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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 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 SV_WAVEFORM_OVERSAMPLER_H
#define SV_WAVEFORM_OVERSAMPLER_H
#include "base/BaseTypes.h"
namespace sv {
class DenseTimeValueModel;
/** Oversample the sample data from a DenseTimeValueModel by an
* integer factor, on the assumption that the model represents
* audio. Oversampling is carried out using a windowed sinc filter
* for a fixed 8x ratio with further linear interpolation to handle
* other ratios. The aim is not to provide the "best-sounding"
* interpolation, but to provide accurate and predictable projections
* of the theoretical waveform shape for display rendering without
* leaving decisions about interpolation up to a resampler library.
*/
class WaveformOversampler
{
public:
/** Return an oversampled version of the audio data from the given
* source sample range. Will query sufficient source audio before
* and after the requested range (where available) to ensure an
* accurate-looking result after filtering. The returned vector
* will have sourceFrameCount * oversampleBy samples, except when
* truncated because the end of the model was reached.
*/
static floatvec_t getOversampledData(const DenseTimeValueModel &source,
int channel,
sv_frame_t sourceStartFrame,
sv_frame_t sourceFrameCount,
int oversampleBy);
private:
static floatvec_t getFixedRatioData(const DenseTimeValueModel &source,
int channel,
sv_frame_t sourceStartFrame,
sv_frame_t sourceFrameCount);
static int m_filterRatio;
static floatvec_t m_filter;
};
} // end namespace sv
#endif
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