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_ECHO_PROBE_H__ 24 #define __GST_WEBRTC_ECHO_PROBE_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_ECHO_PROBE (gst_webrtc_echo_probe_get_type()) 39 #define GST_WEBRTC_ECHO_PROBE(obj) (G_TYPE_CHECK_INSTANCE_CAST((obj),GST_TYPE_WEBRTC_ECHO_PROBE,GstWebrtcEchoProbe)) 40 #define GST_IS_WEBRTC_ECHO_PROBE(obj) (G_TYPE_CHECK_INSTANCE_TYPE((obj),GST_TYPE_WEBRTC_ECHO_PROBE)) 41 #define GST_WEBRTC_ECHO_PROBE_CLASS(klass) (G_TYPE_CHECK_CLASS_CAST((klass) ,GST_TYPE_WEBRTC_ECHO_PROBE,GstWebrtcEchoProbeClass)) 42 #define GST_IS_WEBRTC_ECHO_PROBE_CLASS(klass) (G_TYPE_CHECK_CLASS_TYPE((klass) ,GST_TYPE_WEBRTC_ECHO_PROBE)) 43 #define GST_WEBRTC_ECHO_PROBE_GET_CLASS(obj) (G_TYPE_INSTANCE_GET_CLASS((obj) ,GST_TYPE_WEBRTC_ECHO_PROBE,GstWebrtcEchoProbeClass)) 44 45 #define GST_WEBRTC_ECHO_PROBE_LOCK(obj) g_mutex_lock (&GST_WEBRTC_ECHO_PROBE (obj)->lock) 46 #define GST_WEBRTC_ECHO_PROBE_UNLOCK(obj) g_mutex_unlock (&GST_WEBRTC_ECHO_PROBE (obj)->lock) 47 48 typedef struct _GstWebrtcEchoProbe GstWebrtcEchoProbe; 49 typedef struct _GstWebrtcEchoProbeClass GstWebrtcEchoProbeClass; 50 51 /** 52 * GstWebrtcEchoProbe: 53 * 54 * The adder object structure. 55 */ 56 struct _GstWebrtcEchoProbe 57 { 58 GstAudioFilter parent; 59 60 /* This lock is required as the DSP may need to lock itself using it's 61 * object lock and also lock the probe. The natural order for the DSP is 62 * to lock the DSP and then the echo probe. If we where using the probe 63 * object lock, we'd be racing with GstBin which will lock sink to src, 64 * and may accidently reverse the order. */ 65 GMutex lock; 66 67 /* Protected by the lock */ 68 GstAudioInfo info; 69 guint period_size; 70 guint period_samples; 71 GstClockTime latency; 72 gint delay; 73 gboolean interleaved; 74 75 GstSegment segment; 76 GstAdapter *adapter; 77 GstPlanarAudioAdapter *padapter; 78 79 /* Private */ 80 gboolean acquired; 81 }; 82 83 struct _GstWebrtcEchoProbeClass 84 { 85 GstAudioFilterClass parent_class; 86 }; 87 88 GType gst_webrtc_echo_probe_get_type (void); 89 90 GstWebrtcEchoProbe *gst_webrtc_acquire_echo_probe (const gchar * name); 91 void gst_webrtc_release_echo_probe (GstWebrtcEchoProbe * probe); 92 gint gst_webrtc_echo_probe_read (GstWebrtcEchoProbe * self, 93 GstClockTime rec_time, gpointer frame, GstBuffer ** buf); 94 95 G_END_DECLS 96 #endif /* __GST_WEBRTC_ECHO_PROBE_H__ */ 97