comparison data/fileio/AudioFileSizeEstimator.cpp @ 1126:39019ce29178 tony-2.0-integration

Merge through to branch for Tony 2.0
author Chris Cannam
date Thu, 20 Aug 2015 14:54:21 +0100
parents 393134235fa0
children 513e4d67d8df
comparison
equal deleted inserted replaced
1119:e22bfe8ca248 1126:39019ce29178
1 /* -*- c-basic-offset: 4 indent-tabs-mode: nil -*- vi:set ts=8 sts=4 sw=4: */
2
3 /*
4 Sonic Visualiser
5 An audio file viewer and annotation editor.
6 Centre for Digital Music, Queen Mary, University of London.
7
8 This program is free software; you can redistribute it and/or
9 modify it under the terms of the GNU General Public License as
10 published by the Free Software Foundation; either version 2 of the
11 License, or (at your option) any later version. See the file
12 COPYING included with this distribution for more information.
13 */
14
15 #include "AudioFileSizeEstimator.h"
16
17 #include "WavFileReader.h"
18
19 #include <QFile>
20
21 //#define DEBUG_AUDIO_FILE_SIZE_ESTIMATOR 1
22
23 sv_frame_t
24 AudioFileSizeEstimator::estimate(FileSource source,
25 sv_samplerate_t targetRate)
26 {
27 sv_frame_t estimate = 0;
28
29 // Most of our file readers don't know the sample count until
30 // after they've finished decoding. This is an exception:
31
32 WavFileReader *reader = new WavFileReader(source);
33 if (reader->isOK() &&
34 reader->getChannelCount() > 0 &&
35 reader->getFrameCount() > 0) {
36 sv_frame_t samples =
37 reader->getFrameCount() * reader->getChannelCount();
38 sv_samplerate_t rate = reader->getSampleRate();
39 if (targetRate != 0.0 && targetRate != rate) {
40 samples = sv_frame_t(double(samples) * targetRate / rate);
41 }
42 delete reader;
43 estimate = samples;
44 }
45
46 if (estimate == 0) {
47
48 // The remainder just makes an estimate based on the file size
49 // and extension. We don't even know its sample rate at this
50 // point, so the following is a wild guess.
51
52 double rateRatio = 1.0;
53 if (targetRate != 0.0) {
54 rateRatio = targetRate / 44100.0;
55 }
56
57 QString extension = source.getExtension();
58
59 source.waitForData();
60 if (!source.isOK()) return 0;
61
62 sv_frame_t sz = 0;
63 {
64 QFile f(source.getLocalFilename());
65 if (f.open(QFile::ReadOnly)) {
66 #ifdef DEBUG_AUDIO_FILE_SIZE_ESTIMATOR
67 cerr << "opened file, size is " << f.size() << endl;
68 #endif
69 sz = f.size();
70 f.close();
71 }
72 }
73
74 if (extension == "ogg" || extension == "oga" ||
75 extension == "m4a" || extension == "mp3" ||
76 extension == "wma") {
77
78 // Usually a lossy file. Compression ratios can vary
79 // dramatically, but don't usually exceed about 20x compared
80 // to 16-bit PCM (e.g. a 128kbps mp3 has 11x ratio over WAV at
81 // 44.1kHz). We can estimate the number of samples to be file
82 // size x 20, divided by 2 as we're comparing with 16-bit PCM.
83
84 estimate = sv_frame_t(double(sz) * 10 * rateRatio);
85 }
86
87 if (extension == "flac") {
88
89 // FLAC usually takes up a bit more than half the space of
90 // 16-bit PCM. So the number of 16-bit samples is roughly the
91 // same as the file size in bytes. As above, let's be
92 // conservative.
93
94 estimate = sv_frame_t(double(sz) * 1.2 * rateRatio);
95 }
96
97 #ifdef DEBUG_AUDIO_FILE_SIZE_ESTIMATOR
98 cerr << "AudioFileSizeEstimator: for extension " << extension << ", estimate = " << estimate << endl;
99 #endif
100 }
101
102 #ifdef DEBUG_AUDIO_FILE_SIZE_ESTIMATOR
103 cerr << "estimate = " << estimate << endl;
104 #endif
105
106 return estimate;
107 }
108