Google Git
Sign in
webrtc / src.git / 512bdce7141371a7db54b10d52f90505342efe1f
commit512bdce7141371a7db54b10d52f90505342efe1f[log] [tgz]
authorSebastian Jansson <srte@webrtc.org>Mon Apr 23 11:15:04 2018
committerCommit Bot <commit-bot@chromium.org>Wed Apr 25 10:32:23 2018
treeb2501735d74fdbe6f4302238c19d28cf78a5f0f9
parent09408115cda7e4203cdc06413c0240958b167814 [diff]
Using microseconds in FakeNetworkPipe.

This CL changes the usages of milliseconds as a unit in FakeNetworkPipe
to microseconds. This matches the time unit of the PacketTime struct and
increases the precision of the simulation. The time resolution in
FakeNetworkPipe::Config is kept unchanged to keep the values more human
readable.

This CL prepares refactoring in upcoming CLs.

Bug: webrtc:9054
Change-Id: I103f7a0afa41381f676ea07fcc8c083532e61f1d
Reviewed-on: https://webrtc-review.googlesource.com/64140
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Reviewed-by: Niels Moller <nisse@webrtc.org>
Reviewed-by: Christoffer Rodbro <crodbro@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23018}
2 files changed
tree: b2501735d74fdbe6f4302238c19d28cf78a5f0f9
  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. tools_webrtc/
  25. video/
  26. .clang-format
  27. .git-blame-ignore-revs
  28. .gitignore
  29. .gn
  30. .vpython
  31. AUTHORS
  32. BUILD.gn
  33. CODE_OF_CONDUCT.md
  34. codereview.settings
  35. common_types.cc
  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. typedefs.h
  52. WATCHLISTS
  53. webrtc.gni
  54. 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