Remove WVoE::SetAudioDeviceModule() - it is now set in ctor.

webrtc/media/engine/webrtcvoiceengine.h

Index: webrtc/media/engine/webrtcvoiceengine.h
diff --git a/webrtc/media/engine/webrtcvoiceengine.h b/webrtc/media/engine/webrtcvoiceengine.h
index 056b0780ef16b1af9dd18ffdacec73d79267205d..247669798416439e8d23b4e223b8c818e7392d80 100644
--- a/webrtc/media/engine/webrtcvoiceengine.h
+++ b/webrtc/media/engine/webrtcvoiceengine.h
@@ -43,11 +43,11 @@ class WebRtcVoiceEngine final : public webrtc::TraceCallback  {
   // Exposed for the WVoE/MC unit test.
   static bool ToCodecInst(const AudioCodec& in, webrtc::CodecInst* out);
 
-  WebRtcVoiceEngine();
+  explicit WebRtcVoiceEngine(webrtc::AudioDeviceModule* adm);
   // Dependency injection for testing.
-  explicit WebRtcVoiceEngine(VoEWrapper* voe_wrapper);
-  ~WebRtcVoiceEngine();
-  bool Init(rtc::Thread* worker_thread);
+  WebRtcVoiceEngine(webrtc::AudioDeviceModule* adm, VoEWrapper* voe_wrapper);
+  ~WebRtcVoiceEngine() override;
+  bool Init();
   void Terminate();
 
   rtc::scoped_refptr<webrtc::AudioState> GetAudioState() const;
@@ -75,9 +75,6 @@ class WebRtcVoiceEngine final : public webrtc::TraceCallback  {
   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 an existing file. A maximum file size in bytes can be
   // specified. When the maximum file size is reached, logging is stopped and
   // the file is closed. If max_size_bytes is set to <= 0, no limit will be
@@ -95,8 +92,6 @@ class WebRtcVoiceEngine final : public webrtc::TraceCallback  {
   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.
@@ -120,7 +115,6 @@ class WebRtcVoiceEngine final : public webrtc::TraceCallback  {
   std::vector<AudioCodec> codecs_;
   std::vector<WebRtcVoiceMediaChannel*> channels_;
   webrtc::Config voe_config_;
-  bool initialized_ = false;
   bool is_dumping_aec_ = false;
 
   webrtc::AgcConfig default_agc_config_;
@@ -156,8 +150,6 @@ class WebRtcVoiceMediaChannel final : public VoiceMediaChannel,
   bool PausePlayout();
   bool ResumePlayout();
   void SetSend(bool send) override;
-  bool PauseSend();
-  bool ResumeSend();
   bool SetAudioSend(uint32_t ssrc,
                     bool enable,
                     const AudioOptions* options,