summaryrefslogtreecommitdiff
path: root/examples/test-mp4.c
blob: 34b8906aab2edd4b23417d13a3452e60316d6d55 (plain)
1
2
3
4
5
6
7
8
9
10
11
12
13
14
15
16
17
18
19
20
21
22
23
24
25
26
27
28
29
30
31
32
33
34
35
36
37
38
39
40
41
42
43
44
45
46
47
48
49
50
51
52
53
54
55
56
57
58
59
60
61
62
63
64
65
66
67
68
69
70
71
72
73
74
75
76
77
78
79
80
81
82
83
84
85
86
87
88
89
90
91
92
93
94
95
96
97
98
99
100
101
102
103
104
105
106
107
108
109
110
111
112
113
114
115
116
117
118
119
120
121
122
123
124
125
126
127
128
129
130
131
132
133
134
135
136
137
138
139
140
141
142
143
144
145
146
147
148
149
150
151
152
153
154
155
/* 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>

#define DEFAULT_RTSP_PORT "8554"

static char *port = (char *) DEFAULT_RTSP_PORT;

static GOptionEntry entries[] = {
  {"port", 'p', 0, G_OPTION_ARG_STRING, &port,
      "Port to listen on (default: " DEFAULT_RTSP_PORT ")", "PORT"},
  {NULL}
};

/* 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;
  GOptionContext *optctx;
  GError *error = NULL;
  gchar *str;

  optctx = g_option_context_new ("<filename.mp4> - Test RTSP Server, MP4");
  g_option_context_add_main_entries (optctx, entries, NULL);
  g_option_context_add_group (optctx, gst_init_get_option_group ());
  if (!g_option_context_parse (optctx, &argc, &argv, &error)) {
    g_printerr ("Error parsing options: %s\n", error->message);
    g_option_context_free (optctx);
    g_clear_error (&error);
    return -1;
  }

  if (argc < 2) {
    g_print ("%s\n", g_option_context_get_help (optctx, TRUE, NULL));
    return 1;
  }
  g_option_context_free (optctx);

  loop = g_main_loop_new (NULL, FALSE);

  /* create a server instance */
  server = gst_rtsp_server_new ();
  g_object_set (server, "service", port, NULL);

  /* 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:%s/test\n", port);
  g_main_loop_run (loop);

  return 0;
}