Adding upstream version 124.0.1.upstream/124.0.1

Signed-off-by: Daniel Baumann <daniel.baumann@progress-linux.org>

1 files changed, 259 insertions, 0 deletions

diff --git a/third_party/libwebrtc/modules/pacing/pacing_controller.h b/third_party/libwebrtc/modules/pacing/pacing_controller.h
new file mode 100644
index 0000000000..dd5636ccef
--- /dev/null
+++ b/third_party/libwebrtc/modules/pacing/pacing_controller.h
@@ -0,0 +1,259 @@
+/*
+ *  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.
+ */
+
+#ifndef MODULES_PACING_PACING_CONTROLLER_H_
+#define MODULES_PACING_PACING_CONTROLLER_H_
+
+#include <stddef.h>
+#include <stdint.h>
+
+#include <array>
+#include <atomic>
+#include <memory>
+#include <vector>
+
+#include "absl/types/optional.h"
+#include "api/field_trials_view.h"
+#include "api/function_view.h"
+#include "api/transport/field_trial_based_config.h"
+#include "api/transport/network_types.h"
+#include "api/units/data_size.h"
+#include "modules/pacing/bitrate_prober.h"
+#include "modules/pacing/interval_budget.h"
+#include "modules/pacing/prioritized_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/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. RtpPacketPacer). Furthermore, the
+// forwarding of packets when they are ready to be sent is also handled
+// externally, via the PacingController::PacketSender interface.
+class PacingController {
+ public:
+  class PacketSender {
+   public:
+    virtual ~PacketSender() = default;
+    virtual void SendPacket(std::unique_ptr<RtpPacketToSend> packet,
+                            const PacedPacketInfo& cluster_info) = 0;
+    // Should be called after each call to SendPacket().
+    virtual std::vector<std::unique_ptr<RtpPacketToSend>> FetchFec() = 0;
+    virtual std::vector<std::unique_ptr<RtpPacketToSend>> GeneratePadding(
+        DataSize size) = 0;
+    // TODO(bugs.webrtc.org/1439830): Make pure virtual once subclasses adapt.
+    virtual void OnBatchComplete() {}
+
+    // TODO(bugs.webrtc.org/11340): Make pure virtual once downstream projects
+    // have been updated.
+    virtual void OnAbortedRetransmissions(
+        uint32_t ssrc,
+        rtc::ArrayView<const uint16_t> sequence_numbers) {}
+    virtual absl::optional<uint32_t> GetRtxSsrcForMedia(uint32_t ssrc) const {
+      return absl::nullopt;
+    }
+  };
+
+  // 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;
+  // 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;
+  // The default minimum time that should elapse calls to `ProcessPackets()`.
+  static const TimeDelta kMinSleepTime;
+  // When padding should be generated, add packets to the buffer with a size
+  // corresponding to this duration times the current padding rate.
+  static const TimeDelta kTargetPaddingDuration;
+  // The maximum time that the pacer can use when "replaying" passed time where
+  // padding should have been generated.
+  static const TimeDelta kMaxPaddingReplayDuration;
+  // Allow probes to be processed slightly ahead of inteded send time. Currently
+  // set to 1ms as this is intended to allow times be rounded down to the
+  // nearest millisecond.
+  static const TimeDelta kMaxEarlyProbeProcessing;
+  // Max total size of packets expected to be sent in a burst in order to not
+  // risk loosing packets due to too small send socket buffers. It upper limits
+  // the send burst interval.
+  // Ex: max send burst interval = 63Kb / 10Mbit/s = 50ms.
+  static constexpr DataSize kMaxBurstSize = DataSize::Bytes(63 * 1000);
+
+  PacingController(Clock* clock,
+                   PacketSender* packet_sender,
+                   const FieldTrialsView& field_trials);
+
+  ~PacingController();
+
+  // Adds the packet to the queue and calls PacketRouter::SendPacket() when
+  // it's time to send.
+  void EnqueuePacket(std::unique_ptr<RtpPacketToSend> packet);
+
+  void CreateProbeClusters(
+      rtc::ArrayView<const ProbeClusterConfig> probe_cluster_configs);
+
+  void Pause();   // Temporarily pause all sending.
+  void Resume();  // Resume sending packets.
+  bool IsPaused() const;
+
+  void SetCongested(bool congested);
+
+  // Sets the pacing rates. Must be called once before packets can be sent.
+  void SetPacingRates(DataRate pacing_rate, DataRate padding_rate);
+  DataRate pacing_rate() const { return adjusted_media_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);
+  // The pacer is allowed to send enqued packets in bursts and can build up a
+  // packet "debt" that correspond to approximately the send rate during
+  // 'burst_interval'.
+  void SetSendBurstInterval(TimeDelta burst_interval);
+
+  // Returns the time when the oldest packet was queued.
+  Timestamp OldestPacketEnqueueTime() const;
+
+  // Number of packets in the pacer queue.
+  size_t QueueSizePackets() const;
+  // Number of packets in the pacer queue per media type (RtpPacketMediaType
+  // values are used as lookup index).
+  const std::array<int, kNumMediaTypes>& SizeInPacketsPerRtpPacketMediaType()
+      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 IsProbing() const;
+
+  // Note: Intended for debugging purposes only, will be removed.
+  // Sets the number of iterations of the main loop in `ProcessPackets()` that
+  // is considered erroneous to exceed.
+  void SetCircuitBreakerThreshold(int num_iterations);
+
+  // Remove any pending packets matching this SSRC from the packet queue.
+  void RemovePacketsForSsrc(uint32_t ssrc);
+
+ private:
+  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);
+  void UpdatePaddingBudgetWithSentData(DataSize size);
+
+  DataSize PaddingToAdd(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 MaybeUpdateMediaRateDueToLongQueue(Timestamp now);
+
+  Timestamp CurrentTime() const;
+
+  // Helper methods for packet that may not be paced. Returns a finite Timestamp
+  // if a packet type is configured to not be paced and the packet queue has at
+  // least one packet of that type. Otherwise returns
+  // Timestamp::MinusInfinity().
+  Timestamp NextUnpacedSendTime() const;
+
+  Clock* const clock_;
+  PacketSender* const packet_sender_;
+  const FieldTrialsView& field_trials_;
+
+  const bool drain_large_queues_;
+  const bool send_padding_if_silent_;
+  const bool pace_audio_;
+  const bool ignore_transport_overhead_;
+  const bool fast_retransmissions_;
+  const bool keyframe_flushing_;
+  DataRate max_rate = DataRate::BitsPerSec(100'000'000);
+  DataSize transport_overhead_per_packet_;
+  TimeDelta send_burst_interval_;
+
+  // 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_;
+
+  // Amount of outstanding data for media and padding.
+  DataSize media_debt_;
+  DataSize padding_debt_;
+
+  // The target pacing rate, signaled via SetPacingRates().
+  DataRate pacing_rate_;
+  // The media send rate, which might adjusted from pacing_rate_, e.g. if the
+  // pacing queue is growing too long.
+  DataRate adjusted_media_rate_;
+  // The padding target rate. We aim to fill up to this rate with padding what
+  // is not already used by media.
+  DataRate padding_rate_;
+
+  BitrateProber prober_;
+  bool probing_send_failure_;
+
+  Timestamp last_process_time_;
+  Timestamp last_send_time_;
+  absl::optional<Timestamp> first_sent_packet_time_;
+  bool seen_first_packet_;
+
+  PrioritizedPacketQueue packet_queue_;
+
+  bool congested_;
+
+  TimeDelta queue_time_limit_;
+  bool account_for_audio_;
+  bool include_overhead_;
+
+  int circuit_breaker_threshold_;
+};
+}  // namespace webrtc
+
+#endif  // MODULES_PACING_PACING_CONTROLLER_H_