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gst/rtsp/gstrtspsrc.c: More seeking fixes, mostly passing around the new playback segment in order to configure it pr...
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
parent
f8df008747
commit
92e16a65ae
2 changed files with 57 additions and 38 deletions
12
ChangeLog
12
ChangeLog
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@ -1,3 +1,15 @@
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2007-10-08 Wim Taymans <wim.taymans@gmail.com>
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* gst/rtsp/gstrtspsrc.c: (gst_rtspsrc_flush),
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(gst_rtspsrc_do_seek), (gst_rtspsrc_perform_seek),
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(gst_rtspsrc_configure_caps), (gst_rtspsrc_loop_udp),
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(gst_rtspsrc_parse_range), (gst_rtspsrc_open), (gst_rtspsrc_play),
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(gst_rtspsrc_change_state):
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More seeking fixes, mostly passing around the new playback segment in
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order to configure it properly.
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Also reset base_time of udp sources when setting them back to PLAYING as
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a temporary hack until core supports seek in live sources properly.
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2007-10-08 Wim Taymans <wim.taymans@gmail.com>
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* gst/rtp/gstrtpmp4adepay.c:
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@ -201,7 +201,7 @@ static GstRTSPResult gst_rtspsrc_send_cb (GstRTSPExtension * ext,
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GstRTSPMessage * request, GstRTSPMessage * response, GstRTSPSrc * src);
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static gboolean gst_rtspsrc_open (GstRTSPSrc * src);
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static gboolean gst_rtspsrc_play (GstRTSPSrc * src);
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static gboolean gst_rtspsrc_play (GstRTSPSrc * src, GstSegment * segment);
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static gboolean gst_rtspsrc_pause (GstRTSPSrc * src);
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static gboolean gst_rtspsrc_close (GstRTSPSrc * src);
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@ -1124,6 +1124,8 @@ gst_rtspsrc_flush (GstRTSPSrc * src, gboolean flush)
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gint cmd, i;
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GstState state;
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GList *walk;
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GstClock *clock;
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GstClockTime base_time = -1;
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if (flush) {
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event = gst_event_new_flush_start ();
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@ -1135,6 +1137,11 @@ gst_rtspsrc_flush (GstRTSPSrc * src, gboolean flush)
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GST_DEBUG_OBJECT (src, "stop flush");
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cmd = CMD_WAIT;
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state = GST_STATE_PLAYING;
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clock = gst_element_get_clock (GST_ELEMENT_CAST (src));
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if (clock) {
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base_time = gst_clock_get_time (clock);
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gst_object_unref (clock);
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}
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}
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gst_rtspsrc_push_event (src, event);
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gst_rtspsrc_loop_send_cmd (src, cmd, flush);
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@ -1145,6 +1152,8 @@ gst_rtspsrc_flush (GstRTSPSrc * src, gboolean flush)
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for (i = 0; i < 2; i++) {
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if (stream->udpsrc[i]) {
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if (base_time != -1)
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gst_element_set_base_time (stream->udpsrc[i], base_time);
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gst_element_set_state (stream->udpsrc[i], state);
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}
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}
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@ -1182,11 +1191,11 @@ gst_rtspsrc_do_seek (GstRTSPSrc * src, GstSegment * segment)
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{
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gboolean res;
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/* PLAY will add the range header now. */
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src->state = GST_RTSP_STATE_SEEKING;
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/* PLAY will add the range header now. */
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src->need_range = TRUE;
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res = gst_rtspsrc_play (src);
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res = gst_rtspsrc_play (src, segment);
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return res;
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}
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@ -1203,7 +1212,6 @@ gst_rtspsrc_perform_seek (GstRTSPSrc * src, GstEvent * event)
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gboolean flush;
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gboolean update;
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GstSegment seeksegment = { 0, };
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gint64 last_stop;
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if (event) {
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GST_DEBUG_OBJECT (src, "doing seek with event");
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@ -1250,10 +1258,7 @@ gst_rtspsrc_perform_seek (GstRTSPSrc * src, GstEvent * event)
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/* stop flushing state */
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gst_rtspsrc_loop_send_cmd (src, CMD_WAIT, FALSE);
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/* save current position */
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last_stop = src->segment.last_stop;
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GST_DEBUG_OBJECT (src, "stopped streaming at %" G_GINT64_FORMAT, last_stop);
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GST_DEBUG_OBJECT (src, "stopped streaming");
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/* copy segment, we need this because we still need the old
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* segment when we close the current segment. */
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@ -1318,11 +1323,9 @@ gst_rtspsrc_perform_seek (GstRTSPSrc * src, GstEvent * event)
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src->segment.format, src->segment.last_stop, stop,
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src->segment.last_stop);
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/* mark discont if we are going to stream from another position. */
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if (last_stop != src->segment.last_stop) {
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GST_DEBUG_OBJECT (src, "mark DISCONT, we did a seek to another position");
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//src->discont = TRUE;
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}
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/* mark discont */
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GST_DEBUG_OBJECT (src, "mark DISCONT, we did a seek to another position");
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GST_RTSP_STREAM_UNLOCK (src);
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return TRUE;
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@ -2223,7 +2226,7 @@ gst_rtspsrc_activate_streams (GstRTSPSrc * src)
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}
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static void
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gst_rtspsrc_configure_caps (GstRTSPSrc * src)
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gst_rtspsrc_configure_caps (GstRTSPSrc * src, GstSegment * segment)
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{
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GList *walk;
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guint64 start, stop;
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@ -2231,10 +2234,10 @@ gst_rtspsrc_configure_caps (GstRTSPSrc * src)
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GST_DEBUG_OBJECT (src, "configuring stream caps");
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start = src->segment.last_stop;
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stop = src->segment.duration;
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play_speed = src->segment.rate;
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play_scale = src->segment.applied_rate;
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start = segment->last_stop;
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stop = segment->duration;
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play_speed = segment->rate;
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play_scale = segment->applied_rate;
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for (walk = src->streams; walk; walk = g_list_next (walk)) {
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GstRTSPStream *stream = (GstRTSPStream *) walk->data;
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@ -2260,9 +2263,12 @@ gst_rtspsrc_configure_caps (GstRTSPSrc * src)
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stream->caps = caps;
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}
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}
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GST_DEBUG_OBJECT (src, "stream %p, caps %" GST_PTR_FORMAT, stream, caps);
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}
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if (src->session)
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if (src->session) {
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GST_DEBUG_OBJECT (src, "clear session");
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g_signal_emit_by_name (src->session, "clear-pt-map", NULL);
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}
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}
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static GstFlowReturn
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@ -2787,7 +2793,7 @@ gst_rtspsrc_loop_udp (GstRTSPSrc * src)
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goto open_failed;
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/* start playback */
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if (!gst_rtspsrc_play (src))
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if (!gst_rtspsrc_play (src, &src->segment))
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goto play_failed;
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done:
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@ -3722,7 +3728,8 @@ cleanup_error:
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}
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static void
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gst_rtspsrc_parse_range (GstRTSPSrc * src, const gchar * range)
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gst_rtspsrc_parse_range (GstRTSPSrc * src, const gchar * range,
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GstSegment * segment)
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{
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GstRTSPTimeRange *therange;
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@ -3742,7 +3749,7 @@ gst_rtspsrc_parse_range (GstRTSPSrc * src, const gchar * range)
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GST_DEBUG_OBJECT (src, "range: min %" GST_TIME_FORMAT,
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GST_TIME_ARGS (seconds));
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gst_segment_set_last_stop (&src->segment, GST_FORMAT_TIME, seconds);
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gst_segment_set_last_stop (segment, GST_FORMAT_TIME, seconds);
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if (therange->max.type == GST_RTSP_TIME_NOW)
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seconds = -1;
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@ -3757,7 +3764,7 @@ gst_rtspsrc_parse_range (GstRTSPSrc * src, const gchar * range)
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/* don't change duration with unknown value, we might have a valid value
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* there that we want to keep. */
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if (seconds != -1)
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gst_segment_set_duration (&src->segment, GST_FORMAT_TIME, seconds);
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gst_segment_set_duration (segment, GST_FORMAT_TIME, seconds);
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} else {
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GST_WARNING_OBJECT (src, "could not parse range: '%s'", range);
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}
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@ -3866,7 +3873,7 @@ gst_rtspsrc_open (GstRTSPSrc * src)
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range = gst_sdp_message_get_attribute_val (&sdp, "range");
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if (range)
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gst_rtspsrc_parse_range (src, range);
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gst_rtspsrc_parse_range (src, range, &src->segment);
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}
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/* create streams */
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@ -4163,7 +4170,7 @@ gst_rtspsrc_get_float (const char *str, gfloat * val)
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}
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static gboolean
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gst_rtspsrc_play (GstRTSPSrc * src)
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gst_rtspsrc_play (GstRTSPSrc * src, GstSegment * segment)
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{
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GstRTSPMessage request = { 0 };
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GstRTSPMessage response = { 0 };
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@ -4188,26 +4195,26 @@ gst_rtspsrc_play (GstRTSPSrc * src)
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goto create_request_failed;
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if (src->need_range) {
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if (src->segment.last_stop == 0)
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if (segment->last_stop == 0)
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hval = g_strdup_printf ("npt=0-");
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else
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hval =
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gst_rtspsrc_dup_printf ("npt=%f-",
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((gdouble) src->segment.last_stop) / GST_SECOND);
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((gdouble) segment->last_stop) / GST_SECOND);
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gst_rtsp_message_add_header (&request, GST_RTSP_HDR_RANGE, hval);
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g_free (hval);
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src->need_range = FALSE;
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}
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if (src->segment.rate != 1.0) {
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hval = gst_rtspsrc_dup_printf ("%f", src->segment.rate);
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if (segment->rate != 1.0) {
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hval = gst_rtspsrc_dup_printf ("%f", segment->rate);
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gst_rtsp_message_add_header (&request, GST_RTSP_HDR_SPEED, hval);
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g_free (hval);
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}
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if (src->segment.applied_rate != 1.0) {
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hval = gst_rtspsrc_dup_printf ("%f", src->segment.applied_rate);
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if (segment->applied_rate != 1.0) {
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hval = gst_rtspsrc_dup_printf ("%f", segment->applied_rate);
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gst_rtsp_message_add_header (&request, GST_RTSP_HDR_SCALE, hval);
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g_free (hval);
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}
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@ -4221,7 +4228,7 @@ gst_rtspsrc_play (GstRTSPSrc * src)
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* Play Time) and should be put in the NEWSEGMENT position field. */
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if (gst_rtsp_message_get_header (&response, GST_RTSP_HDR_RANGE, &hval,
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0) == GST_RTSP_OK)
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gst_rtspsrc_parse_range (src, hval);
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gst_rtspsrc_parse_range (src, hval, segment);
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/* parse Speed header. This is the intended playback rate of the stream
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* and should be put in the NEWSEGMENT rate field. */
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@ -4230,9 +4237,9 @@ gst_rtspsrc_play (GstRTSPSrc * src)
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gfloat fval;
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if (gst_rtspsrc_get_float (hval, &fval) > 0)
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src->segment.rate = fval;
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segment->rate = fval;
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} else {
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src->segment.rate = 1.0;
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segment->rate = 1.0;
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}
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/* parse Scale header. This is the playback rate as sent by the server
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@ -4242,9 +4249,9 @@ gst_rtspsrc_play (GstRTSPSrc * src)
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gfloat fval;
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if (gst_rtspsrc_get_float (hval, &fval) > 0)
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src->segment.applied_rate = fval;
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segment->applied_rate = fval;
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} else {
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src->segment.applied_rate = 1.0;
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segment->applied_rate = 1.0;
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}
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/* parse the RTP-Info header field (if ANY) to get the base seqnum and timestamp
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@ -4257,7 +4264,7 @@ gst_rtspsrc_play (GstRTSPSrc * src)
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gst_rtsp_message_unset (&response);
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/* configure the caps of the streams after we parsed all headers. */
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gst_rtspsrc_configure_caps (src);
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gst_rtspsrc_configure_caps (src, segment);
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/* for interleaved transport, we receive the data on the RTSP connection
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* instead of UDP. We start a task to select and read from that connection.
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@ -4480,7 +4487,7 @@ gst_rtspsrc_change_state (GstElement * element, GstStateChange transition)
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gst_rtsp_connection_flush (rtspsrc->connection, FALSE);
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/* FIXME, the server might send UDP packets before we activate the UDP
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* ports */
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gst_rtspsrc_play (rtspsrc);
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gst_rtspsrc_play (rtspsrc, &rtspsrc->segment);
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break;
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case GST_STATE_CHANGE_PLAYING_TO_PAUSED:
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case GST_STATE_CHANGE_PAUSED_TO_READY:
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