webrtcdsp: Offset timestamp with duration

The saved timestamp is used to compute the delay of the probe data.
As it's used at the following incoming buffer, it needs to be offset
with the duration of the buffer to represent the end position. Also,
properly initialize the saved timestamp and protect against TIME_NONE.

diff --git a/ext/webrtcdsp/gstwebrtcdsp.cpp b/ext/webrtcdsp/gstwebrtcdsp.cpp
index 6b6239d..2f87f58 100644 (file)
--- a/ext/webrtcdsp/gstwebrtcdsp.cpp
+++ b/ext/webrtcdsp/gstwebrtcdsp.cpp
@@ -256,6 +256,10 @@ gst_webrtc_dsp_sync_reverse_stream (GstWebrtcDsp * self,
   GstClockTimeDiff diff;
   guint64 distance;
 
+  /* We need to wait for a time reference */
+  if (!GST_CLOCK_TIME_IS_VALID (self->timestamp))
+      return FALSE;
+
   probe_timestamp = gst_adapter_prev_pts (probe->adapter, &distance);
 
   if (!GST_CLOCK_TIME_IS_VALID (probe_timestamp)) {
@@ -359,8 +363,10 @@ gst_webrtc_dsp_process_stream (GstWebrtcDsp * self)
   frame.samples_per_channel_ = self->period_size / self->info.bpf;
 
   timestamp = gst_adapter_prev_pts (self->adapter, &distance);
-  timestamp += gst_util_uint64_scale_int (distance / self->info.bpf,
-      GST_SECOND, self->info.rate);
+
+  if (GST_CLOCK_TIME_IS_VALID (timestamp))
+      timestamp += gst_util_uint64_scale_int (distance / self->info.bpf,
+              GST_SECOND, self->info.rate);
 
   buffer = gst_adapter_take_buffer (self->adapter, self->period_size);
 
@@ -390,7 +396,8 @@ gst_webrtc_dsp_process_stream (GstWebrtcDsp * self)
   else
     GST_BUFFER_FLAG_UNSET (buffer, GST_BUFFER_FLAG_DISCONT);
 
-  self->timestamp = timestamp;
+  if (GST_CLOCK_TIME_IS_VALID (timestamp))
+      self->timestamp = timestamp + GST_BUFFER_DURATION (buffer);
 
   return buffer;
 }
@@ -488,6 +495,7 @@ gst_webrtc_dsp_setup (GstAudioFilter * filter, const GstAudioInfo * info)
   GST_OBJECT_LOCK (self);
 
   gst_adapter_clear (self->adapter);
+  self->timestamp = GST_CLOCK_TIME_NONE;
   self->delay_ms = 0;
   self->info = *info;
   apm = self->apm;