Remove SetVideoLogging/SetAudioLogging from ChannelManager and down the stack.

talk/media/webrtc/webrtcvoiceengine.h

 /*
  * libjingle
  * Copyright 2004 Google Inc.
  *
  * Redistribution and use in source and binary forms, with or without
  * modification, are permitted provided that the following conditions are met:
  *
  *  1. Redistributions of source code must retain the above copyright notice,
  *     this list of conditions and the following disclaimer.
  *  2. Redistributions in binary form must reproduce the above copyright notice,
@@ skipping 68 matching lines @@
   bool GetOutputVolume(int* level);
   bool SetOutputVolume(int level);
   int GetInputLevel();
 
   const std::vector<AudioCodec>& codecs();
   bool FindCodec(const AudioCodec& codec);
   bool FindWebRtcCodec(const AudioCodec& codec, webrtc::CodecInst* gcodec);
 
   const std::vector<RtpHeaderExtension>& rtp_header_extensions() const;
 
-  void SetLogging(int min_sev, const char* filter);
-
   // For tracking WebRtc channels. Needed because we have to pause them
   // all when switching devices.
   // May only be called by WebRtcVoiceMediaChannel.
   void RegisterChannel(WebRtcVoiceMediaChannel* channel);
   void UnregisterChannel(WebRtcVoiceMediaChannel* channel);
 
   // Called by WebRtcVoiceMediaChannel to set a gain offset from
   // the default AGC target level.
   bool AdjustAgcLevel(int delta);
 
@@ skipping 14 matching lines @@
   bool StartRtcEventLog(rtc::PlatformFile file);
 
   // Stops recording the RtcEventLog.
   void StopRtcEventLog();
 
  private:
   void Construct();
   void ConstructCodecs();
   bool GetVoeCodec(int index, webrtc::CodecInst* codec);
   bool InitInternal();
-  void SetTraceFilter(int filter);
-  void SetTraceOptions(const std::string& options);
+  void SetTraceFilter(int filter);                    // !!!!!!!!!! Remove?
+  void SetTraceOptions(const std::string& options);   // !!!!!!!!!! Remove?

[pthatcher1, 2015/11/18 19:26:26]

If these only have an effect when SetLogging is called and SetLogging is going
away, then yes, they should go away.

[the sun, 2015/11/19 15:55:12]

Done.

   // Every option that is "set" will be applied. Every option not "set" will be
   // ignored. This allows us to selectively turn on and off different options
   // easily at any time.
   bool ApplyOptions(const AudioOptions& options);
 
   // webrtc::TraceCallback:
   void Print(webrtc::TraceLevel level, const char* trace, int length) override;
 
   // Given the device type, name, and id, find device id. Return true and
   // set the output parameter rtc_id if successful.
   bool FindWebRtcAudioDeviceId(
       bool is_input, const std::string& dev_name, int dev_id, int* rtc_id);
 
   void StartAecDump(const std::string& filename);
   int CreateVoEChannel();
 
   static const int kDefaultLogSeverity = rtc::LS_WARNING;
 
   rtc::ThreadChecker signal_thread_checker_;
   rtc::ThreadChecker worker_thread_checker_;
 
   // The primary instance of WebRtc VoiceEngine.
   rtc::scoped_ptr<VoEWrapper> voe_wrapper_;
   rtc::scoped_ptr<VoETraceWrapper> tracing_;
   rtc::scoped_refptr<webrtc::AudioState> audio_state_;
   // The external audio device manager
   webrtc::AudioDeviceModule* adm_ = nullptr;
   int log_filter_;
-  std::string log_options_;
+  std::string log_options_;           // !!!!!!!!!!! Remove?

[pthatcher1, 2015/11/18 19:26:26]

If SetLogging is the only thing that sets it, and SetLogging is going away, then
yes, it should go away as well.

[the sun, 2015/11/19 15:55:12]

Yes, as well as some other stuff.

   bool is_dumping_aec_ = false;
   std::vector<AudioCodec> codecs_;
   std::vector<RtpHeaderExtension> rtp_header_extensions_;
   std::vector<WebRtcVoiceMediaChannel*> channels_;
   webrtc::AgcConfig default_agc_config_;
 
   webrtc::Config voe_config_;
 
   bool initialized_ = false;
   AudioOptions options_;
@@ skipping 161 matching lines @@
   // the WebRtc thread must be synchronized with edits on the worker thread.
   // Reads on the worker thread are ok.
   std::vector<RtpHeaderExtension> receive_extensions_;
   std::vector<webrtc::RtpExtension> recv_rtp_extensions_;
 
   RTC_DISALLOW_IMPLICIT_CONSTRUCTORS(WebRtcVoiceMediaChannel);
 };
 }  // namespace cricket
 
 #endif  // TALK_MEDIA_WEBRTCVOICEENGINE_H_