Add NACK rate throttling for audio channels.

webrtc/modules/rtp_rtcp/source/rtcp_sender_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/gmock/include/gmock/gmock.h"
 #include "testing/gtest/include/gtest/gtest.h"
 
+#include "webrtc/base/rate_limiter.h"
 #include "webrtc/common_types.h"
 #include "webrtc/modules/rtp_rtcp/source/rtcp_sender.h"
 #include "webrtc/modules/rtp_rtcp/source/rtcp_utility.h"
 #include "webrtc/modules/rtp_rtcp/source/rtp_rtcp_impl.h"
 #include "webrtc/test/mock_transport.h"
 #include "webrtc/test/rtcp_packet_parser.h"
 
 using ::testing::_;
 using ::testing::ElementsAre;
 using ::testing::Invoke;
@@ skipping 196 matching lines @@
 static const uint32_t kSenderSsrc = 0x11111111;
 static const uint32_t kRemoteSsrc = 0x22222222;
 static const uint32_t kStartRtpTimestamp = 0x34567;
 static const uint32_t kRtpTimestamp = 0x45678;
 }
 
 class RtcpSenderTest : public ::testing::Test {
  protected:
   RtcpSenderTest()
       : clock_(1335900000),
-        receive_statistics_(ReceiveStatistics::Create(&clock_)) {
+        receive_statistics_(ReceiveStatistics::Create(&clock_)),
+        retransmission_rate_limiter_(&clock_, 1000) {
     RtpRtcp::Configuration configuration;
     configuration.audio = false;
     configuration.clock = &clock_;
     configuration.outgoing_transport = &test_transport_;
+    configuration.retransmission_rate_limiter = &retransmission_rate_limiter_;
 
     rtp_rtcp_impl_.reset(new ModuleRtpRtcpImpl(configuration));
     rtcp_sender_.reset(new RTCPSender(false, &clock_, receive_statistics_.get(),
                                       nullptr, nullptr, &test_transport_));
     rtcp_sender_->SetSSRC(kSenderSsrc);
     rtcp_sender_->SetRemoteSSRC(kRemoteSsrc);
     rtcp_sender_->SetStartTimestamp(kStartRtpTimestamp);
     rtcp_sender_->SetLastRtpTime(kRtpTimestamp, clock_.TimeInMilliseconds());
   }
 
@@ skipping 11 matching lines @@
 
   RTCPSender::FeedbackState feedback_state() {
     return rtp_rtcp_impl_->GetFeedbackState();
   }
 
   SimulatedClock clock_;
   TestTransport test_transport_;
   std::unique_ptr<ReceiveStatistics> receive_statistics_;
   std::unique_ptr<ModuleRtpRtcpImpl> rtp_rtcp_impl_;
   std::unique_ptr<RTCPSender> rtcp_sender_;
+  RateLimiter retransmission_rate_limiter_;
 };
 
 TEST_F(RtcpSenderTest, SetRtcpStatus) {
   EXPECT_EQ(RtcpMode::kOff, rtcp_sender_->Status());
   rtcp_sender_->SetRTCPStatus(RtcpMode::kReducedSize);
   EXPECT_EQ(RtcpMode::kReducedSize, rtcp_sender_->Status());
 }
 
 TEST_F(RtcpSenderTest, SetSendingStatus) {
   EXPECT_FALSE(rtcp_sender_->Sending());
@@ skipping 535 matching lines @@
   rtcp_sender_->SetLastRtpTime(kRtpTimestamp, clock_.TimeInMilliseconds());
 
   // Set up XR VoIP metric to be included with BYE
   rtcp_sender_->SetRTCPStatus(RtcpMode::kCompound);
   RTCPVoIPMetric metric;
   EXPECT_EQ(0, rtcp_sender_->SetRTCPVoIPMetrics(&metric));
   EXPECT_EQ(0, rtcp_sender_->SendRTCP(feedback_state(), kRtcpBye));
 }
 
 }  // namespace webrtc