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diff --git a/third_party/libwebrtc/media/base/media_engine.h b/third_party/libwebrtc/media/base/media_engine.h
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+++ b/third_party/libwebrtc/media/base/media_engine.h
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+/*
+ *  Copyright (c) 2004 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 MEDIA_BASE_MEDIA_ENGINE_H_
+#define MEDIA_BASE_MEDIA_ENGINE_H_
+
+#include <memory>
+#include <string>
+#include <vector>
+
+#include "api/audio_codecs/audio_decoder_factory.h"
+#include "api/audio_codecs/audio_encoder_factory.h"
+#include "api/crypto/crypto_options.h"
+#include "api/field_trials_view.h"
+#include "api/rtp_parameters.h"
+#include "api/video/video_bitrate_allocator_factory.h"
+#include "call/audio_state.h"
+#include "media/base/codec.h"
+#include "media/base/media_channel.h"
+#include "media/base/media_channel_impl.h"
+#include "media/base/media_config.h"
+#include "media/base/video_common.h"
+#include "rtc_base/system/file_wrapper.h"
+
+namespace webrtc {
+class AudioDeviceModule;
+class AudioMixer;
+class AudioProcessing;
+class Call;
+}  // namespace webrtc
+
+namespace cricket {
+
+// Checks that the scalability_mode value of each encoding is supported by at
+// least one video codec of the list. If the list is empty, no check is done.
+webrtc::RTCError CheckScalabilityModeValues(
+    const webrtc::RtpParameters& new_parameters,
+    rtc::ArrayView<cricket::Codec> codec_preferences,
+    absl::optional<cricket::Codec> send_codec);
+
+// Checks the parameters have valid and supported values, and checks parameters
+// with CheckScalabilityModeValues().
+webrtc::RTCError CheckRtpParametersValues(
+    const webrtc::RtpParameters& new_parameters,
+    rtc::ArrayView<cricket::Codec> codec_preferences,
+    absl::optional<cricket::Codec> send_codec);
+
+// Checks that the immutable values have not changed in new_parameters and
+// checks all parameters with CheckRtpParametersValues().
+webrtc::RTCError CheckRtpParametersInvalidModificationAndValues(
+    const webrtc::RtpParameters& old_parameters,
+    const webrtc::RtpParameters& new_parameters,
+    rtc::ArrayView<cricket::Codec> codec_preferences,
+    absl::optional<cricket::Codec> send_codec);
+
+// Checks that the immutable values have not changed in new_parameters and
+// checks parameters (except SVC) with CheckRtpParametersValues(). It should
+// usually be paired with a call to CheckScalabilityModeValues().
+webrtc::RTCError CheckRtpParametersInvalidModificationAndValues(
+    const webrtc::RtpParameters& old_parameters,
+    const webrtc::RtpParameters& new_parameters);
+
+struct RtpCapabilities {
+  RtpCapabilities();
+  ~RtpCapabilities();
+  std::vector<webrtc::RtpExtension> header_extensions;
+};
+
+class RtpHeaderExtensionQueryInterface {
+ public:
+  virtual ~RtpHeaderExtensionQueryInterface() = default;
+
+  // Returns a vector of RtpHeaderExtensionCapability, whose direction is
+  // kStopped if the extension is stopped (not used) by default.
+  virtual std::vector<webrtc::RtpHeaderExtensionCapability>
+  GetRtpHeaderExtensions() const = 0;
+};
+
+class VoiceEngineInterface : public RtpHeaderExtensionQueryInterface {
+ public:
+  VoiceEngineInterface() = default;
+  virtual ~VoiceEngineInterface() = default;
+
+  VoiceEngineInterface(const VoiceEngineInterface&) = delete;
+  VoiceEngineInterface& operator=(const VoiceEngineInterface&) = delete;
+
+  // Initialization
+  // Starts the engine.
+  virtual void Init() = 0;
+
+  // TODO(solenberg): Remove once VoE API refactoring is done.
+  virtual rtc::scoped_refptr<webrtc::AudioState> GetAudioState() const = 0;
+
+  virtual std::unique_ptr<VoiceMediaSendChannelInterface> CreateSendChannel(
+      webrtc::Call* call,
+      const MediaConfig& config,
+      const AudioOptions& options,
+      const webrtc::CryptoOptions& crypto_options,
+      webrtc::AudioCodecPairId codec_pair_id) {
+    // TODO(hta): Make pure virtual when all downstream has updated
+    RTC_CHECK_NOTREACHED();
+    return nullptr;
+  }
+
+  virtual std::unique_ptr<VoiceMediaReceiveChannelInterface>
+  CreateReceiveChannel(webrtc::Call* call,
+                       const MediaConfig& config,
+                       const AudioOptions& options,
+                       const webrtc::CryptoOptions& crypto_options,
+                       webrtc::AudioCodecPairId codec_pair_id) {
+    // TODO(hta): Make pure virtual when all downstream has updated
+    RTC_CHECK_NOTREACHED();
+    return nullptr;
+  }
+
+  virtual const std::vector<AudioCodec>& send_codecs() const = 0;
+  virtual const std::vector<AudioCodec>& recv_codecs() const = 0;
+
+  // Starts AEC dump using existing file, a maximum file size in bytes can be
+  // specified. Logging is stopped just before the size limit is exceeded.
+  // If max_size_bytes is set to a value <= 0, no limit will be used.
+  virtual bool StartAecDump(webrtc::FileWrapper file,
+                            int64_t max_size_bytes) = 0;
+
+  // Stops recording AEC dump.
+  virtual void StopAecDump() = 0;
+
+  virtual absl::optional<webrtc::AudioDeviceModule::Stats>
+  GetAudioDeviceStats() = 0;
+};
+
+class VideoEngineInterface : public RtpHeaderExtensionQueryInterface {
+ public:
+  VideoEngineInterface() = default;
+  virtual ~VideoEngineInterface() = default;
+
+  VideoEngineInterface(const VideoEngineInterface&) = delete;
+  VideoEngineInterface& operator=(const VideoEngineInterface&) = delete;
+
+  virtual std::unique_ptr<VideoMediaSendChannelInterface> CreateSendChannel(
+      webrtc::Call* call,
+      const MediaConfig& config,
+      const VideoOptions& options,
+      const webrtc::CryptoOptions& crypto_options,
+      webrtc::VideoBitrateAllocatorFactory* video_bitrate_allocator_factory) {
+    // Default implementation, delete when all is updated
+    RTC_CHECK_NOTREACHED();
+    return nullptr;
+  }
+
+  virtual std::unique_ptr<VideoMediaReceiveChannelInterface>
+  CreateReceiveChannel(webrtc::Call* call,
+                       const MediaConfig& config,
+                       const VideoOptions& options,
+                       const webrtc::CryptoOptions& crypto_options) {
+    // Default implementation, delete when all is updated
+    RTC_CHECK_NOTREACHED();
+    return nullptr;
+  }
+
+  // Retrieve list of supported codecs.
+  virtual std::vector<VideoCodec> send_codecs() const = 0;
+  virtual std::vector<VideoCodec> recv_codecs() const = 0;
+  // As above, but if include_rtx is false, don't include RTX codecs.
+  // TODO(bugs.webrtc.org/13931): Remove default implementation once
+  // upstream subclasses have converted.
+  virtual std::vector<VideoCodec> send_codecs(bool include_rtx) const {
+    RTC_DCHECK(include_rtx);
+    return send_codecs();
+  }
+  virtual std::vector<VideoCodec> recv_codecs(bool include_rtx) const {
+    RTC_DCHECK(include_rtx);
+    return recv_codecs();
+  }
+};
+
+// MediaEngineInterface is an abstraction of a media engine which can be
+// subclassed to support different media componentry backends.
+// It supports voice and video operations in the same class to facilitate
+// proper synchronization between both media types.
+class MediaEngineInterface {
+ public:
+  virtual ~MediaEngineInterface() {}
+
+  // Initialization. Needs to be called on the worker thread.
+  virtual bool Init() = 0;
+
+  virtual VoiceEngineInterface& voice() = 0;
+  virtual VideoEngineInterface& video() = 0;
+  virtual const VoiceEngineInterface& voice() const = 0;
+  virtual const VideoEngineInterface& video() const = 0;
+};
+
+// CompositeMediaEngine constructs a MediaEngine from separate
+// voice and video engine classes.
+// Optionally owns a FieldTrialsView trials map.
+class CompositeMediaEngine : public MediaEngineInterface {
+ public:
+  CompositeMediaEngine(std::unique_ptr<webrtc::FieldTrialsView> trials,
+                       std::unique_ptr<VoiceEngineInterface> audio_engine,
+                       std::unique_ptr<VideoEngineInterface> video_engine);
+  CompositeMediaEngine(std::unique_ptr<VoiceEngineInterface> audio_engine,
+                       std::unique_ptr<VideoEngineInterface> video_engine);
+  ~CompositeMediaEngine() override;
+
+  // Always succeeds.
+  bool Init() override;
+
+  VoiceEngineInterface& voice() override;
+  VideoEngineInterface& video() override;
+  const VoiceEngineInterface& voice() const override;
+  const VideoEngineInterface& video() const override;
+
+ private:
+  const std::unique_ptr<webrtc::FieldTrialsView> trials_;
+  const std::unique_ptr<VoiceEngineInterface> voice_engine_;
+  const std::unique_ptr<VideoEngineInterface> video_engine_;
+};
+
+webrtc::RtpParameters CreateRtpParametersWithOneEncoding();
+webrtc::RtpParameters CreateRtpParametersWithEncodings(StreamParams sp);
+
+// Returns a vector of RTP extensions as visible from RtpSender/Receiver
+// GetCapabilities(). The returned vector only shows what will definitely be
+// offered by default, i.e. the list of extensions returned from
+// GetRtpHeaderExtensions() that are not kStopped.
+std::vector<webrtc::RtpExtension> GetDefaultEnabledRtpHeaderExtensions(
+    const RtpHeaderExtensionQueryInterface& query_interface);
+
+}  // namespace cricket
+
+#endif  // MEDIA_BASE_MEDIA_ENGINE_H_