rtpsession.c

mediastreamer2是开源的网络传输媒体流的库

}



/**
 *	Set the RTP profile to be used for the receiveing 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_recv_profile (RtpSession * session, RtpProfile * profile)
{
	session->rcv.profile = profile;
	rtp_session_recv_telephone_events_supported(session);
}

/**
 *@param session a rtp session
 *
 *	DEPRECATED! Returns current send profile.
 *	Use rtp_session_get_send_profile() or rtp_session_get_recv_profile()
 *
**/
RtpProfile *rtp_session_get_profile(RtpSession *session){
	return session->snd.profile;
}


/**
 *@param session a rtp session
 *
 *	Returns current send profile.
 *
**/
RtpProfile *rtp_session_get_send_profile(RtpSession *session){
	return session->snd.profile;
}

/**
 *@param session a rtp session
 *
 *	Returns current receive profile.
 *
**/
RtpProfile *rtp_session_get_recv_profile(RtpSession *session){
	return session->rcv.profile;
}

/**
 *	The default value is UDP_MAX_SIZE bytes, a value which is working for mostly everyone.
 *	However if your application can make assumption on the sizes of received packet,
 *	it can be interesting to set it to a lower value in order to save memory.
 *
 * @param session a rtp session
 * @param bufsize max size in bytes for receiving packets
**/
void rtp_session_set_recv_buf_size(RtpSession *session, int bufsize){
	session->recv_buf_size=bufsize;
}

/**
 *	Set kernel send maximum buffer size for the rtp socket.
 *	A value of zero defaults to the operating system default.
**/
void rtp_session_set_rtp_socket_send_buffer_size(RtpSession * session, unsigned int size){
	session->rtp.snd_socket_size=size;
}

/**
 *	Set kernel recv maximum buffer size for the rtp socket.
 *	A value of zero defaults to the operating system default.
**/
void rtp_session_set_rtp_socket_recv_buffer_size(RtpSession * session, unsigned int size){
	session->rtp.rcv_socket_size=size;
}

/**
 *	This function provides the way for an application to be informed of various events that
 *	may occur during a rtp session. @signal is a string identifying the event, and @cb is 
 *	a user supplied function in charge of processing it. The application can register
 *	several callbacks for the same signal, in the limit of #RTP_CALLBACK_TABLE_MAX_ENTRIES.
 *	Here are name and meaning of supported signals types:
 *
 *	"ssrc_changed" : the SSRC of the incoming stream has changed.
 *
 *	"payload_type_changed" : the payload type of the incoming stream has changed.
 *
 *	"telephone-event_packet" : a telephone-event rtp packet (RFC2833) is received.
 *
 *	"telephone-event" : a telephone event has occured. This is a high-level shortcut for "telephone-event_packet".
 *
 *	"network_error" : a network error happened on a socket. Arguments of the callback functions are
 *						a const char * explaining the error, an int errno error code and the user_data as usual.
 *
 *	"timestamp_jump" : we have received a packet with timestamp in far future compared to last timestamp received.
 *						The farness of far future is set by rtp_sesssion_set_time_jump_limit()
 *  "rtcp_bye": we have received a RTCP bye packet. Arguments of the callback
 *              functions are a const char * containing the leaving reason and
 *              the user_data.
 * 
 *	Returns: 0 on success, -EOPNOTSUPP if the signal does not exists, -1 if no more callbacks
 *	can be assigned to the signal type.
 *
 * @param session 	a rtp session
 * @param signal_name	the name of a signal
 * @param cb		a RtpCallback
 * @param user_data	a pointer to any data to be passed when invoking the callback.
 *
**/
int
rtp_session_signal_connect (RtpSession * session, const char *signal_name,
			    RtpCallback cb, unsigned long user_data)
{
	OList *elem;
	for (elem=session->signal_tables;elem!=NULL;elem=o_list_next(elem)){
		RtpSignalTable *s=(RtpSignalTable*) elem->data;
		if (strcmp(signal_name,s->signal_name)==0){
			return rtp_signal_table_add(s,cb,user_data);
		}
	}
	ortp_warning ("rtp_session_signal_connect: inexistant signal %s",signal_name);
	return -1;
}


/**
 *	Removes callback function @cb to the list of callbacks for signal @signal.
 *
 * @param session a rtp session
 * @param signal_name	a signal name
 * @param cb	a callback function.
 * @return: 0 on success, a negative value if the callback was not found.
**/
int
rtp_session_signal_disconnect_by_callback (RtpSession * session, const char *signal_name,
					   RtpCallback cb)
{
	OList *elem;
	for (elem=session->signal_tables;elem!=NULL;elem=o_list_next(elem)){
		RtpSignalTable *s=(RtpSignalTable*) elem->data;
		if (strcmp(signal_name,s->signal_name)==0){
			return rtp_signal_table_remove_by_callback(s,cb);
		}
	}
	ortp_warning ("rtp_session_signal_connect: inexistant signal %s",signal_name);
	return -1;
}


/**
 * sets the initial sequence number of a sending session.
 * @param session		a rtp session freshly created.
 * @param addr			a 16 bit unsigned number.
 *
**/
void rtp_session_set_seq_number(RtpSession *session, uint16_t seq){
	session->rtp.snd_seq=seq;
}


uint16_t rtp_session_get_seq_number(RtpSession *session){
	return session->rtp.snd_seq;
}


/**
 *	Sets the SSRC for the outgoing stream.
 *  If not done, a random ssrc is used.
 *
 * @param session a rtp session.
 * @param ssrc an unsigned 32bit integer representing the synchronisation source identifier (SSRC).
**/
void
rtp_session_set_ssrc (RtpSession * session, uint32_t ssrc)
{
	session->snd.ssrc = ssrc;
}


void rtp_session_update_payload_type(RtpSession *session, int paytype){
	/* check if we support this payload type */
	PayloadType *pt=rtp_profile_get_payload(session->rcv.profile,paytype);
	session->hw_recv_pt=paytype;
	if (pt!=0){
		ortp_message ("payload type changed to %i(%s) !",
				 paytype,pt->mime_type);
		payload_type_changed(session,pt);
	}else{
		ortp_warning("Receiving packet with unknown payload type %i.",paytype);
	}
}
/**
 *	Sets the payload type of the rtp session. It decides of the payload types written in the
 *	of the rtp header for the outgoing stream, if the session is SENDRECV or SENDONLY.
 *	For payload type in incoming packets, the application can be informed by registering
 *	for the "payload_type_changed" signal, so that it can make the necessary changes
 *	on the downstream decoder that deals with the payload of the packets.
 *
 * @param session a rtp session
 * @param paytype the payload type number
 * @return 0 on success, -1 if the payload is not defined.
**/

int
rtp_session_set_send_payload_type (RtpSession * session, int paytype)
{
	session->snd.pt=paytype;
	return 0;
}

/**
 *@param session a rtp session
 *
 *@return the payload type currently used in outgoing rtp packets
**/
int rtp_session_get_send_payload_type(const RtpSession *session){
	return session->snd.pt;
}

/**
 *
 *	Sets the expected payload type for incoming packets.
 *	If the actual payload type in incoming packets is different that this expected payload type, thus
 *	the "payload_type_changed" signal is emitted.
 *
 *@param session a rtp session
 *@param paytype the payload type number
 *@return 0 on success, -1 if the payload is not defined.
**/

int
rtp_session_set_recv_payload_type (RtpSession * session, int paytype)
{
	PayloadType *pt;
	session->rcv.pt=paytype;
	session->hw_recv_pt=paytype;
	pt=rtp_profile_get_payload(session->rcv.profile,paytype);
	if (pt!=NULL){
		payload_type_changed(session,pt);
	}
	return 0;
}

/**
 *@param session a rtp session
 *
 * @return the payload type currently used in incoming rtp packets
**/
int rtp_session_get_recv_payload_type(const RtpSession *session){
	return session->rcv.pt;
}

/**
 *	Sets the expected payload type for incoming packets and payload type to be used for outgoing packets.
 *	If the actual payload type in incoming packets is different that this expected payload type, thus
 *	the "payload_type_changed" signal is emitted.
 *
 * @param session a rtp session
 * @param paytype the payload type number
 * @return 0 on success, -1 if the payload is not defined.
**/
int rtp_session_set_payload_type(RtpSession *session, int pt){
	if (rtp_session_set_send_payload_type(session,pt)<0) return -1;
	if (rtp_session_set_recv_payload_type(session,pt)<0) return -1;
	return 0;
}


static void rtp_header_init_from_session(rtp_header_t *rtp, RtpSession *session){
	rtp->version = 2;
	rtp->padbit = 0;
	rtp->extbit = 0;
	rtp->markbit= 0;
	rtp->cc = 0;
	rtp->paytype = session->snd.pt;
	rtp->ssrc = session->snd.ssrc;
	rtp->timestamp = 0;	/* set later, when packet is sended */
	/* set a seq number */
	rtp->seq_number=session->rtp.snd_seq;
}

/**
 *	Allocates a new rtp packet. In the header, ssrc and payload_type according to the session's
 *	context. Timestamp and seq number are not set, there will be set when the packet is going to be
 *	sent with rtp_session_sendm_with_ts().
 *	If payload_size is zero, thus an empty packet (just a RTP header) is returned.
 *
 *@param session a rtp session.
 *@param header_size the rtp header size. For standart size (without extensions), it is RTP_FIXED_HEADER_SIZE
 *@param payload data to be copied into the rtp packet.
 *@param payload_size size of data carried by the rtp packet.
 *@return a rtp packet in a mblk_t (message block) structure.
**/
mblk_t * rtp_session_create_packet(RtpSession *session,int header_size, const uint8_t *payload, int payload_size)
{
	mblk_t *mp;
	int msglen=header_size+payload_size;
	rtp_header_t *rtp;
	
	mp=allocb(msglen,BPRI_MED);
	rtp=(rtp_header_t*)mp->b_rptr;
	rtp_header_init_from_session(rtp,session);
	/*copy the payload, if any */
	mp->b_wptr+=header_size;
	if (payload_size){
		memcpy(mp->b_wptr,payload,payload_size);
		mp->b_wptr+=payload_size;
	}
	return mp;
}

/**
 *	Creates a new rtp packet using the given payload buffer (no copy). The header will be allocated separetely.
 *  In the header, ssrc and payload_type according to the session's
 *	context. Timestamp and seq number are not set, there will be set when the packet is going to be
 *	sent with rtp_session_sendm_with_ts().
 *	oRTP will send this packet using libc's sendmsg() (if this function is availlable!) so that there will be no
 *	packet concatenation involving copies to be done in user-space.
 *  @freefn can be NULL, in that case payload will be kept untouched.
 *
 * @param session a rtp session.
 * @param payload the data to be sent with this packet
 * @param payload_size size of data
 * @param freefn a function that will be called when the payload buffer is no more needed.
 * @return: a rtp packet in a mblk_t (message block) structure.
**/

mblk_t * rtp_session_create_packet_with_data(RtpSession *session, uint8_t *payload, int payload_size, void (*freefn)(void*))
{
	mblk_t *mp,*mpayload;
	int header_size=RTP_FIXED_HEADER_SIZE; /* revisit when support for csrc is done */
	rtp_header_t *rtp;
	
	mp=allocb(header_size,BPRI_MED);
	rtp=(rtp_header_t*)mp->b_rptr;
	rtp_header_init_from_session(rtp,session);
	mp->b_wptr+=header_size;
	/* create a mblk_t around the user supplied payload buffer */
	mpayload=esballoc(payload,payload_size,BPRI_MED,freefn);
	mpayload->b_wptr+=payload_size;
	/* link it with the header */
	mp->b_cont=mpayload;
	return mp;
}


/**
 * Creates a new rtp packet using the buffer given in arguments (no copy). 
 * In the header, ssrc and payload_type according to the session's
 *context. Timestamp and seq number are not set, there will be set when the packet is going to be
 *	sent with rtp_session_sendm_with_ts().
 *  @freefn can be NULL, in that case payload will be kept untouched.
 *
 * @param session a rtp session.
 * @param buffer a buffer that contains first just enough place to write a RTP header, then the data to send.
 * @param size the size of the buffer
 * @param freefn a function that will be called once the buffer is no more needed (the data has been sent).
 * @return a rtp packet in a mblk_t (message block) structure.
**/
mblk_t * rtp_session_create_packet_in_place(RtpSession *session,uint8_t *buffer, int size, void (*freefn)(void*) )
{
	mblk_t *mp;
	rtp_header_t *rtp;
	
	mp=esballoc(buffer,size,BPRI_MED,freefn);

	rtp=(rtp_header_t*)mp->b_rptr;
	rtp_header_init_from_session(rtp,session);
	return mp;
}


int
__rtp_session_sendm_with_ts (RtpSession * session, mblk_t *mp, uint32_t packet_ts, uint32_t send_ts)
{
	rtp_header_t *rtp;
	uint32_t packet_time;
	int error = 0;
	int packsize;
	RtpScheduler *sched=session->sched;
	RtpStream *stream=&session->rtp;

	if (session->flags & RTP_SESSION_SEND_NOT_STARTED)
	{
		session->rtp.snd_ts_offset = send_ts;
		/* Set initial last_rcv_time to first send time. */
		if ((session->flags & RTP_SESSION_RECV_NOT_STARTED)
		|| session->mode == RTP_SESSION_SENDONLY)
		{
		gettimeofday(&session->last_recv_time, NULL);
		}
		if (session->flags & RTP_SESSION_SCHEDULED)
		{
			session->rtp.snd_time_offset = sched->time_;
		}
		rtp_session_unset_flag (session,RTP_SESSION_SEND_NOT_STARTED);