webrtc/modules/audio_processing/aec3/render_buffer_unittest.cc
Mirko Bonadei 92ea95e34a Fixing WebRTC after moving from src/webrtc to src/
In https://webrtc-review.googlesource.com/c/src/+/1560 we moved WebRTC
from src/webrtc to src/ (in order to preserve an healthy git history).
This CL takes care of fixing header guards, #include paths, etc...

NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
TBR=tommi@webrtc.org


Bug: chromium:611808
Change-Id: Iea91618212bee0af16aa3f05071eab8f93706578
Reviewed-on: https://webrtc-review.googlesource.com/1561
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Henrik Kjellander <kjellander@webrtc.org>
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#19846}
2017-09-15 05:02:56 +00:00

46 lines
1.3 KiB
C++

/*
* 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/aec3/render_buffer.h"
#include <algorithm>
#include <functional>
#include <vector>
#include "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