/********************************************************************** * 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  jddevine@aware.com  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 mc@fivebats.com      Changed default bitrate selection so  * *                              it works with the new LSF stuff       * *10/01/95 mc@fivebats.com      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*************************************************************************/voidobtain_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;