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Revert last change as we don't want plugins-good to depend on plugins-base CVS now.
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.
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4 changed files with 51 additions and 3 deletions
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@ -1,3 +1,12 @@
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2007-03-30 Sebastian Dröge <slomo@circular-chaos.org>
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* configure.ac:
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* gst/auparse/gstauparse.c: (gst_au_parse_reset),
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(gst_au_parse_parse_header), (gst_au_parse_chain):
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* gst/auparse/gstauparse.h:
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Revert last change as we don't want plugins-good to depend on
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plugins-base CVS now.
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2007-03-29 Sebastian Dröge <slomo@circular-chaos.org>
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* configure.ac:
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@ -47,7 +47,7 @@ AM_PROG_LIBTOOL
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dnl *** required versions of GStreamer stuff ***
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GST_REQ=0.10.11.1
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GSTPB_REQ=0.10.12.1
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GSTPB_REQ=0.10.11.1
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dnl *** autotools stuff ****
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@ -158,6 +158,7 @@ gst_au_parse_reset (GstAuParse * auparse)
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auparse->encoding = 0;
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auparse->samplerate = 0;
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auparse->channels = 0;
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auparse->float_swap = 0;
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gst_adapter_clear (auparse->adapter);
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@ -282,6 +283,8 @@ gst_au_parse_parse_header (GstAuParse * auparse)
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* http://www.tsp.ece.mcgill.ca/MMSP/Documents/AudioFormats/AU/Samples.html
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*/
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auparse->float_swap = 0;
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switch (auparse->encoding) {
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case 1: /* 8-bit ISDN mu-law G.711 */
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law = 1;
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@ -360,9 +363,13 @@ gst_au_parse_parse_header (GstAuParse * auparse)
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tempcaps = gst_caps_new_simple ("audio/x-raw-float",
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"rate", G_TYPE_INT, auparse->samplerate,
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"channels", G_TYPE_INT, auparse->channels,
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"endianness", G_TYPE_INT, auparse->endianness,
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"endianness", G_TYPE_INT, G_BYTE_ORDER,
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"width", G_TYPE_INT, depth, NULL);
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auparse->sample_size = auparse->channels * depth / 8;
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if (auparse->endianness != G_BYTE_ORDER) {
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GST_DEBUG_OBJECT (auparse, "need to swap float byte order ourselves!");
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auparse->float_swap = depth;
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}
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} else if (layout[0]) {
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tempcaps = gst_caps_new_simple ("audio/x-adpcm",
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"layout", G_TYPE_STRING, layout, NULL);
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@ -467,7 +474,35 @@ gst_au_parse_chain (GstPad * pad, GstBuffer * buf)
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}
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data = gst_adapter_peek (auparse->adapter, sendnow);
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memcpy (GST_BUFFER_DATA (outbuf), data, sendnow);
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/* audioconvert only handles floats in native endianness ... */
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switch (auparse->float_swap) {
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case 32:{
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guint32 *indata = (guint32 *) data;
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guint32 *outdata = (guint32 *) GST_BUFFER_DATA (outbuf);
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gint i;
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for (i = 0; i < (sendnow / sizeof (guint32)); ++i) {
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outdata[i] = GUINT32_SWAP_LE_BE (indata[i]);
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}
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break;
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}
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case 64:{
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guint64 *indata = (guint64 *) data;
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guint64 *outdata = (guint64 *) GST_BUFFER_DATA (outbuf);
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gint i;
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for (i = 0; i < (sendnow / sizeof (guint64)); ++i) {
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outdata[i] = GUINT64_SWAP_LE_BE (indata[i]);
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}
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break;
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}
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default:{
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memcpy (GST_BUFFER_DATA (outbuf), data, sendnow);
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break;
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}
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}
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gst_adapter_flush (auparse->adapter, sendnow);
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auparse->buffer_offset += sendnow;
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@ -62,6 +62,10 @@ struct _GstAuParse {
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guint samplerate;
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guint endianness;
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guint channels;
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/* audioconvert only handles float in native endianness,
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* so we need to swap endianness here ourselves for now */
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guint float_swap; /* 0, 32 or 64 */
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};
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struct _GstAuParseClass {
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