| /* |
| * 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 COMMON_AUDIO_INCLUDE_AUDIO_UTIL_H_ |
| #define COMMON_AUDIO_INCLUDE_AUDIO_UTIL_H_ |
| |
| #include <stdint.h> |
| |
| #include <algorithm> |
| #include <cmath> |
| #include <cstring> |
| #include <limits> |
| |
| #include "rtc_base/checks.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] |
| // Dbfs: float [-20.0*log(10, 32768), 0] = [-90.3, 0] |
| // The ratio conversion functions use this naming convention: |
| // Ratio: float (0, +inf) |
| // Db: float (-inf, +inf) |
| 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); |
| |
| inline float DbToRatio(float v) { |
| return std::pow(10.0f, v / 20.0f); |
| } |
| |
| inline float DbfsToFloatS16(float v) { |
| static constexpr float kMaximumAbsFloatS16 = -limits_int16::min(); |
| return DbToRatio(v) * kMaximumAbsFloatS16; |
| } |
| |
| inline float FloatS16ToDbfs(float v) { |
| RTC_DCHECK_GE(v, 0); |
| |
| // kMinDbfs is equal to -20.0 * log10(-limits_int16::min()) |
| static constexpr float kMinDbfs = -90.30899869919436f; |
| if (v <= 1.0f) { |
| return kMinDbfs; |
| } |
| // Equal to 20 * log10(v / (-limits_int16::min())) |
| return 20.0f * std::log10(v) + kMinDbfs; |
| } |
| |
| // Copy audio from |src| channels to |dest| channels unless |src| and |dest| |
| // point to the same address. |src| and |dest| must have the same number of |
| // channels, and there must be sufficient space allocated in |dest|. |
| template <typename T> |
| void CopyAudioIfNeeded(const T* const* src, |
| int num_frames, |
| int num_channels, |
| T* const* dest) { |
| for (int i = 0; i < num_channels; ++i) { |
| if (src[i] != dest[i]) { |
| std::copy(src[i], src[i] + num_frames, dest[i]); |
| } |
| } |
| } |
| |
| // 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, |
| size_t samples_per_channel, |
| size_t num_channels, |
| T* const* deinterleaved) { |
| for (size_t i = 0; i < num_channels; ++i) { |
| T* channel = deinterleaved[i]; |
| size_t interleaved_idx = i; |
| for (size_t 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, |
| size_t samples_per_channel, |
| size_t num_channels, |
| T* interleaved) { |
| for (size_t i = 0; i < num_channels; ++i) { |
| const T* channel = deinterleaved[i]; |
| size_t interleaved_idx = i; |
| for (size_t j = 0; j < samples_per_channel; ++j) { |
| interleaved[interleaved_idx] = channel[j]; |
| interleaved_idx += num_channels; |
| } |
| } |
| } |
| |
| // Copies audio from a single channel buffer pointed to by |mono| to each |
| // channel of |interleaved|. There must be sufficient space allocated in |
| // |interleaved| (|samples_per_channel| * |num_channels|). |
| template <typename T> |
| void UpmixMonoToInterleaved(const T* mono, |
| int num_frames, |
| int num_channels, |
| T* interleaved) { |
| int interleaved_idx = 0; |
| for (int i = 0; i < num_frames; ++i) { |
| for (int j = 0; j < num_channels; ++j) { |
| interleaved[interleaved_idx++] = mono[i]; |
| } |
| } |
| } |
| |
| template <typename T, typename Intermediate> |
| void DownmixToMono(const T* const* input_channels, |
| size_t num_frames, |
| int num_channels, |
| T* out) { |
| for (size_t i = 0; i < num_frames; ++i) { |
| Intermediate value = input_channels[0][i]; |
| for (int j = 1; j < num_channels; ++j) { |
| value += input_channels[j][i]; |
| } |
| out[i] = value / num_channels; |
| } |
| } |
| |
| // Downmixes an interleaved multichannel signal to a single channel by averaging |
| // all channels. |
| template <typename T, typename Intermediate> |
| void DownmixInterleavedToMonoImpl(const T* interleaved, |
| size_t num_frames, |
| int num_channels, |
| T* deinterleaved) { |
| RTC_DCHECK_GT(num_channels, 0); |
| RTC_DCHECK_GT(num_frames, 0); |
| |
| const T* const end = interleaved + num_frames * num_channels; |
| |
| while (interleaved < end) { |
| const T* const frame_end = interleaved + num_channels; |
| |
| Intermediate value = *interleaved++; |
| while (interleaved < frame_end) { |
| value += *interleaved++; |
| } |
| |
| *deinterleaved++ = value / num_channels; |
| } |
| } |
| |
| template <typename T> |
| void DownmixInterleavedToMono(const T* interleaved, |
| size_t num_frames, |
| int num_channels, |
| T* deinterleaved); |
| |
| template <> |
| void DownmixInterleavedToMono<int16_t>(const int16_t* interleaved, |
| size_t num_frames, |
| int num_channels, |
| int16_t* deinterleaved); |
| |
| } // namespace webrtc |
| |
| #endif // COMMON_AUDIO_INCLUDE_AUDIO_UTIL_H_ |