C++ (Cpp) rtcp_next_packet 예제들

예제 #1

파일: mediastreamer2_audio_stream_tester.c 프로젝트: JonathanRadesa/mediastreamer2

static void event_queue_cb(MediaStream *ms, void *user_pointer) {
	stats_t *st = (stats_t *)user_pointer;
	OrtpEvent *ev = NULL;

	if (st->q != NULL) {
		while ((ev = ortp_ev_queue_get(st->q)) != NULL) {
			OrtpEventType evt = ortp_event_get_type(ev);
			OrtpEventData *d = ortp_event_get_data(ev);
			if (evt == ORTP_EVENT_TMMBR_RECEIVED) {
				do {
					if (rtcp_is_RTPFB(d->packet)) {
						switch (rtcp_RTPFB_get_type(d->packet)) {
							case RTCP_RTPFB_TMMBR:
								st->number_of_TMMBR++;
								break;
							default:
								break;
						}
					}
				} while (rtcp_next_packet(d->packet));
			}
			ortp_event_destroy(ev);
		}
	}
}

예제 #2

파일: mediastreamer2_adaptive_tester.c 프로젝트: xiaolds/VideoCallVoIP

static void event_queue_cb(MediaStream *ms, void *user_pointer) {
	LossRateEstimatorCtx *ctx = (LossRateEstimatorCtx*)user_pointer;
	if (ctx->q != NULL) {
		OrtpEvent *ev = NULL;
		while ((ev = ortp_ev_queue_get(ctx->q)) != NULL) {
			OrtpEventType evt = ortp_event_get_type(ev);
			OrtpEventData *evd = ortp_event_get_data(ev);
			if (evt == ORTP_EVENT_RTCP_PACKET_RECEIVED) {
				do {
					const report_block_t *rb=NULL;
					if (rtcp_is_SR(evd->packet)){
						rb=rtcp_SR_get_report_block(evd->packet,0);
					}else if (rtcp_is_RR(evd->packet)){
						rb=rtcp_RR_get_report_block(evd->packet,0);
					}

					if (rb&&ortp_loss_rate_estimator_process_report_block(ctx->estimator,&ms->sessions.rtp_session->rtp,rb)){
						float diff = fabs(ortp_loss_rate_estimator_get_value(ctx->estimator) - ctx->loss_rate);
						CU_ASSERT_IN_RANGE(diff, 0, 10);
					}
				} while (rtcp_next_packet(evd->packet));
			}
			ortp_event_destroy(ev);
		}
	}
}

예제 #3

파일: mediastream.c 프로젝트: duh931/mediastreamer2

static void media_stream_process_rtcp(MediaStream *stream, mblk_t *m, time_t curtime){
	stream->last_packet_time=curtime;
	ms_message("%s stream [%p]: receiving RTCP %s%s",media_stream_type_str(stream),stream,(rtcp_is_SR(m)?"SR":""),(rtcp_is_RR(m)?"RR":""));
	do{
		if (stream->rc_enable && stream->rc) ms_bitrate_controller_process_rtcp(stream->rc,m);
		if (stream->qi) ms_quality_indicator_update_from_feedback(stream->qi,m);
		stream->process_rtcp(stream,m);
	}while(rtcp_next_packet(m));
}

예제 #4

파일: mediastream.c 프로젝트: cybertk/blackberry-linphone

static void parse_rtcp(mblk_t *m){
	do{
		if (rtcp_is_RR(m)){
			ms_message("Receiving RTCP RR");
		}else if (rtcp_is_SR(m)){
			ms_message("Receiving RTCP SR");
		}else if (rtcp_is_SDES(m)){
			ms_message("Receiving RTCP SDES");
			rtcp_sdes_parse(m,display_items,NULL);
		}else {
			ms_message("Receiving unhandled RTCP message");
		}
	}while(rtcp_next_packet(m));
}

예제 #5

static void video_steam_process_rtcp(VideoStream *stream, mblk_t *m){
	do{
		if (rtcp_is_SR(m)){
			const report_block_t *rb;
			ms_message("video_steam_process_rtcp: receiving RTCP SR");
			rb=rtcp_SR_get_report_block(m,0);
			if (rb){
				unsigned int ij;
				float flost;
				ij=report_block_get_interarrival_jitter(rb);
				flost=100.0*report_block_get_fraction_lost(rb)/256.0;
				ms_message("interarrival jitter=%u , lost packets percentage since last report=%f ",ij,flost);
				if (stream->adapt_bitrate) video_stream_adapt_bitrate(stream,ij,flost);
			}
		}
	}while(rtcp_next_packet(m));
}

예제 #6

파일: audiostream.c 프로젝트: LaughingAngus/linphone-vs2008

static void audio_steam_process_rtcp(AudioStream *stream, mblk_t *m){
	do{
		if (rtcp_is_SR(m)){
			const report_block_t *rb;
			//ms_message("audio_steam_process_rtcp: receiving RTCP SR");
			rb=rtcp_SR_get_report_block(m,0);
			if (rb){
				unsigned int ij;
				float flost;
				ij=report_block_get_interarrival_jitter(rb);
				flost=(float)(100.0*report_block_get_fraction_lost(rb)/256.0);
				if(stream->quality_cb!=NULL && stream->userdata!=NULL) stream->quality_cb(stream->userdata,ij,flost);
				ms_message("Audio RTCP Report: jitter=%u , lost packets percentage=%2.2f%%",ij,flost);
			}
		}
	}while(rtcp_next_packet(m));
}

예제 #7

static void video_steam_process_rtcp(VideoStream *stream, mblk_t *m){
	do{
		if (rtcp_is_SR(m)){
			const report_block_t *rb;
			ms_message("video_steam_process_rtcp: receiving RTCP SR");
			rb=rtcp_SR_get_report_block(m,0);
			if (rb){
				unsigned int ij;
				float rt=rtp_session_get_round_trip_propagation(stream->ms.session);
				float flost;
				ij=report_block_get_interarrival_jitter(rb);
				flost=(float)(100.0*report_block_get_fraction_lost(rb)/256.0);
				ms_message("video_steam_process_rtcp: interarrival jitter=%u , lost packets percentage since last report=%f, round trip time=%f seconds",ij,flost,rt);
				if (stream->ms.rc)
					ms_bitrate_controller_process_rtcp(stream->ms.rc,m);
			}
		}
	}while(rtcp_next_packet(m));
}

예제 #8

static void parse_rtcp(mblk_t *m) {
    const report_block_t *rb;
    do {
        if (rtcp_is_RR(m)) {
            ms_message("Receiving RTCP RR");
        } else if (rtcp_is_SR(m)) {
            ms_message("Receiving RTCP SR");
            rb=rtcp_SR_get_report_block(m,0);
            if (rb) {
                ms_message("interarrival jitter=%u",
                           report_block_get_interarrival_jitter(rb));
            }
        } else if (rtcp_is_SDES(m)) {
            ms_message("Receiving RTCP SDES");
            rtcp_sdes_parse(m,display_items,NULL);
        } else {
            ms_message("Receiving unhandled RTCP message");
        }
    } while(rtcp_next_packet(m));
}

예제 #9

파일: audiostream.c 프로젝트: jgabrams/linphone-mediastreamer2

static void audio_stream_process_rtcp(AudioStream *stream, mblk_t *m){
	do{
		const report_block_t *rb=NULL;
		if (rtcp_is_SR(m)){
			rb=rtcp_SR_get_report_block(m,0);
		}else if (rtcp_is_RR(m)){
			rb=rtcp_RR_get_report_block(m,0);
		}
		if (rb){
			unsigned int ij;
			float rt=rtp_session_get_round_trip_propagation(stream->ms.session);
			float flost;
			ij=report_block_get_interarrival_jitter(rb);
			flost=(float)(100.0*report_block_get_fraction_lost(rb)/256.0);
			ms_message("audio_stream_iterate(): remote statistics available\n\tremote's interarrival jitter=%u\n"
			           "\tremote's lost packets percentage since last report=%f\n\tround trip time=%f seconds",ij,flost,rt);
			if (stream->ms.rc) ms_bitrate_controller_process_rtcp(stream->ms.rc,m);
			if (stream->ms.qi) ms_quality_indicator_update_from_feedback(stream->ms.qi,m);
		}
	}while(rtcp_next_packet(m));
}

예제 #10

파일: mediastreamer2_video_stream_tester.c 프로젝트: tiena2cva/Linphone

static void event_queue_cb(MediaStream *ms, void *user_pointer) {
	video_stream_tester_stats_t *st = (video_stream_tester_stats_t *)user_pointer;
	OrtpEvent *ev = NULL;

	if (st->q != NULL) {
		while ((ev = ortp_ev_queue_get(st->q)) != NULL) {
			OrtpEventType evt = ortp_event_get_type(ev);
			OrtpEventData *d = ortp_event_get_data(ev);
			if (evt == ORTP_EVENT_RTCP_PACKET_EMITTED) {
				do {
					if (rtcp_is_RR(d->packet)) {
						st->number_of_RR++;
					} else if (rtcp_is_SR(d->packet)) {
						st->number_of_SR++;
					} else if (rtcp_is_SDES(d->packet)) {
						st->number_of_SDES++;
					} else if (rtcp_is_PSFB(d->packet)) {
						switch (rtcp_PSFB_get_type(d->packet)) {
							case RTCP_PSFB_PLI:
								st->number_of_PLI++;
								break;
							case RTCP_PSFB_SLI:
								st->number_of_SLI++;
								break;
							case RTCP_PSFB_RPSI:
								st->number_of_RPSI++;
								break;
							default:
								break;
						}
					}
				} while (rtcp_next_packet(d->packet));
			}
			ortp_event_destroy(ev);
		}
	}
}