Refactoring: Use enum for RtpPayloadType.

media/cast/test/utility/default_config.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/test/utility/default_config.h"
 
 #include <stddef.h>
 
 #include "base/bind.h"
 #include "media/cast/cast_config.h"
@@ skipping 16 matching lines @@
 }  // namespace
 
 namespace media {
 namespace cast {
 
 FrameReceiverConfig GetDefaultAudioReceiverConfig() {
   FrameReceiverConfig config;
   config.receiver_ssrc = 2;
   config.sender_ssrc = 1;
   config.rtp_max_delay_ms = kDefaultRtpMaxDelayMs;
-  config.rtp_payload_type = kDefaultRtpAudioPayloadType;
+  config.rtp_payload_type = RtpPayloadType::AUDIO_OPUS;
   config.rtp_timebase = 48000;
   config.channels = 2;
   config.target_frame_rate = 100;  // 10ms of signal per frame
   config.codec = media::cast::CODEC_AUDIO_OPUS;
   return config;
 }
 
 FrameReceiverConfig GetDefaultVideoReceiverConfig() {
   FrameReceiverConfig config;
   config.receiver_ssrc = 12;
   config.sender_ssrc = 11;
   config.rtp_max_delay_ms = kDefaultRtpMaxDelayMs;
-  config.rtp_payload_type = kDefaultRtpVideoPayloadType;
+  config.rtp_payload_type = RtpPayloadType::VIDEO_VP8;
   config.rtp_timebase = kVideoFrequency;
   config.channels = 1;
   config.target_frame_rate = kDefaultMaxFrameRate;
   config.codec = media::cast::CODEC_VIDEO_VP8;
   return config;
 }
 
 AudioSenderConfig GetDefaultAudioSenderConfig() {
   FrameReceiverConfig recv_config = GetDefaultAudioReceiverConfig();
   AudioSenderConfig config;
@@ skipping 33 matching lines @@
 CreateDefaultVideoEncodeAcceleratorCallback() {
   return base::Bind(&CreateVideoEncodeAccelerator);
 }
 
 CreateVideoEncodeMemoryCallback CreateDefaultVideoEncodeMemoryCallback() {
   return base::Bind(&CreateVideoEncodeMemory);
 }
 
 }  // namespace cast
 }  // namespace media