Major AEC3 render pipeline changes

webrtc/modules/audio_processing/aec3/downsampled_render_buffer.h

Index: webrtc/modules/audio_processing/aec3/downsampled_render_buffer.h
diff --git a/webrtc/modules/audio_processing/aec3/downsampled_render_buffer.h b/webrtc/modules/audio_processing/aec3/downsampled_render_buffer.h
new file mode 100644
index 0000000000000000000000000000000000000000..d9a28385288a8136d9b37ec81836508a7270dc91
--- /dev/null
+++ b/webrtc/modules/audio_processing/aec3/downsampled_render_buffer.h
@@ -0,0 +1,30 @@
+/*
+ *  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_DOWNSAMPLED_RENDER_BUFFER_H_
+#define WEBRTC_MODULES_AUDIO_PROCESSING_AEC3_DOWNSAMPLED_RENDER_BUFFER_H_
+
+#include <array>
+
+#include "webrtc/modules/audio_processing/aec3/aec3_common.h"
+
+namespace webrtc {
+
+// Holds the circular buffer of the downsampled render data.
+struct DownsampledRenderBuffer {
+  DownsampledRenderBuffer();
+  ~DownsampledRenderBuffer();
+  std::array<float, kDownsampledRenderBufferSize> buffer;
+  int position = 0;
+};
+
+}  // namespace webrtc
+
+#endif  // WEBRTC_MODULES_AUDIO_PROCESSING_AEC3_DOWNSAMPLED_RENDER_BUFFER_H_