Repo created

TMessagesProj/jni/voip/webrtc/modules/congestion_controller/goog_cc/acknowledged_bitrate_estimator.cc

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+/*
+ *  Copyright (c) 2017 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 "modules/congestion_controller/goog_cc/acknowledged_bitrate_estimator.h"
+
+#include <algorithm>
+#include <memory>
+#include <utility>
+#include <vector>
+
+#include "absl/types/optional.h"
+#include "api/field_trials_view.h"
+#include "api/transport/network_types.h"
+#include "api/units/data_rate.h"
+#include "api/units/data_size.h"
+#include "api/units/timestamp.h"
+#include "modules/congestion_controller/goog_cc/bitrate_estimator.h"
+#include "rtc_base/checks.h"
+
+namespace webrtc {
+
+AcknowledgedBitrateEstimator::AcknowledgedBitrateEstimator(
+    const FieldTrialsView* key_value_config)
+    : AcknowledgedBitrateEstimator(
+          key_value_config,
+          std::make_unique<BitrateEstimator>(key_value_config)) {}
+
+AcknowledgedBitrateEstimator::~AcknowledgedBitrateEstimator() {}
+
+AcknowledgedBitrateEstimator::AcknowledgedBitrateEstimator(
+    const FieldTrialsView* key_value_config,
+    std::unique_ptr<BitrateEstimator> bitrate_estimator)
+    : in_alr_(false), bitrate_estimator_(std::move(bitrate_estimator)) {}
+
+void AcknowledgedBitrateEstimator::IncomingPacketFeedbackVector(
+    const std::vector<PacketResult>& packet_feedback_vector) {
+  RTC_DCHECK(std::is_sorted(packet_feedback_vector.begin(),
+                            packet_feedback_vector.end(),
+                            PacketResult::ReceiveTimeOrder()));
+  for (const auto& packet : packet_feedback_vector) {
+    if (alr_ended_time_ && packet.sent_packet.send_time > *alr_ended_time_) {
+      bitrate_estimator_->ExpectFastRateChange();
+      alr_ended_time_.reset();
+    }
+    DataSize acknowledged_estimate = packet.sent_packet.size;
+    acknowledged_estimate += packet.sent_packet.prior_unacked_data;
+    bitrate_estimator_->Update(packet.receive_time, acknowledged_estimate,
+                               in_alr_);
+  }
+}
+
+absl::optional<DataRate> AcknowledgedBitrateEstimator::bitrate() const {
+  return bitrate_estimator_->bitrate();
+}
+
+absl::optional<DataRate> AcknowledgedBitrateEstimator::PeekRate() const {
+  return bitrate_estimator_->PeekRate();
+}
+
+void AcknowledgedBitrateEstimator::SetAlrEndedTime(Timestamp alr_ended_time) {
+  alr_ended_time_.emplace(alr_ended_time);
+}
+
+void AcknowledgedBitrateEstimator::SetAlr(bool in_alr) {
+  in_alr_ = in_alr;
+}
+
+}  // namespace webrtc

TMessagesProj/jni/voip/webrtc/modules/congestion_controller/goog_cc/acknowledged_bitrate_estimator.h

@@ -0,0 +1,53 @@
+/*
+ *  Copyright (c) 2017 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_CONGESTION_CONTROLLER_GOOG_CC_ACKNOWLEDGED_BITRATE_ESTIMATOR_H_
+#define MODULES_CONGESTION_CONTROLLER_GOOG_CC_ACKNOWLEDGED_BITRATE_ESTIMATOR_H_
+
+#include <memory>
+#include <vector>
+
+#include "absl/types/optional.h"
+#include "api/field_trials_view.h"
+#include "api/transport/network_types.h"
+#include "api/units/data_rate.h"
+#include "api/units/timestamp.h"
+#include "modules/congestion_controller/goog_cc/acknowledged_bitrate_estimator_interface.h"
+#include "modules/congestion_controller/goog_cc/bitrate_estimator.h"
+
+namespace webrtc {
+
+class AcknowledgedBitrateEstimator
+    : public AcknowledgedBitrateEstimatorInterface {
+ public:
+  AcknowledgedBitrateEstimator(
+      const FieldTrialsView* key_value_config,
+      std::unique_ptr<BitrateEstimator> bitrate_estimator);
+
+  explicit AcknowledgedBitrateEstimator(
+      const FieldTrialsView* key_value_config);
+  ~AcknowledgedBitrateEstimator() override;
+
+  void IncomingPacketFeedbackVector(
+      const std::vector<PacketResult>& packet_feedback_vector) override;
+  absl::optional<DataRate> bitrate() const override;
+  absl::optional<DataRate> PeekRate() const override;
+  void SetAlr(bool in_alr) override;
+  void SetAlrEndedTime(Timestamp alr_ended_time) override;
+
+ private:
+  absl::optional<Timestamp> alr_ended_time_;
+  bool in_alr_;
+  std::unique_ptr<BitrateEstimator> bitrate_estimator_;
+};
+
+}  // namespace webrtc
+
+#endif  // MODULES_CONGESTION_CONTROLLER_GOOG_CC_ACKNOWLEDGED_BITRATE_ESTIMATOR_H_

TMessagesProj/jni/voip/webrtc/modules/congestion_controller/goog_cc/acknowledged_bitrate_estimator_interface.cc

@@ -0,0 +1,95 @@
+/*
+ *  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 "modules/congestion_controller/goog_cc/acknowledged_bitrate_estimator_interface.h"
+
+#include <algorithm>
+#include <memory>
+
+#include "api/field_trials_view.h"
+#include "api/units/time_delta.h"
+#include "modules/congestion_controller/goog_cc/acknowledged_bitrate_estimator.h"
+#include "modules/congestion_controller/goog_cc/robust_throughput_estimator.h"
+#include "rtc_base/experiments/struct_parameters_parser.h"
+#include "rtc_base/logging.h"
+
+namespace webrtc {
+
+constexpr char RobustThroughputEstimatorSettings::kKey[];
+
+RobustThroughputEstimatorSettings::RobustThroughputEstimatorSettings(
+    const FieldTrialsView* key_value_config) {
+  Parser()->Parse(
+      key_value_config->Lookup(RobustThroughputEstimatorSettings::kKey));
+  if (window_packets < 10 || 1000 < window_packets) {
+    RTC_LOG(LS_WARNING) << "Window size must be between 10 and 1000 packets";
+    window_packets = 20;
+  }
+  if (max_window_packets < 10 || 1000 < max_window_packets) {
+    RTC_LOG(LS_WARNING)
+        << "Max window size must be between 10 and 1000 packets";
+    max_window_packets = 500;
+  }
+  max_window_packets = std::max(max_window_packets, window_packets);
+
+  if (required_packets < 10 || 1000 < required_packets) {
+    RTC_LOG(LS_WARNING) << "Required number of initial packets must be between "
+                           "10 and 1000 packets";
+    required_packets = 10;
+  }
+  required_packets = std::min(required_packets, window_packets);
+
+  if (min_window_duration < TimeDelta::Millis(100) ||
+      TimeDelta::Millis(3000) < min_window_duration) {
+    RTC_LOG(LS_WARNING) << "Window duration must be between 100 and 3000 ms";
+    min_window_duration = TimeDelta::Millis(750);
+  }
+  if (max_window_duration < TimeDelta::Seconds(1) ||
+      TimeDelta::Seconds(15) < max_window_duration) {
+    RTC_LOG(LS_WARNING) << "Max window duration must be between 1 and 15 s";
+    max_window_duration = TimeDelta::Seconds(5);
+  }
+  min_window_duration = std::min(min_window_duration, max_window_duration);
+
+  if (unacked_weight < 0.0 || 1.0 < unacked_weight) {
+    RTC_LOG(LS_WARNING)
+        << "Weight for prior unacked size must be between 0 and 1.";
+    unacked_weight = 1.0;
+  }
+}
+
+std::unique_ptr<StructParametersParser>
+RobustThroughputEstimatorSettings::Parser() {
+  return StructParametersParser::Create(
+      "enabled", &enabled,                          //
+      "window_packets", &window_packets,            //
+      "max_window_packets", &max_window_packets,    //
+      "window_duration", &min_window_duration,      //
+      "max_window_duration", &max_window_duration,  //
+      "required_packets", &required_packets,        //
+      "unacked_weight", &unacked_weight);
+}
+
+AcknowledgedBitrateEstimatorInterface::
+    ~AcknowledgedBitrateEstimatorInterface() {}
+
+std::unique_ptr<AcknowledgedBitrateEstimatorInterface>
+AcknowledgedBitrateEstimatorInterface::Create(
+    const FieldTrialsView* key_value_config) {
+  RobustThroughputEstimatorSettings simplified_estimator_settings(
+      key_value_config);
+  if (simplified_estimator_settings.enabled) {
+    return std::make_unique<RobustThroughputEstimator>(
+        simplified_estimator_settings);
+  }
+  return std::make_unique<AcknowledgedBitrateEstimator>(key_value_config);
+}
+
+}  // namespace webrtc

TMessagesProj/jni/voip/webrtc/modules/congestion_controller/goog_cc/acknowledged_bitrate_estimator_interface.h

@@ -0,0 +1,86 @@
+/*
+ *  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_CONGESTION_CONTROLLER_GOOG_CC_ACKNOWLEDGED_BITRATE_ESTIMATOR_INTERFACE_H_
+#define MODULES_CONGESTION_CONTROLLER_GOOG_CC_ACKNOWLEDGED_BITRATE_ESTIMATOR_INTERFACE_H_
+
+
+#include <memory>
+#include <vector>
+
+#include "absl/types/optional.h"
+#include "api/field_trials_view.h"
+#include "api/transport/network_types.h"
+#include "api/units/data_rate.h"
+#include "api/units/time_delta.h"
+#include "api/units/timestamp.h"
+#include "rtc_base/experiments/struct_parameters_parser.h"
+
+namespace webrtc {
+
+struct RobustThroughputEstimatorSettings {
+  static constexpr char kKey[] = "WebRTC-Bwe-RobustThroughputEstimatorSettings";
+
+  RobustThroughputEstimatorSettings() = delete;
+  explicit RobustThroughputEstimatorSettings(
+      const FieldTrialsView* key_value_config);
+
+  // Set `enabled` to true to use the RobustThroughputEstimator, false to use
+  // the AcknowledgedBitrateEstimator.
+  bool enabled = true;
+
+  // The estimator keeps the smallest window containing at least
+  // `window_packets` and at least the packets received during the last
+  // `min_window_duration` milliseconds.
+  // (This means that it may store more than `window_packets` at high bitrates,
+  // and a longer duration than `min_window_duration` at low bitrates.)
+  // However, if will never store more than kMaxPackets (for performance
+  // reasons), and never longer than max_window_duration (to avoid very old
+  // packets influencing the estimate for example when sending is paused).
+  unsigned window_packets = 20;
+  unsigned max_window_packets = 500;
+  TimeDelta min_window_duration = TimeDelta::Seconds(1);
+  TimeDelta max_window_duration = TimeDelta::Seconds(5);
+
+  // The estimator window requires at least `required_packets` packets
+  // to produce an estimate.
+  unsigned required_packets = 10;
+
+  // If audio packets aren't included in allocation (i.e. the
+  // estimated available bandwidth is divided only among the video
+  // streams), then `unacked_weight` should be set to 0.
+  // If audio packets are included in allocation, but not in bandwidth
+  // estimation (i.e. they don't have transport-wide sequence numbers,
+  // but we nevertheless divide the estimated available bandwidth among
+  // both audio and video streams), then `unacked_weight` should be set to 1.
+  // If all packets have transport-wide sequence numbers, then the value
+  // of `unacked_weight` doesn't matter.
+  double unacked_weight = 1.0;
+
+  std::unique_ptr<StructParametersParser> Parser();
+};
+
+class AcknowledgedBitrateEstimatorInterface {
+ public:
+  static std::unique_ptr<AcknowledgedBitrateEstimatorInterface> Create(
+      const FieldTrialsView* key_value_config);
+  virtual ~AcknowledgedBitrateEstimatorInterface();
+
+  virtual void IncomingPacketFeedbackVector(
+      const std::vector<PacketResult>& packet_feedback_vector) = 0;
+  virtual absl::optional<DataRate> bitrate() const = 0;
+  virtual absl::optional<DataRate> PeekRate() const = 0;
+  virtual void SetAlr(bool in_alr) = 0;
+  virtual void SetAlrEndedTime(Timestamp alr_ended_time) = 0;
+};
+
+}  // namespace webrtc
+
+#endif  // MODULES_CONGESTION_CONTROLLER_GOOG_CC_ACKNOWLEDGED_BITRATE_ESTIMATOR_INTERFACE_H_

TMessagesProj/jni/voip/webrtc/modules/congestion_controller/goog_cc/alr_detector.cc

@@ -0,0 +1,111 @@
+/*
+ *  Copyright (c) 2016 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 "modules/congestion_controller/goog_cc/alr_detector.h"
+
+#include <cstdint>
+#include <cstdio>
+#include <memory>
+
+#include "api/rtc_event_log/rtc_event.h"
+#include "api/rtc_event_log/rtc_event_log.h"
+#include "logging/rtc_event_log/events/rtc_event_alr_state.h"
+#include "rtc_base/checks.h"
+#include "rtc_base/numerics/safe_conversions.h"
+#include "rtc_base/time_utils.h"
+
+namespace webrtc {
+
+namespace {
+AlrDetectorConfig GetConfigFromTrials(const FieldTrialsView* key_value_config) {
+  RTC_CHECK(AlrExperimentSettings::MaxOneFieldTrialEnabled(*key_value_config));
+  absl::optional<AlrExperimentSettings> experiment_settings =
+      AlrExperimentSettings::CreateFromFieldTrial(
+          *key_value_config,
+          AlrExperimentSettings::kScreenshareProbingBweExperimentName);
+  if (!experiment_settings) {
+    experiment_settings = AlrExperimentSettings::CreateFromFieldTrial(
+        *key_value_config,
+        AlrExperimentSettings::kStrictPacingAndProbingExperimentName);
+  }
+  AlrDetectorConfig conf;
+  if (experiment_settings) {
+    conf.bandwidth_usage_ratio =
+        experiment_settings->alr_bandwidth_usage_percent / 100.0;
+    conf.start_budget_level_ratio =
+        experiment_settings->alr_start_budget_level_percent / 100.0;
+    conf.stop_budget_level_ratio =
+        experiment_settings->alr_stop_budget_level_percent / 100.0;
+  }
+  conf.Parser()->Parse(
+      key_value_config->Lookup("WebRTC-AlrDetectorParameters"));
+  return conf;
+}
+}  //  namespace
+
+std::unique_ptr<StructParametersParser> AlrDetectorConfig::Parser() {
+  return StructParametersParser::Create(   //
+      "bw_usage", &bandwidth_usage_ratio,  //
+      "start", &start_budget_level_ratio,  //
+      "stop", &stop_budget_level_ratio);
+}
+
+AlrDetector::AlrDetector(AlrDetectorConfig config, RtcEventLog* event_log)
+    : conf_(config), alr_budget_(0, true), event_log_(event_log) {}
+
+AlrDetector::AlrDetector(const FieldTrialsView* key_value_config)
+    : AlrDetector(GetConfigFromTrials(key_value_config), nullptr) {}
+
+AlrDetector::AlrDetector(const FieldTrialsView* key_value_config,
+                         RtcEventLog* event_log)
+    : AlrDetector(GetConfigFromTrials(key_value_config), event_log) {}
+AlrDetector::~AlrDetector() {}
+
+void AlrDetector::OnBytesSent(size_t bytes_sent, int64_t send_time_ms) {
+  if (!last_send_time_ms_.has_value()) {
+    last_send_time_ms_ = send_time_ms;
+    // Since the duration for sending the bytes is unknwon, return without
+    // updating alr state.
+    return;
+  }
+  int64_t delta_time_ms = send_time_ms - *last_send_time_ms_;
+  last_send_time_ms_ = send_time_ms;
+
+  alr_budget_.UseBudget(bytes_sent);
+  alr_budget_.IncreaseBudget(delta_time_ms);
+  bool state_changed = false;
+  if (alr_budget_.budget_ratio() > conf_.start_budget_level_ratio &&
+      !alr_started_time_ms_) {
+    alr_started_time_ms_.emplace(rtc::TimeMillis());
+    state_changed = true;
+  } else if (alr_budget_.budget_ratio() < conf_.stop_budget_level_ratio &&
+             alr_started_time_ms_) {
+    state_changed = true;
+    alr_started_time_ms_.reset();
+  }
+  if (event_log_ && state_changed) {
+    event_log_->Log(
+        std::make_unique<RtcEventAlrState>(alr_started_time_ms_.has_value()));
+  }
+}
+
+void AlrDetector::SetEstimatedBitrate(int bitrate_bps) {
+  RTC_DCHECK(bitrate_bps);
+  int target_rate_kbps =
+      static_cast<double>(bitrate_bps) * conf_.bandwidth_usage_ratio / 1000;
+  alr_budget_.set_target_rate_kbps(target_rate_kbps);
+}
+
+absl::optional<int64_t> AlrDetector::GetApplicationLimitedRegionStartTime()
+    const {
+  return alr_started_time_ms_;
+}
+
+}  // namespace webrtc

TMessagesProj/jni/voip/webrtc/modules/congestion_controller/goog_cc/alr_detector.h

@@ -0,0 +1,76 @@
+/*
+ *  Copyright (c) 2016 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_CONGESTION_CONTROLLER_GOOG_CC_ALR_DETECTOR_H_
+#define MODULES_CONGESTION_CONTROLLER_GOOG_CC_ALR_DETECTOR_H_
+
+#include <stddef.h>
+#include <stdint.h>
+
+#include <memory>
+
+#include "absl/types/optional.h"
+#include "api/field_trials_view.h"
+#include "modules/pacing/interval_budget.h"
+#include "rtc_base/experiments/alr_experiment.h"
+#include "rtc_base/experiments/struct_parameters_parser.h"
+
+namespace webrtc {
+
+class RtcEventLog;
+
+struct AlrDetectorConfig {
+  // Sent traffic ratio as a function of network capacity used to determine
+  // application-limited region. ALR region start when bandwidth usage drops
+  // below kAlrStartUsageRatio and ends when it raises above
+  // kAlrEndUsageRatio. NOTE: This is intentionally conservative at the moment
+  // until BW adjustments of application limited region is fine tuned.
+  double bandwidth_usage_ratio = 0.65;
+  double start_budget_level_ratio = 0.80;
+  double stop_budget_level_ratio = 0.50;
+  std::unique_ptr<StructParametersParser> Parser();
+};
+// Application limited region detector is a class that utilizes signals of
+// elapsed time and bytes sent to estimate whether network traffic is
+// currently limited by the application's ability to generate traffic.
+//
+// AlrDetector provides a signal that can be utilized to adjust
+// estimate bandwidth.
+// Note: This class is not thread-safe.
+class AlrDetector {
+ public:
+  AlrDetector(AlrDetectorConfig config, RtcEventLog* event_log);
+  explicit AlrDetector(const FieldTrialsView* key_value_config);
+  AlrDetector(const FieldTrialsView* key_value_config, RtcEventLog* event_log);
+  ~AlrDetector();
+
+  void OnBytesSent(size_t bytes_sent, int64_t send_time_ms);
+
+  // Set current estimated bandwidth.
+  void SetEstimatedBitrate(int bitrate_bps);
+
+  // Returns time in milliseconds when the current application-limited region
+  // started or empty result if the sender is currently not application-limited.
+  absl::optional<int64_t> GetApplicationLimitedRegionStartTime() const;
+
+ private:
+  friend class GoogCcStatePrinter;
+  const AlrDetectorConfig conf_;
+
+  absl::optional<int64_t> last_send_time_ms_;
+
+  IntervalBudget alr_budget_;
+  absl::optional<int64_t> alr_started_time_ms_;
+
+  RtcEventLog* event_log_;
+};
+}  // namespace webrtc
+
+#endif  // MODULES_CONGESTION_CONTROLLER_GOOG_CC_ALR_DETECTOR_H_

TMessagesProj/jni/voip/webrtc/modules/congestion_controller/goog_cc/bitrate_estimator.cc

@@ -0,0 +1,170 @@
+/*
+ *  Copyright (c) 2017 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 "modules/congestion_controller/goog_cc/bitrate_estimator.h"
+
+#include <algorithm>
+#include <cmath>
+#include <cstdint>
+
+#include "absl/types/optional.h"
+#include "api/field_trials_view.h"
+#include "api/units/data_rate.h"
+#include "api/units/data_size.h"
+#include "api/units/time_delta.h"
+#include "api/units/timestamp.h"
+#include "modules/remote_bitrate_estimator/test/bwe_test_logging.h"
+#include "rtc_base/checks.h"
+#include "rtc_base/experiments/field_trial_parser.h"
+
+namespace webrtc {
+
+namespace {
+constexpr int kInitialRateWindowMs = 500;
+constexpr int kRateWindowMs = 150;
+constexpr int kMinRateWindowMs = 150;
+constexpr int kMaxRateWindowMs = 1000;
+
+const char kBweThroughputWindowConfig[] = "WebRTC-BweThroughputWindowConfig";
+
+}  // namespace
+
+BitrateEstimator::BitrateEstimator(const FieldTrialsView* key_value_config)
+    : sum_(0),
+      initial_window_ms_("initial_window_ms",
+                         kInitialRateWindowMs,
+                         kMinRateWindowMs,
+                         kMaxRateWindowMs),
+      noninitial_window_ms_("window_ms",
+                            kRateWindowMs,
+                            kMinRateWindowMs,
+                            kMaxRateWindowMs),
+      uncertainty_scale_("scale", 10.0),
+      uncertainty_scale_in_alr_("scale_alr", uncertainty_scale_),
+      small_sample_uncertainty_scale_("scale_small", uncertainty_scale_),
+      small_sample_threshold_("small_thresh", DataSize::Zero()),
+      uncertainty_symmetry_cap_("symmetry_cap", DataRate::Zero()),
+      estimate_floor_("floor", DataRate::Zero()),
+      current_window_ms_(0),
+      prev_time_ms_(-1),
+      bitrate_estimate_kbps_(-1.0f),
+      bitrate_estimate_var_(50.0f) {
+  // E.g WebRTC-BweThroughputWindowConfig/initial_window_ms:350,window_ms:250/
+  ParseFieldTrial(
+      {&initial_window_ms_, &noninitial_window_ms_, &uncertainty_scale_,
+       &uncertainty_scale_in_alr_, &small_sample_uncertainty_scale_,
+       &small_sample_threshold_, &uncertainty_symmetry_cap_, &estimate_floor_},
+      key_value_config->Lookup(kBweThroughputWindowConfig));
+}
+
+BitrateEstimator::~BitrateEstimator() = default;
+
+void BitrateEstimator::Update(Timestamp at_time, DataSize amount, bool in_alr) {
+  int rate_window_ms = noninitial_window_ms_.Get();
+  // We use a larger window at the beginning to get a more stable sample that
+  // we can use to initialize the estimate.
+  if (bitrate_estimate_kbps_ < 0.f)
+    rate_window_ms = initial_window_ms_.Get();
+  bool is_small_sample = false;
+  float bitrate_sample_kbps = UpdateWindow(at_time.ms(), amount.bytes(),
+                                           rate_window_ms, &is_small_sample);
+  if (bitrate_sample_kbps < 0.0f)
+    return;
+  if (bitrate_estimate_kbps_ < 0.0f) {
+    // This is the very first sample we get. Use it to initialize the estimate.
+    bitrate_estimate_kbps_ = bitrate_sample_kbps;
+    return;
+  }
+  // Optionally use higher uncertainty for very small samples to avoid dropping
+  // estimate and for samples obtained in ALR.
+  float scale = uncertainty_scale_;
+  if (is_small_sample && bitrate_sample_kbps < bitrate_estimate_kbps_) {
+    scale = small_sample_uncertainty_scale_;
+  } else if (in_alr && bitrate_sample_kbps < bitrate_estimate_kbps_) {
+    // Optionally use higher uncertainty for samples obtained during ALR.
+    scale = uncertainty_scale_in_alr_;
+  }
+  // Define the sample uncertainty as a function of how far away it is from the
+  // current estimate. With low values of uncertainty_symmetry_cap_ we add more
+  // uncertainty to increases than to decreases. For higher values we approach
+  // symmetry.
+  float sample_uncertainty =
+      scale * std::abs(bitrate_estimate_kbps_ - bitrate_sample_kbps) /
+      (bitrate_estimate_kbps_ +
+       std::min(bitrate_sample_kbps,
+                uncertainty_symmetry_cap_.Get().kbps<float>()));
+
+  float sample_var = sample_uncertainty * sample_uncertainty;
+  // Update a bayesian estimate of the rate, weighting it lower if the sample
+  // uncertainty is large.
+  // The bitrate estimate uncertainty is increased with each update to model
+  // that the bitrate changes over time.
+  float pred_bitrate_estimate_var = bitrate_estimate_var_ + 5.f;
+  bitrate_estimate_kbps_ = (sample_var * bitrate_estimate_kbps_ +
+                            pred_bitrate_estimate_var * bitrate_sample_kbps) /
+                           (sample_var + pred_bitrate_estimate_var);
+  bitrate_estimate_kbps_ =
+      std::max(bitrate_estimate_kbps_, estimate_floor_.Get().kbps<float>());
+  bitrate_estimate_var_ = sample_var * pred_bitrate_estimate_var /
+                          (sample_var + pred_bitrate_estimate_var);
+  BWE_TEST_LOGGING_PLOT(1, "acknowledged_bitrate", at_time.ms(),
+                        bitrate_estimate_kbps_ * 1000);
+}
+
+float BitrateEstimator::UpdateWindow(int64_t now_ms,
+                                     int bytes,
+                                     int rate_window_ms,
+                                     bool* is_small_sample) {
+  RTC_DCHECK(is_small_sample != nullptr);
+  // Reset if time moves backwards.
+  if (now_ms < prev_time_ms_) {
+    prev_time_ms_ = -1;
+    sum_ = 0;
+    current_window_ms_ = 0;
+  }
+  if (prev_time_ms_ >= 0) {
+    current_window_ms_ += now_ms - prev_time_ms_;
+    // Reset if nothing has been received for more than a full window.
+    if (now_ms - prev_time_ms_ > rate_window_ms) {
+      sum_ = 0;
+      current_window_ms_ %= rate_window_ms;
+    }
+  }
+  prev_time_ms_ = now_ms;
+  float bitrate_sample = -1.0f;
+  if (current_window_ms_ >= rate_window_ms) {
+    *is_small_sample = sum_ < small_sample_threshold_->bytes();
+    bitrate_sample = 8.0f * sum_ / static_cast<float>(rate_window_ms);
+    current_window_ms_ -= rate_window_ms;
+    sum_ = 0;
+  }
+  sum_ += bytes;
+  return bitrate_sample;
+}
+
+absl::optional<DataRate> BitrateEstimator::bitrate() const {
+  if (bitrate_estimate_kbps_ < 0.f)
+    return absl::nullopt;
+  return DataRate::KilobitsPerSec(bitrate_estimate_kbps_);
+}
+
+absl::optional<DataRate> BitrateEstimator::PeekRate() const {
+  if (current_window_ms_ > 0)
+    return DataSize::Bytes(sum_) / TimeDelta::Millis(current_window_ms_);
+  return absl::nullopt;
+}
+
+void BitrateEstimator::ExpectFastRateChange() {
+  // By setting the bitrate-estimate variance to a higher value we allow the
+  // bitrate to change fast for the next few samples.
+  bitrate_estimate_var_ += 200;
+}
+
+}  // namespace webrtc

TMessagesProj/jni/voip/webrtc/modules/congestion_controller/goog_cc/bitrate_estimator.h

@@ -0,0 +1,63 @@
+/*
+ *  Copyright (c) 2017 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_CONGESTION_CONTROLLER_GOOG_CC_BITRATE_ESTIMATOR_H_
+#define MODULES_CONGESTION_CONTROLLER_GOOG_CC_BITRATE_ESTIMATOR_H_
+
+#include <stdint.h>
+
+#include "absl/types/optional.h"
+#include "api/field_trials_view.h"
+#include "api/units/data_rate.h"
+#include "api/units/data_size.h"
+#include "api/units/timestamp.h"
+#include "rtc_base/experiments/field_trial_parser.h"
+
+namespace webrtc {
+
+// Computes a bayesian estimate of the throughput given acks containing
+// the arrival time and payload size. Samples which are far from the current
+// estimate or are based on few packets are given a smaller weight, as they
+// are considered to be more likely to have been caused by, e.g., delay spikes
+// unrelated to congestion.
+class BitrateEstimator {
+ public:
+  explicit BitrateEstimator(const FieldTrialsView* key_value_config);
+  virtual ~BitrateEstimator();
+  virtual void Update(Timestamp at_time, DataSize amount, bool in_alr);
+
+  virtual absl::optional<DataRate> bitrate() const;
+  absl::optional<DataRate> PeekRate() const;
+
+  virtual void ExpectFastRateChange();
+
+ private:
+  float UpdateWindow(int64_t now_ms,
+                     int bytes,
+                     int rate_window_ms,
+                     bool* is_small_sample);
+  int sum_;
+  FieldTrialConstrained<int> initial_window_ms_;
+  FieldTrialConstrained<int> noninitial_window_ms_;
+  FieldTrialParameter<double> uncertainty_scale_;
+  FieldTrialParameter<double> uncertainty_scale_in_alr_;
+  FieldTrialParameter<double> small_sample_uncertainty_scale_;
+  FieldTrialParameter<DataSize> small_sample_threshold_;
+  FieldTrialParameter<DataRate> uncertainty_symmetry_cap_;
+  FieldTrialParameter<DataRate> estimate_floor_;
+  int64_t current_window_ms_;
+  int64_t prev_time_ms_;
+  float bitrate_estimate_kbps_;
+  float bitrate_estimate_var_;
+};
+
+}  // namespace webrtc
+
+#endif  // MODULES_CONGESTION_CONTROLLER_GOOG_CC_BITRATE_ESTIMATOR_H_

TMessagesProj/jni/voip/webrtc/modules/congestion_controller/goog_cc/congestion_window_pushback_controller.cc

@@ -0,0 +1,79 @@
+/*
+ *  Copyright (c) 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.
+ */
+
+#include "modules/congestion_controller/goog_cc/congestion_window_pushback_controller.h"
+
+#include <algorithm>
+#include <cstdint>
+
+#include "absl/strings/match.h"
+#include "api/field_trials_view.h"
+#include "api/units/data_size.h"
+#include "rtc_base/experiments/rate_control_settings.h"
+
+namespace webrtc {
+
+CongestionWindowPushbackController::CongestionWindowPushbackController(
+    const FieldTrialsView* key_value_config)
+    : add_pacing_(
+          absl::StartsWith(key_value_config->Lookup(
+                               "WebRTC-AddPacingToCongestionWindowPushback"),
+                           "Enabled")),
+      min_pushback_target_bitrate_bps_(
+          RateControlSettings::ParseFromKeyValueConfig(key_value_config)
+              .CongestionWindowMinPushbackTargetBitrateBps()),
+      current_data_window_(
+          RateControlSettings::ParseFromKeyValueConfig(key_value_config)
+              .CongestionWindowInitialDataWindow()) {}
+
+void CongestionWindowPushbackController::UpdateOutstandingData(
+    int64_t outstanding_bytes) {
+  outstanding_bytes_ = outstanding_bytes;
+}
+void CongestionWindowPushbackController::UpdatePacingQueue(
+    int64_t pacing_bytes) {
+  pacing_bytes_ = pacing_bytes;
+}
+
+void CongestionWindowPushbackController::SetDataWindow(DataSize data_window) {
+  current_data_window_ = data_window;
+}
+
+uint32_t CongestionWindowPushbackController::UpdateTargetBitrate(
+    uint32_t bitrate_bps) {
+  if (!current_data_window_ || current_data_window_->IsZero())
+    return bitrate_bps;
+  int64_t total_bytes = outstanding_bytes_;
+  if (add_pacing_)
+    total_bytes += pacing_bytes_;
+  double fill_ratio =
+      total_bytes / static_cast<double>(current_data_window_->bytes());
+  if (fill_ratio > 1.5) {
+    encoding_rate_ratio_ *= 0.9;
+  } else if (fill_ratio > 1) {
+    encoding_rate_ratio_ *= 0.95;
+  } else if (fill_ratio < 0.1) {
+    encoding_rate_ratio_ = 1.0;
+  } else {
+    encoding_rate_ratio_ *= 1.05;
+    encoding_rate_ratio_ = std::min(encoding_rate_ratio_, 1.0);
+  }
+  uint32_t adjusted_target_bitrate_bps =
+      static_cast<uint32_t>(bitrate_bps * encoding_rate_ratio_);
+
+  // Do not adjust below the minimum pushback bitrate but do obey if the
+  // original estimate is below it.
+  bitrate_bps = adjusted_target_bitrate_bps < min_pushback_target_bitrate_bps_
+                    ? std::min(bitrate_bps, min_pushback_target_bitrate_bps_)
+                    : adjusted_target_bitrate_bps;
+  return bitrate_bps;
+}
+
+}  // namespace webrtc

TMessagesProj/jni/voip/webrtc/modules/congestion_controller/goog_cc/congestion_window_pushback_controller.h

@@ -0,0 +1,47 @@
+/*
+ *  Copyright (c) 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 MODULES_CONGESTION_CONTROLLER_GOOG_CC_CONGESTION_WINDOW_PUSHBACK_CONTROLLER_H_
+#define MODULES_CONGESTION_CONTROLLER_GOOG_CC_CONGESTION_WINDOW_PUSHBACK_CONTROLLER_H_
+
+#include <stdint.h>
+
+#include "absl/types/optional.h"
+#include "api/field_trials_view.h"
+#include "api/units/data_size.h"
+
+namespace webrtc {
+
+// This class enables pushback from congestion window directly to video encoder.
+// When the congestion window is filling up, the video encoder target bitrate
+// will be reduced accordingly to accommodate the network changes. To avoid
+// pausing video too frequently, a minimum encoder target bitrate threshold is
+// used to prevent video pause due to a full congestion window.
+class CongestionWindowPushbackController {
+ public:
+  explicit CongestionWindowPushbackController(
+      const FieldTrialsView* key_value_config);
+  void UpdateOutstandingData(int64_t outstanding_bytes);
+  void UpdatePacingQueue(int64_t pacing_bytes);
+  uint32_t UpdateTargetBitrate(uint32_t bitrate_bps);
+  void SetDataWindow(DataSize data_window);
+
+ private:
+  const bool add_pacing_;
+  const uint32_t min_pushback_target_bitrate_bps_;
+  absl::optional<DataSize> current_data_window_;
+  int64_t outstanding_bytes_ = 0;
+  int64_t pacing_bytes_ = 0;
+  double encoding_rate_ratio_ = 1.0;
+};
+
+}  // namespace webrtc
+
+#endif  // MODULES_CONGESTION_CONTROLLER_GOOG_CC_CONGESTION_WINDOW_PUSHBACK_CONTROLLER_H_

TMessagesProj/jni/voip/webrtc/modules/congestion_controller/goog_cc/delay_based_bwe.cc

@@ -0,0 +1,314 @@
+/*
+ *  Copyright (c) 2016 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 "modules/congestion_controller/goog_cc/delay_based_bwe.h"
+
+#include <algorithm>
+#include <cstdint>
+#include <memory>
+#include <utility>
+#include <vector>
+
+#include "absl/types/optional.h"
+#include "api/field_trials_view.h"
+#include "api/network_state_predictor.h"
+#include "api/rtc_event_log/rtc_event_log.h"
+#include "api/transport/network_types.h"
+#include "api/units/data_rate.h"
+#include "api/units/data_size.h"
+#include "api/units/time_delta.h"
+#include "api/units/timestamp.h"
+#include "logging/rtc_event_log/events/rtc_event_bwe_update_delay_based.h"
+#include "modules/congestion_controller/goog_cc/delay_increase_detector_interface.h"
+#include "modules/congestion_controller/goog_cc/inter_arrival_delta.h"
+#include "modules/congestion_controller/goog_cc/trendline_estimator.h"
+#include "modules/remote_bitrate_estimator/include/bwe_defines.h"
+#include "modules/remote_bitrate_estimator/test/bwe_test_logging.h"
+#include "rtc_base/checks.h"
+#include "rtc_base/experiments/struct_parameters_parser.h"
+#include "rtc_base/logging.h"
+#include "rtc_base/race_checker.h"
+#include "system_wrappers/include/metrics.h"
+
+namespace webrtc {
+namespace {
+constexpr TimeDelta kStreamTimeOut = TimeDelta::Seconds(2);
+constexpr TimeDelta kSendTimeGroupLength = TimeDelta::Millis(5);
+
+// This ssrc is used to fulfill the current API but will be removed
+// after the API has been changed.
+constexpr uint32_t kFixedSsrc = 0;
+}  // namespace
+
+constexpr char BweSeparateAudioPacketsSettings::kKey[];
+
+BweSeparateAudioPacketsSettings::BweSeparateAudioPacketsSettings(
+    const FieldTrialsView* key_value_config) {
+  Parser()->Parse(
+      key_value_config->Lookup(BweSeparateAudioPacketsSettings::kKey));
+}
+
+std::unique_ptr<StructParametersParser>
+BweSeparateAudioPacketsSettings::Parser() {
+  return StructParametersParser::Create(      //
+      "enabled", &enabled,                    //
+      "packet_threshold", &packet_threshold,  //
+      "time_threshold", &time_threshold);
+}
+
+DelayBasedBwe::Result::Result()
+    : updated(false),
+      probe(false),
+      target_bitrate(DataRate::Zero()),
+      recovered_from_overuse(false),
+      delay_detector_state(BandwidthUsage::kBwNormal) {}
+
+DelayBasedBwe::DelayBasedBwe(const FieldTrialsView* key_value_config,
+                             RtcEventLog* event_log,
+                             NetworkStatePredictor* network_state_predictor)
+    : event_log_(event_log),
+      key_value_config_(key_value_config),
+      separate_audio_(key_value_config),
+      audio_packets_since_last_video_(0),
+      last_video_packet_recv_time_(Timestamp::MinusInfinity()),
+      network_state_predictor_(network_state_predictor),
+      video_delay_detector_(
+          new TrendlineEstimator(key_value_config_, network_state_predictor_)),
+      audio_delay_detector_(
+          new TrendlineEstimator(key_value_config_, network_state_predictor_)),
+      active_delay_detector_(video_delay_detector_.get()),
+      last_seen_packet_(Timestamp::MinusInfinity()),
+      uma_recorded_(false),
+      rate_control_(*key_value_config, /*send_side=*/true),
+      prev_bitrate_(DataRate::Zero()),
+      prev_state_(BandwidthUsage::kBwNormal) {
+  RTC_LOG(LS_INFO)
+      << "Initialized DelayBasedBwe with separate audio overuse detection"
+      << separate_audio_.Parser()->Encode();
+}
+
+DelayBasedBwe::~DelayBasedBwe() {}
+
+DelayBasedBwe::Result DelayBasedBwe::IncomingPacketFeedbackVector(
+    const TransportPacketsFeedback& msg,
+    absl::optional<DataRate> acked_bitrate,
+    absl::optional<DataRate> probe_bitrate,
+    absl::optional<NetworkStateEstimate> network_estimate,
+    bool in_alr) {
+  RTC_DCHECK_RUNS_SERIALIZED(&network_race_);
+
+  auto packet_feedback_vector = msg.SortedByReceiveTime();
+  // TODO(holmer): An empty feedback vector here likely means that
+  // all acks were too late and that the send time history had
+  // timed out. We should reduce the rate when this occurs.
+  if (packet_feedback_vector.empty()) {
+    RTC_LOG(LS_WARNING) << "Very late feedback received.";
+    return DelayBasedBwe::Result();
+  }
+
+  if (!uma_recorded_) {
+    RTC_HISTOGRAM_ENUMERATION(kBweTypeHistogram,
+                              BweNames::kSendSideTransportSeqNum,
+                              BweNames::kBweNamesMax);
+    uma_recorded_ = true;
+  }
+  bool delayed_feedback = true;
+  bool recovered_from_overuse = false;
+  BandwidthUsage prev_detector_state = active_delay_detector_->State();
+  for (const auto& packet_feedback : packet_feedback_vector) {
+    delayed_feedback = false;
+    IncomingPacketFeedback(packet_feedback, msg.feedback_time);
+    if (prev_detector_state == BandwidthUsage::kBwUnderusing &&
+        active_delay_detector_->State() == BandwidthUsage::kBwNormal) {
+      recovered_from_overuse = true;
+    }
+    prev_detector_state = active_delay_detector_->State();
+  }
+
+  if (delayed_feedback) {
+    // TODO(bugs.webrtc.org/10125): Design a better mechanism to safe-guard
+    // against building very large network queues.
+    return Result();
+  }
+  rate_control_.SetInApplicationLimitedRegion(in_alr);
+  rate_control_.SetNetworkStateEstimate(network_estimate);
+  return MaybeUpdateEstimate(acked_bitrate, probe_bitrate,
+                             std::move(network_estimate),
+                             recovered_from_overuse, in_alr, msg.feedback_time);
+}
+
+void DelayBasedBwe::IncomingPacketFeedback(const PacketResult& packet_feedback,
+                                           Timestamp at_time) {
+  // Reset if the stream has timed out.
+  if (last_seen_packet_.IsInfinite() ||
+      at_time - last_seen_packet_ > kStreamTimeOut) {
+    video_inter_arrival_delta_ =
+        std::make_unique<InterArrivalDelta>(kSendTimeGroupLength);
+    audio_inter_arrival_delta_ =
+        std::make_unique<InterArrivalDelta>(kSendTimeGroupLength);
+
+    video_delay_detector_.reset(
+        new TrendlineEstimator(key_value_config_, network_state_predictor_));
+    audio_delay_detector_.reset(
+        new TrendlineEstimator(key_value_config_, network_state_predictor_));
+    active_delay_detector_ = video_delay_detector_.get();
+  }
+  last_seen_packet_ = at_time;
+
+  // As an alternative to ignoring small packets, we can separate audio and
+  // video packets for overuse detection.
+  DelayIncreaseDetectorInterface* delay_detector_for_packet =
+      video_delay_detector_.get();
+  if (separate_audio_.enabled) {
+    if (packet_feedback.sent_packet.audio) {
+      delay_detector_for_packet = audio_delay_detector_.get();
+      audio_packets_since_last_video_++;
+      if (audio_packets_since_last_video_ > separate_audio_.packet_threshold &&
+          packet_feedback.receive_time - last_video_packet_recv_time_ >
+              separate_audio_.time_threshold) {
+        active_delay_detector_ = audio_delay_detector_.get();
+      }
+    } else {
+      audio_packets_since_last_video_ = 0;
+      last_video_packet_recv_time_ =
+          std::max(last_video_packet_recv_time_, packet_feedback.receive_time);
+      active_delay_detector_ = video_delay_detector_.get();
+    }
+  }
+  DataSize packet_size = packet_feedback.sent_packet.size;
+
+  TimeDelta send_delta = TimeDelta::Zero();
+  TimeDelta recv_delta = TimeDelta::Zero();
+  int size_delta = 0;
+
+  InterArrivalDelta* inter_arrival_for_packet =
+      (separate_audio_.enabled && packet_feedback.sent_packet.audio)
+          ? audio_inter_arrival_delta_.get()
+          : video_inter_arrival_delta_.get();
+  bool calculated_deltas = inter_arrival_for_packet->ComputeDeltas(
+      packet_feedback.sent_packet.send_time, packet_feedback.receive_time,
+      at_time, packet_size.bytes(), &send_delta, &recv_delta, &size_delta);
+
+  delay_detector_for_packet->Update(recv_delta.ms<double>(),
+                                    send_delta.ms<double>(),
+                                    packet_feedback.sent_packet.send_time.ms(),
+                                    packet_feedback.receive_time.ms(),
+                                    packet_size.bytes(), calculated_deltas);
+}
+
+DataRate DelayBasedBwe::TriggerOveruse(Timestamp at_time,
+                                       absl::optional<DataRate> link_capacity) {
+  RateControlInput input(BandwidthUsage::kBwOverusing, link_capacity);
+  return rate_control_.Update(input, at_time);
+}
+
+DelayBasedBwe::Result DelayBasedBwe::MaybeUpdateEstimate(
+    absl::optional<DataRate> acked_bitrate,
+    absl::optional<DataRate> probe_bitrate,
+    absl::optional<NetworkStateEstimate> state_estimate,
+    bool recovered_from_overuse,
+    bool in_alr,
+    Timestamp at_time) {
+  Result result;
+
+  // Currently overusing the bandwidth.
+  if (active_delay_detector_->State() == BandwidthUsage::kBwOverusing) {
+    if (acked_bitrate &&
+        rate_control_.TimeToReduceFurther(at_time, *acked_bitrate)) {
+      result.updated =
+          UpdateEstimate(at_time, acked_bitrate, &result.target_bitrate);
+    } else if (!acked_bitrate && rate_control_.ValidEstimate() &&
+               rate_control_.InitialTimeToReduceFurther(at_time)) {
+      // Overusing before we have a measured acknowledged bitrate. Reduce send
+      // rate by 50% every 200 ms.
+      // TODO(tschumim): Improve this and/or the acknowledged bitrate estimator
+      // so that we (almost) always have a bitrate estimate.
+      rate_control_.SetEstimate(rate_control_.LatestEstimate() / 2, at_time);
+      result.updated = true;
+      result.probe = false;
+      result.target_bitrate = rate_control_.LatestEstimate();
+    }
+  } else {
+    if (probe_bitrate) {
+      result.probe = true;
+      result.updated = true;
+      rate_control_.SetEstimate(*probe_bitrate, at_time);
+      result.target_bitrate = rate_control_.LatestEstimate();
+    } else {
+      result.updated =
+          UpdateEstimate(at_time, acked_bitrate, &result.target_bitrate);
+      result.recovered_from_overuse = recovered_from_overuse;
+    }
+  }
+  BandwidthUsage detector_state = active_delay_detector_->State();
+  if ((result.updated && prev_bitrate_ != result.target_bitrate) ||
+      detector_state != prev_state_) {
+    DataRate bitrate = result.updated ? result.target_bitrate : prev_bitrate_;
+
+    BWE_TEST_LOGGING_PLOT(1, "target_bitrate_bps", at_time.ms(), bitrate.bps());
+
+    if (event_log_) {
+      event_log_->Log(std::make_unique<RtcEventBweUpdateDelayBased>(
+          bitrate.bps(), detector_state));
+    }
+
+    prev_bitrate_ = bitrate;
+    prev_state_ = detector_state;
+  }
+
+  result.delay_detector_state = detector_state;
+  return result;
+}
+
+bool DelayBasedBwe::UpdateEstimate(Timestamp at_time,
+                                   absl::optional<DataRate> acked_bitrate,
+                                   DataRate* target_rate) {
+  const RateControlInput input(active_delay_detector_->State(), acked_bitrate);
+  *target_rate = rate_control_.Update(input, at_time);
+  return rate_control_.ValidEstimate();
+}
+
+void DelayBasedBwe::OnRttUpdate(TimeDelta avg_rtt) {
+  rate_control_.SetRtt(avg_rtt);
+}
+
+bool DelayBasedBwe::LatestEstimate(std::vector<uint32_t>* ssrcs,
+                                   DataRate* bitrate) const {
+  // Currently accessed from both the process thread (see
+  // ModuleRtpRtcpImpl::Process()) and the configuration thread (see
+  // Call::GetStats()). Should in the future only be accessed from a single
+  // thread.
+  RTC_DCHECK(ssrcs);
+  RTC_DCHECK(bitrate);
+  if (!rate_control_.ValidEstimate())
+    return false;
+
+  *ssrcs = {kFixedSsrc};
+  *bitrate = rate_control_.LatestEstimate();
+  return true;
+}
+
+void DelayBasedBwe::SetStartBitrate(DataRate start_bitrate) {
+  RTC_LOG(LS_INFO) << "BWE Setting start bitrate to: "
+                   << ToString(start_bitrate);
+  rate_control_.SetStartBitrate(start_bitrate);
+}
+
+void DelayBasedBwe::SetMinBitrate(DataRate min_bitrate) {
+  // Called from both the configuration thread and the network thread. Shouldn't
+  // be called from the network thread in the future.
+  rate_control_.SetMinBitrate(min_bitrate);
+}
+
+TimeDelta DelayBasedBwe::GetExpectedBwePeriod() const {
+  return rate_control_.GetExpectedBandwidthPeriod();
+}
+
+}  // namespace webrtc

TMessagesProj/jni/voip/webrtc/modules/congestion_controller/goog_cc/delay_based_bwe.h

@@ -0,0 +1,136 @@
+/*
+ *  Copyright (c) 2016 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_CONGESTION_CONTROLLER_GOOG_CC_DELAY_BASED_BWE_H_
+#define MODULES_CONGESTION_CONTROLLER_GOOG_CC_DELAY_BASED_BWE_H_
+
+#include <stdint.h>
+
+#include <memory>
+#include <vector>
+
+#include "absl/types/optional.h"
+#include "api/field_trials_view.h"
+#include "api/network_state_predictor.h"
+#include "api/transport/network_types.h"
+#include "api/units/data_rate.h"
+#include "api/units/time_delta.h"
+#include "api/units/timestamp.h"
+#include "modules/congestion_controller/goog_cc/delay_increase_detector_interface.h"
+#include "modules/congestion_controller/goog_cc/inter_arrival_delta.h"
+#include "modules/congestion_controller/goog_cc/link_capacity_estimator.h"
+#include "modules/congestion_controller/goog_cc/probe_bitrate_estimator.h"
+#include "modules/remote_bitrate_estimator/aimd_rate_control.h"
+#include "modules/remote_bitrate_estimator/inter_arrival.h"
+#include "rtc_base/experiments/struct_parameters_parser.h"
+#include "rtc_base/race_checker.h"
+
+namespace webrtc {
+class RtcEventLog;
+
+struct BweSeparateAudioPacketsSettings {
+  static constexpr char kKey[] = "WebRTC-Bwe-SeparateAudioPackets";
+
+  BweSeparateAudioPacketsSettings() = default;
+  explicit BweSeparateAudioPacketsSettings(
+      const FieldTrialsView* key_value_config);
+
+  bool enabled = false;
+  int packet_threshold = 10;
+  TimeDelta time_threshold = TimeDelta::Seconds(1);
+
+  std::unique_ptr<StructParametersParser> Parser();
+};
+
+class DelayBasedBwe {
+ public:
+  struct Result {
+    Result();
+    ~Result() = default;
+    bool updated;
+    bool probe;
+    DataRate target_bitrate = DataRate::Zero();
+    bool recovered_from_overuse;
+    BandwidthUsage delay_detector_state;
+  };
+
+  explicit DelayBasedBwe(const FieldTrialsView* key_value_config,
+                         RtcEventLog* event_log,
+                         NetworkStatePredictor* network_state_predictor);
+
+  DelayBasedBwe() = delete;
+  DelayBasedBwe(const DelayBasedBwe&) = delete;
+  DelayBasedBwe& operator=(const DelayBasedBwe&) = delete;
+
+  virtual ~DelayBasedBwe();
+
+  Result IncomingPacketFeedbackVector(
+      const TransportPacketsFeedback& msg,
+      absl::optional<DataRate> acked_bitrate,
+      absl::optional<DataRate> probe_bitrate,
+      absl::optional<NetworkStateEstimate> network_estimate,
+      bool in_alr);
+  void OnRttUpdate(TimeDelta avg_rtt);
+  bool LatestEstimate(std::vector<uint32_t>* ssrcs, DataRate* bitrate) const;
+  void SetStartBitrate(DataRate start_bitrate);
+  void SetMinBitrate(DataRate min_bitrate);
+  TimeDelta GetExpectedBwePeriod() const;
+  DataRate TriggerOveruse(Timestamp at_time,
+                          absl::optional<DataRate> link_capacity);
+  DataRate last_estimate() const { return prev_bitrate_; }
+  BandwidthUsage last_state() const { return prev_state_; }
+
+ private:
+  friend class GoogCcStatePrinter;
+  void IncomingPacketFeedback(const PacketResult& packet_feedback,
+                              Timestamp at_time);
+  Result MaybeUpdateEstimate(
+      absl::optional<DataRate> acked_bitrate,
+      absl::optional<DataRate> probe_bitrate,
+      absl::optional<NetworkStateEstimate> state_estimate,
+      bool recovered_from_overuse,
+      bool in_alr,
+      Timestamp at_time);
+  // Updates the current remote rate estimate and returns true if a valid
+  // estimate exists.
+  bool UpdateEstimate(Timestamp at_time,
+                      absl::optional<DataRate> acked_bitrate,
+                      DataRate* target_rate);
+
+  rtc::RaceChecker network_race_;
+  RtcEventLog* const event_log_;
+  const FieldTrialsView* const key_value_config_;
+
+  // Alternatively, run two separate overuse detectors for audio and video,
+  // and fall back to the audio one if we haven't seen a video packet in a
+  // while.
+  BweSeparateAudioPacketsSettings separate_audio_;
+  int64_t audio_packets_since_last_video_;
+  Timestamp last_video_packet_recv_time_;
+
+  NetworkStatePredictor* network_state_predictor_;
+  std::unique_ptr<InterArrival> video_inter_arrival_;
+  std::unique_ptr<InterArrivalDelta> video_inter_arrival_delta_;
+  std::unique_ptr<DelayIncreaseDetectorInterface> video_delay_detector_;
+  std::unique_ptr<InterArrival> audio_inter_arrival_;
+  std::unique_ptr<InterArrivalDelta> audio_inter_arrival_delta_;
+  std::unique_ptr<DelayIncreaseDetectorInterface> audio_delay_detector_;
+  DelayIncreaseDetectorInterface* active_delay_detector_;
+
+  Timestamp last_seen_packet_;
+  bool uma_recorded_;
+  AimdRateControl rate_control_;
+  DataRate prev_bitrate_;
+  BandwidthUsage prev_state_;
+};
+
+}  // namespace webrtc
+
+#endif  // MODULES_CONGESTION_CONTROLLER_GOOG_CC_DELAY_BASED_BWE_H_

TMessagesProj/jni/voip/webrtc/modules/congestion_controller/goog_cc/delay_based_bwe_unittest.cc

@@ -0,0 +1,256 @@
+/*
+ *  Copyright (c) 2016 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 "modules/congestion_controller/goog_cc/delay_based_bwe.h"
+
+#include <cstdint>
+
+#include "api/network_state_predictor.h"
+#include "api/transport/network_types.h"
+#include "api/units/data_rate.h"
+#include "api/units/time_delta.h"
+#include "modules/congestion_controller/goog_cc/delay_based_bwe_unittest_helper.h"
+#include "system_wrappers/include/clock.h"
+#include "test/gtest.h"
+
+namespace webrtc {
+
+namespace {
+constexpr int kNumProbesCluster0 = 5;
+constexpr int kNumProbesCluster1 = 8;
+const PacedPacketInfo kPacingInfo0(0, kNumProbesCluster0, 2000);
+const PacedPacketInfo kPacingInfo1(1, kNumProbesCluster1, 4000);
+constexpr float kTargetUtilizationFraction = 0.95f;
+}  // namespace
+
+TEST_F(DelayBasedBweTest, ProbeDetection) {
+  int64_t now_ms = clock_.TimeInMilliseconds();
+
+  // First burst sent at 8 * 1000 / 10 = 800 kbps.
+  for (int i = 0; i < kNumProbesCluster0; ++i) {
+    clock_.AdvanceTimeMilliseconds(10);
+    now_ms = clock_.TimeInMilliseconds();
+    IncomingFeedback(now_ms, now_ms, 1000, kPacingInfo0);
+  }
+  EXPECT_TRUE(bitrate_observer_.updated());
+
+  // Second burst sent at 8 * 1000 / 5 = 1600 kbps.
+  for (int i = 0; i < kNumProbesCluster1; ++i) {
+    clock_.AdvanceTimeMilliseconds(5);
+    now_ms = clock_.TimeInMilliseconds();
+    IncomingFeedback(now_ms, now_ms, 1000, kPacingInfo1);
+  }
+
+  EXPECT_TRUE(bitrate_observer_.updated());
+  EXPECT_GT(bitrate_observer_.latest_bitrate(), 1500000u);
+}
+
+TEST_F(DelayBasedBweTest, ProbeDetectionNonPacedPackets) {
+  int64_t now_ms = clock_.TimeInMilliseconds();
+  // First burst sent at 8 * 1000 / 10 = 800 kbps, but with every other packet
+  // not being paced which could mess things up.
+  for (int i = 0; i < kNumProbesCluster0; ++i) {
+    clock_.AdvanceTimeMilliseconds(5);
+    now_ms = clock_.TimeInMilliseconds();
+    IncomingFeedback(now_ms, now_ms, 1000, kPacingInfo0);
+    // Non-paced packet, arriving 5 ms after.
+    clock_.AdvanceTimeMilliseconds(5);
+    IncomingFeedback(now_ms, now_ms, 100, PacedPacketInfo());
+  }
+
+  EXPECT_TRUE(bitrate_observer_.updated());
+  EXPECT_GT(bitrate_observer_.latest_bitrate(), 800000u);
+}
+
+TEST_F(DelayBasedBweTest, ProbeDetectionFasterArrival) {
+  int64_t now_ms = clock_.TimeInMilliseconds();
+  // First burst sent at 8 * 1000 / 10 = 800 kbps.
+  // Arriving at 8 * 1000 / 5 = 1600 kbps.
+  int64_t send_time_ms = 0;
+  for (int i = 0; i < kNumProbesCluster0; ++i) {
+    clock_.AdvanceTimeMilliseconds(1);
+    send_time_ms += 10;
+    now_ms = clock_.TimeInMilliseconds();
+    IncomingFeedback(now_ms, send_time_ms, 1000, kPacingInfo0);
+  }
+
+  EXPECT_FALSE(bitrate_observer_.updated());
+}
+
+TEST_F(DelayBasedBweTest, ProbeDetectionSlowerArrival) {
+  int64_t now_ms = clock_.TimeInMilliseconds();
+  // First burst sent at 8 * 1000 / 5 = 1600 kbps.
+  // Arriving at 8 * 1000 / 7 = 1142 kbps.
+  // Since the receive rate is significantly below the send rate, we expect to
+  // use 95% of the estimated capacity.
+  int64_t send_time_ms = 0;
+  for (int i = 0; i < kNumProbesCluster1; ++i) {
+    clock_.AdvanceTimeMilliseconds(7);
+    send_time_ms += 5;
+    now_ms = clock_.TimeInMilliseconds();
+    IncomingFeedback(now_ms, send_time_ms, 1000, kPacingInfo1);
+  }
+
+  EXPECT_TRUE(bitrate_observer_.updated());
+  EXPECT_NEAR(bitrate_observer_.latest_bitrate(),
+              kTargetUtilizationFraction * 1140000u, 10000u);
+}
+
+TEST_F(DelayBasedBweTest, ProbeDetectionSlowerArrivalHighBitrate) {
+  int64_t now_ms = clock_.TimeInMilliseconds();
+  // Burst sent at 8 * 1000 / 1 = 8000 kbps.
+  // Arriving at 8 * 1000 / 2 = 4000 kbps.
+  // Since the receive rate is significantly below the send rate, we expect to
+  // use 95% of the estimated capacity.
+  int64_t send_time_ms = 0;
+  for (int i = 0; i < kNumProbesCluster1; ++i) {
+    clock_.AdvanceTimeMilliseconds(2);
+    send_time_ms += 1;
+    now_ms = clock_.TimeInMilliseconds();
+    IncomingFeedback(now_ms, send_time_ms, 1000, kPacingInfo1);
+  }
+
+  EXPECT_TRUE(bitrate_observer_.updated());
+  EXPECT_NEAR(bitrate_observer_.latest_bitrate(),
+              kTargetUtilizationFraction * 4000000u, 10000u);
+}
+
+TEST_F(DelayBasedBweTest, GetExpectedBwePeriodMs) {
+  auto default_interval = bitrate_estimator_->GetExpectedBwePeriod();
+  EXPECT_GT(default_interval.ms(), 0);
+  CapacityDropTestHelper(1, true, 533, 0);
+  auto interval = bitrate_estimator_->GetExpectedBwePeriod();
+  EXPECT_GT(interval.ms(), 0);
+  EXPECT_NE(interval.ms(), default_interval.ms());
+}
+
+TEST_F(DelayBasedBweTest, InitialBehavior) {
+  InitialBehaviorTestHelper(730000);
+}
+
+TEST_F(DelayBasedBweTest, InitializeResult) {
+  DelayBasedBwe::Result result;
+  EXPECT_EQ(result.delay_detector_state, BandwidthUsage::kBwNormal);
+}
+
+TEST_F(DelayBasedBweTest, RateIncreaseReordering) {
+  RateIncreaseReorderingTestHelper(730000);
+}
+TEST_F(DelayBasedBweTest, RateIncreaseRtpTimestamps) {
+  RateIncreaseRtpTimestampsTestHelper(617);
+}
+
+TEST_F(DelayBasedBweTest, CapacityDropOneStream) {
+  CapacityDropTestHelper(1, false, 500, 0);
+}
+
+TEST_F(DelayBasedBweTest, CapacityDropPosOffsetChange) {
+  CapacityDropTestHelper(1, false, 867, 30000);
+}
+
+TEST_F(DelayBasedBweTest, CapacityDropNegOffsetChange) {
+  CapacityDropTestHelper(1, false, 933, -30000);
+}
+
+TEST_F(DelayBasedBweTest, CapacityDropOneStreamWrap) {
+  CapacityDropTestHelper(1, true, 533, 0);
+}
+
+TEST_F(DelayBasedBweTest, TestTimestampGrouping) {
+  TestTimestampGroupingTestHelper();
+}
+
+TEST_F(DelayBasedBweTest, TestShortTimeoutAndWrap) {
+  // Simulate a client leaving and rejoining the call after 35 seconds. This
+  // will make abs send time wrap, so if streams aren't timed out properly
+  // the next 30 seconds of packets will be out of order.
+  TestWrappingHelper(35);
+}
+
+TEST_F(DelayBasedBweTest, TestLongTimeoutAndWrap) {
+  // Simulate a client leaving and rejoining the call after some multiple of
+  // 64 seconds later. This will cause a zero difference in abs send times due
+  // to the wrap, but a big difference in arrival time, if streams aren't
+  // properly timed out.
+  TestWrappingHelper(10 * 64);
+}
+
+TEST_F(DelayBasedBweTest, TestInitialOveruse) {
+  const DataRate kStartBitrate = DataRate::KilobitsPerSec(300);
+  const DataRate kInitialCapacity = DataRate::KilobitsPerSec(200);
+  const uint32_t kDummySsrc = 0;
+  // High FPS to ensure that we send a lot of packets in a short time.
+  const int kFps = 90;
+
+  stream_generator_->AddStream(new test::RtpStream(kFps, kStartBitrate.bps()));
+  stream_generator_->set_capacity_bps(kInitialCapacity.bps());
+
+  // Needed to initialize the AimdRateControl.
+  bitrate_estimator_->SetStartBitrate(kStartBitrate);
+
+  // Produce 40 frames (in 1/3 second) and give them to the estimator.
+  int64_t bitrate_bps = kStartBitrate.bps();
+  bool seen_overuse = false;
+  for (int i = 0; i < 40; ++i) {
+    bool overuse = GenerateAndProcessFrame(kDummySsrc, bitrate_bps);
+    if (overuse) {
+      EXPECT_TRUE(bitrate_observer_.updated());
+      EXPECT_LE(bitrate_observer_.latest_bitrate(), kInitialCapacity.bps());
+      EXPECT_GT(bitrate_observer_.latest_bitrate(),
+                0.8 * kInitialCapacity.bps());
+      bitrate_bps = bitrate_observer_.latest_bitrate();
+      seen_overuse = true;
+      break;
+    } else if (bitrate_observer_.updated()) {
+      bitrate_bps = bitrate_observer_.latest_bitrate();
+      bitrate_observer_.Reset();
+    }
+  }
+  EXPECT_TRUE(seen_overuse);
+  EXPECT_LE(bitrate_observer_.latest_bitrate(), kInitialCapacity.bps());
+  EXPECT_GT(bitrate_observer_.latest_bitrate(), 0.8 * kInitialCapacity.bps());
+}
+
+TEST_F(DelayBasedBweTest, TestTimestampPrecisionHandling) {
+  // This test does some basic checks to make sure that timestamps with higher
+  // than millisecond precision are handled properly and do not cause any
+  // problems in the estimator. Specifically, previously reported in
+  // webrtc:14023 and described in more details there, the rounding to the
+  // nearest milliseconds caused discrepancy in the accumulated delay. This lead
+  // to false-positive overuse detection.
+  // Technical details of the test:
+  // Send times(ms): 0.000,  9.725, 20.000, 29.725, 40.000, 49.725, ...
+  // Recv times(ms): 0.500, 10.000, 20.500, 30.000, 40.500, 50.000, ...
+  // Send deltas(ms):   9.750,  10.250,  9.750, 10.250,  9.750, ...
+  // Recv deltas(ms):   9.500,  10.500,  9.500, 10.500,  9.500, ...
+  // There is no delay building up between the send times and the receive times,
+  // therefore this case should never lead to an overuse detection. However, if
+  // the time deltas were accidentally rounded to the nearest milliseconds, then
+  // all the send deltas would be equal to 10ms while some recv deltas would
+  // round up to 11ms which would lead in a false illusion of delay build up.
+  uint32_t last_bitrate = bitrate_observer_.latest_bitrate();
+  for (int i = 0; i < 1000; ++i) {
+    clock_.AdvanceTimeMicroseconds(500);
+    IncomingFeedback(clock_.CurrentTime(),
+                     clock_.CurrentTime() - TimeDelta::Micros(500), 1000,
+                     PacedPacketInfo());
+    clock_.AdvanceTimeMicroseconds(9500);
+    IncomingFeedback(clock_.CurrentTime(),
+                     clock_.CurrentTime() - TimeDelta::Micros(250), 1000,
+                     PacedPacketInfo());
+    clock_.AdvanceTimeMicroseconds(10000);
+
+    // The bitrate should never decrease in this test.
+    EXPECT_LE(last_bitrate, bitrate_observer_.latest_bitrate());
+    last_bitrate = bitrate_observer_.latest_bitrate();
+  }
+}
+
+}  // namespace webrtc

TMessagesProj/jni/voip/webrtc/modules/congestion_controller/goog_cc/delay_based_bwe_unittest_helper.cc

@@ -0,0 +1,543 @@
+/*
+ *  Copyright (c) 2016 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 "modules/congestion_controller/goog_cc/delay_based_bwe_unittest_helper.h"
+
+#include <algorithm>
+#include <cstddef>
+#include <cstdint>
+#include <memory>
+#include <vector>
+
+#include "absl/types/optional.h"
+#include "api/transport/network_types.h"
+#include "api/units/data_rate.h"
+#include "api/units/data_size.h"
+#include "api/units/time_delta.h"
+#include "api/units/timestamp.h"
+#include "modules/congestion_controller/goog_cc/acknowledged_bitrate_estimator_interface.h"
+#include "modules/congestion_controller/goog_cc/delay_based_bwe.h"
+#include "modules/congestion_controller/goog_cc/probe_bitrate_estimator.h"
+#include "rtc_base/checks.h"
+#include "test/field_trial.h"
+#include "test/gtest.h"
+
+namespace webrtc {
+constexpr size_t kMtu = 1200;
+constexpr uint32_t kAcceptedBitrateErrorBps = 50000;
+
+// Number of packets needed before we have a valid estimate.
+constexpr int kNumInitialPackets = 2;
+
+constexpr int kInitialProbingPackets = 5;
+
+namespace test {
+
+void TestBitrateObserver::OnReceiveBitrateChanged(uint32_t bitrate) {
+  latest_bitrate_ = bitrate;
+  updated_ = true;
+}
+
+RtpStream::RtpStream(int fps, int bitrate_bps)
+    : fps_(fps), bitrate_bps_(bitrate_bps), next_rtp_time_(0) {
+  RTC_CHECK_GT(fps_, 0);
+}
+
+// Generates a new frame for this stream. If called too soon after the
+// previous frame, no frame will be generated. The frame is split into
+// packets.
+int64_t RtpStream::GenerateFrame(int64_t time_now_us,
+                                 int64_t* next_sequence_number,
+                                 std::vector<PacketResult>* packets) {
+  if (time_now_us < next_rtp_time_) {
+    return next_rtp_time_;
+  }
+  RTC_CHECK(packets != NULL);
+  size_t bits_per_frame = (bitrate_bps_ + fps_ / 2) / fps_;
+  size_t n_packets =
+      std::max<size_t>((bits_per_frame + 4 * kMtu) / (8 * kMtu), 1u);
+  size_t payload_size = (bits_per_frame + 4 * n_packets) / (8 * n_packets);
+  for (size_t i = 0; i < n_packets; ++i) {
+    PacketResult packet;
+    packet.sent_packet.send_time =
+        Timestamp::Micros(time_now_us + kSendSideOffsetUs);
+    packet.sent_packet.size = DataSize::Bytes(payload_size);
+    packet.sent_packet.sequence_number = (*next_sequence_number)++;
+    packets->push_back(packet);
+  }
+  next_rtp_time_ = time_now_us + (1000000 + fps_ / 2) / fps_;
+  return next_rtp_time_;
+}
+
+// The send-side time when the next frame can be generated.
+int64_t RtpStream::next_rtp_time() const {
+  return next_rtp_time_;
+}
+
+void RtpStream::set_bitrate_bps(int bitrate_bps) {
+  ASSERT_GE(bitrate_bps, 0);
+  bitrate_bps_ = bitrate_bps;
+}
+
+int RtpStream::bitrate_bps() const {
+  return bitrate_bps_;
+}
+
+bool RtpStream::Compare(const std::unique_ptr<RtpStream>& lhs,
+                        const std::unique_ptr<RtpStream>& rhs) {
+  return lhs->next_rtp_time_ < rhs->next_rtp_time_;
+}
+
+StreamGenerator::StreamGenerator(int capacity, int64_t time_now)
+    : capacity_(capacity), prev_arrival_time_us_(time_now) {}
+
+StreamGenerator::~StreamGenerator() = default;
+
+// Add a new stream.
+void StreamGenerator::AddStream(RtpStream* stream) {
+  streams_.push_back(std::unique_ptr<RtpStream>(stream));
+}
+
+// Set the link capacity.
+void StreamGenerator::set_capacity_bps(int capacity_bps) {
+  ASSERT_GT(capacity_bps, 0);
+  capacity_ = capacity_bps;
+}
+
+// Divides `bitrate_bps` among all streams. The allocated bitrate per stream
+// is decided by the current allocation ratios.
+void StreamGenerator::SetBitrateBps(int bitrate_bps) {
+  ASSERT_GE(streams_.size(), 0u);
+  int total_bitrate_before = 0;
+  for (const auto& stream : streams_) {
+    total_bitrate_before += stream->bitrate_bps();
+  }
+  int64_t bitrate_before = 0;
+  int total_bitrate_after = 0;
+  for (const auto& stream : streams_) {
+    bitrate_before += stream->bitrate_bps();
+    int64_t bitrate_after =
+        (bitrate_before * bitrate_bps + total_bitrate_before / 2) /
+        total_bitrate_before;
+    stream->set_bitrate_bps(bitrate_after - total_bitrate_after);
+    total_bitrate_after += stream->bitrate_bps();
+  }
+  ASSERT_EQ(bitrate_before, total_bitrate_before);
+  EXPECT_EQ(total_bitrate_after, bitrate_bps);
+}
+
+// TODO(holmer): Break out the channel simulation part from this class to make
+// it possible to simulate different types of channels.
+int64_t StreamGenerator::GenerateFrame(int64_t time_now_us,
+                                       int64_t* next_sequence_number,
+                                       std::vector<PacketResult>* packets) {
+  RTC_CHECK(packets != NULL);
+  RTC_CHECK(packets->empty());
+  RTC_CHECK_GT(capacity_, 0);
+  auto it =
+      std::min_element(streams_.begin(), streams_.end(), RtpStream::Compare);
+  (*it)->GenerateFrame(time_now_us, next_sequence_number, packets);
+  for (PacketResult& packet : *packets) {
+    int capacity_bpus = capacity_ / 1000;
+    int64_t required_network_time_us =
+        (8 * 1000 * packet.sent_packet.size.bytes() + capacity_bpus / 2) /
+        capacity_bpus;
+    prev_arrival_time_us_ =
+        std::max(time_now_us + required_network_time_us,
+                 prev_arrival_time_us_ + required_network_time_us);
+    packet.receive_time = Timestamp::Micros(prev_arrival_time_us_);
+  }
+  it = std::min_element(streams_.begin(), streams_.end(), RtpStream::Compare);
+  return std::max((*it)->next_rtp_time(), time_now_us);
+}
+}  // namespace test
+
+DelayBasedBweTest::DelayBasedBweTest()
+    : field_trial(std::make_unique<test::ScopedFieldTrials>(
+          "WebRTC-Bwe-RobustThroughputEstimatorSettings/enabled:true/")),
+      clock_(100000000),
+      acknowledged_bitrate_estimator_(
+          AcknowledgedBitrateEstimatorInterface::Create(&field_trial_config_)),
+      probe_bitrate_estimator_(new ProbeBitrateEstimator(nullptr)),
+      bitrate_estimator_(
+          new DelayBasedBwe(&field_trial_config_, nullptr, nullptr)),
+      stream_generator_(new test::StreamGenerator(1e6,  // Capacity.
+                                                  clock_.TimeInMicroseconds())),
+      arrival_time_offset_ms_(0),
+      next_sequence_number_(0),
+      first_update_(true) {}
+
+DelayBasedBweTest::~DelayBasedBweTest() {}
+
+void DelayBasedBweTest::AddDefaultStream() {
+  stream_generator_->AddStream(new test::RtpStream(30, 3e5));
+}
+
+const uint32_t DelayBasedBweTest::kDefaultSsrc = 0;
+
+void DelayBasedBweTest::IncomingFeedback(int64_t arrival_time_ms,
+                                         int64_t send_time_ms,
+                                         size_t payload_size) {
+  IncomingFeedback(arrival_time_ms, send_time_ms, payload_size,
+                   PacedPacketInfo());
+}
+
+void DelayBasedBweTest::IncomingFeedback(int64_t arrival_time_ms,
+                                         int64_t send_time_ms,
+                                         size_t payload_size,
+                                         const PacedPacketInfo& pacing_info) {
+  RTC_CHECK_GE(arrival_time_ms + arrival_time_offset_ms_, 0);
+  IncomingFeedback(Timestamp::Millis(arrival_time_ms + arrival_time_offset_ms_),
+                   Timestamp::Millis(send_time_ms), payload_size, pacing_info);
+}
+
+void DelayBasedBweTest::IncomingFeedback(Timestamp receive_time,
+                                         Timestamp send_time,
+                                         size_t payload_size,
+                                         const PacedPacketInfo& pacing_info) {
+  PacketResult packet;
+  packet.receive_time = receive_time;
+  packet.sent_packet.send_time = send_time;
+  packet.sent_packet.size = DataSize::Bytes(payload_size);
+  packet.sent_packet.pacing_info = pacing_info;
+  packet.sent_packet.sequence_number = next_sequence_number_++;
+  if (packet.sent_packet.pacing_info.probe_cluster_id !=
+      PacedPacketInfo::kNotAProbe)
+    probe_bitrate_estimator_->HandleProbeAndEstimateBitrate(packet);
+
+  TransportPacketsFeedback msg;
+  msg.feedback_time = Timestamp::Millis(clock_.TimeInMilliseconds());
+  msg.packet_feedbacks.push_back(packet);
+  acknowledged_bitrate_estimator_->IncomingPacketFeedbackVector(
+      msg.SortedByReceiveTime());
+  DelayBasedBwe::Result result =
+      bitrate_estimator_->IncomingPacketFeedbackVector(
+          msg, acknowledged_bitrate_estimator_->bitrate(),
+          probe_bitrate_estimator_->FetchAndResetLastEstimatedBitrate(),
+          /*network_estimate*/ absl::nullopt, /*in_alr*/ false);
+  if (result.updated) {
+    bitrate_observer_.OnReceiveBitrateChanged(result.target_bitrate.bps());
+  }
+}
+
+// Generates a frame of packets belonging to a stream at a given bitrate and
+// with a given ssrc. The stream is pushed through a very simple simulated
+// network, and is then given to the receive-side bandwidth estimator.
+// Returns true if an over-use was seen, false otherwise.
+// The StreamGenerator::updated() should be used to check for any changes in
+// target bitrate after the call to this function.
+bool DelayBasedBweTest::GenerateAndProcessFrame(uint32_t ssrc,
+                                                uint32_t bitrate_bps) {
+  stream_generator_->SetBitrateBps(bitrate_bps);
+  std::vector<PacketResult> packets;
+
+  int64_t next_time_us = stream_generator_->GenerateFrame(
+      clock_.TimeInMicroseconds(), &next_sequence_number_, &packets);
+  if (packets.empty())
+    return false;
+
+  bool overuse = false;
+  bitrate_observer_.Reset();
+  clock_.AdvanceTimeMicroseconds(packets.back().receive_time.us() -
+                                 clock_.TimeInMicroseconds());
+  for (auto& packet : packets) {
+    RTC_CHECK_GE(packet.receive_time.ms() + arrival_time_offset_ms_, 0);
+    packet.receive_time += TimeDelta::Millis(arrival_time_offset_ms_);
+
+    if (packet.sent_packet.pacing_info.probe_cluster_id !=
+        PacedPacketInfo::kNotAProbe)
+      probe_bitrate_estimator_->HandleProbeAndEstimateBitrate(packet);
+  }
+
+  acknowledged_bitrate_estimator_->IncomingPacketFeedbackVector(packets);
+  TransportPacketsFeedback msg;
+  msg.packet_feedbacks = packets;
+  msg.feedback_time = Timestamp::Millis(clock_.TimeInMilliseconds());
+
+  DelayBasedBwe::Result result =
+      bitrate_estimator_->IncomingPacketFeedbackVector(
+          msg, acknowledged_bitrate_estimator_->bitrate(),
+          probe_bitrate_estimator_->FetchAndResetLastEstimatedBitrate(),
+          /*network_estimate*/ absl::nullopt, /*in_alr*/ false);
+  if (result.updated) {
+    bitrate_observer_.OnReceiveBitrateChanged(result.target_bitrate.bps());
+    if (!first_update_ && result.target_bitrate.bps() < bitrate_bps)
+      overuse = true;
+    first_update_ = false;
+  }
+
+  clock_.AdvanceTimeMicroseconds(next_time_us - clock_.TimeInMicroseconds());
+  return overuse;
+}
+
+// Run the bandwidth estimator with a stream of `number_of_frames` frames, or
+// until it reaches `target_bitrate`.
+// Can for instance be used to run the estimator for some time to get it
+// into a steady state.
+uint32_t DelayBasedBweTest::SteadyStateRun(uint32_t ssrc,
+                                           int max_number_of_frames,
+                                           uint32_t start_bitrate,
+                                           uint32_t min_bitrate,
+                                           uint32_t max_bitrate,
+                                           uint32_t target_bitrate) {
+  uint32_t bitrate_bps = start_bitrate;
+  bool bitrate_update_seen = false;
+  // Produce `number_of_frames` frames and give them to the estimator.
+  for (int i = 0; i < max_number_of_frames; ++i) {
+    bool overuse = GenerateAndProcessFrame(ssrc, bitrate_bps);
+    if (overuse) {
+      EXPECT_LT(bitrate_observer_.latest_bitrate(), max_bitrate);
+      EXPECT_GT(bitrate_observer_.latest_bitrate(), min_bitrate);
+      bitrate_bps = bitrate_observer_.latest_bitrate();
+      bitrate_update_seen = true;
+    } else if (bitrate_observer_.updated()) {
+      bitrate_bps = bitrate_observer_.latest_bitrate();
+      bitrate_observer_.Reset();
+    }
+    if (bitrate_update_seen && bitrate_bps > target_bitrate) {
+      break;
+    }
+  }
+  EXPECT_TRUE(bitrate_update_seen);
+  return bitrate_bps;
+}
+
+void DelayBasedBweTest::InitialBehaviorTestHelper(
+    uint32_t expected_converge_bitrate) {
+  const int kFramerate = 50;  // 50 fps to avoid rounding errors.
+  const int kFrameIntervalMs = 1000 / kFramerate;
+  const PacedPacketInfo kPacingInfo(0, 5, 5000);
+  DataRate bitrate = DataRate::Zero();
+  int64_t send_time_ms = 0;
+  std::vector<uint32_t> ssrcs;
+  EXPECT_FALSE(bitrate_estimator_->LatestEstimate(&ssrcs, &bitrate));
+  EXPECT_EQ(0u, ssrcs.size());
+  clock_.AdvanceTimeMilliseconds(1000);
+  EXPECT_FALSE(bitrate_estimator_->LatestEstimate(&ssrcs, &bitrate));
+  EXPECT_FALSE(bitrate_observer_.updated());
+  bitrate_observer_.Reset();
+  clock_.AdvanceTimeMilliseconds(1000);
+  // Inserting packets for 5 seconds to get a valid estimate.
+  for (int i = 0; i < 5 * kFramerate + 1 + kNumInitialPackets; ++i) {
+    // NOTE!!! If the following line is moved under the if case then this test
+    //         wont work on windows realease bots.
+    PacedPacketInfo pacing_info =
+        i < kInitialProbingPackets ? kPacingInfo : PacedPacketInfo();
+
+    if (i == kNumInitialPackets) {
+      EXPECT_FALSE(bitrate_estimator_->LatestEstimate(&ssrcs, &bitrate));
+      EXPECT_EQ(0u, ssrcs.size());
+      EXPECT_FALSE(bitrate_observer_.updated());
+      bitrate_observer_.Reset();
+    }
+    IncomingFeedback(clock_.TimeInMilliseconds(), send_time_ms, kMtu,
+                     pacing_info);
+    clock_.AdvanceTimeMilliseconds(1000 / kFramerate);
+    send_time_ms += kFrameIntervalMs;
+  }
+  EXPECT_TRUE(bitrate_estimator_->LatestEstimate(&ssrcs, &bitrate));
+  ASSERT_EQ(1u, ssrcs.size());
+  EXPECT_EQ(kDefaultSsrc, ssrcs.front());
+  EXPECT_NEAR(expected_converge_bitrate, bitrate.bps(),
+              kAcceptedBitrateErrorBps);
+  EXPECT_TRUE(bitrate_observer_.updated());
+  bitrate_observer_.Reset();
+  EXPECT_EQ(bitrate_observer_.latest_bitrate(), bitrate.bps());
+}
+
+void DelayBasedBweTest::RateIncreaseReorderingTestHelper(
+    uint32_t expected_bitrate_bps) {
+  const int kFramerate = 50;  // 50 fps to avoid rounding errors.
+  const int kFrameIntervalMs = 1000 / kFramerate;
+  const PacedPacketInfo kPacingInfo(0, 5, 5000);
+  int64_t send_time_ms = 0;
+  // Inserting packets for five seconds to get a valid estimate.
+  for (int i = 0; i < 5 * kFramerate + 1 + kNumInitialPackets; ++i) {
+    // NOTE!!! If the following line is moved under the if case then this test
+    //         wont work on windows realease bots.
+    PacedPacketInfo pacing_info =
+        i < kInitialProbingPackets ? kPacingInfo : PacedPacketInfo();
+
+    // TODO(sprang): Remove this hack once the single stream estimator is gone,
+    // as it doesn't do anything in Process().
+    if (i == kNumInitialPackets) {
+      // Process after we have enough frames to get a valid input rate estimate.
+
+      EXPECT_FALSE(bitrate_observer_.updated());  // No valid estimate.
+    }
+    IncomingFeedback(clock_.TimeInMilliseconds(), send_time_ms, kMtu,
+                     pacing_info);
+    clock_.AdvanceTimeMilliseconds(kFrameIntervalMs);
+    send_time_ms += kFrameIntervalMs;
+  }
+  EXPECT_TRUE(bitrate_observer_.updated());
+  EXPECT_NEAR(expected_bitrate_bps, bitrate_observer_.latest_bitrate(),
+              kAcceptedBitrateErrorBps);
+  for (int i = 0; i < 10; ++i) {
+    clock_.AdvanceTimeMilliseconds(2 * kFrameIntervalMs);
+    send_time_ms += 2 * kFrameIntervalMs;
+    IncomingFeedback(clock_.TimeInMilliseconds(), send_time_ms, 1000);
+    IncomingFeedback(clock_.TimeInMilliseconds(),
+                     send_time_ms - kFrameIntervalMs, 1000);
+  }
+  EXPECT_TRUE(bitrate_observer_.updated());
+  EXPECT_NEAR(expected_bitrate_bps, bitrate_observer_.latest_bitrate(),
+              kAcceptedBitrateErrorBps);
+}
+
+// Make sure we initially increase the bitrate as expected.
+void DelayBasedBweTest::RateIncreaseRtpTimestampsTestHelper(
+    int expected_iterations) {
+  // This threshold corresponds approximately to increasing linearly with
+  // bitrate(i) = 1.04 * bitrate(i-1) + 1000
+  // until bitrate(i) > 500000, with bitrate(1) ~= 30000.
+  uint32_t bitrate_bps = 30000;
+  int iterations = 0;
+  AddDefaultStream();
+  // Feed the estimator with a stream of packets and verify that it reaches
+  // 500 kbps at the expected time.
+  while (bitrate_bps < 5e5) {
+    bool overuse = GenerateAndProcessFrame(kDefaultSsrc, bitrate_bps);
+    if (overuse) {
+      EXPECT_GT(bitrate_observer_.latest_bitrate(), bitrate_bps);
+      bitrate_bps = bitrate_observer_.latest_bitrate();
+      bitrate_observer_.Reset();
+    } else if (bitrate_observer_.updated()) {
+      bitrate_bps = bitrate_observer_.latest_bitrate();
+      bitrate_observer_.Reset();
+    }
+    ++iterations;
+  }
+  ASSERT_EQ(expected_iterations, iterations);
+}
+
+void DelayBasedBweTest::CapacityDropTestHelper(
+    int number_of_streams,
+    bool wrap_time_stamp,
+    uint32_t expected_bitrate_drop_delta,
+    int64_t receiver_clock_offset_change_ms) {
+  const int kFramerate = 30;
+  const int kStartBitrate = 900e3;
+  const int kMinExpectedBitrate = 800e3;
+  const int kMaxExpectedBitrate = 1100e3;
+  const uint32_t kInitialCapacityBps = 1000e3;
+  const uint32_t kReducedCapacityBps = 500e3;
+
+  int steady_state_time = 0;
+  if (number_of_streams <= 1) {
+    steady_state_time = 10;
+    AddDefaultStream();
+  } else {
+    steady_state_time = 10 * number_of_streams;
+    int bitrate_sum = 0;
+    int kBitrateDenom = number_of_streams * (number_of_streams - 1);
+    for (int i = 0; i < number_of_streams; i++) {
+      // First stream gets half available bitrate, while the rest share the
+      // remaining half i.e.: 1/2 = Sum[n/(N*(N-1))] for n=1..N-1 (rounded up)
+      int bitrate = kStartBitrate / 2;
+      if (i > 0) {
+        bitrate = (kStartBitrate * i + kBitrateDenom / 2) / kBitrateDenom;
+      }
+      stream_generator_->AddStream(new test::RtpStream(kFramerate, bitrate));
+      bitrate_sum += bitrate;
+    }
+    ASSERT_EQ(bitrate_sum, kStartBitrate);
+  }
+
+  // Run in steady state to make the estimator converge.
+  stream_generator_->set_capacity_bps(kInitialCapacityBps);
+  uint32_t bitrate_bps = SteadyStateRun(
+      kDefaultSsrc, steady_state_time * kFramerate, kStartBitrate,
+      kMinExpectedBitrate, kMaxExpectedBitrate, kInitialCapacityBps);
+  EXPECT_NEAR(kInitialCapacityBps, bitrate_bps, 180000u);
+  bitrate_observer_.Reset();
+
+  // Add an offset to make sure the BWE can handle it.
+  arrival_time_offset_ms_ += receiver_clock_offset_change_ms;
+
+  // Reduce the capacity and verify the decrease time.
+  stream_generator_->set_capacity_bps(kReducedCapacityBps);
+  int64_t overuse_start_time = clock_.TimeInMilliseconds();
+  int64_t bitrate_drop_time = -1;
+  for (int i = 0; i < 100 * number_of_streams; ++i) {
+    GenerateAndProcessFrame(kDefaultSsrc, bitrate_bps);
+    if (bitrate_drop_time == -1 &&
+        bitrate_observer_.latest_bitrate() <= kReducedCapacityBps) {
+      bitrate_drop_time = clock_.TimeInMilliseconds();
+    }
+    if (bitrate_observer_.updated())
+      bitrate_bps = bitrate_observer_.latest_bitrate();
+  }
+
+  EXPECT_NEAR(expected_bitrate_drop_delta,
+              bitrate_drop_time - overuse_start_time, 33);
+}
+
+void DelayBasedBweTest::TestTimestampGroupingTestHelper() {
+  const int kFramerate = 50;  // 50 fps to avoid rounding errors.
+  const int kFrameIntervalMs = 1000 / kFramerate;
+  int64_t send_time_ms = 0;
+  // Initial set of frames to increase the bitrate. 6 seconds to have enough
+  // time for the first estimate to be generated and for Process() to be called.
+  for (int i = 0; i <= 6 * kFramerate; ++i) {
+    IncomingFeedback(clock_.TimeInMilliseconds(), send_time_ms, 1000);
+
+    clock_.AdvanceTimeMilliseconds(kFrameIntervalMs);
+    send_time_ms += kFrameIntervalMs;
+  }
+  EXPECT_TRUE(bitrate_observer_.updated());
+  EXPECT_GE(bitrate_observer_.latest_bitrate(), 400000u);
+
+  // Insert batches of frames which were sent very close in time. Also simulate
+  // capacity over-use to see that we back off correctly.
+  const int kTimestampGroupLength = 15;
+  for (int i = 0; i < 100; ++i) {
+    for (int j = 0; j < kTimestampGroupLength; ++j) {
+      // Insert `kTimestampGroupLength` frames with just 1 timestamp ticks in
+      // between. Should be treated as part of the same group by the estimator.
+      IncomingFeedback(clock_.TimeInMilliseconds(), send_time_ms, 100);
+      clock_.AdvanceTimeMilliseconds(kFrameIntervalMs / kTimestampGroupLength);
+      send_time_ms += 1;
+    }
+    // Increase time until next batch to simulate over-use.
+    clock_.AdvanceTimeMilliseconds(10);
+    send_time_ms += kFrameIntervalMs - kTimestampGroupLength;
+  }
+  EXPECT_TRUE(bitrate_observer_.updated());
+  // Should have reduced the estimate.
+  EXPECT_LT(bitrate_observer_.latest_bitrate(), 400000u);
+}
+
+void DelayBasedBweTest::TestWrappingHelper(int silence_time_s) {
+  const int kFramerate = 100;
+  const int kFrameIntervalMs = 1000 / kFramerate;
+  int64_t send_time_ms = 0;
+
+  for (size_t i = 0; i < 3000; ++i) {
+    IncomingFeedback(clock_.TimeInMilliseconds(), send_time_ms, 1000);
+    clock_.AdvanceTimeMilliseconds(kFrameIntervalMs);
+    send_time_ms += kFrameIntervalMs;
+  }
+  DataRate bitrate_before = DataRate::Zero();
+  std::vector<uint32_t> ssrcs;
+  bitrate_estimator_->LatestEstimate(&ssrcs, &bitrate_before);
+
+  clock_.AdvanceTimeMilliseconds(silence_time_s * 1000);
+  send_time_ms += silence_time_s * 1000;
+
+  for (size_t i = 0; i < 24; ++i) {
+    IncomingFeedback(clock_.TimeInMilliseconds(), send_time_ms, 1000);
+    clock_.AdvanceTimeMilliseconds(2 * kFrameIntervalMs);
+    send_time_ms += kFrameIntervalMs;
+  }
+  DataRate bitrate_after = DataRate::Zero();
+  bitrate_estimator_->LatestEstimate(&ssrcs, &bitrate_after);
+  EXPECT_LT(bitrate_after, bitrate_before);
+}
+}  // namespace webrtc

TMessagesProj/jni/voip/webrtc/modules/congestion_controller/goog_cc/delay_based_bwe_unittest_helper.h

@@ -0,0 +1,190 @@
+/*
+ *  Copyright (c) 2016 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_CONGESTION_CONTROLLER_GOOG_CC_DELAY_BASED_BWE_UNITTEST_HELPER_H_
+#define MODULES_CONGESTION_CONTROLLER_GOOG_CC_DELAY_BASED_BWE_UNITTEST_HELPER_H_
+
+#include <stddef.h>
+#include <stdint.h>
+
+#include <memory>
+#include <vector>
+
+#include "api/transport/field_trial_based_config.h"
+#include "api/transport/network_types.h"
+#include "api/units/timestamp.h"
+#include "modules/congestion_controller/goog_cc/acknowledged_bitrate_estimator.h"
+#include "modules/congestion_controller/goog_cc/delay_based_bwe.h"
+#include "system_wrappers/include/clock.h"
+#include "test/field_trial.h"
+#include "test/gtest.h"
+
+namespace webrtc {
+namespace test {
+
+class TestBitrateObserver {
+ public:
+  TestBitrateObserver() : updated_(false), latest_bitrate_(0) {}
+  ~TestBitrateObserver() {}
+
+  void OnReceiveBitrateChanged(uint32_t bitrate);
+
+  void Reset() { updated_ = false; }
+
+  bool updated() const { return updated_; }
+
+  uint32_t latest_bitrate() const { return latest_bitrate_; }
+
+ private:
+  bool updated_;
+  uint32_t latest_bitrate_;
+};
+
+class RtpStream {
+ public:
+  enum { kSendSideOffsetUs = 1000000 };
+
+  RtpStream(int fps, int bitrate_bps);
+
+  RtpStream(const RtpStream&) = delete;
+  RtpStream& operator=(const RtpStream&) = delete;
+
+  // Generates a new frame for this stream. If called too soon after the
+  // previous frame, no frame will be generated. The frame is split into
+  // packets.
+  int64_t GenerateFrame(int64_t time_now_us,
+                        int64_t* next_sequence_number,
+                        std::vector<PacketResult>* packets);
+
+  // The send-side time when the next frame can be generated.
+  int64_t next_rtp_time() const;
+
+  void set_bitrate_bps(int bitrate_bps);
+
+  int bitrate_bps() const;
+
+  static bool Compare(const std::unique_ptr<RtpStream>& lhs,
+                      const std::unique_ptr<RtpStream>& rhs);
+
+ private:
+  int fps_;
+  int bitrate_bps_;
+  int64_t next_rtp_time_;
+};
+
+class StreamGenerator {
+ public:
+  StreamGenerator(int capacity, int64_t time_now);
+  ~StreamGenerator();
+
+  StreamGenerator(const StreamGenerator&) = delete;
+  StreamGenerator& operator=(const StreamGenerator&) = delete;
+
+  // Add a new stream.
+  void AddStream(RtpStream* stream);
+
+  // Set the link capacity.
+  void set_capacity_bps(int capacity_bps);
+
+  // Divides `bitrate_bps` among all streams. The allocated bitrate per stream
+  // is decided by the initial allocation ratios.
+  void SetBitrateBps(int bitrate_bps);
+
+  // Set the RTP timestamp offset for the stream identified by `ssrc`.
+  void set_rtp_timestamp_offset(uint32_t ssrc, uint32_t offset);
+
+  // TODO(holmer): Break out the channel simulation part from this class to make
+  // it possible to simulate different types of channels.
+  int64_t GenerateFrame(int64_t time_now_us,
+                        int64_t* next_sequence_number,
+                        std::vector<PacketResult>* packets);
+
+ private:
+  // Capacity of the simulated channel in bits per second.
+  int capacity_;
+  // The time when the last packet arrived.
+  int64_t prev_arrival_time_us_;
+  // All streams being transmitted on this simulated channel.
+  std::vector<std::unique_ptr<RtpStream>> streams_;
+};
+}  // namespace test
+
+class DelayBasedBweTest : public ::testing::Test {
+ public:
+  DelayBasedBweTest();
+  ~DelayBasedBweTest() override;
+
+ protected:
+  void AddDefaultStream();
+
+  // Helpers to insert a single packet into the delay-based BWE.
+  void IncomingFeedback(int64_t arrival_time_ms,
+                        int64_t send_time_ms,
+                        size_t payload_size);
+  void IncomingFeedback(int64_t arrival_time_ms,
+                        int64_t send_time_ms,
+                        size_t payload_size,
+                        const PacedPacketInfo& pacing_info);
+  void IncomingFeedback(Timestamp receive_time,
+                        Timestamp send_time,
+                        size_t payload_size,
+                        const PacedPacketInfo& pacing_info);
+
+  // Generates a frame of packets belonging to a stream at a given bitrate and
+  // with a given ssrc. The stream is pushed through a very simple simulated
+  // network, and is then given to the receive-side bandwidth estimator.
+  // Returns true if an over-use was seen, false otherwise.
+  // The StreamGenerator::updated() should be used to check for any changes in
+  // target bitrate after the call to this function.
+  bool GenerateAndProcessFrame(uint32_t ssrc, uint32_t bitrate_bps);
+
+  // Run the bandwidth estimator with a stream of `number_of_frames` frames, or
+  // until it reaches `target_bitrate`.
+  // Can for instance be used to run the estimator for some time to get it
+  // into a steady state.
+  uint32_t SteadyStateRun(uint32_t ssrc,
+                          int number_of_frames,
+                          uint32_t start_bitrate,
+                          uint32_t min_bitrate,
+                          uint32_t max_bitrate,
+                          uint32_t target_bitrate);
+
+  void TestTimestampGroupingTestHelper();
+
+  void TestWrappingHelper(int silence_time_s);
+
+  void InitialBehaviorTestHelper(uint32_t expected_converge_bitrate);
+  void RateIncreaseReorderingTestHelper(uint32_t expected_bitrate);
+  void RateIncreaseRtpTimestampsTestHelper(int expected_iterations);
+  void CapacityDropTestHelper(int number_of_streams,
+                              bool wrap_time_stamp,
+                              uint32_t expected_bitrate_drop_delta,
+                              int64_t receiver_clock_offset_change_ms);
+
+  static const uint32_t kDefaultSsrc;
+  FieldTrialBasedConfig field_trial_config_;
+
+  std::unique_ptr<test::ScopedFieldTrials>
+      field_trial;        // Must be initialized first.
+  SimulatedClock clock_;  // Time at the receiver.
+  test::TestBitrateObserver bitrate_observer_;
+  std::unique_ptr<AcknowledgedBitrateEstimatorInterface>
+      acknowledged_bitrate_estimator_;
+  const std::unique_ptr<ProbeBitrateEstimator> probe_bitrate_estimator_;
+  std::unique_ptr<DelayBasedBwe> bitrate_estimator_;
+  std::unique_ptr<test::StreamGenerator> stream_generator_;
+  int64_t arrival_time_offset_ms_;
+  int64_t next_sequence_number_;
+  bool first_update_;
+};
+
+}  // namespace webrtc
+
+#endif  // MODULES_CONGESTION_CONTROLLER_GOOG_CC_DELAY_BASED_BWE_UNITTEST_HELPER_H_

TMessagesProj/jni/voip/webrtc/modules/congestion_controller/goog_cc/delay_increase_detector_interface.h

@@ -0,0 +1,45 @@
+/*
+ *  Copyright (c) 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 MODULES_CONGESTION_CONTROLLER_GOOG_CC_DELAY_INCREASE_DETECTOR_INTERFACE_H_
+#define MODULES_CONGESTION_CONTROLLER_GOOG_CC_DELAY_INCREASE_DETECTOR_INTERFACE_H_
+
+#include <stdint.h>
+
+#include <cstddef>
+
+#include "api/network_state_predictor.h"
+
+namespace webrtc {
+
+class DelayIncreaseDetectorInterface {
+ public:
+  DelayIncreaseDetectorInterface() {}
+  virtual ~DelayIncreaseDetectorInterface() {}
+
+  DelayIncreaseDetectorInterface(const DelayIncreaseDetectorInterface&) =
+      delete;
+  DelayIncreaseDetectorInterface& operator=(
+      const DelayIncreaseDetectorInterface&) = delete;
+
+  // Update the detector with a new sample. The deltas should represent deltas
+  // between timestamp groups as defined by the InterArrival class.
+  virtual void Update(double recv_delta_ms,
+                      double send_delta_ms,
+                      int64_t send_time_ms,
+                      int64_t arrival_time_ms,
+                      size_t packet_size,
+                      bool calculated_deltas) = 0;
+
+  virtual BandwidthUsage State() const = 0;
+};
+
+}  // namespace webrtc
+
+#endif  // MODULES_CONGESTION_CONTROLLER_GOOG_CC_DELAY_INCREASE_DETECTOR_INTERFACE_H_

TMessagesProj/jni/voip/webrtc/modules/congestion_controller/goog_cc/goog_cc_network_control.cc

@@ -0,0 +1,747 @@
+/*
+ *  Copyright (c) 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.
+ */
+
+#include "modules/congestion_controller/goog_cc/goog_cc_network_control.h"
+
+#include <stdio.h>
+
+#include <algorithm>
+#include <cstdint>
+#include <memory>
+#include <numeric>
+#include <utility>
+#include <vector>
+
+#include "absl/strings/match.h"
+#include "absl/strings/string_view.h"
+#include "absl/types/optional.h"
+#include "api/field_trials_view.h"
+#include "api/network_state_predictor.h"
+#include "api/transport/network_control.h"
+#include "api/transport/network_types.h"
+#include "api/units/data_rate.h"
+#include "api/units/data_size.h"
+#include "api/units/time_delta.h"
+#include "api/units/timestamp.h"
+#include "logging/rtc_event_log/events/rtc_event_remote_estimate.h"
+#include "modules/congestion_controller/goog_cc/acknowledged_bitrate_estimator_interface.h"
+#include "modules/congestion_controller/goog_cc/alr_detector.h"
+#include "modules/congestion_controller/goog_cc/congestion_window_pushback_controller.h"
+#include "modules/congestion_controller/goog_cc/delay_based_bwe.h"
+#include "modules/congestion_controller/goog_cc/loss_based_bwe_v2.h"
+#include "modules/congestion_controller/goog_cc/probe_bitrate_estimator.h"
+#include "modules/congestion_controller/goog_cc/probe_controller.h"
+#include "modules/congestion_controller/goog_cc/send_side_bandwidth_estimation.h"
+#include "modules/remote_bitrate_estimator/include/bwe_defines.h"
+#include "modules/remote_bitrate_estimator/test/bwe_test_logging.h"
+#include "rtc_base/checks.h"
+#include "rtc_base/experiments/field_trial_parser.h"
+#include "rtc_base/experiments/rate_control_settings.h"
+#include "rtc_base/logging.h"
+
+namespace webrtc {
+
+namespace {
+// From RTCPSender video report interval.
+constexpr TimeDelta kLossUpdateInterval = TimeDelta::Millis(1000);
+
+// 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.
+constexpr float kDefaultPaceMultiplier = 2.5f;
+
+// If the probe result is far below the current throughput estimate
+// it's unlikely that the probe is accurate, so we don't want to drop too far.
+// However, if we actually are overusing, we want to drop to something slightly
+// below the current throughput estimate to drain the network queues.
+constexpr double kProbeDropThroughputFraction = 0.85;
+
+bool IsEnabled(const FieldTrialsView* config, absl::string_view key) {
+  return absl::StartsWith(config->Lookup(key), "Enabled");
+}
+
+bool IsNotDisabled(const FieldTrialsView* config, absl::string_view key) {
+  return !absl::StartsWith(config->Lookup(key), "Disabled");
+}
+
+BandwidthLimitedCause GetBandwidthLimitedCause(LossBasedState loss_based_state,
+                                               bool is_rtt_above_limit,
+                                               BandwidthUsage bandwidth_usage) {
+  if (bandwidth_usage == BandwidthUsage::kBwOverusing ||
+      bandwidth_usage == BandwidthUsage::kBwUnderusing) {
+    return BandwidthLimitedCause::kDelayBasedLimitedDelayIncreased;
+  } else if (is_rtt_above_limit) {
+    return BandwidthLimitedCause::kRttBasedBackOffHighRtt;
+  }
+  switch (loss_based_state) {
+    case LossBasedState::kDecreasing:
+      // Probes may not be sent in this state.
+      return BandwidthLimitedCause::kLossLimitedBwe;
+    case webrtc::LossBasedState::kIncreaseUsingPadding:
+      // Probes may not be sent in this state.
+      return BandwidthLimitedCause::kLossLimitedBwe;
+    case LossBasedState::kIncreasing:
+      // Probes may be sent in this state.
+      return BandwidthLimitedCause::kLossLimitedBweIncreasing;
+    case LossBasedState::kDelayBasedEstimate:
+      return BandwidthLimitedCause::kDelayBasedLimited;
+  }
+}
+
+}  // namespace
+
+GoogCcNetworkController::GoogCcNetworkController(NetworkControllerConfig config,
+                                                 GoogCcConfig goog_cc_config)
+    : key_value_config_(config.key_value_config ? config.key_value_config
+                                                : &trial_based_config_),
+      event_log_(config.event_log),
+      packet_feedback_only_(goog_cc_config.feedback_only),
+      safe_reset_on_route_change_("Enabled"),
+      safe_reset_acknowledged_rate_("ack"),
+      use_min_allocatable_as_lower_bound_(
+          IsNotDisabled(key_value_config_, "WebRTC-Bwe-MinAllocAsLowerBound")),
+      ignore_probes_lower_than_network_estimate_(IsNotDisabled(
+          key_value_config_,
+          "WebRTC-Bwe-IgnoreProbesLowerThanNetworkStateEstimate")),
+      limit_probes_lower_than_throughput_estimate_(
+          IsNotDisabled(key_value_config_,
+                        "WebRTC-Bwe-LimitProbesLowerThanThroughputEstimate")),
+      rate_control_settings_(
+          RateControlSettings::ParseFromKeyValueConfig(key_value_config_)),
+      pace_at_max_of_bwe_and_lower_link_capacity_(
+          IsEnabled(key_value_config_,
+                    "WebRTC-Bwe-PaceAtMaxOfBweAndLowerLinkCapacity")),
+      probe_controller_(
+          new ProbeController(key_value_config_, config.event_log)),
+      congestion_window_pushback_controller_(
+          rate_control_settings_.UseCongestionWindowPushback()
+              ? std::make_unique<CongestionWindowPushbackController>(
+                    key_value_config_)
+              : nullptr),
+      bandwidth_estimation_(
+          std::make_unique<SendSideBandwidthEstimation>(key_value_config_,
+                                                        event_log_)),
+      alr_detector_(
+          std::make_unique<AlrDetector>(key_value_config_, config.event_log)),
+      probe_bitrate_estimator_(new ProbeBitrateEstimator(config.event_log)),
+      network_estimator_(std::move(goog_cc_config.network_state_estimator)),
+      network_state_predictor_(
+          std::move(goog_cc_config.network_state_predictor)),
+      delay_based_bwe_(new DelayBasedBwe(key_value_config_,
+                                         event_log_,
+                                         network_state_predictor_.get())),
+      acknowledged_bitrate_estimator_(
+          AcknowledgedBitrateEstimatorInterface::Create(key_value_config_)),
+      initial_config_(config),
+      last_loss_based_target_rate_(*config.constraints.starting_rate),
+      last_pushback_target_rate_(last_loss_based_target_rate_),
+      last_stable_target_rate_(last_loss_based_target_rate_),
+      last_loss_base_state_(LossBasedState::kDelayBasedEstimate),
+      pacing_factor_(config.stream_based_config.pacing_factor.value_or(
+          kDefaultPaceMultiplier)),
+      min_total_allocated_bitrate_(
+          config.stream_based_config.min_total_allocated_bitrate.value_or(
+              DataRate::Zero())),
+      max_padding_rate_(config.stream_based_config.max_padding_rate.value_or(
+          DataRate::Zero())) {
+  RTC_DCHECK(config.constraints.at_time.IsFinite());
+  ParseFieldTrial(
+      {&safe_reset_on_route_change_, &safe_reset_acknowledged_rate_},
+      key_value_config_->Lookup("WebRTC-Bwe-SafeResetOnRouteChange"));
+  if (delay_based_bwe_)
+    delay_based_bwe_->SetMinBitrate(kCongestionControllerMinBitrate);
+}
+
+GoogCcNetworkController::~GoogCcNetworkController() {}
+
+NetworkControlUpdate GoogCcNetworkController::OnNetworkAvailability(
+    NetworkAvailability msg) {
+  NetworkControlUpdate update;
+  update.probe_cluster_configs = probe_controller_->OnNetworkAvailability(msg);
+  return update;
+}
+
+NetworkControlUpdate GoogCcNetworkController::OnNetworkRouteChange(
+    NetworkRouteChange msg) {
+  if (safe_reset_on_route_change_) {
+    absl::optional<DataRate> estimated_bitrate;
+    if (safe_reset_acknowledged_rate_) {
+      estimated_bitrate = acknowledged_bitrate_estimator_->bitrate();
+      if (!estimated_bitrate)
+        estimated_bitrate = acknowledged_bitrate_estimator_->PeekRate();
+    } else {
+      estimated_bitrate = bandwidth_estimation_->target_rate();
+    }
+    if (estimated_bitrate) {
+      if (msg.constraints.starting_rate) {
+        msg.constraints.starting_rate =
+            std::min(*msg.constraints.starting_rate, *estimated_bitrate);
+      } else {
+        msg.constraints.starting_rate = estimated_bitrate;
+      }
+    }
+  }
+
+  acknowledged_bitrate_estimator_ =
+      AcknowledgedBitrateEstimatorInterface::Create(key_value_config_);
+  probe_bitrate_estimator_.reset(new ProbeBitrateEstimator(event_log_));
+  if (network_estimator_)
+    network_estimator_->OnRouteChange(msg);
+  delay_based_bwe_.reset(new DelayBasedBwe(key_value_config_, event_log_,
+                                           network_state_predictor_.get()));
+  bandwidth_estimation_->OnRouteChange();
+  probe_controller_->Reset(msg.at_time);
+  NetworkControlUpdate update;
+  update.probe_cluster_configs = ResetConstraints(msg.constraints);
+  MaybeTriggerOnNetworkChanged(&update, msg.at_time);
+  return update;
+}
+
+NetworkControlUpdate GoogCcNetworkController::OnProcessInterval(
+    ProcessInterval msg) {
+  NetworkControlUpdate update;
+  if (initial_config_) {
+    update.probe_cluster_configs =
+        ResetConstraints(initial_config_->constraints);
+    update.pacer_config = GetPacingRates(msg.at_time);
+
+    if (initial_config_->stream_based_config.requests_alr_probing) {
+      probe_controller_->EnablePeriodicAlrProbing(
+          *initial_config_->stream_based_config.requests_alr_probing);
+    }
+    absl::optional<DataRate> total_bitrate =
+        initial_config_->stream_based_config.max_total_allocated_bitrate;
+    if (total_bitrate) {
+      auto probes = probe_controller_->OnMaxTotalAllocatedBitrate(
+          *total_bitrate, msg.at_time);
+      update.probe_cluster_configs.insert(update.probe_cluster_configs.end(),
+                                          probes.begin(), probes.end());
+    }
+    initial_config_.reset();
+  }
+  if (congestion_window_pushback_controller_ && msg.pacer_queue) {
+    congestion_window_pushback_controller_->UpdatePacingQueue(
+        msg.pacer_queue->bytes());
+  }
+  bandwidth_estimation_->UpdateEstimate(msg.at_time);
+  absl::optional<int64_t> start_time_ms =
+      alr_detector_->GetApplicationLimitedRegionStartTime();
+  probe_controller_->SetAlrStartTimeMs(start_time_ms);
+
+  auto probes = probe_controller_->Process(msg.at_time);
+  update.probe_cluster_configs.insert(update.probe_cluster_configs.end(),
+                                      probes.begin(), probes.end());
+
+  if (rate_control_settings_.UseCongestionWindow() &&
+      !feedback_max_rtts_.empty()) {
+    UpdateCongestionWindowSize();
+  }
+  if (congestion_window_pushback_controller_ && current_data_window_) {
+    congestion_window_pushback_controller_->SetDataWindow(
+        *current_data_window_);
+  } else {
+    update.congestion_window = current_data_window_;
+  }
+  MaybeTriggerOnNetworkChanged(&update, msg.at_time);
+  return update;
+}
+
+NetworkControlUpdate GoogCcNetworkController::OnRemoteBitrateReport(
+    RemoteBitrateReport msg) {
+  if (packet_feedback_only_) {
+    RTC_LOG(LS_ERROR) << "Received REMB for packet feedback only GoogCC";
+    return NetworkControlUpdate();
+  }
+  bandwidth_estimation_->UpdateReceiverEstimate(msg.receive_time,
+                                                msg.bandwidth);
+  BWE_TEST_LOGGING_PLOT(1, "REMB_kbps", msg.receive_time.ms(),
+                        msg.bandwidth.bps() / 1000);
+  return NetworkControlUpdate();
+}
+
+NetworkControlUpdate GoogCcNetworkController::OnRoundTripTimeUpdate(
+    RoundTripTimeUpdate msg) {
+  if (packet_feedback_only_ || msg.smoothed)
+    return NetworkControlUpdate();
+  RTC_DCHECK(!msg.round_trip_time.IsZero());
+  if (delay_based_bwe_)
+    delay_based_bwe_->OnRttUpdate(msg.round_trip_time);
+  bandwidth_estimation_->UpdateRtt(msg.round_trip_time, msg.receive_time);
+  return NetworkControlUpdate();
+}
+
+NetworkControlUpdate GoogCcNetworkController::OnSentPacket(
+    SentPacket sent_packet) {
+  alr_detector_->OnBytesSent(sent_packet.size.bytes(),
+                             sent_packet.send_time.ms());
+  acknowledged_bitrate_estimator_->SetAlr(
+      alr_detector_->GetApplicationLimitedRegionStartTime().has_value());
+
+  if (!first_packet_sent_) {
+    first_packet_sent_ = true;
+    // Initialize feedback time to send time to allow estimation of RTT until
+    // first feedback is received.
+    bandwidth_estimation_->UpdatePropagationRtt(sent_packet.send_time,
+                                                TimeDelta::Zero());
+  }
+  bandwidth_estimation_->OnSentPacket(sent_packet);
+
+  if (congestion_window_pushback_controller_) {
+    congestion_window_pushback_controller_->UpdateOutstandingData(
+        sent_packet.data_in_flight.bytes());
+    NetworkControlUpdate update;
+    MaybeTriggerOnNetworkChanged(&update, sent_packet.send_time);
+    return update;
+  } else {
+    return NetworkControlUpdate();
+  }
+}
+
+NetworkControlUpdate GoogCcNetworkController::OnReceivedPacket(
+    ReceivedPacket received_packet) {
+  return NetworkControlUpdate();
+}
+
+NetworkControlUpdate GoogCcNetworkController::OnStreamsConfig(
+    StreamsConfig msg) {
+  NetworkControlUpdate update;
+  if (msg.requests_alr_probing) {
+    probe_controller_->EnablePeriodicAlrProbing(*msg.requests_alr_probing);
+  }
+  if (msg.max_total_allocated_bitrate) {
+    update.probe_cluster_configs =
+        probe_controller_->OnMaxTotalAllocatedBitrate(
+            *msg.max_total_allocated_bitrate, msg.at_time);
+  }
+
+  bool pacing_changed = false;
+  if (msg.pacing_factor && *msg.pacing_factor != pacing_factor_) {
+    pacing_factor_ = *msg.pacing_factor;
+    pacing_changed = true;
+  }
+  if (msg.min_total_allocated_bitrate &&
+      *msg.min_total_allocated_bitrate != min_total_allocated_bitrate_) {
+    min_total_allocated_bitrate_ = *msg.min_total_allocated_bitrate;
+    pacing_changed = true;
+
+    if (use_min_allocatable_as_lower_bound_) {
+      ClampConstraints();
+      delay_based_bwe_->SetMinBitrate(min_data_rate_);
+      bandwidth_estimation_->SetMinMaxBitrate(min_data_rate_, max_data_rate_);
+    }
+  }
+  if (msg.max_padding_rate && *msg.max_padding_rate != max_padding_rate_) {
+    max_padding_rate_ = *msg.max_padding_rate;
+    pacing_changed = true;
+  }
+
+  if (pacing_changed)
+    update.pacer_config = GetPacingRates(msg.at_time);
+  return update;
+}
+
+NetworkControlUpdate GoogCcNetworkController::OnTargetRateConstraints(
+    TargetRateConstraints constraints) {
+  NetworkControlUpdate update;
+  update.probe_cluster_configs = ResetConstraints(constraints);
+  MaybeTriggerOnNetworkChanged(&update, constraints.at_time);
+  return update;
+}
+
+void GoogCcNetworkController::ClampConstraints() {
+  // TODO(holmer): We should make sure the default bitrates are set to 10 kbps,
+  // and that we don't try to set the min bitrate to 0 from any applications.
+  // The congestion controller should allow a min bitrate of 0.
+  min_data_rate_ = std::max(min_target_rate_, kCongestionControllerMinBitrate);
+  if (use_min_allocatable_as_lower_bound_) {
+    min_data_rate_ = std::max(min_data_rate_, min_total_allocated_bitrate_);
+  }
+  if (max_data_rate_ < min_data_rate_) {
+    RTC_LOG(LS_WARNING) << "max bitrate smaller than min bitrate";
+    max_data_rate_ = min_data_rate_;
+  }
+  if (starting_rate_ && starting_rate_ < min_data_rate_) {
+    RTC_LOG(LS_WARNING) << "start bitrate smaller than min bitrate";
+    starting_rate_ = min_data_rate_;
+  }
+}
+
+std::vector<ProbeClusterConfig> GoogCcNetworkController::ResetConstraints(
+    TargetRateConstraints new_constraints) {
+  min_target_rate_ = new_constraints.min_data_rate.value_or(DataRate::Zero());
+  max_data_rate_ =
+      new_constraints.max_data_rate.value_or(DataRate::PlusInfinity());
+  starting_rate_ = new_constraints.starting_rate;
+  ClampConstraints();
+
+  bandwidth_estimation_->SetBitrates(starting_rate_, min_data_rate_,
+                                     max_data_rate_, new_constraints.at_time);
+
+  if (starting_rate_)
+    delay_based_bwe_->SetStartBitrate(*starting_rate_);
+  delay_based_bwe_->SetMinBitrate(min_data_rate_);
+
+  return probe_controller_->SetBitrates(
+      min_data_rate_, starting_rate_.value_or(DataRate::Zero()), max_data_rate_,
+      new_constraints.at_time);
+}
+
+NetworkControlUpdate GoogCcNetworkController::OnTransportLossReport(
+    TransportLossReport msg) {
+  if (packet_feedback_only_)
+    return NetworkControlUpdate();
+  int64_t total_packets_delta =
+      msg.packets_received_delta + msg.packets_lost_delta;
+  bandwidth_estimation_->UpdatePacketsLost(
+      msg.packets_lost_delta, total_packets_delta, msg.receive_time);
+  return NetworkControlUpdate();
+}
+
+void GoogCcNetworkController::UpdateCongestionWindowSize() {
+  TimeDelta min_feedback_max_rtt = TimeDelta::Millis(
+      *std::min_element(feedback_max_rtts_.begin(), feedback_max_rtts_.end()));
+
+  const DataSize kMinCwnd = DataSize::Bytes(2 * 1500);
+  TimeDelta time_window =
+      min_feedback_max_rtt +
+      TimeDelta::Millis(
+          rate_control_settings_.GetCongestionWindowAdditionalTimeMs());
+
+  DataSize data_window = last_loss_based_target_rate_ * time_window;
+  if (current_data_window_) {
+    data_window =
+        std::max(kMinCwnd, (data_window + current_data_window_.value()) / 2);
+  } else {
+    data_window = std::max(kMinCwnd, data_window);
+  }
+  current_data_window_ = data_window;
+}
+
+NetworkControlUpdate GoogCcNetworkController::OnTransportPacketsFeedback(
+    TransportPacketsFeedback report) {
+  if (report.packet_feedbacks.empty()) {
+    // TODO(bugs.webrtc.org/10125): Design a better mechanism to safe-guard
+    // against building very large network queues.
+    return NetworkControlUpdate();
+  }
+
+  if (congestion_window_pushback_controller_) {
+    congestion_window_pushback_controller_->UpdateOutstandingData(
+        report.data_in_flight.bytes());
+  }
+  TimeDelta max_feedback_rtt = TimeDelta::MinusInfinity();
+  TimeDelta min_propagation_rtt = TimeDelta::PlusInfinity();
+  Timestamp max_recv_time = Timestamp::MinusInfinity();
+
+  std::vector<PacketResult> feedbacks = report.ReceivedWithSendInfo();
+  for (const auto& feedback : feedbacks)
+    max_recv_time = std::max(max_recv_time, feedback.receive_time);
+
+  for (const auto& feedback : feedbacks) {
+    TimeDelta feedback_rtt =
+        report.feedback_time - feedback.sent_packet.send_time;
+    TimeDelta min_pending_time = max_recv_time - feedback.receive_time;
+    TimeDelta propagation_rtt = feedback_rtt - min_pending_time;
+    max_feedback_rtt = std::max(max_feedback_rtt, feedback_rtt);
+    min_propagation_rtt = std::min(min_propagation_rtt, propagation_rtt);
+  }
+
+  if (max_feedback_rtt.IsFinite()) {
+    feedback_max_rtts_.push_back(max_feedback_rtt.ms());
+    const size_t kMaxFeedbackRttWindow = 32;
+    if (feedback_max_rtts_.size() > kMaxFeedbackRttWindow)
+      feedback_max_rtts_.pop_front();
+    // TODO(srte): Use time since last unacknowledged packet.
+    bandwidth_estimation_->UpdatePropagationRtt(report.feedback_time,
+                                                min_propagation_rtt);
+  }
+  if (packet_feedback_only_) {
+    if (!feedback_max_rtts_.empty()) {
+      int64_t sum_rtt_ms =
+          std::accumulate(feedback_max_rtts_.begin(), feedback_max_rtts_.end(),
+                          static_cast<int64_t>(0));
+      int64_t mean_rtt_ms = sum_rtt_ms / feedback_max_rtts_.size();
+      if (delay_based_bwe_)
+        delay_based_bwe_->OnRttUpdate(TimeDelta::Millis(mean_rtt_ms));
+    }
+
+    TimeDelta feedback_min_rtt = TimeDelta::PlusInfinity();
+    for (const auto& packet_feedback : feedbacks) {
+      TimeDelta pending_time = max_recv_time - packet_feedback.receive_time;
+      TimeDelta rtt = report.feedback_time -
+                      packet_feedback.sent_packet.send_time - pending_time;
+      // Value used for predicting NACK round trip time in FEC controller.
+      feedback_min_rtt = std::min(rtt, feedback_min_rtt);
+    }
+    if (feedback_min_rtt.IsFinite()) {
+      bandwidth_estimation_->UpdateRtt(feedback_min_rtt, report.feedback_time);
+    }
+
+    expected_packets_since_last_loss_update_ +=
+        report.PacketsWithFeedback().size();
+    for (const auto& packet_feedback : report.PacketsWithFeedback()) {
+      if (!packet_feedback.IsReceived())
+        lost_packets_since_last_loss_update_ += 1;
+    }
+    if (report.feedback_time > next_loss_update_) {
+      next_loss_update_ = report.feedback_time + kLossUpdateInterval;
+      bandwidth_estimation_->UpdatePacketsLost(
+          lost_packets_since_last_loss_update_,
+          expected_packets_since_last_loss_update_, report.feedback_time);
+      expected_packets_since_last_loss_update_ = 0;
+      lost_packets_since_last_loss_update_ = 0;
+    }
+  }
+  absl::optional<int64_t> alr_start_time =
+      alr_detector_->GetApplicationLimitedRegionStartTime();
+
+  if (previously_in_alr_ && !alr_start_time.has_value()) {
+    int64_t now_ms = report.feedback_time.ms();
+    acknowledged_bitrate_estimator_->SetAlrEndedTime(report.feedback_time);
+    probe_controller_->SetAlrEndedTimeMs(now_ms);
+  }
+  previously_in_alr_ = alr_start_time.has_value();
+  acknowledged_bitrate_estimator_->IncomingPacketFeedbackVector(
+      report.SortedByReceiveTime());
+  auto acknowledged_bitrate = acknowledged_bitrate_estimator_->bitrate();
+  bandwidth_estimation_->SetAcknowledgedRate(acknowledged_bitrate,
+                                             report.feedback_time);
+  for (const auto& feedback : report.SortedByReceiveTime()) {
+    if (feedback.sent_packet.pacing_info.probe_cluster_id !=
+        PacedPacketInfo::kNotAProbe) {
+      probe_bitrate_estimator_->HandleProbeAndEstimateBitrate(feedback);
+    }
+  }
+
+  if (network_estimator_) {
+    network_estimator_->OnTransportPacketsFeedback(report);
+    auto prev_estimate = estimate_;
+    estimate_ = network_estimator_->GetCurrentEstimate();
+    // TODO(srte): Make OnTransportPacketsFeedback signal whether the state
+    // changed to avoid the need for this check.
+    if (estimate_ && (!prev_estimate || estimate_->last_feed_time !=
+                                            prev_estimate->last_feed_time)) {
+      event_log_->Log(std::make_unique<RtcEventRemoteEstimate>(
+          estimate_->link_capacity_lower, estimate_->link_capacity_upper));
+      probe_controller_->SetNetworkStateEstimate(*estimate_);
+    }
+  }
+  absl::optional<DataRate> probe_bitrate =
+      probe_bitrate_estimator_->FetchAndResetLastEstimatedBitrate();
+  if (ignore_probes_lower_than_network_estimate_ && probe_bitrate &&
+      estimate_ && *probe_bitrate < delay_based_bwe_->last_estimate() &&
+      *probe_bitrate < estimate_->link_capacity_lower) {
+    probe_bitrate.reset();
+  }
+  if (limit_probes_lower_than_throughput_estimate_ && probe_bitrate &&
+      acknowledged_bitrate) {
+    // Limit the backoff to something slightly below the acknowledged
+    // bitrate. ("Slightly below" because we want to drain the queues
+    // if we are actually overusing.)
+    // The acknowledged bitrate shouldn't normally be higher than the delay
+    // based estimate, but it could happen e.g. due to packet bursts or
+    // encoder overshoot. We use std::min to ensure that a probe result
+    // below the current BWE never causes an increase.
+    DataRate limit =
+        std::min(delay_based_bwe_->last_estimate(),
+                 *acknowledged_bitrate * kProbeDropThroughputFraction);
+    probe_bitrate = std::max(*probe_bitrate, limit);
+  }
+
+  NetworkControlUpdate update;
+  bool recovered_from_overuse = false;
+
+  DelayBasedBwe::Result result;
+  result = delay_based_bwe_->IncomingPacketFeedbackVector(
+      report, acknowledged_bitrate, probe_bitrate, estimate_,
+      alr_start_time.has_value());
+
+  if (result.updated) {
+    if (result.probe) {
+      bandwidth_estimation_->SetSendBitrate(result.target_bitrate,
+                                            report.feedback_time);
+    }
+    // Since SetSendBitrate now resets the delay-based estimate, we have to
+    // call UpdateDelayBasedEstimate after SetSendBitrate.
+    bandwidth_estimation_->UpdateDelayBasedEstimate(report.feedback_time,
+                                                    result.target_bitrate);
+  }
+  bandwidth_estimation_->UpdateLossBasedEstimator(
+      report, result.delay_detector_state, probe_bitrate,
+      alr_start_time.has_value());
+  if (result.updated) {
+    // Update the estimate in the ProbeController, in case we want to probe.
+    MaybeTriggerOnNetworkChanged(&update, report.feedback_time);
+  }
+
+  recovered_from_overuse = result.recovered_from_overuse;
+
+  if (recovered_from_overuse) {
+    probe_controller_->SetAlrStartTimeMs(alr_start_time);
+    auto probes = probe_controller_->RequestProbe(report.feedback_time);
+    update.probe_cluster_configs.insert(update.probe_cluster_configs.end(),
+                                        probes.begin(), probes.end());
+  }
+
+  // No valid RTT could be because send-side BWE isn't used, in which case
+  // we don't try to limit the outstanding packets.
+  if (rate_control_settings_.UseCongestionWindow() &&
+      max_feedback_rtt.IsFinite()) {
+    UpdateCongestionWindowSize();
+  }
+  if (congestion_window_pushback_controller_ && current_data_window_) {
+    congestion_window_pushback_controller_->SetDataWindow(
+        *current_data_window_);
+  } else {
+    update.congestion_window = current_data_window_;
+  }
+
+  return update;
+}
+
+NetworkControlUpdate GoogCcNetworkController::OnNetworkStateEstimate(
+    NetworkStateEstimate msg) {
+  estimate_ = msg;
+  return NetworkControlUpdate();
+}
+
+NetworkControlUpdate GoogCcNetworkController::GetNetworkState(
+    Timestamp at_time) const {
+  NetworkControlUpdate update;
+  update.target_rate = TargetTransferRate();
+  update.target_rate->network_estimate.at_time = at_time;
+  update.target_rate->network_estimate.loss_rate_ratio =
+      last_estimated_fraction_loss_.value_or(0) / 255.0;
+  update.target_rate->network_estimate.round_trip_time =
+      last_estimated_round_trip_time_;
+  update.target_rate->network_estimate.bwe_period =
+      delay_based_bwe_->GetExpectedBwePeriod();
+
+  update.target_rate->at_time = at_time;
+  update.target_rate->target_rate = last_pushback_target_rate_;
+  update.target_rate->stable_target_rate =
+      bandwidth_estimation_->GetEstimatedLinkCapacity();
+  update.pacer_config = GetPacingRates(at_time);
+  update.congestion_window = current_data_window_;
+  return update;
+}
+
+void GoogCcNetworkController::MaybeTriggerOnNetworkChanged(
+    NetworkControlUpdate* update,
+    Timestamp at_time) {
+  uint8_t fraction_loss = bandwidth_estimation_->fraction_loss();
+  TimeDelta round_trip_time = bandwidth_estimation_->round_trip_time();
+  DataRate loss_based_target_rate = bandwidth_estimation_->target_rate();
+  LossBasedState loss_based_state = bandwidth_estimation_->loss_based_state();
+  DataRate pushback_target_rate = loss_based_target_rate;
+
+  BWE_TEST_LOGGING_PLOT(1, "fraction_loss_%", at_time.ms(),
+                        (fraction_loss * 100) / 256);
+  BWE_TEST_LOGGING_PLOT(1, "rtt_ms", at_time.ms(), round_trip_time.ms());
+  BWE_TEST_LOGGING_PLOT(1, "Target_bitrate_kbps", at_time.ms(),
+                        loss_based_target_rate.kbps());
+
+  double cwnd_reduce_ratio = 0.0;
+  if (congestion_window_pushback_controller_) {
+    int64_t pushback_rate =
+        congestion_window_pushback_controller_->UpdateTargetBitrate(
+            loss_based_target_rate.bps());
+    pushback_rate = std::max<int64_t>(bandwidth_estimation_->GetMinBitrate(),
+                                      pushback_rate);
+    pushback_target_rate = DataRate::BitsPerSec(pushback_rate);
+    if (rate_control_settings_.UseCongestionWindowDropFrameOnly()) {
+      cwnd_reduce_ratio = static_cast<double>(loss_based_target_rate.bps() -
+                                              pushback_target_rate.bps()) /
+                          loss_based_target_rate.bps();
+    }
+  }
+  DataRate stable_target_rate =
+      bandwidth_estimation_->GetEstimatedLinkCapacity();
+  stable_target_rate = std::min(stable_target_rate, pushback_target_rate);
+
+  if ((loss_based_target_rate != last_loss_based_target_rate_) ||
+      (loss_based_state != last_loss_base_state_) ||
+      (fraction_loss != last_estimated_fraction_loss_) ||
+      (round_trip_time != last_estimated_round_trip_time_) ||
+      (pushback_target_rate != last_pushback_target_rate_) ||
+      (stable_target_rate != last_stable_target_rate_)) {
+    last_loss_based_target_rate_ = loss_based_target_rate;
+    last_pushback_target_rate_ = pushback_target_rate;
+    last_estimated_fraction_loss_ = fraction_loss;
+    last_estimated_round_trip_time_ = round_trip_time;
+    last_stable_target_rate_ = stable_target_rate;
+    last_loss_base_state_ = loss_based_state;
+
+    alr_detector_->SetEstimatedBitrate(loss_based_target_rate.bps());
+
+    TimeDelta bwe_period = delay_based_bwe_->GetExpectedBwePeriod();
+
+    TargetTransferRate target_rate_msg;
+    target_rate_msg.at_time = at_time;
+    if (rate_control_settings_.UseCongestionWindowDropFrameOnly()) {
+      target_rate_msg.target_rate = loss_based_target_rate;
+      target_rate_msg.cwnd_reduce_ratio = cwnd_reduce_ratio;
+    } else {
+      target_rate_msg.target_rate = pushback_target_rate;
+    }
+    target_rate_msg.stable_target_rate = stable_target_rate;
+    target_rate_msg.network_estimate.at_time = at_time;
+    target_rate_msg.network_estimate.round_trip_time = round_trip_time;
+    target_rate_msg.network_estimate.loss_rate_ratio = fraction_loss / 255.0f;
+    target_rate_msg.network_estimate.bwe_period = bwe_period;
+
+    update->target_rate = target_rate_msg;
+
+    auto probes = probe_controller_->SetEstimatedBitrate(
+        loss_based_target_rate,
+        GetBandwidthLimitedCause(bandwidth_estimation_->loss_based_state(),
+                                 bandwidth_estimation_->IsRttAboveLimit(),
+                                 delay_based_bwe_->last_state()),
+        at_time);
+    update->probe_cluster_configs.insert(update->probe_cluster_configs.end(),
+                                         probes.begin(), probes.end());
+    update->pacer_config = GetPacingRates(at_time);
+    RTC_LOG(LS_VERBOSE) << "bwe " << at_time.ms() << " pushback_target_bps="
+                        << last_pushback_target_rate_.bps()
+                        << " estimate_bps=" << loss_based_target_rate.bps();
+  }
+}
+
+PacerConfig GoogCcNetworkController::GetPacingRates(Timestamp at_time) const {
+  // Pacing rate is based on target rate before congestion window pushback,
+  // because we don't want to build queues in the pacer when pushback occurs.
+  DataRate pacing_rate = DataRate::Zero();
+  if (pace_at_max_of_bwe_and_lower_link_capacity_ && estimate_ &&
+      !bandwidth_estimation_->PaceAtLossBasedEstimate()) {
+    pacing_rate =
+        std::max({min_total_allocated_bitrate_, estimate_->link_capacity_lower,
+                  last_loss_based_target_rate_}) *
+        pacing_factor_;
+  } else {
+    pacing_rate =
+        std::max(min_total_allocated_bitrate_, last_loss_based_target_rate_) *
+        pacing_factor_;
+  }
+  DataRate padding_rate =
+      (last_loss_base_state_ == LossBasedState::kIncreaseUsingPadding)
+          ? std::max(max_padding_rate_, last_loss_based_target_rate_)
+          : max_padding_rate_;
+  padding_rate = std::min(padding_rate, last_pushback_target_rate_);
+  PacerConfig msg;
+  msg.at_time = at_time;
+  msg.time_window = TimeDelta::Seconds(1);
+  msg.data_window = pacing_rate * msg.time_window;
+  msg.pad_window = padding_rate * msg.time_window;
+  return msg;
+}
+
+}  // namespace webrtc

TMessagesProj/jni/voip/webrtc/modules/congestion_controller/goog_cc/goog_cc_network_control.h

@@ -0,0 +1,149 @@
+/*
+ *  Copyright (c) 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 MODULES_CONGESTION_CONTROLLER_GOOG_CC_GOOG_CC_NETWORK_CONTROL_H_
+#define MODULES_CONGESTION_CONTROLLER_GOOG_CC_GOOG_CC_NETWORK_CONTROL_H_
+
+#include <stdint.h>
+
+#include <deque>
+#include <memory>
+#include <vector>
+
+#include "absl/types/optional.h"
+#include "api/field_trials_view.h"
+#include "api/network_state_predictor.h"
+#include "api/rtc_event_log/rtc_event_log.h"
+#include "api/transport/field_trial_based_config.h"
+#include "api/transport/network_control.h"
+#include "api/transport/network_types.h"
+#include "api/units/data_rate.h"
+#include "api/units/data_size.h"
+#include "api/units/time_delta.h"
+#include "api/units/timestamp.h"
+#include "modules/congestion_controller/goog_cc/acknowledged_bitrate_estimator_interface.h"
+#include "modules/congestion_controller/goog_cc/alr_detector.h"
+#include "modules/congestion_controller/goog_cc/congestion_window_pushback_controller.h"
+#include "modules/congestion_controller/goog_cc/delay_based_bwe.h"
+#include "modules/congestion_controller/goog_cc/probe_bitrate_estimator.h"
+#include "modules/congestion_controller/goog_cc/probe_controller.h"
+#include "modules/congestion_controller/goog_cc/send_side_bandwidth_estimation.h"
+#include "rtc_base/experiments/field_trial_parser.h"
+#include "rtc_base/experiments/rate_control_settings.h"
+
+namespace webrtc {
+struct GoogCcConfig {
+  std::unique_ptr<NetworkStateEstimator> network_state_estimator = nullptr;
+  std::unique_ptr<NetworkStatePredictor> network_state_predictor = nullptr;
+  bool feedback_only = false;
+};
+
+class GoogCcNetworkController : public NetworkControllerInterface {
+ public:
+  GoogCcNetworkController(NetworkControllerConfig config,
+                          GoogCcConfig goog_cc_config);
+
+  GoogCcNetworkController() = delete;
+  GoogCcNetworkController(const GoogCcNetworkController&) = delete;
+  GoogCcNetworkController& operator=(const GoogCcNetworkController&) = delete;
+
+  ~GoogCcNetworkController() override;
+
+  // NetworkControllerInterface
+  NetworkControlUpdate OnNetworkAvailability(NetworkAvailability msg) override;
+  NetworkControlUpdate OnNetworkRouteChange(NetworkRouteChange msg) override;
+  NetworkControlUpdate OnProcessInterval(ProcessInterval msg) override;
+  NetworkControlUpdate OnRemoteBitrateReport(RemoteBitrateReport msg) override;
+  NetworkControlUpdate OnRoundTripTimeUpdate(RoundTripTimeUpdate msg) override;
+  NetworkControlUpdate OnSentPacket(SentPacket msg) override;
+  NetworkControlUpdate OnReceivedPacket(ReceivedPacket msg) override;
+  NetworkControlUpdate OnStreamsConfig(StreamsConfig msg) override;
+  NetworkControlUpdate OnTargetRateConstraints(
+      TargetRateConstraints msg) override;
+  NetworkControlUpdate OnTransportLossReport(TransportLossReport msg) override;
+  NetworkControlUpdate OnTransportPacketsFeedback(
+      TransportPacketsFeedback msg) override;
+  NetworkControlUpdate OnNetworkStateEstimate(
+      NetworkStateEstimate msg) override;
+
+  NetworkControlUpdate GetNetworkState(Timestamp at_time) const;
+
+ private:
+  friend class GoogCcStatePrinter;
+  std::vector<ProbeClusterConfig> ResetConstraints(
+      TargetRateConstraints new_constraints);
+  void ClampConstraints();
+  void MaybeTriggerOnNetworkChanged(NetworkControlUpdate* update,
+                                    Timestamp at_time);
+  void UpdateCongestionWindowSize();
+  PacerConfig GetPacingRates(Timestamp at_time) const;
+  const FieldTrialBasedConfig trial_based_config_;
+
+  const FieldTrialsView* const key_value_config_;
+  RtcEventLog* const event_log_;
+  const bool packet_feedback_only_;
+  FieldTrialFlag safe_reset_on_route_change_;
+  FieldTrialFlag safe_reset_acknowledged_rate_;
+  const bool use_min_allocatable_as_lower_bound_;
+  const bool ignore_probes_lower_than_network_estimate_;
+  const bool limit_probes_lower_than_throughput_estimate_;
+  const RateControlSettings rate_control_settings_;
+  const bool pace_at_max_of_bwe_and_lower_link_capacity_;
+
+  const std::unique_ptr<ProbeController> probe_controller_;
+  const std::unique_ptr<CongestionWindowPushbackController>
+      congestion_window_pushback_controller_;
+
+  std::unique_ptr<SendSideBandwidthEstimation> bandwidth_estimation_;
+  std::unique_ptr<AlrDetector> alr_detector_;
+  std::unique_ptr<ProbeBitrateEstimator> probe_bitrate_estimator_;
+  std::unique_ptr<NetworkStateEstimator> network_estimator_;
+  std::unique_ptr<NetworkStatePredictor> network_state_predictor_;
+  std::unique_ptr<DelayBasedBwe> delay_based_bwe_;
+  std::unique_ptr<AcknowledgedBitrateEstimatorInterface>
+      acknowledged_bitrate_estimator_;
+
+  absl::optional<NetworkControllerConfig> initial_config_;
+
+  DataRate min_target_rate_ = DataRate::Zero();
+  DataRate min_data_rate_ = DataRate::Zero();
+  DataRate max_data_rate_ = DataRate::PlusInfinity();
+  absl::optional<DataRate> starting_rate_;
+
+  bool first_packet_sent_ = false;
+
+  absl::optional<NetworkStateEstimate> estimate_;
+
+  Timestamp next_loss_update_ = Timestamp::MinusInfinity();
+  int lost_packets_since_last_loss_update_ = 0;
+  int expected_packets_since_last_loss_update_ = 0;
+
+  std::deque<int64_t> feedback_max_rtts_;
+
+  DataRate last_loss_based_target_rate_;
+  DataRate last_pushback_target_rate_;
+  DataRate last_stable_target_rate_;
+  LossBasedState last_loss_base_state_;
+
+  absl::optional<uint8_t> last_estimated_fraction_loss_ = 0;
+  TimeDelta last_estimated_round_trip_time_ = TimeDelta::PlusInfinity();
+
+  double pacing_factor_;
+  DataRate min_total_allocated_bitrate_;
+  DataRate max_padding_rate_;
+
+  bool previously_in_alr_ = false;
+
+  absl::optional<DataSize> current_data_window_;
+};
+
+}  // namespace webrtc
+
+#endif  // MODULES_CONGESTION_CONTROLLER_GOOG_CC_GOOG_CC_NETWORK_CONTROL_H_

TMessagesProj/jni/voip/webrtc/modules/congestion_controller/goog_cc/inter_arrival_delta.cc

@@ -0,0 +1,142 @@
+/*
+ *  Copyright (c) 2020 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 "modules/congestion_controller/goog_cc/inter_arrival_delta.h"
+
+#include <algorithm>
+#include <cstddef>
+
+#include "api/units/time_delta.h"
+#include "api/units/timestamp.h"
+#include "rtc_base/checks.h"
+#include "rtc_base/logging.h"
+
+namespace webrtc {
+
+static constexpr TimeDelta kBurstDeltaThreshold = TimeDelta::Millis(5);
+static constexpr TimeDelta kMaxBurstDuration = TimeDelta::Millis(100);
+constexpr TimeDelta InterArrivalDelta::kArrivalTimeOffsetThreshold;
+
+InterArrivalDelta::InterArrivalDelta(TimeDelta send_time_group_length)
+    : send_time_group_length_(send_time_group_length),
+      current_timestamp_group_(),
+      prev_timestamp_group_(),
+      num_consecutive_reordered_packets_(0) {}
+
+bool InterArrivalDelta::ComputeDeltas(Timestamp send_time,
+                                      Timestamp arrival_time,
+                                      Timestamp system_time,
+                                      size_t packet_size,
+                                      TimeDelta* send_time_delta,
+                                      TimeDelta* arrival_time_delta,
+                                      int* packet_size_delta) {
+  bool calculated_deltas = false;
+  if (current_timestamp_group_.IsFirstPacket()) {
+    // We don't have enough data to update the filter, so we store it until we
+    // have two frames of data to process.
+    current_timestamp_group_.send_time = send_time;
+    current_timestamp_group_.first_send_time = send_time;
+    current_timestamp_group_.first_arrival = arrival_time;
+  } else if (current_timestamp_group_.first_send_time > send_time) {
+    // Reordered packet.
+    return false;
+  } else if (NewTimestampGroup(arrival_time, send_time)) {
+    // First packet of a later send burst, the previous packets sample is ready.
+    if (prev_timestamp_group_.complete_time.IsFinite()) {
+      *send_time_delta =
+          current_timestamp_group_.send_time - prev_timestamp_group_.send_time;
+      *arrival_time_delta = current_timestamp_group_.complete_time -
+                            prev_timestamp_group_.complete_time;
+
+      TimeDelta system_time_delta = current_timestamp_group_.last_system_time -
+                                    prev_timestamp_group_.last_system_time;
+
+      if (*arrival_time_delta - system_time_delta >=
+          kArrivalTimeOffsetThreshold) {
+        RTC_LOG(LS_WARNING)
+            << "The arrival time clock offset has changed (diff = "
+            << arrival_time_delta->ms() - system_time_delta.ms()
+            << " ms), resetting.";
+        Reset();
+        return false;
+      }
+      if (*arrival_time_delta < TimeDelta::Zero()) {
+        // The group of packets has been reordered since receiving its local
+        // arrival timestamp.
+        ++num_consecutive_reordered_packets_;
+        if (num_consecutive_reordered_packets_ >= kReorderedResetThreshold) {
+          RTC_LOG(LS_WARNING)
+              << "Packets between send burst arrived out of order, resetting:"
+              << " arrival_time_delta_ms=" << arrival_time_delta->ms()
+              << ", send_time_delta_ms=" << send_time_delta->ms();
+          Reset();
+        }
+        return false;
+      } else {
+        num_consecutive_reordered_packets_ = 0;
+      }
+      *packet_size_delta = static_cast<int>(current_timestamp_group_.size) -
+                           static_cast<int>(prev_timestamp_group_.size);
+      calculated_deltas = true;
+    }
+    prev_timestamp_group_ = current_timestamp_group_;
+    // The new timestamp is now the current frame.
+    current_timestamp_group_.first_send_time = send_time;
+    current_timestamp_group_.send_time = send_time;
+    current_timestamp_group_.first_arrival = arrival_time;
+    current_timestamp_group_.size = 0;
+  } else {
+    current_timestamp_group_.send_time =
+        std::max(current_timestamp_group_.send_time, send_time);
+  }
+  // Accumulate the frame size.
+  current_timestamp_group_.size += packet_size;
+  current_timestamp_group_.complete_time = arrival_time;
+  current_timestamp_group_.last_system_time = system_time;
+
+  return calculated_deltas;
+}
+
+// Assumes that `timestamp` is not reordered compared to
+// `current_timestamp_group_`.
+bool InterArrivalDelta::NewTimestampGroup(Timestamp arrival_time,
+                                          Timestamp send_time) const {
+  if (current_timestamp_group_.IsFirstPacket()) {
+    return false;
+  } else if (BelongsToBurst(arrival_time, send_time)) {
+    return false;
+  } else {
+    return send_time - current_timestamp_group_.first_send_time >
+           send_time_group_length_;
+  }
+}
+
+bool InterArrivalDelta::BelongsToBurst(Timestamp arrival_time,
+                                       Timestamp send_time) const {
+  RTC_DCHECK(current_timestamp_group_.complete_time.IsFinite());
+  TimeDelta arrival_time_delta =
+      arrival_time - current_timestamp_group_.complete_time;
+  TimeDelta send_time_delta = send_time - current_timestamp_group_.send_time;
+  if (send_time_delta.IsZero())
+    return true;
+  TimeDelta propagation_delta = arrival_time_delta - send_time_delta;
+  if (propagation_delta < TimeDelta::Zero() &&
+      arrival_time_delta <= kBurstDeltaThreshold &&
+      arrival_time - current_timestamp_group_.first_arrival < kMaxBurstDuration)
+    return true;
+  return false;
+}
+
+void InterArrivalDelta::Reset() {
+  num_consecutive_reordered_packets_ = 0;
+  current_timestamp_group_ = SendTimeGroup();
+  prev_timestamp_group_ = SendTimeGroup();
+}
+}  // namespace webrtc

TMessagesProj/jni/voip/webrtc/modules/congestion_controller/goog_cc/inter_arrival_delta.h

@@ -0,0 +1,92 @@
+/*
+ *  Copyright (c) 2020 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_CONGESTION_CONTROLLER_GOOG_CC_INTER_ARRIVAL_DELTA_H_
+#define MODULES_CONGESTION_CONTROLLER_GOOG_CC_INTER_ARRIVAL_DELTA_H_
+
+#include <cstddef>
+
+#include "api/units/time_delta.h"
+#include "api/units/timestamp.h"
+
+namespace webrtc {
+
+// Helper class to compute the inter-arrival time delta and the size delta
+// between two send bursts. This code is branched from
+// modules/remote_bitrate_estimator/inter_arrival.
+class InterArrivalDelta {
+ public:
+  // After this many packet groups received out of order InterArrival will
+  // reset, assuming that clocks have made a jump.
+  static constexpr int kReorderedResetThreshold = 3;
+  static constexpr TimeDelta kArrivalTimeOffsetThreshold =
+      TimeDelta::Seconds(3);
+
+  // A send time group is defined as all packets with a send time which are at
+  // most send_time_group_length older than the first timestamp in that
+  // group.
+  explicit InterArrivalDelta(TimeDelta send_time_group_length);
+
+  InterArrivalDelta() = delete;
+  InterArrivalDelta(const InterArrivalDelta&) = delete;
+  InterArrivalDelta& operator=(const InterArrivalDelta&) = delete;
+
+  // This function returns true if a delta was computed, or false if the current
+  // group is still incomplete or if only one group has been completed.
+  // `send_time` is the send time.
+  // `arrival_time` is the time at which the packet arrived.
+  // `packet_size` is the size of the packet.
+  // `timestamp_delta` (output) is the computed send time delta.
+  // `arrival_time_delta` (output) is the computed arrival-time delta.
+  // `packet_size_delta` (output) is the computed size delta.
+  bool ComputeDeltas(Timestamp send_time,
+                     Timestamp arrival_time,
+                     Timestamp system_time,
+                     size_t packet_size,
+                     TimeDelta* send_time_delta,
+                     TimeDelta* arrival_time_delta,
+                     int* packet_size_delta);
+
+ private:
+  struct SendTimeGroup {
+    SendTimeGroup()
+        : size(0),
+          first_send_time(Timestamp::MinusInfinity()),
+          send_time(Timestamp::MinusInfinity()),
+          first_arrival(Timestamp::MinusInfinity()),
+          complete_time(Timestamp::MinusInfinity()),
+          last_system_time(Timestamp::MinusInfinity()) {}
+
+    bool IsFirstPacket() const { return complete_time.IsInfinite(); }
+
+    size_t size;
+    Timestamp first_send_time;
+    Timestamp send_time;
+    Timestamp first_arrival;
+    Timestamp complete_time;
+    Timestamp last_system_time;
+  };
+
+  // Returns true if the last packet was the end of the current batch and the
+  // packet with `send_time` is the first of a new batch.
+  bool NewTimestampGroup(Timestamp arrival_time, Timestamp send_time) const;
+
+  bool BelongsToBurst(Timestamp arrival_time, Timestamp send_time) const;
+
+  void Reset();
+
+  const TimeDelta send_time_group_length_;
+  SendTimeGroup current_timestamp_group_;
+  SendTimeGroup prev_timestamp_group_;
+  int num_consecutive_reordered_packets_;
+};
+}  // namespace webrtc
+
+#endif  // MODULES_CONGESTION_CONTROLLER_GOOG_CC_INTER_ARRIVAL_DELTA_H_

TMessagesProj/jni/voip/webrtc/modules/congestion_controller/goog_cc/link_capacity_estimator.cc

@@ -0,0 +1,79 @@
+/*
+ *  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.
+ */
+#include "modules/congestion_controller/goog_cc/link_capacity_estimator.h"
+
+#include <algorithm>
+#include <cmath>
+
+#include "api/units/data_rate.h"
+#include "rtc_base/numerics/safe_minmax.h"
+
+namespace webrtc {
+LinkCapacityEstimator::LinkCapacityEstimator() {}
+
+DataRate LinkCapacityEstimator::UpperBound() const {
+  if (estimate_kbps_.has_value())
+    return DataRate::KilobitsPerSec(estimate_kbps_.value() +
+                                    3 * deviation_estimate_kbps());
+  return DataRate::Infinity();
+}
+
+DataRate LinkCapacityEstimator::LowerBound() const {
+  if (estimate_kbps_.has_value())
+    return DataRate::KilobitsPerSec(
+        std::max(0.0, estimate_kbps_.value() - 3 * deviation_estimate_kbps()));
+  return DataRate::Zero();
+}
+
+void LinkCapacityEstimator::Reset() {
+  estimate_kbps_.reset();
+}
+
+void LinkCapacityEstimator::OnOveruseDetected(DataRate acknowledged_rate) {
+  Update(acknowledged_rate, 0.05);
+}
+
+void LinkCapacityEstimator::OnProbeRate(DataRate probe_rate) {
+  Update(probe_rate, 0.5);
+}
+
+void LinkCapacityEstimator::Update(DataRate capacity_sample, double alpha) {
+  double sample_kbps = capacity_sample.kbps();
+  if (!estimate_kbps_.has_value()) {
+    estimate_kbps_ = sample_kbps;
+  } else {
+    estimate_kbps_ = (1 - alpha) * estimate_kbps_.value() + alpha * sample_kbps;
+  }
+  // Estimate the variance of the link capacity estimate and normalize the
+  // variance with the link capacity estimate.
+  const double norm = std::max(estimate_kbps_.value(), 1.0);
+  double error_kbps = estimate_kbps_.value() - sample_kbps;
+  deviation_kbps_ =
+      (1 - alpha) * deviation_kbps_ + alpha * error_kbps * error_kbps / norm;
+  // 0.4 ~= 14 kbit/s at 500 kbit/s
+  // 2.5f ~= 35 kbit/s at 500 kbit/s
+  deviation_kbps_ = rtc::SafeClamp(deviation_kbps_, 0.4f, 2.5f);
+}
+
+bool LinkCapacityEstimator::has_estimate() const {
+  return estimate_kbps_.has_value();
+}
+
+DataRate LinkCapacityEstimator::estimate() const {
+  return DataRate::KilobitsPerSec(*estimate_kbps_);
+}
+
+double LinkCapacityEstimator::deviation_estimate_kbps() const {
+  // Calculate the max bit rate std dev given the normalized
+  // variance and the current throughput bitrate. The standard deviation will
+  // only be used if estimate_kbps_ has a value.
+  return sqrt(deviation_kbps_ * estimate_kbps_.value());
+}
+}  // namespace webrtc

TMessagesProj/jni/voip/webrtc/modules/congestion_controller/goog_cc/link_capacity_estimator.h

@@ -0,0 +1,38 @@
+/*
+ *  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 MODULES_CONGESTION_CONTROLLER_GOOG_CC_LINK_CAPACITY_ESTIMATOR_H_
+#define MODULES_CONGESTION_CONTROLLER_GOOG_CC_LINK_CAPACITY_ESTIMATOR_H_
+
+#include "absl/types/optional.h"
+#include "api/units/data_rate.h"
+
+namespace webrtc {
+class LinkCapacityEstimator {
+ public:
+  LinkCapacityEstimator();
+  DataRate UpperBound() const;
+  DataRate LowerBound() const;
+  void Reset();
+  void OnOveruseDetected(DataRate acknowledged_rate);
+  void OnProbeRate(DataRate probe_rate);
+  bool has_estimate() const;
+  DataRate estimate() const;
+
+ private:
+  friend class GoogCcStatePrinter;
+  void Update(DataRate capacity_sample, double alpha);
+
+  double deviation_estimate_kbps() const;
+  absl::optional<double> estimate_kbps_;
+  double deviation_kbps_ = 0.4;
+};
+}  // namespace webrtc
+
+#endif  // MODULES_CONGESTION_CONTROLLER_GOOG_CC_LINK_CAPACITY_ESTIMATOR_H_

TMessagesProj/jni/voip/webrtc/modules/congestion_controller/goog_cc/loss_based_bandwidth_estimation.cc

@@ -0,0 +1,265 @@
+/*
+ *  Copyright (c) 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.
+ */
+
+#include "modules/congestion_controller/goog_cc/loss_based_bandwidth_estimation.h"
+
+#include <algorithm>
+#include <vector>
+
+#include "absl/strings/match.h"
+#include "absl/strings/string_view.h"
+#include "api/field_trials_view.h"
+#include "api/transport/network_types.h"
+#include "api/units/data_rate.h"
+#include "api/units/time_delta.h"
+#include "api/units/timestamp.h"
+#include "rtc_base/checks.h"
+#include "rtc_base/experiments/field_trial_parser.h"
+
+namespace webrtc {
+namespace {
+const char kBweLossBasedControl[] = "WebRTC-Bwe-LossBasedControl";
+
+// Expecting RTCP feedback to be sent with roughly 1s intervals, a 5s gap
+// indicates a channel outage.
+constexpr TimeDelta kMaxRtcpFeedbackInterval = TimeDelta::Millis(5000);
+
+// Increase slower when RTT is high.
+double GetIncreaseFactor(const LossBasedControlConfig& config, TimeDelta rtt) {
+  // Clamp the RTT
+  if (rtt < config.increase_low_rtt) {
+    rtt = config.increase_low_rtt;
+  } else if (rtt > config.increase_high_rtt) {
+    rtt = config.increase_high_rtt;
+  }
+  auto rtt_range = config.increase_high_rtt.Get() - config.increase_low_rtt;
+  if (rtt_range <= TimeDelta::Zero()) {
+    RTC_DCHECK_NOTREACHED();  // Only on misconfiguration.
+    return config.min_increase_factor;
+  }
+  auto rtt_offset = rtt - config.increase_low_rtt;
+  auto relative_offset = std::max(0.0, std::min(rtt_offset / rtt_range, 1.0));
+  auto factor_range = config.max_increase_factor - config.min_increase_factor;
+  return config.min_increase_factor + (1 - relative_offset) * factor_range;
+}
+
+double LossFromBitrate(DataRate bitrate,
+                       DataRate loss_bandwidth_balance,
+                       double exponent) {
+  if (loss_bandwidth_balance >= bitrate)
+    return 1.0;
+  return pow(loss_bandwidth_balance / bitrate, exponent);
+}
+
+DataRate BitrateFromLoss(double loss,
+                         DataRate loss_bandwidth_balance,
+                         double exponent) {
+  if (exponent <= 0) {
+    RTC_DCHECK_NOTREACHED();
+    return DataRate::Infinity();
+  }
+  if (loss < 1e-5)
+    return DataRate::Infinity();
+  return loss_bandwidth_balance * pow(loss, -1.0 / exponent);
+}
+
+double ExponentialUpdate(TimeDelta window, TimeDelta interval) {
+  // Use the convention that exponential window length (which is really
+  // infinite) is the time it takes to dampen to 1/e.
+  if (window <= TimeDelta::Zero()) {
+    RTC_DCHECK_NOTREACHED();
+    return 1.0f;
+  }
+  return 1.0f - exp(interval / window * -1.0);
+}
+
+bool IsEnabled(const webrtc::FieldTrialsView& key_value_config,
+               absl::string_view name) {
+  return absl::StartsWith(key_value_config.Lookup(name), "Enabled");
+}
+
+}  // namespace
+
+LossBasedControlConfig::LossBasedControlConfig(
+    const FieldTrialsView* key_value_config)
+    : enabled(IsEnabled(*key_value_config, kBweLossBasedControl)),
+      min_increase_factor("min_incr", 1.02),
+      max_increase_factor("max_incr", 1.08),
+      increase_low_rtt("incr_low_rtt", TimeDelta::Millis(200)),
+      increase_high_rtt("incr_high_rtt", TimeDelta::Millis(800)),
+      decrease_factor("decr", 0.99),
+      loss_window("loss_win", TimeDelta::Millis(800)),
+      loss_max_window("loss_max_win", TimeDelta::Millis(800)),
+      acknowledged_rate_max_window("ackrate_max_win", TimeDelta::Millis(800)),
+      increase_offset("incr_offset", DataRate::BitsPerSec(1000)),
+      loss_bandwidth_balance_increase("balance_incr",
+                                      DataRate::KilobitsPerSec(0.5)),
+      loss_bandwidth_balance_decrease("balance_decr",
+                                      DataRate::KilobitsPerSec(4)),
+      loss_bandwidth_balance_reset("balance_reset",
+                                   DataRate::KilobitsPerSec(0.1)),
+      loss_bandwidth_balance_exponent("exponent", 0.5),
+      allow_resets("resets", false),
+      decrease_interval("decr_intvl", TimeDelta::Millis(300)),
+      loss_report_timeout("timeout", TimeDelta::Millis(6000)) {
+  ParseFieldTrial(
+      {&min_increase_factor, &max_increase_factor, &increase_low_rtt,
+       &increase_high_rtt, &decrease_factor, &loss_window, &loss_max_window,
+       &acknowledged_rate_max_window, &increase_offset,
+       &loss_bandwidth_balance_increase, &loss_bandwidth_balance_decrease,
+       &loss_bandwidth_balance_reset, &loss_bandwidth_balance_exponent,
+       &allow_resets, &decrease_interval, &loss_report_timeout},
+      key_value_config->Lookup(kBweLossBasedControl));
+}
+LossBasedControlConfig::LossBasedControlConfig(const LossBasedControlConfig&) =
+    default;
+LossBasedControlConfig::~LossBasedControlConfig() = default;
+
+LossBasedBandwidthEstimation::LossBasedBandwidthEstimation(
+    const FieldTrialsView* key_value_config)
+    : config_(key_value_config),
+      average_loss_(0),
+      average_loss_max_(0),
+      loss_based_bitrate_(DataRate::Zero()),
+      acknowledged_bitrate_max_(DataRate::Zero()),
+      acknowledged_bitrate_last_update_(Timestamp::MinusInfinity()),
+      time_last_decrease_(Timestamp::MinusInfinity()),
+      has_decreased_since_last_loss_report_(false),
+      last_loss_packet_report_(Timestamp::MinusInfinity()),
+      last_loss_ratio_(0) {}
+
+void LossBasedBandwidthEstimation::UpdateLossStatistics(
+    const std::vector<PacketResult>& packet_results,
+    Timestamp at_time) {
+  if (packet_results.empty()) {
+    RTC_DCHECK_NOTREACHED();
+    return;
+  }
+  int loss_count = 0;
+  for (const auto& pkt : packet_results) {
+    loss_count += !pkt.IsReceived() ? 1 : 0;
+  }
+  last_loss_ratio_ = static_cast<double>(loss_count) / packet_results.size();
+  const TimeDelta time_passed = last_loss_packet_report_.IsFinite()
+                                    ? at_time - last_loss_packet_report_
+                                    : TimeDelta::Seconds(1);
+  last_loss_packet_report_ = at_time;
+  has_decreased_since_last_loss_report_ = false;
+
+  average_loss_ += ExponentialUpdate(config_.loss_window, time_passed) *
+                   (last_loss_ratio_ - average_loss_);
+  if (average_loss_ > average_loss_max_) {
+    average_loss_max_ = average_loss_;
+  } else {
+    average_loss_max_ +=
+        ExponentialUpdate(config_.loss_max_window, time_passed) *
+        (average_loss_ - average_loss_max_);
+  }
+}
+
+void LossBasedBandwidthEstimation::UpdateAcknowledgedBitrate(
+    DataRate acknowledged_bitrate,
+    Timestamp at_time) {
+  const TimeDelta time_passed =
+      acknowledged_bitrate_last_update_.IsFinite()
+          ? at_time - acknowledged_bitrate_last_update_
+          : TimeDelta::Seconds(1);
+  acknowledged_bitrate_last_update_ = at_time;
+  if (acknowledged_bitrate > acknowledged_bitrate_max_) {
+    acknowledged_bitrate_max_ = acknowledged_bitrate;
+  } else {
+    acknowledged_bitrate_max_ -=
+        ExponentialUpdate(config_.acknowledged_rate_max_window, time_passed) *
+        (acknowledged_bitrate_max_ - acknowledged_bitrate);
+  }
+}
+
+DataRate LossBasedBandwidthEstimation::Update(Timestamp at_time,
+                                              DataRate min_bitrate,
+                                              DataRate wanted_bitrate,
+                                              TimeDelta last_round_trip_time) {
+  if (loss_based_bitrate_.IsZero()) {
+    loss_based_bitrate_ = wanted_bitrate;
+  }
+
+  // Only increase if loss has been low for some time.
+  const double loss_estimate_for_increase = average_loss_max_;
+  // Avoid multiple decreases from averaging over one loss spike.
+  const double loss_estimate_for_decrease =
+      std::min(average_loss_, last_loss_ratio_);
+  const bool allow_decrease =
+      !has_decreased_since_last_loss_report_ &&
+      (at_time - time_last_decrease_ >=
+       last_round_trip_time + config_.decrease_interval);
+  // If packet lost reports are too old, dont increase bitrate.
+  const bool loss_report_valid =
+      at_time - last_loss_packet_report_ < 1.2 * kMaxRtcpFeedbackInterval;
+
+  if (loss_report_valid && config_.allow_resets &&
+      loss_estimate_for_increase < loss_reset_threshold()) {
+    loss_based_bitrate_ = wanted_bitrate;
+  } else if (loss_report_valid &&
+             loss_estimate_for_increase < loss_increase_threshold()) {
+    // Increase bitrate by RTT-adaptive ratio.
+    DataRate new_increased_bitrate =
+        min_bitrate * GetIncreaseFactor(config_, last_round_trip_time) +
+        config_.increase_offset;
+    // The bitrate that would make the loss "just high enough".
+    const DataRate new_increased_bitrate_cap = BitrateFromLoss(
+        loss_estimate_for_increase, config_.loss_bandwidth_balance_increase,
+        config_.loss_bandwidth_balance_exponent);
+    new_increased_bitrate =
+        std::min(new_increased_bitrate, new_increased_bitrate_cap);
+    loss_based_bitrate_ = std::max(new_increased_bitrate, loss_based_bitrate_);
+  } else if (loss_estimate_for_decrease > loss_decrease_threshold() &&
+             allow_decrease) {
+    // The bitrate that would make the loss "just acceptable".
+    const DataRate new_decreased_bitrate_floor = BitrateFromLoss(
+        loss_estimate_for_decrease, config_.loss_bandwidth_balance_decrease,
+        config_.loss_bandwidth_balance_exponent);
+    DataRate new_decreased_bitrate =
+        std::max(decreased_bitrate(), new_decreased_bitrate_floor);
+    if (new_decreased_bitrate < loss_based_bitrate_) {
+      time_last_decrease_ = at_time;
+      has_decreased_since_last_loss_report_ = true;
+      loss_based_bitrate_ = new_decreased_bitrate;
+    }
+  }
+  return loss_based_bitrate_;
+}
+
+void LossBasedBandwidthEstimation::Initialize(DataRate bitrate) {
+  loss_based_bitrate_ = bitrate;
+  average_loss_ = 0;
+  average_loss_max_ = 0;
+}
+
+double LossBasedBandwidthEstimation::loss_reset_threshold() const {
+  return LossFromBitrate(loss_based_bitrate_,
+                         config_.loss_bandwidth_balance_reset,
+                         config_.loss_bandwidth_balance_exponent);
+}
+
+double LossBasedBandwidthEstimation::loss_increase_threshold() const {
+  return LossFromBitrate(loss_based_bitrate_,
+                         config_.loss_bandwidth_balance_increase,
+                         config_.loss_bandwidth_balance_exponent);
+}
+
+double LossBasedBandwidthEstimation::loss_decrease_threshold() const {
+  return LossFromBitrate(loss_based_bitrate_,
+                         config_.loss_bandwidth_balance_decrease,
+                         config_.loss_bandwidth_balance_exponent);
+}
+
+DataRate LossBasedBandwidthEstimation::decreased_bitrate() const {
+  return config_.decrease_factor * acknowledged_bitrate_max_;
+}
+}  // namespace webrtc

TMessagesProj/jni/voip/webrtc/modules/congestion_controller/goog_cc/loss_based_bandwidth_estimation.h

@@ -0,0 +1,97 @@
+/*
+ *  Copyright (c) 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 MODULES_CONGESTION_CONTROLLER_GOOG_CC_LOSS_BASED_BANDWIDTH_ESTIMATION_H_
+#define MODULES_CONGESTION_CONTROLLER_GOOG_CC_LOSS_BASED_BANDWIDTH_ESTIMATION_H_
+
+#include <vector>
+
+#include "api/field_trials_view.h"
+#include "api/transport/network_types.h"
+#include "api/units/data_rate.h"
+#include "api/units/time_delta.h"
+#include "api/units/timestamp.h"
+#include "rtc_base/experiments/field_trial_parser.h"
+
+namespace webrtc {
+
+struct LossBasedControlConfig {
+  explicit LossBasedControlConfig(const FieldTrialsView* key_value_config);
+  LossBasedControlConfig(const LossBasedControlConfig&);
+  LossBasedControlConfig& operator=(const LossBasedControlConfig&) = default;
+  ~LossBasedControlConfig();
+  bool enabled;
+  FieldTrialParameter<double> min_increase_factor;
+  FieldTrialParameter<double> max_increase_factor;
+  FieldTrialParameter<TimeDelta> increase_low_rtt;
+  FieldTrialParameter<TimeDelta> increase_high_rtt;
+  FieldTrialParameter<double> decrease_factor;
+  FieldTrialParameter<TimeDelta> loss_window;
+  FieldTrialParameter<TimeDelta> loss_max_window;
+  FieldTrialParameter<TimeDelta> acknowledged_rate_max_window;
+  FieldTrialParameter<DataRate> increase_offset;
+  FieldTrialParameter<DataRate> loss_bandwidth_balance_increase;
+  FieldTrialParameter<DataRate> loss_bandwidth_balance_decrease;
+  FieldTrialParameter<DataRate> loss_bandwidth_balance_reset;
+  FieldTrialParameter<double> loss_bandwidth_balance_exponent;
+  FieldTrialParameter<bool> allow_resets;
+  FieldTrialParameter<TimeDelta> decrease_interval;
+  FieldTrialParameter<TimeDelta> loss_report_timeout;
+};
+
+// Estimates an upper BWE limit based on loss.
+// It requires knowledge about lost packets and acknowledged bitrate.
+// Ie, this class require transport feedback.
+class LossBasedBandwidthEstimation {
+ public:
+  explicit LossBasedBandwidthEstimation(
+      const FieldTrialsView* key_value_config);
+  // Returns the new estimate.
+  DataRate Update(Timestamp at_time,
+                  DataRate min_bitrate,
+                  DataRate wanted_bitrate,
+                  TimeDelta last_round_trip_time);
+  void UpdateAcknowledgedBitrate(DataRate acknowledged_bitrate,
+                                 Timestamp at_time);
+  void Initialize(DataRate bitrate);
+  bool Enabled() const { return config_.enabled; }
+  // Returns true if LossBasedBandwidthEstimation is enabled and have
+  // received loss statistics. Ie, this class require transport feedback.
+  bool InUse() const {
+    return Enabled() && last_loss_packet_report_.IsFinite();
+  }
+  void UpdateLossStatistics(const std::vector<PacketResult>& packet_results,
+                            Timestamp at_time);
+  DataRate GetEstimate() const { return loss_based_bitrate_; }
+
+ private:
+  friend class GoogCcStatePrinter;
+  void Reset(DataRate bitrate);
+  double loss_increase_threshold() const;
+  double loss_decrease_threshold() const;
+  double loss_reset_threshold() const;
+
+  DataRate decreased_bitrate() const;
+
+  const LossBasedControlConfig config_;
+  double average_loss_;
+  double average_loss_max_;
+  DataRate loss_based_bitrate_;
+  DataRate acknowledged_bitrate_max_;
+  Timestamp acknowledged_bitrate_last_update_;
+  Timestamp time_last_decrease_;
+  bool has_decreased_since_last_loss_report_;
+  Timestamp last_loss_packet_report_;
+  double last_loss_ratio_;
+};
+
+}  // namespace webrtc
+
+#endif  // MODULES_CONGESTION_CONTROLLER_GOOG_CC_LOSS_BASED_BANDWIDTH_ESTIMATION_H_

TMessagesProj/jni/voip/webrtc/modules/congestion_controller/goog_cc/loss_based_bwe_v2.h

@@ -0,0 +1,223 @@
+/*
+ *  Copyright 2021 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_CONGESTION_CONTROLLER_GOOG_CC_LOSS_BASED_BWE_V2_H_
+#define MODULES_CONGESTION_CONTROLLER_GOOG_CC_LOSS_BASED_BWE_V2_H_
+
+#include <vector>
+
+#include "absl/types/optional.h"
+#include "api/array_view.h"
+#include "api/field_trials_view.h"
+#include "api/transport/network_types.h"
+#include "api/units/data_rate.h"
+#include "api/units/data_size.h"
+#include "api/units/time_delta.h"
+#include "api/units/timestamp.h"
+
+namespace webrtc {
+
+// State of the loss based estimate, which can be either increasing/decreasing
+// when network is loss limited, or equal to the delay based estimate.
+enum class LossBasedState {
+  kIncreasing = 0,
+  // TODO(bugs.webrtc.org/12707): Remove one of the increasing states once we
+  // have decided if padding is usefull for ramping up when BWE is loss
+  // limited.
+  kIncreaseUsingPadding = 1,
+  kDecreasing = 2,
+  kDelayBasedEstimate = 3
+};
+
+class LossBasedBweV2 {
+ public:
+  struct Result {
+    ~Result() = default;
+    DataRate bandwidth_estimate = DataRate::Zero();
+    // State is used by goog_cc, which later sends probe requests to probe
+    // controller if state is kIncreasing.
+    LossBasedState state = LossBasedState::kDelayBasedEstimate;
+  };
+  // Creates a disabled `LossBasedBweV2` if the
+  // `key_value_config` is not valid.
+  explicit LossBasedBweV2(const FieldTrialsView* key_value_config);
+
+  LossBasedBweV2(const LossBasedBweV2&) = delete;
+  LossBasedBweV2& operator=(const LossBasedBweV2&) = delete;
+
+  ~LossBasedBweV2() = default;
+
+  bool IsEnabled() const;
+  // Returns true iff a BWE can be calculated, i.e., the estimator has been
+  // initialized with a BWE and then has received enough `PacketResult`s.
+  bool IsReady() const;
+
+  // Returns true if loss based BWE is ready to be used in the start phase.
+  bool ReadyToUseInStartPhase() const;
+
+  // Returns true if loss based BWE can be used in the start phase.
+  bool UseInStartPhase() const;
+
+  // Returns `DataRate::PlusInfinity` if no BWE can be calculated.
+  Result GetLossBasedResult() const;
+
+  void SetAcknowledgedBitrate(DataRate acknowledged_bitrate);
+  void SetMinMaxBitrate(DataRate min_bitrate, DataRate max_bitrate);
+  void UpdateBandwidthEstimate(
+      rtc::ArrayView<const PacketResult> packet_results,
+      DataRate delay_based_estimate,
+      bool in_alr);
+  bool PaceAtLossBasedEstimate() const;
+
+  // For unit testing only.
+  void SetBandwidthEstimate(DataRate bandwidth_estimate);
+
+ private:
+  struct ChannelParameters {
+    double inherent_loss = 0.0;
+    DataRate loss_limited_bandwidth = DataRate::MinusInfinity();
+  };
+
+  struct Config {
+    double bandwidth_rampup_upper_bound_factor = 0.0;
+    double bandwidth_rampup_upper_bound_factor_in_hold = 0;
+    double bandwidth_rampup_hold_threshold = 0;
+    double rampup_acceleration_max_factor = 0.0;
+    TimeDelta rampup_acceleration_maxout_time = TimeDelta::Zero();
+    std::vector<double> candidate_factors;
+    double higher_bandwidth_bias_factor = 0.0;
+    double higher_log_bandwidth_bias_factor = 0.0;
+    double inherent_loss_lower_bound = 0.0;
+    double loss_threshold_of_high_bandwidth_preference = 0.0;
+    double bandwidth_preference_smoothing_factor = 0.0;
+    DataRate inherent_loss_upper_bound_bandwidth_balance =
+        DataRate::MinusInfinity();
+    double inherent_loss_upper_bound_offset = 0.0;
+    double initial_inherent_loss_estimate = 0.0;
+    int newton_iterations = 0;
+    double newton_step_size = 0.0;
+    bool append_acknowledged_rate_candidate = true;
+    bool append_delay_based_estimate_candidate = false;
+    bool append_upper_bound_candidate_in_alr = false;
+    TimeDelta observation_duration_lower_bound = TimeDelta::Zero();
+    int observation_window_size = 0;
+    double sending_rate_smoothing_factor = 0.0;
+    double instant_upper_bound_temporal_weight_factor = 0.0;
+    DataRate instant_upper_bound_bandwidth_balance = DataRate::MinusInfinity();
+    double instant_upper_bound_loss_offset = 0.0;
+    double temporal_weight_factor = 0.0;
+    double bandwidth_backoff_lower_bound_factor = 0.0;
+    double max_increase_factor = 0.0;
+    TimeDelta delayed_increase_window = TimeDelta::Zero();
+    bool not_increase_if_inherent_loss_less_than_average_loss = false;
+    bool not_use_acked_rate_in_alr = false;
+    bool use_in_start_phase = false;
+    int min_num_observations = 0;
+    double lower_bound_by_acked_rate_factor = 0.0;
+    double hold_duration_factor = 0.0;
+    bool use_byte_loss_rate = false;
+    TimeDelta padding_duration = TimeDelta::Zero();
+    bool bound_best_candidate = false;
+    bool pace_at_loss_based_estimate = false;
+  };
+
+  struct Derivatives {
+    double first = 0.0;
+    double second = 0.0;
+  };
+
+  struct Observation {
+    bool IsInitialized() const { return id != -1; }
+
+    int num_packets = 0;
+    int num_lost_packets = 0;
+    int num_received_packets = 0;
+    DataRate sending_rate = DataRate::MinusInfinity();
+    DataSize size = DataSize::Zero();
+    DataSize lost_size = DataSize::Zero();
+    int id = -1;
+  };
+
+  struct PartialObservation {
+    int num_packets = 0;
+    int num_lost_packets = 0;
+    DataSize size = DataSize::Zero();
+    DataSize lost_size = DataSize::Zero();
+  };
+
+  struct PaddingInfo {
+    DataRate padding_rate = DataRate::MinusInfinity();
+    Timestamp padding_timestamp = Timestamp::MinusInfinity();
+  };
+
+  struct HoldInfo {
+    Timestamp timestamp = Timestamp::MinusInfinity();
+    TimeDelta duration = TimeDelta::Zero();
+    DataRate rate = DataRate::PlusInfinity();
+  };
+
+  static absl::optional<Config> CreateConfig(
+      const FieldTrialsView* key_value_config);
+  bool IsConfigValid() const;
+
+  // Returns `0.0` if not enough loss statistics have been received.
+  double GetAverageReportedLossRatio() const;
+  double GetAverageReportedPacketLossRatio() const;
+  double GetAverageReportedByteLossRatio() const;
+  std::vector<ChannelParameters> GetCandidates(bool in_alr) const;
+  DataRate GetCandidateBandwidthUpperBound() const;
+  Derivatives GetDerivatives(const ChannelParameters& channel_parameters) const;
+  double GetFeasibleInherentLoss(
+      const ChannelParameters& channel_parameters) const;
+  double GetInherentLossUpperBound(DataRate bandwidth) const;
+  double AdjustBiasFactor(double loss_rate, double bias_factor) const;
+  double GetHighBandwidthBias(DataRate bandwidth) const;
+  double GetObjective(const ChannelParameters& channel_parameters) const;
+  DataRate GetSendingRate(DataRate instantaneous_sending_rate) const;
+  DataRate GetInstantUpperBound() const;
+  void CalculateInstantUpperBound();
+  DataRate GetInstantLowerBound() const;
+  void CalculateInstantLowerBound();
+
+  void CalculateTemporalWeights();
+  void NewtonsMethodUpdate(ChannelParameters& channel_parameters) const;
+
+  // Returns false if no observation was created.
+  bool PushBackObservation(rtc::ArrayView<const PacketResult> packet_results);
+  bool IsEstimateIncreasingWhenLossLimited(DataRate old_estimate,
+                                           DataRate new_estimate);
+  bool IsInLossLimitedState() const;
+  bool CanKeepIncreasingState(DataRate estimate) const;
+
+  absl::optional<DataRate> acknowledged_bitrate_;
+  absl::optional<Config> config_;
+  ChannelParameters current_best_estimate_;
+  int num_observations_ = 0;
+  std::vector<Observation> observations_;
+  PartialObservation partial_observation_;
+  Timestamp last_send_time_most_recent_observation_ = Timestamp::PlusInfinity();
+  Timestamp last_time_estimate_reduced_ = Timestamp::MinusInfinity();
+  absl::optional<DataRate> cached_instant_upper_bound_;
+  absl::optional<DataRate> cached_instant_lower_bound_;
+  std::vector<double> instant_upper_bound_temporal_weights_;
+  std::vector<double> temporal_weights_;
+  Timestamp recovering_after_loss_timestamp_ = Timestamp::MinusInfinity();
+  DataRate bandwidth_limit_in_current_window_ = DataRate::PlusInfinity();
+  DataRate min_bitrate_ = DataRate::KilobitsPerSec(1);
+  DataRate max_bitrate_ = DataRate::PlusInfinity();
+  DataRate delay_based_estimate_ = DataRate::PlusInfinity();
+  LossBasedBweV2::Result loss_based_result_ = LossBasedBweV2::Result();
+  HoldInfo last_hold_info_ = HoldInfo();
+  PaddingInfo last_padding_info_ = PaddingInfo();
+};
+
+}  // namespace webrtc
+
+#endif  // MODULES_CONGESTION_CONTROLLER_GOOG_CC_LOSS_BASED_BWE_V2_H_

TMessagesProj/jni/voip/webrtc/modules/congestion_controller/goog_cc/probe_bitrate_estimator.cc

@@ -0,0 +1,206 @@
+/*
+ *  Copyright (c) 2016 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 "modules/congestion_controller/goog_cc/probe_bitrate_estimator.h"
+
+#include <algorithm>
+#include <memory>
+
+#include "absl/types/optional.h"
+#include "api/rtc_event_log/rtc_event_log.h"
+#include "api/transport/network_types.h"
+#include "api/units/data_rate.h"
+#include "api/units/data_size.h"
+#include "api/units/time_delta.h"
+#include "api/units/timestamp.h"
+#include "logging/rtc_event_log/events/rtc_event_probe_result_failure.h"
+#include "logging/rtc_event_log/events/rtc_event_probe_result_success.h"
+#include "rtc_base/checks.h"
+#include "rtc_base/logging.h"
+
+namespace webrtc {
+namespace {
+// The minumum number of probes we need to receive feedback about in percent
+// in order to have a valid estimate.
+constexpr double kMinReceivedProbesRatio = .80;
+
+// The minumum number of bytes we need to receive feedback about in percent
+// in order to have a valid estimate.
+constexpr double kMinReceivedBytesRatio = .80;
+
+// The maximum |receive rate| / |send rate| ratio for a valid estimate.
+constexpr float kMaxValidRatio = 2.0f;
+
+// The minimum |receive rate| / |send rate| ratio assuming that the link is
+// not saturated, i.e. we assume that we will receive at least
+// kMinRatioForUnsaturatedLink * |send rate| if |send rate| is less than the
+// link capacity.
+constexpr float kMinRatioForUnsaturatedLink = 0.9f;
+
+// The target utilization of the link. If we know true link capacity
+// we'd like to send at 95% of that rate.
+constexpr float kTargetUtilizationFraction = 0.95f;
+
+// The maximum time period over which the cluster history is retained.
+// This is also the maximum time period beyond which a probing burst is not
+// expected to last.
+constexpr TimeDelta kMaxClusterHistory = TimeDelta::Seconds(1);
+
+// The maximum time interval between first and the last probe on a cluster
+// on the sender side as well as the receive side.
+constexpr TimeDelta kMaxProbeInterval = TimeDelta::Seconds(1);
+
+}  // namespace
+
+ProbeBitrateEstimator::ProbeBitrateEstimator(RtcEventLog* event_log)
+    : event_log_(event_log) {}
+
+ProbeBitrateEstimator::~ProbeBitrateEstimator() = default;
+
+absl::optional<DataRate> ProbeBitrateEstimator::HandleProbeAndEstimateBitrate(
+    const PacketResult& packet_feedback) {
+  int cluster_id = packet_feedback.sent_packet.pacing_info.probe_cluster_id;
+  RTC_DCHECK_NE(cluster_id, PacedPacketInfo::kNotAProbe);
+
+  EraseOldClusters(packet_feedback.receive_time);
+
+  AggregatedCluster* cluster = &clusters_[cluster_id];
+
+  if (packet_feedback.sent_packet.send_time < cluster->first_send) {
+    cluster->first_send = packet_feedback.sent_packet.send_time;
+  }
+  if (packet_feedback.sent_packet.send_time > cluster->last_send) {
+    cluster->last_send = packet_feedback.sent_packet.send_time;
+    cluster->size_last_send = packet_feedback.sent_packet.size;
+  }
+  if (packet_feedback.receive_time < cluster->first_receive) {
+    cluster->first_receive = packet_feedback.receive_time;
+    cluster->size_first_receive = packet_feedback.sent_packet.size;
+  }
+  if (packet_feedback.receive_time > cluster->last_receive) {
+    cluster->last_receive = packet_feedback.receive_time;
+  }
+  cluster->size_total += packet_feedback.sent_packet.size;
+  cluster->num_probes += 1;
+
+  RTC_DCHECK_GT(
+      packet_feedback.sent_packet.pacing_info.probe_cluster_min_probes, 0);
+  RTC_DCHECK_GT(packet_feedback.sent_packet.pacing_info.probe_cluster_min_bytes,
+                0);
+
+  int min_probes =
+      packet_feedback.sent_packet.pacing_info.probe_cluster_min_probes *
+      kMinReceivedProbesRatio;
+  DataSize min_size =
+      DataSize::Bytes(
+          packet_feedback.sent_packet.pacing_info.probe_cluster_min_bytes) *
+      kMinReceivedBytesRatio;
+  if (cluster->num_probes < min_probes || cluster->size_total < min_size)
+    return absl::nullopt;
+
+  TimeDelta send_interval = cluster->last_send - cluster->first_send;
+  TimeDelta receive_interval = cluster->last_receive - cluster->first_receive;
+
+  if (send_interval <= TimeDelta::Zero() || send_interval > kMaxProbeInterval ||
+      receive_interval <= TimeDelta::Zero() ||
+      receive_interval > kMaxProbeInterval) {
+    RTC_LOG(LS_INFO) << "Probing unsuccessful, invalid send/receive interval"
+                        " [cluster id: "
+                     << cluster_id
+                     << "] [send interval: " << ToString(send_interval)
+                     << "]"
+                        " [receive interval: "
+                     << ToString(receive_interval) << "]";
+    if (event_log_) {
+      event_log_->Log(std::make_unique<RtcEventProbeResultFailure>(
+          cluster_id, ProbeFailureReason::kInvalidSendReceiveInterval));
+    }
+    return absl::nullopt;
+  }
+  // Since the `send_interval` does not include the time it takes to actually
+  // send the last packet the size of the last sent packet should not be
+  // included when calculating the send bitrate.
+  RTC_DCHECK_GT(cluster->size_total, cluster->size_last_send);
+  DataSize send_size = cluster->size_total - cluster->size_last_send;
+  DataRate send_rate = send_size / send_interval;
+
+  // Since the `receive_interval` does not include the time it takes to
+  // actually receive the first packet the size of the first received packet
+  // should not be included when calculating the receive bitrate.
+  RTC_DCHECK_GT(cluster->size_total, cluster->size_first_receive);
+  DataSize receive_size = cluster->size_total - cluster->size_first_receive;
+  DataRate receive_rate = receive_size / receive_interval;
+
+  double ratio = receive_rate / send_rate;
+  if (ratio > kMaxValidRatio) {
+    RTC_LOG(LS_INFO) << "Probing unsuccessful, receive/send ratio too high"
+                        " [cluster id: "
+                     << cluster_id << "] [send: " << ToString(send_size)
+                     << " / " << ToString(send_interval) << " = "
+                     << ToString(send_rate)
+                     << "]"
+                        " [receive: "
+                     << ToString(receive_size) << " / "
+                     << ToString(receive_interval) << " = "
+                     << ToString(receive_rate)
+                     << " ]"
+                        " [ratio: "
+                     << ToString(receive_rate) << " / " << ToString(send_rate)
+                     << " = " << ratio << " > kMaxValidRatio ("
+                     << kMaxValidRatio << ")]";
+    if (event_log_) {
+      event_log_->Log(std::make_unique<RtcEventProbeResultFailure>(
+          cluster_id, ProbeFailureReason::kInvalidSendReceiveRatio));
+    }
+    return absl::nullopt;
+  }
+  RTC_LOG(LS_INFO) << "Probing successful"
+                      " [cluster id: "
+                   << cluster_id << "] [send: " << ToString(send_size) << " / "
+                   << ToString(send_interval) << " = " << ToString(send_rate)
+                   << " ]"
+                      " [receive: "
+                   << ToString(receive_size) << " / "
+                   << ToString(receive_interval) << " = "
+                   << ToString(receive_rate) << "]";
+
+  DataRate res = std::min(send_rate, receive_rate);
+  // If we're receiving at significantly lower bitrate than we were sending at,
+  // it suggests that we've found the true capacity of the link. In this case,
+  // set the target bitrate slightly lower to not immediately overuse.
+  if (receive_rate < kMinRatioForUnsaturatedLink * send_rate) {
+    RTC_DCHECK_GT(send_rate, receive_rate);
+    res = kTargetUtilizationFraction * receive_rate;
+  }
+  if (event_log_) {
+    event_log_->Log(
+        std::make_unique<RtcEventProbeResultSuccess>(cluster_id, res.bps()));
+  }
+  estimated_data_rate_ = res;
+  return estimated_data_rate_;
+}
+
+absl::optional<DataRate>
+ProbeBitrateEstimator::FetchAndResetLastEstimatedBitrate() {
+  absl::optional<DataRate> estimated_data_rate = estimated_data_rate_;
+  estimated_data_rate_.reset();
+  return estimated_data_rate;
+}
+
+void ProbeBitrateEstimator::EraseOldClusters(Timestamp timestamp) {
+  for (auto it = clusters_.begin(); it != clusters_.end();) {
+    if (it->second.last_receive + kMaxClusterHistory < timestamp) {
+      it = clusters_.erase(it);
+    } else {
+      ++it;
+    }
+  }
+}
+}  // namespace webrtc

TMessagesProj/jni/voip/webrtc/modules/congestion_controller/goog_cc/probe_bitrate_estimator.h

@@ -0,0 +1,59 @@
+/*
+ *  Copyright (c) 2016 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_CONGESTION_CONTROLLER_GOOG_CC_PROBE_BITRATE_ESTIMATOR_H_
+#define MODULES_CONGESTION_CONTROLLER_GOOG_CC_PROBE_BITRATE_ESTIMATOR_H_
+
+#include <map>
+
+#include "absl/types/optional.h"
+#include "api/transport/network_types.h"
+#include "api/units/data_rate.h"
+#include "api/units/data_size.h"
+#include "api/units/timestamp.h"
+
+namespace webrtc {
+class RtcEventLog;
+
+class ProbeBitrateEstimator {
+ public:
+  explicit ProbeBitrateEstimator(RtcEventLog* event_log);
+  ~ProbeBitrateEstimator();
+
+  // Should be called for every probe packet we receive feedback about.
+  // Returns the estimated bitrate if the probe completes a valid cluster.
+  absl::optional<DataRate> HandleProbeAndEstimateBitrate(
+      const PacketResult& packet_feedback);
+
+  absl::optional<DataRate> FetchAndResetLastEstimatedBitrate();
+
+ private:
+  struct AggregatedCluster {
+    int num_probes = 0;
+    Timestamp first_send = Timestamp::PlusInfinity();
+    Timestamp last_send = Timestamp::MinusInfinity();
+    Timestamp first_receive = Timestamp::PlusInfinity();
+    Timestamp last_receive = Timestamp::MinusInfinity();
+    DataSize size_last_send = DataSize::Zero();
+    DataSize size_first_receive = DataSize::Zero();
+    DataSize size_total = DataSize::Zero();
+  };
+
+  // Erases old cluster data that was seen before `timestamp`.
+  void EraseOldClusters(Timestamp timestamp);
+
+  std::map<int, AggregatedCluster> clusters_;
+  RtcEventLog* const event_log_;
+  absl::optional<DataRate> estimated_data_rate_;
+};
+
+}  // namespace webrtc
+
+#endif  // MODULES_CONGESTION_CONTROLLER_GOOG_CC_PROBE_BITRATE_ESTIMATOR_H_

TMessagesProj/jni/voip/webrtc/modules/congestion_controller/goog_cc/probe_controller.cc

@@ -0,0 +1,555 @@
+/*
+ *  Copyright (c) 2016 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 "modules/congestion_controller/goog_cc/probe_controller.h"
+
+#include <algorithm>
+#include <cstdint>
+#include <initializer_list>
+#include <memory>
+#include <vector>
+
+#include "absl/strings/match.h"
+#include "absl/types/optional.h"
+#include "api/field_trials_view.h"
+#include "api/rtc_event_log/rtc_event_log.h"
+#include "api/transport/network_types.h"
+#include "api/units/data_rate.h"
+#include "api/units/data_size.h"
+#include "api/units/time_delta.h"
+#include "api/units/timestamp.h"
+#include "logging/rtc_event_log/events/rtc_event_probe_cluster_created.h"
+#include "rtc_base/checks.h"
+#include "rtc_base/experiments/field_trial_parser.h"
+#include "rtc_base/logging.h"
+#include "system_wrappers/include/metrics.h"
+
+namespace webrtc {
+
+namespace {
+// Maximum waiting time from the time of initiating probing to getting
+// the measured results back.
+constexpr TimeDelta kMaxWaitingTimeForProbingResult = TimeDelta::Seconds(1);
+
+// Default probing bitrate limit. Applied only when the application didn't
+// specify max bitrate.
+constexpr DataRate kDefaultMaxProbingBitrate = DataRate::KilobitsPerSec(5000);
+
+// If the bitrate drops to a factor `kBitrateDropThreshold` or lower
+// and we recover within `kBitrateDropTimeoutMs`, then we'll send
+// a probe at a fraction `kProbeFractionAfterDrop` of the original bitrate.
+constexpr double kBitrateDropThreshold = 0.66;
+constexpr TimeDelta kBitrateDropTimeout = TimeDelta::Seconds(5);
+constexpr double kProbeFractionAfterDrop = 0.85;
+
+// Timeout for probing after leaving ALR. If the bitrate drops significantly,
+// (as determined by the delay based estimator) and we leave ALR, then we will
+// send a probe if we recover within `kLeftAlrTimeoutMs` ms.
+constexpr TimeDelta kAlrEndedTimeout = TimeDelta::Seconds(3);
+
+// The expected uncertainty of probe result (as a fraction of the target probe
+// This is a limit on how often probing can be done when there is a BW
+// drop detected in ALR.
+constexpr TimeDelta kMinTimeBetweenAlrProbes = TimeDelta::Seconds(5);
+
+// bitrate). Used to avoid probing if the probe bitrate is close to our current
+// estimate.
+constexpr double kProbeUncertainty = 0.05;
+
+// Use probing to recover faster after large bitrate estimate drops.
+constexpr char kBweRapidRecoveryExperiment[] =
+    "WebRTC-BweRapidRecoveryExperiment";
+
+void MaybeLogProbeClusterCreated(RtcEventLog* event_log,
+                                 const ProbeClusterConfig& probe) {
+  RTC_DCHECK(event_log);
+  if (!event_log) {
+    return;
+  }
+
+  DataSize min_data_size = probe.target_data_rate * probe.target_duration;
+  event_log->Log(std::make_unique<RtcEventProbeClusterCreated>(
+      probe.id, probe.target_data_rate.bps(), probe.target_probe_count,
+      min_data_size.bytes()));
+}
+
+}  // namespace
+
+ProbeControllerConfig::ProbeControllerConfig(
+    const FieldTrialsView* key_value_config)
+    : first_exponential_probe_scale("p1", 3.0),
+      second_exponential_probe_scale("p2", 6.0),
+      further_exponential_probe_scale("step_size", 2),
+      further_probe_threshold("further_probe_threshold", 0.7),
+      alr_probing_interval("alr_interval", TimeDelta::Seconds(5)),
+      alr_probe_scale("alr_scale", 2),
+      network_state_estimate_probing_interval("network_state_interval",
+                                              TimeDelta::PlusInfinity()),
+      probe_if_estimate_lower_than_network_state_estimate_ratio(
+          "est_lower_than_network_ratio",
+          0),
+      estimate_lower_than_network_state_estimate_probing_interval(
+          "est_lower_than_network_interval",
+          TimeDelta::Seconds(3)),
+      network_state_probe_scale("network_state_scale", 1.0),
+      network_state_probe_duration("network_state_probe_duration",
+                                   TimeDelta::Millis(15)),
+
+      probe_on_max_allocated_bitrate_change("probe_max_allocation", true),
+      first_allocation_probe_scale("alloc_p1", 1),
+      second_allocation_probe_scale("alloc_p2", 2),
+      allocation_probe_limit_by_current_scale("alloc_current_bwe_limit"),
+      min_probe_packets_sent("min_probe_packets_sent", 5),
+      min_probe_duration("min_probe_duration", TimeDelta::Millis(15)),
+      loss_limited_probe_scale("loss_limited_scale", 1.5),
+      skip_if_estimate_larger_than_fraction_of_max(
+          "skip_if_est_larger_than_fraction_of_max",
+          0.0) {
+  ParseFieldTrial(
+      {&first_exponential_probe_scale, &second_exponential_probe_scale,
+       &further_exponential_probe_scale, &further_probe_threshold,
+       &alr_probing_interval, &alr_probe_scale,
+       &probe_on_max_allocated_bitrate_change, &first_allocation_probe_scale,
+       &second_allocation_probe_scale, &allocation_probe_limit_by_current_scale,
+       &min_probe_duration, &network_state_estimate_probing_interval,
+       &probe_if_estimate_lower_than_network_state_estimate_ratio,
+       &estimate_lower_than_network_state_estimate_probing_interval,
+       &network_state_probe_scale, &network_state_probe_duration,
+       &min_probe_packets_sent, &loss_limited_probe_scale,
+       &skip_if_estimate_larger_than_fraction_of_max},
+      key_value_config->Lookup("WebRTC-Bwe-ProbingConfiguration"));
+
+  // Specialized keys overriding subsets of WebRTC-Bwe-ProbingConfiguration
+  ParseFieldTrial(
+      {&first_exponential_probe_scale, &second_exponential_probe_scale},
+      key_value_config->Lookup("WebRTC-Bwe-InitialProbing"));
+  ParseFieldTrial({&further_exponential_probe_scale, &further_probe_threshold},
+                  key_value_config->Lookup("WebRTC-Bwe-ExponentialProbing"));
+  ParseFieldTrial(
+      {&alr_probing_interval, &alr_probe_scale, &loss_limited_probe_scale},
+      key_value_config->Lookup("WebRTC-Bwe-AlrProbing"));
+  ParseFieldTrial(
+      {&first_allocation_probe_scale, &second_allocation_probe_scale,
+       &allocation_probe_limit_by_current_scale},
+      key_value_config->Lookup("WebRTC-Bwe-AllocationProbing"));
+  ParseFieldTrial({&min_probe_packets_sent, &min_probe_duration},
+                  key_value_config->Lookup("WebRTC-Bwe-ProbingBehavior"));
+}
+
+ProbeControllerConfig::ProbeControllerConfig(const ProbeControllerConfig&) =
+    default;
+ProbeControllerConfig::~ProbeControllerConfig() = default;
+
+ProbeController::ProbeController(const FieldTrialsView* key_value_config,
+                                 RtcEventLog* event_log)
+    : network_available_(false),
+      enable_periodic_alr_probing_(false),
+      in_rapid_recovery_experiment_(absl::StartsWith(
+          key_value_config->Lookup(kBweRapidRecoveryExperiment),
+          "Enabled")),
+      event_log_(event_log),
+      config_(ProbeControllerConfig(key_value_config)) {
+  Reset(Timestamp::Zero());
+}
+
+ProbeController::~ProbeController() {}
+
+std::vector<ProbeClusterConfig> ProbeController::SetBitrates(
+    DataRate min_bitrate,
+    DataRate start_bitrate,
+    DataRate max_bitrate,
+    Timestamp at_time) {
+  if (start_bitrate > DataRate::Zero()) {
+    start_bitrate_ = start_bitrate;
+    estimated_bitrate_ = start_bitrate;
+  } else if (start_bitrate_.IsZero()) {
+    start_bitrate_ = min_bitrate;
+  }
+
+  // The reason we use the variable `old_max_bitrate_pbs` is because we
+  // need to set `max_bitrate_` before we call InitiateProbing.
+  DataRate old_max_bitrate = max_bitrate_;
+  max_bitrate_ =
+      max_bitrate.IsFinite() ? max_bitrate : kDefaultMaxProbingBitrate;
+
+  switch (state_) {
+    case State::kInit:
+      if (network_available_)
+        return InitiateExponentialProbing(at_time);
+      break;
+
+    case State::kWaitingForProbingResult:
+      break;
+
+    case State::kProbingComplete:
+      // If the new max bitrate is higher than both the old max bitrate and the
+      // estimate then initiate probing.
+      if (!estimated_bitrate_.IsZero() && old_max_bitrate < max_bitrate_ &&
+          estimated_bitrate_ < max_bitrate_) {
+        return InitiateProbing(at_time, {max_bitrate_}, false);
+      }
+      break;
+  }
+  return std::vector<ProbeClusterConfig>();
+}
+
+std::vector<ProbeClusterConfig> ProbeController::OnMaxTotalAllocatedBitrate(
+    DataRate max_total_allocated_bitrate,
+    Timestamp at_time) {
+  const bool in_alr = alr_start_time_.has_value();
+  const bool allow_allocation_probe = in_alr;
+
+  if (config_.probe_on_max_allocated_bitrate_change &&
+      state_ == State::kProbingComplete &&
+      max_total_allocated_bitrate != max_total_allocated_bitrate_ &&
+      estimated_bitrate_ < max_bitrate_ &&
+      estimated_bitrate_ < max_total_allocated_bitrate &&
+      allow_allocation_probe) {
+    max_total_allocated_bitrate_ = max_total_allocated_bitrate;
+
+    if (!config_.first_allocation_probe_scale)
+      return std::vector<ProbeClusterConfig>();
+
+    DataRate first_probe_rate = max_total_allocated_bitrate *
+                                config_.first_allocation_probe_scale.Value();
+    DataRate current_bwe_limit =
+        !config_.allocation_probe_limit_by_current_scale
+            ? DataRate::PlusInfinity()
+            : estimated_bitrate_ *
+                  config_.allocation_probe_limit_by_current_scale.Value();
+    bool limited_by_current_bwe = current_bwe_limit < first_probe_rate;
+    if (limited_by_current_bwe) {
+      first_probe_rate = current_bwe_limit;
+    }
+
+    std::vector<DataRate> probes = {first_probe_rate};
+    if (!limited_by_current_bwe && config_.second_allocation_probe_scale) {
+      DataRate second_probe_rate =
+          max_total_allocated_bitrate *
+          config_.second_allocation_probe_scale.Value();
+      limited_by_current_bwe = current_bwe_limit < second_probe_rate;
+      if (limited_by_current_bwe) {
+        second_probe_rate = current_bwe_limit;
+      }
+      if (second_probe_rate > first_probe_rate)
+        probes.push_back(second_probe_rate);
+    }
+    bool allow_further_probing = limited_by_current_bwe;
+
+    return InitiateProbing(at_time, probes, allow_further_probing);
+  }
+  max_total_allocated_bitrate_ = max_total_allocated_bitrate;
+  return std::vector<ProbeClusterConfig>();
+}
+
+std::vector<ProbeClusterConfig> ProbeController::OnNetworkAvailability(
+    NetworkAvailability msg) {
+  network_available_ = msg.network_available;
+
+  if (!network_available_ && state_ == State::kWaitingForProbingResult) {
+    state_ = State::kProbingComplete;
+    min_bitrate_to_probe_further_ = DataRate::PlusInfinity();
+  }
+
+  if (network_available_ && state_ == State::kInit && !start_bitrate_.IsZero())
+    return InitiateExponentialProbing(msg.at_time);
+  return std::vector<ProbeClusterConfig>();
+}
+
+std::vector<ProbeClusterConfig> ProbeController::InitiateExponentialProbing(
+    Timestamp at_time) {
+  RTC_DCHECK(network_available_);
+  RTC_DCHECK(state_ == State::kInit);
+  RTC_DCHECK_GT(start_bitrate_, DataRate::Zero());
+
+  // When probing at 1.8 Mbps ( 6x 300), this represents a threshold of
+  // 1.2 Mbps to continue probing.
+  std::vector<DataRate> probes = {config_.first_exponential_probe_scale *
+                                  start_bitrate_};
+  if (config_.second_exponential_probe_scale &&
+      config_.second_exponential_probe_scale.GetOptional().value() > 0) {
+    probes.push_back(config_.second_exponential_probe_scale.Value() *
+                     start_bitrate_);
+  }
+  return InitiateProbing(at_time, probes, true);
+}
+
+std::vector<ProbeClusterConfig> ProbeController::SetEstimatedBitrate(
+    DataRate bitrate,
+    BandwidthLimitedCause bandwidth_limited_cause,
+    Timestamp at_time) {
+  bandwidth_limited_cause_ = bandwidth_limited_cause;
+  if (bitrate < kBitrateDropThreshold * estimated_bitrate_) {
+    time_of_last_large_drop_ = at_time;
+    bitrate_before_last_large_drop_ = estimated_bitrate_;
+  }
+  estimated_bitrate_ = bitrate;
+
+  if (state_ == State::kWaitingForProbingResult) {
+    // Continue probing if probing results indicate channel has greater
+    // capacity.
+    DataRate network_state_estimate_probe_further_limit =
+        config_.network_state_estimate_probing_interval->IsFinite() &&
+                network_estimate_
+            ? network_estimate_->link_capacity_upper *
+                  config_.further_probe_threshold
+            : DataRate::PlusInfinity();
+    RTC_LOG(LS_INFO) << "Measured bitrate: " << bitrate
+                     << " Minimum to probe further: "
+                     << min_bitrate_to_probe_further_ << " upper limit: "
+                     << network_state_estimate_probe_further_limit;
+
+    if (bitrate > min_bitrate_to_probe_further_ &&
+        bitrate <= network_state_estimate_probe_further_limit) {
+      return InitiateProbing(
+          at_time, {config_.further_exponential_probe_scale * bitrate}, true);
+    }
+  }
+  return {};
+}
+
+void ProbeController::EnablePeriodicAlrProbing(bool enable) {
+  enable_periodic_alr_probing_ = enable;
+}
+
+void ProbeController::SetAlrStartTimeMs(
+    absl::optional<int64_t> alr_start_time_ms) {
+  if (alr_start_time_ms) {
+    alr_start_time_ = Timestamp::Millis(*alr_start_time_ms);
+  } else {
+    alr_start_time_ = absl::nullopt;
+  }
+}
+void ProbeController::SetAlrEndedTimeMs(int64_t alr_end_time_ms) {
+  alr_end_time_.emplace(Timestamp::Millis(alr_end_time_ms));
+}
+
+std::vector<ProbeClusterConfig> ProbeController::RequestProbe(
+    Timestamp at_time) {
+  // Called once we have returned to normal state after a large drop in
+  // estimated bandwidth. The current response is to initiate a single probe
+  // session (if not already probing) at the previous bitrate.
+  //
+  // If the probe session fails, the assumption is that this drop was a
+  // real one from a competing flow or a network change.
+  bool in_alr = alr_start_time_.has_value();
+  bool alr_ended_recently =
+      (alr_end_time_.has_value() &&
+       at_time - alr_end_time_.value() < kAlrEndedTimeout);
+  if (in_alr || alr_ended_recently || in_rapid_recovery_experiment_) {
+    if (state_ == State::kProbingComplete) {
+      DataRate suggested_probe =
+          kProbeFractionAfterDrop * bitrate_before_last_large_drop_;
+      DataRate min_expected_probe_result =
+          (1 - kProbeUncertainty) * suggested_probe;
+      TimeDelta time_since_drop = at_time - time_of_last_large_drop_;
+      TimeDelta time_since_probe = at_time - last_bwe_drop_probing_time_;
+      if (min_expected_probe_result > estimated_bitrate_ &&
+          time_since_drop < kBitrateDropTimeout &&
+          time_since_probe > kMinTimeBetweenAlrProbes) {
+        RTC_LOG(LS_INFO) << "Detected big bandwidth drop, start probing.";
+        // Track how often we probe in response to bandwidth drop in ALR.
+        RTC_HISTOGRAM_COUNTS_10000(
+            "WebRTC.BWE.BweDropProbingIntervalInS",
+            (at_time - last_bwe_drop_probing_time_).seconds());
+        last_bwe_drop_probing_time_ = at_time;
+        return InitiateProbing(at_time, {suggested_probe}, false);
+      }
+    }
+  }
+  return std::vector<ProbeClusterConfig>();
+}
+
+void ProbeController::SetNetworkStateEstimate(
+    webrtc::NetworkStateEstimate estimate) {
+  network_estimate_ = estimate;
+}
+
+void ProbeController::Reset(Timestamp at_time) {
+  bandwidth_limited_cause_ = BandwidthLimitedCause::kDelayBasedLimited;
+  state_ = State::kInit;
+  min_bitrate_to_probe_further_ = DataRate::PlusInfinity();
+  time_last_probing_initiated_ = Timestamp::Zero();
+  estimated_bitrate_ = DataRate::Zero();
+  network_estimate_ = absl::nullopt;
+  start_bitrate_ = DataRate::Zero();
+  max_bitrate_ = kDefaultMaxProbingBitrate;
+  Timestamp now = at_time;
+  last_bwe_drop_probing_time_ = now;
+  alr_end_time_.reset();
+  time_of_last_large_drop_ = now;
+  bitrate_before_last_large_drop_ = DataRate::Zero();
+  max_total_allocated_bitrate_ = DataRate::Zero();
+}
+
+bool ProbeController::TimeForAlrProbe(Timestamp at_time) const {
+  if (enable_periodic_alr_probing_ && alr_start_time_) {
+    Timestamp next_probe_time =
+        std::max(*alr_start_time_, time_last_probing_initiated_) +
+        config_.alr_probing_interval;
+    return at_time >= next_probe_time;
+  }
+  return false;
+}
+
+bool ProbeController::TimeForNetworkStateProbe(Timestamp at_time) const {
+  if (!network_estimate_ ||
+      network_estimate_->link_capacity_upper.IsInfinite()) {
+    return false;
+  }
+
+  bool probe_due_to_low_estimate =
+      bandwidth_limited_cause_ == BandwidthLimitedCause::kDelayBasedLimited &&
+      estimated_bitrate_ <
+          config_.probe_if_estimate_lower_than_network_state_estimate_ratio *
+              network_estimate_->link_capacity_upper;
+  if (probe_due_to_low_estimate &&
+      config_.estimate_lower_than_network_state_estimate_probing_interval
+          ->IsFinite()) {
+    Timestamp next_probe_time =
+        time_last_probing_initiated_ +
+        config_.estimate_lower_than_network_state_estimate_probing_interval;
+    return at_time >= next_probe_time;
+  }
+
+  bool periodic_probe =
+      estimated_bitrate_ < network_estimate_->link_capacity_upper;
+  if (periodic_probe &&
+      config_.network_state_estimate_probing_interval->IsFinite()) {
+    Timestamp next_probe_time = time_last_probing_initiated_ +
+                                config_.network_state_estimate_probing_interval;
+    return at_time >= next_probe_time;
+  }
+
+  return false;
+}
+
+std::vector<ProbeClusterConfig> ProbeController::Process(Timestamp at_time) {
+  if (at_time - time_last_probing_initiated_ >
+      kMaxWaitingTimeForProbingResult) {
+    if (state_ == State::kWaitingForProbingResult) {
+      RTC_LOG(LS_INFO) << "kWaitingForProbingResult: timeout";
+      state_ = State::kProbingComplete;
+      min_bitrate_to_probe_further_ = DataRate::PlusInfinity();
+    }
+  }
+  if (estimated_bitrate_.IsZero() || state_ != State::kProbingComplete) {
+    return {};
+  }
+  if (TimeForAlrProbe(at_time) || TimeForNetworkStateProbe(at_time)) {
+    return InitiateProbing(
+        at_time, {estimated_bitrate_ * config_.alr_probe_scale}, true);
+  }
+  return std::vector<ProbeClusterConfig>();
+}
+
+std::vector<ProbeClusterConfig> ProbeController::InitiateProbing(
+    Timestamp now,
+    std::vector<DataRate> bitrates_to_probe,
+    bool probe_further) {
+  if (config_.skip_if_estimate_larger_than_fraction_of_max > 0) {
+    DataRate network_estimate = network_estimate_
+                                    ? network_estimate_->link_capacity_upper
+                                    : DataRate::PlusInfinity();
+    DataRate max_probe_rate =
+        max_total_allocated_bitrate_.IsZero()
+            ? max_bitrate_
+            : std::min(max_total_allocated_bitrate_, max_bitrate_);
+    if (std::min(network_estimate, estimated_bitrate_) >
+        config_.skip_if_estimate_larger_than_fraction_of_max * max_probe_rate) {
+      state_ = State::kProbingComplete;
+      min_bitrate_to_probe_further_ = DataRate::PlusInfinity();
+      return {};
+    }
+  }
+
+  DataRate max_probe_bitrate = max_bitrate_;
+  if (max_total_allocated_bitrate_ > DataRate::Zero()) {
+    // If a max allocated bitrate has been configured, allow probing up to 2x
+    // that rate. This allows some overhead to account for bursty streams,
+    // which otherwise would have to ramp up when the overshoot is already in
+    // progress.
+    // It also avoids minor quality reduction caused by probes often being
+    // received at slightly less than the target probe bitrate.
+    max_probe_bitrate =
+        std::min(max_probe_bitrate, max_total_allocated_bitrate_ * 2);
+  }
+
+  DataRate estimate_capped_bitrate = DataRate::PlusInfinity();
+  switch (bandwidth_limited_cause_) {
+    case BandwidthLimitedCause::kRttBasedBackOffHighRtt:
+    case BandwidthLimitedCause::kDelayBasedLimitedDelayIncreased:
+    case BandwidthLimitedCause::kLossLimitedBwe:
+      RTC_LOG(LS_INFO) << "Not sending probe in bandwidth limited state.";
+      return {};
+    case BandwidthLimitedCause::kLossLimitedBweIncreasing:
+      estimate_capped_bitrate =
+          std::min(max_probe_bitrate,
+                   estimated_bitrate_ * config_.loss_limited_probe_scale);
+      break;
+    case BandwidthLimitedCause::kDelayBasedLimited:
+      break;
+    default:
+      break;
+  }
+
+  if (config_.network_state_estimate_probing_interval->IsFinite() &&
+      network_estimate_ && network_estimate_->link_capacity_upper.IsFinite()) {
+    if (network_estimate_->link_capacity_upper.IsZero()) {
+      RTC_LOG(LS_INFO) << "Not sending probe, Network state estimate is zero";
+      return {};
+    }
+    estimate_capped_bitrate = std::min(
+        {estimate_capped_bitrate, max_probe_bitrate,
+         std::max(estimated_bitrate_, network_estimate_->link_capacity_upper *
+                                          config_.network_state_probe_scale)});
+  }
+
+  std::vector<ProbeClusterConfig> pending_probes;
+  for (DataRate bitrate : bitrates_to_probe) {
+    RTC_DCHECK(!bitrate.IsZero());
+    bitrate = std::min(bitrate, estimate_capped_bitrate);
+    if (bitrate > max_probe_bitrate) {
+      bitrate = max_probe_bitrate;
+      probe_further = false;
+    }
+
+    ProbeClusterConfig config;
+    config.at_time = now;
+    config.target_data_rate = bitrate;
+    if (network_estimate_ &&
+        config_.network_state_estimate_probing_interval->IsFinite()) {
+      config.target_duration = config_.network_state_probe_duration;
+    } else {
+      config.target_duration = config_.min_probe_duration;
+    }
+
+    config.target_probe_count = config_.min_probe_packets_sent;
+    config.id = next_probe_cluster_id_;
+    next_probe_cluster_id_++;
+    MaybeLogProbeClusterCreated(event_log_, config);
+    pending_probes.push_back(config);
+  }
+  time_last_probing_initiated_ = now;
+  if (probe_further) {
+    state_ = State::kWaitingForProbingResult;
+    // Dont expect probe results to be larger than a fraction of the actual
+    // probe rate.
+    min_bitrate_to_probe_further_ =
+        std::min(estimate_capped_bitrate, (*(bitrates_to_probe.end() - 1))) *
+        config_.further_probe_threshold;
+  } else {
+    state_ = State::kProbingComplete;
+    min_bitrate_to_probe_further_ = DataRate::PlusInfinity();
+  }
+  return pending_probes;
+}
+
+}  // namespace webrtc

TMessagesProj/jni/voip/webrtc/modules/congestion_controller/goog_cc/probe_controller.h

@@ -0,0 +1,187 @@
+/*
+ *  Copyright (c) 2016 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_CONGESTION_CONTROLLER_GOOG_CC_PROBE_CONTROLLER_H_
+#define MODULES_CONGESTION_CONTROLLER_GOOG_CC_PROBE_CONTROLLER_H_
+
+#include <stdint.h>
+
+#include <vector>
+
+#include "absl/base/attributes.h"
+#include "absl/types/optional.h"
+#include "api/field_trials_view.h"
+#include "api/rtc_event_log/rtc_event_log.h"
+#include "api/transport/network_types.h"
+#include "api/units/data_rate.h"
+#include "api/units/time_delta.h"
+#include "api/units/timestamp.h"
+#include "rtc_base/experiments/field_trial_parser.h"
+
+namespace webrtc {
+
+struct ProbeControllerConfig {
+  explicit ProbeControllerConfig(const FieldTrialsView* key_value_config);
+  ProbeControllerConfig(const ProbeControllerConfig&);
+  ProbeControllerConfig& operator=(const ProbeControllerConfig&) = default;
+  ~ProbeControllerConfig();
+
+  // These parameters configure the initial probes. First we send one or two
+  // probes of sizes p1 * start_bitrate_ and p2 * start_bitrate_.
+  // Then whenever we get a bitrate estimate of at least further_probe_threshold
+  // times the size of the last sent probe we'll send another one of size
+  // step_size times the new estimate.
+  FieldTrialParameter<double> first_exponential_probe_scale;
+  FieldTrialOptional<double> second_exponential_probe_scale;
+  FieldTrialParameter<double> further_exponential_probe_scale;
+  FieldTrialParameter<double> further_probe_threshold;
+
+  // Configures how often we send ALR probes and how big they are.
+  FieldTrialParameter<TimeDelta> alr_probing_interval;
+  FieldTrialParameter<double> alr_probe_scale;
+
+  // Configures how often we send probes if NetworkStateEstimate is available.
+  FieldTrialParameter<TimeDelta> network_state_estimate_probing_interval;
+  // Periodically probe as long as the the ratio beteeen current estimate and
+  // NetworkStateEstimate is lower then this.
+  FieldTrialParameter<double>
+      probe_if_estimate_lower_than_network_state_estimate_ratio;
+  FieldTrialParameter<TimeDelta>
+      estimate_lower_than_network_state_estimate_probing_interval;
+  FieldTrialParameter<double> network_state_probe_scale;
+  // Overrides min_probe_duration if network_state_estimate_probing_interval
+  // is set and a network state estimate is known.
+  FieldTrialParameter<TimeDelta> network_state_probe_duration;
+
+  // Configures the probes emitted by changed to the allocated bitrate.
+  FieldTrialParameter<bool> probe_on_max_allocated_bitrate_change;
+  FieldTrialOptional<double> first_allocation_probe_scale;
+  FieldTrialOptional<double> second_allocation_probe_scale;
+  FieldTrialOptional<double> allocation_probe_limit_by_current_scale;
+
+  // The minimum number probing packets used.
+  FieldTrialParameter<int> min_probe_packets_sent;
+  // The minimum probing duration.
+  FieldTrialParameter<TimeDelta> min_probe_duration;
+  FieldTrialParameter<double> loss_limited_probe_scale;
+  // Dont send a probe if min(estimate, network state estimate) is larger than
+  // this fraction of the set max bitrate.
+  FieldTrialParameter<double> skip_if_estimate_larger_than_fraction_of_max;
+};
+
+// Reason that bandwidth estimate is limited. Bandwidth estimate can be limited
+// by either delay based bwe, or loss based bwe when it increases/decreases the
+// estimate.
+enum class BandwidthLimitedCause {
+  kLossLimitedBweIncreasing = 0,
+  kLossLimitedBwe = 1,
+  kDelayBasedLimited = 2,
+  kDelayBasedLimitedDelayIncreased = 3,
+  kRttBasedBackOffHighRtt = 4
+};
+
+// This class controls initiation of probing to estimate initial channel
+// capacity. There is also support for probing during a session when max
+// bitrate is adjusted by an application.
+class ProbeController {
+ public:
+  explicit ProbeController(const FieldTrialsView* key_value_config,
+                           RtcEventLog* event_log);
+  ~ProbeController();
+
+  ProbeController(const ProbeController&) = delete;
+  ProbeController& operator=(const ProbeController&) = delete;
+
+  ABSL_MUST_USE_RESULT std::vector<ProbeClusterConfig> SetBitrates(
+      DataRate min_bitrate,
+      DataRate start_bitrate,
+      DataRate max_bitrate,
+      Timestamp at_time);
+
+  // The total bitrate, as opposed to the max bitrate, is the sum of the
+  // configured bitrates for all active streams.
+  ABSL_MUST_USE_RESULT std::vector<ProbeClusterConfig>
+  OnMaxTotalAllocatedBitrate(DataRate max_total_allocated_bitrate,
+                             Timestamp at_time);
+
+  ABSL_MUST_USE_RESULT std::vector<ProbeClusterConfig> OnNetworkAvailability(
+      NetworkAvailability msg);
+
+  ABSL_MUST_USE_RESULT std::vector<ProbeClusterConfig> SetEstimatedBitrate(
+      DataRate bitrate,
+      BandwidthLimitedCause bandwidth_limited_cause,
+      Timestamp at_time);
+
+  void EnablePeriodicAlrProbing(bool enable);
+
+  void SetAlrStartTimeMs(absl::optional<int64_t> alr_start_time);
+  void SetAlrEndedTimeMs(int64_t alr_end_time);
+
+  ABSL_MUST_USE_RESULT std::vector<ProbeClusterConfig> RequestProbe(
+      Timestamp at_time);
+
+  void SetNetworkStateEstimate(webrtc::NetworkStateEstimate estimate);
+
+  // Resets the ProbeController to a state equivalent to as if it was just
+  // created EXCEPT for `enable_periodic_alr_probing_` and
+  // `network_available_`.
+  void Reset(Timestamp at_time);
+
+  ABSL_MUST_USE_RESULT std::vector<ProbeClusterConfig> Process(
+      Timestamp at_time);
+
+ private:
+  enum class State {
+    // Initial state where no probing has been triggered yet.
+    kInit,
+    // Waiting for probing results to continue further probing.
+    kWaitingForProbingResult,
+    // Probing is complete.
+    kProbingComplete,
+  };
+
+  ABSL_MUST_USE_RESULT std::vector<ProbeClusterConfig>
+  InitiateExponentialProbing(Timestamp at_time);
+  ABSL_MUST_USE_RESULT std::vector<ProbeClusterConfig> InitiateProbing(
+      Timestamp now,
+      std::vector<DataRate> bitrates_to_probe,
+      bool probe_further);
+  bool TimeForAlrProbe(Timestamp at_time) const;
+  bool TimeForNetworkStateProbe(Timestamp at_time) const;
+
+  bool network_available_;
+  BandwidthLimitedCause bandwidth_limited_cause_ =
+      BandwidthLimitedCause::kDelayBasedLimited;
+  State state_;
+  DataRate min_bitrate_to_probe_further_ = DataRate::PlusInfinity();
+  Timestamp time_last_probing_initiated_ = Timestamp::MinusInfinity();
+  DataRate estimated_bitrate_ = DataRate::Zero();
+  absl::optional<webrtc::NetworkStateEstimate> network_estimate_;
+  DataRate start_bitrate_ = DataRate::Zero();
+  DataRate max_bitrate_ = DataRate::PlusInfinity();
+  Timestamp last_bwe_drop_probing_time_ = Timestamp::Zero();
+  absl::optional<Timestamp> alr_start_time_;
+  absl::optional<Timestamp> alr_end_time_;
+  bool enable_periodic_alr_probing_;
+  Timestamp time_of_last_large_drop_ = Timestamp::MinusInfinity();
+  DataRate bitrate_before_last_large_drop_ = DataRate::Zero();
+  DataRate max_total_allocated_bitrate_ = DataRate::Zero();
+
+  const bool in_rapid_recovery_experiment_;
+  RtcEventLog* event_log_;
+
+  int32_t next_probe_cluster_id_ = 1;
+
+  ProbeControllerConfig config_;
+};
+
+}  // namespace webrtc
+
+#endif  // MODULES_CONGESTION_CONTROLLER_GOOG_CC_PROBE_CONTROLLER_H_

TMessagesProj/jni/voip/webrtc/modules/congestion_controller/goog_cc/robust_throughput_estimator.cc

@@ -0,0 +1,204 @@
+/*
+ *  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 "modules/congestion_controller/goog_cc/robust_throughput_estimator.h"
+
+#include <stddef.h>
+
+#include <algorithm>
+#include <utility>
+#include <vector>
+
+#include "absl/types/optional.h"
+#include "api/transport/network_types.h"
+#include "api/units/data_rate.h"
+#include "api/units/data_size.h"
+#include "api/units/time_delta.h"
+#include "api/units/timestamp.h"
+#include "modules/congestion_controller/goog_cc/acknowledged_bitrate_estimator_interface.h"
+#include "rtc_base/checks.h"
+#include "rtc_base/logging.h"
+
+namespace webrtc {
+
+RobustThroughputEstimator::RobustThroughputEstimator(
+    const RobustThroughputEstimatorSettings& settings)
+    : settings_(settings),
+      latest_discarded_send_time_(Timestamp::MinusInfinity()) {
+  RTC_DCHECK(settings.enabled);
+}
+
+RobustThroughputEstimator::~RobustThroughputEstimator() {}
+
+bool RobustThroughputEstimator::FirstPacketOutsideWindow() {
+  if (window_.empty())
+    return false;
+  if (window_.size() > settings_.max_window_packets)
+    return true;
+  TimeDelta current_window_duration =
+      window_.back().receive_time - window_.front().receive_time;
+  if (current_window_duration > settings_.max_window_duration)
+    return true;
+  if (window_.size() > settings_.window_packets &&
+      current_window_duration > settings_.min_window_duration) {
+    return true;
+  }
+  return false;
+}
+
+void RobustThroughputEstimator::IncomingPacketFeedbackVector(
+    const std::vector<PacketResult>& packet_feedback_vector) {
+  RTC_DCHECK(std::is_sorted(packet_feedback_vector.begin(),
+                            packet_feedback_vector.end(),
+                            PacketResult::ReceiveTimeOrder()));
+  for (const auto& packet : packet_feedback_vector) {
+    // Ignore packets without valid send or receive times.
+    // (This should not happen in production since lost packets are filtered
+    // out before passing the feedback vector to the throughput estimator.
+    // However, explicitly handling this case makes the estimator more robust
+    // and avoids a hard-to-detect bad state.)
+    if (packet.receive_time.IsInfinite() ||
+        packet.sent_packet.send_time.IsInfinite()) {
+      continue;
+    }
+
+    // Insert the new packet.
+    window_.push_back(packet);
+    window_.back().sent_packet.prior_unacked_data =
+        window_.back().sent_packet.prior_unacked_data *
+        settings_.unacked_weight;
+    // In most cases, receive timestamps should already be in order, but in the
+    // rare case where feedback packets have been reordered, we do some swaps to
+    // ensure that the window is sorted.
+    for (size_t i = window_.size() - 1;
+         i > 0 && window_[i].receive_time < window_[i - 1].receive_time; i--) {
+      std::swap(window_[i], window_[i - 1]);
+    }
+    constexpr TimeDelta kMaxReorderingTime = TimeDelta::Seconds(1);
+    const TimeDelta receive_delta =
+        (window_.back().receive_time - packet.receive_time);
+    if (receive_delta > kMaxReorderingTime) {
+      RTC_LOG(LS_WARNING)
+          << "Severe packet re-ordering or timestamps offset changed: "
+          << receive_delta;
+      window_.clear();
+      latest_discarded_send_time_ = Timestamp::MinusInfinity();
+    }
+  }
+
+  // Remove old packets.
+  while (FirstPacketOutsideWindow()) {
+    latest_discarded_send_time_ = std::max(
+        latest_discarded_send_time_, window_.front().sent_packet.send_time);
+    window_.pop_front();
+  }
+}
+
+absl::optional<DataRate> RobustThroughputEstimator::bitrate() const {
+  if (window_.empty() || window_.size() < settings_.required_packets)
+    return absl::nullopt;
+
+  TimeDelta largest_recv_gap(TimeDelta::Zero());
+  TimeDelta second_largest_recv_gap(TimeDelta::Zero());
+  for (size_t i = 1; i < window_.size(); i++) {
+    // Find receive time gaps.
+    TimeDelta gap = window_[i].receive_time - window_[i - 1].receive_time;
+    if (gap > largest_recv_gap) {
+      second_largest_recv_gap = largest_recv_gap;
+      largest_recv_gap = gap;
+    } else if (gap > second_largest_recv_gap) {
+      second_largest_recv_gap = gap;
+    }
+  }
+
+  Timestamp first_send_time = Timestamp::PlusInfinity();
+  Timestamp last_send_time = Timestamp::MinusInfinity();
+  Timestamp first_recv_time = Timestamp::PlusInfinity();
+  Timestamp last_recv_time = Timestamp::MinusInfinity();
+  DataSize recv_size = DataSize::Bytes(0);
+  DataSize send_size = DataSize::Bytes(0);
+  DataSize first_recv_size = DataSize::Bytes(0);
+  DataSize last_send_size = DataSize::Bytes(0);
+  size_t num_sent_packets_in_window = 0;
+  for (const auto& packet : window_) {
+    if (packet.receive_time < first_recv_time) {
+      first_recv_time = packet.receive_time;
+      first_recv_size =
+          packet.sent_packet.size + packet.sent_packet.prior_unacked_data;
+    }
+    last_recv_time = std::max(last_recv_time, packet.receive_time);
+    recv_size += packet.sent_packet.size;
+    recv_size += packet.sent_packet.prior_unacked_data;
+
+    if (packet.sent_packet.send_time < latest_discarded_send_time_) {
+      // If we have dropped packets from the window that were sent after
+      // this packet, then this packet was reordered. Ignore it from
+      // the send rate computation (since the send time may be very far
+      // in the past, leading to underestimation of the send rate.)
+      // However, ignoring packets creates a risk that we end up without
+      // any packets left to compute a send rate.
+      continue;
+    }
+    if (packet.sent_packet.send_time > last_send_time) {
+      last_send_time = packet.sent_packet.send_time;
+      last_send_size =
+          packet.sent_packet.size + packet.sent_packet.prior_unacked_data;
+    }
+    first_send_time = std::min(first_send_time, packet.sent_packet.send_time);
+
+    send_size += packet.sent_packet.size;
+    send_size += packet.sent_packet.prior_unacked_data;
+    ++num_sent_packets_in_window;
+  }
+
+  // Suppose a packet of size S is sent every T milliseconds.
+  // A window of N packets would contain N*S bytes, but the time difference
+  // between the first and the last packet would only be (N-1)*T. Thus, we
+  // need to remove the size of one packet to get the correct rate of S/T.
+  // Which packet to remove (if the packets have varying sizes),
+  // depends on the network model.
+  // Suppose that 2 packets with sizes s1 and s2, are received at times t1
+  // and t2, respectively. If the packets were transmitted back to back over
+  // a bottleneck with rate capacity r, then we'd expect t2 = t1 + r * s2.
+  // Thus, r = (t2-t1) / s2, so the size of the first packet doesn't affect
+  // the difference between t1 and t2.
+  // Analoguously, if the first packet is sent at time t1 and the sender
+  // paces the packets at rate r, then the second packet can be sent at time
+  // t2 = t1 + r * s1. Thus, the send rate estimate r = (t2-t1) / s1 doesn't
+  // depend on the size of the last packet.
+  recv_size -= first_recv_size;
+  send_size -= last_send_size;
+
+  // Remove the largest gap by replacing it by the second largest gap.
+  // This is to ensure that spurious "delay spikes" (i.e. when the
+  // network stops transmitting packets for a short period, followed
+  // by a burst of delayed packets), don't cause the estimate to drop.
+  // This could cause an overestimation, which we guard against by
+  // never returning an estimate above the send rate.
+  RTC_DCHECK(first_recv_time.IsFinite());
+  RTC_DCHECK(last_recv_time.IsFinite());
+  TimeDelta recv_duration = (last_recv_time - first_recv_time) -
+                            largest_recv_gap + second_largest_recv_gap;
+  recv_duration = std::max(recv_duration, TimeDelta::Millis(1));
+
+  if (num_sent_packets_in_window < settings_.required_packets) {
+    // Too few send times to calculate a reliable send rate.
+    return recv_size / recv_duration;
+  }
+
+  RTC_DCHECK(first_send_time.IsFinite());
+  RTC_DCHECK(last_send_time.IsFinite());
+  TimeDelta send_duration = last_send_time - first_send_time;
+  send_duration = std::max(send_duration, TimeDelta::Millis(1));
+
+  return std::min(send_size / send_duration, recv_size / recv_duration);
+}
+
+}  // namespace webrtc

TMessagesProj/jni/voip/webrtc/modules/congestion_controller/goog_cc/robust_throughput_estimator.h

@@ -0,0 +1,50 @@
+/*
+ *  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_CONGESTION_CONTROLLER_GOOG_CC_ROBUST_THROUGHPUT_ESTIMATOR_H_
+#define MODULES_CONGESTION_CONTROLLER_GOOG_CC_ROBUST_THROUGHPUT_ESTIMATOR_H_
+
+#include <deque>
+#include <vector>
+
+#include "absl/types/optional.h"
+#include "api/transport/network_types.h"
+#include "api/units/data_rate.h"
+#include "api/units/timestamp.h"
+#include "modules/congestion_controller/goog_cc/acknowledged_bitrate_estimator_interface.h"
+
+namespace webrtc {
+
+class RobustThroughputEstimator : public AcknowledgedBitrateEstimatorInterface {
+ public:
+  explicit RobustThroughputEstimator(
+      const RobustThroughputEstimatorSettings& settings);
+  ~RobustThroughputEstimator() override;
+
+  void IncomingPacketFeedbackVector(
+      const std::vector<PacketResult>& packet_feedback_vector) override;
+
+  absl::optional<DataRate> bitrate() const override;
+
+  absl::optional<DataRate> PeekRate() const override { return bitrate(); }
+  void SetAlr(bool /*in_alr*/) override {}
+  void SetAlrEndedTime(Timestamp /*alr_ended_time*/) override {}
+
+ private:
+  bool FirstPacketOutsideWindow();
+
+  const RobustThroughputEstimatorSettings settings_;
+  std::deque<PacketResult> window_;
+  Timestamp latest_discarded_send_time_ = Timestamp::MinusInfinity();
+};
+
+}  // namespace webrtc
+
+#endif  // MODULES_CONGESTION_CONTROLLER_GOOG_CC_ROBUST_THROUGHPUT_ESTIMATOR_H_

TMessagesProj/jni/voip/webrtc/modules/congestion_controller/goog_cc/send_side_bandwidth_estimation.cc

@@ -0,0 +1,711 @@
+/*
+ *  Copyright (c) 2012 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 "modules/congestion_controller/goog_cc/send_side_bandwidth_estimation.h"
+
+#include <algorithm>
+#include <cstdint>
+#include <cstdio>
+#include <limits>
+#include <memory>
+#include <string>
+#include <utility>
+
+#include "absl/strings/match.h"
+#include "absl/types/optional.h"
+#include "api/field_trials_view.h"
+#include "api/network_state_predictor.h"
+#include "api/rtc_event_log/rtc_event_log.h"
+#include "api/transport/network_types.h"
+#include "api/units/data_rate.h"
+#include "api/units/time_delta.h"
+#include "api/units/timestamp.h"
+#include "logging/rtc_event_log/events/rtc_event_bwe_update_loss_based.h"
+#include "modules/congestion_controller/goog_cc/loss_based_bwe_v2.h"
+#include "modules/remote_bitrate_estimator/include/bwe_defines.h"
+#include "rtc_base/checks.h"
+#include "rtc_base/experiments/field_trial_parser.h"
+#include "rtc_base/logging.h"
+#include "system_wrappers/include/field_trial.h"
+#include "system_wrappers/include/metrics.h"
+
+namespace webrtc {
+namespace {
+constexpr TimeDelta kBweIncreaseInterval = TimeDelta::Millis(1000);
+constexpr TimeDelta kBweDecreaseInterval = TimeDelta::Millis(300);
+constexpr TimeDelta kStartPhase = TimeDelta::Millis(2000);
+constexpr TimeDelta kBweConverganceTime = TimeDelta::Millis(20000);
+constexpr int kLimitNumPackets = 20;
+constexpr DataRate kDefaultMaxBitrate = DataRate::BitsPerSec(1000000000);
+constexpr TimeDelta kLowBitrateLogPeriod = TimeDelta::Millis(10000);
+constexpr TimeDelta kRtcEventLogPeriod = TimeDelta::Millis(5000);
+// Expecting that RTCP feedback is sent uniformly within [0.5, 1.5]s intervals.
+constexpr TimeDelta kMaxRtcpFeedbackInterval = TimeDelta::Millis(5000);
+
+constexpr float kDefaultLowLossThreshold = 0.02f;
+constexpr float kDefaultHighLossThreshold = 0.1f;
+constexpr DataRate kDefaultBitrateThreshold = DataRate::Zero();
+
+struct UmaRampUpMetric {
+  const char* metric_name;
+  int bitrate_kbps;
+};
+
+const UmaRampUpMetric kUmaRampupMetrics[] = {
+    {"WebRTC.BWE.RampUpTimeTo500kbpsInMs", 500},
+    {"WebRTC.BWE.RampUpTimeTo1000kbpsInMs", 1000},
+    {"WebRTC.BWE.RampUpTimeTo2000kbpsInMs", 2000}};
+const size_t kNumUmaRampupMetrics =
+    sizeof(kUmaRampupMetrics) / sizeof(kUmaRampupMetrics[0]);
+
+const char kBweLosExperiment[] = "WebRTC-BweLossExperiment";
+
+bool BweLossExperimentIsEnabled() {
+  std::string experiment_string =
+      webrtc::field_trial::FindFullName(kBweLosExperiment);
+  // The experiment is enabled iff the field trial string begins with "Enabled".
+  return absl::StartsWith(experiment_string, "Enabled");
+}
+
+bool ReadBweLossExperimentParameters(float* low_loss_threshold,
+                                     float* high_loss_threshold,
+                                     uint32_t* bitrate_threshold_kbps) {
+  RTC_DCHECK(low_loss_threshold);
+  RTC_DCHECK(high_loss_threshold);
+  RTC_DCHECK(bitrate_threshold_kbps);
+  std::string experiment_string =
+      webrtc::field_trial::FindFullName(kBweLosExperiment);
+  int parsed_values =
+      sscanf(experiment_string.c_str(), "Enabled-%f,%f,%u", low_loss_threshold,
+             high_loss_threshold, bitrate_threshold_kbps);
+  if (parsed_values == 3) {
+    RTC_CHECK_GT(*low_loss_threshold, 0.0f)
+        << "Loss threshold must be greater than 0.";
+    RTC_CHECK_LE(*low_loss_threshold, 1.0f)
+        << "Loss threshold must be less than or equal to 1.";
+    RTC_CHECK_GT(*high_loss_threshold, 0.0f)
+        << "Loss threshold must be greater than 0.";
+    RTC_CHECK_LE(*high_loss_threshold, 1.0f)
+        << "Loss threshold must be less than or equal to 1.";
+    RTC_CHECK_LE(*low_loss_threshold, *high_loss_threshold)
+        << "The low loss threshold must be less than or equal to the high loss "
+           "threshold.";
+    RTC_CHECK_GE(*bitrate_threshold_kbps, 0)
+        << "Bitrate threshold can't be negative.";
+    RTC_CHECK_LT(*bitrate_threshold_kbps,
+                 std::numeric_limits<int>::max() / 1000)
+        << "Bitrate must be smaller enough to avoid overflows.";
+    return true;
+  }
+  RTC_LOG(LS_WARNING) << "Failed to parse parameters for BweLossExperiment "
+                         "experiment from field trial string. Using default.";
+  *low_loss_threshold = kDefaultLowLossThreshold;
+  *high_loss_threshold = kDefaultHighLossThreshold;
+  *bitrate_threshold_kbps = kDefaultBitrateThreshold.kbps();
+  return false;
+}
+}  // namespace
+
+LinkCapacityTracker::LinkCapacityTracker()
+    : tracking_rate("rate", TimeDelta::Seconds(10)) {
+  ParseFieldTrial({&tracking_rate},
+                  field_trial::FindFullName("WebRTC-Bwe-LinkCapacity"));
+}
+
+LinkCapacityTracker::~LinkCapacityTracker() {}
+
+void LinkCapacityTracker::UpdateDelayBasedEstimate(
+    Timestamp at_time,
+    DataRate delay_based_bitrate) {
+  if (delay_based_bitrate < last_delay_based_estimate_) {
+    capacity_estimate_bps_ =
+        std::min(capacity_estimate_bps_, delay_based_bitrate.bps<double>());
+    last_link_capacity_update_ = at_time;
+  }
+  last_delay_based_estimate_ = delay_based_bitrate;
+}
+
+void LinkCapacityTracker::OnStartingRate(DataRate start_rate) {
+  if (last_link_capacity_update_.IsInfinite())
+    capacity_estimate_bps_ = start_rate.bps<double>();
+}
+
+void LinkCapacityTracker::OnRateUpdate(absl::optional<DataRate> acknowledged,
+                                       DataRate target,
+                                       Timestamp at_time) {
+  if (!acknowledged)
+    return;
+  DataRate acknowledged_target = std::min(*acknowledged, target);
+  if (acknowledged_target.bps() > capacity_estimate_bps_) {
+    TimeDelta delta = at_time - last_link_capacity_update_;
+    double alpha = delta.IsFinite() ? exp(-(delta / tracking_rate.Get())) : 0;
+    capacity_estimate_bps_ = alpha * capacity_estimate_bps_ +
+                             (1 - alpha) * acknowledged_target.bps<double>();
+  }
+  last_link_capacity_update_ = at_time;
+}
+
+void LinkCapacityTracker::OnRttBackoff(DataRate backoff_rate,
+                                       Timestamp at_time) {
+  capacity_estimate_bps_ =
+      std::min(capacity_estimate_bps_, backoff_rate.bps<double>());
+  last_link_capacity_update_ = at_time;
+}
+
+DataRate LinkCapacityTracker::estimate() const {
+  return DataRate::BitsPerSec(capacity_estimate_bps_);
+}
+
+RttBasedBackoff::RttBasedBackoff(const FieldTrialsView* key_value_config)
+    : disabled_("Disabled"),
+      configured_limit_("limit", TimeDelta::Seconds(3)),
+      drop_fraction_("fraction", 0.8),
+      drop_interval_("interval", TimeDelta::Seconds(1)),
+      bandwidth_floor_("floor", DataRate::KilobitsPerSec(5)),
+      rtt_limit_(TimeDelta::PlusInfinity()),
+      // By initializing this to plus infinity, we make sure that we never
+      // trigger rtt backoff unless packet feedback is enabled.
+      last_propagation_rtt_update_(Timestamp::PlusInfinity()),
+      last_propagation_rtt_(TimeDelta::Zero()),
+      last_packet_sent_(Timestamp::MinusInfinity()) {
+  ParseFieldTrial({&disabled_, &configured_limit_, &drop_fraction_,
+                   &drop_interval_, &bandwidth_floor_},
+                  key_value_config->Lookup("WebRTC-Bwe-MaxRttLimit"));
+  if (!disabled_) {
+    rtt_limit_ = configured_limit_.Get();
+  }
+}
+
+void RttBasedBackoff::UpdatePropagationRtt(Timestamp at_time,
+                                           TimeDelta propagation_rtt) {
+  last_propagation_rtt_update_ = at_time;
+  last_propagation_rtt_ = propagation_rtt;
+}
+
+bool RttBasedBackoff::IsRttAboveLimit() const {
+  return CorrectedRtt() > rtt_limit_;
+}
+
+TimeDelta RttBasedBackoff::CorrectedRtt() const {
+  // Avoid timeout when no packets are being sent.
+  TimeDelta timeout_correction = std::max(
+      last_packet_sent_ - last_propagation_rtt_update_, TimeDelta::Zero());
+  return timeout_correction + last_propagation_rtt_;
+}
+
+RttBasedBackoff::~RttBasedBackoff() = default;
+
+SendSideBandwidthEstimation::SendSideBandwidthEstimation(
+    const FieldTrialsView* key_value_config, RtcEventLog* event_log)
+    : key_value_config_(key_value_config),
+      rtt_backoff_(key_value_config),
+      lost_packets_since_last_loss_update_(0),
+      expected_packets_since_last_loss_update_(0),
+      current_target_(DataRate::Zero()),
+      last_logged_target_(DataRate::Zero()),
+      min_bitrate_configured_(kCongestionControllerMinBitrate),
+      max_bitrate_configured_(kDefaultMaxBitrate),
+      last_low_bitrate_log_(Timestamp::MinusInfinity()),
+      has_decreased_since_last_fraction_loss_(false),
+      last_loss_feedback_(Timestamp::MinusInfinity()),
+      last_loss_packet_report_(Timestamp::MinusInfinity()),
+      last_fraction_loss_(0),
+      last_logged_fraction_loss_(0),
+      last_round_trip_time_(TimeDelta::Zero()),
+      receiver_limit_(DataRate::PlusInfinity()),
+      delay_based_limit_(DataRate::PlusInfinity()),
+      time_last_decrease_(Timestamp::MinusInfinity()),
+      first_report_time_(Timestamp::MinusInfinity()),
+      initially_lost_packets_(0),
+      bitrate_at_2_seconds_(DataRate::Zero()),
+      uma_update_state_(kNoUpdate),
+      uma_rtt_state_(kNoUpdate),
+      rampup_uma_stats_updated_(kNumUmaRampupMetrics, false),
+      event_log_(event_log),
+      last_rtc_event_log_(Timestamp::MinusInfinity()),
+      low_loss_threshold_(kDefaultLowLossThreshold),
+      high_loss_threshold_(kDefaultHighLossThreshold),
+      bitrate_threshold_(kDefaultBitrateThreshold),
+      loss_based_bandwidth_estimator_v1_(key_value_config),
+      loss_based_bandwidth_estimator_v2_(new LossBasedBweV2(key_value_config)),
+      loss_based_state_(LossBasedState::kDelayBasedEstimate),
+      disable_receiver_limit_caps_only_("Disabled") {
+  RTC_DCHECK(event_log);
+  if (BweLossExperimentIsEnabled()) {
+    uint32_t bitrate_threshold_kbps;
+    if (ReadBweLossExperimentParameters(&low_loss_threshold_,
+                                        &high_loss_threshold_,
+                                        &bitrate_threshold_kbps)) {
+      RTC_LOG(LS_INFO) << "Enabled BweLossExperiment with parameters "
+                       << low_loss_threshold_ << ", " << high_loss_threshold_
+                       << ", " << bitrate_threshold_kbps;
+      bitrate_threshold_ = DataRate::KilobitsPerSec(bitrate_threshold_kbps);
+    }
+  }
+  ParseFieldTrial({&disable_receiver_limit_caps_only_},
+                  key_value_config->Lookup("WebRTC-Bwe-ReceiverLimitCapsOnly"));
+  if (LossBasedBandwidthEstimatorV2Enabled()) {
+    loss_based_bandwidth_estimator_v2_->SetMinMaxBitrate(
+        min_bitrate_configured_, max_bitrate_configured_);
+  }
+}
+
+SendSideBandwidthEstimation::~SendSideBandwidthEstimation() {}
+
+void SendSideBandwidthEstimation::OnRouteChange() {
+  lost_packets_since_last_loss_update_ = 0;
+  expected_packets_since_last_loss_update_ = 0;
+  current_target_ = DataRate::Zero();
+  min_bitrate_configured_ = kCongestionControllerMinBitrate;
+  max_bitrate_configured_ = kDefaultMaxBitrate;
+  last_low_bitrate_log_ = Timestamp::MinusInfinity();
+  has_decreased_since_last_fraction_loss_ = false;
+  last_loss_feedback_ = Timestamp::MinusInfinity();
+  last_loss_packet_report_ = Timestamp::MinusInfinity();
+  last_fraction_loss_ = 0;
+  last_logged_fraction_loss_ = 0;
+  last_round_trip_time_ = TimeDelta::Zero();
+  receiver_limit_ = DataRate::PlusInfinity();
+  delay_based_limit_ = DataRate::PlusInfinity();
+  time_last_decrease_ = Timestamp::MinusInfinity();
+  first_report_time_ = Timestamp::MinusInfinity();
+  initially_lost_packets_ = 0;
+  bitrate_at_2_seconds_ = DataRate::Zero();
+  uma_update_state_ = kNoUpdate;
+  uma_rtt_state_ = kNoUpdate;
+  last_rtc_event_log_ = Timestamp::MinusInfinity();
+  if (LossBasedBandwidthEstimatorV2Enabled() &&
+      loss_based_bandwidth_estimator_v2_->UseInStartPhase()) {
+    loss_based_bandwidth_estimator_v2_.reset(
+        new LossBasedBweV2(key_value_config_));
+  }
+}
+
+void SendSideBandwidthEstimation::SetBitrates(
+    absl::optional<DataRate> send_bitrate,
+    DataRate min_bitrate,
+    DataRate max_bitrate,
+    Timestamp at_time) {
+  SetMinMaxBitrate(min_bitrate, max_bitrate);
+  if (send_bitrate) {
+    link_capacity_.OnStartingRate(*send_bitrate);
+    SetSendBitrate(*send_bitrate, at_time);
+  }
+}
+
+void SendSideBandwidthEstimation::SetSendBitrate(DataRate bitrate,
+                                                 Timestamp at_time) {
+  RTC_DCHECK_GT(bitrate, DataRate::Zero());
+  // Reset to avoid being capped by the estimate.
+  delay_based_limit_ = DataRate::PlusInfinity();
+  UpdateTargetBitrate(bitrate, at_time);
+  // Clear last sent bitrate history so the new value can be used directly
+  // and not capped.
+  min_bitrate_history_.clear();
+}
+
+void SendSideBandwidthEstimation::SetMinMaxBitrate(DataRate min_bitrate,
+                                                   DataRate max_bitrate) {
+  min_bitrate_configured_ =
+      std::max(min_bitrate, kCongestionControllerMinBitrate);
+  if (max_bitrate > DataRate::Zero() && max_bitrate.IsFinite()) {
+    max_bitrate_configured_ = std::max(min_bitrate_configured_, max_bitrate);
+  } else {
+    max_bitrate_configured_ = kDefaultMaxBitrate;
+  }
+  loss_based_bandwidth_estimator_v2_->SetMinMaxBitrate(min_bitrate_configured_,
+                                                       max_bitrate_configured_);
+}
+
+int SendSideBandwidthEstimation::GetMinBitrate() const {
+  return min_bitrate_configured_.bps<int>();
+}
+
+DataRate SendSideBandwidthEstimation::target_rate() const {
+  DataRate target = current_target_;
+  if (!disable_receiver_limit_caps_only_)
+    target = std::min(target, receiver_limit_);
+  return std::max(min_bitrate_configured_, target);
+}
+
+LossBasedState SendSideBandwidthEstimation::loss_based_state() const {
+  return loss_based_state_;
+}
+
+bool SendSideBandwidthEstimation::IsRttAboveLimit() const {
+  return rtt_backoff_.IsRttAboveLimit();
+}
+
+DataRate SendSideBandwidthEstimation::GetEstimatedLinkCapacity() const {
+  return link_capacity_.estimate();
+}
+
+void SendSideBandwidthEstimation::UpdateReceiverEstimate(Timestamp at_time,
+                                                         DataRate bandwidth) {
+  // TODO(srte): Ensure caller passes PlusInfinity, not zero, to represent no
+  // limitation.
+  receiver_limit_ = bandwidth.IsZero() ? DataRate::PlusInfinity() : bandwidth;
+  ApplyTargetLimits(at_time);
+}
+
+void SendSideBandwidthEstimation::UpdateDelayBasedEstimate(Timestamp at_time,
+                                                           DataRate bitrate) {
+  link_capacity_.UpdateDelayBasedEstimate(at_time, bitrate);
+  // TODO(srte): Ensure caller passes PlusInfinity, not zero, to represent no
+  // limitation.
+  delay_based_limit_ = bitrate.IsZero() ? DataRate::PlusInfinity() : bitrate;
+  ApplyTargetLimits(at_time);
+}
+
+void SendSideBandwidthEstimation::SetAcknowledgedRate(
+    absl::optional<DataRate> acknowledged_rate,
+    Timestamp at_time) {
+  acknowledged_rate_ = acknowledged_rate;
+  if (!acknowledged_rate.has_value()) {
+    return;
+  }
+  if (LossBasedBandwidthEstimatorV1Enabled()) {
+    loss_based_bandwidth_estimator_v1_.UpdateAcknowledgedBitrate(
+        *acknowledged_rate, at_time);
+  }
+  if (LossBasedBandwidthEstimatorV2Enabled()) {
+    loss_based_bandwidth_estimator_v2_->SetAcknowledgedBitrate(
+        *acknowledged_rate);
+  }
+}
+
+void SendSideBandwidthEstimation::UpdateLossBasedEstimator(
+    const TransportPacketsFeedback& report,
+    BandwidthUsage delay_detector_state,
+    absl::optional<DataRate> probe_bitrate,
+    bool in_alr) {
+  if (LossBasedBandwidthEstimatorV1Enabled()) {
+    loss_based_bandwidth_estimator_v1_.UpdateLossStatistics(
+        report.packet_feedbacks, report.feedback_time);
+  }
+  if (LossBasedBandwidthEstimatorV2Enabled()) {
+    loss_based_bandwidth_estimator_v2_->UpdateBandwidthEstimate(
+        report.packet_feedbacks, delay_based_limit_, in_alr);
+    UpdateEstimate(report.feedback_time);
+  }
+}
+
+void SendSideBandwidthEstimation::UpdatePacketsLost(int64_t packets_lost,
+                                                    int64_t number_of_packets,
+                                                    Timestamp at_time) {
+  last_loss_feedback_ = at_time;
+  if (first_report_time_.IsInfinite())
+    first_report_time_ = at_time;
+
+  // Check sequence number diff and weight loss report
+  if (number_of_packets > 0) {
+    int64_t expected =
+        expected_packets_since_last_loss_update_ + number_of_packets;
+
+    // Don't generate a loss rate until it can be based on enough packets.
+    if (expected < kLimitNumPackets) {
+      // Accumulate reports.
+      expected_packets_since_last_loss_update_ = expected;
+      lost_packets_since_last_loss_update_ += packets_lost;
+      return;
+    }
+
+    has_decreased_since_last_fraction_loss_ = false;
+    int64_t lost_q8 =
+        std::max<int64_t>(lost_packets_since_last_loss_update_ + packets_lost,
+                          0)
+        << 8;
+    last_fraction_loss_ = std::min<int>(lost_q8 / expected, 255);
+
+    // Reset accumulators.
+    lost_packets_since_last_loss_update_ = 0;
+    expected_packets_since_last_loss_update_ = 0;
+    last_loss_packet_report_ = at_time;
+    UpdateEstimate(at_time);
+  }
+
+  UpdateUmaStatsPacketsLost(at_time, packets_lost);
+}
+
+void SendSideBandwidthEstimation::UpdateUmaStatsPacketsLost(Timestamp at_time,
+                                                            int packets_lost) {
+  DataRate bitrate_kbps =
+      DataRate::KilobitsPerSec((current_target_.bps() + 500) / 1000);
+  for (size_t i = 0; i < kNumUmaRampupMetrics; ++i) {
+    if (!rampup_uma_stats_updated_[i] &&
+        bitrate_kbps.kbps() >= kUmaRampupMetrics[i].bitrate_kbps) {
+      RTC_HISTOGRAMS_COUNTS_100000(i, kUmaRampupMetrics[i].metric_name,
+                                   (at_time - first_report_time_).ms());
+      rampup_uma_stats_updated_[i] = true;
+    }
+  }
+  if (IsInStartPhase(at_time)) {
+    initially_lost_packets_ += packets_lost;
+  } else if (uma_update_state_ == kNoUpdate) {
+    uma_update_state_ = kFirstDone;
+    bitrate_at_2_seconds_ = bitrate_kbps;
+    RTC_HISTOGRAM_COUNTS("WebRTC.BWE.InitiallyLostPackets",
+                         initially_lost_packets_, 0, 100, 50);
+    RTC_HISTOGRAM_COUNTS("WebRTC.BWE.InitialBandwidthEstimate",
+                         bitrate_at_2_seconds_.kbps(), 0, 2000, 50);
+  } else if (uma_update_state_ == kFirstDone &&
+             at_time - first_report_time_ >= kBweConverganceTime) {
+    uma_update_state_ = kDone;
+    int bitrate_diff_kbps = std::max(
+        bitrate_at_2_seconds_.kbps<int>() - bitrate_kbps.kbps<int>(), 0);
+    RTC_HISTOGRAM_COUNTS("WebRTC.BWE.InitialVsConvergedDiff", bitrate_diff_kbps,
+                         0, 2000, 50);
+  }
+}
+
+void SendSideBandwidthEstimation::UpdateRtt(TimeDelta rtt, Timestamp at_time) {
+  // Update RTT if we were able to compute an RTT based on this RTCP.
+  // FlexFEC doesn't send RTCP SR, which means we won't be able to compute RTT.
+  if (rtt > TimeDelta::Zero())
+    last_round_trip_time_ = rtt;
+
+  if (!IsInStartPhase(at_time) && uma_rtt_state_ == kNoUpdate) {
+    uma_rtt_state_ = kDone;
+    RTC_HISTOGRAM_COUNTS("WebRTC.BWE.InitialRtt", rtt.ms<int>(), 0, 2000, 50);
+  }
+}
+
+void SendSideBandwidthEstimation::UpdateEstimate(Timestamp at_time) {
+  if (rtt_backoff_.IsRttAboveLimit()) {
+    if (at_time - time_last_decrease_ >= rtt_backoff_.drop_interval_ &&
+        current_target_ > rtt_backoff_.bandwidth_floor_) {
+      time_last_decrease_ = at_time;
+      DataRate new_bitrate =
+          std::max(current_target_ * rtt_backoff_.drop_fraction_,
+                   rtt_backoff_.bandwidth_floor_.Get());
+      link_capacity_.OnRttBackoff(new_bitrate, at_time);
+      UpdateTargetBitrate(new_bitrate, at_time);
+      return;
+    }
+    // TODO(srte): This is likely redundant in most cases.
+    ApplyTargetLimits(at_time);
+    return;
+  }
+
+  // We trust the REMB and/or delay-based estimate during the first 2 seconds if
+  // we haven't had any packet loss reported, to allow startup bitrate probing.
+  if (last_fraction_loss_ == 0 && IsInStartPhase(at_time) &&
+      !loss_based_bandwidth_estimator_v2_->ReadyToUseInStartPhase()) {
+    DataRate new_bitrate = current_target_;
+    // TODO(srte): We should not allow the new_bitrate to be larger than the
+    // receiver limit here.
+    if (receiver_limit_.IsFinite())
+      new_bitrate = std::max(receiver_limit_, new_bitrate);
+    if (delay_based_limit_.IsFinite())
+      new_bitrate = std::max(delay_based_limit_, new_bitrate);
+    if (LossBasedBandwidthEstimatorV1Enabled()) {
+      loss_based_bandwidth_estimator_v1_.Initialize(new_bitrate);
+    }
+
+    if (new_bitrate != current_target_) {
+      min_bitrate_history_.clear();
+      if (LossBasedBandwidthEstimatorV1Enabled()) {
+        min_bitrate_history_.push_back(std::make_pair(at_time, new_bitrate));
+      } else {
+        min_bitrate_history_.push_back(
+            std::make_pair(at_time, current_target_));
+      }
+      UpdateTargetBitrate(new_bitrate, at_time);
+      return;
+    }
+  }
+  UpdateMinHistory(at_time);
+  if (last_loss_packet_report_.IsInfinite()) {
+    // No feedback received.
+    // TODO(srte): This is likely redundant in most cases.
+    ApplyTargetLimits(at_time);
+    return;
+  }
+
+  if (LossBasedBandwidthEstimatorV1ReadyForUse()) {
+    DataRate new_bitrate = loss_based_bandwidth_estimator_v1_.Update(
+        at_time, min_bitrate_history_.front().second, delay_based_limit_,
+        last_round_trip_time_);
+    UpdateTargetBitrate(new_bitrate, at_time);
+    return;
+  }
+
+  if (LossBasedBandwidthEstimatorV2ReadyForUse()) {
+    LossBasedBweV2::Result result =
+        loss_based_bandwidth_estimator_v2_->GetLossBasedResult();
+    loss_based_state_ = result.state;
+    UpdateTargetBitrate(result.bandwidth_estimate, at_time);
+    return;
+  }
+
+  TimeDelta time_since_loss_packet_report = at_time - last_loss_packet_report_;
+  if (time_since_loss_packet_report < 1.2 * kMaxRtcpFeedbackInterval) {
+    // We only care about loss above a given bitrate threshold.
+    float loss = last_fraction_loss_ / 256.0f;
+    // We only make decisions based on loss when the bitrate is above a
+    // threshold. This is a crude way of handling loss which is uncorrelated
+    // to congestion.
+    if (current_target_ < bitrate_threshold_ || loss <= low_loss_threshold_) {
+      // Loss < 2%: Increase rate by 8% of the min bitrate in the last
+      // kBweIncreaseInterval.
+      // Note that by remembering the bitrate over the last second one can
+      // rampup up one second faster than if only allowed to start ramping
+      // at 8% per second rate now. E.g.:
+      //   If sending a constant 100kbps it can rampup immediately to 108kbps
+      //   whenever a receiver report is received with lower packet loss.
+      //   If instead one would do: current_bitrate_ *= 1.08^(delta time),
+      //   it would take over one second since the lower packet loss to achieve
+      //   108kbps.
+      DataRate new_bitrate = DataRate::BitsPerSec(
+          min_bitrate_history_.front().second.bps() * 1.08 + 0.5);
+
+      // Add 1 kbps extra, just to make sure that we do not get stuck
+      // (gives a little extra increase at low rates, negligible at higher
+      // rates).
+      new_bitrate += DataRate::BitsPerSec(1000);
+      UpdateTargetBitrate(new_bitrate, at_time);
+      return;
+    } else if (current_target_ > bitrate_threshold_) {
+      if (loss <= high_loss_threshold_) {
+        // Loss between 2% - 10%: Do nothing.
+      } else {
+        // Loss > 10%: Limit the rate decreases to once a kBweDecreaseInterval
+        // + rtt.
+        if (!has_decreased_since_last_fraction_loss_ &&
+            (at_time - time_last_decrease_) >=
+                (kBweDecreaseInterval + last_round_trip_time_)) {
+          time_last_decrease_ = at_time;
+
+          // Reduce rate:
+          //   newRate = rate * (1 - 0.5*lossRate);
+          //   where packetLoss = 256*lossRate;
+          DataRate new_bitrate = DataRate::BitsPerSec(
+              (current_target_.bps() *
+               static_cast<double>(512 - last_fraction_loss_)) /
+              512.0);
+          has_decreased_since_last_fraction_loss_ = true;
+          UpdateTargetBitrate(new_bitrate, at_time);
+          return;
+        }
+      }
+    }
+  }
+  // TODO(srte): This is likely redundant in most cases.
+  ApplyTargetLimits(at_time);
+}
+
+void SendSideBandwidthEstimation::UpdatePropagationRtt(
+    Timestamp at_time,
+    TimeDelta propagation_rtt) {
+  rtt_backoff_.UpdatePropagationRtt(at_time, propagation_rtt);
+}
+
+void SendSideBandwidthEstimation::OnSentPacket(const SentPacket& sent_packet) {
+  // Only feedback-triggering packets will be reported here.
+  rtt_backoff_.last_packet_sent_ = sent_packet.send_time;
+}
+
+bool SendSideBandwidthEstimation::IsInStartPhase(Timestamp at_time) const {
+  return first_report_time_.IsInfinite() ||
+         at_time - first_report_time_ < kStartPhase;
+}
+
+void SendSideBandwidthEstimation::UpdateMinHistory(Timestamp at_time) {
+  // Remove old data points from history.
+  // Since history precision is in ms, add one so it is able to increase
+  // bitrate if it is off by as little as 0.5ms.
+  while (!min_bitrate_history_.empty() &&
+         at_time - min_bitrate_history_.front().first + TimeDelta::Millis(1) >
+             kBweIncreaseInterval) {
+    min_bitrate_history_.pop_front();
+  }
+
+  // Typical minimum sliding-window algorithm: Pop values higher than current
+  // bitrate before pushing it.
+  while (!min_bitrate_history_.empty() &&
+         current_target_ <= min_bitrate_history_.back().second) {
+    min_bitrate_history_.pop_back();
+  }
+
+  min_bitrate_history_.push_back(std::make_pair(at_time, current_target_));
+}
+
+DataRate SendSideBandwidthEstimation::GetUpperLimit() const {
+  DataRate upper_limit = delay_based_limit_;
+  if (disable_receiver_limit_caps_only_)
+    upper_limit = std::min(upper_limit, receiver_limit_);
+  return std::min(upper_limit, max_bitrate_configured_);
+}
+
+void SendSideBandwidthEstimation::MaybeLogLowBitrateWarning(DataRate bitrate,
+                                                            Timestamp at_time) {
+  if (at_time - last_low_bitrate_log_ > kLowBitrateLogPeriod) {
+    RTC_LOG(LS_WARNING) << "Estimated available bandwidth " << ToString(bitrate)
+                        << " is below configured min bitrate "
+                        << ToString(min_bitrate_configured_) << ".";
+    last_low_bitrate_log_ = at_time;
+  }
+}
+
+void SendSideBandwidthEstimation::MaybeLogLossBasedEvent(Timestamp at_time) {
+  if (current_target_ != last_logged_target_ ||
+      last_fraction_loss_ != last_logged_fraction_loss_ ||
+      at_time - last_rtc_event_log_ > kRtcEventLogPeriod) {
+    event_log_->Log(std::make_unique<RtcEventBweUpdateLossBased>(
+        current_target_.bps(), last_fraction_loss_,
+        expected_packets_since_last_loss_update_));
+    last_logged_fraction_loss_ = last_fraction_loss_;
+    last_logged_target_ = current_target_;
+    last_rtc_event_log_ = at_time;
+  }
+}
+
+void SendSideBandwidthEstimation::UpdateTargetBitrate(DataRate new_bitrate,
+                                                      Timestamp at_time) {
+  new_bitrate = std::min(new_bitrate, GetUpperLimit());
+  if (new_bitrate < min_bitrate_configured_) {
+    MaybeLogLowBitrateWarning(new_bitrate, at_time);
+    new_bitrate = min_bitrate_configured_;
+  }
+  current_target_ = new_bitrate;
+  MaybeLogLossBasedEvent(at_time);
+  link_capacity_.OnRateUpdate(acknowledged_rate_, current_target_, at_time);
+}
+
+void SendSideBandwidthEstimation::ApplyTargetLimits(Timestamp at_time) {
+  UpdateTargetBitrate(current_target_, at_time);
+}
+
+bool SendSideBandwidthEstimation::LossBasedBandwidthEstimatorV1Enabled() const {
+  return loss_based_bandwidth_estimator_v1_.Enabled() &&
+         !LossBasedBandwidthEstimatorV2Enabled();
+}
+
+bool SendSideBandwidthEstimation::LossBasedBandwidthEstimatorV1ReadyForUse()
+    const {
+  return LossBasedBandwidthEstimatorV1Enabled() &&
+         loss_based_bandwidth_estimator_v1_.InUse();
+}
+
+bool SendSideBandwidthEstimation::LossBasedBandwidthEstimatorV2Enabled() const {
+  return loss_based_bandwidth_estimator_v2_->IsEnabled();
+}
+
+bool SendSideBandwidthEstimation::LossBasedBandwidthEstimatorV2ReadyForUse()
+    const {
+  return loss_based_bandwidth_estimator_v2_->IsReady();
+}
+
+bool SendSideBandwidthEstimation::PaceAtLossBasedEstimate() const {
+  return LossBasedBandwidthEstimatorV2ReadyForUse() &&
+         loss_based_bandwidth_estimator_v2_->PaceAtLossBasedEstimate();
+}
+
+}  // namespace webrtc

TMessagesProj/jni/voip/webrtc/modules/congestion_controller/goog_cc/send_side_bandwidth_estimation.h

@@ -0,0 +1,219 @@
+/*
+ *  Copyright (c) 2012 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.
+ *
+ *  FEC and NACK added bitrate is handled outside class
+ */
+
+#ifndef MODULES_CONGESTION_CONTROLLER_GOOG_CC_SEND_SIDE_BANDWIDTH_ESTIMATION_H_
+#define MODULES_CONGESTION_CONTROLLER_GOOG_CC_SEND_SIDE_BANDWIDTH_ESTIMATION_H_
+
+#include <stdint.h>
+
+#include <deque>
+#include <memory>
+#include <utility>
+#include <vector>
+
+#include "absl/types/optional.h"
+#include "api/field_trials_view.h"
+#include "api/network_state_predictor.h"
+#include "api/transport/network_types.h"
+#include "api/units/data_rate.h"
+#include "api/units/time_delta.h"
+#include "api/units/timestamp.h"
+#include "modules/congestion_controller/goog_cc/loss_based_bandwidth_estimation.h"
+#include "modules/congestion_controller/goog_cc/loss_based_bwe_v2.h"
+#include "rtc_base/experiments/field_trial_parser.h"
+
+namespace webrtc {
+
+class RtcEventLog;
+
+class LinkCapacityTracker {
+ public:
+  LinkCapacityTracker();
+  ~LinkCapacityTracker();
+  // Call when a new delay-based estimate is available.
+  void UpdateDelayBasedEstimate(Timestamp at_time,
+                                DataRate delay_based_bitrate);
+  void OnStartingRate(DataRate start_rate);
+  void OnRateUpdate(absl::optional<DataRate> acknowledged,
+                    DataRate target,
+                    Timestamp at_time);
+  void OnRttBackoff(DataRate backoff_rate, Timestamp at_time);
+  DataRate estimate() const;
+
+ private:
+  FieldTrialParameter<TimeDelta> tracking_rate;
+  double capacity_estimate_bps_ = 0;
+  Timestamp last_link_capacity_update_ = Timestamp::MinusInfinity();
+  DataRate last_delay_based_estimate_ = DataRate::PlusInfinity();
+};
+
+class RttBasedBackoff {
+ public:
+  explicit RttBasedBackoff(const FieldTrialsView* key_value_config);
+  ~RttBasedBackoff();
+  void UpdatePropagationRtt(Timestamp at_time, TimeDelta propagation_rtt);
+  bool IsRttAboveLimit() const;
+
+  FieldTrialFlag disabled_;
+  FieldTrialParameter<TimeDelta> configured_limit_;
+  FieldTrialParameter<double> drop_fraction_;
+  FieldTrialParameter<TimeDelta> drop_interval_;
+  FieldTrialParameter<DataRate> bandwidth_floor_;
+
+ public:
+  TimeDelta rtt_limit_;
+  Timestamp last_propagation_rtt_update_;
+  TimeDelta last_propagation_rtt_;
+  Timestamp last_packet_sent_;
+
+ private:
+  TimeDelta CorrectedRtt() const;
+};
+
+class SendSideBandwidthEstimation {
+ public:
+  SendSideBandwidthEstimation() = delete;
+  SendSideBandwidthEstimation(const FieldTrialsView* key_value_config,
+                              RtcEventLog* event_log);
+  ~SendSideBandwidthEstimation();
+
+  void OnRouteChange();
+
+  DataRate target_rate() const;
+  LossBasedState loss_based_state() const;
+  // Return whether the current rtt is higher than the rtt limited configured in
+  // RttBasedBackoff.
+  bool IsRttAboveLimit() const;
+  uint8_t fraction_loss() const { return last_fraction_loss_; }
+  TimeDelta round_trip_time() const { return last_round_trip_time_; }
+
+  DataRate GetEstimatedLinkCapacity() const;
+  // Call periodically to update estimate.
+  void UpdateEstimate(Timestamp at_time);
+  void OnSentPacket(const SentPacket& sent_packet);
+  void UpdatePropagationRtt(Timestamp at_time, TimeDelta propagation_rtt);
+
+  // Call when we receive a RTCP message with TMMBR or REMB.
+  void UpdateReceiverEstimate(Timestamp at_time, DataRate bandwidth);
+
+  // Call when a new delay-based estimate is available.
+  void UpdateDelayBasedEstimate(Timestamp at_time, DataRate bitrate);
+
+  // Call when we receive a RTCP message with a ReceiveBlock.
+  void UpdatePacketsLost(int64_t packets_lost,
+                         int64_t number_of_packets,
+                         Timestamp at_time);
+
+  // Call when we receive a RTCP message with a ReceiveBlock.
+  void UpdateRtt(TimeDelta rtt, Timestamp at_time);
+
+  void SetBitrates(absl::optional<DataRate> send_bitrate,
+                   DataRate min_bitrate,
+                   DataRate max_bitrate,
+                   Timestamp at_time);
+  void SetSendBitrate(DataRate bitrate, Timestamp at_time);
+  void SetMinMaxBitrate(DataRate min_bitrate, DataRate max_bitrate);
+  int GetMinBitrate() const;
+  void SetAcknowledgedRate(absl::optional<DataRate> acknowledged_rate,
+                           Timestamp at_time);
+  void UpdateLossBasedEstimator(const TransportPacketsFeedback& report,
+                                BandwidthUsage delay_detector_state,
+                                absl::optional<DataRate> probe_bitrate,
+                                bool in_alr);
+  bool PaceAtLossBasedEstimate() const;
+
+ private:
+  friend class GoogCcStatePrinter;
+
+  enum UmaState { kNoUpdate, kFirstDone, kDone };
+
+  bool IsInStartPhase(Timestamp at_time) const;
+
+  void UpdateUmaStatsPacketsLost(Timestamp at_time, int packets_lost);
+
+  // Updates history of min bitrates.
+  // After this method returns min_bitrate_history_.front().second contains the
+  // min bitrate used during last kBweIncreaseIntervalMs.
+  void UpdateMinHistory(Timestamp at_time);
+
+  // Gets the upper limit for the target bitrate. This is the minimum of the
+  // delay based limit, the receiver limit and the loss based controller limit.
+  DataRate GetUpperLimit() const;
+  // Prints a warning if `bitrate` if sufficiently long time has past since last
+  // warning.
+  void MaybeLogLowBitrateWarning(DataRate bitrate, Timestamp at_time);
+  // Stores an update to the event log if the loss rate has changed, the target
+  // has changed, or sufficient time has passed since last stored event.
+  void MaybeLogLossBasedEvent(Timestamp at_time);
+
+  // Cap `bitrate` to [min_bitrate_configured_, max_bitrate_configured_] and
+  // set `current_bitrate_` to the capped value and updates the event log.
+  void UpdateTargetBitrate(DataRate bitrate, Timestamp at_time);
+  // Applies lower and upper bounds to the current target rate.
+  // TODO(srte): This seems to be called even when limits haven't changed, that
+  // should be cleaned up.
+  void ApplyTargetLimits(Timestamp at_time);
+
+  bool LossBasedBandwidthEstimatorV1Enabled() const;
+  bool LossBasedBandwidthEstimatorV2Enabled() const;
+
+  bool LossBasedBandwidthEstimatorV1ReadyForUse() const;
+  bool LossBasedBandwidthEstimatorV2ReadyForUse() const;
+
+  const FieldTrialsView* key_value_config_;
+  RttBasedBackoff rtt_backoff_;
+  LinkCapacityTracker link_capacity_;
+
+  std::deque<std::pair<Timestamp, DataRate> > min_bitrate_history_;
+
+  // incoming filters
+  int lost_packets_since_last_loss_update_;
+  int expected_packets_since_last_loss_update_;
+
+  absl::optional<DataRate> acknowledged_rate_;
+  DataRate current_target_;
+  DataRate last_logged_target_;
+  DataRate min_bitrate_configured_;
+  DataRate max_bitrate_configured_;
+  Timestamp last_low_bitrate_log_;
+
+  bool has_decreased_since_last_fraction_loss_;
+  Timestamp last_loss_feedback_;
+  Timestamp last_loss_packet_report_;
+  uint8_t last_fraction_loss_;
+  uint8_t last_logged_fraction_loss_;
+  TimeDelta last_round_trip_time_;
+
+  // The max bitrate as set by the receiver in the call. This is typically
+  // signalled using the REMB RTCP message and is used when we don't have any
+  // send side delay based estimate.
+  DataRate receiver_limit_;
+  DataRate delay_based_limit_;
+  Timestamp time_last_decrease_;
+  Timestamp first_report_time_;
+  int initially_lost_packets_;
+  DataRate bitrate_at_2_seconds_;
+  UmaState uma_update_state_;
+  UmaState uma_rtt_state_;
+  std::vector<bool> rampup_uma_stats_updated_;
+  RtcEventLog* const event_log_;
+  Timestamp last_rtc_event_log_;
+  float low_loss_threshold_;
+  float high_loss_threshold_;
+  DataRate bitrate_threshold_;
+  LossBasedBandwidthEstimation loss_based_bandwidth_estimator_v1_;
+  std::unique_ptr<LossBasedBweV2> loss_based_bandwidth_estimator_v2_;
+  LossBasedState loss_based_state_;
+  FieldTrialFlag disable_receiver_limit_caps_only_;
+};
+}  // namespace webrtc
+#endif  // MODULES_CONGESTION_CONTROLLER_GOOG_CC_SEND_SIDE_BANDWIDTH_ESTIMATION_H_

TMessagesProj/jni/voip/webrtc/modules/congestion_controller/goog_cc/trendline_estimator.cc

@@ -0,0 +1,339 @@
+/*
+ *  Copyright (c) 2016 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 "modules/congestion_controller/goog_cc/trendline_estimator.h"
+
+#include <math.h>
+
+#include <algorithm>
+#include <cstddef>
+#include <cstdint>
+#include <cstdio>
+#include <deque>
+#include <memory>
+#include <string>
+#include <utility>
+
+#include "absl/strings/match.h"
+#include "absl/types/optional.h"
+#include "api/field_trials_view.h"
+#include "api/network_state_predictor.h"
+#include "modules/remote_bitrate_estimator/test/bwe_test_logging.h"
+#include "rtc_base/checks.h"
+#include "rtc_base/experiments/struct_parameters_parser.h"
+#include "rtc_base/logging.h"
+#include "rtc_base/numerics/safe_minmax.h"
+
+namespace webrtc {
+
+namespace {
+
+// Parameters for linear least squares fit of regression line to noisy data.
+constexpr double kDefaultTrendlineSmoothingCoeff = 0.9;
+constexpr double kDefaultTrendlineThresholdGain = 4.0;
+const char kBweWindowSizeInPacketsExperiment[] =
+    "WebRTC-BweWindowSizeInPackets";
+
+size_t ReadTrendlineFilterWindowSize(const FieldTrialsView* key_value_config) {
+  std::string experiment_string =
+      key_value_config->Lookup(kBweWindowSizeInPacketsExperiment);
+  size_t window_size;
+  int parsed_values =
+      sscanf(experiment_string.c_str(), "Enabled-%zu", &window_size);
+  if (parsed_values == 1) {
+    if (window_size > 1)
+      return window_size;
+    RTC_LOG(LS_WARNING) << "Window size must be greater than 1.";
+  }
+  RTC_LOG(LS_WARNING) << "Failed to parse parameters for BweWindowSizeInPackets"
+                         " experiment from field trial string. Using default.";
+  return TrendlineEstimatorSettings::kDefaultTrendlineWindowSize;
+}
+
+absl::optional<double> LinearFitSlope(
+    const std::deque<TrendlineEstimator::PacketTiming>& packets) {
+  RTC_DCHECK(packets.size() >= 2);
+  // Compute the "center of mass".
+  double sum_x = 0;
+  double sum_y = 0;
+  for (const auto& packet : packets) {
+    sum_x += packet.arrival_time_ms;
+    sum_y += packet.smoothed_delay_ms;
+  }
+  double x_avg = sum_x / packets.size();
+  double y_avg = sum_y / packets.size();
+  // Compute the slope k = \sum (x_i-x_avg)(y_i-y_avg) / \sum (x_i-x_avg)^2
+  double numerator = 0;
+  double denominator = 0;
+  for (const auto& packet : packets) {
+    double x = packet.arrival_time_ms;
+    double y = packet.smoothed_delay_ms;
+    numerator += (x - x_avg) * (y - y_avg);
+    denominator += (x - x_avg) * (x - x_avg);
+  }
+  if (denominator == 0)
+    return absl::nullopt;
+  return numerator / denominator;
+}
+
+absl::optional<double> ComputeSlopeCap(
+    const std::deque<TrendlineEstimator::PacketTiming>& packets,
+    const TrendlineEstimatorSettings& settings) {
+  RTC_DCHECK(1 <= settings.beginning_packets &&
+             settings.beginning_packets < packets.size());
+  RTC_DCHECK(1 <= settings.end_packets &&
+             settings.end_packets < packets.size());
+  RTC_DCHECK(settings.beginning_packets + settings.end_packets <=
+             packets.size());
+  TrendlineEstimator::PacketTiming early = packets[0];
+  for (size_t i = 1; i < settings.beginning_packets; ++i) {
+    if (packets[i].raw_delay_ms < early.raw_delay_ms)
+      early = packets[i];
+  }
+  size_t late_start = packets.size() - settings.end_packets;
+  TrendlineEstimator::PacketTiming late = packets[late_start];
+  for (size_t i = late_start + 1; i < packets.size(); ++i) {
+    if (packets[i].raw_delay_ms < late.raw_delay_ms)
+      late = packets[i];
+  }
+  if (late.arrival_time_ms - early.arrival_time_ms < 1) {
+    return absl::nullopt;
+  }
+  return (late.raw_delay_ms - early.raw_delay_ms) /
+             (late.arrival_time_ms - early.arrival_time_ms) +
+         settings.cap_uncertainty;
+}
+
+constexpr double kMaxAdaptOffsetMs = 15.0;
+constexpr double kOverUsingTimeThreshold = 10;
+constexpr int kMinNumDeltas = 60;
+constexpr int kDeltaCounterMax = 1000;
+
+}  // namespace
+
+constexpr char TrendlineEstimatorSettings::kKey[];
+
+TrendlineEstimatorSettings::TrendlineEstimatorSettings(
+    const FieldTrialsView* key_value_config) {
+  if (absl::StartsWith(
+          key_value_config->Lookup(kBweWindowSizeInPacketsExperiment),
+          "Enabled")) {
+    window_size = ReadTrendlineFilterWindowSize(key_value_config);
+  }
+  Parser()->Parse(key_value_config->Lookup(TrendlineEstimatorSettings::kKey));
+  if (window_size < 10 || 200 < window_size) {
+    RTC_LOG(LS_WARNING) << "Window size must be between 10 and 200 packets";
+    window_size = kDefaultTrendlineWindowSize;
+  }
+  if (enable_cap) {
+    if (beginning_packets < 1 || end_packets < 1 ||
+        beginning_packets > window_size || end_packets > window_size) {
+      RTC_LOG(LS_WARNING) << "Size of beginning and end must be between 1 and "
+                          << window_size;
+      enable_cap = false;
+      beginning_packets = end_packets = 0;
+      cap_uncertainty = 0.0;
+    }
+    if (beginning_packets + end_packets > window_size) {
+      RTC_LOG(LS_WARNING)
+          << "Size of beginning plus end can't exceed the window size";
+      enable_cap = false;
+      beginning_packets = end_packets = 0;
+      cap_uncertainty = 0.0;
+    }
+    if (cap_uncertainty < 0.0 || 0.025 < cap_uncertainty) {
+      RTC_LOG(LS_WARNING) << "Cap uncertainty must be between 0 and 0.025";
+      cap_uncertainty = 0.0;
+    }
+  }
+}
+
+std::unique_ptr<StructParametersParser> TrendlineEstimatorSettings::Parser() {
+  return StructParametersParser::Create("sort", &enable_sort,  //
+                                        "cap", &enable_cap,    //
+                                        "beginning_packets",
+                                        &beginning_packets,                   //
+                                        "end_packets", &end_packets,          //
+                                        "cap_uncertainty", &cap_uncertainty,  //
+                                        "window_size", &window_size);
+}
+
+TrendlineEstimator::TrendlineEstimator(
+    const FieldTrialsView* key_value_config,
+    NetworkStatePredictor* network_state_predictor)
+    : settings_(key_value_config),
+      smoothing_coef_(kDefaultTrendlineSmoothingCoeff),
+      threshold_gain_(kDefaultTrendlineThresholdGain),
+      num_of_deltas_(0),
+      first_arrival_time_ms_(-1),
+      accumulated_delay_(0),
+      smoothed_delay_(0),
+      delay_hist_(),
+      k_up_(0.0087),
+      k_down_(0.039),
+      overusing_time_threshold_(kOverUsingTimeThreshold),
+      threshold_(12.5),
+      prev_modified_trend_(NAN),
+      last_update_ms_(-1),
+      prev_trend_(0.0),
+      time_over_using_(-1),
+      overuse_counter_(0),
+      hypothesis_(BandwidthUsage::kBwNormal),
+      hypothesis_predicted_(BandwidthUsage::kBwNormal),
+      network_state_predictor_(network_state_predictor) {
+  RTC_LOG(LS_INFO)
+      << "Using Trendline filter for delay change estimation with settings "
+      << settings_.Parser()->Encode() << " and "
+      << (network_state_predictor_ ? "injected" : "no")
+      << " network state predictor";
+}
+
+TrendlineEstimator::~TrendlineEstimator() {}
+
+void TrendlineEstimator::UpdateTrendline(double recv_delta_ms,
+                                         double send_delta_ms,
+                                         int64_t send_time_ms,
+                                         int64_t arrival_time_ms,
+                                         size_t packet_size) {
+  const double delta_ms = recv_delta_ms - send_delta_ms;
+  ++num_of_deltas_;
+  num_of_deltas_ = std::min(num_of_deltas_, kDeltaCounterMax);
+  if (first_arrival_time_ms_ == -1)
+    first_arrival_time_ms_ = arrival_time_ms;
+
+  // Exponential backoff filter.
+  accumulated_delay_ += delta_ms;
+  BWE_TEST_LOGGING_PLOT(1, "accumulated_delay_ms", arrival_time_ms,
+                        accumulated_delay_);
+  smoothed_delay_ = smoothing_coef_ * smoothed_delay_ +
+                    (1 - smoothing_coef_) * accumulated_delay_;
+  BWE_TEST_LOGGING_PLOT(1, "smoothed_delay_ms", arrival_time_ms,
+                        smoothed_delay_);
+
+  // Maintain packet window
+  delay_hist_.emplace_back(
+      static_cast<double>(arrival_time_ms - first_arrival_time_ms_),
+      smoothed_delay_, accumulated_delay_);
+  if (settings_.enable_sort) {
+    for (size_t i = delay_hist_.size() - 1;
+         i > 0 &&
+         delay_hist_[i].arrival_time_ms < delay_hist_[i - 1].arrival_time_ms;
+         --i) {
+      std::swap(delay_hist_[i], delay_hist_[i - 1]);
+    }
+  }
+  if (delay_hist_.size() > settings_.window_size)
+    delay_hist_.pop_front();
+
+  // Simple linear regression.
+  double trend = prev_trend_;
+  if (delay_hist_.size() == settings_.window_size) {
+    // Update trend_ if it is possible to fit a line to the data. The delay
+    // trend can be seen as an estimate of (send_rate - capacity)/capacity.
+    // 0 < trend < 1   ->  the delay increases, queues are filling up
+    //   trend == 0    ->  the delay does not change
+    //   trend < 0     ->  the delay decreases, queues are being emptied
+    trend = LinearFitSlope(delay_hist_).value_or(trend);
+    if (settings_.enable_cap) {
+      absl::optional<double> cap = ComputeSlopeCap(delay_hist_, settings_);
+      // We only use the cap to filter out overuse detections, not
+      // to detect additional underuses.
+      if (trend >= 0 && cap.has_value() && trend > cap.value()) {
+        trend = cap.value();
+      }
+    }
+  }
+  BWE_TEST_LOGGING_PLOT(1, "trendline_slope", arrival_time_ms, trend);
+
+  Detect(trend, send_delta_ms, arrival_time_ms);
+}
+
+void TrendlineEstimator::Update(double recv_delta_ms,
+                                double send_delta_ms,
+                                int64_t send_time_ms,
+                                int64_t arrival_time_ms,
+                                size_t packet_size,
+                                bool calculated_deltas) {
+  if (calculated_deltas) {
+    UpdateTrendline(recv_delta_ms, send_delta_ms, send_time_ms, arrival_time_ms,
+                    packet_size);
+  }
+  if (network_state_predictor_) {
+    hypothesis_predicted_ = network_state_predictor_->Update(
+        send_time_ms, arrival_time_ms, hypothesis_);
+  }
+}
+
+BandwidthUsage TrendlineEstimator::State() const {
+  return network_state_predictor_ ? hypothesis_predicted_ : hypothesis_;
+}
+
+void TrendlineEstimator::Detect(double trend, double ts_delta, int64_t now_ms) {
+  if (num_of_deltas_ < 2) {
+    hypothesis_ = BandwidthUsage::kBwNormal;
+    return;
+  }
+  const double modified_trend =
+      std::min(num_of_deltas_, kMinNumDeltas) * trend * threshold_gain_;
+  prev_modified_trend_ = modified_trend;
+  BWE_TEST_LOGGING_PLOT(1, "T", now_ms, modified_trend);
+  BWE_TEST_LOGGING_PLOT(1, "threshold", now_ms, threshold_);
+  if (modified_trend > threshold_) {
+    if (time_over_using_ == -1) {
+      // Initialize the timer. Assume that we've been
+      // over-using half of the time since the previous
+      // sample.
+      time_over_using_ = ts_delta / 2;
+    } else {
+      // Increment timer
+      time_over_using_ += ts_delta;
+    }
+    overuse_counter_++;
+    if (time_over_using_ > overusing_time_threshold_ && overuse_counter_ > 1) {
+      if (trend >= prev_trend_) {
+        time_over_using_ = 0;
+        overuse_counter_ = 0;
+        hypothesis_ = BandwidthUsage::kBwOverusing;
+      }
+    }
+  } else if (modified_trend < -threshold_) {
+    time_over_using_ = -1;
+    overuse_counter_ = 0;
+    hypothesis_ = BandwidthUsage::kBwUnderusing;
+  } else {
+    time_over_using_ = -1;
+    overuse_counter_ = 0;
+    hypothesis_ = BandwidthUsage::kBwNormal;
+  }
+  prev_trend_ = trend;
+  UpdateThreshold(modified_trend, now_ms);
+}
+
+void TrendlineEstimator::UpdateThreshold(double modified_trend,
+                                         int64_t now_ms) {
+  if (last_update_ms_ == -1)
+    last_update_ms_ = now_ms;
+
+  if (fabs(modified_trend) > threshold_ + kMaxAdaptOffsetMs) {
+    // Avoid adapting the threshold to big latency spikes, caused e.g.,
+    // by a sudden capacity drop.
+    last_update_ms_ = now_ms;
+    return;
+  }
+
+  const double k = fabs(modified_trend) < threshold_ ? k_down_ : k_up_;
+  const int64_t kMaxTimeDeltaMs = 100;
+  int64_t time_delta_ms = std::min(now_ms - last_update_ms_, kMaxTimeDeltaMs);
+  threshold_ += k * (fabs(modified_trend) - threshold_) * time_delta_ms;
+  threshold_ = rtc::SafeClamp(threshold_, 6.f, 600.f);
+  last_update_ms_ = now_ms;
+}
+
+}  // namespace webrtc

TMessagesProj/jni/voip/webrtc/modules/congestion_controller/goog_cc/trendline_estimator.h

@@ -0,0 +1,124 @@
+/*
+ *  Copyright (c) 2016 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_CONGESTION_CONTROLLER_GOOG_CC_TRENDLINE_ESTIMATOR_H_
+#define MODULES_CONGESTION_CONTROLLER_GOOG_CC_TRENDLINE_ESTIMATOR_H_
+
+#include <stddef.h>
+#include <stdint.h>
+
+#include <deque>
+#include <memory>
+
+#include "api/field_trials_view.h"
+#include "api/network_state_predictor.h"
+#include "modules/congestion_controller/goog_cc/delay_increase_detector_interface.h"
+#include "rtc_base/experiments/struct_parameters_parser.h"
+
+namespace webrtc {
+
+struct TrendlineEstimatorSettings {
+  static constexpr char kKey[] = "WebRTC-Bwe-TrendlineEstimatorSettings";
+  static constexpr unsigned kDefaultTrendlineWindowSize = 20;
+
+  TrendlineEstimatorSettings() = delete;
+  explicit TrendlineEstimatorSettings(const FieldTrialsView* key_value_config);
+
+  // Sort the packets in the window. Should be redundant,
+  // but then almost no cost.
+  bool enable_sort = false;
+
+  // Cap the trendline slope based on the minimum delay seen
+  // in the beginning_packets and end_packets respectively.
+  bool enable_cap = false;
+  unsigned beginning_packets = 7;
+  unsigned end_packets = 7;
+  double cap_uncertainty = 0.0;
+
+  // Size (in packets) of the window.
+  unsigned window_size = kDefaultTrendlineWindowSize;
+
+  std::unique_ptr<StructParametersParser> Parser();
+};
+
+class TrendlineEstimator : public DelayIncreaseDetectorInterface {
+ public:
+  TrendlineEstimator(const FieldTrialsView* key_value_config,
+                     NetworkStatePredictor* network_state_predictor);
+
+  ~TrendlineEstimator() override;
+
+  TrendlineEstimator(const TrendlineEstimator&) = delete;
+  TrendlineEstimator& operator=(const TrendlineEstimator&) = delete;
+
+  // Update the estimator with a new sample. The deltas should represent deltas
+  // between timestamp groups as defined by the InterArrival class.
+  void Update(double recv_delta_ms,
+              double send_delta_ms,
+              int64_t send_time_ms,
+              int64_t arrival_time_ms,
+              size_t packet_size,
+              bool calculated_deltas) override;
+
+  void UpdateTrendline(double recv_delta_ms,
+                       double send_delta_ms,
+                       int64_t send_time_ms,
+                       int64_t arrival_time_ms,
+                       size_t packet_size);
+
+  BandwidthUsage State() const override;
+
+  struct PacketTiming {
+    PacketTiming(double arrival_time_ms,
+                 double smoothed_delay_ms,
+                 double raw_delay_ms)
+        : arrival_time_ms(arrival_time_ms),
+          smoothed_delay_ms(smoothed_delay_ms),
+          raw_delay_ms(raw_delay_ms) {}
+    double arrival_time_ms;
+    double smoothed_delay_ms;
+    double raw_delay_ms;
+  };
+
+ private:
+  friend class GoogCcStatePrinter;
+  void Detect(double trend, double ts_delta, int64_t now_ms);
+
+  void UpdateThreshold(double modified_offset, int64_t now_ms);
+
+  // Parameters.
+  TrendlineEstimatorSettings settings_;
+  const double smoothing_coef_;
+  const double threshold_gain_;
+  // Used by the existing threshold.
+  int num_of_deltas_;
+  // Keep the arrival times small by using the change from the first packet.
+  int64_t first_arrival_time_ms_;
+  // Exponential backoff filtering.
+  double accumulated_delay_;
+  double smoothed_delay_;
+  // Linear least squares regression.
+  std::deque<PacketTiming> delay_hist_;
+
+  const double k_up_;
+  const double k_down_;
+  double overusing_time_threshold_;
+  double threshold_;
+  double prev_modified_trend_;
+  int64_t last_update_ms_;
+  double prev_trend_;
+  double time_over_using_;
+  int overuse_counter_;
+  BandwidthUsage hypothesis_;
+  BandwidthUsage hypothesis_predicted_;
+  NetworkStatePredictor* network_state_predictor_;
+};
+}  // namespace webrtc
+
+#endif  // MODULES_CONGESTION_CONTROLLER_GOOG_CC_TRENDLINE_ESTIMATOR_H_

TMessagesProj/jni/voip/webrtc/modules/congestion_controller/goog_cc/trendline_estimator_unittest.cc

@@ -0,0 +1,153 @@
+/*
+ *  Copyright (c) 2016 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 "modules/congestion_controller/goog_cc/trendline_estimator.h"
+
+#include <algorithm>
+#include <cstddef>
+#include <cstdint>
+#include <vector>
+
+#include "api/network_state_predictor.h"
+#include "api/transport/field_trial_based_config.h"
+#include "rtc_base/checks.h"
+#include "test/gtest.h"
+
+namespace webrtc {
+namespace {
+
+class PacketTimeGenerator {
+ public:
+  PacketTimeGenerator(int64_t initial_clock, double time_between_packets)
+      : initial_clock_(initial_clock),
+        time_between_packets_(time_between_packets),
+        packets_(0) {}
+  int64_t operator()() {
+    return initial_clock_ + time_between_packets_ * packets_++;
+  }
+
+ private:
+  const int64_t initial_clock_;
+  const double time_between_packets_;
+  size_t packets_;
+};
+
+class TrendlineEstimatorTest : public testing::Test {
+ public:
+  TrendlineEstimatorTest()
+      : send_times(kPacketCount),
+        recv_times(kPacketCount),
+        packet_sizes(kPacketCount),
+        config(),
+        estimator(&config, nullptr),
+        count(1) {
+    std::fill(packet_sizes.begin(), packet_sizes.end(), kPacketSizeBytes);
+  }
+
+  void RunTestUntilStateChange() {
+    RTC_DCHECK_EQ(send_times.size(), kPacketCount);
+    RTC_DCHECK_EQ(recv_times.size(), kPacketCount);
+    RTC_DCHECK_EQ(packet_sizes.size(), kPacketCount);
+    RTC_DCHECK_GE(count, 1);
+    RTC_DCHECK_LT(count, kPacketCount);
+
+    auto initial_state = estimator.State();
+    for (; count < kPacketCount; count++) {
+      double recv_delta = recv_times[count] - recv_times[count - 1];
+      double send_delta = send_times[count] - send_times[count - 1];
+      estimator.Update(recv_delta, send_delta, send_times[count],
+                       recv_times[count], packet_sizes[count], true);
+      if (estimator.State() != initial_state) {
+        return;
+      }
+    }
+  }
+
+ protected:
+  const size_t kPacketCount = 25;
+  const size_t kPacketSizeBytes = 1200;
+  std::vector<int64_t> send_times;
+  std::vector<int64_t> recv_times;
+  std::vector<size_t> packet_sizes;
+  const FieldTrialBasedConfig config;
+  TrendlineEstimator estimator;
+  size_t count;
+};
+}  // namespace
+
+TEST_F(TrendlineEstimatorTest, Normal) {
+  PacketTimeGenerator send_time_generator(123456789 /*initial clock*/,
+                                          20 /*20 ms between sent packets*/);
+  std::generate(send_times.begin(), send_times.end(), send_time_generator);
+
+  PacketTimeGenerator recv_time_generator(987654321 /*initial clock*/,
+                                          20 /*delivered at the same pace*/);
+  std::generate(recv_times.begin(), recv_times.end(), recv_time_generator);
+
+  EXPECT_EQ(estimator.State(), BandwidthUsage::kBwNormal);
+  RunTestUntilStateChange();
+  EXPECT_EQ(estimator.State(), BandwidthUsage::kBwNormal);
+  EXPECT_EQ(count, kPacketCount);  // All packets processed
+}
+
+TEST_F(TrendlineEstimatorTest, Overusing) {
+  PacketTimeGenerator send_time_generator(123456789 /*initial clock*/,
+                                          20 /*20 ms between sent packets*/);
+  std::generate(send_times.begin(), send_times.end(), send_time_generator);
+
+  PacketTimeGenerator recv_time_generator(987654321 /*initial clock*/,
+                                          1.1 * 20 /*10% slower delivery*/);
+  std::generate(recv_times.begin(), recv_times.end(), recv_time_generator);
+
+  EXPECT_EQ(estimator.State(), BandwidthUsage::kBwNormal);
+  RunTestUntilStateChange();
+  EXPECT_EQ(estimator.State(), BandwidthUsage::kBwOverusing);
+  RunTestUntilStateChange();
+  EXPECT_EQ(estimator.State(), BandwidthUsage::kBwOverusing);
+  EXPECT_EQ(count, kPacketCount);  // All packets processed
+}
+
+TEST_F(TrendlineEstimatorTest, Underusing) {
+  PacketTimeGenerator send_time_generator(123456789 /*initial clock*/,
+                                          20 /*20 ms between sent packets*/);
+  std::generate(send_times.begin(), send_times.end(), send_time_generator);
+
+  PacketTimeGenerator recv_time_generator(987654321 /*initial clock*/,
+                                          0.85 * 20 /*15% faster delivery*/);
+  std::generate(recv_times.begin(), recv_times.end(), recv_time_generator);
+
+  EXPECT_EQ(estimator.State(), BandwidthUsage::kBwNormal);
+  RunTestUntilStateChange();
+  EXPECT_EQ(estimator.State(), BandwidthUsage::kBwUnderusing);
+  RunTestUntilStateChange();
+  EXPECT_EQ(estimator.State(), BandwidthUsage::kBwUnderusing);
+  EXPECT_EQ(count, kPacketCount);  // All packets processed
+}
+
+TEST_F(TrendlineEstimatorTest, IncludesSmallPacketsByDefault) {
+  PacketTimeGenerator send_time_generator(123456789 /*initial clock*/,
+                                          20 /*20 ms between sent packets*/);
+  std::generate(send_times.begin(), send_times.end(), send_time_generator);
+
+  PacketTimeGenerator recv_time_generator(987654321 /*initial clock*/,
+                                          1.1 * 20 /*10% slower delivery*/);
+  std::generate(recv_times.begin(), recv_times.end(), recv_time_generator);
+
+  std::fill(packet_sizes.begin(), packet_sizes.end(), 100);
+
+  EXPECT_EQ(estimator.State(), BandwidthUsage::kBwNormal);
+  RunTestUntilStateChange();
+  EXPECT_EQ(estimator.State(), BandwidthUsage::kBwOverusing);
+  RunTestUntilStateChange();
+  EXPECT_EQ(estimator.State(), BandwidthUsage::kBwOverusing);
+  EXPECT_EQ(count, kPacketCount);  // All packets processed
+}
+
+}  // namespace webrtc