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

2011-05-30 11:22:19 +00:00
parent 01813fe945
commit 77ae29bc81
1153 changed files with 404089 additions and 0 deletions
--- a/modules/video_coding/main/test/rtp_player.h
+++ b/modules/video_coding/main/test/rtp_player.h
@@ -0,0 +1,106 @@
+/*
+ *  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 WEBRTC_MODULES_VIDEO_CODING_TEST_RTP_PLAYER_H_
+#define WEBRTC_MODULES_VIDEO_CODING_TEST_RTP_PLAYER_H_
+
+#include "typedefs.h"
+#include "rtp_rtcp.h"
+#include "list_wrapper.h"
+#include "critical_section_wrapper.h"
+#include "video_coding_defines.h"
+
+#include <stdio.h>
+#include <string>
+
+#define HDR_SIZE 8 // rtpplay packet header size in bytes
+#define FIRSTLINELEN 40
+#define RAND_VEC_LENGTH 4096
+
+struct RawRtpPacket
+{
+public:
+    RawRtpPacket(WebRtc_UWord8* data, WebRtc_UWord16 len);
+    ~RawRtpPacket();
+
+    WebRtc_UWord8* rtpData;
+    WebRtc_UWord16 rtpLen;
+    WebRtc_Word64 resendTimeMs;
+};
+
+class LostPackets : public webrtc::ListWrapper
+{
+public:
+    LostPackets();
+    ~LostPackets();
+
+    WebRtc_UWord32 AddPacket(WebRtc_UWord8* rtpData, WebRtc_UWord16 rtpLen);
+    WebRtc_UWord32 SetResendTime(WebRtc_UWord16 sequenceNumber, WebRtc_Word64 resendTime);
+    WebRtc_UWord32 TotalNumberOfLosses() const { return _lossCount; };
+    WebRtc_UWord32 NumberOfPacketsToResend() const;
+    void ResentPacket(WebRtc_UWord16 seqNo);
+    void Lock()     {_critSect.Enter();};
+    void Unlock()   {_critSect.Leave();};
+private:
+    webrtc::CriticalSectionWrapper& _critSect;
+    WebRtc_UWord32 _lossCount;
+    FILE*        _debugFile;
+};
+
+struct PayloadCodecTuple
+{
+    PayloadCodecTuple(WebRtc_UWord8 plType, std::string codecName, webrtc::VideoCodecType type) :
+        name(codecName), payloadType(plType), codecType(type) {};
+    const std::string name;
+    const WebRtc_UWord8 payloadType;
+    const webrtc::VideoCodecType codecType;
+};
+
+class RTPPlayer : public webrtc::VCMPacketRequestCallback
+{
+public:
+    RTPPlayer(const char* filename, webrtc::RtpData* callback);
+    virtual ~RTPPlayer();
+
+    WebRtc_Word32 Initialize(const webrtc::ListWrapper& payloadList);
+    WebRtc_Word32 NextPacket(const WebRtc_Word64 timeNow);
+    WebRtc_UWord32 TimeUntilNextPacket() const;
+    WebRtc_Word32 SimulatePacketLoss(float lossRate, bool enableNack = false, WebRtc_UWord32 rttMs = 0);
+    WebRtc_Word32 SetReordering(bool enabled);
+    WebRtc_Word32 ResendPackets(const WebRtc_UWord16* sequenceNumbers, WebRtc_UWord16 length);
+    void Print() const;
+
+private:
+    WebRtc_Word32 SendPacket(WebRtc_UWord8* rtpData, WebRtc_UWord16 rtpLen);
+    WebRtc_Word32 ReadPacket(WebRtc_Word16* rtpdata, WebRtc_UWord32* offset);
+    WebRtc_Word32 ReadHeader();
+    FILE*              _rtpFile;
+    webrtc::RtpRtcp&   _rtpModule;
+    WebRtc_UWord32     _nextRtpTime;
+    webrtc::RtpData*   _dataCallback;
+    bool               _firstPacket;
+    float              _lossRate;
+    bool               _nackEnabled;
+    LostPackets        _lostPackets;
+    WebRtc_UWord32     _resendPacketCount;
+    WebRtc_Word32      _noLossStartup;
+    bool               _endOfFile;
+    WebRtc_UWord32     _rttMs;
+    WebRtc_Word64      _firstPacketRtpTime;
+    WebRtc_Word64      _firstPacketTimeMs;
+    RawRtpPacket*      _reorderBuffer;
+    bool               _reordering;
+    WebRtc_Word16      _nextPacket[8000];
+    WebRtc_Word32      _nextPacketLength;
+    int                _randVec[RAND_VEC_LENGTH];
+    int                _randVecPos;
+};
+
+#endif // WEBRTC_MODULES_VIDEO_CODING_TEST_RTP_PLAYER_H_