view audioio/PhaseVocoderTimeStretcher.cpp @ 90:51ea003f8f99

* Add large chunks of context help in the optional status bar
* Add an extra overlay mode in which even the centre frame is disabled
* Fixes to FTP retrieval

/* -*- 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 *)fftwf_malloc(sizeof(float) * (m_wlen / 2 + 1));
    m_prevTransientScore = 0;
    m_prevTransient = false;

    m_tempbuf = (float *)fftwf_malloc(sizeof(float) * m_wlen);

    m_time = new float *[m_channels];
    m_freq = new fftwf_complex *[m_channels];
    m_plan = new fftwf_plan[m_channels];
    m_iplan = new fftwf_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 *)fftwf_malloc(sizeof(float) * m_wlen);
        
    for (size_t c = 0; c < m_channels; ++c) {

        m_prevPhase[c] = (float *)fftwf_malloc(sizeof(float) * (m_wlen / 2 + 1));
        m_prevAdjustedPhase[c] = (float *)fftwf_malloc(sizeof(float) * (m_wlen / 2 + 1));

        m_time[c] = (float *)fftwf_malloc(sizeof(float) * m_wlen);
        m_freq[c] = (fftwf_complex *)fftwf_malloc(sizeof(fftwf_complex) *
                                                  (m_wlen / 2 + 1));
        
        m_plan[c] = fftwf_plan_dft_r2c_1d(m_wlen, m_time[c], m_freq[c], FFTW_ESTIMATE);
        m_iplan[c] = fftwf_plan_dft_c2r_1d(m_wlen, m_freq[c], m_time[c], FFTW_ESTIMATE);

        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 *)fftwf_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 = 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) {

        fftwf_destroy_plan(m_plan[c]);
        fftwf_destroy_plan(m_iplan[c]);

        fftwf_free(m_time[c]);
        fftwf_free(m_freq[c]);

        fftwf_free(m_mashbuf[c]);
        fftwf_free(m_prevPhase[c]);
        fftwf_free(m_prevAdjustedPhase[c]);

        delete m_inbuf[c];
        delete m_outbuf[c];
    }

    fftwf_free(m_tempbuf);
    fftwf_free(m_modulationbuf);
    fftwf_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 squashy = m_n2sum - fixed;

                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];
    }

    fftwf_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 > 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;
    }

    fftwf_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;
        }
    }
}