Revert of Refactored incoming bitrate estimator.

webrtc/modules/congestion_controller/delay_based_bwe.cc

 /*
  *  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.
  */
 
@@ skipping 20 matching lines @@
 constexpr int kTimestampGroupLengthMs = 5;
 constexpr int kAbsSendTimeFraction = 18;
 constexpr int kAbsSendTimeInterArrivalUpshift = 8;
 constexpr int kInterArrivalShift =
     kAbsSendTimeFraction + kAbsSendTimeInterArrivalUpshift;
 constexpr double kTimestampToMs =
     1000.0 / static_cast<double>(1 << kInterArrivalShift);
 // This ssrc is used to fulfill the current API but will be removed
 // after the API has been changed.
 constexpr uint32_t kFixedSsrc = 0;
+constexpr int kInitialRateWindowMs = 500;
+constexpr int kRateWindowMs = 150;
 
 // Parameters for linear least squares fit of regression line to noisy data.
 constexpr size_t kDefaultTrendlineWindowSize = 20;
 constexpr double kDefaultTrendlineSmoothingCoeff = 0.9;
 constexpr double kDefaultTrendlineThresholdGain = 4.0;
 
 constexpr int kMaxConsecutiveFailedLookups = 5;
 
 const char kBweSparseUpdateExperiment[] = "WebRTC-BweSparseUpdateExperiment";
 
 bool BweSparseUpdateExperimentIsEnabled() {
   std::string experiment_string =
       webrtc::field_trial::FindFullName(kBweSparseUpdateExperiment);
   return experiment_string == "Enabled";
 }
+
+class PacketFeedbackComparator {
+ public:
+  inline bool operator()(const webrtc::PacketFeedback& lhs,
+                         const webrtc::PacketFeedback& rhs) {
+    if (lhs.arrival_time_ms != rhs.arrival_time_ms)
+      return lhs.arrival_time_ms < rhs.arrival_time_ms;
+    if (lhs.send_time_ms != rhs.send_time_ms)
+      return lhs.send_time_ms < rhs.send_time_ms;
+    return lhs.sequence_number < rhs.sequence_number;
+  }
+};
+
+void SortPacketFeedbackVector(const std::vector<webrtc::PacketFeedback>& input,
+                              std::vector<webrtc::PacketFeedback>* output) {
+  auto pred = [](const webrtc::PacketFeedback& packet_feedback) {
+    return packet_feedback.arrival_time_ms !=
+           webrtc::PacketFeedback::kNotReceived;
+  };
+  std::copy_if(input.begin(), input.end(), std::back_inserter(*output), pred);
+  std::sort(output->begin(), output->end(), PacketFeedbackComparator());
+}
 }  // namespace
 
 namespace webrtc {
 
+DelayBasedBwe::BitrateEstimator::BitrateEstimator()
+    : sum_(0),
+      current_win_ms_(0),
+      prev_time_ms_(-1),
+      bitrate_estimate_(-1.0f),
+      bitrate_estimate_var_(50.0f) {}
+
+void DelayBasedBwe::BitrateEstimator::Update(int64_t now_ms, int bytes) {
+  int rate_window_ms = kRateWindowMs;
+  // 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_ < 0.f)
+    rate_window_ms = kInitialRateWindowMs;
+  float bitrate_sample = UpdateWindow(now_ms, bytes, rate_window_ms);
+  if (bitrate_sample < 0.0f)
+    return;
+  if (bitrate_estimate_ < 0.0f) {
+    // This is the very first sample we get. Use it to initialize the estimate.
+    bitrate_estimate_ = bitrate_sample;
+    return;
+  }
+  // Define the sample uncertainty as a function of how far away it is from the
+  // current estimate.
+  float sample_uncertainty =
+      10.0f * std::abs(bitrate_estimate_ - bitrate_sample) / bitrate_estimate_;
+  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_ = (sample_var * bitrate_estimate_ +
+                       pred_bitrate_estimate_var * bitrate_sample) /
+                      (sample_var + pred_bitrate_estimate_var);
+  bitrate_estimate_var_ = sample_var * pred_bitrate_estimate_var /
+                          (sample_var + pred_bitrate_estimate_var);
+}
+
+float DelayBasedBwe::BitrateEstimator::UpdateWindow(int64_t now_ms,
+                                                    int bytes,
+                                                    int rate_window_ms) {
+  // Reset if time moves backwards.
+  if (now_ms < prev_time_ms_) {
+    prev_time_ms_ = -1;
+    sum_ = 0;
+    current_win_ms_ = 0;
+  }
+  if (prev_time_ms_ >= 0) {
+    current_win_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_win_ms_ %= rate_window_ms;
+    }
+  }
+  prev_time_ms_ = now_ms;
+  float bitrate_sample = -1.0f;
+  if (current_win_ms_ >= rate_window_ms) {
+    bitrate_sample = 8.0f * sum_ / static_cast<float>(rate_window_ms);
+    current_win_ms_ -= rate_window_ms;
+    sum_ = 0;
+  }
+  sum_ += bytes;
+  return bitrate_sample;
+}
+
+rtc::Optional<uint32_t> DelayBasedBwe::BitrateEstimator::bitrate_bps() const {
+  if (bitrate_estimate_ < 0.f)
+    return rtc::Optional<uint32_t>();
+  return rtc::Optional<uint32_t>(bitrate_estimate_ * 1000);
+}
+
 DelayBasedBwe::DelayBasedBwe(RtcEventLog* event_log, const Clock* clock)
     : event_log_(event_log),
       clock_(clock),
       inter_arrival_(),
       trendline_estimator_(),
       detector_(),
+      receiver_incoming_bitrate_(),
       last_seen_packet_ms_(-1),
       uma_recorded_(false),
       probe_bitrate_estimator_(event_log),
       trendline_window_size_(kDefaultTrendlineWindowSize),
       trendline_smoothing_coeff_(kDefaultTrendlineSmoothingCoeff),
       trendline_threshold_gain_(kDefaultTrendlineThresholdGain),
       consecutive_delayed_feedbacks_(0),
       last_logged_bitrate_(0),
       last_logged_state_(BandwidthUsage::kBwNormal),
       in_sparse_update_experiment_(BweSparseUpdateExperimentIsEnabled()) {
   LOG(LS_INFO) << "Using Trendline filter for delay change estimation.";
   network_thread_.DetachFromThread();
 }
 
 DelayBasedBwe::~DelayBasedBwe() {}
 
 DelayBasedBwe::Result DelayBasedBwe::IncomingPacketFeedbackVector(
-    const std::vector<PacketFeedback>& packet_feedback_vector,
-    rtc::Optional<uint32_t> acked_bitrate_bps) {
-  RTC_DCHECK(std::is_sorted(packet_feedback_vector.begin(),
-                            packet_feedback_vector.end(),
-                            PacketFeedbackComparator()));
+    const std::vector<PacketFeedback>& packet_feedback_vector) {
   RTC_DCHECK(network_thread_.CalledOnValidThread());
 
+  std::vector<PacketFeedback> sorted_packet_feedback_vector;
+  SortPacketFeedbackVector(packet_feedback_vector,
+                           &sorted_packet_feedback_vector);
   // TOOD(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()) {
+  if (sorted_packet_feedback_vector.empty()) {
     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 overusing = false;
   bool delayed_feedback = true;
-  for (const auto& packet_feedback : packet_feedback_vector) {
+  for (const auto& packet_feedback : sorted_packet_feedback_vector) {
     if (packet_feedback.send_time_ms < 0)
       continue;
     delayed_feedback = false;
     IncomingPacketFeedback(packet_feedback);
     if (!in_sparse_update_experiment_)
       overusing |= (detector_.State() == BandwidthUsage::kBwOverusing);
   }
   if (in_sparse_update_experiment_)
     overusing = (detector_.State() == BandwidthUsage::kBwOverusing);
   if (delayed_feedback) {
     ++consecutive_delayed_feedbacks_;
     if (consecutive_delayed_feedbacks_ >= kMaxConsecutiveFailedLookups) {
       consecutive_delayed_feedbacks_ = 0;
-      return OnLongFeedbackDelay(packet_feedback_vector.back().arrival_time_ms);
+      return OnLongFeedbackDelay(
+          sorted_packet_feedback_vector.back().arrival_time_ms);
     }
   } else {
     consecutive_delayed_feedbacks_ = 0;
-    return MaybeUpdateEstimate(overusing, acked_bitrate_bps);
+    return MaybeUpdateEstimate(overusing);
   }
   return Result();
 }
 
 DelayBasedBwe::Result DelayBasedBwe::OnLongFeedbackDelay(
     int64_t arrival_time_ms) {
   // Estimate should always be valid since a start bitrate always is set in the
   // Call constructor. An alternative would be to return an empty Result here,
   // or to estimate the throughput based on the feedback we received.
   RTC_DCHECK(rate_control_.ValidEstimate());
   rate_control_.SetEstimate(rate_control_.LatestEstimate() / 2,
                             arrival_time_ms);
   Result result;
   result.updated = true;
   result.probe = false;
   result.target_bitrate_bps = rate_control_.LatestEstimate();
   LOG(LS_WARNING) << "Long feedback delay detected, reducing BWE to "
                   << result.target_bitrate_bps;
   return result;
 }
 
 void DelayBasedBwe::IncomingPacketFeedback(
     const PacketFeedback& packet_feedback) {
   int64_t now_ms = clock_->TimeInMilliseconds();
+
+  receiver_incoming_bitrate_.Update(packet_feedback.arrival_time_ms,
+                                    packet_feedback.payload_size);
+  Result result;
   // Reset if the stream has timed out.
   if (last_seen_packet_ms_ == -1 ||
       now_ms - last_seen_packet_ms_ > kStreamTimeOutMs) {
     inter_arrival_.reset(
         new InterArrival((kTimestampGroupLengthMs << kInterArrivalShift) / 1000,
                          kTimestampToMs, true));
     trendline_estimator_.reset(new TrendlineEstimator(
         trendline_window_size_, trendline_smoothing_coeff_,
         trendline_threshold_gain_));
   }
@@ skipping 22 matching lines @@
     detector_.Detect(trendline_estimator_->trendline_slope(), ts_delta_ms,
                      trendline_estimator_->num_of_deltas(),
                      packet_feedback.arrival_time_ms);
   }
   if (packet_feedback.pacing_info.probe_cluster_id !=
       PacedPacketInfo::kNotAProbe) {
     probe_bitrate_estimator_.HandleProbeAndEstimateBitrate(packet_feedback);
   }
 }
 
-DelayBasedBwe::Result DelayBasedBwe::MaybeUpdateEstimate(
-    bool overusing,
-    rtc::Optional<uint32_t> acked_bitrate_bps) {
+DelayBasedBwe::Result DelayBasedBwe::MaybeUpdateEstimate(bool overusing) {
   Result result;
   int64_t now_ms = clock_->TimeInMilliseconds();
 
+  rtc::Optional<uint32_t> acked_bitrate_bps =
+      receiver_incoming_bitrate_.bitrate_bps();
   rtc::Optional<int> probe_bitrate_bps =
       probe_bitrate_estimator_.FetchAndResetLastEstimatedBitrateBps();
   // Currently overusing the bandwidth.
   if (overusing) {
     if (acked_bitrate_bps &&
         rate_control_.TimeToReduceFurther(now_ms, *acked_bitrate_bps)) {
       result.updated = UpdateEstimate(now_ms, acked_bitrate_bps, overusing,
                                       &result.target_bitrate_bps);
     }
   } else {
@@ skipping 59 matching lines @@
 void DelayBasedBwe::SetMinBitrate(int min_bitrate_bps) {
   // 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_bps);
 }
 
 int64_t DelayBasedBwe::GetExpectedBwePeriodMs() const {
   return rate_control_.GetExpectedBandwidthPeriodMs();
 }
 }  // namespace webrtc