webrtcdsp: GStreamer Bad Plugins 1.0 Plugins Reference Manual

webrtcdsp

webrtcdsp — Audio Filter using WebRTC Audio Processing library

Properties

gboolean	echo-cancel	Read / Write / Construct
gboolean	experimental-agc	Read / Write / Construct
gboolean	extended-filter	Read / Write / Construct
gboolean	gain-control	Read / Write / Construct
gboolean	high-pass-filter	Read / Write / Construct
gboolean	noise-suppression	Read / Write / Construct
gchar *	probe	Read / Write / Construct
GstWebrtcEchoSuppressionLevel	echo-suppression-level	Read / Write / Construct
GstWebrtcNoiseSuppressionLevel	noise-suppression-level	Read / Write / Construct
gboolean	delay-agnostic	Read / Write / Construct
gint	compression-gain-db	Read / Write / Construct
GstWebrtcGainControlMode	gain-control-mode	Read / Write / Construct
gboolean	limiter	Read / Write / Construct
gint	startup-min-volume	Read / Write / Construct
gint	target-level-dbfs	Read / Write / Construct
gboolean	voice-detection	Read / Write / Construct
gint	voice-detection-frame-size-ms	Read / Write / Construct
GstWebrtcVoiceDetectionLikelihood	voice-detection-likelihood	Read / Write / Construct

Types and Values

GstWebrtcDsp

Object Hierarchy

    GObject
    ╰── GInitiallyUnowned
        ╰── GstObject
            ╰── GstElement
                ╰── GstBaseTransform
                    ╰── GstAudioFilter
                        ╰── GstWebrtcDsp

Description

A voice enhancement filter based on WebRTC Audio Processing library. This library provides a whide variety of enhancement algorithms. This element tries to enable as much as possible. The currently enabled enhancements are High Pass Filter, Echo Canceller, Noise Suppression, Automatic Gain Control, and some extended filters.

While webrtcdsp element can be used alone, there is an exception for the echo canceller. The audio canceller need to be aware of the far end streams that are played to loud speakers. For this, you must place a webrtcechoprobe element at that far end. Note that the sample rate must match between webrtcdsp and the webrtechoprobe. Though, the number of channels can differ. The probe is found by the DSP element using it's object name. By default, webrtcdsp looks for webrtcechoprobe0, which means it just work if you have a single probe and DSP.

The probe can only be used within the same top level GstPipeline. Additonally, to simplify the code, the probe element must be created before the DSP sink pad is activated. It does not need to be in any particular state and does not even need to be added to the pipeline yet.

Example launch line

As a conveniance, the echo canceller can be tested using an echo loop. In this configuration, one would expect a single echo to be heard.

1	gst-launch-1.0 pulsesrc ! webrtcdsp ! webrtcechoprobe ! pulsesink

In real environment, you'll place the probe before the playback, but only process the far end streams. The DSP should be placed as close as possible to the audio capture. The following pipeline is astracted and does not represent a real pipeline.

1 2	gst-launch-1.0 far-end-src ! audio/x-raw,rate=48000 ! webrtcechoprobe ! pulsesink \ pulsesrc ! audio/x-raw,rate=48000 ! webrtcdsp ! far-end-sink

Synopsis

Element Information

plugin	webrtcdsp
author	Nicolas Dufresne <nicolas.dufresne@collabora.com>
class	Generic/Audio

Element Pads

name	sink
direction	sink
presence	always
details	audio/x-raw, format=(string)S16LE, layout=(string)interleaved, rate=(int){ 48000, 32000, 16000, 8000 }, channels=(int)[ 1, 2147483647 ]

name	src
direction	source
presence	always
details	audio/x-raw, format=(string)S16LE, layout=(string)interleaved, rate=(int){ 48000, 32000, 16000, 8000 }, channels=(int)[ 1, 2147483647 ]

Functions

Types and Values

GstWebrtcDsp

typedef struct _GstWebrtcDsp GstWebrtcDsp;