rtspsink: Add rtspclientsink element

Add an rtspclientsink element that accepts streams for which
there is a registered payloader and sends them to
an RTSP server using RECORD.

Sending is synchronised to the pipeline clock. Payload-types
are automatically selected. The 'new-payloader' signal is fired
for custom configuration of payloaders when they are created.

Can now stream a movie like this:

receiver:
  ./test-record "( decodebin name=depay0 ! videoconvert ! autovideosink \
       decodebin name=depay1 ! audioconvert ! autoaudiosink )"
sender:
  gst-launch-1.0 filesrc location=file-with-aac-and-h264.mp4 ! qtdemux name=d ! \
       queue ! aacparse ! rtspclientsink location=rtsp://127.0.0.1:8554/test name=s \

https://bugzilla.gnome.org/show_bug.cgi?id=758180

.gitignore

@@ -62,6 +62,7 @@ stamp-h.in
 /tests/check/gst/stream
 /tests/check/gst/threadpool
 /tests/check/gst/token
+/tests/check/gst/rtspclientsink
 /tests/check/test-registry.reg
 /tests/test-reuse
 

configure.ac

@@ -170,6 +170,8 @@ AC_MSG_NOTICE(Using GStreamer Core Plugins in $GST_PLUGINS_DIR)
 
 AG_GST_CHECK_GST_BASE($GST_API_VERSION, [$GST_REQ], [yes])
 
+AG_GST_CHECK_GST_NET($GST_API_VERSION, [$GST_REQ], yes)
+
 AG_GST_CHECK_GST_PLUGINS_BASE($GST_API_VERSION, [$GSTPB_REQ], [yes])
 GSTPB_PLUGINS_DIR=`$PKG_CONFIG gstreamer-plugins-base-$GST_API_VERSION --variable pluginsdir`
 AC_SUBST(GSTPB_PLUGINS_DIR)
@@ -218,6 +220,31 @@ AG_GST_SET_PACKAGE_RELEASE_DATETIME_WITH_NANO([$PACKAGE_VERSION_NANO],
   ["${srcdir}/gst-rtsp-server.doap"],
   [$PACKAGE_VERSION_MAJOR.$PACKAGE_VERSION_MINOR.$PACKAGE_VERSION_MICRO])
 
+dnl build static plugins or not
+AC_MSG_CHECKING([whether to build static plugins or not])
+AC_ARG_ENABLE(
+  static-plugins,
+  AC_HELP_STRING(
+    [--enable-static-plugins],
+    [build static plugins @<:@default=no@:>@]),
+  [AS_CASE(
+    [$enableval], [no], [], [yes], [],
+    [AC_MSG_ERROR([bad value "$enableval" for --enable-static-plugins])])],
+  [enable_static_plugins=no])
+AC_MSG_RESULT([$enable_static_plugins])
+if test "x$enable_static_plugins" = xyes; then
+  AC_DEFINE(GST_PLUGIN_BUILD_STATIC, 1,
+    [Define if static plugins should be built])
+  GST_PLUGIN_LIBTOOLFLAGS=""
+else
+  GST_PLUGIN_LIBTOOLFLAGS="--tag=disable-static"
+fi
+AC_SUBST(GST_PLUGIN_LIBTOOLFLAGS)
+AM_CONDITIONAL(GST_PLUGIN_BUILD_STATIC, test "x$enable_static_plugins" = "xyes")
+
+GST_PLUGIN_LDFLAGS="-module -avoid-version -export-symbols-regex '^[_]*gst_plugin_.*' $GST_ALL_LDFLAGS"
+AC_SUBST(GST_PLUGIN_LDFLAGS)
+
 # set by AG_GST_PARSE_SUBSYSTEM_DISABLES above
 dnl make sure it doesn't complain about unused variables if debugging is disabled
 NO_WARNINGS=""
@@ -324,6 +351,7 @@ common/Makefile
 common/m4/Makefile
 gst/Makefile
 gst/rtsp-server/Makefile
+gst/rtsp-sink/Makefile
 examples/Makefile
 tests/Makefile
 tests/check/Makefile

gst/Makefile.am

@@ -1 +1 @@
-SUBDIRS = rtsp-server
+SUBDIRS = rtsp-server rtsp-sink

gst/rtsp-sink/Makefile.am

@@ -0,0 +1,18 @@
+plugin_LTLIBRARIES = libgstrtspclientsink.la
+
+libgstrtspclientsink_la_SOURCES = gstrtspclientsink.c plugin.c
+
+libgstrtspclientsink_la_CFLAGS = -I$(top_srcdir) $(GST_PLUGINS_BASE_CFLAGS) $(GST_CFLAGS) $(GIO_CFLAGS)
+
+# FIXME: Hack to avoid having to add GETTEXT_PACKAGE to gst-rtsp
+libgstrtspclientsink_la_CFLAGS += -D"GETTEXT_PACKAGE=gst-rtsp-server-1.0"
+
+libgstrtspclientsink_la_LIBADD = $(top_builddir)/gst/rtsp-server/libgstrtspserver-@GST_API_VERSION@.la  \
+			$(GST_PLUGINS_BASE_LIBS) $(GST_BASE_LIBS) \
+		       -lgstrtp-@GST_API_VERSION@ -lgstrtsp-@GST_API_VERSION@ \
+		       -lgstsdp-@GST_API_VERSION@ $(GST_NET_LIBS) $(GST_LIBS) \
+			$(GIO_LIBS)
+libgstrtspclientsink_la_LDFLAGS = $(GST_PLUGIN_LDFLAGS)
+libgstrtspclientsink_la_LIBTOOLFLAGS = $(GST_PLUGIN_LIBTOOLFLAGS)
+
+noinst_HEADERS = gstrtspclientsink.h

gst/rtsp-sink/gstrtspclientsink.h

@@ -0,0 +1,244 @@
+/* GStreamer
+ * Copyright (C) <1999> Erik Walthinsen <omega@cse.ogi.edu>
+ *               <2006> Wim Taymans <wim@fluendo.com>
+ *               <2015> Jan Schmidt <jan at centricular 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.
+ */
+/*
+ * Unless otherwise indicated, Source Code is licensed under MIT license.
+ * See further explanation attached in License Statement (distributed in the file
+ * LICENSE).
+ *
+ * Permission is hereby granted, free of charge, to any person obtaining a copy of
+ * this software and associated documentation files (the "Software"), to deal in
+ * the Software without restriction, including without limitation the rights to
+ * use, copy, modify, merge, publish, distribute, sublicense, and/or sell copies
+ * of the Software, and to permit persons to whom the Software is furnished to do
+ * so, subject to the following conditions:
+ *
+ * The above copyright notice and this permission notice shall be included in all
+ * copies or substantial portions of the Software.
+ *
+ * THE SOFTWARE IS PROVIDED "AS IS", WITHOUT WARRANTY OF ANY KIND, EXPRESS OR
+ * IMPLIED, INCLUDING BUT NOT LIMITED TO THE WARRANTIES OF MERCHANTABILITY,
+ * FITNESS FOR A PARTICULAR PURPOSE AND NONINFRINGEMENT. IN NO EVENT SHALL THE
+ * AUTHORS OR COPYRIGHT HOLDERS BE LIABLE FOR ANY CLAIM, DAMAGES OR OTHER
+ * LIABILITY, WHETHER IN AN ACTION OF CONTRACT, TORT OR OTHERWISE, ARISING FROM,
+ * OUT OF OR IN CONNECTION WITH THE SOFTWARE OR THE USE OR OTHER DEALINGS IN THE
+ * SOFTWARE.
+ */
+
+#ifndef __GST_RTSP_CLIENT_SINK_H__
+#define __GST_RTSP_CLIENT_SINK_H__
+
+#include <gst/gst.h>
+
+G_BEGIN_DECLS
+
+#include <gst/rtsp-server/rtsp-stream.h>
+#include <gst/rtsp/rtsp.h>
+#include <gio/gio.h>
+
+#define GST_TYPE_RTSP_CLIENT_SINK \
+  (gst_rtsp_client_sink_get_type())
+#define GST_RTSP_CLIENT_SINK(obj) \
+  (G_TYPE_CHECK_INSTANCE_CAST((obj),GST_TYPE_RTSP_CLIENT_SINK,GstRTSPClientSink))
+#define GST_RTSP_CLIENT_SINK_CLASS(klass) \
+  (G_TYPE_CHECK_CLASS_CAST((klass),GST_TYPE_RTSP_CLIENT_SINK,GstRTSPClientSinkClass))
+#define GST_IS_RTSP_CLIENT_SINK(obj) \
+  (G_TYPE_CHECK_INSTANCE_TYPE((obj),GST_TYPE_RTSP_CLIENT_SINK))
+#define GST_IS_RTSP_CLIENT_SINK_CLASS(klass) \
+  (G_TYPE_CHECK_CLASS_TYPE((klass),GST_TYPE_RTSP_CLIENT_SINK))
+#define GST_RTSP_CLIENT_SINK_CAST(obj) \
+  ((GstRTSPClientSink *)(obj))
+
+typedef struct _GstRTSPClientSink GstRTSPClientSink;
+typedef struct _GstRTSPClientSinkClass GstRTSPClientSinkClass;
+
+#define GST_RTSP_STATE_GET_LOCK(rtsp)    (&GST_RTSP_CLIENT_SINK_CAST(rtsp)->state_rec_lock)
+#define GST_RTSP_STATE_LOCK(rtsp)        (g_rec_mutex_lock (GST_RTSP_STATE_GET_LOCK(rtsp)))
+#define GST_RTSP_STATE_UNLOCK(rtsp)      (g_rec_mutex_unlock (GST_RTSP_STATE_GET_LOCK(rtsp)))
+
+#define GST_RTSP_STREAM_GET_LOCK(rtsp)   (&GST_RTSP_CLIENT_SINK_CAST(rtsp)->stream_rec_lock)
+#define GST_RTSP_STREAM_LOCK(rtsp)       (g_rec_mutex_lock (GST_RTSP_STREAM_GET_LOCK(rtsp)))
+#define GST_RTSP_STREAM_UNLOCK(rtsp)     (g_rec_mutex_unlock (GST_RTSP_STREAM_GET_LOCK(rtsp)))
+
+typedef struct _GstRTSPConnInfo GstRTSPConnInfo;
+
+struct _GstRTSPConnInfo {
+  gchar              *location;
+  GstRTSPUrl         *url;
+  gchar              *url_str;
+  GstRTSPConnection  *connection;
+  gboolean            connected;
+  gboolean            flushing;
+};
+
+typedef struct _GstRTSPStreamInfo GstRTSPStreamInfo;
+typedef struct _GstRTSPStreamContext GstRTSPStreamContext;
+
+struct _GstRTSPStreamContext {
+  GstRTSPClientSink *parent;
+
+  guint index;
+  /* Index of the SDPMedia in the stored SDP */
+  guint sdp_index;
+
+  GstElement *payloader;
+  guint payloader_block_id;
+  gboolean prerolled;
+
+  /* Stream management object */
+  GstRTSPStream *stream;
+  gboolean joined;
+
+  /* Secure profile key mgmt */
+  GstCaps      *srtcpparams;
+
+  /* per stream connection */
+  GstRTSPConnInfo  conninfo;
+  /* For interleaved mode */
+  guint8        channel[2];
+
+  GstRTSPStreamTransport *stream_transport;
+};
+
+/**
+ * GstRTSPNatMethod:
+ * @GST_RTSP_NAT_NONE: none
+ * @GST_RTSP_NAT_DUMMY: send dummy packets
+ *
+ * Different methods for trying to traverse firewalls.
+ */
+typedef enum
+{
+  GST_RTSP_NAT_NONE,
+  GST_RTSP_NAT_DUMMY
+} GstRTSPNatMethod;
+
+struct _GstRTSPClientSink {
+  GstBin           parent;
+
+  /* task and mutex for interleaved mode */
+  gboolean         interleaved;
+  GstTask         *task;
+  GRecMutex        stream_rec_lock;
+  GstSegment       segment;
+  gint             free_channel;
+
+  /* UDP mode loop */
+  gint             pending_cmd;
+  gint             busy_cmd;
+  gboolean         ignore_timeout;
+  gboolean         open_error;
+
+  /* mutex for protecting state changes */
+  GRecMutex        state_rec_lock;
+
+  GstSDPMessage    *uri_sdp;
+  gboolean         from_sdp;
+
+  /* properties */
+  GstRTSPLowerTrans protocols;
+  gboolean          debug;
+  guint             retry;
+  guint64           udp_timeout;
+  GTimeVal          tcp_timeout;
+  GTimeVal         *ptcp_timeout;
+  guint             latency;
+  gboolean          do_rtsp_keep_alive;
+  gchar            *proxy_host;
+  guint             proxy_port;
+  gchar            *proxy_user;        /* from url or property */
+  gchar            *proxy_passwd;      /* from url or property */
+  gchar            *prop_proxy_id;     /* set via property */
+  gchar            *prop_proxy_pw;     /* set via property */
+  guint             rtp_blocksize;
+  gchar            *user_id;
+  gchar            *user_pw;
+  GstRTSPRange      client_port_range;
+  gint              udp_buffer_size;
+  gboolean          udp_reconnect;
+  gchar            *multi_iface;
+  gboolean          ntp_sync;
+  gboolean          use_pipeline_clock;
+  GstStructure     *sdes;
+  GTlsCertificateFlags tls_validation_flags;
+  GTlsDatabase     *tls_database;
+  GTlsInteraction  *tls_interaction;
+  gint              ntp_time_source;
+  gchar            *user_agent;
+
+  /* state */
+  GstRTSPState       state;
+  gchar             *content_base;
+  GstRTSPLowerTrans  cur_protocols;
+  gboolean           tried_url_auth;
+  gchar             *addr;
+  gboolean           need_redirect;
+  GstRTSPTimeRange  *range;
+  gchar             *control;
+  guint              next_port_num;
+  GstClock          *provided_clock;
+
+  /* supported methods */
+  gint               methods;
+
+  /* session management */
+  GstRTSPConnInfo  conninfo;
+
+  /* Everything goes in an internal
+   * locked-state bin */
+  GstBin          *internal_bin;
+  /* Set to true when internal bin state
+   * >= PAUSED */
+  gboolean        prerolled;
+
+  /* TRUE if we posted async-start */
+  gboolean         in_async;
+
+  /* TRUE when stream info has been collected */
+  gboolean         streams_collected;
+
+  guint            next_pad_id;
+  gint             next_dyn_pt;
+
+  GstElement      *rtpbin;
+
+  GList           *contexts;
+  GstSDPMessage   cursdp;
+
+  GMutex          send_lock;
+
+  GMutex          preroll_lock;
+  GCond           preroll_cond;
+
+  GstClockTime    rtx_time;
+
+  GstRTSPProfile profiles;
+  gchar          *server_ip;
+};
+
+struct _GstRTSPClientSinkClass {
+  GstBinClass parent_class;
+};
+
+GType gst_rtsp_client_sink_get_type(void);
+
+G_END_DECLS
+
+#endif /* __GST_RTSP_CLIENT_SINK_H__ */

gst/rtsp-sink/plugin.c

@@ -0,0 +1,26 @@
+#ifdef HAVE_CONFIG_H
+#include "config.h"
+#endif
+
+#include "gstrtspclientsink.h"
+
+static gboolean
+plugin_init (GstPlugin * plugin)
+{
+#ifdef ENABLE_NLS
+  bindtextdomain (GETTEXT_PACKAGE, LOCALEDIR);
+  bind_textdomain_codeset (GETTEXT_PACKAGE, "UTF-8");
+#endif /* ENABLE_NLS */
+
+  if (!gst_element_register (plugin, "rtspclientsink", GST_RANK_NONE,
+          GST_TYPE_RTSP_CLIENT_SINK))
+    return FALSE;
+
+  return TRUE;
+}
+
+GST_PLUGIN_DEFINE (GST_VERSION_MAJOR,
+    GST_VERSION_MINOR,
+    rtspclientsink,
+    "RTSP client sink element",
+    plugin_init, VERSION, GST_LICENSE, GST_PACKAGE_NAME, GST_PACKAGE_ORIGIN)

tests/check/Makefile.am

@@ -11,8 +11,8 @@ AM_TESTS_ENVIRONMENT = \
         GST_STATE_IGNORE_ELEMENTS="$(STATE_IGNORE_ELEMENTS)"	\
 	$(REGISTRY_ENVIRONMENT)					\
 	GST_PLUGIN_SYSTEM_PATH_1_0=					\
-	GST_PLUGIN_PATH_1_0=$(GST_PLUGINS_DIR):$(GSTPB_PLUGINS_DIR):$(GSTPG_PLUGINS_DIR):$(GSTPD_PLUGINS_DIR)	\
-	GST_PLUGIN_LOADING_WHITELIST="gstreamer:gst-plugins-base:gst-plugins-good:gst-plugins-bad"
+	GST_PLUGIN_PATH_1_0=$(GST_PLUGINS_DIR):$(GSTPB_PLUGINS_DIR):$(GSTPG_PLUGINS_DIR):$(GSTPD_PLUGINS_DIR):$(top_builddir)/gst	\
+	GST_PLUGIN_LOADING_WHITELIST="gstreamer:gst-plugins-base:gst-plugins-good:gst-plugins-bad:gst-rtsp-server"
 
 
 # ths core dumps of some machines have PIDs appended
@@ -37,7 +37,8 @@ check_PROGRAMS = \
 	gst/permissions \
 	gst/token \
 	gst/sessionmedia \
-	gst/sessionpool
+	gst/sessionpool \
+	gst/rtspclientsink
 
 # these tests don't even pass
 noinst_PROGRAMS =

tests/check/gst/rtspclientsink.c

@@ -0,0 +1,221 @@
+/* GStreamer unit test for rtspclientsink
+ * Copyright (C) 2012 Axis Communications <dev-gstreamer at axis dot com>
+ *   @author David Svensson Fors <davidsf at axis dot com>
+ * Copyright (C) 2015 Centricular Ltd
+ *   @author Tim-Philipp Müller <tim@centricular.com>
+ *   @author Jan Schmidt <jan@centricular.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 <gst/sdp/gstsdpmessage.h>
+#include <gst/rtp/gstrtpbuffer.h>
+#include <gst/rtp/gstrtcpbuffer.h>
+
+#include <stdio.h>
+#include <netinet/in.h>
+
+#include "rtsp-server.h"
+
+#define TEST_MOUNT_POINT  "/test"
+
+/* tested rtsp server */
+static GstRTSPServer *server = NULL;
+
+/* tcp port that the test server listens for rtsp requests on */
+static gint test_port = 0;
+
+/* id of the server's source within the GMainContext */
+static guint source_id;
+
+/* iterate the default main context until there are no events to dispatch */
+static void
+iterate (void)
+{
+  while (g_main_context_iteration (NULL, FALSE)) {
+    GST_DEBUG ("iteration");
+  }
+}
+
+/* start the testing rtsp server for RECORD mode */
+static GstRTSPMediaFactory *
+start_record_server (const gchar * launch_line)
+{
+  GstRTSPMediaFactory *factory;
+  GstRTSPMountPoints *mounts;
+  gchar *service;
+
+  mounts = gst_rtsp_server_get_mount_points (server);
+
+  factory = gst_rtsp_media_factory_new ();
+
+  gst_rtsp_media_factory_set_transport_mode (factory,
+      GST_RTSP_TRANSPORT_MODE_RECORD);
+  gst_rtsp_media_factory_set_launch (factory, launch_line);
+  gst_rtsp_mount_points_add_factory (mounts, TEST_MOUNT_POINT, factory);
+  g_object_unref (mounts);
+
+  /* set port to any */
+  gst_rtsp_server_set_service (server, "0");
+
+  /* attach to default main context */
+  source_id = gst_rtsp_server_attach (server, NULL);
+  fail_if (source_id == 0);
+
+  /* get port */
+  service = gst_rtsp_server_get_service (server);
+  test_port = atoi (service);
+  fail_unless (test_port != 0);
+  g_free (service);
+
+  GST_DEBUG ("rtsp server listening on port %d", test_port);
+  return factory;
+}
+
+/* stop the tested rtsp server */
+static void
+stop_server (void)
+{
+  g_source_remove (source_id);
+  source_id = 0;
+
+  GST_DEBUG ("rtsp server stopped");
+}
+
+/* fixture setup function */
+static void
+setup (void)
+{
+  server = gst_rtsp_server_new ();
+}
+
+/* fixture clean-up function */
+static void
+teardown (void)
+{
+  if (server) {
+    g_object_unref (server);
+    server = NULL;
+  }
+  test_port = 0;
+}
+
+/* create an rtsp connection to the server on test_port */
+static gchar *
+get_server_uri (gint port, const gchar * mount_point)
+{
+  gchar *address;
+  gchar *uri_string;
+  GstRTSPUrl *url = NULL;
+
+  address = gst_rtsp_server_get_address (server);
+  uri_string = g_strdup_printf ("rtsp://%s:%d%s", address, port, mount_point);
+  g_free (address);
+
+  fail_unless (gst_rtsp_url_parse (uri_string, &url) == GST_RTSP_OK);
+  gst_rtsp_url_free (url);
+
+  return uri_string;
+}
+
+static void
+media_constructed_cb (GstRTSPMediaFactory * mfactory, GstRTSPMedia * media,
+    gpointer user_data)
+{
+  GstElement **p_sink = user_data;
+  GstElement *bin;
+
+  bin = gst_rtsp_media_get_element (media);
+  *p_sink = gst_bin_get_by_name (GST_BIN (bin), "sink");
+  GST_INFO ("media constructed!: %" GST_PTR_FORMAT, *p_sink);
+}
+
+#define AUDIO_PIPELINE "audiotestsrc num-buffers=%d ! " \
+  "audio/x-raw,rate=8000 ! alawenc ! rtspclientsink name=sink location=%s"
+#define RECORD_N_BUFS 10
+
+GST_START_TEST (test_record)
+{
+  GstRTSPMediaFactory *mfactory;
+  GstElement *server_sink = NULL;
+  gint i;
+
+  mfactory =
+      start_record_server ("( rtppcmadepay name=depay0 ! appsink name=sink )");
+
+  g_signal_connect (mfactory, "media-constructed",
+      G_CALLBACK (media_constructed_cb), &server_sink);
+
+  /* Create an rtspclientsink and send some data */
+  {
+    gchar *uri = get_server_uri (test_port, TEST_MOUNT_POINT);
+    gchar *pipe_str = g_strdup_printf (AUDIO_PIPELINE,
+        RECORD_N_BUFS, uri);
+    GstMessage *msg;
+    GstElement *pipeline;
+    GstBus *bus;
+
+    pipeline = gst_parse_launch (pipe_str, NULL);
+    fail_unless (pipeline != NULL);
+
+    bus = gst_element_get_bus (pipeline);
+    fail_if (bus == NULL);
+
+    gst_element_set_state (pipeline, GST_STATE_PLAYING);
+
+    msg = gst_bus_poll (bus, GST_MESSAGE_EOS | GST_MESSAGE_ERROR, -1);
+    fail_if (GST_MESSAGE_TYPE (msg) != GST_MESSAGE_EOS);
+    gst_message_unref (msg);
+
+    gst_element_set_state (pipeline, GST_STATE_NULL);
+    gst_object_unref (pipeline);
+  }
+
+  iterate ();
+
+  /* check received data (we assume every buffer created by audiotestsrc and
+   * subsequently encoded by mulawenc results in exactly one RTP packet) */
+  for (i = 0; i < RECORD_N_BUFS; ++i) {
+    GstSample *sample = NULL;
+
+    g_signal_emit_by_name (G_OBJECT (server_sink), "pull-sample", &sample);
+    GST_INFO ("%2d recv sample: %p", i, sample);
+    if (sample)
+      gst_sample_unref (sample);
+  }
+
+  /* clean up and iterate so the clean-up can finish */
+  stop_server ();
+  iterate ();
+}
+
+GST_END_TEST;
+
+static Suite *
+rtspclientsink_suite (void)
+{
+  Suite *s = suite_create ("rtspclientsink");
+  TCase *tc = tcase_create ("general");
+
+  suite_add_tcase (s, tc);
+  tcase_add_checked_fixture (tc, setup, teardown);
+  tcase_set_timeout (tc, 120);
+  tcase_add_test (tc, test_record);
+  return s;
+}
+
+GST_CHECK_MAIN (rtspclientsink);