Index: webrtc/modules/rtp_rtcp/source/rtp_receiver_unittest.cc |
diff --git a/webrtc/modules/rtp_rtcp/source/rtp_receiver_unittest.cc b/webrtc/modules/rtp_rtcp/source/rtp_receiver_unittest.cc |
index a68f06e6848dcd3c3ad5433ad74aceeeb2d788ab..363659efe7f256317b7ebeec4abffff7ca0f5929 100644 |
--- a/webrtc/modules/rtp_rtcp/source/rtp_receiver_unittest.cc |
+++ b/webrtc/modules/rtp_rtcp/source/rtp_receiver_unittest.cc |
@@ -46,7 +46,7 @@ class RtpReceiverTest : public ::testing::Test { |
: fake_clock_(123456), |
rtp_receiver_( |
RtpReceiver::CreateAudioReceiver(&fake_clock_, |
- nullptr, |
+ &null_rtp_data_, |
nullptr, |
&rtp_payload_registry_)) { |
CodecInst voice_codec = {}; |
@@ -73,6 +73,7 @@ class RtpReceiverTest : public ::testing::Test { |
} |
SimulatedClock fake_clock_; |
+ NullRtpData null_rtp_data_; |
RTPPayloadRegistry rtp_payload_registry_; |
std::unique_ptr<RtpReceiver> rtp_receiver_; |
}; |