musicin.c

ISO mp3 sources (distribution 10) Layer 1/2/3, C Source, 512 k Sources of the Mpeg 1,2 layer 1,2

/**********************************************************************
 * ISO MPEG Audio Subgroup Software Simulation Group (1996)
 * ISO 13818-3 MPEG-2 Audio Encoder - Lower Sampling Frequency Extension
 *
 * $Id: musicin.c,v 1.2 1997/01/19 22:28:29 rowlands Exp $
 *
 * $Log: musicin.c,v $
 * Revision 1.2  1997/01/19 22:28:29  rowlands
 * Layer 3 bug fixes from Seymour Shlien
 *
 * 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            *
 *                              removed extraneous variables          *
 * 3/21/91  J.Georges Fritsch   introduction of the bit-stream        *
 *                              package. This package allows you      *
 *                              to generate the bit-stream in a       *
 *                              binary or ascii format                *
 * 3/31/91  Bill Aspromonte     replaced the read of the SB matrix    *
 *                              by an "code generated" one            *
 * 5/10/91  W. Joseph Carter    Ported to Macintosh and Unix.         *
 *                              Incorporated Jean-Georges Fritsch's   *
 *                              "bitstream.c" package.                *
 *                              Modified to strictly adhere to        *
 *                              encoded bitstream specs, including    *
 *                              "Berlin changes".                     *
 *                              Modified user interface dialog & code *
 *                              to accept any input & output          *
 *                              filenames desired.  Also added        *
 *                              de-emphasis prompt and final bail-out *
 *                              opportunity before encoding.          *
 *                              Added AIFF PCM sound file reading     *
 *                              capability.                           *
 *                              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.           *
 * 27jun91  dpwe (Aware Inc)    Used new frame_params struct.         *
 *                              Clear all automatic arrays.           *
 *                              Changed some variable names,          *
 *                              simplified some code.                 *
 *                              Track number of bits actually sent.   *
 *                              Fixed padding slot, stereo bitrate    *
 *                              Added joint-stereo : scales L+R.      *
 * 6/12/91  Earle Jennings      added fix for MS_DOS in obtain_param  *
 * 6/13/91  Earle Jennings      added stack length adjustment before  *
 *                              main for MS_DOS                       *
 * 7/10/91  Earle Jennings      conversion of all float to FLOAT      *
 *                              port to MsDos from MacIntosh completed*
 * 8/ 8/91  Jens Spille         Change for MS-C6.00                   *
 * 8/22/91  Jens Spille         new obtain_parameters()               *
 *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        *
 *                              I_CRC_calc, II_CRC_calc and encode_CRC*
 *                              Additions and revisions are marked    *
 *                              with "dhl" for clarity                *
 *11/11/91 Katherine Wang       Documentation of code.                *
 *                                (variables in documentation are     *
 *                                surround by the # symbol, and an '*'*
 *                                denotes layer I or II versions)     *
 * 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.  *
 * 2/25/92  Masahiro Iwadare    made code cleaner and more consistent *
 * 8/07/92  Mike Coleman        make exit() codes return error status *
 *                              made slight changes for portability   *
 *19 aug 92 Soren H. Nielsen    Changed MS-DOS file name extensions.  *
 * 8/25/92  Shaun Astarabadi    Replaced rint() function with explicit*
 *                              rounding for portability with MSDOS.  *
 * 9/22/92  [e-mail]  Fixed _scale_factor_calc() calls.     *
 *10/19/92  Masahiro Iwadare    added info->mode and info->mode_ext   *
 *                              updates for AIFF format files         *
 * 3/10/93  Kevin Peterson      In parse_args, only set non default   *
 *                              bit rate if specified in arg list.    *
 *                              Use return value from aiff_read_hdrs  *
 *                              to fseek to start of sound data       *
 * 7/26/93  Davis Pan           fixed bug in printing info->mode_ext  *
 *                              value for joint stereo condition      *
 * 8/27/93 Seymour Shlien,      Fixes in Unix and MSDOS ports,        *
 *         Daniel Lauzon, and                                         *
 *         Bill Truerniet                                             *
 * 11/7/95 Soeren H. Nielsen    LSF added. Bug fix in MSDOS ext.      *
 * 8/02/95 [e-mail]      Changed default bitrate selection so  *
 *                              it works with the new LSF stuff       *
 *10/01/95 [e-mail]      Added layer3                          *
 **********************************************************************/

#ifdef MS_DOS
#include <dos.h>
#endif
#include <stdlib.h>
#include "common.h"
#include "encoder.h"
#include "l3psy.h"
#include "mdct.h"
#include "loop.h"
#include "l3bitstream.h"
#include <assert.h>

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

FILE               *musicin;
Bit_stream_struc   bs;
char               *programName;

/* Implementations */

/************************************************************************
*
* obtain_parameters
*
* PURPOSE:  Prompts for and reads user input for encoding parameters
*
* SEMANTICS:  The parameters read are:
* - input and output filenames
* - sampling frequency (if AIFF file, will read from the AIFF file header)
* - layer number
* - mode (stereo, joint stereo, dual channel or mono)
* - psychoacoustic model (I or II)
* - total bitrate, irrespective of the mode
* - de-emphasis, error protection, copyright and original or copy flags
*
************************************************************************/

void
obtain_parameters(fr_ps,psy,num_samples,original_file_name,encoded_file_name)
frame_params    *fr_ps;
int             *psy;
unsigned long   *num_samples;
char            original_file_name[MAX_NAME_SIZE];
char            encoded_file_name[MAX_NAME_SIZE];
{
    int j;
    long int freq;
    int model, brt;
    char t[50];
    IFF_AIFF pcm_aiff_data;
    layer *info = fr_ps->header;
    long soundPosition;
#ifdef  MS_DOS
    char temp_str[MAX_NAME_SIZE];
#endif 

    do  {
       printf("Enter PCM input file name <required>: ");
       gets(original_file_name);
       if (original_file_name[0] == NULL_CHAR)
       printf("PCM input file name is required.\n");
    } while (original_file_name[0] == NULL_CHAR);
    printf(">>> PCM input file name is: %s\n", original_file_name);
 
    if ((musicin = fopen(original_file_name, "rb")) == NULL) {
       printf("Could not find \"%s\".\n", original_file_name);
       exit(1);
    }

#ifdef  MS_DOS
    /* replace old extension with new one, 1992-08-19, 1995-06-12 shn */
    new_ext(original_file_name, DFLT_EXT, temp_str);
    printf("Enter MPEG encoded output file name <%s>: ",
           temp_str);
#else
    printf("Enter MPEG encoded output file name <%s%s>: ",
           original_file_name, DFLT_EXT);
#endif
 
    gets(encoded_file_name);
    
    if (encoded_file_name[0] == NULL_CHAR) {
#ifdef  MS_DOS
  strcpy(encoded_file_name, temp_str);
#else
        strcat(strcpy(encoded_file_name, original_file_name), DFLT_EXT);
#endif
    }
        

    printf(">>> MPEG encoded output file name is: %s\n", encoded_file_name);
 
    open_bit_stream_w(&bs, encoded_file_name, BUFFER_SIZE);
 
    if ((soundPosition = aiff_read_headers(musicin, &pcm_aiff_data)) != -1) {

       printf(">>> Using Audio IFF sound file headers\n");

       aiff_check(original_file_name, &pcm_aiff_data, &info->version);

       if (fseek(musicin, soundPosition, SEEK_SET) != 0) {
          printf("Could not seek to PCM sound data in \"%s\".\n",
                 original_file_name);
          exit(1);
       }

       info->sampling_frequency =
      SmpFrqIndex((long)pcm_aiff_data.sampleRate, &info->version);
       printf(">>> %f Hz sampling frequency selected\n",
              pcm_aiff_data.sampleRate);

       /* Determine number of samples in sound file */
#ifndef MS_DOS
       *num_samples = pcm_aiff_data.numChannels *
                      pcm_aiff_data.numSampleFrames;
#else
       *num_samples = (long)(pcm_aiff_data.numChannels) *
                      (long)(pcm_aiff_data.numSampleFrames);
#endif

    }
    else {    /* Not using Audio IFF sound file headers. */

       printf("What is the sampling frequency? <44100>[Hz]: ");
       gets(t);
       freq = atol(t);
       switch (freq) {
          case 48000 : info->sampling_frequency = 1;
              printf(">>> %ld Hz sampling freq selected\n", freq);
              break;
          case 44100 : info->sampling_frequency = 0;
              printf(">>> %ld Hz sampling freq selected\n", freq);
              break;
          case 32000 : info->sampling_frequency = 2;
              printf(">>> %ld Hz sampling freq selected\n", freq);
              break;
          case 24000 : info->sampling_frequency = 1;
        info->version = MPEG_PHASE2_LSF;
              printf(">>> %ld Hz sampling freq selected\n", freq);
              break;
          case 22050 : info->sampling_frequency = 0;
        info->version = MPEG_PHASE2_LSF;
              printf(">>> %ld Hz sampling freq selected\n", freq);
              break;
          case 16000 : info->sampling_frequency = 2;
        info->version = MPEG_PHASE2_LSF;
              printf(">>> %ld Hz sampling freq selected\n", freq);
              break;
          default:    info->sampling_frequency = 0;
              printf(">>> Default 44.1 kHz samp freq selected\n");
       }
       printf(">>> Encoding algorithm is %s\n", version_names[info->version]);
       if (fseek(musicin, 0, SEEK_SET) != 0) {
          printf("Could not seek to PCM sound data in \"%s\".\n",
                  original_file_name);
          exit(1);
       }
 
       /* Declare sound file to have "infinite" number of samples. */
       *num_samples = MAX_U_32_NUM;

    }

    printf("Which layer do you want to use?\n");
    printf("Available: Layer (1), Layer (<2>), Layer (3): ");
    gets(t);
    switch(*t){
       case '1': info->lay = 1; printf(">>> Using Layer %s\n",t); break;
       case '2': info->lay = 2; printf(">>> Using Layer %s\n",t); break;
     case '3': info->lay = 3; printf(">>> Using Layer %s\n",t); break;
       default:  info->lay = 2; printf(">>> Using default Layer 2\n"); break;
    }

    printf("Which mode do you want?\n");
    printf("Available: (<s>)tereo, (j)oint stereo, ");
    printf("(d)ual channel, s(i)ngle Channel: ");
    gets(t);
    switch(*t){
       case 's':
       case 'S':
          info->mode = MPG_MD_STEREO; info->mode_ext = 0;
          printf(">>> Using mode %s\n",t);
          break;
       case 'j':
       case 'J':
          info->mode = MPG_MD_JOINT_STEREO;
          printf(">>> Using mode %s\n",t);
          break;
       case 'd':
       case 'D':
          info->mode = MPG_MD_DUAL_CHANNEL; info->mode_ext = 0;
          printf(">>> Using mode %s\n",t);
          break;
       case 'i':
       case 'I':
          info->mode = MPG_MD_MONO; info->mode_ext = 0;
          printf(">>> Using mode %s\n",t);
          break;
       default:
          info->mode = MPG_MD_STEREO; info->mode_ext = 0;
          printf(">>> Using default stereo mode\n");
          break;
    }

    printf("Which psychoacoustic model do you want to use? <1>: ");
    gets(t);
    model = atoi(t);
    if (model > 2 || model < 1) {
       printf(">>> Default model 1 selected\n");
       *psy = 1;
    }
    else {
       *psy = model;
       printf(">>> Using psychoacoustic model %d\n", model);
    }
    
    /* set default bitrate to highest allowed, which is index 14 */
  brt = bitrate[info->version][info->lay-1][14];
    printf( "What is the total bitrate? <%u>[kbps]: ", brt );
    gets( t );
    brt = atoi( t );
    if ( brt == 0 )
      j = 15;
    else
      j = 0;
    while ( j < 15 )
    {
    if ( bitrate[info->version][info->lay-1][j] == brt )
          break;
    j++;
    }
    if ( j == 15 )
    {
    brt = bitrate[info->version][info->lay-1][14];
        printf( ">>> Using default %u kbps\n", brt );
        info->bitrate_index = 14;
    }
    else
    {
       info->bitrate_index = j;
       printf( ">>> Bitrate = %d kbps\n", bitrate[info->version][info->lay-1][j] );
    }
 
    printf("What type of de-emphasis should the decoder use?\n");
    printf("Available: (<n>)one, (5)0/15 microseconds, (c)citt j.17: ");
    gets(t);
    if (*t != 'n' && *t != '5' && *t != 'c') {
       printf(">>> Using default no de-emphasis\n");
       info->emphasis = 0;
    }
    else {
       if (*t == 'n')      info->emphasis = 0;
       else if (*t == '5') info->emphasis = 1;
       else if (*t == 'c') info->emphasis = 3;
       printf(">>> Using de-emphasis %s\n",t);
    }
 
/*  Start 2. Part changes for CD Ver 3.2; jsp; 22-Aug-1991 */
 
    printf("Do you want to set the private bit? (y/<n>): ");
    gets(t);
    if (*t == 'y' || *t == 'Y') info->extension = 1;
    else                        info->extension = 0;
    if(info->extension) printf(">>> Private bit set\n");
    else                printf(">>> Private bit not set\n");
 
/*  End changes for CD Ver 3.2; jsp; 22-Aug-1991 */
 
    printf("Do you want error protection? (y/<n>): ");
    gets(t);
    if (*t == 'y' || *t == 'Y') info->error_protection = TRUE;
    else                        info->error_protection = FALSE;
    if(info->error_protection) printf(">>> Error protection used\n");
    else printf(">>> Error protection not used\n");
 
    printf("Is the material copyrighted? (y/<n>): ");
    gets(t);
    if (*t == 'y' || *t == 'Y') info->copyright = 1;
    else                        info->copyright = 0;