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From the Linux Audio Devel mailing list

Message posted to LinuxAudioDevel by Joern Nettingsmeier:

Richard Guenther asked: I'd like to create a virtual 2-(stereo)-channel alsa device from one ISA SB AWE and one on-board VIA alsa device. Has anyone figured out how to do this using .asoundrc magic? (I know Jaroslav knows and told Joern, but I think this is of greater interest) Joern/Jaroslav, can you post a quick howto on this topic? Preferrably including some .asoundrc quoting...

Just to clarify, this will not create the mythical "multi-channel soundcard out of el-cheapo consumer cards". You will drift out of sync over time unless a hardware syncing mechanism like wordclock is in place.

Still, it is sometimes helpful to make applications see one 4channel card to allow for flexible routing if they can't easily be made to talk to multiple cards ( jack being one example). Here's how you do it (thanks to jaroslav and takashi for explaining it to me). Here's an .asoundrc example...

Create a virtual four-channel device with two soundcards: IIUC, this is in fact two interleaved stereo streams in different memory locations, so jack will complain that it cannot get mmap-based access. See below.

pcm.multi {
    type multi;
    slaves.a.pcm "hw:0,0";
    slaves.a.channels 2;
    slaves.b.pcm "hw:1,0";
    slaves.b.channels 2;
    bindings.0.slave a; 0;
    bindings.1.slave a; 1;
    bindings.2.slave b; 0;
    bindings.3.slave b; 1;

I do not really understand what it means to have a ctl interface to a multi device, but jack will be unhappy if there is no mixer to talk to, so we set this to card 0.

ctl.multi {
    type hw;
    card 0;

This creates a 4channel interleaved pcm stream based on the multi device, jack will work with this one...

pcm.ttable {
    type route;
    slave.pcm "multi";
    ttable.0.0 1;
    ttable.1.1 1;
    ttable.2.2 1;
    ttable.3.3 1;
ctl.ttable {
    type hw;
    card 0;

A syntax description is in ALSA lib config. The available plugins (i.e. the "type" entries) are described in the doxygen docs at pcm_plugins. It seems there is now also a LADSPA plugin plugin :) I have not tried it yet, but it looks interesting. To test the above setting, I feed a signal to both soundcards, play it back and listen to it via with my external mixer:

arecord -f cd -D multi -c 4 | aplay -D multi

This will give you loads of xruns, but it's probably ok for testing. To start jack with the new device, use:

jackd [-a] -R [-v] -d alsa -d ttable [-p 1024]

I hope I got it right... the settings Work For Me (tm), although I can't get ardour to record from the device, but then I can't get it to record at all at the moment. jack starts ok, though.


WARNING NOTE added by LudwigSchwardt: (comments welcome!)

Be careful when using this scheme for capturing (recording) audio data. It might work with multiple sample-synchronized cards (i.e. Hammerfall/ICE1712) that have a single sample clock signal driving all of them, or when the dummy soundcard is combined with another soundcard for testing. It will definitely fail when combining two or more unlinked soundcards. When using JACK, I typically got an error message of

snd_pcm_mmap_commit: Assertion `frames <= snd_pcm_mmap_avail(pcm)' failed

and jackd hangs in the process. This error may take a while to pop up, but it is inevitable, since this is a basic hardware limitation.

Each soundcard has its own sample clock. If these aren't linked to the same clock in some way (eg. wordclock), the clock frequencies will differ, sometimes substantially, due to manufacturing tolerances on the crystals, etc. This means that the hardware interrupts announcing the availability of new data will be out of sync. Even worse, the cards with faster clocks will produce more data than the slower cards during the same time interval. Therefore, even if a program like JACK waited until all the soundcards had audio data available and then presented this data to its callback, at some stage data from the soundcard with a faster sample clock will have to be discarded to keep up with the rest. At the very least xruns will occur.

It might be possible to write a software layer (ALSA driver/plugin?) to hide the clock differences by effectively resampling the streams to a common frequency. This will be the only way to keep programs like JACK completely happy. As this will inevitably involve the discarding and/or generation of samples, it is still not a pretty solution. It will probably also increase the latency of the system.

On the other hand, this .asoundrc trick seems to work OK for playback. During playback, programs dump audio data into soundcard buffers and let the hardware decide when the data will reach the output terminals. These buffers in effect hide the difference in clock frequencies. However, xruns are probably still going to happen, for instance when the faster card runs out of data or the slower card's buffer is full.

regards, Ludwig

Hardware mod

With a small hw mod multiple cards can be made running in sync and hence use the above alsa device for recording.

Have a look at

br, Timo

Multiple ice1712

At the other end of the scale (el-expensivo?), here's a guide to using multiple ice1712 (Delta 1010) cards for pro recording. Nothing fancy, just lots of channels in and out.


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