audioaggregator: Improve log messages

Make the level of log messages saner and improve some.

diff --git a/gst/audiomixer/gstaudioaggregator.c b/gst/audiomixer/gstaudioaggregator.c
index 9155b1b..45353d9 100644 (file)
--- a/gst/audiomixer/gstaudioaggregator.c
+++ b/gst/audiomixer/gstaudioaggregator.c
@@ -829,7 +829,7 @@ gst_audio_aggregator_fill_buffer (GstAudioAggregator * aagg,
   if (discont) {
     /* Have discont, need resync */
     if (pad->priv->next_offset != -1)
-      GST_INFO_OBJECT (pad, "Have discont. Expected %"
+      GST_DEBUG_OBJECT (pad, "Have discont. Expected %"
           G_GUINT64_FORMAT ", got %" G_GUINT64_FORMAT,
           pad->priv->next_offset, start_offset);
     pad->priv->output_offset = -1;
@@ -995,7 +995,7 @@ gst_audio_aggregator_mix_buffer (GstAudioAggregator * aagg,
   if (pad->priv->position == pad->priv->size) {
     /* Buffer done, drop it */
     gst_buffer_replace (&pad->priv->buffer, NULL);
-    GST_DEBUG_OBJECT (pad, "Finished mixing buffer, waiting for next");
+    GST_LOG_OBJECT (pad, "Finished mixing buffer, waiting for next");
     return FALSE;
   }
 
@@ -1194,8 +1194,8 @@ gst_audio_aggregator_aggregate (GstAggregator * agg, gboolean timeout)
       if (timeout) {
         if (pad->priv->output_offset < next_offset) {
           gint64 diff = next_offset - pad->priv->output_offset;
-          GST_LOG_OBJECT (pad, "Timeout, missing %" G_GINT64_FORMAT " frames (%"
-              GST_TIME_FORMAT ")", diff,
+          GST_DEBUG_OBJECT (pad, "Timeout, missing %" G_GINT64_FORMAT
+              " frames (%" GST_TIME_FORMAT ")", diff,
               GST_TIME_ARGS (gst_util_uint64_scale (diff, GST_SECOND,
                       GST_AUDIO_INFO_RATE (&aagg->info))));
         }
@@ -1241,7 +1241,7 @@ gst_audio_aggregator_aggregate (GstAggregator * agg, gboolean timeout)
       pad->priv->output_offset += diff;
 
       if (pad->priv->position == pad->priv->size) {
-        GST_LOG_OBJECT (pad, "Buffer was late by %" GST_TIME_FORMAT
+        GST_DEBUG_OBJECT (pad, "Buffer was late by %" GST_TIME_FORMAT
             ", dropping %" GST_PTR_FORMAT,
             GST_TIME_ARGS (gst_util_uint64_scale (odiff, GST_SECOND,
                     GST_AUDIO_INFO_RATE (&aagg->info))), pad->priv->buffer);
@@ -1262,8 +1262,8 @@ gst_audio_aggregator_aggregate (GstAggregator * agg, gboolean timeout)
       drop_buf = !gst_audio_aggregator_mix_buffer (aagg, pad, pad->priv->buffer,
           outbuf);
       if (pad->priv->output_offset >= next_offset) {
-        GST_DEBUG_OBJECT (pad,
-            "Pad is after current offset: %" G_GUINT64_FORMAT " >= %"
+        GST_LOG_OBJECT (pad,
+            "Pad is at or after current offset: %" G_GUINT64_FORMAT " >= %"
             G_GINT64_FORMAT, pad->priv->output_offset, next_offset);
       } else {
         is_done = FALSE;
@@ -1279,15 +1279,15 @@ gst_audio_aggregator_aggregate (GstAggregator * agg, gboolean timeout)
 
   if (dropped) {
     /* We dropped a buffer, retry */
-    GST_INFO_OBJECT (aagg, "A pad dropped a buffer, wait for the next one");
+    GST_LOG_OBJECT (aagg, "A pad dropped a buffer, wait for the next one");
     GST_AUDIO_AGGREGATOR_UNLOCK (aagg);
     return GST_FLOW_OK;
   }
 
   if (!is_done && !is_eos) {
     /* Get more buffers */
-    GST_INFO_OBJECT (aagg,
-        "We're not done yet for the current offset," " waiting for more data");
+    GST_LOG_OBJECT (aagg,
+        "We're not done yet for the current offset, waiting for more data");
     GST_AUDIO_AGGREGATOR_UNLOCK (aagg);
     return GST_FLOW_OK;
   }