Set min BWE bitrate form 10kbps to 5kbps and centralize minimum bitrate.

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.
  */
 
 #include "webrtc/modules/bitrate_controller/send_side_bandwidth_estimation.h"
 
 #include <algorithm>
 #include <cmath>
 
 #include "webrtc/base/checks.h"
 #include "webrtc/base/logging.h"
 #include "webrtc/logging/rtc_event_log/rtc_event_log.h"
+#include "webrtc/modules/congestion_controller/include/defines.h"
 #include "webrtc/system_wrappers/include/field_trial.h"
 #include "webrtc/system_wrappers/include/metrics.h"
 
 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;
 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 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),
+      min_bitrate_configured_(congestion_controller::kMinBitrateBps),
       max_bitrate_configured_(kDefaultMaxBitrateBps),
       last_low_bitrate_log_ms_(-1),
       has_decreased_since_last_fraction_loss_(false),
       last_feedback_ms_(-1),
       last_packet_report_ms_(-1),
       last_timeout_ms_(-1),
       last_fraction_loss_(0),
       last_logged_fraction_loss_(0),
       last_round_trip_time_ms_(0),
       bwe_incoming_(0),
@@ skipping 26 matching lines @@
   bitrate_ = bitrate;
 
   // Clear last sent bitrate history so the new value can be used directly
   // and not capped.
   min_bitrate_history_.clear();
 }
 
 void SendSideBandwidthEstimation::SetMinMaxBitrate(int min_bitrate,
                                                    int max_bitrate) {
   RTC_DCHECK_GE(min_bitrate, 0);
-  min_bitrate_configured_ = std::max(min_bitrate, kDefaultMinBitrateBps);
+  min_bitrate_configured_ =
+      std::max(min_bitrate, congestion_controller::kMinBitrateBps);
   if (max_bitrate > 0) {
     max_bitrate_configured_ =
         std::max<uint32_t>(min_bitrate_configured_, max_bitrate);
   } else {
     max_bitrate_configured_ = kDefaultMaxBitrateBps;
   }
 }
 
 int SendSideBandwidthEstimation::GetMinBitrate() const {
   return min_bitrate_configured_;
@@ skipping 221 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