[Bug 761943] New: gstrtpbasepayload: set DEFAULT_PERFECT_RTPTIME to FALSE

[GStreamer (GNOME Bugzilla)]

https://bugzilla.gnome.org/show_bug.cgi?id=761943

            Bug ID: 761943
           Summary: gstrtpbasepayload: set DEFAULT_PERFECT_RTPTIME to
                    FALSE
    Classification: Platform
           Product: GStreamer
           Version: git master
                OS: Mac OS
            Status: NEW
          Severity: normal
          Priority: Normal
         Component: gst-plugins-base
          Assignee: gstreamer-bugs at lists.freedesktop.org
          Reporter: havard.graff at gmail.com
        QA Contact: gstreamer-bugs at lists.freedesktop.org
                CC: gstreamer at pexip.com
     GNOME version: ---

Created attachment 320983
  --> https://bugzilla.gnome.org/attachment.cgi?id=320983&action=edit
patch

A very common pipeline is feeding an encoder inheriting from
GstAudioEncoder into a payloader inheriting from GstRtpBasePayload.

If this is the case, the rtp-timestamp will, if perfect_rtptime is TRUE,
reflect the amount of bytes in each packet, and not (like it should)
reflect the timestamp of the packets in the clock-rate domain.

Example:
20ms AAC packets with clock-rate of 90000, should have the rtp-timestamp
incrementing with 90000 * 20 / 1000 = 1800.

Currently, because perfect_rtptime is default TRUE, the AAC-packets
will be incrementing with the buffer-size of the packets, which is
typically 64000 * 20 / 1000 = 1280.

In the case that bitrate and clock-rate are the same (like with ALAW and
MULAW) it will work by chance, which is probably why it has not been
discovered earlier.

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