Adding upstream version 115.7.0esr.upstream/115.7.0esrupstream

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

1 files changed, 134 insertions, 0 deletions

diff --git a/third_party/libwebrtc/call/simulated_network.h b/third_party/libwebrtc/call/simulated_network.h
new file mode 100644
index 0000000000..8597367add
--- /dev/null
+++ b/third_party/libwebrtc/call/simulated_network.h
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+/*
+ *  Copyright 2018 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 CALL_SIMULATED_NETWORK_H_
+#define CALL_SIMULATED_NETWORK_H_
+
+#include <stdint.h>
+
+#include <deque>
+#include <queue>
+#include <vector>
+
+#include "absl/types/optional.h"
+#include "api/sequence_checker.h"
+#include "api/test/simulated_network.h"
+#include "api/units/data_size.h"
+#include "api/units/timestamp.h"
+#include "rtc_base/race_checker.h"
+#include "rtc_base/random.h"
+#include "rtc_base/synchronization/mutex.h"
+#include "rtc_base/thread_annotations.h"
+
+namespace webrtc {
+
+// Class simulating a network link.
+//
+// This is a basic implementation of NetworkBehaviorInterface that supports:
+// - Packet loss
+// - Capacity delay
+// - Extra delay with or without packets reorder
+// - Packet overhead
+// - Queue max capacity
+class SimulatedNetwork : public SimulatedNetworkInterface {
+ public:
+  using Config = BuiltInNetworkBehaviorConfig;
+  explicit SimulatedNetwork(Config config, uint64_t random_seed = 1);
+  ~SimulatedNetwork() override;
+
+  // Sets a new configuration. This will affect packets that will be sent with
+  // EnqueuePacket but also packets in the network that have not left the
+  // network emulation. Packets that are ready to be retrieved by
+  // DequeueDeliverablePackets are not affected by the new configuration.
+  // TODO(bugs.webrtc.org/14525): Fix SetConfig and make it apply only to the
+  // part of the packet that is currently being sent (instead of applying to
+  // all of it).
+  void SetConfig(const Config& config) override;
+  void UpdateConfig(std::function<void(BuiltInNetworkBehaviorConfig*)>
+                        config_modifier) override;
+  void PauseTransmissionUntil(int64_t until_us) override;
+
+  // NetworkBehaviorInterface
+  bool EnqueuePacket(PacketInFlightInfo packet) override;
+  std::vector<PacketDeliveryInfo> DequeueDeliverablePackets(
+      int64_t receive_time_us) override;
+
+  absl::optional<int64_t> NextDeliveryTimeUs() const override;
+
+ private:
+  struct PacketInfo {
+    PacketInFlightInfo packet;
+    // Time when the packet has left (or will leave) the network.
+    int64_t arrival_time_us;
+  };
+  // Contains current configuration state.
+  struct ConfigState {
+    // Static link configuration.
+    Config config;
+    // The probability to drop the packet if we are currently dropping a
+    // burst of packet
+    double prob_loss_bursting;
+    // The probability to drop a burst of packets.
+    double prob_start_bursting;
+    // Used for temporary delay spikes.
+    int64_t pause_transmission_until_us = 0;
+  };
+
+  // Moves packets from capacity- to delay link.
+  void UpdateCapacityQueue(ConfigState state, int64_t time_now_us)
+      RTC_RUN_ON(&process_checker_);
+  ConfigState GetConfigState() const;
+
+  mutable Mutex config_lock_;
+
+  // Guards the data structures involved in delay and loss processing, such as
+  // the packet queues.
+  rtc::RaceChecker process_checker_;
+  // Models the capacity of the network by rejecting packets if the queue is
+  // full and keeping them in the queue until they are ready to exit (according
+  // to the link capacity, which cannot be violated, e.g. a 1 kbps link will
+  // only be able to deliver 1000 bits per second).
+  //
+  // Invariant:
+  // The head of the `capacity_link_` has arrival_time_us correctly set to the
+  // time when the packet is supposed to be delivered (without accounting
+  // potential packet loss or potential extra delay and without accounting for a
+  // new configuration of the network, which requires a re-computation of the
+  // arrival_time_us).
+  std::queue<PacketInfo> capacity_link_ RTC_GUARDED_BY(process_checker_);
+  // Models the extra delay of the network (see `queue_delay_ms`
+  // and `delay_standard_deviation_ms` in BuiltInNetworkBehaviorConfig), packets
+  // in the `delay_link_` have technically already left the network and don't
+  // use its capacity but they are not delivered yet.
+  std::deque<PacketInfo> delay_link_ RTC_GUARDED_BY(process_checker_);
+  // Represents the next moment in time when the network is supposed to deliver
+  // packets to the client (either by pulling them from `delay_link_` or
+  // `capacity_link_` or both).
+  absl::optional<int64_t> next_process_time_us_
+      RTC_GUARDED_BY(process_checker_);
+
+  ConfigState config_state_ RTC_GUARDED_BY(config_lock_);
+
+  Random random_ RTC_GUARDED_BY(process_checker_);
+  // Are we currently dropping a burst of packets?
+  bool bursting_;
+
+  // The send time of the last enqueued packet, this is only used to check that
+  // the send time of enqueued packets is monotonically increasing.
+  int64_t last_enqueue_time_us_;
+
+  // The last time a packet left the capacity_link_ (used to enforce
+  // the capacity of the link and avoid packets starts to get sent before
+  // the link it free).
+  int64_t last_capacity_link_exit_time_;
+};
+
+}  // namespace webrtc
+
+#endif  // CALL_SIMULATED_NETWORK_H_