Why is wav file mixed by soundmixer plugin truncated?

[philippe renon]

I have run into similar issues. Seems like clocks provided by sound sinks are broken in some situations.See Bug 788362 – directsoundsrc: provided clock is not monotonic after pause and resume

  
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Bug 788362 – directsoundsrc: provided clock is not monotonic after pause and resume
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Try to add provide-clock=false to your alsasink. The pipeline will then use a system clock which behaves better in my case. 

    Le Lundi 4 décembre 2017 10h32, toub <etienne.laurent at pa.cotte.fr> a écrit :
 

 I develop in C an application relying on soundmixer plugin. My aim is to
trigger sounds on events. As reactivity is critical, I wish to keep
gstreamer pipeline alive, and dynamically plug new sounds on the audiomixer
element (after applying an offset on the new sink of the mixer).
It works pretty well, except that each time if plug a new sound in, the
first 200msec (more or less) of the sound get truncated.
I've been looking into the code, everything looks fine as far as I can see
(no mistake on sound plugin, neither on running time/offset applied).
So I made a try on a simple pipeline with gst-launch, with an offset applied
on the mixer sink: 

gst-launch-1.0 filesrc location=/etc/pa/doublclick_aigu_grave.wav ! wavparse
! audiomixer sink_0::offset=1000000000 ! alsasink

It turned out that this pipeline truncate the beginning of the sound as
well. When I activated the logs with --gst-debug=alsa:5, I got a bunch of
logs before sound started :

alsa gstalsasink.c:1054:gst_alsasink_write:<alsasink0> written 441 frames
out of 441

What is going on here ? Why is the sound truncated ?




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