audiostream.c

基于osip、eXosip、speex、ffmpeg的VoIP源代码

/*
mediastreamer2 library - modular sound and video processing and streaming
Copyright (C) 2006  Simon MORLAT ([e-mail])

This program is free software; you can redistribute it and/or
modify it under the terms of the GNU General Public License
as published by the Free Software Foundation; either version 2
of the License, or (at your option) any later version.

This program 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 General Public License for more details.

You should have received a copy of the GNU General Public License
along with this program; if not, write to the Free Software
Foundation, Inc., 59 Temple Place - Suite 330, Boston, MA  02111-1307, USA.
*/


#ifdef HAVE_CONFIG_H
#include "mediastreamer-config.h"
#endif

#include "mediastreamer2/mediastream.h"

#include "mediastreamer2/dtmfgen.h"
#include "mediastreamer2/mssndcard.h"
#include "mediastreamer2/msrtp.h"
#include "mediastreamer2/msfileplayer.h"
#include "mediastreamer2/msfilerec.h"

#ifdef INET6
	#include <sys/types.h>
#ifndef WIN32
	#include <sys/socket.h>
	#include <netdb.h>
#endif
#endif


#define MAX_RTP_SIZE	1500


/* this code is not part of the library itself, it is part of the mediastream program */
void audio_stream_free(AudioStream *stream)
{
	if (stream->session!=NULL) rtp_session_destroy(stream->session);
	if (stream->rtpsend!=NULL) ms_filter_destroy(stream->rtpsend);
	if (stream->rtprecv!=NULL) ms_filter_destroy(stream->rtprecv);
	if (stream->soundread!=NULL) ms_filter_destroy(stream->soundread);
	if (stream->soundwrite!=NULL) ms_filter_destroy(stream->soundwrite);
	if (stream->encoder!=NULL) ms_filter_destroy(stream->encoder);
	if (stream->decoder!=NULL) ms_filter_destroy(stream->decoder);
	if (stream->dtmfgen!=NULL) ms_filter_destroy(stream->dtmfgen);
	if (stream->ec!=NULL)	ms_filter_destroy(stream->ec);
	if (stream->ticker!=NULL) ms_ticker_destroy(stream->ticker);
	ms_free(stream);
}

static int dtmf_tab[16]={'0','1','2','3','4','5','6','7','8','9','*','#','A','B','C','D'};

static void on_dtmf_received(RtpSession *s, int dtmf, void * user_data)
{
	MSFilter *dtmfgen=(MSFilter*)user_data;
	if (dtmf>15){
		ms_warning("Unsupported telephone-event type.");
		return;
	}
	ms_message("Receiving dtmf %c.",dtmf_tab[dtmf]);
	if (dtmfgen!=NULL){
		ms_filter_call_method(dtmfgen,MS_DTMF_GEN_PUT,&dtmf_tab[dtmf]);
	}
}

#if 0

static void on_timestamp_jump(RtpSession *s,uint32_t* ts, void * user_data)
{
	ms_warning("The remote sip-phone has send data with a future timestamp: %u,"
			"resynchronising session.",*ts);
	rtp_session_reset(s);
}

#endif


bool_t ms_is_ipv6(const char *remote){
	bool_t ret=FALSE;
#ifdef INET6
	struct addrinfo hints, *res0;
	
	int err;
	memset(&hints, 0, sizeof(hints));
	hints.ai_family = PF_UNSPEC;
	hints.ai_socktype = SOCK_DGRAM;
	err = getaddrinfo(remote,"8000", &hints, &res0);
	if (err!=0) {
		ms_warning ("get_local_addr_for: %s", gai_strerror(err));
		return FALSE;
	}
	ret=(res0->ai_addr->sa_family==AF_INET6); 
	freeaddrinfo(res0);
#endif
	return ret;
}

RtpSession * create_duplex_rtpsession( int locport, bool_t ipv6){
	RtpSession *rtpr;
	rtpr=rtp_session_new(RTP_SESSION_SENDRECV);
	rtp_session_set_recv_buf_size(rtpr,MAX_RTP_SIZE);
	rtp_session_set_scheduling_mode(rtpr,0);
	rtp_session_set_blocking_mode(rtpr,0);
	rtp_session_enable_adaptive_jitter_compensation(rtpr,TRUE);
	rtp_session_set_symmetric_rtp(rtpr,TRUE);
	rtp_session_set_local_addr(rtpr,ipv6 ? "::" : "0.0.0.0",locport);
	rtp_session_signal_connect(rtpr,"timestamp_jump",(RtpCallback)rtp_session_resync,(long)NULL);
	rtp_session_signal_connect(rtpr,"ssrc_changed",(RtpCallback)rtp_session_resync,(long)NULL);
	return rtpr;
}

bool_t audio_stream_alive(AudioStream * stream, int timeout){
	RtpSession *session=stream->session;
	const rtp_stats_t *stats=rtp_session_get_stats(session);
	if (stats->recv!=0){
		if (stats->recv!=stream->last_packet_count){
			stream->last_packet_count=stats->recv;
			stream->last_packet_time=time(NULL);
		}else{
			if (time(NULL)-stream->last_packet_time>timeout){
				/* more than timeout seconds of inactivity*/
				return FALSE;
			}
		}
	}
	return TRUE;
}

/*this function must be called from the MSTicker thread:
it replaces one filter by another one.
This is a dirty hack that works anyway.
It would be interesting to have something that does the job
simplier within the MSTicker api
*/
void audio_stream_change_decoder(AudioStream *stream, int payload){
	RtpSession *session=stream->session;
	RtpProfile *prof=rtp_session_get_profile(session);
	PayloadType *pt=rtp_profile_get_payload(prof,payload);
	if (pt!=NULL){
		MSFilter *dec=ms_filter_create_decoder(pt->mime_type);
		if (dec!=NULL){
			ms_filter_unlink(stream->rtprecv, 0, stream->decoder, 0);
			ms_filter_unlink(stream->decoder,0,stream->dtmfgen,0);
			ms_filter_postprocess(stream->decoder);
			ms_filter_destroy(stream->decoder);
			stream->decoder=dec;
			if (pt->recv_fmtp!=NULL)
				ms_filter_call_method(stream->decoder,MS_FILTER_ADD_FMTP,(void*)pt->recv_fmtp);
			ms_filter_link (stream->rtprecv, 0, stream->decoder, 0);
			ms_filter_link (stream->decoder,0 , stream->dtmfgen, 0);
			ms_filter_preprocess(stream->decoder,stream->ticker);
			
		}else{
			ms_warning("No decoder found for %s",pt->mime_type);
		}
	}else{
		ms_warning("No payload defined with number %i",payload);
	}
}

static void payload_type_changed(RtpSession *session, unsigned long data){
	AudioStream *stream=(AudioStream*)data;
	int pt=rtp_session_get_recv_payload_type(stream->session);
	audio_stream_change_decoder(stream,pt);
}


int audio_stream_start_full(AudioStream *stream, RtpProfile *profile, const char *remip,int remport,
	int rem_rtcp_port, int payload,int jitt_comp, const char *infile, const char *outfile,
	MSSndCard *playcard, MSSndCard *captcard, bool_t use_ec)
{
	RtpSession *rtps=stream->session;
	PayloadType *pt;
	int tmp;	

	rtp_session_set_profile(rtps,profile);
	if (remport>0) rtp_session_set_remote_addr_full(rtps,remip,remport,rem_rtcp_port);
	rtp_session_set_payload_type(rtps,payload);
	rtp_session_set_jitter_compensation(rtps,jitt_comp);
	
	if (remport>0)
		ms_filter_call_method(stream->rtpsend,MS_RTP_SEND_SET_SESSION,rtps);
	stream->rtprecv=ms_filter_new(MS_RTP_RECV_ID);
	ms_filter_call_method(stream->rtprecv,MS_RTP_RECV_SET_SESSION,rtps);
	stream->session=rtps;
	
	stream->dtmfgen=ms_filter_new(MS_DTMF_GEN_ID);
	rtp_session_signal_connect(rtps,"telephone-event",(RtpCallback)on_dtmf_received,(unsigned long)stream->dtmfgen);
	rtp_session_signal_connect(rtps,"payload_type_changed",(RtpCallback)payload_type_changed,(unsigned long)stream);
	
	/* creates the local part */
	if (captcard!=NULL) stream->soundread=ms_snd_card_create_reader(captcard);
	else {
		stream->soundread=ms_filter_new(MS_FILE_PLAYER_ID);
		if (infile!=NULL) audio_stream_play(stream,infile);
	}
	if (playcard!=NULL) stream->soundwrite=ms_snd_card_create_writer(playcard);
	else {
		stream->soundwrite=ms_filter_new(MS_FILE_REC_ID);
		if (outfile!=NULL) audio_stream_record(stream,outfile);
	}
	
	/* creates the couple of encoder/decoder */
	pt=rtp_profile_get_payload(profile,payload);