#include "rtsp-sdp.h"
#include "rtsp-params.h"
+#define GST_RTSP_CLIENT_GET_PRIVATE(obj) \
+ (G_TYPE_INSTANCE_GET_PRIVATE ((obj), GST_TYPE_RTSP_CLIENT, GstRTSPClientPrivate))
+
+/* locking order:
+ * send_lock, lock, tunnels_lock
+ */
+
+struct _GstRTSPClientPrivate
+{
+ GMutex lock; /* protects everything else */
+ GMutex send_lock;
+ GstRTSPConnection *connection;
+ GstRTSPWatch *watch;
+ guint close_seq;
+ gchar *server_ip;
+ gboolean is_ipv6;
+ gboolean use_client_settings;
+
+ GstRTSPClientSendFunc send_func; /* protected by send_lock */
+ gpointer send_data; /* protected by send_lock */
+ GDestroyNotify send_notify; /* protected by send_lock */
+
+ GstRTSPSessionPool *session_pool;
+ GstRTSPMountPoints *mount_points;
+ GstRTSPAuth *auth;
+
+ GstRTSPUrl *uri;
+ GstRTSPMedia *media;
+
+ GList *transports;
+ GList *sessions;
+};
+
static GMutex tunnels_lock;
-static GHashTable *tunnels;
+static GHashTable *tunnels; /* protected by tunnels_lock */
#define DEFAULT_SESSION_POOL NULL
#define DEFAULT_MOUNT_POINTS NULL
{
GObjectClass *gobject_class;
+ g_type_class_add_private (klass, sizeof (GstRTSPClientPrivate));
+
gobject_class = G_OBJECT_CLASS (klass);
gobject_class->get_property = gst_rtsp_client_get_property;
static void
gst_rtsp_client_init (GstRTSPClient * client)
{
- client->use_client_settings = DEFAULT_USE_CLIENT_SETTINGS;
- client->close_response_seq = 0;
+ GstRTSPClientPrivate *priv = GST_RTSP_CLIENT_GET_PRIVATE (client);
+
+ client->priv = priv;
+
+ g_mutex_init (&priv->lock);
+ g_mutex_init (&priv->send_lock);
+ priv->use_client_settings = DEFAULT_USE_CLIENT_SETTINGS;
+ priv->close_seq = 0;
+}
+
+static GstRTSPFilterResult
+filter_session (GstRTSPSession * sess, GstRTSPSessionMedia * media,
+ gpointer user_data)
+{
+ GstRTSPClient *client = GST_RTSP_CLIENT (user_data);
+
+ gst_rtsp_session_media_set_state (media, GST_STATE_NULL);
+ unlink_session_transports (client, sess, media);
+
+ /* unmanage the media in the session */
+ return GST_RTSP_FILTER_REMOVE;
}
static void
client_unlink_session (GstRTSPClient * client, GstRTSPSession * session)
{
/* unlink all media managed in this session */
- while (session->medias) {
- GstRTSPSessionMedia *media = session->medias->data;
-
- gst_rtsp_session_media_set_state (media, GST_STATE_NULL);
- 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_rtsp_session_filter (session, filter_session, client);
}
static void
client_cleanup_sessions (GstRTSPClient * client)
{
+ GstRTSPClientPrivate *priv = client->priv;
GList *sessions;
/* remove weak-ref from sessions */
- for (sessions = client->sessions; sessions; sessions = g_list_next (sessions)) {
+ for (sessions = priv->sessions; sessions; sessions = g_list_next (sessions)) {
GstRTSPSession *session = (GstRTSPSession *) sessions->data;
g_object_weak_unref (G_OBJECT (session),
(GWeakNotify) client_session_finalized, client);
client_unlink_session (client, session);
}
- g_list_free (client->sessions);
- client->sessions = NULL;
+ g_list_free (priv->sessions);
+ priv->sessions = NULL;
}
/* A client is finalized when the connection is broken */
gst_rtsp_client_finalize (GObject * obj)
{
GstRTSPClient *client = GST_RTSP_CLIENT (obj);
+ GstRTSPClientPrivate *priv = client->priv;
GST_INFO ("finalize client %p", client);
- if (client->watch)
- g_source_destroy ((GSource *) client->watch);
+ gst_rtsp_client_set_send_func (client, NULL, NULL, NULL);
+
+ if (priv->watch)
+ g_source_destroy ((GSource *) priv->watch);
client_cleanup_sessions (client);
- gst_rtsp_connection_free (client->connection);
- if (client->session_pool)
- g_object_unref (client->session_pool);
- if (client->mount_points)
- g_object_unref (client->mount_points);
- if (client->auth)
- g_object_unref (client->auth);
+ if (priv->connection)
+ gst_rtsp_connection_free (priv->connection);
+ if (priv->session_pool)
+ g_object_unref (priv->session_pool);
+ if (priv->mount_points)
+ g_object_unref (priv->mount_points);
+ if (priv->auth)
+ g_object_unref (priv->auth);
- if (client->uri)
- gst_rtsp_url_free (client->uri);
- if (client->media) {
- gst_rtsp_media_unprepare (client->media);
- g_object_unref (client->media);
+ if (priv->uri)
+ gst_rtsp_url_free (priv->uri);
+ if (priv->media) {
+ gst_rtsp_media_unprepare (priv->media);
+ g_object_unref (priv->media);
}
- g_free (client->server_ip);
+ g_free (priv->server_ip);
+ g_mutex_clear (&priv->lock);
+ g_mutex_clear (&priv->send_lock);
G_OBJECT_CLASS (gst_rtsp_client_parent_class)->finalize (obj);
}
send_response (GstRTSPClient * client, GstRTSPSession * session,
GstRTSPMessage * response, gboolean close)
{
+ GstRTSPClientPrivate *priv = client->priv;
+
gst_rtsp_message_add_header (response, GST_RTSP_HDR_SERVER,
"GStreamer RTSP server");
gst_rtsp_message_dump (response);
}
- if (close) {
- gst_rtsp_message_add_header (state->response, GST_RTSP_HDR_CONNECTION,
- "close");
- }
- /* send the response and store the seq number so we can wait until it's
- * written to the client to close the connection */
- gst_rtsp_watch_send_message (client->watch, response, close ?
- &client->close_response_seq : NULL);
+ if (close)
+ gst_rtsp_message_add_header (response, GST_RTSP_HDR_CONNECTION, "close");
+
+ g_mutex_lock (&priv->send_lock);
+ if (priv->send_func)
+ priv->send_func (client, response, close, priv->send_data);
+ g_mutex_unlock (&priv->send_lock);
+
gst_rtsp_message_unset (response);
}
static GstRTSPMedia *
find_media (GstRTSPClient * client, GstRTSPClientState * state)
{
+ GstRTSPClientPrivate *priv = client->priv;
GstRTSPMediaFactory *factory;
GstRTSPMedia *media;
GstRTSPAuth *auth;
- if (!compare_uri (client->uri, state->uri)) {
+ if (!compare_uri (priv->uri, state->uri)) {
/* remove any previously cached values before we try to construct a new
* media for uri */
- if (client->uri)
- gst_rtsp_url_free (client->uri);
- client->uri = NULL;
- if (client->media) {
- gst_rtsp_media_unprepare (client->media);
- g_object_unref (client->media);
+ if (priv->uri)
+ gst_rtsp_url_free (priv->uri);
+ priv->uri = NULL;
+ if (priv->media) {
+ gst_rtsp_media_unprepare (priv->media);
+ g_object_unref (priv->media);
}
- client->media = NULL;
+ priv->media = NULL;
- if (!client->mount_points)
+ if (!priv->mount_points)
goto no_mount_points;
/* find the factory for the uri first */
if (!(factory =
- gst_rtsp_mount_points_find_factory (client->mount_points,
+ gst_rtsp_mount_points_find_factory (priv->mount_points,
state->uri)))
goto no_factory;
- state->factory = factory;
-
/* check if we have access to the factory */
if ((auth = gst_rtsp_media_factory_get_auth (factory))) {
+ state->factory = factory;
+
if (!gst_rtsp_auth_check (auth, client, 0, state))
goto not_allowed;
+ state->factory = NULL;
g_object_unref (auth);
}
g_object_unref (factory);
factory = NULL;
- state->factory = NULL;
-
- /* set ipv6 on the media before preparing */
- media->is_ipv6 = client->is_ipv6;
- state->media = media;
/* prepare the media */
if (!(gst_rtsp_media_prepare (media)))
goto no_prepare;
/* now keep track of the uri and the media */
- client->uri = gst_rtsp_url_copy (state->uri);
- client->media = media;
+ priv->uri = gst_rtsp_url_copy (state->uri);
+ priv->media = media;
+ state->media = media;
} else {
/* we have seen this uri before, used cached media */
- media = client->media;
+ media = priv->media;
state->media = media;
GST_INFO ("reusing cached media %p", media);
}
/* ERRORS */
no_mount_points:
{
+ GST_ERROR ("client %p: no mount points configured", client);
send_generic_response (client, GST_RTSP_STS_NOT_FOUND, state);
return NULL;
}
no_factory:
{
+ GST_ERROR ("client %p: no factory for uri", client);
send_generic_response (client, GST_RTSP_STS_NOT_FOUND, state);
return NULL;
}
not_allowed:
{
+ GST_ERROR ("client %p: unauthorized request", client);
handle_unauthorized_request (client, auth, state);
g_object_unref (factory);
+ state->factory = NULL;
g_object_unref (auth);
return NULL;
}
no_media:
{
+ GST_ERROR ("client %p: can't create media", client);
send_generic_response (client, GST_RTSP_STS_SERVICE_UNAVAILABLE, state);
g_object_unref (factory);
return NULL;
}
no_prepare:
{
+ GST_ERROR ("client %p: can't prepare media", client);
send_generic_response (client, GST_RTSP_STS_SERVICE_UNAVAILABLE, state);
g_object_unref (media);
return NULL;
static gboolean
do_send_data (GstBuffer * buffer, guint8 channel, GstRTSPClient * client)
{
+ GstRTSPClientPrivate *priv = client->priv;
GstRTSPMessage message = { 0 };
GstMapInfo map_info;
guint8 *data;
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);
+ g_mutex_lock (&priv->send_lock);
+ if (priv->send_func)
+ priv->send_func (client, &message, FALSE, priv->send_data);
+ g_mutex_unlock (&priv->send_lock);
gst_rtsp_message_steal_body (&message, &data, &usize);
gst_buffer_unmap (buffer, &map_info);
link_transport (GstRTSPClient * client, GstRTSPSession * session,
GstRTSPStreamTransport * trans)
{
+ GstRTSPClientPrivate *priv = client->priv;
+
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);
- client->transports = g_list_prepend (client->transports, trans);
+ priv->transports = g_list_prepend (priv->transports, trans);
/* make sure our session can't expire */
gst_rtsp_session_prevent_expire (session);
unlink_transport (GstRTSPClient * client, GstRTSPSession * session,
GstRTSPStreamTransport * trans)
{
+ GstRTSPClientPrivate *priv = client->priv;
+
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);
+ priv->transports = g_list_remove (priv->transports, trans);
/* our session can now expire */
gst_rtsp_session_allow_expire (session);
{
guint n_streams, i;
- n_streams = gst_rtsp_media_n_streams (media->media);
+ n_streams =
+ gst_rtsp_media_n_streams (gst_rtsp_session_media_get_media (media));
for (i = 0; i < n_streams; i++) {
GstRTSPStreamTransport *trans;
- GstRTSPTransport *tr;
+ const GstRTSPTransport *tr;
/* get the transport, if there is no transport configured, skip this stream */
trans = gst_rtsp_session_media_get_transport (media, i);
if (trans == NULL)
continue;
- tr = trans->transport;
+ tr = gst_rtsp_stream_transport_get_transport (trans);
if (tr->lower_transport == GST_RTSP_LOWER_TRANS_TCP) {
/* for TCP, unlink the stream from the TCP connection of the client */
static void
close_connection (GstRTSPClient * client)
{
+ GstRTSPClientPrivate *priv = client->priv;
const gchar *tunnelid;
GST_DEBUG ("client %p: closing connection", client);
- if ((tunnelid = gst_rtsp_connection_get_tunnelid (client->connection))) {
+ if ((tunnelid = gst_rtsp_connection_get_tunnelid (priv->connection))) {
g_mutex_lock (&tunnels_lock);
/* remove from tunnelids */
g_hash_table_remove (tunnels, tunnelid);
g_mutex_unlock (&tunnels_lock);
}
- gst_rtsp_connection_close (client->connection);
+ gst_rtsp_connection_close (priv->connection);
}
static gboolean
handle_teardown_request (GstRTSPClient * client, GstRTSPClientState * state)
{
+ GstRTSPClientPrivate *priv = client->priv;
GstRTSPSession *session;
GstRTSPSessionMedia *media;
GstRTSPStatusCode code;
/* remove the session from the watched sessions */
g_object_weak_unref (G_OBJECT (session),
(GWeakNotify) client_session_finalized, client);
- client->sessions = g_list_remove (client->sessions, session);
+ priv->sessions = g_list_remove (priv->sessions, session);
gst_rtsp_session_media_set_state (media, GST_STATE_NULL);
* are torn down. */
if (!gst_rtsp_session_release_media (session, media)) {
/* remove the session */
- gst_rtsp_session_pool_remove (client->session_pool, session);
+ gst_rtsp_session_pool_remove (priv->session_pool, session);
}
/* construct the response now */
code = GST_RTSP_STS_OK;
/* ERRORS */
no_session:
{
+ GST_ERROR ("client %p: no session", client);
send_generic_response (client, GST_RTSP_STS_SESSION_NOT_FOUND, state);
return FALSE;
}
not_found:
{
+ GST_ERROR ("client %p: no media for uri", client);
send_generic_response (client, GST_RTSP_STS_NOT_FOUND, state);
return FALSE;
}
/* ERRORS */
bad_request:
{
+ GST_ERROR ("client %p: bad request", client);
send_generic_response (client, GST_RTSP_STS_BAD_REQUEST, state);
return FALSE;
}
/* ERRORS */
bad_request:
{
+ GST_ERROR ("client %p: bad request", client);
send_generic_response (client, GST_RTSP_STS_BAD_REQUEST, state);
return FALSE;
}
GstRTSPSession *session;
GstRTSPSessionMedia *media;
GstRTSPStatusCode code;
+ GstRTSPState rtspstate;
if (!(session = state->session))
goto no_session;
state->sessmedia = media;
+ rtspstate = gst_rtsp_session_media_get_rtsp_state (media);
/* the session state must be playing or recording */
- if (media->state != GST_RTSP_STATE_PLAYING &&
- media->state != GST_RTSP_STATE_RECORDING)
+ if (rtspstate != GST_RTSP_STATE_PLAYING &&
+ rtspstate != GST_RTSP_STATE_RECORDING)
goto invalid_state;
/* unlink the all TCP callbacks */
send_response (client, session, state->response, FALSE);
/* the state is now READY */
- media->state = GST_RTSP_STATE_READY;
+ gst_rtsp_session_media_set_rtsp_state (media, GST_RTSP_STATE_READY);
g_signal_emit (client, gst_rtsp_client_signals[SIGNAL_PAUSE_REQUEST],
0, state);
/* ERRORS */
no_session:
{
+ GST_ERROR ("client %p: no seesion", client);
send_generic_response (client, GST_RTSP_STS_SESSION_NOT_FOUND, state);
return FALSE;
}
not_found:
{
+ GST_ERROR ("client %p: no media for uri", client);
send_generic_response (client, GST_RTSP_STS_NOT_FOUND, state);
return FALSE;
}
invalid_state:
{
+ GST_ERROR ("client %p: not PLAYING or RECORDING", client);
send_generic_response (client, GST_RTSP_STS_METHOD_NOT_VALID_IN_THIS_STATE,
state);
return FALSE;
gchar *str;
GstRTSPTimeRange *range;
GstRTSPResult res;
+ GstRTSPState rtspstate;
+ GstRTSPRangeUnit unit = GST_RTSP_RANGE_NPT;
if (!(session = state->session))
goto no_session;
state->sessmedia = media;
/* the session state must be playing or ready */
- if (media->state != GST_RTSP_STATE_PLAYING &&
- media->state != GST_RTSP_STATE_READY)
+ rtspstate = gst_rtsp_session_media_get_rtsp_state (media);
+ if (rtspstate != GST_RTSP_STATE_PLAYING && rtspstate != GST_RTSP_STATE_READY)
goto invalid_state;
/* parse the range header if we have one */
if (res == GST_RTSP_OK) {
if (gst_rtsp_range_parse (str, &range) == GST_RTSP_OK) {
/* we have a range, seek to the position */
- gst_rtsp_media_seek (media->media, range);
+ gst_rtsp_media_seek (gst_rtsp_session_media_get_media (media), range);
+ unit = range->unit;
gst_rtsp_range_free (range);
}
}
/* grab RTPInfo from the payloaders now */
rtpinfo = g_string_new ("");
- n_streams = gst_rtsp_media_n_streams (media->media);
+ n_streams =
+ gst_rtsp_media_n_streams (gst_rtsp_session_media_get_media (media));
for (i = 0, infocount = 0; i < n_streams; i++) {
GstRTSPStreamTransport *trans;
- GstRTSPTransport *tr;
+ GstRTSPStream *stream;
+ const GstRTSPTransport *tr;
gchar *uristr;
guint rtptime, seq;
GST_INFO ("stream %d is not configured", i);
continue;
}
- tr = trans->transport;
+ tr = gst_rtsp_stream_transport_get_transport (trans);
if (tr->lower_transport == GST_RTSP_LOWER_TRANS_TCP) {
/* for TCP, link the stream to the TCP connection of the client */
link_transport (client, session, trans);
}
- if (gst_rtsp_stream_get_rtpinfo (trans->stream, &rtptime, &seq)) {
+ stream = gst_rtsp_stream_transport_get_stream (trans);
+ if (gst_rtsp_stream_get_rtpinfo (stream, &rtptime, &seq)) {
if (infocount > 0)
g_string_append (rtpinfo, ", ");
}
/* add the range */
- str = gst_rtsp_media_get_range_string (media->media, TRUE);
+ str =
+ gst_rtsp_media_get_range_string (gst_rtsp_session_media_get_media (media),
+ TRUE, unit);
gst_rtsp_message_take_header (state->response, GST_RTSP_HDR_RANGE, str);
send_response (client, session, state->response, FALSE);
/* start playing after sending the request */
gst_rtsp_session_media_set_state (media, GST_STATE_PLAYING);
- media->state = GST_RTSP_STATE_PLAYING;
+ gst_rtsp_session_media_set_rtsp_state (media, GST_RTSP_STATE_PLAYING);
g_signal_emit (client, gst_rtsp_client_signals[SIGNAL_PLAY_REQUEST],
0, state);
/* ERRORS */
no_session:
{
+ GST_ERROR ("client %p: no session", client);
send_generic_response (client, GST_RTSP_STS_SESSION_NOT_FOUND, state);
return FALSE;
}
not_found:
{
+ GST_ERROR ("client %p: media not found", client);
send_generic_response (client, GST_RTSP_STS_NOT_FOUND, state);
return FALSE;
}
invalid_state:
{
+ GST_ERROR ("client %p: not PLAYING or READY", client);
send_generic_response (client, GST_RTSP_STS_METHOD_NOT_VALID_IN_THIS_STATE,
state);
return FALSE;
configure_client_transport (GstRTSPClient * client, GstRTSPClientState * state,
GstRTSPTransport * ct)
{
+ GstRTSPClientPrivate *priv = client->priv;
+
/* 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) {
+ if (ct->destination && priv->use_client_settings) {
+ GstRTSPAddress *addr;
+
+ addr = gst_rtsp_stream_reserve_address (state->stream, ct->destination,
+ ct->port.min, ct->port.max - ct->port.min + 1, ct->ttl);
+
+ if (addr == NULL)
+ goto no_address;
+
+ gst_rtsp_address_free (addr);
+ } else {
GstRTSPAddress *addr;
addr = gst_rtsp_stream_get_address (state->stream);
ct->port.min = addr->port;
ct->port.max = addr->port + addr->n_ports - 1;
ct->ttl = addr->ttl;
+
+ gst_rtsp_address_free (addr);
}
} else {
GstRTSPUrl *url;
- url = gst_rtsp_connection_get_url (client->connection);
+ url = gst_rtsp_connection_get_url (priv->connection);
g_free (ct->destination);
ct->destination = g_strdup (url->host);
switch (st->lower_transport) {
case GST_RTSP_LOWER_TRANS_UDP:
st->client_port = ct->client_port;
- st->server_port = state->stream->server_port;
+ gst_rtsp_stream_get_server_port (state->stream, &st->server_port);
break;
case GST_RTSP_LOWER_TRANS_UDP_MCAST:
st->port = ct->port;
break;
}
- if (state->stream->session)
- g_object_get (state->stream->session, "internal-ssrc", &st->ssrc, NULL);
+ gst_rtsp_stream_get_ssrc (state->stream, &st->ssrc);
return st;
}
static gboolean
handle_setup_request (GstRTSPClient * client, GstRTSPClientState * state)
{
+ GstRTSPClientPrivate *priv = client->priv;
GstRTSPResult res;
GstRTSPUrl *uri;
gchar *transport;
GstRTSPSessionMedia *sessmedia;
GstRTSPMedia *media;
GstRTSPStream *stream;
+ GstRTSPState rtspstate;
uri = state->uri;
/* we create the session after parsing stuff so that we don't make
* a session for malformed requests */
- if (client->session_pool == NULL)
+ if (priv->session_pool == NULL)
goto no_pool;
session = state->session;
} else {
/* create a session if this fails we probably reached our session limit or
* something. */
- if (!(session = gst_rtsp_session_pool_create (client->session_pool)))
+ if (!(session = gst_rtsp_session_pool_create (priv->session_pool)))
goto service_unavailable;
state->session = session;
goto not_found;
state->sessmedia = sessmedia;
- state->media = media = sessmedia->media;
+ state->media = media = gst_rtsp_session_media_get_media (sessmedia);
/* now get the stream */
stream = gst_rtsp_media_get_stream (media, streamid);
send_response (client, session, state->response, FALSE);
/* update the state */
- switch (sessmedia->state) {
+ rtspstate = gst_rtsp_session_media_get_rtsp_state (sessmedia);
+ switch (rtspstate) {
case GST_RTSP_STATE_PLAYING:
case GST_RTSP_STATE_RECORDING:
case GST_RTSP_STATE_READY:
/* no state change */
break;
default:
- sessmedia->state = GST_RTSP_STATE_READY;
+ gst_rtsp_session_media_set_rtsp_state (sessmedia, GST_RTSP_STATE_READY);
break;
}
g_object_unref (session);
/* ERRORS */
bad_request:
{
+ GST_ERROR ("client %p: bad request", client);
send_generic_response (client, GST_RTSP_STS_BAD_REQUEST, state);
return FALSE;
}
not_found:
{
+ GST_ERROR ("client %p: media not found", client);
send_generic_response (client, GST_RTSP_STS_NOT_FOUND, state);
g_object_unref (session);
gst_rtsp_transport_free (ct);
}
invalid_blocksize:
{
+ GST_ERROR ("client %p: invalid blocksize", client);
send_generic_response (client, GST_RTSP_STS_BAD_REQUEST, state);
g_object_unref (session);
gst_rtsp_transport_free (ct);
}
unsupported_client_transport:
{
+ GST_ERROR ("client %p: unsupported client transport", client);
send_generic_response (client, GST_RTSP_STS_UNSUPPORTED_TRANSPORT, state);
g_object_unref (session);
gst_rtsp_transport_free (ct);
}
no_transport:
{
+ GST_ERROR ("client %p: no transport", client);
send_generic_response (client, GST_RTSP_STS_UNSUPPORTED_TRANSPORT, state);
return FALSE;
}
unsupported_transports:
{
+ GST_ERROR ("client %p: unsupported transports", client);
send_generic_response (client, GST_RTSP_STS_UNSUPPORTED_TRANSPORT, state);
gst_rtsp_transport_free (ct);
return FALSE;
}
no_pool:
{
- send_generic_response (client, GST_RTSP_STS_SERVICE_UNAVAILABLE, state);
+ GST_ERROR ("client %p: no session pool configured", client);
+ send_generic_response (client, GST_RTSP_STS_SESSION_NOT_FOUND, state);
gst_rtsp_transport_free (ct);
return FALSE;
}
service_unavailable:
{
+ GST_ERROR ("client %p: can't create session", client);
send_generic_response (client, GST_RTSP_STS_SERVICE_UNAVAILABLE, state);
gst_rtsp_transport_free (ct);
return FALSE;
static GstSDPMessage *
create_sdp (GstRTSPClient * client, GstRTSPMedia * media)
{
+ GstRTSPClientPrivate *priv = client->priv;
GstSDPMessage *sdp;
GstSDPInfo info;
const gchar *proto;
/* some standard things first */
gst_sdp_message_set_version (sdp, "0");
- if (client->is_ipv6)
+ if (priv->is_ipv6)
proto = "IP6";
else
proto = "IP4";
gst_sdp_message_set_origin (sdp, "-", "1188340656180883", "1", "IN", proto,
- client->server_ip);
+ priv->server_ip);
gst_sdp_message_set_session_name (sdp, "Session streamed with GStreamer");
gst_sdp_message_set_information (sdp, "rtsp-server");
gst_sdp_message_add_attribute (sdp, "control", "*");
info.server_proto = proto;
- info.server_ip = g_strdup (client->server_ip);
+ info.server_ip = g_strdup (priv->server_ip);
/* create an SDP for the media object */
if (!gst_rtsp_sdp_from_media (sdp, &info, media))
/* ERRORS */
no_sdp:
{
+ GST_ERROR ("client %p: could not create SDP", client);
g_free (info.server_ip);
gst_sdp_message_free (sdp);
return NULL;
/* ERRORS */
no_media:
{
+ GST_ERROR ("client %p: no media", client);
/* error reply is already sent */
return FALSE;
}
no_sdp:
{
+ GST_ERROR ("client %p: can't create SDP", client);
send_generic_response (client, GST_RTSP_STS_SERVICE_UNAVAILABLE, state);
g_object_unref (media);
return FALSE;
static void
client_session_finalized (GstRTSPClient * client, GstRTSPSession * session)
{
+ GstRTSPClientPrivate *priv = client->priv;
+
GST_INFO ("client %p: session %p finished", client, session);
/* unlink all media managed in this session */
client_unlink_session (client, session);
/* remove the session */
- if (!(client->sessions = g_list_remove (client->sessions, session))) {
+ if (!(priv->sessions = g_list_remove (priv->sessions, session))) {
GST_INFO ("client %p: all sessions finalized, close the connection",
client);
close_connection (client);
static void
client_watch_session (GstRTSPClient * client, GstRTSPSession * session)
{
+ GstRTSPClientPrivate *priv = client->priv;
GList *walk;
- for (walk = client->sessions; walk; walk = g_list_next (walk)) {
+ for (walk = priv->sessions; walk; walk = g_list_next (walk)) {
GstRTSPSession *msession = (GstRTSPSession *) walk->data;
/* we already know about this session */
g_object_weak_ref (G_OBJECT (session), (GWeakNotify) client_session_finalized,
client);
- client->sessions = g_list_prepend (client->sessions, session);
+ priv->sessions = g_list_prepend (priv->sessions, session);
g_signal_emit (client, gst_rtsp_client_signals[SIGNAL_NEW_SESSION], 0,
session);
static void
handle_request (GstRTSPClient * client, GstRTSPMessage * request)
{
+ GstRTSPClientPrivate *priv = client->priv;
GstRTSPMethod method;
const gchar *uristr;
- GstRTSPUrl *uri;
+ GstRTSPUrl *uri = NULL;
GstRTSPVersion version;
GstRTSPResult res;
- GstRTSPSession *session;
+ GstRTSPSession *session = NULL;
GstRTSPClientState state = { NULL };
GstRTSPMessage response = { 0 };
gchar *sessid;
gst_rtsp_message_parse_request (request, &method, &uristr, &version);
- if (version != GST_RTSP_VERSION_1_0) {
- /* we can only handle 1.0 requests */
- send_generic_response (client, GST_RTSP_STS_RTSP_VERSION_NOT_SUPPORTED,
- &state);
- return;
- }
+ /* we can only handle 1.0 requests */
+ if (version != GST_RTSP_VERSION_1_0)
+ goto not_supported;
+
state.method = method;
/* we always try to parse the url first */
- if (gst_rtsp_url_parse (uristr, &uri) != GST_RTSP_OK) {
- send_generic_response (client, GST_RTSP_STS_BAD_REQUEST, &state);
- return;
- }
-
- /* sanitize the uri */
- sanitize_uri (uri);
- state.uri = uri;
+ if (gst_rtsp_url_parse (uristr, &uri) != GST_RTSP_OK)
+ goto bad_request;
/* get the session if there is any */
res = gst_rtsp_message_get_header (request, GST_RTSP_HDR_SESSION, &sessid, 0);
if (res == GST_RTSP_OK) {
- if (client->session_pool == NULL)
+ if (priv->session_pool == NULL)
goto no_pool;
/* we had a session in the request, find it again */
- if (!(session = gst_rtsp_session_pool_find (client->session_pool, sessid)))
+ if (!(session = gst_rtsp_session_pool_find (priv->session_pool, sessid)))
goto session_not_found;
/* we add the session to the client list of watched sessions. When a session
* disappears because it times out, we will be notified. If all sessions are
* gone, we will close the connection */
client_watch_session (client, session);
- } else
- session = NULL;
+ }
+ /* sanitize the uri */
+ sanitize_uri (uri);
+ state.uri = uri;
state.session = session;
- if (client->auth) {
- if (!gst_rtsp_auth_check (client->auth, client, 0, &state))
+ if (priv->auth) {
+ if (!gst_rtsp_auth_check (priv->auth, client, 0, &state))
goto not_authorized;
}
case GST_RTSP_ANNOUNCE:
case GST_RTSP_RECORD:
case GST_RTSP_REDIRECT:
- send_generic_response (client, GST_RTSP_STS_NOT_IMPLEMENTED, &state);
- break;
+ goto not_implemented;
case GST_RTSP_INVALID:
default:
- send_generic_response (client, GST_RTSP_STS_BAD_REQUEST, &state);
- break;
+ goto bad_request;
}
+
+done:
if (session)
g_object_unref (session);
-
- gst_rtsp_url_free (uri);
+ if (uri)
+ gst_rtsp_url_free (uri);
return;
/* ERRORS */
+not_supported:
+ {
+ GST_ERROR ("client %p: version %d not supported", client, version);
+ send_generic_response (client, GST_RTSP_STS_RTSP_VERSION_NOT_SUPPORTED,
+ &state);
+ goto done;
+ }
+bad_request:
+ {
+ GST_ERROR ("client %p: bad request", client);
+ send_generic_response (client, GST_RTSP_STS_BAD_REQUEST, &state);
+ goto done;
+ }
no_pool:
{
- send_generic_response (client, GST_RTSP_STS_SERVICE_UNAVAILABLE, &state);
- return;
+ GST_ERROR ("client %p: no pool configured", client);
+ send_generic_response (client, GST_RTSP_STS_SESSION_NOT_FOUND, &state);
+ goto done;
}
session_not_found:
{
+ GST_ERROR ("client %p: session not found", client);
send_generic_response (client, GST_RTSP_STS_SESSION_NOT_FOUND, &state);
- return;
+ goto done;
}
not_authorized:
{
- handle_unauthorized_request (client, client->auth, &state);
- return;
+ GST_ERROR ("client %p: not allowed", client);
+ handle_unauthorized_request (client, priv->auth, &state);
+ goto done;
+ }
+not_implemented:
+ {
+ GST_ERROR ("client %p: method %d not implemented", client, method);
+ send_generic_response (client, GST_RTSP_STS_NOT_IMPLEMENTED, &state);
+ goto done;
}
}
static void
handle_data (GstRTSPClient * client, GstRTSPMessage * message)
{
+ GstRTSPClientPrivate *priv = client->priv;
GstRTSPResult res;
guint8 channel;
GList *walk;
buffer = gst_buffer_new_wrapped (data, size);
handled = FALSE;
- for (walk = client->transports; walk; walk = g_list_next (walk)) {
+ for (walk = priv->transports; walk; walk = g_list_next (walk)) {
GstRTSPStreamTransport *trans;
GstRTSPStream *stream;
- GstRTSPTransport *tr;
+ const GstRTSPTransport *tr;
trans = walk->data;
- /* we only add clients with a transport to the list */
- tr = trans->transport;
- stream = trans->stream;
+ tr = gst_rtsp_stream_transport_get_transport (trans);
+ stream = gst_rtsp_stream_transport_get_stream (trans);
/* check for TCP transport */
if (tr->lower_transport == GST_RTSP_LOWER_TRANS_TCP) {
GstRTSPSessionPool * pool)
{
GstRTSPSessionPool *old;
+ GstRTSPClientPrivate *priv;
- old = client->session_pool;
- if (old != pool) {
- if (pool)
- g_object_ref (pool);
- client->session_pool = pool;
- if (old)
- g_object_unref (old);
- }
+ g_return_if_fail (GST_IS_RTSP_CLIENT (client));
+
+ priv = client->priv;
+
+ if (pool)
+ g_object_ref (pool);
+
+ g_mutex_lock (&priv->lock);
+ old = priv->session_pool;
+ priv->session_pool = pool;
+ g_mutex_unlock (&priv->lock);
+
+ if (old)
+ g_object_unref (old);
}
/**
GstRTSPSessionPool *
gst_rtsp_client_get_session_pool (GstRTSPClient * client)
{
+ GstRTSPClientPrivate *priv;
GstRTSPSessionPool *result;
- if ((result = client->session_pool))
- g_object_ref (result);
-
- return result;
-}
-
-/**
- * gst_rtsp_client_set_server:
- * @client: a #GstRTSPClient
- * @server: a #GstRTSPServer
- *
- * Set @server as the server that created @client.
- */
-void
-gst_rtsp_client_set_server (GstRTSPClient * client, GstRTSPServer * server)
-{
- GstRTSPServer *old;
-
- old = client->server;
- if (old != server) {
- if (server)
- g_object_ref (server);
- client->server = server;
- if (old)
- g_object_unref (old);
- }
-}
+ g_return_val_if_fail (GST_IS_RTSP_CLIENT (client), NULL);
-/**
- * gst_rtsp_client_get_server:
- * @client: a #GstRTSPClient
- *
- * Get the #GstRTSPServer object that @client was created from.
- *
- * Returns: (transfer full): a #GstRTSPServer, unref after usage.
- */
-GstRTSPServer *
-gst_rtsp_client_get_server (GstRTSPClient * client)
-{
- GstRTSPServer *result;
+ priv = client->priv;
- if ((result = client->server))
+ g_mutex_lock (&priv->lock);
+ if ((result = priv->session_pool))
g_object_ref (result);
+ g_mutex_unlock (&priv->lock);
return result;
}
gst_rtsp_client_set_mount_points (GstRTSPClient * client,
GstRTSPMountPoints * mounts)
{
+ GstRTSPClientPrivate *priv;
GstRTSPMountPoints *old;
- old = client->mount_points;
+ g_return_if_fail (GST_IS_RTSP_CLIENT (client));
- if (old != mounts) {
- if (mounts)
- g_object_ref (mounts);
- client->mount_points = mounts;
- if (old)
- g_object_unref (old);
- }
+ priv = client->priv;
+
+ if (mounts)
+ g_object_ref (mounts);
+
+ g_mutex_lock (&priv->lock);
+ old = priv->mount_points;
+ priv->mount_points = mounts;
+ g_mutex_unlock (&priv->lock);
+
+ if (old)
+ g_object_unref (old);
}
/**
GstRTSPMountPoints *
gst_rtsp_client_get_mount_points (GstRTSPClient * client)
{
+ GstRTSPClientPrivate *priv;
GstRTSPMountPoints *result;
- if ((result = client->mount_points))
+ g_return_val_if_fail (GST_IS_RTSP_CLIENT (client), NULL);
+
+ priv = client->priv;
+
+ g_mutex_lock (&priv->lock);
+ if ((result = priv->mount_points))
g_object_ref (result);
+ g_mutex_unlock (&priv->lock);
return result;
}
gst_rtsp_client_set_use_client_settings (GstRTSPClient * client,
gboolean use_client_settings)
{
- client->use_client_settings = use_client_settings;
+ GstRTSPClientPrivate *priv;
+
+ g_return_if_fail (GST_IS_RTSP_CLIENT (client));
+
+ priv = client->priv;
+
+ g_mutex_lock (&priv->lock);
+ priv->use_client_settings = use_client_settings;
+ g_mutex_unlock (&priv->lock);
}
/**
gboolean
gst_rtsp_client_get_use_client_settings (GstRTSPClient * client)
{
- return client->use_client_settings;
+ GstRTSPClientPrivate *priv;
+ gboolean res;
+
+ g_return_val_if_fail (GST_IS_RTSP_CLIENT (client), FALSE);
+
+ priv = client->priv;
+
+ g_mutex_lock (&priv->lock);
+ res = priv->use_client_settings;
+ g_mutex_unlock (&priv->lock);
+
+ return res;
}
/**
void
gst_rtsp_client_set_auth (GstRTSPClient * client, GstRTSPAuth * auth)
{
+ GstRTSPClientPrivate *priv;
GstRTSPAuth *old;
g_return_if_fail (GST_IS_RTSP_CLIENT (client));
- old = client->auth;
+ priv = client->priv;
- if (old != auth) {
- if (auth)
- g_object_ref (auth);
- client->auth = auth;
- if (old)
- g_object_unref (old);
- }
+ if (auth)
+ g_object_ref (auth);
+
+ g_mutex_lock (&priv->lock);
+ old = priv->auth;
+ priv->auth = auth;
+ g_mutex_unlock (&priv->lock);
+
+ if (old)
+ g_object_unref (old);
}
GstRTSPAuth *
gst_rtsp_client_get_auth (GstRTSPClient * client)
{
+ GstRTSPClientPrivate *priv;
GstRTSPAuth *result;
g_return_val_if_fail (GST_IS_RTSP_CLIENT (client), NULL);
- if ((result = client->auth))
+ priv = client->priv;
+
+ g_mutex_lock (&priv->lock);
+ if ((result = priv->auth))
g_object_ref (result);
+ g_mutex_unlock (&priv->lock);
return result;
}
-static GstRTSPResult
-message_received (GstRTSPWatch * watch, GstRTSPMessage * message,
- gpointer user_data)
+/**
+ * gst_rtsp_client_get_uri:
+ * @client: a #GstRTSPClient
+ *
+ * Get the #GstRTSPUrl of @client.
+ *
+ * Returns: (transfer full): the #GstRTSPUrl of @client. Free with
+ * gst_rtsp_url_free () after usage.
+ */
+GstRTSPUrl *
+gst_rtsp_client_get_uri (GstRTSPClient * client)
{
- GstRTSPClient *client = GST_RTSP_CLIENT (user_data);
+ GstRTSPClientPrivate *priv;
+ GstRTSPUrl *result = NULL;
+
+ g_return_val_if_fail (GST_IS_RTSP_CLIENT (client), NULL);
+
+ priv = client->priv;
+
+ g_mutex_lock (&priv->lock);
+ if (priv->uri != NULL)
+ result = gst_rtsp_url_copy (priv->uri);
+ g_mutex_unlock (&priv->lock);
+
+ return result;
+}
+
+/**
+ * gst_rtsp_client_get_connection:
+ * @client: a #GstRTSPClient
+ *
+ * Get the #GstRTSPConnection of @client.
+ *
+ * Returns: (transfer none): the #GstRTSPConnection of @client.
+ * The connection object returned remains valid until the client is freed.
+ */
+GstRTSPConnection *
+gst_rtsp_client_get_connection (GstRTSPClient * client)
+{
+ g_return_val_if_fail (GST_IS_RTSP_CLIENT (client), NULL);
+
+ return client->priv->connection;
+}
+
+/**
+ * gst_rtsp_client_set_send_func:
+ * @client: a #GstRTSPClient
+ * @func: a #GstRTSPClientSendFunc
+ * @user_data: user data passed to @func
+ * @notify: called when @user_data is no longer in use
+ *
+ * Set @func as the callback that will be called when a new message needs to be
+ * sent to the client. @user_data is passed to @func and @notify is called when
+ * @user_data is no longer in use.
+ */
+void
+gst_rtsp_client_set_send_func (GstRTSPClient * client,
+ GstRTSPClientSendFunc func, gpointer user_data, GDestroyNotify notify)
+{
+ GstRTSPClientPrivate *priv;
+ GDestroyNotify old_notify;
+ gpointer old_data;
+
+ g_return_if_fail (GST_IS_RTSP_CLIENT (client));
+
+ priv = client->priv;
+
+ g_mutex_lock (&priv->send_lock);
+ priv->send_func = func;
+ old_notify = priv->send_notify;
+ old_data = priv->send_data;
+ priv->send_notify = notify;
+ priv->send_data = user_data;
+ g_mutex_unlock (&priv->send_lock);
+
+ if (old_notify)
+ old_notify (old_data);
+}
+
+/**
+ * gst_rtsp_client_handle_message:
+ * @client: a #GstRTSPClient
+ * @message: an #GstRTSPMessage
+ *
+ * Let the client handle @message.
+ *
+ * Returns: a #GstRTSPResult.
+ */
+GstRTSPResult
+gst_rtsp_client_handle_message (GstRTSPClient * client,
+ GstRTSPMessage * message)
+{
+ g_return_val_if_fail (GST_IS_RTSP_CLIENT (client), GST_RTSP_EINVAL);
+ g_return_val_if_fail (message != NULL, GST_RTSP_EINVAL);
switch (message->type) {
case GST_RTSP_MESSAGE_REQUEST:
}
static GstRTSPResult
+do_send_message (GstRTSPClient * client, GstRTSPMessage * message,
+ gboolean close, gpointer user_data)
+{
+ GstRTSPClientPrivate *priv = client->priv;
+
+ /* send the response and store the seq number so we can wait until it's
+ * written to the client to close the connection */
+ return gst_rtsp_watch_send_message (priv->watch, message, close ?
+ &priv->close_seq : NULL);
+}
+
+static GstRTSPResult
+message_received (GstRTSPWatch * watch, GstRTSPMessage * message,
+ gpointer user_data)
+{
+ return gst_rtsp_client_handle_message (GST_RTSP_CLIENT (user_data), message);
+}
+
+static GstRTSPResult
message_sent (GstRTSPWatch * watch, guint cseq, gpointer user_data)
{
- GstRTSPClient *client;
+ GstRTSPClient *client = GST_RTSP_CLIENT (user_data);
+ GstRTSPClientPrivate *priv = client->priv;
- client = GST_RTSP_CLIENT (user_data);
- if (client->close_response_seq && client->close_response_seq == cseq) {
- client->close_response_seq = 0;
+ if (priv->close_seq && priv->close_seq == cseq) {
+ priv->close_seq = 0;
close_connection (client);
}
closed (GstRTSPWatch * watch, gpointer user_data)
{
GstRTSPClient *client = GST_RTSP_CLIENT (user_data);
+ GstRTSPClientPrivate *priv = client->priv;
const gchar *tunnelid;
GST_INFO ("client %p: connection closed", client);
- if ((tunnelid = gst_rtsp_connection_get_tunnelid (client->connection))) {
+ if ((tunnelid = gst_rtsp_connection_get_tunnelid (priv->connection))) {
g_mutex_lock (&tunnels_lock);
/* remove from tunnelids */
g_hash_table_remove (tunnels, tunnelid);
g_mutex_unlock (&tunnels_lock);
}
+ gst_rtsp_client_set_send_func (client, NULL, NULL, NULL);
+
return GST_RTSP_OK;
}
static gboolean
remember_tunnel (GstRTSPClient * client)
{
+ GstRTSPClientPrivate *priv = client->priv;
const gchar *tunnelid;
/* store client in the pending tunnels */
- tunnelid = gst_rtsp_connection_get_tunnelid (client->connection);
+ tunnelid = gst_rtsp_connection_get_tunnelid (priv->connection);
if (tunnelid == NULL)
goto no_tunnelid;
static GstRTSPStatusCode
tunnel_start (GstRTSPWatch * watch, gpointer user_data)
{
- GstRTSPClient *client;
-
- client = GST_RTSP_CLIENT (user_data);
+ GstRTSPClient *client = GST_RTSP_CLIENT (user_data);
+ GstRTSPClientPrivate *priv = client->priv;
GST_INFO ("client %p: tunnel start (connection %p)", client,
- client->connection);
+ priv->connection);
if (!remember_tunnel (client))
goto tunnel_error;
static GstRTSPResult
tunnel_lost (GstRTSPWatch * watch, gpointer user_data)
{
- GstRTSPClient *client;
-
- client = GST_RTSP_CLIENT (user_data);
+ GstRTSPClient *client = GST_RTSP_CLIENT (user_data);
+ GstRTSPClientPrivate *priv = client->priv;
- GST_INFO ("client %p: tunnel lost (connection %p)", client,
- client->connection);
+ GST_WARNING ("client %p: tunnel lost (connection %p)", client,
+ priv->connection);
/* ignore error, it'll only be a problem when the client does a POST again */
remember_tunnel (client);
{
const gchar *tunnelid;
GstRTSPClient *client = GST_RTSP_CLIENT (user_data);
+ GstRTSPClientPrivate *priv = client->priv;
GstRTSPClient *oclient;
+ GstRTSPClientPrivate *opriv;
GST_INFO ("client %p: tunnel complete", client);
/* find previous tunnel */
- tunnelid = gst_rtsp_connection_get_tunnelid (client->connection);
+ tunnelid = gst_rtsp_connection_get_tunnelid (priv->connection);
if (tunnelid == NULL)
goto no_tunnelid;
g_object_ref (oclient);
g_hash_table_remove (tunnels, tunnelid);
- if (oclient->watch == NULL)
+ opriv = oclient->priv;
+
+ if (opriv->watch == NULL)
goto tunnel_closed;
g_mutex_unlock (&tunnels_lock);
GST_INFO ("client %p: found tunnel %p (old %p, new %p)", client, oclient,
- oclient->connection, client->connection);
+ opriv->connection, priv->connection);
/* merge the tunnels into the first client */
- gst_rtsp_connection_do_tunnel (oclient->connection, client->connection);
- gst_rtsp_watch_reset (oclient->watch);
+ gst_rtsp_connection_do_tunnel (opriv->connection, priv->connection);
+ gst_rtsp_watch_reset (opriv->watch);
g_object_unref (oclient);
return GST_RTSP_OK;
/* ERRORS */
no_tunnelid:
{
- GST_INFO ("client %p: no tunnelid provided", client);
+ GST_ERROR ("client %p: no tunnelid provided", client);
return GST_RTSP_ERROR;
}
no_tunnel:
{
g_mutex_unlock (&tunnels_lock);
- GST_INFO ("client %p: tunnel session %s not found", client, tunnelid);
+ GST_ERROR ("client %p: tunnel session %s not found", client, tunnelid);
return GST_RTSP_ERROR;
}
tunnel_closed:
{
g_mutex_unlock (&tunnels_lock);
- GST_INFO ("client %p: tunnel session %s was closed", client, tunnelid);
+ GST_ERROR ("client %p: tunnel session %s was closed", client, tunnelid);
g_object_unref (oclient);
return GST_RTSP_ERROR;
}
static void
client_watch_notify (GstRTSPClient * client)
{
+ GstRTSPClientPrivate *priv = client->priv;
+
GST_INFO ("client %p: watch destroyed", client);
- client->watch = NULL;
+ priv->watch = NULL;
g_signal_emit (client, gst_rtsp_client_signals[SIGNAL_CLOSED], 0, NULL);
g_object_unref (client);
}
setup_client (GstRTSPClient * client, GSocket * socket,
GstRTSPConnection * conn, GError ** error)
{
+ GstRTSPClientPrivate *priv = client->priv;
GSocket *read_socket;
GSocketAddress *address;
GstRTSPUrl *url;
read_socket = gst_rtsp_connection_get_read_socket (conn);
- client->is_ipv6 = g_socket_get_family (socket) == G_SOCKET_FAMILY_IPV6;
+ priv->is_ipv6 = g_socket_get_family (socket) == G_SOCKET_FAMILY_IPV6;
if (!(address = g_socket_get_remote_address (read_socket, error)))
goto no_address;
- g_free (client->server_ip);
+ g_free (priv->server_ip);
/* keep the original ip that the client connected to */
if (G_IS_INET_SOCKET_ADDRESS (address)) {
GInetAddress *iaddr;
iaddr = g_inet_socket_address_get_address (G_INET_SOCKET_ADDRESS (address));
- client->server_ip = g_inet_address_to_string (iaddr);
+ priv->server_ip = g_inet_address_to_string (iaddr);
g_object_unref (address);
} else {
- client->server_ip = g_strdup ("unknown");
+ priv->server_ip = g_strdup ("unknown");
}
GST_INFO ("client %p connected to server ip %s, ipv6 = %d", client,
- client->server_ip, client->is_ipv6);
+ priv->server_ip, priv->is_ipv6);
url = gst_rtsp_connection_get_url (conn);
GST_INFO ("added new client %p ip %s:%d", client, url->host, url->port);
- client->connection = conn;
+ priv->connection = conn;
return TRUE;
GstRTSPConnection *conn;
GstRTSPResult res;
+ g_return_val_if_fail (GST_IS_RTSP_CLIENT (client), FALSE);
+ g_return_val_if_fail (G_IS_SOCKET (socket), FALSE);
+
GST_RTSP_CHECK (gst_rtsp_connection_create_from_socket (socket, ip, port,
initial_buffer, &conn), no_connection);
GstRTSPConnection *conn;
GstRTSPResult res;
+ g_return_val_if_fail (GST_IS_RTSP_CLIENT (client), FALSE);
+ g_return_val_if_fail (G_IS_SOCKET (socket), FALSE);
+
/* a new client connected. */
GST_RTSP_CHECK (gst_rtsp_connection_accept (socket, &conn, cancellable),
accept_failed);
guint
gst_rtsp_client_attach (GstRTSPClient * client, GMainContext * context)
{
+ GstRTSPClientPrivate *priv;
guint res;
g_return_val_if_fail (GST_IS_RTSP_CLIENT (client), 0);
- g_return_val_if_fail (client->watch == NULL, 0);
+ priv = client->priv;
+ g_return_val_if_fail (priv->watch == NULL, 0);
/* create watch for the connection and attach */
- client->watch = gst_rtsp_watch_new (client->connection, &watch_funcs,
+ priv->watch = gst_rtsp_watch_new (priv->connection, &watch_funcs,
g_object_ref (client), (GDestroyNotify) client_watch_notify);
+ gst_rtsp_client_set_send_func (client, do_send_message, priv->watch,
+ (GDestroyNotify) gst_rtsp_watch_unref);
+
+ /* FIXME make this configurable. We don't want to do this yet because it will
+ * be superceeded by a cache object later */
+ gst_rtsp_watch_set_send_backlog (priv->watch, 0, 100);
GST_INFO ("attaching to context %p", context);
- res = gst_rtsp_watch_attach (client->watch, context);
- gst_rtsp_watch_unref (client->watch);
+ res = gst_rtsp_watch_attach (priv->watch, context);
return res;
}
projects / platform / upstream / gstreamer.git / blobdiff