/* * Copyright (c) 2015 The WebRTC project authors. All Rights Reserved. * * Use of this source code is governed by a BSD-style license * that can be found in the LICENSE file in the root of the source * tree. An additional intellectual property rights grant can be found * in the file PATENTS. All contributing project authors may * be found in the AUTHORS file in the root of the source tree. */ #ifndef WEBRTC_MODULES_PACING_PACKET_ROUTER_H_ #define WEBRTC_MODULES_PACING_PACKET_ROUTER_H_ #include #include #include "webrtc/base/constructormagic.h" #include "webrtc/base/criticalsection.h" #include "webrtc/base/thread_annotations.h" #include "webrtc/base/race_checker.h" #include "webrtc/common_types.h" #include "webrtc/modules/pacing/paced_sender.h" #include "webrtc/modules/remote_bitrate_estimator/include/remote_bitrate_estimator.h" #include "webrtc/modules/rtp_rtcp/include/rtp_rtcp_defines.h" namespace webrtc { class RtpRtcp; namespace rtcp { class TransportFeedback; } // namespace rtcp // PacketRouter keeps track of rtp send modules to support the pacer. // In addition, it handles feedback messages, which are sent on a send // module if possible (sender report), otherwise on receive module // (receiver report). For the latter case, we also keep track of the // receive modules. class PacketRouter : public PacedSender::PacketSender, public TransportSequenceNumberAllocator, public RemoteBitrateObserver { public: PacketRouter(); ~PacketRouter() override; // TODO(nisse): Delete, as soon as downstream app is updated. RTC_DEPRECATED void AddRtpModule(RtpRtcp* rtp_module) { AddReceiveRtpModule(rtp_module); } RTC_DEPRECATED void RemoveRtpModule(RtpRtcp* rtp_module) { RemoveReceiveRtpModule(rtp_module); } void AddSendRtpModule(RtpRtcp* rtp_module); void RemoveSendRtpModule(RtpRtcp* rtp_module); void AddReceiveRtpModule(RtpRtcp* rtp_module); void RemoveReceiveRtpModule(RtpRtcp* rtp_module); // Implements PacedSender::Callback. bool TimeToSendPacket(uint32_t ssrc, uint16_t sequence_number, int64_t capture_timestamp, bool retransmission, const PacedPacketInfo& packet_info) override; size_t TimeToSendPadding(size_t bytes, const PacedPacketInfo& packet_info) override; void SetTransportWideSequenceNumber(uint16_t sequence_number); uint16_t AllocateSequenceNumber() override; // Called every time there is a new bitrate estimate for a receive channel // group. This call will trigger a new RTCP REMB packet if the bitrate // estimate has decreased or if no RTCP REMB packet has been sent for // a certain time interval. // Implements RtpReceiveBitrateUpdate. void OnReceiveBitrateChanged(const std::vector& ssrcs, uint32_t bitrate_bps) override; // Send REMB feedback. virtual bool SendRemb(uint32_t bitrate_bps, const std::vector& ssrcs); // Send transport feedback packet to send-side. virtual bool SendTransportFeedback(rtcp::TransportFeedback* packet); private: rtc::RaceChecker pacer_race_; rtc::CriticalSection modules_crit_; std::list rtp_send_modules_ GUARDED_BY(modules_crit_); std::vector rtp_receive_modules_ GUARDED_BY(modules_crit_); rtc::CriticalSection remb_crit_; // The last time a REMB was sent. int64_t last_remb_time_ms_ GUARDED_BY(remb_crit_); uint32_t last_send_bitrate_bps_ GUARDED_BY(remb_crit_); // The last bitrate update. uint32_t bitrate_bps_ GUARDED_BY(remb_crit_); volatile int transport_seq_; RTC_DISALLOW_COPY_AND_ASSIGN(PacketRouter); }; } // namespace webrtc #endif // WEBRTC_MODULES_PACING_PACKET_ROUTER_H_