Added empty directory with myself as owner for new mixer.

webrtc/modules/audio_mixer/source/audio_frame_manipulator.cc

+/*
+ *  Copyright (c) 2012 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.
+ */
+
+#include "webrtc/modules/audio_conference_mixer/source/audio_frame_manipulator.h"
+#include "webrtc/modules/include/module_common_types.h"
+#include "webrtc/typedefs.h"
+
+namespace {
+// Linear ramping over 80 samples.
+// TODO(hellner): ramp using fix point?
+const float rampArray[] = {
+    0.0000f, 0.0127f, 0.0253f, 0.0380f, 0.0506f, 0.0633f, 0.0759f, 0.0886f,
+    0.1013f, 0.1139f, 0.1266f, 0.1392f, 0.1519f, 0.1646f, 0.1772f, 0.1899f,
+    0.2025f, 0.2152f, 0.2278f, 0.2405f, 0.2532f, 0.2658f, 0.2785f, 0.2911f,
+    0.3038f, 0.3165f, 0.3291f, 0.3418f, 0.3544f, 0.3671f, 0.3797f, 0.3924f,
+    0.4051f, 0.4177f, 0.4304f, 0.4430f, 0.4557f, 0.4684f, 0.4810f, 0.4937f,
+    0.5063f, 0.5190f, 0.5316f, 0.5443f, 0.5570f, 0.5696f, 0.5823f, 0.5949f,
+    0.6076f, 0.6203f, 0.6329f, 0.6456f, 0.6582f, 0.6709f, 0.6835f, 0.6962f,
+    0.7089f, 0.7215f, 0.7342f, 0.7468f, 0.7595f, 0.7722f, 0.7848f, 0.7975f,
+    0.8101f, 0.8228f, 0.8354f, 0.8481f, 0.8608f, 0.8734f, 0.8861f, 0.8987f,
+    0.9114f, 0.9241f, 0.9367f, 0.9494f, 0.9620f, 0.9747f, 0.9873f, 1.0000f};
+const size_t rampSize = sizeof(rampArray) / sizeof(rampArray[0]);
+}  // namespace
+
+namespace webrtc {
+uint32_t CalculateEnergy(const AudioFrame& audioFrame) {
+  uint32_t energy = 0;
+  for (size_t position = 0; position < audioFrame.samples_per_channel_;
+       position++) {
+    // TODO(andrew): this can easily overflow.
+    energy += audioFrame.data_[position] * audioFrame.data_[position];
+  }
+  return energy;
+}
+
+void RampIn(AudioFrame& audioFrame) {
+  assert(rampSize <= audioFrame.samples_per_channel_);
+  for (size_t i = 0; i < rampSize; i++) {
+    audioFrame.data_[i] =
+        static_cast<int16_t>(rampArray[i] * audioFrame.data_[i]);
+  }
+}
+
+void RampOut(AudioFrame& audioFrame) {
+  assert(rampSize <= audioFrame.samples_per_channel_);
+  for (size_t i = 0; i < rampSize; i++) {
+    const size_t rampPos = rampSize - 1 - i;
+    audioFrame.data_[i] =
+        static_cast<int16_t>(rampArray[rampPos] * audioFrame.data_[i]);
+  }
+  memset(&audioFrame.data_[rampSize], 0,
+         (audioFrame.samples_per_channel_ - rampSize) *
+             sizeof(audioFrame.data_[0]));
+}
+}  // namespace webrtc