musicin.c

MP3编码库

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

#include <stdlib.h>
#include <stdio.h>
#include <string.h>

#define WRITE_BUFFER_SIZE 32000			// must be enough to store one frame compress data
										// bit_writer will never check fullness for this buffer

#define         MAX_NAME_SIZE           81
/* Default Input Arguments (for command line control) */

#define DFLT_LAY        2      /* default encoding layer is II */
#define DFLT_MOD        's'    /* default mode is stereo */
#define DFLT_PSY        1      /* default psych model is 1 */
#define DFLT_SFQ        44.1   /* default input sampling rate is 44.1 kHz */
#define DFLT_EMP        'n'    /* default de-emphasis is none */
#define DFLT_EXT        ".mpg" /* default output file extension */

#define FILETYPE_ENCODE 'TEXT'
#define CREATOR_ENCODE  'MpgD'

#ifdef _DEBUG
#include <assert.h>
#endif
#include "./libmp3enc/libmp3enc.h"
#include "audio_in.h"


/* Global variable definitions for "musicin.c" */
FILE               *musicin;
FILE			   *music_out;
i8				   *programName;

/* The following functions are in the file "musicin.c" */
i32    parse_args();
void   print_config();
void   usage();
void   aiff_check();

/* Implementations */
i32 init_MP3codec(MP3CODEC *mp3codec, layer *input);
i32 MP3_encoding(u8 *f_buf, MP3CODEC *mp3codec);
i32 deinit_MP3codec(MP3CODEC *mp3codec, u8 *buf);

/************************************************************************
*
* parse_args
*
* PURPOSE:  Sets encoding parameters to the specifications of the
* command line.  Default settings are used for parameters
* not specified in the command line.
*
* SEMANTICS:  The command line is parsed according to the following
* syntax:
*
* -l  is followed by the layer number
* -m  is followed by the mode
* -p  is followed by the psychoacoustic model number
* -s  is followed by the sampling rate
* -b  is followed by the total bitrate, irrespective of the mode
* -d  is followed by the emphasis flag
* -c  is followed by the copyright/no_copyright flag
* -o  is followed by the original/not_original flag
* -e  is followed by the error_protection on/off flag
*
* If the input file is in AIFF format, the sampling frequency is read
* from the AIFF header.
*
* The input and output filenames are read into #inpath# and #outpath#.
*
************************************************************************/
 
i32 parse_args(
	i32     argc,
	i8		**argv,
	layer   *info,
	u32		*num_samples,
	i8		inPath[MAX_NAME_SIZE],
	i8		outPath[MAX_NAME_SIZE]
)
{
   i32   file_type;
   i32   srate;
   i32   brate;

   i32   err = 0, i = 0;
   
   IFF_AIFF pcm_aiff_data;

   long  samplerate;
   long  soundPosition;
 
   printf("\nparsing input arguments...\n");
   /* preset defaults */
   inPath[0] = '\0';   outPath[0] = '\0';
   info->lay = DFLT_LAY;
   switch(DFLT_MOD) {
      case 's': info->mode = MPG_MD_STEREO; info->mode_ext = 0; break;
      case 'd': info->mode = MPG_MD_DUAL_CHANNEL; info->mode_ext=0; break;
      case 'j': info->mode = MPG_MD_JOINT_STEREO; break;
      case 'm': info->mode = MPG_MD_MONO; info->mode_ext = 0; break;
      default:
         fprintf(stderr, "%s: Bad mode dflt %c\n", programName, DFLT_MOD);
         abort();
   }

   if((info->sampling_frequency = SmpFrqIndex((long)(1000*DFLT_SFQ), &info->version)) < 0) {
      fprintf(stderr, "%s: bad sfrq default %.2f\n", programName, DFLT_SFQ);
      abort();
   }
  info->bitrate_index = 14;  
  brate = 0;
   switch(DFLT_EMP) {
      case 'n': info->emphasis = 0; break;
      case '5': info->emphasis = 1; break;
      case 'c': info->emphasis = 3; break;
      default: 
         fprintf(stderr, "%s: Bad emph dflt %c\n", programName, DFLT_EMP);
         abort();
   }
   info->copyright = 0; info->original = 0; info->error_protection = FALSE;
 
   /* process args */
   while(++i<argc && err == 0) {
      i8 c, *token, *arg, *nextArg;
      i32  argUsed;
 
      token = argv[i];
      if(*token++ == '-') {
         if(i+1 < argc) nextArg = argv[i+1];
         else           nextArg = "";
         argUsed = 0;
         while( (c = *token++) ) {

            if(*token /* NumericQ(token) */) 
				arg = token;
            else
				arg = nextArg;

            switch(c) {
               case 'l':
				  info->lay = atoi(arg); argUsed = 1;
                  if(info->lay<1 || info->lay>3) {
                     fprintf(stderr,"%s: -l layer must be 1, 2, or 3, not %s\n",
                          programName, arg);
                     err = 1;
                  }
                  break;
               case 'm':
				  argUsed = 1;
                  if (*arg == 's')
                    { info->mode = MPG_MD_STEREO; info->mode_ext = 0; }
                  else if (*arg == 'd')
                    { info->mode = MPG_MD_DUAL_CHANNEL; info->mode_ext=0; }
                  else if (*arg == 'j')
                    { info->mode = MPG_MD_JOINT_STEREO; }
                  else if (*arg == 'm')
                    { info->mode = MPG_MD_MONO; info->mode_ext = 0; }
                  else {
                    fprintf(stderr,"%s: -m mode must be s/d/j/m not %s\n",
                            programName, arg);
                    err = 1;
                  }
                  break;


               case 's':
                  argUsed = 1;
                  srate = atoi( arg );
                  /* samplerate = rint( 1000.0 * srate ); $A  */
                  samplerate =  100 * srate ;
                  if( (info->sampling_frequency =
						SmpFrqIndex((long) samplerate, &info->version)) < 0 )
                      err = 1;
                  break;
                  
               case 'b':        
				  argUsed = 1;
				  brate = atoi(arg); 
				  break;

               case 'd':
				  argUsed = 1;
                  if (*arg == 'n')                    info->emphasis = 0;
                  else if (*arg == '5')               info->emphasis = 1;
                  else if (*arg == 'c')               info->emphasis = 3;
                  else {
                     fprintf(stderr,"%s: -d emp must be n/5/c not %s\n",
                             programName, arg);
                     err = 1;
                  }
                  break;
                case 'c':       info->copyright = 1; break;
                case 'o':       info->original  = 1; break;
                case 'e':       info->error_protection = TRUE; break;
                default:        fprintf(stderr,"%s: unrec option %c\n",
                                        programName, c);
                                err = 1; break;
            }
            if(argUsed) {
               if(arg == token)    token = "";   /* no more from token */
               else                ++i;          /* skip arg we used */
               arg = ""; argUsed = 0;
            }
         }
      }
      else {
         if(inPath[0] == '\0')       strcpy(inPath, argv[i]);
         else if(outPath[0] == '\0') strcpy(outPath, argv[i]);
         else {
            fprintf(stderr,"%s: excess arg %s\n", programName, argv[i]);
            err = 1;
         }
      }
   }

   if(err || inPath[0] == '\0') usage();  /* never returns */
 
   if(outPath[0] == '\0') {
      strcpy(outPath, inPath);
      strcat(outPath, DFLT_EXT);
   }

   if ((musicin = fopen(inPath, "rb")) == NULL) {
      printf("Could not find \"%s\".\n", inPath);
      exit(1);