mirrors/gstreamer
mirrors/gstreamer
1
1
Fork 0
mirror of https://gitlab.freedesktop.org/gstreamer/gstreamer.git synced 2024-07-05 14:16:06 +00:00
Code Issues 1 Projects Releases Wiki Activity

media: set multicast sink parameters

Browse source 
Disable loop and automatic multicast join on the udpsink elements.
Add some more debug info.
Reset some state variables in the right place.
Use the right port numbers for multicast.
This commit is contained in:
Wim Taymans 2010-03-05 13:28:58 +01:00
parent 63addbc278
commit 53f8350b36
1 changed files with 57 additions and 40 deletions
Show stats Download patch file Download diff file Expand all files Collapse all files

97
gst/rtsp-server/rtsp-media.c
View file

@ -107,8 +107,6 @@ static void
gst_rtsp_media_init (GstRTSPMedia * media)
{
media->streams = g_array_new (FALSE, TRUE, sizeof (GstRTSPMediaStream *));
media->is_live = FALSE;
media->buffering = FALSE;
}
static void
@ -248,7 +246,7 @@ collect_media_stats (GstRTSPMedia *media)
}
GST_INFO ("stats: position %"GST_TIME_FORMAT", duration %"GST_TIME_FORMAT,
GST_TIME_ARGS (position), GST_TIME_ARGS (duration));
GST_TIME_ARGS (position), GST_TIME_ARGS (duration));
if (position == -1) {
media->range.min.type = GST_RTSP_TIME_NOW;
@ -461,10 +459,10 @@ gst_rtsp_media_seek (GstRTSPMedia *media, GstRTSPTimeRange *range)
if (start != -1 || stop != -1) {
GST_INFO ("seeking to %"GST_TIME_FORMAT" - %"GST_TIME_FORMAT,
GST_TIME_ARGS (start), GST_TIME_ARGS (stop));
GST_TIME_ARGS (start), GST_TIME_ARGS (stop));
res = gst_element_seek (media->pipeline, 1.0, GST_FORMAT_TIME,
flags, start_type, start, stop_type, stop);
flags, start_type, start, stop_type, stop);
/* and block for the seek to complete */
GST_INFO ("done seeking %d", res);
@ -647,6 +645,11 @@ again:
g_object_set (G_OBJECT (udpsink1), "sync", FALSE, NULL);
g_object_set (G_OBJECT (udpsink1), "async", FALSE, NULL);
g_object_set (G_OBJECT (udpsink0), "auto-multicast", FALSE, NULL);
g_object_set (G_OBJECT (udpsink0), "loop", FALSE, NULL);
g_object_set (G_OBJECT (udpsink1), "auto-multicast", FALSE, NULL);
g_object_set (G_OBJECT (udpsink1), "loop", FALSE, NULL);
/* we keep these elements, we configure all in configure_transport when the
* server told us to really use the UDP ports. */
stream->udpsrc[0] = udpsrc0;
@ -786,8 +789,8 @@ on_new_ssrc (GObject *session, GObject *source, GstRTSPMediaStream *stream)
if ((trans = find_transport (stream, rtcp_from))) {
GST_INFO ("%p: found transport %p for source %p", stream, trans, source);
/* keep ref to the source */
trans->rtpsource = source;
/* keep ref to the source */
trans->rtpsource = source;
g_object_set_qdata (source, ssrc_stream_map_key, trans);
}
@ -927,7 +930,7 @@ setup_stream (GstRTSPMediaStream *stream, guint idx, GstRTSPMedia *media)
gst_bin_add (GST_BIN_CAST (media->pipeline), stream->appsink[i]);
gst_bin_add (GST_BIN_CAST (media->pipeline), stream->appsrc[i]);
gst_app_sink_set_callbacks (GST_APP_SINK_CAST (stream->appsink[i]),
&sink_cb, stream, NULL);
&sink_cb, stream, NULL);
}
/* hook up the stream to the RTP session elements. */
@ -949,7 +952,7 @@ setup_stream (GstRTSPMediaStream *stream, guint idx, GstRTSPMedia *media)
/* get the session */
g_signal_emit_by_name (media->rtpbin, "get-internal-session", idx,
&stream->session);
&stream->session);
g_signal_connect (stream->session, "on-new-ssrc", (GCallback) on_new_ssrc,
stream);
@ -1121,20 +1124,20 @@ default_handle_message (GstRTSPMedia *media, GstMessage *message)
GST_INFO ("Buffering done, setting pipeline to PLAYING");
gst_element_set_state (media->pipeline, GST_STATE_PLAYING);
}
else {
else {
GST_INFO ("Buffering done");
}
}
} else {
/* buffering busy */
if (media->buffering == FALSE) {
if (media->target_state == GST_STATE_PLAYING) {
if (media->target_state == GST_STATE_PLAYING) {
/* we were not buffering but PLAYING, PAUSE the pipeline. */
GST_INFO ("Buffering, setting pipeline to PAUSED ...");
gst_element_set_state (media->pipeline, GST_STATE_PAUSED);
}
else {
}
else {
GST_INFO ("Buffering ...");
}
}
}
media->buffering = TRUE;
}
@ -1273,6 +1276,10 @@ gst_rtsp_media_prepare (GstRTSPMedia *media)
GST_INFO ("preparing media %p", media);
/* reset some variables */
media->is_live = FALSE;
media->buffering = FALSE;
bus = gst_pipeline_get_bus (GST_PIPELINE_CAST (media->pipeline));
/* add the pipeline bus to our custom mainloop */
@ -1306,22 +1313,27 @@ gst_rtsp_media_prepare (GstRTSPMedia *media)
g_signal_connect (elem, "pad-added", (GCallback) pad_added_cb, media);
g_signal_connect (elem, "no-more-pads", (GCallback) no_more_pads_cb, media);
/* we add a fakesink here in order to make the state change async. We remove
* the fakesink again in the no-more-pads callback. */
media->fakesink = gst_element_factory_make ("fakesink", "fakesink");
gst_bin_add (GST_BIN (media->pipeline), media->fakesink);
}
GST_INFO ("setting pipeline to PAUSED for media %p", media);
/* first go to PAUSED */
ret = gst_element_set_state (media->pipeline, GST_STATE_PAUSED);
media->target_state = GST_STATE_PAUSED;
switch (ret) {
case GST_STATE_CHANGE_SUCCESS:
GST_INFO ("SUCCESS state change for media %p", media);
break;
case GST_STATE_CHANGE_ASYNC:
GST_INFO ("ASYNC state change for media %p", media);
break;
case GST_STATE_CHANGE_NO_PREROLL:
/* we need to go to PLAYING */
GST_INFO ("live media %p", media);
GST_INFO ("NO_PREROLL state change: live media %p", media);
media->is_live = TRUE;
ret = gst_element_set_state (media->pipeline, GST_STATE_PLAYING);
if (ret == GST_STATE_CHANGE_FAILURE)
@ -1448,7 +1460,7 @@ gst_rtsp_media_set_state (GstRTSPMedia *media, GstState state, GArray *transport
case GST_STATE_PAUSED:
/* we're going from PLAYING to PAUSED, READY or NULL, remove */
if (media->target_state == GST_STATE_PLAYING)
remove = TRUE;
remove = TRUE;
break;
case GST_STATE_PLAYING:
/* we're going to PLAYING, add */
@ -1479,42 +1491,47 @@ gst_rtsp_media_set_state (GstRTSPMedia *media, GstState state, GArray *transport
case GST_RTSP_LOWER_TRANS_UDP:
case GST_RTSP_LOWER_TRANS_UDP_MCAST:
{
gchar *dest;
gint min, max;
gchar *dest;
gint min, max;
dest = trans->destination;
min = trans->client_port.min;
max = trans->client_port.max;
dest = trans->destination;
if (trans->lower_transport == GST_RTSP_LOWER_TRANS_UDP_MCAST) {
min = trans->port.min;
max = trans->port.max;
} else {
min = trans->client_port.min;
max = trans->client_port.max;
}
if (add && !tr->active) {
if (add && !tr->active) {
GST_INFO ("adding %s:%d-%d", dest, min, max);
g_signal_emit_by_name (stream->udpsink[0], "add", dest, min, NULL);
g_signal_emit_by_name (stream->udpsink[1], "add", dest, max, NULL);
stream->transports = g_list_prepend (stream->transports, tr);
tr->active = TRUE;
media->active++;
} else if (remove && tr->active) {
stream->transports = g_list_prepend (stream->transports, tr);
tr->active = TRUE;
media->active++;
} else if (remove && tr->active) {
GST_INFO ("removing %s:%d-%d", dest, min, max);
g_signal_emit_by_name (stream->udpsink[0], "remove", dest, min, NULL);
g_signal_emit_by_name (stream->udpsink[1], "remove", dest, max, NULL);
stream->transports = g_list_remove (stream->transports, tr);
tr->active = FALSE;
media->active--;
}
stream->transports = g_list_remove (stream->transports, tr);
tr->active = FALSE;
media->active--;
}
break;
}
case GST_RTSP_LOWER_TRANS_TCP:
if (add && !tr->active) {
if (add && !tr->active) {
GST_INFO ("adding TCP %s", trans->destination);
stream->transports = g_list_prepend (stream->transports, tr);
tr->active = TRUE;
media->active++;
} else if (remove && tr->active) {
stream->transports = g_list_prepend (stream->transports, tr);
tr->active = TRUE;
media->active++;
} else if (remove && tr->active) {
GST_INFO ("removing TCP %s", trans->destination);
stream->transports = g_list_remove (stream->transports, tr);
tr->active = FALSE;
media->active--;
}
stream->transports = g_list_remove (stream->transports, tr);
tr->active = FALSE;
media->active--;
}
break;
default:
GST_INFO ("Unknown transport %d", trans->lower_transport);