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webrtc / src / 56db8d09529d5ba92d24954a1d78a90c8ea2cd85
commit56db8d09529d5ba92d24954a1d78a90c8ea2cd85[log] [tgz]
authorTomas Gunnarsson <tommi@webrtc.org>Sat Feb 26 10:32:51 2022
committerWebRTC LUCI CQ <webrtc-scoped@luci-project-accounts.iam.gserviceaccount.com>Sat Feb 26 10:35:13 2022
treea4423677437c14e1082bca12bf41f4863044b72c
parent1db3c7435eb868713c4b362bda032ebf65c22478 [diff]
Revert "Represent RtpPacketToSend::capture_time with Timestamp"

This reverts commit 385eb9714daa80306d2f92d36678c42892dab555.

Reason for revert: Causes problems downstream:

#
# Fatal error in: rtc_base/units/unit_base.h, line 122
# last system error: 0
# Check failed: value >= 0 (-234 vs. 0)

Original change's description:
> Represent RtpPacketToSend::capture_time with Timestamp
>
> Bug: webrtc:13757
> Change-Id: I0ede22cd34e3a59afe1477d8edd495dce64e3242
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/252586
> Reviewed-by: Erik Språng <sprang@webrtc.org>
> Reviewed-by: Rasmus Brandt <brandtr@webrtc.org>
> Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
> Cr-Commit-Position: refs/heads/main@{#36083}

Bug: webrtc:13757
Change-Id: I8442abd438be8726cf671d0f372d50ecfac6847e
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/252720
Auto-Submit: Tomas Gunnarsson <tommi@webrtc.org>
Bot-Commit: rubber-stamper@appspot.gserviceaccount.com <rubber-stamper@appspot.gserviceaccount.com>
Commit-Queue: Tomas Gunnarsson <tommi@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#36087}
19 files changed
tree: a4423677437c14e1082bca12bf41f4863044b72c
  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. .mailmap
  31. .style.yapf
  32. .vpython
  33. .vpython3
  34. AUTHORS
  35. BUILD.gn
  36. CODE_OF_CONDUCT.md
  37. codereview.settings
  38. DEPS
  39. DIR_METADATA
  40. ENG_REVIEW_OWNERS
  41. g3doc.lua
  42. LICENSE
  43. license_template.txt
  44. native-api.md
  45. OWNERS
  46. PATENTS
  47. PRESUBMIT.py
  48. presubmit_test.py
  49. presubmit_test_mocks.py
  50. pylintrc
  51. README.chromium
  52. README.md
  53. WATCHLISTS
  54. webrtc.gni
  55. webrtc_lib_link_test.cc
  56. 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