| /* |
| * Copyright (c) 2012 The WebRTC project authors. All Rights Reserved. |
| * |
| * Use of this source code is governed by a BSD-style license |
| * that can be found in the LICENSE file in the root of the source |
| * tree. An additional intellectual property rights grant can be found |
| * in the file PATENTS. All contributing project authors may |
| * be found in the AUTHORS file in the root of the source tree. |
| */ |
| |
| #include "audio/audio_level.h" |
| |
| #include "api/audio/audio_frame.h" |
| #include "common_audio/signal_processing/include/signal_processing_library.h" |
| |
| namespace webrtc { |
| namespace voe { |
| |
| AudioLevel::AudioLevel() |
| : abs_max_(0), count_(0), current_level_full_range_(0) { |
| } |
| |
| AudioLevel::~AudioLevel() {} |
| |
| void AudioLevel::Reset() { |
| rtc::CritScope cs(&crit_sect_); |
| abs_max_ = 0; |
| count_ = 0; |
| current_level_full_range_ = 0; |
| total_energy_ = 0.0; |
| total_duration_ = 0.0; |
| } |
| |
| int16_t AudioLevel::LevelFullRange() const { |
| rtc::CritScope cs(&crit_sect_); |
| return current_level_full_range_; |
| } |
| |
| void AudioLevel::ResetLevelFullRange() { |
| rtc::CritScope cs(&crit_sect_); |
| abs_max_ = 0; |
| count_ = 0; |
| current_level_full_range_ = 0; |
| } |
| |
| double AudioLevel::TotalEnergy() const { |
| rtc::CritScope cs(&crit_sect_); |
| return total_energy_; |
| } |
| |
| double AudioLevel::TotalDuration() const { |
| rtc::CritScope cs(&crit_sect_); |
| return total_duration_; |
| } |
| |
| void AudioLevel::ComputeLevel(const AudioFrame& audioFrame, double duration) { |
| // Check speech level (works for 2 channels as well) |
| int16_t abs_value = |
| audioFrame.muted() |
| ? 0 |
| : WebRtcSpl_MaxAbsValueW16( |
| audioFrame.data(), |
| audioFrame.samples_per_channel_ * audioFrame.num_channels_); |
| |
| // Protect member access using a lock since this method is called on a |
| // dedicated audio thread in the RecordedDataIsAvailable() callback. |
| rtc::CritScope cs(&crit_sect_); |
| |
| if (abs_value > abs_max_) |
| abs_max_ = abs_value; |
| |
| // Update level approximately 9 times per second, assuming audio frame |
| // duration is approximately 10 ms. (The update frequency is every |
| // 11th (= |kUpdateFrequency+1|) call: 1000/(11*10)=9.09..., we should |
| // probably change this behavior, see https://crbug.com/webrtc/10784). |
| if (count_++ == kUpdateFrequency) { |
| current_level_full_range_ = abs_max_; |
| |
| count_ = 0; |
| |
| // Decay the absolute maximum (divide by 4) |
| abs_max_ >>= 2; |
| } |
| |
| // See the description for "totalAudioEnergy" in the WebRTC stats spec |
| // (https://w3c.github.io/webrtc-stats/#dom-rtcmediastreamtrackstats-totalaudioenergy) |
| // for an explanation of these formulas. In short, we need a value that can |
| // be used to compute RMS audio levels over different time intervals, by |
| // taking the difference between the results from two getStats calls. To do |
| // this, the value needs to be of units "squared sample value * time". |
| double additional_energy = |
| static_cast<double>(current_level_full_range_) / INT16_MAX; |
| additional_energy *= additional_energy; |
| total_energy_ += additional_energy * duration; |
| total_duration_ += duration; |
| } |
| |
| } // namespace voe |
| } // namespace webrtc |