* Initial import

Makefile.in

@@ -0,0 +1,99 @@
+
+all:	bin lib bin/rubberband lib/librubberband.a lib/vamp-rubberband.so lib/ladspa-rubberband.so
+
+CXX		= @CXX@
+CXXFLAGS	= @CXXFLAGS@ @SRC_CFLAGS@ @SNDFILE_CFLAGS@ @FFTW_CFLAGS@ @FFTWF_CFLAGS@ -Irubberband -Isrc
+LDFLAGS		= @LDFLAGS@ @SRC_LIBS@ @SNDFILE_LIBS@ @FFTW_LIBS@ @FFTWF_LIBS@
+
+MKDIR		= mkdir
+AR		= ar
+
+PROGRAM_TARGET := bin/rubberband
+STATIC_TARGET  := lib/librubberband.a
+DYNAMIC_TARGET := lib/librubberband.so
+VAMP_TARGET    := lib/vamp-rubberband.so
+LADSPA_TARGET  := lib/ladspa-rubberband.so
+
+PUBLIC_INCLUDES := \
+	rubberband/TimeStretcher.h \
+	rubberband/RubberBandStretcher.h
+
+LIBRARY_INCLUDES := \
+	src/AudioCurve.h \
+	src/ConstantAudioCurve.h \
+	src/FFT.h \
+	src/HighFrequencyAudioCurve.h \
+	src/PercussiveAudioCurve.h \
+	src/Resampler.h \
+	src/RingBuffer.h \
+	src/Scavenger.h \
+	src/StretchCalculator.h \
+	src/StretcherImpl.h \
+	src/StretcherChannelData.h \
+	src/Thread.h \
+	src/Window.h \
+	src/sysutils.h
+
+LIBRARY_SOURCES := \
+	src/RubberBandStretcher.cpp \
+	src/ConstantAudioCurve.cpp \
+	src/HighFrequencyAudioCurve.cpp \
+	src/PercussiveAudioCurve.cpp \
+	src/AudioCurve.cpp \
+	src/Resampler.cpp \
+	src/StretchCalculator.cpp \
+	src/StretcherImpl.cpp \
+	src/StretcherProcess.cpp \
+	src/StretcherChannelData.cpp \
+	src/FFT.cpp \
+	src/Thread.cpp \
+	src/sysutils.cpp
+
+PROGRAM_SOURCES := \
+	src/main.cpp
+
+VAMP_HEADERS := \
+	src/vamp/RubberBandVampPlugin.h
+
+VAMP_SOURCES := \
+	src/vamp/RubberBandVampPlugin.cpp \
+	src/vamp/libmain.cpp
+
+LADSPA_HEADERS := \
+	src/ladspa/RubberBandPitchShifter.h
+
+LADSPA_SOURCES := \
+	src/ladspa/RubberBandPitchShifter.cpp \
+	src/ladspa/libmain.cpp
+
+LIBRARY_OBJECTS := $(LIBRARY_SOURCES:.cpp=.o)
+PROGRAM_OBJECTS := $(PROGRAM_SOURCES:.cpp=.o)
+VAMP_OBJECTS    := $(VAMP_SOURCES:.cpp=.o)
+LADSPA_OBJECTS  := $(LADSPA_SOURCES:.cpp=.o)
+
+$(PROGRAM_TARGET):	$(LIBRARY_OBJECTS) $(PROGRAM_OBJECTS)
+	$(CXX) -o $@ $^ $(LDFLAGS)
+
+$(STATIC_TARGET):	$(LIBRARY_OBJECTS)
+	$(AR) rsc $@ $^
+
+$(DYNAMIC_TARGET):	$(LIBRARY_OBJECTS)
+	$(CXX) -shared -Wl,-Bsymbolic $^ -o $@ $(LDFLAGS)
+
+$(VAMP_TARGET):		$(LIBRARY_OBJECTS) $(VAMP_OBJECTS)
+	$(CXX) -shared -Wl,-Bsymbolic -o $@ $^ $(VAMP_PLUGIN_LIBS) $(LDFLAGS)
+
+$(LADSPA_TARGET):	$(LIBRARY_OBJECTS) $(LADSPA_OBJECTS)
+	$(CXX) -shared -Wl,-Bsymbolic -o $@ $^ $(LDFLAGS)
+
+bin:
+	$(MKDIR) $@
+lib:
+	$(MKDIR) $@
+
+clean:
+	rm -f $(LIBRARY_OBJECTS) $(PROGRAM_OBJECTS) $(LADSPA_OBJECTS) $(VAMP_OBJECTS)
+
+distclean:	clean
+	rm -f $(PROGRAM_TARGET) $(STATIC_TARGET) $(DYNAMIC_TARGET) $(VAMP_TARGET) $(LADSPA_TARGET)
+

configure.ac

@@ -0,0 +1,35 @@
+
+AC_INIT(RubberBand, 0.1, cannam@all-day-breakfast.com)
+
+AC_CONFIG_SRCDIR(src/StretcherImpl.h)
+AC_PROG_CXX
+AC_HEADER_STDC
+AC_C_BIGENDIAN
+
+PKG_CHECK_MODULES([SRC],[samplerate])
+AC_SUBST(SRC_CFLAGS)
+AC_SUBST(SRC_LIBS)
+
+PKG_CHECK_MODULES([SNDFILE],[sndfile])
+AC_SUBST(SNDFILE_CFLAGS)
+AC_SUBST(SNDFILE_LIBS)
+
+PKG_CHECK_MODULES([FFTWF],[fftw3f])
+AC_SUBST(FFTWF_CFLAGS)
+AC_SUBST(FFTWF_LIBS)
+
+PKG_CHECK_MODULES([FFTW],[fftw3])
+AC_SUBST(FFTW_CFLAGS)
+AC_SUBST(FFTW_LIBS)
+
+changequote(,)dnl
+if test "x$GCC" = "xyes"; then
+  case " $CXXFLAGS " in
+    *[\ \	]-Wall[\ \	]*) ;;
+    *) CXXFLAGS="$CXXFLAGS -Wall" ;;
+  esac
+fi
+changequote([,])dnl
+
+AC_OUTPUT([Makefile])
+

rubberband/RubberBandStretcher.h

@@ -0,0 +1,106 @@
+/* -*- c-basic-offset: 4 indent-tabs-mode: nil -*-  vi:set ts=8 sts=4 sw=4: */
+
+/*
+    Rubber Band
+    An audio time-stretching and pitch-shifting library.
+    Copyright 2007 Chris Cannam.
+    
+    This program is free software; you can redistribute it and/or
+    modify it under the terms of the GNU General Public License as
+    published by the Free Software Foundation; either version 2 of the
+    License, or (at your option) any later version.  See the file
+    COPYING included with this distribution for more information.
+*/
+
+#ifndef _RUBBERBANDSTRETCHER_H_
+#define _RUBBERBANDSTRETCHER_H_
+
+#include "TimeStretcher.h"
+
+#include <vector>
+
+namespace RubberBand
+{
+
+class RubberBandStretcher : public TimeStretcher
+{
+public:
+
+    typedef int Options;
+    
+    static const int OptionProcessOffline   = 0x00000000;
+    static const int OptionProcessRealTime  = 0x00000001;
+    
+    static const int OptionStretchElastic   = 0x00000000;
+    static const int OptionStretchPrecise   = 0x00000010;
+    
+    static const int OptionTransientsCrisp  = 0x00000000;
+    static const int OptionTransientsSmooth = 0x00000100;
+
+    static const int OptionPhasePeakLocked  = 0x00000000;
+    static const int OptionPhaseIndependent = 0x00001000;
+    
+    static const int OptionThreadingAuto    = 0x00000000;
+    static const int OptionThreadingNone    = 0x00010000;
+
+    static const int OptionWindowStandard   = 0x00000000;
+    static const int OptionWindowShort      = 0x00100000;
+    static const int OptionWindowLong       = 0x00200000;
+
+    static const int DefaultOptions         = 0x00000000;
+    static const int PercussiveOptions      = OptionWindowShort | \
+                                              OptionPhaseIndependent;
+
+    RubberBandStretcher(size_t sampleRate,
+                        size_t channels,
+                        Options options = DefaultOptions,
+                        double initialTimeRatio = 1.0,
+                        double initialPitchScale = 1.0);
+    virtual ~RubberBandStretcher();
+    
+    virtual void reset();
+    virtual void setTimeRatio(double ratio);
+    virtual void setPitchScale(double scale); //!!!??? pitch ratio?
+
+    virtual double getTimeRatio() const;
+    virtual double getPitchScale() const;
+
+    virtual size_t getLatency() const;
+
+    virtual void setTransientsOption(Options options);
+    virtual void setPhaseOption(Options options);
+
+    virtual void setExpectedInputDuration(size_t samples);
+    virtual void setMaxProcessBlockSize(size_t samples);
+    virtual size_t getSamplesRequired() const;
+
+    // if samples == 0, input may be null
+    virtual void study(const float *const *input, size_t samples, bool final);
+    virtual void process(const float *const *input, size_t samples, bool final);
+
+    virtual int available() const; // returns -1 if all data processed and nothing further will be available
+    virtual size_t retrieve(float *const *output, size_t samples) const;
+
+    virtual float getFrequencyCutoff(int n) const;
+    virtual void setFrequencyCutoff(int n, float f);
+    
+    //!!! ideally, this stuff wouldn't be here...
+
+    virtual size_t getInputIncrement() const;
+    virtual std::vector<int> getOutputIncrements() const; //!!! document particular meaning in RT mode
+    virtual std::vector<float> getLockCurve() const; //!!! document particular meaning in RT mode
+
+    virtual size_t getChannelCount() const;
+    
+    virtual void calculateStretch();
+
+    virtual void setDebugLevel(int level);
+
+protected:
+    class Impl;
+    Impl *m_d;
+};
+
+}
+
+#endif

rubberband/TimeStretcher.h

@@ -0,0 +1,51 @@
+/* -*- c-basic-offset: 4 indent-tabs-mode: nil -*-  vi:set ts=8 sts=4 sw=4: */
+
+/*
+    Rubber Band
+    An audio time-stretching and pitch-shifting library.
+    Copyright 2007 Chris Cannam.
+    
+    This program is free software; you can redistribute it and/or
+    modify it under the terms of the GNU General Public License as
+    published by the Free Software Foundation; either version 2 of the
+    License, or (at your option) any later version.  See the file
+    COPYING included with this distribution for more information.
+*/
+
+#ifndef _RUBBERBAND_TIMESTRETCHER_H_
+#define _RUBBERBAND_TIMESTRETCHER_H_
+
+#include <sys/types.h>
+
+namespace RubberBand
+{
+
+class TimeStretcher
+{
+public:
+    TimeStretcher(size_t sampleRate, size_t channels) :
+        m_sampleRate(sampleRate),
+        m_channels(channels)
+    { }
+    virtual ~TimeStretcher()
+    { }
+
+    virtual void reset() = 0;
+    virtual void setTimeRatio(double ratio) = 0;
+    virtual size_t getLatency() const = 0;
+
+    virtual void study(const float *const *input, size_t samples, bool final) = 0;
+    virtual size_t getSamplesRequired() const = 0; // to cause processing to happen
+    virtual void process(const float *const *input, size_t samples, bool final) = 0;
+    virtual int available() const = 0;
+    virtual size_t retrieve(float *const *output, size_t samples) const = 0;
+
+protected:
+    size_t m_sampleRate;
+    size_t m_channels;
+};
+
+}
+
+#endif
+    

src/AudioCurve.cpp

@@ -0,0 +1,31 @@
+/* -*- c-basic-offset: 4 indent-tabs-mode: nil -*-  vi:set ts=8 sts=4 sw=4: */
+
+/*
+    Rubber Band
+    An audio time-stretching and pitch-shifting library.
+    Copyright 2007 Chris Cannam.
+    
+    This program is free software; you can redistribute it and/or
+    modify it under the terms of the GNU General Public License as
+    published by the Free Software Foundation; either version 2 of the
+    License, or (at your option) any later version.  See the file
+    COPYING included with this distribution for more information.
+*/
+
+#include "AudioCurve.h"
+
+namespace RubberBand
+{
+
+AudioCurve::AudioCurve(size_t sampleRate, size_t blockSize) :
+    m_sampleRate(sampleRate),
+    m_blockSize(blockSize)
+{
+}
+
+AudioCurve::~AudioCurve()
+{
+}
+
+
+}

src/AudioCurve.h

@@ -0,0 +1,42 @@
+/* -*- c-basic-offset: 4 indent-tabs-mode: nil -*-  vi:set ts=8 sts=4 sw=4: */
+
+/*
+    Rubber Band
+    An audio time-stretching and pitch-shifting library.
+    Copyright 2007 Chris Cannam.
+    
+    This program is free software; you can redistribute it and/or
+    modify it under the terms of the GNU General Public License as
+    published by the Free Software Foundation; either version 2 of the
+    License, or (at your option) any later version.  See the file
+    COPYING included with this distribution for more information.
+*/
+
+#ifndef _AUDIO_CURVE_H_
+#define _AUDIO_CURVE_H_
+
+#include <sys/types.h>
+
+namespace RubberBand 
+{
+
+class AudioCurve
+{
+public:
+    AudioCurve(size_t sampleRate, size_t blockSize);
+    virtual ~AudioCurve();
+
+    virtual void setBlockSize(size_t newSize) = 0;
+    
+    virtual float process(float *mag, size_t increment) = 0;
+    virtual void reset() = 0;
+
+protected:
+    size_t m_sampleRate;
+    size_t m_blockSize;
+};
+
+}
+
+#endif
+

src/ConstantAudioCurve.cpp

@@ -0,0 +1,47 @@
+/* -*- c-basic-offset: 4 indent-tabs-mode: nil -*-  vi:set ts=8 sts=4 sw=4: */
+
+/*
+    Rubber Band
+    An audio time-stretching and pitch-shifting library.
+    Copyright 2007 Chris Cannam.
+    
+    This program is free software; you can redistribute it and/or
+    modify it under the terms of the GNU General Public License as
+    published by the Free Software Foundation; either version 2 of the
+    License, or (at your option) any later version.  See the file
+    COPYING included with this distribution for more information.
+*/
+
+#include "ConstantAudioCurve.h"
+
+namespace RubberBand
+{
+
+ConstantAudioCurve::ConstantAudioCurve(size_t sampleRate, size_t blockSize) :
+    AudioCurve(sampleRate, blockSize)
+{
+}
+
+ConstantAudioCurve::~ConstantAudioCurve()
+{
+}
+
+void
+ConstantAudioCurve::reset()
+{
+}
+
+void
+ConstantAudioCurve::setBlockSize(size_t newSize)
+{
+    m_blockSize = newSize;
+}
+
+float
+ConstantAudioCurve::process(float *, size_t)
+{
+    return 1.f;
+}
+
+}
+

src/ConstantAudioCurve.h

@@ -0,0 +1,37 @@
+/* -*- c-basic-offset: 4 indent-tabs-mode: nil -*-  vi:set ts=8 sts=4 sw=4: */
+
+/*
+    Rubber Band
+    An audio time-stretching and pitch-shifting library.
+    Copyright 2007 Chris Cannam.
+    
+    This program is free software; you can redistribute it and/or
+    modify it under the terms of the GNU General Public License as
+    published by the Free Software Foundation; either version 2 of the
+    License, or (at your option) any later version.  See the file
+    COPYING included with this distribution for more information.
+*/
+
+#ifndef _CONSTANT_AUDIO_CURVE_H_
+#define _CONSTANT_AUDIO_CURVE_H_
+
+#include "AudioCurve.h"
+
+namespace RubberBand
+{
+
+class ConstantAudioCurve : public AudioCurve
+{
+public:
+    ConstantAudioCurve(size_t sampleRate, size_t blockSize);
+    virtual ~ConstantAudioCurve();
+
+    virtual void setBlockSize(size_t newSize);
+
+    virtual float process(float *mag, size_t increment);
+    virtual void reset();
+};
+
+}
+
+#endif

src/FFT.cpp

@@ -0,0 +1,606 @@
+/* -*- c-basic-offset: 4 indent-tabs-mode: nil -*-  vi:set ts=8 sts=4 sw=4: */
+
+/*
+    Rubber Band
+    An audio time-stretching and pitch-shifting library.
+    Copyright 2007 Chris Cannam.
+    
+    This program is free software; you can redistribute it and/or
+    modify it under the terms of the GNU General Public License as
+    published by the Free Software Foundation; either version 2 of the
+    License, or (at your option) any later version.  See the file
+    COPYING included with this distribution for more information.
+*/
+
+#include "FFT.h"
+
+
+#include <fftw3.h>
+
+#include <cmath>
+#include <iostream>
+#include <map>
+#include <vector>
+
+class FFTImpl
+{
+public:
+    virtual ~FFTImpl() { }
+
+    virtual void initFloat() = 0;
+    virtual void initDouble() = 0;
+
+    virtual void forward(double *realIn, double *realOut, double *imagOut) = 0;
+    virtual void forwardPolar(double *realIn, double *magOut, double *phaseOut) = 0;
+    virtual void forwardMagnitude(double *realIn, double *magOut) = 0;
+
+    virtual void forward(float *realIn, float *realOut, float *imagOut) = 0;
+    virtual void forwardPolar(float *realIn, float *magOut, float *phaseOut) = 0;
+    virtual void forwardMagnitude(float *realIn, float *magOut) = 0;
+
+    virtual void inverse(double *realIn, double *imagIn, double *realOut) = 0;
+    virtual void inversePolar(double *magIn, double *phaseIn, double *realOut) = 0;
+
+    virtual void inverse(float *realIn, float *imagIn, float *realOut) = 0;
+    virtual void inversePolar(float *magIn, float *phaseIn, float *realOut) = 0;
+};    
+
+
+
+
+class D_FFTW : public FFTImpl
+{
+public:
+    D_FFTW(unsigned int size) : m_size(size), m_fplanf(0), m_dplanf(0) { 
+    }
+
+    ~D_FFTW() {
+        if (m_fplanf) {
+            fftwf_destroy_plan(m_fplanf);
+            fftwf_destroy_plan(m_fplani);
+            fftwf_free(m_fbuf);
+            fftwf_free(m_fpacked);
+        }
+        if (m_dplanf) {
+            fftw_destroy_plan(m_dplanf);
+            fftw_destroy_plan(m_dplani);
+            fftw_free(m_dbuf);
+            fftw_free(m_dpacked);
+        }
+    }
+
+    //!!! check return values
+
+    void initFloat() {
+        if (m_fplanf) return;
+        m_fbuf = (float *)fftw_malloc(m_size * sizeof(float));
+        m_fpacked = (fftwf_complex *)fftw_malloc
+            ((m_size/2 + 1) * sizeof(fftwf_complex));
+        m_fplanf = fftwf_plan_dft_r2c_1d
+            (m_size, m_fbuf, m_fpacked, FFTW_MEASURE);
+        m_fplani = fftwf_plan_dft_c2r_1d
+            (m_size, m_fpacked, m_fbuf, FFTW_MEASURE);
+    }
+
+    void initDouble() {
+        if (m_dplanf) return;
+        m_dbuf = (double *)fftw_malloc(m_size * sizeof(double));
+        m_dpacked = (fftw_complex *)fftw_malloc
+            ((m_size/2 + 1) * sizeof(fftw_complex));
+        m_dplanf = fftw_plan_dft_r2c_1d
+            (m_size, m_dbuf, m_dpacked, FFTW_MEASURE);
+        m_dplani = fftw_plan_dft_c2r_1d
+            (m_size, m_dpacked, m_dbuf, FFTW_MEASURE);
+    }
+
+    void packFloat(float *re, float *im) {
+        for (unsigned int i = 0; i <= m_size/2; ++i) {
+            m_fpacked[i][0] = re[i];
+            m_fpacked[i][1] = im[i];
+        }
+    }
+
+    void packDouble(double *re, double *im) {
+        for (unsigned int i = 0; i <= m_size/2; ++i) {
+            m_dpacked[i][0] = re[i];
+            m_dpacked[i][1] = im[i];
+        }
+    }
+
+    void unpackFloat(float *re, float *im) {
+        for (unsigned int i = 0; i <= m_size/2; ++i) {
+            re[i] = m_fpacked[i][0];
+            im[i] = m_fpacked[i][1];
+        }
+    }        
+
+    void unpackDouble(double *re, double *im) {
+        for (unsigned int i = 0; i <= m_size/2; ++i) {
+            re[i] = m_dpacked[i][0];
+            im[i] = m_dpacked[i][1];
+        }
+    }        
+
+    void forward(double *realIn, double *realOut, double *imagOut) {
+        if (!m_dplanf) initDouble();
+        for (unsigned int i = 0; i < m_size; ++i) {
+            m_dbuf[i] = realIn[i];
+        }
+        fftw_execute(m_dplanf);
+        unpackDouble(realOut, imagOut);
+    }
+
+    void forwardPolar(double *realIn, double *magOut, double *phaseOut) {
+        if (!m_dplanf) initDouble();
+        for (unsigned int i = 0; i < m_size; ++i) {
+            m_dbuf[i] = realIn[i];
+        }
+        fftw_execute(m_dplanf);
+        for (unsigned int i = 0; i <= m_size/2; ++i) {
+            magOut[i] = sqrt(m_dpacked[i][0] * m_dpacked[i][0] +
+                             m_dpacked[i][1] * m_dpacked[i][1]);
+            phaseOut[i] = atan2(m_dpacked[i][1], m_dpacked[i][0]);
+        }
+    }
+
+    void forwardMagnitude(double *realIn, double *magOut) {
+        if (!m_dplanf) initDouble();
+        for (unsigned int i = 0; i < m_size; ++i) {
+            m_dbuf[i] = realIn[i];
+        }
+        fftw_execute(m_dplanf);
+        for (unsigned int i = 0; i <= m_size/2; ++i) {
+            magOut[i] = sqrt(m_dpacked[i][0] * m_dpacked[i][0] +
+                             m_dpacked[i][1] * m_dpacked[i][1]);
+        }
+    }
+
+    void forward(float *realIn, float *realOut, float *imagOut) {
+        if (!m_fplanf) initFloat();
+        for (unsigned int i = 0; i < m_size; ++i) {
+            m_fbuf[i] = realIn[i];
+        }
+        fftwf_execute(m_fplanf);
+        unpackFloat(realOut, imagOut);
+    }
+
+    void forwardPolar(float *realIn, float *magOut, float *phaseOut) {
+        if (!m_fplanf) initFloat();
+        for (unsigned int i = 0; i < m_size; ++i) {
+            m_fbuf[i] = realIn[i];
+        }
+        fftwf_execute(m_fplanf);
+        for (unsigned int i = 0; i <= m_size/2; ++i) {
+            magOut[i] = sqrtf(m_fpacked[i][0] * m_fpacked[i][0] +
+                              m_fpacked[i][1] * m_fpacked[i][1]);
+          phaseOut[i] = atan2f(m_fpacked[i][1], m_fpacked[i][0]) ;
+        }
+    }
+
+    void forwardMagnitude(float *realIn, float *magOut) {
+        if (!m_fplanf) initFloat();
+        for (unsigned int i = 0; i < m_size; ++i) {
+            m_fbuf[i] = realIn[i];
+        }
+        fftwf_execute(m_fplanf);
+        for (unsigned int i = 0; i <= m_size/2; ++i) {
+            magOut[i] = sqrtf(m_fpacked[i][0] * m_fpacked[i][0] +
+                              m_fpacked[i][1] * m_fpacked[i][1]);
+        }
+    }
+
+    void inverse(double *realIn, double *imagIn, double *realOut) {
+        if (!m_dplanf) initDouble();
+        packDouble(realIn, imagIn);
+        fftw_execute(m_dplani);
+        for (unsigned int i = 0; i < m_size; ++i) {
+            realOut[i] = m_dbuf[i];
+        }
+    }
+
+    void inversePolar(double *magIn, double *phaseIn, double *realOut) {
+        if (!m_dplanf) initDouble();
+        for (unsigned int i = 0; i <= m_size/2; ++i) {
+            m_dpacked[i][0] = magIn[i] * cos(phaseIn[i]);
+            m_dpacked[i][1] = magIn[i] * sin(phaseIn[i]);
+        }
+        fftw_execute(m_dplani);
+        for (unsigned int i = 0; i < m_size; ++i) {
+            realOut[i] = m_dbuf[i];
+        }
+    }
+
+    void inverse(float *realIn, float *imagIn, float *realOut) {
+        if (!m_fplanf) initFloat();
+        packFloat(realIn, imagIn);
+        fftwf_execute(m_fplani);
+        for (unsigned int i = 0; i < m_size; ++i) {
+            realOut[i] = m_fbuf[i];
+        }
+    }
+
+    void inversePolar(float *magIn, float *phaseIn, float *realOut) {
+        if (!m_fplanf) initFloat();
+        for (unsigned int i = 0; i <= m_size/2; ++i) {
+            m_fpacked[i][0] = magIn[i] * cosf(phaseIn[i]);
+            m_fpacked[i][1] = magIn[i] * sinf(phaseIn[i]);
+        }
+        fftwf_execute(m_fplani);
+        for (unsigned int i = 0; i < m_size; ++i) {
+            realOut[i] = m_fbuf[i];
+        }
+    }
+
+private:
+    unsigned int m_size;
+    fftwf_plan m_fplanf;
+    fftwf_plan m_fplani;
+    fftw_plan m_dplanf;
+    fftw_plan m_dplani;
+    float *m_fbuf;
+    fftwf_complex *m_fpacked;
+    double *m_dbuf;
+    fftw_complex *m_dpacked;
+};
+
+
+class D_Cross : public FFTImpl
+{
+public:
+    D_Cross(unsigned int size) : m_size(size), m_table(0) {
+        
+        m_a = new double[size];
+        m_b = new double[size];
+        m_c = new double[size];
+        m_d = new double[size];
+
+        m_table = new int[m_size];
+    
+        unsigned int bits;
+        unsigned int i, j, k, m;
+
+        for (i = 0; ; ++i) {
+            if (m_size & (1 << i)) {
+                bits = i;
+                break;
+            }
+        }
+        
+        for (i = 0; i < m_size; ++i) {
+            
+            m = i;
+            
+            for (j = k = 0; j < bits; ++j) {
+                k = (k << 1) | (m & 1);
+                m >>= 1;
+            }
+            
+            m_table[i] = k;
+        }
+    }
+
+    ~D_Cross() {
+        delete[] m_table;
+        delete[] m_a;
+        delete[] m_b;
+        delete[] m_c;
+        delete[] m_d;
+    }
+
+    void initFloat() { }
+    void initDouble() { }
+
+    void forward(double *realIn, double *realOut, double *imagOut) {
+        basefft(false, realIn, 0, m_c, m_d);
+        for (size_t i = 0; i <= m_size/2; ++i) realOut[i] = m_c[i];
+        for (size_t i = 0; i <= m_size/2; ++i) imagOut[i] = m_d[i];
+    }
+
+    void forwardPolar(double *realIn, double *magOut, double *phaseOut) {
+        basefft(false, realIn, 0, m_c, m_d);
+        for (unsigned int i = 0; i <= m_size/2; ++i) {
+            magOut[i] = sqrt(m_c[i] * m_c[i] + m_d[i] * m_d[i]);
+            phaseOut[i] = atan2(m_d[i], m_c[i]) ;
+        }
+    }
+
+    void forwardMagnitude(double *realIn, double *magOut) {
+        basefft(false, realIn, 0, m_c, m_d);
+        for (unsigned int i = 0; i <= m_size/2; ++i) {
+            magOut[i] = sqrt(m_c[i] * m_c[i] + m_d[i] * m_d[i]);
+        }
+    }
+
+    void forward(float *realIn, float *realOut, float *imagOut) {
+        for (size_t i = 0; i < m_size; ++i) m_a[i] = realIn[i];
+        basefft(false, m_a, 0, m_c, m_d);
+        for (size_t i = 0; i <= m_size/2; ++i) realOut[i] = m_c[i];
+        for (size_t i = 0; i <= m_size/2; ++i) imagOut[i] = m_d[i];
+    }
+
+    void forwardPolar(float *realIn, float *magOut, float *phaseOut) {
+        for (size_t i = 0; i < m_size; ++i) m_a[i] = realIn[i];
+        basefft(false, m_a, 0, m_c, m_d);
+        for (unsigned int i = 0; i <= m_size/2; ++i) {
+            magOut[i] = sqrt(m_c[i] * m_c[i] + m_d[i] * m_d[i]);
+            phaseOut[i] = atan2(m_d[i], m_c[i]) ;
+        }
+    }
+
+    void forwardMagnitude(float *realIn, float *magOut) {
+        for (size_t i = 0; i < m_size; ++i) m_a[i] = realIn[i];
+        basefft(false, m_a, 0, m_c, m_d);
+        for (unsigned int i = 0; i <= m_size/2; ++i) {
+            magOut[i] = sqrt(m_c[i] * m_c[i] + m_d[i] * m_d[i]);
+        }
+    }
+
+    void inverse(double *realIn, double *imagIn, double *realOut) {
+        for (unsigned int i = 0; i <= m_size/2; ++i) {
+            double real = realIn[i];
+            double imag = imagIn[i];
+            m_a[i] = real;
+            m_b[i] = imag;
+            if (i > 0) {
+                m_a[m_size-i] = real;
+                m_b[m_size-i] = -imag;
+            }
+        }
+        basefft(true, m_a, m_b, realOut, m_d);
+    }
+
+    void inversePolar(double *magIn, double *phaseIn, double *realOut) {
+        for (unsigned int i = 0; i <= m_size/2; ++i) {
+            double real = magIn[i] * cos(phaseIn[i]);
+            double imag = magIn[i] * sin(phaseIn[i]);
+            m_a[i] = real;
+            m_b[i] = imag;
+            if (i > 0) {
+                m_a[m_size-i] = real;
+                m_b[m_size-i] = -imag;
+            }
+        }
+        basefft(true, m_a, m_b, realOut, m_d);
+    }
+
+    void inverse(float *realIn, float *imagIn, float *realOut) {
+        for (unsigned int i = 0; i <= m_size/2; ++i) {
+            float real = realIn[i];
+            float imag = imagIn[i];
+            m_a[i] = real;
+            m_b[i] = imag;
+            if (i > 0) {
+                m_a[m_size-i] = real;
+                m_b[m_size-i] = -imag;
+            }
+        }
+        basefft(true, m_a, m_b, m_c, m_d);
+        for (unsigned int i = 0; i < m_size; ++i) realOut[i] = m_c[i];
+    }
+
+    void inversePolar(float *magIn, float *phaseIn, float *realOut) {
+        for (unsigned int i = 0; i <= m_size/2; ++i) {
+            float real = magIn[i] * cosf(phaseIn[i]);
+            float imag = magIn[i] * sinf(phaseIn[i]);
+            m_a[i] = real;
+            m_b[i] = imag;
+            if (i > 0) {
+                m_a[m_size-i] = real;
+                m_b[m_size-i] = -imag;
+            }
+        }
+        basefft(true, m_a, m_b, m_c, m_d);
+        for (unsigned int i = 0; i < m_size; ++i) realOut[i] = m_c[i];
+    }
+
+private:
+    unsigned int m_size;
+    int *m_table;
+    double *m_a;
+    double *m_b;
+    double *m_c;
+    double *m_d;
+    void basefft(bool inverse, double *ri, double *ii, double *ro, double *io);
+};
+
+void
+D_Cross::basefft(bool inverse, double *ri, double *ii, double *ro, double *io)
+{
+    if (!ri || !ro || !io) return;
+
+    unsigned int i, j, k, m;
+    unsigned int blockSize, blockEnd;
+
+    double tr, ti;
+
+    double angle = 2.0 * M_PI;
+    if (inverse) angle = -angle;
+
+    const unsigned int n = m_size;
+
+    if (ii) {
+	for (i = 0; i < n; ++i) {
+	    ro[m_table[i]] = ri[i];
+	    io[m_table[i]] = ii[i];
+	}
+    } else {
+	for (i = 0; i < n; ++i) {
+	    ro[m_table[i]] = ri[i];
+	    io[m_table[i]] = 0.0;
+	}
+    }
+
+    blockEnd = 1;
+
+    for (blockSize = 2; blockSize <= n; blockSize <<= 1) {
+
+	double delta = angle / (double)blockSize;
+	double sm2 = -sin(-2 * delta);
+	double sm1 = -sin(-delta);
+	double cm2 = cos(-2 * delta);
+	double cm1 = cos(-delta);
+	double w = 2 * cm1;
+	double ar[3], ai[3];
+
+	for (i = 0; i < n; i += blockSize) {
+
+	    ar[2] = cm2;
+	    ar[1] = cm1;
+
+	    ai[2] = sm2;
+	    ai[1] = sm1;
+
+	    for (j = i, m = 0; m < blockEnd; j++, m++) {
+
+		ar[0] = w * ar[1] - ar[2];
+		ar[2] = ar[1];
+		ar[1] = ar[0];
+
+		ai[0] = w * ai[1] - ai[2];
+		ai[2] = ai[1];
+		ai[1] = ai[0];
+
+		k = j + blockEnd;
+		tr = ar[0] * ro[k] - ai[0] * io[k];
+		ti = ar[0] * io[k] + ai[0] * ro[k];
+
+		ro[k] = ro[j] - tr;
+		io[k] = io[j] - ti;
+
+		ro[j] += tr;
+		io[j] += ti;
+	    }
+	}
+
+	blockEnd = blockSize;
+    }
+
+/* fftw doesn't rescale, so nor will we
+
+    if (inverse) {
+
+	double denom = (double)n;
+
+	for (i = 0; i < n; i++) {
+	    ro[i] /= denom;
+	    io[i] /= denom;
+	}
+    }
+*/
+}
+
+int
+FFT::m_method = -1;
+
+FFT::FFT(unsigned int size)
+{
+    if (size < 2) throw InvalidSize;
+    if (size & (size-1)) throw InvalidSize;
+
+    if (m_method == -1) {
+        m_method = 1;
+    }
+
+    switch (m_method) {
+
+    case 0:
+        d = new D_Cross(size);
+        break;
+
+    case 1:
+        std::cerr << "FFT::FFT(" << size << "): using FFTW3 implementation"
+                  << std::endl;
+        d = new D_FFTW(size);
+        break;
+
+    default:
+        std::cerr << "FFT::FFT(" << size << "): using built-in implementation"
+                  << std::endl;
+        d = new D_Cross(size);
+        break;
+    }
+}
+
+FFT::~FFT()
+{
+    delete d;
+}
+
+void
+FFT::forward(double *realIn, double *realOut, double *imagOut)
+{
+    d->forward(realIn, realOut, imagOut);
+}
+
+void
+FFT::forwardPolar(double *realIn, double *magOut, double *phaseOut)
+{
+    d->forwardPolar(realIn, magOut, phaseOut);
+}
+
+void
+FFT::forwardMagnitude(double *realIn, double *magOut)
+{
+    d->forwardMagnitude(realIn, magOut);
+}
+
+void
+FFT::forward(float *realIn, float *realOut, float *imagOut)
+{
+    d->forward(realIn, realOut, imagOut);
+}
+
+void
+FFT::forwardPolar(float *realIn, float *magOut, float *phaseOut)
+{
+    d->forwardPolar(realIn, magOut, phaseOut);
+}
+
+void
+FFT::forwardMagnitude(float *realIn, float *magOut)
+{
+    d->forwardMagnitude(realIn, magOut);
+}
+
+void
+FFT::inverse(double *realIn, double *imagIn, double *realOut)
+{
+    d->inverse(realIn, imagIn, realOut);
+}
+
+void
+FFT::inversePolar(double *magIn, double *phaseIn, double *realOut)
+{
+    d->inversePolar(magIn, phaseIn, realOut);
+}
+
+void
+FFT::inverse(float *realIn, float *imagIn, float *realOut)
+{
+    d->inverse(realIn, imagIn, realOut);
+}
+
+void
+FFT::inversePolar(float *magIn, float *phaseIn, float *realOut)
+{
+    d->inversePolar(magIn, phaseIn, realOut);
+}
+
+void
+FFT::initFloat() 
+{
+    d->initFloat();
+}
+
+void
+FFT::initDouble() 
+{
+    d->initDouble();
+}
+
+
+void
+FFT::tune()
+{
+}
+

src/FFT.h

@@ -0,0 +1,70 @@
+/* -*- c-basic-offset: 4 indent-tabs-mode: nil -*-  vi:set ts=8 sts=4 sw=4: */
+
+/*
+    Rubber Band
+    An audio time-stretching and pitch-shifting library.
+    Copyright 2007 Chris Cannam.
+    
+    This program is free software; you can redistribute it and/or
+    modify it under the terms of the GNU General Public License as
+    published by the Free Software Foundation; either version 2 of the
+    License, or (at your option) any later version.  See the file
+    COPYING included with this distribution for more information.
+*/
+
+#ifndef _RUBBERBAND_FFT_H_
+#define _RUBBERBAND_FFT_H_
+
+class FFTImpl;
+
+
+/**
+ * Provide the basic FFT computations we need, using one of a set of
+ * candidate FFT implementations (depending on compile flags).
+ *
+ * Implements real->complex FFTs of power-of-two sizes only.  Note
+ * that only the first half of the output signal is returned (the
+ * complex conjugates half is omitted), so the "complex" arrays need
+ * room for size/2+1 elements.
+ *
+ * Not thread safe: use a separate instance per thread.
+ */
+
+class FFT
+{
+public:
+    enum Exception { InvalidSize };
+
+    FFT(unsigned int size); // may throw InvalidSize
+    ~FFT();
+
+    void forward(double *realIn, double *realOut, double *imagOut);
+    void forwardPolar(double *realIn, double *magOut, double *phaseOut);
+    void forwardMagnitude(double *realIn, double *magOut);
+
+    void forward(float *realIn, float *realOut, float *imagOut);
+    void forwardPolar(float *realIn, float *magOut, float *phaseOut);
+    void forwardMagnitude(float *realIn, float *magOut);
+
+    void inverse(double *realIn, double *imagIn, double *realOut);
+    void inversePolar(double *magIn, double *phaseIn, double *realOut);
+
+    void inverse(float *realIn, float *imagIn, float *realOut);
+    void inversePolar(float *magIn, float *phaseIn, float *realOut);
+
+    // Calling one or both of these is optional -- if neither is
+    // called, the first call to a forward or inverse method will call
+    // init().  You only need call these if you don't want to risk
+    // expensive allocations etc happening in forward or inverse.
+    void initFloat();
+    void initDouble();
+
+    static void tune();
+
+protected:
+    FFTImpl *d;
+    static int m_method;
+};
+
+#endif
+

src/HighFrequencyAudioCurve.cpp

@@ -0,0 +1,62 @@
+/* -*- c-basic-offset: 4 indent-tabs-mode: nil -*-  vi:set ts=8 sts=4 sw=4: */
+
+/*
+    Rubber Band
+    An audio time-stretching and pitch-shifting library.
+    Copyright 2007 Chris Cannam.
+    
+    This program is free software; you can redistribute it and/or
+    modify it under the terms of the GNU General Public License as
+    published by the Free Software Foundation; either version 2 of the
+    License, or (at your option) any later version.  See the file
+    COPYING included with this distribution for more information.
+*/
+
+#include "HighFrequencyAudioCurve.h"
+
+namespace RubberBand
+{
+
+HighFrequencyAudioCurve::HighFrequencyAudioCurve(size_t sampleRate, size_t blockSize) :
+    AudioCurve(sampleRate, blockSize)
+{
+    m_prevMag = new double[m_blockSize/2 + 1];
+
+    for (size_t i = 0; i <= m_blockSize/2; ++i) {
+        m_prevMag[i] = 0.f;
+    }
+}
+
+HighFrequencyAudioCurve::~HighFrequencyAudioCurve()
+{
+    delete[] m_prevMag;
+}
+
+void
+HighFrequencyAudioCurve::reset()
+{
+    for (size_t i = 0; i <= m_blockSize/2; ++i) {
+        m_prevMag[i] = 0;
+    }
+}
+
+void
+HighFrequencyAudioCurve::setBlockSize(size_t newSize)
+{
+    m_blockSize = newSize;
+}
+
+float
+HighFrequencyAudioCurve::process(float *mag, size_t increment)
+{
+    float result = 0.0;
+
+    for (size_t n = 0; n <= m_blockSize / 2; ++n) {
+        result += mag[n];
+    }
+
+    return result;
+}
+
+}
+

src/HighFrequencyAudioCurve.h

@@ -0,0 +1,42 @@
+/* -*- c-basic-offset: 4 indent-tabs-mode: nil -*-  vi:set ts=8 sts=4 sw=4: */
+
+/*
+    Rubber Band
+    An audio time-stretching and pitch-shifting library.
+    Copyright 2007 Chris Cannam.
+    
+    This program is free software; you can redistribute it and/or
+    modify it under the terms of the GNU General Public License as
+    published by the Free Software Foundation; either version 2 of the
+    License, or (at your option) any later version.  See the file
+    COPYING included with this distribution for more information.
+*/
+
+#ifndef _HIGHFREQUENCY_AUDIO_CURVE_H_
+#define _HIGHFREQUENCY_AUDIO_CURVE_H_
+
+#include "AudioCurve.h"
+#include "Window.h"
+
+namespace RubberBand
+{
+
+class HighFrequencyAudioCurve : public AudioCurve
+{
+public:
+    HighFrequencyAudioCurve(size_t sampleRate, size_t blockSize);
+
+    virtual ~HighFrequencyAudioCurve();
+
+    virtual void setBlockSize(size_t newSize);
+
+    virtual float process(float *mag, size_t increment);
+    virtual void reset();
+
+protected:
+    double *m_prevMag;
+};
+
+}
+
+#endif

src/PercussiveAudioCurve.cpp

@@ -0,0 +1,80 @@
+/* -*- c-basic-offset: 4 indent-tabs-mode: nil -*-  vi:set ts=8 sts=4 sw=4: */
+
+/*
+    Rubber Band
+    An audio time-stretching and pitch-shifting library.
+    Copyright 2007 Chris Cannam.
+    
+    This program is free software; you can redistribute it and/or
+    modify it under the terms of the GNU General Public License as
+    published by the Free Software Foundation; either version 2 of the
+    License, or (at your option) any later version.  See the file
+    COPYING included with this distribution for more information.
+*/
+
+#include "PercussiveAudioCurve.h"
+
+#include <cmath>
+
+namespace RubberBand
+{
+
+PercussiveAudioCurve::PercussiveAudioCurve(size_t sampleRate, size_t blockSize) :
+    AudioCurve(sampleRate, blockSize)
+{
+    m_prevMag = new double[m_blockSize/2 + 1];
+
+    for (size_t i = 0; i <= m_blockSize/2; ++i) {
+        m_prevMag[i] = 0.f;
+    }
+}
+
+PercussiveAudioCurve::~PercussiveAudioCurve()
+{
+    delete[] m_prevMag;
+}
+
+void
+PercussiveAudioCurve::reset()
+{
+    for (size_t i = 0; i <= m_blockSize/2; ++i) {
+        m_prevMag[i] = 0;
+    }
+}
+
+void
+PercussiveAudioCurve::setBlockSize(size_t newSize)
+{
+    delete[] m_prevMag;
+    m_blockSize = newSize;
+    
+    m_prevMag = new double[m_blockSize/2 + 1];
+
+    reset();
+}
+
+float
+PercussiveAudioCurve::process(float *mag, size_t increment)
+{
+    static float threshold = pow(10, 0.3);
+    static float zeroThresh = pow(10, -16);
+
+    size_t count = 0;
+    size_t nonZeroCount = 0;
+
+    for (size_t n = 1; n <= m_blockSize / 2; ++n) {
+        //!!! adjust threshold so that this multiplication is unnecessary
+	float sqrmag = mag[n] * mag[n];
+        bool above = ((sqrmag / m_prevMag[n]) >= threshold);
+        if (above) ++count;
+        if (sqrmag > zeroThresh) ++nonZeroCount;
+	m_prevMag[n] = sqrmag;
+    }
+
+//!!!    return float(count) / float(m_blockSize);
+    if (nonZeroCount == 0) return 0;
+    else return float(count) / float(nonZeroCount);
+}
+
+}
+

src/PercussiveAudioCurve.h

@@ -0,0 +1,41 @@
+/* -*- c-basic-offset: 4 indent-tabs-mode: nil -*-  vi:set ts=8 sts=4 sw=4: */
+
+/*
+    Rubber Band
+    An audio time-stretching and pitch-shifting library.
+    Copyright 2007 Chris Cannam.
+    
+    This program is free software; you can redistribute it and/or
+    modify it under the terms of the GNU General Public License as
+    published by the Free Software Foundation; either version 2 of the
+    License, or (at your option) any later version.  See the file
+    COPYING included with this distribution for more information.
+*/
+
+#ifndef _PERCUSSIVE_AUDIO_CURVE_H_
+#define _PERCUSSIVE_AUDIO_CURVE_H_
+
+#include "AudioCurve.h"
+
+namespace RubberBand
+{
+
+class PercussiveAudioCurve : public AudioCurve
+{
+public:
+    PercussiveAudioCurve(size_t sampleRate, size_t blockSize);
+
+    virtual ~PercussiveAudioCurve();
+
+    virtual void setBlockSize(size_t newSize);
+
+    virtual float process(float *mag, size_t increment);
+    virtual void reset();
+
+protected:
+    double *m_prevMag;
+};
+
+}
+
+#endif

src/Resampler.cpp

@@ -0,0 +1,169 @@
+/* -*- c-basic-offset: 4 indent-tabs-mode: nil -*- vi:set ts=8 sts=4 sw=4: */
+
+/*
+    Rubber Band
+    An audio time-stretching and pitch-shifting library.
+    Copyright 2007 Chris Cannam.
+    
+    This program is free software; you can redistribute it and/or
+    modify it under the terms of the GNU General Public License as
+    published by the Free Software Foundation; either version 2 of the
+    License, or (at your option) any later version.  See the file
+    COPYING included with this distribution for more information.
+*/
+
+#include "Resampler.h"
+
+#include <cstdlib>
+#include <cmath>
+
+#include <iostream>
+
+
+#include <samplerate.h>
+
+namespace RubberBand {
+
+class Resampler::D
+{
+public:
+    D(Quality quality, size_t channels, size_t maxBufferSize);
+    ~D();
+
+    size_t resample(float **in, float **out,
+                    size_t incount, float ratio, bool final);
+
+    void reset();
+
+protected:
+    SRC_STATE *m_src;
+    float *m_iin;
+    float *m_iout;
+    size_t m_channels;
+    size_t m_iinsize;
+    size_t m_ioutsize;
+};
+
+Resampler::D::D(Quality quality, size_t channels, size_t maxBufferSize) :
+    m_src(0),
+    m_iin(0),
+    m_iout(0),
+    m_channels(channels),
+    m_iinsize(0),
+    m_ioutsize(0)
+{
+    std::cerr << "Resampler::Resampler: using libsamplerate implementation"
+              << std::endl;
+
+    int err = 0;
+    m_src = src_new(quality == Best ? SRC_SINC_BEST_QUALITY :
+                    quality == Fastest ? SRC_LINEAR :
+                    SRC_SINC_FASTEST,
+                    channels, &err);
+
+    //!!! check err, throw
+
+    if (maxBufferSize > 0 && m_channels > 1) {
+        //!!! alignment?
+        m_iinsize = maxBufferSize * m_channels;
+        m_ioutsize = maxBufferSize * m_channels * 2;
+        m_iin = (float *)malloc(m_iinsize * sizeof(float));
+        m_iout = (float *)malloc(m_ioutsize * sizeof(float));
+    }
+}
+
+Resampler::D::~D()
+{
+    src_delete(m_src);
+    if (m_iinsize > 0) {
+        free(m_iin);
+    }
+    if (m_ioutsize > 0) {
+        free(m_iout);
+    }
+}
+
+size_t
+Resampler::D::resample(float **in, float **out, size_t incount, float ratio,
+                       bool final)
+{
+    SRC_DATA data;
+
+    size_t outcount = lrintf(ceilf(incount * ratio));
+
+    if (m_channels == 1) {
+        data.data_in = *in;
+        data.data_out = *out;
+    } else {
+        if (incount * m_channels > m_iinsize) {
+            m_iinsize = incount * m_channels;
+            m_iin = (float *)realloc(m_iin, m_iinsize * sizeof(float));
+        }
+        if (outcount * m_channels > m_ioutsize) {
+            m_ioutsize = outcount * m_channels;
+            m_iout = (float *)realloc(m_iout, m_ioutsize * sizeof(float));
+        }
+        for (size_t i = 0; i < incount; ++i) {
+            for (size_t c = 0; c < m_channels; ++c) {
+                m_iin[i * m_channels + c] = in[c][i];
+            }
+        }
+        data.data_in = m_iin;
+        data.data_out = m_iout;
+    }
+
+    data.input_frames = incount;
+    data.output_frames = outcount;
+    data.src_ratio = ratio;
+    data.end_of_input = (final ? 1 : 0);
+
+    int err = src_process(m_src, &data);
+
+    //!!! check err, respond appropriately
+
+    if (m_channels > 1) {
+        for (size_t i = 0; i < data.output_frames_gen; ++i) {
+            for (size_t c = 0; c < m_channels; ++c) {
+                out[c][i] = m_iout[i * m_channels + c];
+            }
+        }
+    }
+
+    return data.output_frames_gen;
+}
+
+void
+Resampler::D::reset()
+{
+    src_reset(m_src);
+}
+
+} // end namespace
+
+
+namespace RubberBand {
+
+Resampler::Resampler(Quality quality, size_t channels, size_t maxBufferSize)
+{
+    m_d = new D(quality, channels, maxBufferSize);
+}
+
+Resampler::~Resampler()
+{
+    delete m_d;
+}
+
+size_t 
+Resampler::resample(float **in, float **out,
+                    size_t incount, float ratio, bool final)
+{
+    return m_d->resample(in, out, incount, ratio, final);
+}
+
+void
+Resampler::reset()
+{
+    m_d->reset();
+}
+
+}

src/Resampler.h

@@ -0,0 +1,48 @@
+/* -*- c-basic-offset: 4 indent-tabs-mode: nil -*- vi:set ts=8 sts=4 sw=4: */
+
+/*
+    Rubber Band
+    An audio time-stretching and pitch-shifting library.
+    Copyright 2007 Chris Cannam.
+    
+    This program is free software; you can redistribute it and/or
+    modify it under the terms of the GNU General Public License as
+    published by the Free Software Foundation; either version 2 of the
+    License, or (at your option) any later version.  See the file
+    COPYING included with this distribution for more information.
+*/
+
+#ifndef _RUBBERBAND_RESAMPLER_H_
+#define _RUBBERBAND_RESAMPLER_H_
+
+#include <sys/types.h>
+
+namespace RubberBand {
+
+class Resampler
+{
+public:
+    enum Quality { Best, FastestTolerable, Fastest };
+
+    /**
+     * Construct a resampler with the given quality level and channel
+     * count.  maxBufferSize gives a bound on the maximum incount size
+     * that may be passed to the resample function before the
+     * resampler needs to reallocate its internal buffers.
+     */
+    Resampler(Quality quality, size_t channels, size_t maxBufferSize = 0);
+    ~Resampler();
+
+    size_t resample(float **in, float **out,
+                    size_t incount, float ratio, bool final = false);
+
+    void reset();
+
+protected:
+    class D;
+    D *m_d;
+};
+
+}
+
+#endif

src/RingBuffer.h

@@ -0,0 +1,628 @@
+/* -*- c-basic-offset: 4 indent-tabs-mode: nil -*-  vi:set ts=8 sts=4 sw=4: */
+
+/*
+    Rubber Band
+    An audio time-stretching and pitch-shifting library.
+    Copyright 2007 Chris Cannam.
+    
+    This program is free software; you can redistribute it and/or
+    modify it under the terms of the GNU General Public License as
+    published by the Free Software Foundation; either version 2 of the
+    License, or (at your option) any later version.  See the file
+    COPYING included with this distribution for more information.
+*/
+
+#ifndef _RUBBERBAND_RINGBUFFER_H_
+#define _RUBBERBAND_RINGBUFFER_H_
+
+#include <sys/types.h>
+#include <sys/mman.h>
+
+#include "Scavenger.h"
+
+//#define DEBUG_RINGBUFFER 1
+
+#define MLOCK(a,b) ::mlock(a,b)
+#define MUNLOCK(a,b) ::munlock(a,b)
+
+#ifdef DEBUG_RINGBUFFER
+#include <iostream>
+#endif
+
+namespace RubberBand {
+
+/**
+ * RingBuffer implements a lock-free ring buffer for one writer and N
+ * readers, that is to be used to store a sample type T.
+ */
+
+template <typename T, int N = 1>
+class RingBuffer
+{
+public:
+    /**
+     * Create a ring buffer with room to write n samples.
+     *
+     * Note that the internal storage size will actually be n+1
+     * samples, as one element is unavailable for administrative
+     * reasons.  Since the ring buffer performs best if its size is a
+     * power of two, this means n should ideally be some power of two
+     * minus one.
+     */
+    RingBuffer(size_t n);
+
+    virtual ~RingBuffer();
+
+    /**
+     * Return the total capacity of the ring buffer in samples.
+     * (This is the argument n passed to the constructor.)
+     */
+    size_t getSize() const;
+
+    /**
+     * Resize the ring buffer.  This also empties it; use resized()
+     * below if you do not want this to happen.  Actually swaps in a
+     * new, larger buffer; the old buffer is scavenged after a seemly
+     * delay.  Should be called from the write thread.
+     */
+    void resize(size_t newSize);
+
+    /**
+     * Return a new ring buffer (allocated with "new" -- called must
+     * delete when no longer needed) of the given size, containing the
+     * same data as this one.  If another thread reads from or writes
+     * to this buffer during the call, the results may be incomplete
+     * or inconsistent.  If this buffer's data will not fit in the new
+     * size, the contents are undefined.
+     */
+    RingBuffer<T, N> *resized(size_t newSize, int R = 0) const;
+
+    /**
+     * Lock the ring buffer into physical memory.  Returns true
+     * for success.
+     */
+    bool mlock();
+
+    /**
+     * Reset read and write pointers, thus emptying the buffer.
+     * Should be called from the write thread.
+     */
+    void reset();
+
+    /**
+     * Return the amount of data available for reading by reader R, in
+     * samples.
+     */
+    size_t getReadSpace(int R = 0) const;
+
+    /**
+     * Return the amount of space available for writing, in samples.
+     */
+    size_t getWriteSpace() const;
+
+    /**
+     * Read n samples from the buffer, for reader R.  If fewer than n
+     * are available, the remainder will be zeroed out.  Returns the
+     * number of samples actually read.
+     */
+    size_t read(T *destination, size_t n, int R = 0);
+
+    /**
+     * Read n samples from the buffer, for reader R, adding them to
+     * the destination.  If fewer than n are available, the remainder
+     * will be left alone.  Returns the number of samples actually
+     * read.
+     */
+    size_t readAdding(T *destination, size_t n, int R = 0);
+
+    /**
+     * Read one sample from the buffer, for reader R.  If no sample is
+     * available, this will silently return zero.  Calling this
+     * repeatedly is obviously slower than calling read once, but it
+     * may be good enough if you don't want to allocate a buffer to
+     * read into.
+     */
+    T readOne(int R = 0);
+
+    /**
+     * Read n samples from the buffer, if available, for reader R,
+     * without advancing the read pointer -- i.e. a subsequent read()
+     * or skip() will be necessary to empty the buffer.  If fewer than
+     * n are available, the remainder will be zeroed out.  Returns the
+     * number of samples actually read.
+     */
+    size_t peek(T *destination, size_t n, int R = 0) const;
+
+    /**
+     * Read one sample from the buffer, if available, without
+     * advancing the read pointer -- i.e. a subsequent read() or
+     * skip() will be necessary to empty the buffer.  Returns zero if
+     * no sample was available.
+     */
+    T peekOne(int R = 0) const;
+
+    /**
+     * Pretend to read n samples from the buffer, for reader R,
+     * without actually returning them (i.e. discard the next n
+     * samples).  Returns the number of samples actually available for
+     * discarding.
+     */
+    size_t skip(size_t n, int R = 0);
+
+    /**
+     * Write n samples to the buffer.  If insufficient space is
+     * available, not all samples may actually be written.  Returns
+     * the number of samples actually written.
+     */
+    size_t write(const T *source, size_t n);
+
+    /**
+     * Write n zero-value samples to the buffer.  If insufficient
+     * space is available, not all zeros may actually be written.
+     * Returns the number of zeroes actually written.
+     */
+    size_t zero(size_t n);
+
+protected:
+    T               *m_buffer;
+    volatile size_t  m_writer;
+    volatile size_t  m_readers[N];
+    size_t           m_size;
+    bool             m_mlocked;
+
+    static Scavenger<ScavengerArrayWrapper<T> > m_scavenger;
+
+private:
+    RingBuffer(const RingBuffer &); // not provided
+    RingBuffer &operator=(const RingBuffer &); // not provided
+};
+
+template <typename T, int N>
+Scavenger<ScavengerArrayWrapper<T> > RingBuffer<T, N>::m_scavenger;
+
+template <typename T, int N>
+RingBuffer<T, N>::RingBuffer(size_t n) :
+    m_buffer(new T[n + 1]),
+    m_writer(0),
+    m_size(n + 1),
+    m_mlocked(false)
+{
+#ifdef DEBUG_RINGBUFFER
+    std::cerr << "RingBuffer<T," << N << ">[" << this << "]::RingBuffer(" << n << ")" << std::endl;
+#endif
+
+    for (int i = 0; i < N; ++i) m_readers[i] = 0;
+
+    m_scavenger.scavenge();
+}
+
+template <typename T, int N>
+RingBuffer<T, N>::~RingBuffer()
+{
+#ifdef DEBUG_RINGBUFFER
+    std::cerr << "RingBuffer<T," << N << ">[" << this << "]::~RingBuffer" << std::endl;
+#endif
+
+    if (m_mlocked) {
+	MUNLOCK((void *)m_buffer, m_size * sizeof(T));
+    }
+    delete[] m_buffer;
+
+    m_scavenger.scavenge();
+}
+
+template <typename T, int N>
+size_t
+RingBuffer<T, N>::getSize() const
+{
+#ifdef DEBUG_RINGBUFFER
+    std::cerr << "RingBuffer<T," << N << ">[" << this << "]::getSize(): " << m_size-1 << std::endl;
+#endif
+
+    return m_size - 1;
+}
+
+template <typename T, int N>
+void
+RingBuffer<T, N>::resize(size_t newSize)
+{
+#ifdef DEBUG_RINGBUFFER
+    std::cerr << "RingBuffer<T," << N << ">[" << this << "]::resize(" << newSize << ")" << std::endl;
+#endif
+
+    m_scavenger.scavenge();
+
+    if (m_mlocked) {
+	MUNLOCK((void *)m_buffer, m_size * sizeof(T));
+    }
+
+    m_scavenger.claim(new ScavengerArrayWrapper<T>(m_buffer));
+
+    reset();
+    m_buffer = new T[newSize + 1];
+    m_size = newSize + 1;
+
+    if (m_mlocked) {
+	if (MLOCK((void *)m_buffer, m_size * sizeof(T))) {
+	    m_mlocked = false;
+	}
+    }
+}
+
+template <typename T, int N>
+RingBuffer<T, N> *
+RingBuffer<T, N>::resized(size_t newSize, int R) const
+{
+    RingBuffer<T, N> *newBuffer = new RingBuffer<T, N>(newSize);
+
+    size_t w = m_writer;
+    size_t r = m_readers[R];
+
+    while (r != w) {
+        T value = m_buffer[r];
+        newBuffer->write(&value, 1);
+        if (++r == m_size) r = 0;
+    }
+
+    return newBuffer;
+}
+
+template <typename T, int N>
+bool
+RingBuffer<T, N>::mlock()
+{
+    if (MLOCK((void *)m_buffer, m_size * sizeof(T))) return false;
+    m_mlocked = true;
+    return true;
+}
+
+template <typename T, int N>
+void
+RingBuffer<T, N>::reset()
+{
+#ifdef DEBUG_RINGBUFFER
+    std::cerr << "RingBuffer<T," << N << ">[" << this << "]::reset" << std::endl;
+#endif
+
+    m_writer = 0;
+    for (int i = 0; i < N; ++i) m_readers[i] = 0;
+}
+
+template <typename T, int N>
+size_t
+RingBuffer<T, N>::getReadSpace(int R) const
+{
+    size_t writer = m_writer;
+    size_t reader = m_readers[R];
+    size_t space;
+
+#ifdef DEBUG_RINGBUFFER
+    std::cerr << "RingBuffer<T," << N << ">[" << this << "]::getReadSpace(" << R << "): reader " << reader << ", writer " << writer << std::endl;
+#endif
+
+    if (writer > reader) space = writer - reader;
+    else if (writer < reader) space = (writer + m_size) - reader;
+    else space = 0;
+
+#ifdef DEBUG_RINGBUFFER
+    std::cerr << "RingBuffer<T," << N << ">[" << this << "]::getReadSpace(" << R << "): " << space << std::endl;
+#endif
+
+    return space;
+}
+
+template <typename T, int N>
+size_t
+RingBuffer<T, N>::getWriteSpace() const
+{
+    size_t space = 0;
+    for (int i = 0; i < N; ++i) {
+        size_t writer = m_writer;
+        size_t reader = m_readers[i];
+	size_t here = (reader + m_size - writer - 1);
+        if (here >= m_size) here -= m_size;
+	if (i == 0 || here < space) space = here;
+    }
+
+#ifdef DEBUG_RINGBUFFER
+    size_t rs(getReadSpace()), rp(m_readers[0]);
+
+    std::cerr << "RingBuffer: write space " << space << ", read space "
+	      << rs << ", total " << (space + rs) << ", m_size " << m_size << std::endl;
+    std::cerr << "RingBuffer: reader " << rp << ", writer " << m_writer << std::endl;
+#endif
+
+#ifdef DEBUG_RINGBUFFER
+    std::cerr << "RingBuffer<T," << N << ">[" << this << "]::getWriteSpace(): " << space << std::endl;
+#endif
+
+    return space;
+}
+
+template <typename T, int N>
+size_t
+RingBuffer<T, N>::read(T *destination, size_t n, int R)
+{
+#ifdef DEBUG_RINGBUFFER
+    std::cerr << "RingBuffer<T," << N << ">[" << this << "]::read(dest, " << n << ", " << R << ")" << std::endl;
+#endif
+
+    size_t available = getReadSpace(R);
+    if (n > available) {
+#ifdef DEBUG_RINGBUFFER
+	std::cerr << "WARNING: Only " << available << " samples available"
+		  << std::endl;
+#endif
+        for (size_t i = available; i < n; ++i) {
+            destination[i] = 0;
+        }
+	n = available;
+    }
+    if (n == 0) return n;
+
+    size_t reader = m_readers[R];
+    size_t here = m_size - reader;
+
+    if (here >= n) {
+        for (size_t i = 0; i < n; ++i) {
+            destination[i] = (m_buffer + reader)[i];
+        }
+    } else {
+        for (size_t i = 0; i < here; ++i) {
+            destination[i] = (m_buffer + reader)[i];
+        }
+        for (size_t i = 0; i < (n - here); ++i) {
+            destination[i + here] = m_buffer[i];
+        }
+    }
+
+    reader += n;
+    while (reader >= m_size) reader -= m_size;
+    m_readers[R] = reader;
+
+#ifdef DEBUG_RINGBUFFER
+    std::cerr << "RingBuffer<T," << N << ">[" << this << "]::read: read " << n << ", reader now " << m_readers[R] << std::endl;
+#endif
+
+    return n;
+}
+
+template <typename T, int N>
+size_t
+RingBuffer<T, N>::readAdding(T *destination, size_t n, int R)
+{
+#ifdef DEBUG_RINGBUFFER
+    std::cerr << "RingBuffer<T," << N << ">[" << this << "]::readAdding(dest, " << n << ", " << R << ")" << std::endl;
+#endif
+
+    size_t available = getReadSpace(R);
+    if (n > available) {
+#ifdef DEBUG_RINGBUFFER
+	std::cerr << "WARNING: Only " << available << " samples available"
+		  << std::endl;
+#endif
+	n = available;
+    }
+    if (n == 0) return n;
+
+    size_t reader = m_readers[R];
+    size_t here = m_size - reader;
+
+    if (here >= n) {
+	for (size_t i = 0; i < n; ++i) {
+	    destination[i] += (m_buffer + reader)[i];
+	}
+    } else {
+	for (size_t i = 0; i < here; ++i) {
+	    destination[i] += (m_buffer + reader)[i];
+	}
+	for (size_t i = 0; i < (n - here); ++i) {
+	    destination[i + here] += m_buffer[i];
+	}
+    }
+
+    reader += n;
+    while (reader >= m_size) reader -= m_size;
+    m_readers[R] = reader;
+    return n;
+}
+
+template <typename T, int N>
+T
+RingBuffer<T, N>::readOne(int R)
+{
+#ifdef DEBUG_RINGBUFFER
+    std::cerr << "RingBuffer<T," << N << ">[" << this << "]::readOne(" << R << ")" << std::endl;
+#endif
+
+    if (m_writer == m_readers[R]) {
+#ifdef DEBUG_RINGBUFFER
+	std::cerr << "WARNING: No sample available"
+		  << std::endl;
+#endif
+	return 0;
+    }
+    size_t reader = m_readers[R];
+    T value = m_buffer[reader];
+    if (++reader == m_size) reader = 0;
+    m_readers[R] = reader;
+    return value;
+}
+
+template <typename T, int N>
+size_t
+RingBuffer<T, N>::peek(T *destination, size_t n, int R) const
+{
+#ifdef DEBUG_RINGBUFFER
+    std::cerr << "RingBuffer<T," << N << ">[" << this << "]::peek(dest, " << n << ", " << R << ")" << std::endl;
+#endif
+
+    size_t available = getReadSpace(R);
+    if (n > available) {
+#ifdef DEBUG_RINGBUFFER
+	std::cerr << "WARNING: Only " << available << " samples available"
+		  << std::endl;
+#endif
+	memset(destination + available, 0, (n - available) * sizeof(T));
+	n = available;
+    }
+    if (n == 0) return n;
+
+    size_t reader = m_readers[R];
+    size_t here = m_size - reader;
+
+    if (here >= n) {
+        for (size_t i = 0; i < n; ++i) {
+            destination[i] = (m_buffer + reader)[i];
+        }
+    } else {
+        for (size_t i = 0; i < here; ++i) {
+            destination[i] = (m_buffer + reader)[i];
+        }
+        for (size_t i = 0; i < (n - here); ++i) {
+            destination[i + here] = m_buffer[i];
+        }
+    }
+
+#ifdef DEBUG_RINGBUFFER
+    std::cerr << "RingBuffer<T," << N << ">[" << this << "]::peek: read " << n << std::endl;
+#endif
+
+    return n;
+}
+
+template <typename T, int N>
+T
+RingBuffer<T, N>::peekOne(int R) const
+{
+#ifdef DEBUG_RINGBUFFER
+    std::cerr << "RingBuffer<T," << N << ">[" << this << "]::peek(" << R << ")" << std::endl;
+#endif
+
+    if (m_writer == m_readers[R]) {
+#ifdef DEBUG_RINGBUFFER
+	std::cerr << "WARNING: No sample available"
+		  << std::endl;
+#endif
+	return 0;
+    }
+    T value = m_buffer[m_readers[R]];
+    return value;
+}
+
+template <typename T, int N>
+size_t
+RingBuffer<T, N>::skip(size_t n, int R)
+{
+#ifdef DEBUG_RINGBUFFER
+    std::cerr << "RingBuffer<T," << N << ">[" << this << "]::skip(" << n << ", " << R << ")" << std::endl;
+#endif
+
+    size_t available = getReadSpace(R);
+    if (n > available) {
+#ifdef DEBUG_RINGBUFFER
+	std::cerr << "WARNING: Only " << available << " samples available"
+		  << std::endl;
+#endif
+	n = available;
+    }
+    if (n == 0) return n;
+
+    size_t reader = m_readers[R];
+    reader += n;
+    while (reader >= m_size) reader -= m_size;
+    m_readers[R] = reader;
+    return n;
+}
+
+template <typename T, int N>
+size_t
+RingBuffer<T, N>::write(const T *source, size_t n)
+{
+#ifdef DEBUG_RINGBUFFER
+    std::cerr << "RingBuffer<T," << N << ">[" << this << "]::write(" << n << ")" << std::endl;
+#endif
+
+    size_t available = getWriteSpace();
+    if (n > available) {
+#ifdef DEBUG_RINGBUFFER
+	std::cerr << "WARNING: Only room for " << available << " samples"
+		  << std::endl;
+#endif
+	n = available;
+    }
+    if (n == 0) return n;
+
+    size_t writer = m_writer;
+    size_t here = m_size - writer;
+    if (here >= n) {
+        for (size_t i = 0; i < n; ++i) {
+            (m_buffer + writer)[i] = source[i];
+        }
+    } else {
+        for (size_t i = 0; i < here; ++i) {
+            (m_buffer + writer)[i] = source[i];
+        }
+        for (size_t i = 0; i < (n - here); ++i) {
+            m_buffer[i] = (source + here)[i];
+        }
+    }
+
+    writer += n;
+    while (writer >= m_size) writer -= m_size;
+    m_writer = writer;
+
+#ifdef DEBUG_RINGBUFFER
+    std::cerr << "RingBuffer<T," << N << ">[" << this << "]::write: wrote " << n << ", writer now " << m_writer << std::endl;
+#endif
+
+    return n;
+}
+
+template <typename T, int N>
+size_t
+RingBuffer<T, N>::zero(size_t n)
+{
+#ifdef DEBUG_RINGBUFFER
+    std::cerr << "RingBuffer<T," << N << ">[" << this << "]::zero(" << n << ")" << std::endl;
+#endif
+
+    size_t available = getWriteSpace();
+    if (n > available) {
+#ifdef DEBUG_RINGBUFFER
+	std::cerr << "WARNING: Only room for " << available << " samples"
+		  << std::endl;
+#endif
+	n = available;
+    }
+    if (n == 0) return n;
+
+    size_t writer = m_writer;
+    size_t here = m_size - writer;
+    if (here >= n) {
+        for (size_t i = 0; i < n; ++i) {
+            (m_buffer + writer)[i] = 0;
+        }
+    } else {
+        for (size_t i = 0; i < here; ++i) {
+            (m_buffer + writer)[i] = 0;
+        }
+        for (size_t i = 0; i < (n - here); ++i) {
+            m_buffer[i] = 0;
+        }
+    }
+
+    writer += n;
+    while (writer >= m_size) writer -= m_size;
+    m_writer = writer;
+
+#ifdef DEBUG_RINGBUFFER
+    std::cerr << "writer -> " << m_writer << std::endl;
+#endif
+
+    return n;
+}
+
+}
+
+#endif // _RINGBUFFER_H_

src/RubberBandStretcher.cpp

@@ -0,0 +1,178 @@
+/* -*- c-basic-offset: 4 indent-tabs-mode: nil -*-  vi:set ts=8 sts=4 sw=4: */
+
+/*
+    Rubber Band
+    An audio time-stretching and pitch-shifting library.
+    Copyright 2007 Chris Cannam.
+    
+    This program is free software; you can redistribute it and/or
+    modify it under the terms of the GNU General Public License as
+    published by the Free Software Foundation; either version 2 of the
+    License, or (at your option) any later version.  See the file
+    COPYING included with this distribution for more information.
+*/
+
+#include "StretcherImpl.h"
+
+namespace RubberBand {
+
+
+RubberBandStretcher::RubberBandStretcher(size_t sampleRate,
+                                         size_t channels,
+                                         Options options,
+                                         double initialTimeRatio,
+                                         double initialPitchScale) :
+    TimeStretcher(sampleRate, channels),
+    m_d(new Impl(this, sampleRate, channels, options,
+                 initialTimeRatio, initialPitchScale))
+{
+}
+
+RubberBandStretcher::~RubberBandStretcher()
+{
+    delete m_d;
+}
+
+void
+RubberBandStretcher::reset()
+{
+    m_d->reset();
+}
+
+void
+RubberBandStretcher::setTimeRatio(double ratio)
+{
+    m_d->setTimeRatio(ratio);
+}
+
+void
+RubberBandStretcher::setPitchScale(double scale)
+{
+    m_d->setPitchScale(scale);
+}
+
+double
+RubberBandStretcher::getTimeRatio() const
+{
+    return m_d->getTimeRatio();
+}
+
+double
+RubberBandStretcher::getPitchScale() const
+{
+    return m_d->getPitchScale();
+}
+
+size_t
+RubberBandStretcher::getLatency() const
+{
+    return m_d->getLatency();
+}
+
+void
+RubberBandStretcher::setTransientsOption(Options options) 
+{
+    m_d->setTransientsOption(options);
+}
+
+void
+RubberBandStretcher::setPhaseOption(Options options) 
+{
+    m_d->setPhaseOption(options);
+}
+
+void
+RubberBandStretcher::setExpectedInputDuration(size_t samples) 
+{
+    m_d->setExpectedInputDuration(samples);
+}
+
+void
+RubberBandStretcher::setMaxProcessBlockSize(size_t samples)
+{
+    m_d->setMaxProcessBlockSize(samples);
+}
+
+size_t
+RubberBandStretcher::getSamplesRequired() const
+{
+    return m_d->getSamplesRequired();
+}
+
+void
+RubberBandStretcher::study(const float *const *input, size_t samples,
+                           bool final)
+{
+    m_d->study(input, samples, final);
+}
+
+void
+RubberBandStretcher::process(const float *const *input, size_t samples,
+                             bool final)
+{
+    m_d->process(input, samples, final);
+}
+
+int
+RubberBandStretcher::available() const
+{
+    return m_d->available();
+}
+
+size_t
+RubberBandStretcher::retrieve(float *const *output, size_t samples) const
+{
+    return m_d->retrieve(output, samples);
+}
+
+float
+RubberBandStretcher::getFrequencyCutoff(int n) const
+{
+    return m_d->getFrequencyCutoff(n);
+}
+
+void
+RubberBandStretcher::setFrequencyCutoff(int n, float f) 
+{
+    m_d->setFrequencyCutoff(n, f);
+}
+
+size_t
+RubberBandStretcher::getInputIncrement() const
+{
+    return m_d->getInputIncrement();
+}
+
+std::vector<int>
+RubberBandStretcher::getOutputIncrements() const
+{
+    return m_d->getOutputIncrements();
+}
+
+std::vector<float>
+RubberBandStretcher::getLockCurve() const
+{
+    return m_d->getLockCurve();
+}
+
+size_t
+RubberBandStretcher::getChannelCount() const
+{
+    return m_d->getChannelCount();
+}
+
+void
+RubberBandStretcher::calculateStretch()
+{
+    m_d->calculateStretch();
+}
+
+void
+RubberBandStretcher::setDebugLevel(int level)
+{
+    m_d->setDebugLevel(level);
+}
+
+
+}
+

src/Scavenger.h

@@ -0,0 +1,198 @@
+/* -*- c-basic-offset: 4 indent-tabs-mode: nil -*-  vi:set ts=8 sts=4 sw=4: */
+
+/*
+    Rubber Band
+    An audio time-stretching and pitch-shifting library.
+    Copyright 2007 Chris Cannam.
+    
+    This program is free software; you can redistribute it and/or
+    modify it under the terms of the GNU General Public License as
+    published by the Free Software Foundation; either version 2 of the
+    License, or (at your option) any later version.  See the file
+    COPYING included with this distribution for more information.
+*/
+
+#ifndef _RUBBERBAND_SCAVENGER_H_
+#define _RUBBERBAND_SCAVENGER_H_
+
+#include <vector>
+#include <list>
+#include <sys/time.h>
+#include <pthread.h>
+#include <iostream>
+
+namespace RubberBand {
+
+/**
+ * A very simple class that facilitates running things like plugins
+ * without locking, by collecting unwanted objects and deleting them
+ * after a delay so as to be sure nobody's in the middle of using
+ * them.  Requires scavenge() to be called regularly from a non-RT
+ * thread.
+ *
+ * This is currently not at all suitable for large numbers of objects
+ * -- it's just a quick hack for use with things like plugins.
+ */
+
+template <typename T>
+class Scavenger
+{
+public:
+    Scavenger(int sec = 2, int defaultObjectListSize = 200);
+    ~Scavenger();
+
+    /**
+     * Call from an RT thread etc., to pass ownership of t to us.
+     * Only one thread should be calling this on any given scavenger.
+     */
+    void claim(T *t);
+
+    /**
+     * Call from a non-RT thread.
+     * Only one thread should be calling this on any given scavenger.
+     */
+    void scavenge(bool clearNow = false);
+
+protected:
+    typedef std::pair<T *, int> ObjectTimePair;
+    typedef std::vector<ObjectTimePair> ObjectTimeList;
+    ObjectTimeList m_objects;
+    int m_sec;
+
+    typedef std::list<T *> ObjectList;
+    ObjectList m_excess;
+    int m_lastExcess;
+    pthread_mutex_t m_excessMutex;
+    void pushExcess(T *);
+    void clearExcess(int);
+
+    unsigned int m_claimed;
+    unsigned int m_scavenged;
+};
+
+/**
+ * A wrapper to permit arrays to be scavenged.
+ */
+
+template <typename T>
+class ScavengerArrayWrapper
+{
+public:
+    ScavengerArrayWrapper(T *array) : m_array(array) { }
+    ~ScavengerArrayWrapper() { delete[] m_array; }
+
+private:
+    T *m_array;
+};
+
+
+template <typename T>
+Scavenger<T>::Scavenger(int sec, int defaultObjectListSize) :
+    m_objects(ObjectTimeList(defaultObjectListSize)),
+    m_sec(sec),
+    m_claimed(0),
+    m_scavenged(0)
+{
+    pthread_mutex_init(&m_excessMutex, 0);
+}
+
+template <typename T>
+Scavenger<T>::~Scavenger()
+{
+    if (m_scavenged < m_claimed) {
+	for (size_t i = 0; i < m_objects.size(); ++i) {
+	    ObjectTimePair &pair = m_objects[i];
+	    if (pair.first != 0) {
+		T *ot = pair.first;
+		pair.first = 0;
+		delete ot;
+		++m_scavenged;
+	    }
+	}
+    }
+
+    clearExcess(0);
+}
+
+template <typename T>
+void
+Scavenger<T>::claim(T *t)
+{
+//    std::cerr << "Scavenger::claim(" << t << ")" << std::endl;
+
+    struct timeval tv;
+    (void)gettimeofday(&tv, 0);
+    int sec = tv.tv_sec;
+
+    for (size_t i = 0; i < m_objects.size(); ++i) {
+	ObjectTimePair &pair = m_objects[i];
+	if (pair.first == 0) {
+	    pair.second = sec;
+	    pair.first = t;
+	    ++m_claimed;
+	    return;
+	}
+    }
+
+    std::cerr << "WARNING: Scavenger::claim(" << t << "): run out of slots, "
+	      << "using non-RT-safe method" << std::endl;
+    pushExcess(t);
+}
+
+template <typename T>
+void
+Scavenger<T>::scavenge(bool clearNow)
+{
+//    std::cerr << "Scavenger::scavenge: scavenged " << m_scavenged << ", claimed " << m_claimed << std::endl;
+
+    if (m_scavenged >= m_claimed) return;
+    
+    struct timeval tv;
+    (void)gettimeofday(&tv, 0);
+    int sec = tv.tv_sec;
+
+    for (size_t i = 0; i < m_objects.size(); ++i) {
+	ObjectTimePair &pair = m_objects[i];
+	if (clearNow ||
+	    (pair.first != 0 && pair.second + m_sec < sec)) {
+	    T *ot = pair.first;
+	    pair.first = 0;
+	    delete ot;
+	    ++m_scavenged;
+	}
+    }
+
+    if (sec > m_lastExcess + m_sec) {
+        clearExcess(sec);
+    }
+}
+
+template <typename T>
+void
+Scavenger<T>::pushExcess(T *t)
+{
+    pthread_mutex_lock(&m_excessMutex);
+    m_excess.push_back(t);
+    struct timeval tv;
+    (void)gettimeofday(&tv, 0);
+    m_lastExcess = tv.tv_sec;
+    pthread_mutex_unlock(&m_excessMutex);
+}
+
+template <typename T>
+void
+Scavenger<T>::clearExcess(int sec)
+{
+    pthread_mutex_lock(&m_excessMutex);
+    for (typename ObjectList::iterator i = m_excess.begin();
+	 i != m_excess.end(); ++i) {
+	delete *i;
+    }
+    m_excess.clear();
+    m_lastExcess = sec;
+    pthread_mutex_unlock(&m_excessMutex);
+}
+
+}
+
+#endif

src/StretchCalculator.cpp

@@ -0,0 +1,750 @@
+/* -*- c-basic-offset: 4 indent-tabs-mode: nil -*-  vi:set ts=8 sts=4 sw=4: */
+
+/*
+    Rubber Band
+    An audio time-stretching and pitch-shifting library.
+    Copyright 2007 Chris Cannam.
+    
+    This program is free software; you can redistribute it and/or
+    modify it under the terms of the GNU General Public License as
+    published by the Free Software Foundation; either version 2 of the
+    License, or (at your option) any later version.  See the file
+    COPYING included with this distribution for more information.
+*/
+
+#include "StretchCalculator.h"
+
+#include <math.h>
+#include <iostream>
+#include <deque>
+#include <set>
+#include <cassert>
+
+namespace RubberBand
+{
+	
+StretchCalculator::StretchCalculator(size_t sampleRate,
+                                     size_t inputIncrement,
+                                     bool useHardPeaks) :
+    m_sampleRate(sampleRate),
+    m_increment(inputIncrement),
+    m_prevDf(0),
+    m_divergence(0),
+    m_recovery(0),
+    m_prevRatio(1.0),
+    m_wasTransient(false),
+    m_useHardPeaks(useHardPeaks)
+{
+    std::cerr << "StretchCalculator::StretchCalculator: useHardPeaks = " << useHardPeaks << std::endl;
+}    
+
+StretchCalculator::~StretchCalculator()
+{
+}
+
+std::vector<int>
+StretchCalculator::calculate(double ratio, size_t inputDuration,
+                             const std::vector<float> &lockDf,
+                             const std::vector<float> &stretchDf)
+{
+    // Method:
+
+    //!!! This description is out of date.
+
+    //!!! Rationalise naming -- generally wise to avoid the word
+    //"frame" and instead use "block" / "sample" for processing frame /
+    // audio frame.
+
+    // 1. Pre-process the df array, and for each (say) one second's
+    // worth of values, calculate the number of peaks that would
+    // qualify for phase locking given the default threshold.  Then
+    // reduce or increase the threshold by stages until that number is
+    // in a sensible range (say 1-10 peaks per second -- the low end
+    // is harder to estimate than the high end, so it may be better to
+    // start with a high sensitivity and reduce it).
+
+    // 2. Record the positions of peaks, and separately the positions
+    // of those peaks that qualify for locking using the sliding
+    // threshold window.  Don't permit two locked peaks within a very
+    // short time frame (e.g. 30-50ms).
+
+    // 3. Map each of the locked peaks (or any peaks that are over a
+    // given intensity?), as well as the start and end points, to a
+    // proportionate position in the newly stretched array so as to
+    // ensure that their timing is strictly "correct".
+
+    // 4. Calculate how much time is left in the stretch total, after
+    // each of the locked frames has been allocated its static
+    // allowance.  Also count the non-locked frames.
+
+    // 5. For each region between two locked frames, calculate the
+    // number of samples to allocate that region given the time
+    // available for stretch and the number of non-locked frames.
+    // Then divvy them up... how exactly?
+
+
+    assert(lockDf.size() == stretchDf.size());
+    
+    std::vector<Peak> peaks = findPeaks(lockDf);
+    size_t totalCount = lockDf.size();
+
+    std::vector<int> increments;
+
+    size_t outputDuration = lrint(inputDuration * ratio);
+
+    std::cerr << "debug level: " << m_debugLevel << std::endl;
+
+    if (m_debugLevel > 0) {
+        std::cerr << "StretchCalculator::calculate(): inputDuration " << inputDuration << ", ratio " << ratio << ", outputDuration " << outputDuration;
+    }
+
+    //!!! round down?
+    outputDuration = lrint((lockDf.size() * m_increment) * ratio);
+
+    if (m_debugLevel > 0) {
+        std::cerr << " (rounded up to " << outputDuration << ")";
+        std::cerr << ", df size " << lockDf.size() << std::endl;
+    }
+
+//    size_t stretchable = outputDuration - lockCount * m_increment;
+    
+    std::vector<size_t> fixedAudioFrames;
+    for (size_t i = 0; i < peaks.size(); ++i) {
+        fixedAudioFrames.push_back
+            //!!! this should be rounding down, shouldn't it? not lrint?
+            (lrint((double(peaks[i].frame) * outputDuration) / totalCount));
+    }
+
+//    size_t lockIndex = 0;
+
+    if (m_debugLevel > 1) {
+        std::cerr << "have " << peaks.size() << " fixed positions" << std::endl;
+    }
+
+    size_t totalInput = 0, totalOutput = 0;
+
+    // so for each inter-lock region, we want to take the number of
+    // output frames to be allocated and the detection function values
+    // within the range, and produce a series of increments that sum
+    // to the number of output frames, such that each increment is
+    // displaced from the input increment by an amount inversely
+    // proportional to the magnitude of the detection function at that
+    // input step.  Ideally the detection function would have been
+    // somewhat smoothed for this purpose but we'll start raw.
+
+    //!!! Actually, we would possibly be better off using a fixed
+    // smooth curve than the detection function itself.
+
+    size_t regionTotalFrames = 0;
+
+    for (size_t i = 0; i <= peaks.size(); ++i) {
+        
+        size_t regionStart, regionStartBlock, regionEnd, regionEndBlock;
+        bool phaseLock = false;
+
+        if (i == 0) {
+            regionStartBlock = 0;
+            regionStart = 0;
+        } else {
+            regionStartBlock = peaks[i-1].frame;
+            regionStart = fixedAudioFrames[i-1];
+            phaseLock = peaks[i-1].hard;
+        }
+
+        if (i == peaks.size()) {
+            regionEndBlock = totalCount;
+            regionEnd = outputDuration;
+        } else {
+            regionEndBlock = peaks[i].frame;
+            regionEnd = fixedAudioFrames[i];
+        }
+        
+        size_t regionDuration = regionEnd - regionStart;
+        regionTotalFrames += regionDuration;
+
+        std::vector<float> dfRegion;
+
+        for (size_t j = regionStartBlock; j != regionEndBlock; ++j) {
+            dfRegion.push_back(stretchDf[j]);
+        }
+
+        if (m_debugLevel > 1) {
+            std::cerr << "distributeRegion from " << regionStartBlock << " to " << regionEndBlock << " (frames " << regionStart << " to " << regionEnd << ")" << std::endl;
+        }
+
+        dfRegion = smoothDF(dfRegion);
+        
+        std::vector<int> regionIncrements = distributeRegion
+            (dfRegion, regionDuration, ratio, phaseLock);
+
+        size_t totalForRegion = 0;
+
+        for (size_t j = 0; j < regionIncrements.size(); ++j) {
+
+            int incr = regionIncrements[j];
+
+            if (j == 0 && phaseLock) increments.push_back(-incr);
+            else increments.push_back(incr);
+
+            if (incr > 0) totalForRegion += incr;
+            else totalForRegion += -incr;
+
+            totalInput += m_increment;
+        }
+
+        if (totalForRegion != regionDuration) {
+            std::cerr << "*** WARNING: distributeRegion returned wrong duration " << totalForRegion << ", expected " << regionDuration << std::endl;
+        }
+
+        totalOutput += totalForRegion;
+    }
+
+    if (m_debugLevel > 0) {
+        std::cerr << "total input increment = " << totalInput << " (= " << totalInput / m_increment << " blocks), output = " << totalOutput << ", ratio = " << double(totalOutput)/double(totalInput) << ", ideal output " << ceil(totalInput * ratio) << std::endl;
+        std::cerr << "(region total = " << regionTotalFrames << ")" << std::endl;
+    }
+    return increments;
+}
+
+int
+StretchCalculator::calculateSingle(double ratio,
+                                   size_t inputDurationSoFar,
+                                   float df)
+{
+    bool isTransient = false;
+
+    //!!! We want to ensure, as close as possible, that the lock
+    // points appear at _exactly_ the right frame numbers
+
+    //!!! depends on block size.  larger block sizes need higher
+    //thresholds.  since block size depends on ratio, I suppose we
+    //could in theory calculate the threshold from the ratio directly.
+    //For now we just frig it to work OK for a couple of common cases
+    float transientThreshold = 0.35;
+    if (ratio > 1) transientThreshold = 0.25;
+
+    if (m_useHardPeaks && df > m_prevDf * 1.1 && df > transientThreshold) {
+        isTransient = true;
+    }
+
+    if (m_debugLevel > 2) {
+        std::cerr << "df = " << df << ", prevDf = " << m_prevDf
+                  << ", thresh = " << transientThreshold << std::endl;
+    }
+
+    m_prevDf = df;
+
+    if (isTransient && !m_wasTransient) {
+        if (m_debugLevel > 1) {
+            std::cerr << "StretchCalculator::calculateSingle: transient found at "
+                      << inputDurationSoFar << std::endl;
+        }
+        m_divergence += m_increment - (m_increment * ratio);
+        m_wasTransient = true;
+        m_recovery = m_divergence / ((m_sampleRate / 10.0) / m_increment);
+        return -m_increment;
+    }
+
+    if (m_prevRatio != ratio) {
+        m_recovery = m_divergence / ((m_sampleRate / 10.0) / m_increment);
+        m_prevRatio = ratio;
+    }
+
+    //!!! want transient amnesty as above (hard peak amnesty)
+    m_wasTransient = false;
+
+    int incr = lrint(m_increment * ratio - m_recovery);
+    if (m_debugLevel > 2 || (m_debugLevel > 1 && m_divergence != 0)) {
+        std::cerr << "divergence = " << m_divergence << ", recovery = " << m_recovery << ", incr = " << incr << ", ";
+    }
+    if (incr < (m_increment * ratio) / 2) {
+        incr = (m_increment * ratio) / 2;
+    } else if (incr > m_increment * ratio * 2) {
+        incr = m_increment * ratio * 2;
+    }
+
+    double divdiff = (m_increment * ratio) - incr;
+
+    if (m_debugLevel > 2 || (m_debugLevel > 1 && m_divergence != 0)) {
+        std::cerr << "divdiff = " << divdiff << std::endl;
+    }
+
+    double prevDivergence = m_divergence;
+    m_divergence -= divdiff;
+    if ((prevDivergence < 0 && m_divergence > 0) ||
+        (prevDivergence > 0 && m_divergence < 0)) {
+        m_recovery = m_divergence / ((m_sampleRate / 10.0) / m_increment);
+    }
+
+    return incr;
+}
+
+void
+StretchCalculator::reset()
+{
+    m_prevDf = 0;
+    m_divergence = 0;
+}
+
+std::vector<StretchCalculator::Peak>
+StretchCalculator::findPeaks(const std::vector<float> &rawDf)
+{
+    std::vector<float> df = smoothDF(rawDf);
+
+    std::set<size_t> hardPeakCandidates;
+    std::set<size_t> softPeakCandidates;
+
+    if (m_useHardPeaks) {
+
+    //!!! this should depend on duration based on output increment surely?
+        size_t hardPeakAmnesty = lrint(ceil(double(m_sampleRate) /
+                                            (20 * double(m_increment)))); // 0.05 sec ish
+//    size_t hardPeakAmnesty = 5;
+
+        size_t prevHardPeak = 0;
+        std::cerr << "hardPeakAmnesty = " << hardPeakAmnesty << std::endl;
+        for (size_t i = 1; i + 1 < df.size(); ++i) {
+
+            //!!! this ratio configurable? dependent on block size and sr?
+
+            if (df[i] < 0.1) continue;
+            if (df[i] <= df[i-1] * 1.2) continue;
+
+            if (df[i] > df[i-1] * 1.4 ||
+                (df[i+1] > df[i] && df[i+1] > df[i-1] * 1.8) ||
+                df[i] > 0.4) {
+                if (!hardPeakCandidates.empty() &&
+                    i < prevHardPeak + hardPeakAmnesty) {
+                    continue;
+                }
+                size_t peakLocation = i;
+                if (i + 1 < rawDf.size() &&
+                    rawDf[i + 1] > rawDf[i] * 1.4) {
+                    ++peakLocation;
+                }
+                if (m_debugLevel > 1) {
+                    std::cerr << "hard peak at " << peakLocation << " (" << df[peakLocation] << " > " << df[peakLocation-1] << " * " << 1.4 << ")" << std::endl;
+                }
+                hardPeakCandidates.insert(peakLocation);
+                prevHardPeak = peakLocation;
+            }
+        }
+    }
+
+    //!!! we don't yet do the right thing with soft peaks.  if
+    //!useHardPeaks, we should be locking on soft peaks; if
+    //useHardPeaks, we should be ignoring soft peaks if they occur
+    //shortly after hard ones, otherwise either locking on the, or at
+    //least making sure they fall at the correct sample time
+
+//    int mediansize = lrint(ceil(double(m_sampleRate) /
+//                                (4 * double(m_increment)))); // 0.25 sec ish
+    size_t medianmaxsize = lrint(ceil(double(m_sampleRate) /
+                                 double(m_increment))); // 1 sec ish
+//    int mediansize = lrint(ceil(double(m_sampleRate) /
+//                                (2 * double(m_increment)))); // 0.5 sec ish
+
+    if (m_debugLevel > 1) {
+        std::cerr << "mediansize = " << medianmaxsize << std::endl;
+    }
+    if (medianmaxsize < 7) {
+        medianmaxsize = 7;
+        if (m_debugLevel > 1) {
+            std::cerr << "adjusted mediansize = " << medianmaxsize << std::endl;
+        }
+    }
+
+    int minspacing = lrint(ceil(double(m_sampleRate) /
+                                (20 * double(m_increment)))); // 0.05 sec ish
+    
+    std::deque<float> medianwin;
+    std::vector<float> sorted;
+    int softPeakAmnesty = 0;
+
+    for (size_t i = 0; i < medianmaxsize/2; ++i) {
+        medianwin.push_back(0);
+    }
+    for (size_t i = 0; i < medianmaxsize/2 && i < df.size(); ++i) {
+        medianwin.push_back(df[i]);
+    }
+
+    size_t lastSoftPeak = 0;
+
+    for (size_t i = 0; i < df.size(); ++i) {
+        
+        size_t mediansize = medianmaxsize;
+
+        if (medianwin.size() < mediansize) {
+            mediansize = medianwin.size();
+        }
+
+        if (mediansize < 2) {
+            if (i > medianmaxsize) {
+                medianwin.pop_front();
+            }
+            if (i < df.size()) medianwin.push_back(df[i]);
+            continue;
+        }
+
+        if (m_debugLevel > 2) {
+//            std::cerr << "have " << mediansize << " in median buffer" << std::endl;
+        }
+
+        sorted.clear();
+        for (size_t j = 0; j < mediansize; ++j) {
+            sorted.push_back(medianwin[j]);
+        }
+        std::sort(sorted.begin(), sorted.end());
+
+        size_t n = 90; // percentile above which we pick peaks
+        size_t index = (sorted.size() * n) / 100;
+        if (index >= sorted.size()) index = sorted.size()-1;
+        if (index == sorted.size()-1 && index > 0) --index;
+        float thresh = sorted[index];
+
+        size_t middle = medianmaxsize / 2;
+        if (middle >= mediansize) middle = mediansize-1;
+
+        if (m_debugLevel > 2) {
+            std::cerr << "medianwin[" << middle << "] = " << medianwin[middle] << ", thresh = " << thresh << std::endl;
+        }
+
+        if (medianwin[middle] > thresh &&
+            medianwin[middle] > medianwin[middle-1] &&
+            medianwin[middle] > medianwin[middle+1] &&
+            softPeakAmnesty == 0) {
+
+            size_t maxindex = middle;
+            float maxval = medianwin[middle];
+
+            for (size_t j = middle+1; j < mediansize; ++j) {
+                if (medianwin[j] > maxval) {
+                    maxval = medianwin[j];
+                    maxindex = j;
+                } else if (medianwin[j] < medianwin[middle]) {
+                    break;
+                }
+            }
+
+            //!!! we should distinguish between soft peaks (any found
+            //using the above method) and hard peaks, which also show
+            //a very rapid rise in detection function prior to the
+            //peak (the first value after the rise is not necessarily
+            //the peak itself, but it is probably where we should
+            //locate the lock).  For hard peaks we need to lock in
+            //time to preserve the shape of the transient (unless some
+            //option is set to soft mode), for soft peaks we just want
+            //to avoid phase drift so we build up to the lock at the
+            //exact peak moment.
+            
+//            size_t peak = i + maxindex - mediansize;
+            size_t peak = i + maxindex - middle;
+
+//            if (peak > 0) --peak; //!!! that's a fudge
+
+            if (softPeakCandidates.empty() || lastSoftPeak != peak) {
+
+                if (m_debugLevel > 1) {
+                    std::cerr << "soft peak at " << peak << " (" << peak * m_increment << "): "
+                              << medianwin[middle] << " > " << thresh << " and "
+                              << medianwin[middle] << " > " << medianwin[middle-1] << " and "
+                              << medianwin[middle] << " > " << medianwin[middle+1]
+                              << std::endl;
+                }
+
+                if (peak >= df.size()) {
+                    if (m_debugLevel > 2) {
+                        std::cerr << "peak is beyond end"  << std::endl;
+                    }
+                } else {
+                    softPeakCandidates.insert(peak);
+                    lastSoftPeak = peak;
+                }
+            }
+
+            softPeakAmnesty = minspacing + maxindex - middle;
+            if (m_debugLevel > 2) {
+                std::cerr << "amnesty = " << softPeakAmnesty << std::endl;
+            }
+
+        } else if (softPeakAmnesty > 0) --softPeakAmnesty;
+
+        if (medianwin.size() >= medianmaxsize) {
+            medianwin.pop_front();
+        }
+        if (i < df.size()) medianwin.push_back(df[i]);
+    }
+
+    std::vector<Peak> peaks;
+
+    //!!!
+//    if (!softPeakCandidates.empty()) {
+//        std::cerr << "clearing " << softPeakCandidates.size() << " soft peak candidates" << std::endl;
+//    }
+//    softPeakCandidates.clear();
+
+
+    while (!hardPeakCandidates.empty() || !softPeakCandidates.empty()) {
+        bool haveHardPeak = !hardPeakCandidates.empty();
+        bool haveSoftPeak = !softPeakCandidates.empty();
+        size_t hardPeak = (haveHardPeak ? *hardPeakCandidates.begin() : 0);
+        size_t softPeak = (haveSoftPeak ? *softPeakCandidates.begin() : 0);
+        Peak peak;
+        peak.hard = false;
+        peak.frame = softPeak;
+        if (haveHardPeak &&
+            (!haveSoftPeak || hardPeak <= softPeak)) {
+            if (m_debugLevel > 2) {
+                std::cerr << "Hard peak: " << hardPeak << std::endl;
+            }
+            peak.hard = true;
+            peak.frame = hardPeak;
+            hardPeakCandidates.erase(hardPeakCandidates.begin());
+        } else {
+            if (m_debugLevel > 2) {
+                std::cerr << "Soft peak: " << softPeak << std::endl;
+            }
+        }            
+
+        if (haveSoftPeak && peak.frame == softPeak) {
+            softPeakCandidates.erase(softPeakCandidates.begin());
+        }
+            
+        peaks.push_back(peak);
+    }                
+
+    return peaks;
+}
+
+std::vector<float>
+StretchCalculator::smoothDF(const std::vector<float> &df)
+{
+    std::vector<float> smoothedDF;
+    
+    for (size_t i = 0; i < df.size(); ++i) {
+        // three-value moving mean window for simple smoothing
+        float total = 0.f, count = 0;
+        if (i > 0) { total += df[i-1]; ++count; }
+        total += df[i]; ++count;
+        if (i+1 < df.size()) { total += df[i+1]; ++count; }
+        float mean = total / count;
+//        if (isnan(mean)) {
+//            std::cerr << "ERROR: mean at " << i << " (of " << df.size() << ") is NaN: dfs are: "
+//                      << df[i-1] << ", " << df[i] << ", " << df[i+1] << std::endl;
+//        }
+        smoothedDF.push_back(mean);
+    }
+
+    return smoothedDF;
+}
+
+std::vector<int>
+StretchCalculator::distributeRegion(const std::vector<float> &dfIn,
+                                    size_t duration, float ratio, bool lock)
+{
+    std::vector<float> df(dfIn);
+    std::vector<int> increments;
+
+    // The peak for the stretch detection function may appear after
+    // the peak that we're using to calculate the start of the region.
+    // We don't want that.  If we find a peak in the first half of
+    // the region, we should set all the values up to that point to
+    // the same value as the peak.
+
+    //!!! this is not subtle enough, especially if the region is long
+    //-- we want a bound that corresponds to acoustic perception of
+    //the audible bounce
+
+    for (size_t i = 1; i < df.size()/2; ++i) {
+        if (df[i] < df[i-1]) {
+            if (m_debugLevel > 1) {
+                std::cerr << "stretch peak offset: " << i-1 << " (peak " << df[i-1] << ")" << std::endl;
+            }
+            for (size_t j = 0; j < i-1; ++j) {
+                df[j] = df[i-1];
+            }
+            break;
+        }
+    }
+
+//    for (size_t i = 0; i < df.size(); ++i) {
+//        if (i == 0 || df[i] > maxDf) maxDf = df[i];
+//    }
+
+    long toAllot = long(duration) - long(m_increment * df.size());
+//    bool negative = (toAllot < 0);
+    
+    if (m_debugLevel > 1) {
+        std::cerr << "region of " << df.size() << " blocks, output duration " << duration << ", toAllot " << toAllot << std::endl;
+    }
+
+    size_t totalIncrement = 0;
+
+    //!!! we need to place limits on the amount of displacement per
+    //chunk.  if ratio < 0, no increment should be larger than
+    //increment*ratio or smaller than increment*ratio/2; if ratio > 0,
+    //none should be smaller than increment*ratio or larger than
+    //increment*ratio*2.  We need to enforce this in the assignment of
+    //displacements to allotments, not by trying to respond if
+    //something turns out wrong
+
+    //!!! ratio is only provided to this function for the purposes of
+    //establishing this bound to the displacement
+    
+    // so if maxDisplacement / totalDisplacement > increment * ratio*2 - increment (for ratio > 1)
+    // or maxDisplacement / totalDisplacement < increment * ratio/2 (for ratio < 1)
+    // then we need to adjust... what?
+    
+    bool acceptableSquashRange = false;
+
+    double totalDisplacement = 0;
+    double maxDisplacement = 0; // min displacement will be 0 by definition
+
+    float maxDf = 0;
+    float adj = 0;
+
+    while (!acceptableSquashRange) {
+
+        acceptableSquashRange = true;
+        calculateDisplacements(df, maxDf, totalDisplacement, maxDisplacement,
+                               adj);
+
+        if (m_debugLevel > 1) {
+            std::cerr << "totalDisplacement " << totalDisplacement << ", max " << maxDisplacement << " (maxDf " << maxDf << ", df count " << df.size() << ")" << std::endl;
+        }
+
+        if (totalDisplacement == 0) {
+// Not usually a problem, in fact
+//            std::cerr << "WARNING: totalDisplacement == 0 (duration " << duration << ", " << df.size() << " values in df)" << std::endl;
+            if (!df.empty() && adj == 0) {
+                acceptableSquashRange = false;
+                adj = 1;
+            }
+            continue;
+        }
+
+        int extremeIncrement = m_increment + lrint((toAllot * maxDisplacement) / totalDisplacement);
+        if (ratio < 1.0) {
+            if (extremeIncrement > lrint(ceil(m_increment * ratio))) {
+                std::cerr << "ERROR: extreme increment " << extremeIncrement << " > " << m_increment * ratio << " (I thought this couldn't happen?)" << std::endl;
+            } else if (extremeIncrement < (m_increment * ratio) / 2) {
+                if (m_debugLevel > 0) {
+                    std::cerr << "WARNING: extreme increment " << extremeIncrement << " < " << (m_increment * ratio) / 2 << std::endl;
+                }
+                acceptableSquashRange = false;
+            }
+        } else {
+            if (extremeIncrement > m_increment * ratio * 2) {
+                if (m_debugLevel > 0) {
+                    std::cerr << "WARNING: extreme increment " << extremeIncrement << " > " << m_increment * ratio * 2 << std::endl;
+                }
+                acceptableSquashRange = false;
+            } else if (extremeIncrement < lrint(floor(m_increment * ratio))) {
+                std::cerr << "ERROR: extreme increment " << extremeIncrement << " < " << m_increment * ratio << " (I thought this couldn't happen?)" << std::endl;
+            }
+        }
+
+        if (!acceptableSquashRange) {
+            // Need to make maxDisplacement smaller as a proportion of
+            // the total displacement, yet ensure that the
+            // displacements still sum to the total.  How?
+
+//            std::cerr << "Adjusting df values by " << maxDf/10 << "..." << std::endl;
+
+//            std::cerr << "now: ";
+//            for (size_t i = 0; i < df.size(); ++i) {
+//                df[i] += maxDf/10;
+//                std::cerr << df[i] << " ";
+//            }
+//            std::cerr << std::endl;
+            adj += maxDf/10;
+            
+            //...
+        }
+    }
+
+    for (size_t i = 0; i < df.size(); ++i) {
+
+        double displacement = maxDf - df[i];
+        if (displacement < 0) displacement -= adj;
+        else displacement += adj;
+
+        if (i == 0 && lock) {
+            if (df.size() == 1) {
+                increments.push_back(duration);
+                totalIncrement += duration;
+            } else {
+                increments.push_back(m_increment);
+                totalIncrement += m_increment;
+            }
+            totalDisplacement -= displacement;
+            continue;
+        }
+
+        double theoreticalAllotment = 0;
+
+        if (totalDisplacement != 0) {
+            theoreticalAllotment = (toAllot * displacement) / totalDisplacement;
+        }
+        int allotment = lrint(theoreticalAllotment);
+        if (i + 1 == df.size()) allotment = toAllot;
+
+        int increment = m_increment + allotment;
+
+        if (increment <= 0) {
+            //!!! this is a serious problem, the allocation is quite wrong if it allows increment to diverge so far from the input increment
+            std::cerr << "*** WARNING: increment " << increment << " <= 0, rounding to zero" << std::endl;
+            increment = 0;
+            allotment = increment - m_increment;
+        }
+
+        increments.push_back(increment);
+        totalIncrement += increment;
+
+        toAllot -= allotment;
+        totalDisplacement -= displacement;
+
+        if (m_debugLevel > 2) {
+            std::cerr << "df " << df[i] << ", smoothed " << df[i] << ", disp " << displacement << ", allot " << theoreticalAllotment << ", incr " << increment << ", remain " << toAllot << std::endl;
+        }
+    }
+    
+    if (m_debugLevel > 2) {
+        std::cerr << "total increment: " << totalIncrement << ", left over: " << toAllot << " to allot, displacement " << totalDisplacement << std::endl;
+    }
+
+    if (totalIncrement != duration) {
+        std::cerr << "*** WARNING: calculated output duration " << totalIncrement << " != expected " << duration << std::endl;
+    }
+
+    return increments;
+}
+
+void
+StretchCalculator::calculateDisplacements(const std::vector<float> &df,
+                                          float &maxDf,
+                                          double &totalDisplacement,
+                                          double &maxDisplacement,
+                                          float adj) const
+{
+    totalDisplacement = maxDisplacement = 0;
+
+    maxDf = 0;
+
+    for (size_t i = 0; i < df.size(); ++i) {
+        if (i == 0 || df[i] > maxDf) maxDf = df[i];
+    }
+
+    for (size_t i = 0; i < df.size(); ++i) {
+        double displacement = maxDf - df[i];
+        if (displacement < 0) displacement -= adj;
+        else displacement += adj;
+        totalDisplacement += displacement;
+        if (i == 0 || displacement > maxDisplacement) {
+            maxDisplacement = displacement;
+        }
+    }
+}
+
+}
+

src/StretchCalculator.h

@@ -0,0 +1,89 @@
+/* -*- c-basic-offset: 4 indent-tabs-mode: nil -*-  vi:set ts=8 sts=4 sw=4: */
+
+/*
+    Rubber Band
+    An audio time-stretching and pitch-shifting library.
+    Copyright 2007 Chris Cannam.
+    
+    This program is free software; you can redistribute it and/or
+    modify it under the terms of the GNU General Public License as
+    published by the Free Software Foundation; either version 2 of the
+    License, or (at your option) any later version.  See the file
+    COPYING included with this distribution for more information.
+*/
+
+#ifndef _RUBBERBAND_STRETCH_CALCULATOR_H_
+#define _RUBBERBAND_STRETCH_CALCULATOR_H_
+
+#include <sys/types.h>
+
+#include <vector>
+
+namespace RubberBand
+{
+
+class StretchCalculator
+{
+public:
+    StretchCalculator(size_t sampleRate, size_t inputIncrement, bool useHardPeaks);
+    virtual ~StretchCalculator();
+
+    /**
+     * Calculate phase increments for a region of audio, given the
+     * overall target stretch ratio, input duration in audio samples,
+     * and the audio curves to use for identifying phase lock points
+     * (lockAudioCurve) and for allocating stretches to relatively
+     * less prominent points (stretchAudioCurve).
+     */
+    virtual std::vector<int> calculate(double ratio, size_t inputDuration,
+                                       const std::vector<float> &lockAudioCurve,
+                                       const std::vector<float> &stretchAudioCurve);
+
+    /**
+     * Calculate the phase increment for a single audio block, given
+     * the overall target stretch ratio and the block's value on the
+     * phase-lock audio curve.  State is retained between calls in the
+     * StretchCalculator object; call reset() to reset it.  This uses
+     * a less sophisticated method than the offline calculate().
+     */
+    virtual int calculateSingle(double ratio, size_t inputDurationSoFar,
+                                float curveValue);
+
+    void reset();
+  
+    void setDebugLevel(int level) { m_debugLevel = level; }
+
+    std::vector<float> smoothDF(const std::vector<float> &df);
+
+protected:
+    struct Peak {
+        size_t frame;
+        bool hard;
+    };
+    std::vector<Peak> findPeaks(const std::vector<float> &audioCurve);
+
+    std::vector<int> distributeRegion(const std::vector<float> &regionCurve,
+                                      size_t outputDuration, float ratio,
+                                      bool lock);
+
+    void calculateDisplacements(const std::vector<float> &df,
+                                float &maxDf,
+                                double &totalDisplacement,
+                                double &maxDisplacement,
+                                float adj) const;
+
+    size_t m_sampleRate;
+    size_t m_blockSize;
+    size_t m_increment;
+    float m_prevDf;
+    double m_divergence;
+    float m_recovery;
+    float m_prevRatio;
+    bool m_wasTransient;
+    int m_debugLevel;
+    bool m_useHardPeaks;
+};
+
+}
+
+#endif

src/StretcherChannelData.cpp

@@ -0,0 +1,264 @@
+/* -*- c-basic-offset: 4 indent-tabs-mode: nil -*-  vi:set ts=8 sts=4 sw=4: */
+
+#include "StretcherChannelData.h"
+
+#include "Resampler.h"
+
+namespace RubberBand 
+{
+
+RubberBandStretcher::Impl::ChannelData::ChannelData(size_t blockSize,
+                                                    size_t outbufSize)
+{
+    std::set<size_t> s;
+    construct(s, blockSize, outbufSize);
+}
+
+RubberBandStretcher::Impl::ChannelData::ChannelData(const std::set<size_t> &blockSizes,
+                                                    size_t initialBlockSize,
+                                                    size_t outbufSize)
+{
+    construct(blockSizes, initialBlockSize, outbufSize);
+}
+
+void
+RubberBandStretcher::Impl::ChannelData::construct(const std::set<size_t> &blockSizes,
+                                                  size_t initialBlockSize,
+                                                  size_t outbufSize)
+{
+    size_t maxSize = initialBlockSize;
+
+    if (!blockSizes.empty()) {
+        // std::set is ordered by value
+        std::set<size_t>::const_iterator i = blockSizes.end();
+        maxSize = *--i;
+    }
+    if (blockSizes.find(initialBlockSize) == blockSizes.end()) {
+        if (initialBlockSize > maxSize) maxSize = initialBlockSize;
+    }
+
+    size_t realSize = maxSize/2 + 1; // size of the real "half" of freq data
+
+    std::cerr << "ChannelData::construct([" << blockSizes.size() << "], " << maxSize << ", " << outbufSize << ")" << std::endl;
+    
+    if (outbufSize < maxSize) outbufSize = maxSize;
+
+    inbuf = new RingBuffer<float>(maxSize);
+    outbuf = new RingBuffer<float>(outbufSize);
+
+    mag = new double[realSize];
+    phase = new double[realSize];
+    prevPhase = new double[realSize];
+    unwrappedPhase = new double[realSize];
+    freqPeak = new size_t[realSize];
+
+    accumulator = new float[maxSize];
+    windowAccumulator = new float[maxSize];
+
+    fltbuf = new float[maxSize];
+    dblbuf = new double[maxSize];
+
+    for (std::set<size_t>::const_iterator i = blockSizes.begin();
+         i != blockSizes.end(); ++i) {
+        ffts[*i] = new FFT(*i);
+        ffts[*i]->initDouble();
+    }
+    if (blockSizes.find(initialBlockSize) == blockSizes.end()) {
+        ffts[initialBlockSize] = new FFT(initialBlockSize);
+        ffts[initialBlockSize]->initDouble();
+    }
+    fft = ffts[initialBlockSize];
+
+    resampler = 0;
+    resamplebuf = 0;
+    resamplebufSize = 0;
+
+    reset();
+
+    for (size_t i = 0; i < realSize; ++i) {
+        mag[i] = 0.0;
+        phase[i] = 0.0;
+        prevPhase[i] = 0.0;
+        unwrappedPhase[i] = 0.0;
+        freqPeak[i] = 0;
+    }
+
+    for (size_t i = 0; i < maxSize; ++i) {
+        accumulator[i] = 0.f;
+        windowAccumulator[i] = 0.f;
+        dblbuf[i] = 0.0;
+        fltbuf[i] = 0.0;
+    }
+}
+
+void
+RubberBandStretcher::Impl::ChannelData::setBlockSize(size_t blockSize)
+{
+    size_t oldSize = inbuf->getSize();
+    size_t realSize = blockSize/2 + 1;
+
+    std::cerr << "ChannelData::setBlockSize(" << blockSize << ") [from " << oldSize << "]" << std::endl;
+
+    if (oldSize >= blockSize) {
+
+        // no need to reallocate buffers, just reselect fft
+
+        //!!! we can't actually do this without locking against the
+        //process thread, can we?  we need to zero the mag/phase
+        //buffers without interference
+
+        if (ffts.find(blockSize) == ffts.end()) {
+            //!!! this also requires a lock, but it shouldn't occur in
+            //RT mode with proper initialisation
+            ffts[blockSize] = new FFT(blockSize);
+            ffts[blockSize]->initDouble();
+        }
+        
+        fft = ffts[blockSize];
+
+        for (size_t i = 0; i < realSize; ++i) {
+            mag[i] = 0.0;
+            phase[i] = 0.0;
+            prevPhase[i] = 0.0;
+            unwrappedPhase[i] = 0.0;
+            freqPeak[i] = 0;
+        }
+
+        return;
+    }
+
+    //!!! at this point we need a lock in case a different client
+    //thread is calling process() -- we need this lock even if we
+    //aren't running in threaded mode ourselves -- if we're in RT
+    //mode, then the process call should trylock and fail if the lock
+    //is unavailable (since this should never normally be the case in
+    //general use in RT mode)
+
+    RingBuffer<float> *newbuf = inbuf->resized(blockSize);
+    delete inbuf;
+    inbuf = newbuf;
+
+    // We don't want to preserve data in these arrays
+
+    delete[] mag;
+    delete[] phase;
+    delete[] prevPhase;
+    delete[] unwrappedPhase;
+    delete[] freqPeak;
+
+    mag = new double[realSize];
+    phase = new double[realSize];
+    prevPhase = new double[realSize];
+    unwrappedPhase = new double[realSize];
+    freqPeak = new size_t[realSize];
+
+    delete[] fltbuf;
+    delete[] dblbuf;
+
+    fltbuf = new float[blockSize];
+    dblbuf = new double[blockSize];
+
+    // But we do want to preserve data in these
+
+    float *newAcc = new float[blockSize];
+    for (size_t i = 0; i < oldSize; ++i) newAcc[i] = accumulator[i];
+    delete[] accumulator;
+    accumulator = newAcc;
+
+    newAcc = new float[blockSize];
+    for (size_t i = 0; i < oldSize; ++i) newAcc[i] = windowAccumulator[i];
+    delete[] windowAccumulator;
+    windowAccumulator = newAcc;
+    
+    //!!! and resampler?
+
+    for (size_t i = 0; i < realSize; ++i) {
+        mag[i] = 0.0;
+        phase[i] = 0.0;
+        prevPhase[i] = 0.0;
+        unwrappedPhase[i] = 0.0;
+        freqPeak[i] = 0;
+    }
+
+    for (size_t i = 0; i < blockSize; ++i) {
+        dblbuf[i] = 0.0;
+        fltbuf[i] = 0.0;
+    }
+
+    for (size_t i = oldSize; i < blockSize; ++i) {
+        accumulator[i] = 0.f;
+        windowAccumulator[i] = 0.f;
+    }
+
+    if (ffts.find(blockSize) == ffts.end()) {
+        ffts[blockSize] = new FFT(blockSize);
+        ffts[blockSize]->initDouble();
+    }
+    
+    fft = ffts[blockSize];
+}
+
+void
+RubberBandStretcher::Impl::ChannelData::setOutbufSize(size_t outbufSize)
+{
+    size_t oldSize = outbuf->getSize();
+
+    std::cerr << "ChannelData::setOutbufSize(" << outbufSize << ") [from " << oldSize << "]" << std::endl;
+
+    if (oldSize < outbufSize) {
+
+        //!!! at this point we need a lock in case a different client
+        //thread is calling process()
+
+        //!!! this doesn't do what we want -- we want a locking resize
+        //that preserves the existing data
+//        outbuf->resize(outbufSize);
+
+        RingBuffer<float> *newbuf = outbuf->resized(outbufSize);
+        delete outbuf;
+        outbuf = newbuf;
+    }
+}
+
+RubberBandStretcher::Impl::ChannelData::~ChannelData()
+{
+    delete resampler;
+    delete[] resamplebuf;
+
+    delete inbuf;
+    delete outbuf;
+    delete[] mag;
+    delete[] phase;
+    delete[] prevPhase;
+    delete[] unwrappedPhase;
+    delete[] freqPeak;
+    delete[] accumulator;
+    delete[] windowAccumulator;
+    delete[] fltbuf;
+    delete[] dblbuf;
+
+    for (std::map<size_t, FFT *>::iterator i = ffts.begin();
+         i != ffts.end(); ++i) {
+        delete i->second;
+    }
+}
+
+void
+RubberBandStretcher::Impl::ChannelData::reset()
+{
+    inbuf->reset();
+    outbuf->reset();
+
+    if (resampler) resampler->reset();
+
+    accumulatorFill = 0;
+    prevIncrement = 0;
+    blockCount = 0;
+    inCount = 0;
+    inputSize = -1;
+    outCount = 0;
+    draining = false;
+    outputComplete = false;
+}
+
+}

src/StretcherChannelData.h

@@ -0,0 +1,121 @@
+/* -*- c-basic-offset: 4 indent-tabs-mode: nil -*-  vi:set ts=8 sts=4 sw=4: */
+
+/*
+    Rubber Band
+    An audio time-stretching and pitch-shifting library.
+    Copyright 2007 Chris Cannam.
+    
+    This program is free software; you can redistribute it and/or
+    modify it under the terms of the GNU General Public License as
+    published by the Free Software Foundation; either version 2 of the
+    License, or (at your option) any later version.  See the file
+    COPYING included with this distribution for more information.
+*/
+
+#ifndef _RUBBERBAND_STRETCHERCHANNELDATA_H_
+#define _RUBBERBAND_STRETCHERCHANNELDATA_H_
+
+#include "StretcherImpl.h"
+
+#include <set>
+
+namespace RubberBand
+{
+
+class Resampler;
+	
+class RubberBandStretcher::Impl::ChannelData
+{
+public:        
+    /**
+     * Construct a ChannelData structure.
+     *
+     * The block size passed in here is the size for the FFT
+     * calculation, and most of the buffer sizes also depend on
+     * it.  In practice it is always a power of two and except for
+     * very extreme stretches is always either 1024, 2048 or 4096.
+     *
+     * The outbuf size depends on other factors as well, including
+     * the pitch scale factor and any maximum processing block
+     * size specified by the user of the code.
+     */
+    ChannelData(size_t blockSize, size_t outbufSize);
+
+    /**
+     * Construct a ChannelData structure that can process at
+     * different FFT sizes without requiring reallocation when the
+     * size changes.  The size can subsequently be changed with a
+     * call to setBlockSize.  Reallocation will only be necessary
+     * if setBlockSize is called with a value not equal to one of
+     * those passed in to the constructor.
+     *
+     * The outbufSize should be the maximum possible outbufSize to
+     * avoid reallocation, which will happen if setOutbufSize is
+     * called subsequently.
+     */
+    ChannelData(const std::set<size_t> &blockSizes,
+                size_t initialBlockSize, size_t outbufSize);
+    ~ChannelData();
+
+    /**
+     * Reset buffers
+     */
+    void reset();
+
+    /**
+     * Set the FFT and buffer sizes from the given processing
+     * block size.  If this ChannelData was constructed with a set
+     * of block sizes and the given block size here was among
+     * them, no reallocation will be required.
+     */
+    void setBlockSize(size_t blockSize);
+
+    /**
+     * Set the outbufSize for the channel data.  Reallocation will
+     * occur.
+     */
+    void setOutbufSize(size_t outbufSize);
+
+    RingBuffer<float> *inbuf;
+    RingBuffer<float> *outbuf;
+
+    double *mag;
+    double *phase;
+
+    double *prevPhase;
+    double *unwrappedPhase;
+
+    size_t *freqPeak;
+
+    float *accumulator;
+    size_t accumulatorFill;
+    float *windowAccumulator;
+
+    float *fltbuf;
+    double *dblbuf;
+
+    size_t prevIncrement; // only used in RT mode
+
+    size_t blockCount;
+    size_t inCount;
+    long inputSize; // set only after known (when data ended); -1 previously
+    size_t outCount;
+
+    bool draining;
+    bool outputComplete;
+
+    FFT *fft;
+    std::map<size_t, FFT *> ffts;
+
+    Resampler *resampler;
+    float *resamplebuf;
+    size_t resamplebufSize;
+
+private:
+    void construct(const std::set<size_t> &blockSizes,
+                   size_t initialBlockSize, size_t outbufSize);
+};        
+
+}
+
+#endif

src/StretcherImpl.cpp

@@ -0,0 +1,978 @@
+/* -*- c-basic-offset: 4 indent-tabs-mode: nil -*-  vi:set ts=8 sts=4 sw=4: */
+
+/*
+    Rubber Band
+    An audio time-stretching and pitch-shifting library.
+    Copyright 2007 Chris Cannam.
+    
+    This program is free software; you can redistribute it and/or
+    modify it under the terms of the GNU General Public License as
+    published by the Free Software Foundation; either version 2 of the
+    License, or (at your option) any later version.  See the file
+    COPYING included with this distribution for more information.
+*/
+
+#include "StretcherImpl.h"
+#include "PercussiveAudioCurve.h"
+#include "HighFrequencyAudioCurve.h"
+#include "ConstantAudioCurve.h"
+#include "StretchCalculator.h"
+#include "StretcherChannelData.h"
+#include "Resampler.h"
+
+#include <cassert>
+#include <cmath>
+#include <set>
+#include <map>
+
+using std::cerr;
+using std::endl;
+using std::vector;
+using std::map;
+using std::set;
+using std::max;
+using std::min;
+
+namespace RubberBand {
+
+static const size_t defaultIncrement = 256;
+static const size_t defaultBlockSize = 2048;
+    
+RubberBandStretcher::Impl::Impl(RubberBandStretcher *stretcher,
+                                size_t sampleRate,
+                                size_t channels,
+                                Options options,
+                                double initialTimeRatio,
+                                double initialPitchScale) :
+    m_stretcher(stretcher),
+    m_channels(channels),
+    m_timeRatio(initialTimeRatio),
+    m_pitchScale(initialPitchScale),
+    m_blockSize(defaultBlockSize),
+    m_outbufSize(defaultBlockSize * 2),
+    m_increment(defaultIncrement),
+    m_maxProcessBlockSize(defaultBlockSize),
+    m_expectedInputDuration(0),
+    m_threaded(false),
+    m_realtime(false),
+    m_options(options),
+    m_debugLevel(1),
+    m_mode(JustCreated),
+    m_window(0),
+    m_studyFFT(0),
+    m_inputDuration(0),
+    m_lastProcessOutputIncrements(16),
+    m_lastProcessLockDf(16),
+    m_lockAudioCurve(0),
+    m_stretchAudioCurve(0),
+    m_stretchCalculator(0),
+    m_freq0(600),
+    m_freq1(1200),
+    m_freq2(12000),
+    m_baseBlockSize(defaultBlockSize)
+{
+    cerr << "RubberBandStretcher::Impl::Impl: options = " << options << endl;
+
+    if ((options & OptionWindowShort) || (options & OptionWindowLong)) {
+        if ((options & OptionWindowShort) && (options & OptionWindowLong)) {
+            cerr << "RubberBandStretcher::Impl::Impl: Cannot specify OptionWindowLong and OptionWindowShort together; falling back to OptionWindowStandard" << endl;
+        } else if (options & OptionWindowShort) {
+            m_baseBlockSize = defaultBlockSize / 2;
+            cerr << "setting baseBlockSize to " << m_baseBlockSize << endl;
+        } else if (options & OptionWindowLong) {
+            m_baseBlockSize = defaultBlockSize * 2;
+            cerr << "setting baseBlockSize to " << m_baseBlockSize << endl;
+        }
+        m_blockSize = m_baseBlockSize;
+        m_outbufSize = m_baseBlockSize * 2;
+        m_maxProcessBlockSize = m_baseBlockSize;
+    }
+
+    if (m_options & OptionProcessRealTime) {
+
+        m_realtime = true;
+
+        if (!(m_options & OptionStretchPrecise)) {
+            cerr << "RubberBandStretcher::Impl::Impl: Real-time mode: enabling OptionStretchPrecise" << endl;
+            m_options |= OptionStretchPrecise;
+        }
+    }
+
+    if (m_channels > 1) {
+
+        if (!m_realtime &&
+            !(m_options & OptionThreadingNone) &&
+            Thread::threadingAvailable() &&
+            system_is_multiprocessor()) {
+
+            m_threaded = true;
+        
+            if (m_debugLevel > 0) {
+                cerr << "Going multithreaded..." << endl;
+            }
+        }
+    }
+
+    configure();
+}
+
+RubberBandStretcher::Impl::~Impl()
+{
+    if (m_threaded) {
+        MutexLocker locker(&m_threadSetMutex);
+        for (set<ProcessThread *>::iterator i = m_threadSet.begin();
+             i != m_threadSet.end(); ++i) {
+            if (m_debugLevel > 0) {
+                cerr << "RubberBandStretcher::~RubberBandStretcher: joining (channel " << *i << ")" << endl;
+            }
+            (*i)->wait();
+            delete *i;
+        }
+    }
+
+    for (size_t c = 0; c < m_channels; ++c) {
+        delete m_channelData[c];
+    }
+
+    delete m_lockAudioCurve;
+    delete m_stretchAudioCurve;
+    delete m_stretchCalculator;
+    delete m_studyFFT;
+
+    for (map<size_t, Window<float> *>::iterator i = m_windows.begin();
+         i != m_windows.end(); ++i) {
+        delete i->second;
+    }
+}
+
+void
+RubberBandStretcher::Impl::reset()
+{
+    //!!! does not do the right thing in threaded mode
+
+    if (m_threaded) m_threadSetMutex.lock();
+
+    for (size_t c = 0; c < m_channels; ++c) {
+        delete m_channelData[c];
+        m_channelData[c] = new ChannelData(m_blockSize, m_outbufSize);
+    }
+    m_mode = JustCreated;
+    if (m_lockAudioCurve) m_lockAudioCurve->reset();
+    if (m_stretchAudioCurve) m_stretchAudioCurve->reset();
+    m_inputDuration = 0;
+
+    if (m_threaded) m_threadSetMutex.unlock();
+//    m_done = false;
+}
+
+void
+RubberBandStretcher::Impl::setTimeRatio(double ratio)
+{
+    if (!m_realtime) {
+        if (m_mode == Studying || m_mode == Processing) {
+            cerr << "RubberBandStretcher::Impl::setTimeRatio: Cannot set ratio while studying or processing in non-RT mode" << endl;
+            return;
+        }
+    }
+
+    if (ratio == m_timeRatio) return;
+    m_timeRatio = ratio;
+
+    reconfigure();
+}
+
+void
+RubberBandStretcher::Impl::setPitchScale(double fs)
+{
+    if (!m_realtime) {
+        if (m_mode == Studying || m_mode == Processing) {
+            cerr << "RubberBandStretcher::Impl::setPitchScale: Cannot set ratio while studying or processing in non-RT mode" << endl;
+            return;
+        }
+    }
+
+    if (fs == m_pitchScale) return;
+    m_pitchScale = fs;
+
+    reconfigure();
+}
+
+double
+RubberBandStretcher::Impl::getTimeRatio() const
+{
+    return m_timeRatio;
+}
+
+double
+RubberBandStretcher::Impl::getPitchScale() const
+{
+    return m_pitchScale;
+}
+
+void
+RubberBandStretcher::Impl::setExpectedInputDuration(size_t samples)
+{
+    if (samples == m_expectedInputDuration) return;
+    m_expectedInputDuration = samples;
+
+    reconfigure();
+}
+
+void
+RubberBandStretcher::Impl::setMaxProcessBlockSize(size_t samples)
+{
+    if (samples <= m_maxProcessBlockSize) return;
+    m_maxProcessBlockSize = samples;
+
+    reconfigure();
+}
+
+float
+RubberBandStretcher::Impl::getFrequencyCutoff(int n) const
+{
+    switch (n) {
+    case 0: return m_freq0;
+    case 1: return m_freq1;
+    case 2: return m_freq2;
+    }
+    return 0.f;
+}
+
+void
+RubberBandStretcher::Impl::setFrequencyCutoff(int n, float f)
+{
+    switch (n) {
+    case 0: m_freq0 = f; break;
+    case 1: m_freq1 = f; break;
+    case 2: m_freq2 = f; break;
+    }
+}
+
+double
+RubberBandStretcher::Impl::getEffectiveRatio() const
+{
+    // Returns the ratio that the internal time stretcher needs to
+    // achieve, not the resulting duration ratio of the output (which
+    // is simply m_timeRatio).
+
+    // A frequency shift is achieved using an additional time shift,
+    // followed by resampling back to the original time shift to
+    // change the pitch.  Note that the resulting frequency change is
+    // fixed, as it is effected by the resampler -- in contrast to
+    // time shifting, which is variable aiming to place the majority
+    // of the stretch or squash in low-interest regions of audio.
+
+    return m_timeRatio * m_pitchScale;
+}
+
+size_t
+RubberBandStretcher::Impl::roundUp(size_t value)
+{
+    if (!(value & (value - 1))) return value;
+    int bits = 0;
+    while (value) { ++bits; value >>= 1; }
+    value = 1 << bits;
+    return value;
+}
+
+void
+RubberBandStretcher::Impl::calculateSizes()
+{
+    size_t inputIncrement = defaultIncrement;
+    size_t blockSize = m_baseBlockSize;
+    size_t outputIncrement;
+
+    double r = getEffectiveRatio();
+
+    if (m_realtime) {
+
+        // use a fixed input increment
+
+        inputIncrement = defaultIncrement;
+
+        if (r < 1) {
+            outputIncrement = int(floor(inputIncrement * r));
+            if (outputIncrement < 1) {
+                outputIncrement = 1;
+                inputIncrement = roundUp(lrint(ceil(outputIncrement / r)));
+                blockSize = inputIncrement * 4;
+            }
+        } else {
+            outputIncrement = int(ceil(inputIncrement * r));
+            while (outputIncrement > 1024 && inputIncrement > 1) {
+                inputIncrement /= 2;
+                outputIncrement = lrint(ceil(inputIncrement * r));
+            }
+            blockSize = std::max(blockSize, roundUp(outputIncrement * 4.5));
+        }
+
+    } else {
+
+        // use a variable increment
+
+        if (r < 1) {
+            inputIncrement = blockSize / 4;
+            while (inputIncrement >= 512) inputIncrement /= 2;
+            outputIncrement = int(floor(inputIncrement * r));
+            if (outputIncrement < 1) {
+                outputIncrement = 1;
+                inputIncrement = roundUp(lrint(ceil(outputIncrement / r)));
+                blockSize = inputIncrement * 4;
+            }
+        } else {
+            outputIncrement = blockSize / 6;
+            inputIncrement = int(outputIncrement / r);
+            while (outputIncrement > 1024 && inputIncrement > 1) {
+                outputIncrement /= 2;
+                inputIncrement = int(outputIncrement / r);
+            }
+            blockSize = std::max(blockSize, roundUp(outputIncrement * 6));
+        }
+    }        
+
+    if (m_expectedInputDuration > 0) {
+        while (inputIncrement * 4 > m_expectedInputDuration &&
+               inputIncrement > 1) {
+            inputIncrement /= 2;
+        }
+    }
+
+    // blockSize can be almost anything, but it can't be greater than
+    // 4 * defaultBlockSize unless ratio is less than 1/1024.
+
+    m_blockSize = blockSize;
+    m_increment = inputIncrement;
+
+    // When squashing, the greatest theoretically possible output
+    // increment is the input increment.  When stretching adaptively
+    // the sky's the limit in principle, but we expect
+    // StretchCalculator to restrict itself to using no more than
+    // twice the basic output increment (i.e. input increment times
+    // ratio) for any block.
+
+    if (m_debugLevel > 0) {
+        cerr << "configure: effective ratio = " << getEffectiveRatio() << endl;
+        cerr << "configure: block size = " << m_blockSize << ", increment = " << m_increment << " (approx output increment = " << int(lrint(m_increment * getEffectiveRatio())) << ")" << endl;
+    }
+
+    static size_t maxBlockSize = 0;
+
+    if (m_blockSize > maxBlockSize) {
+        //!!!
+        cerr << "configure: NOTE: max block size so far increased from "
+                  << maxBlockSize << " to " << m_blockSize << endl;
+        maxBlockSize = m_blockSize;
+    }
+
+    if (m_blockSize > m_maxProcessBlockSize) {
+        m_maxProcessBlockSize = m_blockSize;
+    }
+
+    m_outbufSize = max
+        (size_t(ceil(m_maxProcessBlockSize / m_pitchScale)),
+         m_blockSize);
+
+    if (m_realtime) {
+        // This headroom is so as to try to avoid reallocation when
+        // the pitch scale changes
+        m_outbufSize = m_outbufSize * 16;
+    } else {
+        if (m_threaded) {
+            // This headroom is to permit the processing threads to
+            // run ahead of the buffer output drainage; the exact
+            // amount of headroom is a question of tuning rather than
+            // results
+            m_outbufSize = m_outbufSize * 16;
+        }
+    }
+
+    //!!! for very long stretches (e.g. x5), this is necessary; for
+    //even longer ones (e.g. x10), even more of an outbuf is
+    //necessary. clearly something wrong in our calculations... or do
+    //we just need to ensure client calls setMaxProcessBlockSize?
+    if (!m_realtime && !m_threaded) {
+//!!!        m_outbufSize = m_outbufSize * 2;
+    }
+}
+
+void
+RubberBandStretcher::Impl::configure()
+{
+    size_t prevBlockSize = m_blockSize;
+    size_t prevOutbufSize = m_outbufSize;
+    if (m_windows.empty()) {
+        prevBlockSize = 0;
+        prevOutbufSize = 0;
+    }
+
+    calculateSizes();
+
+    bool blockSizeChanged = (prevBlockSize != m_blockSize);
+    bool outbufSizeChanged = (prevOutbufSize != m_outbufSize);
+
+    // This function may be called at any time in non-RT mode, after a
+    // parameter has changed.  It shouldn't be legal to call it after
+    // processing has already begun.
+
+    // This function is only called once (on construction) in RT
+    // mode.  After that reconfigure() does the work in a hopefully
+    // RT-safe way.
+
+    set<size_t> blockSizes;
+    if (m_realtime) {
+        blockSizes.insert(m_baseBlockSize);
+        blockSizes.insert(m_baseBlockSize * 2);
+        blockSizes.insert(m_baseBlockSize * 4);
+    }
+    blockSizes.insert(m_blockSize);
+
+    if (blockSizeChanged) {
+
+        for (set<size_t>::const_iterator i = blockSizes.begin();
+             i != blockSizes.end(); ++i) {
+            if (m_windows.find(*i) == m_windows.end()) {
+                m_windows[*i] = new Window<float>(HanningWindow, *i);
+            }
+        }
+        m_window = m_windows[m_blockSize];
+
+        if (m_debugLevel > 0) {
+            cerr << "Window area: " << m_window->getArea() << "; synthesis window area: " << m_window->getArea() << endl;
+        }
+    }
+
+    if (blockSizeChanged || outbufSizeChanged) {
+        
+        for (size_t c = 0; c < m_channelData.size(); ++c) {
+            delete m_channelData[c];
+        }
+        m_channelData.clear();
+
+        for (size_t c = 0; c < m_channels; ++c) {
+            m_channelData.push_back
+                (new ChannelData(blockSizes, m_blockSize, m_outbufSize));
+        }
+    }
+
+    if (!m_realtime && blockSizeChanged) {
+        delete m_studyFFT;
+        m_studyFFT = new FFT(m_blockSize);
+        m_studyFFT->initFloat();
+    }
+
+    if (m_pitchScale != 1.0 || m_realtime) {
+
+        for (size_t c = 0; c < m_channels; ++c) {
+
+            if (m_channelData[c]->resampler) continue;
+
+            m_channelData[c]->resampler =
+                new Resampler(Resampler::FastestTolerable, 1, 4096 * 16);
+
+            m_channelData[c]->resamplebufSize =
+                lrintf(ceil((m_increment * m_timeRatio * 2) / m_pitchScale));
+            m_channelData[c]->resamplebuf =
+                new float[m_channelData[c]->resamplebufSize];
+        }
+    }
+    
+    delete m_lockAudioCurve;
+    m_lockAudioCurve = new PercussiveAudioCurve(m_stretcher->m_sampleRate,
+                                                m_blockSize);
+
+    // stretchAudioCurve unused in RT mode; lockAudioCurve and
+    // stretchCalculator however are used in all modes
+
+    if (!m_realtime) {
+        delete m_stretchAudioCurve;
+        if (!(m_options & OptionStretchPrecise)) {
+            //!!! probably adaptively-whitened spectral difference curve
+            //would be better
+            m_stretchAudioCurve = new HighFrequencyAudioCurve
+                (m_stretcher->m_sampleRate, m_blockSize);
+        } else {
+            m_stretchAudioCurve = new ConstantAudioCurve
+                (m_stretcher->m_sampleRate, m_blockSize);
+        }
+    }
+
+    delete m_stretchCalculator;
+    m_stretchCalculator = new StretchCalculator
+        (m_stretcher->m_sampleRate, m_increment,
+         !(m_options & OptionTransientsSmooth));
+
+    m_stretchCalculator->setDebugLevel(m_debugLevel);
+    m_inputDuration = 0;
+
+    // Prepare the inbufs with half a block of emptiness.  The centre
+    // point of the first processing block for the onset detector
+    // should be the first sample of the audio, and we continue until
+    // we can no longer centre a block within the input audio.  The
+    // number of onset detector blocks will be the number of audio
+    // samples input, divided by the input increment, plus one.
+
+    // In real-time mode, we don't do this prefill -- it's better to
+    // start with a swoosh than introduce more latency, and we don't
+    // want gaps when the ratio changes.
+
+    if (!m_realtime) {
+        for (size_t c = 0; c < m_channels; ++c) {
+            m_channelData[c]->reset();
+            m_channelData[c]->inbuf->zero(m_blockSize/2);
+        }
+    }
+}
+
+
+//!!! separated out from configure() for the moment so we can look at
+// the logic of it
+void
+RubberBandStretcher::Impl::reconfigure()
+{
+    if (!m_realtime) {
+        if (m_mode == Studying) {
+            // stop and calculate the stretch curve so far, then reset
+            // the df vectors
+            calculateStretch();
+            m_lockDf.clear();
+            m_stretchDf.clear();
+            m_inputDuration = 0;
+        }
+        configure();
+    }
+
+    size_t prevBlockSize = m_blockSize;
+    size_t prevOutbufSize = m_outbufSize;
+
+    calculateSizes();
+
+    // There are various allocations in this function, but they should
+    // never happen in normal use -- they just recover from the case
+    // where not all of the things we need were correctly created when
+    // we first configured (for whatever reason).  This is intended to
+    // be "effectively" realtime safe.  The same goes for
+    // ChannelData::setOutbufSize and setBlockSize.
+
+    if (m_blockSize != prevBlockSize) {
+
+        if (m_windows.find(m_blockSize) == m_windows.end()) {
+            std::cerr << "WARNING: reconfigure(): window allocation (size " << m_blockSize << ") required in RT mode" << std::endl;
+            m_windows[m_blockSize] = new Window<float>(HanningWindow, m_blockSize);
+        }
+        m_window = m_windows[m_blockSize];
+
+        for (size_t c = 0; c < m_channels; ++c) {
+            m_channelData[c]->setBlockSize(m_blockSize);
+        }
+    }
+
+    if (m_outbufSize != prevOutbufSize) {
+        for (size_t c = 0; c < m_channels; ++c) {
+            m_channelData[c]->setOutbufSize(m_outbufSize);
+        }
+    }
+
+    if (m_pitchScale != 1.0) {
+        for (size_t c = 0; c < m_channels; ++c) {
+
+            if (m_channelData[c]->resampler) continue;
+
+            std::cerr << "WARNING: reconfigure(): resampler construction required in RT mode" << std::endl;
+
+            m_channelData[c]->resampler =
+                new Resampler(Resampler::FastestTolerable, 1, m_blockSize);
+
+            m_channelData[c]->resamplebufSize =
+                lrintf(ceil((m_increment * m_timeRatio * 2) / m_pitchScale));
+            m_channelData[c]->resamplebuf =
+                new float[m_channelData[c]->resamplebufSize];
+        }
+    }
+
+    if (m_blockSize != prevBlockSize) {
+        m_lockAudioCurve->setBlockSize(m_blockSize);
+    }
+}
+
+size_t
+RubberBandStretcher::Impl::getLatency() const
+{
+    if (!m_realtime) return 0;
+    return int((m_blockSize/2) / m_pitchScale + 1);
+}
+
+void
+RubberBandStretcher::Impl::setTransientsOption(Options options)
+{
+    if (options & OptionTransientsSmooth) {
+        m_options |= OptionTransientsSmooth;
+    } else {
+        m_options &= ~OptionTransientsSmooth;
+    }
+}
+
+void
+RubberBandStretcher::Impl::setPhaseOption(Options options)
+{
+    if (options & OptionPhaseIndependent) {
+        m_options |= OptionPhaseIndependent;
+    } else {
+        m_options &= ~OptionPhaseIndependent;
+    }
+}
+
+void
+RubberBandStretcher::Impl::study(const float *const *input, size_t samples, bool final)
+{
+    if (m_realtime) {
+        if (m_debugLevel > 1) {
+            cerr << "RubberBandStretcher::Impl::study: Not meaningful in realtime mode" << endl;
+        }
+        return;
+    }
+
+    if (m_mode == Processing || m_mode == Finished) {
+        cerr << "RubberBandStretcher::Impl::study: Cannot study after processing" << endl;
+        return;
+    }
+    m_mode = Studying;
+    
+    size_t consumed = 0;
+
+    ChannelData &cd = *m_channelData[0];
+    RingBuffer<float> &inbuf = *cd.inbuf;
+
+    const float *mixdown;
+    float *mdalloc = 0;
+
+    if (m_channels > 1 || final) {
+        // mix down into a single channel for analysis
+        mdalloc = new float[samples];
+        for (size_t i = 0; i < samples; ++i) {
+            if (i < samples) {
+                mdalloc[i] = input[0][i];
+            } else {
+                mdalloc[i] = 0.f;
+            }
+        }
+        for (size_t c = 1; c < m_channels; ++c) {
+            for (size_t i = 0; i < samples; ++i) {
+                mdalloc[i] += input[c][i];
+            }
+        }
+        for (size_t i = 0; i < samples; ++i) {
+            mdalloc[i] /= m_channels;
+        }
+        mixdown = mdalloc;
+    } else {
+        mixdown = input[0];
+    }
+
+    while (consumed < samples) {
+
+	size_t writable = inbuf.getWriteSpace();
+	writable = min(writable, samples - consumed);
+
+	if (writable == 0) {
+            // warn
+            cerr << "WARNING: writable == 0 (consumed = " << consumed << ", samples = " << samples << ")" << endl;
+	} else {
+            inbuf.write(mixdown + consumed, writable);
+            consumed += writable;
+        }
+
+	while ((inbuf.getReadSpace() >= m_blockSize) ||
+               (final && (inbuf.getReadSpace() >= m_blockSize/2))) {
+
+            //!!! inconsistency throughout -- we are using "blocksize",
+            // but "chunk" instead of "block"
+
+	    // We know we have at least m_blockSize samples available
+	    // in m_inbuf.  We need to peek m_blockSize of them for
+	    // processing, and then skip m_increment to advance the
+	    // read pointer.
+
+            // cd.accumulator is not otherwise used during studying,
+            // so we can use it as a temporary buffer here
+
+            size_t got = inbuf.peek(cd.accumulator, m_blockSize);
+            assert(final || got == m_blockSize);
+
+            m_window->cut(cd.accumulator);
+
+            // We don't need the fftshift for studying, as we're only
+            // interested in magnitude
+
+            m_studyFFT->forwardMagnitude(cd.accumulator, cd.fltbuf);
+
+            float df = m_lockAudioCurve->process(cd.fltbuf, m_increment);
+            m_lockDf.push_back(df);
+
+//            cout << m_lockDf.size() << " [" << final << "] -> " << df << " \t: ";
+
+            df = m_stretchAudioCurve->process(cd.fltbuf, m_increment);
+            m_stretchDf.push_back(df);
+
+//            cout << df << endl;
+
+            // We have augmented the input by m_blockSize/2 so
+            // that the first block is centred on the first audio
+            // sample.  We want to ensure that m_inputDuration
+            // contains the exact input duration without including
+            // this extra bit.  We just add up all the increments
+            // here, and deduct the extra afterwards.
+
+            m_inputDuration += m_increment;
+//                cerr << "incr input duration by increment: " << m_increment << " -> " << m_inputDuration << endl;
+            inbuf.skip(m_increment);
+	}
+    }
+
+    if (final) {
+        int rs = inbuf.getReadSpace();
+        m_inputDuration += rs;
+//        cerr << "incr input duration by read space: " << rs << " -> " << m_inputDuration << endl;
+
+        if (m_inputDuration > m_blockSize/2) { // deducting the extra
+            m_inputDuration -= m_blockSize/2;
+        }
+    }
+
+    if (m_channels > 1) delete[] mdalloc;
+}
+
+vector<int>
+RubberBandStretcher::Impl::getOutputIncrements() const
+{
+    if (!m_realtime) {
+        return m_outputIncrements;
+    } else {
+        vector<int> increments;
+        while (m_lastProcessOutputIncrements.getReadSpace() > 0) {
+            increments.push_back(m_lastProcessOutputIncrements.readOne());
+        }
+        return increments;
+    }
+}
+
+vector<float>
+RubberBandStretcher::Impl::getLockCurve() const
+{
+    if (!m_realtime) {
+        return m_lockDf;
+    } else {
+        vector<float> df;
+        while (m_lastProcessLockDf.getReadSpace() > 0) {
+            df.push_back(m_lastProcessLockDf.readOne());
+        }
+        return df;
+    }
+}
+
+void
+RubberBandStretcher::Impl::calculateStretch()
+{
+    std::vector<int> increments = m_stretchCalculator->calculate
+        (getEffectiveRatio(),
+         m_inputDuration,
+         m_lockDf,
+         m_stretchDf);
+
+    if (m_outputIncrements.empty()) m_outputIncrements = increments;
+    else {
+        for (size_t i = 0; i < increments.size(); ++i) {
+            m_outputIncrements.push_back(increments[i]);
+        }
+    }
+    
+    return;
+}
+
+void
+RubberBandStretcher::Impl::setDebugLevel(int level)
+{
+    m_debugLevel = level;
+    if (m_stretchCalculator) m_stretchCalculator->setDebugLevel(level);
+}	
+
+size_t
+RubberBandStretcher::Impl::getSamplesRequired() const
+{
+    size_t reqd = 0;
+
+    for (size_t c = 0; c < m_channels; ++c) {
+
+        size_t reqdHere = 0;
+
+        ChannelData &cd = *m_channelData[c];
+        RingBuffer<float> &inbuf = *cd.inbuf;
+
+        size_t rs = inbuf.getReadSpace();
+
+        // See notes in testInbufReadSpace below
+
+        if (rs < m_blockSize && !cd.draining) {
+            
+            if (cd.inputSize == -1) {
+                reqdHere = m_blockSize - rs;
+                if (reqdHere > reqd) reqd = reqdHere;
+                continue;
+            }
+        
+            if (rs == 0) {
+                reqdHere = m_blockSize;
+                if (reqdHere > reqd) reqd = reqdHere;
+                continue;
+            }
+        }
+    }
+    
+    return reqd;
+}    
+
+void
+RubberBandStretcher::Impl::process(const float *const *input, size_t samples, bool final)
+{
+    if (m_mode == Finished) {
+        cerr << "RubberBandStretcher::Impl::process: Cannot process again after final block" << endl;
+        return;
+    }
+
+    if (m_mode == JustCreated || m_mode == Studying) {
+
+        //!!! m_studying isn't the right test for "is this the first
+        //time we've processed?"
+
+        if (m_mode == Studying) {
+            calculateStretch();
+        }
+
+        for (size_t c = 0; c < m_channels; ++c) {
+            m_channelData[c]->reset();
+            m_channelData[c]->inbuf->zero(m_blockSize/2);
+        }
+
+        if (m_threaded) {
+            MutexLocker locker(&m_threadSetMutex);
+
+            for (size_t c = 0; c < m_channels; ++c) {
+                ProcessThread *thread = new ProcessThread(this, c);
+                m_threadSet.insert(thread);
+                thread->start();
+            }
+            
+            if (m_debugLevel > 0) {
+                cerr << m_channels << " threads created" << endl;
+            }
+        }
+        
+        m_mode = Processing;
+    }
+
+    bool allConsumed = false;
+
+    map<size_t, size_t> consumed;
+    for (size_t c = 0; c < m_channels; ++c) {
+        consumed[c] = 0;
+    }
+
+    while (!allConsumed) {
+
+//        cerr << "process looping" << endl;
+
+//#ifndef NO_THREADING
+//        if (m_threaded) {
+//            pthread_mutex_lock(&m_inputProcessedMutex);
+//        }
+//#endif
+
+        // In a threaded mode, our "consumed" counters only indicate
+        // the number of samples that have been taken into the input
+        // ring buffers waiting to be processed by the process thread.
+        // In non-threaded mode, "consumed" counts the number that
+        // have actually been processed.
+
+        allConsumed = true;
+        for (size_t c = 0; c < m_channels; ++c) {
+            consumed[c] += consumeChannel(c,
+                                          input[c] + consumed[c],
+                                          samples - consumed[c]);
+            if (consumed[c] < samples) {
+                allConsumed = false;
+//                cerr << "process: waiting on input consumption for channel " << c << endl;
+            } else {
+                if (final) {
+                    m_channelData[c]->inputSize = m_channelData[c]->inCount;
+                }
+//                cerr << "process: happy with channel " << c << endl;
+            }
+            if (!m_threaded && !m_realtime) {
+                processChunks(c);
+            }
+        }
+
+        if (m_realtime) {
+            // When running in real time, we need to process both
+            // channels in step because we will need to use the sum of
+            // their frequency domain representations as the input to
+            // the realtime onset detector
+            processOneChunk();
+        }
+
+        if (m_threaded) {
+            m_dataAvailable.signal();
+            m_spaceAvailable.lock();
+            if (!allConsumed) {
+                m_spaceAvailable.wait(500);
+            }
+/*
+        } else {
+            if (!allConsumed) {
+                cerr << "RubberBandStretcher::Impl::process: ERROR: Too much data provided to process() call -- either call setMaxProcessBlockSize() beforehand, or provide only getSamplesRequired() frames at a time" << endl;
+                for (size_t c = 0; c < m_channels; ++c) {
+                    cerr << "channel " << c << ": " << samples << " provided, " << consumed[c] << " consumed" << endl;
+                }
+//                break;
+            }
+*/
+        }
+    }
+    
+//    cerr << "process returning" << endl;
+
+    if (final) m_mode = Finished;
+}
+
+size_t
+RubberBandStretcher::Impl::consumeChannel(size_t c, const float *input, size_t samples)
+{
+    size_t consumed = 0;
+    
+    ChannelData &cd = *m_channelData[c];
+    RingBuffer<float> &inbuf = *cd.inbuf;
+    
+    while (consumed < samples) {
+
+	size_t writable = inbuf.getWriteSpace();
+
+//        cerr << "channel " << c << ": samples remaining = " << samples - consumed << ", writable space = " << writable << endl;
+
+	writable = min(writable, samples - consumed);
+
+	if (writable == 0) {
+            // warn
+//            cerr << "WARNING: writable == 0 for ch " << c << " (consumed = " << consumed << ", samples = " << samples << ")" << endl;
+            return consumed;
+	} else {
+            inbuf.write(input + consumed, writable);
+            consumed += writable;
+            cd.inCount += writable;
+        }
+    }
+
+    return samples;
+}
+
+
+}
+

src/StretcherImpl.h

@@ -0,0 +1,184 @@
+/* -*- c-basic-offset: 4 indent-tabs-mode: nil -*-  vi:set ts=8 sts=4 sw=4: */
+
+/*
+    Rubber Band
+    An audio time-stretching and pitch-shifting library.
+    Copyright 2007 Chris Cannam.
+    
+    This program is free software; you can redistribute it and/or
+    modify it under the terms of the GNU General Public License as
+    published by the Free Software Foundation; either version 2 of the
+    License, or (at your option) any later version.  See the file
+    COPYING included with this distribution for more information.
+*/
+
+#ifndef _RUBBERBAND_STRETCHERIMPL_H_
+#define _RUBBERBAND_STRETCHERIMPL_H_
+
+#include "RubberBandStretcher.h"
+
+#include "Window.h"
+#include "Thread.h"
+#include "RingBuffer.h"
+#include "FFT.h"
+#include "sysutils.h"
+
+#include <set>
+
+namespace RubberBand
+{
+
+class AudioCurve;
+class StretchCalculator;
+
+class RubberBandStretcher::Impl
+{
+public:
+    Impl(RubberBandStretcher *stretcher,
+         size_t sampleRate, size_t channels, Options options,
+         double initialTimeRatio, double initialPitchScale);
+    ~Impl();
+    
+    void reset();
+    void setTimeRatio(double ratio);
+    void setPitchScale(double scale);
+
+    double getTimeRatio() const;
+    double getPitchScale() const;
+
+    size_t getLatency() const;
+
+    void setTransientsOption(Options);
+    void setPhaseOption(Options);
+
+    void setExpectedInputDuration(size_t samples);
+    void setMaxProcessBlockSize(size_t samples);
+
+    size_t getSamplesRequired() const;
+
+    void study(const float *const *input, size_t samples, bool final);
+    void process(const float *const *input, size_t samples, bool final);
+
+    int available() const;
+    size_t retrieve(float *const *output, size_t samples) const;
+
+    float getFrequencyCutoff(int n) const;
+    void setFrequencyCutoff(int n, float f);
+
+    size_t getInputIncrement() const {
+        return m_increment;
+    }
+
+    std::vector<int> getOutputIncrements() const;
+    std::vector<float> getLockCurve() const;
+
+    size_t getChannelCount() const {
+        return m_channels;
+    }
+    
+    void calculateStretch();
+
+    void setDebugLevel(int level);
+
+protected:
+    RubberBandStretcher *m_stretcher;
+    size_t m_channels;
+
+    size_t consumeChannel(size_t channel, const float *input, size_t samples);
+    bool processChunks(size_t channel); // returns "last"
+    bool processOneChunk(); // across all channels, for real time use
+    bool processChunkForChannel(size_t channel, size_t phaseIncrement,
+                                size_t shiftIncrement, bool lock);
+    bool testInbufReadSpace(size_t channel);
+    void calculateIncrements(size_t &phaseIncrement,
+                             size_t &shiftIncrement, bool &lock);
+    bool getIncrements(size_t channel, size_t &phaseIncrement,
+                       size_t &shiftIncrement, bool &lock);
+    void analyseChunk(size_t channel);
+    void modifyChunk(size_t channel, size_t outputIncrement, bool lock);
+    void synthesiseChunk(size_t channel);
+    void writeChunk(size_t channel, size_t shiftIncrement, bool last);
+
+    void calculateSizes();
+    void configure();
+    void reconfigure();
+
+    double getEffectiveRatio() const;
+    
+    size_t roundUp(size_t value); // to next power of two
+
+    double m_timeRatio;
+    double m_pitchScale;
+
+    size_t m_blockSize;
+    size_t m_outbufSize;
+    size_t m_increment;
+
+    size_t m_maxProcessBlockSize;
+    size_t m_expectedInputDuration;
+    
+    bool m_threaded;
+    bool m_realtime;
+    Options m_options;
+    int m_debugLevel;
+
+    enum ProcessMode {
+        JustCreated,
+        Studying,
+        Processing,
+        Finished
+    };
+
+    ProcessMode m_mode;
+
+    std::map<size_t, Window<float> *> m_windows;
+    Window<float> *m_window;
+    FFT *m_studyFFT;
+
+    Condition m_dataAvailable;
+    Condition m_spaceAvailable;
+    
+    class ProcessThread : public Thread
+    {
+    public:
+        ProcessThread(Impl *s, size_t c) : m_s(s), m_channel(c) { }
+        void run();
+    private:
+        Impl *m_s;
+        size_t m_channel;
+    };
+
+    mutable Mutex m_threadSetMutex;
+    std::set<ProcessThread *> m_threadSet;
+    
+
+    size_t m_inputDuration;
+    std::vector<float> m_lockDf;
+    std::vector<float> m_stretchDf;
+
+    class ChannelData; 
+    std::vector<ChannelData *> m_channelData;
+
+    std::vector<int> m_outputIncrements;
+
+    mutable RingBuffer<int> m_lastProcessOutputIncrements;
+    mutable RingBuffer<float> m_lastProcessLockDf;
+
+    AudioCurve *m_lockAudioCurve;
+    AudioCurve *m_stretchAudioCurve;
+    StretchCalculator *m_stretchCalculator;
+
+    float m_freq0;
+    float m_freq1;
+    float m_freq2;
+
+    size_t m_baseBlockSize;
+
+    void writeOutput(RingBuffer<float> &to, float *from,
+                     size_t qty, size_t &outCount, size_t theoreticalOut);
+
+};
+
+}
+
+#endif

src/StretcherProcess.cpp

@@ -0,0 +1,853 @@
+/* -*- c-basic-offset: 4 indent-tabs-mode: nil -*-  vi:set ts=8 sts=4 sw=4: */
+
+/*
+    Rubber Band
+    An audio time-stretching and pitch-shifting library.
+    Copyright 2007 Chris Cannam.
+    
+    This program is free software; you can redistribute it and/or
+    modify it under the terms of the GNU General Public License as
+    published by the Free Software Foundation; either version 2 of the
+    License, or (at your option) any later version.  See the file
+    COPYING included with this distribution for more information.
+*/
+
+#include "StretcherImpl.h"
+#include "PercussiveAudioCurve.h"
+#include "HighFrequencyAudioCurve.h"
+#include "ConstantAudioCurve.h"
+#include "StretchCalculator.h"
+#include "StretcherChannelData.h"
+#include "Resampler.h"
+
+#include <cassert>
+#include <cmath>
+#include <set>
+#include <map>
+
+using std::cerr;
+using std::endl;
+
+namespace RubberBand {
+
+void
+RubberBandStretcher::Impl::ProcessThread::run()
+{
+    if (m_s->m_debugLevel > 1) {
+        cerr << "thread " << m_channel << " getting going" << endl;
+    }
+
+    ChannelData &cd = *m_s->m_channelData[m_channel];
+
+    while (cd.inputSize == -1 ||
+           cd.inbuf->getReadSpace() > 0) {
+
+        if (cd.inputSize != -1) {
+            cerr << "inputSize == " << cd.inputSize
+                 << ", readSpace == " << cd.inbuf->getReadSpace() << endl;
+        }
+        
+        bool last = m_s->processChunks(m_channel);
+
+        m_s->m_spaceAvailable.signal();
+
+        if (last) break;
+
+        m_s->m_dataAvailable.lock();
+        if (cd.inbuf->getReadSpace() == 0) {
+            m_s->m_dataAvailable.wait();
+        }
+    }
+
+    m_s->processChunks(m_channel);
+    m_s->m_spaceAvailable.signal();
+    
+    if (m_s->m_debugLevel > 1) {
+        cerr << "thread " << m_channel << " done" << endl;
+    }
+}
+
+bool
+RubberBandStretcher::Impl::processChunks(size_t c)
+{
+    // Process as many chunks as there are available on the input
+    // buffer for channel c.  This requires that the increments have
+    // already been calculated.
+
+    ChannelData &cd = *m_channelData[c];
+
+    bool last = false;
+
+    while (!last) {
+
+        if (!testInbufReadSpace(c)) break;
+
+        if (!cd.draining) {
+            size_t got = cd.inbuf->peek(cd.fltbuf, m_blockSize);
+            assert(got == m_blockSize || cd.inputSize >= 0);
+            cd.inbuf->skip(m_increment);
+            analyseChunk(c);
+        }
+
+        bool lock = false;
+        size_t phaseIncrement, shiftIncrement;
+        getIncrements(c, phaseIncrement, shiftIncrement, lock);
+
+        last = processChunkForChannel(c, phaseIncrement, shiftIncrement, lock);
+        cd.blockCount++;
+        if (m_debugLevel > 2) {
+            cerr << "channel " << c << ": last = " << last << ", blockCount = " << cd.blockCount << endl;
+        }
+    }
+
+    return last;
+}
+
+bool
+RubberBandStretcher::Impl::processOneChunk()
+{
+    // Process a single chunk for all channels, provided there is
+    // enough data on each channel for at least one chunk.  This is
+    // able to calculate increments as it goes along.
+
+    for (size_t c = 0; c < m_channels; ++c) {
+        if (!testInbufReadSpace(c)) return false;
+        ChannelData &cd = *m_channelData[c];
+        if (!cd.draining) {
+            size_t got = cd.inbuf->peek(cd.fltbuf, m_blockSize);
+            assert(got == m_blockSize || cd.inputSize >= 0);
+            cd.inbuf->skip(m_increment);
+            analyseChunk(c);
+        }
+    }
+    
+    bool lock = false;
+    size_t phaseIncrement, shiftIncrement;
+    if (!getIncrements(0, phaseIncrement, shiftIncrement, lock)) {
+        calculateIncrements(phaseIncrement, shiftIncrement, lock);
+    }
+
+    bool last = false;
+    for (size_t c = 0; c < m_channels; ++c) {
+        last = processChunkForChannel(c, phaseIncrement, shiftIncrement, lock);
+        m_channelData[c]->blockCount++;
+    }
+
+    return last;
+}
+
+bool
+RubberBandStretcher::Impl::testInbufReadSpace(size_t c)
+{
+    ChannelData &cd = *m_channelData[c];
+    RingBuffer<float> &inbuf = *cd.inbuf;
+
+    size_t rs = inbuf.getReadSpace();
+
+    if (rs < m_blockSize && !cd.draining) {
+            
+        if (cd.inputSize == -1) {
+
+            // Not all the input data has been written to the inbuf
+            // (that's why the input size is not yet set).  We can't
+            // process, because we don't have a full block of data, so
+            // our process block would contain some empty padding in
+            // its input -- and that would give incorrect output, as
+            // we know there is more input to come.
+
+            if (!m_threaded) {
+//                cerr << "WARNING: RubberBandStretcher: read space < block size ("
+//                          << inbuf.getReadSpace() << " < " << m_blockSize
+//                          << ") when not all input written, on processChunks for channel " << c << endl;
+            }
+            return false;
+        }
+        
+        if (rs == 0) {
+
+            if (m_debugLevel > 1) {
+                cerr << "read space = 0, giving up" << endl;
+            }
+            return false;
+
+        } else if (rs < m_blockSize/2) {
+
+            if (m_debugLevel > 1) {
+                cerr << "read space = " << rs << ", setting draining true" << endl;
+            }
+            
+            cd.draining = true;
+        }
+    }
+
+    return true;
+}
+
+bool 
+RubberBandStretcher::Impl::processChunkForChannel(size_t c,
+                                                  size_t phaseIncrement,
+                                                  size_t shiftIncrement,
+                                                  bool lock)
+{
+    // Process a single chunk on a single channel.  This assumes
+    // enough input data is available; caller must have tested this
+    // using e.g. testInbufReadSpace first.  Return true if this is
+    // the last chunk on the channel.
+
+    if (lock && (m_debugLevel > 1)) {
+        cerr << "processChunkForChannel: lock found, incrs "
+                  << phaseIncrement << ":" << shiftIncrement << endl;
+    }
+
+    ChannelData &cd = *m_channelData[c];
+
+    if (!cd.draining) {
+        
+        // This is the normal processing case -- draining is only
+        // set when all the input has been used and we only need
+        // to write from the existing accumulator into the output.
+        
+        // We know we have enough samples available in m_inbuf --
+        // this is usually m_blockSize, but we know that if fewer
+        // are available, it's OK to use zeroes for the rest
+        // (which the ring buffer will provide) because we've
+        // reached the true end of the data.
+        
+        // We need to peek m_blockSize samples for processing, and
+        // then skip m_increment to advance the read pointer.
+        
+        modifyChunk(c, phaseIncrement, lock);
+        synthesiseChunk(c); // reads from cd.mag, cd.phase
+
+        if (m_debugLevel > 2) {
+            if (lock) {
+                for (int i = 0; i < 10; ++i) {
+                    cd.accumulator[i] = ((drand48() * 2 - 1.0) > 0 ? 1 : -1);
+                }
+            }
+        }
+    }
+
+    bool last = false;
+
+    if (cd.draining) {
+        if (m_debugLevel > 1) {
+            cerr << "draining: accumulator fill = " << cd.accumulatorFill << " (shiftIncrement = " << shiftIncrement << ")" <<  endl;
+        }
+        if (shiftIncrement == 0) {
+            cerr << "WARNING: draining: shiftIncrement == 0, can't handle that in this context: setting to " << m_increment << endl;
+            shiftIncrement = m_increment;
+        }
+        if (cd.accumulatorFill <= shiftIncrement) {
+            if (m_debugLevel > 1) {
+                cerr << "reducing shift increment from " << shiftIncrement
+                          << " to " << cd.accumulatorFill
+                          << " and marking as last" << endl;
+            }
+            shiftIncrement = cd.accumulatorFill;
+            last = true;
+        }
+    }
+        
+    if (m_threaded) {
+        size_t required = shiftIncrement;
+        if (m_pitchScale != 1.0) {
+            required = int(required / m_pitchScale) + 1;
+        }
+        
+        if (cd.outbuf->getWriteSpace() < required) {
+            if (m_debugLevel > 0) {
+                cerr << "buffer overrun on output for channel " << c << endl;
+            }
+
+            //!!! The only correct thing we can do here is resize the
+            // buffer.  We can't wait for the client thread to read
+            // some data out from the buffer so as to make more space,
+            // because the client thread is probably stuck in a
+            // process() call waiting for us to stow away enough input
+            // increments to allow the process() call to complete.
+
+        }
+    }
+    
+    writeChunk(c, shiftIncrement, last);
+    return last;
+}
+
+void
+RubberBandStretcher::Impl::calculateIncrements(size_t &phaseIncrementRtn,
+                                               size_t &shiftIncrementRtn,
+                                               bool &lock)
+{
+//    cerr << "calculateIncrements" << endl;
+    
+    // Calculate the next upcoming phase and shift increment, on the
+    // basis that both channels are in sync.  This is in contrast to
+    // getIncrements, which requires that all the increments have been
+    // calculated in advance but can then return increments
+    // corresponding to different blocks in different channels.
+
+    // Requires frequency domain representations of channel data in
+    // the mag and phase buffers in the channel.
+
+    // This function is only used in real-time mode.
+
+    phaseIncrementRtn = m_increment;
+    shiftIncrementRtn = m_increment;
+    lock = false;
+
+    if (m_channels == 0) return;
+
+    ChannelData &cd = *m_channelData[0];
+
+    size_t bc = cd.blockCount;
+    for (size_t c = 1; c < m_channels; ++c) {
+        if (m_channelData[c]->blockCount != bc) {
+            cerr << "ERROR: RubberBandStretcher::Impl::calculateIncrements: Channels are not in sync" << endl;
+            return;
+        }
+    }
+
+    // Normally we would mix down the time-domain signal and apply a
+    // single FFT, or else mix down the Cartesian form of the
+    // frequency-domain signal.  Both of those would be inefficient
+    // from this position.  Fortunately, the onset detectors should
+    // work reasonably well (maybe even better?) if we just sum the
+    // magnitudes of the frequency-domain channel signals and forget
+    // about phase entirely.  Normally we don't expect the channel
+    // phases to cancel each other, and broadband effects will still
+    // be apparent.
+
+    for (size_t i = 0; i <= m_blockSize/2; ++i) {
+        cd.fltbuf[i] = 0.0;
+    }
+
+    for (size_t c = 0; c < m_channels; ++c) {
+        for (size_t i = 0; i <= m_blockSize/2; ++i) {
+            cd.fltbuf[i] += m_channelData[c]->mag[i];
+        }
+    }
+    
+    float df = m_lockAudioCurve->process(cd.fltbuf, m_increment);
+
+    int incr = m_stretchCalculator->calculateSingle
+        (getEffectiveRatio(),
+         m_inputDuration, //!!! no, totally wrong... fortunately it doesn't matter atm
+         df);
+
+    m_lastProcessLockDf.write(&df, 1);
+    m_lastProcessOutputIncrements.write(&incr, 1);
+
+    if (incr < 0) {
+        lock = true;
+        incr = -incr;
+    }
+    
+    // The returned increment is the phase increment.  The shift
+    // increment for one chunk is the same as the phase increment for
+    // the following chunk (see comment below).  This means we don't
+    // actually know the shift increment until we see the following
+    // phase increment... which is a bit of a problem.
+
+    // This implies we should use this increment for the shift
+    // increment, and make the following phase increment the same as
+    // it.  This means in RT mode we'll be one block later with our
+    // phase lock than we would be in non-RT mode.  The sensitivity of
+    // the broadband onset detector may mean that this isn't a problem
+    // -- test it and see.
+
+    shiftIncrementRtn = incr;
+
+    if (cd.prevIncrement == 0) {
+        phaseIncrementRtn = shiftIncrementRtn;
+    } else {
+        phaseIncrementRtn = cd.prevIncrement;
+    }
+
+    cd.prevIncrement = shiftIncrementRtn;
+}
+
+bool
+RubberBandStretcher::Impl::getIncrements(size_t channel,
+                                         size_t &phaseIncrementRtn,
+                                         size_t &shiftIncrementRtn,
+                                         bool &lock)
+{
+    if (channel >= m_channels) {
+        phaseIncrementRtn = m_increment;
+        shiftIncrementRtn = m_increment;
+        lock = false;
+        return false;
+    }
+
+    // There are two relevant output increments here.  The first is
+    // the phase increment which we use when recalculating the phases
+    // for the current chunk; the second is the shift increment used
+    // to determine how far to shift the processing buffer after
+    // writing the chunk.  The shift increment for one chunk is the
+    // same as the phase increment for the following chunk.
+    
+    // When an onset occurs for which we need to lock phases, the
+    // increment given will be negative.
+    
+    // When we lock phases, the previous shift increment (and so
+    // current phase increments) must have been m_increment to ensure
+    // consistency.
+    
+    // m_outputIncrements stores phase increments.
+
+    ChannelData &cd = *m_channelData[channel];
+    bool gotData = true;
+
+    if (cd.blockCount >= m_outputIncrements.size()) {
+        //!!! this is an error if in non-realtime mode
+//        cerr << "*** ERROR: block count " << cd.blockCount << " >= "
+//                  << m_outputIncrements.size() << endl;
+        if (m_outputIncrements.size() == 0) {
+            phaseIncrementRtn = m_increment;
+            shiftIncrementRtn = m_increment;
+            lock = false;
+            return false;
+        } else {
+            cd.blockCount = m_outputIncrements.size()-1;
+            gotData = false;
+        }
+    }
+    
+    int phaseIncrement = m_outputIncrements[cd.blockCount];
+    
+    int shiftIncrement = phaseIncrement;
+    if (cd.blockCount + 1 < m_outputIncrements.size()) {
+        shiftIncrement = m_outputIncrements[cd.blockCount + 1];
+    }
+    
+    if (phaseIncrement < 0) {
+        phaseIncrement = -phaseIncrement;
+        lock = true;
+    }
+    
+    if (shiftIncrement < 0) {
+        shiftIncrement = -shiftIncrement;
+    }
+    
+    if (shiftIncrement >= int(m_blockSize)) {
+        cerr << "*** ERROR: RubberBandStretcher::Impl::processChunks: shiftIncrement " << shiftIncrement << " >= blockSize " << m_blockSize << " at " << cd.blockCount << " (of " << m_outputIncrements.size() << ")" << endl;
+        shiftIncrement = m_blockSize;
+    }
+
+    phaseIncrementRtn = phaseIncrement;
+    shiftIncrementRtn = shiftIncrement;
+    if (cd.blockCount == 0) lock = true; // don't mess with the first block
+    return gotData;
+}
+
+void
+RubberBandStretcher::Impl::analyseChunk(size_t channel)
+{
+    size_t i;
+
+    ChannelData &cd = *m_channelData[channel];
+
+    // cd.fltbuf is known to contain m_blockSize samples
+
+    m_window->cut(cd.fltbuf);
+
+    for (i = 0; i < m_blockSize/2; ++i) {
+	cd.dblbuf[i] = cd.fltbuf[i + m_blockSize/2];
+	cd.dblbuf[i + m_blockSize/2] = cd.fltbuf[i];
+    }
+
+    cd.fft->forwardPolar(cd.dblbuf, cd.mag, cd.phase);
+}
+
+double mod(double x, double y) { return x - (y * floor(x / y)); }
+double princarg(double a) { return mod(a + M_PI, -2 * M_PI) + M_PI; }
+
+void
+RubberBandStretcher::Impl::modifyChunk(size_t channel, size_t outputIncrement,
+                                       bool lock)
+{
+    ChannelData &cd = *m_channelData[channel];
+
+    if (lock && m_debugLevel > 1) {
+        cerr << "lock: leaving phases unmodified" << endl;
+    }
+
+    size_t count = m_blockSize/2;
+    size_t pfp = 0;
+    double rate = m_stretcher->m_sampleRate;
+
+    if (!(m_options & OptionPhaseIndependent)) {
+
+        cd.freqPeak[0] = 0;
+
+        size_t limit0 = lrint((m_freq0 * m_blockSize) / rate);
+        size_t limit1 = lrint((m_freq1 * m_blockSize) / rate);
+        size_t limit2 = lrint((m_freq2 * m_blockSize) / rate);
+        size_t range = 0;
+
+        if (limit1 < limit0) limit1 = limit0;
+        if (limit2 < limit1) limit2 = limit1;
+    
+//        cerr << "limit0 = " << limit0 << " limit1 = " << limit1 << " limit2 = " << limit2 << endl;
+
+        int peakCount = 0;
+
+        for (size_t i = 0; i <= count; ++i) {
+
+            double mag = cd.mag[i];
+
+            //!!! N.B. if the stretch ratio is very long, it's generally
+            //better not to attempt this phase lamination -- stick with
+            //range==0 throughout.
+
+            bool isPeak = true;
+
+            for (size_t j = 1; j <= range; ++j) {
+
+                if (mag < cd.mag[i-j]) {
+                    isPeak = false;
+                    break;
+                }
+
+                if (mag < cd.mag[i+j]) {
+                    isPeak = false;
+                    break;
+                }
+            }        
+
+            if (isPeak) {
+
+                // i is a peak bin.
+
+                // The previous peak bin was at pfp; make freqPeak entries
+                // from pfp to half-way between pfp and i point at pfp, and
+                // those from the half-way mark to i point at i.
+            
+                size_t halfway = (pfp + i) / 2;
+                if (halfway == pfp) halfway = pfp + 1;
+
+                for (size_t j = pfp + 1; j < halfway; ++j) {
+                    cd.freqPeak[j] = pfp;
+                }
+                for (size_t j = halfway; j <= i; ++j) {
+                    cd.freqPeak[j] = i;
+                }
+
+                pfp = i;
+
+                ++peakCount;
+            }
+
+            if (i == limit0) range = 1;
+            if (i == limit1) range = 2;
+            if (i >= limit2) {
+                range = 3;
+                if (i + range + 1 > count) range = count - i;
+            }
+        }
+
+//        cerr << "peakCount = " << peakCount << endl;
+        
+        cd.freqPeak[count-1] = count-1;
+        cd.freqPeak[count] = count;
+    }
+
+    double peakInPhase = 0.0;
+    double peakOutPhase = 0.0;
+    size_t p, pp;
+
+    for (size_t i = 0; i <= count; ++i) {
+        
+        if (m_options & OptionPhaseIndependent) {
+            p = i;
+            pp = i-1;
+        } else {
+            p = cd.freqPeak[i];
+            pp = cd.freqPeak[i-1];
+        }
+
+        if (!lock) {
+
+            if (i == 0 || p != pp) {
+	
+                double omega = (2 * M_PI * m_increment * p) / m_blockSize;
+                double expectedPhase = cd.prevPhase[p] + omega;
+                double phaseError = princarg(cd.phase[p] - expectedPhase);
+                double phaseIncrement = (omega + phaseError) / m_increment;
+            
+                double unwrappedPhase = cd.unwrappedPhase[p] +
+                    outputIncrement * phaseIncrement;
+
+                cd.prevPhase[p] = cd.phase[p];
+                cd.phase[p] = unwrappedPhase;
+                cd.unwrappedPhase[p] = unwrappedPhase;
+
+                peakInPhase = cd.prevPhase[p];
+                peakOutPhase = unwrappedPhase;
+            }
+
+            if (i != p) {
+
+                double diffToPeak = peakInPhase - cd.phase[i];
+                double unwrappedPhase = peakOutPhase - diffToPeak;
+                
+                cd.prevPhase[i] = cd.phase[i];
+                cd.phase[i] = unwrappedPhase;
+                cd.unwrappedPhase[i] = unwrappedPhase;
+            }
+
+        } else {
+            cd.prevPhase[i] = cd.phase[i];
+            cd.unwrappedPhase[i] = cd.phase[i];
+        }
+    }
+}
+
+void
+RubberBandStretcher::Impl::synthesiseChunk(size_t channel)
+{
+    ChannelData &cd = *m_channelData[channel];
+
+    cd.fft->inversePolar(cd.mag, cd.phase, cd.dblbuf);
+
+    for (size_t i = 0; i < m_blockSize/2; ++i) {
+        cd.fltbuf[i] = cd.dblbuf[i + m_blockSize/2];
+        cd.fltbuf[i + m_blockSize/2] = cd.dblbuf[i];
+    }
+
+    // our ffts produced unscaled results
+    for (size_t i = 0; i < m_blockSize; ++i) {
+        cd.fltbuf[i] = cd.fltbuf[i] / m_blockSize;
+    }
+
+    m_window->cut(cd.fltbuf);
+
+    for (size_t i = 0; i < m_blockSize; ++i) {
+        cd.accumulator[i] += cd.fltbuf[i];
+    }
+
+    cd.accumulatorFill = m_blockSize;
+
+    //!!! not much cop, this bit
+
+    float area = m_window->getArea();
+
+    for (size_t i = 0; i < m_blockSize; ++i) {
+        float val = m_window->getValue(i);
+        cd.windowAccumulator[i] += val * area;
+    }
+}
+
+void
+RubberBandStretcher::Impl::writeChunk(size_t channel, size_t shiftIncrement, bool last)
+{
+    ChannelData &cd = *m_channelData[channel];
+
+    if (m_debugLevel > 2) {
+        cerr << "writeChunk(" << channel << ", " << shiftIncrement << ", " << last << ")" << endl;
+    }
+
+    static float windowaccmax = 0; //!!!
+        
+    for (int i = 0; i < shiftIncrement; ++i) {
+        if (cd.windowAccumulator[i] > 0.f) {
+            cd.accumulator[i] /= cd.windowAccumulator[i];
+            if (fabsf(cd.windowAccumulator[i]) > windowaccmax) {
+//                cerr << "window accumulator max " << windowaccmax
+//                          << " -> " << fabsf(cd.windowAccumulator[i]) << endl;
+                //!!!
+                windowaccmax = fabsf(cd.windowAccumulator[i]);
+            }
+        }
+    }
+
+    // for exact sample scaling (probably not meaningful if we
+    // were running in RT mode)
+    size_t theoreticalOut = 0;
+    if (cd.inputSize >= 0) {
+//        cerr << "cd.inputSize = " << cd.inputSize << endl;
+        theoreticalOut = lrint(cd.inputSize * m_timeRatio);
+//        cerr << "theoreticalOut = " << theoreticalOut << endl;
+    }
+
+    if (m_pitchScale != 1.0 && cd.resampler) {
+//            cerr << "resampler: input " << shiftIncrement << ", output max " << int(shiftIncrement / m_pitchScale) + 1 << endl;
+
+        size_t reqSize = int(ceil(shiftIncrement / m_pitchScale));
+        if (reqSize > cd.resamplebufSize) {
+            // This shouldn't normally happen -- the buffer is
+            // supposed to be initialised with enough space in the
+            // first place.  But we retain this check in case the
+            // pitch scale has changed since then, or the stretch
+            // calculator has goneb mad, or something.
+            cerr << "resampler: resizing buffer from "
+                      << cd.resamplebufSize << " to " << reqSize << endl;
+            cd.resamplebufSize = reqSize;
+            if (cd.resamplebuf) delete[] cd.resamplebuf;
+            cd.resamplebuf = new float[cd.resamplebufSize];
+        }
+
+
+        size_t outframes = cd.resampler->resample(&cd.accumulator,
+                                                  &cd.resamplebuf,
+                                                  shiftIncrement,
+                                                  1.0 / m_pitchScale,
+                                                  last);
+
+
+        writeOutput(*cd.outbuf, cd.resamplebuf,
+                    outframes, cd.outCount, theoreticalOut);
+
+    } else {
+        writeOutput(*cd.outbuf, cd.accumulator,
+                    shiftIncrement, cd.outCount, theoreticalOut);
+    }
+    
+    for (size_t i = 0; i < m_blockSize - shiftIncrement; ++i) {
+        cd.accumulator[i] = cd.accumulator[i + shiftIncrement];
+    }
+    
+    for (size_t i = m_blockSize - shiftIncrement; i < m_blockSize; ++i) {
+        cd.accumulator[i] = 0.0f;
+    }
+    
+    for (size_t i = 0; i < m_blockSize - shiftIncrement; ++i) {
+        cd.windowAccumulator[i] = cd.windowAccumulator[i + shiftIncrement];
+    }
+    
+    for (size_t i = m_blockSize - shiftIncrement; i < m_blockSize; ++i) {
+        cd.windowAccumulator[i] = 0.0f;
+    }
+    
+    if (cd.accumulatorFill > shiftIncrement) {
+        cd.accumulatorFill -= shiftIncrement;
+    } else {
+        cd.accumulatorFill = 0;
+        if (cd.draining) {
+            if (m_debugLevel > 1) {
+                cerr << "RubberBandStretcher::Impl::processChunks: setting outputComplete to true" << endl;
+            }
+            cd.outputComplete = true;
+        }
+    }
+}
+
+void
+RubberBandStretcher::Impl::writeOutput(RingBuffer<float> &to, float *from, size_t qty, size_t &outCount, size_t theoreticalOut)
+{
+    // In non-RT mode, we don't want to write the first startSkip
+    // samples, because the first block is centred on the start of the
+    // output.  In RT mode we didn't apply any pre-padding in
+    // configure(), so we don't want to remove any here.
+
+    size_t startSkip = 0;
+    if (!m_realtime) {
+        startSkip = lrintf((m_blockSize/2) / m_pitchScale);
+    }
+
+    if (outCount > startSkip) {
+        if (theoreticalOut > 0) {
+            if (m_debugLevel > 1) {
+                cerr << "theoreticalOut = " << theoreticalOut << ", outCount = " << outCount << ", startSkip = " << startSkip << ", qty = " << qty << endl;
+            }
+            if (outCount - startSkip <= theoreticalOut &&
+                outCount - startSkip + qty > theoreticalOut) {
+                qty = theoreticalOut - (outCount - startSkip);
+                if (m_debugLevel > 1) {
+                    cerr << "reduce qty to " << qty << endl;
+                }
+            }
+        }
+
+        if (m_debugLevel > 2) {
+            cerr << "writing " << qty << endl;
+        }
+        to.write(from, qty);
+        outCount += qty;
+        return;
+    }
+
+    if (outCount + qty <= startSkip) {
+        if (m_debugLevel > 1) {
+            cerr << "qty = " << qty << ", startSkip = " << startSkip << ", outCount = " << outCount << ", discarding" << endl;
+        }
+        outCount += qty;
+        return;
+    }
+
+    size_t off = startSkip - outCount;
+    if (m_debugLevel > 1) {
+        cerr << "qty = " << qty << ", startSkip = " << startSkip << ", outCount = " << outCount << ", writing " << qty - off << " from start offset " << off << endl;
+    }
+    to.write(from + off, qty - off);
+    outCount += qty;
+}
+
+int
+RubberBandStretcher::Impl::available() const
+{
+    if (m_threaded) {
+        MutexLocker locker(&m_threadSetMutex);
+        if (m_channelData.empty()) return 0;
+    } else {
+        if (m_channelData.empty()) return 0;
+    }
+
+    if (!m_threaded) {
+        for (size_t c = 0; c < m_channels; ++c) {
+            if (m_channelData[c]->inputSize >= 0) {
+                cerr << "available: m_done true" << endl;
+                if (m_channelData[c]->inbuf->getReadSpace() > 0) {
+                    cerr << "calling processChunks(" << c << ") from available" << endl;
+                    //!!! do we ever actually do this? if so, this method should not be const
+                    ((RubberBandStretcher::Impl *)this)->processChunks(c);
+                }
+            }
+        }
+    }
+
+    size_t min = 0;
+    bool consumed = true;
+    bool haveResamplers = false;
+
+    for (size_t i = 0; i < m_channels; ++i) {
+        size_t availIn = m_channelData[i]->inbuf->getReadSpace();
+        size_t availOut = m_channelData[i]->outbuf->getReadSpace();
+        if (m_debugLevel > 2) {
+            cerr << "available on channel " << i << ": " << availOut << " (waiting: " << availIn << ")" << endl;
+        }
+        if (i == 0 || availOut < min) min = availOut;
+        if (!m_channelData[i]->outputComplete) consumed = false;
+        if (m_channelData[i]->resampler) haveResamplers = true;
+    }
+
+    if (min == 0 && consumed) return -1;
+    if (m_pitchScale == 1.0) return min;
+
+    if (haveResamplers) return min; // resampling has already happened
+    return int(floor(min / m_pitchScale));
+}
+
+size_t
+RubberBandStretcher::Impl::retrieve(float *const *output, size_t samples) const
+{
+    size_t got = samples;
+
+    for (size_t c = 0; c < m_channels; ++c) {
+        size_t gotHere = m_channelData[c]->outbuf->read(output[c], got);
+        if (gotHere < got) {
+            if (c > 0) {
+                if (m_debugLevel > 0) {
+                    cerr << "RubberBandStretcher::Impl::retrieve: WARNING: channel imbalance detected" << endl;
+                }
+            }
+            got = gotHere;
+        }
+    }
+
+    return got;
+}
+
+}
+

src/Thread.cpp

@@ -0,0 +1,190 @@
+/* -*- c-basic-offset: 4 indent-tabs-mode: nil -*-  vi:set ts=8 sts=4 sw=4: */
+
+/*
+    Rubber Band
+    An audio time-stretching and pitch-shifting library.
+    Copyright 2007 Chris Cannam.
+    
+    This program is free software; you can redistribute it and/or
+    modify it under the terms of the GNU General Public License as
+    published by the Free Software Foundation; either version 2 of the
+    License, or (at your option) any later version.  See the file
+    COPYING included with this distribution for more information.
+*/
+
+#include "Thread.h"
+
+#include <iostream>
+
+#include <sys/time.h>
+#include <time.h>
+
+using std::cerr;
+using std::endl;
+
+namespace RubberBand
+{
+
+
+Thread::Thread() :
+    m_id(0),
+    m_extant(false)
+{
+}
+
+Thread::~Thread()
+{
+    if (m_extant) {
+        pthread_join(m_id, 0);
+    }
+}
+
+void
+Thread::start()
+{
+    if (pthread_create(&m_id, 0, staticRun, this)) {
+        //!!!
+        cerr << "ERROR: thread creation failed" << endl;
+        exit(1);
+    } else {
+        m_extant = true;
+    }
+}    
+
+void 
+Thread::wait()
+{
+    if (m_extant) {
+        pthread_join(m_id, 0);
+        m_extant = false;
+    }
+}
+
+Thread::Id
+Thread::id()
+{
+    return m_id;
+}
+
+bool
+Thread::threadingAvailable()
+{
+    return true;
+}
+
+void *
+Thread::staticRun(void *arg)
+{
+    Thread *thread = (Thread *)arg;
+    thread->run();
+    return 0;
+}
+
+Mutex::Mutex()
+{
+    pthread_mutex_init(&m_mutex, 0);
+}
+
+Mutex::~Mutex()
+{
+    pthread_mutex_destroy(&m_mutex);
+}
+
+void
+Mutex::lock()
+{
+    pthread_mutex_lock(&m_mutex);
+}
+
+void
+Mutex::unlock()
+{
+    pthread_mutex_unlock(&m_mutex);
+}
+
+bool
+Mutex::trylock()
+{
+    if (pthread_mutex_trylock(&m_mutex)) {
+        return false;
+    } else {
+        return true;
+    }
+}
+
+Condition::Condition()
+{
+    pthread_mutex_init(&m_mutex, 0);
+    m_locked = false;
+    pthread_cond_init(&m_condition, 0);
+}
+
+Condition::~Condition()
+{
+    if (m_locked) pthread_mutex_unlock(&m_mutex);
+    pthread_cond_destroy(&m_condition);
+    pthread_mutex_destroy(&m_mutex);
+}
+
+void
+Condition::lock()
+{
+    if (m_locked) return;
+    pthread_mutex_lock(&m_mutex);
+    m_locked = true;
+}
+
+void 
+Condition::wait(int us)
+{
+    lock();
+    if (us == 0) {
+
+        pthread_cond_wait(&m_condition, &m_mutex);
+
+    } else {
+
+        struct timeval now;
+        gettimeofday(&now, 0);
+
+        now.tv_usec += us;
+        while (now.tv_usec > 1000000) {
+            now.tv_usec -= 1000000;
+            ++now.tv_sec;
+        }
+
+        struct timespec timeout;
+        timeout.tv_sec = now.tv_sec;
+        timeout.tv_nsec = now.tv_usec * 1000;
+    
+        pthread_cond_timedwait(&m_condition, &m_mutex, &timeout);
+    }
+
+    pthread_mutex_unlock(&m_mutex);
+    m_locked = false;
+}
+
+void
+Condition::signal()
+{
+    pthread_cond_signal(&m_condition);
+}
+
+
+MutexLocker::MutexLocker(Mutex *mutex) :
+    m_mutex(mutex)
+{
+    if (m_mutex) {
+        m_mutex->lock();
+    }
+}
+
+MutexLocker::~MutexLocker()
+{
+    if (m_mutex) {
+        m_mutex->unlock();
+    }
+}
+
+}
+

src/Thread.h

@@ -0,0 +1,89 @@
+/* -*- c-basic-offset: 4 indent-tabs-mode: nil -*-  vi:set ts=8 sts=4 sw=4: */
+
+/*
+    Rubber Band
+    An audio time-stretching and pitch-shifting library.
+    Copyright 2007 Chris Cannam.
+    
+    This program is free software; you can redistribute it and/or
+    modify it under the terms of the GNU General Public License as
+    published by the Free Software Foundation; either version 2 of the
+    License, or (at your option) any later version.  See the file
+    COPYING included with this distribution for more information.
+*/
+
+#ifndef _RUBBERBAND_THREAD_H_
+#define _RUBBERBAND_THREAD_H_
+
+#include <pthread.h>
+
+namespace RubberBand
+{
+
+class Thread
+{
+public:
+    typedef pthread_t Id;
+
+    Thread();
+    virtual ~Thread();
+
+    Id id();
+
+    void start();
+    void wait();
+
+    static bool threadingAvailable();
+
+protected:
+    virtual void run() = 0;
+
+private:
+    pthread_t m_id;
+    bool m_extant;
+    static void *staticRun(void *);
+};
+
+class Mutex
+{
+public:
+    Mutex();
+    ~Mutex();
+
+    void lock();
+    void unlock();
+    bool trylock();
+
+private:
+    pthread_mutex_t m_mutex;
+};
+
+class MutexLocker
+{
+public:
+    MutexLocker(Mutex *);
+    ~MutexLocker();
+
+private:
+    Mutex *m_mutex;
+};
+
+class Condition
+{
+public:
+    Condition();
+    ~Condition();
+    
+    void lock();
+    void wait(int us = 0);
+    void signal();
+    
+private:
+    pthread_mutex_t m_mutex;
+    bool m_locked;
+    pthread_cond_t m_condition;
+};
+
+}
+
+#endif

src/Window.h

@@ -0,0 +1,161 @@
+/* -*- c-basic-offset: 4 indent-tabs-mode: nil -*-  vi:set ts=8 sts=4 sw=4: */
+
+/*
+    Rubber Band
+    An audio time-stretching and pitch-shifting library.
+    Copyright 2007 Chris Cannam.
+    
+    This program is free software; you can redistribute it and/or
+    modify it under the terms of the GNU General Public License as
+    published by the Free Software Foundation; either version 2 of the
+    License, or (at your option) any later version.  See the file
+    COPYING included with this distribution for more information.
+*/
+
+#ifndef _RUBBERBAND_WINDOW_H_
+#define _RUBBERBAND_WINDOW_H_
+
+#include <cmath>
+#include <iostream>
+#include <map>
+
+enum WindowType {
+    RectangularWindow,
+    BartlettWindow,
+    HammingWindow,
+    HanningWindow,
+    BlackmanWindow,
+    GaussianWindow,
+    ParzenWindow,
+    NuttallWindow,
+    BlackmanHarrisWindow
+};
+
+template <typename T>
+class Window
+{
+public:
+    /**
+     * Construct a windower of the given type.
+     */
+    Window(WindowType type, size_t size) : m_type(type), m_size(size) { encache(); }
+    Window(const Window &w) : m_type(w.m_type), m_size(w.m_size) { encache(); }
+    Window &operator=(const Window &w) {
+	if (&w == this) return *this;
+	m_type = w.m_type;
+	m_size = w.m_size;
+	encache();
+	return *this;
+    }
+    virtual ~Window() { delete[] m_cache; }
+    
+    void cut(T *src) const { cut(src, src); }
+    void cut(T *src, T *dst) const {
+	for (size_t i = 0; i < m_size; ++i) dst[i] = src[i] * m_cache[i];
+    }
+
+    T getArea() { return m_area; }
+    T getValue(size_t i) { return m_cache[i]; }
+
+    WindowType getType() const { return m_type; }
+    size_t getSize() const { return m_size; }
+
+protected:
+    WindowType m_type;
+    size_t m_size;
+    T *m_cache;
+    T m_area;
+    
+    void encache();
+    void cosinewin(T *, T, T, T, T);
+};
+
+template <typename T>
+void Window<T>::encache()
+{
+    int n = int(m_size);
+    T *mult = new T[n];
+    int i;
+    for (i = 0; i < n; ++i) mult[i] = 1.0;
+
+    switch (m_type) {
+		
+    case RectangularWindow:
+	for (i = 0; i < n; ++i) {
+	    mult[i] *= 0.5;
+	}
+	break;
+	    
+    case BartlettWindow:
+	for (i = 0; i < n/2; ++i) {
+	    mult[i] *= (i / T(n/2));
+	    mult[i + n/2] *= (1.0 - (i / T(n/2)));
+	}
+	break;
+	    
+    case HammingWindow:
+        cosinewin(mult, 0.54, 0.46, 0.0, 0.0);
+	break;
+	    
+    case HanningWindow:
+        cosinewin(mult, 0.50, 0.50, 0.0, 0.0);
+	break;
+	    
+    case BlackmanWindow:
+        cosinewin(mult, 0.42, 0.50, 0.08, 0.0);
+	break;
+	    
+    case GaussianWindow:
+	for (i = 0; i < n; ++i) {
+            mult[i] *= pow(2, - pow((i - (n-1)/2.0) / ((n-1)/2.0 / 3), 2));
+	}
+	break;
+	    
+    case ParzenWindow:
+    {
+        int N = n-1;
+        for (i = 0; i < N/4; ++i) {
+            T m = 2 * pow(1.0 - (T(N)/2 - i) / (T(N)/2), 3);
+            mult[i] *= m;
+            mult[N-i] *= m;
+        }
+        for (i = N/4; i <= N/2; ++i) {
+            int wn = i - N/2;
+            T m = 1.0 - 6 * pow(wn / (T(N)/2), 2) * (1.0 - abs(wn) / (T(N)/2));
+            mult[i] *= m;
+            mult[N-i] *= m;
+        }            
+        break;
+    }
+
+    case NuttallWindow:
+        cosinewin(mult, 0.3635819, 0.4891775, 0.1365995, 0.0106411);
+	break;
+
+    case BlackmanHarrisWindow:
+        cosinewin(mult, 0.35875, 0.48829, 0.14128, 0.01168);
+        break;
+    }
+	
+    m_cache = mult;
+
+    m_area = 0;
+    for (int i = 0; i < n; ++i) {
+        m_area += m_cache[i];
+    }
+    m_area /= n;
+}
+
+template <typename T>
+void Window<T>::cosinewin(T *mult, T a0, T a1, T a2, T a3)
+{
+    int n = int(m_size);
+    for (int i = 0; i < n; ++i) {
+        mult[i] *= (a0
+                    - a1 * cos(2 * M_PI * i / n)
+                    + a2 * cos(4 * M_PI * i / n)
+                    - a3 * cos(6 * M_PI * i / n));
+    }
+}
+
+#endif

src/ladspa/RubberBandPitchShifter.cpp

@@ -0,0 +1,358 @@
+/* -*- c-basic-offset: 4 indent-tabs-mode: nil -*-  vi:set ts=8 sts=4 sw=4: */
+
+/*
+    Rubber Band
+    An audio time-stretching and pitch-shifting library.
+    Copyright 2007 Chris Cannam.
+    
+    This program is free software; you can redistribute it and/or
+    modify it under the terms of the GNU General Public License as
+    published by the Free Software Foundation; either version 2 of the
+    License, or (at your option) any later version.  See the file
+    COPYING included with this distribution for more information.
+*/
+
+#include "RubberBandPitchShifter.h"
+
+#include "RubberBandStretcher.h"
+
+#include <iostream>
+#include <cmath>
+
+const char *const
+RubberBandPitchShifter::portNamesMono[PortCountMono] =
+{
+    "_latency",
+    "Cents",
+    "Semitones",
+    "Octaves",
+    "Input",
+    "Output"
+};
+
+const char *const
+RubberBandPitchShifter::portNamesStereo[PortCountStereo] =
+{
+    "_latency",
+    "Cents",
+    "Semitones",
+    "Octaves",
+    "Input L",
+    "Output L",
+    "Input R",
+    "Output R"
+};
+
+const LADSPA_PortDescriptor 
+RubberBandPitchShifter::portsMono[PortCountMono] =
+{
+    LADSPA_PORT_OUTPUT | LADSPA_PORT_CONTROL,
+    LADSPA_PORT_INPUT  | LADSPA_PORT_CONTROL,
+    LADSPA_PORT_INPUT  | LADSPA_PORT_CONTROL,
+    LADSPA_PORT_INPUT  | LADSPA_PORT_CONTROL,
+    LADSPA_PORT_INPUT  | LADSPA_PORT_AUDIO,
+    LADSPA_PORT_OUTPUT | LADSPA_PORT_AUDIO
+};
+
+const LADSPA_PortDescriptor 
+RubberBandPitchShifter::portsStereo[PortCountStereo] =
+{
+    LADSPA_PORT_OUTPUT | LADSPA_PORT_CONTROL,
+    LADSPA_PORT_INPUT  | LADSPA_PORT_CONTROL,
+    LADSPA_PORT_INPUT  | LADSPA_PORT_CONTROL,
+    LADSPA_PORT_INPUT  | LADSPA_PORT_CONTROL,
+    LADSPA_PORT_INPUT  | LADSPA_PORT_AUDIO,
+    LADSPA_PORT_OUTPUT | LADSPA_PORT_AUDIO,
+    LADSPA_PORT_INPUT  | LADSPA_PORT_AUDIO,
+    LADSPA_PORT_OUTPUT | LADSPA_PORT_AUDIO
+};
+
+const LADSPA_PortRangeHint 
+RubberBandPitchShifter::hintsMono[PortCountMono] =
+{
+    { 0, 0, 0 },
+    { LADSPA_HINT_DEFAULT_0 |
+      LADSPA_HINT_BOUNDED_BELOW |
+      LADSPA_HINT_BOUNDED_ABOVE,
+      -100.0, 100.0 },
+    { LADSPA_HINT_DEFAULT_0 |
+      LADSPA_HINT_BOUNDED_BELOW |
+      LADSPA_HINT_BOUNDED_ABOVE |
+      LADSPA_HINT_INTEGER,
+      -12.0, 12.0 },
+    { LADSPA_HINT_DEFAULT_0 |
+      LADSPA_HINT_BOUNDED_BELOW |
+      LADSPA_HINT_BOUNDED_ABOVE |
+      LADSPA_HINT_INTEGER,
+      -4.0, 4.0 },
+    { 0, 0, 0 },
+    { 0, 0, 0 }
+};
+
+const LADSPA_PortRangeHint 
+RubberBandPitchShifter::hintsStereo[PortCountStereo] =
+{
+    { 0, 0, 0 },
+    { LADSPA_HINT_DEFAULT_0 |
+      LADSPA_HINT_BOUNDED_BELOW |
+      LADSPA_HINT_BOUNDED_ABOVE,
+      -100.0, 100.0 },
+    { LADSPA_HINT_DEFAULT_0 |
+      LADSPA_HINT_BOUNDED_BELOW |
+      LADSPA_HINT_BOUNDED_ABOVE |
+      LADSPA_HINT_INTEGER,
+      -12.0, 12.0 },
+    { LADSPA_HINT_DEFAULT_0 |
+      LADSPA_HINT_BOUNDED_BELOW |
+      LADSPA_HINT_BOUNDED_ABOVE |
+      LADSPA_HINT_INTEGER,
+      -4.0, 4.0 },
+    { 0, 0, 0 },
+    { 0, 0, 0 },
+    { 0, 0, 0 },
+    { 0, 0, 0 }
+};
+
+const LADSPA_Properties
+RubberBandPitchShifter::properties = LADSPA_PROPERTY_HARD_RT_CAPABLE;
+
+const LADSPA_Descriptor 
+RubberBandPitchShifter::ladspaDescriptorMono =
+{
+    2979, // "Unique" ID
+    "rubberband-pitchshifter-mono", // Label
+    properties,
+    "Rubber Band Mono Pitch Shifter", // Name
+    "Chris Cannam", //!!! Maker
+    "GPL", //!!! Copyright
+    PortCountMono,
+    portsMono,
+    portNamesMono,
+    hintsMono,
+    0, // Implementation data
+    instantiate,
+    connectPort,
+    activate,
+    run,
+    0, // Run adding
+    0, // Set run adding gain
+    deactivate,
+    cleanup
+};
+
+const LADSPA_Descriptor 
+RubberBandPitchShifter::ladspaDescriptorStereo =
+{
+    9792, // "Unique" ID
+    "rubberband-pitchshifter-stereo", // Label
+    properties,
+    "Rubber Band Stereo Pitch Shifter", // Name
+    "Chris Cannam", //!!! Maker
+    "GPL", //!!! Copyright
+    PortCountStereo,
+    portsStereo,
+    portNamesStereo,
+    hintsStereo,
+    0, // Implementation data
+    instantiate,
+    connectPort,
+    activate,
+    run,
+    0, // Run adding
+    0, // Set run adding gain
+    deactivate,
+    cleanup
+};
+
+const LADSPA_Descriptor *
+RubberBandPitchShifter::getDescriptor(unsigned long index)
+{
+//    std::cerr << "RubberBandPitchShifter::getDescriptor(" << index << ")" << std::endl;
+    if (index == 0) return &ladspaDescriptorMono;
+    if (index == 1) return &ladspaDescriptorStereo;
+    else return 0;
+}
+
+RubberBandPitchShifter::RubberBandPitchShifter(int sampleRate, size_t channels) :
+    m_latency(0),
+    m_cents(0),
+    m_semitones(0),
+    m_octaves(0),
+    m_ratio(1.0),
+    m_prevRatio(1.0),
+    m_extraLatency(8192), //!!! this should be at least the maximum possible displacement from linear at input rates, divided by the pitch scale factor.  It could be very large
+    m_stretcher(new RubberBand::RubberBandStretcher
+                (sampleRate, channels,
+                 RubberBand::RubberBandStretcher::OptionProcessRealTime |
+                 RubberBand::RubberBandStretcher::OptionStretchPrecise  |
+//                 RubberBand::RubberBandStretcher::OptionTransientsSmooth |
+                 RubberBand::RubberBandStretcher::OptionTransientsCrisp |
+                 RubberBand::RubberBandStretcher::OptionPhasePeakLocked |
+                 RubberBand::RubberBandStretcher::OptionThreadingNone)),
+    m_sampleRate(sampleRate),
+    m_channels(channels)
+{
+//    m_stretcher->setMaxProcessBlockSize(4096);
+
+    for (size_t c = 0; c < m_channels; ++c) {
+        m_input[c] = 0;
+        m_output[c] = 0;
+        //!!! size must be at least max process size plus m_extraLatency:
+        m_outputBuffer[c] = new RubberBand::RingBuffer<float>(8092); //!!!
+        m_outputBuffer[c]->zero(m_extraLatency);
+        //!!! size must be at least max process size:
+        m_scratch[c] = new float[16384];//!!!
+    }
+}
+
+RubberBandPitchShifter::~RubberBandPitchShifter()
+{
+    delete m_stretcher;
+    for (size_t c = 0; c < m_channels; ++c) {
+        delete m_outputBuffer[c];
+        delete[] m_scratch[c];
+    }
+}
+    
+LADSPA_Handle
+RubberBandPitchShifter::instantiate(const LADSPA_Descriptor *desc, unsigned long rate)
+{
+    if (desc->PortCount == ladspaDescriptorMono.PortCount) {
+        return new RubberBandPitchShifter(rate, 1);
+    } else if (desc->PortCount == ladspaDescriptorStereo.PortCount) {
+        return new RubberBandPitchShifter(rate, 2);
+    }
+    return 0;
+}
+
+void
+RubberBandPitchShifter::connectPort(LADSPA_Handle handle,
+				    unsigned long port, LADSPA_Data *location)
+{
+    RubberBandPitchShifter *shifter = (RubberBandPitchShifter *)handle;
+
+    float **ports[PortCountStereo] = {
+        &shifter->m_latency,
+	&shifter->m_cents,
+	&shifter->m_semitones,
+	&shifter->m_octaves,
+	&shifter->m_input[0],
+	&shifter->m_output[0],
+	&shifter->m_input[1],
+	&shifter->m_output[1]
+    };
+
+    *ports[port] = (float *)location;
+}
+
+void
+RubberBandPitchShifter::activate(LADSPA_Handle handle)
+{
+    RubberBandPitchShifter *shifter = (RubberBandPitchShifter *)handle;
+//!!!    QMutexLocker locker(&shifter->m_mutex);
+
+    shifter->updateRatio();
+    shifter->m_prevRatio = shifter->m_ratio;
+    shifter->m_stretcher->reset();
+    shifter->m_stretcher->setPitchScale(shifter->m_ratio);
+}
+
+void
+RubberBandPitchShifter::run(LADSPA_Handle handle, unsigned long samples)
+{
+    RubberBandPitchShifter *shifter = (RubberBandPitchShifter *)handle;
+    shifter->runImpl(samples);
+}
+
+void
+RubberBandPitchShifter::updateRatio()
+{
+    double oct = *m_octaves;
+    oct += *m_semitones / 12;
+    oct += *m_cents / 1200;
+    m_ratio = pow(2.0, oct);
+}
+
+void
+RubberBandPitchShifter::runImpl(unsigned long insamples)
+{
+//    std::cerr << "RubberBandPitchShifter::runImpl(" << insamples << ")" << std::endl;
+
+    updateRatio();
+    if (m_ratio != m_prevRatio) {
+        m_stretcher->setPitchScale(m_ratio);
+        m_prevRatio = m_ratio;
+    }
+
+    if (m_latency) {
+        *m_latency = m_stretcher->getLatency() + m_extraLatency;
+//        std::cerr << "latency = " << *m_latency << std::endl;
+    }
+
+    int samples = insamples;
+    int processed = 0;
+    size_t outTotal = 0;
+
+    float *ptrs[2];
+
+    //!!! We have to break up the input into chunks like this because
+    // insamples could be arbitrarily large 
+
+    while (processed < samples) {
+
+        //!!! size_t:
+        int toCauseProcessing = m_stretcher->getSamplesRequired();
+//        std::cout << "to-cause: " << toCauseProcessing << ", remain = " << samples - processed;
+        int inchunk = std::min(samples - processed, toCauseProcessing);
+        for (size_t c = 0; c < m_channels; ++c) {
+            ptrs[c] = &(m_input[c][processed]);
+        }
+        m_stretcher->process(ptrs, inchunk, false);
+        processed += inchunk;
+
+        int avail = m_stretcher->available();
+        int writable = m_outputBuffer[0]->getWriteSpace();
+        int outchunk = std::min(avail, writable);
+        size_t actual = m_stretcher->retrieve(m_scratch, outchunk);
+        outTotal += actual;
+
+//        std::cout << ", avail: " << avail << ", outchunk = " << outchunk;
+//        if (actual != outchunk) std::cout << " (" << actual << ")";
+//        std::cout << std::endl;
+
+        outchunk = actual;
+
+        for (size_t c = 0; c < m_channels; ++c) {
+            if (int(m_outputBuffer[c]->getWriteSpace()) < outchunk) {
+                std::cerr << "RubberBandPitchShifter::runImpl: buffer overrun: chunk = " << outchunk << ", space = " << m_outputBuffer[c]->getWriteSpace() << std::endl;
+            }                
+            m_outputBuffer[c]->write(m_scratch[c], outchunk);
+        }
+    }
+
+//    std::cout << "processed = " << processed << " in, " << outTotal << " out" << ", fill = " << m_outputBuffer[0]->getReadSpace() << " of " << m_outputBuffer[0]->getSize() << std::endl;
+    
+    for (size_t c = 0; c < m_channels; ++c) {
+        int avail = m_outputBuffer[c]->getReadSpace();
+//        std::cout << "avail: " << avail << std::endl;
+        if (avail < samples && c == 0) {
+            std::cerr << "RubberBandPitchShifter::runImpl: buffer underrun: required = " << samples << ", available = " << avail << std::endl;
+        }
+        int chunk = std::min(avail, samples);
+//        std::cout << "out chunk: " << chunk << std::endl;
+        m_outputBuffer[c]->read(m_output[c], chunk);
+    }
+}
+
+void
+RubberBandPitchShifter::deactivate(LADSPA_Handle handle)
+{
+    activate(handle); // both functions just reset the plugin
+}
+
+void
+RubberBandPitchShifter::cleanup(LADSPA_Handle handle)
+{
+    delete (RubberBandPitchShifter *)handle;
+}
+

src/ladspa/RubberBandPitchShifter.h

@@ -0,0 +1,91 @@
+/* -*- c-basic-offset: 4 indent-tabs-mode: nil -*-  vi:set ts=8 sts=4 sw=4: */
+
+/*
+    Rubber Band
+    An audio time-stretching and pitch-shifting library.
+    Copyright 2007 Chris Cannam.
+    
+    This program is free software; you can redistribute it and/or
+    modify it under the terms of the GNU General Public License as
+    published by the Free Software Foundation; either version 2 of the
+    License, or (at your option) any later version.  See the file
+    COPYING included with this distribution for more information.
+*/
+
+#ifndef _RUBBERBAND_PITCH_SHIFTER_H_
+#define _RUBBERBAND_PITCH_SHIFTER_H_
+
+#include <ladspa.h>
+
+#include "RingBuffer.h"
+
+namespace RubberBand {
+class RubberBandStretcher;
+}
+
+class RubberBandPitchShifter
+{
+public:
+    static const LADSPA_Descriptor *getDescriptor(unsigned long index);
+    
+protected:
+    RubberBandPitchShifter(int sampleRate, size_t channels);
+    ~RubberBandPitchShifter();
+
+    enum {
+        LatencyPort      = 0,
+	OctavesPort      = 1,
+	SemitonesPort    = 2,
+	CentsPort        = 3,
+        InputPort1       = 4,
+        OutputPort1      = 5,
+        PortCountMono    = 6,
+        InputPort2       = 6,
+        OutputPort2      = 7,
+        PortCountStereo  = 8
+    };
+
+    static const char *const portNamesMono[PortCountMono];
+    static const LADSPA_PortDescriptor portsMono[PortCountMono];
+    static const LADSPA_PortRangeHint hintsMono[PortCountMono];
+
+    static const char *const portNamesStereo[PortCountStereo];
+    static const LADSPA_PortDescriptor portsStereo[PortCountStereo];
+    static const LADSPA_PortRangeHint hintsStereo[PortCountStereo];
+
+    static const LADSPA_Properties properties;
+
+    static const LADSPA_Descriptor ladspaDescriptorMono;
+    static const LADSPA_Descriptor ladspaDescriptorStereo;
+
+    static LADSPA_Handle instantiate(const LADSPA_Descriptor *, unsigned long);
+    static void connectPort(LADSPA_Handle, unsigned long, LADSPA_Data *);
+    static void activate(LADSPA_Handle);
+    static void run(LADSPA_Handle, unsigned long);
+    static void deactivate(LADSPA_Handle);
+    static void cleanup(LADSPA_Handle);
+
+    void runImpl(unsigned long);
+    void updateRatio();
+
+    float *m_input[2];
+    float *m_output[2];
+    float *m_latency;
+    float *m_cents;
+    float *m_semitones;
+    float *m_octaves;
+    double m_ratio;
+    double m_prevRatio;
+
+    size_t m_extraLatency;
+
+    RubberBand::RubberBandStretcher *m_stretcher;
+    RubberBand::RingBuffer<float> *m_outputBuffer[2];
+    float *m_scratch[2];
+
+    int m_sampleRate;
+    size_t m_channels;
+};
+
+
+#endif

src/ladspa/libmain.cpp

@@ -0,0 +1,28 @@
+/* -*- c-basic-offset: 4 indent-tabs-mode: nil -*-  vi:set ts=8 sts=4 sw=4: */
+
+/*
+    Rubber Band
+    An audio time-stretching and pitch-shifting library.
+    Copyright 2007 Chris Cannam.
+    
+    This program is free software; you can redistribute it and/or
+    modify it under the terms of the GNU General Public License as
+    published by the Free Software Foundation; either version 2 of the
+    License, or (at your option) any later version.  See the file
+    COPYING included with this distribution for more information.
+*/
+
+#include <ladspa.h>
+
+#include "RubberBandPitchShifter.h"
+
+#include <stdio.h>
+
+extern "C" {
+
+const LADSPA_Descriptor *ladspa_descriptor(unsigned long index)
+{
+    return RubberBandPitchShifter::getDescriptor(index);
+}
+
+}

src/main.cpp

@@ -0,0 +1,408 @@
+/* -*- c-basic-offset: 4 indent-tabs-mode: nil -*-  vi:set ts=8 sts=4 sw=4: */
+
+/*
+    Rubber Band
+    An audio time-stretching and pitch-shifting library.
+    Copyright 2007 Chris Cannam.
+    
+    This program is free software; you can redistribute it and/or
+    modify it under the terms of the GNU General Public License as
+    published by the Free Software Foundation; either version 2 of the
+    License, or (at your option) any later version.  See the file
+    COPYING included with this distribution for more information.
+*/
+
+#include "RubberBandStretcher.h"
+
+#include <iostream>
+#include <sndfile.h>
+#include <cmath>
+#include <sys/time.h>
+#include <time.h>
+
+#include <getopt.h>
+
+using namespace std;
+using namespace RubberBand;
+
+int main(int argc, char **argv)
+{
+    int c;
+
+    double ratio = 1.0;
+    double pitchshift = 1.0;
+    double frequencyshift = 1.0;
+    int debug = 1;
+    bool realtime = false;
+    bool precise = false;
+    bool threaded = true;
+    bool transients = true;
+    bool peaklock = true;
+    bool longwin = false;
+    bool shortwin = false;
+    int crispness = -1;
+    bool help = false;
+
+    float fthresh0 = -1.f;
+    float fthresh1 = -1.f;
+    float fthresh2 = -1.f;
+
+    while (1) {
+        int thisOptind = optind ? optind : 1;
+        int optionIndex = 0;
+
+        static struct option longOpts[] = {
+            { "help",          0, 0, 'h' },
+            { "time",          1, 0, 't' },
+            { "tempo",         1, 0, 'T' },
+            { "pitch",         1, 0, 'p' },
+            { "frequency",     1, 0, 'f' },
+            { "crisp",         1, 0, 'c' },
+            { "crispness",     1, 0, 'c' },
+            { "debug",         1, 0, 'd' },
+            { "realtime",      0, 0, 'R' },
+            { "precise",       0, 0, 'P' },
+            { "no-threads",    0, 0, '0' },
+            { "no-transients", 0, 0, '1' },
+            { "no-peaklock",   0, 0, '2' },
+            { "window-long",   0, 0, '3' },
+            { "window-short",  0, 0, '4' },
+            { "thresh0",       1, 0, '5' },
+            { "thresh1",       1, 0, '6' },
+            { "thresh2",       1, 0, '7' },
+            { 0, 0, 0 }
+        };
+
+        c = getopt_long(argc, argv, "t:p:d:RPc:f:", longOpts, &optionIndex);
+        if (c == -1) break;
+
+        switch (c) {
+        case 'h': help = true; break;
+        case 't': ratio *= atof(optarg); break;
+        case 'T': { double m = atof(optarg); if (m != 0.0) ratio /= m; } break;
+        case 'p': pitchshift = atof(optarg); break;
+        case 'f': frequencyshift = atof(optarg); break;
+        case 'd': debug = atoi(optarg); break;
+        case 'R': realtime = true; break;
+        case 'P': precise = true; break;
+        case '0': threaded = false; break;
+        case '1': transients = false; break;
+        case '2': peaklock = false; break;
+        case '3': longwin = true; break;
+        case '4': shortwin = true; break;
+        case '5': fthresh0 = atof(optarg); break;
+        case '6': fthresh1 = atof(optarg); break;
+        case '7': fthresh2 = atof(optarg); break;
+        case 'c': crispness = atoi(optarg); break;
+        default: break;
+        }
+    }
+
+    if (help || optind + 2 != argc) {
+        cerr << endl;
+	cerr << "Rubber Band" << endl;
+        cerr << "An audio time-stretching and pitch-shifting library and utility program." << endl;
+	cerr << "Copyright 2007 Chris Cannam." << endl;
+        cerr << endl;
+	cerr << "Usage: " << argv[0] << " [options] <infile.wav> <outfile.wav>" << endl;
+        cerr << endl;
+        cerr << "where options may be:" << endl;
+        cerr << endl;
+        cerr << "  -t<X>, --time <X>       Stretch to X times original duration, or" << endl;
+        cerr << "  -T<X>, --tempo <X>      Change tempo by multiple X (equivalent to --time 1/X)" << endl;
+        cerr << endl;
+        cerr << "  -p<X>, --pitch <X>      Raise pitch by X semitones, or" << endl;
+        cerr << "  -f<X>, --frequency <X>  Change frequency by multiple X" << endl;
+        cerr << endl;
+        cerr << "  -c<N>, --crisp <N>      Crispness (N = 0,1,2,3); default 2 (see below)" << endl;
+        cerr << endl;
+        cerr << "The following options adjust the processing mode and stretch algorithm." << endl;
+        cerr << "These are mostly included for test purposes; the default settings and standard" << endl;
+        cerr << "crispness parameter are intended to provide the best sounding set of options" << endl;
+        cerr << "for most situations." << endl;
+        cerr << endl;
+        cerr << "  -P,    --precise        Aim for minimal time distortion (implied by -R)" << endl;
+        cerr << "  -R,    --realtime       Select realtime mode (implies -P --no-threads)" << endl;
+        cerr << "         --no-threads     No extra threads regardless of cpus/channel count" << endl;
+        cerr << "         --no-transients  Disable phase resynchronisation at transients" << endl;
+        cerr << "         --no-peaklock    Disable phase locking to peak frequencies" << endl;
+        cerr << "         --window-long    Use longer processing window (actual size may vary)" << endl;
+        cerr << "         --window-short   Use shorter processing window" << endl;
+        cerr << "         --thresh<N> <F>  Set internal freq threshold N (N = 0,1,2) to F Hz" << endl;
+        cerr << endl;
+        cerr << "  -d<N>, --debug <N>      Select debug level (N = 0,1,2,3); default 1, full 3" << std::endl;
+        cerr << "                          (N.B. debug level 3 includes audible ticks in output)" << endl;
+        cerr << endl;
+        cerr << "  -h,    --help           Show this help" << endl;
+        cerr << endl;
+        cerr << "\"Crispness\" levels:" << endl;
+        cerr << "  -c 0   equivalent to --no-transients --no-peaklock" << endl;
+        cerr << "  -c 1   equivalent to --no-peaklock" << endl;
+        cerr << "  -c 2   default processing options" << endl;
+        cerr << "  -c 3   equivalent to --no-peaklock --window-short (may be suitable for drums)" << endl;
+        cerr << endl;
+	return 2;
+    }
+
+    switch (crispness) {
+    case -1: crispness = 2; break;
+    case 0: transients = false; peaklock = false; longwin = false; shortwin = false; break;
+    case 1: transients = true; peaklock = false; longwin = false; shortwin = false; break;
+    case 2: transients = true; peaklock = true; longwin = false; shortwin = false; break;
+    case 3: transients = true; peaklock = false; longwin = false; shortwin = true; break;
+    };
+
+    char *fileName = strdup(argv[optind++]);
+    char *fileNameOut = strdup(argv[optind++]);
+    
+    SNDFILE *sndfile;
+    SNDFILE *sndfileOut;
+    SF_INFO sfinfo;
+    SF_INFO sfinfoOut;
+    memset(&sfinfo, 0, sizeof(SF_INFO));
+
+    sndfile = sf_open(fileName, SFM_READ, &sfinfo);
+    if (!sndfile) {
+	cerr << "ERROR: Failed to open input file \"" << fileName << "\": "
+	     << sf_strerror(sndfile) << endl;
+	return 1;
+    }
+
+    sfinfoOut.channels = sfinfo.channels;
+    sfinfoOut.format = sfinfo.format;
+    sfinfoOut.frames = int(sfinfo.frames * ratio + 0.1);
+    sfinfoOut.samplerate = sfinfo.samplerate;
+    sfinfoOut.sections = sfinfo.sections;
+    sfinfoOut.seekable = sfinfo.seekable;
+
+    sndfileOut = sf_open(fileNameOut, SFM_WRITE, &sfinfoOut) ;
+    if (!sndfileOut) {
+	cerr << "ERROR: Failed to open output file \"" << fileName << "\" for writing: "
+	     << sf_strerror(sndfile) << endl;
+	return 1;
+    }
+    
+    int ibs = 1024;
+    size_t channels = sfinfo.channels;
+
+    RubberBandStretcher::Options options = 0;
+    if (realtime)    options |= RubberBandStretcher::OptionProcessRealTime;
+    if (precise)     options |= RubberBandStretcher::OptionStretchPrecise;
+    if (!transients) options |= RubberBandStretcher::OptionTransientsSmooth;
+    if (!peaklock)   options |= RubberBandStretcher::OptionPhaseIndependent;
+    if (!threaded)   options |= RubberBandStretcher::OptionThreadingNone;
+    if (longwin)     options |= RubberBandStretcher::OptionWindowLong;
+    if (shortwin)    options |= RubberBandStretcher::OptionWindowShort;
+
+    if (pitchshift != 1.0) {
+        frequencyshift *= pow(2.0, pitchshift / 12);
+    }
+
+    RubberBandStretcher ts(sfinfo.samplerate, channels, options,
+                           ratio, frequencyshift);
+    
+    ts.setDebugLevel(debug);
+
+    ts.setExpectedInputDuration(sfinfo.frames);
+
+//    ts.setTimeRatio(ratio);
+//    ts.setPitchScale(pitchshift);
+
+    float *fbuf = new float[channels * ibs];
+    float **ibuf = new float *[channels];
+    for (size_t i = 0; i < channels; ++i) ibuf[i] = new float[ibs];
+
+    int frame = 0;
+    int percent = 0;
+
+    struct timeval tv;
+    (void)gettimeofday(&tv, 0);
+
+    if (!realtime) {
+
+        cerr << "First pass (studying)..." << endl;
+
+        while (frame < sfinfo.frames) {
+
+//        std::cout << "study frame " << frame << std::endl;
+
+            int count = -1;
+
+            if (sf_seek(sndfile, frame, SEEK_SET) < 0) break;
+            if ((count = sf_readf_float(sndfile, fbuf, ibs)) <= 0) break;
+        
+            for (size_t c = 0; c < channels; ++c) {
+                for (int i = 0; i < count; ++i) {
+                    float value = fbuf[i * channels + c];
+                    ibuf[c][i] = value;
+                }
+            }
+
+            bool final = (frame + ibs >= sfinfo.frames);
+
+            ts.study(ibuf, count, final);
+
+            int p = int((double(frame) * 100.0) / sfinfo.frames);
+            if (p > percent || frame == 0) {
+                percent = p;
+                cerr << "\r" << percent << "% ";
+            }
+
+            frame += ibs;
+        }
+
+        cerr << endl;
+
+        cerr << "Second pass (processing)..." << endl;
+    }
+
+    frame = 0;
+    percent = 0;
+    
+    float inpeak = 0;
+    double insum = 0;
+    float outpeak = 0.0;
+    double outsum = 0.0;
+    size_t countIn = 0, countOut = 0;
+
+    while (frame < sfinfo.frames) {
+
+        int count = -1;
+
+	if (sf_seek(sndfile, frame, SEEK_SET) < 0) break;
+	if ((count = sf_readf_float(sndfile, fbuf, ibs)) < 0) break;
+        
+        countIn += count;
+
+        for (size_t c = 0; c < channels; ++c) {
+            for (int i = 0; i < count; ++i) {
+                float value = fbuf[i * channels + c];
+                ibuf[c][i] = value;
+                if (fabsf(value) > inpeak) inpeak = fabsf(value);
+                insum += value * value;
+            }
+        }
+
+        bool final = (frame + ibs >= sfinfo.frames);
+
+        ts.process(ibuf, count, final);
+
+//        if 
+//            std::cerr << frame << " + " << ibs << " >= " << sfinfo.frames << ": calling ts.complete()!" << std::endl;
+//            ts.complete();
+//        }
+
+        int avail = ts.available();
+        if (debug > 1) std::cerr << "available = " << avail << std::endl;
+
+        if (avail > 0) {
+            float **obf = new float *[channels];
+            for (size_t i = 0; i < channels; ++i) {
+                obf[i] = new float[avail];
+            }
+            ts.retrieve(obf, avail);
+            countOut += avail;
+            float *fobf = new float[channels * avail];
+            for (size_t c = 0; c < channels; ++c) {
+                for (size_t i = 0; i < avail; ++i) {
+                    float value = obf[c][i];
+                    if (fabsf(value) > outpeak) outpeak = fabsf(value);
+                    outsum += value * value;
+                    value *= 0.75;
+                    if (value > 1.f) value = 1.f;
+                    if (value < -1.f) value = -1.f;
+                    fobf[i * channels + c] = value;
+                }
+            }
+//            std::cout << "fobf mean: ";
+//    double d = 0;
+//    for (int i = 0; i < avail; ++i) {
+//        d += fobf[i];
+//    }
+//    d /= avail;
+//    std::cout << d << std::endl;
+            sf_writef_float(sndfileOut, fobf, avail);
+            delete[] fobf;
+            for (size_t i = 0; i < channels; ++i) {
+                delete[] obf[i];
+            }
+            delete[] obf;
+        }
+
+	int p = int((double(frame) * 100.0) / sfinfo.frames);
+	if (p > percent || frame == 0) {
+	    percent = p;
+	    cerr << "\r" << percent << "% ";
+	}
+
+        frame += ibs;
+    }
+
+    int avail;
+
+    while ((avail = ts.available()) >= 0) {
+
+        if (debug > 1) std::cerr << "(completing) available = " << avail << std::endl;
+
+        if (avail > 0) {
+            float **obf = new float *[channels];
+            for (size_t i = 0; i < channels; ++i) {
+                obf[i] = new float[avail];
+            }
+            ts.retrieve(obf, avail);
+            countOut += avail;
+            float *fobf = new float[channels * avail];
+            for (size_t c = 0; c < channels; ++c) {
+                for (size_t i = 0; i < avail; ++i) {
+                    float value = obf[c][i];
+                    if (fabsf(value) > outpeak) outpeak = fabsf(value);
+                    outsum += value * value;
+                    value *= 0.75;
+                    if (value > 1.f) value = 1.f;
+                    if (value < -1.f) value = -1.f;
+                    fobf[i * channels + c] = value;
+                }
+            }
+
+            sf_writef_float(sndfileOut, fobf, avail);
+            delete[] fobf;
+            for (size_t i = 0; i < channels; ++i) {
+                delete[] obf[i];
+            }
+            delete[] obf;
+        }
+    }
+
+    sf_close(sndfile);
+    sf_close(sndfileOut);
+
+    double inmean = sqrt(insum / (sfinfo.frames * sfinfo.channels));
+    double outmean = sqrt(outsum / (countOut * sfinfo.channels));
+
+    cerr << endl << "in: " << countIn << ", out: " << countOut << ", ratio: " << float(countOut)/float(countIn) << ", ideal output: " << lrint(countIn * ratio) << ", diff: " << abs(lrint(countIn * ratio) - int(countOut)) << endl;
+
+    cerr << "input peak: " << inpeak << "; output peak " << outpeak << "; gain " << (inpeak > 0 ? outpeak/inpeak : 1) << endl;
+    cerr << "input rms: " << inmean << "; output rms " << outmean << "; gain " << (inmean > 0 ? outmean/inmean : 1) << endl;
+
+    struct timeval etv;
+    (void)gettimeofday(&etv, 0);
+
+    cerr << "\nstart:   " << tv.tv_sec << ":" << tv.tv_usec << endl;
+    cerr << "finish:  " << etv.tv_sec << ":" << etv.tv_usec << endl;
+    
+    etv.tv_sec -= tv.tv_sec;
+    if (etv.tv_usec < tv.tv_usec) {
+        etv.tv_usec += 1000000;
+        etv.tv_sec -= 1;
+    }
+    etv.tv_usec -= tv.tv_usec;
+
+    cerr << "elapsed: " << etv.tv_sec << ":" << etv.tv_usec << endl;
+
+    double sec = double(etv.tv_sec) + (double(etv.tv_usec) / 1000000.0);
+    cerr << "\nin/sec: " << countIn/sec << ", out/sec: " << countOut/sec << endl;
+
+    return 0;
+}
+
+

src/sysutils.cpp

@@ -0,0 +1,53 @@
+/* -*- c-basic-offset: 4 indent-tabs-mode: nil -*-  vi:set ts=8 sts=4 sw=4: */
+
+/*
+    Rubber Band
+    An audio time-stretching and pitch-shifting library.
+    Copyright 2007 Chris Cannam.
+    
+    This program is free software; you can redistribute it and/or
+    modify it under the terms of the GNU General Public License as
+    published by the Free Software Foundation; either version 2 of the
+    License, or (at your option) any later version.  See the file
+    COPYING included with this distribution for more information.
+*/
+
+#include "sysutils.h"
+
+#include <stdio.h>
+#include <string.h>
+
+namespace RubberBand {
+
+bool
+system_is_multiprocessor()
+{
+    static bool tested = false, mp = false;
+
+    if (tested) return mp;
+
+    
+    //...
+
+    FILE *cpuinfo = fopen("/proc/cpuinfo", "r");
+    if (!cpuinfo) return false;
+
+    int count = 0;
+    char buf[256];
+    while (!feof(cpuinfo)) {
+        fgets(buf, 256, cpuinfo);
+        if (!strncmp(buf, "processor", 9)) {
+            ++count;
+        }
+        if (count > 1) break;
+    }
+
+    fclose(cpuinfo);
+    return (count > 1);
+
+}
+
+}
+
+
+

src/sysutils.h

@@ -0,0 +1,24 @@
+/* -*- c-basic-offset: 4 indent-tabs-mode: nil -*-  vi:set ts=8 sts=4 sw=4: */
+
+/*
+    Rubber Band
+    An audio time-stretching and pitch-shifting library.
+    Copyright 2007 Chris Cannam.
+    
+    This program is free software; you can redistribute it and/or
+    modify it under the terms of the GNU General Public License as
+    published by the Free Software Foundation; either version 2 of the
+    License, or (at your option) any later version.  See the file
+    COPYING included with this distribution for more information.
+*/
+
+#ifndef _RUBBERBAND_SYSINFO_H_
+#define _RUBBERBAND_SYSINFO_H_
+
+namespace RubberBand {
+
+extern bool system_is_multiprocessor();
+
+}
+
+#endif

src/vamp/RubberBandVampPlugin.cpp

@@ -0,0 +1,564 @@
+/* -*- c-basic-offset: 4 indent-tabs-mode: nil -*-  vi:set ts=8 sts=4 sw=4: */
+
+/*
+    Rubber Band
+    An audio time-stretching and pitch-shifting library.
+    Copyright 2007 Chris Cannam.
+    
+    This program is free software; you can redistribute it and/or
+    modify it under the terms of the GNU General Public License as
+    published by the Free Software Foundation; either version 2 of the
+    License, or (at your option) any later version.  See the file
+    COPYING included with this distribution for more information.
+*/
+
+#include "RubberBandVampPlugin.h"
+
+#include "StretchCalculator.h"
+
+#include <cmath>
+
+using std::string;
+using std::vector;
+using std::cerr;
+using std::endl;
+
+class RubberBandVampPlugin::Impl
+{
+public:
+    size_t m_stepSize;
+    size_t m_blockSize;
+    size_t m_sampleRate;
+
+    float m_timeRatio;
+    float m_pitchRatio;
+
+    bool m_realtime;
+    bool m_elasticTiming;
+    bool m_crispTransients;
+    bool m_threadingAllowed;
+
+    RubberBand::RubberBandStretcher *m_stretcher;
+
+    int m_incrementsOutput;
+    int m_aggregateIncrementsOutput;
+    int m_divergenceOutput;
+    int m_lockDfOutput;
+    int m_smoothedLockDfOutput;
+    int m_lockPointsOutput;
+
+    size_t m_counter;
+    size_t m_accumulatedIncrement;
+
+    FeatureSet processOffline(const float *const *inputBuffers,
+                              Vamp::RealTime timestamp);
+
+    FeatureSet getRemainingFeaturesOffline();
+
+    FeatureSet processRealTime(const float *const *inputBuffers,
+                               Vamp::RealTime timestamp);
+
+    FeatureSet getRemainingFeaturesRealTime();
+
+    FeatureSet createFeatures(size_t inputIncrement,
+                              std::vector<int> &outputIncrements,
+                              std::vector<float> &lockDf,
+                              std::vector<float> &smoothedDF,
+                              size_t baseCount,
+                              bool includeFinal);
+};
+
+
+RubberBandVampPlugin::RubberBandVampPlugin(float inputSampleRate) :
+    Plugin(inputSampleRate)
+{
+    m_d = new Impl();
+    m_d->m_stepSize = 0;
+    m_d->m_timeRatio = 1.0;
+    m_d->m_pitchRatio = 1.0;
+    m_d->m_realtime = false;
+    m_d->m_elasticTiming = true;
+    m_d->m_crispTransients = true;
+    m_d->m_threadingAllowed = true;
+    m_d->m_stretcher = 0;
+    m_d->m_sampleRate = lrintf(m_inputSampleRate);
+}
+
+RubberBandVampPlugin::~RubberBandVampPlugin()
+{
+    delete m_d->m_stretcher;
+    delete m_d;
+}
+
+string
+RubberBandVampPlugin::getIdentifier() const
+{
+    return "rubberband";
+}
+
+string
+RubberBandVampPlugin::getName() const
+{
+    return "Rubber Band Timestretch Analysis";
+}
+
+string
+RubberBandVampPlugin::getDescription() const
+{
+    return "Carry out analysis phases of time stretcher process";
+}
+
+string
+RubberBandVampPlugin::getMaker() const
+{
+    return "Rubber Band"; ///!!!
+}
+
+int
+RubberBandVampPlugin::getPluginVersion() const
+{
+    return 1;
+}
+
+string
+RubberBandVampPlugin::getCopyright() const
+{
+    return "";//!!!
+}
+
+RubberBandVampPlugin::OutputList
+RubberBandVampPlugin::getOutputDescriptors() const
+{
+    OutputList list;
+
+    size_t rate = 0;
+    if (m_d->m_stretcher) {
+        rate = lrintf(m_inputSampleRate / m_d->m_stretcher->getInputIncrement());
+    }
+
+    OutputDescriptor d;
+    d.identifier = "increments";
+    d.name = "Output Increments";
+    d.description = ""; //!!!
+    d.unit = "samples";
+    d.hasFixedBinCount = true;
+    d.binCount = 1;
+    d.hasKnownExtents = false; //!!!
+    d.isQuantized = true;
+    d.quantizeStep = 1.0;
+    d.sampleType = OutputDescriptor::VariableSampleRate;
+    d.sampleRate = rate;
+    m_d->m_incrementsOutput = list.size();
+    list.push_back(d);
+
+    d.identifier = "aggregate_increments";
+    d.name = "Accumulated Output Increments";
+    d.description = ""; //!!!
+    d.sampleRate = 0;
+    m_d->m_aggregateIncrementsOutput = list.size();
+    list.push_back(d);
+
+    d.identifier = "divergence";
+    d.name = "Divergence from Linear";
+    d.description = ""; //!!!
+    d.isQuantized = false;
+    d.sampleRate = 0;
+    m_d->m_divergenceOutput = list.size();
+    list.push_back(d);
+
+    d.identifier = "lockdf";
+    d.name = "Lock Point Detection Function";
+    d.description = ""; //!!!
+    d.unit = "";
+    d.sampleRate = rate;
+    m_d->m_lockDfOutput = list.size();
+    list.push_back(d);
+
+    d.identifier = "smoothedlockdf";
+    d.name = "Smoothed Lock Point Detection Function";
+    d.description = ""; //!!!
+    d.unit = "";
+    m_d->m_smoothedLockDfOutput = list.size();
+    list.push_back(d);
+
+    d.identifier = "lockpoints";
+    d.name = "Phase Lock Points";
+    d.description = ""; //!!!
+    d.unit = "";
+    d.hasFixedBinCount = true;
+    d.binCount = 0;
+    d.hasKnownExtents = false;
+    d.isQuantized = false;
+    d.sampleRate = 0;
+    m_d->m_lockPointsOutput = list.size();
+    list.push_back(d);
+
+    return list;
+}
+
+RubberBandVampPlugin::ParameterList
+RubberBandVampPlugin::getParameterDescriptors() const
+{
+    ParameterList list;
+
+    ParameterDescriptor d;
+    d.identifier = "timeratio";
+    d.name = "Timestretch Ratio";
+    d.description = ""; //!!!
+    d.unit = "";
+    d.minValue = 0.0000001;
+    d.maxValue = 1000;
+    d.defaultValue = 1.0;
+    d.isQuantized = false;
+    list.push_back(d);
+
+    d.identifier = "pitchratio";
+    d.name = "Pitch Scaling Ratio";
+    d.description = ""; //!!!
+    d.unit = "";
+    d.minValue = 0.0000001;
+    d.maxValue = 1000;
+    d.defaultValue = 1.0;
+    d.isQuantized = false;
+    list.push_back(d);
+
+    d.identifier = "mode";
+    d.name = "Processing Mode";
+    d.description = ""; //!!!
+    d.unit = "";
+    d.minValue = 0;
+    d.maxValue = 1;
+    d.defaultValue = 0;
+    d.isQuantized = true;
+    d.quantizeStep = 1;
+    d.valueNames.clear();
+    d.valueNames.push_back("Offline");
+    d.valueNames.push_back("Real Time");
+    list.push_back(d);
+
+    d.identifier = "stretchtype";
+    d.name = "Stretch Flexibility";
+    d.description = ""; //!!!
+    d.unit = "";
+    d.minValue = 0;
+    d.maxValue = 1;
+    d.defaultValue = 0;
+    d.isQuantized = true;
+    d.quantizeStep = 1;
+    d.valueNames.clear();
+    d.valueNames.push_back("Elastic");
+    d.valueNames.push_back("Precise");
+    list.push_back(d);
+
+    d.identifier = "transientmode";
+    d.name = "Transient Handling";
+    d.description = ""; //!!!
+    d.unit = "";
+    d.minValue = 0;
+    d.maxValue = 1;
+    d.defaultValue = 0;
+    d.isQuantized = true;
+    d.quantizeStep = 1;
+    d.valueNames.clear();
+    d.valueNames.push_back("Crisp");
+    d.valueNames.push_back("Soft");
+    list.push_back(d);
+
+    d.identifier = "threadingmode";
+    d.name = "Threading";
+    d.description = ""; //!!!
+    d.unit = "";
+    d.minValue = 0;
+    d.maxValue = 1;
+    d.defaultValue = 0;
+    d.isQuantized = true;
+    d.quantizeStep = 1;
+    d.valueNames.clear();
+    d.valueNames.push_back("Enabled");
+    d.valueNames.push_back("Disabled");
+    list.push_back(d);
+
+    return list;
+}
+
+float
+RubberBandVampPlugin::getParameter(std::string id) const
+{
+    if (id == "timeratio") return m_d->m_timeRatio;
+    if (id == "pitchratio") return m_d->m_pitchRatio;
+    if (id == "mode") return m_d->m_realtime ? 1 : 0;
+    if (id == "stretchtype") return m_d->m_elasticTiming ? 0 : 1;
+    if (id == "transientmode") return m_d->m_crispTransients ? 0 : 1;
+    if (id == "threadingmode") return m_d->m_threadingAllowed ? 0 : 1;
+    return 0.f;
+}
+
+void
+RubberBandVampPlugin::setParameter(std::string id, float value)
+{
+    if (id == "timeratio") {
+        m_d->m_timeRatio = value;
+    } else if (id == "pitchratio") {
+        m_d->m_pitchRatio = value;
+    } else {
+        bool set = (value > 0.5);
+        if (id == "mode") m_d->m_realtime = set;
+        else if (id == "stretchtype") m_d->m_elasticTiming = !set;
+        else if (id == "transientmode") m_d->m_crispTransients = !set;
+        else if (id == "threadingmode") m_d->m_threadingAllowed = !set;
+    }
+}
+
+bool
+RubberBandVampPlugin::initialise(size_t channels, size_t stepSize, size_t blockSize)
+{
+    if (channels < getMinChannelCount() ||
+	channels > getMaxChannelCount()) return false;
+
+    m_d->m_stepSize = std::min(stepSize, blockSize);
+    m_d->m_blockSize = stepSize;
+
+    RubberBand::RubberBandStretcher::Options options = 0;
+
+    if (m_d->m_realtime)
+         options |= RubberBand::RubberBandStretcher::OptionProcessRealTime;
+    else options |= RubberBand::RubberBandStretcher::OptionProcessOffline;
+
+    if (m_d->m_elasticTiming)
+         options |= RubberBand::RubberBandStretcher::OptionStretchElastic;
+    else options |= RubberBand::RubberBandStretcher::OptionStretchPrecise;
+ 
+    if (m_d->m_crispTransients) 
+         options |= RubberBand::RubberBandStretcher::OptionTransientsCrisp;
+    else options |= RubberBand::RubberBandStretcher::OptionTransientsSmooth;
+
+    if (m_d->m_threadingAllowed)
+         options |= RubberBand::RubberBandStretcher::OptionThreadingAuto;
+    else options |= RubberBand::RubberBandStretcher::OptionThreadingNone;
+
+    delete m_d->m_stretcher;
+    m_d->m_stretcher = new RubberBand::RubberBandStretcher
+        (m_d->m_sampleRate, channels, options);
+    m_d->m_stretcher->setDebugLevel(2);
+    m_d->m_stretcher->setTimeRatio(m_d->m_timeRatio);
+    m_d->m_stretcher->setPitchScale(m_d->m_pitchRatio);
+
+    m_d->m_counter = 0;
+    m_d->m_accumulatedIncrement = 0;
+
+    return true;
+}
+
+void
+RubberBandVampPlugin::reset()
+{
+//    delete m_stretcher;  //!!! or just if (m_stretcher) m_stretcher->reset();
+//    m_stretcher = new RubberBand::RubberBandStretcher(lrintf(m_inputSampleRate), channels);
+    if (m_d->m_stretcher) m_d->m_stretcher->reset();
+}
+
+RubberBandVampPlugin::FeatureSet
+RubberBandVampPlugin::process(const float *const *inputBuffers,
+                              Vamp::RealTime timestamp)
+{
+    if (m_d->m_realtime) {
+        return m_d->processRealTime(inputBuffers, timestamp);
+    } else {
+        return m_d->processOffline(inputBuffers, timestamp);
+    }        
+}
+
+RubberBandVampPlugin::FeatureSet
+RubberBandVampPlugin::getRemainingFeatures()
+{
+    if (m_d->m_realtime) {
+        return m_d->getRemainingFeaturesRealTime();
+    } else {
+        return m_d->getRemainingFeaturesOffline();
+    }
+}
+
+RubberBandVampPlugin::FeatureSet
+RubberBandVampPlugin::Impl::processOffline(const float *const *inputBuffers,
+                                           Vamp::RealTime timestamp)
+{
+    if (!m_stretcher) {
+	cerr << "ERROR: RubberBandVampPlugin::processOffline: "
+	     << "RubberBandVampPlugin has not been initialised"
+	     << endl;
+	return FeatureSet();
+    }
+
+    m_stretcher->study(inputBuffers, m_blockSize, false);
+    return FeatureSet();
+}
+
+RubberBandVampPlugin::FeatureSet
+RubberBandVampPlugin::Impl::getRemainingFeaturesOffline()
+{
+    m_stretcher->study(0, 0, true);
+
+    m_stretcher->calculateStretch();
+
+    int rate = m_sampleRate;
+
+    RubberBand::StretchCalculator sc(rate,
+                                     m_stretcher->getInputIncrement(),
+                                     true);
+
+    size_t inputIncrement = m_stretcher->getInputIncrement();
+    std::vector<int> outputIncrements = m_stretcher->getOutputIncrements();
+    std::vector<float> lockDf = m_stretcher->getLockCurve();
+    std::vector<float> smoothedDf = sc.smoothDF(lockDf);
+
+    FeatureSet features = createFeatures
+        (inputIncrement, outputIncrements, lockDf, smoothedDf, 0, true);
+
+    return features;
+}
+
+RubberBandVampPlugin::FeatureSet
+RubberBandVampPlugin::Impl::processRealTime(const float *const *inputBuffers,
+                                            Vamp::RealTime timestamp)
+{
+    // This function is not in any way a real-time function (i.e. it
+    // has no requirement to be RT safe); it simply operates the
+    // stretcher in RT mode.
+
+    if (!m_stretcher) {
+	cerr << "ERROR: RubberBandVampPlugin::processRealTime: "
+	     << "RubberBandVampPlugin has not been initialised"
+	     << endl;
+	return FeatureSet();
+    }
+
+    m_stretcher->process(inputBuffers, m_blockSize, false);
+    
+    size_t inputIncrement = m_stretcher->getInputIncrement();
+    std::vector<int> outputIncrements = m_stretcher->getOutputIncrements();
+    std::vector<float> lockDf = m_stretcher->getLockCurve();
+    std::vector<float> smoothedDf; // not meaningful in RT mode
+    FeatureSet features = createFeatures(inputIncrement, outputIncrements,
+                                         lockDf, smoothedDf, m_counter, false);
+    m_counter += outputIncrements.size();
+
+    return features;
+}
+
+RubberBandVampPlugin::FeatureSet
+RubberBandVampPlugin::Impl::getRemainingFeaturesRealTime()
+{
+    return FeatureSet();
+}
+
+RubberBandVampPlugin::FeatureSet
+RubberBandVampPlugin::Impl::createFeatures(size_t inputIncrement,
+                                           std::vector<int> &outputIncrements,
+                                           std::vector<float> &lockDf,
+                                           std::vector<float> &smoothedDf,
+                                           size_t baseCount,
+                                           bool includeFinal)
+{
+    size_t actual = m_accumulatedIncrement;
+
+    double overallRatio = m_timeRatio * m_pitchRatio;
+
+    char label[200];
+
+    FeatureSet features;
+
+    int rate = m_sampleRate;
+
+    for (size_t i = 0; i < outputIncrements.size(); ++i) {
+
+        size_t frame = (baseCount + i) * inputIncrement;
+
+        int oi = outputIncrements[i];
+        bool lock = false;
+
+        if (oi < 0) {
+            oi = -oi;
+            lock = true;
+        }
+
+        double linear = (frame * overallRatio);
+
+        Vamp::RealTime t = Vamp::RealTime::frame2RealTime(frame, rate);
+
+        Feature feature;
+        feature.hasTimestamp = true;
+        feature.timestamp = t;
+        feature.values.push_back(oi);
+        feature.label = Vamp::RealTime::frame2RealTime(oi, rate).toText();
+        features[m_incrementsOutput].push_back(feature);
+
+        feature.values.clear();
+        feature.values.push_back(actual);
+        feature.label = Vamp::RealTime::frame2RealTime(actual, rate).toText();
+        features[m_aggregateIncrementsOutput].push_back(feature);
+
+        feature.values.clear();
+        feature.values.push_back(actual - linear);
+
+        sprintf(label, "expected %ld, actual %ld, difference %ld (%s ms)",
+                long(linear), long(actual), long(actual - linear),
+                // frame2RealTime expects an integer frame number,
+                // hence our multiplication factor
+                (Vamp::RealTime::frame2RealTime
+                 (lrintf((actual - linear) * 1000), rate) / 1000)
+                .toText().c_str());
+        feature.label = label;
+
+        features[m_divergenceOutput].push_back(feature);
+        actual += oi;
+        
+        char buf[30];
+
+        if (i < lockDf.size()) {
+            feature.values.clear();
+            feature.values.push_back(lockDf[i]);
+            sprintf(buf, "%d", baseCount + i);
+            feature.label = buf;
+            features[m_lockDfOutput].push_back(feature);
+        }
+
+        if (i < smoothedDf.size()) {
+            feature.values.clear();
+            feature.values.push_back(smoothedDf[i]);
+            features[m_smoothedLockDfOutput].push_back(feature);
+        }
+
+        if (lock) {
+            feature.values.clear();
+            feature.label = buf;
+            features[m_lockPointsOutput].push_back(feature);
+        }
+    }
+
+    if (includeFinal) {
+        Vamp::RealTime t = Vamp::RealTime::frame2RealTime
+            (inputIncrement * (baseCount + outputIncrements.size()), rate);
+        Feature feature;
+        feature.hasTimestamp = true;
+        feature.timestamp = t;
+        feature.label = Vamp::RealTime::frame2RealTime(actual, rate).toText();
+        feature.values.clear();
+        feature.values.push_back(actual);
+        features[m_aggregateIncrementsOutput].push_back(feature);
+
+        float linear = ((baseCount + outputIncrements.size())
+                        * inputIncrement * overallRatio);
+        feature.values.clear();
+        feature.values.push_back(actual - linear);
+        feature.label =  // see earlier comment
+            (Vamp::RealTime::frame2RealTime //!!! update this as earlier label
+             (lrintf((actual - linear) * 1000), rate) / 1000)
+            .toText();
+        features[m_divergenceOutput].push_back(feature);
+    }
+
+    m_accumulatedIncrement = actual;
+
+    return features;
+}
+

src/vamp/RubberBandVampPlugin.h

@@ -0,0 +1,56 @@
+/* -*- c-basic-offset: 4 indent-tabs-mode: nil -*-  vi:set ts=8 sts=4 sw=4: */
+
+/*
+    Rubber Band
+    An audio time-stretching and pitch-shifting library.
+    Copyright 2007 Chris Cannam.
+    
+    This program is free software; you can redistribute it and/or
+    modify it under the terms of the GNU General Public License as
+    published by the Free Software Foundation; either version 2 of the
+    License, or (at your option) any later version.  See the file
+    COPYING included with this distribution for more information.
+*/
+
+#ifndef _RUBBERBAND_VAMP_PLUGIN_H_
+#define _RUBBERBAND_VAMP_PLUGIN_H_
+
+#include <vamp-sdk/Plugin.h>
+
+#include "RubberBandStretcher.h"
+
+class RubberBandVampPlugin : public Vamp::Plugin
+{
+public:
+    RubberBandVampPlugin(float inputSampleRate);
+    virtual ~RubberBandVampPlugin();
+
+    bool initialise(size_t channels, size_t stepSize, size_t blockSize);
+    void reset();
+
+    InputDomain getInputDomain() const { return TimeDomain; }
+
+    std::string getIdentifier() const;
+    std::string getName() const;
+    std::string getDescription() const;
+    std::string getMaker() const;
+    int getPluginVersion() const;
+    std::string getCopyright() const;
+
+    ParameterList getParameterDescriptors() const;
+    float getParameter(std::string id) const;
+    void setParameter(std::string id, float value);
+
+    OutputList getOutputDescriptors() const;
+
+    FeatureSet process(const float *const *inputBuffers,
+                       Vamp::RealTime timestamp);
+
+    FeatureSet getRemainingFeatures();
+
+protected:
+    class Impl;
+    Impl *m_d;
+};
+
+#endif

src/vamp/libmain.cpp

@@ -0,0 +1,32 @@
+/* -*- c-basic-offset: 4 indent-tabs-mode: nil -*-  vi:set ts=8 sts=4 sw=4: */
+
+/*
+    Rubber Band
+    An audio time-stretching and pitch-shifting library.
+    Copyright 2007 Chris Cannam.
+    
+    This program is free software; you can redistribute it and/or
+    modify it under the terms of the GNU General Public License as
+    published by the Free Software Foundation; either version 2 of the
+    License, or (at your option) any later version.  See the file
+    COPYING included with this distribution for more information.
+*/
+
+#include <vamp/vamp.h>
+#include <vamp-sdk/PluginAdapter.h>
+
+#include "RubberBandVampPlugin.h"
+
+static Vamp::PluginAdapter<RubberBandVampPlugin> rubberBandAdapter;
+
+const VampPluginDescriptor *vampGetPluginDescriptor(unsigned int version,
+                                                    unsigned int index)
+{
+    if (version < 1) return 0;
+
+    switch (index) {
+    case  0: return rubberBandAdapter.getDescriptor();
+    default: return 0;
+    }
+}
+