mirror of
https://github.com/thestk/stk
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Version 4.0
This commit is contained in:
Gary Scavone
committed by
Stephen Sinclair
Stephen Sinclair
parent
3f126af4e5
commit
81475b04c5
267
src/RtWvIn.cpp
267
src/RtWvIn.cpp
@@ -1,146 +1,121 @@
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/*******************************************/
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/* RtWvIn Input Class, */
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/* by Gary P. Scavone, 1999-2000 */
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/* */
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/* This object inherits from WvIn and is */
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/* used to read in realtime 16-bit data */
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/* from a computer's audio port. */
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/* */
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/* NOTE: This object is NOT intended for */
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/* use in achieving simultaneous realtime */
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/* audio input/output (together with */
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/* RtWvOut). Under certain circumstances */
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/* such a scheme is possible, though you */
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/* should definitely know what you are */
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/* doing before trying. For safer "full- */
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/* duplex" operation, use the RtDuplex */
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/* class. */
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/*******************************************/
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#include "RtWvIn.h"
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RtWvIn :: RtWvIn(int chans, MY_FLOAT srate, int device)
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{
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chunking = 1;
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looping = 0;
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sound_dev = new RtAudio(chans, srate, "record", device);
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channels = chans;
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bufferSize = RT_BUFFER_SIZE;
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data = 0;
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rtdata = (INT16 *) new INT16[(bufferSize+1)*channels];
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lastSamples = (INT16 *) new INT16[channels];
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for (int i=0;i<channels;i++) {
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lastSamples[i] = (INT16) 0.0;
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}
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this->getData(0);
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rate = (MY_FLOAT) srate / SRATE;
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if (fmod(rate, 1.0) > 0.0) interpolate = 1;
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else interpolate = 0;
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phaseOffset = (MY_FLOAT) 0.0;
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lastOutput = (MY_FLOAT *) new MY_FLOAT[channels];
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this->reset();
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gain = 0.00003052;
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#if (defined(__STK_REALTIME_) && defined(__OS_IRIX_))
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// This is necessary under IRIX because it scales the input by 0.5
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// when using single-channel input.
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if (channels == 1) gain *= 2;
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#endif
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}
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RtWvIn :: ~RtWvIn()
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{
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delete sound_dev;
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if (rtdata) {
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delete [ ] rtdata;
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rtdata = 0;
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}
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if (lastSamples) {
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delete [ ] lastSamples;
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lastSamples = 0;
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}
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}
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void RtWvIn :: setRate(MY_FLOAT aRate)
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{
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// Negative rates not allowed for realtime input
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rate = fabs(aRate);
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if (fmod(rate, 1.0) != 0.0) interpolate = 1;
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else interpolate = 0;
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}
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void RtWvIn :: addTime(MY_FLOAT aTime)
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{
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// Negative time shift no allowed for realtime input
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time += fabs(aTime);
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}
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void RtWvIn :: setLooping(int aLoopStatus)
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{
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// No looping for realtime data.
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looping = 0;
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}
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long RtWvIn :: getSize()
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{
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return bufferSize;
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}
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void RtWvIn :: getData(long index)
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{
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static long temp = RT_BUFFER_SIZE*channels;
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sound_dev->recordBuffer(&rtdata[channels],temp);
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/* Fill in the extra sample frame for interpolation.
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* We do this by pre-pending the last sample frame
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* from the previous input buffer to the current one.
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*/
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for (int i=0;i<channels;i++) {
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rtdata[i] = lastSamples[i];
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lastSamples[i] = rtdata[temp+i];
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}
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}
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int RtWvIn :: informTick()
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{
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static MY_FLOAT alpha;
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static long index;
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if (time >= bufferSize) {
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this->getData(0);
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while (time >= bufferSize)
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time -= bufferSize;
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}
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// integer part of time address
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index = (long) time;
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if (interpolate) {
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// fractional part of time address
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alpha = time - (MY_FLOAT) index;
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index *= channels;
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for (int i=0;i<channels;i++) {
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// Do linear interpolation
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lastOutput[i] = (MY_FLOAT) rtdata[index];
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lastOutput[i] += alpha * ((MY_FLOAT) rtdata[index+channels] - lastOutput[i]);
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lastOutput[i] *= gain;
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index++;
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}
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}
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else {
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index *= channels;
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for (int i=0;i<channels;i++) {
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lastOutput[i] = rtdata[index++];
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lastOutput[i] *= gain;
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}
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}
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// increment time, which can be negative
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time += rate;
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return finished;
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}
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/***************************************************/
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/*! \class RtWvIn
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\brief STK realtime audio input class.
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This class provides a simplified interface to
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RtAudio for realtime audio input. It is a
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protected subclass of WvIn.
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RtWvIn supports multi-channel data in
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interleaved format. It is important to
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distinguish the tick() methods, which return
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samples produced by averaging across sample
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frames, from the tickFrame() methods, which
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return pointers to multi-channel sample frames.
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For single-channel data, these methods return
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equivalent values.
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by Perry R. Cook and Gary P. Scavone, 1995 - 2002.
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*/
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/***************************************************/
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#include "RtWvIn.h"
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RtWvIn :: RtWvIn(int nChannels, MY_FLOAT sampleRate, int device, int bufferFrames, int nBuffers )
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{
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channels = nChannels;
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int size = bufferFrames;
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RtAudio::RTAUDIO_FORMAT format = ( sizeof(MY_FLOAT) == 8 ) ? RtAudio::RTAUDIO_FLOAT64 : RtAudio::RTAUDIO_FLOAT32;
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try {
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audio = new RtAudio(&stream, 0, 0, device, channels, format,
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(int)sampleRate, &size, nBuffers);
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data = (MY_FLOAT *) audio->getStreamBuffer(stream);
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}
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catch (RtError &error) {
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handleError( error.getMessage(), StkError::AUDIO_SYSTEM );
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}
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bufferSize = size;
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lastOutput = (MY_FLOAT *) new MY_FLOAT[channels];
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for (unsigned int i=0; i<channels; i++) lastOutput[i] = 0.0;
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counter = 0;
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stopped = true;
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}
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RtWvIn :: ~RtWvIn()
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{
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if ( !stopped )
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audio->stopStream(stream);
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delete audio;
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data = 0; // RtAudio deletes the buffer itself.
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}
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void RtWvIn :: start()
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{
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if ( stopped ) {
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audio->startStream(stream);
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stopped = false;
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}
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}
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void RtWvIn :: stop()
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{
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if ( !stopped ) {
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audio->stopStream(stream);
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stopped = true;
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}
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}
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MY_FLOAT RtWvIn :: lastOut(void) const
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{
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return WvIn::lastOut();
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}
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MY_FLOAT RtWvIn :: tick(void)
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{
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tickFrame();
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return lastOut();
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}
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MY_FLOAT *RtWvIn :: tick(MY_FLOAT *vector, unsigned int vectorSize)
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{
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for ( unsigned int i=0; i<vectorSize; i++ )
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vector[i] = tick();
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return vector;
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}
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const MY_FLOAT *RtWvIn :: lastFrame() const
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{
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return lastOutput;
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}
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const MY_FLOAT *RtWvIn :: tickFrame(void)
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{
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if ( stopped )
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start();
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if (counter == 0) {
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try {
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audio->tickStream(stream);
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}
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catch (RtError &error) {
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handleError( error.getMessage(), StkError::AUDIO_SYSTEM );
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}
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}
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long temp = counter * channels;
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for (unsigned int i=0; i<channels; i++)
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lastOutput[i] = data[temp++];
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counter++;
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if (counter >= (long) bufferSize)
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counter = 0;
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return lastOutput;
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}
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MY_FLOAT *RtWvIn :: tickFrame(MY_FLOAT *frameVector, unsigned int frames)
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{
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return WvIn::tickFrame( frameVector, frames );
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}
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