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webrtc / src / fb4d66be9726b849f8af61cafb09ed96bb993604
commitfb4d66be9726b849f8af61cafb09ed96bb993604[log] [tgz]
authorSebastian Jansson <srte@webrtc.org>Fri Jun 15 12:16:44 2018
committerCommit Bot <commit-bot@chromium.org>Fri Jun 15 12:39:59 2018
treebd967f1e1ae0cfb3b8492b82272e0f6d71ec5adc
parentb9b146c9feaf6fd610c3d96b6ed532c447f4bb89 [diff]
Improves buffer time calculation in network control tester.

The previous solution caused packet reordering if the bandwidth changed
with large buffers. To avoid this, the buffer time is tracked instead.
This means the the bandwidth is applied per packet and can't be
retroactively changed for packets already handled.

Bug: webrtc:8415
Change-Id: Ib6c97ba9b948220e88c79776aa8d96de289dcfb5
Reviewed-on: https://webrtc-review.googlesource.com/83723
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23629}
2 files changed
tree: bd967f1e1ae0cfb3b8492b82272e0f6d71ec5adc
  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