Moved creation of AudioDecoderFactory to inside PeerConnectionFactory.

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 28 matching lines @@
 class WebRtcVoiceMediaChannel;
 
 // WebRtcVoiceEngine is a class to be used with CompositeMediaEngine.
 // It uses the WebRtc VoiceEngine library for audio handling.
 class WebRtcVoiceEngine final : public webrtc::TraceCallback  {
   friend class WebRtcVoiceMediaChannel;
  public:
   // Exposed for the WVoE/MC unit test.
   static bool ToCodecInst(const AudioCodec& in, webrtc::CodecInst* out);
 
-  explicit WebRtcVoiceEngine(webrtc::AudioDeviceModule* adm);
+  WebRtcVoiceEngine(
+      webrtc::AudioDeviceModule* adm,
+      const rtc::scoped_refptr<webrtc::AudioDecoderFactory>& decoder_factory);
   // Dependency injection for testing.
-  WebRtcVoiceEngine(webrtc::AudioDeviceModule* adm, VoEWrapper* voe_wrapper);
+  WebRtcVoiceEngine(
+      webrtc::AudioDeviceModule* adm,
+      const rtc::scoped_refptr<webrtc::AudioDecoderFactory>& decoder_factory,
+      VoEWrapper* voe_wrapper);
   ~WebRtcVoiceEngine() override;
 
   rtc::scoped_refptr<webrtc::AudioState> GetAudioState() const;
   VoiceMediaChannel* CreateChannel(webrtc::Call* call,
                                    const MediaConfig& config,
                                    const AudioOptions& options);
 
   bool GetOutputVolume(int* level);
   bool SetOutputVolume(int level);
   int GetInputLevel();
@@ skipping 43 matching lines @@
 
   void StartAecDump(const std::string& filename);
   int CreateVoEChannel();
   webrtc::AudioDeviceModule* adm();
 
   rtc::ThreadChecker signal_thread_checker_;
   rtc::ThreadChecker worker_thread_checker_;
 
   // The audio device manager.
   rtc::scoped_refptr<webrtc::AudioDeviceModule> adm_;
+  rtc::scoped_refptr<webrtc::AudioDecoderFactory> decoder_factory_;
   // The primary instance of WebRtc VoiceEngine.
   std::unique_ptr<VoEWrapper> voe_wrapper_;
   rtc::scoped_refptr<webrtc::AudioState> audio_state_;
   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, experimental_ns and
@@ skipping 167 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_