*
* You should have received a copy of the GNU Library General Public
* License along with this library; if not, write to the
- * Free Software Foundation, Inc., 59 Temple Place - Suite 330,
- * Boston, MA 02111-1307, USA.
+ * Free Software Foundation, Inc., 51 Franklin St, Fifth Floor,
+ * Boston, MA 02110-1301, USA.
*/
#include <stdio.h>
-#include <stdlib.h>
-#include <unistd.h>
-#include <errno.h>
#include <string.h>
-#include <sys/time.h>
-#include <sys/types.h>
-#include <netinet/in.h>
-#include <netdb.h>
-#include <sys/socket.h>
-#include <sys/wait.h>
-#include <fcntl.h>
-#include <arpa/inet.h>
-#include <sys/ioctl.h>
#include "rtsp-client.h"
#include "rtsp-sdp.h"
#include "rtsp-params.h"
-/* temporary multicast address until it's configurable somewhere */
-#define MCAST_ADDRESS "224.2.0.1"
-
-static GMutex *tunnels_lock;
+static GMutex tunnels_lock;
static GHashTable *tunnels;
+#define DEFAULT_SESSION_POOL NULL
+#define DEFAULT_MEDIA_MAPPING NULL
+#define DEFAULT_USE_CLIENT_SETTINGS FALSE
+
enum
{
PROP_0,
PROP_SESSION_POOL,
PROP_MEDIA_MAPPING,
+ PROP_USE_CLIENT_SETTINGS,
PROP_LAST
};
enum
{
SIGNAL_CLOSED,
+ SIGNAL_NEW_SESSION,
+ SIGNAL_OPTIONS_REQUEST,
+ SIGNAL_DESCRIBE_REQUEST,
+ SIGNAL_SETUP_REQUEST,
+ SIGNAL_PLAY_REQUEST,
+ SIGNAL_PAUSE_REQUEST,
+ SIGNAL_TEARDOWN_REQUEST,
+ SIGNAL_SET_PARAMETER_REQUEST,
+ SIGNAL_GET_PARAMETER_REQUEST,
SIGNAL_LAST
};
const GValue * value, GParamSpec * pspec);
static void gst_rtsp_client_finalize (GObject * obj);
+static GstSDPMessage *create_sdp (GstRTSPClient * client, GstRTSPMedia * media);
static void client_session_finalized (GstRTSPClient * client,
GstRTSPSession * session);
-static void unlink_session_streams (GstRTSPClient * client,
+static void unlink_session_transports (GstRTSPClient * client,
GstRTSPSession * session, GstRTSPSessionMedia * media);
G_DEFINE_TYPE (GstRTSPClient, gst_rtsp_client, G_TYPE_OBJECT);
gobject_class->set_property = gst_rtsp_client_set_property;
gobject_class->finalize = gst_rtsp_client_finalize;
+ klass->create_sdp = create_sdp;
+
g_object_class_install_property (gobject_class, PROP_SESSION_POOL,
g_param_spec_object ("session-pool", "Session Pool",
"The session pool to use for client session",
GST_TYPE_RTSP_MEDIA_MAPPING,
G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
+ g_object_class_install_property (gobject_class, PROP_USE_CLIENT_SETTINGS,
+ g_param_spec_boolean ("use-client-settings", "Use Client Settings",
+ "Use client settings for ttl and destination in multicast",
+ DEFAULT_USE_CLIENT_SETTINGS,
+ G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
+
gst_rtsp_client_signals[SIGNAL_CLOSED] =
g_signal_new ("closed", G_TYPE_FROM_CLASS (klass), G_SIGNAL_RUN_LAST,
G_STRUCT_OFFSET (GstRTSPClientClass, closed), NULL, NULL,
g_cclosure_marshal_VOID__VOID, G_TYPE_NONE, 0, G_TYPE_NONE);
+ gst_rtsp_client_signals[SIGNAL_NEW_SESSION] =
+ g_signal_new ("new-session", G_TYPE_FROM_CLASS (klass), G_SIGNAL_RUN_LAST,
+ G_STRUCT_OFFSET (GstRTSPClientClass, new_session), NULL, NULL,
+ g_cclosure_marshal_VOID__OBJECT, G_TYPE_NONE, 1, GST_TYPE_RTSP_SESSION);
+
+ gst_rtsp_client_signals[SIGNAL_OPTIONS_REQUEST] =
+ g_signal_new ("options-request", G_TYPE_FROM_CLASS (klass),
+ G_SIGNAL_RUN_LAST, G_STRUCT_OFFSET (GstRTSPClientClass, options_request),
+ NULL, NULL, g_cclosure_marshal_VOID__POINTER, G_TYPE_NONE, 1,
+ G_TYPE_POINTER);
+
+ gst_rtsp_client_signals[SIGNAL_DESCRIBE_REQUEST] =
+ g_signal_new ("describe-request", G_TYPE_FROM_CLASS (klass),
+ G_SIGNAL_RUN_LAST, G_STRUCT_OFFSET (GstRTSPClientClass, describe_request),
+ NULL, NULL, g_cclosure_marshal_VOID__POINTER, G_TYPE_NONE, 1,
+ G_TYPE_POINTER);
+
+ gst_rtsp_client_signals[SIGNAL_SETUP_REQUEST] =
+ g_signal_new ("setup-request", G_TYPE_FROM_CLASS (klass),
+ G_SIGNAL_RUN_LAST, G_STRUCT_OFFSET (GstRTSPClientClass, setup_request),
+ NULL, NULL, g_cclosure_marshal_VOID__POINTER, G_TYPE_NONE, 1,
+ G_TYPE_POINTER);
+
+ gst_rtsp_client_signals[SIGNAL_PLAY_REQUEST] =
+ g_signal_new ("play-request", G_TYPE_FROM_CLASS (klass),
+ G_SIGNAL_RUN_LAST, G_STRUCT_OFFSET (GstRTSPClientClass, play_request),
+ NULL, NULL, g_cclosure_marshal_VOID__POINTER, G_TYPE_NONE, 1,
+ G_TYPE_POINTER);
+
+ gst_rtsp_client_signals[SIGNAL_PAUSE_REQUEST] =
+ g_signal_new ("pause-request", G_TYPE_FROM_CLASS (klass),
+ G_SIGNAL_RUN_LAST, G_STRUCT_OFFSET (GstRTSPClientClass, pause_request),
+ NULL, NULL, g_cclosure_marshal_VOID__POINTER, G_TYPE_NONE, 1,
+ G_TYPE_POINTER);
+
+ gst_rtsp_client_signals[SIGNAL_TEARDOWN_REQUEST] =
+ g_signal_new ("teardown-request", G_TYPE_FROM_CLASS (klass),
+ G_SIGNAL_RUN_LAST, G_STRUCT_OFFSET (GstRTSPClientClass, teardown_request),
+ NULL, NULL, g_cclosure_marshal_VOID__POINTER, G_TYPE_NONE, 1,
+ G_TYPE_POINTER);
+
+ gst_rtsp_client_signals[SIGNAL_SET_PARAMETER_REQUEST] =
+ g_signal_new ("set-parameter-request", G_TYPE_FROM_CLASS (klass),
+ G_SIGNAL_RUN_LAST, G_STRUCT_OFFSET (GstRTSPClientClass,
+ set_parameter_request), NULL, NULL, g_cclosure_marshal_VOID__POINTER,
+ G_TYPE_NONE, 1, G_TYPE_POINTER);
+
+ gst_rtsp_client_signals[SIGNAL_GET_PARAMETER_REQUEST] =
+ g_signal_new ("get-parameter-request", G_TYPE_FROM_CLASS (klass),
+ G_SIGNAL_RUN_LAST, G_STRUCT_OFFSET (GstRTSPClientClass,
+ get_parameter_request), NULL, NULL, g_cclosure_marshal_VOID__POINTER,
+ G_TYPE_NONE, 1, G_TYPE_POINTER);
+
tunnels =
g_hash_table_new_full (g_str_hash, g_str_equal, g_free, g_object_unref);
- tunnels_lock = g_mutex_new ();
+ g_mutex_init (&tunnels_lock);
GST_DEBUG_CATEGORY_INIT (rtsp_client_debug, "rtspclient", 0, "GstRTSPClient");
}
static void
gst_rtsp_client_init (GstRTSPClient * client)
{
+ client->use_client_settings = DEFAULT_USE_CLIENT_SETTINGS;
+ client->teardown_response_seq = 0;
}
static void
client_unlink_session (GstRTSPClient * client, GstRTSPSession * session)
{
- GList *medias;
-
/* unlink all media managed in this session */
- for (medias = session->medias; medias; medias = g_list_next (medias)) {
- GstRTSPSessionMedia *media = medias->data;
+ while (session->medias) {
+ GstRTSPSessionMedia *media = session->medias->data;
gst_rtsp_session_media_set_state (media, GST_STATE_NULL);
- unlink_session_streams (client, session, media);
- /* unmanage the media in the session. */
+ unlink_session_transports (client, session, media);
+ /* unmanage the media in the session. this will modify session->medias */
gst_rtsp_session_release_media (session, media);
}
}
GST_INFO ("finalize client %p", client);
+ if (client->watch)
+ g_source_destroy ((GSource *) client->watch);
+
client_cleanup_sessions (client);
gst_rtsp_connection_free (client->connection);
case PROP_MEDIA_MAPPING:
g_value_take_object (value, gst_rtsp_client_get_media_mapping (client));
break;
+ case PROP_USE_CLIENT_SETTINGS:
+ g_value_set_boolean (value,
+ gst_rtsp_client_get_use_client_settings (client));
+ break;
default:
G_OBJECT_WARN_INVALID_PROPERTY_ID (object, propid, pspec);
}
case PROP_MEDIA_MAPPING:
gst_rtsp_client_set_media_mapping (client, g_value_get_object (value));
break;
+ case PROP_USE_CLIENT_SETTINGS:
+ gst_rtsp_client_set_use_client_settings (client,
+ g_value_get_boolean (value));
+ break;
default:
G_OBJECT_WARN_INVALID_PROPERTY_ID (object, propid, pspec);
}
static void
send_response (GstRTSPClient * client, GstRTSPSession * session,
- GstRTSPMessage * response)
+ GstRTSPMessage * response, guint * id)
{
gst_rtsp_message_add_header (response, GST_RTSP_HDR_SERVER,
"GStreamer RTSP server");
/* add the new session header for new session ids */
if (session) {
- gchar *str;
-
- if (session->timeout != 60)
- str =
- g_strdup_printf ("%s; timeout=%d", session->sessionid,
- session->timeout);
- else
- str = g_strdup (session->sessionid);
-
- gst_rtsp_message_take_header (response, GST_RTSP_HDR_SESSION, str);
+ gst_rtsp_message_take_header (response, GST_RTSP_HDR_SESSION,
+ gst_rtsp_session_get_header (session));
}
if (gst_debug_category_get_threshold (rtsp_client_debug) >= GST_LEVEL_LOG) {
gst_rtsp_message_dump (response);
}
- gst_rtsp_watch_send_message (client->watch, response, NULL);
+ gst_rtsp_watch_send_message (client->watch, response, id);
gst_rtsp_message_unset (response);
}
gst_rtsp_message_init_response (state->response, code,
gst_rtsp_status_as_text (code), state->request);
- send_response (client, NULL, state->response);
+ send_response (client, NULL, state->response, NULL);
}
static void
gst_rtsp_auth_setup_auth (auth, client, 0, state);
}
- send_response (client, state->session, state->response);
+ send_response (client, state->session, state->response, NULL);
}
do_send_data (GstBuffer * buffer, guint8 channel, GstRTSPClient * client)
{
GstRTSPMessage message = { 0 };
+ GstMapInfo map_info;
guint8 *data;
- guint size;
+ guint usize;
gst_rtsp_message_init_data (&message, channel);
- data = GST_BUFFER_DATA (buffer);
- size = GST_BUFFER_SIZE (buffer);
- gst_rtsp_message_take_body (&message, data, size);
+ /* FIXME, need some sort of iovec RTSPMessage here */
+ if (!gst_buffer_map (buffer, &map_info, GST_MAP_READ))
+ return FALSE;
+
+ gst_rtsp_message_take_body (&message, map_info.data, map_info.size);
/* FIXME, client->watch could have been finalized here, we need to keep an
* extra refcount to the watch. */
gst_rtsp_watch_send_message (client->watch, &message, NULL);
- gst_rtsp_message_steal_body (&message, &data, &size);
+ gst_rtsp_message_steal_body (&message, &data, &usize);
+ gst_buffer_unmap (buffer, &map_info);
+
gst_rtsp_message_unset (&message);
return TRUE;
}
-static gboolean
-do_send_data_list (GstBufferList * blist, guint8 channel,
- GstRTSPClient * client)
+static void
+link_transport (GstRTSPClient * client, GstRTSPSession * session,
+ GstRTSPStreamTransport * trans)
{
- GstBufferListIterator *it;
+ GST_DEBUG ("client %p: linking transport %p", client, trans);
+ gst_rtsp_stream_transport_set_callbacks (trans,
+ (GstRTSPSendFunc) do_send_data,
+ (GstRTSPSendFunc) do_send_data, client, NULL);
- it = gst_buffer_list_iterate (blist);
- while (gst_buffer_list_iterator_next_group (it)) {
- GstBuffer *group = gst_buffer_list_iterator_merge_group (it);
+ client->transports = g_list_prepend (client->transports, trans);
- if (group == NULL)
- continue;
-
- do_send_data (group, channel, client);
- }
- gst_buffer_list_iterator_free (it);
-
- return TRUE;
-}
-
-static void
-link_stream (GstRTSPClient * client, GstRTSPSession * session,
- GstRTSPSessionStream * stream)
-{
- GST_DEBUG ("client %p: linking stream %p", client, stream);
- gst_rtsp_session_stream_set_callbacks (stream, (GstRTSPSendFunc) do_send_data,
- (GstRTSPSendFunc) do_send_data, (GstRTSPSendListFunc) do_send_data_list,
- (GstRTSPSendListFunc) do_send_data_list, client, NULL);
- client->streams = g_list_prepend (client->streams, stream);
/* make sure our session can't expire */
gst_rtsp_session_prevent_expire (session);
}
static void
-unlink_stream (GstRTSPClient * client, GstRTSPSession * session,
- GstRTSPSessionStream * stream)
+unlink_transport (GstRTSPClient * client, GstRTSPSession * session,
+ GstRTSPStreamTransport * trans)
{
- GST_DEBUG ("client %p: unlinking stream %p", client, stream);
- gst_rtsp_session_stream_set_callbacks (stream, NULL, NULL, NULL, NULL, NULL,
- NULL);
- client->streams = g_list_remove (client->streams, stream);
+ 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);
}
static void
-unlink_session_streams (GstRTSPClient * client, GstRTSPSession * session,
+unlink_session_transports (GstRTSPClient * client, GstRTSPSession * session,
GstRTSPSessionMedia * media)
{
guint n_streams, i;
n_streams = gst_rtsp_media_n_streams (media->media);
for (i = 0; i < n_streams; i++) {
- GstRTSPSessionStream *sstream;
+ GstRTSPStreamTransport *trans;
GstRTSPTransport *tr;
- /* get the stream as configured in the session */
- sstream = gst_rtsp_session_media_get_stream (media, i);
/* get the transport, if there is no transport configured, skip this stream */
- if (!(tr = sstream->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_stream (client, session, sstream);
+ unlink_transport (client, session, trans);
}
}
}
GST_DEBUG ("client %p: closing connection", client);
if ((tunnelid = gst_rtsp_connection_get_tunnelid (client->connection))) {
- g_mutex_lock (tunnels_lock);
+ g_mutex_lock (&tunnels_lock);
/* remove from tunnelids */
g_hash_table_remove (tunnels, tunnelid);
- g_mutex_unlock (tunnels_lock);
+ g_mutex_unlock (&tunnels_lock);
}
gst_rtsp_connection_close (client->connection);
- if (client->watchid) {
- g_source_destroy ((GSource *) client->watch);
- client->watchid = 0;
- client->watch = NULL;
- }
}
static gboolean
state->sessmedia = media;
/* unlink the all TCP callbacks */
- unlink_session_streams (client, session, media);
+ unlink_session_transports (client, session, media);
/* remove the session from the watched sessions */
g_object_weak_unref (G_OBJECT (session),
gst_rtsp_message_add_header (state->response, GST_RTSP_HDR_CONNECTION,
"close");
- send_response (client, session, state->response);
+ /* send the response and store the seq number so we can wait until it's
+ * written to the client to close the connection */
+ send_response (client, session, state->response,
+ &client->teardown_response_seq);
- close_connection (client);
+ /* we emit the signal before closing the connection */
+ g_signal_emit (client, gst_rtsp_client_signals[SIGNAL_TEARDOWN_REQUEST],
+ 0, state);
return TRUE;
if (res != GST_RTSP_OK)
goto bad_request;
- send_response (client, state->session, state->response);
+ send_response (client, state->session, state->response, NULL);
}
+
+ g_signal_emit (client, gst_rtsp_client_signals[SIGNAL_GET_PARAMETER_REQUEST],
+ 0, state);
+
return TRUE;
/* ERRORS */
if (res != GST_RTSP_OK)
goto bad_request;
- send_response (client, state->session, state->response);
+ send_response (client, state->session, state->response, NULL);
}
+
+ g_signal_emit (client, gst_rtsp_client_signals[SIGNAL_SET_PARAMETER_REQUEST],
+ 0, state);
+
return TRUE;
/* ERRORS */
goto invalid_state;
/* unlink the all TCP callbacks */
- unlink_session_streams (client, session, media);
+ unlink_session_transports (client, session, media);
/* then pause sending */
gst_rtsp_session_media_set_state (media, GST_STATE_PAUSED);
gst_rtsp_message_init_response (state->response, code,
gst_rtsp_status_as_text (code), state->request);
- send_response (client, session, state->response);
+ send_response (client, session, state->response, NULL);
/* the state is now READY */
media->state = GST_RTSP_STATE_READY;
+ g_signal_emit (client, gst_rtsp_client_signals[SIGNAL_PAUSE_REQUEST],
+ 0, state);
+
return TRUE;
/* ERRORS */
GstRTSPStatusCode code;
GString *rtpinfo;
guint n_streams, i, infocount;
- guint timestamp, seqnum;
gchar *str;
GstRTSPTimeRange *range;
GstRTSPResult res;
n_streams = gst_rtsp_media_n_streams (media->media);
for (i = 0, infocount = 0; i < n_streams; i++) {
- GstRTSPSessionStream *sstream;
- GstRTSPMediaStream *stream;
+ GstRTSPStreamTransport *trans;
GstRTSPTransport *tr;
- GObjectClass *payobjclass;
gchar *uristr;
+ guint rtptime, seq;
- /* get the stream as configured in the session */
- sstream = gst_rtsp_session_media_get_stream (media, i);
/* get the transport, if there is no transport configured, skip this stream */
- if (!(tr = sstream->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 */
- link_stream (client, session, sstream);
+ link_transport (client, session, trans);
}
- stream = sstream->media_stream;
-
- payobjclass = G_OBJECT_GET_CLASS (stream->payloader);
-
- if (g_object_class_find_property (payobjclass, "seqnum") &&
- g_object_class_find_property (payobjclass, "timestamp")) {
- GObject *payobj;
-
- payobj = G_OBJECT (stream->payloader);
-
- /* only add RTP-Info for streams with seqnum and timestamp */
- g_object_get (payobj, "seqnum", &seqnum, "timestamp", ×tamp, NULL);
-
+ if (gst_rtsp_stream_get_rtpinfo (trans->stream, &rtptime, &seq)) {
if (infocount > 0)
g_string_append (rtpinfo, ", ");
uristr = gst_rtsp_url_get_request_uri (state->uri);
g_string_append_printf (rtpinfo, "url=%s/stream=%d;seq=%u;rtptime=%u",
- uristr, i, seqnum, timestamp);
+ uristr, i, seq, rtptime);
g_free (uristr);
infocount++;
str = gst_rtsp_media_get_range_string (media->media, TRUE);
gst_rtsp_message_take_header (state->response, GST_RTSP_HDR_RANGE, str);
- send_response (client, session, state->response);
+ send_response (client, session, state->response, NULL);
/* start playing after sending the request */
gst_rtsp_session_media_set_state (media, GST_STATE_PLAYING);
media->state = GST_RTSP_STATE_PLAYING;
+ g_signal_emit (client, gst_rtsp_client_signals[SIGNAL_PLAY_REQUEST],
+ 0, state);
+
return TRUE;
/* ERRORS */
gst_rtsp_session_touch (session);
}
+/* parse @transport and return a valid transport in @tr. only transports
+ * from @supported are returned. Returns FALSE if no valid transport
+ * was found. */
+static gboolean
+parse_transport (const char *transport, GstRTSPLowerTrans supported,
+ GstRTSPTransport * tr)
+{
+ gint i;
+ gboolean res;
+ gchar **transports;
+
+ res = FALSE;
+ gst_rtsp_transport_init (tr);
+
+ GST_DEBUG ("parsing transports %s", transport);
+
+ transports = g_strsplit (transport, ",", 0);
+
+ /* loop through the transports, try to parse */
+ for (i = 0; transports[i]; i++) {
+ res = gst_rtsp_transport_parse (transports[i], tr);
+ if (res != GST_RTSP_OK) {
+ /* no valid transport, search some more */
+ GST_WARNING ("could not parse transport %s", transports[i]);
+ goto next;
+ }
+
+ /* we have a transport, see if it's RTP/AVP */
+ if (tr->trans != GST_RTSP_TRANS_RTP || tr->profile != GST_RTSP_PROFILE_AVP) {
+ GST_WARNING ("invalid transport %s", transports[i]);
+ goto next;
+ }
+
+ if (!(tr->lower_transport & supported)) {
+ GST_WARNING ("unsupported transport %s", transports[i]);
+ goto next;
+ }
+
+ /* we have a valid transport */
+ GST_INFO ("found valid transport %s", transports[i]);
+ res = TRUE;
+ break;
+
+ next:
+ gst_rtsp_transport_init (tr);
+ }
+ g_strfreev (transports);
+
+ return res;
+}
+
+static gboolean
+handle_blocksize (GstRTSPMedia * media, GstRTSPStream * stream,
+ GstRTSPMessage * request)
+{
+ gchar *blocksize_str;
+ gboolean ret = TRUE;
+
+ if (gst_rtsp_message_get_header (request, GST_RTSP_HDR_BLOCKSIZE,
+ &blocksize_str, 0) == GST_RTSP_OK) {
+ guint64 blocksize;
+ gchar *end;
+
+ blocksize = g_ascii_strtoull (blocksize_str, &end, 10);
+ if (end == blocksize_str) {
+ GST_ERROR ("failed to parse blocksize");
+ ret = FALSE;
+ } else {
+ /* we don't want to change the mtu when this media
+ * can be shared because it impacts other clients */
+ if (gst_rtsp_media_is_shared (media))
+ return TRUE;
+
+ if (blocksize > G_MAXUINT)
+ blocksize = G_MAXUINT;
+ gst_rtsp_stream_set_mtu (stream, 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) {
+ GstRTSPAddress *addr;
+
+ addr = gst_rtsp_stream_get_address (state->stream);
+ if (addr == NULL)
+ goto no_address;
+
+ g_free (ct->destination);
+ ct->destination = g_strdup (addr->address);
+ ct->port.min = addr->port;
+ ct->port.max = addr->port + addr->n_ports - 1;
+ ct->ttl = addr->ttl;
+ }
+ } 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;
+
+ /* ERRORS */
+no_address:
+ {
+ GST_ERROR_OBJECT (client, "failed to acquire address for stream");
+ return FALSE;
+ }
+}
+
+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)
{
GstRTSPResult res;
GstRTSPUrl *uri;
gchar *transport;
- gchar **transports;
- gboolean have_transport;
GstRTSPTransport *ct, *st;
- gint i;
GstRTSPLowerTrans supported;
GstRTSPStatusCode code;
GstRTSPSession *session;
- GstRTSPSessionStream *stream;
+ GstRTSPStreamTransport *trans;
gchar *trans_str, *pos;
guint streamid;
- GstRTSPSessionMedia *media;
- GstRTSPUrl *url;
+ GstRTSPSessionMedia *sessmedia;
+ GstRTSPMedia *media;
+ GstRTSPStream *stream;
uri = state->uri;
/* the uri contains the stream number we added in the SDP config, which is
- * always /stream=%d so we need to strip that off
+ * always /stream=%d so we need to strip that off
* parse the stream we need to configure, look for the stream in the abspath
* first and then in the query. */
if (uri->abspath == NULL || !(pos = strstr (uri->abspath, "/stream="))) {
goto bad_request;
}
- /* we can mofify the parse uri in place */
+ /* we can mofify the parsed uri in place */
*pos = '\0';
pos += strlen ("/stream=");
if (res != GST_RTSP_OK)
goto no_transport;
- transports = g_strsplit (transport, ",", 0);
gst_rtsp_transport_new (&ct);
- /* init transports */
- have_transport = FALSE;
- gst_rtsp_transport_init (ct);
-
/* our supported transports */
supported = GST_RTSP_LOWER_TRANS_UDP |
GST_RTSP_LOWER_TRANS_UDP_MCAST | GST_RTSP_LOWER_TRANS_TCP;
- /* loop through the transports, try to parse */
- for (i = 0; transports[i]; i++) {
- res = gst_rtsp_transport_parse (transports[i], ct);
- if (res != GST_RTSP_OK) {
- /* no valid transport, search some more */
- GST_WARNING ("could not parse transport %s", transports[i]);
- goto next;
- }
-
- /* we have a transport, see if it's RTP/AVP */
- if (ct->trans != GST_RTSP_TRANS_RTP || ct->profile != GST_RTSP_PROFILE_AVP) {
- GST_WARNING ("invalid transport %s", transports[i]);
- goto next;
- }
-
- if (!(ct->lower_transport & supported)) {
- GST_WARNING ("unsupported transport %s", transports[i]);
- goto next;
- }
-
- /* we have a valid transport */
- GST_INFO ("found valid transport %s", transports[i]);
- have_transport = TRUE;
- break;
-
- next:
- gst_rtsp_transport_init (ct);
- }
- g_strfreev (transports);
-
- /* we have not found anything usable, error out */
- if (!have_transport)
+ /* parse and find a usable supported transport */
+ 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;
- /* we have a valid transport now, set the destination of the client. */
- g_free (ct->destination);
- if (ct->lower_transport == GST_RTSP_LOWER_TRANS_UDP_MCAST) {
- ct->destination = g_strdup (MCAST_ADDRESS);
- } else {
- url = gst_rtsp_connection_get_url (client->connection);
- ct->destination = g_strdup (url->host);
- }
-
session = state->session;
if (session) {
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. */
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;
- /* fix the transports */
- 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);
- }
- }
+ /* now get the stream */
+ stream = gst_rtsp_media_get_stream (media, streamid);
+ if (stream == NULL)
+ goto not_found;
+
+ state->stream = stream;
+
+ /* set blocksize on this stream */
+ if (!handle_blocksize (media, stream, state->request))
+ goto invalid_blocksize;
- /* get a handle to the stream in the media */
- if (!(stream = gst_rtsp_session_media_get_stream (media, streamid)))
- goto no_stream;
+ /* update the client transport */
+ if (!configure_client_transport (client, state, ct))
+ goto unsupported_client_transport;
- st = gst_rtsp_session_stream_set_transport (stream, ct);
+ /* set in the session media transport */
+ trans = gst_rtsp_session_media_set_transport (sessmedia, stream, ct);
/* configure keepalive for this transport */
- gst_rtsp_session_stream_set_keepalive (stream,
+ 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);
trans_str);
g_free (trans_str);
- send_response (client, session, state->response);
+ send_response (client, session, state->response, NULL);
/* 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);
+ g_signal_emit (client, gst_rtsp_client_signals[SIGNAL_SETUP_REQUEST],
+ 0, state);
+
return TRUE;
/* ERRORS */
{
send_generic_response (client, GST_RTSP_STS_NOT_FOUND, state);
g_object_unref (session);
+ gst_rtsp_transport_free (ct);
return FALSE;
}
-no_stream:
+invalid_blocksize:
{
- send_generic_response (client, GST_RTSP_STS_NOT_FOUND, state);
- g_object_unref (media);
+ send_generic_response (client, GST_RTSP_STS_BAD_REQUEST, state);
g_object_unref (session);
+ gst_rtsp_transport_free (ct);
+ return FALSE;
+ }
+unsupported_client_transport:
+ {
+ send_generic_response (client, GST_RTSP_STS_UNSUPPORTED_TRANSPORT, state);
+ g_object_unref (session);
+ gst_rtsp_transport_free (ct);
return FALSE;
}
no_transport:
no_pool:
{
send_generic_response (client, GST_RTSP_STS_SERVICE_UNAVAILABLE, state);
+ gst_rtsp_transport_free (ct);
return FALSE;
}
service_unavailable:
{
send_generic_response (client, GST_RTSP_STS_SERVICE_UNAVAILABLE, state);
+ gst_rtsp_transport_free (ct);
return FALSE;
}
}
gst_sdp_message_add_attribute (sdp, "control", "*");
info.server_proto = proto;
- if (media->protocols & GST_RTSP_LOWER_TRANS_UDP_MCAST)
- info.server_ip = MCAST_ADDRESS;
- else
- info.server_ip = client->server_ip;
+ info.server_ip = g_strdup (client->server_ip);
/* create an SDP for the media object */
if (!gst_rtsp_sdp_from_media (sdp, &info, media))
goto no_sdp;
+ g_free (info.server_ip);
+
return sdp;
/* ERRORS */
no_sdp:
{
+ g_free (info.server_ip);
gst_sdp_message_free (sdp);
return NULL;
}
guint i, str_len;
gchar *str, *content_base;
GstRTSPMedia *media;
+ GstRTSPClientClass *klass;
+
+ klass = GST_RTSP_CLIENT_GET_CLASS (client);
/* check what kind of format is accepted, we don't really do anything with it
* and always return SDP for now. */
goto no_media;
/* create an SDP for the media object on this client */
- if (!(sdp = create_sdp (client, media)))
+ if (!(sdp = klass->create_sdp (client, media)))
goto no_sdp;
g_object_unref (media);
gst_rtsp_message_take_body (state->response, (guint8 *) str, strlen (str));
gst_sdp_message_free (sdp);
- send_response (client, state->session, state->response);
+ send_response (client, state->session, state->response, NULL);
+
+ g_signal_emit (client, gst_rtsp_client_signals[SIGNAL_DESCRIBE_REQUEST],
+ 0, state);
return TRUE;
gst_rtsp_message_add_header (state->response, GST_RTSP_HDR_PUBLIC, str);
g_free (str);
- send_response (client, state->session, state->response);
+ send_response (client, state->session, state->response, NULL);
+
+ g_signal_emit (client, gst_rtsp_client_signals[SIGNAL_OPTIONS_REQUEST],
+ 0, state);
return TRUE;
}
g_object_weak_ref (G_OBJECT (session), (GWeakNotify) client_session_finalized,
client);
client->sessions = g_list_prepend (client->sessions, session);
+
+ g_signal_emit (client, gst_rtsp_client_signals[SIGNAL_NEW_SESSION], 0,
+ session);
}
static void
state.session = session;
if (client->auth) {
- if (!gst_rtsp_auth_check (client->auth, client, &state))
+ if (!gst_rtsp_auth_check (client->auth, client, 0, &state))
goto not_authorized;
}
gst_rtsp_message_steal_body (message, &data, &size);
- buffer = gst_buffer_new ();
- GST_BUFFER_DATA (buffer) = data;
- GST_BUFFER_MALLOCDATA (buffer) = data;
- GST_BUFFER_SIZE (buffer) = size;
+ buffer = gst_buffer_new_wrapped (data, size);
handled = FALSE;
- for (walk = client->streams; walk; walk = g_list_next (walk)) {
- GstRTSPSessionStream *stream = (GstRTSPSessionStream *) walk->data;
- GstRTSPMediaStream *mstream;
+ for (walk = client->transports; walk; walk = g_list_next (walk)) {
+ GstRTSPStreamTransport *trans;
+ GstRTSPStream *stream;
GstRTSPTransport *tr;
- /* get the transport, if there is no transport configured, skip this stream */
- if (!(tr = stream->trans.transport))
- continue;
+ trans = walk->data;
- /* we also need a media stream */
- if (!(mstream = stream->media_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) {
/* dispatch to the stream based on the channel number */
if (tr->interleaved.min == channel) {
- gst_rtsp_media_stream_rtp (mstream, buffer);
+ gst_rtsp_stream_recv_rtp (stream, buffer);
handled = TRUE;
break;
} else if (tr->interleaved.max == channel) {
- gst_rtsp_media_stream_rtcp (mstream, buffer);
+ gst_rtsp_stream_recv_rtcp (stream, buffer);
handled = TRUE;
break;
}
*
* Get the #GstRTSPSessionPool object that @client uses to manage its sessions.
*
- * Returns: a #GstRTSPSessionPool, unref after usage.
+ * Returns: (transfer full): a #GstRTSPSessionPool, unref after usage.
*/
GstRTSPSessionPool *
gst_rtsp_client_get_session_pool (GstRTSPClient * client)
*
* Get the #GstRTSPServer object that @client was created from.
*
- * Returns: a #GstRTSPServer, unref after usage.
+ * Returns: (transfer full): a #GstRTSPServer, unref after usage.
*/
GstRTSPServer *
gst_rtsp_client_get_server (GstRTSPClient * client)
*
* Get the #GstRTSPMediaMapping object that @client uses to manage its sessions.
*
- * Returns: a #GstRTSPMediaMapping, unref after usage.
+ * Returns: (transfer full): a #GstRTSPMediaMapping, unref after usage.
*/
GstRTSPMediaMapping *
gst_rtsp_client_get_media_mapping (GstRTSPClient * client)
}
/**
+ * gst_rtsp_client_set_use_client_settings:
+ * @client: a #GstRTSPClient
+ * @use_client_settings: whether to use client settings for multicast
+ *
+ * Use client transport settings (destination and ttl) for multicast.
+ * When @use_client_settings is %FALSE, the server settings will be
+ * used.
+ */
+void
+gst_rtsp_client_set_use_client_settings (GstRTSPClient * client,
+ gboolean use_client_settings)
+{
+ client->use_client_settings = use_client_settings;
+}
+
+/**
+ * gst_rtsp_client_get_use_client_settings:
+ * @client: a #GstRTSPClient
+ *
+ * Check if client transport settings (destination and ttl) for multicast
+ * will be used.
+ */
+gboolean
+gst_rtsp_client_get_use_client_settings (GstRTSPClient * client)
+{
+ return client->use_client_settings;
+}
+
+/**
* gst_rtsp_client_set_auth:
* @client: a #GstRTSPClient
* @auth: a #GstRTSPAuth
*
* Get the #GstRTSPAuth used as the authentication manager of @client.
*
- * Returns: the #GstRTSPAuth of @client. g_object_unref() after
+ * Returns: (transfer full): the #GstRTSPAuth of @client. g_object_unref() after
* usage.
*/
GstRTSPAuth *
static GstRTSPResult
message_sent (GstRTSPWatch * watch, guint cseq, gpointer user_data)
{
- /* GstRTSPClient *client; */
-
- /* client = GST_RTSP_CLIENT (user_data); */
+ GstRTSPClient *client;
- /* GST_INFO ("client %p: sent a message with cseq %d", client, cseq); */
+ client = GST_RTSP_CLIENT (user_data);
+ if (client->teardown_response_seq && client->teardown_response_seq == cseq) {
+ client->teardown_response_seq = 0;
+ close_connection (client);
+ }
return GST_RTSP_OK;
}
GST_INFO ("client %p: connection closed", client);
if ((tunnelid = gst_rtsp_connection_get_tunnelid (client->connection))) {
- g_mutex_lock (tunnels_lock);
+ g_mutex_lock (&tunnels_lock);
/* remove from tunnelids */
g_hash_table_remove (tunnels, tunnelid);
- g_mutex_unlock (tunnels_lock);
+ g_mutex_unlock (&tunnels_lock);
}
return GST_RTSP_OK;
GST_INFO ("client %p: inserting tunnel session %s", client, tunnelid);
/* we can't have two clients connecting with the same tunnelid */
- g_mutex_lock (tunnels_lock);
+ g_mutex_lock (&tunnels_lock);
if (g_hash_table_lookup (tunnels, tunnelid))
goto tunnel_existed;
g_hash_table_insert (tunnels, g_strdup (tunnelid), g_object_ref (client));
- g_mutex_unlock (tunnels_lock);
+ g_mutex_unlock (&tunnels_lock);
return TRUE;
}
tunnel_existed:
{
- g_mutex_unlock (tunnels_lock);
+ g_mutex_unlock (&tunnels_lock);
GST_ERROR ("client %p: tunnel session %s already existed", client,
tunnelid);
return FALSE;
if (tunnelid == NULL)
goto no_tunnelid;
- g_mutex_lock (tunnels_lock);
+ g_mutex_lock (&tunnels_lock);
if (!(oclient = g_hash_table_lookup (tunnels, tunnelid)))
goto no_tunnel;
if (oclient->watch == NULL)
goto tunnel_closed;
- g_mutex_unlock (tunnels_lock);
+ g_mutex_unlock (&tunnels_lock);
GST_INFO ("client %p: found tunnel %p (old %p, new %p)", client, oclient,
oclient->connection, client->connection);
gst_rtsp_watch_reset (oclient->watch);
g_object_unref (oclient);
- /* we don't need this watch anymore */
- g_source_destroy ((GSource *) client->watch);
- client->watchid = 0;
- client->watch = NULL;
-
return GST_RTSP_OK;
/* ERRORS */
no_tunnelid:
{
GST_INFO ("client %p: no tunnelid provided", client);
- return GST_RTSP_STS_SERVICE_UNAVAILABLE;
+ return GST_RTSP_ERROR;
}
no_tunnel:
{
- g_mutex_unlock (tunnels_lock);
+ g_mutex_unlock (&tunnels_lock);
GST_INFO ("client %p: tunnel session %s not found", client, tunnelid);
- return GST_RTSP_STS_SERVICE_UNAVAILABLE;
+ return GST_RTSP_ERROR;
}
tunnel_closed:
{
- g_mutex_unlock (tunnels_lock);
+ g_mutex_unlock (&tunnels_lock);
GST_INFO ("client %p: tunnel session %s was closed", client, tunnelid);
g_object_unref (oclient);
- return GST_RTSP_STS_SERVICE_UNAVAILABLE;
+ return GST_RTSP_ERROR;
}
}
client_watch_notify (GstRTSPClient * client)
{
GST_INFO ("client %p: watch destroyed", client);
- client->watchid = 0;
client->watch = NULL;
g_signal_emit (client, gst_rtsp_client_signals[SIGNAL_CLOSED], 0, NULL);
g_object_unref (client);
}
-/**
- * gst_rtsp_client_attach:
- * @client: a #GstRTSPClient
- * @channel: a #GIOChannel
- *
- * Accept a new connection for @client on the socket in @channel.
- *
- * This function should be called when the client properties and urls are fully
- * configured and the client is ready to start.
- *
- * Returns: %TRUE if the client could be accepted.
- */
-gboolean
-gst_rtsp_client_accept (GstRTSPClient * client, GIOChannel * channel)
+static gboolean
+setup_client (GstRTSPClient * client, GSocket * socket,
+ GstRTSPConnection * conn, GError ** error)
{
- int sock, fd;
- GstRTSPConnection *conn;
- GstRTSPResult res;
- GSource *source;
- GMainContext *context;
+ GSocket *read_socket;
+ GSocketAddress *address;
GstRTSPUrl *url;
- struct sockaddr_storage addr;
- socklen_t addrlen;
- gchar ip[INET6_ADDRSTRLEN];
- /* a new client connected. */
- sock = g_io_channel_unix_get_fd (channel);
+ read_socket = gst_rtsp_connection_get_read_socket (conn);
+ client->is_ipv6 = g_socket_get_family (socket) == G_SOCKET_FAMILY_IPV6;
- GST_RTSP_CHECK (gst_rtsp_connection_accept (sock, &conn), accept_failed);
+ if (!(address = g_socket_get_remote_address (read_socket, error)))
+ goto no_address;
- fd = gst_rtsp_connection_get_readfd (conn);
-
- addrlen = sizeof (addr);
- if (getsockname (fd, (struct sockaddr *) &addr, &addrlen) < 0)
- goto getpeername_failed;
-
- client->is_ipv6 = addr.ss_family == AF_INET6;
+ g_free (client->server_ip);
+ /* keep the original ip that the client connected to */
+ if (G_IS_INET_SOCKET_ADDRESS (address)) {
+ GInetAddress *iaddr;
- if (getnameinfo ((struct sockaddr *) &addr, addrlen, ip, sizeof (ip), NULL, 0,
- NI_NUMERICHOST) != 0)
- goto getnameinfo_failed;
+ iaddr = g_inet_socket_address_get_address (G_INET_SOCKET_ADDRESS (address));
- /* keep the original ip that the client connected to */
- g_free (client->server_ip);
- client->server_ip = g_strndup (ip, sizeof (ip));
+ client->server_ip = g_inet_address_to_string (iaddr);
+ g_object_unref (address);
+ } else {
+ client->server_ip = g_strdup ("unknown");
+ }
GST_INFO ("client %p connected to server ip %s, ipv6 = %d", client,
client->server_ip, client->is_ipv6);
client->connection = conn;
- /* create watch for the connection and attach */
- client->watch = gst_rtsp_watch_new (client->connection, &watch_funcs,
- g_object_ref (client), (GDestroyNotify) client_watch_notify);
+ return TRUE;
- /* find the context to add the watch */
- if ((source = g_main_current_source ()))
- context = g_source_get_context (source);
- else
- context = NULL;
+ /* ERRORS */
+no_address:
+ {
+ GST_ERROR ("could not get remote address %s", (*error)->message);
+ return FALSE;
+ }
+}
- GST_INFO ("attaching to context %p", context);
+/**
+ * gst_rtsp_client_use_socket:
+ * @client: a #GstRTSPClient
+ * @socket: a #GSocket
+ * @ip: the IP address of the remote client
+ * @port: the port used by the other end
+ * @initial_buffer: any zero terminated initial data that was already read from
+ * the socket
+ * @error: a #GError
+ *
+ * Take an existing network socket and use it for an RTSP connection.
+ *
+ * Returns: %TRUE on success.
+ */
+gboolean
+gst_rtsp_client_use_socket (GstRTSPClient * client, GSocket * socket,
+ const gchar * ip, gint port, const gchar * initial_buffer, GError ** error)
+{
+ GstRTSPConnection *conn;
+ GstRTSPResult res;
- client->watchid = gst_rtsp_watch_attach (client->watch, context);
- gst_rtsp_watch_unref (client->watch);
+ GST_RTSP_CHECK (gst_rtsp_connection_create_from_socket (socket, ip, port,
+ initial_buffer, &conn), no_connection);
- return TRUE;
+ return setup_client (client, socket, conn, error);
/* ERRORS */
-accept_failed:
+no_connection:
{
gchar *str = gst_rtsp_strresult (res);
- GST_ERROR ("Could not accept client on server socket %d: %s", sock, str);
+ GST_ERROR ("could not create connection from socket %p: %s", socket, str);
g_free (str);
return FALSE;
}
-getpeername_failed:
- {
- GST_ERROR ("getpeername failed: %s", g_strerror (errno));
- return FALSE;
- }
-getnameinfo_failed:
+}
+
+/**
+ * gst_rtsp_client_accept:
+ * @client: a #GstRTSPClient
+ * @socket: a #GSocket
+ * @context: the context to run in
+ * @cancellable: a #GCancellable
+ * @error: a #GError
+ *
+ * Accept a new connection for @client on @socket.
+ *
+ * Returns: %TRUE if the client could be accepted.
+ */
+gboolean
+gst_rtsp_client_accept (GstRTSPClient * client, GSocket * socket,
+ GCancellable * cancellable, GError ** error)
+{
+ GstRTSPConnection *conn;
+ GstRTSPResult res;
+
+ /* a new client connected. */
+ GST_RTSP_CHECK (gst_rtsp_connection_accept (socket, &conn, cancellable),
+ accept_failed);
+
+ return setup_client (client, socket, conn, error);
+
+ /* ERRORS */
+accept_failed:
{
- GST_ERROR ("getnameinfo failed: %s", g_strerror (errno));
+ gchar *str = gst_rtsp_strresult (res);
+
+ GST_ERROR ("Could not accept client on server socket %p: %s", socket, str);
+ g_free (str);
return FALSE;
}
}
+
+/**
+ * gst_rtsp_client_attach:
+ * @client: a #GstRTSPClient
+ * @context: (allow-none): a #GMainContext
+ *
+ * Attaches @client to @context. When the mainloop for @context is run, the
+ * client will be dispatched. When @context is NULL, the default context will be
+ * used).
+ *
+ * This function should be called when the client properties and urls are fully
+ * configured and the client is ready to start.
+ *
+ * Returns: the ID (greater than 0) for the source within the GMainContext.
+ */
+guint
+gst_rtsp_client_attach (GstRTSPClient * client, GMainContext * context)
+{
+ guint res;
+
+ g_return_val_if_fail (GST_IS_RTSP_CLIENT (client), 0);
+ g_return_val_if_fail (client->watch == NULL, 0);
+
+ /* create watch for the connection and attach */
+ client->watch = gst_rtsp_watch_new (client->connection, &watch_funcs,
+ g_object_ref (client), (GDestroyNotify) client_watch_notify);
+
+ GST_INFO ("attaching to context %p", context);
+ res = gst_rtsp_watch_attach (client->watch, context);
+ gst_rtsp_watch_unref (client->watch);
+
+ return res;
+}
projects / platform / upstream / gstreamer.git / blobdiff