| commit | d76a0fc5e92a794f9857f4e0f970f6068695fd07 | [log] [tgz] |
|---|---|---|
| author | erikvarga@webrtc.org <erikvarga@webrtc.org> | Tue Oct 09 12:31:28 2018 +0200 |
| committer | Commit Bot <commit-bot@chromium.org> | Tue Oct 09 12:10:55 2018 +0000 |
| tree | 089bf1f3b83a524819ededd1b6d12cf6d38e8e44 | |
| parent | b674cd103847da8639a58b60aba4df093314ae8c [diff] |
Throttle the RTP decryption error messages in the SrtpSession and SrtpTransport In order to avoid excessive logging when a large percentage of received packets are bad (e.g. when the same packets get sent several times). Bug: webrtc:9839 Change-Id: I2daed89b170adf7252624bf0da9af5a980bacc17 Reviewed-on: https://webrtc-review.googlesource.com/c/104624 Reviewed-by: Per Kjellander <perkj@webrtc.org> Reviewed-by: Danil Chapovalov <danilchap@webrtc.org> Commit-Queue: Erik Varga <erikvarga@webrtc.org> Cr-Commit-Position: refs/heads/master@{#25060}
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.