Send audio and video codecs to RTPPayloadRegistry

webrtc/modules/rtp_rtcp/source/rtp_receiver_video.h

 /*
  *  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 21 matching lines @@
                          size_t packet_length,
                          int64_t timestamp,
                          bool is_first_packet) override;
 
   TelephoneEventHandler* GetTelephoneEventHandler() override { return NULL; }
 
   RTPAliveType ProcessDeadOrAlive(uint16_t last_payload_length) const override;
 
   bool ShouldReportCsrcChanges(uint8_t payload_type) const override;
 
-  int32_t OnNewPayloadTypeCreated(
-      const char payload_name[RTP_PAYLOAD_NAME_SIZE],
-      int8_t payload_type,
-      uint32_t frequency) override;
+  int32_t OnNewPayloadTypeCreated(const CodecInst& audio_codec) override;
 
   int32_t InvokeOnInitializeDecoder(
       RtpFeedback* callback,
       int8_t payload_type,
       const char payload_name[RTP_PAYLOAD_NAME_SIZE],
       const PayloadUnion& specific_payload) const override;
 
   void SetPacketOverHead(uint16_t packet_over_head);
 
  private:
   OneTimeEvent first_packet_received_;
 };
 }  // namespace webrtc
 
 #endif  // WEBRTC_MODULES_RTP_RTCP_SOURCE_RTP_RECEIVER_VIDEO_H_