Tune loss-based BWE to be more compatible with the low frequency loss reports of audio streams.

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 15 matching lines @@
 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 kDefaultMaxBitrateBps = 1000000000;
 const int64_t kLowBitrateLogPeriodMs = 10000;
 const int64_t kRtcEventLogPeriodMs = 5000;
 // Expecting that RTCP feedback is sent uniformly within [0.5, 1.5]s intervals.
-const int64_t kFeedbackIntervalMs = 1500;
+const int64_t kFeedbackIntervalMs = 5000;
 const int64_t kFeedbackTimeoutIntervals = 3;
 const int64_t kTimeoutIntervalMs = 1000;
 
 const float kDefaultLowLossThreshold = 0.02f;
 const float kDefaultHighLossThreshold = 0.1f;
 const int kDefaultBitrateThresholdKbps = 0;
 
 struct UmaRampUpMetric {
   const char* metric_name;
   int bitrate_kbps;
@@ skipping 364 matching lines @@
       last_fraction_loss_ != last_logged_fraction_loss_ ||
       now_ms - last_rtc_event_log_ms_ > kRtcEventLogPeriodMs) {
     event_log_->LogLossBasedBweUpdate(bitrate_bps, last_fraction_loss_,
                                       expected_packets_since_last_loss_update_);
     last_logged_fraction_loss_ = last_fraction_loss_;
     last_rtc_event_log_ms_ = now_ms;
   }
   current_bitrate_bps_ = bitrate_bps;
 }
 }  // namespace webrtc