Enable the bayesian bitrate estimator by default.

webrtc/modules/congestion_controller/delay_based_bwe_unittest_helper.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.
  */
 #include "webrtc/modules/congestion_controller/delay_based_bwe_unittest_helper.h"
@@ skipping 393 matching lines @@
       bitrate_sum += bitrate;
     }
     ASSERT_EQ(bitrate_sum, kStartBitrate);
   }
 
   // Run in steady state to make the estimator converge.
   stream_generator_->set_capacity_bps(kInitialCapacityBps);
   uint32_t bitrate_bps = SteadyStateRun(
       kDefaultSsrc, steady_state_time * kFramerate, kStartBitrate,
       kMinExpectedBitrate, kMaxExpectedBitrate, kInitialCapacityBps);
-  EXPECT_NEAR(kInitialCapacityBps, bitrate_bps, 130000u);
+  EXPECT_NEAR(kInitialCapacityBps, bitrate_bps, 150000u);
   bitrate_observer_.Reset();
 
   // Add an offset to make sure the BWE can handle it.
   arrival_time_offset_ms_ += receiver_clock_offset_change_ms;
 
   // Reduce the capacity and verify the decrease time.
   stream_generator_->set_capacity_bps(kReducedCapacityBps);
   int64_t overuse_start_time = clock_.TimeInMilliseconds();
   int64_t bitrate_drop_time = -1;
   for (int i = 0; i < 100 * number_of_streams; ++i) {
@@ skipping 71 matching lines @@
     IncomingFeedback(clock_.TimeInMilliseconds(), send_time_ms,
                      sequence_number++, 1000);
     clock_.AdvanceTimeMilliseconds(2 * kFrameIntervalMs);
     send_time_ms += kFrameIntervalMs;
   }
   uint32_t bitrate_after = 0;
   bitrate_estimator_->LatestEstimate(&ssrcs, &bitrate_after);
   EXPECT_LT(bitrate_after, bitrate_before);
 }
 }  // namespace webrtc