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Add suport for RTP manager monitoring

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Add the first stage in monitoring the rtp manager.
Make sure we don't update the state to something we don't want.
This commit is contained in:
Wim Taymans 2009-02-13 19:54:18 +01:00
parent 308ad6f6d0
commit f0c047ef94
2 changed files with 63 additions and 1 deletions
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61
gst/rtsp-server/rtsp-media.c
View file

@ -77,6 +77,9 @@ gst_rtsp_media_init (GstRTSPMedia * media)
static void
gst_rtsp_media_stream_free (GstRTSPMediaStream *stream)
{
if (stream->session)
g_object_unref (stream->session);
if (stream->caps)
gst_caps_unref (stream->caps);
@ -421,6 +424,36 @@ caps_notify (GstPad * pad, GParamSpec * unused, GstRTSPMediaStream * stream)
g_free (capsstr);
}
static void
on_new_ssrc (GObject *session, GObject *source, GstRTSPMedia *media)
{
g_message ("%p: new source %p", media, source);
}
static void
on_ssrc_active (GObject *session, GObject *source, GstRTSPMedia *media)
{
g_message ("%p: source %p is active", media, source);
}
static void
on_bye_ssrc (GObject *session, GObject *source, GstRTSPMedia *media)
{
g_message ("%p: source %p bye", media, source);
}
static void
on_bye_timeout (GObject *session, GObject *source, GstRTSPMedia *media)
{
g_message ("%p: source %p bye timeout", media, source);
}
static void
on_timeout (GObject *session, GObject *source, GstRTSPMedia *media)
{
g_message ("%p: source %p timeout", media, source);
}
/* prepare the pipeline objects to handle @stream in @media */
static gboolean
setup_stream (GstRTSPMediaStream *stream, guint idx, GstRTSPMedia *media)
@ -449,6 +482,21 @@ setup_stream (GstRTSPMediaStream *stream, guint idx, GstRTSPMedia *media)
stream->recv_rtcp_sink = gst_element_get_request_pad (media->rtpbin, name);
g_free (name);
/* get the session */
g_signal_emit_by_name (media->rtpbin, "get-internal-session", idx,
&stream->session);
g_signal_connect (stream->session, "on-new-ssrc", (GCallback) on_new_ssrc,
media);
g_signal_connect (stream->session, "on-ssrc-active", (GCallback) on_ssrc_active,
media);
g_signal_connect (stream->session, "on-bye-ssrc", (GCallback) on_bye_ssrc,
media);
g_signal_connect (stream->session, "on-bye-timeout", (GCallback) on_bye_timeout,
media);
g_signal_connect (stream->session, "on-timeout", (GCallback) on_timeout,
media);
/* link the RTP pad to the session manager */
gst_pad_link (stream->srcpad, stream->send_rtp_sink);
@ -665,7 +713,10 @@ gst_rtsp_media_prepare (GstRTSPMedia *media)
ret = gst_element_set_state (media->pipeline, GST_STATE_PLAYING);
break;
case GST_STATE_CHANGE_FAILURE:
{
unlock_streams (media);
goto state_failed;
}
}
/* now wait for all pads to be prerolled */
@ -728,7 +779,6 @@ state_failed:
}
gst_message_unref (message);
}
unlock_streams (media);
gst_element_set_state (media->pipeline, GST_STATE_NULL);
gst_object_unref (bus);
return FALSE;
@ -754,6 +804,9 @@ gst_rtsp_media_play (GstRTSPMedia *media, GArray *transports)
g_return_val_if_fail (transports != NULL, FALSE);
g_return_val_if_fail (media->prepared, FALSE);
if (media->target_state == GST_STATE_PLAYING)
return TRUE;
for (i = 0; i < transports->len; i++) {
GstRTSPMediaTrans *tr;
GstRTSPMediaStream *stream;
@ -803,6 +856,9 @@ gst_rtsp_media_pause (GstRTSPMedia *media, GArray *transports)
g_return_val_if_fail (transports != NULL, FALSE);
g_return_val_if_fail (media->prepared, FALSE);
if (media->target_state == GST_STATE_PAUSED)
return TRUE;
for (i = 0; i < transports->len; i++) {
GstRTSPMediaTrans *tr;
GstRTSPMediaStream *stream;
@ -851,6 +907,9 @@ gst_rtsp_media_stop (GstRTSPMedia *media, GArray *transports)
g_return_val_if_fail (transports != NULL, FALSE);
g_return_val_if_fail (media->prepared, FALSE);
if (media->target_state == GST_STATE_NULL)
return TRUE;
gst_rtsp_media_pause (media, transports);
g_message ("stop");

3
gst/rtsp-server/rtsp-media.h
View file

@ -84,6 +84,9 @@ struct _GstRTSPMediaStream {
GstPad *send_rtp_src;
GstPad *send_rtcp_src;
/* the RTPSession object */
GObject *session;
/* sinks used for sending and receiving RTP and RTCP, they share
* sockets */
GstElement *udpsrc[2];