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client: improve client cleanups

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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
This commit is contained in:
Wim Taymans 2010-04-06 17:08:40 +02:00
parent 4fdd2bf4d1
commit 336ffc0941
1 changed files with 43 additions and 21 deletions
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64
gst/rtsp-server/rtsp-client.c
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@ -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)
{
}
static void
client_unlink_session (GstRTSPClient *client, GstRTSPSession *session)
{
GList *medias;
/* 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 (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);
GList *walk;
GST_INFO ("finalize client %p", client);
/* 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),
(GWeakNotify) client_session_finalized, client);
}
g_list_free (client->sessions);
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);