[Bug 784616] rtpgsmpay: GSM over RTP gives bad result / broken payload data

[GStreamer (GNOME Bugzilla)]

https://bugzilla.gnome.org/show_bug.cgi?id=784616

Tim-Philipp Müller <t.i.m at zen.co.uk> changed:

           What    |Removed                     |Added
----------------------------------------------------------------------------
             Status|REOPENED                    |RESOLVED
         Resolution|---                         |FIXED
   Target Milestone|git master                  |1.12.2
            Summary|GSM over RTP gives bad      |rtpgsmpay: GSM over RTP
                   |result                      |gives bad result / broken
                   |                            |payload data

--- Comment #5 from Tim-Philipp Müller <t.i.m at zen.co.uk> ---
Thanks, should be fixed in master and 1.12 branch now:

commit c7f42cc3bc192ef0122b68d95cb24e187b975623
Author: Yasushi SHOJI <yashi at atmark-techno.com>
Date:   Fri Jul 7 21:15:57 2017 +0900

    rtpgsmpay: fix accidental garbage data before actual payload

    Do not allocate payload size outbuf if appending payload buffer.

    The commit 137672ff1824948bda4b1b1967de8c24a0055b67 attached payload
    to the output buffer but forgot to remove payload allocation.  That
    effectively doubled payload size and add zero'ed or random bytes.

    Makes the following pipeline work again:

    gst-launch-1.0 -v audiotestsrc wave=2 ! gsmenc ! rtpgsmpay ! rtpgsmdepay !
gsmdec ! autoaudiosink

    https://bugzilla.gnome.org/show_bug.cgi?id=784616

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