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webrtc / src / 374b892ec7249db193de1b1d2354e71ffed2e321
commit374b892ec7249db193de1b1d2354e71ffed2e321[log] [tgz]
authorHarald Alvestrand <hta@webrtc.org>Wed Apr 02 19:04:25 2025
committerWebRTC LUCI CQ <webrtc-scoped@luci-project-accounts.iam.gserviceaccount.com>Wed Apr 02 19:10:31 2025
treefd152429a63ce3b8f4a564ad26be3ccb0f162f15
parentc92fc76366d6e1c9eff7074fe5d5c0a4c2144303 [diff]
Revert "Assign supported codecs per section, not globally."

This reverts commit c92fc76366d6e1c9eff7074fe5d5c0a4c2144303.

Reason for revert: Gerritt submitted patch set #10 not #12 - this broke

Original change's description:
> Assign supported codecs per section, not globally.
>
> This uses the PayloadTypeSuggester for PT disambiguation,
> ensuring that conficts are managed across the transport.
>
> Bug: webrtc:360058654
> Change-Id: Ic8f0569d0d96d97c13765169903929ec8a110104
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/383400
> Commit-Queue: Harald Alvestrand <hta@webrtc.org>
> Reviewed-by: Evan Shrubsole <eshr@webrtc.org>
> Reviewed-by: Henrik Boström <hbos@webrtc.org>
> Cr-Commit-Position: refs/heads/main@{#44284}

Bug: webrtc:360058654
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Change-Id: I16d0d92fb9a1ecc65eda060f81d41faefcca1e4c
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/384000
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Danil Chapovalov <danilchap@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@{#44285}
8 files changed
tree: fd152429a63ce3b8f4a564ad26be3ccb0f162f15
  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