Early initialize recording on the ADM from WebRtcVoiceMediaChannel.

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 85 matching lines @@
   // 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;
 
   void StartAecDump(const std::string& filename);
   int CreateVoEChannel();
+  webrtc::AudioDeviceModule* adm();
 
   rtc::ThreadChecker signal_thread_checker_;
   rtc::ThreadChecker worker_thread_checker_;
 
   // The primary instance of WebRtc VoiceEngine.
   std::unique_ptr<VoEWrapper> voe_wrapper_;
   rtc::scoped_refptr<webrtc::AudioState> audio_state_;
   // The external audio device manager
   webrtc::AudioDeviceModule* adm_ = nullptr;
   std::vector<AudioCodec> codecs_;
   std::vector<WebRtcVoiceMediaChannel*> channels_;
   webrtc::Config voe_config_;
   bool is_dumping_aec_ = false;
 
   webrtc::AgcConfig default_agc_config_;
   // 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);
+  RTC_DISALLOW_IMPLICIT_CONSTRUCTORS(WebRtcVoiceEngine);
 };
 
 // WebRtcVoiceMediaChannel is an implementation of VoiceMediaChannel that uses
 // WebRtc Voice Engine.
 class WebRtcVoiceMediaChannel final : public VoiceMediaChannel,
                                       public webrtc::Transport {
  public:
   WebRtcVoiceMediaChannel(WebRtcVoiceEngine* engine,
                           const MediaConfig& config,
                           const AudioOptions& options,
@@ skipping 78 matching lines @@
   bool ChangePlayout(bool playout);
   int CreateVoEChannel();
   bool DeleteVoEChannel(int channel);
   bool IsDefaultRecvStream(uint32_t ssrc) {
     return default_recv_ssrc_ == static_cast<int64_t>(ssrc);
   }
   bool SetSendBitrateInternal(int bps);
   bool HasSendCodec() const {
     return send_codec_spec_.codec_inst.pltype != -1;
   }
+  void SetupRecording();
 
   rtc::ThreadChecker worker_thread_checker_;
 
   WebRtcVoiceEngine* const engine_ = nullptr;
   std::vector<AudioCodec> recv_codecs_;
   bool send_bitrate_setting_ = false;
   int send_bitrate_bps_ = 0;
   AudioOptions options_;
   rtc::Optional<int> dtmf_payload_type_;
   bool desired_playout_ = false;
@@ skipping 36 matching lines @@
     int cng_payload_type = -1;
     int cng_plfreq = -1;
     webrtc::CodecInst codec_inst;
   } send_codec_spec_;
 
   RTC_DISALLOW_IMPLICIT_CONSTRUCTORS(WebRtcVoiceMediaChannel);
 };
 }  // namespace cricket
 
 #endif  // WEBRTC_MEDIA_ENGINE_WEBRTCVOICEENGINE_H_