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/include/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 size_t rampSize = sizeof(rampArray)/sizeof(rampArray[0]); 39 } // namespace 40 41 namespace webrtc { 42 void CalculateEnergy(AudioFrame& audioFrame) 43 { 44 audioFrame.energy_ = 0; 45 for(size_t position = 0; position < audioFrame.samples_per_channel_; 46 position++) 47 { 48 // TODO(andrew): this can easily overflow. 49 audioFrame.energy_ += audioFrame.data_[position] * 50 audioFrame.data_[position]; 51 } 52 } 53 54 void RampIn(AudioFrame& audioFrame) 55 { 56 assert(rampSize <= audioFrame.samples_per_channel_); 57 for(size_t i = 0; i < rampSize; i++) 58 { 59 audioFrame.data_[i] = static_cast<int16_t>(rampArray[i] * 60 audioFrame.data_[i]); 61 } 62 } 63 64 void RampOut(AudioFrame& audioFrame) 65 { 66 assert(rampSize <= audioFrame.samples_per_channel_); 67 for(size_t i = 0; i < rampSize; i++) 68 { 69 const size_t rampPos = rampSize - 1 - i; 70 audioFrame.data_[i] = static_cast<int16_t>(rampArray[rampPos] * 71 audioFrame.data_[i]); 72 } 73 memset(&audioFrame.data_[rampSize], 0, 74 (audioFrame.samples_per_channel_ - rampSize) * 75 sizeof(audioFrame.data_[0])); 76 } 77 } // namespace webrtc 78