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/*
* 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
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