gstreamer/gst/mve/mveaudioenc.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;
}