audio_coding: remove "main" directory

This is the last piece of the old directory layout of the modules.

Duplicated header files are left in audio_coding/main/include until
downstream code is updated to the new location. They have pragma
warnings added to them and identical header guards as the new headers to avoid breaking things.

BUG=webrtc:5095
TESTED=Passing compile-trybots with --clobber flag:
git cl try --clobber --bot=win_compile_rel --bot=linux_compile_rel --bot=android_compile_rel --bot=mac_compile_rel --bot=ios_rel --bot=linux_gn_rel --bot=win_x64_gn_rel --bot=mac_x64_gn_rel --bot=android_gn_rel -m tryserver.webrtc
NOTRY=True
NOPRESUBMIT=True

Review URL: https://codereview.webrtc.org/1481493004

Cr-Commit-Position: refs/heads/master@{#10803}

webrtc/modules/audio_coding/test/RTPFile.h

@@ -0,0 +1,126 @@
+/*
+ *  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_RTPFILE_H_
+#define WEBRTC_MODULES_AUDIO_CODING_TEST_RTPFILE_H_
+
+#include <stdio.h>
+#include <queue>
+
+#include "webrtc/modules/audio_coding/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_TEST_RTPFILE_H_