Reduce send rate to 50% if overusing before we have an acknowledged bitrate.

webrtc/modules/congestion_controller/delay_based_bwe_unittest.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.
  */
 
@@ skipping 176 matching lines @@
 }
 
 TEST_F(DelayBasedBweTest, TestLongTimeoutAndWrap) {
   // Simulate a client leaving and rejoining the call after some multiple of
   // 64 seconds later. This will cause a zero difference in abs send times due
   // to the wrap, but a big difference in arrival time, if streams aren't
   // properly timed out.
   TestWrappingHelper(10 * 64);
 }
 
+TEST_F(DelayBasedBweTest, TestInitialOveruse) {
+  const uint32_t kStartBitrate = 300e3;
+  const uint32_t kInitialCapacityBps = 200e3;
+  const uint32_t kDummySsrc = 0;
+  // High FPS to ensure that we send a lot of packets in a short time.
+  const int kFps = 90;
+
+  stream_generator_->AddStream(new test::RtpStream(kFps, kStartBitrate));
+  stream_generator_->set_capacity_bps(kInitialCapacityBps);
+
+  // Needed to initialize the AimdRateControl.
+  bitrate_estimator_->SetStartBitrate(kStartBitrate);
+
+  // Produce 30 frames (in 1/3 second) and give them to the estimator.
+  uint32_t bitrate_bps = kStartBitrate;
+  bool seen_overuse = false;
+  for (int i = 0; i < 30; ++i) {
+    bool overuse = GenerateAndProcessFrame(kDummySsrc, bitrate_bps);
+    // The purpose of this test is to ensure that we back down even if we don't
+    // have any acknowledged bitrate estimate yet. Hence, if the test works
+    // as expected, we should not have a measured bitrate yet.
+    EXPECT_FALSE(acknowledged_bitrate_estimator_->bitrate_bps().has_value());
+    if (overuse) {
+      EXPECT_TRUE(bitrate_observer_.updated());
+      EXPECT_NEAR(bitrate_observer_.latest_bitrate(), kStartBitrate / 2, 15000);
+      bitrate_bps = bitrate_observer_.latest_bitrate();
+      seen_overuse = true;
+      break;
+    } else if (bitrate_observer_.updated()) {
+      bitrate_bps = bitrate_observer_.latest_bitrate();
+      bitrate_observer_.Reset();
+    }
+  }
+  EXPECT_TRUE(seen_overuse);
+  EXPECT_NEAR(bitrate_observer_.latest_bitrate(), kStartBitrate / 2, 15000);
+}
+
 }  // namespace webrtc