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webrtc / src / / 23df143850f9b6757de71805ce53d738d57f72c6
commit23df143850f9b6757de71805ce53d738d57f72c6[log] [tgz]
authorYves Gerey <yvesg@webrtc.org>Fri Dec 20 10:49:00 2019
committerCommit Bot <commit-bot@chromium.org>Fri Dec 20 11:23:39 2019
tree333c3c1045338fc08d6bab34492969765356bf6b
parent077ee357743e873f2e34acc7d2194f9bd5b7b2bf [diff]
Roll chromium_revision 26cf7e7d6c..e4c6d7fe53 (725941:726742)

Manual tweak:  Don't roll src/third_party (harfbuzz-ng/BUILD.gn broken in WebRTC).

Change log: https://chromium.googlesource.com/chromium/src/+log/26cf7e7d6c..e4c6d7fe53
Full diff: https://chromium.googlesource.com/chromium/src/+/26cf7e7d6c..e4c6d7fe53

Changed dependencies
* src/base: https://chromium.googlesource.com/chromium/src/base/+log/6109a80975..f7fb459610
* src/build: https://chromium.googlesource.com/chromium/src/build/+log/4abd203d72..032b1bd069
* src/ios: https://chromium.googlesource.com/chromium/src/ios/+log/4bc3bea248..d292120bff
* src/testing: https://chromium.googlesource.com/chromium/src/testing/+log/873e02ab19..c5705ae7a3
* src/third_party/catapult: https://chromium.googlesource.com/catapult.git/+log/d04ef219dd..8c77d963b8
* src/third_party/depot_tools: https://chromium.googlesource.com/chromium/tools/depot_tools.git/+log/6037820448..05934953bf
* src/tools: https://chromium.googlesource.com/chromium/src/tools/+log/3c85ad5a73..78adf6b3a0
DEPS diff: https://chromium.googlesource.com/chromium/src/+/26cf7e7d6c..e4c6d7fe53/DEPS

No update to Clang.

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

Change-Id: Iaf9e165f0447555b3b811b8cb1df682fc93664e6
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/162925
Reviewed-by: Yves Gerey <yvesg@webrtc.org>
Reviewed-by: Autoroller <chromium-webrtc-autoroll@webrtc-ci.iam.gserviceaccount.com>
Commit-Queue: Yves Gerey <yvesg@google.com>
Cr-Commit-Position: refs/heads/master@{#30119}
1 file changed
tree: 333c3c1045338fc08d6bab34492969765356bf6b
  1. api/
  2. audio/
  3. build_overrides/
  4. call/
  5. common_audio/
  6. common_video/
  7. data/
  8. docs/
  9. examples/
  10. logging/
  11. media/
  12. modules/
  13. p2p/
  14. pc/
  15. resources/
  16. rtc_base/
  17. rtc_tools/
  18. sdk/
  19. stats/
  20. style-guide/
  21. system_wrappers/
  22. test/
  23. tools_webrtc/
  24. video/
  25. .clang-format
  26. .git-blame-ignore-revs
  27. .gitignore
  28. .gn
  29. .vpython
  30. abseil-in-webrtc.md
  31. AUTHORS
  32. BUILD.gn
  33. CODE_OF_CONDUCT.md
  34. codereview.settings
  35. common_types.h
  36. DEPS
  37. ENG_REVIEW_OWNERS
  38. LICENSE
  39. license_template.txt
  40. native-api.md
  41. OWNERS
  42. PATENTS
  43. PRESUBMIT.py
  44. presubmit_test.py
  45. presubmit_test_mocks.py
  46. pylintrc
  47. README.chromium
  48. README.md
  49. style-guide.md
  50. WATCHLISTS
  51. webrtc.gni
  52. webrtc_lib_link_test.cc
  53. 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