Major AEC3 render pipeline changes

webrtc/modules/audio_processing/aec3/render_delay_controller.h

 /*
  *  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.
  */
 
 #ifndef WEBRTC_MODULES_AUDIO_PROCESSING_AEC3_RENDER_DELAY_CONTROLLER_H_
 #define WEBRTC_MODULES_AUDIO_PROCESSING_AEC3_RENDER_DELAY_CONTROLLER_H_
 
 #include "webrtc/base/array_view.h"
 #include "webrtc/base/optional.h"
+#include "webrtc/modules/audio_processing/aec3/downsampled_render_buffer.h"
 #include "webrtc/modules/audio_processing/aec3/render_delay_buffer.h"
 #include "webrtc/modules/audio_processing/logging/apm_data_dumper.h"
 
 namespace webrtc {
 
 // Class for aligning the render and capture signal using a RenderDelayBuffer.
 class RenderDelayController {
  public:
-  static RenderDelayController* Create(
-      int sample_rate_hz,
-      const RenderDelayBuffer& render_delay_buffer);
+  static RenderDelayController* Create(int sample_rate_hz);
   virtual ~RenderDelayController() = default;
 
+  // Resets the delay controller.
+  virtual void Reset() = 0;
+
+  // Receives the externally used delay.
+  virtual void SetDelay(size_t render_delay) = 0;
+
   // Aligns the render buffer content with the capture signal.
-  virtual size_t GetDelay(rtc::ArrayView<const float> capture) = 0;
-
-  // Analyzes the render signal and returns false if there is a buffer overrun.
-  virtual bool AnalyzeRender(rtc::ArrayView<const float> render) = 0;
+  virtual size_t GetDelay(const DownsampledRenderBuffer& render_buffer,
+                          rtc::ArrayView<const float> capture) = 0;
 
   // Returns an approximate value for the headroom in the buffer alignment.
   virtual rtc::Optional<size_t> AlignmentHeadroomSamples() const = 0;
 };
 }  // namespace webrtc
 
 #endif  // WEBRTC_MODULES_AUDIO_PROCESSING_AEC3_RENDER_DELAY_CONTROLLER_H_