view data/fileio/AudioFileSizeEstimator.cpp @ 1211:5a1198083d9a piper

Pull out model creation into the transformer thread run(), so that all communications with the plugin server happen on a single thread. Then make the model accessor wait for them to be created (which still happens right at the start of processing) before returning.

/* -*- 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;
}