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;
}
