From: Nicolas Dufresne <nicolas.dufresne@collabora.com>
Date: Thu, 23 Jun 2016 01:54:13 +0000 (-0400)
Subject: webrtcdsp: Synchronize with delays
X-Git-Tag: 1.10.4~560
X-Git-Url: http://review.tizen.org/git/?a=commitdiff_plain;h=86aa3b5f9c7016e749e4f389adc6c5a29108b6f3;p=platform%2Fupstream%2Fgst-plugins-bad.git

webrtcdsp: Synchronize with delays

Until now, we were synchronizing both DSP and Probe adapter by
waiting and clipping the probe adapter data. This increases the CPU
usage, can cause copies if the audio is not 10ms aligned and the worst
is that it prevents the processing from compensating for inaccurate
latency. This is also a step forward toward supporting playback
filters.
---

diff --git a/ext/webrtcdsp/gstwebrtcdsp.cpp b/ext/webrtcdsp/gstwebrtcdsp.cpp
index f17947b..6b6239d 100644
--- a/ext/webrtcdsp/gstwebrtcdsp.cpp
+++ b/ext/webrtcdsp/gstwebrtcdsp.cpp
@@ -176,6 +176,7 @@ struct _GstWebrtcDsp
   GstClockTime timestamp;
   GstAdapter *adapter;
   webrtc::AudioProcessing * apm;
+  gint delay_ms;
 
   /* Protected by the object lock */
   gchar *probe_name;
@@ -288,28 +289,11 @@ gst_webrtc_dsp_sync_reverse_stream (GstWebrtcDsp * self,
   probe_timestamp += gst_util_uint64_scale_int (distance / probe->info.bpf,
       GST_SECOND, probe->info.rate);
 
-  probe_timestamp += probe->latency;
-
   diff = GST_CLOCK_DIFF (probe_timestamp, self->timestamp);
-  if (diff < 0) {
-    GST_TRACE_OBJECT (self,
-        "Echo cancellation will start in in %" GST_TIME_FORMAT,
-        GST_TIME_ARGS (-diff));
-    return FALSE;
-  }
-
-  distance = gst_util_uint64_scale_int ((guint64) diff,
-      probe->info.rate * probe->info.bpf, GST_SECOND);
-
-  if (gst_adapter_available (probe->adapter) < distance) {
-    GST_TRACE_OBJECT (self, "Not enough data to synchronize for now.");
-    return FALSE;
-  }
-
-  gst_adapter_flush (probe->adapter, (gsize) distance);
-  probe->synchronized = TRUE;
+  self->delay_ms = (probe->latency - diff) / GST_MSECOND;
 
   GST_DEBUG_OBJECT (probe, "Echo Probe is now synchronized");
+  probe->synchronized = TRUE;
 
   return TRUE;
 }
@@ -387,8 +371,7 @@ gst_webrtc_dsp_process_stream (GstWebrtcDsp * self)
 
   memcpy (frame.data_, info.data, self->period_size);
 
-  /* We synchronize in GStreamer */
-  apm->set_stream_delay_ms (5);
+  apm->set_stream_delay_ms (self->delay_ms);
 
   if ((err = apm->ProcessStream (&frame)) < 0) {
     GST_WARNING_OBJECT (self, "Failed to filter the audio: %s.",
@@ -505,6 +488,7 @@ gst_webrtc_dsp_setup (GstAudioFilter * filter, const GstAudioInfo * info)
   GST_OBJECT_LOCK (self);
 
   gst_adapter_clear (self->adapter);
+  self->delay_ms = 0;
   self->info = *info;
   apm = self->apm;