onvif: Implement and test the Streaming Specification

https://www.onvif.org/specs/stream/ONVIF-Streaming-Spec.pdf

gst/rtsp-server/rtsp-client.c

@@ -1833,6 +1833,8 @@ setup_play_mode (GstRTSPClient * client, GstRTSPContext * ctx,
   GstSeekFlags flags = GST_SEEK_FLAG_NONE;
   GstRTSPClientClass *klass = GST_RTSP_CLIENT_GET_CLASS (client);
   GstRTSPStatusCode rtsp_status_code;
+  GstClockTime trickmode_interval = 0;
+  gboolean enable_rate_control = TRUE;
 
   /* parse the range header if we have one */
   res = gst_rtsp_message_get_header (ctx->request, GST_RTSP_HDR_RANGE, &str, 0);
@@ -1876,13 +1878,17 @@ setup_play_mode (GstRTSPClient * client, GstRTSPContext * ctx,
   /* give the application a chance to tweak range, flags, or rate */
   if (klass->adjust_play_mode != NULL) {
     rtsp_status_code =
-        klass->adjust_play_mode (client, ctx, &range, &flags, &rate);
+        klass->adjust_play_mode (client, ctx, &range, &flags, &rate,
+        &trickmode_interval, &enable_rate_control);
     if (rtsp_status_code != GST_RTSP_STS_OK)
       goto adjust_play_mode_failed;
   }
 
+  gst_rtsp_media_set_rate_control (ctx->media, enable_rate_control);
+
   /* now do the seek with the seek options */
-  (void) gst_rtsp_media_seek_full_with_rate (ctx->media, range, flags, rate);
+  gst_rtsp_media_seek_trickmode (ctx->media, range, flags, rate,
+      trickmode_interval);
   if (range != NULL)
     gst_rtsp_range_free (range);
 
@@ -2031,6 +2037,12 @@ handle_play_request (GstRTSPClient * client, GstRTSPContext * ctx)
           g_strdup_printf ("%1.3f", rate));
   }
 
+  if (klass->adjust_play_response) {
+    code = klass->adjust_play_response (client, ctx);
+    if (code != GST_RTSP_STS_OK)
+      goto adjust_play_response_failed;
+  }
+
   send_message (client, ctx, ctx->response, FALSE);
 
   /* start playing after sending the response */
@@ -2114,6 +2126,12 @@ get_rates_error:
     send_generic_response (client, GST_RTSP_STS_INTERNAL_SERVER_ERROR, ctx);
     return FALSE;
   }
+adjust_play_response_failed:
+  {
+    GST_ERROR ("client %p: failed to adjust play response", client);
+    send_generic_response (client, code, ctx);
+    return FALSE;
+  }
 }
 
 static void

gst/rtsp-server/rtsp-client.h

@@ -109,7 +109,10 @@ struct _GstRTSPClient {
  *    RTSP response(ctx->response) via a call to gst_rtsp_message_init_response()
  * @make_path_from_uri: called to create path from uri.
  * @adjust_play_mode: called to give the application the possibility to adjust
- *    the range, seek flags, and/or rate. Since 1.18
+ *    the range, seek flags, rate and rate-control. Since 1.18
+ * @adjust_play_response: called to give the implementation the possibility to
+ *    adjust the response to a play request, for example if extra headers were
+ *    parsed when #GstRTSPClientClass.adjust_play_mode was called. Since 1.18
  * @tunnel_http_response: called when a response to the GET request is about to
  *   be sent for a tunneled connection. The response can be modified. Since: 1.4
  *
@@ -132,7 +135,12 @@ struct _GstRTSPClientClass {
                                          GstRTSPContext * context,
                                          GstRTSPTimeRange ** range,
                                          GstSeekFlags * flags,
-                                         gdouble * rate);
+                                         gdouble * rate,
+                                         GstClockTime * trickmode_interval,
+                                         gboolean * enable_rate_control);
+  GstRTSPStatusCode (*adjust_play_response) (GstRTSPClient * client,
+                                            GstRTSPContext * context);
+
   /* signals */
   void     (*closed)                  (GstRTSPClient *client);
   void     (*new_session)             (GstRTSPClient *client, GstRTSPSession *session);
@@ -169,7 +177,7 @@ struct _GstRTSPClientClass {
   GstRTSPStatusCode (*pre_record_request)        (GstRTSPClient *client, GstRTSPContext *ctx);
 
   /*< private >*/
-  gpointer _gst_reserved[GST_PADDING_LARGE-17];
+  gpointer _gst_reserved[GST_PADDING_LARGE-18];
 };
 
 GST_RTSP_SERVER_API

gst/rtsp-server/rtsp-media.c

@@ -51,7 +51,7 @@
  *
  * The state of the media can be controlled with gst_rtsp_media_set_state ().
  * Seeking can be done with gst_rtsp_media_seek(), or gst_rtsp_media_seek_full()
- * or gst_rtsp_media_seek_full_with_rate() for finer control of the seek.
+ * or gst_rtsp_media_seek_trickmode() for finer control of the seek.
  *
  * With gst_rtsp_media_unprepare() the pipeline is stopped and shut down. When
  * gst_rtsp_media_set_eos_shutdown() an EOS will be sent to the pipeline to
@@ -146,6 +146,7 @@ struct _GstRTSPMediaPrivate
   gboolean do_retransmission;   /* protected by lock */
   guint latency;                /* protected by lock */
   GstClock *clock;              /* protected by lock */
+  gboolean do_rate_control;     /* protected by lock */
   GstRTSPPublishClockMode publish_clock_mode;
 
   /* Dynamic element handling */
@@ -167,6 +168,7 @@ struct _GstRTSPMediaPrivate
 #define DEFAULT_STOP_ON_DISCONNECT TRUE
 #define DEFAULT_MAX_MCAST_TTL   255
 #define DEFAULT_BIND_MCAST_ADDRESS FALSE
+#define DEFAULT_DO_RATE_CONTROL TRUE
 
 #define DEFAULT_DO_RETRANSMISSION FALSE
 
@@ -471,6 +473,7 @@ gst_rtsp_media_init (GstRTSPMedia * media)
   priv->do_retransmission = DEFAULT_DO_RETRANSMISSION;
   priv->max_mcast_ttl = DEFAULT_MAX_MCAST_TTL;
   priv->bind_mcast_address = DEFAULT_BIND_MCAST_ADDRESS;
+  priv->do_rate_control = DEFAULT_DO_RATE_CONTROL;
 }
 
 static void
@@ -2293,6 +2296,7 @@ gst_rtsp_media_create_stream (GstRTSPMedia * media, GstElement * payloader,
   gst_rtsp_stream_set_retransmission_time (stream, priv->rtx_time);
   gst_rtsp_stream_set_buffer_size (stream, priv->buffer_size);
   gst_rtsp_stream_set_publish_clock_mode (stream, priv->publish_clock_mode);
+  gst_rtsp_stream_set_rate_control (stream, priv->do_rate_control);
 
   g_ptr_array_add (priv->streams, stream);
 
@@ -2662,13 +2666,15 @@ gst_rtsp_media_get_status (GstRTSPMedia * media)
 }
 
 /**
- * gst_rtsp_media_seek_full_with_rate:
+ * gst_rtsp_media_seek_trickmode:
  * @media: a #GstRTSPMedia
  * @range: (transfer none): a #GstRTSPTimeRange
  * @flags: The minimal set of #GstSeekFlags to use
  * @rate: the rate to use in the seek
+ * @trickmode_interval: The trickmode interval to use for KEY_UNITS trick mode
  *
- * Seek the pipeline of @media to @range with the given @flags and @rate.
+ * Seek the pipeline of @media to @range with the given @flags and @rate,
+ * and @trickmode_interval.
  * @media must be prepared with gst_rtsp_media_prepare().
  * In order to perform the seek operation, the pipeline must contain all
  * needed transport parts (transport sinks).
@@ -2678,8 +2684,9 @@ gst_rtsp_media_get_status (GstRTSPMedia * media)
  * Since: 1.18
  */
 gboolean
-gst_rtsp_media_seek_full_with_rate (GstRTSPMedia * media,
-    GstRTSPTimeRange * range, GstSeekFlags flags, gdouble rate)
+gst_rtsp_media_seek_trickmode (GstRTSPMedia * media,
+    GstRTSPTimeRange * range, GstSeekFlags flags, gdouble rate,
+    GstClockTime trickmode_interval)
 {
   GstRTSPMediaClass *klass;
   GstRTSPMediaPrivate *priv;
@@ -2789,6 +2796,8 @@ gst_rtsp_media_seek_full_with_rate (GstRTSPMedia * media,
       GST_DEBUG ("no position change, no flags set by caller, so not seeking");
       res = TRUE;
     } else {
+      GstEvent *seek_event;
+
       gst_rtsp_media_set_status (media, GST_RTSP_MEDIA_STATUS_PREPARING);
 
       if (rate < 0.0) {
@@ -2801,8 +2810,12 @@ gst_rtsp_media_seek_full_with_rate (GstRTSPMedia * media,
         stop_type = temp_type;
       }
 
-      res = gst_element_seek (priv->pipeline, rate, GST_FORMAT_TIME,
-          flags, start_type, start, stop_type, stop);
+      seek_event = gst_event_new_seek (rate, GST_FORMAT_TIME, flags, start_type,
+          start, stop_type, stop);
+
+      gst_event_set_seek_trickmode_interval (seek_event, trickmode_interval);
+
+      res = gst_element_send_event (priv->pipeline, seek_event);
 
       /* and block for the seek to complete */
       GST_INFO ("done seeking %d", res);
@@ -2880,7 +2893,7 @@ gboolean
 gst_rtsp_media_seek_full (GstRTSPMedia * media, GstRTSPTimeRange * range,
     GstSeekFlags flags)
 {
-  return gst_rtsp_media_seek_full_with_rate (media, range, flags, 1.0);
+  return gst_rtsp_media_seek_trickmode (media, range, flags, 1.0, 0);
 }
 
 /**
@@ -2896,8 +2909,8 @@ gst_rtsp_media_seek_full (GstRTSPMedia * media, GstRTSPTimeRange * range,
 gboolean
 gst_rtsp_media_seek (GstRTSPMedia * media, GstRTSPTimeRange * range)
 {
-  return gst_rtsp_media_seek_full_with_rate (media, range, GST_SEEK_FLAG_NONE,
-      1.0);
+  return gst_rtsp_media_seek_trickmode (media, range, GST_SEEK_FLAG_NONE,
+      1.0, 0);
 }
 
 static void
@@ -4728,3 +4741,59 @@ gst_rtsp_media_is_receive_only (GstRTSPMedia * media)
 
   return receive_only;
 }
+
+/**
+ * gst_rtsp_media_set_rate_control:
+ *
+ * Define whether @media will follow the Rate-Control=no behaviour as specified
+ * in the ONVIF replay spec.
+ *
+ * Since: 1.18
+ */
+void
+gst_rtsp_media_set_rate_control (GstRTSPMedia * media, gboolean enabled)
+{
+  GstRTSPMediaPrivate *priv;
+  guint i;
+
+  g_return_if_fail (GST_IS_RTSP_MEDIA (media));
+
+  GST_LOG_OBJECT (media, "%s rate control", enabled ? "Enabling" : "Disabling");
+
+  priv = media->priv;
+
+  g_mutex_lock (&priv->lock);
+  priv->do_rate_control = enabled;
+  for (i = 0; i < priv->streams->len; i++) {
+    GstRTSPStream *stream = g_ptr_array_index (priv->streams, i);
+
+    gst_rtsp_stream_set_rate_control (stream, enabled);
+
+  }
+  g_mutex_unlock (&priv->lock);
+}
+
+/**
+ * gst_rtsp_media_get_rate_control:
+ *
+ * Returns: whether @media will follow the Rate-Control=no behaviour as specified
+ * in the ONVIF replay spec.
+ *
+ * Since: 1.18
+ */
+gboolean
+gst_rtsp_media_get_rate_control (GstRTSPMedia * media)
+{
+  GstRTSPMediaPrivate *priv;
+  gboolean res;
+
+  g_return_val_if_fail (GST_IS_RTSP_MEDIA (media), FALSE);
+
+  priv = media->priv;
+
+  g_mutex_lock (&priv->lock);
+  res = priv->do_rate_control;
+  g_mutex_unlock (&priv->lock);
+
+  return res;
+}

gst/rtsp-server/rtsp-media.h

@@ -388,10 +388,11 @@ gboolean              gst_rtsp_media_seek_full        (GstRTSPMedia *media,
                                                        GstSeekFlags flags);
 
 GST_RTSP_SERVER_API
-gboolean              gst_rtsp_media_seek_full_with_rate (GstRTSPMedia *media,
-                                                          GstRTSPTimeRange *range,
-                                                          GstSeekFlags flags,
-                                                          gdouble rate);
+gboolean              gst_rtsp_media_seek_trickmode   (GstRTSPMedia *media,
+                                                       GstRTSPTimeRange *range,
+                                                       GstSeekFlags flags,
+                                                       gdouble rate,
+                                                       GstClockTime trickmode_interval);
 
 GST_RTSP_SERVER_API
 GstClockTimeDiff      gst_rtsp_media_seekable         (GstRTSPMedia *media);
@@ -420,6 +421,12 @@ gboolean              gst_rtsp_media_complete_pipeline (GstRTSPMedia * media, GP
 GST_RTSP_SERVER_API
 gboolean              gst_rtsp_media_is_receive_only (GstRTSPMedia * media);
 
+GST_RTSP_SERVER_API
+void                  gst_rtsp_media_set_rate_control (GstRTSPMedia * media, gboolean enabled);
+
+GST_RTSP_SERVER_API
+gboolean              gst_rtsp_media_get_rate_control (GstRTSPMedia * media);
+
 #ifdef G_DEFINE_AUTOPTR_CLEANUP_FUNC
 G_DEFINE_AUTOPTR_CLEANUP_FUNC(GstRTSPMedia, gst_object_unref)
 #endif

gst/rtsp-server/rtsp-onvif-client.c

@@ -37,11 +37,14 @@ gst_rtsp_onvif_client_check_requirements (GstRTSPClient * client,
   GstRTSPMediaFactory *factory = NULL;
   gchar *path = NULL;
   gboolean has_backchannel = FALSE;
+  gboolean has_replay = FALSE;
   GString *unsupported = g_string_new ("");
 
   while (*requirements) {
     if (strcmp (*requirements, GST_RTSP_ONVIF_BACKCHANNEL_REQUIREMENT) == 0) {
       has_backchannel = TRUE;
+    } else if (strcmp (*requirements, GST_RTSP_ONVIF_REPLAY_REQUIREMENT) == 0) {
+      has_replay = TRUE;
     } else {
       if (unsupported->len)
         g_string_append (unsupported, ", ");
@@ -75,6 +78,22 @@ gst_rtsp_onvif_client_check_requirements (GstRTSPClient * client,
     }
   }
 
+  if (has_replay && !GST_IS_RTSP_ONVIF_MEDIA_FACTORY (factory)) {
+    if (unsupported->len)
+      g_string_append (unsupported, ", ");
+    g_string_append (unsupported, GST_RTSP_ONVIF_REPLAY_REQUIREMENT);
+  } else if (has_replay) {
+    GstRTSPOnvifMediaFactory *onvif_factory =
+        GST_RTSP_ONVIF_MEDIA_FACTORY (factory);
+
+    if (!gst_rtsp_onvif_media_factory_has_replay_support (onvif_factory)) {
+      if (unsupported->len)
+        g_string_append (unsupported, ", ");
+      g_string_append (unsupported, GST_RTSP_ONVIF_REPLAY_REQUIREMENT);
+    }
+  }
+
+
 out:
   if (path)
     g_free (path);
@@ -86,15 +105,115 @@ out:
   return g_string_free (unsupported, FALSE);
 }
 
+static GstRTSPStatusCode
+gst_rtsp_onvif_client_adjust_play_mode (GstRTSPClient * client,
+    GstRTSPContext * ctx, GstRTSPTimeRange ** range, GstSeekFlags * flags,
+    gdouble * rate, GstClockTime * trickmode_interval,
+    gboolean * enable_rate_control)
+{
+  GstRTSPStatusCode ret = GST_RTSP_STS_BAD_REQUEST;
+  gchar **split = NULL;
+  gchar *str;
+
+  if (gst_rtsp_message_get_header (ctx->request, GST_RTSP_HDR_FRAMES,
+          &str, 0) == GST_RTSP_OK) {
+
+    split = g_strsplit (str, "/", 2);
+
+    if (!g_strcmp0 (split[0], "intra")) {
+      if (split[1]) {
+        guint64 interval;
+        gchar *end;
+
+        interval = g_ascii_strtoull (split[1], &end, 10);
+
+        if (!end || *end != '\0') {
+          GST_ERROR ("Unexpected interval value %s", split[1]);
+          goto done;
+        }
+
+        *trickmode_interval = interval * GST_MSECOND;
+      }
+      *flags |= GST_SEEK_FLAG_TRICKMODE_KEY_UNITS;
+    } else if (!g_strcmp0 (split[0], "predicted")) {
+      if (split[1]) {
+        GST_ERROR ("Predicted frames mode does not allow an interval (%s)",
+            str);
+        goto done;
+      }
+      *flags |= GST_SEEK_FLAG_TRICKMODE_FORWARD_PREDICTED;
+    } else {
+      GST_ERROR ("Invalid frames mode (%s)", str);
+      goto done;
+    }
+  }
+
+  if (gst_rtsp_message_get_header (ctx->request, GST_RTSP_HDR_RATE_CONTROL,
+          &str, 0) == GST_RTSP_OK) {
+    if (!g_strcmp0 (str, "no")) {
+      *enable_rate_control = FALSE;
+    } else if (!g_strcmp0 (str, "yes")) {
+      *enable_rate_control = TRUE;
+    } else {
+      GST_ERROR ("Invalid rate control header: %s", str);
+      goto done;
+    }
+  }
+
+  ret = GST_RTSP_STS_OK;
+
+done:
+  if (split)
+    g_strfreev (split);
+  return ret;
+}
+
+static GstRTSPStatusCode
+gst_rtsp_onvif_client_adjust_play_response (GstRTSPClient * client,
+    GstRTSPContext * ctx)
+{
+  GstRTSPStatusCode ret = GST_RTSP_STS_OK;
+  gchar *str;
+
+  if (gst_rtsp_message_get_header (ctx->request, GST_RTSP_HDR_RATE_CONTROL,
+          &str, 0) == GST_RTSP_OK) {
+    gst_rtsp_message_add_header (ctx->response, GST_RTSP_HDR_RATE_CONTROL,
+        gst_rtsp_media_get_rate_control (ctx->media) ? "yes" : "no");
+  }
+
+  return ret;
+}
+
 static void
 gst_rtsp_onvif_client_class_init (GstRTSPOnvifClientClass * klass)
 {
   GstRTSPClientClass *client_klass = (GstRTSPClientClass *) klass;
 
   client_klass->check_requirements = gst_rtsp_onvif_client_check_requirements;
+  client_klass->adjust_play_mode = gst_rtsp_onvif_client_adjust_play_mode;
+  client_klass->adjust_play_response =
+      gst_rtsp_onvif_client_adjust_play_response;
 }
 
 static void
 gst_rtsp_onvif_client_init (GstRTSPOnvifClient * client)
 {
 }
+
+/**
+ * gst_rtsp_onvif_client_new:
+ *
+ * Create a new #GstRTSPOnvifClient instance.
+ *
+ * Returns: (transfer full): a new #GstRTSPOnvifClient
+ * Since: 1.18
+ */
+GstRTSPClient *
+gst_rtsp_onvif_client_new (void)
+{
+  GstRTSPClient *result;
+
+  result = g_object_new (GST_TYPE_RTSP_ONVIF_CLIENT, NULL);
+
+  return result;
+}

gst/rtsp-server/rtsp-onvif-client.h

@@ -59,4 +59,7 @@ struct GstRTSPOnvifClient
 GST_RTSP_SERVER_API
 GType gst_rtsp_onvif_client_get_type (void);
 
+GST_RTSP_SERVER_API
+GstRTSPClient * gst_rtsp_onvif_client_new (void);
+
 #endif /* __GST_RTSP_ONVIF_CLIENT_H__ */

gst/rtsp-server/rtsp-onvif-media-factory.c

@@ -50,6 +50,7 @@ struct GstRTSPOnvifMediaFactoryPrivate
   GMutex lock;
   gchar *backchannel_launch;
   guint backchannel_bandwidth;
+  gboolean has_replay_support;
 };
 
 G_DEFINE_TYPE_WITH_PRIVATE (GstRTSPOnvifMediaFactory,
@@ -447,6 +448,42 @@ gst_rtsp_onvif_media_factory_has_backchannel_support (GstRTSPOnvifMediaFactory *
   return FALSE;
 }
 
+/**
+ * gst_rtsp_onvif_media_factory_has_replay_support:
+ *
+ * Returns: %TRUE if ONVIF replay is supported by the media factory.
+ *
+ * Since: 1.18
+ */
+gboolean
+gst_rtsp_onvif_media_factory_has_replay_support (GstRTSPOnvifMediaFactory *
+    factory)
+{
+  gboolean has_replay_support;
+
+  g_mutex_lock (&factory->priv->lock);
+  has_replay_support = factory->priv->has_replay_support;
+  g_mutex_unlock (&factory->priv->lock);
+
+  return has_replay_support;
+}
+
+/**
+ * gst_rtsp_onvif_media_factory_set_replay_support:
+ *
+ * Set to %TRUE if ONVIF replay is supported by the media factory.
+ *
+ * Since: 1.18
+ */
+void
+gst_rtsp_onvif_media_factory_set_replay_support (GstRTSPOnvifMediaFactory *
+    factory, gboolean has_replay_support)
+{
+  g_mutex_lock (&factory->priv->lock);
+  factory->priv->has_replay_support = has_replay_support;
+  g_mutex_unlock (&factory->priv->lock);
+}
+
 /**
  * gst_rtsp_onvif_media_factory_set_backchannel_bandwidth:
  * @factory: a #GstRTSPMediaFactory

gst/rtsp-server/rtsp-onvif-media-factory.h

@@ -74,6 +74,12 @@ gchar * gst_rtsp_onvif_media_factory_get_backchannel_launch (GstRTSPOnvifMediaFa
 GST_RTSP_SERVER_API
 gboolean gst_rtsp_onvif_media_factory_has_backchannel_support (GstRTSPOnvifMediaFactory * factory);
 
+GST_RTSP_SERVER_API
+gboolean gst_rtsp_onvif_media_factory_has_replay_support (GstRTSPOnvifMediaFactory * factory);
+
+GST_RTSP_SERVER_API
+void gst_rtsp_onvif_media_factory_set_replay_support (GstRTSPOnvifMediaFactory * factory, gboolean has_replay_support);
+
 GST_RTSP_SERVER_API
 void gst_rtsp_onvif_media_factory_set_backchannel_bandwidth (GstRTSPOnvifMediaFactory * factory, guint bandwidth);
 GST_RTSP_SERVER_API

gst/rtsp-server/rtsp-onvif-media.c

@@ -131,6 +131,16 @@ gst_rtsp_onvif_media_setup_sdp (GstRTSPMedia * media, GstSDPMessage * sdp,
           if (res) {
             GstSDPMedia *smedia =
                 &g_array_index (sdp->medias, GstSDPMedia, sdp->medias->len - 1);
+            gchar *x_onvif_track, *media_str;
+
+            media_str =
+                g_ascii_strup (gst_structure_get_string (s, "media"), -1);
+            x_onvif_track =
+                g_strdup_printf ("%s%03d", media_str, sdp->medias->len - 1);
+            gst_sdp_media_add_attribute (smedia, "x-onvif-track",
+                x_onvif_track);
+            g_free (x_onvif_track);
+            g_free (media_str);
 
             if (sinkpad) {
               GstRTSPOnvifMedia *onvif_media = GST_RTSP_ONVIF_MEDIA (media);

gst/rtsp-server/rtsp-onvif-server.h

@@ -62,6 +62,7 @@ GST_RTSP_SERVER_API
 GstRTSPServer *gst_rtsp_onvif_server_new (void);
 
 #define GST_RTSP_ONVIF_BACKCHANNEL_REQUIREMENT "www.onvif.org/ver20/backchannel"
+#define GST_RTSP_ONVIF_REPLAY_REQUIREMENT "onvif-replay"
 
 #include "rtsp-onvif-client.h"
 #include "rtsp-onvif-media-factory.h"

gst/rtsp-server/rtsp-stream.c

@@ -130,6 +130,9 @@ struct _GstRTSPStreamPrivate
   guint rtx_pt;
   GstClockTime rtx_time;
 
+  /* rate control */
+  gboolean do_rate_control;
+
   /* Forward Error Correction with RFC 5109 */
   GstElement *ulpfec_decoder;
   GstElement *ulpfec_encoder;
@@ -190,6 +193,7 @@ struct _GstRTSPStreamPrivate
                                         GST_RTSP_LOWER_TRANS_TCP
 #define DEFAULT_MAX_MCAST_TTL   255
 #define DEFAULT_BIND_MCAST_ADDRESS FALSE
+#define DEFAULT_DO_RATE_CONTROL TRUE
 
 enum
 {
@@ -291,6 +295,7 @@ gst_rtsp_stream_init (GstRTSPStream * stream)
   priv->publish_clock_mode = GST_RTSP_PUBLISH_CLOCK_MODE_CLOCK;
   priv->max_mcast_ttl = DEFAULT_MAX_MCAST_TTL;
   priv->bind_mcast_address = DEFAULT_BIND_MCAST_ADDRESS;
+  priv->do_rate_control = DEFAULT_DO_RATE_CONTROL;
 
   g_mutex_init (&priv->lock);
 
@@ -3378,6 +3383,11 @@ create_sender_part (GstRTSPStream * stream, const GstRTSPTransport * transport)
     return FALSE;
   }
 
+  if (g_object_class_find_property (G_OBJECT_GET_CLASS (priv->payloader),
+          "onvif-no-rate-control"))
+    g_object_set (priv->payloader, "onvif-no-rate-control",
+        !priv->do_rate_control, NULL);
+
   for (i = 0; i < 2; i++) {
     gboolean link_tee = FALSE;
     /* For the sender we create this bit of pipeline for both
@@ -3436,6 +3446,9 @@ create_sender_part (GstRTSPStream * stream, const GstRTSPTransport * transport)
       g_object_set (priv->appsink[i], "emit-signals", FALSE, "buffer-list",
           TRUE, "max-buffers", 1, NULL);
 
+      if (i == 0)
+        g_object_set (priv->appsink[i], "sync", priv->do_rate_control, NULL);
+
       /* we need to set sync and preroll to FALSE for the sink to avoid
        * deadlock. This is only needed for sink sending RTCP data. */
       if (i == 1)
@@ -3765,6 +3778,9 @@ gst_rtsp_stream_join_bin (GstRTSPStream * stream, GstBin * bin,
   g_signal_connect (priv->session, "on-sender-ssrc-active",
       (GCallback) on_sender_ssrc_active, stream);
 
+  g_object_set (priv->session, "disable-sr-timestamp", !priv->do_rate_control,
+      NULL);
+
   if (priv->srcpad) {
     /* be notified of caps changes */
     priv->caps_sig = g_signal_connect (priv->send_src[0], "notify::caps",
@@ -5919,3 +5935,53 @@ gst_rtsp_stream_get_ulpfec_percentage (GstRTSPStream * stream)
 
   return res;
 }
+
+/**
+ * gst_rtsp_stream_set_rate_control:
+ *
+ * Define whether @stream will follow the Rate-Control=no behaviour as specified
+ * in the ONVIF replay spec.
+ *
+ * Since: 1.18
+ */
+void
+gst_rtsp_stream_set_rate_control (GstRTSPStream * stream, gboolean enabled)
+{
+  GST_DEBUG_OBJECT (stream, "%s rate control",
+      enabled ? "Enabling" : "Disabling");
+
+  g_mutex_lock (&stream->priv->lock);
+  stream->priv->do_rate_control = enabled;
+  if (stream->priv->appsink[0])
+    g_object_set (stream->priv->appsink[0], "sync", enabled, NULL);
+  if (stream->priv->payloader
+      && g_object_class_find_property (G_OBJECT_GET_CLASS (stream->priv->
+              payloader), "onvif-no-rate-control"))
+    g_object_set (stream->priv->payloader, "onvif-no-rate-control", !enabled,
+        NULL);
+  if (stream->priv->session) {
+    g_object_set (stream->priv->session, "disable-sr-timestamp", !enabled,
+        NULL);
+  }
+  g_mutex_unlock (&stream->priv->lock);
+}
+
+/**
+ * gst_rtsp_stream_get_rate_control:
+ *
+ * Returns: whether @stream will follow the Rate-Control=no behaviour as specified
+ * in the ONVIF replay spec.
+ *
+ * Since: 1.18
+ */
+gboolean
+gst_rtsp_stream_get_rate_control (GstRTSPStream * stream)
+{
+  gboolean ret;
+
+  g_mutex_lock (&stream->priv->lock);
+  ret = stream->priv->do_rate_control;
+  g_mutex_unlock (&stream->priv->lock);
+
+  return ret;
+}

gst/rtsp-server/rtsp-stream.h

@@ -359,6 +359,12 @@ void               gst_rtsp_stream_set_ulpfec_percentage (GstRTSPStream *stream,
 GST_RTSP_SERVER_API
 guint              gst_rtsp_stream_get_ulpfec_percentage (GstRTSPStream *stream);
 
+GST_RTSP_SERVER_API
+void               gst_rtsp_stream_set_rate_control (GstRTSPStream * stream, gboolean enabled);
+
+GST_RTSP_SERVER_API
+gboolean           gst_rtsp_stream_get_rate_control (GstRTSPStream * stream);
+
 /**
  * GstRTSPStreamTransportFilterFunc:
  * @stream: a #GstRTSPStream object

tests/check/gst/media.c

@@ -109,16 +109,16 @@ GST_START_TEST (test_media_seek)
   fail_unless (applied_rate == 1.0);
 
   /* seeking with rate set to 1.5 should result in rate == 1.5 */
-  fail_unless (gst_rtsp_media_seek_full_with_rate (media, range,
-          GST_SEEK_FLAG_NONE, 1.5));
+  fail_unless (gst_rtsp_media_seek_trickmode (media, range,
+          GST_SEEK_FLAG_NONE, 1.5, 0));
   fail_unless (gst_rtsp_media_get_rates (media, &rate, &applied_rate));
   fail_unless (rate == 1.5);
   fail_unless (applied_rate == 1.0);
 
   /* seeking with rate set to -2.0 should result in rate == -2.0 */
   fail_unless (gst_rtsp_range_parse ("npt=10-5", &range) == GST_RTSP_OK);
-  fail_unless (gst_rtsp_media_seek_full_with_rate (media, range,
-          GST_SEEK_FLAG_NONE, -2.0));
+  fail_unless (gst_rtsp_media_seek_trickmode (media, range,
+          GST_SEEK_FLAG_NONE, -2.0, 0));
   fail_unless (gst_rtsp_media_get_rates (media, &rate, &applied_rate));
   fail_unless (rate == -2.0);
   fail_unless (applied_rate == 1.0);

tests/check/meson.build

@@ -28,6 +28,7 @@ rtsp_server_tests = [
   'gst/stream',
   'gst/threadpool',
   'gst/token',
+  'gst/onvif',
 ]
 
 if not get_option('rtspclientsink').disabled()