| commit | ae95ff32ff6920660aacf0978a3cb6c4be595a40 | [log] [tgz] |
|---|---|---|
| author | glaznev <glaznev@webrtc.org> | Thu Feb 04 11:47:12 2016 -0800 |
| committer | Commit bot <commit-bot@chromium.org> | Thu Feb 04 19:47:20 2016 +0000 |
| tree | 01df53711c7fcda4623d6f25607891397e006bc0 | |
| parent | a92d6be4111c4b1d3876093ee46bc8713dad5f7f [diff] |
Add more logging and fix PTS overflow for HW decoder. - Reduce maximum pending frames for H.264 decoder to 8. - Log data for next 2 output frames every time frame drop happens or decoder drain is triggered. - When timeout happens for dequeueInputBuffer call try to drain the decoder and get input buffer one more time. - Fix PTS values overflow. Review URL: https://codereview.webrtc.org/1661203002 Cr-Commit-Position: refs/heads/master@{#11492}
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.