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media: keep track of active transports

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Keep track of which transport is active to avoid closing the connection too
soon.
Remove the destination transport also when going to NULL.
Print some stats about the SDES and other RTCP messages we receive from the
clients.
This commit is contained in:
Wim Taymans 2009-05-26 11:42:41 +02:00
parent b0beb1c8d1
commit 5955fc7d12
2 changed files with 42 additions and 5 deletions
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46
gst/rtsp-server/rtsp-media.c
View file

@ -690,16 +690,40 @@ caps_notify (GstPad * pad, GParamSpec * unused, GstRTSPMediaStream * stream)
g_free (capsstr);
}
static void
dump_structure (const GstStructure *s)
{
gchar *sstr;
sstr = gst_structure_to_string (s);
g_message ("structure: %s", sstr);
g_free (sstr);
}
static void
on_new_ssrc (GObject *session, GObject *source, GstRTSPMedia *media)
{
g_message ("%p: new source %p", media, source);
}
static void
on_ssrc_sdes (GObject *session, GObject *source, GstRTSPMedia *media)
{
GstStructure *sdes;
g_message ("%p: new SDES %p", media, source);
g_object_get (source, "sdes", &sdes, NULL);
dump_structure (sdes);
}
static void
on_ssrc_active (GObject *session, GObject *source, GstRTSPMedia *media)
{
GstStructure *stats;
g_message ("%p: source %p is active", media, source);
g_object_get (source, "stats", &stats, NULL);
dump_structure (stats);
}
static void
@ -813,6 +837,8 @@ setup_stream (GstRTSPMediaStream *stream, guint idx, GstRTSPMedia *media)
g_signal_connect (stream->session, "on-new-ssrc", (GCallback) on_new_ssrc,
media);
g_signal_connect (stream->session, "on-ssrc-sdes", (GCallback) on_ssrc_sdes,
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,
@ -1025,6 +1051,10 @@ default_handle_message (GstRTSPMedia *media, GstMessage *message)
g_free (debug);
break;
}
case GST_MESSAGE_ELEMENT:
{
break;
}
default:
g_message ("%p: got message type %s", media, gst_message_type_get_name (type));
break;
@ -1279,11 +1309,13 @@ gst_rtsp_media_set_state (GstRTSPMedia *media, GstState state, GArray *transport
add = remove = FALSE;
g_message ("going to state %s media %p", gst_element_state_get_name (state), media);
switch (state) {
case GST_STATE_NULL:
/* unlock the streams so that they follow the state changes from now on */
unlock_streams (media);
break;
/* fallthrough */
case GST_STATE_PAUSED:
/* we're going from PLAYING to PAUSED, READY or NULL, remove */
if (media->target_state == GST_STATE_PLAYING)
@ -1325,27 +1357,31 @@ gst_rtsp_media_set_state (GstRTSPMedia *media, GstState state, GArray *transport
min = trans->client_port.min;
max = trans->client_port.max;
if (add) {
if (add && !tr->active) {
g_message ("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);
tr->active = TRUE;
media->active++;
} else if (remove) {
} else if (remove && tr->active) {
g_message ("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);
tr->active = FALSE;
media->active--;
}
break;
}
case GST_RTSP_LOWER_TRANS_TCP:
if (add) {
if (add && !tr->active) {
g_message ("adding TCP %s", trans->destination);
stream->transports = g_list_prepend (stream->transports, tr);
tr->active = TRUE;
media->active++;
} else if (remove) {
} else if (remove && tr->active) {
g_message ("removing TCP %s", trans->destination);
stream->transports = g_list_remove (stream->transports, tr);
tr->active = FALSE;
media->active--;
}
break;

1
gst/rtsp-server/rtsp-media.h
View file

@ -61,6 +61,7 @@ struct _GstRTSPMediaTrans {
GstRTSPSendFunc send_rtcp;
gpointer user_data;
GDestroyNotify notify;
gboolean active;
GstRTSPTransport *transport;
};