Danil Chapovalov b438b5a33d 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}
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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

Description
The idea is to make CMake build for WebRTC m130 version - for audio processing module
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