gstreamer/gst/mve/mveaudioenc.c
Jens Granseuer 5dbec4ecf4 Add Interplay MVE format demuxer/decoder and muxer/encoder. Demuxer doesn't support seeking yet, but seems to work fi...
Original commit message from CVS:
Patch by: Jens Granseuer  <jensgr at gmx net>
* configure.ac:
* gst/mve/Makefile.am:
* gst/mve/TODO:
* gst/mve/gstmve.c:
* gst/mve/gstmvedemux.c:
* gst/mve/gstmvedemux.h:
* gst/mve/gstmvemux.c:
* gst/mve/gstmvemux.h:
* gst/mve/mve.h:
* gst/mve/mveaudiodec.c:
* gst/mve/mveaudioenc.c:
* gst/mve/mvevideodec16.c:
* gst/mve/mvevideodec8.c:
* gst/mve/mvevideoenc16.c:
* gst/mve/mvevideoenc8.c:
Add Interplay MVE format demuxer/decoder and muxer/encoder. Demuxer
doesn't support seeking yet, but seems to work fine otherwise.
Closes #348973.
2007-01-11 11:39:56 +00:00

120 lines
4.3 KiB
C

/*
* Interplay MVE audio compressor
* Copyright (C) 2003, 2004 Alexander Belyakov <abel@krasu.ru>
* Copyright (C) 2006 Jens Granseuer <jensgr@gmx.net>
*
* This library is free software; you can redistribute it and/or
* modify it under the terms of the GNU Library General Public
* License as published by the Free Software Foundation; either
* version 2 of the License, or (at your option) any later version.
*
* This library is distributed in the hope that it will be useful,
* but WITHOUT ANY WARRANTY; without even the implied warranty of
* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
* Library General Public License for more details.
*
* You should have received a copy of the GNU Library General Public
* License along with this library; if not, write to the
* Free Software Foundation, Inc., 59 Temple Place - Suite 330,
* Boston, MA 02111-1307, USA.
*/
#include <math.h>
#include <gst/gst.h>
static const gint32 dec_table[256] =
{
0, 1, 2, 3, 4, 5, 6, 7, 8, 9, 10, 11, 12, 13, 14, 15,
16, 17, 18, 19,
20, 21, 22, 23, 24, 25, 26, 27, 28, 29, 30, 31,
32, 33, 34, 35, 36, 37,
38, 39, 40, 41, 42, 43, 47, 51, 56, 61,
66, 72, 79, 86, 94, 102, 112,
122, 133, 145, 158, 173, 189, 206, 225, 245,
267, 292, 318, 348, 379,
414, 452, 493, 538, 587, 640, 699, 763, 832, 908, 991,
1081, 1180, 1288,
1405, 1534, 1673, 1826, 1993, 2175, 2373, 2590, 2826, 3084, 3365, 3672,
4008,
4373, 4772, 5208, 5683, 6202, 6767, 7385, 8059, 8794, 9597, 10472,
11428, 12471, 13609, 14851, 16206,
17685, 19298, 21060, 22981, 25078,
27367, 29864, 32589, 35563, 38808, 42350, 46214, 50431, 55033, 60055,
65535,
1, -65535, -60055, -55033, -50431, -46214, -42350, -38808, -35563,
-32589, -29864, -27367, -25078, -22981, -21060, -19298,
-17685, -16206,
-14851, -13609, -12471, -11428, -10472, -9597, -8794, -8059, -7385, -6767,
-6202, -5683, -5208, -4772,
-4373, -4008, -3672, -3365, -3084, -2826,
-2590, -2373, -2175, -1993, -1826, -1673, -1534, -1405, -1288, -1180,
-1081, -991, -908, -832, -763, -699, -640, -587, -538, -493, -452, -414,
-379, -348, -318, -292,
-267, -245, -225, -206, -189, -173, -158, -145,
-133, -122, -112, -102, -94, -86, -79, -72,
-66, -61, -56, -51, -47, -43,
-42, -41, -40, -39, -38, -37, -36, -35, -34, -33,
-32, -31, -30, -29,
-28, -27, -26, -25, -24, -23, -22, -21, -20, -19, -18, -17,
-16, -15,
-14, -13, -12, -11, -10, -9, -8, -7, -6, -5, -4, -3, -2, -1
};
/* This value could be non-optimal. Without knowledge of the value
distribution in the real signal, the actual optimum cannot be evaluated.
Should be somewhere between 11.458 and 11.542. */
static const gdouble DPCM_SCALE = 11.5131;
static gint8
mve_enc_delta (guint n)
{
if (n < 44)
return n;
return floor (DPCM_SCALE * log (n));
}
gint
mve_compress_audio (guint8 * dest, const guint8 * src, guint16 len,
guint8 channels)
{
gint16 prev[2], s;
gint delta, real_res;
gint cur_chan;
guint8 v;
for (cur_chan = 0; cur_chan < channels; ++cur_chan) {
prev[cur_chan] = GST_READ_UINT16_LE (src);
GST_WRITE_UINT16_LE (dest, prev[cur_chan]);
src += 2;
dest += 2;
len -= 2;
}
cur_chan = 0;
while (len > 0) {
s = GST_READ_UINT16_LE (src);
src += 2;
delta = s - prev[cur_chan];
if (delta >= 0)
v = mve_enc_delta (delta);
else
v = 256 - mve_enc_delta (-delta);
real_res = dec_table[v] + prev[cur_chan];
if (real_res < -32768 || real_res > 32767) {
/* correct overflow */
/* GST_DEBUG ("co:%d + %d = %d -> new v:%d, dec_table:%d will be %d",
prev[cur_chan], dec_table[v], real_res,
v, dec_table[v], prev[cur_chan]+dec_table[v]); */
if (s > 0) {
if (real_res > 32767)
--v;
} else {
if (real_res < -32768)
++v;
}
real_res = dec_table[v] + prev[cur_chan];
}
if (G_UNLIKELY (abs (real_res - s) > 32767)) {
GST_ERROR ("sign loss left unfixed in audio stream, deviation:%ld",
real_res - s);
return -1;
}
*dest++ = v;
--len;
/* use previous output instead of input. That way output will not go too far from input. */
prev[cur_chan] += dec_table[v];
cur_chan = channels - 1 - cur_chan;
}
return 0;
}