Major AEC3 render pipeline changes

webrtc/modules/audio_processing/aec3/aec_state.cc

 /*
  *  Copyright (c) 2017 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 @@
       active_render_counter_(kActiveRenderCounterInitial) {
   bands_with_reliable_filter_.fill(false);
   filter_estimate_strength_.fill(0.f);
 }
 
 AecState::~AecState() = default;
 
 void AecState::Update(const std::vector<std::array<float, kFftLengthBy2Plus1>>&
                           filter_frequency_response,
                       const rtc::Optional<size_t>& external_delay_samples,
-                      const FftBuffer& X_buffer,
+                      const RenderBuffer& X_buffer,
                       const std::array<float, kFftLengthBy2Plus1>& E2_main,
                       const std::array<float, kFftLengthBy2Plus1>& E2_shadow,
                       const std::array<float, kFftLengthBy2Plus1>& Y2,
                       rtc::ArrayView<const float> x,
                       const EchoPathVariability& echo_path_variability,
                       bool echo_leakage_detected) {
   filter_length_ = filter_frequency_response.size();
   AnalyzeFilter(filter_frequency_response, &bands_with_reliable_filter_,
                 &filter_estimate_strength_, &filter_delay_);
   // Compute the externally provided delay in partitions. The truncation is
@@ skipping 41 matching lines @@
         [](float a) { return a <= 10.f; });
     headset_detected_ = low_erl_band_count > 20 && noisy_band_count > 20;
 #endif
     headset_detected_ = false;
   } else {
     headset_detected_ = false;
   }
 }
 
 }  // namespace webrtc