From: Wim Taymans <wim.taymans@collabora.co.uk>
Date: Tue, 6 Apr 2010 15:08:40 +0000 (+0200)
Subject: client: improve client cleanups
X-Git-Tag: 1.19.3~495^2~1467
X-Git-Url: http://review.tizen.org/git/?a=commitdiff_plain;h=336ffc0941308597105f4cd3bc8c1cb497b38108;p=platform%2Fupstream%2Fgstreamer.git

client: improve client cleanups

Make sure the session does not timeout when using TCP. We need to do this
because quicktime player does not send RTCP for some reason in tunneled
mode.
Refactor some cleanup code.

Fixes #612915
---

diff --git a/gst/rtsp-server/rtsp-client.c b/gst/rtsp-server/rtsp-client.c
index 6c575db..b65b5e4 100644
--- a/gst/rtsp-server/rtsp-client.c
+++ b/gst/rtsp-server/rtsp-client.c
@@ -48,6 +48,8 @@ static void gst_rtsp_client_finalize (GObject * obj);
 
 static void client_session_finalized (GstRTSPClient * client,
     GstRTSPSession * session);
+static void unlink_session_streams (GstRTSPClient * client,
+    GstRTSPSession *session, GstRTSPSessionMedia * media);
 
 G_DEFINE_TYPE (GstRTSPClient, gst_rtsp_client, G_TYPE_OBJECT);
 
@@ -86,23 +88,42 @@ gst_rtsp_client_init (GstRTSPClient * client)
 {
 }
 
-/* A client is finalized when the connection is broken */
 static void
-gst_rtsp_client_finalize (GObject * obj)
+client_unlink_session (GstRTSPClient *client, GstRTSPSession *session)
 {
-  GstRTSPClient *client = GST_RTSP_CLIENT (obj);
-  GList *walk;
+  GList *medias;
 
-  GST_INFO ("finalize client %p", client);
+  /* 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);
+  }
+}
+
+static void
+client_cleanup_sessions (GstRTSPClient *client)
+{
+  GList *sessions;
 
   /* remove weak-ref from sessions */
-  for (walk = client->sessions; walk; walk = g_list_next (walk)) {
-    GstRTSPSession *msession = (GstRTSPSession *) walk->data;
-    g_object_weak_unref (G_OBJECT (msession),
+  for (sessions = client->sessions; sessions; sessions = g_list_next (sessions)) {
+    GstRTSPSession *session = (GstRTSPSession *) sessions->data;
+    g_object_weak_unref (G_OBJECT (session),
         (GWeakNotify) client_session_finalized, client);
+    client_unlink_session (client, session);
   }
-
   g_list_free (client->sessions);
+  client->sessions = NULL;
+}
+
+/* A client is finalized when the connection is broken */
+static void
+gst_rtsp_client_finalize (GObject * obj)
+{
+  GstRTSPClient *client = GST_RTSP_CLIENT (obj);
+
+  GST_INFO ("finalize client %p", client);
+
+  client_cleanup_sessions (client);
 
   gst_rtsp_connection_free (client->connection);
   if (client->session_pool)
@@ -329,24 +350,28 @@ do_send_data (GstBuffer * buffer, guint8 channel, GstRTSPClient * client)
 }
 
 static void
-link_stream (GstRTSPClient * client, GstRTSPSessionStream * stream)
+link_stream (GstRTSPClient * client, GstRTSPSession *session, GstRTSPSessionStream * stream)
 {
   GST_DEBUG ("client %p: linking stream %p", client, stream);
   gst_rtsp_session_stream_set_callbacks (stream, (GstRTSPSendFunc) do_send_data,
       (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);
 }
 
 static void
-unlink_stream (GstRTSPClient * client, GstRTSPSessionStream * stream)
+unlink_stream (GstRTSPClient * client, GstRTSPSession *session, GstRTSPSessionStream * stream)
 {
   GST_DEBUG ("client %p: unlinking stream %p", client, stream);
   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);
 }
 
 static void
-unlink_session_streams (GstRTSPClient * client, GstRTSPSessionMedia * media)
+unlink_session_streams (GstRTSPClient * client, GstRTSPSession *session, GstRTSPSessionMedia * media)
 {
   guint n_streams, i;
 
@@ -363,7 +388,7 @@ unlink_session_streams (GstRTSPClient * client, GstRTSPSessionMedia * media)
 
     if (tr->lower_transport == GST_RTSP_LOWER_TRANS_TCP) {
       /* for TCP, unlink the stream from the TCP connection of the client */
-      unlink_stream (client, sstream);
+      unlink_stream (client, session, sstream);
     }
   }
 }
@@ -407,7 +432,7 @@ handle_teardown_request (GstRTSPClient * client, GstRTSPUrl * uri,
     goto not_found;
 
   /* unlink the all TCP callbacks */
-  unlink_session_streams (client, media);
+  unlink_session_streams (client, session, media);
 
   /* remove the session from the watched sessions */
   g_object_weak_unref (G_OBJECT (session),
@@ -541,7 +566,7 @@ handle_pause_request (GstRTSPClient * client, GstRTSPUrl * uri,
     goto invalid_state;
 
   /* unlink the all TCP callbacks */
-  unlink_session_streams (client, media);
+  unlink_session_streams (client, session, media);
 
   /* then pause sending */
   gst_rtsp_session_media_set_state (media, GST_STATE_PAUSED);
@@ -635,7 +660,7 @@ handle_play_request (GstRTSPClient * client, GstRTSPUrl * uri,
 
     if (tr->lower_transport == GST_RTSP_LOWER_TRANS_TCP) {
       /* for TCP, link the stream to the TCP connection of the client */
-      link_stream (client, sstream);
+      link_stream (client, session, sstream);
     }
 
     stream = sstream->media_stream;
@@ -1131,15 +1156,12 @@ santize_uri (GstRTSPUrl * uri)
 static void
 client_session_finalized (GstRTSPClient * client, GstRTSPSession * session)
 {
-  GList *medias;
-
   GST_INFO ("client %p: session %p finished", client, session);
 
   /* unlink all media managed in this session */
-  for (medias = session->medias; medias; medias = g_list_next (medias)) {
-    unlink_session_streams (client, (GstRTSPSessionMedia *) medias->data);
-  }
+  client_unlink_session (client, session);
 
+  /* remove the session */
   if (!(client->sessions = g_list_remove (client->sessions, session))) {
     GST_INFO ("client %p: all sessions finalized, close the connection", client);
     close_connection (client);