| 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),
|
|
|