Rename more death test to *DeathTest
Bug: webrtc:11577
Change-Id: If45e322fed3f2935e64c9e4d7e8c096eccc53ac4
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/176140
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Tommi <tommi@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#31362}
diff --git a/modules/audio_mixer/audio_mixer_impl_unittest.cc b/modules/audio_mixer/audio_mixer_impl_unittest.cc
index 53cd45d..383771c 100644
--- a/modules/audio_mixer/audio_mixer_impl_unittest.cc
+++ b/modules/audio_mixer/audio_mixer_impl_unittest.cc
@@ -606,7 +606,7 @@
};
const int HighOutputRateCalculator::kDefaultFrequency;
-TEST(AudioMixer, MultipleChannelsAndHighRate) {
+TEST(AudioMixerDeathTest, MultipleChannelsAndHighRate) {
constexpr size_t kSamplesPerChannel =
HighOutputRateCalculator::kDefaultFrequency / 100;
// As many channels as an AudioFrame can fit:
diff --git a/modules/audio_mixer/frame_combiner_unittest.cc b/modules/audio_mixer/frame_combiner_unittest.cc
index 5f024a4..4b189a0 100644
--- a/modules/audio_mixer/frame_combiner_unittest.cc
+++ b/modules/audio_mixer/frame_combiner_unittest.cc
@@ -89,7 +89,7 @@
}
// There are DCHECKs in place to check for invalid parameters.
-TEST(FrameCombiner, DebugBuildCrashesWithManyChannels) {
+TEST(FrameCombinerDeathTest, DebugBuildCrashesWithManyChannels) {
FrameCombiner combiner(true);
for (const int rate : {8000, 18000, 34000, 48000}) {
for (const int number_of_channels : {10, 20, 21}) {
@@ -118,7 +118,7 @@
}
}
-TEST(FrameCombiner, DebugBuildCrashesWithHighRate) {
+TEST(FrameCombinerDeathTest, DebugBuildCrashesWithHighRate) {
FrameCombiner combiner(true);
for (const int rate : {50000, 96000, 128000, 196000}) {
for (const int number_of_channels : {1, 2, 3}) {
diff --git a/modules/audio_processing/aec3/adaptive_fir_filter_unittest.cc b/modules/audio_processing/aec3/adaptive_fir_filter_unittest.cc
index 8e4f5d9..39f4e11 100644
--- a/modules/audio_processing/aec3/adaptive_fir_filter_unittest.cc
+++ b/modules/audio_processing/aec3/adaptive_fir_filter_unittest.cc
@@ -285,13 +285,13 @@
#if RTC_DCHECK_IS_ON && GTEST_HAS_DEATH_TEST && !defined(WEBRTC_ANDROID)
// Verifies that the check for non-null data dumper works.
-TEST(AdaptiveFirFilterTest, NullDataDumper) {
+TEST(AdaptiveFirFilterDeathTest, NullDataDumper) {
EXPECT_DEATH(AdaptiveFirFilter(9, 9, 250, 1, DetectOptimization(), nullptr),
"");
}
// Verifies that the check for non-null filter output works.
-TEST(AdaptiveFirFilterTest, NullFilterOutput) {
+TEST(AdaptiveFirFilterDeathTest, NullFilterOutput) {
ApmDataDumper data_dumper(42);
AdaptiveFirFilter filter(9, 9, 250, 1, DetectOptimization(), &data_dumper);
std::unique_ptr<RenderDelayBuffer> render_delay_buffer(
diff --git a/modules/audio_processing/aec3/alignment_mixer_unittest.cc b/modules/audio_processing/aec3/alignment_mixer_unittest.cc
index 832e4ea..03ef066 100644
--- a/modules/audio_processing/aec3/alignment_mixer_unittest.cc
+++ b/modules/audio_processing/aec3/alignment_mixer_unittest.cc
@@ -175,7 +175,7 @@
#if RTC_DCHECK_IS_ON && GTEST_HAS_DEATH_TEST && !defined(WEBRTC_ANDROID)
-TEST(AlignmentMixer, ZeroNumChannels) {
+TEST(AlignmentMixerDeathTest, ZeroNumChannels) {
EXPECT_DEATH(
AlignmentMixer(/*num_channels*/ 0, /*downmix*/ false,
/*adaptive_selection*/ false, /*excitation_limit*/ 1.f,
@@ -183,7 +183,7 @@
, "");
}
-TEST(AlignmentMixer, IncorrectVariant) {
+TEST(AlignmentMixerDeathTest, IncorrectVariant) {
EXPECT_DEATH(
AlignmentMixer(/*num_channels*/ 1, /*downmix*/ true,
/*adaptive_selection*/ true, /*excitation_limit*/ 1.f,
diff --git a/modules/audio_processing/aec3/block_processor_unittest.cc b/modules/audio_processing/aec3/block_processor_unittest.cc
index 2b928e8..911dad4 100644
--- a/modules/audio_processing/aec3/block_processor_unittest.cc
+++ b/modules/audio_processing/aec3/block_processor_unittest.cc
@@ -252,21 +252,21 @@
#if RTC_DCHECK_IS_ON && GTEST_HAS_DEATH_TEST && !defined(WEBRTC_ANDROID)
// TODO(gustaf): Re-enable the test once the issue with memory leaks during
// DEATH tests on test bots has been fixed.
-TEST(BlockProcessor, DISABLED_VerifyRenderBlockSizeCheck) {
+TEST(BlockProcessorDeathTest, DISABLED_VerifyRenderBlockSizeCheck) {
for (auto rate : {16000, 32000, 48000}) {
SCOPED_TRACE(ProduceDebugText(rate));
RunRenderBlockSizeVerificationTest(rate);
}
}
-TEST(BlockProcessor, VerifyCaptureBlockSizeCheck) {
+TEST(BlockProcessorDeathTest, VerifyCaptureBlockSizeCheck) {
for (auto rate : {16000, 32000, 48000}) {
SCOPED_TRACE(ProduceDebugText(rate));
RunCaptureBlockSizeVerificationTest(rate);
}
}
-TEST(BlockProcessor, VerifyRenderNumBandsCheck) {
+TEST(BlockProcessorDeathTest, VerifyRenderNumBandsCheck) {
for (auto rate : {16000, 32000, 48000}) {
SCOPED_TRACE(ProduceDebugText(rate));
RunRenderNumBandsVerificationTest(rate);
@@ -275,7 +275,7 @@
// TODO(peah): Verify the check for correct number of bands in the capture
// signal.
-TEST(BlockProcessor, VerifyCaptureNumBandsCheck) {
+TEST(BlockProcessorDeathTest, VerifyCaptureNumBandsCheck) {
for (auto rate : {16000, 32000, 48000}) {
SCOPED_TRACE(ProduceDebugText(rate));
RunCaptureNumBandsVerificationTest(rate);
@@ -283,7 +283,7 @@
}
// Verifiers that the verification for null ProcessCapture input works.
-TEST(BlockProcessor, NullProcessCaptureParameter) {
+TEST(BlockProcessorDeathTest, NullProcessCaptureParameter) {
EXPECT_DEATH(std::unique_ptr<BlockProcessor>(
BlockProcessor::Create(EchoCanceller3Config(), 16000, 1, 1))
->ProcessCapture(false, false, nullptr, nullptr),
diff --git a/modules/audio_processing/aec3/coarse_filter_update_gain_unittest.cc b/modules/audio_processing/aec3/coarse_filter_update_gain_unittest.cc
index 4185c1a..92775cf 100644
--- a/modules/audio_processing/aec3/coarse_filter_update_gain_unittest.cc
+++ b/modules/audio_processing/aec3/coarse_filter_update_gain_unittest.cc
@@ -138,7 +138,7 @@
#if RTC_DCHECK_IS_ON && GTEST_HAS_DEATH_TEST && !defined(WEBRTC_ANDROID)
// Verifies that the check for non-null output gain parameter works.
-TEST(CoarseFilterUpdateGain, NullDataOutputGain) {
+TEST(CoarseFilterUpdateGainDeathTest, NullDataOutputGain) {
ApmDataDumper data_dumper(42);
FftBuffer fft_buffer(1, 1);
RenderSignalAnalyzer analyzer(EchoCanceller3Config{});
diff --git a/modules/audio_processing/aec3/decimator_unittest.cc b/modules/audio_processing/aec3/decimator_unittest.cc
index 1e279ce..e6f5ea0 100644
--- a/modules/audio_processing/aec3/decimator_unittest.cc
+++ b/modules/audio_processing/aec3/decimator_unittest.cc
@@ -103,7 +103,7 @@
#if RTC_DCHECK_IS_ON && GTEST_HAS_DEATH_TEST && !defined(WEBRTC_ANDROID)
// Verifies the check for the input size.
-TEST(Decimator, WrongInputSize) {
+TEST(DecimatorDeathTest, WrongInputSize) {
Decimator decimator(4);
std::vector<float> x(kBlockSize - 1, 0.f);
std::array<float, kBlockSize / 4> x_downsampled;
@@ -111,14 +111,14 @@
}
// Verifies the check for non-null output parameter.
-TEST(Decimator, NullOutput) {
+TEST(DecimatorDeathTest, NullOutput) {
Decimator decimator(4);
std::vector<float> x(kBlockSize, 0.f);
EXPECT_DEATH(decimator.Decimate(x, nullptr), "");
}
// Verifies the check for the output size.
-TEST(Decimator, WrongOutputSize) {
+TEST(DecimatorDeathTest, WrongOutputSize) {
Decimator decimator(4);
std::vector<float> x(kBlockSize, 0.f);
std::array<float, kBlockSize / 4 - 1> x_downsampled;
@@ -126,7 +126,7 @@
}
// Verifies the check for the correct downsampling factor.
-TEST(Decimator, CorrectDownSamplingFactor) {
+TEST(DecimatorDeathTest, CorrectDownSamplingFactor) {
EXPECT_DEATH(Decimator(3), "");
}
diff --git a/modules/audio_processing/aec3/echo_canceller3_unittest.cc b/modules/audio_processing/aec3/echo_canceller3_unittest.cc
index 21255f1..04d93e4 100644
--- a/modules/audio_processing/aec3/echo_canceller3_unittest.cc
+++ b/modules/audio_processing/aec3/echo_canceller3_unittest.cc
@@ -890,7 +890,7 @@
#if RTC_DCHECK_IS_ON && GTEST_HAS_DEATH_TEST && !defined(WEBRTC_ANDROID)
-TEST(EchoCanceller3InputCheck, WrongCaptureNumBandsCheckVerification) {
+TEST(EchoCanceller3InputCheckDeathTest, WrongCaptureNumBandsCheckVerification) {
for (auto rate : {16000, 32000, 48000}) {
SCOPED_TRACE(ProduceDebugText(rate));
EchoCanceller3Tester(rate).RunProcessCaptureNumBandsCheckVerification();
@@ -899,7 +899,7 @@
// Verifiers that the verification for null input to the capture processing api
// call works.
-TEST(EchoCanceller3InputCheck, NullCaptureProcessingParameter) {
+TEST(EchoCanceller3InputCheckDeathTest, NullCaptureProcessingParameter) {
EXPECT_DEATH(EchoCanceller3(EchoCanceller3Config(), 16000, 1, 1)
.ProcessCapture(nullptr, false),
"");
@@ -908,7 +908,7 @@
// Verifies the check for correct sample rate.
// TODO(peah): Re-enable the test once the issue with memory leaks during DEATH
// tests on test bots has been fixed.
-TEST(EchoCanceller3InputCheck, DISABLED_WrongSampleRate) {
+TEST(EchoCanceller3InputCheckDeathTest, DISABLED_WrongSampleRate) {
ApmDataDumper data_dumper(0);
EXPECT_DEATH(EchoCanceller3(EchoCanceller3Config(), 8001, 1, 1), "");
}
diff --git a/modules/audio_processing/aec3/echo_remover_metrics_unittest.cc b/modules/audio_processing/aec3/echo_remover_metrics_unittest.cc
index 30c6611..45b30a9 100644
--- a/modules/audio_processing/aec3/echo_remover_metrics_unittest.cc
+++ b/modules/audio_processing/aec3/echo_remover_metrics_unittest.cc
@@ -23,7 +23,7 @@
#if RTC_DCHECK_IS_ON && GTEST_HAS_DEATH_TEST && !defined(WEBRTC_ANDROID)
// Verifies the check for non-null input.
-TEST(UpdateDbMetric, NullValue) {
+TEST(UpdateDbMetricDeathTest, NullValue) {
std::array<float, kFftLengthBy2Plus1> value;
value.fill(0.f);
EXPECT_DEATH(aec3::UpdateDbMetric(value, nullptr), "");
diff --git a/modules/audio_processing/aec3/echo_remover_unittest.cc b/modules/audio_processing/aec3/echo_remover_unittest.cc
index e050027..77a2076 100644
--- a/modules/audio_processing/aec3/echo_remover_unittest.cc
+++ b/modules/audio_processing/aec3/echo_remover_unittest.cc
@@ -91,14 +91,14 @@
// Verifies the check for the samplerate.
// TODO(peah): Re-enable the test once the issue with memory leaks during DEATH
// tests on test bots has been fixed.
-TEST(EchoRemover, DISABLED_WrongSampleRate) {
+TEST(EchoRemoverDeathTest, DISABLED_WrongSampleRate) {
EXPECT_DEATH(std::unique_ptr<EchoRemover>(
EchoRemover::Create(EchoCanceller3Config(), 8001, 1, 1)),
"");
}
// Verifies the check for the capture block size.
-TEST(EchoRemover, WrongCaptureBlockSize) {
+TEST(EchoRemoverDeathTest, WrongCaptureBlockSize) {
absl::optional<DelayEstimate> delay_estimate;
for (auto rate : {16000, 32000, 48000}) {
SCOPED_TRACE(ProduceDebugText(rate));
@@ -121,7 +121,7 @@
// Verifies the check for the number of capture bands.
// TODO(peah): Re-enable the test once the issue with memory leaks during DEATH
// tests on test bots has been fixed.c
-TEST(EchoRemover, DISABLED_WrongCaptureNumBands) {
+TEST(EchoRemoverDeathTest, DISABLED_WrongCaptureNumBands) {
absl::optional<DelayEstimate> delay_estimate;
for (auto rate : {16000, 32000, 48000}) {
SCOPED_TRACE(ProduceDebugText(rate));
@@ -143,7 +143,7 @@
}
// Verifies the check for non-null capture block.
-TEST(EchoRemover, NullCapture) {
+TEST(EchoRemoverDeathTest, NullCapture) {
absl::optional<DelayEstimate> delay_estimate;
std::unique_ptr<EchoRemover> remover(
EchoRemover::Create(EchoCanceller3Config(), 16000, 1, 1));
diff --git a/modules/audio_processing/aec3/fft_data_unittest.cc b/modules/audio_processing/aec3/fft_data_unittest.cc
index 0812fd6..9be2680 100644
--- a/modules/audio_processing/aec3/fft_data_unittest.cc
+++ b/modules/audio_processing/aec3/fft_data_unittest.cc
@@ -44,12 +44,12 @@
#if RTC_DCHECK_IS_ON && GTEST_HAS_DEATH_TEST && !defined(WEBRTC_ANDROID)
// Verifies the check for null output in CopyToPackedArray.
-TEST(FftData, NonNullCopyToPackedArrayOutput) {
+TEST(FftDataDeathTest, NonNullCopyToPackedArrayOutput) {
EXPECT_DEATH(FftData().CopyToPackedArray(nullptr), "");
}
// Verifies the check for null output in Spectrum.
-TEST(FftData, NonNullSpectrumOutput) {
+TEST(FftDataDeathTest, NonNullSpectrumOutput) {
EXPECT_DEATH(FftData().Spectrum(Aec3Optimization::kNone, nullptr), "");
}
diff --git a/modules/audio_processing/aec3/matched_filter_lag_aggregator_unittest.cc b/modules/audio_processing/aec3/matched_filter_lag_aggregator_unittest.cc
index e136c89..8e2a12e 100644
--- a/modules/audio_processing/aec3/matched_filter_lag_aggregator_unittest.cc
+++ b/modules/audio_processing/aec3/matched_filter_lag_aggregator_unittest.cc
@@ -144,7 +144,7 @@
#if RTC_DCHECK_IS_ON && GTEST_HAS_DEATH_TEST && !defined(WEBRTC_ANDROID)
// Verifies the check for non-null data dumper.
-TEST(MatchedFilterLagAggregator, NullDataDumper) {
+TEST(MatchedFilterLagAggregatorDeathTest, NullDataDumper) {
EchoCanceller3Config config;
EXPECT_DEATH(MatchedFilterLagAggregator(
nullptr, 10, config.delay.delay_selection_thresholds),
diff --git a/modules/audio_processing/aec3/matched_filter_unittest.cc b/modules/audio_processing/aec3/matched_filter_unittest.cc
index 8a6e22e..7d9a7d4 100644
--- a/modules/audio_processing/aec3/matched_filter_unittest.cc
+++ b/modules/audio_processing/aec3/matched_filter_unittest.cc
@@ -375,7 +375,7 @@
#if RTC_DCHECK_IS_ON && GTEST_HAS_DEATH_TEST && !defined(WEBRTC_ANDROID)
// Verifies the check for non-zero windows size.
-TEST(MatchedFilter, ZeroWindowSize) {
+TEST(MatchedFilterDeathTest, ZeroWindowSize) {
ApmDataDumper data_dumper(0);
EchoCanceller3Config config;
EXPECT_DEATH(MatchedFilter(&data_dumper, DetectOptimization(), 16, 0, 1, 1,
@@ -385,7 +385,7 @@
}
// Verifies the check for non-null data dumper.
-TEST(MatchedFilter, NullDataDumper) {
+TEST(MatchedFilterDeathTest, NullDataDumper) {
EchoCanceller3Config config;
EXPECT_DEATH(MatchedFilter(nullptr, DetectOptimization(), 16, 1, 1, 1, 150,
config.delay.delay_estimate_smoothing,
@@ -395,7 +395,7 @@
// Verifies the check for that the sub block size is a multiple of 4.
// TODO(peah): Activate the unittest once the required code has been landed.
-TEST(MatchedFilter, DISABLED_BlockSizeMultipleOf4) {
+TEST(MatchedFilterDeathTest, DISABLED_BlockSizeMultipleOf4) {
ApmDataDumper data_dumper(0);
EchoCanceller3Config config;
EXPECT_DEATH(MatchedFilter(&data_dumper, DetectOptimization(), 15, 1, 1, 1,
@@ -407,7 +407,7 @@
// Verifies the check for that there is an integer number of sub blocks that add
// up to a block size.
// TODO(peah): Activate the unittest once the required code has been landed.
-TEST(MatchedFilter, DISABLED_SubBlockSizeAddsUpToBlockSize) {
+TEST(MatchedFilterDeathTest, DISABLED_SubBlockSizeAddsUpToBlockSize) {
ApmDataDumper data_dumper(0);
EchoCanceller3Config config;
EXPECT_DEATH(MatchedFilter(&data_dumper, DetectOptimization(), 12, 1, 1, 1,
diff --git a/modules/audio_processing/aec3/refined_filter_update_gain_unittest.cc b/modules/audio_processing/aec3/refined_filter_update_gain_unittest.cc
index 117f345..2393fdd 100644
--- a/modules/audio_processing/aec3/refined_filter_update_gain_unittest.cc
+++ b/modules/audio_processing/aec3/refined_filter_update_gain_unittest.cc
@@ -234,7 +234,7 @@
#if RTC_DCHECK_IS_ON && GTEST_HAS_DEATH_TEST && !defined(WEBRTC_ANDROID)
// Verifies that the check for non-null output gain parameter works.
-TEST(RefinedFilterUpdateGain, NullDataOutputGain) {
+TEST(RefinedFilterUpdateGainDeathTest, NullDataOutputGain) {
ApmDataDumper data_dumper(42);
EchoCanceller3Config config;
RenderSignalAnalyzer analyzer(config);
diff --git a/modules/audio_processing/aec3/render_buffer_unittest.cc b/modules/audio_processing/aec3/render_buffer_unittest.cc
index 6981f6d..4559528 100644
--- a/modules/audio_processing/aec3/render_buffer_unittest.cc
+++ b/modules/audio_processing/aec3/render_buffer_unittest.cc
@@ -21,21 +21,21 @@
#if RTC_DCHECK_IS_ON && GTEST_HAS_DEATH_TEST && !defined(WEBRTC_ANDROID)
// Verifies the check for non-null fft buffer.
-TEST(RenderBuffer, NullExternalFftBuffer) {
+TEST(RenderBufferDeathTest, NullExternalFftBuffer) {
BlockBuffer block_buffer(10, 3, 1, kBlockSize);
SpectrumBuffer spectrum_buffer(10, 1);
EXPECT_DEATH(RenderBuffer(&block_buffer, &spectrum_buffer, nullptr), "");
}
// Verifies the check for non-null spectrum buffer.
-TEST(RenderBuffer, NullExternalSpectrumBuffer) {
+TEST(RenderBufferDeathTest, NullExternalSpectrumBuffer) {
FftBuffer fft_buffer(10, 1);
BlockBuffer block_buffer(10, 3, 1, kBlockSize);
EXPECT_DEATH(RenderBuffer(&block_buffer, nullptr, &fft_buffer), "");
}
// Verifies the check for non-null block buffer.
-TEST(RenderBuffer, NullExternalBlockBuffer) {
+TEST(RenderBufferDeathTest, NullExternalBlockBuffer) {
FftBuffer fft_buffer(10, 1);
SpectrumBuffer spectrum_buffer(10, 1);
EXPECT_DEATH(RenderBuffer(nullptr, &spectrum_buffer, &fft_buffer), "");
diff --git a/modules/audio_processing/aec3/render_delay_buffer_unittest.cc b/modules/audio_processing/aec3/render_delay_buffer_unittest.cc
index 35e8131..efd4a29 100644
--- a/modules/audio_processing/aec3/render_delay_buffer_unittest.cc
+++ b/modules/audio_processing/aec3/render_delay_buffer_unittest.cc
@@ -97,14 +97,14 @@
// Verifies the check for feasible delay.
// TODO(peah): Re-enable the test once the issue with memory leaks during DEATH
// tests on test bots has been fixed.
-TEST(RenderDelayBuffer, DISABLED_WrongDelay) {
+TEST(RenderDelayBufferDeathTest, DISABLED_WrongDelay) {
std::unique_ptr<RenderDelayBuffer> delay_buffer(
RenderDelayBuffer::Create(EchoCanceller3Config(), 48000, 1));
EXPECT_DEATH(delay_buffer->AlignFromDelay(21), "");
}
// Verifies the check for the number of bands in the inserted blocks.
-TEST(RenderDelayBuffer, WrongNumberOfBands) {
+TEST(RenderDelayBufferDeathTest, WrongNumberOfBands) {
for (auto rate : {16000, 32000, 48000}) {
for (size_t num_channels : {1, 2, 8}) {
SCOPED_TRACE(ProduceDebugText(rate));
@@ -120,7 +120,7 @@
}
// Verifies the check for the number of channels in the inserted blocks.
-TEST(RenderDelayBuffer, WrongNumberOfChannels) {
+TEST(RenderDelayBufferDeathTest, WrongNumberOfChannels) {
for (auto rate : {16000, 32000, 48000}) {
for (size_t num_channels : {1, 2, 8}) {
SCOPED_TRACE(ProduceDebugText(rate));
@@ -136,7 +136,7 @@
}
// Verifies the check of the length of the inserted blocks.
-TEST(RenderDelayBuffer, WrongBlockLength) {
+TEST(RenderDelayBufferDeathTest, WrongBlockLength) {
for (auto rate : {16000, 32000, 48000}) {
for (size_t num_channels : {1, 2, 8}) {
SCOPED_TRACE(ProduceDebugText(rate));
diff --git a/modules/audio_processing/aec3/render_delay_controller_unittest.cc b/modules/audio_processing/aec3/render_delay_controller_unittest.cc
index fb7b86a..0d3c856 100644
--- a/modules/audio_processing/aec3/render_delay_controller_unittest.cc
+++ b/modules/audio_processing/aec3/render_delay_controller_unittest.cc
@@ -325,7 +325,7 @@
#if RTC_DCHECK_IS_ON && GTEST_HAS_DEATH_TEST && !defined(WEBRTC_ANDROID)
// Verifies the check for the capture signal block size.
-TEST(RenderDelayController, WrongCaptureSize) {
+TEST(RenderDelayControllerDeathTest, WrongCaptureSize) {
std::vector<std::vector<float>> block(
1, std::vector<float>(kBlockSize - 1, 0.f));
EchoCanceller3Config config;
@@ -345,7 +345,7 @@
// Verifies the check for correct sample rate.
// TODO(peah): Re-enable the test once the issue with memory leaks during DEATH
// tests on test bots has been fixed.
-TEST(RenderDelayController, DISABLED_WrongSampleRate) {
+TEST(RenderDelayControllerDeathTest, DISABLED_WrongSampleRate) {
for (auto rate : {-1, 0, 8001, 16001}) {
SCOPED_TRACE(ProduceDebugText(rate));
EchoCanceller3Config config;
diff --git a/modules/audio_processing/aec3/render_signal_analyzer_unittest.cc b/modules/audio_processing/aec3/render_signal_analyzer_unittest.cc
index f40fade..7a48cc4 100644
--- a/modules/audio_processing/aec3/render_signal_analyzer_unittest.cc
+++ b/modules/audio_processing/aec3/render_signal_analyzer_unittest.cc
@@ -117,7 +117,7 @@
#if RTC_DCHECK_IS_ON && GTEST_HAS_DEATH_TEST && !defined(WEBRTC_ANDROID)
// Verifies that the check for non-null output parameter works.
-TEST(RenderSignalAnalyzer, NullMaskOutput) {
+TEST(RenderSignalAnalyzerDeathTest, NullMaskOutput) {
RenderSignalAnalyzer analyzer(EchoCanceller3Config{});
EXPECT_DEATH(analyzer.MaskRegionsAroundNarrowBands(nullptr), "");
}
diff --git a/modules/audio_processing/aec3/subtractor_unittest.cc b/modules/audio_processing/aec3/subtractor_unittest.cc
index 72e5787..bbc1e4f 100644
--- a/modules/audio_processing/aec3/subtractor_unittest.cc
+++ b/modules/audio_processing/aec3/subtractor_unittest.cc
@@ -189,7 +189,7 @@
#if RTC_DCHECK_IS_ON && GTEST_HAS_DEATH_TEST && !defined(WEBRTC_ANDROID)
// Verifies that the check for non data dumper works.
-TEST(Subtractor, NullDataDumper) {
+TEST(SubtractorDeathTest, NullDataDumper) {
EXPECT_DEATH(
Subtractor(EchoCanceller3Config(), 1, 1, nullptr, DetectOptimization()),
"");
diff --git a/modules/audio_processing/aec3/suppression_filter_unittest.cc b/modules/audio_processing/aec3/suppression_filter_unittest.cc
index b55c719..a160bec 100644
--- a/modules/audio_processing/aec3/suppression_filter_unittest.cc
+++ b/modules/audio_processing/aec3/suppression_filter_unittest.cc
@@ -50,7 +50,7 @@
#if RTC_DCHECK_IS_ON && GTEST_HAS_DEATH_TEST && !defined(WEBRTC_ANDROID)
// Verifies the check for null suppressor output.
-TEST(SuppressionFilter, NullOutput) {
+TEST(SuppressionFilterDeathTest, NullOutput) {
std::vector<FftData> cn(1);
std::vector<FftData> cn_high_bands(1);
std::vector<FftData> E(1);
@@ -62,7 +62,7 @@
}
// Verifies the check for allowed sample rate.
-TEST(SuppressionFilter, ProperSampleRate) {
+TEST(SuppressionFilterDeathTest, ProperSampleRate) {
EXPECT_DEATH(SuppressionFilter(Aec3Optimization::kNone, 16001, 1), "");
}
diff --git a/modules/audio_processing/aec3/suppression_gain_unittest.cc b/modules/audio_processing/aec3/suppression_gain_unittest.cc
index 0452f2e..4fb4cd71 100644
--- a/modules/audio_processing/aec3/suppression_gain_unittest.cc
+++ b/modules/audio_processing/aec3/suppression_gain_unittest.cc
@@ -25,7 +25,7 @@
#if RTC_DCHECK_IS_ON && GTEST_HAS_DEATH_TEST && !defined(WEBRTC_ANDROID)
// Verifies that the check for non-null output gains works.
-TEST(SuppressionGain, NullOutputGains) {
+TEST(SuppressionGainDeathTest, NullOutputGains) {
std::vector<std::array<float, kFftLengthBy2Plus1>> E2(1, {0.f});
std::vector<std::array<float, kFftLengthBy2Plus1>> R2(1, {0.f});
std::vector<std::array<float, kFftLengthBy2Plus1>> S2(1);
diff --git a/modules/audio_processing/audio_buffer_unittest.cc b/modules/audio_processing/audio_buffer_unittest.cc
index 7cb51ca..f3b2ddc 100644
--- a/modules/audio_processing/audio_buffer_unittest.cc
+++ b/modules/audio_processing/audio_buffer_unittest.cc
@@ -40,7 +40,7 @@
}
#if RTC_DCHECK_IS_ON && GTEST_HAS_DEATH_TEST && !defined(WEBRTC_ANDROID)
-TEST(AudioBufferTest, SetNumChannelsDeathTest) {
+TEST(AudioBufferDeathTest, SetNumChannelsDeathTest) {
AudioBuffer ab(kSampleRateHz, kMono, kSampleRateHz, kMono, kSampleRateHz,
kMono);
RTC_EXPECT_DEATH(ab.set_num_channels(kStereo), "num_channels");
diff --git a/modules/audio_processing/audio_processing_unittest.cc b/modules/audio_processing/audio_processing_unittest.cc
index 90413a8..93ddc97 100644
--- a/modules/audio_processing/audio_processing_unittest.cc
+++ b/modules/audio_processing/audio_processing_unittest.cc
@@ -962,49 +962,51 @@
}
#if RTC_DCHECK_IS_ON && GTEST_HAS_DEATH_TEST && !defined(WEBRTC_ANDROID)
-TEST_F(ApmTest, GainControlDiesOnTooLowTargetLevelDbfs) {
+using ApmDeathTest = ApmTest;
+
+TEST_F(ApmDeathTest, GainControlDiesOnTooLowTargetLevelDbfs) {
auto config = apm_->GetConfig();
config.gain_controller1.enabled = true;
config.gain_controller1.target_level_dbfs = -1;
EXPECT_DEATH(apm_->ApplyConfig(config), "");
}
-TEST_F(ApmTest, GainControlDiesOnTooHighTargetLevelDbfs) {
+TEST_F(ApmDeathTest, GainControlDiesOnTooHighTargetLevelDbfs) {
auto config = apm_->GetConfig();
config.gain_controller1.enabled = true;
config.gain_controller1.target_level_dbfs = 32;
EXPECT_DEATH(apm_->ApplyConfig(config), "");
}
-TEST_F(ApmTest, GainControlDiesOnTooLowCompressionGainDb) {
+TEST_F(ApmDeathTest, GainControlDiesOnTooLowCompressionGainDb) {
auto config = apm_->GetConfig();
config.gain_controller1.enabled = true;
config.gain_controller1.compression_gain_db = -1;
EXPECT_DEATH(apm_->ApplyConfig(config), "");
}
-TEST_F(ApmTest, GainControlDiesOnTooHighCompressionGainDb) {
+TEST_F(ApmDeathTest, GainControlDiesOnTooHighCompressionGainDb) {
auto config = apm_->GetConfig();
config.gain_controller1.enabled = true;
config.gain_controller1.compression_gain_db = 91;
EXPECT_DEATH(apm_->ApplyConfig(config), "");
}
-TEST_F(ApmTest, GainControlDiesOnTooLowAnalogLevelLowerLimit) {
+TEST_F(ApmDeathTest, GainControlDiesOnTooLowAnalogLevelLowerLimit) {
auto config = apm_->GetConfig();
config.gain_controller1.enabled = true;
config.gain_controller1.analog_level_minimum = -1;
EXPECT_DEATH(apm_->ApplyConfig(config), "");
}
-TEST_F(ApmTest, GainControlDiesOnTooHighAnalogLevelUpperLimit) {
+TEST_F(ApmDeathTest, GainControlDiesOnTooHighAnalogLevelUpperLimit) {
auto config = apm_->GetConfig();
config.gain_controller1.enabled = true;
config.gain_controller1.analog_level_maximum = 65536;
EXPECT_DEATH(apm_->ApplyConfig(config), "");
}
-TEST_F(ApmTest, GainControlDiesOnInvertedAnalogLevelLimits) {
+TEST_F(ApmDeathTest, GainControlDiesOnInvertedAnalogLevelLimits) {
auto config = apm_->GetConfig();
config.gain_controller1.enabled = true;
config.gain_controller1.analog_level_minimum = 512;
@@ -1012,7 +1014,7 @@
EXPECT_DEATH(apm_->ApplyConfig(config), "");
}
-TEST_F(ApmTest, ApmDiesOnTooLowAnalogLevel) {
+TEST_F(ApmDeathTest, ApmDiesOnTooLowAnalogLevel) {
auto config = apm_->GetConfig();
config.gain_controller1.enabled = true;
config.gain_controller1.analog_level_minimum = 255;
@@ -1021,7 +1023,7 @@
EXPECT_DEATH(apm_->set_stream_analog_level(254), "");
}
-TEST_F(ApmTest, ApmDiesOnTooHighAnalogLevel) {
+TEST_F(ApmDeathTest, ApmDiesOnTooHighAnalogLevel) {
auto config = apm_->GetConfig();
config.gain_controller1.enabled = true;
config.gain_controller1.analog_level_minimum = 255;
@@ -2414,7 +2416,7 @@
EXPECT_EQ(AudioProcessing::RuntimeSetting::Type::kNotSpecified, s.type());
}
-TEST(RuntimeSettingTest, TestCapturePreGain) {
+TEST(RuntimeSettingDeathTest, TestCapturePreGain) {
using Type = AudioProcessing::RuntimeSetting::Type;
{
auto s = AudioProcessing::RuntimeSetting::CreateCapturePreGain(1.25f);
@@ -2429,7 +2431,7 @@
#endif
}
-TEST(RuntimeSettingTest, TestCaptureFixedPostGain) {
+TEST(RuntimeSettingDeathTest, TestCaptureFixedPostGain) {
using Type = AudioProcessing::RuntimeSetting::Type;
{
auto s = AudioProcessing::RuntimeSetting::CreateCaptureFixedPostGain(1.25f);
diff --git a/modules/audio_processing/utility/cascaded_biquad_filter_unittest.cc b/modules/audio_processing/utility/cascaded_biquad_filter_unittest.cc
index 989e362..ff7022d 100644
--- a/modules/audio_processing/utility/cascaded_biquad_filter_unittest.cc
+++ b/modules/audio_processing/utility/cascaded_biquad_filter_unittest.cc
@@ -103,7 +103,7 @@
#if RTC_DCHECK_IS_ON && GTEST_HAS_DEATH_TEST && !defined(WEBRTC_ANDROID)
// Verifies that the check of the lengths for the input and output works for the
// non-in-place call.
-TEST(CascadedBiquadFilter, InputSizeCheckVerification) {
+TEST(CascadedBiquadFilterDeathTest, InputSizeCheckVerification) {
const std::vector<float> input = CreateInputWithIncreasingValues(10);
std::vector<float> output(input.size() - 1);
diff --git a/modules/audio_processing/utility/pffft_wrapper_unittest.cc b/modules/audio_processing/utility/pffft_wrapper_unittest.cc
index 9aed548..2ad6849 100644
--- a/modules/audio_processing/utility/pffft_wrapper_unittest.cc
+++ b/modules/audio_processing/utility/pffft_wrapper_unittest.cc
@@ -125,23 +125,24 @@
#if !defined(NDEBUG) && GTEST_HAS_DEATH_TEST && !defined(WEBRTC_ANDROID)
-class PffftInvalidSizeTest : public ::testing::Test,
- public ::testing::WithParamInterface<size_t> {};
+class PffftInvalidSizeDeathTest : public ::testing::Test,
+ public ::testing::WithParamInterface<size_t> {
+};
-TEST_P(PffftInvalidSizeTest, DoNotCreateRealWrapper) {
+TEST_P(PffftInvalidSizeDeathTest, DoNotCreateRealWrapper) {
size_t fft_size = GetParam();
ASSERT_FALSE(Pffft::IsValidFftSize(fft_size, Pffft::FftType::kReal));
EXPECT_DEATH(CreatePffftWrapper(fft_size, Pffft::FftType::kReal), "");
}
-TEST_P(PffftInvalidSizeTest, DoNotCreateComplexWrapper) {
+TEST_P(PffftInvalidSizeDeathTest, DoNotCreateComplexWrapper) {
size_t fft_size = GetParam();
ASSERT_FALSE(Pffft::IsValidFftSize(fft_size, Pffft::FftType::kComplex));
EXPECT_DEATH(CreatePffftWrapper(fft_size, Pffft::FftType::kComplex), "");
}
INSTANTIATE_TEST_SUITE_P(PffftTest,
- PffftInvalidSizeTest,
+ PffftInvalidSizeDeathTest,
::testing::Values(17,
33,
65,
diff --git a/modules/pacing/packet_router_unittest.cc b/modules/pacing/packet_router_unittest.cc
index 75729cb..79092ea 100644
--- a/modules/pacing/packet_router_unittest.cc
+++ b/modules/pacing/packet_router_unittest.cc
@@ -406,7 +406,8 @@
}
#if RTC_DCHECK_IS_ON && GTEST_HAS_DEATH_TEST && !defined(WEBRTC_ANDROID)
-TEST_F(PacketRouterTest, DoubleRegistrationOfSendModuleDisallowed) {
+using PacketRouterDeathTest = PacketRouterTest;
+TEST_F(PacketRouterDeathTest, DoubleRegistrationOfSendModuleDisallowed) {
NiceMock<MockRtpRtcp> module;
constexpr bool remb_candidate = false; // Value irrelevant.
@@ -417,7 +418,7 @@
packet_router_.RemoveSendRtpModule(&module);
}
-TEST_F(PacketRouterTest, DoubleRegistrationOfReceiveModuleDisallowed) {
+TEST_F(PacketRouterDeathTest, DoubleRegistrationOfReceiveModuleDisallowed) {
NiceMock<MockRtpRtcp> module;
constexpr bool remb_candidate = false; // Value irrelevant.
@@ -428,13 +429,13 @@
packet_router_.RemoveReceiveRtpModule(&module);
}
-TEST_F(PacketRouterTest, RemovalOfNeverAddedSendModuleDisallowed) {
+TEST_F(PacketRouterDeathTest, RemovalOfNeverAddedSendModuleDisallowed) {
NiceMock<MockRtpRtcp> module;
EXPECT_DEATH(packet_router_.RemoveSendRtpModule(&module), "");
}
-TEST_F(PacketRouterTest, RemovalOfNeverAddedReceiveModuleDisallowed) {
+TEST_F(PacketRouterDeathTest, RemovalOfNeverAddedReceiveModuleDisallowed) {
NiceMock<MockRtpRtcp> module;
EXPECT_DEATH(packet_router_.RemoveReceiveRtpModule(&module), "");
diff --git a/rtc_base/bit_buffer_unittest.cc b/rtc_base/bit_buffer_unittest.cc
index b3521b4..441cd26 100644
--- a/rtc_base/bit_buffer_unittest.cc
+++ b/rtc_base/bit_buffer_unittest.cc
@@ -142,7 +142,7 @@
EXPECT_FALSE(buffer.ReadBits(&val, 1));
}
-TEST(BitBufferTest, SetOffsetValues) {
+TEST(BitBufferDeathTest, SetOffsetValues) {
uint8_t bytes[4] = {0};
BitBufferWriter buffer(bytes, 4);
diff --git a/rtc_base/buffer_unittest.cc b/rtc_base/buffer_unittest.cc
index 3e7396d..8beae43 100644
--- a/rtc_base/buffer_unittest.cc
+++ b/rtc_base/buffer_unittest.cc
@@ -447,7 +447,7 @@
EXPECT_EQ(kObsidian, buf[2].stone);
}
-TEST(BufferTest, DieOnUseAfterMove) {
+TEST(BufferDeathTest, DieOnUseAfterMove) {
Buffer buf(17);
Buffer buf2 = std::move(buf);
EXPECT_EQ(buf2.size(), 17u);
diff --git a/rtc_base/checks_unittest.cc b/rtc_base/checks_unittest.cc
index e6e094e..91e04cf6 100644
--- a/rtc_base/checks_unittest.cc
+++ b/rtc_base/checks_unittest.cc
@@ -19,7 +19,7 @@
}
#if GTEST_HAS_DEATH_TEST && !defined(WEBRTC_ANDROID)
-TEST(ChecksTest, Checks) {
+TEST(ChecksDeathTest, Checks) {
#if RTC_CHECK_MSG_ENABLED
EXPECT_DEATH(FATAL() << "message",
"\n\n#\n"
diff --git a/rtc_base/operations_chain_unittest.cc b/rtc_base/operations_chain_unittest.cc
index 968f94c..ed3c924 100644
--- a/rtc_base/operations_chain_unittest.cc
+++ b/rtc_base/operations_chain_unittest.cc
@@ -369,14 +369,15 @@
#if RTC_DCHECK_IS_ON && GTEST_HAS_DEATH_TEST && !defined(WEBRTC_ANDROID)
-TEST(OperationsChainTest, OperationNotInvokingCallbackShouldCrash) {
+TEST(OperationsChainDeathTest, OperationNotInvokingCallbackShouldCrash) {
scoped_refptr<OperationsChain> operations_chain = OperationsChain::Create();
EXPECT_DEATH(
operations_chain->ChainOperation([](std::function<void()> callback) {}),
"");
}
-TEST(OperationsChainTest, OperationInvokingCallbackMultipleTimesShouldCrash) {
+TEST(OperationsChainDeathTest,
+ OperationInvokingCallbackMultipleTimesShouldCrash) {
scoped_refptr<OperationsChain> operations_chain = OperationsChain::Create();
EXPECT_DEATH(
operations_chain->ChainOperation([](std::function<void()> callback) {
diff --git a/rtc_base/strings/string_builder_unittest.cc b/rtc_base/strings/string_builder_unittest.cc
index 84717ad..99dfd86 100644
--- a/rtc_base/strings/string_builder_unittest.cc
+++ b/rtc_base/strings/string_builder_unittest.cc
@@ -59,7 +59,7 @@
// off.
#if (GTEST_HAS_DEATH_TEST && !defined(WEBRTC_ANDROID)) || !RTC_DCHECK_IS_ON
-TEST(SimpleStringBuilder, BufferOverrunConstCharP) {
+TEST(SimpleStringBuilderDeathTest, BufferOverrunConstCharP) {
char sb_buf[4];
SimpleStringBuilder sb(sb_buf);
const char* const msg = "This is just too much";
@@ -71,7 +71,7 @@
#endif
}
-TEST(SimpleStringBuilder, BufferOverrunStdString) {
+TEST(SimpleStringBuilderDeathTest, BufferOverrunStdString) {
char sb_buf[4];
SimpleStringBuilder sb(sb_buf);
sb << 12;
@@ -84,7 +84,7 @@
#endif
}
-TEST(SimpleStringBuilder, BufferOverrunInt) {
+TEST(SimpleStringBuilderDeathTest, BufferOverrunInt) {
char sb_buf[4];
SimpleStringBuilder sb(sb_buf);
constexpr int num = -12345;
@@ -100,7 +100,7 @@
#endif
}
-TEST(SimpleStringBuilder, BufferOverrunDouble) {
+TEST(SimpleStringBuilderDeathTest, BufferOverrunDouble) {
char sb_buf[5];
SimpleStringBuilder sb(sb_buf);
constexpr double num = 123.456;
@@ -113,7 +113,7 @@
#endif
}
-TEST(SimpleStringBuilder, BufferOverrunConstCharPAlreadyFull) {
+TEST(SimpleStringBuilderDeathTest, BufferOverrunConstCharPAlreadyFull) {
char sb_buf[4];
SimpleStringBuilder sb(sb_buf);
sb << 123;
@@ -126,7 +126,7 @@
#endif
}
-TEST(SimpleStringBuilder, BufferOverrunIntAlreadyFull) {
+TEST(SimpleStringBuilderDeathTest, BufferOverrunIntAlreadyFull) {
char sb_buf[4];
SimpleStringBuilder sb(sb_buf);
sb << "xyz";
diff --git a/rtc_base/swap_queue_unittest.cc b/rtc_base/swap_queue_unittest.cc
index 199ac6b..3862d85 100644
--- a/rtc_base/swap_queue_unittest.cc
+++ b/rtc_base/swap_queue_unittest.cc
@@ -135,7 +135,7 @@
}
#if RTC_DCHECK_IS_ON && GTEST_HAS_DEATH_TEST && !defined(WEBRTC_ANDROID)
-TEST(SwapQueueTest, UnsuccessfulItemVerifyFunctor) {
+TEST(SwapQueueDeathTest, UnsuccessfulItemVerifyFunctor) {
// Queue item verifier for the test.
auto minus_2_verifier = [](const int& i) { return i > -2; };
SwapQueue<int, decltype(minus_2_verifier)> queue(2, minus_2_verifier);
@@ -148,7 +148,7 @@
EXPECT_DEATH(result = queue.Insert(&invalid_value), "");
}
-TEST(SwapQueueTest, UnSuccessfulItemVerifyInsert) {
+TEST(SwapQueueDeathTest, UnSuccessfulItemVerifyInsert) {
std::vector<int> template_element(kChunkSize);
SwapQueue<std::vector<int>,
SwapQueueItemVerifier<std::vector<int>, &LengthVerifierFunction>>
@@ -158,7 +158,7 @@
EXPECT_DEATH(result = queue.Insert(&invalid_chunk), "");
}
-TEST(SwapQueueTest, UnSuccessfulItemVerifyRemove) {
+TEST(SwapQueueDeathTest, UnSuccessfulItemVerifyRemove) {
std::vector<int> template_element(kChunkSize);
SwapQueue<std::vector<int>,
SwapQueueItemVerifier<std::vector<int>, &LengthVerifierFunction>>
diff --git a/system_wrappers/source/field_trial_unittest.cc b/system_wrappers/source/field_trial_unittest.cc
index fdabe1b..ada6313 100644
--- a/system_wrappers/source/field_trial_unittest.cc
+++ b/system_wrappers/source/field_trial_unittest.cc
@@ -32,7 +32,7 @@
EXPECT_TRUE(FieldTrialsStringIsValid("Audio/Enabled/B/C/Audio/Enabled/"));
}
-TEST(FieldTrialValidationTest, RejectsBadInputs) {
+TEST(FieldTrialValidationDeathTest, RejectsBadInputs) {
// Bad delimiters
RTC_EXPECT_DEATH(InitFieldTrialsFromString("Audio/EnabledVideo/Disabled/"),
"Invalid field trials string:");
@@ -90,7 +90,7 @@
"Audio/Enabled/Video/Enabled/");
}
-TEST(FieldTrialMergingTest, DchecksBadInput) {
+TEST(FieldTrialMergingDeathTest, DchecksBadInput) {
RTC_EXPECT_DEATH(MergeFieldTrialsStrings("Audio/Enabled/", "garbage"),
"Invalid field trials string:");
}
diff --git a/system_wrappers/source/metrics_unittest.cc b/system_wrappers/source/metrics_unittest.cc
index 9e5bc86..7532b2a 100644
--- a/system_wrappers/source/metrics_unittest.cc
+++ b/system_wrappers/source/metrics_unittest.cc
@@ -114,7 +114,8 @@
}
#if RTC_DCHECK_IS_ON && GTEST_HAS_DEATH_TEST && !defined(WEBRTC_ANDROID)
-TEST_F(MetricsTest, RtcHistogramsCounts_InvalidIndex) {
+using MetricsDeathTest = MetricsTest;
+TEST_F(MetricsDeathTest, RtcHistogramsCounts_InvalidIndex) {
EXPECT_DEATH(RTC_HISTOGRAMS_COUNTS_1000(-1, "Name", kSample), "");
EXPECT_DEATH(RTC_HISTOGRAMS_COUNTS_1000(3, "Name", kSample), "");
EXPECT_DEATH(RTC_HISTOGRAMS_COUNTS_1000(3u, "Name", kSample), "");
diff --git a/video/rtp_video_stream_receiver2_unittest.cc b/video/rtp_video_stream_receiver2_unittest.cc
index c8584fc..d8784e7 100644
--- a/video/rtp_video_stream_receiver2_unittest.cc
+++ b/video/rtp_video_stream_receiver2_unittest.cc
@@ -1112,7 +1112,8 @@
}
#if RTC_DCHECK_IS_ON && GTEST_HAS_DEATH_TEST && !defined(WEBRTC_ANDROID)
-TEST_F(RtpVideoStreamReceiver2Test, RepeatedSecondarySinkDisallowed) {
+using RtpVideoStreamReceiver2DeathTest = RtpVideoStreamReceiver2Test;
+TEST_F(RtpVideoStreamReceiver2DeathTest, RepeatedSecondarySinkDisallowed) {
MockRtpPacketSink secondary_sink;
rtp_video_stream_receiver_->AddSecondarySink(&secondary_sink);
diff --git a/video/rtp_video_stream_receiver_unittest.cc b/video/rtp_video_stream_receiver_unittest.cc
index 510cad3..d561ea4 100644
--- a/video/rtp_video_stream_receiver_unittest.cc
+++ b/video/rtp_video_stream_receiver_unittest.cc
@@ -1110,7 +1110,8 @@
}
#if RTC_DCHECK_IS_ON && GTEST_HAS_DEATH_TEST && !defined(WEBRTC_ANDROID)
-TEST_F(RtpVideoStreamReceiverTest, RepeatedSecondarySinkDisallowed) {
+using RtpVideoStreamReceiverDeathTest = RtpVideoStreamReceiverTest;
+TEST_F(RtpVideoStreamReceiverDeathTest, RepeatedSecondarySinkDisallowed) {
MockRtpPacketSink secondary_sink;
rtp_video_stream_receiver_->AddSecondarySink(&secondary_sink);
