X-Git-Url: http://review.tizen.org/git/?a=blobdiff_plain;f=gst%2Frtsp%2Fgstrtspsrc.c;h=05f532d54bcf6d73d8267c92924eaaed621ea174;hb=a858bf46db860965089611674bfc43289830b646;hp=9293baa1f88953eb78ef0e636149db5d76daec9a;hpb=bca1463a666bfc149a5fa2cf2acaa480f15b906a;p=platform%2Fupstream%2Fgstreamer.git

diff --git a/gst/rtsp/gstrtspsrc.c b/gst/rtsp/gstrtspsrc.c
index 9293baa..05f532d 100644
--- a/gst/rtsp/gstrtspsrc.c
+++ b/gst/rtsp/gstrtspsrc.c
@@ -14,8 +14,8 @@
  *
  * You should have received a copy of the GNU Library General Public
  * License along with this library; if not, write to the
- * Free Software Foundation, Inc., 59 Temple Place - Suite 330,
- * Boston, MA 02111-1307, USA.
+ * Free Software Foundation, Inc., 51 Franklin St, Fifth Floor,
+ * Boston, MA 02110-1301, USA.
  */
 /*
  * Unless otherwise indicated, Source Code is licensed under MIT license.
@@ -69,7 +69,7 @@
  * <refsect2>
  * <title>Example launch line</title>
  * |[
- * gst-launch rtspsrc location=rtsp://some.server/url ! fakesink
+ * gst-launch-1.0 rtspsrc location=rtsp://some.server/url ! fakesink
  * ]| Establish a connection to an RTSP server and send the raw RTP packets to a
  * fakesink.
  * </refsect2>
@@ -86,7 +86,6 @@
 #endif /* HAVE_UNISTD_H */
 #include <stdlib.h>
 #include <string.h>
-#include <locale.h>
 #include <stdio.h>
 #include <stdarg.h>
 
@@ -167,6 +166,7 @@ gst_rtsp_src_buffer_mode_get_type (void)
 #define DEFAULT_UDP_BUFFER_SIZE  0x80000
 #define DEFAULT_TCP_TIMEOUT      20000000
 #define DEFAULT_LATENCY_MS       2000
+#define DEFAULT_DROP_ON_LATENCY  FALSE
 #define DEFAULT_CONNECTION_SPEED 0
 #define DEFAULT_NAT_METHOD       GST_RTSP_NAT_DUMMY
 #define DEFAULT_DO_RTCP          TRUE
@@ -178,6 +178,8 @@ gst_rtsp_src_buffer_mode_get_type (void)
 #define DEFAULT_BUFFER_MODE      BUFFER_MODE_AUTO
 #define DEFAULT_PORT_RANGE       NULL
 #define DEFAULT_SHORT_HEADER     FALSE
+#define DEFAULT_PROBATION        2
+#define DEFAULT_UDP_RECONNECT    TRUE
 
 enum
 {
@@ -189,6 +191,7 @@ enum
   PROP_TIMEOUT,
   PROP_TCP_TIMEOUT,
   PROP_LATENCY,
+  PROP_DROP_ON_LATENCY,
   PROP_CONNECTION_SPEED,
   PROP_NAT_METHOD,
   PROP_DO_RTCP,
@@ -201,6 +204,8 @@ enum
   PROP_PORT_RANGE,
   PROP_UDP_BUFFER_SIZE,
   PROP_SHORT_HEADER,
+  PROP_PROBATION,
+  PROP_UDP_RECONNECT,
   PROP_LAST
 };
 
@@ -248,36 +253,35 @@ static void gst_rtspsrc_handle_message (GstBin * bin, GstMessage * message);
 static gboolean gst_rtspsrc_setup_auth (GstRTSPSrc * src,
     GstRTSPMessage * response);
 
-static void gst_rtspsrc_loop_send_cmd (GstRTSPSrc * src, gint cmd);
+static void gst_rtspsrc_loop_send_cmd (GstRTSPSrc * src, gint cmd, gint mask);
 static GstRTSPResult gst_rtspsrc_send_cb (GstRTSPExtension * ext,
     GstRTSPMessage * request, GstRTSPMessage * response, GstRTSPSrc * src);
 
 static GstRTSPResult gst_rtspsrc_open (GstRTSPSrc * src, gboolean async);
 static GstRTSPResult gst_rtspsrc_play (GstRTSPSrc * src, GstSegment * segment,
     gboolean async);
-static GstRTSPResult gst_rtspsrc_pause (GstRTSPSrc * src, gboolean idle,
-    gboolean async);
+static GstRTSPResult gst_rtspsrc_pause (GstRTSPSrc * src, gboolean async);
 static GstRTSPResult gst_rtspsrc_close (GstRTSPSrc * src, gboolean async,
     gboolean only_close);
 
 static gboolean gst_rtspsrc_uri_set_uri (GstURIHandler * handler,
     const gchar * uri, GError ** error);
+static gchar *gst_rtspsrc_uri_get_uri (GstURIHandler * handler);
 
 static gboolean gst_rtspsrc_activate_streams (GstRTSPSrc * src);
 static gboolean gst_rtspsrc_loop (GstRTSPSrc * src);
 static gboolean gst_rtspsrc_stream_push_event (GstRTSPSrc * src,
-    GstRTSPStream * stream, GstEvent * event, gboolean source);
-static gboolean gst_rtspsrc_push_event (GstRTSPSrc * src, GstEvent * event,
-    gboolean source);
+    GstRTSPStream * stream, GstEvent * event);
+static gboolean gst_rtspsrc_push_event (GstRTSPSrc * src, GstEvent * event);
 
 /* commands we send to out loop to notify it of events */
-#define CMD_OPEN	0
-#define CMD_PLAY	1
-#define CMD_PAUSE	2
-#define CMD_CLOSE	3
-#define CMD_WAIT	4
-#define CMD_RECONNECT	5
-#define CMD_LOOP	6
+#define CMD_OPEN	(1 << 0)
+#define CMD_PLAY	(1 << 1)
+#define CMD_PAUSE	(1 << 2)
+#define CMD_CLOSE	(1 << 3)
+#define CMD_WAIT	(1 << 4)
+#define CMD_RECONNECT	(1 << 5)
+#define CMD_LOOP	(1 << 6)
 
 #define GST_ELEMENT_PROGRESS(el, type, code, text)      \
 G_STMT_START {                                          \
@@ -349,6 +353,12 @@ gst_rtspsrc_class_init (GstRTSPSrcClass * klass)
           "Amount of ms to buffer", 0, G_MAXUINT, DEFAULT_LATENCY_MS,
           G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
 
+  g_object_class_install_property (gobject_class, PROP_DROP_ON_LATENCY,
+      g_param_spec_boolean ("drop-on-latency",
+          "Drop buffers when maximum latency is reached",
+          "Tells the jitterbuffer to never exceed the given latency in size",
+          DEFAULT_DROP_ON_LATENCY, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
+
   g_object_class_install_property (gobject_class, PROP_CONNECTION_SPEED,
       g_param_spec_uint64 ("connection-speed", "Connection Speed",
           "Network connection speed in kbps (0 = unknown)",
@@ -476,13 +486,24 @@ gst_rtspsrc_class_init (GstRTSPSrcClass * klass)
           "Only send the basic RTSP headers for broken encoders",
           DEFAULT_SHORT_HEADER, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
 
+  g_object_class_install_property (gobject_class, PROP_PROBATION,
+      g_param_spec_uint ("probation", "Number of probations",
+          "Consecutive packet sequence numbers to accept the source",
+          0, G_MAXUINT, DEFAULT_PROBATION,
+          G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
+
+  g_object_class_install_property (gobject_class, PROP_UDP_RECONNECT,
+      g_param_spec_boolean ("udp-reconnect", "Reconnect to the server",
+          "Reconnect to the server if RTSP connection is closed when doing UDP",
+          DEFAULT_UDP_RECONNECT, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
+
   gstelement_class->send_event = gst_rtspsrc_send_event;
   gstelement_class->change_state = gst_rtspsrc_change_state;
 
   gst_element_class_add_pad_template (gstelement_class,
       gst_static_pad_template_get (&rtptemplate));
 
-  gst_element_class_set_details_simple (gstelement_class,
+  gst_element_class_set_static_metadata (gstelement_class,
       "RTSP packet receiver", "Source/Network",
       "Receive data over the network via RTSP (RFC 2326)",
       "Wim Taymans <wim@fluendo.com>, "
@@ -505,6 +526,7 @@ gst_rtspsrc_init (GstRTSPSrc * src)
   src->udp_timeout = DEFAULT_TIMEOUT;
   gst_rtspsrc_set_tcp_timeout (src, DEFAULT_TCP_TIMEOUT);
   src->latency = DEFAULT_LATENCY_MS;
+  src->drop_on_latency = DEFAULT_DROP_ON_LATENCY;
   src->connection_speed = DEFAULT_CONNECTION_SPEED;
   src->nat_method = DEFAULT_NAT_METHOD;
   src->do_rtcp = DEFAULT_DO_RTCP;
@@ -518,6 +540,8 @@ gst_rtspsrc_init (GstRTSPSrc * src)
   src->client_port_range.max = 0;
   src->udp_buffer_size = DEFAULT_UDP_BUFFER_SIZE;
   src->short_header = DEFAULT_SHORT_HEADER;
+  src->probation = DEFAULT_PROBATION;
+  src->udp_reconnect = DEFAULT_UDP_RECONNECT;
 
   /* get a list of all extensions */
   src->extensions = gst_rtsp_ext_list_get ();
@@ -658,6 +682,9 @@ gst_rtspsrc_set_property (GObject * object, guint prop_id, const GValue * value,
     case PROP_LATENCY:
       rtspsrc->latency = g_value_get_uint (value);
       break;
+    case PROP_DROP_ON_LATENCY:
+      rtspsrc->drop_on_latency = g_value_get_boolean (value);
+      break;
     case PROP_CONNECTION_SPEED:
       rtspsrc->connection_speed = g_value_get_uint64 (value);
       break;
@@ -709,6 +736,12 @@ gst_rtspsrc_set_property (GObject * object, guint prop_id, const GValue * value,
     case PROP_SHORT_HEADER:
       rtspsrc->short_header = g_value_get_boolean (value);
       break;
+    case PROP_PROBATION:
+      rtspsrc->probation = g_value_get_uint (value);
+      break;
+    case PROP_UDP_RECONNECT:
+      rtspsrc->udp_reconnect = g_value_get_boolean (value);
+      break;
     default:
       G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
       break;
@@ -751,6 +784,9 @@ gst_rtspsrc_get_property (GObject * object, guint prop_id, GValue * value,
     case PROP_LATENCY:
       g_value_set_uint (value, rtspsrc->latency);
       break;
+    case PROP_DROP_ON_LATENCY:
+      g_value_set_boolean (value, rtspsrc->drop_on_latency);
+      break;
     case PROP_CONNECTION_SPEED:
       g_value_set_uint64 (value, rtspsrc->connection_speed);
       break;
@@ -807,6 +843,12 @@ gst_rtspsrc_get_property (GObject * object, guint prop_id, GValue * value,
     case PROP_SHORT_HEADER:
       g_value_set_boolean (value, rtspsrc->short_header);
       break;
+    case PROP_PROBATION:
+      g_value_set_uint (value, rtspsrc->probation);
+      break;
+    case PROP_UDP_RECONNECT:
+      g_value_set_boolean (value, rtspsrc->udp_reconnect);
+      break;
     default:
       G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
       break;
@@ -1205,6 +1247,10 @@ gst_rtspsrc_cleanup (GstRTSPSrc * src)
     gst_sdp_message_free (src->sdp);
     src->sdp = NULL;
   }
+  if (src->start_segment) {
+    gst_event_unref (src->start_segment);
+    src->start_segment = NULL;
+  }
 }
 
 #define PARSE_INT(p, del, res)          \
@@ -1519,7 +1565,7 @@ again:
       tmp_rtp >= src->client_port_range.max)
     goto no_ports;
 
-  udpsrc0 = gst_element_make_from_uri (GST_URI_SRC, host, NULL);
+  udpsrc0 = gst_element_make_from_uri (GST_URI_SRC, host, NULL, NULL);
   if (udpsrc0 == NULL)
     goto no_udp_protocol;
   g_object_set (G_OBJECT (udpsrc0), "port", tmp_rtp, "reuse", FALSE, NULL);
@@ -1540,6 +1586,7 @@ again:
       GST_DEBUG_OBJECT (src, "free RTP udpsrc");
       gst_element_set_state (udpsrc0, GST_STATE_NULL);
       gst_object_unref (udpsrc0);
+      udpsrc0 = NULL;
 
       GST_DEBUG_OBJECT (src, "retry %d", count);
       goto again;
@@ -1561,6 +1608,7 @@ again:
     GST_DEBUG_OBJECT (src, "free RTP udpsrc");
     gst_element_set_state (udpsrc0, GST_STATE_NULL);
     gst_object_unref (udpsrc0);
+    udpsrc0 = NULL;
 
     GST_DEBUG_OBJECT (src, "retry %d", count);
     tmp_rtp++;
@@ -1568,13 +1616,13 @@ again:
   }
 
   /* allocate port+1 for RTCP now */
-  udpsrc1 = gst_element_make_from_uri (GST_URI_SRC, host, NULL);
+  udpsrc1 = gst_element_make_from_uri (GST_URI_SRC, host, NULL, NULL);
   if (udpsrc1 == NULL)
     goto no_udp_rtcp_protocol;
 
   /* set port */
   tmp_rtcp = tmp_rtp + 1;
-  if (src->client_port_range.max > 0 && tmp_rtcp >= src->client_port_range.max)
+  if (src->client_port_range.max > 0 && tmp_rtcp > src->client_port_range.max)
     goto no_ports;
 
   g_object_set (G_OBJECT (udpsrc1), "port", tmp_rtcp, "reuse", FALSE, NULL);
@@ -1591,6 +1639,7 @@ again:
     GST_DEBUG_OBJECT (src, "free RTP udpsrc");
     gst_element_set_state (udpsrc0, GST_STATE_NULL);
     gst_object_unref (udpsrc0);
+    udpsrc0 = NULL;
 
     GST_DEBUG_OBJECT (src, "free RTCP udpsrc");
     gst_element_set_state (udpsrc1, GST_STATE_NULL);
@@ -1666,8 +1715,6 @@ gst_rtspsrc_flush (GstRTSPSrc * src, gboolean flush, gboolean playing)
   gint cmd, i;
   GstState state;
   GList *walk;
-  GstClock *clock;
-  GstClockTime base_time = GST_CLOCK_TIME_NONE;
 
   if (flush) {
     event = gst_event_new_flush_start ();
@@ -1675,31 +1722,20 @@ gst_rtspsrc_flush (GstRTSPSrc * src, gboolean flush, gboolean playing)
     cmd = CMD_WAIT;
     state = GST_STATE_PAUSED;
   } else {
-    event = gst_event_new_flush_stop (TRUE);
+    event = gst_event_new_flush_stop (FALSE);
     GST_DEBUG_OBJECT (src, "stop flush; playing %d", playing);
     cmd = CMD_LOOP;
     if (playing)
       state = GST_STATE_PLAYING;
     else
       state = GST_STATE_PAUSED;
-    clock = gst_element_get_clock (GST_ELEMENT_CAST (src));
-    if (clock) {
-      base_time = gst_clock_get_time (clock);
-      gst_object_unref (clock);
-    }
   }
-  gst_rtspsrc_push_event (src, event, FALSE);
-  gst_rtspsrc_loop_send_cmd (src, cmd);
+  gst_rtspsrc_push_event (src, event);
+  gst_rtspsrc_loop_send_cmd (src, cmd, CMD_LOOP);
 
-  /* set up manager before data-flow resumes */
   /* to manage jitterbuffer buffer mode */
-  if (src->manager) {
-    gst_element_set_base_time (GST_ELEMENT_CAST (src->manager), base_time);
-    /* and to have base_time trickle further down,
-     * e.g. to jitterbuffer for its timeout handling */
-    if (base_time != -1)
-      gst_element_set_state (GST_ELEMENT_CAST (src->manager), state);
-  }
+  if (src->manager)
+    gst_element_set_state (GST_ELEMENT_CAST (src->manager), state);
 
   /* make running time start start at 0 again */
   for (walk = src->streams; walk; walk = g_list_next (walk)) {
@@ -1708,15 +1744,10 @@ gst_rtspsrc_flush (GstRTSPSrc * src, gboolean flush, gboolean playing)
     for (i = 0; i < 2; i++) {
       /* for udp case */
       if (stream->udpsrc[i]) {
-        if (base_time != -1)
-          gst_element_set_base_time (stream->udpsrc[i], base_time);
         gst_element_set_state (stream->udpsrc[i], state);
       }
     }
   }
-  /* for tcp interleaved case */
-  if (base_time != -1)
-    gst_element_set_base_time (GST_ELEMENT_CAST (src), base_time);
 }
 
 static GstRTSPResult
@@ -1866,8 +1897,9 @@ gst_rtspsrc_perform_seek (GstRTSPSrc * src, GstEvent * event)
   if (playing) {
     /* obtain current position in case seek fails */
     gst_rtspsrc_get_position (src);
-    gst_rtspsrc_pause (src, FALSE, FALSE);
+    gst_rtspsrc_pause (src, FALSE);
   }
+  src->skip = skip;
 
   gst_rtspsrc_do_seek (src, &seeksegment);
 
@@ -1896,18 +1928,12 @@ gst_rtspsrc_perform_seek (GstRTSPSrc * src, GstEvent * event)
   GST_DEBUG_OBJECT (src, "Creating newsegment from %" G_GINT64_FORMAT
       " to %" G_GINT64_FORMAT, src->segment.position, stop);
 
-  /* store the newsegment event so it can be sent from the streaming thread. */
-  if (src->start_segment)
-    gst_event_unref (src->start_segment);
-  src->start_segment = gst_event_new_segment (&src->segment);
-
   /* mark discont */
   GST_DEBUG_OBJECT (src, "mark DISCONT, we did a seek to another position");
   for (walk = src->streams; walk; walk = g_list_next (walk)) {
     GstRTSPStream *stream = (GstRTSPStream *) walk->data;
     stream->discont = TRUE;
   }
-  src->skip = skip;
 
   GST_RTSP_STREAM_UNLOCK (src);
 
@@ -2090,6 +2116,18 @@ gst_rtspsrc_handle_src_query (GstPad * pad, GstObject * parent,
       }
       break;
     }
+    case GST_QUERY_URI:
+    {
+      gchar *uri;
+
+      uri = gst_rtspsrc_uri_get_uri (GST_URI_HANDLER (src));
+      if (uri != NULL) {
+        gst_query_set_uri (query, uri);
+        g_free (uri);
+        res = TRUE;
+      }
+      break;
+    }
     default:
     {
       GstPad *target = gst_ghost_pad_get_target (GST_GHOST_PAD_CAST (pad));
@@ -2188,7 +2226,7 @@ new_manager_pad (GstElement * manager, GstPad * pad, GstRTSPSrc * src)
   gchar *name;
   GstPadTemplate *template;
   gint id, ssrc, pt;
-  GList *lstream;
+  GList *ostreams;
   GstRTSPStream *stream;
   gboolean all_added;
 
@@ -2200,41 +2238,43 @@ new_manager_pad (GstElement * manager, GstPad * pad, GstRTSPSrc * src)
   if (sscanf (name, "recv_rtp_src_%u_%u_%u", &id, &ssrc, &pt) != 3)
     goto unknown_stream;
 
-  GST_DEBUG_OBJECT (src, "stream: %u, SSRC %d, PT %d", id, ssrc, pt);
+  GST_DEBUG_OBJECT (src, "stream: %u, SSRC %08x, PT %d", id, ssrc, pt);
 
   stream = find_stream (src, &id, (gpointer) find_stream_by_id);
   if (stream == NULL)
     goto unknown_stream;
 
-  /* create a new pad we will use to stream to */
-  template = gst_static_pad_template_get (&rtptemplate);
-  stream->srcpad = gst_ghost_pad_new_from_template (name, pad, template);
-  gst_object_unref (template);
-  g_free (name);
-
+  /* we'll add it later see below */
   stream->added = TRUE;
-  gst_pad_set_event_function (stream->srcpad, gst_rtspsrc_handle_src_event);
-  gst_pad_set_query_function (stream->srcpad, gst_rtspsrc_handle_src_query);
-  gst_pad_set_active (stream->srcpad, TRUE);
-  gst_element_add_pad (GST_ELEMENT_CAST (src), stream->srcpad);
 
   /* check if we added all streams */
   all_added = TRUE;
-  for (lstream = src->streams; lstream; lstream = g_list_next (lstream)) {
-    stream = (GstRTSPStream *) lstream->data;
+  for (ostreams = src->streams; ostreams; ostreams = g_list_next (ostreams)) {
+    GstRTSPStream *ostream = (GstRTSPStream *) ostreams->data;
 
     GST_DEBUG_OBJECT (src, "stream %p, container %d, disabled %d, added %d",
-        stream, stream->container, stream->disabled, stream->added);
+        ostream, ostream->container, ostream->disabled, ostream->added);
 
     /* a container stream only needs one pad added. Also disabled streams don't
      * count */
-    if (!stream->container && !stream->disabled && !stream->added) {
+    if (!ostream->container && !ostream->disabled && !ostream->added) {
       all_added = FALSE;
       break;
     }
   }
   GST_RTSP_STATE_UNLOCK (src);
 
+  /* create a new pad we will use to stream to */
+  template = gst_static_pad_template_get (&rtptemplate);
+  stream->srcpad = gst_ghost_pad_new_from_template (name, pad, template);
+  gst_object_unref (template);
+  g_free (name);
+
+  gst_pad_set_event_function (stream->srcpad, gst_rtspsrc_handle_src_event);
+  gst_pad_set_query_function (stream->srcpad, gst_rtspsrc_handle_src_query);
+  gst_pad_set_active (stream->srcpad, TRUE);
+  gst_element_add_pad (GST_ELEMENT_CAST (src), stream->srcpad);
+
   if (all_added) {
     GST_DEBUG_OBJECT (src, "We added all streams");
     /* when we get here, all stream are added and we can fire the no-more-pads
@@ -2291,7 +2331,7 @@ gst_rtspsrc_do_stream_eos (GstRTSPSrc * src, GstRTSPStream * stream)
     goto was_eos;
 
   stream->eos = TRUE;
-  gst_rtspsrc_stream_push_event (src, stream, gst_event_new_eos (), TRUE);
+  gst_rtspsrc_stream_push_event (src, stream, gst_event_new_eos ());
   return;
 
   /* ERRORS */
@@ -2390,6 +2430,11 @@ gst_rtspsrc_stream_configure_manager (GstRTSPSrc * src, GstRTSPStream * stream,
       g_object_set (src->manager, "latency", src->latency, NULL);
 
       klass = G_OBJECT_GET_CLASS (G_OBJECT (src->manager));
+      if (g_object_class_find_property (klass, "drop-on-latency")) {
+        g_object_set (src->manager, "drop-on-latency", src->drop_on_latency,
+            NULL);
+      }
+
       if (g_object_class_find_property (klass, "buffer-mode")) {
         if (src->buffer_mode != BUFFER_MODE_AUTO) {
           g_object_set (src->manager, "buffer-mode", src->buffer_mode, NULL);
@@ -2475,6 +2520,9 @@ gst_rtspsrc_stream_configure_manager (GstRTSPSrc * src, GstRTSPStream * stream,
           g_object_set (rtpsession, "rtcp-rs-bandwidth", stream->rs_bandwidth,
               NULL);
         }
+
+        g_object_set (rtpsession, "probation", src->probation, NULL);
+
         g_signal_connect (rtpsession, "on-bye-ssrc", (GCallback) on_bye_ssrc,
             stream);
         g_signal_connect (rtpsession, "on-bye-timeout", (GCallback) on_timeout,
@@ -2518,6 +2566,7 @@ gst_rtspsrc_stream_free_udp (GstRTSPStream * stream)
 
   for (i = 0; i < 2; i++) {
     if (stream->udpsrc[i]) {
+      GST_DEBUG ("free UDP source %d for stream %p", i, stream);
       gst_element_set_state (stream->udpsrc[i], GST_STATE_NULL);
       gst_object_unref (stream->udpsrc[i]);
       stream->udpsrc[i] = NULL;
@@ -2695,7 +2744,8 @@ gst_rtspsrc_stream_configure_mcast (GstRTSPSrc * src, GstRTSPStream * stream,
   /* creating UDP source for RTP */
   if (min != -1) {
     uri = g_strdup_printf ("udp://%s:%d", destination, min);
-    stream->udpsrc[0] = gst_element_make_from_uri (GST_URI_SRC, uri, NULL);
+    stream->udpsrc[0] =
+        gst_element_make_from_uri (GST_URI_SRC, uri, NULL, NULL);
     g_free (uri);
     if (stream->udpsrc[0] == NULL)
       goto no_element;
@@ -2703,6 +2753,10 @@ gst_rtspsrc_stream_configure_mcast (GstRTSPSrc * src, GstRTSPStream * stream,
     /* take ownership */
     gst_object_ref_sink (stream->udpsrc[0]);
 
+    if (src->udp_buffer_size != 0)
+      g_object_set (G_OBJECT (stream->udpsrc[0]), "buffer-size",
+          src->udp_buffer_size, NULL);
+
     /* change state */
     gst_element_set_state (stream->udpsrc[0], GST_STATE_PAUSED);
   }
@@ -2710,7 +2764,8 @@ gst_rtspsrc_stream_configure_mcast (GstRTSPSrc * src, GstRTSPStream * stream,
   /* creating another UDP source for RTCP */
   if (max != -1) {
     uri = g_strdup_printf ("udp://%s:%d", destination, max);
-    stream->udpsrc[1] = gst_element_make_from_uri (GST_URI_SRC, uri, NULL);
+    stream->udpsrc[1] =
+        gst_element_make_from_uri (GST_URI_SRC, uri, NULL, NULL);
     g_free (uri);
     if (stream->udpsrc[1] == NULL)
       goto no_element;
@@ -2755,8 +2810,8 @@ gst_rtspsrc_stream_configure_udp (GstRTSPSrc * src, GstRTSPStream * stream,
     /* configure a timeout on the UDP port. When the timeout message is
      * posted, we assume UDP transport is not possible. We reconnect using TCP
      * if we can. */
-    g_object_set (G_OBJECT (stream->udpsrc[0]), "timeout", src->udp_timeout,
-        NULL);
+    g_object_set (G_OBJECT (stream->udpsrc[0]), "timeout",
+        src->udp_timeout * 1000, NULL);
 
     /* get output pad of the UDP source. */
     *outpad = gst_element_get_static_pad (stream->udpsrc[0], "src");
@@ -2837,7 +2892,8 @@ gst_rtspsrc_stream_configure_udp_sinks (GstRTSPSrc * src,
         rtp_port);
 
     uri = g_strdup_printf ("udp://%s:%d", destination, rtp_port);
-    stream->udpsink[0] = gst_element_make_from_uri (GST_URI_SINK, uri, NULL);
+    stream->udpsink[0] =
+        gst_element_make_from_uri (GST_URI_SINK, uri, NULL, NULL);
     g_free (uri);
     if (stream->udpsink[0] == NULL)
       goto no_sink_element;
@@ -2889,7 +2945,8 @@ gst_rtspsrc_stream_configure_udp_sinks (GstRTSPSrc * src,
         rtcp_port);
 
     uri = g_strdup_printf ("udp://%s:%d", destination, rtcp_port);
-    stream->udpsink[1] = gst_element_make_from_uri (GST_URI_SINK, uri, NULL);
+    stream->udpsink[1] =
+        gst_element_make_from_uri (GST_URI_SINK, uri, NULL, NULL);
     g_free (uri);
     if (stream->udpsink[1] == NULL)
       goto no_sink_element;
@@ -3103,15 +3160,15 @@ gst_rtspsrc_activate_streams (GstRTSPSrc * src)
       g_object_set (G_OBJECT (stream->udpsrc[0]), "timeout", (guint64) 0, NULL);
     }
     if (stream->srcpad) {
+      GST_DEBUG_OBJECT (src, "activating stream pad %p", stream);
+      gst_pad_set_active (stream->srcpad, TRUE);
+
       /* if we don't have a session manager, set the caps now. If we have a
        * session, we will get a notification of the pad and the caps. */
       if (!src->manager) {
         GST_DEBUG_OBJECT (src, "setting pad caps for stream %p", stream);
         gst_pad_set_caps (stream->srcpad, stream->caps);
       }
-
-      GST_DEBUG_OBJECT (src, "activating stream pad %p", stream);
-      gst_pad_set_active (stream->srcpad, TRUE);
       /* add the pad */
       if (!stream->added) {
         GST_DEBUG_OBJECT (src, "adding stream pad %p", stream);
@@ -3136,7 +3193,8 @@ gst_rtspsrc_activate_streams (GstRTSPSrc * src)
 }
 
 static void
-gst_rtspsrc_configure_caps (GstRTSPSrc * src, GstSegment * segment)
+gst_rtspsrc_configure_caps (GstRTSPSrc * src, GstSegment * segment,
+    gboolean reset_manager)
 {
   GList *walk;
   guint64 start, stop;
@@ -3172,7 +3230,7 @@ gst_rtspsrc_configure_caps (GstRTSPSrc * src, GstSegment * segment)
     }
     GST_DEBUG_OBJECT (src, "stream %p, caps %" GST_PTR_FORMAT, stream, caps);
   }
-  if (src->manager) {
+  if (reset_manager && src->manager) {
     GST_DEBUG_OBJECT (src, "clear session");
     g_signal_emit_by_name (src->manager, "clear-pt-map", NULL);
   }
@@ -3214,15 +3272,15 @@ done:
 
 static gboolean
 gst_rtspsrc_stream_push_event (GstRTSPSrc * src, GstRTSPStream * stream,
-    GstEvent * event, gboolean source)
+    GstEvent * event)
 {
   gboolean res = TRUE;
 
   /* only streams that have a connection to the outside world */
-  if (stream->srcpad == NULL)
+  if (stream->container || stream->disabled)
     goto done;
 
-  if (source && stream->udpsrc[0]) {
+  if (stream->udpsrc[0]) {
     gst_event_ref (event);
     res = gst_element_send_event (stream->udpsrc[0], event);
   } else if (stream->channelpad[0]) {
@@ -3233,7 +3291,7 @@ gst_rtspsrc_stream_push_event (GstRTSPSrc * src, GstRTSPStream * stream,
       res = gst_pad_send_event (stream->channelpad[0], event);
   }
 
-  if (source && stream->udpsrc[1]) {
+  if (stream->udpsrc[1]) {
     gst_event_ref (event);
     res &= gst_element_send_event (stream->udpsrc[1], event);
   } else if (stream->channelpad[1]) {
@@ -3251,7 +3309,7 @@ done:
 }
 
 static gboolean
-gst_rtspsrc_push_event (GstRTSPSrc * src, GstEvent * event, gboolean source)
+gst_rtspsrc_push_event (GstRTSPSrc * src, GstEvent * event)
 {
   GList *streams;
   gboolean res = TRUE;
@@ -3260,7 +3318,7 @@ gst_rtspsrc_push_event (GstRTSPSrc * src, GstEvent * event, gboolean source)
     GstRTSPStream *ostream = (GstRTSPStream *) streams->data;
 
     gst_event_ref (event);
-    res &= gst_rtspsrc_stream_push_event (src, ostream, event, source);
+    res &= gst_rtspsrc_stream_push_event (src, ostream, event);
   }
   gst_event_unref (event);
 
@@ -3343,6 +3401,7 @@ static GstRTSPResult
 gst_rtsp_conninfo_close (GstRTSPSrc * src, GstRTSPConnInfo * info,
     gboolean free)
 {
+  GST_RTSP_STATE_LOCK (src);
   if (info->connected) {
     GST_DEBUG_OBJECT (src, "closing connection...");
     gst_rtsp_connection_close (info->connection);
@@ -3354,6 +3413,7 @@ gst_rtsp_conninfo_close (GstRTSPSrc * src, GstRTSPConnInfo * info,
     gst_rtsp_connection_free (info->connection);
     info->connection = NULL;
   }
+  GST_RTSP_STATE_UNLOCK (src);
   return GST_RTSP_OK;
 }
 
@@ -3376,6 +3436,7 @@ gst_rtspsrc_connection_flush (GstRTSPSrc * src, gboolean flush)
   GList *walk;
 
   GST_DEBUG_OBJECT (src, "set flushing %d", flush);
+  GST_RTSP_STATE_LOCK (src);
   if (src->conninfo.connection) {
     GST_DEBUG_OBJECT (src, "connection flush");
     gst_rtsp_connection_flush (src->conninfo.connection, flush);
@@ -3386,6 +3447,7 @@ gst_rtspsrc_connection_flush (GstRTSPSrc * src, gboolean flush)
     if (stream->conninfo.connection)
       gst_rtsp_connection_flush (stream->conninfo.connection, flush);
   }
+  GST_RTSP_STATE_UNLOCK (src);
 }
 
 /* FIXME, handle server request, reply with OK, for now */
@@ -3514,6 +3576,7 @@ gst_rtspsrc_loop_interleaved (GstRTSPSrc * src)
   GstFlowReturn ret = GST_FLOW_OK;
   GstBuffer *buf;
   gboolean is_rtcp, have_data;
+  GstEvent *event;
 
   /* here we are only interested in data messages */
   have_data = FALSE;
@@ -3644,6 +3707,10 @@ gst_rtspsrc_loop_interleaved (GstRTSPSrc * src)
     gst_rtspsrc_activate_streams (src);
     src->need_activate = FALSE;
   }
+  if ((event = src->start_segment) != NULL) {
+    src->start_segment = NULL;
+    gst_rtspsrc_push_event (src, event);
+  }
 
   if (src->base_time == -1) {
     /* Take current running_time. This timestamp will be put on
@@ -3788,10 +3855,13 @@ gst_rtspsrc_loop_udp (GstRTSPSrc * src)
          * see what happens. */
         GST_ELEMENT_WARNING (src, RESOURCE, READ, (NULL),
             ("The server closed the connection."));
-        if ((res =
-                gst_rtsp_conninfo_reconnect (src, &src->conninfo, FALSE)) < 0)
-          goto connect_error;
-
+        if (src->udp_reconnect) {
+          if ((res =
+                  gst_rtsp_conninfo_reconnect (src, &src->conninfo, FALSE)) < 0)
+            goto connect_error;
+        } else {
+          goto server_eof;
+        }
         continue;
       case GST_RTSP_ENET:
         GST_DEBUG_OBJECT (src, "An ethernet problem occured.");
@@ -4066,7 +4136,7 @@ gst_rtspsrc_loop_end_cmd (GstRTSPSrc * src, gint cmd, GstRTSPResult ret)
 }
 
 static void
-gst_rtspsrc_loop_send_cmd (GstRTSPSrc * src, gint cmd)
+gst_rtspsrc_loop_send_cmd (GstRTSPSrc * src, gint cmd, gint mask)
 {
   gint old;
 
@@ -4076,19 +4146,25 @@ gst_rtspsrc_loop_send_cmd (GstRTSPSrc * src, gint cmd)
   GST_DEBUG_OBJECT (src, "sending cmd %d", cmd);
 
   GST_OBJECT_LOCK (src);
-  old = src->loop_cmd;
+  old = src->pending_cmd;
+  if (old == CMD_RECONNECT) {
+    GST_DEBUG_OBJECT (src, "ignore, we were reconnecting");
+    cmd = CMD_RECONNECT;
+  }
   if (old != CMD_WAIT) {
-    src->loop_cmd = CMD_WAIT;
+    src->pending_cmd = CMD_WAIT;
     GST_OBJECT_UNLOCK (src);
     /* cancel previous request */
     gst_rtspsrc_loop_cancel_cmd (src, old);
     GST_OBJECT_LOCK (src);
   }
-  src->loop_cmd = cmd;
+  src->pending_cmd = cmd;
   /* interrupt if allowed */
-  if (src->waiting) {
-    GST_DEBUG_OBJECT (src, "start connection flush");
+  if (src->busy_cmd & mask) {
+    GST_DEBUG_OBJECT (src, "connection flush busy %d", src->busy_cmd);
     gst_rtspsrc_connection_flush (src, TRUE);
+  } else {
+    GST_DEBUG_OBJECT (src, "not interrupting busy cmd %d", src->busy_cmd);
   }
   if (src->task)
     gst_task_start (src->task);
@@ -4132,8 +4208,11 @@ pause:
         gst_element_post_message (GST_ELEMENT_CAST (src),
             gst_message_new_segment_done (GST_OBJECT_CAST (src),
                 src->segment.format, src->segment.position));
+        gst_rtspsrc_push_event (src,
+            gst_event_new_segment_done (src->segment.format,
+                src->segment.position));
       } else {
-        gst_rtspsrc_push_event (src, gst_event_new_eos (), FALSE);
+        gst_rtspsrc_push_event (src, gst_event_new_eos ());
       }
     } else if (ret == GST_FLOW_NOT_LINKED || ret < GST_FLOW_EOS) {
       /* for fatal errors we post an error message, post the error before the
@@ -4141,8 +4220,9 @@ pause:
       GST_ELEMENT_ERROR (src, STREAM, FAILED,
           ("Internal data flow error."),
           ("streaming task paused, reason %s (%d)", reason, ret));
-      gst_rtspsrc_push_event (src, gst_event_new_eos (), FALSE);
+      gst_rtspsrc_push_event (src, gst_event_new_eos ());
     }
+    gst_rtspsrc_loop_send_cmd (src, CMD_WAIT, CMD_LOOP);
     return FALSE;
   }
 }
@@ -4529,8 +4609,8 @@ receive_error:
   {
     switch (res) {
       case GST_RTSP_EEOF:
-        GST_WARNING_OBJECT (src, "server closed connection, doing reconnect");
-        if (try == 0) {
+        GST_WARNING_OBJECT (src, "server closed connection");
+        if ((try == 0) && !src->interleaved && src->udp_reconnect) {
           try++;
           /* if reconnect succeeds, try again */
           if ((res =
@@ -4731,9 +4811,7 @@ gst_rtspsrc_parse_methods (GstRTSPSrc * src, GstRTSPMessage * response)
 {
   GstRTSPHeaderField field;
   gchar *respoptions;
-  gchar **options;
   gint indx = 0;
-  gint i;
 
   /* reset supported methods */
   src->methods = 0;
@@ -4751,25 +4829,7 @@ gst_rtspsrc_parse_methods (GstRTSPSrc * src, GstRTSPMessage * response)
     if (!respoptions)
       break;
 
-    /* If we get here, the server gave a list of supported methods, parse
-     * them here. The string is like:
-     *
-     * OPTIONS, DESCRIBE, ANNOUNCE, PLAY, SETUP, ...
-     */
-    options = g_strsplit (respoptions, ",", 0);
-
-    for (i = 0; options[i]; i++) {
-      gchar *stripped;
-      gint method;
-
-      stripped = g_strstrip (options[i]);
-      method = gst_rtsp_find_method (stripped);
-
-      /* keep bitfield of supported methods */
-      if (method != GST_RTSP_INVALID)
-        src->methods |= method;
-    }
-    g_strfreev (options);
+    src->methods |= gst_rtsp_options_from_text (respoptions);
 
     indx++;
   }
@@ -4865,6 +4925,14 @@ gst_rtspsrc_create_transports_string (GstRTSPSrc * src,
     if (add_udp_str)
       g_string_append (result, "/UDP");
     g_string_append (result, ";multicast");
+    if (src->next_port_num != 0) {
+      if (src->client_port_range.max > 0 &&
+          src->next_port_num >= src->client_port_range.max)
+        goto no_ports;
+
+      g_string_append_printf (result, ";client_port=%d-%d",
+          src->next_port_num, src->next_port_num + 1);
+    }
   } else if (protocols & GST_RTSP_LOWER_TRANS_TCP) {
     GST_DEBUG_OBJECT (src, "adding TCP");
 
@@ -4881,8 +4949,14 @@ gst_rtspsrc_create_transports_string (GstRTSPSrc * src,
   /* ERRORS */
 failed:
   {
+    GST_ERROR ("extension gave error %d", res);
     return res;
   }
+no_ports:
+  {
+    GST_ERROR ("no more ports available");
+    return GST_RTSP_ERROR;
+  }
 }
 
 static GstRTSPResult
@@ -4956,6 +5030,7 @@ done:
   /* ERRORS */
 failed:
   {
+    GST_ERROR ("failed to allocate udp ports");
     return GST_RTSP_ERROR;
   }
 }
@@ -5225,6 +5300,12 @@ gst_rtspsrc_setup_streams (GstRTSPSrc * src, gboolean async)
           /* only allow multicast for other streams */
           GST_DEBUG_OBJECT (src, "stream %p as UDP multicast", stream);
           protocols = GST_RTSP_LOWER_TRANS_UDP_MCAST;
+          /* if the server selected our ports, increment our counters so that
+           * we select a new port later */
+          if (src->next_port_num == transport.port.min &&
+              src->next_port_num + 1 == transport.port.max) {
+            src->next_port_num += 2;
+          }
           break;
         case GST_RTSP_LOWER_TRANS_UDP:
           /* only allow unicast for other streams */
@@ -5512,6 +5593,7 @@ gst_rtspsrc_open_from_sdp (GstRTSPSrc * src, GstSDPMessage * sdp,
 setup_failed:
   {
     GST_ERROR_OBJECT (src, "setup failed");
+    gst_rtspsrc_cleanup (src);
     return res;
   }
 }
@@ -6066,8 +6148,23 @@ gst_rtspsrc_play (GstRTSPSrc * src, GstSegment * segment, gboolean async)
    * only in async case, since receive elements may not have been affected
    * by overall state change (e.g. not around yet),
    * do not mess with state in sync case (e.g. seeking) */
-  if (async)
-    gst_element_set_state (GST_ELEMENT_CAST (src), GST_STATE_PLAYING);
+  if (async) {
+    /* state change might be happening in the application thread. A
+     * specific case is when chaging state to NULL where we will wait
+     * for this task to finish (gst_rtspsrc_stop). However this task
+     * will try to change the state to PLAYING causing a deadlock. */
+
+    /* make sure we are not in the middle of a state change. The
+     * state lock is a recursive lock so it's safe to lock twice from
+     * the same thread */
+    if (GST_STATE_TRYLOCK (src)) {
+      gst_element_set_state (GST_ELEMENT_CAST (src), GST_STATE_PLAYING);
+      GST_STATE_UNLOCK (src);
+    } else {
+      res = GST_RTSP_ERROR;
+      goto changing_state;
+    }
+  }
 
   /* construct a control url */
   if (src->control)
@@ -6104,6 +6201,11 @@ gst_rtspsrc_play (GstRTSPSrc * src, GstSegment * segment, gboolean async)
 
       gst_rtsp_message_add_header (&request, GST_RTSP_HDR_RANGE, hval);
       g_free (hval);
+
+      /* store the newsegment event so it can be sent from the streaming thread. */
+      if (src->start_segment)
+        gst_event_unref (src->start_segment);
+      src->start_segment = gst_event_new_segment (&src->segment);
     }
 
     if (segment->rate != 1.0) {
@@ -6185,12 +6287,13 @@ gst_rtspsrc_play (GstRTSPSrc * src, GstSegment * segment, gboolean async)
     if (control)
       break;
   }
+  /* configure the caps of the streams after we parsed all headers. Only reset
+   * the manager object when we set a new Range header (we did a seek) */
+  gst_rtspsrc_configure_caps (src, segment, src->need_range);
+
   /* set again when needed */
   src->need_range = FALSE;
 
-  /* configure the caps of the streams after we parsed all headers. */
-  gst_rtspsrc_configure_caps (src, segment);
-
   src->running = TRUE;
   src->base_time = -1;
   src->state = GST_RTSP_STATE_PLAYING;
@@ -6224,6 +6327,11 @@ was_playing:
     GST_DEBUG_OBJECT (src, "we were already PLAYING");
     goto done;
   }
+changing_state:
+  {
+    GST_DEBUG_OBJECT (src, "failed going to PLAYING, already changing state");
+    goto done;
+  }
 create_request_failed:
   {
     gchar *str = gst_rtsp_strresult (res);
@@ -6250,7 +6358,7 @@ send_error:
 }
 
 static GstRTSPResult
-gst_rtspsrc_pause (GstRTSPSrc * src, gboolean idle, gboolean async)
+gst_rtspsrc_pause (GstRTSPSrc * src, gboolean async)
 {
   GstRTSPResult res = GST_RTSP_OK;
   GstRTSPMessage request = { 0 };
@@ -6397,7 +6505,7 @@ gst_rtspsrc_handle_message (GstBin * bin, GstMessage * message)
         /* we only act on the first udp timeout message, others are irrelevant
          * and can be ignored. */
         if (!ignore_timeout)
-          gst_rtspsrc_loop_send_cmd (rtspsrc, CMD_RECONNECT);
+          gst_rtspsrc_loop_send_cmd (rtspsrc, CMD_RECONNECT, CMD_LOOP);
         /* eat and free */
         gst_message_unref (message);
         return;
@@ -6454,46 +6562,39 @@ static void
 gst_rtspsrc_thread (GstRTSPSrc * src)
 {
   gint cmd;
-  GstRTSPResult ret;
-  gboolean running = FALSE;
 
   GST_OBJECT_LOCK (src);
-  cmd = src->loop_cmd;
-  src->loop_cmd = CMD_WAIT;
+  cmd = src->pending_cmd;
+  if (cmd == CMD_RECONNECT || CMD_PLAY || cmd == CMD_LOOP)
+    src->pending_cmd = CMD_LOOP;
+  else
+    src->pending_cmd = CMD_WAIT;
   GST_DEBUG_OBJECT (src, "got command %d", cmd);
 
   /* we got the message command, so ensure communication is possible again */
   gst_rtspsrc_connection_flush (src, FALSE);
 
-  /* we allow these to be interrupted */
-  if (cmd == CMD_LOOP || cmd == CMD_CLOSE || cmd == CMD_PAUSE)
-    src->waiting = TRUE;
+  src->busy_cmd = cmd;
   GST_OBJECT_UNLOCK (src);
 
   switch (cmd) {
     case CMD_OPEN:
-      ret = gst_rtspsrc_open (src, TRUE);
+      gst_rtspsrc_open (src, TRUE);
       break;
     case CMD_PLAY:
-      ret = gst_rtspsrc_play (src, &src->segment, TRUE);
-      if (ret == GST_RTSP_OK)
-        running = TRUE;
+      gst_rtspsrc_play (src, &src->segment, TRUE);
       break;
     case CMD_PAUSE:
-      ret = gst_rtspsrc_pause (src, TRUE, TRUE);
-      if (ret == GST_RTSP_OK)
-        running = TRUE;
+      gst_rtspsrc_pause (src, TRUE);
       break;
     case CMD_CLOSE:
-      ret = gst_rtspsrc_close (src, TRUE, FALSE);
+      gst_rtspsrc_close (src, TRUE, FALSE);
       break;
     case CMD_LOOP:
-      running = gst_rtspsrc_loop (src);
+      gst_rtspsrc_loop (src);
       break;
     case CMD_RECONNECT:
-      ret = gst_rtspsrc_reconnect (src, FALSE);
-      if (ret == GST_RTSP_OK)
-        running = TRUE;
+      gst_rtspsrc_reconnect (src, FALSE);
       break;
     default:
       break;
@@ -6501,14 +6602,12 @@ gst_rtspsrc_thread (GstRTSPSrc * src)
 
   GST_OBJECT_LOCK (src);
   /* and go back to sleep */
-  if (src->loop_cmd == CMD_WAIT) {
-    if (running)
-      src->loop_cmd = CMD_LOOP;
-    else if (src->task)
+  if (src->pending_cmd == CMD_WAIT) {
+    if (src->task)
       gst_task_pause (src->task);
   }
   /* reset waiting */
-  src->waiting = FALSE;
+  src->busy_cmd = CMD_WAIT;
   GST_OBJECT_UNLOCK (src);
 }
 
@@ -6519,10 +6618,10 @@ gst_rtspsrc_start (GstRTSPSrc * src)
 
   GST_OBJECT_LOCK (src);
 
-  src->loop_cmd = CMD_WAIT;
+  src->pending_cmd = CMD_WAIT;
 
   if (src->task == NULL) {
-    src->task = gst_task_new ((GstTaskFunction) gst_rtspsrc_thread, src);
+    src->task = gst_task_new ((GstTaskFunction) gst_rtspsrc_thread, src, NULL);
     if (src->task == NULL)
       goto task_error;
 
@@ -6548,7 +6647,7 @@ gst_rtspsrc_stop (GstRTSPSrc * src)
   GST_DEBUG_OBJECT (src, "stopping");
 
   /* also cancels pending task */
-  gst_rtspsrc_loop_send_cmd (src, CMD_WAIT);
+  gst_rtspsrc_loop_send_cmd (src, CMD_WAIT, CMD_CLOSE);
 
   GST_OBJECT_LOCK (src);
   if ((task = src->task)) {
@@ -6596,12 +6695,12 @@ gst_rtspsrc_change_state (GstElement * element, GstStateChange transition)
       /* first attempt, don't ignore timeouts */
       rtspsrc->ignore_timeout = FALSE;
       rtspsrc->open_error = FALSE;
-      gst_rtspsrc_loop_send_cmd (rtspsrc, CMD_OPEN);
+      gst_rtspsrc_loop_send_cmd (rtspsrc, CMD_OPEN, 0);
       break;
     case GST_STATE_CHANGE_PAUSED_TO_PLAYING:
     case GST_STATE_CHANGE_PLAYING_TO_PAUSED:
       /* unblock the tcp tasks and make the loop waiting */
-      gst_rtspsrc_loop_send_cmd (rtspsrc, CMD_WAIT);
+      gst_rtspsrc_loop_send_cmd (rtspsrc, CMD_WAIT, CMD_LOOP);
       break;
     case GST_STATE_CHANGE_PAUSED_TO_READY:
       break;
@@ -6615,18 +6714,18 @@ gst_rtspsrc_change_state (GstElement * element, GstStateChange transition)
 
   switch (transition) {
     case GST_STATE_CHANGE_PAUSED_TO_PLAYING:
-      gst_rtspsrc_loop_send_cmd (rtspsrc, CMD_PLAY);
+      gst_rtspsrc_loop_send_cmd (rtspsrc, CMD_PLAY, 0);
       break;
     case GST_STATE_CHANGE_PLAYING_TO_PAUSED:
       /* send pause request and keep the idle task around */
-      gst_rtspsrc_loop_send_cmd (rtspsrc, CMD_PAUSE);
+      gst_rtspsrc_loop_send_cmd (rtspsrc, CMD_PAUSE, CMD_LOOP);
       ret = GST_STATE_CHANGE_NO_PREROLL;
       break;
     case GST_STATE_CHANGE_READY_TO_PAUSED:
       ret = GST_STATE_CHANGE_NO_PREROLL;
       break;
     case GST_STATE_CHANGE_PAUSED_TO_READY:
-      gst_rtspsrc_loop_send_cmd (rtspsrc, CMD_CLOSE);
+      gst_rtspsrc_loop_send_cmd (rtspsrc, CMD_CLOSE, CMD_PAUSE);
       break;
     case GST_STATE_CHANGE_READY_TO_NULL:
       gst_rtspsrc_stop (rtspsrc);
@@ -6654,7 +6753,7 @@ gst_rtspsrc_send_event (GstElement * element, GstEvent * event)
   rtspsrc = GST_RTSPSRC (element);
 
   if (GST_EVENT_IS_DOWNSTREAM (event)) {
-    res = gst_rtspsrc_push_event (rtspsrc, event, TRUE);
+    res = gst_rtspsrc_push_event (rtspsrc, event);
   } else {
     res = GST_ELEMENT_CLASS (parent_class)->send_event (element, event);
   }