1 files changed, 47 insertions, 0 deletions
diff --git a/third_party/libwebrtc/modules/audio_processing/test/echo_canceller_test_tools.cc b/third_party/libwebrtc/modules/audio_processing/test/echo_canceller_test_tools.cc
new file mode 100644
index 0000000000..1d36b954f9
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_processing/test/echo_canceller_test_tools.cc
@@ -0,0 +1,47 @@
+/*
+ *  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 "modules/audio_processing/test/echo_canceller_test_tools.h"
+
+#include "rtc_base/checks.h"
+
+namespace webrtc {
+
+void RandomizeSampleVector(Random* random_generator, rtc::ArrayView<float> v) {
+  RandomizeSampleVector(random_generator, v,
+                        /*amplitude=*/32767.f);
+}
+
+void RandomizeSampleVector(Random* random_generator,
+                           rtc::ArrayView<float> v,
+                           float amplitude) {
+  for (auto& v_k : v) {
+    v_k = 2 * amplitude * random_generator->Rand<float>() - amplitude;
+  }
+}
+
+template <typename T>
+void DelayBuffer<T>::Delay(rtc::ArrayView<const T> x,
+                           rtc::ArrayView<T> x_delayed) {
+  RTC_DCHECK_EQ(x.size(), x_delayed.size());
+  if (buffer_.empty()) {
+    std::copy(x.begin(), x.end(), x_delayed.begin());
+  } else {
+    for (size_t k = 0; k < x.size(); ++k) {
+      x_delayed[k] = buffer_[next_insert_index_];
+      buffer_[next_insert_index_] = x[k];
+      next_insert_index_ = (next_insert_index_ + 1) % buffer_.size();
+    }
+  }
+}
+
+template class DelayBuffer<float>;
+template class DelayBuffer<int>;
+}  // namespace webrtc