.id = "1852" I: sink.c: device.vendor.name = "GYROCOM C&C Co., LTD" I: sink.c: device.product.id = "7022" I: sink.c: device.product.name = "DigiHug_USB_Audio" I: sink.c: device.serial = "FiiO_DigiHug_USB_Audio" I: sink.c: device.string = "front:1" I: sink.c: device.buffering.buffer_size = "352800" I: sink.c: device.buffering.fragment_size = "176400" I: sink.c: device.access_mode = "mmap+timer" I: sink.c: device.profile.name = "analog-stereo" I: sink.c: device.profile.description = "Analog Stereo" I: sink.c: device.description = "DigiHug_USB_Audio Analog Stereo" I: sink.c: alsa.mixer_name = "USB Mixer" I: sink.c: alsa.components = "USB1852:7022" I: sink.c: module-udev-detect.discovered = "1" I: sink.c: device.icon_name = "audio-card-usb" I: source.c: Created source 0 "alsa_output.usb-FiiO_DigiHug_USB_Audio-01-Audio.analog-stereo.monitor" with sample spec s16le 2ch 44100Hz and channel map front-left,front-right I: source.c: device.description = "Monitor of DigiHug_USB_Audio Analog Stereo" I: source.c: device.class = "monitor" I: source.c: alsa.card = "1" I: source.c: alsa.card_name = "DigiHug USB Audio" I: source.c: alsa.long_card_name = "FiiO DigiHug USB Audio at usb-0000:00:1a.0-1.2, full speed" I: source.c: alsa.driver_name = "snd_usb_audio" I: source.c: device.bus_path = "pci-0000:00:1a.0-usb-0:1.2:1.1" I: source.c: sysfs.path = "/devices/pci0000:00/0000:00:1a.0/usb1/1-1/1-1.2/1-1.2:1.1/sound/card1" I: source.c: udev.id = "usb-FiiO_DigiHug_USB_Audio-01-Audio" I: source.c: device.bus = "usb" I: source.c: device.vendor.id = "1852" I: source.c: device.vendor.name = "GYROCOM C&C Co., LTD" I: source.c: device.product.id = "7022" I: source.c: device.product.name = "DigiHug_USB_Audio" I: source.c: device.serial = "FiiO_DigiHug_USB_Audio" I: source.c: device.string = "1" I: source.c: module-udev-detect.discovered = "1" I: source.c: device.icon_name = "audio-card-usb" I: alsa-sink.c: Using 2.0 fragments of size 176400 bytes (1000.00ms), buffer size is 352800 bytes (2000.00ms) I: alsa-sink.c: Time scheduling watermark is 20.00ms I: alsa-sink.c: Hardware volume ranges from 0 to 110. I: alsa-sink.c: Using hardware volume control. Hardware dB scale not supported. I: alsa-sink.c: Using hardware mute control. I: core-util.c: Successfully enabled SCHED_RR scheduling for thread, with priority 5. I: alsa-sink.c: Starting playback. I: module.c: Loaded "module-alsa-card" (index: #4; argument: "device_id="1" name="usb-FiiO_DigiHug_USB_Audio-01-Audio" card_name="alsa_card.usb-FiiO_DigiHug_USB_Audio-01-Audio" tsched=yes ignore_dB=no card_properties="module-udev-detect.discovered=1""). I: module-udev-detect.c: Card /devices/pci0000:00/0000:00:1a.0/usb1/1-1/1-1.2/1-1.2:1.1/sound/card1 (alsa_card.usb-FiiO_DigiHug_USB_Audio-01-Audio) module loaded. I: (alsa-lib)pcm_hw.c: open '/dev/snd/pcmC2D0p' failed (-2) I: alsa-util.c: Error opening PCM device hw:2: No such file or directory I: (alsa-lib)pcm_hw.c: open '/dev/snd/pcmC2D0p' failed (-2) I: alsa-util.c: Error opening PCM device hw:2: No such file or directory I: (alsa-lib)pcm_hw.c: open '/dev/snd/pcmC2D0p' failed (-2) I: alsa-util.c: Error opening PCM device hw:2: No such file or directory I: (alsa-lib)pcm_hw.c: open '/dev/snd/pcmC2D0p' failed (-2) I: alsa-util.c: Error opening PCM device hw:2: No such file or directory I: (alsa-lib)pcm_hw.c: open '/dev/snd/pcmC2D0p' failed (-2) I: alsa-util.c: Error opening PCM device hw:2: No such file or directory I: (alsa-lib)pcm_hw.c: open '/dev/snd/pcmC2D0p' failed (-2) I: alsa-util.c: Error opening PCM device front:2: No such file or directory I: (alsa-lib)pcm_hw.c: open '/dev/snd/pcmC2D0p' failed (-2) I: alsa-util.c: Error opening PCM device hw:2: No such file or directory I: (alsa-lib)pcm_hw.c: open '/dev/snd/pcmC2D0p' failed (-2) I: alsa-util.c: Error opening PCM device front:2: No such file or directory I: (alsa-lib)pcm_hw.c: open '/dev/snd/pcmC2D0p' failed (-2) I: alsa-util.c: Error opening PCM device hw:2: No such file or directory I: (alsa-lib)pcm_hw.c: open '/dev/snd/pcmC2D0p' failed (-2) I: alsa-util.c: Error opening PCM device front:2: No such file or directory I: (alsa-lib)pcm_hw.c: open '/dev/snd/pcmC2D0p' failed (-2) I: alsa-util.c: Error opening PCM device hw:2: No such file or directory I: (alsa-lib)pcm_hw.c: open '/dev/snd/pcmC2D0p' failed (-2) I: alsa-util.c: Error opening PCM device front:2: No such file or directory I: (alsa-lib)pcm_hw.c: open '/dev/snd/pcmC2D0p' failed (-2) I: alsa-util.c: Error opening PCM device hw:2: No such file or directory I: (alsa-lib)pcm_hw.c: open '/dev/snd/pcmC2D0p' failed (-2) I: alsa-util.c: Error opening PCM device front:2: No such file or directory I: (alsa-lib)pcm_hw.c: open '/dev/snd/pcmC2D0p' failed (-2) I: alsa-util.c: Error opening PCM device hw:2: No such file or directory I: (alsa-lib)pcm_hw.c: open '/dev/snd/pcmC2D0p' failed (-2) I: alsa-util.c: Error opening PCM device surround40:2: No such file or directory I: (alsa-lib)pcm_hw.c: open '/dev/snd/pcmC2D0p' failed (-2) I: alsa-util.c: Error opening PCM device surround40:2: No such file or directory I: (alsa-lib)pcm_hw.c: open '/dev/snd/pcmC2D0p' failed (-2) I: alsa-util.c: Error opening PCM device surround40:2: No such file or directory I: (alsa-lib)pcm_hw.c: open '/dev/snd/pcmC2D0p' failed (-2) I: alsa-util.c: Error opening PCM device surround40:2: No such file or directory I: (alsa-lib)pcm_hw.c: open '/dev/snd/pcmC2D0p' failed (-2) I: alsa-util.c: Error opening PCM device surround40:2: No such file or directory I: (alsa-lib)pcm_hw.c: open '/dev/snd/pcmC2D0p' failed (-2) I: alsa-util.c: Error opening PCM device surround41:2: No such file or directory I: (alsa-lib)pcm_hw.c: open '/dev/snd/pcmC2D0p' failed (-2) I: alsa-util.c: Error opening PCM device surround41:2: No such file or directory I: (alsa-lib)pcm_hw.c: open '/dev/snd/pcmC2D0p' failed (-2) I: alsa-util.c: Error opening PCM device surround41:2: No such file or directory I: (alsa-lib)pcm_hw.c: open '/dev/snd/pcmC2D0p' failed (-2) I: alsa-util.c: Error opening PCM device surround41:2: No such file or directory I: (alsa-lib)pcm_hw.c: open '/dev/snd/pcmC2D0p' failed (-2) I: alsa-util.c: Error opening PCM device surround41:2: No such file or directory I: (alsa-lib)pcm_hw.c: open '/dev/snd/pcmC2D0p' failed (-2) I: alsa-util.c: Error opening PCM device surround50:2: No such file or directory I: (alsa-lib)pcm_hw.c: open '/dev/snd/pcmC2D0p' failed (-2) I: alsa-util.c: Error opening PCM device surround50:2: No such file or directory I: (alsa-lib)pcm_hw.c: open '/dev/snd/pcmC2D0p' failed (-2) I: alsa-util.c: Error opening PCM device surround50:2: No such file or directory I: (alsa-lib)pcm_hw.c: open '/dev/snd/pcmC2D0p' failed (-2) I: alsa-util.c: Error opening PCM device surround50:2: No such file or directory I: (alsa-lib)pcm_hw.c: open '/dev/snd/pcmC2D0p' failed (-2) I: alsa-util.c: Error opening PCM device surround50:2: No such file or directory I: (alsa-lib)pcm_hw.c: open '/dev/snd/pcmC2D0p' failed (-2) I: alsa-util.c: Error opening PCM device surround51:2: No such file or directory I: (alsa-lib)pcm_hw.c: open '/dev/snd/pcmC2D0p' failed (-2) I: alsa-util.c: Error opening PCM device surround51:2: No such file or directory I: (alsa-lib)pcm_hw.c: open '/dev/snd/pcmC2D0p' failed (-2) I: alsa-util.c: Error opening PCM device surround51:2: No such file or directory I: (alsa-lib)pcm_hw.c: open '/dev/snd/pcmC2D0p' failed (-2) I: alsa-util.c: Error opening PCM device surround51:2: No such file or directory I: (alsa-lib)pcm_hw.c: open '/dev/snd/pcmC2D0p' failed (-2) I: alsa-util.c: Error opening PCM device surround51:2: No such file or directory I: (alsa-lib)pcm_hw.c: open '/dev/snd/pcmC2D0p' failed (-2) I: alsa-util.c: Error opening PCM device surround71:2: No such file or directory I: (alsa-lib)pcm_hw.c: open '/dev/snd/pcmC2D0p' failed (-2) I: alsa-util.c: Error opening PCM device surround71:2: No such file or directory I: (alsa-lib)pcm_hw.c: open '/dev/snd/pcmC2D0p' failed (-2) I: alsa-util.c: Error opening PCM device surround71:2: No such file or directory I: (alsa-lib)pcm_hw.c: open '/dev/snd/pcmC2D0p' failed (-2) I: alsa-util.c: Error opening PCM device surround71:2: No such file or directory I: (alsa-lib)pcm_hw.c: open '/dev/snd/pcmC2D0p' failed (-2) I: alsa-util.c: Error opening PCM device surround71:2: No such file or directory I: (alsa-lib)pcm_hw.c: open '/dev/snd/pcmC2D0p' failed (-2) I: alsa-util.c: Error opening PCM device iec958:2: No such file or directory I: (alsa-lib)pcm_hw.c: open '/dev/snd/pcmC2D0p' failed (-2) I: alsa-util.c: Error opening PCM device iec958:2: No such file or directory I: (alsa-lib)pcm_hw.c: open '/dev/snd/pcmC2D0p' failed (-2) I: alsa-util.c: Error opening PCM device iec958:2: No such file or directory I: (alsa-lib)pcm_hw.c: open '/dev/snd/pcmC2D0p' failed (-2) I: alsa-util.c: Error opening PCM device iec958:2: No such file or directory I: (alsa-lib)pcm_hw.c: open '/dev/snd/pcmC2D0p' failed (-2) I: alsa-util.c: Error opening PCM device iec958:2: No such file or directory I: (alsa-lib)pcm_hw.c: open '/dev/snd/pcmC2D0p' failed (-2) I: alsa-util.c: Error opening PCM device iec958:2: No such file or directory I: (alsa-lib)pcm_hw.c: open '/dev/snd/pcmC2D0p' failed (-2) I: alsa-util.c: Error opening PCM device iec958:2: No such file or directory I: (alsa-lib)pcm_hw.c: open '/dev/snd/pcmC2D0p' failed (-2) I: alsa-util.c: Error opening PCM device iec958:2: No such file or directory I: (alsa-lib)pcm_hw.c: open '/dev/snd/pcmC2D0p' failed (-2) I: alsa-util.c: Error opening PCM device iec958:2: No such file or directory I: (alsa-lib)pcm_hw.c: open '/dev/snd/pcmC2D0p' failed (-2) I: alsa-util.c: Error opening PCM device iec958:2: No such file or directory I: (alsa-lib)pcm.c: Unknown PCM a52:2 I: alsa-util.c: Error opening PCM device a52:2: No such file or directory I: (alsa-lib)pcm.c: Unknown PCM a52:2 I: alsa-util.c: Error opening PCM device a52:2: No such file or directory I: (alsa-lib)pcm.c: Unknown PCM a52:2 I: alsa-util.c: Error opening PCM device a52:2: No such file or directory I: (alsa-lib)pcm.c: Unknown PCM a52:2 I: alsa-util.c: Error opening PCM device a52:2: No such file or directory I: (alsa-lib)pcm.c: Unknown PCM a52:2 I: alsa-util.c: Error opening PCM device a52:2: No such file or directory I: (alsa-lib)pcm.c: Unknown PCM a52:2 I: alsa-util.c: Error opening PCM device a52:2: No such file or directory I: (alsa-lib)pcm.c: Unknown PCM a52:2 I: alsa-util.c: Error opening PCM device a52:2: No such file or directory I: (alsa-lib)pcm.c: Unknown PCM a52:2 I: alsa-util.c: Error opening PCM device a52:2: No such file or directory I: (alsa-lib)pcm.c: Unknown PCM a52:2 I: alsa-util.c: Error opening PCM device a52:2: No such file or directory I: (alsa-lib)pcm.c: Unknown PCM a52:2 I: alsa-util.c: Error opening PCM device a52:2: No such file or directory I: (alsa-lib)confmisc.c: Unable to find definition 'cards.USB-Audio.pcm.hdmi.0:CARD=2,AES0=4,AES1=130,AES2=0,AES3=2' I: (alsa-lib)conf.c: function snd_func_refer returned error: No such file or directory I: (alsa-lib)conf.c: Evaluate error: No such file or directory I: (alsa-lib)pcm.c: Unknown PCM hdmi:2 I: alsa-util.c: Error opening PCM device hdmi:2: No such file or directory I: (alsa-lib)confmisc.c: Unable to find definition 'cards.USB-Audio.pcm.hdmi.0:CARD=2,AES0=4,AES1=130,AES2=0,AES3=2' I: (alsa-lib)conf.c: function snd_func_refer returned error: No such file or directory I: (alsa-lib)conf.c: Evaluate error: No such file or directory I: (alsa-lib)pcm.c: Unknown PCM hdmi:2 I: alsa-util.c: Error opening PCM device hdmi:2: No such file or directory I: (alsa-lib)confmisc.c: Unable to find definition 'cards.USB-Audio.pcm.hdmi.0:CARD=2,AES0=4,AES1=130,AES2=0,AES3=2' I: (alsa-lib)conf.c: function snd_func_refer returned error: No such file or directory I: (alsa-lib)conf.c: Evaluate error: No such file or directory I: (alsa-lib)pcm.c: Unknown PCM hdmi:2 I: alsa-util.c: Error opening PCM device hdmi:2: No such file or directory I: (alsa-lib)confmisc.c: Unable to find definition 'cards.USB-Audio.pcm.hdmi.0:CARD=2,AES0=4,AES1=130,AES2=0,AES3=2' I: (alsa-lib)conf.c: function snd_func_refer returned error: No such file or directory I: (alsa-lib)conf.c: Evaluate error: No such file or directory I: (alsa-lib)pcm.c: Unknown PCM hdmi:2 I: alsa-util.c: Error opening PCM device hdmi:2: No such file or directory I: (alsa-lib)confmisc.c: Unable to find definition 'cards.USB-Audio.pcm.hdmi.0:CARD=2,AES0=4,AES1=130,AES2=0,AES3=2' I: (alsa-lib)conf.c: function snd_func_refer returned error: No such file or directory I: (alsa-lib)conf.c: Evaluate error: No such file or directory I: (alsa-lib)pcm.c: Unknown PCM hdmi:2 I: alsa-util.c: Error opening PCM device hdmi:2: No such file or directory I: alsa-util.c: Device hw:2 doesn't support 44100 Hz, changed to 8000 Hz. I: alsa-util.c: Failed to set hardware parameters on plug:front:2: Invalid argument I: alsa-util.c: Failed to set hardware parameters on plug:hw:2: Invalid argument I: alsa-util.c: Failed to set hardware parameters on plug:iec958:2: Invalid argument I: alsa-util.c: Failed to set hardware parameters on plug:iec958:2: Invalid argument I: module-card-restore.c: Restoring profile for card alsa_card.usb-046d_08d7-01-U0x46d0x8d7. I: card.c: Created 1 "alsa_card.usb-046d_08d7-01-U0x46d0x8d7" I: module.c: Loaded "module-alsa-card" (index: #5; argument: "device_id="2" name="usb-046d_08d7-01-U0x46d0x8d7" card_name="alsa_card.usb-046d_08d7-01-U0x46d0x8d7" tsched=yes ignore_dB=no card_properties="module-udev-detect.discovered=1""). I: module-udev-detect.c: Card /devices/pci0000:00/0000:00:1a.0/usb1/1-1/1-1.6/1-1.6:1.1/sound/card2 (alsa_card.usb-046d_08d7-01-U0x46d0x8d7) module loaded. I: module-udev-detect.c: Found 3 cards. I: module.c: Loaded "module-udev-detect" (index: #6; argument: ""). I: module.c: Loaded "module-esound-protocol-unix" (index: #7; argument: ""). I: module.c: Loaded "module-native-protocol-unix" (index: #8; argument: ""). I: module-default-device-restore.c: Saved default sink 'alsa_output.pci-0000_00_1b.0.analog-surround-41' not existant, not restoring default sink setting. I: module-default-device-restore.c: Saved default source 'alsa_output.pci-0000_00_1b.0.analog-surround-41.monitor' not existant, not restoring default source setting. I: module.c: Loaded "module-default-device-restore" (index: #9; argument: ""). I: module.c: Loaded "module-rescue-streams" (index: #10; argument: ""). I: module.c: Loaded "module-always-sink" (index: #11; argument: ""). I: module.c: Loaded "module-intended-roles" (index: #12; argument: ""). I: module.c: Loaded "module-suspend-on-idle" (index: #13; argument: ""). I: client.c: Created 0 "ConsoleKit Session /org/freedesktop/ConsoleKit/Session2" I: module.c: Loaded "module-console-kit" (index: #14; argument: ""). I: module.c: Loaded "module-position-event-sounds" (index: #15; argument: ""). I: module.c: Loaded "module-cork-music-on-phone" (index: #16; argument: ""). E: module.c: Failed to open module "module-combine-sink": file not found E: main.c: Module load failed. E: main.c: Failed to initialize daemon. I: module.c: Unloading "module-device-restore" (index: #0). I: module.c: Unloaded "module-device-restore" (index: #0). I: module.c: Unloading "module-stream-restore" (index: #1). I: module.c: Unloaded "module-stream-restore" (index: #1). I: module.c: Unloading "module-card-restore" (index: #2). I: module.c: Unloaded "module-card-restore" (index: #2). I: module.c: Unloading "module-augment-properties" (index: #3). I: module.c: Unloaded "module-augment-properties" (index: #3). I: module.c: Unloading "module-alsa-card" (index: #4). I: sink.c: Freeing sink 0 "alsa_output.usb-FiiO_DigiHug_USB_Audio-01-Audio.analog-stereo" I: source.c: Freeing source 0 "alsa_output.usb-FiiO_DigiHug_USB_Audio-01-Audio.analog-stereo.monitor" I: card.c: Freed 0 "alsa_card.usb-FiiO_DigiHug_USB_Audio-01-Audio" I: module.c: Unloaded "module-alsa-card" (index: #4). I: module.c: Unloading "module-alsa-card" (index: #5). I: card.c: Freed 1 "alsa_card.usb-046d_08d7-01-U0x46d0x8d7" I: module.c: Unloaded "module-alsa-card" (index: #5). I: module.c: Unloading "module-udev-detect" (index: #6). I: module.c: Unloaded "module-udev-detect" (index: #6). I: module.c: Unloading "module-esound-protocol-unix" (index: #7). I: module.c: Unloaded "module-esound-protocol-unix" (index: #7). I: module.c: Unloading "module-native-protocol-unix" (index: #8). I: module.c: Unloaded "module-native-protocol-unix" (index: #8). I: module.c: Unloading "module-default-device-restore" (index: #9). I: module.c: Unloaded "module-default-device-restore" (index: #9). I: module.c: Unloading "module-rescue-streams" (index: #10). I: module.c: Unloaded "module-rescue-streams" (index: #10). I: module.c: Unloading "module-always-sink" (index: #11). I: module.c: Unloaded "module-always-sink" (index: #11). I: module.c: Unloading "module-intended-roles" (index: #12). I: module.c: Unloaded "module-intended-roles" (index: #12). I: module.c: Unloading "module-suspend-on-idle" (index: #13). I: module.c: Unloaded "module-suspend-on-idle" (index: #13). I: module.c: Unloading "module-console-kit" (index: #14). I: client.c: Freed 0 "ConsoleKit Session /org/freedesktop/ConsoleKit/Session2" I: module.c: Unloaded "module-console-kit" (index: #14). I: module.c: Unloading "module-position-event-sounds" (index: #15). I: module.c: Unloaded "module-position-event-sounds" (index: #15). I: module.c: Unloading "module-cork-music-on-phone" (index: #16). I: module.c: Unloaded "module-cork-music-on-phone" (index: #16). I: main.c: Daemon terminated. I believe the relevant part is this: E: module.c: Failed to open module "module-combine-sink": file not found E: main.c: Module load failed. E: main.c: Failed to initialize daemon. I tried uninstalling and reinstalling pulseaudio, I tried to find a way to install module-combine-sink. Nothing worked. I'm on a Debian Squeeze 32-bit machine. What can I do to fix this?

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  • Convert WAV to PCM(wav)

    - by Marco
    Hi, I'm looking for a small PCM converter tool which I can access by Dos-Console. I have any wave-files and need always this output: PCM 44,1k, 16bit, Mono Is there any program for this? Thx 4 answers

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  • using isight camera in macbookpro(8,2) on ubuntu 12.04 virtualbox VM

    - by Kurt Spindler
    I'm having a lot of trouble using the built-in isight camera on my macbookpro8,2 (early 2011) from an ubuntu 12.04 virtual machine, run inside VirtualBox. The following is the log I get when I try to run guvcview ubuntu@ubuntu:~$ guvcview guvcview 1.5.3 ALSA lib pcm.c:2217:(snd_pcm_open_noupdate) Unknown PCM cards.pcm.rear ALSA lib pcm.c:2217:(snd_pcm_open_noupdate) Unknown PCM cards.pcm.center_lfe ALSA lib pcm.c:2217:(snd_pcm_open_noupdate) Unknown PCM cards.pcm.side ALSA lib pcm.c:2217:(snd_pcm_open_noupdate) Unknown PCM cards.pcm.surround71 ALSA lib setup.c:565:(add_elem) Cannot obtain info for CTL elem (MIXER,'IEC958 Playback Default',0,0,0): No such file or directory ALSA lib setup.c:565:(add_elem) Cannot obtain info for CTL elem (MIXER,'IEC958 Playback Default',0,0,0): No such file or directory ALSA lib setup.c:565:(add_elem) Cannot obtain info for CTL elem (MIXER,'IEC958 Playback Default',0,0,0): No such file or directory ALSA lib pcm.c:2217:(snd_pcm_open_noupdate) Unknown PCM cards.pcm.hdmi ALSA lib pcm.c:2217:(snd_pcm_open_noupdate) Unknown PCM cards.pcm.hdmi ALSA lib pcm.c:2217:(snd_pcm_open_noupdate) Unknown PCM cards.pcm.modem ALSA lib pcm.c:2217:(snd_pcm_open_noupdate) Unknown PCM cards.pcm.modem ALSA lib pcm.c:2217:(snd_pcm_open_noupdate) Unknown PCM cards.pcm.phoneline ALSA lib pcm.c:2217:(snd_pcm_open_noupdate) Unknown PCM cards.pcm.phoneline ALSA lib audio/pcm_bluetooth.c:1614:(audioservice_expect) BT_GET_CAPABILITIES failed : Input/output error(5) ALSA lib audio/pcm_bluetooth.c:1614:(audioservice_expect) BT_GET_CAPABILITIES failed : Input/output error(5) ALSA lib audio/pcm_bluetooth.c:1614:(audioservice_expect) BT_GET_CAPABILITIES failed : Input/output error(5) ALSA lib audio/pcm_bluetooth.c:1614:(audioservice_expect) BT_GET_CAPABILITIES failed : Input/output error(5) ALSA lib pcm_dmix.c:957:(snd_pcm_dmix_open) The dmix plugin supports only playback stream Cannot connect to server socket err = No such file or directory Cannot connect to server socket jack server is not running or cannot be started video device: /dev/video0 Init. FaceTime HD Camera (Built-in) (location: usb-0000:00:0b.0-1) { pixelformat = 'YUYV', description = 'YUV 4:2:2 (YUYV)' } { discrete: width = 160, height = 120 } Time interval between frame: 100/2997, 1/25, 1/24, 1/15, { discrete: width = 176, height = 144 } Time interval between frame: 100/2997, 1/25, 1/24, 1/15, { discrete: width = 320, height = 240 } Time interval between frame: 100/2997, 1/25, 1/24, 1/15, { discrete: width = 352, height = 288 } Time interval between frame: 100/2997, 1/25, 1/24, 1/15, { discrete: width = 640, height = 480 } Time interval between frame: 100/2997, 1/25, 1/24, 1/15, { discrete: width = 1280, height = 720 } Time interval between frame: 1/10, { pixelformat = 'MJPG', description = 'MJPEG' } { discrete: width = 960, height = 540 } Time interval between frame: 100/2997, 1/25, 1/24, 1/15, { discrete: width = 1024, height = 576 } Time interval between frame: 100/2997, 1/25, 1/24, 1/15, { discrete: width = 1280, height = 720 } Time interval between frame: 100/2997, 1/25, 1/24, 1/15, { pixelformat = 'RGB3', description = 'RGB3' } { discrete: width = 160, height = 120 } Time interval between frame: 100/2997, 1/25, 1/24, 1/15, { discrete: width = 176, height = 144 } Time interval between frame: 100/2997, 1/25, 1/24, 1/15, { discrete: width = 320, height = 240 } Time interval between frame: 100/2997, 1/25, 1/24, 1/15, { discrete: width = 352, height = 288 } Time interval between frame: 100/2997, 1/25, 1/24, 1/15, { discrete: width = 640, height = 480 } Time interval between frame: 100/2997, 1/25, 1/24, 1/15, { discrete: width = 1280, height = 720 } Time interval between frame: 100/2997, 1/25, 1/24, 1/15, { discrete: width = 960, height = 540 } Time interval between frame: 100/2997, 1/25, 1/24, 1/15, { discrete: width = 1024, height = 576 } Time interval between frame: 100/2997, 1/25, 1/24, 1/15, { pixelformat = 'BGR3', description = 'BGR3' } { discrete: width = 160, height = 120 } Time interval between frame: 100/2997, 1/25, 1/24, 1/15, { discrete: width = 176, height = 144 } Time interval between frame: 100/2997, 1/25, 1/24, 1/15, { discrete: width = 320, height = 240 } Time interval between frame: 100/2997, 1/25, 1/24, 1/15, { discrete: width = 352, height = 288 } Time interval between frame: 100/2997, 1/25, 1/24, 1/15, { discrete: width = 640, height = 480 } Time interval between frame: 100/2997, 1/25, 1/24, 1/15, { discrete: width = 1280, height = 720 } Time interval between frame: 100/2997, 1/25, 1/24, 1/15, { discrete: width = 960, height = 540 } Time interval between frame: 100/2997, 1/25, 1/24, 1/15, { discrete: width = 1024, height = 576 } Time interval between frame: 100/2997, 1/25, 1/24, 1/15, { pixelformat = 'YU12', description = 'YU12' } { discrete: width = 160, height = 120 } Time interval between frame: 100/2997, 1/25, 1/24, 1/15, { discrete: width = 176, height = 144 } Time interval between frame: 100/2997, 1/25, 1/24, 1/15, { discrete: width = 320, height = 240 } Time interval between frame: 100/2997, 1/25, 1/24, 1/15, { discrete: width = 352, height = 288 } Time interval between frame: 100/2997, 1/25, 1/24, 1/15, { discrete: width = 640, height = 480 } Time interval between frame: 100/2997, 1/25, 1/24, 1/15, { discrete: width = 1280, height = 720 } Time interval between frame: 100/2997, 1/25, 1/24, 1/15, { discrete: width = 960, height = 540 } Time interval between frame: 100/2997, 1/25, 1/24, 1/15, { discrete: width = 1024, height = 576 } Time interval between frame: 100/2997, 1/25, 1/24, 1/15, { pixelformat = 'YV12', description = 'YV12' } { discrete: width = 160, height = 120 } Time interval between frame: 100/2997, 1/25, 1/24, 1/15, { discrete: width = 176, height = 144 } Time interval between frame: 100/2997, 1/25, 1/24, 1/15, { discrete: width = 320, height = 240 } Time interval between frame: 100/2997, 1/25, 1/24, 1/15, { discrete: width = 352, height = 288 } Time interval between frame: 100/2997, 1/25, 1/24, 1/15, { discrete: width = 640, height = 480 } Time interval between frame: 100/2997, 1/25, 1/24, 1/15, { discrete: width = 1280, height = 720 } Time interval between frame: 100/2997, 1/25, 1/24, 1/15, { discrete: width = 960, height = 540 } Time interval between frame: 100/2997, 1/25, 1/24, 1/15, { discrete: width = 1024, height = 576 } Time interval between frame: 100/2997, 1/25, 1/24, 1/15, vid:05ac pid:8509 driver:uvcvideo checking format: 1196444237 VIDIOC_G_COMP:: Invalid argument compression control not supported fps is set to 1/25 drawing controls no codec detected for H264 no codec detected for MP3 - (lavc) Checking video mode 960x540@32bpp : OK Could not grab image (select timeout): Resource temporarily unavailable Could not grab image (select timeout): Resource temporarily unavailable Could not grab image (select timeout): Resource temporarily unavailable Could not grab image (select timeout): Resource temporarily unavailable Could not grab image (select timeout): Resource temporarily unavailable Could not grab image (select timeout): Resource temporarily unavailable Could not grab image (select timeout): Resource temporarily unavailable Could not grab image (select timeout): Resource temporarily unavailable Could not grab image (select timeout): Resource temporarily unavailable write /home/ubuntu/.guvcviewrc OK free controls cleaned allocations - 100% Closing portaudio ...OK Closing GTK... OK ubuntu@ubuntu:~$ Any help would be greatly appreciated. Only clue I have is that I initially was having problems, tried using the old method of fixing isights (involving installing isight-firmware-tools) before realizing that I just hadn't turned on the VM setting to allow the VM to access the webcam. :) Anyway, I wonder if installing that messed something up. However, I think this is a red herring because I've: shut down and turned back on the Mac, restarted the VM, tried a different VM (for which I never installed isight-firmware-tools, and created an entirely new ubuntu vm. All instances have had this problem. Similarly, other viewers, such as cheese, avplay, avconv have had all various kinds of errors.

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  • Pure C# open source PCM to Ogg convertor?

    - by Ole Jak
    Microsoft Silverlight 4 is in beta. It supports PCM audio output. It would be madness to stream PCM over internet (for ex in P2P chart webApp) so we need Pure C# open source PCM to Ogg convertor. No unmanaged code, nothing going out of .net sandbox. So does any one know such Pure C# open source PCM to Ogg convertor? What do I need: Open Source Libs for encoding. Tutorials and blog articles on How to do it, about etc. BTW: why Pure C#? - because Silverlight 4 does not support unmanaged or just not C# DLL's. BTW2: this question is similar to this one but it is different because Ogg is Open Source, free while mp3 will not be free until 2010

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  • detecting pauses in a spoken word audio file using pymad, pcm, vad, etc

    - by james
    First I am going to broadly state what I'm trying to do and ask for advice. Then I will explain my current approach and ask for answers to my current problems. Problem I have an MP3 file of a person speaking. I'd like to split it up into segments roughly corresponding to a sentence or phrase. (I'd do it manually, but we are talking hours of data.) If you have advice on how to do this programatically or for some existing utilities, I'd love to hear it. (I'm aware of voice activity detection and I've looked into it a bit, but I didn't see any freely available utilities.) Current Approach I thought the simplest thing would be to scan the MP3 at certain intervals and identify places where the average volume was below some threshold. Then I would use some existing utility to cut up the mp3 at those locations. I've been playing around with pymad and I believe that I've successfully extracted the PCM (pulse code modulation) data for each frame of the mp3. Now I am stuck because I can't really seem to wrap my head around how the PCM data translates to relative volume. I'm also aware of other complicating factors like multiple channels, big endian vs little, etc. Advice on how to map a group of pcm samples to relative volume would be key. Thanks!

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  • Visualizing volume of PCM samples

    - by genevincent
    I have several chunks of PCM audio (G.711) in my C++ application. I would like to visualize the different audio volume in each of these chunks. My first attempt was to calculate the average of the sample values for each chunk and use that as an a volume indicator, but this doesn't work well. I do get 0 for chunks with silence and differing values for chunks with audio, but the values only differ slighly and don't seem to resemble the actual volume. What would be a better algorithem calculate the volume ? I hear G.711 audio is logarithmic PCM. How should I take that into account ?

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  • Silverlight 4 - encoding PCM data from the microphone

    - by Richard
    Hi I've written a basic SL4 application to capture audio data from the microphone using CaptureSource. The trouble is, it's raw PCM output - which means huge and uncompressed. Given that I need this application to run purely within a SL4 environment, how can I compress the PCM audio data into something that can be delivered to a remote server more easily? In conversation, people have suggested Speex and WMA for instance, but I haven't found any libraries or examples that work without requiring reference to DLL's that won't work in a SL4 project. Thanks, Richard.

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  • Getting PCM values of WAV files

    - by user2431088
    I have a .wav mono file (16bit,44.1kHz) and im using this code below. If im not wrong, this would give me an output of values between -1 and 1 which i can apply FFT on ( to be converted to a spectrogram later on). However, my output is no where near -1 and 1. This is a portion of my output 7.01214599609375 17750.2552337646 8308.42733764648 0.000274658203125 1.00001525878906 0.67291259765625 1.3458251953125 16.0000305175781 24932 758.380676269531 0.0001068115234375 This is the code which i got from another post Edit 1: public static Double[] prepare(String wavePath, out int SampleRate) { Double[] data; byte[] wave; byte[] sR = new byte[4]; System.IO.FileStream WaveFile = System.IO.File.OpenRead(wavePath); wave = new byte[WaveFile.Length]; data = new Double[(wave.Length - 44) / 4];//shifting the headers out of the PCM data; WaveFile.Read(wave, 0, Convert.ToInt32(WaveFile.Length));//read the wave file into the wave variable /***********Converting and PCM accounting***************/ for (int i = 0; i < data.Length; i += 2) { data[i] = BitConverter.ToInt16(wave, i) / 32768.0; } /**************assigning sample rate**********************/ for (int i = 24; i < 28; i++) { sR[i - 24] = wave[i]; } SampleRate = BitConverter.ToInt16(sR, 0); return data; }

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  • Conceal packet loss in PCM stream

    - by ZeroDefect
    I am looking to use 'Packet Loss Concealment' to conceal lost PCM frames in an audio stream. Unfortunately, I cannot find a library that is accessible without all the licensing restrictions and code bloat (...up for some suggestions though). I have located some GPL code written by Steve Underwood for the Asterisk project which implements PLC. There are several limitations; although, as Steve suggests in his code, his algorithm can be applied to different streams with a bit of work. Currently, the code works with 8kHz 16-bit signed mono streams. Variations of the code can be found through a simple search of Google Code Search. My hope is that I can adapt the code to work with other streams. Initially, the goal is to adjust the algorithm for 8+ kHz, 16-bit signed, multichannel audio (all in a C++ environment). Eventually, I'm looking to make the code available under the GPL license in hopes that it could be of benefit to others... Attached is the code below with my efforts. The code includes a main function that will "drop" a number of frames with a given probability. Unfortunately, the code does not quite work as expected. I'm receiving EXC_BAD_ACCESS when running in gdb, but I don't get a trace from gdb when using 'bt' command. Clearly, I'm trampimg on memory some where but not sure exactly where. When I comment out the *amdf_pitch* function, the code runs without crashing... int main (int argc, char *argv[]) { std::ifstream fin("C:\\cc32kHz.pcm"); if(!fin.is_open()) { std::cout << "Failed to open input file" << std::endl; return 1; } std::ofstream fout_repaired("C:\\cc32kHz_repaired.pcm"); if(!fout_repaired.is_open()) { std::cout << "Failed to open output repaired file" << std::endl; return 1; } std::ofstream fout_lossy("C:\\cc32kHz_lossy.pcm"); if(!fout_lossy.is_open()) { std::cout << "Failed to open output repaired file" << std::endl; return 1; } audio::PcmConcealer Concealer; Concealer.Init(1, 16, 32000); //Generate random numbers; srand( time(NULL) ); int value = 0; int probability = 5; while(!fin.eof()) { char arr[2]; fin.read(arr, 2); //Generate's random number; value = rand() % 100 + 1; if(value <= probability) { char blank[2] = {0x00, 0x00}; fout_lossy.write(blank, 2); //Fill in data; Concealer.Fill((int16_t *)blank, 1); fout_repaired.write(blank, 2); } else { //Write data to file; fout_repaired.write(arr, 2); fout_lossy.write(arr, 2); Concealer.Receive((int16_t *)arr, 1); } } fin.close(); fout_repaired.close(); fout_lossy.close(); return 0; } PcmConcealer.hpp /* * Code adapted from Steve Underwood of the Asterisk Project. This code inherits * the same licensing restrictions as the Asterisk Project. */ #ifndef __PCMCONCEALER_HPP__ #define __PCMCONCEALER_HPP__ /** 1. What does it do? The packet loss concealment module provides a suitable synthetic fill-in signal, to minimise the audible effect of lost packets in VoIP applications. It is not tied to any particular codec, and could be used with almost any codec which does not specify its own procedure for packet loss concealment. Where a codec specific concealment procedure exists, the algorithm is usually built around knowledge of the characteristics of the particular codec. It will, therefore, generally give better results for that particular codec than this generic concealer will. 2. How does it work? While good packets are being received, the plc_rx() routine keeps a record of the trailing section of the known speech signal. If a packet is missed, plc_fillin() is called to produce a synthetic replacement for the real speech signal. The average mean difference function (AMDF) is applied to the last known good signal, to determine its effective pitch. Based on this, the last pitch period of signal is saved. Essentially, this cycle of speech will be repeated over and over until the real speech resumes. However, several refinements are needed to obtain smooth pleasant sounding results. - The two ends of the stored cycle of speech will not always fit together smoothly. This can cause roughness, or even clicks, at the joins between cycles. To soften this, the 1/4 pitch period of real speech preceeding the cycle to be repeated is blended with the last 1/4 pitch period of the cycle to be repeated, using an overlap-add (OLA) technique (i.e. in total, the last 5/4 pitch periods of real speech are used). - The start of the synthetic speech will not always fit together smoothly with the tail of real speech passed on before the erasure was identified. Ideally, we would like to modify the last 1/4 pitch period of the real speech, to blend it into the synthetic speech. However, it is too late for that. We could have delayed the real speech a little, but that would require more buffer manipulation, and hurt the efficiency of the no-lost-packets case (which we hope is the dominant case). Instead we use a degenerate form of OLA to modify the start of the synthetic data. The last 1/4 pitch period of real speech is time reversed, and OLA is used to blend it with the first 1/4 pitch period of synthetic speech. The result seems quite acceptable. - As we progress into the erasure, the chances of the synthetic signal being anything like correct steadily fall. Therefore, the volume of the synthesized signal is made to decay linearly, such that after 50ms of missing audio it is reduced to silence. - When real speech resumes, an extra 1/4 pitch period of sythetic speech is blended with the start of the real speech. If the erasure is small, this smoothes the transition. If the erasure is long, and the synthetic signal has faded to zero, the blending softens the start up of the real signal, avoiding a kind of "click" or "pop" effect that might occur with a sudden onset. 3. How do I use it? Before audio is processed, call plc_init() to create an instance of the packet loss concealer. For each received audio packet that is acceptable (i.e. not including those being dropped for being too late) call plc_rx() to record the content of the packet. Note this may modify the packet a little after a period of packet loss, to blend real synthetic data smoothly. When a real packet is not available in time, call plc_fillin() to create a sythetic substitute. That's it! */ /*! Minimum allowed pitch (66 Hz) */ #define PLC_PITCH_MIN(SAMPLE_RATE) ((double)(SAMPLE_RATE) / 66.6) /*! Maximum allowed pitch (200 Hz) */ #define PLC_PITCH_MAX(SAMPLE_RATE) ((SAMPLE_RATE) / 200) /*! Maximum pitch OLA window */ //#define PLC_PITCH_OVERLAP_MAX(SAMPLE_RATE) ((PLC_PITCH_MIN(SAMPLE_RATE)) >> 2) /*! The length over which the AMDF function looks for similarity (20 ms) */ #define CORRELATION_SPAN(SAMPLE_RATE) ((20 * (SAMPLE_RATE)) / 1000) /*! History buffer length. The buffer must also be at leat 1.25 times PLC_PITCH_MIN, but that is much smaller than the buffer needs to be for the pitch assessment. */ //#define PLC_HISTORY_LEN(SAMPLE_RATE) ((CORRELATION_SPAN(SAMPLE_RATE)) + (PLC_PITCH_MIN(SAMPLE_RATE))) namespace audio { typedef struct { /*! Consecutive erased samples */ int missing_samples; /*! Current offset into pitch period */ int pitch_offset; /*! Pitch estimate */ int pitch; /*! Buffer for a cycle of speech */ float *pitchbuf;//[PLC_PITCH_MIN]; /*! History buffer */ short *history;//[PLC_HISTORY_LEN]; /*! Current pointer into the history buffer */ int buf_ptr; } plc_state_t; class PcmConcealer { public: PcmConcealer(); ~PcmConcealer(); void Init(int channels, int bit_depth, int sample_rate); //Process a block of received audio samples. int Receive(short amp[], int frames); //Fill-in a block of missing audio samples. int Fill(short amp[], int frames); void Destroy(); private: int amdf_pitch(int min_pitch, int max_pitch, short amp[], int channel_index, int frames); void save_history(plc_state_t *s, short *buf, int channel_index, int frames); void normalise_history(plc_state_t *s); /** Holds the states of each of the channels **/ std::vector< plc_state_t * > ChannelStates; int plc_pitch_min; int plc_pitch_max; int plc_pitch_overlap_max; int correlation_span; int plc_history_len; int channel_count; int sample_rate; bool Initialized; }; } #endif PcmConcealer.cpp /* * Code adapted from Steve Underwood of the Asterisk Project. This code inherits * the same licensing restrictions as the Asterisk Project. */ #include "audio/PcmConcealer.hpp" /* We do a straight line fade to zero volume in 50ms when we are filling in for missing data. */ #define ATTENUATION_INCREMENT 0.0025 /* Attenuation per sample */ #if !defined(INT16_MAX) #define INT16_MAX (32767) #define INT16_MIN (-32767-1) #endif #ifdef WIN32 inline double rint(double x) { return floor(x + 0.5); } #endif inline short fsaturate(double damp) { if (damp > 32767.0) return INT16_MAX; if (damp < -32768.0) return INT16_MIN; return (short)rint(damp); } namespace audio { PcmConcealer::PcmConcealer() : Initialized(false) { } PcmConcealer::~PcmConcealer() { Destroy(); } void PcmConcealer::Init(int channels, int bit_depth, int sample_rate) { if(Initialized) return; if(channels <= 0 || bit_depth != 16) return; Initialized = true; channel_count = channels; this->sample_rate = sample_rate; ////////////// double min = PLC_PITCH_MIN(sample_rate); int imin = (int)min; double max = PLC_PITCH_MAX(sample_rate); int imax = (int)max; plc_pitch_min = imin; plc_pitch_max = imax; plc_pitch_overlap_max = (plc_pitch_min >> 2); correlation_span = CORRELATION_SPAN(sample_rate); plc_history_len = correlation_span + plc_pitch_min; ////////////// for(int i = 0; i < channel_count; i ++) { plc_state_t *t = new plc_state_t; memset(t, 0, sizeof(plc_state_t)); t->pitchbuf = new float[plc_pitch_min]; t->history = new short[plc_history_len]; ChannelStates.push_back(t); } } void PcmConcealer::Destroy() { if(!Initialized) return; while(ChannelStates.size()) { plc_state_t *s = ChannelStates.at(0); if(s) { if(s->history) delete s->history; if(s->pitchbuf) delete s->pitchbuf; memset(s, 0, sizeof(plc_state_t)); delete s; } ChannelStates.erase(ChannelStates.begin()); } ChannelStates.clear(); Initialized = false; } //Process a block of received audio samples. int PcmConcealer::Receive(short amp[], int frames) { if(!Initialized) return 0; int j = 0; for(int k = 0; k < ChannelStates.size(); k++) { int i; int overlap_len; int pitch_overlap; float old_step; float new_step; float old_weight; float new_weight; float gain; plc_state_t *s = ChannelStates.at(k); if (s->missing_samples) { /* Although we have a real signal, we need to smooth it to fit well with the synthetic signal we used for the previous block */ /* The start of the real data is overlapped with the next 1/4 cycle of the synthetic data. */ pitch_overlap = s->pitch >> 2; if (pitch_overlap > frames) pitch_overlap = frames; gain = 1.0 - s->missing_samples * ATTENUATION_INCREMENT; if (gain < 0.0) gain = 0.0; new_step = 1.0/pitch_overlap; old_step = new_step*gain; new_weight = new_step; old_weight = (1.0 - new_step)*gain; for (i = 0; i < pitch_overlap; i++) { int index = (i * channel_count) + j; amp[index] = fsaturate(old_weight * s->pitchbuf[s->pitch_offset] + new_weight * amp[index]); if (++s->pitch_offset >= s->pitch) s->pitch_offset = 0; new_weight += new_step; old_weight -= old_step; if (old_weight < 0.0) old_weight = 0.0; } s->missing_samples = 0; } save_history(s, amp, j, frames); j++; } return frames; } //Fill-in a block of missing audio samples. int PcmConcealer::Fill(short amp[], int frames) { if(!Initialized) return 0; int j =0; for(int k = 0; k < ChannelStates.size(); k++) { short *tmp = new short[plc_pitch_overlap_max]; int i; int pitch_overlap; float old_step; float new_step; float old_weight; float new_weight; float gain; short *orig_amp; int orig_len; orig_amp = amp; orig_len = frames; plc_state_t *s = ChannelStates.at(k); if (s->missing_samples == 0) { // As the gap in real speech starts we need to assess the last known pitch, //and prepare the synthetic data we will use for fill-in normalise_history(s); s->pitch = amdf_pitch(plc_pitch_min, plc_pitch_max, s->history + plc_history_len - correlation_span - plc_pitch_min, j, correlation_span); // We overlap a 1/4 wavelength pitch_overlap = s->pitch >> 2; // Cook up a single cycle of pitch, using a single of the real signal with 1/4 //cycle OLA'ed to make the ends join up nicely // The first 3/4 of the cycle is a simple copy for (i = 0; i < s->pitch - pitch_overlap; i++) s->pitchbuf[i] = s->history[plc_history_len - s->pitch + i]; // The last 1/4 of the cycle is overlapped with the end of the previous cycle new_step = 1.0/pitch_overlap; new_weight = new_step; for ( ; i < s->pitch; i++) { s->pitchbuf[i] = s->history[plc_history_len - s->pitch + i]*(1.0 - new_weight) + s->history[plc_history_len - 2*s->pitch + i]*new_weight; new_weight += new_step; } // We should now be ready to fill in the gap with repeated, decaying cycles // of what is in pitchbuf // We need to OLA the first 1/4 wavelength of the synthetic data, to smooth // it into the previous real data. To avoid the need to introduce a delay // in the stream, reverse the last 1/4 wavelength, and OLA with that. gain = 1.0; new_step = 1.0/pitch_overlap; old_step = new_step; new_weight = new_step; old_weight = 1.0 - new_step; for (i = 0; i < pitch_overlap; i++) { int index = (i * channel_count) + j; amp[index] = fsaturate(old_weight * s->history[plc_history_len - 1 - i] + new_weight * s->pitchbuf[i]); new_weight += new_step; old_weight -= old_step; if (old_weight < 0.0) old_weight = 0.0; } s->pitch_offset = i; } else { gain = 1.0 - s->missing_samples*ATTENUATION_INCREMENT; i = 0; } for ( ; gain > 0.0 && i < frames; i++) { int index = (i * channel_count) + j; amp[index] = s->pitchbuf[s->pitch_offset]*gain; gain -= ATTENUATION_INCREMENT; if (++s->pitch_offset >= s->pitch) s->pitch_offset = 0; } for ( ; i < frames; i++) { int index = (i * channel_count) + j; amp[i] = 0; } s->missing_samples += orig_len; save_history(s, amp, j, frames); delete [] tmp; j++; } return frames; } void PcmConcealer::save_history(plc_state_t *s, short *buf, int channel_index, int frames) { if (frames >= plc_history_len) { /* Just keep the last part of the new data, starting at the beginning of the buffer */ //memcpy(s->history, buf + len - plc_history_len, sizeof(short)*plc_history_len); int frames_to_copy = plc_history_len; for(int i = 0; i < frames_to_copy; i ++) { int index = (channel_count * (i + frames - plc_history_len)) + channel_index; s->history[i] = buf[index]; } s->buf_ptr = 0; return; } if (s->buf_ptr + frames > plc_history_len) { /* Wraps around - must break into two sections */ //memcpy(s->history + s->buf_ptr, buf, sizeof(short)*(plc_history_len - s->buf_ptr)); short *hist_ptr = s->history + s->buf_ptr; int frames_to_copy = plc_history_len - s->buf_ptr; for(int i = 0; i < frames_to_copy; i ++) { int index = (channel_count * i) + channel_index; hist_ptr[i] = buf[index]; } frames -= (plc_history_len - s->buf_ptr); //memcpy(s->history, buf + (plc_history_len - s->buf_ptr), sizeof(short)*len); frames_to_copy = frames; for(int i = 0; i < frames_to_copy; i ++) { int index = (channel_count * (i + (plc_history_len - s->buf_ptr))) + channel_index; s->history[i] = buf[index]; } s->buf_ptr = frames; return; } /* Can use just one section */ //memcpy(s->history + s->buf_ptr, buf, sizeof(short)*len); short *hist_ptr = s->history + s->buf_ptr; int frames_to_copy = frames; for(int i = 0; i < frames_to_copy; i ++) { int index = (channel_count * i) + channel_index; hist_ptr[i] = buf[index]; } s->buf_ptr += frames; } void PcmConcealer::normalise_history(plc_state_t *s) { short *tmp = new short[plc_history_len]; if (s->buf_ptr == 0) return; memcpy(tmp, s->history, sizeof(short)*s->buf_ptr); memcpy(s->history, s->history + s->buf_ptr, sizeof(short)*(plc_history_len - s->buf_ptr)); memcpy(s->history + plc_history_len - s->buf_ptr, tmp, sizeof(short)*s->buf_ptr); s->buf_ptr = 0; delete [] tmp; } int PcmConcealer::amdf_pitch(int min_pitch, int max_pitch, short amp[], int channel_index, int frames) { int i; int j; int acc; int min_acc; int pitch; pitch = min_pitch; min_acc = INT_MAX; for (i = max_pitch; i <= min_pitch; i++) { acc = 0; for (j = 0; j < frames; j++) { int index1 = (channel_count * (i+j)) + channel_index; int index2 = (channel_count * j) + channel_index; //std::cout << "Index 1: " << index1 << ", Index 2: " << index2 << std::endl; acc += abs(amp[index1] - amp[index2]); } if (acc < min_acc) { min_acc = acc; pitch = i; } } std::cout << "Pitch: " << pitch << std::endl; return pitch; } } P.S. - I must confess that digital audio is not my forte...

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  • Correct way to Convert 16bit PCM Wave data to float

    - by fredley
    I have a wave file in 16bit PCM form. I've got the raw data in a byte[] and a method for extracting samples, and I need them in float format, i.e. a float[] to do a Fourier Transform. Here's my code, does this look right? I'm working on Android so javax.sound.sampled etc. is not available. private static short getSample(byte[] buffer, int position) { return (short) (((buffer[position + 1] & 0xff) << 8) | (buffer[position] & 0xff)); } ... float[] samples = new float[samplesLength]; for (int i = 0;i<input.length/2;i+=2){ samples[i/2] = (float)getSample(input,i) / (float)Short.MAX_VALUE; }

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  • Properly trimming PCM data from a ByteArray

    - by Lowgain
    I have a situation where I need to trim a small amount of audio from the beginning of a recorded clip (generally somewhere between 110-150ms, it is an inconsistent amount). I'm recording in 44100 frequency and 16 bitrate. This is the code I'm using: public function get trimmedData():ByteArray { var ba:ByteArray = new ByteArray(); var bitPosition:uint = 44100 * 16 * (recordGap / 1000); bitPosition -= int(bitPosition % 16); //should keep snapped to nearest sample, I hope ba.writeBytes(_rawData, (bitPosition / 8)); return ba; } This seems to work time-wise, but all the recorded audio gets staticy and gross. Is something off about my rounding? This is the first time I've needed to alter raw PCM data so I'm not sure about the finer details of it. Thanks!

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  • How to convert from wav or mp3 to raw PCM [on hold]

    - by Komyg
    I am developing a game using Cocos2d-X and Marmalade SDK, and I am looking for any recommendations of programs that can convert audio files in mp3 or wav format to raw PCM 16 format. The problem is that I am using the SimpleAudioEngine class to play sounds in my game and in Marmalade it only supports files that are encoded as raw PCM 16. Unfortunately I've been having a very hard time finding a program that can do this type of conversion, so I am looking for a recommendation.

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  • Void* array casting to float, int32, int16, etc.

    - by Griffin
    Hey guys, I've got an array of PCM data, it could be 16 bit, 24 bit packed, 32 bit, etc.. It could be signed, or unsigned, and it could be 32 or 64 bit floating point. It is currently stored as a "void**" matrix, indexed by channel, then by frame. The goal is to allow my library to take in any PCM format and buffer it, without requiring manipulation of the data to fit a designated structure. If the A/D converter spits out 24 bit packed arrays of interleaved PCM, I need to accept it gracefully. I also need to support 16 bit non interleaved, as well as any permutation of the above formats. I know the bit depth and other information at runtime, and I'm trying to code efficiently while not duplicating code. What I need is an effective way to cast the matrix, put PCM data into the matrix, and then pull it out later. I can cast the matrix to int32_t, or int16_t for the 32 and 16 bit signed PCM respectively, I'll probably have to store the 24 bit PCM in an int32_t for 32 bit, 8 bit byte systems as well. Can anyone recommend a good way to put data into this array, and pull it out later? I'd like to avoid large sections of code which look like: switch( mFormat ) { case 1: // unsigned 8 bit for( int i = 0; i < mChannels; i++ ) framesArray = (uint8_t*)pcm[i]; break; case 2: // signed 8 bit for( int i = 0; i < mChannels; i++ ) framesArray = (int8_t*)pcm[i]; break; case 3: // unsigned 16 bit ... Limitations: I'm working in C/C++, no templates, no RTTI, no STL. Think embedded. Things get trickier when I have to port this to a DSP with 16 bit bytes. Does anybody have any useful macros they might be willing to share? Thanks, -Griff

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  • How to convert MP3 tp PCM using delphi code?

    - by XBasic3000
    I have TBass from http://www.un4seen.com/bass.html. I load mp3 and triying to change the format to PCM but it give me same result of mp3? acmForm.wFormatTag :=1; acmForm.nChannels :=1; acmForm.nSamplesPerSec :=8000; acmForm.nAvgBytesPerSec:=16000; acmForm.nBlockAlign := 2; acmForm.wBitsPerSample := 16; acmForm.cbSize := 0;

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  • How to convert pcm to mp3?

    - by avirk
    I have some .pcm files and I want to convert them on high quality .mp3 format. I tried to find tools by Google search but did not get the right one for me. I will prefer the freeware but if there is not a good freeware then I can also consider the shareware. The pcm format has much large files as I have 200-500 mb so the tool should be able to handle the large files. Please help me regard this problem.

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  • Faster way to convert from 24 bit wav pcm format to float?

    - by LMO
    I need to read data in from a wav file in 24 bit pcm format, and convert to float. I'm using Python 2.7.2. The wave package reads the data in as a string, so what I've tried is: # read in entire wav file wdata = f.readframes(nFrames) # unpack into signed integers and convert to float data = array.array('f') for i in range(0,nFrames*3,3): data.append(float(struct.unpack('<i', '\x00'+ wdata[i:i+3])[0])) # normalize sample values data = data / 0x800000 This is quite a bit faster than my earlier approaches, but still quite slow. Can anyone suggest a more efficient method?

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  • Difficulty porting raw PCM output code from Java to Android AudioTrack API.

    - by IndigoParadox
    I'm attempting to port an application that plays chiptunes (NSF, SPC, etc) music files from Java SE to Android. The Android API seems to lack the javax multimedia classes that this application uses to output raw PCM audio. The closest analog I've found in the API is AudioTrack and so I've been wrestling with that. However, when I try to run one of my sample music files through my port-in-progress, all I get back is static. My suspicion is that it's the AudioTrack I've setup which is at fault. I've tried various different constructors but it all just outputs static in the end. The DataLine setup in the original code is something like: AudioFormat audioFormat = new AudioFormat( AudioFormat.Encoding.PCM_SIGNED, 44100, 16, 2, 4, 44100, true ); DataLine.Info lineInfo = new DataLine.Info( SourceDataLine.class, audioFormat ); DataLine line = (SourceDataLine)AudioSystem.getLine( lineInfo ); The constructor I'm using right now is: AudioTrack = new AudioTrack( AudioManager.STREAM_MUSIC, 44100, AudioFormat.CHANNEL_CONFIGURATION_STEREO, AudioFormat.ENCODING_PCM_16BIT, AudioTrack.getMinBufferSize( 44100, AudioFormat.CHANNEL_CONFIGURATION_STEREO, AudioFormat.ENCODING_PCM_16BIT ), AudioTrack.MODE_STREAM ); I've replaced constants and variables in those so they make sense as concisely as possible, but my basic question is if there are any obvious problems in the assumptions I made when going from one format to the other.

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  • Amnesia doesn't start due to audio problems

    - by james
    I have a problem with amnesia game. After Intro and clicking continue button few times, when game is supposed to start it crashes. Here is console output: ALSA lib pcm_dmix.c:1018:(snd_pcm_dmix_open) unable to open slave ALSA lib pcm.c:2217:(snd_pcm_open_noupdate) Unknown PCM cards.pcm.rear ALSA lib pcm.c:2217:(snd_pcm_open_noupdate) Unknown PCM cards.pcm.center_lfe ALSA lib pcm.c:2217:(snd_pcm_open_noupdate) Unknown PCM cards.pcm.side ALSA lib audio/pcm_bluetooth.c:1614:(audioservice_expect) BT_GET_CAPABILITIES failed : Input/output error(5) ALSA lib audio/pcm_bluetooth.c:1614:(audioservice_expect) BT_GET_CAPABILITIES failed : Input/output error(5) ALSA lib audio/pcm_bluetooth.c:1614:(audioservice_expect) BT_GET_CAPABILITIES failed : Input/output error(5) ALSA lib audio/pcm_bluetooth.c:1614:(audioservice_expect) BT_GET_CAPABILITIES failed : Input/output error(5) ALSA lib pcm_dmix.c:957:(snd_pcm_dmix_open) The dmix plugin supports only playback stream ALSA lib pcm_dmix.c:1018:(snd_pcm_dmix_open) unable to open slave Cannot connect to server socket err = No such file or directory Cannot connect to server socket jack server is not running or cannot be started I should mention I have integrated both graphic and sound card.

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