glaznev ae95ff32ff 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}
2016-02-04 19:47:20 +00:00
2014-06-17 08:54:03 +00:00
2015-09-11 09:04:09 +00:00

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.

Development

See http://www.webrtc.org/native-code/development for instructions on how to get started developing with the native code.

More info

Description
The idea is to make CMake build for WebRTC m130 version - for audio processing module
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