call/rtp_transport_controller_send.cc - src.git - Git at Google

 /*
  *  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
	/*
	* 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