Variable audio bitrate.

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 44 matching lines @@
   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);
   }
   test->OnCallsCreated(sender_call_.get(), receiver_call_.get());
+  receive_transport_.reset(test->CreateReceiveTransport());
   send_transport_.reset(test->CreateSendTransport(sender_call_.get()));
-  receive_transport_.reset(test->CreateReceiveTransport());
 
   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 353 matching lines @@
 
 EndToEndTest::EndToEndTest(unsigned int timeout_ms) : BaseTest(timeout_ms) {
 }
 
 bool EndToEndTest::ShouldCreateReceivers() const {
   return true;
 }
 
 }  // namespace test
 }  // namespace webrtc