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

modules/rtp_rtcp/source/rtcp_receiver_help.h

@@ -0,0 +1,122 @@
+/*
+ *  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_RTP_RTCP_SOURCE_RTCP_RECEIVER_HELP_H_
+#define WEBRTC_MODULES_RTP_RTCP_SOURCE_RTCP_RECEIVER_HELP_H_
+
+#include "typedefs.h"
+
+#include "rtp_rtcp_defines.h"   // RTCPReportBlock
+#include "rtp_rtcp_config.h"           // RTCP_MAX_REPORT_BLOCKS
+#include "rtcp_utility.h"
+#include "tmmbr_help.h"
+
+namespace webrtc {
+namespace RTCPHelp
+{
+
+class RTCPPacketInformation
+{
+public:
+    RTCPPacketInformation();
+    ~RTCPPacketInformation();
+
+    void AddVoIPMetric(const RTCPVoIPMetric*  metric);
+
+    void AddApplicationData(const WebRtc_UWord8* data, const WebRtc_UWord16 size);
+
+    void AddNACKPacket(const WebRtc_UWord16 packetID);
+    void ResetNACKPacketIdArray();
+
+    void AddReportInfo(const WebRtc_UWord8 fractionLost,
+                       const WebRtc_UWord16 rtt,
+                       const WebRtc_UWord32 extendedHighSeqNum,
+                       const WebRtc_UWord32 jitter);
+
+    WebRtc_UWord32  rtcpPacketTypeFlags; // RTCPPacketTypeFlags bit field
+    WebRtc_UWord32  remoteSSRC;
+
+    WebRtc_UWord16* nackSequenceNumbers;
+    WebRtc_UWord16  nackSequenceNumbersLength;
+
+    WebRtc_UWord8   applicationSubType;
+    WebRtc_UWord32  applicationName;
+    WebRtc_UWord8*  applicationData;
+    WebRtc_UWord16  applicationLength;
+
+    bool          reportBlock;
+    WebRtc_UWord8   fractionLost;
+    WebRtc_UWord16  roundTripTime;
+    WebRtc_UWord32  lastReceivedExtendedHighSeqNum;
+    WebRtc_UWord32  jitter;
+
+    WebRtc_UWord8   sliPictureId;
+    WebRtc_UWord64  rpsiPictureId;
+
+    RTCPVoIPMetric*  VoIPMetric;
+};
+
+
+class RTCPReportBlockInformation
+{
+public:
+    RTCPReportBlockInformation();
+    ~RTCPReportBlockInformation();
+
+    // Statistics
+    RTCPReportBlock remoteReceiveBlock;
+    WebRtc_UWord32        remoteMaxJitter;
+
+    // RTT
+    WebRtc_UWord16    RTT;
+    WebRtc_UWord16    minRTT;
+    WebRtc_UWord16    maxRTT;
+    WebRtc_UWord16    avgRTT;
+    WebRtc_UWord32    numAverageCalcs;
+};
+
+class RTCPReceiveInformation
+{
+public:
+    RTCPReceiveInformation();
+    ~RTCPReceiveInformation();
+
+    void VerifyAndAllocateBoundingSet(const WebRtc_UWord32 minimumSize);
+    void VerifyAndAllocateTMMBRSet(const WebRtc_UWord32 minimumSize);
+
+    void InsertTMMBRItem(const WebRtc_UWord32 senderSSRC,
+                         const RTCPUtility::RTCPPacketRTPFBTMMBRItem& TMMBRItem);
+
+    // get
+    WebRtc_Word32 GetTMMBRSet(const WebRtc_UWord32 sourceIdx,
+                            const WebRtc_UWord32 targetIdx,
+                            TMMBRSet* candidateSet);
+
+    WebRtc_UWord32    lastTimeReceived;
+
+    // FIR
+    WebRtc_Word32     lastFIRSequenceNumber;
+    WebRtc_UWord32    lastFIRRequest;
+
+    // TMMBN
+    TMMBRSet        TmmbnBoundingSet;
+
+    // TMMBR
+    TMMBRSet        TmmbrSet;
+
+    bool            readyForDelete;
+private:
+    WebRtc_UWord32*   _tmmbrSetTimeouts;
+};
+
+} // end namespace RTCPHelp
+} // namespace webrtc
+
+#endif // WEBRTC_MODULES_RTP_RTCP_SOURCE_RTCP_RECEIVER_HELP_H_