Move src/ -> webrtc/

TBR=niklas.enbom@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/915006

git-svn-id: http://webrtc.googlecode.com/svn/trunk@2963 4adac7df-926f-26a2-2b94-8c16560cd09d

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

@@ -0,0 +1,103 @@
+/*
+ *  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 RTPFILE_H
+#define RTPFILE_H
+
+#include "audio_coding_module.h"
+#include "module_common_types.h"
+#include "typedefs.h"
+#include "rw_lock_wrapper.h"
+#include <stdio.h>
+#include <queue>
+
+namespace webrtc {
+
+class RTPStream
+{
+public:
+    virtual ~RTPStream(){}
+
+    virtual void Write(const WebRtc_UWord8 payloadType, const WebRtc_UWord32 timeStamp,
+                                     const WebRtc_Word16 seqNo, const WebRtc_UWord8* payloadData,
+                                     const WebRtc_UWord16 payloadSize, WebRtc_UWord32 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 WebRtc_UWord16 Read(WebRtcRTPHeader* rtpInfo,
+                    WebRtc_UWord8* payloadData,
+                    WebRtc_UWord16 payloadSize,
+                    WebRtc_UWord32* offset) = 0;
+    virtual bool EndOfFile() const = 0;
+
+protected:
+    void MakeRTPheader(WebRtc_UWord8* rtpHeader, 
+                                      WebRtc_UWord8 payloadType, WebRtc_Word16 seqNo, 
+                                      WebRtc_UWord32 timeStamp, WebRtc_UWord32 ssrc);
+    void ParseRTPHeader(WebRtcRTPHeader* rtpInfo, const WebRtc_UWord8* rtpHeader);
+};
+
+class RTPPacket
+{
+public:
+    RTPPacket(WebRtc_UWord8 payloadType, WebRtc_UWord32 timeStamp,
+                                     WebRtc_Word16 seqNo, const WebRtc_UWord8* payloadData,
+                                     WebRtc_UWord16 payloadSize, WebRtc_UWord32 frequency);
+    ~RTPPacket();
+    WebRtc_UWord8 payloadType;
+    WebRtc_UWord32 timeStamp;
+    WebRtc_Word16 seqNo;
+    WebRtc_UWord8* payloadData;
+    WebRtc_UWord16 payloadSize;
+    WebRtc_UWord32 frequency;
+};
+
+class RTPBuffer : public RTPStream
+{
+public:
+    RTPBuffer();
+    ~RTPBuffer();
+    void Write(const WebRtc_UWord8 payloadType, const WebRtc_UWord32 timeStamp,
+                                     const WebRtc_Word16 seqNo, const WebRtc_UWord8* payloadData,
+                                     const WebRtc_UWord16 payloadSize, WebRtc_UWord32 frequency);
+    WebRtc_UWord16 Read(WebRtcRTPHeader* rtpInfo,
+                    WebRtc_UWord8* payloadData,
+                    WebRtc_UWord16 payloadSize,
+                    WebRtc_UWord32* offset);
+    virtual bool EndOfFile() const;
+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 WebRtc_UWord8 payloadType, const WebRtc_UWord32 timeStamp,
+                                     const WebRtc_Word16 seqNo, const WebRtc_UWord8* payloadData,
+                                     const WebRtc_UWord16 payloadSize, WebRtc_UWord32 frequency);
+    WebRtc_UWord16 Read(WebRtcRTPHeader* rtpInfo,
+                    WebRtc_UWord8* payloadData,
+                    WebRtc_UWord16 payloadSize,
+                    WebRtc_UWord32* offset);
+    bool EndOfFile() const { return _rtpEOF; }
+private:
+    FILE*   _rtpFile;
+    bool    _rtpEOF;
+};
+
+} // namespace webrtc
+#endif