2005-05-11 07:44:44 +00:00
|
|
|
/* GStreamer
|
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
|
|
|
* Copyright (C) <2005,2006> Wim Taymans <wim@fluendo.com>
|
2005-05-11 07:44:44 +00:00
|
|
|
*
|
|
|
|
* 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.
|
|
|
|
*/
|
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
|
|
|
/*
|
|
|
|
* Unless otherwise indicated, Source Code is licensed under MIT license.
|
|
|
|
* See further explanation attached in License Statement (distributed in the file
|
|
|
|
* LICENSE).
|
|
|
|
*
|
|
|
|
* Permission is hereby granted, free of charge, to any person obtaining a copy of
|
|
|
|
* this software and associated documentation files (the "Software"), to deal in
|
|
|
|
* the Software without restriction, including without limitation the rights to
|
|
|
|
* use, copy, modify, merge, publish, distribute, sublicense, and/or sell copies
|
|
|
|
* of the Software, and to permit persons to whom the Software is furnished to do
|
|
|
|
* so, subject to the following conditions:
|
|
|
|
*
|
|
|
|
* The above copyright notice and this permission notice shall be included in all
|
|
|
|
* copies or substantial portions of the Software.
|
|
|
|
*
|
|
|
|
* THE SOFTWARE IS PROVIDED "AS IS", WITHOUT WARRANTY OF ANY KIND, EXPRESS OR
|
|
|
|
* IMPLIED, INCLUDING BUT NOT LIMITED TO THE WARRANTIES OF MERCHANTABILITY,
|
|
|
|
* FITNESS FOR A PARTICULAR PURPOSE AND NONINFRINGEMENT. IN NO EVENT SHALL THE
|
|
|
|
* AUTHORS OR COPYRIGHT HOLDERS BE LIABLE FOR ANY CLAIM, DAMAGES OR OTHER
|
|
|
|
* LIABILITY, WHETHER IN AN ACTION OF CONTRACT, TORT OR OTHERWISE, ARISING FROM,
|
|
|
|
* OUT OF OR IN CONNECTION WITH THE SOFTWARE OR THE USE OR OTHER DEALINGS IN THE
|
|
|
|
* SOFTWARE.
|
|
|
|
*/
|
2005-05-11 07:44:44 +00:00
|
|
|
|
|
|
|
#ifndef __RTSP_CONNECTION_H__
|
|
|
|
#define __RTSP_CONNECTION_H__
|
|
|
|
|
|
|
|
#include <glib.h>
|
|
|
|
|
|
|
|
#include <rtspdefs.h>
|
|
|
|
#include <rtspurl.h>
|
|
|
|
#include <rtspmessage.h>
|
|
|
|
|
|
|
|
G_BEGIN_DECLS
|
|
|
|
|
|
|
|
typedef struct _RTSPConnection
|
|
|
|
{
|
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
|
|
|
/* URL for the connection */
|
|
|
|
RTSPUrl *url;
|
|
|
|
|
|
|
|
/* connection state */
|
gst/rtsp/gstrtspsrc.*: Fix sending RTCP to the right place.
Original commit message from CVS:
* gst/rtsp/gstrtspsrc.c: (gst_rtspsrc_init),
(gst_rtspsrc_finalize), (gst_rtspsrc_stream_configure_transport),
(gst_rtspsrc_handle_request), (gst_rtspsrc_loop_interleaved),
(gst_rtspsrc_send_keep_alive), (gst_rtspsrc_loop_udp),
(gst_rtspsrc_loop_send_cmd), (gst_rtspsrc_try_send),
(gst_rtspsrc_open), (gst_rtspsrc_handle_message):
* gst/rtsp/gstrtspsrc.h:
Fix sending RTCP to the right place.
Fix bug in reffing the wrong UDP element.
Use new pad names for the session manager.
Implement handling server requests in interleaved and UDP modes.
Handle session keep-alive in UDP modes.
Remove GCond for handling UDP timeouts.
* gst/rtsp/rtspconnection.c: (rtsp_connection_connect),
(rtsp_connection_send), (rtsp_connection_read), (read_body),
(rtsp_connection_receive), (rtsp_connection_close):
* gst/rtsp/rtspconnection.h:
Store connection IP address for later.
Add timeout args to all operations that might block forever.
Parse session timeout.
Only close sockets when not already closed.
* gst/rtsp/rtspdefs.c:
* gst/rtsp/rtspdefs.h:
Add timeout return value and error string.
* gst/rtsp/rtspmessage.c: (rtsp_message_init_response):
Add small comment.
2007-05-02 13:32:57 +00:00
|
|
|
gint fd;
|
|
|
|
gint control_sock[2];
|
|
|
|
gchar *ip;
|
2005-05-11 07:44:44 +00:00
|
|
|
|
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
|
|
|
/* Session state */
|
2005-12-06 19:44:58 +00:00
|
|
|
gint cseq; /* sequence number */
|
|
|
|
gchar session_id[512]; /* session id */
|
gst/rtsp/gstrtspsrc.*: Fix sending RTCP to the right place.
Original commit message from CVS:
* gst/rtsp/gstrtspsrc.c: (gst_rtspsrc_init),
(gst_rtspsrc_finalize), (gst_rtspsrc_stream_configure_transport),
(gst_rtspsrc_handle_request), (gst_rtspsrc_loop_interleaved),
(gst_rtspsrc_send_keep_alive), (gst_rtspsrc_loop_udp),
(gst_rtspsrc_loop_send_cmd), (gst_rtspsrc_try_send),
(gst_rtspsrc_open), (gst_rtspsrc_handle_message):
* gst/rtsp/gstrtspsrc.h:
Fix sending RTCP to the right place.
Fix bug in reffing the wrong UDP element.
Use new pad names for the session manager.
Implement handling server requests in interleaved and UDP modes.
Handle session keep-alive in UDP modes.
Remove GCond for handling UDP timeouts.
* gst/rtsp/rtspconnection.c: (rtsp_connection_connect),
(rtsp_connection_send), (rtsp_connection_read), (read_body),
(rtsp_connection_receive), (rtsp_connection_close):
* gst/rtsp/rtspconnection.h:
Store connection IP address for later.
Add timeout args to all operations that might block forever.
Parse session timeout.
Only close sockets when not already closed.
* gst/rtsp/rtspdefs.c:
* gst/rtsp/rtspdefs.h:
Add timeout return value and error string.
* gst/rtsp/rtspmessage.c: (rtsp_message_init_response):
Add small comment.
2007-05-02 13:32:57 +00:00
|
|
|
gint timeout; /* session timeout in seconds */
|
2007-05-18 10:36:12 +00:00
|
|
|
GTimer *timer; /* timeout timer */
|
gst/rtsp/: Implement simple Basic Authentication support so that urls like rtsp://user:pass@hostname/rtspstream work ...
Original commit message from CVS:
* gst/rtsp/gstrtspsrc.c: (gst_rtspsrc_finalize),
(gst_rtspsrc_create_stream), (rtsp_auth_method_to_string),
(gst_rtspsrc_parse_auth_hdr), (gst_rtspsrc_setup_auth),
(gst_rtspsrc_send), (gst_rtspsrc_try_send), (gst_rtspsrc_open),
(gst_rtspsrc_close), (gst_rtspsrc_play), (gst_rtspsrc_pause),
(gst_rtspsrc_uri_set_uri):
* gst/rtsp/gstrtspsrc.h:
* gst/rtsp/rtspconnection.c: (rtsp_connection_create),
(append_auth_header), (rtsp_connection_send),
(rtsp_connection_free), (rtsp_connection_set_auth):
* gst/rtsp/rtspconnection.h:
* gst/rtsp/rtspdefs.h:
* gst/rtsp/rtspurl.c: (rtsp_url_get_request_uri):
* gst/rtsp/rtspurl.h:
Implement simple Basic Authentication support so that urls like
rtsp://user:pass@hostname/rtspstream work on hosts that require
authentication.
2007-02-23 18:12:27 +00:00
|
|
|
|
|
|
|
/* Authentication */
|
2007-05-18 10:36:12 +00:00
|
|
|
RTSPAuthMethod auth_method;
|
|
|
|
gchar *username;
|
|
|
|
gchar *passwd;
|
2005-05-11 07:44:44 +00:00
|
|
|
} RTSPConnection;
|
|
|
|
|
gst/rtsp/: Added README
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/rtsp/rtsp.h:
* gst/rtsp/rtspconnection.c: (rtsp_connection_create),
(rtsp_connection_send), (read_line), (parse_request_line),
(parse_line), (read_body), (rtsp_connection_receive),
(rtsp_connection_free):
* gst/rtsp/rtspconnection.h:
* gst/rtsp/rtspdefs.c: (rtsp_find_method):
* gst/rtsp/rtspdefs.h:
* gst/rtsp/rtspmessage.c: (rtsp_message_set_body),
(rtsp_message_take_body):
* gst/rtsp/rtspmessage.h:
* gst/rtsp/rtspstream.h:
* gst/rtsp/sdpmessage.c: (sdp_parse_line):
Added README
Some cleanups.
2005-05-11 12:01:10 +00:00
|
|
|
/* opening/closing a connection */
|
gst/rtsp/gstrtspsrc.*: Fix sending RTCP to the right place.
Original commit message from CVS:
* gst/rtsp/gstrtspsrc.c: (gst_rtspsrc_init),
(gst_rtspsrc_finalize), (gst_rtspsrc_stream_configure_transport),
(gst_rtspsrc_handle_request), (gst_rtspsrc_loop_interleaved),
(gst_rtspsrc_send_keep_alive), (gst_rtspsrc_loop_udp),
(gst_rtspsrc_loop_send_cmd), (gst_rtspsrc_try_send),
(gst_rtspsrc_open), (gst_rtspsrc_handle_message):
* gst/rtsp/gstrtspsrc.h:
Fix sending RTCP to the right place.
Fix bug in reffing the wrong UDP element.
Use new pad names for the session manager.
Implement handling server requests in interleaved and UDP modes.
Handle session keep-alive in UDP modes.
Remove GCond for handling UDP timeouts.
* gst/rtsp/rtspconnection.c: (rtsp_connection_connect),
(rtsp_connection_send), (rtsp_connection_read), (read_body),
(rtsp_connection_receive), (rtsp_connection_close):
* gst/rtsp/rtspconnection.h:
Store connection IP address for later.
Add timeout args to all operations that might block forever.
Parse session timeout.
Only close sockets when not already closed.
* gst/rtsp/rtspdefs.c:
* gst/rtsp/rtspdefs.h:
Add timeout return value and error string.
* gst/rtsp/rtspmessage.c: (rtsp_message_init_response):
Add small comment.
2007-05-02 13:32:57 +00:00
|
|
|
RTSPResult rtsp_connection_create (RTSPUrl *url, RTSPConnection **conn);
|
|
|
|
RTSPResult rtsp_connection_connect (RTSPConnection *conn, GTimeVal *timeout);
|
|
|
|
RTSPResult rtsp_connection_close (RTSPConnection *conn);
|
|
|
|
RTSPResult rtsp_connection_free (RTSPConnection *conn);
|
2005-05-11 07:44:44 +00:00
|
|
|
|
gst/rtsp/gstrtspsrc.c: Send RTCP messages back to the server over the TCP connection.
Original commit message from CVS:
* gst/rtsp/gstrtspsrc.c: (gst_rtspsrc_handle_src_event),
(gst_rtspsrc_handle_src_query), (gst_rtspsrc_sink_chain),
(gst_rtspsrc_stream_configure_manager),
(gst_rtspsrc_stream_free_udp), (gst_rtspsrc_stream_configure_tcp),
(gst_rtspsrc_stream_configure_mcast),
(gst_rtspsrc_stream_configure_udp),
(gst_rtspsrc_stream_configure_udp_sink),
(gst_rtspsrc_stream_configure_transport):
Send RTCP messages back to the server over the TCP connection.
* gst/rtsp/rtspconnection.c: (rtsp_connection_write),
(rtsp_connection_send), (rtsp_connection_read), (read_body),
(rtsp_connection_receive):
* gst/rtsp/rtspconnection.h:
Factor out and expose lowlevel _write and _read methods.
Implement sending data messages to the server.
2007-05-04 12:31:32 +00:00
|
|
|
/* sending/receiving raw bytes */
|
|
|
|
RTSPResult rtsp_connection_read (RTSPConnection * conn, guint8 * data,
|
|
|
|
guint size, GTimeVal * timeout);
|
|
|
|
RTSPResult rtsp_connection_write (RTSPConnection * conn, const guint8 * data,
|
|
|
|
guint size, GTimeVal * timeout);
|
|
|
|
|
gst/rtsp/: Added README
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/rtsp/rtsp.h:
* gst/rtsp/rtspconnection.c: (rtsp_connection_create),
(rtsp_connection_send), (read_line), (parse_request_line),
(parse_line), (read_body), (rtsp_connection_receive),
(rtsp_connection_free):
* gst/rtsp/rtspconnection.h:
* gst/rtsp/rtspdefs.c: (rtsp_find_method):
* gst/rtsp/rtspdefs.h:
* gst/rtsp/rtspmessage.c: (rtsp_message_set_body),
(rtsp_message_take_body):
* gst/rtsp/rtspmessage.h:
* gst/rtsp/rtspstream.h:
* gst/rtsp/sdpmessage.c: (sdp_parse_line):
Added README
Some cleanups.
2005-05-11 12:01:10 +00:00
|
|
|
/* sending/receiving messages */
|
gst/rtsp/gstrtspsrc.*: Fix sending RTCP to the right place.
Original commit message from CVS:
* gst/rtsp/gstrtspsrc.c: (gst_rtspsrc_init),
(gst_rtspsrc_finalize), (gst_rtspsrc_stream_configure_transport),
(gst_rtspsrc_handle_request), (gst_rtspsrc_loop_interleaved),
(gst_rtspsrc_send_keep_alive), (gst_rtspsrc_loop_udp),
(gst_rtspsrc_loop_send_cmd), (gst_rtspsrc_try_send),
(gst_rtspsrc_open), (gst_rtspsrc_handle_message):
* gst/rtsp/gstrtspsrc.h:
Fix sending RTCP to the right place.
Fix bug in reffing the wrong UDP element.
Use new pad names for the session manager.
Implement handling server requests in interleaved and UDP modes.
Handle session keep-alive in UDP modes.
Remove GCond for handling UDP timeouts.
* gst/rtsp/rtspconnection.c: (rtsp_connection_connect),
(rtsp_connection_send), (rtsp_connection_read), (read_body),
(rtsp_connection_receive), (rtsp_connection_close):
* gst/rtsp/rtspconnection.h:
Store connection IP address for later.
Add timeout args to all operations that might block forever.
Parse session timeout.
Only close sockets when not already closed.
* gst/rtsp/rtspdefs.c:
* gst/rtsp/rtspdefs.h:
Add timeout return value and error string.
* gst/rtsp/rtspmessage.c: (rtsp_message_init_response):
Add small comment.
2007-05-02 13:32:57 +00:00
|
|
|
RTSPResult rtsp_connection_send (RTSPConnection *conn, RTSPMessage *message, GTimeVal *timeout);
|
|
|
|
RTSPResult rtsp_connection_receive (RTSPConnection *conn, RTSPMessage *message, GTimeVal *timeout);
|
2005-05-11 07:44:44 +00:00
|
|
|
|
2007-05-18 10:36:12 +00:00
|
|
|
/* reset the timeout */
|
|
|
|
RTSPResult rtsp_connection_next_timeout (RTSPConnection *conn, GTimeVal *timeout);
|
|
|
|
RTSPResult rtsp_connection_reset_timeout (RTSPConnection *conn);
|
|
|
|
|
gst/rtsp/gstrtspsrc.c: Send RTCP messages back to the server over the TCP connection.
Original commit message from CVS:
* gst/rtsp/gstrtspsrc.c: (gst_rtspsrc_handle_src_event),
(gst_rtspsrc_handle_src_query), (gst_rtspsrc_sink_chain),
(gst_rtspsrc_stream_configure_manager),
(gst_rtspsrc_stream_free_udp), (gst_rtspsrc_stream_configure_tcp),
(gst_rtspsrc_stream_configure_mcast),
(gst_rtspsrc_stream_configure_udp),
(gst_rtspsrc_stream_configure_udp_sink),
(gst_rtspsrc_stream_configure_transport):
Send RTCP messages back to the server over the TCP connection.
* gst/rtsp/rtspconnection.c: (rtsp_connection_write),
(rtsp_connection_send), (rtsp_connection_read), (read_body),
(rtsp_connection_receive):
* gst/rtsp/rtspconnection.h:
Factor out and expose lowlevel _write and _read methods.
Implement sending data messages to the server.
2007-05-04 12:31:32 +00:00
|
|
|
/* flushing state */
|
gst/rtsp/gstrtspsrc.*: Fix sending RTCP to the right place.
Original commit message from CVS:
* gst/rtsp/gstrtspsrc.c: (gst_rtspsrc_init),
(gst_rtspsrc_finalize), (gst_rtspsrc_stream_configure_transport),
(gst_rtspsrc_handle_request), (gst_rtspsrc_loop_interleaved),
(gst_rtspsrc_send_keep_alive), (gst_rtspsrc_loop_udp),
(gst_rtspsrc_loop_send_cmd), (gst_rtspsrc_try_send),
(gst_rtspsrc_open), (gst_rtspsrc_handle_message):
* gst/rtsp/gstrtspsrc.h:
Fix sending RTCP to the right place.
Fix bug in reffing the wrong UDP element.
Use new pad names for the session manager.
Implement handling server requests in interleaved and UDP modes.
Handle session keep-alive in UDP modes.
Remove GCond for handling UDP timeouts.
* gst/rtsp/rtspconnection.c: (rtsp_connection_connect),
(rtsp_connection_send), (rtsp_connection_read), (read_body),
(rtsp_connection_receive), (rtsp_connection_close):
* gst/rtsp/rtspconnection.h:
Store connection IP address for later.
Add timeout args to all operations that might block forever.
Parse session timeout.
Only close sockets when not already closed.
* gst/rtsp/rtspdefs.c:
* gst/rtsp/rtspdefs.h:
Add timeout return value and error string.
* gst/rtsp/rtspmessage.c: (rtsp_message_init_response):
Add small comment.
2007-05-02 13:32:57 +00:00
|
|
|
RTSPResult rtsp_connection_flush (RTSPConnection *conn, gboolean flush);
|
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
|
|
|
|
gst/rtsp/: Implement simple Basic Authentication support so that urls like rtsp://user:pass@hostname/rtspstream work ...
Original commit message from CVS:
* gst/rtsp/gstrtspsrc.c: (gst_rtspsrc_finalize),
(gst_rtspsrc_create_stream), (rtsp_auth_method_to_string),
(gst_rtspsrc_parse_auth_hdr), (gst_rtspsrc_setup_auth),
(gst_rtspsrc_send), (gst_rtspsrc_try_send), (gst_rtspsrc_open),
(gst_rtspsrc_close), (gst_rtspsrc_play), (gst_rtspsrc_pause),
(gst_rtspsrc_uri_set_uri):
* gst/rtsp/gstrtspsrc.h:
* gst/rtsp/rtspconnection.c: (rtsp_connection_create),
(append_auth_header), (rtsp_connection_send),
(rtsp_connection_free), (rtsp_connection_set_auth):
* gst/rtsp/rtspconnection.h:
* gst/rtsp/rtspdefs.h:
* gst/rtsp/rtspurl.c: (rtsp_url_get_request_uri):
* gst/rtsp/rtspurl.h:
Implement simple Basic Authentication support so that urls like
rtsp://user:pass@hostname/rtspstream work on hosts that require
authentication.
2007-02-23 18:12:27 +00:00
|
|
|
/* Configure Authentication data */
|
gst/rtsp/gstrtspsrc.*: Fix sending RTCP to the right place.
Original commit message from CVS:
* gst/rtsp/gstrtspsrc.c: (gst_rtspsrc_init),
(gst_rtspsrc_finalize), (gst_rtspsrc_stream_configure_transport),
(gst_rtspsrc_handle_request), (gst_rtspsrc_loop_interleaved),
(gst_rtspsrc_send_keep_alive), (gst_rtspsrc_loop_udp),
(gst_rtspsrc_loop_send_cmd), (gst_rtspsrc_try_send),
(gst_rtspsrc_open), (gst_rtspsrc_handle_message):
* gst/rtsp/gstrtspsrc.h:
Fix sending RTCP to the right place.
Fix bug in reffing the wrong UDP element.
Use new pad names for the session manager.
Implement handling server requests in interleaved and UDP modes.
Handle session keep-alive in UDP modes.
Remove GCond for handling UDP timeouts.
* gst/rtsp/rtspconnection.c: (rtsp_connection_connect),
(rtsp_connection_send), (rtsp_connection_read), (read_body),
(rtsp_connection_receive), (rtsp_connection_close):
* gst/rtsp/rtspconnection.h:
Store connection IP address for later.
Add timeout args to all operations that might block forever.
Parse session timeout.
Only close sockets when not already closed.
* gst/rtsp/rtspdefs.c:
* gst/rtsp/rtspdefs.h:
Add timeout return value and error string.
* gst/rtsp/rtspmessage.c: (rtsp_message_init_response):
Add small comment.
2007-05-02 13:32:57 +00:00
|
|
|
RTSPResult rtsp_connection_set_auth (RTSPConnection *conn, RTSPAuthMethod method,
|
|
|
|
gchar *user, gchar *pass);
|
gst/rtsp/: Implement simple Basic Authentication support so that urls like rtsp://user:pass@hostname/rtspstream work ...
Original commit message from CVS:
* gst/rtsp/gstrtspsrc.c: (gst_rtspsrc_finalize),
(gst_rtspsrc_create_stream), (rtsp_auth_method_to_string),
(gst_rtspsrc_parse_auth_hdr), (gst_rtspsrc_setup_auth),
(gst_rtspsrc_send), (gst_rtspsrc_try_send), (gst_rtspsrc_open),
(gst_rtspsrc_close), (gst_rtspsrc_play), (gst_rtspsrc_pause),
(gst_rtspsrc_uri_set_uri):
* gst/rtsp/gstrtspsrc.h:
* gst/rtsp/rtspconnection.c: (rtsp_connection_create),
(append_auth_header), (rtsp_connection_send),
(rtsp_connection_free), (rtsp_connection_set_auth):
* gst/rtsp/rtspconnection.h:
* gst/rtsp/rtspdefs.h:
* gst/rtsp/rtspurl.c: (rtsp_url_get_request_uri):
* gst/rtsp/rtspurl.h:
Implement simple Basic Authentication support so that urls like
rtsp://user:pass@hostname/rtspstream work on hosts that require
authentication.
2007-02-23 18:12:27 +00:00
|
|
|
|
2005-05-11 07:44:44 +00:00
|
|
|
G_END_DECLS
|
|
|
|
|
|
|
|
#endif /* __RTSP_CONNECTION_H__ */
|