1 /*
2 * Copyright (c) 2013 The WebRTC project authors. All Rights Reserved.
3 *
4 * Use of this source code is governed by a BSD-style license
5 * that can be found in the LICENSE file in the root of the source
6 * tree. An additional intellectual property rights grant can be found
7 * in the file PATENTS. All contributing project authors may
8 * be found in the AUTHORS file in the root of the source tree.
9 */
10
11 #ifndef WEBRTC_COMMON_AUDIO_INCLUDE_AUDIO_UTIL_H_
12 #define WEBRTC_COMMON_AUDIO_INCLUDE_AUDIO_UTIL_H_
13
14 #include <limits>
15
16 #include "webrtc/system_wrappers/interface/scoped_ptr.h"
17 #include "webrtc/typedefs.h"
18
19 namespace webrtc {
20
21 typedef std::numeric_limits<int16_t> limits_int16;
22
RoundToInt16(float v)23 static inline int16_t RoundToInt16(float v) {
24 const float kMaxRound = limits_int16::max() - 0.5f;
25 const float kMinRound = limits_int16::min() + 0.5f;
26 if (v > 0)
27 return v >= kMaxRound ? limits_int16::max() :
28 static_cast<int16_t>(v + 0.5f);
29 return v <= kMinRound ? limits_int16::min() :
30 static_cast<int16_t>(v - 0.5f);
31 }
32
33 // Scale (from [-1, 1]) and round to full-range int16 with clamping.
ScaleAndRoundToInt16(float v)34 static inline int16_t ScaleAndRoundToInt16(float v) {
35 if (v > 0)
36 return v >= 1 ? limits_int16::max() :
37 static_cast<int16_t>(v * limits_int16::max() + 0.5f);
38 return v <= -1 ? limits_int16::min() :
39 static_cast<int16_t>(-v * limits_int16::min() - 0.5f);
40 }
41
42 // Scale to float [-1, 1].
ScaleToFloat(int16_t v)43 static inline float ScaleToFloat(int16_t v) {
44 const float kMaxInt16Inverse = 1.f / limits_int16::max();
45 const float kMinInt16Inverse = 1.f / limits_int16::min();
46 return v * (v > 0 ? kMaxInt16Inverse : -kMinInt16Inverse);
47 }
48
49 // Round |size| elements of |src| to int16 with clamping and write to |dest|.
50 void RoundToInt16(const float* src, int size, int16_t* dest);
51
52 // Scale (from [-1, 1]) and round |size| elements of |src| to full-range int16
53 // with clamping and write to |dest|.
54 void ScaleAndRoundToInt16(const float* src, int size, int16_t* dest);
55
56 // Scale |size| elements of |src| to float [-1, 1] and write to |dest|.
57 void ScaleToFloat(const int16_t* src, int size, float* dest);
58
59 // Deinterleave audio from |interleaved| to the channel buffers pointed to
60 // by |deinterleaved|. There must be sufficient space allocated in the
61 // |deinterleaved| buffers (|num_channel| buffers with |samples_per_channel|
62 // per buffer).
63 template <typename T>
Deinterleave(const T * interleaved,int samples_per_channel,int num_channels,T ** deinterleaved)64 void Deinterleave(const T* interleaved, int samples_per_channel,
65 int num_channels, T** deinterleaved) {
66 for (int i = 0; i < num_channels; ++i) {
67 T* channel = deinterleaved[i];
68 int interleaved_idx = i;
69 for (int j = 0; j < samples_per_channel; ++j) {
70 channel[j] = interleaved[interleaved_idx];
71 interleaved_idx += num_channels;
72 }
73 }
74 }
75
76 // Interleave audio from the channel buffers pointed to by |deinterleaved| to
77 // |interleaved|. There must be sufficient space allocated in |interleaved|
78 // (|samples_per_channel| * |num_channels|).
79 template <typename T>
Interleave(const T * const * deinterleaved,int samples_per_channel,int num_channels,T * interleaved)80 void Interleave(const T* const* deinterleaved, int samples_per_channel,
81 int num_channels, T* interleaved) {
82 for (int i = 0; i < num_channels; ++i) {
83 const T* channel = deinterleaved[i];
84 int interleaved_idx = i;
85 for (int j = 0; j < samples_per_channel; ++j) {
86 interleaved[interleaved_idx] = channel[j];
87 interleaved_idx += num_channels;
88 }
89 }
90 }
91
92 } // namespace webrtc
93
94 #endif // WEBRTC_COMMON_AUDIO_INCLUDE_AUDIO_UTIL_H_
95