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Unified Diff: webrtc/call/rtc_event_log_unittest.cc

Issue 1803923003: Clean away use of RtpAudioFeedback interface from RTP/RTCP sender code. (Closed) Base URL: https://chromium.googlesource.com/external/webrtc.git@master
Patch Set: rebase Created 4 years, 9 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 3039c8babb83d5f61e48191a4addfed7c2d00897..e3104591d9ab55847f424d97c8dae5451dc20f7f 100644
--- a/webrtc/call/rtc_event_log_unittest.cc
+++ b/webrtc/call/rtc_event_log_unittest.cc
@@ -308,7 +308,6 @@ size_t GenerateRtpPacket(uint32_t extensions_bitvector,
RTPSender rtp_sender(false, // bool audio
clock, // Clock* clock
nullptr, // Transport*
- nullptr, // RtpAudioFeedback*
nullptr, // PacedSender*
nullptr, // PacketRouter*
nullptr, // SendTimeObserver*
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