NetEq: only update current_rtp_payload_type_ when validated

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 580 matching lines @@
     dtmf_buffer_->Flush();
 
     // Store new SSRC.
     ssrc_ = main_header.ssrc;
 
     // Update audio buffer timestamp.
     sync_buffer_->IncreaseEndTimestamp(main_header.timestamp - timestamp_);
 
     // Update codecs.
     timestamp_ = main_header.timestamp;
-    current_rtp_payload_type_ = main_header.payloadType;
 
     // Reset timestamp scaling.
     timestamp_scaler_->Reset();
 
     // Trigger an update of sampling rate and the number of channels.
     update_sample_rate_and_channels = true;
   }
 
   // Update RTCP statistics, only for regular packets.
   if (!is_sync_packet)
@@ skipping 125 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;
   }
 
-  if (current_rtp_payload_type_ != 0xFF) {
-    const DecoderDatabase::DecoderInfo* dec_info =
-        decoder_database_->GetDecoderInfo(current_rtp_payload_type_);
-    if (!dec_info) {
-      assert(false);  // Already checked that the payload type is known.
-    }
-  }
+  RTC_DCHECK(current_rtp_payload_type_ == 0xFF ||
+             decoder_database_->GetDecoderInfo(current_rtp_payload_type_))
+      << "Payload type " << static_cast<int>(current_rtp_payload_type_)
+      << " is unknown where it shouldn't be";
 
   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