Moving src/webrtc into src/.

In order to eliminate the WebRTC Subtree mirror in Chromium, 
WebRTC is moving the content of the src/webrtc directory up
to the src/ directory.

NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
TBR=tommi@webrtc.org

Bug: chromium:611808
Change-Id: Iac59c5b51b950f174119565bac87955a7994bc38
Reviewed-on: https://webrtc-review.googlesource.com/1560
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Henrik Kjellander <kjellander@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#19845}

modules/audio_processing/aec3/render_delay_buffer.h

@@ -0,0 +1,59 @@
+/*
+ *  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_BUFFER_H_
+#define WEBRTC_MODULES_AUDIO_PROCESSING_AEC3_RENDER_DELAY_BUFFER_H_
+
+#include <stddef.h>
+#include <array>
+#include <vector>
+
+#include "webrtc/api/array_view.h"
+#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/fft_data.h"
+#include "webrtc/modules/audio_processing/aec3/render_buffer.h"
+
+namespace webrtc {
+
+// Class for buffering the incoming render blocks such that these may be
+// extracted with a specified delay.
+class RenderDelayBuffer {
+ public:
+  static RenderDelayBuffer* Create(size_t num_bands);
+  virtual ~RenderDelayBuffer() = default;
+
+  // Resets the buffer data.
+  virtual void Reset() = 0;
+
+  // Inserts a block into the buffer and returns true if the insert is
+  // successful.
+  virtual bool Insert(const std::vector<std::vector<float>>& block) = 0;
+
+  // Updates the buffers one step based on the specified buffer delay. Returns
+  // true if there was no overrun, otherwise returns false.
+  virtual bool UpdateBuffers() = 0;
+
+  // Sets the buffer delay.
+  virtual void SetDelay(size_t delay) = 0;
+
+  // Gets the buffer delay.
+  virtual size_t Delay() const = 0;
+
+  // Returns the render buffer for the echo remover.
+  virtual const RenderBuffer& GetRenderBuffer() const = 0;
+
+  // Returns the downsampled render buffer.
+  virtual const DownsampledRenderBuffer& GetDownsampledRenderBuffer() const = 0;
+};
+
+}  // namespace webrtc
+
+#endif  // WEBRTC_MODULES_AUDIO_PROCESSING_AEC3_RENDER_DELAY_BUFFER_H_