Nuke SetSenderBufferingMode.

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 72 matching lines @@
   // 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 SetRTCPMode(const RtcpMode rtcp_mode);
   void SetProtectionMode(bool enable_nack,
                          bool enable_fec,
                          int payload_type_red,
                          int payload_type_fec);
   bool IsSendingFecEnabled();
-  int SetSenderBufferingMode(int target_delay_ms);
   int SetSendTimestampOffsetStatus(bool enable, int id);
   int SetSendAbsoluteSendTimeStatus(bool enable, int id);
   int SetSendVideoRotationStatus(bool enable, int id);
   int SetSendTransportSequenceNumber(bool enable, int id);
 
   // Sets SSRC for outgoing stream.
   int32_t SetSSRC(const uint32_t SSRC,
                   const StreamType usage,
                   const unsigned char simulcast_idx);
 
@@ skipping 265 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_