Reland "Change ReceiveStatistics reaction to large sequence numbers jumps"

This reverts commit 7e0299e2452b021fcd14a8fdb86257459eeacf90.

Reason for revert: audio receive stream fix not to use 0 reordering threshold

Original change's description:
> Revert "Change ReceiveStatistics reaction to large sequence numbers jumps"
> 
> This reverts commit c4f120130f495e9726bf221356642de69125f4a2.
> 
> Reason for revert: breaks downstream tests due to zero max reordering for audio receive channels
> 
> Original change's description:
> > Change ReceiveStatistics reaction to large sequence numbers jumps
> > 
> > Consider stream restart when two sequential packets arrived far from
> > previous packets' sequence numbers.
> > instead of resetting on single one.
> > For packet loss calculation ignore sequence number gap during reset.
> > 
> > Bug: webrtc:9445, b/38179459
> > Change-Id: I0c2717ef8f9ec182b280ae757b5582f56d9afcef
> > Reviewed-on: https://webrtc-review.googlesource.com/c/111962
> > Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
> > Reviewed-by: Åsa Persson <asapersson@webrtc.org>
> > Cr-Commit-Position: refs/heads/master@{#25890}
> 
> TBR=danilchap@webrtc.org,asapersson@webrtc.org
> 
> Change-Id: Icc9f4d86d9f0b07f0fa2f3d443f9a90aa91f5e21
> No-Presubmit: true
> No-Tree-Checks: true
> No-Try: true
> Bug: webrtc:9445, b/38179459
> Reviewed-on: https://webrtc-review.googlesource.com/c/113067
> Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
> Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#25897}

TBR=danilchap@webrtc.org,asapersson@webrtc.org

Change-Id: I8747aa5cb6209b92fafefed077bc19d305d11db6
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: webrtc:9445, b/38179459
Reviewed-on: https://webrtc-review.googlesource.com/c/113263
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25907}
3 files changed
tree: 061194d2ce5342b7ff0bd56d4bc9fcf3ff5cd1f9
  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. 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