[alsa-devel] Can module snd-aloop (virtual looback device) be made clock slave of a real audio device ?

[James Warden]

----- Original Message -----
From: Jaroslav Kysela <perex at perex.cz>
To: alsa-devel at alsa-project.org
Cc: warjamy at 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.