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* Copyright (c) 2019 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 <stddef.h>
#include <stdint.h>
#include <atomic>
#include <memory>
#include <vector>
#include "absl/types/optional.h"
#include "api/function_view.h"
#include "api/rtc_event_log/rtc_event_log.h"
#include "api/transport/field_trial_based_config.h"
#include "api/transport/network_types.h"
#include "api/transport/webrtc_key_value_config.h"
#include "modules/pacing/bitrate_prober.h"
#include "modules/pacing/interval_budget.h"
#include "modules/pacing/round_robin_packet_queue.h"
#include "modules/pacing/rtp_packet_pacer.h"
#include "modules/rtp_rtcp/include/rtp_packet_sender.h"
#include "modules/rtp_rtcp/include/rtp_rtcp_defines.h"
#include "modules/rtp_rtcp/source/rtp_packet_to_send.h"
#include "rtc_base/critical_section.h"
#include "rtc_base/experiments/field_trial_parser.h"
#include "rtc_base/thread_annotations.h"
namespace webrtc {
// This class implements a leaky-bucket packet pacing algorithm. It handles the
// logic of determining which packets to send when, but the actual timing of
// the processing is done externally (e.g. PacedSender). Furthermore, the
// forwarding of packets when they are ready to be sent is also handled
// externally, via the PacedSendingController::PacketSender interface.
class PacingController {
// Periodic mode uses the IntervalBudget class for tracking bitrate
// budgets, and expected ProcessPackets() to be called a fixed rate,
// e.g. every 5ms as implemented by PacedSender.
// Dynamic mode allows for arbitrary time delta between calls to
// ProcessPackets.
enum class ProcessMode { kPeriodic, kDynamic };
class PacketSender {
virtual ~PacketSender() = default;
virtual void SendPacket(std::unique_ptr<RtpPacketToSend> packet,
const PacedPacketInfo& cluster_info) = 0;
virtual std::vector<std::unique_ptr<RtpPacketToSend>> GeneratePadding(
DataSize size) = 0;
// Expected max pacer delay. If ExpectedQueueTime() is higher than
// this value, the packet producers should wait (eg drop frames rather than
// encoding them). Bitrate sent may temporarily exceed target set by
// UpdateBitrate() so that this limit will be upheld.
static const TimeDelta kMaxExpectedQueueLength;
// Pacing-rate relative to our target send rate.
// Multiplicative factor that is applied to the target bitrate to calculate
// the number of bytes that can be transmitted per interval.
// Increasing this factor will result in lower delays in cases of bitrate
// overshoots from the encoder.
static const float kDefaultPaceMultiplier;
// If no media or paused, wake up at least every |kPausedProcessIntervalMs| in
// order to send a keep-alive packet so we don't get stuck in a bad state due
// to lack of feedback.
static const TimeDelta kPausedProcessInterval;
static const TimeDelta kMinSleepTime;
PacingController(Clock* clock,
PacketSender* packet_sender,
RtcEventLog* event_log,
const WebRtcKeyValueConfig* field_trials,
ProcessMode mode);
// Adds the packet to the queue and calls PacketRouter::SendPacket() when
// it's time to send.
void EnqueuePacket(std::unique_ptr<RtpPacketToSend> packet);
void CreateProbeCluster(DataRate bitrate, int cluster_id);
void Pause(); // Temporarily pause all sending.
void Resume(); // Resume sending packets.
bool IsPaused() const;
void SetCongestionWindow(DataSize congestion_window_size);
void UpdateOutstandingData(DataSize outstanding_data);
// Sets the pacing rates. Must be called once before packets can be sent.
void SetPacingRates(DataRate pacing_rate, DataRate padding_rate);
// Currently audio traffic is not accounted by pacer and passed through.
// With the introduction of audio BWE audio traffic will be accounted for
// the pacer budget calculation. The audio traffic still will be injected
// at high priority.
void SetAccountForAudioPackets(bool account_for_audio);
void SetIncludeOverhead();
void SetTransportOverhead(DataSize overhead_per_packet);
// Returns the time since the oldest queued packet was enqueued.
TimeDelta OldestPacketWaitTime() const;
// Number of packets in the pacer queue.
size_t QueueSizePackets() const;
// Totals size of packets in the pacer queue.
DataSize QueueSizeData() const;
// Current buffer level, i.e. max of media and padding debt.
DataSize CurrentBufferLevel() const;
// Returns the time when the first packet was sent;
absl::optional<Timestamp> FirstSentPacketTime() const;
// Returns the number of milliseconds it will take to send the current
// packets in the queue, given the current size and bitrate, ignoring prio.
TimeDelta ExpectedQueueTime() const;
void SetQueueTimeLimit(TimeDelta limit);
// Enable bitrate probing. Enabled by default, mostly here to simplify
// testing. Must be called before any packets are being sent to have an
// effect.
void SetProbingEnabled(bool enabled);
// Returns the next time we expect ProcessPackets() to be called.
Timestamp NextSendTime() const;
// Check queue of pending packets and send them or padding packets, if budget
// is available.
void ProcessPackets();
bool Congested() const;
bool IsProbing() const;
void EnqueuePacketInternal(std::unique_ptr<RtpPacketToSend> packet,
int priority);
TimeDelta UpdateTimeAndGetElapsed(Timestamp now);
bool ShouldSendKeepalive(Timestamp now) const;
// Updates the number of bytes that can be sent for the next time interval.
void UpdateBudgetWithElapsedTime(TimeDelta delta);
void UpdateBudgetWithSentData(DataSize size);
DataSize PaddingToAdd(absl::optional<DataSize> recommended_probe_size,
DataSize data_sent) const;
std::unique_ptr<RtpPacketToSend> GetPendingPacket(
const PacedPacketInfo& pacing_info,
Timestamp target_send_time,
Timestamp now);
void OnPacketSent(RtpPacketMediaType packet_type,
DataSize packet_size,
Timestamp send_time);
void OnPaddingSent(DataSize padding_sent);
Timestamp CurrentTime() const;
const ProcessMode mode_;
Clock* const clock_;
PacketSender* const packet_sender_;
const std::unique_ptr<FieldTrialBasedConfig> fallback_field_trials_;
const WebRtcKeyValueConfig* field_trials_;
const bool drain_large_queues_;
const bool send_padding_if_silent_;
const bool pace_audio_;
const bool small_first_probe_packet_;
const bool ignore_transport_overhead_;
// In dynamic mode, indicates the target size when requesting padding,
// expressed as a duration in order to adjust for varying padding rate.
const TimeDelta padding_target_duration_;
TimeDelta min_packet_limit_;
DataSize transport_overhead_per_packet_;
// TODO(webrtc:9716): Remove this when we are certain clocks are monotonic.
// The last millisecond timestamp returned by |clock_|.
mutable Timestamp last_timestamp_;
bool paused_;
// If |use_interval_budget_| is true, |media_budget_| and |padding_budget_|
// will be used to track when packets can be sent. Otherwise the media and
// padding debt counters will be used together with the target rates.
// This is the media budget, keeping track of how many bits of media
// we can pace out during the current interval.
IntervalBudget media_budget_;
// This is the padding budget, keeping track of how many bits of padding we're
// allowed to send out during the current interval. This budget will be
// utilized when there's no media to send.
IntervalBudget padding_budget_;
DataSize media_debt_;
DataSize padding_debt_;
DataRate media_rate_;
DataRate padding_rate_;
BitrateProber prober_;
bool probing_send_failure_;
DataRate pacing_bitrate_;
Timestamp last_process_time_;
Timestamp last_send_time_;
absl::optional<Timestamp> first_sent_packet_time_;
RoundRobinPacketQueue packet_queue_;
uint64_t packet_counter_;
DataSize congestion_window_size_;
DataSize outstanding_data_;
TimeDelta queue_time_limit;
bool account_for_audio_;
bool include_overhead_;
} // namespace webrtc