| /* |
| * Copyright (c) 2018 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 "test/test_main_lib.h" |
| |
| #include <fstream> |
| #include <string> |
| |
| #include "absl/memory/memory.h" |
| #include "rtc_base/checks.h" |
| #include "rtc_base/flags.h" |
| #include "rtc_base/logging.h" |
| #include "rtc_base/ssl_adapter.h" |
| #include "rtc_base/ssl_stream_adapter.h" |
| #include "rtc_base/thread.h" |
| #include "system_wrappers/include/field_trial.h" |
| #include "system_wrappers/include/metrics.h" |
| #include "test/field_trial.h" |
| #include "test/gmock.h" |
| #include "test/gtest.h" |
| #include "test/testsupport/file_utils.h" |
| #include "test/testsupport/perf_test.h" |
| |
| #if defined(WEBRTC_WIN) |
| #include "rtc_base/win32_socket_init.h" |
| #endif |
| |
| #if defined(WEBRTC_IOS) |
| #include "test/ios/test_support.h" |
| |
| WEBRTC_DEFINE_string(NSTreatUnknownArgumentsAsOpen, |
| "", |
| "Intentionally ignored flag intended for iOS simulator."); |
| WEBRTC_DEFINE_string(ApplePersistenceIgnoreState, |
| "", |
| "Intentionally ignored flag intended for iOS simulator."); |
| WEBRTC_DEFINE_bool( |
| save_chartjson_result, |
| false, |
| "Store the perf results in Documents/perf_result.json in the format " |
| "described by " |
| "https://github.com/catapult-project/catapult/blob/master/dashboard/docs/" |
| "data-format.md."); |
| |
| #else |
| |
| WEBRTC_DEFINE_string( |
| isolated_script_test_output, |
| "", |
| "Path to output an empty JSON file which Chromium infra requires."); |
| |
| WEBRTC_DEFINE_string( |
| isolated_script_test_perf_output, |
| "", |
| "Path where the perf results should be stored in the JSON format described " |
| "by " |
| "https://github.com/catapult-project/catapult/blob/master/dashboard/docs/" |
| "data-format.md."); |
| |
| #endif |
| |
| WEBRTC_DEFINE_bool(logs, true, "print logs to stderr"); |
| WEBRTC_DEFINE_bool(verbose, false, "verbose logs to stderr"); |
| |
| WEBRTC_DEFINE_string( |
| force_fieldtrials, |
| "", |
| "Field trials control experimental feature code which can be forced. " |
| "E.g. running with --force_fieldtrials=WebRTC-FooFeature/Enable/" |
| " will assign the group Enable to field trial WebRTC-FooFeature."); |
| |
| WEBRTC_DEFINE_bool(help, false, "Print this message."); |
| |
| namespace webrtc { |
| |
| namespace { |
| |
| class TestMainImpl : public TestMain { |
| public: |
| int Init(int* argc, char* argv[]) override { |
| ::testing::InitGoogleMock(argc, argv); |
| |
| // Default to LS_INFO, even for release builds to provide better test |
| // logging. |
| if (rtc::LogMessage::GetLogToDebug() > rtc::LS_INFO) |
| rtc::LogMessage::LogToDebug(rtc::LS_INFO); |
| |
| if (rtc::FlagList::SetFlagsFromCommandLine(argc, argv, false)) { |
| return 1; |
| } |
| if (FLAG_help) { |
| rtc::FlagList::Print(nullptr, false); |
| return 0; |
| } |
| |
| if (FLAG_verbose) |
| rtc::LogMessage::LogToDebug(rtc::LS_VERBOSE); |
| |
| rtc::LogMessage::SetLogToStderr(FLAG_logs || FLAG_verbose); |
| |
| // TODO(bugs.webrtc.org/9792): we need to reference something from |
| // fileutils.h so that our downstream hack where we replace fileutils.cc |
| // works. Otherwise the downstream flag implementation will take over and |
| // botch the flag introduced by the hack. Remove this awful thing once the |
| // downstream implementation has been eliminated. |
| (void)webrtc::test::JoinFilename("horrible", "hack"); |
| |
| // InitFieldTrialsFromString stores the char*, so the char array must |
| // outlive the application. |
| webrtc::field_trial::InitFieldTrialsFromString(FLAG_force_fieldtrials); |
| webrtc::metrics::Enable(); |
| |
| #if defined(WEBRTC_WIN) |
| winsock_init_ = absl::make_unique<rtc::WinsockInitializer>(); |
| #endif |
| |
| // Initialize SSL which are used by several tests. |
| rtc::InitializeSSL(); |
| rtc::SSLStreamAdapter::EnableTimeCallbackForTesting(); |
| |
| // Ensure that main thread gets wrapped as an rtc::Thread. |
| // TODO(bugs.webrt.org/9714): It might be better to avoid wrapping the main |
| // thread, or leave it to individual tests that need it. But as long as we |
| // have automatic thread wrapping, we need this to avoid that some other |
| // random thread (which one depending on which tests are run) gets |
| // automatically wrapped. |
| rtc::ThreadManager::Instance()->WrapCurrentThread(); |
| RTC_CHECK(rtc::Thread::Current()); |
| return 0; |
| } |
| |
| int Run(int argc, char* argv[]) override { |
| #if defined(WEBRTC_IOS) |
| rtc::test::InitTestSuite(RUN_ALL_TESTS, argc, argv, |
| FLAG_save_chartjson_result); |
| rtc::test::RunTestsFromIOSApp(); |
| return 0; |
| #else |
| int exit_code = RUN_ALL_TESTS(); |
| |
| std::string chartjson_result_file = FLAG_isolated_script_test_perf_output; |
| if (!chartjson_result_file.empty()) { |
| webrtc::test::WritePerfResults(chartjson_result_file); |
| } |
| |
| std::string result_filename = FLAG_isolated_script_test_output; |
| if (!result_filename.empty()) { |
| std::ofstream result_file(result_filename); |
| result_file << "{\"version\": 3}"; |
| result_file.close(); |
| } |
| |
| #if defined(ADDRESS_SANITIZER) || defined(LEAK_SANITIZER) || \ |
| defined(MEMORY_SANITIZER) || defined(THREAD_SANITIZER) || \ |
| defined(UNDEFINED_SANITIZER) |
| // We want the test flagged as failed only for sanitizer defects, |
| // in which case the sanitizer will override exit code with 66. |
| return 0; |
| #endif |
| |
| return exit_code; |
| #endif |
| } |
| |
| ~TestMainImpl() override = default; |
| |
| private: |
| #if defined(WEBRTC_WIN) |
| std::unique_ptr<rtc::WinsockInitializer> winsock_init_; |
| #endif |
| }; |
| |
| } // namespace |
| |
| std::unique_ptr<TestMain> TestMain::Create() { |
| return absl::make_unique<TestMainImpl>(); |
| } |
| |
| } // namespace webrtc |