10a7d23be5
RtpSenderBase::SetParametersInternal stores init_parameters_ if media_channel_ does not exist. When RtpSenderBase::SetSsrc is called, init_parameters_ is used to set the initial encoding parameters and degradation_preference. However, if no encoding parameter is specified, degradation_preference will not be set. This CL modifies the RtpSender so that degradation_preference is not ignored even in this case. Bug: webrtc:14279 Change-Id: I7e95ecdf5fcb19037e4f118981d1314d78ffca5a Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/268960 Reviewed-by: Henrik Boström <hbos@webrtc.org> Commit-Queue: Daniel.L (Byoungchan) Lee <daniel.l@hpcnt.com> Cr-Commit-Position: refs/heads/main@{#37574}
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: https://ci.chromium.org/p/webrtc/g/ci/console
- Coding style guide
- Code of conduct
- Reporting bugs
- Documentation