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

webrtc/modules/audio_coding/neteq/packet_buffer.cc

 /*
  *  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 90 matching lines @@
       packet->header.timestamp == (*it)->header.timestamp) {
     delete [] (*it)->payload;
     delete *it;
     it = buffer_.erase(it);
   }
   buffer_.insert(it, packet);  // Insert the packet at that position.
 
   return return_val;
 }
 
-int PacketBuffer::InsertPacketList(PacketList* packet_list,
-                                   const DecoderDatabase& decoder_database,
-                                   uint8_t* current_rtp_payload_type,
-                                   uint8_t* current_cng_rtp_payload_type) {
+int PacketBuffer::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) {
   bool flushed = false;
   while (!packet_list->empty()) {
     Packet* packet = packet_list->front();
     if (decoder_database.IsComfortNoise(packet->header.payloadType)) {
-      if (*current_cng_rtp_payload_type != 0xFF &&
-          *current_cng_rtp_payload_type != packet->header.payloadType) {
+      if (*current_cng_rtp_payload_type &&
+          **current_cng_rtp_payload_type != packet->header.payloadType) {
         // New CNG payload type implies new codec type.
-        *current_rtp_payload_type = 0xFF;
+        *current_rtp_payload_type = rtc::Optional<uint8_t>();
         Flush();
         flushed = true;
       }
-      *current_cng_rtp_payload_type = packet->header.payloadType;
+      *current_cng_rtp_payload_type =
+          rtc::Optional<uint8_t>(packet->header.payloadType);
     } else if (!decoder_database.IsDtmf(packet->header.payloadType)) {
       // This must be speech.
-      if (*current_rtp_payload_type != 0xFF &&
-          *current_rtp_payload_type != packet->header.payloadType) {
-        *current_cng_rtp_payload_type = 0xFF;
+      if (*current_rtp_payload_type &&
+          **current_rtp_payload_type != packet->header.payloadType) {
+        *current_cng_rtp_payload_type = rtc::Optional<uint8_t>();
         Flush();
         flushed = true;
       }
-      *current_rtp_payload_type = packet->header.payloadType;
+      *current_rtp_payload_type =
+          rtc::Optional<uint8_t>(packet->header.payloadType);
     }
     int return_val = InsertPacket(packet);
     packet_list->pop_front();
     if (return_val == kFlushed) {
       // The buffer flushed, but this is not an error. We can still continue.
       flushed = true;
     } else if (return_val != kOK) {
       // An error occurred. Delete remaining packets in list and return.
       DeleteAllPackets(packet_list);
       return return_val;
@@ skipping 142 matching lines @@
     // Continue while the list is not empty.
   }
 }
 
 void PacketBuffer::BufferStat(int* num_packets, int* max_num_packets) const {
   *num_packets = static_cast<int>(buffer_.size());
   *max_num_packets = static_cast<int>(max_number_of_packets_);
 }
 
 }  // namespace webrtc