Index: webrtc/audio/test/low_bandwidth_audio_test.cc |
diff --git a/webrtc/audio/test/low_bandwidth_audio_test.cc b/webrtc/audio/test/low_bandwidth_audio_test.cc |
index c1b1a02235610b51cfa708aa934eb6183ccc3275..50d7e674e83d11833f43808e6572cc86c2df842b 100644 |
--- a/webrtc/audio/test/low_bandwidth_audio_test.cc |
+++ b/webrtc/audio/test/low_bandwidth_audio_test.cc |
@@ -80,12 +80,14 @@ test::PacketTransport* AudioQualityTest::CreateSendTransport( |
Call* sender_call) { |
return new test::PacketTransport( |
sender_call, this, test::PacketTransport::kSender, |
+ MediaType::AUDIO, |
GetNetworkPipeConfig()); |
} |
test::PacketTransport* AudioQualityTest::CreateReceiveTransport() { |
return new test::PacketTransport(nullptr, this, |
- test::PacketTransport::kReceiver, GetNetworkPipeConfig()); |
+ test::PacketTransport::kReceiver, MediaType::AUDIO, |
+ GetNetworkPipeConfig()); |
} |
void AudioQualityTest::ModifyAudioConfigs( |