e1lama/stk
1
0
Fork 0
mirror of https://github.com/thestk/stk synced 2026-04-20 14:36:55 +00:00
Code Issues Packages Projects Releases Wiki Activity

Version 4.3.0

Browse Source
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
Download Patch File Download Diff File Expand all files Collapse all files

45
projects/effects/effects.cpp
View File

@@ -149,11 +149,13 @@ void processMessage( TickData* data )
// The tick() function handles sample computation and scheduling of
// control updates. It will be called automatically by RtAudio 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 *dataPointer )
{
TickData *data = (TickData *) dataPointer;
register StkFloat sample, *samples = (StkFloat *) buffer;
int i, counter, nTicks = bufferSize;
register StkFloat *oSamples = (StkFloat *) outputBuffer, *iSamples = (StkFloat *) inputBuffer;
register StkFloat sample;
int i, counter, nTicks = (int) nBufferFrames;
while ( nTicks > 0 && !done ) {
@@ -170,9 +172,9 @@ int tick(char *buffer, int bufferSize, void *dataPointer)
counter = min( nTicks, data->counter );
data->counter -= counter;
for ( i=0; i<counter; i++ ) {
sample = data->envelope.tick() * data->effect->tick( *samples );
*samples++ = sample; // two channels interleaved
*samples++ = sample;
sample = data->envelope.tick() * data->effect->tick( *iSamples++ );
*oSamples++ = sample; // two channels interleaved
*oSamples++ = sample;
nTicks--;
}
if ( nTicks == 0 ) break;
@@ -188,10 +190,10 @@ int tick(char *buffer, int bufferSize, void *dataPointer)
int main( int argc, char *argv[] )
{
TickData data;
RtAudio *adac = 0;
RtAudio adac;
int i;
if (argc < 2 || argc > 6) usage();
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
@@ -215,27 +217,31 @@ int main( int argc, char *argv[] )
// Allocate the adac here.
RtAudioFormat format = ( sizeof(StkFloat) == 8 ) ? RTAUDIO_FLOAT64 : RTAUDIO_FLOAT32;
int bufferSize = RT_BUFFER_SIZE;
RtAudio::StreamParameters oparameters, iparameters;
oparameters.deviceId = adac.getDefaultOutputDevice();
oparameters.nChannels = 2;
iparameters.deviceId = adac.getDefaultInputDevice();
iparameters.nChannels = 1;
unsigned int bufferFrames = RT_BUFFER_SIZE;
try {
adac = new RtAudio(0, 2, 0, 2, format, (int)Stk::sampleRate(), &bufferSize, 4);
adac.openStream( &oparameters, &iparameters, format, (unsigned int)Stk::sampleRate(), &bufferFrames, &tick, (void *)&data );
}
catch (RtError& error) {
catch ( RtError& error ) {
error.printMessage();
goto cleanup;
}
data.envelope.setRate( 0.001 );
data.effect = &(data.echo);
data.effect = &( data.echo );
// Install an interrupt handler function.
(void) signal( SIGINT, finish );
// If realtime output, set our callback function and start the dac.
try {
adac->setStreamCallback( &tick, (void *)&data );
adac->startStream();
adac.startStream();
}
catch (RtError &error) {
catch ( RtError &error ) {
error.printMessage();
goto cleanup;
}
@@ -246,19 +252,16 @@ int main( int argc, char *argv[] )
Stk::sleep( 50 );
}
// Shut down the callback and output stream.
// Shut down the output stream.
try {
adac->cancelStreamCallback();
adac->closeStream();
adac.closeStream();
}
catch (RtError& error) {
catch ( RtError& error ) {
error.printMessage();
}
cleanup:
delete adac;
std::cout << "\neffects finished ... goodbye.\n\n";
return 0;
}