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This reverts commit65792c5a5c
. Reason for revert: <INSERT REASONING HERE> Original change's description: > Revert "Revert "Reland "Moved congestion controller to task queue.""" > > This reverts commit4e849f6925
. > > Reason for revert: <INSERT REASONING HERE> > > Original change's description: > > Revert "Reland "Moved congestion controller to task queue."" > > > > This reverts commit57daeb7ac7
. > > > > Reason for revert: Cause increased congestion and deadlocks in downstream project > > > > Original change's description: > > > Reland "Moved congestion controller to task queue." > > > > > > This is a reland of0cbcba7ea0
. > > > > > > Original change's description: > > > > Moved congestion controller to task queue. > > > > > > > > The goal of this work is to make it easier to experiment with the > > > > bandwidth estimation implementation. For this reason network control > > > > functionality is moved from SendSideCongestionController(SSCC), > > > > PacedSender and BitrateController to the newly created > > > > GoogCcNetworkController which implements the newly created > > > > NetworkControllerInterface. This allows the implementation to be > > > > replaced at runtime in the future. > > > > > > > > This is the first part of a split of a larger CL, see: > > > > https://webrtc-review.googlesource.com/c/src/+/39788/8 > > > > For further explanations. > > > > > > > > Bug: webrtc:8415 > > > > Change-Id: I770189c04cc31b313bd4e57821acff55fbcb1ad3 > > > > Reviewed-on: https://webrtc-review.googlesource.com/43840 > > > > Commit-Queue: Sebastian Jansson <srte@webrtc.org> > > > > Reviewed-by: Björn Terelius <terelius@webrtc.org> > > > > Reviewed-by: Stefan Holmer <stefan@webrtc.org> > > > > Cr-Commit-Position: refs/heads/master@{#21868} > > > > > > Bug: webrtc:8415 > > > Change-Id: I1d1756a30deed5b421b1c91c1918a13b6bb455da > > > Reviewed-on: https://webrtc-review.googlesource.com/48000 > > > Reviewed-by: Stefan Holmer <stefan@webrtc.org> > > > Commit-Queue: Sebastian Jansson <srte@webrtc.org> > > > Cr-Commit-Position: refs/heads/master@{#21899} > > > > TBR=terelius@webrtc.org,stefan@webrtc.org,srte@webrtc.org > > > > # Not skipping CQ checks because original CL landed > 1 day ago. > > > > Bug: webrtc:8415 > > Change-Id: Ida8074dcac2cc28b3629228eb22846d8a8e81b83 > > Reviewed-on: https://webrtc-review.googlesource.com/52980 > > Reviewed-by: Danil Chapovalov <danilchap@webrtc.org> > > Commit-Queue: Danil Chapovalov <danilchap@webrtc.org> > > Cr-Commit-Position: refs/heads/master@{#22017} > > TBR=danilchap@webrtc.org,terelius@webrtc.org,stefan@webrtc.org,srte@webrtc.org > > Change-Id: I3393b74370c4f4d0955f50728005b2b925be169b > No-Presubmit: true > No-Tree-Checks: true > No-Try: true > Bug: webrtc:8415 > Reviewed-on: https://webrtc-review.googlesource.com/53262 > Reviewed-by: Sebastian Jansson <srte@webrtc.org> > Commit-Queue: Sebastian Jansson <srte@webrtc.org> > Cr-Commit-Position: refs/heads/master@{#22023} TBR=danilchap@webrtc.org,terelius@webrtc.org,stefan@webrtc.org,srte@webrtc.org Change-Id: Id68ad986ee51142b7be3381d0793709b4392fe2c No-Presubmit: true No-Tree-Checks: true No-Try: true Bug: webrtc:8415 Reviewed-on: https://webrtc-review.googlesource.com/53360 Reviewed-by: Sebastian Jansson <srte@webrtc.org> Commit-Queue: Sebastian Jansson <srte@webrtc.org> Cr-Commit-Position: refs/heads/master@{#22024}
205 lines
7.4 KiB
C++
205 lines
7.4 KiB
C++
/*
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* Copyright (c) 2015 The WebRTC project authors. All Rights Reserved.
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*
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* Use of this source code is governed by a BSD-style license
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* that can be found in the LICENSE file in the root of the source
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* tree. An additional intellectual property rights grant can be found
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* in the file PATENTS. All contributing project authors may
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* be found in the AUTHORS file in the root of the source tree.
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*/
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#include "modules/congestion_controller/transport_feedback_adapter.h"
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#include <algorithm>
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#include "modules/rtp_rtcp/include/rtp_rtcp_defines.h"
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#include "modules/rtp_rtcp/source/rtcp_packet/transport_feedback.h"
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#include "rtc_base/checks.h"
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#include "rtc_base/logging.h"
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#include "rtc_base/numerics/mod_ops.h"
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namespace webrtc {
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const int64_t kNoTimestamp = -1;
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const int64_t kSendTimeHistoryWindowMs = 60000;
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const int64_t kBaseTimestampScaleFactor =
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rtcp::TransportFeedback::kDeltaScaleFactor * (1 << 8);
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const int64_t kBaseTimestampRangeSizeUs = kBaseTimestampScaleFactor * (1 << 24);
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TransportFeedbackAdapter::TransportFeedbackAdapter(const Clock* clock)
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: send_time_history_(clock, kSendTimeHistoryWindowMs),
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clock_(clock),
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current_offset_ms_(kNoTimestamp),
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last_timestamp_us_(kNoTimestamp),
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local_net_id_(0),
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remote_net_id_(0) {}
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TransportFeedbackAdapter::~TransportFeedbackAdapter() {
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RTC_DCHECK(observers_.empty());
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}
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void TransportFeedbackAdapter::RegisterPacketFeedbackObserver(
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PacketFeedbackObserver* observer) {
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rtc::CritScope cs(&observers_lock_);
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RTC_DCHECK(observer);
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RTC_DCHECK(std::find(observers_.begin(), observers_.end(), observer) ==
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observers_.end());
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observers_.push_back(observer);
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}
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void TransportFeedbackAdapter::DeRegisterPacketFeedbackObserver(
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PacketFeedbackObserver* observer) {
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rtc::CritScope cs(&observers_lock_);
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RTC_DCHECK(observer);
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const auto it = std::find(observers_.begin(), observers_.end(), observer);
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RTC_DCHECK(it != observers_.end());
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observers_.erase(it);
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}
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void TransportFeedbackAdapter::AddPacket(uint32_t ssrc,
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uint16_t sequence_number,
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size_t length,
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const PacedPacketInfo& pacing_info) {
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{
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rtc::CritScope cs(&lock_);
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const int64_t creation_time_ms = clock_->TimeInMilliseconds();
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send_time_history_.AddAndRemoveOld(
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PacketFeedback(creation_time_ms, sequence_number, length, local_net_id_,
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remote_net_id_, pacing_info));
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}
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{
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rtc::CritScope cs(&observers_lock_);
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for (auto* observer : observers_) {
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observer->OnPacketAdded(ssrc, sequence_number);
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}
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}
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}
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void TransportFeedbackAdapter::OnSentPacket(uint16_t sequence_number,
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int64_t send_time_ms) {
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rtc::CritScope cs(&lock_);
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send_time_history_.OnSentPacket(sequence_number, send_time_ms);
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}
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void TransportFeedbackAdapter::SetNetworkIds(uint16_t local_id,
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uint16_t remote_id) {
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rtc::CritScope cs(&lock_);
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local_net_id_ = local_id;
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remote_net_id_ = remote_id;
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}
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std::vector<PacketFeedback> TransportFeedbackAdapter::GetPacketFeedbackVector(
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const rtcp::TransportFeedback& feedback) {
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int64_t timestamp_us = feedback.GetBaseTimeUs();
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int64_t now_ms = clock_->TimeInMilliseconds();
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// Add timestamp deltas to a local time base selected on first packet arrival.
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// This won't be the true time base, but makes it easier to manually inspect
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// time stamps.
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if (last_timestamp_us_ == kNoTimestamp) {
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current_offset_ms_ = now_ms;
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} else {
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int64_t delta = timestamp_us - last_timestamp_us_;
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// Detect and compensate for wrap-arounds in base time.
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if (std::abs(delta - kBaseTimestampRangeSizeUs) < std::abs(delta)) {
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delta -= kBaseTimestampRangeSizeUs; // Wrap backwards.
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} else if (std::abs(delta + kBaseTimestampRangeSizeUs) < std::abs(delta)) {
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delta += kBaseTimestampRangeSizeUs; // Wrap forwards.
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}
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current_offset_ms_ += delta / 1000;
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}
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last_timestamp_us_ = timestamp_us;
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std::vector<PacketFeedback> packet_feedback_vector;
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if (feedback.GetPacketStatusCount() == 0) {
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RTC_LOG(LS_INFO) << "Empty transport feedback packet received.";
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return packet_feedback_vector;
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}
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packet_feedback_vector.reserve(feedback.GetPacketStatusCount());
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int64_t feedback_rtt = -1;
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{
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rtc::CritScope cs(&lock_);
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size_t failed_lookups = 0;
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int64_t offset_us = 0;
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int64_t timestamp_ms = 0;
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uint16_t seq_num = feedback.GetBaseSequence();
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for (const auto& packet : feedback.GetReceivedPackets()) {
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// Insert into the vector those unreceived packets which precede this
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// iteration's received packet.
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for (; seq_num != packet.sequence_number(); ++seq_num) {
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PacketFeedback packet_feedback(PacketFeedback::kNotReceived, seq_num);
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// Note: Element not removed from history because it might be reported
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// as received by another feedback.
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if (!send_time_history_.GetFeedback(&packet_feedback, false))
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++failed_lookups;
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if (packet_feedback.local_net_id == local_net_id_ &&
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packet_feedback.remote_net_id == remote_net_id_) {
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packet_feedback_vector.push_back(packet_feedback);
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}
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}
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// Handle this iteration's received packet.
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offset_us += packet.delta_us();
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timestamp_ms = current_offset_ms_ + (offset_us / 1000);
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PacketFeedback packet_feedback(timestamp_ms, packet.sequence_number());
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if (!send_time_history_.GetFeedback(&packet_feedback, true))
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++failed_lookups;
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if (packet_feedback.local_net_id == local_net_id_ &&
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packet_feedback.remote_net_id == remote_net_id_) {
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if (packet_feedback.send_time_ms >= 0) {
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int64_t rtt = now_ms - packet_feedback.send_time_ms;
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// max() is used to account for feedback being delayed by the
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// receiver.
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feedback_rtt = std::max(rtt, feedback_rtt);
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}
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packet_feedback_vector.push_back(packet_feedback);
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}
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++seq_num;
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}
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if (failed_lookups > 0) {
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RTC_LOG(LS_WARNING) << "Failed to lookup send time for " << failed_lookups
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<< " packet" << (failed_lookups > 1 ? "s" : "")
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<< ". Send time history too small?";
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}
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if (feedback_rtt > -1) {
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feedback_rtts_.push_back(feedback_rtt);
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const size_t kFeedbackRttWindow = 32;
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if (feedback_rtts_.size() > kFeedbackRttWindow)
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feedback_rtts_.pop_front();
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min_feedback_rtt_.emplace(
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*std::min_element(feedback_rtts_.begin(), feedback_rtts_.end()));
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}
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}
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return packet_feedback_vector;
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}
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void TransportFeedbackAdapter::OnTransportFeedback(
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const rtcp::TransportFeedback& feedback) {
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last_packet_feedback_vector_ = GetPacketFeedbackVector(feedback);
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{
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rtc::CritScope cs(&observers_lock_);
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for (auto* observer : observers_) {
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observer->OnPacketFeedbackVector(last_packet_feedback_vector_);
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}
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}
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}
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std::vector<PacketFeedback>
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TransportFeedbackAdapter::GetTransportFeedbackVector() const {
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return last_packet_feedback_vector_;
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}
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rtc::Optional<int64_t> TransportFeedbackAdapter::GetMinFeedbackLoopRtt() const {
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rtc::CritScope cs(&lock_);
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return min_feedback_rtt_;
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}
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size_t TransportFeedbackAdapter::GetOutstandingBytes() const {
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rtc::CritScope cs(&lock_);
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return send_time_history_.GetOutstandingBytes(local_net_id_, remote_net_id_);
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}
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} // namespace webrtc
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