Andreas Pehrson b1a174041d Relax VideoCaptureImpl::IncomingFrame size check
When testing manually with gstreamer and v4l2loopback, the incoming
buffer is often larger than the expected size. This change allows
such frames, while still logging the error.

Bug: webrtc:14830
Change-Id: I399aa55af6437d75b50830166a667547f6d144d4
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/291530
Commit-Queue: Stefan Holmer <stefan@webrtc.org>
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#39972}
2023-05-03 06:27:25 +00:00
2023-04-27 09:53:00 +00:00
2023-03-13 13:16:22 +00:00
2023-04-27 09:53:00 +00:00
2023-04-27 09:53:00 +00:00
2023-04-26 13:01:13 +00:00
2023-04-27 09:53:00 +00:00
2023-02-13 10:30:38 +00:00
.gn
2023-03-13 12:37:57 +00:00
2022-02-20 14:22:13 +00:00
2021-12-08 08:53:00 +00:00
2022-12-02 09:21:47 +00:00
2022-12-02 09:21:47 +00:00
2022-05-13 09:01:34 +00:00
2023-04-12 07:09:41 +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
Languages
C++ 90.3%
Java 2.9%
C 2.2%
Objective-C++ 2%
Python 1.3%
Other 1%