modules/pacing/packet_router.cc - src - Git at Google

 /*
  *  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.
  */

 #include "modules/pacing/packet_router.h"

 #include <algorithm>
 #include <cstdint>
 #include <limits>
 #include <utility>

 #include "absl/types/optional.h"
 #include "modules/rtp_rtcp/include/rtp_rtcp.h"
 #include "modules/rtp_rtcp/include/rtp_rtcp_defines.h"
 #include "modules/rtp_rtcp/source/rtcp_packet/transport_feedback.h"
 #include "rtc_base/atomic_ops.h"
 #include "rtc_base/checks.h"
 #include "rtc_base/logging.h"
 #include "rtc_base/time_utils.h"

 namespace webrtc {
 namespace {

 constexpr int kRembSendIntervalMs = 200;

 }  // namespace

 PacketRouter::PacketRouter()
     : last_send_module_(nullptr),
       last_remb_time_ms_(rtc::TimeMillis()),
       last_send_bitrate_bps_(0),
       bitrate_bps_(0),
       max_bitrate_bps_(std::numeric_limits<decltype(max_bitrate_bps_)>::max()),
       active_remb_module_(nullptr),
       transport_seq_(0) {}

 PacketRouter::~PacketRouter() {
   RTC_DCHECK(rtp_send_modules_.empty());
   RTC_DCHECK(rtcp_feedback_senders_.empty());
   RTC_DCHECK(sender_remb_candidates_.empty());
   RTC_DCHECK(receiver_remb_candidates_.empty());
   RTC_DCHECK(active_remb_module_ == nullptr);
 }

 void PacketRouter::AddSendRtpModule(RtpRtcp* rtp_module, bool remb_candidate) {
   rtc::CritScope cs(&modules_crit_);
   RTC_DCHECK(std::find(rtp_send_modules_.begin(), rtp_send_modules_.end(),
                        rtp_module) == rtp_send_modules_.end());
   // Put modules which can use regular payload packets (over rtx) instead of
   // padding first as it's less of a waste
   if ((rtp_module->RtxSendStatus() & kRtxRedundantPayloads) > 0) {
     rtp_send_modules_.push_front(rtp_module);
   } else {
     rtp_send_modules_.push_back(rtp_module);
   }

   if (remb_candidate) {
     AddRembModuleCandidate(rtp_module, /* media_sender = */ true);
   }
 }

 void PacketRouter::RemoveSendRtpModule(RtpRtcp* rtp_module) {
   rtc::CritScope cs(&modules_crit_);
   MaybeRemoveRembModuleCandidate(rtp_module, /* media_sender = */ true);
   auto it =
       std::find(rtp_send_modules_.begin(), rtp_send_modules_.end(), rtp_module);
   RTC_DCHECK(it != rtp_send_modules_.end());
   rtp_send_modules_.erase(it);
   if (last_send_module_ == rtp_module) {
     last_send_module_ = nullptr;
   }
 }

 void PacketRouter::AddReceiveRtpModule(RtcpFeedbackSenderInterface* rtcp_sender,
                                        bool remb_candidate) {
   rtc::CritScope cs(&modules_crit_);
   RTC_DCHECK(std::find(rtcp_feedback_senders_.begin(),
                        rtcp_feedback_senders_.end(),
                        rtcp_sender) == rtcp_feedback_senders_.end());

   rtcp_feedback_senders_.push_back(rtcp_sender);

   if (remb_candidate) {
     AddRembModuleCandidate(rtcp_sender, /* media_sender = */ false);
   }
 }

 void PacketRouter::RemoveReceiveRtpModule(
     RtcpFeedbackSenderInterface* rtcp_sender) {
   rtc::CritScope cs(&modules_crit_);
   MaybeRemoveRembModuleCandidate(rtcp_sender, /* media_sender = */ false);
   auto it = std::find(rtcp_feedback_senders_.begin(),
                       rtcp_feedback_senders_.end(), rtcp_sender);
   RTC_DCHECK(it != rtcp_feedback_senders_.end());
   rtcp_feedback_senders_.erase(it);
 }

 RtpPacketSendResult PacketRouter::TimeToSendPacket(
     uint32_t ssrc,
     uint16_t sequence_number,
     int64_t capture_timestamp,
     bool retransmission,
     const PacedPacketInfo& pacing_info) {
   rtc::CritScope cs(&modules_crit_);
   for (auto* rtp_module : rtp_send_modules_) {
     if (!rtp_module->SendingMedia()) {
       continue;
     }
     if (ssrc == rtp_module->SSRC() || ssrc == rtp_module->FlexfecSsrc()) {
       if ((rtp_module->RtxSendStatus() & kRtxRedundantPayloads) &&
           rtp_module->HasBweExtensions()) {
         // This is now the last module to send media, and has the desired
         // properties needed for payload based padding. Cache it for later use.
         last_send_module_ = rtp_module;
       }
       return rtp_module->TimeToSendPacket(ssrc, sequence_number,
                                           capture_timestamp, retransmission,
                                           pacing_info);
     }
   }
   return RtpPacketSendResult::kPacketNotFound;
 }

 void PacketRouter::SendPacket(std::unique_ptr<RtpPacketToSend> packet,
                               const PacedPacketInfo& cluster_info) {
   rtc::CritScope cs(&modules_crit_);
   for (auto* rtp_module : rtp_send_modules_) {
     if (rtp_module->TrySendPacket(packet.get(), cluster_info)) {
       const bool can_send_padding =
           (rtp_module->RtxSendStatus() & kRtxRedundantPayloads) &&
           rtp_module->HasBweExtensions();
       if (can_send_padding) {
         // This is now the last module to send media, and has the desired
         // properties needed for payload based padding. Cache it for later use.
         last_send_module_ = rtp_module;
       }
       return;
     }
   }

   RTC_LOG(LS_WARNING) << "Failed to send packet, matching RTP module not found "
                          "or transport error. SSRC = "
                       << packet->Ssrc() << ", sequence number "
                       << packet->SequenceNumber();
 }

 size_t PacketRouter::TimeToSendPadding(size_t bytes_to_send,
                                        const PacedPacketInfo& pacing_info) {
   size_t total_bytes_sent = 0;
   rtc::CritScope cs(&modules_crit_);
   // First try on the last rtp module to have sent media. This increases the
   // the chance that any payload based padding will be useful as it will be
   // somewhat distributed over modules according the packet rate, even if it
   // will be more skewed towards the highest bitrate stream. At the very least
   // this prevents sending payload padding on a disabled stream where it's
   // guaranteed not to be useful.
   if (last_send_module_ != nullptr) {
     RTC_DCHECK(std::find(rtp_send_modules_.begin(), rtp_send_modules_.end(),
                          last_send_module_) != rtp_send_modules_.end());
     RTC_DCHECK(last_send_module_->HasBweExtensions());
     total_bytes_sent += last_send_module_->TimeToSendPadding(
         bytes_to_send - total_bytes_sent, pacing_info);
     if (total_bytes_sent >= bytes_to_send) {
       return total_bytes_sent;
     }
   }

   // Rtp modules are ordered by which stream can most benefit from padding.
   for (RtpRtcp* module : rtp_send_modules_) {
     if (module->SendingMedia() && module->HasBweExtensions()) {
       size_t bytes_sent = module->TimeToSendPadding(
           bytes_to_send - total_bytes_sent, pacing_info);
       total_bytes_sent += bytes_sent;
       if (total_bytes_sent >= bytes_to_send)
         break;
     }
   }
   return total_bytes_sent;
 }

 void PacketRouter::SetTransportWideSequenceNumber(uint16_t sequence_number) {
   rtc::AtomicOps::ReleaseStore(&transport_seq_, sequence_number);
 }

 uint16_t PacketRouter::AllocateSequenceNumber() {
   int prev_seq = rtc::AtomicOps::AcquireLoad(&transport_seq_);
   int desired_prev_seq;
   int new_seq;
   do {
     desired_prev_seq = prev_seq;
     new_seq = (desired_prev_seq + 1) & 0xFFFF;
     // Note: CompareAndSwap returns the actual value of transport_seq at the
     // time the CAS operation was executed. Thus, if prev_seq is returned, the
     // operation was successful - otherwise we need to retry. Saving the
     // return value saves us a load on retry.
     prev_seq = rtc::AtomicOps::CompareAndSwap(&transport_seq_, desired_prev_seq,
                                               new_seq);
   } while (prev_seq != desired_prev_seq);

   return new_seq;
 }

 void PacketRouter::OnReceiveBitrateChanged(const std::vector<uint32_t>& ssrcs,
                                            uint32_t bitrate_bps) {
   // % threshold for if we should send a new REMB asap.
   const int64_t kSendThresholdPercent = 97;
   // TODO(danilchap): Remove receive_bitrate_bps variable and the cast
   // when OnReceiveBitrateChanged takes bitrate as int64_t.
   int64_t receive_bitrate_bps = static_cast<int64_t>(bitrate_bps);

   int64_t now_ms = rtc::TimeMillis();
   {
     rtc::CritScope lock(&remb_crit_);

     // If we already have an estimate, check if the new total estimate is below
     // kSendThresholdPercent of the previous estimate.
     if (last_send_bitrate_bps_ > 0) {
       int64_t new_remb_bitrate_bps =
           last_send_bitrate_bps_ - bitrate_bps_ + receive_bitrate_bps;

       if (new_remb_bitrate_bps <
           kSendThresholdPercent * last_send_bitrate_bps_ / 100) {
         // The new bitrate estimate is less than kSendThresholdPercent % of the
         // last report. Send a REMB asap.
         last_remb_time_ms_ = now_ms - kRembSendIntervalMs;
       }
     }
     bitrate_bps_ = receive_bitrate_bps;

     if (now_ms - last_remb_time_ms_ < kRembSendIntervalMs) {
       return;
     }
     // NOTE: Updated if we intend to send the data; we might not have
     // a module to actually send it.
     last_remb_time_ms_ = now_ms;
     last_send_bitrate_bps_ = receive_bitrate_bps;
     // Cap the value to send in remb with configured value.
     receive_bitrate_bps = std::min(receive_bitrate_bps, max_bitrate_bps_);
   }
   SendRemb(receive_bitrate_bps, ssrcs);
 }

 void PacketRouter::SetMaxDesiredReceiveBitrate(int64_t bitrate_bps) {
   RTC_DCHECK_GE(bitrate_bps, 0);
   {
     rtc::CritScope lock(&remb_crit_);
     max_bitrate_bps_ = bitrate_bps;
     if (rtc::TimeMillis() - last_remb_time_ms_ < kRembSendIntervalMs &&
         last_send_bitrate_bps_ > 0 &&
         last_send_bitrate_bps_ <= max_bitrate_bps_) {
       // Recent measured bitrate is already below the cap.
       return;
     }
   }
   SendRemb(bitrate_bps, /*ssrcs=*/{});
 }

 bool PacketRouter::SendRemb(int64_t bitrate_bps,
                             const std::vector<uint32_t>& ssrcs) {
   rtc::CritScope lock(&modules_crit_);

   if (!active_remb_module_) {
     return false;
   }

   // The Add* and Remove* methods above ensure that REMB is disabled on all
   // other modules, because otherwise, they will send REMB with stale info.
   active_remb_module_->SetRemb(bitrate_bps, ssrcs);

   return true;
 }

 bool PacketRouter::SendTransportFeedback(rtcp::TransportFeedback* packet) {
   rtc::CritScope cs(&modules_crit_);
   // Prefer send modules.
   for (auto* rtp_module : rtp_send_modules_) {
     packet->SetSenderSsrc(rtp_module->SSRC());
     if (rtp_module->SendFeedbackPacket(*packet)) {
       return true;
     }
   }
   for (auto* rtcp_sender : rtcp_feedback_senders_) {
     packet->SetSenderSsrc(rtcp_sender->SSRC());
     if (rtcp_sender->SendFeedbackPacket(*packet)) {
       return true;
     }
   }
   return false;
 }

 void PacketRouter::AddRembModuleCandidate(
     RtcpFeedbackSenderInterface* candidate_module,
     bool media_sender) {
   RTC_DCHECK(candidate_module);
   std::vector<RtcpFeedbackSenderInterface*>& candidates =
       media_sender ? sender_remb_candidates_ : receiver_remb_candidates_;
   RTC_DCHECK(std::find(candidates.cbegin(), candidates.cend(),
                        candidate_module) == candidates.cend());
   candidates.push_back(candidate_module);
   DetermineActiveRembModule();
 }

 void PacketRouter::MaybeRemoveRembModuleCandidate(
     RtcpFeedbackSenderInterface* candidate_module,
     bool media_sender) {
   RTC_DCHECK(candidate_module);
   std::vector<RtcpFeedbackSenderInterface*>& candidates =
       media_sender ? sender_remb_candidates_ : receiver_remb_candidates_;
   auto it = std::find(candidates.begin(), candidates.end(), candidate_module);

   if (it == candidates.end()) {
     return;  // Function called due to removal of non-REMB-candidate module.
   }

   if (*it == active_remb_module_) {
     UnsetActiveRembModule();
   }
   candidates.erase(it);
   DetermineActiveRembModule();
 }

 void PacketRouter::UnsetActiveRembModule() {
   RTC_CHECK(active_remb_module_);
   active_remb_module_->UnsetRemb();
   active_remb_module_ = nullptr;
 }

 void PacketRouter::DetermineActiveRembModule() {
   // Sender modules take precedence over receiver modules, because SRs (sender
   // reports) are sent more frequently than RR (receiver reports).
   // When adding the first sender module, we should change the active REMB
   // module to be that. Otherwise, we remain with the current active module.

   RtcpFeedbackSenderInterface* new_active_remb_module;

   if (!sender_remb_candidates_.empty()) {
     new_active_remb_module = sender_remb_candidates_.front();
   } else if (!receiver_remb_candidates_.empty()) {
     new_active_remb_module = receiver_remb_candidates_.front();
   } else {
     new_active_remb_module = nullptr;
   }

   if (new_active_remb_module != active_remb_module_ && active_remb_module_) {
     UnsetActiveRembModule();
   }

   active_remb_module_ = new_active_remb_module;
 }

 }  // namespace webrtc
	/*
	* 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.
	*/

	#include "modules/pacing/packet_router.h"

	#include <algorithm>
	#include <cstdint>
	#include <limits>
	#include <utility>

	#include "absl/types/optional.h"
	#include "modules/rtp_rtcp/include/rtp_rtcp.h"
	#include "modules/rtp_rtcp/include/rtp_rtcp_defines.h"
	#include "modules/rtp_rtcp/source/rtcp_packet/transport_feedback.h"
	#include "rtc_base/atomic_ops.h"
	#include "rtc_base/checks.h"
	#include "rtc_base/logging.h"
	#include "rtc_base/time_utils.h"

	namespace webrtc {
	namespace {

	constexpr int kRembSendIntervalMs = 200;

	} // namespace

	PacketRouter::PacketRouter()
	: last_send_module_(nullptr),
	last_remb_time_ms_(rtc::TimeMillis()),
	last_send_bitrate_bps_(0),
	bitrate_bps_(0),
	max_bitrate_bps_(std::numeric_limits<decltype(max_bitrate_bps_)>::max()),
	active_remb_module_(nullptr),
	transport_seq_(0) {}

	PacketRouter::~PacketRouter() {
	RTC_DCHECK(rtp_send_modules_.empty());
	RTC_DCHECK(rtcp_feedback_senders_.empty());
	RTC_DCHECK(sender_remb_candidates_.empty());
	RTC_DCHECK(receiver_remb_candidates_.empty());
	RTC_DCHECK(active_remb_module_ == nullptr);
	}

	void PacketRouter::AddSendRtpModule(RtpRtcp* rtp_module, bool remb_candidate) {
	rtc::CritScope cs(&modules_crit_);
	RTC_DCHECK(std::find(rtp_send_modules_.begin(), rtp_send_modules_.end(),
	rtp_module) == rtp_send_modules_.end());
	// Put modules which can use regular payload packets (over rtx) instead of
	// padding first as it's less of a waste
	if ((rtp_module->RtxSendStatus() & kRtxRedundantPayloads) > 0) {
	rtp_send_modules_.push_front(rtp_module);
	} else {
	rtp_send_modules_.push_back(rtp_module);
	}

	if (remb_candidate) {
	AddRembModuleCandidate(rtp_module, /* media_sender = */ true);
	}
	}

	void PacketRouter::RemoveSendRtpModule(RtpRtcp* rtp_module) {
	rtc::CritScope cs(&modules_crit_);
	MaybeRemoveRembModuleCandidate(rtp_module, /* media_sender = */ true);
	auto it =
	std::find(rtp_send_modules_.begin(), rtp_send_modules_.end(), rtp_module);
	RTC_DCHECK(it != rtp_send_modules_.end());
	rtp_send_modules_.erase(it);
	if (last_send_module_ == rtp_module) {
	last_send_module_ = nullptr;
	}
	}

	void PacketRouter::AddReceiveRtpModule(RtcpFeedbackSenderInterface* rtcp_sender,
	bool remb_candidate) {
	rtc::CritScope cs(&modules_crit_);
	RTC_DCHECK(std::find(rtcp_feedback_senders_.begin(),
	rtcp_feedback_senders_.end(),
	rtcp_sender) == rtcp_feedback_senders_.end());

	rtcp_feedback_senders_.push_back(rtcp_sender);

	if (remb_candidate) {
	AddRembModuleCandidate(rtcp_sender, /* media_sender = */ false);
	}
	}

	void PacketRouter::RemoveReceiveRtpModule(
	RtcpFeedbackSenderInterface* rtcp_sender) {
	rtc::CritScope cs(&modules_crit_);
	MaybeRemoveRembModuleCandidate(rtcp_sender, /* media_sender = */ false);
	auto it = std::find(rtcp_feedback_senders_.begin(),
	rtcp_feedback_senders_.end(), rtcp_sender);
	RTC_DCHECK(it != rtcp_feedback_senders_.end());
	rtcp_feedback_senders_.erase(it);
	}

	RtpPacketSendResult PacketRouter::TimeToSendPacket(
	uint32_t ssrc,
	uint16_t sequence_number,
	int64_t capture_timestamp,
	bool retransmission,
	const PacedPacketInfo& pacing_info) {
	rtc::CritScope cs(&modules_crit_);
	for (auto* rtp_module : rtp_send_modules_) {
	if (!rtp_module->SendingMedia()) {
	continue;
	}
	if (ssrc == rtp_module->SSRC() \|\| ssrc == rtp_module->FlexfecSsrc()) {
	if ((rtp_module->RtxSendStatus() & kRtxRedundantPayloads) &&
	rtp_module->HasBweExtensions()) {
	// This is now the last module to send media, and has the desired
	// properties needed for payload based padding. Cache it for later use.
	last_send_module_ = rtp_module;
	}
	return rtp_module->TimeToSendPacket(ssrc, sequence_number,
	capture_timestamp, retransmission,
	pacing_info);
	}
	}
	return RtpPacketSendResult::kPacketNotFound;
	}

	void PacketRouter::SendPacket(std::unique_ptr<RtpPacketToSend> packet,
	const PacedPacketInfo& cluster_info) {
	rtc::CritScope cs(&modules_crit_);
	for (auto* rtp_module : rtp_send_modules_) {
	if (rtp_module->TrySendPacket(packet.get(), cluster_info)) {
	const bool can_send_padding =
	(rtp_module->RtxSendStatus() & kRtxRedundantPayloads) &&
	rtp_module->HasBweExtensions();
	if (can_send_padding) {
	// This is now the last module to send media, and has the desired
	// properties needed for payload based padding. Cache it for later use.
	last_send_module_ = rtp_module;
	}
	return;
	}
	}

	RTC_LOG(LS_WARNING) << "Failed to send packet, matching RTP module not found "
	"or transport error. SSRC = "
	<< packet->Ssrc() << ", sequence number "
	<< packet->SequenceNumber();
	}

	size_t PacketRouter::TimeToSendPadding(size_t bytes_to_send,
	const PacedPacketInfo& pacing_info) {
	size_t total_bytes_sent = 0;
	rtc::CritScope cs(&modules_crit_);
	// First try on the last rtp module to have sent media. This increases the
	// the chance that any payload based padding will be useful as it will be
	// somewhat distributed over modules according the packet rate, even if it
	// will be more skewed towards the highest bitrate stream. At the very least
	// this prevents sending payload padding on a disabled stream where it's
	// guaranteed not to be useful.
	if (last_send_module_ != nullptr) {
	RTC_DCHECK(std::find(rtp_send_modules_.begin(), rtp_send_modules_.end(),
	last_send_module_) != rtp_send_modules_.end());
	RTC_DCHECK(last_send_module_->HasBweExtensions());
	total_bytes_sent += last_send_module_->TimeToSendPadding(
	bytes_to_send - total_bytes_sent, pacing_info);
	if (total_bytes_sent >= bytes_to_send) {
	return total_bytes_sent;
	}
	}

	// Rtp modules are ordered by which stream can most benefit from padding.
	for (RtpRtcp* module : rtp_send_modules_) {
	if (module->SendingMedia() && module->HasBweExtensions()) {
	size_t bytes_sent = module->TimeToSendPadding(
	bytes_to_send - total_bytes_sent, pacing_info);
	total_bytes_sent += bytes_sent;
	if (total_bytes_sent >= bytes_to_send)
	break;
	}
	}
	return total_bytes_sent;
	}

	void PacketRouter::SetTransportWideSequenceNumber(uint16_t sequence_number) {
	rtc::AtomicOps::ReleaseStore(&transport_seq_, sequence_number);
	}

	uint16_t PacketRouter::AllocateSequenceNumber() {
	int prev_seq = rtc::AtomicOps::AcquireLoad(&transport_seq_);
	int desired_prev_seq;
	int new_seq;
	do {
	desired_prev_seq = prev_seq;
	new_seq = (desired_prev_seq + 1) & 0xFFFF;
	// Note: CompareAndSwap returns the actual value of transport_seq at the
	// time the CAS operation was executed. Thus, if prev_seq is returned, the
	// operation was successful - otherwise we need to retry. Saving the
	// return value saves us a load on retry.
	prev_seq = rtc::AtomicOps::CompareAndSwap(&transport_seq_, desired_prev_seq,
	new_seq);
	} while (prev_seq != desired_prev_seq);

	return new_seq;
	}

	void PacketRouter::OnReceiveBitrateChanged(const std::vector<uint32_t>& ssrcs,
	uint32_t bitrate_bps) {
	// % threshold for if we should send a new REMB asap.
	const int64_t kSendThresholdPercent = 97;
	// TODO(danilchap): Remove receive_bitrate_bps variable and the cast
	// when OnReceiveBitrateChanged takes bitrate as int64_t.
	int64_t receive_bitrate_bps = static_cast<int64_t>(bitrate_bps);

	int64_t now_ms = rtc::TimeMillis();
	{
	rtc::CritScope lock(&remb_crit_);

	// If we already have an estimate, check if the new total estimate is below
	// kSendThresholdPercent of the previous estimate.
	if (last_send_bitrate_bps_ > 0) {
	int64_t new_remb_bitrate_bps =
	last_send_bitrate_bps_ - bitrate_bps_ + receive_bitrate_bps;

	if (new_remb_bitrate_bps <
	kSendThresholdPercent * last_send_bitrate_bps_ / 100) {
	// The new bitrate estimate is less than kSendThresholdPercent % of the
	// last report. Send a REMB asap.
	last_remb_time_ms_ = now_ms - kRembSendIntervalMs;
	}
	}
	bitrate_bps_ = receive_bitrate_bps;

	if (now_ms - last_remb_time_ms_ < kRembSendIntervalMs) {
	return;
	}
	// NOTE: Updated if we intend to send the data; we might not have
	// a module to actually send it.
	last_remb_time_ms_ = now_ms;
	last_send_bitrate_bps_ = receive_bitrate_bps;
	// Cap the value to send in remb with configured value.
	receive_bitrate_bps = std::min(receive_bitrate_bps, max_bitrate_bps_);
	}
	SendRemb(receive_bitrate_bps, ssrcs);
	}

	void PacketRouter::SetMaxDesiredReceiveBitrate(int64_t bitrate_bps) {
	RTC_DCHECK_GE(bitrate_bps, 0);
	{
	rtc::CritScope lock(&remb_crit_);
	max_bitrate_bps_ = bitrate_bps;
	if (rtc::TimeMillis() - last_remb_time_ms_ < kRembSendIntervalMs &&
	last_send_bitrate_bps_ > 0 &&
	last_send_bitrate_bps_ <= max_bitrate_bps_) {
	// Recent measured bitrate is already below the cap.
	return;
	}
	}
	SendRemb(bitrate_bps, /ssrcs=/{});
	}

	bool PacketRouter::SendRemb(int64_t bitrate_bps,
	const std::vector<uint32_t>& ssrcs) {
	rtc::CritScope lock(&modules_crit_);

	if (!active_remb_module_) {
	return false;
	}

	// The Add* and Remove* methods above ensure that REMB is disabled on all
	// other modules, because otherwise, they will send REMB with stale info.
	active_remb_module_->SetRemb(bitrate_bps, ssrcs);

	return true;
	}

	bool PacketRouter::SendTransportFeedback(rtcp::TransportFeedback* packet) {
	rtc::CritScope cs(&modules_crit_);
	// Prefer send modules.
	for (auto* rtp_module : rtp_send_modules_) {
	packet->SetSenderSsrc(rtp_module->SSRC());
	if (rtp_module->SendFeedbackPacket(*packet)) {
	return true;
	}
	}
	for (auto* rtcp_sender : rtcp_feedback_senders_) {
	packet->SetSenderSsrc(rtcp_sender->SSRC());
	if (rtcp_sender->SendFeedbackPacket(*packet)) {
	return true;
	}
	}
	return false;
	}

	void PacketRouter::AddRembModuleCandidate(
	RtcpFeedbackSenderInterface* candidate_module,
	bool media_sender) {
	RTC_DCHECK(candidate_module);
	std::vector<RtcpFeedbackSenderInterface*>& candidates =
	media_sender ? sender_remb_candidates_ : receiver_remb_candidates_;
	RTC_DCHECK(std::find(candidates.cbegin(), candidates.cend(),
	candidate_module) == candidates.cend());
	candidates.push_back(candidate_module);
	DetermineActiveRembModule();
	}

	void PacketRouter::MaybeRemoveRembModuleCandidate(
	RtcpFeedbackSenderInterface* candidate_module,
	bool media_sender) {
	RTC_DCHECK(candidate_module);
	std::vector<RtcpFeedbackSenderInterface*>& candidates =
	media_sender ? sender_remb_candidates_ : receiver_remb_candidates_;
	auto it = std::find(candidates.begin(), candidates.end(), candidate_module);

	if (it == candidates.end()) {
	return; // Function called due to removal of non-REMB-candidate module.
	}

	if (*it == active_remb_module_) {
	UnsetActiveRembModule();
	}
	candidates.erase(it);
	DetermineActiveRembModule();
	}

	void PacketRouter::UnsetActiveRembModule() {
	RTC_CHECK(active_remb_module_);
	active_remb_module_->UnsetRemb();
	active_remb_module_ = nullptr;
	}

	void PacketRouter::DetermineActiveRembModule() {
	// Sender modules take precedence over receiver modules, because SRs (sender
	// reports) are sent more frequently than RR (receiver reports).
	// When adding the first sender module, we should change the active REMB
	// module to be that. Otherwise, we remain with the current active module.

	RtcpFeedbackSenderInterface* new_active_remb_module;

	if (!sender_remb_candidates_.empty()) {
	new_active_remb_module = sender_remb_candidates_.front();
	} else if (!receiver_remb_candidates_.empty()) {
	new_active_remb_module = receiver_remb_candidates_.front();
	} else {
	new_active_remb_module = nullptr;
	}

	if (new_active_remb_module != active_remb_module_ && active_remb_module_) {
	UnsetActiveRembModule();
	}

	active_remb_module_ = new_active_remb_module;
	}

	} // namespace webrtc