/*  The oRTP library is an RTP (Realtime Transport Protocol - rfc3550) stack.  Copyright (C) 2001  Simon MORLAT simon.morlat@linphone.org  This library 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 2.1 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  Lesser General Public License for more details.  You should have received a copy of the GNU Lesser 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*/#ifndef RTP_H#define RTP_H#include <ortp/port.h>#include <ortp/str_utils.h>#define IPMAXLEN 20#define UDP_MAX_SIZE 65536#define RTP_FIXED_HEADER_SIZE 12#define RTP_DEFAULT_JITTER_TIME 80	/*miliseconds*/#define RTP_DEFAULT_MULTICAST_TTL 5	/*hops*/#define RTP_DEFAULT_MULTICAST_LOOPBACK 0  /*false*/#define RTP_DEFAULT_DSCP 0x00  /*best effort*/typedef struct rtp_header{#ifdef ORTP_BIGENDIAN	uint16_t version:2;	uint16_t padbit:1;	uint16_t extbit:1;	uint16_t cc:4;	uint16_t markbit:1;	uint16_t paytype:7;#else	uint16_t cc:4;	uint16_t extbit:1;	uint16_t padbit:1;	uint16_t version:2;	uint16_t paytype:7;	uint16_t markbit:1;#endif	uint16_t seq_number;	uint32_t timestamp;	uint32_t ssrc;	uint32_t csrc[16];} rtp_header_t;typedef struct rtp_stats{	uint64_t packet_sent;	uint64_t sent;		/* bytes sent */	uint64_t recv; 		/* bytes of payload received and delivered in time to the application */	uint64_t hw_recv;		/* bytes of payload received */	uint64_t packet_recv;	/* number of packets received */	uint64_t unavaillable;	/* packets not availlable when they were queried */	uint64_t outoftime;		/* number of packets that were received too late */	uint64_t skipped;		/* number of packets skipped (that the application never queried 											or that need to be skipped in order to compensate a clock slide (see adaptive jitter control)) */	uint64_t cum_packet_loss; /* cumulative number of packet lost */	uint64_t bad;			/* packets that did not appear to be RTP */	uint64_t discarded;		/* incoming packets discarded because the queue exceeds its max size */} rtp_stats_t;#define RTP_TIMESTAMP_IS_NEWER_THAN(ts1,ts2) \	((uint32_t)((uint32_t)(ts1) - (uint32_t)(ts2))< (uint32_t)(1<<31))#define RTP_TIMESTAMP_IS_STRICTLY_NEWER_THAN(ts1,ts2) \	( ((uint32_t)((uint32_t)(ts1) - (uint32_t)(ts2))< (uint32_t)(1<<31)) && (ts1)!=(ts2) )#define TIME_IS_NEWER_THAN(t1,t2) RTP_TIMESTAMP_IS_NEWER_THAN(t1,t2)#define TIME_IS_STRICTLY_NEWER_THAN(t1,t2) RTP_TIMESTAMP_IS_STRICTLY_NEWER_THAN(t1,t2)/* packet api *//* the first argument is a mblk_t. The header is supposed to be not splitted  */#define rtp_set_markbit(mp,value)		((rtp_header_t*)((mp)->b_rptr))->markbit=(value)#define rtp_set_seqnumber(mp,seq)	((rtp_header_t*)((mp)->b_rptr))->seq_number=(seq)#define rtp_set_timestamp(mp,ts)	((rtp_header_t*)((mp)->b_rptr))->timestamp=(ts)#define rtp_set_ssrc(mp,_ssrc)		((rtp_header_t*)((mp)->b_rptr))->ssrc=(_ssrc)void rtp_add_csrc(mblk_t *mp ,uint32_t csrc);#define rtp_set_payload_type(mp,pt)	((rtp_header_t*)((mp)->b_rptr))->paytype=(pt)#define rtp_get_markbit(mp)	(((rtp_header_t*)((mp)->b_rptr))->markbit)	#define rtp_get_timestamp(mp)	(((rtp_header_t*)((mp)->b_rptr))->timestamp)	#define rtp_get_seqnumber(mp)	(((rtp_header_t*)((mp)->b_rptr))->seq_number)#define rtp_get_payload_type(mp)	(((rtp_header_t*)((mp)->b_rptr))->paytype)#define rtp_get_ssrc(mp)		(((rtp_header_t*)((mp)->b_rptr))->ssrc)#endif