6dc82e8f8b
NetEq tapers down the audio produced through loss concealment when the expansion has been going on for some time. When the audio packets starts coming in again, there is a ramp-up that happens. This ramp-up could before this change extend over more than one 10 ms block, which made keeping track of the scaling factor necessary. With this change, we make this ramp-up quicker in the rare cases when it lasted more than 10 ms, so that it always ramps up to 100% within one block. This way, we can remove the mute_factor_array. This change breaks bit-exactness, but careful listening could not reveal an audible difference. This change is a part of a larger refactoring of NetEq's PLC code. Bug: webrtc:9180 Change-Id: I4c513ce3ed8d66f9beec2abfb1f0c7ffaac7a21e Reviewed-on: https://webrtc-review.googlesource.com/77180 Commit-Queue: Henrik Lundin <henrik.lundin@webrtc.org> Reviewed-by: Minyue Li <minyue@webrtc.org> Cr-Commit-Position: refs/heads/master@{#23342}
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
- 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
- Code of conduct