[gst-devel] JACK Audio Dropouts

[Esben Stien]

Tristan Matthews <tristan at sat.qc.ca> writes:

> You may need a queue element somewhere in your launch line. 

Yes, tried that and didn't help: 

gst-launch -v filesrc location=audiodump.wav ! decodebin ! queue ! audioconvert  ! audioresample ! jackaudiosink sync=false

> Also I would recommend writing a C program to build and run your
> launch-line instead of using gst-launch, as gst-launch is primarily
> used for debugging.

Yes. 

> https://trac.sat.qc.ca/trac/miville/browser/trunk/prototypes/gstreamer/shellScripts/jackHarmonics.sh?rev=195

This played perfectly and got me thinking. This is a generator, so it
creates the sample rate requested. 

I tried playing a file with the same sample rate I use on JACK, which
is 96000 Hz and it played perfectly. So, every file that has the same
sample rate as JACK plays fine. So, I guess the offender is the sample
rate conversion library. There is no secret rabbit code plugin for
gstreamer?.

So, this plays perfectly: 

gst-launch filesrc location=new-wav/a_little_loop_fo_ya.wav ! decodebin ! queue ! audioconvert  ! audioresample ! jackaudiosink

, where the .wav file is 96000 Hz. Didn't even have to use sync=false. 

Is there any way to change the parameters of the sample rate
conversion plugin and maybe use a less expensive algorithm?.

-- 
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