Merge min_ms and max_ms accessors in PlayoutDelayOracle

webrtc/modules/rtp_rtcp/source/rtp_sender.cc

 /*
  *  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 463 matching lines @@
 
     if (rtp_header) {
       playout_delay_oracle_.UpdateRequest(ssrc, rtp_header->playout_delay,
                                           sequence_number);
     }
 
     // Update the active/inactive status of playout delay extension based
     // on what the oracle indicates.
     {
       rtc::CritScope lock(&send_critsect_);
-      if (playout_delay_active_ != playout_delay_oracle_.send_playout_delay()) {
-        playout_delay_active_ = playout_delay_oracle_.send_playout_delay();
+      bool send_playout_delay = playout_delay_oracle_.send_playout_delay();
+      if (playout_delay_active_ != send_playout_delay) {
+        playout_delay_active_ = send_playout_delay;
         rtp_header_extension_map_.SetActive(kRtpExtensionPlayoutDelay,
                                             playout_delay_active_);
       }
     }
 
     result = video_->SendVideo(video_type, frame_type, payload_type,
                                rtp_timestamp, capture_time_ms, payload_data,
                                payload_size, fragmentation, rtp_header);
   }
 
@@ skipping 674 matching lines @@
       case kRtpExtensionAbsoluteSendTime:
         block_length = BuildAbsoluteSendTimeExtension(extension_data);
         break;
       case kRtpExtensionVideoRotation:
         block_length = BuildVideoRotationExtension(extension_data);
         break;
       case kRtpExtensionTransportSequenceNumber:
         block_length = BuildTransportSequenceNumberExtension(
             extension_data, transport_sequence_number_);
         break;
-      case kRtpExtensionPlayoutDelay:
+      case kRtpExtensionPlayoutDelay: {
+        PlayoutDelay playout_delay = playout_delay_oracle_.playout_delay();
         block_length = BuildPlayoutDelayExtension(
-            extension_data, playout_delay_oracle_.min_playout_delay_ms(),
-            playout_delay_oracle_.max_playout_delay_ms());
+            extension_data, playout_delay.min_ms, playout_delay.max_ms);
         break;
+      }
       default:
         assert(false);
     }
     total_block_length += block_length;
     type = rtp_header_extension_map_.Next(type);
   }
   if (total_block_length == 0) {
     // No extension added.
     return 0;
   }
@@ skipping 556 matching lines @@
   rtc::CritScope lock(&send_critsect_);
 
   RtpState state;
   state.sequence_number = sequence_number_rtx_;
   state.start_timestamp = timestamp_offset_;
 
   return state;
 }
 
 }  // namespace webrtc