Cast: Make receiver use cast_transport

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 <stdint.h>
 
 #include "base/bind.h"
 #include "base/bind_helpers.h"
 #include "base/memory/scoped_ptr.h"
 #include "base/test/simple_test_tick_clock.h"
@@ skipping 59 matching lines @@
     audio_config_.frequency = kDefaultAudioSamplingRate;
     audio_config_.channels = 2;
     audio_config_.bitrate = kDefaultAudioEncoderBitrate;
     audio_config_.rtp_payload_type = 127;
 
     net::IPEndPoint dummy_endpoint;
 
     transport_sender_.reset(new CastTransportSenderImpl(
         NULL,
         testing_clock_,
+        net::IPEndPoint(),
         dummy_endpoint,
         make_scoped_ptr(new base::DictionaryValue),
         base::Bind(&UpdateCastTransportStatus),
         BulkRawEventsCallback(),
         base::TimeDelta(),
         task_runner_,
+        PacketReceiverCallback(),
         &transport_));
     audio_sender_.reset(new AudioSender(
         cast_environment_, audio_config_, transport_sender_.get()));
     task_runner_->RunTasks();
   }
 
   ~AudioSenderTest() override {}
 
   static void UpdateCastTransportStatus(CastTransportStatus status) {
     EXPECT_EQ(TRANSPORT_AUDIO_INITIALIZED, status);
@@ skipping 37 matching lines @@
   base::TimeDelta max_rtcp_timeout =
       base::TimeDelta::FromMilliseconds(1 + kDefaultRtcpIntervalMs * 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