mirror of
https://gitlab.freedesktop.org/gstreamer/gstreamer.git
synced 2024-09-29 23:32:33 +00:00
Use ASYNC RTSP io
Use the async RTSP channels instead of spawning a new thread for each client. If a sessionid is specified in a request, fail if we don't have the session.
This commit is contained in:
Wim Taymans
parent
b70a6c9d83
commit
39c2e31e65
2 changed files with 170 additions and 201 deletions
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@ -184,7 +184,10 @@ send_response (GstRTSPClient *client, GstRTSPSession *session, GstRTSPMessage *r
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gst_rtsp_message_remove_header (response, GST_RTSP_HDR_SESSION, -1);
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}
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#if 0
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gst_rtsp_connection_send (client->connection, response, &timeout);
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#endif
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gst_rtsp_channel_queue_message (client->channel, response);
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gst_rtsp_message_unset (response);
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}
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@ -287,45 +290,6 @@ no_prepare:
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}
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}
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/* Get the session or NULL when there was no session */
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static GstRTSPSession *
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find_session (GstRTSPClient *client, GstRTSPMessage *request)
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{
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GstRTSPResult res;
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GstRTSPSession *session;
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gchar *sessid;
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res = gst_rtsp_message_get_header (request, GST_RTSP_HDR_SESSION, &sessid, 0);
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if (res == GST_RTSP_OK) {
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if (client->session_pool == NULL)
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goto no_pool;
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/* we had a session in the request, find it again */
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if (!(session = gst_rtsp_session_pool_find (client->session_pool, sessid)))
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goto session_not_found;
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client->timeout = gst_rtsp_session_get_timeout (session);
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}
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else
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goto service_unavailable;
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return session;
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/* ERRORS */
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no_pool:
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{
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return NULL;
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}
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session_not_found:
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{
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return NULL;
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}
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service_unavailable:
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{
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return NULL;
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}
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}
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static gboolean
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handle_teardown_request (GstRTSPClient *client, GstRTSPUrl *uri, GstRTSPSession *session, GstRTSPMessage *request)
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{
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@ -490,7 +454,7 @@ handle_play_request (GstRTSPClient *client, GstRTSPUrl *uri, GstRTSPSession *ses
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/* ERRORS */
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no_session:
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{
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/* error was sent */
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send_generic_response (client, GST_RTSP_STS_SESSION_NOT_FOUND, request);
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return FALSE;
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}
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not_found:
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@ -580,7 +544,7 @@ handle_setup_request (GstRTSPClient *client, GstRTSPUrl *uri, GstRTSPSession *se
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/* we have a valid transport now, set the destination of the client. */
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g_free (ct->destination);
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ct->destination = g_strdup (inet_ntoa (client->address.sin_addr));
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ct->destination = g_strdup (client->connection->url->host);
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if (session) {
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g_object_ref (session);
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@ -813,163 +777,101 @@ santize_uri (GstRTSPUrl *uri)
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*d = '\0';
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}
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/* this function runs in a client specific thread and handles all rtsp messages
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* with the client */
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static gpointer
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handle_client (GstRTSPClient *client)
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static void
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handle_request (GstRTSPClient *client, GstRTSPMessage *request)
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{
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GstRTSPMessage request = { 0 };
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GstRTSPResult res;
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GstRTSPMethod method;
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const gchar *uristr;
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GstRTSPUrl *uri;
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GstRTSPVersion version;
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while (TRUE) {
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GTimeVal timeout;
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GstRTSPSession *session;
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timeout.tv_sec = client->timeout;
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timeout.tv_usec = 0;
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/* start by waiting for a message from the client */
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res = gst_rtsp_connection_receive (client->connection, &request, &timeout);
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if (res < 0) {
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if (res == GST_RTSP_ETIMEOUT)
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goto timeout;
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goto receive_failed;
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}
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GstRTSPResult res;
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GstRTSPSession *session;
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gchar *sessid;
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#ifdef DEBUG
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gst_rtsp_message_dump (&request);
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gst_rtsp_message_dump (request);
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#endif
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gst_rtsp_message_parse_request (&request, &method, &uristr, &version);
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gst_rtsp_message_parse_request (request, &method, &uristr, &version);
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if (version != GST_RTSP_VERSION_1_0) {
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/* we can only handle 1.0 requests */
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send_generic_response (client, GST_RTSP_STS_RTSP_VERSION_NOT_SUPPORTED, &request);
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continue;
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}
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/* we always try to parse the url first */
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if ((res = gst_rtsp_url_parse (uristr, &uri)) != GST_RTSP_OK) {
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send_generic_response (client, GST_RTSP_STS_BAD_REQUEST, &request);
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continue;
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}
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/* sanitize the uri */
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santize_uri (uri);
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/* get the session if there is any */
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session = find_session (client, &request);
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/* now see what is asked and dispatch to a dedicated handler */
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switch (method) {
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case GST_RTSP_OPTIONS:
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handle_options_request (client, uri, session, &request);
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break;
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case GST_RTSP_DESCRIBE:
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handle_describe_request (client, uri, session, &request);
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break;
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case GST_RTSP_SETUP:
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handle_setup_request (client, uri, session, &request);
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break;
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case GST_RTSP_PLAY:
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handle_play_request (client, uri, session, &request);
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break;
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case GST_RTSP_PAUSE:
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handle_pause_request (client, uri, session, &request);
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break;
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case GST_RTSP_TEARDOWN:
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handle_teardown_request (client, uri, session, &request);
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break;
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case GST_RTSP_ANNOUNCE:
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case GST_RTSP_GET_PARAMETER:
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case GST_RTSP_RECORD:
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case GST_RTSP_REDIRECT:
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case GST_RTSP_SET_PARAMETER:
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send_generic_response (client, GST_RTSP_STS_NOT_IMPLEMENTED, &request);
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break;
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case GST_RTSP_INVALID:
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default:
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send_generic_response (client, GST_RTSP_STS_BAD_REQUEST, &request);
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break;
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}
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if (session)
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g_object_unref (session);
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gst_rtsp_url_free (uri);
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if (version != GST_RTSP_VERSION_1_0) {
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/* we can only handle 1.0 requests */
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send_generic_response (client, GST_RTSP_STS_RTSP_VERSION_NOT_SUPPORTED, request);
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return;
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}
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g_object_unref (client);
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return NULL;
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/* we always try to parse the url first */
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if ((res = gst_rtsp_url_parse (uristr, &uri)) != GST_RTSP_OK) {
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send_generic_response (client, GST_RTSP_STS_BAD_REQUEST, request);
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return;
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}
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/* sanitize the uri */
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santize_uri (uri);
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/* get the session if there is any */
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res = gst_rtsp_message_get_header (request, GST_RTSP_HDR_SESSION, &sessid, 0);
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if (res == GST_RTSP_OK) {
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if (client->session_pool == NULL)
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goto no_pool;
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/* we had a session in the request, find it again */
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if (!(session = gst_rtsp_session_pool_find (client->session_pool, sessid)))
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goto session_not_found;
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client->timeout = gst_rtsp_session_get_timeout (session);
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}
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else
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session = NULL;
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/* now see what is asked and dispatch to a dedicated handler */
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switch (method) {
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case GST_RTSP_OPTIONS:
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handle_options_request (client, uri, session, request);
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break;
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case GST_RTSP_DESCRIBE:
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handle_describe_request (client, uri, session, request);
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break;
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case GST_RTSP_SETUP:
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handle_setup_request (client, uri, session, request);
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break;
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case GST_RTSP_PLAY:
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handle_play_request (client, uri, session, request);
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break;
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case GST_RTSP_PAUSE:
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handle_pause_request (client, uri, session, request);
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break;
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case GST_RTSP_TEARDOWN:
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handle_teardown_request (client, uri, session, request);
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break;
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case GST_RTSP_ANNOUNCE:
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case GST_RTSP_GET_PARAMETER:
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case GST_RTSP_RECORD:
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case GST_RTSP_REDIRECT:
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case GST_RTSP_SET_PARAMETER:
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send_generic_response (client, GST_RTSP_STS_NOT_IMPLEMENTED, request);
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break;
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case GST_RTSP_INVALID:
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default:
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send_generic_response (client, GST_RTSP_STS_BAD_REQUEST, request);
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break;
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}
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if (session)
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g_object_unref (session);
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gst_rtsp_url_free (uri);
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return;
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/* ERRORS */
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timeout:
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no_pool:
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{
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g_message ("client timed out");
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if (client->session_pool)
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gst_rtsp_session_pool_cleanup (client->session_pool);
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goto cleanup;
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send_generic_response (client, GST_RTSP_STS_SERVICE_UNAVAILABLE, request);
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return;
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}
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receive_failed:
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session_not_found:
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{
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gchar *str;
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str = gst_rtsp_strresult (res);
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g_message ("receive failed %d (%s), disconnect client %p", res,
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str, client);
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g_free (str);
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goto cleanup;
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}
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cleanup:
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{
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gst_rtsp_message_unset (&request);
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gst_rtsp_connection_close (client->connection);
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g_object_unref (client);
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return NULL;
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}
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}
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/* called when we need to accept a new request from a client */
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static gboolean
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client_accept (GstRTSPClient *client, GIOChannel *channel)
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{
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/* a new client connected. */
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int server_sock_fd, fd;
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unsigned int address_len;
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GstRTSPConnection *conn;
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server_sock_fd = g_io_channel_unix_get_fd (channel);
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address_len = sizeof (client->address);
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memset (&client->address, 0, address_len);
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fd = accept (server_sock_fd, (struct sockaddr *) &client->address,
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&address_len);
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if (fd == -1)
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goto accept_failed;
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/* now create the connection object */
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gst_rtsp_connection_create (NULL, &conn);
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conn->fd.fd = fd;
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/* FIXME some hackery, we need to have a connection method to accept server
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* connections */
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gst_poll_add_fd (conn->fdset, &conn->fd);
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g_message ("added new client %p ip %s with fd %d", client,
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inet_ntoa (client->address.sin_addr), conn->fd.fd);
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client->connection = conn;
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return TRUE;
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/* ERRORS */
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accept_failed:
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{
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g_error ("Could not accept client on server socket %d: %s (%d)",
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server_sock_fd, g_strerror (errno), errno);
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return FALSE;
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send_generic_response (client, GST_RTSP_STS_SESSION_NOT_FOUND, request);
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return;
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}
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}
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@ -1087,6 +989,58 @@ gst_rtsp_client_get_media_mapping (GstRTSPClient *client)
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return result;
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}
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static GstRTSPResult
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message_received (GstRTSPChannel *channel, GstRTSPMessage *message, gpointer user_data)
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{
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GstRTSPClient *client = GST_RTSP_CLIENT (user_data);
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g_message ("client %p: received a message", client);
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handle_request (client, message);
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return GST_RTSP_OK;
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}
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static GstRTSPResult
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message_sent (GstRTSPChannel *channel, guint cseq, gpointer user_data)
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{
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GstRTSPClient *client = GST_RTSP_CLIENT (user_data);
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g_message ("client %p: sent a message with cseq %d", client, cseq);
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return GST_RTSP_OK;
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}
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static GstRTSPResult
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closed (GstRTSPChannel *channel, gpointer user_data)
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{
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GstRTSPClient *client = GST_RTSP_CLIENT (user_data);
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g_message ("client %p: connection closed", client);
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return GST_RTSP_OK;
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}
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static GstRTSPResult
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error (GstRTSPChannel *channel, GstRTSPResult result, gpointer user_data)
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{
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GstRTSPClient *client = GST_RTSP_CLIENT (user_data);
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gchar *str;
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str = gst_rtsp_strresult (result);
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g_message ("client %p: received an error %s", client, str);
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g_free (str);
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return GST_RTSP_OK;
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}
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static GstRTSPChannelFuncs channel_funcs = {
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message_received,
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message_sent,
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closed,
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error
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};
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/**
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* gst_rtsp_client_attach:
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* @client: a #GstRTSPClient
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@ -1102,32 +1056,47 @@ gst_rtsp_client_get_media_mapping (GstRTSPClient *client)
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gboolean
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gst_rtsp_client_accept (GstRTSPClient *client, GIOChannel *channel)
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{
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GError *error = NULL;
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int sock;
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GstRTSPConnection *conn;
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GstRTSPResult res;
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GSource *source;
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GMainContext *context;
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if (!client_accept (client, channel))
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goto accept_failed;
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/* a new client connected. */
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sock = g_io_channel_unix_get_fd (channel);
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/* client accepted, spawn a thread for the client, we don't need to join the
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* thread */
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g_object_ref (client);
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client->thread = g_thread_create ((GThreadFunc)handle_client, client, FALSE, &error);
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if (client->thread == NULL)
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goto no_thread;
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GST_RTSP_CHECK (gst_rtsp_connection_accept (sock, &conn), accept_failed);
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g_message ("added new client %p ip %s:%d with fd %d", client,
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conn->url->host, conn->url->port, conn->fd.fd);
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client->connection = conn;
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/* create channel for the connection and attach */
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client->channel = gst_rtsp_channel_new (client->connection, &channel_funcs,
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g_object_ref (client), g_object_unref);
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/* find the context to add the channel */
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if ((source = g_main_current_source ()))
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context = g_source_get_context (source);
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else
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context = NULL;
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g_message ("attaching to context %p", context);
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gst_rtsp_channel_attach (client->channel, context);
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gst_rtsp_channel_unref (client->channel);
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return TRUE;
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/* ERRORS */
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accept_failed:
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{
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return FALSE;
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}
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no_thread:
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{
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if (error) {
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g_warning ("could not create thread for client %p: %s", client, error->message);
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g_error_free (error);
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}
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g_object_unref (client);
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gchar *str = gst_rtsp_strresult (res);
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g_error ("Could not accept client on server socket %d: %s",
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sock, str);
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g_free (str);
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return FALSE;
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}
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}
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@ -70,7 +70,7 @@ struct _GstRTSPClient {
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GObject parent;
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GstRTSPConnection *connection;
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struct sockaddr_in address;
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GstRTSPChannel *channel;
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GThread *thread;
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guint timeout;
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