Index: webrtc/modules/rtp_rtcp/test/testAPI/test_api_audio.cc |
diff --git a/webrtc/modules/rtp_rtcp/test/testAPI/test_api_audio.cc b/webrtc/modules/rtp_rtcp/test/testAPI/test_api_audio.cc |
index 9b44c4f40dbe934d9b229a24f49f64c0f8e12478..11b592afccca9efa4eca4cab4ebe39f79f019dbf 100644 |
--- a/webrtc/modules/rtp_rtcp/test/testAPI/test_api_audio.cc |
+++ b/webrtc/modules/rtp_rtcp/test/testAPI/test_api_audio.cc |
@@ -86,6 +86,7 @@ |
~RtpRtcpAudioTest() {} |
void SetUp() override { |
+ audioFeedback = new NullRtpAudioFeedback(); |
data_receiver1 = new VerifyingAudioReceiver(); |
data_receiver2 = new VerifyingAudioReceiver(); |
rtp_callback = new RTPCallback(); |
@@ -108,14 +109,16 @@ |
module1 = RtpRtcp::CreateRtpRtcp(configuration); |
rtp_receiver1_.reset(RtpReceiver::CreateAudioReceiver( |
- &fake_clock, data_receiver1, NULL, rtp_payload_registry1_.get())); |
+ &fake_clock, audioFeedback, data_receiver1, NULL, |
+ rtp_payload_registry1_.get())); |
configuration.receive_statistics = receive_statistics2_.get(); |
configuration.outgoing_transport = transport2; |
module2 = RtpRtcp::CreateRtpRtcp(configuration); |
rtp_receiver2_.reset(RtpReceiver::CreateAudioReceiver( |
- &fake_clock, data_receiver2, NULL, rtp_payload_registry2_.get())); |
+ &fake_clock, audioFeedback, data_receiver2, NULL, |
+ rtp_payload_registry2_.get())); |
transport1->SetSendModule(module2, rtp_payload_registry2_.get(), |
rtp_receiver2_.get(), receive_statistics2_.get()); |
@@ -128,6 +131,7 @@ |
delete module2; |
delete transport1; |
delete transport2; |
+ delete audioFeedback; |
delete data_receiver1; |
delete data_receiver2; |
delete rtp_callback; |
@@ -145,6 +149,7 @@ |
VerifyingAudioReceiver* data_receiver2; |
LoopBackTransport* transport1; |
LoopBackTransport* transport2; |
+ NullRtpAudioFeedback* audioFeedback; |
RTPCallback* rtp_callback; |
uint32_t test_ssrc; |
uint32_t test_timestamp; |