Index: webrtc/call/rtc_event_log_unittest.cc |
diff --git a/webrtc/call/rtc_event_log_unittest.cc b/webrtc/call/rtc_event_log_unittest.cc |
index f8b555b63d2e956d4603fce19968c30f428d7dde..57d18b7e72e5b783debfc9f9ff2a1b901bc0c88f 100644 |
--- a/webrtc/call/rtc_event_log_unittest.cc |
+++ b/webrtc/call/rtc_event_log_unittest.cc |
@@ -20,6 +20,7 @@ |
#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" |
@@ -111,19 +112,20 @@ size_t GenerateRtpPacket(uint32_t extensions_bitvector, |
Random* prng) { |
RTC_CHECK_GE(packet_size, 16 + 4 * csrcs_count + 4 * kNumExtensions); |
Clock* clock = Clock::GetRealTimeClock(); |
- |
- 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* |
+ 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* |
+ &retranmission_rate_limiter); |
std::vector<uint32_t> csrcs; |
for (unsigned i = 0; i < csrcs_count; i++) { |