Refactoring: Use enum for RtpPayloadType.

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 88 matching lines @@
     testing_clock_->Advance(base::TimeTicks::Now() - base::TimeTicks());
     task_runner_ = new FakeSingleThreadTaskRunner(testing_clock_);
     cast_environment_ =
         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_.frequency = kDefaultAudioSamplingRate;
     audio_config_.channels = 2;
     audio_config_.bitrate = kDefaultAudioEncoderBitrate;
-    audio_config_.rtp_payload_type = 127;
+    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_));
     OperationalStatus operational_status = STATUS_UNINITIALIZED;
     audio_sender_.reset(new AudioSender(
         cast_environment_,
         audio_config_,
@@ skipping 41 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