[Bug 796508] New: RTP OPUS payloader/depayloader introduce delay which is not compensated properly

[GStreamer (GNOME Bugzilla)]

https://bugzilla.gnome.org/show_bug.cgi?id=796508

            Bug ID: 796508
           Summary: RTP OPUS payloader/depayloader introduce delay which
                    is not compensated properly
    Classification: Platform
           Product: GStreamer
           Version: 1.14.1
                OS: Linux
            Status: NEW
          Severity: normal
          Priority: Normal
         Component: gst-plugins-good
          Assignee: gstreamer-bugs at lists.freedesktop.org
          Reporter: bugzilla at poradnik-webmastera.com
        QA Contact: gstreamer-bugs at lists.freedesktop.org
     GNOME version: ---

Following pipeline creates testout.wav file with two channels. First one comes
from audiotestsrc element, without extra processing. 2nd one is the same audio,
passed via opusenc/rtpopuspay/rtpopusdepay/opusdec. When you check generated
file, you will find that there is 6ms delay between 2nd and 1st channel:

gst-launch-1.0 audiointerleave name=int start-time-selection=first audiotestsrc
wave=ticks is-live=true do-timestamp=true ! tee name=tee ! audioconvert !
audioresample ! capsfilter
caps="audio/x-raw,channels=(int)1,channel-mask=(bitmask)0x1,rate=(int)48000" !
queue ! int.sink_0 tee. ! queue ! audioconvert ! audioresample ! opusenc !
rtpopuspay min-ptime=20000000 max-ptime=20000000 ! capsfilter
caps="application/x-rtp,media=(string)audio,encoding-name=(string)OPUS,payload=(int)97"
! rtpopusdepay ! opusdec ! audioconvert ! audioresample ! capsfilter
caps="audio/x-raw,channels=(int)1,channel-mask=(bitmask)0x2" ! queue !
int.sink_1 int.src ! audioconvert ! audioresample ! wavenc ! filesink
location=testout.wav sync=true

When I removed rtpopuspay/rtpopusdepay from pipeline, this extra delay
disappeared and both channels were properly synchronized:

gst-launch-1.0 audiointerleave name=int start-time-selection=first audiotestsrc
wave=ticks is-live=true do-timestamp=true ! tee name=tee ! audioconvert !
audioresample ! capsfilter
caps="audio/x-raw,channels=(int)1,channel-mask=(bitmask)0x1,rate=(int)48000" !
queue ! int.sink_0 tee. ! queue ! audioconvert ! audioresample ! opusenc !
opusdec ! audioconvert ! audioresample ! capsfilter
caps="audio/x-raw,channels=(int)1,channel-mask=(bitmask)0x2" ! queue !
int.sink_1 int.src ! audioconvert ! audioresample ! wavenc ! filesink
location=testout.wav sync=true

I also tried to create pipeline with PCMA codecs/payloaders, and this one also
is properly synchronized:

gst-launch-1.0 audiointerleave name=int start-time-selection=first audiotestsrc
wave=ticks is-live=true do-timestamp=true ! tee name=tee ! audioconvert !
audioresample ! capsfilter
caps="audio/x-raw,channels=(int)1,channel-mask=(bitmask)0x1,rate=(int)8000" !
queue ! int.sink_0 tee. ! queue ! audioconvert ! audioresample ! alawenc !
rtppcmapay min-ptime=20000000 max-ptime=20000000 ! capsfilter
caps="application/x-rtp,media=(string)audio,encoding-name=(string)PCMA,payload=(int)8"
! rtppcmadepay ! alawdec ! audioconvert ! audioresample ! capsfilter
caps="audio/x-raw,channels=(int)1,channel-mask=(bitmask)0x2" ! queue !
int.sink_1 int.src ! audioconvert ! audioresample ! wavenc ! filesink
location=testout.wav sync=true

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