Add NACK rate throttling for audio channels.

webrtc/call/rtc_event_log_unittest.cc

 /*
  *  Copyright (c) 2015 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.
  */
 
 #ifdef ENABLE_RTC_EVENT_LOG
 
 #include <map>
 #include <memory>
 #include <string>
 #include <utility>
 #include <vector>
 
 #include "testing/gtest/include/gtest/gtest.h"
 #include "webrtc/base/buffer.h"
 #include "webrtc/base/checks.h"
 #include "webrtc/base/random.h"
+#include "webrtc/base/rate_limiter.h"
 #include "webrtc/call.h"
 #include "webrtc/call/rtc_event_log.h"
 #include "webrtc/call/rtc_event_log_parser.h"
 #include "webrtc/call/rtc_event_log_unittest_helper.h"
 #include "webrtc/modules/rtp_rtcp/source/rtcp_packet.h"
 #include "webrtc/modules/rtp_rtcp/source/rtcp_packet/sender_report.h"
 #include "webrtc/modules/rtp_rtcp/source/rtp_sender.h"
 #include "webrtc/system_wrappers/include/clock.h"
 #include "webrtc/test/test_suite.h"
 #include "webrtc/test/testsupport/fileutils.h"
@@ skipping 71 matching lines @@
  * presence of extension number i from kExtensionTypes / kExtensionNames.
  * The least significant bit extension_bitvector has number 0.
  */
 size_t GenerateRtpPacket(uint32_t extensions_bitvector,
                          uint32_t csrcs_count,
                          uint8_t* packet,
                          size_t packet_size,
                          Random* prng) {
   RTC_CHECK_GE(packet_size, 16 + 4 * csrcs_count + 4 * kNumExtensions);
   Clock* clock = Clock::GetRealTimeClock();
+  RateLimiter retranmission_rate_limiter(clock, 1000);
 
-  RTPSender rtp_sender(false,     // bool audio
-                       clock,     // Clock* clock
-                       nullptr,   // Transport*
-                       nullptr,   // PacedSender*
-                       nullptr,   // PacketRouter*
-                       nullptr,   // SendTimeObserver*
-                       nullptr,   // BitrateStatisticsObserver*
-                       nullptr,   // FrameCountObserver*
-                       nullptr,   // SendSideDelayObserver*
-                       nullptr,   // RtcEventLog*
-                       nullptr,   // SendPacketObserver*
-                       nullptr);  // NackRateLimiter*
+  RTPSender rtp_sender(false,    // bool audio
+                       clock,    // Clock* clock
+                       nullptr,  // Transport*
+                       nullptr,  // PacedSender*
+                       nullptr,  // PacketRouter*
+                       nullptr,  // SendTimeObserver*
+                       nullptr,  // BitrateStatisticsObserver*
+                       nullptr,  // FrameCountObserver*
+                       nullptr,  // SendSideDelayObserver*
+                       nullptr,  // RtcEventLog*
+                       nullptr,  // SendPacketObserver*
+                       &retranmission_rate_limiter);
 
   std::vector<uint32_t> csrcs;
   for (unsigned i = 0; i < csrcs_count; i++) {
     csrcs.push_back(prng->Rand<uint32_t>());
   }
   rtp_sender.SetCsrcs(csrcs);
   rtp_sender.SetSSRC(prng->Rand<uint32_t>());
   rtp_sender.SetStartTimestamp(prng->Rand<uint32_t>(), true);
   rtp_sender.SetSequenceNumber(prng->Rand<uint16_t>());
 
@@ skipping 331 matching lines @@
                                          rtcp_packet.size());
 
   RtcEventLogTestHelper::VerifyLogEndEvent(parsed_log, 3);
 
   // Clean up temporary file - can be pretty slow.
   remove(temp_filename.c_str());
 }
 }  // namespace webrtc
 
 #endif  // ENABLE_RTC_EVENT_LOG