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webrtc / src / d78a854c567e72c9686e2aef16bd2f4ea8bcdb29
commitd78a854c567e72c9686e2aef16bd2f4ea8bcdb29[log] [tgz]
authorHarald Alvestrand <hta@webrtc.org>Thu May 15 04:35:20 2025
committerWebRTC LUCI CQ <webrtc-scoped@luci-project-accounts.iam.gserviceaccount.com>Thu May 15 08:01:27 2025
tree32e1167678f061e6ddc54a1563237c4a948bb75b
parentdff5f9c27f4bfad368e8db0eb7dfd25391fe8bcc [diff]
Revert "Use only PayloadTypeSuggester for PT assigment"

This reverts commit e12bd20849d40501e17c33b74adce2aaf6c2a755.

Reason for revert: b/409355977 - sometimes caused PT clashes
Bug: webrtc:360058654
Original change's description:
> Use only PayloadTypeSuggester for PT assigment
>
> This merges two relands:
> Reland "Stop using the UsedPayloadTypes class"
> This reverts commit ebb67395c1a7c81ec592f45d2abfdaf7f8f3e722.
> Reland "Use only one kind of codec list merging in CodecVendor."
> This reverts commit 4aa2908fee6a0d286449172565604a04ebfdea7c.
>
> The UsedPayloadTypes class has been removed.
>
> Bug: webrtc:360058654
> Change-Id: I72b8e4f336e37e9a276c6a449482c3b7f46fdc65
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/385223
> Reviewed-by: Evan Shrubsole <eshr@webrtc.org>
> Commit-Queue: Harald Alvestrand <hta@webrtc.org>
> Cr-Commit-Position: refs/heads/main@{#44547}

No-IWYU: Rollback
Bug: webrtc:360058654
Change-Id: Ifd62f359b457f8a88208fb4c082f6bcac3275e8f
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/391700
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Bot-Commit: rubber-stamper@appspot.gserviceaccount.com <rubber-stamper@appspot.gserviceaccount.com>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#44630}
3 files changed
tree: 32e1167678f061e6ddc54a1563237c4a948bb75b
  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. .rustfmt.toml
  34. .style.yapf
  35. .vpython3
  36. AUTHORS
  37. BUILD.gn
  38. CODE_OF_CONDUCT.md
  39. codereview.settings
  40. DEPS
  41. DIR_METADATA
  42. ENG_REVIEW_OWNERS
  43. LICENSE
  44. license_template.txt
  45. native-api.md
  46. OWNERS
  47. OWNERS_INFRA
  48. PATENTS
  49. PRESUBMIT.py
  50. presubmit_test.py
  51. presubmit_test_mocks.py
  52. pylintrc
  53. pylintrc_old_style
  54. README.chromium
  55. README.md
  56. WATCHLISTS
  57. webrtc.gni
  58. webrtc_lib_link_test.cc
  59. 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