Dear ALSA devel,
I'm forwarding a message I sent to ALSA-user a week ago; it didn't get any replies. It asks some questions about combining stereo devices into a single multi-channel device. Is this the right place to ask? I'll summarize because maybe the original message was a bit verbose:
1. Is this document still correct (i.e. saying that ALSA doesn't do any resampling when you combine two devices in a "multi" configuration): https://github.com/opensrc/alsa/blob/master/lib/md/TwoCardsAsOne.md
2. What is the best way to combine output devices? Pulseaudio?
3. Is there a way to configure an ALSA pcm which routes to a non-default Pulseaudio sink?
Thanks,
Frederick
----- Forwarded message from frederik@ofb.net -----
Date: Mon, 19 Mar 2018 19:10:47 -0700 From: frederik@ofb.net To: alsa-user@lists.sourceforge.net Subject: [Alsa-user] the mythical "el-cheapo" X-Spam-Status: No, score=-4.9 required=5.0 tests=BAYES_00,DKIM_SIGNED, HEADER_FROM_DIFFERENT_DOMAINS,RCVD_IN_DNSWL_NONE,RP_MATCHES_RCVD, SPF_HELO_PASS,SPF_PASS,T_DKIM_INVALID autolearn=unavailable autolearn_force=no version=3.4.1 X-Spam-Level: User-Agent: Mutt/1.9.4 (2018-02-28) X-My-Message-ID: 20180320021047.GA4304@ofb.net#2 X-My-Tags: inbox alsa-user
Dear ALSA user,
I am trying to develop some software to use for a custom sound-masking system where I live (to e.g. mask the sound of construction beeping going on outside). It is crucial to have more than two speakers, because if you can tell where the noise is coming from then it becomes distracting. But I would want this to be an inexpensive solution for students and such and so I don't want to depend on a surround sound system being present. So currently I'm using three $7 USB stereo sound adapters, which control six speakers.
For various reasons it would be good for me to be able to combine the USB cards into one audio device. One reason is that users might want to play music on the same speakers that they're using for sound-masking. Another is that having one device with one sample rate would simplify the coding of the noise generation, which might be based on some adaptive algorithm that e.g. listens to a microphone signal, and which might be written in a high-level language where threads are inefficient.
However, there are a few documents discouraging me from trying to combine devices in ALSA, saying "you will drift out of sync over time" because of differences in the crystals of the various sound cards.
https://www.alsa-project.org/main/index.php/Asoundrc#Virtual_multi_channel_d... https://github.com/opensrc/alsa/blob/master/lib/md/TwoCardsAsOne.md
But they describe a "route/multi" configuration using ALSA which often seems to work OK for me. For example, I can generate six independent pink noise streams with SoX and it doesn't usually sound glitchy:
AUDIODEV=noise play -c 6 -r 44100 -n synth pinknoise gain -10 channels 6
Music also seems to play OK. I've experienced glitches with movie audio, which resolve when I pause/unpause, but sometimes these issues seemed to persist even when I routed the movie audio to a single pair of speakers. I never tracked down the problem but I thought it might be a sample rate issue.
Pulseaudio advertises the ability to combine devices correctly using module-combine-sink which tracks the actual frequencies of the various sink crystals:
https://www.freedesktop.org/wiki/Software/PulseAudio/Documentation/User/Modu...
However, the pulseaudio latency is very high and it makes movie audio out of sync with the picture. Furthermore, it seems difficult to use with ALSA because the "pulse" ALSA module doesn't seem to take a parameter which would let me specify which pulse sink or source I am interested in. Thus it seems impossible to use ALSA programs like SoX's play command (above) to output to a non-default Pulseaudio sink. Is this correct?
I want to be able to write software which other people can easily install and which can coexist alongside other software they have installed, including audio software. For example I don't want my system to force them to adopt a certain Pulseaudio sink as "default". Even depending on Pulseaudio seems somewhat undesirable. But since I lack familiarity with audio software, I am hoping someone on this list can point me to useful projects or sources of information so that I can solve these problems in a standard way, and produce software that might actually end up being useful to others.
Thank you,
Frederick Eaton
pcm.mix1 { type dmix ipc_key 2852 # must be unique slave { pcm "hw:3" </home/frederik/hw-settings> } }
pcm.mix2 { type dmix ipc_key 2853 slave { pcm "hw:1" </home/frederik/hw-settings> } }
pcm.mix3 { type dmix ipc_key 2842 # must be unique slave { pcm "hw:0" </home/frederik/hw-settings> } }
pcm.multi_mix { type multi; slaves.a.pcm "mix1"; slaves.b.pcm "mix2"; slaves.c.pcm "mix3"; slaves.a.channels 2; slaves.b.channels 2; slaves.c.channels 2; bindings.0.slave a; bindings.0.channel 0; bindings.1.slave a; bindings.1.channel 1;
bindings.2.slave b; bindings.2.channel 0; bindings.3.slave b; bindings.3.channel 1;
bindings.4.slave c; bindings.4.channel 0; bindings.5.slave c; bindings.5.channel 1; } # pcm.multi_mix { # type multi; # slaves.a.pcm "mix2"; # slaves.b.pcm "mix1"; # slaves.a.channels 2; # slaves.b.channels 2; # bindings.0.slave a; # bindings.0.channel 0; # bindings.1.slave a; # bindings.1.channel 1;
# bindings.2.slave b; # bindings.2.channel 0; # bindings.3.slave b; # bindings.3.channel 1; # }
pcm.upmix { type route; slave.pcm "multi_mix";
ttable.0.0 1; ttable.1.1 1;
ttable.0.2 1; ttable.1.3 1;
ttable.0.4 1; ttable.1.5 1; } # pcm.upmix { # type route; # slave.pcm "multi_mix";
# ttable.0.0 1; # ttable.1.1 1;
# ttable.0.2 1; # ttable.1.3 1; # }
pcm.noise { type softvol ## FHE 01 Mar 2018 # why is plug: needed here? if i put plug:multi_mix in the upmix # device it breaks slave.pcm "plug:multi_mix" control { name "Noise" card 1 } }
ctl.noise { type hw card 1 }
pcm.music { type softvol # slave.pcm "plug:mix1" slave.pcm "plug:upmix"
## the following gives "invalid argument", why? # slave { # pcm "upmix"; # channels 2; # }
control { name "Music" card 1 } }
ctl.music { type hw card 1 }
pcm.music_no_bass { type plug; slave.pcm { type ladspa # slave.pcm "plughw:0,0"; slave.pcm "plug:music"; path "/usr/lib/ladspa/"; plugins [{ label hpf input { controls [ 1000 ] } }] hint { show on description "Highpass filter for music" } } }
# pcm.!default { # type asym # #playback.pcm "music" # playback.pcm "plug:music_no_bass" # #playback.pcm "plug:hw0mix" # capture.pcm "hw:0" # }
# FHE 10 Mar 2018 ## we need an "asym" with capture and playback to make pulseaudio happy pcm.!default { type asym playback.pcm { ## from https://wiki.archlinux.org/index.php/Advanced_Linux_Sound_Architecture/Examp... # set the default device according to the environment # variable ALSA_DEFAULT_PCM and default to plug:music_no_bass @func refer name { @func concat strings [ "pcm." { @func getenv vars [ ALSA_DEFAULT_PCM ] default "music_no_bass" } ] } } capture.pcm "hw:0" }
# FHE 10 Mar 2018 old version ## FHE 01 Mar 2018 # pcm.!default { # @func refer # name { @func concat # strings [ "pcm." # { @func getenv # vars [ ALSA_DEFAULT_PCM ] # default "music_no_bass" # } # ] # } # }
rate 44100 format S16_LE
## FHE 01 Mar 2018 # Eshelman's settings period_time 0 period_size 2048 buffer_size 65536 buffer_time 0 periods 128
## overrides # https://www.alsa-project.org/main/index.php/FramesPeriods # buffer size must be >= 2*period_size*(channels*bytes) # = 8*period_size # interrupt frequency is period_size * rate # we seem to get errors with period_size <= 256 # these settings seem to increase CPU usage only slightly, e.g. from # 5.3% (for above settings) to 7.3% (for below) with mpv # period_size 512 # buffer_size 4096 # period_size 1024 # buffer_size 8192 # This will cause the device to interrupt 86 times per second
#!/bin/zsh
#front-left,front-right,rear-left,rear-right,front-center,lfe
pacmd load-module module-alsa-sink device=mix1 sink_name=mix1 channels=2 channel_map=front-left,front-right pacmd load-module module-alsa-sink device=mix2 sink_name=mix2 channels=2 channel_map=rear-left,rear-right pacmd load-module module-alsa-sink device=mix3 sink_name=mix3 channels=2 channel_map=front-center,lfe
pacmd load-module module-combine-sink sink_name=mix123 slaves=mix1,mix2,mix3 channels=6 channel_map=front-left,front-right,rear-left,rear-right,front-center,lfe
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