| Index: talk/media/webrtc/webrtcvoiceengine.h
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| diff --git a/talk/media/webrtc/webrtcvoiceengine.h b/talk/media/webrtc/webrtcvoiceengine.h
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| index 3222861b21e7c61b5a087b59076396e7942896ad..1de4fb972af70a9453d198ffec5fc45c2e1ed88c 100644
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| --- a/talk/media/webrtc/webrtcvoiceengine.h
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| +++ b/talk/media/webrtc/webrtcvoiceengine.h
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| @@ -29,7 +29,6 @@
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|  #define TALK_MEDIA_WEBRTCVOICEENGINE_H_
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|  
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|  #include <map>
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| -#include <set>
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|  #include <string>
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|  #include <vector>
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|  
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| @@ -72,9 +71,6 @@ class WebRtcVoiceEngine final : public webrtc::TraceCallback  {
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|    VoiceMediaChannel* CreateChannel(webrtc::Call* call,
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|                                     const AudioOptions& options);
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|  
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| -  AudioOptions GetOptions() const { return options_; }
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| -  bool SetOptions(const AudioOptions& options);
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| -  bool SetDevices(const Device* in_device, const Device* out_device);
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|    bool GetOutputVolume(int* level);
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|    bool SetOutputVolume(int level);
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|    int GetInputLevel();
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| @@ -118,15 +114,11 @@ class WebRtcVoiceEngine final : public webrtc::TraceCallback  {
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|    // ignored. This allows us to selectively turn on and off different options
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|    // easily at any time.
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|    bool ApplyOptions(const AudioOptions& options);
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| +  void SetDefaultDevices();
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|  
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|    // webrtc::TraceCallback:
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|    void Print(webrtc::TraceLevel level, const char* trace, int length) override;
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|  
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| -  // Given the device type, name, and id, find device id. Return true and
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| -  // set the output parameter rtc_id if successful.
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| -  bool FindWebRtcAudioDeviceId(
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| -      bool is_input, const std::string& dev_name, int dev_id, int* rtc_id);
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| -
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|    void StartAecDump(const std::string& filename);
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|    int CreateVoEChannel();
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|  
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| @@ -138,16 +130,13 @@ class WebRtcVoiceEngine final : public webrtc::TraceCallback  {
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|    rtc::scoped_refptr<webrtc::AudioState> audio_state_;
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|    // The external audio device manager
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|    webrtc::AudioDeviceModule* adm_ = nullptr;
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| -  bool is_dumping_aec_ = false;
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|    std::vector<AudioCodec> codecs_;
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|    std::vector<WebRtcVoiceMediaChannel*> channels_;
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| -  webrtc::AgcConfig default_agc_config_;
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| -
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|    webrtc::Config voe_config_;
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| -
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|    bool initialized_ = false;
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| -  AudioOptions options_;
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| +  bool is_dumping_aec_ = false;
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|  
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| +  webrtc::AgcConfig default_agc_config_;
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|    // Cache received extended_filter_aec, delay_agnostic_aec and experimental_ns
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|    // values, and apply them in case they are missing in the audio options. We
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|    // need to do this because SetExtraOptions() will revert to defaults for
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| 
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