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/OnDemandServerMediaSubsession.hh |  227 ++++++++++++++++++++++++++++++++++++++++++++++++++++++++
 1 files changed, 227 insertions(+), 0 deletions(-)

diff --git a/VisitFace/RtspNativeCodec/app/libs/live555/include/liveMedia/OnDemandServerMediaSubsession.hh b/VisitFace/RtspNativeCodec/app/libs/live555/include/liveMedia/OnDemandServerMediaSubsession.hh
new file mode 100644
index 0000000..9a1210b
--- /dev/null
+++ b/VisitFace/RtspNativeCodec/app/libs/live555/include/liveMedia/OnDemandServerMediaSubsession.hh
@@ -0,0 +1,227 @@
+/**********
+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.
+// A 'ServerMediaSubsession' object that creates new, unicast, "RTPSink"s
+// on demand.
+// C++ header
+
+#ifndef _ON_DEMAND_SERVER_MEDIA_SUBSESSION_HH
+#define _ON_DEMAND_SERVER_MEDIA_SUBSESSION_HH
+
+#ifndef _SERVER_MEDIA_SESSION_HH
+#include "ServerMediaSession.hh"
+#endif
+#ifndef _RTP_SINK_HH
+#include "RTPSink.hh"
+#endif
+#ifndef _BASIC_UDP_SINK_HH
+#include "BasicUDPSink.hh"
+#endif
+#ifndef _RTCP_HH
+#include "RTCP.hh"
+#endif
+
+class OnDemandServerMediaSubsession: public ServerMediaSubsession {
+protected: // we're a virtual base class
+  OnDemandServerMediaSubsession(UsageEnvironment& env, Boolean reuseFirstSource,
+				portNumBits initialPortNum = 6970,
+				Boolean multiplexRTCPWithRTP = False);
+  virtual ~OnDemandServerMediaSubsession();
+
+protected: // redefined virtual functions
+  virtual char const* sdpLines();
+  virtual void getStreamParameters(unsigned clientSessionId,
+				   netAddressBits clientAddress,
+                                   Port const& clientRTPPort,
+                                   Port const& clientRTCPPort,
+				   int tcpSocketNum,
+                                   unsigned char rtpChannelId,
+                                   unsigned char rtcpChannelId,
+                                   netAddressBits& destinationAddress,
+				   u_int8_t& destinationTTL,
+                                   Boolean& isMulticast,
+                                   Port& serverRTPPort,
+                                   Port& serverRTCPPort,
+                                   void*& streamToken);
+  virtual void startStream(unsigned clientSessionId, void* streamToken,
+			   TaskFunc* rtcpRRHandler,
+			   void* rtcpRRHandlerClientData,
+			   unsigned short& rtpSeqNum,
+                           unsigned& rtpTimestamp,
+			   ServerRequestAlternativeByteHandler* serverRequestAlternativeByteHandler,
+                           void* serverRequestAlternativeByteHandlerClientData);
+  virtual void pauseStream(unsigned clientSessionId, void* streamToken);
+  virtual void seekStream(unsigned clientSessionId, void* streamToken, double& seekNPT, double streamDuration, u_int64_t& numBytes);
+  virtual void seekStream(unsigned clientSessionId, void* streamToken, char*& absStart, char*& absEnd);
+  virtual void nullSeekStream(unsigned clientSessionId, void* streamToken,
+			      double streamEndTime, u_int64_t& numBytes);
+  virtual void setStreamScale(unsigned clientSessionId, void* streamToken, float scale);
+  virtual float getCurrentNPT(void* streamToken);
+  virtual FramedSource* getStreamSource(void* streamToken);
+  virtual void getRTPSinkandRTCP(void* streamToken,
+				 RTPSink const*& rtpSink, RTCPInstance const*& rtcp);
+  virtual void deleteStream(unsigned clientSessionId, void*& streamToken);
+
+protected: // new virtual functions, possibly redefined by subclasses
+  virtual char const* getAuxSDPLine(RTPSink* rtpSink,
+				    FramedSource* inputSource);
+  virtual void seekStreamSource(FramedSource* inputSource, double& seekNPT, double streamDuration, u_int64_t& numBytes);
+    // This routine is used to seek by relative (i.e., NPT) time.
+    // "streamDuration", if >0.0, specifies how much data to stream, past "seekNPT".  (If <=0.0, all remaining data is streamed.)
+    // "numBytes" returns the size (in bytes) of the data to be streamed, or 0 if unknown or unlimited.
+  virtual void seekStreamSource(FramedSource* inputSource, char*& absStart, char*& absEnd);
+    // This routine is used to seek by 'absolute' time.
+    // "absStart" should be a string of the form "YYYYMMDDTHHMMSSZ" or "YYYYMMDDTHHMMSS.<frac>Z".
+    // "absEnd" should be either NULL (for no end time), or a string of the same form as "absStart".
+    // These strings may be modified in-place, or can be reassigned to a newly-allocated value (after delete[]ing the original).
+  virtual void setStreamSourceScale(FramedSource* inputSource, float scale);
+  virtual void setStreamSourceDuration(FramedSource* inputSource, double streamDuration, u_int64_t& numBytes);
+  virtual void closeStreamSource(FramedSource* inputSource);
+
+protected: // new virtual functions, defined by all subclasses
+  virtual FramedSource* createNewStreamSource(unsigned clientSessionId,
+					      unsigned& estBitrate) = 0;
+      // "estBitrate" is the stream's estimated bitrate, in kbps
+  virtual RTPSink* createNewRTPSink(Groupsock* rtpGroupsock,
+				    unsigned char rtpPayloadTypeIfDynamic,
+				    FramedSource* inputSource) = 0;
+
+protected: // new virtual functions, may be redefined by a subclass:
+  virtual Groupsock* createGroupsock(struct in_addr const& addr, Port port);
+  virtual RTCPInstance* createRTCP(Groupsock* RTCPgs, unsigned totSessionBW, /* in kbps */
+				   unsigned char const* cname, RTPSink* sink);
+
+public:
+  void multiplexRTCPWithRTP() { fMultiplexRTCPWithRTP = True; }
+    // An alternative to passing the "multiplexRTCPWithRTP" parameter as True in the constructor
+
+  void setRTCPAppPacketHandler(RTCPAppHandlerFunc* handler, void* clientData);
+    // Sets a handler to be called if a RTCP "APP" packet arrives from any future client.
+    // (Any current clients are not affected; any "APP" packets from them will continue to be
+    // handled by whatever handler existed when the client sent its first RTSP "PLAY" command.)
+    // (Call with (NULL, NULL) to remove an existing handler - for future clients only)
+
+  void sendRTCPAppPacket(u_int8_t subtype, char const* name,
+			 u_int8_t* appDependentData, unsigned appDependentDataSize);
+    // Sends a custom RTCP "APP" packet to the most recent client (if "reuseFirstSource" was False),
+    // or to all current clients (if "reuseFirstSource" was True).
+    // The parameters correspond to their
+    // respective fields as described in the RTP/RTCP definition (RFC 3550).
+    // Note that only the low-order 5 bits of "subtype" are used, and only the first 4 bytes
+    // of "name" are used.  (If "name" has fewer than 4 bytes, or is NULL,
+    // then the remaining bytes are '\0'.)
+
+private:
+  void setSDPLinesFromRTPSink(RTPSink* rtpSink, FramedSource* inputSource,
+			      unsigned estBitrate);
+      // used to implement "sdpLines()"
+
+protected:
+  char* fSDPLines;
+  HashTable* fDestinationsHashTable; // indexed by client session id
+
+private:
+  Boolean fReuseFirstSource;
+  portNumBits fInitialPortNum;
+  Boolean fMultiplexRTCPWithRTP;
+  void* fLastStreamToken;
+  char fCNAME[100]; // for RTCP
+  RTCPAppHandlerFunc* fAppHandlerTask;
+  void* fAppHandlerClientData;
+  friend class StreamState;
+};
+
+
+// A class that represents the state of an ongoing stream.  This is used only internally, in the implementation of
+// "OnDemandServerMediaSubsession", but we expose the definition here, in case subclasses of "OnDemandServerMediaSubsession"
+// want to access it.
+
+class Destinations {
+public:
+  Destinations(struct in_addr const& destAddr,
+               Port const& rtpDestPort,
+               Port const& rtcpDestPort)
+    : isTCP(False), addr(destAddr), rtpPort(rtpDestPort), rtcpPort(rtcpDestPort) {
+  }
+  Destinations(int tcpSockNum, unsigned char rtpChanId, unsigned char rtcpChanId)
+    : isTCP(True), rtpPort(0) /*dummy*/, rtcpPort(0) /*dummy*/,
+      tcpSocketNum(tcpSockNum), rtpChannelId(rtpChanId), rtcpChannelId(rtcpChanId) {
+  }
+
+public:
+  Boolean isTCP;
+  struct in_addr addr;
+  Port rtpPort;
+  Port rtcpPort;
+  int tcpSocketNum;
+  unsigned char rtpChannelId, rtcpChannelId;
+};
+
+class StreamState {
+public:
+  StreamState(OnDemandServerMediaSubsession& master,
+              Port const& serverRTPPort, Port const& serverRTCPPort,
+	      RTPSink* rtpSink, BasicUDPSink* udpSink,
+	      unsigned totalBW, FramedSource* mediaSource,
+	      Groupsock* rtpGS, Groupsock* rtcpGS);
+  virtual ~StreamState();
+
+  void startPlaying(Destinations* destinations, unsigned clientSessionId,
+		    TaskFunc* rtcpRRHandler, void* rtcpRRHandlerClientData,
+		    ServerRequestAlternativeByteHandler* serverRequestAlternativeByteHandler,
+                    void* serverRequestAlternativeByteHandlerClientData);
+  void pause();
+  void sendRTCPAppPacket(u_int8_t subtype, char const* name,
+			 u_int8_t* appDependentData, unsigned appDependentDataSize);
+  void endPlaying(Destinations* destinations, unsigned clientSessionId);
+  void reclaim();
+
+  unsigned& referenceCount() { return fReferenceCount; }
+
+  Port const& serverRTPPort() const { return fServerRTPPort; }
+  Port const& serverRTCPPort() const { return fServerRTCPPort; }
+
+  RTPSink* rtpSink() const { return fRTPSink; }
+  RTCPInstance* rtcpInstance() const { return fRTCPInstance; }
+
+  float streamDuration() const { return fStreamDuration; }
+
+  FramedSource* mediaSource() const { return fMediaSource; }
+  float& startNPT() { return fStartNPT; }
+
+private:
+  OnDemandServerMediaSubsession& fMaster;
+  Boolean fAreCurrentlyPlaying;
+  unsigned fReferenceCount;
+
+  Port fServerRTPPort, fServerRTCPPort;
+
+  RTPSink* fRTPSink;
+  BasicUDPSink* fUDPSink;
+
+  float fStreamDuration;
+  unsigned fTotalBW;
+  RTCPInstance* fRTCPInstance;
+
+  FramedSource* fMediaSource;
+  float fStartNPT; // initial 'normal play time'; reset after each seek
+
+  Groupsock* fRTPgs;
+  Groupsock* fRTCPgs;
+};
+
+#endif

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