| commit | 156e3afa7f945069b206c3e252ea3609869bfad5 | [log] [tgz] |
|---|---|---|
| author | kthelgason <kthelgason@webrtc.org> | Mon Mar 06 08:04:32 2017 |
| committer | Commit bot <commit-bot@chromium.org> | Mon Mar 06 08:04:32 2017 |
| tree | 8679f51fa294bd0131ec48a767a1a5d0cb62d639 | |
| parent | 0a3e69a9a4ab1c4d5fa8fe523a8ce960846605ef [diff] |
Optimize ParseRbsp method in H264 bitstream parser. After profiling, sakal@ found that this method was taking very long, and causing the bitstream parsing to take up to 1ms per frame. The culprit proved to be rtc::Buffer::AppendData, which was called for every byte and subsequently calls memcpy. BUG=webrtc:7293 Review-Url: https://codereview.webrtc.org/2728093002 Cr-Commit-Position: refs/heads/master@{#17051}
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.