Add NACK rate throttling for audio channels.

webrtc/modules/rtp_rtcp/source/rtcp_receiver_unittest.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 <memory>
 
 #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/remote_bitrate_estimator/include/mock/mock_remote_bitrate_observer.h"
 #include "webrtc/modules/remote_bitrate_estimator/remote_bitrate_estimator_single_stream.h"
 #include "webrtc/modules/rtp_rtcp/source/byte_io.h"
 #include "webrtc/modules/rtp_rtcp/source/rtcp_packet.h"
 #include "webrtc/modules/rtp_rtcp/source/rtcp_packet/app.h"
 #include "webrtc/modules/rtp_rtcp/source/rtcp_packet/bye.h"
 #include "webrtc/modules/rtp_rtcp/source/rtcp_packet/compound_packet.h"
 #include "webrtc/modules/rtp_rtcp/source/rtcp_packet/extended_jitter_report.h"
 #include "webrtc/modules/rtp_rtcp/source/rtcp_packet/extended_reports.h"
@@ skipping 47 matching lines @@
 };
 
 class RtcpReceiverTest : public ::testing::Test {
  protected:
   RtcpReceiverTest()
       : over_use_detector_options_(),
         system_clock_(1335900000),
         remote_bitrate_observer_(),
         remote_bitrate_estimator_(
             new RemoteBitrateEstimatorSingleStream(&remote_bitrate_observer_,
-                                                   &system_clock_)) {
-    test_transport_ = new TestTransport();
+                                                   &system_clock_)),
+        retransmission_rate_limiter_(&system_clock_, 1000) {
+    test_transport_.reset(new TestTransport());
 
     RtpRtcp::Configuration configuration;
     configuration.audio = false;
     configuration.clock = &system_clock_;
-    configuration.outgoing_transport = test_transport_;
+    configuration.outgoing_transport = test_transport_.get();
     configuration.remote_bitrate_estimator = remote_bitrate_estimator_.get();
-    rtp_rtcp_impl_ = new ModuleRtpRtcpImpl(configuration);
-    rtcp_receiver_ = new RTCPReceiver(&system_clock_, false, nullptr, nullptr,
-                                      nullptr, nullptr, rtp_rtcp_impl_);
-    test_transport_->SetRTCPReceiver(rtcp_receiver_);
+    configuration.retransmission_rate_limiter = &retransmission_rate_limiter_;
+    rtp_rtcp_impl_.reset(new ModuleRtpRtcpImpl(configuration));
+    rtcp_receiver_.reset(new RTCPReceiver(&system_clock_, false, nullptr,
+                                          nullptr, nullptr, nullptr,
+                                          rtp_rtcp_impl_.get()));
+    test_transport_->SetRTCPReceiver(rtcp_receiver_.get());
   }
-  ~RtcpReceiverTest() {
-    delete rtcp_receiver_;
-    delete rtp_rtcp_impl_;
-    delete test_transport_;
-  }
+
+  ~RtcpReceiverTest() {}
 
   // Injects an RTCP packet into the receiver.
   // Returns 0 for OK, non-0 for failure.
   int InjectRtcpPacket(const uint8_t* packet,
                        uint16_t packet_len) {
     RTCPUtility::RTCPParserV2 rtcpParser(packet,
                                          packet_len,
                                          true);  // Allow non-compound RTCP
 
     RTCPHelp::RTCPPacketInformation rtcpPacketInformation;
@@ skipping 24 matching lines @@
     rtcp_packet_info_.xr_dlrr_item = rtcpPacketInformation.xr_dlrr_item;
     if (rtcpPacketInformation.VoIPMetric)
       rtcp_packet_info_.AddVoIPMetric(rtcpPacketInformation.VoIPMetric.get());
     rtcp_packet_info_.transport_feedback_.reset(
         rtcpPacketInformation.transport_feedback_.release());
     return 0;
   }
 
   OverUseDetectorOptions over_use_detector_options_;
   SimulatedClock system_clock_;
-  ModuleRtpRtcpImpl* rtp_rtcp_impl_;
-  RTCPReceiver* rtcp_receiver_;
-  TestTransport* test_transport_;
+  std::unique_ptr<TestTransport> test_transport_;
+  std::unique_ptr<ModuleRtpRtcpImpl> rtp_rtcp_impl_;
+  std::unique_ptr<RTCPReceiver> rtcp_receiver_;
   RTCPHelp::RTCPPacketInformation rtcp_packet_info_;
   MockRemoteBitrateObserver remote_bitrate_observer_;
   std::unique_ptr<RemoteBitrateEstimator> remote_bitrate_estimator_;
+  RateLimiter retransmission_rate_limiter_;
 };
 
 
 TEST_F(RtcpReceiverTest, BrokenPacketIsIgnored) {
   const uint8_t bad_packet[] = {0, 0, 0, 0};
   EXPECT_EQ(0, InjectRtcpPacket(bad_packet, sizeof(bad_packet)));
   EXPECT_EQ(0U, rtcp_packet_info_.rtcpPacketTypeFlags);
 }
 
 TEST_F(RtcpReceiverTest, InvalidFeedbackPacketIsIgnored) {
@@ skipping 1128 matching lines @@
 
   // Transport feedback should be ignored, but next packet should work.
   EXPECT_EQ(0u, rtcp_packet_info_.rtcpPacketTypeFlags & kRtcpTransportFeedback);
   EXPECT_NE(0u, rtcp_packet_info_.rtcpPacketTypeFlags & kRtcpRemb);
   EXPECT_EQ(kBitrateBps, rtcp_packet_info_.receiverEstimatedMaxBitrate);
 }
 
 }  // Anonymous namespace
 
 }  // namespace webrtc