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webrtc / src / 5298f72f97d76275f3dfb01ca1c01585d274573e
commit5298f72f97d76275f3dfb01ca1c01585d274573e[log] [tgz]
authorPer Kjellander <perkj@webrtc.org>Thu Jul 03 13:58:26 2025
committerWebRTC LUCI CQ <webrtc-scoped@luci-project-accounts.iam.gserviceaccount.com>Thu Jul 03 17:32:40 2025
treea53c12b42c52806aeb0c3a0435a6d8c062c61686
parent83b748da4fd1fe3ca8609afc8db49a404726e486 [diff]
Revert "Refactor class SendSideBandwidthEstimation"

This reverts commit c432ba14dbb2cdd2b90a6f18043254cbcfa2fd0e.

Reason for revert: Investigate downstream test breakage - b/429391976

Bug: webrtc:423841921, webrtc:42222445
Original change's description:
> Refactor class SendSideBandwidthEstimation
>
> The goal is to make states more clear and be able to log where a certain decision come from.
> In this cl:
>  - loss bases BWE handling moved to separate files
>  - remove field trial WebRTC-Bwe-ReceiverLimitCapsOnly
>
> Bug: webrtc:423841921, webrtc:42222445
> Change-Id: I502bee094e18606f8a188214fafa421a868023ca
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/396342
> Reviewed-by: Diep Bui <diepbp@webrtc.org>
> Commit-Queue: Per Kjellander <perkj@webrtc.org>
> Cr-Commit-Position: refs/heads/main@{#45056}

Bug: webrtc:423841921, webrtc:42222445
Change-Id: I8dcda24877dd8b1fbab4e1bb5235e2e7b903dabf
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/399080
Bot-Commit: rubber-stamper@appspot.gserviceaccount.com <rubber-stamper@appspot.gserviceaccount.com>
Reviewed-by: Diep Bui <diepbp@webrtc.org>
Commit-Queue: Per Kjellander <perkj@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#45088}
9 files changed
tree: a53c12b42c52806aeb0c3a0435a6d8c062c61686
  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. .rustfmt.toml
  34. .style.yapf
  35. .vpython3
  36. AUTHORS
  37. BUILD.gn
  38. CODE_OF_CONDUCT.md
  39. codereview.settings
  40. DEPS
  41. DIR_METADATA
  42. ENG_REVIEW_OWNERS
  43. LICENSE
  44. license_template.txt
  45. native-api.md
  46. OWNERS
  47. OWNERS_INFRA
  48. PATENTS
  49. PRESUBMIT.py
  50. presubmit_test.py
  51. presubmit_test_mocks.py
  52. pylintrc
  53. pylintrc_old_style
  54. README.chromium
  55. README.md
  56. WATCHLISTS
  57. webrtc.gni
  58. webrtc_lib_link_test.cc
  59. 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