Index: webrtc/pc/rtpsenderreceiver_unittest.cc |
diff --git a/webrtc/pc/rtpsenderreceiver_unittest.cc b/webrtc/pc/rtpsenderreceiver_unittest.cc |
index 99380e922089238187b3155f2d49975d5a1ad942..4bcbb51884e1889ed3713c651a165eee2cc6c6c6 100644 |
--- a/webrtc/pc/rtpsenderreceiver_unittest.cc |
+++ b/webrtc/pc/rtpsenderreceiver_unittest.cc |
@@ -304,8 +304,7 @@ TEST_F(RtpSenderReceiverTest, AddAndDestroyVideoRtpReceiver) { |
TEST_F(RtpSenderReceiverTest, LocalAudioSourceOptionsApplied) { |
cricket::AudioOptions options; |
options.echo_cancellation = rtc::Optional<bool>(true); |
- auto source = LocalAudioSource::Create( |
- PeerConnectionFactoryInterface::Options(), &options); |
+ auto source = LocalAudioSource::Create(&options); |
CreateAudioRtpSender(source.get()); |
EXPECT_EQ(rtc::Optional<bool>(true), |