Mercurial > hg > svcore
view data/model/WaveformOversampler.cpp @ 1717:417528c41e66
Build fix for Travis
author | Chris Cannam |
---|---|
date | Fri, 17 May 2019 12:41:06 +0100 |
parents | ca43af0dcab7 |
children | 074b860a7828 |
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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. */ #include "WaveformOversampler.h" #include "base/Profiler.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) { Profiler profiler("WaveformOversampler::getOversampledData"); // 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)); double interpolated = (1.0 - diff) * fixedRatio[ix]; if (in_range_for(fixedRatio, ix + 1)) { interpolated += diff * fixedRatio[ix + 1]; } result[i] = float(interpolated); } return result; } floatvec_t WaveformOversampler::getFixedRatioData(const DenseTimeValueModel *source, int channel, sv_frame_t sourceStartFrame, sv_frame_t sourceFrameCount) { Profiler profiler("WaveformOversampler::getFixedRatioData"); 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 = filterTailOut / 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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