Major AEC3 render pipeline changes

webrtc/modules/audio_processing/aec3/mock/mock_render_delay_buffer.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_MOCK_MOCK_RENDER_DELAY_BUFFER_H_
 #define WEBRTC_MODULES_AUDIO_PROCESSING_AEC3_MOCK_MOCK_RENDER_DELAY_BUFFER_H_
 
 #include <vector>
 
 #include "webrtc/modules/audio_processing/aec3/aec3_common.h"
+#include "webrtc/modules/audio_processing/aec3/downsampled_render_buffer.h"
+#include "webrtc/modules/audio_processing/aec3/render_buffer.h"
 #include "webrtc/modules/audio_processing/aec3/render_delay_buffer.h"
 #include "webrtc/test/gmock.h"
 
 namespace webrtc {
 namespace test {
 
 class MockRenderDelayBuffer : public RenderDelayBuffer {
  public:
   explicit MockRenderDelayBuffer(int sample_rate_hz)
-      : block_(std::vector<std::vector<float>>(
-            NumBandsForRate(sample_rate_hz),
-            std::vector<float>(kBlockSize, 0.f))) {
-    ON_CALL(*this, GetNext())
+      : render_buffer_(Aec3Optimization::kNone,
+                       NumBandsForRate(sample_rate_hz),
+                       kRenderDelayBufferSize,
+                       std::vector<size_t>(1, kAdaptiveFilterLength)) {
+    ON_CALL(*this, GetRenderBuffer())
         .WillByDefault(
-            testing::Invoke(this, &MockRenderDelayBuffer::FakeGetNext));
+            testing::Invoke(this, &MockRenderDelayBuffer::FakeGetRenderBuffer));
+    ON_CALL(*this, GetDownsampledRenderBuffer())
+        .WillByDefault(testing::Invoke(
+            this, &MockRenderDelayBuffer::FakeGetDownsampledRenderBuffer));
   }
   virtual ~MockRenderDelayBuffer() = default;
 
+  MOCK_METHOD0(Reset, void());
   MOCK_METHOD1(Insert, bool(std::vector<std::vector<float>>* block));
-  MOCK_METHOD0(GetNext, const std::vector<std::vector<float>>&());
+  MOCK_METHOD0(UpdateBuffers, bool());
   MOCK_METHOD1(SetDelay, void(size_t delay));
   MOCK_CONST_METHOD0(Delay, size_t());
   MOCK_CONST_METHOD0(MaxDelay, size_t());
   MOCK_CONST_METHOD0(IsBlockAvailable, bool());
-  MOCK_CONST_METHOD0(MaxApiJitter, size_t());
+  MOCK_CONST_METHOD0(GetRenderBuffer, const RenderBuffer&());
+  MOCK_CONST_METHOD0(GetDownsampledRenderBuffer,
+                     const DownsampledRenderBuffer&());
 
  private:
-  const std::vector<std::vector<float>>& FakeGetNext() const { return block_; }
-  std::vector<std::vector<float>> block_;
+  const RenderBuffer& FakeGetRenderBuffer() const { return render_buffer_; }
+  const DownsampledRenderBuffer& FakeGetDownsampledRenderBuffer() const {
+    return downsampled_render_buffer_;
+  }
+  RenderBuffer render_buffer_;
+  DownsampledRenderBuffer downsampled_render_buffer_;
 };
 
 }  // namespace test
 }  // namespace webrtc
 
 #endif  // WEBRTC_MODULES_AUDIO_PROCESSING_AEC3_MOCK_MOCK_RENDER_DELAY_BUFFER_H_