Modernize WebRTC integration tests to support simulated time - Introduce internal::PeerConnectionIntegrationTestBase to allow injecting TimeController and customizing thread creation. - Add PeerConnectionIntegrationTestWithSimulatedTime for tests that run in simulated time. - Refactor VirtualSocketServer and FirewallSocketServer management to use std::unique_ptr and make ownership explicit, avoiding leaks. - Update FakeAudioCaptureModule to support running on a simulated thread. Note that a task queue does not work - switching to task queue caused timeouts on many bots - Update various integration tests to use the new base classes and support simulated time. - Add TODO for future unification of NewGetStats implementations (https://issues.webrtc.org/501310910). Bug: webrtc:42223992 Change-Id: I15a23b30aad728e9fe2c890f5195bff46a6a6964 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/462820 Reviewed-by: Tomas Gunnarsson <tommi@webrtc.org> Commit-Queue: Evan Shrubsole <eshr@webrtc.org> Cr-Commit-Position: refs/heads/main@{#47462}
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 here for instructions on how to get started developing with the native code.
Authoritative list of directories that contain the native API header files.