void Audio::closeInputDevice()
{
if (!adc.isStreamOpen()) return;
adc.stopStream();
while (adc.isStreamRunning());
adc.closeStream();
}
int main()
{
// Set the global sample rate before creating class instances.
Stk::setSampleRate( 44100.0 );
SineWave sine;
RtAudio dac;
// Figure out how many bytes in an StkFloat and setup the RtAudio stream.
RtAudio::StreamParameters parameters;
parameters.deviceId = dac.getDefaultOutputDevice();
parameters.deviceId = 3;
parameters.nChannels = 1;
RtAudioFormat format = ( sizeof(StkFloat) == 8 ) ? RTAUDIO_FLOAT64 : RTAUDIO_FLOAT32;
unsigned int bufferFrames = RT_BUFFER_SIZE;
try {
dac.openStream( ¶meters, NULL, format, (unsigned int)Stk::sampleRate(),
&bufferFrames, &tick, (void *)&sine );
}
catch ( RtAudioError &error ) {
error.printMessage();
goto cleanup;
}
// configuration of oscilator
sine.setFrequency(440.0);
// start the main real time loop
try {
dac.startStream();
}
catch ( RtAudioError &error ) {
error.printMessage();
goto cleanup;
}
// USER interface
// Block waiting here.
char keyhit;
std::cout << "\nPlaying ... press <enter> to quit.\n";
std::cin.get( keyhit );
// SYSTEM shutdown
// Shut down the output stream.
try {
dac.closeStream();
}
catch ( RtAudioError &error ) {
error.printMessage();
}
cleanup:
return 0;
}
void av_audio_start() {
av_audio_get();
if (rta.isStreamRunning()) {
rta.stopStream();
}
if (rta.isStreamOpen()) {
// close it:
rta.closeStream();
}
unsigned int devices = rta.getDeviceCount();
if (devices < 1) {
printf("No audio devices found\n");
return;
}
RtAudio::DeviceInfo info;
RtAudio::StreamParameters iParams, oParams;
printf("Available audio devices (%d):\n", devices);
for (unsigned int i=0; i<devices; i++) {
info = rta.getDeviceInfo(i);
printf("Device %d: %dx%d (%d) %s\n", i, info.inputChannels, info.outputChannels, info.duplexChannels, info.name.c_str());
}
printf("device %d\n", audio.indevice);
info = rta.getDeviceInfo(audio.indevice);
printf("Using audio input %d: %dx%d (%d) %s\n", audio.indevice, info.inputChannels, info.outputChannels, info.duplexChannels, info.name.c_str());
audio.inchannels = info.inputChannels;
iParams.deviceId = audio.indevice;
iParams.nChannels = audio.inchannels;
iParams.firstChannel = 0;
info = rta.getDeviceInfo(audio.outdevice);
printf("Using audio output %d: %dx%d (%d) %s\n", audio.outdevice, info.inputChannels, info.outputChannels, info.duplexChannels, info.name.c_str());
audio.outchannels = info.outputChannels;
oParams.deviceId = audio.outdevice;
oParams.nChannels = audio.outchannels;
oParams.firstChannel = 0;
RtAudio::StreamOptions options;
//options.flags |= RTAUDIO_NONINTERLEAVED;
options.streamName = "av";
try {
rta.openStream( &oParams, &iParams, RTAUDIO_FLOAT32, audio.samplerate, &audio.blocksize, &av_rtaudio_callback, NULL, &options );
rta.startStream();
printf("Audio started\n");
}
catch ( RtError& e ) {
fprintf(stderr, "%s\n", e.getMessage().c_str());
}
}
int
playsin(void)
{
RtAudio *audio;
unsigned int bufsize = 4096;
CallbackData data;
try {
audio = new RtAudio(RtAudio::WINDOWS_WASAPI);
}
catch (...) {
return 1;
}
if (!audio) {
fprintf(stderr, "fail to allocate RtAudio¥n");
return 1;
}
/* probe audio devices */
unsigned int devId = audio->getDefaultOutputDevice();
/* Setup output stream parameters */
RtAudio::StreamParameters *outParam = new RtAudio::StreamParameters();
outParam->deviceId = devId;
outParam->nChannels = 2;
audio->openStream(outParam, NULL, RTAUDIO_FLOAT32, 44100,
&bufsize, rtaudio_callback, &data);
/* Create Wave Form Table */
data.nRate = 44100;
/* Frame Number is based on Freq(440Hz) and Sampling Rate(44100) */
/* hmm... nFrame = 44100 is enough approximation, maybe... */
data.nFrame = 44100;
data.nChannel = outParam->nChannels;
data.cur = 0;
data.wftable = (float *)calloc(data.nChannel * data.nFrame, sizeof(float));
if (!data.wftable)
{
delete audio;
fprintf(stderr, "fail to allocate memory¥n");
return 1;
}
for (unsigned int i = 0; i < data.nFrame; i++) {
float v = sin(i * 3.1416 * 2 * 440 / data.nRate);
for (unsigned int j = 0; j < data.nChannel; j++) {
data.wftable[i*data.nChannel + j] = v;
}
}
audio->startStream();
// sleep(10);
audio->stopStream();
audio->closeStream();
delete audio;
return 0;
}
void Audio::closeOutputDevice()
{
if (!dac.isStreamOpen()) return;
dac.stopStream();
while (dac.isStreamRunning());
dac.closeStream();
outIsOpened = false;
}
int main( int argc, char *argv[] )
{
unsigned int channels, fs, bufferFrames, device = 0, offset = 0;
char *file;
// minimal command-line checking
if ( argc < 4 || argc > 6 ) usage();
RtAudio dac;
if ( dac.getDeviceCount() < 1 ) {
std::cout << "\nNo audio devices found!\n";
exit( 0 );
}
channels = (unsigned int) atoi( argv[1]) ;
fs = (unsigned int) atoi( argv[2] );
file = argv[3];
if ( argc > 4 )
device = (unsigned int) atoi( argv[4] );
if ( argc > 5 )
offset = (unsigned int) atoi( argv[5] );
OutputData data;
data.fd = fopen( file, "rb" );
if ( !data.fd ) {
std::cout << "Unable to find or open file!\n";
exit( 1 );
}
// Set our stream parameters for output only.
bufferFrames = 512;
RtAudio::StreamParameters oParams;
oParams.deviceId = device;
oParams.nChannels = channels;
oParams.firstChannel = offset;
data.channels = channels;
try {
dac.openStream( &oParams, NULL, FORMAT, fs, &bufferFrames, &output, (void *)&data );
dac.startStream();
}
catch ( RtError& e ) {
std::cout << '\n' << e.getMessage() << '\n' << std::endl;
goto cleanup;
}
std::cout << "\nPlaying raw file " << file << " (buffer frames = " << bufferFrames << ")." << std::endl;
while ( 1 ) {
SLEEP( 100 ); // wake every 100 ms to check if we're done
if ( dac.isStreamRunning() == false ) break;
}
cleanup:
fclose( data.fd );
dac.closeStream();
return 0;
}
void stop() {
try {
// Stop the stream
dac.stopStream();
} catch (RtAudioError &e) {
e.printMessage();
}
if (dac.isStreamOpen()) dac.closeStream();
}
int main(int argc, char *argv[])
{
int buffer_size, fs, device = 0;
RtAudio *audio;
double *data;
char input;
// minimal command-line checking
if (argc != 3 && argc != 4 ) usage();
chans = (int) atoi(argv[1]);
fs = (int) atoi(argv[2]);
if ( argc == 4 )
device = (int) atoi(argv[3]);
// Open the realtime output device
buffer_size = 1024;
try {
audio = new RtAudio(device, chans, 0, 0,
FORMAT, fs, &buffer_size, 4);
}
catch (RtError &error) {
error.printMessage();
exit(EXIT_FAILURE);
}
data = (double *) calloc(chans, sizeof(double));
try {
audio->setStreamCallback(&saw, (void *)data);
audio->startStream();
}
catch (RtError &error) {
error.printMessage();
goto cleanup;
}
std::cout << "\nPlaying ... press <enter> to quit (buffer size = " << buffer_size << ").\n";
std::cin.get(input);
// Stop the stream.
try {
audio->stopStream();
}
catch (RtError &error) {
error.printMessage();
}
cleanup:
audio->closeStream();
delete audio;
if (data) free(data);
return 0;
}
void
stop_audio(void)
{
try {
audio.stopStream();
} catch(RtError& e) {
e.printMessage();
}
if(audio.isStreamOpen())
audio.closeStream();
}
int main( int argc, char *argv[] )
{
if ( argc != 2 ) usage();
// Set the global sample rate and rawwave path before creating class instances.
Stk::setSampleRate( 44100.0 );
Stk::setRawwavePath( "rawwaves/" );
TickData data;
RtAudio dac;
// Figure out how many bytes in an StkFloat and setup the RtAudio stream.
RtAudio::StreamParameters parameters;
parameters.deviceId = dac.getDefaultOutputDevice();
parameters.nChannels = 1;
RtAudioFormat format = ( sizeof(StkFloat) == 8 ) ? RTAUDIO_FLOAT64 : RTAUDIO_FLOAT32;
unsigned int bufferFrames = RT_BUFFER_SIZE;
try {
dac.openStream( ¶meters, NULL, format, (unsigned int)Stk::sampleRate(), &bufferFrames, &tick, (void *)&data );
}
catch ( RtAudioError &error ) {
error.printMessage();
goto cleanup;
}
try {
// Define and load the BeeThree instrument
data.instrument = new BeeThree();
}
catch ( StkError & ) {
goto cleanup;
}
if ( data.messager.setScoreFile( argv[1] ) == false )
goto cleanup;
try {
dac.startStream();
}
catch ( RtAudioError &error ) {
error.printMessage();
goto cleanup;
}
// Block waiting until callback signals done.
while ( !data.done )
Stk::sleep( 100 );
// Shut down the output stream.
try {
dac.closeStream();
}
catch ( RtAudioError &error ) {
error.printMessage();
}
cleanup:
delete data.instrument;
return 0;
}
int main(int argc, char *argv[])
{
int chans, fs, device = 0;
RtAudio *audio;
char input;
// minimal command-line checking
if (argc != 3 && argc != 4 ) usage();
chans = (int) atoi(argv[1]);
fs = (int) atoi(argv[2]);
if ( argc == 4 )
device = (int) atoi(argv[3]);
// Open the realtime output device
int buffer_size = 512;
try {
audio = new RtAudio(device, chans, device, chans,
FORMAT, fs, &buffer_size, 8);
}
catch (RtError &error) {
error.printMessage();
exit(EXIT_FAILURE);
}
try {
audio->setStreamCallback(&inout, NULL);
audio->startStream();
}
catch (RtError &error) {
error.printMessage();
goto cleanup;
}
std::cout << "\nRunning ... press <enter> to quit (buffer size = " << buffer_size << ").\n";
std::cin.get(input);
try {
audio->stopStream();
}
catch (RtError &error) {
error.printMessage();
}
cleanup:
audio->closeStream();
delete audio;
return 0;
}
int main()
{
RtAudio dac;
//std::cout << dac.getDeviceCount() << std::endl; //2
if (dac.getDeviceCount() < 1) {
std::cout << "\nNo audio devices found!\n";
exit(0);
}
RtAudio::StreamParameters parameters;
//std::cout << dac.getDefaultOutputDevice() << std::endl;
parameters.deviceId = dac.getDefaultOutputDevice(); //0
parameters.nChannels = 2;
parameters.firstChannel = 0;
unsigned int sampleRate = 44100;
unsigned int bufferFrames = 256; // 256 sample frames
RtAudio::StreamParameters input;
input.deviceId = dac.getDefaultInputDevice();
input.nChannels = 2;
input.firstChannel = 0;
double data[2];
try {
dac.openStream(¶meters, &input, RTAUDIO_SINT16,
sampleRate, &bufferFrames, &saw, (void *)&data);
dac.startStream();
}
catch (RtAudioError& e) {
e.printMessage();
exit(0);
}
char input1;
std::cout << "\nPlaying ... press <enter> to quit.\n";
std::cin.get(input1);
try {
// Stop the stream
dac.stopStream();
}
catch (RtAudioError& e) {
e.printMessage();
}
if (dac.isStreamOpen()) dac.closeStream();
system("pause");
return 0;
}
static void close_output(va_list args)
{
try
{
// Stop the stream
vessl_out.stopStream();
if ( vessl_out.isStreamOpen() )
{
vessl_out.closeStream();
printf("[vessl] output was closed.\n");
}
}
catch (RtAudioError& e)
{
e.printMessage();
}
}
void keyboard (unsigned char key, int x, int y)
{
switch (key) {
case 'a':
autorotate = ! autorotate;
break;
case 't' : theString->Ktension *= 1.05946;
theString->Ktension = fmin ( 1.0, theString->Ktension );
break;
case 'l' : theString->Ktension *= 0.943876; break;
case 'P' : theString->pluck(); break;
case 'p' : theString->pluckvel(); break;
case 'r' : theString->reset(); break;
case 'd' : theString->print(); break;
case 'v' : theString->toggleVibrator(); break;
case 'f' : theString->vibratorFreq *= 0.943875;
std::cout << "vib freq = " << theString->vibratorFreq << std::endl;
break;
case 'F' : theString->vibratorFreq /= 0.943875;
std::cout << "vib freq = " << theString->vibratorFreq << std::endl;
break;
case 27: /* ESC */
try {
// Stop the stream
dac.stopStream();
}
catch (RtError& e) {
e.printMessage();
}
if (dac.isStreamOpen()) dac.closeStream();
exit(0);
break;
default: break;
}
}
int main( int argc, char *argv[] )
{
unsigned int bufferFrames, fs, device = 0, offset = 0;
// minimal command-line checking
if (argc < 3 || argc > 6 ) usage();
RtAudio dac;
if ( dac.getDeviceCount() < 1 ) {
std::cout << "\nNo audio devices found!\n";
exit( 1 );
}
channels = (unsigned int) atoi( argv[1] );
fs = (unsigned int) atoi( argv[2] );
if ( argc > 3 )
device = (unsigned int) atoi( argv[3] );
if ( argc > 4 )
offset = (unsigned int) atoi( argv[4] );
if ( argc > 5 )
nFrames = (unsigned int) (fs * atof( argv[5] ));
if ( nFrames > 0 ) checkCount = true;
double *data = (double *) calloc( channels, sizeof( double ) );
// Let RtAudio print messages to stderr.
dac.showWarnings( true );
// Set our stream parameters for output only.
bufferFrames = 256;
RtAudio::StreamParameters oParams;
oParams.deviceId = device;
oParams.nChannels = channels;
oParams.firstChannel = offset;
options.flags |= RTAUDIO_HOG_DEVICE;
options.flags |= RTAUDIO_SCHEDULE_REALTIME;
#if !defined( USE_INTERLEAVED )
options.flags |= RTAUDIO_NONINTERLEAVED;
#endif
try {
dac.openStream( &oParams, NULL, FORMAT, fs, &bufferFrames, &saw, (void *)data, &options );
dac.startStream();
}
catch ( RtError& e ) {
e.printMessage();
goto cleanup;
}
if ( checkCount ) {
while ( dac.isStreamRunning() == true ) SLEEP( 100 );
}
else {
char input;
//std::cout << "Stream latency = " << dac.getStreamLatency() << "\n" << std::endl;
std::cout << "\nPlaying ... press <enter> to quit (buffer size = " << bufferFrames << ").\n";
std::cin.get( input );
try {
// Stop the stream
dac.stopStream();
}
catch ( RtError& e ) {
e.printMessage();
}
}
cleanup:
if ( dac.isStreamOpen() ) dac.closeStream();
free( data );
return 0;
}
// entry point
int main( int argc, char ** argv )
{
// initialize GLUT
glutInit( &argc, argv );
// double buffer, use rgb color, enable depth buffer
glutInitDisplayMode( GLUT_DOUBLE | GLUT_RGB | GLUT_DEPTH );
// initialize the window size
glutInitWindowSize( g_width, g_height );
// set the window postion
glutInitWindowPosition( 100, 100 );
// create the window
glutCreateWindow( "VisualSine" );
// set the idle function - called when idle
glutIdleFunc( idleFunc );
// set the display function - called when redrawing
glutDisplayFunc( displayFunc );
// set the reshape function - called when client area changes
glutReshapeFunc( reshapeFunc );
// set the keyboard function - called on keyboard events
glutKeyboardFunc( keyboardFunc );
// set the mouse function - called on mouse stuff
glutMouseFunc( mouseFunc );
// RtAudio pointer
RtAudio * audio = NULL;
// buffer size
int buffer_size = 512;
// create the RtAudio
try {
audio = new RtAudio(
0, // device number of output
1, // number of output channels
1, // device number for input
1, // number of input channels
RTAUDIO_FLOAT64, // format
MY_SRATE, // sample rate
&buffer_size, // buffer size
8 // number of buffers
);
} catch( RtError & err ) {
err.printMessage();
exit(1);
}
// allocate global buffer
g_buffer = new SAMPLE[buffer_size];
g_bufferSize = buffer_size;
// set the callback
try {
audio->setStreamCallback( &callme, NULL );
audio->startStream();
} catch( RtError & err ) {
// do stuff
err.printMessage();
goto cleanup;
}
// let GLUT handle the current thread from here
glutMainLoop();
// if we get here, then stop!
try {
audio->stopStream();
} catch( RtError & err ) {
// do stuff
err.printMessage();
}
cleanup:
audio->closeStream();
delete audio;
return 0;
}
int main () {
// Damit das Programm funktioniert, muss eine 16Bit PCM Wave-Datei im
// gleichen Ordner liegen !
const char * fname = "test.flac" ;
// Soundfile-Handle aus der libsndfile-Bibliothek
SndfileHandle file = SndfileHandle (fname) ;
// Alle möglichen Infos über die Audio-Datei ausgeben !
std::cout << "Reading file: " << fname << std::endl;
std::cout << "File format: " << file.format() << std::endl;
std::cout << "PCM 16 BIT: " << (SF_FORMAT_WAV | SF_FORMAT_PCM_16) << std::endl;
std::cout << "Samples in file: " << file.frames() << std::endl;
std::cout << "Samplerate " << file.samplerate() << std::endl;
std::cout << "Channels: " << file.channels() << std::endl;
// Die RtAudio-Klasse ist gleichermassen dac und adc, wird hier aber nur als dac verwendet !
RtAudio dac;
if ( dac.getDeviceCount() < 1 ) {
std::cout << "\nNo audio devices found!\n";
return 0;
}
// Output params ...
RtAudio::StreamParameters parameters;
parameters.deviceId = dac.getDefaultOutputDevice();
parameters.nChannels = 2;
parameters.firstChannel = 0;
unsigned int sampleRate = 44100;
// ACHTUNG! Frames != Samples
// ein Frame = Samples für alle Kanäle
// d.h. |Samples| = Kanäle*Frames !
unsigned int bufferFrames = 1024;
// Da wir 16 Bit PCM-Daten lesen, sollte als Datenformat RTAUDIO_SINT16 genutzt
// werden.
// Als Daten wird der Callback-Struktur hier das Soundfile-Handle übergeben.
// Sollte man in einer "ernsthaften" Lösung anders machen !
// Inkompatible Formate können übrigens "interessante" Effekte ergeben !
try {
dac.openStream( ¶meters, NULL, RTAUDIO_SINT16,
sampleRate, &bufferFrames, &fplay, (void *)&file);
dac.startStream();
}
catch ( RtAudioError& e ) {
e.printMessage();
return 0;
}
char input;
std::cout << "\nPlaying ... press <enter> to quit.\n";
std::cin.get( input );
try {
// Stop the stream
dac.stopStream();
}
catch (RtAudioError& e) {
e.printMessage();
}
if ( dac.isStreamOpen() ) dac.closeStream();
return 0 ;
}
//-----------------------------------------------------------------------------
// name: main()
// desc: entry point
//-----------------------------------------------------------------------------
int main( int argc, char ** argv )
{
callbackData data;
// global for frequency
data.g_freq=440;
// global sample number variable
data.g_t = 0;
// global for width;
data.g_width = 0;
//global for input
data.g_input=0;
//check parameters and parse input
if (!parse(argc,argv,data))
{
exit(0);
}
// instantiate RtAudio object
RtAudio adac;
// variables
unsigned int bufferBytes = 0;
// frame size
unsigned int bufferFrames = 512;
// check for audio devices
if( adac.getDeviceCount() < 1 )
{
// nopes
cout << "no audio devices found!" << endl;
exit( 1 );
}
// let RtAudio print messages to stderr.
adac.showWarnings( true );
// set input and output parameters
RtAudio::StreamParameters iParams, oParams;
iParams.deviceId = adac.getDefaultInputDevice();
iParams.nChannels = MY_CHANNELS;
iParams.firstChannel = 0;
oParams.deviceId = adac.getDefaultOutputDevice();
oParams.nChannels = MY_CHANNELS;
oParams.firstChannel = 0;
// create stream options
RtAudio::StreamOptions options;
// go for it
try {
// open a stream
adac.openStream( &oParams, &iParams, MY_FORMAT, MY_SRATE, &bufferFrames, &callme, (void *)&data, &options );
}
catch( RtError& e )
{
// error!
cout << e.getMessage() << endl;
exit( 1 );
}
// compute
bufferBytes = bufferFrames * MY_CHANNELS * sizeof(SAMPLE);
// test RtAudio functionality for reporting latency.
cout << "stream latency: " << adac.getStreamLatency() << " frames" << endl;
// go for it
try {
// start stream
adac.startStream();
// get input
char input;
std::cout << "running... press <enter> to quit (buffer frames: " << bufferFrames << ")" << endl;
std::cin.get(input);
// stop the stream.
adac.stopStream();
}
catch( RtError& e )
{
// print error message
cout << e.getMessage() << endl;
goto cleanup;
}
cleanup:
// close if open
if( adac.isStreamOpen() )
adac.closeStream();
// done
outfile<<"];\nplot(x)";
return 0;
}
int main( int argc, char *argv[] )
{
//Dekrispator init
randomGen_init();
Synth_Init();
//end Dekrispator init
// FILE* f = fopen("bla.txt","wb");
// fclose(f);
TickData data;
RtAudio dac;
int i;
//if ( argc < 2 || argc > 6 ) usage();
// If you want to change the default sample rate (set in Stk.h), do
// it before instantiating any objects! If the sample rate is
// specified in the command line, it will override this setting.
Stk::setSampleRate( 44100.0 );
{
RtMidiIn *midiin = 0;
midiin = new RtMidiIn();
unsigned int i = 0, nPorts = midiin->getPortCount();
if ( nPorts == 0 ) {
std::cout << "No input Midi ports available, just running demo mode." << std::endl;
delete midiin;
midiin = 0;
} else
{
for ( i=0; i<nPorts; i++ ) {
std::string portName = midiin->getPortName(i);
std::cout << " Input port #" << i << ": " << portName << '\n';
}
delete midiin;
midiin = 0;
for ( i=0; i<nPorts && i<MAX_MIDI_DEVICES; i++ ) {
data.messagers[data.numMessagers++].startMidiInput(i);
}
}
}
// Parse the command-line arguments.
unsigned int port = 2001;
for ( i=1; i<argc; i++ ) {
if ( !strcmp( argv[i], "-is" ) ) {
if ( i+1 < argc && argv[i+1][0] != '-' ) port = atoi(argv[++i]);
if (data.numMessagers<MAX_MIDI_DEVICES)
{
data.messagers[data.numMessagers++].startSocketInput( port );
}
}
else if (!strcmp( argv[i], "-ip" ) )
{
if (data.numMessagers<MAX_MIDI_DEVICES)
{
data.messagers[data.numMessagers++].startStdInput();
}
}
else if ( !strcmp( argv[i], "-s" ) && ( i+1 < argc ) && argv[i+1][0] != '-')
Stk::setSampleRate( atoi(argv[++i]) );
else
usage();
}
// Allocate the dac here.
RtAudioFormat format = ( sizeof(StkFloat) == 8 ) ? RTAUDIO_FLOAT64 : RTAUDIO_FLOAT32;
RtAudio::StreamParameters parameters;
parameters.deviceId = dac.getDefaultOutputDevice();
parameters.nChannels = 2;
unsigned int bufferFrames = RT_BUFFER_SIZE;
try {
dac.openStream( ¶meters, NULL, format, (unsigned int)Stk::sampleRate(), &bufferFrames, &tick, (void *)&data );
}
catch ( RtAudioError& error ) {
error.printMessage();
goto cleanup;
}
data.reverbs[0].setT60( data.t60 );
data.reverbs[0].setEffectMix( 0.5 );
data.reverbs[1].setT60( 2.0 );
data.reverbs[1].setEffectMix( 0.2 );
data.rateScaler = 22050.0 / Stk::sampleRate();
// Install an interrupt handler function.
(void) signal( SIGINT, finish );
// If realtime output, set our callback function and start the dac.
try {
dac.startStream();
}
catch ( RtAudioError &error ) {
error.printMessage();
goto cleanup;
}
// Setup finished.
while ( !done ) {
// Periodically check "done" status.
Stk::sleep( 50 );
}
// Shut down the output stream.
try {
dac.closeStream();
}
catch ( RtAudioError& error ) {
error.printMessage();
}
cleanup:
return 0;
}
int main()
{
// Set the global sample rate before creating class instances.
Stk::setSampleRate( 44100.0 );
SineWave sine;
RtAudio dac;
// Figure out how many bytes in an StkFloat and setup the RtAudio stream.
RtAudio::StreamParameters parameters;
parameters.deviceId = dac.getDefaultOutputDevice();
parameters.nChannels = 1;
RtAudioFormat format = ( sizeof(StkFloat) == 8 ) ? RTAUDIO_FLOAT64 : RTAUDIO_FLOAT32;
unsigned int bufferFrames = RT_BUFFER_SIZE;
try {
dac.openStream( ¶meters, NULL, format, (unsigned int)Stk::sampleRate(), &bufferFrames, &tick, (void *)&sine );
}
catch ( RtError &error ) {
error.printMessage();
goto cleanup;
}
double f = 440.0;
double twelveRoot2 = 1.0594630943592952645618252949463;
sine.setFrequency(f);
try {
dac.startStream();
}
catch ( RtError &error ) {
error.printMessage();
goto cleanup;
}
// Block waiting here.
int keyhit = 0;
std::cout << "\nPlaying ... press <esc> to quit.\n";
while (keyhit != 32 && keyhit != 27)
{
keyhit = _getch();
if (tolower(keyhit) == 'a')
{
f = 220.0;
sine.setFrequency(f);
}
else if (tolower(keyhit) == 'g')
{
f /= twelveRoot2;
sine.setFrequency(f);
}
else if (tolower(keyhit) == 'h')
{
f *= twelveRoot2;
sine.setFrequency(f);
}
else if (tolower(keyhit) == 'f')
{
for (int i = 0; i < 2; ++i)
f /= twelveRoot2;
sine.setFrequency(f);
}
else if (tolower(keyhit) == 'j')
{
for (int i = 0; i < 2; ++i)
f *= twelveRoot2;
sine.setFrequency(f);
}
else if (tolower(keyhit) == 'd')
{
for (int i = 0; i < 3; ++i)
f /= twelveRoot2;
sine.setFrequency(f);
}
else if (tolower(keyhit) == 'k')
{
for (int i = 0; i < 3; ++i)
f *= twelveRoot2;
sine.setFrequency(f);
}
else if (tolower(keyhit) == 's')
{
for (int i = 0; i < 4; ++i)
f /= twelveRoot2;
sine.setFrequency(f);
}
else if (tolower(keyhit) == 'l')
{
for (int i = 0; i < 4; ++i)
f *= twelveRoot2;
sine.setFrequency(f);
}
else
{
std::cout << "Freq: " << f << std::endl;
}
}
// Shut down the output stream.
try {
dac.closeStream();
}
catch ( RtError &error ) {
error.printMessage();
}
cleanup:
return 0;
}
int main(const int argc, const char *argv[]) {
RtAudio adc;
unsigned int deviceCount = adc.getDeviceCount();
if (deviceCount < 1) {
cout << endl << "No audio devices found!" << endl;
exit(0);
}
unsigned int inputDevice = adc.getDefaultInputDevice();
unsigned int outputDevice = adc.getDefaultOutputDevice();
for (int i=0; i<argc; i++) {
if (strcmp(argv[i], "-devices") == 0) {
// Scan through devices for various capabilities
showDevices(deviceCount, adc);
exit(0);
}
if (strcmp(argv[i], "-input") == 0) {
if (i == argc-1) {
usage();
exit(0);
}
inputDevice=atoi(argv[++i]);
validateDevice(inputDevice, deviceCount, adc, true);
}
if (strcmp(argv[i], "-output") == 0) {
if (i == argc-1) {
usage();
exit(0);
}
outputDevice=atoi(argv[++i]);
validateDevice(outputDevice, deviceCount, adc, false);
}
}
// Initialise DSP thread
// Initialise GUI
unsigned int sampleRate = 44100;
unsigned int bufferFrames = 512;
unsigned int bufferBytes = 0;
RtAudio::StreamParameters inputParameters;
inputParameters.deviceId = inputDevice;
inputParameters.nChannels = 2;
inputParameters.firstChannel = 0;
RtAudio::StreamParameters outputParameters;
outputParameters.deviceId = outputDevice;
outputParameters.nChannels = 2;
outputParameters.firstChannel = 0;
try {
adc.openStream(&outputParameters, &inputParameters, RTAUDIO_SINT16, sampleRate,
&bufferFrames, &inout, &bufferBytes);
adc.startStream();
} catch (RtAudioError& e) {
e.printMessage();
exit(0);
}
// adc.openStream could have adjusted the bufferFrames.
// Set the user data buffer to the sample buffer size in bytes, so that the
// inout callback function knows how much data to copy. The example code
// uses this - 2 is Stereo, 4 is signed int (4 bytes on OSX)
bufferBytes = bufferFrames * 2 * 4;
// Can now initialise buffer management. inout could have been asking for
// buffers but buffer management won't give them until it has been
// initialised.
cout << "buffer size in bytes is " << bufferBytes << endl;
// TODO protect with mutex
bufferManager = new BufferManager(bufferBytes, maxBuffers);
char input;
cout << endl << "Recording ... press <enter> to quit." << endl;
cin.get(input);
cout << "Terminating" << endl;
try {
// Stop the stream
adc.stopStream();
} catch (RtAudioError& e) {
e.printMessage();
}
if (adc.isStreamOpen())
adc.closeStream();
// TODO shut down DSP chain, release all buffers
// TODO shut down Display chain, release all buffers
delete bufferManager;
cout << "Terminated" << endl;
return 0;
}
int main( int argc, char *argv[] )
{
TickData data;
int i;
#if defined(__STK_REALTIME__)
RtAudio dac;
#endif
// If you want to change the default sample rate (set in Stk.h), do
// it before instantiating any objects! If the sample rate is
// specified in the command line, it will override this setting.
Stk::setSampleRate( 44100.0 );
// By default, warning messages are not printed. If we want to see
// them, we need to specify that here.
Stk::showWarnings( true );
// Check the command-line arguments for errors and to determine
// the number of WvOut objects to be instantiated (in utilities.cpp).
data.nWvOuts = checkArgs( argc, argv );
data.wvout = (WvOut **) calloc( data.nWvOuts, sizeof(WvOut *) );
// Parse the command-line flags, instantiate WvOut objects, and
// instantiate the input message controller (in utilities.cpp).
try {
data.realtime = parseArgs( argc, argv, data.wvout, data.messager );
}
catch (StkError &) {
goto cleanup;
}
// If realtime output, allocate the dac here.
#if defined(__STK_REALTIME__)
if ( data.realtime ) {
RtAudioFormat format = ( sizeof(StkFloat) == 8 ) ? RTAUDIO_FLOAT64 : RTAUDIO_FLOAT32;
RtAudio::StreamParameters parameters;
parameters.deviceId = dac.getDefaultOutputDevice();
parameters.nChannels = data.channels;
unsigned int bufferFrames = RT_BUFFER_SIZE;
try {
dac.openStream( ¶meters, NULL, format, (unsigned int)Stk::sampleRate(), &bufferFrames, &tick, (void *)&data );
}
catch ( RtAudioError& error ) {
error.printMessage();
goto cleanup;
}
}
#endif
// Set the reverb parameters.
data.reverb.setT60( data.t60 );
data.reverb.setEffectMix( 0.2 );
// Allocate guitar
data.guitar = new Guitar( nStrings );
// Configure distortion and feedback.
data.distortion.setThreshold( 2.0 / 3.0 );
data.distortion.setA1( 1.0 );
data.distortion.setA2( 0.0 );
data.distortion.setA3( -1.0 / 3.0 );
data.distortionMix = 0.9;
data.distortionGain = 1.0;
data.feedbackDelay.setMaximumDelay( (unsigned long int)( 1.1 * Stk::sampleRate() ) );
data.feedbackDelay.setDelay( 20000 );
data.feedbackGain = 0.001;
data.oldFeedbackGain = 0.001;
// Install an interrupt handler function.
(void) signal(SIGINT, finish);
// If realtime output, set our callback function and start the dac.
#if defined(__STK_REALTIME__)
if ( data.realtime ) {
try {
dac.startStream();
}
catch ( RtAudioError &error ) {
error.printMessage();
goto cleanup;
}
}
#endif
// Setup finished.
while ( !done ) {
#if defined(__STK_REALTIME__)
if ( data.realtime )
// Periodically check "done" status.
Stk::sleep( 200 );
else
#endif
// Call the "tick" function to process data.
tick( NULL, NULL, 256, 0, 0, (void *)&data );
}
// Shut down the output stream.
#if defined(__STK_REALTIME__)
if ( data.realtime ) {
try {
dac.closeStream();
}
catch ( RtAudioError& error ) {
error.printMessage();
}
}
#endif
cleanup:
for ( i=0; i<(int)data.nWvOuts; i++ ) delete data.wvout[i];
free( data.wvout );
delete data.guitar;
std::cout << "\nStk eguitar finished ... goodbye.\n\n";
return 0;
}
int main( int argc, char *argv[])
{
// COMMAND LINE ARG HANDLING
map<string, ugen> ugens;
ugens["--sine"] = &sine;
ugens["--saw"] = &saw;
ugens["--pulse"] = &pulse;
ugens["--noise"] = &noise;
ugens["--impulse"] = &impulse;
if (argc < 4 || argc > 10 ) usage();
string type_arg = argv[1];
g_active_ugen = ugens[type_arg];
if (g_active_ugen == NULL)
usage();
double freq_arg = atof(argv[2]);
if (freq_arg <= 0)
usage();
g_frequency = freq_arg;
double width_arg = atof(argv[3]);
if (width_arg < 0 || width_arg > 1)
usage();
g_width = width_arg;
if (argc > 4) { // modulation parameters present
for (int i = 4; i < argc;) {
if (string(argv[i]).compare("--input") == 0) {
g_modulate_input = true;
i++;
}
else if (string(argv[i]).compare("--fm") == 0) {
g_fm_on = true;
string fm_type_arg = argv[++i];
g_active_fm_ugen = ugens[fm_type_arg];
if (g_active_fm_ugen == NULL)
usage();
double fm_freq_arg = atof(argv[++i]);
if (fm_freq_arg <= 0)
usage();
g_fm_frequency = fm_freq_arg;
double fm_width_arg = atof(argv[++i]);
if (fm_width_arg < 0 || fm_width_arg > 1)
usage();
g_fm_width = fm_width_arg;
double fm_index_arg = atoi(argv[++i]);
g_fm_index = fm_index_arg;
i++;
}
else
usage();
}
}
// AUDIO SETUP
RtAudio audio;
audio.showWarnings( true );
RtAudio::StreamParameters output_params;
RtAudio::StreamParameters input_params;
// Choose an audio device and a sample rate
unsigned int sample_rate;
unsigned int devices = audio.getDeviceCount();
if ( devices < 1 ) {
cerr << "No audio device found!" << endl;
exit(1);
}
RtAudio::DeviceInfo info;
for (unsigned int i = 0; i < devices; i++ ) {
info = audio.getDeviceInfo(i);
if ( info.isDefaultOutput ) {
output_params.deviceId = i;
output_params.nChannels = 2;
if (info.sampleRates.size() < 1) {
cerr << "No supported sample rates found!" << endl;
exit(1);
}
for (int i = 0; i < info.sampleRates.size(); i++) {
sample_rate = info.sampleRates[i];
if (sample_rate == 44100 || sample_rate == 48000) {
// Found a nice sample rate, stop looking
break;
}
}
cout << "Using sample rate: " << sample_rate << endl;
}
if ( info.isDefaultInput ) {
input_params.deviceId = i;
input_params.nChannels = 1;
}
}
cout << "Using output device ID " << output_params.deviceId << " which has " <<
output_params.nChannels << " output channels." << endl;
cout << "Using input device ID " << input_params.deviceId << " which has " <<
input_params.nChannels << " input channels." << endl;
RtAudio::StreamOptions options;
options.flags |= RTAUDIO_HOG_DEVICE;
options.flags |= RTAUDIO_SCHEDULE_REALTIME;
unsigned int buffer_frames = 256;
try {
audio.openStream( &output_params, // output params
&input_params, // input params
RTAUDIO_FLOAT64, // audio format
sample_rate, // sample rate
&buffer_frames, // num frames per buffer (mutable by rtaudio)
&callback, // audio callback
&audio, // user data pointer HACK HACK :D
&options); // stream options
audio.startStream();
} catch ( RtError &e ) {
e.printMessage();
goto cleanup;
}
char input;
cout << "Playing, press enter to quit (buffer frames = " << buffer_frames << ")." << endl;
cin.get( input );
try {
audio.stopStream();
}
catch ( RtError &e ) {
e.printMessage();
}
cleanup:
if ( audio.isStreamOpen() ) {
audio.closeStream();
}
return 0;
}
int main(int argc, char *argv[])
{
int chans, fs, buffer_size, count, device = 0;
long counter = 0;
MY_TYPE *buffer;
char *file;
FILE *fd;
RtAudio *audio;
// minimal command-line checking
if (argc != 4 && argc != 5 ) usage();
chans = (int) atoi(argv[1]);
fs = (int) atoi(argv[2]);
file = argv[3];
if ( argc == 5 )
device = (int) atoi(argv[4]);
fd = fopen(file,"rb");
if (!fd) {
std::cout << "can't find file!\n";
exit(0);
}
// Open the realtime output device
buffer_size = 512;
try {
audio = new RtAudio(device, chans, 0, 0,
FORMAT, fs, &buffer_size, 2);
}
catch (RtError &error) {
fclose(fd);
error.printMessage();
exit(EXIT_FAILURE);
}
try {
buffer = (MY_TYPE *) audio->getStreamBuffer();
audio->startStream();
}
catch (RtError &error) {
error.printMessage();
goto cleanup;
}
while (1) {
count = fread(buffer, chans * sizeof(MY_TYPE), buffer_size, fd);
if (count == buffer_size) {
try {
audio->tickStream();
}
catch (RtError &error) {
error.printMessage();
goto cleanup;
}
}
else
break;
counter += buffer_size;
}
try {
audio->stopStream();
}
catch (RtError &error) {
error.printMessage();
}
cleanup:
audio->closeStream();
delete audio;
fclose(fd);
return 0;
}
// ========
// = Main =
// ========
// Entry point
int main (int argc, char *argv[])
{
cout<<argc<<" "<<argv[0];
if (argc>3) {cerr<<"\nERROR - wrong number of arguments\n";exit(1);}
if (argc==3)
g_audio_history = atoi(argv[2]);
else
g_audio_history = 30;
if (argc>1)
g_fft_history = atoi(argv[1]);
else
g_fft_history = 100;
help();
// RtAudio config + init
// pointer to RtAudio object
RtAudio * audio = NULL;
// create the object
try
{
audio = new RtAudio();
}
catch( RtError & err ) {
err.printMessage();
exit(1);
}
if( audio->getDeviceCount() < 1 )
{
// nopes
cout << "no audio devices found!" << endl;
exit( 1 );
}
// let RtAudio print messages to stderr.
audio->showWarnings( true );
// set input and output parameters
RtAudio::StreamParameters iParams, oParams;
iParams.deviceId = audio->getDefaultInputDevice();
iParams.nChannels = 1;
iParams.firstChannel = 0;
oParams.deviceId = audio->getDefaultOutputDevice();
oParams.nChannels = 1;
oParams.firstChannel = 0;
// create stream options
RtAudio::StreamOptions options;
// set the callback and start stream
try
{
audio->openStream( &oParams, &iParams, RTAUDIO_FLOAT64, MY_SRATE, &g_buffSize, &audioCallback, NULL, &options);
cerr << "Buffer size defined by RtAudio: " << g_buffSize << endl;
// allocate the buffer for the fft
g_fftBuff = new float[g_buffSize * ZPF];
g_audioBuff = new float[g_buffSize * ZPF];
if ( g_fftBuff == NULL ) {
cerr << "Something went wrong when creating the fft and audio buffers" << endl;
exit (1);
}
// allocate the buffer for the time domain window
g_window = new float[g_buffSize];
if ( g_window == NULL ) {
cerr << "Something went wrong when creating the window" << endl;
exit (1);
}
// create a hanning window
make_window( g_window, g_buffSize );
// start the audio stream
audio->startStream();
// test RtAudio functionality for reporting latency.
cout << "stream latency: " << audio->getStreamLatency() << " frames" << endl;
}
catch( RtError & err )
{
err.printMessage();
goto cleanup;
}
// ============
// = GL stuff =
// ============
// initialize GLUT
glutInit( &argc, argv );
// double buffer, use rgb color, enable depth buffer
glutInitDisplayMode( GLUT_DOUBLE | GLUT_RGB | GLUT_DEPTH );
// initialize the window size
glutInitWindowSize( g_width, g_height );
// set the window postion
glutInitWindowPosition( 100, 100 );
// create the window
glutCreateWindow( "Hello GL" );
//glutEnterGameMode();
// set the idle function - called when idle
glutIdleFunc( idleFunc );
// set the display function - called when redrawing
glutDisplayFunc( displayFunc );
// set the reshape function - called when client area changes
glutReshapeFunc( reshapeFunc );
// set the keyboard function - called on keyboard events
glutKeyboardFunc( keyboardFunc );
// set the mouse function - called on mouse stuff
glutMouseFunc( mouseFunc );
// set the special function - called on special keys events (fn, arrows, pgDown, etc)
glutSpecialFunc( specialFunc );
// do our own initialization
initialize();
// let GLUT handle the current thread from here
glutMainLoop();
// if we get here, stop!
try
{
audio->stopStream();
}
catch( RtError & err )
{
err.printMessage();
}
// Clean up
cleanup:
if(audio)
{
audio->closeStream();
delete audio;
}
return 0;
}
//-----------------------------------------------------------------------------
// Name: main( )
// Desc: starting point
//-----------------------------------------------------------------------------
int main( int argc, char ** argv )
{
// Get RtAudio Instance with default API
RtAudio *audio = new RtAudio();
// buffer size
unsigned int buffer_size = 512;
// Output Stream Parameters
RtAudio::StreamParameters outputStreamParams;
outputStreamParams.deviceId = audio->getDefaultOutputDevice();
outputStreamParams.nChannels = 1;
// Input Stream Parameters
RtAudio::StreamParameters inputStreamParams;
inputStreamParams.deviceId = audio->getDefaultInputDevice();
inputStreamParams.nChannels = 1;
// Get RtAudio Stream
try {
audio->openStream(
NULL,
&inputStreamParams,
RTAUDIO_FLOAT32,
MY_FREQ,
&buffer_size,
callback_func,
NULL
);
}
catch(RtError &err) {
err.printMessage();
exit(1);
}
g_bufferSize = buffer_size;
// Samples for Feature Extraction in a Buffer
g_samples = (SAMPLE *)malloc(sizeof(SAMPLE)*g_bufferSize*g_numMaxBuffersToUse);
g_audio_buffer = (SAMPLE *)malloc(sizeof(SAMPLE)*g_bufferSize*g_numMaxBuffersToUse);
g_another_buffer = (SAMPLE *)malloc(sizeof(SAMPLE)*g_bufferSize*g_numMaxBuffersToUse);
g_buffest = (SAMPLE *)malloc(sizeof(SAMPLE)*g_bufferSize*g_numMaxBuffersToUse);
g_residue = (SAMPLE *)malloc(sizeof(SAMPLE)*g_bufferSize*g_numMaxBuffersToUse);
g_coeff = (SAMPLE *)malloc(sizeof(SAMPLE)*g_order);
g_dwt = (SAMPLE *)malloc(sizeof(SAMPLE)*g_bufferSize*g_numMaxBuffersToUse);
// initialize GLUT
glutInit( &argc, argv );
// double buffer, use rgb color, enable depth buffer
glutInitDisplayMode( GLUT_DOUBLE | GLUT_RGB | GLUT_DEPTH );
// initialize the window size
glutInitWindowSize( g_width, g_height );
// set the window postion
glutInitWindowPosition( 100, 100 );
// create the window
glutCreateWindow( "The New File" );
// set the idle function - called when idle
glutIdleFunc( idleFunc );
// set the display function - called when redrawing
glutDisplayFunc( displayFunc );
// set the reshape function - called when client area changes
glutReshapeFunc( reshapeFunc );
// set the keyboard function - called on keyboard events
glutKeyboardFunc( keyboardFunc );
// set the mouse function - called on mouse stuff
glutMouseFunc( mouseFunc );
// do our own initialization
initialize();
// initialize mfcc
initMFCC();
//init lpc
initialize_lpc();
// initialize osc
// Initialize a socket to get a port
g_transmitSocket = new UdpTransmitSocket( IpEndpointName( g_ADDRESS.c_str(), SERVERPORT ) );
// // Set the global sample rate before creating class instances.
// Stk::setSampleRate( 44100.0 );
// // Read In File
// try
// {
// // read the file
// g_fin.openFile( "TomVega.wav" );
// // change the rate
// g_fin.setRate( 1 );
// // normalize the peak
// g_fin.normalize();
// } catch( stk::StkError & e )
// {
// cerr << "baaaaaaaaad..." << endl;
// return 1;
// }
// Start Stream
try {
audio->startStream();
} catch( RtError & err ) {
// do stuff
err.printMessage();
goto cleanup;
}
// let GLUT handle the current thread from here
glutMainLoop();
// if we get here, then stop!
try{
audio->stopStream();
}
catch( RtError & err ) {
// do stuff
err.printMessage();
}
cleanup:
audio->closeStream();
delete audio;
return 0;
}
int main( int argc, char *argv[] )
{
unsigned int bufferFrames, fs, oDevice = 0, iDevice = 0, iOffset = 0, oOffset = 0;
char input;
// minimal command-line checking
if (argc < 3 || argc > 7 ) usage();
RtAudio dac;
if ( dac.getDeviceCount() < 1 ) {
std::cout << "\nNo audio devices found!\n";
exit( 1 );
}
channels = (unsigned int) atoi( argv[1] );
fs = (unsigned int) atoi( argv[2] );
if ( argc > 3 )
iDevice = (unsigned int) atoi( argv[3] );
if ( argc > 4 )
oDevice = (unsigned int) atoi(argv[4]);
if ( argc > 5 )
iOffset = (unsigned int) atoi(argv[5]);
if ( argc > 6 )
oOffset = (unsigned int) atoi(argv[6]);
double *data = (double *) calloc( channels, sizeof( double ) );
// Let RtAudio print messages to stderr.
dac.showWarnings( true );
// Set our stream parameters for output only.
bufferFrames = 256;
RtAudio::StreamParameters oParams, iParams;
oParams.deviceId = oDevice;
oParams.nChannels = channels;
oParams.firstChannel = oOffset;
RtAudio::StreamOptions options;
options.flags = RTAUDIO_HOG_DEVICE;
try {
dac.openStream( &oParams, NULL, RTAUDIO_FLOAT64, fs, &bufferFrames, &sawi, (void *)data, &options );
std::cout << "\nStream latency = " << dac.getStreamLatency() << std::endl;
// Start the stream
dac.startStream();
std::cout << "\nPlaying ... press <enter> to stop.\n";
std::cin.get( input );
// Stop the stream
dac.stopStream();
// Restart again
std::cout << "Press <enter> to restart.\n";
std::cin.get( input );
dac.startStream();
// Test abort function
std::cout << "Playing again ... press <enter> to abort.\n";
std::cin.get( input );
dac.abortStream();
// Restart another time
std::cout << "Press <enter> to restart again.\n";
std::cin.get( input );
dac.startStream();
std::cout << "Playing again ... press <enter> to close the stream.\n";
std::cin.get( input );
}
catch ( RtError& e ) {
e.printMessage();
goto cleanup;
}
if ( dac.isStreamOpen() ) dac.closeStream();
// Test non-interleaved functionality
options.flags = RTAUDIO_NONINTERLEAVED;
try {
dac.openStream( &oParams, NULL, RTAUDIO_FLOAT64, fs, &bufferFrames, &sawni, (void *)data, &options );
std::cout << "Press <enter> to start non-interleaved playback.\n";
std::cin.get( input );
// Start the stream
dac.startStream();
std::cout << "\nPlaying ... press <enter> to stop.\n";
std::cin.get( input );
}
catch ( RtError& e ) {
e.printMessage();
goto cleanup;
}
if ( dac.isStreamOpen() ) dac.closeStream();
// Now open a duplex stream.
unsigned int bufferBytes;
iParams.deviceId = iDevice;
iParams.nChannels = channels;
iParams.firstChannel = iOffset;
options.flags = RTAUDIO_NONINTERLEAVED;
try {
dac.openStream( &oParams, &iParams, RTAUDIO_SINT32, fs, &bufferFrames, &inout, (void *)&bufferBytes, &options );
bufferBytes = bufferFrames * channels * 4;
std::cout << "Press <enter> to start duplex operation.\n";
std::cin.get( input );
// Start the stream
dac.startStream();
std::cout << "\nRunning ... press <enter> to stop.\n";
std::cin.get( input );
// Stop the stream
dac.stopStream();
std::cout << "\nStopped ... press <enter> to restart.\n";
std::cin.get( input );
// Restart the stream
dac.startStream();
std::cout << "\nRunning ... press <enter> to stop.\n";
std::cin.get( input );
}
catch ( RtError& e ) {
e.printMessage();
}
cleanup:
if ( dac.isStreamOpen() ) dac.closeStream();
free( data );
return 0;
}
//-----------------------------------------------------------------------------
// name: main()
// desc: entry point
//-----------------------------------------------------------------------------
int main( int argc, char ** argv )
{
RtMidiIn *midiin = new RtMidiIn();
// Check available ports.
unsigned int nPorts = midiin->getPortCount();
if ( nPorts == 0 ) {
std::cout << "No ports available!\n";
//goto cleanup;
}
midiin->openPort( 0 );
// Set our callback function. This should be done immediately after
// opening the port to avoid having incoming messages written to the
// queue.
midiin->setCallback( &mycallback );
// Don't ignore sysex, timing, or active sensing messages.
midiin->ignoreTypes( false, false, false );
std::cout << "\nReading MIDI input ... press <enter> to quit.\n";
char input;
std::cin.get(input);
// instantiate RtAudio object
RtAudio audio;
// variables
unsigned int bufferBytes = 0;
// frame size
unsigned int numFrames = 512;
// check for audio devices
if( audio.getDeviceCount() < 1 )
{
// nopes
cout << "no audio devices found!" << endl;
exit( 1 );
}
// let RtAudio print messages to stderr.
audio.showWarnings( true );
// set input and output parameters
RtAudio::StreamParameters iParams, oParams;
iParams.deviceId = audio.getDefaultInputDevice();
iParams.nChannels = MY_CHANNELS;
iParams.firstChannel = 0;
oParams.deviceId = audio.getDefaultOutputDevice();
oParams.nChannels = MY_CHANNELS;
oParams.firstChannel = 0;
// create stream options
RtAudio::StreamOptions options;
// go for it
try {
// open a stream
audio.openStream( &oParams, &iParams, MY_FORMAT, MY_SRATE, &numFrames, &callme, NULL, &options );
}
catch( RtError& e )
{
// error!
cout << e.getMessage() << endl;
exit( 1 );
}
// compute
bufferBytes = numFrames * MY_CHANNELS * sizeof(SAMPLE);
// test RtAudio functionality for reporting latency.
cout << "stream latency: " << audio.getStreamLatency() << " frames" << endl;
for( int i = 0; i < MY_NUMSTRINGS; i++ )
{
// intialize
g_ks[i].init( MY_SRATE*2, 440, MY_SRATE );
}
// go for it
try {
// start stream
audio.startStream();
char input;
std::cout << "Press any key to quit ";
std::cin.get(input);
// stop the stream.
audio.stopStream();
}
catch( RtError& e )
{
// print error message
cout << e.getMessage() << endl;
goto cleanup;
}
cleanup:
// close if open
if( audio.isStreamOpen() )
audio.closeStream();
delete midiin;
// done
return 0;
}
int main(int argc, char ** argv)
{
if ( argc==1 )
{
std::cerr << "f " << freq << "\t frequency" << std::endl;
std::cerr << "s " << scale << "\t scale" << std::endl;
std::cerr << "t " << steps << "\t steps" << std::endl;
std::cerr << "r " << rthresh << "\t right thresh" << std::endl;
std::cerr << "f " << lthresh << "\t left thresh" << std::endl;
std::cerr << "i " << inp_audio << "\t inp_audio device id" << std::endl;
std::cerr << "o " << out_audio << "\t out_audio device id" << std::endl;
}
for ( int i = 1; i<argc-1; i++ )
{
if ( !strcmp(argv[i],"f") ) { freq=atoi(argv[++i]); continue; }
if ( !strcmp(argv[i],"l") ) { lthresh=atoi(argv[++i]); continue; }
if ( !strcmp(argv[i],"r") ) { rthresh=atoi(argv[++i]); continue; }
if ( !strcmp(argv[i],"s") ) { scale=atoi(argv[++i]); continue; }
if ( !strcmp(argv[i],"t") ) { steps=atoi(argv[++i]); continue; }
if ( !strcmp(argv[i],"i") ) { inp_audio=atoi(argv[++i]); continue; }
if ( !strcmp(argv[i],"o") ) { out_audio=atoi(argv[++i]); continue; }
}
unsigned int bufferFrames = NUM_FREQ*2;
Data data;
data.fft.Init(bufferFrames, NUM_FREQ, 1, 2.5f);
RtAudio adac;
if ( adac.getDeviceCount() < 1 )
{
std::cout << "\nNo audio devices found!\n";
exit( 0 );
}
RtAudio::StreamParameters iParams, oParams;
iParams.deviceId = inp_audio; // <----------- put them on
iParams.nChannels = 1; // different devices
oParams.deviceId = out_audio; // <----------- for duplex mode
oParams.nChannels = 1; //
try
{
adac.openStream( &oParams, &iParams, RTAUDIO_FLOAT32, 44100, &bufferFrames, &inout, &data );
}
catch ( RtError& e )
{
e.printMessage();
exit( 0 );
}
try
{
adac.startStream();
main_init();
int k = 0;
int kcen = 658;
int lmean=0,rmean=0;
while(true)
{
// find the global max:
float m = 0;
int mi=-1;
for ( int i=64; i<NUM_FREQ; i++ ) // skip low freq
{
if ( data.freqs[i] > m )
{
m = data.freqs[i];
mi = i;
}
}
kcen = ipol(kcen,mi,4);
// get the mean of the lower and the higher neighbours
int lsum=0,rsum=0;
for( int i=-steps; i<-2; i++ )
{
lsum += data.value(kcen+i,scale);
}
for( int i=2; i<steps; i++ )
{
rsum += data.value(kcen+i,scale);
}
rsum /= (steps-2);
lsum /= (steps-2);
int rd = rsum-rmean;
int ld = lsum-lmean;
lmean=ipol(lmean,lsum,256);
rmean=ipol(rmean,rsum,256);
int lc=' ',rc=' ';
if ( rd>rthresh )
rc='r';
if ( ld>lthresh )
lc='l';
//if ( ld>lthresh || ld>lthresh )
std::cerr << char(lc) << " " << char(rc) << std::endl;
main_idle(data,kcen);
}
// Stop the stream.
adac.stopStream();
}
catch ( RtError& e )
{
e.printMessage();
goto cleanup;
}
cleanup:
if ( adac.isStreamOpen() ) adac.closeStream();
return 0;
}
int main(int argc, char *argv[])
{
// Minimal command-line checking.
if ( argc < 3 || argc > 4 ) usage();
// Set the global sample rate before creating class instances.
Stk::setSampleRate( (StkFloat) atof( argv[2] ) );
// Initialize our WvIn and RtAudio pointers.
RtAudio dac;
FileWvIn input;
FileLoop inputLoop;
// Try to load the soundfile.
try {
input.openFile( argv[1] );
inputLoop.openFile( argv[1] );
}
catch ( StkError & ) {
exit( 1 );
}
// Set input read rate based on the default STK sample rate.
double rate = 1.0;
rate = input.getFileRate() / Stk::sampleRate();
rate = inputLoop.getFileRate() / Stk::sampleRate();
if ( argc == 4 ) rate *= atof( argv[3] );
input.setRate( rate );
input.ignoreSampleRateChange();
// Find out how many channels we have.
int channels = input.channelsOut();
// Figure out how many bytes in an StkFloat and setup the RtAudio stream.
RtAudio::StreamParameters parameters;
parameters.deviceId = dac.getDefaultOutputDevice();
parameters.nChannels = channels;
RtAudioFormat format = ( sizeof(StkFloat) == 8 ) ? RTAUDIO_FLOAT64 : RTAUDIO_FLOAT32;
unsigned int bufferFrames = RT_BUFFER_SIZE;
try {
dac.openStream( ¶meters, NULL, format, (unsigned int)Stk::sampleRate(), &bufferFrames, &tick, (void *)&inputLoop );
}
catch ( RtAudioError &error ) {
error.printMessage();
goto cleanup;
}
// Install an interrupt handler function.
(void) signal(SIGINT, finish);
// Resize the StkFrames object appropriately.
frames.resize( bufferFrames, channels );
try {
dac.startStream();
}
catch ( RtAudioError &error ) {
error.printMessage();
goto cleanup;
}
// Block waiting until callback signals done.
while ( !done )
Stk::sleep( 100 );
// By returning a non-zero value in the callback above, the stream
// is automatically stopped. But we should still close it.
try {
dac.closeStream();
}
catch ( RtAudioError &error ) {
error.printMessage();
}
cleanup:
return 0;
}