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author | Daniel Baumann <daniel.baumann@progress-linux.org> | 2024-04-07 17:32:43 +0000 |
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committer | Daniel Baumann <daniel.baumann@progress-linux.org> | 2024-04-07 17:32:43 +0000 |
commit | 6bf0a5cb5034a7e684dcc3500e841785237ce2dd (patch) | |
tree | a68f146d7fa01f0134297619fbe7e33db084e0aa /third_party/libwebrtc/common_audio/resampler/push_resampler_unittest.cc | |
parent | Initial commit. (diff) | |
download | thunderbird-6bf0a5cb5034a7e684dcc3500e841785237ce2dd.tar.xz thunderbird-6bf0a5cb5034a7e684dcc3500e841785237ce2dd.zip |
Adding upstream version 1:115.7.0.upstream/1%115.7.0upstream
Signed-off-by: Daniel Baumann <daniel.baumann@progress-linux.org>
Diffstat (limited to 'third_party/libwebrtc/common_audio/resampler/push_resampler_unittest.cc')
-rw-r--r-- | third_party/libwebrtc/common_audio/resampler/push_resampler_unittest.cc | 48 |
1 files changed, 48 insertions, 0 deletions
diff --git a/third_party/libwebrtc/common_audio/resampler/push_resampler_unittest.cc b/third_party/libwebrtc/common_audio/resampler/push_resampler_unittest.cc new file mode 100644 index 0000000000..91f2233aad --- /dev/null +++ b/third_party/libwebrtc/common_audio/resampler/push_resampler_unittest.cc @@ -0,0 +1,48 @@ +/* + * 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> resampler; + EXPECT_EQ(0, resampler.InitializeIfNeeded(16000, 16000, 1)); + EXPECT_EQ(0, resampler.InitializeIfNeeded(16000, 16000, 2)); + EXPECT_EQ(0, resampler.InitializeIfNeeded(16000, 16000, 8)); +} + +#if RTC_DCHECK_IS_ON && GTEST_HAS_DEATH_TEST && !defined(WEBRTC_ANDROID) +TEST(PushResamplerDeathTest, VerifiesBadInputParameters1) { + PushResampler<int16_t> resampler; + RTC_EXPECT_DEATH(resampler.InitializeIfNeeded(-1, 16000, 1), + "src_sample_rate_hz"); +} + +TEST(PushResamplerDeathTest, VerifiesBadInputParameters2) { + PushResampler<int16_t> resampler; + RTC_EXPECT_DEATH(resampler.InitializeIfNeeded(16000, -1, 1), + "dst_sample_rate_hz"); +} + +TEST(PushResamplerDeathTest, VerifiesBadInputParameters3) { + PushResampler<int16_t> resampler; + RTC_EXPECT_DEATH(resampler.InitializeIfNeeded(16000, 16000, 0), + "num_channels"); +} +#endif + +} // namespace webrtc |