Refactor handling of AudioOptions.

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 11 matching lines @@
  * OR BUSINESS INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF LIABILITY,
  * WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING NEGLIGENCE OR
  * OTHERWISE) ARISING IN ANY WAY OUT OF THE USE OF THIS SOFTWARE, EVEN IF
  * ADVISED OF THE POSSIBILITY OF SUCH DAMAGE.
  */
 
 #ifndef TALK_MEDIA_WEBRTCVOICEENGINE_H_
 #define TALK_MEDIA_WEBRTCVOICEENGINE_H_
 
 #include <map>
-#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/scoped_ptr.h"
@@ skipping 22 matching lines @@
   // Dependency injection for testing.
   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();
   RtpCapabilities GetCapabilities() const;
 
   // For tracking WebRtc channels. Needed because we have to pause them
   // all when switching devices.
   // May only be called by WebRtcVoiceMediaChannel.
@@ skipping 23 matching lines @@
   // Stops recording the RtcEventLog.
   void StopRtcEventLog();
 
  private:
   void Construct();
   bool InitInternal();
   // 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);
+  void SetDefaultDevices();
 
   // 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();
 
   rtc::ThreadChecker signal_thread_checker_;
   rtc::ThreadChecker worker_thread_checker_;
 
   // The primary instance of WebRtc VoiceEngine.
   rtc::scoped_ptr<VoEWrapper> voe_wrapper_;
   rtc::scoped_refptr<webrtc::AudioState> audio_state_;
   // The external audio device manager
   webrtc::AudioDeviceModule* adm_ = nullptr;
   bool is_dumping_aec_ = false;
   std::vector<AudioCodec> codecs_;
   std::vector<WebRtcVoiceMediaChannel*> channels_;
+  webrtc::Config voe_config_;
+  bool initialized_ = false;

[tommi, 2015/12/08 09:06:03]

nit: could group together with is_dumping_aec for slightly improved layout

[the sun, 2015/12/08 10:21:08]

Done.

+
   webrtc::AgcConfig default_agc_config_;
-
-  webrtc::Config voe_config_;
-
-  bool initialized_ = false;
-  AudioOptions options_;
-
   // 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.
   rtc::Optional<bool> extended_filter_aec_;
   rtc::Optional<bool> delay_agnostic_aec_;
   rtc::Optional<bool> experimental_ns_;
 
   RTC_DISALLOW_COPY_AND_ASSIGN(WebRtcVoiceEngine);
 };
@@ skipping 126 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_