void start(unsigned int bufferFrames = 512, unsigned int sampleRate = 44100) {
this->bufferFrames = bufferFrames;
this->sampleRate = sampleRate;
if (dac.getDeviceCount() < 1) {
std::cout << "\nNo audio devices found!\n";
exit(0);
}
parameters.deviceId = (id == 1) ? 0 : 1;
RtAudio::DeviceInfo info;
info = dac.getDeviceInfo(parameters.deviceId);
std::cout << "device = " << info.name << std::endl;
//parameters.deviceId = dac.getDefaultOutputDevice();
parameters.nChannels = 2;
parameters.firstChannel = 0;
//RtApiAlsa::getDeviceInfo: snd_pcm_open error for device (default), No such file or directory.
try {
unsigned int got = bufferFrames;
dac.openStream(¶meters, NULL, RTAUDIO_SINT16, sampleRate, &got, &process, (void *)&data);
//dac.openStream(¶meters, NULL, RTAUDIO_FLOAT32, sampleRate, &got, &process, (void *)&data);
dac.startStream();
std::cout << "requested " << bufferFrames << " but got " << got << std::endl;
} catch (RtAudioError &e) {
e.printMessage();
exit(0);
}
}
void slgAudio::info(){
RtAudio *audioTemp = NULL;
audioTemp = new RtAudio();
unsigned int devices = audioTemp->getDeviceCount();
RtAudio::DeviceInfo info;
for (int i=0;i<devices;i++){
info = audioTemp->getDeviceInfo(i);
// std::cout<<"default input: "<<m_audio->getDefaultInputDevice()<<std::endl;
// std::cout<<"default output: "<<m_audio->getDefaultOutputDevice()<<std::endl;
if (info.probed ==true){
std::cout<<"----------------------------- Device "<<i<<" ---------------------------"<<std::endl;
if (info.isDefaultInput)
std::cout << "--Default Input"<<std::endl;
if (info.isDefaultOutput)
std::cout << "--Default Output"<<std::endl;
std::cout << "Name = " << info.name << '\n';
std::cout << "Max Input Channels = " << info.inputChannels << '\n';
std::cout << "Max Output Channels = " << info.outputChannels << '\n';
std::cout << "Max Duplex Channels = " << info.duplexChannels << '\n';
}
}
delete audioTemp;
audioTemp = NULL;
}
GOrgueSoundPort* GOrgueSoundRtPort::create(GOrgueSound* sound, wxString name)
{
try
{
std::vector<RtAudio::Api> rtaudio_apis;
RtAudio::getCompiledApi(rtaudio_apis);
for (unsigned k = 0; k < rtaudio_apis.size(); k++)
{
RtAudio* audioDevice = 0;
try
{
audioDevice = new RtAudio(rtaudio_apis[k]);
for (unsigned i = 0; i < audioDevice->getDeviceCount(); i++)
if (getName(rtaudio_apis[k], audioDevice, i) == name)
return new GOrgueSoundRtPort(sound, name, rtaudio_apis[k]);
}
catch (RtAudioError &e)
{
wxString error = wxString::FromAscii(e.getMessage().c_str());
wxLogError(_("RtAudio error: %s"), error.c_str());
}
if (audioDevice)
delete audioDevice;
}
}
catch (RtAudioError &e)
{
wxString error = wxString::FromAscii(e.getMessage().c_str());
wxLogError(_("RtAudio error: %s"), error.c_str());
}
return NULL;
}
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;
}
/* 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;
}
bool Audio::openAudioInputDevice(unsigned int device)
{
RtAudio::StreamParameters p;
double data[2];
unsigned int num = 0;
if (inIsOpened) return false;
for (unsigned int i=0; i<adc.getDeviceCount(); i++)
{
if (adc.getDeviceInfo(i).inputChannels)
{
if (device == num)
{
p.deviceId = i;
break;
}
num++;
}
}
p.firstChannel = 0;
p.nChannels = 2;
try
{
adc.openStream(NULL, &p, RTAUDIO_FLOAT32, sampleRate, &bufferFrames, &audioInputCallback, (void*)&data);
adc.startStream();
}
catch (RtAudioError& e)
{
inError = e.getMessage();
return false;
}
inBuffer=(float*)calloc(bufferFrames, sizeof(float));
inIsOpened=true;
return true;
}
//---------------------------------------------------------
void ofSoundStreamListDevices(){
RtAudio *audioTemp = 0;
try {
audioTemp = new RtAudio();
} catch (RtError &error) {
error.printMessage();
}
int devices = audioTemp->getDeviceCount();
RtAudio::DeviceInfo info;
for (int i=0; i< devices; i++) {
try {
info = audioTemp->getDeviceInfo(i);
} catch (RtError &error) {
error.printMessage();
break;
}
std::cout << "device = " << i << " (" << info.name << ")\n";
if (info.isDefaultInput) std::cout << "----* default ----* \n";
std::cout << "maximum output channels = " << info.outputChannels << "\n";
std::cout << "maximum input channels = " << info.inputChannels << "\n";
std::cout << "-----------------------------------------\n";
}
delete audioTemp;
}
bool DeviceManager::getAudioDevices(bool input, std::vector<Device>& devs)
{
devs.clear();
#if defined(ANDROID)
// Under Android, we don't access the device file directly.
// Arbitrary use 0 for the mic and 1 for the output.
// These ids are used in MediaEngine::SetSoundDevices(in, out);
// The strings are for human consumption.
if (input) {
devs.push_back(Device("audioin", "audiorecord", 0));
} else {
devs.push_back(Device("audioout", "audiotrack", 1));
}
return true;
#elif defined(HAVE_RTAUDIO)
// Since we are using RtAudio for audio capture it's best to
// use RtAudio to enumerate devices to ensure indexes match.
RtAudio audio;
// Determine the number of devices available
auto ndevices = audio.getDeviceCount();
TraceS(this) << "Get audio devices: " << ndevices << endl;
// Scan through devices for various capabilities
RtAudio::DeviceInfo info;
for (unsigned i = 0; i <= ndevices; i++) {
try {
info = audio.getDeviceInfo(i); // may throw RtAudioError
TraceS(this) << "Device:"
<< "\n\tName: " << info.name
<< "\n\tOutput Channels: " << info.outputChannels
<< "\n\tInput Channels: " << info.inputChannels
<< "\n\tDuplex Channels: " << info.duplexChannels
<< "\n\tDefault Output: " << info.isDefaultOutput
<< "\n\tDefault Input: " << info.isDefaultInput
<< "\n\tProbed: " << info.probed
<< endl;
if (info.probed == true && (
(input && info.inputChannels > 0) ||
(!input && info.outputChannels > 0))) {
TraceS(this) << "Adding device: " << info.name << endl;
Device dev((input ? "audioin" : "audioout"), i, info.name, "",
(input ? info.isDefaultInput : info.isDefaultOutput));
devs.push_back(dev);
}
}
catch (RtAudioError& e) {
ErrorS(this) << "Cannot probe audio device: " << e.getMessage() << endl;
}
}
return filterDevices(devs, kFilteredAudioDevicesName);
#endif
}
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 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;
}
// サポートしているデバイスリストを表示
void listDevices() {
RtAudio audio;
unsigned int devices = audio.getDeviceCount();
RtAudio::DeviceInfo info;
for(unsigned int i=0; i<devices; i++) {
info = audio.getDeviceInfo(i);
std::cout << "============================" << std::endl;
std::cout << "\nDevide ID:" << i << std::endl;
std::cout << "Name:" << info.name << std::endl;
if ( info.probed == false )
std::cout << "Probe Status = UNsuccessful\n";
else {
std::cout << "Probe Status = Successful\n";
std::cout << "Output Channels = " << info.outputChannels << '\n';
std::cout << "Input Channels = " << info.inputChannels << '\n';
std::cout << "Duplex Channels = " << info.duplexChannels << '\n';
if ( info.isDefaultOutput ) {
std::cout << "This is the default output device.\n";
} else {
std::cout << "This is NOT the default output device.\n";
}
if ( info.isDefaultInput ) { std::cout << "This is the default input device.\n";
} else {
std::cout << "This is NOT the default input device.\n";
}
if ( info.nativeFormats == 0 ) {
std::cout << "No natively supported data formats(?)!";
} else {
std::cout << "Natively supported data formats:\n";
if ( info.nativeFormats & RTAUDIO_SINT8 )
std::cout << " 8-bit int\n";
if ( info.nativeFormats & RTAUDIO_SINT16 )
std::cout << " 16-bit int\n";
if ( info.nativeFormats & RTAUDIO_SINT24 )
std::cout << " 24-bit int\n";
if ( info.nativeFormats & RTAUDIO_SINT32 )
std::cout << " 32-bit int\n";
if ( info.nativeFormats & RTAUDIO_FLOAT32 )
std::cout << " 32-bit float\n";
if ( info.nativeFormats & RTAUDIO_FLOAT64 )
std::cout << " 64-bit float\n";
}
if ( info.sampleRates.size() < 1 ) {
std::cout << "No supported sample rates found!";
} else {
std::cout << "Supported sample rates = ";
for (unsigned int j=0; j<info.sampleRates.size(); j++)
std::cout << info.sampleRates[j] << " ";
}
std::cout << std::endl;
}
}
}
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;
}
unsigned int Audio::getAudioInputCount()
{
unsigned int num = 0;
unsigned int deviceCount=adc.getDeviceCount();
for (unsigned int i=0; i<deviceCount; i++)
{
if(adc.getDeviceInfo(i).inputChannels) num++;
}
return num;
}
QString Audio::getInDeviceName(unsigned int device)
{
unsigned int num = 0;
for (unsigned int i=0; i<adc.getDeviceCount(); i++)
{
if (adc.getDeviceInfo(i).inputChannels)
{
if (device == num) return QString::fromStdString(adc.getDeviceInfo(i).name);
num++;
}
}
return QObject::tr("Unknown audio input device");
}
void AudioThread::enumerateDevices(std::vector<RtAudio::DeviceInfo> &devs) {
RtAudio endac;
int numDevices = endac.getDeviceCount();
for (int i = 0; i < numDevices; i++) {
RtAudio::DeviceInfo info = endac.getDeviceInfo(i);
devs.push_back(info);
std::cout << std::endl;
std::cout << "Audio Device #" << i << " " << info.name << std::endl;
std::cout << "\tDefault Output? " << (info.isDefaultOutput ? "Yes" : "No") << std::endl;
std::cout << "\tDefault Input? " << (info.isDefaultOutput ? "Yes" : "No") << std::endl;
std::cout << "\tInput channels: " << info.inputChannels << std::endl;
std::cout << "\tOutput channels: " << info.outputChannels << std::endl;
std::cout << "\tDuplex channels: " << info.duplexChannels << std::endl;
std::cout << "\t" << "Native formats:" << std::endl;
RtAudioFormat nFormats = info.nativeFormats;
if (nFormats & RTAUDIO_SINT8) {
std::cout << "\t\t8-bit signed integer." << std::endl;
}
if (nFormats & RTAUDIO_SINT16) {
std::cout << "\t\t16-bit signed integer." << std::endl;
}
if (nFormats & RTAUDIO_SINT24) {
std::cout << "\t\t24-bit signed integer." << std::endl;
}
if (nFormats & RTAUDIO_SINT32) {
std::cout << "\t\t32-bit signed integer." << std::endl;
}
if (nFormats & RTAUDIO_FLOAT32) {
std::cout << "\t\t32-bit float normalized between plus/minus 1.0." << std::endl;
}
if (nFormats & RTAUDIO_FLOAT64) {
std::cout << "\t\t32-bit float normalized between plus/minus 1.0." << std::endl;
}
std::vector<unsigned int>::iterator srate;
std::cout << "\t" << "Supported sample rates:" << std::endl;
for (srate = info.sampleRates.begin(); srate != info.sampleRates.end(); srate++) {
std::cout << "\t\t" << (*srate) << "hz" << std::endl;
}
std::cout << std::endl;
}
}
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;
}
//-----------------------------------------------------------------------------
// name: probe()
// desc: ...
//-----------------------------------------------------------------------------
void Digitalio::probe()
{
#ifndef __DISABLE_RTAUDIO__
RtAudio * rta = NULL;
RtAudio::DeviceInfo info;
// allocate RtAudio
try { rta = new RtAudio( ); }
catch( RtError err )
{
// problem finding audio devices, most likely
EM_error2b( 0, "%s", err.getMessage().c_str() );
return;
}
// get count
int devices = rta->getDeviceCount();
EM_error2b( 0, "found %d device(s) ...", devices );
// EM_error2( 0, "--------------------------" );
EM_reset_msg();
// loop
for( int i = 0; i < devices; i++ )
{
try { info = rta->getDeviceInfo(i); }
catch( RtError & error )
{
error.printMessage();
break;
}
// print
EM_error2b( 0, "------( audio device: %d )---------------", i+1 );
print( info );
// skip
if( i < devices ) EM_error2( 0, "" );
EM_reset_msg();
}
delete rta;
#endif // __DISABLE_RTAUDIO__
return;
}
QHash<int, QString> BleAudioCapture::availableDevices()
{
RtAudio rtAudio;
int deviceCount = rtAudio.getDeviceCount();
RtAudio::DeviceInfo info;
QHash<int, QString> devices;
for (int i = 0; i < deviceCount; ++i) {
info = rtAudio.getDeviceInfo(i);
if (info.inputChannels > 0) {
devices.insert(i, QString::fromStdString(info.name));
}
}
return devices;
}
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);
}
}
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;
}
void initaudio() {
using namespace std;
/*
vector<RtAudio::Api> apis;
RtAudio::getCompiledApi(apis);
cout << apis.size() << endl;
for (size_t i = 0; i < apis.size(); ++i) {
cout << apis[i] << endl;
}
*/
int ndevices = adc.getDeviceCount();
if ( ndevices < 1 ) {
std::cout << "\nNo audio devices found!\n";
return;
}
cout << endl;
for (int i = 0; i < ndevices; ++i) {
RtAudio::DeviceInfo devinfo;
devinfo = adc.getDeviceInfo(i);
cout << i << " " << devinfo.name << " " << devinfo.inputChannels << endl;
}
RtAudio::StreamParameters parameters;
parameters.deviceId = 2;
parameters.nChannels = 1;
parameters.firstChannel = 0;
unsigned int sampleRate = 1200;
unsigned int bufferFrames = 8; // 256 sample frames
try {
adc.openStream( NULL, ¶meters, RTAUDIO_SINT16,
sampleRate, &bufferFrames, &record );
adc.startStream();
}
catch ( exception& e ) {
std::cout << "EXCEPTION" << endl << e.what();
return;
}
}
void GOrgueSoundRtPort::addDevices(std::vector<GOrgueSoundDevInfo>& result)
{
try
{
std::vector<RtAudio::Api> rtaudio_apis;
RtAudio::getCompiledApi(rtaudio_apis);
for (unsigned k = 0; k < rtaudio_apis.size(); k++)
{
RtAudio* audioDevice = 0;
try
{
audioDevice = new RtAudio(rtaudio_apis[k]);
for (unsigned i = 0; i < audioDevice->getDeviceCount(); i++)
{
RtAudio::DeviceInfo dev_info = audioDevice->getDeviceInfo(i);
if (!dev_info.probed)
continue;
if (dev_info.outputChannels < 1)
continue;
GOrgueSoundDevInfo info;
info.channels = dev_info.outputChannels;
info.isDefault = dev_info.isDefaultOutput;
info.name = getName(rtaudio_apis[k], audioDevice, i);
result.push_back(info);
}
}
catch (RtAudioError &e)
{
wxString error = wxString::FromAscii(e.getMessage().c_str());
wxLogError(_("RtAudio error: %s"), error.c_str());
}
if (audioDevice)
delete audioDevice;
}
}
catch (RtAudioError &e)
{
wxString error = wxString::FromAscii(e.getMessage().c_str());
wxLogError(_("RtAudio error: %s"), error.c_str());
}
}
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[])
{
// 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[] )
{
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;
}
// ========
// = 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;
}
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;
}
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 ;
}
unsigned int getAudioDeviceCount() {
RtAudio audio;
return audio.getDeviceCount();
}
//-----------------------------------------------------------------------------
// 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;
}