Reland "Fix overflow due to rounding in AbsoluteSendTime::To24Bits"

This reverts commit 791294a647bfba8ebd26821a78020a2bb2f82b9b.

Reason for revert: downstream test adjusted

Original change's description:
> Revert "Fix overflow due to rounding in AbsoluteSendTime::To24Bits"
>
> This reverts commit a17651f7d8748905d902eedf34471a0c227ca789.
>
> Reason for revert: triggers failure in downstream test
>
> Original change's description:
> > Fix overflow due to rounding in AbsoluteSendTime::To24Bits
> >
> > Actual rounding is not important for this time as long it is consistent
> > during the call: only difference between two absolute send time matter
> > Rounding down avoids producing 1 < 24 when value is close to the wrap around boundary.
> >
> > Bug: None
> > Change-Id: Ibbf5bae21bc37eccdc5d4c130a59796ee5108017
> > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/268001
> > Commit-Queue: Åsa Persson <asapersson@webrtc.org>
> > Auto-Submit: Danil Chapovalov <danilchap@webrtc.org>
> > Reviewed-by: Åsa Persson <asapersson@webrtc.org>
> > Cr-Commit-Position: refs/heads/main@{#37468}
>
> Bug: None
> Change-Id: I90a9c1b174b918b7ede58c3bbdb879b1b67da7b2
> No-Presubmit: true
> No-Tree-Checks: true
> No-Try: true
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/268120
> Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
> Bot-Commit: rubber-stamper@appspot.gserviceaccount.com <rubber-stamper@appspot.gserviceaccount.com>
> Auto-Submit: Danil Chapovalov <danilchap@webrtc.org>
> Cr-Commit-Position: refs/heads/main@{#37473}

Bug: None
Change-Id: I99bcc6c6b7c08cd9621bdce336cd5793f78ee657
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/268190
Bot-Commit: rubber-stamper@appspot.gserviceaccount.com <rubber-stamper@appspot.gserviceaccount.com>
Reviewed-by: Åsa Persson <asapersson@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#37498}
1 file changed
tree: c239a6e07b4211e8854e6a6e541af40481fda2e9
  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