[gstreamer-bugs] [Bug 576524] New: vorbisenc audio quality problems when recording from RTP streams

[GStreamer (bugzilla.gnome.org)]

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  GStreamer | gst-plugins-base | Ver: 0.10.21
           Summary: vorbisenc audio quality problems when recording from RTP
                    streams
           Product: GStreamer
           Version: 0.10.21
          Platform: Other
        OS/Version: Linux
            Status: UNCONFIRMED
          Severity: normal
          Priority: Normal
         Component: gst-plugins-base
        AssignedTo: gstreamer-bugs at lists.sourceforge.net
        ReportedBy: jordijp at gmail.com
         QAContact: gstreamer-bugs at lists.sourceforge.net
                CC: jordijp at gmail.com
     GNOME version: Unspecified
   GNOME milestone: Unspecified


Trying to record audio from a RTP stream.


Sender:

gst-launch-0.10 -v audiotestsrc ! lame ! rtpmpapay ! udpsink 


Receiver:

gst-launch-0.10 --gst-debug-level=3 udpsrc ! application/x-rtp, payload=14,
clock-rate=90000, media=audio ! gstrtpjitterbuffer ! rtpmpadepay ! flump3dec !
audioconvert ! vorbisenc ! oggmux ! filesink location=audio.ogg

is getting this errors:

0:00:07.465505183 10321      0x2573100 WARN             vorbisenc
vorbisenc.c:1194:gst_vorbis_enc_chain:<vorbisenc0> Buffer is discontinuous,
flushing encoder and restarting (Discont from 0:00:07.177071624 to
0:00:07.233669678)

which relays on the recorded audio quality :-(

NOTE: using "audiorate" removes these errors, but does not fix the audio
quality.

Workaround:

Prior to filing this report, I compared the audio quality with:

gst-launch-0.10 --gst-debug-level=3 udpsrc ! application/x-rtp, payload=14,
clock-rate=90000, media=audio ! gstrtpjitterbuffer ! rtpmpadepay ! mp3parse !
filesink location=audio.mp3

Recording in MP3 gives a better audio quality.


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