Version 4.3.0

src/ADSR.cpp

@@ -11,7 +11,7 @@
     envelope value reaches 0.0 in the
     ADSR::RELEASE state.
 
-    by Perry R. Cook and Gary P. Scavone, 1995 - 2005.
+    by Perry R. Cook and Gary P. Scavone, 1995 - 2007.
 */
 /***************************************************/
 
@@ -32,6 +32,15 @@ ADSR :: ~ADSR()
 {
 }
 
+void ADSR :: sampleRateChanged( StkFloat newRate, StkFloat oldRate )
+{
+  if ( !ignoreSampleRateChange_ ) {
+    attackRate_ = oldRate * attackRate_ / newRate;
+    decayRate_ = oldRate * decayRate_ / newRate;
+    releaseRate_ = oldRate * releaseRate_ / newRate;
+  }
+}
+
 void ADSR :: keyOn()
 {
   target_ = 1.0;

src/Asymp.cpp

@@ -19,12 +19,12 @@
     to \e keyOn and \e keyOff messages by ramping to
     1.0 on keyOn and to 0.0 on keyOff.
 
-    by Perry R. Cook and Gary P. Scavone, 1995 - 2005.
+    by Perry R. Cook and Gary P. Scavone, 1995 - 2007.
 */
 /***************************************************/
 
 #include "Asymp.h"
-#include <math.h>
+#include <cmath>
 
 Asymp :: Asymp(void) : Envelope()
 {
@@ -36,6 +36,14 @@ Asymp :: ~Asymp(void)
 {    
 }
 
+void Asymp :: sampleRateChanged( StkFloat newRate, StkFloat oldRate )
+{
+  if ( !ignoreSampleRateChange_ ) {
+    StkFloat tau = -1.0 / ( std::log( factor_ ) * oldRate );
+    factor_ = std::exp( -1.0 / ( tau * newRate ) );
+  }
+}
+
 void Asymp :: keyOn(void)
 {
   Envelope::keyOn();
@@ -56,7 +64,7 @@ void Asymp :: setTau(StkFloat tau)
     return;
   }
 
-  factor_ = exp( -1.0 / ( tau * Stk::sampleRate() ) );
+ factor_ = std::exp( -1.0 / ( tau * Stk::sampleRate() ) );
   constant_ = ( 1.0 - factor_ ) * target_;
 }
 
@@ -68,8 +76,8 @@ void Asymp :: setTime(StkFloat time)
     return;
   }
 
-  StkFloat tau = -time / log( TARGET_THRESHOLD );
-  factor_ = exp( -1.0 / ( tau * Stk::sampleRate() ) );
+  StkFloat tau = -time / std::log( TARGET_THRESHOLD );
+  factor_ = std::exp( -1.0 / ( tau * Stk::sampleRate() ) );
   constant_ = ( 1.0 - factor_ ) * target_;
 }
 

src/BeeThree.cpp

@@ -28,7 +28,7 @@
     type who should worry about this (making
     money) worry away.
 
-    by Perry R. Cook and Gary P. Scavone, 1995 - 2005.
+    by Perry R. Cook and Gary P. Scavone, 1995 - 2007.
 */
 /***************************************************/
 

src/BiQuad.cpp

@@ -8,7 +8,7 @@
     frequency response while maintaining a constant
     filter gain.
 
-    by Perry R. Cook and Gary P. Scavone, 1995 - 2005.
+    by Perry R. Cook and Gary P. Scavone, 1995 - 2007.
 */
 /***************************************************/
 
@@ -22,10 +22,20 @@ BiQuad :: BiQuad() : Filter()
   b[0] = 1.0;
   a[0] = 1.0;
   Filter::setCoefficients( b, a );
+  Stk::addSampleRateAlert( this );
 }
 
 BiQuad :: ~BiQuad()
 {
+  Stk::removeSampleRateAlert( this );
+}
+
+void BiQuad :: sampleRateChanged( StkFloat newRate, StkFloat oldRate )
+{
+  if ( !ignoreSampleRateChange_ ) {
+    errorString_ << "BiQuad::sampleRateChanged: you may need to recompute filter coefficients!";
+    handleError( StkError::WARNING );
+  }
 }
 
 void BiQuad :: clear(void)
@@ -99,26 +109,3 @@ StkFloat BiQuad :: lastOut(void) const
 {
   return Filter::lastOut();
 }
-
-StkFloat BiQuad :: computeSample( StkFloat input )
-{
-  inputs_[0] = gain_ * input;
-  outputs_[0] = b_[0] * inputs_[0] + b_[1] * inputs_[1] + b_[2] * inputs_[2];
-  outputs_[0] -= a_[2] * outputs_[2] + a_[1] * outputs_[1];
-  inputs_[2] = inputs_[1];
-  inputs_[1] = inputs_[0];
-  outputs_[2] = outputs_[1];
-  outputs_[1] = outputs_[0];
-
-  return outputs_[0];
-}
-
-StkFloat BiQuad :: tick( StkFloat input )
-{
-  return this->computeSample( input );
-}
-
-StkFrames& BiQuad :: tick( StkFrames& frames, unsigned int channel )
-{
-  return Filter::tick( frames, channel );
-}

src/BlitSquare.cpp

@@ -8,8 +8,8 @@
     The algorithm implemented in this class uses a SincM function with
     an even M value to achieve a bipolar bandlimited impulse train.
     This signal is then integrated to achieve a square waveform.  The
-    integration process has an associated DC offset but that is
-    subtracted off the output signal.
+    integration process has an associated DC offset so a DC blocking
+    filter is applied at the output.
 
     The user can specify both the fundamental frequency of the
     waveform and the number of harmonics contained in the resulting
@@ -19,17 +19,21 @@
     to half the sample rate.  Note, however, that this setting may
     produce aliasing in the signal when the frequency is changing (no
     automatic modification of the number of harmonics is performed by
-    the setFrequency() function).
+    the setFrequency() function).  Also note that the harmonics of a
+    square wave fall at odd integer multiples of the fundamental, so
+    aliasing will happen with a lower fundamental than with the other
+    Blit waveforms.  This class is not guaranteed to be well behaved
+    in the presence of significant aliasing.
 
     Based on initial code of Robin Davies, 2005.
-    Modified algorithm code by Gary Scavone, 2005.
+    Modified algorithm code by Gary Scavone, 2005 - 2006.
 */
 /***************************************************/
 
 #include "BlitSquare.h"
 #include <cmath>
 #include <limits>
- 
+
 BlitSquare:: BlitSquare( StkFloat frequency )
 {
   nHarmonics_ = 0;
@@ -45,6 +49,8 @@ void BlitSquare :: reset()
 {
   phase_ = 0.0;
   lastOutput_ = 0;
+  dcbState_ = 0.0;
+  lastBlitOutput_ = 0;
 }
 
 void BlitSquare :: setFrequency( StkFloat frequency )
@@ -56,8 +62,8 @@ void BlitSquare :: setFrequency( StkFloat frequency )
 
   // By using an even value of the parameter M, we get a bipolar blit
   // waveform at half the blit frequency.  Thus, we need to scale the
-  // frequency value here by 2.0. (GPS, 2005).
-  p_ = 2.0 * Stk::sampleRate() / frequency;
+  // frequency value here by 0.5. (GPS, 2006).
+  p_ = 0.5 * Stk::sampleRate() / frequency;
   rate_ = PI / p_;
   this->updateHarmonics();
 }
@@ -73,13 +79,12 @@ void BlitSquare :: updateHarmonics( void )
   // Make sure we end up with an even value of the parameter M here.
   if ( nHarmonics_ <= 0 ) {
     unsigned int maxHarmonics = (unsigned int) floor( 0.5 * p_ );
-    m_ = 2 * maxHarmonics;
+    m_ = 2 * (maxHarmonics + 1);
   }
   else
-    m_ = 2 * nHarmonics_;
+    m_ = 2 * (nHarmonics_ + 1);
 
-  // This offset value was derived empirically. (GPS, 2005)
-  offset_ = 1.0 - 0.5 * m_ / p_;
+  a_ = m_ / p_;
 
 #if defined(_STK_DEBUG_)
   errorString_ << "BlitSquare::updateHarmonics: nHarmonics_ = " << nHarmonics_ << ", m_ = " << m_ << '.';
@@ -89,7 +94,7 @@ void BlitSquare :: updateHarmonics( void )
 
 StkFloat BlitSquare :: computeSample( void )
 {
-  StkFloat temp = lastOutput_;
+  StkFloat temp = lastBlitOutput_;
 
   // A fully  optimized version of this would replace the two sin calls
   // with a pair of fast sin oscillators, for which stable fast 
@@ -103,20 +108,24 @@ StkFloat BlitSquare :: computeSample( void )
   if ( fabs( denominator )  < std::numeric_limits<StkFloat>::epsilon() ) {
     // Inexact comparison safely distinguishes betwen *close to zero*, and *close to PI*.
     if ( phase_ < 0.1f || phase_ > TWO_PI - 0.1f )
-      lastOutput_ = 1.0;
+      lastBlitOutput_ = a_;
     else
-      lastOutput_ = -1.0;
+      lastBlitOutput_ = -a_;
   }
   else {
-    lastOutput_ =  sin( m_ * phase_ );
-    lastOutput_ /= p_ * denominator;
+    lastBlitOutput_ =  sin( m_ * phase_ );
+    lastBlitOutput_ /= p_ * denominator;
   }
 
-  lastOutput_ += temp;
+  lastBlitOutput_ += temp;
+
+  // Now apply DC blocker.
+  lastOutput_ = lastBlitOutput_ - dcbState_ + 0.999 * lastOutput_;
+  dcbState_ = lastBlitOutput_;
 
   phase_ += rate_;
   if ( phase_ >= TWO_PI ) phase_ -= TWO_PI;
-    
-	return lastOutput_ - offset_;
+
+	return lastOutput_;
 }
 

src/BlowBotl.cpp

@@ -12,7 +12,7 @@
        - Vibrato Gain = 1
        - Volume = 128
 
-    by Perry R. Cook and Gary P. Scavone, 1995 - 2005.
+    by Perry R. Cook and Gary P. Scavone, 1995 - 2007.
 */
 /***************************************************/
 

src/BlowHole.cpp

@@ -29,7 +29,7 @@
        - Register State = 1
        - Breath Pressure = 128
 
-    by Perry R. Cook and Gary P. Scavone, 1995 - 2005.
+    by Perry R. Cook and Gary P. Scavone, 1995 - 2007.
 */
 /***************************************************/
 

src/BowTable.cpp

@@ -5,7 +5,7 @@
     This class implements a simple bowed string
     non-linear function, as described by Smith (1986).
 
-    by Perry R. Cook and Gary P. Scavone, 1995 - 2005.
+    by Perry R. Cook and Gary P. Scavone, 1995 - 2007.
 */
 /***************************************************/
 

src/Bowed.cpp

@@ -17,7 +17,7 @@
        - Vibrato Gain = 1
        - Volume = 128
 
-    by Perry R. Cook and Gary P. Scavone, 1995 - 2005.
+    by Perry R. Cook and Gary P. Scavone, 1995 - 2007.
 */
 /***************************************************/
 

src/Brass.cpp

@@ -16,7 +16,7 @@
        - Vibrato Gain = 1
        - Volume = 128
 
-    by Perry R. Cook and Gary P. Scavone, 1995 - 2005.
+    by Perry R. Cook and Gary P. Scavone, 1995 - 2007.
 */
 /***************************************************/
 

src/Chorus.cpp

@@ -4,7 +4,7 @@
 
     This class implements a chorus effect.
 
-    by Perry R. Cook and Gary P. Scavone, 1995 - 2005.
+    by Perry R. Cook and Gary P. Scavone, 1995 - 2007.
 */
 /***************************************************/
 

src/Clarinet.cpp

@@ -18,7 +18,7 @@
        - Vibrato Gain = 1
        - Breath Pressure = 128
 
-    by Perry R. Cook and Gary P. Scavone, 1995 - 2005.
+    by Perry R. Cook and Gary P. Scavone, 1995 - 2007.
 */
 /***************************************************/
 
@@ -46,7 +46,7 @@ Clarinet :: ~Clarinet()
 void Clarinet :: clear()
 {
   delayLine_.clear();
-  filter_.tick((StkFloat) 0.0);
+  filter_.tick( 0.0 );
 }
 
 void Clarinet :: setFrequency(StkFloat frequency)

src/Delay.cpp

@@ -14,7 +14,7 @@
     used in fixed delay-length applications, such
     as for reverberation.
 
-    by Perry R. Cook and Gary P. Scavone, 1995 - 2005.
+    by Perry R. Cook and Gary P. Scavone, 1995 - 2007.
 */
 /***************************************************/
 

src/DelayA.cpp

@@ -15,7 +15,7 @@
     minimum delay possible in this implementation is limited to a
     value of 0.5.
 
-    by Perry R. Cook and Gary P. Scavone, 1995 - 2005.
+    by Perry R. Cook and Gary P. Scavone, 1995 - 2007.
 */
 /***************************************************/
 

src/DelayL.cpp

@@ -18,7 +18,7 @@
     order Lagrange interpolators can typically
     improve (minimize) this attenuation characteristic.
 
-    by Perry R. Cook and Gary P. Scavone, 1995 - 2005.
+    by Perry R. Cook and Gary P. Scavone, 1995 - 2007.
 */
 /***************************************************/
 

src/Drummer.cpp

@@ -11,7 +11,7 @@
     of simultaneous voices) via a #define in the
     Drummer.h.
 
-    by Perry R. Cook and Gary P. Scavone, 1995 - 2005.
+    by Perry R. Cook and Gary P. Scavone, 1995 - 2007.
 */
 /***************************************************/
 

src/Echo.cpp

@@ -4,7 +4,7 @@
 
     This class implements an echo effect.
 
-    by Perry R. Cook and Gary P. Scavone, 1995 - 2005.
+    by Perry R. Cook and Gary P. Scavone, 1995 - 2007.
 */
 /***************************************************/
 

src/Effect.cpp

@@ -5,7 +5,7 @@
     This class provides common functionality for
     STK effects subclasses.
 
-    by Perry R. Cook and Gary P. Scavone, 1995 - 2005.
+    by Perry R. Cook and Gary P. Scavone, 1995 - 2007.
 */
 /***************************************************/
 

src/Envelope.cpp

@@ -9,7 +9,7 @@
     \e keyOff messages, ramping to 1.0 on
     keyOn and to 0.0 on keyOff.
 
-    by Perry R. Cook and Gary P. Scavone, 1995 - 2005.
+    by Perry R. Cook and Gary P. Scavone, 1995 - 2007.
 */
 /***************************************************/
 
@@ -21,6 +21,7 @@ Envelope :: Envelope(void) : Generator()
   value_ = 0.0;
   rate_ = 0.001;
   state_ = 0;
+  Stk::addSampleRateAlert( this );
 }
 
 Envelope :: Envelope ( const Envelope& e )
@@ -29,10 +30,12 @@ Envelope :: Envelope ( const Envelope& e )
   value_ = 0.0;
   rate_ = 0.001;
   state_ = 0;
+  Stk::addSampleRateAlert( this );
 }
 
 Envelope :: ~Envelope(void)
 {
+  Stk::removeSampleRateAlert( this );
 }
 
 Envelope& Envelope :: operator= ( const Envelope& e )
@@ -47,6 +50,12 @@ Envelope& Envelope :: operator= ( const Envelope& e )
   return *this;
 }
 
+void Envelope :: sampleRateChanged( StkFloat newRate, StkFloat oldRate )
+{
+  if ( !ignoreSampleRateChange_ )
+    rate_ = oldRate * rate_ / newRate;
+}
+
 void Envelope :: keyOn(void)
 {
   target_ = 1.0;

src/FM.cpp

@@ -19,7 +19,7 @@
     type who should worry about this (making
     money) worry away.
 
-    by Perry R. Cook and Gary P. Scavone, 1995 - 2005.
+    by Perry R. Cook and Gary P. Scavone, 1995 - 2007.
 */
 /***************************************************/
 

src/FMVoices.cpp

@@ -26,7 +26,7 @@
     type who should worry about this (making
     money) worry away.
 
-    by Perry R. Cook and Gary P. Scavone, 1995 - 2005.
+    by Perry R. Cook and Gary P. Scavone, 1995 - 2007.
 */
 /***************************************************/
 

src/FileRead.cpp

@@ -25,7 +25,7 @@
     filling a matrix row.  The sample rate for
     MAT-files is assumed to be 44100 Hz.
 
-    by Perry R. Cook and Gary P. Scavone, 1995 - 2005.
+    by Perry R. Cook and Gary P. Scavone, 1995 - 2007.
 */
 /***************************************************/
 
@@ -469,11 +469,12 @@ bool FileRead :: getMatInfo( const char *fileName )
   // http://www.mathworks.com/access/helpdesk/help/pdf_doc/matlab/matfile_format.pdf
 
   // Verify this is a version 5 MAT-file format.
-  char head[4];
+  char head[5];
   if ( fseek(fd_, 0, SEEK_SET) == -1 ) goto error;
   if ( fread(&head, 4, 1, fd_) != 1 ) goto error;
   // If any of the first 4 characters of the header = 0, then this is
   // a Version 4 MAT-file.
+  head[4] = '\0';
   if ( strstr(head, "0") ) {
     errorString_ << "FileRead: " << fileName << " appears to be a Version 4 MAT-file, which is not currently supported.";
     return false;

src/FileWrite.cpp

@@ -17,7 +17,7 @@
     type, the data type will automatically be modified.  Compressed
     data types are not supported.
 
-    by Perry R. Cook and Gary P. Scavone, 1995 - 2005.
+    by Perry R. Cook and Gary P. Scavone, 1995 - 2007.
 */
 /***************************************************/
 

src/FileWvIn.cpp

@@ -26,7 +26,7 @@
     See the FileRead class for a description of the supported audio
     file formats.
 
-    by Perry R. Cook and Gary P. Scavone, 1995 - 2005.
+    by Perry R. Cook and Gary P. Scavone, 1995 - 2007.
 */
 /***************************************************/
 
@@ -37,6 +37,7 @@ FileWvIn :: FileWvIn( unsigned long chunkThreshold, unsigned long chunkSize )
   : finished_(true), interpolate_(false), time_(0.0),
     chunkThreshold_(chunkThreshold), chunkSize_(chunkSize)
 {
+  Stk::addSampleRateAlert( this );
 }
 
 FileWvIn :: FileWvIn( std::string fileName, bool raw, bool doNormalize,
@@ -45,11 +46,19 @@ FileWvIn :: FileWvIn( std::string fileName, bool raw, bool doNormalize,
     chunkThreshold_(chunkThreshold), chunkSize_(chunkSize)
 {
   openFile( fileName, raw, doNormalize );
+  Stk::addSampleRateAlert( this );
 }
 
 FileWvIn :: ~FileWvIn()
 {
   this->closeFile();
+  Stk::removeSampleRateAlert( this );
+}
+
+void FileWvIn :: sampleRateChanged( StkFloat newRate, StkFloat oldRate )
+{
+  if ( !ignoreSampleRateChange_ )
+    this->setRate( oldRate * rate_ / newRate );
 }
 
 void FileWvIn :: closeFile( void )
@@ -130,6 +139,11 @@ void FileWvIn :: normalize( StkFloat peak )
 
 void FileWvIn :: setRate( StkFloat rate )
 {
+#if defined(_STK_DEBUG_)
+  errorString_ << "FileWvIn::setRate: changing file read rate from " << rate_ << " to " << rate << '.';
+  handleError( StkError::DEBUG_WARNING );
+#endif
+
   rate_ = rate;
 
   // If negative rate and at beginning of sound, move pointer to end

src/FileWvOut.cpp

@@ -17,7 +17,7 @@
     Currently, FileWvOut is non-interpolating and the output rate is
     always Stk::sampleRate().
 
-    by Perry R. Cook and Gary P. Scavone, 1995 - 2005.
+    by Perry R. Cook and Gary P. Scavone, 1995 - 2007.
 */
 /***************************************************/
 

src/Filter.cpp

@@ -23,7 +23,7 @@
     results in one extra multiply per computed sample,
     but allows easy control of the overall filter gain.
 
-    by Perry R. Cook and Gary P. Scavone, 1995 - 2005.
+    by Perry R. Cook and Gary P. Scavone, 1995 - 2007.
 */
 /***************************************************/
 

src/Flute.cpp

@@ -18,7 +18,7 @@
        - Vibrato Gain = 1
        - Breath Pressure = 128
 
-    by Perry R. Cook and Gary P. Scavone, 1995 - 2005.
+    by Perry R. Cook and Gary P. Scavone, 1995 - 2007.
 */
 /***************************************************/
 
@@ -31,7 +31,6 @@ Flute :: Flute(StkFloat lowestFrequency)
   boreDelay_.setMaximumDelay( length_ );
   boreDelay_.setDelay( 100.0 );
 
-  length_ >>= 1;
   jetDelay_.setMaximumDelay( length_ );
   jetDelay_.setDelay( 49.0 );
 
@@ -133,7 +132,7 @@ void Flute :: setEndReflection(StkFloat coefficient)
   endReflection_ = coefficient;
 }               
 
-void Flute :: setJetDelay(StkFloat aRatio)
+void Flute :: setJetDelay( StkFloat aRatio )
 {
   // Delay = length - approximate filter delay.
   StkFloat temp = Stk::sampleRate() / lastFrequency_ - (StkFloat) 2.0;
@@ -149,7 +148,6 @@ StkFloat Flute :: computeSample()
   // Calculate the breath pressure (envelope + noise + vibrato)
   breathPressure = maxPressure_ * adsr_.tick();
   breathPressure += breathPressure * ( noiseGain_ * noise_.tick() + vibratoGain_ * vibrato_.tick() );
-  //breathPressure += breathPressure * vibratoGain_ * vibrato_.tick();
 
   StkFloat temp = filter_.tick( boreDelay_.lastOut() );
   temp = dcBlock_.tick( temp ); // Block DC on reflection.

src/FormSwep.cpp

@@ -8,7 +8,7 @@
     It provides methods for controlling the sweep
     rate and target frequency.
 
-    by Perry R. Cook and Gary P. Scavone, 1995 - 2005.
+    by Perry R. Cook and Gary P. Scavone, 1995 - 2007.
 */
 /***************************************************/
 

src/Function.cpp

@@ -6,7 +6,7 @@
     implement tables or other types of input to output function
     mappings.
 
-    by Perry R. Cook and Gary P. Scavone, 1995 - 2005.
+    by Perry R. Cook and Gary P. Scavone, 1995 - 2007.
 */
 /***************************************************/
 

src/Generator.cpp

@@ -5,7 +5,7 @@
     This class provides common functionality for
     STK unit generator sample-source subclasses.
 
-    by Perry R. Cook and Gary P. Scavone, 1995 - 2005.
+    by Perry R. Cook and Gary P. Scavone, 1995 - 2007.
 */
 /***************************************************/
 

src/Granulate.cpp

@@ -17,6 +17,7 @@
 
 #include "Granulate.h"
 #include "FileRead.h"
+#include <cmath>
 
 Granulate :: Granulate( void )
 {
@@ -120,7 +121,7 @@ void Granulate :: reset()
 void Granulate :: setVoices( unsigned int nVoices )
 {
 #if defined(_STK_DEBUG_)
-  errorString_ << "Granulate::setGrains: nGrains = " << nGrains << ", existing grains = " << grains_.size() << '.';
+  errorString_ << "Granulate::setVoices: nVoices = " << nVoices << ", existing voices = " << grains_.size() << '.';
   handleError( StkError::DEBUG_WARNING );
 #endif
 
@@ -133,6 +134,7 @@ void Granulate :: setVoices( unsigned int nVoices )
     grains_[i].repeats = 0;
     count = (unsigned int ) ( i * gDuration_ * 0.001 * Stk::sampleRate() / nVoices );
     grains_[i].counter = count;
+    grains_[i].pointer = gPointer_;
     grains_[i].state = GRAIN_STOPPED;
   }
 
@@ -186,9 +188,13 @@ void Granulate :: calculateGrain( Granulate::Grain& grain )
 
   // Calculate offset parameter.
   seconds = gOffset_ * 0.001;
-  seconds += ( seconds * gRandomFactor_ * noise.tick() );
+  seconds += ( seconds * gRandomFactor_ * std::abs( noise.tick() ) );
   int offset = (int) ( seconds * Stk::sampleRate() );
-  grain.pointer = gPointer_ + offset;
+
+  // Add some randomization to the pointer start position.
+  seconds = gDuration_ * 0.001 * gRandomFactor_ * noise.tick();
+  offset += (int) ( seconds * Stk::sampleRate() );
+  grain.pointer += offset;
   while ( grain.pointer >= data_.frames() ) grain.pointer -= data_.frames();
   if ( grain.pointer <  0 ) grain.pointer = 0;
   grain.startPointer = grain.pointer;

src/HevyMetl.cpp

@@ -22,7 +22,7 @@
     type who should worry about this (making
     money) worry away.
 
-    by Perry R. Cook and Gary P. Scavone, 1995 - 2005.
+    by Perry R. Cook and Gary P. Scavone, 1995 - 2007.
 */
 /***************************************************/
 

src/InetWvIn.cpp

@@ -20,7 +20,7 @@
     data type for the incoming stream is signed 16-bit integers,
     though any of the defined StkFormats are permissible.
 
-    by Perry R. Cook and Gary P. Scavone, 1995 - 2005.
+    by Perry R. Cook and Gary P. Scavone, 1995 - 2007.
 */
 /***************************************************/
 

src/InetWvOut.cpp

@@ -16,7 +16,7 @@
     data type is signed 16-bit integers but any of the defined
     StkFormats are permissible.
 
-    by Perry R. Cook and Gary P. Scavone, 1995 - 2005.
+    by Perry R. Cook and Gary P. Scavone, 1995 - 2007.
 */
 /***************************************************/
 

src/Instrmnt.cpp

@@ -5,7 +5,7 @@
     This class provides a common interface for
     all STK instruments.
 
-    by Perry R. Cook and Gary P. Scavone, 1995 - 2005.
+    by Perry R. Cook and Gary P. Scavone, 1995 - 2007.
 */
 /***************************************************/
 

src/JCRev.cpp

@@ -10,7 +10,7 @@
     filters, and two decorrelation delay lines in
     parallel at the output.
 
-    by Perry R. Cook and Gary P. Scavone, 1995 - 2005.
+    by Perry R. Cook and Gary P. Scavone, 1995 - 2007.
 */
 /***************************************************/
 

src/JetTable.cpp

@@ -9,7 +9,7 @@
     Consult Fletcher and Rossing, Karjalainen,
     Cook, and others for more information.
 
-    by Perry R. Cook and Gary P. Scavone, 1995 - 2005.
+    by Perry R. Cook and Gary P. Scavone, 1995 - 2007.
 */
 /***************************************************/
 

src/Makefile.in

@@ -42,7 +42,7 @@ CFLAGS  += @warn@ $(INCLUDE)
 
 REALTIME = @realtime@
 ifeq ($(REALTIME),yes)
-	OBJECTS += RtMidi.o RtAudio.o RtWvOut.o RtWvIn.o RtDuplex.o InetWvOut.o InetWvIn.o Thread.o Mutex.o Socket.o TcpClient.o TcpServer.o UdpSocket.o
+	OBJECTS += RtMidi.o RtAudio.o RtWvOut.o RtWvIn.o InetWvOut.o InetWvIn.o Thread.o Mutex.o Socket.o TcpClient.o TcpServer.o UdpSocket.o
 	DEFS    += @audio_apis@
 endif
 

src/Mandolin.cpp

@@ -23,7 +23,7 @@
        - String Detuning = 1
        - Microphone Position = 128
 
-    by Perry R. Cook and Gary P. Scavone, 1995 - 2005.
+    by Perry R. Cook and Gary P. Scavone, 1995 - 2007.
 */
 /***************************************************/
 
@@ -58,7 +58,7 @@ Mandolin :: ~Mandolin()
     delete soundfile_[i];
 }
 
-void Mandolin :: pluck(StkFloat amplitude)
+void Mandolin :: pluck( StkFloat amplitude )
 {
   // This function gets interesting, because pluck
   // may be longer than string length, so we just

src/Messager.cpp

@@ -28,7 +28,7 @@
     This class is primarily for use in STK example programs but it is
     generic enough to work in many other contexts.
 
-    by Perry R. Cook and Gary P. Scavone, 1995 - 2005.
+    by Perry R. Cook and Gary P. Scavone, 1995 - 2007.
 */
 /***************************************************/
 

src/MidiFileIn.cpp

@@ -298,7 +298,7 @@ unsigned long MidiFileIn :: getNextEvent( std::vector<unsigned char> *event, uns
       // Update track counter and check the tempo map.
       trackCounters_[track] += ticks;
       TempoChange tempoEvent = tempoEvents_[ trackTempoIndex_[track] ];
-      if ( trackCounters_[track] >= tempoEvent.count ) {
+      if ( trackCounters_[track] >= tempoEvent.count && trackTempoIndex_[track] < tempoEvents_.size() - 1 ) {
         trackTempoIndex_[track]++;
         tickSeconds_[track] = tempoEvent.tickSeconds;
       }

src/Modal.cpp

@@ -7,7 +7,7 @@
     (non-sweeping BiQuad filters), where N is set
     during instantiation.
 
-    by Perry R. Cook and Gary P. Scavone, 1995 - 2005.
+    by Perry R. Cook and Gary P. Scavone, 1995 - 2007.
 */
 /***************************************************/
 

src/ModalBar.cpp

@@ -24,7 +24,7 @@
          - Two Fixed = 7
          - Clump = 8
 
-    by Perry R. Cook and Gary P. Scavone, 1995 - 2005.
+    by Perry R. Cook and Gary P. Scavone, 1995 - 2007.
 */
 /***************************************************/
 

src/Modulate.cpp

@@ -6,7 +6,7 @@
     modulations to give a nice, natural human
     modulation function.
 
-    by Perry R. Cook and Gary P. Scavone, 1995 - 2005.
+    by Perry R. Cook and Gary P. Scavone, 1995 - 2007.
 */
 /***************************************************/
 

src/Moog.cpp

@@ -14,7 +14,7 @@
        - Vibrato Gain = 1
        - Gain = 128
 
-    by Perry R. Cook and Gary P. Scavone, 1995 - 2005.
+    by Perry R. Cook and Gary P. Scavone, 1995 - 2007.
 */
 /***************************************************/
 

src/Mutex.cpp

@@ -7,7 +7,7 @@
     systems, the pthread library is used. Under
     Windows, critical sections are used.
 
-    by Perry R. Cook and Gary P. Scavone, 1995 - 2005.
+    by Perry R. Cook and Gary P. Scavone, 1995 - 2007.
 */
 /***************************************************/
 

src/NRev.cpp

@@ -12,7 +12,7 @@
     filters in parallel with corresponding right
     and left outputs.
 
-    by Perry R. Cook and Gary P. Scavone, 1995 - 2005.
+    by Perry R. Cook and Gary P. Scavone, 1995 - 2007.
 */
 /***************************************************/
 

src/Noise.cpp

@@ -6,7 +6,7 @@
     C rand() function.  The quality of the rand()
     function varies from one OS to another.
 
-    by Perry R. Cook and Gary P. Scavone, 1995 - 2005.
+    by Perry R. Cook and Gary P. Scavone, 1995 - 2007.
 */
 /***************************************************/
 

src/OnePole.cpp

@@ -8,7 +8,7 @@
     the real axis of the z-plane while maintaining
     a constant peak filter gain.
 
-    by Perry R. Cook and Gary P. Scavone, 1995 - 2005.
+    by Perry R. Cook and Gary P. Scavone, 1995 - 2007.
 */
 /***************************************************/
 

src/OneZero.cpp

@@ -8,7 +8,7 @@
     along the real axis of the z-plane while
     maintaining a constant filter gain.
 
-    by Perry R. Cook and Gary P. Scavone, 1995 - 2005.
+    by Perry R. Cook and Gary P. Scavone, 1995 - 2007.
 */
 /***************************************************/
 

src/PRCRev.cpp

@@ -10,7 +10,7 @@
     two series allpass units and two parallel comb
     filters.
 
-    by Perry R. Cook and Gary P. Scavone, 1995 - 2005.
+    by Perry R. Cook and Gary P. Scavone, 1995 - 2007.
 */
 /***************************************************/
 

src/PercFlut.cpp

@@ -22,7 +22,7 @@
     type who should worry about this (making
     money) worry away.
 
-    by Perry R. Cook and Gary P. Scavone, 1995 - 2005.
+    by Perry R. Cook and Gary P. Scavone, 1995 - 2007.
 */
 /***************************************************/
 

src/Phonemes.cpp

@@ -6,7 +6,7 @@
     set of 32 static phoneme formant parameters
     and provide access to those values.
 
-    by Perry R. Cook and Gary P. Scavone, 1995 - 2005.
+    by Perry R. Cook and Gary P. Scavone, 1995 - 2007.
 */
 /***************************************************/
 

src/PitShift.cpp

@@ -5,7 +5,7 @@
     This class implements a simple pitch shifter
     using delay lines.
 
-    by Perry R. Cook and Gary P. Scavone, 1995 - 2005.
+    by Perry R. Cook and Gary P. Scavone, 1995 - 2007.
 */
 /***************************************************/
 

src/PluckTwo.cpp

@@ -14,7 +14,7 @@
     use possibly subject to patents held by
     Stanford University, Yamaha, and others.
 
-    by Perry R. Cook and Gary P. Scavone, 1995 - 2005.
+    by Perry R. Cook and Gary P. Scavone, 1995 - 2007.
 */
 /***************************************************/
 

src/Plucked.cpp

@@ -13,13 +13,13 @@
     Stanford, bearing the names of Karplus and/or
     Strong.
 
-    by Perry R. Cook and Gary P. Scavone, 1995 - 2005.
+    by Perry R. Cook and Gary P. Scavone, 1995 - 2007.
 */
 /***************************************************/
 
 #include "Plucked.h"
 
-Plucked :: Plucked(StkFloat lowestFrequency)
+Plucked :: Plucked( StkFloat lowestFrequency )
 {
   length_ = (unsigned long) (Stk::sampleRate() / lowestFrequency + 1);
   loopGain_ = 0.999;
@@ -39,7 +39,7 @@ void Plucked :: clear()
   pickFilter_.clear();
 }
 
-void Plucked :: setFrequency(StkFloat frequency)
+void Plucked :: setFrequency( StkFloat frequency )
 {
   StkFloat freakency = frequency;
   if ( frequency <= 0.0 ) {
@@ -60,7 +60,7 @@ void Plucked :: setFrequency(StkFloat frequency)
   if ( loopGain_ >= 1.0 ) loopGain_ = 0.99999;
 }
 
-void Plucked :: pluck(StkFloat amplitude)
+void Plucked :: pluck( StkFloat amplitude )
 {
   StkFloat gain = amplitude;
   if ( gain > 1.0 ) {
@@ -81,7 +81,7 @@ void Plucked :: pluck(StkFloat amplitude)
     delayLine_.tick( 0.6 * delayLine_.lastOut() + pickFilter_.tick( noise_.tick() ) );
 }
 
-void Plucked :: noteOn(StkFloat frequency, StkFloat amplitude)
+void Plucked :: noteOn( StkFloat frequency, StkFloat amplitude )
 {
   this->setFrequency( frequency );
   this->pluck( amplitude );

src/PoleZero.cpp

@@ -8,7 +8,7 @@
     filter with a given coefficient.  Another
     method is provided to create a DC blocking filter.
 
-    by Perry R. Cook and Gary P. Scavone, 1995 - 2005.
+    by Perry R. Cook and Gary P. Scavone, 1995 - 2007.
 */
 /***************************************************/
 

src/ReedTable.cpp

@@ -13,7 +13,7 @@
     Smith (1986), Hirschman, Cook, Scavone, and
     others for more information.
 
-    by Perry R. Cook and Gary P. Scavone, 1995 - 2005.
+    by Perry R. Cook and Gary P. Scavone, 1995 - 2007.
 */
 /***************************************************/
 

src/Resonate.cpp

@@ -13,7 +13,7 @@
        - Zero Radii = 1
        - Envelope Gain = 128
 
-    by Perry R. Cook and Gary P. Scavone, 1995 - 2005.
+    by Perry R. Cook and Gary P. Scavone, 1995 - 2007.
 */
 /***************************************************/
 

src/Rhodey.cpp

@@ -26,7 +26,7 @@
     type who should worry about this (making
     money) worry away.
 
-    by Perry R. Cook and Gary P. Scavone, 1995 - 2005.
+    by Perry R. Cook and Gary P. Scavone, 1995 - 2007.
 */
 /***************************************************/
 

src/RtDuplex.cpp

@@ -1,245 +0,0 @@
-/***************************************************/
-/*! \class RtDuplex
-    \brief STK realtime audio (blocking) input/output class.
-
-    This class provides a simplified interface to
-    RtAudio for realtime audio input/output.  It
-    may also be possible to achieve duplex
-    operation using separate RtWvIn and RtWvOut
-    classes, but this class ensures better
-    input/output synchronization.
-
-    Because this class makes use of RtAudio's
-    blocking input/output routines, its
-    performance is less robust on systems where
-    the audio API is callback-based (Macintosh
-    CoreAudio and Windows ASIO).
-
-    RtDuplex supports multi-channel data in
-    interleaved format.  It is important to
-    distinguish the tick() methods, which output
-    single samples to all channels in a sample
-    frame and return samples produced by averaging
-    across sample frames, from the tickFrame()
-    methods, which take/return pointers to
-    multi-channel sample frames.
-
-    by Perry R. Cook and Gary P. Scavone, 1995 - 2005.
-*/
-/***************************************************/
-
-#include "RtDuplex.h"
-
-RtDuplex :: RtDuplex(int nChannels, StkFloat sampleRate, int device, int bufferFrames, int nBuffers )
-{
-  channels_ = nChannels;
-  bufferSize_ = bufferFrames;
-  RtAudioFormat format = ( sizeof(StkFloat) == 8 ) ? RTAUDIO_FLOAT64 : RTAUDIO_FLOAT32;
-
-  audio_ = 0;
-  try {
-    audio_ = new RtAudio();
-  }
-  catch (RtError &error) {
-    handleError( error.getMessageString(), StkError::AUDIO_SYSTEM );
-  }
-
-  // Now open a stream and get the buffer pointer.
-  try {
-    audio_->openStream(device, channels_, device, channels_, format,
-                      (int)sampleRate, &bufferSize_, nBuffers);
-    data_ = (StkFloat *) audio_->getStreamBuffer();
-  }
-  catch (RtError &error) {
-    handleError( error.getMessageString(), StkError::AUDIO_SYSTEM );
-  }
-
-  lastOutput_ = (StkFloat *) new StkFloat[channels_];
-  for (unsigned int i=0; i<channels_; i++) lastOutput_[i] = 0.0;
-  counter_ = 0;
-  stopped_ = true;
-}
-
-RtDuplex :: ~RtDuplex()
-{
-  if ( !stopped_ )
-    audio_->stopStream();
-  delete audio_;
-  delete [] lastOutput_;
-  data_ = 0; // RtAudio deletes the buffer itself.
-}
-
-void RtDuplex :: start()
-{
-  if ( stopped_ ) {
-    audio_->startStream();
-    stopped_ = false;
-  }
-}
-
-void RtDuplex :: stop()
-{
-  if ( !stopped_ ) {
-    audio_->stopStream();
-    stopped_ = true;
-  }
-}
-
-StkFloat RtDuplex :: lastOut(void) const
-{
-  if ( channels_ == 1 )
-    return *lastOutput_;
-
-  StkFloat output = 0.0;
-  for (unsigned int i=0; i<channels_; i++ ) {
-    output += lastOutput_[i];
-  }
-  return output / channels_;
-}
-
-StkFloat RtDuplex :: tick(const StkFloat sample)
-{
-  if ( stopped_ )
-    start();
-
-  if (counter_ == 0) {
-    try {
-      audio_->tickStream();
-    }
-    catch (RtError &error) {
-      handleError( error.getMessageString(), StkError::AUDIO_SYSTEM );
-    }
-  }
-
-  unsigned long temp = counter_ * channels_;
-  for (unsigned int i=0; i<channels_; i++) {
-    lastOutput_[i] = data_[temp];
-    data_[temp++] = sample;
-  }
-
-  counter_++;
-  if (counter_ >= (long) bufferSize_)
-    counter_ = 0;
-
-  return lastOut();
-}
-
-StkFloat *RtDuplex :: tick(StkFloat *vector, unsigned int vectorSize)
-{
-  for ( unsigned int i=0; i<vectorSize; i++ )
-    vector[i] = tick(vector[i]);
-
-  return vector;
-}
-
-StkFrames& RtDuplex :: tick( StkFrames& frames, unsigned int channel )
-{
-  if ( channel == 0 || frames.channels() < channel ) {
-    errorString_ << "RtDuplex::tick(): channel argument (" << channel << ") is zero or > channels in StkFrames argument!";
-    handleError( StkError::FUNCTION_ARGUMENT );
-  }
-
-  if ( frames.channels() == 1 ) {
-    for ( unsigned int i=0; i<frames.frames(); i++ )
-      frames[i] = tick( frames[i] );
-  }
-  else if ( frames.interleaved() ) {
-    unsigned int hop = frames.channels();
-    unsigned int index = channel - 1;
-    for ( unsigned int i=0; i<frames.frames(); i++ ) {
-      frames[index] = tick( frames[index] );
-      index += hop;
-    }
-  }
-  else {
-    unsigned int iStart = (channel - 1) * frames.frames();
-    for ( unsigned int i=0; i<frames.frames(); i++ )
-      frames[iStart + i] = tick( frames[iStart + i] );
-  }
-
-  return frames;
-}
-
-const StkFloat *RtDuplex :: lastFrame() const
-{
-  return lastOutput_;
-}
-
-StkFloat *RtDuplex :: tickFrame(StkFloat *frameVector, unsigned int frames)
-{
-  if ( stopped_ )
-    start();
-
-  unsigned int i;
-  unsigned long temp;
-  for (unsigned int j=0; j<frames; j++ ) {
-    if (counter_ == 0) {
-      try {
-        audio_->tickStream();
-      }
-      catch (RtError &error) {
-        handleError( error.getMessageString(), StkError::AUDIO_SYSTEM );
-      }
-    }
-
-    temp = counter_ * channels_;
-    for (i=0; i<channels_; i++) {
-      lastOutput_[i] = data_[temp];
-      data_[temp++] = frameVector[j*channels_+i];
-      frameVector[j*channels_+i] = lastOutput_[i];
-    }
-
-    counter_++;
-    if (counter_ >= (long) bufferSize_)
-      counter_ = 0;
-  }
-
-  return frameVector;
-}
-
-StkFrames& RtDuplex :: tickFrame( StkFrames& frames )
-{
-  if ( channels_ != frames.channels() ) {
-    errorString_ << "RtDuplex::tickFrame(): incompatible channel value in StkFrames argument!";
-    handleError( StkError::FUNCTION_ARGUMENT );
-  }
-
-  if ( stopped_ )
-    start();
-
-  unsigned long j, index = 0;
-  unsigned long k, hop = frames.frames();
-  for (unsigned int i=0; i<frames.frames(); i++ ) {
-    if (counter_ == 0) {
-      try {
-        audio_->tickStream();
-      }
-      catch (RtError &error) {
-        handleError( error.getMessageString(), StkError::AUDIO_SYSTEM );
-      }
-    }
-
-    if ( channels_ > 1 && frames.interleaved() == false ) {
-      k = i;
-      for (j=0; j<channels_; j++) {
-        lastOutput_[j] = data_[index];
-        data_[index++] = frames[k];
-        frames[k] = lastOutput_[j];
-        k += hop;
-      }
-    }
-    else {
-      for (j=0; j<channels_; j++) {
-        lastOutput_[j] = data_[index];
-        data_[index] = frames[index];
-        frames[index++] = lastOutput_[j];
-      }
-    }
-
-    counter_++;
-    if (counter_ >= (long) bufferSize_)
-      counter_ = 0;
-  }
-
-  return frames;
-}

src/RtMidi.cpp

@@ -35,7 +35,7 @@
 */
 /**********************************************************************/
 
-// RtMidi: Version 1.0.4, 14 October 2005
+// RtMidi: Version 1.0.5, in development
 
 #include "RtMidi.h"
 #include <sstream>
@@ -383,13 +383,15 @@ void RtMidiIn :: openPort( unsigned int portNumber )
   connected_ = true;
 }
 
-void RtMidiIn :: openVirtualPort()
+void RtMidiIn :: openVirtualPort( const std::string portName )
 {
   CoreMidiData *data = static_cast<CoreMidiData *> (apiData_);
 
   // Create a virtual MIDI input destination.
   MIDIEndpointRef endpoint;
-  OSStatus result = MIDIDestinationCreate( data->client, CFSTR("RtMidi Input"), midiInputCallback, (void *)&inputData_, &endpoint );
+  OSStatus result = MIDIDestinationCreate( data->client,
+                                           CFStringCreateWithCString( NULL, portName.c_str(), kCFStringEncodingASCII ),
+                                           midiInputCallback, (void *)&inputData_, &endpoint );
   if ( result != noErr ) {
     errorString_ = "RtMidiIn::openVirtualPort: error creating virtual OS-X MIDI destination.";
     error( RtError::DRIVER_ERROR );
@@ -548,7 +550,7 @@ void RtMidiOut :: closePort( void )
   }
 }
 
-void RtMidiOut :: openVirtualPort()
+void RtMidiOut :: openVirtualPort( std::string portName )
 {
   CoreMidiData *data = static_cast<CoreMidiData *> (apiData_);
 
@@ -560,7 +562,9 @@ void RtMidiOut :: openVirtualPort()
 
   // Create a virtual MIDI output source.
   MIDIEndpointRef endpoint;
-  OSStatus result = MIDISourceCreate( data->client, CFSTR("RtMidi Output"), &endpoint );
+  OSStatus result = MIDISourceCreate( data->client,
+                                      CFStringCreateWithCString( NULL, portName.c_str(), kCFStringEncodingASCII ),
+                                      &endpoint );
   if ( result != noErr ) {
     errorString_ = "RtMidiOut::initialize: error creating OS-X virtual MIDI source.";
     error( RtError::DRIVER_ERROR );
@@ -957,7 +961,7 @@ void RtMidiIn :: openPort( unsigned int portNumber )
   connected_ = true;
 }
 
-void RtMidiIn :: openVirtualPort()
+void RtMidiIn :: openVirtualPort( std::string portName )
 {
   AlsaMidiData *data = static_cast<AlsaMidiData *> (apiData_);
   if ( data->vport < 0 ) {
@@ -973,7 +977,7 @@ void RtMidiIn :: openVirtualPort()
     snd_seq_port_info_set_timestamping(pinfo, 1);
     snd_seq_port_info_set_timestamp_real(pinfo, 1);    
     snd_seq_port_info_set_timestamp_queue(pinfo, data->queue_id);
-    snd_seq_port_info_set_name(pinfo, "RtMidi Input");
+    snd_seq_port_info_set_name(pinfo, portName.c_str());
     data->vport = snd_seq_create_port(data->seq, pinfo);
 
     if ( data->vport < 0 ) {
@@ -1195,11 +1199,11 @@ void RtMidiOut :: closePort( void )
   }
 }
 
-void RtMidiOut :: openVirtualPort()
+void RtMidiOut :: openVirtualPort( std::string portName )
 {
   AlsaMidiData *data = static_cast<AlsaMidiData *> (apiData_);
   if ( data->vport < 0 ) {
-    data->vport = snd_seq_create_simple_port( data->seq, "RtMidi Output",
+    data->vport = snd_seq_create_simple_port( data->seq, portName.c_str(),
                                               SND_SEQ_PORT_CAP_READ|SND_SEQ_PORT_CAP_SUBS_READ,
                                               SND_SEQ_PORT_TYPE_MIDI_GENERIC );
 
@@ -1484,7 +1488,7 @@ void RtMidiIn :: openPort( unsigned int portNumber )
   connected_ = true;
 }
 
-void RtMidiIn :: openVirtualPort()
+void RtMidiIn :: openVirtualPort( std::string portName )
 {
   // This function cannot be implemented for the Irix MIDI API.
   errorString_ = "RtMidiIn::openVirtualPort: cannot be implemented in Irix MIDI API!";
@@ -1617,7 +1621,7 @@ void RtMidiOut :: closePort( void )
   }
 }
 
-void RtMidiOut :: openVirtualPort()
+void RtMidiOut :: openVirtualPort( std::string portName )
 {
   // This function cannot be implemented for the Irix MIDI API.
   errorString_ = "RtMidiOut::openVirtualPort: cannot be implemented in Irix MIDI API!";
@@ -1676,6 +1680,8 @@ void RtMidiOut :: sendMessage( std::vector<unsigned char> *message )
 // API information deciphered from:
 //  - http://msdn.microsoft.com/library/default.asp?url=/library/en-us/multimed/htm/_win32_midi_reference.asp
 
+// Thanks to Jean-Baptiste Berruchon for the sysex code.
+
 #if defined(__WINDOWS_MM__)
 
 // The Windows MM API is based on the use of a callback function for
@@ -1693,8 +1699,11 @@ struct WinMidiData {
   HMIDIOUT outHandle;  // Handle to Midi Output Device
   DWORD lastTime;
   RtMidiIn::MidiMessage message;
+  LPMIDIHDR sysexBuffer;
 };
 
+#define  RT_SYSEX_BUFFER_SIZE 1024
+
 //*********************************************************************//
 //  API: Windows MM
 //  Class Definitions: RtMidiIn
@@ -1748,11 +1757,21 @@ static void CALLBACK midiInputCallback( HMIDIOUT hmin,
     unsigned char *ptr = (unsigned char *) &midiMessage;
     for ( int i=0; i<nBytes; i++ ) apiData->message.bytes.push_back( *ptr++ );
   }
-  else { // Sysex message
+  else if ( !(data->ignoreFlags & 0x01) ) {
+    // Sysex message and we're not ignoring it
     MIDIHDR *sysex = ( MIDIHDR *) midiMessage;
     for ( int i=0; i<(int)sysex->dwBytesRecorded; i++ )
       apiData->message.bytes.push_back( sysex->lpData[i] );
-    if ( apiData->message.bytes.back() != 0xF7 ) return;
+
+    // When the callback has to be unaffected (application closes), 
+    // it seems WinMM calls it with an empty sysex to de-queue the buffer
+    // If the buffer is requeued afer that message, the PC suddenly reboots
+    // after one or two minutes (JB).
+    if ( apiData->sysexBuffer->dwBytesRecorded > 0 ) {
+      MMRESULT result = midiInAddBuffer( apiData->inHandle, apiData->sysexBuffer, sizeof(MIDIHDR) );
+      if ( result != MMSYSERR_NOERROR )
+        std::cerr << "\nRtMidiIn::midiInputCallback: error sending sysex to Midi device!!\n\n";
+    }
   }
 
   if ( data->usingCallback ) {
@@ -1822,6 +1841,27 @@ void RtMidiIn :: openPort( unsigned int portNumber )
     error( RtError::DRIVER_ERROR );
   }
 
+  // Allocate and init the sysex buffer.
+  data->sysexBuffer = (MIDIHDR*) new char[ sizeof(MIDIHDR) ];
+  data->sysexBuffer->lpData = new char[1024];
+  data->sysexBuffer->dwBufferLength = 1024;
+  data->sysexBuffer->dwFlags = 0;
+
+  result = midiInPrepareHeader( data->inHandle, data->sysexBuffer, sizeof(MIDIHDR) );
+  if ( result != MMSYSERR_NOERROR ) {
+    midiInClose( data->inHandle );
+    errorString_ = "RtMidiIn::openPort: error starting Windows MM MIDI input port (PrepareHeader).";
+    error( RtError::DRIVER_ERROR );
+  }
+
+  // Register the buffer.
+  result = midiInAddBuffer( data->inHandle, data->sysexBuffer, sizeof(MIDIHDR) );
+  if ( result != MMSYSERR_NOERROR ) {
+    midiInClose( data->inHandle );
+    errorString_ = "RtMidiIn::openPort: error starting Windows MM MIDI input port (AddBuffer).";
+    error( RtError::DRIVER_ERROR );
+  }
+
   result = midiInStart( data->inHandle );
   if ( result != MMSYSERR_NOERROR ) {
     midiInClose( data->inHandle );
@@ -1832,7 +1872,7 @@ void RtMidiIn :: openPort( unsigned int portNumber )
   connected_ = true;
 }
 
-void RtMidiIn :: openVirtualPort()
+void RtMidiIn :: openVirtualPort( std::string portName )
 {
   // This function cannot be implemented for the Windows MM MIDI API.
   errorString_ = "RtMidiIn::openVirtualPort: cannot be implemented in Windows MM MIDI API!";
@@ -1845,6 +1885,16 @@ void RtMidiIn :: closePort( void )
     WinMidiData *data = static_cast<WinMidiData *> (apiData_);
     midiInReset( data->inHandle );
     midiInStop( data->inHandle );
+
+    int result = midiInUnprepareHeader(data->inHandle, data->sysexBuffer, sizeof(MIDIHDR));
+    delete [] data->sysexBuffer->lpData;
+    delete [] data->sysexBuffer;
+    if ( result != MMSYSERR_NOERROR ) {
+      midiInClose( data->inHandle );
+      errorString_ = "RtMidiIn::openPort: error closing Windows MM MIDI input port (midiInUnprepareHeader).";
+      error( RtError::DRIVER_ERROR );
+    }
+
     midiInClose( data->inHandle );
     connected_ = false;
   }
@@ -1878,7 +1928,14 @@ std::string RtMidiIn :: getPortName( unsigned int portNumber )
   MIDIINCAPS deviceCaps;
   MMRESULT result = midiInGetDevCaps( portNumber, &deviceCaps, sizeof(MIDIINCAPS));
 
-  std::string stringName = std::string( deviceCaps.szPname );
+  // For some reason, we need to copy character by character with
+  // UNICODE (thanks to Eduardo Coutinho!).
+  //std::string stringName = std::string( deviceCaps.szPname );
+  char nameString[MAXPNAMELEN];
+  for( int i=0; i<MAXPNAMELEN; i++ )
+    nameString[i] = (char)( deviceCaps.szPname[i] );
+
+  std::string stringName( nameString );
   return stringName;
 }
 
@@ -1905,7 +1962,14 @@ std::string RtMidiOut :: getPortName( unsigned int portNumber )
   MIDIOUTCAPS deviceCaps;
   MMRESULT result = midiOutGetDevCaps( portNumber, &deviceCaps, sizeof(MIDIOUTCAPS));
 
-  std::string stringName = std::string( deviceCaps.szPname );
+  // For some reason, we need to copy character by character with
+  // UNICODE (thanks to Eduardo Coutinho!).
+  //std::string stringName = std::string( deviceCaps.szPname );
+  char nameString[MAXPNAMELEN];
+  for( int i=0; i<MAXPNAMELEN; i++ )
+    nameString[i] = (char)( deviceCaps.szPname[i] );
+
+  std::string stringName( nameString );
   return stringName;
 }
 
@@ -1969,7 +2033,7 @@ void RtMidiOut :: closePort( void )
   }
 }
 
-void RtMidiOut :: openVirtualPort()
+void RtMidiOut :: openVirtualPort( std::string portName )
 {
   // This function cannot be implemented for the Windows MM MIDI API.
   errorString_ = "RtMidiOut::openVirtualPort: cannot be implemented in Windows MM MIDI API!";

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;
 }

src/RtWvOut.cpp

@@ -3,62 +3,124 @@
     \brief STK realtime audio (blocking) output class.
 
     This class provides a simplified interface to RtAudio for realtime
-    audio output.  It is a subclass of WvOut.  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 output.  It is a subclass of WvOut.  This class makes use of
+    RtAudio's callback functionality by creating a large ring-buffer
+    into which data is written.  This class should not be used when
+    low-latency is desired.
 
-    RtWvOut supports multi-channel data in interleaved format.  It is
-    important to distinguish the tick() methods, which output single
-    samples to all channels in a sample frame, from the tickFrame()
-    method, which take a pointer or reference to multi-channel sample
-    frame data.
+    RtWvOut supports multi-channel data in both interleaved and
+    non-interleaved formats.  It is important to distinguish the
+    tick() methods, which output single samples to all channels in a
+    sample frame, from the tickFrame() method, which take a pointer or
+    reference to multi-channel sample frame data.
 
-    by Perry R. Cook and Gary P. Scavone, 1995 - 2005.
+    by Perry R. Cook and Gary P. Scavone, 1995 - 2007.
 */
 /***************************************************/
 
 #include "RtWvOut.h"
 
+// Streaming status states.
+enum { RUNNING, EMPTYING, FINISHED };
+
+// This function is automatically called by RtAudio to get audio data for output.
+int write( void *outputBuffer, void *inputBuffer, unsigned int nBufferFrames,
+           double streamTime, RtAudioStreamStatus status, void *dataPointer )
+{
+  return ( (RtWvOut *) dataPointer )->readBuffer( outputBuffer, nBufferFrames );
+}
+
+// This function does not block.  If the user does not write output
+// data to the buffer fast enough, previous data will be re-output
+// (data underrun).
+int RtWvOut :: readBuffer( void *buffer, unsigned int frameCount )
+{
+  unsigned int nSamples, nChannels = data_.channels();
+  unsigned int nFrames = frameCount;
+  StkFloat *input = (StkFloat *) &data_[ readIndex_ * nChannels ];
+  StkFloat *output = (StkFloat *) buffer;
+  long counter;
+
+  while ( nFrames > 0 ) {
+
+    // I'm assuming that both the RtAudio and StkFrames buffers
+    // contain interleaved data.
+    counter = nFrames;
+
+    // Pre-increment read pointer and check bounds.
+    readIndex_ += nFrames;
+    if ( readIndex_ >= data_.frames() ) {
+      counter -= readIndex_ - data_.frames();
+      readIndex_ = 0;
+    }
+
+    // Copy data from the StkFrames container.
+    if ( status_ == EMPTYING && framesFilled_ <= counter ) {
+      nSamples = framesFilled_ * nChannels;
+      unsigned int i;
+      for ( i=0; i<nSamples; i++ ) *output++ = *input++;
+      nSamples = (counter - framesFilled_) * nChannels;
+      for ( i=0; i<nSamples; i++ ) *output++ = 0.0;
+      status_ = FINISHED;
+      return 1;
+    }
+    else {
+      nSamples = counter * nChannels;
+      for ( unsigned int i=0; i<nSamples; i++ )
+        *output++ = *input++;
+    }
+
+    nFrames -= counter;
+  }
+
+  framesFilled_ -= frameCount;
+  if ( framesFilled_ < 0 ) {
+    framesFilled_ = 0;
+    //    writeIndex_ = readIndex_;
+    errorString_ << "RtWvOut: audio buffer underrun!";
+    handleError( StkError::WARNING );
+  }
+
+  return 0;
+}
+
+
 RtWvOut :: RtWvOut( unsigned int nChannels, StkFloat sampleRate, int device, int bufferFrames, int nBuffers )
-  : stopped_( true ), nChannels_(nChannels), bufferIndex_( 0 ), iBuffer_( 0 )
+  : stopped_( true ), readIndex_( 0 ), writeIndex_( 0 ), framesFilled_( 0 ), status_(0)
 {
   // We'll let RtAudio deal with channel and sample rate limitations.
-  int size = bufferFrames;
+  RtAudio::StreamParameters parameters;
+  parameters.deviceId = device;
+  parameters.nChannels = nChannels;
+  unsigned int size = bufferFrames;
   RtAudioFormat format = ( sizeof(StkFloat) == 8 ) ? RTAUDIO_FLOAT64 : RTAUDIO_FLOAT32;
 
-  dac_ = 0;
+  // Open a stream and set the callback function.
   try {
-    dac_ = new RtAudio();
+    dac_.openStream( &parameters, NULL, format, (unsigned int)Stk::sampleRate(), &size, &write, (void *)this );
   }
-  catch (RtError &error) {
-    handleError( error.getMessageString(), StkError::AUDIO_SYSTEM );
+  catch ( RtError &error ) {
+    handleError( error.what(), StkError::AUDIO_SYSTEM );
   }
 
-  // Now open a stream and get the buffer pointer.
-  try {
-    dac_->openStream( device, (int)nChannels, 0, 0, format,
-                      (int)sampleRate, &size, nBuffers );
-    buffer_ = (StkFloat *) dac_->getStreamBuffer();
-  }
-  catch (RtError &error) {
-    handleError( error.getMessageString(), StkError::AUDIO_SYSTEM );
-  }
-
-  bufferFrames_ = size;
+  data_.resize( size * nBuffers, nChannels );
+  unsigned int offset = (unsigned int ) (data_.size() / 2.0);
+  writeIndex_ = offset;    // start writing half-way into buffer
+  framesFilled_ = offset;
 }
 
 RtWvOut :: ~RtWvOut()
 {
-  if ( !stopped_ ) dac_->stopStream();
-  dac_->closeStream();
-  delete dac_;
+  // Change status flag to signal callback to clear the buffer and close.
+  status_ = EMPTYING;
+  while ( status_ != FINISHED || dac_.isStreamRunning() == true ) Stk::sleep( 100 );
+  dac_.closeStream();
 }
 
 void RtWvOut :: start()
 {
   if ( stopped_ ) {
-    dac_->startStream();
+    dac_.startStream();
     stopped_ = false;
   }
 }
@@ -66,77 +128,81 @@ void RtWvOut :: start()
 void RtWvOut :: stop()
 {
   if ( !stopped_ ) {
-    dac_->stopStream();
+    dac_.stopStream();
     stopped_ = true;
   }
 }
 
-void RtWvOut :: incrementFrame( void )
-{
-  frameCounter_++;
-  bufferIndex_++;
-
-  if ( bufferIndex_ == bufferFrames_ ) {
-    try {
-      dac_->tickStream();
-    }
-    catch (RtError &error) {
-      handleError( error.getMessageString(), StkError::AUDIO_SYSTEM );
-    }
-    bufferIndex_ = 0;
-    iBuffer_ = 0;
-  }
-}
-
 void RtWvOut :: computeSample( const StkFloat sample )
 {
-  if ( stopped_ ) start();
+  if ( stopped_ ) this->start();
 
+  // Block until we have room for at least one frame of output data.
+  while ( framesFilled_ == (long) data_.frames() ) Stk::sleep( 1 );
+
+  unsigned int nChannels = data_.channels();
   StkFloat input = sample;
   clipTest( input );
-  for ( unsigned int j=0; j<nChannels_; j++ )
-    buffer_[iBuffer_++] = input;
+  unsigned long index = writeIndex_ * nChannels;
+  for ( unsigned int j=0; j<nChannels; j++ )
+    data_[index++] = input;
 
-  this->incrementFrame();
+  framesFilled_++;
+  frameCounter_++;
+  writeIndex_++;
+  if ( writeIndex_ == data_.frames() )
+    writeIndex_ = 0;
 }
 
 void RtWvOut :: computeFrames( const StkFrames& frames )
 {
-  if ( stopped_ ) start();
-
-  if ( frames.channels() != nChannels_ ) {
+  if ( data_.channels() != frames.channels() ) {
     errorString_ << "RtWvOut::computeFrames(): incompatible channel value in StkFrames argument!";
     handleError( StkError::FUNCTION_ARGUMENT );
   }
 
-  unsigned int j;
-  if ( nChannels_ == 1 || frames.interleaved() ) {
+  if ( stopped_ ) this->start();
 
-    unsigned int iFrames = 0;
+  // Block until we have room for the frames of output data.
+  while ( data_.frames() - framesFilled_ < frames.frames() ) Stk::sleep( 1 );
+
+  unsigned int j, nChannels = data_.channels();
+  if ( nChannels == 1 || frames.interleaved() ) {
+
+    unsigned int index, iFrames = 0;
     for ( unsigned int i=0; i<frames.frames(); i++ ) {
 
-      for ( j=0; j<nChannels_; j++ ) {
-        buffer_[iBuffer_] = frames[iFrames++];
-        clipTest( buffer_[iBuffer_++] );
-      }
+      index = writeIndex_ * nChannels;
+      for ( j=0; j<nChannels; j++ )
+        data_[index] = frames[iFrames++];
+        clipTest( data_[index++] );
 
-      this->incrementFrame();
+      framesFilled_++;
+      frameCounter_++;
+      writeIndex_++;
+      if ( writeIndex_ == data_.frames() )
+        writeIndex_ = 0;
     }
   }
   else { // non-interleaved frames
 
     unsigned long hop = frames.frames();
-    unsigned int index;
+    unsigned int index, iFrame;
     for ( unsigned int i=0; i<frames.frames(); i++ ) {
 
-      index = i;
-      for ( j=0; j<nChannels_; j++ ) {
-        buffer_[iBuffer_] = frames[index];
-        clipTest( buffer_[iBuffer_++] );
-        index += hop;
+      iFrame = i;
+      index = writeIndex_ * nChannels;
+      for ( j=0; j<nChannels; j++ ) {
+        data_[index] = frames[iFrame];
+        clipTest( data_[index++] );
+        iFrame += hop;
       }
 
-      this->incrementFrame();
+      framesFilled_++;
+      frameCounter_++;
+      writeIndex_++;
+      if ( writeIndex_ == data_.frames() )
+        writeIndex_ = 0;
     }
   }
 }

src/Sampler.cpp

@@ -5,7 +5,7 @@
     This instrument provides an ADSR envelope, a one-pole filter, and
     structures for an arbitrary number of attack and loop waves.
 
-    by Perry R. Cook and Gary P. Scavone, 1995 - 2005.
+    by Perry R. Cook and Gary P. Scavone, 1995 - 2007.
 */
 /***************************************************/
 

src/Saxofony.cpp

@@ -31,7 +31,7 @@
        - Vibrato Gain = 1
        - Breath Pressure = 128
 
-    by Perry R. Cook and Gary P. Scavone, 1995 - 2005.
+    by Perry R. Cook and Gary P. Scavone, 1995 - 2007.
 */
 /***************************************************/
 

src/Simple.cpp

@@ -13,7 +13,7 @@
        - Envelope Rate = 11
        - Gain = 128
 
-    by Perry R. Cook and Gary P. Scavone, 1995 - 2005.
+    by Perry R. Cook and Gary P. Scavone, 1995 - 2007.
 */
 /***************************************************/
 

src/SineWave.cpp

@@ -9,7 +9,7 @@
 
     The "table" length, set in SineWave.h, is 2048 samples by default.
 
-    by Perry R. Cook and Gary P. Scavone, 1995 - 2005.
+    by Perry R. Cook and Gary P. Scavone, 1995 - 2007.
 */
 /***************************************************/
 
@@ -27,10 +27,19 @@ SineWave :: SineWave( void )
     for ( unsigned long i=0; i<=TABLE_SIZE; i++ )
       table_[i] = sin( TWO_PI * i * temp );
   }
+
+  Stk::addSampleRateAlert( this );
 }
 
 SineWave :: ~SineWave()
 {
+  Stk::removeSampleRateAlert( this );
+}
+
+void SineWave :: sampleRateChanged( StkFloat newRate, StkFloat oldRate )
+{
+  if ( !ignoreSampleRateChange_ )
+    this->setRate( oldRate * rate_ / newRate );
 }
 
 void SineWave :: reset(void)

src/SingWave.cpp

@@ -9,7 +9,7 @@
     from pitch shifting.  It will be used as an
     excitation source for other instruments.
 
-    by Perry R. Cook and Gary P. Scavone, 1995 - 2005.
+    by Perry R. Cook and Gary P. Scavone, 1995 - 2007.
 */
 /***************************************************/
 

src/Sitar.cpp

@@ -13,7 +13,7 @@
     Stanford, bearing the names of Karplus and/or
     Strong.
 
-    by Perry R. Cook and Gary P. Scavone, 1995 - 2005.
+    by Perry R. Cook and Gary P. Scavone, 1995 - 2007.
 */
 /***************************************************/
 

src/Skini.cpp

@@ -19,7 +19,7 @@
 
     See also SKINI.txt.
 
-    by Perry R. Cook and Gary P. Scavone, 1995 - 2005.
+    by Perry R. Cook and Gary P. Scavone, 1995 - 2007.
 */
 /***************************************************/
 

src/Socket.cpp

@@ -5,7 +5,7 @@
     This class provides common functionality for TCP and UDP internet
     socket server and client subclasses.
 
-    by Perry R. Cook and Gary P. Scavone, 1995 - 2005.
+    by Perry R. Cook and Gary P. Scavone, 1995 - 2007.
 */
 /***************************************************/
 

src/StifKarp.cpp

@@ -17,7 +17,7 @@
        - String Sustain = 11
        - String Stretch = 1
 
-    by Perry R. Cook and Gary P. Scavone, 1995 - 2005.
+    by Perry R. Cook and Gary P. Scavone, 1995 - 2007.
 */
 /***************************************************/
 

src/Stk.cpp

@@ -8,7 +8,31 @@
     provides error handling and byte-swapping
     functions.
 
-    by Perry R. Cook and Gary P. Scavone, 1995 - 2005.
+    by Perry R. Cook and Gary P. Scavone, 1995 - 2007.
+
+    Permission is hereby granted, free of charge, to any person
+    obtaining a copy of this software and associated documentation files
+    (the "Software"), to deal in the Software without restriction,
+    including without limitation the rights to use, copy, modify, merge,
+    publish, distribute, sublicense, and/or sell copies of the Software,
+    and to permit persons to whom the Software is furnished to do so,
+    subject to the following conditions:
+
+    The above copyright notice and this permission notice shall be
+    included in all copies or substantial portions of the Software.
+
+    Any person wishing to distribute modifications to the Software is
+    asked to send the modifications to the original developer so that
+    they can be incorporated into the canonical version.  This is,
+    however, not a binding provision of this license.
+
+    THE SOFTWARE IS PROVIDED "AS IS", WITHOUT WARRANTY OF ANY KIND,
+    EXPRESS OR IMPLIED, INCLUDING BUT NOT LIMITED TO THE WARRANTIES OF
+    MERCHANTABILITY, FITNESS FOR A PARTICULAR PURPOSE AND NONINFRINGEMENT.
+    IN NO EVENT SHALL THE AUTHORS OR COPYRIGHT HOLDERS BE LIABLE FOR
+    ANY CLAIM, DAMAGES OR OTHER LIABILITY, WHETHER IN AN ACTION OF
+    CONTRACT, TORT OR OTHERWISE, ARISING FROM, OUT OF OR IN CONNECTION
+    WITH THE SOFTWARE OR THE USE OR OTHER DEALINGS IN THE SOFTWARE.
 */
 /***************************************************/
 
@@ -22,17 +46,55 @@ const Stk::StkFormat Stk :: STK_SINT24  = 0x4;
 const Stk::StkFormat Stk :: STK_SINT32  = 0x8;
 const Stk::StkFormat Stk :: STK_FLOAT32 = 0x10;
 const Stk::StkFormat Stk :: STK_FLOAT64 = 0x20;
-bool Stk :: showWarnings_ = false;
+bool Stk :: showWarnings_ = true;
 bool Stk :: printErrors_ = true;
+std::vector<Stk *> Stk :: alertList_;
 
-Stk :: Stk(void)
+Stk :: Stk( void )
+  : ignoreSampleRateChange_(false)
 {
 }
 
-Stk :: ~Stk(void)
+Stk :: ~Stk( void )
 {
 }
 
+void Stk :: setSampleRate( StkFloat rate )
+{
+  if ( rate > 0.0 && rate != srate_ ) {
+    StkFloat oldRate = srate_;
+    srate_ = rate;
+
+    for ( unsigned int i=0; i<alertList_.size(); i++ )
+      alertList_[i]->sampleRateChanged( srate_, oldRate );
+  }
+}
+
+void Stk :: sampleRateChanged( StkFloat newRate, StkFloat oldRate )
+{
+  // This function should be reimplemented in classes that need to
+  // make internal variable adjustments in response to a global sample
+  // rate change.
+}
+
+void Stk :: addSampleRateAlert( Stk *ptr )
+{
+  for ( unsigned int i=0; i<alertList_.size(); i++ )
+    if ( alertList_[i] == ptr ) return;
+
+  alertList_.push_back( ptr );
+}
+
+void Stk :: removeSampleRateAlert( Stk *ptr )
+{
+  for ( unsigned int i=0; i<alertList_.size(); i++ ) {
+    if ( alertList_[i] == ptr ) {
+      alertList_.erase( alertList_.begin() + i );
+      return;
+    }
+  }
+}
+
 void Stk :: setRawwavePath( std::string path )
 {
   if ( !path.empty() )
@@ -304,7 +366,7 @@ StkFloat StkFrames :: interpolate( StkFloat frame, unsigned int channel ) const
   else {
     iIndex += channel * nFrames_;
     output = data_[ iIndex ];
-    output += ( alpha * ( data_[ iIndex++ ] - output ) );
+    output += ( alpha * ( data_[ ++iIndex ] - output ) );
   }
 
   return output;

src/SubNoise.cpp

@@ -6,7 +6,7 @@
     using the C rand() function.  The quality of the
     rand() function varies from one OS to another.
 
-    by Perry R. Cook and Gary P. Scavone, 1995 - 2005.
+    by Perry R. Cook and Gary P. Scavone, 1995 - 2007.
 */
 /***************************************************/
 

src/TcpClient.cpp

@@ -19,7 +19,7 @@
     less than or equal to zero indicate a closed
     or lost connection or the occurence of an error.
 
-    by Perry R. Cook and Gary P. Scavone, 1995 - 2005.
+    by Perry R. Cook and Gary P. Scavone, 1995 - 2007.
 */
 /***************************************************/
 

src/TcpServer.cpp

@@ -19,7 +19,7 @@
     less than or equal to zero indicate a closed
     or lost connection or the occurence of an error.
 
-    by Perry R. Cook and Gary P. Scavone, 1995 - 2005.
+    by Perry R. Cook and Gary P. Scavone, 1995 - 2007.
 */
 /***************************************************/
 

src/Thread.cpp

@@ -7,7 +7,7 @@
     systems, the pthread library is used.  Under Windows,
     the Windows thread library is used.
 
-    by Perry R. Cook and Gary P. Scavone, 1995 - 2005.
+    by Perry R. Cook and Gary P. Scavone, 1995 - 2007.
 */
 /***************************************************/
 

src/TubeBell.cpp

@@ -26,7 +26,7 @@
     type who should worry about this (making
     money) worry away.
 
-    by Perry R. Cook and Gary P. Scavone, 1995 - 2005.
+    by Perry R. Cook and Gary P. Scavone, 1995 - 2007.
 */
 /***************************************************/
 

src/TwoPole.cpp

@@ -8,7 +8,7 @@
     frequency response while maintaining a nearly
     constant filter gain.
 
-    by Perry R. Cook and Gary P. Scavone, 1995 - 2005.
+    by Perry R. Cook and Gary P. Scavone, 1995 - 2007.
 */
 /***************************************************/
 
@@ -21,10 +21,20 @@ TwoPole :: TwoPole() : Filter()
   std::vector<StkFloat> a(3, 0.0);
   a[0] = 1.0;
   Filter::setCoefficients( b, a );
+  Stk::addSampleRateAlert( this );
 }
 
 TwoPole :: ~TwoPole()
 {
+  Stk::removeSampleRateAlert( this );
+}
+
+void TwoPole :: sampleRateChanged( StkFloat newRate, StkFloat oldRate )
+{
+  if ( !ignoreSampleRateChange_ ) {
+    errorString_ << "TwoPole::sampleRateChanged: you may need to recompute filter coefficients!";
+    handleError( StkError::WARNING );
+  }
 }
 
 void TwoPole :: clear(void)

src/TwoZero.cpp

@@ -8,7 +8,7 @@
     frequency response while maintaining a
     constant filter gain.
 
-    by Perry R. Cook and Gary P. Scavone, 1995 - 2005.
+    by Perry R. Cook and Gary P. Scavone, 1995 - 2007.
 */
 /***************************************************/
 
@@ -21,10 +21,20 @@ TwoZero :: TwoZero() : Filter()
   b[0] = 1.0;
   std::vector<StkFloat> a(1, 1.0);
   Filter::setCoefficients( b, a );
+  Stk::addSampleRateAlert( this );
 }
 
 TwoZero :: ~TwoZero()
 {
+  Stk::removeSampleRateAlert( this );
+}
+
+void TwoZero :: sampleRateChanged( StkFloat newRate, StkFloat oldRate )
+{
+  if ( !ignoreSampleRateChange_ ) {
+    errorString_ << "TwoZero::sampleRateChanged: you may need to recompute filter coefficients!";
+    handleError( StkError::WARNING );
+  }
 }
 
 void TwoZero :: clear(void)

src/UdpSocket.cpp

@@ -17,7 +17,7 @@
     read/write methods.  Values less than or equal to zero indicate
     the occurence of an error.
 
-    by Perry R. Cook and Gary P. Scavone, 1995 - 2005.
+    by Perry R. Cook and Gary P. Scavone, 1995 - 2007.
 */
 /***************************************************/
 

src/VoicForm.cpp

@@ -21,7 +21,7 @@
        - Vibrato Gain = 1
        - Loudness (Spectral Tilt) = 128
 
-    by Perry R. Cook and Gary P. Scavone, 1995 - 2005.
+    by Perry R. Cook and Gary P. Scavone, 1995 - 2007.
 */
 /***************************************************/
 

src/Voicer.cpp

@@ -25,7 +25,7 @@
     an ensemble.  Alternately, control changes can
     be sent to all voices on a given channel.
 
-    by Perry R. Cook and Gary P. Scavone, 1995 - 2005.
+    by Perry R. Cook and Gary P. Scavone, 1995 - 2007.
 */
 /***************************************************/
 

src/WaveLoop.cpp

@@ -12,7 +12,7 @@
     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.
 */
 /***************************************************/
 
@@ -22,6 +22,7 @@
 WaveLoop :: WaveLoop( unsigned long chunkThreshold, unsigned long chunkSize )
   : FileWvIn( chunkThreshold, chunkSize ), phaseOffset_(0.0)
 {
+  Stk::addSampleRateAlert( this );
 }
 
 WaveLoop :: WaveLoop( std::string fileName, bool raw, bool doNormalize,
@@ -29,10 +30,18 @@ WaveLoop :: WaveLoop( std::string fileName, bool raw, bool doNormalize,
   : FileWvIn( chunkThreshold, chunkSize ), phaseOffset_(0.0)
 {
   this->openFile( fileName, raw, doNormalize );
+  Stk::addSampleRateAlert( this );
 }
 
 WaveLoop :: ~WaveLoop()
 {
+  Stk::removeSampleRateAlert( this );
+}
+
+void WaveLoop :: sampleRateChanged( StkFloat newRate, StkFloat oldRate )
+{
+  if ( !ignoreSampleRateChange_ )
+    this->setRate( oldRate * rate_ / newRate );
 }
 
 void WaveLoop :: openFile( std::string fileName, bool raw, bool doNormalize )

src/Whistle.cpp

@@ -139,7 +139,7 @@ StkFloat Whistle :: computeSample()
     frameCount += 1;
     if ( frameCount >= (1470 / subSample_) ) {
       frameCount = 0;
-      printf("%f %f %f\n",tempVectorP->getX(),tempVectorP->getY(),envOut);
+      printf("%f %f %f\n",tempVectorP_->getX(),tempVectorP_->getY(),envOut);
       fflush(stdout);
     }
 #endif

src/Wurley.cpp

@@ -26,7 +26,7 @@
     type who should worry about this (making
     money) worry away.
 
-    by Perry R. Cook and Gary P. Scavone, 1995 - 2005.
+    by Perry R. Cook and Gary P. Scavone, 1995 - 2007.
 */
 /***************************************************/
 

src/WvIn.cpp

@@ -13,7 +13,7 @@
     Both interleaved and non-interleaved data is supported via the use
     of StkFrames objects.
 
-    by Perry R. Cook and Gary P. Scavone, 1995 - 2005.
+    by Perry R. Cook and Gary P. Scavone, 1995 - 2007.
 */
 /***************************************************/
 

src/WvOut.cpp

@@ -16,7 +16,7 @@
     Currently, WvOut is non-interpolating and the output rate is
     always Stk::sampleRate().
 
-    by Perry R. Cook and Gary P. Scavone, 1995 - 2005.
+    by Perry R. Cook and Gary P. Scavone, 1995 - 2007.
 */
 /***************************************************/
 

src/asio/asio.h

@@ -3,9 +3,12 @@
 
 /*
 	Steinberg Audio Stream I/O API
-	(c) 1997 - 1999, Steinberg Soft- und Hardware GmbH
+	(c) 1997 - 2005, Steinberg Media Technologies GmbH
+
+	ASIO Interface Specification v 2.1
+
+	2005 - Added support for DSD sample data (in cooperation with Sony)
 
-	ASIO Interface Specification v 2.0
 
 	basic concept is an i/o synchronous double-buffer scheme:
 	
@@ -131,7 +134,7 @@ enum {
 
 	// these are used for 32 bit data buffer, with different alignment of the data inside
 	// 32 bit PCI bus systems can be more easily used with these
-	ASIOSTInt32MSB16 = 8,		// 32 bit data with 18 bit alignment
+	ASIOSTInt32MSB16 = 8,		// 32 bit data with 16 bit alignment
 	ASIOSTInt32MSB18 = 9,		// 32 bit data with 18 bit alignment
 	ASIOSTInt32MSB20 = 10,		// 32 bit data with 20 bit alignment
 	ASIOSTInt32MSB24 = 11,		// 32 bit data with 24 bit alignment
@@ -147,9 +150,56 @@ enum {
 	ASIOSTInt32LSB16 = 24,		// 32 bit data with 18 bit alignment
 	ASIOSTInt32LSB18 = 25,		// 32 bit data with 18 bit alignment
 	ASIOSTInt32LSB20 = 26,		// 32 bit data with 20 bit alignment
-	ASIOSTInt32LSB24 = 27		// 32 bit data with 24 bit alignment
+	ASIOSTInt32LSB24 = 27,		// 32 bit data with 24 bit alignment
+
+	//	ASIO DSD format.
+	ASIOSTDSDInt8LSB1   = 32,		// DSD 1 bit data, 8 samples per byte. First sample in Least significant bit.
+	ASIOSTDSDInt8MSB1   = 33,		// DSD 1 bit data, 8 samples per byte. First sample in Most significant bit.
+	ASIOSTDSDInt8NER8	= 40,		// DSD 8 bit data, 1 sample per byte. No Endianness required.
+
+	ASIOSTLastEntry
 };
 
+/*-----------------------------------------------------------------------------
+// DSD operation and buffer layout
+// Definition by Steinberg/Sony Oxford.
+//
+// We have tried to treat DSD as PCM and so keep a consistant structure across
+// the ASIO interface.
+//
+// DSD's sample rate is normally referenced as a multiple of 44.1Khz, so
+// the standard sample rate is refered to as 64Fs (or 2.8224Mhz). We looked
+// at making a special case for DSD and adding a field to the ASIOFuture that
+// would allow the user to select the Over Sampleing Rate (OSR) as a seperate
+// entity but decided in the end just to treat it as a simple value of
+// 2.8224Mhz and use the standard interface to set it.
+//
+// The second problem was the "word" size, in PCM the word size is always a
+// greater than or equal to 8 bits (a byte). This makes life easy as we can
+// then pack the samples into the "natural" size for the machine.
+// In DSD the "word" size is 1 bit. This is not a major problem and can easily
+// be dealt with if we ensure that we always deal with a multiple of 8 samples.
+//
+// DSD brings with it another twist to the Endianness religion. How are the
+// samples packed into the byte. It would be nice to just say the most significant
+// bit is always the first sample, however there would then be a performance hit
+// on little endian machines. Looking at how some of the processing goes...
+// Little endian machines like the first sample to be in the Least Significant Bit,
+//   this is because when you write it to memory the data is in the correct format
+//   to be shifted in and out of the words.
+// Big endian machine prefer the first sample to be in the Most Significant Bit,
+//   again for the same reasion.
+//
+// And just when things were looking really muddy there is a proposed extension to
+// DSD that uses 8 bit word sizes. It does not care what endianness you use.
+//
+// Switching the driver between DSD and PCM mode
+// ASIOFuture allows for extending the ASIO API quite transparently.
+// See kAsioSetIoFormat, kAsioGetIoFormat, kAsioCanDoIoFormat
+//
+//-----------------------------------------------------------------------------*/
+
+
 //- - - - - - - - - - - - - - - - - - - - - - - - -
 // Error codes
 //- - - - - - - - - - - - - - - - - - - - - - - - -
@@ -403,9 +453,14 @@ enum
 	kAsioSupportsTimeInfo,		// if host returns true here, it will expect the
 								// callback bufferSwitchTimeInfo to be called instead
 								// of bufferSwitch
-	kAsioSupportsTimeCode,		// supports time code reading/writing
-
-	kAsioSupportsInputMonitor,	// supports input monitoring
+	kAsioSupportsTimeCode,		// 
+	kAsioMMCCommand,			// unused - value: number of commands, message points to mmc commands
+	kAsioSupportsInputMonitor,	// kAsioSupportsXXX return 1 if host supports this
+	kAsioSupportsInputGain,     // unused and undefined
+	kAsioSupportsInputMeter,    // unused and undefined
+	kAsioSupportsOutputGain,    // unused and undefined
+	kAsioSupportsOutputMeter,   // unused and undefined
+	kAsioOverload,              // driver detected an overload
 
 	kAsioNumMessageSelectors
 };
@@ -860,7 +915,14 @@ enum
 	kAsioCanInputGain,
 	kAsioCanInputMeter,
 	kAsioCanOutputGain,
-	kAsioCanOutputMeter
+	kAsioCanOutputMeter,
+
+	//	DSD support
+	//	The following extensions are required to allow switching
+	//	and control of the DSD subsystem.
+	kAsioSetIoFormat			= 0x23111961,		/* ASIOIoFormat * in params.			*/
+	kAsioGetIoFormat			= 0x23111983,		/* ASIOIoFormat * in params.			*/
+	kAsioCanDoIoFormat			= 0x23112004,		/* ASIOIoFormat * in params.			*/
 };
 
 typedef struct ASIOInputMonitor
@@ -905,6 +967,43 @@ enum
 	kTransMonitor		// trackSwitches
 };
 
+/*
+// DSD support
+//	Some notes on how to use ASIOIoFormatType.
+//
+//	The caller will fill the format with the request types.
+//	If the board can do the request then it will leave the
+//	values unchanged. If the board does not support the
+//	request then it will change that entry to Invalid (-1)
+//
+//	So to request DSD then
+//
+//	ASIOIoFormat NeedThis={kASIODSDFormat};
+//
+//	if(ASE_SUCCESS != ASIOFuture(kAsioSetIoFormat,&NeedThis) ){
+//		// If the board did not accept one of the parameters then the
+//		// whole call will fail and the failing parameter will
+//		// have had its value changes to -1.
+//	}
+//
+// Note: Switching between the formats need to be done before the "prepared"
+// state (see ASIO 2 documentation) is entered.
+*/
+typedef long int ASIOIoFormatType;
+enum ASIOIoFormatType_e
+{
+	kASIOFormatInvalid = -1,
+	kASIOPCMFormat = 0,
+	kASIODSDFormat = 1,
+};
+
+typedef struct ASIOIoFormat_s
+{
+	ASIOIoFormatType	FormatType;
+	char				future[512-sizeof(ASIOIoFormatType)];
+} ASIOIoFormat;
+
+
 ASIOError ASIOOutputReady(void);
 /* Purpose:
 	  this tells the driver that the host has completed processing

src/asio/asiolist.cpp

@@ -19,7 +19,7 @@ static LONG findDrvPath (char *clsidstr,char *dllpath,int dllpathsize)
 	DWORD			index;
 	OFSTRUCT		ofs;
 	HFILE			hfile;
-	BOOL			found = false;
+	BOOL			found = FALSE;
 
 	CharLowerBuff(clsidstr,strlen(clsidstr));
 	if ((cr = RegOpenKey(HKEY_CLASSES_ROOT,COM_CLSID,&hkEnum)) == ERROR_SUCCESS) {
@@ -44,7 +44,7 @@ static LONG findDrvPath (char *clsidstr,char *dllpath,int dllpathsize)
 						}
 						RegCloseKey(hksub);
 					}
-					found = true;	// break out 
+					found = TRUE;	// break out 
 				}
 			}
 		}				
@@ -134,7 +134,7 @@ AsioDriverList::AsioDriverList ()
 	LPASIODRVSTRUCT	pdl;
 	LONG 			cr;
 	DWORD			index = 0;
-	BOOL			fin = false;
+	BOOL			fin = FALSE;
 
 	numdrv		= 0;
 	lpdrvlist	= 0;
@@ -144,7 +144,7 @@ AsioDriverList::AsioDriverList ()
 		if ((cr = RegEnumKey(hkEnum,index++,(LPTSTR)keyname,MAXDRVNAMELEN))== ERROR_SUCCESS) {
 			lpdrvlist = newDrvStruct (hkEnum,keyname,0,lpdrvlist);
 		}
-		else fin = true;
+		else fin = TRUE;
 	}
 	if (hkEnum) RegCloseKey(hkEnum);
 

src/asio/iasiothiscallresolver.cpp

@@ -0,0 +1,563 @@
+/*
+	IASIOThiscallResolver.cpp see the comments in iasiothiscallresolver.h for
+    the top level description - this comment describes the technical details of
+    the implementation.
+
+    The latest version of this file is available from:
+    http://www.audiomulch.com/~rossb/code/calliasio
+
+    please email comments to Ross Bencina <rossb@audiomulch.com>
+
+    BACKGROUND
+
+    The IASIO interface declared in the Steinberg ASIO 2 SDK declares
+    functions with no explicit calling convention. This causes MSVC++ to default
+    to using the thiscall convention, which is a proprietary convention not
+    implemented by some non-microsoft compilers - notably borland BCC,
+    C++Builder, and gcc. MSVC++ is the defacto standard compiler used by
+    Steinberg. As a result of this situation, the ASIO sdk will compile with
+    any compiler, however attempting to execute the compiled code will cause a
+    crash due to different default calling conventions on non-Microsoft
+    compilers.
+
+    IASIOThiscallResolver solves the problem by providing an adapter class that
+    delegates to the IASIO interface using the correct calling convention
+    (thiscall). Due to the lack of support for thiscall in the Borland and GCC
+    compilers, the calls have been implemented in assembly language.
+
+    A number of macros are defined for thiscall function calls with different
+    numbers of parameters, with and without return values - it may be possible
+    to modify the format of these macros to make them work with other inline
+    assemblers.
+
+
+    THISCALL DEFINITION
+
+    A number of definitions of the thiscall calling convention are floating
+    around the internet. The following definition has been validated against
+    output from the MSVC++ compiler:
+
+    For non-vararg functions, thiscall works as follows: the object (this)
+    pointer is passed in ECX. All arguments are passed on the stack in
+    right to left order. The return value is placed in EAX. The callee
+    clears the passed arguments from the stack.
+
+
+    FINDING FUNCTION POINTERS FROM AN IASIO POINTER
+
+    The first field of a COM object is a pointer to its vtble. Thus a pointer
+    to an object implementing the IASIO interface also points to a pointer to
+    that object's vtbl. The vtble is a table of function pointers for all of
+    the virtual functions exposed by the implemented interfaces.
+
+    If we consider a variable declared as a pointer to IASO:
+
+    IASIO *theAsioDriver
+
+    theAsioDriver points to:
+
+    object implementing IASIO
+    {
+        IASIOvtbl *vtbl
+        other data
+    }
+
+    in other words, theAsioDriver points to a pointer to an IASIOvtbl
+
+    vtbl points to a table of function pointers:
+
+    IASIOvtbl ( interface IASIO : public IUnknown )
+    {
+    (IUnknown functions)
+    0   virtual HRESULT STDMETHODCALLTYPE (*QueryInterface)(REFIID riid, void **ppv) = 0;
+    4   virtual ULONG STDMETHODCALLTYPE (*AddRef)() = 0;
+    8   virtual ULONG STDMETHODCALLTYPE (*Release)() = 0;      
+
+    (IASIO functions)
+    12	virtual ASIOBool (*init)(void *sysHandle) = 0;
+    16	virtual void (*getDriverName)(char *name) = 0;
+    20	virtual long (*getDriverVersion)() = 0;
+    24	virtual void (*getErrorMessage)(char *string) = 0;
+    28	virtual ASIOError (*start)() = 0;
+    32	virtual ASIOError (*stop)() = 0;
+    36	virtual ASIOError (*getChannels)(long *numInputChannels, long *numOutputChannels) = 0;
+    40	virtual ASIOError (*getLatencies)(long *inputLatency, long *outputLatency) = 0;
+    44	virtual ASIOError (*getBufferSize)(long *minSize, long *maxSize,
+            long *preferredSize, long *granularity) = 0;
+    48	virtual ASIOError (*canSampleRate)(ASIOSampleRate sampleRate) = 0;
+    52	virtual ASIOError (*getSampleRate)(ASIOSampleRate *sampleRate) = 0;
+    56	virtual ASIOError (*setSampleRate)(ASIOSampleRate sampleRate) = 0;
+    60	virtual ASIOError (*getClockSources)(ASIOClockSource *clocks, long *numSources) = 0;
+    64	virtual ASIOError (*setClockSource)(long reference) = 0;
+    68	virtual ASIOError (*getSamplePosition)(ASIOSamples *sPos, ASIOTimeStamp *tStamp) = 0;
+    72	virtual ASIOError (*getChannelInfo)(ASIOChannelInfo *info) = 0;
+    76	virtual ASIOError (*createBuffers)(ASIOBufferInfo *bufferInfos, long numChannels,
+            long bufferSize, ASIOCallbacks *callbacks) = 0;
+    80	virtual ASIOError (*disposeBuffers)() = 0;
+    84	virtual ASIOError (*controlPanel)() = 0;
+    88	virtual ASIOError (*future)(long selector,void *opt) = 0;
+    92	virtual ASIOError (*outputReady)() = 0;
+    };
+
+    The numbers in the left column show the byte offset of each function ptr
+    from the beginning of the vtbl. These numbers are used in the code below
+    to select different functions.
+
+    In order to find the address of a particular function, theAsioDriver
+    must first be dereferenced to find the value of the vtbl pointer:
+
+    mov     eax, theAsioDriver
+    mov     edx, [theAsioDriver]  // edx now points to vtbl[0]
+
+    Then an offset must be added to the vtbl pointer to select a
+    particular function, for example vtbl+44 points to the slot containing
+    a pointer to the getBufferSize function.
+
+    Finally vtbl+x must be dereferenced to obtain the value of the function
+    pointer stored in that address:
+
+    call    [edx+44]    // call the function pointed to by
+                        // the value in the getBufferSize field of the vtbl
+
+
+    SEE ALSO
+
+    Martin Fay's OpenASIO DLL at http://www.martinfay.com solves the same
+    problem by providing a new COM interface which wraps IASIO with an
+    interface that uses portable calling conventions. OpenASIO must be compiled
+    with MSVC, and requires that you ship the OpenASIO DLL with your
+    application.
+
+    
+    ACKNOWLEDGEMENTS
+
+    Ross Bencina: worked out the thiscall details above, wrote the original
+    Borland asm macros, and a patch for asio.cpp (which is no longer needed).
+    Thanks to Martin Fay for introducing me to the issues discussed here,
+    and to Rene G. Ceballos for assisting with asm dumps from MSVC++.
+
+    Antti Silvast: converted the original calliasio to work with gcc and NASM
+    by implementing the asm code in a separate file.
+
+	Fraser Adams: modified the original calliasio containing the Borland inline
+    asm to add inline asm for gcc i.e. Intel syntax for Borland and AT&T syntax
+    for gcc. This seems a neater approach for gcc than to have a separate .asm
+    file and it means that we only need one version of the thiscall patch.
+
+    Fraser Adams: rewrote the original calliasio patch in the form of the
+    IASIOThiscallResolver class in order to avoid modifications to files from
+    the Steinberg SDK, which may have had potential licence issues.
+
+    Andrew Baldwin: contributed fixes for compatibility problems with more
+    recent versions of the gcc assembler.
+*/
+
+
+// We only need IASIOThiscallResolver at all if we are on Win32. For other
+// platforms we simply bypass the IASIOThiscallResolver definition to allow us
+// to be safely #include'd whatever the platform to keep client code portable
+#if defined(WIN32) || defined(_WIN32) || defined(__WIN32__)
+
+
+// If microsoft compiler we can call IASIO directly so IASIOThiscallResolver
+// is not used.
+#if !defined(_MSC_VER)
+
+
+#include <new>
+#include <assert.h>
+
+// We have a mechanism in iasiothiscallresolver.h to ensure that asio.h is
+// #include'd before it in client code, we do NOT want to do this test here.
+#define iasiothiscallresolver_sourcefile 1
+#include "iasiothiscallresolver.h"
+#undef iasiothiscallresolver_sourcefile
+
+// iasiothiscallresolver.h redefines ASIOInit for clients, but we don't want
+// this macro defined in this translation unit.
+#undef ASIOInit
+
+
+// theAsioDriver is a global pointer to the current IASIO instance which the
+// ASIO SDK uses to perform all actions on the IASIO interface. We substitute
+// our own forwarding interface into this pointer.
+extern IASIO* theAsioDriver;
+
+
+// The following macros define the inline assembler for BORLAND first then gcc
+
+#if defined(__BCPLUSPLUS__) || defined(__BORLANDC__)          
+
+
+#define CALL_THISCALL_0( resultName, thisPtr, funcOffset )\
+    void *this_ = (thisPtr);                                                \
+    __asm {                                                                 \
+        mov     ecx, this_            ;                                     \
+        mov     eax, [ecx]            ;                                     \
+        call    [eax+funcOffset]      ;                                     \
+        mov     resultName, eax       ;                                     \
+    }
+
+
+#define CALL_VOID_THISCALL_1( thisPtr, funcOffset, param1 )\
+    void *this_ = (thisPtr);                                                \
+    __asm {                                                                 \
+        mov     eax, param1           ;                                     \
+        push    eax                   ;                                     \
+        mov     ecx, this_            ;                                     \
+        mov     eax, [ecx]            ;                                     \
+        call    [eax+funcOffset]      ;                                     \
+    }
+
+
+#define CALL_THISCALL_1( resultName, thisPtr, funcOffset, param1 )\
+    void *this_ = (thisPtr);                                                \
+    __asm {                                                                 \
+        mov     eax, param1           ;                                     \
+        push    eax                   ;                                     \
+        mov     ecx, this_            ;                                     \
+        mov     eax, [ecx]            ;                                     \
+        call    [eax+funcOffset]      ;                                     \
+        mov     resultName, eax       ;                                     \
+    }
+
+
+#define CALL_THISCALL_1_DOUBLE( resultName, thisPtr, funcOffset, param1 )\
+    void *this_ = (thisPtr);                                                \
+    void *doubleParamPtr_ (&param1);                                        \
+    __asm {                                                                 \
+        mov     eax, doubleParamPtr_  ;                                     \
+        push    [eax+4]               ;                                     \
+        push    [eax]                 ;                                     \
+        mov     ecx, this_            ;                                     \
+        mov     eax, [ecx]            ;                                     \
+        call    [eax+funcOffset]      ;                                     \
+        mov     resultName, eax       ;                                     \
+    }
+
+
+#define CALL_THISCALL_2( resultName, thisPtr, funcOffset, param1, param2 )\
+    void *this_ = (thisPtr);                                                \
+    __asm {                                                                 \
+        mov     eax, param2           ;                                     \
+        push    eax                   ;                                     \
+        mov     eax, param1           ;                                     \
+        push    eax                   ;                                     \
+        mov     ecx, this_            ;                                     \
+        mov     eax, [ecx]            ;                                     \
+        call    [eax+funcOffset]      ;                                     \
+        mov     resultName, eax       ;                                     \
+    }
+
+
+#define CALL_THISCALL_4( resultName, thisPtr, funcOffset, param1, param2, param3, param4 )\
+    void *this_ = (thisPtr);                                                \
+    __asm {                                                                 \
+        mov     eax, param4           ;                                     \
+        push    eax                   ;                                     \
+        mov     eax, param3           ;                                     \
+        push    eax                   ;                                     \
+        mov     eax, param2           ;                                     \
+        push    eax                   ;                                     \
+        mov     eax, param1           ;                                     \
+        push    eax                   ;                                     \
+        mov     ecx, this_            ;                                     \
+        mov     eax, [ecx]            ;                                     \
+        call    [eax+funcOffset]      ;                                     \
+        mov     resultName, eax       ;                                     \
+    }
+
+
+#elif defined(__GNUC__)
+
+
+#define CALL_THISCALL_0( resultName, thisPtr, funcOffset )                  \
+    __asm__ __volatile__ ("movl (%1), %%edx\n\t"                            \
+                          "call *"#funcOffset"(%%edx)\n\t"                  \
+                          :"=a"(resultName) /* Output Operands */           \
+                          :"c"(thisPtr)     /* Input Operands */            \
+                         );                                                 \
+
+
+#define CALL_VOID_THISCALL_1( thisPtr, funcOffset, param1 )                 \
+    __asm__ __volatile__ ("pushl %0\n\t"                                    \
+                          "movl (%1), %%edx\n\t"                            \
+                          "call *"#funcOffset"(%%edx)\n\t"                  \
+                          :                 /* Output Operands */           \
+                          :"r"(param1),     /* Input Operands */            \
+                           "c"(thisPtr)                                     \
+                         );                                                 \
+
+
+#define CALL_THISCALL_1( resultName, thisPtr, funcOffset, param1 )          \
+    __asm__ __volatile__ ("pushl %1\n\t"                                    \
+                          "movl (%2), %%edx\n\t"                            \
+                          "call *"#funcOffset"(%%edx)\n\t"                  \
+                          :"=a"(resultName) /* Output Operands */           \
+                          :"r"(param1),     /* Input Operands */            \
+                           "c"(thisPtr)                                     \
+                          );                                                \
+
+
+#define CALL_THISCALL_1_DOUBLE( resultName, thisPtr, funcOffset, param1 )   \
+    __asm__ __volatile__ ("pushl 4(%1)\n\t"                                 \
+                          "pushl (%1)\n\t"                                  \
+                          "movl (%2), %%edx\n\t"                            \
+                          "call *"#funcOffset"(%%edx);\n\t"                 \
+                          :"=a"(resultName) /* Output Operands */           \
+                          :"a"(&param1),    /* Input Operands */            \
+                           /* Note: Using "r" above instead of "a" fails */ \
+                           /* when using GCC 3.3.3, and maybe later versions*/\
+                           "c"(thisPtr)                                     \
+                          );                                                \
+
+
+#define CALL_THISCALL_2( resultName, thisPtr, funcOffset, param1, param2 )  \
+    __asm__ __volatile__ ("pushl %1\n\t"                                    \
+                          "pushl %2\n\t"                                    \
+                          "movl (%3), %%edx\n\t"                            \
+                          "call *"#funcOffset"(%%edx)\n\t"                  \
+                          :"=a"(resultName) /* Output Operands */           \
+                          :"r"(param2),     /* Input Operands */            \
+                           "r"(param1),                                     \
+                           "c"(thisPtr)                                     \
+                          );                                                \
+
+
+#define CALL_THISCALL_4( resultName, thisPtr, funcOffset, param1, param2, param3, param4 )\
+    __asm__ __volatile__ ("pushl %1\n\t"                                    \
+                          "pushl %2\n\t"                                    \
+                          "pushl %3\n\t"                                    \
+                          "pushl %4\n\t"                                    \
+                          "movl (%5), %%edx\n\t"                            \
+                          "call *"#funcOffset"(%%edx)\n\t"                  \
+                          :"=a"(resultName) /* Output Operands */           \
+                          :"r"(param4),     /* Input Operands  */           \
+                           "r"(param3),                                     \
+                           "r"(param2),                                     \
+                           "r"(param1),                                     \
+                           "c"(thisPtr)                                     \
+                          );                                                \
+
+#endif
+
+
+
+// Our static singleton instance.
+IASIOThiscallResolver IASIOThiscallResolver::instance;
+
+// Constructor called to initialize static Singleton instance above. Note that
+// it is important not to clear that_ incase it has already been set by the call
+// to placement new in ASIOInit().
+IASIOThiscallResolver::IASIOThiscallResolver()
+{
+}
+
+// Constructor called from ASIOInit() below
+IASIOThiscallResolver::IASIOThiscallResolver(IASIO* that)
+: that_( that )
+{
+}
+
+// Implement IUnknown methods as assert(false). IASIOThiscallResolver is not
+// really a COM object, just a wrapper which will work with the ASIO SDK.
+// If you wanted to use ASIO without the SDK you might want to implement COM
+// aggregation in these methods.
+HRESULT STDMETHODCALLTYPE IASIOThiscallResolver::QueryInterface(REFIID riid, void **ppv)
+{
+    (void)riid;     // suppress unused variable warning
+
+    assert( false ); // this function should never be called by the ASIO SDK.
+
+    *ppv = NULL;
+    return E_NOINTERFACE;
+}
+
+ULONG STDMETHODCALLTYPE IASIOThiscallResolver::AddRef()
+{
+    assert( false ); // this function should never be called by the ASIO SDK.
+
+    return 1;
+}
+
+ULONG STDMETHODCALLTYPE IASIOThiscallResolver::Release()
+{
+    assert( false ); // this function should never be called by the ASIO SDK.
+    
+    return 1;
+}
+
+
+// Implement the IASIO interface methods by performing the vptr manipulation
+// described above then delegating to the real implementation.
+ASIOBool IASIOThiscallResolver::init(void *sysHandle)
+{
+    ASIOBool result;
+    CALL_THISCALL_1( result, that_, 12, sysHandle );
+    return result;
+}
+
+void IASIOThiscallResolver::getDriverName(char *name)
+{
+    CALL_VOID_THISCALL_1( that_, 16, name );
+}
+
+long IASIOThiscallResolver::getDriverVersion()
+{
+    ASIOBool result;
+    CALL_THISCALL_0( result, that_, 20 );
+    return result;
+}
+
+void IASIOThiscallResolver::getErrorMessage(char *string)
+{
+     CALL_VOID_THISCALL_1( that_, 24, string );
+}
+
+ASIOError IASIOThiscallResolver::start()
+{
+    ASIOBool result;
+    CALL_THISCALL_0( result, that_, 28 );
+    return result;
+}
+
+ASIOError IASIOThiscallResolver::stop()
+{
+    ASIOBool result;
+    CALL_THISCALL_0( result, that_, 32 );
+    return result;
+}
+
+ASIOError IASIOThiscallResolver::getChannels(long *numInputChannels, long *numOutputChannels)
+{
+    ASIOBool result;
+    CALL_THISCALL_2( result, that_, 36, numInputChannels, numOutputChannels );
+    return result;
+}
+
+ASIOError IASIOThiscallResolver::getLatencies(long *inputLatency, long *outputLatency)
+{
+    ASIOBool result;
+    CALL_THISCALL_2( result, that_, 40, inputLatency, outputLatency );
+    return result;
+}
+
+ASIOError IASIOThiscallResolver::getBufferSize(long *minSize, long *maxSize,
+        long *preferredSize, long *granularity)
+{
+    ASIOBool result;
+    CALL_THISCALL_4( result, that_, 44, minSize, maxSize, preferredSize, granularity );
+    return result;
+}
+
+ASIOError IASIOThiscallResolver::canSampleRate(ASIOSampleRate sampleRate)
+{
+    ASIOBool result;
+    CALL_THISCALL_1_DOUBLE( result, that_, 48, sampleRate );
+    return result;
+}
+
+ASIOError IASIOThiscallResolver::getSampleRate(ASIOSampleRate *sampleRate)
+{
+    ASIOBool result;
+    CALL_THISCALL_1( result, that_, 52, sampleRate );
+    return result;
+}
+
+ASIOError IASIOThiscallResolver::setSampleRate(ASIOSampleRate sampleRate)
+{    
+    ASIOBool result;
+    CALL_THISCALL_1_DOUBLE( result, that_, 56, sampleRate );
+    return result;
+}
+
+ASIOError IASIOThiscallResolver::getClockSources(ASIOClockSource *clocks, long *numSources)
+{
+    ASIOBool result;
+    CALL_THISCALL_2( result, that_, 60, clocks, numSources );
+    return result;
+}
+
+ASIOError IASIOThiscallResolver::setClockSource(long reference)
+{
+    ASIOBool result;
+    CALL_THISCALL_1( result, that_, 64, reference );
+    return result;
+}
+
+ASIOError IASIOThiscallResolver::getSamplePosition(ASIOSamples *sPos, ASIOTimeStamp *tStamp)
+{
+    ASIOBool result;
+    CALL_THISCALL_2( result, that_, 68, sPos, tStamp );
+    return result;
+}
+
+ASIOError IASIOThiscallResolver::getChannelInfo(ASIOChannelInfo *info)
+{
+    ASIOBool result;
+    CALL_THISCALL_1( result, that_, 72, info );
+    return result;
+}
+
+ASIOError IASIOThiscallResolver::createBuffers(ASIOBufferInfo *bufferInfos,
+        long numChannels, long bufferSize, ASIOCallbacks *callbacks)
+{
+    ASIOBool result;
+    CALL_THISCALL_4( result, that_, 76, bufferInfos, numChannels, bufferSize, callbacks );
+    return result;
+}
+
+ASIOError IASIOThiscallResolver::disposeBuffers()
+{
+    ASIOBool result;
+    CALL_THISCALL_0( result, that_, 80 );
+    return result;
+}
+
+ASIOError IASIOThiscallResolver::controlPanel()
+{
+    ASIOBool result;
+    CALL_THISCALL_0( result, that_, 84 );
+    return result;
+}
+
+ASIOError IASIOThiscallResolver::future(long selector,void *opt)
+{
+    ASIOBool result;
+    CALL_THISCALL_2( result, that_, 88, selector, opt );
+    return result;
+}
+
+ASIOError IASIOThiscallResolver::outputReady()
+{
+    ASIOBool result;
+    CALL_THISCALL_0( result, that_, 92 );
+    return result;
+}
+
+
+// Implement our substitute ASIOInit() method
+ASIOError IASIOThiscallResolver::ASIOInit(ASIODriverInfo *info)
+{
+    // To ensure that our instance's vptr is correctly constructed, even if
+    // ASIOInit is called prior to main(), we explicitly call its constructor
+    // (potentially over the top of an existing instance). Note that this is
+    // pretty ugly, and is only safe because IASIOThiscallResolver has no
+    // destructor and contains no objects with destructors.
+    new((void*)&instance) IASIOThiscallResolver( theAsioDriver );
+
+    // Interpose between ASIO client code and the real driver.
+    theAsioDriver = &instance;
+
+    // Note that we never need to switch theAsioDriver back to point to the
+    // real driver because theAsioDriver is reset to zero in ASIOExit().
+
+    // Delegate to the real ASIOInit
+	return ::ASIOInit(info);
+}
+
+
+#endif /* !defined(_MSC_VER) */
+
+#endif /* Win32 */
+

src/asio/iasiothiscallresolver.h

@@ -0,0 +1,201 @@
+// ****************************************************************************
+//
+// Changed:         I have modified this file slightly (includes) to work  with
+//                  RtAudio. RtAudio.cpp must include this file after asio.h.                                                    
+//
+// File:			IASIOThiscallResolver.h
+// Description:     The IASIOThiscallResolver class implements the IASIO
+//					interface and acts as a proxy to the real IASIO interface by
+//                  calling through its vptr table using the thiscall calling
+//                  convention. To put it another way, we interpose
+//                  IASIOThiscallResolver between ASIO SDK code and the driver.
+//                  This is necessary because most non-Microsoft compilers don't
+//                  implement the thiscall calling convention used by IASIO.
+//
+//					iasiothiscallresolver.cpp contains the background of this
+//					problem plus a technical description of the vptr
+//                  manipulations.
+//
+//					In order to use this mechanism one simply has to add
+//					iasiothiscallresolver.cpp to the list of files to compile
+//                  and #include <iasiothiscallresolver.h>
+//
+//					Note that this #include must come after the other ASIO SDK
+//                  #includes, for example:
+//
+//					#include <windows.h>
+//					#include <asiosys.h>
+//					#include <asio.h>
+//					#include <asiodrivers.h>
+//					#include <iasiothiscallresolver.h>
+//
+//					Actually the important thing is to #include
+//                  <iasiothiscallresolver.h> after <asio.h>. We have
+//                  incorporated a test to enforce this ordering.
+//
+//					The code transparently takes care of the interposition by
+//                  using macro substitution to intercept calls to ASIOInit()
+//                  and ASIOExit(). We save the original ASIO global
+//                  "theAsioDriver" in our "that" variable, and then set
+//                  "theAsioDriver" to equal our IASIOThiscallResolver instance.
+//
+// 					Whilst this method of resolving the thiscall problem requires
+//					the addition of #include <iasiothiscallresolver.h> to client
+//                  code it has the advantage that it does not break the terms
+//                  of the ASIO licence by publishing it. We are NOT modifying
+//                  any Steinberg code here, we are merely implementing the IASIO
+//					interface in the same way that we would need to do if we
+//					wished to provide an open source ASIO driver.
+//
+//					For compilation with MinGW -lole32 needs to be added to the
+//                  linker options. For BORLAND, linking with Import32.lib is
+//                  sufficient.
+//
+//					The dependencies are with: CoInitialize, CoUninitialize,
+//					CoCreateInstance, CLSIDFromString - used by asiolist.cpp
+//					and are required on Windows whether ThiscallResolver is used
+//					or not.
+//
+//					Searching for the above strings in the root library path
+//					of your compiler should enable the correct libraries to be
+//					identified if they aren't immediately obvious.
+//
+//                  Note that the current implementation of IASIOThiscallResolver
+//                  is not COM compliant - it does not correctly implement the
+//                  IUnknown interface. Implementing it is not necessary because
+//                  it is not called by parts of the ASIO SDK which call through
+//                  theAsioDriver ptr. The IUnknown methods are implemented as
+//                  assert(false) to ensure that the code fails if they are
+//                  ever called.
+// Restrictions:	None. Public Domain & Open Source distribute freely
+//					You may use IASIOThiscallResolver commercially as well as
+//                  privately.
+//					You the user assume the responsibility for the use of the
+//					files, binary or text, and there is no guarantee or warranty,
+//					expressed or implied, including but not limited to the
+//					implied warranties of merchantability and fitness for a
+//					particular purpose. You assume all responsibility and agree
+//					to hold no entity, copyright holder or distributors liable
+//					for any loss of data or inaccurate representations of data
+//					as a result of using IASIOThiscallResolver.
+// Version:         1.4 Added separate macro CALL_THISCALL_1_DOUBLE from
+//                  Andrew Baldwin, and volatile for whole gcc asm blocks,
+//                  both for compatibility with newer gcc versions. Cleaned up
+//                  Borland asm to use one less register.
+//                  1.3 Switched to including assert.h for better compatibility.
+//                  Wrapped entire .h and .cpp contents with a check for
+//                  _MSC_VER to provide better compatibility with MS compilers.
+//                  Changed Singleton implementation to use static instance
+//                  instead of freestore allocated instance. Removed ASIOExit
+//                  macro as it is no longer needed.
+//                  1.2 Removed semicolons from ASIOInit and ASIOExit macros to
+//                  allow them to be embedded in expressions (if statements).
+//                  Cleaned up some comments. Removed combase.c dependency (it
+//                  doesn't compile with BCB anyway) by stubbing IUnknown.
+//                  1.1 Incorporated comments from Ross Bencina including things
+//					such as changing name from ThiscallResolver to
+//					IASIOThiscallResolver, tidying up the constructor, fixing
+//					a bug in IASIOThiscallResolver::ASIOExit() and improving
+//					portability through the use of conditional compilation
+//					1.0 Initial working version.
+// Created:			6/09/2003
+// Authors:         Fraser Adams
+//                  Ross Bencina
+//                  Rene G. Ceballos
+//                  Martin Fay
+//                  Antti Silvast
+//                  Andrew Baldwin
+//
+// ****************************************************************************
+
+
+#ifndef included_iasiothiscallresolver_h
+#define included_iasiothiscallresolver_h
+
+// We only need IASIOThiscallResolver at all if we are on Win32. For other
+// platforms we simply bypass the IASIOThiscallResolver definition to allow us
+// to be safely #include'd whatever the platform to keep client code portable
+#if defined(WIN32) || defined(_WIN32) || defined(__WIN32__)
+
+
+// If microsoft compiler we can call IASIO directly so IASIOThiscallResolver
+// is not used.
+#if !defined(_MSC_VER)
+
+
+// The following is in order to ensure that this header is only included after
+// the other ASIO headers (except for the case of iasiothiscallresolver.cpp).
+// We need to do this because IASIOThiscallResolver works by eclipsing the
+// original definition of ASIOInit() with a macro (see below).
+#if !defined(iasiothiscallresolver_sourcefile)
+	#if !defined(__ASIO_H)
+	#error iasiothiscallresolver.h must be included AFTER asio.h
+	#endif
+#endif
+
+#include <windows.h>
+#include "iasiodrv.h" /* From ASIO SDK */
+
+
+class IASIOThiscallResolver : public IASIO {
+private:
+	IASIO* that_; // Points to the real IASIO
+
+	static IASIOThiscallResolver instance; // Singleton instance
+
+	// Constructors - declared private so construction is limited to
+    // our Singleton instance
+    IASIOThiscallResolver();
+	IASIOThiscallResolver(IASIO* that);
+public:
+
+    // Methods from the IUnknown interface. We don't fully implement IUnknown
+    // because the ASIO SDK never calls these methods through theAsioDriver ptr.
+    // These methods are implemented as assert(false).
+    virtual HRESULT STDMETHODCALLTYPE QueryInterface(REFIID riid, void **ppv);
+    virtual ULONG STDMETHODCALLTYPE AddRef();
+    virtual ULONG STDMETHODCALLTYPE Release();
+
+    // Methods from the IASIO interface, implemented as forwarning calls to that.
+	virtual ASIOBool init(void *sysHandle);
+	virtual void getDriverName(char *name);
+	virtual long getDriverVersion();
+	virtual void getErrorMessage(char *string);
+	virtual ASIOError start();
+	virtual ASIOError stop();
+	virtual ASIOError getChannels(long *numInputChannels, long *numOutputChannels);
+	virtual ASIOError getLatencies(long *inputLatency, long *outputLatency);
+	virtual ASIOError getBufferSize(long *minSize, long *maxSize, long *preferredSize, long *granularity);
+	virtual ASIOError canSampleRate(ASIOSampleRate sampleRate);
+	virtual ASIOError getSampleRate(ASIOSampleRate *sampleRate);
+	virtual ASIOError setSampleRate(ASIOSampleRate sampleRate);
+	virtual ASIOError getClockSources(ASIOClockSource *clocks, long *numSources);
+	virtual ASIOError setClockSource(long reference);
+	virtual ASIOError getSamplePosition(ASIOSamples *sPos, ASIOTimeStamp *tStamp);
+	virtual ASIOError getChannelInfo(ASIOChannelInfo *info);
+	virtual ASIOError createBuffers(ASIOBufferInfo *bufferInfos, long numChannels, long bufferSize, ASIOCallbacks *callbacks);
+	virtual ASIOError disposeBuffers();
+	virtual ASIOError controlPanel();
+	virtual ASIOError future(long selector,void *opt);
+	virtual ASIOError outputReady();
+
+    // Class method, see ASIOInit() macro below.
+    static ASIOError ASIOInit(ASIODriverInfo *info); // Delegates to ::ASIOInit
+};
+
+
+// Replace calls to ASIOInit with our interposing version.
+// This macro enables us to perform thiscall resolution simply by #including
+// <iasiothiscallresolver.h> after the asio #includes (this file _must_ be
+// included _after_ the asio #includes)
+
+#define ASIOInit(name) IASIOThiscallResolver::ASIOInit((name))
+
+
+#endif /* !defined(_MSC_VER) */
+
+#endif /* Win32 */
+
+#endif /* included_iasiothiscallresolver_h */
+
+