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Version 4.3.0

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This commit is contained in:
Gary Scavone
2009-03-24 23:02:15 -04:00
committed by Stephen Sinclair
parent 2cbce2d8bd
commit 27d9b79dc7
271 changed files with 22219 additions and 8834 deletions
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49
projects/examples/play.cpp
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@@ -38,10 +38,11 @@ void usage(void) {
// This tick() function handles sample computation only. It will be
// called automatically when the system needs a new buffer of audio
// samples.
int tick(char *buffer, int bufferSize, void *dataPointer)
int tick( void *outputBuffer, void *inputBuffer, unsigned int nBufferFrames,
double streamTime, RtAudioStreamStatus status, void *userData )
{
FileWvIn *input = (FileWvIn *) dataPointer;
register StkFloat *samples = (StkFloat *) buffer;
FileWvIn *input = (FileWvIn *) userData;
register StkFloat *samples = (StkFloat *) outputBuffer;
input->tickFrame( frames );
for ( unsigned int i=0; i<frames.size(); i++ )
@@ -58,18 +59,18 @@ int tick(char *buffer, int bufferSize, void *dataPointer)
int main(int argc, char *argv[])
{
// Minimal command-line checking.
if (argc < 3 || argc > 4) usage();
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 = 0;
FileWvIn *input = 0;
RtAudio dac;
FileWvIn input;
// Try to load the soundfile.
try {
input = new FileWvIn( argv[1] );
input.openFile( argv[1] );
}
catch (StkError &) {
exit(0);
@@ -77,21 +78,25 @@ int main(int argc, char *argv[])
// Set input read rate based on the default STK sample rate.
double rate = 1.0;
rate = input->getFileRate() / Stk::sampleRate();
rate = input.getFileRate() / Stk::sampleRate();
if ( argc == 4 ) rate *= atof(argv[3]);
input->setRate( rate );
input.setRate( rate );
input.ignoreSampleRateChange();
// Find out how many channels we have.
int channels = input->getChannels();
int channels = input.getChannels();
// Define and open the realtime output device.
// Figure out how many bytes in an StkFloat and setup the RtAudio object.
// 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;
int bufferSize = RT_BUFFER_SIZE;
unsigned int bufferFrames = RT_BUFFER_SIZE;
try {
dac = new RtAudio(0, channels, 0, 0, format, (int)Stk::sampleRate(), &bufferSize, 4);
dac.openStream( &parameters, NULL, format, (unsigned int)Stk::sampleRate(), &bufferFrames, &tick, (void *)&input );
}
catch (RtError &error) {
catch ( RtError &error ) {
error.printMessage();
goto cleanup;
}
@@ -100,13 +105,12 @@ int main(int argc, char *argv[])
(void) signal(SIGINT, finish);
// Resize the StkFrames object appropriately.
frames.resize( bufferSize, channels );
frames.resize( bufferFrames, channels );
try {
dac->setStreamCallback(&tick, (void *)input);
dac->startStream();
dac.startStream();
}
catch (RtError &error) {
catch ( RtError &error ) {
error.printMessage();
goto cleanup;
}
@@ -118,15 +122,12 @@ int main(int argc, char *argv[])
// By returning a non-zero value in the callback above, the stream
// is automatically stopped. But we should still close it.
try {
dac->cancelStreamCallback();
dac->closeStream();
dac.closeStream();
}
catch (RtError &error) {
catch ( RtError &error ) {
error.printMessage();
}
cleanup:
delete input;
delete dac;
return 0;
}