mirrors/gstreamer
mirrors/gstreamer
1
1
Fork 0
mirror of https://gitlab.freedesktop.org/gstreamer/gstreamer.git synced 2024-12-25 09:40:37 +00:00
Code Issues 1 Projects Releases Wiki Activity

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

Browse source 
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
This commit is contained in:
Andrew 2017-02-08 13:36:00 +00:00 committed by Sebastian Dröge
parent 16941255e7
commit 76792a5c20
3 changed files with 10 additions and 7 deletions
Show stats Download patch file Download diff file Expand all files Collapse all files

11
gst/rtpmanager/gstrtpjitterbuffer.c
View file

@ -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 */

4
gst/rtpmanager/rtpjitterbuffer.c
View file

@ -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;

2
gst/rtpmanager/rtpjitterbuffer.h
View file

@ -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__ */