Google Git
Sign in
webrtc / src / 69c8d3c843326aff9dee32cc639741c1cd7f8ae9
commit69c8d3c843326aff9dee32cc639741c1cd7f8ae9[log] [tgz]
authorChristoffer Jansson <jansson@google.com>Tue Apr 18 07:16:32 2023
committerWebRTC LUCI CQ <webrtc-scoped@luci-project-accounts.iam.gserviceaccount.com>Tue Apr 18 07:18:16 2023
tree203862e69aaadef53b115b1f188e62b32cb36eec
parent8b5bf6dd059f39c00cca8d5be4cf59794f8da402 [diff]
Revert "Migrate TestAudioDeviceModule on AudioDeviceModuleImpl"

This reverts commit e42bf81486d2f08b6dcbf1442287202e937ce52b.

Reason for revert: Breaks iOS simulator bots and thus blocks chromium roll, https://chromium-review.googlesource.com/c/chromium/src/+/4433814

Original change's description:
> Migrate TestAudioDeviceModule on AudioDeviceModuleImpl
>
> Bug: b/272350185
> Change-Id: Ia3d85d6fa3b0d4809e987a39d60d3eb022687132
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/300363
> Commit-Queue: Artem Titov <titovartem@webrtc.org>
> Reviewed-by: Henrik Andreassson <henrika@webrtc.org>
> Cr-Commit-Position: refs/heads/main@{#39877}

Bug: b/272350185
Change-Id: I1e3b542fc1278797f283afedeae01cbb7412d353
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/301701
Commit-Queue: Jeremy Leconte <jleconte@google.com>
Bot-Commit: rubber-stamper@appspot.gserviceaccount.com <rubber-stamper@appspot.gserviceaccount.com>
Reviewed-by: Jeremy Leconte <jleconte@google.com>
Auto-Submit: Christoffer Jansson <jansson@google.com>
Owners-Override: Christoffer Jansson <jansson@google.com>
Cr-Commit-Position: refs/heads/main@{#39881}
11 files changed
tree: 203862e69aaadef53b115b1f188e62b32cb36eec
  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. .vpython
  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. 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