2005-05-11 07:44:44 +00:00
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/* GStreamer
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* Copyright (C) <1999> Erik Walthinsen <omega@cse.ogi.edu>
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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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* <2006> Wim Taymans <wim@fluendo.com>
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2005-05-11 07:44:44 +00:00
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*
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* This library is free software; you can redistribute it and/or
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* modify it under the terms of the GNU Library General Public
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* License as published by the Free Software Foundation; either
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* version 2 of the License, or (at your option) any later version.
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*
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* This library is distributed in the hope that it will be useful,
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* but WITHOUT ANY WARRANTY; without even the implied warranty of
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* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
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* Library General Public License for more details.
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*
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* You should have received a copy of the GNU Library General Public
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* License along with this library; if not, write to the
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* Free Software Foundation, Inc., 59 Temple Place - Suite 330,
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* Boston, MA 02111-1307, USA.
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*/
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gst/rtsp/URLS: Added some test URLS.
Original commit message from CVS:
* gst/rtsp/URLS:
Added some test URLS.
* gst/rtsp/gstrtspsrc.c: (gst_rtspsrc_create_stream),
(gst_rtspsrc_loop), (gst_rtspsrc_open):
* gst/rtsp/gstrtspsrc.h:
When creating streams, give access to the complete SDP.
Fix some leaks.
Collect and merge global stream properties in stream caps.
Preliminary support for WMServer.
* gst/rtsp/rtspconnection.c: (rtsp_connection_create),
(rtsp_connection_connect), (rtsp_connection_read), (read_body),
(rtsp_connection_receive):
* gst/rtsp/rtspconnection.h:
Make connection interruptable.
Refactor to make it reconnectable.
Don't fail on short reads when reading data packets.
* gst/rtsp/rtspurl.c: (rtsp_url_parse), (rtsp_url_set_port),
(rtsp_url_get_port):
* gst/rtsp/rtspurl.h:
Add methods for getting/setting the port.
* gst/rtsp/sdpmessage.c: (sdp_message_get_attribute_val_n),
(sdp_message_get_attribute_val), (sdp_media_get_attribute),
(sdp_media_get_attribute_val_n), (sdp_media_get_attribute_val),
(sdp_media_get_format), (sdp_parse_line),
(sdp_message_parse_buffer):
Fix headers.
Add methods for getting multiple attributes with the same name.
Increase buffer size when parsing.
Fix parsing of a=foo fields.
* gst/rtsp/test.c: (main):
Update to new connection API.
* gst/rtsp/rtspmessage.c: (rtsp_message_new_response),
(rtsp_message_init_response), (rtsp_message_init_data),
(rtsp_message_unset), (rtsp_message_free), (rtsp_message_dump):
* gst/rtsp/rtspmessage.h:
* gst/rtsp/rtsptransport.c: (rtsp_transport_free):
* gst/rtsp/rtsptransport.h:
* gst/rtsp/sdp.h:
* gst/rtsp/sdpmessage.h:
* gst/rtsp/gstrtsp.c:
* gst/rtsp/gstrtsp.h:
* gst/rtsp/gstrtpdec.c:
* gst/rtsp/gstrtpdec.h:
* gst/rtsp/rtsp.h:
* gst/rtsp/rtspdefs.c:
* gst/rtsp/rtspdefs.h:
Dual licensed under MIT and LGPL now.
2006-09-20 16:06:27 +00:00
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/*
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* Unless otherwise indicated, Source Code is licensed under MIT license.
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* See further explanation attached in License Statement (distributed in the file
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* LICENSE).
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*
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* Permission is hereby granted, free of charge, to any person obtaining a copy of
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* this software and associated documentation files (the "Software"), to deal in
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* the Software without restriction, including without limitation the rights to
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* use, copy, modify, merge, publish, distribute, sublicense, and/or sell copies
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* of the Software, and to permit persons to whom the Software is furnished to do
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* so, subject to the following conditions:
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*
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* The above copyright notice and this permission notice shall be included in all
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* copies or substantial portions of the Software.
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*
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* THE SOFTWARE IS PROVIDED "AS IS", WITHOUT WARRANTY OF ANY KIND, EXPRESS OR
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* IMPLIED, INCLUDING BUT NOT LIMITED TO THE WARRANTIES OF MERCHANTABILITY,
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* FITNESS FOR A PARTICULAR PURPOSE AND NONINFRINGEMENT. IN NO EVENT SHALL THE
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* AUTHORS OR COPYRIGHT HOLDERS BE LIABLE FOR ANY CLAIM, DAMAGES OR OTHER
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* LIABILITY, WHETHER IN AN ACTION OF CONTRACT, TORT OR OTHERWISE, ARISING FROM,
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* OUT OF OR IN CONNECTION WITH THE SOFTWARE OR THE USE OR OTHER DEALINGS IN THE
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* SOFTWARE.
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*/
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2005-05-11 07:44:44 +00:00
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#ifdef HAVE_CONFIG_H
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#include "config.h"
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#endif
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2006-02-09 14:20:14 +00:00
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#include "gstrtpdec.h"
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2005-05-11 07:44:44 +00:00
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#include "gstrtspsrc.h"
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gst/rtsp/: Add google RTSP extension, it can only handle udp and responds with unsupported if we do anything else. Fi...
Original commit message from CVS:
* gst/rtsp/Makefile.am:
* gst/rtsp/gstrtsp.c: (plugin_init):
* gst/rtsp/gstrtspgoogle.c: (gst_rtsp_google_before_send),
(gst_rtsp_google_after_send), (gst_rtsp_google_get_transports),
(_do_init), (gst_rtsp_google_base_init),
(gst_rtsp_google_class_init), (gst_rtsp_google_init),
(gst_rtsp_google_finalize), (gst_rtsp_google_change_state),
(gst_rtsp_google_extension_init):
* gst/rtsp/gstrtspgoogle.h:
Add google RTSP extension, it can only handle udp and responds with
unsupported if we do anything else. Fixes #546465.
* gst/rtsp/gstrtspsrc.c: (gst_rtspsrc_connection_send),
(gst_rtspsrc_connection_receive), (gst_rtspsrc_loop_send_cmd),
(gst_rtspsrc_create_transports_string),
(gst_rtspsrc_setup_streams), (gst_rtspsrc_open),
(gst_rtspsrc_close), (gst_rtspsrc_pause):
Make transport setup code a bit better using GString.
Add some more debug.
Check for closed connections before doing anything on them.
2008-08-20 17:30:19 +00:00
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#include "gstrtspgoogle.h"
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2005-05-11 07:44:44 +00:00
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static gboolean
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plugin_init (GstPlugin * plugin)
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{
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if (!gst_element_register (plugin, "rtspsrc", GST_RANK_NONE,
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GST_TYPE_RTSPSRC))
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return FALSE;
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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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if (!gst_element_register (plugin, "rtpdec", GST_RANK_NONE, GST_TYPE_RTP_DEC))
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2006-02-09 14:20:14 +00:00
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return FALSE;
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2005-05-11 07:44:44 +00:00
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gst/rtsp/: Add google RTSP extension, it can only handle udp and responds with unsupported if we do anything else. Fi...
Original commit message from CVS:
* gst/rtsp/Makefile.am:
* gst/rtsp/gstrtsp.c: (plugin_init):
* gst/rtsp/gstrtspgoogle.c: (gst_rtsp_google_before_send),
(gst_rtsp_google_after_send), (gst_rtsp_google_get_transports),
(_do_init), (gst_rtsp_google_base_init),
(gst_rtsp_google_class_init), (gst_rtsp_google_init),
(gst_rtsp_google_finalize), (gst_rtsp_google_change_state),
(gst_rtsp_google_extension_init):
* gst/rtsp/gstrtspgoogle.h:
Add google RTSP extension, it can only handle udp and responds with
unsupported if we do anything else. Fixes #546465.
* gst/rtsp/gstrtspsrc.c: (gst_rtspsrc_connection_send),
(gst_rtspsrc_connection_receive), (gst_rtspsrc_loop_send_cmd),
(gst_rtspsrc_create_transports_string),
(gst_rtspsrc_setup_streams), (gst_rtspsrc_open),
(gst_rtspsrc_close), (gst_rtspsrc_pause):
Make transport setup code a bit better using GString.
Add some more debug.
Check for closed connections before doing anything on them.
2008-08-20 17:30:19 +00:00
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if (!gst_element_register (plugin, "rtspgoogle", GST_RANK_SECONDARY,
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GST_TYPE_RTSP_GOOGLE)) {
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return FALSE;
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}
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2005-05-11 07:44:44 +00:00
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return TRUE;
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}
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GST_PLUGIN_DEFINE (GST_VERSION_MAJOR,
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GST_VERSION_MINOR,
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"rtsp",
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"transfer data via RTSP",
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2005-11-14 02:13:35 +00:00
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plugin_init, VERSION, GST_LICENSE, GST_PACKAGE_NAME, GST_PACKAGE_ORIGIN)
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