407 lines
15 KiB
C
407 lines
15 KiB
C
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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: RTPStream.h
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Contains: Represents a single client stream (audio, video, etc).
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Control API is similar to overall session API.
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Contains all stream-specific data & resources, used by Session when it
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wants to send out or receive data for this stream
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This is also the class that implements the RTP stream dictionary
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for QTSS API.
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*/
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#ifndef __RTPSTREAM_H__
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#define __RTPSTREAM_H__
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#include "QTSS.h"
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#include "QTSSDictionary.h"
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#include "QTSS_Private.h"
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#include "UDPDemuxer.h"
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#include "UDPSocketPool.h"
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#include "RTSPRequestInterface.h"
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#include "RTPSessionInterface.h"
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#include "RTPPacketResender.h"
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#include "QTSServerInterface.h"
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#include "RTPStream3GPP.h"
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#include "RTCPPacket.h"
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#include "RTSPRequest3GPP.h"
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#ifndef MIN
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#define MIN(a,b) (((a)<(b))?(a):(b))
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#endif /* MIN */
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#ifndef MAX
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#define MAX(a,b) (((a)>(b))?(a):(b))
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#endif /* MAX */
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class RTPStream : public QTSSDictionary, public UDPDemuxerTask
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{
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public:
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// Initializes dictionary resources
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static void Initialize();
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//
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// CONSTRUCTOR / DESTRUCTOR
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RTPStream(UInt32 inSSRC, RTPSessionInterface* inSession);
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virtual ~RTPStream();
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//
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//ACCESS FUNCTIONS
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UInt32 GetSSRC() { return fSsrc; }
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UInt8 GetRTPChannelNum() { return fRTPChannel; }
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UInt8 GetRTCPChannelNum() { return fRTCPChannel; }
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RTPPacketResender* GetResender() { return &fResender; }
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QTSS_RTPTransportType GetTransportType() { return fTransportType; }
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UInt32 GetStalePacketsDropped() { return fStalePacketsDropped; }
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UInt32 GetTotalPacketsRecv() { return fTotalPacketsRecv; }
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UInt32 GetSDPStreamID() { return fTrackID; } //streamID is trackID
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RTPSessionInterface &GetSession() { return *fSession; }
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// Setup uses the info in the RTSPRequestInterface to associate
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// all the necessary resources, ports, sockets, etc, etc, with this
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// stream.
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QTSS_Error Setup(RTSPRequestInterface* request, QTSS_AddStreamFlags inFlags);
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// Write sends RTP data to the client. Caller must specify
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// either qtssWriteFlagsIsRTP or qtssWriteFlagsIsRTCP
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virtual QTSS_Error Write(void* inBuffer, UInt32 inLen,
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UInt32* outLenWritten, QTSS_WriteFlags inFlags);
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//UTILITY FUNCTIONS:
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//These are not necessary to call and do not manipulate the state of the
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//stream. They may, however, be useful services exported by the server
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// Formats a standard setup response.
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void SendSetupResponse(RTSPRequestInterface* request);
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//Formats a transport header for this stream.
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void AppendTransport(RTSPRequestInterface* request);
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//Formats a RTP-Info header for this stream.
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//Isn't useful unless you've already called Play()
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void AppendRTPInfo(QTSS_RTSPHeader inHeader,
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RTSPRequestInterface* request, UInt32 inFlags, Bool16 lastInfo);
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//
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// When we get an incoming Interleaved Packet for this stream, this
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// function should be called
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void ProcessIncomingInterleavedData(UInt8 inChannelNum, RTSPSessionInterface* inRTSPSession, StrPtrLen* inPacket);
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//When we get a new RTCP packet, we can directly invoke the RTP session and tell it
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//to process the packet right now!
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void ProcessIncomingRTCPPacket(StrPtrLen* inPacket);
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//Process the incoming ack RTCP packet
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Bool16 ProcessAckPacket(RTCPPacket &rtcpPacket, SInt64 &curTime);
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//Process the incoming qtss app RTCP packet
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Bool16 ProcessCompressedQTSSPacket(RTCPPacket &rtcpPacket, SInt64 &curTime, StrPtrLen ¤tPtr);
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Bool16 ProcessNADUPacket(RTCPPacket &rtcpPacket, SInt64 &curTime, StrPtrLen ¤tPtr, UInt32 highestSeqNum);
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// Send a RTCP SR on this stream. Pass in true if this SR should also have a BYE
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void SendRTCPSR(const SInt64& inTime, Bool16 inAppendBye = false);
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//
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// Retransmits get sent when there is new data to be sent, but this function
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// should be called periodically even if there is no new packet data, as
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// the pipe should have a steady stream of data in it.
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void SendRetransmits();
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//
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// Update the thinning parameters for this stream to match current prefs
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void SetThinningParams();
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//
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// Reset the delay parameters that are stored for the thinning calculations
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void ResetThinningDelayParams() { fLastCurrentPacketDelay = 0; }
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void SetLateTolerance(Float32 inLateToleranceInSec) { fLateToleranceInSec = inLateToleranceInSec; }
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void EnableSSRC() { fEnableSSRC = true; }
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void DisableSSRC() { fEnableSSRC = false; }
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void SetRateAdaptData(RateAdapationStreamDataFields *rateAdaptStreamData);
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void SetBitRateData(UInt32 movieBitRate) { fStream3GPP->SetBitRateData(movieBitRate); }
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//Tells the stream that it has been paused; the next Write will restart the stream.
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void Pause() { fStream3GPP->Pause(); }
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void SetMinQuality() { SetQualityLevel(fNumQualityLevels); }
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void SetMaxQuality() { SetQualityLevel(kMaxQualityLevel); }
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SInt32 GetQualityLevel();
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void SetQualityLevel(SInt32 level);
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void HalveQualityLevel()
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{
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UInt32 minLevel = fNumQualityLevels - 1;
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SetQualityLevel(minLevel - (minLevel - GetQualityLevel()) / 2);
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}
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void SetMaxQualityLevelLimit(SInt32 newMaxLimit) //Changes what is the best quality level possible
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{
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SInt32 minLevel = MAX(0, (SInt32) fNumQualityLevels - 2); //do not drop down to key frames
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fMaxQualityLevel = MAX(MIN(minLevel, newMaxLimit), 0);
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SetQualityLevel(GetQualityLevel());
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}
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SInt32 GetMaxQualityLevelLimit() { return fMaxQualityLevel; }
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UInt32 GetNumQualityLevels() { return fNumQualityLevels; }
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QTSS_RTPPayloadType GetPayLoadType() { return fPayloadType; }
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private:
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enum
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{
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kMaxSsrcSizeInBytes = 12,
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kMaxStreamURLSizeInBytes = 32,
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kDefaultPayloadBufSize = 32,
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kSenderReportIntervalInSecs = 7,
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kSenderReportInterval3GPPInSecs = 1,
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kNumPrebuiltChNums = 10,
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kMaxQualityLevel = 0,
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kIsRTCPPacket = TRUE,
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kIsRTPPacket = FALSE
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};
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SInt64 fLastQualityChange;
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SInt32 fQualityInterval;
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//either pointers to the statically allocated sockets (maintained by the server)
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//or fresh ones (only fresh in extreme special cases)
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UDPSocketPair* fSockets;
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RTPSessionInterface* fSession;
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// info for kinda reliable UDP
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//DssDurationTimer fInfoDisplayTimer;
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SInt32 fBytesSentThisInterval;
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SInt32 fDisplayCount;
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Bool16 fSawFirstPacket;
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SInt64 fStreamCumDuration;
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// manages UDP retransmits
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RTPPacketResender fResender;
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RTPBandwidthTracker* fTracker;
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//who am i sending to?
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UInt32 fRemoteAddr;
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UInt16 fRemoteRTPPort;
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UInt16 fRemoteRTCPPort;
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UInt16 fLocalRTPPort;
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UInt32 fMonitorAddr;
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int fMonitorSocket;
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UInt32 fPlayerToMonitorAddr;
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//RTCP stuff
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SInt64 fLastSenderReportTime;
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UInt32 fPacketCount;
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UInt32 fLastPacketCount;
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UInt32 fPacketCountInRTCPInterval;
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UInt32 fByteCount;
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// DICTIONARY ATTRIBUTES
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//Module assigns a streamID to this object
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UInt32 fTrackID;
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//low-level RTP stuff
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UInt32 fSsrc;
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char fSsrcString[kMaxSsrcSizeInBytes];
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StrPtrLen fSsrcStringPtr;
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Bool16 fEnableSSRC;
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//Payload name and codec type.
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char fPayloadNameBuf[kDefaultPayloadBufSize];
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QTSS_RTPPayloadType fPayloadType;
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//Media information.
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UInt16 fFirstSeqNumber;//used in sending the play response
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UInt32 fFirstTimeStamp;//RTP time
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UInt32 fTimescale;
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//what is the URL for this stream?
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char fStreamURL[kMaxStreamURLSizeInBytes];
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StrPtrLen fStreamURLPtr;
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SInt32 fQualityLevel;
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UInt32 fNumQualityLevels;
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UInt32 fLastRTPTimestamp;
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SInt64 fLastNTPTimeStamp;
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UInt32 fEstRTT; //The estimated RTT calculated from RTCP's DLSR and LSR fields
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// RTCP data
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UInt32 fFractionLostPackets;
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UInt32 fTotalLostPackets;
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UInt32 fJitter;
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UInt32 fReceiverBitRate;
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UInt16 fAvgLateMsec;
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UInt16 fPercentPacketsLost;
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UInt16 fAvgBufDelayMsec;
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UInt16 fIsGettingBetter;
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UInt16 fIsGettingWorse;
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UInt32 fNumEyes;
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UInt32 fNumEyesActive;
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UInt32 fNumEyesPaused;
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UInt32 fTotalPacketsRecv;
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UInt32 fPriorTotalPacketsRecv;
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UInt16 fTotalPacketsDropped;
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UInt16 fTotalPacketsLost;
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UInt32 fCurPacketsLostInRTCPInterval;
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UInt16 fClientBufferFill;
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UInt16 fFrameRate;
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UInt16 fExpectedFrameRate;
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UInt16 fAudioDryCount;
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UInt32 fClientSSRC;
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Bool16 fIsTCP;
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QTSS_RTPTransportType fTransportType;
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// HTTP params
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// Each stream has a set of thinning related tolerances,
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// that are dependent on prefs and parameters in the SETUP.
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// These params, as well as the current packet delay determine
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// whether a packet gets dropped.
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SInt32 fTurnThinningOffDelay_TCP;
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SInt32 fIncreaseThinningDelay_TCP;
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SInt32 fDropAllPacketsForThisStreamDelay_TCP;
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UInt32 fStalePacketsDropped_TCP;
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SInt64 fTimeStreamCaughtUp_TCP;
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SInt64 fLastQualityLevelIncreaseTime_TCP;
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//
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// Each stream has a set of thinning related tolerances,
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// that are dependent on prefs and parameters in the SETUP.
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// These params, as well as the current packet delay determine
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// whether a packet gets dropped.
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SInt32 fThinAllTheWayDelay;
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SInt32 fAlwaysThinDelay;
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SInt32 fStartThinningDelay;
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SInt32 fStartThickingDelay;
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SInt32 fThickAllTheWayDelay;
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SInt32 fQualityCheckInterval;
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SInt32 fDropAllPacketsForThisStreamDelay;
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UInt32 fStalePacketsDropped;
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SInt64 fLastCurrentPacketDelay;
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Bool16 fWaitOnLevelAdjustment;
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Float32 fBufferDelay; // from the sdp
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Float32 fLateToleranceInSec;
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// Pointer to the stream ref (this is just a this pointer)
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QTSS_StreamRef fStreamRef;
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UInt32 fCurrentAckTimeout;
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SInt32 fMaxSendAheadTimeMSec;
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#if DEBUG
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UInt32 fNumPacketsDroppedOnTCPFlowControl;
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SInt64 fFlowControlStartedMsec;
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SInt64 fFlowControlDurationMsec;
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#endif
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// If we are interleaving RTP data over the TCP connection,
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// these are channel numbers to use for RTP & RTCP
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UInt8 fRTPChannel;
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UInt8 fRTCPChannel;
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QTSS_RTPNetworkMode fNetworkMode;
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SInt64 fStreamStartTimeOSms;
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RTPStream3GPP* fStream3GPP;
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SInt32 fLastQualityLevel;
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SInt32 fLastRateLevel;
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Bool16 fDisableThinning;
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SInt64 fLastQualityUpdate;
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UInt32 fDefaultQualityLevel;
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SInt32 fMaxQualityLevel;
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Bool16 fInitialMaxQualityLevelIsSet;
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Bool16 fUDPMonitorEnabled;
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UInt16 fMonitorVideoDestPort;
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UInt16 fMonitorAudioDestPort;
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//-----------------------------------------------------------
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// acutally write the data out that way
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QTSS_Error InterleavedWrite(void* inBuffer, UInt32 inLen, UInt32* outLenWritten, unsigned char channel );
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// implements the ReliableRTP protocol
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QTSS_Error ReliableRTPWrite(void* inBuffer, UInt32 inLen, const SInt64& curPacketDelay);
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void SetTCPThinningParams();
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QTSS_Error TCPWrite(void* inBuffer, UInt32 inLen, UInt32* outLenWritten, UInt32 inFlags);
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static QTSSAttrInfoDict::AttrInfo sAttributes[];
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static StrPtrLen sChannelNums[];
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static QTSS_ModuleState sRTCPProcessModuleState;
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static char *noType;
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static char *UDP;
|
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static char *RUDP;
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static char *TCP;
|
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Bool16 UpdateQualityLevel(const SInt64& inTransmitTime, const SInt64& inCurrentPacketDelay,
|
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const SInt64& inCurrentTime, UInt32 inPacketSize);
|
||
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void DisableThinning() { fDisableThinning = true; }
|
||
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void Update3GPPQualityLevels(QTSS_PacketStruct* thePacket, SInt64 theTime);
|
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Bool16 Supports3GPPQualityLevels();
|
||
|
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void SetInitialMaxQualityLevel();
|
||
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|
||
|
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char *GetStreamTypeStr();
|
||
|
|
enum { rtp = 0, rtcpSR = 1, rtcpRR = 2, rtcpACK = 3, rtcpAPP = 4 };
|
||
|
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Float32 GetStreamStartTimeSecs() { return (Float32) ((OS::Milliseconds() - this->fSession->GetSessionCreateTime())/1000.0); }
|
||
|
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void PrintPacket(char *inBuffer, UInt32 inLen, SInt32 inType);
|
||
|
|
void PrintRTP(char* packetBuff, UInt32 inLen);
|
||
|
|
void PrintRTCPSenderReport(char* packetBuff, UInt32 inLen);
|
||
|
|
inline void PrintPacketPrefEnabled(char *inBuffer,UInt32 inLen, SInt32 inType) { if (QTSServerInterface::GetServer()->GetPrefs()->PacketHeaderPrintfsEnabled() ) this->PrintPacket(inBuffer,inLen, inType); }
|
||
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|
||
|
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void SetOverBufferState(RTSPRequestInterface* request);
|
||
|
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|
||
|
|
Bool16 TestRTCPPackets(StrPtrLen* inPacketPtr, UInt32 itemName);
|
||
|
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|
||
|
|
void UDPMonitorWrite(void* thePacketData, UInt32 inLen, Bool16 isRTCP);
|
||
|
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|
||
|
|
|
||
|
|
};
|
||
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|
||
|
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#endif // __RTPSTREAM_H__
|