Victor Boivie adb224b3e9 dcsctp: Simplify congestion control algorithm
There was a feature in the retransmission queue to avoid fragmenting
large messages when the congestion window was large. This was a feature
that intended to improve data channel performance under Chrome, where
communication with the network process (over MOJO) was lossy and losing
messages with small fragments could result in unnecessary
retransmissions. But back then, the implementation for fast retransmit
wasn't implemented correctly, so the benchmarking result don't
reproduce any longer.

So just improve the algorithm by removing this code. This aligns it with
the RFC and makes it easier to implement pluggable congestion control
algorithms (that wouldn't want this feature).

Bug: webrtc:42223116
Change-Id: Ifaaa82dac4b8fe2f55418158ae8b3da97199212f
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/359681
Reviewed-by: Florent Castelli <orphis@webrtc.org>
Commit-Queue: Victor Boivie <boivie@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#42797}
2024-08-18 15:20:35 +00:00
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2022-12-02 09:21:47 +00:00
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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 here for instructions on how to get started developing with the native code.

Authoritative list of directories that contain the native API header files.

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