Stop using old AudioCodingModule::RegisterReceiveCodec overloads

webrtc/modules/audio_coding/test/insert_packet_with_timing.cc

 /*
  *  Copyright (c) 2013 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 76 matching lines @@
     ASSERT_TRUE(playout_timing_fid_ != NULL);
 
     next_receive_ts_ = ReceiveTimestamp();
 
     CodecInst codec;
     ASSERT_EQ(0, AudioCodingModule::Codec(FLAGS_codec.c_str(), &codec,
                              FLAGS_codec_sample_rate_hz,
                              FLAGS_codec_channels));
     ASSERT_EQ(0, receive_acm_->InitializeReceiver());
     ASSERT_EQ(0, send_acm_->RegisterSendCodec(codec));
-    ASSERT_EQ(0, receive_acm_->RegisterReceiveCodec(codec));
+    ASSERT_EQ(true, receive_acm_->RegisterReceiveCodec(codec.pltype,
+                                                       CodecInstToSdp(codec)));
 
     // Set codec-dependent parameters.
     samples_in_1ms_ = codec.plfreq / 1000;
     num_10ms_in_codec_frame_ = codec.pacsize / (codec.plfreq / 100);
 
     channel_->RegisterReceiverACM(receive_acm_.get());
     send_acm_->RegisterTransportCallback(channel_);
 
     if (FLAGS_input.size() == 0) {
       std::string file_name = test::ResourcePath("audio_coding/testfile32kHz",
@@ skipping 193 matching lines @@
       if (delay_log != NULL) {
         fprintf(delay_log, "%3d %3d\n", optimal_delay_ms, current_delay_ms);
       }
     }
   }
   std::cout << std::endl;
   test.TearDown();
   if (delay_log != NULL)
     fclose(delay_log);
 }