| 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();
|
|
|
|
|