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webrtc / src / 876d0c9881eab8e7f8389812eb3738bdd374aa22
commit876d0c9881eab8e7f8389812eb3738bdd374aa22[log] [tgz]
authorMirko Bonadei <mbonadei@webrtc.org>Tue Jun 04 21:15:56 2024
committerWebRTC LUCI CQ <webrtc-scoped@luci-project-accounts.iam.gserviceaccount.com>Wed Jun 05 07:07:37 2024
treea1bc3934507ab3b7ccd5933f8f5d25a68baac23a
parent06e88bbb5aabdbc2614e51739d79a756c9e4d34b [diff]
Fix use-of-uninitialized-value in NetEq tests.

The new version of MSan (rolled by [1]) detects the following:

```
==39908==WARNING: MemorySanitizer: use-of-uninitialized-value
    #0 0x5591400a52ef in GetPlayoutDelayMs ./../../modules/audio_coding/neteq/decision_logic.cc:466:35
    #1 0x5591400a52ef in webrtc::DecisionLogic::ExpectedPacketAvailable(webrtc::NetEqController::NetEqStatus) ./../../modules/audio_coding/neteq/decision_logic.cc:311:36
    #2 0x5591400a39e9 in webrtc::DecisionLogic::GetDecision(webrtc::NetEqController::NetEqStatus const&, bool*) ./../../modules/audio_coding/neteq/decision_logic.cc:0:0
    #3 0x55913cf590c9 in webrtc::DecisionLogicTest_PreemptiveExpand_Test::TestBody() ./../../modules/audio_coding/neteq/decision_logic_unittest.cc:139:3
    #4 0x55913ef28283 in HandleExceptionsInMethodIfSupported<testing::Test, void> ./../../third_party/googletest/src/googletest/src/gtest.cc:0:3
    #5 0x55913ef28283 in testing::Test::Run() ./../../third_party/googletest/src/googletest/src/gtest.cc:2710:5
    #6 0x55913ef2ab46 in testing::TestInfo::Run() ./../../third_party/googletest/src/googletest/src/gtest.cc:2856:11
    #7 0x55913ef2da34 in testing::TestSuite::Run() ./../../third_party/googletest/src/googletest/src/gtest.cc:3034:30
    #8 0x55913ef621e8 in testing::internal::UnitTestImpl::RunAllTests() ./../../third_party/googletest/src/googletest/src/gtest.cc:5964:44
    #9 0x55913ef60f54 in HandleExceptionsInMethodIfSupported<testing::internal::UnitTestImpl, bool> ./../../third_party/googletest/src/googletest/src/gtest.cc:0:0
    #10 0x55913ef60f54 in testing::UnitTest::Run() ./../../third_party/googletest/src/googletest/src/gtest.cc:5543:10
    #11 0x55913ee1a944 in RUN_ALL_TESTS ./../../third_party/googletest/src/googletest/include/gtest/gtest.h:2334:73
    #12 0x55913ee1a944 in webrtc::(anonymous namespace)::TestMainImpl::Run(int, char**) ./../../test/test_main_lib.cc:203:21
    #13 0x55913cbd36b8 in main ./../../test/test_main.cc:72:16
    #14 0x7fdb18c73082 in __libc_start_main /build/glibc-LcI20x/glibc-2.31/csu/../csu/libc-start.c:308:16
    #15 0x55913cb3e1a9 in _start ??:0:0
```

[1] - https://webrtc-review.googlesource.com/c/src/+/353620

Bug: b/344970813
Change-Id: I9b5d7791e68b4c494168ba9f007a3099ae21fed4
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/353581
Auto-Submit: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Jakob Ivarsson‎ <jakobi@webrtc.org>
Commit-Queue: Jakob Ivarsson‎ <jakobi@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#42433}
1 file changed
tree: a1bc3934507ab3b7ccd5933f8f5d25a68baac23a
  1. api/
  2. audio/
  3. build_overrides/
  4. call/
  5. common_audio/
  6. common_video/
  7. data/
  8. docs/
  9. examples/
  10. experiments/
  11. g3doc/
  12. infra/
  13. logging/
  14. media/
  15. modules/
  16. net/
  17. p2p/
  18. pc/
  19. resources/
  20. rtc_base/
  21. rtc_tools/
  22. sdk/
  23. stats/
  24. system_wrappers/
  25. test/
  26. tools_webrtc/
  27. video/
  28. .clang-format
  29. .git-blame-ignore-revs
  30. .gitignore
  31. .gn
  32. .mailmap
  33. .style.yapf
  34. .vpython3
  35. AUTHORS
  36. BUILD.gn
  37. CODE_OF_CONDUCT.md
  38. codereview.settings
  39. DEPS
  40. DIR_METADATA
  41. ENG_REVIEW_OWNERS
  42. LICENSE
  43. license_template.txt
  44. native-api.md
  45. OWNERS
  46. OWNERS_INFRA
  47. PATENTS
  48. PRESUBMIT.py
  49. presubmit_test.py
  50. presubmit_test_mocks.py
  51. pylintrc
  52. pylintrc_old_style
  53. README.chromium
  54. README.md
  55. WATCHLISTS
  56. webrtc.gni
  57. webrtc_lib_link_test.cc
  58. 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