gstreamer-tips-picture-in-picture-compositing

http://www.oz9aec.net/index.php/gstreamer/347-more-gstreamer-tips-picture-in-picture-compositing

http://blog.sina.com.cn/s/blog_5106eff101018lsu.html

1. RTSP协议建立服务器（该代码是C，但看看我的客户端端代码，看看它如何的API是相当直截了当） 我修改了代码的URL

/* GStreamer
 * Copyright (C) 2008 Wim Taymans <wim.taymans at gmail.com>
 * Copyright (c) 2012 enthusiasticgeek <enthusiasticgeek@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., 59 Temple Place - Suite 330,
 * Boston, MA 02111-1307, USA.
 */

//Edited by: enthusiasticgeek (c) 2012 for Stack Overflow Sept 11, 2012
//###########################################################################
//Important
//###########################################################################
//On ubuntu: sudo apt-get install libgstrtspserver-0.10-0 libgstrtspserver-0.10-dev
//Play with VLC
// CodeGo.net
//video decode only: gst-launch -v rtspsrc location=" CodeGo.net  ! rtph264depay ! ffdec_h264 ! autovideosink
//audio and video:
//gst-launch -v rtspsrc location=" CodeGo.net  name=demux demux. ! queue ! rtph264depay ! ffdec_h264 ! ffmpegcolorspace ! autovideosink sync=false demux. ! queue ! rtppcmadepay ! alawdec ! autoaudiosink
//###########################################################################
#include <gst/gst.h>
#include <gst/rtsp-server/rtsp-server.h>
/* define this if you want the resource to only be available when using
 * user/admin as the password */
#undef WITH_AUTH
/* this timeout is periodically run to clean up the expired sessions from the
 * pool. This needs to be run explicitly currently but might be done
 * automatically as part of the mainloop. */
static gboolean
timeout (GstRTSPServer * server, gboolean ignored)
{
 GstRTSPSessionPool *pool;
 pool = gst_rtsp_server_get_session_pool (server);
 gst_rtsp_session_pool_cleanup (pool);
 g_object_unref (pool);
 return TRUE;
}
int
main (int argc, char *argv[])
{
 GMainLoop *loop;
 GstRTSPServer *server;
 GstRTSPMediaMapping *mapping;
 GstRTSPMediaFactory *factory;
#ifdef WITH_AUTH
 GstRTSPAuth *auth;
 gchar *basic;
#endif
 gst_init (&argc, &argv);
 loop = g_main_loop_new (NULL, FALSE);
 /* create a server instance */
 server = gst_rtsp_server_new ();
 gst_rtsp_server_set_service(server,"8554"); //set the port #
 /* get the mapping for this server, every server has a default mapper object
 * that be used to map uri mount points to media factories */
 mapping = gst_rtsp_server_get_media_mapping (server);
#ifdef WITH_AUTH
 /* make a new authentication manager. it can be added to control access to all
 * the factories on the server or on individual factories. */
 auth = gst_rtsp_auth_new ();
 basic = gst_rtsp_auth_make_basic ("user", "admin");
 gst_rtsp_auth_set_basic (auth, basic);
 g_free (basic);
 /* configure in the server */
 gst_rtsp_server_set_auth (server, auth);
#endif
 /* 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, "( "
  "videotestsrc ! video/x-raw-yuv,width=320,height=240,framerate=10/1 ! "
  "x264enc ! queue ! rtph264pay name=pay0 pt=96 ! audiotestsrc ! audio/x-raw-int,rate=8000 ! alawenc ! rtppcmapay name=pay1 pt=97 "")");
 /* attach the test factory to the /test url */
 gst_rtsp_media_mapping_add_factory (mapping, "/test", factory);
 /* don't need the ref to the mapper anymore */
 g_object_unref (mapping);
 /* attach the server to the default maincontext */
 if (gst_rtsp_server_attach (server, NULL) == 0)
 goto failed;
 /* add a timeout for the session cleanup */
 g_timeout_add_seconds (2, (GSourceFunc) timeout, server);
 /* start serving, this never stops */
 g_main_loop_run (loop);
 return 0;
 /* ERRORS */
failed:
 {
 g_print ("failed to attach the server\n");
 return -1;
 }
}

Makefile文件

# Copyright (c) 2012 enthusiasticgeek
# RTSP demo for Stack Overflow
sample:
 gcc -Wall -I/usr/include/gstreamer-0.10 rtsp.c -o rtsp `pkg-config --libs --cflags gstreamer-0.10 gstreamer-rtsp-0.10` -lglib-2.0 -lgstrtspserver-0.10 -lgstreamer-0.10

一旦你建立了二进制，简单来说它./rtsp然后打开另一个选项卡中的终端测试以下的pipeline。 测试解码流水线。它工作得很好！

gst-launch -v rtspsrc location=" CodeGo.net  name=demux demux. ! queue ! rtph264depay ! ffdec_h264 ! ffmpegcolorspace ! autovideosink sync=false demux. ! queue ! rtppcmadepay ! alawdec ! autoaudiosink

 gst_rtsp_media_factory_set_launch (factory,
      "( videotestsrc is-live=1 ! x264enc ! rtph264pay name=pay0 pt=96 )");