| commit | 6c2c13af06b32778b86950681758a7970d1c5d9e | [log] [tgz] |
|---|---|---|
| author | Niels Möller <nisse@webrtc.org> | Thu Mar 29 13:06:51 2018 +0200 |
| committer | Commit Bot <commit-bot@chromium.org> | Thu Mar 29 11:45:18 2018 +0000 |
| tree | 977f70b5e5d5571d6118e8cef06ef6046fe2f3ff | |
| parent | 27e41c52f597aefaf1946808e0cdff1848be0593 [diff] |
Revert "Reland "Move rtp-specific config out of EncoderSettings."" This reverts commit 04dd1768625eb2241d1fb97fd0137897e703e266. Reason for revert: Regression in ramp up perf tests. Original change's description: > Reland "Move rtp-specific config out of EncoderSettings." > > This is a reland of bc900cb1d1810fcf678fe41cf1e3966daa39c88c > > Original change's description: > > Move rtp-specific config out of EncoderSettings. > > > > In VideoSendStream::Config, move payload_name and payload_type from > > EncoderSettings to Rtp. > > > > EncoderSettings now contains configuration for VideoStreamEncoder only, > > and should perhaps be renamed in a follow up cl. It's no longer > > passed as an argument to VideoCodecInitializer::SetupCodec. > > > > The latter then needs a different way to know the codec type, > > which is provided by a new codec_type member in VideoEncoderConfig. > > > > Bug: webrtc:8830 > > Change-Id: Ifcc691aef1ee6a95e43c0452c5e630d92a511cd6 > > Reviewed-on: https://webrtc-review.googlesource.com/62062 > > Commit-Queue: Niels Moller <nisse@webrtc.org> > > Reviewed-by: Magnus Jedvert <magjed@webrtc.org> > > Reviewed-by: Stefan Holmer <stefan@webrtc.org> > > Reviewed-by: Rasmus Brandt <brandtr@webrtc.org> > > Cr-Commit-Position: refs/heads/master@{#22532} > > Bug: webrtc:8830 > Change-Id: If88ef7d57cdaa4fae3c7b2a97ea5a6e1b833e019 > Reviewed-on: https://webrtc-review.googlesource.com/63721 > Reviewed-by: Rasmus Brandt <brandtr@webrtc.org> > Reviewed-by: Stefan Holmer <stefan@webrtc.org> > Commit-Queue: Niels Moller <nisse@webrtc.org> > Cr-Commit-Position: refs/heads/master@{#22595} TBR=brandtr@webrtc.org,magjed@webrtc.org,nisse@webrtc.org,stefan@webrtc.org Bug: webrtc:8830,chromium:827080 Change-Id: Iaaf146de91ec5c0d741b8efdf143f7e173084fef Reviewed-on: https://webrtc-review.googlesource.com/65520 Commit-Queue: Niels Moller <nisse@webrtc.org> Reviewed-by: Niels Moller <nisse@webrtc.org> Cr-Commit-Position: refs/heads/master@{#22677}
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.