git-svn-id: http://webrtc.googlecode.com/svn/trunk@4 4adac7df-926f-26a2-2b94-8c16560cd09d
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niklase@google.com
236
modules/rtp_rtcp/source/rtp_receiver.h
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236
modules/rtp_rtcp/source/rtp_receiver.h
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/*
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* Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
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*
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* Use of this source code is governed by a BSD-style license
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* that can be found in the LICENSE file in the root of the source
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* tree. An additional intellectual property rights grant can be found
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* in the file PATENTS. All contributing project authors may
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* be found in the AUTHORS file in the root of the source tree.
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*/
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#ifndef WEBRTC_MODULES_RTP_RTCP_SOURCE_RTP_RECEIVER_H_
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#define WEBRTC_MODULES_RTP_RTCP_SOURCE_RTP_RECEIVER_H_
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#include "typedefs.h"
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#include "rtp_utility.h"
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#include "rtp_rtcp.h"
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#include "rtp_receiver_audio.h"
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#include "rtp_receiver_video.h"
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#include "rtcp_receiver_help.h"
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#include "Bitrate.h"
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namespace webrtc {
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class RtpRtcpFeedback;
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class Trace;
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class RTPReceiver : public RTPReceiverAudio, public RTPReceiverVideo, public Bitrate
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{
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public:
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RTPReceiver(const WebRtc_Word32 id,
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const bool audio,
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ModuleRtpRtcpPrivate& callback);
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virtual ~RTPReceiver();
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virtual void ChangeUniqueId(const WebRtc_Word32 id);
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WebRtc_Word32 Init();
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RtpVideoCodecTypes VideoCodecType() const;
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WebRtc_UWord32 MaxConfiguredBitrate() const;
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WebRtc_Word32 SetPacketTimeout(const WebRtc_UWord32 timeoutMS);
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void PacketTimeout();
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void ProcessDeadOrAlive(const bool RTCPalive, const WebRtc_UWord32 now);
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void ProcessBitrate();
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WebRtc_Word32 RegisterIncomingDataCallback(RtpData* incomingDataCallback);
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WebRtc_Word32 RegisterIncomingRTPCallback(RtpFeedback* incomingMessagesCallback);
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WebRtc_Word32 RegisterReceivePayload( const WebRtc_Word8 payloadName[RTP_PAYLOAD_NAME_SIZE],
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const WebRtc_Word8 payloadType,
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const WebRtc_UWord32 frequency,
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const WebRtc_UWord8 channels,
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const WebRtc_UWord32 rate);
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WebRtc_Word32 DeRegisterReceivePayload(const WebRtc_Word8 payloadType);
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WebRtc_Word32 ReceivePayloadType(const WebRtc_Word8 payloadName[RTP_PAYLOAD_NAME_SIZE],
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const WebRtc_UWord32 frequency,
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const WebRtc_UWord8 channels,
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WebRtc_Word8* payloadType,
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const WebRtc_UWord32 rate) const;
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WebRtc_Word32 ReceivePayload(const WebRtc_Word8 payloadType,
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WebRtc_Word8 payloadName[RTP_PAYLOAD_NAME_SIZE],
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WebRtc_UWord32* frequency,
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WebRtc_UWord8* channels,
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WebRtc_UWord32* rate) const;
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WebRtc_Word32 RemotePayload(WebRtc_Word8 payloadName[RTP_PAYLOAD_NAME_SIZE],
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WebRtc_Word8* payloadType,
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WebRtc_UWord32* frequency,
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WebRtc_UWord8* channels) const;
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WebRtc_Word32 IncomingRTPPacket(WebRtcRTPHeader* rtpheader,
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const WebRtc_UWord8* incomingRtpPacket,
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const WebRtc_UWord16 incomingRtpPacketLengt);
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NACKMethod NACK() const ;
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// Turn negative acknowledgement requests on/off
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WebRtc_Word32 SetNACKStatus(const NACKMethod method);
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// last received
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virtual WebRtc_UWord32 TimeStamp() const;
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virtual WebRtc_UWord16 SequenceNumber() const;
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WebRtc_Word32 EstimatedRemoteTimeStamp(WebRtc_UWord32& timestamp) const;
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WebRtc_UWord32 SSRC() const;
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WebRtc_Word32 CSRCs( WebRtc_UWord32 arrOfCSRC[kRtpCsrcSize]) const;
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WebRtc_Word32 Energy( WebRtc_UWord8 arrOfEnergy[kRtpCsrcSize]) const;
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// get the currently configured SSRC filter
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WebRtc_Word32 SSRCFilter(WebRtc_UWord32& allowedSSRC) const;
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// set a SSRC to be used as a filter for incoming RTP streams
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WebRtc_Word32 SetSSRCFilter(const bool enable, const WebRtc_UWord32 allowedSSRC);
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WebRtc_Word32 Statistics(WebRtc_UWord8 *fraction_lost,
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WebRtc_UWord32 *cum_lost,
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WebRtc_UWord32 *ext_max,
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WebRtc_UWord32 *jitter, // will be moved from JB
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WebRtc_UWord32 *max_jitter,
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bool reset = false) const;
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WebRtc_Word32 Statistics(WebRtc_UWord8 *fraction_lost,
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WebRtc_UWord32 *cum_lost,
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WebRtc_UWord32 *ext_max,
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WebRtc_UWord32 *jitter, // will be moved from JB
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WebRtc_UWord32 *max_jitter,
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WebRtc_Word32 *missing,
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bool reset = false) const;
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WebRtc_Word32 DataCounters(WebRtc_UWord32 *bytesReceived,
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WebRtc_UWord32 *packetsReceived) const;
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WebRtc_Word32 ResetStatistics();
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WebRtc_Word32 ResetDataCounters();
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WebRtc_UWord16 PacketOHReceived() const;
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WebRtc_UWord32 PacketCountReceived() const;
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WebRtc_UWord32 ByteCountReceived() const;
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virtual WebRtc_UWord32 PayloadTypeToPayload(const WebRtc_UWord8 payloadType,
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ModuleRTPUtility::Payload*& payload) const;
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protected:
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virtual WebRtc_Word32 CallbackOfReceivedPayloadData(const WebRtc_UWord8* payloadData,
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const WebRtc_UWord16 payloadSize,
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const WebRtcRTPHeader* rtpHeader);
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virtual bool RetransmitOfOldPacket(const WebRtc_UWord16 sequenceNumber,
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const WebRtc_UWord32 rtpTimeStamp) const;
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void UpdateStatistics(const WebRtcRTPHeader* rtpHeader,
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const WebRtc_UWord16 bytes,
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const bool oldPacket);
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virtual WebRtc_Word8 REDPayloadType() const;
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private:
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// Is RED configured with payload type payloadType
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bool REDPayloadType(const WebRtc_Word8 payloadType) const;
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bool InOrderPacket(const WebRtc_UWord16 sequenceNumber) const;
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void CheckSSRCChanged(const WebRtcRTPHeader* rtpHeader);
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void CheckCSRC(const WebRtcRTPHeader* rtpHeader);
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WebRtc_Word32 CheckPayloadChanged(const WebRtcRTPHeader* rtpHeader,
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const WebRtc_Word8 firstPayloadByte,
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bool& isRED,
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ModuleRTPUtility::AudioPayload& audioSpecific,
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ModuleRTPUtility::VideoPayload& videoSpecific);
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void UpdateNACKBitRate(WebRtc_Word32 bytes, WebRtc_UWord32 now);
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bool ProcessNACKBitRate(WebRtc_UWord32 now);
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private:
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WebRtc_Word32 _id;
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const bool _audio;
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CriticalSectionWrapper& _criticalSectionCbs;
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ModuleRtpRtcpPrivate& _cbPrivateFeedback;
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RtpFeedback* _cbRtpFeedback;
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RtpData* _cbRtpData;
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CriticalSectionWrapper& _criticalSectionRTPReceiver;
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mutable WebRtc_UWord32 _lastReceiveTime;
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WebRtc_UWord16 _lastReceivedPayloadLength;
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WebRtc_Word8 _lastReceivedPayloadType;
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WebRtc_Word8 _lastReceivedMediaPayloadType;
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ModuleRTPUtility::AudioPayload _lastReceivedAudioSpecific;
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ModuleRTPUtility::VideoPayload _lastReceivedVideoSpecific;
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WebRtc_UWord32 _packetTimeOutMS;
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WebRtc_Word8 _redPayloadType;
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//
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MapWrapper _payloadTypeMap;
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// SSRCs
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WebRtc_UWord32 _SSRC;
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WebRtc_UWord8 _numCSRCs;
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WebRtc_UWord32 _currentRemoteCSRC[kRtpCsrcSize];
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WebRtc_UWord8 _numEnergy;
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WebRtc_UWord8 _currentRemoteEnergy[kRtpCsrcSize];
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bool _useSSRCFilter;
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WebRtc_UWord32 _SSRCFilter;
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// stats on received RTP packets
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WebRtc_UWord32 _jitterQ4;
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mutable WebRtc_UWord32 _jitterMaxQ4;
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mutable WebRtc_UWord32 _cumulativeLoss;
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WebRtc_UWord32 _localTimeLastReceivedTimestamp;
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WebRtc_UWord32 _lastReceivedTimestamp;
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WebRtc_UWord16 _lastReceivedSequenceNumber;
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WebRtc_UWord16 _receivedSeqFirst;
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WebRtc_UWord16 _receivedSeqMax;
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WebRtc_UWord16 _receivedSeqWraps;
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// current counter values
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WebRtc_UWord16 _receivedPacketOH;
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WebRtc_UWord32 _receivedByteCount;
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WebRtc_UWord32 _receivedOldPacketCount;
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WebRtc_UWord32 _receivedInorderPacketCount;
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// counter values when we sent the last report
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mutable WebRtc_UWord32 _lastReportInorderPackets;
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mutable WebRtc_UWord32 _lastReportOldPackets;
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mutable WebRtc_UWord16 _lastReportSeqMax;
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mutable WebRtc_UWord8 _lastReportFractionLost;
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mutable WebRtc_UWord32 _lastReportCumulativeLost; // 24 bits valid
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mutable WebRtc_UWord32 _lastReportExtendedHighSeqNum;
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mutable WebRtc_UWord32 _lastReportJitter;
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// NACK
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NACKMethod _nackMethod;
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};
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} // namespace webrtc
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#endif // WEBRTC_MODULES_RTP_RTCP_SOURCE_RTP_RECEIVER_H_
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