modules/audio_conference_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)
{
    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