rtpjitterbuffer: Don't always reset PTS to 0 after a gap

In function rtp_jitter_buffer_calculate_pts: If gap in incoming RTP
timestamps is more than (3 * jbuf->clock_rate) we call
rtp_jitter_buffer_reset_skew which resets pts to 0. So components down
the pipeline (playes, mixers) just skip frames/samples until pts becomes
equal to pts before gap.

In version 1.10.2 and before this checking was bypassed for packets with
"estimated dts", and gaps were handled correctly.

https://bugzilla.gnome.org/show_bug.cgi?id=778341

3 files changed, 10 insertions, 7 deletions

diff --git a/gst/rtpmanager/gstrtpjitterbuffer.c b/gst/rtpmanager/gstrtpjitterbuffer.c
index 2ca9fe742..64f05dc41 100644
--- a/gst/rtpmanager/gstrtpjitterbuffer.c
+++ b/gst/rtpmanager/gstrtpjitterbuffer.c
@@ -2929,8 +2929,9 @@ gst_rtp_jitter_buffer_chain (GstPad * pad, GstObject * parent,
     GST_DEBUG_OBJECT (jitterbuffer, "First buffer #%d", seqnum);
 
     /* calculate a pts based on rtptime and arrival time (dts) */
-    pts = rtp_jitter_buffer_calculate_pts (priv->jbuf, dts, rtptime,
-        gst_element_get_base_time (GST_ELEMENT_CAST (jitterbuffer)));
+    pts =
+        rtp_jitter_buffer_calculate_pts (priv->jbuf, dts, estimated_dts,
+        rtptime, gst_element_get_base_time (GST_ELEMENT_CAST (jitterbuffer)));
 
     /* we don't know what the next_in_seqnum should be, wait for the last
      * possible moment to push this buffer, maybe we get an earlier seqnum
@@ -2981,8 +2982,10 @@ gst_rtp_jitter_buffer_chain (GstPad * pad, GstObject * parent,
     g_queue_clear (&priv->gap_packets);
 
     /* calculate a pts based on rtptime and arrival time (dts) */
-    pts = rtp_jitter_buffer_calculate_pts (priv->jbuf, dts, rtptime,
-        gst_element_get_base_time (GST_ELEMENT_CAST (jitterbuffer)));
+    /* If we estimated the DTS, don't consider it in the clock skew calculations */
+    pts =
+        rtp_jitter_buffer_calculate_pts (priv->jbuf, dts, estimated_dts,
+        rtptime, gst_element_get_base_time (GST_ELEMENT_CAST (jitterbuffer)));
 
     if (G_LIKELY (gap == 0)) {
       /* packet is expected */
diff --git a/gst/rtpmanager/rtpjitterbuffer.c b/gst/rtpmanager/rtpjitterbuffer.c
index 03b7ebc1e..a706d6283 100644
--- a/gst/rtpmanager/rtpjitterbuffer.c
+++ b/gst/rtpmanager/rtpjitterbuffer.c
@@ -690,7 +690,7 @@ queue_do_insert (RTPJitterBuffer * jbuf, GList * list, GList * item)
 
 GstClockTime
 rtp_jitter_buffer_calculate_pts (RTPJitterBuffer * jbuf, GstClockTime dts,
-      guint32 rtptime, GstClockTime base_time)
+    gboolean estimated_dts, guint32 rtptime, GstClockTime base_time)
 {
   guint64 ext_rtptime;
   GstClockTime gstrtptime, pts;
@@ -703,7 +703,7 @@ rtp_jitter_buffer_calculate_pts (RTPJitterBuffer * jbuf, GstClockTime dts,
    * Only reset if valid input time, which is likely for UDP input
    * where we expect this might happen due to async thread effects
    * (in seek and state change cycles), but not so much for TCP input */
-  if (GST_CLOCK_TIME_IS_VALID (dts) &&
+  if (GST_CLOCK_TIME_IS_VALID (dts) && !estimated_dts &&
       jbuf->mode != RTP_JITTER_BUFFER_MODE_SLAVE &&
       jbuf->base_time != -1 && jbuf->last_rtptime != -1) {
     GstClockTime ext_rtptime = jbuf->ext_rtptime;
diff --git a/gst/rtpmanager/rtpjitterbuffer.h b/gst/rtpmanager/rtpjitterbuffer.h
index d04b5fd96..4d3c6fe16 100644
--- a/gst/rtpmanager/rtpjitterbuffer.h
+++ b/gst/rtpmanager/rtpjitterbuffer.h
@@ -189,7 +189,7 @@ void                  rtp_jitter_buffer_get_sync         (RTPJitterBuffer *jbuf,
                                                           guint64 *timestamp, guint32 *clock_rate,
                                                           guint64 *last_rtptime);
 
-GstClockTime          rtp_jitter_buffer_calculate_pts    (RTPJitterBuffer * jbuf, GstClockTime dts,
+GstClockTime          rtp_jitter_buffer_calculate_pts    (RTPJitterBuffer * jbuf, GstClockTime dts, gboolean estimated_dts,
                                                           guint32 rtptime, GstClockTime base_time);
 
 #endif /* __RTP_JITTER_BUFFER_H__ */