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238 lines
8.6 KiB
C++
238 lines
8.6 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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#ifndef MODULES_PACING_PACING_CONTROLLER_H_
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#define MODULES_PACING_PACING_CONTROLLER_H_
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#include <stddef.h>
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#include <stdint.h>
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#include <array>
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#include <atomic>
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#include <memory>
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#include <vector>
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#include "absl/types/optional.h"
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#include "api/field_trials_view.h"
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#include "api/function_view.h"
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#include "api/transport/field_trial_based_config.h"
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#include "api/transport/network_types.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/pacing/prioritized_packet_queue.h"
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#include "modules/pacing/rtp_packet_pacer.h"
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#include "modules/rtp_rtcp/include/rtp_packet_sender.h"
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#include "modules/rtp_rtcp/include/rtp_rtcp_defines.h"
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#include "modules/rtp_rtcp/source/rtp_packet_to_send.h"
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#include "rtc_base/experiments/field_trial_parser.h"
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#include "rtc_base/thread_annotations.h"
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namespace webrtc {
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// This class implements a leaky-bucket packet pacing algorithm. It handles the
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// logic of determining which packets to send when, but the actual timing of
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// the processing is done externally (e.g. RtpPacketPacer). Furthermore, the
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// forwarding of packets when they are ready to be sent is also handled
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// externally, via the PacingController::PacketSender interface.
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class PacingController {
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public:
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class PacketSender {
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public:
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virtual ~PacketSender() = default;
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virtual void SendPacket(std::unique_ptr<RtpPacketToSend> packet,
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const PacedPacketInfo& cluster_info) = 0;
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// Should be called after each call to SendPacket().
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virtual std::vector<std::unique_ptr<RtpPacketToSend>> FetchFec() = 0;
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virtual std::vector<std::unique_ptr<RtpPacketToSend>> GeneratePadding(
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DataSize size) = 0;
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// TODO(bugs.webrtc.org/11340): Make pure virtual once downstream projects
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// have been updated.
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virtual void OnAbortedRetransmissions(
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uint32_t ssrc,
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rtc::ArrayView<const uint16_t> sequence_numbers) {}
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virtual absl::optional<uint32_t> GetRtxSsrcForMedia(uint32_t ssrc) const {
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return absl::nullopt;
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}
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};
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// Expected max pacer delay. If ExpectedQueueTime() is higher than
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// this value, the packet producers should wait (eg drop frames rather than
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// encoding them). Bitrate sent may temporarily exceed target set by
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// UpdateBitrate() so that this limit will be upheld.
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static const TimeDelta kMaxExpectedQueueLength;
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// If no media or paused, wake up at least every `kPausedProcessIntervalMs` in
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// order to send a keep-alive packet so we don't get stuck in a bad state due
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// to lack of feedback.
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static const TimeDelta kPausedProcessInterval;
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static const TimeDelta kMinSleepTime;
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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
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// nearest millisecond.
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static const TimeDelta kMaxEarlyProbeProcessing;
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PacingController(Clock* clock,
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PacketSender* packet_sender,
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const FieldTrialsView& field_trials);
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~PacingController();
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// Adds the packet to the queue and calls PacketRouter::SendPacket() when
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// it's time to send.
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void EnqueuePacket(std::unique_ptr<RtpPacketToSend> packet);
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// ABSL_DEPRECATED("Use CreateProbeClusters instead")
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void CreateProbeCluster(DataRate bitrate, int cluster_id);
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void CreateProbeClusters(
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rtc::ArrayView<const ProbeClusterConfig> probe_cluster_configs);
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void Pause(); // Temporarily pause all sending.
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void Resume(); // Resume sending packets.
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bool IsPaused() const;
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void SetCongested(bool congested);
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// Sets the pacing rates. Must be called once before packets can be sent.
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void SetPacingRates(DataRate pacing_rate, DataRate padding_rate);
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DataRate pacing_rate() const { return adjusted_media_rate_; }
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// Currently audio traffic is not accounted by pacer and passed through.
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// With the introduction of audio BWE audio traffic will be accounted for
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// the pacer budget calculation. The audio traffic still will be injected
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// at high priority.
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void SetAccountForAudioPackets(bool account_for_audio);
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void SetIncludeOverhead();
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void SetTransportOverhead(DataSize overhead_per_packet);
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// The pacer is allowed to send enqued packets in bursts and can build up a
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// packet "debt" that correspond to approximately the send rate during
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// 'burst_interval'.
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void SetSendBurstInterval(TimeDelta burst_interval);
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// Returns the time when the oldest packet was queued.
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Timestamp OldestPacketEnqueueTime() const;
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// Number of packets in the pacer queue.
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size_t QueueSizePackets() const;
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// Number of packets in the pacer queue per media type (RtpPacketMediaType
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// values are used as lookup index).
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const std::array<int, kNumMediaTypes>& SizeInPacketsPerRtpPacketMediaType()
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const;
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// Totals size of packets in the pacer queue.
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DataSize QueueSizeData() const;
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// Current buffer level, i.e. max of media and padding debt.
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DataSize CurrentBufferLevel() const;
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// Returns the time when the first packet was sent.
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absl::optional<Timestamp> FirstSentPacketTime() const;
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// Returns the number of milliseconds it will take to send the current
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// packets in the queue, given the current size and bitrate, ignoring prio.
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TimeDelta ExpectedQueueTime() const;
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void SetQueueTimeLimit(TimeDelta limit);
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// Enable bitrate probing. Enabled by default, mostly here to simplify
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// testing. Must be called before any packets are being sent to have an
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// effect.
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void SetProbingEnabled(bool enabled);
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// Returns the next time we expect ProcessPackets() to be called.
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Timestamp NextSendTime() const;
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// Check queue of pending packets and send them or padding packets, if budget
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// is available.
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void ProcessPackets();
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bool IsProbing() const;
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private:
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TimeDelta UpdateTimeAndGetElapsed(Timestamp now);
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bool ShouldSendKeepalive(Timestamp now) const;
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// Updates the number of bytes that can be sent for the next time interval.
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void UpdateBudgetWithElapsedTime(TimeDelta delta);
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void UpdateBudgetWithSentData(DataSize size);
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void UpdatePaddingBudgetWithSentData(DataSize size);
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DataSize PaddingToAdd(DataSize recommended_probe_size,
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DataSize data_sent) const;
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std::unique_ptr<RtpPacketToSend> GetPendingPacket(
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const PacedPacketInfo& pacing_info,
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Timestamp target_send_time,
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Timestamp now);
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void OnPacketSent(RtpPacketMediaType packet_type,
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DataSize packet_size,
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Timestamp send_time);
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void MaybeUpdateMediaRateDueToLongQueue(Timestamp now);
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Timestamp CurrentTime() const;
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// Helper methods for packet that may not be paced. Returns a finite Timestamp
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// if a packet type is configured to not be paced and the packet queue has at
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// least one packet of that type. Otherwise returns
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// Timestamp::MinusInfinity().
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Timestamp NextUnpacedSendTime() const;
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Clock* const clock_;
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PacketSender* const packet_sender_;
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const FieldTrialsView& field_trials_;
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const bool drain_large_queues_;
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const bool send_padding_if_silent_;
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const bool pace_audio_;
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const bool ignore_transport_overhead_;
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const bool fast_retransmissions_;
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TimeDelta min_packet_limit_;
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DataSize transport_overhead_per_packet_;
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TimeDelta send_burst_interval_;
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// TODO(webrtc:9716): Remove this when we are certain clocks are monotonic.
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// The last millisecond timestamp returned by `clock_`.
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mutable Timestamp last_timestamp_;
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bool paused_;
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// Amount of outstanding data for media and padding.
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DataSize media_debt_;
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DataSize padding_debt_;
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// The target pacing rate, signaled via SetPacingRates().
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DataRate pacing_rate_;
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// The media send rate, which might adjusted from pacing_rate_, e.g. if the
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// pacing queue is growing too long.
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DataRate adjusted_media_rate_;
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// The padding target rate. We aim to fill up to this rate with padding what
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// is not already used by media.
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DataRate padding_rate_;
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BitrateProber prober_;
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bool probing_send_failure_;
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Timestamp last_process_time_;
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Timestamp last_send_time_;
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absl::optional<Timestamp> first_sent_packet_time_;
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bool seen_first_packet_;
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PrioritizedPacketQueue packet_queue_;
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bool congested_;
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TimeDelta queue_time_limit_;
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bool account_for_audio_;
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bool include_overhead_;
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};
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} // namespace webrtc
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#endif // MODULES_PACING_PACING_CONTROLLER_H_
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