Update audio_processing module

Corresponds to upstream commit 524e9b043e7e86fd72353b987c9d5f6a1ebf83e1

Update notes:

 * Pull in third party license file

 * Replace .gypi files with BUILD.gn to keep track of what changes
   upstream

 * Bunch of new filse pulled in as dependencies

 * Won't build yet due to changes needed on top of these

webrtc/common_audio/include/audio_util.h

@@ -0,0 +1,188 @@
+/*
+ *  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 WEBRTC_COMMON_AUDIO_INCLUDE_AUDIO_UTIL_H_
+#define WEBRTC_COMMON_AUDIO_INCLUDE_AUDIO_UTIL_H_
+
+#include <limits>
+#include <cstring>
+
+#include "webrtc/base/checks.h"
+#include "webrtc/base/scoped_ptr.h"
+#include "webrtc/typedefs.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, 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);
+}
+
+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);
+}
+
+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);
+}
+
+static inline float FloatToFloatS16(float v) {
+  return v * (v > 0 ? limits_int16::max() : -limits_int16::min());
+}
+
+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);
+}
+
+void FloatToS16(const float* src, size_t size, int16_t* dest);
+void S16ToFloat(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);
+
+// 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,
+                  int num_channels,
+                  T* const* deinterleaved) {
+  for (int i = 0; i < num_channels; ++i) {
+    T* channel = deinterleaved[i];
+    int 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,
+                int num_channels,
+                T* interleaved) {
+  for (int i = 0; i < num_channels; ++i) {
+    const T* channel = deinterleaved[i];
+    int 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, 0u);
+
+  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  // WEBRTC_COMMON_AUDIO_INCLUDE_AUDIO_UTIL_H_