/* * 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); G_END_DECLS #endif /* __GST_WEBRTC_DSP_H__ */