element = GST_ELEMENT (agg);
aagg = GST_AUDIO_AGGREGATOR (agg);
- blocksize = gst_util_uint64_scale (aagg->priv->output_buffer_duration,
- GST_AUDIO_INFO_RATE (&aagg->info), GST_SECOND);
- blocksize = MAX (1, blocksize);
-
GST_AUDIO_AGGREGATOR_LOCK (aagg);
GST_OBJECT_LOCK (agg);
rate = GST_AUDIO_INFO_RATE (&aagg->info);
bpf = GST_AUDIO_INFO_BPF (&aagg->info);
+ blocksize = gst_util_uint64_scale (aagg->priv->output_buffer_duration,
+ GST_AUDIO_INFO_RATE (&aagg->info), GST_SECOND);
+ blocksize = MAX (1, blocksize);
/* for the next timestamp, use the sample counter, which will
* never accumulate rounding errors */