Add suport for RTP manager monitoring

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

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);
@ -665,8 +713,11 @@ 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 */
ret = gst_element_get_state (media->pipeline, NULL, NULL, -1);
@ -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");

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];