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

Index: webrtc/modules/bitrate_controller/send_side_bandwidth_estimation.cc
diff --git a/webrtc/modules/bitrate_controller/send_side_bandwidth_estimation.cc b/webrtc/modules/bitrate_controller/send_side_bandwidth_estimation.cc
index e7bb6361f2ab609254edcab79b9948d85486b0d9..dc1490589078acac93ba3575af814ccd8331f1fe 100644
--- a/webrtc/modules/bitrate_controller/send_side_bandwidth_estimation.cc
+++ b/webrtc/modules/bitrate_controller/send_side_bandwidth_estimation.cc
@@ -33,7 +33,7 @@ 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;