1d50cb61d8
This is a reland of 19df870d924662e3b6efb86078d31a8e086b38b5 Patchset 1 is the original. Subsequent patchset changes threadchecker that crashed with downstream code. Original change's description: > Reland "Allows FEC generation after pacer step." > > This is a reland of 75fd127640bdf1729af6b4a25875e6d01f1570e0 > > Patchset 2 contains a fix. Old code can in factor call > RtpRtcpImpl::FetchFec(). It should only be a noop since deferred fec > is not supported there - we shouldn't crash. > > Original change's description: > > Allows FEC generation after pacer step. > > > > Split out from https://webrtc-review.googlesource.com/c/src/+/173708 > > This CL enables FEC packets to be generated as media packets are sent, > > rather than generated, i.e. media packets are inserted into the fec > > generator after the pacing stage rather than at packetization time. > > > > This may have some small impact of performance. FEC packets are > > typically only generated when a new packet with a marker bit is added, > > which means FEC packets protecting a frame will now be sent after all > > of the media packets, rather than (potentially) interleaved with them. > > Therefore this feature is currently behind a flag so we can examine the > > impact. Once we are comfortable with the behavior we'll make it default > > and remove the old code. > > > > Note that this change does not include the "protect all header > > extensions" part of the original CL - that will be a follow-up. > > > > Bug: webrtc:11340 > > Change-Id: I3fe139c5d53968579b75b91e2612075451ff0f5d > > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/177760 > > Commit-Queue: Erik Språng <sprang@webrtc.org> > > Reviewed-by: Sebastian Jansson <srte@webrtc.org> > > Cr-Commit-Position: refs/heads/master@{#31558} > > Bug: webrtc:11340 > Change-Id: I2ea49ee87ee9ff409044e34a777a7dd0ae0a077f > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/177984 > Commit-Queue: Erik Språng <sprang@webrtc.org> > Reviewed-by: Sebastian Jansson <srte@webrtc.org> > Cr-Commit-Position: refs/heads/master@{#31613} Bug: webrtc:11340 Change-Id: Ib741c8c284f523c959f8aca454088d9eee7b17f8 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/178600 Reviewed-by: Sebastian Jansson <srte@webrtc.org> Commit-Queue: Erik Språng <sprang@webrtc.org> Cr-Commit-Position: refs/heads/master@{#31619}
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