Merge min_ms and max_ms accessors in PlayoutDelayOracle

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"
 
 namespace webrtc {
 
 PlayoutDelayOracle::PlayoutDelayOracle()
     : high_sequence_number_(0),
       send_playout_delay_(false),
       ssrc_(0),
-      min_playout_delay_ms_(-1),
-      max_playout_delay_ms_(-1) {}
+      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, playout_delay.max_ms);
   int64_t unwrapped_seq_num = unwrapper_.Unwrap(seq_num);
   if (playout_delay.min_ms >= 0 &&
-      playout_delay.min_ms != min_playout_delay_ms_) {
+      playout_delay.min_ms != playout_delay_.min_ms) {
     send_playout_delay_ = true;
-    min_playout_delay_ms_ = playout_delay.min_ms;
+    playout_delay_.min_ms = playout_delay.min_ms;
     high_sequence_number_ = unwrapped_seq_num;
   }
 
   if (playout_delay.max_ms >= 0 &&
-      playout_delay.max_ms != max_playout_delay_ms_) {
+      playout_delay.max_ms != playout_delay_.max_ms) {
     send_playout_delay_ = true;
-    max_playout_delay_ms_ = playout_delay.max_ms;
+    playout_delay_.max_ms = playout_delay.max_ms;
     high_sequence_number_ = unwrapped_seq_num;
   }
   ssrc_ = ssrc;
 }
 
 // If an ACK is received on the packet containing the playout delay extension,
 // we stop sending the extension on future packets.
 void PlayoutDelayOracle::OnReceivedRtcpReportBlocks(
     const ReportBlockList& report_blocks) {
   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