int main()
{
RtAudio dac;
if ( dac.getDeviceCount() == 0 ) exit( 0 );
RtAudio::StreamParameters parameters;
parameters.deviceId = dac.getDefaultOutputDevice();
parameters.nChannels = 2;
unsigned int sampleRate = 44100;
unsigned int bufferFrames = 256; // 256 sample frames
RtAudio::StreamOptions options;
options.flags = RTAUDIO_NONINTERLEAVED;
try {
dac.openStream( ¶meters, NULL, RTAUDIO_FLOAT32,
sampleRate, &bufferFrames, &myCallback, NULL, &options );
}
catch ( RtError& e ) {
std::cout << '\n' << e.getMessage() << '\n' << std::endl;
exit( 0 );
}
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.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;
}
/* returns 0 on failure */
int
start_audio(AudioCallback _callback, int sample_rate, void *data)
{
if(audio.getDeviceCount() < 1) {
std::cout << "No audio devices found!\n";
return 0;
}
RtAudio::StreamParameters iparams, oparams;
/* configure input (microphone) */
iparams.deviceId = audio.getDefaultInputDevice();
iparams.nChannels = 1;
iparams.firstChannel = 0;
/* configure output */
oparams.deviceId = audio.getDefaultOutputDevice();
oparams.nChannels = 2;
oparams.firstChannel = 0;
unsigned int bufferFrames = 256;
callback = _callback;
try {
audio.openStream(&oparams, &iparams, RTAUDIO_FLOAT64 /* double */, sample_rate, &bufferFrames, &render, data);
audio.startStream();
} catch(RtError& e) {
e.printMessage();
return 0;
}
return 1;
}
int main(int argc, const char * argv[])
{
RtAudio dac;
RtAudio::StreamParameters rtParams;
rtParams.deviceId = dac.getDefaultOutputDevice();
rtParams.nChannels = nChannels;
#if RASPI
unsigned int sampleRate = 22000;
#else
unsigned int sampleRate = 44100;
#endif
unsigned int bufferFrames = 512; // 512 sample frames
Tonic::setSampleRate(sampleRate);
std::vector<Synth> synths;
synths.push_back(*new BassDrum());
synths.push_back(*new Snare());
synths.push_back(*new HiHat());
synths.push_back(*new Funky());
// Test write pattern
DrumMachine *drumMachine = new DrumMachine(synths);
drumMachine->loadPattern(0);
ControlMetro metro = ControlMetro().bpm(480);
ControlCallback drumMachineTick = ControlCallback(&mixer, [&](ControlGeneratorOutput output){
drumMachine->tick();
}).input(metro);
Generator mixedSignal;
for(int i = 0; i < NUM_TRACKS; i++)
{
mixedSignal = mixedSignal + synths[i];
}
mixer.setOutputGen(mixedSignal);
try
{
dac.openStream( &rtParams, NULL, RTAUDIO_FLOAT32, sampleRate, &bufferFrames, &renderCallback, NULL, NULL );
dac.startStream();
// Send a pointer to our global drumMachine instance
// to the serial communications layer.
listenForMessages( drumMachine );
dac.stopStream();
}
catch ( RtError& e )
{
std::cout << '\n' << e.getMessage() << '\n' << std::endl;
exit( 0 );
}
return 0;
}
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;
}
int main(int argc, char** argv)
{
if (argc != 2) {
printf("Usage: synth file.sf2\n");
exit(0);
}
LightFluidSynth *usynth;
usynth = new LightFluidSynth();
usynth->loadSF2(argv[1]);
// usynth->loadSF2("tim.sf2");
RtMidiIn *midiIn = new RtMidiIn();
if (midiIn->getPortCount() == 0) {
std::cout << "No MIDI ports available!\n";
}
midiIn->openPort(0);
midiIn->setCallback( &midiCallback, (void *)usynth );
midiIn->ignoreTypes( false, false, false );
// RtAudio dac(RtAudio::LINUX_PULSE);
RtAudio dac;
RtAudio::StreamParameters rtParams;
// Determine the number of devices available
unsigned int devices = dac.getDeviceCount();
// Scan through devices for various capabilities
RtAudio::DeviceInfo info;
for ( unsigned int i = 0; i < devices; i++ ) {
info = dac.getDeviceInfo( i );
if ( info.probed == true ) {
std::cout << "device " << " = " << info.name;
std::cout << ": maximum output channels = " << info.outputChannels << "\n";
}
}
// rtParams.deviceId = 3;
rtParams.deviceId = dac.getDefaultOutputDevice();
rtParams.nChannels = 2;
unsigned int bufferFrames = FRAME_SIZE;
RtAudio::StreamOptions options;
options.flags = RTAUDIO_SCHEDULE_REALTIME;
dac.openStream( &rtParams, NULL, AUDIO_FORMAT, SAMPLE_RATE, &bufferFrames, &audioCallback, (void *)usynth, &options );
dac.startStream();
printf("\n\nPress Enter to stop\n\n");
cin.get();
dac.stopStream();
delete(usynth);
return 0;
}
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()
{
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 open_output(va_list args)
{
RtAudio::StreamParameters parameters;
parameters.deviceId = vessl_out.getDefaultOutputDevice();
parameters.nChannels = 2;
parameters.firstChannel = 0;
RtAudio::StreamOptions options;
if ( vessl_out_bufferFrames == 0 )
{
options.flags |= RTAUDIO_MINIMIZE_LATENCY;
}
options.numberOfBuffers = 2;
parameters.deviceId = vessl_out.getDefaultOutputDevice();
try
{
vessl_out.openStream( ¶meters, NULL, RTAUDIO_FLOAT32, vessl_out_sampleRate, &vessl_out_bufferFrames, &output_render_callback, (void*)0, &options );
}
catch( RtAudioError& e )
{
printf("[vessl] FAILED TO OPEN OUTPUT: %s\n", e.getMessage().c_str());
}
if ( vessl_out.isStreamOpen() )
{
try
{
vessl_out.startStream();
}
catch( RtAudioError& e )
{
printf("[vessl] FAILED TO START OUTPUT: %s\n", e.getMessage().c_str());
}
}
}
void init(){
unsigned int sampleRate = 44100;
unsigned int bufferFrames = 128;
// init pd
if(!lpd.init(0, 2, sampleRate)) {
std::cerr << "Could not init pd" << std::endl;
exit(1);
}
// receive messages from pd
lpd.setReceiver(&pdObject);
lpd.subscribe("cursor");
// send DSP 1 message to pd
lpd.computeAudio(true);
// load the patch
pd::Patch patch = lpd.openPatch("test.pd", "./pd");
std::cout << patch << std::endl;
// use the RtAudio API to connect to the default audio device
if(audio.getDeviceCount()==0){
std::cout << "There are no available sound devices." << std::endl;
exit(1);
}
RtAudio::StreamParameters parameters;
parameters.deviceId = audio.getDefaultOutputDevice();
parameters.nChannels = 2;
RtAudio::StreamOptions options;
options.streamName = "libpd rtaudio test";
options.flags = RTAUDIO_SCHEDULE_REALTIME;
if(audio.getCurrentApi() != RtAudio::MACOSX_CORE) {
options.flags |= RTAUDIO_MINIMIZE_LATENCY; // CoreAudio doesn't seem to like this
}
try {
audio.openStream( ¶meters, NULL, RTAUDIO_FLOAT32, sampleRate, &bufferFrames, &audioCallback, NULL, &options );
audio.startStream();
}
catch(RtAudioError& e) {
std::cerr << e.getMessage() << std::endl;
exit(1);
}
}
av_Audio * av_audio_get() {
static bool initialized = false;
if (!initialized) {
initialized = true;
rta.showWarnings( true );
// defaults:
audio.samplerate = 44100;
audio.blocksize = 256;
audio.inchannels = 2;
audio.outchannels = 2;
audio.time = 0;
audio.lag = 0.04;
audio.indevice = rta.getDefaultInputDevice();
audio.outdevice = rta.getDefaultOutputDevice();
/*
audio.msgbuffer.size = AV_AUDIO_MSGBUFFER_SIZE_DEFAULT;
audio.msgbuffer.read = 0;
audio.msgbuffer.write = 0;
audio.msgbuffer.data = (unsigned char *)malloc(audio.msgbuffer.size);
*/
audio.onframes = 0;
// one second of ringbuffer:
int blockspersecond = audio.samplerate / audio.blocksize;
audio.blocks = blockspersecond + 1;
audio.blockstep = audio.blocksize * audio.outchannels;
int len = audio.blockstep * audio.blocks;
audio.buffer = (float *)calloc(len, sizeof(float));
audio.blockread = 0;
audio.blockwrite = 0;
printf("audio initialized\n");
//AL = lua_open(); //av_init_lua();
// unique to audio thread:
//if (luaL_dostring(AL, "require 'audioprocess'")) {
// printf("error: %s\n", lua_tostring(AL, -1));
// initialized = false;
//}
}
return &audio;
}
int startAudio() {
// Determine the number of devices available
unsigned int devices = audio.getDeviceCount();
if(devices==0) {
printf("please run 'sudo modprobe snd_bcm2835' to enable the alsa driver\n");
return 1;
}
// Scan through devices for various capabilities
RtAudio::DeviceInfo info;
for ( unsigned int i=0; i<devices; i++ ) {
info = audio.getDeviceInfo( i );
if ( info.probed == true ) {
// Print, for example, the maximum number of output channels for each device
std::cout << "device = " << i;
std::cout << ": maximum output channels = " << info.outputChannels << "\n";
}
}
RtAudio::StreamParameters parameters;
parameters.deviceId = audio.getDefaultOutputDevice();
parameters.nChannels = 2;
parameters.firstChannel = 0;
unsigned int sampleRate = SAMPLERATE;
unsigned int bufferFrames = BUFFERSIZE;
double data[2];
try {
audio.openStream( ¶meters, NULL, RTAUDIO_FLOAT32,
sampleRate, &bufferFrames, &audioCallback, (void *)&data );
audio.startStream();
} catch ( RtError& e ) {
e.printMessage();
return 1;
}
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()
{
// 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( 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[])
{
// 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;
}
//-----------------------------------------------------------------------------
// name: main()
// desc: entry point
//-----------------------------------------------------------------------------
int main( int argc, char ** argv )
{
// instantiate RtAudio object
RtAudio audio;
// variables
unsigned int bufferBytes = 0;
// frame size
unsigned int bufferFrames = 512;
// check for audio devices
if( audio.getDeviceCount() < 1 )
{
// nopes
cout << "no audio devices found!" << endl;
exit( 1 );
}
// initialize GLUT
glutInit( &argc, argv );
// init gfx
initGfx();
// 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, &bufferFrames, &callme, (void *)&bufferBytes, &options );
}
catch( RtError& e )
{
// error!
cout << e.getMessage() << endl;
exit( 1 );
}
// compute
bufferBytes = bufferFrames * MY_CHANNELS * sizeof(SAMPLE);
// allocate global buffer
g_bufferSize = bufferFrames;
g_buffer = new SAMPLE[g_bufferSize];
memset( g_buffer, 0, sizeof(SAMPLE)*g_bufferSize );
// go for it
try {
// start stream
audio.startStream();
// let GLUT handle the current thread from here
glutMainLoop();
// 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();
// done
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(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)
{
//parse tempo
if (argc>2)
{
cerr<<"Error in arguments\n";
printHelp();
exit(1);
}
else if (argc==2)
{
g_tempo = atoi(argv[1]);
if (g_tempo<40 && g_tempo>200)
{
cerr<<"Tempo out of bounds!\n";
printHelp();
exit(1);
}
tempoChange();
}
// set up fluid synth stuff
// TODO: error checking!!!!
g_settings = new_fluid_settings();
g_synth = new_fluid_synth( g_settings );
g_metronome = new_fluid_synth( g_settings );
//fluid_player_t* player;
//player = new_fluid_player(g_synth);
//fluid_player_add(player, "backing.mid");
//fluid_player_play(player);
if (fluid_synth_sfload(g_synth, "piano.sf2", 1) == -1)
{
cerr << "Error loading sound font" << endl;
exit(1);
}
if (fluid_synth_sfload(g_metronome, "drum.sf2", 1) == -1)
{
cerr << "Error loading sound font" << endl;
exit(1);
}
// RtAudio config + init
// pointer to RtAudio object
RtMidiIn * midiin = NULL;
RtAudio * audio = NULL;
unsigned int bufferSize = 512;//g_sixteenth/100;
// MIDI config + init
try
{
midiin = new RtMidiIn();
}
catch( RtError & err ) {
err.printMessage();
// goto cleanup;
}
// Check available ports.
if ( midiin->getPortCount() == 0 )
{
std::cout << "No ports available!\n";
// goto cleanup;
}
// use the first available port
if ( midiin->getPortCount() > 2)
midiin->openPort( 1 );
else
midiin->openPort( 0 );
// set midi callback
midiin->setCallback( &midi_callback );
// Don't ignore sysex, timing, or active sensing messages.
midiin->ignoreTypes( false, false, false );
// create the object
try
{
audio = new RtAudio();
cerr << "buffer size: " << bufferSize << endl;
}
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 = 2;
oParams.firstChannel = 0;
// create stream options
RtAudio::StreamOptions options;
// set the callback and start stream
try
{
audio->openStream( &oParams, &iParams, RTAUDIO_FLOAT32, MY_SRATE, &bufferSize, &audioCallback, NULL, &options);
audio->startStream();
// test RtAudio functionality for reporting latency.
cout << "stream latency: " << audio->getStreamLatency() << " frames" << endl;
}
catch( RtError & err )
{
err.printMessage();
goto cleanup;
}
// wait for user input
cout << "Type CTRL+C to quit:";
//initialize graphics
gfxInit(&argc,argv);
// if we get here, stop!
try
{
audio->stopStream();
}
catch( RtError & err )
{
err.printMessage();
}
// Clean up
cleanup:
if(audio)
{
audio->closeStream();
delete audio;
}
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 );
// Depending on how you compile STK, you may need to explicitly set
// the path to the rawwave directory.
Stk::setRawwavePath( "../../rawwaves/" );
// 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 *) );
// Instantiate the instrument(s) type from the command-line argument
// (in utilities.cpp).
data.nVoices = countVoices( argc, argv );
data.instrument = (Instrmnt **) calloc( data.nVoices, sizeof(Instrmnt *) );
data.currentVoice = voiceByName( argv[1], &data.instrument[0] );
if ( data.currentVoice < 0 ) {
free( data.wvout );
free( data.instrument );
usage(argv[0]);
}
// If there was no error allocating the first voice, we should be fine for more.
for ( i=1; i<data.nVoices; i++ )
voiceByName( argv[1], &data.instrument[i] );
data.voicer = (Voicer *) new Voicer( 0.0 );
for ( i=0; i<data.nVoices; i++ )
data.voicer->addInstrument( data.instrument[i] );
// 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);
// 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.voicer;
for ( i=0; i<data.nVoices; i++ ) delete data.instrument[i];
free( data.instrument );
std::cout << "\nStk demo finished ... goodbye.\n\n";
return 0;
}
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;
if ( device == 0 )
oParams.deviceId = dac.getDefaultOutputDevice();
data.channels = channels;
try {
dac.openStream( &oParams, NULL, FORMAT, fs, &bufferFrames, &output, (void *)&data );
dac.startStream();
}
catch ( RtAudioError& 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;
}
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( 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 = 512;
RtAudio::StreamParameters oParams;
oParams.deviceId = device;
oParams.nChannels = channels;
oParams.firstChannel = offset;
if ( device == 0 )
oParams.deviceId = dac.getDefaultOutputDevice();
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, &errorCallback );
dac.startStream();
}
catch ( RtAudioError& 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 ( RtAudioError& e ) {
e.printMessage();
}
}
cleanup:
if ( dac.isStreamOpen() ) dac.closeStream();
free( data );
return 0;
}
unsigned int Audio::getDefaultOutputDevice()
{
return dac.getDefaultOutputDevice();
}
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 )
{
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;
}
// ========
// = 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;
}
void init(int argc, char **argv) {
/////
theString = new StringModel ( 1000, 0.5, 0.99999, 8 );
// *** test the rtaudio callback
if ( dac.getDeviceCount() < 1 ) {
std::cout << "\nNo audio devices found!\n";
exit( 0 );
}
parameters.deviceId = dac.getDefaultOutputDevice();
parameters.nChannels = 2;
parameters.firstChannel = 0;
sampleRate = 44100;
bufferFrames = 256; // 256 sample frames
try {
dac.openStream ( ¶meters,
NULL,
RTAUDIO_FLOAT32,
sampleRate,
&bufferFrames,
StringModel::audioCallback,
(void *)theString );
dac.startStream();
}
catch ( RtError& e ) {
std::cout << "\nexception on dac:\n";
e.printMessage();
exit(0);
}
//////
GLfloat pos[] = {5.0, 5.0, 10.0, 0.0};
glLightfv(GL_LIGHT0, GL_POSITION, pos);
glEnable(GL_CULL_FACE);
glEnable(GL_LIGHTING);
glEnable(GL_LIGHT0);
glEnable(GL_DEPTH_TEST);
glEnable(GL_NORMALIZE);
glEnable(GL_COLOR_MATERIAL);
glEnable(GL_BLEND);
glBlendFunc(GL_SRC_ALPHA, GL_ONE_MINUS_SRC_ALPHA);
glEnable(GL_FOG);
float FogCol[3]={0.0,0.0,0.0};
glFogfv(GL_FOG_COLOR,FogCol);
glFogi(GL_FOG_MODE, GL_LINEAR);
glFogf(GL_FOG_START, 10.0f);
glFogf(GL_FOG_END, 40.f);
glClearColor (0.0, 0.0, 0.0, 0.0);
set_to_ident(g_trackball_transform);
std::cout << "\nPlaying ... press \n";
std::cout << "t to tighten\n";
std::cout << "l to loosen\n";
std::cout << "p to pluck\n";
std::cout << "r to reset\n";
std::cout << "d to dump velocities\n";
std::cout << "f/F to change vibrator freq\n";
std::cout << "ESC to quit.\n";
}