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-rw-r--r--gst/mve/mveaudioenc.c154
1 files changed, 0 insertions, 154 deletions
diff --git a/gst/mve/mveaudioenc.c b/gst/mve/mveaudioenc.c
deleted file mode 100644
index 1cff20a71..000000000
--- a/gst/mve/mveaudioenc.c
+++ /dev/null
@@ -1,154 +0,0 @@
-/*
- * 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., 51 Franklin St, Fifth Floor,
- * Boston, MA 02110-1301, USA.
- */
-
-#include <math.h>
-#include <stdlib.h>
-
-#include "gstmvemux.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:%d",
- 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;
-}