audiomixer: Implement get_next_time()
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@ -235,6 +235,15 @@ gst_audiomixer_do_clip (GstAggregator * agg,
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GstAggregatorPad * bpad, GstBuffer * buffer, GstBuffer ** outbuf);
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GstAggregatorPad * bpad, GstBuffer * buffer, GstBuffer ** outbuf);
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static GstFlowReturn gst_audiomixer_aggregate (GstAggregator * agg);
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static GstFlowReturn gst_audiomixer_aggregate (GstAggregator * agg);
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static GstClockTime
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gst_audiomixer_get_next_time (GstAggregator * agg)
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{
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if (agg->segment.position == -1)
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return agg->segment.start;
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else
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return agg->segment.position;
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}
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/* we can only accept caps that we and downstream can handle.
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/* we can only accept caps that we and downstream can handle.
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* if we have filtercaps set, use those to constrain the target caps.
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* if we have filtercaps set, use those to constrain the target caps.
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*/
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*/
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@ -815,6 +824,8 @@ gst_audiomixer_class_init (GstAudioMixerClass * klass)
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agg_class->start = gst_audiomixer_start;
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agg_class->start = gst_audiomixer_start;
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agg_class->stop = gst_audiomixer_stop;
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agg_class->stop = gst_audiomixer_stop;
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agg_class->get_next_time = gst_audiomixer_get_next_time;
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agg_class->sink_query = GST_DEBUG_FUNCPTR (gst_audiomixer_sink_query);
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agg_class->sink_query = GST_DEBUG_FUNCPTR (gst_audiomixer_sink_query);
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agg_class->sink_event = GST_DEBUG_FUNCPTR (gst_audiomixer_sink_event);
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agg_class->sink_event = GST_DEBUG_FUNCPTR (gst_audiomixer_sink_event);
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