Rename {,Set}GenericFECStatus to {,Set}UlpfecConfig.

webrtc/modules/rtp_rtcp/source/rtp_sender_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 42 matching lines @@
                  const uint8_t* payload_data,
                  size_t payload_size,
                  const RTPFragmentationHeader* fragmentation,
                  const RTPVideoHeader* video_header);
 
   int32_t SendRTPIntraRequest();
 
   void SetVideoCodecType(RtpVideoCodecTypes type);
 
   // FEC
-  void SetGenericFECStatus(bool enable,
-                           uint8_t payload_type_red,
-                           uint8_t payload_type_fec);
+  void SetUlpfecConfig(bool enabled,
+                       int red_payload_type,
+                       int ulpfec_payload_type);
 
-  void GenericFECStatus(bool* enable,
-                        uint8_t* payload_type_red,
-                        uint8_t* payload_type_fec) const;
+  void UlpfecConfig(bool* enabled,

[stefan-webrtc, 2016/10/31 09:33:16]

Perhaps GetUlpfecConfig

[brandtr, 2016/10/31 12:47:03]

Done.

+                    int* red_payload_type,
+                    int* ulpfec_payload_type) const;
 
   void SetFecParameters(const FecProtectionParams* delta_params,
                         const FecProtectionParams* key_params);
 
   uint32_t VideoBitrateSent() const;
   uint32_t FecOverheadRate() const;
 
   int SelectiveRetransmissions() const;
   void SetSelectiveRetransmissions(uint8_t settings);
 
@@ skipping 11 matching lines @@
   // Should never be held when calling out of this class.
   rtc::CriticalSection crit_;
   rtc::SequencedTaskChecker encoder_checker_;
 
   RtpVideoCodecTypes video_type_ = kRtpVideoGeneric;
   int32_t retransmission_settings_ GUARDED_BY(crit_) = kRetransmitBaseLayer;
   VideoRotation last_rotation_ GUARDED_BY(encoder_checker_) = kVideoRotation_0;
 
   // FEC
   bool fec_enabled_ GUARDED_BY(crit_) = false;
-  int8_t red_payload_type_ GUARDED_BY(crit_) = 0;
-  int8_t fec_payload_type_ GUARDED_BY(crit_) = 0;
+  int red_payload_type_ GUARDED_BY(crit_) = -1;
+  int fec_payload_type_ GUARDED_BY(crit_) = -1;

[stefan-webrtc, 2016/10/31 09:33:16]

I'd prefer moving these initializations to the initializer list. Feel free to do
that...

[brandtr, 2016/10/31 12:47:03]

Done!

This was originally my plan, but then I saw this CL:
https://codereview.webrtc.org/2067673004/diff/200001/webrtc/modules/rtp_rtcp/...

But with your mandate, I'm happy to do this :)

[danilchap, 2016/10/31 13:22:00]

just curious, what are benefits of initializer list over default Member
Initializers?
Default member initializers make it easier to add new member, reorder existent
members, make it easier to see when a POD member miss initial value.

[stefan-webrtc, 2016/11/03 13:52:25]

We discussed this a lot at some point, and I think the conclusion was that it
harms readability if some members are initialized in the initializer list while
others are initialized in the .h file. At that point I remember that there
wasn't a clear consensus on whether this also would apply for const variables,
but I tend to just prefer everything in initializer lists to keep it simple.

Feel free to bring it up for discussion again if you think it's wrong though!

   FecProtectionParams delta_fec_params_ GUARDED_BY(crit_) = FecProtectionParams{
       0, 1, kFecMaskRandom};
   FecProtectionParams key_fec_params_ GUARDED_BY(crit_) = FecProtectionParams{
       0, 1, kFecMaskRandom};
   ProducerFec producer_fec_ GUARDED_BY(crit_);
 
   rtc::CriticalSection stats_crit_;
   // Bitrate used for FEC payload, RED headers, RTP headers for FEC packets
   // and any padding overhead.
   RateStatistics fec_bitrate_ GUARDED_BY(stats_crit_);
   // Bitrate used for video payload and RTP headers.
   RateStatistics video_bitrate_ GUARDED_BY(stats_crit_);
   OneTimeEvent first_frame_sent_;
 };
 
 }  // namespace webrtc
 
 #endif  // WEBRTC_MODULES_RTP_RTCP_SOURCE_RTP_SENDER_VIDEO_H_