karllen.zheng@ringcentral.com f92f39e618 Increase the default maximum jitter buffer size to 200 packets for Android.
Bug: webrtc:42220461
Change-Id: I6dadd8357173c79af0290199d9ed12a0e0417f4f
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/361673
Reviewed-by: Sergey Silkin <ssilkin@webrtc.org>
Reviewed-by: Henrik Lundin <henrik.lundin@webrtc.org>
Reviewed-by: Jakob Ivarsson‎ <jakobi@webrtc.org>
Commit-Queue: Sergey Silkin <ssilkin@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#42970}
2024-09-06 11:21:34 +00:00
2024-09-02 20:50:58 +00:00
2024-09-04 07:58:47 +00:00
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2022-02-20 14:22:13 +00:00
2024-09-02 20:50:58 +00:00
2022-12-02 09:21:47 +00:00
2023-09-25 15:56:09 +00:00
2024-05-27 12:46:57 +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 here 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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