From a5e07cc3db44bcb31b142ca33cd6130a8f25ba6e Mon Sep 17 00:00:00 2001 From: Tommi Date: Tue, 26 May 2020 21:40:37 +0200 Subject: [PATCH] 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 Commit-Queue: Tommi Reviewed-by: Mirko Bonadei Cr-Commit-Position: refs/heads/master@{#31362} --- .../audio_mixer/audio_mixer_impl_unittest.cc | 2 +- .../audio_mixer/frame_combiner_unittest.cc | 4 ++-- .../aec3/adaptive_fir_filter_unittest.cc | 4 ++-- .../aec3/alignment_mixer_unittest.cc | 4 ++-- .../aec3/block_processor_unittest.cc | 10 ++++---- .../coarse_filter_update_gain_unittest.cc | 2 +- .../aec3/decimator_unittest.cc | 8 +++---- .../aec3/echo_canceller3_unittest.cc | 6 ++--- .../aec3/echo_remover_metrics_unittest.cc | 2 +- .../aec3/echo_remover_unittest.cc | 8 +++---- .../aec3/fft_data_unittest.cc | 4 ++-- .../matched_filter_lag_aggregator_unittest.cc | 2 +- .../aec3/matched_filter_unittest.cc | 8 +++---- .../refined_filter_update_gain_unittest.cc | 2 +- .../aec3/render_buffer_unittest.cc | 6 ++--- .../aec3/render_delay_buffer_unittest.cc | 8 +++---- .../aec3/render_delay_controller_unittest.cc | 4 ++-- .../aec3/render_signal_analyzer_unittest.cc | 2 +- .../aec3/subtractor_unittest.cc | 2 +- .../aec3/suppression_filter_unittest.cc | 4 ++-- .../aec3/suppression_gain_unittest.cc | 2 +- .../audio_processing/audio_buffer_unittest.cc | 2 +- .../audio_processing_unittest.cc | 24 ++++++++++--------- .../cascaded_biquad_filter_unittest.cc | 2 +- .../utility/pffft_wrapper_unittest.cc | 11 +++++---- modules/pacing/packet_router_unittest.cc | 9 +++---- rtc_base/bit_buffer_unittest.cc | 2 +- rtc_base/buffer_unittest.cc | 2 +- rtc_base/checks_unittest.cc | 2 +- rtc_base/operations_chain_unittest.cc | 5 ++-- rtc_base/strings/string_builder_unittest.cc | 12 +++++----- rtc_base/swap_queue_unittest.cc | 6 ++--- .../source/field_trial_unittest.cc | 4 ++-- system_wrappers/source/metrics_unittest.cc | 3 ++- video/rtp_video_stream_receiver2_unittest.cc | 3 ++- video/rtp_video_stream_receiver_unittest.cc | 3 ++- 36 files changed, 96 insertions(+), 88 deletions(-) diff --git a/modules/audio_mixer/audio_mixer_impl_unittest.cc b/modules/audio_mixer/audio_mixer_impl_unittest.cc index 53cd45dc5a..383771ce60 100644 --- a/modules/audio_mixer/audio_mixer_impl_unittest.cc +++ b/modules/audio_mixer/audio_mixer_impl_unittest.cc @@ -606,7 +606,7 @@ class HighOutputRateCalculator : public OutputRateCalculator { }; 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 5f024a4a55..4b189a052e 100644 --- a/modules/audio_mixer/frame_combiner_unittest.cc +++ b/modules/audio_mixer/frame_combiner_unittest.cc @@ -89,7 +89,7 @@ TEST(FrameCombiner, BasicApiCallsLimiter) { } // 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, DebugBuildCrashesWithManyChannels) { } } -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 8e4f5d9644..39f4e11192 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 @@ TEST_P(AdaptiveFirFilterOneTwoFourEightRenderChannels, #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 render_delay_buffer( diff --git a/modules/audio_processing/aec3/alignment_mixer_unittest.cc b/modules/audio_processing/aec3/alignment_mixer_unittest.cc index 832e4ea884..03ef06614b 100644 --- a/modules/audio_processing/aec3/alignment_mixer_unittest.cc +++ b/modules/audio_processing/aec3/alignment_mixer_unittest.cc @@ -175,7 +175,7 @@ TEST(AlignmentMixer, FixedMode) { #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, ZeroNumChannels) { , ""); } -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 2b928e877b..911dad4c81 100644 --- a/modules/audio_processing/aec3/block_processor_unittest.cc +++ b/modules/audio_processing/aec3/block_processor_unittest.cc @@ -252,21 +252,21 @@ TEST(BlockProcessor, TestLongerCall) { #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 @@ TEST(BlockProcessor, VerifyRenderNumBandsCheck) { // 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 @@ TEST(BlockProcessor, VerifyCaptureNumBandsCheck) { } // Verifiers that the verification for null ProcessCapture input works. -TEST(BlockProcessor, NullProcessCaptureParameter) { +TEST(BlockProcessorDeathTest, NullProcessCaptureParameter) { EXPECT_DEATH(std::unique_ptr( 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 4185c1adb8..92775cf702 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 @@ std::string ProduceDebugText(size_t delay, int filter_length_blocks) { #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 1e279cea3e..e6f5ea0403 100644 --- a/modules/audio_processing/aec3/decimator_unittest.cc +++ b/modules/audio_processing/aec3/decimator_unittest.cc @@ -103,7 +103,7 @@ TEST(Decimator, NoLeakageFromUpperFrequencies) { #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 x(kBlockSize - 1, 0.f); std::array x_downsampled; @@ -111,14 +111,14 @@ TEST(Decimator, WrongInputSize) { } // Verifies the check for non-null output parameter. -TEST(Decimator, NullOutput) { +TEST(DecimatorDeathTest, NullOutput) { Decimator decimator(4); std::vector 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 x(kBlockSize, 0.f); std::array x_downsampled; @@ -126,7 +126,7 @@ TEST(Decimator, WrongOutputSize) { } // 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 21255f192e..04d93e4db4 100644 --- a/modules/audio_processing/aec3/echo_canceller3_unittest.cc +++ b/modules/audio_processing/aec3/echo_canceller3_unittest.cc @@ -890,7 +890,7 @@ TEST(EchoCanceller3FieldTrials, Aec3SuppressorTuningOverrideOneParam) { #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 @@ TEST(EchoCanceller3InputCheck, WrongCaptureNumBandsCheckVerification) { // 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 @@ TEST(EchoCanceller3InputCheck, NullCaptureProcessingParameter) { // 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 30c6611869..45b30a9c74 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 @@ namespace webrtc { #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 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 e050027c63..77a207659c 100644 --- a/modules/audio_processing/aec3/echo_remover_unittest.cc +++ b/modules/audio_processing/aec3/echo_remover_unittest.cc @@ -91,14 +91,14 @@ TEST_P(EchoRemoverMultiChannel, BasicApiCalls) { // 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::Create(EchoCanceller3Config(), 8001, 1, 1)), ""); } // Verifies the check for the capture block size. -TEST(EchoRemover, WrongCaptureBlockSize) { +TEST(EchoRemoverDeathTest, WrongCaptureBlockSize) { absl::optional delay_estimate; for (auto rate : {16000, 32000, 48000}) { SCOPED_TRACE(ProduceDebugText(rate)); @@ -121,7 +121,7 @@ TEST(EchoRemover, WrongCaptureBlockSize) { // 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 delay_estimate; for (auto rate : {16000, 32000, 48000}) { SCOPED_TRACE(ProduceDebugText(rate)); @@ -143,7 +143,7 @@ TEST(EchoRemover, DISABLED_WrongCaptureNumBands) { } // Verifies the check for non-null capture block. -TEST(EchoRemover, NullCapture) { +TEST(EchoRemoverDeathTest, NullCapture) { absl::optional delay_estimate; std::unique_ptr 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 0812fd6420..9be2680453 100644 --- a/modules/audio_processing/aec3/fft_data_unittest.cc +++ b/modules/audio_processing/aec3/fft_data_unittest.cc @@ -44,12 +44,12 @@ TEST(FftData, TestOptimizations) { #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 e136c89877..8e2a12e6c5 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 @@ TEST(MatchedFilterLagAggregator, DISABLED_PersistentAggregatedLag) { #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 8a6e22eeca..7d9a7d4d0a 100644 --- a/modules/audio_processing/aec3/matched_filter_unittest.cc +++ b/modules/audio_processing/aec3/matched_filter_unittest.cc @@ -375,7 +375,7 @@ TEST(MatchedFilter, NumberOfLagEstimates) { #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 @@ TEST(MatchedFilter, ZeroWindowSize) { } // 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 @@ TEST(MatchedFilter, NullDataDumper) { // 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 @@ TEST(MatchedFilter, DISABLED_BlockSizeMultipleOf4) { // 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 117f34508e..2393fddd6f 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 @@ std::string ProduceDebugText(size_t delay, int filter_length_blocks) { #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 6981f6d510..4559528600 100644 --- a/modules/audio_processing/aec3/render_buffer_unittest.cc +++ b/modules/audio_processing/aec3/render_buffer_unittest.cc @@ -21,21 +21,21 @@ namespace webrtc { #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 35e81319cf..efd4a29920 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 @@ TEST(RenderDelayBuffer, AlignFromDelay) { // 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 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 @@ TEST(RenderDelayBuffer, WrongNumberOfBands) { } // 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 @@ TEST(RenderDelayBuffer, WrongNumberOfChannels) { } // 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 fb7b86a75d..0d3c856466 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 @@ TEST(RenderDelayController, DISABLED_AlignmentWithJitter) { #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> block( 1, std::vector(kBlockSize - 1, 0.f)); EchoCanceller3Config config; @@ -345,7 +345,7 @@ TEST(RenderDelayController, WrongCaptureSize) { // 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 f40fade830..7a48cc4b69 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 @@ std::string ProduceDebugText(size_t num_channels) { #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 72e57879a0..bbc1e4ffc6 100644 --- a/modules/audio_processing/aec3/subtractor_unittest.cc +++ b/modules/audio_processing/aec3/subtractor_unittest.cc @@ -189,7 +189,7 @@ std::string ProduceDebugText(size_t num_render_channels, #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 b55c719fa9..a160bec045 100644 --- a/modules/audio_processing/aec3/suppression_filter_unittest.cc +++ b/modules/audio_processing/aec3/suppression_filter_unittest.cc @@ -50,7 +50,7 @@ void ProduceSinusoid(int sample_rate_hz, #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 cn(1); std::vector cn_high_bands(1); std::vector E(1); @@ -62,7 +62,7 @@ TEST(SuppressionFilter, NullOutput) { } // 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 0452f2e1fb..4fb4cd7142 100644 --- a/modules/audio_processing/aec3/suppression_gain_unittest.cc +++ b/modules/audio_processing/aec3/suppression_gain_unittest.cc @@ -25,7 +25,7 @@ namespace aec3 { #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> E2(1, {0.f}); std::vector> R2(1, {0.f}); std::vector> S2(1); diff --git a/modules/audio_processing/audio_buffer_unittest.cc b/modules/audio_processing/audio_buffer_unittest.cc index 7cb51ca5f1..f3b2ddc689 100644 --- a/modules/audio_processing/audio_buffer_unittest.cc +++ b/modules/audio_processing/audio_buffer_unittest.cc @@ -40,7 +40,7 @@ TEST(AudioBufferTest, SetNumChannelsSetsChannelBuffersNumChannels) { } #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 90413a84be..93ddc97366 100644 --- a/modules/audio_processing/audio_processing_unittest.cc +++ b/modules/audio_processing/audio_processing_unittest.cc @@ -962,49 +962,51 @@ TEST_F(ApmTest, GainControl) { } #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 @@ TEST_F(ApmTest, GainControlDiesOnInvertedAnalogLevelLimits) { 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 @@ TEST_F(ApmTest, ApmDiesOnTooLowAnalogLevel) { 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 @@ TEST(RuntimeSettingTest, TestDefaultCtor) { 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 @@ TEST(RuntimeSettingTest, TestCapturePreGain) { #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 989e362a49..ff7022dba4 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 @@ TEST(CascadedBiquadFilter, TransparentConfiguration) { #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 input = CreateInputWithIncreasingValues(10); std::vector 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 9aed548934..2ad6849cd4 100644 --- a/modules/audio_processing/utility/pffft_wrapper_unittest.cc +++ b/modules/audio_processing/utility/pffft_wrapper_unittest.cc @@ -125,23 +125,24 @@ TEST(PffftTest, CreateWrapperWithValidSize) { #if !defined(NDEBUG) && GTEST_HAS_DEATH_TEST && !defined(WEBRTC_ANDROID) -class PffftInvalidSizeTest : public ::testing::Test, - public ::testing::WithParamInterface {}; +class PffftInvalidSizeDeathTest : public ::testing::Test, + public ::testing::WithParamInterface { +}; -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 75729cb544..79092ea7ee 100644 --- a/modules/pacing/packet_router_unittest.cc +++ b/modules/pacing/packet_router_unittest.cc @@ -406,7 +406,8 @@ TEST_F(PacketRouterTest, SendPacketAssignsTransportSequenceNumbers) { } #if RTC_DCHECK_IS_ON && GTEST_HAS_DEATH_TEST && !defined(WEBRTC_ANDROID) -TEST_F(PacketRouterTest, DoubleRegistrationOfSendModuleDisallowed) { +using PacketRouterDeathTest = PacketRouterTest; +TEST_F(PacketRouterDeathTest, DoubleRegistrationOfSendModuleDisallowed) { NiceMock module; constexpr bool remb_candidate = false; // Value irrelevant. @@ -417,7 +418,7 @@ TEST_F(PacketRouterTest, DoubleRegistrationOfSendModuleDisallowed) { packet_router_.RemoveSendRtpModule(&module); } -TEST_F(PacketRouterTest, DoubleRegistrationOfReceiveModuleDisallowed) { +TEST_F(PacketRouterDeathTest, DoubleRegistrationOfReceiveModuleDisallowed) { NiceMock module; constexpr bool remb_candidate = false; // Value irrelevant. @@ -428,13 +429,13 @@ TEST_F(PacketRouterTest, DoubleRegistrationOfReceiveModuleDisallowed) { packet_router_.RemoveReceiveRtpModule(&module); } -TEST_F(PacketRouterTest, RemovalOfNeverAddedSendModuleDisallowed) { +TEST_F(PacketRouterDeathTest, RemovalOfNeverAddedSendModuleDisallowed) { NiceMock module; EXPECT_DEATH(packet_router_.RemoveSendRtpModule(&module), ""); } -TEST_F(PacketRouterTest, RemovalOfNeverAddedReceiveModuleDisallowed) { +TEST_F(PacketRouterDeathTest, RemovalOfNeverAddedReceiveModuleDisallowed) { NiceMock module; EXPECT_DEATH(packet_router_.RemoveReceiveRtpModule(&module), ""); diff --git a/rtc_base/bit_buffer_unittest.cc b/rtc_base/bit_buffer_unittest.cc index b3521b4951..441cd26495 100644 --- a/rtc_base/bit_buffer_unittest.cc +++ b/rtc_base/bit_buffer_unittest.cc @@ -142,7 +142,7 @@ TEST(BitBufferTest, ReadBits) { 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 3e7396dd2c..8beae43cf9 100644 --- a/rtc_base/buffer_unittest.cc +++ b/rtc_base/buffer_unittest.cc @@ -447,7 +447,7 @@ TEST(BufferTest, TestStruct) { 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 e6e094e597..91e04cf6a1 100644 --- a/rtc_base/checks_unittest.cc +++ b/rtc_base/checks_unittest.cc @@ -19,7 +19,7 @@ TEST(ChecksTest, ExpressionNotEvaluatedWhenCheckPassing) { } #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 968f94c060..ed3c924998 100644 --- a/rtc_base/operations_chain_unittest.cc +++ b/rtc_base/operations_chain_unittest.cc @@ -369,14 +369,15 @@ TEST(OperationsChainTest, FunctorIsNotDestroyedWhileExecuting) { #if RTC_DCHECK_IS_ON && GTEST_HAS_DEATH_TEST && !defined(WEBRTC_ANDROID) -TEST(OperationsChainTest, OperationNotInvokingCallbackShouldCrash) { +TEST(OperationsChainDeathTest, OperationNotInvokingCallbackShouldCrash) { scoped_refptr operations_chain = OperationsChain::Create(); EXPECT_DEATH( operations_chain->ChainOperation([](std::function callback) {}), ""); } -TEST(OperationsChainTest, OperationInvokingCallbackMultipleTimesShouldCrash) { +TEST(OperationsChainDeathTest, + OperationInvokingCallbackMultipleTimesShouldCrash) { scoped_refptr operations_chain = OperationsChain::Create(); EXPECT_DEATH( operations_chain->ChainOperation([](std::function callback) { diff --git a/rtc_base/strings/string_builder_unittest.cc b/rtc_base/strings/string_builder_unittest.cc index 84717ad1d1..99dfd86292 100644 --- a/rtc_base/strings/string_builder_unittest.cc +++ b/rtc_base/strings/string_builder_unittest.cc @@ -59,7 +59,7 @@ TEST(SimpleStringBuilder, StdString) { // 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 @@ TEST(SimpleStringBuilder, BufferOverrunConstCharP) { #endif } -TEST(SimpleStringBuilder, BufferOverrunStdString) { +TEST(SimpleStringBuilderDeathTest, BufferOverrunStdString) { char sb_buf[4]; SimpleStringBuilder sb(sb_buf); sb << 12; @@ -84,7 +84,7 @@ TEST(SimpleStringBuilder, BufferOverrunStdString) { #endif } -TEST(SimpleStringBuilder, BufferOverrunInt) { +TEST(SimpleStringBuilderDeathTest, BufferOverrunInt) { char sb_buf[4]; SimpleStringBuilder sb(sb_buf); constexpr int num = -12345; @@ -100,7 +100,7 @@ TEST(SimpleStringBuilder, BufferOverrunInt) { #endif } -TEST(SimpleStringBuilder, BufferOverrunDouble) { +TEST(SimpleStringBuilderDeathTest, BufferOverrunDouble) { char sb_buf[5]; SimpleStringBuilder sb(sb_buf); constexpr double num = 123.456; @@ -113,7 +113,7 @@ TEST(SimpleStringBuilder, BufferOverrunDouble) { #endif } -TEST(SimpleStringBuilder, BufferOverrunConstCharPAlreadyFull) { +TEST(SimpleStringBuilderDeathTest, BufferOverrunConstCharPAlreadyFull) { char sb_buf[4]; SimpleStringBuilder sb(sb_buf); sb << 123; @@ -126,7 +126,7 @@ TEST(SimpleStringBuilder, BufferOverrunConstCharPAlreadyFull) { #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 199ac6b185..3862d850fa 100644 --- a/rtc_base/swap_queue_unittest.cc +++ b/rtc_base/swap_queue_unittest.cc @@ -135,7 +135,7 @@ TEST(SwapQueueTest, SuccessfulItemVerifyFunctor) { } #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 queue(2, minus_2_verifier); @@ -148,7 +148,7 @@ TEST(SwapQueueTest, UnsuccessfulItemVerifyFunctor) { EXPECT_DEATH(result = queue.Insert(&invalid_value), ""); } -TEST(SwapQueueTest, UnSuccessfulItemVerifyInsert) { +TEST(SwapQueueDeathTest, UnSuccessfulItemVerifyInsert) { std::vector template_element(kChunkSize); SwapQueue, SwapQueueItemVerifier, &LengthVerifierFunction>> @@ -158,7 +158,7 @@ TEST(SwapQueueTest, UnSuccessfulItemVerifyInsert) { EXPECT_DEATH(result = queue.Insert(&invalid_chunk), ""); } -TEST(SwapQueueTest, UnSuccessfulItemVerifyRemove) { +TEST(SwapQueueDeathTest, UnSuccessfulItemVerifyRemove) { std::vector template_element(kChunkSize); SwapQueue, SwapQueueItemVerifier, &LengthVerifierFunction>> diff --git a/system_wrappers/source/field_trial_unittest.cc b/system_wrappers/source/field_trial_unittest.cc index fdabe1b7e6..ada6313e67 100644 --- a/system_wrappers/source/field_trial_unittest.cc +++ b/system_wrappers/source/field_trial_unittest.cc @@ -32,7 +32,7 @@ TEST(FieldTrialValidationTest, AcceptsValidInputs) { 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 @@ TEST(FieldTrialMergingTest, MergesValidInput) { "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 9e5bc86ba9..7532b2ad83 100644 --- a/system_wrappers/source/metrics_unittest.cc +++ b/system_wrappers/source/metrics_unittest.cc @@ -114,7 +114,8 @@ TEST_F(MetricsTest, RtcHistogramsCounts_AddSample) { } #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 c8584fcd55..d8784e7d45 100644 --- a/video/rtp_video_stream_receiver2_unittest.cc +++ b/video/rtp_video_stream_receiver2_unittest.cc @@ -1112,7 +1112,8 @@ TEST_F(RtpVideoStreamReceiver2DependencyDescriptorTest, } #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 510cad37c1..d561ea4d69 100644 --- a/video/rtp_video_stream_receiver_unittest.cc +++ b/video/rtp_video_stream_receiver_unittest.cc @@ -1110,7 +1110,8 @@ TEST_F(RtpVideoStreamReceiverDependencyDescriptorTest, } #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);