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Unified Diff: webrtc/call/rtc_event_log_unittest.cc

Issue 2181383002: Add NACK rate throttling for audio channels. (Closed) Base URL: https://chromium.googlesource.com/external/webrtc.git@master
Patch Set: Fixed race in test Created 4 years, 5 months ago
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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 2d583a928e0859b4e239895f2ab3a59f15af0d71..16d9159dc7bfd65af9f9e94db1e04cf7f6102c52 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);
stefan-webrtc 2016/07/28 15:06:15 retransmission
sprang_webrtc 2016/07/28 15:17:17 Done.
+
+ 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++) {
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