commit | ce86e2965c15ffc6f054a0ea5daab89f4fc622d8 | [log] [tgz] |
---|---|---|
author | Sami Kalliomäki <sakal@webrtc.org> | Wed Nov 22 12:20:11 2017 |
committer | Commit Bot <commit-bot@chromium.org> | Wed Nov 22 16:10:27 2017 |
tree | 677d86db6ff57d5bea15634036e821ce12839760 | |
parent | 1882d8509a138468c6dd8437506973d236c80e62 [diff] |
Android: Use RTP timestamp instead of capture_time_ms_ in VideoDecoderWrapper. capture_time_ms_ is always 0 for frames received from the network. This caused a bug because it was used as an unique identified. Bug: b/68271454 Change-Id: Ic4417a52e61cf2b0cd796a89207a90b603a16590 Reviewed-on: https://webrtc-review.googlesource.com/24940 Reviewed-by: Magnus Jedvert <magjed@webrtc.org> Commit-Queue: Sami Kalliomäki <sakal@webrtc.org> Cr-Commit-Position: refs/heads/master@{#20837}
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.