AnotherFoxGuy/webrtc-audio-processing
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Update to current webrtc library

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This is from the upstream library commit id
3326535126e435f1ba647885ce43a8f0f3d317eb, corresponding to Chromium
88.0.4290.1.
This commit is contained in:
Arun Raghavan
2020-10-12 18:08:02 -04:00
parent b1b02581d3
commit bcec8b0b21
859 changed files with 76187 additions and 49580 deletions
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98
webrtc/common_audio/include/audio_util.h
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@ -8,15 +8,17 @@
* 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_
#ifndef COMMON_AUDIO_INCLUDE_AUDIO_UTIL_H_
#define COMMON_AUDIO_INCLUDE_AUDIO_UTIL_H_
#include <limits>
#include <stdint.h>
#include <algorithm>
#include <cmath>
#include <cstring>
#include <limits>
#include "webrtc/base/checks.h"
#include "webrtc/base/scoped_ptr.h"
#include "webrtc/typedefs.h"
#include "rtc_base/checks.h"
namespace webrtc {
@ -25,46 +27,70 @@ 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);
}
// 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) {
static const float kMaxInt16Inverse = 1.f / limits_int16::max();
static const float kMinInt16Inverse = 1.f / limits_int16::min();
return v * (v > 0 ? kMaxInt16Inverse : -kMinInt16Inverse);
constexpr float kScaling = 1.f / 32768.f;
return v * kScaling;
}
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);
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) {
return v * (v > 0 ? limits_int16::max() : -limits_int16::min());
v = std::min(v, 1.f);
v = std::max(v, -1.f);
return v * 32768.f;
}
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);
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|.
@ -87,11 +113,11 @@ void CopyAudioIfNeeded(const T* const* src,
template <typename T>
void Deinterleave(const T* interleaved,
size_t samples_per_channel,
int num_channels,
size_t num_channels,
T* const* deinterleaved) {
for (int i = 0; i < num_channels; ++i) {
for (size_t i = 0; i < num_channels; ++i) {
T* channel = deinterleaved[i];
int interleaved_idx = 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;
@ -105,11 +131,11 @@ void Deinterleave(const T* interleaved,
template <typename T>
void Interleave(const T* const* deinterleaved,
size_t samples_per_channel,
int num_channels,
size_t num_channels,
T* interleaved) {
for (int i = 0; i < num_channels; ++i) {
for (size_t i = 0; i < num_channels; ++i) {
const T* channel = deinterleaved[i];
int interleaved_idx = 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;
@ -155,7 +181,7 @@ void DownmixInterleavedToMonoImpl(const T* interleaved,
int num_channels,
T* deinterleaved) {
RTC_DCHECK_GT(num_channels, 0);
RTC_DCHECK_GT(num_frames, 0u);
RTC_DCHECK_GT(num_frames, 0);
const T* const end = interleaved + num_frames * num_channels;
@ -185,4 +211,4 @@ void DownmixInterleavedToMono<int16_t>(const int16_t* interleaved,
} // namespace webrtc
#endif // WEBRTC_COMMON_AUDIO_INCLUDE_AUDIO_UTIL_H_
#endif // COMMON_AUDIO_INCLUDE_AUDIO_UTIL_H_