Fix Multiple SSL clocks issue when running PeerConnectionIntegrationTestWithFakeClock.

This is needed because reference deletion of DtlsTransport and OpenSSLStreamAdapter is queued asynchronously on the network_thread.

# Fatal error in: ../../rtc_base/openssl_stream_adapter.cc, line 1121
# last system error: 0
# Check failed: g_clock_for_testing == clock
# Multiple SSL clocks for testing is not implemented*** SIGABRT received at time=1781709018 on cpu 4 ***

https://ci.chromium.org/ui/tests/p/webrtc/clusters?interval=24h&selectedMetrics=human-cls-failed-presubmit%2Ccritical-failures-exonerated%2Cfailures&orderBy=critical-failures-exonerated&orderDir=desc&q=PeerConnectionIntegrationTestWithFakeClock

Bug: None
Change-Id: Ie1068ddd338fbfb33883761fa19d0495162eddff
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/483460
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Commit-Queue: Jeremy Leconte <jleconte@google.com>
Cr-Commit-Position: refs/heads/main@{#48038}
1 file changed
tree: 945f2b3c519d5e41dfc7a8258f2c1a7302d5633e
  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. .vpython3
  41. .yapfignore
  42. AUTHORS
  43. BUILD.gn
  44. CODE_OF_CONDUCT.md
  45. codereview.settings
  46. DEPS
  47. DIR_METADATA
  48. ENG_REVIEW_OWNERS
  49. GEMINI.md
  50. LICENSE
  51. license_template.txt
  52. native-api.md
  53. OWNERS
  54. OWNERS_INFRA
  55. PATENTS
  56. PRESUBMIT.py
  57. presubmit_test.py
  58. presubmit_test_mocks.py
  59. pylintrc
  60. pylintrc_old_style
  61. README.chromium
  62. README.md
  63. unsafe_buffers_paths.txt
  64. WATCHLISTS
  65. webrtc.gni
  66. webrtc_lib_link_test.cc
  67. 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