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audioaggregator: Improve log messages

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Make the level of log messages saner and improve some.
This commit is contained in:
Olivier Crête 2015-11-02 19:40:28 -05:00
parent 7161795a44
commit 1369924fa0
1 changed files with 10 additions and 10 deletions
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20
gst/audiomixer/gstaudioaggregator.c
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@ -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;
}