| /* |
| * 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/base/scoped_ptr.h" |
| #include "webrtc/typedefs.h" |
| |
| namespace webrtc { |
| |
| typedef std::numeric_limits<int16_t> limits_int16; |
| |
| // The conversion functions use the following naming convention: |
| // S16: int16_t [-32768, 32767] |
| // Float: float [-1.0, 1.0] |
| // FloatS16: float [-32768.0, 32767.0] |
| static inline int16_t FloatToS16(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); |
| } |
| |
| static inline float S16ToFloat(int16_t v) { |
| static const float kMaxInt16Inverse = 1.f / limits_int16::max(); |
| static const float kMinInt16Inverse = 1.f / limits_int16::min(); |
| return v * (v > 0 ? kMaxInt16Inverse : -kMinInt16Inverse); |
| } |
| |
| static inline int16_t FloatS16ToS16(float v) { |
| static const float kMaxRound = limits_int16::max() - 0.5f; |
| static 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); |
| } |
| |
| static inline float FloatToFloatS16(float v) { |
| return v * (v > 0 ? limits_int16::max() : -limits_int16::min()); |
| } |
| |
| static inline float FloatS16ToFloat(float v) { |
| static const float kMaxInt16Inverse = 1.f / limits_int16::max(); |
| static const float kMinInt16Inverse = 1.f / limits_int16::min(); |
| return v * (v > 0 ? kMaxInt16Inverse : -kMinInt16Inverse); |
| } |
| |
| void FloatToS16(const float* src, size_t size, int16_t* dest); |
| void S16ToFloat(const int16_t* src, size_t size, float* dest); |
| void FloatS16ToS16(const float* src, size_t size, int16_t* dest); |
| void FloatToFloatS16(const float* src, size_t size, float* dest); |
| void FloatS16ToFloat(const float* 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_ |