Henrik Boström 89f095cb07 Deflake PeerConnectionSimulcastMediaFlowTests due to unstopped sources.
Only in testing environments are the task queues shut down while sources
still have media flowing. It's still not clear why heap-use-after-free
happens, since it should be enough to close the PC, but it is clear that
the crash is happening due to frames flowing while the test is shutting
down, which is not something happening outside of testing.

In an attempt to deflake, this CL makes sure to manually stop the
test-only sources before closing the peer connection.

Bug: webrtc:15018
Change-Id: I48ee131a8994c9c4caee1bb4875580d255b97da1
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/299944
Commit-Queue: Henrik Boström <hbos@webrtc.org>
Auto-Submit: Henrik Boström <hbos@webrtc.org>
Reviewed-by: Jeremy Leconte <jleconte@google.com>
Cr-Commit-Position: refs/heads/main@{#39752}
2023-04-03 16:33:01 +00:00
2023-03-13 13:16:22 +00:00
2023-04-03 07:12:11 +00:00
2023-02-13 10:30:38 +00:00
.gn
2023-03-13 12:37:57 +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
2022-11-17 21:29:53 +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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