Remove the SetLocalMonitor() API.

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 35 matching lines @@
 #include "webrtc/call.h"
 #include "webrtc/common.h"
 #include "webrtc/config.h"
 
 namespace webrtc {
 class VideoEngine;
 }
 
 namespace cricket {
 
-// WebRtcMonitorStream is used to monitor a stream coming from WebRtc.
-// For now we just dump the data.
-class WebRtcMonitorStream : public webrtc::OutStream {
-  bool Write(const void* buf, size_t len) override { return true; }
-};
-
 class AudioDeviceModule;
 class AudioRenderer;
 class VoETraceWrapper;
 class VoEWrapper;
 class VoiceProcessor;
 class WebRtcVoiceMediaChannel;
 
 // WebRtcVoiceEngine is a class to be used with CompositeMediaEngine.
 // It uses the WebRtc VoiceEngine library for audio handling.
 class WebRtcVoiceEngine
@@ skipping 15 matching lines @@
   VoiceMediaChannel* CreateChannel(webrtc::Call* call,
                                    const AudioOptions& options);
 
   AudioOptions GetOptions() const { return options_; }
   bool SetOptions(const AudioOptions& options);
   bool SetDelayOffset(int offset);
   bool SetDevices(const Device* in_device, const Device* out_device);
   bool GetOutputVolume(int* level);
   bool SetOutputVolume(int level);
   int GetInputLevel();
-  bool SetLocalMonitor(bool enable);
 
   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);
 
   bool RegisterProcessor(uint32 ssrc,
@@ skipping 75 matching lines @@
   // 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);
   bool FindChannelAndSsrc(int channel_num,
                           WebRtcVoiceMediaChannel** channel,
                           uint32* ssrc) const;
   bool FindChannelNumFromSsrc(uint32 ssrc,
                               MediaProcessorDirection direction,
                               int* channel_num);
-  bool ChangeLocalMonitor(bool enable);
-  bool PauseLocalMonitor();
-  bool ResumeLocalMonitor();
 
   bool UnregisterProcessorChannel(MediaProcessorDirection channel_direction,
                                   uint32 ssrc,
                                   VoiceProcessor* voice_processor,
                                   MediaProcessorDirection processor_direction);
 
   void StartAecDump(const std::string& filename);
   void StopAecDump();
   int CreateVoiceChannel(VoEWrapper* voe);
 
   // When a voice processor registers with the engine, it is connected
   // to either the Rx or Tx signals, based on the direction parameter.
   // SignalXXMediaFrame will be invoked for every audio packet.
   FrameSignal SignalRxMediaFrame;
   FrameSignal SignalTxMediaFrame;
 
   static const int kDefaultLogSeverity = rtc::LS_WARNING;
 
   // The primary instance of WebRtc VoiceEngine.
   rtc::scoped_ptr<VoEWrapper> voe_wrapper_;
   rtc::scoped_ptr<VoETraceWrapper> tracing_;
   // The external audio device manager
   webrtc::AudioDeviceModule* adm_;
   int log_filter_;
   std::string log_options_;
   bool is_dumping_aec_;
   std::vector<AudioCodec> codecs_;
   std::vector<RtpHeaderExtension> rtp_header_extensions_;
-  bool desired_local_monitor_enable_;
-  rtc::scoped_ptr<WebRtcMonitorStream> monitor_;
   ChannelList channels_;
   // channels_ can be read from WebRtc callback thread. We need a lock on that
   // callback as well as the RegisterChannel/UnregisterChannel.
   rtc::CriticalSection channels_cs_;
   webrtc::AgcConfig default_agc_config_;
 
   webrtc::Config voe_config_;
 
   bool initialized_;
   // See SetOptions and SetOptionOverrides for a description of the
@@ skipping 200 matching lines @@
   std::vector<webrtc::RtpExtension> recv_rtp_extensions_;
 
   // Do not lock this on the VoE media processor thread; potential for deadlock
   // exists.
   mutable rtc::CriticalSection receive_channels_cs_;
 };
 
 }  // namespace cricket
 
 #endif  // TALK_MEDIA_WEBRTCVOICEENGINE_H_