コード例 #1
0
void call_destroy(Call *obj)
{
	obj->root->calls=ms_list_remove(obj->root->calls,obj);
	rtp_profile_destroy(obj->profile);
	sdp_context_free(obj->sdpc);
	ms_free(obj);
}
コード例 #2
0
static void remove_local_endpoint(LinphoneConference *ctx){
	if (ctx->local_endpoint){
		ms_audio_conference_remove_member(ctx->conf,ctx->local_endpoint);
		ms_audio_endpoint_release_from_stream(ctx->local_endpoint);
		ctx->local_endpoint=NULL;
		audio_stream_stop(ctx->local_participant);
		ctx->local_participant=NULL;
		rtp_profile_destroy(ctx->local_dummy_profile);
	}
}
コード例 #3
0
ファイル: linphonecall.c プロジェクト: ApOgEE/linphone-sdk
void linphone_call_stop_media_streams(LinphoneCall *call){
	if (call->audiostream!=NULL) {
		rtp_session_unregister_event_queue(call->audiostream->session,call->audiostream_app_evq);
		ortp_ev_queue_flush(call->audiostream_app_evq);
		ortp_ev_queue_destroy(call->audiostream_app_evq);

		if (call->audiostream->ec){
			const char *state_str=NULL;
			ms_filter_call_method(call->audiostream->ec,MS_ECHO_CANCELLER_GET_STATE_STRING,&state_str);
			if (state_str){
				ms_message("Writing echo canceller state, %i bytes",strlen(state_str));
				lp_config_set_string(call->core->config,"sound","ec_state",state_str);
			}
		}
		linphone_call_log_fill_stats (call->log,call->audiostream);
		audio_stream_stop(call->audiostream);
		call->audiostream=NULL;
	}


#ifdef VIDEO_ENABLED
	if (call->videostream!=NULL){
		rtp_session_unregister_event_queue(call->videostream->session,call->videostream_app_evq);
		ortp_ev_queue_flush(call->videostream_app_evq);
		ortp_ev_queue_destroy(call->videostream_app_evq);
		video_stream_stop(call->videostream);
		call->videostream=NULL;
	}
	ms_event_queue_skip(call->core->msevq);
	
#endif
	if (call->audio_profile){
		rtp_profile_clear_all(call->audio_profile);
		rtp_profile_destroy(call->audio_profile);
		call->audio_profile=NULL;
	}
	if (call->video_profile){
		rtp_profile_clear_all(call->video_profile);
		rtp_profile_destroy(call->video_profile);
		call->video_profile=NULL;
	}
}
コード例 #4
0
ファイル: pcap_playback.c プロジェクト: biddyweb/azfone-ios
static void clear_mediastreams(MediastreamDatas *args)
{
	MSConnectionHelper h;

	ms_message("stopping all...\n");

	if (args->read) {
		ms_ticker_detach(args->ticker, args->read);
		ms_connection_helper_start(&h);
		ms_connection_helper_unlink(&h, args->read, -1, 0);
		ms_connection_helper_unlink(&h, args->decoder, 0, 0);
		ms_connection_helper_unlink(&h, args->write, 0, -1);
	}
	rtp_profile_destroy(args->profile);

	ms_exit();
}
コード例 #5
0
Call * call_new(Sipomatic *root, eXosip_event_t *ev)
{
	Call *obj;
	char *sdpans;
	int status;
	sdp_message_t *sdp;
	sdp_context_t *sdpc;
	
	sdp=eXosip_get_sdp_info(ev->request);
	sdpc=sdp_handler_create_context(&sipomatic_sdp_handler,NULL,"sipomatic");
	obj=ms_new0(Call,1);
	obj->profile=rtp_profile_new("remote");
	eXosip_call_send_answer(ev->tid,100,NULL);
	sdp_context_set_user_pointer(sdpc,obj);
	sdpans=sdp_context_get_answer(sdpc,sdp);
	if (sdpans!=NULL){
		eXosip_call_send_answer(ev->tid,180,NULL);
		
	}else{
		status=sdp_context_get_status(sdpc);
		eXosip_call_send_answer(ev->tid,status,NULL);
		sdp_context_free(sdpc);
		rtp_profile_destroy(obj->profile);
		ms_free(obj);
		return NULL;
	}
	obj->sdpc=sdpc;
	obj->did=ev->did;
	obj->tid=ev->tid;
	obj->time=time(NULL);
	obj->audio_stream=NULL;
	obj->state=CALL_STATE_INIT;
	obj->eof=0;
	obj->root=root;
	root->calls=ms_list_append(root->calls,obj);
	return obj;
}
コード例 #6
0
static void encrypted_audio_stream_base( bool_t change_ssrc,
										 bool_t change_send_key_in_the_middle
										,bool_t set_both_send_recv_key
										,bool_t send_key_first) {
	AudioStream * 	marielle = audio_stream_new (MARIELLE_RTP_PORT, MARIELLE_RTCP_PORT,FALSE);
	AudioStream * 	margaux = audio_stream_new (MARGAUX_RTP_PORT,MARGAUX_RTCP_PORT, FALSE);
	RtpProfile* profile = rtp_profile_new("default profile");
	char* hello_file = ms_strdup_printf("%s/%s", mediastreamer2_tester_get_file_root(), HELLO_8K_1S_FILE);
	char* recorded_file = ms_strdup_printf("%s/%s", mediastreamer2_tester_get_writable_dir(), RECORDED_8K_1S_FILE);
	stats_t marielle_stats;
	stats_t margaux_stats;
	int dummy=0;

	if (ms_srtp_supported()) {
		reset_stats(&marielle_stats);
		reset_stats(&margaux_stats);

		rtp_profile_set_payload (profile,0,&payload_type_pcmu8000);

		CU_ASSERT_EQUAL(audio_stream_start_full(margaux
				, profile
				, MARIELLE_IP
				, MARIELLE_RTP_PORT
				, MARIELLE_IP
				, MARIELLE_RTCP_PORT
				, 0
				, 50
				, NULL
				, recorded_file
				, NULL
				, NULL
				, 0),0);

		CU_ASSERT_EQUAL(audio_stream_start_full(marielle
				, profile
				, MARGAUX_IP
				, MARGAUX_RTP_PORT
				, MARGAUX_IP
				, MARGAUX_RTCP_PORT
				, 0
				, 50
				, hello_file
				, NULL
				, NULL
				, NULL
				, 0),0);

		if (send_key_first) {
			CU_ASSERT_TRUE(media_stream_set_srtp_send_key_b64(&(marielle->ms.sessions), MS_AES_128_SHA1_32, "d0RmdmcmVCspeEc3QGZiNWpVLFJhQX1cfHAwJSoj") == 0);
			if (set_both_send_recv_key)
				CU_ASSERT_TRUE(media_stream_set_srtp_send_key_b64(&(margaux->ms.sessions), MS_AES_128_SHA1_32, "6jCLmtRkVW9E/BUuJtYj/R2z6+4iEe06/DWohQ9F") == 0);

			CU_ASSERT_TRUE(media_stream_set_srtp_recv_key_b64(&(margaux->ms.sessions), MS_AES_128_SHA1_32, "d0RmdmcmVCspeEc3QGZiNWpVLFJhQX1cfHAwJSoj") ==0);
			if (set_both_send_recv_key)
				CU_ASSERT_TRUE(media_stream_set_srtp_recv_key_b64(&(marielle->ms.sessions), MS_AES_128_SHA1_32, "6jCLmtRkVW9E/BUuJtYj/R2z6+4iEe06/DWohQ9F") ==0);

		} else {
			CU_ASSERT_TRUE(media_stream_set_srtp_recv_key_b64(&(margaux->ms.sessions), MS_AES_128_SHA1_32, "d0RmdmcmVCspeEc3QGZiNWpVLFJhQX1cfHAwJSoj") ==0);
			if (set_both_send_recv_key)
				CU_ASSERT_TRUE(media_stream_set_srtp_recv_key_b64(&(marielle->ms.sessions), MS_AES_128_SHA1_32, "6jCLmtRkVW9E/BUuJtYj/R2z6+4iEe06/DWohQ9F") ==0);

			CU_ASSERT_TRUE(media_stream_set_srtp_send_key_b64(&(marielle->ms.sessions), MS_AES_128_SHA1_32, "d0RmdmcmVCspeEc3QGZiNWpVLFJhQX1cfHAwJSoj") == 0);
			if (set_both_send_recv_key)
				CU_ASSERT_TRUE(media_stream_set_srtp_send_key_b64(&(margaux->ms.sessions), MS_AES_128_SHA1_32, "6jCLmtRkVW9E/BUuJtYj/R2z6+4iEe06/DWohQ9F") == 0);

		}

		ms_filter_add_notify_callback(marielle->soundread, notify_cb, &marielle_stats,TRUE);
		if (change_send_key_in_the_middle) {
			int dummy=0;
			wait_for_until(&marielle->ms,&margaux->ms,&dummy,1,2000);
			CU_ASSERT_TRUE(media_stream_set_srtp_send_key_b64(&(marielle->ms.sessions), MS_AES_128_SHA1_32, "eCYF4nYyCvmCpFWjUeDaxI2GWp2BzCRlIPfg52Te") == 0);
			CU_ASSERT_TRUE(media_stream_set_srtp_recv_key_b64(&(margaux->ms.sessions), MS_AES_128_SHA1_32, "eCYF4nYyCvmCpFWjUeDaxI2GWp2BzCRlIPfg52Te") ==0);
		}
		CU_ASSERT_TRUE(wait_for_until(&marielle->ms,&margaux->ms,&marielle_stats.number_of_EndOfFile,1,12000));

		/*make sure packets can cross from sender to receiver*/
		wait_for_until(&marielle->ms,&margaux->ms,&dummy,1,500);

		audio_stream_get_local_rtp_stats(marielle,&marielle_stats.rtp);
		audio_stream_get_local_rtp_stats(margaux,&margaux_stats.rtp);

		/* No packet loss is assumed */
		if (change_send_key_in_the_middle) {
			/*we can accept one or 2 error in such case*/
			CU_ASSERT_TRUE((marielle_stats.rtp.packet_sent-margaux_stats.rtp.packet_recv)<3);
		} else
			CU_ASSERT_EQUAL(marielle_stats.rtp.sent,margaux_stats.rtp.recv);

		if (change_ssrc) {
			audio_stream_stop(marielle);
			marielle = audio_stream_new (MARIELLE_RTP_PORT, MARIELLE_RTCP_PORT,FALSE);
			CU_ASSERT_EQUAL(audio_stream_start_full(marielle
							, profile
							, MARGAUX_IP
							, MARGAUX_RTP_PORT
							, MARGAUX_IP
							, MARGAUX_RTCP_PORT
							, 0
							, 50
							, hello_file
							, NULL
							, NULL
							, NULL
							, 0),0);
			CU_ASSERT_FATAL(media_stream_set_srtp_send_key_b64(&(marielle->ms.sessions), MS_AES_128_SHA1_32, "d0RmdmcmVCspeEc3QGZiNWpVLFJhQX1cfHAwJSoj") == 0);

			ms_filter_add_notify_callback(marielle->soundread, notify_cb, &marielle_stats,TRUE);

			CU_ASSERT_TRUE(wait_for_until(&marielle->ms,&margaux->ms,&marielle_stats.number_of_EndOfFile,2,12000));

			/*make sure packets can cross from sender to receiver*/
			wait_for_until(&marielle->ms,&margaux->ms,&dummy,1,500);

			audio_stream_get_local_rtp_stats(marielle,&marielle_stats.rtp);
			audio_stream_get_local_rtp_stats(margaux,&margaux_stats.rtp);

			/* No packet loss is assumed */
			CU_ASSERT_EQUAL(marielle_stats.rtp.sent*2,margaux_stats.rtp.recv);

		}

		unlink(recorded_file);
		ms_free(recorded_file);
		ms_free(hello_file);
	} else {
		ms_warning("srtp not available, skiping...");
	}
	audio_stream_stop(marielle);
	audio_stream_stop(margaux);
	rtp_profile_destroy(profile);

}
コード例 #7
0
static void run_media_streams(int localport, const char *remote_ip, int remoteport, int payload, const char *fmtp,
          int jitter, int bitrate, MSVideoSize vs, bool_t ec, bool_t agc, bool_t eq)
{
	AudioStream *audio=NULL;
#ifdef VIDEO_ENABLED
	VideoStream *video=NULL;
#endif
	RtpSession *session=NULL;
	PayloadType *pt;
	RtpProfile *profile=rtp_profile_clone_full(&av_profile);
	OrtpEvQueue *q=ortp_ev_queue_new();	

	ms_init();
	signal(SIGINT,stop_handler);
	pt=rtp_profile_get_payload(profile,payload);
	if (pt==NULL){
		printf("Error: no payload defined with number %i.",payload);
		exit(-1);
	}
	if (fmtp!=NULL) payload_type_set_send_fmtp(pt,fmtp);
	if (bitrate>0) pt->normal_bitrate=bitrate;

	if (pt->type!=PAYLOAD_VIDEO){
		MSSndCardManager *manager=ms_snd_card_manager_get();
		MSSndCard *capt= capture_card==NULL ? ms_snd_card_manager_get_default_capture_card(manager) :
				ms_snd_card_manager_get_card(manager,capture_card);
		MSSndCard *play= playback_card==NULL ? ms_snd_card_manager_get_default_playback_card(manager) :
				ms_snd_card_manager_get_card(manager,playback_card);
		audio=audio_stream_new(localport,ms_is_ipv6(remote_ip));
		audio_stream_enable_automatic_gain_control(audio,agc);
		audio_stream_enable_noise_gate(audio,use_ng);
		audio_stream_set_echo_canceller_params(audio,ec_len_ms,ec_delay_ms,ec_framesize);
		printf("Starting audio stream.\n");
	
		audio_stream_start_full(audio,profile,remote_ip,remoteport,remoteport+1, payload, jitter,infile,outfile,
		                        outfile==NULL ? play : NULL ,infile==NULL ? capt : NULL,infile!=NULL ? FALSE: ec);
		
		if (audio) {
			if (use_ng && ng_threshold!=-1)
				ms_filter_call_method(audio->volsend,MS_VOLUME_SET_NOISE_GATE_THRESHOLD,&ng_threshold);
			session=audio->session;
		}
	}else{
#ifdef VIDEO_ENABLED
		if (eq){
			ms_fatal("Cannot put an audio equalizer in a video stream !");
			exit(-1);
		}
		printf("Starting video stream.\n");
		video=video_stream_new(localport, ms_is_ipv6(remote_ip));
		video_stream_set_sent_video_size(video,vs);
		video_stream_use_preview_video_window(video,two_windows);
		video_stream_start(video,profile,
					remote_ip,
					remoteport,remoteport+1,
					payload,
					jitter,
					ms_web_cam_manager_get_default_cam(ms_web_cam_manager_get()));
		session=video->session;
#else
		printf("Error: video support not compiled.\n");
#endif
	}
  if (eq || ec){ /*read from stdin interactive commands */
    char commands[128];
    commands[127]='\0';
    ms_sleep(1);  /* ensure following text be printed after ortp messages */
    if (eq)
      printf("\nPlease enter equalizer requests, such as 'eq active 1', 'eq active 0', 'eq 1200 0.1 200'\n");
    if (ec)
      printf("\nPlease enter echo canceller requests: ec reset; ec <delay ms> <tail_length ms'\n");
    while(fgets(commands,sizeof(commands)-1,stdin)!=NULL){
      int active,freq,freq_width;
      int delay_ms, tail_ms;
      float gain;
      if (sscanf(commands,"eq active %i",&active)==1){
        audio_stream_enable_equalizer(audio,active);
        printf("OK\n");
      }else if (sscanf(commands,"eq %i %f %i",&freq,&gain,&freq_width)==3){
        audio_stream_equalizer_set_gain(audio,freq,gain,freq_width);
        printf("OK\n");
      }else if (sscanf(commands,"eq %i %f",&freq,&gain)==2){
        audio_stream_equalizer_set_gain(audio,freq,gain,0);
        printf("OK\n");
      }else if (strstr(commands,"dump")){
        int n=0,i;
        float *t;
        ms_filter_call_method(audio->equalizer,MS_EQUALIZER_GET_NUM_FREQUENCIES,&n);
        t=(float*)alloca(sizeof(float)*n);
        ms_filter_call_method(audio->equalizer,MS_EQUALIZER_DUMP_STATE,t);
        for(i=0;i<n;++i){
          if (fabs(t[i]-1)>0.01){
            printf("%i:%f:0 ",(i*pt->clock_rate)/(2*n),t[i]);
          }
        }
        printf("\nOK\n");
      }else if (sscanf(commands,"ec reset %i",&active)==1){
          //audio_stream_enable_equalizer(audio,active);
          //printf("OK\n");
      }else if (sscanf(commands,"ec active %i",&active)==1){
          //audio_stream_enable_equalizer(audio,active);
          //printf("OK\n");
      }else if (sscanf(commands,"ec %i %i",&delay_ms,&tail_ms)==2){
        audio_stream_set_echo_canceller_params(audio,tail_ms,delay_ms,128);
        // revisit: workaround with old method call to force echo reset
        delay_ms*=8;
        ms_filter_call_method(audio->ec,MS_FILTER_SET_PLAYBACKDELAY,&delay_ms);
        printf("OK\n");
      }else if (strstr(commands,"quit")){
        break;
      }else printf("Cannot understand this.\n");
    }
	}else{  /* no interactive stuff - continuous debug output */
		rtp_session_register_event_queue(session,q);
		while(cond)
		{
			int n;
			for(n=0;n<100;++n){
	#ifdef WIN32
				MSG msg;
				Sleep(10);
				while (PeekMessage(&msg, NULL, 0, 0,1)){
					TranslateMessage(&msg);
					DispatchMessage(&msg);
				}
	#else
				struct timespec ts;
				ts.tv_sec=0;
				ts.tv_nsec=10000000;
				nanosleep(&ts,NULL);
	#endif
	#if defined(VIDEO_ENABLED)
				if (video) video_stream_iterate(video);
	#endif
			}
			ortp_global_stats_display();
			if (session){
				printf("Bandwidth usage: download=%f kbits/sec, upload=%f kbits/sec\n",
					rtp_session_compute_recv_bandwidth(session)*1e-3,
					rtp_session_compute_send_bandwidth(session)*1e-3);
				parse_events(q);
			}
		}
					}
	
	printf("stopping all...\n");
	
	if (audio) audio_stream_stop(audio);
#ifdef VIDEO_ENABLED
	if (video) video_stream_stop(video);
#endif
	ortp_ev_queue_destroy(q);
	rtp_profile_destroy(profile);
}
コード例 #8
0
void run_media_streams(int localport,  const char *remote_ip, int remoteport, int payload, const char *fmtp, int jitter, bool_t ec, int bitrate)
{
    AudioStream *audio=NULL;
#ifdef VIDEO_ENABLED
    VideoStream *video=NULL;
#endif
    RtpSession *session=NULL;
    PayloadType *pt;
    RtpProfile *profile=rtp_profile_clone_full(&av_profile);
    OrtpEvQueue *q=ortp_ev_queue_new();

    ms_init();
    signal(SIGINT,stop_handler);
    pt=rtp_profile_get_payload(profile,payload);
    if (pt==NULL) {
        printf("Error: no payload defined with number %i.",payload);
        exit(-1);
    }
    if (fmtp!=NULL) payload_type_set_send_fmtp(pt,fmtp);
    if (bitrate>0) pt->normal_bitrate=bitrate;

    if (pt->type!=PAYLOAD_VIDEO) {
        printf("Starting audio stream.\n");
        audio=audio_stream_start(profile,localport,remote_ip,remoteport,payload,jitter, ec);
        if (audio) session=audio->session;
    } else {
#ifdef VIDEO_ENABLED
        printf("Starting video stream.\n");
        video=video_stream_new(localport, ms_is_ipv6(remote_ip));
        video_stream_start(video,profile,
                           remote_ip,
                           remoteport,
                           payload,
                           jitter,
                           "/dev/video0");
        session=video->session;
#else
        printf("Error: video support not compiled.\n");
#endif
    }
    rtp_session_register_event_queue(session,q);
    while(cond)
    {
        /* sleep until we receive SIGINT */
#ifdef WIN32
        int n;
        MSG msg;
        for(n=0; n<100; ++n) {
            Sleep(10);
            while (PeekMessage(&msg, NULL, 0, 0,1)) {
                TranslateMessage(&msg);
                DispatchMessage(&msg);
            }
        }
#else
        sleep(1);
#endif
        ortp_global_stats_display();
        if (session) {
            printf("Bandwidth usage: download=%f kbits/sec, upload=%f kbits/sec\n",
                   rtp_session_compute_recv_bandwidth(session)*1e-3,
                   rtp_session_compute_send_bandwidth(session)*1e-3);
            parse_events(q);
        }
    }

    printf("stoping all...\n");

    if (audio) audio_stream_stop(audio);
#ifdef VIDEO_ENABLED
    if (video) video_stream_stop(video);
#endif
    ortp_ev_queue_destroy(q);
    rtp_profile_destroy(profile);
}
コード例 #9
0
ファイル: mkvstream.c プロジェクト: Accontech/mediastreamer2
/*
 * This small program starts a video stream to either
 * - read an H264 video track from mkv file and stream it out with RTP to specified destination
 * - receive H264 RTP packets on a local port and record them into an mkv file
 */
int main(int argc, char *argv[]){
	const char *command;
	const char *file;
	const char *ip;
	int port;
	VideoStream *stream;
	RtpProfile *profile;
	PayloadType *pt;
	Mode mode = INVALID_MODE;
	int local_port = 7778;
	MSFactory *factory ;
	
	MSMediaStreamIO io = MS_MEDIA_STREAM_IO_INITIALIZER;
	int err;
	
	/*parse command line arguments*/
	
	if (argc<4) usage(argv[0]);
	
	command = argv[1];
	if (strcasecmp(command,"play")==0) mode = PLAY_MODE;
	else if (strcasecmp(command, "record")==0) mode = RECORD_MODE;
	else usage(argv[0]);
	
	file = argv[2];
	
	if (mode == PLAY_MODE){
		ip = argv[3];
		if (argc<5) usage(argv[0]);
		port = atoi(argv[4]);
	}else{
		local_port = atoi(argv[3]);
		ip = "127.0.0.1"; port = 9990; /*dummy destination address, we won't send anything here anyway*/
	}
	
	/*set a signal handler to interrupt the program cleanly*/
	signal(SIGINT,stop_handler);
	
	/*initialize mediastreamer2*/
	factory = ms_factory_new_with_voip();
	
	/*create the video stream */
	stream = video_stream_new(factory, local_port, local_port+1, FALSE);
	
	/*define its local input and outputs with the MSMediaStreamIO structure*/
	if (mode == PLAY_MODE){
		io.input.type = MSResourceFile;
		io.input.file = file; /*the file we want to stream out via rtp*/
		io.output.type = MSResourceFile;
		io.output.file = NULL; /*we don't set a record file in PLAY_MODE, we just want the received video stream to be ignored, if something is received*/
	}else{
		io.input.type = MSResourceFile;
		io.input.file = NULL; /*We don't want to send anything via RTP in RECORD_MODE*/
		io.output.type = MSResourceFile;
		io.output.file = file; /*The file to which we want to record the received video stream*/
	}
	
	/*define the RTP profile to use: in this case we just want to use H264 codec*/
	profile = rtp_profile_new("My RTP profile");
	pt = payload_type_clone(&payload_type_h264);
	rtp_profile_set_payload(profile, payload_type_number, pt); /*we assign H264 to payload type number payload_type_number*/
	
	media_stream_set_target_network_bitrate(&stream->ms, 500000); /*set a target IP bitrate in bits/second */
	
	/*By default, the VideoStream will show up a display window where the received video is played, with a local preview as well.
	 * If you don't need this, assign (void*)-1 as window id, which explicitely disable the display feature.*/
	
	/*video_stream_set_native_window_id(stream, (void*)-1);*/
	
	/*start the video stream, given the RtpProfile and "io" definition */
	err = video_stream_start_from_io(stream, profile, ip, port, ip, port+1, payload_type_number, &io);
	if (err !=0 ){
		fprintf(stderr,"Could not start video stream.");
		goto end;
	}
	/*Register an event handler on the player to be notified of end of file*/
	ms_filter_add_notify_callback(stream->source, on_end_of_play, NULL, FALSE);
	
	/*program's main loop*/
	while (active){
		/*handle video stream background activity. This is non blocking*/
		video_stream_iterate(stream);
		/*process event callbacks*/
		ms_event_queue_pump(ms_factory_get_event_queue(factory));
		ms_usleep(50000); /*pause 50ms to avoid busy loop*/
	}
	
end:
	/*stop and destroy the video stream object*/
	if (stream) video_stream_stop(stream);
	/*free the RTP profile and payload type inside*/
	if (profile) rtp_profile_destroy(profile);
	
	ms_factory_destroy(factory);
	
	return err;
}