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diff --git a/webrtc/modules/audio_coding/codecs/opus/audio_encoder_opus.h b/webrtc/modules/audio_coding/codecs/opus/audio_encoder_opus.h
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+++ b/webrtc/modules/audio_coding/codecs/opus/audio_encoder_opus.h
@@ -0,0 +1,102 @@
+/*
+ *  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 WEBRTC_MODULES_AUDIO_CODING_CODECS_OPUS_AUDIO_ENCODER_OPUS_H_
+#define WEBRTC_MODULES_AUDIO_CODING_CODECS_OPUS_AUDIO_ENCODER_OPUS_H_
+
+#include <vector>
+
+#include "webrtc/base/constructormagic.h"
+#include "webrtc/modules/audio_coding/codecs/opus/opus_interface.h"
+#include "webrtc/modules/audio_coding/codecs/audio_encoder.h"
+
+namespace webrtc {
+
+struct CodecInst;
+
+class AudioEncoderOpus final : public AudioEncoder {
+ public:
+  enum ApplicationMode {
+    kVoip = 0,
+    kAudio = 1,
+  };
+
+  struct Config {
+    bool IsOk() const;
+    int frame_size_ms = 20;
+    size_t num_channels = 1;
+    int payload_type = 120;
+    ApplicationMode application = kVoip;
+    int bitrate_bps = 64000;
+    bool fec_enabled = false;
+    int max_playback_rate_hz = 48000;
+    int complexity = kDefaultComplexity;
+    bool dtx_enabled = false;
+
+   private:
+#if defined(WEBRTC_ANDROID) || defined(WEBRTC_IOS) || defined(WEBRTC_ARCH_ARM)
+    // If we are on Android, iOS and/or ARM, use a lower complexity setting as
+    // default, to save encoder complexity.
+    static const int kDefaultComplexity = 5;
+#else
+    static const int kDefaultComplexity = 9;
+#endif
+  };
+
+  explicit AudioEncoderOpus(const Config& config);
+  explicit AudioEncoderOpus(const CodecInst& codec_inst);
+  ~AudioEncoderOpus() override;
+
+  size_t MaxEncodedBytes() const override;
+  int SampleRateHz() const override;
+  size_t NumChannels() const override;
+  size_t Num10MsFramesInNextPacket() const override;
+  size_t Max10MsFramesInAPacket() const override;
+  int GetTargetBitrate() const override;
+
+  EncodedInfo EncodeInternal(uint32_t rtp_timestamp,
+                             rtc::ArrayView<const int16_t> audio,
+                             size_t max_encoded_bytes,
+                             uint8_t* encoded) 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 SetApplication(Application application) override;
+  void SetMaxPlaybackRate(int frequency_hz) override;
+  void SetProjectedPacketLossRate(double fraction) override;
+  void SetTargetBitrate(int target_bps) override;
+
+  // Getters for testing.
+  double packet_loss_rate() const { return packet_loss_rate_; }
+  ApplicationMode application() const { return config_.application; }
+  bool dtx_enabled() const { return config_.dtx_enabled; }
+
+ private:
+  size_t Num10msFramesPerPacket() const;
+  size_t SamplesPer10msFrame() const;
+  bool RecreateEncoderInstance(const Config& config);
+
+  Config config_;
+  double packet_loss_rate_;
+  std::vector<int16_t> input_buffer_;
+  OpusEncInst* inst_;
+  uint32_t first_timestamp_in_buffer_;
+  RTC_DISALLOW_COPY_AND_ASSIGN(AudioEncoderOpus);
+};
+
+}  // namespace webrtc
+
+#endif  // WEBRTC_MODULES_AUDIO_CODING_CODECS_OPUS_AUDIO_ENCODER_OPUS_H_