Google Git
Sign in
webrtc / src / d546186b89806ce3972128cc460ea02f82ba3575
commitd546186b89806ce3972128cc460ea02f82ba3575[log] [tgz]
authorBjörn Terelius <terelius@webrtc.org>Tue Oct 27 10:51:34 2020
committerCommit Bot <commit-bot@chromium.org>Tue Oct 27 10:51:46 2020
tree7e338870c204501e9a6c20988cd41e48a0de8698
parentf8b5bfeaf2471c79744605fca5851839fda726df [diff]
Revert "Reland "Default enable WebRTC-SendSideBwe-WithOverhead.""

This reverts commit 1dbe30c7e895c7eb4da51c968a7a8897f25ad7e6.

Reason for revert: Speculative revert due to failing tests.

Original change's description:
> Reland "Default enable WebRTC-SendSideBwe-WithOverhead."
>
> This is a reland of 87c1950841c3f5e465e1663cc922717ce191e192
>
> Original change's description:
> > Default enable WebRTC-SendSideBwe-WithOverhead.
> >
> > Bug: webrtc:6762
> > Change-Id: I18ace06a33b3b60d5a19796d4769f70cd977d604
> > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/188801
> > Commit-Queue: Jakob Ivarsson <jakobi@webrtc.org>
> > Reviewed-by: Stefan Holmer <stefan@webrtc.org>
> > Reviewed-by: Ali Tofigh <alito@webrtc.org>
> > Cr-Commit-Position: refs/heads/master@{#32472}
>
> Bug: webrtc:6762
> Change-Id: Icf096a8755d29600a13bd08b1f22f5a79de21e90
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/190143
> Reviewed-by: Ali Tofigh <alito@webrtc.org>
> Reviewed-by: Stefan Holmer <stefan@webrtc.org>
> Commit-Queue: Jakob Ivarsson <jakobi@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#32492}

TBR=stefan@webrtc.org,jakobi@webrtc.org,alito@webrtc.org

Change-Id: I7e0378788576236059627cf8c3bad58cd70aff7e
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: webrtc:6762
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/190500
Reviewed-by: Björn Terelius <terelius@webrtc.org>
Commit-Queue: Björn Terelius <terelius@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#32504}
11 files changed
tree: 7e338870c204501e9a6c20988cd41e48a0de8698
  1. api/
  2. audio/
  3. build_overrides/
  4. call/
  5. common_audio/
  6. common_video/
  7. data/
  8. docs/
  9. examples/
  10. logging/
  11. media/
  12. modules/
  13. p2p/
  14. pc/
  15. resources/
  16. rtc_base/
  17. rtc_tools/
  18. sdk/
  19. stats/
  20. style-guide/
  21. system_wrappers/
  22. test/
  23. tools_webrtc/
  24. video/
  25. .clang-format
  26. .git-blame-ignore-revs
  27. .gitignore
  28. .gn
  29. .vpython
  30. abseil-in-webrtc.md
  31. AUTHORS
  32. BUILD.gn
  33. CODE_OF_CONDUCT.md
  34. codereview.settings
  35. DEPS
  36. ENG_REVIEW_OWNERS
  37. LICENSE
  38. license_template.txt
  39. native-api.md
  40. OWNERS
  41. PATENTS
  42. PRESUBMIT.py
  43. presubmit_test.py
  44. presubmit_test_mocks.py
  45. pylintrc
  46. README.chromium
  47. README.md
  48. style-guide.md
  49. WATCHLISTS
  50. webrtc.gni
  51. webrtc_lib_link_test.cc
  52. 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