NetEq: Change member variables for current RTP types to rtc::Optionals

webrtc/modules/audio_coding/neteq/mock/mock_packet_buffer.h

 /*
  *  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 14 matching lines @@
   MOCK_METHOD0(Die, void());
   MOCK_METHOD0(Flush,
       void());
   MOCK_CONST_METHOD0(Empty,
       bool());
   MOCK_METHOD1(InsertPacket,
       int(Packet* packet));
   MOCK_METHOD4(InsertPacketList,
       int(PacketList* packet_list,
           const DecoderDatabase& decoder_database,
-          uint8_t* current_rtp_payload_type,
-          uint8_t* current_cng_rtp_payload_type));
+          rtc::Optional<uint8_t>* current_rtp_payload_type,
+          rtc::Optional<uint8_t>* current_cng_rtp_payload_type));
   MOCK_CONST_METHOD1(NextTimestamp,
       int(uint32_t* next_timestamp));
   MOCK_CONST_METHOD2(NextHigherTimestamp,
       int(uint32_t timestamp, uint32_t* next_timestamp));
   MOCK_CONST_METHOD0(NextRtpHeader,
       const RTPHeader*());
   MOCK_METHOD1(GetNextPacket,
       Packet*(size_t* discard_count));
   MOCK_METHOD0(DiscardNextPacket,
       int());
   MOCK_METHOD2(DiscardOldPackets,
       int(uint32_t timestamp_limit, uint32_t horizon_samples));
   MOCK_METHOD1(DiscardAllOldPackets,
       int(uint32_t timestamp_limit));
   MOCK_CONST_METHOD0(NumPacketsInBuffer,
       size_t());
   MOCK_METHOD1(IncrementWaitingTimes,
       void(int));
   MOCK_CONST_METHOD0(current_memory_bytes,
       int());
 };
 
 }  // namespace webrtc
 #endif  // WEBRTC_MODULES_AUDIO_CODING_NETEQ_MOCK_MOCK_PACKET_BUFFER_H_