[Bug 796810] New: webrtc SDP RTX processing bug with no ssrc

[GStreamer (GNOME Bugzilla)]

https://bugzilla.gnome.org/show_bug.cgi?id=796810

            Bug ID: 796810
           Summary: webrtc SDP RTX processing bug with no ssrc
    Classification: Platform
           Product: GStreamer
           Version: git master
                OS: Linux
            Status: NEW
          Severity: normal
          Priority: Normal
         Component: gst-plugins-bad
          Assignee: gstreamer-bugs at lists.freedesktop.org
          Reporter: thaytan at noraisin.net
        QA Contact: gstreamer-bugs at lists.freedesktop.org
     GNOME version: ---

The unit test for webrtc encounters a case where it uses an uninitialised
variable here in sdp_media_from_transceiver(), because the caps it generates
for type == GST_WEBRTC_SDP_TYPE_OFFER don't contain an ssrc.

This patch adds a warning when that happens, but I'm not sure why it's
happening:

diff --git a/ext/webrtc/gstwebrtcbin.c b/ext/webrtc/gstwebrtcbin.c
index 9baebede3..ffe0bb1c7 100644
--- a/ext/webrtc/gstwebrtcbin.c
+++ b/ext/webrtc/gstwebrtcbin.c
@@ -1701,15 +1701,17 @@ sdp_media_from_transceiver (GstWebRTCBin * webrtc,
GstSDPMedia * media,
     gint clockrate = -1;
     gint rtx_target_pt;
     gint original_rtx_target_pt;        /* Workaround chrome bug:
https://bugs.chromium.org/p/webrtc/issues/detail?id=6196 */
-    guint rtx_target_ssrc;
+    guint rtx_target_ssrc = -1;

     if (gst_structure_get_int (s, "payload", &rtx_target_pt))
       g_array_append_val (reserved_pts, rtx_target_pt);

     original_rtx_target_pt = rtx_target_pt;

-    gst_structure_get_int (s, "clock-rate", &clockrate);
-    gst_structure_get_uint (s, "ssrc", &rtx_target_ssrc);
+    if (!gst_structure_get_int (s, "clock-rate", &clockrate))
+      GST_WARNING_OBJECT (webrtc, "Caps %" GST_PTR_FORMAT " are missing
clock-rate", caps);
+    if (!gst_structure_get_uint (s, "ssrc", &rtx_target_ssrc))
+      GST_WARNING_OBJECT (webrtc, "Caps %" GST_PTR_FORMAT " are missing ssrc",
caps);

     _pick_fec_payload_types (webrtc, WEBRTC_TRANSCEIVER (trans), reserved_pts,
         clockrate, &rtx_target_pt, media);


Here's the warning:

0:00:00.874198556 16302      0x1976720 WARN               webrtcbin
gstwebrtcbin.c:1714:sdp_media_from_transceiver:<webrtcbin0> Caps
application/x-rtp, payload=(int)96, encoding-name=(string)OPUS,
media=(string)audio, clock-rate=(int)48000, rtcp-fb-nack-pli=(boolean)true are
missing ssrc

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