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rtspsrc: Don't reset time in flush-stop

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Don't reset the time in flush-stop. Live sources can do this flush in the
playing state and so the pipeline will never have a chance to update the
base_time of the elements, which only happens when going from paused to
playing.
This commit is contained in:
Wim Taymans 2012-06-13 13:36:45 +02:00
parent 59191412eb
commit 694be55c05
1 changed files with 3 additions and 21 deletions
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24
gst/rtsp/gstrtspsrc.c
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@ -1664,8 +1664,6 @@ gst_rtspsrc_flush (GstRTSPSrc * src, gboolean flush, gboolean playing)
gint cmd, i;
GstState state;
GList *walk;
GstClock *clock;
GstClockTime base_time = GST_CLOCK_TIME_NONE;
if (flush) {
event = gst_event_new_flush_start ();
@ -1673,31 +1671,20 @@ gst_rtspsrc_flush (GstRTSPSrc * src, gboolean flush, gboolean playing)
cmd = CMD_WAIT;
state = GST_STATE_PAUSED;
} else {
event = gst_event_new_flush_stop (TRUE);
event = gst_event_new_flush_stop (FALSE);
GST_DEBUG_OBJECT (src, "stop flush; playing %d", playing);
cmd = CMD_LOOP;
if (playing)
state = GST_STATE_PLAYING;
else
state = GST_STATE_PAUSED;
clock = gst_element_get_clock (GST_ELEMENT_CAST (src));
if (clock) {
base_time = gst_clock_get_time (clock);
gst_object_unref (clock);
}
}
gst_rtspsrc_push_event (src, event);
gst_rtspsrc_loop_send_cmd (src, cmd, CMD_LOOP);
/* set up manager before data-flow resumes */
/* to manage jitterbuffer buffer mode */
if (src->manager) {
gst_element_set_base_time (GST_ELEMENT_CAST (src->manager), base_time);
/* and to have base_time trickle further down,
* e.g. to jitterbuffer for its timeout handling */
if (base_time != -1)
gst_element_set_state (GST_ELEMENT_CAST (src->manager), state);
}
if (src->manager)
gst_element_set_state (GST_ELEMENT_CAST (src->manager), state);
/* make running time start start at 0 again */
for (walk = src->streams; walk; walk = g_list_next (walk)) {
@ -1706,15 +1693,10 @@ gst_rtspsrc_flush (GstRTSPSrc * src, gboolean flush, gboolean playing)
for (i = 0; i < 2; i++) {
/* for udp case */
if (stream->udpsrc[i]) {
if (base_time != -1)
gst_element_set_base_time (stream->udpsrc[i], base_time);
gst_element_set_state (stream->udpsrc[i], state);
}
}
}
/* for tcp interleaved case */
if (base_time != -1)
gst_element_set_base_time (GST_ELEMENT_CAST (src), base_time);
}
static GstRTSPResult