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diff --git a/webrtc/modules/audio_coding/codecs/g722/audio_decoder_g722.h b/webrtc/modules/audio_coding/codecs/g722/audio_decoder_g722.h
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index 0000000000..7cc2ea9877
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+++ b/webrtc/modules/audio_coding/codecs/g722/audio_decoder_g722.h
@@ -0,0 +1,72 @@
+/*
+ *  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 WEBRTC_MODULES_AUDIO_CODING_CODECS_G722_AUDIO_DECODER_G722_H_
+#define WEBRTC_MODULES_AUDIO_CODING_CODECS_G722_AUDIO_DECODER_G722_H_
+
+#include "webrtc/modules/audio_coding/codecs/audio_decoder.h"
+
+typedef struct WebRtcG722DecInst G722DecInst;
+
+namespace webrtc {
+
+class AudioDecoderG722 final : public AudioDecoder {
+ public:
+  AudioDecoderG722();
+  ~AudioDecoderG722() override;
+  bool HasDecodePlc() const override;
+  void Reset() override;
+  int PacketDuration(const uint8_t* encoded, size_t encoded_len) const override;
+  size_t Channels() const override;
+
+ protected:
+  int DecodeInternal(const uint8_t* encoded,
+                     size_t encoded_len,
+                     int sample_rate_hz,
+                     int16_t* decoded,
+                     SpeechType* speech_type) override;
+
+ private:
+  G722DecInst* dec_state_;
+  RTC_DISALLOW_COPY_AND_ASSIGN(AudioDecoderG722);
+};
+
+class AudioDecoderG722Stereo final : public AudioDecoder {
+ public:
+  AudioDecoderG722Stereo();
+  ~AudioDecoderG722Stereo() override;
+  void Reset() override;
+
+ protected:
+  int DecodeInternal(const uint8_t* encoded,
+                     size_t encoded_len,
+                     int sample_rate_hz,
+                     int16_t* decoded,
+                     SpeechType* speech_type) override;
+  size_t Channels() const override;
+
+ private:
+  // Splits the stereo-interleaved payload in |encoded| into separate payloads
+  // for left and right channels. The separated payloads are written to
+  // |encoded_deinterleaved|, which must hold at least |encoded_len| samples.
+  // The left channel starts at offset 0, while the right channel starts at
+  // offset encoded_len / 2 into |encoded_deinterleaved|.
+  void SplitStereoPacket(const uint8_t* encoded,
+                         size_t encoded_len,
+                         uint8_t* encoded_deinterleaved);
+
+  G722DecInst* dec_state_left_;
+  G722DecInst* dec_state_right_;
+  RTC_DISALLOW_COPY_AND_ASSIGN(AudioDecoderG722Stereo);
+};
+
+}  // namespace webrtc
+
+#endif  // WEBRTC_MODULES_AUDIO_CODING_CODECS_G722_AUDIO_DECODER_G722_H_