Taylor Brandstetter 210b752fb6 Temporarily increase DTLS buffer size to 2.
It's not expected this will make a difference, since the packet should
be read from the queue if possible as soon as it's added to it.

But we're doing this as an added precaution in case we overlooked
something. See linked bug.

Bug: chromium:1063834
Change-Id: I7a3a6d86a97683cbcbeed5ef1aaa8090cf6bf8c0
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/172661
Commit-Queue: Taylor <deadbeef@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30990}
2020-04-03 02:48:44 +00:00
2020-02-27 14:27:23 +00:00
2019-10-28 12:27:50 +00:00
.gn
2020-03-18 18:04:41 +00:00
2020-03-30 12:15:56 +00:00
2020-01-28 07:53:15 +00:00

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
Readme BSD-3-Clause 446 MiB
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