| Index: webrtc/audio/audio_receive_stream_unittest.cc
|
| diff --git a/webrtc/audio/audio_receive_stream_unittest.cc b/webrtc/audio/audio_receive_stream_unittest.cc
|
| index 8e26dd908376374b79ac9ff2fab5df5129b9aeab..aed1d1ad20b61fc5d566ef9fb321203e6200f454 100644
|
| --- a/webrtc/audio/audio_receive_stream_unittest.cc
|
| +++ b/webrtc/audio/audio_receive_stream_unittest.cc
|
| @@ -279,7 +279,7 @@ TEST(AudioReceiveStreamTest, ReceiveRtpPacket) {
|
| EXPECT_CALL(*helper.remote_bitrate_estimator(),
|
| IncomingPacket(packet_time.timestamp / 1000,
|
| rtp_packet.size() - kExpectedHeaderLength,
|
| - VerifyHeaderExtension(expected_extension), false))
|
| + VerifyHeaderExtension(expected_extension)))
|
| .Times(1);
|
| EXPECT_CALL(*helper.channel_proxy(),
|
| ReceivedRTPPacket(&rtp_packet[0],
|
|
|