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audiorate: also fill up segments if possible

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This commit is contained in:
Mark Nauwelaerts 2009-12-14 18:47:27 +01:00
parent a11a1858ed
commit 56d4534554
1 changed files with 27 additions and 9 deletions
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36
gst/audiorate/gstaudiorate.c
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@ -294,6 +294,26 @@ gst_audio_rate_init (GstAudioRate * audiorate)
audiorate->silent = DEFAULT_SILENT;
}
static void
gst_audio_rate_fill_to_time (GstAudioRate * audiorate, GstClockTime time)
{
GstBuffer *buf;
GST_DEBUG_OBJECT (audiorate, "next_ts: %" GST_TIME_FORMAT
", filling to %" GST_TIME_FORMAT, GST_TIME_ARGS (audiorate->next_ts),
GST_TIME_ARGS (time));
if (!GST_CLOCK_TIME_IS_VALID (time) ||
!GST_CLOCK_TIME_IS_VALID (audiorate->next_ts))
return;
/* feed an empty buffer to chain with the given timestamp,
* it will take care of filling */
buf = gst_buffer_new ();
GST_BUFFER_TIMESTAMP (buf) = time;
gst_audio_rate_chain (audiorate->sinkpad, buf);
}
static gboolean
gst_audio_rate_sink_event (GstPad * pad, GstEvent * event)
{
@ -319,21 +339,16 @@ gst_audio_rate_sink_event (GstPad * pad, GstEvent * event)
&start, &stop, &time);
GST_DEBUG_OBJECT (audiorate, "handle NEWSEGMENT");
/* FIXME:
* - sparse stream support. For this, the update flag is TRUE and the
* start/time positions are updated, meaning that time progressed by
* time - old_time amount and we need to fill that gap with empty
* samples.
* - fill the current segment if it has a valid stop position. This
* happens when the update flag is FALSE. With the segment helper we can
* calculate the accumulated time and compare this to the next_offset.
*/
/* FIXME: bad things will likely happen if rate < 0 ... */
if (!update) {
/* a new segment starts. We need to figure out what will be the next
* sample offset. We mark the offsets as invalid so that the _chain
* function will perform this calculation. */
gst_audio_rate_fill_to_time (audiorate, audiorate->src_segment.stop);
audiorate->next_offset = -1;
audiorate->next_ts = -1;
} else {
gst_audio_rate_fill_to_time (audiorate, audiorate->src_segment.start);
}
/* we accept all formats */
@ -633,6 +648,9 @@ gst_audio_rate_chain (GstPad * pad, GstBuffer * buf)
}
}
if (GST_BUFFER_SIZE (buf) == 0)
goto beach;
/* Now calculate parameters for whichever buffer (either the original
* or truncated one) we're pushing. */
GST_BUFFER_OFFSET (buf) = audiorate->next_offset;