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2015年11月1日 (日) 13:53時点における版

マンゴー実況
カテゴリ	隔離
サーバ	maguro
フォルダ	liveanarchy
開設日	2014年 7月2日
名無しの名前	穴カス（無能）
ID	表示なし
板URL	http://maguro.2ch.net/mango/

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目次

1 戦犯リスト

2 赤座あかりちゃんのチンコバッキバキでワロタｗｗｗ😓

3 諸注意

4 ISO 13818-3 MPEG-2 Audio Encoder - Lower Sampling Frequency Extension

5 記事本文ここまで

6 <a href="http://anazitu.blog.jp/archives/1038397374.html" itemprop="url">ちはやふる 28巻　漫画感想</a>

6.1 あっ...(察し)

6.2 タバ速のこの画像どういう意味なのか教えて

6.3 自

6.4 Berners-Lee & Connolly Standards Track [Page 61]

6.4.1 ‮Berners-Lee & Connolly Standards Track [Page 62]

6.4.2 Berners-Lee & Connolly Standards Track [Page 63]

6.4.3 Berners-Lee & Connolly Standards Track [Page 64]

戦犯リスト
http://info.2ch.net/index.php?title=Anarchy%E5%AE%9F%E6%B3%81%E6%9D%BF&action=history
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‮WarというのはこのWikiの荒らしなので荒らしていたら注意してあげよう。that any increase in the speed with which insight and improvements can be distributed to others can result in a dramatic acceleration in those improvements. I believe wiki -

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←

†

R.I.P 赤座あかり 💩😅

赤座あかりちゃんのチンコバッキバキでワロタｗｗｗ😓

諸注意

なお、ならないことを承知した上で読み進めて下さい。
穴実は聞けば教えてくれない厳しい世界です。文化は口伝で脈々と受け継がれています。

あんなかわいい顔して立派なものつけてるとかぐう興奮する

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ISO 13818-3 MPEG-2 Audio Encoder - Lower Sampling Frequency Extension

*
* $Id: encode.c,v 1.1 1996/02/14 04:04:23 rowlands Exp $
*
* $Log: encode.c,v $
* Revision 1.1  1996/02/14 04:04:23  rowlands
* Initial revision
*
* Received from Mike Coleman
**********************************************************************/

/**********************************************************************

*   date   programmers         comment                               *
* 3/01/91  Douglas Wong,       start of version 1.1 records          *
*          Davis Pan                                                 *
* 3/06/91  Douglas Wong        rename: setup.h to endef.h            *
*                                      efilter to enfilter           *
*                                      ewindow to enwindow           *
*                              integrated "quantizer", "scalefactor",*
*                              and "transmission" files              *
*                              update routine "window_subband"       *
* 3/31/91  Bill Aspromonte     replaced read_filter by               *
*                              create_an_filter                      *
* 5/10/91  W. Joseph Carter    Ported to Macintosh and Unix.         *
*                              Incorporated Jean-Georges Fritsch's   *
*                              "bitstream.c" package.                *
*                              Incorporated Bill Aspromonte's        *
*                              filterbank coefficient matrix         *
*                              calculation routines and added        *
*                              roundoff to coincide with specs.      *
*                              Modified to strictly adhere to        *
*                              encoded bitstream specs, including    *
*                              "Berlin changes".                     *
*                              Modified PCM sound file handling to   *
*                              process all incoming samples and fill *
*                              out last encoded frame with zeros     *
*                              (silence) if needed.                  *
*                              Located and fixed numerous software   *
*                              bugs and table data errors.           *
* 19jun91  dpwe (Aware)        moved "alloc_*" reader to common.c    *
*                              Globals sblimit, alloc replaced by new*
*                              struct 'frame_params' passed as arg.  *
*                              Added JOINT STEREO coding, layers I,II*
*                              Affects: *_bit_allocation,            *
*                              subband_quantization, encode_bit_alloc*
*                              sample_encoding                       *
* 6/10/91  Earle Jennings      modified II_subband_quantization to   *
*                              resolve type cast problem for MS_DOS  *
* 6/11/91  Earle Jennings      modified to avoid overflow on MS_DOS  *
*                              in routine filter_subband             *
* 7/10/91  Earle Jennings      port to MsDos from MacIntosh version  *
* 8/ 8/91  Jens Spille         Change for MS-C6.00                   *
*10/ 1/91  S.I. Sudharsanan,   Ported to IBM AIX platform.           *
*          Don H. Lee,                                               *
*          Peter W. Farrett                                          *
*10/ 3/91  Don H. Lee          implemented CRC-16 error protection   *
*                              newly introduced function encode_CRC  *
*11/ 8/91  Kathy Wang          Documentation of code                 *
*                              All variablenames are referred to     *
*                              with surrounding pound (#) signs      *
* 2/11/92  W. Joseph Carter    Ported new code to Macintosh.  Most   *
*                              important fixes involved changing     *
*                              16-bit ints to long or unsigned in    *
*                              bit alloc routines for quant of 65535 *
*                              and passing proper function args.     *
*                              Removed "Other Joint Stereo" option   *
*                              and made bitrate be total channel     *
*                              bitrate, irrespective of the mode.    *
*                              Fixed many small bugs & reorganized.  *
* 6/16/92  Shaun Astarabadi    Changed I_scale_factor_calc() and     *
*                              II_scale_factor_calc() to use scale   *
*                              factor 0 thru 62 only and not to      *
*                              encode index 63 into the bit stream.  *
* 7/27/92  Mike Li             (re-)Port to MS-DOS                   *
* 9/22/92  jddevine@aware.com  Fixed _scale_factor_calc() defs       *
* 3/31/93  Giogio Dimino       changed II_a_bit_allocation() from:   *
*                              if( ad > ...) to if(ad >= ...)        *
* 8/05/93  TEST                changed I_a_bit_allocation() from:    *
*                              if( ad > ...) to if(ad >= ...)        *
* 8/02/95  mc@fivebats.com     Changed audio file reading code to    *
*                              read samples big-endian               *
*10/15/95  mc@fivebats.com     Modified get_audio() for layer3-LSF   *
**********************************************************************/

include "common.h"
include "encoder.h"

ifdef MS_DOS

extern unsigned _stklen = 16384;

endif

/*=======================================================================\ | | | This segment contains all the core routines of the encoder, | | except for the psychoacoustic models. | | | | The user can select either one of the two psychoacoustic | | models. Model I is a simple tonal and noise masking threshold | | generator, and Model II is a more sophisticated cochlear masking | | threshold generator. Model I is recommended for lower complexity | | applications whereas Model II gives better subjective quality at low | | bit rates. | | | | Layers I and II of mono, stereo, and joint stereo modes are supported.| | Routines associated with a given layer are prefixed by "I_" for layer | | 1 and "II_" for layer 2. | \=======================================================================*/

/************************************************************************

read_samples()
PURPOSE: reads the PCM samples from a file to the buffer
SEMANTICS:
Reads #samples_read# number of shorts from #musicin# filepointer
into #sample_buffer[]#. Returns the number of samples read.
- - - - /

unsigned long read_samples(musicin, sample_buffer, num_samples, frame_size) FILE *musicin; short sample_buffer[2304]; unsigned long num_samples, frame_size; {

   unsigned long samples_read;
   static unsigned long samples_to_read;
   static char init = TRUE;

   if (init) {
       samples_to_read = num_samples;
       init = FALSE;
   }
   if (samples_to_read >= frame_size)
       samples_read = frame_size;
   else
       samples_read = samples_to_read;
   if ((samples_read =
        fread(sample_buffer, sizeof(short), (int)samples_read, musicin)) == 0)
       printf("Hit end of audio data\n");
   /*
      Samples are big-endian. If this is a little-endian machine
      we must swap
    */
   if ( NativeByteOrder == order_unknown )
     {

NativeByteOrder = DetermineByteOrder(); if ( NativeByteOrder == order_unknown ) { fprintf( stderr, "byte order not determined\n" ); exit( 1 ); }

     }
   if ( NativeByteOrder == order_littleEndian )
     SwapBytesInWords( sample_buffer, samples_read );

   samples_to_read -= samples_read;
   if (samples_read < frame_size && samples_read > 0) {
       printf("Insufficient PCM input for one frame - fillout with zeros\n");
       for (; samples_read < frame_size; sample_buffer[samples_read++] = 0);
       samples_to_read = 0;
   }
   return(samples_read);

}

/************************************************************************

get_audio()
PURPOSE: reads a frame of audio data from a file to the buffer,
aligns the data for future processing, and separates the
left and right channels
- - - - /

unsigned long get_audio( musicin, buffer, num_samples, stereo, info ) FILE *musicin; short FAR buffer[2][1152]; unsigned long num_samples; int stereo; layer *info; {

   int j;
   short insamp[2304];
   unsigned long samples_read;
   int lay;
   lay = info->lay;

   if ( (lay == 3) && (info->version == 0) )
   {

if ( stereo == 2 ) { samples_read = read_samples( musicin, insamp, num_samples, (unsigned long) 1152 ); for ( j = 0; j < 576; j++ ) { buffer[0][j] = insamp[2 * j]; buffer[1][j] = insamp[2 * j + 1]; } } else { samples_read = read_samples( musicin, insamp, num_samples, (unsigned long) 576 ); for ( j = 0; j < 576; j++ ) { buffer[0][j] = insamp[j]; buffer[1][j] = 0; } }

   }
   else
   {

if (lay == 1){ if(stereo == 2){ /* layer 1, stereo */ samples_read = read_samples(musicin, insamp, num_samples, (unsigned long) 768); for(j=0;j<448;j++) { if(j<64) { buffer[0][j] = buffer[0][j+384]; buffer[1][j] = buffer[1][j+384]; } else { buffer[0][j] = insamp[2*j-128]; buffer[1][j] = insamp[2*j-127]; } } } else { /* layer 1, mono */ samples_read = read_samples(musicin, insamp, num_samples, (unsigned long) 384); for(j=0;j<448;j++){ if(j<64) { buffer[0][j] = buffer[0][j+384]; buffer[1][j] = 0; } else { buffer[0][j] = insamp[j-64]; buffer[1][j] = 0; } } } } else { if(stereo == 2){ /* layer 2 (or 3), stereo */ samples_read = read_samples(musicin, insamp, num_samples, (unsigned long) 2304); for(j=0;j<1152;j++) { buffer[0][j] = insamp[2*j]; buffer[1][j] = insamp[2*j+1]; } } else { /* layer 2 (or 3), mono */ samples_read = read_samples(musicin, insamp, num_samples, (unsigned long) 1152); for(j=0;j<1152;j++){ buffer[0][j] = insamp[j]; buffer[1][j] = 0; } } }

   }
   return(samples_read);

}

/************************************************************************

read_ana_window()
PURPOSE: Reads encoder window file "enwindow" into array #ana_win#
- - - - /

void read_ana_window(ana_win) double FAR ana_win[HAN_SIZE]; {

   int i,j[4];
   FILE *fp;
   double f[4];
   char t[150];

   if (!(fp = OpenTableFile("enwindow") ) ) {
      printf("Please check analysis window table 'enwindow'\n");
      exit(1);
   }
   for (i=0;i<512;i+=4) {
      fgets(t, 150, fp);
      sscanf(t,"C[%d] = %lf C[%d] = %lf C[%d] = %lf C[%d] = %lf\n",
             j, f,j+1,f+1,j+2,f+2,j+3,f+3);
      if (i==j[0]) {
         ana_win[i] = f[0];
         ana_win[i+1] = f[1];
         ana_win[i+2] = f[2];
         ana_win[i+3] = f[3];
      }
      else {
         printf("Check index in analysis window table\n");
         exit(1);
      }
      fgets(t,150,fp);
   }
   fclose(fp);

}

/************************************************************************

window_subband()
PURPOSE: Overlapping window on PCM samples
SEMANTICS:
32 16-bit pcm samples are scaled to fractional 2's complement and
concatenated to the end of the window buffer #x#. The updated window
buffer #x# is then windowed by the analysis window #c# to produce the
windowed sample #z#
- - - - /

void window_subband(buffer, z, k) short FAR **buffer; double FAR z[HAN_SIZE]; int k; {

   typedef double FAR XX[2][HAN_SIZE];
   static XX FAR *x;
   int i, j;
   static off[2] = {0,0};
   static char init = 0;
   static double FAR *c;
   if (!init) {
       c = (double FAR *) mem_alloc(sizeof(double) * HAN_SIZE, "window");
       read_ana_window(c);
       x = (XX FAR *) mem_alloc(sizeof(XX),"x");
       for (i=0;i<2;i++)
           for (j=0;j<HAN_SIZE;j++)
               (*x)[i][j] = 0;
       init = 1;
   }

   /* replace 32 oldest samples with 32 new samples */
   for (i=0;i<32;i++) (*x)[k][31-i+off[k]] = (double) *(*buffer)++/SCALE;
   /* shift samples into proper window positions */
   for (i=0;i<HAN_SIZE;i++) z[i] = (*x)[k][(i+off[k])&HAN_SIZE-1] * c[i];
   off[k] += 480;              /*offset is modulo (HAN_SIZE-1)*/
   off[k] &= HAN_SIZE-1;

}

/************************************************************************

create_ana_filter()
PURPOSE: Calculates the analysis filter bank coefficients
SEMANTICS:
Calculates the analysis filterbank coefficients and rounds to the
9th decimal place accuracy of the filterbank tables in the ISO
document. The coefficients are stored in #filter#

- - - - /

void create_ana_filter(filter) double FAR filter[SBLIMIT][64]; {

  register int i,k;

  for (i=0; i<32; i++)
     for (k=0; k<64; k++) {
         if ((filter[i][k] = 1e9*cos((double)((2*i+1)*(16-k)*PI64))) >= 0)
            modf(filter[i][k]+0.5, &filter[i][k]);
         else
            modf(filter[i][k]-0.5, &filter[i][k]);
         filter[i][k] *= 1e-9;
  }

}

/************************************************************************

filter_subband()
PURPOSE: Calculates the analysis filter bank coefficients
SEMANTICS:
The windowed samples #z# is filtered by the digital filter matrix #m#
to produce the subband samples #s#. This done by first selectively
picking out values from the windowed samples, and then multiplying
them by the filter matrix, producing 32 subband samples.
- - - - /

void filter_subband(z,s) double FAR z[HAN_SIZE], s[SBLIMIT]; {

  double y[64];
  int i,j;

static char init = 0;

  typedef double MM[SBLIMIT][64];

static MM FAR *m;

ifdef MS_DOS

  long    SIZE_OF_MM;
  SIZE_OF_MM      = SBLIMIT*64;
  SIZE_OF_MM      *= 8;
  if (!init) {
      m = (MM FAR *) mem_alloc(SIZE_OF_MM, "filter");
      create_ana_filter(*m);
      init = 1;
  }

else

  if (!init) {
      m = (MM FAR *) mem_alloc(sizeof(MM), "filter");
      create_ana_filter(*m);
      init = 1;
  }

endif

  for (i=0;i<64;i++) for (j=0, y[i] = 0;j<8;j++) y[i] += z[i+64*j];
  for (i=0;i<SBLIMIT;i++)
      for (j=0, s[i]= 0;j<64;j++) s[i] += (*m)[i][j] * y[j];

}

/************************************************************************

encode_info()
PURPOSE: Puts the syncword and header information on the output
bitstream.
- - - - /

void encode_info(fr_ps,bs) frame_params *fr_ps; Bit_stream_struc *bs; {

       layer *info = fr_ps->header;

       putbits(bs,0xfff,12);                    /* syncword 12 bits */
       put1bit(bs,info->version);               /* ID        1 bit  */
       putbits(bs,4-info->lay,2);               /* layer     2 bits */
       put1bit(bs,!info->error_protection);     /* bit set => no err prot */
       putbits(bs,info->bitrate_index,4);
       putbits(bs,info->sampling_frequency,2);
       put1bit(bs,info->padding);
       put1bit(bs,info->extension);             /* private_bit */
       putbits(bs,info->mode,2);
       putbits(bs,info->mode_ext,2);
       put1bit(bs,info->copyright);
       put1bit(bs,info->original);
       putbits(bs,info->emphasis,2);

}

/************************************************************************

mod()
PURPOSE: Returns the absolute value of its argument
- - - - /

double mod(a) double a; {

   return (a > 0) ? a : -a;

}

/************************************************************************

I_combine_LR (Layer I)
II_combine_LR (Layer II)
PURPOSE:Combines left and right channels into a mono channel
SEMANTICS: The average of left and right subband samples is put into
#joint_sample#
Layer I and II differ in frame length and # subbands used
- - - - /

void I_combine_LR(sb_sample, joint_sample) double FAR sb_sample[2][3][SCALE_BLOCK][SBLIMIT]; double FAR joint_sample[3][SCALE_BLOCK][SBLIMIT]; { /* make a filtered mono for joint stereo */

   int sb, smp;

  for(sb = 0; sb<SBLIMIT; ++sb)
     for(smp = 0; smp<SCALE_BLOCK; ++smp)
       joint_sample[0][smp][sb] = .5 *
                   (sb_sample[0][0][smp][sb] + sb_sample[1][0][smp][sb]);

}

void II_combine_LR(sb_sample, joint_sample, sblimit) double FAR sb_sample[2][3][SCALE_BLOCK][SBLIMIT]; double FAR joint_sample[3][SCALE_BLOCK][SBLIMIT]; int sblimit; { /* make a filtered mono for joint stereo */

  int sb, smp, sufr;

  for(sb = 0; sb<sblimit; ++sb)
     for(smp = 0; smp<SCALE_BLOCK; ++smp)
        for(sufr = 0; sufr<3; ++sufr)
           joint_sample[sufr][smp][sb] = .5 * (sb_sample[0][sufr][smp][sb]
                                          + sb_sample[1][sufr][smp][sb]);

}

/************************************************************************

I_scale_factor_calc (Layer I)
II_scale_factor_calc (Layer II)
PURPOSE:For each subband, calculate the scale factor for each set
of the 12 subband samples
SEMANTICS: Pick the scalefactor #multiple[]# just larger than the
absolute value of the peak subband sample of 12 samples,
and store the corresponding scalefactor index in #scalar#.
Layer II has three sets of 12-subband samples for a given
subband.
- - - - /

void I_scale_factor_calc(sb_sample,scalar,stereo) double FAR sb_sample[][3][SCALE_BLOCK][SBLIMIT]; unsigned int scalar[][3][SBLIMIT]; int stereo; {

  int i,j, k;
  double s[SBLIMIT];

  for (k=0;k<stereo;k++) {
    for (i=0;i<SBLIMIT;i++)
      for (j=1, s[i] = mod(sb_sample[k][0][0][i]);j<SCALE_BLOCK;j++)
        if (mod(sb_sample[k][0][j][i]) > s[i])
           s[i] = mod(sb_sample[k][0][j][i]);

    for (i=0;i<SBLIMIT;i++)
      for (j=SCALE_RANGE-2,scalar[k][0][i]=0;j>=0;j--) /* $A 6/16/92 */
        if (s[i] <= multiple[j]) {
           scalar[k][0][i] = j;
           break;
        }
  }

}

/******************************** Layer II ******************************/

void II_scale_factor_calc(sb_sample,scalar,stereo,sblimit) double FAR sb_sample[][3][SCALE_BLOCK][SBLIMIT]; unsigned int scalar[][3][SBLIMIT]; int stereo,sblimit; {

 int i,j, k,t;
 double s[SBLIMIT];

 for (k=0;k<stereo;k++) for (t=0;t<3;t++) {
   for (i=0;i<sblimit;i++)
     for (j=1, s[i] = mod(sb_sample[k][t][0][i]);j<SCALE_BLOCK;j++)
       if (mod(sb_sample[k][t][j][i]) > s[i])
            s[i] = mod(sb_sample[k][t][j][i]);

 for (i=0;i<sblimit;i++)
   for (j=SCALE_RANGE-2,scalar[k][t][i]=0;j>=0;j--)    /* $A 6/16/92 */
     if (s[i] <= multiple[j]) {
        scalar[k][t][i] = j;
        break;
     }
     for (i=sblimit;i<SBLIMIT;i++) scalar[k][t][i] = SCALE_RANGE-1;
   }

}

/************************************************************************

pick_scale (Layer II)
PURPOSE:For each subband, puts the smallest scalefactor of the 3
associated with a frame into #max_sc#. This is used
used by Psychoacoustic Model I.
(I would recommend changin max_sc to min_sc)
- - - - /

void pick_scale(scalar, fr_ps, max_sc) unsigned int scalar[2][3][SBLIMIT]; frame_params *fr_ps; double FAR max_sc[2][SBLIMIT]; {

 int i,j,k,max;
 int stereo  = fr_ps->stereo;
 int sblimit = fr_ps->sblimit;

 for (k=0;k<stereo;k++)
   for (i=0;i<sblimit;max_sc[k][i] = multiple[max],i++)
     for (j=1, max = scalar[k][0][i];j<3;j++)
        if (max > scalar[k][j][i]) max = scalar[k][j][i];
 for (i=sblimit;i<SBLIMIT;i++) max_sc[0][i] = max_sc[1][i] = 1E-20;

}

/************************************************************************

put_scale (Layer I)
PURPOSE:Sets #max_sc# to the scalefactor index in #scalar.
This is used by Psychoacoustic Model I
- - - - /

void put_scale(scalar, fr_ps, max_sc) unsigned int scalar[2][3][SBLIMIT]; frame_params *fr_ps; double FAR max_sc[2][SBLIMIT]; {

  int i,j,k, max;
  int stereo  = fr_ps->stereo;
  int sblimit = fr_ps->sblimit;

  for (k=0;k<stereo;k++) for (i=0;i<SBLIMIT;i++)
       max_sc[k][i] = multiple[scalar[k][0][i]];

}

/************************************************************************

II_transmission_pattern (Layer II only)
PURPOSE:For a given subband, determines whether to send 1, 2, or
all 3 of the scalefactors, and fills in the scalefactor
select information accordingly
SEMANTICS: The subbands and channels are classified based on how much
the scalefactors changes over its three values (corresponding
to the 3 sets of 12 samples per subband). The classification
will send 1 or 2 scalefactors instead of three if the scalefactors
do not change much. The scalefactor select information,
#scfsi#, is filled in accordingly.
- - - - /

void II_transmission_pattern(scalar, scfsi, fr_ps) unsigned int scalar[2][3][SBLIMIT]; unsigned int scfsi[2][SBLIMIT]; frame_params *fr_ps; {

  int stereo  = fr_ps->stereo;
  int sblimit = fr_ps->sblimit;
  int dscf[2];
  int class[2],i,j,k;

static int pattern[5][5] = {0x123, 0x122, 0x122, 0x133, 0x123,

                           0x113, 0x111, 0x111, 0x444, 0x113,
                           0x111, 0x111, 0x111, 0x333, 0x113,
                           0x222, 0x222, 0x222, 0x333, 0x123,
                           0x123, 0x122, 0x122, 0x133, 0x123};

  for (k=0;k<stereo;k++)
    for (i=0;i<sblimit;i++) {
      dscf[0] =  (scalar[k][0][i]-scalar[k][1][i]);
      dscf[1] =  (scalar[k][1][i]-scalar[k][2][i]);
      for (j=0;j<2;j++) {
        if (dscf[j]<=-3) class[j] = 0;
        else if (dscf[j] > -3 && dscf[j] <0) class[j] = 1;
             else if (dscf[j] == 0) class[j] = 2;
                  else if (dscf[j] > 0 && dscf[j] < 3) class[j] = 3;
                       else class[j] = 4;
      }
      switch (pattern[class[0]][class[1]]) {
        case 0x123 :    scfsi[k][i] = 0;
                        break;
        case 0x122 :    scfsi[k][i] = 3;
                        scalar[k][2][i] = scalar[k][1][i];
                        break;
        case 0x133 :    scfsi[k][i] = 3;
                        scalar[k][1][i] = scalar[k][2][i];
                        break;
        case 0x113 :    scfsi[k][i] = 1;
                        scalar[k][1][i] = scalar[k][0][i];
                        break;
        case 0x111 :    scfsi[k][i] = 2;
                        scalar[k][1][i] = scalar[k][2][i] = scalar[k][0][i];
                        break;
        case 0x222 :    scfsi[k][i] = 2;
                        scalar[k][0][i] = scalar[k][2][i] = scalar[k][1][i];
                        break;
        case 0x333 :    scfsi[k][i] = 2;
                        scalar[k][0][i] = scalar[k][1][i] = scalar[k][2][i];
                        break;
        case 0x444 :    scfsi[k][i] = 2;
                        if (scalar[k][0][i] > scalar[k][2][i])
                             scalar[k][0][i] = scalar[k][2][i];
                        scalar[k][1][i] = scalar[k][2][i] = scalar[k][0][i];
     }
  }

}

/************************************************************************

I_encode_scale (Layer I)
II_encode_scale (Layer II)
PURPOSE:The encoded scalar factor information is arranged and
queued into the output fifo to be transmitted.
For Layer II, the three scale factors associated with
a given subband and channel are transmitted in accordance
with the scfsi, which is transmitted first.
- - - - /

void I_encode_scale(scalar, bit_alloc, fr_ps, bs) unsigned int scalar[2][3][SBLIMIT]; unsigned int bit_alloc[2][SBLIMIT]; frame_params *fr_ps; Bit_stream_struc *bs; {

  int stereo  = fr_ps->stereo;
  int sblimit = fr_ps->sblimit;
  int i,j;

  for (i=0;i<SBLIMIT;i++) for (j=0;j<stereo;j++)
     if (bit_alloc[j][i]) putbits(bs,scalar[j][0][i],6);

}

/***************************** Layer II ********************************/

void II_encode_scale(bit_alloc, scfsi, scalar, fr_ps, bs) unsigned int bit_alloc[2][SBLIMIT], scfsi[2][SBLIMIT]; unsigned int scalar[2][3][SBLIMIT]; frame_params *fr_ps; Bit_stream_struc *bs; {

  int stereo  = fr_ps->stereo;
  int sblimit = fr_ps->sblimit;
  int jsbound = fr_ps->jsbound;
  int i,j,k;

  for (i=0;i<sblimit;i++) for (k=0;k<stereo;k++)
    if (bit_alloc[k][i])  putbits(bs,scfsi[k][i],2);

  for (i=0;i<sblimit;i++) for (k=0;k<stereo;k++)
    if (bit_alloc[k][i])  /* above jsbound, bit_alloc[0][i] == ba[1][i] */
       switch (scfsi[k][i]) {
          case 0: for (j=0;j<3;j++)
                    putbits(bs,scalar[k][j][i],6);
                  break;
          case 1:
          case 3: putbits(bs,scalar[k][0][i],6);
                  putbits(bs,scalar[k][2][i],6);
                  break;
          case 2: putbits(bs,scalar[k][0][i],6);
       }

}

/*=======================================================================\ | | | The following routines are done after the masking threshold | | has been calculated by the fft analysis routines in the Psychoacoustic | | model. Using the MNR calculated, the actual number of bits allocated | | to each subband is found iteratively. | | | \=======================================================================*/

/************************************************************************

I_bits_for_nonoise (Layer I)
II_bits_for_nonoise (Layer II)
PURPOSE:Returns the number of bits required to produce a
mask-to-noise ratio better or equal to the noise/no_noise threshold.
SEMANTICS:
bbal = # bits needed for encoding bit allocation
bsel = # bits needed for encoding scalefactor select information
banc = # bits needed for ancillary data (header info included)
For each subband and channel, will add bits until one of the
following occurs:
- Hit maximum number of bits we can allocate for that subband
- MNR is better than or equal to the minimum masking level
(NOISY_MIN_MNR)
Then the bits required for scalefactors, scfsi, bit allocation,
and the subband samples are tallied (#req_bits#) and returned.
(NOISY_MIN_MNR) is the smallest MNR a subband can have before it is
counted as 'noisy' by the logic which chooses the number of JS
subbands.
Joint stereo is supported.
- - - - /

static double snr[18] = {0.00, 7.00, 11.00, 16.00, 20.84,

                        25.28, 31.59, 37.75, 43.84,
                        49.89, 55.93, 61.96, 67.98, 74.01,
                        80.03, 86.05, 92.01, 98.01};

int I_bits_for_nonoise(perm_smr, fr_ps) double FAR perm_smr[2][SBLIMIT]; frame_params *fr_ps; {

  int i,j,k;
  int stereo  = fr_ps->stereo;
  int sblimit = fr_ps->sblimit;
  int jsbound = fr_ps->jsbound;
  int req_bits = 0;

  /* initial b_anc (header) allocation bits */
  req_bits = 32 + 4 * ( (jsbound * stereo) + (SBLIMIT-jsbound) );

  for(i=0; i<SBLIMIT; ++i)
    for(j=0; j<((i<jsbound)?stereo:1); ++j) {
      for(k=0;k<14; ++k)
        if( (-perm_smr[j][i] + snr[k]) >= NOISY_MIN_MNR)
          break; /* we found enough bits */
        if(stereo == 2 && i >= jsbound)     /* check other JS channel */
          for(;k<14; ++k)
            if( (-perm_smr[1-j][i] + snr[k]) >= NOISY_MIN_MNR) break;
        if(k>0) req_bits += (k+1)*SCALE_BLOCK + 6*((i>=jsbound)?stereo:1);
  }
  return req_bits;

}

/***************************** Layer II ********************************/

int II_bits_for_nonoise(perm_smr, scfsi, fr_ps) double FAR perm_smr[2][SBLIMIT]; unsigned int scfsi[2][SBLIMIT]; frame_params *fr_ps; {

  int sb,ch,ba;
  int stereo  = fr_ps->stereo;
  int sblimit = fr_ps->sblimit;
  int jsbound = fr_ps->jsbound;
  al_table *alloc = fr_ps->alloc;
  int req_bits = 0, bbal = 0, berr = 0, banc = 32;
  int maxAlloc, sel_bits, sc_bits, smp_bits;

static int sfsPerScfsi[] = { 3,2,1,2 }; /* lookup # sfs per scfsi */

  /* added 92-08-11 shn */
  if (fr_ps->header->error_protection) berr=16; else berr=0; 

  for (sb=0; sb<jsbound; ++sb)
    bbal += stereo * (*alloc)[sb][0].bits;
  for (sb=jsbound; sb<sblimit; ++sb)
    bbal += (*alloc)[sb][0].bits;
  req_bits = banc + bbal + berr;

  for(sb=0; sb<sblimit; ++sb)
    for(ch=0; ch<((sb<jsbound)?stereo:1); ++ch) {
      maxAlloc = (1<<(*alloc)[sb][0].bits)-1;
      sel_bits = sc_bits = smp_bits = 0;
      for(ba=0;ba<maxAlloc-1; ++ba)
        if( (-perm_smr[ch][sb] + snr[(*alloc)[sb][ba].quant+((ba>0)?1:0)])
            >= NOISY_MIN_MNR)
           break;      /* we found enough bits */
      if(stereo == 2 && sb >= jsbound) /* check other JS channel */
        for(;ba<maxAlloc-1; ++ba)
          if( (-perm_smr[1-ch][sb]+ snr[(*alloc)[sb][ba].quant+((ba>0)?1:0)])
              >= NOISY_MIN_MNR)
            break;
      if(ba>0) {
        smp_bits = SCALE_BLOCK * ((*alloc)[sb][ba].group * (*alloc)[sb][ba].bits);
        /* scale factor bits required for subband */
        sel_bits = 2;
        sc_bits  = 6 * sfsPerScfsi[scfsi[ch][sb]];
        if(stereo == 2 && sb >= jsbound) {
          /* each new js sb has L+R scfsis */
          sel_bits += 2;
          sc_bits  += 6 * sfsPerScfsi[scfsi[1-ch][sb]];
        }
        req_bits += smp_bits+sel_bits+sc_bits;
      }
  }
  return req_bits;

}

/************************************************************************

I_main_bit_allocation (Layer I)
II_main_bit_allocation (Layer II)
PURPOSE:For joint stereo mode, determines which of the 4 joint
stereo modes is needed. Then calls *_a_bit_allocation(), which
allocates bits for each of the subbands until there are no more bits
left, or the MNR is at the noise/no_noise threshold.
SEMANTICS:
For joint stereo mode, joint stereo is changed to stereo if
there are enough bits to encode stereo at or better than the
no-noise threshold (NOISY_MIN_MNR). Otherwise, the system
iteratively allocates less bits by using joint stereo until one
of the following occurs:
- there are no more noisy subbands (MNR >= NOISY_MIN_MNR)
- mode_ext has been reduced to 0, which means that all but the
lowest 4 subbands have been converted from stereo to joint
stereo, and no more subbands may be converted
This function calls *_bits_for_nonoise() and *_a_bit_allocation().
- - - - /

void I_main_bit_allocation(perm_smr, bit_alloc, adb, fr_ps) double FAR perm_smr[2][SBLIMIT]; unsigned int bit_alloc[2][SBLIMIT]; int *adb; frame_params *fr_ps; {

  int  noisy_sbs;
  int  mode, mode_ext, lay, i;
  int  rq_db, av_db = *adb;

static int init = 0;

  if(init == 0) {
    /* rearrange snr for layer I */
    snr[2] = snr[3];
    for (i=3;i<16;i++) snr[i] = snr[i+2];
    init = 1;
  }

  if((mode = fr_ps->actual_mode) == MPG_MD_JOINT_STEREO) {
    fr_ps->header->mode = MPG_MD_STEREO;
    fr_ps->header->mode_ext = 0;
    fr_ps->jsbound = fr_ps->sblimit;
    if(rq_db = I_bits_for_nonoise(perm_smr, fr_ps) > *adb) {
      fr_ps->header->mode = MPG_MD_JOINT_STEREO;
      mode_ext = 4;           /* 3 is least severe reduction */
      lay = fr_ps->header->lay;
      do {
         --mode_ext;
         fr_ps->jsbound = js_bound(lay, mode_ext);
         rq_db = I_bits_for_nonoise(perm_smr, fr_ps);
      } while( (rq_db > *adb) && (mode_ext > 0));
      fr_ps->header->mode_ext = mode_ext;
    }    /* well we either eliminated noisy sbs or mode_ext == 0 */
  }
  noisy_sbs = I_a_bit_allocation(perm_smr, bit_alloc, adb, fr_ps);

}

/***************************** Layer II ********************************/

void II_main_bit_allocation(perm_smr, scfsi, bit_alloc, adb, fr_ps) double FAR perm_smr[2][SBLIMIT]; unsigned int scfsi[2][SBLIMIT]; unsigned int bit_alloc[2][SBLIMIT]; int *adb; frame_params *fr_ps; {

  int  noisy_sbs, nn;
  int  mode, mode_ext, lay;
  int  rq_db, av_db = *adb;

  if((mode = fr_ps->actual_mode) == MPG_MD_JOINT_STEREO) {
    fr_ps->header->mode = MPG_MD_STEREO;
    fr_ps->header->mode_ext = 0;
    fr_ps->jsbound = fr_ps->sblimit;
    if((rq_db=II_bits_for_nonoise(perm_smr, scfsi, fr_ps)) > *adb) {
      fr_ps->header->mode = MPG_MD_JOINT_STEREO;
      mode_ext = 4;           /* 3 is least severe reduction */
      lay = fr_ps->header->lay;
      do {
        --mode_ext;
        fr_ps->jsbound = js_bound(lay, mode_ext);
        rq_db = II_bits_for_nonoise(perm_smr, scfsi, fr_ps);
      } while( (rq_db > *adb) && (mode_ext > 0));
      fr_ps->header->mode_ext = mode_ext;
    }    /* well we either eliminated noisy sbs or mode_ext == 0 */
  }
  noisy_sbs = II_a_bit_allocation(perm_smr, scfsi, bit_alloc, adb, fr_ps);

}

/************************************************************************

I_a_bit_allocation (Layer I)
II_a_bit_allocation (Layer II)
PURPOSE:Adds bits to the subbands with the lowest mask-to-noise
ratios, until the maximum number of bits for the subband has
been allocated.
SEMANTICS:
1. Find the subband and channel with the smallest MNR (#min_sb#,
and #min_ch#)
2. Calculate the increase in bits needed if we increase the bit
allocation to the next higher level
3. If there are enough bits available for increasing the resolution
in #min_sb#, #min_ch#, and the subband has not yet reached its
maximum allocation, update the bit allocation, MNR, and bits

   available accordingly

4. Repeat until there are no more bits left, or no more available
subbands. (A subband is still available until the maximum
number of bits for the subband has been allocated, or there
aren't enough bits to go to the next higher resolution in the

   subband.)

- - - - /

int I_a_bit_allocation(perm_smr, bit_alloc, adb, fr_ps) /* return noisy sbs */ double FAR perm_smr[2][SBLIMIT]; unsigned int bit_alloc[2][SBLIMIT]; int *adb; frame_params *fr_ps; {

  int i, k, smpl_bits, scale_bits, min_sb, min_ch, oth_ch;
  int bspl, bscf, ad, noisy_sbs, done = 0, bbal ;
  double mnr[2][SBLIMIT], small;
  char used[2][SBLIMIT];
  int stereo  = fr_ps->stereo;
  int sblimit = fr_ps->sblimit;
  int jsbound = fr_ps->jsbound;
  al_table *alloc = fr_ps->alloc;

static char init= 0; static int banc=32, berr=0;

  if (!init) {
     init = 1;
     if (fr_ps->header->error_protection) berr = 16;  /* added 92-08-11 shn */
  }
  bbal = 4 * ( (jsbound * stereo) + (SBLIMIT-jsbound) );
  *adb -= bbal + berr + banc;
  ad= *adb;

  for (i=0;i<SBLIMIT;i++) for (k=0;k<stereo;k++) {
    mnr[k][i]=snr[0]-perm_smr[k][i];
    bit_alloc[k][i] = 0;
    used[k][i] = 0;
  }
  bspl = bscf = 0;

  do  {
    /* locate the subband with minimum SMR */
    small = mnr[0][0]+1;    min_sb = -1; min_ch = -1;
    for (i=0;i<SBLIMIT;i++) for (k=0;k<stereo;k++)
      /* go on only if there are bits left */
      if (used[k][i] != 2 && small > mnr[k][i]) {
        small = mnr[k][i];
        min_sb = i;  min_ch = k;
      }
    if(min_sb > -1) {   /* there was something to find */
      /* first step of bit allocation is biggest */
      if (used[min_ch][min_sb])  { smpl_bits = SCALE_BLOCK; scale_bits = 0; }
      else                       { smpl_bits = 24; scale_bits = 6; }
      if(min_sb >= jsbound)        scale_bits *= stereo;

      /* check to see enough bits were available for */
      /* increasing resolution in the minimum band */

      if (ad >= bspl + bscf + scale_bits + smpl_bits) {
        bspl += smpl_bits; /* bit for subband sample */
        bscf += scale_bits; /* bit for scale factor */
        bit_alloc[min_ch][min_sb]++;
        used[min_ch][min_sb] = 1; /* subband has bits */
        mnr[min_ch][min_sb] = -perm_smr[min_ch][min_sb]
                              + snr[bit_alloc[min_ch][min_sb]];
        /* Check if subband has been fully allocated max bits */
        if (bit_alloc[min_ch][min_sb] ==  14 ) used[min_ch][min_sb] = 2;
      }
      else            /* no room to improve this band */
        used[min_ch][min_sb] = 2; /*   for allocation anymore */
      if(stereo == 2 && min_sb >= jsbound) {
        oth_ch = 1-min_ch;  /* joint-st : fix other ch */
        bit_alloc[oth_ch][min_sb] = bit_alloc[min_ch][min_sb];
        used[oth_ch][min_sb] = used[min_ch][min_sb];
        mnr[oth_ch][min_sb] = -perm_smr[oth_ch][min_sb]
                              + snr[bit_alloc[oth_ch][min_sb]];
      }
    }
  } while(min_sb>-1);     /* i.e. still some sub-bands to find */

  /* Calculate the number of bits left, add on to pointed var */
  ad -= bspl+bscf;
  *adb = ad;

  /* see how many channels are noisy */
  noisy_sbs = 0; small = mnr[0][0];
  for(k=0; k<stereo; ++k) {
    for(i = 0; i< SBLIMIT; ++i) {
      if(mnr[k][i] < NOISY_MIN_MNR)   ++noisy_sbs;
      if(small > mnr[k][i])           small = mnr[k][i];
    }
  }
  return noisy_sbs;

}

/***************************** Layer II ********************************/

int II_a_bit_allocation(perm_smr, scfsi, bit_alloc, adb, fr_ps) double FAR perm_smr[2][SBLIMIT]; unsigned int scfsi[2][SBLIMIT]; unsigned int bit_alloc[2][SBLIMIT]; int *adb; frame_params *fr_ps; {

  int i, min_ch, min_sb, oth_ch, k, increment, scale, seli, ba;
  int bspl, bscf, bsel, ad, noisy_sbs, bbal=0;
  double mnr[2][SBLIMIT], small;
  char used[2][SBLIMIT];
  int stereo  = fr_ps->stereo;
  int sblimit = fr_ps->sblimit;
  int jsbound = fr_ps->jsbound;
  al_table *alloc = fr_ps->alloc;

static char init= 0; static int banc=32, berr=0; static int sfsPerScfsi[] = { 3,2,1,2 }; /* lookup # sfs per scfsi */

  if (!init) { 
      init = 1;  
      if (fr_ps->header->error_protection) berr=16; /* added 92-08-11 shn */
  }
  for (i=0; i<jsbound; ++i)
    bbal += stereo * (*alloc)[i][0].bits;
  for (i=jsbound; i<sblimit; ++i)
    bbal += (*alloc)[i][0].bits;
  *adb -= bbal + berr + banc;
  ad = *adb;

  for (i=0;i<sblimit;i++) for (k=0;k<stereo;k++) {
    mnr[k][i]=snr[0]-perm_smr[k][i];
    bit_alloc[k][i] = 0;
    used[k][i] = 0;
  }
  bspl = bscf = bsel = 0;

  do  {
    /* locate the subband with minimum SMR */
    small = 999999.0; min_sb = -1; min_ch = -1;
    for (i=0;i<sblimit;i++) for(k=0;k<stereo;++k)
      if (used[k][i]  != 2 && small > mnr[k][i]) {
        small = mnr[k][i];
        min_sb = i;  min_ch = k;
    }
    if(min_sb > -1) {   /* there was something to find */
      /* find increase in bit allocation in subband [min] */
      increment = SCALE_BLOCK * ((*alloc)[min_sb][bit_alloc[min_ch][min_sb]+1].group *
                       (*alloc)[min_sb][bit_alloc[min_ch][min_sb]+1].bits);
      if (used[min_ch][min_sb])
        increment -= SCALE_BLOCK * ((*alloc)[min_sb][bit_alloc[min_ch][min_sb]].group*
                          (*alloc)[min_sb][bit_alloc[min_ch][min_sb]].bits);

      /* scale factor bits required for subband [min] */
      oth_ch = 1 - min_ch;    /* above js bound, need both chans */
      if (used[min_ch][min_sb]) scale = seli = 0;
      else {          /* this channel had no bits or scfs before */
        seli = 2;
        scale = 6 * sfsPerScfsi[scfsi[min_ch][min_sb]];
        if(stereo == 2 && min_sb >= jsbound) {
          /* each new js sb has L+R scfsis */
          seli += 2;
          scale += 6 * sfsPerScfsi[scfsi[oth_ch][min_sb]];
        }
      }
      /* check to see enough bits were available for */
      /* increasing resolution in the minimum band */
      if (ad >= bspl + bscf + bsel + seli + scale + increment) {
        ba = ++bit_alloc[min_ch][min_sb]; /* next up alloc */
        bspl += increment;  /* bits for subband sample */
        bscf += scale;      /* bits for scale factor */
        bsel += seli;       /* bits for scfsi code */
        used[min_ch][min_sb] = 1; /* subband has bits */
        mnr[min_ch][min_sb] = -perm_smr[min_ch][min_sb] +
                              snr[(*alloc)[min_sb][ba].quant+1];
        /* Check if subband has been fully allocated max bits */
        if (ba >= (1<<(*alloc)[min_sb][0].bits)-1) used[min_ch][min_sb] = 2;
      }
      else used[min_ch][min_sb] = 2; /* can't increase this alloc */
      if(min_sb >= jsbound && stereo == 2) {
        /* above jsbound, alloc applies L+R */
        ba = bit_alloc[oth_ch][min_sb] = bit_alloc[min_ch][min_sb];
        used[oth_ch][min_sb] = used[min_ch][min_sb];
        mnr[oth_ch][min_sb] = -perm_smr[oth_ch][min_sb] +
                              snr[(*alloc)[min_sb][ba].quant+1];
      }
    }
  } while(min_sb > -1);   /* until could find no channel */
  /* Calculate the number of bits left */
  ad -= bspl+bscf+bsel;   *adb = ad;
  for (i=sblimit;i<SBLIMIT;i++) for (k=0;k<stereo;k++) bit_alloc[k][i]=0;

  noisy_sbs = 0;  small = mnr[0][0];      /* calc worst noise in case */
  for(k=0;k<stereo;++k) {
    for (i=0;i<sblimit;i++) {
      if (small > mnr[k][i]) small = mnr[k][i];
      if(mnr[k][i] < NOISY_MIN_MNR) ++noisy_sbs; /* noise is not masked */

    }
  }
  return noisy_sbs;

}

/************************************************************************

I_subband_quantization (Layer I)
II_subband_quantization (Layer II)
PURPOSE:Quantizes subband samples to appropriate number of bits
SEMANTICS: Subband samples are divided by their scalefactors, which

makes the quantization more efficient. The scaled samples are

quantized by the function a*x+b, where a and b are functions of
the number of quantization levels. The result is then truncated
to the appropriate number of bits and the MSB is inverted.
Note that for fractional 2's complement, inverting the MSB for a

negative number x is equivalent to adding 1 to it.

- - - - /

static double a[17] = {

 0.750000000, 0.625000000, 0.875000000, 0.562500000, 0.937500000,
 0.968750000, 0.984375000, 0.992187500, 0.996093750, 0.998046875,
 0.999023438, 0.999511719, 0.999755859, 0.999877930, 0.999938965,
 0.999969482, 0.999984741 };

static double b[17] = {

 -0.250000000, -0.375000000, -0.125000000, -0.437500000, -0.062500000,
 -0.031250000, -0.015625000, -0.007812500, -0.003906250, -0.001953125,
 -0.000976563, -0.000488281, -0.000244141, -0.000122070, -0.000061035,
 -0.000030518, -0.000015259 };

void I_subband_quantization(scalar, sb_samples, j_scale, j_samps,

                           bit_alloc, sbband, fr_ps)

unsigned int scalar[2][3][SBLIMIT]; double FAR sb_samples[2][3][SCALE_BLOCK][SBLIMIT]; unsigned int j_scale[3][SBLIMIT]; double FAR j_samps[3][SCALE_BLOCK][SBLIMIT]; /* L+R for j-stereo if necess */ unsigned int bit_alloc[2][SBLIMIT]; unsigned int FAR sbband[2][3][SCALE_BLOCK][SBLIMIT]; frame_params *fr_ps; {

  int i, j, k, n, sig;
  int stereo  = fr_ps->stereo;
  int sblimit = fr_ps->sblimit;
  int jsbound = fr_ps->jsbound;
  double d;

static char init = 0;

  if (!init) {
    init = 1;
    /* rearrange quantization coef to correspond to layer I table */
    a[1] = a[2]; b[1] = b[2];
    for (i=2;i<15;i++) { a[i] = a[i+2]; b[i] = b[i+2]; }
  }
  for (j=0;j<SCALE_BLOCK;j++) for (i=0;i<SBLIMIT;i++)
    for (k=0;k<((i<jsbound)?stereo:1);k++)
      if (bit_alloc[k][i]) {
        /* for joint stereo mode, have to construct a single subband stream
           for the js channels.  At present, we calculate a set of mono
           subband samples and pass them through the scaling system to
           generate an alternate normalised sample stream.

           Could normalise both streams (divide by their scfs), then average
           them.  In bad conditions, this could give rise to spurious
           cancellations.  Instead, we could just select the sb stream from
           the larger channel (higher scf), in which case _that_ channel
           would be 'properly' reconstructed, and the mate would just be a
           scaled version.  Spec recommends averaging the two (unnormalised)
           subband channels, then normalising this new signal without
           actually sending this scale factor... This means looking ahead.
        */
        if(stereo == 2 && i>=jsbound)
          /* use the joint data passed in */
          d = j_samps[0][j][i] / multiple[j_scale[0][i]];
        else
          d = sb_samples[k][0][j][i] / multiple[scalar[k][0][i]];
        /* scale and quantize floating point sample */
        n = bit_alloc[k][i];
        d = d * a[n-1] + b[n-1];
        /* extract MSB N-1 bits from the floating point sample */
        if (d >= 0) sig = 1;
        else { sig = 0; d += 1.0; }
        sbband[k][0][j][i] = (unsigned int) (d * (double) (1L<<n));
        /* tag the inverted sign bit to sbband at position N */
        if (sig) sbband[k][0][j][i] |= 1<<n;
      }

}

/***************************** Layer II ********************************/

void II_subband_quantization(scalar, sb_samples, j_scale, j_samps,

                            bit_alloc, sbband, fr_ps)

unsigned int scalar[2][3][SBLIMIT]; double FAR sb_samples[2][3][SCALE_BLOCK][SBLIMIT]; unsigned int j_scale[3][SBLIMIT]; double FAR j_samps[3][SCALE_BLOCK][SBLIMIT]; unsigned int bit_alloc[2][SBLIMIT]; unsigned int FAR sbband[2][3][SCALE_BLOCK][SBLIMIT]; frame_params *fr_ps; {

  int i, j, k, s, n, qnt, sig;
  int stereo  = fr_ps->stereo;
  int sblimit = fr_ps->sblimit;
  int jsbound = fr_ps->jsbound;
  unsigned int stps;
  double d;
  al_table *alloc = fr_ps->alloc;

  for (s=0;s<3;s++)
    for (j=0;j<SCALE_BLOCK;j++)
      for (i=0;i<sblimit;i++)
        for (k=0;k<((i<jsbound)?stereo:1);k++)
          if (bit_alloc[k][i]) {
            /* scale and quantize floating point sample */
            if(stereo == 2 && i>=jsbound)       /* use j-stereo samples */
              d = j_samps[s][j][i] / multiple[j_scale[s][i]];
            else
              d = sb_samples[k][s][j][i] / multiple[scalar[k][s][i]];
            if (mod(d) > 1.0)
              printf("Not scaled properly %d %d %d %d\n",k,s,j,i);
            qnt = (*alloc)[i][bit_alloc[k][i]].quant;
            d = d * a[qnt] + b[qnt];
            /* extract MSB N-1 bits from the floating point sample */
            if (d >= 0) sig = 1;
            else { sig = 0; d += 1.0; }
            n = 0;

ifndef MS_DOS

            stps = (*alloc)[i][bit_alloc[k][i]].steps;
            while ((1L<<n) < stps) n++;

else

            while  ( ( (unsigned long)(1L<<(long)n) <
                      ((unsigned long) ((*alloc)[i][bit_alloc[k][i]].steps)
                       & 0xffff
                       )
                      ) && ( n <16)
                    ) n++;

endif

            n--;
            sbband[k][s][j][i] = (unsigned int) (d * (double) (1L<<n));
            /* tag the inverted sign bit to sbband at position N */
            /* The bit inversion is a must for grouping with 3,5,9 steps
               so it is done for all subbands */
            if (sig) sbband[k][s][j][i] |= 1<<n;
          }
          for (s=0;s<3;s++)
            for (j=sblimit;j<SBLIMIT;j++)
              for (i=0;i<SCALE_BLOCK;i++) for (k=0;k<stereo;k++) sbband[k][s][i][j] = 0;

}

/*************************************************************************

I_encode_bit_alloc (Layer I)
II_encode_bit_alloc (Layer II)
PURPOSE:Writes bit allocation information onto bitstream
Layer I uses 4 bits/subband for bit allocation information,
and Layer II uses 4,3,2, or 0 bits depending on the
quantization table used.
- - - - /

void I_encode_bit_alloc(bit_alloc, fr_ps, bs) unsigned int bit_alloc[2][SBLIMIT]; frame_params *fr_ps; Bit_stream_struc *bs; {

  int i,k;
  int stereo  = fr_ps->stereo;
  int sblimit = fr_ps->sblimit;
  int jsbound = fr_ps->jsbound;

  for (i=0;i<SBLIMIT;i++)
    for (k=0;k<((i<jsbound)?stereo:1);k++) putbits(bs,bit_alloc[k][i],4);

}

/***************************** Layer II ********************************/

void II_encode_bit_alloc(bit_alloc, fr_ps, bs) unsigned int bit_alloc[2][SBLIMIT]; frame_params *fr_ps; Bit_stream_struc *bs; {

  int i,k;
  int stereo  = fr_ps->stereo;
  int sblimit = fr_ps->sblimit;
  int jsbound = fr_ps->jsbound;
  al_table *alloc = fr_ps->alloc;

  for (i=0;i<sblimit;i++)
    for (k=0;k<((i<jsbound)?stereo:1);k++)
      putbits(bs,bit_alloc[k][i],(*alloc)[i][0].bits);

}

/************************************************************************

I_sample_encoding (Layer I)
II_sample_encoding (Layer II)
PURPOSE:Put one frame of subband samples on to the bitstream
SEMANTICS: The number of bits allocated per sample is read from
the bit allocation information #bit_alloc#. Layer 2
supports writing grouped samples for quantization steps
that are not a power of 2.
- - - - /

void I_sample_encoding(sbband, bit_alloc, fr_ps, bs) unsigned int FAR sbband[2][3][SCALE_BLOCK][SBLIMIT]; unsigned int bit_alloc[2][SBLIMIT]; frame_params *fr_ps; Bit_stream_struc *bs; {

  int i,j,k;
  int stereo  = fr_ps->stereo;
  int sblimit = fr_ps->sblimit;
  int jsbound = fr_ps->jsbound;

  for(j=0;j<SCALE_BLOCK;j++) {
    for(i=0;i<SBLIMIT;i++)
      for(k=0;k<((i<jsbound)?stereo:1);k++)
        if(bit_alloc[k][i]) putbits(bs,sbband[k][0][j][i],bit_alloc[k][i]+1);
  }

}

/***************************** Layer II ********************************/

void II_sample_encoding(sbband, bit_alloc, fr_ps, bs) unsigned int FAR sbband[2][3][SCALE_BLOCK][SBLIMIT]; unsigned int bit_alloc[2][SBLIMIT]; frame_params *fr_ps; Bit_stream_struc *bs; {

  unsigned int temp;
  unsigned int i,j,k,s,x,y;
  int stereo  = fr_ps->stereo;
  int sblimit = fr_ps->sblimit;
  int jsbound = fr_ps->jsbound;
  al_table *alloc = fr_ps->alloc;

  for (s=0;s<3;s++)
    for (j=0;j<SCALE_BLOCK;j+=3)
      for (i=0;i<sblimit;i++)
        for (k=0;k<((i<jsbound)?stereo:1);k++)
          if (bit_alloc[k][i]) {
            if ((*alloc)[i][bit_alloc[k][i]].group == 3) {
              for (x=0;x<3;x++) putbits(bs,sbband[k][s][j+x][i],
                                        (*alloc)[i][bit_alloc[k][i]].bits);
            }
            else {
              y =(*alloc)[i][bit_alloc[k][i]].steps;
              temp = sbband[k][s][j][i] +
                     sbband[k][s][j+1][i] * y +
                     sbband[k][s][j+2][i] * y * y;
              putbits(bs,temp,(*alloc)[i][bit_alloc[k][i]].bits);
            }
          }

}

/************************************************************************

encode_CRC
- - - - /

void encode_CRC(crc, bs) unsigned int crc; Bit_stream_struc *bs; {

  putbits(bs, crc, 16);

}

                   </header>

久々にマックのポテト食べたらこれかよまじで最悪 <a href="ht‮tp://t.co/ELnO7q7ZMh">pic.twitter.com/ELnO7q7ZMh</a>
— いとーともき (@1121_ito) <a href="https://twitter.com/1121_ito/status/637496727528800256">2015, 8月 29</a>

                       <a href="http://anazitu.blog.jp/archives/1038450082.html#more">【マクドナルドのフライドポテトの中からゴキブリ　穴実反応】の続きを読む</a>

タグ：: <a href="http://anazitu.blog.jp/tag/%E7%A9%B4%E5%AE%9F" title="穴実タグの一覧へ">穴実</a>; <a href="http://anazitu.blog.jp/tag/%E3%81%BE%E3%81%A8%E3%82%81" title="まとめタグの一覧へ">まとめ</a>

記事本文ここまで

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   dc:title="ちはやふる 28巻　漫画感想"
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<a href="http://anazitu.blog.jp/archives/1038397374.html" itemprop="url">ちはやふる 28巻　漫画感想</a>

2015年08月29日

カテゴリ:
<a href="http://anazitu.blog.jp/archives/cat_1105978.html">ちはやふる</a>
<a href="http://anazitu.blog.jp/archives/cat_1105979.html">感想</a>

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                       <a href="http://anazitu.blog.jp/archives/1038397374.html#more">【ちはやふる 28巻　漫画感想】の続きを読む</a>

タグ：: <a href="http://anazitu.blog.jp/tag/%E3%81%A1%E3%81%AF%E3%82%84%E3%81%B5%E3%82%8B" title="ちはやふるタグの一覧へ">ちはやふる</a>; <a href="http://anazitu.blog.jp/tag/%E6%84%9F%E6%83%B3" title="感想タグの一覧へ">感想</a>

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日赤座←左右対称に近い 1 名前：ライブ・アナーキーさん[] 投稿日：2015/08/30(日) 21:26:07.67 エラ吊り目ファビョ染まる

2 名前：ライブ・アナーキーさん[] 投稿日：2015/08/30(日) 21:26:45.16

あっ...(察し)

もっと読む »
投稿者 暇つぶしさん 時刻: 23:13 0 件のコメント: メールで送信BlogThis!Twitter で共有するFacebook で共有するPinterest に共有
2015年8月30日日曜日

タバ速のこの画像どういう意味なのか教えて

1 ：ライブ・アナーキーさん：2015/08/30(日) 18:59:52.89

   http://livedoor.blogimg.jp/live_anarchy/imgs/5/c/5cf515e1.png

   運営がコメントしてるの

2 ：ライブ・アナーキーさん：2015/08/30(日) 19:01:19.16

自

もっと読む » 投稿者暇つぶしさん時刻: 19:01 0 件のコメント: メールで送信BlogThis!Twitter で共有するFacebook で共有するPinterest に共有 2015年8月29日土曜日去年の夏の穴実からいるヤツって今どれくらいいる？ 1 名前：ライブ・アナーキーさん[] 投稿日：2015/05/04(月) 16:10:44.62 俺は去年の8月からおるけど、1月くらいから急に人減ったな

2 名前：ライブ・アナーキーさん[] 投稿日：2015/05/04(月) 16:12:19.30 カラコロから入ったで確かに人減ったな

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   ▼  2015 (19)
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           赤座←左右対称に近い

Berners-Lee & Connolly Standards Track [Page 61]

RFC 1866 Hypertext Markup Language - 2.0 November 1995

<!ENTITY % HTML.Version

       "-//IETF//DTD HTML 2.0 Strict//EN"

       -- Typical usage:

           <!DOCTYPE HTML PUBLIC
               "-//IETF//DTD HTML Strict//EN">
           <html>
           ...
           </html>
       --
       >

<!ENTITY % HTML.Recommended "INCLUDE">

<!ENTITY % html PUBLIC "-//IETF//DTD HTML 2.0//EN"> %html;

　　　　　　 llq,,_　　　　　　　　　　　　　　　　　　　　　　　　　　　　　　　　　　　　　　　　__,,ggy　　___
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　　　　　　　]llllllll][　　　　　　　　　　　　　　　　　　　　　　　　　　　　　　　　　　　　＾^".lllll)ﾟllilillglili「゜.;llllllllll
　　　　　　　 lililllllili　　　 ,,gggy,,　　　　　　　 ,,,,,,,,,,,,,,gggg,,ggggggggg、　　　　　　　 .＼llll]lﾟ　 ＾)llllllq,,lili(「゜
　　.;g,,　　　　llllllll[　　　 glilillllllll]　　　　　　.ヽllllllll)((ﾟ^^^""""＾＾＾＾＾　　　　　　　　　 」llll(ggllllllllk＾ﾟllilillg,,,,,,
　　 lilill;;　　　.lllll][　　　.l[[.lllllllll][　　　　　　　　　　　　　　　_　_______　　　　　　　　　lili]ﾟl((]llllliliﾟ,,,,__　 ^^^^ﾟl4=
　　 ]llllllll　.lkqllll]゜　__,,lilillllllllllll](　　　　　　　tgglllllllllllllllllli;i;i;i;i;i;i;i;lililllll.　　　　　　　gllll(゛.／lllllilillll(ﾟ^
　　 lilillll](,,glilillllliligllllllll]("lilillll][　　　　　　　　＾゛＾　　　　　　　　　　　　　　　　.ヽll[ﾟ^ vlilillll][ﾟlllll[
　　 lilillllllllllllllll]ll((fﾟ^"＾　.llllllllllili　　　　　　　　　　　　　　　　　　　　　　　　　　　　　　 ^^^](゜]llll[　　　　 .;gg,,,,
　　　^lllllili^＾　　　　　　　.＼](ﾟl'

9.3. Level 1 HTML DTD

  This document type declaration refers to the HTML DTD with the
  `HTML.Forms' entity defined as `IGNORE' rather than `INCLUDE'.
  Documents which contain <FORM> elements do not conform to this DTD,
  and must use the level 2 DTD.

<!ENTITY % HTML.Version

       "-//IETF//DTD HTML 2.0 Level 1//EN"

       -- Typical usage:

           <!DOCTYPE HTML PUBLIC
               "-//IETF//DTD HTML Level 1//EN">
           <html>
           ...
           </html>

‮Berners-Lee & Connolly Standards Track [Page 62]

RFC 1866 Hypertext Markup Language - 2.0 November 1995

       --
       >

<!ENTITY % HTML.Forms "IGNORE">

<!ENTITY % html PUBLIC "-//IETF//DTD HTML 2.0//EN"> %html;

9.4. Strict Level 1 HTML DTD

  This document type declaration refers to the level 1 HTML DTD with
  the `HTML.Recommended' entity defined as `INCLUDE' rather than
  IGNORE; that is, it refers to the more structurally rigid definition
  of HTML.

<!ENTITY % HTML.Version

       "-//IETF//DTD HTML 2.0 Strict Level 1//EN"

       -- Typical usage:

           <!DOCTYPE HTML PUBLIC
               "-//IETF//DTD HTML Strict Level 1//EN">
           <html>
           ...
           </html>
       --
       >

<!ENTITY % HTML.Recommended "INCLUDE">

<!ENTITY % html-1 PUBLIC "-//IETF//DTD HTML 2.0 Level 1//EN"> %html-1;

Berners-Lee & Connolly Standards Track [Page 63]

RFC 1866 Hypertext Markup Language - 2.0 November 1995

9.5. SGML Declaration for HTML

  This is the SGML Declaration for HyperText Markup Language.

<!SGML "ISO 8879:1986" --

       SGML Declaration for HyperText Markup Language (HTML).

CHARSET

        BASESET  "ISO 646:1983//CHARSET
                  International Reference Version
                  (IRV)//ESC 2/5 4/0"
        DESCSET  0   9   UNUSED
                 9   2   9
                 11  2   UNUSED
                 13  1   13
                 14  18  UNUSED
                 32  95  32
                 127 1   UNUSED
    BASESET   "ISO Registration Number 100//CHARSET
               ECMA-94 Right Part of
               Latin Alphabet Nr. 1//ESC 2/13 4/1"

        DESCSET  128  32   UNUSED
                 160  96    32

CAPACITY SGMLREF

               TOTALCAP        150000
               GRPCAP          150000
               ENTCAP          150000

SCOPE DOCUMENT SYNTAX

        SHUNCHAR CONTROLS 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 31 127
        BASESET  "ISO 646:1983//CHARSET
                  International Reference Version
                  (IRV)//ESC 2/5 4/0"
        DESCSET  0 128 0
        FUNCTION
                 RE          13
                 RS          10
                 SPACE       32
                 TAB SEPCHAR  9
        NAMING   LCNMSTRT ""
                 UCNMSTRT ""

Berners-Lee & Connolly Standards Track [Page 64]

RFC 1866 Hypertext Markup Language - 2.0 November 1995

                 LCNMCHAR ".-"
                 UCNMCHAR ".-"
                 NAMECASE GENERAL YES
                          ENTITY  NO
        DELIM    GENERAL  SGMLREF
                 SHORTREF SGMLREF
        NAMES    SGMLREF
        QUANTITY SGMLREF
                 ATTSPLEN 2100
                 LITLEN   1024
                 NAMELEN  72    -- somewhat arbitrary; taken from
                               internet line length conventions --
                 PILEN    1024
                 TAGLVL   100
                 TAGLEN   2100
                 GRPGTCNT 150
                 GRPCNT   64

FEATURES

 MINIMIZE
   DATATAG  NO
   OMITTAG  YES
   RANK     NO
   SHORTTAG YES
 LINK
   SIMPLE   NO
   IMPLICIT NO
   EXPLICIT NO
 OTHER
   CONCUR   NO
   SUBDOC   NO
   FORMAL   YES
 APPINFO    "SDA"  -- conforming SGML Document Access application
                   --

9.6. Sample SGML Open Entity Catalog for HTML

  The SGML standard describes an "entity manager" as the portion or
  component of an SGML system that maps SGML entities into the actual
  storage model (e.g., the file system). The standard itself does not

   3 名前: 風吹けば名無し＠転載禁止 Mail: 投稿日: 2015/07/31(金) 15:47:39.05 ID: 0ROMJ/zc0
   >>2
   舐めってるぞ

   滑川ゥー！

   ぬめってるぞ

   ガッツリフェラなんJ民

   なんJ民ぬめぬめでわろたｗｗｗｗｗ

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諸注意

ISO 13818-3 MPEG-2 Audio Encoder - Lower Sampling Frequency Extension

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Berners-Lee & Connolly Standards Track [Page 61]

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Berners-Lee & Connolly Standards Track [Page 64]

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