/* * Copyright (c) 2011 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. */ #ifndef AUDIO_AUDIO_LEVEL_H_ #define AUDIO_AUDIO_LEVEL_H_ #include "rtc_base/synchronization/mutex.h" #include "rtc_base/thread_annotations.h" namespace webrtc { class AudioFrame; namespace voe { // This class is thread-safe. However, TotalEnergy() and TotalDuration() are // related, so if you call ComputeLevel() on a different thread than you read // these values, you still need to use lock to read them as a pair. class AudioLevel { public: AudioLevel(); ~AudioLevel(); void Reset(); // Returns the current audio level linearly [0,32767], which gets updated // every "kUpdateFrequency+1" call to ComputeLevel() based on the maximum // audio level of any audio frame, decaying by a factor of 1/4 each time // LevelFullRange() gets updated. // Called on "API thread(s)" from APIs like VoEBase::CreateChannel(), // VoEBase::StopSend(). int16_t LevelFullRange() const; void ResetLevelFullRange(); // See the description for "totalAudioEnergy" in the WebRTC stats spec // (https://w3c.github.io/webrtc-stats/#dom-rtcaudiohandlerstats-totalaudioenergy) // In our implementation, the total audio energy increases by the // energy-equivalent of LevelFullRange() at the time of ComputeLevel(), rather // than the energy of the samples in that specific audio frame. As a result, // we may report a higher audio energy and audio level than the spec mandates. // TODO(https://crbug.com/webrtc/10784): We should either do what the spec // says or update the spec to match our implementation. If we want to have a // decaying audio level we should probably update both the spec and the // implementation to reduce the complexity of the definition. If we want to // continue to have decaying audio we should have unittests covering the // behavior of the decay. double TotalEnergy() const; double TotalDuration() const; // Called on a native capture audio thread (platform dependent) from the // AudioTransport::RecordedDataIsAvailable() callback. // In Chrome, this method is called on the AudioInputDevice thread. void ComputeLevel(const AudioFrame& audioFrame, double duration); private: enum { kUpdateFrequency = 10 }; mutable Mutex mutex_; int16_t abs_max_ RTC_GUARDED_BY(mutex_); int16_t count_ RTC_GUARDED_BY(mutex_); int16_t current_level_full_range_ RTC_GUARDED_BY(mutex_); double total_energy_ RTC_GUARDED_BY(mutex_) = 0.0; double total_duration_ RTC_GUARDED_BY(mutex_) = 0.0; }; } // namespace voe } // namespace webrtc #endif // AUDIO_AUDIO_LEVEL_H_