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rtspsrc: send keep alive when paused

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When we are paused, send keep alive messages to the server so that our session
doesn't time out when we go back to playing later.
This commit is contained in:
Wim Taymans 2010-02-26 17:13:49 +01:00
parent 8599aff951
commit 38f2b4735d
1 changed files with 34 additions and 42 deletions
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76
gst/rtsp/gstrtspsrc.c
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@ -241,7 +241,7 @@ static GstRTSPResult gst_rtspsrc_send_cb (GstRTSPExtension * ext,
static gboolean gst_rtspsrc_open (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, gboolean idle);
static gboolean gst_rtspsrc_close (GstRTSPSrc * src);
static gboolean gst_rtspsrc_uri_set_uri (GstURIHandler * handler,
@ -1590,7 +1590,7 @@ gst_rtspsrc_perform_seek (GstRTSPSrc * src, GstEvent * event)
/* if we were playing, pause first */
if (playing)
gst_rtspsrc_pause (src);
gst_rtspsrc_pause (src, FALSE);
gst_rtspsrc_do_seek (src, &seeksegment);
@ -2947,30 +2947,12 @@ gst_rtspsrc_loop_interleaved (GstRTSPSrc * src)
/* get the next timeout interval */
gst_rtsp_connection_next_timeout (src->connection, &tv_timeout);
/* see if the timeout period expired */
if ((tv_timeout.tv_sec | tv_timeout.tv_usec) == 0) {
GST_DEBUG_OBJECT (src, "timout, sending keep-alive");
/* send keep-alive, ignore the result, a warning will be posted. */
gst_rtspsrc_send_keep_alive (src);
}
GST_DEBUG_OBJECT (src, "doing receive with timeout %d seconds",
(gint) tv_timeout.tv_sec);
GST_DEBUG_OBJECT (src, "doing receive");
/* We need to check if playback has been paused while we have been
* doing something else in our own GstTask (e.g. pushing buffer). There
* is a slight chance that we have just received data buffer when PAUSE
* state change happens (in another thread). In this case we well be
* totally ignorant of that unless we explicitly check it here. */
GST_RTSP_STATE_LOCK (src);
if (src->state == GST_RTSP_STATE_READY) {
/* We are looping in a paused mode */
GST_RTSP_STATE_UNLOCK (src);
goto already_paused;
}
/* protect the connection with the connection lock so that we can see when
* we are finished doing server communication */
res = gst_rtspsrc_connection_receive (src, &message, src->ptcp_timeout);
GST_RTSP_STATE_UNLOCK (src);
switch (res) {
case GST_RTSP_OK:
@ -2980,8 +2962,10 @@ gst_rtspsrc_loop_interleaved (GstRTSPSrc * src)
/* we got interrupted this means we need to stop */
goto interrupt;
case GST_RTSP_ETIMEOUT:
/* no reply, go EOS */
goto timeout;
/* no reply, send keep alive */
GST_DEBUG_OBJECT (src, "timeout, sending keep-alive");
gst_rtspsrc_send_keep_alive (src);
continue;
case GST_RTSP_EEOF:
/* go EOS when the server closed the connection */
goto server_eof;
@ -3131,14 +3115,6 @@ unknown_stream:
gst_rtsp_message_unset (&message);
return GST_FLOW_OK;
}
timeout:
{
GST_DEBUG_OBJECT (src, "we got a timeout");
GST_ELEMENT_WARNING (src, RESOURCE, READ, (NULL),
("Timeout while waiting for server message."));
gst_rtsp_message_unset (&message);
return GST_FLOW_UNEXPECTED;
}
server_eof:
{
GST_DEBUG_OBJECT (src, "we got an eof from the server");
@ -3156,11 +3132,6 @@ interrupt:
gst_rtsp_connection_flush (src->connection, FALSE);
return GST_FLOW_WRONG_STATE;
}
already_paused:
{
GST_DEBUG_OBJECT (src, "got interrupted: playback already paused");
return GST_FLOW_WRONG_STATE;
}
receive_error:
{
gchar *str = gst_rtsp_strresult (res);
@ -3305,7 +3276,7 @@ gst_rtspsrc_loop_udp (GstRTSPSrc * src)
src->cur_protocols = GST_RTSP_LOWER_TRANS_TCP;
/* pause to prepare for a restart */
gst_rtspsrc_pause (src);
gst_rtspsrc_pause (src, FALSE);
if (src->task) {
/* stop task, we cannot join as this would deadlock, the task will stop when
@ -5176,6 +5147,19 @@ gst_rtspsrc_play (GstRTSPSrc * src, GstSegment * segment)
if (src->state == GST_RTSP_STATE_PLAYING)
goto was_playing;
if (!src->connection || !src->connected)
goto done;
/* waiting for connection idle, we were flushing so any attempt at doing data
* transfer will result in pausing the tasks. */
GST_DEBUG_OBJECT (src, "wait for connection idle");
GST_RTSP_CONN_LOCK (src);
GST_DEBUG_OBJECT (src, "connection is idle now");
GST_RTSP_CONN_UNLOCK (src);
GST_DEBUG_OBJECT (src, "stop connection flush");
gst_rtsp_connection_flush (src->connection, FALSE);
/* do play */
res =
gst_rtsp_message_init_request (&request, GST_RTSP_PLAY,
@ -5286,7 +5270,7 @@ send_error:
}
static gboolean
gst_rtspsrc_pause (GstRTSPSrc * src)
gst_rtspsrc_pause (GstRTSPSrc * src, gboolean idle)
{
GstRTSPMessage request = { 0 };
GstRTSPMessage response = { 0 };
@ -5325,6 +5309,13 @@ gst_rtspsrc_pause (GstRTSPSrc * src)
gst_rtsp_message_unset (&request);
gst_rtsp_message_unset (&response);
if (idle && src->task) {
GST_DEBUG_OBJECT (src, "starting idle task again");
src->base_time = -1;
gst_rtspsrc_loop_send_cmd (src, CMD_WAIT, FALSE);
gst_task_start (src->task);
}
no_connection:
src->state = GST_RTSP_STATE_READY;
@ -5461,14 +5452,14 @@ gst_rtspsrc_change_state (GstElement * element, GstStateChange transition)
break;
case GST_STATE_CHANGE_PAUSED_TO_PLAYING:
GST_DEBUG_OBJECT (rtspsrc, "PAUSED->PLAYING: stop connection flush");
gst_rtsp_connection_flush (rtspsrc->connection, FALSE);
gst_rtspsrc_loop_send_cmd (rtspsrc, CMD_STOP, TRUE);
/* send some dummy packets before we chain up to the parent to activate
* the receive in the udp sources */
gst_rtspsrc_send_dummy_packets (rtspsrc);
break;
case GST_STATE_CHANGE_PLAYING_TO_PAUSED:
case GST_STATE_CHANGE_PAUSED_TO_READY:
GST_DEBUG_OBJECT (rtspsrc, "shutdown: sending stop command");
GST_DEBUG_OBJECT (rtspsrc, "state change: sending stop command");
gst_rtspsrc_loop_send_cmd (rtspsrc, CMD_STOP, TRUE);
break;
default:
@ -5485,7 +5476,8 @@ gst_rtspsrc_change_state (GstElement * element, GstStateChange transition)
gst_rtspsrc_play (rtspsrc, &rtspsrc->segment);
break;
case GST_STATE_CHANGE_PLAYING_TO_PAUSED:
gst_rtspsrc_pause (rtspsrc);
/* send pause request and keep the idle task around */
gst_rtspsrc_pause (rtspsrc, TRUE);
ret = GST_STATE_CHANGE_NO_PREROLL;
break;
case GST_STATE_CHANGE_READY_TO_PAUSED: