Mercurial > hg > svgui
changeset 1339:b0555b434575 zoom
Add custom waveform oversampler method
author | Chris Cannam |
---|---|
date | Wed, 26 Sep 2018 11:54:25 +0100 |
parents | 6d091348e4e9 |
children | fc3c9971a43a |
files | files.pri layer/WaveformLayer.cpp layer/WaveformLayer.h layer/WaveformOversampler.cpp layer/WaveformOversampler.h |
diffstat | 5 files changed, 353 insertions(+), 27 deletions(-) [+] |
line wrap: on
line diff
--- a/files.pri Tue Sep 25 16:38:50 2018 +0100 +++ b/files.pri Wed Sep 26 11:54:25 2018 +0100 @@ -36,6 +36,7 @@ layer/TimeValueLayer.h \ layer/VerticalScaleLayer.h \ layer/WaveformLayer.h \ + layer/WaveformOversampler.h \ view/AlignmentView.h \ view/Overview.h \ view/Pane.h \ @@ -124,6 +125,7 @@ layer/TimeRulerLayer.cpp \ layer/TimeValueLayer.cpp \ layer/WaveformLayer.cpp \ + layer/WaveformOversampler.cpp \ view/AlignmentView.cpp \ view/Overview.cpp \ view/Pane.cpp \
--- a/layer/WaveformLayer.cpp Tue Sep 25 16:38:50 2018 +0100 +++ b/layer/WaveformLayer.cpp Wed Sep 26 11:54:25 2018 +0100 @@ -23,6 +23,7 @@ #include "ColourDatabase.h" #include "PaintAssistant.h" +#include "WaveformOversampler.h" #include <QPainter> #include <QPixmap> @@ -51,7 +52,6 @@ m_scale(LinearScale), m_middleLineHeight(0.5), m_aggressive(false), - m_oversampler(0), m_cache(0), m_cacheValid(false) { @@ -60,7 +60,6 @@ WaveformLayer::~WaveformLayer() { delete m_cache; - delete m_oversampler; } void @@ -84,14 +83,6 @@ m_cacheValid = false; if (!m_model || !m_model->isOK()) return; - delete m_oversampler; - breakfastquay::Resampler::Parameters params; - params.initialSampleRate = m_model->getSampleRate(); - // Oversampler is initialised with 1 channel: we resample the - // channels individually for practical reasons (because channel - // configuration may vary) - m_oversampler = new breakfastquay::Resampler(params, 1); - connectSignals(m_model); emit modelReplaced(); @@ -709,23 +700,12 @@ } for (int ch = minChannel; ch <= maxChannel; ++ch) { - floatvec_t raw = m_model->getData(ch, rf0, rf1 - rf0); - floatvec_t oversampled(oversampleBy * (frame1 - frame0 + 2 * tail), - 0.f); - //!!! return value, err etc - m_oversampler->reset(); - m_oversampler->resampleInterleaved(oversampled.data(), - oversampled.size(), - raw.data(), - raw.size(), - oversampleBy, - true); + floatvec_t oversampled = WaveformOversampler::getOversampledData + (m_model, ch, frame0, frame1 - frame0, oversampleBy); RangeSummarisableTimeValueModel::RangeBlock rr; - for (sv_frame_t ix = leftTail * oversampleBy; - in_range_for(oversampled, ix + (rightTail * oversampleBy)); - ++ix) { + for (float v: oversampled) { RangeSummarisableTimeValueModel::Range r; - r.sample(oversampled[ix]); + r.sample(v); rr.push_back(r); } ranges.push_back(rr);
--- a/layer/WaveformLayer.h Tue Sep 25 16:38:50 2018 +0100 +++ b/layer/WaveformLayer.h Wed Sep 26 11:54:25 2018 +0100 @@ -258,8 +258,6 @@ mutable std::vector<float> m_effectiveGains; - mutable breakfastquay::Resampler *m_oversampler; - mutable QPixmap *m_cache; mutable bool m_cacheValid; mutable ZoomLevel m_cacheZoomLevel;
--- /dev/null Thu Jan 01 00:00:00 1970 +0000 +++ b/layer/WaveformOversampler.cpp Wed Sep 26 11:54:25 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; + +/// Pre-designed 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/layer/WaveformOversampler.h Wed Sep 26 11:54:25 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