1 /*
2  * WebRTC Audio Processing Elements
3  *
4  *  Copyright 2016 Collabora Ltd
5  *    @author: Nicolas Dufresne <nicolas.dufresne@collabora.com>
6  *
7  * This library is free software; you can redistribute it and/or
8  * modify it under the terms of the GNU Lesser General Public
9  * License as published by the Free Software Foundation; either
10  * version 2.1 of the License, or (at your option) any later version.
11  *
12  * This library is distributed in the hope that it will be useful,
13  * but WITHOUT ANY WARRANTY; without even the implied warranty of
14  * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the GNU
15  * Lesser General Public License for more details.
16  *
17  * You should have received a copy of the GNU Lesser General Public
18  * License along with this library; if not, write to the Free Software
19  * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA  02110-1301 USA
20  *
21  */
22 
23 #ifndef __GST_WEBRTC_DSP_H__
24 #define __GST_WEBRTC_DSP_H__
25 
26 #include <gst/gst.h>
27 #include <gst/base/gstadapter.h>
28 #include <gst/base/gstbasetransform.h>
29 #include <gst/audio/audio.h>
30 
31 #ifndef GST_USE_UNSTABLE_API
32 #define GST_USE_UNSTABLE_API
33 #endif
34 #include <gst/audio/gstplanaraudioadapter.h>
35 
36 G_BEGIN_DECLS
37 
38 #define GST_TYPE_WEBRTC_DSP            (gst_webrtc_dsp_get_type())
39 #define GST_WEBRTC_DSP(obj)            (G_TYPE_CHECK_INSTANCE_CAST((obj),GST_TYPE_WEBRTC_DSP,GstWebrtcDsp))
40 #define GST_IS_WEBRTC_DSP(obj)         (G_TYPE_CHECK_INSTANCE_TYPE((obj),GST_TYPE_WEBRTC_DSP))
41 #define GST_WEBRTC_DSP_CLASS(klass)    (G_TYPE_CHECK_CLASS_CAST((klass) ,GST_TYPE_WEBRTC_DSP,GstWebrtcDspClass))
42 #define GST_IS_WEBRTC_DSP_CLASS(klass) (G_TYPE_CHECK_CLASS_TYPE((klass) ,GST_TYPE_WEBRTC_DSP))
43 #define GST_WEBRTC_DSP_GET_CLASS(obj)  (G_TYPE_INSTANCE_GET_CLASS((obj) ,GST_TYPE_WEBRTC_DSP,GstWebrtcDspClass))
44 
45 typedef struct _GstWebrtcDsp GstWebrtcDsp;
46 typedef struct _GstWebrtcDspClass GstWebrtcDspClass;
47 
48 struct _GstWebrtcDspClass
49 {
50   GstAudioFilterClass parent_class;
51 };
52 
53 GType gst_webrtc_dsp_get_type (void);
54 
55 G_END_DECLS
56 
57 #endif /* __GST_WEBRTC_DSP_H__ */
58