Pulseaudio + ALSA Configuration

Okay… if you have read my previous post on ALSA Configuration you will know that I am trying to solve my sound setup problem in linux. The basic problem can be summarized as:

• Use the sound cards at their native sample-rate to improve over their rather poor hardware re-samplers.
• Mix audio from multiple applications on the cards.
• Be able to easily designate audio cards on a per application basis at application launch time.

I had commented earlier that I decided against pulseaudio because of its lack of 32 bit signed integer audio support. I realized after some general use of my previous alsa configuration that even though some applications use 32 bit audio well, most other applications have problems. Typically mplayer and ut2004 (through aoss) couldn’t correctly figure out that the device was running at 32 bit precision and sent it 16 bit audio data. This caused both the applications to run audio at half the speed with the audio crackling all over. The fix was simple: change S32_LE in .asoundrc to S16_LE. However, that would be the end of any 32 bit audio on my system. I also realized that the only application which I used (heavily) with 32 bit audio was madplay (piped through aplay; madplay directly outputs to hardware alone). I grudgingly agreed to sacrifice that .

Pulseaudio configuration is pretty straight forward. I configured the /etc/pulse/default.pa for the two cards in the following way. I happily noticed that pulseaudio used a good resampler by default (speex-float-3 which has similar CPU usage as alsa’s plugin speexrate).

/etc/pulse/default.pa

 
#!/usr/bin/pulseaudio -nF
.nofail

### Load something into the sample cache
#load-sample-lazy x11-bell /usr/share/sounds/gtk-events/activate.wav
#load-sample-lazy pulse-hotplug /usr/share/sounds/startup3.wav
#load-sample-lazy pulse-coldplug /usr/share/sounds/startup3.wav
#load-sample-lazy pulse-access /usr/share/sounds/generic.wav

.fail

### Load audio drivers statically
load-module module-alsa-sink device=intelHDA sink_name=Spkr rate=192000
load-module module-alsa-sink device=iMic sink_name=HeadPh rate=48000

### Load several protocols
#load-module module-esound-protocol-unix
load-module module-native-protocol-unix

### Automatically restore the volume of playback streams
#load-module module-volume-restore

### Automatically restore the default sink/source when changed by the user during runtime
#load-module module-default-device-restore

### Automatically move streams to the default sink if the sink they are
### connected to dies, similar for sources
load-module module-rescue-streams

### Automatically suspend sinks/sources that become idle for too long
load-module module-suspend-on-idle

### Make some devices default
set-default-sink Spkr
#set-default-source input

To make alsa applications work directly with pulseaudio we just need to set the default pcm and ctl to the pulse alsa plugin (configuration shown later). The pulse plugin creates a new audio device called pulseaudio and maps its audio I/O to the default pulse sink which can be changed at runtime. The pulseaudio device also has two mixer controls; a “Master” for playback and a “Capture” for recording.

The problem with the above is that pulseaudio starts looking for mixer controls “Master” for every configured sink hardware device (with a fallback to “PCM”). The default alsa configuration doesn’t provide a “Master” control for either the intelHDA card or the iMic card. The “PCM” on iMic controls the hardware mixer directly and that for intelHDA controls a software volume for alsa’s default dmix configuration. Thus, the pulseaudio fallback to “PCM” works correctly for iMic (changing the pulseaudio “Master” control changes the volume) but not for intelHDA (changing pulseaudio “Master” control has no change on volume, but changing “Front” directly on the hardware using alsamixer -c does work). To work around this I configured two virtual softvol (software volume) devices for my two cards naming the softvol control “Master”. The configuration is shown below.

.asoundrc

pcm.!default {
        type pulse
}

ctl.!default {
        type pulse
}

# Create softvol devices for intelHDA and iMic.
# For intelHDA, its "PCM" control controls its default (not used) softvol.
# For iMic, its "PCM" control controls its hardware directly.
pcm.intelHDA {
        type softvol
        slave.pcm "hw:0,0"
        control.name "Master"
        control.card 0
}

pcm.iMic {
        type softvol
        slave.pcm "hw:1,0"
        control.name "Master"
        control.card 1
}

# It seems we require these dummies as pulseaudio tries to look up mixer devices.
ctl.intelHDA {
        type hw
        card 0
}

ctl.iMic {
        type hw
        card 1
}

# For UT2004 (which uses dsp0 through aoss), always go directly to headphone hardware.
pcm.dsp0 {
        type plug
        slave.pcm "hw:1,0"
}

ctl.dsp0 {
        type hw
        card 1
}

Note that the pcm names used (intelHDA and iMic) are used in /etc/pulse/default.pa as sinks. More information on the softvol alsa plugin can be found here. There is some nice documentation about the various alsa plugins here. There some good documentation about alsa is also present here, here and here.

To change the current default pulseaudio sink I rewrote the audio.sh from the previous post to the following:

audio.sh

#!/bin/bash

pulse_set_default_sink () {
        pacmd << EOF
set-default-sink $1
EOF
}

h=`cat $HOME/.dwm/status_audio`;

if [ "x$1" == "x" ]; then
        if [ "x$h" == "x[ Spkr ] " ]; then
                echo -n "[ HeadPh ] " > $HOME/.dwm/status_audio ;
                pulse_set_default_sink HeadPh ;
        else
                echo -n "[ Spkr ] " > $HOME/.dwm/status_audio ;
                pulse_set_default_sink Spkr ;
        fi
elif [ $1 == "up" ]; then
        amixer set Master 2%+ ;
elif [ $1 == "down" ]; then
        amixer set Master 5%- ;
elif [ $1 == "toggle" ]; then
        amixer set Master toggle ;
else
        echo "$0: Unknown command $1";
fi

v=`amixer get Master | grep "Front Left:" | egrep -o "[0-9]+%"`
m=`amixer get Master | grep "Front Left:" | egrep -o "off"`

if [ "x$m" == "x" ]; then
        echo -n "vol:$v "
else
        echo -n "vol:(M) "
fi > $HOME/.dwm/status_volume

Note that now environment based device settings are gone as the current default pulseaudio sink automatically maps to the global alsa default pcm device. Once an application connects to pulseaudio, it retains its sink mapping even after the default sink is changed. (Update: It seems that module-volume-restore remembers the volume and sink of a client and restores both when it sees the client again. This behavior is definitely useful but we need to disable module-volume-restore if we want our current default sink logic to work). Pulseaudio also support hot swapping an application’s audio from one sink to another. I am still thinking of a way to toggle the sink device of the currently focussed application using dwm shortcuts… that would be some nifty configuration .

About these ads