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audiomixer: Implement get_next_time()

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Sebastian Dröge 2014-12-16 17:37:12 +01:00
parent d5a171cae9
commit cf90f534f1
1 changed files with 11 additions and 0 deletions
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11
gst/audiomixer/gstaudiomixer.c
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@ -235,6 +235,15 @@ gst_audiomixer_do_clip (GstAggregator * agg,
GstAggregatorPad * bpad, GstBuffer * buffer, GstBuffer ** outbuf); GstAggregatorPad * bpad, GstBuffer * buffer, GstBuffer ** outbuf);
static GstFlowReturn gst_audiomixer_aggregate (GstAggregator * agg); static GstFlowReturn gst_audiomixer_aggregate (GstAggregator * agg);
static GstClockTime
gst_audiomixer_get_next_time (GstAggregator * agg)
{
if (agg->segment.position == -1)
return agg->segment.start;
else
return agg->segment.position;
}
/* we can only accept caps that we and downstream can handle. /* we can only accept caps that we and downstream can handle.
* if we have filtercaps set, use those to constrain the target caps. * if we have filtercaps set, use those to constrain the target caps.
*/ */
@ -815,6 +824,8 @@ gst_audiomixer_class_init (GstAudioMixerClass * klass)
agg_class->start = gst_audiomixer_start; agg_class->start = gst_audiomixer_start;
agg_class->stop = gst_audiomixer_stop; agg_class->stop = gst_audiomixer_stop;
agg_class->get_next_time = gst_audiomixer_get_next_time;
agg_class->sink_query = GST_DEBUG_FUNCPTR (gst_audiomixer_sink_query); agg_class->sink_query = GST_DEBUG_FUNCPTR (gst_audiomixer_sink_query);
agg_class->sink_event = GST_DEBUG_FUNCPTR (gst_audiomixer_sink_event); agg_class->sink_event = GST_DEBUG_FUNCPTR (gst_audiomixer_sink_event);