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
|
/*
* Copyright (c) 2014 The WebRTC project authors. All Rights Reserved.
*
* Use of this source code is governed by a BSD-style license
* that can be found in the LICENSE file in the root of the source
* tree. An additional intellectual property rights grant can be found
* in the file PATENTS. All contributing project authors may
* be found in the AUTHORS file in the root of the source tree.
*/
#include <math.h>
#include <limits>
#include "webrtc/audio_processing/debug.pb.h"
#include "webrtc/common_audio/include/audio_util.h"
#include "webrtc/common_audio/wav_file.h"
#include "webrtc/modules/audio_processing/channel_buffer.h"
#include "webrtc/modules/audio_processing/include/audio_processing.h"
#include "webrtc/modules/interface/module_common_types.h"
#include "webrtc/system_wrappers/interface/scoped_ptr.h"
namespace webrtc {
static const AudioProcessing::Error kNoErr = AudioProcessing::kNoError;
#define EXPECT_NOERR(expr) EXPECT_EQ(kNoErr, (expr))
class RawFile {
public:
RawFile(const std::string& filename)
: file_handle_(fopen(filename.c_str(), "wb")) {}
~RawFile() {
fclose(file_handle_);
}
void WriteSamples(const int16_t* samples, size_t num_samples) {
#ifndef WEBRTC_ARCH_LITTLE_ENDIAN
#error "Need to convert samples to little-endian when writing to PCM file"
#endif
fwrite(samples, sizeof(*samples), num_samples, file_handle_);
}
void WriteSamples(const float* samples, size_t num_samples) {
fwrite(samples, sizeof(*samples), num_samples, file_handle_);
}
private:
FILE* file_handle_;
};
static inline void WriteIntData(const int16_t* data,
size_t length,
WavWriter* wav_file,
RawFile* raw_file) {
if (wav_file) {
wav_file->WriteSamples(data, length);
}
if (raw_file) {
raw_file->WriteSamples(data, length);
}
}
static inline void WriteFloatData(const float* const* data,
size_t samples_per_channel,
int num_channels,
WavWriter* wav_file,
RawFile* raw_file) {
size_t length = num_channels * samples_per_channel;
scoped_ptr<float[]> buffer(new float[length]);
Interleave(data, samples_per_channel, num_channels, buffer.get());
if (raw_file) {
raw_file->WriteSamples(buffer.get(), length);
}
// TODO(aluebs): Use ScaleToInt16Range() from audio_util
for (size_t i = 0; i < length; ++i) {
buffer[i] = buffer[i] > 0 ?
buffer[i] * std::numeric_limits<int16_t>::max() :
-buffer[i] * std::numeric_limits<int16_t>::min();
}
if (wav_file) {
wav_file->WriteSamples(buffer.get(), length);
}
}
// Exits on failure; do not use in unit tests.
static inline FILE* OpenFile(const std::string& filename, const char* mode) {
FILE* file = fopen(filename.c_str(), mode);
if (!file) {
printf("Unable to open file %s\n", filename.c_str());
exit(1);
}
return file;
}
static inline int SamplesFromRate(int rate) {
return AudioProcessing::kChunkSizeMs * rate / 1000;
}
static inline void SetFrameSampleRate(AudioFrame* frame,
int sample_rate_hz) {
frame->sample_rate_hz_ = sample_rate_hz;
frame->samples_per_channel_ = AudioProcessing::kChunkSizeMs *
sample_rate_hz / 1000;
}
template <typename T>
void SetContainerFormat(int sample_rate_hz,
int num_channels,
AudioFrame* frame,
scoped_ptr<ChannelBuffer<T> >* cb) {
SetFrameSampleRate(frame, sample_rate_hz);
frame->num_channels_ = num_channels;
cb->reset(new ChannelBuffer<T>(frame->samples_per_channel_, num_channels));
}
static inline AudioProcessing::ChannelLayout LayoutFromChannels(
int num_channels) {
switch (num_channels) {
case 1:
return AudioProcessing::kMono;
case 2:
return AudioProcessing::kStereo;
default:
assert(false);
return AudioProcessing::kMono;
}
}
// Allocates new memory in the scoped_ptr to fit the raw message and returns the
// number of bytes read.
static inline size_t ReadMessageBytesFromFile(FILE* file,
scoped_ptr<uint8_t[]>* bytes) {
// The "wire format" for the size is little-endian. Assume we're running on
// a little-endian machine.
int32_t size = 0;
if (fread(&size, sizeof(size), 1, file) != 1)
return 0;
if (size <= 0)
return 0;
bytes->reset(new uint8_t[size]);
return fread(bytes->get(), sizeof((*bytes)[0]), size, file);
}
// Returns true on success, false on error or end-of-file.
static inline bool ReadMessageFromFile(FILE* file,
::google::protobuf::MessageLite* msg) {
scoped_ptr<uint8_t[]> bytes;
size_t size = ReadMessageBytesFromFile(file, &bytes);
if (!size)
return false;
msg->Clear();
return msg->ParseFromArray(bytes.get(), size);
}
template <typename T>
float ComputeSNR(const T* ref, const T* test, int length, float* variance) {
float mse = 0;
float mean = 0;
*variance = 0;
for (int i = 0; i < length; ++i) {
T error = ref[i] - test[i];
mse += error * error;
*variance += ref[i] * ref[i];
mean += ref[i];
}
mse /= length;
*variance /= length;
mean /= length;
*variance -= mean * mean;
float snr = 100; // We assign 100 dB to the zero-error case.
if (mse > 0)
snr = 10 * log10(*variance / mse);
return snr;
}
} // namespace webrtc
|
