2ae5fdff86
This required quite a few small changes in the mixing algorithm structure, the mixer interface and the mixer unit tests. BUG=webrtc:6346 Review-Url: https://codereview.webrtc.org/2396483002 Cr-Commit-Position: refs/heads/master@{#14567}
63 lines
2.6 KiB
C++
63 lines
2.6 KiB
C++
/*
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* Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
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*
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* Use of this source code is governed by a BSD-style license
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* that can be found in the LICENSE file in the root of the source
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* tree. An additional intellectual property rights grant can be found
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* in the file PATENTS. All contributing project authors may
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* be found in the AUTHORS file in the root of the source tree.
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*/
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#include "webrtc/modules/audio_mixer/audio_frame_manipulator.h"
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#include "webrtc/modules/include/module_common_types.h"
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#include "webrtc/typedefs.h"
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namespace webrtc {
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namespace {
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// Linear ramping over 80 samples.
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// TODO(hellner): ramp using fix point?
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const float kRampArray[] = {
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0.0000f, 0.0127f, 0.0253f, 0.0380f, 0.0506f, 0.0633f, 0.0759f, 0.0886f,
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0.1013f, 0.1139f, 0.1266f, 0.1392f, 0.1519f, 0.1646f, 0.1772f, 0.1899f,
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0.2025f, 0.2152f, 0.2278f, 0.2405f, 0.2532f, 0.2658f, 0.2785f, 0.2911f,
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0.3038f, 0.3165f, 0.3291f, 0.3418f, 0.3544f, 0.3671f, 0.3797f, 0.3924f,
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0.4051f, 0.4177f, 0.4304f, 0.4430f, 0.4557f, 0.4684f, 0.4810f, 0.4937f,
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0.5063f, 0.5190f, 0.5316f, 0.5443f, 0.5570f, 0.5696f, 0.5823f, 0.5949f,
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0.6076f, 0.6203f, 0.6329f, 0.6456f, 0.6582f, 0.6709f, 0.6835f, 0.6962f,
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0.7089f, 0.7215f, 0.7342f, 0.7468f, 0.7595f, 0.7722f, 0.7848f, 0.7975f,
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0.8101f, 0.8228f, 0.8354f, 0.8481f, 0.8608f, 0.8734f, 0.8861f, 0.8987f,
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0.9114f, 0.9241f, 0.9367f, 0.9494f, 0.9620f, 0.9747f, 0.9873f, 1.0000f};
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const size_t kRampSize = sizeof(kRampArray) / sizeof(kRampArray[0]);
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} // namespace
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uint32_t AudioMixerCalculateEnergy(const AudioFrame& audio_frame) {
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uint32_t energy = 0;
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for (size_t position = 0; position < audio_frame.samples_per_channel_;
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position++) {
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// TODO(aleloi): This can overflow. Convert to floats.
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energy += audio_frame.data_[position] * audio_frame.data_[position];
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}
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return energy;
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}
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void NewMixerRampIn(AudioFrame* audio_frame) {
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assert(kRampSize <= audio_frame->samples_per_channel_);
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for (size_t i = 0; i < kRampSize; i++) {
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audio_frame->data_[i] =
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static_cast<int16_t>(kRampArray[i] * audio_frame->data_[i]);
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}
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}
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void NewMixerRampOut(AudioFrame* audio_frame) {
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assert(kRampSize <= audio_frame->samples_per_channel_);
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for (size_t i = 0; i < kRampSize; i++) {
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const size_t kRampPos = kRampSize - 1 - i;
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audio_frame->data_[i] =
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static_cast<int16_t>(kRampArray[kRampPos] * audio_frame->data_[i]);
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}
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memset(&audio_frame->data_[kRampSize], 0,
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(audio_frame->samples_per_channel_ - kRampSize) *
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sizeof(audio_frame->data_[0]));
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}
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} // namespace webrtc
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