Simplify handling of options in WebRtcVoiceMediaEngine.

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 74 matching lines @@
   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);
+  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);
 
   VoEWrapper* voe() { return voe_wrapper_.get(); }
   int GetLastEngineError();
 
   // Set the external ADM. This can only be called before Init.
   bool SetAudioDeviceModule(webrtc::AudioDeviceModule* adm);
 
   // Starts AEC dump using existing file.
   bool StartAecDump(rtc::PlatformFile file);
 
   // Create a VoiceEngine Channel.
   int CreateMediaVoiceChannel();
 
  private:
-  typedef std::vector<WebRtcVoiceMediaChannel*> ChannelList;
-
   void Construct();
   void ConstructCodecs();
   bool GetVoeCodec(int index, webrtc::CodecInst* codec);
   bool InitInternal();
   void SetTraceFilter(int filter);
   void SetTraceOptions(const std::string& options);
-  // Applies either options or overrides.  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.
+  // 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);
-  // Overrides, when set, take precedence over the options on a
-  // per-option basis.  For example, if AGC is set in options and AEC
-  // is set in overrides, AGC and AEC will be both be set.  Overrides
-  // can also turn off options.  For example, if AGC is set to "on" in
-  // options and AGC is set to "off" in overrides, the result is that
-  // AGC will be off until different overrides are applied or until
-  // the overrides are cleared.  Only one set of overrides is present
-  // at a time (they do not "stack").  And when the overrides are
-  // cleared, the media engine's state reverts back to the options set
-  // via SetOptions.  This allows us to have both "persistent options"
-  // (the normal options) and "temporary options" (overrides).
-  bool SetOptionOverrides(const AudioOptions& options);
-  bool ClearOptionOverrides();
 
   // webrtc::TraceCallback:
   void Print(webrtc::TraceLevel level, const char* trace, int length) override;
 
   // webrtc::VoiceEngineObserver:
   void CallbackOnError(int channel, int errCode) override;
 
   // Given the device type, name, and id, find device id. Return true and
   // set the output parameter rtc_id if successful.
   bool FindWebRtcAudioDeviceId(
@@ skipping 11 matching lines @@
   // 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_;
-  ChannelList channels_;
+  std::vector<WebRtcVoiceMediaChannel*> 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
-  // difference between options and overrides.
-  // options_ are the base options, which combined with the
-  // option_overrides_, create the current options being used.
-  // options_ is stored so that when option_overrides_ is cleared, we
-  // can restore the options_ without the option_overrides.
   AudioOptions options_;
-  AudioOptions option_overrides_;
 
   // Cache received extended_filter_aec, delay_agnostic_aec and experimental_ns
   // values, and apply them in case they are missing in the audio options. We
   // need to do this because SetExtraOptions() will revert to defaults for
   // options which are not provided.
   Settable<bool> extended_filter_aec_;
   Settable<bool> delay_agnostic_aec_;
   Settable<bool> experimental_ns_;
 };
 
@@ skipping 173 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_