Adding upstream version 115.7.0esr.upstream/115.7.0esrupstream

Signed-off-by: Daniel Baumann <daniel.baumann@progress-linux.org>

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_