| /* |
| * Copyright (c) 2012 The WebRTC project authors. All Rights Reserved. |
| * |
| * Use of this source code is governed by a BSD-style license |
| * that can be found in the LICENSE file in the root of the source |
| * tree. An additional intellectual property rights grant can be found |
| * in the file PATENTS. All contributing project authors may |
| * be found in the AUTHORS file in the root of the source tree. |
| */ |
| |
| #ifndef WEBRTC_MODULES_RTP_RTCP_SOURCE_RTP_RTCP_IMPL_H_ |
| #define WEBRTC_MODULES_RTP_RTCP_SOURCE_RTP_RTCP_IMPL_H_ |
| |
| #include <list> |
| |
| #include "modules/rtp_rtcp/interface/rtp_rtcp.h" |
| #include "modules/rtp_rtcp/source/rtcp_receiver.h" |
| #include "modules/rtp_rtcp/source/rtcp_sender.h" |
| #include "modules/rtp_rtcp/source/rtp_receiver.h" |
| #include "modules/rtp_rtcp/source/rtp_sender.h" |
| #include "system_wrappers/interface/scoped_ptr.h" |
| |
| #ifdef MATLAB |
| class MatlabPlot; |
| #endif |
| |
| namespace webrtc { |
| |
| class ModuleRtpRtcpImpl : public RtpRtcp { |
| public: |
| explicit ModuleRtpRtcpImpl(const RtpRtcp::Configuration& configuration); |
| |
| virtual ~ModuleRtpRtcpImpl(); |
| |
| // returns the number of milliseconds until the module want a worker thread to call Process |
| virtual WebRtc_Word32 TimeUntilNextProcess(); |
| |
| // Process any pending tasks such as timeouts |
| virtual WebRtc_Word32 Process(); |
| |
| /** |
| * Receiver |
| */ |
| // configure a timeout value |
| virtual WebRtc_Word32 SetPacketTimeout(const WebRtc_UWord32 RTPtimeoutMS, |
| const WebRtc_UWord32 RTCPtimeoutMS); |
| |
| // Set periodic dead or alive notification |
| virtual WebRtc_Word32 SetPeriodicDeadOrAliveStatus( |
| const bool enable, |
| const WebRtc_UWord8 sampleTimeSeconds); |
| |
| // Get periodic dead or alive notification status |
| virtual WebRtc_Word32 PeriodicDeadOrAliveStatus( |
| bool &enable, |
| WebRtc_UWord8 &sampleTimeSeconds); |
| |
| virtual WebRtc_Word32 RegisterReceivePayload(const CodecInst& voiceCodec); |
| |
| virtual WebRtc_Word32 RegisterReceivePayload(const VideoCodec& videoCodec); |
| |
| virtual WebRtc_Word32 ReceivePayloadType(const CodecInst& voiceCodec, |
| WebRtc_Word8* plType); |
| |
| virtual WebRtc_Word32 ReceivePayloadType(const VideoCodec& videoCodec, |
| WebRtc_Word8* plType); |
| |
| virtual WebRtc_Word32 DeRegisterReceivePayload( |
| const WebRtc_Word8 payloadType); |
| |
| // register RTP header extension |
| virtual WebRtc_Word32 RegisterReceiveRtpHeaderExtension( |
| const RTPExtensionType type, |
| const WebRtc_UWord8 id); |
| |
| virtual WebRtc_Word32 DeregisterReceiveRtpHeaderExtension( |
| const RTPExtensionType type); |
| |
| // get the currently configured SSRC filter |
| virtual WebRtc_Word32 SSRCFilter(WebRtc_UWord32& allowedSSRC) const; |
| |
| // set a SSRC to be used as a filter for incoming RTP streams |
| virtual WebRtc_Word32 SetSSRCFilter(const bool enable, const WebRtc_UWord32 allowedSSRC); |
| |
| // Get last received remote timestamp |
| virtual WebRtc_UWord32 RemoteTimestamp() const; |
| |
| // Get the local time of the last received remote timestamp. |
| virtual int64_t LocalTimeOfRemoteTimeStamp() const; |
| |
| // Get the current estimated remote timestamp |
| virtual WebRtc_Word32 EstimatedRemoteTimeStamp(WebRtc_UWord32& timestamp) const; |
| |
| virtual WebRtc_UWord32 RemoteSSRC() const; |
| |
| virtual WebRtc_Word32 RemoteCSRCs( WebRtc_UWord32 arrOfCSRC[kRtpCsrcSize]) const ; |
| |
| virtual WebRtc_Word32 SetRTXReceiveStatus(const bool enable, |
| const WebRtc_UWord32 SSRC); |
| |
| virtual WebRtc_Word32 RTXReceiveStatus(bool* enable, |
| WebRtc_UWord32* SSRC) const; |
| |
| // called by the network module when we receive a packet |
| virtual WebRtc_Word32 IncomingPacket( const WebRtc_UWord8* incomingPacket, |
| const WebRtc_UWord16 packetLength); |
| |
| /** |
| * Sender |
| */ |
| virtual WebRtc_Word32 RegisterSendPayload(const CodecInst& voiceCodec); |
| |
| virtual WebRtc_Word32 RegisterSendPayload(const VideoCodec& videoCodec); |
| |
| virtual WebRtc_Word32 DeRegisterSendPayload(const WebRtc_Word8 payloadType); |
| |
| virtual WebRtc_Word8 SendPayloadType() const; |
| |
| // register RTP header extension |
| virtual WebRtc_Word32 RegisterSendRtpHeaderExtension( |
| const RTPExtensionType type, |
| const WebRtc_UWord8 id); |
| |
| virtual WebRtc_Word32 DeregisterSendRtpHeaderExtension( |
| const RTPExtensionType type); |
| |
| virtual void SetTransmissionSmoothingStatus(const bool enable); |
| |
| virtual bool TransmissionSmoothingStatus() const; |
| |
| // get start timestamp |
| virtual WebRtc_UWord32 StartTimestamp() const; |
| |
| // configure start timestamp, default is a random number |
| virtual WebRtc_Word32 SetStartTimestamp(const WebRtc_UWord32 timestamp); |
| |
| virtual WebRtc_UWord16 SequenceNumber() const; |
| |
| // Set SequenceNumber, default is a random number |
| virtual WebRtc_Word32 SetSequenceNumber(const WebRtc_UWord16 seq); |
| |
| virtual WebRtc_UWord32 SSRC() const; |
| |
| // configure SSRC, default is a random number |
| virtual WebRtc_Word32 SetSSRC(const WebRtc_UWord32 ssrc); |
| |
| virtual WebRtc_Word32 CSRCs( WebRtc_UWord32 arrOfCSRC[kRtpCsrcSize]) const ; |
| |
| virtual WebRtc_Word32 SetCSRCs( const WebRtc_UWord32 arrOfCSRC[kRtpCsrcSize], |
| const WebRtc_UWord8 arrLength); |
| |
| virtual WebRtc_Word32 SetCSRCStatus(const bool include); |
| |
| virtual WebRtc_UWord32 PacketCountSent() const; |
| |
| virtual int CurrentSendFrequencyHz() const; |
| |
| virtual WebRtc_UWord32 ByteCountSent() const; |
| |
| virtual WebRtc_Word32 SetRTXSendStatus(const bool enable, |
| const bool setSSRC, |
| const WebRtc_UWord32 SSRC); |
| |
| virtual WebRtc_Word32 RTXSendStatus(bool* enable, |
| WebRtc_UWord32* SSRC) const; |
| |
| // sends kRtcpByeCode when going from true to false |
| virtual WebRtc_Word32 SetSendingStatus(const bool sending); |
| |
| virtual bool Sending() const; |
| |
| // Drops or relays media packets |
| virtual WebRtc_Word32 SetSendingMediaStatus(const bool sending); |
| |
| virtual bool SendingMedia() const; |
| |
| // Used by the codec module to deliver a video or audio frame for packetization |
| virtual WebRtc_Word32 SendOutgoingData( |
| const FrameType frameType, |
| const WebRtc_Word8 payloadType, |
| const WebRtc_UWord32 timeStamp, |
| int64_t capture_time_ms, |
| const WebRtc_UWord8* payloadData, |
| const WebRtc_UWord32 payloadSize, |
| const RTPFragmentationHeader* fragmentation = NULL, |
| const RTPVideoHeader* rtpVideoHdr = NULL); |
| |
| /* |
| * RTCP |
| */ |
| |
| // Get RTCP status |
| virtual RTCPMethod RTCP() const; |
| |
| // configure RTCP status i.e on/off |
| virtual WebRtc_Word32 SetRTCPStatus(const RTCPMethod method); |
| |
| // Set RTCP CName |
| virtual WebRtc_Word32 SetCNAME(const char cName[RTCP_CNAME_SIZE]); |
| |
| // Get RTCP CName |
| virtual WebRtc_Word32 CNAME(char cName[RTCP_CNAME_SIZE]); |
| |
| // Get remote CName |
| virtual WebRtc_Word32 RemoteCNAME(const WebRtc_UWord32 remoteSSRC, |
| char cName[RTCP_CNAME_SIZE]) const; |
| |
| // Get remote NTP |
| virtual WebRtc_Word32 RemoteNTP(WebRtc_UWord32 *ReceivedNTPsecs, |
| WebRtc_UWord32 *ReceivedNTPfrac, |
| WebRtc_UWord32 *RTCPArrivalTimeSecs, |
| WebRtc_UWord32 *RTCPArrivalTimeFrac, |
| WebRtc_UWord32 *rtcp_timestamp) const; |
| |
| virtual WebRtc_Word32 AddMixedCNAME(const WebRtc_UWord32 SSRC, |
| const char cName[RTCP_CNAME_SIZE]); |
| |
| virtual WebRtc_Word32 RemoveMixedCNAME(const WebRtc_UWord32 SSRC); |
| |
| // Get RoundTripTime |
| virtual WebRtc_Word32 RTT(const WebRtc_UWord32 remoteSSRC, |
| WebRtc_UWord16* RTT, |
| WebRtc_UWord16* avgRTT, |
| WebRtc_UWord16* minRTT, |
| WebRtc_UWord16* maxRTT) const; |
| |
| // Reset RoundTripTime statistics |
| virtual WebRtc_Word32 ResetRTT(const WebRtc_UWord32 remoteSSRC); |
| |
| // Force a send of an RTCP packet |
| // normal SR and RR are triggered via the process function |
| virtual WebRtc_Word32 SendRTCP(WebRtc_UWord32 rtcpPacketType = kRtcpReport); |
| |
| // statistics of our localy created statistics of the received RTP stream |
| virtual WebRtc_Word32 StatisticsRTP(WebRtc_UWord8 *fraction_lost, |
| WebRtc_UWord32 *cum_lost, |
| WebRtc_UWord32 *ext_max, |
| WebRtc_UWord32 *jitter, |
| WebRtc_UWord32 *max_jitter = NULL) const; |
| |
| // Reset RTP statistics |
| virtual WebRtc_Word32 ResetStatisticsRTP(); |
| |
| virtual WebRtc_Word32 ResetReceiveDataCountersRTP(); |
| |
| virtual WebRtc_Word32 ResetSendDataCountersRTP(); |
| |
| // statistics of the amount of data sent and received |
| virtual WebRtc_Word32 DataCountersRTP(WebRtc_UWord32 *bytesSent, |
| WebRtc_UWord32 *packetsSent, |
| WebRtc_UWord32 *bytesReceived, |
| WebRtc_UWord32 *packetsReceived) const; |
| |
| virtual WebRtc_Word32 ReportBlockStatistics( |
| WebRtc_UWord8 *fraction_lost, |
| WebRtc_UWord32 *cum_lost, |
| WebRtc_UWord32 *ext_max, |
| WebRtc_UWord32 *jitter, |
| WebRtc_UWord32 *jitter_transmission_time_offset); |
| |
| // Get received RTCP report, sender info |
| virtual WebRtc_Word32 RemoteRTCPStat( RTCPSenderInfo* senderInfo); |
| |
| // Get received RTCP report, report block |
| virtual WebRtc_Word32 RemoteRTCPStat( |
| std::vector<RTCPReportBlock>* receiveBlocks) const; |
| |
| // Set received RTCP report block |
| virtual WebRtc_Word32 AddRTCPReportBlock(const WebRtc_UWord32 SSRC, |
| const RTCPReportBlock* receiveBlock); |
| |
| virtual WebRtc_Word32 RemoveRTCPReportBlock(const WebRtc_UWord32 SSRC); |
| |
| /* |
| * (REMB) Receiver Estimated Max Bitrate |
| */ |
| virtual bool REMB() const; |
| |
| virtual WebRtc_Word32 SetREMBStatus(const bool enable); |
| |
| virtual WebRtc_Word32 SetREMBData(const WebRtc_UWord32 bitrate, |
| const WebRtc_UWord8 numberOfSSRC, |
| const WebRtc_UWord32* SSRC); |
| |
| /* |
| * (IJ) Extended jitter report. |
| */ |
| virtual bool IJ() const; |
| |
| virtual WebRtc_Word32 SetIJStatus(const bool enable); |
| |
| /* |
| * (TMMBR) Temporary Max Media Bit Rate |
| */ |
| virtual bool TMMBR() const ; |
| |
| virtual WebRtc_Word32 SetTMMBRStatus(const bool enable); |
| |
| WebRtc_Word32 SetTMMBN(const TMMBRSet* boundingSet); |
| |
| virtual WebRtc_UWord16 MaxPayloadLength() const; |
| |
| virtual WebRtc_UWord16 MaxDataPayloadLength() const; |
| |
| virtual WebRtc_Word32 SetMaxTransferUnit(const WebRtc_UWord16 size); |
| |
| virtual WebRtc_Word32 SetTransportOverhead(const bool TCP, |
| const bool IPV6, |
| const WebRtc_UWord8 authenticationOverhead = 0); |
| |
| /* |
| * (NACK) Negative acknowledgement |
| */ |
| |
| // Is Negative acknowledgement requests on/off? |
| virtual NACKMethod NACK() const ; |
| |
| // Turn negative acknowledgement requests on/off |
| virtual WebRtc_Word32 SetNACKStatus(const NACKMethod method); |
| |
| virtual int SelectiveRetransmissions() const; |
| |
| virtual int SetSelectiveRetransmissions(uint8_t settings); |
| |
| // Send a Negative acknowledgement packet |
| virtual WebRtc_Word32 SendNACK(const WebRtc_UWord16* nackList, |
| const WebRtc_UWord16 size); |
| |
| // Store the sent packets, needed to answer to a Negative acknowledgement requests |
| virtual WebRtc_Word32 SetStorePacketsStatus(const bool enable, const WebRtc_UWord16 numberToStore = 200); |
| |
| /* |
| * (APP) Application specific data |
| */ |
| virtual WebRtc_Word32 SetRTCPApplicationSpecificData(const WebRtc_UWord8 subType, |
| const WebRtc_UWord32 name, |
| const WebRtc_UWord8* data, |
| const WebRtc_UWord16 length); |
| /* |
| * (XR) VOIP metric |
| */ |
| virtual WebRtc_Word32 SetRTCPVoIPMetrics(const RTCPVoIPMetric* VoIPMetric); |
| |
| /* |
| * Audio |
| */ |
| |
| // set audio packet size, used to determine when it's time to send a DTMF packet in silence (CNG) |
| virtual WebRtc_Word32 SetAudioPacketSize(const WebRtc_UWord16 packetSizeSamples); |
| |
| // Outband DTMF detection |
| virtual WebRtc_Word32 SetTelephoneEventStatus(const bool enable, |
| const bool forwardToDecoder, |
| const bool detectEndOfTone = false); |
| |
| // Is outband DTMF turned on/off? |
| virtual bool TelephoneEvent() const; |
| |
| // Is forwarding of outband telephone events turned on/off? |
| virtual bool TelephoneEventForwardToDecoder() const; |
| |
| virtual bool SendTelephoneEventActive(WebRtc_Word8& telephoneEvent) const; |
| |
| // Send a TelephoneEvent tone using RFC 2833 (4733) |
| virtual WebRtc_Word32 SendTelephoneEventOutband(const WebRtc_UWord8 key, |
| const WebRtc_UWord16 time_ms, |
| const WebRtc_UWord8 level); |
| |
| // Set payload type for Redundant Audio Data RFC 2198 |
| virtual WebRtc_Word32 SetSendREDPayloadType(const WebRtc_Word8 payloadType); |
| |
| // Get payload type for Redundant Audio Data RFC 2198 |
| virtual WebRtc_Word32 SendREDPayloadType(WebRtc_Word8& payloadType) const; |
| |
| // Set status and ID for header-extension-for-audio-level-indication. |
| virtual WebRtc_Word32 SetRTPAudioLevelIndicationStatus(const bool enable, |
| const WebRtc_UWord8 ID); |
| |
| // Get status and ID for header-extension-for-audio-level-indication. |
| virtual WebRtc_Word32 GetRTPAudioLevelIndicationStatus(bool& enable, |
| WebRtc_UWord8& ID) const; |
| |
| // Store the audio level in dBov for header-extension-for-audio-level-indication. |
| virtual WebRtc_Word32 SetAudioLevel(const WebRtc_UWord8 level_dBov); |
| |
| /* |
| * Video |
| */ |
| virtual RtpVideoCodecTypes ReceivedVideoCodec() const; |
| |
| virtual RtpVideoCodecTypes SendVideoCodec() const; |
| |
| virtual WebRtc_Word32 SendRTCPSliceLossIndication(const WebRtc_UWord8 pictureID); |
| |
| // Set method for requestion a new key frame |
| virtual WebRtc_Word32 SetKeyFrameRequestMethod(const KeyFrameRequestMethod method); |
| |
| // send a request for a keyframe |
| virtual WebRtc_Word32 RequestKeyFrame(); |
| |
| virtual WebRtc_Word32 SetCameraDelay(const WebRtc_Word32 delayMS); |
| |
| virtual void SetTargetSendBitrate(const WebRtc_UWord32 bitrate); |
| |
| virtual WebRtc_Word32 SetGenericFECStatus(const bool enable, |
| const WebRtc_UWord8 payloadTypeRED, |
| const WebRtc_UWord8 payloadTypeFEC); |
| |
| virtual WebRtc_Word32 GenericFECStatus(bool& enable, |
| WebRtc_UWord8& payloadTypeRED, |
| WebRtc_UWord8& payloadTypeFEC); |
| |
| virtual WebRtc_Word32 SetFecParameters( |
| const FecProtectionParams* delta_params, |
| const FecProtectionParams* key_params); |
| |
| virtual WebRtc_Word32 LastReceivedNTP(WebRtc_UWord32& NTPsecs, |
| WebRtc_UWord32& NTPfrac, |
| WebRtc_UWord32& remoteSR); |
| |
| virtual WebRtc_Word32 BoundingSet(bool &tmmbrOwner, |
| TMMBRSet*& boundingSetRec); |
| |
| virtual void BitrateSent(WebRtc_UWord32* totalRate, |
| WebRtc_UWord32* videoRate, |
| WebRtc_UWord32* fecRate, |
| WebRtc_UWord32* nackRate) const; |
| |
| virtual int EstimatedReceiveBandwidth( |
| WebRtc_UWord32* available_bandwidth) const; |
| |
| virtual void SetRemoteSSRC(const WebRtc_UWord32 SSRC); |
| |
| virtual WebRtc_UWord32 SendTimeOfSendReport(const WebRtc_UWord32 sendReport); |
| |
| // good state of RTP receiver inform sender |
| virtual WebRtc_Word32 SendRTCPReferencePictureSelection(const WebRtc_UWord64 pictureID); |
| |
| void OnReceivedTMMBR(); |
| |
| // bad state of RTP receiver request a keyframe |
| void OnRequestIntraFrame(); |
| |
| // received a request for a new SLI |
| void OnReceivedSliceLossIndication(const WebRtc_UWord8 pictureID); |
| |
| // received a new refereence frame |
| void OnReceivedReferencePictureSelectionIndication( |
| const WebRtc_UWord64 pitureID); |
| |
| void OnReceivedNACK(const WebRtc_UWord16 nackSequenceNumbersLength, |
| const WebRtc_UWord16* nackSequenceNumbers); |
| |
| void OnRequestSendReport(); |
| |
| // Following function is only called when constructing the object so no |
| // need to worry about data race. |
| void OwnsClock() { _owns_clock = true; } |
| |
| protected: |
| void RegisterChildModule(RtpRtcp* module); |
| |
| void DeRegisterChildModule(RtpRtcp* module); |
| |
| bool UpdateRTCPReceiveInformationTimers(); |
| |
| void ProcessDeadOrAliveTimer(); |
| |
| WebRtc_UWord32 BitrateReceivedNow() const; |
| |
| // Get remote SequenceNumber |
| WebRtc_UWord16 RemoteSequenceNumber() const; |
| |
| // only for internal testing |
| WebRtc_UWord32 LastSendReport(WebRtc_UWord32& lastRTCPTime); |
| |
| RTPSender _rtpSender; |
| RTPReceiver _rtpReceiver; |
| |
| RTCPSender _rtcpSender; |
| RTCPReceiver _rtcpReceiver; |
| |
| bool _owns_clock; |
| RtpRtcpClock& _clock; |
| private: |
| WebRtc_Word32 _id; |
| const bool _audio; |
| bool _collisionDetected; |
| WebRtc_Word64 _lastProcessTime; |
| WebRtc_Word64 _lastBitrateProcessTime; |
| WebRtc_Word64 _lastPacketTimeoutProcessTime; |
| WebRtc_UWord16 _packetOverHead; |
| |
| scoped_ptr<CriticalSectionWrapper> _criticalSectionModulePtrs; |
| scoped_ptr<CriticalSectionWrapper> _criticalSectionModulePtrsFeedback; |
| ModuleRtpRtcpImpl* _defaultModule; |
| std::list<ModuleRtpRtcpImpl*> _childModules; |
| |
| // Dead or alive |
| bool _deadOrAliveActive; |
| WebRtc_UWord32 _deadOrAliveTimeoutMS; |
| WebRtc_Word64 _deadOrAliveLastTimer; |
| // send side |
| NACKMethod _nackMethod; |
| WebRtc_UWord32 _nackLastTimeSent; |
| WebRtc_UWord16 _nackLastSeqNumberSent; |
| |
| bool _simulcast; |
| VideoCodec _sendVideoCodec; |
| KeyFrameRequestMethod _keyFrameReqMethod; |
| |
| RemoteBitrateEstimator* remote_bitrate_; |
| |
| #ifdef MATLAB |
| MatlabPlot* _plot1; |
| #endif |
| }; |
| } // namespace webrtc |
| #endif // WEBRTC_MODULES_RTP_RTCP_SOURCE_RTP_RTCP_IMPL_H_ |