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rtspsrc: manage element state ourselves

Browse source 
Lock the state of the all our elements and manage their states
outselves. Because we are working async, we can't rely on the state
change function to set the state at the right time or to return the
right return value from the state change function.

Fixes https://bugzilla.gnome.org/show_bug.cgi?id=702046
This commit is contained in:
Wim Taymans 2013-06-16 05:40:13 +02:00
parent e89a48616b
commit d9bc48edc9
1 changed files with 44 additions and 27 deletions
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71
gst/rtsp/gstrtspsrc.c
View file

@ -1789,6 +1789,8 @@ again:
* server told us to really use the UDP ports. */
stream->udpsrc[0] = gst_object_ref_sink (udpsrc0);
stream->udpsrc[1] = gst_object_ref_sink (udpsrc1);
gst_element_set_locked_state (stream->udpsrc[0], TRUE);
gst_element_set_locked_state (stream->udpsrc[1], TRUE);
/* keep track of next available port number when we have a range
* configured */
@ -1834,13 +1836,31 @@ cleanup:
}
}
static void
gst_rtspsrc_set_state (GstRTSPSrc * src, GstState state)
{
GList *walk;
if (src->manager)
gst_element_set_state (GST_ELEMENT_CAST (src->manager), state);
for (walk = src->streams; walk; walk = g_list_next (walk)) {
GstRTSPStream *stream = (GstRTSPStream *) walk->data;
gint i;
for (i = 0; i < 2; i++) {
if (stream->udpsrc[i])
gst_element_set_state (stream->udpsrc[i], state);
}
}
}
static void
gst_rtspsrc_flush (GstRTSPSrc * src, gboolean flush, gboolean playing)
{
GstEvent *event;
gint cmd, i;
gint cmd;
GstState state;
GList *walk;
if (flush) {
event = gst_event_new_flush_start ();
@ -1858,22 +1878,7 @@ gst_rtspsrc_flush (GstRTSPSrc * src, gboolean flush, gboolean playing)
}
gst_rtspsrc_push_event (src, event);
gst_rtspsrc_loop_send_cmd (src, cmd, CMD_LOOP);
/* to manage jitterbuffer buffer mode */
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)) {
GstRTSPStream *stream = (GstRTSPStream *) walk->data;
for (i = 0; i < 2; i++) {
/* for udp case */
if (stream->udpsrc[i]) {
gst_element_set_state (stream->udpsrc[i], state);
}
}
}
gst_rtspsrc_set_state (src, state);
}
static GstRTSPResult
@ -2541,7 +2546,6 @@ gst_rtspsrc_stream_configure_manager (GstRTSPSrc * src, GstRTSPStream * stream,
/* configure the manager */
if (src->manager == NULL) {
GObjectClass *klass;
GstState target;
if (!(src->manager = gst_element_factory_make (manager, "manager"))) {
/* fallback */
@ -2556,13 +2560,10 @@ gst_rtspsrc_stream_configure_manager (GstRTSPSrc * src, GstRTSPStream * stream,
}
/* we manage this element */
gst_element_set_locked_state (src->manager, TRUE);
gst_bin_add (GST_BIN_CAST (src), src->manager);
GST_OBJECT_LOCK (src);
target = GST_STATE_TARGET (src);
GST_OBJECT_UNLOCK (src);
ret = gst_element_set_state (src->manager, target);
ret = gst_element_set_state (src->manager, GST_STATE_PAUSED);
if (ret == GST_STATE_CHANGE_FAILURE)
goto start_manager_failure;
@ -2906,6 +2907,7 @@ gst_rtspsrc_stream_configure_mcast (GstRTSPSrc * src, GstRTSPStream * stream,
src->multi_iface, NULL);
/* change state */
gst_element_set_locked_state (stream->udpsrc[0], TRUE);
gst_element_set_state (stream->udpsrc[0], GST_STATE_PAUSED);
}
@ -2961,6 +2963,7 @@ gst_rtspsrc_stream_configure_udp (GstRTSPSrc * src, GstRTSPStream * stream,
/* we manage the UDP elements now. For unicast, the UDP sources where
* allocated in the stream when we suggested a transport. */
if (stream->udpsrc[0]) {
gst_element_set_locked_state (stream->udpsrc[0], TRUE);
gst_bin_add (GST_BIN_CAST (src), stream->udpsrc[0]);
GST_DEBUG_OBJECT (src, "setting up UDP source");
@ -3001,6 +3004,7 @@ gst_rtspsrc_stream_configure_udp (GstRTSPSrc * src, GstRTSPStream * stream,
if (stream->udpsrc[1]) {
GstCaps *caps;
gst_element_set_locked_state (stream->udpsrc[1], TRUE);
gst_bin_add (GST_BIN_CAST (src), stream->udpsrc[1]);
caps = gst_caps_new_empty_simple ("application/x-rtcp");
@ -6085,6 +6089,8 @@ gst_rtspsrc_close (GstRTSPSrc * src, gboolean async, gboolean only_close)
GST_DEBUG_OBJECT (src, "TEARDOWN...");
gst_rtspsrc_set_state (src, GST_STATE_READY);
if (src->state < GST_RTSP_STATE_READY) {
GST_DEBUG_OBJECT (src, "not ready, doing cleanup");
goto close;
@ -6404,6 +6410,8 @@ gst_rtspsrc_play (GstRTSPSrc * src, GstSegment * segment, gboolean async)
* udp sources */
gst_rtspsrc_send_dummy_packets (src);
gst_rtspsrc_set_state (src, GST_STATE_PLAYING);
/* construct a control url */
if (src->control)
control = src->control;
@ -6660,6 +6668,9 @@ gst_rtspsrc_pause (GstRTSPSrc * src, gboolean async)
break;
}
/* change element states now */
gst_rtspsrc_set_state (src, GST_STATE_PAUSED);
no_connection:
src->state = GST_RTSP_STATE_READY;
@ -6946,22 +6957,28 @@ gst_rtspsrc_change_state (GstElement * element, GstStateChange transition)
goto done;
switch (transition) {
case GST_STATE_CHANGE_NULL_TO_READY:
ret = GST_STATE_CHANGE_SUCCESS;
break;
case GST_STATE_CHANGE_READY_TO_PAUSED:
ret = GST_STATE_CHANGE_NO_PREROLL;
break;
case GST_STATE_CHANGE_PAUSED_TO_PLAYING:
gst_rtspsrc_loop_send_cmd (rtspsrc, CMD_PLAY, 0);
ret = GST_STATE_CHANGE_SUCCESS;
break;
case GST_STATE_CHANGE_PLAYING_TO_PAUSED:
/* send pause request and keep the idle task around */
gst_rtspsrc_loop_send_cmd (rtspsrc, CMD_PAUSE, CMD_LOOP);
ret = GST_STATE_CHANGE_NO_PREROLL;
break;
case GST_STATE_CHANGE_READY_TO_PAUSED:
ret = GST_STATE_CHANGE_NO_PREROLL;
break;
case GST_STATE_CHANGE_PAUSED_TO_READY:
gst_rtspsrc_loop_send_cmd (rtspsrc, CMD_CLOSE, CMD_PAUSE);
ret = GST_STATE_CHANGE_SUCCESS;
break;
case GST_STATE_CHANGE_READY_TO_NULL:
gst_rtspsrc_stop (rtspsrc);
ret = GST_STATE_CHANGE_SUCCESS;
break;
default:
break;