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audiorate: Use gst_util_uint64_scale_int_round()

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Using gst_util_uint64_scale_int() causes slight drift
which accumulates over time.

https://bugzilla.gnome.org/show_bug.cgi?id=741045
This commit is contained in:
Chad 2014-12-02 15:58:00 -05:00 committed by Olivier Crête
parent 49611eb364
commit e397b03f35
1 changed files with 11 additions and 9 deletions
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20
gst/audiorate/gstaudiorate.c
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@ -445,7 +445,7 @@ gst_audio_rate_chain (GstPad * pad, GstObject * parent, GstBuffer * buf)
* buffer in that segment, which is the offset expressed in DEFAULT units. * buffer in that segment, which is the offset expressed in DEFAULT units.
*/ */
/* convert first timestamp of segment to sample position */ /* convert first timestamp of segment to sample position */
pos = gst_util_uint64_scale_int (audiorate->src_segment.start, pos = gst_util_uint64_scale_int_round (audiorate->src_segment.start,
GST_AUDIO_INFO_RATE (&audiorate->info), GST_SECOND); GST_AUDIO_INFO_RATE (&audiorate->info), GST_SECOND);
GST_DEBUG_OBJECT (audiorate, "resync to offset %" G_GINT64_FORMAT, pos); GST_DEBUG_OBJECT (audiorate, "resync to offset %" G_GINT64_FORMAT, pos);
@ -454,12 +454,13 @@ gst_audio_rate_chain (GstPad * pad, GstObject * parent, GstBuffer * buf)
audiorate->discont = TRUE; audiorate->discont = TRUE;
audiorate->next_offset = pos; audiorate->next_offset = pos;
audiorate->next_ts = gst_util_uint64_scale_int (audiorate->next_offset, audiorate->next_ts =
GST_SECOND, GST_AUDIO_INFO_RATE (&audiorate->info)); gst_util_uint64_scale_int_round (audiorate->next_offset, GST_SECOND,
GST_AUDIO_INFO_RATE (&audiorate->info));
if (audiorate->skip_to_first && GST_BUFFER_TIMESTAMP_IS_VALID (buf)) { if (audiorate->skip_to_first && GST_BUFFER_TIMESTAMP_IS_VALID (buf)) {
GST_DEBUG_OBJECT (audiorate, "but skipping to first buffer instead"); GST_DEBUG_OBJECT (audiorate, "but skipping to first buffer instead");
pos = gst_util_uint64_scale_int (GST_BUFFER_TIMESTAMP (buf), pos = gst_util_uint64_scale_int_round (GST_BUFFER_TIMESTAMP (buf),
GST_AUDIO_INFO_RATE (&audiorate->info), GST_SECOND); GST_AUDIO_INFO_RATE (&audiorate->info), GST_SECOND);
GST_DEBUG_OBJECT (audiorate, "so resync to offset %" G_GINT64_FORMAT, GST_DEBUG_OBJECT (audiorate, "so resync to offset %" G_GINT64_FORMAT,
pos); pos);
@ -539,10 +540,11 @@ gst_audio_rate_chain (GstPad * pad, GstObject * parent, GstBuffer * buf)
* offset to get duration. Necessary complexity to get 'perfect' * offset to get duration. Necessary complexity to get 'perfect'
* streams */ * streams */
GST_BUFFER_TIMESTAMP (fill) = audiorate->next_ts; GST_BUFFER_TIMESTAMP (fill) = audiorate->next_ts;
audiorate->next_ts = gst_util_uint64_scale_int (audiorate->next_offset, audiorate->next_ts =
GST_SECOND, rate); gst_util_uint64_scale_int_round (audiorate->next_offset, GST_SECOND,
GST_BUFFER_DURATION (fill) = audiorate->next_ts - rate);
GST_BUFFER_TIMESTAMP (fill); GST_BUFFER_DURATION (fill) =
audiorate->next_ts - GST_BUFFER_TIMESTAMP (fill);
/* we created this buffer to fill a gap */ /* we created this buffer to fill a gap */
GST_BUFFER_FLAG_SET (fill, GST_BUFFER_FLAG_GAP); GST_BUFFER_FLAG_SET (fill, GST_BUFFER_FLAG_GAP);
@ -621,7 +623,7 @@ send:
GST_BUFFER_OFFSET_END (buf) = in_offset_end; GST_BUFFER_OFFSET_END (buf) = in_offset_end;
GST_BUFFER_TIMESTAMP (buf) = audiorate->next_ts; GST_BUFFER_TIMESTAMP (buf) = audiorate->next_ts;
audiorate->next_ts = gst_util_uint64_scale_int (in_offset_end, audiorate->next_ts = gst_util_uint64_scale_int_round (in_offset_end,
GST_SECOND, rate); GST_SECOND, rate);
GST_BUFFER_DURATION (buf) = audiorate->next_ts - GST_BUFFER_TIMESTAMP (buf); GST_BUFFER_DURATION (buf) = audiorate->next_ts - GST_BUFFER_TIMESTAMP (buf);