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webrtc / src.git / 536378bf37c7c7624832e4deabd4fcf032ebcefb
commit536378bf37c7c7624832e4deabd4fcf032ebcefb[log] [tgz]
authorphilipel <philipel@webrtc.org>Tue May 31 10:20:23 2016
committerCommit bot <commit-bot@chromium.org>Tue May 31 10:20:28 2016
treefae8205bc9df3c897753d030bf64649d9b9c05d0
parent6c2eab34f8ff1f7a724622e3473858a4c0724bd8 [diff]
Allow FakeNetworkPipe to drop packets in bursts.

The fake network pipe will still only drop packets at an average rate of
|loss_percent| but in bursts at an average length specified by
|avg_burst_loss_length|.

Also added the flag -avg_burst_loss_length to video loopback.

BUG=

Review-Url: https://codereview.webrtc.org/1995683003
Cr-Commit-Position: refs/heads/master@{#12969}
4 files changed
tree: fae8205bc9df3c897753d030bf64649d9b9c05d0
  1. build_overrides/
  2. chromium/
  3. data/
  4. infra/
  5. resources/
  6. talk/
  7. third_party/
  8. tools/
  9. webrtc/
  10. .clang-format
  11. .gitignore
  12. .gn
  13. all.gyp
  14. AUTHORS
  15. BUILD.gn
  16. check_root_dir.py
  17. codereview.settings
  18. COPYING
  19. DEPS
  20. LICENSE
  21. license_template.txt
  22. LICENSE_THIRD_PARTY
  23. OWNERS
  24. PATENTS
  25. PRESUBMIT.py
  26. pylintrc
  27. README.md
  28. setup_links.py
  29. sync_chromium.py
  30. WATCHLISTS
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. This page is maintained by the Google Chrome team.

Development

See http://www.webrtc.org/native-code/development for instructions on how to get started developing with the native code.

More info