[alsa-devel] ALSA application programming: route audio from one PCM to another
Hi,
I finally managed to write an ALSA I/O plugin that does what I want. The plugin supports both playback and capture.
Now I would like to write a simple audio application that takes audio samples
* from the microphone and plays it back on my plugin and * from the plugin (capture) and plays it back on the speakers
Hence as long as the application is running, it should do the above.
* Is there a special ALSA way to route audio from one PCM to another ?
* If not, I suppose it would just work if I open the plugin PCM and the hw PCM at the same time and copy audio frames between them ?
cheers, stefan
'Twas brillig, and Stefan Schoenleitner at 13/04/10 16:23 did gyre and gimble:
Hi,
I finally managed to write an ALSA I/O plugin that does what I want. The plugin supports both playback and capture.
Now I would like to write a simple audio application that takes audio samples
- from the microphone and plays it back on my plugin
and
- from the plugin (capture) and plays it back on the speakers
This sounds like something that would be more appropriate for jack http://jackaudio.org/
Hence as long as the application is running, it should do the above.
Is there a special ALSA way to route audio from one PCM to another ?
If not, I suppose it would just work if I open the plugin PCM and the
hw PCM at the same time and copy audio frames between them ?
Dealing with this can be quite complex, especially if the pcms are clocked of different sources, you have to deal with a degree of resampling to ensure that clock skew doesn't get out of control.
The module-loopback plugin in PulseAudio does a similar thing (routes audio from a source to a sink) and as such has to deal with these clock skew problems.
Col
Colin Guthrie wrote:
'Twas brillig, and Stefan Schoenleitner at 13/04/10 16:23 did gyre and gimble:
Hi,
I finally managed to write an ALSA I/O plugin that does what I want. The plugin supports both playback and capture.
Now I would like to write a simple audio application that takes audio samples
- from the microphone and plays it back on my plugin
and
- from the plugin (capture) and plays it back on the speakers
This sounds like something that would be more appropriate for jack http://jackaudio.org/
Thanks for your response, that really sounds like a job for JACK.
However, due to the nature of jack it seems that running the jack-daemon is always necessary. As my code is supposed to work on a very small scale embedded target, I would prefer to have a small stand-alone application that does not require a running jack-daemon.
* Do you know if it is possible to use the jack functionality without having to run the jack-daemon ?
Hence as long as the application is running, it should do the above.
Is there a special ALSA way to route audio from one PCM to another ?
If not, I suppose it would just work if I open the plugin PCM and the
hw PCM at the same time and copy audio frames between them ?
Dealing with this can be quite complex, especially if the pcms are clocked of different sources, you have to deal with a degree of resampling to ensure that clock skew doesn't get out of control.
Both PCMs are on the same machine, hence they should be clocked from the same source as well ?
cheers, stefan
'Twas brillig, and Stefan Schoenleitner at 14/04/10 09:44 did gyre and gimble:
Colin Guthrie wrote:
'Twas brillig, and Stefan Schoenleitner at 13/04/10 16:23 did gyre and gimble:
Hi,
I finally managed to write an ALSA I/O plugin that does what I want. The plugin supports both playback and capture.
Now I would like to write a simple audio application that takes audio samples
- from the microphone and plays it back on my plugin
and
- from the plugin (capture) and plays it back on the speakers
This sounds like something that would be more appropriate for jack http://jackaudio.org/
Thanks for your response, that really sounds like a job for JACK.
However, due to the nature of jack it seems that running the jack-daemon is always necessary. As my code is supposed to work on a very small scale embedded target, I would prefer to have a small stand-alone application that does not require a running jack-daemon.
- Do you know if it is possible to use the jack functionality without
having to run the jack-daemon ?
I'm afraid not. Like PulseAudio, Jack needs the daemon to do all the hard stuff.
I guess you'll just have to write some small app that does the record/playback. I doubt it'll be that hard.
That said, with this approach, the fact that you've got an alsa plugin doing the processing in the middle seems slightly irrelevant... why not have your app sample the input, do the DSP it needs to do, then play the output? No need to: sample, play [process] sample, play. Rather: sample [process] play.
Perhaps I'm not understanding the other needs/use cases of the plugin tho'. Just idle thoughts :)
Hence as long as the application is running, it should do the above.
Is there a special ALSA way to route audio from one PCM to another ?
If not, I suppose it would just work if I open the plugin PCM and the
hw PCM at the same time and copy audio frames between them ?
Dealing with this can be quite complex, especially if the pcms are clocked of different sources, you have to deal with a degree of resampling to ensure that clock skew doesn't get out of control.
Both PCMs are on the same machine, hence they should be clocked from the same source as well ?
I believe this is quite likely, but I'm not really certain that it holds true in all cases.
Someone more clued up than me may be able to comment.
Col
Colin Guthrie wrote:
- Do you know if it is possible to use the jack functionality without
having to run the jack-daemon ?
I'm afraid not. Like PulseAudio, Jack needs the daemon to do all the hard stuff.
I guess you'll just have to write some small app that does the record/playback. I doubt it'll be that hard.
I guess I will try that first. If it doesn't work I can still try jack.
That said, with this approach, the fact that you've got an alsa plugin doing the processing in the middle seems slightly irrelevant... why not have your app sample the input, do the DSP it needs to do, then play the output? No need to: sample, play [process] sample, play. Rather: sample [process] play.
Perhaps I'm not understanding the other needs/use cases of the plugin tho'. Just idle thoughts :)
In the beginning I had a similar idea as well ;) However, in more detail the DSP actually does speech compression. The compressed speech should then be sent to a remote host where it is decompressed again.
Hence one host samples speech coming from the microphone and plays it back via the io plugin. The plugin takes the samples and forwards them to the DSP so that they can be compressed. After some processing delay the DSP will output the compressed speech packets. These compressed packets are then read by the plugin and written to a socket.
The remote host eventually receives the compressed speech packets and the same process can be performed in reverse so that the decompressed speech will come out of the speakers.
As this should work in both directions at the same time, each side needs to capture and playback at the same time as well.
Of course, like you said, it would be possible to do that without the io plugin in between (so that the audio application would directly communicate with the DSP).
The drawback of this solution is that only the written audio application could actually use the DSP.
However, with the ALSA io plugin a lot of flexibility is gained as just about any audio application can use the speech compression DSP without having to change the application itself. Another thing is that the DSP only allows a single sampling rate and audio format. As it is possible to chain different alsa plugins together, it would also be easy to resample and convert just about any audio format to the format the DSP requires ;)
Hence in the end you could fire up your favorite media player and play an mp3 or a movie over the plugin. After that speech compression would be performed and in the end you could hear that sound on the remote host. (Of course that will only work well for speech as the DSP does speech compression and not generic audio compression.)
At least that's the plan for now ;) Until then there is some more work to be done.
So what do think ?
cheers, stefan
participants (2)
-
Colin Guthrie -
Stefan Schoenleitner