gstreamer/subprojects/gst-plugins-bad/ext/webrtcdsp/gstwebrtcdsp.h

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/*
* WebRTC Audio Processing Elements
*
* Copyright 2016 Collabora Ltd
* @author: Nicolas Dufresne <nicolas.dufresne@collabora.com>
*
* This library is free software; you can redistribute it and/or
* modify it under the terms of the GNU Lesser General Public
* License as published by the Free Software Foundation; either
* version 2.1 of the License, or (at your option) any later version.
*
* This library is distributed in the hope that it will be useful,
* but WITHOUT ANY WARRANTY; without even the implied warranty of
* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
* Lesser General Public License for more details.
*
* You should have received a copy of the GNU Lesser General Public
* License along with this library; if not, write to the Free Software
* Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
*
*/
#ifndef __GST_WEBRTC_DSP_H__
#define __GST_WEBRTC_DSP_H__
#include <gst/gst.h>
#include <gst/base/gstadapter.h>
#include <gst/base/gstbasetransform.h>
#include <gst/audio/audio.h>
#ifndef GST_USE_UNSTABLE_API
#define GST_USE_UNSTABLE_API
#endif
#include <gst/audio/gstplanaraudioadapter.h>
G_BEGIN_DECLS
#define GST_TYPE_WEBRTC_DSP (gst_webrtc_dsp_get_type())
#define GST_WEBRTC_DSP(obj) (G_TYPE_CHECK_INSTANCE_CAST((obj),GST_TYPE_WEBRTC_DSP,GstWebrtcDsp))
#define GST_IS_WEBRTC_DSP(obj) (G_TYPE_CHECK_INSTANCE_TYPE((obj),GST_TYPE_WEBRTC_DSP))
#define GST_WEBRTC_DSP_CLASS(klass) (G_TYPE_CHECK_CLASS_CAST((klass) ,GST_TYPE_WEBRTC_DSP,GstWebrtcDspClass))
#define GST_IS_WEBRTC_DSP_CLASS(klass) (G_TYPE_CHECK_CLASS_TYPE((klass) ,GST_TYPE_WEBRTC_DSP))
#define GST_WEBRTC_DSP_GET_CLASS(obj) (G_TYPE_INSTANCE_GET_CLASS((obj) ,GST_TYPE_WEBRTC_DSP,GstWebrtcDspClass))
typedef struct _GstWebrtcDsp GstWebrtcDsp;
typedef struct _GstWebrtcDspClass GstWebrtcDspClass;
struct _GstWebrtcDspClass
{
GstAudioFilterClass parent_class;
};
GType gst_webrtc_dsp_get_type (void);
GST_ELEMENT_REGISTER_DECLARE (webrtcdsp);
G_END_DECLS
#endif /* __GST_WEBRTC_DSP_H__ */