diff --git a/modules/audio_coding/codecs/opus/audio_decoder_opus.cc b/modules/audio_coding/codecs/opus/audio_decoder_opus.cc
new file mode 100644
index 0000000..e78fc04
--- /dev/null
+++ b/modules/audio_coding/codecs/opus/audio_decoder_opus.cc
@@ -0,0 +1,94 @@
+/*
+ *  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.
+ */
+
+#include "webrtc/modules/audio_coding/codecs/opus/interface/audio_decoder_opus.h"
+
+#include "webrtc/base/checks.h"
+
+namespace webrtc {
+
+AudioDecoderOpus::AudioDecoderOpus(size_t num_channels)
+    : channels_(num_channels) {
+  DCHECK(num_channels == 1 || num_channels == 2);
+  WebRtcOpus_DecoderCreate(&dec_state_, static_cast<int>(channels_));
+  WebRtcOpus_DecoderInit(dec_state_);
+}
+
+AudioDecoderOpus::~AudioDecoderOpus() {
+  WebRtcOpus_DecoderFree(dec_state_);
+}
+
+int AudioDecoderOpus::DecodeInternal(const uint8_t* encoded,
+                                     size_t encoded_len,
+                                     int sample_rate_hz,
+                                     int16_t* decoded,
+                                     SpeechType* speech_type) {
+  DCHECK_EQ(sample_rate_hz, 48000);
+  int16_t temp_type = 1;  // Default is speech.
+  int ret =
+      WebRtcOpus_Decode(dec_state_, encoded, encoded_len, decoded, &temp_type);
+  if (ret > 0)
+    ret *= static_cast<int>(channels_);  // Return total number of samples.
+  *speech_type = ConvertSpeechType(temp_type);
+  return ret;
+}
+
+int AudioDecoderOpus::DecodeRedundantInternal(const uint8_t* encoded,
+                                              size_t encoded_len,
+                                              int sample_rate_hz,
+                                              int16_t* decoded,
+                                              SpeechType* speech_type) {
+  if (!PacketHasFec(encoded, encoded_len)) {
+    // This packet is a RED packet.
+    return DecodeInternal(encoded, encoded_len, sample_rate_hz, decoded,
+                          speech_type);
+  }
+
+  DCHECK_EQ(sample_rate_hz, 48000);
+  int16_t temp_type = 1;  // Default is speech.
+  int ret = WebRtcOpus_DecodeFec(dec_state_, encoded, encoded_len, decoded,
+                                 &temp_type);
+  if (ret > 0)
+    ret *= static_cast<int>(channels_);  // Return total number of samples.
+  *speech_type = ConvertSpeechType(temp_type);
+  return ret;
+}
+
+void AudioDecoderOpus::Reset() {
+  WebRtcOpus_DecoderInit(dec_state_);
+}
+
+int AudioDecoderOpus::PacketDuration(const uint8_t* encoded,
+                                     size_t encoded_len) const {
+  return WebRtcOpus_DurationEst(dec_state_, encoded, encoded_len);
+}
+
+int AudioDecoderOpus::PacketDurationRedundant(const uint8_t* encoded,
+                                              size_t encoded_len) const {
+  if (!PacketHasFec(encoded, encoded_len)) {
+    // This packet is a RED packet.
+    return PacketDuration(encoded, encoded_len);
+  }
+
+  return WebRtcOpus_FecDurationEst(encoded, encoded_len);
+}
+
+bool AudioDecoderOpus::PacketHasFec(const uint8_t* encoded,
+                                    size_t encoded_len) const {
+  int fec;
+  fec = WebRtcOpus_PacketHasFec(encoded, encoded_len);
+  return (fec == 1);
+}
+
+size_t AudioDecoderOpus::Channels() const {
+  return channels_;
+}
+
+}  // namespace webrtc
diff --git a/modules/audio_coding/codecs/opus/interface/audio_decoder_opus.h b/modules/audio_coding/codecs/opus/interface/audio_decoder_opus.h
new file mode 100644
index 0000000..9fa77b0
--- /dev/null
+++ b/modules/audio_coding/codecs/opus/interface/audio_decoder_opus.h
@@ -0,0 +1,51 @@
+/*
+ *  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_OPUS_INTERFACE_AUDIO_DECODER_OPUS_H
+#define WEBRTC_MODULES_AUDIO_CODING_CODECS_OPUS_INTERFACE_AUDIO_DECODER_OPUS_H
+
+#include "webrtc/modules/audio_coding/codecs/audio_decoder.h"
+#include "webrtc/modules/audio_coding/codecs/opus/interface/opus_interface.h"
+
+namespace webrtc {
+
+class AudioDecoderOpus : public AudioDecoder {
+ public:
+  explicit AudioDecoderOpus(size_t num_channels);
+  ~AudioDecoderOpus() override;
+
+  void Reset() override;
+  int PacketDuration(const uint8_t* encoded, size_t encoded_len) const override;
+  int PacketDurationRedundant(const uint8_t* encoded,
+                              size_t encoded_len) const override;
+  bool PacketHasFec(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;
+  int DecodeRedundantInternal(const uint8_t* encoded,
+                              size_t encoded_len,
+                              int sample_rate_hz,
+                              int16_t* decoded,
+                              SpeechType* speech_type) override;
+
+ private:
+  OpusDecInst* dec_state_;
+  const size_t channels_;
+  DISALLOW_COPY_AND_ASSIGN(AudioDecoderOpus);
+};
+
+}  // namespace webrtc
+
+#endif  // WEBRTC_MODULES_AUDIO_CODING_CODECS_OPUS_INTERFACE_AUDIO_DECODER_OPUS_H
diff --git a/modules/audio_coding/codecs/opus/opus.gypi b/modules/audio_coding/codecs/opus/opus.gypi
index 4ae4340..5a420b4 100644
--- a/modules/audio_coding/codecs/opus/opus.gypi
+++ b/modules/audio_coding/codecs/opus/opus.gypi
@@ -43,7 +43,9 @@
         '<(webrtc_root)',
       ],
       'sources': [
+        'audio_decoder_opus.cc',
         'audio_encoder_opus.cc',
+        'interface/audio_decoder_opus.h',
         'interface/audio_encoder_opus.h',
         'interface/opus_interface.h',
         'opus_inst.h',