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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>
98 lines
3.4 KiB
C
98 lines
3.4 KiB
C
/*
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* WebRTC Audio Processing Elements
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*
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* Copyright 2016 Collabora Ltd
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* @author: Nicolas Dufresne <nicolas.dufresne@collabora.com>
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*
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* This library is free software; you can redistribute it and/or
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* modify it under the terms of the GNU Lesser General Public
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* License as published by the Free Software Foundation; either
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* version 2.1 of the License, or (at your option) any later version.
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*
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* This library is distributed in the hope that it will be useful,
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* but WITHOUT ANY WARRANTY; without even the implied warranty of
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* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
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* Lesser General Public License for more details.
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*
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* You should have received a copy of the GNU Lesser General Public
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* License along with this library; if not, write to the Free Software
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* Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
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*
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*/
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#ifndef __GST_WEBRTC_ECHO_PROBE_H__
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#define __GST_WEBRTC_ECHO_PROBE_H__
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#include <gst/gst.h>
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#include <gst/base/gstadapter.h>
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#include <gst/base/gstbasetransform.h>
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#include <gst/audio/audio.h>
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#ifndef GST_USE_UNSTABLE_API
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#define GST_USE_UNSTABLE_API
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#endif
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#include <gst/audio/gstplanaraudioadapter.h>
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G_BEGIN_DECLS
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#define GST_TYPE_WEBRTC_ECHO_PROBE (gst_webrtc_echo_probe_get_type())
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#define GST_WEBRTC_ECHO_PROBE(obj) (G_TYPE_CHECK_INSTANCE_CAST((obj),GST_TYPE_WEBRTC_ECHO_PROBE,GstWebrtcEchoProbe))
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#define GST_IS_WEBRTC_ECHO_PROBE(obj) (G_TYPE_CHECK_INSTANCE_TYPE((obj),GST_TYPE_WEBRTC_ECHO_PROBE))
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#define GST_WEBRTC_ECHO_PROBE_CLASS(klass) (G_TYPE_CHECK_CLASS_CAST((klass) ,GST_TYPE_WEBRTC_ECHO_PROBE,GstWebrtcEchoProbeClass))
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#define GST_IS_WEBRTC_ECHO_PROBE_CLASS(klass) (G_TYPE_CHECK_CLASS_TYPE((klass) ,GST_TYPE_WEBRTC_ECHO_PROBE))
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#define GST_WEBRTC_ECHO_PROBE_GET_CLASS(obj) (G_TYPE_INSTANCE_GET_CLASS((obj) ,GST_TYPE_WEBRTC_ECHO_PROBE,GstWebrtcEchoProbeClass))
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#define GST_WEBRTC_ECHO_PROBE_LOCK(obj) g_mutex_lock (&GST_WEBRTC_ECHO_PROBE (obj)->lock)
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#define GST_WEBRTC_ECHO_PROBE_UNLOCK(obj) g_mutex_unlock (&GST_WEBRTC_ECHO_PROBE (obj)->lock)
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typedef struct _GstWebrtcEchoProbe GstWebrtcEchoProbe;
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typedef struct _GstWebrtcEchoProbeClass GstWebrtcEchoProbeClass;
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/**
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* GstWebrtcEchoProbe:
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*
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* The adder object structure.
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*/
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struct _GstWebrtcEchoProbe
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{
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GstAudioFilter parent;
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/* This lock is required as the DSP may need to lock itself using it's
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* object lock and also lock the probe. The natural order for the DSP is
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* to lock the DSP and then the echo probe. If we where using the probe
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* object lock, we'd be racing with GstBin which will lock sink to src,
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* and may accidentally reverse the order. */
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GMutex lock;
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/* Protected by the lock */
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GstAudioInfo info;
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guint period_size;
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guint period_samples;
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GstClockTime latency;
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gint delay;
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gboolean interleaved;
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GstSegment segment;
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GstAdapter *adapter;
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GstPlanarAudioAdapter *padapter;
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/* Private */
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gboolean acquired;
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};
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struct _GstWebrtcEchoProbeClass
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{
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GstAudioFilterClass parent_class;
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};
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GType gst_webrtc_echo_probe_get_type (void);
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GST_ELEMENT_REGISTER_DECLARE (webrtcechoprobe);
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GstWebrtcEchoProbe *gst_webrtc_acquire_echo_probe (const gchar * name);
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void gst_webrtc_release_echo_probe (GstWebrtcEchoProbe * probe);
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gint gst_webrtc_echo_probe_read (GstWebrtcEchoProbe * self,
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GstClockTime rec_time, gpointer frame, GstBuffer ** buf);
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G_END_DECLS
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#endif /* __GST_WEBRTC_ECHO_PROBE_H__ */
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