Update audio_processing module
Corresponds to upstream commit 524e9b043e7e86fd72353b987c9d5f6a1ebf83e1 Update notes: * Pull in third party license file * Replace .gypi files with BUILD.gn to keep track of what changes upstream * Bunch of new filse pulled in as dependencies * Won't build yet due to changes needed on top of these
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webrtc/modules/audio_processing/rms_level.cc
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61
webrtc/modules/audio_processing/rms_level.cc
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/*
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* Copyright (c) 2014 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_processing/rms_level.h"
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#include <assert.h>
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#include <math.h>
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namespace webrtc {
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static const float kMaxSquaredLevel = 32768 * 32768;
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RMSLevel::RMSLevel()
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: sum_square_(0),
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sample_count_(0) {}
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RMSLevel::~RMSLevel() {}
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void RMSLevel::Reset() {
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sum_square_ = 0;
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sample_count_ = 0;
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}
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void RMSLevel::Process(const int16_t* data, size_t length) {
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for (size_t i = 0; i < length; ++i) {
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sum_square_ += data[i] * data[i];
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}
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sample_count_ += length;
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}
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void RMSLevel::ProcessMuted(size_t length) {
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sample_count_ += length;
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}
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int RMSLevel::RMS() {
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if (sample_count_ == 0 || sum_square_ == 0) {
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Reset();
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return kMinLevel;
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}
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// Normalize by the max level.
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float rms = sum_square_ / (sample_count_ * kMaxSquaredLevel);
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// 20log_10(x^0.5) = 10log_10(x)
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rms = 10 * log10(rms);
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assert(rms <= 0);
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if (rms < -kMinLevel)
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rms = -kMinLevel;
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rms = -rms;
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Reset();
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return static_cast<int>(rms + 0.5);
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}
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} // namespace webrtc
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