[alsa-devel] Can module snd-aloop (virtual looback device) be made clock slave of a real audio device ?
Hello alsa-devel's,
A while back, I had written a rather hackish method to hook up the ALSA virtual loopback device to jack (jackaudio.org) via Jack clients alsa_in / alsa_out so that non jack-aware applications that can however use ALSA devices "directly" (via alsa-lib) can have their audio stream redirected to jack (see my rather lengthy prose at http://alsa.opensrc.org/Jack_and_Loopback_device_as_Alsa-to-Jack_bridge).
There are a few advantages in this method (depends on your use-cases of course). But it could be improved (or so I think) by removing the following drawbacks:
The alsa_in/out jack clients are resampling the audio stream to synchronize with the device that jack is using. The resampling, done by libsamplerate, takes some CPU power. Since many users are using my recipe on laptops, this is not always welcome.
Because snd-aloop is a virtual device, couldn't it be made a clock slave of the h/w device that jack is operating on ? This would in principle remove the need for resampling. I could imagine that the module could be passed an option at loading time (e.g. modprobe snd-aloop clock-master=<real h/w device index> or something like that)
Since I have no clue as to the kernel code required to implement / extend snd-aloop, I would like some opinions about this.
Thanks for any hints.
J.
Date 20.8.2012 11:51, James Warden wrote:
Hello alsa-devel's,
A while back, I had written a rather hackish method to hook up the ALSA virtual loopback device to jack (jackaudio.org) via Jack clients alsa_in / alsa_out so that non jack-aware applications that can however use ALSA devices "directly" (via alsa-lib) can have their audio stream redirected to jack (see my rather lengthy prose at http://alsa.opensrc.org/Jack_and_Loopback_device_as_Alsa-to-Jack_bridge).
There are a few advantages in this method (depends on your use-cases of course). But it could be improved (or so I think) by removing the following drawbacks:
The alsa_in/out jack clients are resampling the audio stream to synchronize with the device that jack is using. The resampling, done by libsamplerate, takes some CPU power. Since many users are using my recipe on laptops, this is not always welcome.
Because snd-aloop is a virtual device, couldn't it be made a clock slave of the h/w device that jack is operating on ? This would in principle remove the need for resampling. I could imagine that the module could be passed an option at loading time (e.g. modprobe snd-aloop clock-master=<real h/w device index> or something like that)
Use 'PCM Rate Shift 100000' mixer control to fine tune the rate (timing) for the snd-aloop PCM devices. An example using this solution is the alsaloop application from the alsa-utils package.
Also note that other controls can expose the audio format on the playback side, so you can avoid resampling in the alsa-lib completely for ALSA audio apps. The alsaloop application can give you more hints.
Jaroslav
----- Original Message ----- From: Jaroslav Kysela perex@perex.cz To: alsa-devel@alsa-project.org Cc: warjamy@yahoo.com Sent: Monday, August 20, 2012 12:26 PM Subject: Re: [alsa-devel] Can module snd-aloop (virtual looback device) be made clock slave of a real audio device ?
Date 20.8.2012 11:51, James Warden wrote:
Hello alsa-devel's,
A while back, I had written a rather hackish method to hook up the ALSA virtual loopback device to jack (jackaudio.org) via Jack clients alsa_in / alsa_out so that non jack-aware applications that can however use ALSA devices "directly" (via alsa-lib) can have their audio stream redirected to jack (see my rather lengthy prose at http://alsa.opensrc.org/Jack_and_Loopback_device_as_Alsa-to-Jack_bridge).
There are a few advantages in this method (depends on your use-cases of course). But it could be improved (or so I think) by removing the following drawbacks:
The alsa_in/out jack clients are resampling the audio stream to synchronize with the device that jack is using. The resampling, done by libsamplerate, takes some CPU power. Since many users are using my recipe on laptops, this is not always welcome.
Because snd-aloop is a virtual device, couldn't it be made a clock slave of the h/w device that jack is operating on ? This would in principle remove the need for resampling. I could imagine that the module could be passed an option at loading time (e.g. modprobe snd-aloop clock-master=<real h/w device index> or something like that)
Use 'PCM Rate Shift 100000' mixer control to fine tune the rate (timing) for the snd-aloop PCM devices. An example using this solution is the alsaloop application from the alsa-utils package.
Also note that other controls can expose the audio format on the playback side, so you can avoid resampling in the alsa-lib completely for ALSA audio apps. The alsaloop application can give you more hints.
Thanks for the hints!
I have a few questions:
1- I can see the "PCM Rate Shift 100000" control when I list it with amixer.
is there a way to have these (hidden) controls show up in alsamixer as well ?
2- The range of this control is 80%-120%. Do I understand that it is a constant ? Once set, it just stays there ?
3- How would I know which value I would have to set anyway ?
4- something a bit orthogonal: my main card is a Hammerfall HDSP + Multiface II. The access mode is MAP_NONINTERLEAVED according to the hw_params. This prevents me from e.g. combining this card and the virtual loopback into one (not that I want to do that in the end but I was experimenting with the merge plugin at some point in the past) because the access mode is MAP_INTERLEAVED for snd-aloop. This also prevents the alsaloop utility from being used like in this example:
alsaloop -C hw:Loopback,1,0 -P hw:DSP
This fires up OK, but then when I try to play something on hw:Loopback,0,0, alsaloop gives up with an error message regarding the stream format being wrong or the like (I am not sitting at home now, so I can't say preceisely).
Thanks for clarifying these bits. Maybe this conversation should take place in alsa-users ?
J.
Date 21.8.2012 10:28, James Warden wrote:
----- Original Message ----- From: Jaroslav Kysela perex@perex.cz To: alsa-devel@alsa-project.org Cc: warjamy@yahoo.com Sent: Monday, August 20, 2012 12:26 PM Subject: Re: [alsa-devel] Can module snd-aloop (virtual looback device) be made clock slave of a real audio device ?
Date 20.8.2012 11:51, James Warden wrote:
Hello alsa-devel's,
A while back, I had written a rather hackish method to hook up the ALSA virtual loopback device to jack (jackaudio.org) via Jack clients alsa_in / alsa_out so that non jack-aware applications that can however use ALSA devices "directly" (via alsa-lib) can have their audio stream redirected to jack (see my rather lengthy prose at http://alsa.opensrc.org/Jack_and_Loopback_device_as_Alsa-to-Jack_bridge).
There are a few advantages in this method (depends on your use-cases of course). But it could be improved (or so I think) by removing the following drawbacks:
The alsa_in/out jack clients are resampling the audio stream to synchronize with the device that jack is using. The resampling, done by libsamplerate, takes some CPU power. Since many users are using my recipe on laptops, this is not always welcome.
Because snd-aloop is a virtual device, couldn't it be made a clock slave of the h/w device that jack is operating on ? This would in principle remove the need for resampling. I could imagine that the module could be passed an option at loading time (e.g. modprobe snd-aloop clock-master=<real h/w device index> or something like that)
Use 'PCM Rate Shift 100000' mixer control to fine tune the rate (timing) for the snd-aloop PCM devices. An example using this solution is the alsaloop application from the alsa-utils package.
Also note that other controls can expose the audio format on the playback side, so you can avoid resampling in the alsa-lib completely for ALSA audio apps. The alsaloop application can give you more hints.
Thanks for the hints!
I have a few questions:
1- I can see the "PCM Rate Shift 100000" control when I list it with amixer.
is there a way to have these (hidden) controls show up in alsamixer as well ?
No, it isn't, it's marked as a part of the PCM API, not mixer API. This control was introduced to set the drift programatically.
2- The range of this control is 80%-120%. Do I understand that it is a constant ? Once set, it just stays there ?
This value can be changed (corrected) at any time.
3- How would I know which value I would have to set anyway ?
alsaloop measures the frequency from the real hardware and computes the actual drift which is passed to snd-aloop. The drift is handled all times (it's not just one shot measure).
4- something a bit orthogonal: my main card is a Hammerfall HDSP + Multiface II. The access mode is MAP_NONINTERLEAVED according to the hw_params. This prevents me from e.g. combining this card and the virtual loopback into one (not that I want to do that in the end but I was experimenting with the merge plugin at some point in the past) because the access mode is MAP_INTERLEAVED for snd-aloop. This also prevents the alsaloop utility from being used like in this example:
alsaloop -C hw:Loopback,1,0 -P hw:DSP
This fires up OK, but then when I try to play something on hw:Loopback,0,0, alsaloop gives up with an error message regarding the stream format being wrong or the like (I am not sitting at home now, so I can't say preceisely).
Use the plughw:DSP device which does the conversions.
Jaroslav
participants (2)
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James Warden -
Jaroslav Kysela