rtsp-server: separate tizen feature code using pre-processor

[Version] 1.22.8-5

Change-Id: I5db791f8e8e7c4b13b037e2dedd99486a5dbc003
Signed-off-by: Youngwoo Cho <young222.cho@samsung.com>

diff --git a/packaging/gstreamer.spec b/packaging/gstreamer.spec
index b2899b9..fe3b1fa 100644 (file)
--- a/packaging/gstreamer.spec
+++ b/packaging/gstreamer.spec
@@ -62,7 +62,7 @@
 
 Name:           %{_name}
 Version:        1.22.8
-Release:        4
+Release:        5
 Summary:        Streaming-Media Framework Runtime
 License:        LGPL-2.0+
 Group:          Multimedia/Framework

diff --git a/subprojects/gst-rtsp-server/gst/rtsp-server/meson.build b/subprojects/gst-rtsp-server/gst/rtsp-server/meson.build
index 30c9da9..12c8bcf 100644 (file)
--- a/subprojects/gst-rtsp-server/gst/rtsp-server/meson.build
+++ b/subprojects/gst-rtsp-server/gst/rtsp-server/meson.build
@@ -1,3 +1,5 @@
+extra_defs = []
+
 rtsp_server_sources = files(
   'rtsp-address-pool.c',
   'rtsp-auth.c',
@@ -76,11 +78,14 @@ gst_rtsp_server = library('gstrtspserver-@0@'.format(api_version),
   install : true,
   dependencies : gst_rtsp_server_deps)
 
+extra_defs += ['-DTIZEN_FEATURE_RTSP_SERVER_MODIFICATION']
+
 library_def = {'lib': gst_rtsp_server}
 pkg_name = 'gstreamer-rtsp-server-' + api_version
 pkgconfig.generate(gst_rtsp_server,
   libraries : [gst_dep],
   subdirs : pkgconfig_subdirs,
+  extra_cflags : extra_defs,
   name : 'gstreamer-rtsp-server-1.0',
   description : 'GStreamer based RTSP server',
 )

diff --git a/subprojects/gst-rtsp-server/gst/rtsp-server/rtsp-client.c b/subprojects/gst-rtsp-server/gst/rtsp-server/rtsp-client.c
index 382ef2f..ccba62e 100644 (file)
--- a/subprojects/gst-rtsp-server/gst/rtsp-server/rtsp-client.c
+++ b/subprojects/gst-rtsp-server/gst/rtsp-server/rtsp-client.c
@@ -202,6 +202,8 @@ static GstRTSPResult default_params_get (GstRTSPClient * client,
     GstRTSPContext * ctx);
 static gchar *default_make_path_from_uri (GstRTSPClient * client,
     const GstRTSPUrl * uri);
+
+#ifdef TIZEN_FEATURE_RTSP_SERVER_MODIFICATION
 static gboolean default_handle_options_request (GstRTSPClient * client,
     GstRTSPContext * ctx, GstRTSPVersion version);
 static gboolean default_handle_set_param_request (GstRTSPClient * client,
@@ -210,6 +212,7 @@ static gboolean default_handle_get_param_request (GstRTSPClient * client,
     GstRTSPContext * ctx);
 static gboolean default_handle_play_request (GstRTSPClient * client,
     GstRTSPContext * ctx);
+#endif
 
 static void client_session_removed (GstRTSPSessionPool * pool,
     GstRTSPSession * session, GstRTSPClient * client);
@@ -238,10 +241,13 @@ gst_rtsp_client_class_init (GstRTSPClientClass * klass)
   klass->params_set = default_params_set;
   klass->params_get = default_params_get;
   klass->make_path_from_uri = default_make_path_from_uri;
+
+#ifdef TIZEN_FEATURE_RTSP_SERVER_MODIFICATION
   klass->handle_options_request = default_handle_options_request;
   klass->handle_set_param_request = default_handle_set_param_request;
   klass->handle_get_param_request = default_handle_get_param_request;
   klass->handle_play_request = default_handle_play_request;
+#endif
 
   klass->pre_options_request = default_pre_signal_handler;
   klass->pre_describe_request = default_pre_signal_handler;
@@ -1611,7 +1617,11 @@ default_params_get (GstRTSPClient * client, GstRTSPContext * ctx)
 }
 
 static gboolean
+#ifdef TIZEN_FEATURE_RTSP_SERVER_MODIFICATION
 default_handle_get_param_request (GstRTSPClient * client, GstRTSPContext * ctx)
+#else
+handle_get_param_request (GstRTSPClient * client, GstRTSPContext * ctx)
+#endif
 {
   GstRTSPResult res;
   guint8 *data;
@@ -1669,7 +1679,11 @@ bad_request:
 }
 
 static gboolean
+#ifdef TIZEN_FEATURE_RTSP_SERVER_MODIFICATION
 default_handle_set_param_request (GstRTSPClient * client, GstRTSPContext * ctx)
+#else
+handle_set_param_request (GstRTSPClient * client, GstRTSPContext * ctx)
+#endif
 {
   GstRTSPResult res;
   guint8 *data;
@@ -2088,7 +2102,11 @@ seek_failed:
 }
 
 static gboolean
+#ifdef TIZEN_FEATURE_RTSP_SERVER_MODIFICATION
 default_handle_play_request (GstRTSPClient * client, GstRTSPContext * ctx)
+#else
+handle_play_request (GstRTSPClient * client, GstRTSPContext * ctx)
+#endif
 {
   GstRTSPSession *session;
   GstRTSPClientClass *klass;
@@ -3054,8 +3072,10 @@ handle_setup_request (GstRTSPClient * client, GstRTSPContext * ctx)
   st = make_server_transport (client, media, ctx, ct);
   trans_str = gst_rtsp_transport_as_text (st);
 
+#ifdef TIZEN_FEATURE_RTSP_SERVER_MODIFICATION
   /* FIXME-WFD : Temporarily force to set profile string */
   trans_str = g_strjoinv ("RTP/AVP/UDP", g_strsplit (trans_str, "RTP/AVP", -1));
+#endif
 
   gst_rtsp_transport_free (st);
 
@@ -3817,7 +3837,11 @@ unsuspend_failed:
 }
 
 static gboolean
+#ifdef TIZEN_FEATURE_RTSP_SERVER_MODIFICATION
 default_handle_options_request (GstRTSPClient * client, GstRTSPContext * ctx,
+#else
+handle_options_request (GstRTSPClient * client, GstRTSPContext * ctx,
+#endif
     GstRTSPVersion version)
 {
   GstRTSPMethod options;
@@ -4016,9 +4040,11 @@ handle_request (GstRTSPClient * client, GstRTSPMessage * request)
   GstRTSPMessage response = { 0 };
   gchar *unsupported_reqs = NULL;
   gchar *sessid = NULL, *pipelined_request_id = NULL;
+#ifdef TIZEN_FEATURE_RTSP_SERVER_MODIFICATION
   GstRTSPClientClass *klass;
 
   klass = GST_RTSP_CLIENT_GET_CLASS (client);
+#endif
   if (!(ctx = gst_rtsp_context_get_current ())) {
     ctx = &sctx;
     ctx->auth = priv->auth;
@@ -4125,7 +4151,11 @@ handle_request (GstRTSPClient * client, GstRTSPMessage * request)
   switch (method) {
     case GST_RTSP_OPTIONS:
       priv->version = version;
+#ifdef TIZEN_FEATURE_RTSP_SERVER_MODIFICATION
       klass->handle_options_request (client, ctx, version);
+#else
+      handle_options_request (client, ctx, version);
+#endif
       break;
     case GST_RTSP_DESCRIBE:
       handle_describe_request (client, ctx);
@@ -4134,7 +4164,11 @@ handle_request (GstRTSPClient * client, GstRTSPMessage * request)
       handle_setup_request (client, ctx);
       break;
     case GST_RTSP_PLAY:
+#ifdef TIZEN_FEATURE_RTSP_SERVER_MODIFICATION
       klass->handle_play_request (client, ctx);
+#else
+      handle_play_request (client, ctx);
+#endif
       break;
     case GST_RTSP_PAUSE:
       handle_pause_request (client, ctx);
@@ -4143,10 +4177,18 @@ handle_request (GstRTSPClient * client, GstRTSPMessage * request)
       handle_teardown_request (client, ctx);
       break;
     case GST_RTSP_SET_PARAMETER:
+#ifdef TIZEN_FEATURE_RTSP_SERVER_MODIFICATION
       klass->handle_set_param_request (client, ctx);
+#else
+      handle_set_param_request (client, ctx);
+#endif
       break;
     case GST_RTSP_GET_PARAMETER:
+#ifdef TIZEN_FEATURE_RTSP_SERVER_MODIFICATION
       klass->handle_get_param_request (client, ctx);
+#else
+      handle_get_param_request (client, ctx);
+#endif
       break;
     case GST_RTSP_ANNOUNCE:
       if (version >= GST_RTSP_VERSION_2_0)
@@ -4711,7 +4753,9 @@ gst_rtsp_client_set_connection (GstRTSPClient * client,
   } else {
     priv->is_ipv6 = g_socket_get_family (read_socket) == G_SOCKET_FAMILY_IPV6;
     priv->server_ip = g_strdup ("unknown");
+#ifdef TIZEN_FEATURE_BUG_FIX
     g_object_unref (address);
+#endif
   }
 
   GST_INFO ("client %p connected to server ip %s, ipv6 = %d", client,
@@ -5487,6 +5531,7 @@ restart:
   return result;
 }
 
+#ifdef TIZEN_FEATURE_RTSP_SERVER_MODIFICATION
 /**
  * gst_rtsp_client_set_watch_flushing:
  * @client: a #GstRTSPClient
@@ -5508,3 +5553,4 @@ gst_rtsp_client_set_watch_flushing (GstRTSPClient * client, gboolean val)
     gst_rtsp_watch_set_flushing (priv->watch, val);
   }
 }
+#endif

diff --git a/subprojects/gst-rtsp-server/gst/rtsp-server/rtsp-client.h b/subprojects/gst-rtsp-server/gst/rtsp-server/rtsp-client.h
index 7df4c52..1d5d8ec 100644 (file)
--- a/subprojects/gst-rtsp-server/gst/rtsp-server/rtsp-client.h
+++ b/subprojects/gst-rtsp-server/gst/rtsp-server/rtsp-client.h
@@ -131,10 +131,12 @@ struct _GstRTSPClientClass {
   GstRTSPResult   (*params_set) (GstRTSPClient *client, GstRTSPContext *ctx);
   GstRTSPResult   (*params_get) (GstRTSPClient *client, GstRTSPContext *ctx);
   gchar *         (*make_path_from_uri) (GstRTSPClient *client, const GstRTSPUrl *uri);
+#ifdef TIZEN_FEATURE_RTSP_SERVER_MODIFICATION
   gboolean        (*handle_options_request) (GstRTSPClient * client, GstRTSPContext * ctx, GstRTSPVersion version);
   gboolean        (*handle_set_param_request) (GstRTSPClient * client, GstRTSPContext * ctx);
   gboolean        (*handle_get_param_request) (GstRTSPClient * client, GstRTSPContext * ctx);
   gboolean        (*handle_play_request) (GstRTSPClient * client, GstRTSPContext * ctx);
+#endif
   GstRTSPStatusCode (*adjust_play_mode) (GstRTSPClient * client,
                                          GstRTSPContext * context,
                                          GstRTSPTimeRange ** range,
@@ -271,8 +273,10 @@ GstRTSPResult         gst_rtsp_client_send_message      (GstRTSPClient * client,
                                                          GstRTSPSession *session,
                                                          GstRTSPMessage *message);
 
+#ifdef TIZEN_FEATURE_RTSP_SERVER_MODIFICATION
 GST_RTSP_SERVER_API
 void                  gst_rtsp_client_set_watch_flushing (GstRTSPClient * client, gboolean val);
+#endif
 /**
  * GstRTSPClientSessionFilterFunc:
  * @client: a #GstRTSPClient object

diff --git a/subprojects/gst-rtsp-server/gst/rtsp-server/rtsp-media-factory-uri.c b/subprojects/gst-rtsp-server/gst/rtsp-server/rtsp-media-factory-uri.c
index 22befcc..fe6b99c 100644 (file)
--- a/subprojects/gst-rtsp-server/gst/rtsp-server/rtsp-media-factory-uri.c
+++ b/subprojects/gst-rtsp-server/gst/rtsp-server/rtsp-media-factory-uri.c
@@ -641,7 +641,9 @@ no_uridecodebin:
   {
     g_critical ("can't create uridecodebin element");
     gst_object_unref (element);
+#ifdef TIZEN_FEATURE_BUG_FIX
     gst_object_unref (topbin);
+#endif
     return NULL;
   }
 }

diff --git a/subprojects/gst-rtsp-server/gst/rtsp-server/rtsp-media.c b/subprojects/gst-rtsp-server/gst/rtsp-server/rtsp-media.c
index b5bf472..995e0d9 100644 (file)
--- a/subprojects/gst-rtsp-server/gst/rtsp-server/rtsp-media.c
+++ b/subprojects/gst-rtsp-server/gst/rtsp-server/rtsp-media.c
@@ -218,8 +218,10 @@ enum
   SIGNAL_TARGET_STATE,
   SIGNAL_NEW_STATE,
   SIGNAL_HANDLE_MESSAGE,
+#ifdef TIZEN_FEATURE_RTSP_SERVER_MODIFICATION
   SIGNAL_PREPARING,
   SIGNAL_UNPREPARING,
+#endif
   SIGNAL_LAST
 };
 
@@ -236,7 +238,9 @@ static gboolean default_handle_message (GstRTSPMedia * media,
     GstMessage * message);
 static void finish_unprepare (GstRTSPMedia * media);
 static gboolean default_prepare (GstRTSPMedia * media, GstRTSPThread * thread);
+#ifdef TIZEN_FEATURE_RTSP_SERVER_MODIFICATION
 static gboolean default_start_preroll (GstRTSPMedia * media);
+#endif
 static gboolean default_unprepare (GstRTSPMedia * media);
 static gboolean default_suspend (GstRTSPMedia * media);
 static gboolean default_unsuspend (GstRTSPMedia * media);
@@ -249,7 +253,9 @@ static GstElement *default_create_rtpbin (GstRTSPMedia * media);
 static gboolean default_setup_sdp (GstRTSPMedia * media, GstSDPMessage * sdp,
     GstSDPInfo * info);
 static gboolean default_handle_sdp (GstRTSPMedia * media, GstSDPMessage * sdp);
+#ifdef TIZEN_FEATURE_RTSP_SERVER_MODIFICATION
 static gboolean default_start_prepare (GstRTSPMedia * media);
+#endif
 
 static gboolean wait_preroll (GstRTSPMedia * media);
 
@@ -474,6 +480,7 @@ gst_rtsp_media_class_init (GstRTSPMediaClass * klass)
           handle_message), NULL, NULL, NULL, G_TYPE_BOOLEAN, 1,
       GST_TYPE_MESSAGE);
 
+#ifdef TIZEN_FEATURE_RTSP_SERVER_MODIFICATION
   gst_rtsp_media_signals[SIGNAL_PREPARING] =
       g_signal_new ("preparing", G_TYPE_FROM_CLASS (klass), G_SIGNAL_RUN_LAST,
       G_STRUCT_OFFSET (GstRTSPMediaClass, preparing), NULL, NULL,
@@ -485,12 +492,15 @@ gst_rtsp_media_class_init (GstRTSPMediaClass * klass)
       G_STRUCT_OFFSET (GstRTSPMediaClass, unpreparing), NULL, NULL,
       NULL, G_TYPE_NONE, 2, GST_TYPE_RTSP_STREAM,
       G_TYPE_UINT);
+#endif
 
   GST_DEBUG_CATEGORY_INIT (rtsp_media_debug, "rtspmedia", 0, "GstRTSPMedia");
 
   klass->handle_message = default_handle_message;
   klass->prepare = default_prepare;
+#ifdef TIZEN_FEATURE_RTSP_SERVER_MODIFICATION
   klass->start_preroll = default_start_preroll;
+#endif
   klass->unprepare = default_unprepare;
   klass->suspend = default_suspend;
   klass->unsuspend = default_unsuspend;
@@ -500,7 +510,9 @@ gst_rtsp_media_class_init (GstRTSPMediaClass * klass)
   klass->create_rtpbin = default_create_rtpbin;
   klass->setup_sdp = default_setup_sdp;
   klass->handle_sdp = default_handle_sdp;
+#ifdef TIZEN_FEATURE_RTSP_SERVER_MODIFICATION
   klass->start_prepare = default_start_prepare;
+#endif
 }
 
 static void
@@ -2822,7 +2834,11 @@ media_streams_set_blocked (GstRTSPMedia * media, gboolean blocked)
     priv->blocking_msg_received = 0;
 }
 
+#ifdef TIZEN_FEATURE_RTSP_SERVER_MODIFICATION
 void
+#else
+static void
+#endif
 gst_rtsp_media_set_status (GstRTSPMedia * media, GstRTSPMediaStatus status)
 {
   GstRTSPMediaPrivate *priv = media->priv;
@@ -3180,6 +3196,7 @@ set_state (GstRTSPMedia * media, GstState state)
       media);
   ret = gst_element_set_state (priv->pipeline, state);
 
+#ifdef TIZEN_FEATURE_RTSP_SERVER_MODIFICATION
   {
     gchar *filename = NULL;
     filename = g_strdup_printf ("media_%s", gst_element_state_get_name (state));
@@ -3188,12 +3205,18 @@ set_state (GstRTSPMedia * media, GstState state)
 
     g_free (filename);
   }
+#endif
 
   return ret;
 }
 
+#ifdef TIZEN_FEATURE_RTSP_SERVER_MODIFICATION
 GstStateChangeReturn
 gst_rtsp_media_set_target_state (GstRTSPMedia * media, GstState state, gboolean do_state)
+#else
+static GstStateChangeReturn
+set_target_state (GstRTSPMedia * media, GstState state, gboolean do_state)
+#endif
 {
   GstRTSPMediaPrivate *priv = media->priv;
   GstStateChangeReturn ret;
@@ -3604,7 +3627,11 @@ struct _DynPaySignalHandlers
 };
 
 static gboolean
+#ifdef TIZEN_FEATURE_RTSP_SERVER_MODIFICATION
 default_start_preroll (GstRTSPMedia * media)
+#else
+start_preroll (GstRTSPMedia * media)
+#endif
 {
   GstRTSPMediaPrivate *priv = media->priv;
   GstStateChangeReturn ret;
@@ -3613,7 +3640,11 @@ default_start_preroll (GstRTSPMedia * media)
 
   /* start blocked since it is possible that there are no sink elements yet */
   media_streams_set_blocked (media, TRUE);
+#ifdef TIZEN_FEATURE_RTSP_SERVER_MODIFICATION
   ret = gst_rtsp_media_set_target_state (media, GST_STATE_PAUSED, TRUE);
+#else
+  ret = set_target_state (media, GST_STATE_PAUSED, TRUE);
+#endif
 
   switch (ret) {
     case GST_STATE_CHANGE_SUCCESS:
@@ -3745,10 +3776,16 @@ request_fec_decoder (GstElement * rtpbin, guint sessid, GstRTSPMedia * media)
 }
 
 static gboolean
+#ifdef TIZEN_FEATURE_RTSP_SERVER_MODIFICATION
 default_start_prepare (GstRTSPMedia * media)
+#else
+start_prepare (GstRTSPMedia * media)
+#endif
 {
   GstRTSPMediaPrivate *priv = media->priv;
+#ifdef TIZEN_FEATURE_RTSP_SERVER_MODIFICATION
   GstRTSPMediaClass *klass;
+#endif
   guint i;
   GList *walk;
 
@@ -3758,8 +3795,9 @@ default_start_prepare (GstRTSPMedia * media)
 
   g_signal_connect (priv->rtpbin, "request-fec-decoder",
       G_CALLBACK (request_fec_decoder), media);
-
+#ifdef TIZEN_FEATURE_RTSP_SERVER_MODIFICATION
   klass = GST_RTSP_MEDIA_GET_CLASS (media);
+#endif
 
   /* link streams we already have, other streams might appear when we have
    * dynamic elements */
@@ -3784,8 +3822,10 @@ default_start_prepare (GstRTSPMedia * media)
       goto join_bin_failed;
     }
 
+#ifdef TIZEN_FEATURE_RTSP_SERVER_MODIFICATION
     g_signal_emit (media, gst_rtsp_media_signals[SIGNAL_PREPARING], 0, stream,
         i, NULL);
+#endif
   }
 
   if (priv->rtpbin)
@@ -3813,7 +3853,11 @@ default_start_prepare (GstRTSPMedia * media)
      * a second ASYNC state change failing */
     priv->is_live = TRUE;
     gst_rtsp_media_set_status (media, GST_RTSP_MEDIA_STATUS_PREPARED);
+#ifdef TIZEN_FEATURE_RTSP_SERVER_MODIFICATION
   } else if (!klass->start_preroll (media)) {
+#else
+  } else if (!start_preroll (media)) {
+#endif
     goto preroll_failed;
   }
 
@@ -3895,9 +3939,14 @@ default_prepare (GstRTSPMedia * media, GstRTSPThread * thread)
 
   /* do remainder in context */
   source = g_idle_source_new ();
+#ifdef TIZEN_FEATURE_RTSP_SERVER_MODIFICATION
   if (klass->start_prepare)
     g_source_set_callback (source, (GSourceFunc) klass->start_prepare,
         g_object_ref (media), (GDestroyNotify) g_object_unref);
+#else
+  g_source_set_callback (source, (GSourceFunc) start_prepare,
+      g_object_ref (media), (GDestroyNotify) g_object_unref);
+#endif
   g_source_attach (source, context);
   g_source_unref (source);
 
@@ -4077,8 +4126,10 @@ finish_unprepare (GstRTSPMedia * media)
 
     gst_rtsp_stream_leave_bin (stream, GST_BIN (priv->pipeline), priv->rtpbin);
 
+#ifdef TIZEN_FEATURE_RTSP_SERVER_MODIFICATION
     g_signal_emit (media, gst_rtsp_media_signals[SIGNAL_UNPREPARING], 0, stream,
         i, NULL);
+#endif
   }
 
   /* remove the pad signal handlers */
@@ -4187,7 +4238,11 @@ gst_rtsp_media_unprepare (GstRTSPMedia * media)
     goto is_unpreparing;
 
   GST_INFO ("unprepare media %p", media);
+#ifdef TIZEN_FEATURE_RTSP_SERVER_MODIFICATION
   gst_rtsp_media_set_target_state (media, GST_STATE_NULL, FALSE);
+#else
+  set_target_state (media, GST_STATE_NULL, FALSE);
+#endif
   success = TRUE;
 
   if (priv->status == GST_RTSP_MEDIA_STATUS_PREPARED
@@ -4607,13 +4662,21 @@ default_suspend (GstRTSPMedia * media)
       break;
     case GST_RTSP_SUSPEND_MODE_PAUSE:
       GST_DEBUG ("media %p suspend to PAUSED", media);
+#ifdef TIZEN_FEATURE_RTSP_SERVER_MODIFICATION
       ret = gst_rtsp_media_set_target_state (media, GST_STATE_PAUSED, TRUE);
+#else
+      ret = set_target_state (media, GST_STATE_PAUSED, TRUE);
+#endif
       if (ret == GST_STATE_CHANGE_FAILURE)
         goto state_failed;
       break;
     case GST_RTSP_SUSPEND_MODE_RESET:
       GST_DEBUG ("media %p suspend to NULL", media);
+#ifdef TIZEN_FEATURE_RTSP_SERVER_MODIFICATION
       ret = gst_rtsp_media_set_target_state (media, GST_STATE_NULL, TRUE);
+#else
+      ret = set_target_state (media, GST_STATE_NULL, TRUE);
+#endif
       if (ret == GST_STATE_CHANGE_FAILURE)
         goto state_failed;
       /* Because payloader needs to set the sequence number as
@@ -4720,9 +4783,11 @@ default_unsuspend (GstRTSPMedia * media)
 {
   GstRTSPMediaPrivate *priv = media->priv;
   gboolean preroll_ok;
+#ifdef TIZEN_FEATURE_RTSP_SERVER_MODIFICATION
   GstRTSPMediaClass *klass;
 
   klass = GST_RTSP_MEDIA_GET_CLASS (media);
+#endif
 
   switch (priv->suspend_mode) {
     case GST_RTSP_SUSPEND_MODE_NONE:
@@ -4747,9 +4812,14 @@ default_unsuspend (GstRTSPMedia * media)
        * specific transport parts: all active streams contain at least one sink
        * element and it's safe to unblock all blocked streams */
       media_streams_set_blocked (media, FALSE);
+#ifdef TIZEN_FEATURE_RTSP_SERVER_MODIFICATION
        if (klass->start_preroll)
         if (!klass->start_preroll (media))
           goto start_failed;
+#else
+      if (!start_preroll (media))
+        goto start_failed;
+#endif
 
       g_rec_mutex_unlock (&priv->state_lock);
       preroll_ok = wait_preroll (media);
@@ -4847,7 +4917,11 @@ media_set_pipeline_state_locked (GstRTSPMedia * media, GstState state)
     gst_rtsp_media_unprepare (media);
   } else {
     GST_INFO ("state %s media %p", gst_element_state_get_name (state), media);
+#ifdef TIZEN_FEATURE_RTSP_SERVER_MODIFICATION
     gst_rtsp_media_set_target_state (media, state, FALSE);
+#else
+    set_target_state (media, state, FALSE);
+#endif
 
     if (state == GST_STATE_PLAYING) {
       /* make sure pads are not blocking anymore when going to PLAYING */
@@ -5288,6 +5362,7 @@ gst_rtsp_media_get_rate_control (GstRTSPMedia * media)
   return res;
 }
 
+#ifdef TIZEN_FEATURE_RTSP_SERVER_MODIFICATION
 GstElement *
 gst_rtsp_media_get_pipeline (GstRTSPMedia * media)
 {
@@ -5320,3 +5395,4 @@ gst_rtsp_media_get_rtpbin (GstRTSPMedia * media)
 
   return priv->rtpbin;
 }
+#endif

diff --git a/subprojects/gst-rtsp-server/gst/rtsp-server/rtsp-media.h b/subprojects/gst-rtsp-server/gst/rtsp-server/rtsp-media.h
index d2d2c99..8e33669 100644 (file)
--- a/subprojects/gst-rtsp-server/gst/rtsp-server/rtsp-media.h
+++ b/subprojects/gst-rtsp-server/gst/rtsp-server/rtsp-media.h
@@ -164,7 +164,9 @@ struct _GstRTSPMediaClass {
   /* vmethods */
   gboolean        (*handle_message)  (GstRTSPMedia *media, GstMessage *message);
   gboolean        (*prepare)         (GstRTSPMedia *media, GstRTSPThread *thread);
+#ifdef TIZEN_FEATURE_RTSP_SERVER_MODIFICATION
   gboolean        (*start_preroll)   (GstRTSPMedia *media);
+#endif
   gboolean        (*unprepare)       (GstRTSPMedia *media);
   gboolean        (*suspend)         (GstRTSPMedia *media);
   gboolean        (*unsuspend)       (GstRTSPMedia *media);
@@ -175,7 +177,9 @@ struct _GstRTSPMediaClass {
   GstElement *    (*create_rtpbin)   (GstRTSPMedia *media);
   gboolean        (*setup_rtpbin)    (GstRTSPMedia *media, GstElement *rtpbin);
   gboolean        (*setup_sdp)       (GstRTSPMedia *media, GstSDPMessage *sdp, GstSDPInfo *info);
+#ifdef TIZEN_FEATURE_RTSP_SERVER_MODIFICATION
   gboolean        (*start_prepare)   (GstRTSPMedia *media);
+#endif
 
   /* signals */
   void            (*new_stream)      (GstRTSPMedia *media, GstRTSPStream * stream);
@@ -189,8 +193,11 @@ struct _GstRTSPMediaClass {
 
   gboolean        (*handle_sdp)      (GstRTSPMedia *media, GstSDPMessage *sdp);
 
+#ifdef TIZEN_FEATURE_RTSP_SERVER_MODIFICATION
   void            (*preparing)       (GstRTSPMedia *media, GstRTSPStream * stream, guint idx);
   void            (*unpreparing)     (GstRTSPMedia *media, GstRTSPStream * stream, guint idx);
+#endif
+
   /*< private >*/
   gpointer         _gst_reserved[GST_PADDING_LARGE-1];
 };
@@ -429,6 +436,7 @@ GST_RTSP_SERVER_API
 void                  gst_rtsp_media_set_pipeline_state (GstRTSPMedia * media,
                                                          GstState state);
 
+#ifdef TIZEN_FEATURE_RTSP_SERVER_MODIFICATION
 GST_RTSP_SERVER_API
 GstStateChangeReturn  gst_rtsp_media_set_target_state (GstRTSPMedia * media, GstState state, gboolean do_state);
 
@@ -440,6 +448,7 @@ GstElement *          gst_rtsp_media_get_pipeline     (GstRTSPMedia * media);
 
 GST_RTSP_SERVER_API
 GstElement *          gst_rtsp_media_get_rtpbin       (GstRTSPMedia * media);
+#endif
 
 GST_RTSP_SERVER_API
 gboolean              gst_rtsp_media_complete_pipeline (GstRTSPMedia * media, GPtrArray * transports);

diff --git a/subprojects/gst-rtsp-server/gst/rtsp-server/rtsp-server-object.h b/subprojects/gst-rtsp-server/gst/rtsp-server/rtsp-server-object.h
index 44f9055..2e31b30 100644 (file)
--- a/subprojects/gst-rtsp-server/gst/rtsp-server/rtsp-server-object.h
+++ b/subprojects/gst-rtsp-server/gst/rtsp-server/rtsp-server-object.h
@@ -85,7 +85,9 @@ struct _GstRTSPServerClass {
   GObjectClass  parent_class;
 
   GstRTSPClient * (*create_client)      (GstRTSPServer *server);
+#ifdef TIZEN_FEATURE_RTSP_SERVER_MODIFICATION
   GSocket * (*create_socket)      (GstRTSPServer * server,  GCancellable * cancellable, GError ** error);
+#endif
 
   /* signals */
   void            (*client_connected)   (GstRTSPServer *server, GstRTSPClient *client);

diff --git a/subprojects/gst-rtsp-server/gst/rtsp-server/rtsp-server.c b/subprojects/gst-rtsp-server/gst/rtsp-server/rtsp-server.c
index af317ac..7086cc2 100644 (file)
--- a/subprojects/gst-rtsp-server/gst/rtsp-server/rtsp-server.c
+++ b/subprojects/gst-rtsp-server/gst/rtsp-server/rtsp-server.c
@@ -246,7 +246,9 @@ gst_rtsp_server_class_init (GstRTSPServerClass * klass)
       NULL, NULL, NULL, G_TYPE_NONE, 1, GST_TYPE_RTSP_CLIENT);
 
   klass->create_client = default_create_client;
+#ifdef TIZEN_FEATURE_RTSP_SERVER_MODIFICATION
   klass->create_socket = gst_rtsp_server_create_socket;
+#endif
 
   GST_DEBUG_CATEGORY_INIT (rtsp_server_debug, "rtspserver", 0, "GstRTSPServer");
 }
@@ -1340,17 +1342,25 @@ gst_rtsp_server_create_source (GstRTSPServer * server,
     GCancellable * cancellable, GError ** error)
 {
   GstRTSPServerPrivate *priv;
+#ifdef TIZEN_FEATURE_RTSP_SERVER_MODIFICATION
   GstRTSPServerClass *klass;
   GSocket *socket = NULL, *old;
+#else
+  GSocket *socket, *old;
+#endif
   GSource *source;
 
   g_return_val_if_fail (GST_IS_RTSP_SERVER (server), NULL);
 
   priv = server->priv;
+#ifdef TIZEN_FEATURE_RTSP_SERVER_MODIFICATION
   klass = GST_RTSP_SERVER_GET_CLASS (server);
 
   if (klass->create_socket)
     socket = klass->create_socket (server, NULL, error);
+#else
+  socket = gst_rtsp_server_create_socket (server, NULL, error);
+#endif
 
   if (socket == NULL)
     goto no_socket;

diff --git a/subprojects/gst-rtsp-server/gst/rtsp-server/rtsp-stream.c b/subprojects/gst-rtsp-server/gst/rtsp-server/rtsp-stream.c
index 355bb03..e5d0a00 100644 (file)
--- a/subprojects/gst-rtsp-server/gst/rtsp-server/rtsp-stream.c
+++ b/subprojects/gst-rtsp-server/gst/rtsp-server/rtsp-stream.c
@@ -255,7 +255,9 @@ enum
   SIGNAL_NEW_RTP_ENCODER,
   SIGNAL_NEW_RTCP_ENCODER,
   SIGNAL_NEW_RTP_RTCP_DECODER,
+#ifdef TIZEN_FEATURE_RTSP_SERVER_MODIFICATION
   SIGNAL_RTCP_STATS,
+#endif
   SIGNAL_LAST
 };
 
@@ -317,10 +319,12 @@ gst_rtsp_stream_class_init (GstRTSPStreamClass * klass)
       g_signal_new ("new-rtp-rtcp-decoder", G_TYPE_FROM_CLASS (klass),
       G_SIGNAL_RUN_LAST, 0, NULL, NULL, NULL, G_TYPE_NONE, 1, GST_TYPE_ELEMENT);
 
+#ifdef TIZEN_FEATURE_RTSP_SERVER_MODIFICATION
   gst_rtsp_stream_signals[SIGNAL_RTCP_STATS] =
       g_signal_new ("rtcp-statistics", G_TYPE_FROM_CLASS (klass),
       G_SIGNAL_RUN_LAST, 0, NULL, NULL, g_cclosure_marshal_generic,
       G_TYPE_NONE, 1, GST_TYPE_STRUCTURE);
+#endif
 
   GST_DEBUG_CATEGORY_INIT (rtsp_stream_debug, "rtspstream", 0, "GstRTSPStream");
 
@@ -2432,8 +2436,10 @@ check_transport (GObject * source, GstRTSPStream * stream)
 
       dump_structure (stats);
 
+#ifdef TIZEN_FEATURE_RTSP_SERVER_MODIFICATION
       g_signal_emit (stream, gst_rtsp_stream_signals[SIGNAL_RTCP_STATS], 0,
           stats);
+#endif
 
       rtcp_from = gst_structure_get_string (stats, "rtcp-from");
       if ((trans = find_transport (stream, rtcp_from))) {
@@ -2484,8 +2490,10 @@ on_ssrc_active (GObject * session, GObject * source, GstRTSPStream * stream)
     GstStructure *stats;
     g_object_get (source, "stats", &stats, NULL);
     if (stats) {
+#ifdef TIZEN_FEATURE_RTSP_SERVER_MODIFICATION
       g_signal_emit (stream, gst_rtsp_stream_signals[SIGNAL_RTCP_STATS], 0,
           stats);
+#endif
 
       dump_structure (stats);
       gst_structure_free (stats);
@@ -5223,6 +5231,7 @@ gst_rtsp_stream_get_current_seqnum (GstRTSPStream * stream)
   return seqnum;
 }
 
+#ifdef TIZEN_FEATURE_RTSP_SERVER_MODIFICATION
 guint64
 gst_rtsp_stream_get_udp_sent_bytes (GstRTSPStream * stream)
 {
@@ -5237,6 +5246,7 @@ gst_rtsp_stream_get_udp_sent_bytes (GstRTSPStream * stream)
 
   return bytes;
 }
+#endif
 
 /**
  * gst_rtsp_stream_transport_filter:

diff --git a/subprojects/gst-rtsp-server/gst/rtsp-server/rtsp-stream.h b/subprojects/gst-rtsp-server/gst/rtsp-server/rtsp-stream.h
index db9d5cf..919e11a 100644 (file)
--- a/subprojects/gst-rtsp-server/gst/rtsp-server/rtsp-stream.h
+++ b/subprojects/gst-rtsp-server/gst/rtsp-server/rtsp-stream.h
@@ -269,8 +269,10 @@ void              gst_rtsp_stream_set_seqnum_offset          (GstRTSPStream *str
 GST_RTSP_SERVER_API
 guint16           gst_rtsp_stream_get_current_seqnum          (GstRTSPStream *stream);
 
+#ifdef TIZEN_FEATURE_RTSP_SERVER_MODIFICATION
 GST_RTSP_SERVER_API
 guint64           gst_rtsp_stream_get_udp_sent_bytes          (GstRTSPStream *stream);
+#endif
 
 GST_RTSP_SERVER_API
 void              gst_rtsp_stream_set_retransmission_time     (GstRTSPStream *stream, GstClockTime time);

diff --git a/subprojects/gst-rtsp-server/meson.build b/subprojects/gst-rtsp-server/meson.build
index ae6b59e..4b1f80f 100644 (file)
--- a/subprojects/gst-rtsp-server/meson.build
+++ b/subprojects/gst-rtsp-server/meson.build
@@ -155,6 +155,7 @@ if host_machine.system() != 'windows'
 endif
 
 # TIZEN_BUILD_OPTION
+cdata.set('TIZEN_FEATURE_RTSP_SERVER_MODIFICATION', true)
 cdata.set('TIZEN_FEATURE_BUG_FIX', true)
 
 if get_option('tv-profile')