| commit | b93a2451e06f8068ddb6495c0c28b7845c6ce560 | [log] [tgz] |
|---|---|---|
| author | Yves Gerey <yvesg@webrtc.org> | Fri Jul 19 20:46:13 2019 |
| committer | Commit Bot <commit-bot@chromium.org> | Mon Jul 22 14:22:48 2019 |
| tree | dd7c91ea73ed7f1fe6cfef449c8e5563e8a86a7e | |
| parent | e34d62caecb44bed9207e5563b9aef0de2c150f9 [diff] |
[Unit tests] Remove race condition and dangling pointer to mock.
Lifetime issue: "webrtc_audio_module_rec_thread" was still accessing
AudioTransport mock at and after its destruction.
Bug: webrtc:9751
Change-Id: I24308077cdeb77e570b8ec74098f1ae3397b7155
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/146217
Reviewed-by: Henrik Andreassson <henrika@webrtc.org>
Commit-Queue: Yves Gerey <yvesg@google.com>
Cr-Commit-Position: refs/heads/master@{#28635}
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.
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.