Send audio and video codecs to RTPPayloadRegistry

webrtc/modules/rtp_rtcp/test/testAPI/test_api_video.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 53 matching lines @@
 
     transport_->SetSendModule(video_module_, &rtp_payload_registry_,
                               rtp_receiver_.get(), receive_statistics_.get());
 
     VideoCodec video_codec;
     memset(&video_codec, 0, sizeof(video_codec));
     video_codec.plType = 123;
     memcpy(video_codec.plName, "I420", 5);
 
     EXPECT_EQ(0, video_module_->RegisterSendPayload(video_codec));
-    EXPECT_EQ(0, rtp_receiver_->RegisterReceivePayload(video_codec.plName,
-                                                       video_codec.plType,
-                                                       90000,
-                                                       0,
-                                                       video_codec.maxBitrate));
+    EXPECT_EQ(0, rtp_payload_registry_.RegisterReceivePayload(video_codec));
 
     payload_data_length_ = sizeof(video_frame_);
 
     for (size_t n = 0; n < payload_data_length_; n++) {
       video_frame_[n] = n%10;
     }
   }
 
   size_t BuildRTPheader(uint8_t* dataBuffer,
                          uint32_t timestamp,
@@ skipping 65 matching lines @@
 
 TEST_F(RtpRtcpVideoTest, PaddingOnlyFrames) {
   const size_t kPadSize = 255;
   uint8_t padding_packet[kPadSize];
   uint32_t seq_num = 0;
   uint32_t timestamp = 3000;
   VideoCodec codec;
   codec.codecType = kVideoCodecVP8;
   codec.plType = kPayloadType;
   strncpy(codec.plName, "VP8", 4);
-  EXPECT_EQ(0, rtp_receiver_->RegisterReceivePayload(codec.plName,
-                                                     codec.plType,
-                                                     90000,
-                                                     0,
-                                                     codec.maxBitrate));
+  EXPECT_EQ(0, rtp_payload_registry_.RegisterReceivePayload(codec));
   for (int frame_idx = 0; frame_idx < 10; ++frame_idx) {
     for (int packet_idx = 0; packet_idx < 5; ++packet_idx) {
       size_t packet_size = PaddingPacket(padding_packet, timestamp, seq_num,
                                          kPadSize);
       ++seq_num;
       RTPHeader header;
       std::unique_ptr<RtpHeaderParser> parser(RtpHeaderParser::Create());
       EXPECT_TRUE(parser->Parse(padding_packet, packet_size, &header));
       PayloadUnion payload_specific;
       EXPECT_TRUE(rtp_payload_registry_.GetPayloadSpecifics(header.payloadType,
                                                             &payload_specific));
       const uint8_t* payload = padding_packet + header.headerLength;
       const size_t payload_length = packet_size - header.headerLength;
       EXPECT_TRUE(rtp_receiver_->IncomingRtpPacket(header, payload,
                                                    payload_length,
                                                    payload_specific, true));
       EXPECT_EQ(0u, receiver_->payload_size());
       EXPECT_EQ(payload_length, receiver_->rtp_header().header.paddingLength);
     }
     timestamp += 3000;
     fake_clock.AdvanceTimeMilliseconds(33);
   }
 }
 
 }  // namespace webrtc