1 files changed, 217 insertions, 64 deletions

diff --git a/gst/rtsp/gstrtspsrc.c b/gst/rtsp/gstrtspsrc.c
index ee383a45..a99a102a 100644
--- a/gst/rtsp/gstrtspsrc.c
+++ b/gst/rtsp/gstrtspsrc.c
@@ -290,8 +290,6 @@ gst_rtspsrc_init (GstRTSPSrc * src, GstRTSPSrcClass * g_class)
   src->stream_rec_lock = g_new (GStaticRecMutex, 1);
   g_static_rec_mutex_init (src->stream_rec_lock);
 
-  src->loop_cond = g_cond_new ();
-
   src->location = g_strdup (DEFAULT_LOCATION);
   src->url = NULL;
 
@@ -316,7 +314,6 @@ gst_rtspsrc_finalize (GObject * object)
 
   g_static_rec_mutex_free (rtspsrc->stream_rec_lock);
   g_free (rtspsrc->stream_rec_lock);
-  g_cond_free (rtspsrc->loop_cond);
   g_free (rtspsrc->location);
   g_free (rtspsrc->req_location);
   g_free (rtspsrc->content_base);
@@ -1280,8 +1277,8 @@ use_no_manager:
           goto no_element;
 
         /* take ownership */
-        gst_object_ref (stream->udpsrc[0]);
-        gst_object_sink (stream->udpsrc[0]);
+        gst_object_ref (stream->udpsrc[1]);
+        gst_object_sink (stream->udpsrc[1]);
 
         gst_element_set_state (stream->udpsrc[1], GST_STATE_READY);
       }
@@ -1350,9 +1347,11 @@ use_no_manager:
       else
         port = transport->server_port.max;
 
-      destination = transport->destination;
+      /* first take the source, then the endpoint to figure out where to send
+       * the RTCP. */
+      destination = transport->source;
       if (destination == NULL)
-        destination = src->addr;
+        destination = src->connection->ip;
 
       GST_DEBUG_OBJECT (src, "configure UDP sink for %s:%d", destination, port);
 
@@ -1375,7 +1374,7 @@ use_no_manager:
       stream->rtcppad = gst_element_get_pad (stream->udpsink, "sink");
 
       /* get session RTCP pad */
-      name = g_strdup_printf ("rtcp_src_%d", stream->id);
+      name = g_strdup_printf ("send_rtcp_src_%d", stream->id);
       pad = gst_element_get_request_pad (src->session, name);
       g_free (name);
 
@@ -1557,10 +1556,43 @@ gst_rtspsrc_push_event (GstRTSPSrc * src, GstEvent * event)
   gst_event_unref (event);
 }
 
+/* FIXME, handle server request, reply with OK, for now */
+static RTSPResult
+gst_rtspsrc_handle_request (GstRTSPSrc * src, RTSPMessage * request)
+{
+  RTSPMessage response = { 0 };
+  RTSPResult res;
+
+  GST_DEBUG_OBJECT (src, "got server request message");
+
+  if (src->debug)
+    rtsp_message_dump (request);
+
+  res = rtsp_message_init_response (&response, RTSP_STS_OK, "OK", request);
+  if (res < 0)
+    goto send_error;
+
+  GST_DEBUG_OBJECT (src, "replying with OK");
+
+  if (src->debug)
+    rtsp_message_dump (&response);
+
+  if ((res = rtsp_connection_send (src->connection, &response, NULL)) < 0)
+    goto send_error;
+
+  return RTSP_OK;
+
+  /* ERRORS */
+send_error:
+  {
+    return res;
+  }
+}
+
 static void
 gst_rtspsrc_loop_interleaved (GstRTSPSrc * src)
 {
-  RTSPMessage response = { 0 };
+  RTSPMessage message = { 0 };
   RTSPResult res;
   gint channel;
   GList *lstream;
@@ -1570,18 +1602,38 @@ gst_rtspsrc_loop_interleaved (GstRTSPSrc * src)
   guint size;
   GstFlowReturn ret = GST_FLOW_OK;
   GstBuffer *buf;
-  gboolean is_rtcp = FALSE;
+  gboolean is_rtcp, have_data;
 
+  have_data = FALSE;
   do {
     GST_DEBUG_OBJECT (src, "doing receive");
-    if ((res = rtsp_connection_receive (src->connection, &response)) < 0)
+
+    if ((res = rtsp_connection_receive (src->connection, &message, NULL)) < 0)
       goto receive_error;
 
-    GST_DEBUG_OBJECT (src, "got packet type %d", response.type);
+    switch (message.type) {
+      case RTSP_MESSAGE_REQUEST:
+        /* server sends us a request message, handle it */
+        if ((res = gst_rtspsrc_handle_request (src, &message)) < 0)
+          goto handle_request_failed;
+        break;
+      case RTSP_MESSAGE_RESPONSE:
+        /* we ignore response messages */
+        GST_DEBUG_OBJECT (src, "ignoring response message");
+        break;
+      case RTSP_MESSAGE_DATA:
+        GST_DEBUG_OBJECT (src, "got data message");
+        have_data = TRUE;
+        break;
+      default:
+        GST_WARNING_OBJECT (src, "ignoring unknown message type %d",
+            message.type);
+        break;
+    }
   }
-  while (response.type != RTSP_MESSAGE_DATA);
+  while (!have_data);
 
-  channel = response.type_data.data.channel;
+  channel = message.type_data.data.channel;
 
   lstream = g_list_find_custom (src->streams, GINT_TO_POINTER (channel),
       (GCompareFunc) find_stream_by_channel);
@@ -1591,13 +1643,16 @@ gst_rtspsrc_loop_interleaved (GstRTSPSrc * src)
   stream = (GstRTSPStream *) lstream->data;
   if (channel == stream->channel[0]) {
     outpad = stream->channelpad[0];
+    is_rtcp = FALSE;
   } else if (channel == stream->channel[1]) {
     outpad = stream->channelpad[1];
     is_rtcp = TRUE;
+  } else {
+    is_rtcp = FALSE;
   }
 
   /* take a look at the body to figure out what we have */
-  rtsp_message_get_body (&response, &data, &size);
+  rtsp_message_get_body (&message, &data, &size);
   if (size < 2)
     goto invalid_length;
 
@@ -1612,8 +1667,8 @@ gst_rtspsrc_loop_interleaved (GstRTSPSrc * src)
   if (outpad == NULL)
     goto unknown_stream;
 
-  /* and chain buffer to internal element */
-  rtsp_message_steal_body (&response, &data, &size);
+  /* take the message body for further processing */
+  rtsp_message_steal_body (&message, &data, &size);
 
   /* strip the trailing \0 */
   size -= 1;
@@ -1624,7 +1679,7 @@ gst_rtspsrc_loop_interleaved (GstRTSPSrc * src)
   GST_BUFFER_SIZE (buf) = size;
 
   /* don't need message anymore */
-  rtsp_message_unset (&response);
+  rtsp_message_unset (&message);
 
   GST_DEBUG_OBJECT (src, "pushing data of size %d on channel %d", size,
       channel);
@@ -1652,7 +1707,7 @@ gst_rtspsrc_loop_interleaved (GstRTSPSrc * src)
 unknown_stream:
   {
     GST_DEBUG_OBJECT (src, "unknown stream on channel %d, ignored", channel);
-    rtsp_message_unset (&response);
+    rtsp_message_unset (&message);
     return;
   }
 receive_error:
@@ -1664,9 +1719,20 @@ receive_error:
     g_free (str);
 
     if (src->debug)
-      rtsp_message_dump (&response);
+      rtsp_message_dump (&message);
 
-    rtsp_message_unset (&response);
+    rtsp_message_unset (&message);
+    ret = GST_FLOW_UNEXPECTED;
+    goto need_pause;
+  }
+handle_request_failed:
+  {
+    gchar *str = rtsp_strresult (res);
+
+    GST_ELEMENT_ERROR (src, RESOURCE, WRITE, (NULL),
+        ("Could not send message. (%s)", str));
+    g_free (str);
+    rtsp_message_unset (&message);
     ret = GST_FLOW_UNEXPECTED;
     goto need_pause;
   }
@@ -1674,7 +1740,7 @@ invalid_length:
   {
     GST_ELEMENT_WARNING (src, RESOURCE, READ, (NULL),
         ("Short message received."));
-    rtsp_message_unset (&response);
+    rtsp_message_unset (&message);
     return;
   }
 need_pause:
@@ -1707,21 +1773,118 @@ need_pause:
   }
 }
 
+/* send server keep-alive */
+static RTSPResult
+gst_rtspsrc_send_keep_alive (GstRTSPSrc * src)
+{
+  RTSPMessage request = { 0 };
+  RTSPMessage response = { 0 };
+  RTSPResult res;
+
+  GST_DEBUG_OBJECT (src, "creating server keep-alive");
+
+  res =
+      rtsp_message_init_request (&request, RTSP_GET_PARAMETER,
+      src->req_location);
+  if (res < 0)
+    goto send_error;
+
+  if (!gst_rtspsrc_send (src, &request, &response, NULL))
+    goto send_error;
+
+  rtsp_message_unset (&request);
+
+  return RTSP_OK;
+
+  /* ERRORS */
+send_error:
+  {
+    gchar *str = rtsp_strresult (res);
+
+    rtsp_message_unset (&request);
+    GST_ELEMENT_WARNING (src, RESOURCE, WRITE, (NULL),
+        ("Could not send keep-alive. (%s)", str));
+    g_free (str);
+    return res;
+  }
+}
+
 static void
 gst_rtspsrc_loop_udp (GstRTSPSrc * src)
 {
   gboolean restart = FALSE;
+  RTSPResult res;
 
   GST_OBJECT_LOCK (src);
   if (src->loop_cmd == CMD_STOP)
     goto stopping;
 
-  /* FIXME, we should continue reading the TCP socket because the server might
-   * send us requests */
   while (src->loop_cmd == CMD_WAIT) {
-    GST_DEBUG_OBJECT (src, "waiting");
-    GST_RTSP_LOOP_WAIT (src);
-    GST_DEBUG_OBJECT (src, "waiting done");
+    GTimeVal tv_timeout;
+    gint timeout;
+
+    GST_OBJECT_UNLOCK (src);
+
+    while (TRUE) {
+      RTSPMessage message = { 0 };
+
+      /* calculate the session timeout. We should send the keep-alive request a
+       * little earlier to compensate for the round trip time to the server. We
+       * subtract 1 second here. */
+      timeout = src->connection->timeout;
+      if (timeout > 1)
+        timeout -= 1;
+
+      /* use the session timeout for receiving data */
+      tv_timeout.tv_sec = timeout;
+      tv_timeout.tv_usec = 0;
+
+      GST_DEBUG_OBJECT (src, "doing receive with timeout %d seconds", timeout);
+
+      /* we should continue reading the TCP socket because the server might
+       * send us requests. When the session timeout expires, we need to send a
+       * keep-alive request to keep the session open. */
+      res = rtsp_connection_receive (src->connection, &message, &tv_timeout);
+
+      switch (res) {
+        case RTSP_OK:
+          GST_DEBUG_OBJECT (src, "we received a server message");
+          break;
+        case RTSP_EINTR:
+          /* we got interrupted, see what we have to do */
+          GST_DEBUG_OBJECT (src, "we got interrupted");
+          /* unset flushing so we can do something else */
+          rtsp_connection_flush (src->connection, FALSE);
+          goto interrupt;
+        case RTSP_ETIMEOUT:
+          /* ignore result, a warning was posted */
+          GST_DEBUG_OBJECT (src, "timout, sending keep-alive");
+          res = gst_rtspsrc_send_keep_alive (src);
+          continue;
+        default:
+          goto receive_error;
+      }
+
+      switch (message.type) {
+        case RTSP_MESSAGE_REQUEST:
+          /* server sends us a request message, handle it */
+          if ((res = gst_rtspsrc_handle_request (src, &message)) < 0)
+            goto handle_request_failed;
+          break;
+        case RTSP_MESSAGE_RESPONSE:
+        case RTSP_MESSAGE_DATA:
+          /* we ignore response and data messages */
+          GST_DEBUG_OBJECT (src, "ignoring message");
+          break;
+        default:
+          GST_WARNING_OBJECT (src, "ignoring unknown message type %d",
+              message.type);
+          break;
+      }
+    }
+  interrupt:
+    GST_OBJECT_LOCK (src);
+    GST_DEBUG_OBJECT (src, "we have command %d", src->loop_cmd);
     if (src->loop_cmd == CMD_STOP)
       goto stopping;
   }
@@ -1762,7 +1925,7 @@ gst_rtspsrc_loop_udp (GstRTSPSrc * src)
   gst_rtspsrc_close (src);
 
   /* see if we have TCP left to try */
-  if (!(src->cur_protocols & RTSP_LOWER_TRANS_TCP))
+  if (!(src->protocols & RTSP_LOWER_TRANS_TCP))
     goto no_protocols;
 
   /* open new connection using tcp */
@@ -1790,6 +1953,24 @@ stopping:
     gst_task_pause (src->task);
     return;
   }
+receive_error:
+  {
+    gchar *str = rtsp_strresult (res);
+
+    GST_ELEMENT_WARNING (src, RESOURCE, READ, (NULL),
+        ("Could not receive message. (%s)", str));
+    g_free (str);
+    return;
+  }
+handle_request_failed:
+  {
+    gchar *str = rtsp_strresult (res);
+
+    GST_ELEMENT_WARNING (src, RESOURCE, READ, (NULL),
+        ("Could not handle server message. (%s)", str));
+    g_free (str);
+    return;
+  }
 no_protocols:
   {
     src->cur_protocols = 0;
@@ -1815,7 +1996,8 @@ gst_rtspsrc_loop_send_cmd (GstRTSPSrc * src, gint cmd)
 {
   GST_OBJECT_LOCK (src);
   src->loop_cmd = cmd;
-  GST_RTSP_LOOP_SIGNAL (src);
+  if (cmd != CMD_WAIT)
+    rtsp_connection_flush (src->connection, TRUE);
   GST_OBJECT_UNLOCK (src);
 }
 
@@ -1828,36 +2010,6 @@ gst_rtspsrc_loop (GstRTSPSrc * src)
     gst_rtspsrc_loop_udp (src);
 }
 
-static RTSPResult
-gst_rtspsrc_handle_request (GstRTSPSrc * src, RTSPMessage * request)
-{
-  RTSPMessage response = { 0 };
-  RTSPResult res;
-
-  res = rtsp_message_init_response (&response, RTSP_STS_OK, "OK", request);
-  if (res < 0)
-    goto send_error;
-
-  if (src->debug)
-    rtsp_message_dump (&response);
-
-  if ((res = rtsp_connection_send (src->connection, &response)) < 0)
-    goto send_error;
-
-  return RTSP_OK;
-
-  /* ERRORS */
-send_error:
-  {
-    gchar *str = rtsp_strresult (res);
-
-    GST_ELEMENT_ERROR (src, RESOURCE, WRITE, (NULL),
-        ("Could not send message. (%s)", str));
-    g_free (str);
-    return res;
-  }
-}
-
 #ifndef GST_DISABLE_GST_DEBUG
 const gchar *
 rtsp_auth_method_to_string (RTSPAuthMethod method)
@@ -2085,11 +2237,11 @@ gst_rtspsrc_try_send (GstRTSPSrc * src, RTSPMessage * request,
   if (src->debug)
     rtsp_message_dump (request);
 
-  if ((res = rtsp_connection_send (src->connection, request)) < 0)
+  if ((res = rtsp_connection_send (src->connection, request, NULL)) < 0)
     goto send_error;
 
 next:
-  if ((res = rtsp_connection_receive (src->connection, response)) < 0)
+  if ((res = rtsp_connection_receive (src->connection, response, NULL)) < 0)
     goto receive_error;
 
   if (src->debug)
@@ -2097,16 +2249,17 @@ next:
 
   switch (response->type) {
     case RTSP_MESSAGE_REQUEST:
-      /* FIXME, handle server request, reply with OK, for now */
       if ((res = gst_rtspsrc_handle_request (src, response)) < 0)
         goto handle_request_failed;
       goto next;
     case RTSP_MESSAGE_RESPONSE:
       /* ok, a response is good */
+      GST_DEBUG_OBJECT (src, "received response message");
       break;
     default:
     case RTSP_MESSAGE_DATA:
       /* get next response */
+      GST_DEBUG_OBJECT (src, "ignoring data response message");
       goto next;
   }
 
@@ -2587,7 +2740,7 @@ gst_rtspsrc_open (GstRTSPSrc * src)
 
   /* connect */
   GST_DEBUG_OBJECT (src, "connecting (%s)...", src->req_location);
-  if ((res = rtsp_connection_connect (src->connection)) < 0)
+  if ((res = rtsp_connection_connect (src->connection, NULL)) < 0)
     goto could_not_connect;
 
   /* create OPTIONS */
@@ -3116,8 +3269,8 @@ gst_rtspsrc_handle_message (GstBin * bin, GstMessage * message)
       gst_message_unref (message);
       break;
 
-      /* fatal our not our message, forward */
     forward:
+      /* fatal but not our message, forward */
       GST_BIN_CLASS (parent_class)->handle_message (bin, message);
       break;
     }