/* GStreamer
 * Copyright (C) 2008 Wim Taymans <wim.taymans at gmail.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/rtsp-server/rtsp-server.h>

/* called when a stream has received an RTCP packet from the client */
static void
on_ssrc_active (GObject * session, GObject * source, GstRTSPMedia * media)
{
  GstStructure *stats;

  GST_INFO ("source %p in session %p is active", source, session);

  g_object_get (source, "stats", &stats, NULL);
  if (stats) {
    gchar *sstr;

    sstr = gst_structure_to_string (stats);
    g_print ("structure: %s\n", sstr);
    g_free (sstr);

    gst_structure_free (stats);
  }
}

/* signal callback when the media is prepared for streaming. We can get the
 * session manager for each of the streams and connect to some signals. */
static void
media_prepared_cb (GstRTSPMedia * media)
{
  guint i, n_streams;

  n_streams = gst_rtsp_media_n_streams (media);

  GST_INFO ("media %p is prepared and has %u streams", media, n_streams);

  for (i = 0; i < n_streams; i++) {
    GstRTSPStream *stream;
    GObject *session;

    stream = gst_rtsp_media_get_stream (media, i);
    if (stream == NULL)
      continue;

    session = gst_rtsp_stream_get_rtpsession (stream);
    GST_INFO ("watching session %p on stream %u", session, i);

    g_signal_connect (session, "on-ssrc-active",
        (GCallback) on_ssrc_active, media);
  }
}

static void
media_configure_cb (GstRTSPMediaFactory * factory, GstRTSPMedia * media)
{
  /* connect our prepared signal so that we can see when this media is
   * prepared for streaming */
  g_signal_connect (media, "prepared", (GCallback) media_prepared_cb, factory);
}

int
main (int argc, char *argv[])
{
  GMainLoop *loop;
  GstRTSPServer *server;
  GstRTSPMountPoints *mounts;
  GstRTSPMediaFactory *factory;
  gchar *str;

  gst_init (&argc, &argv);

  if (argc < 2) {
    g_message ("usage: %s <filename.mp4>", argv[0]);
    return -1;
  }

  loop = g_main_loop_new (NULL, FALSE);

  /* 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);

  str = g_strdup_printf ("( "
      "filesrc location=%s ! qtdemux name=d "
      "d. ! queue ! rtph264pay pt=96 name=pay0 "
      "d. ! queue ! rtpmp4apay pt=97 name=pay1 " ")", argv[1]);

  /* 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 = gst_rtsp_media_factory_new ();
  gst_rtsp_media_factory_set_launch (factory, str);
  g_signal_connect (factory, "media-configure", (GCallback) media_configure_cb,
      factory);
  g_free (str);

  /* 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;
}