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Use the new buffer clipping function from gstaudio here and require gst-plugins-base CVS.

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Original commit message from CVS:
* configure.ac:
* ext/faad/gstfaad.c: (gst_faad_chain), (gst_faad_change_state):
Use the new buffer clipping function from gstaudio here and
require gst-plugins-base CVS.
This commit is contained in:
Sebastian Dröge 2007-07-24 05:15:41 +00:00
parent 1bbb76ef8e
commit 48ca19fbd6
3 changed files with 11 additions and 53 deletions
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7
ChangeLog
View file

@ -1,3 +1,10 @@
2007-07-24 Sebastian Dröge <slomo@circular-chaos.org>
* configure.ac:
* ext/faad/gstfaad.c: (gst_faad_chain), (gst_faad_change_state):
Use the new buffer clipping function from gstaudio here and
require gst-plugins-base CVS.
2007-07-23 Stefan Kost <ensonic@users.sf.net>
* configure.ac:

2
configure.ac
View file

@ -46,7 +46,7 @@ AM_PROG_LIBTOOL
dnl *** required versions of GStreamer stuff ***
dnl *** remove rtpmanager/equalizer stuff below when this is updated
GST_REQ=0.10.13
GSTPB_REQ=0.10.13
GSTPB_REQ=0.10.13.1
dnl *** autotools stuff ****

55
ext/faad/gstfaad.c
View file

@ -1132,56 +1132,6 @@ looks_like_valid_header (guint8 * input_data, guint input_size)
return TRUE;
}
/*
clips buffer to currently configured segment. Returns FALSE if the buffer
has to be dropped.
*/
static gboolean
clip_outgoing_buffer (GstFaad * faad, GstBuffer * buffer)
{
gint64 start, stop, cstart, cstop, diff;
gboolean res = TRUE;
if (faad->segment->format != GST_FORMAT_TIME)
goto beach;
start = GST_BUFFER_TIMESTAMP (buffer);
stop = start + GST_BUFFER_DURATION (buffer);
if (gst_segment_clip (faad->segment, GST_FORMAT_TIME,
start, stop, &cstart, &cstop)) {
diff = cstart - start;
if (diff > 0) {
GST_BUFFER_TIMESTAMP (buffer) = cstart;
GST_BUFFER_DURATION (buffer) -= diff;
/* time->frames->bytes */
diff =
faad->bps * faad->channels * GST_CLOCK_TIME_TO_FRAMES (diff,
faad->samplerate);
GST_BUFFER_DATA (buffer) += diff;
GST_BUFFER_SIZE (buffer) -= diff;
}
diff = stop - cstop;
if (diff > 0) {
GST_BUFFER_DURATION (buffer) -= diff;
/* time->frames->bytes */
diff =
faad->bps * faad->channels * GST_CLOCK_TIME_TO_FRAMES (diff,
faad->samplerate);
/* update size */
GST_BUFFER_SIZE (buffer) -= diff;
}
} else {
GST_DEBUG_OBJECT (faad, "buffer is outside configured segment");
res = FALSE;
}
beach:
return res;
}
static GstFlowReturn
gst_faad_chain (GstPad * pad, GstBuffer * buffer)
{
@ -1389,7 +1339,8 @@ gst_faad_chain (GstPad * pad, GstBuffer * buffer)
faad->sum_dur_out += GST_BUFFER_DURATION (outbuf);
GST_OBJECT_UNLOCK (faad);
if (clip_outgoing_buffer (faad, outbuf)) {
if ((outbuf = gst_audio_buffer_clip (outbuf, faad->segment,
faad->samplerate, faad->bps * faad->channels))) {
GST_LOG_OBJECT (faad,
"pushing buffer, off=%" G_GUINT64_FORMAT ", ts=%" GST_TIME_FORMAT,
GST_BUFFER_OFFSET (outbuf),
@ -1493,7 +1444,7 @@ gst_faad_change_state (GstElement * element, GstStateChange transition)
return GST_STATE_CHANGE_FAILURE;
break;
case GST_STATE_CHANGE_READY_TO_PAUSED:
gst_segment_init (faad->segment, GST_FORMAT_UNDEFINED);
gst_segment_init (faad->segment, GST_FORMAT_TIME);
break;
default:
break;