Stop increasing loss-based BWE if no feedback is received.

webrtc/modules/bitrate_controller/send_side_bandwidth_estimation.cc

 /*
  *  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.
  */
 
@@ skipping 11 matching lines @@
 namespace webrtc {
 namespace {
 const int64_t kBweIncreaseIntervalMs = 1000;
 const int64_t kBweDecreaseIntervalMs = 300;
 const int64_t kStartPhaseMs = 2000;
 const int64_t kBweConverganceTimeMs = 20000;
 const int kLimitNumPackets = 20;
 const int kDefaultMinBitrateBps = 10000;
 const int kDefaultMaxBitrateBps = 1000000000;
 const int64_t kLowBitrateLogPeriodMs = 10000;
+// Expecting that RTCP feedback is sent uniformly within [0.5, 1.5]s intervals.
+const int64_t kFeedbackIntervalMs = 1500;
+const int64_t kFeedbackTimeoutIntervals = 3;
+const int64_t kTimeoutIntervalMs = 1000;
 
 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]);
 
 }  // namespace
 
 SendSideBandwidthEstimation::SendSideBandwidthEstimation(RtcEventLog* event_log)
     : lost_packets_since_last_loss_update_Q8_(0),
       expected_packets_since_last_loss_update_(0),
       bitrate_(0),
       min_bitrate_configured_(kDefaultMinBitrateBps),
       max_bitrate_configured_(kDefaultMaxBitrateBps),
       last_low_bitrate_log_ms_(-1),
       has_decreased_since_last_fraction_loss_(false),
-      time_last_receiver_block_ms_(-1),
+      last_feedback_ms_(-1),
+      last_packet_report_ms_(-1),
+      last_timeout_ms_(-1),
       last_fraction_loss_(0),
       last_round_trip_time_ms_(0),
       bwe_incoming_(0),
       delay_based_bitrate_bps_(0),
       time_last_decrease_ms_(0),
       first_report_time_ms_(-1),
       initially_lost_packets_(0),
       bitrate_at_2_seconds_kbps_(0),
       uma_update_state_(kNoUpdate),
       rampup_uma_stats_updated_(kNumUmaRampupMetrics, false),
-      event_log_(event_log) {
+      event_log_(event_log),
+      in_timeout_experiment_(webrtc::field_trial::FindFullName(
+                                 "WebRTC-SendSideBwe") == "Enabled") {
   RTC_DCHECK(event_log);
 }
 
 SendSideBandwidthEstimation::~SendSideBandwidthEstimation() {}
 
 void SendSideBandwidthEstimation::SetBitrates(int send_bitrate,
                                               int min_bitrate,
                                               int max_bitrate) {
   if (send_bitrate > 0)
     SetSendBitrate(send_bitrate);
@@ skipping 43 matching lines @@
     int64_t now_ms,
     uint32_t bitrate_bps) {
   delay_based_bitrate_bps_ = bitrate_bps;
   bitrate_ = CapBitrateToThresholds(now_ms, bitrate_);
 }
 
 void SendSideBandwidthEstimation::UpdateReceiverBlock(uint8_t fraction_loss,
                                                       int64_t rtt,
                                                       int number_of_packets,
                                                       int64_t now_ms) {
+  last_feedback_ms_ = now_ms;
   if (first_report_time_ms_ == -1)
     first_report_time_ms_ = now_ms;
 
   // Update RTT.
   last_round_trip_time_ms_ = rtt;
 
   // Check sequence number diff and weight loss report
   if (number_of_packets > 0) {
     // Calculate number of lost packets.
     const int num_lost_packets_Q8 = fraction_loss * number_of_packets;
     // Accumulate reports.
     lost_packets_since_last_loss_update_Q8_ += num_lost_packets_Q8;
     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_packets_since_last_loss_update_ < kLimitNumPackets)
       return;
 
     has_decreased_since_last_fraction_loss_ = false;
     last_fraction_loss_ = lost_packets_since_last_loss_update_Q8_ /
                           expected_packets_since_last_loss_update_;
 
     // Reset accumulators.
     lost_packets_since_last_loss_update_Q8_ = 0;
     expected_packets_since_last_loss_update_ = 0;
+    last_packet_report_ms_ = now_ms;
+    UpdateEstimate(now_ms);

[terelius, 2016/08/23 17:33:29]

Nice. I've been meaning to try moving the UpdateEstimate into the "if
(number_of_packets > 0)" block.

   }
-  time_last_receiver_block_ms_ = now_ms;
-  UpdateEstimate(now_ms);
   UpdateUmaStats(now_ms, rtt, (fraction_loss * number_of_packets) >> 8);
 }
 
 void SendSideBandwidthEstimation::UpdateUmaStats(int64_t now_ms,
                                                  int64_t rtt,
                                                  int lost_packets) {
   int bitrate_kbps = static_cast<int>((bitrate_ + 500) / 1000);
   for (size_t i = 0; i < kNumUmaRampupMetrics; ++i) {
     if (!rampup_uma_stats_updated_[i] &&
         bitrate_kbps >= kUmaRampupMetrics[i].bitrate_kbps) {
@@ skipping 23 matching lines @@
   }
 }
 
 void SendSideBandwidthEstimation::UpdateEstimate(int64_t now_ms) {
   // 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(now_ms)) {
     uint32_t prev_bitrate = bitrate_;
     if (bwe_incoming_ > bitrate_)
       bitrate_ = CapBitrateToThresholds(now_ms, bwe_incoming_);
-    if (delay_based_bitrate_bps_ > bitrate_)
+    if (delay_based_bitrate_bps_ > bitrate_) {
       bitrate_ = CapBitrateToThresholds(now_ms, delay_based_bitrate_bps_);
+    }
     if (bitrate_ != prev_bitrate) {
       min_bitrate_history_.clear();
       min_bitrate_history_.push_back(std::make_pair(now_ms, bitrate_));
       return;
     }
   }
   UpdateMinHistory(now_ms);
-  // Only start updating bitrate when receiving receiver blocks.
-  // TODO(pbos): Handle the case when no receiver report is received for a very
-  // long time.
-  if (time_last_receiver_block_ms_ != -1) {
+  if (last_packet_report_ms_ == -1) {
+    // No feedback received.
+    bitrate_ = CapBitrateToThresholds(now_ms, bitrate_);
+    return;
+  }
+  int64_t time_since_packet_report_ms = now_ms - last_packet_report_ms_;
+  int64_t time_since_feedback_ms = now_ms - last_feedback_ms_;
+  if (time_since_packet_report_ms < 1.2 * kFeedbackIntervalMs) {
     if (last_fraction_loss_ <= 5) {
       // Loss < 2%: Increase rate by 8% of the min bitrate in the last
       // kBweIncreaseIntervalMs.
       // 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 immediatly to 108kbps
       //   whenever a receiver report is received with lower packet loss.
       //   If instead one would do: bitrate_ *= 1.08^(delta time), it would
-      //   take over one second since the lower packet loss to achieve 108kbps.
+      //   take over one second since the lower packet loss to achieve
+      //   108kbps.
       bitrate_ = static_cast<uint32_t>(
           min_bitrate_history_.front().second * 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).
       bitrate_ += 1000;
 
       event_log_->LogBwePacketLossEvent(
           bitrate_, last_fraction_loss_,
           expected_packets_since_last_loss_update_);
     } else if (last_fraction_loss_ <= 26) {
       // Loss between 2% - 10%: Do nothing.
     } else {
-      // Loss > 10%: Limit the rate decreases to once a kBweDecreaseIntervalMs +
-      // rtt.
+      // Loss > 10%: Limit the rate decreases to once a kBweDecreaseIntervalMs
+      // + rtt.
       if (!has_decreased_since_last_fraction_loss_ &&
           (now_ms - time_last_decrease_ms_) >=
               (kBweDecreaseIntervalMs + last_round_trip_time_ms_)) {
         time_last_decrease_ms_ = now_ms;
 
         // Reduce rate:
         //   newRate = rate * (1 - 0.5*lossRate);
         //   where packetLoss = 256*lossRate;
         bitrate_ = static_cast<uint32_t>(
             (bitrate_ * static_cast<double>(512 - last_fraction_loss_)) /
             512.0);
         has_decreased_since_last_fraction_loss_ = true;
       }
       event_log_->LogBwePacketLossEvent(
           bitrate_, last_fraction_loss_,
           expected_packets_since_last_loss_update_);
     }
+  } else if (time_since_feedback_ms >
+                 kFeedbackTimeoutIntervals * kFeedbackIntervalMs &&
+             (last_timeout_ms_ == -1 ||
+              now_ms - last_timeout_ms_ > kTimeoutIntervalMs)) {
+    if (in_timeout_experiment_) {
+      LOG(LS_WARNING) << "Feedback timed out (" << time_since_feedback_ms
+                      << " ms), reducing bitrate.";
+      bitrate_ *= 0.8;
+      last_timeout_ms_ = now_ms;

[terelius, 2016/08/23 17:33:29]

Should the accumulators (expected_packets_since_last_loss_update_,
lost_packets_since_last_loss_update_Q8_) be reset too, or is it better to keep
whatever data we have even if it is old?

[stefan-webrtc, 2016/09/04 11:17:35]

Perhaps we should reset those too.

+    }
   }
   bitrate_ = CapBitrateToThresholds(now_ms, bitrate_);
 }
 
 bool SendSideBandwidthEstimation::IsInStartPhase(int64_t now_ms) const {
   return first_report_time_ms_ == -1 ||
          now_ms - first_report_time_ms_ < kStartPhaseMs;
 }
 
 void SendSideBandwidthEstimation::UpdateMinHistory(int64_t now_ms) {
@@ skipping 33 matching lines @@
       LOG(LS_WARNING) << "Estimated available bandwidth " << bitrate / 1000
                       << " kbps is below configured min bitrate "
                       << min_bitrate_configured_ / 1000 << " kbps.";
       last_low_bitrate_log_ms_ = now_ms;
     }
     bitrate = min_bitrate_configured_;
   }
   return bitrate;
 }
 }  // namespace webrtc