Add NACK rate throttling for audio channels.

webrtc/modules/rtp_rtcp/source/rtcp_format_remb_unittest.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 <memory>
 
 #include "testing/gtest/include/gtest/gtest.h"
 
+#include "webrtc/base/rate_limiter.h"
 #include "webrtc/common_types.h"
 #include "webrtc/modules/remote_bitrate_estimator/include/bwe_defines.h"
 #include "webrtc/modules/remote_bitrate_estimator/include/mock/mock_remote_bitrate_observer.h"
 #include "webrtc/modules/remote_bitrate_estimator/remote_bitrate_estimator_single_stream.h"
 #include "webrtc/modules/rtp_rtcp/source/rtcp_receiver.h"
 #include "webrtc/modules/rtp_rtcp/source/rtcp_sender.h"
 #include "webrtc/modules/rtp_rtcp/source/rtp_rtcp_impl.h"
 #include "webrtc/modules/rtp_rtcp/source/rtp_utility.h"
 #include "webrtc/test/null_transport.h"
 #include "webrtc/typedefs.h"
@@ skipping 37 matching lines @@
       : over_use_detector_options_(),
         system_clock_(Clock::GetRealTimeClock()),
         dummy_rtp_rtcp_impl_(nullptr),
         receive_statistics_(ReceiveStatistics::Create(system_clock_)),
         rtcp_sender_(nullptr),
         rtcp_receiver_(nullptr),
         test_transport_(nullptr),
         remote_bitrate_observer_(),
         remote_bitrate_estimator_(
             new RemoteBitrateEstimatorSingleStream(&remote_bitrate_observer_,
-                                                   system_clock_)) {}
+                                                   system_clock_)),
+        retransmission_rate_limiter_(Clock::GetRealTimeClock(), 1000) {}
   void SetUp() override;
   void TearDown() override;
 
   OverUseDetectorOptions over_use_detector_options_;
   Clock* system_clock_;
   ModuleRtpRtcpImpl* dummy_rtp_rtcp_impl_;
   std::unique_ptr<ReceiveStatistics> receive_statistics_;
   RTCPSender* rtcp_sender_;
   RTCPReceiver* rtcp_receiver_;
   TestTransport* test_transport_;
   test::NullTransport null_transport_;
   MockRemoteBitrateObserver remote_bitrate_observer_;
   std::unique_ptr<RemoteBitrateEstimator> remote_bitrate_estimator_;
+  RateLimiter retransmission_rate_limiter_;
 };
 
 void RtcpFormatRembTest::SetUp() {
   RtpRtcp::Configuration configuration;
   configuration.audio = false;
   configuration.clock = system_clock_;
   configuration.remote_bitrate_estimator = remote_bitrate_estimator_.get();
   configuration.outgoing_transport = &null_transport_;
+  configuration.retransmission_rate_limiter = &retransmission_rate_limiter_;
   dummy_rtp_rtcp_impl_ = new ModuleRtpRtcpImpl(configuration);
   rtcp_receiver_ = new RTCPReceiver(system_clock_, false, nullptr, nullptr,
                                     nullptr, nullptr, dummy_rtp_rtcp_impl_);
   test_transport_ = new TestTransport(rtcp_receiver_);
   rtcp_sender_ = new RTCPSender(false, system_clock_, receive_statistics_.get(),
                                 nullptr, nullptr, test_transport_);
 }
 
 void RtcpFormatRembTest::TearDown() {
   delete rtcp_sender_;
@@ skipping 22 matching lines @@
 TEST_F(RtcpFormatRembTest, TestCompund) {
   uint32_t SSRCs[2] = {456789, 98765};
   rtcp_sender_->SetRTCPStatus(RtcpMode::kCompound);
   rtcp_sender_->SetREMBData(1234, std::vector<uint32_t>(SSRCs, SSRCs + 2));
   RTCPSender::FeedbackState feedback_state =
       dummy_rtp_rtcp_impl_->GetFeedbackState();
   EXPECT_EQ(0, rtcp_sender_->SendRTCP(feedback_state, kRtcpRemb));
 }
 }  // namespace
 }  // namespace webrtc