InputVolumeStatsReporter: replace WebRTC.Audio.AgcSetLevel
The `WebRTC.Audio.AgcSetLevel` name is misleading and the histogram is logged for each channel - but the input volume is one for all the channels. Changes: - `WebRTC.Audio.Apm.RecommendedInputVolume.OnChangeToMatchTarget` is the new name - Now available not only in `AgcManagerDirect` (AGC1), but also in `InputVolumeController` (AGC2) - Logged once and not for each channel - Also add the following AGC implementation agnostic histograms - `WebRTC.Audio.Apm.AppliedInputVolume.OnChange` - `WebRTC.Audio.Apm.RecommendedInputVolume.OnChange` - Fix `SpeechSamplesReader::Feed()` in the unit tests, which did not set the applied input volume and apply the recommended one The histogram definitions are updated in crrev.com/c/4087426. Bug: webrtc:7494 Change-Id: I03c5dfb08165805215ca2c4bb6509b16de8d68da Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/287081 Commit-Queue: Alessio Bazzica <alessiob@webrtc.org> Reviewed-by: Hanna Silen <silen@webrtc.org> Cr-Commit-Position: refs/heads/main@{#38852}
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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
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- Reporting bugs
- Documentation
Description
The idea is to make CMake build for WebRTC m130 version - for audio processing module
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