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android / platform / external / webrtc / 85a126ec5d76457381333dd5abdde36da0f06c1c
commit85a126ec5d76457381333dd5abdde36da0f06c1c[log] [tgz]
authorAlessio Bazzica <alessiob@webrtc.org>Thu Aug 11 15:48:54 2022 +0200
committerWebRTC LUCI CQ <webrtc-scoped@luci-project-accounts.iam.gserviceaccount.com>Fri Aug 12 12:42:43 2022 +0000
tree28f3f15d2dfebb0706212eec7f5e37ee9f4db25d
parentb69b81944c4abdb65294624435d48fbd2d83a64a [diff]
`ApmTest.VerifyDebugDump(Int|Float)` more insightful with failures

Instead of showing individual byte differences, this CL detects
differences in the expected and actual byte streams of the evaluated
AEC dump and, if detected, parses the `audioproc::Event` proto lite
messages and calls EXPECT_EQ() for a subset of individual (sub-)fields.

Note that messages are parsed only if the byte streams of each message
pair do not match, so with no failures the test runs at no extra cost.
Plus, the the added funcionality can only be enabled for local
debugging by flipping the `kDumpWhenExpectMessageEqFails` flag - a
code change cannot land if the flag is set to true.

Note that `MessageDifferencer` (see [1]) could not be used because
it is not implemented for `MessageLite` protos.

[1] https://developers.google.com/protocol-buffers/docs/reference/cpp/google.protobuf.util.message_differencer

Bug: b/241923537
Change-Id: I8e0eda3b1ecfe06945b6dad5ee8871f8200d76d7
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/270922
Reviewed-by: Jesus de Vicente Pena <devicentepena@webrtc.org>
Commit-Queue: Alessio Bazzica <alessiob@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#37765}
1 file changed
tree: 28f3f15d2dfebb0706212eec7f5e37ee9f4db25d
  1. api/
  2. audio/
  3. build_overrides/
  4. call/
  5. common_audio/
  6. common_video/
  7. data/
  8. docs/
  9. examples/
  10. g3doc/
  11. infra/
  12. logging/
  13. media/
  14. modules/
  15. net/
  16. p2p/
  17. pc/
  18. resources/
  19. rtc_base/
  20. rtc_tools/
  21. sdk/
  22. stats/
  23. system_wrappers/
  24. test/
  25. tools_webrtc/
  26. video/
  27. .clang-format
  28. .git-blame-ignore-revs
  29. .gitignore
  30. .gn
  31. .mailmap
  32. .style.yapf
  33. .vpython
  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. g3doc.lua
  43. LICENSE
  44. license_template.txt
  45. native-api.md
  46. OWNERS
  47. PATENTS
  48. PRESUBMIT.py
  49. presubmit_test.py
  50. presubmit_test_mocks.py
  51. pylintrc
  52. README.chromium
  53. README.md
  54. WATCHLISTS
  55. webrtc.gni
  56. webrtc_lib_link_test.cc
  57. 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