Reason for revert: Bot breakage caused by TickTime::UseFakeClock has been removed. Original issue's description: > Revert of Merge webrtc/video_engine/ into webrtc/video/ (patchset #2 id:20001 of https://codereview.webrtc.org/1506773002/ ) > > Reason for revert: > Breaks Dr Memory Light https://build.chromium.org/p/client.webrtc/builders/Win%20DrMemory%20Light/builds/4643 and all the Android Tests bots. > > Original issue's description: > > Merge webrtc/video_engine/ into webrtc/video/ > > > > BUG=webrtc:1695 > > R=mflodman@webrtc.org > > > > Committed: https://crrev.com/03ef053202bc5d5ab43460eebf5403232f157646 > > Cr-Commit-Position: refs/heads/master@{#10926} > > TBR=mflodman@webrtc.org,pbos@webrtc.org > NOPRESUBMIT=true > NOTREECHECKS=true > NOTRY=true > BUG=webrtc:1695 > > Committed: https://crrev.com/8237abf563bf4782ee104408b53cc8e55ce44518 > Cr-Commit-Position: refs/heads/master@{#10937} BUG=webrtc:1695 TBR=mflodman@webrtc.org,kjellander@webrtc.org NOPRESUBMIT=true Review URL: https://codereview.webrtc.org/1510183002 . Cr-Commit-Position: refs/heads/master@{#10948}
295 lines
11 KiB
C++
295 lines
11 KiB
C++
/*
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* Copyright (c) 2012 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 "webrtc/call/congestion_controller.h"
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#include "webrtc/base/checks.h"
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#include "webrtc/base/logging.h"
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#include "webrtc/base/thread_annotations.h"
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#include "webrtc/common.h"
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#include "webrtc/modules/bitrate_controller/include/bitrate_controller.h"
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#include "webrtc/modules/pacing/paced_sender.h"
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#include "webrtc/modules/pacing/packet_router.h"
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#include "webrtc/modules/remote_bitrate_estimator/include/send_time_history.h"
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#include "webrtc/modules/remote_bitrate_estimator/remote_bitrate_estimator_abs_send_time.h"
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#include "webrtc/modules/remote_bitrate_estimator/remote_bitrate_estimator_single_stream.h"
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#include "webrtc/modules/remote_bitrate_estimator/remote_estimator_proxy.h"
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#include "webrtc/modules/remote_bitrate_estimator/transport_feedback_adapter.h"
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#include "webrtc/modules/rtp_rtcp/include/rtp_rtcp.h"
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#include "webrtc/modules/utility/include/process_thread.h"
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#include "webrtc/system_wrappers/include/critical_section_wrapper.h"
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#include "webrtc/video/call_stats.h"
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#include "webrtc/video/payload_router.h"
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#include "webrtc/video/vie_encoder.h"
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#include "webrtc/video/vie_remb.h"
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#include "webrtc/voice_engine/include/voe_video_sync.h"
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namespace webrtc {
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namespace {
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static const uint32_t kTimeOffsetSwitchThreshold = 30;
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class WrappingBitrateEstimator : public RemoteBitrateEstimator {
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public:
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WrappingBitrateEstimator(RemoteBitrateObserver* observer, Clock* clock)
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: observer_(observer),
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clock_(clock),
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crit_sect_(CriticalSectionWrapper::CreateCriticalSection()),
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rbe_(new RemoteBitrateEstimatorSingleStream(observer_, clock_)),
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using_absolute_send_time_(false),
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packets_since_absolute_send_time_(0),
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min_bitrate_bps_(RemoteBitrateEstimator::kDefaultMinBitrateBps) {}
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virtual ~WrappingBitrateEstimator() {}
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void IncomingPacket(int64_t arrival_time_ms,
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size_t payload_size,
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const RTPHeader& header,
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bool was_paced) override {
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CriticalSectionScoped cs(crit_sect_.get());
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PickEstimatorFromHeader(header);
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rbe_->IncomingPacket(arrival_time_ms, payload_size, header, was_paced);
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}
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int32_t Process() override {
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CriticalSectionScoped cs(crit_sect_.get());
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return rbe_->Process();
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}
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int64_t TimeUntilNextProcess() override {
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CriticalSectionScoped cs(crit_sect_.get());
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return rbe_->TimeUntilNextProcess();
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}
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void OnRttUpdate(int64_t avg_rtt_ms, int64_t max_rtt_ms) override {
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CriticalSectionScoped cs(crit_sect_.get());
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rbe_->OnRttUpdate(avg_rtt_ms, max_rtt_ms);
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}
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void RemoveStream(unsigned int ssrc) override {
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CriticalSectionScoped cs(crit_sect_.get());
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rbe_->RemoveStream(ssrc);
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}
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bool LatestEstimate(std::vector<unsigned int>* ssrcs,
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unsigned int* bitrate_bps) const override {
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CriticalSectionScoped cs(crit_sect_.get());
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return rbe_->LatestEstimate(ssrcs, bitrate_bps);
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}
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bool GetStats(ReceiveBandwidthEstimatorStats* output) const override {
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CriticalSectionScoped cs(crit_sect_.get());
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return rbe_->GetStats(output);
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}
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void SetMinBitrate(int min_bitrate_bps) {
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CriticalSectionScoped cs(crit_sect_.get());
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rbe_->SetMinBitrate(min_bitrate_bps);
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min_bitrate_bps_ = min_bitrate_bps;
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}
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private:
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void PickEstimatorFromHeader(const RTPHeader& header)
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EXCLUSIVE_LOCKS_REQUIRED(crit_sect_.get()) {
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if (header.extension.hasAbsoluteSendTime) {
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// If we see AST in header, switch RBE strategy immediately.
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if (!using_absolute_send_time_) {
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LOG(LS_INFO) <<
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"WrappingBitrateEstimator: Switching to absolute send time RBE.";
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using_absolute_send_time_ = true;
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PickEstimator();
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}
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packets_since_absolute_send_time_ = 0;
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} else {
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// When we don't see AST, wait for a few packets before going back to TOF.
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if (using_absolute_send_time_) {
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++packets_since_absolute_send_time_;
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if (packets_since_absolute_send_time_ >= kTimeOffsetSwitchThreshold) {
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LOG(LS_INFO) << "WrappingBitrateEstimator: Switching to transmission "
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<< "time offset RBE.";
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using_absolute_send_time_ = false;
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PickEstimator();
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}
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}
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}
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}
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// Instantiate RBE for Time Offset or Absolute Send Time extensions.
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void PickEstimator() EXCLUSIVE_LOCKS_REQUIRED(crit_sect_.get()) {
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if (using_absolute_send_time_) {
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rbe_.reset(new RemoteBitrateEstimatorAbsSendTime(observer_, clock_));
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} else {
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rbe_.reset(new RemoteBitrateEstimatorSingleStream(observer_, clock_));
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}
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rbe_->SetMinBitrate(min_bitrate_bps_);
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}
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RemoteBitrateObserver* observer_;
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Clock* clock_;
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rtc::scoped_ptr<CriticalSectionWrapper> crit_sect_;
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rtc::scoped_ptr<RemoteBitrateEstimator> rbe_;
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bool using_absolute_send_time_;
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uint32_t packets_since_absolute_send_time_;
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int min_bitrate_bps_;
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RTC_DISALLOW_IMPLICIT_CONSTRUCTORS(WrappingBitrateEstimator);
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};
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} // namespace
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CongestionController::CongestionController(ProcessThread* process_thread,
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CallStats* call_stats,
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BitrateObserver* bitrate_observer)
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: remb_(new VieRemb(Clock::GetRealTimeClock())),
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packet_router_(new PacketRouter()),
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pacer_(new PacedSender(Clock::GetRealTimeClock(),
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packet_router_.get(),
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BitrateController::kDefaultStartBitrateKbps,
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PacedSender::kDefaultPaceMultiplier *
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BitrateController::kDefaultStartBitrateKbps,
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0)),
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remote_bitrate_estimator_(
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new WrappingBitrateEstimator(remb_.get(), Clock::GetRealTimeClock())),
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remote_estimator_proxy_(
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new RemoteEstimatorProxy(Clock::GetRealTimeClock(),
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packet_router_.get())),
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process_thread_(process_thread),
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call_stats_(call_stats),
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pacer_thread_(ProcessThread::Create("PacerThread")),
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// Constructed last as this object calls the provided callback on
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// construction.
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bitrate_controller_(
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BitrateController::CreateBitrateController(Clock::GetRealTimeClock(),
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bitrate_observer)),
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min_bitrate_bps_(RemoteBitrateEstimator::kDefaultMinBitrateBps) {
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call_stats_->RegisterStatsObserver(remote_bitrate_estimator_.get());
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pacer_thread_->RegisterModule(pacer_.get());
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pacer_thread_->Start();
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process_thread->RegisterModule(remote_estimator_proxy_.get());
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process_thread->RegisterModule(remote_bitrate_estimator_.get());
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process_thread->RegisterModule(bitrate_controller_.get());
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}
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CongestionController::~CongestionController() {
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pacer_thread_->Stop();
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pacer_thread_->DeRegisterModule(pacer_.get());
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process_thread_->DeRegisterModule(bitrate_controller_.get());
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process_thread_->DeRegisterModule(remote_bitrate_estimator_.get());
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process_thread_->DeRegisterModule(remote_estimator_proxy_.get());
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call_stats_->DeregisterStatsObserver(remote_bitrate_estimator_.get());
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if (transport_feedback_adapter_.get())
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call_stats_->DeregisterStatsObserver(transport_feedback_adapter_.get());
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RTC_DCHECK(!remb_->InUse());
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RTC_DCHECK(encoders_.empty());
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}
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void CongestionController::AddEncoder(ViEEncoder* encoder) {
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rtc::CritScope lock(&encoder_crit_);
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encoders_.push_back(encoder);
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}
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void CongestionController::RemoveEncoder(ViEEncoder* encoder) {
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rtc::CritScope lock(&encoder_crit_);
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for (auto it = encoders_.begin(); it != encoders_.end(); ++it) {
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if (*it == encoder) {
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encoders_.erase(it);
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return;
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}
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}
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}
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void CongestionController::SetBweBitrates(int min_bitrate_bps,
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int start_bitrate_bps,
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int max_bitrate_bps) {
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if (start_bitrate_bps > 0)
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bitrate_controller_->SetStartBitrate(start_bitrate_bps);
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bitrate_controller_->SetMinMaxBitrate(min_bitrate_bps, max_bitrate_bps);
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if (remote_bitrate_estimator_.get())
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remote_bitrate_estimator_->SetMinBitrate(min_bitrate_bps);
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if (transport_feedback_adapter_.get())
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transport_feedback_adapter_->GetBitrateEstimator()->SetMinBitrate(
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min_bitrate_bps);
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min_bitrate_bps_ = min_bitrate_bps;
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}
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BitrateController* CongestionController::GetBitrateController() const {
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return bitrate_controller_.get();
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}
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RemoteBitrateEstimator* CongestionController::GetRemoteBitrateEstimator(
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bool send_side_bwe) const {
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if (send_side_bwe)
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return remote_estimator_proxy_.get();
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else
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return remote_bitrate_estimator_.get();
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}
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TransportFeedbackObserver*
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CongestionController::GetTransportFeedbackObserver() {
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if (transport_feedback_adapter_.get() == nullptr) {
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transport_feedback_adapter_.reset(new TransportFeedbackAdapter(
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bitrate_controller_->CreateRtcpBandwidthObserver(),
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Clock::GetRealTimeClock(), process_thread_));
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transport_feedback_adapter_->SetBitrateEstimator(
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new RemoteBitrateEstimatorAbsSendTime(
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transport_feedback_adapter_.get(), Clock::GetRealTimeClock()));
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transport_feedback_adapter_->GetBitrateEstimator()->SetMinBitrate(
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min_bitrate_bps_);
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call_stats_->RegisterStatsObserver(transport_feedback_adapter_.get());
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}
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return transport_feedback_adapter_.get();
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}
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void CongestionController::UpdatePacerBitrate(int bitrate_kbps,
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int max_bitrate_kbps,
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int min_bitrate_kbps) {
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pacer_->UpdateBitrate(bitrate_kbps, max_bitrate_kbps, min_bitrate_kbps);
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}
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int64_t CongestionController::GetPacerQueuingDelayMs() const {
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return pacer_->QueueInMs();
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}
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// TODO(mflodman): Move out of this class.
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void CongestionController::SetChannelRembStatus(bool sender,
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bool receiver,
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RtpRtcp* rtp_module) {
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rtp_module->SetREMBStatus(sender || receiver);
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if (sender) {
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remb_->AddRembSender(rtp_module);
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} else {
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remb_->RemoveRembSender(rtp_module);
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}
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if (receiver) {
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remb_->AddReceiveChannel(rtp_module);
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} else {
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remb_->RemoveReceiveChannel(rtp_module);
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}
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}
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void CongestionController::SignalNetworkState(NetworkState state) {
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if (state == kNetworkUp) {
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pacer_->Resume();
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} else {
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pacer_->Pause();
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}
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}
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void CongestionController::OnSentPacket(const rtc::SentPacket& sent_packet) {
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if (transport_feedback_adapter_) {
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transport_feedback_adapter_->OnSentPacket(sent_packet.packet_id,
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sent_packet.send_time_ms);
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}
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}
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} // namespace webrtc
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