Add sender controlled playout delay limits

webrtc/modules/rtp_rtcp/include/rtp_rtcp_defines.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.
  */
 
@@ skipping 49 matching lines @@
   kAllowRetransmission
 };
 
 enum RTPExtensionType {
   kRtpExtensionNone,
   kRtpExtensionTransmissionTimeOffset,
   kRtpExtensionAudioLevel,
   kRtpExtensionAbsoluteSendTime,
   kRtpExtensionVideoRotation,
   kRtpExtensionTransportSequenceNumber,
+  kRtpExtensionPlayoutDelay,
 };
 
 enum RTCPAppSubTypes { kAppSubtypeBwe = 0x00 };
 
 // TODO(sprang): Make this an enum class once rtcp_receiver has been cleaned up.
 enum RTCPPacketType : uint32_t {
   kRtcpReport = 0x0001,
   kRtcpSr = 0x0002,
   kRtcpRr = 0x0004,
   kRtcpSdes = 0x0008,
@@ skipping 304 matching lines @@
 class TransportSequenceNumberAllocator {
  public:
   TransportSequenceNumberAllocator() {}
   virtual ~TransportSequenceNumberAllocator() {}
 
   virtual uint16_t AllocateSequenceNumber() = 0;
 };
 
 }  // namespace webrtc
 #endif  // WEBRTC_MODULES_RTP_RTCP_INCLUDE_RTP_RTCP_DEFINES_H_