Major AEC3 render pipeline changes

webrtc/modules/audio_processing/aec3/render_buffer_unittest.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/render_buffer.h"
+
+#include <algorithm>
+#include <functional>
+#include <vector>
+
+#include "webrtc/test/gtest.h"
+
+namespace webrtc {
+
+#if RTC_DCHECK_IS_ON && GTEST_HAS_DEATH_TEST && !defined(WEBRTC_ANDROID)
+
+// Verifies the check for the provided numbers of Ffts to include in the
+// spectral sum.
+TEST(RenderBuffer, TooLargeNumberOfSpectralSums) {
+  EXPECT_DEATH(
+      RenderBuffer(Aec3Optimization::kNone, 3, 1, std::vector<size_t>(2, 1)),
+      "");
+}
+
+TEST(RenderBuffer, TooSmallNumberOfSpectralSums) {
+  EXPECT_DEATH(
+      RenderBuffer(Aec3Optimization::kNone, 3, 1, std::vector<size_t>()), "");
+}
+
+// Verifies the feasibility check for the provided number of Ffts to include in
+// the spectral.
+TEST(RenderBuffer, FeasibleNumberOfFftsInSum) {
+  EXPECT_DEATH(
+      RenderBuffer(Aec3Optimization::kNone, 3, 1, std::vector<size_t>(1, 2)),
+      "");
+}
+
+#endif
+
+}  // namespace webrtc