[Bug 646167] New: [gstrtpjitterbuffer] stops processing when mode=buffer and drop-on-latency=1

[GStreamer (bugzilla.gnome.org)]

https://bugzilla.gnome.org/show_bug.cgi?id=646167
  GStreamer | gst-plugins-good | 0.10.30

           Summary: [gstrtpjitterbuffer] stops processing when mode=buffer
                    and drop-on-latency=1
    Classification: Platform
           Product: GStreamer
           Version: 0.10.30
        OS/Version: Linux
            Status: UNCONFIRMED
          Severity: normal
          Priority: Normal
         Component: gst-plugins-good
        AssignedTo: gstreamer-bugs at lists.freedesktop.org
        ReportedBy: tiagokatcipis at gmail.com
         QAContact: gstreamer-bugs at lists.freedesktop.org
      GNOME target: ---
     GNOME version: ---


I was doing some tests with gstrtpjitterbuffer, basically generating audio with
audiotestsrc and sending it using rtp (ptime must be 20ms on my tests).

The pipeline used to generate audio is:

gst-launch -m audiotestsrc is_live=true ! alawenc !
audio/x-alaw,rate=8000,channels=1 ! rtppcmapay min-ptime=20000000
max-ptime=20000000 ! udpsink host=127.0.0.1 port=5000

And on the receive side i have an application that does some measurements using
pad probing on the gstrtpjitterbuffer element. Everything was ok until i
started to set the gstrtpjitterbuffer "mode" property to "buffer". I got a
little audio and them silence. First thing i thought was "my test must be
broken". But when i started to debug to find out what i was doing wrong i found
that what was making my test getting locked was:

g_object_set(G_OBJECT (jitter_buffer), "drop-on-latency", TRUE, NULL);

so i tried to reproduce the problem using gst-launch and it really stops
processing, (the "buffering"  message gets on 71% and stops, no audio is
heard). If i don't set drop-on-latency=1, this don't happens, or if the mode is
not buffer it works with drop-on-latency=1.

Basically:


works: gst-launch udpsrc port=5000 caps=application/x-rtp,clock-rate=8000 !
gstrtpptdemux ! gstrtpjitterbuffer mode=none drop-on-latency=1 ! decodebin !
autoaudiosink

works: gst-launch udpsrc port=5000 caps=application/x-rtp,clock-rate=8000 !
gstrtpptdemux ! gstrtpjitterbuffer mode=none drop-on-latency=0 ! decodebin !
autoaudiosink

works: gst-launch udpsrc port=5000 caps=application/x-rtp,clock-rate=8000 !
gstrtpptdemux ! gstrtpjitterbuffer mode=slave drop-on-latency=1 ! decodebin !
autoaudiosink

works: gst-launch udpsrc port=5000 caps=application/x-rtp,clock-rate=8000 !
gstrtpptdemux ! gstrtpjitterbuffer mode=slave drop-on-latency=0 ! decodebin !
autoaudiosink

works: gst-launch udpsrc port=5000 caps=application/x-rtp,clock-rate=8000 !
gstrtpptdemux ! gstrtpjitterbuffer mode=buffer drop-on-latency=0 ! decodebin !
autoaudiosink

stop processing on 71%: gst-launch udpsrc port=5000
caps=application/x-rtp,clock-rate=8000 ! gstrtpptdemux ! gstrtpjitterbuffer
mode=buffer drop-on-latency=1 ! decodebin ! autoaudiosink


i didn't find anything saying that the drop-on-latency property must not be set
to true when the buffering mode is "buffer", so i thought this must be a bug
(it is pretty simple to reproduce and the pipelines are quite simple too).

The problem happened on Ubuntu 10.10/Gstreamer 0.10.30.

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