Use SingleThreadedTaskQueue in DirectTransport

webrtc/audio/test/low_bandwidth_audio_test.cc

 /*
  *  Copyright (c) 2017 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.
  */
 
@@ skipping 68 matching lines @@
     test::FakeAudioDevice* send_audio_device,
     test::FakeAudioDevice* recv_audio_device) {
   send_audio_device_ = send_audio_device;
 }
 
 FakeNetworkPipe::Config AudioQualityTest::GetNetworkPipeConfig() {
   return FakeNetworkPipe::Config();
 }
 
 test::PacketTransport* AudioQualityTest::CreateSendTransport(
+    SingleThreadedTaskQueueForTesting* task_queue,
     Call* sender_call) {
   return new test::PacketTransport(
-      sender_call, this, test::PacketTransport::kSender,
+      task_queue, sender_call, this, test::PacketTransport::kSender,
       test::CallTest::payload_type_map_, GetNetworkPipeConfig());
 }
 
-test::PacketTransport* AudioQualityTest::CreateReceiveTransport() {
+test::PacketTransport* AudioQualityTest::CreateReceiveTransport(
+    SingleThreadedTaskQueueForTesting* task_queue) {
   return new test::PacketTransport(
-      nullptr, this, test::PacketTransport::kReceiver,
+      task_queue, nullptr, this, test::PacketTransport::kReceiver,
       test::CallTest::payload_type_map_, GetNetworkPipeConfig());
 }
 
 void AudioQualityTest::ModifyAudioConfigs(
   AudioSendStream::Config* send_config,
   std::vector<AudioReceiveStream::Config>* receive_configs) {
   // Large bitrate by default.
   const webrtc::SdpAudioFormat kDefaultFormat("OPUS", 48000, 2,
                                               {{"stereo", "1"}});
   send_config->send_codec_spec =
@@ skipping 55 matching lines @@
   }
 };
 
 TEST_F(LowBandwidthAudioTest, Mobile2GNetwork) {
   Mobile2GNetworkTest test;
   RunBaseTest(&test);
 }
 
 }  // namespace test
 }  // namespace webrtc