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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
53 lines
1.9 KiB
C
53 lines
1.9 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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#ifndef __GST_WEBRTC_DSP_H__
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#define __GST_WEBRTC_DSP_H__
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#include <gst/gst.h>
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#include <gst/base/gstadapter.h>
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#include <gst/base/gstbasetransform.h>
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#include <gst/audio/audio.h>
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G_BEGIN_DECLS
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#define GST_TYPE_WEBRTC_DSP (gst_webrtc_dsp_get_type())
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#define GST_WEBRTC_DSP(obj) (G_TYPE_CHECK_INSTANCE_CAST((obj),GST_TYPE_WEBRTC_DSP,GstWebrtcDsp))
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#define GST_IS_WEBRTC_DSP(obj) (G_TYPE_CHECK_INSTANCE_TYPE((obj),GST_TYPE_WEBRTC_DSP))
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#define GST_WEBRTC_DSP_CLASS(klass) (G_TYPE_CHECK_CLASS_CAST((klass) ,GST_TYPE_WEBRTC_DSP,GstWebrtcDspClass))
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#define GST_IS_WEBRTC_DSP_CLASS(klass) (G_TYPE_CHECK_CLASS_TYPE((klass) ,GST_TYPE_WEBRTC_DSP))
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#define GST_WEBRTC_DSP_GET_CLASS(obj) (G_TYPE_INSTANCE_GET_CLASS((obj) ,GST_TYPE_WEBRTC_DSP,GstWebrtcDspClass))
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typedef struct _GstWebrtcDsp GstWebrtcDsp;
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typedef struct _GstWebrtcDspClass GstWebrtcDspClass;
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struct _GstWebrtcDspClass
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{
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GstAudioFilterClass parent_class;
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};
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GType gst_webrtc_dsp_get_type (void);
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G_END_DECLS
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#endif /* __GST_WEBRTC_DSP_H__ */
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