Add CreateSend/ReceiveTransport() methods to CallTest.

webrtc/test/call_test.cc

 /*
  *  Copyright (c) 2014 The WebRTC project authors. All Rights Reserved.
  *
  *  Use of this source code is governed by a BSD-style license
  *  that can be found in the LICENSE file in the root of the source
  *  tree. An additional intellectual property rights grant can be found
  *  in the file PATENTS.  All contributing project authors may
  *  be found in the AUTHORS file in the root of the source tree.
  */
 #include "webrtc/base/checks.h"
@@ skipping 21 matching lines @@
       audio_send_stream_(nullptr),
       fake_encoder_(clock_),
       num_video_streams_(0),
       num_audio_streams_(0),
       fake_send_audio_device_(nullptr),
       fake_recv_audio_device_(nullptr) {}
 
 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);
   Call::Config send_config(test->GetSenderCallConfig());
   if (num_audio_streams_ > 0) {
     CreateVoiceEngines();
     AudioState::Config audio_state_config;
     audio_state_config.voice_engine = voe_send_.voice_engine;
     send_config.audio_state = AudioState::Create(audio_state_config);
   }
   CreateSenderCall(send_config);
   if (test->ShouldCreateReceivers()) {
     Call::Config recv_config(test->GetReceiverCallConfig());
     if (num_audio_streams_ > 0) {
       AudioState::Config audio_state_config;
       audio_state_config.voice_engine = voe_recv_.voice_engine;
       recv_config.audio_state = AudioState::Create(audio_state_config);
     }
     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()) {
     send_transport_->SetReceiver(receiver_call_->Receiver());
     receive_transport_->SetReceiver(sender_call_->Receiver());
   } else {
     // Sender-only call delivers to itself.
     send_transport_->SetReceiver(sender_call_->Receiver());
     receive_transport_->SetReceiver(nullptr);
   }
@@ skipping 298 matching lines @@
   return Call::Config();
 }
 
 Call::Config BaseTest::GetReceiverCallConfig() {
   return Call::Config();
 }
 
 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;
 }
 
 size_t BaseTest::GetNumAudioStreams() const {
   return 0;
 }
 
 void BaseTest::ModifyVideoConfigs(
@@ skipping 26 matching lines @@
 
 EndToEndTest::EndToEndTest(unsigned int timeout_ms) : BaseTest(timeout_ms) {
 }
 
 bool EndToEndTest::ShouldCreateReceivers() const {
   return true;
 }
 
 }  // namespace test
 }  // namespace webrtc