Add NACK rate throttling for audio channels.

webrtc/modules/rtp_rtcp/test/testAPI/test_api_video.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 <stdlib.h>
 
 #include <algorithm>
 #include <memory>
 #include <vector>
 
 #include "testing/gtest/include/gtest/gtest.h"
+#include "webrtc/base/rate_limiter.h"
 #include "webrtc/common_types.h"
 #include "webrtc/modules/rtp_rtcp/include/rtp_payload_registry.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/byte_io.h"
 #include "webrtc/modules/rtp_rtcp/source/rtp_receiver_video.h"
 #include "webrtc/modules/rtp_rtcp/test/testAPI/test_api.h"
 
 namespace {
 
 const unsigned char kPayloadType = 100;
 
 };
 
 namespace webrtc {
 
 class RtpRtcpVideoTest : public ::testing::Test {
  protected:
   RtpRtcpVideoTest()
       : rtp_payload_registry_(RTPPayloadStrategy::CreateStrategy(false)),
         test_ssrc_(3456),
         test_timestamp_(4567),
         test_sequence_number_(2345),
-        fake_clock(123456) {}
+        fake_clock(123456),
+        retransmission_rate_limiter_(&fake_clock, 1000) {}
   ~RtpRtcpVideoTest() {}
 
   virtual void SetUp() {
     transport_ = new LoopBackTransport();
     receiver_ = new TestRtpReceiver();
     receive_statistics_.reset(ReceiveStatistics::Create(&fake_clock));
     RtpRtcp::Configuration configuration;
     configuration.audio = false;
     configuration.clock = &fake_clock;
     configuration.outgoing_transport = transport_;
+    configuration.retransmission_rate_limiter = &retransmission_rate_limiter_;
 
     video_module_ = RtpRtcp::CreateRtpRtcp(configuration);
     rtp_receiver_.reset(RtpReceiver::CreateVideoReceiver(
         &fake_clock, receiver_, NULL, &rtp_payload_registry_));
 
     video_module_->SetRTCPStatus(RtcpMode::kCompound);
     video_module_->SetSSRC(test_ssrc_);
     video_module_->SetStorePacketsStatus(true, 600);
     EXPECT_EQ(0, video_module_->SetSendingStatus(true));
 
@@ skipping 70 matching lines @@
   std::unique_ptr<RtpReceiver> rtp_receiver_;
   RtpRtcp* video_module_;
   LoopBackTransport* transport_;
   TestRtpReceiver* receiver_;
   uint32_t test_ssrc_;
   uint32_t test_timestamp_;
   uint16_t test_sequence_number_;
   uint8_t  video_frame_[65000];
   size_t payload_data_length_;
   SimulatedClock fake_clock;
+  RateLimiter retransmission_rate_limiter_;
 };
 
 TEST_F(RtpRtcpVideoTest, BasicVideo) {
   uint32_t timestamp = 3000;
   EXPECT_EQ(0, video_module_->SendOutgoingData(kVideoFrameDelta, 123,
                                                timestamp,
                                                timestamp / 90,
                                                video_frame_,
                                                payload_data_length_));
 }
@@ skipping 30 matching lines @@
                                                    payload_specific, true));
       EXPECT_EQ(0u, receiver_->payload_size());
       EXPECT_EQ(payload_length, receiver_->rtp_header().header.paddingLength);
     }
     timestamp += 3000;
     fake_clock.AdvanceTimeMilliseconds(33);
   }
 }
 
 }  // namespace webrtc