stefan 8497fdde43 Add functionality which limits the number of bytes on the network.
The limit is based on the bandwidth delay product, but also adds some additional slack to compensate for the sawtooth-like BWE pattern and the slowness of the encoder rate control. The delay is estimated based on the time from sending a packet until an ack is received. Since acks are received in bursts (feedback is only sent periodically), a min filter is used to estimate the rtt.

Whenever the in flight bytes reaches the congestion window, the pacer is paused, which in turn will result in send-side queues growing. Eventually the encoders will be paused as the pacer queue grows large (currently 2 seconds).

BUG=webrtc:7926

Review-Url: https://codereview.webrtc.org/2918323002
Cr-Commit-Position: refs/heads/master@{#19289}
2017-08-09 14:17:33 +00:00
2017-06-30 10:04:59 +00:00
.gn
2017-08-02 08:26:18 +00:00
2017-06-30 10:04:59 +00:00
2017-01-20 20:45:07 +00:00
2017-07-19 17:40:47 +00:00
2017-03-23 10:46:00 +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.

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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