static void basic_audio_stream() { AudioStream * marielle = audio_stream_new (MARIELLE_RTP_PORT, MARIELLE_RTCP_PORT,FALSE); stats_t marielle_stats; AudioStream * margaux = audio_stream_new (MARGAUX_RTP_PORT,MARGAUX_RTCP_PORT, FALSE); stats_t margaux_stats; RtpProfile* profile = rtp_profile_new("default profile"); 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_8K_1S_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_8K_1S_FILE , NULL , NULL , NULL , 0),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,1,12000)); 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,margaux_stats.rtp.recv); audio_stream_stop(marielle); audio_stream_stop(margaux); unlink(RECORDED_8K_1S_FILE); }
static void add_local_endpoint(LinphoneConference *conf,LinphoneCore *lc){ /*create a dummy audiostream in order to extract the local part of it */ /* network address and ports have no meaning and are not used here. */ AudioStream *st=audio_stream_new(65000,65001,FALSE); MSSndCard *playcard=lc->sound_conf.lsd_card ? lc->sound_conf.lsd_card : lc->sound_conf.play_sndcard; MSSndCard *captcard=lc->sound_conf.capt_sndcard; const MSAudioConferenceParams *params=ms_audio_conference_get_params(conf->conf); conf->local_dummy_profile=make_dummy_profile(params->samplerate); audio_stream_start_full(st, conf->local_dummy_profile, "127.0.0.1", 65000, "127.0.0.1", 65001, 0, 40, NULL, NULL, playcard, captcard, linphone_core_echo_cancellation_enabled(lc) ); _post_configure_audio_stream(st,lc,FALSE); conf->local_participant=st; conf->local_endpoint=ms_audio_endpoint_get_from_stream(st,FALSE); ms_audio_conference_add_member(conf->conf,conf->local_endpoint); }
// Wrap streams in Ruby objects VALUE streams_to_ruby_array(VALUE self, AVFormatContext * format) { VALUE streams = rb_ary_new(); unsigned i = 0; for(; i < format->nb_streams; ++i) { switch (format->streams[i]->codec->codec_type) { case AVMEDIA_TYPE_VIDEO: { // Video stream rb_ary_push(streams, video_stream_new(self, format->streams[i])); break; } case AVMEDIA_TYPE_AUDIO: { // Audio stream rb_ary_push(streams, audio_stream_new(self, format->streams[i])); break; } default: { // All other streams rb_ary_push(streams, stream_new(self, format->streams[i])); break; } } } return streams; }
void linphone_gtk_start_record_sound(GtkWidget *w, gpointer data){ LinphoneCore *lc = linphone_gtk_get_core(); AudioStream *stream = NULL; MSSndCardManager *manager = ms_snd_card_manager_get(); gboolean active=gtk_toggle_button_get_active(GTK_TOGGLE_BUTTON(w)); if(active){ gchar *path = get_record_file(); stream=audio_stream_new(8888, 8889, FALSE); if(stream != NULL){ audio_stream_start_full(stream,&av_profile,"127.0.0.1",8888,"127.0.0.1",8889,0,0,NULL, path,NULL,ms_snd_card_manager_get_card(manager,linphone_core_get_capture_device(lc)),FALSE); g_object_set_data(G_OBJECT(audio_assistant),"record_stream",stream); } gint timeout_id = gtk_timeout_add(6000,(GtkFunction)linphone_gtk_stop_record,NULL); g_object_set_data(G_OBJECT(audio_assistant),"timeout_id",GINT_TO_POINTER(timeout_id)); g_object_set_data(G_OBJECT(audio_assistant),"path",path); } else { stream = (AudioStream *)g_object_get_data(G_OBJECT(audio_assistant),"record_stream"); gint timeout_id = GPOINTER_TO_INT(g_object_get_data(G_OBJECT(audio_assistant),"timeout_id")); gtk_timeout_remove(timeout_id); if(stream != NULL){ audio_stream_stop(stream); g_object_set_data(G_OBJECT(audio_assistant),"record_stream",NULL); } update_record_button(FALSE); update_play_button(TRUE); } }
static stream_manager_t * stream_manager_new() { stream_manager_t * mgr = ms_new0(stream_manager_t,1); mgr->local_rtp= (rand() % ((2^16)-1024) + 1024) & ~0x1; mgr->local_rtcp=mgr->local_rtp+1; mgr->stream = audio_stream_new (mgr->local_rtp, mgr->local_rtcp,FALSE); return mgr; }
void linphone_call_init_media_streams(LinphoneCall *call){ LinphoneCore *lc=call->core; SalMediaDescription *md=call->localdesc; AudioStream *audiostream; call->audiostream=audiostream=audio_stream_new(md->streams[0].port,linphone_core_ipv6_enabled(lc)); if (linphone_core_echo_limiter_enabled(lc)){ const char *type=lp_config_get_string(lc->config,"sound","el_type","mic"); if (strcasecmp(type,"mic")==0) audio_stream_enable_echo_limiter(audiostream,ELControlMic); else if (strcasecmp(type,"full")==0) audio_stream_enable_echo_limiter(audiostream,ELControlFull); } audio_stream_enable_gain_control(audiostream,TRUE); if (linphone_core_echo_cancellation_enabled(lc)){ int len,delay,framesize; const char *statestr=lp_config_get_string(lc->config,"sound","ec_state",NULL); len=lp_config_get_int(lc->config,"sound","ec_tail_len",0); delay=lp_config_get_int(lc->config,"sound","ec_delay",0); framesize=lp_config_get_int(lc->config,"sound","ec_framesize",0); audio_stream_set_echo_canceller_params(audiostream,len,delay,framesize); if (statestr && audiostream->ec){ ms_filter_call_method(audiostream->ec,MS_ECHO_CANCELLER_SET_STATE_STRING,(void*)statestr); } } audio_stream_enable_automatic_gain_control(audiostream,linphone_core_agc_enabled(lc)); { int enabled=lp_config_get_int(lc->config,"sound","noisegate",0); audio_stream_enable_noise_gate(audiostream,enabled); } if (lc->a_rtp) rtp_session_set_transports(audiostream->session,lc->a_rtp,lc->a_rtcp); call->audiostream_app_evq = ortp_ev_queue_new(); rtp_session_register_event_queue(audiostream->session,call->audiostream_app_evq); #ifdef VIDEO_ENABLED if ((lc->video_conf.display || lc->video_conf.capture) && md->streams[1].port>0){ call->videostream=video_stream_new(md->streams[1].port,linphone_core_ipv6_enabled(lc)); if( lc->video_conf.displaytype != NULL) video_stream_set_display_filter_name(call->videostream,lc->video_conf.displaytype); video_stream_set_event_callback(call->videostream,video_stream_event_cb, call); if (lc->v_rtp) rtp_session_set_transports(call->videostream->session,lc->v_rtp,lc->v_rtcp); call->videostream_app_evq = ortp_ev_queue_new(); rtp_session_register_event_queue(call->videostream->session,call->videostream_app_evq); #ifdef TEST_EXT_RENDERER video_stream_set_render_callback(call->videostream,rendercb,NULL); #endif } #else call->videostream=NULL; #endif }
AudioStream * audio_stream_start(RtpProfile *prof,int locport,const char *remip,int remport,int profile,int jitt_comp,bool_t use_ec) { MSSndCard *sndcard_playback; MSSndCard *sndcard_capture; AudioStream *stream; sndcard_capture=ms_snd_card_manager_get_default_capture_card(ms_snd_card_manager_get()); sndcard_playback=ms_snd_card_manager_get_default_playback_card(ms_snd_card_manager_get()); if (sndcard_capture==NULL || sndcard_playback==NULL) return NULL; stream=audio_stream_new(locport, locport+1, ms_is_ipv6(remip)); if (audio_stream_start_full(stream,prof,remip,remport,remip,remport+1,profile,jitt_comp,NULL,NULL,sndcard_playback,sndcard_capture,use_ec)==0) return stream; audio_stream_free(stream); return NULL; }
AudioStream *audio_stream_start_with_sndcards(RtpProfile *prof,int locport,const char *remip,int remport,int profile,int jitt_comp,MSSndCard *playcard, MSSndCard *captcard, bool_t use_ec) { AudioStream *stream; if (playcard==NULL) { ms_error("No playback card."); return NULL; } if (captcard==NULL) { ms_error("No capture card."); return NULL; } stream=audio_stream_new(locport, locport+1, ms_is_ipv6(remip)); if (audio_stream_start_full(stream,prof,remip,remport,remip,remport+1,profile,jitt_comp,NULL,NULL,playcard,captcard,use_ec)==0) return stream; audio_stream_free(stream); return NULL; }
stream_manager_t * stream_manager_new(MSFormatType type) { stream_manager_t * mgr = ms_new0(stream_manager_t,1); mgr->type=type; mgr->local_rtp=(rand() % ((2^16)-1024) + 1024) & ~0x1; mgr->local_rtcp=mgr->local_rtp+1; mgr->user_data = 0; if (mgr->type==MSAudio){ mgr->audio_stream=audio_stream_new (mgr->local_rtp, mgr->local_rtcp,FALSE); }else{ #if VIDEO_ENABLED mgr->video_stream=video_stream_new (mgr->local_rtp, mgr->local_rtcp,FALSE); #else ms_fatal("Unsupported stream type [%s]",ms_format_type_to_string(mgr->type)); #endif } return mgr; }
void call_accept(Call *call) { sdp_context_t *ctx; PayloadType *payload; char *hellofile; static int call_count=0; char record_file[250]; osip_message_t *msg=NULL; sprintf(record_file,"/tmp/sipomatic%i.wav",call_count); ctx=call->sdpc; payload=rtp_profile_get_payload(call->profile,call->audio.pt); if (strcmp(payload->mime_type,"telephone-event")==0){ /* telephone-event is not enough to accept a call */ ms_message("Cannot accept call with only telephone-event.\n"); eXosip_call_send_answer(call->did,415,NULL); call->state=CALL_STATE_FINISHED; return; } if (payload->clock_rate==16000){ hellofile=call->root->file_path16000hz; }else hellofile=call->root->file_path8000hz; eXosip_call_build_answer(call->tid,200,&msg); osip_message_set_content_type(msg,"application/sdp"); osip_message_set_body(msg,call->sdpc->answerstr,strlen(call->sdpc->answerstr)); eXosip_call_send_answer(call->tid,200,msg); call->audio_stream=audio_stream_new(call->audio.localport,call->audio.localport+1,call->root->ipv6); audio_stream_start_with_files(call->audio_stream, call->profile, call->audio.remaddr,call->audio.remoteport,call->audio.remoteport+1, call->audio.pt,20,hellofile,record_file); call_count++; #ifdef VIDEO_ENABLED if (call->video.remoteport!=0){ video_stream_send_only_start(call->video_stream,call->profile, call->video.remaddr,call->video.remoteport,call->video.remoteport+1,call->video.pt, 60, ms_web_cam_manager_get_default_cam(ms_web_cam_manager_get())); } #endif call->time=time(NULL); call->state=CALL_STATE_RUNNING; ms_filter_set_notify_callback(call->audio_stream->soundread,endoffile_cb,(void*)call); }
static stream_manager_t * stream_manager_new(StreamType type) { stream_manager_t * mgr = ms_new0(stream_manager_t,1); mgr->type=type; mgr->local_rtp=(rand() % ((2^16)-1024) + 1024) & ~0x1; mgr->local_rtcp=mgr->local_rtp+1; mgr->evq=ortp_ev_queue_new(); if (mgr->type==AudioStreamType){ mgr->audio_stream=audio_stream_new (mgr->local_rtp, mgr->local_rtcp,FALSE); rtp_session_register_event_queue(mgr->audio_stream->ms.sessions.rtp_session,mgr->evq); }else{ #if VIDEO_ENABLED mgr->video_stream=video_stream_new (mgr->local_rtp, mgr->local_rtcp,FALSE); rtp_session_register_event_queue(mgr->video_stream->ms.sessions.rtp_session,mgr->evq); #else ms_fatal("Unsupported stream type [%s]",ms_stream_type_to_string(mgr->type)); #endif } return mgr; }
void linphone_gtk_start_play_record_sound(GtkWidget *w,gpointer data){ LinphoneCore *lc = linphone_gtk_get_core(); gboolean active=gtk_toggle_button_get_active(GTK_TOGGLE_BUTTON(w)); AudioStream *stream = NULL; MSSndCardManager *manager = ms_snd_card_manager_get(); if(active){ gchar *path = g_object_get_data(G_OBJECT(audio_assistant),"path"); stream=audio_stream_new(8888, 8889, FALSE); if(path != NULL){ audio_stream_start_full(stream,&av_profile,"127.0.0.1",8888,"127.0.0.1",8889,0,0,path, NULL,ms_snd_card_manager_get_card(manager,linphone_core_get_playback_device(lc)),NULL,FALSE); ms_filter_add_notify_callback(stream->soundread,endoffile_cb,stream,FALSE); g_object_set_data(G_OBJECT(audio_assistant),"play_stream",stream); } } else { stream = (AudioStream *)g_object_get_data(G_OBJECT(audio_assistant),"play_stream"); if(stream != NULL){ audio_stream_stop(stream); g_object_set_data(G_OBJECT(audio_assistant),"play_stream",NULL); } } }
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); }
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); }