gstreamer/docs/plugins/inspect/plugin-webrtcdsp.xml

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<plugin>
<name>webrtcdsp</name>
<description>Voice pre-processing using WebRTC Audio Processing Library</description>
<filename>../../ext/webrtcdsp/.libs/libgstwebrtcdsp.so</filename>
<basename>libgstwebrtcdsp.so</basename>
<version>1.9.0.1</version>
<license>LGPL</license>
<source>gst-plugins-bad</source>
<package>WebRTCDsp</package>
<origin>http://git.collabora.com</origin>
<elements>
<element>
<name>webrtcdsp</name>
<longname>Voice Processor (AGC, AEC, filters, etc.)</longname>
<class>Generic/Audio</class>
<description>Pre-processes voice with WebRTC Audio Processing Library</description>
<author>Nicolas Dufresne &lt;nicolas.dufresne@collabora.com&gt;</author>
<pads>
<caps>
<name>sink</name>
<direction>sink</direction>
<presence>always</presence>
<details>audio/x-raw, format=(string)S16LE, layout=(string)interleaved, rate=(int){ 48000, 32000, 16000, 8000 }, channels=(int)[ 1, 2147483647 ]</details>
</caps>
<caps>
<name>src</name>
<direction>source</direction>
<presence>always</presence>
<details>audio/x-raw, format=(string)S16LE, layout=(string)interleaved, rate=(int){ 48000, 32000, 16000, 8000 }, channels=(int)[ 1, 2147483647 ]</details>
</caps>
</pads>
</element>
<element>
<name>webrtcechoprobe</name>
<longname>Accoustic Echo Canceller probe</longname>
<class>Generic/Audio</class>
<description>Gathers playback buffers for webrtcdsp</description>
<author>Nicolas Dufresne &lt;nicolas.dufrsesne@collabora.com&gt;</author>
<pads>
<caps>
<name>sink</name>
<direction>sink</direction>
<presence>always</presence>
<details>audio/x-raw, format=(string)S16LE, layout=(string)interleaved, rate=(int){ 48000, 32000, 16000, 8000 }, channels=(int)[ 1, 2147483647 ]</details>
</caps>
<caps>
<name>src</name>
<direction>source</direction>
<presence>always</presence>
<details>audio/x-raw, format=(string)S16LE, layout=(string)interleaved, rate=(int){ 48000, 32000, 16000, 8000 }, channels=(int)[ 1, 2147483647 ]</details>
</caps>
</pads>
</element>
</elements>
</plugin>