neteq_rtpplay: Add one more RTP header extension and fix some stats

The extension ID for transport sequence number is added to the list of
known RTP header extensions. Also, the minimum and maximum waiting
time for packets is now aggregated as minimum and maximum,
respectively, not as averages.

BUG=none

Review-Url: https://codereview.webrtc.org/3004783003
Cr-Commit-Position: refs/heads/master@{#19593}
1 file changed
tree: 15bc89642ca411b3d7c091c65e297229d7a81a45
  1. build_overrides/
  2. data/
  3. infra/
  4. resources/
  5. tools_webrtc/
  6. webrtc/
  7. .clang-format
  8. .git-blame-ignore-revs
  9. .gitignore
  10. .gn
  11. AUTHORS
  12. BUILD.gn
  13. CODE_OF_CONDUCT.md
  14. codereview.settings
  15. DEPS
  16. LICENSE
  17. license_template.txt
  18. LICENSE_THIRD_PARTY
  19. OWNERS
  20. PATENTS
  21. PRESUBMIT.py
  22. pylintrc
  23. README.md
  24. 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.

Development

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

More info