Why is wav file mixed by soundmixer plugin truncated?

[Nicolas Dufresne]

Le jeudi 07 décembre 2017 à 02:45 -0700, toub a écrit :
> I made a try with  gst_pipeline_set_latency(). First played sound is
> truncated, next one (while first sound is still playing) is delaued a little
> but not truncated.
> However, as sometimes sound is truncated much more than 200ms, I don't think
> it is a solution to have latency set this way.
> 
> In any way, why do I have to set latency ? As far a I know, buffers produced
> by mixer a timelapse T are not delayed before arriving to alsasink.

As you have live sources you need latency, because by the time we have
capture the audio, the data is already late. The latency is the amount
of time you give to your pipeline to transport data from source to sink
and render it, plus the extra time needed to synchronize to the sink
that renders last.

> 
> Also, I could not find how to use the do-latency signal to modify alsa sink.
> Could you give me a sample ?

There is no example that I know of, I usually start from the default
implementation:

https://cgit.freedesktop.org/gstreamer/gstreamer/tree/gst/gstpipeline.c#n619

> 
> Thanks in any case,
> 
> Étienne
> 
> 
> 
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