Stop using old AudioCodingModule::RegisterReceiveCodec overloads

webrtc/modules/audio_coding/test/target_delay_unittest.cc

Index: webrtc/modules/audio_coding/test/target_delay_unittest.cc
diff --git a/webrtc/modules/audio_coding/test/target_delay_unittest.cc b/webrtc/modules/audio_coding/test/target_delay_unittest.cc
index 2939453e4f9a7011a6db48a6e88e341ef72f3774..1b400262ae0c633b187cbb8b41c6fa3c78ad5cdf 100644
--- a/webrtc/modules/audio_coding/test/target_delay_unittest.cc
+++ b/webrtc/modules/audio_coding/test/target_delay_unittest.cc
@@ -30,12 +30,12 @@ class TargetDelayTest : public ::testing::Test {
   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;