mirrors/gstreamer
mirrors/gstreamer
1
1
Fork 0
mirror of https://gitlab.freedesktop.org/gstreamer/gstreamer.git synced 2024-11-27 12:11:13 +00:00
Code Issues 1 Projects Releases Wiki Activity

audioaggregator: Fix build error

Browse source 
Build error due to wrong argument type in debug message
aagg->priv->offset and next_offset are of type int64, but uint64
formatter is being used in logs. Changing all those to int64

https://bugzilla.gnome.org/show_bug.cgi?id=756065
This commit is contained in:
Vineeth TM 2015-10-07 08:48:15 +09:00 committed by Sebastian Dröge
parent 52ea2667c6
commit 11ab8b7965
1 changed files with 8 additions and 7 deletions
Show stats Download patch file Download diff file Expand all files Collapse all files

15
gst/audiomixer/gstaudioaggregator.c
View file

@ -892,7 +892,7 @@ gst_audio_aggregator_fill_buffer (GstAudioAggregator * aagg,
pad->priv->output_offset = -1;
GST_DEBUG_OBJECT (pad,
"Buffer before segment or current position: %" G_GUINT64_FORMAT " < %"
G_GUINT64_FORMAT, end_output_offset, aagg->priv->offset);
G_GINT64_FORMAT, end_output_offset, aagg->priv->offset);
return FALSE;
}
@ -922,7 +922,7 @@ gst_audio_aggregator_fill_buffer (GstAudioAggregator * aagg,
pad->priv->output_offset = -1;
GST_DEBUG_OBJECT (pad,
"Buffer before segment or current position: %" G_GUINT64_FORMAT
" < %" G_GUINT64_FORMAT, end_output_offset, aagg->priv->offset);
" < %" G_GINT64_FORMAT, end_output_offset, aagg->priv->offset);
return FALSE;
}
}
@ -934,7 +934,7 @@ gst_audio_aggregator_fill_buffer (GstAudioAggregator * aagg,
GST_DEBUG_OBJECT (pad,
"Buffer resynced: Pad offset %" G_GUINT64_FORMAT
", current audio aggregator offset %" G_GUINT64_FORMAT,
", current audio aggregator offset %" G_GINT64_FORMAT,
pad->priv->output_offset, aagg->priv->offset);
}
@ -1139,7 +1139,8 @@ gst_audio_aggregator_aggregate (GstAggregator * agg, gboolean timeout)
aagg->priv->offset =
gst_util_uint64_scale (agg->segment.position - agg->segment.start, rate,
GST_SECOND);
GST_DEBUG_OBJECT (aagg, "Starting at offset %lu", aagg->priv->offset);
GST_DEBUG_OBJECT (aagg, "Starting at offset %" G_GINT64_FORMAT,
aagg->priv->offset);
}
blocksize = gst_util_uint64_scale (aagg->priv->output_buffer_duration,
@ -1172,7 +1173,7 @@ gst_audio_aggregator_aggregate (GstAggregator * agg, gboolean timeout)
outbuf = aagg->priv->current_buffer;
GST_LOG_OBJECT (agg,
"Starting to mix %u samples for offset %" G_GUINT64_FORMAT
"Starting to mix %u samples for offset %" G_GINT64_FORMAT
" with timestamp %" GST_TIME_FORMAT, blocksize,
aagg->priv->offset, GST_TIME_ARGS (agg->segment.position));
@ -1263,7 +1264,7 @@ gst_audio_aggregator_aggregate (GstAggregator * agg, gboolean timeout)
if (pad->priv->output_offset >= next_offset) {
GST_DEBUG_OBJECT (pad,
"Pad is after current offset: %" G_GUINT64_FORMAT " >= %"
G_GUINT64_FORMAT, pad->priv->output_offset, next_offset);
G_GINT64_FORMAT, pad->priv->output_offset, next_offset);
} else {
is_done = FALSE;
}
@ -1314,7 +1315,7 @@ gst_audio_aggregator_aggregate (GstAggregator * agg, gboolean timeout)
if (max_offset <= next_offset) {
GST_DEBUG_OBJECT (aagg,
"Last buffer is incomplete: %" G_GUINT64_FORMAT " <= %"
G_GUINT64_FORMAT, max_offset, next_offset);
G_GINT64_FORMAT, max_offset, next_offset);
next_offset = max_offset;
next_timestamp =
agg->segment.start + gst_util_uint64_scale (next_offset, GST_SECOND,