int audio_stream_start_full(AudioStream *stream, RtpProfile *profile, const char *rem_rtp_ip,int rem_rtp_port,
	const char *rem_rtcp_ip, 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->ms.session;
	PayloadType *pt,*tel_ev;
	int tmp;
	MSConnectionHelper h;
	int sample_rate;
	MSRtpPayloadPickerContext picker_context;
	bool_t has_builtin_ec=FALSE;

	rtp_session_set_profile(rtps,profile);
	if (rem_rtp_port>0) rtp_session_set_remote_addr_full(rtps,rem_rtp_ip,rem_rtp_port,rem_rtcp_ip,rem_rtcp_port);
	if (rem_rtcp_port<=0){
		rtp_session_enable_rtcp(rtps,FALSE);
	}
	rtp_session_set_payload_type(rtps,payload);
	rtp_session_set_jitter_compensation(rtps,jitt_comp);

	if (rem_rtp_port>0)
		ms_filter_call_method(stream->ms.rtpsend,MS_RTP_SEND_SET_SESSION,rtps);
	stream->ms.rtprecv=ms_filter_new(MS_RTP_RECV_ID);
	ms_filter_call_method(stream->ms.rtprecv,MS_RTP_RECV_SET_SESSION,rtps);
	stream->ms.session=rtps;

	if((stream->features & AUDIO_STREAM_FEATURE_DTMF) != 0)
		stream->dtmfgen=ms_filter_new(MS_DTMF_GEN_ID);
	else
		stream->dtmfgen=NULL;
	rtp_session_signal_connect(rtps,"telephone-event",(RtpCallback)on_dtmf_received,(unsigned long)stream);
	rtp_session_signal_connect(rtps,"payload_type_changed",(RtpCallback)mediastream_payload_type_changed,(unsigned long)&stream->ms);
	/* creates the local part */
	if (captcard!=NULL){
		if (stream->soundread==NULL)
			stream->soundread=ms_snd_card_create_reader(captcard);
		has_builtin_ec=!!(ms_snd_card_get_capabilities(captcard) & MS_SND_CARD_CAP_BUILTIN_ECHO_CANCELLER);
	}else {
		stream->soundread=ms_filter_new(MS_FILE_PLAYER_ID);
		stream->read_resampler=ms_filter_new(MS_RESAMPLE_ID);
		if (infile!=NULL) audio_stream_play(stream,infile);
	}
	if (playcard!=NULL) {
		if (stream->soundwrite==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);
	if (pt==NULL){
		ms_error("audiostream.c: undefined payload type.");
		return -1;
	}
	tel_ev=rtp_profile_get_payload_from_mime (profile,"telephone-event");

	if ((stream->features & AUDIO_STREAM_FEATURE_DTMF_ECHO) != 0 && (tel_ev==NULL || ( (tel_ev->flags & PAYLOAD_TYPE_FLAG_CAN_RECV) && !(tel_ev->flags & PAYLOAD_TYPE_FLAG_CAN_SEND)))
	    && ( strcasecmp(pt->mime_type,"pcmu")==0 || strcasecmp(pt->mime_type,"pcma")==0)){
		/*if no telephone-event payload is usable and pcma or pcmu is used, we will generate
		  inband dtmf*/
		stream->dtmfgen_rtp=ms_filter_new (MS_DTMF_GEN_ID);
	} else {
		stream->dtmfgen_rtp=NULL;
	}
	
	if (ms_filter_call_method(stream->ms.rtpsend,MS_FILTER_GET_SAMPLE_RATE,&sample_rate)!=0){
		ms_error("Sample rate is unknown for RTP side !");
		return -1;
	}
	
	stream->ms.encoder=ms_filter_create_encoder(pt->mime_type);
	stream->ms.decoder=ms_filter_create_decoder(pt->mime_type);
	if ((stream->ms.encoder==NULL) || (stream->ms.decoder==NULL)){
		/* big problem: we have not a registered codec for this payload...*/
		ms_error("audio_stream_start_full: No decoder or encoder available for payload %s.",pt->mime_type);
		return -1;
	}
	if (ms_filter_has_method(stream->ms.decoder, MS_FILTER_SET_RTP_PAYLOAD_PICKER)) {
		ms_message(" decoder has FEC capabilities");
		picker_context.filter_graph_manager=stream;
		picker_context.picker=&audio_stream_payload_picker;
		ms_filter_call_method(stream->ms.decoder,MS_FILTER_SET_RTP_PAYLOAD_PICKER, &picker_context);
	}
	if((stream->features & AUDIO_STREAM_FEATURE_VOL_SND) != 0)
		stream->volsend=ms_filter_new(MS_VOLUME_ID);
	else
		stream->volsend=NULL;
	if((stream->features & AUDIO_STREAM_FEATURE_VOL_RCV) != 0)
		stream->volrecv=ms_filter_new(MS_VOLUME_ID);
	else
		stream->volrecv=NULL;
	audio_stream_enable_echo_limiter(stream,stream->el_type);
	audio_stream_enable_noise_gate(stream,stream->use_ng);

	if (stream->use_agc){
		int tmp=1;
		if (stream->volsend==NULL)
			stream->volsend=ms_filter_new(MS_VOLUME_ID);
		ms_filter_call_method(stream->volsend,MS_VOLUME_ENABLE_AGC,&tmp);
	}

	if (stream->dtmfgen) {
		ms_filter_call_method(stream->dtmfgen,MS_FILTER_SET_SAMPLE_RATE,&sample_rate);
		ms_filter_call_method(stream->dtmfgen,MS_FILTER_SET_NCHANNELS,&pt->channels);
	}
	if (stream->dtmfgen_rtp) {
		ms_filter_call_method(stream->dtmfgen_rtp,MS_FILTER_SET_SAMPLE_RATE,&sample_rate);
		ms_filter_call_method(stream->dtmfgen_rtp,MS_FILTER_SET_NCHANNELS,&pt->channels);
	}
	/* give the sound filters some properties */
	if (ms_filter_call_method(stream->soundread,MS_FILTER_SET_SAMPLE_RATE,&sample_rate) != 0) {
		/* need to add resampler*/
		if (stream->read_resampler == NULL) stream->read_resampler=ms_filter_new(MS_RESAMPLE_ID);
	}
	ms_filter_call_method(stream->soundread,MS_FILTER_SET_NCHANNELS,&pt->channels);

	if (ms_filter_call_method(stream->soundwrite,MS_FILTER_SET_SAMPLE_RATE,&sample_rate) != 0) {
		/* need to add resampler*/
		if (stream->write_resampler == NULL) stream->write_resampler=ms_filter_new(MS_RESAMPLE_ID);
	}
	ms_filter_call_method(stream->soundwrite,MS_FILTER_SET_NCHANNELS,&pt->channels);

	// Override feature
	if ( ((stream->features & AUDIO_STREAM_FEATURE_EC) && !use_ec) || has_builtin_ec )
		stream->features &=~AUDIO_STREAM_FEATURE_EC;

	/*configure the echo canceller if required */
	if ((stream->features & AUDIO_STREAM_FEATURE_EC) == 0 && stream->ec != NULL) {
		ms_filter_destroy(stream->ec);
		stream->ec=NULL;
	}
	if (stream->ec){
		if (!stream->is_ec_delay_set){
			int delay_ms=ms_snd_card_get_minimal_latency(captcard);
			if (delay_ms!=0){
				ms_message("Setting echo canceller delay with value provided by soundcard: %i ms",delay_ms);
				ms_filter_call_method(stream->ec,MS_ECHO_CANCELLER_SET_DELAY,&delay_ms);
			}
		}
		ms_filter_call_method(stream->ec,MS_FILTER_SET_SAMPLE_RATE,&sample_rate);
	}
	
	if (stream->features & AUDIO_STREAM_FEATURE_MIXED_RECORDING){
		int val=0;
		int pin=1;
		stream->recorder=ms_filter_new(MS_FILE_REC_ID);
		stream->recorder_mixer=ms_filter_new(MS_AUDIO_MIXER_ID);
		stream->recv_tee=ms_filter_new(MS_TEE_ID);
		stream->send_tee=ms_filter_new(MS_TEE_ID);
		ms_filter_call_method(stream->recorder_mixer,MS_AUDIO_MIXER_ENABLE_CONFERENCE_MODE,&val);
		ms_filter_call_method(stream->recorder_mixer,MS_FILTER_SET_SAMPLE_RATE,&sample_rate);
		ms_filter_call_method(stream->recorder_mixer,MS_FILTER_SET_NCHANNELS,&pt->channels);
		ms_filter_call_method(stream->recv_tee,MS_TEE_MUTE,&pin);
		ms_filter_call_method(stream->send_tee,MS_TEE_MUTE,&pin);
		ms_filter_call_method(stream->recorder,MS_FILTER_SET_SAMPLE_RATE,&sample_rate);
		ms_filter_call_method(stream->recorder,MS_FILTER_SET_NCHANNELS,&pt->channels);
		
	}

	/* give the encoder/decoder some parameters*/
	ms_filter_call_method(stream->ms.encoder,MS_FILTER_SET_SAMPLE_RATE,&sample_rate);
	ms_message("Payload's bitrate is %i",pt->normal_bitrate);
	if (pt->normal_bitrate>0){
		ms_message("Setting audio encoder network bitrate to %i",pt->normal_bitrate);
		ms_filter_call_method(stream->ms.encoder,MS_FILTER_SET_BITRATE,&pt->normal_bitrate);
	}
	ms_filter_call_method(stream->ms.encoder,MS_FILTER_SET_NCHANNELS,&pt->channels);
	ms_filter_call_method(stream->ms.decoder,MS_FILTER_SET_SAMPLE_RATE,&sample_rate);
	ms_filter_call_method(stream->ms.decoder,MS_FILTER_SET_NCHANNELS,&pt->channels);

	if (pt->send_fmtp!=NULL) {
		char value[16]={0};
		int ptime;
		if (ms_filter_has_method(stream->ms.encoder,MS_AUDIO_ENCODER_SET_PTIME)){
			if (fmtp_get_value(pt->send_fmtp,"ptime",value,sizeof(value)-1)){
				ptime=atoi(value);
				ms_filter_call_method(stream->ms.encoder,MS_AUDIO_ENCODER_SET_PTIME,&ptime);
			}
		}
		ms_filter_call_method(stream->ms.encoder,MS_FILTER_ADD_FMTP, (void*)pt->send_fmtp);
	}
	if (pt->recv_fmtp!=NULL) ms_filter_call_method(stream->ms.decoder,MS_FILTER_ADD_FMTP,(void*)pt->recv_fmtp);

	/*create the equalizer*/
	if ((stream->features & AUDIO_STREAM_FEATURE_EQUALIZER) != 0){
		stream->equalizer=ms_filter_new(MS_EQUALIZER_ID);
		if(stream->equalizer) {
			tmp=stream->eq_active;
			ms_filter_call_method(stream->equalizer,MS_EQUALIZER_SET_ACTIVE,&tmp);
		}
	}else
		stream->equalizer=NULL;
	

	/*configure resampler if needed*/
	ms_filter_call_method(stream->ms.rtpsend, MS_FILTER_SET_NCHANNELS, &pt->channels);
	ms_filter_call_method(stream->ms.rtprecv, MS_FILTER_SET_NCHANNELS, &pt->channels);
	if (stream->read_resampler){
		audio_stream_configure_resampler(stream->read_resampler,stream->soundread,stream->ms.rtpsend);
	}

	if (stream->write_resampler){
		audio_stream_configure_resampler(stream->write_resampler,stream->ms.rtprecv,stream->soundwrite);
	}

	if (stream->ms.use_rc){
		stream->ms.rc=ms_audio_bitrate_controller_new(stream->ms.session,stream->ms.encoder,0);
	}
	
	/* Create PLC */
	if ((stream->features & AUDIO_STREAM_FEATURE_PLC) != 0) {
		int decoder_have_plc = 0;
		if (ms_filter_has_method(stream->ms.decoder, MS_AUDIO_DECODER_HAVE_PLC)) {
			if (ms_filter_call_method(stream->ms.decoder, MS_AUDIO_DECODER_HAVE_PLC, &decoder_have_plc) != 0) {
				ms_warning("MS_AUDIO_DECODER_HAVE_PLC function error: enable default plc");
			}
		} else {
			ms_warning("MS_DECODER_HAVE_PLC function not implemented by the decoder: enable default plc");
		}
		if (decoder_have_plc == 0) {
			stream->plc = ms_filter_new(MS_GENERIC_PLC_ID);
		}

		if (stream->plc) {
			ms_filter_call_method(stream->plc, MS_FILTER_SET_NCHANNELS, &pt->channels);
			ms_filter_call_method(stream->plc, MS_FILTER_SET_SAMPLE_RATE, &sample_rate);
		}
	} else {
		stream->plc = NULL;
	}

	/* create ticker */
	if (stream->ms.ticker==NULL) start_ticker(&stream->ms);
	else{
		/*we were using the dummy preload graph, destroy it*/
		if (stream->dummy) stop_preload_graph(stream);
	}
	
	/* and then connect all */
	/* tip: draw yourself the picture if you don't understand */

	/*sending graph*/
	ms_connection_helper_start(&h);
	ms_connection_helper_link(&h,stream->soundread,-1,0);
	if (stream->read_resampler)
		ms_connection_helper_link(&h,stream->read_resampler,0,0);
	if (stream->ec)
		ms_connection_helper_link(&h,stream->ec,1,1);
	if (stream->volsend)
		ms_connection_helper_link(&h,stream->volsend,0,0);
	if (stream->dtmfgen_rtp)
		ms_connection_helper_link(&h,stream->dtmfgen_rtp,0,0);
	if (stream->send_tee)
		ms_connection_helper_link(&h,stream->send_tee,0,0);
	ms_connection_helper_link(&h,stream->ms.encoder,0,0);
	ms_connection_helper_link(&h,stream->ms.rtpsend,0,-1);

	/*receiving graph*/
	ms_connection_helper_start(&h);
	ms_connection_helper_link(&h,stream->ms.rtprecv,-1,0);
	ms_connection_helper_link(&h,stream->ms.decoder,0,0);
	if (stream->plc)
		ms_connection_helper_link(&h,stream->plc,0,0);
	if (stream->dtmfgen)
		ms_connection_helper_link(&h,stream->dtmfgen,0,0);
	if (stream->volrecv)
		ms_connection_helper_link(&h,stream->volrecv,0,0);
	if (stream->recv_tee)
		ms_connection_helper_link(&h,stream->recv_tee,0,0);
	if (stream->equalizer)
		ms_connection_helper_link(&h,stream->equalizer,0,0);
	if (stream->ec)
		ms_connection_helper_link(&h,stream->ec,0,0);
	if (stream->write_resampler)
		ms_connection_helper_link(&h,stream->write_resampler,0,0);
	ms_connection_helper_link(&h,stream->soundwrite,0,-1);

	/*call recording part, attached to both outgoing and incoming graphs*/
	if (stream->recorder){
		ms_filter_link(stream->send_tee,1,stream->recorder_mixer,0);
		ms_filter_link(stream->recv_tee,1,stream->recorder_mixer,1);
		ms_filter_link(stream->recorder_mixer,0,stream->recorder,0);
	}
	
	/*to make sure all preprocess are done before befre processing audio*/
	ms_ticker_attach_multiple(stream->ms.ticker
				,stream->soundread
				,stream->ms.rtprecv
				,NULL);

	stream->ms.start_time=ms_time(NULL);
	stream->ms.is_beginning=TRUE;

	return 0;
}
示例#2
0
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);
	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);
	if (pt==NULL){
		ms_error("audiostream.c: undefined payload type.");
		return -1;
	}
	stream->encoder=ms_filter_create_encoder(pt->mime_type);
	stream->decoder=ms_filter_create_decoder(pt->mime_type);
	if ((stream->encoder==NULL) || (stream->decoder==NULL)){
		/* big problem: we have not a registered codec for this payload...*/
		ms_error("mediastream.c: No decoder available for payload %i.",payload);
		return -1;
	}
	
	if (use_ec) {
		stream->ec=ms_filter_new(MS_SPEEX_EC_ID);
		ms_filter_call_method(stream->ec,MS_FILTER_SET_SAMPLE_RATE,&pt->clock_rate);
	}	

	/* give the sound filters some properties */
	ms_filter_call_method(stream->soundread,MS_FILTER_SET_SAMPLE_RATE,&pt->clock_rate);
	ms_filter_call_method(stream->soundwrite,MS_FILTER_SET_SAMPLE_RATE,&pt->clock_rate);
	tmp=1;
	ms_filter_call_method(stream->soundwrite,MS_FILTER_SET_NCHANNELS, &tmp);
	
	/* give the encoder/decoder some parameters*/
	ms_filter_call_method(stream->encoder,MS_FILTER_SET_SAMPLE_RATE,&pt->clock_rate);
	ms_message("Payload's bitrate is %i",pt->normal_bitrate);
	if (pt->normal_bitrate>0){
		ms_message("Setting audio encoder network bitrate to %i",pt->normal_bitrate);
		ms_filter_call_method(stream->encoder,MS_FILTER_SET_BITRATE,&pt->normal_bitrate);
	}
	ms_filter_call_method(stream->decoder,MS_FILTER_SET_SAMPLE_RATE,&pt->clock_rate);
	
	if (pt->send_fmtp!=NULL) ms_filter_call_method(stream->encoder,MS_FILTER_ADD_FMTP, (void*)pt->send_fmtp);
	if (pt->recv_fmtp!=NULL) ms_filter_call_method(stream->decoder,MS_FILTER_ADD_FMTP,(void*)pt->recv_fmtp);
	
	/* and then connect all */
	/* tip: draw yourself the picture if you don't understand */
	if (stream->ec){
		ms_filter_link(stream->soundread,0,stream->ec,1);
		ms_filter_link(stream->ec,1,stream->encoder,0);
		ms_filter_link(stream->dtmfgen,0,stream->ec,0);
		ms_filter_link(stream->ec,0,stream->soundwrite,0);
	}else{
		ms_filter_link(stream->soundread,0,stream->encoder,0);
		ms_filter_link(stream->dtmfgen,0,stream->soundwrite,0);
	}
	
	ms_filter_link(stream->encoder,0,stream->rtpsend,0);
	ms_filter_link(stream->rtprecv,0,stream->decoder,0);
	ms_filter_link(stream->decoder,0,stream->dtmfgen,0);
	
	/* create ticker */
	stream->ticker=ms_ticker_new();
	ms_ticker_set_name(stream->ticker,"Audio MSTicker");
	ms_ticker_attach(stream->ticker,stream->soundread);
	ms_ticker_attach(stream->ticker,stream->rtprecv);
	
	return 0;
}
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;
	MSConnectionHelper h;

	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);
	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);
		stream->read_resampler=ms_filter_new(MS_RESAMPLE_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);
	if (pt==NULL){
		ms_error("audiostream.c: undefined payload type.");
		return -1;
	}
	stream->encoder=ms_filter_create_encoder(pt->mime_type);
	stream->decoder=ms_filter_create_decoder(pt->mime_type);
	if ((stream->encoder==NULL) || (stream->decoder==NULL)){
		/* big problem: we have not a registered codec for this payload...*/
		ms_error("mediastream.c: No decoder available for payload %i.",payload);
		return -1;
	}

	if (stream->el_type!=ELInactive || stream->use_gc || stream->use_ng){
		stream->volsend=ms_filter_new(MS_VOLUME_ID);
		stream->volrecv=ms_filter_new(MS_VOLUME_ID);
		if (stream->el_type!=ELInactive){
			if (stream->el_type==ELControlFull) {
				/* also reduce speaker gain when no signal - same parameters as std. noise gate */
				int tmp=1;
				ms_filter_call_method(stream->volrecv,MS_VOLUME_ENABLE_NOISE_GATE,&tmp);
			}
			ms_filter_call_method(stream->volsend,MS_VOLUME_SET_PEER,stream->volrecv);
		}
		if (stream->use_ng){
			int tmp=1;
			ms_filter_call_method(stream->volsend,MS_VOLUME_ENABLE_NOISE_GATE,&tmp);
		}
	}

	if (stream->use_agc || stream->use_nr){
		int tmp=1;
		if (stream->volsend==NULL)
			stream->volsend=ms_filter_new(MS_VOLUME_ID);

		if(stream->use_agc) ms_filter_call_method(stream->volsend,MS_VOLUME_ENABLE_AGC,&tmp);
		/*Noise Reduction*/
		if(stream->use_nr) ms_filter_call_method(stream->volsend,MS_VOLUME_ENABLE_NR,&tmp);
	}

	/* give the sound filters some properties */
	if (ms_filter_call_method(stream->soundread,MS_FILTER_SET_SAMPLE_RATE,&pt->clock_rate) != 0) {
		/* need to add resampler*/
		if (stream->read_resampler == NULL) stream->read_resampler=ms_filter_new(MS_RESAMPLE_ID);
	}

	if (ms_filter_call_method(stream->soundwrite,MS_FILTER_SET_SAMPLE_RATE,&pt->clock_rate) != 0) {
		/* need to add resampler*/
		if (stream->write_resampler == NULL) stream->write_resampler=ms_filter_new(MS_RESAMPLE_ID);
	}

	tmp=1;
	ms_filter_call_method(stream->soundwrite,MS_FILTER_SET_NCHANNELS, &tmp);

	if(stream->record_enabled)
	{
		stream->filewriter = ms_filter_new(MS_FILE_REC_ID);
		stream->recordmixer= ms_filter_new(MS_AUDIO_MIXER_ID);
		stream->mic_tee = ms_filter_new(MS_TEE_ID);
		stream->spk_tee = ms_filter_new(MS_TEE_ID);
		ms_filter_call_method(stream->filewriter,MS_FILTER_SET_SAMPLE_RATE,&pt->clock_rate);
		ms_filter_call_method(stream->recordmixer,MS_FILTER_SET_SAMPLE_RATE,&pt->clock_rate);
		tmp=1;
		ms_filter_call_method(stream->recordmixer,MS_FILTER_SET_NCHANNELS,&tmp);
	}

	/*configure the echo canceller if required */
	if (use_ec) {
		stream->ec=ms_filter_new(MS_SPEEX_EC_ID);
		ms_filter_call_method(stream->ec,MS_FILTER_SET_SAMPLE_RATE,&pt->clock_rate);
		if (stream->ec_tail_len!=0)
			ms_filter_call_method(stream->ec,MS_ECHO_CANCELLER_SET_TAIL_LENGTH,&stream->ec_tail_len);
		if (stream->ec_delay!=0){
			ms_filter_call_method(stream->ec,MS_ECHO_CANCELLER_SET_DELAY,&stream->ec_delay);
		}else{
			/*configure from latency of sound card in case it is availlable */
			int latency=0;
			ms_filter_call_method(stream->soundread,MS_FILTER_GET_LATENCY,&latency);
			latency-=30; /*keep 30 milliseconds security margin*/
			if (latency<0) latency=0;
			ms_filter_call_method(stream->ec,MS_ECHO_CANCELLER_SET_DELAY,&latency);
		}
		if (stream->ec_framesize!=0)
			ms_filter_call_method(stream->ec,MS_ECHO_CANCELLER_SET_FRAMESIZE,&stream->ec_framesize);
	}

	/* give the encoder/decoder some parameters*/
	ms_filter_call_method(stream->encoder,MS_FILTER_SET_SAMPLE_RATE,&pt->clock_rate);
	ms_message("Payload's bitrate is %i",pt->normal_bitrate);
	if (pt->normal_bitrate>0){
		ms_message("Setting audio encoder network bitrate to %i",pt->normal_bitrate);
		ms_filter_call_method(stream->encoder,MS_FILTER_SET_BITRATE,&pt->normal_bitrate);
	}
	ms_filter_call_method(stream->decoder,MS_FILTER_SET_SAMPLE_RATE,&pt->clock_rate);

	if (pt->send_fmtp!=NULL) ms_filter_call_method(stream->encoder,MS_FILTER_ADD_FMTP, (void*)pt->send_fmtp);
	if (pt->recv_fmtp!=NULL) ms_filter_call_method(stream->decoder,MS_FILTER_ADD_FMTP,(void*)pt->recv_fmtp);

	/*create the equalizer*/
	stream->equalizer=ms_filter_new(MS_EQUALIZER_ID);
	tmp=stream->eq_active;
	ms_filter_call_method(stream->equalizer,MS_EQUALIZER_SET_ACTIVE,&tmp);
	/*configure resampler if needed*/
	if (stream->read_resampler){
		audio_stream_configure_resampler(stream->read_resampler,stream->soundread,stream->rtpsend);
	}

	if (stream->write_resampler){
		audio_stream_configure_resampler(stream->write_resampler,stream->rtprecv,stream->soundwrite);
	}
	/* and then connect all */
	/* tip: draw yourself the picture if you don't understand */

	/*sending graph*/
	ms_connection_helper_start(&h);
	ms_connection_helper_link(&h,stream->soundread,-1,0);
	if (stream->read_resampler)
		ms_connection_helper_link(&h,stream->read_resampler,0,0);
	if (stream->ec)
		ms_connection_helper_link(&h,stream->ec,1,1);
	if (stream->volsend)
		ms_connection_helper_link(&h,stream->volsend,0,0);
	if(stream->mic_tee)
		ms_connection_helper_link(&h,stream->mic_tee,0,0);
	if(stream->mic_tee && stream->recordmixer)
		ms_filter_link(stream->mic_tee,1,stream->recordmixer,0);

	ms_connection_helper_link(&h,stream->encoder,0,0);
	ms_connection_helper_link(&h,stream->rtpsend,0,-1);

	/*receiving graph*/
	ms_connection_helper_start(&h);
	ms_connection_helper_link(&h,stream->rtprecv,-1,0);
	ms_connection_helper_link(&h,stream->decoder,0,0);
	ms_connection_helper_link(&h,stream->dtmfgen,0,0);
	if (stream->equalizer)
		ms_connection_helper_link(&h,stream->equalizer,0,0);
	if (stream->volrecv)
		ms_connection_helper_link(&h,stream->volrecv,0,0);
	if (stream->ec)
		ms_connection_helper_link(&h,stream->ec,0,0);
	if (stream->write_resampler)
		ms_connection_helper_link(&h,stream->write_resampler,0,0);
	if(stream->spk_tee)
		ms_connection_helper_link(&h,stream->spk_tee,0,0);
	if(stream->mic_tee && stream->recordmixer)
		ms_filter_link(stream->spk_tee,1,stream->recordmixer,1);
	ms_connection_helper_link(&h,stream->soundwrite,0,-1);

	if (stream->filewriter && stream->spk_tee && stream->mic_tee && stream->recordmixer){
		ms_filter_link(stream->recordmixer,0,stream->filewriter,0);
	}
	/* create ticker */
	stream->ticker=ms_ticker_new();
	ms_ticker_set_name(stream->ticker,"Audio MSTicker");
	ms_ticker_attach(stream->ticker,stream->soundread);
	ms_ticker_attach(stream->ticker,stream->rtprecv);

	return 0;
}
示例#4
0
static void setup_ring_player(LinphoneCore *lc, LinphoneCall *call){
	int pause_time=3000;
	audio_stream_play(call->audiostream,lc->sound_conf.ringback_tone);
	ms_filter_call_method(call->audiostream->soundread,MS_FILE_PLAYER_LOOP,&pause_time);
}