Generate random rtp packets with RtpPacketToSend instead of RtpSender
in rtc event log unittest

R=stefan@webrtc.org, terelius@webrtc.org

Review URL: https://codereview.webrtc.org/2228493002 .

Cr-Commit-Position: refs/heads/master@{#13705}
1 file changed
tree: 6408c3e7a90d65ab8abe67375a19337b4d678f8b
  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