Add NACK rate throttling for audio channels.

webrtc/modules/rtp_rtcp/test/testAPI/test_api.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 "webrtc/modules/rtp_rtcp/test/testAPI/test_api.h"
 
 #include <algorithm>
 #include <memory>
 #include <vector>
 
+#include "webrtc/base/rate_limiter.h"
 #include "webrtc/test/null_transport.h"
 
 namespace webrtc {
 
 void LoopBackTransport::SetSendModule(RtpRtcp* rtp_rtcp_module,
                                       RTPPayloadRegistry* payload_registry,
                                       RtpReceiver* receiver,
                                       ReceiveStatistics* receive_statistics) {
   rtp_rtcp_module_ = rtp_rtcp_module;
   rtp_payload_registry_ = payload_registry;
@@ skipping 46 matching lines @@
     const webrtc::WebRtcRTPHeader* rtp_header) {
   EXPECT_LE(payload_size, sizeof(payload_data_));
   memcpy(payload_data_, payload_data, payload_size);
   memcpy(&rtp_header_, rtp_header, sizeof(rtp_header_));
   payload_size_ = payload_size;
   return 0;
 }
 
 class RtpRtcpAPITest : public ::testing::Test {
  protected:
-  RtpRtcpAPITest() : fake_clock_(123456) {
+  RtpRtcpAPITest()
+      : 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;
   }
   ~RtpRtcpAPITest() {}
 
   void SetUp() override {
     RtpRtcp::Configuration configuration;
     configuration.audio = true;
     configuration.clock = &fake_clock_;
     configuration.outgoing_transport = &null_transport_;
+    configuration.retransmission_rate_limiter = &retransmission_rate_limiter_;
     module_.reset(RtpRtcp::CreateRtpRtcp(configuration));
     rtp_payload_registry_.reset(new RTPPayloadRegistry(
             RTPPayloadStrategy::CreateStrategy(true)));
     rtp_receiver_.reset(RtpReceiver::CreateAudioReceiver(
         &fake_clock_, NULL, NULL, rtp_payload_registry_.get()));
   }
 
   std::unique_ptr<RTPPayloadRegistry> rtp_payload_registry_;
   std::unique_ptr<RtpReceiver> rtp_receiver_;
   std::unique_ptr<RtpRtcp> module_;
   uint32_t test_ssrc_;
   uint32_t test_timestamp_;
   uint16_t test_sequence_number_;
   std::vector<uint32_t> test_csrcs_;
   SimulatedClock fake_clock_;
   test::NullTransport null_transport_;
+  RateLimiter retransmission_rate_limiter_;
 };
 
 TEST_F(RtpRtcpAPITest, Basic) {
   module_->SetSequenceNumber(test_sequence_number_);
   EXPECT_EQ(test_sequence_number_, module_->SequenceNumber());
 
   module_->SetStartTimestamp(test_timestamp_);
   EXPECT_EQ(test_timestamp_, module_->StartTimestamp());
 
   EXPECT_FALSE(module_->Sending());
@@ skipping 55 matching lines @@
   rtx_header.payloadType = kRtxPayloadType;
   EXPECT_TRUE(rtp_payload_registry_->IsRtx(rtx_header));
   rtx_header.ssrc = 0;
   EXPECT_FALSE(rtp_payload_registry_->IsRtx(rtx_header));
   rtx_header.ssrc = kRtxSsrc;
   rtx_header.payloadType = 0;
   EXPECT_TRUE(rtp_payload_registry_->IsRtx(rtx_header));
 }
 
 }  // namespace webrtc