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diff --git a/webrtc/modules/audio_coding/codecs/g722/audio_encoder_g722.h b/webrtc/modules/audio_coding/codecs/g722/audio_encoder_g722.h
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index 0000000000..07d767e778
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+++ b/webrtc/modules/audio_coding/codecs/g722/audio_encoder_g722.h
@@ -0,0 +1,73 @@
+/*
+ *  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_G722_AUDIO_ENCODER_G722_H_
+#define WEBRTC_MODULES_AUDIO_CODING_CODECS_G722_AUDIO_ENCODER_G722_H_
+
+#include "webrtc/base/buffer.h"
+#include "webrtc/base/scoped_ptr.h"
+#include "webrtc/modules/audio_coding/codecs/audio_encoder.h"
+#include "webrtc/modules/audio_coding/codecs/g722/g722_interface.h"
+
+namespace webrtc {
+
+struct CodecInst;
+
+class AudioEncoderG722 final : public AudioEncoder {
+ public:
+  struct Config {
+    bool IsOk() const;
+
+    int payload_type = 9;
+    int frame_size_ms = 20;
+    size_t num_channels = 1;
+  };
+
+  explicit AudioEncoderG722(const Config& config);
+  explicit AudioEncoderG722(const CodecInst& codec_inst);
+  ~AudioEncoderG722() override;
+
+  size_t MaxEncodedBytes() const override;
+  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;
+  EncodedInfo EncodeInternal(uint32_t rtp_timestamp,
+                             rtc::ArrayView<const int16_t> audio,
+                             size_t max_encoded_bytes,
+                             uint8_t* encoded) override;
+  void Reset() override;
+
+ private:
+  // The encoder state for one channel.
+  struct EncoderState {
+    G722EncInst* encoder;
+    rtc::scoped_ptr<int16_t[]> speech_buffer;   // Queued up for encoding.
+    rtc::Buffer encoded_buffer;                 // Already encoded.
+    EncoderState();
+    ~EncoderState();
+  };
+
+  size_t SamplesPerChannel() const;
+
+  const size_t num_channels_;
+  const int payload_type_;
+  const size_t num_10ms_frames_per_packet_;
+  size_t num_10ms_frames_buffered_;
+  uint32_t first_timestamp_in_buffer_;
+  const rtc::scoped_ptr<EncoderState[]> encoders_;
+  rtc::Buffer interleave_buffer_;
+  RTC_DISALLOW_COPY_AND_ASSIGN(AudioEncoderG722);
+};
+
+}  // namespace webrtc
+#endif  // WEBRTC_MODULES_AUDIO_CODING_CODECS_G722_AUDIO_ENCODER_G722_H_