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gecko-dev/third_party/libwebrtc/call/rtp_transport_controller_se...

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/*
* 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 <memory>
#include <utility>
#include <vector>
#include "absl/strings/match.h"
#include "absl/types/optional.h"
#include "api/transport/goog_cc_factory.h"
#include "api/transport/network_types.h"
#include "api/units/data_rate.h"
#include "api/units/time_delta.h"
#include "api/units/timestamp.h"
#include "call/rtp_video_sender.h"
#include "logging/rtc_event_log/events/rtc_event_remote_estimate.h"
#include "logging/rtc_event_log/events/rtc_event_route_change.h"
#include "modules/rtp_rtcp/source/rtcp_packet/transport_feedback.h"
#include "rtc_base/checks.h"
#include "rtc_base/logging.h"
#include "rtc_base/rate_limiter.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 IsEnabled(const FieldTrialsView& trials, absl::string_view key) {
return absl::StartsWith(trials.Lookup(key), "Enabled");
}
bool IsDisabled(const FieldTrialsView& trials, absl::string_view key) {
return absl::StartsWith(trials.Lookup(key), "Disabled");
}
bool IsRelayed(const rtc::NetworkRoute& route) {
return route.local.uses_turn() || route.remote.uses_turn();
}
} // namespace
RtpTransportControllerSend::PacerSettings::PacerSettings(
const FieldTrialsView& trials)
: tq_disabled("Disabled"),
holdback_window("holdback_window", TimeDelta::Millis(5)),
holdback_packets("holdback_packets", 3) {
ParseFieldTrial({&tq_disabled, &holdback_window, &holdback_packets},
trials.Lookup("WebRTC-TaskQueuePacer"));
}
RtpTransportControllerSend::RtpTransportControllerSend(
Clock* clock,
webrtc::RtcEventLog* event_log,
NetworkStatePredictorFactoryInterface* predictor_factory,
NetworkControllerFactoryInterface* controller_factory,
const BitrateConstraints& bitrate_config,
std::unique_ptr<ProcessThread> process_thread,
TaskQueueFactory* task_queue_factory,
const FieldTrialsView& trials)
: clock_(clock),
event_log_(event_log),
bitrate_configurator_(bitrate_config),
pacer_started_(false),
process_thread_(std::move(process_thread)),
pacer_settings_(trials),
process_thread_pacer_(pacer_settings_.use_task_queue_pacer()
? nullptr
: new PacedSender(clock,
&packet_router_,
trials,
process_thread_.get())),
task_queue_pacer_(
pacer_settings_.use_task_queue_pacer()
? new TaskQueuePacedSender(clock,
&packet_router_,
trials,
task_queue_factory,
pacer_settings_.holdback_window.Get(),
pacer_settings_.holdback_packets.Get())
: nullptr),
observer_(nullptr),
controller_factory_override_(controller_factory),
controller_factory_fallback_(
std::make_unique<GoogCcNetworkControllerFactory>(predictor_factory)),
process_interval_(controller_factory_fallback_->GetProcessInterval()),
last_report_block_time_(Timestamp::Millis(clock_->TimeInMilliseconds())),
reset_feedback_on_route_change_(
!IsEnabled(trials, "WebRTC-Bwe-NoFeedbackReset")),
send_side_bwe_with_overhead_(
!IsDisabled(trials, "WebRTC-SendSideBwe-WithOverhead")),
add_pacing_to_cwin_(
IsEnabled(trials, "WebRTC-AddPacingToCongestionWindowPushback")),
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_(clock, kRetransmitWindowSizeMs),
task_queue_(task_queue_factory->CreateTaskQueue(
"rtp_send_controller",
TaskQueueFactory::Priority::NORMAL)),
field_trials_(trials) {
ParseFieldTrial({&relay_bandwidth_cap_},
trials.Lookup("WebRTC-Bwe-NetworkRouteConstraints"));
initial_config_.constraints = ConvertConstraints(bitrate_config, clock_);
initial_config_.event_log = event_log;
initial_config_.key_value_config = &trials;
RTC_DCHECK(bitrate_config.start_bitrate_bps > 0);
pacer()->SetPacingRates(
DataRate::BitsPerSec(bitrate_config.start_bitrate_bps), DataRate::Zero());
if (absl::StartsWith(trials.Lookup("WebRTC-LazyPacerStart"), "Disabled")) {
EnsureStarted();
}
}
RtpTransportControllerSend::~RtpTransportControllerSend() {
RTC_DCHECK(video_rtp_senders_.empty());
process_thread_->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,
RtcEventLog* event_log,
std::unique_ptr<FecController> fec_controller,
const RtpSenderFrameEncryptionConfig& frame_encryption_config,
rtc::scoped_refptr<FrameTransformerInterface> frame_transformer) {
RTC_DCHECK_RUN_ON(&main_thread_);
video_rtp_senders_.push_back(std::make_unique<RtpVideoSender>(
clock_, 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, event_log, &retransmission_rate_limiter_, std::move(fec_controller),
frame_encryption_config.frame_encryptor,
frame_encryption_config.crypto_options, std::move(frame_transformer),
field_trials_));
return video_rtp_senders_.back().get();
}
void RtpTransportControllerSend::DestroyRtpVideoSender(
RtpVideoSenderInterface* rtp_video_sender) {
RTC_DCHECK_RUN_ON(&main_thread_);
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::UpdateControlState() {
absl::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() {
bool congested = transport_feedback_adapter_.GetOutstandingData() >=
congestion_window_size_;
if (congested != is_congested_) {
is_congested_ = congested;
pacer()->SetCongested(congested);
}
}
RtpPacketPacer* RtpTransportControllerSend::pacer() {
if (pacer_settings_.use_task_queue_pacer()) {
return task_queue_pacer_.get();
}
return process_thread_pacer_.get();
}
const RtpPacketPacer* RtpTransportControllerSend::pacer() const {
if (pacer_settings_.use_task_queue_pacer()) {
return task_queue_pacer_.get();
}
return process_thread_pacer_.get();
}
rtc::TaskQueue* RtpTransportControllerSend::GetWorkerQueue() {
return &task_queue_;
}
PacketRouter* RtpTransportControllerSend::packet_router() {
return &packet_router_;
}
NetworkStateEstimateObserver*
RtpTransportControllerSend::network_state_estimate_observer() {
return this;
}
TransportFeedbackObserver*
RtpTransportControllerSend::transport_feedback_observer() {
return this;
}
RtpPacketSender* RtpTransportControllerSend::packet_sender() {
if (pacer_settings_.use_task_queue_pacer()) {
return task_queue_pacer_.get();
}
return process_thread_pacer_.get();
}
void RtpTransportControllerSend::SetAllocatedSendBitrateLimits(
BitrateAllocationLimits limits) {
RTC_DCHECK_RUN_ON(&task_queue_);
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(&task_queue_);
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::RegisterTargetTransferRateObserver(
TargetTransferRateObserver* observer) {
task_queue_.PostTask([this, observer] {
RTC_DCHECK_RUN_ON(&task_queue_);
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 {
// TODO(bugs.webrtc.org/11438): Experiment with using more information/
// other conditions.
bool connected_changed = old_route.connected != new_route.connected;
bool 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()) {
bool relaying_changed = IsRelayed(old_route) != IsRelayed(new_route);
return connected_changed || route_ids_changed || relaying_changed;
} else {
return connected_changed || route_ids_changed;
}
}
void RtpTransportControllerSend::OnNetworkRouteChanged(
const std::string& transport_name,
const rtc::NetworkRoute& network_route) {
// 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;
}
absl::optional<BitrateConstraints> relay_constraint_update =
ApplyOrLiftRelayCap(IsRelayed(network_route));
// Check whether the network route has changed on each transport.
auto result =
network_routes_.insert(std::make_pair(transport_name, network_route));
auto kv = result.first;
bool inserted = result.second;
if (inserted || !(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);
}
task_queue_.PostTask([this, network_route] {
RTC_DCHECK_RUN_ON(&task_queue_);
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);
if (event_log_) {
event_log_->Log(std::make_unique<RtcEventRouteChange>(
network_route.connected, network_route.packet_overhead));
}
NetworkRouteChange msg;
msg.at_time = Timestamp::Millis(clock_->TimeInMilliseconds());
msg.constraints = ConvertConstraints(bitrate_config, clock_);
task_queue_.PostTask([this, msg, network_route] {
RTC_DCHECK_RUN_ON(&task_queue_);
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(&main_thread_);
RTC_LOG(LS_VERBOSE) << "SignalNetworkState "
<< (network_available ? "Up" : "Down");
NetworkAvailability msg;
msg.at_time = Timestamp::Millis(clock_->TimeInMilliseconds());
msg.network_available = network_available;
task_queue_.PostTask([this, msg]() {
RTC_DCHECK_RUN_ON(&task_queue_);
if (network_available_ == msg.network_available)
return;
network_available_ = msg.network_available;
if (network_available_) {
pacer()->Resume();
} else {
pacer()->Pause();
}
is_congested_ = false;
pacer()->SetCongested(false);
if (controller_) {
control_handler_->SetNetworkAvailability(network_available_);
PostUpdates(controller_->OnNetworkAvailability(msg));
UpdateControlState();
} else {
MaybeCreateControllers();
}
});
for (auto& rtp_sender : video_rtp_senders_) {
rtp_sender->OnNetworkAvailability(network_available);
}
}
RtcpBandwidthObserver* RtpTransportControllerSend::GetBandwidthObserver() {
return this;
}
int64_t RtpTransportControllerSend::GetPacerQueuingDelayMs() const {
return pacer()->OldestPacketWaitTime().ms();
}
absl::optional<Timestamp> RtpTransportControllerSend::GetFirstPacketTime()
const {
return pacer()->FirstSentPacketTime();
}
void RtpTransportControllerSend::EnablePeriodicAlrProbing(bool enable) {
task_queue_.PostTask([this, enable]() {
RTC_DCHECK_RUN_ON(&task_queue_);
streams_config_.requests_alr_probing = enable;
UpdateStreamsConfig();
});
}
void RtpTransportControllerSend::OnSentPacket(
const rtc::SentPacket& sent_packet) {
task_queue_.PostTask([this, sent_packet]() {
RTC_DCHECK_RUN_ON(&task_queue_);
absl::optional<SentPacket> packet_msg =
transport_feedback_adapter_.ProcessSentPacket(sent_packet);
if (packet_msg) {
// Only update outstanding data if:
// 1. Packet feadback 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(controller_->OnSentPacket(*packet_msg));
}
});
}
void RtpTransportControllerSend::OnReceivedPacket(
const ReceivedPacket& packet_msg) {
task_queue_.PostTask([this, packet_msg]() {
RTC_DCHECK_RUN_ON(&task_queue_);
if (controller_)
PostUpdates(controller_->OnReceivedPacket(packet_msg));
});
}
void RtpTransportControllerSend::UpdateBitrateConstraints(
const BitrateConstraints& updated) {
TargetRateConstraints msg = ConvertConstraints(updated, clock_);
task_queue_.PostTask([this, msg]() {
RTC_DCHECK_RUN_ON(&task_queue_);
if (controller_) {
PostUpdates(controller_->OnTargetRateConstraints(msg));
} else {
UpdateInitialConstraints(msg);
}
});
}
void RtpTransportControllerSend::SetSdpBitrateParameters(
const BitrateConstraints& constraints) {
absl::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) {
absl::optional<BitrateConstraints> updated =
bitrate_configurator_.UpdateWithClientPreferences(preferences);
if (updated.has_value()) {
UpdateBitrateConstraints(*updated);
} else {
RTC_LOG(LS_VERBOSE)
<< "WebRTC.RtpTransportControllerSend.SetClientBitratePreferences: "
"nothing to update";
}
}
absl::optional<BitrateConstraints>
RtpTransportControllerSend::ApplyOrLiftRelayCap(bool is_relayed) {
DataRate cap = is_relayed ? static_cast<DataRate>(relay_bandwidth_cap_)
: DataRate::PlusInfinity();
return bitrate_configurator_.UpdateWithRelayCap(cap);
}
void RtpTransportControllerSend::OnTransportOverheadChanged(
size_t transport_overhead_bytes_per_packet) {
RTC_DCHECK_RUN_ON(&main_thread_);
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() {
if (!pacer_started_) {
pacer_started_ = true;
if (pacer_settings_.use_task_queue_pacer()) {
task_queue_pacer_->EnsureStarted();
} else {
process_thread_->Start();
}
}
}
void RtpTransportControllerSend::OnReceivedEstimatedBitrate(uint32_t bitrate) {
RemoteBitrateReport msg;
msg.receive_time = Timestamp::Millis(clock_->TimeInMilliseconds());
msg.bandwidth = DataRate::BitsPerSec(bitrate);
task_queue_.PostTask([this, msg]() {
RTC_DCHECK_RUN_ON(&task_queue_);
if (controller_)
PostUpdates(controller_->OnRemoteBitrateReport(msg));
});
}
void RtpTransportControllerSend::OnReceivedRtcpReceiverReport(
const ReportBlockList& report_blocks,
int64_t rtt_ms,
int64_t now_ms) {
task_queue_.PostTask([this, report_blocks, now_ms]() {
RTC_DCHECK_RUN_ON(&task_queue_);
OnReceivedRtcpReceiverReportBlocks(report_blocks, now_ms);
});
task_queue_.PostTask([this, now_ms, rtt_ms]() {
RTC_DCHECK_RUN_ON(&task_queue_);
RoundTripTimeUpdate report;
report.receive_time = Timestamp::Millis(now_ms);
report.round_trip_time = TimeDelta::Millis(rtt_ms);
report.smoothed = false;
if (controller_ && !report.round_trip_time.IsZero())
PostUpdates(controller_->OnRoundTripTimeUpdate(report));
});
}
void RtpTransportControllerSend::OnAddPacket(
const RtpPacketSendInfo& packet_info) {
Timestamp creation_time = Timestamp::Millis(clock_->TimeInMilliseconds());
task_queue_.PostTask([this, packet_info, creation_time]() {
RTC_DCHECK_RUN_ON(&task_queue_);
feedback_demuxer_.AddPacket(packet_info);
transport_feedback_adapter_.AddPacket(
packet_info,
send_side_bwe_with_overhead_ ? transport_overhead_bytes_per_packet_ : 0,
creation_time);
});
}
void RtpTransportControllerSend::OnTransportFeedback(
const rtcp::TransportFeedback& feedback) {
auto feedback_time = Timestamp::Millis(clock_->TimeInMilliseconds());
task_queue_.PostTask([this, feedback, feedback_time]() {
RTC_DCHECK_RUN_ON(&task_queue_);
feedback_demuxer_.OnTransportFeedback(feedback);
absl::optional<TransportPacketsFeedback> feedback_msg =
transport_feedback_adapter_.ProcessTransportFeedback(feedback,
feedback_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) {
if (event_log_) {
event_log_->Log(std::make_unique<RtcEventRemoteEstimate>(
estimate.link_capacity_lower, estimate.link_capacity_upper));
}
estimate.update_time = Timestamp::Millis(clock_->TimeInMilliseconds());
task_queue_.PostTask([this, estimate] {
RTC_DCHECK_RUN_ON(&task_queue_);
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(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::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() {
if (!pacer_queue_update_task_.Running()) {
pacer_queue_update_task_ = RepeatingTaskHandle::DelayedStart(
task_queue_.Get(), kPacerQueueUpdateInterval, [this]() {
RTC_DCHECK_RUN_ON(&task_queue_);
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_.Get(), process_interval_, [this]() {
RTC_DCHECK_RUN_ON(&task_queue_);
UpdateControllerWithTimeInterval();
return process_interval_;
});
}
}
void RtpTransportControllerSend::UpdateControllerWithTimeInterval() {
RTC_DCHECK(controller_);
ProcessInterval msg;
msg.at_time = Timestamp::Millis(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(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());
}
for (const auto& probe : update.probe_cluster_configs) {
pacer()->CreateProbeCluster(probe.target_data_rate, probe.id);
}
if (update.target_rate) {
control_handler_->SetTargetRate(*update.target_rate);
UpdateControlState();
}
}
void RtpTransportControllerSend::OnReceivedRtcpReceiverReportBlocks(
const ReportBlockList& report_blocks,
int64_t now_ms) {
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 RTCPReportBlock& report_block : report_blocks) {
auto it = last_report_blocks_.find(report_block.source_ssrc);
if (it != last_report_blocks_.end()) {
auto number_of_packets = report_block.extended_highest_sequence_number -
it->second.extended_highest_sequence_number;
total_packets_delta += number_of_packets;
auto lost_delta = report_block.packets_lost - it->second.packets_lost;
total_packets_lost_delta += lost_delta;
}
last_report_blocks_[report_block.source_ssrc] = report_block;
}
// 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;
Timestamp now = Timestamp::Millis(now_ms);
TransportLossReport msg;
msg.packets_lost_delta = total_packets_lost_delta;
msg.packets_received_delta = packets_received_delta;
msg.receive_time = now;
msg.start_time = last_report_block_time_;
msg.end_time = now;
if (controller_)
PostUpdates(controller_->OnTransportLossReport(msg));
last_report_block_time_ = now;
}
} // namespace webrtc
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