Forcing alsa dmix to 24 bit 96 kHz with softvol [SOLVED]

[selig]

I have bought a Creative Sound Blaster X-Fi Surround 5.1 Pro USB. There are several problems with this card:

1) it cannot mix several streams simultaneously
2) it cannot set volume (via snd-usb-audio)

It works out of the box and I was able to add a volume control using the Alsa softvol plugin:
.asoundrc:

Code: Select all

pcm.!default {
    type            plug
    slave.pcm       "softvol"   #make use of softvol
}

pcm.softvol {
    type            softvol
    slave {
        pcm        "dmix"      #redirect the output to dmix (instead of "hw:0,0")
    }
    control {
        name        "Master"       #override the PCM slider to set the softvol volume level globally
        card        0
    }
}

The problem is that with dmix, everything gets downsampled to 48 kHz and 16 bit. I was able to force conversion to 24bit this using this .asoundrc (:

Code: Select all

pcm_slave.slave_format_s24le {
  pcm "plug:front"
  format "S24_3LE"
}

pcm.front24 {
  @args [ CARD DEV ]
  @args.CARD {
    type string
  }
  type plug
  slave slave_format_s24le
  hint {
    show {
      @func refer
      name defaults.namehint.basic
    }
    description "Front speakers, converted to S24_3LE"
    device 0
  }
}

pcm.!default front24

But then I am unable to use softvol volume control. Similarly, I can disable dmix and get a native sample rate (for example 96 kHz):

Code: Select all

pcm.!default {
type plug
slave.pcm hw
}

But I have no idea how to combine these three configurations so that I get softvol volume control, 24 bit and 96 kHz output (the soundcard can do either 16 or 24 bit and 48 or 96 kHz). Could anyone help me, please?

[PaulBredbury]

Take a look at my ~/.asoundrc

Change the format and rate within pcm.dmixed

Add softvol into the middle - hopefully you can figure that out, from the examples and notes.

[selig]

Thanks a lot! That asoundrc really saved me. The relevant sections from my asoundrc are here, if anyone needs them:

Code: Select all

pcm.softvol {
    type            softvol
    slave.pcm "dmixed"      #redirect the output to dmix (instead of "hw:0,0")

    control {
        name        "Master"       #override the PCM slider to set the softvol volume level globally
        card        0
    }
}



pcm.dmixed {
        type asym
        playback.pcm {
                # See plugin:dmix at http://www.alsa-project.org/alsa-doc/alsa-lib/pcm_plugins.html
                type dmix
                ipc_key 5678293
                ipc_perm 0660
                ipc_gid audio
                slave {
                        channels 2
                        pcm {
                                format S24_3LE
                                rate 96000

                                type hw
                                card 0
                                device 0
                                subdevice 0
                        }

                        period_size 1024
                         buffer_size 8192
                }

                bindings {
                        0 0
                        1 1
                }
        }
        capture.pcm "hw:0"
}


pcm.!default {
        type plug
        slave.pcm "softvol"
}

Now I am just not sure what kind of sample rate converter dmix uses. Can this be configured or do I need to put a resampler before dmix and specify converter "samplerate_medium" in it?

[PaulBredbury]

Look at my pcm.headphones entry, as an example:

Code: Select all

pcm.headphones {
    type rate
    slave {
        pcm "plug:bs2b"
        #rate 44100
        rate 48000
    }
    # Choices: samplerate_best samplerate_medium samplerate samplerate_order samplerate_linear
    converter "samplerate_medium"

    hint {
        show on
        description "Headphones"
    }
}

Or the easier way, at the top of my ~/.asoundrc:

Code: Select all

defaults.pcm.rate_converter "samplerate_medium"

But don't worry about the samplerate - I've never noticed the slightest difference, on anything expect a single demonstration file (from Ubuntu bug) consisting of crazy audio tones, which was created solely to illustrate the difference in the algorithms.

Edit: Added link to wav file.

[aCOSwt]

But don't worry about the samplerate - I've never noticed the slightest difference, on anything expect a single demonstration file consisting of crazy audio tones, which was created solely to illustrate the difference in the algorithms.

I'd word this differently : If you can notice the difference, then, you should not use an Alsa plugin for resampling.