blob: 53cd495cd065549bfac41a5867c104d456d99840 [file] [log] [blame]
/*
* 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 "testing/gtest/include/gtest/gtest.h"
#include "webrtc/base/checks.h" // force defintion of RTC_DCHECK_IS_ON
#include "webrtc/common_audio/resampler/include/push_resampler.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