Unify rtcp packet setters

webrtc/modules/rtp_rtcp/source/rtcp_packet/extended_jitter_report.cc

 /*
  *  Copyright (c) 2015 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/rtcp_packet/extended_jitter_report.h"
 
+#include <utility>
+
 #include "webrtc/base/checks.h"
 #include "webrtc/base/logging.h"
 #include "webrtc/modules/rtp_rtcp/source/byte_io.h"
 #include "webrtc/modules/rtp_rtcp/source/rtcp_packet/common_header.h"
 
 namespace webrtc {
 namespace rtcp {
 constexpr uint8_t ExtendedJitterReport::kPacketType;
 // Transmission Time Offsets in RTP Streams (RFC 5450).
 //
@@ skipping 26 matching lines @@
 
   inter_arrival_jitters_.resize(number_of_jitters);
   for (size_t index = 0; index < number_of_jitters; ++index) {
     inter_arrival_jitters_[index] = ByteReader<uint32_t>::ReadBigEndian(
         &packet.payload()[index * kJitterSizeBytes]);
   }
 
   return true;
 }
 
-bool ExtendedJitterReport::WithJitter(uint32_t jitter) {
-  if (inter_arrival_jitters_.size() >= kMaxNumberOfJitters) {
-    LOG(LS_WARNING) << "Max inter-arrival jitter items reached.";
+bool ExtendedJitterReport::SetJitterValues(std::vector<uint32_t> values) {
+  if (values.size() > kMaxNumberOfJitterValues) {
+    LOG(LS_WARNING) << "Too many inter-arrival jitter items.";
     return false;
   }
-  inter_arrival_jitters_.push_back(jitter);
+  inter_arrival_jitters_ = std::move(values);
   return true;
 }
 
 bool ExtendedJitterReport::Create(
     uint8_t* packet,
     size_t* index,
     size_t max_length,
     RtcpPacket::PacketReadyCallback* callback) const {
   while (*index + BlockLength() > max_length) {
     if (!OnBufferFull(packet, index, callback))
       return false;
   }
   const size_t index_end = *index + BlockLength();
   size_t length = inter_arrival_jitters_.size();
   CreateHeader(length, kPacketType, length, packet, index);
 
   for (uint32_t jitter : inter_arrival_jitters_) {
     ByteWriter<uint32_t>::WriteBigEndian(packet + *index, jitter);
     *index += kJitterSizeBytes;
   }
   // Sanity check.
   RTC_DCHECK_EQ(index_end, *index);
   return true;
 }
 
 }  // namespace rtcp
 }  // namespace webrtc