/* GStreamer * Copyright (C) 2008 Wim Taymans * * 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 #include 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); } /* 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; }