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gst/rtsp/gstrtspsrc.*: In TCP mode, only timestamp the first buffer. TCP is not real time and it does not make sense ...

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Original commit message from CVS:
* gst/rtsp/gstrtspsrc.c: (gst_rtspsrc_loop_interleaved),
(gst_rtspsrc_play):
* gst/rtsp/gstrtspsrc.h:
In TCP mode, only timestamp the first buffer. TCP is not real time and
it does not make sense to try to skew compensate, also some servers send
the first batch of data in a burst.
This commit is contained in:
Wim Taymans 2007-09-28 14:56:19 +00:00
parent c57ce8b9d5
commit bea9010658
3 changed files with 31 additions and 9 deletions
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9
ChangeLog
View file

@ -1,3 +1,12 @@
2007-09-28 Wim Taymans <wim.taymans@gmail.com>
* gst/rtsp/gstrtspsrc.c: (gst_rtspsrc_loop_interleaved),
(gst_rtspsrc_play):
* gst/rtsp/gstrtspsrc.h:
In TCP mode, only timestamp the first buffer. TCP is not real time and
it does not make sense to try to skew compensate, also some servers send
the first batch of data in a burst.
2007-09-27 Tim-Philipp Müller <tim at centricular dot net>
* gst/matroska/matroska-demux.c:

30
gst/rtsp/gstrtspsrc.c
View file

@ -2450,15 +2450,6 @@ gst_rtspsrc_loop_interleaved (GstRTSPSrc * src)
/* don't need message anymore */
gst_rtsp_message_unset (&message);
/* set reception timestamp */
GST_OBJECT_LOCK (src);
if (GST_ELEMENT_CLOCK (src)) {
GstClockTime now = gst_clock_get_time (GST_ELEMENT_CLOCK (src));
GST_BUFFER_TIMESTAMP (buf) = now - GST_ELEMENT_CAST (src)->base_time;
}
GST_OBJECT_UNLOCK (src);
GST_DEBUG_OBJECT (src, "pushing data of size %d on channel %d", size,
channel);
@ -2473,10 +2464,30 @@ gst_rtspsrc_loop_interleaved (GstRTSPSrc * src)
gst_buffer_set_caps (buf, stream->caps);
}
if (src->base_time == -1) {
/* Take current running_time. This timestamp will be put on
* the first buffer of each stream because we are a live source and so we
* timestamp with the running_time. When we are dealing with TCP, we also
* only timestamp the first buffer (using the DISCONT flag) because a server
* typically bursts data, for which we don't want to compensate by speeding
* up the media. The other timestamps will be interpollated from this one
* using the RTP timestamps. */
GST_OBJECT_LOCK (src);
if (GST_ELEMENT_CLOCK (src)) {
GstClockTime now = gst_clock_get_time (GST_ELEMENT_CLOCK (src));
src->base_time = now - GST_ELEMENT_CAST (src)->base_time;
}
GST_OBJECT_UNLOCK (src);
}
if (stream->discont && !is_rtcp) {
/* mark first RTP buffer as discont */
GST_BUFFER_FLAG_SET (buf, GST_BUFFER_FLAG_DISCONT);
stream->discont = FALSE;
/* first buffer gets the timestamp, other buffers are not timestamped and
* their presentation time will be interpollated from the rtp timestamps. */
GST_BUFFER_TIMESTAMP (buf) = src->base_time;
}
/* chain to the peer pad */
@ -4157,6 +4168,7 @@ gst_rtspsrc_play (GstRTSPSrc * src)
gst_task_set_lock (src->task, GST_RTSP_STREAM_GET_LOCK (src));
}
src->running = TRUE;
src->base_time = -1;
src->state = GST_RTSP_STATE_PLAYING;
gst_rtspsrc_loop_send_cmd (src, CMD_WAIT, FALSE);
gst_task_start (src->task);

1
gst/rtsp/gstrtspsrc.h
View file

@ -135,6 +135,7 @@ struct _GstRTSPSrc {
gint free_channel;
GstEvent *close_segment;
GstEvent *start_segment;
GstClockTime base_time;
/* UDP mode loop */
gint loop_cmd;