mirrors/gstreamer
mirrors/gstreamer
1
1
Fork 0
mirror of https://gitlab.freedesktop.org/gstreamer/gstreamer.git synced 2025-01-26 00:58:12 +00:00
Code Issues 1 Projects Releases Wiki Activity

gst/rtsp/gstrtspsrc.c: More seeking fixes, mostly passing around the new playback segment in order to configure it pr...

Browse source 
Original commit message from CVS:
* gst/rtsp/gstrtspsrc.c: (gst_rtspsrc_flush),
(gst_rtspsrc_do_seek), (gst_rtspsrc_perform_seek),
(gst_rtspsrc_configure_caps), (gst_rtspsrc_loop_udp),
(gst_rtspsrc_parse_range), (gst_rtspsrc_open), (gst_rtspsrc_play),
(gst_rtspsrc_change_state):
More seeking fixes, mostly passing around the new playback segment in
order to configure it properly.
Also reset base_time of udp sources when setting them back to PLAYING as
a temporary hack until core supports seek in live sources properly.
This commit is contained in:
Wim Taymans 2007-10-08 11:58:51 +00:00
parent f8df008747
commit 92e16a65ae
2 changed files with 57 additions and 38 deletions
Show stats Download patch file Download diff file Expand all files Collapse all files

12
ChangeLog
View file

@ -1,3 +1,15 @@
2007-10-08 Wim Taymans <wim.taymans@gmail.com>
* gst/rtsp/gstrtspsrc.c: (gst_rtspsrc_flush),
(gst_rtspsrc_do_seek), (gst_rtspsrc_perform_seek),
(gst_rtspsrc_configure_caps), (gst_rtspsrc_loop_udp),
(gst_rtspsrc_parse_range), (gst_rtspsrc_open), (gst_rtspsrc_play),
(gst_rtspsrc_change_state):
More seeking fixes, mostly passing around the new playback segment in
order to configure it properly.
Also reset base_time of udp sources when setting them back to PLAYING as
a temporary hack until core supports seek in live sources properly.
2007-10-08 Wim Taymans <wim.taymans@gmail.com> 2007-10-08 Wim Taymans <wim.taymans@gmail.com>
* gst/rtp/gstrtpmp4adepay.c: * gst/rtp/gstrtpmp4adepay.c:

81
gst/rtsp/gstrtspsrc.c
View file

@ -201,7 +201,7 @@ static GstRTSPResult gst_rtspsrc_send_cb (GstRTSPExtension * ext,
GstRTSPMessage * request, GstRTSPMessage * response, GstRTSPSrc * src); GstRTSPMessage * request, GstRTSPMessage * response, GstRTSPSrc * src);
static gboolean gst_rtspsrc_open (GstRTSPSrc * src); static gboolean gst_rtspsrc_open (GstRTSPSrc * src);
static gboolean gst_rtspsrc_play (GstRTSPSrc * src); static gboolean gst_rtspsrc_play (GstRTSPSrc * src, GstSegment * segment);
static gboolean gst_rtspsrc_pause (GstRTSPSrc * src); static gboolean gst_rtspsrc_pause (GstRTSPSrc * src);
static gboolean gst_rtspsrc_close (GstRTSPSrc * src); static gboolean gst_rtspsrc_close (GstRTSPSrc * src);
@ -1124,6 +1124,8 @@ gst_rtspsrc_flush (GstRTSPSrc * src, gboolean flush)
gint cmd, i; gint cmd, i;
GstState state; GstState state;
GList *walk; GList *walk;
GstClock *clock;
GstClockTime base_time = -1;
if (flush) { if (flush) {
event = gst_event_new_flush_start (); event = gst_event_new_flush_start ();
@ -1135,6 +1137,11 @@ gst_rtspsrc_flush (GstRTSPSrc * src, gboolean flush)
GST_DEBUG_OBJECT (src, "stop flush"); GST_DEBUG_OBJECT (src, "stop flush");
cmd = CMD_WAIT; cmd = CMD_WAIT;
state = GST_STATE_PLAYING; state = GST_STATE_PLAYING;
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_push_event (src, event);
gst_rtspsrc_loop_send_cmd (src, cmd, flush); gst_rtspsrc_loop_send_cmd (src, cmd, flush);
@ -1145,6 +1152,8 @@ gst_rtspsrc_flush (GstRTSPSrc * src, gboolean flush)
for (i = 0; i < 2; i++) { for (i = 0; i < 2; i++) {
if (stream->udpsrc[i]) { 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); gst_element_set_state (stream->udpsrc[i], state);
} }
} }
@ -1182,11 +1191,11 @@ gst_rtspsrc_do_seek (GstRTSPSrc * src, GstSegment * segment)
{ {
gboolean res; gboolean res;
/* PLAY will add the range header now. */
src->state = GST_RTSP_STATE_SEEKING; src->state = GST_RTSP_STATE_SEEKING;
/* PLAY will add the range header now. */
src->need_range = TRUE; src->need_range = TRUE;
res = gst_rtspsrc_play (src); res = gst_rtspsrc_play (src, segment);
return res; return res;
} }
@ -1203,7 +1212,6 @@ gst_rtspsrc_perform_seek (GstRTSPSrc * src, GstEvent * event)
gboolean flush; gboolean flush;
gboolean update; gboolean update;
GstSegment seeksegment = { 0, }; GstSegment seeksegment = { 0, };
gint64 last_stop;
if (event) { if (event) {
GST_DEBUG_OBJECT (src, "doing seek with event"); GST_DEBUG_OBJECT (src, "doing seek with event");
@ -1250,10 +1258,7 @@ gst_rtspsrc_perform_seek (GstRTSPSrc * src, GstEvent * event)
/* stop flushing state */ /* stop flushing state */
gst_rtspsrc_loop_send_cmd (src, CMD_WAIT, FALSE); gst_rtspsrc_loop_send_cmd (src, CMD_WAIT, FALSE);
/* save current position */ GST_DEBUG_OBJECT (src, "stopped streaming");
last_stop = src->segment.last_stop;
GST_DEBUG_OBJECT (src, "stopped streaming at %" G_GINT64_FORMAT, last_stop);
/* copy segment, we need this because we still need the old /* copy segment, we need this because we still need the old
* segment when we close the current segment. */ * segment when we close the current segment. */
@ -1318,11 +1323,9 @@ gst_rtspsrc_perform_seek (GstRTSPSrc * src, GstEvent * event)
src->segment.format, src->segment.last_stop, stop, src->segment.format, src->segment.last_stop, stop,
src->segment.last_stop); src->segment.last_stop);
/* mark discont if we are going to stream from another position. */ /* mark discont */
if (last_stop != src->segment.last_stop) {
GST_DEBUG_OBJECT (src, "mark DISCONT, we did a seek to another position"); GST_DEBUG_OBJECT (src, "mark DISCONT, we did a seek to another position");
//src->discont = TRUE;
}
GST_RTSP_STREAM_UNLOCK (src); GST_RTSP_STREAM_UNLOCK (src);
return TRUE; return TRUE;
@ -2223,7 +2226,7 @@ gst_rtspsrc_activate_streams (GstRTSPSrc * src)
} }
static void static void
gst_rtspsrc_configure_caps (GstRTSPSrc * src) gst_rtspsrc_configure_caps (GstRTSPSrc * src, GstSegment * segment)
{ {
GList *walk; GList *walk;
guint64 start, stop; guint64 start, stop;
@ -2231,10 +2234,10 @@ gst_rtspsrc_configure_caps (GstRTSPSrc * src)
GST_DEBUG_OBJECT (src, "configuring stream caps"); GST_DEBUG_OBJECT (src, "configuring stream caps");
start = src->segment.last_stop; start = segment->last_stop;
stop = src->segment.duration; stop = segment->duration;
play_speed = src->segment.rate; play_speed = segment->rate;
play_scale = src->segment.applied_rate; play_scale = segment->applied_rate;
for (walk = src->streams; walk; walk = g_list_next (walk)) { for (walk = src->streams; walk; walk = g_list_next (walk)) {
GstRTSPStream *stream = (GstRTSPStream *) walk->data; GstRTSPStream *stream = (GstRTSPStream *) walk->data;
@ -2260,10 +2263,13 @@ gst_rtspsrc_configure_caps (GstRTSPSrc * src)
stream->caps = caps; stream->caps = caps;
} }
} }
GST_DEBUG_OBJECT (src, "stream %p, caps %" GST_PTR_FORMAT, stream, caps);
} }
if (src->session) if (src->session) {
GST_DEBUG_OBJECT (src, "clear session");
g_signal_emit_by_name (src->session, "clear-pt-map", NULL); g_signal_emit_by_name (src->session, "clear-pt-map", NULL);
} }
}
static GstFlowReturn static GstFlowReturn
gst_rtspsrc_combine_flows (GstRTSPSrc * src, GstRTSPStream * stream, gst_rtspsrc_combine_flows (GstRTSPSrc * src, GstRTSPStream * stream,
@ -2787,7 +2793,7 @@ gst_rtspsrc_loop_udp (GstRTSPSrc * src)
goto open_failed; goto open_failed;
/* start playback */ /* start playback */
if (!gst_rtspsrc_play (src)) if (!gst_rtspsrc_play (src, &src->segment))
goto play_failed; goto play_failed;
done: done:
@ -3722,7 +3728,8 @@ cleanup_error:
} }
static void static void
gst_rtspsrc_parse_range (GstRTSPSrc * src, const gchar * range) gst_rtspsrc_parse_range (GstRTSPSrc * src, const gchar * range,
GstSegment * segment)
{ {
GstRTSPTimeRange *therange; GstRTSPTimeRange *therange;
@ -3742,7 +3749,7 @@ gst_rtspsrc_parse_range (GstRTSPSrc * src, const gchar * range)
GST_DEBUG_OBJECT (src, "range: min %" GST_TIME_FORMAT, GST_DEBUG_OBJECT (src, "range: min %" GST_TIME_FORMAT,
GST_TIME_ARGS (seconds)); GST_TIME_ARGS (seconds));
gst_segment_set_last_stop (&src->segment, GST_FORMAT_TIME, seconds); gst_segment_set_last_stop (segment, GST_FORMAT_TIME, seconds);
if (therange->max.type == GST_RTSP_TIME_NOW) if (therange->max.type == GST_RTSP_TIME_NOW)
seconds = -1; seconds = -1;
@ -3757,7 +3764,7 @@ gst_rtspsrc_parse_range (GstRTSPSrc * src, const gchar * range)
/* don't change duration with unknown value, we might have a valid value /* don't change duration with unknown value, we might have a valid value
* there that we want to keep. */ * there that we want to keep. */
if (seconds != -1) if (seconds != -1)
gst_segment_set_duration (&src->segment, GST_FORMAT_TIME, seconds); gst_segment_set_duration (segment, GST_FORMAT_TIME, seconds);
} else { } else {
GST_WARNING_OBJECT (src, "could not parse range: '%s'", range); GST_WARNING_OBJECT (src, "could not parse range: '%s'", range);
} }
@ -3866,7 +3873,7 @@ gst_rtspsrc_open (GstRTSPSrc * src)
range = gst_sdp_message_get_attribute_val (&sdp, "range"); range = gst_sdp_message_get_attribute_val (&sdp, "range");
if (range) if (range)
gst_rtspsrc_parse_range (src, range); gst_rtspsrc_parse_range (src, range, &src->segment);
} }
/* create streams */ /* create streams */
@ -4163,7 +4170,7 @@ gst_rtspsrc_get_float (const char *str, gfloat * val)
} }
static gboolean static gboolean
gst_rtspsrc_play (GstRTSPSrc * src) gst_rtspsrc_play (GstRTSPSrc * src, GstSegment * segment)
{ {
GstRTSPMessage request = { 0 }; GstRTSPMessage request = { 0 };
GstRTSPMessage response = { 0 }; GstRTSPMessage response = { 0 };
@ -4188,26 +4195,26 @@ gst_rtspsrc_play (GstRTSPSrc * src)
goto create_request_failed; goto create_request_failed;
if (src->need_range) { if (src->need_range) {
if (src->segment.last_stop == 0) if (segment->last_stop == 0)
hval = g_strdup_printf ("npt=0-"); hval = g_strdup_printf ("npt=0-");
else else
hval = hval =
gst_rtspsrc_dup_printf ("npt=%f-", gst_rtspsrc_dup_printf ("npt=%f-",
((gdouble) src->segment.last_stop) / GST_SECOND); ((gdouble) segment->last_stop) / GST_SECOND);
gst_rtsp_message_add_header (&request, GST_RTSP_HDR_RANGE, hval); gst_rtsp_message_add_header (&request, GST_RTSP_HDR_RANGE, hval);
g_free (hval); g_free (hval);
src->need_range = FALSE; src->need_range = FALSE;
} }
if (src->segment.rate != 1.0) { if (segment->rate != 1.0) {
hval = gst_rtspsrc_dup_printf ("%f", src->segment.rate); hval = gst_rtspsrc_dup_printf ("%f", segment->rate);
gst_rtsp_message_add_header (&request, GST_RTSP_HDR_SPEED, hval); gst_rtsp_message_add_header (&request, GST_RTSP_HDR_SPEED, hval);
g_free (hval); g_free (hval);
} }
if (src->segment.applied_rate != 1.0) { if (segment->applied_rate != 1.0) {
hval = gst_rtspsrc_dup_printf ("%f", src->segment.applied_rate); hval = gst_rtspsrc_dup_printf ("%f", segment->applied_rate);
gst_rtsp_message_add_header (&request, GST_RTSP_HDR_SCALE, hval); gst_rtsp_message_add_header (&request, GST_RTSP_HDR_SCALE, hval);
g_free (hval); g_free (hval);
} }
@ -4221,7 +4228,7 @@ gst_rtspsrc_play (GstRTSPSrc * src)
* Play Time) and should be put in the NEWSEGMENT position field. */ * Play Time) and should be put in the NEWSEGMENT position field. */
if (gst_rtsp_message_get_header (&response, GST_RTSP_HDR_RANGE, &hval, if (gst_rtsp_message_get_header (&response, GST_RTSP_HDR_RANGE, &hval,
0) == GST_RTSP_OK) 0) == GST_RTSP_OK)
gst_rtspsrc_parse_range (src, hval); gst_rtspsrc_parse_range (src, hval, segment);
/* parse Speed header. This is the intended playback rate of the stream /* parse Speed header. This is the intended playback rate of the stream
* and should be put in the NEWSEGMENT rate field. */ * and should be put in the NEWSEGMENT rate field. */
@ -4230,9 +4237,9 @@ gst_rtspsrc_play (GstRTSPSrc * src)
gfloat fval; gfloat fval;
if (gst_rtspsrc_get_float (hval, &fval) > 0) if (gst_rtspsrc_get_float (hval, &fval) > 0)
src->segment.rate = fval; segment->rate = fval;
} else { } else {
src->segment.rate = 1.0; segment->rate = 1.0;
} }
/* parse Scale header. This is the playback rate as sent by the server /* parse Scale header. This is the playback rate as sent by the server
@ -4242,9 +4249,9 @@ gst_rtspsrc_play (GstRTSPSrc * src)
gfloat fval; gfloat fval;
if (gst_rtspsrc_get_float (hval, &fval) > 0) if (gst_rtspsrc_get_float (hval, &fval) > 0)
src->segment.applied_rate = fval; segment->applied_rate = fval;
} else { } else {
src->segment.applied_rate = 1.0; segment->applied_rate = 1.0;
} }
/* parse the RTP-Info header field (if ANY) to get the base seqnum and timestamp /* parse the RTP-Info header field (if ANY) to get the base seqnum and timestamp
@ -4257,7 +4264,7 @@ gst_rtspsrc_play (GstRTSPSrc * src)
gst_rtsp_message_unset (&response); gst_rtsp_message_unset (&response);
/* configure the caps of the streams after we parsed all headers. */ /* configure the caps of the streams after we parsed all headers. */
gst_rtspsrc_configure_caps (src); gst_rtspsrc_configure_caps (src, segment);
/* for interleaved transport, we receive the data on the RTSP connection /* for interleaved transport, we receive the data on the RTSP connection
* instead of UDP. We start a task to select and read from that connection. * instead of UDP. We start a task to select and read from that connection.
@ -4480,7 +4487,7 @@ gst_rtspsrc_change_state (GstElement * element, GstStateChange transition)
gst_rtsp_connection_flush (rtspsrc->connection, FALSE); gst_rtsp_connection_flush (rtspsrc->connection, FALSE);
/* FIXME, the server might send UDP packets before we activate the UDP /* FIXME, the server might send UDP packets before we activate the UDP
* ports */ * ports */
gst_rtspsrc_play (rtspsrc); gst_rtspsrc_play (rtspsrc, &rtspsrc->segment);
break; break;
case GST_STATE_CHANGE_PLAYING_TO_PAUSED: case GST_STATE_CHANGE_PLAYING_TO_PAUSED:
case GST_STATE_CHANGE_PAUSED_TO_READY: case GST_STATE_CHANGE_PAUSED_TO_READY: