Don't use default aec_dump in webrtcvoiceengine when non is provided

talk/media/webrtc/webrtcvoiceengine.cc

 /*
  * libjingle
  * Copyright 2004 Google Inc.
  *
  * Redistribution and use in source and binary forms, with or without
  * modification, are permitted provided that the following conditions are met:
  *
  *  1. Redistributions of source code must retain the above copyright notice,
  *     this list of conditions and the following disclaimer.
  *  2. Redistributions in binary form must reproduce the above copyright notice,
@@ skipping 567 matching lines @@
   options.highpass_filter = rtc::Optional<bool>(true);
   options.stereo_swapping = rtc::Optional<bool>(false);
   options.audio_jitter_buffer_max_packets = rtc::Optional<int>(50);
   options.audio_jitter_buffer_fast_accelerate = rtc::Optional<bool>(false);
   options.typing_detection = rtc::Optional<bool>(true);
   options.adjust_agc_delta = rtc::Optional<int>(0);
   options.experimental_agc = rtc::Optional<bool>(false);
   options.extended_filter_aec = rtc::Optional<bool>(false);
   options.delay_agnostic_aec = rtc::Optional<bool>(false);
   options.experimental_ns = rtc::Optional<bool>(false);
-  options.aec_dump = rtc::Optional<bool>(false);
 
   // Apply any given options on top.
   options.SetAll(options_in);
 
   // kEcConference is AEC with high suppression.
   webrtc::EcModes ec_mode = webrtc::kEcConference;
   webrtc::AecmModes aecm_mode = webrtc::kAecmSpeakerphone;
   webrtc::AgcModes agc_mode = webrtc::kAgcAdaptiveAnalog;
   webrtc::NsModes ns_mode = webrtc::kNsHighSuppression;
   if (options.aecm_generate_comfort_noise) {
@@ skipping 1901 matching lines @@
     }
   } else {
     LOG(LS_INFO) << "Stopping playout for channel #" << channel;
     engine()->voe()->base()->StopPlayout(channel);
   }
   return true;
 }
 }  // namespace cricket
 
 #endif  // HAVE_WEBRTC_VOICE