Exposing RtpSenders and RtpReceivers from PeerConnection.

This CL essentially converts [Local|Remote]TrackHandler to
Rtp[Sender|Receiver], and adds a "SetTrack" method for RtpSender.

It also gets rid of MediaStreamHandler and MediaStreamHandlerContainer,
since these classes weren't really anything more than containers.
PeerConnection now manages the RtpSenders and RtpReceivers directly.

Review URL: https://codereview.webrtc.org/1351803002

Cr-Commit-Position: refs/heads/master@{#10100}
21 files changed
tree: f3dcf359eba295e94225e507929d626cb9933100
  1. chromium/
  2. data/
  3. infra/
  4. resources/
  5. talk/
  6. third_party/
  7. tools/
  8. webrtc/
  9. .clang-format
  10. .gitignore
  11. .gn
  12. all.gyp
  13. AUTHORS
  14. BUILD.gn
  15. check_root_dir.py
  16. codereview.settings
  17. COPYING
  18. DEPS
  19. LICENSE
  20. license_template.txt
  21. LICENSE_THIRD_PARTY
  22. OWNERS
  23. PATENTS
  24. PRESUBMIT.py
  25. pylintrc
  26. README.md
  27. setup_links.py
  28. sync_chromium.py
  29. 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