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diff --git a/third_party/libwebrtc/webrtc/audio/audio_receive_stream.h b/third_party/libwebrtc/webrtc/audio/audio_receive_stream.h
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+++ b/third_party/libwebrtc/webrtc/audio/audio_receive_stream.h
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+/*
+ *  Copyright (c) 2015 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_RECEIVE_STREAM_H_
+#define AUDIO_AUDIO_RECEIVE_STREAM_H_
+
+#include <memory>
+#include <vector>
+
+#include "api/audio/audio_mixer.h"
+#include "audio/audio_state.h"
+#include "call/audio_receive_stream.h"
+#include "call/rtp_packet_sink_interface.h"
+#include "call/syncable.h"
+#include "rtc_base/constructormagic.h"
+#include "rtc_base/thread_checker.h"
+
+namespace webrtc {
+class PacketRouter;
+class RtcEventLog;
+class RtpPacketReceived;
+class RtpStreamReceiverControllerInterface;
+class RtpStreamReceiverInterface;
+
+namespace voe {
+class ChannelProxy;
+}  // namespace voe
+
+namespace internal {
+class AudioSendStream;
+
+class AudioReceiveStream final : public webrtc::AudioReceiveStream,
+                                 public AudioMixer::Source,
+                                 public Syncable {
+ public:
+  AudioReceiveStream(RtpStreamReceiverControllerInterface* receiver_controller,
+                     PacketRouter* packet_router,
+                     const webrtc::AudioReceiveStream::Config& config,
+                     const rtc::scoped_refptr<webrtc::AudioState>& audio_state,
+                     webrtc::RtcEventLog* event_log);
+  ~AudioReceiveStream() override;
+
+  // webrtc::AudioReceiveStream implementation.
+  void Start() override;
+  void Stop() override;
+  webrtc::AudioReceiveStream::Stats GetStats() const override;
+  int GetOutputLevel() const override;
+  void SetSink(std::unique_ptr<AudioSinkInterface> sink) override;
+  void SetGain(float gain) override;
+  std::vector<webrtc::RtpSource> GetSources() const override;
+
+  // TODO(nisse): We don't formally implement RtpPacketSinkInterface, and this
+  // method shouldn't be needed. But it's currently used by the
+  // AudioReceiveStreamTest.ReceiveRtpPacket unittest. Figure out if that test
+  // shuld be refactored or deleted, and then delete this method.
+  void OnRtpPacket(const RtpPacketReceived& packet);
+
+  // AudioMixer::Source
+  AudioFrameInfo GetAudioFrameWithInfo(int sample_rate_hz,
+                                       AudioFrame* audio_frame) override;
+  int Ssrc() const override;
+  int PreferredSampleRate() const override;
+
+  // Syncable
+  int id() const override;
+  rtc::Optional<Syncable::Info> GetInfo() const override;
+  uint32_t GetPlayoutTimestamp() const override;
+  void SetMinimumPlayoutDelay(int delay_ms) override;
+
+  void AssociateSendStream(AudioSendStream* send_stream);
+  void SignalNetworkState(NetworkState state);
+  bool DeliverRtcp(const uint8_t* packet, size_t length);
+  const webrtc::AudioReceiveStream::Config& config() const;
+
+ private:
+  VoiceEngine* voice_engine() const;
+  AudioState* audio_state() const;
+  int SetVoiceEnginePlayout(bool playout);
+
+  rtc::ThreadChecker worker_thread_checker_;
+  rtc::ThreadChecker module_process_thread_checker_;
+  const webrtc::AudioReceiveStream::Config config_;
+  rtc::scoped_refptr<webrtc::AudioState> audio_state_;
+  std::unique_ptr<voe::ChannelProxy> channel_proxy_;
+
+  bool playing_ RTC_ACCESS_ON(worker_thread_checker_) = false;
+
+  std::unique_ptr<RtpStreamReceiverInterface> rtp_stream_receiver_;
+
+  RTC_DISALLOW_IMPLICIT_CONSTRUCTORS(AudioReceiveStream);
+};
+}  // namespace internal
+}  // namespace webrtc
+
+#endif  // AUDIO_AUDIO_RECEIVE_STREAM_H_