Refactoring: CastTransport InitializeAudio/InitializeVideo.

media/cast/sender/audio_sender_unittest.cc

 // Copyright 2014 The Chromium Authors. All rights reserved.
 // Use of this source code is governed by a BSD-style license that can be
 // found in the LICENSE file.
 
 #include "media/cast/sender/audio_sender.h"
 
 #include <stdint.h>
 
 #include <memory>
 #include <utility>
@@ skipping 24 matching lines @@
   DVLOG(1) << "OperationalStatus transitioning from " << *out_status << " to "
            << in_status;
   *out_status = in_status;
 }
 
 class TransportClient : public CastTransport::Client {
  public:
   TransportClient() {}
 
   void OnStatusChanged(CastTransportStatus status) final {
-    EXPECT_EQ(TRANSPORT_AUDIO_INITIALIZED, status);
+    EXPECT_EQ(TRANSPORT_STREAM_INITIALIZED, status);
   };
   void OnLoggingEventsReceived(
       std::unique_ptr<std::vector<FrameEvent>> frame_events,
       std::unique_ptr<std::vector<PacketEvent>> packet_events) final{};
   void ProcessRtpPacket(std::unique_ptr<Packet> packet) final{};
 
   DISALLOW_COPY_AND_ASSIGN(TransportClient);
 };
 
 }  // namespace
@@ skipping 46 matching lines @@
         new CastEnvironment(std::unique_ptr<base::TickClock>(testing_clock_),
                             task_runner_, task_runner_, task_runner_);
     audio_config_.codec = CODEC_AUDIO_OPUS;
     audio_config_.use_external_encoder = false;
     audio_config_.rtp_timebase = kDefaultAudioSamplingRate;
     audio_config_.channels = 2;
     audio_config_.max_bitrate = kDefaultAudioEncoderBitrate;
     audio_config_.rtp_payload_type = RtpPayloadType::AUDIO_OPUS;
 
     transport_ = new TestPacketSender();
-    transport_sender_.reset(
-        new CastTransportImpl(testing_clock_, base::TimeDelta(),
-                              base::WrapUnique(new TransportClient()),
-                              base::WrapUnique(transport_), task_runner_));
+    transport_sender_.reset(new CastTransportImpl(
+        testing_clock_, base::TimeDelta(), base::MakeUnique<TransportClient>(),
+        base::WrapUnique(transport_), task_runner_));
     OperationalStatus operational_status = STATUS_UNINITIALIZED;
     audio_sender_.reset(new AudioSender(
         cast_environment_,
         audio_config_,
         base::Bind(&SaveOperationalStatus, &operational_status),
         transport_sender_.get()));
     task_runner_->RunTasks();
     CHECK_EQ(STATUS_INITIALIZED, operational_status);
   }
 
@@ skipping 35 matching lines @@
   base::TimeDelta max_rtcp_timeout =
       base::TimeDelta::FromMilliseconds(1 + kRtcpReportIntervalMs * 3 / 2);
   testing_clock_->Advance(max_rtcp_timeout);
   task_runner_->RunTasks();
   EXPECT_LE(1, transport_->number_of_rtp_packets());
   EXPECT_LE(1, transport_->number_of_rtcp_packets());
 }
 
 }  // namespace cast
 }  // namespace media