Mercurial > hg > sonic-visualiser
view audioio/PhaseVocoderTimeStretcher.h @ 45:6b6bca31ad53
* When setting a new model on a layer, don't delete the old one until after
the new one has been set (a layer's setModel may want to compare against the
old one, as WaveformLayer does)
author | Chris Cannam |
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date | Thu, 05 Oct 2006 11:00:59 +0000 |
parents | e3b32dc5180b |
children | bedc7517b6e8 |
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/* -*- 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> #include <QMutex> /** * A time stretcher that alters the performance speed of audio, * preserving pitch. * * This is based on the straightforward phase vocoder with phase * unwrapping (as in e.g. the DAFX book pp275-), with optional * percussive transient detection to avoid smearing percussive notes * and resynchronise phases, and adding a stream API for real-time * use. Principles and methods from Chris Duxbury, AES 2002 and 2004 * thesis; Emmanuel Ravelli, DAFX 2005; Dan Barry, ISSC 2005 on * percussion detection; code by Chris Cannam. */ class PhaseVocoderTimeStretcher { public: PhaseVocoderTimeStretcher(size_t sampleRate, size_t channels, float ratio, bool sharpen, size_t maxOutputBlockSize); virtual ~PhaseVocoderTimeStretcher(); /** * Return the number of samples that would need to be added via * putInput in order to provoke the time stretcher into doing some * time stretching and making more output samples available. * This will be an estimate, if transient sharpening is on; the * caller may need to do the put/get/test cycle more than once. */ size_t getRequiredInputSamples() const; /** * Put (and possibly process) a given number of input samples. * Number should usually equal the value returned from * getRequiredInputSamples(). */ void putInput(float **input, size_t samples); /** * Get the number of processed samples ready for reading. */ size_t getAvailableOutputSamples() const; /** * Get some processed samples. */ void getOutput(float **output, size_t samples); //!!! and reset? /** * Change the time stretch ratio. */ void setRatio(float ratio); /** * Get the hop size for input. */ size_t getInputIncrement() const { return m_n1; } /** * Get the hop size for output. */ size_t getOutputIncrement() const { return m_n2; } /** * Get the window size for FFT processing. */ size_t getWindowSize() const { return m_wlen; } /** * Get the stretch ratio. */ float getRatio() const { return float(m_n2) / float(m_n1); } /** * Return whether this time stretcher will attempt to sharpen transients. */ bool getSharpening() const { return m_sharpen; } /** * Return the number of channels for this time stretcher. */ size_t getChannelCount() const { return m_channels; } /** * Get the latency added by the time stretcher, in sample frames. * This will be exact if transient sharpening is off, or approximate * if it is on. */ size_t getProcessingLatency() const; protected: /** * Process a single phase vocoder frame from "in" into * m_freq[channel]. */ void analyseBlock(size_t channel, float *in); // into m_freq[channel] /** * Examine m_freq[0..m_channels-1] and return whether a percussive * transient is found. */ bool isTransient(); /** * Resynthesise from m_freq[channel] adding in to "out", * adjusting phases on the basis of a prior step size of lastStep. * Also add the window shape in to the modulation array (if * present) -- for use in ensuring the output has the correct * magnitude afterwards. */ void synthesiseBlock(size_t channel, float *out, float *modulation, size_t lastStep); void initialise(); void calculateParameters(); void cleanup(); bool shouldSharpen() { return m_sharpen && (m_ratio > 0.25); } size_t m_sampleRate; size_t m_channels; size_t m_maxOutputBlockSize; float m_ratio; bool m_sharpen; size_t m_n1; size_t m_n2; size_t m_wlen; Window<float> *m_analysisWindow; Window<float> *m_synthesisWindow; int m_totalCount; int m_transientCount; int m_n2sum; float **m_prevPhase; float **m_prevAdjustedPhase; float *m_prevTransientMag; int m_prevTransientScore; int m_transientThreshold; bool m_prevTransient; float *m_tempbuf; float **m_time; fftwf_complex **m_freq; fftwf_plan *m_plan; fftwf_plan *m_iplan; RingBuffer<float> **m_inbuf; RingBuffer<float> **m_outbuf; float **m_mashbuf; float *m_modulationbuf; QMutex *m_mutex; }; #endif