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/*
* 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"
#include "test/testsupport/rtc_expect_death.h"
// Quality testing of PushResampler is done in audio/remix_resample_unittest.cc.
namespace webrtc {
TEST(PushResamplerTest, VerifiesInputParameters) {
PushResampler<int16_t> resampler1(160, 160, 1);
PushResampler<int16_t> resampler2(160, 160, 2);
PushResampler<int16_t> resampler3(160, 160, 8);
}
#if RTC_DCHECK_IS_ON && GTEST_HAS_DEATH_TEST && !defined(WEBRTC_ANDROID)
TEST(PushResamplerDeathTest, VerifiesBadInputParameters1) {
RTC_EXPECT_DEATH(PushResampler<int16_t>(-1, 160, 1),
"src_samples_per_channel");
}
TEST(PushResamplerDeathTest, VerifiesBadInputParameters2) {
RTC_EXPECT_DEATH(PushResampler<int16_t>(160, -1, 1),
"dst_samples_per_channel");
}
TEST(PushResamplerDeathTest, VerifiesBadInputParameters3) {
RTC_EXPECT_DEATH(PushResampler<int16_t>(160, 16000, 0), "num_channels");
}
#endif
} // namespace webrtc