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webrtc / src.git / 8b7ca4abb20179a952669cb123a1bdfac5ba89c8
commit8b7ca4abb20179a952669cb123a1bdfac5ba89c8[log] [tgz]
authorErik Språng <sprang@webrtc.org>Thu May 17 11:43:35 2018
committerCommit Bot <commit-bot@chromium.org>Thu May 17 12:21:11 2018
tree8b17f8a9c2ce15b6c815917023bffaf130b021d2
parente1c3c01a9087e85b3581832da09ebea6c718e5d8 [diff]
Make packet router send padding on rtp module that last sent media.

Currently we prefer the last added rtp module that supports rtx, and
assume this is the HD stream.

If we suffer a network degradation and stop sending HD, the current
behavior will trigger RTX padding on an inactive stream, which is not
very useful.

With this change, we will prefer the rtp module that last sent media,
which will spread the load a bit across active media streams, but will
be biased toward the one with highest packet rate.

Bug: webrtc:8975
Change-Id: Id52865ccd5263722c66d327b8c80457f63b90385
Reviewed-on: https://webrtc-review.googlesource.com/77360
Commit-Queue: Erik Språng <sprang@webrtc.org>
Reviewed-by: Philip Eliasson <philipel@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23281}
3 files changed
tree: 8b17f8a9c2ce15b6c815917023bffaf130b021d2
  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.cc
  37. common_types.h
  38. DEPS
  39. LICENSE
  40. license_template.txt
  41. LICENSE_THIRD_PARTY
  42. native-api.md
  43. OWNERS
  44. PATENTS
  45. PRESUBMIT.py
  46. presubmit_test.py
  47. presubmit_test_mocks.py
  48. pylintrc
  49. README.chromium
  50. README.md
  51. style-guide.md
  52. THIRD_PARTY_CHROMIUM_DEPS.json
  53. THIRD_PARTY_WEBRTC_DEPS.json
  54. typedefs.h
  55. WATCHLISTS
  56. webrtc.gni
  57. 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