In preparation of coming CLs that will add an AGC interface to make the gain controller injectable. This CL simplifies AGC2 (dummy sub-module of audioproc_f) since it only implements the fixed digital mode with hard-clipping - i.e., no limiter is used. The AGC2 config now includes the fixed gain to apply and audioproc_f has been adapted accordingly. Finally, this CL slightly simplifies the AGC2 integration into APM. This CL is a continuation of https://codereview.webrtc.org/2995043002/ Bug: webrtc:7494 Change-Id: I3d554ea4dc6208928352059feb14987edabf14c7 Reviewed-on: https://webrtc-review.googlesource.com/4661 Commit-Queue: Alessio Bazzica <alessiob@webrtc.org> Reviewed-by: Per Åhgren <peah@webrtc.org> Cr-Commit-Position: refs/heads/master@{#20278}
Reland of Fix the video buffer size should take rtt into consideration (patchset #2 id:160001 of https://codereview.chromium.org/3002033002/ )
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
- Official web site: http://www.webrtc.org
- Master source code repo: https://webrtc.googlesource.com/src
- 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
- Coding style guide
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Description
The idea is to make CMake build for WebRTC m130 version - for audio processing module
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