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audiomixer: Remove weird and wrong segment handling

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There's no reason why audiomixer should override the segment
base of upstream with whatever value it got from a SEEK event,
or even worse... with 0 if there was no SEEK event yet. This
broke synchronization if upstream provided a segment base other
than 0, e.g. when using pad offsets.
Also that this code did things conditional on the element's state
should've been a big warning already that something is just wrong.
If this breaks anything else now, let's fix it properly :)

Also don't do fancy segment position trickery when receiving a
segment event. It's just not correct.
This commit is contained in:
Sebastian Dröge 2015-02-05 16:02:54 +01:00
parent b1eef4f436
commit 6d6c693254
1 changed files with 0 additions and 34 deletions
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34
gst/audiomixer/gstaudiomixer.c
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@ -639,9 +639,6 @@ gst_audiomixer_src_event (GstAggregator * agg, GstEvent * event)
/* Link up */ /* Link up */
result = GST_AGGREGATOR_CLASS (parent_class)->src_event (agg, event); result = GST_AGGREGATOR_CLASS (parent_class)->src_event (agg, event);
if (result)
audiomixer->base_time = agg->segment.start;
goto done; goto done;
} }
break; break;
@ -693,17 +690,6 @@ gst_audiomixer_sink_event (GstAggregator * agg, GstAggregatorPad * aggpad,
gst_event_unref (event); gst_event_unref (event);
event = NULL; event = NULL;
} }
if (event) {
res =
GST_AGGREGATOR_CLASS (parent_class)->sink_event (agg, aggpad,
event);
if (res)
aggpad->segment.position = segment->start + segment->offset;
event = NULL;
}
break; break;
} }
default: default:
@ -761,23 +747,6 @@ gst_audiomixer_flush (GstAggregator * agg)
return GST_FLOW_OK; return GST_FLOW_OK;
} }
static gboolean
gst_audiomixer_send_event (GstElement * element, GstEvent * event)
{
GstAudioMixer *audiomixer = GST_AUDIO_MIXER (element);
gboolean res = GST_ELEMENT_CLASS (parent_class)->send_event (element, event);
GST_STATE_LOCK (element);
if (GST_EVENT_TYPE (event) == GST_EVENT_SEEK &&
GST_STATE (element) < GST_STATE_PAUSED) {
audiomixer->base_time = GST_AGGREGATOR (element)->segment.start;
}
GST_STATE_UNLOCK (element);
return res;
}
static void static void
gst_audiomixer_class_init (GstAudioMixerClass * klass) gst_audiomixer_class_init (GstAudioMixerClass * klass)
{ {
@ -827,8 +796,6 @@ gst_audiomixer_class_init (GstAudioMixerClass * klass)
GST_DEBUG_FUNCPTR (gst_audiomixer_request_new_pad); GST_DEBUG_FUNCPTR (gst_audiomixer_request_new_pad);
gstelement_class->release_pad = gstelement_class->release_pad =
GST_DEBUG_FUNCPTR (gst_audiomixer_release_pad); GST_DEBUG_FUNCPTR (gst_audiomixer_release_pad);
gstelement_class->send_event = GST_DEBUG_FUNCPTR (gst_audiomixer_send_event);
agg_class->sinkpads_type = GST_TYPE_AUDIO_MIXER_PAD; agg_class->sinkpads_type = GST_TYPE_AUDIO_MIXER_PAD;
agg_class->start = gst_audiomixer_start; agg_class->start = gst_audiomixer_start;
@ -1093,7 +1060,6 @@ gst_audio_mixer_fill_buffer (GstAudioMixer * audiomixer, GstAudioMixerPad * pad,
guint64 start_running_time_offset; guint64 start_running_time_offset;
guint64 end_running_time_offset; guint64 end_running_time_offset;
aggpad->segment.base = audiomixer->base_time;
start_running_time = start_running_time =
gst_segment_to_running_time (&aggpad->segment, gst_segment_to_running_time (&aggpad->segment,
GST_FORMAT_TIME, start_time); GST_FORMAT_TIME, start_time);