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webrtc / src / 9a12ee5d37ced49a3b84357ac7ae8a9ddd8d01a6
commit9a12ee5d37ced49a3b84357ac7ae8a9ddd8d01a6[log] [tgz]
authorJakob Ivarsson <jakobi@webrtc.org>Thu Nov 26 15:04:18 2020
committerCommit Bot <commit-bot@chromium.org>Tue Dec 01 09:42:40 2020
tree3faaffedcd95e353e34483d3539048746c5be71e
parenta4205478657d79ac0090d2db2a0103fb55230f7b [diff]
Use frame rate in video overhead calculation.

Assume bitrate is evenly distributed between frames. This is wrong for
uneven frame sizes and will underestimate the overhead for simulcast.
However, it will be more correct than the current calculation,
especially for low bitrates when each frame is smaller than one packet.
This is also when overhead matters more since it is a larger fraction
of the total bitrate.

It is also unclear what will happen when using FEC.

Bug: b/166341943
Change-Id: I247b9d0fc7a8ad5daa9b577f55ec16c56efa34c3
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/195221
Reviewed-by: Erik Språng <sprang@webrtc.org>
Commit-Queue: Jakob Ivarsson <jakobi@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#32725}
3 files changed
tree: 3faaffedcd95e353e34483d3539048746c5be71e
  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. DEPS
  36. ENG_REVIEW_OWNERS
  37. LICENSE
  38. license_template.txt
  39. native-api.md
  40. OWNERS
  41. PATENTS
  42. PRESUBMIT.py
  43. presubmit_test.py
  44. presubmit_test_mocks.py
  45. pylintrc
  46. README.chromium
  47. README.md
  48. style-guide.md
  49. WATCHLISTS
  50. webrtc.gni
  51. webrtc_lib_link_test.cc
  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 here 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