git-svn-id: http://webrtc.googlecode.com/svn/trunk@156 4adac7df-926f-26a2-2b94-8c16560cd09d
diff --git a/src/modules/audio_coding/main/test/RTPFile.h b/src/modules/audio_coding/main/test/RTPFile.h
new file mode 100644
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+++ b/src/modules/audio_coding/main/test/RTPFile.h
@@ -0,0 +1,99 @@
+/*
+ * Copyright (c) 2011 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>
+
+using 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;
+ virtual WebRtc_UWord16 Read(WebRtcRTPHeader* rtpInfo,
+ WebRtc_Word8* 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_Word8* 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(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_Word8* payloadData,
+ WebRtc_UWord16 payloadSize,
+ WebRtc_UWord32* offset);
+ bool EndOfFile() const { return _rtpEOF; }
+private:
+ FILE* _rtpFile;
+ bool _rtpEOF;
+};
+
+#endif