MUSICOUT.C
资源名称:ampegsrc.zip [点击查看]
上传用户:dshsh2009
上传日期:2007-01-07
资源大小:155k
文件大小:24k
源码类别:
mpeg/mp3
开发平台:
Unix_Linux
- /**********************************************************************
- Copyright (c) 1991 MPEG/audio software simulation group, All Rights Reserved
- musicout.c
- **********************************************************************/
- /**********************************************************************
- * MPEG/audio coding/decoding software, work in progress *
- * NOT for public distribution until verified and approved by the *
- * MPEG/audio committee. For further information, please contact *
- * Davis Pan, 708-538-5671, e-mail: pan@ukraine.corp.mot.com *
- * *
- * VERSION 4.3 *
- * changes made since last update: *
- * 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 *
- **********************************************************************/
- #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.
- /*
- /*****************************************************************/
- /*********************************************************************
- /*
- /* 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 short 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, x, 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;
- char encoded_file_name[MAX_NAME_SIZE];
- char decoded_file_name[MAX_NAME_SIZE];
- char default_file_name[MAX_NAME_SIZE];
- char t[50];
- int need_aiff;
- int need_esps; /* MI */
- int topSb = 0;
- 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;
- programName = argv[0];
- if(argc==1) { /* no command line args -> interact */
- do {
- printf ("Enter encoded file name <required>: ");
- gets (encoded_file_name);
- if (encoded_file_name[0] == NULL_CHAR)
- printf ("Encoded file name is required. n");
- } while (encoded_file_name[0] == NULL_CHAR);
- printf (">>> Encoded file name is: %s n", encoded_file_name);
- x=0;
- while (x <= MAX_NAME_SIZE)
- {
- default_file_name[x] = NULL_CHAR;
- ++x;
- }
- x=0;
- while (x <= 8)
- {
- default_file_name[x] = encoded_file_name[x];
- if (encoded_file_name[++x] == '.')
- x = 9;
- }
- strcat(default_file_name,DFLT_OPEXT);
- printf("Enter MPEG decoded output file name <%s>: ",
- default_file_name); /* 92-08-19 shn */
- gets(decoded_file_name);
- if (decoded_file_name[0] == NULL_CHAR) {
- /* replace old extension with new one, 92-08-19 shn */
- strcpy(decoded_file_name,default_file_name);
- }
- printf (">>> MPEG decoded file name is: %s n", decoded_file_name);
- printf(
- "Do you wish to write an AIFF compatible sound file ? (y/<n>) : ");
- gets(t);
- if (*t == 'y' || *t == 'Y') need_aiff = TRUE;
- else need_aiff = FALSE;
- if (need_aiff)
- printf(">>> An AIFF compatible sound file will be writtenn");
- else printf(">>> A non-headered PCM sound file will be writtenn");
- printf(
- "Do you wish to exit (last chance before decoding) ? (y/<n>) : ");
- gets(t);
- if (*t == 'y' || *t == 'Y') exit(0);
- }
- else { /* interpret CL Args */
- int i=0, err=0;
- need_aiff = FALSE;
- need_esps = FALSE; /* MI */
- encoded_file_name[0] = ' ';
- decoded_file_name[0] = ' ';
- while(++i<argc && err == 0) {
- char c, *token, *arg, *nextArg;
- int 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 's': topSb = atoi(arg); argUsed = 1;
- if(topSb<1 || topSb>SBLIMIT) {
- fprintf(stderr, "%s: -s band %s not %d..%dn",
- programName, arg, 1, SBLIMIT);
- err = 1;
- }
- break;
- case 'A': need_aiff = TRUE; break;
- case 'E': need_esps = TRUE; break; /* MI */
- default: fprintf(stderr,"%s: unrecognized option %cn",
- 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(encoded_file_name[0] == ' ')
- strcpy(encoded_file_name, argv[i]);
- else
- if(decoded_file_name[0] == ' ')
- strcpy(decoded_file_name, argv[i]);
- else {
- fprintf(stderr,
- "%s: excess arg %sn", programName, argv[i]);
- err = 1;
- }
- }
- }
- if(err || encoded_file_name[0] == ' ') usage(); /* never returns */
- if(decoded_file_name[0] == ' ') {
- strcpy(decoded_file_name, encoded_file_name);
- strcat(decoded_file_name, DFLT_OPEXT);
- }
- }
- /* report results of dialog / command line */
- printf("Input file = '%s' output file = '%s'n",
- encoded_file_name, decoded_file_name);
- if(need_aiff) printf("Output file written in AIFF formatn");
- if(need_esps) printf("Output file written in ESPS formatn"); /* MI */
- if ((musicout = fopen(decoded_file_name, "w+b")) == NULL) {
- printf ("Could not create "%s".n", decoded_file_name);
- exit(1);
- }
- open_bit_stream_r(&bs, encoded_file_name, BUFFER_SIZE);
- if (need_aiff)
- if (aiff_seek_to_sound_data(musicout) == -1) {
- printf("Could not seek to PCM sound data in "%s".n",
- decoded_file_name);
- exit(1);
- }
- sample_frames = 0;
- while (!end_bs(&bs)) {
- 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 %ldn",
- frameBits, frameBits/bitsPerSlot, frameBits%bitsPerSlot);
- gotBits += frameBits;
- if (!sync) {
- printf("Frame cannot be locatedn");
- printf("Input stream may be emptyn");
- 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;
- 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 && need_esps) {
- esps_write_header(musicout,(long) sample_frames, (double)
- s_freq[info.sampling_frequency] * 1000,
- (int) stereo, 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(topSb>0) /* clear channels to 0 */
- for(j=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 clippedn", 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(topSb>0) /* debug : clear channels to 0 */
- for(j=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 clippedn", 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; 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 ;
- if( main_data_end > 4096 )
- { frame_start -= 4096;
- rewindNbytes( 4096 );
- }
- frame_start += main_data_slots(fr_ps);
- 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<2;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();
- III_get_scale_factors(III_scalefac,&III_side_info,gr,ch,
- &fr_ps);
- III_hufman_decode(is, &III_side_info, ch, gr, part2_start,
- &fr_ps);
- III_dequantize_sample(is, ro[ch], &III_scalefac,
- &(III_side_info.ch[ch].gr[gr]), ch, &fr_ps);
- }
- III_stereo(ro,lr,III_scalefac,
- &(III_side_info.ch[0].gr[gr]), &fr_ps);
- 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);
- III_antialias(re, hybridIn, /* Antialias butterflies. */
- &(III_side_info.ch[ch].gr[gr]), &fr_ps);
- for (sb=0; sb<SBLIMIT; sb++) { /* Hybrid synthesis. */
- III_hybrid(hybridIn[sb], hybridOut[sb], sb, ch,
- &(III_side_info.ch[ch].gr[gr]), &fr_ps);
- }
- for (ss=0;ss<18;ss++) /*Frequency inversion for polyphase.*/
- for (sb=0; sb<SBLIMIT; sb++)
- if ((ss%2) && (sb%2))
- hybridOut[sb][ss] = -hybridOut[sb][ss];
- for (ss=0;ss<18;ss++) { /* Polyphase synthesis */
- for (sb=0; sb<SBLIMIT; sb++)
- polyPhaseIn[sb] = hybridOut[sb][ss];
- clip += SubBandSynthesis (polyPhaseIn, ch,
- &((*pcm_sample)[ch][ss][0]));
- }
- }
- /* Output PCM sample points for one granule. */
- out_fifo(*pcm_sample, 18, &fr_ps, done, musicout,
- &sample_frames);
- }
- if(clip > 0) printf("%d samples clipped.n", clip);
- break;
- }
- }
- }
- if (need_aiff) {
- pcm_aiff_data.numChannels = stereo;
- pcm_aiff_data.numSampleFrames = sample_frames;
- pcm_aiff_data.sampleSize = 16;
- pcm_aiff_data.sampleRate = s_freq[info.sampling_frequency]*1000;
- #ifdef IFF_LONG
- pcm_aiff_data.sampleType = IFF_ID_SSND;
- #else
- strncpy(&pcm_aiff_data.sampleType,IFF_ID_SSND,4);
- #endif
- pcm_aiff_data.blkAlgn.offset = 0;
- pcm_aiff_data.blkAlgn.blockSize = 0;
- if (aiff_write_headers(musicout, &pcm_aiff_data) == -1) {
- printf("Could not write AIFF headers to "%s"n",
- decoded_file_name);
- exit(2);
- }
- }
- printf("Avg slots/frame = %.3f; b/smp = %.2f; br = %.3f kbpsn",
- (FLOAT) gotBits / (frameNum * bitsPerSlot),
- (FLOAT) gotBits / (frameNum * samplesPerFrame),
- (FLOAT) gotBits / (frameNum * samplesPerFrame) *
- s_freq[info.sampling_frequency]);
- close_bit_stream_r(&bs);
- fclose(musicout);
- /* for the correct AIFF header information */
- /* on the Macintosh */
- /* the file type and the file creator for */
- /* Macintosh compatible Digidesign is set */
- #ifdef MACINTOSH
- if (need_aiff) set_mac_file_attr(decoded_file_name, VOL_REF_NUM,
- CREATR_DEC_AIFF, FILTYP_DEC_AIFF);
- else set_mac_file_attr(decoded_file_name, VOL_REF_NUM,
- CREATR_DEC_BNRY, FILTYP_DEC_BNRY);
- #endif
- printf("Decoding of "%s" is finishedn", encoded_file_name);
- printf("The decoded PCM output file name is "%s"n", decoded_file_name);
- if (need_aiff)
- printf(""%s" has been written with AIFF header informationn",
- decoded_file_name);
- exit( 0 );
- }
- static void usage() /* print syntax & exit */
- {
- fprintf(stderr,
- "usage: %s queries for all arguments, orn",
- programName);
- fprintf(stderr,
- " %s [-A][-s sb] inputBS [outPCM]n", programName);
- fprintf(stderr,"wheren");
- fprintf(stderr," -A write an AIFF output PCM sound filen");
- fprintf(stderr," -s sb resynth only up to this sb (debugging only)n");
- fprintf(stderr," inputBS input bit stream of encoded audion");
- fprintf(stderr," outPCM output PCM sound file (dflt inName+%s)n",
- DFLT_OPEXT);
- exit(1);
- }