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+/* -*- 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 <[email protected]>,
+ Chris Cannam <[email protected]>,
+ Richard Bown <[email protected]>
+
+ 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