Add even more of the source
This should be about everything needed to build so far?
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172
Server.tproj/RTPSession.h
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172
Server.tproj/RTPSession.h
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/*
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*
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* @APPLE_LICENSE_HEADER_START@
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*
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* Copyright (c) 1999-2008 Apple Inc. All Rights Reserved.
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*
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* This file contains Original Code and/or Modifications of Original Code
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* as defined in and that are subject to the Apple Public Source License
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* Version 2.0 (the 'License'). You may not use this file except in
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* compliance with the License. Please obtain a copy of the License at
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* http://www.opensource.apple.com/apsl/ and read it before using this
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* file.
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*
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* The Original Code and all software distributed under the License are
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* distributed on an 'AS IS' basis, WITHOUT WARRANTY OF ANY KIND, EITHER
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* EXPRESS OR IMPLIED, AND APPLE HEREBY DISCLAIMS ALL SUCH WARRANTIES,
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* INCLUDING WITHOUT LIMITATION, ANY WARRANTIES OF MERCHANTABILITY,
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* FITNESS FOR A PARTICULAR PURPOSE, QUIET ENJOYMENT OR NON-INFRINGEMENT.
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* Please see the License for the specific language governing rights and
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* limitations under the License.
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*
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* @APPLE_LICENSE_HEADER_END@
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*
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*/
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/*
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File: RTPSession.h
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Contains: RTPSession represents an, well, an RTP session. The server creates
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one of these when a new client connects, and it lives for the duration
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of an RTP presentation. It contains all the resources associated with
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that presentation, such as RTPStream objects. It also provides an
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API for manipulating a session (playing it, pausing it, stopping it, etc)
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It is also the active element, ie. it is the object that gets scheduled
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to send out & receive RTP & RTCP packets
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Change History (most recent first):
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*/
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#ifndef _RTPSESSION_H_
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#define _RTPSESSION_H_
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#include "RTPSessionInterface.h"
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#include "RTSPRequestInterface.h"
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#include "RTPStream.h"
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#include "QTSSModule.h"
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class RTPSession : public RTPSessionInterface
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{
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public:
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RTPSession();
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virtual ~RTPSession();
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//
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//ACCESS FUNCTIONS
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QTSSModule* GetPacketSendingModule() { return fModule; }
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Bool16 HasAnRTPStream() { return fHasAnRTPStream; }
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RTPStream* FindRTPStreamForChannelNum(UInt8 inChannelNum);
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//
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// MODIFIERS
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//This puts this session into the session map (called by the server! not the module!)
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//If this function fails (it can return QTSS_DupName), it means that there is already
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//a session with this session ID in the map.
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QTSS_Error Activate(const StrPtrLen& inSessionID);
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// The way this object is implemented currently, only one module can send the
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// packets for a session.
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void SetPacketSendingModule(QTSSModule* inModule) { fModule = inModule; }
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//Once the session is bound, a module can add streams to it.
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//It must pass in a trackID that uniquely identifies this stream.
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//This call can only be made during an RTSP Setup request, and the
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//RTSPRequestInterface must be provided.
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//You may also opt to attach a codec name and type to this stream.
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QTSS_Error AddStream( RTSPRequestInterface* request, RTPStream** outStream,
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QTSS_AddStreamFlags inFlags);
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//Reset the thinning params for all streams using the late tolerance value
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void SetStreamThinningParams(Float32 inLateTolerance);
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//Find the appropriate stream and set the rate adaptation stream data
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void Set3GPPRateAdaptionData(RateAdapationStreamDataFields *rateAdaptStreamData);
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// Store the Session Interface's Link Char values
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void Set3GPPLinkCharData(LinkCharDataFields *linkCharData);
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//Begins playing all streams. Currently must be associated with an RTSP Play
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//request, and the request interface must be provided.
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QTSS_Error Play(RTSPRequestInterface* request, QTSS_PlayFlags inFlags);
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//Pauses all streams.
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void Pause();
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// Tears down the session. This will cause QTSS_SessionClosing_Role to run
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void Teardown();
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//Utility functions. Modules aren't required to use these, but can be useful
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void SendDescribeResponse(RTSPRequestInterface* request);
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void SendAnnounceResponse(RTSPRequestInterface* request);
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void SendPlayResponse(RTSPRequestInterface* request, UInt32 inFlags);
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void SendPauseResponse(RTSPRequestInterface* request)
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{ request->SendHeader(); }
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void SendTeardownResponse(RTSPRequestInterface* request)
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{ request->SetKeepAlive(false); request->SendHeader(); }
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SInt32 GetQualityLevel();
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void SetQualityLevel(SInt32 level);
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void SetMovieBitRateData();
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private:
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//where timeouts, deletion conditions get processed
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virtual SInt64 Run();
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// Utility function used by Play
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UInt32 PowerOf2Floor(UInt32 inNumToFloor);
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//overbuffer logging function
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void LogOverbufferStats();
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enum
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{
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kRTPStreamArraySize = 20,
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kCantGetMutexIdleTime = 10
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};
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QTSSModule* fModule;
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Bool16 fHasAnRTPStream;
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SInt32 fSessionQualityLevel;
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char fRTPStreamArray[kRTPStreamArraySize];
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// Module invocation and module state.
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// This info keeps track of our current state so that
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// the state machine works properly.
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enum
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{
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kStart = 0,
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kSendingPackets = 1
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};
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UInt32 fCurrentModuleIndex;
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UInt32 fCurrentState;
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QTSS_ModuleState fModuleState;
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QTSS_CliSesClosingReason fClosingReason;
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UInt32 fCurrentModule;
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// This is here to give the ability to the ClientSessionClosing role to
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// do asynchronous I/O
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Bool16 fModuleDoingAsyncStuff;
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#if DEBUG
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Bool16 fActivateCalled;
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#endif
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SInt64 fLastBandwidthTrackerStatsUpdate;
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};
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#endif //_RTPSESSION_H_
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