[0.11] gst-plugins-good: rtpjitterbuffer/rtpbin: relax dropping rtcp packets

[Wim Taymans]

Module: gst-plugins-good
Branch: 0.11
Commit: 77ebd33991df40c239827f5666f2bbfc558518c4
URL:    http://cgit.freedesktop.org/gstreamer/gst-plugins-good/commit/?id=77ebd33991df40c239827f5666f2bbfc558518c4

Author: Mark Nauwelaerts <mark.nauwelaerts at collabora.co.uk>
Date:   Wed Aug 24 14:37:52 2011 +0200

rtpjitterbuffer/rtpbin: relax dropping rtcp packets

... to at least having it trigger a/v synchronization, possibly without
using provided values which are still not considered sane
(as previously dropped).

---

 gst/rtpmanager/gstrtpbin.c          |   13 +++++++++++++
 gst/rtpmanager/gstrtpjitterbuffer.c |    8 ++++++--
 2 files changed, 19 insertions(+), 2 deletions(-)

diff --git a/gst/rtpmanager/gstrtpbin.c b/gst/rtpmanager/gstrtpbin.c
index c3ca387..beac56c 100644
--- a/gst/rtpmanager/gstrtpbin.c
+++ b/gst/rtpmanager/gstrtpbin.c
@@ -1052,6 +1052,19 @@ gst_rtp_bin_associate (GstRtpBin * bin, GstRtpBinStream * stream, guint8 len,
         stream->ssrc, client, client->cname);
   }
 
+  if (!GST_CLOCK_TIME_IS_VALID (last_extrtptime)) {
+    GST_DEBUG_OBJECT (bin, "invalidated sync data");
+    if (bin->rtcp_sync == GST_RTP_BIN_RTCP_SYNC_RTP) {
+      /* we don't need that data, so carry on,
+       * but make some values look saner */
+      last_extrtptime = base_rtptime;
+    } else {
+      /* nothing we can do with this data in this case */
+      GST_DEBUG_OBJECT (bin, "bailing out");
+      return;
+    }
+  }
+
   /* Take the extended rtptime we found in the SR packet and map it to the
    * local rtptime. The local rtp time is used to construct timestamps on the
    * buffers so we will calculate what running_time corresponds to the RTP
diff --git a/gst/rtpmanager/gstrtpjitterbuffer.c b/gst/rtpmanager/gstrtpjitterbuffer.c
index 7a76853..3e1b546 100644
--- a/gst/rtpmanager/gstrtpjitterbuffer.c
+++ b/gst/rtpmanager/gstrtpjitterbuffer.c
@@ -2032,8 +2032,12 @@ gst_rtp_jitter_buffer_chain_rtcp (GstPad * pad, GstBuffer * buffer)
         diff = ext_rtptime - last_rtptime;
         /* if bigger than 1 second, we drop it */
         if (diff > clock_rate) {
-          GST_DEBUG_OBJECT (jitterbuffer, "dropping, too far ahead");
-          drop = TRUE;
+          GST_DEBUG_OBJECT (jitterbuffer, "too far ahead");
+          /* should drop this, but some RTSP servers end up with bogus
+           * way too ahead RTCP packet when repeated PAUSE/PLAY,
+           * so still trigger rptbin sync but invalidate RTCP data
+           * (sync might use other methods) */
+          ext_rtptime = -1;
         }
         GST_DEBUG_OBJECT (jitterbuffer, "ext last %" G_GUINT64_FORMAT ", diff %"
             G_GUINT64_FORMAT, last_rtptime, diff);