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744 lines
26 KiB
C++
744 lines
26 KiB
C++
/*
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* Copyright (c) 2019 The WebRTC project authors. All Rights Reserved.
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*
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* Use of this source code is governed by a BSD-style license
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* that can be found in the LICENSE file in the root of the source
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* tree. An additional intellectual property rights grant can be found
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* in the file PATENTS. All contributing project authors may
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* be found in the AUTHORS file in the root of the source tree.
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*/
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#include "modules/pacing/pacing_controller.h"
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#include <algorithm>
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#include <memory>
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#include <utility>
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#include <vector>
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#include "absl/strings/match.h"
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#include "modules/pacing/bitrate_prober.h"
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#include "modules/pacing/interval_budget.h"
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#include "modules/utility/include/process_thread.h"
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#include "rtc_base/checks.h"
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#include "rtc_base/experiments/field_trial_parser.h"
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#include "rtc_base/logging.h"
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#include "rtc_base/time_utils.h"
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#include "system_wrappers/include/clock.h"
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namespace webrtc {
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namespace {
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// Time limit in milliseconds between packet bursts.
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constexpr TimeDelta kDefaultMinPacketLimit = TimeDelta::Millis(5);
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constexpr TimeDelta kCongestedPacketInterval = TimeDelta::Millis(500);
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// TODO(sprang): Consider dropping this limit.
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// The maximum debt level, in terms of time, capped when sending packets.
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constexpr TimeDelta kMaxDebtInTime = TimeDelta::Millis(500);
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constexpr TimeDelta kMaxElapsedTime = TimeDelta::Seconds(2);
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// Upper cap on process interval, in case process has not been called in a long
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// time. Applies only to periodic mode.
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constexpr TimeDelta kMaxProcessingInterval = TimeDelta::Millis(30);
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// Allow probes to be processed slightly ahead of inteded send time. Currently
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// set to 1ms as this is intended to allow times be rounded down to the nearest
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// millisecond.
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constexpr TimeDelta kMaxEarlyProbeProcessing = TimeDelta::Millis(1);
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constexpr int kFirstPriority = 0;
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bool IsDisabled(const WebRtcKeyValueConfig& field_trials,
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absl::string_view key) {
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return absl::StartsWith(field_trials.Lookup(key), "Disabled");
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}
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bool IsEnabled(const WebRtcKeyValueConfig& field_trials,
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absl::string_view key) {
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return absl::StartsWith(field_trials.Lookup(key), "Enabled");
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}
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TimeDelta GetDynamicPaddingTarget(const WebRtcKeyValueConfig& field_trials) {
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FieldTrialParameter<TimeDelta> padding_target("timedelta",
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TimeDelta::Millis(5));
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ParseFieldTrial({&padding_target},
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field_trials.Lookup("WebRTC-Pacer-DynamicPaddingTarget"));
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return padding_target.Get();
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}
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int GetPriorityForType(RtpPacketMediaType type) {
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// Lower number takes priority over higher.
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switch (type) {
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case RtpPacketMediaType::kAudio:
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// Audio is always prioritized over other packet types.
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return kFirstPriority + 1;
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case RtpPacketMediaType::kRetransmission:
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// Send retransmissions before new media.
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return kFirstPriority + 2;
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case RtpPacketMediaType::kVideo:
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case RtpPacketMediaType::kForwardErrorCorrection:
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// Video has "normal" priority, in the old speak.
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// Send redundancy concurrently to video. If it is delayed it might have a
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// lower chance of being useful.
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return kFirstPriority + 3;
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case RtpPacketMediaType::kPadding:
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// Packets that are in themselves likely useless, only sent to keep the
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// BWE high.
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return kFirstPriority + 4;
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}
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RTC_CHECK_NOTREACHED();
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}
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} // namespace
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const TimeDelta PacingController::kMaxExpectedQueueLength =
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TimeDelta::Millis(2000);
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const float PacingController::kDefaultPaceMultiplier = 2.5f;
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const TimeDelta PacingController::kPausedProcessInterval =
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kCongestedPacketInterval;
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const TimeDelta PacingController::kMinSleepTime = TimeDelta::Millis(1);
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PacingController::PacingController(Clock* clock,
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PacketSender* packet_sender,
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RtcEventLog* event_log,
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const WebRtcKeyValueConfig* field_trials,
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ProcessMode mode)
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: mode_(mode),
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clock_(clock),
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packet_sender_(packet_sender),
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fallback_field_trials_(
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!field_trials ? std::make_unique<FieldTrialBasedConfig>() : nullptr),
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field_trials_(field_trials ? field_trials : fallback_field_trials_.get()),
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drain_large_queues_(
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!IsDisabled(*field_trials_, "WebRTC-Pacer-DrainQueue")),
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send_padding_if_silent_(
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IsEnabled(*field_trials_, "WebRTC-Pacer-PadInSilence")),
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pace_audio_(IsEnabled(*field_trials_, "WebRTC-Pacer-BlockAudio")),
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ignore_transport_overhead_(
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IsEnabled(*field_trials_, "WebRTC-Pacer-IgnoreTransportOverhead")),
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padding_target_duration_(GetDynamicPaddingTarget(*field_trials_)),
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min_packet_limit_(kDefaultMinPacketLimit),
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transport_overhead_per_packet_(DataSize::Zero()),
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last_timestamp_(clock_->CurrentTime()),
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paused_(false),
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media_budget_(0),
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padding_budget_(0),
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media_debt_(DataSize::Zero()),
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padding_debt_(DataSize::Zero()),
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media_rate_(DataRate::Zero()),
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padding_rate_(DataRate::Zero()),
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prober_(*field_trials_),
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probing_send_failure_(false),
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pacing_bitrate_(DataRate::Zero()),
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last_process_time_(clock->CurrentTime()),
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last_send_time_(last_process_time_),
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packet_queue_(last_process_time_, field_trials_),
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packet_counter_(0),
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congestion_window_size_(DataSize::PlusInfinity()),
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outstanding_data_(DataSize::Zero()),
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queue_time_limit(kMaxExpectedQueueLength),
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account_for_audio_(false),
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include_overhead_(false) {
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if (!drain_large_queues_) {
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RTC_LOG(LS_WARNING) << "Pacer queues will not be drained,"
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"pushback experiment must be enabled.";
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}
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FieldTrialParameter<int> min_packet_limit_ms("", min_packet_limit_.ms());
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ParseFieldTrial({&min_packet_limit_ms},
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field_trials_->Lookup("WebRTC-Pacer-MinPacketLimitMs"));
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min_packet_limit_ = TimeDelta::Millis(min_packet_limit_ms.Get());
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UpdateBudgetWithElapsedTime(min_packet_limit_);
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}
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PacingController::~PacingController() = default;
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void PacingController::CreateProbeCluster(DataRate bitrate, int cluster_id) {
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prober_.CreateProbeCluster(bitrate, CurrentTime(), cluster_id);
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}
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void PacingController::Pause() {
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if (!paused_)
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RTC_LOG(LS_INFO) << "PacedSender paused.";
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paused_ = true;
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packet_queue_.SetPauseState(true, CurrentTime());
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}
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void PacingController::Resume() {
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if (paused_)
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RTC_LOG(LS_INFO) << "PacedSender resumed.";
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paused_ = false;
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packet_queue_.SetPauseState(false, CurrentTime());
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}
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bool PacingController::IsPaused() const {
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return paused_;
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}
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void PacingController::SetCongestionWindow(DataSize congestion_window_size) {
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const bool was_congested = Congested();
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congestion_window_size_ = congestion_window_size;
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if (was_congested && !Congested()) {
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TimeDelta elapsed_time = UpdateTimeAndGetElapsed(CurrentTime());
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UpdateBudgetWithElapsedTime(elapsed_time);
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}
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}
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void PacingController::UpdateOutstandingData(DataSize outstanding_data) {
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const bool was_congested = Congested();
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outstanding_data_ = outstanding_data;
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if (was_congested && !Congested()) {
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TimeDelta elapsed_time = UpdateTimeAndGetElapsed(CurrentTime());
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UpdateBudgetWithElapsedTime(elapsed_time);
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}
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}
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bool PacingController::Congested() const {
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if (congestion_window_size_.IsFinite()) {
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return outstanding_data_ >= congestion_window_size_;
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}
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return false;
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}
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bool PacingController::IsProbing() const {
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return prober_.is_probing();
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}
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Timestamp PacingController::CurrentTime() const {
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Timestamp time = clock_->CurrentTime();
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if (time < last_timestamp_) {
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RTC_LOG(LS_WARNING)
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<< "Non-monotonic clock behavior observed. Previous timestamp: "
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<< last_timestamp_.ms() << ", new timestamp: " << time.ms();
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RTC_DCHECK_GE(time, last_timestamp_);
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time = last_timestamp_;
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}
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last_timestamp_ = time;
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return time;
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}
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void PacingController::SetProbingEnabled(bool enabled) {
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RTC_CHECK_EQ(0, packet_counter_);
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prober_.SetEnabled(enabled);
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}
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void PacingController::SetPacingRates(DataRate pacing_rate,
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DataRate padding_rate) {
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RTC_DCHECK_GT(pacing_rate, DataRate::Zero());
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media_rate_ = pacing_rate;
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padding_rate_ = padding_rate;
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pacing_bitrate_ = pacing_rate;
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padding_budget_.set_target_rate_kbps(padding_rate.kbps());
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RTC_LOG(LS_VERBOSE) << "bwe:pacer_updated pacing_kbps="
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<< pacing_bitrate_.kbps()
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<< " padding_budget_kbps=" << padding_rate.kbps();
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}
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void PacingController::EnqueuePacket(std::unique_ptr<RtpPacketToSend> packet) {
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RTC_DCHECK(pacing_bitrate_ > DataRate::Zero())
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<< "SetPacingRate must be called before InsertPacket.";
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RTC_CHECK(packet->packet_type());
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// Get priority first and store in temporary, to avoid chance of object being
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// moved before GetPriorityForType() being called.
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const int priority = GetPriorityForType(*packet->packet_type());
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EnqueuePacketInternal(std::move(packet), priority);
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}
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void PacingController::SetAccountForAudioPackets(bool account_for_audio) {
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account_for_audio_ = account_for_audio;
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}
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void PacingController::SetIncludeOverhead() {
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include_overhead_ = true;
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packet_queue_.SetIncludeOverhead();
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}
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void PacingController::SetTransportOverhead(DataSize overhead_per_packet) {
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if (ignore_transport_overhead_)
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return;
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transport_overhead_per_packet_ = overhead_per_packet;
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packet_queue_.SetTransportOverhead(overhead_per_packet);
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}
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TimeDelta PacingController::ExpectedQueueTime() const {
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RTC_DCHECK_GT(pacing_bitrate_, DataRate::Zero());
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return TimeDelta::Millis(
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(QueueSizeData().bytes() * 8 * rtc::kNumMillisecsPerSec) /
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pacing_bitrate_.bps());
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}
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size_t PacingController::QueueSizePackets() const {
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return packet_queue_.SizeInPackets();
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}
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DataSize PacingController::QueueSizeData() const {
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return packet_queue_.Size();
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}
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DataSize PacingController::CurrentBufferLevel() const {
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return std::max(media_debt_, padding_debt_);
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}
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absl::optional<Timestamp> PacingController::FirstSentPacketTime() const {
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return first_sent_packet_time_;
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}
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TimeDelta PacingController::OldestPacketWaitTime() const {
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Timestamp oldest_packet = packet_queue_.OldestEnqueueTime();
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if (oldest_packet.IsInfinite()) {
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return TimeDelta::Zero();
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}
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return CurrentTime() - oldest_packet;
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}
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void PacingController::EnqueuePacketInternal(
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std::unique_ptr<RtpPacketToSend> packet,
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int priority) {
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prober_.OnIncomingPacket(DataSize::Bytes(packet->payload_size()));
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Timestamp now = CurrentTime();
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if (mode_ == ProcessMode::kDynamic && packet_queue_.Empty() &&
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NextSendTime() <= now) {
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TimeDelta elapsed_time = UpdateTimeAndGetElapsed(now);
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UpdateBudgetWithElapsedTime(elapsed_time);
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}
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packet_queue_.Push(priority, now, packet_counter_++, std::move(packet));
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}
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TimeDelta PacingController::UpdateTimeAndGetElapsed(Timestamp now) {
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// If no previous processing, or last process was "in the future" because of
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// early probe processing, then there is no elapsed time to add budget for.
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if (last_process_time_.IsMinusInfinity() || now < last_process_time_) {
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return TimeDelta::Zero();
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}
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RTC_DCHECK_GE(now, last_process_time_);
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TimeDelta elapsed_time = now - last_process_time_;
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last_process_time_ = now;
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if (elapsed_time > kMaxElapsedTime) {
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RTC_LOG(LS_WARNING) << "Elapsed time (" << elapsed_time.ms()
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<< " ms) longer than expected, limiting to "
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<< kMaxElapsedTime.ms();
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elapsed_time = kMaxElapsedTime;
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}
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return elapsed_time;
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}
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bool PacingController::ShouldSendKeepalive(Timestamp now) const {
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if (send_padding_if_silent_ || paused_ || Congested() ||
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packet_counter_ == 0) {
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// We send a padding packet every 500 ms to ensure we won't get stuck in
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// congested state due to no feedback being received.
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TimeDelta elapsed_since_last_send = now - last_send_time_;
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if (elapsed_since_last_send >= kCongestedPacketInterval) {
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return true;
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}
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}
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return false;
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}
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Timestamp PacingController::NextSendTime() const {
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const Timestamp now = CurrentTime();
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if (paused_) {
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return last_send_time_ + kPausedProcessInterval;
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}
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// If probing is active, that always takes priority.
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if (prober_.is_probing()) {
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Timestamp probe_time = prober_.NextProbeTime(now);
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// |probe_time| == PlusInfinity indicates no probe scheduled.
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if (probe_time != Timestamp::PlusInfinity() && !probing_send_failure_) {
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return probe_time;
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}
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}
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if (mode_ == ProcessMode::kPeriodic) {
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// In periodic non-probing mode, we just have a fixed interval.
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return last_process_time_ + min_packet_limit_;
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}
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// In dynamic mode, figure out when the next packet should be sent,
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// given the current conditions.
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if (!pace_audio_) {
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// Not pacing audio, if leading packet is audio its target send
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// time is the time at which it was enqueued.
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absl::optional<Timestamp> audio_enqueue_time =
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packet_queue_.LeadingAudioPacketEnqueueTime();
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if (audio_enqueue_time.has_value()) {
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return *audio_enqueue_time;
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}
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}
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if (Congested() || packet_counter_ == 0) {
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// We need to at least send keep-alive packets with some interval.
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return last_send_time_ + kCongestedPacketInterval;
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}
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// Check how long until we can send the next media packet.
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if (media_rate_ > DataRate::Zero() && !packet_queue_.Empty()) {
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return std::min(last_send_time_ + kPausedProcessInterval,
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last_process_time_ + media_debt_ / media_rate_);
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}
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// If we _don't_ have pending packets, check how long until we have
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// bandwidth for padding packets. Both media and padding debts must
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// have been drained to do this.
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if (padding_rate_ > DataRate::Zero() && packet_queue_.Empty()) {
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TimeDelta drain_time =
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std::max(media_debt_ / media_rate_, padding_debt_ / padding_rate_);
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return std::min(last_send_time_ + kPausedProcessInterval,
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last_process_time_ + drain_time);
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}
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if (send_padding_if_silent_) {
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return last_send_time_ + kPausedProcessInterval;
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}
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return last_process_time_ + kPausedProcessInterval;
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}
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void PacingController::ProcessPackets() {
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Timestamp now = CurrentTime();
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Timestamp target_send_time = now;
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if (mode_ == ProcessMode::kDynamic) {
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target_send_time = NextSendTime();
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TimeDelta early_execute_margin =
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prober_.is_probing() ? kMaxEarlyProbeProcessing : TimeDelta::Zero();
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if (target_send_time.IsMinusInfinity()) {
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target_send_time = now;
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} else if (now < target_send_time - early_execute_margin) {
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// We are too early, but if queue is empty still allow draining some debt.
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// Probing is allowed to be sent up to kMinSleepTime early.
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TimeDelta elapsed_time = UpdateTimeAndGetElapsed(now);
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UpdateBudgetWithElapsedTime(elapsed_time);
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return;
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}
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if (target_send_time < last_process_time_) {
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// After the last process call, at time X, the target send time
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// shifted to be earlier than X. This should normally not happen
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// but we want to make sure rounding errors or erratic behavior
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// of NextSendTime() does not cause issue. In particular, if the
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// buffer reduction of
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// rate * (target_send_time - previous_process_time)
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// in the main loop doesn't clean up the existing debt we may not
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// be able to send again. We don't want to check this reordering
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// there as it is the normal exit condtion when the buffer is
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// exhausted and there are packets in the queue.
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UpdateBudgetWithElapsedTime(last_process_time_ - target_send_time);
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target_send_time = last_process_time_;
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}
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}
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Timestamp previous_process_time = last_process_time_;
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TimeDelta elapsed_time = UpdateTimeAndGetElapsed(now);
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if (ShouldSendKeepalive(now)) {
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// We can not send padding unless a normal packet has first been sent. If
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// we do, timestamps get messed up.
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if (packet_counter_ == 0) {
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last_send_time_ = now;
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} else {
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DataSize keepalive_data_sent = DataSize::Zero();
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std::vector<std::unique_ptr<RtpPacketToSend>> keepalive_packets =
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packet_sender_->GeneratePadding(DataSize::Bytes(1));
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for (auto& packet : keepalive_packets) {
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keepalive_data_sent +=
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DataSize::Bytes(packet->payload_size() + packet->padding_size());
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packet_sender_->SendPacket(std::move(packet), PacedPacketInfo());
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for (auto& packet : packet_sender_->FetchFec()) {
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EnqueuePacket(std::move(packet));
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}
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}
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OnPaddingSent(keepalive_data_sent);
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}
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}
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if (paused_) {
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return;
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}
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if (elapsed_time > TimeDelta::Zero()) {
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DataRate target_rate = pacing_bitrate_;
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DataSize queue_size_data = packet_queue_.Size();
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if (queue_size_data > DataSize::Zero()) {
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// Assuming equal size packets and input/output rate, the average packet
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// has avg_time_left_ms left to get queue_size_bytes out of the queue, if
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// time constraint shall be met. Determine bitrate needed for that.
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packet_queue_.UpdateQueueTime(now);
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if (drain_large_queues_) {
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TimeDelta avg_time_left =
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std::max(TimeDelta::Millis(1),
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queue_time_limit - packet_queue_.AverageQueueTime());
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DataRate min_rate_needed = queue_size_data / avg_time_left;
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if (min_rate_needed > target_rate) {
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target_rate = min_rate_needed;
|
|
RTC_LOG(LS_VERBOSE) << "bwe:large_pacing_queue pacing_rate_kbps="
|
|
<< target_rate.kbps();
|
|
}
|
|
}
|
|
}
|
|
|
|
if (mode_ == ProcessMode::kPeriodic) {
|
|
// In periodic processing mode, the IntevalBudget allows positive budget
|
|
// up to (process interval duration) * (target rate), so we only need to
|
|
// update it once before the packet sending loop.
|
|
media_budget_.set_target_rate_kbps(target_rate.kbps());
|
|
UpdateBudgetWithElapsedTime(elapsed_time);
|
|
} else {
|
|
media_rate_ = target_rate;
|
|
}
|
|
}
|
|
|
|
bool first_packet_in_probe = false;
|
|
PacedPacketInfo pacing_info;
|
|
DataSize recommended_probe_size = DataSize::Zero();
|
|
bool is_probing = prober_.is_probing();
|
|
if (is_probing) {
|
|
// Probe timing is sensitive, and handled explicitly by BitrateProber, so
|
|
// use actual send time rather than target.
|
|
pacing_info = prober_.CurrentCluster(now).value_or(PacedPacketInfo());
|
|
if (pacing_info.probe_cluster_id != PacedPacketInfo::kNotAProbe) {
|
|
first_packet_in_probe = pacing_info.probe_cluster_bytes_sent == 0;
|
|
recommended_probe_size = prober_.RecommendedMinProbeSize();
|
|
RTC_DCHECK_GT(recommended_probe_size, DataSize::Zero());
|
|
} else {
|
|
// No valid probe cluster returned, probe might have timed out.
|
|
is_probing = false;
|
|
}
|
|
}
|
|
|
|
DataSize data_sent = DataSize::Zero();
|
|
|
|
// The paused state is checked in the loop since it leaves the critical
|
|
// section allowing the paused state to be changed from other code.
|
|
while (!paused_) {
|
|
if (first_packet_in_probe) {
|
|
// If first packet in probe, insert a small padding packet so we have a
|
|
// more reliable start window for the rate estimation.
|
|
auto padding = packet_sender_->GeneratePadding(DataSize::Bytes(1));
|
|
// If no RTP modules sending media are registered, we may not get a
|
|
// padding packet back.
|
|
if (!padding.empty()) {
|
|
// Insert with high priority so larger media packets don't preempt it.
|
|
EnqueuePacketInternal(std::move(padding[0]), kFirstPriority);
|
|
// We should never get more than one padding packets with a requested
|
|
// size of 1 byte.
|
|
RTC_DCHECK_EQ(padding.size(), 1u);
|
|
}
|
|
first_packet_in_probe = false;
|
|
}
|
|
|
|
if (mode_ == ProcessMode::kDynamic &&
|
|
previous_process_time < target_send_time) {
|
|
// Reduce buffer levels with amount corresponding to time between last
|
|
// process and target send time for the next packet.
|
|
// If the process call is late, that may be the time between the optimal
|
|
// send times for two packets we should already have sent.
|
|
UpdateBudgetWithElapsedTime(target_send_time - previous_process_time);
|
|
previous_process_time = target_send_time;
|
|
}
|
|
|
|
// Fetch the next packet, so long as queue is not empty or budget is not
|
|
// exhausted.
|
|
std::unique_ptr<RtpPacketToSend> rtp_packet =
|
|
GetPendingPacket(pacing_info, target_send_time, now);
|
|
|
|
if (rtp_packet == nullptr) {
|
|
// No packet available to send, check if we should send padding.
|
|
DataSize padding_to_add = PaddingToAdd(recommended_probe_size, data_sent);
|
|
if (padding_to_add > DataSize::Zero()) {
|
|
std::vector<std::unique_ptr<RtpPacketToSend>> padding_packets =
|
|
packet_sender_->GeneratePadding(padding_to_add);
|
|
if (padding_packets.empty()) {
|
|
// No padding packets were generated, quite send loop.
|
|
break;
|
|
}
|
|
for (auto& packet : padding_packets) {
|
|
EnqueuePacket(std::move(packet));
|
|
}
|
|
// Continue loop to send the padding that was just added.
|
|
continue;
|
|
}
|
|
|
|
// Can't fetch new packet and no padding to send, exit send loop.
|
|
break;
|
|
}
|
|
|
|
RTC_DCHECK(rtp_packet);
|
|
RTC_DCHECK(rtp_packet->packet_type().has_value());
|
|
const RtpPacketMediaType packet_type = *rtp_packet->packet_type();
|
|
DataSize packet_size = DataSize::Bytes(rtp_packet->payload_size() +
|
|
rtp_packet->padding_size());
|
|
|
|
if (include_overhead_) {
|
|
packet_size += DataSize::Bytes(rtp_packet->headers_size()) +
|
|
transport_overhead_per_packet_;
|
|
}
|
|
|
|
packet_sender_->SendPacket(std::move(rtp_packet), pacing_info);
|
|
for (auto& packet : packet_sender_->FetchFec()) {
|
|
EnqueuePacket(std::move(packet));
|
|
}
|
|
data_sent += packet_size;
|
|
|
|
// Send done, update send/process time to the target send time.
|
|
OnPacketSent(packet_type, packet_size, target_send_time);
|
|
|
|
// If we are currently probing, we need to stop the send loop when we have
|
|
// reached the send target.
|
|
if (is_probing && data_sent >= recommended_probe_size) {
|
|
break;
|
|
}
|
|
|
|
if (mode_ == ProcessMode::kDynamic) {
|
|
// Update target send time in case that are more packets that we are late
|
|
// in processing.
|
|
Timestamp next_send_time = NextSendTime();
|
|
if (next_send_time.IsMinusInfinity()) {
|
|
target_send_time = now;
|
|
} else {
|
|
target_send_time = std::min(now, next_send_time);
|
|
}
|
|
}
|
|
}
|
|
|
|
last_process_time_ = std::max(last_process_time_, previous_process_time);
|
|
|
|
if (is_probing) {
|
|
probing_send_failure_ = data_sent == DataSize::Zero();
|
|
if (!probing_send_failure_) {
|
|
prober_.ProbeSent(CurrentTime(), data_sent);
|
|
}
|
|
}
|
|
}
|
|
|
|
DataSize PacingController::PaddingToAdd(DataSize recommended_probe_size,
|
|
DataSize data_sent) const {
|
|
if (!packet_queue_.Empty()) {
|
|
// Actual payload available, no need to add padding.
|
|
return DataSize::Zero();
|
|
}
|
|
|
|
if (Congested()) {
|
|
// Don't add padding if congested, even if requested for probing.
|
|
return DataSize::Zero();
|
|
}
|
|
|
|
if (packet_counter_ == 0) {
|
|
// We can not send padding unless a normal packet has first been sent. If we
|
|
// do, timestamps get messed up.
|
|
return DataSize::Zero();
|
|
}
|
|
|
|
if (!recommended_probe_size.IsZero()) {
|
|
if (recommended_probe_size > data_sent) {
|
|
return recommended_probe_size - data_sent;
|
|
}
|
|
return DataSize::Zero();
|
|
}
|
|
|
|
if (mode_ == ProcessMode::kPeriodic) {
|
|
return DataSize::Bytes(padding_budget_.bytes_remaining());
|
|
} else if (padding_rate_ > DataRate::Zero() &&
|
|
padding_debt_ == DataSize::Zero()) {
|
|
return padding_target_duration_ * padding_rate_;
|
|
}
|
|
return DataSize::Zero();
|
|
}
|
|
|
|
std::unique_ptr<RtpPacketToSend> PacingController::GetPendingPacket(
|
|
const PacedPacketInfo& pacing_info,
|
|
Timestamp target_send_time,
|
|
Timestamp now) {
|
|
if (packet_queue_.Empty()) {
|
|
return nullptr;
|
|
}
|
|
|
|
// First, check if there is any reason _not_ to send the next queued packet.
|
|
|
|
// Unpaced audio packets and probes are exempted from send checks.
|
|
bool unpaced_audio_packet =
|
|
!pace_audio_ && packet_queue_.LeadingAudioPacketEnqueueTime().has_value();
|
|
bool is_probe = pacing_info.probe_cluster_id != PacedPacketInfo::kNotAProbe;
|
|
if (!unpaced_audio_packet && !is_probe) {
|
|
if (Congested()) {
|
|
// Don't send anything if congested.
|
|
return nullptr;
|
|
}
|
|
|
|
if (mode_ == ProcessMode::kPeriodic) {
|
|
if (media_budget_.bytes_remaining() <= 0) {
|
|
// Not enough budget.
|
|
return nullptr;
|
|
}
|
|
} else {
|
|
// Dynamic processing mode.
|
|
if (now <= target_send_time) {
|
|
// We allow sending slightly early if we think that we would actually
|
|
// had been able to, had we been right on time - i.e. the current debt
|
|
// is not more than would be reduced to zero at the target sent time.
|
|
TimeDelta flush_time = media_debt_ / media_rate_;
|
|
if (now + flush_time > target_send_time) {
|
|
return nullptr;
|
|
}
|
|
}
|
|
}
|
|
}
|
|
|
|
return packet_queue_.Pop();
|
|
}
|
|
|
|
void PacingController::OnPacketSent(RtpPacketMediaType packet_type,
|
|
DataSize packet_size,
|
|
Timestamp send_time) {
|
|
if (!first_sent_packet_time_) {
|
|
first_sent_packet_time_ = send_time;
|
|
}
|
|
bool audio_packet = packet_type == RtpPacketMediaType::kAudio;
|
|
if (!audio_packet || account_for_audio_) {
|
|
// Update media bytes sent.
|
|
UpdateBudgetWithSentData(packet_size);
|
|
}
|
|
last_send_time_ = send_time;
|
|
last_process_time_ = send_time;
|
|
}
|
|
|
|
void PacingController::OnPaddingSent(DataSize data_sent) {
|
|
if (data_sent > DataSize::Zero()) {
|
|
UpdateBudgetWithSentData(data_sent);
|
|
}
|
|
Timestamp now = CurrentTime();
|
|
last_send_time_ = now;
|
|
last_process_time_ = now;
|
|
}
|
|
|
|
void PacingController::UpdateBudgetWithElapsedTime(TimeDelta delta) {
|
|
if (mode_ == ProcessMode::kPeriodic) {
|
|
delta = std::min(kMaxProcessingInterval, delta);
|
|
media_budget_.IncreaseBudget(delta.ms());
|
|
padding_budget_.IncreaseBudget(delta.ms());
|
|
} else {
|
|
media_debt_ -= std::min(media_debt_, media_rate_ * delta);
|
|
padding_debt_ -= std::min(padding_debt_, padding_rate_ * delta);
|
|
}
|
|
}
|
|
|
|
void PacingController::UpdateBudgetWithSentData(DataSize size) {
|
|
outstanding_data_ += size;
|
|
if (mode_ == ProcessMode::kPeriodic) {
|
|
media_budget_.UseBudget(size.bytes());
|
|
padding_budget_.UseBudget(size.bytes());
|
|
} else {
|
|
media_debt_ += size;
|
|
media_debt_ = std::min(media_debt_, media_rate_ * kMaxDebtInTime);
|
|
padding_debt_ += size;
|
|
padding_debt_ = std::min(padding_debt_, padding_rate_ * kMaxDebtInTime);
|
|
}
|
|
}
|
|
|
|
void PacingController::SetQueueTimeLimit(TimeDelta limit) {
|
|
queue_time_limit = limit;
|
|
}
|
|
|
|
} // namespace webrtc
|