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diff --git a/third_party/libwebrtc/modules/audio_coding/codecs/opus/audio_encoder_opus.h b/third_party/libwebrtc/modules/audio_coding/codecs/opus/audio_encoder_opus.h
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index 0000000000..8c5c235016
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+++ b/third_party/libwebrtc/modules/audio_coding/codecs/opus/audio_encoder_opus.h
@@ -0,0 +1,184 @@
+/*
+ *  Copyright (c) 2014 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_CODING_CODECS_OPUS_AUDIO_ENCODER_OPUS_H_
+#define MODULES_AUDIO_CODING_CODECS_OPUS_AUDIO_ENCODER_OPUS_H_
+
+#include <functional>
+#include <memory>
+#include <string>
+#include <vector>
+
+#include "absl/strings/string_view.h"
+#include "absl/types/optional.h"
+#include "api/audio_codecs/audio_encoder.h"
+#include "api/audio_codecs/audio_format.h"
+#include "api/audio_codecs/opus/audio_encoder_opus_config.h"
+#include "common_audio/smoothing_filter.h"
+#include "modules/audio_coding/audio_network_adaptor/include/audio_network_adaptor.h"
+#include "modules/audio_coding/codecs/opus/opus_interface.h"
+
+namespace webrtc {
+
+class RtcEventLog;
+
+class AudioEncoderOpusImpl final : public AudioEncoder {
+ public:
+  // Returns empty if the current bitrate falls within the hysteresis window,
+  // defined by complexity_threshold_bps +/- complexity_threshold_window_bps.
+  // Otherwise, returns the current complexity depending on whether the
+  // current bitrate is above or below complexity_threshold_bps.
+  static absl::optional<int> GetNewComplexity(
+      const AudioEncoderOpusConfig& config);
+
+  // Returns OPUS_AUTO if the the current bitrate is above wideband threshold.
+  // Returns empty if it is below, but bandwidth coincides with the desired one.
+  // Otherwise returns the desired bandwidth.
+  static absl::optional<int> GetNewBandwidth(
+      const AudioEncoderOpusConfig& config,
+      OpusEncInst* inst);
+
+  using AudioNetworkAdaptorCreator =
+      std::function<std::unique_ptr<AudioNetworkAdaptor>(absl::string_view,
+                                                         RtcEventLog*)>;
+
+  AudioEncoderOpusImpl(const AudioEncoderOpusConfig& config, int payload_type);
+
+  // Dependency injection for testing.
+  AudioEncoderOpusImpl(
+      const AudioEncoderOpusConfig& config,
+      int payload_type,
+      const AudioNetworkAdaptorCreator& audio_network_adaptor_creator,
+      std::unique_ptr<SmoothingFilter> bitrate_smoother);
+
+  AudioEncoderOpusImpl(int payload_type, const SdpAudioFormat& format);
+  ~AudioEncoderOpusImpl() override;
+
+  AudioEncoderOpusImpl(const AudioEncoderOpusImpl&) = delete;
+  AudioEncoderOpusImpl& operator=(const AudioEncoderOpusImpl&) = delete;
+
+  int SampleRateHz() const override;
+  size_t NumChannels() const override;
+  int RtpTimestampRateHz() const override;
+  size_t Num10MsFramesInNextPacket() const override;
+  size_t Max10MsFramesInAPacket() const override;
+  int GetTargetBitrate() const override;
+
+  void Reset() override;
+  bool SetFec(bool enable) override;
+
+  // Set Opus DTX. Once enabled, Opus stops transmission, when it detects
+  // voice being inactive. During that, it still sends 2 packets (one for
+  // content, one for signaling) about every 400 ms.
+  bool SetDtx(bool enable) override;
+  bool GetDtx() const override;
+
+  bool SetApplication(Application application) override;
+  void SetMaxPlaybackRate(int frequency_hz) override;
+  bool EnableAudioNetworkAdaptor(const std::string& config_string,
+                                 RtcEventLog* event_log) override;
+  void DisableAudioNetworkAdaptor() override;
+  void OnReceivedUplinkPacketLossFraction(
+      float uplink_packet_loss_fraction) override;
+  void OnReceivedTargetAudioBitrate(int target_audio_bitrate_bps) override;
+  void OnReceivedUplinkBandwidth(
+      int target_audio_bitrate_bps,
+      absl::optional<int64_t> bwe_period_ms) override;
+  void OnReceivedUplinkAllocation(BitrateAllocationUpdate update) override;
+  void OnReceivedRtt(int rtt_ms) override;
+  void OnReceivedOverhead(size_t overhead_bytes_per_packet) override;
+  void SetReceiverFrameLengthRange(int min_frame_length_ms,
+                                   int max_frame_length_ms) override;
+  ANAStats GetANAStats() const override;
+  absl::optional<std::pair<TimeDelta, TimeDelta> > GetFrameLengthRange()
+      const override;
+  rtc::ArrayView<const int> supported_frame_lengths_ms() const {
+    return config_.supported_frame_lengths_ms;
+  }
+
+  // Getters for testing.
+  float packet_loss_rate() const { return packet_loss_rate_; }
+  AudioEncoderOpusConfig::ApplicationMode application() const {
+    return config_.application;
+  }
+  bool fec_enabled() const { return config_.fec_enabled; }
+  size_t num_channels_to_encode() const { return num_channels_to_encode_; }
+  int next_frame_length_ms() const { return next_frame_length_ms_; }
+
+ protected:
+  EncodedInfo EncodeImpl(uint32_t rtp_timestamp,
+                         rtc::ArrayView<const int16_t> audio,
+                         rtc::Buffer* encoded) override;
+
+ private:
+  class PacketLossFractionSmoother;
+
+  static absl::optional<AudioEncoderOpusConfig> SdpToConfig(
+      const SdpAudioFormat& format);
+  static void AppendSupportedEncoders(std::vector<AudioCodecSpec>* specs);
+  static AudioCodecInfo QueryAudioEncoder(const AudioEncoderOpusConfig& config);
+  static std::unique_ptr<AudioEncoder> MakeAudioEncoder(
+      const AudioEncoderOpusConfig&,
+      int payload_type);
+
+  size_t Num10msFramesPerPacket() const;
+  size_t SamplesPer10msFrame() const;
+  size_t SufficientOutputBufferSize() const;
+  bool RecreateEncoderInstance(const AudioEncoderOpusConfig& config);
+  void SetFrameLength(int frame_length_ms);
+  void SetNumChannelsToEncode(size_t num_channels_to_encode);
+  void SetProjectedPacketLossRate(float fraction);
+
+  void OnReceivedUplinkBandwidth(
+      int target_audio_bitrate_bps,
+      absl::optional<int64_t> bwe_period_ms,
+      absl::optional<int64_t> link_capacity_allocation);
+
+  // TODO(minyue): remove "override" when we can deprecate
+  // `AudioEncoder::SetTargetBitrate`.
+  void SetTargetBitrate(int target_bps) override;
+
+  void ApplyAudioNetworkAdaptor();
+  std::unique_ptr<AudioNetworkAdaptor> DefaultAudioNetworkAdaptorCreator(
+      absl::string_view config_string,
+      RtcEventLog* event_log) const;
+
+  void MaybeUpdateUplinkBandwidth();
+
+  AudioEncoderOpusConfig config_;
+  const int payload_type_;
+  const bool use_stable_target_for_adaptation_;
+  const bool adjust_bandwidth_;
+  bool bitrate_changed_;
+  // A multiplier for bitrates at 5 kbps and higher. The target bitrate
+  // will be multiplied by these multipliers, each multiplier is applied to a
+  // 1 kbps range.
+  std::vector<float> bitrate_multipliers_;
+  float packet_loss_rate_;
+  std::vector<int16_t> input_buffer_;
+  OpusEncInst* inst_;
+  uint32_t first_timestamp_in_buffer_;
+  size_t num_channels_to_encode_;
+  int next_frame_length_ms_;
+  int complexity_;
+  std::unique_ptr<PacketLossFractionSmoother> packet_loss_fraction_smoother_;
+  const AudioNetworkAdaptorCreator audio_network_adaptor_creator_;
+  std::unique_ptr<AudioNetworkAdaptor> audio_network_adaptor_;
+  absl::optional<size_t> overhead_bytes_per_packet_;
+  const std::unique_ptr<SmoothingFilter> bitrate_smoother_;
+  absl::optional<int64_t> bitrate_smoother_last_update_time_;
+  int consecutive_dtx_frames_;
+
+  friend struct AudioEncoderOpus;
+};
+
+}  // namespace webrtc
+
+#endif  // MODULES_AUDIO_CODING_CODECS_OPUS_AUDIO_ENCODER_OPUS_H_