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Unified Diff: webrtc/audio/audio_send_stream_unittest.cc

Issue 1628683002: Use separate rtp module lists for send and receive in PacketRouter. (Closed) Base URL: https://chromium.googlesource.com/external/webrtc.git@master
Patch Set: Comments addressed. Created 4 years, 11 months ago
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Index: webrtc/audio/audio_send_stream_unittest.cc
diff --git a/webrtc/audio/audio_send_stream_unittest.cc b/webrtc/audio/audio_send_stream_unittest.cc
index 466c1571acd0af6091abfded1fb04ab0c8a25b9b..a01ef025d3be84a85f30aa5cec0d225b7608bc89 100644
--- a/webrtc/audio/audio_send_stream_unittest.cc
+++ b/webrtc/audio/audio_send_stream_unittest.cc
@@ -83,13 +83,12 @@ struct ConfigHelper {
kTransportSequenceNumberId))
.Times(1);
EXPECT_CALL(*channel_proxy_,
- SetCongestionControlObjects(
+ RegisterSenderCongestionControlObjects(
congestion_controller_.pacer(),
congestion_controller_.GetTransportFeedbackObserver(),
congestion_controller_.packet_router()))
.Times(1);
- EXPECT_CALL(*channel_proxy_,
- SetCongestionControlObjects(nullptr, nullptr, nullptr))
+ EXPECT_CALL(*channel_proxy_, ResetCongestionControlObjects())
.Times(1);
return channel_proxy_;
}));
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