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author | Daniel Baumann <daniel.baumann@progress-linux.org> | 2024-04-19 00:47:55 +0000 |
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committer | Daniel Baumann <daniel.baumann@progress-linux.org> | 2024-04-19 00:47:55 +0000 |
commit | 26a029d407be480d791972afb5975cf62c9360a6 (patch) | |
tree | f435a8308119effd964b339f76abb83a57c29483 /third_party/libwebrtc/common_audio/include | |
parent | Initial commit. (diff) | |
download | firefox-26a029d407be480d791972afb5975cf62c9360a6.tar.xz firefox-26a029d407be480d791972afb5975cf62c9360a6.zip |
Adding upstream version 124.0.1.upstream/124.0.1
Signed-off-by: Daniel Baumann <daniel.baumann@progress-linux.org>
Diffstat (limited to 'third_party/libwebrtc/common_audio/include')
-rw-r--r-- | third_party/libwebrtc/common_audio/include/audio_util.h | 214 |
1 files changed, 214 insertions, 0 deletions
diff --git a/third_party/libwebrtc/common_audio/include/audio_util.h b/third_party/libwebrtc/common_audio/include/audio_util.h new file mode 100644 index 0000000000..4ce46800f1 --- /dev/null +++ b/third_party/libwebrtc/common_audio/include/audio_util.h @@ -0,0 +1,214 @@ +/* + * 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_ |