Moving src/webrtc into src/.
In order to eliminate the WebRTC Subtree mirror in Chromium, WebRTC is moving the content of the src/webrtc directory up to the src/ directory. NOPRESUBMIT=true NOTREECHECKS=true NOTRY=true TBR=tommi@webrtc.org Bug: chromium:611808 Change-Id: Iac59c5b51b950f174119565bac87955a7994bc38 Reviewed-on: https://webrtc-review.googlesource.com/1560 Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org> Reviewed-by: Henrik Kjellander <kjellander@webrtc.org> Cr-Commit-Position: refs/heads/master@{#19845}
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modules/rtp_rtcp/source/rtcp_sender.h
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modules/rtp_rtcp/source/rtcp_sender.h
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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_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 <memory>
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#include <set>
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#include <sstream>
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#include <string>
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#include <vector>
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#include "webrtc/api/call/transport.h"
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#include "webrtc/api/optional.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/include/receive_statistics.h"
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#include "webrtc/modules/rtp_rtcp/include/rtp_rtcp_defines.h"
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#include "webrtc/modules/rtp_rtcp/source/rtcp_nack_stats.h"
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#include "webrtc/modules/rtp_rtcp/source/rtcp_packet.h"
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#include "webrtc/modules/rtp_rtcp/source/rtcp_packet/dlrr.h"
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#include "webrtc/modules/rtp_rtcp/source/rtcp_packet/report_block.h"
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#include "webrtc/modules/rtp_rtcp/source/rtcp_packet/tmmb_item.h"
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#include "webrtc/rtc_base/constructormagic.h"
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#include "webrtc/rtc_base/criticalsection.h"
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#include "webrtc/rtc_base/random.h"
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#include "webrtc/rtc_base/thread_annotations.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 RtcEventLog;
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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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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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rtcp::ReceiveTimeInfo 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(bool audio,
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Clock* clock,
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ReceiveStatisticsProvider* receive_statistics,
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RtcpPacketTypeCounterObserver* packet_type_counter_observer,
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RtcEventLog* event_log,
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Transport* outgoing_transport);
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virtual ~RTCPSender();
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RtcpMode Status() const;
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void SetRTCPStatus(RtcpMode 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 SetTimestampOffset(uint32_t timestamp_offset);
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void SetLastRtpTime(uint32_t rtp_timestamp, int64_t capture_time_ms);
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uint32_t SSRC() const;
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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);
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int32_t AddMixedCNAME(uint32_t SSRC, const char* c_name);
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int32_t RemoveMixedCNAME(uint32_t SSRC);
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bool TimeToSendRTCPReport(bool sendKeyframeBeforeRTP = false) const;
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int32_t SendRTCP(const FeedbackState& feedback_state,
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RTCPPacketType packetType,
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int32_t nackSize = 0,
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const uint16_t* nackList = 0);
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int32_t SendCompoundRTCP(const FeedbackState& feedback_state,
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const std::set<RTCPPacketType>& packetTypes,
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int32_t nackSize = 0,
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const uint16_t* nackList = 0);
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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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bool TMMBR() const;
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void SetTMMBRStatus(bool enable);
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void SetMaxRtpPacketSize(size_t max_packet_size);
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void SetTmmbn(std::vector<rtcp::TmmbItem> bounding_set);
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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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void SetVideoBitrateAllocation(const BitrateAllocation& bitrate);
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bool SendFeedbackPacket(const rtcp::TransportFeedback& packet);
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private:
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class RtcpContext;
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// Determine which RTCP messages should be sent and setup flags.
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void PrepareReport(const FeedbackState& feedback_state)
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RTC_EXCLUSIVE_LOCKS_REQUIRED(critical_section_rtcp_sender_);
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std::vector<rtcp::ReportBlock> CreateReportBlocks(
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const FeedbackState& feedback_state)
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RTC_EXCLUSIVE_LOCKS_REQUIRED(critical_section_rtcp_sender_);
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std::unique_ptr<rtcp::RtcpPacket> BuildSR(const RtcpContext& context)
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RTC_EXCLUSIVE_LOCKS_REQUIRED(critical_section_rtcp_sender_);
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std::unique_ptr<rtcp::RtcpPacket> BuildRR(const RtcpContext& context)
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RTC_EXCLUSIVE_LOCKS_REQUIRED(critical_section_rtcp_sender_);
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std::unique_ptr<rtcp::RtcpPacket> BuildSDES(const RtcpContext& context)
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RTC_EXCLUSIVE_LOCKS_REQUIRED(critical_section_rtcp_sender_);
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std::unique_ptr<rtcp::RtcpPacket> BuildPLI(const RtcpContext& context)
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RTC_EXCLUSIVE_LOCKS_REQUIRED(critical_section_rtcp_sender_);
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std::unique_ptr<rtcp::RtcpPacket> BuildREMB(const RtcpContext& context)
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RTC_EXCLUSIVE_LOCKS_REQUIRED(critical_section_rtcp_sender_);
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std::unique_ptr<rtcp::RtcpPacket> BuildTMMBR(const RtcpContext& context)
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RTC_EXCLUSIVE_LOCKS_REQUIRED(critical_section_rtcp_sender_);
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std::unique_ptr<rtcp::RtcpPacket> BuildTMMBN(const RtcpContext& context)
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RTC_EXCLUSIVE_LOCKS_REQUIRED(critical_section_rtcp_sender_);
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std::unique_ptr<rtcp::RtcpPacket> BuildAPP(const RtcpContext& context)
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RTC_EXCLUSIVE_LOCKS_REQUIRED(critical_section_rtcp_sender_);
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std::unique_ptr<rtcp::RtcpPacket> BuildExtendedReports(
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const RtcpContext& context)
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RTC_EXCLUSIVE_LOCKS_REQUIRED(critical_section_rtcp_sender_);
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std::unique_ptr<rtcp::RtcpPacket> BuildBYE(const RtcpContext& context)
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RTC_EXCLUSIVE_LOCKS_REQUIRED(critical_section_rtcp_sender_);
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std::unique_ptr<rtcp::RtcpPacket> BuildFIR(const RtcpContext& context)
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RTC_EXCLUSIVE_LOCKS_REQUIRED(critical_section_rtcp_sender_);
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std::unique_ptr<rtcp::RtcpPacket> BuildNACK(const RtcpContext& context)
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RTC_EXCLUSIVE_LOCKS_REQUIRED(critical_section_rtcp_sender_);
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private:
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const bool audio_;
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Clock* const clock_;
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Random random_ RTC_GUARDED_BY(critical_section_rtcp_sender_);
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RtcpMode method_ RTC_GUARDED_BY(critical_section_rtcp_sender_);
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RtcEventLog* const event_log_;
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Transport* const transport_;
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rtc::CriticalSection critical_section_rtcp_sender_;
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bool using_nack_ RTC_GUARDED_BY(critical_section_rtcp_sender_);
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bool sending_ RTC_GUARDED_BY(critical_section_rtcp_sender_);
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bool remb_enabled_ RTC_GUARDED_BY(critical_section_rtcp_sender_);
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int64_t next_time_to_send_rtcp_ RTC_GUARDED_BY(critical_section_rtcp_sender_);
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uint32_t timestamp_offset_ RTC_GUARDED_BY(critical_section_rtcp_sender_);
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uint32_t last_rtp_timestamp_ RTC_GUARDED_BY(critical_section_rtcp_sender_);
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int64_t last_frame_capture_time_ms_
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RTC_GUARDED_BY(critical_section_rtcp_sender_);
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uint32_t ssrc_ RTC_GUARDED_BY(critical_section_rtcp_sender_);
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// SSRC that we receive on our RTP channel
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uint32_t remote_ssrc_ RTC_GUARDED_BY(critical_section_rtcp_sender_);
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std::string cname_ RTC_GUARDED_BY(critical_section_rtcp_sender_);
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ReceiveStatisticsProvider* receive_statistics_
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RTC_GUARDED_BY(critical_section_rtcp_sender_);
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std::map<uint32_t, std::string> csrc_cnames_
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RTC_GUARDED_BY(critical_section_rtcp_sender_);
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// send CSRCs
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std::vector<uint32_t> csrcs_ RTC_GUARDED_BY(critical_section_rtcp_sender_);
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// Full intra request
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uint8_t sequence_number_fir_ RTC_GUARDED_BY(critical_section_rtcp_sender_);
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// REMB
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uint32_t remb_bitrate_ RTC_GUARDED_BY(critical_section_rtcp_sender_);
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std::vector<uint32_t> remb_ssrcs_
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RTC_GUARDED_BY(critical_section_rtcp_sender_);
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std::vector<rtcp::TmmbItem> tmmbn_to_send_
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RTC_GUARDED_BY(critical_section_rtcp_sender_);
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uint32_t tmmbr_send_bps_ RTC_GUARDED_BY(critical_section_rtcp_sender_);
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uint32_t packet_oh_send_ RTC_GUARDED_BY(critical_section_rtcp_sender_);
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size_t max_packet_size_ RTC_GUARDED_BY(critical_section_rtcp_sender_);
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// APP
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uint8_t app_sub_type_ RTC_GUARDED_BY(critical_section_rtcp_sender_);
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uint32_t app_name_ RTC_GUARDED_BY(critical_section_rtcp_sender_);
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std::unique_ptr<uint8_t[]> app_data_
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RTC_GUARDED_BY(critical_section_rtcp_sender_);
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uint16_t app_length_ RTC_GUARDED_BY(critical_section_rtcp_sender_);
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// True if sending of XR Receiver reference time report is enabled.
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bool xr_send_receiver_reference_time_enabled_
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RTC_GUARDED_BY(critical_section_rtcp_sender_);
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// XR VoIP metric
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rtc::Optional<RTCPVoIPMetric> xr_voip_metric_
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RTC_GUARDED_BY(critical_section_rtcp_sender_);
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RtcpPacketTypeCounterObserver* const packet_type_counter_observer_;
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RtcpPacketTypeCounter packet_type_counter_
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RTC_GUARDED_BY(critical_section_rtcp_sender_);
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RtcpNackStats nack_stats_ RTC_GUARDED_BY(critical_section_rtcp_sender_);
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rtc::Optional<BitrateAllocation> video_bitrate_allocation_
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RTC_GUARDED_BY(critical_section_rtcp_sender_);
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void SetFlag(uint32_t type, bool is_volatile)
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RTC_EXCLUSIVE_LOCKS_REQUIRED(critical_section_rtcp_sender_);
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void SetFlags(const std::set<RTCPPacketType>& types, bool is_volatile)
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RTC_EXCLUSIVE_LOCKS_REQUIRED(critical_section_rtcp_sender_);
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bool IsFlagPresent(uint32_t type) const
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RTC_EXCLUSIVE_LOCKS_REQUIRED(critical_section_rtcp_sender_);
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bool ConsumeFlag(uint32_t type, bool forced = false)
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RTC_EXCLUSIVE_LOCKS_REQUIRED(critical_section_rtcp_sender_);
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bool AllVolatileFlagsConsumed() const
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RTC_EXCLUSIVE_LOCKS_REQUIRED(critical_section_rtcp_sender_);
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struct ReportFlag {
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ReportFlag(uint32_t type, bool is_volatile)
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: type(type), is_volatile(is_volatile) {}
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bool operator<(const ReportFlag& flag) const { return type < flag.type; }
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bool operator==(const ReportFlag& flag) const { return type == flag.type; }
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const uint32_t type;
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const bool is_volatile;
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};
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std::set<ReportFlag> report_flags_
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RTC_GUARDED_BY(critical_section_rtcp_sender_);
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typedef std::unique_ptr<rtcp::RtcpPacket> (RTCPSender::*BuilderFunc)(
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const RtcpContext&);
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// Map from RTCPPacketType to builder.
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std::map<uint32_t, BuilderFunc> builders_;
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RTC_DISALLOW_IMPLICIT_CONSTRUCTORS(RTCPSender);
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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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