de5507d31b0886631de2bbb6f081764d3beb39f2

Currently test code passes pointer to temporary objects, while RtcpSender passes raw pointers to objects that are then seen as owned, and will be manually deleted by a overloaded destructor, which is scary and fragile. This CL moves all usage to std::unique_ptr<RtcpPacket> instead, which may create some heap churn in unit tests but that should be fine. Bug: webrtc:11925 Change-Id: I981bc7ccd6a74115c5a3de64b8427adbf3f16cc7 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/183920 Commit-Queue: Erik Språng <sprang@webrtc.org> Reviewed-by: Danil Chapovalov <danilchap@webrtc.org> Cr-Commit-Position: refs/heads/master@{#32084}
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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
- 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
Description
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