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a4d873786f
This CL was generated by running git ls-files | grep -P "(\.h|\.cc)$" | grep -v 'sdk/' | grep -v 'rtc_base/ssl_' | \ grep -v 'fake_rtc_certificate_generator.h' | grep -v 'modules/audio_device/win/' | \ grep -v 'system_wrappers/source/clock.cc' | grep -v 'rtc_base/trace_event.h' | \ grep -v 'modules/audio_coding/codecs/ilbc/' | grep -v 'screen_capturer_mac.h' | \ grep -v 'spl_inl_mips.h' | grep -v 'data_size_unittest.cc' | grep -v 'timestamp_unittest.cc' \ | xargs clang-format -i ; git cl format Most of these changes are clang-format grouping and reordering includes differently. Bug: webrtc:9340 Change-Id: Ic83ddbc169bfacd21883e381b5181c3dd4fe8a63 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/144051 Commit-Queue: Jonas Olsson <jonasolsson@webrtc.org> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Cr-Commit-Position: refs/heads/master@{#28505}
98 lines
3.1 KiB
C++
98 lines
3.1 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 "audio/audio_level.h"
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#include "api/audio/audio_frame.h"
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#include "common_audio/signal_processing/include/signal_processing_library.h"
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namespace webrtc {
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namespace voe {
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AudioLevel::AudioLevel()
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: abs_max_(0), count_(0), current_level_full_range_(0) {}
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AudioLevel::~AudioLevel() {}
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void AudioLevel::Reset() {
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rtc::CritScope cs(&crit_sect_);
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abs_max_ = 0;
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count_ = 0;
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current_level_full_range_ = 0;
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total_energy_ = 0.0;
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total_duration_ = 0.0;
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}
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int16_t AudioLevel::LevelFullRange() const {
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rtc::CritScope cs(&crit_sect_);
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return current_level_full_range_;
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}
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void AudioLevel::ResetLevelFullRange() {
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rtc::CritScope cs(&crit_sect_);
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abs_max_ = 0;
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count_ = 0;
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current_level_full_range_ = 0;
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}
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double AudioLevel::TotalEnergy() const {
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rtc::CritScope cs(&crit_sect_);
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return total_energy_;
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}
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double AudioLevel::TotalDuration() const {
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rtc::CritScope cs(&crit_sect_);
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return total_duration_;
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}
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void AudioLevel::ComputeLevel(const AudioFrame& audioFrame, double duration) {
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// Check speech level (works for 2 channels as well)
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int16_t abs_value =
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audioFrame.muted()
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? 0
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: WebRtcSpl_MaxAbsValueW16(
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audioFrame.data(),
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audioFrame.samples_per_channel_ * audioFrame.num_channels_);
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// Protect member access using a lock since this method is called on a
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// dedicated audio thread in the RecordedDataIsAvailable() callback.
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rtc::CritScope cs(&crit_sect_);
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if (abs_value > abs_max_)
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abs_max_ = abs_value;
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// Update level approximately 9 times per second, assuming audio frame
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// duration is approximately 10 ms. (The update frequency is every
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// 11th (= |kUpdateFrequency+1|) call: 1000/(11*10)=9.09..., we should
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// probably change this behavior, see https://crbug.com/webrtc/10784).
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if (count_++ == kUpdateFrequency) {
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current_level_full_range_ = abs_max_;
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count_ = 0;
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// Decay the absolute maximum (divide by 4)
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abs_max_ >>= 2;
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}
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// See the description for "totalAudioEnergy" in the WebRTC stats spec
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// (https://w3c.github.io/webrtc-stats/#dom-rtcmediastreamtrackstats-totalaudioenergy)
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// for an explanation of these formulas. In short, we need a value that can
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// be used to compute RMS audio levels over different time intervals, by
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// taking the difference between the results from two getStats calls. To do
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// this, the value needs to be of units "squared sample value * time".
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double additional_energy =
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static_cast<double>(current_level_full_range_) / INT16_MAX;
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additional_energy *= additional_energy;
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total_energy_ += additional_energy * duration;
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total_duration_ += duration;
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}
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} // namespace voe
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} // namespace webrtc
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