Support for two audio codec lists down into WebRtcVoiceEngine.

Added the plumbing necessary to get two different lists of codecs from
WebRtcVoiceEngine up to MediaSessionDescriptionFactory.

This should be the last step in this set of CLs. Once
https://codereview.webrtc.org/1991233004/ has landed, it's possible to
implement the ReceiveCodecs getter with the info from the
AudioDecoderFactory. The factory needs to be updated to actually
produce the correct list, as well.

BUG=webrtc:5805

Review-Url: https://codereview.webrtc.org/2013053002
Cr-Commit-Position: refs/heads/master@{#13131}
11 files changed
tree: c134018d4974dc5930db70af95617b3733fc70e6
  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