Jonas Oreland 93a9d19d4e p2p_transport_channel: Add estimated disconnected time to CandidatePairChangeEvent
This patch adds a computed estimate on how long the ice stack
was disconnected before switching to a new connection.

The metric is currently computed as now - max(connection->last_data_recevied())
and has resonably good precision.

Bug: webrtc:11862
Change-Id: I8950d55f0eadcf164de089cdb715b4f7eed0a4c3
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/182002
Reviewed-by: Sami Kalliomäki <sakal@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Jonas Oreland <jonaso@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#31969}
2020-08-20 11:40:01 +00:00
2018-10-05 14:40:21 +00:00
2020-02-27 14:27:23 +00:00
2019-10-28 12:27:50 +00:00
.gn
2020-03-18 18:04:41 +00:00
2020-03-30 12:15:56 +00:00
2020-07-13 11:42:07 +00:00
2020-04-16 11:08:43 +00:00
2020-07-15 17:50:55 +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 here 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

Description
The idea is to make CMake build for WebRTC m130 version - for audio processing module
Readme BSD-3-Clause 446 MiB
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