Stop using old AudioCodingModule::RegisterReceiveCodec overloads

webrtc/modules/audio_coding/test/opus_test.cc

Index: webrtc/modules/audio_coding/test/opus_test.cc
diff --git a/webrtc/modules/audio_coding/test/opus_test.cc b/webrtc/modules/audio_coding/test/opus_test.cc
index d3f7b7aafa4b5321983900bd6c335fefeb8cd99b..78b00e22e2638e84b0f1af908da2ad053d7a6d17 100644
--- a/webrtc/modules/audio_coding/test/opus_test.cc
+++ b/webrtc/modules/audio_coding/test/opus_test.cc
@@ -94,7 +94,9 @@ void OpusTest::Perform() {
   int codec_id = acm_receiver_->Codec("opus", 48000, 2);
   EXPECT_EQ(0, acm_receiver_->Codec(codec_id, &opus_codec_param));
   payload_type_ = opus_codec_param.pltype;
-  EXPECT_EQ(0, acm_receiver_->RegisterReceiveCodec(opus_codec_param));
+  EXPECT_EQ(true,
+            acm_receiver_->RegisterReceiveCodec(
+                opus_codec_param.pltype, CodecInstToSdp(opus_codec_param)));
 
   // Create and connect the channel.
   channel_a2b_ = new TestPackStereo;
@@ -159,7 +161,9 @@ void OpusTest::Perform() {
 
   // Register Opus mono as receiving codec.
   opus_codec_param.channels = 1;
-  EXPECT_EQ(0, acm_receiver_->RegisterReceiveCodec(opus_codec_param));
+  EXPECT_EQ(true,
+            acm_receiver_->RegisterReceiveCodec(
+                opus_codec_param.pltype, CodecInstToSdp(opus_codec_param)));
 
   // Run Opus with 2.5 ms frame size.
   Run(channel_a2b_, audio_channels, 32000, 120);