ff761fba82
This changes the following module directories:
* webrtc/modules/audio_conference_mixer/interface
* webrtc/modules/interface
* webrtc/modules/media_file/interface
* webrtc/modules/rtp_rtcp/interface
* webrtc/modules/utility/interface
To avoid breaking downstream, I followed this recipe:
1. Copy the interface dir to a new sibling directory: include
2. Update the header guards in the include directory to match the style guide.
3. Update the header guards in the interface directory to match the ones in include. This is required to avoid getting redefinitions in the not-yet-updated downstream code.
4. Add a pragma warning in the header files in the interface dir. Example:
#pragma message("WARNING: webrtc/modules/interface is DEPRECATED; "
"use webrtc/modules/include")
5. Search for all source references to webrtc/modules/interface and update them to webrtc/modules/include (*.c*,*.h,*.mm,*.S)
6. Update all GYP+GN files. This required manual inspection since many subdirectories of webrtc/modules referenced the interface dir using ../interface etc(*.gyp*,*.gn*)
BUG=5095
TESTED=Passing compile-trybots with --clobber flag:
git cl try --clobber --bot=win_compile_rel --bot=linux_compile_rel --bot=android_compile_rel --bot=mac_compile_rel --bot=ios_rel -m tryserver.webrtc
R=stefan@webrtc.org, tommi@webrtc.org
Review URL: https://codereview.webrtc.org/1417683006 .
Cr-Commit-Position: refs/heads/master@{#10500}
136 lines
3.4 KiB
C++
136 lines
3.4 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/include/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 rtcp {
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class TransportFeedback;
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}
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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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rtc::scoped_ptr<rtcp::TransportFeedback> transport_feedback_;
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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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