Mercurial > hg > svcore
changeset 1536:eb10ed56d5a4 zoom
Add custom waveform oversampler method
author | Chris Cannam |
---|---|
date | Wed, 26 Sep 2018 13:03:46 +0100 |
parents | 22fe34dd7f26 |
children | c5092ca1c6e5 |
files | data/model/WaveformOversampler.cpp data/model/WaveformOversampler.h files.pri |
diffstat | 3 files changed, 348 insertions(+), 0 deletions(-) [+] |
line wrap: on
line diff
--- /dev/null Thu Jan 01 00:00:00 1970 +0000 +++ b/data/model/WaveformOversampler.cpp Wed Sep 26 13:03:46 2018 +0100 @@ -0,0 +1,289 @@ +/* -*- 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. +*/ + +#include "WaveformOversampler.h" + +#include "data/model/DenseTimeValueModel.h" + +floatvec_t +WaveformOversampler::getOversampledData(const DenseTimeValueModel *source, + int channel, + sv_frame_t sourceStartFrame, + sv_frame_t sourceFrameCount, + int oversampleBy) +{ + // Oversampled at a fixed ratio of m_filterRatio + floatvec_t fixedRatio = getFixedRatioData(source, channel, + sourceStartFrame, + sourceFrameCount); + sv_frame_t fixedCount = fixedRatio.size(); + sv_frame_t targetCount = (fixedCount / m_filterRatio) * oversampleBy; + + // And apply linear interpolation to the desired factor + + floatvec_t result(targetCount, 0.f); + + for (int i = 0; i < targetCount; ++i) { + double pos = (double(i) / oversampleBy) * m_filterRatio; + double diff = pos - floor(pos); + int ix = int(floor(pos)); + float interpolated = (1.0 - diff) * fixedRatio[ix]; + if (in_range_for(fixedRatio, ix + 1)) { + interpolated += diff * fixedRatio[ix + 1]; + } + result[i] = interpolated; + } + + return result; +} + +floatvec_t +WaveformOversampler::getFixedRatioData(const DenseTimeValueModel *source, + int channel, + sv_frame_t sourceStartFrame, + sv_frame_t sourceFrameCount) +{ + sv_frame_t sourceLength = source->getEndFrame(); + + if (sourceStartFrame + sourceFrameCount > sourceLength) { + sourceFrameCount = sourceLength - sourceStartFrame; + if (sourceFrameCount <= 0) return {}; + } + + sv_frame_t targetFrameCount = sourceFrameCount * m_filterRatio; + + sv_frame_t filterLength = m_filter.size(); // NB this is known to be odd + sv_frame_t filterTailOut = (filterLength - 1) / 2; + sv_frame_t filterTailIn = filterTailIn / m_filterRatio; + + floatvec_t oversampled(targetFrameCount, 0.f); + + sv_frame_t i0 = sourceStartFrame - filterTailIn; + if (i0 < 0) { + i0 = 0; + } + sv_frame_t i1 = sourceStartFrame + sourceFrameCount + filterTailIn; + if (i1 > sourceLength) { + i1 = sourceLength; + } + + floatvec_t sourceData = source->getData(channel, i0, i1 - i0); + + for (sv_frame_t i = i0; i < i1; ++i) { + float v = sourceData[i - i0]; + sv_frame_t outOffset = + (i - sourceStartFrame) * m_filterRatio - filterTailOut; + for (sv_frame_t j = 0; j < filterLength; ++j) { + sv_frame_t outIndex = outOffset + j; + if (outIndex < 0 || outIndex >= targetFrameCount) { + continue; + } + oversampled[outIndex] += v * m_filter[j]; + } + } + + return oversampled; +} + +int +WaveformOversampler::m_filterRatio = 8; + +/// Precalculated windowed sinc FIR filter for oversampling ratio of 8 +floatvec_t +WaveformOversampler::m_filter { + 2.0171043153063023E-4, 2.887198196326776E-4, + 3.410439309101285E-4, 3.4267123819805857E-4, + 2.843462511901066E-4, 1.6636986363946504E-4, + -4.5940658605786285E-18, -1.9299665002484582E-4, + -3.8279951732549946E-4, -5.357990649609105E-4, + -6.201170748425957E-4, -6.11531555444137E-4, + -4.987822892899791E-4, -2.872272251922189E-4, + -7.822991648518709E-19, 3.2382854144162815E-4, 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--- /dev/null Thu Jan 01 00:00:00 1970 +0000 +++ b/data/model/WaveformOversampler.h Wed Sep 26 13:03:46 2018 +0100 @@ -0,0 +1,57 @@ +/* -*- 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" + +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; +}; + +#endif
--- a/files.pri Mon Sep 24 14:40:22 2018 +0100 +++ b/files.pri Wed Sep 26 13:03:46 2018 +0100 @@ -96,6 +96,7 @@ data/model/SparseValueModel.h \ data/model/TabularModel.h \ data/model/TextModel.h \ + data/model/WaveformOversampler.h \ data/model/WaveFileModel.h \ data/model/ReadOnlyWaveFileModel.h \ data/model/WritableWaveFileModel.h \ @@ -211,6 +212,7 @@ data/model/PowerOfSqrtTwoZoomConstraint.cpp \ data/model/PowerOfTwoZoomConstraint.cpp \ data/model/RangeSummarisableTimeValueModel.cpp \ + data/model/WaveformOversampler.cpp \ data/model/WaveFileModel.cpp \ data/model/ReadOnlyWaveFileModel.cpp \ data/model/WritableWaveFileModel.cpp \