view data/fileio/AudioFileSizeEstimator.cpp @ 1247:8f076d02569a piper

Make SVDEBUG always write to a log file -- formerly this was disabled in NDEBUG builds. I think there's little use to that, it just means that we keep adding more cerr debug output because we aren't getting the log we need. And SVDEBUG logging is not usually used in tight loops, I don't think the performance overhead is too serious.

Also update the About box.

/* -*- 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>

//#define DEBUG_AUDIO_FILE_SIZE_ESTIMATOR 1

sv_frame_t
AudioFileSizeEstimator::estimate(FileSource source,
				 sv_samplerate_t targetRate)
{
    sv_frame_t estimate = 0;
    
    // 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);
	}
	delete reader;
	estimate = samples;
    }

    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)) {
#ifdef DEBUG_AUDIO_FILE_SIZE_ESTIMATOR
		cerr << "opened file, size is "  << f.size() << endl;
#endif
		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);
	}

#ifdef DEBUG_AUDIO_FILE_SIZE_ESTIMATOR
	cerr << "AudioFileSizeEstimator: for extension " << extension << ", estimate = " << estimate << endl;
#endif
    }

#ifdef DEBUG_AUDIO_FILE_SIZE_ESTIMATOR
    cerr << "estimate = " << estimate << endl;
#endif
    
    return estimate;
}