[Bug 796810] New: webrtc SDP RTX processing bug with no ssrc
GStreamer (GNOME Bugzilla)
bugzilla at gnome.org
Sat Jul 14 13:14:19 UTC 2018
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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