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webrtc / src / / f6703c4dcb5735343ef56ec80e97e6af598cf225
commitf6703c4dcb5735343ef56ec80e97e6af598cf225[log] [tgz]
authorMirko Bonadei <mbonadei@webrtc.org>Wed Nov 15 16:14:28 2017
committerCommit Bot <commit-bot@chromium.org>Wed Nov 15 16:14:42 2017
tree3a5377ce1b28aa82a2a3584f8f3408a617572afd
parente403212ede4fa68994a6c591a959ffd44c1057f2 [diff]
Revert "Use moving median filters in RemoteNtpTimeEstimator, RtpToNtpEstimator"

This reverts commit 550b666e20a13f9c22effce878a8e0078a0f7bad.

Reason for revert: breaks downstream projects.

Original change's description:
> Use moving median filters in RemoteNtpTimeEstimator, RtpToNtpEstimator
> 
> If Webrtc-ClockEstimation experiment is enabled, median filtering is
> applied to results of RtpToNtpEstimator and RemoteNtpEstimator to smooth
> out random errors introduced by incorrect RTCP SR reports and networking
> delays.
> 
> Bug: webrtc:8468
> Change-Id: Iec6d094d2809d1efeb0b9483059167d9a03880da
> Reviewed-on: https://webrtc-review.googlesource.com/22682
> Commit-Queue: Ilya Nikolaevskiy <ilnik@webrtc.org>
> Reviewed-by: Åsa Persson <asapersson@webrtc.org>
> Reviewed-by: Per Kjellander <perkj@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#20682}

TBR=ilnik@webrtc.org,asapersson@webrtc.org,perkj@webrtc.org

Change-Id: I17345d912bbaf635612c9b399d5f9166de818190
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: webrtc:8468
Reviewed-on: https://webrtc-review.googlesource.com/23320
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20689}
7 files changed
tree: 3a5377ce1b28aa82a2a3584f8f3408a617572afd
  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