blob: 9f173b1ad1a09ee38d3b9ab42b04964d3a0ca438 [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 <utility>
#include <vector>
#include "absl/memory/memory.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_transport_controller_send.h"
#include "call/rtp_video_sender.h"
#include "rtc_base/checks.h"
#include "rtc_base/location.h"
#include "rtc_base/logging.h"
#include "rtc_base/rate_limiter.h"
#include "system_wrappers/include/field_trial.h"
namespace webrtc {
class RtpTransportControllerSend::PeriodicTask : public rtc::QueuedTask {
public:
virtual void Stop() = 0;
};
namespace {
static const int64_t kRetransmitWindowSizeMs = 500;
static const size_t kMaxOverheadBytes = 500;
const int64_t PacerQueueUpdateIntervalMs = 25;
TargetRateConstraints ConvertConstraints(int min_bitrate_bps,
int max_bitrate_bps,
int start_bitrate_bps,
const Clock* clock) {
TargetRateConstraints msg;
msg.at_time = Timestamp::ms(clock->TimeInMilliseconds());
msg.min_data_rate =
min_bitrate_bps >= 0 ? DataRate::bps(min_bitrate_bps) : DataRate::Zero();
msg.max_data_rate = max_bitrate_bps > 0 ? DataRate::bps(max_bitrate_bps)
: DataRate::Infinity();
if (start_bitrate_bps > 0)
msg.starting_rate = DataRate::bps(start_bitrate_bps);
return msg;
}
TargetRateConstraints ConvertConstraints(const BitrateConstraints& contraints,
const Clock* clock) {
return ConvertConstraints(contraints.min_bitrate_bps,
contraints.max_bitrate_bps,
contraints.start_bitrate_bps, clock);
}
// The template closure pattern is based on rtc::ClosureTask.
template <class Closure>
class PeriodicTaskImpl final : public RtpTransportControllerSend::PeriodicTask {
public:
PeriodicTaskImpl(rtc::TaskQueue* task_queue,
int64_t period_ms,
Closure&& closure)
: task_queue_(task_queue),
period_ms_(period_ms),
closure_(std::forward<Closure>(closure)) {}
bool Run() override {
if (!running_)
return true;
closure_();
// absl::WrapUnique lets us repost this task on the TaskQueue.
task_queue_->PostDelayedTask(absl::WrapUnique(this), period_ms_);
// Return false to tell TaskQueue to not destruct this object, since we have
// taken ownership with absl::WrapUnique.
return false;
}
void Stop() override {
if (task_queue_->IsCurrent()) {
RTC_DCHECK(running_);
running_ = false;
} else {
task_queue_->PostTask([this] { Stop(); });
}
}
private:
rtc::TaskQueue* const task_queue_;
const int64_t period_ms_;
typename std::remove_const<
typename std::remove_reference<Closure>::type>::type closure_;
bool running_ = true;
};
template <class Closure>
static RtpTransportControllerSend::PeriodicTask* StartPeriodicTask(
rtc::TaskQueue* task_queue,
int64_t period_ms,
Closure&& closure) {
auto periodic_task = absl::make_unique<PeriodicTaskImpl<Closure>>(
task_queue, period_ms, std::forward<Closure>(closure));
RtpTransportControllerSend::PeriodicTask* periodic_task_ptr =
periodic_task.get();
task_queue->PostDelayedTask(std::move(periodic_task), period_ms);
return periodic_task_ptr;
}
} // namespace
RtpTransportControllerSend::RtpTransportControllerSend(
Clock* clock,
webrtc::RtcEventLog* event_log,
NetworkControllerFactoryInterface* controller_factory,
const BitrateConstraints& bitrate_config)
: clock_(clock),
pacer_(clock, &packet_router_, event_log),
bitrate_configurator_(bitrate_config),
process_thread_(ProcessThread::Create("SendControllerThread")),
observer_(nullptr),
transport_feedback_adapter_(clock_),
controller_factory_override_(controller_factory),
controller_factory_fallback_(
absl::make_unique<GoogCcNetworkControllerFactory>(event_log)),
process_interval_(controller_factory_fallback_->GetProcessInterval()),
last_report_block_time_(Timestamp::ms(clock_->TimeInMilliseconds())),
reset_feedback_on_route_change_(
!field_trial::IsEnabled("WebRTC-Bwe-NoFeedbackReset")),
send_side_bwe_with_overhead_(
webrtc::field_trial::IsEnabled("WebRTC-SendSideBwe-WithOverhead")),
add_pacing_to_cwin_(
field_trial::IsEnabled("WebRTC-AddPacingToCongestionWindowPushback")),
transport_overhead_bytes_per_packet_(0),
network_available_(false),
packet_feedback_available_(false),
pacer_queue_update_task_(nullptr),
controller_task_(nullptr),
retransmission_rate_limiter_(clock, kRetransmitWindowSizeMs),
task_queue_("rtp_send_controller") {
initial_config_.constraints = ConvertConstraints(bitrate_config, clock_);
RTC_DCHECK(bitrate_config.start_bitrate_bps > 0);
pacer_.SetPacingRates(bitrate_config.start_bitrate_bps, 0);
process_thread_->RegisterModule(&pacer_, RTC_FROM_HERE);
process_thread_->Start();
}
RtpTransportControllerSend::~RtpTransportControllerSend() {
process_thread_->Stop();
process_thread_->DeRegisterModule(&pacer_);
}
RtpVideoSenderInterface* RtpTransportControllerSend::CreateRtpVideoSender(
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) {
video_rtp_senders_.push_back(absl::make_unique<RtpVideoSender>(
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));
return video_rtp_senders_.back().get();
}
void RtpTransportControllerSend::DestroyRtpVideoSender(
RtpVideoSenderInterface* rtp_video_sender) {
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->network_estimate.bandwidth.bps());
// We won't create control_handler_ until we have an observers.
RTC_DCHECK(observer_ != nullptr);
observer_->OnTargetTransferRate(*update);
}
rtc::TaskQueue* RtpTransportControllerSend::GetWorkerQueue() {
return &task_queue_;
}
PacketRouter* RtpTransportControllerSend::packet_router() {
return &packet_router_;
}
TransportFeedbackObserver*
RtpTransportControllerSend::transport_feedback_observer() {
return this;
}
RtpPacketSender* RtpTransportControllerSend::packet_sender() {
return &pacer_;
}
const RtpKeepAliveConfig& RtpTransportControllerSend::keepalive_config() const {
return keepalive_;
}
void RtpTransportControllerSend::SetAllocatedSendBitrateLimits(
int min_send_bitrate_bps,
int max_padding_bitrate_bps,
int max_total_bitrate_bps) {
RTC_DCHECK_RUN_ON(&task_queue_);
streams_config_.min_pacing_rate = DataRate::bps(min_send_bitrate_bps);
streams_config_.max_padding_rate = DataRate::bps(max_padding_bitrate_bps);
streams_config_.max_total_allocated_bitrate =
DataRate::bps(max_total_bitrate_bps);
UpdateStreamsConfig();
}
void RtpTransportControllerSend::SetKeepAliveConfig(
const RtpKeepAliveConfig& config) {
keepalive_ = config;
}
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(limit_ms);
}
CallStatsObserver* RtpTransportControllerSend::GetCallStatsObserver() {
return this;
}
void RtpTransportControllerSend::RegisterPacketFeedbackObserver(
PacketFeedbackObserver* observer) {
transport_feedback_adapter_.RegisterPacketFeedbackObserver(observer);
}
void RtpTransportControllerSend::DeRegisterPacketFeedbackObserver(
PacketFeedbackObserver* observer) {
transport_feedback_adapter_.DeRegisterPacketFeedbackObserver(observer);
}
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();
});
}
void RtpTransportControllerSend::OnNetworkRouteChanged(
const std::string& transport_name,
const rtc::NetworkRoute& network_route) {
// Check if the network route is connected.
if (!network_route.connected) {
RTC_LOG(LS_INFO) << "Transport " << transport_name << " is disconnected";
// TODO(honghaiz): Perhaps handle this in SignalChannelNetworkState and
// consider merging these two methods.
return;
}
// 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) {
// No need to reset BWE if this is the first time the network connects.
return;
}
if (kv->second.connected != network_route.connected ||
kv->second.local_network_id != network_route.local_network_id ||
kv->second.remote_network_id != network_route.remote_network_id) {
kv->second = network_route;
BitrateConstraints bitrate_config = bitrate_configurator_.GetConfig();
RTC_LOG(LS_INFO) << "Network route changed on transport " << transport_name
<< ": new local network id "
<< network_route.local_network_id
<< " new remote network id "
<< network_route.remote_network_id
<< " 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 (reset_feedback_on_route_change_)
transport_feedback_adapter_.SetNetworkIds(
network_route.local_network_id, network_route.remote_network_id);
transport_overhead_bytes_per_packet_ = network_route.packet_overhead;
NetworkRouteChange msg;
msg.at_time = Timestamp::ms(clock_->TimeInMilliseconds());
msg.constraints = ConvertConstraints(bitrate_config, clock_);
task_queue_.PostTask([this, msg] {
RTC_DCHECK_RUN_ON(&task_queue_);
if (controller_) {
PostUpdates(controller_->OnNetworkRouteChange(msg));
} else {
UpdateInitialConstraints(msg.constraints);
}
pacer_.UpdateOutstandingData(0);
});
}
}
void RtpTransportControllerSend::OnNetworkAvailability(bool network_available) {
RTC_LOG(LS_INFO) << "SignalNetworkState "
<< (network_available ? "Up" : "Down");
NetworkAvailability msg;
msg.at_time = Timestamp::ms(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();
}
pacer_.UpdateOutstandingData(0);
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_.QueueInMs();
}
int64_t RtpTransportControllerSend::GetFirstPacketTimeMs() const {
return pacer_.FirstSentPacketTimeMs();
}
void RtpTransportControllerSend::SetPerPacketFeedbackAvailable(bool available) {
RTC_DCHECK_RUN_ON(&task_queue_);
packet_feedback_available_ = available;
if (!controller_)
MaybeCreateControllers();
}
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) {
absl::optional<SentPacket> packet_msg =
transport_feedback_adapter_.ProcessSentPacket(sent_packet);
if (packet_msg) {
task_queue_.PostTask([this, packet_msg]() {
RTC_DCHECK_RUN_ON(&task_queue_);
if (controller_)
PostUpdates(controller_->OnSentPacket(*packet_msg));
});
}
pacer_.UpdateOutstandingData(
transport_feedback_adapter_.GetOutstandingData().bytes());
}
void RtpTransportControllerSend::SetSdpBitrateParameters(
const BitrateConstraints& constraints) {
absl::optional<BitrateConstraints> updated =
bitrate_configurator_.UpdateWithSdpParameters(constraints);
if (updated.has_value()) {
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);
}
});
} 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()) {
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);
}
});
} else {
RTC_LOG(LS_VERBOSE)
<< "WebRTC.RtpTransportControllerSend.SetClientBitratePreferences: "
<< "nothing to update";
}
}
void RtpTransportControllerSend::SetAllocatedBitrateWithoutFeedback(
uint32_t bitrate_bps) {
// Audio transport feedback will not be reported in this mode, instead update
// acknowledged bitrate estimator with the bitrate allocated for audio.
if (field_trial::IsEnabled("WebRTC-Audio-ABWENoTWCC")) {
// TODO(srte): Make sure it's safe to always report this and remove the
// field trial check.
task_queue_.PostTask([this, bitrate_bps]() {
RTC_DCHECK_RUN_ON(&task_queue_);
streams_config_.unacknowledged_rate_allocation =
DataRate::bps(bitrate_bps);
UpdateStreamsConfig();
});
}
}
void RtpTransportControllerSend::OnTransportOverheadChanged(
size_t transport_overhead_bytes_per_packet) {
if (transport_overhead_bytes_per_packet >= kMaxOverheadBytes) {
RTC_LOG(LS_ERROR) << "Transport overhead exceeds " << kMaxOverheadBytes;
return;
}
// 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::OnReceivedEstimatedBitrate(uint32_t bitrate) {
RemoteBitrateReport msg;
msg.receive_time = Timestamp::ms(clock_->TimeInMilliseconds());
msg.bandwidth = DataRate::bps(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::ms(now_ms);
report.round_trip_time = TimeDelta::ms(rtt_ms);
report.smoothed = false;
if (controller_)
PostUpdates(controller_->OnRoundTripTimeUpdate(report));
});
}
void RtpTransportControllerSend::AddPacket(uint32_t ssrc,
uint16_t sequence_number,
size_t length,
const PacedPacketInfo& pacing_info) {
if (send_side_bwe_with_overhead_) {
length += transport_overhead_bytes_per_packet_;
}
transport_feedback_adapter_.AddPacket(ssrc, sequence_number, length,
pacing_info);
}
void RtpTransportControllerSend::OnTransportFeedback(
const rtcp::TransportFeedback& feedback) {
RTC_DCHECK_RUNS_SERIALIZED(&worker_race_);
absl::optional<TransportPacketsFeedback> feedback_msg =
transport_feedback_adapter_.ProcessTransportFeedback(feedback);
if (feedback_msg) {
task_queue_.PostTask([this, feedback_msg]() {
RTC_DCHECK_RUN_ON(&task_queue_);
if (controller_)
PostUpdates(controller_->OnTransportPacketsFeedback(*feedback_msg));
});
}
pacer_.UpdateOutstandingData(
transport_feedback_adapter_.GetOutstandingData().bytes());
}
void RtpTransportControllerSend::OnRttUpdate(int64_t avg_rtt_ms,
int64_t max_rtt_ms) {
int64_t now_ms = clock_->TimeInMilliseconds();
RoundTripTimeUpdate report;
report.receive_time = Timestamp::ms(now_ms);
report.round_trip_time = TimeDelta::ms(avg_rtt_ms);
report.smoothed = true;
task_queue_.PostTask([this, report]() {
RTC_DCHECK_RUN_ON(&task_queue_);
if (controller_)
PostUpdates(controller_->OnRoundTripTimeUpdate(report));
});
}
void RtpTransportControllerSend::MaybeCreateControllers() {
RTC_DCHECK(!controller_);
RTC_DCHECK(!control_handler_);
if (!network_available_ || !observer_)
return;
control_handler_ = absl::make_unique<CongestionControlHandler>();
initial_config_.constraints.at_time =
Timestamp::ms(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_) {
pacer_queue_update_task_ =
StartPeriodicTask(&task_queue_, PacerQueueUpdateIntervalMs, [this]() {
RTC_DCHECK_RUN_ON(&task_queue_);
TimeDelta expected_queue_time =
TimeDelta::ms(pacer_.ExpectedQueueTimeMs());
control_handler_->SetPacerQueue(expected_queue_time);
UpdateControlState();
});
}
if (controller_task_) {
// Stop is not synchronous, but is guaranteed to occur before the first
// invocation of the new controller task started below.
controller_task_->Stop();
controller_task_ = nullptr;
}
if (process_interval_.IsFinite()) {
// The controller task is owned by the task queue and lives until the task
// queue is destroyed or some time after Stop() is called, whichever comes
// first.
controller_task_ =
StartPeriodicTask(&task_queue_, process_interval_.ms(), [this]() {
RTC_DCHECK_RUN_ON(&task_queue_);
UpdateControllerWithTimeInterval();
});
}
}
void RtpTransportControllerSend::UpdateControllerWithTimeInterval() {
RTC_DCHECK(controller_);
ProcessInterval msg;
msg.at_time = Timestamp::ms(clock_->TimeInMilliseconds());
if (add_pacing_to_cwin_)
msg.pacer_queue = DataSize::bytes(pacer_.QueueSizeBytes());
PostUpdates(controller_->OnProcessInterval(msg));
}
void RtpTransportControllerSend::UpdateStreamsConfig() {
streams_config_.at_time = Timestamp::ms(clock_->TimeInMilliseconds());
if (controller_)
PostUpdates(controller_->OnStreamsConfig(streams_config_));
}
void RtpTransportControllerSend::PostUpdates(NetworkControlUpdate update) {
if (update.congestion_window) {
if (update.congestion_window->IsFinite())
pacer_.SetCongestionWindow(update.congestion_window->bytes());
else
pacer_.SetCongestionWindow(PacedSender::kNoCongestionWindow);
}
if (update.pacer_config) {
pacer_.SetPacingRates(update.pacer_config->data_rate().bps(),
update.pacer_config->pad_rate().bps());
}
for (const auto& probe : update.probe_cluster_configs) {
int64_t bitrate_bps = probe.target_data_rate.bps();
pacer_.CreateProbeCluster(bitrate_bps);
}
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::ms(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