[Bug 692953] alsasink does not synchronise properly with a live streaming source, yet (timing skew)

[GStreamer (bugzilla.gnome.org)]

https://bugzilla.gnome.org/show_bug.cgi?id=692953
  GStreamer | gst-plugins-base | 1.0.10

--- Comment #224 from Rémi Lefèvre <remi.lefevre at parrot.com> 2014-06-02 13:19:09 UTC ---
Hi Tom, 

I had two issues:

1. No audio or choppy audio from the start of the pipeline
2. Audio lost after a while

With your patch, I reproduce the first problem but not the second one yet.
However after more tests, I reproduce the first issue even without having
xruns, so I'm not sure this problem is directly related to the one you are
fixing.


I use the following pipeline to test my issues:

gst-launch-1.0 mpgtssrc ! queue max-size-buffers=0 max-size-time=0 ! tsdemux !
audio/x-eac3 ! ac3parse ! avdec_eac3 ! audioconvert
 ! audio/x-raw,rate=48000,channels=2 ! alsasink latency-time=21334
buffer-time=42667

mpegtssrc is a custom tuner plugin comparable to a simple v4l2src.


On a side note, I get the following warning with your patch:
gst-libs/gst/audio/gstaudioringbuffer.c:1544:10: warning: suggest explicit
braces to avoid ambiguous 'else' [-Wparentheses]

that seems justified for the following code:
      if (!wait_segment (buf))
        if (buf->underrun)
          break;
        else
          goto not_started;

-- 
Configure bugmail: https://bugzilla.gnome.org/userprefs.cgi?tab=email
------- You are receiving this mail because: -------
You are the QA contact for the bug.
You are the assignee for the bug.