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diff --git a/webrtc/modules/audio_coding/test/opus_test.h b/webrtc/modules/audio_coding/test/opus_test.h
new file mode 100644
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+++ b/webrtc/modules/audio_coding/test/opus_test.h
@@ -0,0 +1,60 @@
+/*
+ *  Copyright (c) 2012 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_TEST_OPUS_TEST_H_
+#define WEBRTC_MODULES_AUDIO_CODING_TEST_OPUS_TEST_H_
+
+#include <math.h>
+
+#include "webrtc/base/scoped_ptr.h"
+#include "webrtc/modules/audio_coding/codecs/opus/opus_interface.h"
+#include "webrtc/modules/audio_coding/acm2/acm_resampler.h"
+#include "webrtc/modules/audio_coding/test/ACMTest.h"
+#include "webrtc/modules/audio_coding/test/Channel.h"
+#include "webrtc/modules/audio_coding/test/PCMFile.h"
+#include "webrtc/modules/audio_coding/test/TestStereo.h"
+
+namespace webrtc {
+
+class OpusTest : public ACMTest {
+ public:
+  OpusTest();
+  ~OpusTest();
+
+  void Perform();
+
+ private:
+  void Run(TestPackStereo* channel,
+           size_t channels,
+           int bitrate,
+           size_t frame_length,
+           int percent_loss = 0);
+
+  void OpenOutFile(int test_number);
+
+  rtc::scoped_ptr<AudioCodingModule> acm_receiver_;
+  TestPackStereo* channel_a2b_;
+  PCMFile in_file_stereo_;
+  PCMFile in_file_mono_;
+  PCMFile out_file_;
+  PCMFile out_file_standalone_;
+  int counter_;
+  uint8_t payload_type_;
+  uint32_t rtp_timestamp_;
+  acm2::ACMResampler resampler_;
+  WebRtcOpusEncInst* opus_mono_encoder_;
+  WebRtcOpusEncInst* opus_stereo_encoder_;
+  WebRtcOpusDecInst* opus_mono_decoder_;
+  WebRtcOpusDecInst* opus_stereo_decoder_;
+};
+
+}  // namespace webrtc
+
+#endif  // WEBRTC_MODULES_AUDIO_CODING_TEST_OPUS_TEST_H_