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_buffer.h

@@ -0,0 +1,81 @@
+/*
+ *  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_BUFFER_H_
+#define WEBRTC_MODULES_AUDIO_PROCESSING_AEC3_RENDER_BUFFER_H_
+
+#include <memory>
+#include <vector>
+
+#include "webrtc/api/array_view.h"
+#include "webrtc/modules/audio_processing/aec3/aec3_fft.h"
+#include "webrtc/modules/audio_processing/aec3/fft_data.h"
+#include "webrtc/rtc_base/constructormagic.h"
+
+namespace webrtc {
+
+// Provides a buffer of the render data for the echo remover.
+class RenderBuffer {
+ public:
+  // The constructor takes, besides from the other parameters, a vector
+  // containing the number of FFTs that will be included in the spectral sums in
+  // the call to SpectralSum.
+  RenderBuffer(Aec3Optimization optimization,
+               size_t num_bands,
+               size_t size,
+               const std::vector<size_t> num_ffts_for_spectral_sums);
+  ~RenderBuffer();
+
+  // Clears the buffer.
+  void Clear();
+
+  // Insert a block into the buffer.
+  void Insert(const std::vector<std::vector<float>>& block);
+
+  // Gets the last inserted block.
+  const std::vector<std::vector<float>>& MostRecentBlock() const {
+    return last_block_;
+  }
+
+  // Get the spectrum from one of the FFTs in the buffer
+  const std::array<float, kFftLengthBy2Plus1>& Spectrum(
+      size_t buffer_offset_ffts) const {
+    return spectrum_buffer_[(position_ + buffer_offset_ffts) %
+                            fft_buffer_.size()];
+  }
+
+  // Returns the sum of the spectrums for a certain number of FFTs.
+  const std::array<float, kFftLengthBy2Plus1>& SpectralSum(
+      size_t num_ffts) const {
+    RTC_DCHECK_EQ(spectral_sums_length_, num_ffts);
+    return spectral_sums_[0];
+  }
+
+  // Returns the circular buffer.
+  rtc::ArrayView<const FftData> Buffer() const { return fft_buffer_; }
+
+  // Returns the current position in the circular buffer
+  size_t Position() const { return position_; }
+
+ private:
+  const Aec3Optimization optimization_;
+  std::vector<FftData> fft_buffer_;
+  std::vector<std::array<float, kFftLengthBy2Plus1>> spectrum_buffer_;
+  size_t spectral_sums_length_;
+  std::vector<std::array<float, kFftLengthBy2Plus1>> spectral_sums_;
+  size_t position_ = 0;
+  std::vector<std::vector<float>> last_block_;
+  const Aec3Fft fft_;
+  RTC_DISALLOW_IMPLICIT_CONSTRUCTORS(RenderBuffer);
+};
+
+}  // namespace webrtc
+
+#endif  // WEBRTC_MODULES_AUDIO_PROCESSING_AEC3_RENDER_BUFFER_H_