2005-05-11 07:44:44 +00:00
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/* GStreamer
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2007-05-14 09:01:05 +00:00
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* Copyright (C) <2005,2006,2007> Wim Taymans <wim@fluendo.com>
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* <2007> Peter Kjellerstedt <pkj at axis 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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2005-06-29 16:27:27 +00:00
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#include <string.h>
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#include <stdlib.h>
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2005-05-11 07:44:44 +00:00
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#include "rtsptransport.h"
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gst/rtsp/: Morph RTPDec into something compatible with RTPBin as a fallback.
Original commit message from CVS:
* gst/rtsp/Makefile.am:
* gst/rtsp/gstrtpdec.c: (find_session_by_id), (create_session),
(free_session), (gst_rtp_dec_base_init), (gst_rtp_dec_class_init),
(gst_rtp_dec_init), (gst_rtp_dec_finalize),
(gst_rtp_dec_query_src), (gst_rtp_dec_chain_rtp),
(gst_rtp_dec_chain_rtcp), (gst_rtp_dec_set_property),
(gst_rtp_dec_get_property), (gst_rtp_dec_provide_clock),
(gst_rtp_dec_change_state), (create_recv_rtp), (create_recv_rtcp),
(create_rtcp), (gst_rtp_dec_request_new_pad),
(gst_rtp_dec_release_pad):
* gst/rtsp/gstrtpdec.h:
* gst/rtsp/gstrtsp.c: (plugin_init):
Morph RTPDec into something compatible with RTPBin as a fallback.
Various other style fixes.
* gst/rtsp/gstrtspsrc.c: (find_stream_by_id),
(find_stream_by_udpsrc), (gst_rtspsrc_stream_free),
(gst_rtspsrc_cleanup), (gst_rtspsrc_media_to_caps),
(new_session_pad), (gst_rtspsrc_stream_configure_transport),
(gst_rtspsrc_activate_streams), (gst_rtspsrc_loop_interleaved),
(gst_rtspsrc_loop_udp), (gst_rtspsrc_setup_auth),
(gst_rtspsrc_handle_message), (gst_rtspsrc_change_state):
* gst/rtsp/gstrtspsrc.h:
Implement RTPBin session manager handling.
Don't try to add empty properties to caps.
Implement fallback session manager, handling.
Don't combine errors from RTCP streams, just ignore them.
* gst/rtsp/rtsptransport.c: (rtsp_transport_get_manager):
* gst/rtsp/rtsptransport.h:
Implement fallback session manager.
Make RTPBin the default one when available.
2007-04-06 12:54:16 +00:00
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#define MAX_MANAGERS 2
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gst/rtsp/rtsptransport.*: Add validation to rtsp_transport_parse().
Original commit message from CVS:
Patch by: Peter Kjellerstedt <pkj at axis com>
* gst/rtsp/rtsptransport.c: (rtsp_transport_init), (parse_mode),
(parse_range), (range_as_text), (rtsp_transport_mode_as_text),
(rtsp_transport_profile_as_text), (rtsp_transport_ltrans_as_text),
(rtsp_transport_parse), (rtsp_transport_as_text):
* gst/rtsp/rtsptransport.h:
Add validation to rtsp_transport_parse().
Add rtsp_transport_as_text() to generate an RTSP header from an
RTSPTransport.
Change ssrc to guint (was a string) since that is what it is, even
though it is sent as a hex string.
Correctly identify PLAY|RECORD mode parameters (the syntax in the RFC is
incorrect, which can be seen when looking at the examples in the RFC).
Fixes #437670.
2007-05-12 16:26:06 +00:00
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typedef enum
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{
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RTSP_TRANSPORT_DELIVERY = 1 << 0, /* multicast | unicast */
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RTSP_TRANSPORT_DESTINATION = 1 << 1,
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RTSP_TRANSPORT_SOURCE = 1 << 2,
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RTSP_TRANSPORT_INTERLEAVED = 1 << 3,
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RTSP_TRANSPORT_APPEND = 1 << 4,
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RTSP_TRANSPORT_TTL = 1 << 5,
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RTSP_TRANSPORT_LAYERS = 1 << 6,
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RTSP_TRANSPORT_PORT = 1 << 7,
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RTSP_TRANSPORT_CLIENT_PORT = 1 << 8,
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RTSP_TRANSPORT_SERVER_PORT = 1 << 9,
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RTSP_TRANSPORT_SSRC = 1 << 10,
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RTSP_TRANSPORT_MODE = 1 << 11,
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} RTSPTransportParameter;
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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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typedef struct
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{
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const gchar *name;
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const RTSPTransMode mode;
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const gchar *gst_mime;
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gst/rtsp/: Morph RTPDec into something compatible with RTPBin as a fallback.
Original commit message from CVS:
* gst/rtsp/Makefile.am:
* gst/rtsp/gstrtpdec.c: (find_session_by_id), (create_session),
(free_session), (gst_rtp_dec_base_init), (gst_rtp_dec_class_init),
(gst_rtp_dec_init), (gst_rtp_dec_finalize),
(gst_rtp_dec_query_src), (gst_rtp_dec_chain_rtp),
(gst_rtp_dec_chain_rtcp), (gst_rtp_dec_set_property),
(gst_rtp_dec_get_property), (gst_rtp_dec_provide_clock),
(gst_rtp_dec_change_state), (create_recv_rtp), (create_recv_rtcp),
(create_rtcp), (gst_rtp_dec_request_new_pad),
(gst_rtp_dec_release_pad):
* gst/rtsp/gstrtpdec.h:
* gst/rtsp/gstrtsp.c: (plugin_init):
Morph RTPDec into something compatible with RTPBin as a fallback.
Various other style fixes.
* gst/rtsp/gstrtspsrc.c: (find_stream_by_id),
(find_stream_by_udpsrc), (gst_rtspsrc_stream_free),
(gst_rtspsrc_cleanup), (gst_rtspsrc_media_to_caps),
(new_session_pad), (gst_rtspsrc_stream_configure_transport),
(gst_rtspsrc_activate_streams), (gst_rtspsrc_loop_interleaved),
(gst_rtspsrc_loop_udp), (gst_rtspsrc_setup_auth),
(gst_rtspsrc_handle_message), (gst_rtspsrc_change_state):
* gst/rtsp/gstrtspsrc.h:
Implement RTPBin session manager handling.
Don't try to add empty properties to caps.
Implement fallback session manager, handling.
Don't combine errors from RTCP streams, just ignore them.
* gst/rtsp/rtsptransport.c: (rtsp_transport_get_manager):
* gst/rtsp/rtsptransport.h:
Implement fallback session manager.
Make RTPBin the default one when available.
2007-04-06 12:54:16 +00:00
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const gchar *manager[MAX_MANAGERS];
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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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} RTSPTransMap;
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static const RTSPTransMap transports[] = {
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gst/rtsp/: Morph RTPDec into something compatible with RTPBin as a fallback.
Original commit message from CVS:
* gst/rtsp/Makefile.am:
* gst/rtsp/gstrtpdec.c: (find_session_by_id), (create_session),
(free_session), (gst_rtp_dec_base_init), (gst_rtp_dec_class_init),
(gst_rtp_dec_init), (gst_rtp_dec_finalize),
(gst_rtp_dec_query_src), (gst_rtp_dec_chain_rtp),
(gst_rtp_dec_chain_rtcp), (gst_rtp_dec_set_property),
(gst_rtp_dec_get_property), (gst_rtp_dec_provide_clock),
(gst_rtp_dec_change_state), (create_recv_rtp), (create_recv_rtcp),
(create_rtcp), (gst_rtp_dec_request_new_pad),
(gst_rtp_dec_release_pad):
* gst/rtsp/gstrtpdec.h:
* gst/rtsp/gstrtsp.c: (plugin_init):
Morph RTPDec into something compatible with RTPBin as a fallback.
Various other style fixes.
* gst/rtsp/gstrtspsrc.c: (find_stream_by_id),
(find_stream_by_udpsrc), (gst_rtspsrc_stream_free),
(gst_rtspsrc_cleanup), (gst_rtspsrc_media_to_caps),
(new_session_pad), (gst_rtspsrc_stream_configure_transport),
(gst_rtspsrc_activate_streams), (gst_rtspsrc_loop_interleaved),
(gst_rtspsrc_loop_udp), (gst_rtspsrc_setup_auth),
(gst_rtspsrc_handle_message), (gst_rtspsrc_change_state):
* gst/rtsp/gstrtspsrc.h:
Implement RTPBin session manager handling.
Don't try to add empty properties to caps.
Implement fallback session manager, handling.
Don't combine errors from RTCP streams, just ignore them.
* gst/rtsp/rtsptransport.c: (rtsp_transport_get_manager):
* gst/rtsp/rtsptransport.h:
Implement fallback session manager.
Make RTPBin the default one when available.
2007-04-06 12:54:16 +00:00
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{"rtp", RTSP_TRANS_RTP, "application/x-rtp", {"rtpbin", "rtpdec"}},
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{"x-real-rdt", RTSP_TRANS_RDT, "application/x-rdt", {NULL, NULL}},
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{"x-pn-tng", RTSP_TRANS_RDT, "application/x-rdt", {NULL, NULL}},
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{NULL, RTSP_TRANS_UNKNOWN, NULL, {NULL, NULL}}
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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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};
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typedef struct
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{
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const gchar *name;
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const RTSPProfile profile;
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} RTSPProfileMap;
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static const RTSPProfileMap profiles[] = {
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{"avp", RTSP_PROFILE_AVP},
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{"savp", RTSP_PROFILE_SAVP},
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{NULL, RTSP_PROFILE_UNKNOWN}
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};
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typedef struct
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{
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const gchar *name;
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const RTSPLowerTrans ltrans;
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} RTSPLTransMap;
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static const RTSPLTransMap ltrans[] = {
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{"udp", RTSP_LOWER_TRANS_UDP},
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{"mcast", RTSP_LOWER_TRANS_UDP_MCAST},
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{"tcp", RTSP_LOWER_TRANS_TCP},
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gst/rtsp/rtsptransport.*: Add validation to rtsp_transport_parse().
Original commit message from CVS:
Patch by: Peter Kjellerstedt <pkj at axis com>
* gst/rtsp/rtsptransport.c: (rtsp_transport_init), (parse_mode),
(parse_range), (range_as_text), (rtsp_transport_mode_as_text),
(rtsp_transport_profile_as_text), (rtsp_transport_ltrans_as_text),
(rtsp_transport_parse), (rtsp_transport_as_text):
* gst/rtsp/rtsptransport.h:
Add validation to rtsp_transport_parse().
Add rtsp_transport_as_text() to generate an RTSP header from an
RTSPTransport.
Change ssrc to guint (was a string) since that is what it is, even
though it is sent as a hex string.
Correctly identify PLAY|RECORD mode parameters (the syntax in the RFC is
incorrect, which can be seen when looking at the examples in the RFC).
Fixes #437670.
2007-05-12 16:26:06 +00:00
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{NULL, RTSP_LOWER_TRANS_UNKNOWN}
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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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};
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gst/rtsp/rtsptransport.*: Add validation to rtsp_transport_parse().
Original commit message from CVS:
Patch by: Peter Kjellerstedt <pkj at axis com>
* gst/rtsp/rtsptransport.c: (rtsp_transport_init), (parse_mode),
(parse_range), (range_as_text), (rtsp_transport_mode_as_text),
(rtsp_transport_profile_as_text), (rtsp_transport_ltrans_as_text),
(rtsp_transport_parse), (rtsp_transport_as_text):
* gst/rtsp/rtsptransport.h:
Add validation to rtsp_transport_parse().
Add rtsp_transport_as_text() to generate an RTSP header from an
RTSPTransport.
Change ssrc to guint (was a string) since that is what it is, even
though it is sent as a hex string.
Correctly identify PLAY|RECORD mode parameters (the syntax in the RFC is
incorrect, which can be seen when looking at the examples in the RFC).
Fixes #437670.
2007-05-12 16:26:06 +00:00
|
|
|
#define RTSP_TRANSPORT_PARAMETER_IS_UNIQUE(param) \
|
|
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G_STMT_START { \
|
|
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if ((transport_params & (param)) != 0) \
|
|
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goto invalid_transport; \
|
|
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transport_params |= (param); \
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|
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} G_STMT_END
|
|
|
|
|
2005-05-11 07:44:44 +00:00
|
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RTSPResult
|
|
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|
rtsp_transport_new (RTSPTransport ** transport)
|
|
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|
{
|
|
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|
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
|
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g_return_val_if_fail (transport != NULL, RTSP_EINVAL);
|
2005-05-11 07:44:44 +00:00
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trans = g_new0 (RTSPTransport, 1);
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|
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*transport = trans;
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|
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|
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return rtsp_transport_init (trans);
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|
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}
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|
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RTSPResult
|
|
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|
rtsp_transport_init (RTSPTransport * transport)
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|
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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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g_return_val_if_fail (transport != NULL, RTSP_EINVAL);
|
|
|
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|
2005-05-11 07:44:44 +00:00
|
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g_free (transport->destination);
|
|
|
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g_free (transport->source);
|
|
|
|
|
|
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memset (transport, 0, sizeof (RTSPTransport));
|
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|
|
|
|
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transport->trans = RTSP_TRANS_RTP;
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|
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transport->profile = RTSP_PROFILE_AVP;
|
gst/rtsp/rtsptransport.*: Add validation to rtsp_transport_parse().
Original commit message from CVS:
Patch by: Peter Kjellerstedt <pkj at axis com>
* gst/rtsp/rtsptransport.c: (rtsp_transport_init), (parse_mode),
(parse_range), (range_as_text), (rtsp_transport_mode_as_text),
(rtsp_transport_profile_as_text), (rtsp_transport_ltrans_as_text),
(rtsp_transport_parse), (rtsp_transport_as_text):
* gst/rtsp/rtsptransport.h:
Add validation to rtsp_transport_parse().
Add rtsp_transport_as_text() to generate an RTSP header from an
RTSPTransport.
Change ssrc to guint (was a string) since that is what it is, even
though it is sent as a hex string.
Correctly identify PLAY|RECORD mode parameters (the syntax in the RFC is
incorrect, which can be seen when looking at the examples in the RFC).
Fixes #437670.
2007-05-12 16:26:06 +00:00
|
|
|
transport->lower_transport = RTSP_LOWER_TRANS_UDP_MCAST;
|
2005-05-11 07:44:44 +00:00
|
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|
transport->mode_play = TRUE;
|
|
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transport->mode_record = FALSE;
|
gst/rtsp/rtsptransport.*: Add validation to rtsp_transport_parse().
Original commit message from CVS:
Patch by: Peter Kjellerstedt <pkj at axis com>
* gst/rtsp/rtsptransport.c: (rtsp_transport_init), (parse_mode),
(parse_range), (range_as_text), (rtsp_transport_mode_as_text),
(rtsp_transport_profile_as_text), (rtsp_transport_ltrans_as_text),
(rtsp_transport_parse), (rtsp_transport_as_text):
* gst/rtsp/rtsptransport.h:
Add validation to rtsp_transport_parse().
Add rtsp_transport_as_text() to generate an RTSP header from an
RTSPTransport.
Change ssrc to guint (was a string) since that is what it is, even
though it is sent as a hex string.
Correctly identify PLAY|RECORD mode parameters (the syntax in the RFC is
incorrect, which can be seen when looking at the examples in the RFC).
Fixes #437670.
2007-05-12 16:26:06 +00:00
|
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transport->interleaved.min = -1;
|
|
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transport->interleaved.max = -1;
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transport->port.min = -1;
|
|
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transport->port.max = -1;
|
|
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transport->client_port.min = -1;
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|
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transport->client_port.max = -1;
|
|
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transport->server_port.min = -1;
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transport->server_port.max = -1;
|
2005-05-11 07:44:44 +00:00
|
|
|
|
|
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return RTSP_OK;
|
|
|
|
}
|
|
|
|
|
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
|
|
|
RTSPResult
|
|
|
|
rtsp_transport_get_mime (RTSPTransMode trans, const gchar ** mime)
|
|
|
|
{
|
|
|
|
gint i;
|
|
|
|
|
|
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g_return_val_if_fail (mime != NULL, RTSP_EINVAL);
|
|
|
|
|
|
|
|
for (i = 0; transports[i].name; i++)
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|
|
|
if (transports[i].mode == trans)
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|
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|
break;
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|
|
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*mime = transports[i].gst_mime;
|
|
|
|
|
|
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|
return RTSP_OK;
|
|
|
|
}
|
|
|
|
|
|
|
|
RTSPResult
|
gst/rtsp/: Morph RTPDec into something compatible with RTPBin as a fallback.
Original commit message from CVS:
* gst/rtsp/Makefile.am:
* gst/rtsp/gstrtpdec.c: (find_session_by_id), (create_session),
(free_session), (gst_rtp_dec_base_init), (gst_rtp_dec_class_init),
(gst_rtp_dec_init), (gst_rtp_dec_finalize),
(gst_rtp_dec_query_src), (gst_rtp_dec_chain_rtp),
(gst_rtp_dec_chain_rtcp), (gst_rtp_dec_set_property),
(gst_rtp_dec_get_property), (gst_rtp_dec_provide_clock),
(gst_rtp_dec_change_state), (create_recv_rtp), (create_recv_rtcp),
(create_rtcp), (gst_rtp_dec_request_new_pad),
(gst_rtp_dec_release_pad):
* gst/rtsp/gstrtpdec.h:
* gst/rtsp/gstrtsp.c: (plugin_init):
Morph RTPDec into something compatible with RTPBin as a fallback.
Various other style fixes.
* gst/rtsp/gstrtspsrc.c: (find_stream_by_id),
(find_stream_by_udpsrc), (gst_rtspsrc_stream_free),
(gst_rtspsrc_cleanup), (gst_rtspsrc_media_to_caps),
(new_session_pad), (gst_rtspsrc_stream_configure_transport),
(gst_rtspsrc_activate_streams), (gst_rtspsrc_loop_interleaved),
(gst_rtspsrc_loop_udp), (gst_rtspsrc_setup_auth),
(gst_rtspsrc_handle_message), (gst_rtspsrc_change_state):
* gst/rtsp/gstrtspsrc.h:
Implement RTPBin session manager handling.
Don't try to add empty properties to caps.
Implement fallback session manager, handling.
Don't combine errors from RTCP streams, just ignore them.
* gst/rtsp/rtsptransport.c: (rtsp_transport_get_manager):
* gst/rtsp/rtsptransport.h:
Implement fallback session manager.
Make RTPBin the default one when available.
2007-04-06 12:54:16 +00:00
|
|
|
rtsp_transport_get_manager (RTSPTransMode trans, const gchar ** manager,
|
|
|
|
guint option)
|
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
|
|
|
{
|
|
|
|
gint i;
|
|
|
|
|
|
|
|
g_return_val_if_fail (manager != NULL, RTSP_EINVAL);
|
|
|
|
|
|
|
|
for (i = 0; transports[i].name; i++)
|
|
|
|
if (transports[i].mode == trans)
|
|
|
|
break;
|
gst/rtsp/: Morph RTPDec into something compatible with RTPBin as a fallback.
Original commit message from CVS:
* gst/rtsp/Makefile.am:
* gst/rtsp/gstrtpdec.c: (find_session_by_id), (create_session),
(free_session), (gst_rtp_dec_base_init), (gst_rtp_dec_class_init),
(gst_rtp_dec_init), (gst_rtp_dec_finalize),
(gst_rtp_dec_query_src), (gst_rtp_dec_chain_rtp),
(gst_rtp_dec_chain_rtcp), (gst_rtp_dec_set_property),
(gst_rtp_dec_get_property), (gst_rtp_dec_provide_clock),
(gst_rtp_dec_change_state), (create_recv_rtp), (create_recv_rtcp),
(create_rtcp), (gst_rtp_dec_request_new_pad),
(gst_rtp_dec_release_pad):
* gst/rtsp/gstrtpdec.h:
* gst/rtsp/gstrtsp.c: (plugin_init):
Morph RTPDec into something compatible with RTPBin as a fallback.
Various other style fixes.
* gst/rtsp/gstrtspsrc.c: (find_stream_by_id),
(find_stream_by_udpsrc), (gst_rtspsrc_stream_free),
(gst_rtspsrc_cleanup), (gst_rtspsrc_media_to_caps),
(new_session_pad), (gst_rtspsrc_stream_configure_transport),
(gst_rtspsrc_activate_streams), (gst_rtspsrc_loop_interleaved),
(gst_rtspsrc_loop_udp), (gst_rtspsrc_setup_auth),
(gst_rtspsrc_handle_message), (gst_rtspsrc_change_state):
* gst/rtsp/gstrtspsrc.h:
Implement RTPBin session manager handling.
Don't try to add empty properties to caps.
Implement fallback session manager, handling.
Don't combine errors from RTCP streams, just ignore them.
* gst/rtsp/rtsptransport.c: (rtsp_transport_get_manager):
* gst/rtsp/rtsptransport.h:
Implement fallback session manager.
Make RTPBin the default one when available.
2007-04-06 12:54:16 +00:00
|
|
|
|
|
|
|
if (option < MAX_MANAGERS)
|
|
|
|
*manager = transports[i].manager[option];
|
|
|
|
else
|
|
|
|
*manager = NULL;
|
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
|
|
|
|
|
|
|
return RTSP_OK;
|
|
|
|
}
|
|
|
|
|
2005-05-11 07:44:44 +00:00
|
|
|
static void
|
gst/rtsp/rtsptransport.*: Add validation to rtsp_transport_parse().
Original commit message from CVS:
Patch by: Peter Kjellerstedt <pkj at axis com>
* gst/rtsp/rtsptransport.c: (rtsp_transport_init), (parse_mode),
(parse_range), (range_as_text), (rtsp_transport_mode_as_text),
(rtsp_transport_profile_as_text), (rtsp_transport_ltrans_as_text),
(rtsp_transport_parse), (rtsp_transport_as_text):
* gst/rtsp/rtsptransport.h:
Add validation to rtsp_transport_parse().
Add rtsp_transport_as_text() to generate an RTSP header from an
RTSPTransport.
Change ssrc to guint (was a string) since that is what it is, even
though it is sent as a hex string.
Correctly identify PLAY|RECORD mode parameters (the syntax in the RFC is
incorrect, which can be seen when looking at the examples in the RFC).
Fixes #437670.
2007-05-12 16:26:06 +00:00
|
|
|
parse_mode (RTSPTransport * transport, const gchar * str)
|
2005-05-11 07:44:44 +00:00
|
|
|
{
|
gst/rtsp/rtsptransport.*: Add validation to rtsp_transport_parse().
Original commit message from CVS:
Patch by: Peter Kjellerstedt <pkj at axis com>
* gst/rtsp/rtsptransport.c: (rtsp_transport_init), (parse_mode),
(parse_range), (range_as_text), (rtsp_transport_mode_as_text),
(rtsp_transport_profile_as_text), (rtsp_transport_ltrans_as_text),
(rtsp_transport_parse), (rtsp_transport_as_text):
* gst/rtsp/rtsptransport.h:
Add validation to rtsp_transport_parse().
Add rtsp_transport_as_text() to generate an RTSP header from an
RTSPTransport.
Change ssrc to guint (was a string) since that is what it is, even
though it is sent as a hex string.
Correctly identify PLAY|RECORD mode parameters (the syntax in the RFC is
incorrect, which can be seen when looking at the examples in the RFC).
Fixes #437670.
2007-05-12 16:26:06 +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
|
gst/rtsp/rtsptransport.*: Add validation to rtsp_transport_parse().
Original commit message from CVS:
Patch by: Peter Kjellerstedt <pkj at axis com>
* gst/rtsp/rtsptransport.c: (rtsp_transport_init), (parse_mode),
(parse_range), (range_as_text), (rtsp_transport_mode_as_text),
(rtsp_transport_profile_as_text), (rtsp_transport_ltrans_as_text),
(rtsp_transport_parse), (rtsp_transport_as_text):
* gst/rtsp/rtsptransport.h:
Add validation to rtsp_transport_parse().
Add rtsp_transport_as_text() to generate an RTSP header from an
RTSPTransport.
Change ssrc to guint (was a string) since that is what it is, even
though it is sent as a hex string.
Correctly identify PLAY|RECORD mode parameters (the syntax in the RFC is
incorrect, which can be seen when looking at the examples in the RFC).
Fixes #437670.
2007-05-12 16:26:06 +00:00
|
|
|
parse_range (const gchar * str, RTSPRange * range)
|
2005-05-11 07:44:44 +00:00
|
|
|
{
|
|
|
|
gchar *minus;
|
gst/rtsp/rtsptransport.*: Add validation to rtsp_transport_parse().
Original commit message from CVS:
Patch by: Peter Kjellerstedt <pkj at axis com>
* gst/rtsp/rtsptransport.c: (rtsp_transport_init), (parse_mode),
(parse_range), (range_as_text), (rtsp_transport_mode_as_text),
(rtsp_transport_profile_as_text), (rtsp_transport_ltrans_as_text),
(rtsp_transport_parse), (rtsp_transport_as_text):
* gst/rtsp/rtsptransport.h:
Add validation to rtsp_transport_parse().
Add rtsp_transport_as_text() to generate an RTSP header from an
RTSPTransport.
Change ssrc to guint (was a string) since that is what it is, even
though it is sent as a hex string.
Correctly identify PLAY|RECORD mode parameters (the syntax in the RFC is
incorrect, which can be seen when looking at the examples in the RFC).
Fixes #437670.
2007-05-12 16:26:06 +00:00
|
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|
gchar *tmp;
|
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|
|
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|
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/* even though strtol() allows white space, plus and minus in front of
|
|
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* the number, we do not allow it
|
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*/
|
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|
if (g_ascii_isspace (*str) || *str == '+' || *str == '-')
|
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goto invalid_range;
|
2005-05-11 07:44:44 +00:00
|
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minus = strstr (str, "-");
|
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if (minus) {
|
gst/rtsp/rtsptransport.*: Add validation to rtsp_transport_parse().
Original commit message from CVS:
Patch by: Peter Kjellerstedt <pkj at axis com>
* gst/rtsp/rtsptransport.c: (rtsp_transport_init), (parse_mode),
(parse_range), (range_as_text), (rtsp_transport_mode_as_text),
(rtsp_transport_profile_as_text), (rtsp_transport_ltrans_as_text),
(rtsp_transport_parse), (rtsp_transport_as_text):
* gst/rtsp/rtsptransport.h:
Add validation to rtsp_transport_parse().
Add rtsp_transport_as_text() to generate an RTSP header from an
RTSPTransport.
Change ssrc to guint (was a string) since that is what it is, even
though it is sent as a hex string.
Correctly identify PLAY|RECORD mode parameters (the syntax in the RFC is
incorrect, which can be seen when looking at the examples in the RFC).
Fixes #437670.
2007-05-12 16:26:06 +00:00
|
|
|
if (g_ascii_isspace (minus[1]) || minus[1] == '+' || minus[1] == '-')
|
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|
|
goto invalid_range;
|
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|
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range->min = strtol (str, &tmp, 10);
|
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|
if (str == tmp || tmp != minus)
|
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goto invalid_range;
|
|
|
|
|
|
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range->max = strtol (minus + 1, &tmp, 10);
|
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|
if (*tmp && *tmp != ';')
|
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goto invalid_range;
|
2005-05-11 07:44:44 +00:00
|
|
|
} else {
|
gst/rtsp/rtsptransport.*: Add validation to rtsp_transport_parse().
Original commit message from CVS:
Patch by: Peter Kjellerstedt <pkj at axis com>
* gst/rtsp/rtsptransport.c: (rtsp_transport_init), (parse_mode),
(parse_range), (range_as_text), (rtsp_transport_mode_as_text),
(rtsp_transport_profile_as_text), (rtsp_transport_ltrans_as_text),
(rtsp_transport_parse), (rtsp_transport_as_text):
* gst/rtsp/rtsptransport.h:
Add validation to rtsp_transport_parse().
Add rtsp_transport_as_text() to generate an RTSP header from an
RTSPTransport.
Change ssrc to guint (was a string) since that is what it is, even
though it is sent as a hex string.
Correctly identify PLAY|RECORD mode parameters (the syntax in the RFC is
incorrect, which can be seen when looking at the examples in the RFC).
Fixes #437670.
2007-05-12 16:26:06 +00:00
|
|
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range->min = strtol (str, &tmp, 10);
|
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|
if (str == tmp || (*tmp && *tmp != ';'))
|
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goto invalid_range;
|
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|
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range->max = -1;
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}
|
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return;
|
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|
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invalid_range:
|
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{
|
|
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range->min = -1;
|
2005-05-11 07:44:44 +00:00
|
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range->max = -1;
|
gst/rtsp/rtsptransport.*: Add validation to rtsp_transport_parse().
Original commit message from CVS:
Patch by: Peter Kjellerstedt <pkj at axis com>
* gst/rtsp/rtsptransport.c: (rtsp_transport_init), (parse_mode),
(parse_range), (range_as_text), (rtsp_transport_mode_as_text),
(rtsp_transport_profile_as_text), (rtsp_transport_ltrans_as_text),
(rtsp_transport_parse), (rtsp_transport_as_text):
* gst/rtsp/rtsptransport.h:
Add validation to rtsp_transport_parse().
Add rtsp_transport_as_text() to generate an RTSP header from an
RTSPTransport.
Change ssrc to guint (was a string) since that is what it is, even
though it is sent as a hex string.
Correctly identify PLAY|RECORD mode parameters (the syntax in the RFC is
incorrect, which can be seen when looking at the examples in the RFC).
Fixes #437670.
2007-05-12 16:26:06 +00:00
|
|
|
return;
|
2005-05-11 07:44:44 +00:00
|
|
|
}
|
|
|
|
}
|
|
|
|
|
gst/rtsp/rtsptransport.*: Add validation to rtsp_transport_parse().
Original commit message from CVS:
Patch by: Peter Kjellerstedt <pkj at axis com>
* gst/rtsp/rtsptransport.c: (rtsp_transport_init), (parse_mode),
(parse_range), (range_as_text), (rtsp_transport_mode_as_text),
(rtsp_transport_profile_as_text), (rtsp_transport_ltrans_as_text),
(rtsp_transport_parse), (rtsp_transport_as_text):
* gst/rtsp/rtsptransport.h:
Add validation to rtsp_transport_parse().
Add rtsp_transport_as_text() to generate an RTSP header from an
RTSPTransport.
Change ssrc to guint (was a string) since that is what it is, even
though it is sent as a hex string.
Correctly identify PLAY|RECORD mode parameters (the syntax in the RFC is
incorrect, which can be seen when looking at the examples in the RFC).
Fixes #437670.
2007-05-12 16:26:06 +00:00
|
|
|
static gchar *
|
|
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|
range_as_text (const RTSPRange * range)
|
|
|
|
{
|
|
|
|
if (range->min < 0)
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return NULL;
|
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|
else if (range->max < 0)
|
|
|
|
return g_strdup_printf ("%d", range->min);
|
|
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|
else
|
|
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|
return g_strdup_printf ("%d-%d", range->min, range->max);
|
|
|
|
}
|
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|
|
|
|
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|
static const gchar *
|
|
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rtsp_transport_mode_as_text (const RTSPTransport * transport)
|
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|
{
|
|
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|
gint i;
|
|
|
|
|
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for (i = 0; transports[i].name; i++)
|
|
|
|
if (transports[i].mode == transport->trans)
|
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|
return transports[i].name;
|
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return NULL;
|
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}
|
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static const gchar *
|
|
|
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rtsp_transport_profile_as_text (const RTSPTransport * transport)
|
|
|
|
{
|
|
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|
gint i;
|
|
|
|
|
|
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|
for (i = 0; profiles[i].name; i++)
|
|
|
|
if (profiles[i].profile == transport->profile)
|
|
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|
return profiles[i].name;
|
|
|
|
|
|
|
|
return NULL;
|
|
|
|
}
|
|
|
|
|
|
|
|
static const gchar *
|
|
|
|
rtsp_transport_ltrans_as_text (const RTSPTransport * transport)
|
|
|
|
{
|
|
|
|
gint i;
|
|
|
|
|
|
|
|
/* need to special case RTSP_LOWER_TRANS_UDP_MCAST */
|
|
|
|
if (transport->lower_transport == RTSP_LOWER_TRANS_UDP_MCAST)
|
|
|
|
return "udp";
|
|
|
|
|
|
|
|
for (i = 0; ltrans[i].name; i++)
|
|
|
|
if (ltrans[i].ltrans == transport->lower_transport)
|
|
|
|
return ltrans[i].name;
|
|
|
|
|
|
|
|
return NULL;
|
|
|
|
}
|
|
|
|
|
2005-05-11 07:44:44 +00:00
|
|
|
RTSPResult
|
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
|
|
|
rtsp_transport_parse (const gchar * str, RTSPTransport * transport)
|
2005-05-11 07:44:44 +00:00
|
|
|
{
|
gst/rtsp/rtsptransport.*: Add validation to rtsp_transport_parse().
Original commit message from CVS:
Patch by: Peter Kjellerstedt <pkj at axis com>
* gst/rtsp/rtsptransport.c: (rtsp_transport_init), (parse_mode),
(parse_range), (range_as_text), (rtsp_transport_mode_as_text),
(rtsp_transport_profile_as_text), (rtsp_transport_ltrans_as_text),
(rtsp_transport_parse), (rtsp_transport_as_text):
* gst/rtsp/rtsptransport.h:
Add validation to rtsp_transport_parse().
Add rtsp_transport_as_text() to generate an RTSP header from an
RTSPTransport.
Change ssrc to guint (was a string) since that is what it is, even
though it is sent as a hex string.
Correctly identify PLAY|RECORD mode parameters (the syntax in the RFC is
incorrect, which can be seen when looking at the examples in the RFC).
Fixes #437670.
2007-05-12 16:26:06 +00:00
|
|
|
gchar **split, *down, **transp = NULL;
|
|
|
|
guint transport_params = 0;
|
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);
|
gst/rtsp/rtsptransport.*: Add validation to rtsp_transport_parse().
Original commit message from CVS:
Patch by: Peter Kjellerstedt <pkj at axis com>
* gst/rtsp/rtsptransport.c: (rtsp_transport_init), (parse_mode),
(parse_range), (range_as_text), (rtsp_transport_mode_as_text),
(rtsp_transport_profile_as_text), (rtsp_transport_ltrans_as_text),
(rtsp_transport_parse), (rtsp_transport_as_text):
* gst/rtsp/rtsptransport.h:
Add validation to rtsp_transport_parse().
Add rtsp_transport_as_text() to generate an RTSP header from an
RTSPTransport.
Change ssrc to guint (was a string) since that is what it is, even
though it is sent as a hex string.
Correctly identify PLAY|RECORD mode parameters (the syntax in the RFC is
incorrect, which can be seen when looking at the examples in the RFC).
Fixes #437670.
2007-05-12 16:26:06 +00:00
|
|
|
g_free (down);
|
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
|
|
|
|
|
|
|
/* First field contains the transport/profile/lower_transport */
|
gst/rtsp/rtsptransport.*: Add validation to rtsp_transport_parse().
Original commit message from CVS:
Patch by: Peter Kjellerstedt <pkj at axis com>
* gst/rtsp/rtsptransport.c: (rtsp_transport_init), (parse_mode),
(parse_range), (range_as_text), (rtsp_transport_mode_as_text),
(rtsp_transport_profile_as_text), (rtsp_transport_ltrans_as_text),
(rtsp_transport_parse), (rtsp_transport_as_text):
* gst/rtsp/rtsptransport.h:
Add validation to rtsp_transport_parse().
Add rtsp_transport_as_text() to generate an RTSP header from an
RTSPTransport.
Change ssrc to guint (was a string) since that is what it is, even
though it is sent as a hex string.
Correctly identify PLAY|RECORD mode parameters (the syntax in the RFC is
incorrect, which can be seen when looking at the examples in the RFC).
Fixes #437670.
2007-05-12 16:26:06 +00:00
|
|
|
if (split[0] == NULL)
|
|
|
|
goto invalid_transport;
|
|
|
|
|
|
|
|
transp = g_strsplit (split[0], "/", 0);
|
|
|
|
|
|
|
|
if (transp[0] == NULL || transp[1] == NULL)
|
|
|
|
goto invalid_transport;
|
|
|
|
|
|
|
|
for (i = 0; transports[i].name; i++)
|
|
|
|
if (strcmp (transp[0], transports[i].name) == 0)
|
|
|
|
break;
|
|
|
|
transport->trans = transports[i].mode;
|
|
|
|
|
|
|
|
for (i = 0; profiles[i].name; i++)
|
|
|
|
if (strcmp (transp[1], profiles[i].name) == 0)
|
|
|
|
break;
|
|
|
|
transport->profile = profiles[i].profile;
|
|
|
|
|
|
|
|
if (transp[2] != NULL) {
|
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
|
|
|
for (i = 0; ltrans[i].name; i++)
|
gst/rtsp/rtsptransport.*: Add validation to rtsp_transport_parse().
Original commit message from CVS:
Patch by: Peter Kjellerstedt <pkj at axis com>
* gst/rtsp/rtsptransport.c: (rtsp_transport_init), (parse_mode),
(parse_range), (range_as_text), (rtsp_transport_mode_as_text),
(rtsp_transport_profile_as_text), (rtsp_transport_ltrans_as_text),
(rtsp_transport_parse), (rtsp_transport_as_text):
* gst/rtsp/rtsptransport.h:
Add validation to rtsp_transport_parse().
Add rtsp_transport_as_text() to generate an RTSP header from an
RTSPTransport.
Change ssrc to guint (was a string) since that is what it is, even
though it is sent as a hex string.
Correctly identify PLAY|RECORD mode parameters (the syntax in the RFC is
incorrect, which can be seen when looking at the examples in the RFC).
Fixes #437670.
2007-05-12 16:26:06 +00:00
|
|
|
if (strcmp (transp[2], ltrans[i].name) == 0)
|
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
|
|
|
break;
|
|
|
|
transport->lower_transport = ltrans[i].ltrans;
|
gst/rtsp/rtsptransport.*: Add validation to rtsp_transport_parse().
Original commit message from CVS:
Patch by: Peter Kjellerstedt <pkj at axis com>
* gst/rtsp/rtsptransport.c: (rtsp_transport_init), (parse_mode),
(parse_range), (range_as_text), (rtsp_transport_mode_as_text),
(rtsp_transport_profile_as_text), (rtsp_transport_ltrans_as_text),
(rtsp_transport_parse), (rtsp_transport_as_text):
* gst/rtsp/rtsptransport.h:
Add validation to rtsp_transport_parse().
Add rtsp_transport_as_text() to generate an RTSP header from an
RTSPTransport.
Change ssrc to guint (was a string) since that is what it is, even
though it is sent as a hex string.
Correctly identify PLAY|RECORD mode parameters (the syntax in the RFC is
incorrect, which can be seen when looking at the examples in the RFC).
Fixes #437670.
2007-05-12 16:26:06 +00:00
|
|
|
} else {
|
|
|
|
/* specifying the lower transport is optional */
|
|
|
|
if (transport->trans == RTSP_TRANS_RTP &&
|
|
|
|
transport->profile == RTSP_PROFILE_AVP)
|
|
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transport->lower_transport = RTSP_LOWER_TRANS_UDP_MCAST;
|
|
|
|
else
|
|
|
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transport->lower_transport = RTSP_LOWER_TRANS_UNKNOWN;
|
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
|
|
|
}
|
gst/rtsp/rtsptransport.*: Add validation to rtsp_transport_parse().
Original commit message from CVS:
Patch by: Peter Kjellerstedt <pkj at axis com>
* gst/rtsp/rtsptransport.c: (rtsp_transport_init), (parse_mode),
(parse_range), (range_as_text), (rtsp_transport_mode_as_text),
(rtsp_transport_profile_as_text), (rtsp_transport_ltrans_as_text),
(rtsp_transport_parse), (rtsp_transport_as_text):
* gst/rtsp/rtsptransport.h:
Add validation to rtsp_transport_parse().
Add rtsp_transport_as_text() to generate an RTSP header from an
RTSPTransport.
Change ssrc to guint (was a string) since that is what it is, even
though it is sent as a hex string.
Correctly identify PLAY|RECORD mode parameters (the syntax in the RFC is
incorrect, which can be seen when looking at the examples in the RFC).
Fixes #437670.
2007-05-12 16:26:06 +00:00
|
|
|
|
|
|
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g_strfreev (transp);
|
|
|
|
transp = NULL;
|
|
|
|
|
|
|
|
if (transport->trans == RTSP_TRANS_UNKNOWN ||
|
|
|
|
transport->profile == RTSP_PROFILE_UNKNOWN ||
|
|
|
|
transport->lower_transport == RTSP_LOWER_TRANS_UNKNOWN)
|
|
|
|
goto unsupported_transport;
|
|
|
|
|
|
|
|
i = 1;
|
2005-05-11 07:44:44 +00:00
|
|
|
while (split[i]) {
|
gst/rtsp/rtsptransport.*: Add validation to rtsp_transport_parse().
Original commit message from CVS:
Patch by: Peter Kjellerstedt <pkj at axis com>
* gst/rtsp/rtsptransport.c: (rtsp_transport_init), (parse_mode),
(parse_range), (range_as_text), (rtsp_transport_mode_as_text),
(rtsp_transport_profile_as_text), (rtsp_transport_ltrans_as_text),
(rtsp_transport_parse), (rtsp_transport_as_text):
* gst/rtsp/rtsptransport.h:
Add validation to rtsp_transport_parse().
Add rtsp_transport_as_text() to generate an RTSP header from an
RTSPTransport.
Change ssrc to guint (was a string) since that is what it is, even
though it is sent as a hex string.
Correctly identify PLAY|RECORD mode parameters (the syntax in the RFC is
incorrect, which can be seen when looking at the examples in the RFC).
Fixes #437670.
2007-05-12 16:26:06 +00:00
|
|
|
if (strcmp (split[i], "multicast") == 0) {
|
|
|
|
RTSP_TRANSPORT_PARAMETER_IS_UNIQUE (RTSP_TRANSPORT_DELIVERY);
|
|
|
|
if (transport->lower_transport == RTSP_LOWER_TRANS_TCP)
|
|
|
|
goto invalid_transport;
|
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
|
|
|
transport->lower_transport = RTSP_LOWER_TRANS_UDP_MCAST;
|
gst/rtsp/rtsptransport.*: Add validation to rtsp_transport_parse().
Original commit message from CVS:
Patch by: Peter Kjellerstedt <pkj at axis com>
* gst/rtsp/rtsptransport.c: (rtsp_transport_init), (parse_mode),
(parse_range), (range_as_text), (rtsp_transport_mode_as_text),
(rtsp_transport_profile_as_text), (rtsp_transport_ltrans_as_text),
(rtsp_transport_parse), (rtsp_transport_as_text):
* gst/rtsp/rtsptransport.h:
Add validation to rtsp_transport_parse().
Add rtsp_transport_as_text() to generate an RTSP header from an
RTSPTransport.
Change ssrc to guint (was a string) since that is what it is, even
though it is sent as a hex string.
Correctly identify PLAY|RECORD mode parameters (the syntax in the RFC is
incorrect, which can be seen when looking at the examples in the RFC).
Fixes #437670.
2007-05-12 16:26:06 +00:00
|
|
|
} else if (strcmp (split[i], "unicast") == 0) {
|
|
|
|
RTSP_TRANSPORT_PARAMETER_IS_UNIQUE (RTSP_TRANSPORT_DELIVERY);
|
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
|
|
|
if (transport->lower_transport == RTSP_LOWER_TRANS_UDP_MCAST)
|
|
|
|
transport->lower_transport = RTSP_LOWER_TRANS_UDP;
|
2005-05-11 07:44:44 +00:00
|
|
|
} else if (g_str_has_prefix (split[i], "destination=")) {
|
gst/rtsp/rtsptransport.*: Add validation to rtsp_transport_parse().
Original commit message from CVS:
Patch by: Peter Kjellerstedt <pkj at axis com>
* gst/rtsp/rtsptransport.c: (rtsp_transport_init), (parse_mode),
(parse_range), (range_as_text), (rtsp_transport_mode_as_text),
(rtsp_transport_profile_as_text), (rtsp_transport_ltrans_as_text),
(rtsp_transport_parse), (rtsp_transport_as_text):
* gst/rtsp/rtsptransport.h:
Add validation to rtsp_transport_parse().
Add rtsp_transport_as_text() to generate an RTSP header from an
RTSPTransport.
Change ssrc to guint (was a string) since that is what it is, even
though it is sent as a hex string.
Correctly identify PLAY|RECORD mode parameters (the syntax in the RFC is
incorrect, which can be seen when looking at the examples in the RFC).
Fixes #437670.
2007-05-12 16:26:06 +00:00
|
|
|
RTSP_TRANSPORT_PARAMETER_IS_UNIQUE (RTSP_TRANSPORT_DESTINATION);
|
2005-05-11 07:44:44 +00:00
|
|
|
transport->destination = g_strdup (split[i] + 12);
|
|
|
|
} else if (g_str_has_prefix (split[i], "source=")) {
|
gst/rtsp/rtsptransport.*: Add validation to rtsp_transport_parse().
Original commit message from CVS:
Patch by: Peter Kjellerstedt <pkj at axis com>
* gst/rtsp/rtsptransport.c: (rtsp_transport_init), (parse_mode),
(parse_range), (range_as_text), (rtsp_transport_mode_as_text),
(rtsp_transport_profile_as_text), (rtsp_transport_ltrans_as_text),
(rtsp_transport_parse), (rtsp_transport_as_text):
* gst/rtsp/rtsptransport.h:
Add validation to rtsp_transport_parse().
Add rtsp_transport_as_text() to generate an RTSP header from an
RTSPTransport.
Change ssrc to guint (was a string) since that is what it is, even
though it is sent as a hex string.
Correctly identify PLAY|RECORD mode parameters (the syntax in the RFC is
incorrect, which can be seen when looking at the examples in the RFC).
Fixes #437670.
2007-05-12 16:26:06 +00:00
|
|
|
RTSP_TRANSPORT_PARAMETER_IS_UNIQUE (RTSP_TRANSPORT_SOURCE);
|
2005-05-11 07:44:44 +00:00
|
|
|
transport->source = g_strdup (split[i] + 7);
|
|
|
|
} else if (g_str_has_prefix (split[i], "layers=")) {
|
gst/rtsp/rtsptransport.*: Add validation to rtsp_transport_parse().
Original commit message from CVS:
Patch by: Peter Kjellerstedt <pkj at axis com>
* gst/rtsp/rtsptransport.c: (rtsp_transport_init), (parse_mode),
(parse_range), (range_as_text), (rtsp_transport_mode_as_text),
(rtsp_transport_profile_as_text), (rtsp_transport_ltrans_as_text),
(rtsp_transport_parse), (rtsp_transport_as_text):
* gst/rtsp/rtsptransport.h:
Add validation to rtsp_transport_parse().
Add rtsp_transport_as_text() to generate an RTSP header from an
RTSPTransport.
Change ssrc to guint (was a string) since that is what it is, even
though it is sent as a hex string.
Correctly identify PLAY|RECORD mode parameters (the syntax in the RFC is
incorrect, which can be seen when looking at the examples in the RFC).
Fixes #437670.
2007-05-12 16:26:06 +00:00
|
|
|
RTSP_TRANSPORT_PARAMETER_IS_UNIQUE (RTSP_TRANSPORT_LAYERS);
|
|
|
|
transport->layers = strtoul (split[i] + 7, NULL, 10);
|
2005-05-11 07:44:44 +00:00
|
|
|
} else if (g_str_has_prefix (split[i], "mode=")) {
|
gst/rtsp/rtsptransport.*: Add validation to rtsp_transport_parse().
Original commit message from CVS:
Patch by: Peter Kjellerstedt <pkj at axis com>
* gst/rtsp/rtsptransport.c: (rtsp_transport_init), (parse_mode),
(parse_range), (range_as_text), (rtsp_transport_mode_as_text),
(rtsp_transport_profile_as_text), (rtsp_transport_ltrans_as_text),
(rtsp_transport_parse), (rtsp_transport_as_text):
* gst/rtsp/rtsptransport.h:
Add validation to rtsp_transport_parse().
Add rtsp_transport_as_text() to generate an RTSP header from an
RTSPTransport.
Change ssrc to guint (was a string) since that is what it is, even
though it is sent as a hex string.
Correctly identify PLAY|RECORD mode parameters (the syntax in the RFC is
incorrect, which can be seen when looking at the examples in the RFC).
Fixes #437670.
2007-05-12 16:26:06 +00:00
|
|
|
RTSP_TRANSPORT_PARAMETER_IS_UNIQUE (RTSP_TRANSPORT_MODE);
|
2005-05-11 07:44:44 +00:00
|
|
|
parse_mode (transport, split[i] + 5);
|
gst/rtsp/rtsptransport.*: Add validation to rtsp_transport_parse().
Original commit message from CVS:
Patch by: Peter Kjellerstedt <pkj at axis com>
* gst/rtsp/rtsptransport.c: (rtsp_transport_init), (parse_mode),
(parse_range), (range_as_text), (rtsp_transport_mode_as_text),
(rtsp_transport_profile_as_text), (rtsp_transport_ltrans_as_text),
(rtsp_transport_parse), (rtsp_transport_as_text):
* gst/rtsp/rtsptransport.h:
Add validation to rtsp_transport_parse().
Add rtsp_transport_as_text() to generate an RTSP header from an
RTSPTransport.
Change ssrc to guint (was a string) since that is what it is, even
though it is sent as a hex string.
Correctly identify PLAY|RECORD mode parameters (the syntax in the RFC is
incorrect, which can be seen when looking at the examples in the RFC).
Fixes #437670.
2007-05-12 16:26:06 +00:00
|
|
|
if (!transport->mode_play && !transport->mode_record)
|
|
|
|
goto invalid_transport;
|
|
|
|
} else if (strcmp (split[i], "append") == 0) {
|
|
|
|
RTSP_TRANSPORT_PARAMETER_IS_UNIQUE (RTSP_TRANSPORT_APPEND);
|
2005-05-11 07:44:44 +00:00
|
|
|
transport->append = TRUE;
|
|
|
|
} else if (g_str_has_prefix (split[i], "interleaved=")) {
|
gst/rtsp/rtsptransport.*: Add validation to rtsp_transport_parse().
Original commit message from CVS:
Patch by: Peter Kjellerstedt <pkj at axis com>
* gst/rtsp/rtsptransport.c: (rtsp_transport_init), (parse_mode),
(parse_range), (range_as_text), (rtsp_transport_mode_as_text),
(rtsp_transport_profile_as_text), (rtsp_transport_ltrans_as_text),
(rtsp_transport_parse), (rtsp_transport_as_text):
* gst/rtsp/rtsptransport.h:
Add validation to rtsp_transport_parse().
Add rtsp_transport_as_text() to generate an RTSP header from an
RTSPTransport.
Change ssrc to guint (was a string) since that is what it is, even
though it is sent as a hex string.
Correctly identify PLAY|RECORD mode parameters (the syntax in the RFC is
incorrect, which can be seen when looking at the examples in the RFC).
Fixes #437670.
2007-05-12 16:26:06 +00:00
|
|
|
RTSP_TRANSPORT_PARAMETER_IS_UNIQUE (RTSP_TRANSPORT_INTERLEAVED);
|
|
|
|
parse_range (split[i] + 12, &transport->interleaved);
|
|
|
|
if (transport->interleaved.min < 0 ||
|
|
|
|
transport->interleaved.min >= 256 ||
|
|
|
|
transport->interleaved.max >= 256)
|
|
|
|
goto invalid_transport;
|
2005-05-11 07:44:44 +00:00
|
|
|
} else if (g_str_has_prefix (split[i], "ttl=")) {
|
gst/rtsp/rtsptransport.*: Add validation to rtsp_transport_parse().
Original commit message from CVS:
Patch by: Peter Kjellerstedt <pkj at axis com>
* gst/rtsp/rtsptransport.c: (rtsp_transport_init), (parse_mode),
(parse_range), (range_as_text), (rtsp_transport_mode_as_text),
(rtsp_transport_profile_as_text), (rtsp_transport_ltrans_as_text),
(rtsp_transport_parse), (rtsp_transport_as_text):
* gst/rtsp/rtsptransport.h:
Add validation to rtsp_transport_parse().
Add rtsp_transport_as_text() to generate an RTSP header from an
RTSPTransport.
Change ssrc to guint (was a string) since that is what it is, even
though it is sent as a hex string.
Correctly identify PLAY|RECORD mode parameters (the syntax in the RFC is
incorrect, which can be seen when looking at the examples in the RFC).
Fixes #437670.
2007-05-12 16:26:06 +00:00
|
|
|
RTSP_TRANSPORT_PARAMETER_IS_UNIQUE (RTSP_TRANSPORT_TTL);
|
|
|
|
transport->ttl = strtoul (split[i] + 4, NULL, 10);
|
|
|
|
if (transport->ttl >= 256)
|
|
|
|
goto invalid_transport;
|
2005-05-11 07:44:44 +00:00
|
|
|
} else if (g_str_has_prefix (split[i], "port=")) {
|
gst/rtsp/rtsptransport.*: Add validation to rtsp_transport_parse().
Original commit message from CVS:
Patch by: Peter Kjellerstedt <pkj at axis com>
* gst/rtsp/rtsptransport.c: (rtsp_transport_init), (parse_mode),
(parse_range), (range_as_text), (rtsp_transport_mode_as_text),
(rtsp_transport_profile_as_text), (rtsp_transport_ltrans_as_text),
(rtsp_transport_parse), (rtsp_transport_as_text):
* gst/rtsp/rtsptransport.h:
Add validation to rtsp_transport_parse().
Add rtsp_transport_as_text() to generate an RTSP header from an
RTSPTransport.
Change ssrc to guint (was a string) since that is what it is, even
though it is sent as a hex string.
Correctly identify PLAY|RECORD mode parameters (the syntax in the RFC is
incorrect, which can be seen when looking at the examples in the RFC).
Fixes #437670.
2007-05-12 16:26:06 +00:00
|
|
|
RTSP_TRANSPORT_PARAMETER_IS_UNIQUE (RTSP_TRANSPORT_PORT);
|
|
|
|
parse_range (split[i] + 5, &transport->port);
|
|
|
|
if (transport->port.min < 0 ||
|
|
|
|
transport->port.min >= 65536 || transport->port.max >= 65536)
|
|
|
|
goto invalid_transport;
|
2005-05-11 07:44:44 +00:00
|
|
|
} else if (g_str_has_prefix (split[i], "client_port=")) {
|
gst/rtsp/rtsptransport.*: Add validation to rtsp_transport_parse().
Original commit message from CVS:
Patch by: Peter Kjellerstedt <pkj at axis com>
* gst/rtsp/rtsptransport.c: (rtsp_transport_init), (parse_mode),
(parse_range), (range_as_text), (rtsp_transport_mode_as_text),
(rtsp_transport_profile_as_text), (rtsp_transport_ltrans_as_text),
(rtsp_transport_parse), (rtsp_transport_as_text):
* gst/rtsp/rtsptransport.h:
Add validation to rtsp_transport_parse().
Add rtsp_transport_as_text() to generate an RTSP header from an
RTSPTransport.
Change ssrc to guint (was a string) since that is what it is, even
though it is sent as a hex string.
Correctly identify PLAY|RECORD mode parameters (the syntax in the RFC is
incorrect, which can be seen when looking at the examples in the RFC).
Fixes #437670.
2007-05-12 16:26:06 +00:00
|
|
|
RTSP_TRANSPORT_PARAMETER_IS_UNIQUE (RTSP_TRANSPORT_CLIENT_PORT);
|
|
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parse_range (split[i] + 12, &transport->client_port);
|
|
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|
if (transport->client_port.min < 0 ||
|
|
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transport->client_port.min >= 65536 ||
|
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transport->client_port.max >= 65536)
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goto invalid_transport;
|
2005-05-11 07:44:44 +00:00
|
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} else if (g_str_has_prefix (split[i], "server_port=")) {
|
gst/rtsp/rtsptransport.*: Add validation to rtsp_transport_parse().
Original commit message from CVS:
Patch by: Peter Kjellerstedt <pkj at axis com>
* gst/rtsp/rtsptransport.c: (rtsp_transport_init), (parse_mode),
(parse_range), (range_as_text), (rtsp_transport_mode_as_text),
(rtsp_transport_profile_as_text), (rtsp_transport_ltrans_as_text),
(rtsp_transport_parse), (rtsp_transport_as_text):
* gst/rtsp/rtsptransport.h:
Add validation to rtsp_transport_parse().
Add rtsp_transport_as_text() to generate an RTSP header from an
RTSPTransport.
Change ssrc to guint (was a string) since that is what it is, even
though it is sent as a hex string.
Correctly identify PLAY|RECORD mode parameters (the syntax in the RFC is
incorrect, which can be seen when looking at the examples in the RFC).
Fixes #437670.
2007-05-12 16:26:06 +00:00
|
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|
RTSP_TRANSPORT_PARAMETER_IS_UNIQUE (RTSP_TRANSPORT_SERVER_PORT);
|
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parse_range (split[i] + 12, &transport->server_port);
|
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if (transport->server_port.min < 0 ||
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transport->server_port.min >= 65536 ||
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transport->server_port.max >= 65536)
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goto invalid_transport;
|
2005-05-11 07:44:44 +00:00
|
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} else if (g_str_has_prefix (split[i], "ssrc=")) {
|
gst/rtsp/rtsptransport.*: Add validation to rtsp_transport_parse().
Original commit message from CVS:
Patch by: Peter Kjellerstedt <pkj at axis com>
* gst/rtsp/rtsptransport.c: (rtsp_transport_init), (parse_mode),
(parse_range), (range_as_text), (rtsp_transport_mode_as_text),
(rtsp_transport_profile_as_text), (rtsp_transport_ltrans_as_text),
(rtsp_transport_parse), (rtsp_transport_as_text):
* gst/rtsp/rtsptransport.h:
Add validation to rtsp_transport_parse().
Add rtsp_transport_as_text() to generate an RTSP header from an
RTSPTransport.
Change ssrc to guint (was a string) since that is what it is, even
though it is sent as a hex string.
Correctly identify PLAY|RECORD mode parameters (the syntax in the RFC is
incorrect, which can be seen when looking at the examples in the RFC).
Fixes #437670.
2007-05-12 16:26:06 +00:00
|
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RTSP_TRANSPORT_PARAMETER_IS_UNIQUE (RTSP_TRANSPORT_SSRC);
|
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transport->ssrc = strtoul (split[i] + 5, NULL, 16);
|
2005-05-11 07:44:44 +00:00
|
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} else {
|
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/* unknown field... */
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g_warning ("unknown transport field \"%s\"", split[i]);
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}
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i++;
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}
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g_strfreev (split);
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return RTSP_OK;
|
gst/rtsp/rtsptransport.*: Add validation to rtsp_transport_parse().
Original commit message from CVS:
Patch by: Peter Kjellerstedt <pkj at axis com>
* gst/rtsp/rtsptransport.c: (rtsp_transport_init), (parse_mode),
(parse_range), (range_as_text), (rtsp_transport_mode_as_text),
(rtsp_transport_profile_as_text), (rtsp_transport_ltrans_as_text),
(rtsp_transport_parse), (rtsp_transport_as_text):
* gst/rtsp/rtsptransport.h:
Add validation to rtsp_transport_parse().
Add rtsp_transport_as_text() to generate an RTSP header from an
RTSPTransport.
Change ssrc to guint (was a string) since that is what it is, even
though it is sent as a hex string.
Correctly identify PLAY|RECORD mode parameters (the syntax in the RFC is
incorrect, which can be seen when looking at the examples in the RFC).
Fixes #437670.
2007-05-12 16:26:06 +00:00
|
|
|
|
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unsupported_transport:
|
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{
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g_strfreev (split);
|
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return RTSP_ERROR;
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}
|
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invalid_transport:
|
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{
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g_strfreev (transp);
|
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g_strfreev (split);
|
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return RTSP_EINVAL;
|
|
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}
|
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}
|
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gchar *
|
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rtsp_transport_as_text (RTSPTransport * transport)
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|
{
|
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GPtrArray *strs;
|
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gchar *res;
|
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const gchar *tmp;
|
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|
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g_return_val_if_fail (transport != NULL, NULL);
|
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|
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strs = g_ptr_array_new ();
|
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|
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|
|
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/* add the transport specifier */
|
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if ((tmp = rtsp_transport_mode_as_text (transport)) == NULL)
|
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goto invalid_transport;
|
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g_ptr_array_add (strs, g_ascii_strup (tmp, -1));
|
|
|
|
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|
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g_ptr_array_add (strs, g_strdup ("/"));
|
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|
|
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|
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if ((tmp = rtsp_transport_profile_as_text (transport)) == NULL)
|
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goto invalid_transport;
|
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g_ptr_array_add (strs, g_ascii_strup (tmp, -1));
|
|
|
|
|
|
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if (transport->trans != RTSP_TRANS_RTP ||
|
|
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transport->profile != RTSP_PROFILE_AVP ||
|
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transport->lower_transport == RTSP_LOWER_TRANS_TCP) {
|
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g_ptr_array_add (strs, g_strdup ("/"));
|
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|
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if ((tmp = rtsp_transport_ltrans_as_text (transport)) == NULL)
|
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goto invalid_transport;
|
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g_ptr_array_add (strs, g_ascii_strup (tmp, -1));
|
|
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}
|
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|
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|
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/*
|
|
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|
* the order of the following parameters is the same as the one specified in
|
|
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|
* RFC 2326 to please some weird RTSP clients that require it
|
|
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|
*/
|
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|
|
|
|
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/* add the unicast/multicast parameter */
|
|
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if (transport->lower_transport == RTSP_LOWER_TRANS_UDP_MCAST)
|
|
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g_ptr_array_add (strs, g_strdup (";multicast"));
|
|
|
|
else
|
|
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g_ptr_array_add (strs, g_strdup (";unicast"));
|
|
|
|
|
|
|
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/* add the destination parameter */
|
|
|
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if (transport->destination != NULL) {
|
|
|
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g_ptr_array_add (strs, g_strdup (";destination="));
|
|
|
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g_ptr_array_add (strs, g_strdup (transport->destination));
|
|
|
|
}
|
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|
|
|
|
|
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/* add the source parameter */
|
|
|
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if (transport->source != NULL) {
|
|
|
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g_ptr_array_add (strs, g_strdup (";source="));
|
|
|
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g_ptr_array_add (strs, g_strdup (transport->source));
|
|
|
|
}
|
|
|
|
|
|
|
|
/* add the interleaved parameter */
|
|
|
|
if (transport->lower_transport == RTSP_LOWER_TRANS_TCP &&
|
|
|
|
transport->interleaved.min >= 0) {
|
|
|
|
if (transport->interleaved.min < 256 && transport->interleaved.max < 256) {
|
|
|
|
g_ptr_array_add (strs, g_strdup (";interleaved="));
|
|
|
|
g_ptr_array_add (strs, range_as_text (&transport->interleaved));
|
|
|
|
} else
|
|
|
|
goto invalid_transport;
|
|
|
|
}
|
|
|
|
|
|
|
|
/* add the append parameter */
|
|
|
|
if (transport->mode_record && transport->append)
|
|
|
|
g_ptr_array_add (strs, g_strdup (";append"));
|
|
|
|
|
|
|
|
/* add the ttl parameter */
|
|
|
|
if (transport->lower_transport == RTSP_LOWER_TRANS_UDP_MCAST &&
|
|
|
|
transport->ttl != 0) {
|
|
|
|
if (transport->ttl < 256) {
|
|
|
|
g_ptr_array_add (strs, g_strdup (";ttl="));
|
|
|
|
g_ptr_array_add (strs, g_strdup_printf ("%u", transport->ttl));
|
|
|
|
} else
|
|
|
|
goto invalid_transport;
|
|
|
|
}
|
|
|
|
|
|
|
|
/* add the layers parameter */
|
|
|
|
if (transport->layers != 0) {
|
|
|
|
g_ptr_array_add (strs, g_strdup (";layers="));
|
|
|
|
g_ptr_array_add (strs, g_strdup_printf ("%u", transport->layers));
|
|
|
|
}
|
|
|
|
|
|
|
|
/* add the port parameter */
|
|
|
|
if (transport->trans == RTSP_TRANS_RTP && transport->port.min >= 0) {
|
|
|
|
if (transport->port.min < 65536 && transport->port.max < 65536) {
|
|
|
|
g_ptr_array_add (strs, g_strdup (";port="));
|
|
|
|
g_ptr_array_add (strs, range_as_text (&transport->port));
|
|
|
|
} else
|
|
|
|
goto invalid_transport;
|
|
|
|
}
|
|
|
|
|
|
|
|
/* add the client_port parameter */
|
|
|
|
if (transport->trans == RTSP_TRANS_RTP && transport->client_port.min >= 0) {
|
|
|
|
if (transport->client_port.min < 65536 &&
|
|
|
|
transport->client_port.max < 65536) {
|
|
|
|
g_ptr_array_add (strs, g_strdup (";client_port="));
|
|
|
|
g_ptr_array_add (strs, range_as_text (&transport->client_port));
|
|
|
|
} else
|
|
|
|
goto invalid_transport;
|
|
|
|
}
|
|
|
|
|
|
|
|
/* add the server_port parameter */
|
|
|
|
if (transport->trans == RTSP_TRANS_RTP && transport->server_port.min >= 0) {
|
|
|
|
if (transport->server_port.min < 65536 &&
|
|
|
|
transport->server_port.max < 65536) {
|
|
|
|
g_ptr_array_add (strs, g_strdup (";server_port="));
|
|
|
|
g_ptr_array_add (strs, range_as_text (&transport->server_port));
|
|
|
|
} else
|
|
|
|
goto invalid_transport;
|
|
|
|
}
|
|
|
|
|
|
|
|
/* add the ssrc parameter */
|
|
|
|
if (transport->lower_transport != RTSP_LOWER_TRANS_UDP_MCAST &&
|
|
|
|
transport->ssrc != 0) {
|
|
|
|
g_ptr_array_add (strs, g_strdup (";ssrc="));
|
|
|
|
g_ptr_array_add (strs, g_strdup_printf ("%08X", transport->ssrc));
|
|
|
|
}
|
|
|
|
|
|
|
|
/* add the mode parameter */
|
|
|
|
if (transport->mode_play && transport->mode_record)
|
|
|
|
g_ptr_array_add (strs, g_strdup (";mode=\"PLAY,RECORD\""));
|
|
|
|
else if (transport->mode_record)
|
|
|
|
g_ptr_array_add (strs, g_strdup (";mode=\"RECORD\""));
|
|
|
|
else if (transport->mode_play)
|
|
|
|
g_ptr_array_add (strs, g_strdup (";mode=\"PLAY\""));
|
|
|
|
|
|
|
|
/* add a terminating NULL */
|
|
|
|
g_ptr_array_add (strs, NULL);
|
|
|
|
|
|
|
|
res = g_strjoinv (NULL, (gchar **) strs->pdata);
|
|
|
|
g_strfreev ((gchar **) g_ptr_array_free (strs, FALSE));
|
|
|
|
|
|
|
|
return res;
|
|
|
|
|
|
|
|
invalid_transport:
|
|
|
|
{
|
|
|
|
g_ptr_array_add (strs, NULL);
|
|
|
|
g_strfreev ((gchar **) g_ptr_array_free (strs, FALSE));
|
|
|
|
return NULL;
|
|
|
|
}
|
2005-05-11 07:44:44 +00:00
|
|
|
}
|
|
|
|
|
|
|
|
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);
|
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
|
|
|
|
2005-05-11 07:44:44 +00:00
|
|
|
return RTSP_OK;
|
|
|
|
}
|