webrtcdsp: Rewrite echo data synchronization

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

1 files changed, 4 insertions, 2 deletions

diff --git a/ext/webrtcdsp/gstwebrtcechoprobe.h b/ext/webrtcdsp/gstwebrtcechoprobe.h
index a18547d47..7c8a24022 100644
--- a/ext/webrtcdsp/gstwebrtcechoprobe.h
+++ b/ext/webrtcdsp/gstwebrtcechoprobe.h
@@ -62,8 +62,8 @@ struct _GstWebrtcEchoProbe
   /* Protected by the lock */
   GstAudioInfo info;
   guint period_size;
-  gint latency;
-  gboolean synchronized;
+  GstClockTime latency;
+  gint delay;
 
   GstSegment segment;
   GstAdapter *adapter;
@@ -81,6 +81,8 @@ GType gst_webrtc_echo_probe_get_type (void);
 
 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);
 
 G_END_DECLS
 #endif /* __GST_WEBRTC_ECHO_PROBE_H__ */