| File: | root/firefox-clang/third_party/libwebrtc/modules/pacing/pacing_controller.h |
| Warning: | line 43, column 7 Excessive padding in 'class webrtc::PacingController' (32 padding bytes, where 0 is optimal). Optimal fields order: clock_, packet_sender_, field_trials_, max_rate, transport_overhead_per_packet_, send_burst_interval_, last_timestamp_, media_debt_, padding_debt_, pacing_rate_, adjusted_media_rate_, padding_rate_, last_process_time_, last_send_time_, queue_time_limit_, first_sent_packet_time_, prober_, packet_queue_, circuit_breaker_threshold_, drain_large_queues_, send_padding_if_silent_, pace_audio_, ignore_transport_overhead_, fast_retransmissions_, keyframe_flushing_, paused_, probing_send_failure_, seen_first_packet_, congested_, account_for_audio_, include_overhead_, consider reordering the fields or adding explicit padding members |
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| 1 | /* |
| 2 | * Copyright (c) 2019 The WebRTC project authors. All Rights Reserved. |
| 3 | * |
| 4 | * Use of this source code is governed by a BSD-style license |
| 5 | * that can be found in the LICENSE file in the root of the source |
| 6 | * tree. An additional intellectual property rights grant can be found |
| 7 | * in the file PATENTS. All contributing project authors may |
| 8 | * be found in the AUTHORS file in the root of the source tree. |
| 9 | */ |
| 10 | |
| 11 | #ifndef MODULES_PACING_PACING_CONTROLLER_H_ |
| 12 | #define MODULES_PACING_PACING_CONTROLLER_H_ |
| 13 | |
| 14 | #include <stddef.h> |
| 15 | #include <stdint.h> |
| 16 | |
| 17 | #include <array> |
| 18 | #include <memory> |
| 19 | #include <optional> |
| 20 | #include <vector> |
| 21 | |
| 22 | #include "api/array_view.h" |
| 23 | #include "api/field_trials_view.h" |
| 24 | #include "api/rtp_packet_sender.h" |
| 25 | #include "api/transport/network_types.h" |
| 26 | #include "api/units/data_rate.h" |
| 27 | #include "api/units/data_size.h" |
| 28 | #include "api/units/time_delta.h" |
| 29 | #include "api/units/timestamp.h" |
| 30 | #include "modules/pacing/bitrate_prober.h" |
| 31 | #include "modules/pacing/prioritized_packet_queue.h" |
| 32 | #include "modules/rtp_rtcp/include/rtp_rtcp_defines.h" |
| 33 | #include "modules/rtp_rtcp/source/rtp_packet_to_send.h" |
| 34 | #include "system_wrappers/include/clock.h" |
| 35 | |
| 36 | namespace webrtc { |
| 37 | |
| 38 | // This class implements a leaky-bucket packet pacing algorithm. It handles the |
| 39 | // logic of determining which packets to send when, but the actual timing of |
| 40 | // the processing is done externally (e.g. RtpPacketPacer). Furthermore, the |
| 41 | // forwarding of packets when they are ready to be sent is also handled |
| 42 | // externally, via the PacingController::PacketSender interface. |
| 43 | class PacingController { |
Excessive padding in 'class webrtc::PacingController' (32 padding bytes, where 0 is optimal). Optimal fields order: clock_, packet_sender_, field_trials_, max_rate, transport_overhead_per_packet_, send_burst_interval_, last_timestamp_, media_debt_, padding_debt_, pacing_rate_, adjusted_media_rate_, padding_rate_, last_process_time_, last_send_time_, queue_time_limit_, first_sent_packet_time_, prober_, packet_queue_, circuit_breaker_threshold_, drain_large_queues_, send_padding_if_silent_, pace_audio_, ignore_transport_overhead_, fast_retransmissions_, keyframe_flushing_, paused_, probing_send_failure_, seen_first_packet_, congested_, account_for_audio_, include_overhead_, consider reordering the fields or adding explicit padding members | |
| 44 | public: |
| 45 | class PacketSender { |
| 46 | public: |
| 47 | virtual ~PacketSender() = default; |
| 48 | virtual void SendPacket(std::unique_ptr<RtpPacketToSend> packet, |
| 49 | const PacedPacketInfo& cluster_info) = 0; |
| 50 | // Should be called after each call to SendPacket(). |
| 51 | virtual std::vector<std::unique_ptr<RtpPacketToSend>> FetchFec() = 0; |
| 52 | virtual std::vector<std::unique_ptr<RtpPacketToSend>> GeneratePadding( |
| 53 | DataSize size) = 0; |
| 54 | // TODO(bugs.webrtc.org/1439830): Make pure virtual once subclasses adapt. |
| 55 | virtual void OnBatchComplete() {} |
| 56 | |
| 57 | // TODO(bugs.webrtc.org/11340): Make pure virtual once downstream projects |
| 58 | // have been updated. |
| 59 | virtual void OnAbortedRetransmissions( |
| 60 | uint32_t /* ssrc */, |
| 61 | rtc::ArrayView<const uint16_t> /* sequence_numbers */) {} |
| 62 | virtual std::optional<uint32_t> GetRtxSsrcForMedia( |
| 63 | uint32_t /* ssrc */) const { |
| 64 | return std::nullopt; |
| 65 | } |
| 66 | }; |
| 67 | |
| 68 | // If no media or paused, wake up at least every `kPausedProcessIntervalMs` in |
| 69 | // order to send a keep-alive packet so we don't get stuck in a bad state due |
| 70 | // to lack of feedback. |
| 71 | static const TimeDelta kPausedProcessInterval; |
| 72 | // The default minimum time that should elapse calls to `ProcessPackets()`. |
| 73 | static const TimeDelta kMinSleepTime; |
| 74 | // When padding should be generated, add packets to the buffer with a size |
| 75 | // corresponding to this duration times the current padding rate. |
| 76 | static const TimeDelta kTargetPaddingDuration; |
| 77 | // The maximum time that the pacer can use when "replaying" passed time where |
| 78 | // padding should have been generated. |
| 79 | static const TimeDelta kMaxPaddingReplayDuration; |
| 80 | // Allow probes to be processed slightly ahead of inteded send time. Currently |
| 81 | // set to 1ms as this is intended to allow times be rounded down to the |
| 82 | // nearest millisecond. |
| 83 | static const TimeDelta kMaxEarlyProbeProcessing; |
| 84 | // Max total size of packets expected to be sent in a burst in order to not |
| 85 | // risk loosing packets due to too small send socket buffers. It upper limits |
| 86 | // the send burst interval. |
| 87 | // Ex: max send burst interval = 63Kb / 10Mbit/s = 50ms. |
| 88 | static constexpr DataSize kMaxBurstSize = DataSize::Bytes(63 * 1000); |
| 89 | |
| 90 | // Configuration default values. |
| 91 | static constexpr TimeDelta kDefaultBurstInterval = TimeDelta::Millis(40); |
| 92 | static constexpr TimeDelta kMaxExpectedQueueLength = TimeDelta::Millis(2000); |
| 93 | |
| 94 | struct Configuration { |
| 95 | // If the pacer queue grows longer than the configured max queue limit, |
| 96 | // pacer sends at the minimum rate needed to keep the max queue limit and |
| 97 | // ignore the current bandwidth estimate. |
| 98 | bool drain_large_queues = true; |
| 99 | // Expected max pacer delay. If ExpectedQueueTime() is higher than |
| 100 | // this value, the packet producers should wait (eg drop frames rather than |
| 101 | // encoding them). Bitrate sent may temporarily exceed target set by |
| 102 | // SetPacingRates() so that this limit will be upheld if |
| 103 | // `drain_large_queues` is set. |
| 104 | TimeDelta queue_time_limit = kMaxExpectedQueueLength; |
| 105 | // If the first packet of a keyframe is enqueued on a RTP stream, pacer |
| 106 | // skips forward to that packet and drops other enqueued packets on that |
| 107 | // stream, unless a keyframe is already being paced. |
| 108 | bool keyframe_flushing = false; |
| 109 | // Audio retransmission is prioritized before video retransmission packets. |
| 110 | bool prioritize_audio_retransmission = false; |
| 111 | // Configure separate timeouts per priority. After a timeout, a packet of |
| 112 | // that sort will not be paced and instead dropped. |
| 113 | // Note: to set TTL on audio retransmission, |
| 114 | // `prioritize_audio_retransmission` must be true. |
| 115 | PacketQueueTTL packet_queue_ttl; |
| 116 | // The pacer is allowed to send enqueued packets in bursts and can build up |
| 117 | // a packet "debt" that correspond to approximately the send rate during the |
| 118 | // burst interval. |
| 119 | TimeDelta send_burst_interval = kDefaultBurstInterval; |
| 120 | }; |
| 121 | |
| 122 | static Configuration DefaultConfiguration() { return Configuration{}; } |
| 123 | |
| 124 | PacingController(Clock* clock, |
| 125 | PacketSender* packet_sender, |
| 126 | const FieldTrialsView& field_trials, |
| 127 | Configuration configuration = DefaultConfiguration()); |
| 128 | |
| 129 | ~PacingController(); |
| 130 | |
| 131 | // Adds the packet to the queue and calls PacketRouter::SendPacket() when |
| 132 | // it's time to send. |
| 133 | void EnqueuePacket(std::unique_ptr<RtpPacketToSend> packet); |
| 134 | |
| 135 | void CreateProbeClusters( |
| 136 | rtc::ArrayView<const ProbeClusterConfig> probe_cluster_configs); |
| 137 | |
| 138 | void Pause(); // Temporarily pause all sending. |
| 139 | void Resume(); // Resume sending packets. |
| 140 | bool IsPaused() const; |
| 141 | |
| 142 | void SetCongested(bool congested); |
| 143 | |
| 144 | // Sets the pacing rates. Must be called once before packets can be sent. |
| 145 | void SetPacingRates(DataRate pacing_rate, DataRate padding_rate); |
| 146 | DataRate pacing_rate() const { return adjusted_media_rate_; } |
| 147 | |
| 148 | // Currently audio traffic is not accounted by pacer and passed through. |
| 149 | // With the introduction of audio BWE audio traffic will be accounted for |
| 150 | // the pacer budget calculation. The audio traffic still will be injected |
| 151 | // at high priority. |
| 152 | void SetAccountForAudioPackets(bool account_for_audio); |
| 153 | void SetIncludeOverhead(); |
| 154 | |
| 155 | void SetTransportOverhead(DataSize overhead_per_packet); |
| 156 | // The pacer is allowed to send enqued packets in bursts and can build up a |
| 157 | // packet "debt" that correspond to approximately the send rate during |
| 158 | // 'burst_interval'. |
| 159 | void SetSendBurstInterval(TimeDelta burst_interval); |
| 160 | |
| 161 | // A probe may be sent without first waing for a media packet. |
| 162 | void SetAllowProbeWithoutMediaPacket(bool allow); |
| 163 | |
| 164 | // Returns the time when the oldest packet was queued. |
| 165 | Timestamp OldestPacketEnqueueTime() const; |
| 166 | |
| 167 | // Number of packets in the pacer queue. |
| 168 | size_t QueueSizePackets() const; |
| 169 | // Number of packets in the pacer queue per media type (RtpPacketMediaType |
| 170 | // values are used as lookup index). |
| 171 | const std::array<int, kNumMediaTypes>& SizeInPacketsPerRtpPacketMediaType() |
| 172 | const; |
| 173 | // Totals size of packets in the pacer queue. |
| 174 | DataSize QueueSizeData() const; |
| 175 | |
| 176 | // Current buffer level, i.e. max of media and padding debt. |
| 177 | DataSize CurrentBufferLevel() const; |
| 178 | |
| 179 | // Returns the time when the first packet was sent. |
| 180 | std::optional<Timestamp> FirstSentPacketTime() const; |
| 181 | |
| 182 | // Returns the number of milliseconds it will take to send the current |
| 183 | // packets in the queue, given the current size and bitrate, ignoring prio. |
| 184 | TimeDelta ExpectedQueueTime() const; |
| 185 | |
| 186 | void SetQueueTimeLimit(TimeDelta limit); |
| 187 | |
| 188 | // Enable bitrate probing. Enabled by default, mostly here to simplify |
| 189 | // testing. Must be called before any packets are being sent to have an |
| 190 | // effect. |
| 191 | void SetProbingEnabled(bool enabled); |
| 192 | |
| 193 | // Returns the next time we expect ProcessPackets() to be called. |
| 194 | Timestamp NextSendTime() const; |
| 195 | |
| 196 | // Check queue of pending packets and send them or padding packets, if budget |
| 197 | // is available. |
| 198 | void ProcessPackets(); |
| 199 | |
| 200 | bool IsProbing() const; |
| 201 | |
| 202 | // Note: Intended for debugging purposes only, will be removed. |
| 203 | // Sets the number of iterations of the main loop in `ProcessPackets()` that |
| 204 | // is considered erroneous to exceed. |
| 205 | void SetCircuitBreakerThreshold(int num_iterations); |
| 206 | |
| 207 | // Remove any pending packets matching this SSRC from the packet queue. |
| 208 | void RemovePacketsForSsrc(uint32_t ssrc); |
| 209 | |
| 210 | private: |
| 211 | TimeDelta UpdateTimeAndGetElapsed(Timestamp now); |
| 212 | bool ShouldSendKeepalive(Timestamp now) const; |
| 213 | |
| 214 | // Updates the number of bytes that can be sent for the next time interval. |
| 215 | void UpdateBudgetWithElapsedTime(TimeDelta delta); |
| 216 | void UpdateBudgetWithSentData(DataSize size); |
| 217 | void UpdatePaddingBudgetWithSentData(DataSize size); |
| 218 | |
| 219 | DataSize PaddingToAdd(DataSize recommended_probe_size, |
| 220 | DataSize data_sent) const; |
| 221 | |
| 222 | std::unique_ptr<RtpPacketToSend> GetPendingPacket( |
| 223 | const PacedPacketInfo& pacing_info, |
| 224 | Timestamp target_send_time, |
| 225 | Timestamp now); |
| 226 | void OnPacketSent(RtpPacketMediaType packet_type, |
| 227 | DataSize packet_size, |
| 228 | Timestamp send_time); |
| 229 | void MaybeUpdateMediaRateDueToLongQueue(Timestamp now); |
| 230 | |
| 231 | Timestamp CurrentTime() const; |
| 232 | |
| 233 | // Helper methods for packet that may not be paced. Returns a finite Timestamp |
| 234 | // if a packet type is configured to not be paced and the packet queue has at |
| 235 | // least one packet of that type. Otherwise returns |
| 236 | // Timestamp::MinusInfinity(). |
| 237 | Timestamp NextUnpacedSendTime() const; |
| 238 | |
| 239 | Clock* const clock_; |
| 240 | PacketSender* const packet_sender_; |
| 241 | const FieldTrialsView& field_trials_; |
| 242 | |
| 243 | const bool drain_large_queues_; |
| 244 | const bool send_padding_if_silent_; |
| 245 | const bool pace_audio_; |
| 246 | const bool ignore_transport_overhead_; |
| 247 | const bool fast_retransmissions_; |
| 248 | const bool keyframe_flushing_; |
| 249 | DataRate max_rate = DataRate::BitsPerSec(100'000'000); |
| 250 | DataSize transport_overhead_per_packet_; |
| 251 | TimeDelta send_burst_interval_; |
| 252 | |
| 253 | // TODO(webrtc:9716): Remove this when we are certain clocks are monotonic. |
| 254 | // The last millisecond timestamp returned by `clock_`. |
| 255 | mutable Timestamp last_timestamp_; |
| 256 | bool paused_; |
| 257 | |
| 258 | // Amount of outstanding data for media and padding. |
| 259 | DataSize media_debt_; |
| 260 | DataSize padding_debt_; |
| 261 | |
| 262 | // The target pacing rate, signaled via SetPacingRates(). |
| 263 | DataRate pacing_rate_; |
| 264 | // The media send rate, which might adjusted from pacing_rate_, e.g. if the |
| 265 | // pacing queue is growing too long. |
| 266 | DataRate adjusted_media_rate_; |
| 267 | // The padding target rate. We aim to fill up to this rate with padding what |
| 268 | // is not already used by media. |
| 269 | DataRate padding_rate_; |
| 270 | |
| 271 | BitrateProber prober_; |
| 272 | bool probing_send_failure_; |
| 273 | |
| 274 | Timestamp last_process_time_; |
| 275 | Timestamp last_send_time_; |
| 276 | std::optional<Timestamp> first_sent_packet_time_; |
| 277 | bool seen_first_packet_; |
| 278 | |
| 279 | PrioritizedPacketQueue packet_queue_; |
| 280 | |
| 281 | bool congested_; |
| 282 | |
| 283 | TimeDelta queue_time_limit_; |
| 284 | bool account_for_audio_; |
| 285 | bool include_overhead_; |
| 286 | |
| 287 | int circuit_breaker_threshold_; |
| 288 | }; |
| 289 | } // namespace webrtc |
| 290 | |
| 291 | #endif // MODULES_PACING_PACING_CONTROLLER_H_ |