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audioaggregator: Don't overwrite already written samples

Browse source 
On re-sync, don't forget what has already been written. Instead, just
drop any samples that overlap with parts that were already filled.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-base/-/merge_requests/1180>
This commit is contained in:
Olivier Crête 2021-05-27 16:05:04 -04:00
parent 5ad59ce725
commit fd73fd05ca
1 changed files with 11 additions and 6 deletions
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17
gst-libs/gst/audio/gstaudioaggregator.c
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@ -1674,13 +1674,12 @@ gst_audio_aggregator_fill_buffer (GstAudioAggregator * aagg,
GST_DEBUG_OBJECT (pad, "Have discont. Expected %"
G_GUINT64_FORMAT ", got %" G_GUINT64_FORMAT,
pad->priv->next_offset, start_offset);
pad->priv->output_offset = -1;
pad->priv->next_offset = end_offset;
} else {
pad->priv->next_offset += pad->priv->size;
}
if (pad->priv->output_offset == -1) {
if (pad->priv->output_offset == -1 || discont) {
GstClockTime start_running_time;
GstClockTime end_running_time;
GstClockTime segment_pos;
@ -1716,7 +1715,6 @@ gst_audio_aggregator_fill_buffer (GstAudioAggregator * aagg,
/* Outside output segment, drop */
pad->priv->position = 0;
pad->priv->size = 0;
pad->priv->output_offset = -1;
GST_DEBUG_OBJECT (pad, "Buffer outside output segment");
return FALSE;
}
@ -1728,14 +1726,16 @@ gst_audio_aggregator_fill_buffer (GstAudioAggregator * aagg,
if (end_output_offset < aagg->priv->offset) {
pad->priv->position = 0;
pad->priv->size = 0;
pad->priv->output_offset = -1;
GST_DEBUG_OBJECT (pad,
"Buffer before segment or current position: %" G_GUINT64_FORMAT " < %"
G_GINT64_FORMAT, end_output_offset, aagg->priv->offset);
return FALSE;
}
if (start_output_offset == -1 || start_output_offset < aagg->priv->offset) {
if (start_output_offset == -1 ||
start_output_offset < aagg->priv->offset ||
(pad->priv->output_offset != -1 &&
start_output_offset < pad->priv->output_offset)) {
guint diff;
if (start_output_offset == -1 && end_output_offset < pad->priv->size) {
@ -1747,6 +1747,9 @@ gst_audio_aggregator_fill_buffer (GstAudioAggregator * aagg,
diff = aagg->priv->offset - start_output_offset;
else
diff = 0;
} else if (pad->priv->output_offset != -1 &&
start_output_offset < pad->priv->output_offset) {
diff = pad->priv->output_offset - start_output_offset;
} else {
diff = aagg->priv->offset - start_output_offset;
}
@ -1756,7 +1759,6 @@ gst_audio_aggregator_fill_buffer (GstAudioAggregator * aagg,
/* Empty buffer, drop */
pad->priv->position = 0;
pad->priv->size = 0;
pad->priv->output_offset = -1;
GST_DEBUG_OBJECT (pad,
"Buffer before segment or current position: %" G_GUINT64_FORMAT
" < %" G_GINT64_FORMAT, end_output_offset, aagg->priv->offset);
@ -1766,6 +1768,9 @@ gst_audio_aggregator_fill_buffer (GstAudioAggregator * aagg,
if (start_output_offset == -1 || start_output_offset < aagg->priv->offset)
pad->priv->output_offset = aagg->priv->offset;
else if (pad->priv->output_offset != -1)
pad->priv->output_offset = MAX (pad->priv->output_offset,
start_output_offset);
else
pad->priv->output_offset = start_output_offset;