Migrate LogTest to LoggingConfig

InitializeLogging() succeeds only once per process, so the logging
initialization tests ran as EXPECT_EXIT death tests and relied on a
WEBRTC_TEST_SKIP_LOGGING_INIT environment variable in
TestMainImpl::Init(). The other LogTest tests used legacy static setters
such as AddLogToStream(), SetLogQueueNames() and the deprecated
LogThreads().

- Add ResetLoggingForTesting(). It swaps the active LoggingConfig for a
  default one and lets the next InitializeLogging() call succeed, or
  restores a previously saved config. The test-only state is only read
  by InitializeLogging(), so the per-message logging path is unchanged.
- Convert LogTest to a fixture that saves and restores the config set up
  by TestMainImpl::Init() and logs through LoggingConfig sinks. Remove
  the death tests, the environment variable, a duplicate queue name test
  and ImplicitInitialization, which cannot run in-process and whose
  death-test version passed regardless.
- Filter LoggingConfig sinks by LoggingConfig::min_severity(). They were
  filtered by LogSink::min_severity_, which only AddLogToStream() sets,
  so config sinks always used LS_INFO and kept the global threshold at
  LS_INFO or lower.
- Log the --verbose warning in TestMainImpl::Init() after
  InitializeLogging(). Logging it first triggered implicit
  initialization in debug builds, so the logging flags were ignored.

Bug: webrtc:42234107
Change-Id: Icd992a3e1d0e252df09ccf52609ff91106eecd97
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/510041
Reviewed-by: Fredrik Solenberg <solenberg@webrtc.org>
Commit-Queue: Tomas Gunnarsson <tommi@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#49043}
5 files changed
tree: fccf668f8b44e710543f32076b82dc173e13556f
  1. .agents/
  2. agents/
  3. api/
  4. audio/
  5. build_overrides/
  6. call/
  7. common_audio/
  8. common_video/
  9. data/
  10. docs/
  11. examples/
  12. experiments/
  13. g3doc/
  14. infra/
  15. logging/
  16. media/
  17. modules/
  18. net/
  19. p2p/
  20. pc/
  21. resources/
  22. rtc_base/
  23. rtc_tools/
  24. rust/
  25. sdk/
  26. stats/
  27. system_wrappers/
  28. test/
  29. tools_webrtc/
  30. video/
  31. .clang-format
  32. .clang-tidy
  33. .git-blame-ignore-revs
  34. .gitignore
  35. .gn
  36. .mailmap
  37. .rustfmt.toml
  38. .style.mdformat
  39. .style.yapf
  40. .yapfignore
  41. AUTHORS
  42. BUILD.gn
  43. CODE_OF_CONDUCT.md
  44. codereview.settings
  45. DEPS
  46. DIR_METADATA
  47. ENG_REVIEW_OWNERS
  48. GEMINI.md
  49. LICENSE
  50. license_template.txt
  51. native-api.md
  52. OWNERS
  53. OWNERS_INFRA
  54. PATENTS
  55. PRESUBMIT.py
  56. presubmit_test.py
  57. presubmit_test_mocks.py
  58. pylintrc
  59. pylintrc_old_style
  60. README.chromium
  61. README.md
  62. unsafe_buffers_paths.txt
  63. vpython.toml
  64. vpython.toml.uv.lock
  65. WATCHLISTS
  66. webrtc.gni
  67. webrtc_lib_link_test.cc
  68. whitespace.txt
README.md

WebRTC is a free, open software project that provides browsers and mobile applications with Real-Time Communications (RTC) capabilities via simple APIs. The WebRTC components have been optimized to best serve this purpose.

Our mission: To enable rich, high-quality RTC applications to be developed for the browser, mobile platforms, and IoT devices, and allow them all to communicate via a common set of protocols.

The WebRTC initiative is a project supported by Google, Mozilla and Opera, amongst others.

Development

See here for instructions on how to get started developing with the native code.

Authoritative list of directories that contain the native API header files.

More info