Move src/ -> webrtc/
TBR=niklas.enbom@webrtc.org Review URL: https://webrtc-codereview.appspot.com/915006 git-svn-id: http://webrtc.googlecode.com/svn/trunk@2963 4adac7df-926f-26a2-2b94-8c16560cd09d
This commit is contained in:
andrew@webrtc.org
519
webrtc/modules/rtp_rtcp/source/rtp_rtcp_impl.h
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519
webrtc/modules/rtp_rtcp/source/rtp_rtcp_impl.h
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@ -0,0 +1,519 @@
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/*
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* Copyright (c) 2012 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_RTCP_IMPL_H_
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#define WEBRTC_MODULES_RTP_RTCP_SOURCE_RTP_RTCP_IMPL_H_
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#include <list>
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#include "modules/rtp_rtcp/interface/rtp_rtcp.h"
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#include "modules/rtp_rtcp/source/rtcp_receiver.h"
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#include "modules/rtp_rtcp/source/rtcp_sender.h"
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#include "modules/rtp_rtcp/source/rtp_receiver.h"
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#include "modules/rtp_rtcp/source/rtp_sender.h"
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#include "system_wrappers/interface/scoped_ptr.h"
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#ifdef MATLAB
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class MatlabPlot;
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#endif
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namespace webrtc {
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class ModuleRtpRtcpImpl : public RtpRtcp {
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public:
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explicit ModuleRtpRtcpImpl(const RtpRtcp::Configuration& configuration);
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virtual ~ModuleRtpRtcpImpl();
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// returns the number of milliseconds until the module want a worker thread to call Process
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virtual WebRtc_Word32 TimeUntilNextProcess();
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// Process any pending tasks such as timeouts
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virtual WebRtc_Word32 Process();
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/**
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* Receiver
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*/
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// configure a timeout value
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virtual WebRtc_Word32 SetPacketTimeout(const WebRtc_UWord32 RTPtimeoutMS,
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const WebRtc_UWord32 RTCPtimeoutMS);
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// Set periodic dead or alive notification
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virtual WebRtc_Word32 SetPeriodicDeadOrAliveStatus(
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const bool enable,
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const WebRtc_UWord8 sampleTimeSeconds);
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// Get periodic dead or alive notification status
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virtual WebRtc_Word32 PeriodicDeadOrAliveStatus(
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bool &enable,
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WebRtc_UWord8 &sampleTimeSeconds);
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virtual WebRtc_Word32 RegisterReceivePayload(const CodecInst& voiceCodec);
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virtual WebRtc_Word32 RegisterReceivePayload(const VideoCodec& videoCodec);
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virtual WebRtc_Word32 ReceivePayloadType(const CodecInst& voiceCodec,
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WebRtc_Word8* plType);
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virtual WebRtc_Word32 ReceivePayloadType(const VideoCodec& videoCodec,
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WebRtc_Word8* plType);
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virtual WebRtc_Word32 DeRegisterReceivePayload(
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const WebRtc_Word8 payloadType);
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// register RTP header extension
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virtual WebRtc_Word32 RegisterReceiveRtpHeaderExtension(
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const RTPExtensionType type,
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const WebRtc_UWord8 id);
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virtual WebRtc_Word32 DeregisterReceiveRtpHeaderExtension(
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const RTPExtensionType type);
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// get the currently configured SSRC filter
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virtual 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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virtual WebRtc_Word32 SetSSRCFilter(const bool enable, const WebRtc_UWord32 allowedSSRC);
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// Get last received remote timestamp
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virtual WebRtc_UWord32 RemoteTimestamp() const;
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// Get the local time of the last received remote timestamp.
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virtual int64_t LocalTimeOfRemoteTimeStamp() const;
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// Get the current estimated remote timestamp
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virtual WebRtc_Word32 EstimatedRemoteTimeStamp(WebRtc_UWord32& timestamp) const;
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virtual WebRtc_UWord32 RemoteSSRC() const;
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virtual WebRtc_Word32 RemoteCSRCs( WebRtc_UWord32 arrOfCSRC[kRtpCsrcSize]) const ;
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virtual WebRtc_Word32 SetRTXReceiveStatus(const bool enable,
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const WebRtc_UWord32 SSRC);
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virtual WebRtc_Word32 RTXReceiveStatus(bool* enable,
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WebRtc_UWord32* SSRC) const;
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// called by the network module when we receive a packet
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virtual WebRtc_Word32 IncomingPacket( const WebRtc_UWord8* incomingPacket,
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const WebRtc_UWord16 packetLength);
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/**
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* Sender
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*/
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virtual WebRtc_Word32 RegisterSendPayload(const CodecInst& voiceCodec);
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virtual WebRtc_Word32 RegisterSendPayload(const VideoCodec& videoCodec);
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virtual WebRtc_Word32 DeRegisterSendPayload(const WebRtc_Word8 payloadType);
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virtual WebRtc_Word8 SendPayloadType() const;
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// register RTP header extension
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virtual WebRtc_Word32 RegisterSendRtpHeaderExtension(
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const RTPExtensionType type,
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const WebRtc_UWord8 id);
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virtual WebRtc_Word32 DeregisterSendRtpHeaderExtension(
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const RTPExtensionType type);
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virtual void SetTransmissionSmoothingStatus(const bool enable);
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virtual bool TransmissionSmoothingStatus() const;
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// get start timestamp
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virtual WebRtc_UWord32 StartTimestamp() const;
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// configure start timestamp, default is a random number
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virtual WebRtc_Word32 SetStartTimestamp(const WebRtc_UWord32 timestamp);
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virtual WebRtc_UWord16 SequenceNumber() const;
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// Set SequenceNumber, default is a random number
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virtual WebRtc_Word32 SetSequenceNumber(const WebRtc_UWord16 seq);
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virtual WebRtc_UWord32 SSRC() const;
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// configure SSRC, default is a random number
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virtual WebRtc_Word32 SetSSRC(const WebRtc_UWord32 ssrc);
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virtual WebRtc_Word32 CSRCs( WebRtc_UWord32 arrOfCSRC[kRtpCsrcSize]) const ;
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virtual WebRtc_Word32 SetCSRCs( const WebRtc_UWord32 arrOfCSRC[kRtpCsrcSize],
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const WebRtc_UWord8 arrLength);
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virtual WebRtc_Word32 SetCSRCStatus(const bool include);
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virtual WebRtc_UWord32 PacketCountSent() const;
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virtual int CurrentSendFrequencyHz() const;
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virtual WebRtc_UWord32 ByteCountSent() const;
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virtual WebRtc_Word32 SetRTXSendStatus(const bool enable,
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const bool setSSRC,
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const WebRtc_UWord32 SSRC);
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virtual WebRtc_Word32 RTXSendStatus(bool* enable,
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WebRtc_UWord32* SSRC) const;
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// sends kRtcpByeCode when going from true to false
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virtual WebRtc_Word32 SetSendingStatus(const bool sending);
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virtual bool Sending() const;
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// Drops or relays media packets
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virtual WebRtc_Word32 SetSendingMediaStatus(const bool sending);
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virtual bool SendingMedia() const;
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// Used by the codec module to deliver a video or audio frame for packetization
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virtual WebRtc_Word32 SendOutgoingData(
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const FrameType frameType,
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const WebRtc_Word8 payloadType,
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const WebRtc_UWord32 timeStamp,
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int64_t capture_time_ms,
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const WebRtc_UWord8* payloadData,
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const WebRtc_UWord32 payloadSize,
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const RTPFragmentationHeader* fragmentation = NULL,
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const RTPVideoHeader* rtpVideoHdr = NULL);
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/*
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* RTCP
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*/
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// Get RTCP status
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virtual RTCPMethod RTCP() const;
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// configure RTCP status i.e on/off
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virtual WebRtc_Word32 SetRTCPStatus(const RTCPMethod method);
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// Set RTCP CName
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virtual WebRtc_Word32 SetCNAME(const char cName[RTCP_CNAME_SIZE]);
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// Get RTCP CName
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virtual WebRtc_Word32 CNAME(char cName[RTCP_CNAME_SIZE]);
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// Get remote CName
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virtual WebRtc_Word32 RemoteCNAME(const WebRtc_UWord32 remoteSSRC,
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char cName[RTCP_CNAME_SIZE]) const;
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// Get remote NTP
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virtual WebRtc_Word32 RemoteNTP(WebRtc_UWord32 *ReceivedNTPsecs,
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WebRtc_UWord32 *ReceivedNTPfrac,
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WebRtc_UWord32 *RTCPArrivalTimeSecs,
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WebRtc_UWord32 *RTCPArrivalTimeFrac,
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WebRtc_UWord32 *rtcp_timestamp) const;
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virtual WebRtc_Word32 AddMixedCNAME(const WebRtc_UWord32 SSRC,
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const char cName[RTCP_CNAME_SIZE]);
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virtual WebRtc_Word32 RemoveMixedCNAME(const WebRtc_UWord32 SSRC);
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// Get RoundTripTime
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virtual WebRtc_Word32 RTT(const WebRtc_UWord32 remoteSSRC,
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WebRtc_UWord16* RTT,
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WebRtc_UWord16* avgRTT,
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WebRtc_UWord16* minRTT,
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WebRtc_UWord16* maxRTT) const;
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// Reset RoundTripTime statistics
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virtual WebRtc_Word32 ResetRTT(const WebRtc_UWord32 remoteSSRC);
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// Force a send of an RTCP packet
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// normal SR and RR are triggered via the process function
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virtual WebRtc_Word32 SendRTCP(WebRtc_UWord32 rtcpPacketType = kRtcpReport);
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// statistics of our localy created statistics of the received RTP stream
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virtual WebRtc_Word32 StatisticsRTP(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,
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WebRtc_UWord32 *max_jitter = NULL) const;
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// Reset RTP statistics
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virtual WebRtc_Word32 ResetStatisticsRTP();
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virtual WebRtc_Word32 ResetReceiveDataCountersRTP();
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virtual WebRtc_Word32 ResetSendDataCountersRTP();
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// statistics of the amount of data sent and received
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virtual WebRtc_Word32 DataCountersRTP(WebRtc_UWord32 *bytesSent,
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WebRtc_UWord32 *packetsSent,
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WebRtc_UWord32 *bytesReceived,
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WebRtc_UWord32 *packetsReceived) const;
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virtual WebRtc_Word32 ReportBlockStatistics(
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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,
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WebRtc_UWord32 *jitter_transmission_time_offset);
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// Get received RTCP report, sender info
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virtual WebRtc_Word32 RemoteRTCPStat( RTCPSenderInfo* senderInfo);
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// Get received RTCP report, report block
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virtual WebRtc_Word32 RemoteRTCPStat(
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std::vector<RTCPReportBlock>* receiveBlocks) const;
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// Set received RTCP report block
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virtual WebRtc_Word32 AddRTCPReportBlock(const WebRtc_UWord32 SSRC,
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const RTCPReportBlock* receiveBlock);
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virtual WebRtc_Word32 RemoveRTCPReportBlock(const WebRtc_UWord32 SSRC);
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/*
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* (REMB) Receiver Estimated Max Bitrate
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*/
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virtual bool REMB() const;
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virtual WebRtc_Word32 SetREMBStatus(const bool enable);
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virtual WebRtc_Word32 SetREMBData(const WebRtc_UWord32 bitrate,
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const WebRtc_UWord8 numberOfSSRC,
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const WebRtc_UWord32* SSRC);
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/*
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* (IJ) Extended jitter report.
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*/
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virtual bool IJ() const;
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virtual WebRtc_Word32 SetIJStatus(const bool enable);
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/*
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* (TMMBR) Temporary Max Media Bit Rate
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*/
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virtual bool TMMBR() const ;
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virtual WebRtc_Word32 SetTMMBRStatus(const bool enable);
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WebRtc_Word32 SetTMMBN(const TMMBRSet* boundingSet);
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virtual WebRtc_UWord16 MaxPayloadLength() const;
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virtual WebRtc_UWord16 MaxDataPayloadLength() const;
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virtual WebRtc_Word32 SetMaxTransferUnit(const WebRtc_UWord16 size);
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virtual WebRtc_Word32 SetTransportOverhead(const bool TCP,
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const bool IPV6,
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const WebRtc_UWord8 authenticationOverhead = 0);
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/*
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* (NACK) Negative acknowledgement
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*/
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// Is Negative acknowledgement requests on/off?
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virtual NACKMethod NACK() const ;
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// Turn negative acknowledgement requests on/off
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virtual WebRtc_Word32 SetNACKStatus(const NACKMethod method);
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virtual int SelectiveRetransmissions() const;
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virtual int SetSelectiveRetransmissions(uint8_t settings);
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// Send a Negative acknowledgement packet
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virtual WebRtc_Word32 SendNACK(const WebRtc_UWord16* nackList,
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const WebRtc_UWord16 size);
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// Store the sent packets, needed to answer to a Negative acknowledgement requests
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virtual WebRtc_Word32 SetStorePacketsStatus(const bool enable, const WebRtc_UWord16 numberToStore = 200);
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/*
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* (APP) Application specific data
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*/
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virtual WebRtc_Word32 SetRTCPApplicationSpecificData(const WebRtc_UWord8 subType,
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const WebRtc_UWord32 name,
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const WebRtc_UWord8* data,
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const WebRtc_UWord16 length);
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/*
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* (XR) VOIP metric
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*/
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||||
virtual WebRtc_Word32 SetRTCPVoIPMetrics(const RTCPVoIPMetric* VoIPMetric);
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||||
/*
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||||
* Audio
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*/
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// set audio packet size, used to determine when it's time to send a DTMF packet in silence (CNG)
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virtual WebRtc_Word32 SetAudioPacketSize(const WebRtc_UWord16 packetSizeSamples);
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// Outband DTMF detection
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virtual WebRtc_Word32 SetTelephoneEventStatus(const bool enable,
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const bool forwardToDecoder,
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const bool detectEndOfTone = false);
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||||
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// Is outband DTMF turned on/off?
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||||
virtual bool TelephoneEvent() const;
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||||
// Is forwarding of outband telephone events turned on/off?
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virtual bool TelephoneEventForwardToDecoder() const;
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||||
virtual bool SendTelephoneEventActive(WebRtc_Word8& telephoneEvent) const;
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||||
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// Send a TelephoneEvent tone using RFC 2833 (4733)
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virtual WebRtc_Word32 SendTelephoneEventOutband(const WebRtc_UWord8 key,
|
||||
const WebRtc_UWord16 time_ms,
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||||
const WebRtc_UWord8 level);
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||||
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// Set payload type for Redundant Audio Data RFC 2198
|
||||
virtual WebRtc_Word32 SetSendREDPayloadType(const WebRtc_Word8 payloadType);
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||||
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||||
// Get payload type for Redundant Audio Data RFC 2198
|
||||
virtual WebRtc_Word32 SendREDPayloadType(WebRtc_Word8& payloadType) const;
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||||
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||||
// Set status and ID for header-extension-for-audio-level-indication.
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||||
virtual WebRtc_Word32 SetRTPAudioLevelIndicationStatus(const bool enable,
|
||||
const WebRtc_UWord8 ID);
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||||
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||||
// Get status and ID for header-extension-for-audio-level-indication.
|
||||
virtual WebRtc_Word32 GetRTPAudioLevelIndicationStatus(bool& enable,
|
||||
WebRtc_UWord8& ID) const;
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||||
|
||||
// Store the audio level in dBov for header-extension-for-audio-level-indication.
|
||||
virtual WebRtc_Word32 SetAudioLevel(const WebRtc_UWord8 level_dBov);
|
||||
|
||||
/*
|
||||
* Video
|
||||
*/
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||||
virtual RtpVideoCodecTypes ReceivedVideoCodec() const;
|
||||
|
||||
virtual RtpVideoCodecTypes SendVideoCodec() const;
|
||||
|
||||
virtual WebRtc_Word32 SendRTCPSliceLossIndication(const WebRtc_UWord8 pictureID);
|
||||
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||||
// 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();
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||||
|
||||
virtual WebRtc_Word32 SetCameraDelay(const WebRtc_Word32 delayMS);
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||||
|
||||
virtual void SetTargetSendBitrate(const WebRtc_UWord32 bitrate);
|
||||
|
||||
virtual WebRtc_Word32 SetGenericFECStatus(const bool enable,
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||||
const WebRtc_UWord8 payloadTypeRED,
|
||||
const WebRtc_UWord8 payloadTypeFEC);
|
||||
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||||
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_
|
||||
Reference in New Issue
Block a user