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Diffstat (limited to 'src/sound/AudioTimeStretcher.h')
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diff --git a/src/sound/AudioTimeStretcher.h b/src/sound/AudioTimeStretcher.h new file mode 100644 index 0000000..c5d0170 --- /dev/null +++ b/src/sound/AudioTimeStretcher.h @@ -0,0 +1,221 @@ +/* -*- c-basic-offset: 4 indent-tabs-mode: nil -*- vi:set ts=8 sts=4 sw=4: */ + +/* + Rosegarden + A MIDI and audio sequencer and musical notation editor. + + This program is Copyright 2000-2008 + Guillaume Laurent <glaurent@telegraph-road.org>, + Chris Cannam <cannam@all-day-breakfast.com>, + Richard Bown <richard.bown@ferventsoftware.com> + + The moral rights of Guillaume Laurent, Chris Cannam, and Richard + Bown to claim authorship of this work have been asserted. + + Other copyrights also apply to some parts of this work. Please + see the AUTHORS file and individual file headers for details. + + 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. +*/ + +/* + This file is derived from + + 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. +*/ + +#ifndef _AUDIO_TIME_STRETCHER_H_ +#define _AUDIO_TIME_STRETCHER_H_ + +#include "SampleWindow.h" +#include "RingBuffer.h" + +#include <fftw3.h> +#include <pthread.h> +#include <list> + +namespace Rosegarden +{ + +/** + * 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 AudioTimeStretcher +{ +public: + AudioTimeStretcher(size_t sampleRate, + size_t channels, + float ratio, + bool sharpen, + size_t maxOutputBlockSize); + virtual ~AudioTimeStretcher(); + + /** + * 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; + SampleWindow<float> *m_analysisWindow; + SampleWindow<float> *m_synthesisWindow; + + int m_totalCount; + int m_transientCount; + + int m_n2sum; + int m_n2total; + std::list<int> m_n2list; + int m_adjustCount; + + 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; + + mutable pthread_mutex_t m_mutex; +}; + +} + + +#endif |