gstreamer/gst/rtsp/rtspurl.c

117 lines
2.6 KiB
C
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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 <stdlib.h>
#include <string.h>
#include "rtspurl.h"
#define RTSP_PROTO "rtsp://"
#define RTSP_PROTO_LEN 7
#define RTSPU_PROTO "rtspu://"
#define RTSPU_PROTO_LEN 8
/* format is rtsp[u]://[user:passwd@]host[:port]/abspath */
RTSPResult
rtsp_url_parse (const gchar * urlstr, RTSPUrl ** url)
{
RTSPUrl *res;
gchar *p, *slash, *at, *col;
gst/rtsp/gstrtspsrc.*: Small cleanups, added documentation. Original commit message from CVS: * gst/rtsp/gstrtspsrc.c: (gst_rtspsrc_media_to_caps), (gst_rtspsrc_send), (gst_rtspsrc_parse_methods), (gst_rtspsrc_open), (gst_rtspsrc_close), (gst_rtspsrc_play), (gst_rtspsrc_pause), (gst_rtspsrc_change_state), (gst_rtspsrc_uri_get_uri), (gst_rtspsrc_uri_set_uri): * gst/rtsp/gstrtspsrc.h: Small cleanups, added documentation. Try to clean up the requests and responses. Refactor parsing the supported methods. * gst/rtsp/rtspconnection.c: (rtsp_connection_open), (rtsp_connection_create), (rtsp_connection_send), (parse_response_status), (parse_request_line), (rtsp_connection_receive), (rtsp_connection_close), (rtsp_connection_free): * gst/rtsp/rtsptransport.c: (rtsp_transport_new), (rtsp_transport_init), (rtsp_transport_parse), (rtsp_transport_free): * gst/rtsp/rtspurl.c: (rtsp_url_parse): * gst/rtsp/sdpmessage.c: (sdp_message_new), (sdp_message_init), (sdp_message_clean), (sdp_message_free), (sdp_media_new), (sdp_media_init), (sdp_message_parse_buffer), (sdp_message_dump): Use g_return_val some more. * gst/rtsp/rtspdefs.h: Add more enum values to track initial states. * gst/rtsp/rtspmessage.c: (rtsp_message_new_request), (rtsp_message_init_request), (rtsp_message_new_response), (rtsp_message_init_response), (rtsp_message_init_data), (rtsp_message_unset), (rtsp_message_free), (rtsp_message_add_header), (rtsp_message_remove_header), (rtsp_message_get_header), (rtsp_message_set_body), (rtsp_message_take_body), (rtsp_message_get_body), (rtsp_message_steal_body), (rtsp_message_dump): * gst/rtsp/rtspmessage.h: Reorder arguments, object goes as the first one. Use g_return_val some more.
2006-09-18 17:37:46 +00:00
g_return_val_if_fail (urlstr != NULL, RTSP_EINVAL);
g_return_val_if_fail (url != NULL, RTSP_EINVAL);
gst/rtsp/gstrtspsrc.*: Small cleanups, added documentation. Original commit message from CVS: * gst/rtsp/gstrtspsrc.c: (gst_rtspsrc_media_to_caps), (gst_rtspsrc_send), (gst_rtspsrc_parse_methods), (gst_rtspsrc_open), (gst_rtspsrc_close), (gst_rtspsrc_play), (gst_rtspsrc_pause), (gst_rtspsrc_change_state), (gst_rtspsrc_uri_get_uri), (gst_rtspsrc_uri_set_uri): * gst/rtsp/gstrtspsrc.h: Small cleanups, added documentation. Try to clean up the requests and responses. Refactor parsing the supported methods. * gst/rtsp/rtspconnection.c: (rtsp_connection_open), (rtsp_connection_create), (rtsp_connection_send), (parse_response_status), (parse_request_line), (rtsp_connection_receive), (rtsp_connection_close), (rtsp_connection_free): * gst/rtsp/rtsptransport.c: (rtsp_transport_new), (rtsp_transport_init), (rtsp_transport_parse), (rtsp_transport_free): * gst/rtsp/rtspurl.c: (rtsp_url_parse): * gst/rtsp/sdpmessage.c: (sdp_message_new), (sdp_message_init), (sdp_message_clean), (sdp_message_free), (sdp_media_new), (sdp_media_init), (sdp_message_parse_buffer), (sdp_message_dump): Use g_return_val some more. * gst/rtsp/rtspdefs.h: Add more enum values to track initial states. * gst/rtsp/rtspmessage.c: (rtsp_message_new_request), (rtsp_message_init_request), (rtsp_message_new_response), (rtsp_message_init_response), (rtsp_message_init_data), (rtsp_message_unset), (rtsp_message_free), (rtsp_message_add_header), (rtsp_message_remove_header), (rtsp_message_get_header), (rtsp_message_set_body), (rtsp_message_take_body), (rtsp_message_get_body), (rtsp_message_steal_body), (rtsp_message_dump): * gst/rtsp/rtspmessage.h: Reorder arguments, object goes as the first one. Use g_return_val some more.
2006-09-18 17:37:46 +00:00
res = g_new0 (RTSPUrl, 1);
p = (gchar *) urlstr;
if (g_str_has_prefix (p, RTSP_PROTO)) {
res->protocol = RTSP_PROTO_TCP;
p += RTSP_PROTO_LEN;
} else if (g_str_has_prefix (p, RTSPU_PROTO)) {
res->protocol = RTSP_PROTO_UDP;
p += RTSPU_PROTO_LEN;
} else
goto invalid;
slash = strstr (p, "/");
at = strstr (p, "@");
if (at && slash && at > slash)
at = NULL;
if (at) {
col = strstr (p, ":");
/* must have a ':' and it must be before the '@' */
if (col == NULL || col > at)
goto invalid;
res->user = g_strndup (p, col - p);
col++;
res->passwd = g_strndup (col, col - at);
/* move to host */
p = at + 1;
}
col = strstr (p, ":");
if (col) {
res->host = g_strndup (p, col - p);
p = col + 1;
res->port = strtoul (p, (char **) &p, 10);
if (slash)
p = slash + 1;
} else {
res->port = RTSP_DEFAULT_PORT;
if (!slash) {
res->host = g_strdup (p);
p = NULL;
} else {
res->host = g_strndup (p, slash - p);
p = slash + 1;
}
}
if (p)
res->abspath = g_strdup (p);
*url = res;
return RTSP_OK;
/* ERRORS */
invalid:
{
rtsp_url_free (res);
return RTSP_EINVAL;
}
}
void
rtsp_url_free (RTSPUrl * url)
{
if (url == NULL)
return;
g_free (url->user);
g_free (url->passwd);
g_free (url->host);
g_free (url->abspath);
g_free (url);
}