#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;
enum
PROP_LAST
};
+enum
+{
+ SIGNAL_CLOSED,
+ SIGNAL_LAST
+};
+
GST_DEBUG_CATEGORY_STATIC (rtsp_client_debug);
#define GST_CAT_DEFAULT rtsp_client_debug
+static guint gst_rtsp_client_signals[SIGNAL_LAST] = { 0 };
+
static void gst_rtsp_client_get_property (GObject * object, guint propid,
GValue * value, GParamSpec * pspec);
static void gst_rtsp_client_set_property (GObject * object, guint propid,
GST_TYPE_RTSP_MEDIA_MAPPING,
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);
+
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
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)) {
- unlink_session_streams (client, session,
- (GstRTSPSessionMedia *) medias->data);
+ while (g_list_length (session->medias) > 0) {
+ GstRTSPSessionMedia *media = g_list_first (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. this will modify session->medias */
+ gst_rtsp_session_release_media (session, media);
}
}
g_object_unref (client->session_pool);
if (client->media_mapping)
g_object_unref (client->media_mapping);
+ if (client->auth)
+ g_object_unref (client->auth);
if (client->uri)
gst_rtsp_url_free (client->uri);
static void
send_generic_response (GstRTSPClient * client, GstRTSPStatusCode code,
- GstRTSPMessage * request)
+ GstRTSPClientState * state)
{
- GstRTSPMessage response = { 0 };
+ gst_rtsp_message_init_response (state->response, code,
+ gst_rtsp_status_as_text (code), state->request);
- gst_rtsp_message_init_response (&response, code,
- gst_rtsp_status_as_text (code), request);
-
- send_response (client, NULL, &response);
+ send_response (client, NULL, state->response);
}
static void
-handle_unauthorized_request (GstRTSPClient * client, GstRTSPUrl * uri,
- GstRTSPSession * session, GstRTSPMessage * request)
+handle_unauthorized_request (GstRTSPClient * client, GstRTSPAuth * auth,
+ GstRTSPClientState * state)
{
- GstRTSPMessage response = { 0 };
+ gst_rtsp_message_init_response (state->response, GST_RTSP_STS_UNAUTHORIZED,
+ gst_rtsp_status_as_text (GST_RTSP_STS_UNAUTHORIZED), state->request);
- gst_rtsp_message_init_response (&response, GST_RTSP_STS_UNAUTHORIZED,
- gst_rtsp_status_as_text (GST_RTSP_STS_UNAUTHORIZED), request);
-
- if (client->auth) {
+ if (auth) {
/* and let the authentication manager setup the auth tokens */
- gst_rtsp_auth_setup_auth (client->auth, client, uri, session, request,
- &response);
+ gst_rtsp_auth_setup_auth (auth, client, 0, state);
}
- send_response (client, session, &response);
+ send_response (client, state->session, state->response);
}
* 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, GstRTSPUrl * uri, GstRTSPMessage * request)
+find_media (GstRTSPClient * client, GstRTSPClientState * state)
{
GstRTSPMediaFactory *factory;
GstRTSPMedia *media;
+ GstRTSPAuth *auth;
- if (!compare_uri (client->uri, uri)) {
+ if (!compare_uri (client->uri, state->uri)) {
/* remove any previously cached values before we try to construct a new
* media for uri */
if (client->uri)
/* find the factory for the uri first */
if (!(factory =
- gst_rtsp_media_mapping_find_factory (client->media_mapping, uri)))
+ gst_rtsp_media_mapping_find_factory (client->media_mapping,
+ 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))) {
+ if (!gst_rtsp_auth_check (auth, client, 0, state))
+ goto not_allowed;
+
+ g_object_unref (auth);
+ }
+
/* prepare the media and add it to the pipeline */
- if (!(media = gst_rtsp_media_factory_construct (factory, uri)))
+ if (!(media = gst_rtsp_media_factory_construct (factory, state->uri)))
goto no_media;
+ 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 (uri);
+ client->uri = gst_rtsp_url_copy (state->uri);
client->media = media;
} else {
/* we have seen this uri before, used cached media */
media = client->media;
+ state->media = media;
GST_INFO ("reusing cached media %p", media);
}
/* ERRORS */
no_mapping:
{
- send_generic_response (client, GST_RTSP_STS_NOT_FOUND, request);
+ send_generic_response (client, GST_RTSP_STS_NOT_FOUND, state);
return NULL;
}
no_factory:
{
- send_generic_response (client, GST_RTSP_STS_NOT_FOUND, request);
+ send_generic_response (client, GST_RTSP_STS_NOT_FOUND, state);
+ return NULL;
+ }
+not_allowed:
+ {
+ handle_unauthorized_request (client, auth, state);
+ g_object_unref (factory);
+ g_object_unref (auth);
return NULL;
}
no_media:
{
- send_generic_response (client, GST_RTSP_STS_SERVICE_UNAVAILABLE, request);
+ send_generic_response (client, GST_RTSP_STS_SERVICE_UNAVAILABLE, state);
g_object_unref (factory);
return NULL;
}
no_prepare:
{
- send_generic_response (client, GST_RTSP_STS_SERVICE_UNAVAILABLE, request);
+ send_generic_response (client, GST_RTSP_STS_SERVICE_UNAVAILABLE, state);
g_object_unref (media);
- g_object_unref (factory);
return 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);
+ 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_unset (&message);
+ gst_rtsp_message_steal_body (&message, &data, &usize);
+ gst_buffer_unmap (buffer, &map_info);
- return TRUE;
-}
-
-static gboolean
-do_send_data_list (GstBufferList * blist, guint8 channel,
- GstRTSPClient * client)
-{
- GstBufferListIterator *it;
-
- it = gst_buffer_list_iterate (blist);
- while (gst_buffer_list_iterator_next_group (it)) {
- GstBuffer *group = gst_buffer_list_iterator_merge_group (it);
-
- if (group == NULL)
- continue;
-
- do_send_data (group, channel, client);
- }
- gst_buffer_list_iterator_free (it);
+ gst_rtsp_message_unset (&message);
return TRUE;
}
{
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);
+ (GstRTSPSendFunc) do_send_data, client, NULL);
client->streams = g_list_prepend (client->streams, stream);
/* make sure our session can't expire */
gst_rtsp_session_prevent_expire (session);
GstRTSPSessionStream * stream)
{
GST_DEBUG ("client %p: unlinking stream %p", client, stream);
- gst_rtsp_session_stream_set_callbacks (stream, NULL, NULL, NULL, NULL, NULL,
- NULL);
+ gst_rtsp_session_stream_set_callbacks (stream, NULL, NULL, NULL, NULL);
client->streams = g_list_remove (client->streams, stream);
/* our session can now expire */
gst_rtsp_session_allow_expire (session);
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);
}
static gboolean
-handle_teardown_request (GstRTSPClient * client, GstRTSPUrl * uri,
- GstRTSPSession * session, GstRTSPMessage * request)
+handle_teardown_request (GstRTSPClient * client, GstRTSPClientState * state)
{
+ GstRTSPSession *session;
GstRTSPSessionMedia *media;
- GstRTSPMessage response = { 0 };
GstRTSPStatusCode code;
- if (!session)
+ if (!state->session)
goto no_session;
+ session = state->session;
+
/* get a handle to the configuration of the media in the session */
- media = gst_rtsp_session_get_media (session, uri);
+ media = gst_rtsp_session_get_media (session, state->uri);
if (!media)
goto not_found;
+ state->sessmedia = media;
+
/* unlink the all TCP callbacks */
unlink_session_streams (client, session, media);
}
/* construct the response now */
code = GST_RTSP_STS_OK;
- gst_rtsp_message_init_response (&response, code,
- gst_rtsp_status_as_text (code), request);
+ gst_rtsp_message_init_response (state->response, code,
+ gst_rtsp_status_as_text (code), state->request);
- gst_rtsp_message_add_header (&response, GST_RTSP_HDR_CONNECTION, "close");
+ gst_rtsp_message_add_header (state->response, GST_RTSP_HDR_CONNECTION,
+ "close");
- send_response (client, session, &response);
+ send_response (client, session, state->response);
close_connection (client);
/* ERRORS */
no_session:
{
- send_generic_response (client, GST_RTSP_STS_SESSION_NOT_FOUND, request);
+ send_generic_response (client, GST_RTSP_STS_SESSION_NOT_FOUND, state);
return FALSE;
}
not_found:
{
- send_generic_response (client, GST_RTSP_STS_NOT_FOUND, request);
+ send_generic_response (client, GST_RTSP_STS_NOT_FOUND, state);
return FALSE;
}
}
static gboolean
-handle_get_param_request (GstRTSPClient * client, GstRTSPUrl * uri,
- GstRTSPSession * session, GstRTSPMessage * request)
+handle_get_param_request (GstRTSPClient * client, GstRTSPClientState * state)
{
GstRTSPResult res;
guint8 *data;
guint size;
- res = gst_rtsp_message_get_body (request, &data, &size);
+ res = gst_rtsp_message_get_body (state->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, request);
+ send_generic_response (client, GST_RTSP_STS_OK, state);
} else {
- /* there is a body */
- GstRTSPMessage response = { 0 };
-
/* there is a body, handle the params */
- res = gst_rtsp_params_get (client, uri, session, request, &response);
+ res = gst_rtsp_params_get (client, state);
if (res != GST_RTSP_OK)
goto bad_request;
- send_response (client, session, &response);
+ send_response (client, state->session, state->response);
}
return TRUE;
/* ERRORS */
bad_request:
{
- send_generic_response (client, GST_RTSP_STS_BAD_REQUEST, request);
+ send_generic_response (client, GST_RTSP_STS_BAD_REQUEST, state);
return FALSE;
}
}
static gboolean
-handle_set_param_request (GstRTSPClient * client, GstRTSPUrl * uri,
- GstRTSPSession * session, GstRTSPMessage * request)
+handle_set_param_request (GstRTSPClient * client, GstRTSPClientState * state)
{
GstRTSPResult res;
guint8 *data;
guint size;
- res = gst_rtsp_message_get_body (request, &data, &size);
+ res = gst_rtsp_message_get_body (state->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, request);
+ send_generic_response (client, GST_RTSP_STS_OK, state);
} else {
- GstRTSPMessage response = { 0 };
-
/* there is a body, handle the params */
- res = gst_rtsp_params_set (client, uri, session, request, &response);
+ res = gst_rtsp_params_set (client, state);
if (res != GST_RTSP_OK)
goto bad_request;
- send_response (client, session, &response);
+ send_response (client, state->session, state->response);
}
return TRUE;
/* ERRORS */
bad_request:
{
- send_generic_response (client, GST_RTSP_STS_BAD_REQUEST, request);
+ send_generic_response (client, GST_RTSP_STS_BAD_REQUEST, state);
return FALSE;
}
}
static gboolean
-handle_pause_request (GstRTSPClient * client, GstRTSPUrl * uri,
- GstRTSPSession * session, GstRTSPMessage * request)
+handle_pause_request (GstRTSPClient * client, GstRTSPClientState * state)
{
+ GstRTSPSession *session;
GstRTSPSessionMedia *media;
- GstRTSPMessage response = { 0 };
GstRTSPStatusCode code;
- if (!session)
+ if (!(session = state->session))
goto no_session;
/* get a handle to the configuration of the media in the session */
- media = gst_rtsp_session_get_media (session, uri);
+ media = gst_rtsp_session_get_media (session, state->uri);
if (!media)
goto not_found;
+ state->sessmedia = media;
+
/* the session state must be playing or recording */
if (media->state != GST_RTSP_STATE_PLAYING &&
media->state != GST_RTSP_STATE_RECORDING)
/* construct the response now */
code = GST_RTSP_STS_OK;
- gst_rtsp_message_init_response (&response, code,
- gst_rtsp_status_as_text (code), request);
+ gst_rtsp_message_init_response (state->response, code,
+ gst_rtsp_status_as_text (code), state->request);
- send_response (client, session, &response);
+ send_response (client, session, state->response);
/* the state is now READY */
media->state = GST_RTSP_STATE_READY;
/* ERRORS */
no_session:
{
- send_generic_response (client, GST_RTSP_STS_SESSION_NOT_FOUND, request);
+ send_generic_response (client, GST_RTSP_STS_SESSION_NOT_FOUND, state);
return FALSE;
}
not_found:
{
- send_generic_response (client, GST_RTSP_STS_NOT_FOUND, request);
+ send_generic_response (client, GST_RTSP_STS_NOT_FOUND, state);
return FALSE;
}
invalid_state:
{
send_generic_response (client, GST_RTSP_STS_METHOD_NOT_VALID_IN_THIS_STATE,
- request);
+ state);
return FALSE;
}
}
static gboolean
-handle_play_request (GstRTSPClient * client, GstRTSPUrl * uri,
- GstRTSPSession * session, GstRTSPMessage * request)
+handle_play_request (GstRTSPClient * client, GstRTSPClientState * state)
{
+ GstRTSPSession *session;
GstRTSPSessionMedia *media;
- GstRTSPMessage response = { 0 };
GstRTSPStatusCode code;
GString *rtpinfo;
guint n_streams, i, infocount;
GstRTSPTimeRange *range;
GstRTSPResult res;
- if (!session)
+ if (!(session = state->session))
goto no_session;
/* get a handle to the configuration of the media in the session */
- media = gst_rtsp_session_get_media (session, uri);
+ media = gst_rtsp_session_get_media (session, state->uri);
if (!media)
goto not_found;
+ state->sessmedia = media;
+
/* the session state must be playing or ready */
if (media->state != GST_RTSP_STATE_PLAYING &&
media->state != GST_RTSP_STATE_READY)
goto invalid_state;
/* parse the range header if we have one */
- res = gst_rtsp_message_get_header (request, GST_RTSP_HDR_RANGE, &str, 0);
+ res =
+ gst_rtsp_message_get_header (state->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 */
if (infocount > 0)
g_string_append (rtpinfo, ", ");
- uristr = gst_rtsp_url_get_request_uri (uri);
+ 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);
g_free (uristr);
/* construct the response now */
code = GST_RTSP_STS_OK;
- gst_rtsp_message_init_response (&response, code,
- gst_rtsp_status_as_text (code), request);
+ gst_rtsp_message_init_response (state->response, code,
+ gst_rtsp_status_as_text (code), state->request);
/* add the RTP-Info header */
if (infocount > 0) {
str = g_string_free (rtpinfo, FALSE);
- gst_rtsp_message_take_header (&response, GST_RTSP_HDR_RTP_INFO, str);
+ gst_rtsp_message_take_header (state->response, GST_RTSP_HDR_RTP_INFO, str);
} else {
g_string_free (rtpinfo, TRUE);
}
/* add the range */
str = gst_rtsp_media_get_range_string (media->media, TRUE);
- gst_rtsp_message_take_header (&response, GST_RTSP_HDR_RANGE, str);
+ gst_rtsp_message_take_header (state->response, GST_RTSP_HDR_RANGE, str);
- send_response (client, session, &response);
+ send_response (client, session, state->response);
/* start playing after sending the request */
gst_rtsp_session_media_set_state (media, GST_STATE_PLAYING);
/* ERRORS */
no_session:
{
- send_generic_response (client, GST_RTSP_STS_SESSION_NOT_FOUND, request);
+ send_generic_response (client, GST_RTSP_STS_SESSION_NOT_FOUND, state);
return FALSE;
}
not_found:
{
- send_generic_response (client, GST_RTSP_STS_NOT_FOUND, request);
+ send_generic_response (client, GST_RTSP_STS_NOT_FOUND, state);
return FALSE;
}
invalid_state:
{
send_generic_response (client, GST_RTSP_STS_METHOD_NOT_VALID_IN_THIS_STATE,
- request);
+ state);
return FALSE;
}
}
}
static gboolean
-handle_setup_request (GstRTSPClient * client, GstRTSPUrl * uri,
- GstRTSPSession * session, GstRTSPMessage * request)
+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;
- GstRTSPMessage response = { 0 };
GstRTSPStatusCode code;
+ GstRTSPSession *session;
GstRTSPSessionStream *stream;
gchar *trans_str, *pos;
guint streamid;
GstRTSPSessionMedia *media;
- GstRTSPUrl *url;
+
+ 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
/* parse the transport */
res =
- gst_rtsp_message_get_header (request, GST_RTSP_HDR_TRANSPORT, &transport,
- 0);
+ gst_rtsp_message_get_header (state->request, GST_RTSP_HDR_TRANSPORT,
+ &transport, 0);
if (res != GST_RTSP_OK)
goto no_transport;
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);
if (!(session = gst_rtsp_session_pool_create (client->session_pool)))
goto service_unavailable;
+ state->session = session;
+
/* we need a new media configuration in this session */
media = NULL;
}
GstRTSPMedia *m;
/* get a handle to the configuration of the media in the session */
- if ((m = find_media (client, uri, request))) {
+ if ((m = find_media (client, state))) {
/* manage the media in our session now */
media = gst_rtsp_session_manage_media (session, uri, m);
}
if (media == NULL)
goto not_found;
- /* 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);
+ state->sessmedia = media;
+
+ /* 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 = gst_rtsp_media_get_multicast_group (media->media);
+ } else {
+ GstRTSPUrl *url;
+
+ url = gst_rtsp_connection_get_url (client->connection);
+ ct->destination = g_strdup (url->host);
+
+ if (ct->lower_transport & GST_RTSP_LOWER_TRANS_TCP) {
+ /* check if the client selected channels for TCP */
+ if (ct->interleaved.min == -1 || ct->interleaved.max == -1) {
+ gst_rtsp_session_media_alloc_channels (media, &ct->interleaved);
+ }
}
}
/* construct the response now */
code = GST_RTSP_STS_OK;
- gst_rtsp_message_init_response (&response, code,
- gst_rtsp_status_as_text (code), request);
+ gst_rtsp_message_init_response (state->response, code,
+ gst_rtsp_status_as_text (code), state->request);
- gst_rtsp_message_add_header (&response, GST_RTSP_HDR_TRANSPORT, trans_str);
+ gst_rtsp_message_add_header (state->response, GST_RTSP_HDR_TRANSPORT,
+ trans_str);
g_free (trans_str);
- send_response (client, session, &response);
+ send_response (client, session, state->response);
/* update the state */
switch (media->state) {
/* ERRORS */
bad_request:
{
- send_generic_response (client, GST_RTSP_STS_BAD_REQUEST, request);
+ send_generic_response (client, GST_RTSP_STS_BAD_REQUEST, state);
return FALSE;
}
not_found:
{
- send_generic_response (client, GST_RTSP_STS_NOT_FOUND, request);
+ send_generic_response (client, GST_RTSP_STS_NOT_FOUND, state);
g_object_unref (session);
return FALSE;
}
no_stream:
{
- send_generic_response (client, GST_RTSP_STS_NOT_FOUND, request);
- g_object_unref (media);
+ send_generic_response (client, GST_RTSP_STS_NOT_FOUND, state);
g_object_unref (session);
+ gst_rtsp_transport_free (ct);
return FALSE;
}
no_transport:
{
- send_generic_response (client, GST_RTSP_STS_UNSUPPORTED_TRANSPORT, request);
+ send_generic_response (client, GST_RTSP_STS_UNSUPPORTED_TRANSPORT, state);
return FALSE;
}
unsupported_transports:
{
- send_generic_response (client, GST_RTSP_STS_UNSUPPORTED_TRANSPORT, request);
+ 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, request);
+ send_generic_response (client, GST_RTSP_STS_SERVICE_UNAVAILABLE, state);
return FALSE;
}
service_unavailable:
{
- send_generic_response (client, GST_RTSP_STS_SERVICE_UNAVAILABLE, request);
+ send_generic_response (client, GST_RTSP_STS_SERVICE_UNAVAILABLE, state);
return FALSE;
}
}
GstSDPMessage *sdp;
GstSDPInfo info;
const gchar *proto;
+ GstRTSPLowerTrans protocols;
gst_sdp_message_new (&sdp);
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;
+ protocols = gst_rtsp_media_get_protocols (media);
+ if (protocols & GST_RTSP_LOWER_TRANS_UDP_MCAST)
+ info.server_ip = gst_rtsp_media_get_multicast_group (media);
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;
}
/* for the describe we must generate an SDP */
static gboolean
-handle_describe_request (GstRTSPClient * client, GstRTSPUrl * uri,
- GstRTSPSession * session, GstRTSPMessage * request)
+handle_describe_request (GstRTSPClient * client, GstRTSPClientState * state)
{
- GstRTSPMessage response = { 0 };
GstRTSPResult res;
GstSDPMessage *sdp;
guint i, str_len;
gchar *accept;
res =
- gst_rtsp_message_get_header (request, GST_RTSP_HDR_ACCEPT, &accept, i);
+ gst_rtsp_message_get_header (state->request, GST_RTSP_HDR_ACCEPT,
+ &accept, i);
if (res == GST_RTSP_ENOTIMPL)
break;
}
/* find the media object for the uri */
- if (!(media = find_media (client, uri, request)))
+ if (!(media = find_media (client, state)))
goto no_media;
/* create an SDP for the media object on this client */
g_object_unref (media);
- gst_rtsp_message_init_response (&response, GST_RTSP_STS_OK,
- gst_rtsp_status_as_text (GST_RTSP_STS_OK), request);
+ 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_add_header (&response, GST_RTSP_HDR_CONTENT_TYPE,
+ gst_rtsp_message_add_header (state->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 (uri);
+ str = gst_rtsp_url_get_request_uri (state->uri);
str_len = strlen (str);
/* check for trailing '/' and append one */
GST_INFO ("adding content-base: %s", content_base);
- gst_rtsp_message_add_header (&response, GST_RTSP_HDR_CONTENT_BASE,
+ gst_rtsp_message_add_header (state->response, GST_RTSP_HDR_CONTENT_BASE,
content_base);
g_free (content_base);
/* add SDP to the response body */
str = gst_sdp_message_as_text (sdp);
- gst_rtsp_message_take_body (&response, (guint8 *) str, strlen (str));
+ gst_rtsp_message_take_body (state->response, (guint8 *) str, strlen (str));
gst_sdp_message_free (sdp);
- send_response (client, session, &response);
+ send_response (client, state->session, state->response);
return TRUE;
}
no_sdp:
{
- send_generic_response (client, GST_RTSP_STS_SERVICE_UNAVAILABLE, request);
+ send_generic_response (client, GST_RTSP_STS_SERVICE_UNAVAILABLE, state);
g_object_unref (media);
return FALSE;
}
}
static gboolean
-handle_options_request (GstRTSPClient * client, GstRTSPUrl * uri,
- GstRTSPSession * session, GstRTSPMessage * request)
+handle_options_request (GstRTSPClient * client, GstRTSPClientState * state)
{
- GstRTSPMessage response = { 0 };
GstRTSPMethod options;
gchar *str;
str = gst_rtsp_options_as_text (options);
- gst_rtsp_message_init_response (&response, GST_RTSP_STS_OK,
- gst_rtsp_status_as_text (GST_RTSP_STS_OK), request);
+ 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_add_header (&response, GST_RTSP_HDR_PUBLIC, str);
+ gst_rtsp_message_add_header (state->response, GST_RTSP_HDR_PUBLIC, str);
g_free (str);
- send_response (client, session, &response);
+ send_response (client, state->session, state->response);
return TRUE;
}
GstRTSPVersion version;
GstRTSPResult res;
GstRTSPSession *session;
+ GstRTSPClientState state = { NULL };
+ GstRTSPMessage response = { 0 };
gchar *sessid;
+ state.request = request;
+ state.response = &response;
+
if (gst_debug_category_get_threshold (rtsp_client_debug) >= GST_LEVEL_LOG) {
gst_rtsp_message_dump (request);
}
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,
- request);
+ &state);
return;
}
+ 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, request);
+ send_generic_response (client, GST_RTSP_STS_BAD_REQUEST, &state);
return;
}
/* sanitize the uri */
sanitize_uri (uri);
+ state.uri = uri;
/* get the session if there is any */
res = gst_rtsp_message_get_header (request, GST_RTSP_HDR_SESSION, &sessid, 0);
} else
session = NULL;
+ state.session = session;
+
if (client->auth) {
- if (!gst_rtsp_auth_check_method (client->auth, method, client, uri, session,
- request))
+ if (!gst_rtsp_auth_check (client->auth, client, 0, &state))
goto not_authorized;
}
/* now see what is asked and dispatch to a dedicated handler */
switch (method) {
case GST_RTSP_OPTIONS:
- handle_options_request (client, uri, session, request);
+ handle_options_request (client, &state);
break;
case GST_RTSP_DESCRIBE:
- handle_describe_request (client, uri, session, request);
+ handle_describe_request (client, &state);
break;
case GST_RTSP_SETUP:
- handle_setup_request (client, uri, session, request);
+ handle_setup_request (client, &state);
break;
case GST_RTSP_PLAY:
- handle_play_request (client, uri, session, request);
+ handle_play_request (client, &state);
break;
case GST_RTSP_PAUSE:
- handle_pause_request (client, uri, session, request);
+ handle_pause_request (client, &state);
break;
case GST_RTSP_TEARDOWN:
- handle_teardown_request (client, uri, session, request);
+ handle_teardown_request (client, &state);
break;
case GST_RTSP_SET_PARAMETER:
- handle_set_param_request (client, uri, session, request);
+ handle_set_param_request (client, &state);
break;
case GST_RTSP_GET_PARAMETER:
- handle_get_param_request (client, uri, session, request);
+ handle_get_param_request (client, &state);
break;
case GST_RTSP_ANNOUNCE:
case GST_RTSP_RECORD:
case GST_RTSP_REDIRECT:
- send_generic_response (client, GST_RTSP_STS_NOT_IMPLEMENTED, request);
+ send_generic_response (client, GST_RTSP_STS_NOT_IMPLEMENTED, &state);
break;
case GST_RTSP_INVALID:
default:
- send_generic_response (client, GST_RTSP_STS_BAD_REQUEST, request);
+ send_generic_response (client, GST_RTSP_STS_BAD_REQUEST, &state);
break;
}
if (session)
/* ERRORS */
no_pool:
{
- send_generic_response (client, GST_RTSP_STS_SERVICE_UNAVAILABLE, request);
+ send_generic_response (client, GST_RTSP_STS_SERVICE_UNAVAILABLE, &state);
return;
}
session_not_found:
{
- send_generic_response (client, GST_RTSP_STS_SESSION_NOT_FOUND, request);
+ send_generic_response (client, GST_RTSP_STS_SESSION_NOT_FOUND, &state);
return;
}
not_authorized:
{
- handle_unauthorized_request (client, uri, session, request);
+ handle_unauthorized_request (client, client->auth, &state);
return;
}
}
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)) {
}
/**
+ * 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);
+ }
+}
+
+/**
+ * gst_rtsp_client_get_server:
+ * @client: a #GstRTSPClient
+ *
+ * Get the #GstRTSPServer object that @client was created from.
+ *
+ * Returns: a #GstRTSPServer, unref after usage.
+ */
+GstRTSPServer *
+gst_rtsp_client_get_server (GstRTSPClient * client)
+{
+ GstRTSPServer *result;
+
+ if ((result = client->server))
+ g_object_ref (result);
+
+ return result;
+}
+
+/**
* gst_rtsp_client_set_media_mapping:
* @client: a #GstRTSPClient
* @mapping: a #GstRTSPMediaMapping
static GstRTSPResult
message_sent (GstRTSPWatch * watch, guint cseq, gpointer user_data)
{
- GstRTSPClient *client;
+ /* GstRTSPClient *client; */
- client = GST_RTSP_CLIENT (user_data);
+ /* client = GST_RTSP_CLIENT (user_data); */
/* GST_INFO ("client %p: sent a message with cseq %d", client, cseq); */
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);
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;
}
}
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
+ * @socket: a #GSocket
+ * @cancellable: a #GCancellable
+ * @error: a #GError
*
- * Accept a new connection for @client on the socket in @channel.
+ * Accept a new connection for @client on @socket.
*
* 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)
+gst_rtsp_client_accept (GstRTSPClient * client, GSocket * socket,
+ GCancellable * cancellable, GError ** error)
{
- int sock, fd;
GstRTSPConnection *conn;
GstRTSPResult res;
+ GSocket *read_socket;
+ GSocketAddress *addres;
GSource *source;
GMainContext *context;
GstRTSPUrl *url;
gchar ip[INET6_ADDRSTRLEN];
/* a new client connected. */
- sock = g_io_channel_unix_get_fd (channel);
+ GST_RTSP_CHECK (gst_rtsp_connection_accept (socket, &conn, cancellable),
+ accept_failed);
- GST_RTSP_CHECK (gst_rtsp_connection_accept (sock, &conn), accept_failed);
+ read_socket = gst_rtsp_connection_get_read_socket (conn);
+ client->is_ipv6 = g_socket_get_family (socket) == G_SOCKET_FAMILY_IPV6;
- fd = gst_rtsp_connection_get_readfd (conn);
+ if (!(addres = g_socket_get_remote_address (read_socket, error)))
+ goto no_address;
addrlen = sizeof (addr);
- if (getsockname (fd, (struct sockaddr *) &addr, &addrlen) < 0)
- goto getpeername_failed;
-
- client->is_ipv6 = addr.ss_family == AF_INET6;
+ if (!g_socket_address_to_native (addres, &addr, addrlen, error))
+ goto native_failed;
+ g_object_unref (addres);
if (getnameinfo ((struct sockaddr *) &addr, addrlen, ip, sizeof (ip), NULL, 0,
NI_NUMERICHOST) != 0)
{
gchar *str = gst_rtsp_strresult (res);
- GST_ERROR ("Could not accept client on server socket %d: %s", sock, str);
+ GST_ERROR ("Could not accept client on server socket %p: %s", socket, str);
g_free (str);
return FALSE;
}
-getpeername_failed:
+no_address:
+ {
+ GST_ERROR ("could not get remote address %s", (*error)->message);
+ return FALSE;
+ }
+native_failed:
{
- GST_ERROR ("getpeername failed: %s", g_strerror (errno));
+ g_object_unref (addres);
+ GST_ERROR ("could not get native address %s", (*error)->message);
return FALSE;
}
getnameinfo_failed:
projects / platform / upstream / gstreamer.git / blobdiff