Remove GetFeedbackInterval in sender side BWE.

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 338 matching lines @@
   // bitrate(i) = 1.04 * bitrate(i-1) + 1000
   // until bitrate(i) > 500000, with bitrate(1) ~= 30000.
   uint32_t bitrate_bps = 30000;
   int iterations = 0;
   AddDefaultStream();
   // Feed the estimator with a stream of packets and verify that it reaches
   // 500 kbps at the expected time.
   while (bitrate_bps < 5e5) {
     bool overuse = GenerateAndProcessFrame(kDefaultSsrc, bitrate_bps);
     if (overuse) {
-      EXPECT_GT(bitrate_observer_.latest_bitrate(), bitrate_bps);
+      EXPECT_LT(bitrate_observer_.latest_bitrate(), bitrate_bps);

[stefan-webrtc, 2016/10/26 15:19:20]

I don't really understand this expectation. I also don't fully understand the
previous expectation at the moment... WDYT the reasoning behind it is?

       bitrate_bps = bitrate_observer_.latest_bitrate();
       bitrate_observer_.Reset();
     } else if (bitrate_observer_.updated()) {
       bitrate_bps = bitrate_observer_.latest_bitrate();
       bitrate_observer_.Reset();
     }
     ++iterations;
     // ASSERT_LE(iterations, expected_iterations);
   }
   ASSERT_EQ(expected_iterations, iterations);
 }
 
 void DelayBasedBweTest::CapacityDropTestHelper(
     int number_of_streams,
     bool wrap_time_stamp,
     uint32_t expected_bitrate_drop_delta,
     int64_t receiver_clock_offset_change_ms) {
-  const int kFramerate = 30;
+  const int kFramerate = 50;

[stefan-webrtc, 2016/10/26 15:19:20]

Could you comment on this change?

   const int kStartBitrate = 900e3;
   const int kMinExpectedBitrate = 800e3;
   const int kMaxExpectedBitrate = 1100e3;
   const uint32_t kInitialCapacityBps = 1000e3;
   const uint32_t kReducedCapacityBps = 500e3;
 
   int steady_state_time = 0;
   if (number_of_streams <= 1) {
     steady_state_time = 10;
     AddDefaultStream();
@@ skipping 12 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