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webrtc / src / / a08189b948fb1e820e8bfa772bf70d8c598b87e5
commita08189b948fb1e820e8bfa772bf70d8c598b87e5[log] [tgz]
authorJakob Ivarsson‎ <jakobi@webrtc.org>Mon Nov 25 09:00:51 2024
committerWebRTC LUCI CQ <webrtc-scoped@luci-project-accounts.iam.gserviceaccount.com>Mon Nov 25 15:25:10 2024
tree555d18cb6926d6cf77827d8d23c6277e693e61d3
parentb40c5598580478b44b5e14ebd25d74ae3f5f1217 [diff]
Revert "Add InsertPacket method that takes RtpPacketInfo."

This reverts commit 38ddea5ee3320bf3441aeb3654e099b3695c9789.

Reason for revert: not backwards compatible

Original change's description:
> Add InsertPacket method that takes RtpPacketInfo.
>
> The version which only passes receive_time will be removed (once migrated).
> Keeping the version that only passes header and payload for convenience.
>
> This will allow us to attach more metadata on the worker thread before InsertPacket, instead of on the playout thread after GetAudio. Eventually, the plan is to split the RTP handling on the worker thread into a separate class.
>
> Bug: webrtc:42223109
> Change-Id: I5399b53b9fc5c2f1c996e109054b1b0877ecca05
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/369000
> Reviewed-by: Henrik Lundin <henrik.lundin@webrtc.org>
> Commit-Queue: Jakob Ivarsson‎ <jakobi@webrtc.org>
> Cr-Commit-Position: refs/heads/main@{#43445}

Bug: webrtc:42223109
Change-Id: Ie7cf397cfbe5dedca009f16e5e9e3af40adbe99b
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/369200
Reviewed-by: Henrik Lundin <henrik.lundin@webrtc.org>
Bot-Commit: rubber-stamper@appspot.gserviceaccount.com <rubber-stamper@appspot.gserviceaccount.com>
Commit-Queue: Jakob Ivarsson‎ <jakobi@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#43450}
7 files changed
tree: 555d18cb6926d6cf77827d8d23c6277e693e61d3
  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. .vpython3
  35. AUTHORS
  36. BUILD.gn
  37. CODE_OF_CONDUCT.md
  38. codereview.settings
  39. DEPS
  40. DIR_METADATA
  41. ENG_REVIEW_OWNERS
  42. LICENSE
  43. license_template.txt
  44. native-api.md
  45. OWNERS
  46. OWNERS_INFRA
  47. PATENTS
  48. PRESUBMIT.py
  49. presubmit_test.py
  50. presubmit_test_mocks.py
  51. pylintrc
  52. pylintrc_old_style
  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