erikvarga@webrtc.org d76a0fc5e9 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}
2018-10-09 12:10:55 +00:00
2018-10-05 14:40:21 +00:00
2018-10-01 07:03:25 +00:00
2018-08-13 13:54:05 +00:00
2017-09-15 04:25:06 +00:00
2017-09-15 04:25:06 +00:00
2018-07-23 15:28:48 +00:00
2018-07-23 15:28:48 +00:00
2017-09-15 04:25:06 +00:00
2018-02-23 10:34:16 +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.

Development

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.

More info

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