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
|
/*
* Copyright (c) 2013 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.
*/
#ifndef WEBRTC_COMMON_AUDIO_INCLUDE_AUDIO_UTIL_H_
#define WEBRTC_COMMON_AUDIO_INCLUDE_AUDIO_UTIL_H_
#include <limits>
#include "webrtc/system_wrappers/interface/scoped_ptr.h"
#include "webrtc/typedefs.h"
namespace webrtc {
typedef std::numeric_limits<int16_t> limits_int16;
static inline int16_t RoundToInt16(float v) {
const float kMaxRound = limits_int16::max() - 0.5f;
const float kMinRound = limits_int16::min() + 0.5f;
if (v > 0)
return v >= kMaxRound ? limits_int16::max() :
static_cast<int16_t>(v + 0.5f);
return v <= kMinRound ? limits_int16::min() :
static_cast<int16_t>(v - 0.5f);
}
// Scale (from [-1, 1]) and round to full-range int16 with clamping.
static inline int16_t ScaleAndRoundToInt16(float v) {
if (v > 0)
return v >= 1 ? limits_int16::max() :
static_cast<int16_t>(v * limits_int16::max() + 0.5f);
return v <= -1 ? limits_int16::min() :
static_cast<int16_t>(-v * limits_int16::min() - 0.5f);
}
// Scale to float [-1, 1].
static inline float ScaleToFloat(int16_t v) {
const float kMaxInt16Inverse = 1.f / limits_int16::max();
const float kMinInt16Inverse = 1.f / limits_int16::min();
return v * (v > 0 ? kMaxInt16Inverse : -kMinInt16Inverse);
}
// Round |size| elements of |src| to int16 with clamping and write to |dest|.
void RoundToInt16(const float* src, size_t size, int16_t* dest);
// Scale (from [-1, 1]) and round |size| elements of |src| to full-range int16
// with clamping and write to |dest|.
void ScaleAndRoundToInt16(const float* src, size_t size, int16_t* dest);
// Scale |size| elements of |src| to float [-1, 1] and write to |dest|.
void ScaleToFloat(const int16_t* src, size_t size, float* dest);
// Deinterleave audio from |interleaved| to the channel buffers pointed to
// by |deinterleaved|. There must be sufficient space allocated in the
// |deinterleaved| buffers (|num_channel| buffers with |samples_per_channel|
// per buffer).
template <typename T>
void Deinterleave(const T* interleaved, int samples_per_channel,
int num_channels, T* const* deinterleaved) {
for (int i = 0; i < num_channels; ++i) {
T* channel = deinterleaved[i];
int interleaved_idx = i;
for (int j = 0; j < samples_per_channel; ++j) {
channel[j] = interleaved[interleaved_idx];
interleaved_idx += num_channels;
}
}
}
// Interleave audio from the channel buffers pointed to by |deinterleaved| to
// |interleaved|. There must be sufficient space allocated in |interleaved|
// (|samples_per_channel| * |num_channels|).
template <typename T>
void Interleave(const T* const* deinterleaved, int samples_per_channel,
int num_channels, T* interleaved) {
for (int i = 0; i < num_channels; ++i) {
const T* channel = deinterleaved[i];
int interleaved_idx = i;
for (int j = 0; j < samples_per_channel; ++j) {
interleaved[interleaved_idx] = channel[j];
interleaved_idx += num_channels;
}
}
}
} // namespace webrtc
#endif // WEBRTC_COMMON_AUDIO_INCLUDE_AUDIO_UTIL_H_
|
