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

webrtc/modules/audio_coding/neteq/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_PACKET_BUFFER_H_
 #define WEBRTC_MODULES_AUDIO_CODING_NETEQ_PACKET_BUFFER_H_
 
 #include "webrtc/base/constructormagic.h"
+#include "webrtc/base/optional.h"
 #include "webrtc/modules/audio_coding/neteq/packet.h"
 #include "webrtc/typedefs.h"
 
 namespace webrtc {
 
 class DecoderDatabase;
 class TickTimer;
 
 // This is the actual buffer holding the packets before decoding.
 class PacketBuffer {
@@ skipping 27 matching lines @@
   virtual int InsertPacket(Packet* packet);
 
   // Inserts a list of packets into the buffer. The buffer will take over
   // ownership of the packet objects.
   // Returns PacketBuffer::kOK if all packets were inserted successfully.
   // If the buffer was flushed due to overfilling, only a subset of the list is
   // inserted, and PacketBuffer::kFlushed is returned.
   // The last three parameters are included for legacy compatibility.
   // TODO(hlundin): Redesign to not use current_*_payload_type and
   // decoder_database.
-  virtual int InsertPacketList(PacketList* packet_list,
-                               const DecoderDatabase& decoder_database,
-                               uint8_t* current_rtp_payload_type,
-                               uint8_t* current_cng_rtp_payload_type);
+  virtual int InsertPacketList(
+      PacketList* packet_list,
+      const DecoderDatabase& decoder_database,
+      rtc::Optional<uint8_t>* current_rtp_payload_type,
+      rtc::Optional<uint8_t>* current_cng_rtp_payload_type);
 
   // Gets the timestamp for the first packet in the buffer and writes it to the
   // output variable |next_timestamp|.
   // Returns PacketBuffer::kBufferEmpty if the buffer is empty,
   // PacketBuffer::kOK otherwise.
   virtual int NextTimestamp(uint32_t* next_timestamp) const;
 
   // Gets the timestamp for the first packet in the buffer with a timestamp no
   // lower than the input limit |timestamp|. The result is written to the output
   // variable |next_timestamp|.
@@ skipping 67 matching lines @@
 
  private:
   size_t max_number_of_packets_;
   PacketList buffer_;
   const TickTimer* tick_timer_;
   RTC_DISALLOW_COPY_AND_ASSIGN(PacketBuffer);
 };
 
 }  // namespace webrtc
 #endif  // WEBRTC_MODULES_AUDIO_CODING_NETEQ_PACKET_BUFFER_H_