Unify FillAudioEncoderTimeSeries and Pointwise.

webrtc/tools/event_log_visualizer/analyzer.h

 /*
  *  Copyright (c) 2016 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 122 matching lines @@
       const std::string& label_prefix);
 
   bool IsRtxSsrc(StreamId stream_id) const;
 
   bool IsVideoSsrc(StreamId stream_id) const;
 
   bool IsAudioSsrc(StreamId stream_id) const;
 
   std::string GetStreamName(StreamId) const;
 
-  void FillAudioEncoderTimeSeries(
-      Plot* plot,
-      rtc::FunctionView<rtc::Optional<float>(
-          const AudioNetworkAdaptationEvent& ana_event)> get_y) const;
-
   const ParsedRtcEventLog& parsed_log_;
 
   // A list of SSRCs we are interested in analysing.
   // If left empty, all SSRCs will be considered relevant.
   std::vector<uint32_t> desired_ssrc_;
 
   // Tracks what each stream is configured for. Note that a single SSRC can be
   // in several sets. For example, the SSRC used for sending video over RTX
   // will appear in both video_ssrcs_ and rtx_ssrcs_. In the unlikely case that
   // an SSRC is reconfigured to a different media type mid-call, it will also
@@ skipping 26 matching lines @@
   uint64_t end_time_;
 
   // Duration (in seconds) of log file.
   float call_duration_s_;
 };
 
 }  // namespace plotting
 }  // namespace webrtc
 
 #endif  // WEBRTC_TOOLS_EVENT_LOG_VISUALIZER_ANALYZER_H_