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

webrtc/modules/rtp_rtcp/source/rtp_sender.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 166 matching lines @@
 
   // Store the audio level in d_bov for
   // header-extension-for-audio-level-indication.
   int32_t SetAudioLevel(uint8_t level_d_bov);
 
   RtpVideoCodecTypes VideoCodecType() const;
 
   uint32_t MaxConfiguredBitrateVideo() const;
 
   // FEC.
-  void SetGenericFECStatus(bool enable,
-                           uint8_t payload_type_red,
-                           uint8_t payload_type_fec);
-
-  void GenericFECStatus(bool* enable,
-                        uint8_t* payload_type_red,
-                        uint8_t* payload_type_fec) const;
+  void SetUlpfecConfig(bool enabled,
+                       int red_payload_type,
+                       int ulpfec_payload_type);
 
   int32_t SetFecParameters(const FecProtectionParams *delta_params,
                            const FecProtectionParams *key_params);
 
   RTC_DEPRECATED
   size_t SendPadData(size_t bytes,
                      bool timestamp_provided,
                      uint32_t timestamp,
                      int64_t capture_time_ms);
 
@@ skipping 60 matching lines @@
   const bool audio_configured_;
   const std::unique_ptr<RTPSenderAudio> audio_;
   const std::unique_ptr<RTPSenderVideo> video_;
 
   RtpPacketSender* const paced_sender_;
   TransportSequenceNumberAllocator* const transport_sequence_number_allocator_;
   TransportFeedbackObserver* const transport_feedback_observer_;
   int64_t last_capture_time_ms_sent_;
   rtc::CriticalSection send_critsect_;
 
-  Transport *transport_;
+  Transport* transport_;
   bool sending_media_ GUARDED_BY(send_critsect_);
 
   size_t max_payload_length_;
 
   int8_t payload_type_ GUARDED_BY(send_critsect_);
   std::map<int8_t, RtpUtility::Payload*> payload_type_map_;
 
   RtpHeaderExtensionMap rtp_header_extension_map_ GUARDED_BY(send_critsect_);
 
   // Tracks the current request for playout delay limits from application
@@ skipping 39 matching lines @@
   std::map<int8_t, int8_t> rtx_payload_type_map_ GUARDED_BY(send_critsect_);
 
   RateLimiter* const retransmission_rate_limiter_;
 
   RTC_DISALLOW_IMPLICIT_CONSTRUCTORS(RTPSender);
 };
 
 }  // namespace webrtc
 
 #endif  // WEBRTC_MODULES_RTP_RTCP_SOURCE_RTP_SENDER_H_