Sergey Silkin 41c650bea2 Use bitrate limits provided by encoder.
- Use minimum start bitrate to drop frame and adapt resolution in the
beginning of call.

- Use minimum bitrate to decide whether or not resolution should be
increased based on quality in MAINTAIN_FRAMERATE and BALANCED modes.
In BALANCED mode bitrate limits provided by the corresponding field
trial are prioritized over the limits provided by encoder.

Bug: webrtc:10853
Change-Id: I8257eb64565bcafa6ae9887a1af18e90f8400cac
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/156302
Commit-Queue: Sergey Silkin <ssilkin@webrtc.org>
Reviewed-by: Åsa Persson <asapersson@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29461}
2019-10-14 12:57:24 +00:00
2018-10-05 14:40:21 +00:00
2019-09-10 10:03:50 +00:00
2019-10-14 12:24:01 +00:00
2019-10-11 13:11:11 +00:00
2019-07-08 13:45:15 +00:00
2018-12-18 12:30:58 +00:00
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2018-07-23 15:28:48 +00:00
2019-09-03 14:55:43 +00:00
2019-08-20 14:00:49 +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
Readme BSD-3-Clause 446 MiB
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