Åsa Persson f06baccf81 Add test that verifies that VideoEncoderConfig max_framerate is reported to source.
Update test::CreateVideoStreams to use the configured framerate if set.

Bug: webrtc:9597
Change-Id: I3c49dbf38e6c4935e864c4168be1d7e19a054a2c
Reviewed-on: https://webrtc-review.googlesource.com/c/105621
Commit-Queue: Åsa Persson <asapersson@webrtc.org>
Reviewed-by: Rasmus Brandt <brandtr@webrtc.org>
Reviewed-by: Sebastian Jansson <srte@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25195}
2018-10-16 08:50:39 +00:00
2018-10-05 14:40:21 +00:00
2018-10-15 06:59:19 +00:00
.gn
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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
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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
Readme BSD-3-Clause 446 MiB
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