/usr/include/gpac/internal/ietf_dev.h is in libgpac-dev 0.5.0+svn4288~dfsg1-4ubuntu1.
This file is owned by root:root, with mode 0o644.
The actual contents of the file can be viewed below.
1 2 3 4 5 6 7 8 9 10 11 12 13 14 15 16 17 18 19 20 21 22 23 24 25 26 27 28 29 30 31 32 33 34 35 36 37 38 39 40 41 42 43 44 45 46 47 48 49 50 51 52 53 54 55 56 57 58 59 60 61 62 63 64 65 66 67 68 69 70 71 72 73 74 75 76 77 78 79 80 81 82 83 84 85 86 87 88 89 90 91 92 93 94 95 96 97 98 99 100 101 102 103 104 105 106 107 108 109 110 111 112 113 114 115 116 117 118 119 120 121 122 123 124 125 126 127 128 129 130 131 132 133 134 135 136 137 138 139 140 141 142 143 144 145 146 147 148 149 150 151 152 153 154 155 156 157 158 159 160 161 162 163 164 165 166 167 168 169 170 171 172 173 174 175 176 177 178 179 180 181 182 183 184 185 186 187 188 189 190 191 192 193 194 195 196 197 198 199 200 201 202 203 204 205 206 207 208 209 210 211 212 213 214 215 216 217 218 219 220 221 222 223 224 225 226 227 228 229 230 231 232 233 234 235 236 237 238 239 240 241 242 243 244 245 246 247 248 249 250 251 252 253 254 255 256 257 258 259 260 261 262 263 264 265 266 267 268 269 270 271 272 273 274 275 276 277 278 279 280 281 282 283 284 285 286 287 288 289 290 291 292 293 294 295 296 297 298 299 300 301 302 303 304 305 306 307 308 309 310 311 312 313 314 315 316 317 318 319 320 321 322 323 324 325 326 327 328 329 330 331 332 333 334 335 336 337 338 339 340 341 342 343 344 | /*
* GPAC - Multimedia Framework C SDK
*
* Authors: Jean Le Feuvre
* Copyright (c) Telecom ParisTech 2000-2012
* All rights reserved
*
* This file is part of GPAC / IETF RTP/RTSP/SDP sub-project
*
* GPAC is free software; you can redistribute it and/or modify
* it under the terms of the GNU Lesser General Public License as published by
* the Free Software Foundation; either version 2, or (at your option)
* any later version.
*
* GPAC 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 Lesser General Public License for more details.
*
* You should have received a copy of the GNU Lesser General Public
* License along with this library; see the file COPYING. If not, write to
* the Free Software Foundation, 675 Mass Ave, Cambridge, MA 02139, USA.
*
*/
#ifndef _GF_IETF_DEV_H_
#define _GF_IETF_DEV_H_
#include <gpac/ietf.h>
#ifndef GPAC_DISABLE_STREAMING
#include <gpac/thread.h>
/*
RTP intern
*/
typedef struct
{
/*version of the packet. Must be 2*/
u8 Version;
/*padding bits at the end of the payload*/
u8 Padding;
/*number of reports*/
u8 Count;
/*payload type of RTCP pck*/
u8 PayloadType;
/*The length of this RTCP packet in 32-bit words minus one including the header and any padding*/
u16 Length;
/*sync source identifier*/
u32 SSRC;
} GF_RTCPHeader;
typedef struct __PRO_item
{
struct __PRO_item *next;
u32 pck_seq_num;
void *pck;
u32 size;
} GF_POItem;
typedef struct __PO
{
struct __PRO_item *in;
u32 head_seqnum;
u32 Count;
u32 MaxCount;
u32 IsInit;
u32 MaxDelay, LastTime;
} GF_RTPReorder;
/* creates new RTP reorderer
@MaxCount: forces automatic packet flush. 0 means no flush
@MaxDelay: is the max time in ms the queue will wait for a missing packet
*/
GF_RTPReorder *gf_rtp_reorderer_new(u32 MaxCount, u32 MaxDelay);
void gf_rtp_reorderer_del(GF_RTPReorder *po);
/*reset the Queue*/
void gf_rtp_reorderer_reset(GF_RTPReorder *po);
/*Adds a packet to the queue. Packet Data is memcopied*/
GF_Err gf_rtp_reorderer_add(GF_RTPReorder *po, const void * pck, u32 pck_size, u32 pck_seqnum);
/*gets the output of the queue. Packet Data IS YOURS to delete*/
void *gf_rtp_reorderer_get(GF_RTPReorder *po, u32 *pck_size);
/*the RTP channel with both RTP and RTCP sockets and buffers
each channel is identified by a control string given in RTSP Describe
this control string is used with Darwin
*/
struct __tag_rtp_channel
{
/*global transport info for the session*/
GF_RTSPTransport net_info;
/*RTP CHANNEL*/
GF_Socket *rtp;
/*RTCP CHANNEL*/
GF_Socket *rtcp;
/*RTP Packet reordering. Turned on/off during initialization. The library forces a 200 ms
max latency at the reordering queue*/
GF_RTPReorder *po;
/*RTCP report times*/
u32 last_report_time;
u32 next_report_time;
/*NAT keep-alive*/
u32 last_nat_keepalive_time, nat_keepalive_time_period;
/*the seq number of the first packet as signaled by the server if any, or first
RTP SN received (RTP multicast)*/
u32 rtp_first_SN;
/*the TS of the associated first packet as signaled by the server if any, or first
RTP TS received (RTP multicast)*/
u32 rtp_time;
/*NPT from the rtp_time*/
u32 CurrentTime;
/*num loops of pck sn*/
u32 num_sn_loops;
/*some mapping info - we should support # payloads*/
u8 PayloadType;
u32 TimeScale;
/*static buffer for RTP sending*/
char *send_buffer;
u32 send_buffer_size;
u32 pck_sent_since_last_sr;
u32 last_pck_ts;
u32 last_pck_ntp_sec, last_pck_ntp_frac;
u32 num_pck_sent, num_payload_bytes;
Bool no_auto_rtcp;
/*RTCP info*/
char *s_name, *s_email, *s_location, *s_phone, *s_tool, *s_note, *s_priv;
// s8 first_rtp_pck;
s8 first_SR;
u32 SSRC;
u32 SenderSSRC;
u32 last_pck_sn;
/*indicates if a packet loss is detected between current and previous packet*/
Bool packet_loss;
char *CName;
u32 rtcp_bytes_sent;
/*total pck rcv*/
u32 tot_num_pck_rcv, tot_num_pck_expected;
/*stats since last SR*/
u32 last_num_pck_rcv, last_num_pck_expected, last_num_pck_loss;
/*jitter compute*/
u32 Jitter, ntp_init;
s32 last_deviance;
/*NTP of last SR*/
u32 last_SR_NTP_sec, last_SR_NTP_frac;
/*RTP time at last SR as indicated in SR*/
u32 last_SR_rtp_time;
/*payload info*/
u32 total_pck, total_bytes;
};
/*gets UTC in the channel RTP timescale*/
u32 gf_rtp_channel_time(GF_RTPChannel *ch);
/*gets time in 1/65536 seconds (for reports)*/
u32 gf_rtp_get_report_time();
/*updates the time for the next report (SR, RR)*/
void gf_rtp_get_next_report_time(GF_RTPChannel *ch);
/*
RTSP intern
*/
#define GF_RTSP_DEFAULT_BUFFER 2048
#define GF_RTSP_VERSION "RTSP/1.0"
/*macros for RTSP command and response formmating*/
#define RTSP_WRITE_STEPALLOC 250
#define RTSP_WRITE_ALLOC_STR_WITHOUT_CHECK(buf, buf_size, pos, str) \
if (strlen(str)+pos >= buf_size) { \
buf_size += RTSP_WRITE_STEPALLOC; \
buf = (char *) gf_realloc(buf, buf_size); \
} \
strcpy(buf+pos, (const char *) str); \
pos += strlen((const char *) str); \
#define RTSP_WRITE_ALLOC_STR(buf, buf_size, pos, str) \
if (str){ \
RTSP_WRITE_ALLOC_STR_WITHOUT_CHECK(buf, buf_size, pos, str); \
} \
#define RTSP_WRITE_HEADER(buf, buf_size, pos, type, str) \
if( str ) { \
RTSP_WRITE_ALLOC_STR(buf, buf_size, pos, type); \
RTSP_WRITE_ALLOC_STR(buf, buf_size, pos, ": "); \
RTSP_WRITE_ALLOC_STR(buf, buf_size, pos, str); \
RTSP_WRITE_ALLOC_STR(buf, buf_size, pos, "\r\n"); \
} \
#define RTSP_WRITE_INT(buf, buf_size, pos, d, sig) \
if (sig < 0) { \
sprintf(temp, "%d", d); \
} else { \
sprintf(temp, "%d", d); \
} \
RTSP_WRITE_ALLOC_STR_WITHOUT_CHECK(buf, buf_size, pos, temp);
#define RTSP_WRITE_FLOAT_WITHOUT_CHECK(buf, buf_size, pos, d) \
sprintf(temp, "%.4f", d); \
RTSP_WRITE_ALLOC_STR_WITHOUT_CHECK(buf, buf_size, pos, temp);
#define RTSP_WRITE_FLOAT(buf, buf_size, pos, d) \
sprintf(temp, "%.4f", d); \
RTSP_WRITE_ALLOC_STR(buf, buf_size, pos, temp);
/*default packet size, but resize on the fly if needed*/
#define RTSP_PCK_SIZE 6000
#define RTSP_TCP_BUF_SIZE 0x10000ul
typedef struct
{
u8 rtpID;
u8 rtcpID;
void *ch_ptr;
} GF_TCPChan;
/**************************************
RTSP Session
***************************************/
struct _tag_rtsp_session
{
/*service name (extracted from URL) ex: news/latenight.mp4, vod.mp4 ...*/
char *Service;
/*server name (extracted from URL)*/
char *Server;
/*server port (extracted from URL)*/
u16 Port;
/*if RTSP is on UDP*/
u8 ConnectionType;
/*TCP interleaving ID*/
u8 InterID;
/*http tunnel*/
Bool HasTunnel;
GF_Socket *http;
char HTTP_Cookie[30];
u32 CookieRadLen;
/*RTSP CHANNEL*/
GF_Socket *connection;
u32 SockBufferSize;
/*needs connection*/
u32 NeedConnection;
/*the RTSP sequence number*/
u32 CSeq;
/*this is for aggregated request in order to check SeqNum*/
u32 NbPending;
/*RTSP sessionID, arbitrary length, alpha-numeric*/
const char *last_session_id;
/*RTSP STATE machine*/
u32 RTSP_State;
char RTSPLastRequest[40];
/*current buffer from TCP if any*/
char TCPBuffer[RTSP_TCP_BUF_SIZE];
u32 CurrentSize, CurrentPos;
/*RTSP interleaving*/
GF_Err (*RTSP_SignalData)(GF_RTSPSession *sess, void *chan, char *buffer, u32 bufferSize, Bool IsRTCP);
/*buffer for pck reconstruction*/
char *rtsp_pck_buf;
u32 rtsp_pck_size;
u32 pck_start, payloadSize;
/*all RTP channels in an interleaved RTP on RTSP session*/
GF_List *TCPChannels;
/*thread-safe, full duplex library for PLAY and RECORD*/
GF_Mutex *mx;
char *MobileIP;
};
GF_RTSPSession *gf_rtsp_session_new(char *sURL, u16 DefaultPort);
/*check connection status*/
GF_Err gf_rtsp_check_connection(GF_RTSPSession *sess);
/*send data on RTSP*/
GF_Err gf_rtsp_send_data(GF_RTSPSession *sess, char *buffer, u32 Size);
/*
Common RTSP tools
*/
/*locate body-start and body size in response/commands*/
void gf_rtsp_get_body_info(GF_RTSPSession *sess, u32 *body_start, u32 *body_size);
/*read TCP until a full command/response is received*/
GF_Err gf_rtsp_read_reply(GF_RTSPSession *sess);
/*fill the TCP buffer*/
GF_Err gf_rtsp_fill_buffer(GF_RTSPSession *sess);
/*force a fill on TCP buffer - used for de-interleaving and TCP-fragmented RTSP messages*/
GF_Err gf_rtsp_refill_buffer(GF_RTSPSession *sess);
/*parses a transport string and returns a transport structure*/
GF_RTSPTransport *gf_rtsp_transport_parse(char *buffer);
/*parsing of header for com and rsp*/
GF_Err gf_rtsp_parse_header(char *buffer, u32 BufferSize, u32 BodyStart, GF_RTSPCommand *com, GF_RTSPResponse *rsp);
void gf_rtsp_set_command_value(GF_RTSPCommand *com, char *Header, char *Value);
void gf_rtsp_set_response_value(GF_RTSPResponse *rsp, char *Header, char *Value);
/*deinterleave a data packet*/
GF_Err gf_rtsp_set_deinterleave(GF_RTSPSession *sess);
/*start session through HTTP tunnel (QTSS)*/
GF_Err gf_rtsp_http_tunnel_start(GF_RTSPSession *sess, char *UserAgent);
/*packetization routines*/
GF_Err gp_rtp_builder_do_mpeg4(GP_RTPPacketizer *builder, char *data, u32 data_size, u8 IsAUEnd, u32 FullAUSize);
GF_Err gp_rtp_builder_do_h263(GP_RTPPacketizer *builder, char *data, u32 data_size, u8 IsAUEnd, u32 FullAUSize);
GF_Err gp_rtp_builder_do_amr(GP_RTPPacketizer *builder, char *data, u32 data_size, u8 IsAUEnd, u32 FullAUSize);
#ifndef GPAC_DISABLE_AV_PARSERS
GF_Err gp_rtp_builder_do_mpeg12_video(GP_RTPPacketizer *builder, char *data, u32 data_size, u8 IsAUEnd, u32 FullAUSize);
#endif
GF_Err gp_rtp_builder_do_mpeg12_audio(GP_RTPPacketizer *builder, char *data, u32 data_size, u8 IsAUEnd, u32 FullAUSize);
GF_Err gp_rtp_builder_do_tx3g(GP_RTPPacketizer *builder, char *data, u32 data_size, u8 IsAUEnd, u32 FullAUSize, u32 duration, u8 descIndex);
GF_Err gp_rtp_builder_do_avc(GP_RTPPacketizer *builder, char *data, u32 data_size, u8 IsAUEnd, u32 FullAUSize);
GF_Err gp_rtp_builder_do_qcelp(GP_RTPPacketizer *builder, char *data, u32 data_size, u8 IsAUEnd, u32 FullAUSize);
GF_Err gp_rtp_builder_do_smv(GP_RTPPacketizer *builder, char *data, u32 data_size, u8 IsAUEnd, u32 FullAUSize);
GF_Err gp_rtp_builder_do_latm(GP_RTPPacketizer *builder, char *data, u32 data_size, u8 IsAUEnd, u32 FullAUSize, u32 duration);
GF_Err gp_rtp_builder_do_dims(GP_RTPPacketizer *builder, char *data, u32 data_size, u8 IsAUEnd, u32 FullAUSize, u32 duration);
GF_Err gp_rtp_builder_do_ac3(GP_RTPPacketizer *builder, char *data, u32 data_size, u8 IsAUEnd, u32 FullAUSize);
#endif /*GPAC_DISABLE_STREAMING*/
#endif /*_GF_IETF_DEV_H_*/
|