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webrtc / src / f39c815a1d65c6b1c387a4f33f92bfc5fd143251
commitf39c815a1d65c6b1c387a4f33f92bfc5fd143251[log] [tgz]
authorSebastian Jansson <srte@webrtc.org>Mon Oct 14 15:32:21 2019
committerCommit Bot <commit-bot@chromium.org>Tue Oct 15 12:55:46 2019
tree29d903c5d31aaa883b560acad3d6c867a06e03ff
parentffc84527309d73c2779ffad9b3910a83d03b5f06 [diff]
Cleanup: Replacing set extension status bool with CHECK.

This was just checked in all places were it was used, moving the check
into RtpRtcp reduces the boiler plate required at the call sites.

Also changing to always register and unregister extensions by URI to
synchronize the code in AudioSendStream with the code in RtpVideoSender.

This prepares for reducing the scope of ChannelSend.

Bug: webrtc:9883
Change-Id: Ia64d79f20eb98f46cbbbe8318770e4fcf9caa1ec
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/155620
Reviewed-by: Erik Språng <sprang@webrtc.org>
Reviewed-by: Oskar Sundbom <ossu@webrtc.org>
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29490}
11 files changed
tree: 29d903c5d31aaa883b560acad3d6c867a06e03ff
  1. api/
  2. audio/
  3. build_overrides/
  4. call/
  5. common_audio/
  6. common_video/
  7. data/
  8. docs/
  9. examples/
  10. logging/
  11. media/
  12. modules/
  13. p2p/
  14. pc/
  15. resources/
  16. rtc_base/
  17. rtc_tools/
  18. sdk/
  19. stats/
  20. style-guide/
  21. system_wrappers/
  22. test/
  23. tools_webrtc/
  24. video/
  25. .clang-format
  26. .git-blame-ignore-revs
  27. .gitignore
  28. .gn
  29. .vpython
  30. abseil-in-webrtc.md
  31. AUTHORS
  32. BUILD.gn
  33. CODE_OF_CONDUCT.md
  34. codereview.settings
  35. common_types.h
  36. DEPS
  37. ENG_REVIEW_OWNERS
  38. LICENSE
  39. license_template.txt
  40. native-api.md
  41. OWNERS
  42. PATENTS
  43. PRESUBMIT.py
  44. presubmit_test.py
  45. presubmit_test_mocks.py
  46. pylintrc
  47. README.chromium
  48. README.md
  49. style-guide.md
  50. WATCHLISTS
  51. webrtc.gni
  52. 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