// This tick() function handles sample computation only. It will be
// called automatically when the system needs a new buffer of audio
// samples.
int tick( void *outputBuffer, void *inputBuffer, unsigned int nBufferFrames,
double streamTime, RtAudioStreamStatus status, void *userData )
{
FileWvIn *input = (FileWvIn *) userData;
register StkFloat *samples = (StkFloat *) outputBuffer;
input->tick( frames );
for ( unsigned int i=0; i<frames.size(); i++ )
*samples++ = frames[i];
//if ( input->isFinished() ) {
if ( false ) {
done = true;
return 1;
}
else
return 0;
}
int main( int argc, char* argv[] )
{
cout << "*** Arguments ***" << endl;
for ( int a = 0; a < argc; a++ )
cout << a << " : " << argv[a] << endl;
if ( argc == 1 ) return 1;
string ifname = string( argv[1] );
FileWvIn filewvin = FileWvIn( ifname, false, true );
//size_t pos = ifname.rfind("/");
//string dirname = ifname.substr(0, pos);
size_t extpos = ifname.rfind(".wav");
string basename = ifname.substr(0, extpos);
for ( unsigned int i=0; i<filewvin.channelsOut(); i++ )
{
string ofname = basename + "-" + to_string(i) + ".wav";
FileWvOut filewvout = FileWvOut( ofname );
cout << "channel " << i << endl;
vector<StkFloat> data;
unsigned int sn = 0;
int mod = 0;
float delta = 0.0f;
float max = 0.0f;
float corrected;
filewvin.reset();
StkFloat previous;
StkFloat value = filewvin.tick(i);
//~ filewvout.tick( previous );
while ( not( filewvin.isFinished() ) )
{
//~ cout << "value : " << value << endl;
corrected = value + mod;
data.push_back( corrected );
if ( abs( corrected ) > max ) max = abs( corrected );
previous = value;
sn++;
value = filewvin.tick(i);
delta = value - previous;
if ( delta > 1.7 )
{
if ( mod == 0 ) mod = -2;
else mod = 0;
cout << sn << " | value : " << value << " | delta : " << delta << endl;
}
else if ( delta < -1.7 )
{
if ( mod == 0 ) mod = 2;
else mod = 0;
cout << sn << " | value : " << value << " | delta : " << delta << endl;
}
}
cout << "maximum amp : " << max << endl;
for ( vector<StkFloat>::iterator f=data.begin(); f != data.end(); f++ )
filewvout.tick( *f / max );
filewvout.closeFile();
}
filewvin.closeFile();
return 0;
}
// This tick() function handles sample computation only. It will be
// called automatically when the system needs a new buffer of audio
// samples.
int tick( void *outputBuffer, void *inputBuffer, unsigned int nBufferFrames,
double streamTime, RtAudioStreamStatus status, void *userData )
{
FileWvIn *input = (FileWvIn *) userData;
StkFloat *samples = (StkFloat *) outputBuffer;
input->tick( frames );
for ( unsigned int i=0; i<frames.size(); i++ ) {
*samples++ = frames[i];
if ( input->channelsOut() == 1 ) *samples++ = frames[i]; // play mono files in stereo
}
if ( input->isFinished() ) {
done = true;
return 1;
}
else
return 0;
}
int main( int argc, char *argv[] )
{
// Minimal command-line checking.
if ( argc < 3 || argc > 4 ) usage();
FileWvIn input;
InetWvOut output;
// Load the file.
try {
input.openFile( (char *)argv[1] );
}
catch ( StkError & ) {
exit( 1 );
}
// Set the global STK sample rate to the file rate.
Stk::setSampleRate( input.getFileRate() );
// Set input read rate.
double rate = 1.0;
if ( argc == 4 ) rate = atof( argv[3] );
input.setRate( rate );
// Find out how many channels we have.
int channels = input.channelsOut();
StkFrames frames( 4096, channels );
// Attempt to connect to the socket server.
try {
//output.connect( 2006, Socket::PROTO_UDP, (char *)argv[2], channels, Stk::STK_SINT16 );
output.connect( 2006, Socket::PROTO_TCP, (char *)argv[2], channels, Stk::STK_SINT16 );
}
catch ( StkError & ) {
exit( 1 );
}
// Here's the runtime loop
while ( !input.isFinished() )
output.tick( input.tick( frames ) );
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;
}