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webrtc / src.git / 15df2774f4e85cf8900768c1793edcf17d651dcd
commit15df2774f4e85cf8900768c1793edcf17d651dcd[log] [tgz]
authorEmircan Uysaler <emircan@webrtc.org>Wed Jan 09 22:57:40 2019
committerCommit Bot <commit-bot@chromium.org>Wed Jan 09 23:49:03 2019
tree87c63150b9ec6fef7f41cd1e55de0fda0211be06
parent026c8b773d1ef7c53dd9d6ea389319cc6488c2ed [diff]
Add a high bitrate full stack test with fake codec

This CL adds a fake codec factory  in WebRTC that can be used in tests to
produce target bitrate output.

We also add a high bitrate test that makes use of fake codec. This test assumes
ideal network conditions with target bandwidth being available and exercises
WebRTC calls with a high target bitrate(100 Mbps) end-to-end.

Bug: chromium:879723
Change-Id: I981124e2087054ed72c5447e239f28aae0878e29
Reviewed-on: https://webrtc-review.googlesource.com/c/97185
Commit-Queue: Emircan Uysaler <emircan@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Reviewed-by: Johannes Kron <kron@webrtc.org>
Reviewed-by: Sebastian Jansson <srte@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26182}
9 files changed
tree: 87c63150b9ec6fef7f41cd1e55de0fda0211be06
  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. 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