From 9e5babf9db52e64bdae60137be7696e56241fca6 Mon Sep 17 00:00:00 2001
From: xingzilong <xingzilong@454eff88-639b-444f-9e54-f578c98de674>
Date: 星期五, 18 八月 2017 18:12:17 +0800
Subject: [PATCH] H264 NALU解析  并在RTSPServer判断

---
 VisitFace/RtspNativeCodec/app/libs/live555/include/liveMedia/RTPSink.hh |  233 ++++++++++++++++++++++++++++++++++++++++++++++++++++++++++
 1 files changed, 233 insertions(+), 0 deletions(-)

diff --git a/VisitFace/RtspNativeCodec/app/libs/live555/include/liveMedia/RTPSink.hh b/VisitFace/RtspNativeCodec/app/libs/live555/include/liveMedia/RTPSink.hh
new file mode 100644
index 0000000..f1e20e6
--- /dev/null
+++ b/VisitFace/RtspNativeCodec/app/libs/live555/include/liveMedia/RTPSink.hh
@@ -0,0 +1,233 @@
+/**********
+This library is free software; you can redistribute it and/or modify it under
+the terms of the GNU Lesser General Public License as published by the
+Free Software Foundation; either version 3 of the License, or (at your
+option) any later version. (See <http://www.gnu.org/copyleft/lesser.html>.)
+
+This library is distributed in the hope that it will be useful, but WITHOUT
+ANY WARRANTY; without even the implied warranty of MERCHANTABILITY or FITNESS
+FOR A PARTICULAR PURPOSE.  See the GNU Lesser General Public License for
+more details.
+
+You should have received a copy of the GNU Lesser General Public License
+along with this library; if not, write to the Free Software Foundation, Inc.,
+51 Franklin Street, Fifth Floor, Boston, MA 02110-1301  USA
+**********/
+// "liveMedia"
+// Copyright (c) 1996-2017 Live Networks, Inc.  All rights reserved.
+// RTP Sinks
+// C++ header
+
+#ifndef _RTP_SINK_HH
+#define _RTP_SINK_HH
+
+#ifndef _MEDIA_SINK_HH
+#include "MediaSink.hh"
+#endif
+#ifndef _RTP_INTERFACE_HH
+#include "RTPInterface.hh"
+#endif
+
+class RTPTransmissionStatsDB; // forward
+
+class RTPSink: public MediaSink {
+public:
+  static Boolean lookupByName(UsageEnvironment& env, char const* sinkName,
+			      RTPSink*& resultSink);
+
+  // used by RTSP servers:
+  Groupsock const& groupsockBeingUsed() const { return *(fRTPInterface.gs()); }
+  Groupsock& groupsockBeingUsed() { return *(fRTPInterface.gs()); }
+
+  unsigned char rtpPayloadType() const { return fRTPPayloadType; }
+  unsigned rtpTimestampFrequency() const { return fTimestampFrequency; }
+  void setRTPTimestampFrequency(unsigned freq) {
+    fTimestampFrequency = freq;
+  }
+  char const* rtpPayloadFormatName() const {return fRTPPayloadFormatName;}
+
+  unsigned numChannels() const { return fNumChannels; }
+
+  virtual char const* sdpMediaType() const; // for use in SDP m= lines
+  virtual char* rtpmapLine() const; // returns a string to be delete[]d
+  virtual char const* auxSDPLine();
+      // optional SDP line (e.g. a=fmtp:...)
+
+  u_int16_t currentSeqNo() const { return fSeqNo; }
+  u_int32_t presetNextTimestamp();
+      // ensures that the next timestamp to be used will correspond to
+      // the current 'wall clock' time.
+
+  RTPTransmissionStatsDB& transmissionStatsDB() const {
+    return *fTransmissionStatsDB;
+  }
+
+  Boolean nextTimestampHasBeenPreset() const { return fNextTimestampHasBeenPreset; }
+  Boolean& enableRTCPReports() { return fEnableRTCPReports; }
+
+  void getTotalBitrate(unsigned& outNumBytes, double& outElapsedTime);
+      // returns the number of bytes sent since the last time that we
+      // were called, and resets the counter.
+
+  struct timeval const& creationTime() const { return fCreationTime; }
+  struct timeval const& initialPresentationTime() const { return fInitialPresentationTime; }
+  struct timeval const& mostRecentPresentationTime() const { return fMostRecentPresentationTime; }
+  void resetPresentationTimes();
+
+  // Hacks to allow sending RTP over TCP (RFC 2236, section 10.12):
+  void setStreamSocket(int sockNum, unsigned char streamChannelId) {
+    fRTPInterface.setStreamSocket(sockNum, streamChannelId);
+  }
+  void addStreamSocket(int sockNum, unsigned char streamChannelId) {
+    fRTPInterface.addStreamSocket(sockNum, streamChannelId);
+  }
+  void removeStreamSocket(int sockNum, unsigned char streamChannelId) {
+    fRTPInterface.removeStreamSocket(sockNum, streamChannelId);
+  }
+  unsigned& estimatedBitrate() { return fEstimatedBitrate; } // kbps; usually 0 (i.e., unset)
+
+  u_int32_t SSRC() const {return fSSRC;}
+     // later need a means of changing the SSRC if there's a collision #####
+
+protected:
+  RTPSink(UsageEnvironment& env,
+	  Groupsock* rtpGS, unsigned char rtpPayloadType,
+	  u_int32_t rtpTimestampFrequency,
+	  char const* rtpPayloadFormatName,
+	  unsigned numChannels);
+	// abstract base class
+
+  virtual ~RTPSink();
+
+  // used by RTCP:
+  friend class RTCPInstance;
+  friend class RTPTransmissionStats;
+  u_int32_t convertToRTPTimestamp(struct timeval tv);
+  unsigned packetCount() const {return fPacketCount;}
+  unsigned octetCount() const {return fOctetCount;}
+
+protected:
+  RTPInterface fRTPInterface;
+  unsigned char fRTPPayloadType;
+  unsigned fPacketCount, fOctetCount, fTotalOctetCount /*incl RTP hdr*/;
+  struct timeval fTotalOctetCountStartTime, fInitialPresentationTime, fMostRecentPresentationTime;
+  u_int32_t fCurrentTimestamp;
+  u_int16_t fSeqNo;
+
+private:
+  // redefined virtual functions:
+  virtual Boolean isRTPSink() const;
+
+private:
+  u_int32_t fSSRC, fTimestampBase;
+  unsigned fTimestampFrequency;
+  Boolean fNextTimestampHasBeenPreset;
+  Boolean fEnableRTCPReports; // whether RTCP "SR" reports should be sent for this sink (default: True)
+  char const* fRTPPayloadFormatName;
+  unsigned fNumChannels;
+  struct timeval fCreationTime;
+  unsigned fEstimatedBitrate; // set on creation if known; otherwise 0
+
+  RTPTransmissionStatsDB* fTransmissionStatsDB;
+};
+
+
+class RTPTransmissionStats; // forward
+
+class RTPTransmissionStatsDB {
+public:
+  unsigned numReceivers() const { return fNumReceivers; }
+
+  class Iterator {
+  public:
+    Iterator(RTPTransmissionStatsDB& receptionStatsDB);
+    virtual ~Iterator();
+
+    RTPTransmissionStats* next();
+        // NULL if none
+
+  private:
+    HashTable::Iterator* fIter;
+  };
+
+  // The following is called whenever a RTCP RR packet is received:
+  void noteIncomingRR(u_int32_t SSRC, struct sockaddr_in const& lastFromAddress,
+                      unsigned lossStats, unsigned lastPacketNumReceived,
+                      unsigned jitter, unsigned lastSRTime, unsigned diffSR_RRTime);
+
+  // The following is called when a RTCP BYE packet is received:
+  void removeRecord(u_int32_t SSRC);
+
+  RTPTransmissionStats* lookup(u_int32_t SSRC) const;
+
+private: // constructor and destructor, called only by RTPSink:
+  friend class RTPSink;
+  RTPTransmissionStatsDB(RTPSink& rtpSink);
+  virtual ~RTPTransmissionStatsDB();
+
+private:
+  void add(u_int32_t SSRC, RTPTransmissionStats* stats);
+
+private:
+  friend class Iterator;
+  unsigned fNumReceivers;
+  RTPSink& fOurRTPSink;
+  HashTable* fTable;
+};
+
+class RTPTransmissionStats {
+public:
+  u_int32_t SSRC() const {return fSSRC;}
+  struct sockaddr_in const& lastFromAddress() const {return fLastFromAddress;}
+  unsigned lastPacketNumReceived() const {return fLastPacketNumReceived;}
+  unsigned firstPacketNumReported() const {return fFirstPacketNumReported;}
+  unsigned totNumPacketsLost() const {return fTotNumPacketsLost;}
+  unsigned jitter() const {return fJitter;}
+  unsigned lastSRTime() const { return fLastSRTime; }
+  unsigned diffSR_RRTime() const { return fDiffSR_RRTime; }
+  unsigned roundTripDelay() const;
+      // The round-trip delay (in units of 1/65536 seconds) computed from
+      // the most recently-received RTCP RR packet.
+  struct timeval const& timeCreated() const {return fTimeCreated;}
+  struct timeval const& lastTimeReceived() const {return fTimeReceived;}
+  void getTotalOctetCount(u_int32_t& hi, u_int32_t& lo);
+  void getTotalPacketCount(u_int32_t& hi, u_int32_t& lo);
+
+  // Information which requires at least two RRs to have been received:
+  unsigned packetsReceivedSinceLastRR() const;
+  u_int8_t packetLossRatio() const { return fPacketLossRatio; }
+     // as an 8-bit fixed-point number
+  int packetsLostBetweenRR() const;
+
+private:
+  // called only by RTPTransmissionStatsDB:
+  friend class RTPTransmissionStatsDB;
+  RTPTransmissionStats(RTPSink& rtpSink, u_int32_t SSRC);
+  virtual ~RTPTransmissionStats();
+
+  void noteIncomingRR(struct sockaddr_in const& lastFromAddress,
+		      unsigned lossStats, unsigned lastPacketNumReceived,
+                      unsigned jitter,
+		      unsigned lastSRTime, unsigned diffSR_RRTime);
+
+private:
+  RTPSink& fOurRTPSink;
+  u_int32_t fSSRC;
+  struct sockaddr_in fLastFromAddress;
+  unsigned fLastPacketNumReceived;
+  u_int8_t fPacketLossRatio;
+  unsigned fTotNumPacketsLost;
+  unsigned fJitter;
+  unsigned fLastSRTime;
+  unsigned fDiffSR_RRTime;
+  struct timeval fTimeCreated, fTimeReceived;
+  Boolean fAtLeastTwoRRsHaveBeenReceived;
+  unsigned fOldLastPacketNumReceived;
+  unsigned fOldTotNumPacketsLost;
+  Boolean fFirstPacket;
+  unsigned fFirstPacketNumReported;
+  u_int32_t fLastOctetCount, fTotalOctetCount_hi, fTotalOctetCount_lo;
+  u_int32_t fLastPacketCount, fTotalPacketCount_hi, fTotalPacketCount_lo;
+};
+
+#endif

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