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webrtc / src / 4aeab708bb9bd41f5626b05965bfd13fe8a6c458
commit4aeab708bb9bd41f5626b05965bfd13fe8a6c458[log] [tgz]
authorHarald Alvestrand <hta@webrtc.org>Mon Feb 17 18:06:32 2025
committerWebRTC LUCI CQ <webrtc-scoped@luci-project-accounts.iam.gserviceaccount.com>Mon Feb 17 19:46:42 2025
tree3dc4be9e133f5341aa94f9ad70e1b7c77bfdf8a4
parentba0d9713ba37fb72a305e21ede201350b6ba7e2e [diff]
Reland "Fix codec collision on reoffer after munged codec on offer."

This reverts commit 20bd702ebeb13a709832463fe5aadd623b7dc71b.

Reason for revert: Fixed test to not fail when AV1 is missing

Original change's description:
> Revert "Fix codec collision on reoffer after munged codec on offer."
>
> This reverts commit b9ddaa154b91b5d1cbe38bf38fce544a87e00d1a.
>
> Reason for revert: Downstream failure.
>
> Original change's description:
> > Fix codec collision on reoffer after munged codec on offer.
> >
> > Bug: chromium:395077842
> > Change-Id: I7665e593fa0f6883150363cb75103facd62f4fea
> > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/377141
> > Reviewed-by: Henrik Boström <hbos@webrtc.org>
> > Commit-Queue: Harald Alvestrand <hta@webrtc.org>
> > Cr-Commit-Position: refs/heads/main@{#43889}
>
> Bug: chromium:395077842
> Change-Id: I10184a0d521add123838861a5c5e7929864537bb
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/377500
> Bot-Commit: rubber-stamper@appspot.gserviceaccount.com <rubber-stamper@appspot.gserviceaccount.com>
> Reviewed-by: Jeremy Leconte <jleconte@webrtc.org>
> Commit-Queue: Harald Alvestrand <hta@webrtc.org>
> Reviewed-by: Henrik Boström <hbos@webrtc.org>
> Cr-Commit-Position: refs/heads/main@{#43901}

Bug: chromium:395077842
Change-Id: I3ba5cacebc7eb608edffea2de54cf1c1e9355a81
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/377281
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#43907}
3 files changed
tree: 3dc4be9e133f5341aa94f9ad70e1b7c77bfdf8a4
  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