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gerrit-public.fairphone.software / platform / external / webrtc / e448a3fb54eec298d9113b2bd85e71e2fe69bb77
commite448a3fb54eec298d9113b2bd85e71e2fe69bb77[log] [tgz]
authorMarina Ciocea <marinaciocea@webrtc.org>Mon Mar 04 15:52:21 2019 +0100
committerCommit Bot <commit-bot@chromium.org>Mon Mar 11 13:21:36 2019 +0000
treee0ff3c30cdc67cdfa277a49c07e9b1cc52d8eb56
parentad89528051f9478cf8738c0dfd264d4a716b4698 [diff]
Update DataChannel bufferedamount implementation.

Call DataChannelObserver::OnBufferedAmountChange on each successful send.
Previously, the observer would get notified of buffered amount changes only when
queued send data is consumed. Data gets queued only if it cannot be sent right
away. According to the WebRTC standard[1], bufferedamount should be increased
before each sent and decreased after each successful sent. Update implementation
to be standard compliant.

Design doc: http://doc/1lorHBn-GMn5U0T0RQANxrsW0pXhw8XGZM-xZyVUOW90

[1] https://w3c.github.io/webrtc-pc/#dom-datachannel-bufferedamount

Bug: chromium:878682
Change-Id: Ife009d30c4a18dced9a54cf600a445bb1f02561d
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/123237
Commit-Queue: Marina Ciocea <marinaciocea@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27057}
4 files changed
tree: e0ff3c30cdc67cdfa277a49c07e9b1cc52d8eb56
  1. api/
  2. audio/
  3. build_overrides/
  4. call/
  5. common_audio/
  6. common_video/
  7. crypto/
  8. data/
  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. 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