gstreamer/gst/rtsp/rtspdefs.h

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/* GStreamer
gst/rtsp/URLS: Added some test URLS. Original commit message from CVS: * gst/rtsp/URLS: Added some test URLS. * gst/rtsp/gstrtspsrc.c: (gst_rtspsrc_create_stream), (gst_rtspsrc_loop), (gst_rtspsrc_open): * gst/rtsp/gstrtspsrc.h: When creating streams, give access to the complete SDP. Fix some leaks. Collect and merge global stream properties in stream caps. Preliminary support for WMServer. * gst/rtsp/rtspconnection.c: (rtsp_connection_create), (rtsp_connection_connect), (rtsp_connection_read), (read_body), (rtsp_connection_receive): * gst/rtsp/rtspconnection.h: Make connection interruptable. Refactor to make it reconnectable. Don't fail on short reads when reading data packets. * gst/rtsp/rtspurl.c: (rtsp_url_parse), (rtsp_url_set_port), (rtsp_url_get_port): * gst/rtsp/rtspurl.h: Add methods for getting/setting the port. * gst/rtsp/sdpmessage.c: (sdp_message_get_attribute_val_n), (sdp_message_get_attribute_val), (sdp_media_get_attribute), (sdp_media_get_attribute_val_n), (sdp_media_get_attribute_val), (sdp_media_get_format), (sdp_parse_line), (sdp_message_parse_buffer): Fix headers. Add methods for getting multiple attributes with the same name. Increase buffer size when parsing. Fix parsing of a=foo fields. * gst/rtsp/test.c: (main): Update to new connection API. * gst/rtsp/rtspmessage.c: (rtsp_message_new_response), (rtsp_message_init_response), (rtsp_message_init_data), (rtsp_message_unset), (rtsp_message_free), (rtsp_message_dump): * gst/rtsp/rtspmessage.h: * gst/rtsp/rtsptransport.c: (rtsp_transport_free): * gst/rtsp/rtsptransport.h: * gst/rtsp/sdp.h: * gst/rtsp/sdpmessage.h: * gst/rtsp/gstrtsp.c: * gst/rtsp/gstrtsp.h: * gst/rtsp/gstrtpdec.c: * gst/rtsp/gstrtpdec.h: * gst/rtsp/rtsp.h: * gst/rtsp/rtspdefs.c: * gst/rtsp/rtspdefs.h: Dual licensed under MIT and LGPL now.
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* Copyright (C) <2005,2006> Wim Taymans <wim@fluendo.com>
*
* This library is free software; you can redistribute it and/or
* modify it under the terms of the GNU Library General Public
* License as published by the Free Software Foundation; either
* version 2 of the License, or (at your option) any later version.
*
* This library is distributed in the hope that it will be useful,
* but WITHOUT ANY WARRANTY; without even the implied warranty of
* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
* Library General Public License for more details.
*
* You should have received a copy of the GNU Library General Public
* License along with this library; if not, write to the
* Free Software Foundation, Inc., 59 Temple Place - Suite 330,
* Boston, MA 02111-1307, USA.
*/
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.
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/*
* Unless otherwise indicated, Source Code is licensed under MIT license.
* See further explanation attached in License Statement (distributed in the file
* LICENSE).
*
* Permission is hereby granted, free of charge, to any person obtaining a copy of
* this software and associated documentation files (the "Software"), to deal in
* the Software without restriction, including without limitation the rights to
* use, copy, modify, merge, publish, distribute, sublicense, and/or sell copies
* of the Software, and to permit persons to whom the Software is furnished to do
* so, subject to the following conditions:
*
* The above copyright notice and this permission notice shall be included in all
* copies or substantial portions of the Software.
*
* THE SOFTWARE IS PROVIDED "AS IS", WITHOUT WARRANTY OF ANY KIND, EXPRESS OR
* IMPLIED, INCLUDING BUT NOT LIMITED TO THE WARRANTIES OF MERCHANTABILITY,
* FITNESS FOR A PARTICULAR PURPOSE AND NONINFRINGEMENT. IN NO EVENT SHALL THE
* AUTHORS OR COPYRIGHT HOLDERS BE LIABLE FOR ANY CLAIM, DAMAGES OR OTHER
* LIABILITY, WHETHER IN AN ACTION OF CONTRACT, TORT OR OTHERWISE, ARISING FROM,
* OUT OF OR IN CONNECTION WITH THE SOFTWARE OR THE USE OR OTHER DEALINGS IN THE
* SOFTWARE.
*/
#ifndef __RTSP_DEFS_H__
#define __RTSP_DEFS_H__
#include <glib.h>
G_BEGIN_DECLS
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.
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#define RTSP_CHECK(stmt, label) \
if (G_UNLIKELY ((res = (stmt)) != RTSP_OK)) goto label
typedef enum {
RTSP_OK = 0,
/* errors */
RTSP_ERROR = -1,
RTSP_EINVAL = -2,
RTSP_EINTR = -3,
RTSP_ENOMEM = -4,
RTSP_ERESOLV = -5,
RTSP_ENOTIMPL = -6,
RTSP_ESYS = -7,
RTSP_EPARSE = -8,
RTSP_EWSASTART = -9,
RTSP_EWSAVERSION = -10,
RTSP_EEOF = -11,
RTSP_ENET = -12,
RTSP_ENOTIP = -13,
RTSP_ELAST = -14,
} RTSPResult;
typedef enum {
RTSP_PROTO_TCP,
RTSP_PROTO_UDP,
} RTSPProto;
typedef enum {
RTSP_FAM_NONE,
RTSP_FAM_INET,
RTSP_FAM_INET6,
} RTSPFamily;
typedef enum {
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.
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RTSP_STATE_INVALID,
RTSP_STATE_INIT,
RTSP_STATE_READY,
RTSP_STATE_PLAYING,
RTSP_STATE_RECORDING,
} RTSPState;
typedef enum {
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.
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RTSP_INVALID = 0,
RTSP_DESCRIBE = (1 << 0),
RTSP_ANNOUNCE = (1 << 1),
RTSP_GET_PARAMETER = (1 << 2),
RTSP_OPTIONS = (1 << 3),
RTSP_PAUSE = (1 << 4),
RTSP_PLAY = (1 << 5),
RTSP_RECORD = (1 << 6),
RTSP_REDIRECT = (1 << 7),
RTSP_SETUP = (1 << 8),
RTSP_SET_PARAMETER = (1 << 9),
RTSP_TEARDOWN = (1 << 10),
} RTSPMethod;
typedef enum {
/*
* R = Request
* r = response
* g = general
* e = entity
*/
RTSP_HDR_ACCEPT, /* Accept R opt. entity */
RTSP_HDR_ACCEPT_ENCODING, /* Accept-Encoding R opt. entity */
RTSP_HDR_ACCEPT_LANGUAGE, /* Accept-Language R opt. all */
RTSP_HDR_ALLOW, /* Allow r opt. all */
RTSP_HDR_AUTHORIZATION, /* Authorization R opt. all */
RTSP_HDR_BANDWIDTH, /* Bandwidth R opt. all */
RTSP_HDR_BLOCKSIZE, /* Blocksize R opt. all but OPTIONS, TEARDOWN */
RTSP_HDR_CACHE_CONTROL, /* Cache-Control g opt. SETUP */
RTSP_HDR_CONFERENCE, /* Conference R opt. SETUP */
RTSP_HDR_CONNECTION, /* Connection g req. all */
RTSP_HDR_CONTENT_BASE, /* Content-Base e opt. entity */
RTSP_HDR_CONTENT_ENCODING, /* Content-Encoding e req. SET_PARAMETER, DESCRIBE, ANNOUNCE */
RTSP_HDR_CONTENT_LANGUAGE, /* Content-Language e req. DESCRIBE, ANNOUNCE */
RTSP_HDR_CONTENT_LENGTH, /* Content-Length e req. SET_PARAMETER, ANNOUNCE, entity */
RTSP_HDR_CONTENT_LOCATION, /* Content-Location e opt. entity */
RTSP_HDR_CONTENT_TYPE, /* Content-Type e req. SET_PARAMETER, ANNOUNCE, entity */
RTSP_HDR_CSEQ, /* CSeq g req. all */
RTSP_HDR_DATE, /* Date g opt. all */
RTSP_HDR_EXPIRES, /* Expires e opt. DESCRIBE, ANNOUNCE */
RTSP_HDR_FROM, /* From R opt. all */
RTSP_HDR_IF_MODIFIED_SINCE, /* If-Modified-Since R opt. DESCRIBE, SETUP */
RTSP_HDR_LAST_MODIFIED, /* Last-Modified e opt. entity */
RTSP_HDR_PROXY_AUTHENTICATE, /* Proxy-Authenticate */
RTSP_HDR_PROXY_REQUIRE, /* Proxy-Require R req. all */
RTSP_HDR_PUBLIC, /* Public r opt. all */
RTSP_HDR_RANGE, /* Range Rr opt. PLAY, PAUSE, RECORD */
RTSP_HDR_REFERER, /* Referer R opt. all */
RTSP_HDR_REQUIRE, /* Require R req. all */
RTSP_HDR_RETRY_AFTER, /* Retry-After r opt. all */
RTSP_HDR_RTP_INFO, /* RTP-Info r req. PLAY */
RTSP_HDR_SCALE, /* Scale Rr opt. PLAY, RECORD */
RTSP_HDR_SESSION, /* Session Rr req. all but SETUP, OPTIONS */
RTSP_HDR_SERVER, /* Server r opt. all */
RTSP_HDR_SPEED, /* Speed Rr opt. PLAY */
RTSP_HDR_TRANSPORT, /* Transport Rr req. SETUP */
RTSP_HDR_UNSUPPORTED, /* Unsupported r req. all */
RTSP_HDR_USER_AGENT, /* User-Agent R opt. all */
RTSP_HDR_VIA, /* Via g opt. all */
RTSP_HDR_WWW_AUTHENTICATE, /* WWW-Authenticate r opt. all */
/* Real extensions */
RTSP_HDR_CLIENT_CHALLENGE, /* ClientChallenge */
RTSP_HDR_REAL_CHALLENGE1, /* RealChallenge1 */
RTSP_HDR_REAL_CHALLENGE2, /* RealChallenge2 */
RTSP_HDR_SUBSCRIBE, /* Subscribe */
} RTSPHeaderField;
typedef enum {
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.
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RTSP_STS_INVALID = 0,
RTSP_STS_CONTINUE = 100,
RTSP_STS_OK = 200,
RTSP_STS_CREATED = 201,
RTSP_STS_LOW_ON_STORAGE = 250,
RTSP_STS_MULTIPLE_CHOICES = 300,
RTSP_STS_MOVED_PERMANENTLY = 301,
RTSP_STS_MOVE_TEMPORARILY = 302,
RTSP_STS_SEE_OTHER = 303,
RTSP_STS_NOT_MODIFIED = 304,
RTSP_STS_USE_PROXY = 305,
RTSP_STS_BAD_REQUEST = 400,
RTSP_STS_UNAUTHORIZED = 401,
RTSP_STS_PAYMENT_REQUIRED = 402,
RTSP_STS_FORBIDDEN = 403,
RTSP_STS_NOT_FOUND = 404,
RTSP_STS_METHOD_NOT_ALLOWED = 405,
RTSP_STS_NOT_ACCEPTABLE = 406,
RTSP_STS_PROXY_AUTH_REQUIRED = 407,
RTSP_STS_REQUEST_TIMEOUT = 408,
RTSP_STS_GONE = 410,
RTSP_STS_LENGTH_REQUIRED = 411,
RTSP_STS_PRECONDITION_FAILED = 412,
RTSP_STS_REQUEST_ENTITY_TOO_LARGE = 413,
RTSP_STS_REQUEST_URI_TOO_LARGE = 414,
RTSP_STS_UNSUPPORTED_MEDIA_TYPE = 415,
RTSP_STS_PARAMETER_NOT_UNDERSTOOD = 451,
RTSP_STS_CONFERENCE_NOT_FOUND = 452,
RTSP_STS_NOT_ENOUGH_BANDWIDTH = 453,
RTSP_STS_SESSION_NOT_FOUND = 454,
RTSP_STS_METHOD_NOT_VALID_IN_THIS_STATE = 455,
RTSP_STS_HEADER_FIELD_NOT_VALID_FOR_RESOURCE = 456,
RTSP_STS_INVALID_RANGE = 457,
RTSP_STS_PARAMETER_IS_READONLY = 458,
RTSP_STS_AGGREGATE_OPERATION_NOT_ALLOWED = 459,
RTSP_STS_ONLY_AGGREGATE_OPERATION_ALLOWED = 460,
RTSP_STS_UNSUPPORTED_TRANSPORT = 461,
RTSP_STS_DESTINATION_UNREACHABLE = 462,
RTSP_STS_INTERNAL_SERVER_ERROR = 500,
RTSP_STS_NOT_IMPLEMENTED = 501,
RTSP_STS_BAD_GATEWAY = 502,
RTSP_STS_SERVICE_UNAVAILABLE = 503,
RTSP_STS_GATEWAY_TIMEOUT = 504,
RTSP_STS_RTSP_VERSION_NOT_SUPPORTED = 505,
RTSP_STS_OPTION_NOT_SUPPORTED = 551,
} RTSPStatusCode;
gchar* rtsp_strresult (RTSPResult result);
const gchar* rtsp_method_as_text (RTSPMethod method);
const gchar* rtsp_header_as_text (RTSPHeaderField field);
const gchar* rtsp_status_as_text (RTSPStatusCode code);
const gchar* rtsp_status_to_string (RTSPStatusCode code);
RTSPHeaderField rtsp_find_header_field (gchar *header);
RTSPMethod rtsp_find_method (gchar *method);
G_END_DECLS
#endif /* __RTSP_DEFS_H__ */