Fix Chromium clang plugin warnings

webrtc/modules/audio_coding/acm2/acm_send_test_oldapi.cc

 /*
  *  Copyright (c) 2014 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 29 matching lines @@
       timestamp_(0),
       sequence_number_(0) {
   input_frame_.sample_rate_hz_ = source_rate_hz_;
   input_frame_.num_channels_ = 1;
   input_frame_.samples_per_channel_ = input_block_size_samples_;
   assert(input_block_size_samples_ * input_frame_.num_channels_ <=
          AudioFrame::kMaxDataSizeSamples);
   acm_->RegisterTransportCallback(this);
 }
 
+AcmSendTestOldApi::~AcmSendTestOldApi() = default;
+
 bool AcmSendTestOldApi::RegisterCodec(const char* payload_name,
                                       int sampling_freq_hz,
                                       int channels,
                                       int payload_type,
                                       int frame_size_samples) {
   CodecInst codec;
   RTC_CHECK_EQ(0, AudioCodingModule::Codec(payload_name, &codec,
                                            sampling_freq_hz, channels));
   codec.pltype = payload_type;
   codec.pacsize = frame_size_samples;
@@ skipping 89 matching lines @@
          last_payload_vec_.size());
   std::unique_ptr<Packet> packet(
       new Packet(packet_memory, allocated_bytes, clock_.TimeInMilliseconds()));
   RTC_DCHECK(packet);
   RTC_DCHECK(packet->valid_header());
   return packet;
 }
 
 }  // namespace test
 }  // namespace webrtc