[Bug 780044] New: Audio stalls receiving Opus over RTP in a Raspberry Pi 3

GStreamer (GNOME Bugzilla) bugzilla at gnome.org
Tue Mar 14 17:28:13 UTC 2017


https://bugzilla.gnome.org/show_bug.cgi?id=780044

            Bug ID: 780044
           Summary: Audio stalls receiving Opus over RTP in a Raspberry Pi
                    3
    Classification: Platform
           Product: GStreamer
           Version: 1.10.4
                OS: Linux
            Status: NEW
          Severity: normal
          Priority: Normal
         Component: gst-plugins-base
          Assignee: gstreamer-bugs at lists.freedesktop.org
          Reporter: aperez at igalia.com
        QA Contact: gstreamer-bugs at lists.freedesktop.org
     GNOME version: ---

I have been able to consistently reproduce this issue, which has been
reported before in the mailing list:

  
http://gstreamer-devel.966125.n4.nabble.com/Audio-stalling-when-using-Opus-to-send-audio-to-Raspberry-Pi-3-tt4681213.html

To reproduce, launch the commands suggested in the email above. A recap:

 - In a computer:
   gst-launch-1.0 audiotestsrc ! opusenc ! rtpopuspay ! udpsink host=<IP of
RPi> port=5555 

 - In the Raspberry Pi:
   GST_DEBUG=4 gst-launch-1.0 udpsrc port=5555 caps="application/x-rtp" !
rtpopusdepay ! opusdec ! audioconvert ! pulsesink

Now, in order to reproduce the issue very easily, repeteadly disable and
re-enable the network interface of the *computer*. After 3-4 cycles, the
warning messages start to appear:

  0:00:46.587793889   355 0x712014c0 WARN                   pulse
pulsesink.c:702:gst_pulsering_stream_underflow_cb:<pulsesink0> Got underflow
  0:00:54.536071281   355 0x712014c0 WARN                   pulse
pulsesink.c:702:gst_pulsering_stream_underflow_cb:<pulsesink0> Got underflow
  0:01:15.323500284   355   0x65a260 WARN           audiobasesink
gstaudiobasesink.c:1512:gst_audio_base_sink_skew_slaving:<pulsesink0> correct
clock skew -0:00:00.021637386 < -+0:00:00.020000000
  0:01:15.463535544   355   0x65a260 WARN           audiobasesink
gstaudiobasesink.c:1512:gst_audio_base_sink_skew_slaving:<pulsesink0> correct
clock skew -0:00:00.024740252 < -+0:00:00.020000000
  0:01:15.563458930   355   0x65a260 WARN           audiobasesink
gstaudiobasesink.c:1512:gst_audio_base_sink_skew_slaving:<pulsesink0> correct
clock skew -0:00:00.027595592 < -+0:00:00.020000000
  ...

After making a few more tests, I could find out that the issue is *not*
trigger in any of the following cases (i.e. they work fine):

 - Sending audio in the other direction (RPi → PC) and fiddling with the
   network interface of the RPi.
 - Sending audio from a PC to another PC.
 - Using “alsasink” in the RPi.

-- 
You are receiving this mail because:
You are the QA Contact for the bug.
You are the assignee for the bug.


More information about the gstreamer-bugs mailing list