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

diff --git a/gst/rtsp-server/rtsp-client.c b/gst/rtsp-server/rtsp-client.c
index 77dc1d0..eb87c84 100644 (file)
--- a/gst/rtsp-server/rtsp-client.c
+++ b/gst/rtsp-server/rtsp-client.c
@@ -82,7 +82,7 @@ struct _GstRTSPClientPrivate
   gchar *path;
   GstRTSPMedia *media;
 
-  GList *transports;
+  GHashTable *transports;
   GList *sessions;
   guint sessions_cookie;
 
@@ -134,8 +134,6 @@ static void gst_rtsp_client_set_property (GObject * object, guint propid,
 static void gst_rtsp_client_finalize (GObject * obj);
 
 static GstSDPMessage *create_sdp (GstRTSPClient * client, GstRTSPMedia * media);
-static void unlink_session_transports (GstRTSPClient * client,
-    GstRTSPSession * session, GstRTSPSessionMedia * sessmedia);
 static gboolean default_configure_client_media (GstRTSPClient * client,
     GstRTSPMedia * media, GstRTSPStream * stream, GstRTSPContext * ctx);
 static gboolean default_configure_client_transport (GstRTSPClient * client,
@@ -282,18 +280,15 @@ gst_rtsp_client_init (GstRTSPClient * client)
   g_mutex_init (&priv->watch_lock);
   priv->close_seq = 0;
   priv->drop_backlog = DEFAULT_DROP_BACKLOG;
+  priv->transports = g_hash_table_new (g_direct_hash, g_direct_equal);
 }
 
 static GstRTSPFilterResult
 filter_session_media (GstRTSPSession * sess, GstRTSPSessionMedia * sessmedia,
     gpointer user_data)
 {
-  GstRTSPClient *client = GST_RTSP_CLIENT (user_data);
-
   gst_rtsp_session_media_set_state (sessmedia, GST_STATE_NULL);
-  unlink_session_transports (client, sess, sessmedia);
 
-  /* unmanage the media in the session */
   return GST_RTSP_FILTER_REMOVE;
 }
 
@@ -386,6 +381,8 @@ gst_rtsp_client_finalize (GObject * obj)
   g_assert (priv->sessions == NULL);
   g_assert (priv->session_removed_id == 0);
 
+  g_hash_table_unref (priv->transports);
+
   if (priv->connection)
     gst_rtsp_connection_free (priv->connection);
   if (priv->session_pool) {
@@ -715,92 +712,6 @@ do_send_data (GstBuffer * buffer, guint8 channel, GstRTSPClient * client)
   return TRUE;
 }
 
-static void
-link_transport (GstRTSPClient * client, GstRTSPSession * session,
-    GstRTSPStreamTransport * trans)
-{
-  GstRTSPClientPrivate *priv = client->priv;
-
-  GST_DEBUG ("client %p: linking transport %p", client, trans);
-
-  gst_rtsp_stream_transport_set_callbacks (trans,
-      (GstRTSPSendFunc) do_send_data,
-      (GstRTSPSendFunc) do_send_data, client, NULL);
-
-  priv->transports = g_list_prepend (priv->transports, trans);
-
-  /* make sure our session can't expire */
-  gst_rtsp_session_prevent_expire (session);
-}
-
-static void
-link_session_transports (GstRTSPClient * client, GstRTSPSession * session,
-    GstRTSPSessionMedia * sessmedia)
-{
-  guint n_streams, i;
-
-  n_streams =
-      gst_rtsp_media_n_streams (gst_rtsp_session_media_get_media (sessmedia));
-  for (i = 0; i < n_streams; i++) {
-    GstRTSPStreamTransport *trans;
-    const GstRTSPTransport *tr;
-
-    /* get the transport, if there is no transport configured, skip this stream */
-    trans = gst_rtsp_session_media_get_transport (sessmedia, i);
-    if (trans == NULL)
-      continue;
-
-    tr = gst_rtsp_stream_transport_get_transport (trans);
-
-    if (tr->lower_transport == GST_RTSP_LOWER_TRANS_TCP) {
-      /* for TCP, link the stream to the TCP connection of the client */
-      link_transport (client, session, trans);
-    }
-  }
-}
-
-static void
-unlink_transport (GstRTSPClient * client, GstRTSPSession * session,
-    GstRTSPStreamTransport * trans)
-{
-  GstRTSPClientPrivate *priv = client->priv;
-
-  GST_DEBUG ("client %p: unlinking transport %p", client, trans);
-
-  gst_rtsp_stream_transport_set_callbacks (trans, NULL, NULL, NULL, NULL);
-
-  priv->transports = g_list_remove (priv->transports, trans);
-
-  /* our session can now expire */
-  gst_rtsp_session_allow_expire (session);
-}
-
-static void
-unlink_session_transports (GstRTSPClient * client, GstRTSPSession * session,
-    GstRTSPSessionMedia * sessmedia)
-{
-  guint n_streams, i;
-
-  n_streams =
-      gst_rtsp_media_n_streams (gst_rtsp_session_media_get_media (sessmedia));
-  for (i = 0; i < n_streams; i++) {
-    GstRTSPStreamTransport *trans;
-    const GstRTSPTransport *tr;
-
-    /* get the transport, if there is no transport configured, skip this stream */
-    trans = gst_rtsp_session_media_get_transport (sessmedia, i);
-    if (trans == NULL)
-      continue;
-
-    tr = gst_rtsp_stream_transport_get_transport (trans);
-
-    if (tr->lower_transport == GST_RTSP_LOWER_TRANS_TCP) {
-      /* for TCP, unlink the stream from the TCP connection of the client */
-      unlink_transport (client, session, trans);
-    }
-  }
-}
-
 /**
  * gst_rtsp_client_close:
  * @client: a #GstRTSPClient
@@ -895,9 +806,6 @@ handle_teardown_request (GstRTSPClient * client, GstRTSPContext * ctx)
   if (priv->watch != NULL)
     gst_rtsp_watch_set_flushing (priv->watch, TRUE);
 
-  /* unlink the all TCP callbacks */
-  unlink_session_transports (client, session, sessmedia);
-
   gst_rtsp_session_media_set_state (sessmedia, GST_STATE_NULL);
 
   /* allow messages again so that we can send the reply */
@@ -1084,9 +992,6 @@ handle_pause_request (GstRTSPClient * client, GstRTSPContext * ctx)
       rtspstate != GST_RTSP_STATE_RECORDING)
     goto invalid_state;
 
-  /* unlink the all TCP callbacks */
-  unlink_session_transports (client, session, sessmedia);
-
   /* then pause sending */
   gst_rtsp_session_media_set_state (sessmedia, GST_STATE_PAUSED);
 
@@ -1223,9 +1128,6 @@ handle_play_request (GstRTSPClient * client, GstRTSPContext * ctx)
     }
   }
 
-  /* link the all TCP callbacks */
-  link_session_transports (client, session, sessmedia);
-
   /* grab RTPInfo from the media now */
   rtpinfo = gst_rtsp_session_media_get_rtpinfo (sessmedia);
 
@@ -1926,11 +1828,22 @@ handle_setup_request (GstRTSPClient * client, GstRTSPContext * ctx)
   /* configure the url used to set this transport, this we will use when
    * generating the response for the PLAY request */
   gst_rtsp_stream_transport_set_url (trans, uri);
-
   /* configure keepalive for this transport */
   gst_rtsp_stream_transport_set_keepalive (trans,
       (GstRTSPKeepAliveFunc) do_keepalive, session, NULL);
 
+  if (ct->lower_transport == GST_RTSP_LOWER_TRANS_TCP) {
+    /* our callbacks to send data on this TCP connection */
+    gst_rtsp_stream_transport_set_callbacks (trans,
+        (GstRTSPSendFunc) do_send_data,
+        (GstRTSPSendFunc) do_send_data, client, NULL);
+
+    g_hash_table_insert (priv->transports,
+        GINT_TO_POINTER (ct->interleaved.min), trans);
+    g_hash_table_insert (priv->transports,
+        GINT_TO_POINTER (ct->interleaved.max), trans);
+  }
+
   /* create and serialize the server transport */
   st = make_server_transport (client, ctx, ct);
   trans_str = gst_rtsp_transport_as_text (st);
@@ -2610,11 +2523,10 @@ handle_data (GstRTSPClient * client, GstRTSPMessage * message)
   GstRTSPClientPrivate *priv = client->priv;
   GstRTSPResult res;
   guint8 channel;
-  GList *walk;
   guint8 *data;
   guint size;
   GstBuffer *buffer;
-  gboolean handled;
+  GstRTSPStreamTransport *trans;
 
   /* find the stream for this message */
   res = gst_rtsp_message_parse_data (message, &channel);
@@ -2625,33 +2537,13 @@ handle_data (GstRTSPClient * client, GstRTSPMessage * message)
 
   buffer = gst_buffer_new_wrapped (data, size);
 
-  handled = FALSE;
-  for (walk = priv->transports; walk; walk = g_list_next (walk)) {
-    GstRTSPStreamTransport *trans;
-    GstRTSPStream *stream;
-    const GstRTSPTransport *tr;
-
-    trans = walk->data;
-
-    tr = gst_rtsp_stream_transport_get_transport (trans);
-    stream = gst_rtsp_stream_transport_get_stream (trans);
-
-    /* check for TCP transport */
-    if (tr->lower_transport == GST_RTSP_LOWER_TRANS_TCP) {
-      /* dispatch to the stream based on the channel number */
-      if (tr->interleaved.min == channel) {
-        gst_rtsp_stream_recv_rtp (stream, buffer);
-        handled = TRUE;
-        break;
-      } else if (tr->interleaved.max == channel) {
-        gst_rtsp_stream_recv_rtcp (stream, buffer);
-        handled = TRUE;
-        break;
-      }
-    }
-  }
-  if (!handled)
+  trans = g_hash_table_lookup (priv->transports, GINT_TO_POINTER (channel));
+  if (trans) {
+    /* dispatch to the stream based on the channel number */
+    gst_rtsp_stream_transport_recv_data (trans, channel, buffer);
+  } else {
     gst_buffer_unref (buffer);
+  }
 }
 
 /**