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/*
* 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, 32768.0]
// 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 float S16ToFloat(int16_t v) {
constexpr float kScaling = 1.f / 32768.f;
return v * kScaling;
}
static inline int16_t FloatS16ToS16(float v) {
v = std::min(v, 32767.f);
v = std::max(v, -32768.f);
return static_cast<int16_t>(v + std::copysign(0.5f, v));
}
static inline int16_t FloatToS16(float v) {
v *= 32768.f;
v = std::min(v, 32767.f);
v = std::max(v, -32768.f);
return static_cast<int16_t>(v + std::copysign(0.5f, v));
}
static inline float FloatToFloatS16(float v) {
v = std::min(v, 1.f);
v = std::max(v, -1.f);
return v * 32768.f;
}
static inline float FloatS16ToFloat(float v) {
v = std::min(v, 32768.f);
v = std::max(v, -32768.f);
constexpr float kScaling = 1.f / 32768.f;
return v * kScaling;
}
void FloatToS16(const float* src, size_t size, int16_t* dest);
void S16ToFloat(const int16_t* src, size_t size, float* dest);
void S16ToFloatS16(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_