Rename RtpPayloadTypeRegistry::SetCodec() to ::AddCodec().

webrtc/video/rtp_stream_receiver.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 64 matching lines @@
       const VideoReceiveStream::Config* config,
       ReceiveStatisticsProxy* receive_stats_proxy,
       ProcessThread* process_thread,
       RateLimiter* retransmission_rate_limiter,
       NackSender* nack_sender,
       KeyFrameRequestSender* keyframe_request_sender,
       video_coding::OnCompleteFrameCallback* complete_frame_callback,
       VCMTiming* timing);
   ~RtpStreamReceiver();
 
-  bool SetReceiveCodec(const VideoCodec& video_codec);
+  bool AddReceiveCodec(const VideoCodec& video_codec);
 
   uint32_t GetRemoteSsrc() const;
   int GetCsrcs(uint32_t* csrcs) const;
 
   RtpReceiver* GetRtpReceiver() const;
   RtpRtcp* rtp_rtcp() const { return rtp_rtcp_.get(); }
 
   void StartReceive();
   void StopReceive();
 
@@ skipping 98 matching lines @@
   std::unique_ptr<video_coding::RtpFrameReferenceFinder> reference_finder_;
   rtc::CriticalSection last_seq_num_cs_;
   std::map<uint16_t, uint16_t, DescendingSeqNumComp<uint16_t>>
       last_seq_num_for_pic_id_ GUARDED_BY(last_seq_num_cs_);
   video_coding::H264SpsPpsTracker tracker_;
 };
 
 }  // namespace webrtc
 
 #endif  // WEBRTC_VIDEO_RTP_STREAM_RECEIVER_H_