Reverse |rtx_payload_types| map, and rename.

webrtc/call/video_receive_stream.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 150 matching lines @@
 
       // See UlpfecConfig for description.
       UlpfecConfig ulpfec;
 
       // SSRC for retransmissions.
       uint32_t rtx_ssrc = 0;
 
       // Set if the stream is protected using FlexFEC.
       bool protected_by_flexfec = false;
 
-      // Map from video payload type (apt) -> RTX payload type (pt).
+      // Map from rtx payload type -> media payload type.
       // For RTX to be enabled, both an SSRC and this mapping are needed.
-      std::map<int, int> rtx_payload_types;
+      std::map<int, int> rtx_associated_payload_types;
       // TODO(nisse): This is a temporary accessor function to enable
       // reversing and renaming of the rtx_payload_types mapping.
       void AddRtxBinding(int rtx_payload_type, int media_payload_type) {
-        rtx_payload_types[media_payload_type] = rtx_payload_type;
+        rtx_associated_payload_types[rtx_payload_type] = media_payload_type;
       }
+
       // RTP header extensions used for the received stream.
       std::vector<RtpExtension> extensions;
     } rtp;
 
     // Transport for outgoing packets (RTCP).
     Transport* rtcp_send_transport = nullptr;
 
     // Must not be 'nullptr' when the stream is started.
     rtc::VideoSinkInterface<VideoFrame>* renderer = nullptr;
 
@@ skipping 51 matching lines @@
   virtual void AddSecondarySink(RtpPacketSinkInterface* sink) = 0;
   virtual void RemoveSecondarySink(const RtpPacketSinkInterface* sink) = 0;
 
  protected:
   virtual ~VideoReceiveStream() {}
 };
 
 }  // namespace webrtc
 
 #endif  // WEBRTC_CALL_VIDEO_RECEIVE_STREAM_H_