- Removes voe_conference_test.
- Adds a new AudioStatsTest, with better coverage of the same features, based on call_test.
- Adds an AudioEndToEndTest utility, which AudioStatsTest and LowBandwidthAudioTest uses.

BUG=webrtc:4690
R=kwiberg@webrtc.org

Review-Url: https://codereview.webrtc.org/3008273002 .
Cr-Commit-Position: refs/heads/master@{#19833}
15 files changed
tree: 2a2e9edab7a943bea5d6189befacf1d66d8aadff
  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. .vpython
  12. AUTHORS
  13. BUILD.gn
  14. CODE_OF_CONDUCT.md
  15. codereview.settings
  16. DEPS
  17. LICENSE
  18. license_template.txt
  19. LICENSE_THIRD_PARTY
  20. OWNERS
  21. PATENTS
  22. PRESUBMIT.py
  23. presubmit_test.py
  24. presubmit_test_mocks.py
  25. pylintrc
  26. README.md
  27. style-guide.md
  28. 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