examples: Add a simple example of network synch for live streams.

An example server and client that works for synchronising live streams
only - as it can't support pause/play.

4 files changed, 298 insertions, 1 deletions

diff --git a/examples/.gitignore b/examples/.gitignore
index 978258f..b8c175a 100644
--- a/examples/.gitignore
+++ b/examples/.gitignore
@@ -10,3 +10,5 @@ test-video
 test-video-rtx
 test-uri
 test-auth
+test-netclock
+test-netclock-client
diff --git a/examples/Makefile.am b/examples/Makefile.am
index ef33adf..0b1495d 100644
--- a/examples/Makefile.am
+++ b/examples/Makefile.am
@@ -1,7 +1,8 @@
 noinst_PROGRAMS = test-video test-ogg test-mp4 test-readme \
 		  test-launch test-sdp test-uri test-auth \
 		  test-multicast test-multicast2 test-appsrc \
-		  test-video-rtx test-record
+		  test-video-rtx test-record \
+		  test-netclock test-netclock-client
 
 #INCLUDES = -I$(top_srcdir) -I$(srcdir)
 
@@ -13,3 +14,11 @@ noinst_PROGRAMS += test-cgroups
 LDADD += $(LIBCGROUP_LIBS)
 endif
 
+test_netclock_LDFLAGS = \
+        $(GST_LIBS) \
+        -lgstnet-@GST_API_VERSION@ \
+        $(top_builddir)/gst/rtsp-server/libgstrtspserver-@GST_API_VERSION@.la
+
+test_netclock_client_LDFLAGS = \
+        $(GST_LIBS) \
+        -lgstnet-@GST_API_VERSION@
diff --git a/examples/test-netclock-client.c b/examples/test-netclock-client.c
new file mode 100644
index 0000000..f91773e
--- /dev/null
+++ b/examples/test-netclock-client.c
@@ -0,0 +1,105 @@
+/* GStreamer
+ * Copyright (C) 2008 Wim Taymans <wim.taymans at gmail.com>
+ * Copyright (C) 2014 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 <stdlib.h>
+
+#include <gst/gst.h>
+#include <gst/net/gstnet.h>
+
+#define PLAYBACK_DELAY_MS 40
+
+static void
+source_created (GstElement * pipe, GParamSpec * pspec)
+{
+  GstElement *source;
+
+  g_object_get (pipe, "source", &source, NULL);
+  g_assert (source != NULL);
+
+  g_object_set (source, "latency", PLAYBACK_DELAY_MS,
+      "use-pipeline-clock", TRUE, "buffer-mode", 1, NULL);
+
+  gst_object_unref (source);
+}
+
+int
+main (int argc, char *argv[])
+{
+  GstClock *net_clock;
+  gchar *server;
+  gint clock_port;
+  GstElement *pipe;
+  GstMessage *msg;
+
+  gst_init (&argc, &argv);
+
+  if (argc < 2) {
+    g_print ("usage: %s rtsp://URI clock-IP clock-PORT\n"
+        "example: %s rtsp://localhost:8554/test 127.0.0.1 8554\n",
+        argv[0], argv[0]);
+    return -1;
+  }
+
+  server = argv[2];
+  clock_port = atoi (argv[3]);
+
+  net_clock = gst_net_client_clock_new ("net_clock", server, clock_port, 0);
+  if (net_clock == NULL) {
+    g_print ("Failed to create net clock client for %s:%d\n",
+        server, clock_port);
+    return 1;
+  }
+
+  /* Wait 0.5 seconds for the clock to stabilise */
+  g_usleep (G_USEC_PER_SEC / 2);
+
+  pipe = gst_element_factory_make ("playbin", NULL);
+  g_object_set (pipe, "uri", argv[1], NULL);
+  g_signal_connect (pipe, "notify::source", (GCallback) source_created, NULL);
+
+  gst_element_set_start_time (pipe, GST_CLOCK_TIME_NONE);
+  gst_element_set_base_time (pipe, 0);
+  gst_pipeline_use_clock (GST_PIPELINE (pipe), net_clock);
+
+  if (gst_element_set_state (pipe,
+          GST_STATE_PLAYING) == GST_STATE_CHANGE_FAILURE) {
+    g_print ("Failed to set state to PLAYING\n");
+    goto exit;
+  };
+
+  msg = gst_bus_timed_pop_filtered (GST_ELEMENT_BUS (pipe),
+      GST_CLOCK_TIME_NONE, GST_MESSAGE_EOS | GST_MESSAGE_ERROR);
+
+  if (GST_MESSAGE_TYPE (msg) == GST_MESSAGE_ERROR) {
+    GError *err = NULL;
+    gchar *debug = NULL;
+    gst_message_parse_error (msg, &err, &debug);
+    g_print ("\nERROR: %s\n%s\n\n", err->message, debug);
+    g_error_free (err);
+    g_free (debug);
+  }
+  gst_message_unref (msg);
+
+exit:
+  gst_element_set_state (pipe, GST_STATE_NULL);
+  gst_object_unref (pipe);
+
+  return 0;
+}
diff --git a/examples/test-netclock.c b/examples/test-netclock.c
new file mode 100644
index 0000000..2e53635
--- /dev/null
+++ b/examples/test-netclock.c
@@ -0,0 +1,181 @@
+/* GStreamer
+ * Copyright (C) 2008 Wim Taymans <wim.taymans at gmail.com>
+ * Copyright (C) 2014 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/gst.h>
+
+#include <gst/net/gstnettimeprovider.h>
+#include <gst/rtsp-server/rtsp-server.h>
+
+GstClock *global_clock;
+
+#define TEST_TYPE_RTSP_MEDIA_FACTORY      (test_rtsp_media_factory_get_type ())
+#define TEST_TYPE_RTSP_MEDIA              (test_rtsp_media_get_type ())
+
+GType test_rtsp_media_factory_get_type (void);
+GType test_rtsp_media_get_type (void);
+
+static GstRTSPMediaFactory *test_rtsp_media_factory_new (void);
+static GstElement *create_pipeline (GstRTSPMediaFactory * factory,
+    GstRTSPMedia * media);
+
+typedef struct TestRTSPMediaFactoryClass TestRTSPMediaFactoryClass;
+typedef struct TestRTSPMediaFactory TestRTSPMediaFactory;
+
+struct TestRTSPMediaFactoryClass
+{
+  GstRTSPMediaFactoryClass parent;
+};
+
+struct TestRTSPMediaFactory
+{
+  GstRTSPMediaFactory parent;
+};
+
+typedef struct TestRTSPMediaClass TestRTSPMediaClass;
+typedef struct TestRTSPMedia TestRTSPMedia;
+
+struct TestRTSPMediaClass
+{
+  GstRTSPMediaClass parent;
+};
+
+struct TestRTSPMedia
+{
+  GstRTSPMedia parent;
+};
+
+GstRTSPMediaFactory *
+test_rtsp_media_factory_new (void)
+{
+  GstRTSPMediaFactory *result;
+
+  result = g_object_new (TEST_TYPE_RTSP_MEDIA_FACTORY, NULL);
+
+  return result;
+}
+
+G_DEFINE_TYPE (TestRTSPMediaFactory, test_rtsp_media_factory,
+    GST_TYPE_RTSP_MEDIA_FACTORY);
+
+static gboolean custom_setup_rtpbin (GstRTSPMedia * media, GstElement * rtpbin);
+
+static void
+test_rtsp_media_factory_class_init (TestRTSPMediaFactoryClass * test_klass)
+{
+  GstRTSPMediaFactoryClass *mf_klass =
+      (GstRTSPMediaFactoryClass *) (test_klass);
+  mf_klass->create_pipeline = create_pipeline;
+}
+
+static void
+test_rtsp_media_factory_init (TestRTSPMediaFactory * factory)
+{
+}
+
+static GstElement *
+create_pipeline (GstRTSPMediaFactory * factory, GstRTSPMedia * media)
+{
+  GstElement *pipeline;
+
+  pipeline = gst_pipeline_new ("media-pipeline");
+  gst_pipeline_use_clock (GST_PIPELINE (pipeline), global_clock);
+  gst_element_set_base_time (pipeline, 0);
+  gst_element_set_start_time (pipeline, GST_CLOCK_TIME_NONE);
+  gst_rtsp_media_take_pipeline (media, GST_PIPELINE_CAST (pipeline));
+
+  return pipeline;
+}
+
+G_DEFINE_TYPE (TestRTSPMedia, test_rtsp_media, GST_TYPE_RTSP_MEDIA);
+
+static void
+test_rtsp_media_class_init (TestRTSPMediaClass * test_klass)
+{
+  GstRTSPMediaClass *klass = (GstRTSPMediaClass *) (test_klass);
+  klass->setup_rtpbin = custom_setup_rtpbin;
+}
+
+static void
+test_rtsp_media_init (TestRTSPMedia * media)
+{
+}
+
+static gboolean
+custom_setup_rtpbin (GstRTSPMedia * media, GstElement * rtpbin)
+{
+  g_object_set (rtpbin, "use-pipeline-clock", TRUE, NULL);
+  return TRUE;
+}
+
+int
+main (int argc, char *argv[])
+{
+  GMainLoop *loop;
+  GstRTSPServer *server;
+  GstRTSPMountPoints *mounts;
+  GstRTSPMediaFactory *factory;
+
+  gst_init (&argc, &argv);
+
+  if (argc < 2) {
+    g_print ("usage: %s <launch line> \n"
+        "example: %s \"( videotestsrc is-live=true ! x264enc ! rtph264pay name=pay0 pt=96 )\"\n"
+        "Pipeline must be live for synchronisation to work properly with this method!\n",
+        argv[0], argv[0]);
+    return -1;
+  }
+
+  loop = g_main_loop_new (NULL, FALSE);
+
+  global_clock = gst_system_clock_obtain ();
+  gst_net_time_provider_new (global_clock, "0.0.0.0", 8554);
+
+  /* create a server instance */
+  server = gst_rtsp_server_new ();
+
+  /* get the mount points for this server, every server has a default object
+   * that be used to map uri mount points to media factories */
+  mounts = gst_rtsp_server_get_mount_points (server);
+
+  /* make a media factory for a test stream. The default media factory can use
+   * gst-launch syntax to create pipelines.
+   * any launch line works as long as it contains elements named pay%d. Each
+   * element with pay%d names will be a stream */
+  factory = test_rtsp_media_factory_new ();
+  gst_rtsp_media_factory_set_launch (factory, argv[1]);
+  gst_rtsp_media_factory_set_shared (GST_RTSP_MEDIA_FACTORY (factory), TRUE);
+  gst_rtsp_media_factory_set_media_gtype (GST_RTSP_MEDIA_FACTORY (factory),
+      TEST_TYPE_RTSP_MEDIA);
+
+  /* attach the test factory to the /test url */
+  gst_rtsp_mount_points_add_factory (mounts, "/test", factory);
+
+  /* don't need the ref to the mapper anymore */
+  g_object_unref (mounts);
+
+  /* attach the server to the default maincontext */
+  gst_rtsp_server_attach (server, NULL);
+
+  /* start serving */
+  g_print ("stream ready at rtsp://127.0.0.1:8554/test\n");
+  g_main_loop_run (loop);
+
+  return 0;
+}