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rtspsrc: Fix segment in TCP mode

Browse source 
It is expected that buffers are time-stamped with running time. Set
a segment accordingly. In this case we pick 0,-1 as this is what udpsrc
would do. Depayloaders will update the segment to reflect the playback
position.

https://bugzilla.gnome.org/show_bug.cgi?id=635701
This commit is contained in:
Nicolas Dufresne 2015-03-25 15:11:34 -04:00
parent c3bb399fd3
commit 84725d62b5
2 changed files with 15 additions and 24 deletions
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37
gst/rtsp/gstrtspsrc.c
View file

@ -1661,10 +1661,9 @@ gst_rtspsrc_cleanup (GstRTSPSrc * src)
gst_sdp_message_free (src->sdp);
src->sdp = NULL;
}
if (src->start_segment) {
gst_event_unref (src->start_segment);
src->start_segment = NULL;
}
src->need_segment = FALSE;
if (src->provided_clock) {
gst_object_unref (src->provided_clock);
src->provided_clock = NULL;
@ -4541,7 +4540,6 @@ gst_rtspsrc_handle_data (GstRTSPSrc * src, GstRTSPMessage * message)
guint size;
GstBuffer *buf;
gboolean is_rtcp;
GstEvent *event;
channel = message->type_data.data.channel;
@ -4598,15 +4596,12 @@ gst_rtspsrc_handle_data (GstRTSPSrc * src, GstRTSPMessage * message)
gchar *uri;
GList *streams;
guint group_id = gst_util_group_id_next ();
GstSegment segment;
/* generate an SHA256 sum of the URI */
cs = g_checksum_new (G_CHECKSUM_SHA256);
uri = src->conninfo.location;
g_checksum_update (cs, (const guchar *) uri, strlen (uri));
gst_segment_init (&segment, GST_FORMAT_TIME);
for (streams = src->streams; streams; streams = g_list_next (streams)) {
GstRTSPStream *ostream = (GstRTSPStream *) streams->data;
GstCaps *caps;
@ -4649,23 +4644,12 @@ gst_rtspsrc_handle_data (GstRTSPSrc * src, GstRTSPMessage * message)
}
}
}
/* Push a SEGMENT event if we don't have one pending, if we have one
* pending we will just send that one a few lines below to all pads
*/
if (!src->start_segment)
gst_rtspsrc_stream_push_event (src, ostream,
gst_event_new_segment (&segment));
}
g_checksum_free (cs);
gst_rtspsrc_activate_streams (src);
src->need_activate = FALSE;
}
if ((event = src->start_segment) != NULL) {
src->start_segment = NULL;
gst_rtspsrc_push_event (src, event);
src->need_segment = TRUE;
}
if (src->base_time == -1) {
@ -4692,6 +4676,15 @@ gst_rtspsrc_handle_data (GstRTSPSrc * src, GstRTSPMessage * message)
GST_OBJECT_UNLOCK (src);
}
/* If needed send a new segment, don't forget we are live and buffer are
* timestamped with running time */
if (src->need_segment) {
GstSegment segment;
src->need_segment = FALSE;
gst_segment_init (&segment, GST_FORMAT_TIME);
gst_rtspsrc_push_event (src, gst_event_new_segment (&segment));
}
if (stream->discont && !is_rtcp) {
/* mark first RTP buffer as discont */
GST_BUFFER_FLAG_SET (buf, GST_BUFFER_FLAG_DISCONT);
@ -7535,9 +7528,7 @@ gst_rtspsrc_play (GstRTSPSrc * src, GstSegment * segment, gboolean async)
gst_rtsp_message_take_header (&request, GST_RTSP_HDR_RANGE, hval);
/* store the newsegment event so it can be sent from the streaming thread. */
if (src->start_segment)
gst_event_unref (src->start_segment);
src->start_segment = gst_event_new_segment (segment);
src->need_segment = TRUE;
}
if (segment->rate != 1.0) {

2
gst/rtsp/gstrtspsrc.h
View file

@ -185,7 +185,7 @@ struct _GstRTSPSrc {
gboolean need_range;
gboolean skip;
gint free_channel;
GstEvent *start_segment;
gboolean need_segment;
GstClockTime base_time;
/* UDP mode loop */