Commit graph

3 commits

Author SHA1 Message Date
Nicolas Dufresne
bdd1dd04ea webrtcdsp: Avoid redefinition of GST_USE_UNSTABLE_API 2018-08-03 07:57:02 -04:00
George Kiagiadakis
d299c27892 webrtcdsp: add support for using F32/non-interleaved buffers
This is the native format that is in use by the webrtc audio processing
library internally, so this avoids internal {de,}interleaving and
format conversion (S16->F32 and back)

https://bugzilla.gnome.org/show_bug.cgi?id=793605
2018-08-03 13:20:12 +03:00
Nicolas Dufresne
398f7059fc webrtcdsp: Add WebRTC Audio Processing support
This DSP library can be used to enhance voice signal for real time
communication call. In implements multiple filters like noise reduction,
high pass filter, echo cancellation, automatic gain control, etc.

The webrtcdsp element can be used along, or with the help of the
webrtcechoprobe if echo cancellation is enabled. The echo probe should
be placed as close as possible to the audio sink, while the DSP is
generally place close to the audio capture. For local testing, one can
use an echo loop pipeline like the following:

  autoaudiosrc ! webrtcdsp ! webrtcechoprobe ! autoaudiosink

This pipeline should produce a single echo rather then repeated echo.
Those elements works if they are placed in the same top level pipeline.

https://bugzilla.gnome.org/show_bug.cgi?id=767800
2016-06-21 13:46:00 -04:00