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

webrtc/modules/audio_coding/neteq/neteq_impl.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 77 matching lines @@
       expand_factory_(std::move(deps.expand_factory)),
       accelerate_factory_(std::move(deps.accelerate_factory)),
       preemptive_expand_factory_(std::move(deps.preemptive_expand_factory)),
       last_mode_(kModeNormal),
       decoded_buffer_length_(kMaxFrameSize),
       decoded_buffer_(new int16_t[decoded_buffer_length_]),
       playout_timestamp_(0),
       new_codec_(false),
       timestamp_(0),
       reset_decoder_(false),
-      current_rtp_payload_type_(0xFF),      // Invalid RTP payload type.
-      current_cng_rtp_payload_type_(0xFF),  // Invalid RTP payload type.
       ssrc_(0),
       first_packet_(true),
       error_code_(0),
       decoder_error_code_(0),
       background_noise_mode_(config.background_noise_mode),
       playout_mode_(config.playout_mode),
       enable_fast_accelerate_(config.enable_fast_accelerate),
       nack_enabled_(false),
       enable_muted_state_(config.enable_muted_state) {
   LOG(LS_INFO) << "NetEq config: " << config.ToString();
@@ skipping 420 matching lines @@
   }
   // Sanity checks for sync-packets.
   if (is_sync_packet) {
     if (decoder_database_->IsDtmf(rtp_header.header.payloadType) ||
         decoder_database_->IsRed(rtp_header.header.payloadType) ||
         decoder_database_->IsComfortNoise(rtp_header.header.payloadType)) {
       LOG_F(LS_ERROR) << "Sync-packet with an unacceptable payload type "
                       << static_cast<int>(rtp_header.header.payloadType);
       return kSyncPacketNotAccepted;
     }
-    if (first_packet_ ||
-        rtp_header.header.payloadType != current_rtp_payload_type_ ||
+    if (first_packet_ || !current_rtp_payload_type_ ||
+        rtp_header.header.payloadType != *current_rtp_payload_type_ ||
         rtp_header.header.ssrc != ssrc_) {
-      // Even if |current_rtp_payload_type_| is 0xFF, sync-packet isn't
+      // Even if |current_rtp_payload_type_| is empty, sync-packet isn't
       // accepted.
       LOG_F(LS_ERROR)
           << "Changing codec, SSRC or first packet with sync-packet.";
       return kSyncPacketNotAccepted;
     }
   }
   PacketList packet_list;
   RTPHeader main_header;
   {
     // Convert to Packet.
@@ skipping 182 matching lines @@
     PacketBuffer::DeleteAllPackets(&packet_list);
     return kOtherError;
   }
 
   if (first_packet_) {
     first_packet_ = false;
     // Update the codec on the next GetAudio call.
     new_codec_ = true;
   }
 
-  RTC_DCHECK(current_rtp_payload_type_ == 0xFF ||
-             decoder_database_->GetDecoderInfo(current_rtp_payload_type_))
-      << "Payload type " << static_cast<int>(current_rtp_payload_type_)
+  RTC_DCHECK(!current_rtp_payload_type_ ||
+             decoder_database_->GetDecoderInfo(*current_rtp_payload_type_))
+      << "Payload type " << static_cast<int>(*current_rtp_payload_type_)
       << " is unknown where it shouldn't be";

[kwiberg-webrtc, 2016/08/30 10:29:32]

If !current_rtp_payload_type_, you evaluate *current_rtp_payload_type_ when
producing the error message. Maybe

  if (current_rtp_payload_type_) {
    RTC_DCHECK(decoder_database_
        ->GetDecoderInfo(*current_rtp_payload_type_))
        << "Payload type "
        << static_cast<int>(*current_rtp_payload_type_)
        << " is unknown where it shouldn't be";
  }

[hlundin-webrtc, 2016/08/30 10:53:16]

Oh, that's not good. Changed. Done.

 
   if (update_sample_rate_and_channels && !packet_buffer_->Empty()) {
     // We do not use |current_rtp_payload_type_| to |set payload_type|, but
     // get the next RTP header from |packet_buffer_| to obtain the payload type.
     // The reason for it is the following corner case. If NetEq receives a
     // CNG packet with a sample rate different than the current CNG then it
     // flushes its buffer, assuming send codec must have been changed. However,
     // payload type of the hypothetically new send codec is not known.
     const RTPHeader* rtp_header = packet_buffer_->NextRtpHeader();
     assert(rtp_header);
@@ skipping 1375 matching lines @@
   }
 }
 
 void NetEqImpl::CreateDecisionLogic() {
   decision_logic_.reset(DecisionLogic::Create(
       fs_hz_, output_size_samples_, playout_mode_, decoder_database_.get(),
       *packet_buffer_.get(), delay_manager_.get(), buffer_level_filter_.get(),
       tick_timer_.get()));
 }
 }  // namespace webrtc