Allow WebRtcMediaEngine to be created from any thread.

webrtc/media/engine/webrtcvoiceengine.h

 /*
  *  Copyright (c) 2004 The WebRTC project authors. All Rights Reserved.
  *
  *  Use of this source code is governed by a BSD-style license
  *  that can be found in the LICENSE file in the root of the source
  *  tree. An additional intellectual property rights grant can be found
  *  in the file PATENTS.  All contributing project authors may
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
@@ skipping 49 matching lines @@
       rtc::scoped_refptr<webrtc::AudioMixer> audio_mixer);
   // Dependency injection for testing.
   WebRtcVoiceEngine(
       webrtc::AudioDeviceModule* adm,
       const rtc::scoped_refptr<webrtc::AudioEncoderFactory>& encoder_factory,
       const rtc::scoped_refptr<webrtc::AudioDecoderFactory>& decoder_factory,
       rtc::scoped_refptr<webrtc::AudioMixer> audio_mixer,
       VoEWrapper* voe_wrapper);
   ~WebRtcVoiceEngine() override;
 
+  // Does initialization that needs to occur on the worker thread.
+  void Init();
+
   rtc::scoped_refptr<webrtc::AudioState> GetAudioState() const;
   VoiceMediaChannel* CreateChannel(webrtc::Call* call,
                                    const MediaConfig& config,
                                    const AudioOptions& options);
 
   int GetInputLevel();
 
   const std::vector<AudioCodec>& send_codecs() const;
   const std::vector<AudioCodec>& recv_codecs() const;
   RtpCapabilities GetCapabilities() const;
@@ skipping 25 matching lines @@
   // 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;
 
   void StartAecDump(const std::string& filename);
   int CreateVoEChannel();
 
-  rtc::TaskQueue low_priority_worker_queue_;
+  std::unique_ptr<rtc::TaskQueue> low_priority_worker_queue_;
 
   webrtc::AudioDeviceModule* adm();
   webrtc::AudioProcessing* apm();
   webrtc::voe::TransmitMixer* transmit_mixer();
 
   AudioCodecs CollectCodecs(
       const std::vector<webrtc::AudioCodecSpec>& specs) const;
 
   rtc::ThreadChecker signal_thread_checker_;
   rtc::ThreadChecker worker_thread_checker_;
 
   // The audio device manager.
   rtc::scoped_refptr<webrtc::AudioDeviceModule> adm_;
   rtc::scoped_refptr<webrtc::AudioEncoderFactory> encoder_factory_;
   rtc::scoped_refptr<webrtc::AudioDecoderFactory> decoder_factory_;
+  rtc::scoped_refptr<webrtc::AudioMixer> audio_mixer_;
   // Reference to the APM, owned by VoE.
   webrtc::AudioProcessing* apm_ = nullptr;
   // Reference to the TransmitMixer, owned by VoE.
   webrtc::voe::TransmitMixer* transmit_mixer_ = nullptr;
   // The primary instance of WebRtc VoiceEngine.
   std::unique_ptr<VoEWrapper> voe_wrapper_;
   rtc::scoped_refptr<webrtc::AudioState> audio_state_;
   std::vector<AudioCodec> send_codecs_;
   std::vector<AudioCodec> recv_codecs_;
   std::vector<WebRtcVoiceMediaChannel*> channels_;
   webrtc::VoEBase::ChannelConfig channel_config_;
   bool is_dumping_aec_ = false;
+  bool initialized_ = false;
 
   webrtc::AgcConfig default_agc_config_;
   // Cache received extended_filter_aec, delay_agnostic_aec, experimental_ns
   // level controller, and intelligibility_enhancer 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_;
@@ skipping 145 matching lines @@
   std::vector<webrtc::RtpExtension> recv_rtp_extensions_;
 
   rtc::Optional<webrtc::AudioSendStream::Config::SendCodecSpec>
       send_codec_spec_;
 
   RTC_DISALLOW_IMPLICIT_CONSTRUCTORS(WebRtcVoiceMediaChannel);
 };
 }  // namespace cricket
 
 #endif  // WEBRTC_MEDIA_ENGINE_WEBRTCVOICEENGINE_H_