diff audioio/PhaseVocoderTimeStretcher.cpp @ 43:3c5756fb6a68

* Move some things around to facilitate plundering libraries for other
  applications without needing to duplicate so much code.
    sv/osc -> data/osc
    sv/audioio -> audioio
    sv/transform -> plugin/transform
    sv/document -> document (will rename to framework in next commit)

--- /dev/null	Thu Jan 01 00:00:00 1970 +0000
+++ b/audioio/PhaseVocoderTimeStretcher.cpp	Wed Oct 24 16:34:31 2007 +0000
@@ -0,0 +1,626 @@
+/* -*- c-basic-offset: 4 indent-tabs-mode: nil -*-  vi:set ts=8 sts=4 sw=4: */
+
+/*
+    Sonic Visualiser
+    An audio file viewer and annotation editor.
+    Centre for Digital Music, Queen Mary, University of London.
+    This file copyright 2006 Chris Cannam and QMUL.
+    
+    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 "PhaseVocoderTimeStretcher.h"
+
+#include <iostream>
+#include <cassert>
+
+#include <QMutexLocker>
+
+//#define DEBUG_PHASE_VOCODER_TIME_STRETCHER 1
+
+PhaseVocoderTimeStretcher::PhaseVocoderTimeStretcher(size_t sampleRate,
+                                                     size_t channels,
+                                                     float ratio,
+                                                     bool sharpen,
+                                                     size_t maxOutputBlockSize) :
+    m_sampleRate(sampleRate),
+    m_channels(channels),
+    m_maxOutputBlockSize(maxOutputBlockSize),
+    m_ratio(ratio),
+    m_sharpen(sharpen),
+    m_totalCount(0),
+    m_transientCount(0),
+    m_n2sum(0),
+    m_mutex(new QMutex())
+{
+    initialise();
+}
+
+PhaseVocoderTimeStretcher::~PhaseVocoderTimeStretcher()
+{
+    std::cerr << "PhaseVocoderTimeStretcher::~PhaseVocoderTimeStretcher" << std::endl;
+
+    cleanup();
+    
+    delete m_mutex;
+}
+
+void
+PhaseVocoderTimeStretcher::initialise()
+{
+    std::cerr << "PhaseVocoderTimeStretcher::initialise" << std::endl;
+
+    calculateParameters();
+        
+    m_analysisWindow = new Window<float>(HanningWindow, m_wlen);
+    m_synthesisWindow = new Window<float>(HanningWindow, m_wlen);
+
+    m_prevPhase = new float *[m_channels];
+    m_prevAdjustedPhase = new float *[m_channels];
+
+    m_prevTransientMag = (float *)fftf_malloc(sizeof(float) * (m_wlen / 2 + 1));
+    m_prevTransientScore = 0;
+    m_prevTransient = false;
+
+    m_tempbuf = (float *)fftf_malloc(sizeof(float) * m_wlen);
+
+    m_time = new float *[m_channels];
+    m_freq = new fftf_complex *[m_channels];
+    m_plan = new fftf_plan[m_channels];
+    m_iplan = new fftf_plan[m_channels];
+
+    m_inbuf = new RingBuffer<float> *[m_channels];
+    m_outbuf = new RingBuffer<float> *[m_channels];
+    m_mashbuf = new float *[m_channels];
+
+    m_modulationbuf = (float *)fftf_malloc(sizeof(float) * m_wlen);
+        
+    for (size_t c = 0; c < m_channels; ++c) {
+
+        m_prevPhase[c] = (float *)fftf_malloc(sizeof(float) * (m_wlen / 2 + 1));
+        m_prevAdjustedPhase[c] = (float *)fftf_malloc(sizeof(float) * (m_wlen / 2 + 1));
+
+        m_time[c] = (float *)fftf_malloc(sizeof(float) * m_wlen);
+        m_freq[c] = (fftf_complex *)fftf_malloc(sizeof(fftf_complex) *
+                                                  (m_wlen / 2 + 1));
+        
+        m_plan[c] = fftf_plan_dft_r2c_1d(m_wlen, m_time[c], m_freq[c], FFTW_MEASURE);
+        m_iplan[c] = fftf_plan_dft_c2r_1d(m_wlen, m_freq[c], m_time[c], FFTW_MEASURE);
+
+        m_outbuf[c] = new RingBuffer<float>
+            ((m_maxOutputBlockSize + m_wlen) * 2);
+        m_inbuf[c] = new RingBuffer<float>
+            (lrintf(m_outbuf[c]->getSize() / m_ratio) + m_wlen);
+
+        std::cerr << "making inbuf size " << m_inbuf[c]->getSize() << " (outbuf size is " << m_outbuf[c]->getSize() << ", ratio " << m_ratio << ")" << std::endl;
+
+           
+        m_mashbuf[c] = (float *)fftf_malloc(sizeof(float) * m_wlen);
+        
+        for (size_t i = 0; i < m_wlen; ++i) {
+            m_mashbuf[c][i] = 0.0;
+        }
+
+        for (size_t i = 0; i <= m_wlen/2; ++i) {
+            m_prevPhase[c][i] = 0.0;
+            m_prevAdjustedPhase[c][i] = 0.0;
+        }
+    }
+
+    for (size_t i = 0; i < m_wlen; ++i) {
+        m_modulationbuf[i] = 0.0;
+    }
+
+    for (size_t i = 0; i <= m_wlen/2; ++i) {
+        m_prevTransientMag[i] = 0.0;
+    }
+}
+
+void
+PhaseVocoderTimeStretcher::calculateParameters()
+{
+    std::cerr << "PhaseVocoderTimeStretcher::calculateParameters" << std::endl;
+
+    m_wlen = 1024;
+
+    //!!! In transient sharpening mode, we need to pick the window
+    //length so as to be more or less fixed in audio duration (i.e. we
+    //need to exploit the sample rate)
+
+    //!!! have to work out the relationship between wlen and transient
+    //threshold
+
+    if (m_ratio < 1) {
+        if (m_ratio < 0.4) {
+            m_n1 = 1024;
+            m_wlen = 2048;
+        } else if (m_ratio < 0.8) {
+            m_n1 = 512;
+        } else {
+            m_n1 = 256;
+        }
+        if (shouldSharpen()) {
+            m_wlen = 2048;
+        }
+        m_n2 = lrintf(m_n1 * m_ratio);
+    } else {
+        if (m_ratio > 2) {
+            m_n2 = 512;
+            m_wlen = 4096; 
+        } else if (m_ratio > 1.6) {
+            m_n2 = 384;
+            m_wlen = 2048;
+        } else {
+            m_n2 = 256;
+        }
+        if (shouldSharpen()) {
+            if (m_wlen < 2048) m_wlen = 2048;
+        }
+        m_n1 = lrintf(m_n2 / m_ratio);
+        if (m_n1 == 0) {
+            m_n1 = 1;
+            m_n2 = lrintf(m_ratio);
+        }
+    }
+
+    m_transientThreshold = lrintf(m_wlen / 4.5);
+
+    m_totalCount = 0;
+    m_transientCount = 0;
+    m_n2sum = 0;
+
+
+    std::cerr << "PhaseVocoderTimeStretcher: channels = " << m_channels
+              << ", ratio = " << m_ratio
+              << ", n1 = " << m_n1 << ", n2 = " << m_n2 << ", wlen = "
+              << m_wlen << ", max = " << m_maxOutputBlockSize << std::endl;
+//              << ", outbuflen = " << m_outbuf[0]->getSize() << std::endl;
+}
+
+void
+PhaseVocoderTimeStretcher::cleanup()
+{
+    std::cerr << "PhaseVocoderTimeStretcher::cleanup" << std::endl;
+
+    for (size_t c = 0; c < m_channels; ++c) {
+
+        fftf_destroy_plan(m_plan[c]);
+        fftf_destroy_plan(m_iplan[c]);
+
+        fftf_free(m_time[c]);
+        fftf_free(m_freq[c]);
+
+        fftf_free(m_mashbuf[c]);
+        fftf_free(m_prevPhase[c]);
+        fftf_free(m_prevAdjustedPhase[c]);
+
+        delete m_inbuf[c];
+        delete m_outbuf[c];
+    }
+
+    fftf_free(m_tempbuf);
+    fftf_free(m_modulationbuf);
+    fftf_free(m_prevTransientMag);
+
+    delete[] m_prevPhase;
+    delete[] m_prevAdjustedPhase;
+    delete[] m_inbuf;
+    delete[] m_outbuf;
+    delete[] m_mashbuf;
+    delete[] m_time;
+    delete[] m_freq;
+    delete[] m_plan;
+    delete[] m_iplan;
+
+    delete m_analysisWindow;
+    delete m_synthesisWindow;
+}	
+
+void
+PhaseVocoderTimeStretcher::setRatio(float ratio)
+{
+    QMutexLocker locker(m_mutex);
+
+    size_t formerWlen = m_wlen;
+    m_ratio = ratio;
+
+    std::cerr << "PhaseVocoderTimeStretcher::setRatio: new ratio " << ratio
+              << std::endl;
+
+    calculateParameters();
+
+    if (m_wlen == formerWlen) {
+
+        // This is the only container whose size depends on m_ratio
+
+        RingBuffer<float> **newin = new RingBuffer<float> *[m_channels];
+
+        size_t formerSize = m_inbuf[0]->getSize();
+        size_t newSize = lrintf(m_outbuf[0]->getSize() / m_ratio) + m_wlen;
+
+        std::cerr << "resizing inbuf from " << formerSize << " to "
+                  << newSize << " (outbuf size is " << m_outbuf[0]->getSize() << ", ratio " << m_ratio << ")" << std::endl;
+
+        if (formerSize != newSize) {
+
+            size_t ready = m_inbuf[0]->getReadSpace();
+
+            for (size_t c = 0; c < m_channels; ++c) {
+                newin[c] = new RingBuffer<float>(newSize);
+            }
+
+            if (ready > 0) {
+
+                size_t copy = std::min(ready, newSize);
+                float *tmp = new float[ready];
+
+                for (size_t c = 0; c < m_channels; ++c) {
+                    m_inbuf[c]->read(tmp, ready);
+                    newin[c]->write(tmp + ready - copy, copy);
+                }
+                
+                delete[] tmp;
+            }
+            
+            for (size_t c = 0; c < m_channels; ++c) {
+                delete m_inbuf[c];
+            }
+            
+            delete[] m_inbuf;
+            m_inbuf = newin;
+        }
+
+    } else {
+        
+        std::cerr << "wlen changed" << std::endl;
+        cleanup();
+        initialise();
+    }
+}
+
+size_t
+PhaseVocoderTimeStretcher::getProcessingLatency() const
+{
+    return getWindowSize() - getInputIncrement();
+}
+
+size_t
+PhaseVocoderTimeStretcher::getRequiredInputSamples() const
+{
+    QMutexLocker locker(m_mutex);
+
+    if (m_inbuf[0]->getReadSpace() >= m_wlen) return 0;
+    return m_wlen - m_inbuf[0]->getReadSpace();
+}
+
+void
+PhaseVocoderTimeStretcher::putInput(float **input, size_t samples)
+{
+    QMutexLocker locker(m_mutex);
+
+    // We need to add samples from input to our internal buffer.  When
+    // we have m_windowSize samples in the buffer, we can process it,
+    // move the samples back by m_n1 and write the output onto our
+    // internal output buffer.  If we have (samples * ratio) samples
+    // in that, we can write m_n2 of them back to output and return
+    // (otherwise we have to write zeroes).
+
+    // When we process, we write m_wlen to our fixed output buffer
+    // (m_mashbuf).  We then pull out the first m_n2 samples from that
+    // buffer, push them into the output ring buffer, and shift
+    // m_mashbuf left by that amount.
+
+    // The processing latency is then m_wlen - m_n2.
+
+    size_t consumed = 0;
+
+    while (consumed < samples) {
+
+	size_t writable = m_inbuf[0]->getWriteSpace();
+	writable = std::min(writable, samples - consumed);
+
+	if (writable == 0) {
+#ifdef DEBUG_PHASE_VOCODER_TIME_STRETCHER
+	    std::cerr << "WARNING: PhaseVocoderTimeStretcher::putInput: writable == 0 (inbuf has " << m_inbuf[0]->getReadSpace() << " samples available for reading, space for " << m_inbuf[0]->getWriteSpace() << " more)" << std::endl;
+#endif
+            if (m_inbuf[0]->getReadSpace() < m_wlen ||
+                m_outbuf[0]->getWriteSpace() < m_n2) {
+                std::cerr << "WARNING: PhaseVocoderTimeStretcher::putInput: Inbuf has " << m_inbuf[0]->getReadSpace() << ", outbuf has space for " << m_outbuf[0]->getWriteSpace() << " (n2 = " << m_n2 << ", wlen = " << m_wlen << "), won't be able to process" << std::endl;
+                break;
+            }
+	} else {
+
+#ifdef DEBUG_PHASE_VOCODER_TIME_STRETCHER
+            std::cerr << "writing " << writable << " from index " << consumed << " to inbuf, consumed will be " << consumed + writable << std::endl;
+#endif
+
+            for (size_t c = 0; c < m_channels; ++c) {
+                m_inbuf[c]->write(input[c] + consumed, writable);
+            }
+            consumed += writable;
+        }
+
+	while (m_inbuf[0]->getReadSpace() >= m_wlen &&
+	       m_outbuf[0]->getWriteSpace() >= m_n2) {
+
+	    // We know we have at least m_wlen samples available
+	    // in m_inbuf.  We need to peek m_wlen of them for
+	    // processing, and then read m_n1 to advance the read
+	    // pointer.
+            
+            for (size_t c = 0; c < m_channels; ++c) {
+
+                size_t got = m_inbuf[c]->peek(m_tempbuf, m_wlen);
+                assert(got == m_wlen);
+
+                analyseBlock(c, m_tempbuf);
+            }
+
+            bool transient = false;
+            if (shouldSharpen()) transient = isTransient();
+
+            size_t n2 = m_n2;
+
+            if (transient) {
+                n2 = m_n1;
+            }
+
+            ++m_totalCount;
+            if (transient) ++m_transientCount;
+            m_n2sum += n2;
+
+//            std::cerr << "ratio for last 10: " <<last10num << "/" << (10 * m_n1) << " = " << float(last10num) / float(10 * m_n1) << " (should be " << m_ratio << ")" << std::endl;
+            
+            if (m_totalCount > 50 && m_transientCount < m_totalCount) {
+
+                int fixed = lrintf(m_transientCount * m_n1);
+
+                int idealTotal = lrintf(m_totalCount * m_n1 * m_ratio);
+                int idealSquashy = idealTotal - fixed;
+
+                int squashyCount = m_totalCount - m_transientCount;
+                
+                n2 = lrintf(idealSquashy / squashyCount);
+
+#ifdef DEBUG_PHASE_VOCODER_TIME_STRETCHER
+                if (n2 != m_n2) {
+                    std::cerr << m_n2 << " -> " << n2 << std::endl;
+                }
+#endif
+            }
+
+            for (size_t c = 0; c < m_channels; ++c) {
+
+                synthesiseBlock(c, m_mashbuf[c],
+                                c == 0 ? m_modulationbuf : 0,
+                                m_prevTransient ? m_n1 : m_n2);
+
+
+#ifdef DEBUG_PHASE_VOCODER_TIME_STRETCHER
+                std::cerr << "writing first " << m_n2 << " from mashbuf, skipping " << m_n1 << " on inbuf " << std::endl;
+#endif
+                m_inbuf[c]->skip(m_n1);
+
+                for (size_t i = 0; i < n2; ++i) {
+                    if (m_modulationbuf[i] > 0.f) {
+                        m_mashbuf[c][i] /= m_modulationbuf[i];
+                    }
+                }
+
+                m_outbuf[c]->write(m_mashbuf[c], n2);
+
+                for (size_t i = 0; i < m_wlen - n2; ++i) {
+                    m_mashbuf[c][i] = m_mashbuf[c][i + n2];
+                }
+
+                for (size_t i = m_wlen - n2; i < m_wlen; ++i) {
+                    m_mashbuf[c][i] = 0.0f;
+                }
+            }
+
+            m_prevTransient = transient;
+
+            for (size_t i = 0; i < m_wlen - n2; ++i) {
+                m_modulationbuf[i] = m_modulationbuf[i + n2];
+	    }
+
+	    for (size_t i = m_wlen - n2; i < m_wlen; ++i) {
+                m_modulationbuf[i] = 0.0f;
+	    }
+
+            if (!transient) m_n2 = n2;
+	}
+
+
+#ifdef DEBUG_PHASE_VOCODER_TIME_STRETCHER
+	std::cerr << "loop ended: inbuf read space " << m_inbuf[0]->getReadSpace() << ", outbuf write space " << m_outbuf[0]->getWriteSpace() << std::endl;
+#endif
+    }
+
+#ifdef DEBUG_PHASE_VOCODER_TIME_STRETCHER
+    std::cerr << "PhaseVocoderTimeStretcher::putInput returning" << std::endl;
+#endif
+
+//    std::cerr << "ratio: nominal: " << getRatio() << " actual: "
+//              << m_total2 << "/" << m_total1 << " = " << float(m_total2) / float(m_total1) << " ideal: " << m_ratio << std::endl;
+}
+
+size_t
+PhaseVocoderTimeStretcher::getAvailableOutputSamples() const
+{
+    QMutexLocker locker(m_mutex);
+
+    return m_outbuf[0]->getReadSpace();
+}
+
+void
+PhaseVocoderTimeStretcher::getOutput(float **output, size_t samples)
+{
+    QMutexLocker locker(m_mutex);
+
+    if (m_outbuf[0]->getReadSpace() < samples) {
+	std::cerr << "WARNING: PhaseVocoderTimeStretcher::getOutput: not enough data (yet?) (" << m_outbuf[0]->getReadSpace() << " < " << samples << ")" << std::endl;
+	size_t fill = samples - m_outbuf[0]->getReadSpace();
+        for (size_t c = 0; c < m_channels; ++c) {
+            for (size_t i = 0; i < fill; ++i) {
+                output[c][i] = 0.0;
+            }
+            m_outbuf[c]->read(output[c] + fill, m_outbuf[c]->getReadSpace());
+        }
+    } else {
+#ifdef DEBUG_PHASE_VOCODER_TIME_STRETCHER
+	std::cerr << "enough data - writing " << samples << " from outbuf" << std::endl;
+#endif
+        for (size_t c = 0; c < m_channels; ++c) {
+            m_outbuf[c]->read(output[c], samples);
+        }
+    }
+
+#ifdef DEBUG_PHASE_VOCODER_TIME_STRETCHER
+    std::cerr << "PhaseVocoderTimeStretcher::getOutput returning" << std::endl;
+#endif
+}
+
+void
+PhaseVocoderTimeStretcher::analyseBlock(size_t c, float *buf)
+{
+    size_t i;
+
+    // buf contains m_wlen samples
+
+#ifdef DEBUG_PHASE_VOCODER_TIME_STRETCHER
+    std::cerr << "PhaseVocoderTimeStretcher::analyseBlock (channel " << c << ")" << std::endl;
+#endif
+
+    m_analysisWindow->cut(buf);
+
+    for (i = 0; i < m_wlen/2; ++i) {
+	float temp = buf[i];
+	buf[i] = buf[i + m_wlen/2];
+	buf[i + m_wlen/2] = temp;
+    }
+
+    for (i = 0; i < m_wlen; ++i) {
+	m_time[c][i] = buf[i];
+    }
+
+    fftf_execute(m_plan[c]); // m_time -> m_freq
+}
+
+bool
+PhaseVocoderTimeStretcher::isTransient()
+{
+    int count = 0;
+
+    for (size_t i = 0; i <= m_wlen/2; ++i) {
+
+        float real = 0.f, imag = 0.f;
+
+        for (size_t c = 0; c < m_channels; ++c) {
+            real += m_freq[c][i][0];
+            imag += m_freq[c][i][1];
+        }
+
+        float sqrmag = (real * real + imag * imag);
+
+        if (m_prevTransientMag[i] > 0.f) {
+            float diff = 10.f * log10f(sqrmag / m_prevTransientMag[i]);
+            if (diff > 3.f) ++count;
+        }
+
+        m_prevTransientMag[i] = sqrmag;
+    }
+
+    bool isTransient = false;
+
+//    if (count > m_transientThreshold &&
+//        count > m_prevTransientScore * 1.2) {
+    if (count > m_prevTransientScore &&
+        count > m_transientThreshold &&
+        count - m_prevTransientScore > int(m_wlen) / 20) {
+        isTransient = true;
+
+
+//        std::cerr << "isTransient (count = " << count << ", prev = " << m_prevTransientScore << ", diff = " << count - m_prevTransientScore << ", ratio = " << (m_totalCount > 0 ? (float (m_n2sum) / float(m_totalCount * m_n1)) : 1.f) << ", ideal = " << m_ratio << ")" << std::endl;
+//    } else {
+//        std::cerr << " !transient (count = " << count << ", prev = " << m_prevTransientScore << ", diff = " << count - m_prevTransientScore << ")" << std::endl;
+    }
+
+    m_prevTransientScore = count;
+
+    return isTransient;
+}
+
+void
+PhaseVocoderTimeStretcher::synthesiseBlock(size_t c,
+                                           float *out,
+                                           float *modulation,
+                                           size_t lastStep)
+{
+    bool unchanged = (lastStep == m_n1);
+
+    for (size_t i = 0; i <= m_wlen/2; ++i) {
+		
+        float phase = princargf(atan2f(m_freq[c][i][1], m_freq[c][i][0]));
+        float adjustedPhase = phase;
+
+        if (!unchanged) {
+
+            float omega = (2 * M_PI * m_n1 * i) / m_wlen;
+	
+            float expectedPhase = m_prevPhase[c][i] + omega;
+
+            float phaseError = princargf(phase - expectedPhase);
+
+            float phaseIncrement = (omega + phaseError) / m_n1;
+            
+            adjustedPhase = m_prevAdjustedPhase[c][i] +
+                lastStep * phaseIncrement;
+            
+            float mag = sqrtf(m_freq[c][i][0] * m_freq[c][i][0] +
+                              m_freq[c][i][1] * m_freq[c][i][1]);
+            
+            float real = mag * cosf(adjustedPhase);
+            float imag = mag * sinf(adjustedPhase);
+            m_freq[c][i][0] = real;
+            m_freq[c][i][1] = imag;
+        }
+
+        m_prevPhase[c][i] = phase;
+        m_prevAdjustedPhase[c][i] = adjustedPhase;
+    }
+
+    fftf_execute(m_iplan[c]); // m_freq -> m_time, inverse fft
+
+    for (size_t i = 0; i < m_wlen/2; ++i) {
+        float temp = m_time[c][i];
+        m_time[c][i] = m_time[c][i + m_wlen/2];
+        m_time[c][i + m_wlen/2] = temp;
+    }
+    
+    for (size_t i = 0; i < m_wlen; ++i) {
+        m_time[c][i] = m_time[c][i] / m_wlen;
+    }
+
+    m_synthesisWindow->cut(m_time[c]);
+
+    for (size_t i = 0; i < m_wlen; ++i) {
+        out[i] += m_time[c][i];
+    }
+
+    if (modulation) {
+
+        float area = m_analysisWindow->getArea();
+
+        for (size_t i = 0; i < m_wlen; ++i) {
+            float val = m_synthesisWindow->getValue(i);
+            modulation[i] += val * area;
+        }
+    }
+}
+
+