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webrtc / src / / f5d13267aee114aa60e9718fc6f5032c8a5450f3
commitf5d13267aee114aa60e9718fc6f5032c8a5450f3[log] [tgz]
authorHarald Alvestrand <hta@webrtc.org>Wed Jan 15 08:16:28 2025
committerWebRTC LUCI CQ <webrtc-scoped@luci-project-accounts.iam.gserviceaccount.com>Wed Jan 15 10:17:56 2025
tree434c73d9c003f9db785fcdbb748986c4ef85b67f
parent3e384c743649346ea1719881860356493fe5d577 [diff]
Reland "Use Payload Type suggester for all codec merging"

This reverts commit b7abaee819771ca297bac4c51ec0daf62bd9a3fc.

Reason for revert: Suspicion that suspected breakage wasn't real

Original change's description:
> Revert "Use Payload Type suggester for all codec merging"
>
> This reverts commit 0bac2aae596771db020f01a57fee4828081fbc38.
>
> Reason for revert: Suspected breakages downstream
>
> Original change's description:
> > Use Payload Type suggester for all codec merging
> >
> > Bug: webrtc:360058654
> > Change-Id: Id475762253c427c1800c2352a60fc0121c2dc388
> > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/364783
> > Reviewed-by: Florent Castelli <orphis@webrtc.org>
> > Commit-Queue: Harald Alvestrand <hta@webrtc.org>
> > Cr-Commit-Position: refs/heads/main@{#43267}
>
> Bug: webrtc:360058654, b/375132036
> Change-Id: Ieda626270193e7e6c93903b3c03a691b2bf0c1e2
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/366540
> Bot-Commit: rubber-stamper@appspot.gserviceaccount.com <rubber-stamper@appspot.gserviceaccount.com>
> Commit-Queue: Harald Alvestrand <hta@webrtc.org>
> Reviewed-by: Florent Castelli <orphis@webrtc.org>
> Cr-Commit-Position: refs/heads/main@{#43290}

Bug: webrtc:360058654, b/375132036
Change-Id: Id6e72f7aac81023da43de7627c24dd1a792ea461
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/374304
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#43739}
3 files changed
tree: 434c73d9c003f9db785fcdbb748986c4ef85b67f
  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