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], |