audio_coding: remove "main" directory

webrtc/modules/audio_coding/main/test/opus_test.h

Index: webrtc/modules/audio_coding/main/test/opus_test.h
diff --git a/webrtc/modules/audio_coding/main/test/opus_test.h b/webrtc/modules/audio_coding/main/test/opus_test.h
deleted file mode 100644
index 0b960092413d67fdebb90170b4f81b19a2c2929a..0000000000000000000000000000000000000000
--- a/webrtc/modules/audio_coding/main/test/opus_test.h
+++ /dev/null
@@ -1,57 +0,0 @@
-/*
- *  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_MAIN_TEST_OPUS_TEST_H_
-#define WEBRTC_MODULES_AUDIO_CODING_MAIN_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/main/acm2/acm_resampler.h"
-#include "webrtc/modules/audio_coding/main/test/ACMTest.h"
-#include "webrtc/modules/audio_coding/main/test/Channel.h"
-#include "webrtc/modules/audio_coding/main/test/PCMFile.h"
-#include "webrtc/modules/audio_coding/main/test/TestStereo.h"
-
-namespace webrtc {
-
-class OpusTest : public ACMTest {
- public:
-  OpusTest();
-  ~OpusTest();
-
-  void Perform();
-
- private:
-  void Run(TestPackStereo* channel, int channels, int bitrate, int 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_;
-  int 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_MAIN_TEST_OPUS_TEST_H_