| commit | 1642620a7d9fef7c1dca3bb0d7b172debe3b3d4c | [log] [tgz] |
|---|---|---|
| author | Erik Språng <sprang@webrtc.org> | Thu Jun 16 15:54:44 2016 +0200 |
| committer | Erik Språng <sprang@webrtc.org> | Thu Jun 16 13:54:58 2016 +0000 |
| tree | 8070de4a4d53e8f6ebbce15f4573d3cb2cc4a5a2 | |
| parent | fc3a8ee47b5998e07844da04bf9c3f1bd46040a2 [diff] |
Performance fix for H264 RBSP parsing. An rtc::Buffer is used to write output of RBSP parsing, usually one byte at a time. It turns out that container will then expand its capacity one byte at a time, for each byte reallocating the whole buffer and copying the contents over, turning this into an O(n^2) operation. Fix is for now only to preallocate the container storage. Longer term, I think we should mull over if we really need custom containers... R=pbos@webrtc.org TBR=mflodman@webrtc.org BUG= Review URL: https://codereview.webrtc.org/2073763002 . Cr-Commit-Position: refs/heads/master@{#13167}
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. This page is maintained by the Google Chrome team.
See http://www.webrtc.org/native-code/development for instructions on how to get started developing with the native code.