musicout.c

mpeg layerI II III, support window and linux

       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;

    for( bbq=0;bbq<20;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;
#ifdef  DEBUG_PRINT_XING
#ifdef  DEBUG_SCALEFAC
      fprintf(fp_scalefac,"******************************\n");
      fprintf(fp_scalefac,"decode_fram = %d\n",bbq);
#endif
#ifdef  DEBUG_HUFFMAN
      fprintf(fp_huffman,"totbit = %d\n",hsstell());
      fprintf(fp_huffman,"******************************\n");
      fprintf(fp_huffman,"decode_fram = %d\n",bbq);
#endif 
#ifdef  DEBUG_REQUAN
      fprintf(fp_requan,"******************************\n");
      fprintf(fp_requan,"decode_fram = %d\n",bbq);
#endif 
#ifdef  CAPT_CODE_BOOK
      fprintf(fp_capt_codebook,"******************************\n");
      fprintf(fp_capt_codebook,"decode_fram = %d\n",bbq);
#endif       
#endif 
       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,
                         &sample_frames);
             }
             if(clip > 0) printf("%d samples clipped\n", clip);
             break;
          }

          case 3: {
             int nSlots;
             int gr, ch, ss, sb, main_data_end, flush_main ;
	     int  bytes_to_discard ;
	     static int frame_start = 0;

             bitsPerSlot = 8;        
             if(fr_ps.header->version == MPEG_PHASE2_LSF)
		samplesPerFrame = 576;
	     else
		samplesPerFrame = 1152;
 
             III_get_side_info(&bs, &III_side_info, &fr_ps);
             nSlots = main_data_slots(fr_ps);
             for (; nSlots > 0; nSlots--)  /* read main data. */
                hputbuf((unsigned int) getbits(&bs,8), 8);
	     main_data_end = hsstell() / 8; /*of privious frame*/
             if ( flush_main=(hsstell() % bitsPerSlot) ) { 
                hgetbits((int)(bitsPerSlot - flush_main));
		main_data_end ++;
	     }
             bytes_to_discard = frame_start - main_data_end
 			            - III_side_info.main_data_begin ;
#ifdef DEBUG_MAIN
             printf("bytes_to_discard = %d\n",bytes_to_discard);
#endif 
             if( main_data_end > 4096 )
             {   frame_start -= 4096;
                 rewindNbytes( 4096 );
             }

             frame_start += main_data_slots(fr_ps);
#ifdef DEBUG_MAIN 
             printf("frame_start = %d\n",frame_start);
#endif             
             if (bytes_to_discard < 0) {
         printf("Not enough main data to decode frame %d.  Frame discarded.\n", 
                        frameNum - 1); break;
             }
             for (; bytes_to_discard > 0; bytes_to_discard--) hgetbits(8);

             clip = 0;
             for (gr=0;gr<Max_gr;gr++) {
               double lr[2][SBLIMIT][SSLIMIT],ro[2][SBLIMIT][SSLIMIT];

               for (ch=0; ch<stereo; ch++) {
                 long int is[SBLIMIT][SSLIMIT];   /* Quantized samples. */
                 int part2_start;
                 part2_start = hsstell();
#ifdef DEBUG_MAIN
                 printf("gr = %d, ch = %d,part2_start = %d\n",gr,ch,part2_start);
#endif
                 if(fr_ps.header->version != MPEG_PHASE2_LSF)
                 {
                    III_get_scale_factors(&III_scalefac,&III_side_info,gr,ch,
			   &fr_ps,fp_scalefac);
                 }
                 else
                 {
                    III_get_LSF_scale_factors(&III_scalefac,
                                                    &III_side_info,
                                                    gr,ch,&fr_ps);
                 }

#ifdef CAPT_BS 
                huffman_start = hsstell();
                bit_num = 0;
                for (i=huffman_start; i<huffman_start+2800; i++)
                {
                    fprintf(fp_capt_bs,"%x",hget1bit() & 0x1);
                    bit_num ++;
                    if (bit_num == 8)
                    {
                        fprintf(fp_capt_bs,"\n");
                        bit_num = 0;
                    } 
                    if (i == (huffman_start+2800 - 1))
                    {
                        exit(0);
                    }
                }
#endif 
                 
                 III_hufman_decode(is, &III_side_info, ch, gr, part2_start,
                                   &fr_ps,fp_huffman,fp_capt_codebook);
                 III_dequantize_sample(is, ro[ch], &III_scalefac,
                                   &(III_side_info.ch[ch].gr[gr]), ch, &fr_ps,fp_requan);
               }
               III_stereo(ro,lr,&III_scalefac,
                            &(III_side_info.ch[0].gr[gr]), &fr_ps,fp_stereo);

               for (ch=0; ch<stereo; ch++) {
                    double re[SBLIMIT][SSLIMIT];
                    double hybridIn[SBLIMIT][SSLIMIT];/* Hybrid filter input */
                    double hybridOut[SBLIMIT][SSLIMIT];/* Hybrid filter out */
                    double polyPhaseIn[SBLIMIT];     /* PolyPhase Input. */

                    III_reorder (lr[ch],re,&(III_side_info.ch[ch].gr[gr]),
                                  &fr_ps,fp_reorder);

                    III_antialias(re, hybridIn, /* Antialias butterflies. */
                                  &(III_side_info.ch[ch].gr[gr]), &fr_ps,fp_antialias);