2005-05-11 07:44:44 +00:00
|
|
|
/* 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.
|
|
|
|
*/
|
|
|
|
|
2005-06-29 16:27:27 +00:00
|
|
|
#include <string.h>
|
|
|
|
#include <stdlib.h>
|
|
|
|
|
2005-05-11 07:44:44 +00:00
|
|
|
#include "rtsptransport.h"
|
|
|
|
|
|
|
|
RTSPResult
|
|
|
|
rtsp_transport_new (RTSPTransport ** transport)
|
|
|
|
{
|
|
|
|
RTSPTransport *trans;
|
|
|
|
|
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 (transport != NULL, RTSP_EINVAL);
|
2005-05-11 07:44:44 +00:00
|
|
|
|
|
|
|
trans = g_new0 (RTSPTransport, 1);
|
|
|
|
|
|
|
|
*transport = trans;
|
|
|
|
|
|
|
|
return rtsp_transport_init (trans);
|
|
|
|
}
|
|
|
|
|
|
|
|
RTSPResult
|
|
|
|
rtsp_transport_init (RTSPTransport * transport)
|
|
|
|
{
|
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 (transport != NULL, RTSP_EINVAL);
|
|
|
|
|
2005-05-11 07:44:44 +00:00
|
|
|
g_free (transport->destination);
|
|
|
|
g_free (transport->source);
|
|
|
|
g_free (transport->ssrc);
|
|
|
|
|
|
|
|
memset (transport, 0, sizeof (RTSPTransport));
|
|
|
|
|
|
|
|
transport->trans = RTSP_TRANS_RTP;
|
|
|
|
transport->profile = RTSP_PROFILE_AVP;
|
|
|
|
transport->lower_transport = RTSP_LOWER_TRANS_UNKNOWN;
|
|
|
|
transport->mode_play = TRUE;
|
|
|
|
transport->mode_record = FALSE;
|
|
|
|
|
|
|
|
return RTSP_OK;
|
|
|
|
}
|
|
|
|
|
|
|
|
static void
|
|
|
|
parse_mode (RTSPTransport * transport, gchar * str)
|
|
|
|
{
|
gst/rtsp/: RTSP cleanups.
Original commit message from CVS:
* gst/rtsp/README:
* gst/rtsp/gstrtspsrc.c: (gst_rtsp_proto_get_type),
(gst_rtspsrc_class_init), (gst_rtspsrc_create_stream),
(gst_rtspsrc_add_element), (gst_rtspsrc_set_state),
(gst_rtspsrc_stream_setup_rtp),
(gst_rtspsrc_stream_configure_transport), (find_stream),
(gst_rtspsrc_loop), (gst_rtspsrc_open), (gst_rtspsrc_play),
(gst_rtspsrc_change_state):
* gst/rtsp/rtsptransport.c: (rtsp_transport_new),
(rtsp_transport_init), (parse_mode), (parse_range),
(rtsp_transport_parse), (rtsp_transport_free):
RTSP cleanups.
2005-06-02 13:26:36 +00:00
|
|
|
transport->mode_play = (strstr (str, "\"PLAY\"") != NULL);
|
|
|
|
transport->mode_record = (strstr (str, "\"RECORD\"") != NULL);
|
2005-05-11 07:44:44 +00:00
|
|
|
}
|
|
|
|
|
|
|
|
static void
|
|
|
|
parse_range (RTSPTransport * transport, gchar * str, RTSPRange * range)
|
|
|
|
{
|
|
|
|
gchar *minus;
|
|
|
|
|
|
|
|
minus = strstr (str, "-");
|
|
|
|
if (minus) {
|
|
|
|
range->min = atoi (str);
|
|
|
|
range->max = atoi (minus + 1);
|
|
|
|
} else {
|
|
|
|
range->min = atoi (str);
|
|
|
|
range->max = -1;
|
|
|
|
}
|
|
|
|
}
|
|
|
|
|
|
|
|
RTSPResult
|
|
|
|
rtsp_transport_parse (gchar * str, RTSPTransport * transport)
|
|
|
|
{
|
2006-02-15 10:15:47 +00:00
|
|
|
gchar **split, *down;
|
2005-05-11 07:44:44 +00:00
|
|
|
gint i;
|
|
|
|
|
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 (transport != NULL, RTSP_EINVAL);
|
|
|
|
g_return_val_if_fail (str != NULL, RTSP_EINVAL);
|
2005-05-11 07:44:44 +00:00
|
|
|
|
|
|
|
rtsp_transport_init (transport);
|
|
|
|
|
2006-02-15 10:15:47 +00:00
|
|
|
/* case insensitive */
|
|
|
|
down = g_ascii_strdown (str, -1);
|
|
|
|
|
|
|
|
split = g_strsplit (down, ";", 0);
|
2005-05-11 07:44:44 +00:00
|
|
|
i = 0;
|
|
|
|
while (split[i]) {
|
2006-02-15 10:15:47 +00:00
|
|
|
if (g_str_has_prefix (split[i], "rtp/avp/udp")) {
|
2005-05-11 07:44:44 +00:00
|
|
|
transport->lower_transport = RTSP_LOWER_TRANS_UDP;
|
2006-02-15 10:15:47 +00:00
|
|
|
} else if (g_str_has_prefix (split[i], "rtp/avp/tcp")) {
|
2005-05-11 07:44:44 +00:00
|
|
|
transport->lower_transport = RTSP_LOWER_TRANS_TCP;
|
2006-02-15 10:15:47 +00:00
|
|
|
} else if (g_str_has_prefix (split[i], "rtp/avp")) {
|
2005-08-18 16:53:14 +00:00
|
|
|
transport->lower_transport = RTSP_LOWER_TRANS_UDP;
|
2005-05-11 07:44:44 +00:00
|
|
|
} else if (g_str_has_prefix (split[i], "multicast")) {
|
|
|
|
transport->multicast = TRUE;
|
|
|
|
} else if (g_str_has_prefix (split[i], "unicast")) {
|
|
|
|
transport->multicast = FALSE;
|
|
|
|
} else if (g_str_has_prefix (split[i], "destination=")) {
|
|
|
|
transport->destination = g_strdup (split[i] + 12);
|
|
|
|
} else if (g_str_has_prefix (split[i], "source=")) {
|
|
|
|
transport->source = g_strdup (split[i] + 7);
|
|
|
|
} else if (g_str_has_prefix (split[i], "layers=")) {
|
|
|
|
transport->layers = atoi (split[i] + 7);
|
|
|
|
} else if (g_str_has_prefix (split[i], "mode=")) {
|
|
|
|
parse_mode (transport, split[i] + 5);
|
|
|
|
} else if (g_str_has_prefix (split[i], "append")) {
|
|
|
|
transport->append = TRUE;
|
|
|
|
} else if (g_str_has_prefix (split[i], "interleaved=")) {
|
|
|
|
parse_range (transport, split[i] + 12, &transport->interleaved);
|
|
|
|
} else if (g_str_has_prefix (split[i], "ttl=")) {
|
|
|
|
transport->ttl = atoi (split[i] + 4);
|
|
|
|
} else if (g_str_has_prefix (split[i], "port=")) {
|
|
|
|
parse_range (transport, split[i] + 5, &transport->port);
|
|
|
|
} else if (g_str_has_prefix (split[i], "client_port=")) {
|
|
|
|
parse_range (transport, split[i] + 12, &transport->client_port);
|
|
|
|
} else if (g_str_has_prefix (split[i], "server_port=")) {
|
|
|
|
parse_range (transport, split[i] + 12, &transport->server_port);
|
|
|
|
} else if (g_str_has_prefix (split[i], "ssrc=")) {
|
|
|
|
transport->ssrc = g_strdup (split[i] + 5);
|
|
|
|
} else {
|
|
|
|
/* unknown field... */
|
|
|
|
g_warning ("unknown transport field \"%s\"", split[i]);
|
|
|
|
}
|
|
|
|
i++;
|
|
|
|
}
|
|
|
|
g_strfreev (split);
|
2006-02-15 10:15:47 +00:00
|
|
|
g_free (down);
|
2005-05-11 07:44:44 +00:00
|
|
|
|
|
|
|
return RTSP_OK;
|
|
|
|
}
|
|
|
|
|
|
|
|
RTSPResult
|
|
|
|
rtsp_transport_free (RTSPTransport * transport)
|
|
|
|
{
|
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 (transport != NULL, RTSP_EINVAL);
|
|
|
|
|
2005-05-11 07:44:44 +00:00
|
|
|
rtsp_transport_init (transport);
|
|
|
|
g_free (transport);
|
|
|
|
return RTSP_OK;
|
|
|
|
}
|