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diff --git a/third_party/libwebrtc/modules/audio_device/include/audio_device.h b/third_party/libwebrtc/modules/audio_device/include/audio_device.h
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index 0000000000..936ee6cb04
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+++ b/third_party/libwebrtc/modules/audio_device/include/audio_device.h
@@ -0,0 +1,194 @@
+/*
+ *  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_INCLUDE_AUDIO_DEVICE_H_
+#define MODULES_AUDIO_DEVICE_INCLUDE_AUDIO_DEVICE_H_
+
+#include "absl/types/optional.h"
+#include "api/scoped_refptr.h"
+#include "api/task_queue/task_queue_factory.h"
+#include "modules/audio_device/include/audio_device_defines.h"
+#include "rtc_base/ref_count.h"
+
+namespace webrtc {
+
+class AudioDeviceModuleForTest;
+
+class AudioDeviceModule : public rtc::RefCountInterface {
+ public:
+  enum AudioLayer {
+    kPlatformDefaultAudio = 0,
+    kWindowsCoreAudio,
+    kWindowsCoreAudio2,
+    kLinuxAlsaAudio,
+    kLinuxPulseAudio,
+    kAndroidJavaAudio,
+    kAndroidOpenSLESAudio,
+    kAndroidJavaInputAndOpenSLESOutputAudio,
+    kAndroidAAudioAudio,
+    kAndroidJavaInputAndAAudioOutputAudio,
+    kDummyAudio,
+  };
+
+  enum WindowsDeviceType {
+    kDefaultCommunicationDevice = -1,
+    kDefaultDevice = -2
+  };
+
+  struct Stats {
+    // The fields below correspond to similarly-named fields in the WebRTC stats
+    // spec. https://w3c.github.io/webrtc-stats/#playoutstats-dict*
+    double synthesized_samples_duration_s = 0;
+    uint64_t synthesized_samples_events = 0;
+    double total_samples_duration_s = 0;
+    double total_playout_delay_s = 0;
+    uint64_t total_samples_count = 0;
+  };
+
+ public:
+  // Creates a default ADM for usage in production code.
+  static rtc::scoped_refptr<AudioDeviceModule> Create(
+      AudioLayer audio_layer,
+      TaskQueueFactory* task_queue_factory);
+  // Creates an ADM with support for extra test methods. Don't use this factory
+  // in production code.
+  static rtc::scoped_refptr<AudioDeviceModuleForTest> CreateForTest(
+      AudioLayer audio_layer,
+      TaskQueueFactory* task_queue_factory);
+
+  // Retrieve the currently utilized audio layer
+  virtual int32_t ActiveAudioLayer(AudioLayer* audioLayer) const = 0;
+
+  // Full-duplex transportation of PCM audio
+  virtual int32_t RegisterAudioCallback(AudioTransport* audioCallback) = 0;
+
+  // Main initialization and termination
+  virtual int32_t Init() = 0;
+  virtual int32_t Terminate() = 0;
+  virtual bool Initialized() const = 0;
+
+  // Device enumeration
+  virtual int16_t PlayoutDevices() = 0;
+  virtual int16_t RecordingDevices() = 0;
+  virtual int32_t PlayoutDeviceName(uint16_t index,
+                                    char name[kAdmMaxDeviceNameSize],
+                                    char guid[kAdmMaxGuidSize]) = 0;
+  virtual int32_t RecordingDeviceName(uint16_t index,
+                                      char name[kAdmMaxDeviceNameSize],
+                                      char guid[kAdmMaxGuidSize]) = 0;
+
+  // Device selection
+  virtual int32_t SetPlayoutDevice(uint16_t index) = 0;
+  virtual int32_t SetPlayoutDevice(WindowsDeviceType device) = 0;
+  virtual int32_t SetRecordingDevice(uint16_t index) = 0;
+  virtual int32_t SetRecordingDevice(WindowsDeviceType device) = 0;
+
+  // Audio transport initialization
+  virtual int32_t PlayoutIsAvailable(bool* available) = 0;
+  virtual int32_t InitPlayout() = 0;
+  virtual bool PlayoutIsInitialized() const = 0;
+  virtual int32_t RecordingIsAvailable(bool* available) = 0;
+  virtual int32_t InitRecording() = 0;
+  virtual bool RecordingIsInitialized() const = 0;
+
+  // Audio transport control
+  virtual int32_t StartPlayout() = 0;
+  virtual int32_t StopPlayout() = 0;
+  virtual bool Playing() const = 0;
+  virtual int32_t StartRecording() = 0;
+  virtual int32_t StopRecording() = 0;
+  virtual bool Recording() const = 0;
+
+  // Audio mixer initialization
+  virtual int32_t InitSpeaker() = 0;
+  virtual bool SpeakerIsInitialized() const = 0;
+  virtual int32_t InitMicrophone() = 0;
+  virtual bool MicrophoneIsInitialized() const = 0;
+
+  // Speaker volume controls
+  virtual int32_t SpeakerVolumeIsAvailable(bool* available) = 0;
+  virtual int32_t SetSpeakerVolume(uint32_t volume) = 0;
+  virtual int32_t SpeakerVolume(uint32_t* volume) const = 0;
+  virtual int32_t MaxSpeakerVolume(uint32_t* maxVolume) const = 0;
+  virtual int32_t MinSpeakerVolume(uint32_t* minVolume) const = 0;
+
+  // Microphone volume controls
+  virtual int32_t MicrophoneVolumeIsAvailable(bool* available) = 0;
+  virtual int32_t SetMicrophoneVolume(uint32_t volume) = 0;
+  virtual int32_t MicrophoneVolume(uint32_t* volume) const = 0;
+  virtual int32_t MaxMicrophoneVolume(uint32_t* maxVolume) const = 0;
+  virtual int32_t MinMicrophoneVolume(uint32_t* minVolume) const = 0;
+
+  // Speaker mute control
+  virtual int32_t SpeakerMuteIsAvailable(bool* available) = 0;
+  virtual int32_t SetSpeakerMute(bool enable) = 0;
+  virtual int32_t SpeakerMute(bool* enabled) const = 0;
+
+  // Microphone mute control
+  virtual int32_t MicrophoneMuteIsAvailable(bool* available) = 0;
+  virtual int32_t SetMicrophoneMute(bool enable) = 0;
+  virtual int32_t MicrophoneMute(bool* enabled) const = 0;
+
+  // Stereo support
+  virtual int32_t StereoPlayoutIsAvailable(bool* available) const = 0;
+  virtual int32_t SetStereoPlayout(bool enable) = 0;
+  virtual int32_t StereoPlayout(bool* enabled) const = 0;
+  virtual int32_t StereoRecordingIsAvailable(bool* available) const = 0;
+  virtual int32_t SetStereoRecording(bool enable) = 0;
+  virtual int32_t StereoRecording(bool* enabled) const = 0;
+
+  // Playout delay
+  virtual int32_t PlayoutDelay(uint16_t* delayMS) const = 0;
+
+  // Only supported on Android.
+  virtual bool BuiltInAECIsAvailable() const = 0;
+  virtual bool BuiltInAGCIsAvailable() const = 0;
+  virtual bool BuiltInNSIsAvailable() const = 0;
+
+  // Enables the built-in audio effects. Only supported on Android.
+  virtual int32_t EnableBuiltInAEC(bool enable) = 0;
+  virtual int32_t EnableBuiltInAGC(bool enable) = 0;
+  virtual int32_t EnableBuiltInNS(bool enable) = 0;
+
+  // Play underrun count. Only supported on Android.
+  // TODO(alexnarest): Make it abstract after upstream projects support it.
+  virtual int32_t GetPlayoutUnderrunCount() const { return -1; }
+
+  // Used to generate RTC stats. If not implemented, RTCAudioPlayoutStats will
+  // not be present in the stats.
+  virtual absl::optional<Stats> GetStats() const { return absl::nullopt; }
+
+// Only supported on iOS.
+#if defined(WEBRTC_IOS)
+  virtual int GetPlayoutAudioParameters(AudioParameters* params) const = 0;
+  virtual int GetRecordAudioParameters(AudioParameters* params) const = 0;
+#endif  // WEBRTC_IOS
+
+ protected:
+  ~AudioDeviceModule() override {}
+};
+
+// Extends the default ADM interface with some extra test methods.
+// Intended for usage in tests only and requires a unique factory method.
+class AudioDeviceModuleForTest : public AudioDeviceModule {
+ public:
+  // Triggers internal restart sequences of audio streaming. Can be used by
+  // tests to emulate events corresponding to e.g. removal of an active audio
+  // device or other actions which causes the stream to be disconnected.
+  virtual int RestartPlayoutInternally() = 0;
+  virtual int RestartRecordingInternally() = 0;
+
+  virtual int SetPlayoutSampleRate(uint32_t sample_rate) = 0;
+  virtual int SetRecordingSampleRate(uint32_t sample_rate) = 0;
+};
+
+}  // namespace webrtc
+
+#endif  // MODULES_AUDIO_DEVICE_INCLUDE_AUDIO_DEVICE_H_