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rtsp: refactor configuration of transport

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Move the configuration of the transport to a place where it makes
more sense.
This commit is contained in:
Wim Taymans 2012-10-27 23:49:24 +02:00
parent 8c30d050fa
commit fb117a4f75
4 changed files with 118 additions and 99 deletions
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166
gst/rtsp-server/rtsp-client.c
View file

@ -506,7 +506,9 @@ link_transport (GstRTSPClient * client, GstRTSPSession * session,
gst_rtsp_stream_transport_set_callbacks (trans,
(GstRTSPSendFunc) do_send_data,
(GstRTSPSendFunc) do_send_data, client, NULL);
client->transports = g_list_prepend (client->transports, trans);
/* make sure our session can't expire */
gst_rtsp_session_prevent_expire (session);
}
@ -517,7 +519,9 @@ unlink_transport (GstRTSPClient * client, GstRTSPSession * session,
{
GST_DEBUG ("client %p: unlinking transport %p", client, trans);
gst_rtsp_stream_transport_set_callbacks (trans, NULL, NULL, NULL, NULL);
client->transports = g_list_remove (client->transports, trans);
/* our session can now expire */
gst_rtsp_session_allow_expire (session);
}
@ -971,10 +975,77 @@ handle_blocksize (GstRTSPMedia * media, GstRTSPMessage * request)
gst_rtsp_media_set_mtu (media, blocksize);
}
}
return ret;
}
static gboolean
configure_client_transport (GstRTSPClient * client, GstRTSPClientState * state,
GstRTSPTransport * ct)
{
/* we have a valid transport now, set the destination of the client. */
if (ct->lower_transport == GST_RTSP_LOWER_TRANS_UDP_MCAST) {
if (ct->destination == NULL || !client->use_client_settings) {
g_free (ct->destination);
ct->destination = gst_rtsp_media_get_multicast_group (state->media);
}
/* reset ttl and port if client settings are not allowed */
if (!client->use_client_settings) {
ct->port = state->stream->server_port;
ct->ttl = 0;
}
} else {
GstRTSPUrl *url;
url = gst_rtsp_connection_get_url (client->connection);
g_free (ct->destination);
ct->destination = g_strdup (url->host);
if (ct->lower_transport & GST_RTSP_LOWER_TRANS_TCP) {
/* check if the client selected channels for TCP */
if (ct->interleaved.min == -1 || ct->interleaved.max == -1) {
gst_rtsp_session_media_alloc_channels (state->sessmedia,
&ct->interleaved);
}
}
}
return TRUE;
}
static GstRTSPTransport *
make_server_transport (GstRTSPClient * client, GstRTSPClientState * state,
GstRTSPTransport * ct)
{
GstRTSPTransport *st;
/* prepare the server transport */
gst_rtsp_transport_new (&st);
st->trans = ct->trans;
st->profile = ct->profile;
st->lower_transport = ct->lower_transport;
switch (st->lower_transport) {
case GST_RTSP_LOWER_TRANS_UDP:
st->client_port = ct->client_port;
st->server_port = state->stream->server_port;
break;
case GST_RTSP_LOWER_TRANS_UDP_MCAST:
st->port = ct->port;
st->destination = g_strdup (ct->destination);
st->ttl = ct->ttl;
break;
case GST_RTSP_LOWER_TRANS_TCP:
st->interleaved = ct->interleaved;
default:
break;
}
if (state->stream->session)
g_object_get (state->stream->session, "internal-ssrc", &st->ssrc, NULL);
return st;
}
static gboolean
handle_setup_request (GstRTSPClient * client, GstRTSPClientState * state)
{
@ -988,7 +1059,9 @@ handle_setup_request (GstRTSPClient * client, GstRTSPClientState * state)
GstRTSPStreamTransport *trans;
gchar *trans_str, *pos;
guint streamid;
GstRTSPSessionMedia *media;
GstRTSPSessionMedia *sessmedia;
GstRTSPMedia *media;
GstRTSPStream *stream;
uri = state->uri;
@ -1025,6 +1098,8 @@ handle_setup_request (GstRTSPClient * client, GstRTSPClientState * state)
if (!parse_transport (transport, supported, ct))
goto unsupported_transports;
/* we create the session after parsing stuff so that we don't make
* a session for malformed requests */
if (client->session_pool == NULL)
goto no_pool;
@ -1034,7 +1109,7 @@ handle_setup_request (GstRTSPClient * client, GstRTSPClientState * state)
g_object_ref (session);
/* get a handle to the configuration of the media in the session, this can
* return NULL if this is a new url to manage in this session. */
media = gst_rtsp_session_get_media (session, uri);
sessmedia = gst_rtsp_session_get_media (session, uri);
} else {
/* create a session if this fails we probably reached our session limit or
* something. */
@ -1044,65 +1119,49 @@ handle_setup_request (GstRTSPClient * client, GstRTSPClientState * state)
state->session = session;
/* we need a new media configuration in this session */
media = NULL;
sessmedia = NULL;
}
/* we have no media, find one and manage it */
if (media == NULL) {
GstRTSPMedia *m;
if (sessmedia == NULL) {
/* get a handle to the configuration of the media in the session */
if ((m = find_media (client, state))) {
if ((media = find_media (client, state))) {
/* manage the media in our session now */
media = gst_rtsp_session_manage_media (session, uri, m);
sessmedia = gst_rtsp_session_manage_media (session, uri, media);
}
}
/* if we stil have no media, error */
if (media == NULL)
if (sessmedia == NULL)
goto not_found;
state->sessmedia = media;
state->sessmedia = sessmedia;
state->media = media = sessmedia->media;
if (!handle_blocksize (media->media, state->request))
/* now get the stream */
stream = gst_rtsp_media_get_stream (media, streamid);
if (stream == NULL)
goto not_found;
state->stream = stream;
/* FIXME set only on this stream */
if (!handle_blocksize (media, state->request))
goto invalid_blocksize;
/* we have a valid transport now, set the destination of the client. */
if (ct->lower_transport == GST_RTSP_LOWER_TRANS_UDP_MCAST) {
if (ct->destination == NULL || !client->use_client_settings) {
g_free (ct->destination);
ct->destination = gst_rtsp_media_get_multicast_group (media->media);
}
/* reset ttl if client settings are not allowed */
if (!client->use_client_settings) {
ct->ttl = 0;
}
} else {
GstRTSPUrl *url;
/* update the client transport */
configure_client_transport (client, state, ct);
url = gst_rtsp_connection_get_url (client->connection);
g_free (ct->destination);
ct->destination = g_strdup (url->host);
if (ct->lower_transport & GST_RTSP_LOWER_TRANS_TCP) {
/* check if the client selected channels for TCP */
if (ct->interleaved.min == -1 || ct->interleaved.max == -1) {
gst_rtsp_session_media_alloc_channels (media, &ct->interleaved);
}
}
}
/* get a handle to the transport of the media in this session */
if (!(trans = gst_rtsp_session_media_get_transport (media, streamid)))
goto no_stream_transport;
st = gst_rtsp_stream_transport_set_transport (trans, ct);
/* set in the session media transport */
trans = gst_rtsp_session_media_get_transport (sessmedia, streamid);
gst_rtsp_stream_transport_set_transport (trans, ct);
/* configure keepalive for this transport */
gst_rtsp_stream_transport_set_keepalive (trans,
(GstRTSPKeepAliveFunc) do_keepalive, session, NULL);
/* serialize the server transport */
/* create and serialize the server transport */
st = make_server_transport (client, state, ct);
trans_str = gst_rtsp_transport_as_text (st);
gst_rtsp_transport_free (st);
@ -1118,14 +1177,14 @@ handle_setup_request (GstRTSPClient * client, GstRTSPClientState * state)
send_response (client, session, state->response);
/* update the state */
switch (media->state) {
switch (sessmedia->state) {
case GST_RTSP_STATE_PLAYING:
case GST_RTSP_STATE_RECORDING:
case GST_RTSP_STATE_READY:
/* no state change */
break;
default:
media->state = GST_RTSP_STATE_READY;
sessmedia->state = GST_RTSP_STATE_READY;
break;
}
g_object_unref (session);
@ -1155,13 +1214,6 @@ invalid_blocksize:
gst_rtsp_transport_free (ct);
return FALSE;
}
no_stream_transport:
{
send_generic_response (client, GST_RTSP_STS_NOT_FOUND, state);
g_object_unref (session);
gst_rtsp_transport_free (ct);
return FALSE;
}
no_transport:
{
send_generic_response (client, GST_RTSP_STS_UNSUPPORTED_TRANSPORT, state);
@ -1573,17 +1625,15 @@ handle_data (GstRTSPClient * client, GstRTSPMessage * message)
handled = FALSE;
for (walk = client->transports; walk; walk = g_list_next (walk)) {
GstRTSPStreamTransport *trans = (GstRTSPStreamTransport *) walk->data;
GstRTSPStreamTransport *trans;
GstRTSPStream *stream;
GstRTSPTransport *tr;
/* get the transport, if there is no transport configured, skip this stream */
if (!(tr = trans->transport))
continue;
trans = walk->data;
/* we also need a media stream */
if (!(stream = trans->stream))
continue;
/* we only add clients with a transport to the list */
tr = trans->transport;
stream = trans->stream;
/* check for TCP transport */
if (tr->lower_transport == GST_RTSP_LOWER_TRANS_TCP) {

4
gst/rtsp-server/rtsp-client.h
View file

@ -54,7 +54,8 @@ typedef struct _GstRTSPClientState GstRTSPClientState;
* @session: the session, can be NULL
* @sessmedia: the session media for the url can be NULL
* @factory: the media factory for the url, can be NULL.
* @media: the session media for the url can be NULL
* @media: the media for the url can be NULL
* @stream: the stream for the url can be NULL
* @response: the response
*
* Information passed around containing the client state of a request.
@ -67,6 +68,7 @@ struct _GstRTSPClientState {
GstRTSPSessionMedia *sessmedia;
GstRTSPMediaFactory *factory;
GstRTSPMedia *media;
GstRTSPStream *stream;
GstRTSPMessage *response;
};

45
gst/rtsp-server/rtsp-stream-transport.c
View file

@ -83,7 +83,7 @@ gst_rtsp_stream_transport_finalize (GObject * obj)
* gst_rtsp_stream_transport_new:
* @stream: a #GstRTSPStream
*
* Create a new #GstRTSPStreamTransport that can be used for
* Create a new #GstRTSPStreamTransport that can be used to manage
* @stream.
*
* Returns: a new #GstRTSPStreamTransport
@ -152,51 +152,18 @@ gst_rtsp_stream_transport_set_keepalive (GstRTSPStreamTransport * trans,
* @trans: a #GstRTSPStreamTransport
* @ct: a client #GstRTSPTransport
*
* Set @ct as the client transport and create and return a matching server
* transport. This function takes ownership of the passed @ct.
*
* Returns: a server transport or NULL if something went wrong. Use
* gst_rtsp_transport_free () after usage.
* Set @ct as the client transport. This function takes ownership of
* the passed @ct.
*/
GstRTSPTransport *
void
gst_rtsp_stream_transport_set_transport (GstRTSPStreamTransport * trans,
GstRTSPTransport * ct)
{
GstRTSPTransport *st;
g_return_val_if_fail (GST_IS_RTSP_STREAM_TRANSPORT (trans), NULL);
g_return_val_if_fail (ct != NULL, NULL);
/* prepare the server transport */
gst_rtsp_transport_new (&st);
st->trans = ct->trans;
st->profile = ct->profile;
st->lower_transport = ct->lower_transport;
switch (st->lower_transport) {
case GST_RTSP_LOWER_TRANS_UDP:
st->client_port = ct->client_port;
st->server_port = trans->stream->server_port;
break;
case GST_RTSP_LOWER_TRANS_UDP_MCAST:
ct->port = st->port = trans->stream->server_port;
st->destination = g_strdup (ct->destination);
st->ttl = ct->ttl;
break;
case GST_RTSP_LOWER_TRANS_TCP:
st->interleaved = ct->interleaved;
default:
break;
}
if (trans->stream->session)
g_object_get (trans->stream->session, "internal-ssrc", &st->ssrc, NULL);
g_return_if_fail (GST_IS_RTSP_STREAM_TRANSPORT (trans));
g_return_if_fail (ct != NULL);
/* keep track of the transports in the stream. */
if (trans->transport)
gst_rtsp_transport_free (trans->transport);
trans->transport = ct;
return st;
}

2
gst/rtsp-server/rtsp-stream-transport.h
View file

@ -91,7 +91,7 @@ GType gst_rtsp_stream_transport_get_type (void);
GstRTSPStreamTransport * gst_rtsp_stream_transport_new (GstRTSPStream *stream);
GstRTSPTransport * gst_rtsp_stream_transport_set_transport (GstRTSPStreamTransport *trans,
void gst_rtsp_stream_transport_set_transport (GstRTSPStreamTransport *trans,
GstRTSPTransport * ct);
void gst_rtsp_stream_transport_set_callbacks (GstRTSPStreamTransport *trans,