Index: webrtc/test/call_test.cc |
diff --git a/webrtc/test/call_test.cc b/webrtc/test/call_test.cc |
index 83fd844db90bba2fcef6de84e98d67e605985cc6..850e487caf9d5f10219ad1695dda7fe8e74acad0 100644 |
--- a/webrtc/test/call_test.cc |
+++ b/webrtc/test/call_test.cc |
@@ -39,8 +39,7 @@ CallTest::CallTest() |
CallTest::~CallTest() { |
} |
-void CallTest::RunBaseTest(BaseTest* test, |
- const FakeNetworkPipe::Config& config) { |
+void CallTest::RunBaseTest(BaseTest* test) { |
num_video_streams_ = test->GetNumVideoStreams(); |
num_audio_streams_ = test->GetNumAudioStreams(); |
RTC_DCHECK(num_video_streams_ > 0 || num_audio_streams_ > 0); |
@@ -61,11 +60,8 @@ void CallTest::RunBaseTest(BaseTest* test, |
} |
CreateReceiverCall(recv_config); |
} |
- send_transport_.reset(new PacketTransport( |
- sender_call_.get(), test, test::PacketTransport::kSender, config)); |
- receive_transport_.reset(new PacketTransport( |
- nullptr, test, test::PacketTransport::kReceiver, config)); |
- test->OnTransportsCreated(send_transport_.get(), receive_transport_.get()); |
+ send_transport_.reset(test->CreateSendTransport(sender_call_.get())); |
+ receive_transport_.reset(test->CreateReceiveTransport()); |
test->OnCallsCreated(sender_call_.get(), receiver_call_.get()); |
if (test->ShouldCreateReceivers()) { |
@@ -384,8 +380,15 @@ Call::Config BaseTest::GetReceiverCallConfig() { |
void BaseTest::OnCallsCreated(Call* sender_call, Call* receiver_call) { |
} |
-void BaseTest::OnTransportsCreated(PacketTransport* send_transport, |
- PacketTransport* receive_transport) {} |
+test::PacketTransport* BaseTest::CreateSendTransport(Call* sender_call) { |
+ return new PacketTransport(sender_call, this, test::PacketTransport::kSender, |
+ FakeNetworkPipe::Config()); |
+} |
+ |
+test::PacketTransport* BaseTest::CreateReceiveTransport() { |
+ return new PacketTransport(nullptr, this, test::PacketTransport::kReceiver, |
+ FakeNetworkPipe::Config()); |
+} |
size_t BaseTest::GetNumVideoStreams() const { |
return 1; |