c9bbeb0354
Reason for revert: Breaking some Android bots. https://chromegw.corp.google.com/i/client.webrtc/builders/Android32%20Tests%20%28L%20Nexus5%29 Original issue's description: > Wire up send-side bandwidth estimation. > > BUG=webrtc:4173 > > Committed: https://crrev.com/ef165eefc79cf28bb67779afe303cc2365885547 > Cr-Commit-Position: refs/heads/master@{#10012} TBR=stefan@webrtc.org, kjellander@webrtc.org NOPRESUBMIT=false NOTREECHECKS=false NOTRY=false BUG=webrtc:4173 Review URL: https://codereview.webrtc.org/1362923002 . Cr-Commit-Position: refs/heads/master@{#10029}
131 lines
3.3 KiB
C++
131 lines
3.3 KiB
C++
/*
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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_RTCP_RECEIVER_HELP_H_
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#define WEBRTC_MODULES_RTP_RTCP_SOURCE_RTCP_RECEIVER_HELP_H_
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#include "webrtc/base/constructormagic.h"
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#include "webrtc/base/scoped_ptr.h"
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#include "webrtc/modules/rtp_rtcp/interface/rtp_rtcp_defines.h" // RTCPReportBlock
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#include "webrtc/modules/rtp_rtcp/source/rtcp_utility.h"
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#include "webrtc/modules/rtp_rtcp/source/tmmbr_help.h"
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#include "webrtc/typedefs.h"
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namespace webrtc {
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namespace RTCPHelp
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{
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class RTCPReportBlockInformation
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{
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public:
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RTCPReportBlockInformation();
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~RTCPReportBlockInformation();
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// Statistics
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RTCPReportBlock remoteReceiveBlock;
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uint32_t remoteMaxJitter;
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// RTT
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int64_t RTT;
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int64_t minRTT;
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int64_t maxRTT;
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int64_t avgRTT;
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uint32_t numAverageCalcs;
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};
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class RTCPPacketInformation
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{
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public:
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RTCPPacketInformation();
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~RTCPPacketInformation();
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void AddVoIPMetric(const RTCPVoIPMetric* metric);
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void AddApplicationData(const uint8_t* data,
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const uint16_t size);
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void AddNACKPacket(const uint16_t packetID);
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void ResetNACKPacketIdArray();
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void AddReportInfo(const RTCPReportBlockInformation& report_block_info);
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uint32_t rtcpPacketTypeFlags; // RTCPPacketTypeFlags bit field
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uint32_t remoteSSRC;
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std::list<uint16_t> nackSequenceNumbers;
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uint8_t applicationSubType;
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uint32_t applicationName;
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uint8_t* applicationData;
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uint16_t applicationLength;
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ReportBlockList report_blocks;
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int64_t rtt;
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uint32_t interArrivalJitter;
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uint8_t sliPictureId;
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uint64_t rpsiPictureId;
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uint32_t receiverEstimatedMaxBitrate;
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uint32_t ntp_secs;
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uint32_t ntp_frac;
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uint32_t rtp_timestamp;
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uint32_t xr_originator_ssrc;
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bool xr_dlrr_item;
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RTCPVoIPMetric* VoIPMetric;
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private:
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RTC_DISALLOW_COPY_AND_ASSIGN(RTCPPacketInformation);
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};
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class RTCPReceiveInformation
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{
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public:
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RTCPReceiveInformation();
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~RTCPReceiveInformation();
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void VerifyAndAllocateBoundingSet(const uint32_t minimumSize);
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void VerifyAndAllocateTMMBRSet(const uint32_t minimumSize);
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void InsertTMMBRItem(const uint32_t senderSSRC,
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const RTCPUtility::RTCPPacketRTPFBTMMBRItem& TMMBRItem,
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const int64_t currentTimeMS);
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// get
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int32_t GetTMMBRSet(const uint32_t sourceIdx,
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const uint32_t targetIdx,
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TMMBRSet* candidateSet,
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const int64_t currentTimeMS);
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int64_t lastTimeReceived;
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// FIR
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int32_t lastFIRSequenceNumber;
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int64_t lastFIRRequest;
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// TMMBN
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TMMBRSet TmmbnBoundingSet;
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// TMMBR
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TMMBRSet TmmbrSet;
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bool readyForDelete;
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private:
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std::vector<int64_t> _tmmbrSetTimeouts;
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};
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} // end namespace RTCPHelp
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} // namespace webrtc
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#endif // WEBRTC_MODULES_RTP_RTCP_SOURCE_RTCP_RECEIVER_HELP_H_
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