gstreamer/gst/rtsp/test.c

188 lines
4.8 KiB
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/* GStreamer
* Copyright (C) <2005> Wim Taymans <wim@fluendo.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.
*/
#include <stdio.h>
#include "sdp.h"
#include "rtsp.h"
int
main (int argc, gchar * argv[])
{
RTSPUrl *url;
RTSPConnection *conn;
RTSPResult res;
RTSPMessage request = { 0 };
gchar *urlstr;
RTSPMessage response = { 0 };
SDPMessage sdp = { 0 };
urlstr = "rtsp://thread:5454/south-rtsp.mp3";
/* create url */
g_print ("parsing url \"%s\"...\n", urlstr);
res = rtsp_url_parse (urlstr, &url);
if (res != RTSP_OK) {
g_print ("error parsing url \"%s\"\n", urlstr);
return (-1);
}
g_print (" url host: %s\n", url->host);
g_print (" url port: %d\n", url->port);
g_print (" url path: %s\n", url->abspath);
gst/rtsp/URLS: Added some test URLS. Original commit message from CVS: * gst/rtsp/URLS: Added some test URLS. * gst/rtsp/gstrtspsrc.c: (gst_rtspsrc_create_stream), (gst_rtspsrc_loop), (gst_rtspsrc_open): * gst/rtsp/gstrtspsrc.h: When creating streams, give access to the complete SDP. Fix some leaks. Collect and merge global stream properties in stream caps. Preliminary support for WMServer. * gst/rtsp/rtspconnection.c: (rtsp_connection_create), (rtsp_connection_connect), (rtsp_connection_read), (read_body), (rtsp_connection_receive): * gst/rtsp/rtspconnection.h: Make connection interruptable. Refactor to make it reconnectable. Don't fail on short reads when reading data packets. * gst/rtsp/rtspurl.c: (rtsp_url_parse), (rtsp_url_set_port), (rtsp_url_get_port): * gst/rtsp/rtspurl.h: Add methods for getting/setting the port. * gst/rtsp/sdpmessage.c: (sdp_message_get_attribute_val_n), (sdp_message_get_attribute_val), (sdp_media_get_attribute), (sdp_media_get_attribute_val_n), (sdp_media_get_attribute_val), (sdp_media_get_format), (sdp_parse_line), (sdp_message_parse_buffer): Fix headers. Add methods for getting multiple attributes with the same name. Increase buffer size when parsing. Fix parsing of a=foo fields. * gst/rtsp/test.c: (main): Update to new connection API. * gst/rtsp/rtspmessage.c: (rtsp_message_new_response), (rtsp_message_init_response), (rtsp_message_init_data), (rtsp_message_unset), (rtsp_message_free), (rtsp_message_dump): * gst/rtsp/rtspmessage.h: * gst/rtsp/rtsptransport.c: (rtsp_transport_free): * gst/rtsp/rtsptransport.h: * gst/rtsp/sdp.h: * gst/rtsp/sdpmessage.h: * gst/rtsp/gstrtsp.c: * gst/rtsp/gstrtsp.h: * gst/rtsp/gstrtpdec.c: * gst/rtsp/gstrtpdec.h: * gst/rtsp/rtsp.h: * gst/rtsp/rtspdefs.c: * gst/rtsp/rtspdefs.h: Dual licensed under MIT and LGPL now.
2006-09-20 16:06:27 +00:00
/* create and open connection */
g_print ("creating connection...\n");
res = rtsp_connection_create (url, &conn);
if (res != RTSP_OK) {
g_print ("error creating connection to \"%s\"\n", urlstr);
return (-1);
}
/* open connection */
g_print ("opening connection...\n");
gst/rtsp/URLS: Added some test URLS. Original commit message from CVS: * gst/rtsp/URLS: Added some test URLS. * gst/rtsp/gstrtspsrc.c: (gst_rtspsrc_create_stream), (gst_rtspsrc_loop), (gst_rtspsrc_open): * gst/rtsp/gstrtspsrc.h: When creating streams, give access to the complete SDP. Fix some leaks. Collect and merge global stream properties in stream caps. Preliminary support for WMServer. * gst/rtsp/rtspconnection.c: (rtsp_connection_create), (rtsp_connection_connect), (rtsp_connection_read), (read_body), (rtsp_connection_receive): * gst/rtsp/rtspconnection.h: Make connection interruptable. Refactor to make it reconnectable. Don't fail on short reads when reading data packets. * gst/rtsp/rtspurl.c: (rtsp_url_parse), (rtsp_url_set_port), (rtsp_url_get_port): * gst/rtsp/rtspurl.h: Add methods for getting/setting the port. * gst/rtsp/sdpmessage.c: (sdp_message_get_attribute_val_n), (sdp_message_get_attribute_val), (sdp_media_get_attribute), (sdp_media_get_attribute_val_n), (sdp_media_get_attribute_val), (sdp_media_get_format), (sdp_parse_line), (sdp_message_parse_buffer): Fix headers. Add methods for getting multiple attributes with the same name. Increase buffer size when parsing. Fix parsing of a=foo fields. * gst/rtsp/test.c: (main): Update to new connection API. * gst/rtsp/rtspmessage.c: (rtsp_message_new_response), (rtsp_message_init_response), (rtsp_message_init_data), (rtsp_message_unset), (rtsp_message_free), (rtsp_message_dump): * gst/rtsp/rtspmessage.h: * gst/rtsp/rtsptransport.c: (rtsp_transport_free): * gst/rtsp/rtsptransport.h: * gst/rtsp/sdp.h: * gst/rtsp/sdpmessage.h: * gst/rtsp/gstrtsp.c: * gst/rtsp/gstrtsp.h: * gst/rtsp/gstrtpdec.c: * gst/rtsp/gstrtpdec.h: * gst/rtsp/rtsp.h: * gst/rtsp/rtspdefs.c: * gst/rtsp/rtspdefs.h: Dual licensed under MIT and LGPL now.
2006-09-20 16:06:27 +00:00
res = rtsp_connection_connect (conn);
if (res != RTSP_OK) {
g_print ("error opening connection to \"%s\"\n", urlstr);
return (-1);
}
/* do describe */
{
res = rtsp_message_init_request (&request, RTSP_DESCRIBE, urlstr);
if (res != RTSP_OK) {
g_print ("error creating request\n");
return (-1);
}
rtsp_message_add_header (&request, RTSP_HDR_ACCEPT, "application/sdp");
rtsp_message_dump (&request);
res = rtsp_connection_send (conn, &request);
if (res != RTSP_OK) {
g_print ("error sending request\n");
return (-1);
}
res = rtsp_connection_receive (conn, &response);
if (res != RTSP_OK) {
g_print ("error receiving response\n");
return (-1);
}
rtsp_message_dump (&response);
}
/* parse SDP */
{
guint8 *data;
guint size;
rtsp_message_get_body (&response, &data, &size);
sdp_message_init (&sdp);
sdp_message_parse_buffer (data, size, &sdp);
sdp_message_dump (&sdp);
}
/* do setup */
{
gint i;
for (i = 0; i < sdp_message_medias_len (&sdp); i++) {
SDPMedia *media;
gchar *setup_url;
gchar *control_url;
media = sdp_message_get_media (&sdp, i);
g_print ("setup media %d\n", i);
control_url = sdp_media_get_attribute_val (media, "control");
setup_url = g_strdup_printf ("%s/%s", urlstr, control_url);
g_print ("setup %s\n", setup_url);
res = rtsp_message_init_request (&request, RTSP_SETUP, setup_url);
if (res != RTSP_OK) {
g_print ("error creating request\n");
return (-1);
}
rtsp_message_add_header (&request, RTSP_HDR_TRANSPORT,
//"RTP/AVP/UDP;unicast;client_port=5000-5001,RTP/AVP/UDP;multicast,RTP/AVP/TCP");
"RTP/AVP/TCP");
rtsp_message_dump (&request);
res = rtsp_connection_send (conn, &request);
if (res != RTSP_OK) {
g_print ("error sending request\n");
return (-1);
}
res = rtsp_connection_receive (conn, &response);
if (res != RTSP_OK) {
g_print ("error receiving response\n");
return (-1);
}
rtsp_message_dump (&response);
}
}
/* do play */
{
res = rtsp_message_init_request (&request, RTSP_PLAY, urlstr);
if (res != RTSP_OK) {
g_print ("error creating request\n");
return (-1);
}
rtsp_message_dump (&request);
res = rtsp_connection_send (conn, &request);
if (res != RTSP_OK) {
g_print ("error sending request\n");
return (-1);
}
res = rtsp_connection_receive (conn, &response);
if (res != RTSP_OK) {
g_print ("error receiving response\n");
return (-1);
}
rtsp_message_dump (&response);
}
while (TRUE) {
res = rtsp_connection_receive (conn, &response);
if (res != RTSP_OK) {
g_print ("error receiving response\n");
return (-1);
}
rtsp_message_dump (&response);
}
/* close connection */
g_print ("closing connection...\n");
res = rtsp_connection_close (conn);
if (res != RTSP_OK) {
g_print ("error closing connection to \"%s\"\n", urlstr);
return (-1);
}
return 0;
}