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
19namespace webrtc {
20
21typedef std::numeric_limits<int16_t> limits_int16;
22
23static 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.
34static 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].
43static 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|.
50void RoundToInt16(const float* src, size_t 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|.
54void ScaleAndRoundToInt16(const float* src, size_t size, int16_t* dest);
55
56// Scale |size| elements of |src| to float [-1, 1] and write to |dest|.
57void ScaleToFloat(const int16_t* src, size_t 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).
63template <typename T>
64void Deinterleave(const T* interleaved, int samples_per_channel,
65                  int num_channels, T* const* 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|).
79template <typename T>
80void 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