w~1

mpeg layerI II III, support window and linux

/**********************************************************************
 * ISO MPEG Audio Subgroup Software Simulation Group (1996)
 * ISO 13818-3 MPEG-2 Audio Decoder - Lower Sampling Frequency Extension
 *
 * $Id: musicout.c.rca 1.2 Fri Dec 13 00:30:54 2002 franklej Experimental $
 *
 * Received from FhG
 **********************************************************************/
/**********************************************************************
 *   date   programmers                comment                        *
 * 2/25/91  Douglas Wong        start of version 1.0 records          *
 * 3/06/91  Douglas Wong        rename setup.h to dedef.h             *
 *                              removed extraneous variables          *
 *                              removed window_samples (now part of   *
 *                              filter_samples)                       *
 * 3/07/91  Davis Pan           changed output file to "codmusic"     *
 * 5/10/91  Vish (PRISM)        Ported to Macintosh and Unix.         *
 *                              Incorporated new "out_fifo()" which   *
 *                              writes out last incomplete buffer.    *
 *                              Incorporated all AIFF routines which  *
 *                              are also compatible with SUN.         *
 *                              Incorporated user interface for       *
 *                              specifying sound file names.          *
 *                              Also incorporated user interface for  *
 *                              writing AIFF compatible sound files.  *
 * 27jun91  dpwe (Aware)        Added musicout and &sample_frames as  *
 *                              args to out_fifo (were glob refs).    *
 *                              Used new 'frame_params' struct.       *
 *                              Clean,simplify, track clipped output  *
 *                              and total bits/frame received.        *
 * 7/10/91  Earle Jennings      changed to floats to FLOAT            *
 *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 functions are        *
 *                              buffer_CRC and recover_CRC_error      *
 *                              Additions and revisions are marked    *
 *                              with "dhl" for clarity                *
 * 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.  *
 *19 aug 92 Soren H. Nielsen    Changed MS-DOS file name extensions.  *
 * 8/27/93 Seymour Shlien,      Fixes in Unix and MSDOS ports,        *
 *         Daniel Lauzon, and                                         *
 *         Bill Truerniet                                             *
 *--------------------------------------------------------------------*
 * 4/23/92  J. Pineda           Added code for layer III.  LayerIII   *
 *          Amit Gulati         decoding is currently performed in    *
 *                              two-passes for ease of sideinfo and   *
 *                              maindata buffering and decoding.      *
 *                              The second (computation) pass is      *
 *                              activated with "decode -3 <outfile>"  *
 * 10/25/92 Amit Gulati         Modified usage() for layerIII         *
 * 12/10/92 Amit Gulati         Changed processing order of re-order- *
 *                              -ing step.  Fixed adjustment of       *
 *                              main_data_end pointer to exclude      *
 *                              side information.                     *
 *  9/07/93 Toshiyuki Ishino    Integrated Layer III with Ver 3.9.    *
 *--------------------------------------------------------------------*
 * 11/20/93 Masahiro Iwadare    Integrated Layer III with Ver 4.0.    *
 *--------------------------------------------------------------------*
 *  7/14/94 Juergen Koller      Bug fixes in Layer III code           *
 *--------------------------------------------------------------------*
 * 08/11/94 IIS                 Bug fixes in Layer III code           *
 *--------------------------------------------------------------------*
 * 11/04/94 Jon Rowlands        Prototype fixes                       *
 *--------------------------------------------------------------------*
 *  7/12/95 Soeren H. Nielsen   Changes for LSF Layer I and II        *
 *--------------------------------------------------------------------*
 *  7/14/94 Juergen Koller      Bug fixes in Layer III code           *
 *--------------------------------------------------------------------*
 *     8/95 Roland Bitto        addapdet to MPEG 2                    *
 *--------------------------------------------------------------------*
 * 11/22/95 Heiko Purnhagen     skip ancillary data in bitstream      *
 **********************************************************************/

#include        "common.h"
#include        "decoder.h"

/********************************************************************
/*
/*        This part contains the MPEG I decoder for Layers I & II.
/*
/*********************************************************************/

/****************************************************************
/*
/*        For MS-DOS user (Turbo c) change all instance of malloc
/*        to _farmalloc and free to _farfree. Compiler model hugh
/*        Also make sure all the pointer specified are changed to far.
/*
/*****************************************************************/
/* local functions definition */

static void usage();
static void GetArguments();
  

/*********************************************************************
/*
/* Core of the Layer II decoder.  Default layer is Layer II.
/*
/*********************************************************************/

/* Global variable definitions for "musicout.c" */

char *programName;
int main_data_slots();
int side_info_slots();

/* Implementations */

main(argc, argv)
int argc;
char **argv;
{
/*typedef short PCM[2][3][SBLIMIT];*/
typedef OUTPUT_TYPE PCM[2][SSLIMIT][SBLIMIT];
    PCM FAR *pcm_sample;
typedef unsigned int SAM[2][3][SBLIMIT];
    SAM FAR *sample;
typedef double FRA[2][3][SBLIMIT];
    FRA FAR *fraction;
typedef double VE[2][HAN_SIZE];
    VE FAR *w;

    Bit_stream_struc  bs;
    frame_params      fr_ps;
    layer             info;
    FILE              *musicout;
    unsigned long     sample_frames;

    int               i, j, k, stereo, done=FALSE, clip, sync; 
    int               error_protection, crc_error_count, total_error_count;
    unsigned int      old_crc, new_crc;
    unsigned int      bit_alloc[2][SBLIMIT], scfsi[2][SBLIMIT],
                      scale_index[2][3][SBLIMIT];
    unsigned long     bitsPerSlot, samplesPerFrame, frameNum = 0;
    unsigned long     frameBits, gotBits = 0;
    IFF_AIFF          pcm_aiff_data;
    Arguments_t       Arguments;
    int Max_gr;

    int bbq;

III_scalefac_t III_scalefac;
III_side_info_t III_side_info;

#ifdef  MACINTOSH
    console_options.nrows = MAC_WINDOW_SIZE;
    argc = ccommand(&argv);
#endif

    /* Most large variables are declared dynamically to ensure
       compatibility with smaller machines */

    pcm_sample = (PCM FAR *) mem_alloc((long) sizeof(PCM), "PCM Samp");
    sample = (SAM FAR *) mem_alloc((long) sizeof(SAM), "Sample");
    fraction = (FRA FAR *) mem_alloc((long) sizeof(FRA), "fraction");
    w = (VE FAR *) mem_alloc((long) sizeof(VE), "w");

    fr_ps.header = &info;
    fr_ps.tab_num = -1;                /* no table loaded */
    fr_ps.alloc = NULL;
    for (i=0;i<HAN_SIZE;i++) for (j=0;j<2;j++) (*w)[j][i] = 0.0;

    Arguments.topSb = 0;
    GetArguments(argc, argv, &Arguments);
    if ((musicout = fopen(Arguments.decoded_file_name, "w+b")) == NULL) {
          printf ("Could not create \"%s\".\n", Arguments.decoded_file_name);
          exit(1);
        }
    open_bit_stream_r(&bs, Arguments.encoded_file_name, BUFFER_SIZE);

    if (Arguments.need_aiff)
       if (aiff_seek_to_sound_data(musicout) == -1) {
          printf("Could not seek to PCM sound data in \"%s\".\n",
                 Arguments.decoded_file_name);
          exit(1);
       }

    sample_frames = 0;

    //while (!end_bs(&bs)) {
    for( bbq=0;bbq<10;bbq++) {

       sync = seek_sync(&bs, SYNC_WORD, SYNC_WORD_LNGTH);
       frameBits = sstell(&bs) - gotBits;
       if(frameNum > 0)        /* don't want to print on 1st loop; no lay */
          if(frameBits%bitsPerSlot)
             fprintf(stderr,"Got %ld bits = %ld slots plus %ld\n",
                     frameBits, frameBits/bitsPerSlot, frameBits%bitsPerSlot);
       gotBits += frameBits;

       if (!sync) {
          printf("Frame cannot be located\n");
          printf("Input stream may be empty\n");
          done = TRUE;
          /* finally write out the buffer */
          if (info.lay != 1) out_fifo(*pcm_sample, 3, &fr_ps, done,
                                      musicout, &sample_frames);
          else               out_fifo(*pcm_sample, 1, &fr_ps, done,
                                      musicout, &sample_frames);
          break;
       }

       decode_info(&bs, &fr_ps);
       hdr_to_frps(&fr_ps);
       stereo = fr_ps.stereo;
       if(fr_ps.header->version == MPEG_PHASE2_LSF) {
		Max_gr = 1;
       }
       else
       { 
		Max_gr = 2;
       }

       error_protection = info.error_protection;
       crc_error_count = 0;
       total_error_count = 0;
       if(frameNum == 0) WriteHdr(&fr_ps, stdout);  /* printout layer/mode */

#ifdef ESPS
if (frameNum == 0 && Arguments.need_esps) {
	esps_write_header(musicout,(long) sample_frames, (double)
		s_freq[info.version][info.sampling_frequency] * 1000,
		(int) stereo, Arguments.decoded_file_name );
} /* MI */
#endif

       fprintf(stderr, "{%4lu}", frameNum++); fflush(stderr); 
       if (error_protection) buffer_CRC(&bs, &old_crc);

       switch (info.lay) {

          case 1: {
             bitsPerSlot = 32;        samplesPerFrame = 384;
             I_decode_bitalloc(&bs,bit_alloc,&fr_ps);
             I_decode_scale(&bs, bit_alloc, scale_index, &fr_ps);

             if (error_protection) {
                I_CRC_calc(&fr_ps, bit_alloc, &new_crc);
                if (new_crc != old_crc) {
                   crc_error_count++;
                   total_error_count++;
                   recover_CRC_error(*pcm_sample, crc_error_count,
                                     &fr_ps, musicout, &sample_frames);
                   break;
                }
                else crc_error_count = 0;
             }

             clip = 0;
             for (i=0;i<SCALE_BLOCK;i++) {
                I_buffer_sample(&bs,(*sample),bit_alloc,&fr_ps);
                I_dequantize_sample(*sample,*fraction,bit_alloc,&fr_ps);
                I_denormalize_sample((*fraction),scale_index,&fr_ps);

                if(Arguments.topSb>0)        /* clear channels to 0 */
                   for(j=Arguments.topSb; j<fr_ps.sblimit; ++j)
                      for(k=0; k<stereo; ++k)
                         (*fraction)[k][0][j] = 0;

                for (j=0;j<stereo;j++) {
                   clip += SubBandSynthesis (&((*fraction)[j][0][0]), j,
                                             &((*pcm_sample)[j][0][0]));
                }
                out_fifo(*pcm_sample, 1, &fr_ps, done,
                         musicout, &sample_frames);
             }
             if(clip > 0) printf("%d output samples clipped\n", clip);
             break;
          }

          case 2: {
             bitsPerSlot = 8;        samplesPerFrame = 1152;
             II_decode_bitalloc(&bs, bit_alloc, &fr_ps);
             II_decode_scale(&bs, scfsi, bit_alloc, scale_index, &fr_ps);

             if (error_protection) { 
                II_CRC_calc(&fr_ps, bit_alloc, scfsi, &new_crc);
                if (new_crc != old_crc) {
                   crc_error_count++;
                   total_error_count++;
                   recover_CRC_error(*pcm_sample, crc_error_count,
                                     &fr_ps, musicout, &sample_frames);
                   break;
                }
                else crc_error_count = 0;
             }

             clip = 0;
             for (i=0;i<SCALE_BLOCK;i++) {
                II_buffer_sample(&bs,(*sample),bit_alloc,&fr_ps);
                II_dequantize_sample((*sample),bit_alloc,(*fraction),&fr_ps);
                II_denormalize_sample((*fraction),scale_index,&fr_ps,i>>2);

                if(Arguments.topSb>0)        /* debug : clear channels to 0 */
                   for(j=Arguments.topSb; j<fr_ps.sblimit; ++j)
                      for(k=0; k<stereo; ++k)
                         (*fraction)[k][0][j] =
                         (*fraction)[k][1][j] =
                         (*fraction)[k][2][j] = 0;

                for (j=0;j<3;j++) for (k=0;k<stereo;k++) {
                   clip += SubBandSynthesis (&((*fraction)[k][j][0]), k,
                                             &((*pcm_sample)[k][j][0]));
                }
                out_fifo(*pcm_sample, 3, &fr_ps, done, musicout,