annotate data/fileio/AudioFileSizeEstimator.cpp @ 1833:21c792334c2e sensible-delimited-data-strings

Rewrite all the DelimitedDataString stuff so as to return vectors of individual cell strings rather than having the classes add the delimiters themselves. Rename accordingly to names based on StringExport. Take advantage of this in the CSV writer code so as to properly quote cells that contain delimiter characters.

/* -*- 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 = nullptr;

    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;
}