0c0c602653
Switching to an AGC implementation agnostic solution for the input volume emulation functionality offered by the `capture_levels_adjuster` sub-module. This CL also fixes a (silent) bug due to which, when the input volume is emulated via the capture adjuster sub-module, AGC2 reads an incorrect value for the applied input volume. Tested: audioproc_f with `--analog_mic_gain_emulation 1` used to verify bit-exactness for one Wav file and one AEC dump for which the input volume varies. Bug: webrtc:7494, b/241923537 Change-Id: Ide3085f9a5dfd85888ad812ebd56faa175fb2ba7 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/273902 Reviewed-by: Per Åhgren <peah@webrtc.org> Commit-Queue: Alessio Bazzica <alessiob@webrtc.org> Cr-Commit-Position: refs/heads/main@{#38053}
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 here 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
- 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: https://ci.chromium.org/p/webrtc/g/ci/console
- Coding style guide
- Code of conduct
- Reporting bugs
- Documentation