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/*
* 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_
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