Add sender controlled playout delay limits

webrtc/modules/rtp_rtcp/source/rtp_header_extension.h

 /*
  *  Copyright (c) 2012 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_EXTENSION_H_
 #define WEBRTC_MODULES_RTP_RTCP_SOURCE_RTP_HEADER_EXTENSION_H_
 
 #include <map>
 
 #include "webrtc/modules/rtp_rtcp/include/rtp_rtcp_defines.h"
 #include "webrtc/typedefs.h"
 
 namespace webrtc {
 
 const uint16_t kRtpOneByteHeaderExtensionId = 0xBEDE;
 
 const size_t kRtpOneByteHeaderLength = 4;
 const size_t kTransmissionTimeOffsetLength = 4;
 const size_t kAudioLevelLength = 2;
 const size_t kAbsoluteSendTimeLength = 4;
 const size_t kVideoRotationLength = 2;
 const size_t kTransportSequenceNumberLength = 3;
+const size_t kPlayoutDelayLength = 4;
+
+// 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.
+const int kPlayoutDelayGranularityMs = 10;
+// Maximum playout delay value in milliseconds.
+const int kPlayoutDelayMaxMs = 40950;
 
 struct HeaderExtension {
   explicit HeaderExtension(RTPExtensionType extension_type)
       : type(extension_type), length(0), active(true) {
     Init();
   }
 
   HeaderExtension(RTPExtensionType extension_type, bool active)
       : type(extension_type), length(0), active(active) {
     Init();
@@ skipping 12 matching lines @@
         break;
       case kRtpExtensionAbsoluteSendTime:
         length = kAbsoluteSendTimeLength;
         break;
       case kRtpExtensionVideoRotation:
         length = kVideoRotationLength;
         break;
       case kRtpExtensionTransportSequenceNumber:
         length = kTransportSequenceNumberLength;
         break;
+      case kRtpExtensionPlayoutDelay:
+        length = kPlayoutDelayLength;
+        break;
       default:
         assert(false);
     }
   }
 
   const RTPExtensionType type;
   uint8_t length;
   bool active;
 };
 
@@ skipping 44 matching lines @@
   RTPExtensionType Next(RTPExtensionType type) const;
 
  private:
   int32_t Register(const RTPExtensionType type, const uint8_t id, bool active);
   std::map<uint8_t, HeaderExtension*> extensionMap_;
 };
 }  // namespace webrtc
 
 #endif  // WEBRTC_MODULES_RTP_RTCP_SOURCE_RTP_HEADER_EXTENSION_H_