Update config names for WebRTC extended filter echo cancellation

content/renderer/media/media_stream_audio_processor.cc

 // Copyright 2013 The Chromium Authors. All rights reserved.
 // Use of this source code is governed by a BSD-style license that can be
 // found in the LICENSE file.
 
 #include "content/renderer/media/media_stream_audio_processor.h"
 
 #include "base/command_line.h"
 #include "base/metrics/field_trial.h"
 #include "base/metrics/histogram.h"
 #include "base/trace_event/trace_event.h"
@@ skipping 470 matching lines @@
   if (!echo_cancellation && !goog_experimental_aec && !goog_ns &&
       !goog_high_pass_filter && !goog_typing_detection &&
       !goog_agc && !goog_experimental_ns && !goog_beamforming) {
     RecordProcessingState(AUDIO_PROCESSING_DISABLED);
     return;
   }
 
   // Experimental options provided at creation.
   webrtc::Config config;
   if (goog_experimental_aec)
-    config.Set<webrtc::DelayCorrection>(new webrtc::DelayCorrection(true));
+    config.Set<webrtc::ExtendedFilter>(new webrtc::ExtendedFilter(true));
   if (goog_experimental_ns)
     config.Set<webrtc::ExperimentalNs>(new webrtc::ExperimentalNs(true));
   if (IsDelayAgnosticAecEnabled())
     config.Set<webrtc::ReportedDelay>(new webrtc::ReportedDelay(false));
   if (goog_beamforming) {
     ConfigureBeamforming(&config, audio_constraints.GetPropertyAsString(
         MediaAudioConstraints::kGoogArrayGeometry));
   }
   config.Set<webrtc::AudioProcessing48kHzSupport>(
       new webrtc::AudioProcessing48kHzSupport(audio_proc_48kHz_support_));
@@ skipping 235 matching lines @@
   if (echo_information_) {
     echo_information_.get()->UpdateAecDelayStats(ap->echo_cancellation());
   }
 
   // Return 0 if the volume hasn't been changed, and otherwise the new volume.
   return (agc->stream_analog_level() == volume) ?
       0 : agc->stream_analog_level();
 }
 
 }  // namespace content