Re: [Jack-Devel] Connecting alsa-loopback to jack

This is my asound.conf inspired from some thread in LinuxMusicians :

==========================================
[note 1 : The following line is not compulsory for the redirection to work ]
defaults.pcm.rate_converter "samplerate_best"

[note 2 : Alsa default is to be concerned because many jack-unaware packages 
output on the sole Alsa Default. (Adobe-Flash being an example) ]

pcm.!default{
        type plug
        slave.pcm "aduplex"
        hint{
                show on
                description "Alsa Default"}}

pcm.aduplex{
        type asym
        playback.pcm "amix"
        capture.pcm "asnoop"}

pcm.amix{
        type dmix
        ipc_key 219345
        slave{
                pcm "hw:Loopback,0,0"
                format S32_LE

[note 3 : You absolutely want to adjust the three following values according 
to your # Jack instance sampling rate and tests you'll have completed on your 
system.]

                period_size 512
                buffer_size 16384
                rate 44100}}

pcm.asnoop{
        type dsnoop
        ipc_key 219346
        slave.pcm "hw:Loopback,0,1"}

pcm.ploop{
        type plug
        slave.pcm "hw:Loopback,1,1"}

pcm.cloop{
        type dsnoop
        ipc_key 219348
        slave{
                pcm "hw:Loopback,1,0"
                format S32_LE

[note : idem note 3]

                period_size 2048
                buffer_size 8192
                rate 44100}}

[note 4 : Of course, you will adapt the values given as ipc_key to your 
system. I mean, choosing values that won't ever be used by other programs you 
run. ]
=========================================
In addition to this, you must create jack permanent clients => create a script 
named for example Loop2Jack :


/usr/bin/alsa_out -j ploop -dploop -q 1 2>&1 1> /dev/null &
/usr/bin/alsa_in -j cloop -dcloop -q 1 2>&1 1> /dev/null &
sleep 1
jack_connect cloop:capture_1 system:playback_1
jack_connect cloop:capture_2 system:playback_2
jack_connect system:capture_1 ploop:playback_1
jack_connect system:capture_1 ploop:playback_2
exit 0
==========================================
You'll then ensure that this script is executed after jackd's startup.
Launch it manually or, if you use qjackctl there is an option field for 
allowing this as part of the setup/options page.

Having (I hope so) helped you to do what you wish, I must tell that having 
used this facility myself, I came to realize that I was nothing but stupid.
As a matter of fact, I think we can state that nowadays, the applications not 
jack-aware are consumer grade applications playing medias of relatively poor 
audio quality.
You will have additionally noticed that the dmix plugin was used in the 
asound.conf with a rather huge buffer that will have the consequence to add 
some significant latency.
Why then, would you want to bother you real-time-low-latency sound server with 
this sort of sound ?
Because your jackd might almost probably be "wired" to your best quality sound 
device, why would you want <=22Khz samples from your speech-synth, Skype, DE's 
sound notifier... to be driven to this device ?
Worse ! Why would you want these sounds of somehow random occurrence to 
spuriously pollute your precious 24bits >=96KHz recording ?

OK then, of course this is nothing but an opinion from my personal experience.
I only suggest you wonder before bothering too much.

Regards,

aCOSwt

1342597751.14534_0.ltw:2,a <201207180949.12949.acoswt at orange dot fr>