Stop using old AudioCodingModule::RegisterReceiveCodec overloads

webrtc/modules/audio_coding/test/delay_test.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 89 matching lines @@
           "Failed to get codec.";
       if (STR_CASE_CMP(my_codec_param.plname, "opus") == 0)
         my_codec_param.channels = 1;
       else if (my_codec_param.channels > 1)
         continue;
       if (STR_CASE_CMP(my_codec_param.plname, "CN") == 0 &&
           my_codec_param.plfreq == 48000)
         continue;
       if (STR_CASE_CMP(my_codec_param.plname, "telephone-event") == 0)
         continue;
-      ASSERT_EQ(0, acm_b_->RegisterReceiveCodec(my_codec_param)) <<
-          "Couldn't register receive codec.\n";
+      ASSERT_EQ(true,
+                acm_b_->RegisterReceiveCodec(my_codec_param.pltype,
+                                             CodecInstToSdp(my_codec_param)))
+          << "Couldn't register receive codec.\n";

[ossu, 2016/10/05 13:39:58]

Completely unrelated: It can't be the case that this call is the only one in the
world that requires a newline at the end of it. :)

[kwiberg-webrtc, 2016/10/06 12:14:52]

Probably not (although I'd be surprised if no other ASSERT message had a
trailing newline...). Removed the message altogether.

     }
 
     // Create and connect the channel
     ASSERT_EQ(0, acm_a_->RegisterTransportCallback(channel_a2b_)) <<
         "Couldn't register Transport callback.\n";
     channel_a2b_->RegisterReceiverACM(acm_b_.get());
   }
 
   void Perform(const TestSettings* config, size_t num_tests, int duration_sec,
                const char* output_prefix) {
@@ skipping 136 matching lines @@
   test_setting.codec.num_channels = FLAGS_num_channels;
   test_setting.acm.dtx = FLAGS_dtx;
   test_setting.acm.fec = FLAGS_fec;
   test_setting.packet_loss = FLAGS_packet_loss;
 
   webrtc::DelayTest delay_test;
   delay_test.Initialize();
   delay_test.Perform(&test_setting, 1, 240, "delay_test");
   return 0;
 }