Reason for revert: Temporarily reverting as this causes some issues with perf tests. Especially tests with packet loss no longer works. Original issue's description: > Adding support for simulcast and spatial layers into VideoQualityTest > > The CL includes several changes: > - Adding flags describing the streams and spatial layers. > - Reorganizing the order of the flags, to make them easier to maintain. > - Adding a member .params_ to VideoQualityAnalyzer. > (instead of passing it to every member function manually) > - Updating VideoAnalyzer to support simulcast. > (select appropriate ssrc and fix timestamps which are sometimes increased by 1) > - VP9EncoderImpl already had code for automatic calculation of bitrate for each layer. > Changing to first read bitrates and resolution ratios from the flags, if specified. > If not specified, reverting to the old code are setting the values automatically. > - Changing the parameters in LayerFilteringTransport, replacing > xx_discard_thresholds with selected_xx, to make it easier to use for the end user. > > Committed: https://crrev.com/87f83a9a27d657731ccb54025bc04ccad0da136e > Cr-Commit-Position: refs/heads/master@{#10215} TBR=pbos@webrtc.org,mflodman@webrtc.org,ivica@webrtc.org NOPRESUBMIT=true NOTREECHECKS=true NOTRY=true Review URL: https://codereview.webrtc.org/1397363002 Cr-Commit-Position: refs/heads/master@{#10252}
Revert of Adding support for simulcast and spatial layers into VideoQualityTest (patchset #10 id:180001 of https://codereview.webrtc.org/1353263005/ )
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. This page is maintained by the Google Chrome team.
Development
See http://www.webrtc.org/native-code/development for instructions on how to get started developing with the native code.
More info
- Official web site: http://www.webrtc.org
- Master source code repo: https://chromium.googlesource.com/external/webrtc
- Samples and reference apps: https://github.com/webrtc
- Mailing list: http://groups.google.com/group/discuss-webrtc
- Continuous build: http://build.chromium.org/p/client.webrtc
Description
The idea is to make CMake build for WebRTC m130 version - for audio processing module
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