Commit graph

7 commits

Author SHA1 Message Date
Philippe Normand
158a2b3fd1 webrtcdsp: Fix documentation markup 2020-03-15 12:44:31 +00:00
Aaron Boxer
6d3429af34 documentation: fixed a heap o' typos 2019-11-05 09:11:25 -05:00
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
George Kiagiadakis
da44fea1d1 webrtcechoprobe: return from _read() early if the probe is not configured yet
https://bugzilla.gnome.org/show_bug.cgi?id=780642
2017-03-29 16:45:12 +03:00
Nicolas Dufresne
74c0d5fdd2 webrtcdsp: Add delay-agnostic property
In this mode, we let WebRTC Audio Processing figure-out the delay. This
is useful when the latency reported by the stack cannot be trusted. Note
that in this mode, the leaking of echo during packet lost is much worst.
It is recommanded to use PLC (e.g. spanplc, or opus built-in plc).

In this mode, we don't do any synchronization. Instead, we simply process all
the available reverse stream data as it comes.
2016-07-13 23:17:21 -04:00
Nicolas Dufresne
71c9cdeff4 webrtcdsp: Rewrite echo data synchronization
The previous code would run out of sync if there was packet lost
or clock skews. When that happened, the echo cancellation feature would
completely stop working. As this is crucial for audio calls, this patch
re-implement synchronization completely.

Instead of letting it drift until next discont, we now synchronize
against the record data at every iteration. This way we simply never
let the stream drift for longer then 10ms period. We also shorter the
delay by using the latency up the probe (basically excluding the sink
latency. This is a decent delay to avoid starving in the probe queue.

https://bugzilla.gnome.org/show_bug.cgi?id=768009
2016-06-30 09:27:03 -04:00
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