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webrtc / src / 5308652c73c5b76101b59bbfbb1b36ea6e25507c
commit5308652c73c5b76101b59bbfbb1b36ea6e25507c[log] [tgz]
authorHarald Alvestrand <hta@webrtc.org>Sun Aug 25 16:43:08 2024
committerWebRTC LUCI CQ <webrtc-scoped@luci-project-accounts.iam.gserviceaccount.com>Mon Aug 26 11:11:43 2024
tree144e86e01e9fe2c841fb5953e4ac2f2b9de6d09d
parent7348f820a90a42b7ef89f46302bdbab0abd4f5c5 [diff]
Reland "Add recording of PT->Codec mappings on setting SDP for transport"

This reverts commit 6793f831ffdc598e12aced80a4d97956ca50e436.

Reason for revert: Removed the check that caused the error.

Original change's description:
> Revert "Add recording of PT->Codec mappings on setting SDP for transport"
>
> This reverts commit 15717236c8621cb684bb7753acfedbf34d931c80.
>
> Reason for revert: pr-answer
>
> Original change's description:
> > Add recording of PT->Codec mappings on setting SDP for transport
> >
> > Bug: webrtc:360058654
> > Change-Id: I2aa5e0058346cd3fcda47a8ea5115848fbc4f3e2
> > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/360041
> > Commit-Queue: Harald Alvestrand <hta@webrtc.org>
> > Reviewed-by: Florent Castelli <orphis@webrtc.org>
> > Cr-Commit-Position: refs/heads/main@{#42819}
>
> Bug: webrtc:360058654
> Change-Id: I1fea51b3a0cecfa7e7de75f94f47a85fa064be59
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/360380
> Reviewed-by: Jonas Oreland <jonaso@google.com>
> Commit-Queue: Jonas Oreland <jonaso@google.com>
> Reviewed-by: Per Kjellander <perkj@webrtc.org>
> Bot-Commit: rubber-stamper@appspot.gserviceaccount.com <rubber-stamper@appspot.gserviceaccount.com>
> Reviewed-by: Henrik Boström <hbos@webrtc.org>
> Cr-Commit-Position: refs/heads/main@{#42835}

Bug: webrtc:360058654
Change-Id: I2b60ccd60df3bacbeecd848c3cb86f6725b1505a
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/360400
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Florent Castelli <orphis@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#42847}
18 files changed
tree: 144e86e01e9fe2c841fb5953e4ac2f2b9de6d09d
  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