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webrtc / src / c235a8d2bb176061b27876c3520a1afbb048d3a4
commitc235a8d2bb176061b27876c3520a1afbb048d3a4[log] [tgz]
authorSebastian Jansson <srte@webrtc.org>Fri Jun 15 12:46:11 2018
committerCommit Bot <commit-bot@chromium.org>Fri Jun 15 13:29:21 2018
tree75dd2dec10d8efd2b1854c9b8595d84e610d49ce
parentfc50110df6091f68ffda621a8427610131392bd8 [diff]
Adds trial to always send padding packets when not sending video.

This can be used to avoid getting stuck in a state where the encoder
is paused due to low bandwidth estimate which means no additional
feedback is received to update the bandwidth estimate. This could
happen if feedback packets are lost.

Bug: webrtc:8415
Change-Id: I59cd60c0277e8b31a6b911b25e8e488af9008fc2
Reviewed-on: https://webrtc-review.googlesource.com/80880
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Reviewed-by: Christoffer Rodbro <crodbro@webrtc.org>
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23632}
3 files changed
tree: 75dd2dec10d8efd2b1854c9b8595d84e610d49ce
  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. style-guide/
  22. system_wrappers/
  23. test/
  24. third_party/
  25. tools_webrtc/
  26. video/
  27. .clang-format
  28. .git-blame-ignore-revs
  29. .gitignore
  30. .gn
  31. .vpython
  32. AUTHORS
  33. BUILD.gn
  34. CODE_OF_CONDUCT.md
  35. codereview.settings
  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. THIRD_PARTY_CHROMIUM_DEPS.json
  52. THIRD_PARTY_WEBRTC_DEPS.json
  53. typedefs.h
  54. WATCHLISTS
  55. webrtc.gni
  56. 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