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diff --git a/third_party/libwebrtc/webrtc/call/audio_receive_stream.h b/third_party/libwebrtc/webrtc/call/audio_receive_stream.h
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+++ b/third_party/libwebrtc/webrtc/call/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 CALL_AUDIO_RECEIVE_STREAM_H_
+#define CALL_AUDIO_RECEIVE_STREAM_H_
+
+#include <map>
+#include <memory>
+#include <string>
+#include <vector>
+
+#include "api/audio_codecs/audio_decoder_factory.h"
+#include "api/call/transport.h"
+#include "api/optional.h"
+#include "api/rtpparameters.h"
+#include "api/rtpreceiverinterface.h"
+#include "call/rtp_config.h"
+#include "common_types.h"  // NOLINT(build/include)
+#include "rtc_base/scoped_ref_ptr.h"
+#include "typedefs.h"  // NOLINT(build/include)
+
+namespace webrtc {
+class AudioSinkInterface;
+
+// WORK IN PROGRESS
+// This class is under development and is not yet intended for for use outside
+// of WebRtc/Libjingle. Please use the VoiceEngine API instead.
+// See: https://bugs.chromium.org/p/webrtc/issues/detail?id=4690
+
+class AudioReceiveStream {
+ public:
+  struct Stats {
+    uint32_t remote_ssrc = 0;
+    int64_t bytes_rcvd = 0;
+    uint32_t packets_rcvd = 0;
+    uint32_t packets_lost = 0;
+    float fraction_lost = 0.0f;
+    std::string codec_name;
+    rtc::Optional<int> codec_payload_type;
+    uint32_t ext_seqnum = 0;
+    uint32_t jitter_ms = 0;
+    uint32_t jitter_buffer_ms = 0;
+    uint32_t jitter_buffer_preferred_ms = 0;
+    uint32_t delay_estimate_ms = 0;
+    int32_t audio_level = -1;
+    // Stats below correspond to similarly-named fields in the WebRTC stats
+    // spec. https://w3c.github.io/webrtc-stats/#dom-rtcmediastreamtrackstats
+    double total_output_energy = 0.0;
+    uint64_t total_samples_received = 0;
+    double total_output_duration = 0.0;
+    uint64_t concealed_samples = 0;
+    uint64_t concealment_events = 0;
+    double jitter_buffer_delay_seconds = 0.0;
+    // Stats below DO NOT correspond directly to anything in the WebRTC stats
+    float expand_rate = 0.0f;
+    float speech_expand_rate = 0.0f;
+    float secondary_decoded_rate = 0.0f;
+    float secondary_discarded_rate = 0.0f;
+    float accelerate_rate = 0.0f;
+    float preemptive_expand_rate = 0.0f;
+    int32_t decoding_calls_to_silence_generator = 0;
+    int32_t decoding_calls_to_neteq = 0;
+    int32_t decoding_normal = 0;
+    int32_t decoding_plc = 0;
+    int32_t decoding_cng = 0;
+    int32_t decoding_plc_cng = 0;
+    int32_t decoding_muted_output = 0;
+    int64_t capture_start_ntp_time_ms = 0;
+  };
+
+  struct Config {
+    std::string ToString() const;
+
+    // Receive-stream specific RTP settings.
+    struct Rtp {
+      std::string ToString() const;
+
+      // Synchronization source (stream identifier) to be received.
+      uint32_t remote_ssrc = 0;
+
+      // Sender SSRC used for sending RTCP (such as receiver reports).
+      uint32_t local_ssrc = 0;
+
+      // Enable feedback for send side bandwidth estimation.
+      // See
+      // https://tools.ietf.org/html/draft-holmer-rmcat-transport-wide-cc-extensions
+      // for details.
+      bool transport_cc = false;
+
+      // See NackConfig for description.
+      NackConfig nack;
+
+      // RTP header extensions used for the received stream.
+      std::vector<RtpExtension> extensions;
+    } rtp;
+
+    Transport* rtcp_send_transport = nullptr;
+
+    // Underlying VoiceEngine handle, used to map AudioReceiveStream to lower-
+    // level components.
+    // TODO(solenberg): Remove when VoiceEngine channels are created outside
+    // of Call.
+    int voe_channel_id = -1;
+
+    // Identifier for an A/V synchronization group. Empty string to disable.
+    // TODO(pbos): Synchronize streams in a sync group, not just one video
+    // stream to one audio stream. Tracked by issue webrtc:4762.
+    std::string sync_group;
+
+    // Decoder specifications for every payload type that we can receive.
+    std::map<int, SdpAudioFormat> decoder_map;
+
+    rtc::scoped_refptr<AudioDecoderFactory> decoder_factory;
+  };
+
+  // Starts stream activity.
+  // When a stream is active, it can receive, process and deliver packets.
+  virtual void Start() = 0;
+  // Stops stream activity.
+  // When a stream is stopped, it can't receive, process or deliver packets.
+  virtual void Stop() = 0;
+
+  virtual Stats GetStats() const = 0;
+  // TODO(solenberg): Remove, once AudioMonitor is gone.
+  virtual int GetOutputLevel() const = 0;
+
+  // Sets an audio sink that receives unmixed audio from the receive stream.
+  // Ownership of the sink is passed to the stream and can be used by the
+  // caller to do lifetime management (i.e. when the sink's dtor is called).
+  // Only one sink can be set and passing a null sink clears an existing one.
+  // NOTE: Audio must still somehow be pulled through AudioTransport for audio
+  // to stream through this sink. In practice, this happens if mixed audio
+  // is being pulled+rendered and/or if audio is being pulled for the purposes
+  // of feeding to the AEC.
+  virtual void SetSink(std::unique_ptr<AudioSinkInterface> sink) = 0;
+
+  // Sets playback gain of the stream, applied when mixing, and thus after it
+  // is potentially forwarded to any attached AudioSinkInterface implementation.
+  virtual void SetGain(float gain) = 0;
+
+  virtual std::vector<RtpSource> GetSources() const = 0;
+
+ protected:
+  virtual ~AudioReceiveStream() {}
+};
+}  // namespace webrtc
+
+#endif  // CALL_AUDIO_RECEIVE_STREAM_H_