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 21 matching lines @@
 #include <set>
 #include <string>
 #include <vector>
 
 #include "talk/media/base/rtputils.h"
 #include "talk/media/webrtc/webrtccommon.h"
 #include "talk/media/webrtc/webrtcvoe.h"
 #include "talk/session/media/channel.h"
 #include "webrtc/audio_state.h"
 #include "webrtc/base/buffer.h"
-#include "webrtc/base/byteorder.h"
-#include "webrtc/base/logging.h"
 #include "webrtc/base/scoped_ptr.h"
 #include "webrtc/base/stream.h"
 #include "webrtc/base/thread_checker.h"
 #include "webrtc/call.h"
 #include "webrtc/common.h"
 #include "webrtc/config.h"
 
 namespace cricket {
 
 class AudioDeviceModule;
 class AudioRenderer;
-class VoETraceWrapper;
 class VoEWrapper;
 class WebRtcVoiceMediaChannel;
 
 // WebRtcVoiceEngine is a class to be used with CompositeMediaEngine.
 // It uses the WebRtc VoiceEngine library for audio handling.
 class WebRtcVoiceEngine final : public webrtc::TraceCallback  {
   friend class WebRtcVoiceMediaChannel;
 
  public:
   WebRtcVoiceEngine();
   // Dependency injection for testing.
-  WebRtcVoiceEngine(VoEWrapper* voe_wrapper, VoETraceWrapper* tracing);
+  explicit WebRtcVoiceEngine(VoEWrapper* voe_wrapper);
   ~WebRtcVoiceEngine();
   bool Init(rtc::Thread* worker_thread);
   void Terminate();
 
   rtc::scoped_refptr<webrtc::AudioState> GetAudioState() const;
   VoiceMediaChannel* CreateChannel(webrtc::Call* call,
                                    const AudioOptions& options);
 
   AudioOptions GetOptions() const { return options_; }
   bool SetOptions(const AudioOptions& options);
   bool SetDevices(const Device* in_device, const Device* out_device);
   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);
   // 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_;
   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 138 matching lines @@
 
   class WebRtcAudioReceiveStream;
   std::map<uint32_t, WebRtcAudioReceiveStream*> recv_streams_;
   std::vector<webrtc::RtpExtension> recv_rtp_extensions_;
 
   RTC_DISALLOW_IMPLICIT_CONSTRUCTORS(WebRtcVoiceMediaChannel);
 };
 }  // namespace cricket
 
 #endif  // TALK_MEDIA_WEBRTCVOICEENGINE_H_