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diff --git a/third_party/libwebrtc/modules/audio_device/audio_device_buffer.h b/third_party/libwebrtc/modules/audio_device/audio_device_buffer.h
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+++ b/third_party/libwebrtc/modules/audio_device/audio_device_buffer.h
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+/*
+ *  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.
+ */
+
+#ifndef MODULES_AUDIO_DEVICE_AUDIO_DEVICE_BUFFER_H_
+#define MODULES_AUDIO_DEVICE_AUDIO_DEVICE_BUFFER_H_
+
+#include <stddef.h>
+#include <stdint.h>
+
+#include <atomic>
+
+#include "api/sequence_checker.h"
+#include "api/task_queue/task_queue_factory.h"
+#include "modules/audio_device/include/audio_device_defines.h"
+#include "rtc_base/buffer.h"
+#include "rtc_base/synchronization/mutex.h"
+#include "rtc_base/task_queue.h"
+#include "rtc_base/thread_annotations.h"
+#include "rtc_base/timestamp_aligner.h"
+
+namespace webrtc {
+
+// Delta times between two successive playout callbacks are limited to this
+// value before added to an internal array.
+const size_t kMaxDeltaTimeInMs = 500;
+// TODO(henrika): remove when no longer used by external client.
+const size_t kMaxBufferSizeBytes = 3840;  // 10ms in stereo @ 96kHz
+
+class AudioDeviceBuffer {
+ public:
+  enum LogState {
+    LOG_START = 0,
+    LOG_STOP,
+    LOG_ACTIVE,
+  };
+
+  struct Stats {
+    void ResetRecStats() {
+      rec_callbacks = 0;
+      rec_samples = 0;
+      max_rec_level = 0;
+    }
+
+    void ResetPlayStats() {
+      play_callbacks = 0;
+      play_samples = 0;
+      max_play_level = 0;
+    }
+
+    // Total number of recording callbacks where the source provides 10ms audio
+    // data each time.
+    uint64_t rec_callbacks = 0;
+
+    // Total number of playback callbacks where the sink asks for 10ms audio
+    // data each time.
+    uint64_t play_callbacks = 0;
+
+    // Total number of recorded audio samples.
+    uint64_t rec_samples = 0;
+
+    // Total number of played audio samples.
+    uint64_t play_samples = 0;
+
+    // Contains max level (max(abs(x))) of recorded audio packets over the last
+    // 10 seconds where a new measurement is done twice per second. The level
+    // is reset to zero at each call to LogStats().
+    int16_t max_rec_level = 0;
+
+    // Contains max level of recorded audio packets over the last 10 seconds
+    // where a new measurement is done twice per second.
+    int16_t max_play_level = 0;
+  };
+
+  // If `create_detached` is true, the created buffer can be used on another
+  // thread compared to the one on which it was created. It's useful for
+  // testing.
+  explicit AudioDeviceBuffer(TaskQueueFactory* task_queue_factory,
+                             bool create_detached = false);
+  virtual ~AudioDeviceBuffer();
+
+  int32_t RegisterAudioCallback(AudioTransport* audio_callback);
+
+  void StartPlayout();
+  void StartRecording();
+  void StopPlayout();
+  void StopRecording();
+
+  int32_t SetRecordingSampleRate(uint32_t fsHz);
+  int32_t SetPlayoutSampleRate(uint32_t fsHz);
+  uint32_t RecordingSampleRate() const;
+  uint32_t PlayoutSampleRate() const;
+
+  int32_t SetRecordingChannels(size_t channels);
+  int32_t SetPlayoutChannels(size_t channels);
+  size_t RecordingChannels() const;
+  size_t PlayoutChannels() const;
+
+  // TODO(bugs.webrtc.org/13621) Deprecate this function
+  virtual int32_t SetRecordedBuffer(const void* audio_buffer,
+                                    size_t samples_per_channel);
+
+  virtual int32_t SetRecordedBuffer(
+      const void* audio_buffer,
+      size_t samples_per_channel,
+      absl::optional<int64_t> capture_timestamp_ns);
+  virtual void SetVQEData(int play_delay_ms, int rec_delay_ms);
+  virtual int32_t DeliverRecordedData();
+  uint32_t NewMicLevel() const;
+
+  virtual int32_t RequestPlayoutData(size_t samples_per_channel);
+  virtual int32_t GetPlayoutData(void* audio_buffer);
+
+  int32_t SetTypingStatus(bool typing_status);
+
+ private:
+  // Starts/stops periodic logging of audio stats.
+  void StartPeriodicLogging();
+  void StopPeriodicLogging();
+
+  // Called periodically on the internal thread created by the TaskQueue.
+  // Updates some stats but dooes it on the task queue to ensure that access of
+  // members is serialized hence avoiding usage of locks.
+  // state = LOG_START => members are initialized and the timer starts.
+  // state = LOG_STOP => no logs are printed and the timer stops.
+  // state = LOG_ACTIVE => logs are printed and the timer is kept alive.
+  void LogStats(LogState state);
+
+  // Updates counters in each play/record callback. These counters are later
+  // (periodically) read by LogStats() using a lock.
+  void UpdateRecStats(int16_t max_abs, size_t samples_per_channel);
+  void UpdatePlayStats(int16_t max_abs, size_t samples_per_channel);
+
+  // Clears all members tracking stats for recording and playout.
+  // These methods both run on the task queue.
+  void ResetRecStats();
+  void ResetPlayStats();
+
+  // This object lives on the main (creating) thread and most methods are
+  // called on that same thread. When audio has started some methods will be
+  // called on either a native audio thread for playout or a native thread for
+  // recording. Some members are not annotated since they are "protected by
+  // design" and adding e.g. a race checker can cause failures for very few
+  // edge cases and it is IMHO not worth the risk to use them in this class.
+  // TODO(henrika): see if it is possible to refactor and annotate all members.
+
+  // Main thread on which this object is created.
+  SequenceChecker main_thread_checker_;
+
+  Mutex lock_;
+
+  // Task queue used to invoke LogStats() periodically. Tasks are executed on a
+  // worker thread but it does not necessarily have to be the same thread for
+  // each task.
+  rtc::TaskQueue task_queue_;
+
+  // Raw pointer to AudioTransport instance. Supplied to RegisterAudioCallback()
+  // and it must outlive this object. It is not possible to change this member
+  // while any media is active. It is possible to start media without calling
+  // RegisterAudioCallback() but that will lead to ignored audio callbacks in
+  // both directions where native audio will be active but no audio samples will
+  // be transported.
+  AudioTransport* audio_transport_cb_;
+
+  // Sample rate in Hertz. Accessed atomically.
+  std::atomic<uint32_t> rec_sample_rate_;
+  std::atomic<uint32_t> play_sample_rate_;
+
+  // Number of audio channels. Accessed atomically.
+  std::atomic<size_t> rec_channels_;
+  std::atomic<size_t> play_channels_;
+
+  // Keeps track of if playout/recording are active or not. A combination
+  // of these states are used to determine when to start and stop the timer.
+  // Only used on the creating thread and not used to control any media flow.
+  bool playing_ RTC_GUARDED_BY(main_thread_checker_);
+  bool recording_ RTC_GUARDED_BY(main_thread_checker_);
+
+  // Buffer used for audio samples to be played out. Size can be changed
+  // dynamically. The 16-bit samples are interleaved, hence the size is
+  // proportional to the number of channels.
+  rtc::BufferT<int16_t> play_buffer_;
+
+  // Byte buffer used for recorded audio samples. Size can be changed
+  // dynamically.
+  rtc::BufferT<int16_t> rec_buffer_;
+
+  // Contains true of a key-press has been detected.
+  bool typing_status_;
+
+  // Delay values used by the AEC.
+  int play_delay_ms_;
+  int rec_delay_ms_;
+
+  // Capture timestamp.
+  absl::optional<int64_t> capture_timestamp_ns_;
+  // The last time the Timestamp Aligner was used to estimate clock offset
+  // between system clock and capture time from audio.
+  // This is used to prevent estimating the clock offset too often.
+  absl::optional<int64_t> align_offsync_estimation_time_;
+
+  // Counts number of times LogStats() has been called.
+  size_t num_stat_reports_ RTC_GUARDED_BY(task_queue_);
+
+  // Time stamp of last timer task (drives logging).
+  int64_t last_timer_task_time_ RTC_GUARDED_BY(task_queue_);
+
+  // Counts number of audio callbacks modulo 50 to create a signal when
+  // a new storage of audio stats shall be done.
+  int16_t rec_stat_count_;
+  int16_t play_stat_count_;
+
+  // Time stamps of when playout and recording starts.
+  int64_t play_start_time_ RTC_GUARDED_BY(main_thread_checker_);
+  int64_t rec_start_time_ RTC_GUARDED_BY(main_thread_checker_);
+
+  // Contains counters for playout and recording statistics.
+  Stats stats_ RTC_GUARDED_BY(lock_);
+
+  // Stores current stats at each timer task. Used to calculate differences
+  // between two successive timer events.
+  Stats last_stats_ RTC_GUARDED_BY(task_queue_);
+
+  // Set to true at construction and modified to false as soon as one audio-
+  // level estimate larger than zero is detected.
+  bool only_silence_recorded_;
+
+  // Set to true when logging of audio stats is enabled for the first time in
+  // StartPeriodicLogging() and set to false by StopPeriodicLogging().
+  // Setting this member to false prevents (possiby invalid) log messages from
+  // being printed in the LogStats() task.
+  bool log_stats_ RTC_GUARDED_BY(task_queue_);
+
+  // Used for converting capture timestaps (received from AudioRecordThread
+  // via AudioRecordJni::DataIsRecorded) to RTC clock.
+  rtc::TimestampAligner timestamp_aligner_;
+
+// Should *never* be defined in production builds. Only used for testing.
+// When defined, the output signal will be replaced by a sinus tone at 440Hz.
+#ifdef AUDIO_DEVICE_PLAYS_SINUS_TONE
+  double phase_;
+#endif
+};
+
+}  // namespace webrtc
+
+#endif  // MODULES_AUDIO_DEVICE_AUDIO_DEVICE_BUFFER_H_