00b8f6b364
BUG= R=andrew@webrtc.org Review URL: https://webrtc-codereview.appspot.com/36229004 Cr-Commit-Position: refs/heads/master@{#8517} git-svn-id: http://webrtc.googlecode.com/svn/trunk@8517 4adac7df-926f-26a2-2b94-8c16560cd09d
352 lines
12 KiB
C++
352 lines
12 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_SENDER_H_
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#define WEBRTC_MODULES_RTP_RTCP_SOURCE_RTCP_SENDER_H_
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#include <map>
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#include <sstream>
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#include <string>
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#include "webrtc/base/scoped_ptr.h"
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#include "webrtc/base/thread_annotations.h"
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#include "webrtc/modules/remote_bitrate_estimator/include/bwe_defines.h"
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#include "webrtc/modules/remote_bitrate_estimator/include/remote_bitrate_estimator.h"
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#include "webrtc/modules/rtp_rtcp/interface/receive_statistics.h"
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#include "webrtc/modules/rtp_rtcp/interface/rtp_rtcp_defines.h"
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#include "webrtc/modules/rtp_rtcp/source/rtcp_utility.h"
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#include "webrtc/modules/rtp_rtcp/source/rtp_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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class ModuleRtpRtcpImpl;
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class RTCPReceiver;
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class NACKStringBuilder {
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public:
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NACKStringBuilder();
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~NACKStringBuilder();
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void PushNACK(uint16_t nack);
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std::string GetResult();
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private:
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std::ostringstream _stream;
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int _count;
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uint16_t _prevNack;
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bool _consecutive;
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};
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class RTCPSender {
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public:
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struct FeedbackState {
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FeedbackState();
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uint8_t send_payload_type;
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uint32_t frequency_hz;
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uint32_t packets_sent;
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size_t media_bytes_sent;
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uint32_t send_bitrate;
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uint32_t last_rr_ntp_secs;
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uint32_t last_rr_ntp_frac;
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uint32_t remote_sr;
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bool has_last_xr_rr;
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RtcpReceiveTimeInfo last_xr_rr;
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// Used when generating TMMBR.
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ModuleRtpRtcpImpl* module;
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};
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RTCPSender(int32_t id,
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bool audio,
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Clock* clock,
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ReceiveStatistics* receive_statistics,
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RtcpPacketTypeCounterObserver* packet_type_counter_observer);
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virtual ~RTCPSender();
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int32_t RegisterSendTransport(Transport* outgoingTransport);
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RTCPMethod Status() const;
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void SetRTCPStatus(RTCPMethod method);
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bool Sending() const;
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int32_t SetSendingStatus(const FeedbackState& feedback_state,
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bool enabled); // combine the functions
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int32_t SetNackStatus(bool enable);
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void SetStartTimestamp(uint32_t start_timestamp);
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void SetLastRtpTime(uint32_t rtp_timestamp,
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int64_t capture_time_ms);
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void SetSSRC(uint32_t ssrc);
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void SetRemoteSSRC(uint32_t ssrc);
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int32_t SetCNAME(const char cName[RTCP_CNAME_SIZE]);
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int32_t AddMixedCNAME(uint32_t SSRC, const char cName[RTCP_CNAME_SIZE]);
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int32_t RemoveMixedCNAME(uint32_t SSRC);
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int64_t SendTimeOfSendReport(uint32_t sendReport);
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bool SendTimeOfXrRrReport(uint32_t mid_ntp, int64_t* time_ms) const;
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bool TimeToSendRTCPReport(bool sendKeyframeBeforeRTP = false) const;
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uint32_t LastSendReport(int64_t& lastRTCPTime);
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int32_t SendRTCP(
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const FeedbackState& feedback_state,
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uint32_t rtcpPacketTypeFlags,
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int32_t nackSize = 0,
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const uint16_t* nackList = 0,
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bool repeat = false,
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uint64_t pictureID = 0);
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int32_t AddExternalReportBlock(
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uint32_t SSRC,
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const RTCPReportBlock* receiveBlock);
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int32_t RemoveExternalReportBlock(uint32_t SSRC);
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/*
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* REMB
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*/
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bool REMB() const;
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void SetREMBStatus(bool enable);
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void SetREMBData(uint32_t bitrate, const std::vector<uint32_t>& ssrcs);
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/*
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* TMMBR
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*/
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bool TMMBR() const;
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void SetTMMBRStatus(bool enable);
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int32_t SetTMMBN(const TMMBRSet* boundingSet, uint32_t maxBitrateKbit);
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/*
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* Extended jitter report
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*/
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bool IJ() const;
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void SetIJStatus(bool enable);
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/*
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*
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*/
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int32_t SetApplicationSpecificData(uint8_t subType,
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uint32_t name,
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const uint8_t* data,
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uint16_t length);
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int32_t SetRTCPVoIPMetrics(const RTCPVoIPMetric* VoIPMetric);
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void SendRtcpXrReceiverReferenceTime(bool enable);
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bool RtcpXrReceiverReferenceTime() const;
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void SetCsrcs(const std::vector<uint32_t>& csrcs);
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void SetTargetBitrate(unsigned int target_bitrate);
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private:
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int32_t SendToNetwork(const uint8_t* dataBuffer, size_t length);
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int32_t WriteAllReportBlocksToBuffer(uint8_t* rtcpbuffer,
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int pos,
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uint8_t& numberOfReportBlocks,
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uint32_t NTPsec,
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uint32_t NTPfrac)
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EXCLUSIVE_LOCKS_REQUIRED(_criticalSectionRTCPSender);
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int32_t WriteReportBlocksToBuffer(
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uint8_t* rtcpbuffer,
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int32_t position,
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const std::map<uint32_t, RTCPReportBlock*>& report_blocks);
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int32_t AddReportBlock(
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uint32_t SSRC,
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std::map<uint32_t, RTCPReportBlock*>* report_blocks,
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const RTCPReportBlock* receiveBlock);
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bool PrepareReport(const FeedbackState& feedback_state,
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StreamStatistician* statistician,
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RTCPReportBlock* report_block,
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uint32_t* ntp_secs, uint32_t* ntp_frac);
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int32_t BuildSR(const FeedbackState& feedback_state,
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uint8_t* rtcpbuffer,
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int& pos,
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uint32_t NTPsec,
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uint32_t NTPfrac)
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EXCLUSIVE_LOCKS_REQUIRED(_criticalSectionRTCPSender);
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int32_t BuildRR(uint8_t* rtcpbuffer,
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int& pos,
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uint32_t NTPsec,
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uint32_t NTPfrac)
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EXCLUSIVE_LOCKS_REQUIRED(_criticalSectionRTCPSender);
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int PrepareRTCP(
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const FeedbackState& feedback_state,
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uint32_t packetTypeFlags,
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int32_t nackSize,
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const uint16_t* nackList,
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bool repeat,
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uint64_t pictureID,
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uint8_t* rtcp_buffer,
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int buffer_size);
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bool ShouldSendReportBlocks(uint32_t rtcp_packet_type) const;
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int32_t BuildExtendedJitterReport(uint8_t* rtcpbuffer,
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int& pos,
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uint32_t jitterTransmissionTimeOffset)
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EXCLUSIVE_LOCKS_REQUIRED(_criticalSectionRTCPSender);
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int32_t BuildSDEC(uint8_t* rtcpbuffer, int& pos)
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EXCLUSIVE_LOCKS_REQUIRED(_criticalSectionRTCPSender);
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int32_t BuildPLI(uint8_t* rtcpbuffer, int& pos)
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EXCLUSIVE_LOCKS_REQUIRED(_criticalSectionRTCPSender);
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int32_t BuildREMB(uint8_t* rtcpbuffer, int& pos)
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EXCLUSIVE_LOCKS_REQUIRED(_criticalSectionRTCPSender);
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int32_t BuildTMMBR(ModuleRtpRtcpImpl* module, uint8_t* rtcpbuffer, int& pos)
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EXCLUSIVE_LOCKS_REQUIRED(_criticalSectionRTCPSender);
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int32_t BuildTMMBN(uint8_t* rtcpbuffer, int& pos)
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EXCLUSIVE_LOCKS_REQUIRED(_criticalSectionRTCPSender);
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int32_t BuildAPP(uint8_t* rtcpbuffer, int& pos)
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EXCLUSIVE_LOCKS_REQUIRED(_criticalSectionRTCPSender);
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int32_t BuildVoIPMetric(uint8_t* rtcpbuffer, int& pos)
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EXCLUSIVE_LOCKS_REQUIRED(_criticalSectionRTCPSender);
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int32_t BuildBYE(uint8_t* rtcpbuffer, int& pos)
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EXCLUSIVE_LOCKS_REQUIRED(_criticalSectionRTCPSender);
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int32_t BuildFIR(uint8_t* rtcpbuffer, int& pos, bool repeat)
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EXCLUSIVE_LOCKS_REQUIRED(_criticalSectionRTCPSender);
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int32_t BuildSLI(uint8_t* rtcpbuffer, int& pos, uint8_t pictureID)
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EXCLUSIVE_LOCKS_REQUIRED(_criticalSectionRTCPSender);
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int32_t BuildRPSI(uint8_t* rtcpbuffer,
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int& pos,
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uint64_t pictureID,
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uint8_t payloadType)
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EXCLUSIVE_LOCKS_REQUIRED(_criticalSectionRTCPSender);
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int32_t BuildNACK(uint8_t* rtcpbuffer,
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int& pos,
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int32_t nackSize,
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const uint16_t* nackList,
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std::string* nackString)
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EXCLUSIVE_LOCKS_REQUIRED(_criticalSectionRTCPSender);
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int32_t BuildReceiverReferenceTime(uint8_t* buffer,
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int& pos,
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uint32_t ntp_sec,
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uint32_t ntp_frac)
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EXCLUSIVE_LOCKS_REQUIRED(_criticalSectionRTCPSender);
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int32_t BuildDlrr(uint8_t* buffer,
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int& pos,
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const RtcpReceiveTimeInfo& info)
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EXCLUSIVE_LOCKS_REQUIRED(_criticalSectionRTCPSender);
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private:
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const int32_t _id;
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const bool _audio;
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Clock* const _clock;
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RTCPMethod _method GUARDED_BY(_criticalSectionRTCPSender);
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CriticalSectionWrapper* _criticalSectionTransport;
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Transport* _cbTransport GUARDED_BY(_criticalSectionTransport);
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CriticalSectionWrapper* _criticalSectionRTCPSender;
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bool _usingNack GUARDED_BY(_criticalSectionRTCPSender);
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bool _sending GUARDED_BY(_criticalSectionRTCPSender);
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bool _sendTMMBN GUARDED_BY(_criticalSectionRTCPSender);
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bool _REMB GUARDED_BY(_criticalSectionRTCPSender);
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bool _sendREMB GUARDED_BY(_criticalSectionRTCPSender);
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bool _TMMBR GUARDED_BY(_criticalSectionRTCPSender);
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bool _IJ GUARDED_BY(_criticalSectionRTCPSender);
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int64_t _nextTimeToSendRTCP GUARDED_BY(_criticalSectionRTCPSender);
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uint32_t start_timestamp_ GUARDED_BY(_criticalSectionRTCPSender);
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uint32_t last_rtp_timestamp_ GUARDED_BY(_criticalSectionRTCPSender);
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int64_t last_frame_capture_time_ms_ GUARDED_BY(_criticalSectionRTCPSender);
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uint32_t _SSRC GUARDED_BY(_criticalSectionRTCPSender);
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// SSRC that we receive on our RTP channel
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uint32_t _remoteSSRC GUARDED_BY(_criticalSectionRTCPSender);
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char _CNAME[RTCP_CNAME_SIZE] GUARDED_BY(_criticalSectionRTCPSender);
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ReceiveStatistics* receive_statistics_
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GUARDED_BY(_criticalSectionRTCPSender);
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std::map<uint32_t, RTCPReportBlock*> internal_report_blocks_
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GUARDED_BY(_criticalSectionRTCPSender);
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std::map<uint32_t, RTCPReportBlock*> external_report_blocks_
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GUARDED_BY(_criticalSectionRTCPSender);
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std::map<uint32_t, RTCPUtility::RTCPCnameInformation*> _csrcCNAMEs
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GUARDED_BY(_criticalSectionRTCPSender);
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// Sent
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uint32_t _lastSendReport[RTCP_NUMBER_OF_SR] GUARDED_BY(
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_criticalSectionRTCPSender); // allow packet loss and RTT above 1 sec
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int64_t _lastRTCPTime[RTCP_NUMBER_OF_SR] GUARDED_BY(
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_criticalSectionRTCPSender);
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// Sent XR receiver reference time report.
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// <mid ntp (mid 32 bits of the 64 bits NTP timestamp), send time in ms>.
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std::map<uint32_t, int64_t> last_xr_rr_
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GUARDED_BY(_criticalSectionRTCPSender);
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// send CSRCs
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std::vector<uint32_t> csrcs_ GUARDED_BY(_criticalSectionRTCPSender);
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// Full intra request
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uint8_t _sequenceNumberFIR GUARDED_BY(_criticalSectionRTCPSender);
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// REMB
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uint32_t _rembBitrate GUARDED_BY(_criticalSectionRTCPSender);
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std::vector<uint32_t> remb_ssrcs_ GUARDED_BY(_criticalSectionRTCPSender);
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TMMBRHelp _tmmbrHelp GUARDED_BY(_criticalSectionRTCPSender);
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uint32_t _tmmbr_Send GUARDED_BY(_criticalSectionRTCPSender);
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uint32_t _packetOH_Send GUARDED_BY(_criticalSectionRTCPSender);
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// APP
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bool _appSend GUARDED_BY(_criticalSectionRTCPSender);
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uint8_t _appSubType GUARDED_BY(_criticalSectionRTCPSender);
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uint32_t _appName GUARDED_BY(_criticalSectionRTCPSender);
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uint8_t* _appData GUARDED_BY(_criticalSectionRTCPSender);
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uint16_t _appLength GUARDED_BY(_criticalSectionRTCPSender);
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// True if sending of XR Receiver reference time report is enabled.
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bool xrSendReceiverReferenceTimeEnabled_
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GUARDED_BY(_criticalSectionRTCPSender);
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// XR VoIP metric
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bool _xrSendVoIPMetric GUARDED_BY(_criticalSectionRTCPSender);
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RTCPVoIPMetric _xrVoIPMetric GUARDED_BY(_criticalSectionRTCPSender);
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RtcpPacketTypeCounterObserver* const packet_type_counter_observer_;
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RtcpPacketTypeCounter packet_type_counter_
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GUARDED_BY(_criticalSectionRTCPSender);
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RTCPUtility::NackStats nack_stats_ GUARDED_BY(_criticalSectionRTCPSender);
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};
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} // namespace webrtc
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#endif // WEBRTC_MODULES_RTP_RTCP_SOURCE_RTCP_SENDER_H_
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