comparison audioio/PhaseVocoderTimeStretcher.cpp @ 14:085f34c73939

* IntegerTimeStretcher -> PhaseVocoderTimeStretcher (no longer confined to
  integer multiples)

+/* -*- 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.
+    
+    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>
+
+//#define DEBUG_PHASE_VOCODER_TIME_STRETCHER 1
+
+PhaseVocoderTimeStretcher::PhaseVocoderTimeStretcher(float ratio,
+					   size_t maxProcessInputBlockSize,
+					   size_t inputIncrement,
+					   size_t windowSize,
+					   WindowType windowType) :
+    m_ratio(ratio),
+    m_n1(inputIncrement),
+    m_n2(lrintf(m_n1 * ratio)),
+    m_wlen(std::max(windowSize, m_n2 * 2)),
+    m_inbuf(m_wlen),
+    m_outbuf(maxProcessInputBlockSize * ratio + 1024) //!!!
+{
+    m_window = new Window<float>(windowType, m_wlen),
+
+    m_time = (fftwf_complex *)fftwf_malloc(sizeof(fftwf_complex) * m_wlen);
+    m_freq = (fftwf_complex *)fftwf_malloc(sizeof(fftwf_complex) * m_wlen);
+    m_dbuf = (float *)fftwf_malloc(sizeof(float) * m_wlen);
+    m_mashbuf = (float *)fftwf_malloc(sizeof(float) * m_wlen);
+    m_modulationbuf = (float *)fftwf_malloc(sizeof(float) * m_wlen);
+    m_prevPhase = (float *)fftwf_malloc(sizeof(float) * m_wlen);
+    m_prevAdjustedPhase = (float *)fftwf_malloc(sizeof(float) * m_wlen);
+
+    m_plan = fftwf_plan_dft_1d(m_wlen, m_time, m_freq, FFTW_FORWARD, FFTW_ESTIMATE);
+    m_iplan = fftwf_plan_dft_c2r_1d(m_wlen, m_freq, m_dbuf, FFTW_ESTIMATE);
+
+    for (int i = 0; i < m_wlen; ++i) {
+	m_mashbuf[i] = 0.0;
+	m_modulationbuf[i] = 0.0;
+        m_prevPhase[i] = 0.0;
+        m_prevAdjustedPhase[i] = 0.0;
+    }
+}
+
+PhaseVocoderTimeStretcher::~PhaseVocoderTimeStretcher()
+{
+    std::cerr << "PhaseVocoderTimeStretcher::~PhaseVocoderTimeStretcher" << std::endl;
+
+    fftwf_destroy_plan(m_plan);
+    fftwf_destroy_plan(m_iplan);
+
+    fftwf_free(m_time);
+    fftwf_free(m_freq);
+    fftwf_free(m_dbuf);
+    fftwf_free(m_mashbuf);
+    fftwf_free(m_modulationbuf);
+    fftwf_free(m_prevPhase);
+    fftwf_free(m_prevAdjustedPhase);
+
+    delete m_window;
+}	
+
+size_t
+PhaseVocoderTimeStretcher::getProcessingLatency() const
+{
+    return getWindowSize() - getInputIncrement();
+}
+
+void
+PhaseVocoderTimeStretcher::process(float *input, float *output, size_t samples)
+{
+    // 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;
+
+#ifdef DEBUG_PHASE_VOCODER_TIME_STRETCHER
+    std::cerr << "PhaseVocoderTimeStretcher::process(" << samples << ", consumed = " << consumed << "), writable " << m_inbuf.getWriteSpace() <<", readable "<< m_outbuf.getReadSpace() << std::endl;
+#endif
+
+    while (consumed < samples) {
+
+	size_t writable = m_inbuf.getWriteSpace();
+	writable = std::min(writable, samples - consumed);
+
+	if (writable == 0) {
+	    //!!! then what? I don't think this should happen, but
+	    std::cerr << "WARNING: PhaseVocoderTimeStretcher::process: writable == 0" << std::endl;
+	    break;
+	}
+
+#ifdef DEBUG_PHASE_VOCODER_TIME_STRETCHER
+	std::cerr << "writing " << writable << " from index " << consumed << " to inbuf, consumed will be " << consumed + writable << std::endl;
+#endif
+	m_inbuf.write(input + consumed, writable);
+	consumed += writable;
+
+	while (m_inbuf.getReadSpace() >= m_wlen &&
+	       m_outbuf.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.
+
+	    size_t got = m_inbuf.peek(m_dbuf, m_wlen);
+	    assert(got == m_wlen);
+		
+	    processBlock(m_dbuf, m_mashbuf, m_modulationbuf);
+
+#ifdef DEBUG_PHASE_VOCODER_TIME_STRETCHER
+	    std::cerr << "writing first " << m_n2 << " from mashbuf, skipping " << m_n1 << " on inbuf " << std::endl;
+#endif
+	    m_inbuf.skip(m_n1);
+
+            for (size_t i = 0; i < m_n2; ++i) {
+                if (m_modulationbuf[i] > 0.f) {
+                    m_mashbuf[i] /= m_modulationbuf[i];
+                }
+            }
+
+	    m_outbuf.write(m_mashbuf, m_n2);
+
+	    for (size_t i = 0; i < m_wlen - m_n2; ++i) {
+		m_mashbuf[i] = m_mashbuf[i + m_n2];
+                m_modulationbuf[i] = m_modulationbuf[i + m_n2];
+	    }
+
+	    for (size_t i = m_wlen - m_n2; i < m_wlen; ++i) {
+		m_mashbuf[i] = 0.0f;
+                m_modulationbuf[i] = 0.0f;
+	    }
+	}
+
+//	std::cerr << "WARNING: PhaseVocoderTimeStretcher::process: writespace not enough for output increment (" << m_outbuf.getWriteSpace() << " < " << m_n2 << ")" << std::endl;
+//	}
+
+#ifdef DEBUG_PHASE_VOCODER_TIME_STRETCHER
+	std::cerr << "loop ended: inbuf read space " << m_inbuf.getReadSpace() << ", outbuf write space " << m_outbuf.getWriteSpace() << std::endl;
+#endif
+    }
+
+    size_t toRead = lrintf(samples * m_ratio);
+
+    if (m_outbuf.getReadSpace() < toRead) {
+	std::cerr << "WARNING: PhaseVocoderTimeStretcher::process: not enough data (yet?) (" << m_outbuf.getReadSpace() << " < " << toRead << ")" << std::endl;
+	size_t fill = toRead - m_outbuf.getReadSpace();
+	for (size_t i = 0; i < fill; ++i) {
+	    output[i] = 0.0;
+	}
+	m_outbuf.read(output + fill, m_outbuf.getReadSpace());
+    } else {
+#ifdef DEBUG_PHASE_VOCODER_TIME_STRETCHER
+	std::cerr << "enough data - writing " << toRead << " from outbuf" << std::endl;
+#endif
+	m_outbuf.read(output, toRead);
+    }
+
+#ifdef DEBUG_PHASE_VOCODER_TIME_STRETCHER
+    std::cerr << "PhaseVocoderTimeStretcher::process returning" << std::endl;
+#endif
+}
+
+void
+PhaseVocoderTimeStretcher::processBlock(float *buf, float *out, float *modulation)
+{
+    size_t i;
+
+    // buf contains m_wlen samples; out contains enough space for
+    // m_wlen * ratio samples (we mix into out, rather than replacing)
+
+#ifdef DEBUG_PHASE_VOCODER_TIME_STRETCHER
+    std::cerr << "PhaseVocoderTimeStretcher::processBlock" << std::endl;
+#endif
+
+    m_window->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[i][0] = buf[i];
+	m_time[i][1] = 0.0;
+    }
+
+    fftwf_execute(m_plan); // m_time -> m_freq
+
+    for (i = 0; i < m_wlen; ++i) {
+	
+	float mag = sqrtf(m_freq[i][0] * m_freq[i][0] +
+			  m_freq[i][1] * m_freq[i][1]);
+		
+        float phase = princargf(atan2f(m_freq[i][1], m_freq[i][0]));
+
+        float omega = (2 * M_PI * m_n1 * i) / m_wlen;
+	
+        float expectedPhase = m_prevPhase[i] + omega;
+
+        float phaseError = princargf(phase - expectedPhase);
+
+        float phaseIncrement = (omega + phaseError) / m_n1;
+
+        float adjustedPhase = m_prevAdjustedPhase[i] + m_n2 * phaseIncrement;
+	
+	float real = mag * cosf(adjustedPhase);
+	float imag = mag * sinf(adjustedPhase);
+	m_freq[i][0] = real;
+	m_freq[i][1] = imag;
+
+        m_prevPhase[i] = phase;
+        m_prevAdjustedPhase[i] = adjustedPhase;
+    }
+    
+    fftwf_execute(m_iplan); // m_freq -> in, inverse fft
+    
+    for (i = 0; i < m_wlen/2; ++i) {
+	float temp = buf[i] / m_wlen;
+	buf[i] = buf[i + m_wlen/2] / m_wlen;
+	buf[i + m_wlen/2] = temp;
+    }
+    
+    m_window->cut(buf);
+/*    
+    int div = m_wlen / m_n2;
+    if (div > 1) div /= 2;
+    for (i = 0; i < m_wlen; ++i) {
+	buf[i] /= div;
+    }
+*/
+    for (i = 0; i < m_wlen; ++i) {
+	out[i] += buf[i];
+        modulation[i] += m_window->getValue(i);
+    }
+}