Remove VCMRenderBufferSizeCallback.

webrtc/modules/video_coding/jitter_buffer.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 194 matching lines @@
 
   // Returns a list of the sequence numbers currently missing.
   std::vector<uint16_t> GetNackList(bool* request_key_frame);
 
   // Set decode error mode - Should not be changed in the middle of the
   // session. Changes will not influence frames already in the buffer.
   void SetDecodeErrorMode(VCMDecodeErrorMode error_mode);
   int64_t LastDecodedTimestamp() const;
   VCMDecodeErrorMode decode_error_mode() const { return decode_error_mode_; }
 
-  // Used to compute time of complete continuous frames. Returns the timestamps
-  // corresponding to the start and end of the continuous complete buffer.
-  void RenderBufferSize(uint32_t* timestamp_start, uint32_t* timestamp_end);
-
   void RegisterStatsCallback(VCMReceiveStatisticsCallback* callback);
 
   int64_t TimeUntilNextProcess();
   void Process();
 
  private:
   class SequenceNumberLessThan {
    public:
     bool operator()(const uint16_t& sequence_number1,
                     const uint16_t& sequence_number2) const {
@@ skipping 163 matching lines @@
   // frames.
   int frame_counter_;
 
   std::unique_ptr<NackModule> nack_module_;
 
   RTC_DISALLOW_COPY_AND_ASSIGN(VCMJitterBuffer);
 };
 }  // namespace webrtc
 
 #endif  // WEBRTC_MODULES_VIDEO_CODING_JITTER_BUFFER_H_