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rtsp: only create transport when needed

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Only create the StreamTransport when configured.
This commit is contained in:
Wim Taymans 2012-10-28 00:23:57 +02:00
parent 66a29c7ed9
commit 543aa383e7
5 changed files with 56 additions and 38 deletions
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16
gst/rtsp-server/rtsp-client.c
View file

@ -537,12 +537,13 @@ unlink_session_transports (GstRTSPClient * client, GstRTSPSession * session,
GstRTSPStreamTransport *trans;
GstRTSPTransport *tr;
/* get the stream as configured in the session */
trans = gst_rtsp_session_media_get_transport (media, i);
/* get the transport, if there is no transport configured, skip this stream */
if (!(tr = trans->transport))
trans = gst_rtsp_session_media_get_transport (media, i);
if (trans == NULL)
continue;
tr = trans->transport;
if (tr->lower_transport == GST_RTSP_LOWER_TRANS_TCP) {
/* for TCP, unlink the stream from the TCP connection of the client */
unlink_transport (client, session, trans);
@ -815,13 +816,13 @@ handle_play_request (GstRTSPClient * client, GstRTSPClientState * state)
gchar *uristr;
guint rtptime, seq;
/* get the stream as configured in the session */
trans = gst_rtsp_session_media_get_transport (media, i);
/* get the transport, if there is no transport configured, skip this stream */
if (!(tr = trans->transport)) {
trans = gst_rtsp_session_media_get_transport (media, i);
if (trans == NULL) {
GST_INFO ("stream %d is not configured", i);
continue;
}
tr = trans->transport;
if (tr->lower_transport == GST_RTSP_LOWER_TRANS_TCP) {
/* for TCP, link the stream to the TCP connection of the client */
@ -1153,8 +1154,7 @@ handle_setup_request (GstRTSPClient * client, GstRTSPClientState * state)
configure_client_transport (client, state, ct);
/* set in the session media transport */
trans = gst_rtsp_session_media_get_transport (sessmedia, streamid);
gst_rtsp_stream_transport_set_transport (trans, ct);
trans = gst_rtsp_session_media_set_transport (sessmedia, stream, ct);
/* configure keepalive for this transport */
gst_rtsp_stream_transport_set_keepalive (trans,

53
gst/rtsp-server/rtsp-session-media.c
View file

@ -116,12 +116,42 @@ gst_rtsp_session_media_new (const GstRTSPUrl * url, GstRTSPMedia * media)
return result;
}
/**
* gst_rtsp_session_media_set_transport:
* @media: a #GstRTSPSessionMedia
* @stream: a #GstRTSPStream
* @tr: a #GstRTSPTransport
*
* Configure the transport for @stream to @tr in @media.
*
* Returns: the new or updated #GstRTSPStreamTransport for @stream.
*/
GstRTSPStreamTransport *
gst_rtsp_session_media_set_transport (GstRTSPSessionMedia * media,
GstRTSPStream * stream, GstRTSPTransport * tr)
{
GstRTSPStreamTransport *result;
g_return_val_if_fail (GST_IS_RTSP_SESSION_MEDIA (media), NULL);
g_return_val_if_fail (GST_IS_RTSP_STREAM (stream), NULL);
g_return_val_if_fail (stream->idx < media->transports->len, NULL);
result = g_ptr_array_index (media->transports, stream->idx);
if (result == NULL) {
result = gst_rtsp_stream_transport_new (stream, tr);
g_ptr_array_index (media->transports, stream->idx) = result;
} else {
gst_rtsp_stream_transport_set_transport (result, tr);
}
return result;
}
/**
* gst_rtsp_session_media_get_transport:
* @media: a #GstRTSPSessionMedia
* @idx: the stream index
*
* Get a previously created or create a new #GstRTSPStreamTransport at @idx.
* Get a previously created #GstRTSPStreamTransport for the stream at @idx.
*
* Returns: a #GstRTSPStreamTransport that is valid until the session of @media
* is unreffed.
@ -132,30 +162,11 @@ gst_rtsp_session_media_get_transport (GstRTSPSessionMedia * media, guint idx)
GstRTSPStreamTransport *result;
g_return_val_if_fail (GST_IS_RTSP_SESSION_MEDIA (media), NULL);
g_return_val_if_fail (media->media != NULL, NULL);
if (idx >= media->transports->len)
return NULL;
g_return_val_if_fail (idx < media->transports->len, NULL);
result = g_ptr_array_index (media->transports, idx);
if (result == NULL) {
GstRTSPStream *stream;
stream = gst_rtsp_media_get_stream (media->media, idx);
if (stream == NULL)
goto no_media;
result = gst_rtsp_stream_transport_new (stream);
g_ptr_array_index (media->transports, idx) = result;
}
return result;
/* ERRORS */
no_media:
{
return NULL;
}
}
/**

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

@ -75,6 +75,9 @@ gboolean gst_rtsp_session_media_set_state (GstRTSPSessionMe
GstState state);
/* get stream transport config */
GstRTSPStreamTransport * gst_rtsp_session_media_set_transport (GstRTSPSessionMedia *media,
GstRTSPStream *stream,
GstRTSPTransport *tr);
GstRTSPStreamTransport * gst_rtsp_session_media_get_transport (GstRTSPSessionMedia *media,
guint idx);

17
gst/rtsp-server/rtsp-stream-transport.c
View file

@ -82,21 +82,24 @@ gst_rtsp_stream_transport_finalize (GObject * obj)
/**
* gst_rtsp_stream_transport_new:
* @stream: a #GstRTSPStream
* @tr: (transfer full): a GstRTSPTransport
*
* Create a new #GstRTSPStreamTransport that can be used to manage
* @stream.
* @stream with transport @tr.
*
* Returns: a new #GstRTSPStreamTransport
*/
GstRTSPStreamTransport *
gst_rtsp_stream_transport_new (GstRTSPStream * stream)
gst_rtsp_stream_transport_new (GstRTSPStream * stream, GstRTSPTransport * tr)
{
GstRTSPStreamTransport *trans;
g_return_val_if_fail (GST_IS_RTSP_STREAM (stream), NULL);
g_return_val_if_fail (tr != NULL, NULL);
trans = g_object_new (GST_TYPE_RTSP_STREAM_TRANSPORT, NULL);
trans->stream = stream;
trans->transport = tr;
return trans;
}
@ -150,20 +153,20 @@ gst_rtsp_stream_transport_set_keepalive (GstRTSPStreamTransport * trans,
/**
* gst_rtsp_stream_transport_set_transport:
* @trans: a #GstRTSPStreamTransport
* @ct: a client #GstRTSPTransport
* @tr: (transfer full): a client #GstRTSPTransport
*
* Set @ct as the client transport. This function takes ownership of
* the passed @ct.
* the passed @tr.
*/
void
gst_rtsp_stream_transport_set_transport (GstRTSPStreamTransport * trans,
GstRTSPTransport * ct)
GstRTSPTransport * tr)
{
g_return_if_fail (GST_IS_RTSP_STREAM_TRANSPORT (trans));
g_return_if_fail (ct != NULL);
g_return_if_fail (tr != NULL);
/* keep track of the transports in the stream. */
if (trans->transport)
gst_rtsp_transport_free (trans->transport);
trans->transport = ct;
trans->transport = tr;
}

5
gst/rtsp-server/rtsp-stream-transport.h
View file

@ -89,10 +89,11 @@ struct _GstRTSPStreamTransportClass {
GType gst_rtsp_stream_transport_get_type (void);
GstRTSPStreamTransport * gst_rtsp_stream_transport_new (GstRTSPStream *stream);
GstRTSPStreamTransport * gst_rtsp_stream_transport_new (GstRTSPStream *stream,
GstRTSPTransport *tr);
void gst_rtsp_stream_transport_set_transport (GstRTSPStreamTransport *trans,
GstRTSPTransport * ct);
GstRTSPTransport * tr);
void gst_rtsp_stream_transport_set_callbacks (GstRTSPStreamTransport *trans,
GstRTSPSendFunc send_rtp,