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);
}
