/* readers/writers for various sound file headers
*
* TODO: avi, asf, and soundfont code seriously incomplete
* avr doesn't match actual files
* quicktime is just a guess
* esignal not tackled yet (need an example file)
* check resource info on Mac (for SoundEdit)
* get example of little endian DEC header
*
* --------------------------------
* int mus_read_header (char *name): reads file's header
* int mus_write_header (char *name, int type, int in_srate, int in_chans, int loc, int size, int format, char *comment, int len): writes file's header
* int mus_update_header (char *name, int type, int size, int srate, int format, int chans, int loc): update file's header (unset fields should be 0)
* int mus_create_header_buffer (void): initialize (allocate) various buffers -- should be called before using mus_read_header and others
*
* Once mus_read_header has been called, the data in it can be accessed through:
*
* int mus_header_samples (void): samples
* int mus_header_data_location (void) location of data (bytes)
* int mus_header_chans (void) channels
* int mus_header_srate (void) srate
* int mus_header_type (void) header type (i.e. aiff, wave, etc) (see sndlib.h)
* int mus_header_format (void) data format (see sndlib.h)
* int mus_header_distributed (void) true if header info is scattered around in the file
* int mus_header_comment_start (void) comment start location (if any) (bytes)
* int mus_header_comment_end (void) comment end location
* int mus_header_aux_comment_start (int n) if multiple comments, nth start location
* int mus_header_aux_comment_end (int n) if multiple comments, nth end location
* int mus_header_type_specifier (void) original (header-specific) type ID
* int mus_header_bits_per_sample (void) sample width in bits
* int mus_true_file_length (void) true (lseek) file length
* int mus_header_format2bytes (void) sample width in bytes
*
* int mus_header_aiff_p(void) is header actually old-style AIFF, not AIFC
* int mus_header_writable(int type, int format)
* --------------------------------
*
* "Linear" below means 2's complement integer.
*
* Currently supported read/write (in standard data formats):
* NeXT/Sun/DEC/AFsp
* AIFF/AIFC
* RIFF (microsoft wave)
* IRCAM (old style)
* NIST-sphere
* no header
*
* Currently supported read-only (in selected data formats):
* 8SVX (IFF), IRCAM Vax float, EBICSF, INRS, ESPS, SPPACK, ADC (OGI), AVR, VOC,
* Sound Tools, Turtle Beach SMP, SoundFont 2.0, Sound Designer I and II, PSION alaw, MAUD,
* Tandy DeskMate new and old style, Gravis Ultrasound, Comdisco SPW, Goldwave sample, OMF,
* Sonic Foundry, SBStudio II, Delusion digital, Digiplayer ST3, Farandole Composer WaveSample,
* Ultratracker WaveSample, Sample Dump exchange, Yamaha SY85 and SY99 (buggy), Yamaha TX16,
* Covox v8, SPL, AVI, Kurzweil 2000
*
* for a few of these I'm still trying to get documentation -- best sources of info
* are ftp.cwi.nl:pub/audio (info files), the AFsp sources, and the SOX sources.
* sox and gsm are at ftp.cwi.nl, AFsp is from kabal@Polaris.EE.McGill.CA (Peter Kabal) as
* ftp.TSP.EE.McGill.CA:/pub/AFsp/AFsp-V3R2.tar.Z. The Sound Designer formats are described
* in the "Developer Documentation" from Digidesign. Other useful sources can be found at
* ftp.x.org:/contrib/audio/nas, svr-ftp.eng.cam.ac.uk:/comp.speech/tools, and
* at http://www.wotsit.org/music.htm. I put many of my test cases in
* ccrma-ftp.stanford.edu:/pub/Lisp/sf.tar.gz. The RIFF format is described in the
* Microsoft Multimedia Programmer's Reference Manual at ftp.microsoft.com:/SoftLib/MSLFILES/MDRK.EXE.
* AVI format is described in http://www.rahul.net/jfm/avi.html.
*
* The main problem with compressed sound files is that you can't do reliable
* random access to the data, can't easily read backwards, and most of the compression
* schemes are proprietary (and appalling), but to translate Mus10/Sam, HCOM, IEEE text,
* IBM CVSD, MIDI sample dumps, various adpcm cases, NIST shortpack files, and AVI see snd-trans.c
* in the sound editor (snd-3.tar.gz).
*
* If anyone has information on any other header or data formats, I would be most interested in it,
* but only if it can be included in this file.
*/
#if defined(HAVE_CONFIG_H)
#include "config.h"
#endif
#include <math.h>
#include <stdio.h>
#if (!defined(HAVE_CONFIG_H)) || (defined(HAVE_FCNTL_H))
#include <fcntl.h>
#endif
#include <signal.h>
#if (!defined(HAVE_CONFIG_H)) || (defined(HAVE_LIMITS_H))
#include <limits.h>
#endif
#include <errno.h>
#include <stdlib.h>
#if (!defined(HAVE_CONFIG_H)) || (defined(HAVE_STRING_H))
#include <string.h>
#endif
#if (defined(NEXT) || (defined(HAVE_LIBC_H) && (!defined(HAVE_UNISTD_H))))
#include <libc.h>
#else
#if (!(defined(_MSC_VER))) && (!(defined(MPW_C)))
#include <unistd.h>
#endif
#endif
#include "sndlib.h"
static int hdrbuf_is_inited = 0;
#define HDRBUFSIZ 256
#ifndef MACOS
static unsigned char *hdrbuf;
#else
static char *hdrbuf;
#endif
#define INITIAL_READ_SIZE 32
/* AIFF files can have any number of ANNO chunks, so we'll grab at least 4 of them */
#define AUX_COMMENTS 4
static int *aux_comment_start = NULL, *aux_comment_end = NULL;
#define LOOPS 2
static int *loop_modes = NULL,*loop_starts = NULL,*loop_ends = NULL;
static int markers = 0;
static int *marker_ids = NULL,*marker_positions = NULL;
#ifdef CLM
void reset_headers_c(void)
{
hdrbuf_is_inited = 0;
markers = 0;
}
#endif
int mus_create_header_buffer (void)
{
if (hdrbuf_is_inited == 0)
{
hdrbuf_is_inited = 1;
#ifndef MACOS
hdrbuf = (unsigned char *)CALLOC(HDRBUFSIZ,sizeof(unsigned char));
#else
hdrbuf = (char *)CALLOC(HDRBUFSIZ,sizeof(unsigned char));
#endif
aux_comment_start = (int *)CALLOC(AUX_COMMENTS,sizeof(int));
aux_comment_end = (int *)CALLOC(AUX_COMMENTS,sizeof(int));
loop_modes = (int *)CALLOC(LOOPS,sizeof(int));
loop_starts = (int *)CALLOC(LOOPS,sizeof(int));
loop_ends = (int *)CALLOC(LOOPS,sizeof(int));
if ((hdrbuf == NULL) || (aux_comment_start == NULL) || (aux_comment_end == NULL) ||
(loop_modes == NULL) || (loop_starts == NULL) || (loop_ends == NULL))
{
mus_error(MUS_MEMORY_ALLOCATION_FAILED,"header buffer allocation trouble");
return(-1);
}
}
return(0);
}
static const unsigned char I_DSND[4] = {'.','s','n','d'}; /* NeXT/Sun/Dec/SGI/AFsp first word */
static const unsigned char I_FORM[4] = {'F','O','R','M'}; /* AIFF first word */
static const unsigned char I_AIFF[4] = {'A','I','F','F'}; /* AIFF second word */
static const unsigned char I_AIFC[4] = {'A','I','F','C'}; /* ditto but might be compressed data */
static const unsigned char I_COMM[4] = {'C','O','M','M'};
static const unsigned char I_COMT[4] = {'C','O','M','T'};
static const unsigned char I_INFO[4] = {'I','N','F','O'};
static const unsigned char I_INST[4] = {'I','N','S','T'};
static const unsigned char I_inst[4] = {'i','n','s','t'}; /* RIFF wants lower case, just to be different */
static const unsigned char I_MARK[4] = {'M','A','R','K'};
static const unsigned char I_SSND[4] = {'S','S','N','D'};
static const unsigned char I_FVER[4] = {'F','V','E','R'};
static const unsigned char I_NONE[4] = {'N','O','N','E'};
static const unsigned char I_ULAW[4] = {'U','L','A','W'}; /* AIFC compression types that we can handle */
static const unsigned char I_ulaw[4] = {'u','l','a','w'}; /* or maybe it's lowercase (Apple) ... */
static const unsigned char I_ima4[4] = {'i','m','a','4'}; /* AIFC IMA adpcm apparently */
static const unsigned char I_raw_[4] = {'r','a','w',' '}; /* AIFC offset binary OS 8.5 (others are 'MAC3' 'MAC6' 'cdx4' 'cdx2' 'str4') */
static const unsigned char I_sowt[4] = {'s','o','w','t'}; /* AIFC little endian? */
static const unsigned char I_fl32[4] = {'f','l','3','2'}; /* AIFC 32-bit float? */
static const unsigned char I_fl64[4] = {'f','l','6','4'}; /* AIFC 64-bit float? */
static const unsigned char I_twos[4] = {'t','w','o','s'}; /* AIFC big endian? */
static const unsigned char I_ALAW[4] = {'A','L','A','W'};
static const unsigned char I_alaw[4] = {'a','l','a','w'}; /* apple */
static const unsigned char I_APPL[4] = {'A','P','P','L'};
static const unsigned char I_MUS_[4] = {'C','L','M',' '}; /* I hereby claim this AIFF chunk name */
static const unsigned char I_RIFF[4] = {'R','I','F','F'}; /* RIFF first word */
static const unsigned char I_RIFX[4] = {'R','I','F','X'}; /* RIFX first word (big-endian RIFF file) */
static const unsigned char I_WAVE[4] = {'W','A','V','E'};
static const unsigned char I_fmt_[4] = {'f','m','t',' '};
static const unsigned char I_data[4] = {'d','a','t','a'};
static const unsigned char I_fact[4] = {'f','a','c','t'}; /* used by compressed RIFF files */
static const unsigned char I_clm_[4] = {'c','l','m',' '};
static const unsigned char I_NIST[4] = {'N','I','S','T'}; /* first word of NIST SPHERE files */
static const unsigned char I_8SVX[4] = {'8','S','V','X'}; /* AIFF other choice */
static const unsigned char I_VOC0[4] = {'C','r','e','a'}; /* Actual text is "Creative Voice File" */
static const unsigned char I_VOC1[4] = {'t','i','v','e'};
static const unsigned char I_SOUN[4] = {'S','O','U','N'}; /* Sound Tools first word="SOUND" -- not unique as SMP files start with "SOUND SAMPLE" */
static const unsigned char I_SMP1[4] = {'D',' ','S','A'};
static const unsigned char I_SMP2[4] = {'M','P','L','E'};
static const unsigned char I_BODY[4] = {'B','O','D','Y'}; /* next 4 for 8svx chunk names */
static const unsigned char I_VHDR[4] = {'V','H','D','R'};
static const unsigned char I_CHAN[4] = {'C','H','A','N'};
static const unsigned char I_ANNO[4] = {'A','N','N','O'};
static const unsigned char I_NAME[4] = {'N','A','M','E'};
static const unsigned char I_AVR_[4] = {'2','B','I','T'}; /* first word of AVR files */
static const unsigned char I_HCOM[4] = {'H','C','O','M'};
static const unsigned char I_FSSD[4] = {'F','S','S','D'};
static const unsigned char I_SPIB[4] = {'%','/','/','\n'}; /* first word of IEEE spib text sound files */
static const unsigned char I_S___[4] = {'%','-','-','-'}; /* first word of other IEEE spib text sound files */
static const unsigned char I_ALaw[4] = {'A','L','a','w'}; /* first word of PSION alaw files */
static const unsigned char I_Soun[4] = {'S','o','u','n'}; /* second */
static const unsigned char I_MAUD[4] = {'M','A','U','D'}; /* MAUD specialization of AIFF */
static const unsigned char I_MHDR[4] = {'M','H','D','R'};
static const unsigned char I_MDAT[4] = {'M','D','A','T'};
static const unsigned char I_mdat[4] = {'m','d','a','t'}; /* quicktime */
static const unsigned char I_AFsp[4] = {'A','F','s','p'};
static const unsigned char I_MThd[4] = {'M','T','h','d'}; /* sigh -- the M word */
static const unsigned char I_DVSM[4] = {'D','V','S','M'}; /* first word of DVSM files */
static const unsigned char I_DECN[4] = {'.','s','d','\0'}; /* first word of DEC files (?) */
static const unsigned char I_Esig[4] = {'E','s','i','g'}; /* first word of Esignal files */
static const unsigned char I_nalc[4] = {'n','a','l','\n'}; /* second word of Esignal files */
static const unsigned char I_sfbk[4] = {'s','f','b','k'}; /* SoundFont 2.0 */
static const unsigned char I_sdta[4] = {'s','d','t','a'};
static const unsigned char I_shdr[4] = {'s','h','d','r'};
static const unsigned char I_smpl[4] = {'s','m','p','l'};
static const unsigned char I_pdta[4] = {'p','d','t','a'};
static const unsigned char I_LIST[4] = {'L','I','S','T'};
static const unsigned char I_GF1P[4] = {'G','F','1','P'}; /* first word of Gravis Ultrsound patch files */
static const unsigned char I_ATCH[4] = {'A','T','C','H'}; /* second word */
static const unsigned char I_DSIG[4] = {'$','S','I','G'}; /* first word of Comdisco SPW file */
static const unsigned char I_NAL_[4] = {'N','A','L','_'}; /* second word */
static const unsigned char I_GOLD[4] = {'G','O','L','D'}; /* first word Goldwave(?) sample file */
static const unsigned char I__WAV[4] = {' ','S','A','M'}; /* second word */
static const unsigned char I_SRFS[4] = {'S','R','F','S'}; /* first word Sonic Resource Foundry file(?) */
static const unsigned char I_Diam[4] = {'D','i','a','m'}; /* first word DiamondWare file */
static const unsigned char I_ondW[4] = {'o','n','d','W'}; /* second word */
static const unsigned char I_Drat[4] = {'.','r','a','\xfd'}; /* first word real audio file */
static const unsigned char I_CSRE[4] = {'C','S','R','E'}; /* adf first word -- second starts with "40" */
static const unsigned char I_SND_[4] = {'S','N','D',' '}; /* SBStudio II */
static const unsigned char I_SNIN[4] = {'S','N','I','N'};
static const unsigned char I_SNDT[4] = {'S','N','D','T'};
static const unsigned char I_DDSF[4] = {'D','D','S','F'}; /* Delusion Digital Sound File */
static const unsigned char I_FSMt[4] = {'F','S','M','\376'}; /* Farandole Composer WaveSample */
static const unsigned char I_SDXc[4] = {'S','D','X',':'}; /* Sample dump exchange format */
static const unsigned char I_UWFD[4] = {'U','W','F','D'}; /* Ultratracker Wavesample */
static const unsigned char I_LM89[4] = {'L','M','8','9'}; /* Yamaha TX-16 */
static const unsigned char I_SY80[4] = {'S','Y','8','0'}; /* Yamaha SY-99 */
static const unsigned char I_SY85[4] = {'S','Y','8','5'}; /* Yamaha SY-85 */
static const unsigned char I_SCRS[4] = {'S','C','R','S'}; /* Digiplayer ST3 */
static const unsigned char I_covox[4] = {'\377','\125','\377','\252'};
static const unsigned char I_DSPL[4] = {'D','S','P','L'}; /* Digitracker SPL (now obsolete) */
static const unsigned char I_AVI_[4] = {'A','V','I',' '}; /* RIFF AVI */
static const unsigned char I_strf[4] = {'s','t','r','f'};
static const unsigned char I_movi[4] = {'m','o','v','i'};
static const unsigned char I_PRAM[4] = {'P','R','A','M'}; /* Kurzweil 2000 */
static const unsigned char I_ones[4] = {'\377','\377','\377','\377'};
static const unsigned char I_zeros[4] = {'\0','\0','\0','\0'};
static const unsigned char I_asf0[4] = {'\321','\051','\342','\326'};
static const unsigned char I_asf1[4] = {'\332','\065','\321','\021'};
static const unsigned char I_asf2[4] = {'\220','\064','\000','\240'};
static const unsigned char I_asf3[4] = {'\311','\003','\111','\276'};
/* .glt and .shp -> Perry Cook's SPASM data files */
#define I_IRCAM_VAX 0x0001a364
#define I_IRCAM_SUN 0x0002a364
#define I_IRCAM_MIPS 0x0003a364
#define I_IRCAM_NEXT 0x0004a364
#define NINRS 7
static const unsigned int I_INRS[NINRS]={0xcb460020,0xd0465555,0xfa460000,0x1c470040,0x3b470080,0x7a470000,0x9c470040};
static int data_size=0, data_location=0, srate=0, chans=0, header_type=0, data_format=0, original_data_format=0, true_file_length=0;
static int comment_start=0, comment_end=0, header_distributed=0, type_specifier=0, bits_per_sample=0, fact_samples=0, block_align=0;
static int base_detune = 0, base_note = 0;
int mus_header_samples (void) {return(data_size);}
int mus_header_data_location (void) {return(data_location);}
int mus_header_chans (void) {return(chans);}
int mus_header_srate (void) {return(srate);}
int mus_header_type (void) {return(header_type);}
int mus_header_format (void) {return(data_format);}
int mus_header_distributed (void) {return(header_distributed);}
int mus_header_comment_start (void) {return(comment_start);}
int mus_header_comment_end (void) {return(comment_end);}
int mus_header_aux_comment_start (int n) {return(aux_comment_start[n]);}
int mus_header_aux_comment_end (int n) {return(aux_comment_end[n]);}
int mus_header_type_specifier (void) {return(type_specifier);}
int mus_header_bits_per_sample (void) {return(bits_per_sample);}
int mus_header_fact_samples (void) {return(fact_samples);}
int mus_header_block_align (void) {return(block_align);}
int mus_true_file_length (void) {return(true_file_length);}
int mus_header_original_format (void) {return(original_data_format);}
int mus_header_loop_mode(int which) {if (loop_modes) return(loop_modes[which]); else return(-1);}
int mus_header_loop_start(int which) {if (loop_starts) return(loop_starts[which]); else return(-1);}
int mus_header_loop_end(int which) {if (loop_ends) return(loop_ends[which]); else return(-1);}
int mus_header_mark_position(int id) {int i; for (i=0;i<markers;i++) {if (marker_ids[i] == id) return(marker_positions[i]);} return(-1);}
int mus_header_base_detune(void) {return(base_detune);}
int mus_header_base_note(void) {return(base_note);}
int mus_format2bytes (int format)
{
switch (format)
{
case SNDLIB_8_LINEAR: return(1); break;
case SNDLIB_16_LINEAR: return(2); break;
case SNDLIB_8_UNSIGNED: return(1); break;
case SNDLIB_8_MULAW: return(1); break;
case SNDLIB_8_ALAW: return(1); break;
case SNDLIB_32_LINEAR: return(4); break;
case SNDLIB_32_FLOAT: return(4); break;
case SNDLIB_24_LINEAR: return(3); break;
case SNDLIB_64_DOUBLE: return(8); break;
case SNDLIB_16_LINEAR_LITTLE_ENDIAN: return(2); break;
case SNDLIB_32_LINEAR_LITTLE_ENDIAN: return(4); break;
case SNDLIB_32_FLOAT_LITTLE_ENDIAN: return(4); break;
case SNDLIB_64_DOUBLE_LITTLE_ENDIAN: return(8); break;
case SNDLIB_24_LINEAR_LITTLE_ENDIAN: return(3); break;
case SNDLIB_16_UNSIGNED: return(2); break;
case SNDLIB_16_UNSIGNED_LITTLE_ENDIAN: return(2); break;
case SNDLIB_32_VAX_FLOAT: return(4); break;
default: return(1); break; /* we divide by this number, so 0 is not safe */
}
}
int mus_header_format2bytes (void) {return(mus_format2bytes(data_format));}
int mus_samples2bytes (int format, int size) {return(size*(mus_format2bytes(format)));}
int mus_bytes2samples (int format, int size) {return((int)(size/(mus_format2bytes(format))));}
void mus_set_snd_header (int in_srate, int in_chans, int in_format)
{
srate = in_srate;
chans = in_chans;
data_format = in_format;
data_size = mus_bytes2samples(in_format,true_file_length);
}
static int equal_big_or_little_endian(unsigned char *n1, const unsigned int n2)
{
return((mus_big_endian_unsigned_int(n1) == n2) || (mus_little_endian_unsigned_int(n1) == n2));
}
static short big_or_little_endian_short (unsigned char *n, int little)
{
if (little) return(mus_little_endian_short(n));
return(mus_big_endian_short(n));
}
static int big_or_little_endian_int (unsigned char *n, int little)
{
if (little) return(mus_little_endian_int(n));
return(mus_big_endian_int(n));
}
static float big_or_little_endian_float (unsigned char *n, int little)
{
if (little) return(mus_little_endian_float(n));
return(mus_big_endian_float(n));
}
static int match_four_chars(unsigned char *head, const unsigned char *match)
{
int i;
for (i=0;i<4;i++) if (head[i] != match[i]) return(0);
return(1);
}
static void write_four_chars(unsigned char *head, const unsigned char *match)
{
int i;
for (i=0;i<4;i++) head[i] = match[i];
}
char *mus_header_type2string(int type)
{
switch (type)
{
case NeXT_sound_file: return("Sun"); break;
case AIFC_sound_file: return("AIFC"); break;
case RIFF_sound_file: return("RIFF"); break;
case BICSF_sound_file: return("BICSF"); break;
case NIST_sound_file: return("NIST"); break;
case INRS_sound_file: return("INRS"); break;
case ESPS_sound_file: return("ESPS"); break;
case SVX_sound_file: return("SVX8"); break;
case VOC_sound_file: return("VOC"); break;
case SNDT_sound_file: return("SNDT"); break;
case raw_sound_file: return("raw (no header)"); break;
case SMP_sound_file: return("SMP"); break;
case SD2_sound_file: return("Sound Designer 2"); break;
case AVR_sound_file: return("AVR"); break;
case IRCAM_sound_file: return("IRCAM"); break;
case SD1_sound_file: return("Sound Designer 1"); break;
case SPPACK_sound_file: return("SPPACK"); break;
case MUS10_sound_file: return("Mus10"); break;
case HCOM_sound_file: return("HCOM"); break;
case PSION_sound_file: return("PSION"); break;
case MAUD_sound_file: return("MAUD"); break;
case IEEE_sound_file: return("IEEE text"); break;
case DeskMate_sound_file: return("DeskMate"); break;
case DeskMate_2500_sound_file: return("DeskMate_2500"); break;
case Matlab_sound_file: return("Matlab"); break;
case ADC_sound_file: return("ADC/OGI"); break;
case SoundEdit_sound_file: return("SoundEdit"); break;
case SoundEdit_16_sound_file: return("SoundEdit 16"); break;
case DVSM_sound_file: return("DVSM"); break;
case MIDI_file: return("MIDI"); break;
case Esignal_file: return("Esignal"); break;
case soundfont_sound_file: return("SoundFont"); break;
case gravis_sound_file: return("Gravis Ultrasound patch"); break;
case comdisco_sound_file: return("Comdisco SPW signal"); break;
case goldwave_sound_file: return("Goldwave sample"); break;
case srfs_sound_file: return("SRFS"); break;
case MIDI_sample_dump: return("MIDI sample dump"); break;
case DiamondWare_sound_file: return("DiamondWare"); break;
case RealAudio_sound_file: return("RealAudio"); break;
case ADF_sound_file: return("CSRE adf"); break;
case SBStudioII_sound_file: return("SBStudioII"); break;
case Delusion_sound_file: return("Delusion"); break;
case Farandole_sound_file: return("Farandole"); break;
case Sample_dump_sound_file: return("Sample dump"); break;
case Ultratracker_sound_file: return("Ultratracker"); break;
case Yamaha_TX16_sound_file: return("TX-16"); break;
case Yamaha_SY85_sound_file: return("Sy-85"); break;
case Yamaha_SY99_sound_file: return("Sy-99"); break;
case Kurzweil_2000_sound_file: return("Kurzweil 2000"); break;
case digiplayer_sound_file: return("Digiplayer ST3"); break;
case Covox_sound_file: return("Covox V8"); break;
case SPL_sound_file: return("Digitracker SPL"); break;
case AVI_sound_file: return("AVI"); break;
case OMF_sound_file: return("OMF"); break;
case Quicktime_sound_file: return("Quicktime"); break;
case asf_sound_file: return("asf"); break;
case AIFF_sound_file: return("AIFF"); break;
default: return("unknown"); break;
}
}
char *mus_header_data_format2string(int format)
{
switch (format)
{
case SNDLIB_UNSUPPORTED: return("unsupported"); break;
case SNDLIB_NO_SND: return("no_snd"); break;
case SNDLIB_16_LINEAR: return("16-bit big endian"); break;
case SNDLIB_8_MULAW: return("mulaw"); break;
case SNDLIB_8_LINEAR: return("8-bit"); break;
case SNDLIB_32_FLOAT: return("32-bit big endian float"); break;
case SNDLIB_32_LINEAR: return("32-bit big endian"); break;
case SNDLIB_8_ALAW: return("alaw"); break;
case SNDLIB_8_UNSIGNED: return("8-bit unsigned"); break;
case SNDLIB_24_LINEAR: return("24-bit big endian"); break;
case SNDLIB_64_DOUBLE: return("64-bit big endian double"); break;
case SNDLIB_16_LINEAR_LITTLE_ENDIAN: return("16-bit little endian"); break;
case SNDLIB_32_LINEAR_LITTLE_ENDIAN: return("32-bit little endian"); break;
case SNDLIB_32_FLOAT_LITTLE_ENDIAN: return("32-bit little endian float"); break;
case SNDLIB_64_DOUBLE_LITTLE_ENDIAN: return("64-bit little endian double"); break;
case SNDLIB_16_UNSIGNED: return("16-bit big endian unsigned"); break;
case SNDLIB_16_UNSIGNED_LITTLE_ENDIAN: return("16-bit little endian unsigned"); break;
case SNDLIB_12_UNSIGNED: return("12-bit big endian unsigned"); break;
case SNDLIB_12_UNSIGNED_LITTLE_ENDIAN: return("12-bit little endian unsigned"); break;
case SNDLIB_12_LINEAR: return("12-bit big endian"); break;
case SNDLIB_12_LINEAR_LITTLE_ENDIAN: return("12-bit little endian"); break;
case SNDLIB_24_LINEAR_LITTLE_ENDIAN: return("24-bit little endian"); break;
case SNDLIB_32_VAX_FLOAT: return("vax float"); break;
default: return("unknown"); break;
}
}
static int read_bicsf_header (int chan);
/* ------------------------------------ NeXT (or Sun) --------------------------------
*
* 0: ".snd"
* 4: data_location (bytes) (not necessarily word aligned on Sun)
* 8: data_size (bytes) -- sometimes incorrect ("advisory")
* 12: data format indicator -- see below
* 16: srate (int)
* 20: chans
* 24: comment start
*
* in an AFsp file, the first 4 bytes of the comment are "AFsp",
* see headers.lisp for readers/writers of AFsp fields
* for bicsf, the integer at 28 is 107364 or 107415
*
* on NeXTStep, always big-endian. ".snd"==0x2e736e64 on big-endian machines.
*
* formats are:
* 0 unspecified, 1 mulaw_8, 2 linear_8, 3 linear_16, 4 linear_24, 5 linear_32, 6 float,
* 7 double, 8 indirect, 9 nested, 10 dsp_core, 11 dsp_data_8, 12 dsp_data_16, 13 dsp_data_24,
* 14 dsp_data_32, 16 display, 17 mulaw_squelch, 18 emphasized, 19 compressed, 20 compressed_emphasized
* 21 dsp_commands, 22 dsp_commands_samples, 23 adpcm_g721, 24 adpcm_g722, 25 adpcm_g723,
* 26 adpcm_g723_5, 27 alaw_8, 28 aes, 29 delat_mulaw_8 {internal Snd-secret format: 30=linear_32_little_endian}
*
*/
/* according to the file /usr/share/magic, the DECN versions were little endian */
static int read_next_header (int chan)
{
int maybe_bicsf;
type_specifier = mus_uninterpreted_int((unsigned char *)hdrbuf);
data_location = mus_big_endian_int((unsigned char *)(hdrbuf+4));
data_size = mus_big_endian_int((unsigned char *)(hdrbuf+8));
/* can be bogus -- fixup if possible */
true_file_length = lseek(chan,0L,SEEK_END);
if (data_size <= 24) data_size = (true_file_length-data_location);
original_data_format = mus_big_endian_int((unsigned char *)(hdrbuf+12));
switch (original_data_format)
{ /* defined in /usr/include/sound/soundstruct.h */
/* see headers.lisp for a table of all known format names */
case 1: data_format = SNDLIB_8_MULAW; break;
case 2: data_format = SNDLIB_8_LINEAR; break;
case 3: data_format = SNDLIB_16_LINEAR; break;
case 4: data_format = SNDLIB_24_LINEAR; break;
case 5: data_format = SNDLIB_32_LINEAR; break;
case 6: data_format = SNDLIB_32_FLOAT; break;
case 7: data_format = SNDLIB_64_DOUBLE; break;
case 18: data_format = SNDLIB_16_LINEAR; break;
/* "emphasized": Xavier Serra's de-emphasis filter: y(n) = x(n) + .9 y(n-1) */
case 27: data_format = SNDLIB_8_ALAW; break;
case 30: data_format = SNDLIB_32_LINEAR_LITTLE_ENDIAN; break; /* this is for Snd's internal benefit -- it is not used elsewhere */
default: data_format = SNDLIB_UNSUPPORTED; break;
}
srate = mus_big_endian_int((unsigned char *)(hdrbuf+16));
chans = mus_big_endian_int((unsigned char *)(hdrbuf+20));
comment_start = 24;
comment_end = data_location - 1;
if (comment_end < comment_start) comment_end = comment_start;
if (match_four_chars((unsigned char *)(hdrbuf+24),I_AFsp)) header_distributed = 1; else header_distributed = 0;
maybe_bicsf = mus_big_endian_int((unsigned char *)(hdrbuf+28));
if (maybe_bicsf == 107364) read_bicsf_header(chan);
data_size = mus_bytes2samples(data_format,data_size);
return(0);
}
int mus_write_next_header (int chan, int srate, int chans, int loc, int siz, int format, char *comment, int len)
{
int i,j;
write_four_chars((unsigned char *)hdrbuf,I_DSND); /* ".snd" */
i = len/4;
j = 24 + (4 * (i+1));
if (loc < j) loc = j;
mus_set_big_endian_int((unsigned char *)(hdrbuf+4),loc);
mus_set_big_endian_int((unsigned char *)(hdrbuf+8),siz);
switch (format)
{
case SNDLIB_8_MULAW: mus_set_big_endian_int((unsigned char *)(hdrbuf+12),1); break;
case SNDLIB_8_LINEAR: mus_set_big_endian_int((unsigned char *)(hdrbuf+12),2); break;
case SNDLIB_16_LINEAR: mus_set_big_endian_int((unsigned char *)(hdrbuf+12),3); break;
case SNDLIB_24_LINEAR: mus_set_big_endian_int((unsigned char *)(hdrbuf+12),4); break;
case SNDLIB_32_LINEAR: mus_set_big_endian_int((unsigned char *)(hdrbuf+12),5); break;
case SNDLIB_32_FLOAT: mus_set_big_endian_int((unsigned char *)(hdrbuf+12),6); break;
case SNDLIB_64_DOUBLE: mus_set_big_endian_int((unsigned char *)(hdrbuf+12),7); break;
case SNDLIB_32_LINEAR_LITTLE_ENDIAN: mus_set_big_endian_int((unsigned char *)(hdrbuf+12),30); break; /* see above */
case SNDLIB_8_ALAW: mus_set_big_endian_int((unsigned char *)(hdrbuf+12),27); break;
default: mus_error(MUS_UNSUPPORTED_DATA_FORMAT,"can't write NeXT/Sun data format: %d",format); return(-1); break;
}
mus_set_big_endian_int((unsigned char *)(hdrbuf+16),srate);
mus_set_big_endian_int((unsigned char *)(hdrbuf+20),chans);
write(chan,hdrbuf,24);
j = 0;
for (i=0;i<len;i++)
{
hdrbuf[j]=comment[i];
j++;
if (j == HDRBUFSIZ)
{
write(chan,hdrbuf,HDRBUFSIZ);
j = 0;
}
}
for (i=0;i<(loc-(len+24));i++) /* now fill left over bytes with nulls */
{
hdrbuf[j]=0;
j++;
if (j == HDRBUFSIZ)
{
write(chan,hdrbuf,HDRBUFSIZ);
j = 0;
}
}
if (j != 0) write(chan,hdrbuf,j);
data_location = loc;
return(0);
}
static void update_next_header (int chan, int size)
{
lseek(chan,8L,SEEK_SET);
mus_set_big_endian_int((unsigned char *)(hdrbuf+0),size);
write(chan,hdrbuf,4);
}
static void update_next_header_comment (int chan, int loc, char *comment, int len)
{
int i,j;
lseek(chan,(long)(loc-4),SEEK_SET);
read(chan,hdrbuf,4);
lseek(chan,(long)(loc-4),SEEK_SET);
for (j=0;j<4;j++) if (hdrbuf[j]==0) hdrbuf[j] = 32;
j = 4;
for (i=0;i<len;i++)
{
hdrbuf[j]=comment[i];
j++;
if (j == HDRBUFSIZ)
{
write(chan,hdrbuf,HDRBUFSIZ);
j = 0;
}
}
if (j != 0) write(chan,hdrbuf,j);
}
/* ------------------------------------ AIFF ------------------------------------
*
* 0: "FORM"
* 4: size (bytes)
* 8: "AIFF" or "AIFC" -- the latter includes compressed formats (list extended for 8.5 Sound.h)
*
* Thereafter the file is organized into "chunks", each chunk being
* a 4-byte identifer followed by an int (4-bytes) giving the chunk size
* not including the 8-byte header. AIFF data is signed. If the chunk
* size is odd, an extra (unaccounted-for) null byte is added at the end.
*
* The chunks we want are "COMM", "SSND", and "APPL".
*
* COMM: 0: chans
* 2: frames
* 6: bits per sample
* 8: srate as 80-bit IEEE float
* then if AIFC (not AIFF), 4 bytes giving compression id ("NONE"=not compressed)
* followed by Pascal string giving long name of compression type
*
* SSND: 0: data location (offset within SSND chunk)
*
* Other chunks include: ANNO: a comment, INST: loop control, MARK: marker, MIDI: midi,
* COMT: comment (max 65536 chars), NAME: sound name, AUTH: author's name
* (c), AESD: recording data, APPL: application specific stuff
* "MARK" size short-#marks {marks} -- latter are short-ID long-position pstring-name.
* "INST" size chars[baseNote detune lowNote highNote lowVelocity HighVelocity] short-gain loops[sustain release]
* loop: short-playMode marker-begin marker-end (signed?) shorts)
* playMode: 0 no loop, 1 forward loop, 2 forward/backward loop
* chars are MIDI data (detune is in cents)
* "MIDI" size MIDI-data...
* "AESD" size AES Channel Status Data (24 bytes as specified by AES)
* see "AES: Guidelines for the use of the AES3 interface"
* byte 0: bit 0: 0=consumer, 1=pro
* bit 1: 0=audio, 1=non-audio
* bits 2:4: emphasis: 0:none, 4:none, 6:CD, 7:CCITT J17
* bits 6:7: srate: 00=48KHz, 01=48, 10=44.1, 11=32
* byte 1: bits 0:3: chans: 2:mono, else stereo
* byte 2 for word size stuff (always ends up 16-bit): bits 3-5=sample length where 4=16-bit
* byte 3: multi-channels modes, 4: AES sync ref, 5:unused, 6-9:ASCII source ID, 10-13:ASCII destination ID
* byte 14-17:local sample addr, 18-21:time of day addr, then CRC checks
* "APPL" size signature data
* "COMT" size short-#comments {comments} -- the latter are long-time marker short-text-length char-text
* time is in seconds since 1-Jan-1904
* "NAME"/"AUTH"/"(c) "/"ANNO" size char-name
* "FVER" size(4) AIFC-format-version -- currently always 0xA2805140
* "SAXL" -- a desperate kludge to get around Apple's own compression schemes!
*
* always big-endian
* There was also (briefly) an AIFS file, now deprecated.
*
* originally I thought AIFF (as opposed to AIFC) files were obsolete, but lots of
* "legacy" sotfware insists on AIFF still, so mus_header_aiff_p was added to check this case
*/
/* ieee-80 conversions -- design by committee! */
/* this code taken from CSound sources -- apparently originally written by Malcolm Slaney at Apple */
#define ULPOW2TO31 ((unsigned int)0x80000000)
#define DPOW2TO31 ((double)2147483648.0) /* 2^31 */
static double myUlongToDouble(unsigned int ul)
{
double val;
if(ul & ULPOW2TO31) val = DPOW2TO31 + (ul & (~ULPOW2TO31));
else val = ul;
return val;
}
static unsigned int myDoubleToUlong(double val)
{
unsigned int ul;
if(val < DPOW2TO31) ul = (unsigned int)val;
else ul = ULPOW2TO31 | (unsigned int)(val-DPOW2TO31);
return ul;
}
static double ieee_80_to_double(unsigned char *p)
{
unsigned char sign;
short exp = 0;
unsigned int mant1 = 0;
unsigned int mant0 = 0;
double val;
exp = *p++; exp <<= 8; exp |= *p++; sign = (exp & 0x8000) ? 1 : 0; exp &= 0x7FFF;
mant1 = *p++; mant1 <<= 8; mant1 |= *p++; mant1 <<= 8; mant1 |= *p++; mant1 <<= 8; mant1 |= *p++;
mant0 = *p++; mant0 <<= 8; mant0 |= *p++; mant0 <<= 8; mant0 |= *p++; mant0 <<= 8; mant0 |= *p++;
if(mant1 == 0 && mant0 == 0 && exp == 0 && sign == 0)
return 0.0;
else
{
val = myUlongToDouble(mant0) * pow(2.0,-63.0);
val += myUlongToDouble(mant1) * pow(2.0,-31.0);
val *= pow(2.0,((double) exp) - 16383.0);
return sign ? -val : val;
}
}
static void double_to_ieee_80(double val, unsigned char *p)
{
unsigned char sign = 0;
short exp = 0;
unsigned int mant1 = 0;
unsigned int mant0 = 0;
if(val < 0.0) { sign = 1; val = -val; }
if(val != 0.0) /* val identically zero -> all elements zero */
{
exp = (short)(log(val)/log(2.0) + 16383.0);
val *= pow(2.0, 31.0+16383.0-(double)exp);
mant1 = myDoubleToUlong(val);
val -= myUlongToDouble(mant1);
val *= pow(2.0, 32.0);
mant0 = myDoubleToUlong(val);
}
*p++ = ((sign<<7)|(exp>>8)); *p++ = 0xFF & exp;
*p++ = 0xFF & (mant1>>24); *p++ = 0xFF & (mant1>>16); *p++ = 0xFF & (mant1>> 8); *p++ = 0xFF & (mant1);
*p++ = 0xFF & (mant0>>24); *p++ = 0xFF & (mant0>>16); *p++ = 0xFF & (mant0>> 8); *p++ = 0xFF & (mant0);
}
static int update_form_size, update_frames_location, update_ssnd_location;
static int seek_and_read(int chan, unsigned char *buf, int offset, int nbytes)
{
if (offset < 0) return(-1);
lseek(chan, offset, SEEK_SET);
#ifndef MACOS
return(read(chan,buf,nbytes));
#else
return(read(chan,(char *)buf,nbytes));
#endif
}
static int read_aiff_marker(int m, unsigned char *buf)
{
int psize;
marker_ids[m] = mus_big_endian_short((unsigned char *)buf);
marker_positions[m] = mus_big_endian_int((unsigned char *)(buf+2));
psize = (int)buf[6] + 1;
if (psize & 1) psize++;
return(psize+6);
}
static int read_aiff_header (int chan, int overall_offset)
{
/* we know we have checked for FORM xxxx AIFF|AIFC when we arrive here */
/* as far as I can tell, the COMM block has the header data we seek, and the SSND block has the sound data */
/* everything else will be ignored -- apparently we can depend on seeing a "chunk" name, then size */
int chunksize,offset,frames,chunkloc,happy,i,j,num_marks,m,moff,msize;
type_specifier = mus_uninterpreted_int((unsigned char *)(hdrbuf+8+overall_offset));
update_ssnd_location = 0;
chunkloc = 12 + overall_offset;
offset = 0;
for (i=0;i<AUX_COMMENTS;i++) aux_comment_start[i] = 0;
data_format = SNDLIB_16_LINEAR;
header_distributed = 1;
srate = 0;
chans = 0;
happy = 1;
data_size = 0;
if (loop_modes)
{
loop_modes[0] = 0;
loop_modes[1] = 0;
}
true_file_length = lseek(chan,0L,SEEK_END);
update_form_size = mus_big_endian_int((unsigned char *)(hdrbuf+4+overall_offset)); /* should be file-size-8 unless there are multiple forms */
while (happy)
{
offset += chunkloc;
if (seek_and_read(chan,(unsigned char *)hdrbuf,offset,32) <= 0)
{
mus_error(MUS_HEADER_READ_FAILED,"AIFF header chunks confused at %d",offset);
return(-1);
}
chunksize = mus_big_endian_int((unsigned char *)(hdrbuf+4));
if (match_four_chars((unsigned char *)hdrbuf,I_COMM))
{
chans = mus_big_endian_short((unsigned char *)(hdrbuf+8));
frames = mus_big_endian_int((unsigned char *)(hdrbuf+10));
update_frames_location = 10+offset;
original_data_format = mus_big_endian_short((unsigned char *)(hdrbuf+14));
if ((original_data_format%8) != 0)
{
/* weird sizes are legal --
* these samples are left-justified (and zero padded on the right), so
* we can handle any bit size by rounding up to the nearest byte.
*/
original_data_format=8*(1+(original_data_format>>3));
}
if (original_data_format == 8) data_format = SNDLIB_8_LINEAR;
else if (original_data_format == 16) data_format = SNDLIB_16_LINEAR;
else if (original_data_format == 24) data_format = SNDLIB_24_LINEAR;
else if (original_data_format == 32) data_format = SNDLIB_32_LINEAR;
else data_format = SNDLIB_UNSUPPORTED;
srate = (int)ieee_80_to_double((unsigned char *)(hdrbuf+16));
/* if AIFC, compression type over-rides (possibly bogus) original_data_format */
if (type_specifier == mus_uninterpreted_int((unsigned char *)I_AIFC))
{
/* some aifc files assume the compression field is a new and very weird chunk!! -- surely a bug? */
/* AIFF spec says COMM size is always 18, but this is amended in the newer AIFC spec */
if (chunksize == 18) chunksize += (5+((int)hdrbuf[30])); /* 5=chunk header length in this case */
if ((!(match_four_chars((unsigned char *)(hdrbuf+26),I_NONE))) &&
(!(match_four_chars((unsigned char *)(hdrbuf+26),I_twos))))
{
original_data_format = mus_uninterpreted_int((unsigned char *)(hdrbuf+26));
if ((match_four_chars((unsigned char *)(hdrbuf+26),I_ALAW)) ||
(match_four_chars((unsigned char *)(hdrbuf+26),I_alaw)))
data_format = SNDLIB_8_ALAW;
else
{
if ((match_four_chars((unsigned char *)(hdrbuf+26),I_ULAW)) ||
(match_four_chars((unsigned char *)(hdrbuf+26),I_ulaw)))
data_format = SNDLIB_8_MULAW;
else
{
/* taken from Sound.h in OS 8.5 -- just guessing... */
if (match_four_chars((unsigned char *)(hdrbuf+26),I_sowt))
{
if (data_format == SNDLIB_16_LINEAR) data_format = SNDLIB_16_LINEAR_LITTLE_ENDIAN;
else if (data_format == SNDLIB_24_LINEAR) data_format = SNDLIB_24_LINEAR_LITTLE_ENDIAN;
else if (data_format == SNDLIB_32_LINEAR) data_format = SNDLIB_32_LINEAR_LITTLE_ENDIAN;
}
else
{
if (match_four_chars((unsigned char *)(hdrbuf+26),I_raw_))
{
if (data_format == SNDLIB_8_LINEAR) data_format = SNDLIB_8_UNSIGNED;
else if (data_format == SNDLIB_16_LINEAR) data_format = SNDLIB_16_UNSIGNED;
}
else
{
if (match_four_chars((unsigned char *)(hdrbuf+26),I_fl32))
data_format = SNDLIB_32_FLOAT;
else
{
if (match_four_chars((unsigned char *)(hdrbuf+26),I_fl64))
data_format = SNDLIB_64_DOUBLE;
else
{
if (match_four_chars((unsigned char *)(hdrbuf+26),I_ima4))
{
block_align = 34;
original_data_format = SNDLIB_AIFF_IMA_ADPCM;
}
data_format = SNDLIB_UNSUPPORTED;
}
}
}
}
}
}
}
}
data_size = (frames*mus_format2bytes(data_format)*chans);
}
else
{
if (match_four_chars((unsigned char *)hdrbuf,I_SSND))
{
update_ssnd_location = offset+4;
data_location = mus_big_endian_int((unsigned char *)(hdrbuf+8)) + offset + 16; /* Baroque! */
/* offset is where the hdrbuf is positioned in the file, the sound data offset itself is at loc+8 and the */
/* 0-based location of the sound data is at the end of the chunk = 16 (8=header+4=offset+4=blocksize) */
/* the next int can be the block size if the data is block-aligned */
/* only one SSND per AIFF is allowed */
}
else
{
if ((match_four_chars((unsigned char *)hdrbuf,I_ANNO)) || (match_four_chars((unsigned char *)hdrbuf,I_COMT)))
{
j=0;
for (i=0;i<AUX_COMMENTS;i++) if (aux_comment_start[i] == 0) {j=i; break;}
if (j >= AUX_COMMENTS) {mus_error(MUS_HEADER_TOO_MANY_AUXILIARY_COMMENTS,"ran out of auxiliary comment space"); j=0;}
aux_comment_start[j] = offset+8;
if (match_four_chars((unsigned char *)hdrbuf,I_COMT)) aux_comment_start[j] += 8; /* skip time stamp and markerId (not ID, I assume!) */
aux_comment_end[j] = offset+7+chunksize;
}
else
{
if (match_four_chars((unsigned char *)hdrbuf,I_APPL))
{
if (match_four_chars((unsigned char *)(hdrbuf+8),I_MUS_))
{
/* my own chunk has the arbitrary length comment I use (actually the ASCII */
/* representation of a lisp program evaluated in the CLM package) to handle mix et al. */
/* It is nothing more than the actual string -- remember to pad to even length here. */
comment_start = offset + 12;
comment_end = comment_start + chunksize - 5;
}
}
else
{
if (match_four_chars((unsigned char *)hdrbuf,I_INST))
{
base_note = hdrbuf[8];
base_detune = hdrbuf[9];
loop_modes[0] = mus_big_endian_short((unsigned char *)(hdrbuf+16));
loop_starts[0] = mus_big_endian_short((unsigned char *)(hdrbuf+18));
loop_ends[0] = mus_big_endian_short((unsigned char *)(hdrbuf+20));
loop_modes[1] = mus_big_endian_short((unsigned char *)(hdrbuf+22));
loop_starts[1] = mus_big_endian_short((unsigned char *)(hdrbuf+24));
loop_ends[1] = mus_big_endian_short((unsigned char *)(hdrbuf+26));
/* these are mark numbers */
}
else
{
if (match_four_chars((unsigned char *)hdrbuf,I_MARK))
{
/* unsigned short #marks, each mark: id pos name (pstring damn it) */
num_marks = mus_big_endian_unsigned_short((unsigned char *)(hdrbuf+8));
if (num_marks > markers)
{
if (markers > 0) {if (marker_ids) FREE(marker_ids); if (marker_positions) FREE(marker_positions);}
markers = num_marks;
marker_ids = (int *)CALLOC(markers,sizeof(int));
marker_positions = (int *)CALLOC(markers,sizeof(int));
}
moff = 10;
for (m=0;m<num_marks;m++)
{
if (seek_and_read(chan,(unsigned char *)hdrbuf,offset+moff,8) > 0)
{
msize = read_aiff_marker(m,(unsigned char *)hdrbuf);
moff += msize;
}
}
}
}
}
}
}
}
chunkloc = (8+chunksize);
if (chunksize&1) chunkloc++; /* extra null appended to odd-length chunks */
if ((offset+chunkloc) >= update_form_size) happy=0;
}
if (true_file_length < data_size) data_size = true_file_length - data_location;
data_size = mus_bytes2samples(data_format,data_size);
return(0);
}
int mus_header_aiff_p(void) {return(type_specifier == mus_uninterpreted_int((unsigned char *)I_AIFF));}
static int write_aif_header (int chan, int srate, int chans, int siz, int format, char *comment, int len, int aifc_header)
{
/* we write the simplest possible AIFC header: AIFC | COMM | APPL-MUS_ if needed | SSND eof. */
/* the assumption being that we're going to be appending sound data once the header is out */
/* INST and MARK chunks added Jul-95 for various programs that expect them (MixView). */
int i,j,lenhdr,curend,extra,err=0; /* set aifc to 0 to get old-style AIFF header */
char *str;
lenhdr=0;
extra=0;
curend=0;
if (len != 0)
{
lenhdr = 12;
if ((len % 4) != 0)
extra = (4 - (len % 4));
}
write_four_chars((unsigned char *)hdrbuf,I_FORM);
if (aifc_header)
mus_set_big_endian_int((unsigned char *)(hdrbuf+4),len+30+16+38+siz+lenhdr+extra+12+10);
else mus_set_big_endian_int((unsigned char *)(hdrbuf+4),len+30+16+38+siz+lenhdr+extra);
/*
* comment length + 4 for AIFF 18+8 for I_COMM info + 16 for I_SSND info + 38 for INST and MARK +
* siz for data + 12 for comment header if any + padding == total size - 8 (i.e. FORM header).
* INST+MARK (38) added 3-Jul-95 for Notam software compatibility
*/
if (aifc_header)
{
write_four_chars((unsigned char *)(hdrbuf+8),I_AIFC);
write(chan,hdrbuf,12);
curend=12;
write_four_chars((unsigned char *)hdrbuf,I_FVER);
mus_set_big_endian_int((unsigned char *)(hdrbuf+4),4);
mus_set_big_endian_int((unsigned char *)(hdrbuf+8),0xA2805140);
}
else write_four_chars((unsigned char *)(hdrbuf+8),I_AIFF);
write_four_chars((unsigned char *)(hdrbuf+12),I_COMM);
if (aifc_header) mus_set_big_endian_int((unsigned char *)(hdrbuf+16),18+10); else mus_set_big_endian_int((unsigned char *)(hdrbuf+16),18);
mus_set_big_endian_short((unsigned char *)(hdrbuf+20),(short)chans);
mus_set_big_endian_int((unsigned char *)(hdrbuf+22),siz / (chans*mus_format2bytes(format)));
switch (format)
{
case SNDLIB_16_LINEAR: case SNDLIB_16_LINEAR_LITTLE_ENDIAN: case SNDLIB_16_UNSIGNED:
mus_set_big_endian_short((unsigned char *)(hdrbuf+26),16);
break;
case SNDLIB_24_LINEAR: case SNDLIB_24_LINEAR_LITTLE_ENDIAN:
mus_set_big_endian_short((unsigned char *)(hdrbuf+26),24);
break;
case SNDLIB_32_LINEAR: case SNDLIB_32_LINEAR_LITTLE_ENDIAN: case SNDLIB_32_FLOAT:
mus_set_big_endian_short((unsigned char *)(hdrbuf+26),32);
break;
case SNDLIB_64_DOUBLE:
mus_set_big_endian_short((unsigned char *)(hdrbuf+26),64);
break;
case SNDLIB_8_LINEAR: case SNDLIB_8_UNSIGNED:
mus_set_big_endian_short((unsigned char *)(hdrbuf+26),8);
break;
case SNDLIB_8_MULAW:
mus_set_big_endian_short((unsigned char *)(hdrbuf+26),8);
break;
case SNDLIB_8_ALAW:
mus_set_big_endian_short((unsigned char *)(hdrbuf+26),8);
break;
default:
mus_error(MUS_UNSUPPORTED_DATA_FORMAT,"can't write AIFF data format: %d",format);
err = -1;
break;
}
double_to_ieee_80((double)srate,(unsigned char *)(hdrbuf+28));
if (aifc_header)
{
switch (format)
{
case SNDLIB_16_LINEAR: case SNDLIB_24_LINEAR: case SNDLIB_32_LINEAR: case SNDLIB_8_LINEAR: str = (char *)I_NONE; break;
case SNDLIB_16_LINEAR_LITTLE_ENDIAN: case SNDLIB_24_LINEAR_LITTLE_ENDIAN: case SNDLIB_32_LINEAR_LITTLE_ENDIAN: str = (char *)I_sowt; break;
case SNDLIB_32_FLOAT: str = (char *)I_fl32; break;
case SNDLIB_64_DOUBLE: str = (char *)I_fl64; break;
case SNDLIB_8_UNSIGNED: case SNDLIB_16_UNSIGNED: str = (char *)I_raw_; break;
case SNDLIB_8_MULAW: str = (char *)I_ulaw; break;
case SNDLIB_8_ALAW: str = (char *)I_alaw; break;
default: str = (char *)I_NONE; break;
}
write_four_chars((unsigned char *)(hdrbuf+38),(const unsigned char *)str);
(*(unsigned char *)(hdrbuf+42)) = 4; /* final pad null not accounted-for */
write_four_chars((unsigned char *)(hdrbuf+43),(const unsigned char *)str);
(*(unsigned char *)(hdrbuf+47)) = 0;
i=48;
}
else i = 38;
if (len != 0)
{
if (aifc_header)
{
write_four_chars((unsigned char *)(hdrbuf+48),I_APPL);
mus_set_big_endian_int((unsigned char *)(hdrbuf+52),len+4+extra);
write_four_chars((unsigned char *)(hdrbuf+56),I_MUS_);
i = 60;
}
else
{
write_four_chars((unsigned char *)(hdrbuf+38),I_APPL);
mus_set_big_endian_int((unsigned char *)(hdrbuf+42),len+4+extra);
write_four_chars((unsigned char *)(hdrbuf+46),I_MUS_);
i = 50;
}
for (j=0;j<len;j++)
{
if (i == HDRBUFSIZ)
{
curend += HDRBUFSIZ;
write(chan,hdrbuf,HDRBUFSIZ);
i=0;
}
hdrbuf[i]=comment[j];
i++;
}
if (extra != 0)
{
if ((i+extra) > HDRBUFSIZ)
{
curend += i;
write(chan,hdrbuf,i);
i=0;
}
for (j=0;j<extra;j++)
{
hdrbuf[i] = 0;
i++;
}
}
}
curend += i;
write(chan,hdrbuf,i);
write_four_chars((unsigned char *)hdrbuf,I_MARK); /* SoundHack includes a blank MARK chunk for some reason */
mus_set_big_endian_int((unsigned char *)(hdrbuf+4),2);
mus_set_big_endian_short((unsigned char *)(hdrbuf+8),0);
write_four_chars((unsigned char *)(hdrbuf+10),I_INST);
mus_set_big_endian_int((unsigned char *)(hdrbuf+14),20);
mus_set_big_endian_int((unsigned char *)(hdrbuf+18),0x3c00007f); /* base-note=middle C, detune=0, lownote=0, highnote=0x7f */
mus_set_big_endian_int((unsigned char *)(hdrbuf+22),0x017f0000); /* lowvelocity=1, highvelocity=0x7f, gain=0 */
mus_set_big_endian_int((unsigned char *)(hdrbuf+26),0); /* no loops */
mus_set_big_endian_int((unsigned char *)(hdrbuf+30),0);
mus_set_big_endian_int((unsigned char *)(hdrbuf+34),0);
write_four_chars((unsigned char *)(hdrbuf+38),I_SSND);
mus_set_big_endian_int((unsigned char *)(hdrbuf+38+4),siz+8);
mus_set_big_endian_int((unsigned char *)(hdrbuf+38+8),0); /* "offset" */
mus_set_big_endian_int((unsigned char *)(hdrbuf+38+12),0); /* "blocksize " */
data_location = 38+16+curend;
write(chan,hdrbuf,38+16);
return(err);
}
static void update_aiff_header (int chan, int siz)
{
/* we apparently have to make sure the form size and the data size are correct
* assumed here that we'll only be updating our own AIFF files
* There are 3 such locations -- the 2nd word of the file which is the overall form size,
* the frames variable in the COMM chunk, and the chunk-size variable in the SSND chunk
* an unexpected hassle for CLM is that we can open/close the output file many times if running mix,
* so we have to update the various size fields taking into account the old size
*/
read(chan,hdrbuf,INITIAL_READ_SIZE);
read_aiff_header(chan,0);
lseek(chan,4L,SEEK_SET);
mus_set_big_endian_int((unsigned char *)hdrbuf,siz+update_form_size-mus_samples2bytes(data_format,data_size));
/* cancel old data_size from previous possible write */
write(chan,hdrbuf,4);
lseek(chan,update_frames_location,SEEK_SET);
mus_set_big_endian_int((unsigned char *)hdrbuf,siz/(chans*mus_format2bytes(data_format)));
write(chan,hdrbuf,4);
lseek(chan,update_ssnd_location,SEEK_SET);
mus_set_big_endian_int((unsigned char *)hdrbuf,siz+8);
write(chan,hdrbuf,4);
}
static void update_aiff_header_comment (int chan, char *comment, int len)
{
/* save-stats in CLM appends a comment after the sound data has been written */
int i,j,true_len,old_len;
if (len&1) true_len = len+1; else true_len=len;
lseek(chan,0L,SEEK_END);
write_four_chars((unsigned char *)hdrbuf,I_ANNO);
mus_set_big_endian_int((unsigned char *)(hdrbuf+4),len);
for (i=0,j=8;i<len;i++,j++) hdrbuf[j]=comment[i];
write(chan,hdrbuf,8+true_len);
lseek(chan,4L,SEEK_SET);
read(chan,hdrbuf,4);
old_len = mus_big_endian_int((unsigned char *)hdrbuf);
mus_set_big_endian_int((unsigned char *)hdrbuf,old_len+true_len+8);
lseek(chan,4L,SEEK_SET);
write(chan,hdrbuf,4);
}
/* ------------------------------------ RIFF (wave) ------------------------------------
*
* see ftp.microsoft.com:/SoftLib/MSLFILES/MDRK.EXE (also MMSYSTEM.H and MMREG.H)
* ftp://ftp.isi.edu/in-notes/rfc2361.txt
*
* 0: "RIFF" (little-endian) or "RIFX" (big-endian)
* 4: size
* 8: "WAVE" ("RMID" = midi data, others are AVI, CPPO, ACON etc)
* AVI chunk can include audio data
*
* rest very similar to AIFF (odd-sized chunks are padded)
*
* fmt 0: format code (see below)
* 2: chans
* 4: srate (long)
* 8: average rate "for buffer estimation"
* 12: alignment "block size"
* 14: data size (bits per sample) (PCM only)
* 16: count (bytes) of extra info in the header (i.e. trailing info added to this basic header)
* 20: samples per block (short) in dvi_adpcm
*
* formats are: 0: unknown, 1: PCM, 2: ADPCM, 3: IEEE float, 4: VSELP, 5: IBM_CVSD, 6: alaw, 7: mulaw
* 0x10: OKI_ADPCM, 0x11: DVI_ADPCM, 0x12: MediaSpace_ADPCM,
* 0x13: Sierra_ADPCM, 0x14: G723_ADPCM, 0x15: DIGISTD, 0x16: DIGIFIX, 0x17: Dialogic ADPCM,
* 0x18: Mediavision ADPCM, 0x19: HP cu codec,
* 0x20: Yamaha_ADPCM, 0x21: SONARC, 0x22: DSPGroup_TrueSpeech
* 0x23: EchoSC1, 0x24: AudioFile_AF36, 0x25: APTX, 0x26: AudioFile_AF10
* 0x27: prosody 1612, 0x28: lrc,
* 0x30: Dolby_Ac2, 0x31: GSM610, 0x32: MSN audio codec, 0x33: Antext_ADPCM, 0x34: Control_res_vqlpc,
* 0x35: DIGIREAL, 0x36: DIGIADPCM, 0x37: Control_res_cr10, 0x38: NMS_VBXADPCM, 0x39:Roland rdac,
* 0x3a: echo sc3, 0x3b: Rockwell adpcm, 0x3c: Rockwell digitalk codec, 0x3d: Xebec,
* 0x40: G721_ADPCM, 0x41: G728 CELP, 0x42: MS G723, 0x50: MPEG,
* 0x52: RT24, 0x53: PAC, 0x55: Mpeg layer 3, 0x59: Lucent G723, 0x60: Cirrus,
* 0x61: ESS Tech pcm, 0x62: voxware (obsolete), 0x63: canopus atrac,
* 0x64: G726, 0x65: G722, 0x66: DSAT, 0x67: DSAT display,
* 0x69: voxware (obsolete), 0x70: voxware ac8 (obsolete), 0x71: voxware ac10 (obsolete),
* 0x72: voxware ac16 (obsolete), 0x73: voxware ac20 (obsolete), 0x74: voxware rt24,
* 0x75: voxware rt29, 0x76: voxware rt29hw (obsolete), 0x77: voxware vr12 (obsolete),
* 0x78: voxware vr18 (obsolete), 0x79: voxware tq40 (obsolete),
* 0x80: softsound, 0x81: voxware tq60 (obsolete), 0x82: MS RT24, 0x83: G729A,
* 0x84: MVI_MVI2, 0x85: DF G726, 0x86: DF GSM610, 0x88: isaudio, 0x89: onlive,
* 0x91: sbc24, 0x92: dolby ac3 spdif, 0x97: zyxel adpcm, 0x98: philips lpcbb,
* 0x99: packed, 0x100: rhetorex adpcm,
* 0x101: Irat, 0x102: IBM_alaw?, 0x103: IBM_ADPCM?,
* 0x111: vivo G723, 0x112: vivo siren, 0x123: digital g273
* 0x200: Creative_ADPCM, 0x202: Creative fastspeech 8, 0x203: Creative fastspeech 10,
* 0x220: quarterdeck, 0x300: FM_TOWNS_SND, 0x400: BTV digital, 0x680: VME vmpcm,
* 0x1000: OLIGSM, 0x1001: OLIADPCM, 0x1002: OLICELP, 0x1003: OLISBC, 0x1004: OLIOPR
* 0x1100: LH codec, 0x1400: Norris, 0x1401: isaudio, 0x1500: Soundspace musicompression, 0x2000: DVM
* (see http://www.microsoft.com/asf/resources/draft-ietf-fleischman-codec-subtree-00.txt)
*
* RIFF and LIST chunks have nested chunks. Registered chunk names include:
* LIST with subchunks, one of which can be:
* INFO itself containing:
* IARL: archival location, IART: artist, ICMS: commissioned, ICMT: comment, ICOP: copyright, ICRD: creation date,
* ICRP: uh...cropped, IDIM: dimensions, IDPI: dpi, IENG: engineer, IGNR: genre, IKEY: keywords, ILGT: lightness,
* IMED: medium, INAM: name, IPLT: palette, IPRD: product, ISBJ: subject, ISFT: software, ISHP: sharpness,
* ISRC: source, ISRF: source form, ITCH: technician, ISMP: SMPTE time code, IDIT: digitization time
*
* data chunk has the samples
* other (currently ignored) chunks are wavl = waveform data, fact, cues of some sort, slnt = silence,
* plst = playlist, adtl = associated data list, labl = cue label, note = cue comments,
* ltxt = text associated with data segment (cue), file, DISP = displayable object,
* JUNK = outdated info, PAD = padding, etc
* fact chunk generally has number of samples (used in compressed files)
*/
static int wave_to_sndlib_format(int osf, int bps, int little)
{
switch (osf)
{
case 1:
switch (bps)
{
case 8: return(SNDLIB_8_UNSIGNED); break;
case 16: if (little) return(SNDLIB_16_LINEAR_LITTLE_ENDIAN); else return(SNDLIB_16_LINEAR); break;
case 32: if (little) return(SNDLIB_32_LINEAR_LITTLE_ENDIAN); else return(SNDLIB_32_LINEAR); break;
case 24: if (little) return(SNDLIB_24_LINEAR_LITTLE_ENDIAN); else return(SNDLIB_24_LINEAR); break;
default: return(SNDLIB_8_UNSIGNED); break;
}
break;
case 3: if (little) return(SNDLIB_32_FLOAT_LITTLE_ENDIAN); else return(SNDLIB_32_FLOAT); break;
case 6: if (bps == 8) return(SNDLIB_8_ALAW); break;
case 7: if (bps == 8) return(SNDLIB_8_MULAW); break;
/* IBM mulaw follows G711 specs like other versions (this info direct from IBM) */
case 0x101: return(SNDLIB_8_MULAW); break;
case 0x102: return(SNDLIB_8_ALAW); break;
}
return(SNDLIB_UNSUPPORTED);
}
static int read_riff_header (int chan)
{
/* we know we have checked for RIFF xxxx WAVE when we arrive here */
int chunksize,offset,chunkloc,happy,little;
little = 1;
if (match_four_chars((unsigned char *)hdrbuf,I_RIFX)) little=0; /* big-endian data in this case, but I've never seen one */
type_specifier = mus_uninterpreted_int((unsigned char *)(hdrbuf+8));
chunkloc = 12;
offset = 0;
header_distributed = 1;
data_format = SNDLIB_UNSUPPORTED;
srate = 0;
chans = 0;
happy = 1;
data_size = 0;
fact_samples = 0;
bits_per_sample = 0;
true_file_length = lseek(chan,0L,SEEK_END);
update_form_size = big_or_little_endian_int((unsigned char *)(hdrbuf+4),little);
while (happy)
{
offset += chunkloc;
if (seek_and_read(chan,(unsigned char *)hdrbuf,offset,32) <= 0)
{
mus_error(MUS_HEADER_READ_FAILED,"RIFF header chunks confused at %d",offset);
return(-1);
}
chunksize = big_or_little_endian_int((unsigned char *)(hdrbuf+4),little);
if (match_four_chars((unsigned char *)hdrbuf,I_fmt_))
{
/*
* 8: short format code --1=PCM for example
* 10: short chans --1
* 12: long rate --48000 (0xbb80)
* 16: long ave rate --65655 (0x10077)
* 20: short align --2
* 22: short data size (bits) --16
* 24: bytes of extra
* ... some extra data dependent on format
*
* R I F F # # # # W A V E f m t sp
* 5249 4646 f851 0500 5741 5645 666d 7420
* e40f 0000 0100 0100 80bb 0000 0077 0100
* 0200 1000 0000 0000 0000 0000 0000 0000
*
* #x000551f8 = 348664 = size in bytes - 8
* #x00000fe4 = 4068 [fmt_ chunk size?]
*/
original_data_format = big_or_little_endian_short((unsigned char *)(hdrbuf+8),little);
chans = big_or_little_endian_short((unsigned char *)(hdrbuf+10),little);
srate = big_or_little_endian_int((unsigned char *)(hdrbuf+12),little);
block_align = big_or_little_endian_short((unsigned char *)(hdrbuf+20),little);
bits_per_sample = big_or_little_endian_short((unsigned char *)(hdrbuf+22),little);
data_format = wave_to_sndlib_format(original_data_format,bits_per_sample,little);
}
else
{
if (match_four_chars((unsigned char *)hdrbuf,I_data))
{
update_ssnd_location = offset+4;
data_location = offset + 8;
data_size = big_or_little_endian_int((unsigned char *)(hdrbuf+4),little);
happy = 0;
}
else
{
if (match_four_chars((unsigned char *)hdrbuf,I_fact))
{
fact_samples = big_or_little_endian_int((unsigned char *)(hdrbuf+8),little);
}
else
{
if (match_four_chars((unsigned char *)hdrbuf,I_inst))
{
base_note = hdrbuf[8];
base_detune = hdrbuf[9];
/* rest is gain low-note high-note low-velocity high-velocity */
}
else
{
if (match_four_chars((unsigned char *)hdrbuf,I_clm_))
{
comment_start = offset + 8;
comment_end = comment_start + chunksize - 1; /* end of comment not start of next chunk */
}
}
}
}
}
chunkloc = (8+chunksize);
if (chunksize&1) chunkloc++; /* extra null appended to odd-length chunks */
}
if (true_file_length < data_size) data_size = true_file_length - data_location;
data_size = mus_bytes2samples(data_format,data_size);
return(0);
}
static int write_riff_header (int chan, int srate, int chans, int siz, int format, char *comment, int len)
{
int offset,i,j,lenhdr,extra,curend,err=0;
lenhdr=0;
extra=0;
if (len != 0)
{
lenhdr = 12;
if ((len % 4) != 0)
extra = (4 - (len % 4));
}
write_four_chars((unsigned char *)hdrbuf,I_RIFF);
mus_set_little_endian_int((unsigned char *)(hdrbuf+4),len+36+siz+lenhdr+extra);
write_four_chars((unsigned char *)(hdrbuf+8),I_WAVE);
write_four_chars((unsigned char *)(hdrbuf+12),I_fmt_);
mus_set_little_endian_int((unsigned char *)(hdrbuf+16),24-8);
switch (format)
{
case SNDLIB_8_MULAW:
mus_set_little_endian_short((unsigned char *)(hdrbuf+20),7);
mus_set_little_endian_short((unsigned char *)(hdrbuf+34),8);
break;
case SNDLIB_8_ALAW:
mus_set_little_endian_short((unsigned char *)(hdrbuf+20),6);
mus_set_little_endian_short((unsigned char *)(hdrbuf+34),8);
break;
case SNDLIB_8_UNSIGNED:
mus_set_little_endian_short((unsigned char *)(hdrbuf+20),1);
mus_set_little_endian_short((unsigned char *)(hdrbuf+34),8);
break;
case SNDLIB_16_LINEAR_LITTLE_ENDIAN:
mus_set_little_endian_short((unsigned char *)(hdrbuf+20),1);
mus_set_little_endian_short((unsigned char *)(hdrbuf+34),16);
break;
case SNDLIB_24_LINEAR_LITTLE_ENDIAN:
mus_set_little_endian_short((unsigned char *)(hdrbuf+20),1);
mus_set_little_endian_short((unsigned char *)(hdrbuf+34),24);
break;
case SNDLIB_32_LINEAR_LITTLE_ENDIAN:
mus_set_little_endian_short((unsigned char *)(hdrbuf+20),1);
mus_set_little_endian_short((unsigned char *)(hdrbuf+34),32);
break;
case SNDLIB_32_FLOAT_LITTLE_ENDIAN:
mus_set_little_endian_short((unsigned char *)(hdrbuf+20),3);
mus_set_little_endian_short((unsigned char *)(hdrbuf+34),32);
break;
default:
mus_error(MUS_UNSUPPORTED_DATA_FORMAT,"can't write RIFF data format: %d",format);
err = -1;
break;
}
mus_set_little_endian_short((unsigned char *)(hdrbuf+22),(short)chans);
mus_set_little_endian_int((unsigned char *)(hdrbuf+24),srate);
mus_set_little_endian_int((unsigned char *)(hdrbuf+28),srate * chans * mus_format2bytes(format)); /* added chans 10-Mar-99 */
mus_set_little_endian_short((unsigned char *)(hdrbuf+32),(short)(chans * mus_format2bytes(format)));
offset = 36;
i = 36;
curend = 0;
if (len != 0)
{
offset += len+12;
write_four_chars((unsigned char *)(hdrbuf+36),I_clm_);
mus_set_little_endian_int((unsigned char *)(hdrbuf+40),len+extra);
i = 44;
for (j=0;j<len;j++)
{
if (i == HDRBUFSIZ)
{
curend += HDRBUFSIZ;
write(chan,hdrbuf,HDRBUFSIZ);
i=0;
}
hdrbuf[i]=comment[j];
i++;
}
if (extra != 0)
{
if ((i+extra) > HDRBUFSIZ)
{
curend += i;
write(chan,hdrbuf,i);
i=0;
}
for (j=0;j<extra;j++)
{
hdrbuf[i] = 0;
i++;
}
}
}
curend += i;
write(chan,hdrbuf,i);
write_four_chars((unsigned char *)hdrbuf,I_data);
mus_set_little_endian_int((unsigned char *)(hdrbuf+4),siz);
data_location = 8+curend;
write(chan,hdrbuf,8);
return(err);
}
static void update_riff_header (int chan, int siz)
{
read(chan,hdrbuf,INITIAL_READ_SIZE);
read_riff_header(chan);
lseek(chan,4L,SEEK_SET);
mus_set_little_endian_int((unsigned char *)hdrbuf,(siz+update_form_size-mus_samples2bytes(data_format,data_size)));
/* see update_aiff_header for explanation */
write(chan,hdrbuf,4);
lseek(chan,update_ssnd_location,SEEK_SET);
mus_set_little_endian_int((unsigned char *)hdrbuf,siz);
write(chan,hdrbuf,4);
}
static void update_riff_header_comment (int chan, char *comment, int len)
{
/* save-stats in CLM appends a comment after the sound data has been written */
int i,j,true_len,old_len;
if (len&1) true_len = len+1; else true_len=len;
lseek(chan,0L,SEEK_END);
write_four_chars((unsigned char *)hdrbuf,I_INFO);
mus_set_little_endian_int((unsigned char *)(hdrbuf+4),len);
for (i=0,j=8;i<len;i++,j++) hdrbuf[j]=comment[i];
write(chan,hdrbuf,8+true_len);
lseek(chan,4L,SEEK_SET);
read(chan,hdrbuf,4);
old_len = mus_little_endian_int((unsigned char *)hdrbuf);
mus_set_little_endian_int((unsigned char *)hdrbuf,old_len+true_len+8);
lseek(chan,4L,SEEK_SET);
write(chan,hdrbuf,4);
}
/* ------------------------------------ AVI ------------------------------------
* actually a video format, but it sometimes contains embedded 'wave' data
*
* RIFF xxx AVI
* <various LISTs>
* LIST xxxx hdr1 LIST strl(?) strh | strf | strn etc
* strf is the WAVE header starting with the sound format
* LIST xxxx movi ##db|##wb -- wb subblocks have the audio data (these need to be collected as a single stream)
* there are many complications that we make no effort to handle here
*
* described in http://www.rahul.net/jfm/avi.html
*/
static int read_avi_header (int chan)
{
/* we know we have checked for RIFF xxxx AVI when we arrive here */
int chunksize,offset,chunkloc,happy,cksize,rdsize,ckoff,cktotal;
int cksizer,ckoffr,cktotalr,bits;
type_specifier = mus_uninterpreted_int((unsigned char *)(hdrbuf+8));
chunkloc = 12;
offset = 0;
header_distributed = 1;
data_format = SNDLIB_UNSUPPORTED;
srate = 0;
chans = 1;
happy = 1;
data_size = 0;
data_location = 0;
true_file_length = lseek(chan,0L,SEEK_END);
while (happy)
{
offset += chunkloc;
if (seek_and_read(chan,(unsigned char *)hdrbuf,offset,32) <= 0)
{
mus_error(MUS_HEADER_READ_FAILED,"AVI header chunks confused at %d",offset);
return(-1);
}
chunksize = mus_little_endian_int((unsigned char *)(hdrbuf+4));
if (match_four_chars((unsigned char *)hdrbuf,I_LIST))
{
ckoff = offset+12;
cktotal = 12;
if (match_four_chars((unsigned char *)(hdrbuf+8),I_movi))
{
while (cktotal < chunksize)
{
lseek(chan,ckoff,SEEK_SET);
read(chan,hdrbuf,8);
cksize = mus_little_endian_int((unsigned char *)(hdrbuf+4));
if ((hdrbuf[2] == 'w') && (hdrbuf[3] == 'b'))
{
data_location = ckoff;
if (srate != 0) happy=0;
break;
}
ckoff += (8+cksize);
cktotal += (8+cksize);
}
}
else
{
while (cktotal < chunksize)
{
lseek(chan,ckoff,SEEK_SET);
read(chan,hdrbuf,8);
cksize = mus_little_endian_int((unsigned char *)(hdrbuf+4));
ckoff += (8+cksize);
cktotal += (8+cksize);
if (match_four_chars((unsigned char *)hdrbuf,I_LIST))
{
ckoffr = ckoff+12;
cktotalr = 12;
while (cktotalr < cksize)
{
lseek(chan,ckoffr,SEEK_SET);
read(chan,hdrbuf,8);
cksizer = mus_little_endian_int((unsigned char *)(hdrbuf+4));
ckoffr += (8+cksizer);
cktotalr += (8+cksizer);
if (match_four_chars((unsigned char *)hdrbuf,I_strf))
{
if (cksizer < HDRBUFSIZ) rdsize = cksizer; else rdsize = HDRBUFSIZ;
read(chan,hdrbuf,rdsize);
original_data_format = mus_little_endian_short((unsigned char *)hdrbuf);
chans = mus_little_endian_short((unsigned char *)(hdrbuf+2));
srate = mus_little_endian_int((unsigned char *)(hdrbuf+4));
/* block_align = mus_little_endian_short((unsigned char *)(hdrbuf+12)); */
bits = mus_little_endian_short((unsigned char *)(hdrbuf+14));
/* only 16 bit linear little endian for now */
if ((bits == 16) && (original_data_format == 1))
original_data_format = SNDLIB_16_LINEAR_LITTLE_ENDIAN;
if (data_location != 0) happy = 0;
break;
}
}
}
}
}
}
chunkloc = (8+chunksize);
if (chunksize&1) chunkloc++; /* extra null appended to odd-length chunks */
}
return(0);
}
/* ------------------------------------ SoundFont 2.0 ------------------------------------
*
* Emu's SoundFont(tm) format uses RIFF -- at ftp.creaf.com:/pub/emu/sf2_00a.ps)
*
* RIFF xxxx sfbk followed by
* LIST xxxx INFO chunk (nothing of interest -- icmt subchunk might have comments)
* LIST xxxx sdta chunk = data
* smpl chunk (16 bit linear little-endian)
* LIST xxxx pdta list chunk
* shdr subchunk has srate at 40 (int), samples at 28
*/
static int read_soundfont_header (int chan)
{
/* we know we have checked for RIFF xxxx sfbk when we arrive here */
int chunksize,offset,chunkloc,happy,type,cksize,rdsize,ckoff;
type_specifier = mus_uninterpreted_int((unsigned char *)(hdrbuf+8));
chunkloc = 12;
offset = 0;
header_distributed = 1;
data_format = SNDLIB_16_LINEAR_LITTLE_ENDIAN;
srate = 0;
chans = 1; /* to hell with soundfont stereo */
happy = 1;
data_size = 0;
data_location = 0;
true_file_length = lseek(chan,0L,SEEK_END);
while (happy)
{
offset += chunkloc;
if (seek_and_read(chan,(unsigned char *)hdrbuf,offset,32) <= 0)
{
mus_error(MUS_HEADER_READ_FAILED,"SoundFont header chunks confused at %d",offset);
return(-1);
}
chunksize = mus_little_endian_int((unsigned char *)(hdrbuf+4));
if (match_four_chars((unsigned char *)hdrbuf,I_LIST))
{
/* everything is squirreled away in LIST chunks in this format */
if (match_four_chars((unsigned char *)(hdrbuf+8),I_pdta))
{
/* go searching for I_shdr -- headers this complicated should be illegal. */
ckoff = offset+12;
lseek(chan,ckoff,SEEK_SET);
while (!srate)
{
read(chan,hdrbuf,8);
cksize = mus_little_endian_int((unsigned char *)(hdrbuf+4));
ckoff += (8+cksize);
/* here we need to jump over subchunks! -- 4-Aug-97 */
if (match_four_chars((unsigned char *)hdrbuf,I_shdr))
{
if (cksize < HDRBUFSIZ) rdsize = cksize; else rdsize = HDRBUFSIZ;
read(chan,hdrbuf,rdsize);
data_size = 2 * mus_little_endian_int((unsigned char *)(hdrbuf+24));
srate = mus_little_endian_int((unsigned char *)(hdrbuf+36));
type = mus_little_endian_short((unsigned char *)(hdrbuf+44));
if (type != 1)
{
mus_error(MUS_UNSUPPORTED_DATA_FORMAT,"can't handle this SoundFont data type: %x",type);
return(-1);
}
happy = (!data_location);
}
else lseek(chan,ckoff,SEEK_SET);
}
}
else
{
if (match_four_chars((unsigned char *)(hdrbuf+8),I_sdta))
{
/* assume smpl follows + subchunk size */
/* Convert 1.4 appears to create a separate smpl chunk */
data_location = offset+20; /* LIST xxxx sdta smpl xxxx ... */
happy = (!srate);
}
}
}
chunkloc = (8+chunksize);
if (chunksize&1) chunkloc++; /* extra null appended to odd-length chunks */
}
if (true_file_length < data_size) data_size = true_file_length - data_location;
data_size = mus_bytes2samples(data_format,data_size);
return(0);
}
/* ------------------------------------ NIST ------------------------------------
*
* code available in ogitools-v1.0.tar.gz at svr-ftp.eng.cam.ac.uk:comp.speech/sources
*
* 0: "NIST_1A"
* 8: data_location as ASCII representation of integer (apparently always " 1024")
* 16: start of complicated header -- see below for details
*
* The most recent version of the SPHERE package is available
* via anonymous ftp from jaguar.ncsl.nist.gov [129.6.48.157] in the pub directory
* in compressed tar form as "sphere-v.tar.Z" (where "v" is the version
* code 2.6a last I looked). shortpack is also at this site.
*
* here's an example:
*
* NIST_1A
* 1024
* database_id -s5 TIMIT
* database_version -s3 1.0
* utterance_id -s8 aks0_sa1
* channel_count -i 1
* sample_count -i 63488
* sample_rate -i 16000
* sample_min -i -6967
* sample_max -i 7710
* sample_n_bytes -i 2
* sample_byte_format -s2 01
* sample_sig_bits -i 16
* end_head
*
* the sample_byte_format can be "10"=big-endian or "01"=little-endian, or "shortpack-v0"=compressed via shortpack
* other formats are wavpack and shorten.
*
* another field is 'sample_coding' which can be pcm (i.e. linear), 'pcm,embedded-shorten-v1.09', mu-law, ulaw, pculaw etc --
* so unpredictable as to be totally useless. This means we sometimes try to decode shorten-encoded files because
* we ignore this field. And worse, there's a 'channels_interleaved' field that (apparently) can be FALSE. Tough.
*/
#define MAX_FIELD_LENGTH 80
static int decode_nist_value (char *str,int base,int end)
{
/* can be -i -r or -snnn where nnn=ascii rep of integer = len of string (!) */
/* we'll deal only with integer fields (and well-behaved string fields) */
int i,j;
char value[MAX_FIELD_LENGTH];
i=base;
while ((i<end) && (i<MAX_FIELD_LENGTH) && (str[i] != '-')) i++; /* look for -i or whatever */
while ((i<end) && (i<MAX_FIELD_LENGTH) && (str[i] != ' ')) i++; /* look for space after it */
i++;
if (i>=MAX_FIELD_LENGTH) return(0);
for (j=0;i<end;j++,i++)
value[j]=str[i];
value[j]=0;
if (value[0]=='s') return(SNDLIB_NIST_shortpack);
sscanf(value,"%d",&i);
return(i);
}
static int read_nist_header (int chan)
{
char str[MAX_FIELD_LENGTH],name[MAX_FIELD_LENGTH];
int happy = 1;
int k,hend,curbase,j,n,nm,samples,bytes,byte_format;
type_specifier = mus_uninterpreted_int((unsigned char *)hdrbuf); /* the actual id is "NIST_1A" */
for (k=8;k<16;k++) str[k-8]=hdrbuf[k];
sscanf(str,"%d",&data_location); /* always "1024" */
n = 16;
hend = INITIAL_READ_SIZE;
k=0;
curbase = 0;
samples = 0;
bytes = 0;
srate = 0;
chans = 0;
comment_start = 16;
comment_end = 16;
byte_format = 10;
for (j=0;j<MAX_FIELD_LENGTH;j++) str[j]=' ';
while (happy)
{
/* much as in xIFF files, march through the file looking for the data we're after */
/* in this case we munch a character at a time... */
str[k] = hdrbuf[n];
if ((((str[k] == '\0') || (str[k] == '\n')) || ((curbase+n+1) >= data_location)) || (k == 79))
{
/* got a complete record (assuming no embedded newlines, of course) */
/* now look for a record we care about and decode it */
nm = 0;
while ((str[nm] != ' ') && (nm < MAX_FIELD_LENGTH))
{
name[nm] = str[nm];
nm++;
}
if (nm >= MAX_FIELD_LENGTH)
{
header_type = raw_sound_file;
data_format = SNDLIB_UNSUPPORTED;
mus_error(MUS_UNSUPPORTED_HEADER_TYPE,"invalid NIST header?");
return(-1);
}
name[nm]=0;
if (strcmp(name,"sample_rate") == 0) srate = decode_nist_value(str,nm,k); else
if (strcmp(name,"channel_count") == 0) chans = decode_nist_value(str,nm,k); else
if (strcmp(name,"end_head") == 0) {happy = 0; comment_end=curbase+n-1;} else
if (strcmp(name,"sample_count") == 0) samples = decode_nist_value(str,nm,k); else
if ((bytes == 0) && (strcmp(name,"sample_n_bytes") == 0)) bytes = decode_nist_value(str,nm,k); else
if ((bytes == 0) && (strcmp(name,"sample_sig_bits") == 0)) {bytes = decode_nist_value(str,nm,k); bytes = (bytes>>3);} else
if (strcmp(name,"sample_byte_format") == 0) byte_format = decode_nist_value(str,nm,k); else
if (strcmp(name,"end_head") == 0) happy = 0;
for (j=0;j<=k;j++) str[j]=' ';
k=0;
if ((curbase+n+1) > 1024) happy=0;
}
else
k++;
n++;
if (n >= hend)
{
curbase += hend;
n = 0;
read(chan,hdrbuf,HDRBUFSIZ);
hend = HDRBUFSIZ;
}
}
data_size = samples*bytes;
if (byte_format == SNDLIB_NIST_shortpack)
{
data_format = SNDLIB_UNSUPPORTED;
original_data_format = SNDLIB_NIST_shortpack;
}
else
{
if (bytes == 2)
{
if (byte_format == 10)
data_format = SNDLIB_16_LINEAR;
else data_format = SNDLIB_16_LINEAR_LITTLE_ENDIAN;
}
else data_format = SNDLIB_8_MULAW;
}
data_size = mus_bytes2samples(data_format,data_size);
header_distributed = 1;
return(0);
}
static int write_nist_header (int chan, int srate, int chans, int siz, int format)
{
char *header;
int datum;
datum = mus_format2bytes(format);
header = (char *)CALLOC(1024,sizeof(char));
sprintf(header,"NIST_1A\n 1024\nchannel_count -i %d\nsample_rate -i %d\nsample_n_bytes -i %d\nsample_byte_format -s2 %s\nsample_sig_bits -i %d\nsample_count -i %d\nend_head\n",
chans,srate,datum,
((format == SNDLIB_16_LINEAR) || (format == SNDLIB_24_LINEAR) || (format == SNDLIB_32_LINEAR)) ? "10" : "01",
datum* 8,siz/datum);
#ifndef MACOS
write(chan,(unsigned char *)header,1024);
#else
write(chan,header,1024);
#endif
data_location = 1024;
FREE(header);
return(0);
}
static void update_nist_header (int chan, int siz)
{
lseek(chan,0L,SEEK_SET);
read(chan,hdrbuf,INITIAL_READ_SIZE);
read_nist_header(chan);
lseek(chan,0L,SEEK_SET);
write_nist_header(chan,mus_header_srate(),mus_header_chans(),siz,mus_header_format());
}
/* ------------------------------------ BICSF ------------------------------------
* (actually, this is EBICSF and the old BICSF is called IRCAM below)
*
* 0-28: NeXT-compatible header, read by read_next_header above.
* 28: bicsf magic number (107364 or trouble)
* 32: srate as a 32-bit float
* 36: chans
* 40: data format indicator (2=16-bit linear, 4=32-bit float)
* 44: begin chunks, if any
*
* followed by AIFF-style chunked header info with chunks like:
*
* COMM size comment
* MAXA size {max amps (up to 4)} (frame offsets) time-tag unix msec counter
* CUE, PRNT, ENV etc
*/
static int read_bicsf_header (int chan)
{
int chunksize,chunkname,offset,chunkloc,happy;
type_specifier = mus_uninterpreted_int((unsigned char *)(hdrbuf+28));
header_type = BICSF_sound_file;
data_location = 1024;
if (data_size == 0) data_size = (true_file_length-data_location);
lseek(chan,40,SEEK_SET);
read(chan,hdrbuf,HDRBUFSIZ);
original_data_format = mus_big_endian_int((unsigned char *)hdrbuf);
switch (original_data_format)
{
case 2: data_format = SNDLIB_16_LINEAR; break;
case 4: data_format = SNDLIB_32_FLOAT; break;
default: break;
}
/* now check for a COMM chunk, setting the comment pointers */
chunkloc = 4; /* next header + magic number, srate, chans, packing, then chunks, I think */
offset = 40;
header_distributed = 1;
happy = 1;
while (happy)
{
if ((offset+chunkloc) >= data_location)
happy = 0;
else
{
offset += chunkloc;
if (seek_and_read(chan,(unsigned char *)hdrbuf,offset,32) <= 0)
{
mus_error(MUS_HEADER_READ_FAILED,"Bicsf header chunks confused at %d",offset);
return(-1);
}
chunkname = mus_uninterpreted_int((unsigned char *)hdrbuf);
chunksize = mus_big_endian_int((unsigned char *)(hdrbuf+4));
if (match_four_chars((unsigned char *)hdrbuf,I_COMM))
{
comment_start = 8+offset;
comment_end = comment_start + chunksize -1;
happy = 0;
}
else
{
if ((chunkname == 0) || (chunksize == 0))
happy = 0;
}
chunkloc = (8+chunksize);
}
}
return(0);
/* from here we fall back into read_next_header */
}
/* ------------------------------------ IRCAM ------------------------------------
* read/write CLM (old-style BICSF) -- added write option for Sun port 12-Dec-94
*
* 0: 0x1a364 or variations thereof -- byte order gives big/little_endian decision,
* ^ digit gives machine info, according to AFsp sources -- see IRCAM ints above
* 4: srate as a 32-bit float
* 8: chans
* 12: data format indicator (2=16-bit linear, 4=32-bit float)
* according to new Sox (version 11), these packing modes are now bytes/sample in low short, code in high
* so 1 = char, 0x10001 = alaw, 0x20001 = mulaw, 2 = short, 0x40004 = long, 4 = float
* 16: comment start -- how to tell if it's a real comment?
* apparently these are separated as short code, short blocksize, then data
* codes: 0=end, 1=maxamp, 2=comment, 3=pvdata, 4=audioencode and codemax??
* 1024: data start
*
* apparently the byte order depends on the machine.
* and yet... convert 1.4 makes a .sf file with little endian header, the VAX id, and big endian data?
*/
static int read_ircam_header (int chan)
{
short bcode,bloc,bsize;
int happy,offset,little;
type_specifier = mus_uninterpreted_int((unsigned char *)hdrbuf);
if ((mus_little_endian_int((unsigned char *)hdrbuf) == I_IRCAM_VAX) ||
(mus_little_endian_int((unsigned char *)hdrbuf) == I_IRCAM_MIPS))
little=1;
else little=0;
data_location = 1024;
true_file_length=lseek(chan,0L,SEEK_END);
data_size = (true_file_length-1024);
original_data_format = big_or_little_endian_int((unsigned char *)(hdrbuf+12),little);
data_format = SNDLIB_UNSUPPORTED;
if (original_data_format == 2)
{
if (little) data_format = SNDLIB_16_LINEAR_LITTLE_ENDIAN; else data_format = SNDLIB_16_LINEAR;
}
else if (original_data_format == 4)
{
if (little)
{
if (mus_little_endian_int((unsigned char *)hdrbuf) == I_IRCAM_VAX)
data_format = SNDLIB_32_VAX_FLOAT;
else data_format = SNDLIB_32_FLOAT_LITTLE_ENDIAN;
}
else data_format = SNDLIB_32_FLOAT;
}
else if (original_data_format == 0x40004)
{
if (little) data_format = SNDLIB_32_LINEAR_LITTLE_ENDIAN;
else data_format = SNDLIB_32_LINEAR;
}
else if (original_data_format == 0x10001) data_format = SNDLIB_8_ALAW;
else if (original_data_format == 0x20001) data_format = SNDLIB_8_MULAW;
else if (original_data_format == 1) data_format = SNDLIB_8_LINEAR;
srate = (int)big_or_little_endian_float((unsigned char *)(hdrbuf+4),little);
chans = big_or_little_endian_int((unsigned char *)(hdrbuf+8),little);
bloc=16;
happy=1;
offset=0;
while (happy)
{
offset += bloc;
if (seek_and_read(chan,(unsigned char *)hdrbuf,offset,32) <= 0)
{
mus_error(MUS_HEADER_READ_FAILED,"IRCAM header chunks confused at %d",offset);
return(-1);
}
bcode = big_or_little_endian_short((unsigned char *)hdrbuf,little);
bsize = big_or_little_endian_short((unsigned char *)(hdrbuf+2),little);
if (bcode == 2)
{
happy = 0;
comment_start = 4+offset;
comment_end = comment_start+bsize-1; /* was -5? */
}
bloc = bsize;
if ((bsize <= 0) || (bcode <= 0) || ((offset+bloc) > 1023)) happy = 0;
}
header_distributed = 0;
data_size = mus_bytes2samples(data_format,data_size);
return(0);
}
static int write_ircam_header (int chan, int srate, int chans, int format, char *comment, int len)
{
int i,j,err=0;
#ifdef NEXT
mus_set_big_endian_int((unsigned char *)hdrbuf,0x4a364);
#else
mus_set_big_endian_int((unsigned char *)hdrbuf,0x2a364); /* SUN id */
#endif
mus_set_big_endian_float((unsigned char *)(hdrbuf+4),(float)srate);
mus_set_big_endian_int((unsigned char *)(hdrbuf+8),chans);
switch (format)
{
case SNDLIB_8_MULAW: mus_set_big_endian_int((unsigned char *)(hdrbuf+12),0x20001); break;
case SNDLIB_8_ALAW: mus_set_big_endian_int((unsigned char *)(hdrbuf+12),0x10001); break;
case SNDLIB_16_LINEAR: mus_set_big_endian_int((unsigned char *)(hdrbuf+12),2); break;
case SNDLIB_32_LINEAR: mus_set_big_endian_int((unsigned char *)(hdrbuf+12),0x40004); break;
case SNDLIB_32_FLOAT: mus_set_big_endian_int((unsigned char *)(hdrbuf+12),4); break;
default:
mus_error(MUS_UNSUPPORTED_DATA_FORMAT,"IRCAM unsupported sound data format type: %d",format);
err = -1;
break;
}
if (len > 0)
{
mus_set_big_endian_short((unsigned char *)(hdrbuf+16),2);
mus_set_big_endian_short((unsigned char *)(hdrbuf+18),(short)len);
}
else
{
mus_set_big_endian_int((unsigned char *)(hdrbuf+16),0);
}
write(chan,hdrbuf,20);
data_location = 1024;
j = 0;
for (i=0;i<len;i++)
{
hdrbuf[j]=comment[i];
j++;
if (j == HDRBUFSIZ)
{
write(chan,hdrbuf,HDRBUFSIZ);
j = 0;
}
}
for (i=0;i<(1024-(len+20));i++) /* now fill left over bytes with nulls */
{
hdrbuf[j]=0;
j++;
if (j == HDRBUFSIZ)
{
write(chan,hdrbuf,HDRBUFSIZ);
j = 0;
}
}
if (j != 0) write(chan,hdrbuf,j);
return(err);
}
static void update_ircam_header (void)
{
/* size is implicit in file size - header */
}
/* ------------------------------------ 8SVX -------------------------------------
* (also known as IFF)
*
* very similar to AIFF:
* "BODY" => [4] samples [n] data
* "VHDR" => srate (short)
* "CHAN" => chans
* "ANNO" and "NAME"
*
* big_endian throughout
*/
static int read_8svx_header (int chan)
{
int chunksize,offset,chunkloc,happy;
type_specifier = mus_uninterpreted_int((unsigned char *)hdrbuf);
chunkloc = 12;
offset = 0;
data_format = SNDLIB_8_LINEAR;
header_distributed = 1;
srate = 0;
chans = 1;
happy = 1;
update_form_size = mus_big_endian_int((unsigned char *)(hdrbuf+4));
while (happy)
{
offset += chunkloc;
if (seek_and_read(chan,(unsigned char *)hdrbuf,offset,32) <= 0)
{
mus_error(MUS_HEADER_READ_FAILED,"IFF header chunks confused at %d",offset);
return(-1);
}
chunksize = mus_big_endian_int((unsigned char *)(hdrbuf+4));
if (match_four_chars((unsigned char *)hdrbuf,I_CHAN))
{
chans = mus_big_endian_int((unsigned char *)(hdrbuf+8));
chans = (chans & 0x01) + ((chans & 0x02) >> 1) + ((chans & 0x04) >> 2) + ((chans & 0x08) >> 3);
/* what in heaven's name is this? Each bit corresponds to a channel? */
}
else
{
if (match_four_chars((unsigned char *)hdrbuf,I_VHDR))
{
/* num_samples (int) at hdrbuf+8 */
srate = mus_big_endian_unsigned_short((unsigned char *)(hdrbuf+20));
original_data_format = hdrbuf[23];
if (original_data_format != 0) data_format = SNDLIB_UNSUPPORTED;
}
else
{
if ((match_four_chars((unsigned char *)hdrbuf,I_ANNO)) || (match_four_chars((unsigned char *)hdrbuf,I_NAME)))
{
comment_start = offset+8;
comment_end = comment_start+chunksize-1;
}
else
{
if (match_four_chars((unsigned char *)hdrbuf,I_BODY))
{
data_size = chunksize;
data_location = offset+12;
happy = 0;
}
}
}
}
chunkloc = (8+chunksize);
if (chunksize&1) chunkloc++; /* extra null appended to odd-length chunks */
}
data_size = mus_bytes2samples(data_format,data_size);
return(0);
}
/* ------------------------------------ VOC --------------------------------------
*
* 0: "Creative Voice File" followed by a couple ctrl-Z ('32) (swapped data)
* 20: header end (short) {8svx, 26=data_offset, 0x10a=version, ((~version + 0x1234) & 0xffff) = 0x1129}
* [20]: first block:
* block code, 1=data, 0=end, 9=data_16 (2=continue, 3=silence, 4=marker, 5=text, 6=loop, 7=loop-end, 8=extended)
* block len as 24 bit int(?)
* if data, then rate code (byte), then data (assuming 8-bit unsigned, mono)
* if data_16, long srate, byte: data size (8 or 16), byte chans
* if text, ascii text (a comment)
* if extended (8) precedes 1 (data): 8 4 then time constant (short), byte: packing code (0), byte chans (0=mono)
*
* apparently always little_endian
* updated extensively 29-Aug-95 from sox10 voc.c
*/
static int read_voc_header(int chan)
{
int type,happy,len,curbase,voc_extended,bits,code;
data_format = SNDLIB_8_UNSIGNED;
chans = 1;
happy = 1;
voc_extended = 0;
true_file_length=lseek(chan,0L,SEEK_END);
curbase = mus_little_endian_short((unsigned char *)(hdrbuf+20));
header_distributed = 1;
lseek(chan,curbase,SEEK_SET);
read(chan,hdrbuf,HDRBUFSIZ);
type = (int)(hdrbuf[0]);
len=(((int)hdrbuf[3])<<16)+(((int)hdrbuf[2])<<8)+(((int)hdrbuf[1]));
while (happy)
{
if (type == 1) /* voc_data */
{
data_size = len-1; /* was -3 */
data_location = curbase+6;
if (voc_extended == 0)
{
srate = (int)(1000000.0/(256 - ((int)(hdrbuf[4]&0xff))));
original_data_format = hdrbuf[5];
if (hdrbuf[5] == 0) data_format = SNDLIB_8_UNSIGNED; else data_format = SNDLIB_UNSUPPORTED;
}
happy = 0;
}
else
{
if (type == 9) /* voc_data_16 */
{
data_size = len-1; /* was -3 */
data_location = curbase+6;
srate = mus_little_endian_int((unsigned char *)(hdrbuf+4));
bits = ((int)hdrbuf[8]);
if (bits == 8)
{
code = mus_little_endian_short((unsigned char *)(hdrbuf+10));
if (code == 6)
data_format = SNDLIB_8_ALAW;
else
if (code == 7)
data_format = SNDLIB_8_MULAW;
else data_format = SNDLIB_8_UNSIGNED;
}
else
if (bits == 16)
data_format = SNDLIB_16_LINEAR_LITTLE_ENDIAN;
else data_format = SNDLIB_UNSUPPORTED;
chans = (int)hdrbuf[9];
if (chans == 0) chans = 1;
happy = 0;
}
else
{
if (((len+curbase)<true_file_length) && (type != 0))
{
if (type == 5) /* voc_text */
{
comment_start = curbase+4;
comment_end = comment_start + len - 1;
}
else
{
if (type == 8) /* voc_extended */
{
srate = (256000000 / (65536 - mus_little_endian_short((unsigned char *)(hdrbuf+4))));
if ((int)(hdrbuf[7]) == 0) chans=1; else chans=2;
if ((int)(hdrbuf[6]) != 0) data_format = SNDLIB_UNSUPPORTED;
}
}
if (seek_and_read(chan,(unsigned char *)hdrbuf,curbase+len+4,HDRBUFSIZ) <= 0)
{
mus_error(MUS_HEADER_READ_FAILED,"VOC header looks wrong");
return(-1);
}
curbase += len;
}
else happy = 0;
}
}
}
data_size = mus_bytes2samples(data_format,data_size);
return(0);
}
/* ------------------------------------ ADC ------------------------------------
* also known as OGI format
* TIMIT format is identical except it omits the data format field (header size claims to be bytes)
*
* from ad.h and other files, ogitools-v1.0.tar.gz
* we'll look for the big/little endian sequence (short) 8 1 1-or-2 given big/little decision
*
* 0: header size in shorts (8 = 16 bytes) (OGI says this is in bytes)
* 2: version (1)
* 4: chans
* 6: rate (srate = 4000000/rate)
* 8: samples (int) -- seems to be off by 2 -- are they counting ints here?
* 12: data format (0=big-endian)
* 16: data start
*/
static int read_adc_header(int chan)
{
int little;
little = mus_uninterpreted_int((unsigned char *)(hdrbuf+12)); /* 0=big endian */
data_location = 16;
if (little) data_format = SNDLIB_16_LINEAR_LITTLE_ENDIAN; else data_format = SNDLIB_16_LINEAR;
chans = big_or_little_endian_short((unsigned char *)(hdrbuf+4),little);
srate = 4000000 / big_or_little_endian_short((unsigned char *)(hdrbuf+6),little);
data_size = 2*big_or_little_endian_int((unsigned char *)(hdrbuf+8),little);
comment_start=0;
comment_end=0;
header_distributed = 0;
true_file_length=lseek(chan,0L,SEEK_END);
data_size = mus_bytes2samples(data_format,data_size);
return(0);
}
/* ------------------------------------ AVR --------------------------------------
*
* 0: "2BIT"
* 4: sample name (null padded ASCII)
* 12: chans (short) (0=mono, -1=stereo)
* 14: sample size (8 or 16 bit) (short) (value is 8, 12, or 16)
* 16: sample format (signed or unsigned) (short) (0=unsigned, -1=signed)
* 18: loop (on/off), 20: midi (-1 = no MIDI)
* 22: srate
* avr.txt has:
* 22: Replay speed 0=5.485 Khz, 1=8.084 Khz, 2=10.971 Khz, 3=16.168 Khz, 4=21.942 Khz, 5=32.336 Khz, 6=43.885 Khz, 7=47.261 Khz
* 23: sample rate in Hertz (as a 3 byte quantity??)
* 26: length in samples
* 30: loop beg, 34: loop end, 38: midi (keyboard split), 40: compression, 42: nada ("reserved"), 44: name
* 64: comment (limited to 64 bytes)
* 128: data start
*
* the Atari .avr files appear to be 8000 Hz, mono, 8-bit linear unsigned data with an unknown header of 128 words
* apparently there was a change in format sometime in the 90's.
*
* The actual avr files I've found on the net are either garbled, or
* something is wrong with this definition (taken from CMJ and www.wotsit.org's avr.txt).
* SGI dmconvert assumes big-endian here -- this is an Atari format, so it's probably safe to assume big-endian.
*/
static int read_avr_header(int chan)
{
int dsize,dsigned,i;
chans = mus_big_endian_short((unsigned char *)(hdrbuf+12));
if (chans == 0) chans=1; else chans=2;
data_location = 128;
data_size = mus_big_endian_int((unsigned char *)(hdrbuf+26));
header_distributed = 0;
srate = mus_big_endian_unsigned_short((unsigned char *)(hdrbuf+24));
dsize = mus_big_endian_short((unsigned char *)(hdrbuf+14));
dsigned = mus_big_endian_short((unsigned char *)(hdrbuf+16));
if (dsize == 16)
{
if (dsigned == 0)
data_format = SNDLIB_16_UNSIGNED;
else data_format = SNDLIB_16_LINEAR;
}
else
{
if (dsigned == 0)
data_format = SNDLIB_8_UNSIGNED;
else data_format = SNDLIB_8_LINEAR;
}
if (seek_and_read(chan,(unsigned char *)hdrbuf,64,64) <= 0)
{
mus_error(MUS_HEADER_READ_FAILED,"AVR header looks wrong");
return(-1);
}
comment_start = 64;
i = 0;
while ((i < 64) && (hdrbuf[i] != 0)) i++;
comment_end = 64+(i-1);
true_file_length=lseek(chan,0L,SEEK_END);
data_size = mus_bytes2samples(data_format,data_size);
return(0);
}
/* ------------------------------------ SNDT -------------------------------------
*
* this taken from sndrtool.c (sox-10): (modified 6-Feb-98)
* 0: "SOUND" (or off by two throughout if not "0x1a"?)
* 5: 0x1a
* 6-7: 0
* 8-11: nsamps (at 12)
* 12-15: 0
* 16-19: nsamps
* 20-21: srate (little endian short) (at 22)
* 22-23: 0
* 24-25: 10
* 26-27: 4
* 28-> : <filename> "- File created by Sound Exchange"
* .->95: 0 ?
*/
/* similar is Sounder format:
* 0: 0
* 2: short srate (little endian)
* 4: 10
* 6: 4
* then data
* but this format can't be distinguished from a raw sound file
*/
static int read_sndt_header(int chan)
{
data_format = SNDLIB_8_UNSIGNED;
chans = 1;
srate = mus_little_endian_unsigned_short((unsigned char *)(hdrbuf+20));
data_location = 126;
data_size = mus_little_endian_int((unsigned char *)(hdrbuf+8));
if (data_size < 0) data_size = mus_little_endian_int((unsigned char *)(hdrbuf+10));
if (srate <= 1) srate = mus_little_endian_unsigned_short((unsigned char *)(hdrbuf+22));
header_distributed = 0;
true_file_length=lseek(chan,0L,SEEK_END);
return(0);
}
/* ------------------------------------ Covox v8 -------------------------------------
*
* 0: 377 125 377 252 377 125 377 252 x x 0's to 16
* then 8-bit unsigned data
*/
static int read_covox_header(int chan)
{
data_format = SNDLIB_8_UNSIGNED;
chans = 1;
header_distributed = 0;
data_location = 16;
srate = 8000;
true_file_length=lseek(chan,0L,SEEK_END);
data_size = true_file_length - data_location;
return(0);
}
/* ------------------------------------ Digitracker SPL -------------------------------------
* (Atari-related, I think, and now obsolete)
* 0: DSPL
* 4: version
* 5: sample name
* 37: filename
* 45: c-4 freq (srate?)
* 47: length
* 51: loop data
* 59: vol
* 60: bit 0: 0=8 bit, 1=16
* 61: data (in sample data chunks or as straight data??)
*/
static int read_spl_header(int chan)
{
chans = 1;
header_distributed = 0;
data_location = 61;
srate = 8000; /* I need an example to decode this */
true_file_length=lseek(chan,0L,SEEK_END);
if (hdrbuf[60]&1) data_format = SNDLIB_16_LINEAR; else data_format = SNDLIB_8_LINEAR; /* unsigned? */
data_size = mus_bytes2samples(data_format,true_file_length - data_location);
return(0);
}
/* ------------------------------------ SMP -------------------------------------
*
* 0: "SOUND SAMPLE DATA "
* 18: "2.1 "
* 22-81: comment
* 82-111: sample name
* header 112 bytes
* long samples (bytes=samples*2)
* then data start
* data
* always little endian
*/
static int read_smp_header(int chan)
{
data_format = SNDLIB_16_LINEAR_LITTLE_ENDIAN;
chans = 1;
comment_start=22;
comment_end=81;
header_distributed = 1;
data_location = 116;
lseek(chan,112,SEEK_SET);
read(chan,hdrbuf,4);
data_size = 2*(mus_little_endian_int((unsigned char *)hdrbuf));
srate = 8000; /* docs mention an srate floating around at the end of the file, but I can't find it in any example */
true_file_length=lseek(chan,0L,SEEK_END);
data_size = mus_bytes2samples(data_format,data_size);
return(0);
}
/* ------------------------------------ SPPACK -------------------------------------
*
* from AF docs:
* Bytes Type Contents
* 0 160 char Text strings (2 * 80)
* 160 80 char Command line
* 240 2 int Domain
* 242 2 int Frame size
* 244 4 float Sampling frequency
* 252 2 int File identifier (i.e. #o100 #o303)
* 254 2 int Data type (0xfc0e = sampled data file)
* 256 2 int Resolution (in bits)
* 258 2 int Companding flag (1=linear, 2=alaw, 3=mulaw)
* 272 240 char Text strings (3 * 80)
* 512 ... -- Audio data
*
*/
static int read_sppack_header(int chan)
{
int typ,bits;
float sr;
data_location = 512;
chans = 1;
lseek(chan,240,SEEK_SET);
read(chan,hdrbuf,22);
typ = mus_big_endian_short((unsigned char *)hdrbuf);
data_format = SNDLIB_UNSUPPORTED;
header_distributed = 0;
if (typ == 1)
{
if (((hdrbuf[254])==252) && ((hdrbuf[255])==14)) /* #xfc and #x0e */
{
typ = mus_big_endian_short((unsigned char *)(hdrbuf+18));
bits = mus_big_endian_short((unsigned char *)(hdrbuf+16));
sr = mus_big_endian_float((unsigned char *)(hdrbuf+4));
srate = (int)sr;
switch (typ)
{
case 1: if (bits == 16) data_format = SNDLIB_16_LINEAR; else data_format = SNDLIB_8_LINEAR; break;
case 2: data_format = SNDLIB_8_ALAW; break;
case 3: data_format = SNDLIB_8_MULAW; break;
default: data_format = SNDLIB_UNSUPPORTED; break;
}
data_size=lseek(chan,0L,SEEK_END);
data_size=mus_bytes2samples(data_format,data_size-512);
comment_start=0;
comment_end=0;
}
}
true_file_length=lseek(chan,0L,SEEK_END);
return(0);
}
/* ------------------------------------ ESPS (Entropic Signal Processing System) -------------------------------------
*
* specs at ftp.entropic.com
* from AFgetInfoES.c:
*
* Bytes Type Contents
* 8 -> 11 -- Header size (bytes)
* 12 -> 15 int Sampled data record size
* 16 -> 19 int File identifier: 0x00006a1a or 0x1a6a0000
* 40 -> 65 char File creation date
* 124 -> 127 int Number of samples
* 132 -> 135 int Number of doubles in a data record
* 136 -> 139 int Number of floats in a data record
* 140 -> 143 int Number of longs in a data record
* 144 -> 147 int Number of shorts in a data record
* 148 -> 151 int Number of chars in a data record
* 160 -> 167 char User name
* 333 -> H-1 -- "Generic" header items, including "record_freq" {followed by a "double8"=64-bit ?}
* H -> ... -- Audio data
*/
static int read_esps_header (int chan)
{
char str[80];
int happy = 1;
int k,hend,curbase,j,n,chars,floats,shorts,doubles,little;
data_location = mus_big_endian_int((unsigned char *)(hdrbuf+8));
little = (hdrbuf[18] == 0);
true_file_length = lseek(chan,0L,SEEK_END);
data_size = true_file_length - data_location;
header_distributed = 0;
srate = 8000;
chans = 1;
lseek(chan,132,SEEK_SET);
read(chan,hdrbuf,HDRBUFSIZ);
if (little)
{
doubles = mus_little_endian_int((unsigned char *)hdrbuf);
floats = mus_little_endian_int((unsigned char *)(hdrbuf+4));
shorts = mus_little_endian_int((unsigned char *)(hdrbuf+12));
chars = mus_little_endian_int((unsigned char *)(hdrbuf+16));
}
else
{
doubles = mus_big_endian_int((unsigned char *)hdrbuf);
floats = mus_big_endian_int((unsigned char *)(hdrbuf+4));
shorts = mus_big_endian_int((unsigned char *)(hdrbuf+12));
chars = mus_big_endian_int((unsigned char *)(hdrbuf+16));
}
if (shorts != 0)
{
data_format = ((little) ? SNDLIB_16_LINEAR_LITTLE_ENDIAN : SNDLIB_16_LINEAR);
chans = shorts;
}
else
{
if (doubles != 0)
{
data_format = ((little) ? SNDLIB_64_DOUBLE_LITTLE_ENDIAN : SNDLIB_64_DOUBLE);
chans = doubles;
}
else
{
if (floats != 0)
{
data_format = ((little) ? SNDLIB_32_FLOAT_LITTLE_ENDIAN : SNDLIB_32_FLOAT);
chans = floats;
}
else
{
if (chars != 0)
{
data_format = SNDLIB_8_LINEAR; /* ?? */
chans = chars;
}
}
}
}
/* search for "record_freq" to get srate */
lseek(chan,333,SEEK_SET);
read(chan,hdrbuf,HDRBUFSIZ);
curbase=333;
hend=curbase+HDRBUFSIZ;
k=0;
n=0;
for (j=0;j<80;j++) str[j]=' ';
while (happy)
{
str[k] = hdrbuf[n];
if ((str[k] == 'q') || (str[k] == 3) || ((curbase+n+1) >= data_location) || (k == 78))
{ /* 3 = C-C marks end of record */
str[k+1]=0;
if (strcmp(str,"record_freq") == 0)
{
if (seek_and_read(chan,(unsigned char *)hdrbuf,curbase+n,32) <= 0)
{
mus_error(MUS_HEADER_READ_FAILED,"ESPS header looks wrong");
return(-1);
}
n=0;
if (little)
srate = (int)mus_little_endian_double((unsigned char *)(hdrbuf+8));
else srate = (int)mus_big_endian_double((unsigned char *)(hdrbuf+8));
happy = 0;
}
if ((curbase+n+1) >= data_location) happy = 0;
k=0;
}
else
k++;
n++;
if (n >= hend)
{
curbase += hend;
n = 0;
read(chan,hdrbuf,HDRBUFSIZ);
hend = HDRBUFSIZ;
}
}
if (srate == 0) srate = 8000;
data_size = mus_bytes2samples(data_format,data_size);
return(0);
}
/* ------------------------------------ INRS -------------------------------------
*
* from AFgetInfoIN.c:
*
* INRS-Telecommunications audio file:
* Bytes Type Contents
* 0 -> 3 float Sampling Frequency (VAX float format)
* 6 -> 25 char Creation time (e.g. Jun 12 16:52:50 1990)
* 26 -> 29 int Number of speech samples in the file (? -- old INRS files omit this)
* The data in an INRS-Telecommunications audio file is in 16-bit integer (little-endian)
* format. Header is always 512 bytes. Always mono.
*
*/
static int inrs_srates[NINRS] = {6500, 6667, 8000, 10000, 12000, 16000, 20000};
static int read_inrs_header (int chan, int loc)
{
true_file_length = lseek(chan,0L,SEEK_END);
comment_start = 6;
comment_end = 25;
header_distributed = 0;
data_format = SNDLIB_16_LINEAR_LITTLE_ENDIAN;
srate = loc;
chans = 1;
data_location = 512;
data_size = mus_bytes2samples(data_format,true_file_length-512);
return(0);
}
/* ------------------------------------ MAUD -------------------------------------
*
* very similar to AIFF:
* "MHDR" => 4: chunksize (32)
* 8: samples
* 12: bits
* 14: ditto
* 16: clock freq
* 20: clock div (srate = freq/div)
* 22: chan info (0=mono, 1=stereo)
* 24: ditto(?!)
* 26: format (0=unsigned 8 or signed 16 (see bits), 2=alaw, 3=mulaw)
* 28-40: unused
* "MDAT" => data
* "ANNO" => comment
*/
static int read_maud_header (int chan)
{
int chunksize,offset,chunkloc,happy,num,den;
type_specifier = mus_uninterpreted_int((unsigned char *)hdrbuf);
chunkloc = 12;
offset = 0;
data_format = SNDLIB_8_LINEAR;
header_distributed = 1;
srate = 0;
chans = 1;
happy = 1;
update_form_size = mus_big_endian_int((unsigned char *)(hdrbuf+4));
while (happy)
{
offset += chunkloc;
if (seek_and_read(chan,(unsigned char *)hdrbuf,offset,32) <= 0)
{
mus_error(MUS_HEADER_READ_FAILED,"MAUD header chunk bogus at %d",offset);
return(-1);
}
chunksize = mus_big_endian_int((unsigned char *)(hdrbuf+4));
if (match_four_chars((unsigned char *)hdrbuf,I_MHDR))
{
data_size = mus_big_endian_int((unsigned char *)(hdrbuf+8));
num = mus_big_endian_int((unsigned char *)(hdrbuf+16));
den = mus_big_endian_short((unsigned char *)(hdrbuf+20));
srate = (int)(num/den);
num = mus_big_endian_short((unsigned char *)(hdrbuf+12));
den = mus_big_endian_short((unsigned char *)(hdrbuf+26));
if (num == 8)
{
switch (den)
{
case 0: data_format = SNDLIB_8_UNSIGNED; break;
case 2: data_format = SNDLIB_8_ALAW; break;
case 3: data_format = SNDLIB_8_MULAW; break;
default: data_format = SNDLIB_UNSUPPORTED; break;
}
}
else data_format = SNDLIB_16_LINEAR;
num = mus_big_endian_short((unsigned char *)(hdrbuf+22));
if (num == 0) chans = 1; else chans = 2;
}
else
{
if (match_four_chars((unsigned char *)hdrbuf,I_ANNO))
{
comment_start = offset+8;
comment_end = comment_start+chunksize-1;
}
else
{
if (match_four_chars((unsigned char *)hdrbuf,I_MDAT))
{
data_location = offset+12;
happy = 0;
}
}
}
chunkloc = (8+chunksize);
if (chunksize&1) chunkloc++; /* extra null appended to odd-length chunks */
}
data_size = mus_bytes2samples(data_format,data_size);
return(0);
}
/* ------------------------------------ Sound Designer I -------------------------------------
*
* complicated and defined in terms of Pascal records, so the following is a stab in the dark:
*
* 0: 1336 (i.e. header size)
* 764: comment (str255)
* 1020: sample rate (long)
* 1028: data size (short)
* 1030: a code string describing the data type (i.e. "linear") (str32)
* 1064: user comment (str255)
*
* file type: 'SFIL'
*
* always big_endian
*/
static int read_sd1_header (int chan)
{
int n;
chans = 1;
header_distributed = 0;
data_location = 1336;
lseek(chan,1020,SEEK_SET);
read(chan,hdrbuf,64);
srate = mus_big_endian_int((unsigned char *)hdrbuf);
n = mus_big_endian_short((unsigned char *)(hdrbuf+8));
if (n == 16)
data_format = SNDLIB_16_LINEAR;
else data_format = SNDLIB_8_LINEAR;
true_file_length = lseek(chan,0L,SEEK_END);
data_size = mus_bytes2samples(data_format,true_file_length-1336);
n = ((unsigned char)hdrbuf[44]);
if (n != 0)
{
comment_start = 1064;
comment_end = comment_start+n-1;
}
return(0);
}
/* ------------------------------------ PSION alaw -------------------------------------
*
* 0: "ALawSoundFile**"
* 16: version
* 18: length (bytes)
* 22: padding
* 24: repeats
* 26-32: nada
* 32: data
*
* always mono 8-bit alaw 8000 Hz. All the examples on the psion net site appear to be little endian.
*/
static int read_psion_header (int chan)
{
chans = 1;
header_distributed = 0;
data_location = 32;
srate = 8000;
data_format = SNDLIB_8_ALAW;
data_size = mus_little_endian_int((unsigned char *)(hdrbuf+18)); /* always little-endian? */
true_file_length=lseek(chan,0,SEEK_END);
if ((true_file_length+32) != data_size) data_size = (true_file_length-32);
data_size = mus_bytes2samples(data_format,data_size);
return(0);
}
/* ------------------------------------ Tandy DeskMate -------------------------------------
*
* 0: short 1A (c-Z = .snd ID byte)
* 1: data type (0=no compression, 1=music compression, 2=speech compression)
* 2: number of notes (= 1 if sound)
* 3: instrument number (= 0 if sound, otherwise this is an "instrument" file)
* 4: sound/instrument name (ASCIZ -- 10 bytes)
* 14: sampling rate (apparently one of 5500, 11000, 22000 -- is this unsigned?)
* 16: begin variable 28-byte records, after which is sound data as unsigned 8-bit linear ("PCM")
* in sound files:
* 16: 0xFF 0 0xFF 0xFF
* 20: data location if compressed
* 24: 8 bytes of non-sound related things
* 32: sound size
* 36: 8 more useless bytes
*
* always 8-bit unsigned linear mono
* This info from oak.oakland.edu:/SimTel/msdos/sound/tspak17.zip (now at ftp.coast.net:/pub/SimTel/msdos/sound)
*/
static int read_deskmate_header (int chan)
{
/* if we reach here, we're at 0 with HDRBUFSIZ read in */
chans = 1;
header_distributed = 0;
data_location = 44;
srate = mus_little_endian_short((unsigned char *)(hdrbuf+14)); /* unsigned? */
data_format = SNDLIB_8_UNSIGNED;
data_size = mus_little_endian_int((unsigned char *)(hdrbuf+32));
true_file_length=lseek(chan,0,SEEK_END);
data_size = mus_bytes2samples(data_format,data_size);
return(0);
}
/* ------------------------------------ Tandy DeskMate 2500 -------------------------------------
* the "new .snd file format"
*
* 0: name of sound, 10 byte ASCIZ
* 10: nada (probably to be backwards compatible...)
* 44: snd ID -- 0x1A80
* 46: number of samples
* 48: sound number
* 50: nada
* 66: compression code (see above)
* 68: nada
* 88: sampling rate
* 90: nada
* 114: sample descriptors:
* 0: link to next (0 if last)
* 4: nada
* 10: data location
* 12: number of bytes in sample
* 46: end of descriptor (presumably start of data)
*/
static int read_deskmate_2500_header (int chan)
{
/* if we reach here, we're at 0 with HDRBUFSIZ read in */
chans = 1;
header_distributed = 0;
data_location = 114+46;
srate = mus_little_endian_short((unsigned char *)(hdrbuf+88));
data_format = SNDLIB_8_UNSIGNED;
data_size = mus_little_endian_int((unsigned char *)(hdrbuf+46));
true_file_length=lseek(chan,0,SEEK_END);
data_size = mus_bytes2samples(data_format,data_size);
return(0);
}
/* ------------------------------------ Gravis Ultrasound Patch -------------------------------------
*
* http://www.gravis.com/Public/sdk/PATCHKIT.ZIP
*
* header [128], instruments [62], layers [49], waveheaders (nested)
* always little endian, actual files don't match exactly with any documentation
*
* Header block:
* 0: "GF1PATCH100" or "GF1PATCH110"
* 12: "ID#000002"
* 22: comment (copyright notice) (60 bytes ASCIZ)
* 82: number of instruments
* 83: number of voices
* 84: wave channels
* 85: number of waves
* 87: vol
* 89: size?
* 93: reserved (36? bytes)
*
* Instrument block:
* 0: id
* 2: name (16 bytes)
* 18: size
* 22: number of layers
* 23: reserved (40? bytes)
*
* Layer block:
* 0: "previous"
* 1: id
* 2: size
* 6: number of wave samples
* 10: reserved (40? bytes)
*
* Wave block:
* 0: name (7 bytes ASCIZ)
* 7: "fractions"
* 8: data size of wave sample
* 12: loop start
* 16: loop end
* 20: sample rate
* 22: low freq
* 26: high freq
* 30: root freq
* 34: tune
* 36: balance
* 37: envelope data (6+6 bytes I think)
* 49: tremolo and vibrato data (6 bytes)
* 55: mode bit 0: 8/16, 1: signed/unsigned
* 56: scale freq
* 58: scale factor
* 60: reserved (36 bytes)
* followed by data presumably
*/
static int read_gravis_header(int chan)
{
int mode;
lseek(chan,0,SEEK_SET);
read(chan,hdrbuf,128);
chans = hdrbuf[84];
if (chans == 0) chans=1;
comment_start = 22;
comment_end = 81;
lseek(chan,239,SEEK_SET); /* try to jump to wave sample block (128+62+49) */
read(chan,hdrbuf,128);
srate = mus_little_endian_unsigned_short((unsigned char *)(hdrbuf+20));
data_size = mus_little_endian_unsigned_short((unsigned char *)(hdrbuf+8));
mode = hdrbuf[55];
if (mode&1)
{
if (mode&2)
data_format = SNDLIB_16_UNSIGNED_LITTLE_ENDIAN;
else data_format = SNDLIB_16_LINEAR_LITTLE_ENDIAN;
}
else
{
if (mode&2)
data_format = SNDLIB_8_UNSIGNED;
else data_format = SNDLIB_8_LINEAR;
}
data_location = 337;
header_distributed = 0;
true_file_length = lseek(chan,0L,SEEK_END);
data_size = mus_bytes2samples(data_format,data_size);
return(0);
}
/* ------------------------------------ Goldwave -------------------------------------
*
* http://web.cs.mun.ca/~chris3/goldwave/goldwave.html
*/
static int read_goldwave_header(int chan)
{
chans = 1;
header_distributed = 0;
data_location = 28;
data_size = mus_little_endian_int((unsigned char *)(hdrbuf+22));
true_file_length = lseek(chan,0L,SEEK_END);
if (data_size <= 24) data_size = (true_file_length-data_location);
srate = mus_little_endian_int((unsigned char *)(hdrbuf+18));
data_format = SNDLIB_16_LINEAR_LITTLE_ENDIAN;
return(0);
}
/* ------------------------------------ Sonic Resource Foundry -------------------------------------
*
* more reverse engineering...
* http://www.sfoundry.com/
*/
static int read_srfs_header(int chan)
{
chans = 1; /* might be short at header[4] */
header_distributed = 0;
data_location = 32;
true_file_length = lseek(chan,0L,SEEK_END);
data_size = (true_file_length-data_location);
srate = mus_little_endian_int((unsigned char *)(hdrbuf+6));
data_format = SNDLIB_16_LINEAR_LITTLE_ENDIAN;
return(0);
}
/* ------------------------------------ Quicktime -------------------------------------
*
* infinitely complicated -- see Quicktime File Format doc from Apple.
* there's no relation between this document and actual files -- a bizarre joke?
*/
static int read_qt_header(int chan)
{
chans = 1;
header_distributed = 0;
data_location = 12;
true_file_length = lseek(chan,0L,SEEK_END);
data_size = (true_file_length-data_location);
srate = 11025; /* ?? */
data_format = SNDLIB_8_UNSIGNED;
return(0);
}
/* ------------------------------------ SBStudioII -------------------------------------
*
* from a file created by Convert 1.4
* 0: SND <space>
* 8: file size - 8
* SNNA SNIN SNDT blocks:
*
* built in blocks, other names are SNIN, SNDT
* need to scan for SNDT, block length, data
* SNNA len name
* supposedly ends with END (but my examples don't)
* SNIN:
* num (2), reserved (2), tuning (1), vol (2), type (2) bit 0: 1=PCM, bit 1: 1=16,0=8 (then loop data)
* info from Pac.txt (pac.zip) at http://www.wotsit.org/music.htm
*/
static int read_sbstudio_header(int chan)
{
int i,happy,tmp;
unsigned char *bp;
lseek(chan,0,SEEK_SET);
read(chan,hdrbuf,HDRBUFSIZ);
chans = 1;
header_distributed = 0;
srate = 8000; /* no sampling rate field in this header */
data_format = SNDLIB_16_LINEAR_LITTLE_ENDIAN; /* might change later */
true_file_length = lseek(chan,0L,SEEK_END);
data_size = true_file_length - 56; /* first guess */
happy = 1;
i=8;
bp = (unsigned char *)(hdrbuf+8);
while (happy)
{
if (match_four_chars(bp,I_SNDT))
{
data_size = mus_little_endian_int((unsigned char *)(bp+4));
data_location = i+8;
happy = 0;
}
else
{
if (match_four_chars(bp,I_SNIN))
{
tmp = mus_little_endian_short((unsigned char *)(bp+15));
if ((tmp&1)==0)
data_format = SNDLIB_UNSUPPORTED;
else
{
if ((tmp&2)==0) data_format = SNDLIB_8_LINEAR;
}
i+=26;
bp+=26;
}
else
{
i++;
bp++;
}
}
if (i >= HDRBUFSIZ)
{
data_format = SNDLIB_UNSUPPORTED;
happy = 0;
}
}
data_size = mus_bytes2samples(data_format,data_size);
return(0);
}
/* ------------------------------------ Delusion Sound -------------------------------------
*
* more reverse engineering...
* from a file created by Convert 1.4
* 0: DDSF
* 5: name (text)
* 55: data
* probaby similar to DMF format describe in Dmf-form.txt but I don't see any other block names in the data
*/
static int read_delusion_header(int chan)
{
chans = 1;
header_distributed = 0;
data_location = 55;
true_file_length = lseek(chan,0L,SEEK_END);
data_size = (true_file_length-data_location);
srate = 8000;
data_format = SNDLIB_16_LINEAR_LITTLE_ENDIAN;
data_size = mus_bytes2samples(data_format,data_size);
return(0);
}
/* ------------------------------------ Farandole Composer WaveSample -------------------------------------
*
* 0: FSM 254
* 4: name (text) (32 bytes)
* 36: 10,13,26 or something like that
* 39: len?
* 40: volume
* 41: looping data
* 49: type (0=8-bit, else 16)
* 50: loop mode
* 51: data
* described in Fsm.txt and Far-form.txt http://www.wotsit.org/music.htm
*/
static int read_farandole_header(int chan)
{
chans = 1;
header_distributed = 0;
data_location = 51;
true_file_length = lseek(chan,0L,SEEK_END);
data_size = (true_file_length-data_location);
srate = 8000;
if (hdrbuf[49] == 0)
data_format = SNDLIB_8_LINEAR;
else data_format = SNDLIB_16_LINEAR_LITTLE_ENDIAN;
data_size = mus_bytes2samples(data_format,data_size);
return(0);
}
/* ------------------------------------ Yamaha TX-16 -------------------------------------
*
* ftp://ftp.t0.or.at/pub/sound/tx16w/samples.yamaha
* http://www.t0.or.at/~mpakesch/tx16w/
*
* from tx16w.c sox 12.15: (7-Oct-98) (Mark Lakata and Leigh Smith)
* char filetype[6] "LM8953"
* nulls[10],
* dummy_aeg[6]
* format 0x49 = looped, 0xC9 = non-looped
* sample_rate 1 = 33 kHz, 2 = 50 kHz, 3 = 16 kHz
* atc_length[3] if sample rate 0, [2]&0xfe = 6: 33kHz, 0x10:50, 0xf6: 16, depending on [5] but to heck with it
* rpt_length[3] (these are for looped samples, attack and loop lengths)
* unused[2]
*/
static int read_tx16_header(int chan)
{
chans = 1;
header_distributed = 0;
data_location = 32;
true_file_length = lseek(chan,0L,SEEK_END);
data_size = (true_file_length-data_location);
srate = 16000;
if (hdrbuf[23] == 1) srate = 33000;
else if (hdrbuf[23] == 2) srate = 50000;
else if (hdrbuf[23] == 3) srate = 16000;
else if (hdrbuf[23] == 0)
{
if ((hdrbuf[26]&0xFE) == 6) srate = 33000;
else if ((hdrbuf[26]&0xFE) == 0x10) srate = 50000;
else if ((hdrbuf[26]&0xFE) == 0xf6) srate = 16000;
}
original_data_format = SNDLIB_12_LINEAR_LITTLE_ENDIAN; /* can't read this format yet */
data_format = SNDLIB_UNSUPPORTED;
data_size = (int)((float)data_size / 1.5);
return(0);
}
/* ------------------------------------ Yamaha SY-85 and SY-99 -------------------------------------
*
* more reverse engineering...
* 0: SY85 (SY80 is SY-99) SY85ALL SY80 SYALL
* 5: name ("WAVE1")
* (26 int len)
* (33: comment or prompt?)
* data in 16-bit little endian (?)
*/
static int read_sy85_header(int chan)
{
chans = 1;
header_distributed = 0;
data_location = 1024;
true_file_length = lseek(chan,0L,SEEK_END);
data_size = (true_file_length-data_location);
srate = 8000; /* unknown */
data_format = SNDLIB_16_LINEAR; /* not right */
data_size = mus_bytes2samples(data_format,data_size);
return(0);
}
/* ------------------------------------ Kurzweil 2000 -------------------------------------
*
* "PRAM" then header len as big endian int??
* from krz2tx.c (Mark Lakata)
*/
static int read_kurzweil_2000_header(int chan)
{
chans = 1;
header_distributed = 0;
data_location = mus_big_endian_int((unsigned char *)(hdrbuf+4));
true_file_length = lseek(chan,0L,SEEK_END);
data_size = (true_file_length-data_location);
srate = 44100; /* unknown */
data_format = SNDLIB_16_LINEAR;
data_size = mus_bytes2samples(data_format,data_size);
return(0);
}
/* ------------------------------------ Ultratracker WaveSample -------------------------------------
*
* 0..31: name (32=ctrl-Z?)
* 33: PMUWFD (but docs say this is "dos name" -- perhaps we can't recognize this header type reliably)
* 44: 4 ints giving loop and size data
* 60: vol
* 61: "bidi" 0|8|24->8 bit else 16 -- but actual example has 0 with 16-bit
* 62: finetune
* 64: data (or 68?)
* described in Ult-form.txt http://www.wotsit.org/music.htm
*/
static int read_ultratracker_header(int chan)
{
chans = 1;
header_distributed = 0;
data_location = 64;
true_file_length = lseek(chan,0L,SEEK_END);
data_size = (true_file_length-data_location);
srate = 8000;
data_format = SNDLIB_16_LINEAR_LITTLE_ENDIAN;
data_size = mus_bytes2samples(data_format,data_size);
return(0);
}
/* ------------------------------------ Sample dump exchange -------------------------------------
*
* 0: SDX:
* sdx2tx.c (Mark Lakata) reads from 4 for 26 (^z), then
* version (1)
* comment as pascal-style string (byte len, bytes chars)
* then 23 bytes:
* 0: packing (0=pcm)
* 1: midi channel
* 2 + 256*[3]: sample number
* 4: sample format (15: 16 bit unsigned(?), 8: 8bit unsigned(?)
* 5: sample rate (big int?)
* 9: sample length
* 13: loop start
* 17: loop end
* 21: loop type
* 22: reserved
*/
static int read_sample_dump_header(int chan)
{
int i,len;
for (i=4;i<HDRBUFSIZ;i++) if (hdrbuf[i] == 26) break;
len = hdrbuf[i+2];
if (len>0)
{
comment_start = i+3;
comment_end = i+3+len;
}
seek_and_read(chan, (unsigned char *)hdrbuf, i+3+len, HDRBUFSIZ);
srate = mus_little_endian_int((unsigned char *)(hdrbuf+5));
/* data_size = mus_little_endian_int((unsigned char *)(hdrbuf+9)); */
if ((srate<100) || (srate>100000)) srate = 8000;
chans = 1;
header_distributed = 0;
data_location = i+3+len+23;;
true_file_length = lseek(chan,0L,SEEK_END);
data_size = (true_file_length-data_location);
data_format = SNDLIB_16_UNSIGNED_LITTLE_ENDIAN;
data_size = mus_bytes2samples(data_format,data_size);
return(0);
}
/* ------------------------------------ Digiplayer ST3 -------------------------------------
*
* 0: 1 (use 'SCRS' at 76)
* 1: name
* 13: nada
* 14: "paragraph" offset of sample data
* 16: length in bytes (looks like #samples in the actual files...)
* 20: loop start
* 24: loop end
* 28: vol
* 29: ?
* 30: 0=unpacked, 1=DP30ADPCM
* 31: bits: 0=loop, 1=stereo (chans not interleaved!), 2=16-bit samples (little endian)
* 32: freq
* 36: nada
* 40: nada
* 42: 512
* 44: date?
* 48: sample name (28 char ASCIZ)
* 76: 'SCRS'
* 80: data starts
*
* info from http://www.wotsit.org/music.htm (I forgot to write down the file name)
*/
static int read_digiplayer_header(int chan)
{
chans = 1;
header_distributed = 0;
data_location = 80;
true_file_length = lseek(chan,0L,SEEK_END);
data_size = (true_file_length-data_location);
srate = 8000;
data_format = SNDLIB_16_UNSIGNED_LITTLE_ENDIAN;
if (hdrbuf[30]&2) chans=2;
if (hdrbuf[30]&1)
data_format = SNDLIB_UNSUPPORTED;
else
{
if (hdrbuf[30]&4) data_format = SNDLIB_8_UNSIGNED; /* may be backwards -- using Convert 1.4 output here */
}
data_size = mus_bytes2samples(data_format,data_size);
return(0);
}
/* ------------------------------------ CSRE adf -------------------------------------
*
* Info from Stuart Rosen
*
* 0-7: CSRE40
* 8: samples in file (long)
* 12: center line(?) (long)
* 16: start channel(?) (unsigned)
* 18: bits -- 12 or 16 (unsigned) -- is 12 bit sample file packed?
* 20: number system (0=signed, 1=unsigned)
* 22: srate in kHz (float)
* 26: peak sample in file (long) (can be 0)
* 30-511: comment possibly
*
* probably always little-endian (S.R. reads each sample using sizeof(int) -> 16 bits I think)
* if 12-bit unsigned we need to handle the offset somewhere
*/
static int read_adf_header(int chan)
{
int bits,numsys;
lseek(chan,0,SEEK_SET);
read(chan,hdrbuf,30);
chans = 1;
numsys = mus_little_endian_unsigned_short((unsigned char *)(hdrbuf+20));
bits = mus_little_endian_unsigned_short((unsigned char *)(hdrbuf+18));
if ((bits == 16) || (bits == 12))
{
if (numsys == 0)
data_format = SNDLIB_16_LINEAR_LITTLE_ENDIAN;
else data_format = SNDLIB_16_UNSIGNED_LITTLE_ENDIAN;
}
else data_format = SNDLIB_UNSUPPORTED;
srate = (int)mus_little_endian_float((unsigned char *)(hdrbuf+22));
data_size = mus_little_endian_int((unsigned char *)(hdrbuf+8));
data_location = 512;
true_file_length = lseek(chan,0L,SEEK_END);
return(0);
}
/* ------------------------------------ Diamondware -------------------------------------
*
* info from Keith Weiner at DiamondWare (www.dw.com):
*
* 0-22: DWD Header Byte "DiamondWare Digitized\n\0"
* 23: 1A (EOF to abort printing of file)
* 24: Major version number
* 25: Minor version number
* 26-29: Unique sound ID (checksum XOR timestamp)
* 30: Reserved
* 31: Compression type (0=none)
* 32-33: Sampling rate (in Hz)
* 34: Number of channels (1=mono, 2=stereo) (interleaved)
* 35: Number of bits per sample (8, 16) (all data signed)
* 36-37: Absolute value of largest sample in file
* 38-41: length of data section (in bytes)
* 42-45: # samples (16-bit stereo is 4 bytes/sample)
* 46-49: Offset of data from start of file (in bytes)
* 50-53: Reserved for future expansion (markers)
* 54-55: Padding
* 56:offset -- additional text: field=value
* suggested fields: TITLE, ORGARTIST, GENRE, KEYWORDS, ORGMEDIUM, EDITOR, DIGITIZER, COMMENT, SUBJECT, COPYRIGHT, SOFTWARE, CREATEDATE
*
* since this is all Windows/DOS oriented, I'll assume little-endian byte order.
*/
static int read_diamondware_header(int chan)
{
lseek(chan,0,SEEK_SET);
read(chan,hdrbuf,64);
chans = hdrbuf[34];
if (hdrbuf[31] == 0)
{
if (hdrbuf[35] == 8) data_format = SNDLIB_8_LINEAR;
else data_format = SNDLIB_16_LINEAR_LITTLE_ENDIAN;
}
else data_format = SNDLIB_UNSUPPORTED;
srate = mus_little_endian_unsigned_short((unsigned char *)(hdrbuf+32));
data_size = mus_little_endian_int((unsigned char *)(hdrbuf+38));
if (data_format != SNDLIB_UNSUPPORTED)
data_size = mus_bytes2samples(data_format,data_size);
data_location = mus_little_endian_int((unsigned char *)(hdrbuf+46));
true_file_length = lseek(chan,0L,SEEK_END);
return(0);
}
/* ------------------------------------ Comdisco SPW -------------------------------------
* info from AFsp libtsp/AF/nucleus/AFgetSWpar.c
*
* header is text as in NIST:
*
* $SIGNAL_FILE 9\n (12 chars)
* $USER_COMMENT
* <comment line(s)>
* $COMMON_INFO
* SPW Version = 3.10
* System Type = <machine> (e.g. "sun4","hp700")
* Sampling Frequency = <Sfreq> (e.g. "8000")
* Starting Time = 0
* $DATA_INFO
* Number of points = <Nsamp> (e.g. "2000")
* Signal Type = <type> ("Double", "Float", "Fixed-point", "Integer", "Logical")
* Fixed Point Format = <16,0,t> <16,16,t> <8,8,t> <8,0,t> (optional)
* Complex Format = Real_Imag (optional)
* $DATA <data_type> ("ASCII", "BINARY")
*
* the fixed point <n,m,b> is decoded as n=number of bits total per sample, m=integer bits, b=t: signed, u: unsigned
* if $DATA ASCII, data is ascii text as in IEEE text files.
* There are other complications as well. We'll just hack up a stop-gap until someone complains.
*/
static int read_comdisco_header (int chan)
{
/* need to grab a line at a time, call strcmp over and over. This is very tedious. */
char *line = NULL;
char portion[32];
char value[32];
int i,j,k,m,n,curend,commenting,offset,len,little,type;
int happy = 1;
k = 15;
line = (char *)CALLOC(256,sizeof(char));
little = 0;
offset = 0;
type = 0;
curend = INITIAL_READ_SIZE;
commenting = 0;
while (happy)
{
for (i=0;i<256;i++)
{
if (k == curend)
{
offset += curend;
read(chan,hdrbuf,HDRBUFSIZ);
k = 0;
curend = HDRBUFSIZ;
}
if (hdrbuf[k] == '\n') {k++; break;}
line[i] = hdrbuf[k++];
}
line[i] = '\0';
if ((strcmp(line,"$DATA BINARY") == 0) || (strcmp(line,"$DATA ASCII") == 0)) {happy = 0; data_location = offset+k;}
if (strcmp(line,"$USER_COMMENT") == 0)
{
comment_start = offset+k;
commenting = 1;
}
else
{
if (commenting)
{
if (line[0] == '$')
{
comment_end = offset+k-2-strlen(line);
commenting = 0;
}
}
}
if (line[0] != '$')
{
len = strlen(line);
for (j=0;j<8;j++) portion[j]=line[j];
portion[8]='\0';
for (j=8;j<len;j++) if (line[j] == '=') break;
for (n=0,m=j+2;m<len;m++,n++) value[n]=line[m];
value[n]='\0';
if (strcmp(portion,"Sampling") == 0) sscanf(value,"%d",&srate); else
if (strcmp(portion,"Number o") == 0) sscanf(value,"%d",&data_size); else
if (strcmp(portion,"Signal T") == 0) {if (value[1] == 'o') type=2; else if (value[1] == 'l') type=1;} else
if (strcmp(portion,"Fixed Po") == 0) {if (value[1] == '8') type=3;}
}
}
/* now clean up this mess */
chans = 1;
header_distributed = 0;
switch (type)
{
case 0: if (little) data_format = SNDLIB_16_LINEAR_LITTLE_ENDIAN; else data_format = SNDLIB_16_LINEAR; break;
case 1: if (little) data_format = SNDLIB_32_FLOAT_LITTLE_ENDIAN; else data_format = SNDLIB_32_FLOAT; break;
case 2: if (little) data_format = SNDLIB_64_DOUBLE_LITTLE_ENDIAN; else data_format = SNDLIB_64_DOUBLE; break;
case 3: data_format = SNDLIB_8_LINEAR; break;
}
true_file_length = lseek(chan,0L,SEEK_END);
FREE(line);
return(0);
}
/* ------------------------------------ MS ASF -------------------------------------
*
* asf format is described at http://www.microsoft.com/asf/specs.htm
* http://www.microsoft.com/asf/spec3/ASF0198ps.exe
*
* this header is completely insane
*/
static int read_asf_header (int chan)
{
/* a chunked data format, so not really acceptable here or elsewhere -- needs to be unchunked */
int len,ilen = 0,i,j,asf_huge,present,bits = 0;
/* apparently "huge" has some meaning in Windoze C */
len = mus_little_endian_int((unsigned char *)(hdrbuf+16)); /* actually 64 bits */
i = (128+64) / 8;
asf_huge = 0;
srate = 0;
chans = 0;
while (i<len)
{
seek_and_read(chan,(unsigned char *)hdrbuf,i,HDRBUFSIZ);
if ((unsigned int)(hdrbuf[1]) == 0x29)
switch (hdrbuf[0])
{
case 0xd0:
asf_huge = (hdrbuf[((128+64+128+64+64+64+64+32)/8)] & 2);
break;
case 0xd4:
present = ((hdrbuf[16+8+16+8+8+ 4+4+4+4+ 4+4] >> 3) & 0x3);
if (present)
j = 16+8+16+8+8+ 4+4+4+4+ 4+4+ 4+ (4+4+4) + 2;
else j = 16+8+16+8+8+ 4+4+4+4+ 4+4+ 4+ 2;
srate = mus_little_endian_int((unsigned char *)(hdrbuf+j+11+36));
bits = mus_little_endian_int((unsigned char *)(hdrbuf+j+11+32));
chans = mus_little_endian_unsigned_short((unsigned char *)(hdrbuf+j+65));
original_data_format = mus_little_endian_int((unsigned char *)(hdrbuf+j+11));
break;
default: break;
}
ilen = mus_little_endian_int((unsigned char *)(hdrbuf+16));
if (ilen <= 0) break;
if ((chans>0) && (srate>0)) break;
i+=ilen;
}
i=len;
seek_and_read(chan,(unsigned char *)hdrbuf,i,HDRBUFSIZ);
data_format = SNDLIB_UNSUPPORTED;
if (((unsigned int)(hdrbuf[1]) == 0x29) && ((unsigned int)(hdrbuf[0]) == 0xd2))
{
ilen = mus_little_endian_int((unsigned char *)(hdrbuf+16));
if (asf_huge) asf_huge=4; else asf_huge=2;
data_location = i+20+asf_huge+2+4+3+1;
if (bits == 0) bits=8;
data_format = wave_to_sndlib_format(original_data_format,bits,1);
}
if (data_format != SNDLIB_UNSUPPORTED)
data_size = mus_bytes2samples(data_format,(ilen - data_location));
return(0);
}
/* ------------------------------------ no header, Sound Designer II, SoundEdit, SoundEdit 16 -------------------------------------
*
* Sound Designer II data fork is a raw data file -- interleaved channels, bytes in sequence.
* The associated resource fork under "STR " has "sample-size" "sample-rate" and "channels".
* Similarly the resources "sdDD" id 1000, "sdML" id 1000, and "sdLL" id 1000 return pascal records of
* respectively Document Data, Markers and text, Loop data. I don't see any useful info in any of these
* except perhaps the file comment in the "sdDD" record of type str255 10 bytes in (I think).
* See headers.lisp for a reader for some of this stuff.
* file type: 'Sd2f'
*
* SoundEdit data fork contains 8-bit unsigned linear samples. The resource fork has:
* REPT: 0: mouse pos
* 4: length of sound in samples
* 8: sample number of beginning of selection (if any)
* 12: ditto end of selection
* 16: ditto loopback beginning (if any)
* 20: ditto loopback end
* INFO: 0: nada
* 8: "record frequency"
* 12: ditto for playback
* 16: compression mode: 0=none
* 20: chans (0=mono, 1=stereo)
* 24: window position
* 28: sample rate: 1=22KHz, 2=11, 3=7, 4=5
* 32: length of left channel in samples
* 36: if stereo, ditto right channel
* file type: 'FSDD', creator: 'SFX!'
*
* SoundEdit 16 data fork has either 8 or 16-bit linear unsigned samples. The resource fork has:
* TRKS: 0: -7 (version number)
* 4: track length in bytes
* 8: track offset in samples
* 12: left/right stereo
* 16: track selected flag
* 20: sample rate
* 22: pstring -- track title (31 chars, fixed size)
* 54: (?) gain
* one TRKS resource for each track in the file
* INFO: essentially the same as above, but with added version number at
* 40: 0=SoundEdit, -2=SoundEdit 16
* 44: reserved
* 52: sample rate
* and then other stuff that doesn't look interesting
* LABS: 16: pstring (31 chars) track label
* CUES: 0: sample location
* 4: pstring (31 chars) cue point text
* PReS: spectral stuff
* PRnt: printer stuff
* CLRS: colors stuff
* REPT: same as above
* file type: 'jB1 ', creator: 'jBox'.
*/
static int header_raw_srate = 44100;
static int header_raw_chans = 2;
static int header_raw_format = SNDLIB_16_LINEAR;
static int read_no_header (int chan)
{
srate = header_raw_srate;
chans = header_raw_chans;
data_format = header_raw_format;
header_distributed = 0;
data_location = 0;
data_size = lseek(chan,0L,SEEK_END);
true_file_length = data_size;
data_size = mus_bytes2samples(data_format,data_size);
return(0);
}
void mus_set_raw_header_defaults(int sr, int chn, int frm)
{
header_raw_srate = sr;
header_raw_chans = chn;
header_raw_format = frm;
}
#ifdef MACOS
/* try to read SDII resources */
/* apparently c2pstr or the reverse operate in place!! so we copy the strings first */
#include <Resources.h>
#include <Sound.h>
static int mus_resource(char *resname)
{
Handle sndH;
OSErr err;
char *val,*tmp;
int rtn = -1;
tmp = (char *)CALLOC(strlen(resname)+2,sizeof(char));
strcpy(tmp,resname);
sndH = GetNamedResource('STR ',(const unsigned char *)c2pstr(tmp));
FREE(tmp);
err = ResError();
if ((err == noErr) && (sndH))
{
LoadResource(sndH);
val = p2cstr((unsigned char *)(*sndH));
sscanf(val,"%d",&rtn);
ReleaseResource(sndH);
}
return(rtn);
}
#endif
/* ------------------------------------ all together now ------------------------------------ */
int mus_read_header_with_fd (int chan)
{
/* returns 0 on success (at least to the extent that we can report the header type), -1 for error */
int i,happy,loc = 0;
read(chan,hdrbuf,INITIAL_READ_SIZE);
header_type = unsupported_sound_file;
data_format = SNDLIB_UNSUPPORTED;
comment_start = 0;
comment_end = 0;
if ((match_four_chars((unsigned char *)hdrbuf,I_DSND)) || (match_four_chars((unsigned char *)hdrbuf,I_DECN)))
{
header_type = NeXT_sound_file;
return(read_next_header(chan));
}
if (match_four_chars((unsigned char *)hdrbuf,I_FORM))
{
/* next 4 bytes are apparently the file size or something equally useless */
if (match_four_chars((unsigned char *)(hdrbuf+8),I_AIFF))
{
header_type = AIFF_sound_file;
return(read_aiff_header(chan,0));
}
if (match_four_chars((unsigned char *)(hdrbuf+8),I_AIFC))
{
header_type = AIFC_sound_file;
return(read_aiff_header(chan,0));
}
if (match_four_chars((unsigned char *)(hdrbuf+8),I_8SVX))
{
header_type = SVX_sound_file;
return(read_8svx_header(chan));
}
if (match_four_chars((unsigned char *)(hdrbuf+8),I_MAUD))
{
header_type = MAUD_sound_file;
return(read_maud_header(chan));
}
/* apparently SAMP here -> sampled audio data (?) */
mus_error(MUS_UNSUPPORTED_HEADER_TYPE,"unrecognized header");
return(-1); /* i.e. unknown FORM header */
}
if ((match_four_chars((unsigned char *)hdrbuf,I_RIFF)) || match_four_chars((unsigned char *)hdrbuf,I_RIFX))
{
if (match_four_chars((unsigned char *)(hdrbuf+8),I_WAVE))
{
header_type = RIFF_sound_file;
return(read_riff_header(chan));
}
if (match_four_chars((unsigned char *)(hdrbuf+8),I_sfbk))
{
header_type = soundfont_sound_file;
return(read_soundfont_header(chan));
}
if (match_four_chars((unsigned char *)(hdrbuf+8),I_AVI_))
{
header_type = AVI_sound_file;
return(read_avi_header(chan));
}
mus_error(MUS_UNSUPPORTED_HEADER_TYPE,"unrecognized header");
return(-1); /* unknown RIFF header */
}
if ((equal_big_or_little_endian((unsigned char *)hdrbuf,I_IRCAM_VAX)) ||
(equal_big_or_little_endian((unsigned char *)hdrbuf,I_IRCAM_SUN)) ||
(equal_big_or_little_endian((unsigned char *)hdrbuf,I_IRCAM_MIPS)) ||
(equal_big_or_little_endian((unsigned char *)hdrbuf,I_IRCAM_NEXT)))
{
header_type = IRCAM_sound_file;
return(read_ircam_header(chan));
}
if (match_four_chars((unsigned char *)hdrbuf,I_NIST))
{
header_type = NIST_sound_file;
return(read_nist_header(chan));
}
if (match_four_chars((unsigned char *)hdrbuf,I_SOUN))
{
if ((match_four_chars((unsigned char *)(hdrbuf+4),I_SMP1)) && (match_four_chars((unsigned char *)(hdrbuf+8),I_SMP2)))
{
header_type = SMP_sound_file;
return(read_smp_header(chan));
}
else
{
header_type = SNDT_sound_file;
return(read_sndt_header(chan));
}
mus_error(MUS_UNSUPPORTED_HEADER_TYPE,"unrecognized header");
return(-1);
}
if ((match_four_chars((unsigned char *)hdrbuf,I_VOC0)) && (match_four_chars((unsigned char *)(hdrbuf+4),I_VOC1)))
{
header_type = VOC_sound_file;
return(read_voc_header(chan));
}
if (match_four_chars((unsigned char *)hdrbuf,I_AVR_))
{
header_type = AVR_sound_file;
return(read_avr_header(chan));
}
if (mus_big_endian_short((unsigned char *)hdrbuf) == 1336)
{
header_type = SD1_sound_file;
return(read_sd1_header(chan));
}
if ((match_four_chars((unsigned char *)hdrbuf,I_ALaw)) && (match_four_chars((unsigned char *)(hdrbuf+4),I_Soun)))
{
header_type = PSION_sound_file;
return(read_psion_header(chan));
}
if ((match_four_chars((unsigned char *)hdrbuf,I_GF1P)) && (match_four_chars((unsigned char *)(hdrbuf+4),I_ATCH)))
{
header_type = gravis_sound_file;
return(read_gravis_header(chan));
}
if ((match_four_chars((unsigned char *)hdrbuf,I_DSIG)) && (match_four_chars((unsigned char *)(hdrbuf+4),I_NAL_)))
{
header_type = comdisco_sound_file;
return(read_comdisco_header(chan));
}
if ((match_four_chars((unsigned char *)hdrbuf,I_GOLD)) && (match_four_chars((unsigned char *)(hdrbuf+4),I__WAV)))
{
header_type = goldwave_sound_file;
return(read_goldwave_header(chan));
}
if ((match_four_chars((unsigned char *)hdrbuf,I_Diam)) && (match_four_chars((unsigned char *)(hdrbuf+4),I_ondW)))
{
header_type = DiamondWare_sound_file;
return(read_diamondware_header(chan));
}
if (match_four_chars((unsigned char *)hdrbuf,I_SRFS))
{
header_type = srfs_sound_file;
return(read_srfs_header(chan));
}
if (match_four_chars((unsigned char *)hdrbuf,I_CSRE))
{
header_type = ADF_sound_file;
return(read_adf_header(chan));
}
if ((hdrbuf[0] == 0xf0) && (hdrbuf[1] == 0x7e) && (hdrbuf[3] == 0x01))
{
header_type = MIDI_sample_dump;
chans = 1;
srate = srate = (int)(1.0e9 / (float)((hdrbuf[7] + (hdrbuf[8]<<7) + (hdrbuf[9]<<14))));
data_size = (hdrbuf[10] + (hdrbuf[11]<<7) + (hdrbuf[12]<<14));
/* since this file type has embedded blocks, we have to translate it elsewhere */
return(0);
}
/* no recognized magic number at start -- poke around in possible header for other types */
/* ESPS is either 0x00006a1a or 0x1a6a0000 at byte 16 */
if (equal_big_or_little_endian((unsigned char *)(hdrbuf+16),0x00006a1a))
{
header_type = ESPS_sound_file;
return(read_esps_header(chan));
}
lseek(chan,0,SEEK_SET);
read(chan,hdrbuf,256);
if ((hdrbuf[252]==64) && (hdrbuf[253]==195)) /* #o100 and #o303 */
{
header_type = SPPACK_sound_file;
return(read_sppack_header(chan));
}
if ((match_four_chars((unsigned char *)(hdrbuf+65),I_FSSD)) && (match_four_chars((unsigned char *)(hdrbuf+128),I_HCOM)))
{
header_type = HCOM_sound_file;
mus_error(MUS_UNSUPPORTED_HEADER_TYPE,"can't read HCOM header");
return(-1);
}
happy = 0;
for (i=0;i<NINRS;i++)
{
if (equal_big_or_little_endian((unsigned char *)hdrbuf,I_INRS[i]))
{
happy = 1;
loc = inrs_srates[i];
}
}
if (happy)
{
header_type = INRS_sound_file;
return(read_inrs_header(chan,loc));
}
if (mus_big_endian_unsigned_int((unsigned char *)hdrbuf) == 0xAAAAAAAA)
{
header_type = MUS10_sound_file;
mus_error(MUS_UNSUPPORTED_HEADER_TYPE,"can't read Mus10 header");
return(-1);
}
if ((match_four_chars((unsigned char *)hdrbuf,I_SPIB)) || (match_four_chars((unsigned char *)hdrbuf,I_S___)))
{
header_type = IEEE_sound_file;
mus_error(MUS_UNSUPPORTED_HEADER_TYPE,"can't read IEEE header");
return(-1);
}
if (match_four_chars((unsigned char *)hdrbuf,I_DVSM))
{
header_type=DVSM_sound_file;
mus_error(MUS_UNSUPPORTED_HEADER_TYPE,"can't read DVSM header");
return(-1);
}
if (match_four_chars((unsigned char *)hdrbuf,I_Drat))
{
header_type=RealAudio_sound_file;
mus_error(MUS_UNSUPPORTED_HEADER_TYPE,"can't read real audio header");
return(-1);
}
if (match_four_chars((unsigned char *)hdrbuf,I_MThd))
{
header_type=MIDI_file;
mus_error(MUS_UNSUPPORTED_HEADER_TYPE,"can't read Midi header");
return(-1);
}
if ((match_four_chars((unsigned char *)hdrbuf,I_Esig)) && (match_four_chars((unsigned char *)(hdrbuf+4),I_nalc)))
{
header_type=Esignal_file;
mus_error(MUS_UNSUPPORTED_HEADER_TYPE,"can't read Esignal header");
return(-1);
}
if (match_four_chars((unsigned char *)hdrbuf,I_SND_))
{
header_type = SBStudioII_sound_file;
return(read_sbstudio_header(chan));
}
if (match_four_chars((unsigned char *)hdrbuf,I_FSMt))
{
header_type = Farandole_sound_file;
return(read_farandole_header(chan));
}
if (match_four_chars((unsigned char *)hdrbuf,I_SDXc))
{
header_type = Sample_dump_sound_file;
return(read_sample_dump_header(chan));
}
if (match_four_chars((unsigned char *)hdrbuf,I_DDSF))
{
header_type = Delusion_sound_file;
return(read_delusion_header(chan));
}
if (match_four_chars((unsigned char *)hdrbuf,I_LM89))
{
header_type = Yamaha_TX16_sound_file;
return(read_tx16_header(chan));
}
if (match_four_chars((unsigned char *)hdrbuf,I_SY85))
{
header_type = Yamaha_SY85_sound_file;
return(read_sy85_header(chan));
}
if (match_four_chars((unsigned char *)hdrbuf,I_SY80))
{
header_type = Yamaha_SY99_sound_file;
return(read_sy85_header(chan));
}
if (match_four_chars((unsigned char *)hdrbuf,I_PRAM))
{
header_type = Kurzweil_2000_sound_file;
return(read_kurzweil_2000_header(chan));
}
if (match_four_chars((unsigned char *)(hdrbuf+35),I_UWFD))
{
header_type = Ultratracker_sound_file;
return(read_ultratracker_header(chan));
}
if (match_four_chars((unsigned char *)(hdrbuf+76),I_SCRS))
{
header_type = digiplayer_sound_file;
return(read_digiplayer_header(chan));
}
if (match_four_chars((unsigned char *)hdrbuf,I_covox))
{
header_type = Covox_sound_file;
return(read_covox_header(chan));
}
if (match_four_chars((unsigned char *)hdrbuf,I_DSPL))
{
header_type = SPL_sound_file;
return(read_spl_header(chan));
}
if (mus_big_endian_short((unsigned char *)hdrbuf) == 0x1A00)
{
header_type = DeskMate_sound_file;
return(read_deskmate_header(chan));
}
if (mus_big_endian_short((unsigned char *)(hdrbuf+44)) == 0x1A80)
{
header_type = DeskMate_2500_sound_file;
return(read_deskmate_2500_header(chan));
}
#ifdef SNDLIB_LITTLE_ENDIAN
if (mus_uninterpreted_int((unsigned char *)hdrbuf) == 0x01000800)
#else
if (mus_uninterpreted_int((unsigned char *)hdrbuf) == 0x00080001)
#endif
{
header_type = ADC_sound_file;
return(read_adc_header(chan));
}
if ((match_four_chars((unsigned char *)hdrbuf,I_ones)) &&
(match_four_chars((unsigned char *)(hdrbuf+12),I_FORM)))
{
/* possibly an OMF file with an embedded AIFF data file -- this is just a guess... */
header_type = OMF_sound_file;
return(read_aiff_header(chan,12));
/* another (apparently) along these lines is TOC */
}
if ((match_four_chars((unsigned char *)hdrbuf,I_zeros)) &&
(match_four_chars((unsigned char *)(hdrbuf+4),I_mdat)))
{
/* possibly quicktime?? */
header_type = Quicktime_sound_file;
return(read_qt_header(chan));
}
if ((match_four_chars((unsigned char *)hdrbuf,I_asf0)) &&
(match_four_chars((unsigned char *)(hdrbuf+4),I_asf1)) &&
(match_four_chars((unsigned char *)(hdrbuf+8),I_asf2)) &&
(match_four_chars((unsigned char *)(hdrbuf+12),I_asf3)))
{
header_type = asf_sound_file;
return(read_asf_header(chan));
}
header_type = raw_sound_file;
return(read_no_header(chan));
}
int mus_read_header (char *name)
{
int chan,err = 0;
#ifdef MACOS
int loc,happy;
char *dupname;
#endif
chan = mus_open_read(name);
if (chan == -1) return(-1);
err = mus_read_header_with_fd(chan);
#ifdef MACOS
/* on the Mac, we should look for the resource fork. */
/* we'll at least check SDII possibilities */
/* if I had an example of SoundEdit output, I'd add support for that too */
if (header_type == raw_sound_file)
{
dupname = (char *)CALLOC(strlen(name)+2,sizeof(char));
strcpy(dupname,name);
loc = OpenResFile((const unsigned char *)c2pstr(dupname));
FREE(dupname);
if (loc != -1)
{
happy = mus_resource("channels");
if (happy != -1)
{
/* found an SDII file?!? */
chans = happy;
srate = mus_resource("sample-rate");
happy = mus_resource("sample-size");
if (happy == 1)
data_format = SNDLIB_8_UNSIGNED;
else data_format = SNDLIB_16_LINEAR;
data_size = mus_bytes2samples(data_format,true_file_length);
header_type = SD2_sound_file;
}
CloseResFile(loc);
}
}
/* we could also check the file type:
* FInfo fi;
* err = GetFInfo((const unsigned char *)c2pstr(name),fd,&fi);
* if (err == noErr) <check fi.fdType against possibiliites given above>
*/
#endif
close(chan);
return(err);
}
int mus_write_header_with_fd (int chan, int type, int in_srate, int in_chans, int loc, int size, int format, char *comment, int len)
{
int siz;
siz = mus_samples2bytes(format,size);
switch (type)
{
case NeXT_sound_file: return(mus_write_next_header(chan,in_srate,in_chans,loc,siz,format,comment,len)); break;
case AIFC_sound_file: return(write_aif_header(chan,in_srate,in_chans,siz,format,comment,len,1)); break;
case AIFF_sound_file: return(write_aif_header(chan,in_srate,in_chans,siz,format,comment,len,0)); break;
case RIFF_sound_file: return(write_riff_header(chan,in_srate,in_chans,siz,format,comment,len)); break;
case IRCAM_sound_file: return(write_ircam_header(chan,in_srate,in_chans,format,comment,len)); break;
case NIST_sound_file: return(write_nist_header(chan,in_srate,in_chans,siz,format)); break;
case raw_sound_file:
data_location = 0;
data_size = mus_bytes2samples(format,siz);
srate = in_srate;
chans = in_chans;
header_type = raw_sound_file;
data_format = format;
break;
default:
{
mus_error(MUS_UNSUPPORTED_HEADER_TYPE,"can't write %s header",mus_header_type2string(type));
return(-1);
break;
}
}
return(0);
}
int mus_write_header (char *name, int type, int in_srate, int in_chans, int loc, int size, int format, char *comment, int len)
{
int chan,err;
chan = mus_create(name);
if (chan == -1) return(-1);
err = mus_write_header_with_fd(chan,type,in_srate,in_chans,loc,size,format,comment,len);
close(chan);
return(err);
}
int mus_update_header_with_fd(int chan, int type, int size)
{
/* do not export unless you remember that size here is in bytes! */
lseek(chan,0L,SEEK_SET);
switch (type)
{
case NeXT_sound_file: update_next_header(chan,size); break;
case AIFC_sound_file: case AIFF_sound_file: update_aiff_header(chan,size); break;
case RIFF_sound_file: update_riff_header(chan,size); break;
case IRCAM_sound_file: update_ircam_header(); break;
case NIST_sound_file: update_nist_header(chan,size); break;
case raw_sound_file: break;
default:
{
mus_error(MUS_UNSUPPORTED_HEADER_TYPE,"can't udpate %s header",mus_header_type2string(type));
return(-1);
break;
}
}
return(0);
}
int mus_update_header (char *name, int type, int size, int srate, int format, int chans, int loc)
{
int chan,siz,err;
chan = mus_reopen_write(name);
if (chan == -1) return(-1);
siz = mus_samples2bytes(format,size);
err = mus_update_header_with_fd(chan,type,siz);
if (type == NeXT_sound_file)
{
if (srate != 0)
{
lseek(chan,16L,SEEK_SET);
mus_set_big_endian_int((unsigned char *)hdrbuf,srate);
write(chan,hdrbuf,4);
}
if (chans != 0)
{
lseek(chan,20L,SEEK_SET);
mus_set_big_endian_int((unsigned char *)hdrbuf,chans);
write(chan,hdrbuf,4);
}
if (loc != 0)
{
lseek(chan,4L,SEEK_SET);
mus_set_big_endian_int((unsigned char *)hdrbuf,loc);
write(chan,hdrbuf,4);
}
}
close(chan);
return(err);
}
int mus_update_header_comment (char *name, int loc, char *comment, int len, int typ)
{
int chan;
chan = mus_reopen_write(name);
if (chan == -1) return(-1);
switch (typ)
{
case NeXT_sound_file: update_next_header_comment(chan,loc,comment,len); break;
case AIFC_sound_file: case AIFF_sound_file: update_aiff_header_comment(chan,comment,len); break;
case RIFF_sound_file: update_riff_header_comment(chan,comment,len); break;
case raw_sound_file: break;
default:
{
mus_error(MUS_UNSUPPORTED_HEADER_TYPE,"can't udpate %s header comment",mus_header_type2string(typ));
return(-1);
break;
}
}
return(0);
}
int mus_header_writable(int type, int format) /* -2 to ignore format for this call */
{
switch (type)
{
case NeXT_sound_file:
if (format == -2) return(1);
switch (format)
{
case SNDLIB_8_MULAW: case SNDLIB_8_LINEAR: case SNDLIB_16_LINEAR: case SNDLIB_24_LINEAR:
case SNDLIB_32_LINEAR: case SNDLIB_32_FLOAT: case SNDLIB_64_DOUBLE: case SNDLIB_8_ALAW:
case SNDLIB_32_LINEAR_LITTLE_ENDIAN:
return(1); break;
default:
return(0); break;
}
break;
case NIST_sound_file:
if (format == -2) return(1);
switch (format)
{
case SNDLIB_8_LINEAR: case SNDLIB_16_LINEAR: case SNDLIB_24_LINEAR: case SNDLIB_32_LINEAR:
case SNDLIB_16_LINEAR_LITTLE_ENDIAN: case SNDLIB_24_LINEAR_LITTLE_ENDIAN: case SNDLIB_32_LINEAR_LITTLE_ENDIAN:
return(1); break;
default:
return(0); break;
}
break;
case AIFC_sound_file:
if (format == -2) return(1);
switch (format)
{
case SNDLIB_16_LINEAR: case SNDLIB_24_LINEAR: case SNDLIB_32_LINEAR:
case SNDLIB_8_LINEAR: case SNDLIB_8_MULAW: case SNDLIB_8_ALAW:
case SNDLIB_32_FLOAT: case SNDLIB_64_DOUBLE: case SNDLIB_8_UNSIGNED: case SNDLIB_16_UNSIGNED:
case SNDLIB_16_LINEAR_LITTLE_ENDIAN: case SNDLIB_24_LINEAR_LITTLE_ENDIAN: case SNDLIB_32_LINEAR_LITTLE_ENDIAN:
return(1);
break;
default:
return(0);
break;
}
break;
case AIFF_sound_file:
if (format == -2) return(1);
switch (format)
{
case SNDLIB_16_LINEAR: case SNDLIB_24_LINEAR: case SNDLIB_32_LINEAR: case SNDLIB_8_LINEAR:
return(1);
break;
default:
return(0);
break;
}
break;
case RIFF_sound_file:
if (format == -2) return(1);
switch (format)
{
case SNDLIB_8_MULAW: case SNDLIB_8_ALAW: case SNDLIB_8_UNSIGNED: case SNDLIB_32_FLOAT_LITTLE_ENDIAN:
case SNDLIB_16_LINEAR_LITTLE_ENDIAN: case SNDLIB_24_LINEAR_LITTLE_ENDIAN: case SNDLIB_32_LINEAR_LITTLE_ENDIAN:
return(1);
break;
default:
return(0);
break;
}
break;
case IRCAM_sound_file:
if (format == -2) return(1);
switch (format)
{
case SNDLIB_8_MULAW: case SNDLIB_8_ALAW: case SNDLIB_16_LINEAR: case SNDLIB_32_LINEAR: case SNDLIB_32_FLOAT:
return(1);
break;
default:
return(0);
break;
}
break;
case raw_sound_file:
return(format != SNDLIB_32_VAX_FLOAT); /* the only type we don't write */
break;
default: return(0); break;
}
return(0);
}
/* ------------------------------------------------------------------------
* old Mus10, SAM formats, just for completeness
*
* These were used for sound data on the PDP-10s at SAIL and CCRMA in the 70's and 80's.
* The word length was 36-bits. (SAM here refers to the Samson box, the Systems Concepts
* Digital Synthesizer, built by Peter Samson), not the SAM format associated (apparently)
* with ModEdit.
*
* "New" format as used by nearly all CCRMA software pre-1990:
*
* WD 0 - '525252525252 (chosen because "it is unlikely to occur at random"!)
* WD 1 - Clock rate in Hz (PDP-10 36-bit floating point)
* PDP-10 floating point format was sign in bit 0, excess 128 exponent in 1-8, fraction in 9-35
* WD 2 - #samples per word,,pack-code, (has # samples per word in LH, pack-code in RH)
* 0 for 12-bit fixed point (this was the main one)
* 1 for 18-bit fixed point
* 2 for 9-bit floating point incremental (never used)
* 3 for 36-bit floating point (never used)
* 4 for 16-bit sambox fixed point, right justified
* 5 for 20-bit sambox fixed point
* 6 for 20-bit right-adjusted fixed point (sambox SAT format=SAM output)
* 7 for 16-bit fixed point, left justified (never used)
* N>9 for N bit bytes in ILDB format (never used)
* WD 3 - # channels
* WD 4 - Maximum amplitude (if known) (PDP-10 float)
* WD 5 - number of Sambox ticks per pass (inverse of Sambox clock rate, sort of)
* WD 6 - Total #samples in file. If 0 then #wds_in_file*#samps_per_wd assumed.
* WD 7 - Block size (if any). 0 means sound is not blocked.
* WDs '10-'77 Reserved for EDSND usage, but I used it anyway.
* WDs '100-'177 Text description of file (in ASCIZ format) (ASCIZ=ASCII+null if I remember right = C string(?))
*
* "Old" format (pre-1977)
*
* WD 0 - '525252525252
* WD 1 - Clock rate => rate as integer,,code
* code=0 for 6.4Kc (or anything else)
* =1 for 12.8Kc, =2 for 25.6Kc, =3 for 51.2Kc
* =5 for 102.4Kc, =6 for 204.8Kc
* WD 2 - #sample per word,,pack
* 0 for 12 bit
* 1 for 16 bit (18 bit)
* 2 for 9 bit floating point incremental
* 3 for 36-bit floating point
* N>9 for N bit bytes in ILDB format
* WD 3 - # channels
* WD 4 - Maximum amplitude (if known, otherwise 0)
* WDs 5-77 Reserved for future expansion
* WDs 100-177 Text description of file (in ASCIZ format)
*
* ------------------------------------
* and even more esoteric... from the MIT PDP-6 (1971-76):
* JRST 1
* 0
* blkcnt,,2000
* duration,, --- ; negative halfword number to be counted down for duration
* v0 v1 v2 v3 v4 v5 ; 6 six-bit byes, each encoding pitch no. from 2.-63, 63 not used
* duration,, --
* v0 v1 v2 v3 v4 v5
* ...
* checksum
* ...
* blkcnt,,blkaddr ; neg half word in block, not counting itself or checksum,,addr of block start
* ...
* 0,,--
* 0
* checksum ;=ROT 1 and ADD including blkcnt,,blkaddr
* -1,,41
* JRST 101
* checksum
* -1,,41
* JRST 101
* checksum
* JUMPA 101
* JUMPA 101
*
* ah, the good old days...
*/
#if 0
One supported sample type is the DVSM-Format of the programs WinRec, WinCut
and Fortune. Those are programs for the ATARI FALCON.
A DVSM sample file has the following structure:
char magic[6]; /* "DVSM" */
int headlen; /* Headlen in Bytes*/
int freq; /* Sample freqency 0=8kHz 7=50kHz*/
char pack; /* 0 unpacked, 2=DVS packmethod*/
char mode; /* 0=Stereo 8Bit,1=Stereo 16Bit,2=Mono 8Bit,3=Mono 16*/
long blocklen; /* if pack>0: Length of a packed block*/
followed by cookies and the sound data.
The sample frequencies 0 to 7 correspond to the following frequencies:
sam_freq[8]={8195,9834,12292,16390,19668,24585,32778,49170};
For further information refer to the WinRec documentation or take a look at
the article series 'Sound Sample Formate' in the german magazin ST Computer
(3/94 - 6/94).
Confusingly enough there also appears to be a DVMS header (see sox11 cvsd.c)
struct dvms_header {
char Filename[14];
unsigned Id;
unsigned State;
time_t Unixtime;
unsigned Usender;
unsigned Ureceiver;
ULONG Length;
unsigned Srate;
unsigned Days;
unsigned Custom1;
unsigned Custom2;
char Info[16];
char extend[64];
unsigned Crc;
};
#define DVMS_HEADER_LEN 120
#endif
#if 0
/*
SCRIBE is headerless, 16-bit little endian data
(Hidden Markov Model ToolKit output -- obscure!)
HTK format files consist of a contiguous sequence of samples preceded by a header. Each sample is a vector of either 2-byte
integers or 4-byte floats. 2-byte integers are used for compressed forms as described below and for vector quantised data as
described later in section 5.11. HTK format data files can also be used to store speech waveforms as described in section 5.8.
The HTK file format header is 12 bytes long and contains the following data
nSamples -- number of samples in file (4-byte integer)
sampPeriod -- sample period in 100ns units (4-byte integer)
sampSize -- number of bytes per sample (2-byte integer)
parmKind -- a code indicating the sample kind (2-byte integer)
The parameter kind consists of a 6 bit code representing the basic parameter kind plus additional bits for each of the possible
qualifiers . The basic parameter kind codes are
0 WAVEFORM sampled waveform
1 LPC linear prediction filter coefficients
2 LPREFC linear prediction reflection coefficients
3 LPCEPSTRA LPC cepstral coefficients
4 LPDELCEP LPC cepstra plus delta coefficients
5 IREFC LPC reflection coef in 16 bit integer format
6 MFCC mel-frequency cepstral coefficients
7 FBANK log mel-filter bank channel outputs
8 MELSPEC linear mel-filter bank channel outputs
9 USER user defined sample kind
10 DISCRETE vector quantised data
and the bit-encoding for the qualifiers (in octal) is
_E 000100 has energy
_N 000200 absolute energy suppressed
_D 000400 has delta coefficients
_A 001000 has acceleration coefficients
_C 002000 is compressed
_Z 004000 has zero mean static coef.
_K 010000 has CRC checksum
_O 020000 has 0'th cepstral coef.
*/
#endif
#if 0
/*
Esignal starts with 48 byte ASCII preamble:
"Esignal"\n
7 byte version id\n
7 byte left padded data format choice (i.e. " EDR1", EDR2, ASCII, or machine name) EDR1=4byte ints, EDR2=8 byte longs\n
7 byte preamble size (i.e. " 48")\n
7 byte left padded header size (bytes, includes preamble)\n
7 byte left padded record size\n
for example:
Esignal
0.0B
ASCII
48
502
-1
default formats are big endian ints and shorts, IEEE floats
array desc:
short data type code
short rank (i.e. nchans in this context I guess)
long dimensions (i.e. sample number? -- rank of these)
<data>
type codes:
1 NO_TYPE 10 UCHAR
2 ARRAY 11 BOOL
3 DOUBLE 12 DOUBLE_COMPLEX
4 FLOAT 13 FLOAT_COMPLEX
5 LONG 14 LONG_COMPLEX
6 ULONG 15 SHORT_COMPLEX
7 SHORT 16 SCHAR_COMPLEX
8 USHORT 17 CHAR
9 SCHAR 18 WCHAR
here's a simple example:
PREAMBLE
0 45 73 69 67 6e 61 6c 0a "Esignal\n" "magic number"
8 20 20 20 30 2e 30 42 0a " 0.0B\n" version
16 20 20 20 45 44 52 31 0a " EDR1\n" architecture
24 20 20 20 20 20 34 38 0a " 48\n" preamble size
32 20 20 20 20 32 38 35 0a " 285\n" header size
40 20 20 20 20 20 20 32 0a " 2\n" record size
START OF FIELD LIST
48 00 00 00 04 4 number of fields
FIRST FIELD: commandLine
52 00 00 00 0b 11 length of name
56 63 6f 6d 6d 61 6e 64 4c "commandL" field name
64 69 6e 65 "ine" ...
67 00 11 17 (CHAR) data type
69 00 01 1 number of dimensions
71 00 00 00 11 17 dimension
75 00 00 00 00 0 length of units
79 3f f0 00 00 00 00 00 00 1.0 scale
87 00 00 00 00 00 00 00 00 0.0 offset
95 00 00 00 00 0 number of axis names
99 00 01 1 (GLOBAL) occurrence class
101 45 72 65 63 6f 72 64 20 "Erecord " data
109 73 70 65 65 63 68 2e 73 "speech.s" ...
117 64 "d" ...
118 00 00 00 00 0 number of subfields
SECOND FIELD: startTime
122 00 00 00 09 9 length of name
126 73 74 61 72 74 54 69 6d "startTim" field name
134 65 "e" ...
135 00 03 3 (DOUBLE) data type
137 00 00 0 number of dimensions
139 00 00 00 01 1 length of units
143 73 "s" units
144 3f f0 00 00 00 00 00 00 1.0 scale
152 00 00 00 00 00 00 00 00 0.0 offset
160 00 00 00 00 0 number of axis names
164 00 01 1 (GLOBAL) occurrence class
166 00 00 00 00 00 00 00 00 0.0 data
174 00 00 00 00 0 number of subfields
THIRD FIELD: recordFreq
178 00 00 00 0a 10 length of name
182 72 65 63 6f 72 64 46 72 "recordFr" field name
190 65 71 "eq"
192 00 03 3 (DOUBLE) data type
194 00 00 0 number of dimensions
196 00 00 00 02 2 length of units
200 48 7a "Hz" units
202 3f f0 00 00 00 00 00 00 1.0 scale
210 00 00 00 00 00 00 00 00 0.0 offset
218 00 00 00 00 0 number of axis names
222 00 01 1 (GLOBAL) occurrence class
224 40 bf 40 00 00 00 00 00 8000.0 data
232 00 00 00 00 0 number of subfields
FOURTH FIELD: samples
236 00 00 00 07 7 length of name
240 73 61 6d 70 6c 65 73 "samples" field name
247 00 07 7 (SHORT) data type
249 00 01 1 number of dimensions
251 00 00 00 01 1 dimension
255 00 00 00 00 0 length of units
so we might 'read' these guys by getting the header length from the preamble,
and the data type by searching blindly for "samples" followed by the type
indication, then just guess at a sampling rate.
*/
#endif
#if 0
/* MIME uses base64 encoding which is something like: */
int decode_char (unsigned char c)
{
if (isupper (c)) return(c - 'A');
else if (islower (c)) return(c - 'a' + 26);
else if (isdigit (c)) return(c - '0' + 52);
else if (c == '+') return(62);
else if (c == '/') return(63);
return(0);
}
from_base64(char* cs, int *buf)
{
/* assume we grab 4 chars and set 3 ints on each call */
int dc1,dc2,dc3,dc4;
dc1=decode_char(cs[0]); dc2=decode_char(cs[1]); dc3=decode_char(cs[2]); dc4=decode_char(cs[3]);
buf[0]=((dc1<<2) | (dc2>>4)); buf[1]=(((dc2&0xf)<<4) | (dc3>>2)); buf[2]=(((dc3&0x3)<<6) | dc4);
}
#endif
/* ILS headers (info from P Kabal): Interactive Laboratory System.
* in 1988 info from ils@hub.ucsb.edu
* v4 is in longs, v3 in shorts
* data is 16-bit linear (probably big-endian) (locs below are in version format)
* 0: #pts in analysis window, 1: #autoregressive coeffs, 2: preemphasis, 3: shift per frame, 4: hamming if 'y'
* 5: #data blocks, 6: #resonances, 7: start frame, 8: #frames, 9: start sector, 10: #autoregressive coeffs
* 11: #pts past whole block, 12-15: 'field chars', 16: #frames, 17: autoreg flag, 18: #disk num of file
* 19: another disk number, 20: analysis mnemonic, 22-26: id chars, 35-36: scaling mult, 37-38: scaling adder
* 57: starting chan, 58: #chans, 59: mulaw if 50, 60: power of ten mult for sampling freq, 61: int srate,
* 62: -29000 if not sampled, -32000 if sampled, 63: 32149 if initialized.
* 128: data starts (or 256 in v3, I think) -- byte 512 in any case.
*/
/* for MIDI and Synth-related formats by the dozen see http://www.wotsit.org/music.htm */
/*
* real audio: http://www1.real.com/devzone/sdks/rmsdk/guide/index.html
*/
/*
{
int i;
lseek(chan,0,SEEK_SET);
read(chan,hdrbuf,HDRBUFSIZ);
for (i=0;i<HDRBUFSIZ;i++)
{
fprintf(stderr,"%d: %d %c %d %d %d %d\n",
i,
hdrbuf[i],hdrbuf[i],
mus_little_endian_short((unsigned char *)(hdrbuf+i)),
mus_big_endian_short((unsigned char *)(hdrbuf+i)),
mus_little_endian_int((unsigned char *)(hdrbuf+i)),
mus_big_endian_int(unsigned char *)(hdrbuf+i));
}
}
*/
/* from /usr/share/magic:
# Real Audio (Magic .ra\0375)
0 belong 0x2e7261fd realaudio sound file
0 string .RMF\0\0\0 realmedia file
*/