audio_coding: remove "main" directory

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

-/*
- *  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_RTPFILE_H_
-#define WEBRTC_MODULES_AUDIO_CODING_MAIN_TEST_RTPFILE_H_
-
-#include <stdio.h>
-#include <queue>
-
-#include "webrtc/modules/audio_coding/main/include/audio_coding_module.h"
-#include "webrtc/modules/include/module_common_types.h"
-#include "webrtc/system_wrappers/include/rw_lock_wrapper.h"
-#include "webrtc/typedefs.h"
-
-namespace webrtc {
-
-class RTPStream {
- public:
-  virtual ~RTPStream() {
-  }
-
-  virtual void Write(const uint8_t payloadType, const uint32_t timeStamp,
-                     const int16_t seqNo, const uint8_t* payloadData,
-                     const size_t payloadSize, uint32_t frequency) = 0;
-
-  // Returns the packet's payload size. Zero should be treated as an
-  // end-of-stream (in the case that EndOfFile() is true) or an error.
-  virtual size_t Read(WebRtcRTPHeader* rtpInfo, uint8_t* payloadData,
-                      size_t payloadSize, uint32_t* offset) = 0;
-  virtual bool EndOfFile() const = 0;
-
- protected:
-  void MakeRTPheader(uint8_t* rtpHeader, uint8_t payloadType, int16_t seqNo,
-                     uint32_t timeStamp, uint32_t ssrc);
-
-  void ParseRTPHeader(WebRtcRTPHeader* rtpInfo, const uint8_t* rtpHeader);
-};
-
-class RTPPacket {
- public:
-  RTPPacket(uint8_t payloadType, uint32_t timeStamp, int16_t seqNo,
-            const uint8_t* payloadData, size_t payloadSize,
-            uint32_t frequency);
-
-  ~RTPPacket();
-
-  uint8_t payloadType;
-  uint32_t timeStamp;
-  int16_t seqNo;
-  uint8_t* payloadData;
-  size_t payloadSize;
-  uint32_t frequency;
-};
-
-class RTPBuffer : public RTPStream {
- public:
-  RTPBuffer();
-
-  ~RTPBuffer();
-
-  void Write(const uint8_t payloadType,
-             const uint32_t timeStamp,
-             const int16_t seqNo,
-             const uint8_t* payloadData,
-             const size_t payloadSize,
-             uint32_t frequency) override;
-
-  size_t Read(WebRtcRTPHeader* rtpInfo,
-              uint8_t* payloadData,
-              size_t payloadSize,
-              uint32_t* offset) override;
-
-  bool EndOfFile() const override;
-
- private:
-  RWLockWrapper* _queueRWLock;
-  std::queue<RTPPacket *> _rtpQueue;
-};
-
-class RTPFile : public RTPStream {
- public:
-  ~RTPFile() {
-  }
-
-  RTPFile()
-      : _rtpFile(NULL),
-        _rtpEOF(false) {
-  }
-
-  void Open(const char *outFilename, const char *mode);
-
-  void Close();
-
-  void WriteHeader();
-
-  void ReadHeader();
-
-  void Write(const uint8_t payloadType,
-             const uint32_t timeStamp,
-             const int16_t seqNo,
-             const uint8_t* payloadData,
-             const size_t payloadSize,
-             uint32_t frequency) override;
-
-  size_t Read(WebRtcRTPHeader* rtpInfo,
-              uint8_t* payloadData,
-              size_t payloadSize,
-              uint32_t* offset) override;
-
-  bool EndOfFile() const override { return _rtpEOF; }
-
- private:
-  FILE* _rtpFile;
-  bool _rtpEOF;
-};
-
-}  // namespace webrtc
-
-#endif  // WEBRTC_MODULES_AUDIO_CODING_MAIN_TEST_RTPFILE_H_