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-rw-r--r--gst/audiomixer/gstaudioaggregator.c20
1 files changed, 10 insertions, 10 deletions
diff --git a/gst/audiomixer/gstaudioaggregator.c b/gst/audiomixer/gstaudioaggregator.c
index 9155b1baa..45353d9fe 100644
--- 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;
}