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-rw-r--r--ChangeLog7
-rw-r--r--configure.ac2
-rw-r--r--ext/faad/gstfaad.c55
3 files changed, 11 insertions, 53 deletions
diff --git a/ChangeLog b/ChangeLog
index 27ed9210..9b2c1816 100644
--- a/ChangeLog
+++ b/ChangeLog
@@ -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:
diff --git a/configure.ac b/configure.ac
index 1311a102..68510f49 100644
--- a/configure.ac
+++ b/configure.ac
@@ -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 ****
diff --git a/ext/faad/gstfaad.c b/ext/faad/gstfaad.c
index 5863e590..3ab62b12 100644
--- a/ext/faad/gstfaad.c
+++ b/ext/faad/gstfaad.c
@@ -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;