2005-05-11 07:44:44 +00:00
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/* GStreamer
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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
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* Copyright (C) <2005,2006> Wim Taymans <wim@fluendo.com>
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2005-05-11 07:44:44 +00:00
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*
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* This library is free software; you can redistribute it and/or
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* modify it under the terms of the GNU Library General Public
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* License as published by the Free Software Foundation; either
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* version 2 of the License, or (at your option) any later version.
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*
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* This library is distributed in the hope that it will be useful,
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* but WITHOUT ANY WARRANTY; without even the implied warranty of
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* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
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* Library General Public License for more details.
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*
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* You should have received a copy of the GNU Library General Public
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* License along with this library; if not, write to the
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* Free Software Foundation, Inc., 59 Temple Place - Suite 330,
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* Boston, MA 02111-1307, USA.
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*/
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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
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/*
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* Unless otherwise indicated, Source Code is licensed under MIT license.
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* See further explanation attached in License Statement (distributed in the file
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* LICENSE).
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*
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* Permission is hereby granted, free of charge, to any person obtaining a copy of
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* this software and associated documentation files (the "Software"), to deal in
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* the Software without restriction, including without limitation the rights to
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* use, copy, modify, merge, publish, distribute, sublicense, and/or sell copies
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* of the Software, and to permit persons to whom the Software is furnished to do
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* so, subject to the following conditions:
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*
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* The above copyright notice and this permission notice shall be included in all
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* copies or substantial portions of the Software.
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*
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* THE SOFTWARE IS PROVIDED "AS IS", WITHOUT WARRANTY OF ANY KIND, EXPRESS OR
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* IMPLIED, INCLUDING BUT NOT LIMITED TO THE WARRANTIES OF MERCHANTABILITY,
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* FITNESS FOR A PARTICULAR PURPOSE AND NONINFRINGEMENT. IN NO EVENT SHALL THE
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* AUTHORS OR COPYRIGHT HOLDERS BE LIABLE FOR ANY CLAIM, DAMAGES OR OTHER
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* LIABILITY, WHETHER IN AN ACTION OF CONTRACT, TORT OR OTHERWISE, ARISING FROM,
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* OUT OF OR IN CONNECTION WITH THE SOFTWARE OR THE USE OR OTHER DEALINGS IN THE
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* SOFTWARE.
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*/
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2005-05-11 07:44:44 +00:00
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#ifndef __RTSP_DEFS_H__
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#define __RTSP_DEFS_H__
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#include <glib.h>
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G_BEGIN_DECLS
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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
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#define RTSP_CHECK(stmt, label) \
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if (G_UNLIKELY ((res = (stmt)) != RTSP_OK)) goto label
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2005-05-11 07:44:44 +00:00
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typedef enum {
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2006-07-10 16:41:57 +00:00
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RTSP_OK = 0,
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2005-05-11 07:44:44 +00:00
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/* errors */
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gst/rtsp/gstrtspsrc.*: Rework how the transport string is constructed, try to share channels and udp ports.
Original commit message from CVS:
* gst/rtsp/gstrtspsrc.c: (gst_rtsp_proto_get_type),
(gst_rtspsrc_class_init), (gst_rtspsrc_init),
(gst_rtspsrc_create_stream), (gst_rtspsrc_media_to_caps),
(gst_rtspsrc_alloc_udp_ports),
(gst_rtspsrc_stream_configure_transport), (find_stream_by_channel),
(gst_rtspsrc_push_event), (gst_rtspsrc_loop_interleaved),
(gst_rtspsrc_create_transports_string),
(gst_rtspsrc_configure_transports), (gst_rtspsrc_open),
(gst_rtspsrc_handle_message), (gst_rtspsrc_change_state):
* gst/rtsp/gstrtspsrc.h:
Rework how the transport string is constructed, try to share channels
and udp ports.
Make most of the stuff less dependant on RTP as we are also going to use
it for RDT.
Add support for transport specific session managers.
* gst/rtsp/rtspconnection.c: (rtsp_connection_flush):
Implement _flush().
* gst/rtsp/rtspdefs.c: (rtsp_strresult):
* gst/rtsp/rtspdefs.h:
Add generic error return code.
* gst/rtsp/rtspext.h:
Add support for pluggable tranport strings.
* gst/rtsp/rtspextwms.c: (rtsp_ext_wms_before_send),
(rtsp_ext_wms_after_send), (rtsp_ext_wms_parse_sdp),
(rtsp_ext_wms_get_context):
Detect WMServer and activate the extension.
* gst/rtsp/rtsptransport.c: (rtsp_transport_get_mime),
(rtsp_transport_get_manager), (rtsp_transport_parse):
* gst/rtsp/rtsptransport.h:
Added methods to get mime/manager for certain transports.
2006-10-06 12:55:53 +00:00
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RTSP_ERROR = -1,
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RTSP_EINVAL = -2,
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RTSP_EINTR = -3,
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RTSP_ENOMEM = -4,
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RTSP_ERESOLV = -5,
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RTSP_ENOTIMPL = -6,
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RTSP_ESYS = -7,
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RTSP_EPARSE = -8,
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RTSP_EWSASTART = -9,
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RTSP_EWSAVERSION = -10,
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RTSP_EEOF = -11,
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RTSP_ENET = -12,
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RTSP_ENOTIP = -13,
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2006-07-10 16:41:57 +00:00
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gst/rtsp/gstrtspsrc.*: Rework how the transport string is constructed, try to share channels and udp ports.
Original commit message from CVS:
* gst/rtsp/gstrtspsrc.c: (gst_rtsp_proto_get_type),
(gst_rtspsrc_class_init), (gst_rtspsrc_init),
(gst_rtspsrc_create_stream), (gst_rtspsrc_media_to_caps),
(gst_rtspsrc_alloc_udp_ports),
(gst_rtspsrc_stream_configure_transport), (find_stream_by_channel),
(gst_rtspsrc_push_event), (gst_rtspsrc_loop_interleaved),
(gst_rtspsrc_create_transports_string),
(gst_rtspsrc_configure_transports), (gst_rtspsrc_open),
(gst_rtspsrc_handle_message), (gst_rtspsrc_change_state):
* gst/rtsp/gstrtspsrc.h:
Rework how the transport string is constructed, try to share channels
and udp ports.
Make most of the stuff less dependant on RTP as we are also going to use
it for RDT.
Add support for transport specific session managers.
* gst/rtsp/rtspconnection.c: (rtsp_connection_flush):
Implement _flush().
* gst/rtsp/rtspdefs.c: (rtsp_strresult):
* gst/rtsp/rtspdefs.h:
Add generic error return code.
* gst/rtsp/rtspext.h:
Add support for pluggable tranport strings.
* gst/rtsp/rtspextwms.c: (rtsp_ext_wms_before_send),
(rtsp_ext_wms_after_send), (rtsp_ext_wms_parse_sdp),
(rtsp_ext_wms_get_context):
Detect WMServer and activate the extension.
* gst/rtsp/rtsptransport.c: (rtsp_transport_get_mime),
(rtsp_transport_get_manager), (rtsp_transport_parse):
* gst/rtsp/rtsptransport.h:
Added methods to get mime/manager for certain transports.
2006-10-06 12:55:53 +00:00
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RTSP_ELAST = -14,
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2005-05-11 07:44:44 +00:00
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} RTSPResult;
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typedef enum {
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RTSP_FAM_NONE,
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RTSP_FAM_INET,
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RTSP_FAM_INET6,
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} RTSPFamily;
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typedef enum {
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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
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RTSP_STATE_INVALID,
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2005-05-11 07:44:44 +00:00
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RTSP_STATE_INIT,
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RTSP_STATE_READY,
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RTSP_STATE_PLAYING,
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RTSP_STATE_RECORDING,
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} RTSPState;
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typedef enum {
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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
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RTSP_INVALID = 0,
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2005-12-06 19:44:58 +00:00
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RTSP_DESCRIBE = (1 << 0),
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RTSP_ANNOUNCE = (1 << 1),
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RTSP_GET_PARAMETER = (1 << 2),
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RTSP_OPTIONS = (1 << 3),
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RTSP_PAUSE = (1 << 4),
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RTSP_PLAY = (1 << 5),
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RTSP_RECORD = (1 << 6),
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RTSP_REDIRECT = (1 << 7),
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RTSP_SETUP = (1 << 8),
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RTSP_SET_PARAMETER = (1 << 9),
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RTSP_TEARDOWN = (1 << 10),
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2005-05-11 07:44:44 +00:00
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} RTSPMethod;
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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
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/* Authentication methods, ordered by strength */
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typedef enum {
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RTSP_AUTH_NONE = 0x00,
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RTSP_AUTH_BASIC = 0x01,
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RTSP_AUTH_DIGEST = 0x02
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} RTSPAuthMethod;
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/* Strongest available authentication method */
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#define RTSP_AUTH_MAX RTSP_AUTH_DIGEST
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2005-05-11 07:44:44 +00:00
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typedef enum {
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gst/rtsp/: Allow url to be NULL to be able to use it for server connections.
Original commit message from CVS:
Patch by: Peter Kjellerstedt <pkj at axis com>
* gst/rtsp/COPYING.MIT:
* gst/rtsp/gstrtspsrc.c: (gst_rtspsrc_create_stream),
(gst_rtspsrc_stream_free), (gst_rtspsrc_cleanup),
(gst_rtspsrc_alloc_udp_ports), (pad_unblocked), (pad_blocked),
(gst_rtspsrc_stream_configure_transport),
(gst_rtspsrc_activate_streams), (gst_rtspsrc_loop_interleaved),
(gst_rtspsrc_loop_udp), (gst_rtspsrc_send),
(gst_rtspsrc_parse_methods),
(gst_rtspsrc_create_transports_string),
(gst_rtspsrc_prepare_transports), (gst_rtspsrc_setup_streams),
(gst_rtspsrc_open), (gst_rtspsrc_close):
* gst/rtsp/gstrtspsrc.h:
* gst/rtsp/rtspconnection.c: (rtsp_connection_create),
(rtsp_connection_connect), (rtsp_connection_send), (read_line),
(parse_request_line), (parse_line), (rtsp_connection_read),
(rtsp_connection_close):
* gst/rtsp/rtspdefs.c: (rtsp_init_status), (rtsp_strresult),
(rtsp_method_as_text), (rtsp_header_as_text),
(rtsp_status_as_text), (rtsp_find_header_field),
(rtsp_find_method):
* gst/rtsp/rtspdefs.h:
* gst/rtsp/rtspextwms.c: (rtsp_ext_wms_after_send),
(rtsp_ext_wms_configure_stream):
* gst/rtsp/rtspmessage.c: (rtsp_message_new), (rtsp_message_init),
(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_get_header), (rtsp_message_set_body),
(rtsp_message_get_body), (dump_mem), (rtsp_message_dump):
* gst/rtsp/rtspmessage.h:
* gst/rtsp/sdpmessage.c: (sdp_message_get_attribute_val_n),
(sdp_media_get_attribute_val_n), (read_string), (read_string_del),
(sdp_parse_line), (sdp_message_parse_buffer), (print_media),
(sdp_message_dump):
Allow url to be NULL to be able to use it for server connections.
Can now send responses as well as requests.
No longer hangs in an endless loop if EOF is received.
Can now convert a status code to a text string.
Return RTSP_HDR_INVALID for unknown headers.
Return RTSP_INVALID for unknown methods.
Copy CSeq and Session headers from the request.
Only free memory corresponding to the currently set message type.
Added const to function arguments as appropriate.
Avoid a compiler warning when initializing nmedia.
Use guint rather than gint to avoid compiler warnings.
Fix crasher in wms extension.
Factor out stream setup from open_connection.
Delay activation of streams when actual data is received from the
server, this prepares us to do proper protocol switching.
Added new license.
Fixes #380895.
2007-01-10 15:19:48 +00:00
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RTSP_HDR_INVALID,
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2005-05-11 07:44:44 +00:00
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/*
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* R = Request
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* r = response
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* g = general
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* e = entity
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*/
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2005-12-06 19:44:58 +00:00
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RTSP_HDR_ACCEPT, /* Accept R opt. entity */
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RTSP_HDR_ACCEPT_ENCODING, /* Accept-Encoding R opt. entity */
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RTSP_HDR_ACCEPT_LANGUAGE, /* Accept-Language R opt. all */
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RTSP_HDR_ALLOW, /* Allow r opt. all */
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RTSP_HDR_AUTHORIZATION, /* Authorization R opt. all */
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RTSP_HDR_BANDWIDTH, /* Bandwidth R opt. all */
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RTSP_HDR_BLOCKSIZE, /* Blocksize R opt. all but OPTIONS, TEARDOWN */
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RTSP_HDR_CACHE_CONTROL, /* Cache-Control g opt. SETUP */
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RTSP_HDR_CONFERENCE, /* Conference R opt. SETUP */
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RTSP_HDR_CONNECTION, /* Connection g req. all */
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RTSP_HDR_CONTENT_BASE, /* Content-Base e opt. entity */
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RTSP_HDR_CONTENT_ENCODING, /* Content-Encoding e req. SET_PARAMETER, DESCRIBE, ANNOUNCE */
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RTSP_HDR_CONTENT_LANGUAGE, /* Content-Language e req. DESCRIBE, ANNOUNCE */
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RTSP_HDR_CONTENT_LENGTH, /* Content-Length e req. SET_PARAMETER, ANNOUNCE, entity */
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RTSP_HDR_CONTENT_LOCATION, /* Content-Location e opt. entity */
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RTSP_HDR_CONTENT_TYPE, /* Content-Type e req. SET_PARAMETER, ANNOUNCE, entity */
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RTSP_HDR_CSEQ, /* CSeq g req. all */
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RTSP_HDR_DATE, /* Date g opt. all */
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RTSP_HDR_EXPIRES, /* Expires e opt. DESCRIBE, ANNOUNCE */
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RTSP_HDR_FROM, /* From R opt. all */
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RTSP_HDR_IF_MODIFIED_SINCE, /* If-Modified-Since R opt. DESCRIBE, SETUP */
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RTSP_HDR_LAST_MODIFIED, /* Last-Modified e opt. entity */
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RTSP_HDR_PROXY_AUTHENTICATE, /* Proxy-Authenticate */
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RTSP_HDR_PROXY_REQUIRE, /* Proxy-Require R req. all */
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RTSP_HDR_PUBLIC, /* Public r opt. all */
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RTSP_HDR_RANGE, /* Range Rr opt. PLAY, PAUSE, RECORD */
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RTSP_HDR_REFERER, /* Referer R opt. all */
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RTSP_HDR_REQUIRE, /* Require R req. all */
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RTSP_HDR_RETRY_AFTER, /* Retry-After r opt. all */
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RTSP_HDR_RTP_INFO, /* RTP-Info r req. PLAY */
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RTSP_HDR_SCALE, /* Scale Rr opt. PLAY, RECORD */
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RTSP_HDR_SESSION, /* Session Rr req. all but SETUP, OPTIONS */
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RTSP_HDR_SERVER, /* Server r opt. all */
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RTSP_HDR_SPEED, /* Speed Rr opt. PLAY */
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RTSP_HDR_TRANSPORT, /* Transport Rr req. SETUP */
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RTSP_HDR_UNSUPPORTED, /* Unsupported r req. all */
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RTSP_HDR_USER_AGENT, /* User-Agent R opt. all */
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RTSP_HDR_VIA, /* Via g opt. all */
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RTSP_HDR_WWW_AUTHENTICATE, /* WWW-Authenticate r opt. all */
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2005-05-11 07:44:44 +00:00
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gst/rtsp/: Factor out extension in separate module.
Original commit message from CVS:
* gst/rtsp/Makefile.am:
* gst/rtsp/gstrtpdec.c: (gst_rtpdec_getcaps),
(gst_rtpdec_chain_rtp), (gst_rtpdec_chain_rtcp):
* gst/rtsp/gstrtspsrc.c: (gst_rtspsrc_init),
(gst_rtspsrc_finalize), (gst_rtspsrc_create_stream),
(gst_rtspsrc_parse_rtpmap),
(gst_rtspsrc_stream_configure_transport), (gst_rtspsrc_send),
(gst_rtspsrc_parse_methods), (gst_rtspsrc_open),
(gst_rtspsrc_play), (gst_rtspsrc_handle_message):
* gst/rtsp/gstrtspsrc.h:
* gst/rtsp/rtspdefs.c: (rtsp_strresult):
* gst/rtsp/rtspdefs.h:
* gst/rtsp/rtspext.h:
* gst/rtsp/rtspextwms.c: (rtsp_ext_wms_parse_sdp),
(rtsp_ext_wms_get_context):
* gst/rtsp/rtspextwms.h:
* gst/rtsp/rtsptransport.c: (rtsp_transport_init), (parse_mode),
(rtsp_transport_parse):
* gst/rtsp/rtsptransport.h:
Factor out extension in separate module.
Fix getcaps to filter against the padtemplate.
Use Content-Base if the server gives one.
Rework the transport parsing a bit for future extensions.
Added some Real Header field definitions.
2006-10-04 17:24:40 +00:00
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/* Real extensions */
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RTSP_HDR_CLIENT_CHALLENGE, /* ClientChallenge */
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RTSP_HDR_REAL_CHALLENGE1, /* RealChallenge1 */
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RTSP_HDR_REAL_CHALLENGE2, /* RealChallenge2 */
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2006-10-11 16:21:53 +00:00
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RTSP_HDR_REAL_CHALLENGE3, /* RealChallenge3 */
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gst/rtsp/: Factor out extension in separate module.
Original commit message from CVS:
* gst/rtsp/Makefile.am:
* gst/rtsp/gstrtpdec.c: (gst_rtpdec_getcaps),
(gst_rtpdec_chain_rtp), (gst_rtpdec_chain_rtcp):
* gst/rtsp/gstrtspsrc.c: (gst_rtspsrc_init),
(gst_rtspsrc_finalize), (gst_rtspsrc_create_stream),
(gst_rtspsrc_parse_rtpmap),
(gst_rtspsrc_stream_configure_transport), (gst_rtspsrc_send),
(gst_rtspsrc_parse_methods), (gst_rtspsrc_open),
(gst_rtspsrc_play), (gst_rtspsrc_handle_message):
* gst/rtsp/gstrtspsrc.h:
* gst/rtsp/rtspdefs.c: (rtsp_strresult):
* gst/rtsp/rtspdefs.h:
* gst/rtsp/rtspext.h:
* gst/rtsp/rtspextwms.c: (rtsp_ext_wms_parse_sdp),
(rtsp_ext_wms_get_context):
* gst/rtsp/rtspextwms.h:
* gst/rtsp/rtsptransport.c: (rtsp_transport_init), (parse_mode),
(rtsp_transport_parse):
* gst/rtsp/rtsptransport.h:
Factor out extension in separate module.
Fix getcaps to filter against the padtemplate.
Use Content-Base if the server gives one.
Rework the transport parsing a bit for future extensions.
Added some Real Header field definitions.
2006-10-04 17:24:40 +00:00
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RTSP_HDR_SUBSCRIBE, /* Subscribe */
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2006-10-11 16:21:53 +00:00
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RTSP_HDR_ALERT, /* Alert */
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RTSP_HDR_CLIENT_ID, /* ClientID */
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RTSP_HDR_COMPANY_ID, /* CompanyID */
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RTSP_HDR_GUID, /* GUID */
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RTSP_HDR_REGION_DATA, /* RegionData */
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RTSP_HDR_MAX_ASM_WIDTH, /* SupportsMaximumASMBandwidth */
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RTSP_HDR_LANGUAGE, /* Language */
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RTSP_HDR_PLAYER_START_TIME, /* PlayerStarttime */
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gst/rtsp/: Factor out extension in separate module.
Original commit message from CVS:
* gst/rtsp/Makefile.am:
* gst/rtsp/gstrtpdec.c: (gst_rtpdec_getcaps),
(gst_rtpdec_chain_rtp), (gst_rtpdec_chain_rtcp):
* gst/rtsp/gstrtspsrc.c: (gst_rtspsrc_init),
(gst_rtspsrc_finalize), (gst_rtspsrc_create_stream),
(gst_rtspsrc_parse_rtpmap),
(gst_rtspsrc_stream_configure_transport), (gst_rtspsrc_send),
(gst_rtspsrc_parse_methods), (gst_rtspsrc_open),
(gst_rtspsrc_play), (gst_rtspsrc_handle_message):
* gst/rtsp/gstrtspsrc.h:
* gst/rtsp/rtspdefs.c: (rtsp_strresult):
* gst/rtsp/rtspdefs.h:
* gst/rtsp/rtspext.h:
* gst/rtsp/rtspextwms.c: (rtsp_ext_wms_parse_sdp),
(rtsp_ext_wms_get_context):
* gst/rtsp/rtspextwms.h:
* gst/rtsp/rtsptransport.c: (rtsp_transport_init), (parse_mode),
(rtsp_transport_parse):
* gst/rtsp/rtsptransport.h:
Factor out extension in separate module.
Fix getcaps to filter against the padtemplate.
Use Content-Base if the server gives one.
Rework the transport parsing a bit for future extensions.
Added some Real Header field definitions.
2006-10-04 17:24:40 +00:00
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2005-05-11 07:44:44 +00:00
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} RTSPHeaderField;
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typedef enum {
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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
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RTSP_STS_INVALID = 0,
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2005-12-06 19:44:58 +00:00
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RTSP_STS_CONTINUE = 100,
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RTSP_STS_OK = 200,
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RTSP_STS_CREATED = 201,
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RTSP_STS_LOW_ON_STORAGE = 250,
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RTSP_STS_MULTIPLE_CHOICES = 300,
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RTSP_STS_MOVED_PERMANENTLY = 301,
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RTSP_STS_MOVE_TEMPORARILY = 302,
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RTSP_STS_SEE_OTHER = 303,
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RTSP_STS_NOT_MODIFIED = 304,
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RTSP_STS_USE_PROXY = 305,
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RTSP_STS_BAD_REQUEST = 400,
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RTSP_STS_UNAUTHORIZED = 401,
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RTSP_STS_PAYMENT_REQUIRED = 402,
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RTSP_STS_FORBIDDEN = 403,
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RTSP_STS_NOT_FOUND = 404,
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RTSP_STS_METHOD_NOT_ALLOWED = 405,
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RTSP_STS_NOT_ACCEPTABLE = 406,
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RTSP_STS_PROXY_AUTH_REQUIRED = 407,
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RTSP_STS_REQUEST_TIMEOUT = 408,
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RTSP_STS_GONE = 410,
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RTSP_STS_LENGTH_REQUIRED = 411,
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RTSP_STS_PRECONDITION_FAILED = 412,
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RTSP_STS_REQUEST_ENTITY_TOO_LARGE = 413,
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RTSP_STS_REQUEST_URI_TOO_LARGE = 414,
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RTSP_STS_UNSUPPORTED_MEDIA_TYPE = 415,
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RTSP_STS_PARAMETER_NOT_UNDERSTOOD = 451,
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RTSP_STS_CONFERENCE_NOT_FOUND = 452,
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RTSP_STS_NOT_ENOUGH_BANDWIDTH = 453,
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RTSP_STS_SESSION_NOT_FOUND = 454,
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RTSP_STS_METHOD_NOT_VALID_IN_THIS_STATE = 455,
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RTSP_STS_HEADER_FIELD_NOT_VALID_FOR_RESOURCE = 456,
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RTSP_STS_INVALID_RANGE = 457,
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RTSP_STS_PARAMETER_IS_READONLY = 458,
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RTSP_STS_AGGREGATE_OPERATION_NOT_ALLOWED = 459,
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RTSP_STS_ONLY_AGGREGATE_OPERATION_ALLOWED = 460,
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RTSP_STS_UNSUPPORTED_TRANSPORT = 461,
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RTSP_STS_DESTINATION_UNREACHABLE = 462,
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RTSP_STS_INTERNAL_SERVER_ERROR = 500,
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RTSP_STS_NOT_IMPLEMENTED = 501,
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RTSP_STS_BAD_GATEWAY = 502,
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RTSP_STS_SERVICE_UNAVAILABLE = 503,
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RTSP_STS_GATEWAY_TIMEOUT = 504,
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RTSP_STS_RTSP_VERSION_NOT_SUPPORTED = 505,
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RTSP_STS_OPTION_NOT_SUPPORTED = 551,
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2005-05-11 07:44:44 +00:00
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} RTSPStatusCode;
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2006-09-23 15:31:56 +00:00
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gchar* rtsp_strresult (RTSPResult result);
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2005-12-06 19:44:58 +00:00
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const gchar* rtsp_method_as_text (RTSPMethod method);
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const gchar* rtsp_header_as_text (RTSPHeaderField field);
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const gchar* rtsp_status_as_text (RTSPStatusCode code);
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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
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2005-12-06 19:44:58 +00:00
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RTSPHeaderField rtsp_find_header_field (gchar *header);
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RTSPMethod rtsp_find_method (gchar *method);
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2005-05-11 07:44:44 +00:00
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G_END_DECLS
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#endif /* __RTSP_DEFS_H__ */
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