Revert "Add AddTransceiver and GetTransceivers to PeerConnection"

This reverts commit f93d2800d9b0d5818a5a383def0aaef3d441df3a.

Reason for revert: https://logs.chromium.org/v/?s=chromium%2Fbb%2Fchromium.webrtc.fyi%2Fios-device%2F5804%2F%2B%2Frecipes%2Fsteps%2Fcompile%2F0%2Fstdout

Original change's description:
> Add AddTransceiver and GetTransceivers to PeerConnection
> 
> WebRTC 1.0 has added the transceiver API to PeerConnection. This
> is the first step towards exposing this to WebRTC consumers. For
> now, transceivers can be added and fetched but there is not yet
> support for creating offers/answers or setting local/remote
> descriptions. That support ("Unified Plan") will be added in
> follow-up CLs.
> 
> The transceiver API is currently only available if the application
> opts in by specifying the kUnifiedPlan SDP semantics when creating
> the PeerConnection.
> 
> Bug: webrtc:7600
> Change-Id: I0b8ee24b489b45bb4c5f60b699bd20c61af01d8e
> Reviewed-on: https://webrtc-review.googlesource.com/23880
> Commit-Queue: Steve Anton <steveanton@webrtc.org>
> Reviewed-by: Peter Thatcher <pthatcher@webrtc.org>
> Reviewed-by: Henrik Boström <hbos@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#20896}

TBR=steveanton@webrtc.org,zhihuang@webrtc.org,hbos@webrtc.org,pthatcher@webrtc.org

Change-Id: Ie91ea4988dba25c20e2532114d3a9d859a932d4c
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: webrtc:7600
Reviewed-on: https://webrtc-review.googlesource.com/26400
Reviewed-by: Steve Anton <steveanton@webrtc.org>
Commit-Queue: Steve Anton <steveanton@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20897}
10 files changed
tree: 2622ffea2f4b2d0e103cccbc08d95758d3024575
  1. api/
  2. audio/
  3. build_overrides/
  4. call/
  5. common_audio/
  6. common_video/
  7. data/
  8. examples/
  9. infra/
  10. logging/
  11. media/
  12. modules/
  13. ortc/
  14. p2p/
  15. pc/
  16. resources/
  17. rtc_base/
  18. rtc_tools/
  19. sdk/
  20. stats/
  21. system_wrappers/
  22. test/
  23. tools_webrtc/
  24. video/
  25. voice_engine/
  26. .clang-format
  27. .git-blame-ignore-revs
  28. .gitignore
  29. .gn
  30. .vpython
  31. AUTHORS
  32. BUILD.gn
  33. CODE_OF_CONDUCT.md
  34. codereview.settings
  35. common_types.cc
  36. common_types.h
  37. DEPS
  38. LICENSE
  39. license_template.txt
  40. LICENSE_THIRD_PARTY
  41. native-api.md
  42. OWNERS
  43. PATENTS
  44. PRESUBMIT.py
  45. presubmit_test.py
  46. presubmit_test_mocks.py
  47. pylintrc
  48. README.chromium
  49. README.md
  50. style-guide.md
  51. typedefs.h
  52. WATCHLISTS
  53. webrtc.gni
  54. whitespace.txt
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.

Authoritative list of directories that contain the native API header files.

More info