Index: webrtc/test/call_test.h |
diff --git a/webrtc/test/call_test.h b/webrtc/test/call_test.h |
index b4f665821ee2ffa420a6465835a4dc310e8b3961..b96e5e6e112a7e831c08f15ea8b5eb2b87d95259 100644 |
--- a/webrtc/test/call_test.h |
+++ b/webrtc/test/call_test.h |
@@ -192,7 +192,10 @@ class BaseTest : public RtpRtcpObserver { |
virtual Call::Config GetReceiverCallConfig(); |
virtual void OnCallsCreated(Call* sender_call, Call* receiver_call); |
- // The default implementation creates MediaType::VIDEO transports. |
+ // Returns VIDEO for video-only tests, AUDIO for audio-only tests, |
+ // and ANY for tests sending audio and video over the same |
+ // transport. |
+ virtual MediaType SelectMediaType(); |
virtual test::PacketTransport* CreateSendTransport(Call* sender_call); |
virtual test::PacketTransport* CreateReceiveTransport(); |