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author | Daniel Baumann <daniel.baumann@progress-linux.org> | 2024-04-19 00:47:55 +0000 |
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committer | Daniel Baumann <daniel.baumann@progress-linux.org> | 2024-04-19 00:47:55 +0000 |
commit | 26a029d407be480d791972afb5975cf62c9360a6 (patch) | |
tree | f435a8308119effd964b339f76abb83a57c29483 /third_party/libwebrtc/audio/audio_level.h | |
parent | Initial commit. (diff) | |
download | firefox-26a029d407be480d791972afb5975cf62c9360a6.tar.xz firefox-26a029d407be480d791972afb5975cf62c9360a6.zip |
Adding upstream version 124.0.1.upstream/124.0.1
Signed-off-by: Daniel Baumann <daniel.baumann@progress-linux.org>
Diffstat (limited to 'third_party/libwebrtc/audio/audio_level.h')
-rw-r--r-- | third_party/libwebrtc/audio/audio_level.h | 75 |
1 files changed, 75 insertions, 0 deletions
diff --git a/third_party/libwebrtc/audio/audio_level.h b/third_party/libwebrtc/audio/audio_level.h new file mode 100644 index 0000000000..acd1231fe2 --- /dev/null +++ b/third_party/libwebrtc/audio/audio_level.h @@ -0,0 +1,75 @@ +/* + * 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_ |