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/*
* 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 {
void CalculateEnergy(AudioFrame& audioFrame)
{
audioFrame.energy_ = 0;
for(size_t position = 0; position < audioFrame.samples_per_channel_;
position++)
{
// TODO(andrew): this can easily overflow.
audioFrame.energy_ += audioFrame.data_[position] *
audioFrame.data_[position];
}
}
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
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