Remove GetFeedbackInterval in sender side BWE.

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 17 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 int64_t kDefaultRttMs = 200;
 }  // namespace
 
 namespace webrtc {
 
 DelayBasedBwe::DelayBasedBwe(Clock* clock)
     : clock_(clock),
       inter_arrival_(),
       estimator_(),
       detector_(OverUseDetectorOptions()),
       incoming_bitrate_(kBitrateWindowMs, 8000),
       last_update_ms_(-1),
       last_seen_packet_ms_(-1),
-      uma_recorded_(false) {
+      uma_recorded_(false),
+      rtt_ms_(kDefaultRttMs) {
   network_thread_.DetachFromThread();
 }
 
 DelayBasedBwe::Result DelayBasedBwe::IncomingPacketFeedbackVector(
     const std::vector<PacketInfo>& packet_feedback_vector) {
   RTC_DCHECK(network_thread_.CalledOnValidThread());
   if (!uma_recorded_) {
     RTC_HISTOGRAM_ENUMERATION(kBweTypeHistogram,
                               BweNames::kSendSideTransportSeqNum,
                               BweNames::kBweNamesMax);
@@ skipping 64 matching lines @@
   } else if (probing_bps > 0) {
     // No overuse, but probing measured a bitrate.
     remote_rate_.SetEstimate(probing_bps, info.arrival_time_ms);
     result.probe = true;
     result.updated = UpdateEstimate(info.arrival_time_ms, now_ms,
                                     &result.target_bitrate_bps);
   }
   rtc::Optional<uint32_t> incoming_rate =
       incoming_bitrate_.Rate(info.arrival_time_ms);
   if (!result.updated &&
-      (last_update_ms_ == -1 ||
-       now_ms - last_update_ms_ > remote_rate_.GetFeedbackInterval())) {
+      (last_update_ms_ == -1 || now_ms - last_update_ms_ > rtt_ms_)) {

[stefan-webrtc, 2016/10/17 18:59:49]

Do you think there's a true value having this check? I think it's possible to
remove it as it doesn't seem meaningful when the BWE is running on the send-side
and already is limited by the frequency of the feedback reports.

     result.updated = UpdateEstimate(info.arrival_time_ms, now_ms,
                                     &result.target_bitrate_bps);
   }
   if (result.updated)
     last_update_ms_ = now_ms;
 
   return result;
 }
 
 bool DelayBasedBwe::UpdateEstimate(int64_t arrival_time_ms,
                                    int64_t now_ms,
                                    uint32_t* target_bitrate_bps) {
   // The first overuse should immediately trigger a new estimate.
   // We also have to update the estimate immediately if we are overusing
   // and the target bitrate is too high compared to what we are receiving.
   const RateControlInput input(detector_.State(),
                                incoming_bitrate_.Rate(arrival_time_ms),
                                estimator_->var_noise());
   remote_rate_.Update(&input, now_ms);
   *target_bitrate_bps = remote_rate_.UpdateBandwidthEstimate(now_ms);
   return remote_rate_.ValidEstimate();
 }
 
 void DelayBasedBwe::OnRttUpdate(int64_t avg_rtt_ms, int64_t max_rtt_ms) {
+  rtt_ms_ = avg_rtt_ms;
   remote_rate_.SetRtt(avg_rtt_ms);
 }
 
 bool DelayBasedBwe::LatestEstimate(std::vector<uint32_t>* ssrcs,
                                    uint32_t* bitrate_bps) 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_bps);
   if (!remote_rate_.ValidEstimate())
     return false;
 
   *ssrcs = {kFixedSsrc};
   *bitrate_bps = remote_rate_.LatestEstimate();
   return true;
 }
 
 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.
   remote_rate_.SetMinBitrate(min_bitrate_bps);
 }
 }  // namespace webrtc