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audiotestsrc: Report our latency properly in live mode

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While we have no latency at all in theory, any other live source has the
duration of one buffer as minimum latency. Do the same in audiotestsrc.

https://bugzilla.gnome.org/show_bug.cgi?id=741879
This commit is contained in:
Sebastian Dröge 2014-12-22 22:00:26 +01:00
parent e9c6c833c9
commit 631d356845
1 changed files with 17 additions and 0 deletions
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17
gst/audiotestsrc/gstaudiotestsrc.c
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@ -341,6 +341,23 @@ gst_audio_test_src_query (GstBaseSrc * basesrc, GstQuery * query)
res = TRUE;
break;
}
case GST_QUERY_LATENCY:
{
if (src->info.rate > 0) {
GstClockTime latency;
latency =
gst_util_uint64_scale (src->generate_samples_per_buffer, GST_SECOND,
src->info.rate);
gst_query_set_latency (query,
gst_base_src_is_live (GST_BASE_SRC_CAST (src)), latency,
GST_CLOCK_TIME_NONE);
GST_DEBUG_OBJECT (src, "Reporting latency of %" GST_TIME_FORMAT,
GST_TIME_ARGS (latency));
res = TRUE;
}
break;
}
default:
res = GST_BASE_SRC_CLASS (parent_class)->query (basesrc, query);
break;