Avoid that previous settings in APM are overwritten by WebRtcVoiceEngine

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 83 matching lines @@
 
   // 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
   // used.
   bool StartAecDump(rtc::PlatformFile file, int64_t max_size_bytes);
 
   // Stops AEC dump.
   void StopAecDump();
 
-  const webrtc::AudioProcessing::Config& GetApmConfigForTest() const {
-    return apm_config_;
+  const webrtc::AudioProcessing::Config GetApmConfigForTest() const {
+    return apm()->GetConfig();
   }
 
  private:
   // 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);
 
   // webrtc::TraceCallback:
   void Print(webrtc::TraceLevel level, const char* trace, int length) override;
 
   void StartAecDump(const std::string& filename);
   int CreateVoEChannel();
 
   std::unique_ptr<rtc::TaskQueue> low_priority_worker_queue_;
 
   webrtc::AudioDeviceModule* adm();
-  webrtc::AudioProcessing* apm();
+  webrtc::AudioProcessing* apm() const;
   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_;
@@ skipping 19 matching lines @@
   // 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_;
   rtc::Optional<bool> intelligibility_enhancer_;
   rtc::Optional<bool> level_control_;
 
-  webrtc::AudioProcessing::Config apm_config_;
-
   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,
@@ skipping 130 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_