Mercurial > hg > svcore
view data/fileio/AudioFileSizeEstimator.cpp @ 1591:6e68bd92ee21 bqaudiostream
Add flac metadata example
author | Chris Cannam |
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date | Mon, 21 Jan 2019 11:45:51 +0000 |
parents | aadfb395e933 |
children | 70e172e6cc59 f8e3dcbafb4d |
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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" 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; // 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 = 0; 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") { // 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; }