webrtcdsp

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

 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.

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

Hierarchy

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

Factory details

Authors: – Nicolas Dufresne

Classification: – Generic/Audio

Rank – none

Plugin – webrtcdsp

Package – GStreamer Bad Plug-ins

Pad Templates

`sink`

audio/x-raw:
         format: S16LE
         layout: interleaved
           rate: { (int)48000, (int)32000, (int)16000, (int)8000 }
       channels: [ 1, 2147483647 ]
audio/x-raw:
         format: F32LE
         layout: non-interleaved
           rate: { (int)48000, (int)32000, (int)16000, (int)8000 }
       channels: [ 1, 2147483647 ]

Presence – always

Direction – sink

Object type – GstPad

`src`

audio/x-raw:
         format: S16LE
         layout: interleaved
           rate: { (int)48000, (int)32000, (int)16000, (int)8000 }
       channels: [ 1, 2147483647 ]
audio/x-raw:
         format: F32LE
         layout: non-interleaved
           rate: { (int)48000, (int)32000, (int)16000, (int)8000 }
       channels: [ 1, 2147483647 ]

Presence – always

Direction – src

Object type – GstPad