Simplify registration of RTP-header extensions.

webrtc/video/vie_channel.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 70 matching lines @@
   int32_t Init();
 
   // Sets the encoder to use for the channel. |new_stream| indicates the encoder
   // type has changed and we should start a new RTP stream.
   int32_t SetSendCodec(const VideoCodec& video_codec, bool new_stream = true);
 
   void SetProtectionMode(bool enable_nack,
                          bool enable_fec,
                          int payload_type_red,
                          int payload_type_fec);
-  int EnableSendTimestampOffset(int id);
-  int EnableSendAbsoluteSendTime(int id);
-  int EnableSendVideoRotation(int id);
-  int EnableSendTransportSequenceNumber(int id);
 
   RtpState GetRtpStateForSsrc(uint32_t ssrc) const;
 
   // Gets send statistics for the rtp and rtx stream.
   void GetSendStreamDataCounters(StreamDataCounters* rtp_counters,
                                  StreamDataCounters* rtx_counters) const;
 
   void RegisterSendSideDelayObserver(SendSideDelayObserver* observer);
 
   // Called on any new send bitrate estimate.
@@ skipping 211 matching lines @@
   int64_t last_rtt_ms_ GUARDED_BY(crit_);
 
   // RtpRtcp modules, declared last as they use other members on construction.
   const std::vector<RtpRtcp*> rtp_rtcp_modules_;
   size_t num_active_rtp_rtcp_modules_ GUARDED_BY(crit_);
 };
 
 }  // namespace webrtc
 
 #endif  // WEBRTC_VIDEO_VIE_CHANNEL_H_