/* * 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 #include #include #include #include "api/transport/network_control.h" #include "call/rtp_bitrate_configurator.h" #include "call/rtp_transport_controller_send_interface.h" #include "call/rtp_video_sender.h" #include "modules/congestion_controller/include/send_side_congestion_controller_interface.h" #include "modules/pacing/packet_router.h" #include "modules/utility/include/process_thread.h" #include "rtc_base/constructormagic.h" #include "rtc_base/networkroute.h" #include "rtc_base/task_queue.h" namespace webrtc { class Clock; class FrameEncryptorInterface; class RtcEventLog; // TODO(nisse): When we get the underlying transports here, we should // have one object implementing RtpTransportControllerSendInterface // per transport, sharing the same congestion controller. class RtpTransportControllerSend final : public RtpTransportControllerSendInterface, public NetworkChangedObserver { public: RtpTransportControllerSend( Clock* clock, RtcEventLog* event_log, NetworkControllerFactoryInterface* controller_factory, const BitrateConstraints& bitrate_config); ~RtpTransportControllerSend() override; RtpVideoSenderInterface* CreateRtpVideoSender( const std::vector& ssrcs, std::map suspended_ssrcs, const std::map& states, // move states into RtpTransportControllerSend const RtpConfig& rtp_config, int rtcp_report_interval_ms, Transport* send_transport, const RtpSenderObservers& observers, RtcEventLog* event_log, std::unique_ptr fec_controller, const RtpSenderFrameEncryptionConfig& frame_encryption_config) override; void DestroyRtpVideoSender( RtpVideoSenderInterface* rtp_video_sender) override; // Implements NetworkChangedObserver interface. void OnNetworkChanged(uint32_t bitrate_bps, uint8_t fraction_loss, int64_t rtt_ms, int64_t probing_interval_ms) override; // Implements RtpTransportControllerSendInterface rtc::TaskQueue* GetWorkerQueue() override; PacketRouter* packet_router() override; TransportFeedbackObserver* transport_feedback_observer() override; RtpPacketSender* packet_sender() override; const RtpKeepAliveConfig& keepalive_config() const override; void SetAllocatedSendBitrateLimits(int min_send_bitrate_bps, int max_padding_bitrate_bps, int max_total_bitrate_bps) override; void SetKeepAliveConfig(const RtpKeepAliveConfig& config); void SetPacingFactor(float pacing_factor) override; void SetQueueTimeLimit(int limit_ms) override; CallStatsObserver* GetCallStatsObserver() override; void RegisterPacketFeedbackObserver( PacketFeedbackObserver* observer) override; void DeRegisterPacketFeedbackObserver( PacketFeedbackObserver* observer) override; void RegisterTargetTransferRateObserver( TargetTransferRateObserver* observer) override; void OnNetworkRouteChanged(const std::string& transport_name, const rtc::NetworkRoute& network_route) override; void OnNetworkAvailability(bool network_available) override; RtcpBandwidthObserver* GetBandwidthObserver() override; int64_t GetPacerQueuingDelayMs() const override; int64_t GetFirstPacketTimeMs() const override; void SetPerPacketFeedbackAvailable(bool available) override; void EnablePeriodicAlrProbing(bool enable) override; void OnSentPacket(const rtc::SentPacket& sent_packet) override; void SetSdpBitrateParameters(const BitrateConstraints& constraints) override; void SetClientBitratePreferences(const BitrateSettings& preferences) override; void SetAllocatedBitrateWithoutFeedback(uint32_t bitrate_bps) override; void OnTransportOverheadChanged( size_t transport_overhead_per_packet) override; private: const Clock* const clock_; PacketRouter packet_router_; std::vector> video_rtp_senders_; PacedSender pacer_; RtpKeepAliveConfig keepalive_; RtpBitrateConfigurator bitrate_configurator_; std::map network_routes_; const std::unique_ptr process_thread_; rtc::CriticalSection observer_crit_; TargetTransferRateObserver* observer_ RTC_GUARDED_BY(observer_crit_); std::unique_ptr send_side_cc_; RateLimiter retransmission_rate_limiter_; // TODO(perkj): |task_queue_| is supposed to replace |process_thread_|. // |task_queue_| is defined last to ensure all pending tasks are cancelled // and deleted before any other members. rtc::TaskQueue task_queue_; RTC_DISALLOW_COPY_AND_ASSIGN(RtpTransportControllerSend); }; } // namespace webrtc #endif // CALL_RTP_TRANSPORT_CONTROLLER_SEND_H_