Let AudioCodingModule::SendCodec return Maybe<CodecInst>

webrtc/modules/audio_coding/main/test/PacketLossTest.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 125 matching lines @@
 
   // Encode to file
   rtpFile.Open(fileName.c_str(), "wb+");
   rtpFile.WriteHeader();
 
   sender_->testMode = 0;
   sender_->codeId = codec_id;
 
   sender_->Setup(acm.get(), &rtpFile, in_file_name_, sample_rate_hz_, channels_,
                  expected_loss_rate_);
-  struct CodecInst sendCodecInst;
-  if (acm->SendCodec(&sendCodecInst) >= 0) {
+  if (acm->SendCodec()) {
     sender_->Run();
   }
   sender_->Teardown();
   rtpFile.Close();
 
   // Decode to file
   rtpFile.Open(fileName.c_str(), "rb");
   rtpFile.ReadHeader();
 
   receiver_->testMode = 0;
   receiver_->codeId = codec_id;
 
   receiver_->Setup(acm.get(), &rtpFile, "packetLoss_out", channels_,
                    actual_loss_rate_, burst_length_);
   receiver_->Run();
   receiver_->Teardown();
   rtpFile.Close();
 #endif
 }
 
 }  // namespace webrtc