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webrtc / src / 5bf0bb3ed26fbe9eeb752ec55159045b5c6a3c5f
commit5bf0bb3ed26fbe9eeb752ec55159045b5c6a3c5f[log] [tgz]
authorMirko Bonadei <mbonadei@webrtc.org>Tue Aug 17 08:22:09 2021
committerWebRTC LUCI CQ <webrtc-scoped@luci-project-accounts.iam.gserviceaccount.com>Tue Aug 17 09:22:57 2021
tree80280d517e0bb139d7e0a7fe413fd6bd5842d13b
parent14ef6338b088e0d5b0d82879a889a478d937f9c6 [diff]
Enable WebRTC-UseStandardBytesStats in E2E tests by default.

Before this CL PeerConnectionE2EQualityTestSmokeTest was actually
overwriting this field trial:
[ RUN      ] PeerConnectionE2EQualityTestSmokeTest.Smoke
(field_trial.cc:140): Setting field trial string:WebRTC-UseStandardBytesStats/Enabled/
(field_trial.cc:140): Setting field trial string:
(network_emulation.cc:480): Created emulated endpoint 192.168.0.0 (); id=1
(network_emulation.cc:480): Created emulated endpoint 192.168.0.1 (); id=2
(field_trial.cc:140): Setting field trial string:WebRTC-FlexFEC-03-Advertised/Enabled/WebRTC-FlexFEC-03/Enabled/

After this CL it is instead used:
[ RUN      ] PeerConnectionE2EQualityTestSmokeTest.Smoke
(network_emulation.cc:480): Created emulated endpoint 192.168.0.0 (); id=1
(network_emulation.cc:480): Created emulated endpoint 192.168.0.1 (); id=2
(field_trial.cc:140): Setting field trial string:WebRTC-UseStandardBytesStats/Enabled/WebRTC-FlexFEC-03-Advertised/Enabled/WebRTC-FlexFEC-03/Enabled/

This CL also removes the non effective field trial override in
test/pc/e2e/peer_connection_e2e_smoke_test.cc which was unset as soon
as the variable was going out of scope.

Bug: b/186198412
Change-Id: I1698407e2c490a80c1f835cd591624446cf993fc
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/229023
Reviewed-by: Artem Titov <titovartem@webrtc.org>
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#34779}
2 files changed
tree: 80280d517e0bb139d7e0a7fe413fd6bd5842d13b
  1. api/
  2. audio/
  3. build_overrides/
  4. call/
  5. common_audio/
  6. common_video/
  7. data/
  8. docs/
  9. examples/
  10. g3doc/
  11. logging/
  12. media/
  13. modules/
  14. net/
  15. p2p/
  16. pc/
  17. resources/
  18. rtc_base/
  19. rtc_tools/
  20. sdk/
  21. stats/
  22. system_wrappers/
  23. test/
  24. tools_webrtc/
  25. video/
  26. .clang-format
  27. .git-blame-ignore-revs
  28. .gitignore
  29. .gn
  30. .vpython
  31. AUTHORS
  32. BUILD.gn
  33. CODE_OF_CONDUCT.md
  34. codereview.settings
  35. DEPS
  36. DIR_METADATA
  37. ENG_REVIEW_OWNERS
  38. g3doc.lua
  39. LICENSE
  40. license_template.txt
  41. native-api.md
  42. OWNERS
  43. PATENTS
  44. PRESUBMIT.py
  45. presubmit_test.py
  46. presubmit_test_mocks.py
  47. pylintrc
  48. README.chromium
  49. README.md
  50. WATCHLISTS
  51. webrtc.gni
  52. webrtc_lib_link_test.cc
  53. 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