Commit 631d3568 authored by Sebastian Dröge's avatar Sebastian Dröge

audiotestsrc: Report our latency properly in live mode

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
parent e9c6c833
......@@ -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;
......
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