comparison audioio/PhaseVocoderTimeStretcher.h @ 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.
+*/
+
+#ifndef _PHASE_VOCODER_TIME_STRETCHER_H_
+#define _PHASE_VOCODER_TIME_STRETCHER_H_
+
+#include "base/Window.h"
+#include "base/RingBuffer.h"
+
+#include <fftw3.h>
+
+/**
+ * A time stretcher that alters the performance speed of audio,
+ * preserving pitch.  This uses the simple phase vocoder technique
+ * from DAFX pp275-276, adding a block-based stream oriented API.
+ *
+ * Causes significant transient smearing, but sounds good for steady
+ * notes and is generally predictable.
+ */
+
+class PhaseVocoderTimeStretcher
+{
+public:
+    PhaseVocoderTimeStretcher(float ratio,
+			 size_t maxProcessInputBlockSize,
+			 size_t inputIncrement = 64,
+			 size_t windowSize = 2048,
+			 WindowType windowType = HanningWindow);
+    virtual ~PhaseVocoderTimeStretcher();
+
+    /**
+     * Process a block.  The input array contains the given number of
+     * samples; the output has enough space for samples * m_ratio.
+     */
+    void process(float *input, float *output, size_t samples);
+
+    /**
+     * Get the hop size for input.  Smaller values may produce better
+     * results, at a cost in processing time.  Larger values are
+     * faster but increase the likelihood of echo-like effects.  The
+     * default is 64, which is usually pretty good, though heavy on
+     * processor power.
+     */
+    size_t getInputIncrement() const { return m_n1; }
+
+    /**
+     * Get the window size for FFT processing.  Must be larger than
+     * the input and output increments.  The default is 2048.
+     */
+    size_t getWindowSize() const { return m_wlen; }
+
+    /**
+     * Get the window type.  The default is a Hanning window.
+     */
+    WindowType getWindowType() const { return m_window->getType(); }
+
+    float getRatio() const { return m_ratio; }
+    size_t getOutputIncrement() const { return getInputIncrement() * getRatio(); }
+    size_t getProcessingLatency() const;
+
+protected:
+    /**
+     * Process a single phase vocoder frame.
+     * 
+     * Take m_wlen time-domain source samples from in, perform an FFT,
+     * phase shift, and IFFT, and add the results to out (presumably
+     * overlapping parts of existing data from prior frames).
+     *
+     * Also add to the modulation output the results of windowing a
+     * set of 1s with the resynthesis window -- this can then be used
+     * to ensure the output has the correct magnitude in cases where
+     * the window overlap varies or otherwise results in something
+     * other than a flat sum.
+     */
+    void processBlock(float *in, float *out, float *modulation);
+
+    float m_ratio;
+    size_t m_n1;
+    size_t m_n2;
+    size_t m_wlen;
+    Window<float> *m_window;
+
+    fftwf_complex *m_time;
+    fftwf_complex *m_freq;
+    float *m_dbuf;
+    float *m_prevPhase;
+    float *m_prevAdjustedPhase;
+
+    fftwf_plan m_plan;
+    fftwf_plan m_iplan;
+    
+    RingBuffer<float> m_inbuf;
+    RingBuffer<float> m_outbuf;
+    float *m_mashbuf;
+    float *m_modulationbuf;
+};
+
+#endif