| commit | f87536c9de24ce25c398c1f7a413dc8b80208362 | [log] [tgz] |
|---|---|---|
| author | Erik Språng <sprang@webrtc.org> | Thu Mar 05 09:14:04 2020 |
| committer | Commit Bot <commit-bot@chromium.org> | Mon Mar 09 13:41:35 2020 |
| tree | 31d8a53ac4edc37923018021ab5ca65e1b35fcbb | |
| parent | 269d68f521762d7993a83d20fa42be2f4f5d9092 [diff] |
Reland "Reland "Refactors UlpFec and FlexFec to use a common interface."" This is a reland of 49734dc0faa69616a58a1a95c7fc61a4610793cf Patchset 2 contains a fix for the fuzzer set up. Since we now parse an RtpPacket out of the fuzzer data, the header needs to be correct, otherwise we fail before even reaching the FEC code that we actually want to test. Bug: webrtc:11340, chromium:1052323, chromium:1055974 TBR=stefan@webrtc.org Original change's description: > Reland "Refactors UlpFec and FlexFec to use a common interface." > > This is a reland of 11af1d7444fd7438766b7bc52cbd64752d72e32e > > Original change's description: > > Refactors UlpFec and FlexFec to use a common interface. > > > > The new VideoFecGenerator is now injected into RtpSenderVideo, > > and generalizes the usage. > > This also prepares for being able to genera FEC in the RTP egress > > module. > > > > Bug: webrtc:11340 > > Change-Id: I8aa873129b2fb4131eb3399ee88f6ea2747155a3 > > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/168347 > > Reviewed-by: Stefan Holmer <stefan@webrtc.org> > > Reviewed-by: Sebastian Jansson <srte@webrtc.org> > > Reviewed-by: Rasmus Brandt <brandtr@webrtc.org> > > Commit-Queue: Erik Språng <sprang@webrtc.org> > > Cr-Commit-Position: refs/heads/master@{#30515} > > Bug: webrtc:11340, chromium:1052323 > Change-Id: Id646047365f1c46cca9e6f3e8eefa5151207b4a0 > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/168608 > Commit-Queue: Erik Språng <sprang@webrtc.org> > Reviewed-by: Stefan Holmer <stefan@webrtc.org> > Cr-Commit-Position: refs/heads/master@{#30593} Bug: webrtc:11340, chromium:1052323 Change-Id: Ib8925f44e2edfcfeadc95c845c3bfc23822604ed Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/169222 Commit-Queue: Erik Språng <sprang@webrtc.org> Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org> Cr-Commit-Position: refs/heads/master@{#30724}
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 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.