Stop using old AudioCodingModule::RegisterReceiveCodec overloads

webrtc/modules/audio_coding/test/EncodeDecodeTest.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 114 matching lines @@
 void Receiver::Setup(AudioCodingModule *acm, RTPStream *rtpStream,
                      std::string out_file_name, size_t channels) {
   struct CodecInst recvCodec = CodecInst();
   int noOfCodecs;
   EXPECT_EQ(0, acm->InitializeReceiver());
 
   noOfCodecs = acm->NumberOfCodecs();
   for (int i = 0; i < noOfCodecs; i++) {
     EXPECT_EQ(0, acm->Codec(i, &recvCodec));
     if (recvCodec.channels == channels)
-      EXPECT_EQ(0, acm->RegisterReceiveCodec(recvCodec));
+      EXPECT_EQ(true, acm->RegisterReceiveCodec(recvCodec.pltype,
+                                                CodecInstToSdp(recvCodec)));
     // Forces mono/stereo for Opus.
     if (!strcmp(recvCodec.plname, "opus")) {
       recvCodec.channels = channels;
-      EXPECT_EQ(0, acm->RegisterReceiveCodec(recvCodec));
+      EXPECT_EQ(true, acm->RegisterReceiveCodec(recvCodec.pltype,
+                                                CodecInstToSdp(recvCodec)));
     }
   }
 
   int playSampFreq;
   std::string file_name;
   std::stringstream file_stream;
   file_stream << webrtc::test::OutputPath() << out_file_name
       << static_cast<int>(codeId) << ".pcm";
   file_name = file_stream.str();
   _rtpStream = rtpStream;
@@ skipping 200 matching lines @@
   if (acm->SendCodec()) {
     _sender.Run();
   }
   _sender.Teardown();
   rtpFile.Close();
 
   return fileName;
 }
 
 }  // namespace webrtc