Send audio and video codecs to RTPPayloadRegistry

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

 /*
  *  Copyright (c) 2012 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 116 matching lines @@
         rtp_payload_registry2_.get()));
 
     transport1.SetSendModule(module2.get(), rtp_payload_registry2_.get(),
                              rtp_receiver2_.get(), receive_statistics2_.get());
     transport2.SetSendModule(module1.get(), rtp_payload_registry1_.get(),
                              rtp_receiver1_.get(), receive_statistics1_.get());
   }
 
   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;
   VerifyingAudioReceiver data_receiver2;
   RTPCallback rtp_callback;
   std::unique_ptr<ReceiveStatistics> receive_statistics1_;
   std::unique_ptr<ReceiveStatistics> receive_statistics2_;
   std::unique_ptr<RTPPayloadRegistry> rtp_payload_registry1_;
   std::unique_ptr<RTPPayloadRegistry> rtp_payload_registry2_;
   std::unique_ptr<RtpReceiver> rtp_receiver1_;
@@ skipping 55 matching lines @@
   module1->SetSSRC(test_ssrc);
   module1->SetStartTimestamp(test_timestamp);
   EXPECT_EQ(0, module1->SetSendingStatus(true));
 
   // Prepare for DTMF.
   voice_codec.pltype = kDtmfPayloadType;
   voice_codec.plfreq = 8000;
   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;
 
   // Send a DTMF tone using RFC 2833 (4733).
   for (int i = 0; i < 16; i++) {
     EXPECT_EQ(0, module1->SendTelephoneEventOutband(i, timeStamp, 10));
   }
   timeStamp += 160;  // Prepare for next packet.
 
@@ skipping 57 matching lines @@
                                           nullptr, nullptr, nullptr));
 
     EXPECT_EQ(test_ssrc, rtp_receiver2_->SSRC());
     EXPECT_TRUE(rtp_receiver2_->Timestamp(&timestamp));
     EXPECT_EQ(test_timestamp + in_timestamp, timestamp);
     in_timestamp += 10;
   }
 }
 
 }  // namespace webrtc