Send audio and video codecs to RTPPayloadRegistry

webrtc/modules/rtp_rtcp/source/rtp_receiver_impl.h

 /*
  *  Copyright (c) 2013 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 15 matching lines @@
   // Callbacks passed in here may not be NULL (use Null Object callbacks if you
   // want callbacks to do nothing). This class takes ownership of the media
   // receiver but nothing else.
   RtpReceiverImpl(Clock* clock,
                   RtpFeedback* incoming_messages_callback,
                   RTPPayloadRegistry* rtp_payload_registry,
                   RTPReceiverStrategy* rtp_media_receiver);
 
   virtual ~RtpReceiverImpl();
 
+  int32_t RegisterReceivePayload(const CodecInst& audio_codec) override;
+  // TODO(magjed): Remove once external code is updated.
   int32_t RegisterReceivePayload(const char payload_name[RTP_PAYLOAD_NAME_SIZE],
                                  const int8_t payload_type,
                                  const uint32_t frequency,
                                  const size_t channels,
                                  const uint32_t rate) override;
 
   int32_t DeRegisterReceivePayload(const int8_t payload_type) override;
 
   bool IncomingRtpPacket(const RTPHeader& rtp_header,
                          const uint8_t* payload,
@@ skipping 37 matching lines @@
   uint32_t ssrc_;
   uint8_t num_csrcs_;
   uint32_t current_remote_csrc_[kRtpCsrcSize];
 
   uint32_t last_received_timestamp_;
   int64_t last_received_frame_time_ms_;
   uint16_t last_received_sequence_number_;
 };
 }  // namespace webrtc
 #endif  // WEBRTC_MODULES_RTP_RTCP_SOURCE_RTP_RECEIVER_IMPL_H_