1 files changed, 33 insertions, 0 deletions
diff --git a/third_party/libwebrtc/modules/audio_processing/aec3/filter_analyzer_unittest.cc b/third_party/libwebrtc/modules/audio_processing/aec3/filter_analyzer_unittest.cc
new file mode 100644
index 0000000000..f1e2e4c188
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_processing/aec3/filter_analyzer_unittest.cc
@@ -0,0 +1,33 @@
+/*
+ *  Copyright (c) 2019 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/aec3/filter_analyzer.h"
+
+#include <algorithm>
+
+#include "test/gmock.h"
+#include "test/gtest.h"
+
+namespace webrtc {
+
+// Verifies that the filter analyzer handles filter resizes properly.
+TEST(FilterAnalyzer, FilterResize) {
+  EchoCanceller3Config c;
+  std::vector<float> filter(65, 0.f);
+  for (size_t num_capture_channels : {1, 2, 4}) {
+    FilterAnalyzer fa(c, num_capture_channels);
+    fa.SetRegionToAnalyze(filter.size());
+    fa.SetRegionToAnalyze(filter.size());
+    filter.resize(32);
+    fa.SetRegionToAnalyze(filter.size());
+  }
+}
+
+}  // namespace webrtc