Move implmentation specific constants out of rtp_header_extension.h

webrtc/modules/rtp_rtcp/source/playout_delay_oracle.cc

 /*
  *  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.
  */
 
 #include "webrtc/modules/rtp_rtcp/source/playout_delay_oracle.h"
 
 #include "webrtc/base/checks.h"
 #include "webrtc/base/logging.h"
 #include "webrtc/modules/rtp_rtcp/include/rtp_rtcp_defines.h"
-#include "webrtc/modules/rtp_rtcp/source/rtp_header_extension.h"
+#include "webrtc/modules/rtp_rtcp/source/rtp_header_extensions.h"
 
 namespace webrtc {
 
 PlayoutDelayOracle::PlayoutDelayOracle()
     : high_sequence_number_(0),
       send_playout_delay_(false),
       ssrc_(0),
       playout_delay_{-1, -1} {}
 
 PlayoutDelayOracle::~PlayoutDelayOracle() {}
 
 void PlayoutDelayOracle::UpdateRequest(uint32_t ssrc,
                                        PlayoutDelay playout_delay,
                                        uint16_t seq_num) {
   rtc::CritScope lock(&crit_sect_);
-  RTC_DCHECK_LE(playout_delay.min_ms, kPlayoutDelayMaxMs);
-  RTC_DCHECK_LE(playout_delay.max_ms, kPlayoutDelayMaxMs);
+  RTC_DCHECK_LE(playout_delay.min_ms, PlayoutDelayLimits::kMaxMs);
+  RTC_DCHECK_LE(playout_delay.max_ms, PlayoutDelayLimits::kMaxMs);
   RTC_DCHECK_LE(playout_delay.min_ms, playout_delay.max_ms);
   int64_t unwrapped_seq_num = unwrapper_.Unwrap(seq_num);
   if (playout_delay.min_ms >= 0 &&
       playout_delay.min_ms != playout_delay_.min_ms) {
     send_playout_delay_ = true;
     playout_delay_.min_ms = playout_delay.min_ms;
     high_sequence_number_ = unwrapped_seq_num;
   }
 
   if (playout_delay.max_ms >= 0 &&
@@ skipping 12 matching lines @@
   rtc::CritScope lock(&crit_sect_);
   for (const RTCPReportBlock& report_block : report_blocks) {
     if ((ssrc_ == report_block.sourceSSRC) && send_playout_delay_ &&
         (report_block.extendedHighSeqNum > high_sequence_number_)) {
       send_playout_delay_ = false;
     }
   }
 }
 
 }  // namespace webrtc