Remove voe::OutputMixer and AudioConferenceMixer.

modules/audio_conference_mixer/source/audio_frame_manipulator.cc

Index: modules/audio_conference_mixer/source/audio_frame_manipulator.cc
diff --git a/modules/audio_conference_mixer/source/audio_frame_manipulator.cc b/modules/audio_conference_mixer/source/audio_frame_manipulator.cc
deleted file mode 100644
index a16afb70e91158c3e31788128d86985aebc14a10..0000000000000000000000000000000000000000
--- a/modules/audio_conference_mixer/source/audio_frame_manipulator.cc
+++ /dev/null
@@ -1,85 +0,0 @@
-/*
- *  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 "modules/audio_conference_mixer/source/audio_frame_manipulator.h"
-#include "modules/include/module_common_types.h"
-#include "typedefs.h"  // NOLINT(build/include)
-
-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)
-{
-    if (audioFrame.muted()) return 0;
-
-    uint32_t energy = 0;
-    const int16_t* frame_data = audioFrame.data();
-    for(size_t position = 0; position < audioFrame.samples_per_channel_;
-        position++)
-    {
-        // TODO(andrew): this can easily overflow.
-        energy += frame_data[position] * frame_data[position];
-    }
-    return energy;
-}
-
-void RampIn(AudioFrame& audioFrame)
-{
-    assert(rampSize <= audioFrame.samples_per_channel_);
-    if (audioFrame.muted()) return;
-
-    int16_t* frame_data = audioFrame.mutable_data();
-    for(size_t i = 0; i < rampSize; i++)
-    {
-        frame_data[i] = static_cast<int16_t>(rampArray[i] * frame_data[i]);
-    }
-}
-
-void RampOut(AudioFrame& audioFrame)
-{
-    assert(rampSize <= audioFrame.samples_per_channel_);
-    if (audioFrame.muted()) return;
-
-    int16_t* frame_data = audioFrame.mutable_data();
-    for(size_t i = 0; i < rampSize; i++)
-    {
-        const size_t rampPos = rampSize - 1 - i;
-        frame_data[i] = static_cast<int16_t>(rampArray[rampPos] *
-                                             frame_data[i]);
-    }
-    memset(&frame_data[rampSize], 0,
-           (audioFrame.samples_per_channel_ - rampSize) *
-           sizeof(frame_data[0]));
-}
-}  // namespace webrtc