mirror of
https://gitlab.freedesktop.org/gstreamer/gstreamer.git
synced 2025-02-22 14:06:23 +00:00
Add support for session keepalive
Get and update the session timeout for all requests. get the session as early as possible.
This commit is contained in:
Wim Taymans
parent
cd29e2a454
commit
308ad6f6d0
1 changed files with 28 additions and 47 deletions
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@ -289,7 +289,7 @@ no_prepare:
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/* Get the session or NULL when there was no session */
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static GstRTSPSession *
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ensure_session (GstRTSPClient *client, GstRTSPMessage *request)
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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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@ -314,30 +314,26 @@ ensure_session (GstRTSPClient *client, GstRTSPMessage *request)
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/* ERRORS */
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no_pool:
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{
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send_generic_response (client, GST_RTSP_STS_SESSION_NOT_FOUND, request);
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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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send_generic_response (client, GST_RTSP_STS_SESSION_NOT_FOUND, request);
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return NULL;
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}
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service_unavailable:
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{
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send_generic_response (client, GST_RTSP_STS_SERVICE_UNAVAILABLE, request);
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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, GstRTSPMessage *request)
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handle_teardown_request (GstRTSPClient *client, GstRTSPUrl *uri, GstRTSPSession *session, GstRTSPMessage *request)
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{
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GstRTSPSessionMedia *media;
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GstRTSPSession *session;
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GstRTSPMessage response = { 0 };
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GstRTSPStatusCode code;
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if (!(session = ensure_session (client, request)))
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if (!session)
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goto no_session;
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/* get a handle to the configuration of the media in the session */
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@ -359,14 +355,12 @@ handle_teardown_request (GstRTSPClient *client, GstRTSPUrl *uri, GstRTSPMessage
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send_response (client, session, &response);
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g_object_unref (session);
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return FALSE;
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/* ERRORS */
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no_session:
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{
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/* error was sent already */
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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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@ -377,14 +371,13 @@ not_found:
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}
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static gboolean
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handle_pause_request (GstRTSPClient *client, GstRTSPUrl *uri, GstRTSPMessage *request)
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handle_pause_request (GstRTSPClient *client, GstRTSPUrl *uri, GstRTSPSession *session, GstRTSPMessage *request)
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{
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GstRTSPSessionMedia *media;
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GstRTSPSession *session;
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GstRTSPMessage response = { 0 };
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GstRTSPStatusCode code;
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if (!(session = ensure_session (client, request)))
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if (!session)
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goto no_session;
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/* get a handle to the configuration of the media in the session */
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@ -407,34 +400,31 @@ handle_pause_request (GstRTSPClient *client, GstRTSPUrl *uri, GstRTSPMessage *re
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/* the state is now READY */
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media->state = GST_RTSP_STATE_READY;
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g_object_unref (session);
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return FALSE;
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/* ERRORS */
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no_session:
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{
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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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{
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send_generic_response (client, GST_RTSP_STS_NOT_FOUND, request);
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g_object_unref (session);
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return FALSE;
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}
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invalid_state:
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{
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send_generic_response (client, GST_RTSP_STS_METHOD_NOT_VALID_IN_THIS_STATE, request);
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g_object_unref (session);
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return FALSE;
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}
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}
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static gboolean
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handle_play_request (GstRTSPClient *client, GstRTSPUrl *uri, GstRTSPMessage *request)
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handle_play_request (GstRTSPClient *client, GstRTSPUrl *uri, GstRTSPSession *session, GstRTSPMessage *request)
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{
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GstRTSPSessionMedia *media;
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GstRTSPSession *session;
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GstRTSPMessage response = { 0 };
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GstRTSPStatusCode code;
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GString *rtpinfo;
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@ -442,7 +432,7 @@ handle_play_request (GstRTSPClient *client, GstRTSPUrl *uri, GstRTSPMessage *req
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guint timestamp, seqnum;
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gchar *str;
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if (!(session = ensure_session (client, request)))
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if (!session)
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goto no_session;
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/* get a handle to the configuration of the media in the session */
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@ -494,7 +484,6 @@ handle_play_request (GstRTSPClient *client, GstRTSPUrl *uri, GstRTSPMessage *req
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gst_rtsp_session_media_play (media);
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media->state = GST_RTSP_STATE_PLAYING;
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g_object_unref (session);
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return FALSE;
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@ -507,27 +496,23 @@ no_session:
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not_found:
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{
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send_generic_response (client, GST_RTSP_STS_NOT_FOUND, request);
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g_object_unref (session);
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return FALSE;
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}
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invalid_state:
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{
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send_generic_response (client, GST_RTSP_STS_METHOD_NOT_VALID_IN_THIS_STATE, request);
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g_object_unref (session);
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return FALSE;
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}
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}
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static gboolean
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handle_setup_request (GstRTSPClient *client, GstRTSPUrl *uri, GstRTSPMessage *request)
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handle_setup_request (GstRTSPClient *client, GstRTSPUrl *uri, GstRTSPSession *session, GstRTSPMessage *request)
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{
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GstRTSPResult res;
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gchar *sessid;
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gchar *transport;
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gchar **transports;
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gboolean have_transport;
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GstRTSPTransport *ct, *st;
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GstRTSPSession *session;
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gint i;
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GstRTSPLowerTrans supported;
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GstRTSPMessage response = { 0 };
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@ -597,14 +582,8 @@ handle_setup_request (GstRTSPClient *client, GstRTSPUrl *uri, GstRTSPMessage *re
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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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/* a setup request creates a session for a client, check if the client already
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* sent a session id to us */
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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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/* 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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if (session) {
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g_object_ref (session);
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/* get a handle to the configuration of the media in the session, this can
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* return NULL if this is a new url to manage in this session. */
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media = gst_rtsp_session_get_media (session, uri);
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@ -669,7 +648,6 @@ handle_setup_request (GstRTSPClient *client, GstRTSPUrl *uri, GstRTSPMessage *re
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media->state = GST_RTSP_STATE_READY;
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break;
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}
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g_object_unref (session);
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return TRUE;
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@ -683,17 +661,14 @@ bad_request:
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not_found:
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{
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send_generic_response (client, GST_RTSP_STS_NOT_FOUND, request);
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g_object_unref (session);
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return FALSE;
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}
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no_stream:
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{
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send_generic_response (client, GST_RTSP_STS_NOT_FOUND, request);
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g_object_unref (media);
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return FALSE;
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}
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session_not_found:
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{
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send_generic_response (client, GST_RTSP_STS_SESSION_NOT_FOUND, request);
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g_object_unref (session);
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return FALSE;
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}
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no_transport:
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@ -721,7 +696,7 @@ service_unavailable:
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/* for the describe we must generate an SDP */
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static gboolean
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handle_describe_request (GstRTSPClient *client, GstRTSPUrl *uri, GstRTSPMessage *request)
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handle_describe_request (GstRTSPClient *client, GstRTSPUrl *uri, GstRTSPSession *session, GstRTSPMessage *request)
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{
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GstRTSPMessage response = { 0 };
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GstRTSPResult res;
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@ -787,7 +762,7 @@ no_sdp:
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}
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static void
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handle_options_request (GstRTSPClient *client, GstRTSPUrl *uri, GstRTSPMessage *request)
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handle_options_request (GstRTSPClient *client, GstRTSPUrl *uri, GstRTSPSession *session, GstRTSPMessage *request)
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{
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GstRTSPMessage response = { 0 };
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GstRTSPMethod options;
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@ -852,6 +827,7 @@ handle_client (GstRTSPClient *client)
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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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@ -886,25 +862,28 @@ handle_client (GstRTSPClient *client)
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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, &request);
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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, &request);
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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, &request);
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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, &request);
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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, &request);
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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, &request);
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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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@ -918,6 +897,8 @@ handle_client (GstRTSPClient *client)
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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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}
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g_object_unref (client);
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