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webrtc / src / 8234ead6d9671ba01861c0e11042a02c7335c8dd
commit8234ead6d9671ba01861c0e11042a02c7335c8dd[log] [tgz]
authorSeth Hampson <shampson@webrtc.org>Fri Feb 02 23:16:24 2018
committerCommit Bot <commit-bot@chromium.org>Tue Feb 06 18:51:14 2018
tree377909eef18d034559194f60d0f900597d0c96d4
parentc0216b8e68346f8264e96159dbdd06f4228b7653 [diff]
Allows the application to set active simulcast streams.

Currently all simulcast streams are set as active by default. This
update takes the values of the rtp encoding parameters and wires those
values down to the VideoSendStream and VideoStreamEncoder, so that the
appropriate simulcast streams can be turned off and on. This includes
adding more application specific controls in the EncoderStreamFactory.

Bug: webrtc:8653
Change-Id: Iaa7da3209cea0f0db72543981a319e319705cb00
Reviewed-on: https://webrtc-review.googlesource.com/47245
Reviewed-by: Erik Språng <sprang@webrtc.org>
Reviewed-by: Peter Thatcher <pthatcher@webrtc.org>
Commit-Queue: Seth Hampson <shampson@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#21920}
3 files changed
tree: 377909eef18d034559194f60d0f900597d0c96d4
  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. .clang-format
  26. .git-blame-ignore-revs
  27. .gitignore
  28. .gn
  29. .vpython
  30. AUTHORS
  31. BUILD.gn
  32. CODE_OF_CONDUCT.md
  33. codereview.settings
  34. common_types.cc
  35. common_types.h
  36. DEPS
  37. LICENSE
  38. license_template.txt
  39. LICENSE_THIRD_PARTY
  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. typedefs.h
  51. WATCHLISTS
  52. webrtc.gni
  53. 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