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-rw-r--r--third_party/libwebrtc/common_audio/include/audio_util.h214
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
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+++ b/third_party/libwebrtc/common_audio/include/audio_util.h
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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_