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| 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 "modules/audio_conference_mixer/source/audio_frame_manipulator.h" | |
| 12 #include "modules/include/module_common_types.h" | |
| 13 #include "typedefs.h" // NOLINT(build/include) | |
| 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 uint32_t CalculateEnergy(const AudioFrame& audioFrame) | |
| 43 { | |
| 44 if (audioFrame.muted()) return 0; | |
| 45 | |
| 46 uint32_t energy = 0; | |
| 47 const int16_t* frame_data = audioFrame.data(); | |
| 48 for(size_t position = 0; position < audioFrame.samples_per_channel_; | |
| 49 position++) | |
| 50 { | |
| 51 // TODO(andrew): this can easily overflow. | |
| 52 energy += frame_data[position] * frame_data[position]; | |
| 53 } | |
| 54 return energy; | |
| 55 } | |
| 56 | |
| 57 void RampIn(AudioFrame& audioFrame) | |
| 58 { | |
| 59 assert(rampSize <= audioFrame.samples_per_channel_); | |
| 60 if (audioFrame.muted()) return; | |
| 61 | |
| 62 int16_t* frame_data = audioFrame.mutable_data(); | |
| 63 for(size_t i = 0; i < rampSize; i++) | |
| 64 { | |
| 65 frame_data[i] = static_cast<int16_t>(rampArray[i] * frame_data[i]); | |
| 66 } | |
| 67 } | |
| 68 | |
| 69 void RampOut(AudioFrame& audioFrame) | |
| 70 { | |
| 71 assert(rampSize <= audioFrame.samples_per_channel_); | |
| 72 if (audioFrame.muted()) return; | |
| 73 | |
| 74 int16_t* frame_data = audioFrame.mutable_data(); | |
| 75 for(size_t i = 0; i < rampSize; i++) | |
| 76 { | |
| 77 const size_t rampPos = rampSize - 1 - i; | |
| 78 frame_data[i] = static_cast<int16_t>(rampArray[rampPos] * | |
| 79 frame_data[i]); | |
| 80 } | |
| 81 memset(&frame_data[rampSize], 0, | |
| 82 (audioFrame.samples_per_channel_ - rampSize) * | |
| 83 sizeof(frame_data[0])); | |
| 84 } | |
| 85 } // namespace webrtc | |
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