| 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_;
|
| };
|
|
|