1
2
3
4
5
6
7
8
9
10
11
12
13
14
15
16
17
18
19
20
21
22
23
24
25
26
27
28
29
30
31
32
33
34
35
36
37
38
39
40
41
42
43
44
45
46
47
48
49
50
51
52
53
54
55
56
57
58
59
60
61
62
63
64
65
66
67
68
69
70
71
72
73
74
75
76
77
78
79
80
81
82
83
84
85
86
87
88
89
90
91
92
93
94
95
96
97
98
99
100
101
102
103
104
105
106
107
108
109
110
111
112
113
114
115
116
117
118
119
120
121
122
123
124
125
126
127
128
129
130
131
132
133
134
135
136
137
138
139
140
141
142
143
144
145
146
147
148
149
150
151
152
153
154
155
156
157
158
159
160
161
162
163
164
165
166
167
168
169
170
171
172
173
174
175
176
177
178
179
180
181
182
183
184
185
186
187
188
189
190
191
192
193
194
195
196
197
198
199
200
201
202
203
204
205
206
207
208
209
210
211
212
213
|
/* Copyright (C) 2004-2012 Savoir-Faire Linux Inc.
* Author: Yan Morin <yan.morin@savoirfairelinux.com>
*
* Inspired by tonegenerator of
* Laurielle Lea <laurielle.lea@savoirfairelinux.com> (2004)
* Inspired by ringbuffer of Audacity Project
*
* 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 3 of the License, or
* (at your option) any later version.
*
* This program is distributed in the hope that it will be useful,
* but WITHOUT ANY WARRANTY; without even the implied warranty of
* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
* GNU General Public License for more details.
*
* You should have received a copy of the GNU General Public License
* along with this program; if not, write to the Free Software
* Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA.
*
* Additional permission under GNU GPL version 3 section 7:
*
* If you modify this program, or any covered work, by linking or
* combining it with the OpenSSL project's OpenSSL library (or a
* modified version of that library), containing parts covered by the
* terms of the OpenSSL or SSLeay licenses, Savoir-Faire Linux Inc.
* grants you additional permission to convey the resulting work.
* Corresponding Source for a non-source form of such a combination
* shall include the source code for the parts of OpenSSL used as well
* as that of the covered work.
*/
#include <fstream>
#include <cmath>
#include <samplerate.h>
#include <cstring>
#include <vector>
#include <climits>
#include "audiofile.h"
#include "audio/codecs/audiocodec.h"
#include "audio/samplerateconverter.h"
#include "logger.h"
RawFile::RawFile(const std::string& name, sfl::AudioCodec *codec, unsigned int sampleRate)
: AudioFile(name, sampleRate), audioCodec_(codec)
{
if (filepath_.empty())
throw AudioFileException("Unable to open audio file: filename is empty");
std::fstream file;
file.open(filepath_.c_str(), std::fstream::in);
if (!file.is_open())
throw AudioFileException("Unable to open audio file");
file.seekg(0, std::ios::end);
size_t length = file.tellg();
file.seekg(0, std::ios::beg);
std::vector<char> fileBuffer(length);
file.read(&fileBuffer[0], length);
file.close();
const unsigned int frameSize = audioCodec_->getFrameSize();
const unsigned int bitrate = audioCodec_->getBitRate() * 1000 / 8;
const unsigned int audioRate = audioCodec_->getClockRate();
const unsigned int encFrameSize = frameSize * bitrate / audioRate;
const unsigned int decodedSize = length * (frameSize / encFrameSize);
SFLDataFormat *monoBuffer = new SFLDataFormat[decodedSize];
SFLDataFormat *bufpos = monoBuffer;
unsigned char *filepos = reinterpret_cast<unsigned char *>(&fileBuffer[0]);
size_ = decodedSize;
while (length >= encFrameSize) {
bufpos += audioCodec_->decode(bufpos, filepos, encFrameSize);
filepos += encFrameSize;
length -= encFrameSize;
}
if (sampleRate == audioRate)
buffer_ = monoBuffer;
else {
double factord = (double) sampleRate / audioRate;
float* floatBufferIn = new float[size_];
int sizeOut = ceil(factord * size_);
src_short_to_float_array(monoBuffer, floatBufferIn, size_);
delete [] monoBuffer;
delete [] buffer_;
buffer_ = new SFLDataFormat[sizeOut];
SRC_DATA src_data;
src_data.data_in = floatBufferIn;
src_data.input_frames = size_;
src_data.output_frames = sizeOut;
src_data.src_ratio = factord;
float* floatBufferOut = new float[sizeOut];
src_data.data_out = floatBufferOut;
src_simple(&src_data, SRC_SINC_BEST_QUALITY, 1);
src_float_to_short_array(floatBufferOut, buffer_, src_data.output_frames_gen);
delete [] floatBufferOut;
delete [] floatBufferIn;
size_ = src_data.output_frames_gen;
}
}
WaveFile::WaveFile(const std::string &fileName, unsigned int sampleRate) : AudioFile(fileName, sampleRate)
{
const std::fstream fs(fileName.c_str(), std::ios_base::in);
if (!fs)
throw AudioFileException("File " + fileName + " doesn't exist");
std::fstream fileStream;
fileStream.open(fileName.c_str(), std::ios::in | std::ios::binary);
char riff[4] = { 0, 0, 0, 0 };
fileStream.read(riff, sizeof riff / sizeof *riff);
if (strncmp("RIFF", riff, sizeof riff / sizeof *riff) != 0)
throw AudioFileException("File is not of RIFF format");
char fmt[4] = { 0, 0, 0, 0 };
int maxIteration = 10;
while (maxIteration-- and strncmp("fmt ", fmt, sizeof fmt / sizeof *fmt))
fileStream.read(fmt, sizeof fmt / sizeof *fmt);
if (maxIteration == 0)
throw AudioFileException("Could not find \"fmt \" chunk");
SINT32 chunkSize; // fmt chunk size
fileStream.read(reinterpret_cast<char *>(&chunkSize), sizeof chunkSize); // Read fmt chunk size.
unsigned short formatTag; // data compression tag
fileStream.read(reinterpret_cast<char *>(&formatTag), sizeof formatTag);
if (formatTag != 1) // PCM = 1, FLOAT = 3
throw AudioFileException("File contains an unsupported data format type");
// Get number of channels from the header.
SINT16 chan;
fileStream.read(reinterpret_cast<char *>(&chan), sizeof chan);
if (chan > 2)
throw AudioFileException("WaveFile: unsupported number of channels");
// Get file sample rate from the header.
SINT32 fileRate;
fileStream.read(reinterpret_cast<char *>(&fileRate), sizeof fileRate);
SINT32 avgb;
fileStream.read(reinterpret_cast<char *>(&avgb), sizeof avgb);
SINT16 blockal;
fileStream.read(reinterpret_cast<char *>(&blockal), sizeof blockal);
// Determine the data type
SINT16 dt;
fileStream.read(reinterpret_cast<char *>(&dt), sizeof dt);
if (dt != 8 && dt != 16 && dt != 32)
throw AudioFileException("File's bits per sample with is not supported");
// Find the "data" chunk
char data[4] = { 0, 0, 0, 0 };
maxIteration = 10;
while (maxIteration-- && strncmp("data", data, sizeof data / sizeof *data))
fileStream.read(data, sizeof data / sizeof *data);
// Samplerate converter initialized with 88200 sample long
const int rate = static_cast<SINT32>(sampleRate_);
SamplerateConverter converter(std::max(fileRate, rate));
// Get length of data from the header.
SINT32 bytes;
fileStream.read(reinterpret_cast<char *>(&bytes), sizeof bytes);
// sample frames, should not be longer than a minute
int nbSamples = std::min(60 * fileRate, 8 * bytes / dt / chan);
DEBUG("Frame size %ld, data size %d align %d rate %d avgbyte %d "
"chunk size %d dt %d", nbSamples, bytes, blockal, fileRate, avgb,
chunkSize, dt);
size_ = nbSamples;
SFLDataFormat *tempBuffer = new SFLDataFormat[size_];
fileStream.read(reinterpret_cast<char *>(tempBuffer),
nbSamples * sizeof(SFLDataFormat));
// mix two channels together if stereo
if (chan == 2) {
for (int i = 0, j = 0; i < nbSamples - 1; i += 2, ++j)
tempBuffer[j] = (tempBuffer[i] + tempBuffer[i + 1]) * 0.5;
nbSamples *= 0.5;
}
if (fileRate != rate) {
const float ratio = sampleRate_ / (float) fileRate;
const int outSamples = ceil(nbSamples * ratio);
size_ = outSamples;
buffer_ = new SFLDataFormat[size_];
converter.resample(tempBuffer, buffer_, size_, fileRate, sampleRate_, nbSamples);
delete [] tempBuffer;
} else
buffer_ = tempBuffer;
}
|