rtpsession.c

mediastreamer2是开源的网络传输媒体流的库

/*
  The oRTP library is an RTP (Realtime Transport Protocol - rfc3550) stack.
  Copyright (C) 2001  Simon MORLAT [e-mail]

  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
*/


#if defined(WIN32) || defined(_WIN32_WCE)
#include "ortp-config-win32.h"
#else
#include "ortp-config.h"
#endif

#include "ortp/ortp.h"
#include "ortp/telephonyevents.h"
#include "ortp/rtcp.h"
#include "jitterctl.h"
#include "scheduler.h"
#include "utils.h"
#include "rtpsession_priv.h"

extern mblk_t *rtcp_create_simple_bye_packet(uint32_t ssrc, const char *reason);
extern int rtcp_sr_init(RtpSession *session, char *buf, int size);
extern int rtcp_rr_init(RtpSession *session, char *buf, int size);



/* this function initialize all session parameter's that depend on the payload type */
static void payload_type_changed(RtpSession *session, PayloadType *pt){
	jitter_control_set_payload(&session->rtp.jittctl,pt);
	session->rtp.rtcp_report_snt_interval=RTCP_DEFAULT_REPORT_INTERVAL*pt->clock_rate;
	rtp_session_set_time_jump_limit(session,session->rtp.time_jump);
	if (pt->type==PAYLOAD_VIDEO){
		session->permissive=TRUE;
		ortp_message("Using permissive algorithm");
	}
	else session->permissive=FALSE;
}

void wait_point_init(WaitPoint *wp){
	ortp_mutex_init(&wp->lock,NULL);
	ortp_cond_init(&wp->cond,NULL);
	wp->time=0;
	wp->wakeup=FALSE;
}
void wait_point_uninit(WaitPoint *wp){
	ortp_cond_destroy(&wp->cond);
	ortp_mutex_destroy(&wp->lock);
}

#define wait_point_lock(wp) ortp_mutex_lock(&(wp)->lock)
#define wait_point_unlock(wp) ortp_mutex_unlock(&(wp)->lock)

void wait_point_wakeup_at(WaitPoint *wp, uint32_t t, bool_t dosleep){
	wp->time=t;
	wp->wakeup=TRUE;
	if (dosleep) ortp_cond_wait(&wp->cond,&wp->lock);
}


bool_t wait_point_check(WaitPoint *wp, uint32_t t){
	bool_t ok=FALSE;
	
	if (wp->wakeup){
		if (TIME_IS_NEWER_THAN(t,wp->time)){
			wp->wakeup=FALSE;
			ok=TRUE;
			
		}
	}
	return ok;
}
#define wait_point_wakeup(wp) ortp_cond_signal(&(wp)->cond);

extern void rtp_parse(RtpSession *session, mblk_t *mp, uint32_t local_str_ts,
		struct sockaddr *addr, socklen_t addrlen);


static uint32_t uint32_t_random(){
	return random();
}


#define RTP_SEQ_IS_GREATER(seq1,seq2)\
	((uint16_t)((uint16_t)(seq1) - (uint16_t)(seq2))< (uint16_t)(1<<15))

/* put an rtp packet in queue. It is called by rtp_parse()*/
void rtp_putq(queue_t *q, mblk_t *mp)
{
	mblk_t *tmp;
	rtp_header_t *rtp=(rtp_header_t*)mp->b_rptr,*tmprtp;
	/* insert message block by increasing time stamp order : the last (at the bottom)
		message of the queue is the newest*/
	ortp_debug("rtp_putq(): Enqueuing packet with ts=%i and seq=%i",rtp->timestamp,rtp->seq_number);
	
	if (qempty(q)) {
		putq(q,mp);
		return;
	}
	tmp=qlast(q);
	/* we look at the queue from bottom to top, because enqueued packets have a better chance
	to be enqueued at the bottom, since there are surely newer */
	while (!qend(q,tmp))
	{
		tmprtp=(rtp_header_t*)tmp->b_rptr;
		ortp_debug("rtp_putq(): Seeing packet with seq=%i",tmprtp->seq_number);
		
 		if (rtp->seq_number == tmprtp->seq_number)
 		{
 			/* this is a duplicated packet. Don't queue it */
 			ortp_debug("rtp_putq: duplicated message.");
 			freemsg(mp);
 			return;
		}else if (RTP_SEQ_IS_GREATER(rtp->seq_number,tmprtp->seq_number)){
			
			insq(q,tmp->b_next,mp);
			return;
 		}
		tmp=tmp->b_prev;
	}
	/* this packet is the oldest, it has to be 
	placed on top of the queue */
	insq(q,qfirst(q),mp);
	
}



mblk_t *rtp_getq(queue_t *q,uint32_t timestamp, int *rejected)
{
	mblk_t *tmp,*ret=NULL,*old=NULL;
	rtp_header_t *tmprtp;
	uint32_t ts_found=0;
	
	*rejected=0;
	ortp_debug("rtp_getq(): Timestamp %i wanted.",timestamp);

	if (qempty(q))
	{
		/*ortp_debug("rtp_getq: q is empty.");*/
		return NULL;
	}
	/* return the packet with ts just equal or older than the asked timestamp */
	/* packets with older timestamps are discarded */
	while ((tmp=qfirst(q))!=NULL)
	{
		tmprtp=(rtp_header_t*)tmp->b_rptr;
		ortp_debug("rtp_getq: Seeing packet with ts=%i",tmprtp->timestamp);
		if ( RTP_TIMESTAMP_IS_NEWER_THAN(timestamp,tmprtp->timestamp) )
		{
			if (ret!=NULL && tmprtp->timestamp==ts_found) {
				/* we've found two packets with same timestamp. return the first one */
				break;
			}
			if (old!=NULL) {
				ortp_debug("rtp_getq: discarding too old packet with ts=%i",ts_found);
				(*rejected)++;
				freemsg(old);
			}
			ret=getq(q); /* dequeue the packet, since it has an interesting timestamp*/
			ts_found=tmprtp->timestamp;
			ortp_debug("rtp_getq: Found packet with ts=%i",tmprtp->timestamp);
			old=ret;
		}
		else
		{
			break;
		}
	}
	return ret;
}

mblk_t *rtp_getq_permissive(queue_t *q,uint32_t timestamp, int *rejected)
{
	mblk_t *tmp,*ret=NULL;
	rtp_header_t *tmprtp;
	
	*rejected=0;
	ortp_debug("rtp_getq_permissive(): Timestamp %i wanted.",timestamp);

	if (qempty(q))
	{
		/*ortp_debug("rtp_getq: q is empty.");*/
		return NULL;
	}
	/* return the packet with the older timestamp (provided that it is older than
	the asked timestamp) */
	tmp=qfirst(q);
	tmprtp=(rtp_header_t*)tmp->b_rptr;
	ortp_debug("rtp_getq_permissive: Seeing packet with ts=%i",tmprtp->timestamp);
	if ( RTP_TIMESTAMP_IS_NEWER_THAN(timestamp,tmprtp->timestamp) )
	{
		ret=getq(q); /* dequeue the packet, since it has an interesting timestamp*/
		ortp_debug("rtp_getq_permissive: Found packet with ts=%i",tmprtp->timestamp);
	}
	return ret;
}


void
rtp_session_init (RtpSession * session, int mode)
{
	JBParameters jbp;
	memset (session, 0, sizeof (RtpSession));
	session->mode = (RtpSessionMode) mode;
	if ((mode == RTP_SESSION_RECVONLY) || (mode == RTP_SESSION_SENDRECV))
	{
		rtp_session_set_flag (session, RTP_SESSION_RECV_SYNC);
		rtp_session_set_flag (session, RTP_SESSION_RECV_NOT_STARTED);
		
	}
	if ((mode == RTP_SESSION_SENDONLY) || (mode == RTP_SESSION_SENDRECV))
	{
		rtp_session_set_flag (session, RTP_SESSION_SEND_NOT_STARTED);
		session->snd.ssrc=uint32_t_random();
		/* set default source description */
		rtp_session_set_source_description(session,"unknown@unknown",NULL,NULL,
				NULL,NULL,"oRTP-" ORTP_VERSION,"This is free sofware (LGPL) !");
	}
	session->snd.telephone_events_pt=-1;	/* not defined a priori */
	session->rcv.telephone_events_pt=-1;	/* not defined a priori */
	rtp_session_set_profile (session, &av_profile); /*the default profile to work with */
	session->rtp.socket=-1;
	session->rtcp.socket=-1;
#ifndef WIN32
	session->rtp.snd_socket_size=0;	/*use OS default value unless on windows where they are definitely too short*/
	session->rtp.rcv_socket_size=0;
#else
	session->rtp.snd_socket_size=session->rtp.rcv_socket_size=65536;
#endif
	session->dscp=RTP_DEFAULT_DSCP;
	session->multicast_ttl=RTP_DEFAULT_MULTICAST_TTL;
	session->multicast_loopback=RTP_DEFAULT_MULTICAST_LOOPBACK;
	qinit(&session->rtp.rq);
	qinit(&session->rtp.tev_rq);
	qinit(&session->contributing_sources);
	session->eventqs=NULL;
	/* init signal tables */
	rtp_signal_table_init (&session->on_ssrc_changed, session,"ssrc_changed");
	rtp_signal_table_init (&session->on_payload_type_changed, session,"payload_type_changed");
	rtp_signal_table_init (&session->on_telephone_event, session,"telephone-event");
	rtp_signal_table_init (&session->on_telephone_event_packet, session,"telephone-event_packet");
	rtp_signal_table_init (&session->on_timestamp_jump,session,"timestamp_jump");
	rtp_signal_table_init (&session->on_network_error,session,"network_error");
	rtp_signal_table_init (&session->on_rtcp_bye,session,"rtcp_bye");
	wait_point_init(&session->snd.wp);
	wait_point_init(&session->rcv.wp);
	/*defaults send payload type to 0 (pcmu)*/
	rtp_session_set_send_payload_type(session,0);
	/*sets supposed recv payload type to undefined */
	rtp_session_set_recv_payload_type(session,-1);
	/* configure jitter buffer with working default parameters */
	jbp.min_size=RTP_DEFAULT_JITTER_TIME;
	jbp.nom_size=RTP_DEFAULT_JITTER_TIME;
	jbp.max_size=-1;
	jbp.max_packets= 100;/* maximum number of packet allowed to be queued */
	jbp.adaptive=TRUE;
	rtp_session_enable_jitter_buffer(session,TRUE);
	rtp_session_set_jitter_buffer_params(session,&jbp);
	rtp_session_set_time_jump_limit(session,5000);
	rtp_session_enable_rtcp(session,TRUE);
	session->recv_buf_size = UDP_MAX_SIZE;
	session->symmetric_rtp = FALSE;
	session->permissive=FALSE;
	msgb_allocator_init(&session->allocator);
}


/**
 * Creates a new rtp session.
 * If the session is able to send data (RTP_SESSION_SENDONLY or
 * RTP_SESSION_SENDRECV), then a random SSRC number is choosed for 
 * the outgoing stream.
 * @param mode One of the RtpSessionMode flags.	
 *
 * @return the newly created rtp session.
**/
RtpSession *
rtp_session_new (int mode)
{
	RtpSession *session;
	session = (RtpSession *) ortp_malloc (sizeof (RtpSession));
	rtp_session_init (session, mode);
	return session;
}

/**
 * Sets the scheduling mode of the rtp session. If @yesno is TRUE, the rtp session is in
 *	the scheduled mode, that means that you can use session_set_select() to block until it's time
 *	to receive or send on this session according to the timestamp passed to the respective functions.
 *  You can also use blocking mode (see rtp_session_set_blocking_mode() ), to simply block within
 *	the receive and send functions.
 *	If @yesno is FALSE, the ortp scheduler will not manage those sessions, meaning that blocking mode 
 *  and the use of session_set_select() for this session are disabled.
 *@param session a rtp session.
 *@param yesno 	a boolean to indicate the scheduling mode.
 *
 *
**/
void
rtp_session_set_scheduling_mode (RtpSession * session, int yesno)
{
	if (yesno)
	{
		RtpScheduler *sched;
		sched = ortp_get_scheduler ();
		if (sched != NULL)
		{
			rtp_session_set_flag (session, RTP_SESSION_SCHEDULED);
			session->sched = sched;
			rtp_scheduler_add_session (sched, session);
		}
		else
			ortp_warning
				("rtp_session_set_scheduling_mode: Cannot use scheduled mode because the "
				 "scheduler is not started. Use ortp_scheduler_init() before.");
	}
	else
		rtp_session_unset_flag (session, RTP_SESSION_SCHEDULED);
}


/**
 *	This function implicitely enables the scheduling mode if yesno is TRUE.
 *	rtp_session_set_blocking_mode() defines the behaviour of the rtp_session_recv_with_ts() and 
 *	rtp_session_send_with_ts() functions. If @yesno is TRUE, rtp_session_recv_with_ts()
 *	will block until it is time for the packet to be received, according to the timestamp
 *	passed to the function. After this time, the function returns.
 *	For rtp_session_send_with_ts(), it will block until it is time for the packet to be sent.
 *	If @yesno is FALSE, then the two functions will return immediately.
 *
 *  @param session a rtp session
 *  @param yesno a boolean
**/
void
rtp_session_set_blocking_mode (RtpSession * session, int yesno)
{
	if (yesno){
		rtp_session_set_scheduling_mode(session,TRUE);
		rtp_session_set_flag (session, RTP_SESSION_BLOCKING_MODE);
	}else
		rtp_session_unset_flag (session, RTP_SESSION_BLOCKING_MODE);
}

/**
 *	Set the RTP profile to be used for the session. By default, all session are created by
 *	rtp_session_new() are initialized with the AV profile, as defined in RFC 3551. The application
 *	can set any other profile instead using that function.
 *
 * @param session a rtp session
 * @param profile a rtp profile
**/

void
rtp_session_set_profile (RtpSession * session, RtpProfile * profile)
{
	session->snd.profile = profile;
	session->rcv.profile = profile;
	rtp_session_telephone_events_supported(session);
}

/**
 *	By default oRTP automatically sends RTCP SR or RR packets. If
 *	yesno is set to FALSE, the RTCP sending of packet is disabled.
 *	This functionnality might be needed for some equipments that do not
 *	support RTCP, leading to a traffic of ICMP errors on the network.
 *	It can also be used to save bandwidth despite the RTCP bandwidth is 
 *	actually and usually very very low.
**/
void rtp_session_enable_rtcp(RtpSession *session, bool_t yesno){
	session->rtcp.enabled=yesno;
}

/**
 *	Set the RTP profile to be used for the sending by this session. By default, all session are created by
 *	rtp_session_new() are initialized with the AV profile, as defined in RFC 3551. The application
 *	can set any other profile instead using that function.
 * @param session a rtp session
 * @param profile a rtp profile
 *
**/

void
rtp_session_set_send_profile (RtpSession * session, RtpProfile * profile)
{
	session->snd.profile = profile;
	rtp_session_send_telephone_events_supported(session);