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https://github.com/thestk/stk
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Version 4.2.1
This commit is contained in:
Gary Scavone
committed by
Stephen Sinclair
Stephen Sinclair
parent
a6381b9d38
commit
2cbce2d8bd
221
src/RtWvOut.cpp
221
src/RtWvOut.cpp
@@ -2,35 +2,34 @@
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/*! \class RtWvOut
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\brief STK realtime audio (blocking) output class.
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This class provides a simplified interface to
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RtAudio for realtime audio output. It is a
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protected subclass of WvOut.
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This class provides a simplified interface to RtAudio for realtime
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audio output. It is a subclass of WvOut. Because this class
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makes use of RtAudio's blocking output routines, its performance
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is less robust on systems where the audio API is callback-based
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(Macintosh CoreAudio and Windows ASIO).
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RtWvOut supports multi-channel data in
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interleaved format. It is important to
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distinguish the tick() methods, which output
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single samples to all channels in a sample
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frame, from the tickFrame() method, which
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takes a pointer to multi-channel sample
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RtWvOut supports multi-channel data in interleaved format. It is
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important to distinguish the tick() methods, which output single
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samples to all channels in a sample frame, from the tickFrame()
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method, which take a pointer or reference to multi-channel sample
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frame data.
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by Perry R. Cook and Gary P. Scavone, 1995 - 2004.
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by Perry R. Cook and Gary P. Scavone, 1995 - 2005.
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*/
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/***************************************************/
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#include "RtWvOut.h"
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#include <iostream>
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RtWvOut :: RtWvOut( unsigned int nChannels, StkFloat sampleRate, int device, int bufferFrames, int nBuffers )
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: stopped_( true ), nChannels_(nChannels), bufferIndex_( 0 ), iBuffer_( 0 )
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{
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// We'll let RtAudio deal with channel and srate limitations.
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channels_ = nChannels;
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bufferSize_ = bufferFrames;
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// We'll let RtAudio deal with channel and sample rate limitations.
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int size = bufferFrames;
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RtAudioFormat format = ( sizeof(StkFloat) == 8 ) ? RTAUDIO_FLOAT64 : RTAUDIO_FLOAT32;
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audio_ = 0;
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dac_ = 0;
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try {
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audio_ = new RtAudio();
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dac_ = new RtAudio();
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}
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catch (RtError &error) {
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handleError( error.getMessageString(), StkError::AUDIO_SYSTEM );
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@@ -38,28 +37,28 @@ RtWvOut :: RtWvOut( unsigned int nChannels, StkFloat sampleRate, int device, int
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// Now open a stream and get the buffer pointer.
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try {
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audio_->openStream(device, (int)channels_, 0, 0, format,
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(int)sampleRate, &bufferSize_, nBuffers);
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dataPtr_ = (StkFloat *) audio_->getStreamBuffer();
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dac_->openStream( device, (int)nChannels, 0, 0, format,
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(int)sampleRate, &size, nBuffers );
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buffer_ = (StkFloat *) dac_->getStreamBuffer();
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}
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catch (RtError &error) {
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handleError( error.getMessageString(), StkError::AUDIO_SYSTEM );
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}
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stopped_ = true;
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bufferFrames_ = size;
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}
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RtWvOut :: ~RtWvOut()
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{
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if ( !stopped_ )
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audio_->stopStream();
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delete audio_;
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if ( !stopped_ ) dac_->stopStream();
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dac_->closeStream();
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delete dac_;
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}
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void RtWvOut :: start()
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{
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if ( stopped_ ) {
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audio_->startStream();
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dac_->startStream();
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stopped_ = false;
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}
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}
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@@ -67,162 +66,80 @@ void RtWvOut :: start()
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void RtWvOut :: stop()
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{
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if ( !stopped_ ) {
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audio_->stopStream();
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dac_->stopStream();
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stopped_ = true;
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}
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}
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unsigned long RtWvOut :: getFrames( void ) const
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void RtWvOut :: incrementFrame( void )
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{
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return totalCount_;
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}
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frameCounter_++;
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bufferIndex_++;
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StkFloat RtWvOut :: getTime( void ) const
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{
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return (StkFloat) totalCount_ / Stk::sampleRate();
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}
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void RtWvOut :: tick( const StkFloat sample )
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{
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if ( stopped_ )
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start();
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StkFloat input = sample;
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this->clipTest( input );
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for ( unsigned int j=0; j<channels_; j++ )
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dataPtr_[counter_*channels_+j] = input;
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counter_++;
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totalCount_++;
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if ( counter_ >= (unsigned int )bufferSize_ ) {
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if ( bufferIndex_ == bufferFrames_ ) {
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try {
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audio_->tickStream();
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dac_->tickStream();
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}
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catch (RtError &error) {
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handleError( error.getMessageString(), StkError::AUDIO_SYSTEM );
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}
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counter_ = 0;
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bufferIndex_ = 0;
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iBuffer_ = 0;
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}
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}
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void RtWvOut :: tick( const StkFloat *vector, unsigned int vectorSize )
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void RtWvOut :: computeSample( const StkFloat sample )
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{
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for (unsigned int i=0; i<vectorSize; i++)
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tick( vector[i] );
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if ( stopped_ ) start();
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StkFloat input = sample;
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clipTest( input );
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for ( unsigned int j=0; j<nChannels_; j++ )
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buffer_[iBuffer_++] = input;
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this->incrementFrame();
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}
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void RtWvOut :: tick( const StkFrames& frames, unsigned int channel )
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void RtWvOut :: computeFrames( const StkFrames& frames )
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{
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if ( channel == 0 || frames.channels() < channel ) {
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errorString_ << "RtWvOut::tick(): channel argument (" << channel << ") is zero or > channels in StkFrames argument!";
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if ( stopped_ ) start();
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if ( frames.channels() != nChannels_ ) {
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errorString_ << "RtWvOut::computeFrames(): incompatible channel value in StkFrames argument!";
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handleError( StkError::FUNCTION_ARGUMENT );
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}
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if ( stopped_ )
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start();
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if ( frames.channels() == 1 ) {
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for ( unsigned int i=0; i<frames.frames(); i++ )
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tick( frames[i] );
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}
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else if ( frames.interleaved() ) {
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unsigned int hop = frames.channels();
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unsigned int index = channel - 1;
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for ( unsigned int i=0; i<frames.frames(); i++ ) {
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tick( frames[index] );
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index += hop;
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}
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}
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else {
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unsigned int iStart = (channel - 1) * frames.frames();
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for ( unsigned int i=0; i<frames.frames(); i++ )
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tick( frames[iStart + i] );
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}
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}
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void RtWvOut :: tickFrame( const StkFloat *frameVector, unsigned int frames )
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{
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if ( stopped_ )
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start();
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StkFloat sample;
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for ( unsigned int i=0; i<frames; i++ ) {
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for ( unsigned int j=0; j<channels_; j++ ) {
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sample = frameVector[i*channels_+j];
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this->clipTest( sample );
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dataPtr_[counter_*channels_+j] = sample;
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}
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counter_++;
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totalCount_++;
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if ( counter_ >= (unsigned int)bufferSize_ ) {
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try {
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audio_->tickStream();
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}
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catch (RtError &error) {
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handleError( error.getMessageString(), StkError::AUDIO_SYSTEM );
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}
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counter_ = 0;
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}
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}
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}
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void RtWvOut :: tickFrame( const StkFrames& frames )
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{
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if ( channels_ != frames.channels() ) {
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errorString_ << "RtWvOut::tickFrame(): incompatible channel value in StkFrames argument!";
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handleError( StkError::FUNCTION_ARGUMENT );
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}
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if ( stopped_ )
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start();
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unsigned int j;
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StkFloat sample;
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if ( channels_ == 1 || frames.interleaved() ) {
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unsigned long iFrames = 0, iData = counter_;
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for ( unsigned int i=0; i<frames.frames(); i++ ) {
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for ( j=0; j<channels_; j++ ) {
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sample = frames[iFrames++];
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this->clipTest( sample );
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dataPtr_[iData++] = sample;
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}
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counter_++;
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totalCount_++;
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if ( nChannels_ == 1 || frames.interleaved() ) {
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if ( counter_ >= (unsigned int)bufferSize_ ) {
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try {
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audio_->tickStream();
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}
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catch (RtError &error) {
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handleError( error.getMessageString(), StkError::AUDIO_SYSTEM );
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}
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counter_ = 0;
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unsigned int iFrames = 0;
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for ( unsigned int i=0; i<frames.frames(); i++ ) {
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for ( j=0; j<nChannels_; j++ ) {
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buffer_[iBuffer_] = frames[iFrames++];
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clipTest( buffer_[iBuffer_++] );
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}
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this->incrementFrame();
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}
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}
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else {
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unsigned int hop = frames.frames();
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unsigned long iData = counter_;
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for ( unsigned int i=0; i<frames.frames(); i++ ) {
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for ( j=0; j<channels_; j++ ) {
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sample = frames[i + j*hop];
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this->clipTest( sample );
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dataPtr_[iData++] = sample;
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}
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counter_++;
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totalCount_++;
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else { // non-interleaved frames
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if ( counter_ >= (unsigned int)bufferSize_ ) {
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try {
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audio_->tickStream();
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}
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catch (RtError &error) {
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handleError( error.getMessageString(), StkError::AUDIO_SYSTEM );
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}
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counter_ = 0;
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unsigned long hop = frames.frames();
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unsigned int index;
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for ( unsigned int i=0; i<frames.frames(); i++ ) {
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index = i;
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for ( j=0; j<nChannels_; j++ ) {
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buffer_[iBuffer_] = frames[index];
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clipTest( buffer_[iBuffer_++] );
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index += hop;
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}
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this->incrementFrame();
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}
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}
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}
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