stream: Add functions to get rtp and rtcp sockets

Fixes https://bugzilla.gnome.org/show_bug.cgi?id=710100

docs/libs/gst-rtsp-server-sections.txt

@@ -522,6 +522,9 @@ gst_rtsp_stream_recv_rtp
 
 gst_rtsp_stream_add_transport
 gst_rtsp_stream_remove_transport
+
+gst_rtsp_stream_get_rtp_socket
+gst_rtsp_stream_get_rtcp_socket
 <SUBSECTION Standard>
 GST_RTSP_STREAM_CAST
 GST_RTSP_STREAM_CLASS_CAST

gst/rtsp-server/rtsp-stream.c

@@ -1996,3 +1996,71 @@ gst_rtsp_stream_remove_transport (GstRTSPStream * stream,
 
   return res;
 }
+
+/**
+ * gst_rtsp_stream_get_rtp_socket:
+ * @stream: a #GstRTSPStream
+ * @family: the socket family
+ *
+ * Get the RTP socket from @stream for a @family.
+ *
+ * @stream must be joined to a bin.
+ *
+ * Returns: the RTP socket or %NULL if no socket could be allocated for @family.
+ *     Unref after usage
+ */
+GSocket *
+gst_rtsp_stream_get_rtp_socket (GstRTSPStream * stream, GSocketFamily family)
+{
+  GstRTSPStreamPrivate *priv = GST_RTSP_STREAM_GET_PRIVATE (stream);
+  GSocket *socket;
+  gchar *name;
+
+  g_return_val_if_fail (GST_IS_RTSP_STREAM (stream), NULL);
+  g_return_val_if_fail (family == G_SOCKET_FAMILY_IPV4 ||
+      family == G_SOCKET_FAMILY_IPV6, NULL);
+  g_return_val_if_fail (priv->udpsink[0], NULL);
+
+  if (family == G_SOCKET_FAMILY_IPV6)
+    name = "socket-v6";
+  else
+    name = "socket";
+
+  g_object_get (priv->udpsink[0], name, &socket, NULL);
+
+  return socket;
+}
+
+/**
+ * gst_rtsp_stream_get_rtcp_socket:
+ * @stream: a #GstRTSPStream
+ * @family: the socket family
+ *
+ * Get the RTCP socket from @stream for a @family.
+ *
+ * @stream must be joined to a bin.
+ *
+ * Returns: the RTCP socket or %NULL if no socket could be allocated for
+ *     @family. Unref after usage
+ */
+GSocket *
+gst_rtsp_stream_get_rtcp_socket (GstRTSPStream * stream, GSocketFamily family)
+{
+  GstRTSPStreamPrivate *priv = GST_RTSP_STREAM_GET_PRIVATE (stream);
+  GSocket *socket;
+  gchar *name;
+
+  g_return_val_if_fail (GST_IS_RTSP_STREAM (stream), NULL);
+  g_return_val_if_fail (family == G_SOCKET_FAMILY_IPV4 ||
+      family == G_SOCKET_FAMILY_IPV6, NULL);
+  g_return_val_if_fail (priv->udpsink[1], NULL);
+
+  if (family == G_SOCKET_FAMILY_IPV6)
+    name = "socket-v6";
+  else
+    name = "socket";
+
+  g_object_get (priv->udpsink[1], name, &socket, NULL);
+
+  return socket;
+}

gst/rtsp-server/rtsp-stream.h

@@ -122,6 +122,11 @@ gboolean          gst_rtsp_stream_add_transport    (GstRTSPStream *stream,
 gboolean          gst_rtsp_stream_remove_transport (GstRTSPStream *stream,
                                                     GstRTSPStreamTransport *trans);
 
+GSocket *         gst_rtsp_stream_get_rtp_socket   (GstRTSPStream *stream,
+                                                    GSocketFamily family);
+GSocket *         gst_rtsp_stream_get_rtcp_socket  (GstRTSPStream *stream,
+                                                    GSocketFamily family);
+
 G_END_DECLS
 
 #endif /* __GST_RTSP_STREAM_H__ */

tests/check/Makefile.am

@@ -31,6 +31,7 @@ check_PROGRAMS = \
 	gst/mountpoints \
 	gst/mediafactory \
 	gst/media \
+	gst/stream \
 	gst/addresspool \
 	gst/threadpool \
 	gst/permissions \

tests/check/gst/stream.c

@@ -0,0 +1,107 @@
+/* GStreamer
+ * Copyright (C) 2013 Axis Communications AB <dev-gstreamer at axis dot com>
+ *
+ * This library is free software; you can redistribute it and/or
+ * modify it under the terms of the GNU Library General Public
+ * License as published by the Free Software Foundation; either
+ * version 2 of the License, or (at your option) any later version.
+ *
+ * This library is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the GNU
+ * Library General Public License for more details.
+ *
+ * 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., 51 Franklin St, Fifth Floor,
+ * Boston, MA 02110-1301, USA.
+ */
+
+#include <gst/check/gstcheck.h>
+
+#include <rtsp-stream.h>
+
+GST_START_TEST (test_get_sockets)
+{
+  GstPad *srcpad;
+  GstElement *pay;
+  GstRTSPStream *stream;
+  GstBin *bin;
+  GstElement *rtpbin;
+  GSocket *socket;
+  gboolean have_ipv4;
+  gboolean have_ipv6;
+
+  srcpad = gst_pad_new ("testsrcpad", GST_PAD_SRC);
+  fail_unless (srcpad != NULL);
+  gst_pad_set_active (srcpad, TRUE);
+  pay = gst_element_factory_make ("rtpgstpay", "testpayloader");
+  fail_unless (pay != NULL);
+  stream = gst_rtsp_stream_new (0, pay, srcpad);
+  fail_unless (stream != NULL);
+  gst_object_unref (pay);
+  gst_object_unref (srcpad);
+  rtpbin = gst_element_factory_make ("rtpbin", "testrtpbin");
+  fail_unless (rtpbin != NULL);
+  bin = GST_BIN (gst_bin_new ("testbin"));
+  fail_unless (bin != NULL);
+  fail_unless (gst_bin_add (bin, rtpbin));
+
+  fail_unless (gst_rtsp_stream_join_bin (stream, bin, rtpbin, GST_STATE_NULL));
+
+  socket = gst_rtsp_stream_get_rtp_socket (stream, G_SOCKET_FAMILY_IPV4);
+  have_ipv4 = (socket != NULL);
+  if (have_ipv4) {
+    fail_unless (g_socket_get_fd (socket) >= 0);
+    g_object_unref (socket);
+  }
+
+  socket = gst_rtsp_stream_get_rtcp_socket (stream, G_SOCKET_FAMILY_IPV4);
+  if (have_ipv4) {
+    fail_unless (socket != NULL);
+    fail_unless (g_socket_get_fd (socket) >= 0);
+    g_object_unref (socket);
+  } else {
+    fail_unless (socket == NULL);
+  }
+
+  socket = gst_rtsp_stream_get_rtp_socket (stream, G_SOCKET_FAMILY_IPV6);
+  have_ipv6 = (socket != NULL);
+  if (have_ipv6) {
+    fail_unless (g_socket_get_fd (socket) >= 0);
+    g_object_unref (socket);
+  }
+
+  socket = gst_rtsp_stream_get_rtcp_socket (stream, G_SOCKET_FAMILY_IPV6);
+  if (have_ipv6) {
+    fail_unless (socket != NULL);
+    fail_unless (g_socket_get_fd (socket) >= 0);
+    g_object_unref (socket);
+  } else {
+    fail_unless (socket == NULL);
+  }
+
+  /* check that at least one family is available */
+  fail_unless (have_ipv4 || have_ipv6);
+
+  fail_unless (gst_rtsp_stream_leave_bin (stream, bin, rtpbin));
+
+  gst_object_unref (bin);
+  gst_object_unref (stream);
+}
+
+GST_END_TEST;
+
+static Suite *
+rtspstream_suite (void)
+{
+  Suite *s = suite_create ("rtspstream");
+  TCase *tc = tcase_create ("general");
+
+  suite_add_tcase (s, tc);
+  tcase_add_test (tc, test_get_sockets);
+
+  return s;
+}
+
+GST_CHECK_MAIN (rtspstream);