Add NACK rate throttling for audio channels.

webrtc/modules/rtp_rtcp/test/testAPI/test_api_audio.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/gtest/include/gtest/gtest.h"
 
 #include "webrtc/modules/rtp_rtcp/test/testAPI/test_api.h"
 
+#include "webrtc/base/rate_limiter.h"
 #include "webrtc/common_types.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"
 
 namespace webrtc {
 namespace {
 #define test_rate 64000u
 
 class VerifyingAudioReceiver : public NullRtpData {
@@ skipping 42 matching lines @@
     if (payloadType == 96) {
       EXPECT_EQ(test_rate, rate) <<
           "The rate should be 64K for this payloadType";
     }
     return 0;
   }
 };
 
 class RtpRtcpAudioTest : public ::testing::Test {
  protected:
-  RtpRtcpAudioTest() : fake_clock(123456) {
+  RtpRtcpAudioTest()
+      : fake_clock(123456), retransmission_rate_limiter_(&fake_clock, 1000) {
     test_CSRC[0] = 1234;
     test_CSRC[2] = 2345;
     test_ssrc = 3456;
     test_timestamp = 4567;
     test_sequence_number = 2345;
   }
   ~RtpRtcpAudioTest() {}
 
   void SetUp() override {
     data_receiver1 = new VerifyingAudioReceiver();
     data_receiver2 = new VerifyingAudioReceiver();
     rtp_callback = new RTPCallback();
     transport1 = new LoopBackTransport();
     transport2 = new LoopBackTransport();
 
     receive_statistics1_.reset(ReceiveStatistics::Create(&fake_clock));
     receive_statistics2_.reset(ReceiveStatistics::Create(&fake_clock));
 
     rtp_payload_registry1_.reset(new RTPPayloadRegistry(
         RTPPayloadStrategy::CreateStrategy(true)));
     rtp_payload_registry2_.reset(new RTPPayloadRegistry(
         RTPPayloadStrategy::CreateStrategy(true)));
 
     RtpRtcp::Configuration configuration;
     configuration.audio = true;
     configuration.clock = &fake_clock;
     configuration.receive_statistics = receive_statistics1_.get();
     configuration.outgoing_transport = transport1;
+    configuration.retransmission_rate_limiter = &retransmission_rate_limiter_;
 
     module1 = RtpRtcp::CreateRtpRtcp(configuration);
     rtp_receiver1_.reset(RtpReceiver::CreateAudioReceiver(
         &fake_clock, data_receiver1, NULL, rtp_payload_registry1_.get()));
 
     configuration.receive_statistics = receive_statistics2_.get();
     configuration.outgoing_transport = transport2;
 
     module2 = RtpRtcp::CreateRtpRtcp(configuration);
     rtp_receiver2_.reset(RtpReceiver::CreateAudioReceiver(
@@ skipping 26 matching lines @@
   VerifyingAudioReceiver* data_receiver1;
   VerifyingAudioReceiver* data_receiver2;
   LoopBackTransport* transport1;
   LoopBackTransport* transport2;
   RTPCallback* rtp_callback;
   uint32_t test_ssrc;
   uint32_t test_timestamp;
   uint16_t test_sequence_number;
   uint32_t test_CSRC[webrtc::kRtpCsrcSize];
   SimulatedClock fake_clock;
+  RateLimiter retransmission_rate_limiter_;
 };
 
 TEST_F(RtpRtcpAudioTest, Basic) {
   module1->SetSSRC(test_ssrc);
   module1->SetStartTimestamp(test_timestamp);
 
   // Test detection at the end of a DTMF tone.
   // EXPECT_EQ(0, module2->SetTelephoneEventForwardToDecoder(true));
 
   EXPECT_EQ(0, module1->SetSendingStatus(true));
@@ skipping 174 matching lines @@
   for (; timeStamp <= 740 * 160; timeStamp += 160) {
     EXPECT_EQ(0, module1->SendOutgoingData(webrtc::kAudioFrameSpeech, 96,
                                            timeStamp, -1, test, 4));
     fake_clock.AdvanceTimeMilliseconds(20);
     module1->Process();
   }
 }
 
 }  // namespace
 }  // namespace webrtc