Google Git
Sign in
webrtc / src.git / 7ee493622bb721cc14bf45edf22a69118402f971
commit7ee493622bb721cc14bf45edf22a69118402f971[log] [tgz]
authorphilipel <philipel@webrtc.org>Fri Jan 26 14:10:57 2018
committerPhilip Eliasson <philipel@webrtc.org>Fri Jan 26 14:16:39 2018
treed63336131fb8aacf18b4f06a220d689e898030aa
parentafdf3d0c44650802b340518d665e89a7f8a682ea [diff]
Don't overwrite RTP packets in history within one second or 3x RTT.

This prevents us from prematurely overwriting the packets in the history
if the RTT is underestimated.

(cherry picked from commit 788ac70c1f8a34bf375b09dee69a4e7d547dc4f9)

Bug: webrtc:8766
Change-Id: I042e8ce74cdce2a0451596f4217779fc856b51f4
Reviewed-on: https://webrtc-review.googlesource.com/42960
Commit-Queue: Erik Språng <sprang@webrtc.org>
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Cr-Original-Commit-Position: refs/heads/master@{#21735}
Reviewed-on: https://webrtc-review.googlesource.com/43863
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Cr-Commit-Position: refs/branch-heads/65@{#8}
Cr-Branched-From: 3ac67a736bb200ecf7c116a88b2f8d5c542973c8-refs/heads/master@{#21637}
2 files changed
tree: d63336131fb8aacf18b4f06a220d689e898030aa
  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. system_wrappers/
  22. test/
  23. tools_webrtc/
  24. video/
  25. voice_engine/
  26. .clang-format
  27. .git-blame-ignore-revs
  28. .gitignore
  29. .gn
  30. .vpython
  31. AUTHORS
  32. BUILD.gn
  33. CODE_OF_CONDUCT.md
  34. codereview.settings
  35. common_types.cc
  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. typedefs.h
  52. WATCHLISTS
  53. webrtc.gni
  54. 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