commit | deb866360aeee5e0504b8c229b8c952476665a73 | [log] [tgz] |
---|---|---|
author | Philip Eliasson <philipel@webrtc.org> | Wed Nov 29 11:39:27 2017 |
committer | Commit Bot <commit-bot@chromium.org> | Wed Nov 29 11:39:41 2017 |
tree | 04599a7b13d9fe0f9ad19285a5d91d99fe26ff0f | |
parent | f82000328d01f5bd679ccac431205a485221eaa2 [diff] |
Revert "Add stereo codec header and pass it through RTP" This reverts commit 20f2133d5dbd1591b89425b24db3b1e09fbcf0b1. Reason for revert: Breaks downstream project. Original change's description: > Add stereo codec header and pass it through RTP > > - Defines CodecSpecificInfoStereo that carries stereo specific header info from > encoded image. > - Defines RTPVideoHeaderStereo that carries the above info to packetizer, > see module_common_types.h. > - Adds an RTPPacketizer and RTPDepacketizer that supports passing specific stereo > header. > - Uses new data containers in StereoAdapter classes. > > This CL is the step 3 for adding alpha channel support over the wire in webrtc. > See https://webrtc-review.googlesource.com/c/src/+/7800 for the experimental > CL that gives an idea about how it will come together. > Design Doc: https://goo.gl/sFeSUT > > Bug: webrtc:7671 > Change-Id: Ia932568fdd7065ba104afd2bc0ecf25a765748ab > Reviewed-on: https://webrtc-review.googlesource.com/22900 > Reviewed-by: Emircan Uysaler <emircan@webrtc.org> > Reviewed-by: Erik Språng <sprang@webrtc.org> > Reviewed-by: Danil Chapovalov <danilchap@webrtc.org> > Reviewed-by: Niklas Enbom <niklas.enbom@webrtc.org> > Commit-Queue: Emircan Uysaler <emircan@webrtc.org> > Cr-Commit-Position: refs/heads/master@{#20920} TBR=danilchap@webrtc.org,sprang@webrtc.org,stefan@webrtc.org,niklas.enbom@webrtc.org,emircan@webrtc.org Change-Id: I57f3172ca3c60a84537d577a574dc8018e12d634 No-Presubmit: true No-Tree-Checks: true No-Try: true Bug: webrtc:7671 Reviewed-on: https://webrtc-review.googlesource.com/26940 Reviewed-by: Philip Eliasson <philipel@webrtc.org> Commit-Queue: Philip Eliasson <philipel@webrtc.org> Cr-Commit-Position: refs/heads/master@{#20931}
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.