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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 "AudioFileSizeEstimator.h"
#include "WavFileReader.h"
#include <QFile>
#include "base/Debug.h"
namespace sv {
sv_frame_t
AudioFileSizeEstimator::estimate(FileSource source,
sv_samplerate_t targetRate)
{
sv_frame_t estimate = 0;
SVDEBUG << "AudioFileSizeEstimator: Sample count estimate requested for file \""
<< source.getLocalFilename() << "\"" << endl;
#ifndef WITHOUT_LIBSNDFILE
// Most of our file readers don't know the sample count until
// after they've finished decoding. This is an exception:
WavFileReader *reader = new WavFileReader(source);
if (reader->isOK() &&
reader->getChannelCount() > 0 &&
reader->getFrameCount() > 0) {
sv_frame_t samples =
reader->getFrameCount() * reader->getChannelCount();
sv_samplerate_t rate = reader->getSampleRate();
if (targetRate != 0.0 && targetRate != rate) {
samples = sv_frame_t(double(samples) * targetRate / rate);
}
SVDEBUG << "AudioFileSizeEstimator: WAV file reader accepts this file, reports "
<< samples << " samples" << endl;
estimate = samples;
} else {
SVDEBUG << "AudioFileSizeEstimator: WAV file reader doesn't like this file, "
<< "estimating from file size and extension instead" << endl;
}
delete reader;
reader = nullptr;
#endif
if (estimate == 0) {
// The remainder just makes an estimate based on the file size
// and extension. We don't even know its sample rate at this
// point, so the following is a wild guess.
double rateRatio = 1.0;
if (targetRate != 0.0) {
rateRatio = targetRate / 44100.0;
}
QString extension = source.getExtension();
source.waitForData();
if (!source.isOK()) return 0;
sv_frame_t sz = 0;
{
QFile f(source.getLocalFilename());
if (f.open(QFile::ReadOnly)) {
SVDEBUG << "AudioFileSizeEstimator: opened file, size is "
<< f.size() << endl;
sz = f.size();
f.close();
}
}
if (extension == "ogg" || extension == "oga" ||
extension == "m4a" || extension == "mp3" ||
extension == "wma" || extension == "opus") {
// Usually a lossy file. Compression ratios can vary
// dramatically, but don't usually exceed about 20x compared
// to 16-bit PCM (e.g. a 128kbps mp3 has 11x ratio over WAV at
// 44.1kHz). We can estimate the number of samples to be file
// size x 20, divided by 2 as we're comparing with 16-bit PCM.
estimate = sv_frame_t(double(sz) * 10 * rateRatio);
}
if (extension == "flac") {
// FLAC usually takes up a bit more than half the space of
// 16-bit PCM. So the number of 16-bit samples is roughly the
// same as the file size in bytes. As above, let's be
// conservative.
estimate = sv_frame_t(double(sz) * 1.2 * rateRatio);
}
SVDEBUG << "AudioFileSizeEstimator: for extension \""
<< extension << "\", estimate = " << estimate << " samples" << endl;
}
return estimate;
}
} // end namespace sv
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