Commit 9137e989 authored by Sebastian Dröge's avatar Sebastian Dröge

ext/vorbis/vorbisdec.c: Use the new buffer clipping function from gstaudio here.

Original commit message from CVS:
* ext/vorbis/vorbisdec.c: (vorbis_dec_push_forward),
(vorbis_dec_flush_decode):
Use the new buffer clipping function from gstaudio here.
parent 6be25240
2007-07-23 Sebastian Dröge <slomo@circular-chaos.org>
* ext/vorbis/vorbisdec.c: (vorbis_dec_push_forward),
(vorbis_dec_flush_decode):
Use the new buffer clipping function from gstaudio here.
2007-07-23 Sebastian Dröge <slomo@circular-chaos.org>
* docs/libs/gst-plugins-base-libs-sections.txt:
......@@ -802,58 +802,6 @@ copy_samples (float *out, float **in, guint samples, gint channels)
#endif
}
/* clip output samples to the segment boundaries
*/
static gboolean
vorbis_do_clip (GstVorbisDec * dec, GstBuffer * buf)
{
gint64 start, stop, cstart, cstop, diff;
start = GST_BUFFER_TIMESTAMP (buf);
stop = start + GST_BUFFER_DURATION (buf);
if (!gst_segment_clip (&dec->segment, GST_FORMAT_TIME,
start, stop, &cstart, &cstop))
goto clipped;
/* see if some clipping happened */
diff = cstart - start;
if (diff > 0) {
GST_BUFFER_TIMESTAMP (buf) = cstart;
GST_BUFFER_DURATION (buf) -= diff;
/* bring clipped time to samples */
diff = gst_util_uint64_scale_int (diff, dec->vi.rate, GST_SECOND);
/* samples to bytes */
diff *= (sizeof (float) * dec->vi.channels);
GST_DEBUG_OBJECT (dec, "clipping start to %" GST_TIME_FORMAT " %"
G_GUINT64_FORMAT " bytes", GST_TIME_ARGS (cstart), diff);
GST_BUFFER_DATA (buf) += diff;
GST_BUFFER_SIZE (buf) -= diff;
}
diff = stop - cstop;
if (diff > 0) {
GST_BUFFER_DURATION (buf) -= diff;
/* bring clipped time to samples and then to bytes */
diff = gst_util_uint64_scale_int (diff, dec->vi.rate, GST_SECOND);
diff *= (sizeof (float) * dec->vi.channels);
GST_DEBUG_OBJECT (dec, "clipping stop to %" GST_TIME_FORMAT " %"
G_GUINT64_FORMAT " bytes", GST_TIME_ARGS (cstop), diff);
GST_BUFFER_SIZE (buf) -= diff;
}
return FALSE;
/* dropped buffer */
clipped:
{
GST_DEBUG_OBJECT (dec, "clipped buffer");
gst_buffer_unref (buf);
return TRUE;
}
}
static GstFlowReturn
vorbis_dec_push_forward (GstVorbisDec * dec, GstBuffer * buf)
{
......@@ -899,9 +847,10 @@ vorbis_dec_push_forward (GstVorbisDec * dec, GstBuffer * buf)
for (walk = dec->queued; walk; walk = g_list_next (walk)) {
GstBuffer *buffer = GST_BUFFER (walk->data);
/* clips or returns FALSE with buffer unreffed when completely
/* clips or returns NULL with buffer unreffed when completely
* clipped */
if (vorbis_do_clip (dec, buffer))
if (!(buffer = gst_audio_buffer_clip (buffer, &dec->segment,
dec->vi.rate, dec->vi.channels * sizeof (float))))
continue;
if (dec->discont) {
......@@ -916,7 +865,8 @@ vorbis_dec_push_forward (GstVorbisDec * dec, GstBuffer * buf)
}
/* clip */
if (vorbis_do_clip (dec, buf))
if (!(buf = gst_audio_buffer_clip (buf, &dec->segment, dec->vi.rate,
dec->vi.channels * sizeof (float))))
return GST_FLOW_OK;
if (dec->discont) {
......@@ -1297,7 +1247,8 @@ vorbis_dec_flush_decode (GstVorbisDec * dec)
GST_BUFFER_DURATION (buf) = endts - GST_BUFFER_TIMESTAMP (buf);
/* clip, this will unref the buffer in case of clipping */
if (vorbis_do_clip (dec, buf)) {
if (!(buf = gst_audio_buffer_clip (buf, &dec->segment, dec->vi.rate,
dec->vi.channels * sizeof (float)))) {
GST_DEBUG_OBJECT (dec, "clipped buffer %p", buf);
goto next;
}
......
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