141 lines
		
	
	
		
			4.5 KiB
		
	
	
	
		
			C
		
	
	
	
	
	
			
		
		
	
	
			141 lines
		
	
	
		
			4.5 KiB
		
	
	
	
		
			C
		
	
	
	
	
	
| /* 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>
 | |
| 
 | |
| typedef struct
 | |
| {
 | |
|   gboolean white;
 | |
|   GstClockTime timestamp;
 | |
| } MyContext;
 | |
| 
 | |
| /* called when we need to give data to appsrc */
 | |
| static void
 | |
| need_data (GstElement * appsrc, guint unused, MyContext * ctx)
 | |
| {
 | |
|   GstBuffer *buffer;
 | |
|   guint size;
 | |
|   GstFlowReturn ret;
 | |
| 
 | |
|   size = 385 * 288 * 2;
 | |
| 
 | |
|   buffer = gst_buffer_new_allocate (NULL, size, NULL);
 | |
| 
 | |
|   /* this makes the image black/white */
 | |
|   gst_buffer_memset (buffer, 0, ctx->white ? 0xff : 0x0, size);
 | |
| 
 | |
|   ctx->white = !ctx->white;
 | |
| 
 | |
|   /* increment the timestamp every 1/2 second */
 | |
|   GST_BUFFER_PTS (buffer) = ctx->timestamp;
 | |
|   GST_BUFFER_DURATION (buffer) = gst_util_uint64_scale_int (1, GST_SECOND, 2);
 | |
|   ctx->timestamp += GST_BUFFER_DURATION (buffer);
 | |
| 
 | |
|   g_signal_emit_by_name (appsrc, "push-buffer", buffer, &ret);
 | |
|   gst_buffer_unref (buffer);
 | |
| }
 | |
| 
 | |
| /* called when a new media pipeline is constructed. We can query the
 | |
|  * pipeline and configure our appsrc */
 | |
| static void
 | |
| media_configure (GstRTSPMediaFactory * factory, GstRTSPMedia * media,
 | |
|     gpointer user_data)
 | |
| {
 | |
|   GstElement *element, *appsrc;
 | |
|   MyContext *ctx;
 | |
| 
 | |
|   /* get the element used for providing the streams of the media */
 | |
|   element = gst_rtsp_media_get_element (media);
 | |
| 
 | |
|   /* get our appsrc, we named it 'mysrc' with the name property */
 | |
|   appsrc = gst_bin_get_by_name_recurse_up (GST_BIN (element), "mysrc");
 | |
| 
 | |
|   /* this instructs appsrc that we will be dealing with timed buffer */
 | |
|   gst_util_set_object_arg (G_OBJECT (appsrc), "format", "time");
 | |
|   /* configure the caps of the video */
 | |
|   g_object_set (G_OBJECT (appsrc), "caps",
 | |
|       gst_caps_new_simple ("video/x-raw",
 | |
|           "format", G_TYPE_STRING, "RGB16",
 | |
|           "width", G_TYPE_INT, 384,
 | |
|           "height", G_TYPE_INT, 288,
 | |
|           "framerate", GST_TYPE_FRACTION, 0, 1, NULL), NULL);
 | |
| 
 | |
|   ctx = g_new0 (MyContext, 1);
 | |
|   ctx->white = FALSE;
 | |
|   ctx->timestamp = 0;
 | |
|   /* make sure ther datais freed when the media is gone */
 | |
|   g_object_set_data_full (G_OBJECT (media), "my-extra-data", ctx,
 | |
|       (GDestroyNotify) g_free);
 | |
| 
 | |
|   /* install the callback that will be called when a buffer is needed */
 | |
|   g_signal_connect (appsrc, "need-data", (GCallback) need_data, ctx);
 | |
|   gst_object_unref (appsrc);
 | |
|   gst_object_unref (element);
 | |
| }
 | |
| 
 | |
| int
 | |
| main (int argc, char *argv[])
 | |
| {
 | |
|   GMainLoop *loop;
 | |
|   GstRTSPServer *server;
 | |
|   GstRTSPMountPoints *mounts;
 | |
|   GstRTSPMediaFactory *factory;
 | |
| 
 | |
|   gst_init (&argc, &argv);
 | |
| 
 | |
|   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);
 | |
| 
 | |
|   /* 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,
 | |
|       "( appsrc name=mysrc ! videoconvert ! x264enc ! rtph264pay name=pay0 pt=96 )");
 | |
| 
 | |
|   /* notify when our media is ready, This is called whenever someone asks for
 | |
|    * the media and a new pipeline with our appsrc is created */
 | |
|   g_signal_connect (factory, "media-configure", (GCallback) media_configure,
 | |
|       NULL);
 | |
| 
 | |
|   /* attach the test factory to the /test url */
 | |
|   gst_rtsp_mount_points_add_factory (mounts, "/test", factory);
 | |
| 
 | |
|   /* don't need the ref to the mounts 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;
 | |
| }
 |