From 6d227be7a9143ab41d0c56bb6c85f7edb4ed7e03 Mon Sep 17 00:00:00 2001 From: =?UTF-8?q?Sebastian=20P=C3=B6lsterl?= Date: Sat, 21 Nov 2009 19:20:23 +0100 Subject: [PATCH] Use GStreamer's debugging subsystem --- gst/rtsp-server/rtsp-client.c | 49 +++++++------ gst/rtsp-server/rtsp-media-factory.c | 13 ++-- gst/rtsp-server/rtsp-media-mapping.c | 5 +- gst/rtsp-server/rtsp-media.c | 101 ++++++++++++++------------- gst/rtsp-server/rtsp-server.c | 11 ++- gst/rtsp-server/rtsp-session-pool.c | 22 +++--- gst/rtsp-server/rtsp-session.c | 11 +-- 7 files changed, 120 insertions(+), 92 deletions(-) diff --git a/gst/rtsp-server/rtsp-client.c b/gst/rtsp-server/rtsp-client.c index 050969df4d..22477b28e9 100644 --- a/gst/rtsp-server/rtsp-client.c +++ b/gst/rtsp-server/rtsp-client.c @@ -36,6 +36,9 @@ enum PROP_LAST }; +GST_DEBUG_CATEGORY_STATIC (rtsp_client_debug); +#define GST_CAT_DEFAULT rtsp_client_debug + static void gst_rtsp_client_get_property (GObject * object, guint propid, GValue * value, GParamSpec * pspec); static void gst_rtsp_client_set_property (GObject * object, guint propid, @@ -73,6 +76,8 @@ gst_rtsp_client_class_init (GstRTSPClientClass * klass) tunnels = g_hash_table_new_full (g_str_hash, g_str_equal, g_free, g_object_unref); tunnels_lock = g_mutex_new (); + + GST_DEBUG_CATEGORY_INIT (rtsp_client_debug, "rtspclient", 0, "GstRTSPClient"); } static void @@ -87,7 +92,7 @@ gst_rtsp_client_finalize (GObject * obj) GstRTSPClient *client = GST_RTSP_CLIENT (obj); GList *walk; - g_message ("finalize client %p", client); + GST_INFO ("finalize client %p", client); /* remove weak-ref from sessions */ for (walk = client->sessions; walk; walk = g_list_next (walk)) { @@ -259,7 +264,7 @@ find_media (GstRTSPClient * client, GstRTSPUrl * uri, GstRTSPMessage * request) } else { /* we have seen this uri before, used cached media */ media = client->media; - g_message ("reusing cached media %p", media); + GST_INFO ("reusing cached media %p", media); } if (media) @@ -601,7 +606,7 @@ handle_play_request (GstRTSPClient * client, GstRTSPUrl * uri, sstream = gst_rtsp_session_media_get_stream (media, i); /* get the transport, if there is no transport configured, skip this stream */ if (!(tr = sstream->trans.transport)) { - g_message ("stream %d is not configured", i); + GST_INFO ("stream %d is not configured", i); continue; } @@ -633,7 +638,7 @@ handle_play_request (GstRTSPClient * client, GstRTSPUrl * uri, infocount++; } else { - g_warning ("RTP-Info cannot be determined for stream %d", i); + GST_WARNING ("RTP-Info cannot be determined for stream %d", i); } } @@ -685,7 +690,7 @@ invalid_state: static void do_keepalive (GstRTSPSession * session) { - g_message ("keep session %p alive", session); + GST_INFO ("keep session %p alive", session); gst_rtsp_session_touch (session); } @@ -1023,7 +1028,7 @@ static void client_session_finalized (GstRTSPClient * client, GstRTSPSession * session) { if (!(client->sessions = g_list_remove (client->sessions, session))) { - g_message ("all sessions finalized, close the connection"); + GST_INFO ("all sessions finalized, close the connection"); g_source_destroy ((GSource *) client->watch); } } @@ -1041,7 +1046,7 @@ client_watch_session (GstRTSPClient * client, GstRTSPSession * session) return; } - g_message ("watching session %p", session); + GST_INFO ("watching session %p", session); g_object_weak_ref (G_OBJECT (session), (GWeakNotify) client_session_finalized, client); @@ -1063,7 +1068,7 @@ handle_request (GstRTSPClient * client, GstRTSPMessage * request) gst_rtsp_message_dump (request); #endif - g_message ("client %p: received a request", client); + GST_INFO ("client %p: received a request", client); gst_rtsp_message_parse_request (request, &method, &uristr, &version); @@ -1327,7 +1332,7 @@ message_sent (GstRTSPWatch * watch, guint cseq, gpointer user_data) client = GST_RTSP_CLIENT (user_data); - /* g_message ("client %p: sent a message with cseq %d", client, cseq); */ + /* GST_INFO ("client %p: sent a message with cseq %d", client, cseq); */ return GST_RTSP_OK; } @@ -1338,7 +1343,7 @@ closed (GstRTSPWatch * watch, gpointer user_data) GstRTSPClient *client = GST_RTSP_CLIENT (user_data); const gchar *tunnelid; - g_message ("client %p: connection closed", client); + GST_INFO ("client %p: connection closed", client); if ((tunnelid = gst_rtsp_connection_get_tunnelid (client->connection))) { g_mutex_lock (tunnels_lock); @@ -1359,7 +1364,7 @@ error (GstRTSPWatch * watch, GstRTSPResult result, gpointer user_data) gchar *str; str = gst_rtsp_strresult (result); - g_message ("client %p: received an error %s", client, str); + GST_INFO ("client %p: received an error %s", client, str); g_free (str); return GST_RTSP_OK; @@ -1373,14 +1378,14 @@ tunnel_start (GstRTSPWatch * watch, gpointer user_data) client = GST_RTSP_CLIENT (user_data); - g_message ("client %p: tunnel start", client); + GST_INFO ("client %p: tunnel start", client); /* store client in the pending tunnels */ tunnelid = gst_rtsp_connection_get_tunnelid (client->connection); if (tunnelid == NULL) goto no_tunnelid; - g_message ("client %p: inserting %s", client, tunnelid); + GST_INFO ("client %p: inserting %s", client, tunnelid); /* we can't have two clients connecting with the same tunnelid */ g_mutex_lock (tunnels_lock); @@ -1395,13 +1400,13 @@ tunnel_start (GstRTSPWatch * watch, gpointer user_data) /* ERRORS */ no_tunnelid: { - g_message ("client %p: no tunnelid provided", client); + GST_INFO ("client %p: no tunnelid provided", client); return GST_RTSP_STS_SERVICE_UNAVAILABLE; } tunnel_existed: { g_mutex_unlock (tunnels_lock); - g_message ("client %p: tunnel session %s existed", client, tunnelid); + GST_INFO ("client %p: tunnel session %s existed", client, tunnelid); return GST_RTSP_STS_SERVICE_UNAVAILABLE; } } @@ -1413,7 +1418,7 @@ tunnel_complete (GstRTSPWatch * watch, gpointer user_data) GstRTSPClient *client = GST_RTSP_CLIENT (user_data); GstRTSPClient *oclient; - g_message ("client %p: tunnel complete", client); + GST_INFO ("client %p: tunnel complete", client); /* find previous tunnel */ tunnelid = gst_rtsp_connection_get_tunnelid (client->connection); @@ -1430,7 +1435,7 @@ tunnel_complete (GstRTSPWatch * watch, gpointer user_data) g_hash_table_remove (tunnels, tunnelid); g_mutex_unlock (tunnels_lock); - g_message ("client %p: found tunnel %p", client, oclient); + GST_INFO ("client %p: found tunnel %p", client, oclient); /* merge the tunnels into the first client */ gst_rtsp_connection_do_tunnel (oclient->connection, client->connection); @@ -1446,13 +1451,13 @@ tunnel_complete (GstRTSPWatch * watch, gpointer user_data) /* ERRORS */ no_tunnelid: { - g_message ("client %p: no tunnelid provided", client); + GST_INFO ("client %p: no tunnelid provided", client); return GST_RTSP_STS_SERVICE_UNAVAILABLE; } no_tunnel: { g_mutex_unlock (tunnels_lock); - g_message ("client %p: tunnel session %s not found", client, tunnelid); + GST_INFO ("client %p: tunnel session %s not found", client, tunnelid); return GST_RTSP_STS_SERVICE_UNAVAILABLE; } } @@ -1494,7 +1499,7 @@ gst_rtsp_client_accept (GstRTSPClient * client, GIOChannel * channel) GST_RTSP_CHECK (gst_rtsp_connection_accept (sock, &conn), accept_failed); url = gst_rtsp_connection_get_url (conn); - g_message ("added new client %p ip %s:%d", client, url->host, url->port); + GST_INFO ("added new client %p ip %s:%d", client, url->host, url->port); client->connection = conn; @@ -1508,7 +1513,7 @@ gst_rtsp_client_accept (GstRTSPClient * client, GIOChannel * channel) else context = NULL; - g_message ("attaching to context %p", context); + GST_INFO ("attaching to context %p", context); client->watchid = gst_rtsp_watch_attach (client->watch, context); gst_rtsp_watch_unref (client->watch); @@ -1520,7 +1525,7 @@ accept_failed: { gchar *str = gst_rtsp_strresult (res); - g_error ("Could not accept client on server socket %d: %s", sock, str); + GST_ERROR ("Could not accept client on server socket %d: %s", sock, str); g_free (str); return FALSE; } diff --git a/gst/rtsp-server/rtsp-media-factory.c b/gst/rtsp-server/rtsp-media-factory.c index 31823ce8fd..52f94714b8 100644 --- a/gst/rtsp-server/rtsp-media-factory.c +++ b/gst/rtsp-server/rtsp-media-factory.c @@ -30,6 +30,9 @@ enum PROP_LAST }; +GST_DEBUG_CATEGORY (rtsp_media_debug); +#define GST_CAT_DEFAULT rtsp_media_debug + static void gst_rtsp_media_factory_get_property (GObject *object, guint propid, GValue *value, GParamSpec *pspec); static void gst_rtsp_media_factory_set_property (GObject *object, guint propid, @@ -81,6 +84,8 @@ gst_rtsp_media_factory_class_init (GstRTSPMediaFactoryClass * klass) klass->construct = default_construct; klass->configure = default_configure; klass->create_pipeline = default_create_pipeline; + + GST_DEBUG_CATEGORY_INIT (rtsp_media_debug, "rtspmedia", 0, "GstRTSPMedia"); } static void @@ -340,7 +345,7 @@ gst_rtsp_media_factory_construct (GstRTSPMediaFactory *factory, const GstRTSPUrl if (key) g_free (key); - g_message ("constructed media %p for url %s", media, url->abspath); + GST_INFO ("constructed media %p for url %s", media, url->abspath); return media; } @@ -375,7 +380,7 @@ default_get_element (GstRTSPMediaFactory *factory, const GstRTSPUrl *url) if (error != NULL) { /* a recoverable error was encountered */ - g_warning ("recoverable parsing error: %s", error->message); + GST_WARNING ("recoverable parsing error: %s", error->message); g_error_free (error); } return element; @@ -424,7 +429,7 @@ gst_rtsp_media_factory_collect_streams (GstRTSPMediaFactory *factory, const GstR stream = g_new0 (GstRTSPMediaStream, 1); stream->payloader = elem; - g_message ("found stream %d with payloader %p", i, elem); + GST_INFO ("found stream %d with payloader %p", i, elem); pad = gst_element_get_static_pad (elem, "src"); @@ -444,7 +449,7 @@ gst_rtsp_media_factory_collect_streams (GstRTSPMediaFactory *factory, const GstR if ((elem = gst_bin_get_by_name (GST_BIN (element), name))) { /* a stream that will dynamically create pads to provide RTP packets */ - g_message ("found dynamic element %d, %p", i, elem); + GST_INFO ("found dynamic element %d, %p", i, elem); media->dynamic = g_list_prepend (media->dynamic, elem); diff --git a/gst/rtsp-server/rtsp-media-mapping.c b/gst/rtsp-server/rtsp-media-mapping.c index 52490cd329..b0020fd62a 100644 --- a/gst/rtsp-server/rtsp-media-mapping.c +++ b/gst/rtsp-server/rtsp-media-mapping.c @@ -21,6 +21,9 @@ G_DEFINE_TYPE (GstRTSPMediaMapping, gst_rtsp_media_mapping, G_TYPE_OBJECT); +GST_DEBUG_CATEGORY_EXTERN (rtsp_media_debug); +#define GST_CAT_DEFAULT rtsp_media_debug + static void gst_rtsp_media_mapping_finalize (GObject * obj); static GstRTSPMediaFactory * find_media (GstRTSPMediaMapping *mapping, const GstRTSPUrl *url); @@ -76,7 +79,7 @@ find_media (GstRTSPMediaMapping *mapping, const GstRTSPUrl *url) if (result) g_object_ref (result); - g_message ("found media %p for url abspath %s", result, url->abspath); + GST_INFO ("found media %p for url abspath %s", result, url->abspath); return result; } diff --git a/gst/rtsp-server/rtsp-media.c b/gst/rtsp-server/rtsp-media.c index fe79296b60..70de6fd652 100644 --- a/gst/rtsp-server/rtsp-media.c +++ b/gst/rtsp-server/rtsp-media.c @@ -44,6 +44,9 @@ enum SIGNAL_LAST }; +GST_DEBUG_CATEGORY_EXTERN (rtsp_media_debug); +#define GST_CAT_DEFAULT rtsp_media_debug + static GQuark ssrc_stream_map_key; static void gst_rtsp_media_get_property (GObject *object, guint propid, @@ -141,7 +144,7 @@ gst_rtsp_media_finalize (GObject * obj) media = GST_RTSP_MEDIA (obj); - g_message ("finalize media %p", media); + GST_INFO ("finalize media %p", media); if (media->pipeline) { unlock_streams (media); @@ -208,9 +211,9 @@ gst_rtsp_media_set_property (GObject *object, guint propid, static gpointer do_loop (GstRTSPMediaClass *klass) { - g_message ("enter mainloop"); + GST_INFO ("enter mainloop"); g_main_loop_run (klass->loop); - g_message ("exit mainloop"); + GST_INFO ("exit mainloop"); return NULL; } @@ -233,18 +236,18 @@ collect_media_stats (GstRTSPMedia *media) /* get the position */ format = GST_FORMAT_TIME; if (!gst_element_query_position (media->pipeline, &format, &position)) { - g_message ("position query failed"); + GST_INFO ("position query failed"); position = 0; } /* get the duration */ format = GST_FORMAT_TIME; if (!gst_element_query_duration (media->pipeline, &format, &duration)) { - g_message ("duration query failed"); + GST_INFO ("duration query failed"); duration = -1; } - g_message ("stats: position %"GST_TIME_FORMAT", duration %"GST_TIME_FORMAT, + GST_INFO ("stats: position %"GST_TIME_FORMAT", duration %"GST_TIME_FORMAT, GST_TIME_ARGS (position), GST_TIME_ARGS (duration)); if (position == -1) { @@ -457,21 +460,21 @@ gst_rtsp_media_seek (GstRTSPMedia *media, GstRTSPTimeRange *range) } if (start != -1 || stop != -1) { - g_message ("seeking to %"GST_TIME_FORMAT" - %"GST_TIME_FORMAT, + GST_INFO ("seeking to %"GST_TIME_FORMAT" - %"GST_TIME_FORMAT, GST_TIME_ARGS (start), GST_TIME_ARGS (stop)); res = gst_element_seek (media->pipeline, 1.0, GST_FORMAT_TIME, flags, start_type, start, stop_type, stop); /* and block for the seek to complete */ - g_message ("done seeking %d", res); + GST_INFO ("done seeking %d", res); gst_element_get_state (media->pipeline, NULL, NULL, -1); - g_message ("prerolled again"); + GST_INFO ("prerolled again"); collect_media_stats (media); } else { - g_message ("no seek needed"); + GST_INFO ("no seek needed"); res = TRUE; } @@ -480,12 +483,12 @@ gst_rtsp_media_seek (GstRTSPMedia *media, GstRTSPTimeRange *range) /* ERRORS */ not_supported: { - g_warning ("seek unit %d not supported", range->unit); + GST_WARNING ("seek unit %d not supported", range->unit); return FALSE; } weird_type: { - g_warning ("weird range type %d not supported", range->min.type); + GST_WARNING ("weird range type %d not supported", range->min.type); return FALSE; } } @@ -710,7 +713,7 @@ caps_notify (GstPad * pad, GParamSpec * unused, GstRTSPMediaStream * stream) gst_caps_unref (oldcaps); capsstr = gst_caps_to_string (newcaps); - g_message ("stream %p received caps %p, %s", stream, newcaps, capsstr); + GST_INFO ("stream %p received caps %p, %s", stream, newcaps, capsstr); g_free (capsstr); } @@ -720,7 +723,7 @@ dump_structure (const GstStructure *s) gchar *sstr; sstr = gst_structure_to_string (s); - g_message ("structure: %s", sstr); + GST_INFO ("structure: %s", sstr); g_free (sstr); } @@ -743,7 +746,7 @@ find_transport (GstRTSPMediaStream *stream, const gchar *rtcp_from) 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); diff --git a/gst/rtsp-server/rtsp-server.c b/gst/rtsp-server/rtsp-server.c index 0d1353d89b..3d690f8d53 100644 --- a/gst/rtsp-server/rtsp-server.c +++ b/gst/rtsp-server/rtsp-server.c @@ -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; diff --git a/gst/rtsp-server/rtsp-session-pool.c b/gst/rtsp-server/rtsp-session-pool.c index 7c68da1396..8fec319697 100644 --- a/gst/rtsp-server/rtsp-session-pool.c +++ b/gst/rtsp-server/rtsp-session-pool.c @@ -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; diff --git a/gst/rtsp-server/rtsp-session.c b/gst/rtsp-server/rtsp-session.c index 7525036445..ea33fa05bb 100644 --- a/gst/rtsp-server/rtsp-session.c +++ b/gst/rtsp-server/rtsp-session.c @@ -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; }