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webrtc / src / 110a4de4e21ac8f3d53b11e2869612b403829bfb
commit110a4de4e21ac8f3d53b11e2869612b403829bfb[log] [tgz]
authorYves Gerey <yvesg@webrtc.org>Mon Aug 12 15:19:09 2019
committerCommit Bot <commit-bot@chromium.org>Mon Aug 12 15:53:01 2019
treeb0a75f839d0aea84e6d8201ec1ff48b7e6d6cdf0
parent40dc98a117c1f050c4b95ab443af984ec2c33449 [diff]
Roll chromium_revision 8f0166a59b..f0fd984a31 (685582:685691)

!! **Manual change** Less strict audio codec tests to accommodate opus fix [1].
!! This is meant to be a temporary mitigation.
[1] https://chromium-review.googlesource.com/c/chromium/src/+/1746617

Change log: https://chromium.googlesource.com/chromium/src/+log/8f0166a59b..f0fd984a31
Full diff: https://chromium.googlesource.com/chromium/src/+/8f0166a59b..f0fd984a31

Changed dependencies
* src/base: https://chromium.googlesource.com/chromium/src/base/+log/17d8ac209c..f6cc884505
* src/build: https://chromium.googlesource.com/chromium/src/build/+log/d6837de8f1..956965a6ea
* src/ios: https://chromium.googlesource.com/chromium/src/ios/+log/76e0b0bc60..6780db9c3e
* src/testing: https://chromium.googlesource.com/chromium/src/testing/+log/5d328647a1..48823ed18a
* src/third_party: https://chromium.googlesource.com/chromium/src/third_party/+log/d70201c684..82063e79f0
* src/third_party/depot_tools: https://chromium.googlesource.com/chromium/tools/depot_tools.git/+log/1b4c7e9f38..6d98232fde
* src/tools: https://chromium.googlesource.com/chromium/src/tools/+log/b8953a5bf5..2aa12eadc5
DEPS diff: https://chromium.googlesource.com/chromium/src/+/8f0166a59b..f0fd984a31/DEPS

No update to Clang.

TBR=chromium-webrtc-autoroll@webrtc-ci.iam.gserviceaccount.com,
BUG=None

Change-Id: I9869cc3f493bc82361e4f93ad846b32390edb340
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/148700
Reviewed-by: Yves Gerey <yvesg@webrtc.org>
Reviewed-by: Autoroller <chromium-webrtc-autoroll@webrtc-ci.iam.gserviceaccount.com>
Commit-Queue: Yves Gerey <yvesg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28833}
3 files changed
tree: b0a75f839d0aea84e6d8201ec1ff48b7e6d6cdf0
  1. api/
  2. audio/
  3. build_overrides/
  4. call/
  5. common_audio/
  6. common_video/
  7. data/
  8. examples/
  9. logging/
  10. media/
  11. modules/
  12. p2p/
  13. pc/
  14. resources/
  15. rtc_base/
  16. rtc_tools/
  17. sdk/
  18. stats/
  19. style-guide/
  20. system_wrappers/
  21. test/
  22. tools_webrtc/
  23. video/
  24. .clang-format
  25. .git-blame-ignore-revs
  26. .gitignore
  27. .gn
  28. .vpython
  29. abseil-in-webrtc.md
  30. AUTHORS
  31. BUILD.gn
  32. CODE_OF_CONDUCT.md
  33. codereview.settings
  34. common_types.h
  35. DEPS
  36. ENG_REVIEW_OWNERS
  37. LICENSE
  38. license_template.txt
  39. native-api.md
  40. OWNERS
  41. PATENTS
  42. PRESUBMIT.py
  43. presubmit_test.py
  44. presubmit_test_mocks.py
  45. pylintrc
  46. README.chromium
  47. README.md
  48. style-guide.md
  49. WATCHLISTS
  50. webrtc.gni
  51. 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 http://www.webrtc.org/native-code/development 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