NetEq now works with packets as values, rather than pointers.

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.
  */
 
 #ifndef WEBRTC_MODULES_AUDIO_CODING_NETEQ_MOCK_MOCK_PACKET_BUFFER_H_
 #define WEBRTC_MODULES_AUDIO_CODING_NETEQ_MOCK_MOCK_PACKET_BUFFER_H_
 
 #include "webrtc/modules/audio_coding/neteq/packet_buffer.h"
 
 #include "webrtc/test/gmock.h"
 
 namespace webrtc {
 
 class MockPacketBuffer : public PacketBuffer {
  public:
   MockPacketBuffer(size_t max_number_of_packets, const TickTimer* tick_timer)
       : PacketBuffer(max_number_of_packets, tick_timer) {}
   virtual ~MockPacketBuffer() { Die(); }
   MOCK_METHOD0(Die, void());
   MOCK_METHOD0(Flush,
       void());
   MOCK_CONST_METHOD0(Empty,
       bool());
-  MOCK_METHOD1(InsertPacket,
+  int InsertPacket(Packet&& packet) {
+    return InsertPacketWrapped(&packet);
+  }
+  // Since gtest does not properly support move-only types, InsertPacket is
+  // implemented as a wrapper. You'll have to implement InsertPacketWrapped
+  // instead and move from |*packet|.
+  MOCK_METHOD1(InsertPacketWrapped,
       int(Packet* packet));
   MOCK_METHOD4(InsertPacketList,
       int(PacketList* packet_list,
           const DecoderDatabase& decoder_database,
           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(PeekNextPacket,
       const Packet*());
-  MOCK_METHOD1(GetNextPacket,
-      Packet*(size_t* discard_count));
+  MOCK_METHOD0(GetNextPacket,
+      rtc::Optional<Packet>());
   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_