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webrtc / src / 3b75c39d59f3f29aa388db2c306061cad42eaba5
commit3b75c39d59f3f29aa388db2c306061cad42eaba5[log] [tgz]
authorDanil Chapovalov <danilchap@webrtc.org>Thu Aug 10 12:52:14 2023
committerWebRTC LUCI CQ <webrtc-scoped@luci-project-accounts.iam.gserviceaccount.com>Thu Aug 10 14:18:52 2023
treeb534a1226bb8c83477f3652d269c649ca2111a0f
parentda705cea9bab359b9e79dcab6b830e114a8fffb1 [diff]
Revert "Cleanup calculating time between RTCP reports"

This reverts commit 762f193ca4cf7f32c665461e92b36811b1ef6fbd.

Reason for revert: breaks downstream test

Original change's description:
> Cleanup calculating time between RTCP reports
>
> Move that calculation into dedicated function, move comment why it is calculated the way it is into the same function.
> Cleanup that comment - remove parts unused by current code, in particular remove description of code that was deleted a while ago
> Use more strict types for the calculation to make it clearer.
> Replace DCHECK result can't be zero with a clamp to ensure it can't be zero, because with large bitrates it may.
>
> Bug: None
> Change-Id: Ie8c6b9720095cd1cc3f9814b9df16700119337c5
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/315143
> Reviewed-by: Åsa Persson <asapersson@webrtc.org>
> Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
> Cr-Commit-Position: refs/heads/main@{#40529}

Bug: None
Change-Id: I8c83013523120a84f236e8efa0d122363e7a228b
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/315381
Bot-Commit: rubber-stamper@appspot.gserviceaccount.com <rubber-stamper@appspot.gserviceaccount.com>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Reviewed-by: Åsa Persson <asapersson@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#40535}
5 files changed
tree: b534a1226bb8c83477f3652d269c649ca2111a0f
  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. .vpython
  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. 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