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webrtc / src.git / dab50c6fe8f13a19f4dee31a5338ad70a81f4f74
commitdab50c6fe8f13a19f4dee31a5338ad70a81f4f74[log] [tgz]
authorMirko Bonadei <mbonadei@webrtc.org>Mon May 16 13:40:26 2022
committerWebRTC LUCI CQ <webrtc-scoped@luci-project-accounts.iam.gserviceaccount.com>Mon May 16 14:08:04 2022
tree5a991fe951ecd0ab38a75cbcaa4870c181b8516e
parent6928f4ccc6fb334e8fc4cf8b87ea53cbe3dc8261 [diff]
Revert "Use ADM internal state for init state check."

This reverts commit 0e2221eb2f02ed950f4fd9c7fea40b382ea0a0c8.

Reason for revert: Speculative revert, breaks downstream.

Original change's description:
> Use ADM internal state for init state check.
>
> When ADM is terminated and its state requires reinitialized, VoipCore::initialized_ field will falsely skip required reinitializing.
>
> Bug: webrtc:14054
> Change-Id: Ibeb4987a7e9763b8e40926acc4d7eaabde7a3478
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/261924
> Reviewed-by: Per Åhgren <peah@webrtc.org>
> Reviewed-by: Tim Na <natim@google.com>
> Commit-Queue: Tim Na <natim@google.com>
> Cr-Commit-Position: refs/heads/main@{#36893}

Bug: webrtc:14054
Change-Id: I1fa0a1ff440b9619aba60ec25970ce88a67739db
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/262660
Bot-Commit: rubber-stamper@appspot.gserviceaccount.com <rubber-stamper@appspot.gserviceaccount.com>
Owners-Override: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#36896}
2 files changed
tree: 5a991fe951ecd0ab38a75cbcaa4870c181b8516e
  1. api/
  2. audio/
  3. build_overrides/
  4. call/
  5. common_audio/
  6. common_video/
  7. data/
  8. docs/
  9. examples/
  10. g3doc/
  11. infra/
  12. logging/
  13. media/
  14. modules/
  15. net/
  16. p2p/
  17. pc/
  18. resources/
  19. rtc_base/
  20. rtc_tools/
  21. sdk/
  22. stats/
  23. system_wrappers/
  24. test/
  25. tools_webrtc/
  26. video/
  27. .clang-format
  28. .git-blame-ignore-revs
  29. .gitignore
  30. .gn
  31. .mailmap
  32. .style.yapf
  33. .vpython
  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. g3doc.lua
  43. LICENSE
  44. license_template.txt
  45. native-api.md
  46. OWNERS
  47. PATENTS
  48. PRESUBMIT.py
  49. presubmit_test.py
  50. presubmit_test_mocks.py
  51. pylintrc
  52. README.chromium
  53. README.md
  54. WATCHLISTS
  55. webrtc.gni
  56. webrtc_lib_link_test.cc
  57. 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