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133 lines
3.9 KiB
C
133 lines
3.9 KiB
C
/* GStreamer
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* Copyright (C) 2008 Wim Taymans <wim.taymans at gmail.com>
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*
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* This library is free software; you can redistribute it and/or
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* modify it under the terms of the GNU Library General Public
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* License as published by the Free Software Foundation; either
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* version 2 of the License, or (at your option) any later version.
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*
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* This library is distributed in the hope that it will be useful,
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* but WITHOUT ANY WARRANTY; without even the implied warranty of
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* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
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* Library General Public License for more details.
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*
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* You should have received a copy of the GNU Library General Public
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* License along with this library; if not, write to the
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* Free Software Foundation, Inc., 51 Franklin St, Fifth Floor,
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* Boston, MA 02110-1301, USA.
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*/
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#include <gst/gst.h>
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#include <gst/rtsp-server/rtsp-server.h>
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/* called when a stream has received an RTCP packet from the client */
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static void
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on_ssrc_active (GObject * session, GObject * source, GstRTSPMedia * media)
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{
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GstStructure *stats;
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GST_INFO ("source %p in session %p is active", source, session);
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g_object_get (source, "stats", &stats, NULL);
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if (stats) {
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gchar *sstr;
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sstr = gst_structure_to_string (stats);
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g_print ("structure: %s\n", sstr);
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g_free (sstr);
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gst_structure_free (stats);
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}
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}
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/* signal callback when the media is prepared for streaming. We can get the
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* session manager for each of the streams and connect to some signals. */
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static void
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media_prepared_cb (GstRTSPMedia * media)
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{
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guint i, n_streams;
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n_streams = gst_rtsp_media_n_streams (media);
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GST_INFO ("media %p is prepared and has %u streams", media, n_streams);
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for (i = 0; i < n_streams; i++) {
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GstRTSPStream *stream;
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GObject *session;
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stream = gst_rtsp_media_get_stream (media, i);
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if (stream == NULL)
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continue;
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session = gst_rtsp_stream_get_rtpsession (stream);
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GST_INFO ("watching session %p on stream %u", session, i);
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g_signal_connect (session, "on-ssrc-active",
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(GCallback) on_ssrc_active, media);
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}
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}
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static void
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media_configure_cb (GstRTSPMediaFactory * factory, GstRTSPMedia * media)
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{
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/* connect our prepared signal so that we can see when this media is
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* prepared for streaming */
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g_signal_connect (media, "prepared", (GCallback) media_prepared_cb, factory);
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}
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int
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main (int argc, char *argv[])
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{
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GMainLoop *loop;
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GstRTSPServer *server;
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GstRTSPMountPoints *mounts;
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GstRTSPMediaFactory *factory;
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gchar *str;
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gst_init (&argc, &argv);
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if (argc < 2) {
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g_message ("usage: %s <filename.mp4>", argv[0]);
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return -1;
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}
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loop = g_main_loop_new (NULL, FALSE);
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/* create a server instance */
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server = gst_rtsp_server_new ();
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/* get the mount points for this server, every server has a default object
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* that be used to map uri mount points to media factories */
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mounts = gst_rtsp_server_get_mount_points (server);
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str = g_strdup_printf ("( "
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"filesrc location=%s ! qtdemux name=d "
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"d. ! queue ! rtph264pay pt=96 name=pay0 "
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"d. ! queue ! rtpmp4apay pt=97 name=pay1 " ")", argv[1]);
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/* make a media factory for a test stream. The default media factory can use
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* gst-launch syntax to create pipelines.
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* any launch line works as long as it contains elements named pay%d. Each
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* element with pay%d names will be a stream */
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factory = gst_rtsp_media_factory_new ();
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gst_rtsp_media_factory_set_launch (factory, str);
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g_signal_connect (factory, "media-configure", (GCallback) media_configure_cb,
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factory);
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g_free (str);
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/* attach the test factory to the /test url */
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gst_rtsp_mount_points_add_factory (mounts, "/test", factory);
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/* don't need the ref to the mapper anymore */
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g_object_unref (mounts);
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/* attach the server to the default maincontext */
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gst_rtsp_server_attach (server, NULL);
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/* start serving */
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g_print ("stream ready at rtsp://127.0.0.1:8554/test\n");
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g_main_loop_run (loop);
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return 0;
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}
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