| /* |
| * 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. |
| */ |
| #include <utility> |
| #include <vector> |
| |
| #include "absl/memory/memory.h" |
| #include "call/rtp_transport_controller_send.h" |
| #include "modules/congestion_controller/include/send_side_congestion_controller.h" |
| #include "modules/congestion_controller/rtp/include/send_side_congestion_controller.h" |
| #include "rtc_base/location.h" |
| #include "rtc_base/logging.h" |
| #include "rtc_base/rate_limiter.h" |
| #include "system_wrappers/include/field_trial.h" |
| |
| namespace webrtc { |
| namespace { |
| static const int64_t kRetransmitWindowSizeMs = 500; |
| static const size_t kMaxOverheadBytes = 500; |
| const char kTaskQueueExperiment[] = "WebRTC-TaskQueueCongestionControl"; |
| using TaskQueueController = webrtc::webrtc_cc::SendSideCongestionController; |
| |
| std::unique_ptr<SendSideCongestionControllerInterface> CreateController( |
| Clock* clock, |
| rtc::TaskQueue* task_queue, |
| webrtc::RtcEventLog* event_log, |
| PacedSender* pacer, |
| const BitrateConstraints& bitrate_config, |
| bool task_queue_controller, |
| NetworkControllerFactoryInterface* controller_factory) { |
| if (task_queue_controller) { |
| RTC_LOG(LS_INFO) << "Using TaskQueue based SSCC"; |
| return absl::make_unique<webrtc::webrtc_cc::SendSideCongestionController>( |
| clock, task_queue, event_log, pacer, bitrate_config.start_bitrate_bps, |
| bitrate_config.min_bitrate_bps, bitrate_config.max_bitrate_bps, |
| controller_factory); |
| } |
| RTC_LOG(LS_INFO) << "Using Legacy SSCC"; |
| auto cc = absl::make_unique<webrtc::SendSideCongestionController>( |
| clock, nullptr /* observer */, event_log, pacer); |
| cc->SignalNetworkState(kNetworkDown); |
| cc->SetBweBitrates(bitrate_config.min_bitrate_bps, |
| bitrate_config.start_bitrate_bps, |
| bitrate_config.max_bitrate_bps); |
| return std::move(cc); |
| } |
| } // namespace |
| |
| RtpTransportControllerSend::RtpTransportControllerSend( |
| Clock* clock, |
| webrtc::RtcEventLog* event_log, |
| NetworkControllerFactoryInterface* controller_factory, |
| const BitrateConstraints& bitrate_config) |
| : clock_(clock), |
| pacer_(clock, &packet_router_, event_log), |
| bitrate_configurator_(bitrate_config), |
| process_thread_(ProcessThread::Create("SendControllerThread")), |
| observer_(nullptr), |
| retransmission_rate_limiter_(clock, kRetransmitWindowSizeMs), |
| task_queue_("rtp_send_controller") { |
| // Created after task_queue to be able to post to the task queue internally. |
| send_side_cc_ = CreateController( |
| clock, &task_queue_, event_log, &pacer_, bitrate_config, |
| !field_trial::IsDisabled(kTaskQueueExperiment), controller_factory); |
| |
| process_thread_->RegisterModule(&pacer_, RTC_FROM_HERE); |
| process_thread_->RegisterModule(send_side_cc_.get(), RTC_FROM_HERE); |
| process_thread_->Start(); |
| } |
| |
| RtpTransportControllerSend::~RtpTransportControllerSend() { |
| process_thread_->Stop(); |
| process_thread_->DeRegisterModule(send_side_cc_.get()); |
| process_thread_->DeRegisterModule(&pacer_); |
| } |
| |
| RtpVideoSenderInterface* RtpTransportControllerSend::CreateRtpVideoSender( |
| const std::vector<uint32_t>& ssrcs, |
| std::map<uint32_t, RtpState> suspended_ssrcs, |
| const std::map<uint32_t, RtpPayloadState>& states, |
| const RtpConfig& rtp_config, |
| const RtcpConfig& rtcp_config, |
| Transport* send_transport, |
| const RtpSenderObservers& observers, |
| RtcEventLog* event_log, |
| std::unique_ptr<FecController> fec_controller, |
| const RtpSenderFrameEncryptionConfig& frame_encryption_config) { |
| video_rtp_senders_.push_back(absl::make_unique<RtpVideoSender>( |
| ssrcs, suspended_ssrcs, states, rtp_config, rtcp_config, send_transport, |
| observers, |
| // TODO(holmer): Remove this circular dependency by injecting |
| // the parts of RtpTransportControllerSendInterface that are really used. |
| this, event_log, &retransmission_rate_limiter_, std::move(fec_controller), |
| frame_encryption_config.frame_encryptor, |
| frame_encryption_config.crypto_options)); |
| return video_rtp_senders_.back().get(); |
| } |
| |
| void RtpTransportControllerSend::DestroyRtpVideoSender( |
| RtpVideoSenderInterface* rtp_video_sender) { |
| std::vector<std::unique_ptr<RtpVideoSenderInterface>>::iterator it = |
| video_rtp_senders_.end(); |
| for (it = video_rtp_senders_.begin(); it != video_rtp_senders_.end(); ++it) { |
| if (it->get() == rtp_video_sender) { |
| break; |
| } |
| } |
| RTC_DCHECK(it != video_rtp_senders_.end()); |
| video_rtp_senders_.erase(it); |
| } |
| |
| void RtpTransportControllerSend::OnNetworkChanged(uint32_t bitrate_bps, |
| uint8_t fraction_loss, |
| int64_t rtt_ms, |
| int64_t probing_interval_ms) { |
| // TODO(srte): Skip this step when old SendSideCongestionController is |
| // deprecated. |
| TargetTransferRate msg; |
| msg.at_time = Timestamp::ms(clock_->TimeInMilliseconds()); |
| msg.target_rate = DataRate::bps(bitrate_bps); |
| msg.network_estimate.at_time = msg.at_time; |
| msg.network_estimate.bwe_period = TimeDelta::ms(probing_interval_ms); |
| uint32_t bandwidth_bps; |
| if (send_side_cc_->AvailableBandwidth(&bandwidth_bps)) |
| msg.network_estimate.bandwidth = DataRate::bps(bandwidth_bps); |
| msg.network_estimate.loss_rate_ratio = fraction_loss / 255.0; |
| msg.network_estimate.round_trip_time = TimeDelta::ms(rtt_ms); |
| |
| retransmission_rate_limiter_.SetMaxRate(bandwidth_bps); |
| |
| if (!task_queue_.IsCurrent()) { |
| task_queue_.PostTask([this, msg] { |
| rtc::CritScope cs(&observer_crit_); |
| // We won't register as observer until we have an observers. |
| RTC_DCHECK(observer_ != nullptr); |
| observer_->OnTargetTransferRate(msg); |
| }); |
| } else { |
| rtc::CritScope cs(&observer_crit_); |
| // We won't register as observer until we have an observers. |
| RTC_DCHECK(observer_ != nullptr); |
| observer_->OnTargetTransferRate(msg); |
| } |
| } |
| |
| rtc::TaskQueue* RtpTransportControllerSend::GetWorkerQueue() { |
| return &task_queue_; |
| } |
| |
| PacketRouter* RtpTransportControllerSend::packet_router() { |
| return &packet_router_; |
| } |
| |
| TransportFeedbackObserver* |
| RtpTransportControllerSend::transport_feedback_observer() { |
| return send_side_cc_.get(); |
| } |
| |
| RtpPacketSender* RtpTransportControllerSend::packet_sender() { |
| return &pacer_; |
| } |
| |
| const RtpKeepAliveConfig& RtpTransportControllerSend::keepalive_config() const { |
| return keepalive_; |
| } |
| |
| void RtpTransportControllerSend::SetAllocatedSendBitrateLimits( |
| int min_send_bitrate_bps, |
| int max_padding_bitrate_bps, |
| int max_total_bitrate_bps) { |
| send_side_cc_->SetAllocatedSendBitrateLimits( |
| min_send_bitrate_bps, max_padding_bitrate_bps, max_total_bitrate_bps); |
| } |
| |
| void RtpTransportControllerSend::SetKeepAliveConfig( |
| const RtpKeepAliveConfig& config) { |
| keepalive_ = config; |
| } |
| void RtpTransportControllerSend::SetPacingFactor(float pacing_factor) { |
| send_side_cc_->SetPacingFactor(pacing_factor); |
| } |
| void RtpTransportControllerSend::SetQueueTimeLimit(int limit_ms) { |
| pacer_.SetQueueTimeLimit(limit_ms); |
| } |
| CallStatsObserver* RtpTransportControllerSend::GetCallStatsObserver() { |
| return send_side_cc_.get(); |
| } |
| void RtpTransportControllerSend::RegisterPacketFeedbackObserver( |
| PacketFeedbackObserver* observer) { |
| send_side_cc_->RegisterPacketFeedbackObserver(observer); |
| } |
| void RtpTransportControllerSend::DeRegisterPacketFeedbackObserver( |
| PacketFeedbackObserver* observer) { |
| send_side_cc_->DeRegisterPacketFeedbackObserver(observer); |
| } |
| |
| void RtpTransportControllerSend::RegisterTargetTransferRateObserver( |
| TargetTransferRateObserver* observer) { |
| { |
| rtc::CritScope cs(&observer_crit_); |
| RTC_DCHECK(observer_ == nullptr); |
| observer_ = observer; |
| } |
| send_side_cc_->RegisterNetworkObserver(this); |
| } |
| void RtpTransportControllerSend::OnNetworkRouteChanged( |
| const std::string& transport_name, |
| const rtc::NetworkRoute& network_route) { |
| // Check if the network route is connected. |
| if (!network_route.connected) { |
| RTC_LOG(LS_INFO) << "Transport " << transport_name << " is disconnected"; |
| // TODO(honghaiz): Perhaps handle this in SignalChannelNetworkState and |
| // consider merging these two methods. |
| return; |
| } |
| |
| // Check whether the network route has changed on each transport. |
| auto result = |
| network_routes_.insert(std::make_pair(transport_name, network_route)); |
| auto kv = result.first; |
| bool inserted = result.second; |
| if (inserted) { |
| // No need to reset BWE if this is the first time the network connects. |
| return; |
| } |
| if (kv->second != network_route) { |
| kv->second = network_route; |
| BitrateConstraints bitrate_config = bitrate_configurator_.GetConfig(); |
| RTC_LOG(LS_INFO) << "Network route changed on transport " << transport_name |
| << ": new local network id " |
| << network_route.local_network_id |
| << " new remote network id " |
| << network_route.remote_network_id |
| << " Reset bitrates to min: " |
| << bitrate_config.min_bitrate_bps |
| << " bps, start: " << bitrate_config.start_bitrate_bps |
| << " bps, max: " << bitrate_config.max_bitrate_bps |
| << " bps."; |
| RTC_DCHECK_GT(bitrate_config.start_bitrate_bps, 0); |
| send_side_cc_->OnNetworkRouteChanged( |
| network_route, bitrate_config.start_bitrate_bps, |
| bitrate_config.min_bitrate_bps, bitrate_config.max_bitrate_bps); |
| } |
| } |
| void RtpTransportControllerSend::OnNetworkAvailability(bool network_available) { |
| send_side_cc_->SignalNetworkState(network_available ? kNetworkUp |
| : kNetworkDown); |
| for (auto& rtp_sender : video_rtp_senders_) { |
| rtp_sender->OnNetworkAvailability(network_available); |
| } |
| } |
| RtcpBandwidthObserver* RtpTransportControllerSend::GetBandwidthObserver() { |
| return send_side_cc_->GetBandwidthObserver(); |
| } |
| int64_t RtpTransportControllerSend::GetPacerQueuingDelayMs() const { |
| return pacer_.QueueInMs(); |
| } |
| int64_t RtpTransportControllerSend::GetFirstPacketTimeMs() const { |
| return pacer_.FirstSentPacketTimeMs(); |
| } |
| void RtpTransportControllerSend::SetPerPacketFeedbackAvailable(bool available) { |
| send_side_cc_->SetPerPacketFeedbackAvailable(available); |
| } |
| void RtpTransportControllerSend::EnablePeriodicAlrProbing(bool enable) { |
| send_side_cc_->EnablePeriodicAlrProbing(enable); |
| } |
| void RtpTransportControllerSend::OnSentPacket( |
| const rtc::SentPacket& sent_packet) { |
| send_side_cc_->OnSentPacket(sent_packet); |
| } |
| |
| void RtpTransportControllerSend::SetSdpBitrateParameters( |
| const BitrateConstraints& constraints) { |
| absl::optional<BitrateConstraints> updated = |
| bitrate_configurator_.UpdateWithSdpParameters(constraints); |
| if (updated.has_value()) { |
| send_side_cc_->SetBweBitrates(updated->min_bitrate_bps, |
| updated->start_bitrate_bps, |
| updated->max_bitrate_bps); |
| } else { |
| RTC_LOG(LS_VERBOSE) |
| << "WebRTC.RtpTransportControllerSend.SetSdpBitrateParameters: " |
| << "nothing to update"; |
| } |
| } |
| |
| void RtpTransportControllerSend::SetClientBitratePreferences( |
| const BitrateSettings& preferences) { |
| absl::optional<BitrateConstraints> updated = |
| bitrate_configurator_.UpdateWithClientPreferences(preferences); |
| if (updated.has_value()) { |
| send_side_cc_->SetBweBitrates(updated->min_bitrate_bps, |
| updated->start_bitrate_bps, |
| updated->max_bitrate_bps); |
| } else { |
| RTC_LOG(LS_VERBOSE) |
| << "WebRTC.RtpTransportControllerSend.SetClientBitratePreferences: " |
| << "nothing to update"; |
| } |
| } |
| |
| void RtpTransportControllerSend::SetAllocatedBitrateWithoutFeedback( |
| uint32_t bitrate_bps) { |
| // Audio transport feedback will not be reported in this mode, instead update |
| // acknowledged bitrate estimator with the bitrate allocated for audio. |
| if (field_trial::IsEnabled("WebRTC-Audio-ABWENoTWCC")) { |
| // TODO(srte): Make sure it's safe to always report this and remove the |
| // field trial check. |
| send_side_cc_->SetAllocatedBitrateWithoutFeedback(bitrate_bps); |
| } |
| } |
| |
| void RtpTransportControllerSend::OnTransportOverheadChanged( |
| size_t transport_overhead_bytes_per_packet) { |
| if (transport_overhead_bytes_per_packet >= kMaxOverheadBytes) { |
| RTC_LOG(LS_ERROR) << "Transport overhead exceeds " << kMaxOverheadBytes; |
| return; |
| } |
| |
| // TODO(holmer): Call AudioRtpSenders when they have been moved to |
| // RtpTransportControllerSend. |
| for (auto& rtp_video_sender : video_rtp_senders_) { |
| rtp_video_sender->OnTransportOverheadChanged( |
| transport_overhead_bytes_per_packet); |
| } |
| } |
| } // namespace webrtc |