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90 lines
3.4 KiB
C++
90 lines
3.4 KiB
C++
/*
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* WebRTC Audio Processing Elements
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*
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* Copyright 2016 Collabora Ltd
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* @author: Nicolas Dufresne <nicolas.dufresne@collabora.com>
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*
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* This library is free software; you can redistribute it and/or
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* modify it under the terms of the GNU Lesser General Public
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* License as published by the Free Software Foundation; either
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* version 2.1 of the License, or (at your option) any later version.
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*
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* This library is distributed in the hope that it will be useful,
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* but WITHOUT ANY WARRANTY; without even the implied warranty of
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* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
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* Lesser General Public License for more details.
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*
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* You should have received a copy of the GNU Lesser General Public
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* License along with this library; if not, write to the Free Software
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* Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
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*
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*/
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/**
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* SECTION:element-webrtcdsp
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* @short_description: Audio Filter using WebRTC Audio Processing library
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*
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* A voice enhancement filter based on WebRTC Audio Processing library. This
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* library provides a whide variety of enhancement algorithms. This element
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* tries to enable as much as possible. The currently enabled enhancements are
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* High Pass Filter, Echo Canceller, Noise Suppression, Automatic Gain Control,
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* and some extended filters.
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*
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* While webrtcdsp element can be used alone, there is an exception for the
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* echo canceller. The audio canceller need to be aware of the far end streams
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* that are played to loud speakers. For this, you must place a webrtcechoprobe
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* element at that far end. Note that the sample rate must match between
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* webrtcdsp and the webrtechoprobe. Though, the number of channels can differ.
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* The probe is found by the DSP element using it's object name. By default,
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* webrtcdsp looks for webrtcechoprobe0, which means it just work if you have
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* a single probe and DSP.
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*
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* The probe can only be used within the same top level GstPipeline.
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* Additionally, to simplify the code, the probe element must be created
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* before the DSP sink pad is activated. It does not need to be in any
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* particular state and does not even need to be added to the pipeline yet.
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*
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* # Example launch line
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*
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* As a convenience, the echo canceller can be tested using an echo loop. In
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* this configuration, one would expect a single echo to be heard.
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*
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* |[
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* gst-launch-1.0 pulsesrc ! webrtcdsp ! webrtcechoprobe ! pulsesink
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* ]|
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*
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* In real environment, you'll place the probe before the playback, but only
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* process the far end streams. The DSP should be placed as close as possible
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* to the audio capture. The following pipeline is astracted and does not
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* represent a real pipeline.
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*
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* |[
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* gst-launch-1.0 far-end-src ! audio/x-raw,rate=48000 ! webrtcechoprobe ! pulsesink \
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* pulsesrc ! audio/x-raw,rate=48000 ! webrtcdsp ! far-end-sink
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* ]|
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*/
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#ifdef HAVE_CONFIG_H
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#include "config.h"
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#endif
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#include "gstwebrtcdsp.h"
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#include "gstwebrtcechoprobe.h"
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static gboolean
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plugin_init (GstPlugin * plugin)
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{
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gboolean ret = FALSE;
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ret |= GST_ELEMENT_REGISTER (webrtcdsp, plugin);
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ret |= GST_ELEMENT_REGISTER (webrtcechoprobe, plugin);
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return ret;
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}
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GST_PLUGIN_DEFINE (GST_VERSION_MAJOR,
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GST_VERSION_MINOR,
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webrtcdsp,
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"Voice pre-processing using WebRTC Audio Processing Library",
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plugin_init, VERSION, "LGPL", GST_PACKAGE_NAME, GST_PACKAGE_ORIGIN)
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