SIGNAL_TEARDOWN_REQUEST,
SIGNAL_SET_PARAMETER_REQUEST,
SIGNAL_GET_PARAMETER_REQUEST,
+ SIGNAL_HANDLE_RESPONSE,
SIGNAL_LAST
};
GstRTSPSession * session);
static void unlink_session_transports (GstRTSPClient * client,
GstRTSPSession * session, GstRTSPSessionMedia * sessmedia);
+static gboolean default_configure_client_media (GstRTSPClient * client,
+ GstRTSPMedia * media, GstRTSPStream * stream, GstRTSPContext * ctx);
static gboolean default_configure_client_transport (GstRTSPClient * client,
- GstRTSPClientState * state, GstRTSPTransport * ct);
+ GstRTSPContext * ctx, GstRTSPTransport * ct);
static GstRTSPResult default_params_set (GstRTSPClient * client,
- GstRTSPClientState * state);
+ GstRTSPContext * ctx);
static GstRTSPResult default_params_get (GstRTSPClient * client,
- GstRTSPClientState * state);
+ GstRTSPContext * ctx);
+static gchar *default_make_path_from_uri (GstRTSPClient * client,
+ const GstRTSPUrl * uri);
G_DEFINE_TYPE (GstRTSPClient, gst_rtsp_client, G_TYPE_OBJECT);
gobject_class->finalize = gst_rtsp_client_finalize;
klass->create_sdp = create_sdp;
+ klass->configure_client_media = default_configure_client_media;
klass->configure_client_transport = default_configure_client_transport;
klass->params_set = default_params_set;
klass->params_get = default_params_get;
+ klass->make_path_from_uri = default_make_path_from_uri;
g_object_class_install_property (gobject_class, PROP_SESSION_POOL,
g_param_spec_object ("session-pool", "Session Pool",
get_parameter_request), NULL, NULL, g_cclosure_marshal_VOID__POINTER,
G_TYPE_NONE, 1, G_TYPE_POINTER);
+ gst_rtsp_client_signals[SIGNAL_HANDLE_RESPONSE] =
+ g_signal_new ("handle-response", G_TYPE_FROM_CLASS (klass),
+ G_SIGNAL_RUN_LAST, G_STRUCT_OFFSET (GstRTSPClientClass,
+ handle_response), 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);
g_mutex_init (&tunnels_lock);
g_object_unref (priv->mount_points);
if (priv->auth)
g_object_unref (priv->auth);
+ if (priv->thread_pool)
+ g_object_unref (priv->thread_pool);
if (priv->path)
g_free (priv->path);
static void
send_generic_response (GstRTSPClient * client, GstRTSPStatusCode code,
- GstRTSPClientState * state)
+ GstRTSPContext * ctx)
{
- gst_rtsp_message_init_response (state->response, code,
- gst_rtsp_status_as_text (code), state->request);
+ gst_rtsp_message_init_response (ctx->response, code,
+ gst_rtsp_status_as_text (code), ctx->request);
- send_message (client, NULL, state->response, FALSE);
+ send_message (client, NULL, ctx->response, FALSE);
}
static gboolean
* but is cached for when the same client (without breaking the connection) is
* doing a setup for the exact same url. */
static GstRTSPMedia *
-find_media (GstRTSPClient * client, GstRTSPClientState * state, gint * matched)
+find_media (GstRTSPClient * client, GstRTSPContext * ctx, gchar * path,
+ gint * matched)
{
GstRTSPClientPrivate *priv = client->priv;
GstRTSPMediaFactory *factory;
GstRTSPMedia *media;
- gchar *path;
gint path_len;
- if (!priv->mount_points)
- goto no_mount_points;
-
- path = state->uri->abspath;
-
/* find the longest matching factory for the uri first */
if (!(factory = gst_rtsp_mount_points_match (priv->mount_points,
path, matched)))
goto no_factory;
- state->factory = factory;
+ ctx->factory = factory;
if (!gst_rtsp_auth_check (GST_RTSP_AUTH_CHECK_MEDIA_FACTORY_ACCESS))
goto no_factory_access;
priv->media = NULL;
/* prepare the media and add it to the pipeline */
- if (!(media = gst_rtsp_media_factory_construct (factory, state->uri)))
+ if (!(media = gst_rtsp_media_factory_construct (factory, ctx->uri)))
goto no_media;
- state->media = media;
+ ctx->media = media;
thread = gst_rtsp_thread_pool_get_thread (priv->thread_pool,
- GST_RTSP_THREAD_TYPE_MEDIA, state);
+ GST_RTSP_THREAD_TYPE_MEDIA, ctx);
if (thread == NULL)
goto no_thread;
} else {
/* we have seen this path before, used cached media */
media = priv->media;
- state->media = media;
+ ctx->media = media;
GST_INFO ("reusing cached media %p for path %s", media, priv->path);
}
g_object_unref (factory);
- state->factory = NULL;
+ ctx->factory = NULL;
if (media)
g_object_ref (media);
return 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 %s", client, path);
- send_generic_response (client, GST_RTSP_STS_NOT_FOUND, state);
+ GST_ERROR ("client %p: no factory for path %s", client, path);
+ send_generic_response (client, GST_RTSP_STS_NOT_FOUND, ctx);
return NULL;
}
no_factory_access:
{
- GST_ERROR ("client %p: not authorized to see factory uri %s", client, path);
+ GST_ERROR ("client %p: not authorized to see factory path %s", client,
+ path);
+ /* error reply is already sent */
return NULL;
}
not_authorized:
{
- GST_ERROR ("client %p: not authorized for factory uri %s", client, path);
+ GST_ERROR ("client %p: not authorized for factory path %s", client, path);
+ /* error reply is already sent */
return NULL;
}
no_media:
{
GST_ERROR ("client %p: can't create media", client);
- send_generic_response (client, GST_RTSP_STS_SERVICE_UNAVAILABLE, state);
+ send_generic_response (client, GST_RTSP_STS_BAD_REQUEST, ctx);
g_object_unref (factory);
- state->factory = NULL;
+ ctx->factory = NULL;
return NULL;
}
no_thread:
{
GST_ERROR ("client %p: can't create thread", client);
- send_generic_response (client, GST_RTSP_STS_SERVICE_UNAVAILABLE, state);
+ send_generic_response (client, GST_RTSP_STS_SERVICE_UNAVAILABLE, ctx);
g_object_unref (media);
- state->media = NULL;
+ ctx->media = NULL;
g_object_unref (factory);
- state->factory = NULL;
+ ctx->factory = NULL;
return NULL;
}
no_prepare:
{
GST_ERROR ("client %p: can't prepare media", client);
- send_generic_response (client, GST_RTSP_STS_SERVICE_UNAVAILABLE, state);
+ send_generic_response (client, GST_RTSP_STS_SERVICE_UNAVAILABLE, ctx);
g_object_unref (media);
- state->media = NULL;
+ ctx->media = NULL;
g_object_unref (factory);
- state->factory = NULL;
+ ctx->factory = NULL;
return NULL;
}
}
gst_rtsp_connection_close (priv->connection);
}
+static gchar *
+default_make_path_from_uri (GstRTSPClient * client, const GstRTSPUrl * uri)
+{
+ gchar *path;
+
+ if (uri->query)
+ path = g_strconcat (uri->abspath, "?", uri->query, NULL);
+ else
+ path = g_strdup (uri->abspath);
+
+ return path;
+}
+
static gboolean
-handle_teardown_request (GstRTSPClient * client, GstRTSPClientState * state)
+handle_teardown_request (GstRTSPClient * client, GstRTSPContext * ctx)
{
GstRTSPClientPrivate *priv = client->priv;
+ GstRTSPClientClass *klass;
GstRTSPSession *session;
GstRTSPSessionMedia *sessmedia;
GstRTSPStatusCode code;
- const gchar *path;
+ gchar *path;
gint matched;
- if (!state->session)
+ if (!ctx->session)
goto no_session;
- session = state->session;
+ session = ctx->session;
- if (!state->uri)
+ if (!ctx->uri)
goto no_uri;
- path = state->uri->abspath;
+ klass = GST_RTSP_CLIENT_GET_CLASS (client);
+ path = klass->make_path_from_uri (client, ctx->uri);
/* get a handle to the configuration of the media in the session */
sessmedia = gst_rtsp_session_get_media (session, path, &matched);
if (path[matched] != '\0')
goto no_aggregate;
- state->sessmedia = sessmedia;
+ g_free (path);
+
+ ctx->sessmedia = sessmedia;
/* we emit the signal before closing the connection */
g_signal_emit (client, gst_rtsp_client_signals[SIGNAL_TEARDOWN_REQUEST],
- 0, state);
+ 0, ctx);
/* unlink the all TCP callbacks */
unlink_session_transports (client, session, sessmedia);
}
/* construct the response now */
code = GST_RTSP_STS_OK;
- gst_rtsp_message_init_response (state->response, code,
- gst_rtsp_status_as_text (code), state->request);
+ gst_rtsp_message_init_response (ctx->response, code,
+ gst_rtsp_status_as_text (code), ctx->request);
- send_message (client, session, state->response, TRUE);
+ send_message (client, session, ctx->response, TRUE);
return TRUE;
no_session:
{
GST_ERROR ("client %p: no session", client);
- send_generic_response (client, GST_RTSP_STS_SESSION_NOT_FOUND, state);
+ send_generic_response (client, GST_RTSP_STS_SESSION_NOT_FOUND, ctx);
return FALSE;
}
no_uri:
{
GST_ERROR ("client %p: no uri supplied", client);
- send_generic_response (client, GST_RTSP_STS_BAD_REQUEST, state);
+ send_generic_response (client, GST_RTSP_STS_BAD_REQUEST, ctx);
return FALSE;
}
not_found:
{
GST_ERROR ("client %p: no media for uri", client);
- send_generic_response (client, GST_RTSP_STS_NOT_FOUND, state);
+ send_generic_response (client, GST_RTSP_STS_NOT_FOUND, ctx);
+ g_free (path);
return FALSE;
}
no_aggregate:
{
GST_ERROR ("client %p: no aggregate path %s", client, path);
send_generic_response (client,
- GST_RTSP_STS_ONLY_AGGREGATE_OPERATION_ALLOWED, state);
+ GST_RTSP_STS_ONLY_AGGREGATE_OPERATION_ALLOWED, ctx);
+ g_free (path);
return FALSE;
}
}
static GstRTSPResult
-default_params_set (GstRTSPClient * client, GstRTSPClientState * state)
+default_params_set (GstRTSPClient * client, GstRTSPContext * ctx)
{
GstRTSPResult res;
- res = gst_rtsp_params_set (client, state);
+ res = gst_rtsp_params_set (client, ctx);
return res;
}
static GstRTSPResult
-default_params_get (GstRTSPClient * client, GstRTSPClientState * state)
+default_params_get (GstRTSPClient * client, GstRTSPContext * ctx)
{
GstRTSPResult res;
- res = gst_rtsp_params_get (client, state);
+ res = gst_rtsp_params_get (client, ctx);
return res;
}
static gboolean
-handle_get_param_request (GstRTSPClient * client, GstRTSPClientState * state)
+handle_get_param_request (GstRTSPClient * client, GstRTSPContext * ctx)
{
GstRTSPResult res;
guint8 *data;
guint size;
- res = gst_rtsp_message_get_body (state->request, &data, &size);
+ res = gst_rtsp_message_get_body (ctx->request, &data, &size);
if (res != GST_RTSP_OK)
goto bad_request;
if (size == 0) {
/* no body, keep-alive request */
- send_generic_response (client, GST_RTSP_STS_OK, state);
+ send_generic_response (client, GST_RTSP_STS_OK, ctx);
} else {
/* there is a body, handle the params */
- res = GST_RTSP_CLIENT_GET_CLASS (client)->params_get (client, state);
+ res = GST_RTSP_CLIENT_GET_CLASS (client)->params_get (client, ctx);
if (res != GST_RTSP_OK)
goto bad_request;
- send_message (client, state->session, state->response, FALSE);
+ send_message (client, ctx->session, ctx->response, FALSE);
}
g_signal_emit (client, gst_rtsp_client_signals[SIGNAL_GET_PARAMETER_REQUEST],
- 0, state);
+ 0, ctx);
return TRUE;
bad_request:
{
GST_ERROR ("client %p: bad request", client);
- send_generic_response (client, GST_RTSP_STS_BAD_REQUEST, state);
+ send_generic_response (client, GST_RTSP_STS_BAD_REQUEST, ctx);
return FALSE;
}
}
static gboolean
-handle_set_param_request (GstRTSPClient * client, GstRTSPClientState * state)
+handle_set_param_request (GstRTSPClient * client, GstRTSPContext * ctx)
{
GstRTSPResult res;
guint8 *data;
guint size;
- res = gst_rtsp_message_get_body (state->request, &data, &size);
+ res = gst_rtsp_message_get_body (ctx->request, &data, &size);
if (res != GST_RTSP_OK)
goto bad_request;
if (size == 0) {
/* no body, keep-alive request */
- send_generic_response (client, GST_RTSP_STS_OK, state);
+ send_generic_response (client, GST_RTSP_STS_OK, ctx);
} else {
/* there is a body, handle the params */
- res = GST_RTSP_CLIENT_GET_CLASS (client)->params_set (client, state);
+ res = GST_RTSP_CLIENT_GET_CLASS (client)->params_set (client, ctx);
if (res != GST_RTSP_OK)
goto bad_request;
- send_message (client, state->session, state->response, FALSE);
+ send_message (client, ctx->session, ctx->response, FALSE);
}
g_signal_emit (client, gst_rtsp_client_signals[SIGNAL_SET_PARAMETER_REQUEST],
- 0, state);
+ 0, ctx);
return TRUE;
bad_request:
{
GST_ERROR ("client %p: bad request", client);
- send_generic_response (client, GST_RTSP_STS_BAD_REQUEST, state);
+ send_generic_response (client, GST_RTSP_STS_BAD_REQUEST, ctx);
return FALSE;
}
}
static gboolean
-handle_pause_request (GstRTSPClient * client, GstRTSPClientState * state)
+handle_pause_request (GstRTSPClient * client, GstRTSPContext * ctx)
{
GstRTSPSession *session;
+ GstRTSPClientClass *klass;
GstRTSPSessionMedia *sessmedia;
GstRTSPStatusCode code;
GstRTSPState rtspstate;
- const gchar *path;
+ gchar *path;
gint matched;
- if (!(session = state->session))
+ if (!(session = ctx->session))
goto no_session;
- if (!state->uri)
+ if (!ctx->uri)
goto no_uri;
- path = state->uri->abspath;
+ klass = GST_RTSP_CLIENT_GET_CLASS (client);
+ path = klass->make_path_from_uri (client, ctx->uri);
/* get a handle to the configuration of the media in the session */
sessmedia = gst_rtsp_session_get_media (session, path, &matched);
if (path[matched] != '\0')
goto no_aggregate;
- state->sessmedia = sessmedia;
+ g_free (path);
+
+ ctx->sessmedia = sessmedia;
rtspstate = gst_rtsp_session_media_get_rtsp_state (sessmedia);
/* the session state must be playing or recording */
/* construct the response now */
code = GST_RTSP_STS_OK;
- gst_rtsp_message_init_response (state->response, code,
- gst_rtsp_status_as_text (code), state->request);
+ gst_rtsp_message_init_response (ctx->response, code,
+ gst_rtsp_status_as_text (code), ctx->request);
- send_message (client, session, state->response, FALSE);
+ send_message (client, session, ctx->response, FALSE);
/* the state is now READY */
gst_rtsp_session_media_set_rtsp_state (sessmedia, GST_RTSP_STATE_READY);
- g_signal_emit (client, gst_rtsp_client_signals[SIGNAL_PAUSE_REQUEST],
- 0, state);
+ g_signal_emit (client, gst_rtsp_client_signals[SIGNAL_PAUSE_REQUEST], 0, ctx);
return TRUE;
no_session:
{
GST_ERROR ("client %p: no seesion", client);
- send_generic_response (client, GST_RTSP_STS_SESSION_NOT_FOUND, state);
+ send_generic_response (client, GST_RTSP_STS_SESSION_NOT_FOUND, ctx);
return FALSE;
}
no_uri:
{
GST_ERROR ("client %p: no uri supplied", client);
- send_generic_response (client, GST_RTSP_STS_BAD_REQUEST, state);
+ send_generic_response (client, GST_RTSP_STS_BAD_REQUEST, ctx);
return FALSE;
}
not_found:
{
GST_ERROR ("client %p: no media for uri", client);
- send_generic_response (client, GST_RTSP_STS_NOT_FOUND, state);
+ send_generic_response (client, GST_RTSP_STS_NOT_FOUND, ctx);
+ g_free (path);
return FALSE;
}
no_aggregate:
{
GST_ERROR ("client %p: no aggregate path %s", client, path);
send_generic_response (client,
- GST_RTSP_STS_ONLY_AGGREGATE_OPERATION_ALLOWED, state);
+ GST_RTSP_STS_ONLY_AGGREGATE_OPERATION_ALLOWED, ctx);
+ g_free (path);
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);
+ ctx);
return FALSE;
}
}
+/* convert @url and @path to a URL used as a content base for the factory
+ * located at @path */
+static gchar *
+make_base_url (GstRTSPClient * client, GstRTSPUrl * url, gchar * path)
+{
+ GstRTSPUrl tmp;
+ gchar *result, *trail;
+
+ /* check for trailing '/' and append one */
+ trail = (path[strlen (path) - 1] != '/' ? "/" : "");
+
+ tmp = *url;
+ tmp.user = NULL;
+ tmp.passwd = NULL;
+ tmp.abspath = g_strdup_printf ("%s%s", path, trail);
+ tmp.query = NULL;
+ result = gst_rtsp_url_get_request_uri (&tmp);
+ g_free (tmp.abspath);
+
+ return result;
+}
+
static gboolean
-handle_play_request (GstRTSPClient * client, GstRTSPClientState * state)
+handle_play_request (GstRTSPClient * client, GstRTSPContext * ctx)
{
GstRTSPSession *session;
+ GstRTSPClientClass *klass;
GstRTSPSessionMedia *sessmedia;
GstRTSPMedia *media;
GstRTSPStatusCode code;
+ GstRTSPUrl *uri;
GString *rtpinfo;
guint n_streams, i, infocount;
gchar *str;
GstRTSPResult res;
GstRTSPState rtspstate;
GstRTSPRangeUnit unit = GST_RTSP_RANGE_NPT;
- const gchar *path;
+ gchar *path;
gint matched;
- if (!(session = state->session))
+ if (!(session = ctx->session))
goto no_session;
- if (!state->uri)
+ if (!(uri = ctx->uri))
goto no_uri;
- path = state->uri->abspath;
+ klass = GST_RTSP_CLIENT_GET_CLASS (client);
+ path = klass->make_path_from_uri (client, uri);
/* get a handle to the configuration of the media in the session */
sessmedia = gst_rtsp_session_get_media (session, path, &matched);
if (path[matched] != '\0')
goto no_aggregate;
- state->sessmedia = sessmedia;
- state->media = media = gst_rtsp_session_media_get_media (sessmedia);
+ ctx->sessmedia = sessmedia;
+ ctx->media = media = gst_rtsp_session_media_get_media (sessmedia);
/* the session state must be playing or ready */
rtspstate = gst_rtsp_session_media_get_rtsp_state (sessmedia);
if (rtspstate != GST_RTSP_STATE_PLAYING && rtspstate != GST_RTSP_STATE_READY)
goto invalid_state;
+ /* in play we first unsuspend, media could be suspended from SDP or PAUSED */
+ if (!gst_rtsp_media_unsuspend (media))
+ goto unsuspend_failed;
+
/* parse the range header if we have one */
- res =
- gst_rtsp_message_get_header (state->request, GST_RTSP_HDR_RANGE, &str, 0);
+ res = gst_rtsp_message_get_header (ctx->request, GST_RTSP_HDR_RANGE, &str, 0);
if (res == GST_RTSP_OK) {
if (gst_rtsp_range_parse (str, &range) == GST_RTSP_OK) {
/* we have a range, seek to the position */
GstRTSPStreamTransport *trans;
GstRTSPStream *stream;
const GstRTSPTransport *tr;
- gchar *uristr;
guint rtptime, seq;
/* get the transport, if there is no transport configured, skip this stream */
stream = gst_rtsp_stream_transport_get_stream (trans);
if (gst_rtsp_stream_get_rtpinfo (stream, &rtptime, &seq)) {
+ const GstRTSPUrl *url;
+ gchar *url_str;
+
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, seq, rtptime);
- g_free (uristr);
+ url = gst_rtsp_stream_transport_get_url (trans);
+ url_str = gst_rtsp_url_get_request_uri (url);
+ g_string_append_printf (rtpinfo, "url=%s;seq=%u;rtptime=%u",
+ url_str, seq, rtptime);
+ g_free (url_str);
infocount++;
} else {
GST_WARNING ("RTP-Info cannot be determined for stream %d", i);
}
}
+ g_free (path);
/* construct the response now */
code = GST_RTSP_STS_OK;
- gst_rtsp_message_init_response (state->response, code,
- gst_rtsp_status_as_text (code), state->request);
+ gst_rtsp_message_init_response (ctx->response, code,
+ gst_rtsp_status_as_text (code), ctx->request);
/* add the RTP-Info header */
if (infocount > 0) {
str = g_string_free (rtpinfo, FALSE);
- gst_rtsp_message_take_header (state->response, GST_RTSP_HDR_RTP_INFO, str);
+ gst_rtsp_message_take_header (ctx->response, GST_RTSP_HDR_RTP_INFO, str);
} else {
g_string_free (rtpinfo, TRUE);
}
/* add the range */
str = gst_rtsp_media_get_range_string (media, TRUE, unit);
- gst_rtsp_message_take_header (state->response, GST_RTSP_HDR_RANGE, str);
+ if (str)
+ gst_rtsp_message_take_header (ctx->response, GST_RTSP_HDR_RANGE, str);
- send_message (client, session, state->response, FALSE);
+ send_message (client, session, ctx->response, FALSE);
/* start playing after sending the request */
gst_rtsp_session_media_set_state (sessmedia, GST_STATE_PLAYING);
gst_rtsp_session_media_set_rtsp_state (sessmedia, GST_RTSP_STATE_PLAYING);
- g_signal_emit (client, gst_rtsp_client_signals[SIGNAL_PLAY_REQUEST],
- 0, state);
+ g_signal_emit (client, gst_rtsp_client_signals[SIGNAL_PLAY_REQUEST], 0, ctx);
return TRUE;
no_session:
{
GST_ERROR ("client %p: no session", client);
- send_generic_response (client, GST_RTSP_STS_SESSION_NOT_FOUND, state);
+ send_generic_response (client, GST_RTSP_STS_SESSION_NOT_FOUND, ctx);
return FALSE;
}
no_uri:
{
GST_ERROR ("client %p: no uri supplied", client);
- send_generic_response (client, GST_RTSP_STS_BAD_REQUEST, state);
+ send_generic_response (client, GST_RTSP_STS_BAD_REQUEST, ctx);
return FALSE;
}
not_found:
{
GST_ERROR ("client %p: media not found", client);
- send_generic_response (client, GST_RTSP_STS_NOT_FOUND, state);
+ send_generic_response (client, GST_RTSP_STS_NOT_FOUND, ctx);
return FALSE;
}
no_aggregate:
{
GST_ERROR ("client %p: no aggregate path %s", client, path);
send_generic_response (client,
- GST_RTSP_STS_ONLY_AGGREGATE_OPERATION_ALLOWED, state);
+ GST_RTSP_STS_ONLY_AGGREGATE_OPERATION_ALLOWED, ctx);
+ g_free (path);
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);
+ ctx);
+ g_free (path);
+ return FALSE;
+ }
+unsuspend_failed:
+ {
+ GST_ERROR ("client %p: unsuspend failed", client);
+ send_generic_response (client, GST_RTSP_STS_SERVICE_UNAVAILABLE, ctx);
+ g_free (path);
return FALSE;
}
}
}
static gboolean
-handle_blocksize (GstRTSPMedia * media, GstRTSPStream * stream,
- GstRTSPMessage * request)
+default_configure_client_media (GstRTSPClient * client, GstRTSPMedia * media,
+ GstRTSPStream * stream, GstRTSPContext * ctx)
{
+ GstRTSPMessage *request = ctx->request;
gchar *blocksize_str;
- gboolean ret = TRUE;
if (gst_rtsp_message_get_header (request, GST_RTSP_HDR_BLOCKSIZE,
&blocksize_str, 0) == GST_RTSP_OK) {
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);
- }
+ if (end == blocksize_str)
+ goto parse_failed;
+
+ /* 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))
+ goto done;
+
+ if (blocksize > G_MAXUINT)
+ blocksize = G_MAXUINT;
+
+ gst_rtsp_stream_set_mtu (stream, blocksize);
+ }
+done:
+ return TRUE;
+
+ /* ERRORS */
+parse_failed:
+ {
+ GST_ERROR_OBJECT (client, "failed to parse blocksize");
+ send_generic_response (client, GST_RTSP_STS_BAD_REQUEST, ctx);
+ return FALSE;
}
- return ret;
}
static gboolean
default_configure_client_transport (GstRTSPClient * client,
- GstRTSPClientState * state, GstRTSPTransport * ct)
+ GstRTSPContext * ctx, GstRTSPTransport * ct)
{
GstRTSPClientPrivate *priv = client->priv;
if (ct->destination && use_client_settings) {
GstRTSPAddress *addr;
- addr = gst_rtsp_stream_reserve_address (state->stream, ct->destination,
+ addr = gst_rtsp_stream_reserve_address (ctx->stream, ct->destination,
ct->port.min, ct->port.max - ct->port.min + 1, ct->ttl);
if (addr == NULL)
family = priv->is_ipv6 ? G_SOCKET_FAMILY_IPV6 : G_SOCKET_FAMILY_IPV4;
- addr = gst_rtsp_stream_get_multicast_address (state->stream, family);
+ addr = gst_rtsp_stream_get_multicast_address (ctx->stream, family);
if (addr == NULL)
goto no_address;
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,
+ gst_rtsp_session_media_alloc_channels (ctx->sessmedia,
&ct->interleaved);
}
}
}
static GstRTSPTransport *
-make_server_transport (GstRTSPClient * client, GstRTSPClientState * state,
+make_server_transport (GstRTSPClient * client, GstRTSPContext * ctx,
GstRTSPTransport * ct)
{
GstRTSPTransport *st;
switch (st->lower_transport) {
case GST_RTSP_LOWER_TRANS_UDP:
st->client_port = ct->client_port;
- gst_rtsp_stream_get_server_port (state->stream, &st->server_port, family);
+ gst_rtsp_stream_get_server_port (ctx->stream, &st->server_port, family);
break;
case GST_RTSP_LOWER_TRANS_UDP_MCAST:
st->port = ct->port;
break;
}
- gst_rtsp_stream_get_ssrc (state->stream, &st->ssrc);
+ gst_rtsp_stream_get_ssrc (ctx->stream, &st->ssrc);
return st;
}
static gboolean
-handle_setup_request (GstRTSPClient * client, GstRTSPClientState * state)
+handle_setup_request (GstRTSPClient * client, GstRTSPContext * ctx)
{
GstRTSPClientPrivate *priv = client->priv;
GstRTSPResult res;
gchar *path, *control;
gint matched;
- if (!state->uri)
+ if (!ctx->uri)
goto no_uri;
- uri = state->uri;
- path = uri->abspath;
+ uri = ctx->uri;
+ klass = GST_RTSP_CLIENT_GET_CLASS (client);
+ path = klass->make_path_from_uri (client, uri);
/* parse the transport */
res =
- gst_rtsp_message_get_header (state->request, GST_RTSP_HDR_TRANSPORT,
+ gst_rtsp_message_get_header (ctx->request, GST_RTSP_HDR_TRANSPORT,
&transport, 0);
if (res != GST_RTSP_OK)
goto no_transport;
if (priv->session_pool == NULL)
goto no_pool;
- session = state->session;
+ session = ctx->session;
if (session) {
g_object_ref (session);
/* we have no session media, find one and manage it */
if (sessmedia == NULL) {
/* get a handle to the configuration of the media in the session */
- media = find_media (client, state, &matched);
+ media = find_media (client, ctx, path, &matched);
} else {
if ((media = gst_rtsp_session_media_get_media (sessmedia)))
g_object_ref (media);
+ else
+ goto media_not_found;
}
/* no media, not found then */
if (media == NULL)
- goto media_not_found;
+ goto media_not_found_no_reply;
+
+ if (path[matched] == '\0')
+ goto control_not_found;
- /* path is what matched. We can modify the parsed uri in place */
+ /* path is what matched. */
path[matched] = '\0';
/* control is remainder */
control = &path[matched + 1];
goto stream_not_found;
/* now we have a uri identifying a valid media and stream */
- state->stream = stream;
- state->media = media;
+ ctx->stream = stream;
+ ctx->media = media;
if (session == NULL) {
/* create a session if this fails we probably reached our session limit or
g_signal_emit (client, gst_rtsp_client_signals[SIGNAL_NEW_SESSION], 0,
session);
- state->session = session;
+ ctx->session = session;
}
if (sessmedia == NULL) {
g_object_unref (media);
}
- state->sessmedia = sessmedia;
+ ctx->sessmedia = sessmedia;
- /* set blocksize on this stream */
- if (!handle_blocksize (media, stream, state->request))
- goto invalid_blocksize;
+ if (!klass->configure_client_media (client, media, stream, ctx))
+ goto configure_media_failed_no_reply;
gst_rtsp_transport_new (&ct);
/* our supported transports */
- supported = GST_RTSP_LOWER_TRANS_UDP |
- GST_RTSP_LOWER_TRANS_UDP_MCAST | GST_RTSP_LOWER_TRANS_TCP;
+ supported = gst_rtsp_stream_get_protocols (stream);
/* parse and find a usable supported transport */
if (!parse_transport (transport, supported, ct))
goto unsupported_transports;
/* update the client transport */
- klass = GST_RTSP_CLIENT_GET_CLASS (client);
- if (!klass->configure_client_transport (client, state, ct))
+ if (!klass->configure_client_transport (client, ctx, ct))
goto unsupported_client_transport;
/* set in the session media transport */
trans = gst_rtsp_session_media_set_transport (sessmedia, stream, ct);
+ /* configure the url used to set this transport, this we will use when
+ * generating the response for the PLAY request */
+ gst_rtsp_stream_transport_set_url (trans, uri);
+
/* configure keepalive for this transport */
gst_rtsp_stream_transport_set_keepalive (trans,
(GstRTSPKeepAliveFunc) do_keepalive, session, NULL);
/* create and serialize the server transport */
- st = make_server_transport (client, state, ct);
+ st = make_server_transport (client, ctx, ct);
trans_str = gst_rtsp_transport_as_text (st);
gst_rtsp_transport_free (st);
/* construct the response now */
code = GST_RTSP_STS_OK;
- gst_rtsp_message_init_response (state->response, code,
- gst_rtsp_status_as_text (code), state->request);
+ gst_rtsp_message_init_response (ctx->response, code,
+ gst_rtsp_status_as_text (code), ctx->request);
- gst_rtsp_message_add_header (state->response, GST_RTSP_HDR_TRANSPORT,
+ gst_rtsp_message_add_header (ctx->response, GST_RTSP_HDR_TRANSPORT,
trans_str);
g_free (trans_str);
- send_message (client, session, state->response, FALSE);
+ send_message (client, session, ctx->response, FALSE);
/* update the state */
rtspstate = gst_rtsp_session_media_get_rtsp_state (sessmedia);
break;
}
g_object_unref (session);
+ g_free (path);
- g_signal_emit (client, gst_rtsp_client_signals[SIGNAL_SETUP_REQUEST],
- 0, state);
+ g_signal_emit (client, gst_rtsp_client_signals[SIGNAL_SETUP_REQUEST], 0, ctx);
return TRUE;
no_uri:
{
GST_ERROR ("client %p: no uri", client);
- send_generic_response (client, GST_RTSP_STS_BAD_REQUEST, state);
+ send_generic_response (client, GST_RTSP_STS_BAD_REQUEST, ctx);
return FALSE;
}
no_transport:
{
GST_ERROR ("client %p: no transport", client);
- send_generic_response (client, GST_RTSP_STS_UNSUPPORTED_TRANSPORT, state);
+ send_generic_response (client, GST_RTSP_STS_UNSUPPORTED_TRANSPORT, ctx);
+ g_free (path);
return FALSE;
}
no_pool:
{
GST_ERROR ("client %p: no session pool configured", client);
- send_generic_response (client, GST_RTSP_STS_SESSION_NOT_FOUND, state);
+ send_generic_response (client, GST_RTSP_STS_SESSION_NOT_FOUND, ctx);
+ g_free (path);
+ return FALSE;
+ }
+media_not_found_no_reply:
+ {
+ GST_ERROR ("client %p: media '%s' not found", client, path);
+ g_free (path);
+ /* error reply is already sent */
return FALSE;
}
media_not_found:
{
GST_ERROR ("client %p: media '%s' not found", client, path);
- send_generic_response (client, GST_RTSP_STS_NOT_FOUND, state);
+ send_generic_response (client, GST_RTSP_STS_NOT_FOUND, ctx);
+ g_free (path);
+ return FALSE;
+ }
+control_not_found:
+ {
+ GST_ERROR ("client %p: no control in path '%s'", client, path);
+ send_generic_response (client, GST_RTSP_STS_NOT_FOUND, ctx);
+ g_object_unref (media);
+ g_free (path);
return FALSE;
}
stream_not_found:
{
GST_ERROR ("client %p: stream '%s' not found", client, control);
- send_generic_response (client, GST_RTSP_STS_NOT_FOUND, state);
+ send_generic_response (client, GST_RTSP_STS_NOT_FOUND, ctx);
g_object_unref (media);
+ g_free (path);
return FALSE;
}
service_unavailable:
{
GST_ERROR ("client %p: can't create session", client);
- send_generic_response (client, GST_RTSP_STS_SERVICE_UNAVAILABLE, state);
+ send_generic_response (client, GST_RTSP_STS_SERVICE_UNAVAILABLE, ctx);
g_object_unref (media);
+ g_free (path);
return FALSE;
}
sessmedia_unavailable:
{
GST_ERROR ("client %p: can't create session media", client);
- send_generic_response (client, GST_RTSP_STS_SERVICE_UNAVAILABLE, state);
+ send_generic_response (client, GST_RTSP_STS_SERVICE_UNAVAILABLE, ctx);
g_object_unref (media);
g_object_unref (session);
+ g_free (path);
return FALSE;
}
-invalid_blocksize:
+configure_media_failed_no_reply:
{
- GST_ERROR ("client %p: invalid blocksize", client);
- send_generic_response (client, GST_RTSP_STS_BAD_REQUEST, state);
+ GST_ERROR ("client %p: configure_media failed", client);
g_object_unref (session);
+ g_free (path);
+ /* error reply is already sent */
return FALSE;
}
unsupported_transports:
{
GST_ERROR ("client %p: unsupported transports", client);
- send_generic_response (client, GST_RTSP_STS_UNSUPPORTED_TRANSPORT, state);
+ send_generic_response (client, GST_RTSP_STS_UNSUPPORTED_TRANSPORT, ctx);
gst_rtsp_transport_free (ct);
g_object_unref (session);
+ g_free (path);
return FALSE;
}
unsupported_client_transport:
{
GST_ERROR ("client %p: unsupported client transport", client);
- send_generic_response (client, GST_RTSP_STS_UNSUPPORTED_TRANSPORT, state);
+ send_generic_response (client, GST_RTSP_STS_UNSUPPORTED_TRANSPORT, ctx);
gst_rtsp_transport_free (ct);
g_object_unref (session);
+ g_free (path);
return FALSE;
}
}
/* for the describe we must generate an SDP */
static gboolean
-handle_describe_request (GstRTSPClient * client, GstRTSPClientState * state)
+handle_describe_request (GstRTSPClient * client, GstRTSPContext * ctx)
{
+ GstRTSPClientPrivate *priv = client->priv;
GstRTSPResult res;
GstSDPMessage *sdp;
- guint i, str_len;
- gchar *str, *content_base;
+ guint i;
+ gchar *path, *str;
GstRTSPMedia *media;
GstRTSPClientClass *klass;
klass = GST_RTSP_CLIENT_GET_CLASS (client);
- if (!state->uri)
+ if (!ctx->uri)
goto no_uri;
/* check what kind of format is accepted, we don't really do anything with it
gchar *accept;
res =
- gst_rtsp_message_get_header (state->request, GST_RTSP_HDR_ACCEPT,
+ gst_rtsp_message_get_header (ctx->request, GST_RTSP_HDR_ACCEPT,
&accept, i);
if (res == GST_RTSP_ENOTIMPL)
break;
break;
}
+ if (!priv->mount_points)
+ goto no_mount_points;
+
+ if (!(path = gst_rtsp_mount_points_make_path (priv->mount_points, ctx->uri)))
+ goto no_path;
+
/* find the media object for the uri */
- if (!(media = find_media (client, state, NULL)))
+ if (!(media = find_media (client, ctx, path, NULL)))
goto no_media;
/* create an SDP for the media object on this client */
if (!(sdp = klass->create_sdp (client, media)))
goto no_sdp;
+ /* we suspend after the describe */
+ gst_rtsp_media_suspend (media);
g_object_unref (media);
- gst_rtsp_message_init_response (state->response, GST_RTSP_STS_OK,
- gst_rtsp_status_as_text (GST_RTSP_STS_OK), state->request);
+ gst_rtsp_message_init_response (ctx->response, GST_RTSP_STS_OK,
+ gst_rtsp_status_as_text (GST_RTSP_STS_OK), ctx->request);
- gst_rtsp_message_add_header (state->response, GST_RTSP_HDR_CONTENT_TYPE,
+ gst_rtsp_message_add_header (ctx->response, GST_RTSP_HDR_CONTENT_TYPE,
"application/sdp");
/* content base for some clients that might screw up creating the setup uri */
- str = gst_rtsp_url_get_request_uri (state->uri);
- str_len = strlen (str);
-
- /* check for trailing '/' and append one */
- if (str[str_len - 1] != '/') {
- content_base = g_malloc (str_len + 2);
- memcpy (content_base, str, str_len);
- content_base[str_len] = '/';
- content_base[str_len + 1] = '\0';
- g_free (str);
- } else {
- content_base = str;
- }
+ str = make_base_url (client, ctx->uri, path);
+ g_free (path);
- GST_INFO ("adding content-base: %s", content_base);
-
- gst_rtsp_message_add_header (state->response, GST_RTSP_HDR_CONTENT_BASE,
- content_base);
- g_free (content_base);
+ GST_INFO ("adding content-base: %s", str);
+ gst_rtsp_message_take_header (ctx->response, GST_RTSP_HDR_CONTENT_BASE, str);
/* add SDP to the response body */
str = gst_sdp_message_as_text (sdp);
- gst_rtsp_message_take_body (state->response, (guint8 *) str, strlen (str));
+ gst_rtsp_message_take_body (ctx->response, (guint8 *) str, strlen (str));
gst_sdp_message_free (sdp);
- send_message (client, state->session, state->response, FALSE);
+ send_message (client, ctx->session, ctx->response, FALSE);
g_signal_emit (client, gst_rtsp_client_signals[SIGNAL_DESCRIBE_REQUEST],
- 0, state);
+ 0, ctx);
return TRUE;
no_uri:
{
GST_ERROR ("client %p: no uri", client);
- send_generic_response (client, GST_RTSP_STS_BAD_REQUEST, state);
+ send_generic_response (client, GST_RTSP_STS_BAD_REQUEST, ctx);
+ return FALSE;
+ }
+no_mount_points:
+ {
+ GST_ERROR ("client %p: no mount points configured", client);
+ send_generic_response (client, GST_RTSP_STS_NOT_FOUND, ctx);
+ return FALSE;
+ }
+no_path:
+ {
+ GST_ERROR ("client %p: can't find path for url", client);
+ send_generic_response (client, GST_RTSP_STS_NOT_FOUND, ctx);
return FALSE;
}
no_media:
{
GST_ERROR ("client %p: no media", client);
+ g_free (path);
/* 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);
+ send_generic_response (client, GST_RTSP_STS_SERVICE_UNAVAILABLE, ctx);
+ g_free (path);
g_object_unref (media);
return FALSE;
}
}
static gboolean
-handle_options_request (GstRTSPClient * client, GstRTSPClientState * state)
+handle_options_request (GstRTSPClient * client, GstRTSPContext * ctx)
{
GstRTSPMethod options;
gchar *str;
str = gst_rtsp_options_as_text (options);
- gst_rtsp_message_init_response (state->response, GST_RTSP_STS_OK,
- gst_rtsp_status_as_text (GST_RTSP_STS_OK), state->request);
+ gst_rtsp_message_init_response (ctx->response, GST_RTSP_STS_OK,
+ gst_rtsp_status_as_text (GST_RTSP_STS_OK), ctx->request);
- gst_rtsp_message_add_header (state->response, GST_RTSP_HDR_PUBLIC, str);
+ gst_rtsp_message_add_header (ctx->response, GST_RTSP_HDR_PUBLIC, str);
g_free (str);
- send_message (client, state->session, state->response, FALSE);
+ send_message (client, ctx->session, ctx->response, FALSE);
g_signal_emit (client, gst_rtsp_client_signals[SIGNAL_OPTIONS_REQUEST],
- 0, state);
+ 0, ctx);
return TRUE;
}
}
}
-static GPrivate current_state;
-
-/**
- * gst_rtsp_client_state_get_current:
- *
- * Get the current #GstRTSPClientState. This object is retrieved from the
- * current thread that is handling the request for a client.
- *
- * Returns: a #GstRTSPClientState
- */
-GstRTSPClientState *
-gst_rtsp_client_state_get_current (void)
-{
- GSList *l;
-
- l = g_private_get (¤t_state);
- if (l == NULL)
- return NULL;
-
- return (GstRTSPClientState *) (l->data);
-
-}
-
-/**
- * gst_rtsp_client_state_push_current:
- * @state: a ##GstRTSPClientState
- *
- * Pushes @state onto the state stack. The current
- * state can then be received using gst_rtsp_client_state_get_current().
- **/
-void
-gst_rtsp_client_state_push_current (GstRTSPClientState * state)
-{
- GSList *l;
-
- g_return_if_fail (state != NULL);
-
- l = g_private_get (¤t_state);
- l = g_slist_prepend (l, state);
- g_private_set (¤t_state, l);
-}
-
-/**
- * gst_rtsp_client_state_pop_current:
- * @state: a #GstRTSPClientState
- *
- * Pops @state off the state stack (verifying that @state
- * is on the top of the stack).
- **/
-void
-gst_rtsp_client_state_pop_current (GstRTSPClientState * state)
-{
- GSList *l;
-
- l = g_private_get (¤t_state);
-
- g_return_if_fail (l != NULL);
- g_return_if_fail (l->data == state);
-
- l = g_slist_delete_link (l, l);
- g_private_set (¤t_state, l);
-}
-
static void
handle_request (GstRTSPClient * client, GstRTSPMessage * request)
{
GstRTSPVersion version;
GstRTSPResult res;
GstRTSPSession *session = NULL;
- GstRTSPClientState state = { NULL };
+ GstRTSPContext sctx = { NULL }, *ctx;
GstRTSPMessage response = { 0 };
gchar *sessid;
- state.conn = priv->connection;
- state.client = client;
- state.request = request;
- state.response = &response;
- state.auth = priv->auth;
- gst_rtsp_client_state_push_current (&state);
+ if (!(ctx = gst_rtsp_context_get_current ())) {
+ ctx = &sctx;
+ ctx->auth = priv->auth;
+ gst_rtsp_context_push_current (ctx);
+ }
+
+ ctx->conn = priv->connection;
+ ctx->client = client;
+ ctx->request = request;
+ ctx->response = &response;
if (gst_debug_category_get_threshold (rtsp_client_debug) >= GST_LEVEL_LOG) {
gst_rtsp_message_dump (request);
}
- GST_INFO ("client %p: received a request", client);
-
gst_rtsp_message_parse_request (request, &method, &uristr, &version);
+ GST_INFO ("client %p: received a request %s %s %s", client,
+ gst_rtsp_method_as_text (method), uristr,
+ gst_rtsp_version_as_text (version));
+
/* we can only handle 1.0 requests */
if (version != GST_RTSP_VERSION_1_0)
goto not_supported;
- state.method = method;
+ ctx->method = method;
/* we always try to parse the url first */
if (strcmp (uristr, "*") == 0) {
/* special case where we have * as uri, keep uri = NULL */
- } else if (gst_rtsp_url_parse (uristr, &uri) != GST_RTSP_OK)
- goto bad_request;
+ } else if (gst_rtsp_url_parse (uristr, &uri) != GST_RTSP_OK) {
+ /* check if the uristr is an absolute path <=> scheme and host information
+ * is missing */
+ gchar *scheme;
+
+ scheme = g_uri_parse_scheme (uristr);
+ if (scheme == NULL && g_str_has_prefix (uristr, "/")) {
+ gchar *absolute_uristr = NULL;
+
+ GST_WARNING_OBJECT (client, "request doesn't contain absolute url");
+ if (priv->server_ip == NULL) {
+ GST_WARNING_OBJECT (client, "host information missing");
+ goto bad_request;
+ }
+
+ absolute_uristr =
+ g_strdup_printf ("rtsp://%s%s", priv->server_ip, uristr);
+
+ GST_DEBUG_OBJECT (client, "absolute url: %s", absolute_uristr);
+ if (gst_rtsp_url_parse (absolute_uristr, &uri) != GST_RTSP_OK) {
+ g_free (absolute_uristr);
+ goto bad_request;
+ }
+ g_free (absolute_uristr);
+ } else {
+ g_free (scheme);
+ goto bad_request;
+ }
+ }
/* get the session if there is any */
res = gst_rtsp_message_get_header (request, GST_RTSP_HDR_SESSION, &sessid, 0);
/* sanitize the uri */
if (uri)
sanitize_uri (uri);
- state.uri = uri;
- state.session = session;
+ ctx->uri = uri;
+ ctx->session = session;
if (!gst_rtsp_auth_check (GST_RTSP_AUTH_CHECK_URL))
goto not_authorized;
/* now see what is asked and dispatch to a dedicated handler */
switch (method) {
case GST_RTSP_OPTIONS:
- handle_options_request (client, &state);
+ handle_options_request (client, ctx);
break;
case GST_RTSP_DESCRIBE:
- handle_describe_request (client, &state);
+ handle_describe_request (client, ctx);
break;
case GST_RTSP_SETUP:
- handle_setup_request (client, &state);
+ handle_setup_request (client, ctx);
break;
case GST_RTSP_PLAY:
- handle_play_request (client, &state);
+ handle_play_request (client, ctx);
break;
case GST_RTSP_PAUSE:
- handle_pause_request (client, &state);
+ handle_pause_request (client, ctx);
break;
case GST_RTSP_TEARDOWN:
- handle_teardown_request (client, &state);
+ handle_teardown_request (client, ctx);
break;
case GST_RTSP_SET_PARAMETER:
- handle_set_param_request (client, &state);
+ handle_set_param_request (client, ctx);
break;
case GST_RTSP_GET_PARAMETER:
- handle_get_param_request (client, &state);
+ handle_get_param_request (client, ctx);
break;
case GST_RTSP_ANNOUNCE:
case GST_RTSP_RECORD:
}
done:
- gst_rtsp_client_state_pop_current (&state);
+ if (ctx == &sctx)
+ gst_rtsp_context_pop_current (ctx);
if (session)
g_object_unref (session);
if (uri)
{
GST_ERROR ("client %p: version %d not supported", client, version);
send_generic_response (client, GST_RTSP_STS_RTSP_VERSION_NOT_SUPPORTED,
- &state);
+ ctx);
goto done;
}
bad_request:
{
GST_ERROR ("client %p: bad request", client);
- send_generic_response (client, GST_RTSP_STS_BAD_REQUEST, &state);
+ send_generic_response (client, GST_RTSP_STS_BAD_REQUEST, ctx);
goto done;
}
no_pool:
{
GST_ERROR ("client %p: no pool configured", client);
- send_generic_response (client, GST_RTSP_STS_SESSION_NOT_FOUND, &state);
+ send_generic_response (client, GST_RTSP_STS_SESSION_NOT_FOUND, ctx);
goto done;
}
session_not_found:
{
GST_ERROR ("client %p: session not found", client);
- send_generic_response (client, GST_RTSP_STS_SESSION_NOT_FOUND, &state);
+ send_generic_response (client, GST_RTSP_STS_SESSION_NOT_FOUND, ctx);
goto done;
}
not_authorized:
{
GST_ERROR ("client %p: not allowed", client);
+ /* error reply is already sent */
goto done;
}
not_implemented:
{
GST_ERROR ("client %p: method %d not implemented", client, method);
- send_generic_response (client, GST_RTSP_STS_NOT_IMPLEMENTED, &state);
+ send_generic_response (client, GST_RTSP_STS_NOT_IMPLEMENTED, ctx);
+ goto done;
+ }
+}
+
+
+static void
+handle_response (GstRTSPClient * client, GstRTSPMessage * response)
+{
+ GstRTSPClientPrivate *priv = client->priv;
+ GstRTSPResult res;
+ GstRTSPSession *session = NULL;
+ GstRTSPContext sctx = { NULL }, *ctx;
+ gchar *sessid;
+
+ if (!(ctx = gst_rtsp_context_get_current ())) {
+ ctx = &sctx;
+ ctx->auth = priv->auth;
+ gst_rtsp_context_push_current (ctx);
+ }
+
+ ctx->conn = priv->connection;
+ ctx->client = client;
+ ctx->request = NULL;
+ ctx->uri = NULL;
+ ctx->method = GST_RTSP_INVALID;
+ ctx->response = response;
+
+ if (gst_debug_category_get_threshold (rtsp_client_debug) >= GST_LEVEL_LOG) {
+ gst_rtsp_message_dump (response);
+ }
+
+ GST_INFO ("client %p: received a response", client);
+
+ /* get the session if there is any */
+ res =
+ gst_rtsp_message_get_header (response, GST_RTSP_HDR_SESSION, &sessid, 0);
+ if (res == GST_RTSP_OK) {
+ 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 (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);
+ }
+
+ ctx->session = session;
+
+ g_signal_emit (client, gst_rtsp_client_signals[SIGNAL_HANDLE_RESPONSE],
+ 0, ctx);
+
+done:
+ if (ctx == &sctx)
+ gst_rtsp_context_pop_current (ctx);
+ if (session)
+ g_object_unref (session);
+ return;
+
+no_pool:
+ {
+ GST_ERROR ("client %p: no pool configured", client);
+ goto done;
+ }
+session_not_found:
+ {
+ GST_ERROR ("client %p: session not found", client);
goto done;
}
}
handle_request (client, message);
break;
case GST_RTSP_MESSAGE_RESPONSE:
+ handle_response (client, message);
break;
case GST_RTSP_MESSAGE_DATA:
handle_data (client, message);
* @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
+ * 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
/**
* gst_rtsp_client_session_filter:
- * @client: a #GstRTSPclient
- * @func: (scope call): a callback
+ * @client: a #GstRTSPClient
+ * @func: (scope call) (allow-none): a callback
* @user_data: user data passed to @func
*
* Call @func for each session managed by @client. The result value of @func
* will also be added with an additional ref to the result #GList of this
* function..
*
+ * When @func is %NULL, #GST_RTSP_FILTER_REF will be assumed for each session.
+ *
* Returns: (element-type GstRTSPSession) (transfer full): a #GList with all
* sessions for which @func returned #GST_RTSP_FILTER_REF. After usage, each
* element in the #GList should be unreffed before the list is freed.
GList *result, *walk, *next;
g_return_val_if_fail (GST_IS_RTSP_CLIENT (client), NULL);
- g_return_val_if_fail (func != NULL, NULL);
priv = client->priv;
g_mutex_lock (&priv->lock);
for (walk = priv->sessions; walk; walk = next) {
GstRTSPSession *sess = walk->data;
+ GstRTSPFilterResult res;
next = g_list_next (walk);
- switch (func (client, sess, user_data)) {
+ if (func)
+ res = func (client, sess, user_data);
+ else
+ res = GST_RTSP_FILTER_REF;
+
+ switch (res) {
case GST_RTSP_FILTER_REMOVE:
/* stop watching the session and pretent it went away */
client_cleanup_session (client, sess);
projects / platform / upstream / gstreamer.git / blobdiff