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webrtc / src / c0741e9f12df679a2ec634d064edd212d7e72ed3
commitc0741e9f12df679a2ec634d064edd212d7e72ed3[log] [tgz]
authorOlga Sharonova <olka@webrtc.org>Wed Jan 31 12:49:22 2024
committerWebRTC LUCI CQ <webrtc-scoped@luci-project-accounts.iam.gserviceaccount.com>Wed Jan 31 14:35:19 2024
treea97b226a728cbe45b784d3dc6fa7941146f4be5b
parent7cb4ce0079b3db66c892971189f17eb2abe78427 [diff]
Revert "Take out Fuchsia-only SDES-enabling parameters"

This reverts commit 59f3b35013a29f8c73a46fa6fd06aadc96aad892.

Broke WebRTC into Chrome rolls:

https://chromium-review.googlesource.com/c/chromium/src/+/5248171?tab=checks

/../third_party/blink/renderer/modules/peerconnection/rtc_peer_connection_handler.cc:216:18: error: no member named 'enable_dtls_srtp' in 'webrtc::PeerConnectionInterface::RTCConfiguration'
  216 |   configuration->enable_dtls_srtp = dtls_srtp_key_agreement;
      |   ~~~~~~~~~~~~~  ^

Original change's description:
> Take out Fuchsia-only SDES-enabling parameters
>
> This does not remove all traces of SDES - we still need to delete
> the cricket::CryptoParams struct and all code that uses it.
>
> Bug: webrtc:11066, chromium:804275
> Change-Id: I811c8d40da7f4af714d53376f24cd53332a15945
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/336780
> Reviewed-by: Henrik Boström <hbos@webrtc.org>
> Commit-Queue: Harald Alvestrand <hta@webrtc.org>
> Cr-Commit-Position: refs/heads/main@{#41634}

Bug: webrtc:11066, chromium:804275
Change-Id: I2c2114873091e0c662977a6ef5723e6447166a65
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/337181
Commit-Queue: Olga Sharonova <olka@webrtc.org>
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Bot-Commit: rubber-stamper@appspot.gserviceaccount.com <rubber-stamper@appspot.gserviceaccount.com>
Owners-Override: Olga Sharonova <olka@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#41643}
5 files changed
tree: a97b226a728cbe45b784d3dc6fa7941146f4be5b
  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