Henrik Boström 23aff9b737 Implement RTCOutboundRtpStreamStats.totalEncodedBytesTarget.
This is a standardized metric:
https://w3c.github.io/webrtc-stats/#dom-rtcoutboundrtpstreamstats-totalencodedbytestarget

We estimate the target frame size in bytes from the current encoder
target bitrate and encoder framerate.

We would expect that the average bytes produced by the encoder would
over time match the average target, which is calculated by polling
getStats() twice and dividing the delta totalEncodedBytesTarget with
the delta framesEncoded. This is meant to make googTargetEncBitrate
obsolete.

Bug: webrtc:10446
Change-Id: Ib10ce236476a2f965582d5c536f419952926d4e6
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/137200
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Commit-Queue: Henrik Boström <hbos@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28022}
2019-05-22 10:59:39 +00:00
2018-10-05 14:40:21 +00:00
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2019-04-29 12:55:02 +00:00
2019-05-21 18:58:33 +00:00
.gn
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2019-05-21 14:44:11 +00:00
2018-12-18 12:30:58 +00:00
2019-05-17 18:11:58 +00:00
2018-07-23 15:28:48 +00:00
2018-07-23 15:28:48 +00:00
2019-04-26 12:45:06 +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

Description
The idea is to make CMake build for WebRTC m130 version - for audio processing module
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