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Thread: pulseaudio - no sound

  1. #1
    Join Date
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    Distro
    Ubuntu 11.04 Natty Narwhal

    pulseaudio - no sound

    I hear no sound with pulseaudio, but I hear the "ready for login" sound after the x server starts.

    OS: Ubuntu x86 10.04

    In syslog stands:
    Code:
    May 26 13:09:44 Gamma pulseaudio[2435]: module-alsa-card.c: Failed to find a working profile.
    May 26 13:09:44 Gamma pulseaudio[2435]: module.c: Failed to load  module "module-alsa-card" (argument: "device_id="1" name="1" card_name="alsa_card.1" tsched=yes ignore_dB=no card_properties="module-udev-detect.discovered=1""): initialization failed.
    My /etc/pulse/daemon.conf is:
    Code:
    ; daemonize = yes
    ; fail = yes
    ; disallow-module-loading = no
    ; disallow-exit = no
    ; use-pid-file = yes
    ; system-instance = no
    ; disable-shm = no
    ; shm-size-bytes = 0 # setting this 0 will use the system-default, usually 64 MiB
    ; high-priority = yes
    ; nice-level = -11
    
    ; realtime-scheduling = no
    ; realtime-priority = 5
    
    ; exit-idle-time = 20
    ; module-idle-time = 20
    ; scache-idle-time = 20
    
    ; dl-search-path = (depends on architecture)
    
    ; load-default-script-file = yes
    ; default-script-file = 
    
    ; log-target = auto
    ; log-level = notice
    
    resample-method = speex-float-1
    ; disable-remixing = no
    ; disable-lfe-remixing = yes
    
    ; no-cpu-limit = no
    
    ; rlimit-fsize = -1
    ; rlimit-data = -1
    ; rlimit-stack = -1
    ; rlimit-core = -1
    ; rlimit-as = -1
    ; rlimit-rss = -1
    ; rlimit-nproc = -1
    ; rlimit-nofile = 256
    ; rlimit-memlock = -1
    ; rlimit-locks = -1
    ; rlimit-sigpending = -1
    ; rlimit-msgqueue = -1
    ; rlimit-nice = 31
    ; rlimit-rtprio = 9
    ; rlimit-rtttime = 1000000
    
    default-sample-format = s16le
    ; default-sample-rate = 44100
    default-sample-channels = 6
    
    default-fragments = 8
    default-fragment-size-msec = 10
    My sound card:
    Code:
    lspci | grep -i audio
    01:08.0 Multimedia audio controller: C-Media Electronics Inc CM8738 (rev 10)
    Code:
    cat /proc/asound/cards
     0 [CMI8738        ]: CMI8738-MC6 - C-Media CMI8738
                          C-Media CMI8738 (model 55) at 0xd000, irq 18
     1 [UART           ]: MPU-401 UART - MPU-401 UART
                          MPU-401 UART at 0x330, irq 10
    Code:
    aplay -l
    card 0: CMI8738 [C-Media CMI8738], device 0: CMI8738-MC6 [C-Media PCI DAC/ADC]
      subdevice: 1/1
      subdevice #0: subdevice #0
    card 0: CMI8738 [C-Media CMI8738], device 1: CMI8738-MC6 [C-Media PCI 2nd DAC]
      subdevice: 1/1
      subdevice #0: subdevice #0
    card 0: CMI8738 [C-Media CMI8738], device 2: CMI8738-MC6 [C-Media PCI IEC958]
      subdevice: 1/1
      subdevice #0: subdevice #0
    Do anyone have a idea how I get the sound work? (without removing pulseaudio)

  2. #2
    Join Date
    Jun 2007
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    14,885

    Re: pulseaudio - no sound

    It looks like pulseaudio is grabbing the wrong alsa device (using device 1 when it should use device 0). First, back up your configuration file:
    Code:
    sudo cp /etc/pulse/default.pa /etc/pulse/default.pa.bak
    Now, edit your /etc/pulse/default.pa file:
    Code:
    gksu gedit /etc/pulse/default.pa
    Find the section of lines that describe module detection and make them look like this:
    Code:
    ### Load audio drivers statically (it's probably better to not load
    ### these drivers manually, but instead use module-hal-detect --
    ### see below -- for doing this automatically)
    load-module module-alsa-sink device=hw:0,0
    load-module module-alsa-source device=hw:1,0
    load-module module-oss device="/dev/dsp" sink_name=output source_name=input
    load-module module-oss-mmap device="/dev/dsp" sink_name=output source_name=input
    load-module module-null-sink
    load-module module-pipe-sink
    
    ### Automatically load driver modules depending on the hardware available
    #.ifexists module-udev-detect.so
    #load-module module-udev-detect
    #.else
    ### Alternatively use the static hardware detection module (for systems that
    ### lack udev support)
    load-module module-detect
    #.endif
    Save. Quit. Reboot. If that fails to work, restore your original default.pa:
    Code:
    sudo mv /etc/pulse/default.pa.bak /etc/pulse/default.pa
    Maybe someone more experienced with pulseaudio can then help you.

  3. #3
    Join Date
    Jan 2010
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    Germany
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    Distro
    Ubuntu 11.04 Natty Narwhal

    Re: pulseaudio - no sound

    I deactivated the UART sound card in BIOS (called "Midiport") and the error messages at pulseaudio and syslog disappeared.

    Your fix didn't work for me. I still get no audio. So I restored my /etc/pulse/default.pa.

  4. #4
    Join Date
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    Ubuntu Gnome 15.04 Vivid Vervet

    Re: pulseaudio - no sound

    Code:
             Brief Notes on C-Media 8338/8738/8768/8770 Driver
             =================================================
    
                       Takashi Iwai <tiwai@suse.de>
    
    
    Front/Rear Multi-channel Playback
    ---------------------------------
    
    CM8x38 chip can use ADC as the second DAC so that two different stereo
    channels can be used for front/rear playbacks.  Since there are two
    DACs, both streams are handled independently unlike the 4/6ch multi-
    channel playbacks in the section below.
    
    As default, ALSA driver assigns the first PCM device (i.e. hw:0,0 for
    card#0) for front and 4/6ch playbacks, while the second PCM device
    (hw:0,1) is assigned to the second DAC for rear playback.
    
    There are slight differences between the two DACs:
    
    - The first DAC supports U8 and S16LE formats, while the second DAC
      supports only S16LE.
    - The second DAC supports only two channel stereo.
    
    Please note that the CM8x38 DAC doesn't support continuous playback
    rate but only fixed rates: 5512, 8000, 11025, 16000, 22050, 32000,
    44100 and 48000 Hz.
    
    The rear output can be heard only when "Four Channel Mode" switch is
    disabled.  Otherwise no signal will be routed to the rear speakers.
    As default it's turned on.
    
    *** WARNING ***
    When "Four Channel Mode" switch is off, the output from rear speakers
    will be FULL VOLUME regardless of Master and PCM volumes.
    This might damage your audio equipment.  Please disconnect speakers
    before your turn off this switch.
    *** WARNING ***
    
    [ Well.. I once got the output with correct volume (i.e. same with the
      front one) and was so excited.  It was even with "Four Channel" bit
      on and "double DAC" mode.  Actually I could hear separate 4 channels
      from front and rear speakers!  But.. after reboot, all was gone.
      It's a very pity that I didn't save the register dump at that
      time..  Maybe there is an unknown register to achieve this... ]
    
    If your card has an extra output jack for the rear output, the rear
    playback should be routed there as default.  If not, there is a
    control switch in the driver "Line-In As Rear", which you can change
    via alsamixer or somewhat else.  When this switch is on, line-in jack
    is used as rear output.
    
    There are two more controls regarding to the rear output.
    The "Exchange DAC" switch is used to exchange front and rear playback
    routes, i.e. the 2nd DAC is output from front output.
    
    
    4/6 Multi-Channel Playback
    --------------------------
    
    The recent CM8738 chips support for the 4/6 multi-channel playback
    function.  This is useful especially for AC3 decoding.
    
    When the multi-channel is supported, the driver name has a suffix
    "-MC" such like "CMI8738-MC6".  You can check this name from
    /proc/asound/cards.
    
    When the 4/6-ch output is enabled, the second DAC accepts up to 6 (or
    4) channels.  While the dual DAC supports two different rates or
    formats, the 4/6-ch playback supports only the same condition for all
    channels.  Since the multi-channel playback mode uses both DACs, you
    cannot operate with full-duplex.
    
    The 4.0 and 5.1 modes are defined as the pcm "surround40" and "surround51"
    in alsa-lib.  For example, you can play a WAV file with 6 channels like
    
    	% aplay -Dsurround51 sixchannels.wav
    
    For programming the 4/6 channel playback, you need to specify the PCM
    channels as you like and set the format S16LE.  For example, for playback
    with 4 channels,
    
    	snd_pcm_hw_params_set_access(pcm, hw, SND_PCM_ACCESS_RW_INTERLEAVED);
    	    // or mmap if you like
    	snd_pcm_hw_params_set_format(pcm, hw, SND_PCM_FORMAT_S16_LE);
    	snd_pcm_hw_params_set_channels(pcm, hw, 4);
    
    and use the interleaved 4 channel data.
    
    There are some control switchs affecting to the speaker connections:
    
    "Line-In Mode"	- an enum control to change the behavior of line-in
    	jack.  Either "Line-In", "Rear Output" or "Bass Output" can
    	be selected.  The last item is available only with model 039
    	or newer. 
    	When "Rear Output" is chosen, the surround channels 3 and 4
    	are output to line-in jack.
    "Mic-In Mode"	- an enum control to change the behavior of mic-in
    	jack.  Either "Mic-In" or "Center/LFE Output" can be
    	selected. 
    	When "Center/LFE Output" is chosen, the center and bass
    	channels (channels 5 and 6) are output to mic-in jack. 
    
    Digital I/O
    -----------
    
    The CM8x38 provides the excellent SPDIF capability with very cheap
    price (yes, that's the reason I bought the card :)
    
    The SPDIF playback and capture are done via the third PCM device
    (hw:0,2).  Usually this is assigned to the PCM device "spdif".
    The available rates are 44100 and 48000 Hz.
    For playback with aplay, you can run like below:
    
    	% aplay -Dhw:0,2 foo.wav
    
    or
    
    	% aplay -Dspdif foo.wav
    
    24bit format is also supported experimentally.
    
    The playback and capture over SPDIF use normal DAC and ADC,
    respectively, so you cannot playback both analog and digital streams
    simultaneously.
    
    To enable SPDIF output, you need to turn on "IEC958 Output Switch"
    control via mixer or alsactl ("IEC958" is the official name of
    so-called S/PDIF).  Then you'll see the red light on from the card so
    you know that's working obviously :)
    The SPDIF input is always enabled, so you can hear SPDIF input data
    from line-out with "IEC958 In Monitor" switch at any time (see
    below).
    
    You can play via SPDIF even with the first device (hw:0,0),
    but SPDIF is enabled only when the proper format (S16LE), sample rate
    (441100 or 48000) and channels (2) are used.  Otherwise it's turned
    off.  (Also don't forget to turn on "IEC958 Output Switch", too.)
    
    
    Additionally there are relevant control switches:
    
    "IEC958 Mix Analog" - Mix analog PCM playback and FM-OPL/3 streams and
    	output through SPDIF.  This switch appears only on old chip
    	models (CM8738 033 and 037).
    	Note: without this control you can output PCM to SPDIF.
    	This is "mixing" of streams, so e.g. it's not for AC3 output
    	(see the next section).
    
    "IEC958 In Select"  - Select SPDIF input, the internal CD-in (false)
    	and the external input (true).
    
    "IEC958 Loop"       - SPDIF input data is loop back into SPDIF
    	output (aka bypass)
    
    "IEC958 Copyright"  - Set the copyright bit.
    
    "IEC958 5V"         - Select 0.5V (coax) or 5V (optical) interface.
    	On some cards this doesn't work and you need to change the
    	configuration with hardware dip-switch.
    
    "IEC958 In Monitor" - SPDIF input is routed to DAC.
    
    "IEC958 In Phase Inverse" - Set SPDIF input format as inverse.
    	[FIXME: this doesn't work on all chips..]
    
    "IEC958 In Valid"   - Set input validity flag detection.
    
    Note: When "PCM Playback Switch" is on, you'll hear the digital output
    stream through analog line-out.
    
    
    The AC3 (RAW DIGITAL) OUTPUT
    ----------------------------
    
    The driver supports raw digital (typically AC3) i/o over SPDIF.  This
    can be toggled via IEC958 playback control, but usually you need to
    access it via alsa-lib.  See alsa-lib documents for more details.
    
    On the raw digital mode, the "PCM Playback Switch" is automatically
    turned off so that non-audio data is heard from the analog line-out.
    Similarly the following switches are off: "IEC958 Mix Analog" and
    "IEC958 Loop".  The switches are resumed after closing the SPDIF PCM
    device automatically to the previous state.
    
    On the model 033, AC3 is implemented by the software conversion in
    the alsa-lib.  If you need to bypass the software conversion of IEC958
    subframes, pass the "soft_ac3=0" module option.  This doesn't matter
    on the newer models.
    
    
    ANALOG MIXER INTERFACE
    ----------------------
    
    The mixer interface on CM8x38 is similar to SB16.
    There are Master, PCM, Synth, CD, Line, Mic and PC Speaker playback
    volumes.  Synth, CD, Line and Mic have playback and capture switches,
    too, as well as SB16.
    
    In addition to the standard SB mixer, CM8x38 provides more functions.
    - PCM playback switch
    - PCM capture switch (to capture the data sent to DAC)
    - Mic Boost switch
    - Mic capture volume
    - Aux playback volume/switch and capture switch
    - 3D control switch
    
    
    MIDI CONTROLLER
    ---------------
    
    With CMI8338 chips, the MPU401-UART interface is disabled as default.
    You need to set the module option "mpu_port" to a valid I/O port address
    to enable MIDI support.  Valid I/O ports are 0x300, 0x310, 0x320 and
    0x330.  Choose a value that doesn't conflict with other cards.
    
    With CMI8738 and newer chips, the MIDI interface is enabled by default
    and the driver automatically chooses a port address.
    
    There is _no_ hardware wavetable function on this chip (except for
    OPL3 synth below).
    What's said as MIDI synth on Windows is a software synthesizer
    emulation.  On Linux use TiMidity or other softsynth program for
    playing MIDI music.
    
    
    FM OPL/3 Synth
    --------------
    
    The FM OPL/3 is also enabled as default only for the first card.
    Set "fm_port" module option for more cards.
    
    The output quality of FM OPL/3 is, however, very weird.
    I don't know why..
    
    CMI8768 and newer chips do not have the FM synth.
    
    
    Joystick and Modem
    ------------------
    
    The legacy joystick is supported.  To enable the joystick support, pass
    joystick_port=1 module option.  The value 1 means the auto-detection.
    If the auto-detection fails, try to pass the exact I/O address.
    
    The modem is enabled dynamically via a card control switch "Modem".
    
    
    Debugging Information
    ---------------------
    
    The registers are shown in /proc/asound/cardX/cmipci.  If you have any
    problem (especially unexpected behavior of mixer), please attach the
    output of this proc file together with the bug report.

  5. #5
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    Re: pulseaudio - no sound

    I tried pulseaudio at another user account and it works!

    My conclusion: There is a error at a configuration file of pulseaudio in my home folder.
    So I deleted the files in ~/.pulse and restarted pulseaudio.

    Now I hear sound! Problem solved!

  6. #6
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    Re: pulseaudio - no sound

    Quote Originally Posted by fireandfuel View Post
    I tried pulseaudio at another user account and it works!

    My conclusion: There is a error at a configuration file of pulseaudio in my home folder.
    So I deleted the files in ~/.pulse and restarted pulseaudio.

    Now I hear sound! Problem solved!
    Thanks! That just solved my problem!

  7. #7
    Join Date
    May 2012
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    Re: pulseaudio - no sound

    Quote Originally Posted by karatedog View Post
    Thanks! That just solved my problem!
    Funny how this worked for me:
    moved files to trash. restarted pulse.
    "refused connection"
    restored files from trash. NOW the appropriate audio devices showed up, at least. Still no sound, but I'm almost there.

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