Stop using old AudioCodingModule::RegisterReceiveCodec overloads

webrtc/modules/audio_coding/test/target_delay_unittest.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 12 matching lines @@
 
 class TargetDelayTest : public ::testing::Test {
  protected:
   TargetDelayTest() : acm_(AudioCodingModule::Create(0)) {}
 
   ~TargetDelayTest() {}
 
   void SetUp() {
     EXPECT_TRUE(acm_.get() != NULL);
 
-    CodecInst codec;
-    ASSERT_EQ(0, AudioCodingModule::Codec("L16", &codec, kSampleRateHz, 1));
     ASSERT_EQ(0, acm_->InitializeReceiver());
-    ASSERT_EQ(0, acm_->RegisterReceiveCodec(codec));
+    constexpr int pltype = 108;
+    ASSERT_EQ(true,
+              acm_->RegisterReceiveCodec(pltype, {"L16", kSampleRateHz, 1}));
 
-    rtp_info_.header.payloadType = codec.pltype;
+    rtp_info_.header.payloadType = pltype;
     rtp_info_.header.timestamp = 0;
     rtp_info_.header.ssrc = 0x12345678;
     rtp_info_.header.markerBit = false;
     rtp_info_.header.sequenceNumber = 0;
     rtp_info_.type.Audio.channel = 1;
     rtp_info_.type.Audio.isCNG = false;
     rtp_info_.frameType = kAudioFrameSpeech;
 
     int16_t audio[kFrameSizeSamples];
     const int kRange = 0x7FF;  // 2047, easy for masking.
@@ skipping 194 matching lines @@
 #define MAYBE_TargetDelayBufferMinMax DISABLED_TargetDelayBufferMinMax
 #else
 #define MAYBE_TargetDelayBufferMinMax TargetDelayBufferMinMax
 #endif
 TEST_F(TargetDelayTest, MAYBE_TargetDelayBufferMinMax) {
   TargetDelayBufferMinMax();
 }
 
 }  // namespace webrtc