FaSTTUBe/GST-Tensordecoder-ov_ep
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Revert last change as we don't want plugins-good to depend on plugins-base CVS now.

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Original commit message from CVS:
* configure.ac:
* gst/auparse/gstauparse.c: (gst_au_parse_reset),
(gst_au_parse_parse_header), (gst_au_parse_chain):
* gst/auparse/gstauparse.h:
Revert last change as we don't want plugins-good to depend on
plugins-base CVS now.
This commit is contained in:
Sebastian Dröge 2007-03-30 15:59:27 +00:00
parent 6d8e6c9bb0
commit 6632cdb003
4 changed files with 51 additions and 3 deletions
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9
ChangeLog
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@ -1,3 +1,12 @@
2007-03-30 Sebastian Dröge <slomo@circular-chaos.org>
* configure.ac:
* gst/auparse/gstauparse.c: (gst_au_parse_reset),
(gst_au_parse_parse_header), (gst_au_parse_chain):
* gst/auparse/gstauparse.h:
Revert last change as we don't want plugins-good to depend on
plugins-base CVS now.
2007-03-29 Sebastian Dröge <slomo@circular-chaos.org> 2007-03-29 Sebastian Dröge <slomo@circular-chaos.org>
* configure.ac: * configure.ac:

2
configure.ac
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@ -47,7 +47,7 @@ AM_PROG_LIBTOOL
dnl *** required versions of GStreamer stuff *** dnl *** required versions of GStreamer stuff ***
GST_REQ=0.10.11.1 GST_REQ=0.10.11.1
GSTPB_REQ=0.10.12.1 GSTPB_REQ=0.10.11.1
dnl *** autotools stuff **** dnl *** autotools stuff ****

37
gst/auparse/gstauparse.c
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@ -158,6 +158,7 @@ gst_au_parse_reset (GstAuParse * auparse)
auparse->encoding = 0; auparse->encoding = 0;
auparse->samplerate = 0; auparse->samplerate = 0;
auparse->channels = 0; auparse->channels = 0;
auparse->float_swap = 0;
gst_adapter_clear (auparse->adapter); gst_adapter_clear (auparse->adapter);
@ -282,6 +283,8 @@ gst_au_parse_parse_header (GstAuParse * auparse)
* http://www.tsp.ece.mcgill.ca/MMSP/Documents/AudioFormats/AU/Samples.html * http://www.tsp.ece.mcgill.ca/MMSP/Documents/AudioFormats/AU/Samples.html
*/ */
auparse->float_swap = 0;
switch (auparse->encoding) { switch (auparse->encoding) {
case 1: /* 8-bit ISDN mu-law G.711 */ case 1: /* 8-bit ISDN mu-law G.711 */
law = 1; law = 1;
@ -360,9 +363,13 @@ gst_au_parse_parse_header (GstAuParse * auparse)
tempcaps = gst_caps_new_simple ("audio/x-raw-float", tempcaps = gst_caps_new_simple ("audio/x-raw-float",
"rate", G_TYPE_INT, auparse->samplerate, "rate", G_TYPE_INT, auparse->samplerate,
"channels", G_TYPE_INT, auparse->channels, "channels", G_TYPE_INT, auparse->channels,
"endianness", G_TYPE_INT, auparse->endianness, "endianness", G_TYPE_INT, G_BYTE_ORDER,
"width", G_TYPE_INT, depth, NULL); "width", G_TYPE_INT, depth, NULL);
auparse->sample_size = auparse->channels * depth / 8; auparse->sample_size = auparse->channels * depth / 8;
if (auparse->endianness != G_BYTE_ORDER) {
GST_DEBUG_OBJECT (auparse, "need to swap float byte order ourselves!");
auparse->float_swap = depth;
}
} else if (layout[0]) { } else if (layout[0]) {
tempcaps = gst_caps_new_simple ("audio/x-adpcm", tempcaps = gst_caps_new_simple ("audio/x-adpcm",
"layout", G_TYPE_STRING, layout, NULL); "layout", G_TYPE_STRING, layout, NULL);
@ -467,7 +474,35 @@ gst_au_parse_chain (GstPad * pad, GstBuffer * buf)
} }
data = gst_adapter_peek (auparse->adapter, sendnow); data = gst_adapter_peek (auparse->adapter, sendnow);
/* audioconvert only handles floats in native endianness ... */
switch (auparse->float_swap) {
case 32:{
guint32 *indata = (guint32 *) data;
guint32 *outdata = (guint32 *) GST_BUFFER_DATA (outbuf);
gint i;
for (i = 0; i < (sendnow / sizeof (guint32)); ++i) {
outdata[i] = GUINT32_SWAP_LE_BE (indata[i]);
}
break;
}
case 64:{
guint64 *indata = (guint64 *) data;
guint64 *outdata = (guint64 *) GST_BUFFER_DATA (outbuf);
gint i;
for (i = 0; i < (sendnow / sizeof (guint64)); ++i) {
outdata[i] = GUINT64_SWAP_LE_BE (indata[i]);
}
break;
}
default:{
memcpy (GST_BUFFER_DATA (outbuf), data, sendnow); memcpy (GST_BUFFER_DATA (outbuf), data, sendnow);
break;
}
}
gst_adapter_flush (auparse->adapter, sendnow); gst_adapter_flush (auparse->adapter, sendnow);
auparse->buffer_offset += sendnow; auparse->buffer_offset += sendnow;

4
gst/auparse/gstauparse.h
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@ -62,6 +62,10 @@ struct _GstAuParse {
guint samplerate; guint samplerate;
guint endianness; guint endianness;
guint channels; guint channels;
/* audioconvert only handles float in native endianness,
* so we need to swap endianness here ourselves for now */
guint float_swap; /* 0, 32 or 64 */
}; };
struct _GstAuParseClass { struct _GstAuParseClass {