audio_coding: remove "main" directory

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

Index: webrtc/modules/audio_coding/main/test/EncodeDecodeTest.h
diff --git a/webrtc/modules/audio_coding/main/test/EncodeDecodeTest.h b/webrtc/modules/audio_coding/main/test/EncodeDecodeTest.h
deleted file mode 100644
index 4ad92cec154ba429da316b3528a42647f64e92e5..0000000000000000000000000000000000000000
--- a/webrtc/modules/audio_coding/main/test/EncodeDecodeTest.h
+++ /dev/null
@@ -1,123 +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_ENCODEDECODETEST_H_
-#define WEBRTC_MODULES_AUDIO_CODING_MAIN_TEST_ENCODEDECODETEST_H_
-
-#include <stdio.h>
-#include <string.h>
-
-#include "webrtc/modules/audio_coding/main/include/audio_coding_module.h"
-#include "webrtc/modules/audio_coding/main/test/ACMTest.h"
-#include "webrtc/modules/audio_coding/main/test/PCMFile.h"
-#include "webrtc/modules/audio_coding/main/test/RTPFile.h"
-#include "webrtc/typedefs.h"
-
-namespace webrtc {
-
-#define MAX_INCOMING_PAYLOAD 8096
-
-// TestPacketization callback which writes the encoded payloads to file
-class TestPacketization : public AudioPacketizationCallback {
- public:
-  TestPacketization(RTPStream *rtpStream, uint16_t frequency);
-  ~TestPacketization();
-  int32_t SendData(const FrameType frameType,
-                   const uint8_t payloadType,
-                   const uint32_t timeStamp,
-                   const uint8_t* payloadData,
-                   const size_t payloadSize,
-                   const RTPFragmentationHeader* fragmentation) override;
-
- private:
-  static void MakeRTPheader(uint8_t* rtpHeader, uint8_t payloadType,
-                            int16_t seqNo, uint32_t timeStamp, uint32_t ssrc);
-  RTPStream* _rtpStream;
-  int32_t _frequency;
-  int16_t _seqNo;
-};
-
-class Sender {
- public:
-  Sender();
-  void Setup(AudioCodingModule *acm, RTPStream *rtpStream,
-             std::string in_file_name, int sample_rate, int channels);
-  void Teardown();
-  void Run();
-  bool Add10MsData();
-
-  //for auto_test and logging
-  uint8_t testMode;
-  uint8_t codeId;
-
- protected:
-  AudioCodingModule* _acm;
-
- private:
-  PCMFile _pcmFile;
-  AudioFrame _audioFrame;
-  TestPacketization* _packetization;
-};
-
-class Receiver {
- public:
-  Receiver();
-  virtual ~Receiver() {};
-  void Setup(AudioCodingModule *acm, RTPStream *rtpStream,
-             std::string out_file_name, int channels);
-  void Teardown();
-  void Run();
-  virtual bool IncomingPacket();
-  bool PlayoutData();
-
-  //for auto_test and logging
-  uint8_t codeId;
-  uint8_t testMode;
-
- private:
-  PCMFile _pcmFile;
-  int16_t* _playoutBuffer;
-  uint16_t _playoutLengthSmpls;
-  int32_t _frequency;
-  bool _firstTime;
-
- protected:
-  AudioCodingModule* _acm;
-  uint8_t _incomingPayload[MAX_INCOMING_PAYLOAD];
-  RTPStream* _rtpStream;
-  WebRtcRTPHeader _rtpInfo;
-  size_t _realPayloadSizeBytes;
-  size_t _payloadSizeBytes;
-  uint32_t _nextTime;
-};
-
-class EncodeDecodeTest : public ACMTest {
- public:
-  EncodeDecodeTest();
-  explicit EncodeDecodeTest(int testMode);
-  void Perform() override;
-
-  uint16_t _playoutFreq;
-  uint8_t _testMode;
-
- private:
-  std::string EncodeToFile(int fileType,
-                           int codeId,
-                           int* codePars,
-                           int testMode);
-
- protected:
-  Sender _sender;
-  Receiver _receiver;
-};
-
-}  // namespace webrtc
-
-#endif  // WEBRTC_MODULES_AUDIO_CODING_MAIN_TEST_ENCODEDECODETEST_H_