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71c9cdeff4
The previous code would run out of sync if there was packet lost or clock skews. When that happened, the echo cancellation feature would completely stop working. As this is crucial for audio calls, this patch re-implement synchronization completely. Instead of letting it drift until next discont, we now synchronize against the record data at every iteration. This way we simply never let the stream drift for longer then 10ms period. We also shorter the delay by using the latency up the probe (basically excluding the sink latency. This is a decent delay to avoid starving in the probe queue. https://bugzilla.gnome.org/show_bug.cgi?id=768009
88 lines
3.1 KiB
C
88 lines
3.1 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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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 accidently 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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GstClockTime latency;
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gint delay;
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GstSegment segment;
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GstAdapter *adapter;
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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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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);
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G_END_DECLS
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#endif /* __GST_WEBRTC_ECHO_PROBE_H__ */
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