Add NACK rate throttling for audio channels.

webrtc/modules/rtp_rtcp/source/nack_rtx_unittest.cc

 /*
 *  Copyright (c) 2013 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.
 */
 
@@ skipping 157 matching lines @@
 
 class RtpRtcpRtxNackTest : public ::testing::Test {
  protected:
   RtpRtcpRtxNackTest()
       : rtp_payload_registry_(RTPPayloadStrategy::CreateStrategy(false)),
         rtp_rtcp_module_(nullptr),
         transport_(kTestSsrc + 1),
         receiver_(),
         payload_data_length(sizeof(payload_data)),
         fake_clock(123456),
-        retranmission_rate_limiter_(&fake_clock, kMaxRttMs) {}
+        retransmission_rate_limiter_(&fake_clock, kMaxRttMs) {}
   ~RtpRtcpRtxNackTest() {}
 
   void SetUp() override {
     RtpRtcp::Configuration configuration;
     configuration.audio = false;
     configuration.clock = &fake_clock;
     receive_statistics_.reset(ReceiveStatistics::Create(&fake_clock));
     configuration.receive_statistics = receive_statistics_.get();
     configuration.outgoing_transport = &transport_;
-    configuration.retransmission_rate_limiter = &retranmission_rate_limiter_;
+    configuration.retransmission_rate_limiter = &retransmission_rate_limiter_;
     rtp_rtcp_module_ = RtpRtcp::CreateRtpRtcp(configuration);
 
     rtp_feedback_.reset(new TestRtpFeedback(rtp_rtcp_module_));
 
     rtp_receiver_.reset(RtpReceiver::CreateVideoReceiver(
         &fake_clock, &receiver_, rtp_feedback_.get(), &rtp_payload_registry_));
 
     rtp_rtcp_module_->SetSSRC(kTestSsrc);
     rtp_rtcp_module_->SetRTCPStatus(RtcpMode::kCompound);
     rtp_rtcp_module_->SetStorePacketsStatus(true, 600);
@@ skipping 86 matching lines @@
   std::unique_ptr<ReceiveStatistics> receive_statistics_;
   RTPPayloadRegistry rtp_payload_registry_;
   std::unique_ptr<RtpReceiver> rtp_receiver_;
   RtpRtcp* rtp_rtcp_module_;
   std::unique_ptr<TestRtpFeedback> rtp_feedback_;
   RtxLoopBackTransport transport_;
   VerifyingRtxReceiver receiver_;
   uint8_t payload_data[65000];
   size_t payload_data_length;
   SimulatedClock fake_clock;
-  RateLimiter retranmission_rate_limiter_;
+  RateLimiter retransmission_rate_limiter_;
 };
 
 TEST_F(RtpRtcpRtxNackTest, LongNackList) {
   const int kNumPacketsToDrop = 900;
   const int kNumRequiredRtcp = 4;
   uint32_t timestamp = 3000;
   uint16_t nack_list[kNumPacketsToDrop];
   // Disable StorePackets to be able to set a larger packet history.
   rtp_rtcp_module_->SetStorePacketsStatus(false, 0);
   // Enable StorePackets with a packet history of 2000 packets.
@@ skipping 31 matching lines @@
   RunRtxTest(kRtxRetransmitted, 10);
   EXPECT_EQ(kTestSequenceNumber, *(receiver_.sequence_numbers_.begin()));
   EXPECT_EQ(kTestSequenceNumber + kTestNumberOfPackets - 1,
             *(receiver_.sequence_numbers_.rbegin()));
   EXPECT_EQ(kTestNumberOfPackets, receiver_.sequence_numbers_.size());
   EXPECT_EQ(kTestNumberOfRtxPackets, transport_.count_rtx_ssrc_);
   EXPECT_TRUE(ExpectedPacketsReceived());
 }
 
 }  // namespace webrtc