webrtc-audio-processing/webrtc/common_audio/include/audio_util.h

/*
 *  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 "api/audio/audio_view.h"
#include "rtc_base/checks.h"

namespace webrtc {

typedef std::numeric_limits<int16_t> limits_int16;

// TODO(tommi, peah): Move these constants to their own header, e.g.
// `audio_constants.h`. Also consider if they should be in api/.

// Absolute highest acceptable sample rate supported for audio processing,
// capture and codecs. Note that for some components some cases a lower limit
// applies which typically is 48000 but in some cases is lower.
constexpr int kMaxSampleRateHz = 384000;

// Number of samples per channel for 10ms of audio at the highest sample rate.
constexpr size_t kMaxSamplesPerChannel10ms = kMaxSampleRateHz / 100u;

// 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`.
// TODO: b/335805780 - Accept ArrayView.
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 InterleavedView<const T>& interleaved,
                  const DeinterleavedView<T>& deinterleaved) {
  RTC_DCHECK_EQ(NumChannels(interleaved), NumChannels(deinterleaved));
  RTC_DCHECK_EQ(SamplesPerChannel(interleaved),
                SamplesPerChannel(deinterleaved));
  const auto num_channels = NumChannels(interleaved);
  const auto samples_per_channel = SamplesPerChannel(interleaved);
  for (size_t i = 0; i < num_channels; ++i) {
    MonoView<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 DeinterleavedView<const T>& deinterleaved,
                const InterleavedView<T>& interleaved) {
  RTC_DCHECK_EQ(NumChannels(interleaved), NumChannels(deinterleaved));
  RTC_DCHECK_EQ(SamplesPerChannel(interleaved),
                SamplesPerChannel(deinterleaved));
  for (size_t i = 0; i < deinterleaved.num_channels(); ++i) {
    const auto channel = deinterleaved[i];
    size_t interleaved_idx = i;
    for (size_t j = 0; j < deinterleaved.samples_per_channel(); ++j) {
      interleaved[interleaved_idx] = channel[j];
      interleaved_idx += deinterleaved.num_channels();
    }
  }
}

// Downmixes an interleaved multichannel signal to a single channel by averaging
// all channels.
// TODO: b/335805780 - Accept InterleavedView and DeinterleavedView.
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;
  }
}

// TODO: b/335805780 - Accept InterleavedView and DeinterleavedView.
template <typename T>
void DownmixInterleavedToMono(const T* interleaved,
                              size_t num_frames,
                              int num_channels,
                              T* deinterleaved);

// TODO: b/335805780 - Accept InterleavedView and DeinterleavedView.
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_