mirror of
https://gitlab.freedesktop.org/gstreamer/gstreamer.git
synced 2025-02-17 03:35:21 +00:00
Use GStreamer's debugging subsystem
This commit is contained in:
Sebastian Pölsterl
parent
87fbfa54a0
commit
6d227be7a9
7 changed files with 120 additions and 92 deletions
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@ -36,6 +36,9 @@ enum
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PROP_LAST
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};
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GST_DEBUG_CATEGORY_STATIC (rtsp_client_debug);
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#define GST_CAT_DEFAULT rtsp_client_debug
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static void gst_rtsp_client_get_property (GObject * object, guint propid,
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GValue * value, GParamSpec * pspec);
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static void gst_rtsp_client_set_property (GObject * object, guint propid,
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@ -73,6 +76,8 @@ gst_rtsp_client_class_init (GstRTSPClientClass * klass)
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tunnels =
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g_hash_table_new_full (g_str_hash, g_str_equal, g_free, g_object_unref);
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tunnels_lock = g_mutex_new ();
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GST_DEBUG_CATEGORY_INIT (rtsp_client_debug, "rtspclient", 0, "GstRTSPClient");
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}
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static void
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@ -87,7 +92,7 @@ gst_rtsp_client_finalize (GObject * obj)
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GstRTSPClient *client = GST_RTSP_CLIENT (obj);
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GList *walk;
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g_message ("finalize client %p", client);
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GST_INFO ("finalize client %p", client);
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/* remove weak-ref from sessions */
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for (walk = client->sessions; walk; walk = g_list_next (walk)) {
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@ -259,7 +264,7 @@ find_media (GstRTSPClient * client, GstRTSPUrl * uri, GstRTSPMessage * request)
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} else {
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/* we have seen this uri before, used cached media */
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media = client->media;
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g_message ("reusing cached media %p", media);
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GST_INFO ("reusing cached media %p", media);
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}
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if (media)
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@ -601,7 +606,7 @@ handle_play_request (GstRTSPClient * client, GstRTSPUrl * uri,
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sstream = gst_rtsp_session_media_get_stream (media, i);
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/* get the transport, if there is no transport configured, skip this stream */
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if (!(tr = sstream->trans.transport)) {
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g_message ("stream %d is not configured", i);
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GST_INFO ("stream %d is not configured", i);
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continue;
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}
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@ -633,7 +638,7 @@ handle_play_request (GstRTSPClient * client, GstRTSPUrl * uri,
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infocount++;
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} else {
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g_warning ("RTP-Info cannot be determined for stream %d", i);
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GST_WARNING ("RTP-Info cannot be determined for stream %d", i);
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}
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}
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@ -685,7 +690,7 @@ invalid_state:
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static void
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do_keepalive (GstRTSPSession * session)
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{
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g_message ("keep session %p alive", session);
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GST_INFO ("keep session %p alive", session);
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gst_rtsp_session_touch (session);
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}
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@ -1023,7 +1028,7 @@ static void
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client_session_finalized (GstRTSPClient * client, GstRTSPSession * session)
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{
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if (!(client->sessions = g_list_remove (client->sessions, session))) {
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g_message ("all sessions finalized, close the connection");
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GST_INFO ("all sessions finalized, close the connection");
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g_source_destroy ((GSource *) client->watch);
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}
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}
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@ -1041,7 +1046,7 @@ client_watch_session (GstRTSPClient * client, GstRTSPSession * session)
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return;
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}
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g_message ("watching session %p", session);
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GST_INFO ("watching session %p", session);
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g_object_weak_ref (G_OBJECT (session), (GWeakNotify) client_session_finalized,
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client);
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@ -1063,7 +1068,7 @@ handle_request (GstRTSPClient * client, GstRTSPMessage * request)
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gst_rtsp_message_dump (request);
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#endif
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g_message ("client %p: received a request", client);
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GST_INFO ("client %p: received a request", client);
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gst_rtsp_message_parse_request (request, &method, &uristr, &version);
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@ -1327,7 +1332,7 @@ message_sent (GstRTSPWatch * watch, guint cseq, gpointer user_data)
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client = GST_RTSP_CLIENT (user_data);
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/* g_message ("client %p: sent a message with cseq %d", client, cseq); */
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/* GST_INFO ("client %p: sent a message with cseq %d", client, cseq); */
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return GST_RTSP_OK;
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}
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@ -1338,7 +1343,7 @@ closed (GstRTSPWatch * watch, gpointer user_data)
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GstRTSPClient *client = GST_RTSP_CLIENT (user_data);
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const gchar *tunnelid;
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g_message ("client %p: connection closed", client);
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GST_INFO ("client %p: connection closed", client);
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if ((tunnelid = gst_rtsp_connection_get_tunnelid (client->connection))) {
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g_mutex_lock (tunnels_lock);
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@ -1359,7 +1364,7 @@ error (GstRTSPWatch * watch, GstRTSPResult result, gpointer user_data)
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gchar *str;
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str = gst_rtsp_strresult (result);
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g_message ("client %p: received an error %s", client, str);
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GST_INFO ("client %p: received an error %s", client, str);
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g_free (str);
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return GST_RTSP_OK;
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@ -1373,14 +1378,14 @@ tunnel_start (GstRTSPWatch * watch, gpointer user_data)
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client = GST_RTSP_CLIENT (user_data);
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g_message ("client %p: tunnel start", client);
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GST_INFO ("client %p: tunnel start", client);
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/* store client in the pending tunnels */
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tunnelid = gst_rtsp_connection_get_tunnelid (client->connection);
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if (tunnelid == NULL)
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goto no_tunnelid;
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g_message ("client %p: inserting %s", client, tunnelid);
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GST_INFO ("client %p: inserting %s", client, tunnelid);
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/* we can't have two clients connecting with the same tunnelid */
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g_mutex_lock (tunnels_lock);
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@ -1395,13 +1400,13 @@ tunnel_start (GstRTSPWatch * watch, gpointer user_data)
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/* ERRORS */
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no_tunnelid:
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{
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g_message ("client %p: no tunnelid provided", client);
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GST_INFO ("client %p: no tunnelid provided", client);
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return GST_RTSP_STS_SERVICE_UNAVAILABLE;
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}
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tunnel_existed:
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{
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g_mutex_unlock (tunnels_lock);
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g_message ("client %p: tunnel session %s existed", client, tunnelid);
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GST_INFO ("client %p: tunnel session %s existed", client, tunnelid);
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return GST_RTSP_STS_SERVICE_UNAVAILABLE;
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}
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}
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@ -1413,7 +1418,7 @@ tunnel_complete (GstRTSPWatch * watch, gpointer user_data)
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GstRTSPClient *client = GST_RTSP_CLIENT (user_data);
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GstRTSPClient *oclient;
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g_message ("client %p: tunnel complete", client);
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GST_INFO ("client %p: tunnel complete", client);
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/* find previous tunnel */
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tunnelid = gst_rtsp_connection_get_tunnelid (client->connection);
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@ -1430,7 +1435,7 @@ tunnel_complete (GstRTSPWatch * watch, gpointer user_data)
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g_hash_table_remove (tunnels, tunnelid);
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g_mutex_unlock (tunnels_lock);
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g_message ("client %p: found tunnel %p", client, oclient);
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GST_INFO ("client %p: found tunnel %p", client, oclient);
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/* merge the tunnels into the first client */
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gst_rtsp_connection_do_tunnel (oclient->connection, client->connection);
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@ -1446,13 +1451,13 @@ tunnel_complete (GstRTSPWatch * watch, gpointer user_data)
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/* ERRORS */
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no_tunnelid:
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{
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g_message ("client %p: no tunnelid provided", client);
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GST_INFO ("client %p: no tunnelid provided", client);
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return GST_RTSP_STS_SERVICE_UNAVAILABLE;
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}
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no_tunnel:
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{
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g_mutex_unlock (tunnels_lock);
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g_message ("client %p: tunnel session %s not found", client, tunnelid);
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GST_INFO ("client %p: tunnel session %s not found", client, tunnelid);
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return GST_RTSP_STS_SERVICE_UNAVAILABLE;
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}
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}
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@ -1494,7 +1499,7 @@ gst_rtsp_client_accept (GstRTSPClient * client, GIOChannel * channel)
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GST_RTSP_CHECK (gst_rtsp_connection_accept (sock, &conn), accept_failed);
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url = gst_rtsp_connection_get_url (conn);
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g_message ("added new client %p ip %s:%d", client, url->host, url->port);
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GST_INFO ("added new client %p ip %s:%d", client, url->host, url->port);
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client->connection = conn;
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@ -1508,7 +1513,7 @@ gst_rtsp_client_accept (GstRTSPClient * client, GIOChannel * channel)
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else
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context = NULL;
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g_message ("attaching to context %p", context);
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GST_INFO ("attaching to context %p", context);
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client->watchid = gst_rtsp_watch_attach (client->watch, context);
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gst_rtsp_watch_unref (client->watch);
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@ -1520,7 +1525,7 @@ accept_failed:
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{
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gchar *str = gst_rtsp_strresult (res);
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g_error ("Could not accept client on server socket %d: %s", sock, str);
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GST_ERROR ("Could not accept client on server socket %d: %s", sock, str);
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g_free (str);
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return FALSE;
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}
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@ -30,6 +30,9 @@ enum
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PROP_LAST
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};
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GST_DEBUG_CATEGORY (rtsp_media_debug);
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#define GST_CAT_DEFAULT rtsp_media_debug
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static void gst_rtsp_media_factory_get_property (GObject *object, guint propid,
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GValue *value, GParamSpec *pspec);
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static void gst_rtsp_media_factory_set_property (GObject *object, guint propid,
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@ -81,6 +84,8 @@ gst_rtsp_media_factory_class_init (GstRTSPMediaFactoryClass * klass)
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klass->construct = default_construct;
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klass->configure = default_configure;
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klass->create_pipeline = default_create_pipeline;
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GST_DEBUG_CATEGORY_INIT (rtsp_media_debug, "rtspmedia", 0, "GstRTSPMedia");
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}
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static void
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@ -340,7 +345,7 @@ gst_rtsp_media_factory_construct (GstRTSPMediaFactory *factory, const GstRTSPUrl
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if (key)
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g_free (key);
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g_message ("constructed media %p for url %s", media, url->abspath);
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GST_INFO ("constructed media %p for url %s", media, url->abspath);
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return media;
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}
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@ -375,7 +380,7 @@ default_get_element (GstRTSPMediaFactory *factory, const GstRTSPUrl *url)
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if (error != NULL) {
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/* a recoverable error was encountered */
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g_warning ("recoverable parsing error: %s", error->message);
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GST_WARNING ("recoverable parsing error: %s", error->message);
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g_error_free (error);
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}
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return element;
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@ -424,7 +429,7 @@ gst_rtsp_media_factory_collect_streams (GstRTSPMediaFactory *factory, const GstR
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stream = g_new0 (GstRTSPMediaStream, 1);
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stream->payloader = elem;
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g_message ("found stream %d with payloader %p", i, elem);
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GST_INFO ("found stream %d with payloader %p", i, elem);
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pad = gst_element_get_static_pad (elem, "src");
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@ -444,7 +449,7 @@ gst_rtsp_media_factory_collect_streams (GstRTSPMediaFactory *factory, const GstR
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if ((elem = gst_bin_get_by_name (GST_BIN (element), name))) {
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/* a stream that will dynamically create pads to provide RTP packets */
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g_message ("found dynamic element %d, %p", i, elem);
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GST_INFO ("found dynamic element %d, %p", i, elem);
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media->dynamic = g_list_prepend (media->dynamic, elem);
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@ -21,6 +21,9 @@
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G_DEFINE_TYPE (GstRTSPMediaMapping, gst_rtsp_media_mapping, G_TYPE_OBJECT);
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GST_DEBUG_CATEGORY_EXTERN (rtsp_media_debug);
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#define GST_CAT_DEFAULT rtsp_media_debug
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static void gst_rtsp_media_mapping_finalize (GObject * obj);
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static GstRTSPMediaFactory * find_media (GstRTSPMediaMapping *mapping, const GstRTSPUrl *url);
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@ -76,7 +79,7 @@ find_media (GstRTSPMediaMapping *mapping, const GstRTSPUrl *url)
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if (result)
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g_object_ref (result);
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g_message ("found media %p for url abspath %s", result, url->abspath);
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GST_INFO ("found media %p for url abspath %s", result, url->abspath);
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return result;
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}
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@ -44,6 +44,9 @@ enum
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SIGNAL_LAST
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};
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GST_DEBUG_CATEGORY_EXTERN (rtsp_media_debug);
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#define GST_CAT_DEFAULT rtsp_media_debug
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static GQuark ssrc_stream_map_key;
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static void gst_rtsp_media_get_property (GObject *object, guint propid,
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@ -141,7 +144,7 @@ gst_rtsp_media_finalize (GObject * obj)
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media = GST_RTSP_MEDIA (obj);
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g_message ("finalize media %p", media);
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GST_INFO ("finalize media %p", media);
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if (media->pipeline) {
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unlock_streams (media);
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@ -208,9 +211,9 @@ gst_rtsp_media_set_property (GObject *object, guint propid,
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static gpointer
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do_loop (GstRTSPMediaClass *klass)
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{
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g_message ("enter mainloop");
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GST_INFO ("enter mainloop");
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g_main_loop_run (klass->loop);
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g_message ("exit mainloop");
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GST_INFO ("exit mainloop");
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return NULL;
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}
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@ -233,18 +236,18 @@ collect_media_stats (GstRTSPMedia *media)
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/* get the position */
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format = GST_FORMAT_TIME;
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if (!gst_element_query_position (media->pipeline, &format, &position)) {
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g_message ("position query failed");
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GST_INFO ("position query failed");
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position = 0;
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}
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/* get the duration */
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format = GST_FORMAT_TIME;
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if (!gst_element_query_duration (media->pipeline, &format, &duration)) {
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g_message ("duration query failed");
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GST_INFO ("duration query failed");
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duration = -1;
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}
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g_message ("stats: position %"GST_TIME_FORMAT", duration %"GST_TIME_FORMAT,
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GST_INFO ("stats: position %"GST_TIME_FORMAT", duration %"GST_TIME_FORMAT,
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GST_TIME_ARGS (position), GST_TIME_ARGS (duration));
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if (position == -1) {
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@ -457,21 +460,21 @@ gst_rtsp_media_seek (GstRTSPMedia *media, GstRTSPTimeRange *range)
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}
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if (start != -1 || stop != -1) {
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g_message ("seeking to %"GST_TIME_FORMAT" - %"GST_TIME_FORMAT,
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GST_INFO ("seeking to %"GST_TIME_FORMAT" - %"GST_TIME_FORMAT,
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GST_TIME_ARGS (start), GST_TIME_ARGS (stop));
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res = gst_element_seek (media->pipeline, 1.0, GST_FORMAT_TIME,
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flags, start_type, start, stop_type, stop);
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/* and block for the seek to complete */
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g_message ("done seeking %d", res);
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GST_INFO ("done seeking %d", res);
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gst_element_get_state (media->pipeline, NULL, NULL, -1);
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g_message ("prerolled again");
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GST_INFO ("prerolled again");
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collect_media_stats (media);
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}
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else {
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g_message ("no seek needed");
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GST_INFO ("no seek needed");
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res = TRUE;
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}
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@ -480,12 +483,12 @@ gst_rtsp_media_seek (GstRTSPMedia *media, GstRTSPTimeRange *range)
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/* ERRORS */
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not_supported:
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{
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g_warning ("seek unit %d not supported", range->unit);
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GST_WARNING ("seek unit %d not supported", range->unit);
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return FALSE;
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}
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weird_type:
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{
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g_warning ("weird range type %d not supported", range->min.type);
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GST_WARNING ("weird range type %d not supported", range->min.type);
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return FALSE;
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}
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}
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@ -710,7 +713,7 @@ caps_notify (GstPad * pad, GParamSpec * unused, GstRTSPMediaStream * stream)
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gst_caps_unref (oldcaps);
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capsstr = gst_caps_to_string (newcaps);
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g_message ("stream %p received caps %p, %s", stream, newcaps, capsstr);
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GST_INFO ("stream %p received caps %p, %s", stream, newcaps, capsstr);
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g_free (capsstr);
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}
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@ -720,7 +723,7 @@ dump_structure (const GstStructure *s)
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gchar *sstr;
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sstr = gst_structure_to_string (s);
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g_message ("structure: %s", sstr);
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GST_INFO ("structure: %s", sstr);
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g_free (sstr);
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}
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@ -743,7 +746,7 @@ find_transport (GstRTSPMediaStream *stream, const gchar *rtcp_from)
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port = atoi (tmp + 1);
|
||||
dest = g_strndup (rtcp_from, tmp - rtcp_from);
|
||||
|
||||
g_message ("finding %s:%d", dest, port);
|
||||
GST_INFO ("finding %s:%d", dest, port);
|
||||
|
||||
for (walk = stream->transports; walk; walk = g_list_next (walk)) {
|
||||
GstRTSPMediaTrans *trans = walk->data;
|
||||
|
|
@ -768,7 +771,7 @@ on_new_ssrc (GObject *session, GObject *source, GstRTSPMediaStream *stream)
|
|||
GstStructure *stats;
|
||||
GstRTSPMediaTrans *trans;
|
||||
|
||||
g_message ("%p: new source %p", stream, source);
|
||||
GST_INFO ("%p: new source %p", stream, source);
|
||||
|
||||
/* see if we have a stream to match with the origin of the RTCP packet */
|
||||
trans = g_object_get_qdata (source, ssrc_stream_map_key);
|
||||
|
|
@ -781,7 +784,7 @@ on_new_ssrc (GObject *session, GObject *source, GstRTSPMediaStream *stream)
|
|||
|
||||
rtcp_from = gst_structure_get_string (stats, "rtcp-from");
|
||||
if ((trans = find_transport (stream, rtcp_from))) {
|
||||
g_message ("%p: found transport %p for source %p", stream, trans, source);
|
||||
GST_INFO ("%p: found transport %p for source %p", stream, trans, source);
|
||||
|
||||
/* keep ref to the source */
|
||||
trans->rtpsource = source;
|
||||
|
|
@ -791,14 +794,14 @@ on_new_ssrc (GObject *session, GObject *source, GstRTSPMediaStream *stream)
|
|||
gst_structure_free (stats);
|
||||
}
|
||||
} else {
|
||||
g_message ("%p: source %p for transport %p", stream, source, trans);
|
||||
GST_INFO ("%p: source %p for transport %p", stream, source, trans);
|
||||
}
|
||||
}
|
||||
|
||||
static void
|
||||
on_ssrc_sdes (GObject *session, GObject *source, GstRTSPMediaStream *stream)
|
||||
{
|
||||
g_message ("%p: new SDES %p", stream, source);
|
||||
GST_INFO ("%p: new SDES %p", stream, source);
|
||||
}
|
||||
|
||||
static void
|
||||
|
|
@ -808,7 +811,7 @@ on_ssrc_active (GObject *session, GObject *source, GstRTSPMediaStream *stream)
|
|||
|
||||
trans = g_object_get_qdata (source, ssrc_stream_map_key);
|
||||
|
||||
g_message ("%p: source %p in transport %p is active", stream, source, trans);
|
||||
GST_INFO ("%p: source %p in transport %p is active", stream, source, trans);
|
||||
|
||||
if (trans && trans->keep_alive)
|
||||
trans->keep_alive (trans->ka_user_data);
|
||||
|
|
@ -828,7 +831,7 @@ on_ssrc_active (GObject *session, GObject *source, GstRTSPMediaStream *stream)
|
|||
static void
|
||||
on_bye_ssrc (GObject *session, GObject *source, GstRTSPMediaStream *stream)
|
||||
{
|
||||
g_message ("%p: source %p bye", stream, source);
|
||||
GST_INFO ("%p: source %p bye", stream, source);
|
||||
}
|
||||
|
||||
static void
|
||||
|
|
@ -836,7 +839,7 @@ on_bye_timeout (GObject *session, GObject *source, GstRTSPMediaStream *stream)
|
|||
{
|
||||
GstRTSPMediaTrans *trans;
|
||||
|
||||
g_message ("%p: source %p bye timeout", stream, source);
|
||||
GST_INFO ("%p: source %p bye timeout", stream, source);
|
||||
|
||||
if ((trans = g_object_get_qdata (source, ssrc_stream_map_key))) {
|
||||
trans->rtpsource = NULL;
|
||||
|
|
@ -849,7 +852,7 @@ on_timeout (GObject *session, GObject *source, GstRTSPMediaStream *stream)
|
|||
{
|
||||
GstRTSPMediaTrans *trans;
|
||||
|
||||
g_message ("%p: source %p timeout", stream, source);
|
||||
GST_INFO ("%p: source %p timeout", stream, source);
|
||||
|
||||
if ((trans = g_object_get_qdata (source, ssrc_stream_map_key))) {
|
||||
trans->rtpsource = NULL;
|
||||
|
|
@ -1068,7 +1071,7 @@ setup_stream (GstRTSPMediaStream *stream, guint idx, GstRTSPMedia *media)
|
|||
/* ERRORS */
|
||||
link_failed:
|
||||
{
|
||||
g_warning ("failed to link stream %d", idx);
|
||||
GST_WARNING ("failed to link stream %d", idx);
|
||||
return FALSE;
|
||||
}
|
||||
}
|
||||
|
|
@ -1115,22 +1118,22 @@ default_handle_message (GstRTSPMedia *media, GstMessage *message)
|
|||
media->buffering = FALSE;
|
||||
/* if the desired state is playing, go back */
|
||||
if (media->target_state == GST_STATE_PLAYING) {
|
||||
g_message ("Buffering done, setting pipeline to PLAYING");
|
||||
GST_INFO ("Buffering done, setting pipeline to PLAYING");
|
||||
gst_element_set_state (media->pipeline, GST_STATE_PLAYING);
|
||||
}
|
||||
else {
|
||||
g_message ("Buffering done");
|
||||
GST_INFO ("Buffering done");
|
||||
}
|
||||
} else {
|
||||
/* buffering busy */
|
||||
if (media->buffering == FALSE) {
|
||||
if (media->target_state == GST_STATE_PLAYING) {
|
||||
/* we were not buffering but PLAYING, PAUSE the pipeline. */
|
||||
g_message ("Buffering, setting pipeline to PAUSED ...");
|
||||
GST_INFO ("Buffering, setting pipeline to PAUSED ...");
|
||||
gst_element_set_state (media->pipeline, GST_STATE_PAUSED);
|
||||
}
|
||||
else {
|
||||
g_message ("Buffering ...");
|
||||
GST_INFO ("Buffering ...");
|
||||
}
|
||||
}
|
||||
media->buffering = TRUE;
|
||||
|
|
@ -1148,7 +1151,7 @@ default_handle_message (GstRTSPMedia *media, GstMessage *message)
|
|||
gchar *debug;
|
||||
|
||||
gst_message_parse_error (message, &gerror, &debug);
|
||||
g_warning ("%p: got error %s (%s)", media, gerror->message, debug);
|
||||
GST_WARNING ("%p: got error %s (%s)", media, gerror->message, debug);
|
||||
g_error_free (gerror);
|
||||
g_free (debug);
|
||||
break;
|
||||
|
|
@ -1159,7 +1162,7 @@ default_handle_message (GstRTSPMedia *media, GstMessage *message)
|
|||
gchar *debug;
|
||||
|
||||
gst_message_parse_warning (message, &gerror, &debug);
|
||||
g_warning ("%p: got warning %s (%s)", media, gerror->message, debug);
|
||||
GST_WARNING ("%p: got warning %s (%s)", media, gerror->message, debug);
|
||||
g_error_free (gerror);
|
||||
g_free (debug);
|
||||
break;
|
||||
|
|
@ -1169,7 +1172,7 @@ default_handle_message (GstRTSPMedia *media, GstMessage *message)
|
|||
case GST_MESSAGE_STREAM_STATUS:
|
||||
break;
|
||||
default:
|
||||
g_message ("%p: got message type %s", media, gst_message_type_get_name (type));
|
||||
GST_INFO ("%p: got message type %s", media, gst_message_type_get_name (type));
|
||||
break;
|
||||
}
|
||||
return TRUE;
|
||||
|
|
@ -1200,7 +1203,7 @@ pad_added_cb (GstElement *element, GstPad *pad, GstRTSPMedia *media)
|
|||
|
||||
i = media->streams->len + 1;
|
||||
|
||||
g_message ("pad added %s:%s, stream %d", GST_DEBUG_PAD_NAME (pad), i);
|
||||
GST_INFO ("pad added %s:%s, stream %d", GST_DEBUG_PAD_NAME (pad), i);
|
||||
|
||||
stream = g_new0 (GstRTSPMediaStream, 1);
|
||||
stream->payloader = element;
|
||||
|
|
@ -1230,14 +1233,14 @@ pad_added_cb (GstElement *element, GstPad *pad, GstRTSPMedia *media)
|
|||
static void
|
||||
no_more_pads_cb (GstElement *element, GstRTSPMedia *media)
|
||||
{
|
||||
g_message ("no more pads");
|
||||
GST_INFO ("no more pads");
|
||||
if (media->fakesink) {
|
||||
gst_object_ref (media->fakesink);
|
||||
gst_bin_remove (GST_BIN (media->pipeline), media->fakesink);
|
||||
gst_element_set_state (media->fakesink, GST_STATE_NULL);
|
||||
gst_object_unref (media->fakesink);
|
||||
media->fakesink = NULL;
|
||||
g_message ("removed fakesink");
|
||||
GST_INFO ("removed fakesink");
|
||||
}
|
||||
}
|
||||
|
||||
|
|
@ -1268,7 +1271,7 @@ gst_rtsp_media_prepare (GstRTSPMedia *media)
|
|||
if (!media->reusable && media->reused)
|
||||
goto is_reused;
|
||||
|
||||
g_message ("preparing media %p", media);
|
||||
GST_INFO ("preparing media %p", media);
|
||||
|
||||
bus = gst_pipeline_get_bus (GST_PIPELINE_CAST (media->pipeline));
|
||||
|
||||
|
|
@ -1318,7 +1321,7 @@ gst_rtsp_media_prepare (GstRTSPMedia *media)
|
|||
break;
|
||||
case GST_STATE_CHANGE_NO_PREROLL:
|
||||
/* we need to go to PLAYING */
|
||||
g_message ("live media %p", media);
|
||||
GST_INFO ("live media %p", media);
|
||||
media->is_live = TRUE;
|
||||
ret = gst_element_set_state (media->pipeline, GST_STATE_PLAYING);
|
||||
if (ret == GST_STATE_CHANGE_FAILURE)
|
||||
|
|
@ -1336,7 +1339,7 @@ gst_rtsp_media_prepare (GstRTSPMedia *media)
|
|||
/* collect stats about the media */
|
||||
collect_media_stats (media);
|
||||
|
||||
g_message ("object %p is prerolled", media);
|
||||
GST_INFO ("object %p is prerolled", media);
|
||||
|
||||
media->prepared = TRUE;
|
||||
|
||||
|
|
@ -1350,14 +1353,14 @@ was_prepared:
|
|||
/* ERRORS */
|
||||
state_failed:
|
||||
{
|
||||
g_warning ("failed to preroll pipeline");
|
||||
GST_WARNING ("failed to preroll pipeline");
|
||||
unlock_streams (media);
|
||||
gst_element_set_state (media->pipeline, GST_STATE_NULL);
|
||||
return FALSE;
|
||||
}
|
||||
is_reused:
|
||||
{
|
||||
g_warning ("can not reuse media %p", media);
|
||||
GST_WARNING ("can not reuse media %p", media);
|
||||
return FALSE;
|
||||
}
|
||||
}
|
||||
|
|
@ -1381,7 +1384,7 @@ gst_rtsp_media_unprepare (GstRTSPMedia *media)
|
|||
if (!media->prepared)
|
||||
return TRUE;
|
||||
|
||||
g_message ("unprepare media %p", media);
|
||||
GST_INFO ("unprepare media %p", media);
|
||||
media->target_state = GST_STATE_NULL;
|
||||
|
||||
klass = GST_RTSP_MEDIA_GET_CLASS (media);
|
||||
|
|
@ -1435,7 +1438,7 @@ gst_rtsp_media_set_state (GstRTSPMedia *media, GstState state, GArray *transport
|
|||
|
||||
add = remove = FALSE;
|
||||
|
||||
g_message ("going to state %s media %p", gst_element_state_get_name (state), media);
|
||||
GST_INFO ("going to state %s media %p", gst_element_state_get_name (state), media);
|
||||
|
||||
switch (state) {
|
||||
case GST_STATE_NULL:
|
||||
|
|
@ -1484,14 +1487,14 @@ gst_rtsp_media_set_state (GstRTSPMedia *media, GstState state, GArray *transport
|
|||
max = trans->client_port.max;
|
||||
|
||||
if (add && !tr->active) {
|
||||
g_message ("adding %s:%d-%d", dest, min, max);
|
||||
GST_INFO ("adding %s:%d-%d", dest, min, max);
|
||||
g_signal_emit_by_name (stream->udpsink[0], "add", dest, min, NULL);
|
||||
g_signal_emit_by_name (stream->udpsink[1], "add", dest, max, NULL);
|
||||
stream->transports = g_list_prepend (stream->transports, tr);
|
||||
tr->active = TRUE;
|
||||
media->active++;
|
||||
} else if (remove && tr->active) {
|
||||
g_message ("removing %s:%d-%d", dest, min, max);
|
||||
GST_INFO ("removing %s:%d-%d", dest, min, max);
|
||||
g_signal_emit_by_name (stream->udpsink[0], "remove", dest, min, NULL);
|
||||
g_signal_emit_by_name (stream->udpsink[1], "remove", dest, max, NULL);
|
||||
stream->transports = g_list_remove (stream->transports, tr);
|
||||
|
|
@ -1502,19 +1505,19 @@ gst_rtsp_media_set_state (GstRTSPMedia *media, GstState state, GArray *transport
|
|||
}
|
||||
case GST_RTSP_LOWER_TRANS_TCP:
|
||||
if (add && !tr->active) {
|
||||
g_message ("adding TCP %s", trans->destination);
|
||||
GST_INFO ("adding TCP %s", trans->destination);
|
||||
stream->transports = g_list_prepend (stream->transports, tr);
|
||||
tr->active = TRUE;
|
||||
media->active++;
|
||||
} else if (remove && tr->active) {
|
||||
g_message ("removing TCP %s", trans->destination);
|
||||
GST_INFO ("removing TCP %s", trans->destination);
|
||||
stream->transports = g_list_remove (stream->transports, tr);
|
||||
tr->active = FALSE;
|
||||
media->active--;
|
||||
}
|
||||
break;
|
||||
default:
|
||||
g_message ("Unknown transport %d", trans->lower_transport);
|
||||
GST_INFO ("Unknown transport %d", trans->lower_transport);
|
||||
break;
|
||||
}
|
||||
}
|
||||
|
|
@ -1528,13 +1531,13 @@ gst_rtsp_media_set_state (GstRTSPMedia *media, GstState state, GArray *transport
|
|||
else
|
||||
do_state = FALSE;
|
||||
|
||||
g_message ("active %d media %p", media->active, media);
|
||||
GST_INFO ("active %d media %p", media->active, media);
|
||||
|
||||
if (do_state && media->target_state != state) {
|
||||
if (state == GST_STATE_NULL) {
|
||||
gst_rtsp_media_unprepare (media);
|
||||
} else {
|
||||
g_message ("state %s media %p", gst_element_state_get_name (state), media);
|
||||
GST_INFO ("state %s media %p", gst_element_state_get_name (state), media);
|
||||
media->target_state = state;
|
||||
ret = gst_element_set_state (media->pipeline, state);
|
||||
}
|
||||
|
|
@ -1563,7 +1566,7 @@ gst_rtsp_media_remove_elements (GstRTSPMedia *media)
|
|||
for (i = 0; i < media->streams->len; i++) {
|
||||
GstRTSPMediaStream *stream;
|
||||
|
||||
g_message ("Removing elements of stream %d from pipeline", i);
|
||||
GST_INFO ("Removing elements of stream %d from pipeline", i);
|
||||
|
||||
stream = g_array_index (media->streams, GstRTSPMediaStream *, i);
|
||||
|
||||
|
|
|
|||
|
|
@ -37,6 +37,9 @@ enum
|
|||
|
||||
G_DEFINE_TYPE (GstRTSPServer, gst_rtsp_server, G_TYPE_OBJECT);
|
||||
|
||||
GST_DEBUG_CATEGORY_STATIC (rtsp_server_debug);
|
||||
#define GST_CAT_DEFAULT rtsp_server_debug
|
||||
|
||||
static void gst_rtsp_server_get_property (GObject *object, guint propid,
|
||||
GValue *value, GParamSpec *pspec);
|
||||
static void gst_rtsp_server_set_property (GObject *object, guint propid,
|
||||
|
|
@ -102,6 +105,8 @@ gst_rtsp_server_class_init (GstRTSPServerClass * klass)
|
|||
GST_TYPE_RTSP_MEDIA_MAPPING, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
|
||||
|
||||
klass->accept_client = default_accept_client;
|
||||
|
||||
GST_DEBUG_CATEGORY_INIT (rtsp_server_debug, "rtspserver", 0, "GstRTSPServer");
|
||||
}
|
||||
|
||||
static void
|
||||
|
|
@ -399,7 +404,7 @@ gst_rtsp_server_sink_init_send (GstRTSPServer * server)
|
|||
"listened on server socket %d, returning from connection setup",
|
||||
server->server_sock.fd);
|
||||
|
||||
g_message ("listening on port %d", server->port);
|
||||
GST_INFO_OBJECT (server, "listening on port %d", server->port);
|
||||
|
||||
return TRUE;
|
||||
|
||||
|
|
@ -474,7 +479,7 @@ default_accept_client (GstRTSPServer *server, GIOChannel *channel)
|
|||
/* ERRORS */
|
||||
accept_failed:
|
||||
{
|
||||
g_error ("Could not accept client on server socket %d: %s (%d)",
|
||||
GST_ERROR_OBJECT (server, "Could not accept client on server socket %d: %s (%d)",
|
||||
server->server_sock.fd, g_strerror (errno), errno);
|
||||
gst_object_unref (client);
|
||||
return NULL;
|
||||
|
|
@ -511,7 +516,7 @@ gst_rtsp_server_io_func (GIOChannel *channel, GIOCondition condition, GstRTSPSer
|
|||
gst_object_unref (client);
|
||||
}
|
||||
else {
|
||||
g_print ("received unknown event %08x", condition);
|
||||
GST_WARNING_OBJECT (server, "received unknown event %08x", condition);
|
||||
}
|
||||
return TRUE;
|
||||
|
||||
|
|
|
|||
|
|
@ -30,6 +30,9 @@ enum
|
|||
PROP_LAST
|
||||
};
|
||||
|
||||
GST_DEBUG_CATEGORY (rtsp_session_debug);
|
||||
#define GST_CAT_DEFAULT rtsp_session_debug
|
||||
|
||||
static void gst_rtsp_session_pool_get_property (GObject *object, guint propid,
|
||||
GValue *value, GParamSpec *pspec);
|
||||
static void gst_rtsp_session_pool_set_property (GObject *object, guint propid,
|
||||
|
|
@ -59,6 +62,7 @@ gst_rtsp_session_pool_class_init (GstRTSPSessionPoolClass * klass)
|
|||
|
||||
klass->create_session_id = create_session_id;
|
||||
|
||||
GST_DEBUG_CATEGORY_INIT (rtsp_session_debug, "rtspsession", 0, "GstRTSPSession");
|
||||
}
|
||||
|
||||
static void
|
||||
|
|
@ -300,24 +304,24 @@ gst_rtsp_session_pool_create (GstRTSPSessionPool *pool)
|
|||
/* ERRORS */
|
||||
no_function:
|
||||
{
|
||||
g_warning ("no create_session_id vmethod in GstRTSPSessionPool %p", pool);
|
||||
GST_WARNING ("no create_session_id vmethod in GstRTSPSessionPool %p", pool);
|
||||
return NULL;
|
||||
}
|
||||
no_session:
|
||||
{
|
||||
g_warning ("can't create session id with GstRTSPSessionPool %p", pool);
|
||||
GST_WARNING ("can't create session id with GstRTSPSessionPool %p", pool);
|
||||
return NULL;
|
||||
}
|
||||
collision:
|
||||
{
|
||||
g_warning ("can't find unique sessionid for GstRTSPSessionPool %p", pool);
|
||||
GST_WARNING ("can't find unique sessionid for GstRTSPSessionPool %p", pool);
|
||||
g_mutex_unlock (pool->lock);
|
||||
g_free (id);
|
||||
return NULL;
|
||||
}
|
||||
too_many_sessions:
|
||||
{
|
||||
g_warning ("session pool reached max sessions of %d", pool->max_sessions);
|
||||
GST_WARNING ("session pool reached max sessions of %d", pool->max_sessions);
|
||||
g_mutex_unlock (pool->lock);
|
||||
g_free (id);
|
||||
return NULL;
|
||||
|
|
@ -464,7 +468,7 @@ collect_timeout (gchar *sessionid, GstRTSPSession *sess, GstPoolSource *psrc)
|
|||
g_source_get_current_time ((GSource*)psrc, &now);
|
||||
|
||||
timeout = gst_rtsp_session_next_timeout (sess, &now);
|
||||
g_message ("%p: next timeout: %d", sess, timeout);
|
||||
GST_INFO ("%p: next timeout: %d", sess, timeout);
|
||||
if (psrc->timeout == -1 || timeout < psrc->timeout)
|
||||
psrc->timeout = timeout;
|
||||
}
|
||||
|
|
@ -487,7 +491,7 @@ gst_pool_source_prepare (GSource * source, gint * timeout)
|
|||
|
||||
result = psrc->timeout == 0;
|
||||
|
||||
g_message ("prepare %d, %d", psrc->timeout, result);
|
||||
GST_INFO ("prepare %d, %d", psrc->timeout, result);
|
||||
|
||||
return result;
|
||||
}
|
||||
|
|
@ -495,7 +499,7 @@ gst_pool_source_prepare (GSource * source, gint * timeout)
|
|||
static gboolean
|
||||
gst_pool_source_check (GSource * source)
|
||||
{
|
||||
g_message ("check");
|
||||
GST_INFO ("check");
|
||||
|
||||
return gst_pool_source_prepare (source, NULL);
|
||||
}
|
||||
|
|
@ -508,7 +512,7 @@ gst_pool_source_dispatch (GSource * source, GSourceFunc callback,
|
|||
GstPoolSource *psrc = (GstPoolSource *) source;
|
||||
GstRTSPSessionPoolFunc func = (GstRTSPSessionPoolFunc) callback;
|
||||
|
||||
g_message ("dispatch");
|
||||
GST_INFO ("dispatch");
|
||||
|
||||
if (func)
|
||||
res = func (psrc->pool, user_data);
|
||||
|
|
@ -523,7 +527,7 @@ gst_pool_source_finalize (GSource * source)
|
|||
{
|
||||
GstPoolSource *psrc = (GstPoolSource *) source;
|
||||
|
||||
g_message ("finalize %p", psrc);
|
||||
GST_INFO ("finalize %p", psrc);
|
||||
|
||||
g_object_unref (psrc->pool);
|
||||
psrc->pool = NULL;
|
||||
|
|
|
|||
|
|
@ -32,6 +32,9 @@ enum
|
|||
PROP_LAST
|
||||
};
|
||||
|
||||
GST_DEBUG_CATEGORY_EXTERN (rtsp_session_debug);
|
||||
#define GST_CAT_DEFAULT rtsp_session_debug
|
||||
|
||||
static void gst_rtsp_session_get_property (GObject *object, guint propid,
|
||||
GValue *value, GParamSpec *pspec);
|
||||
static void gst_rtsp_session_set_property (GObject *object, guint propid,
|
||||
|
|
@ -72,7 +75,7 @@ gst_rtsp_session_init (GstRTSPSession * session)
|
|||
static void
|
||||
gst_rtsp_session_free_stream (GstRTSPSessionStream *stream)
|
||||
{
|
||||
g_message ("free session stream %p", stream);
|
||||
GST_INFO ("free session stream %p", stream);
|
||||
|
||||
/* remove callbacks now */
|
||||
gst_rtsp_session_stream_set_callbacks (stream, NULL, NULL, NULL, NULL);
|
||||
|
|
@ -91,7 +94,7 @@ gst_rtsp_session_free_media (GstRTSPSessionMedia *media, GstRTSPSession *session
|
|||
|
||||
size = media->streams->len;
|
||||
|
||||
g_message ("free session media %p", media);
|
||||
GST_INFO ("free session media %p", media);
|
||||
|
||||
gst_rtsp_session_media_set_state (media, GST_STATE_NULL);
|
||||
|
||||
|
|
@ -121,7 +124,7 @@ gst_rtsp_session_finalize (GObject * obj)
|
|||
|
||||
session = GST_RTSP_SESSION (obj);
|
||||
|
||||
g_message ("finalize session %p", session);
|
||||
GST_INFO ("finalize session %p", session);
|
||||
|
||||
/* free all media */
|
||||
g_list_foreach (session->medias, (GFunc) gst_rtsp_session_free_media,
|
||||
|
|
@ -208,7 +211,7 @@ gst_rtsp_session_manage_media (GstRTSPSession *sess, const GstRTSPUrl *uri,
|
|||
|
||||
sess->medias = g_list_prepend (sess->medias, result);
|
||||
|
||||
g_message ("manage new media %p in session %p", media, result);
|
||||
GST_INFO ("manage new media %p in session %p", media, result);
|
||||
|
||||
return result;
|
||||
}
|
||||
|
|
|
|||
Loading…
Reference in a new issue