Adding full initial version of delay estimation functionality in echo canceller 3

webrtc/modules/audio_processing/aec3/decimator_by_4.cc

+/*
+ *  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.
+ */
+#include "webrtc/modules/audio_processing/aec3/decimator_by_4.h"
+
+#include "webrtc/base/checks.h"
+
+namespace webrtc {
+namespace {
+
+// [B,A] = butter(2,1500/16000) which are the same as [B,A] =
+// butter(2,750/8000).
+const CascadedBiQuadFilter::BiQuadCoefficients kLowPassFilterCoefficients = {
+    {0.0179f, 0.0357f, 0.0179f},
+    {-1.5879f, 0.6594f}};
+
+}  // namespace
+
+DecimatorBy4::DecimatorBy4()
+    : low_pass_filter_(kLowPassFilterCoefficients, 3) {}
+
+void DecimatorBy4::Decimate(rtc::ArrayView<const float> in,
+                            std::array<float, kSubBlockSize>* out) {
+  RTC_DCHECK_EQ(kBlockSize, in.size());
+  RTC_DCHECK(out);
+  std::array<float, kBlockSize> x;
+
+  // Limit the frequency content of the signal to avoid aliasing.
+  low_pass_filter_.Process(in, x);
+
+  // Downsample the signal.
+  for (size_t j = 0, k = 0; j < out->size(); ++j, k += 4) {
+    RTC_DCHECK_GT(kBlockSize, k);
+    (*out)[j] = x[k];
+  }
+}
+
+}  // namespace webrtc