Add NACK rate throttling for audio channels.

webrtc/modules/rtp_rtcp/test/testAPI/test_api_rtcp.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 <algorithm>
 #include <memory>
 #include <vector>
 
 #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/include/receive_statistics.h"
 #include "webrtc/modules/rtp_rtcp/include/rtp_rtcp.h"
 #include "webrtc/modules/rtp_rtcp/include/rtp_rtcp_defines.h"
 #include "webrtc/modules/rtp_rtcp/source/rtp_receiver_audio.h"
 #include "webrtc/modules/rtp_rtcp/test/testAPI/test_api.h"
 
 namespace webrtc {
 namespace {
 
@@ skipping 32 matching lines @@
   void OnIncomingSSRCChanged(const uint32_t ssrc) override {
     rtp_rtcp_->SetRemoteSSRC(ssrc);
   }
 
  private:
   RtpRtcp* rtp_rtcp_;
 };
 
 class RtpRtcpRtcpTest : public ::testing::Test {
  protected:
-  RtpRtcpRtcpTest() : fake_clock(123456) {
+  RtpRtcpRtcpTest()
+      : fake_clock(123456), retransmission_rate_limiter_(&fake_clock, 1000) {
     test_csrcs.push_back(1234);
     test_csrcs.push_back(2345);
     test_ssrc = 3456;
     test_timestamp = 4567;
     test_sequence_number = 2345;
   }
   ~RtpRtcpRtcpTest() {}
 
   virtual void SetUp() {
     receiver = new TestRtpReceiver();
     transport1 = new LoopBackTransport();
     transport2 = new LoopBackTransport();
     myRTCPFeedback1 = new RtcpCallback();
     myRTCPFeedback2 = new RtcpCallback();
 
     receive_statistics1_.reset(ReceiveStatistics::Create(&fake_clock));
     receive_statistics2_.reset(ReceiveStatistics::Create(&fake_clock));
 
     RtpRtcp::Configuration configuration;
     configuration.audio = true;
     configuration.clock = &fake_clock;
     configuration.receive_statistics = receive_statistics1_.get();
     configuration.outgoing_transport = transport1;
     configuration.intra_frame_callback = myRTCPFeedback1;
+    configuration.retransmission_rate_limiter = &retransmission_rate_limiter_;
 
     rtp_payload_registry1_.reset(new RTPPayloadRegistry(
             RTPPayloadStrategy::CreateStrategy(true)));
     rtp_payload_registry2_.reset(new RTPPayloadRegistry(
             RTPPayloadStrategy::CreateStrategy(true)));
 
     module1 = RtpRtcp::CreateRtpRtcp(configuration);
 
     rtp_feedback1_.reset(new TestRtpFeedback(module1));
 
@@ skipping 86 matching lines @@
   LoopBackTransport* transport1;
   LoopBackTransport* transport2;
   RtcpCallback* myRTCPFeedback1;
   RtcpCallback* myRTCPFeedback2;
 
   uint32_t test_ssrc;
   uint32_t test_timestamp;
   uint16_t test_sequence_number;
   std::vector<uint32_t> test_csrcs;
   SimulatedClock fake_clock;
+  RateLimiter retransmission_rate_limiter_;
 };
 
 TEST_F(RtpRtcpRtcpTest, RTCP_PLI_RPSI) {
   EXPECT_EQ(0, module1->SendRTCPReferencePictureSelection(kTestPictureId));
   EXPECT_EQ(0, module1->SendRTCPSliceLossIndication(kSliPictureId));
 }
 
 TEST_F(RtpRtcpRtcpTest, RTCP_CNAME) {
   uint32_t testOfCSRC[webrtc::kRtpCsrcSize];
   EXPECT_EQ(2, rtp_receiver2_->CSRCs(testOfCSRC));
@@ skipping 53 matching lines @@
   EXPECT_EQ(test_ssrc, report_blocks[0].sourceSSRC);
 
   EXPECT_EQ(0u, report_blocks[0].cumulativeLost);
   EXPECT_LT(0u, report_blocks[0].delaySinceLastSR);
   EXPECT_EQ(test_sequence_number, report_blocks[0].extendedHighSeqNum);
   EXPECT_EQ(0u, report_blocks[0].fractionLost);
 }
 
 }  // namespace
 }  // namespace webrtc