Refactoring: Use enum for RtpPayloadType.

media/cast/net/cast_transport_config.h

 // 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.
 
 #ifndef MEDIA_CAST_NET_CAST_TRANSPORT_CONFIG_H_
 #define MEDIA_CAST_NET_CAST_TRANSPORT_CONFIG_H_
 
 #include <stddef.h>
 #include <stdint.h>
 
@@ skipping 12 matching lines @@
   CODEC_UNKNOWN,
   CODEC_AUDIO_OPUS,
   CODEC_AUDIO_PCM16,
   CODEC_AUDIO_AAC,
   CODEC_VIDEO_FAKE,
   CODEC_VIDEO_VP8,
   CODEC_VIDEO_H264,
   CODEC_LAST = CODEC_VIDEO_H264
 };
 
+// Describes the content being transported over RTP streams.
+enum RtpPayloadType {

[miu, 2016/06/20 21:00:27]

nit style suggestion: Use an enum class:

  enum class RtpPayloadType : int {
    UNKNOWN,
    AUDIO_OPUS = 96,
    AUDIO_AAC,
    ...
  };

[xjz, 2016/06/21 17:24:31]

Done.

+  RTP_PAYLOAD_UNKNOWN,
+
+  // Cast Streaming will encode raw audio frames using one of its available
+  // codec implementations, and transport encoded data in the RTP stream.
+  RTP_PAYLOAD_AUDIO_OPUS = 96,

[miu, 2016/06/20 21:00:27]

If we add new ones in the future, the numbering might become messed up (if any
code depends on it). You might just assign an explicit value to each of these
(97 for AAC, 98 for PCM16, and so on).

[xjz, 2016/06/21 17:24:31]

Done.

+  RTP_PAYLOAD_AUDIO_AAC,
+  RTP_PAYLOAD_AUDIO_PCM16,
+
+  // Audio frame data is not modified, and should be transported reliably and
+  // in-sequence. No assumptions about the data can be made.
+  RTP_PAYLOAD_REMOTE_AUDIO,
+
+  RTP_PAYLOAD_AUDIO_LAST = RTP_PAYLOAD_REMOTE_AUDIO,
+
+  // Cast Streaming will encode raw video frames using one of its available
+  // codec implementations, and transport encoded data in the RTP stream.
+  RTP_PAYLOAD_VIDEO_VP8,
+  RTP_PAYLOAD_VIDEO_H264,
+
+  // Video frame data is not modified, and should be transported reliably and
+  // in-sequence. No assumptions about the data can be made.
+  RTP_PAYLOAD_REMOTE_VIDEO,
+
+  RTP_PAYLOAD_VIDEO_LAST = RTP_PAYLOAD_REMOTE_VIDEO
+};
+
 struct CastTransportRtpConfig {
   CastTransportRtpConfig();
   ~CastTransportRtpConfig();
 
   // Identifier refering to this sender.
   uint32_t ssrc;
 
   // Identifier for incoming RTCP traffic.
   uint32_t feedback_ssrc;
 
   // RTP payload type enum: Specifies the type/encoding of frame data.
-  int rtp_payload_type;
+  RtpPayloadType rtp_payload_type;
 
   // The AES crypto key and initialization vector.  Each of these strings
   // contains the data in binary form, of size kAesKeySize.  If they are empty
   // strings, crypto is not being used.
   std::string aes_key;
   std::string aes_iv_mask;
 };
 
 // A combination of metadata and data for one encoded frame.  This can contain
 // audio data or video data or other.
@@ skipping 132 matching lines @@
          lhs.ntp_fraction == rhs.ntp_fraction &&
          lhs.rtp_timestamp == rhs.rtp_timestamp &&
          lhs.send_packet_count == rhs.send_packet_count &&
          lhs.send_octet_count == rhs.send_octet_count;
 }
 
 }  // namespace cast
 }  // namespace media
 
 #endif  // MEDIA_CAST_NET_CAST_TRANSPORT_CONFIG_H_