Reland "Use absl::optional instead or rtc::Optional"

This reverts commit 28e6a164bf9d2a42545d058bd50d39e1767f7398.

Reason for revert: the static initializer removed from abseil

Original change's description:
> Revert "Use absl::optional instead or rtc::Optional"
>
> This reverts commit 7ba9e92fa0dfb16579f4f6ecd746397bdfdd174d.
>
> Reason for revert: Breaks Chromium static initialized regression test.
> https://ci.chromium.org/p/chromium/builders/luci.chromium.try/android-marshmallow-arm64-rel/5068
>
> Original change's description:
> > Use absl::optional instead or rtc::Optional
> >
> > BUG: webrtc:9078
> > Change-Id: I69aedce324d86e8894b81210a2de17c5ef68fd11
> > Reviewed-on: https://webrtc-review.googlesource.com/77082
> > Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
> > Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
> > Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
> > Cr-Commit-Position: refs/heads/master@{#23440}
>
> TBR=danilchap@webrtc.org,mbonadei@webrtc.org,kwiberg@webrtc.org
>
> Change-Id: I09ae74bddc69d0b25c8dfbcacc4ec906b34ca748
> No-Presubmit: true
> No-Tree-Checks: true
> No-Try: true
> Reviewed-on: https://webrtc-review.googlesource.com/79980
> Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
> Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#23449}

TBR=danilchap@webrtc.org,mbonadei@webrtc.org,kwiberg@webrtc.org

# Not skipping CQ checks because original CL landed > 1 day ago.

Change-Id: Ib5dc71fb63fe02b78743b03f8252b962616eead0
Bug: webrtc:9078
Reviewed-on: https://webrtc-review.googlesource.com/82760
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23586}
5 files changed
tree: 3204cfbecd3364c9b4478312224bbf19dead80ee
  1. api/
  2. audio/
  3. build_overrides/
  4. call/
  5. common_audio/
  6. common_video/
  7. data/
  8. examples/
  9. infra/
  10. logging/
  11. media/
  12. modules/
  13. ortc/
  14. p2p/
  15. pc/
  16. resources/
  17. rtc_base/
  18. rtc_tools/
  19. sdk/
  20. stats/
  21. style-guide/
  22. system_wrappers/
  23. test/
  24. third_party/
  25. tools_webrtc/
  26. video/
  27. .clang-format
  28. .git-blame-ignore-revs
  29. .gitignore
  30. .gn
  31. .vpython
  32. AUTHORS
  33. BUILD.gn
  34. CODE_OF_CONDUCT.md
  35. codereview.settings
  36. common_types.h
  37. DEPS
  38. LICENSE
  39. license_template.txt
  40. LICENSE_THIRD_PARTY
  41. native-api.md
  42. OWNERS
  43. PATENTS
  44. PRESUBMIT.py
  45. presubmit_test.py
  46. presubmit_test_mocks.py
  47. pylintrc
  48. README.chromium
  49. README.md
  50. style-guide.md
  51. THIRD_PARTY_CHROMIUM_DEPS.json
  52. THIRD_PARTY_WEBRTC_DEPS.json
  53. typedefs.h
  54. WATCHLISTS
  55. webrtc.gni
  56. 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