blob: c0150f842b80d73a496acc1a9e20653a2b7a2e27 [file] [log] [blame]
/*
* 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 "call/rtp_transport_controller_send.h"
#include <cstddef>
#include <cstdint>
#include <map>
#include <memory>
#include <optional>
#include <string>
#include <utility>
#include <vector>
#include "absl/strings/string_view.h"
#include "api/array_view.h"
#include "api/call/transport.h"
#include "api/fec_controller.h"
#include "api/frame_transformer_interface.h"
#include "api/rtc_event_log/rtc_event_log.h"
#include "api/rtp_packet_sender.h"
#include "api/scoped_refptr.h"
#include "api/sequence_checker.h"
#include "api/task_queue/pending_task_safety_flag.h"
#include "api/task_queue/task_queue_base.h"
#include "api/transport/bandwidth_estimation_settings.h"
#include "api/transport/bitrate_settings.h"
#include "api/transport/goog_cc_factory.h"
#include "api/transport/network_control.h"
#include "api/transport/network_types.h"
#include "api/units/data_rate.h"
#include "api/units/data_size.h"
#include "api/units/time_delta.h"
#include "api/units/timestamp.h"
#include "call/rtp_config.h"
#include "call/rtp_transport_config.h"
#include "call/rtp_transport_controller_send_interface.h"
#include "call/rtp_video_sender.h"
#include "call/rtp_video_sender_interface.h"
#include "logging/rtc_event_log/events/rtc_event_route_change.h"
#include "modules/congestion_controller/rtp/control_handler.h"
#include "modules/pacing/packet_router.h"
#include "modules/rtp_rtcp/include/report_block_data.h"
#include "modules/rtp_rtcp/include/rtp_rtcp_defines.h"
#include "modules/rtp_rtcp/source/rtcp_packet/transport_feedback.h"
#include "modules/rtp_rtcp/source/rtp_rtcp_interface.h"
#include "rtc_base/checks.h"
#include "rtc_base/experiments/field_trial_parser.h"
#include "rtc_base/logging.h"
#include "rtc_base/network/sent_packet.h"
#include "rtc_base/network_route.h"
#include "rtc_base/rate_limiter.h"
#include "rtc_base/task_utils/repeating_task.h"
namespace webrtc {
namespace {
static const int64_t kRetransmitWindowSizeMs = 500;
static const size_t kMaxOverheadBytes = 500;
constexpr TimeDelta kPacerQueueUpdateInterval = TimeDelta::Millis(25);
TargetRateConstraints ConvertConstraints(int min_bitrate_bps,
int max_bitrate_bps,
int start_bitrate_bps,
Clock* clock) {
TargetRateConstraints msg;
msg.at_time = Timestamp::Millis(clock->TimeInMilliseconds());
msg.min_data_rate = min_bitrate_bps >= 0
? DataRate::BitsPerSec(min_bitrate_bps)
: DataRate::Zero();
msg.max_data_rate = max_bitrate_bps > 0
? DataRate::BitsPerSec(max_bitrate_bps)
: DataRate::Infinity();
if (start_bitrate_bps > 0)
msg.starting_rate = DataRate::BitsPerSec(start_bitrate_bps);
return msg;
}
TargetRateConstraints ConvertConstraints(const BitrateConstraints& contraints,
Clock* clock) {
return ConvertConstraints(contraints.min_bitrate_bps,
contraints.max_bitrate_bps,
contraints.start_bitrate_bps, clock);
}
bool IsRelayed(const rtc::NetworkRoute& route) {
return route.local.uses_turn() || route.remote.uses_turn();
}
} // namespace
RtpTransportControllerSend::RtpTransportControllerSend(
const RtpTransportConfig& config)
: env_(config.env),
task_queue_(TaskQueueBase::Current()),
bitrate_configurator_(config.bitrate_config),
pacer_started_(false),
pacer_(&env_.clock(),
&packet_router_,
env_.field_trials(),
TimeDelta::Millis(5),
3),
observer_(nullptr),
controller_factory_override_(config.network_controller_factory),
controller_factory_fallback_(
std::make_unique<GoogCcNetworkControllerFactory>(
GoogCcFactoryConfig{.network_state_predictor_factory =
config.network_state_predictor_factory})),
process_interval_(controller_factory_fallback_->GetProcessInterval()),
last_report_block_time_(
Timestamp::Millis(env_.clock().TimeInMilliseconds())),
initial_config_(env_),
reset_feedback_on_route_change_(
!env_.field_trials().IsEnabled("WebRTC-Bwe-NoFeedbackReset")),
add_pacing_to_cwin_(env_.field_trials().IsEnabled(
"WebRTC-AddPacingToCongestionWindowPushback")),
reset_bwe_on_adapter_id_change_(
env_.field_trials().IsEnabled("WebRTC-Bwe-ResetOnAdapterIdChange")),
relay_bandwidth_cap_("relay_cap", DataRate::PlusInfinity()),
transport_overhead_bytes_per_packet_(0),
network_available_(false),
congestion_window_size_(DataSize::PlusInfinity()),
is_congested_(false),
retransmission_rate_limiter_(&env_.clock(), kRetransmitWindowSizeMs) {
ParseFieldTrial(
{&relay_bandwidth_cap_},
env_.field_trials().Lookup("WebRTC-Bwe-NetworkRouteConstraints"));
initial_config_.constraints =
ConvertConstraints(config.bitrate_config, &env_.clock());
RTC_DCHECK(config.bitrate_config.start_bitrate_bps > 0);
pacer_.SetPacingRates(
DataRate::BitsPerSec(config.bitrate_config.start_bitrate_bps),
DataRate::Zero());
if (config.pacer_burst_interval) {
// Default burst interval overriden by config.
pacer_.SetSendBurstInterval(*config.pacer_burst_interval);
}
packet_router_.RegisterNotifyBweCallback(
[this](const RtpPacketToSend& packet,
const PacedPacketInfo& pacing_info) {
return NotifyBweOfPacedSentPacket(packet, pacing_info);
});
}
RtpTransportControllerSend::~RtpTransportControllerSend() {
RTC_DCHECK_RUN_ON(&sequence_checker_);
RTC_DCHECK(video_rtp_senders_.empty());
pacer_queue_update_task_.Stop();
controller_task_.Stop();
}
RtpVideoSenderInterface* RtpTransportControllerSend::CreateRtpVideoSender(
const std::map<uint32_t, RtpState>& suspended_ssrcs,
const std::map<uint32_t, RtpPayloadState>& states,
const RtpConfig& rtp_config,
int rtcp_report_interval_ms,
Transport* send_transport,
const RtpSenderObservers& observers,
std::unique_ptr<FecController> fec_controller,
const RtpSenderFrameEncryptionConfig& frame_encryption_config,
rtc::scoped_refptr<FrameTransformerInterface> frame_transformer) {
RTC_DCHECK_RUN_ON(&sequence_checker_);
video_rtp_senders_.push_back(std::make_unique<RtpVideoSender>(
env_, suspended_ssrcs, states, rtp_config, rtcp_report_interval_ms,
send_transport, observers,
// TODO(holmer): Remove this circular dependency by injecting
// the parts of RtpTransportControllerSendInterface that are really used.
this, &retransmission_rate_limiter_, std::move(fec_controller),
frame_encryption_config.frame_encryptor,
frame_encryption_config.crypto_options, std::move(frame_transformer)));
return video_rtp_senders_.back().get();
}
void RtpTransportControllerSend::DestroyRtpVideoSender(
RtpVideoSenderInterface* rtp_video_sender) {
RTC_DCHECK_RUN_ON(&sequence_checker_);
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::RegisterSendingRtpStream(
RtpRtcpInterface& rtp_module) {
RTC_DCHECK_RUN_ON(&sequence_checker_);
// Allow pacer to send packets using this module.
packet_router_.AddSendRtpModule(&rtp_module,
/*remb_candidate=*/true);
pacer_.SetAllowProbeWithoutMediaPacket(
bwe_settings_.allow_probe_without_media &&
packet_router_.SupportsRtxPayloadPadding());
}
void RtpTransportControllerSend::DeRegisterSendingRtpStream(
RtpRtcpInterface& rtp_module) {
RTC_DCHECK_RUN_ON(&sequence_checker_);
// Disabling media, remove from packet router map to reduce size and
// prevent any stray packets in the pacer from asynchronously arriving
// to a disabled module.
packet_router_.RemoveSendRtpModule(&rtp_module);
// Clear the pacer queue of any packets pertaining to this module.
pacer_.RemovePacketsForSsrc(rtp_module.SSRC());
if (rtp_module.RtxSsrc().has_value()) {
pacer_.RemovePacketsForSsrc(*rtp_module.RtxSsrc());
}
if (rtp_module.FlexfecSsrc().has_value()) {
pacer_.RemovePacketsForSsrc(*rtp_module.FlexfecSsrc());
}
pacer_.SetAllowProbeWithoutMediaPacket(
bwe_settings_.allow_probe_without_media &&
packet_router_.SupportsRtxPayloadPadding());
}
void RtpTransportControllerSend::UpdateControlState() {
std::optional<TargetTransferRate> update = control_handler_->GetUpdate();
if (!update)
return;
retransmission_rate_limiter_.SetMaxRate(update->target_rate.bps());
// We won't create control_handler_ until we have an observers.
RTC_DCHECK(observer_ != nullptr);
observer_->OnTargetTransferRate(*update);
}
void RtpTransportControllerSend::UpdateCongestedState() {
if (auto update = GetCongestedStateUpdate()) {
is_congested_ = update.value();
pacer_.SetCongested(update.value());
}
}
std::optional<bool> RtpTransportControllerSend::GetCongestedStateUpdate()
const {
bool congested = transport_feedback_adapter_.GetOutstandingData() >=
congestion_window_size_;
if (congested != is_congested_)
return congested;
return std::nullopt;
}
PacketRouter* RtpTransportControllerSend::packet_router() {
return &packet_router_;
}
NetworkStateEstimateObserver*
RtpTransportControllerSend::network_state_estimate_observer() {
return this;
}
RtpPacketSender* RtpTransportControllerSend::packet_sender() {
return &pacer_;
}
void RtpTransportControllerSend::SetAllocatedSendBitrateLimits(
BitrateAllocationLimits limits) {
RTC_DCHECK_RUN_ON(&sequence_checker_);
streams_config_.min_total_allocated_bitrate = limits.min_allocatable_rate;
streams_config_.max_padding_rate = limits.max_padding_rate;
streams_config_.max_total_allocated_bitrate = limits.max_allocatable_rate;
UpdateStreamsConfig();
}
void RtpTransportControllerSend::SetPacingFactor(float pacing_factor) {
RTC_DCHECK_RUN_ON(&sequence_checker_);
streams_config_.pacing_factor = pacing_factor;
UpdateStreamsConfig();
}
void RtpTransportControllerSend::SetQueueTimeLimit(int limit_ms) {
pacer_.SetQueueTimeLimit(TimeDelta::Millis(limit_ms));
}
StreamFeedbackProvider*
RtpTransportControllerSend::GetStreamFeedbackProvider() {
return &feedback_demuxer_;
}
void RtpTransportControllerSend::ReconfigureBandwidthEstimation(
const BandwidthEstimationSettings& settings) {
RTC_DCHECK_RUN_ON(&sequence_checker_);
bwe_settings_ = settings;
streams_config_.enable_repeated_initial_probing =
bwe_settings_.allow_probe_without_media;
bool allow_probe_without_media = bwe_settings_.allow_probe_without_media &&
packet_router_.SupportsRtxPayloadPadding();
pacer_.SetAllowProbeWithoutMediaPacket(allow_probe_without_media);
if (controller_) {
// Recreate the controller and handler.
control_handler_ = nullptr;
controller_ = nullptr;
// The BWE controller is created when/if the network is available.
MaybeCreateControllers();
if (controller_) {
BitrateConstraints constraints = bitrate_configurator_.GetConfig();
UpdateBitrateConstraints(constraints);
UpdateStreamsConfig();
UpdateNetworkAvailability();
}
}
}
void RtpTransportControllerSend::RegisterTargetTransferRateObserver(
TargetTransferRateObserver* observer) {
RTC_DCHECK_RUN_ON(&sequence_checker_);
RTC_DCHECK(observer_ == nullptr);
observer_ = observer;
observer_->OnStartRateUpdate(*initial_config_.constraints.starting_rate);
MaybeCreateControllers();
}
bool RtpTransportControllerSend::IsRelevantRouteChange(
const rtc::NetworkRoute& old_route,
const rtc::NetworkRoute& new_route) const {
bool connected_changed = old_route.connected != new_route.connected;
bool route_ids_changed = false;
bool relaying_changed = false;
if (reset_bwe_on_adapter_id_change_) {
route_ids_changed =
old_route.local.adapter_id() != new_route.local.adapter_id() ||
old_route.remote.adapter_id() != new_route.remote.adapter_id();
} else {
route_ids_changed =
old_route.local.network_id() != new_route.local.network_id() ||
old_route.remote.network_id() != new_route.remote.network_id();
}
if (relay_bandwidth_cap_->IsFinite()) {
relaying_changed = IsRelayed(old_route) != IsRelayed(new_route);
}
return connected_changed || route_ids_changed || relaying_changed;
}
void RtpTransportControllerSend::OnNetworkRouteChanged(
absl::string_view transport_name,
const rtc::NetworkRoute& network_route) {
RTC_DCHECK_RUN_ON(&sequence_checker_);
// Check if the network route is connected.
if (!network_route.connected) {
// TODO(honghaiz): Perhaps handle this in SignalChannelNetworkState and
// consider merging these two methods.
return;
}
std::optional<BitrateConstraints> relay_constraint_update =
ApplyOrLiftRelayCap(IsRelayed(network_route));
// Check whether the network route has changed on each transport.
auto result = network_routes_.insert(
// Explicit conversion of transport_name to std::string here is necessary
// to support some platforms that cannot yet deal with implicit
// conversion in these types of situations.
std::make_pair(std::string(transport_name), network_route));
auto kv = result.first;
bool inserted = result.second;
if (inserted || !(kv->second == network_route)) {
RTC_LOG(LS_INFO) << "Network route changed on transport " << transport_name
<< ": new_route = " << network_route.DebugString();
if (!inserted) {
RTC_LOG(LS_INFO) << "old_route = " << kv->second.DebugString();
}
}
if (inserted) {
if (relay_constraint_update.has_value()) {
UpdateBitrateConstraints(*relay_constraint_update);
}
transport_overhead_bytes_per_packet_ = network_route.packet_overhead;
// No need to reset BWE if this is the first time the network connects.
return;
}
const rtc::NetworkRoute old_route = kv->second;
kv->second = network_route;
// Check if enough conditions of the new/old route has changed
// to trigger resetting of bitrates (and a probe).
if (IsRelevantRouteChange(old_route, network_route)) {
BitrateConstraints bitrate_config = bitrate_configurator_.GetConfig();
RTC_LOG(LS_INFO) << "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);
env_.event_log().Log(std::make_unique<RtcEventRouteChange>(
network_route.connected, network_route.packet_overhead));
NetworkRouteChange msg;
msg.at_time = Timestamp::Millis(env_.clock().TimeInMilliseconds());
msg.constraints = ConvertConstraints(bitrate_config, &env_.clock());
transport_overhead_bytes_per_packet_ = network_route.packet_overhead;
if (reset_feedback_on_route_change_) {
transport_feedback_adapter_.SetNetworkRoute(network_route);
}
if (controller_) {
PostUpdates(controller_->OnNetworkRouteChange(msg));
} else {
UpdateInitialConstraints(msg.constraints);
}
is_congested_ = false;
pacer_.SetCongested(false);
}
}
void RtpTransportControllerSend::OnNetworkAvailability(bool network_available) {
RTC_DCHECK_RUN_ON(&sequence_checker_);
RTC_LOG(LS_VERBOSE) << "SignalNetworkState "
<< (network_available ? "Up" : "Down");
network_available_ = network_available;
if (network_available) {
pacer_.Resume();
} else {
pacer_.Pause();
}
is_congested_ = false;
pacer_.SetCongested(false);
if (!controller_) {
MaybeCreateControllers();
}
UpdateNetworkAvailability();
for (auto& rtp_sender : video_rtp_senders_) {
rtp_sender->OnNetworkAvailability(network_available);
}
}
NetworkLinkRtcpObserver* RtpTransportControllerSend::GetRtcpObserver() {
return this;
}
int64_t RtpTransportControllerSend::GetPacerQueuingDelayMs() const {
return pacer_.OldestPacketWaitTime().ms();
}
std::optional<Timestamp> RtpTransportControllerSend::GetFirstPacketTime()
const {
return pacer_.FirstSentPacketTime();
}
void RtpTransportControllerSend::EnablePeriodicAlrProbing(bool enable) {
RTC_DCHECK_RUN_ON(&sequence_checker_);
streams_config_.requests_alr_probing = enable;
UpdateStreamsConfig();
}
void RtpTransportControllerSend::OnSentPacket(
const rtc::SentPacket& sent_packet) {
// Normally called on the network thread!
// TODO(crbug.com/1373439): Clarify other thread contexts calling in,
// and simplify task posting logic when the combined network/worker project
// launches.
if (TaskQueueBase::Current() != task_queue_) {
task_queue_->PostTask(SafeTask(safety_.flag(), [this, sent_packet]() {
RTC_DCHECK_RUN_ON(&sequence_checker_);
ProcessSentPacket(sent_packet);
}));
return;
}
RTC_DCHECK_RUN_ON(&sequence_checker_);
ProcessSentPacket(sent_packet);
}
void RtpTransportControllerSend::ProcessSentPacket(
const rtc::SentPacket& sent_packet) {
RTC_DCHECK_RUN_ON(&sequence_checker_);
std::optional<SentPacket> packet_msg =
transport_feedback_adapter_.ProcessSentPacket(sent_packet);
if (!packet_msg)
return;
auto congestion_update = GetCongestedStateUpdate();
NetworkControlUpdate control_update;
if (controller_)
control_update = controller_->OnSentPacket(*packet_msg);
if (!congestion_update && !control_update.has_updates())
return;
ProcessSentPacketUpdates(std::move(control_update));
}
// RTC_RUN_ON(task_queue_)
void RtpTransportControllerSend::ProcessSentPacketUpdates(
NetworkControlUpdate updates) {
RTC_DCHECK_RUN_ON(&sequence_checker_);
// Only update outstanding data if:
// 1. Packet feedback is used.
// 2. The packet has not yet received an acknowledgement.
// 3. It is not a retransmission of an earlier packet.
UpdateCongestedState();
if (controller_) {
PostUpdates(std::move(updates));
}
}
void RtpTransportControllerSend::OnReceivedPacket(
const ReceivedPacket& packet_msg) {
RTC_DCHECK_RUN_ON(&sequence_checker_);
if (controller_)
PostUpdates(controller_->OnReceivedPacket(packet_msg));
}
void RtpTransportControllerSend::UpdateBitrateConstraints(
const BitrateConstraints& updated) {
RTC_DCHECK_RUN_ON(&sequence_checker_);
TargetRateConstraints msg = ConvertConstraints(updated, &env_.clock());
if (controller_) {
PostUpdates(controller_->OnTargetRateConstraints(msg));
} else {
UpdateInitialConstraints(msg);
}
}
void RtpTransportControllerSend::SetSdpBitrateParameters(
const BitrateConstraints& constraints) {
RTC_DCHECK_RUN_ON(&sequence_checker_);
std::optional<BitrateConstraints> updated =
bitrate_configurator_.UpdateWithSdpParameters(constraints);
if (updated.has_value()) {
UpdateBitrateConstraints(*updated);
} else {
RTC_LOG(LS_VERBOSE)
<< "WebRTC.RtpTransportControllerSend.SetSdpBitrateParameters: "
"nothing to update";
}
}
void RtpTransportControllerSend::SetClientBitratePreferences(
const BitrateSettings& preferences) {
RTC_DCHECK_RUN_ON(&sequence_checker_);
std::optional<BitrateConstraints> updated =
bitrate_configurator_.UpdateWithClientPreferences(preferences);
if (updated.has_value()) {
UpdateBitrateConstraints(*updated);
} else {
RTC_LOG(LS_VERBOSE)
<< "WebRTC.RtpTransportControllerSend.SetClientBitratePreferences: "
"nothing to update";
}
}
std::optional<BitrateConstraints>
RtpTransportControllerSend::ApplyOrLiftRelayCap(bool is_relayed) {
DataRate cap = is_relayed ? relay_bandwidth_cap_ : DataRate::PlusInfinity();
return bitrate_configurator_.UpdateWithRelayCap(cap);
}
void RtpTransportControllerSend::OnTransportOverheadChanged(
size_t transport_overhead_bytes_per_packet) {
RTC_DCHECK_RUN_ON(&sequence_checker_);
if (transport_overhead_bytes_per_packet >= kMaxOverheadBytes) {
RTC_LOG(LS_ERROR) << "Transport overhead exceeds " << kMaxOverheadBytes;
return;
}
pacer_.SetTransportOverhead(
DataSize::Bytes(transport_overhead_bytes_per_packet));
// 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);
}
}
void RtpTransportControllerSend::AccountForAudioPacketsInPacedSender(
bool account_for_audio) {
pacer_.SetAccountForAudioPackets(account_for_audio);
}
void RtpTransportControllerSend::IncludeOverheadInPacedSender() {
pacer_.SetIncludeOverhead();
}
void RtpTransportControllerSend::EnsureStarted() {
RTC_DCHECK_RUN_ON(&sequence_checker_);
if (!pacer_started_) {
pacer_started_ = true;
pacer_.EnsureStarted();
}
}
void RtpTransportControllerSend::OnReceiverEstimatedMaxBitrate(
Timestamp receive_time,
DataRate bitrate) {
RTC_DCHECK_RUN_ON(&sequence_checker_);
RemoteBitrateReport msg;
msg.receive_time = receive_time;
msg.bandwidth = bitrate;
if (controller_)
PostUpdates(controller_->OnRemoteBitrateReport(msg));
}
void RtpTransportControllerSend::OnRttUpdate(Timestamp receive_time,
TimeDelta rtt) {
RTC_DCHECK_RUN_ON(&sequence_checker_);
RoundTripTimeUpdate report;
report.receive_time = receive_time;
report.round_trip_time = rtt.RoundTo(TimeDelta::Millis(1));
report.smoothed = false;
if (controller_ && !report.round_trip_time.IsZero())
PostUpdates(controller_->OnRoundTripTimeUpdate(report));
}
void RtpTransportControllerSend::NotifyBweOfPacedSentPacket(
const RtpPacketToSend& packet,
const PacedPacketInfo& pacing_info) {
RTC_DCHECK_RUN_ON(&sequence_checker_);
if (!packet.transport_sequence_number()) {
return;
}
if (!packet.packet_type()) {
RTC_DCHECK_NOTREACHED() << "Unknown packet type";
return;
}
RtpPacketSendInfo packet_info = RtpPacketSendInfo::From(packet, pacing_info);
Timestamp creation_time =
Timestamp::Millis(env_.clock().TimeInMilliseconds());
feedback_demuxer_.AddPacket(packet_info);
transport_feedback_adapter_.AddPacket(
packet_info, transport_overhead_bytes_per_packet_, creation_time);
}
void RtpTransportControllerSend::OnTransportFeedback(
Timestamp receive_time,
const rtcp::TransportFeedback& feedback) {
RTC_DCHECK_RUN_ON(&sequence_checker_);
feedback_demuxer_.OnTransportFeedback(feedback);
std::optional<TransportPacketsFeedback> feedback_msg =
transport_feedback_adapter_.ProcessTransportFeedback(feedback,
receive_time);
if (feedback_msg) {
if (controller_)
PostUpdates(controller_->OnTransportPacketsFeedback(*feedback_msg));
// Only update outstanding data if any packet is first time acked.
UpdateCongestedState();
}
}
void RtpTransportControllerSend::OnRemoteNetworkEstimate(
NetworkStateEstimate estimate) {
RTC_DCHECK_RUN_ON(&sequence_checker_);
estimate.update_time = Timestamp::Millis(env_.clock().TimeInMilliseconds());
if (controller_)
PostUpdates(controller_->OnNetworkStateEstimate(estimate));
}
void RtpTransportControllerSend::MaybeCreateControllers() {
RTC_DCHECK(!controller_);
RTC_DCHECK(!control_handler_);
if (!network_available_ || !observer_)
return;
control_handler_ = std::make_unique<CongestionControlHandler>();
initial_config_.constraints.at_time =
Timestamp::Millis(env_.clock().TimeInMilliseconds());
initial_config_.stream_based_config = streams_config_;
// TODO(srte): Use fallback controller if no feedback is available.
if (controller_factory_override_) {
RTC_LOG(LS_INFO) << "Creating overridden congestion controller";
controller_ = controller_factory_override_->Create(initial_config_);
process_interval_ = controller_factory_override_->GetProcessInterval();
} else {
RTC_LOG(LS_INFO) << "Creating fallback congestion controller";
controller_ = controller_factory_fallback_->Create(initial_config_);
process_interval_ = controller_factory_fallback_->GetProcessInterval();
}
UpdateControllerWithTimeInterval();
StartProcessPeriodicTasks();
}
void RtpTransportControllerSend::UpdateNetworkAvailability() {
if (!controller_) {
return;
}
NetworkAvailability msg;
msg.at_time = Timestamp::Millis(env_.clock().TimeInMilliseconds());
msg.network_available = network_available_;
control_handler_->SetNetworkAvailability(network_available_);
PostUpdates(controller_->OnNetworkAvailability(msg));
UpdateControlState();
}
void RtpTransportControllerSend::UpdateInitialConstraints(
TargetRateConstraints new_contraints) {
if (!new_contraints.starting_rate)
new_contraints.starting_rate = initial_config_.constraints.starting_rate;
RTC_DCHECK(new_contraints.starting_rate);
initial_config_.constraints = new_contraints;
}
void RtpTransportControllerSend::StartProcessPeriodicTasks() {
RTC_DCHECK_RUN_ON(&sequence_checker_);
if (!pacer_queue_update_task_.Running()) {
pacer_queue_update_task_ = RepeatingTaskHandle::DelayedStart(
task_queue_, kPacerQueueUpdateInterval, [this]() {
RTC_DCHECK_RUN_ON(&sequence_checker_);
TimeDelta expected_queue_time = pacer_.ExpectedQueueTime();
control_handler_->SetPacerQueue(expected_queue_time);
UpdateControlState();
return kPacerQueueUpdateInterval;
});
}
controller_task_.Stop();
if (process_interval_.IsFinite()) {
controller_task_ = RepeatingTaskHandle::DelayedStart(
task_queue_, process_interval_, [this]() {
RTC_DCHECK_RUN_ON(&sequence_checker_);
UpdateControllerWithTimeInterval();
return process_interval_;
});
}
}
void RtpTransportControllerSend::UpdateControllerWithTimeInterval() {
RTC_DCHECK(controller_);
ProcessInterval msg;
msg.at_time = Timestamp::Millis(env_.clock().TimeInMilliseconds());
if (add_pacing_to_cwin_)
msg.pacer_queue = pacer_.QueueSizeData();
PostUpdates(controller_->OnProcessInterval(msg));
}
void RtpTransportControllerSend::UpdateStreamsConfig() {
streams_config_.at_time =
Timestamp::Millis(env_.clock().TimeInMilliseconds());
if (controller_)
PostUpdates(controller_->OnStreamsConfig(streams_config_));
}
void RtpTransportControllerSend::PostUpdates(NetworkControlUpdate update) {
if (update.congestion_window) {
congestion_window_size_ = *update.congestion_window;
UpdateCongestedState();
}
if (update.pacer_config) {
pacer_.SetPacingRates(update.pacer_config->data_rate(),
update.pacer_config->pad_rate());
}
if (!update.probe_cluster_configs.empty()) {
pacer_.CreateProbeClusters(std::move(update.probe_cluster_configs));
}
if (update.target_rate) {
control_handler_->SetTargetRate(*update.target_rate);
UpdateControlState();
}
}
void RtpTransportControllerSend::OnReport(
Timestamp receive_time,
rtc::ArrayView<const ReportBlockData> report_blocks) {
RTC_DCHECK_RUN_ON(&sequence_checker_);
if (report_blocks.empty())
return;
int total_packets_lost_delta = 0;
int total_packets_delta = 0;
// Compute the packet loss from all report blocks.
for (const ReportBlockData& report_block : report_blocks) {
auto [it, inserted] =
last_report_blocks_.try_emplace(report_block.source_ssrc());
LossReport& last_loss_report = it->second;
if (!inserted) {
total_packets_delta += report_block.extended_highest_sequence_number() -
last_loss_report.extended_highest_sequence_number;
total_packets_lost_delta +=
report_block.cumulative_lost() - last_loss_report.cumulative_lost;
}
last_loss_report.extended_highest_sequence_number =
report_block.extended_highest_sequence_number();
last_loss_report.cumulative_lost = report_block.cumulative_lost();
}
// Can only compute delta if there has been previous blocks to compare to. If
// not, total_packets_delta will be unchanged and there's nothing more to do.
if (!total_packets_delta)
return;
int packets_received_delta = total_packets_delta - total_packets_lost_delta;
// To detect lost packets, at least one packet has to be received. This check
// is needed to avoid bandwith detection update in
// VideoSendStreamTest.SuspendBelowMinBitrate
if (packets_received_delta < 1)
return;
TransportLossReport msg;
msg.packets_lost_delta = total_packets_lost_delta;
msg.packets_received_delta = packets_received_delta;
msg.receive_time = receive_time;
msg.start_time = last_report_block_time_;
msg.end_time = receive_time;
if (controller_)
PostUpdates(controller_->OnTransportLossReport(msg));
last_report_block_time_ = receive_time;
}
} // namespace webrtc