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  The oRTP library is an RTP (Realtime Transport Protocol - rfc1889) 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
  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


#include <ortp/rtpport.h>
#include <ortp/rtp.h>
#include <ortp/payloadtype.h>
#include <ortp/sessionset.h>
#include <ortp/rtcp.h>
#include <ortp/str_utils.h>
#include <ortp/rtpsignaltable.h>

#include <stdio.h>

#ifndef _WIN32
# include <sys/types.h>
# include <sys/socket.h>
# include <errno.h>
# include <netinet/in.h>
#  include <arpa/inet.h>
# endif
# include <unistd.h>
# include <sys/time.h>
# include <winsock2.h>
#endif   /* _WIN32 */

typedef enum {
} RtpSessionMode;

typedef enum {
      RTP_SESSION_RECV_SYNC=1,      /* the rtp session is synchronising in the incoming stream */
      RTP_SESSION_SEND_SYNC=1<<1, /* the rtp session is synchronising in the outgoing stream */
      RTP_SESSION_SCHEDULED=1<<2, /* the rtp session has to be scheduled */
      RTP_SESSION_BLOCKING_MODE=1<<3, /* in blocking mode */
      RTP_SESSION_RECV_NOT_STARTED=1<<4,  /* the application has not started to try to recv */
      RTP_SESSION_SEND_NOT_STARTED=1<<5,  /* the application has not started to send something */
      RTP_SESSION_IN_SCHEDULER=1<<6,      /* the rtp session is in the scheduler list */
      RTP_SESSION_USING_EXT_SOCKETS=1<<7 /* the session is using externaly supplied sockets */

typedef struct _JitterControl
      gint jitt_comp;   /* the user jitt_comp in miliseconds*/
      gint jitt_comp_ts; /* the jitt_comp converted in rtp time (same unit as timestamp) */
      gint adapt_jitt_comp_ts;
      float slide;
      float jitter;
      gint count;
      gint olddiff;
      float inter_jitter;     /* interarrival jitter as defined in the RFC */
      gint corrective_step;
      gint corrective_slide;
      gboolean adaptive;
} JitterControl;

typedef struct _WaitPoint
      GMutex *lock;
      GCond *cond;
      guint32 time;
      gboolean wakeup;
} WaitPoint;
typedef struct _RtpStream
      gint socket;
      gint sockfamily;
      gint max_rq_size;
      gint time_jump;
      guint32 ts_jump;
      queue_t rq;
      queue_t tev_rq;
      mblk_t *cached_mp;
      int loc_port;
#ifdef INET6
      struct sockaddr_storage rem_addr;
      struct sockaddr_in rem_addr;
      int rem_addrlen;
      JitterControl jittctl;
      guint32 snd_time_offset;/*the scheduler time when the application send its first timestamp*/    
      guint32 snd_ts_offset;  /* the first application timestamp sent by the application */
      guint32 snd_rand_offset;      /* a random number added to the user offset to make the stream timestamp*/
      guint32 snd_last_ts;    /* the last stream timestamp sended */
      guint32 rcv_time_offset; /*the scheduler time when the application ask for its first timestamp*/
      guint32 rcv_ts_offset;  /* the first stream timestamp */
      guint32 rcv_query_ts_offset;  /* the first user timestamp asked by the application */
      guint32 rcv_diff_ts;    /* difference between the first user timestamp and first stream timestamp */
      guint32 hwrcv_diff_ts;
      guint32 rcv_ts;               /* to be unused */
      guint32 rcv_last_ts;    /* the last stream timestamp got by the application */
      guint32 rcv_last_app_ts; /* the last application timestamp asked by the application */    
      guint32 rcv_last_ret_ts; /* the timestamp of the last sample returned (only for continuous audio)*/
      poly32_t hwrcv_extseq; /* last received on socket extended sequence number */
      guint32 hwrcv_seq_at_last_SR;
      guint hwrcv_since_last_SR;
      guint32 last_rcv_SR_ts;     /* NTP timestamp (middle 32 bits) of last received SR */
      struct timeval last_rcv_SR_time;   /* time at which last SR was received  */
      guint16 snd_seq; /* send sequence number */
      guint32 last_rtcp_report_snt_r;     /* the time of the last rtcp report sent, in recv timestamp unit */
      guint32 last_rtcp_report_snt_s;     /* the time of the last rtcp report sent, in send timestamp unit */
      guint32 rtcp_report_snt_interval; /* the interval in timestamp unit between rtcp report sent */
      rtp_stats_t stats;

typedef struct _RtcpStream
      gint socket;
      gint sockfamily;
      mblk_t *cached_mp;
#ifdef INET6
      struct sockaddr_storage rem_addr;
      struct sockaddr_in rem_addr;
      int rem_addrlen;
} RtcpStream;

typedef struct _RtpSession RtpSession;

struct _RtpSession
      RtpSession *next; /* next RtpSession, when the session are enqueued by the scheduler */
      RtpProfile *profile;
      WaitPoint recv_wp;
      WaitPoint send_wp;
      GMutex *lock;
      guint32 send_ssrc;
      guint32 recv_ssrc;
      gint send_pt;/* sent payload type */
      gint recv_pt;/* recv payload type */
      gint max_buf_size;
      RtpSignalTable on_ssrc_changed;
      RtpSignalTable on_payload_type_changed;
      RtpSignalTable on_telephone_event_packet;
      RtpSignalTable on_telephone_event;
      RtpSignalTable on_timestamp_jump;
      RtpSignalTable on_network_error;
      struct _OList *signal_tables;
      RtpStream rtp;
      RtcpStream rtcp;
      RtpSessionMode mode;
      struct _RtpScheduler *sched;
      guint32 flags;
      gint mask_pos;    /* the position in the scheduler mask of RtpSession */
      gpointer user_data;
      /* telephony events extension */
      gint telephone_events_pt;     /* the payload type used for telephony events */
      mblk_t *current_tev;          /* the pending telephony events */
      mblk_t *sd;
      queue_t contributing_sources;

#ifdef __cplusplus
extern "C"

/*private */
void rtp_session_init(RtpSession *session, gint mode);
#define rtp_session_lock(session)   g_mutex_lock(session->lock)
#define rtp_session_unlock(session) g_mutex_unlock(session->lock)
#define rtp_session_set_flag(session,flag) (session)->flags|=(flag)
#define rtp_session_unset_flag(session,flag) (session)->flags&=~(flag)
void rtp_session_uninit(RtpSession *session);

/* public API */
RtpSession *rtp_session_new(gint mode);
void rtp_session_set_scheduling_mode(RtpSession *session, gint yesno);
void rtp_session_set_blocking_mode(RtpSession *session, gint yesno);
void rtp_session_set_profile(RtpSession *session,RtpProfile *profile);
RtpProfile *rtp_session_get_profile(RtpSession *session);
int rtp_session_signal_connect(RtpSession *session,const gchar *signal, RtpCallback cb, gpointer user_data);
int rtp_session_signal_disconnect_by_callback(RtpSession *session,const gchar *signal, RtpCallback cb);
void rtp_session_set_ssrc(RtpSession *session, guint32 ssrc);
void rtp_session_set_seq_number(RtpSession *session, guint16 seq);
guint16 rtp_session_get_seq_number(RtpSession *session);
void rtp_session_set_jitter_compensation(RtpSession *session, int milisec);
void rtp_session_enable_adaptive_jitter_compensation(RtpSession *session, gboolean val);
gboolean rtp_session_adaptive_jitter_compensation_enabled(RtpSession *session);
void rtp_session_set_time_jump_limit(RtpSession *session, gint miliseconds);
int rtp_session_set_local_addr(RtpSession *session,const gchar *addr, gint port);
int rtp_session_get_local_port(const RtpSession *session);
gint rtp_session_set_remote_addr(RtpSession *session,const gchar *addr, gint port);
/* alternatively to the set_remote_addr() and set_local_addr(), an application can give
a valid socket (potentially connect()ed )to be used by the RtpSession */
void rtp_session_set_sockets(RtpSession *session, gint rtpfd, gint rtcpfd);

int rtp_session_set_send_payload_type(RtpSession *session, int paytype);
int rtp_session_get_send_payload_type(const RtpSession *session);

int rtp_session_get_recv_payload_type(const RtpSession *session);
int rtp_session_set_recv_payload_type(RtpSession *session, int pt);

int rtp_session_set_payload_type(RtpSession *session, int pt);

/* deprecated API:
int rtp_session_set_payload_type(RtpSession *session, int paytype);
int rtp_session_get_payload_type(RtpSession *session);
int rtp_session_set_payload_type_with_string (RtpSession * session, const char * mime);

/*low level recv and send functions */
mblk_t * rtp_session_recvm_with_ts (RtpSession * session, guint32 user_ts);
mblk_t * rtp_session_create_packet(RtpSession *session,gint header_size, const char *payload, gint payload_size);
mblk_t * rtp_session_create_packet_with_data(RtpSession *session, char *payload, gint payload_size, void (*freefn)(void*));
mblk_t * rtp_session_create_packet_in_place(RtpSession *session,char *buffer, gint size, void (*freefn)(void*) );
gint rtp_session_sendm_with_ts (RtpSession * session, mblk_t *mp, guint32 userts);
/* high level recv and send functions */
gint rtp_session_recv_with_ts(RtpSession *session, gchar *buffer, gint len, guint32 time, gint *have_more);
gint rtp_session_send_with_ts(RtpSession *session, const gchar *buffer, gint len, guint32 userts);

guint32 rtp_session_get_current_send_ts(RtpSession *session);
guint32 rtp_session_get_current_recv_ts(RtpSession *session);
void rtp_session_flush_sockets(RtpSession *session);
void rtp_session_release_sockets(RtpSession *session);
void rtp_session_reset(RtpSession *session);
void rtp_session_destroy(RtpSession *session);

#define rtp_session_get_stats(session) (&(session)->stats)
#define rtp_session_reset_stats(session)  memset(&(session)->stats,0,sizeof(rtp_stats_t))
void rtp_session_set_data(RtpSession *session, void *data);
void *rtp_session_get_data(const RtpSession *session);

#define rtp_session_max_buf_size_set(session,bufsize) (session)->max_buf_size=(bufsize)

/* in use with the scheduler to convert a timestamp in scheduler time unit (ms) */
guint32 rtp_session_ts_to_time(RtpSession *session,guint32 timestamp);
guint32 rtp_session_time_to_ts(RtpSession *session, gint time);
/* this function aims at simulating senders with "imprecise" clocks, resulting in 
rtp packets sent with timestamp uncorrelated with the system clock .
This is only availlable to sessions working with the oRTP scheduler */
void rtp_session_make_time_distorsion(RtpSession *session, gint milisec);

/*RTCP functions */
void rtp_session_set_source_description(RtpSession *session, const gchar *cname,
      const gchar *name, const gchar *email, const gchar *phone, 
    const gchar *loc, const gchar *tool, const gchar *note);
void rtp_session_add_contributing_source(RtpSession *session, guint32 csrc, 
    const gchar *cname, const gchar *name, const gchar *email, const gchar *phone, 
    const gchar *loc, const gchar *tool, const gchar *note);
void rtp_session_remove_contributing_sources(RtpSession *session, guint32 csrc);
mblk_t* rtp_session_create_rtcp_sdes_packet(RtpSession *session);

/* 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,guint32 csrc);
#define rtp_set_payload_type(mp,pt) ((rtp_header_t*)((mp)->b_rptr))->paytype=(pt)

#ifdef __cplusplus


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