Mercurial > hg > svcore
changeset 1341:513e4d67d8df 3.0-integration
Untabify
author | Chris Cannam |
---|---|
date | Fri, 06 Jan 2017 09:15:36 +0000 |
parents | f5f83fb49852 |
children | c0fece5e7755 |
files | data/fileio/AudioFileSizeEstimator.cpp |
diffstat | 1 files changed, 50 insertions(+), 50 deletions(-) [+] |
line wrap: on
line diff
--- a/data/fileio/AudioFileSizeEstimator.cpp Thu Jan 05 14:30:49 2017 +0000 +++ b/data/fileio/AudioFileSizeEstimator.cpp Fri Jan 06 09:15:36 2017 +0000 @@ -22,7 +22,7 @@ sv_frame_t AudioFileSizeEstimator::estimate(FileSource source, - sv_samplerate_t targetRate) + sv_samplerate_t targetRate) { sv_frame_t estimate = 0; @@ -31,71 +31,71 @@ 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; + 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; - } + // 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(); + QString extension = source.getExtension(); - source.waitForData(); - if (!source.isOK()) return 0; + source.waitForData(); + if (!source.isOK()) return 0; - sv_frame_t sz = 0; - { - QFile f(source.getLocalFilename()); - if (f.open(QFile::ReadOnly)) { + 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; + cerr << "opened file, size is " << f.size() << endl; #endif - sz = f.size(); - f.close(); - } - } + sz = f.size(); + f.close(); + } + } - if (extension == "ogg" || extension == "oga" || - extension == "m4a" || extension == "mp3" || - extension == "wma") { + 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. + // 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); - } + 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. + 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); - } + estimate = sv_frame_t(double(sz) * 1.2 * rateRatio); + } #ifdef DEBUG_AUDIO_FILE_SIZE_ESTIMATOR - cerr << "AudioFileSizeEstimator: for extension " << extension << ", estimate = " << estimate << endl; + cerr << "AudioFileSizeEstimator: for extension " << extension << ", estimate = " << estimate << endl; #endif }