Send audio and video codecs to RTPPayloadRegistry

webrtc/modules/rtp_rtcp/test/testAPI/test_api_audio.cc

Index: webrtc/modules/rtp_rtcp/test/testAPI/test_api_audio.cc
diff --git a/webrtc/modules/rtp_rtcp/test/testAPI/test_api_audio.cc b/webrtc/modules/rtp_rtcp/test/testAPI/test_api_audio.cc
index 5d66fd41898d0571d6cf89a386020c4957310a27..175771755e1701c4453640aa361c6d23ab8dd68e 100644
--- a/webrtc/modules/rtp_rtcp/test/testAPI/test_api_audio.cc
+++ b/webrtc/modules/rtp_rtcp/test/testAPI/test_api_audio.cc
@@ -134,19 +134,9 @@ class RtpRtcpAudioTest : public ::testing::Test {
 
   void RegisterPayload(const CodecInst& codec) {
     EXPECT_EQ(0, module1->RegisterSendPayload(codec));
-    EXPECT_EQ(0, rtp_receiver1_->RegisterReceivePayload(
-        codec.plname,
-        codec.pltype,
-        codec.plfreq,
-        codec.channels,
-        codec.rate));
+    EXPECT_EQ(0, rtp_receiver1_->RegisterReceivePayload(codec));
     EXPECT_EQ(0, module2->RegisterSendPayload(codec));
-    EXPECT_EQ(0, rtp_receiver2_->RegisterReceivePayload(
-        codec.plname,
-        codec.pltype,
-        codec.plfreq,
-        codec.channels,
-        codec.rate));
+    EXPECT_EQ(0, rtp_receiver2_->RegisterReceivePayload(codec));
   }
 
   VerifyingAudioReceiver data_receiver1;
@@ -222,12 +212,7 @@ TEST_F(RtpRtcpAudioTest, DTMF) {
   memcpy(voice_codec.plname, "telephone-event", 16);
 
   EXPECT_EQ(0, module1->RegisterSendPayload(voice_codec));
-  EXPECT_EQ(0, rtp_receiver2_->RegisterReceivePayload(
-        voice_codec.plname,
-        voice_codec.pltype,
-        voice_codec.plfreq,
-        voice_codec.channels,
-        voice_codec.rate));
+  EXPECT_EQ(0, rtp_receiver2_->RegisterReceivePayload(voice_codec));
 
   // Start DTMF test.
   int timeStamp = 160;