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_controller.h

@@ -0,0 +1,46 @@
+/*
+ *  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/api/array_view.h"
+#include "webrtc/api/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/include/audio_processing.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(
+      const AudioProcessing::Config::EchoCanceller3& config,
+      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(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_