| /* |
| * Copyright (c) 2017 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 CALL_RTP_TRANSPORT_CONTROLLER_SEND_H_ |
| #define CALL_RTP_TRANSPORT_CONTROLLER_SEND_H_ |
| |
| #include <cstddef> |
| #include <cstdint> |
| #include <map> |
| #include <memory> |
| #include <optional> |
| #include <string> |
| #include <vector> |
| |
| #include "absl/strings/string_view.h" |
| #include "api/array_view.h" |
| #include "api/environment/environment.h" |
| #include "api/fec_controller.h" |
| #include "api/frame_transformer_interface.h" |
| #include "api/scoped_refptr.h" |
| #include "api/sequence_checker.h" |
| #include "api/task_queue/pending_task_safety_flag.h" |
| #include "api/task_queue/task_queue_base.h" |
| #include "api/transport/bandwidth_estimation_settings.h" |
| #include "api/transport/bitrate_settings.h" |
| #include "api/transport/network_control.h" |
| #include "api/transport/network_types.h" |
| #include "api/units/data_rate.h" |
| #include "api/units/data_size.h" |
| #include "api/units/time_delta.h" |
| #include "api/units/timestamp.h" |
| #include "call/rtp_bitrate_configurator.h" |
| #include "call/rtp_config.h" |
| #include "call/rtp_transport_config.h" |
| #include "call/rtp_transport_controller_send_interface.h" |
| #include "call/rtp_video_sender.h" |
| #include "modules/congestion_controller/rtp/control_handler.h" |
| #include "modules/congestion_controller/rtp/transport_feedback_adapter.h" |
| #include "modules/congestion_controller/rtp/transport_feedback_demuxer.h" |
| #include "modules/pacing/packet_router.h" |
| #include "modules/pacing/task_queue_paced_sender.h" |
| #include "modules/rtp_rtcp/include/report_block_data.h" |
| #include "modules/rtp_rtcp/include/rtp_rtcp_defines.h" |
| #include "rtc_base/experiments/field_trial_parser.h" |
| #include "rtc_base/network_route.h" |
| #include "rtc_base/rate_limiter.h" |
| #include "rtc_base/task_utils/repeating_task.h" |
| #include "rtc_base/thread_annotations.h" |
| |
| namespace webrtc { |
| class FrameEncryptorInterface; |
| |
| class RtpTransportControllerSend final |
| : public RtpTransportControllerSendInterface, |
| public NetworkLinkRtcpObserver, |
| public NetworkStateEstimateObserver { |
| public: |
| explicit RtpTransportControllerSend(const RtpTransportConfig& config); |
| ~RtpTransportControllerSend() override; |
| |
| RtpTransportControllerSend(const RtpTransportControllerSend&) = delete; |
| RtpTransportControllerSend& operator=(const RtpTransportControllerSend&) = |
| delete; |
| |
| // TODO(tommi): Change to std::unique_ptr<>. |
| RtpVideoSenderInterface* CreateRtpVideoSender( |
| const std::map<uint32_t, RtpState>& suspended_ssrcs, |
| const std::map<uint32_t, RtpPayloadState>& |
| states, // move states into RtpTransportControllerSend |
| const RtpConfig& rtp_config, |
| int rtcp_report_interval_ms, |
| Transport* send_transport, |
| const RtpSenderObservers& observers, |
| std::unique_ptr<FecController> fec_controller, |
| const RtpSenderFrameEncryptionConfig& frame_encryption_config, |
| rtc::scoped_refptr<FrameTransformerInterface> frame_transformer) override; |
| void DestroyRtpVideoSender( |
| RtpVideoSenderInterface* rtp_video_sender) override; |
| |
| // Implements RtpTransportControllerSendInterface |
| void RegisterSendingRtpStream(RtpRtcpInterface& rtp_module) override; |
| void DeRegisterSendingRtpStream(RtpRtcpInterface& rtp_module) override; |
| PacketRouter* packet_router() override; |
| |
| NetworkStateEstimateObserver* network_state_estimate_observer() override; |
| RtpPacketSender* packet_sender() override; |
| |
| void SetAllocatedSendBitrateLimits(BitrateAllocationLimits limits) override; |
| void ReconfigureBandwidthEstimation( |
| const BandwidthEstimationSettings& settings) override; |
| |
| void SetPacingFactor(float pacing_factor) override; |
| void SetQueueTimeLimit(int limit_ms) override; |
| StreamFeedbackProvider* GetStreamFeedbackProvider() override; |
| void RegisterTargetTransferRateObserver( |
| TargetTransferRateObserver* observer) override; |
| void OnNetworkRouteChanged(absl::string_view transport_name, |
| const rtc::NetworkRoute& network_route) override; |
| void OnNetworkAvailability(bool network_available) override; |
| NetworkLinkRtcpObserver* GetRtcpObserver() override; |
| int64_t GetPacerQueuingDelayMs() const override; |
| std::optional<Timestamp> GetFirstPacketTime() const override; |
| void EnablePeriodicAlrProbing(bool enable) override; |
| void OnSentPacket(const rtc::SentPacket& sent_packet) override; |
| void OnReceivedPacket(const ReceivedPacket& packet_msg) override; |
| |
| void SetSdpBitrateParameters(const BitrateConstraints& constraints) override; |
| void SetClientBitratePreferences(const BitrateSettings& preferences) override; |
| |
| void OnTransportOverheadChanged( |
| size_t transport_overhead_bytes_per_packet) override; |
| |
| void AccountForAudioPacketsInPacedSender(bool account_for_audio) override; |
| void IncludeOverheadInPacedSender() override; |
| void EnsureStarted() override; |
| |
| // Implements NetworkLinkRtcpObserver interface |
| void OnReceiverEstimatedMaxBitrate(Timestamp receive_time, |
| DataRate bitrate) override; |
| void OnReport(Timestamp receive_time, |
| rtc::ArrayView<const ReportBlockData> report_blocks) override; |
| void OnRttUpdate(Timestamp receive_time, TimeDelta rtt) override; |
| void OnTransportFeedback(Timestamp receive_time, |
| const rtcp::TransportFeedback& feedback) override; |
| |
| // Implements NetworkStateEstimateObserver interface |
| void OnRemoteNetworkEstimate(NetworkStateEstimate estimate) override; |
| |
| NetworkControllerInterface* GetNetworkController() override { |
| RTC_DCHECK_RUN_ON(&sequence_checker_); |
| return controller_.get(); |
| } |
| |
| private: |
| void MaybeCreateControllers() RTC_RUN_ON(sequence_checker_); |
| void UpdateNetworkAvailability() RTC_RUN_ON(sequence_checker_); |
| void UpdateInitialConstraints(TargetRateConstraints new_contraints) |
| RTC_RUN_ON(sequence_checker_); |
| |
| void StartProcessPeriodicTasks() RTC_RUN_ON(sequence_checker_); |
| void UpdateControllerWithTimeInterval() RTC_RUN_ON(sequence_checker_); |
| |
| std::optional<BitrateConstraints> ApplyOrLiftRelayCap(bool is_relayed); |
| bool IsRelevantRouteChange(const rtc::NetworkRoute& old_route, |
| const rtc::NetworkRoute& new_route) const; |
| void UpdateBitrateConstraints(const BitrateConstraints& updated); |
| void UpdateStreamsConfig() RTC_RUN_ON(sequence_checker_); |
| void PostUpdates(NetworkControlUpdate update) RTC_RUN_ON(sequence_checker_); |
| void UpdateControlState() RTC_RUN_ON(sequence_checker_); |
| void UpdateCongestedState() RTC_RUN_ON(sequence_checker_); |
| std::optional<bool> GetCongestedStateUpdate() const |
| RTC_RUN_ON(sequence_checker_); |
| |
| // Called by packet router just before packet is sent to the RTP modules. |
| void NotifyBweOfPacedSentPacket(const RtpPacketToSend& packet, |
| const PacedPacketInfo& pacing_info); |
| void ProcessSentPacket(const rtc::SentPacket& sent_packet) |
| RTC_RUN_ON(sequence_checker_); |
| void ProcessSentPacketUpdates(NetworkControlUpdate updates) |
| RTC_RUN_ON(sequence_checker_); |
| |
| const Environment env_; |
| SequenceChecker sequence_checker_; |
| TaskQueueBase* task_queue_; |
| PacketRouter packet_router_; |
| |
| std::vector<std::unique_ptr<RtpVideoSenderInterface>> video_rtp_senders_ |
| RTC_GUARDED_BY(&sequence_checker_); |
| RtpBitrateConfigurator bitrate_configurator_; |
| std::map<std::string, rtc::NetworkRoute> network_routes_ |
| RTC_GUARDED_BY(sequence_checker_); |
| BandwidthEstimationSettings bwe_settings_ RTC_GUARDED_BY(sequence_checker_); |
| bool pacer_started_ RTC_GUARDED_BY(sequence_checker_); |
| TaskQueuePacedSender pacer_; |
| |
| TargetTransferRateObserver* observer_ RTC_GUARDED_BY(sequence_checker_); |
| TransportFeedbackDemuxer feedback_demuxer_; |
| |
| TransportFeedbackAdapter transport_feedback_adapter_ |
| RTC_GUARDED_BY(sequence_checker_); |
| |
| NetworkControllerFactoryInterface* const controller_factory_override_ |
| RTC_PT_GUARDED_BY(sequence_checker_); |
| const std::unique_ptr<NetworkControllerFactoryInterface> |
| controller_factory_fallback_ RTC_PT_GUARDED_BY(sequence_checker_); |
| |
| std::unique_ptr<CongestionControlHandler> control_handler_ |
| RTC_GUARDED_BY(sequence_checker_) RTC_PT_GUARDED_BY(sequence_checker_); |
| |
| std::unique_ptr<NetworkControllerInterface> controller_ |
| RTC_GUARDED_BY(sequence_checker_) RTC_PT_GUARDED_BY(sequence_checker_); |
| |
| TimeDelta process_interval_ RTC_GUARDED_BY(sequence_checker_); |
| |
| struct LossReport { |
| uint32_t extended_highest_sequence_number = 0; |
| int cumulative_lost = 0; |
| }; |
| std::map<uint32_t, LossReport> last_report_blocks_ |
| RTC_GUARDED_BY(sequence_checker_); |
| Timestamp last_report_block_time_ RTC_GUARDED_BY(sequence_checker_); |
| |
| NetworkControllerConfig initial_config_ RTC_GUARDED_BY(sequence_checker_); |
| StreamsConfig streams_config_ RTC_GUARDED_BY(sequence_checker_); |
| |
| const bool reset_feedback_on_route_change_; |
| const bool add_pacing_to_cwin_; |
| const bool reset_bwe_on_adapter_id_change_; |
| |
| FieldTrialParameter<DataRate> relay_bandwidth_cap_; |
| |
| size_t transport_overhead_bytes_per_packet_ RTC_GUARDED_BY(sequence_checker_); |
| bool network_available_ RTC_GUARDED_BY(sequence_checker_); |
| RepeatingTaskHandle pacer_queue_update_task_ |
| RTC_GUARDED_BY(sequence_checker_); |
| RepeatingTaskHandle controller_task_ RTC_GUARDED_BY(sequence_checker_); |
| |
| DataSize congestion_window_size_ RTC_GUARDED_BY(sequence_checker_); |
| bool is_congested_ RTC_GUARDED_BY(sequence_checker_); |
| |
| // Protected by internal locks. |
| RateLimiter retransmission_rate_limiter_; |
| |
| ScopedTaskSafety safety_; |
| }; |
| |
| } // namespace webrtc |
| |
| #endif // CALL_RTP_TRANSPORT_CONTROLLER_SEND_H_ |