Version 4.3.0

src/RtWvIn.cpp

@@ -3,62 +3,98 @@
     \brief STK realtime audio (blocking) input class.
 
     This class provides a simplified interface to RtAudio for realtime
-    audio input.  It is a protected subclass of WvIn.  Because this
-    class makes use of RtAudio's blocking output routines, its
-    performance is less robust on systems where the audio API is
-    callback-based (Macintosh CoreAudio and Windows ASIO).
+    audio input.  It is a subclass of WvIn.  This class makes use of
+    RtAudio's callback functionality by creating a large ring-buffer
+    from which data is read.  This class should not be used when
+    low-latency is desired.
 
-    RtWvIn supports multi-channel data in interleaved format.  It is
-    important to distinguish the tick() methods, which return samples
-    produced by averaging across sample frames, from the tickFrame()
-    methods, which return references or pointers to multi-channel
-    sample frames.
+    RtWvIn supports multi-channel data in both interleaved and
+    non-interleaved formats.  It is important to distinguish the
+    tick() methods, which return samples produced by averaging across
+    sample frames, from the tickFrame() methods, which return
+    references or pointers to multi-channel sample frames.
 
-    by Perry R. Cook and Gary P. Scavone, 1995 - 2005.
+    by Perry R. Cook and Gary P. Scavone, 1995 - 2007.
 */
 /***************************************************/
 
 #include "RtWvIn.h"
 
-RtWvIn :: RtWvIn( unsigned int nChannels, StkFloat sampleRate, int device, int bufferFrames, int nBuffers )
-  : stopped_( true ), bufferIndex_( 0 )
+// This function is automatically called by RtAudio to supply input audio data.
+int read( void *outputBuffer, void *inputBuffer, unsigned int nBufferFrames,
+          double streamTime, RtAudioStreamStatus status, void *dataPointer )
 {
-  int size = bufferFrames;
+  ( (RtWvIn *) dataPointer )->fillBuffer( inputBuffer, nBufferFrames );
+  return 0;
+}
+
+// This function does not block.  If the user does not read the buffer
+// data fast enough, unread data will be overwritten (data overrun).
+void RtWvIn :: fillBuffer( void *buffer, unsigned int nFrames )
+{
+  StkFloat *samples = (StkFloat *) buffer;
+  unsigned int counter, iStart, nSamples = nFrames * data_.channels();
+
+  while ( nSamples > 0 ) {
+
+    // I'm assuming that both the RtAudio and StkFrames buffers
+    // contain interleaved data.
+    iStart = writeIndex_ * data_.channels();
+    counter = nSamples;
+
+    // Pre-increment write pointer and check bounds.
+    writeIndex_ += nSamples / data_.channels();
+    if ( writeIndex_ >= data_.frames() ) {
+      writeIndex_ = 0;
+      counter = data_.size() - iStart;
+    }
+
+    // Copy data to the StkFrames container.
+    for ( unsigned int i=0; i<counter; i++ )
+      data_[iStart++] = *samples++;
+
+    nSamples -= counter;
+  }
+
+  framesFilled_ += nFrames;
+  if ( framesFilled_ > data_.frames() ) {
+    framesFilled_ = data_.frames();
+    errorString_ << "RtWvIn: audio buffer overrun!";
+    handleError( StkError::WARNING );
+  }
+}
+
+RtWvIn :: RtWvIn( unsigned int nChannels, StkFloat sampleRate, int device, int bufferFrames, int nBuffers )
+  : stopped_( true ), readIndex_( 0 ), writeIndex_( 0 ), framesFilled_( 0 )
+{
+  // We'll let RtAudio deal with channel and sample rate limitations.
+  RtAudio::StreamParameters parameters;
+  parameters.deviceId = device;
+  parameters.nChannels = nChannels;
+  unsigned int size = bufferFrames;
   RtAudioFormat format = ( sizeof(StkFloat) == 8 ) ? RTAUDIO_FLOAT64 : RTAUDIO_FLOAT32;
 
-  adc_ = 0;
   try {
-    adc_ = new RtAudio();
+    adc_.openStream( NULL, &parameters, format, (unsigned int)Stk::sampleRate(), &size, &read, (void *)this );
   }
-  catch (RtError &error) {
-    handleError( error.getMessageString(), StkError::AUDIO_SYSTEM );
+  catch ( RtError &error ) {
+    handleError( error.what(), StkError::AUDIO_SYSTEM );
   }
 
-  // Allocate the adc and get the buffer pointer.
-  try {
-    adc_->openStream( 0, 0, device, (int)nChannels, format,
-                        (int)sampleRate, &size, nBuffers );
-    buffer_ = (StkFloat *) adc_->getStreamBuffer();
-  }
-  catch (RtError &error) {
-    handleError( error.getMessageString(), StkError::AUDIO_SYSTEM );
-  }
-
-  bufferFrames_ = size;
+  data_.resize( size * nBuffers, nChannels );
   lastOutputs_.resize( 1, nChannels );
 }
 
 RtWvIn :: ~RtWvIn()
 {
-  if ( !stopped_ ) adc_->stopStream();
-  adc_->closeStream();
-  delete adc_;
+  if ( !stopped_ ) adc_.stopStream();
+  adc_.closeStream();
 }
 
 void RtWvIn :: start()
 {
   if ( stopped_ ) {
-    adc_->startStream();
+    adc_.startStream();
     stopped_ = false;
   }
 }
@@ -66,7 +102,7 @@ void RtWvIn :: start()
 void RtWvIn :: stop()
 {
   if ( !stopped_ ) {
-    adc_->stopStream();
+    adc_.stopStream();
     stopped_ = true;
   }
 }
@@ -75,20 +111,15 @@ void RtWvIn :: computeFrame( void )
 {
   if ( stopped_ ) this->start();
 
-  if ( bufferIndex_ == 0) {
-    try {
-      adc_->tickStream();
-    }
-    catch (RtError &error) {
-      handleError( error.getMessageString(), StkError::AUDIO_SYSTEM );
-    }
-  }
+  // Block until at least one frame is available.
+  while ( framesFilled_ == 0 )
+    Stk::sleep( 1 );
 
-  long iBuffer = bufferIndex_ * lastOutputs_.size();
-  for (unsigned int i=0; i<lastOutputs_.size(); i++)
-    lastOutputs_[i] = buffer_[iBuffer++];
+  for ( unsigned int i=0; i<lastOutputs_.size(); i++ )
+    lastOutputs_[i] = data_( readIndex_, i );
 
-  bufferIndex_++;
-  if ( bufferIndex_ >= bufferFrames_ )
-    bufferIndex_ = 0;
+  framesFilled_--;
+  readIndex_++;
+  if ( readIndex_ >= data_.frames() )
+    readIndex_ = 0;
 }