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webrtc / src / / 2dfc42d7b6be051bc6216ef9fccd0c961bde854b
commit2dfc42d7b6be051bc6216ef9fccd0c961bde854b[log] [tgz]
authorZhi Huang <zhihuang@webrtc.org>Mon Dec 04 21:38:48 2017
committerCommit Bot <commit-bot@chromium.org>Mon Dec 04 22:27:39 2017
treeb39e940e9b1dbcfe77526115de15b20835202d25
parent75432b38b48a45554b5b0630fea5c038191410f6 [diff]
Prepare to make BaseChannel depend on RtpTransportInternal only.

Eventually we want BaseChannel to depend on the RtpTransportInternal
instead of DtlsTransportInternal and share RtpTransport when bundling.
This CL is the first step.

Add SetRtpTransport and Init_w(RtptransportInternal*) to BaseChannel.
These two methods would replace the existing SetTransports and Init_w
methods.

Add new CreateVoice/VideoChannel methods to the ChannelManager which
 take RtpTransportInternal instead of Dtls/PacketTransportInternal.

|cotnent_name| is removed from the SrtpTransport to simplify to code
since it is only used for debugging.

InitNetwork_n is removed from BaseChannel in CL as well.

Bug: webrtc:7013
Change-Id: I35b1565958548bd4896854c49e61d3ee160b7634
Reviewed-on: https://webrtc-review.googlesource.com/27840
Commit-Queue: Zhi Huang <zhihuang@webrtc.org>
Reviewed-by: Peter Thatcher <pthatcher@webrtc.org>
Reviewed-by: Steve Anton <steveanton@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#21057}
9 files changed
tree: b39e940e9b1dbcfe77526115de15b20835202d25
  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