[rtp_rtcp] rtcp::Empty moved into own file and renamed to CompoundPacket on the way

webrtc/modules/rtp_rtcp/source/rtcp_packet.cc

 /*
  *  Copyright (c) 2014 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 398 matching lines @@
     size_t length,
     uint8_t* buffer,
     size_t* pos) {
   assert(length <= 0xffff);
   const uint8_t kVersion = 2;
   AssignUWord8(buffer, pos, (kVersion << 6) + count_or_format);
   AssignUWord8(buffer, pos, packet_type);
   AssignUWord16(buffer, pos, length);
 }
 
-bool Empty::Create(uint8_t* packet,
-                   size_t* index,
-                   size_t max_length,
-                   RtcpPacket::PacketReadyCallback* callback) const {
-  return true;
-}
-
-size_t Empty::BlockLength() const {
-  return 0;
-}
-
 bool SenderReport::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;
   }
   CreateHeader(sr_.NumberOfReportBlocks, PT_SR, HeaderLength(), packet, index);
   CreateSenderReport(sr_, packet, index);
@@ skipping 236 matching lines @@
   return length_;
 }
 
 void RawPacket::SetLength(size_t length) {
   assert(length <= buffer_length_);
   length_ = length;
 }
 
 }  // namespace rtcp
 }  // namespace webrtc