mirror of
https://gitlab.freedesktop.org/gstreamer/gstreamer.git
synced 2024-11-08 18:39:54 +00:00
63dc81d000
Split plugin into features including dynamic types which can be indiviually registered during a static build. More details here: https://gitlab.freedesktop.org/gstreamer/gst-build/-/merge_requests/199 https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/661 Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/2038>
59 lines
2 KiB
C
59 lines
2 KiB
C
/*
|
|
* 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__ */
|