| 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++) {
|
|
|