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client: simplify session transport handling
link/unlink of the transport in a session was done to keep track of all TCP transports and to send RTP/RTCP data to the streams. We can simplify that by putting all the TCP transports in a hashtable indexed with the channel number. We also don't need to link/unlink the transports when we pause/resume the streams. The same effect is already achieved when we pause/play the media. Indeed, when we pause the media, the transport is removed from the media and the callbacks will not be called anymore. See https://bugzilla.gnome.org/show_bug.cgi?id=736041
This commit is contained in:
Wim Taymans
parent
ea5d4cfc7e
commit
0292be09ea
1 changed files with 23 additions and 131 deletions
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@ -82,7 +82,7 @@ struct _GstRTSPClientPrivate
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gchar *path;
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GstRTSPMedia *media;
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GList *transports;
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GHashTable *transports;
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GList *sessions;
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guint sessions_cookie;
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@ -134,8 +134,6 @@ static void gst_rtsp_client_set_property (GObject * object, guint propid,
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static void gst_rtsp_client_finalize (GObject * obj);
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static GstSDPMessage *create_sdp (GstRTSPClient * client, GstRTSPMedia * media);
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static void unlink_session_transports (GstRTSPClient * client,
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GstRTSPSession * session, GstRTSPSessionMedia * sessmedia);
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static gboolean default_configure_client_media (GstRTSPClient * client,
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GstRTSPMedia * media, GstRTSPStream * stream, GstRTSPContext * ctx);
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static gboolean default_configure_client_transport (GstRTSPClient * client,
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@ -282,18 +280,15 @@ gst_rtsp_client_init (GstRTSPClient * client)
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g_mutex_init (&priv->watch_lock);
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priv->close_seq = 0;
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priv->drop_backlog = DEFAULT_DROP_BACKLOG;
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priv->transports = g_hash_table_new (g_direct_hash, g_direct_equal);
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}
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static GstRTSPFilterResult
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filter_session_media (GstRTSPSession * sess, GstRTSPSessionMedia * sessmedia,
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gpointer user_data)
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{
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GstRTSPClient *client = GST_RTSP_CLIENT (user_data);
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gst_rtsp_session_media_set_state (sessmedia, GST_STATE_NULL);
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unlink_session_transports (client, sess, sessmedia);
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/* unmanage the media in the session */
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return GST_RTSP_FILTER_REMOVE;
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}
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@ -386,6 +381,8 @@ gst_rtsp_client_finalize (GObject * obj)
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g_assert (priv->sessions == NULL);
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g_assert (priv->session_removed_id == 0);
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g_hash_table_unref (priv->transports);
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if (priv->connection)
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gst_rtsp_connection_free (priv->connection);
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if (priv->session_pool) {
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@ -715,92 +712,6 @@ do_send_data (GstBuffer * buffer, guint8 channel, GstRTSPClient * client)
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return TRUE;
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}
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static void
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link_transport (GstRTSPClient * client, GstRTSPSession * session,
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GstRTSPStreamTransport * trans)
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{
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GstRTSPClientPrivate *priv = client->priv;
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GST_DEBUG ("client %p: linking transport %p", client, trans);
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gst_rtsp_stream_transport_set_callbacks (trans,
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(GstRTSPSendFunc) do_send_data,
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(GstRTSPSendFunc) do_send_data, client, NULL);
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priv->transports = g_list_prepend (priv->transports, trans);
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/* make sure our session can't expire */
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gst_rtsp_session_prevent_expire (session);
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}
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static void
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link_session_transports (GstRTSPClient * client, GstRTSPSession * session,
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GstRTSPSessionMedia * sessmedia)
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{
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guint n_streams, i;
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n_streams =
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gst_rtsp_media_n_streams (gst_rtsp_session_media_get_media (sessmedia));
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for (i = 0; i < n_streams; i++) {
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GstRTSPStreamTransport *trans;
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const GstRTSPTransport *tr;
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/* get the transport, if there is no transport configured, skip this stream */
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trans = gst_rtsp_session_media_get_transport (sessmedia, i);
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if (trans == NULL)
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continue;
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tr = gst_rtsp_stream_transport_get_transport (trans);
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if (tr->lower_transport == GST_RTSP_LOWER_TRANS_TCP) {
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/* for TCP, link the stream to the TCP connection of the client */
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link_transport (client, session, trans);
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}
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}
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}
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static void
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unlink_transport (GstRTSPClient * client, GstRTSPSession * session,
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GstRTSPStreamTransport * trans)
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{
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GstRTSPClientPrivate *priv = client->priv;
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GST_DEBUG ("client %p: unlinking transport %p", client, trans);
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gst_rtsp_stream_transport_set_callbacks (trans, NULL, NULL, NULL, NULL);
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priv->transports = g_list_remove (priv->transports, trans);
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/* our session can now expire */
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gst_rtsp_session_allow_expire (session);
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}
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static void
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unlink_session_transports (GstRTSPClient * client, GstRTSPSession * session,
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GstRTSPSessionMedia * sessmedia)
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{
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guint n_streams, i;
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n_streams =
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gst_rtsp_media_n_streams (gst_rtsp_session_media_get_media (sessmedia));
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for (i = 0; i < n_streams; i++) {
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GstRTSPStreamTransport *trans;
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const GstRTSPTransport *tr;
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/* get the transport, if there is no transport configured, skip this stream */
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trans = gst_rtsp_session_media_get_transport (sessmedia, i);
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if (trans == NULL)
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continue;
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tr = gst_rtsp_stream_transport_get_transport (trans);
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if (tr->lower_transport == GST_RTSP_LOWER_TRANS_TCP) {
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/* for TCP, unlink the stream from the TCP connection of the client */
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unlink_transport (client, session, trans);
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}
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}
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}
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/**
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* gst_rtsp_client_close:
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* @client: a #GstRTSPClient
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@ -895,9 +806,6 @@ handle_teardown_request (GstRTSPClient * client, GstRTSPContext * ctx)
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if (priv->watch != NULL)
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gst_rtsp_watch_set_flushing (priv->watch, TRUE);
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/* unlink the all TCP callbacks */
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unlink_session_transports (client, session, sessmedia);
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gst_rtsp_session_media_set_state (sessmedia, GST_STATE_NULL);
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/* allow messages again so that we can send the reply */
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@ -1084,9 +992,6 @@ handle_pause_request (GstRTSPClient * client, GstRTSPContext * ctx)
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rtspstate != GST_RTSP_STATE_RECORDING)
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goto invalid_state;
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/* unlink the all TCP callbacks */
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unlink_session_transports (client, session, sessmedia);
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/* then pause sending */
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gst_rtsp_session_media_set_state (sessmedia, GST_STATE_PAUSED);
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@ -1223,9 +1128,6 @@ handle_play_request (GstRTSPClient * client, GstRTSPContext * ctx)
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}
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}
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/* link the all TCP callbacks */
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link_session_transports (client, session, sessmedia);
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/* grab RTPInfo from the media now */
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rtpinfo = gst_rtsp_session_media_get_rtpinfo (sessmedia);
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@ -1926,11 +1828,22 @@ handle_setup_request (GstRTSPClient * client, GstRTSPContext * ctx)
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/* configure the url used to set this transport, this we will use when
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* generating the response for the PLAY request */
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gst_rtsp_stream_transport_set_url (trans, uri);
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/* configure keepalive for this transport */
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gst_rtsp_stream_transport_set_keepalive (trans,
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(GstRTSPKeepAliveFunc) do_keepalive, session, NULL);
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if (ct->lower_transport == GST_RTSP_LOWER_TRANS_TCP) {
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/* our callbacks to send data on this TCP connection */
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gst_rtsp_stream_transport_set_callbacks (trans,
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(GstRTSPSendFunc) do_send_data,
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(GstRTSPSendFunc) do_send_data, client, NULL);
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g_hash_table_insert (priv->transports,
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GINT_TO_POINTER (ct->interleaved.min), trans);
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g_hash_table_insert (priv->transports,
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GINT_TO_POINTER (ct->interleaved.max), trans);
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}
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/* create and serialize the server transport */
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st = make_server_transport (client, ctx, ct);
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trans_str = gst_rtsp_transport_as_text (st);
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@ -2610,11 +2523,10 @@ handle_data (GstRTSPClient * client, GstRTSPMessage * message)
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GstRTSPClientPrivate *priv = client->priv;
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GstRTSPResult res;
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guint8 channel;
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GList *walk;
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guint8 *data;
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guint size;
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GstBuffer *buffer;
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gboolean handled;
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GstRTSPStreamTransport *trans;
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/* find the stream for this message */
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res = gst_rtsp_message_parse_data (message, &channel);
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@ -2625,33 +2537,13 @@ handle_data (GstRTSPClient * client, GstRTSPMessage * message)
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buffer = gst_buffer_new_wrapped (data, size);
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handled = FALSE;
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for (walk = priv->transports; walk; walk = g_list_next (walk)) {
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GstRTSPStreamTransport *trans;
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GstRTSPStream *stream;
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const GstRTSPTransport *tr;
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trans = walk->data;
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tr = gst_rtsp_stream_transport_get_transport (trans);
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stream = gst_rtsp_stream_transport_get_stream (trans);
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/* check for TCP transport */
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if (tr->lower_transport == GST_RTSP_LOWER_TRANS_TCP) {
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/* dispatch to the stream based on the channel number */
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if (tr->interleaved.min == channel) {
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gst_rtsp_stream_recv_rtp (stream, buffer);
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handled = TRUE;
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break;
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} else if (tr->interleaved.max == channel) {
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gst_rtsp_stream_recv_rtcp (stream, buffer);
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handled = TRUE;
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break;
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}
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}
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}
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if (!handled)
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trans = g_hash_table_lookup (priv->transports, GINT_TO_POINTER (channel));
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if (trans) {
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/* dispatch to the stream based on the channel number */
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gst_rtsp_stream_transport_recv_data (trans, channel, buffer);
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} else {
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gst_buffer_unref (buffer);
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}
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}
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/**
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