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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_ECHO_PROBE_H__
#define __GST_WEBRTC_ECHO_PROBE_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_ECHO_PROBE (gst_webrtc_echo_probe_get_type())
#define GST_WEBRTC_ECHO_PROBE(obj) (G_TYPE_CHECK_INSTANCE_CAST((obj),GST_TYPE_WEBRTC_ECHO_PROBE,GstWebrtcEchoProbe))
#define GST_IS_WEBRTC_ECHO_PROBE(obj) (G_TYPE_CHECK_INSTANCE_TYPE((obj),GST_TYPE_WEBRTC_ECHO_PROBE))
#define GST_WEBRTC_ECHO_PROBE_CLASS(klass) (G_TYPE_CHECK_CLASS_CAST((klass) ,GST_TYPE_WEBRTC_ECHO_PROBE,GstWebrtcEchoProbeClass))
#define GST_IS_WEBRTC_ECHO_PROBE_CLASS(klass) (G_TYPE_CHECK_CLASS_TYPE((klass) ,GST_TYPE_WEBRTC_ECHO_PROBE))
#define GST_WEBRTC_ECHO_PROBE_GET_CLASS(obj) (G_TYPE_INSTANCE_GET_CLASS((obj) ,GST_TYPE_WEBRTC_ECHO_PROBE,GstWebrtcEchoProbeClass))
#define GST_WEBRTC_ECHO_PROBE_LOCK(obj) g_mutex_lock (&GST_WEBRTC_ECHO_PROBE (obj)->lock)
#define GST_WEBRTC_ECHO_PROBE_UNLOCK(obj) g_mutex_unlock (&GST_WEBRTC_ECHO_PROBE (obj)->lock)
typedef struct _GstWebrtcEchoProbe GstWebrtcEchoProbe;
typedef struct _GstWebrtcEchoProbeClass GstWebrtcEchoProbeClass;
/**
* GstWebrtcEchoProbe:
*
* The adder object structure.
*/
struct _GstWebrtcEchoProbe
{
GstAudioFilter parent;
/* This lock is required as the DSP may need to lock itself using it's
* object lock and also lock the probe. The natural order for the DSP is
* to lock the DSP and then the echo probe. If we where using the probe
* object lock, we'd be racing with GstBin which will lock sink to src,
* and may accidentally reverse the order. */
GMutex lock;
/* Protected by the lock */
GstAudioInfo info;
guint period_size;
guint period_samples;
GstClockTime latency;
gint delay;
gboolean interleaved;
GstSegment segment;
GstAdapter *adapter;
GstPlanarAudioAdapter *padapter;
/* Private */
gboolean acquired;
};
struct _GstWebrtcEchoProbeClass
{
GstAudioFilterClass parent_class;
};
GType gst_webrtc_echo_probe_get_type (void);
GST_ELEMENT_REGISTER_DECLARE (webrtcechoprobe);
GstWebrtcEchoProbe *gst_webrtc_acquire_echo_probe (const gchar * name);
void gst_webrtc_release_echo_probe (GstWebrtcEchoProbe * probe);
gint gst_webrtc_echo_probe_read (GstWebrtcEchoProbe * self,
GstClockTime rec_time, gpointer frame, GstBuffer ** buf);
G_END_DECLS
#endif /* __GST_WEBRTC_ECHO_PROBE_H__ */
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