357f596558
This will allow clients to include only the software codecs they need rather than being forced to bundle them all. - libjingle_peerconnection_jni keeps its allow_poison for now, until dependent targets bundle their own codecs explicitly. - native_api_codecs and native_api_video lose their allow_poison because dependent targets are already bundling codecs explicitly. - libjingle_peerconnection_metrics_default_jni and native_api_peerconnection lose their allow_poison because they were not actually poisoned. legacy_hwcodecs_jni and default_video_codec_factory_jni exist for clients that want to continue bundling the same codecs they get by default today. Bug: webrtc:7925 Change-Id: Idf853a6bc77f43decd35ad2a0f467937fec8f8b5 Reviewed-on: https://webrtc-review.googlesource.com/c/108221 Reviewed-by: Sami Kalliomäki <sakal@webrtc.org> Commit-Queue: Jonathan Yu <yujo@chromium.org> Cr-Commit-Position: refs/heads/master@{#25564}
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 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.
More info
- Official web site: http://www.webrtc.org
- Master source code repo: https://webrtc.googlesource.com/src
- Samples and reference apps: https://github.com/webrtc
- Mailing list: http://groups.google.com/group/discuss-webrtc
- Continuous build: http://build.chromium.org/p/client.webrtc
- Coding style guide
- Code of conduct