webrtc/common_audio/resampler/push_resampler_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

55 lines
1.9 KiB
C++

/*
* Copyright (c) 2013 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 "common_audio/resampler/include/push_resampler.h"
#include "rtc_base/checks.h" // RTC_DCHECK_IS_ON
#include "test/gtest.h"
// Quality testing of PushResampler is handled through output_mixer_unittest.cc.
namespace webrtc {
// The below tests are temporarily disabled on WEBRTC_WIN due to problems
// with clang debug builds.
// TODO(tommi): Re-enable when we've figured out what the problem is.
// http://crbug.com/615050
#if !defined(WEBRTC_WIN) && defined(__clang__) && !defined(NDEBUG)
TEST(PushResamplerTest, VerifiesInputParameters) {
PushResampler<int16_t> resampler;
EXPECT_EQ(0, resampler.InitializeIfNeeded(16000, 16000, 1));
EXPECT_EQ(0, resampler.InitializeIfNeeded(16000, 16000, 2));
}
#if RTC_DCHECK_IS_ON && GTEST_HAS_DEATH_TEST && !defined(WEBRTC_ANDROID)
TEST(PushResamplerTest, VerifiesBadInputParameters1) {
PushResampler<int16_t> resampler;
EXPECT_DEATH(resampler.InitializeIfNeeded(-1, 16000, 1),
"src_sample_rate_hz");
}
TEST(PushResamplerTest, VerifiesBadInputParameters2) {
PushResampler<int16_t> resampler;
EXPECT_DEATH(resampler.InitializeIfNeeded(16000, -1, 1),
"dst_sample_rate_hz");
}
TEST(PushResamplerTest, VerifiesBadInputParameters3) {
PushResampler<int16_t> resampler;
EXPECT_DEATH(resampler.InitializeIfNeeded(16000, 16000, 0), "num_channels");
}
TEST(PushResamplerTest, VerifiesBadInputParameters4) {
PushResampler<int16_t> resampler;
EXPECT_DEATH(resampler.InitializeIfNeeded(16000, 16000, 3), "num_channels");
}
#endif
#endif
} // namespace webrtc