Implement PlayoutDelay extension as a trait

webrtc/modules/rtp_rtcp/source/rtp_header_extensions.h

 /*
  *  Copyright (c) 2016 The WebRTC project authors. All Rights Reserved.
  *
  *  Use of this source code is governed by a BSD-style license
  *  that can be found in the LICENSE file in the root of the source
  *  tree. An additional intellectual property rights grant can be found
  *  in the file PATENTS.  All contributing project authors may
  *  be found in the AUTHORS file in the root of the source tree.
  */
 #ifndef WEBRTC_MODULES_RTP_RTCP_SOURCE_RTP_HEADER_EXTENSIONS_H_
@@ skipping 53 matching lines @@
   static constexpr RTPExtensionType kId = kRtpExtensionVideoRotation;
   static constexpr uint8_t kValueSizeBytes = 1;
   static const char* kName;
   static bool IsSupportedFor(MediaType type);
   static bool Parse(const uint8_t* data, VideoRotation* value);
   static bool Write(uint8_t* data, VideoRotation value);
   static bool Parse(const uint8_t* data, uint8_t* value);
   static bool Write(uint8_t* data, uint8_t value);
 };
 
+class PlayoutDelayLimits {
+ public:
+  static constexpr RTPExtensionType kId = kRtpExtensionPlayoutDelay;
+  static constexpr uint8_t kValueSizeBytes = 3;
+  static const char* kName;
+  static bool IsSupportedFor(MediaType type);
+  // Playout delay in milliseconds. A playout delay limit (min or max)
+  // has 12 bits allocated. This allows a range of 0-4095 values which
+  // translates to a range of 0-40950 in milliseconds.
+  static constexpr int kGranularityMs = 10;
+  // Maximum playout delay value in milliseconds.
+  static constexpr int kMaxMs = 0xfff * kGranularityMs;  // 40950.
+
+  static bool Parse(const uint8_t* data, int* min_ms, int* max_ms);

[Irfan, 2016/08/16 17:22:14]

Parse it into PlayoutDelay variable ?

[danilchap, 2016/08/17 07:42:38]

Done.

+  static bool Write(uint8_t* data, int min_ms, int max_ms);

[Irfan, 2016/08/16 17:22:13]

Similarly, Write a PlayoutDelay variable ?

[danilchap, 2016/08/17 07:42:38]

Done.

+};
+
 }  // namespace webrtc
 #endif  // WEBRTC_MODULES_RTP_RTCP_SOURCE_RTP_HEADER_EXTENSIONS_H_