audio_util.h revision 60d9b332a5391045439bfb6a3a5447973e3d5603 
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#include <cstring>
16
17#include "webrtc/base/checks.h"
18#include "webrtc/base/scoped_ptr.h"
19#include "webrtc/typedefs.h"
20
21namespace webrtc {
22
23typedef std::numeric_limits<int16_t> limits_int16;
24
25// The conversion functions use the following naming convention:
26// S16:      int16_t [-32768, 32767]
27// Float:    float   [-1.0, 1.0]
28// FloatS16: float   [-32768.0, 32767.0]
29static inline int16_t FloatToS16(float v) {
30  if (v > 0)
31    return v >= 1 ? limits_int16::max()
32                  : static_cast<int16_t>(v * limits_int16::max() + 0.5f);
33  return v <= -1 ? limits_int16::min()
34                 : static_cast<int16_t>(-v * limits_int16::min() - 0.5f);
35}
36
37static inline float S16ToFloat(int16_t v) {
38  static const float kMaxInt16Inverse = 1.f / limits_int16::max();
39  static const float kMinInt16Inverse = 1.f / limits_int16::min();
40  return v * (v > 0 ? kMaxInt16Inverse : -kMinInt16Inverse);
41}
42
43static inline int16_t FloatS16ToS16(float v) {
44  static const float kMaxRound = limits_int16::max() - 0.5f;
45  static const float kMinRound = limits_int16::min() + 0.5f;
46  if (v > 0)
47    return v >= kMaxRound ? limits_int16::max()
48                          : static_cast<int16_t>(v + 0.5f);
49  return v <= kMinRound ? limits_int16::min() : static_cast<int16_t>(v - 0.5f);
50}
51
52static inline float FloatToFloatS16(float v) {
53  return v * (v > 0 ? limits_int16::max() : -limits_int16::min());
54}
55
56static inline float FloatS16ToFloat(float v) {
57  static const float kMaxInt16Inverse = 1.f / limits_int16::max();
58  static const float kMinInt16Inverse = 1.f / limits_int16::min();
59  return v * (v > 0 ? kMaxInt16Inverse : -kMinInt16Inverse);
60}
61
62void FloatToS16(const float* src, size_t size, int16_t* dest);
63void S16ToFloat(const int16_t* src, size_t size, float* dest);
64void FloatS16ToS16(const float* src, size_t size, int16_t* dest);
65void FloatToFloatS16(const float* src, size_t size, float* dest);
66void FloatS16ToFloat(const float* src, size_t size, float* dest);
67
68// Copy audio from |src| channels to |dest| channels unless |src| and |dest|
69// point to the same address. |src| and |dest| must have the same number of
70// channels, and there must be sufficient space allocated in |dest|.
71template <typename T>
72void CopyAudioIfNeeded(const T* const* src,
73                       int num_frames,
74                       int num_channels,
75                       T* const* dest) {
76  for (int i = 0; i < num_channels; ++i) {
77    if (src[i] != dest[i]) {
78      std::copy(src[i], src[i] + num_frames, dest[i]);
79    }
80  }
81}
82
83// Deinterleave audio from |interleaved| to the channel buffers pointed to
84// by |deinterleaved|. There must be sufficient space allocated in the
85// |deinterleaved| buffers (|num_channel| buffers with |samples_per_channel|
86// per buffer).
87template <typename T>
88void Deinterleave(const T* interleaved,
89                  int samples_per_channel,
90                  int num_channels,
91                  T* const* deinterleaved) {
92  for (int i = 0; i < num_channels; ++i) {
93    T* channel = deinterleaved[i];
94    int interleaved_idx = i;
95    for (int j = 0; j < samples_per_channel; ++j) {
96      channel[j] = interleaved[interleaved_idx];
97      interleaved_idx += num_channels;
98    }
99  }
100}
101
102// Interleave audio from the channel buffers pointed to by |deinterleaved| to
103// |interleaved|. There must be sufficient space allocated in |interleaved|
104// (|samples_per_channel| * |num_channels|).
105template <typename T>
106void Interleave(const T* const* deinterleaved,
107                int samples_per_channel,
108                int num_channels,
109                T* interleaved) {
110  for (int i = 0; i < num_channels; ++i) {
111    const T* channel = deinterleaved[i];
112    int interleaved_idx = i;
113    for (int j = 0; j < samples_per_channel; ++j) {
114      interleaved[interleaved_idx] = channel[j];
115      interleaved_idx += num_channels;
116    }
117  }
118}
119
120// Copies audio from a single channel buffer pointed to by |mono| to each
121// channel of |interleaved|. There must be sufficient space allocated in
122// |interleaved| (|samples_per_channel| * |num_channels|).
123template <typename T>
124void UpmixMonoToInterleaved(const T* mono,
125                            int num_frames,
126                            int num_channels,
127                            T* interleaved) {
128  int interleaved_idx = 0;
129  for (int i = 0; i < num_frames; ++i) {
130    for (int j = 0; j < num_channels; ++j) {
131      interleaved[interleaved_idx++] = mono[i];
132    }
133  }
134}
135
136template <typename T, typename Intermediate>
137void DownmixToMono(const T* const* input_channels,
138                   int num_frames,
139                   int num_channels,
140                   T* out) {
141  for (int i = 0; i < num_frames; ++i) {
142    Intermediate value = input_channels[0][i];
143    for (int j = 1; j < num_channels; ++j) {
144      value += input_channels[j][i];
145    }
146    out[i] = value / num_channels;
147  }
148}
149
150// Downmixes an interleaved multichannel signal to a single channel by averaging
151// all channels.
152template <typename T, typename Intermediate>
153void DownmixInterleavedToMonoImpl(const T* interleaved,
154                                  int num_frames,
155                                  int num_channels,
156                                  T* deinterleaved) {
157  DCHECK_GT(num_channels, 0);
158  DCHECK_GT(num_frames, 0);
159
160  const T* const end = interleaved + num_frames * num_channels;
161
162  while (interleaved < end) {
163    const T* const frame_end = interleaved + num_channels;
164
165    Intermediate value = *interleaved++;
166    while (interleaved < frame_end) {
167      value += *interleaved++;
168    }
169
170    *deinterleaved++ = value / num_channels;
171  }
172}
173
174template <typename T>
175void DownmixInterleavedToMono(const T* interleaved,
176                              int num_frames,
177                              int num_channels,
178                              T* deinterleaved);
179
180template <>
181void DownmixInterleavedToMono<int16_t>(const int16_t* interleaved,
182                                       int num_frames,
183                                       int num_channels,
184                                       int16_t* deinterleaved);
185
186}  // namespace webrtc
187
188#endif  // WEBRTC_COMMON_AUDIO_INCLUDE_AUDIO_UTIL_H_
189