src/third_party/webrtc/modules/audio_conference_mixer/source/audio_frame_manipulator.cc

   1 /*
   2  *  Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
   3  *
   4  *  Use of this source code is governed by a BSD-style license
   5  *  that can be found in the LICENSE file in the root of the source
   6  *  tree. An additional intellectual property rights grant can be found
   7  *  in the file PATENTS.  All contributing project authors may
   8  *  be found in the AUTHORS file in the root of the source tree.
   9  */
  10
  11 #include "webrtc/modules/audio_conference_mixer/source/audio_frame_manipulator.h"
  12 #include "webrtc/modules/interface/module_common_types.h"
  13 #include "webrtc/typedefs.h"
  14
  15 namespace {
  16 // Linear ramping over 80 samples.
  17 // TODO(hellner): ramp using fix point?
  18 const float rampArray[] = {0.0000f, 0.0127f, 0.0253f, 0.0380f,
  19                            0.0506f, 0.0633f, 0.0759f, 0.0886f,
  20                            0.1013f, 0.1139f, 0.1266f, 0.1392f,
  21                            0.1519f, 0.1646f, 0.1772f, 0.1899f,
  22                            0.2025f, 0.2152f, 0.2278f, 0.2405f,
  23                            0.2532f, 0.2658f, 0.2785f, 0.2911f,
  24                            0.3038f, 0.3165f, 0.3291f, 0.3418f,
  25                            0.3544f, 0.3671f, 0.3797f, 0.3924f,
  26                            0.4051f, 0.4177f, 0.4304f, 0.4430f,
  27                            0.4557f, 0.4684f, 0.4810f, 0.4937f,
  28                            0.5063f, 0.5190f, 0.5316f, 0.5443f,
  29                            0.5570f, 0.5696f, 0.5823f, 0.5949f,
  30                            0.6076f, 0.6203f, 0.6329f, 0.6456f,
  31                            0.6582f, 0.6709f, 0.6835f, 0.6962f,
  32                            0.7089f, 0.7215f, 0.7342f, 0.7468f,
  33                            0.7595f, 0.7722f, 0.7848f, 0.7975f,
  34                            0.8101f, 0.8228f, 0.8354f, 0.8481f,
  35                            0.8608f, 0.8734f, 0.8861f, 0.8987f,
  36                            0.9114f, 0.9241f, 0.9367f, 0.9494f,
  37                            0.9620f, 0.9747f, 0.9873f, 1.0000f};
  38 const int rampSize = sizeof(rampArray)/sizeof(rampArray[0]);
  39 }  // namespace
  40
  41 namespace webrtc {
  42 void CalculateEnergy(AudioFrame& audioFrame)
  43 {
  44     if(audioFrame.energy_ != 0xffffffff)
  45     {
  46         return;
  47     }
  48     audioFrame.energy_ = 0;
  49     for(int position = 0; position < audioFrame.samples_per_channel_;
  50         position++)
  51     {
  52         // TODO(andrew): this can easily overflow.
  53         audioFrame.energy_ += audioFrame.data_[position] *
  54                               audioFrame.data_[position];
  55     }
  56 }
  57
  58 void RampIn(AudioFrame& audioFrame)
  59 {
  60     assert(rampSize <= audioFrame.samples_per_channel_);
  61     for(int i = 0; i < rampSize; i++)
  62     {
  63         audioFrame.data_[i] = static_cast<int16_t>(rampArray[i] *
  64                                                    audioFrame.data_[i]);
  65     }
  66 }
  67
  68 void RampOut(AudioFrame& audioFrame)
  69 {
  70     assert(rampSize <= audioFrame.samples_per_channel_);
  71     for(int i = 0; i < rampSize; i++)
  72     {
  73         const int rampPos = rampSize - 1 - i;
  74         audioFrame.data_[i] = static_cast<int16_t>(rampArray[rampPos] *
  75                                                    audioFrame.data_[i]);
  76     }
  77     memset(&audioFrame.data_[rampSize], 0,
  78            (audioFrame.samples_per_channel_ - rampSize) *
  79            sizeof(audioFrame.data_[0]));
  80 }
  81 }  // namespace webrtc