gstreamer/gst/rtsp/rtsptransport.c

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/* GStreamer
* Copyright (C) <2005,2006,2007> Wim Taymans <wim@fluendo.com>
* <2007> Peter Kjellerstedt <pkj at axis 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.
2006-09-20 16:06:27 +00:00
/*
* Unless otherwise indicated, Source Code is licensed under MIT license.
* See further explanation attached in License Statement (distributed in the file
* LICENSE).
*
* Permission is hereby granted, free of charge, to any person obtaining a copy of
* this software and associated documentation files (the "Software"), to deal in
* the Software without restriction, including without limitation the rights to
* use, copy, modify, merge, publish, distribute, sublicense, and/or sell copies
* of the Software, and to permit persons to whom the Software is furnished to do
* so, subject to the following conditions:
*
* The above copyright notice and this permission notice shall be included in all
* copies or substantial portions of the Software.
*
* THE SOFTWARE IS PROVIDED "AS IS", WITHOUT WARRANTY OF ANY KIND, EXPRESS OR
* IMPLIED, INCLUDING BUT NOT LIMITED TO THE WARRANTIES OF MERCHANTABILITY,
* FITNESS FOR A PARTICULAR PURPOSE AND NONINFRINGEMENT. IN NO EVENT SHALL THE
* AUTHORS OR COPYRIGHT HOLDERS BE LIABLE FOR ANY CLAIM, DAMAGES OR OTHER
* LIABILITY, WHETHER IN AN ACTION OF CONTRACT, TORT OR OTHERWISE, ARISING FROM,
* OUT OF OR IN CONNECTION WITH THE SOFTWARE OR THE USE OR OTHER DEALINGS IN THE
* SOFTWARE.
*/
#include <string.h>
#include <stdlib.h>
#include "rtsptransport.h"
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.
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#define MAX_MANAGERS 2
typedef enum
{
RTSP_TRANSPORT_DELIVERY = 1 << 0, /* multicast | unicast */
RTSP_TRANSPORT_DESTINATION = 1 << 1,
RTSP_TRANSPORT_SOURCE = 1 << 2,
RTSP_TRANSPORT_INTERLEAVED = 1 << 3,
RTSP_TRANSPORT_APPEND = 1 << 4,
RTSP_TRANSPORT_TTL = 1 << 5,
RTSP_TRANSPORT_LAYERS = 1 << 6,
RTSP_TRANSPORT_PORT = 1 << 7,
RTSP_TRANSPORT_CLIENT_PORT = 1 << 8,
RTSP_TRANSPORT_SERVER_PORT = 1 << 9,
RTSP_TRANSPORT_SSRC = 1 << 10,
RTSP_TRANSPORT_MODE = 1 << 11,
} RTSPTransportParameter;
typedef struct
{
const gchar *name;
const RTSPTransMode mode;
const gchar *gst_mime;
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.
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const gchar *manager[MAX_MANAGERS];
} RTSPTransMap;
static const RTSPTransMap transports[] = {
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.
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{"rtp", RTSP_TRANS_RTP, "application/x-rtp", {"rtpbin", "rtpdec"}},
{"x-real-rdt", RTSP_TRANS_RDT, "application/x-rdt", {NULL, NULL}},
{"x-pn-tng", RTSP_TRANS_RDT, "application/x-rdt", {NULL, NULL}},
{NULL, RTSP_TRANS_UNKNOWN, NULL, {NULL, NULL}}
};
typedef struct
{
const gchar *name;
const RTSPProfile profile;
} RTSPProfileMap;
static const RTSPProfileMap profiles[] = {
{"avp", RTSP_PROFILE_AVP},
{"savp", RTSP_PROFILE_SAVP},
{NULL, RTSP_PROFILE_UNKNOWN}
};
typedef struct
{
const gchar *name;
const RTSPLowerTrans ltrans;
} RTSPLTransMap;
static const RTSPLTransMap ltrans[] = {
{"udp", RTSP_LOWER_TRANS_UDP},
{"mcast", RTSP_LOWER_TRANS_UDP_MCAST},
{"tcp", RTSP_LOWER_TRANS_TCP},
{NULL, RTSP_LOWER_TRANS_UNKNOWN}
};
#define RTSP_TRANSPORT_PARAMETER_IS_UNIQUE(param) \
G_STMT_START { \
if ((transport_params & (param)) != 0) \
goto invalid_transport; \
transport_params |= (param); \
} G_STMT_END
RTSPResult
rtsp_transport_new (RTSPTransport ** transport)
{
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
g_return_val_if_fail (transport != NULL, RTSP_EINVAL);
trans = g_new0 (RTSPTransport, 1);
*transport = trans;
return rtsp_transport_init (trans);
}
RTSPResult
rtsp_transport_init (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);
g_free (transport->destination);
g_free (transport->source);
memset (transport, 0, sizeof (RTSPTransport));
transport->trans = RTSP_TRANS_RTP;
transport->profile = RTSP_PROFILE_AVP;
transport->lower_transport = RTSP_LOWER_TRANS_UDP_MCAST;
transport->mode_play = TRUE;
transport->mode_record = FALSE;
transport->interleaved.min = -1;
transport->interleaved.max = -1;
transport->port.min = -1;
transport->port.max = -1;
transport->client_port.min = -1;
transport->client_port.max = -1;
transport->server_port.min = -1;
transport->server_port.max = -1;
return RTSP_OK;
}
RTSPResult
rtsp_transport_get_mime (RTSPTransMode trans, const gchar ** mime)
{
gint i;
g_return_val_if_fail (mime != NULL, RTSP_EINVAL);
for (i = 0; transports[i].name; i++)
if (transports[i].mode == trans)
break;
*mime = transports[i].gst_mime;
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)
{
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;
return RTSP_OK;
}
static void
parse_mode (RTSPTransport * transport, const gchar * str)
{
transport->mode_play = (strstr (str, "play") != NULL);
transport->mode_record = (strstr (str, "record") != NULL);
}
static void
parse_range (const gchar * str, RTSPRange * range)
{
gchar *minus;
gchar *tmp;
/* even though strtol() allows white space, plus and minus in front of
* the number, we do not allow it
*/
if (g_ascii_isspace (*str) || *str == '+' || *str == '-')
goto invalid_range;
minus = strstr (str, "-");
if (minus) {
if (g_ascii_isspace (minus[1]) || minus[1] == '+' || minus[1] == '-')
goto invalid_range;
range->min = strtol (str, &tmp, 10);
if (str == tmp || tmp != minus)
goto invalid_range;
range->max = strtol (minus + 1, &tmp, 10);
if (*tmp && *tmp != ';')
goto invalid_range;
} else {
range->min = strtol (str, &tmp, 10);
if (str == tmp || (*tmp && *tmp != ';'))
goto invalid_range;
range->max = -1;
}
return;
invalid_range:
{
range->min = -1;
range->max = -1;
return;
}
}
static gchar *
range_as_text (const RTSPRange * range)
{
if (range->min < 0)
return NULL;
else if (range->max < 0)
return g_strdup_printf ("%d", range->min);
else
return g_strdup_printf ("%d-%d", range->min, range->max);
}
static const gchar *
rtsp_transport_mode_as_text (const RTSPTransport * transport)
{
gint i;
for (i = 0; transports[i].name; i++)
if (transports[i].mode == transport->trans)
return transports[i].name;
return NULL;
}
static const gchar *
rtsp_transport_profile_as_text (const RTSPTransport * transport)
{
gint i;
for (i = 0; profiles[i].name; i++)
if (profiles[i].profile == transport->profile)
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;
}
RTSPResult
rtsp_transport_parse (const gchar * str, RTSPTransport * transport)
{
gchar **split, *down, **transp = NULL;
guint transport_params = 0;
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);
rtsp_transport_init (transport);
/* case insensitive */
down = g_ascii_strdown (str, -1);
split = g_strsplit (down, ";", 0);
g_free (down);
/* First field contains the transport/profile/lower_transport */
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) {
for (i = 0; ltrans[i].name; i++)
if (strcmp (transp[2], ltrans[i].name) == 0)
break;
transport->lower_transport = ltrans[i].ltrans;
} else {
/* specifying the lower transport is optional */
if (transport->trans == RTSP_TRANS_RTP &&
transport->profile == RTSP_PROFILE_AVP)
transport->lower_transport = RTSP_LOWER_TRANS_UDP_MCAST;
else
transport->lower_transport = RTSP_LOWER_TRANS_UNKNOWN;
}
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;
while (split[i]) {
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;
transport->lower_transport = RTSP_LOWER_TRANS_UDP_MCAST;
} else if (strcmp (split[i], "unicast") == 0) {
RTSP_TRANSPORT_PARAMETER_IS_UNIQUE (RTSP_TRANSPORT_DELIVERY);
if (transport->lower_transport == RTSP_LOWER_TRANS_UDP_MCAST)
transport->lower_transport = RTSP_LOWER_TRANS_UDP;
} else if (g_str_has_prefix (split[i], "destination=")) {
RTSP_TRANSPORT_PARAMETER_IS_UNIQUE (RTSP_TRANSPORT_DESTINATION);
transport->destination = g_strdup (split[i] + 12);
} else if (g_str_has_prefix (split[i], "source=")) {
RTSP_TRANSPORT_PARAMETER_IS_UNIQUE (RTSP_TRANSPORT_SOURCE);
transport->source = g_strdup (split[i] + 7);
} else if (g_str_has_prefix (split[i], "layers=")) {
RTSP_TRANSPORT_PARAMETER_IS_UNIQUE (RTSP_TRANSPORT_LAYERS);
transport->layers = strtoul (split[i] + 7, NULL, 10);
} else if (g_str_has_prefix (split[i], "mode=")) {
RTSP_TRANSPORT_PARAMETER_IS_UNIQUE (RTSP_TRANSPORT_MODE);
parse_mode (transport, split[i] + 5);
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);
transport->append = TRUE;
} else if (g_str_has_prefix (split[i], "interleaved=")) {
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;
} else if (g_str_has_prefix (split[i], "ttl=")) {
RTSP_TRANSPORT_PARAMETER_IS_UNIQUE (RTSP_TRANSPORT_TTL);
transport->ttl = strtoul (split[i] + 4, NULL, 10);
if (transport->ttl >= 256)
goto invalid_transport;
} else if (g_str_has_prefix (split[i], "port=")) {
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;
} else if (g_str_has_prefix (split[i], "client_port=")) {
RTSP_TRANSPORT_PARAMETER_IS_UNIQUE (RTSP_TRANSPORT_CLIENT_PORT);
parse_range (split[i] + 12, &transport->client_port);
if (transport->client_port.min < 0 ||
transport->client_port.min >= 65536 ||
transport->client_port.max >= 65536)
goto invalid_transport;
} else if (g_str_has_prefix (split[i], "server_port=")) {
RTSP_TRANSPORT_PARAMETER_IS_UNIQUE (RTSP_TRANSPORT_SERVER_PORT);
parse_range (split[i] + 12, &transport->server_port);
if (transport->server_port.min < 0 ||
transport->server_port.min >= 65536 ||
transport->server_port.max >= 65536)
goto invalid_transport;
} else if (g_str_has_prefix (split[i], "ssrc=")) {
RTSP_TRANSPORT_PARAMETER_IS_UNIQUE (RTSP_TRANSPORT_SSRC);
transport->ssrc = strtoul (split[i] + 5, NULL, 16);
} else {
/* unknown field... */
g_warning ("unknown transport field \"%s\"", split[i]);
}
i++;
}
g_strfreev (split);
return RTSP_OK;
unsupported_transport:
{
g_strfreev (split);
return RTSP_ERROR;
}
invalid_transport:
{
g_strfreev (transp);
g_strfreev (split);
return RTSP_EINVAL;
}
}
gchar *
rtsp_transport_as_text (RTSPTransport * transport)
{
GPtrArray *strs;
gchar *res;
const gchar *tmp;
g_return_val_if_fail (transport != NULL, NULL);
strs = g_ptr_array_new ();
/* add the transport specifier */
if ((tmp = rtsp_transport_mode_as_text (transport)) == NULL)
goto invalid_transport;
g_ptr_array_add (strs, g_ascii_strup (tmp, -1));
g_ptr_array_add (strs, g_strdup ("/"));
if ((tmp = rtsp_transport_profile_as_text (transport)) == NULL)
goto invalid_transport;
g_ptr_array_add (strs, g_ascii_strup (tmp, -1));
if (transport->trans != RTSP_TRANS_RTP ||
transport->profile != RTSP_PROFILE_AVP ||
transport->lower_transport == RTSP_LOWER_TRANS_TCP) {
g_ptr_array_add (strs, g_strdup ("/"));
if ((tmp = rtsp_transport_ltrans_as_text (transport)) == NULL)
goto invalid_transport;
g_ptr_array_add (strs, g_ascii_strup (tmp, -1));
}
/*
* the order of the following parameters is the same as the one specified in
* RFC 2326 to please some weird RTSP clients that require it
*/
/* add the unicast/multicast parameter */
if (transport->lower_transport == RTSP_LOWER_TRANS_UDP_MCAST)
g_ptr_array_add (strs, g_strdup (";multicast"));
else
g_ptr_array_add (strs, g_strdup (";unicast"));
/* add the destination parameter */
if (transport->destination != NULL) {
g_ptr_array_add (strs, g_strdup (";destination="));
g_ptr_array_add (strs, g_strdup (transport->destination));
}
/* add the source parameter */
if (transport->source != NULL) {
g_ptr_array_add (strs, g_strdup (";source="));
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;
}
}
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);
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
return RTSP_OK;
}