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 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. 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]. 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, 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|. 54 void 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|. 57 void 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). 63 template <typename T> 64 void 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|). 79 template <typename T> 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
