Unify rtcp packet setters

webrtc/modules/rtp_rtcp/source/rtcp_packet/sdes.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/rtcp_packet/sdes.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 {
 // Source Description (SDES) (RFC 3550).
 //
 //         0                   1                   2                   3
@@ skipping 107 matching lines @@
     }
     // Adjust to 32bit boundary.
     looking_at += (payload_end - looking_at) % 4;
   }
 
   chunks_ = std::move(chunks);
   block_length_ = block_length;
   return true;
 }
 
-bool Sdes::WithCName(uint32_t ssrc, const std::string& cname) {
+bool Sdes::AddCName(uint32_t ssrc, std::string cname) {
   RTC_DCHECK_LE(cname.length(), 0xffu);
   if (chunks_.size() >= kMaxNumberOfChunks) {
     LOG(LS_WARNING) << "Max SDES chunks reached.";
     return false;
   }
   Chunk chunk;
   chunk.ssrc = ssrc;
-  chunk.cname = cname;
+  chunk.cname = std::move(cname);
   chunks_.push_back(chunk);
   block_length_ += ChunkSize(chunk);
   return true;
 }
 
 bool Sdes::Create(uint8_t* packet,
                   size_t* index,
                   size_t max_length,
                   RtcpPacket::PacketReadyCallback* callback) const {
   while (*index + BlockLength() > max_length) {
@@ skipping 18 matching lines @@
     const int kPadding = 0;
     memset(packet + *index, kPadding, padding_size);
     *index += padding_size;
   }
 
   RTC_CHECK_EQ(*index, index_end);
   return true;
 }
 }  // namespace rtcp
 }  // namespace webrtc