removed

dev/externals/CMakeLists.txt

@@ -1,8 +0,0 @@
-add_subdirectory(portsf)
-add_subdirectory(fastconv)
-add_subdirectory(reverb)
-add_subdirectory(mctools)
-
-##if(USE_LOCAL_PORTAUDIO)
-  add_subdirectory(paprogs)
-##endif()

dev/externals/fastconv/CMakeLists.txt

@@ -1,21 +0,0 @@
-if(APPLE)
-  set(CMAKE_C_FLAGS "-O2 -Wall -mmacosx-version-min=10.8 -Dunix -fomit-frame-pointer -funroll-loops")
-  set(CMAKE_CXX_FLAGS "-O2 -Wall -mmacosx-version-min=10.8 -Dunix -fomit-frame-pointer -funroll-loops  -std=c++11 -stdlib=libc++")
-else()
-  if(MINGW)
-    set(CMAKE_C_FLAGS "-O3 -DWIN32 -D_WIN32 -fomit-frame-pointer  -funroll-loops")
-  else()
-    set(CMAKE_C_FLAGS "-O3 -Wall -Dlinux -Dunix -fomit-frame-pointer -funroll-loops")
-  endif()
-endif()
-
-link_directories(../lib)
-
-include_directories(../include)
-
-add_executable(fconv fconv.cpp genrespframe2.cpp mxfftd.c)
-
-target_link_libraries(fconv portsf sfsys ${EXTRA_LIBRARIES})
-
-my_install(fconv)
-

dev/externals/fastconv/fconv-fftw.cpp

@@ -1,768 +0,0 @@
-/*
- * Copyright (c) 1983-2013 Richard Dobson and Composers Desktop Project Ltd
- * http://people.bath.ac.uk/masrwd
- * http://www.composersdesktop.com
- * This file is part of the CDP System.
- * The CDP System is free software; you can redistribute it
- * and/or modify it under the terms of the GNU Lesser General Public 
- * License as published by the Free Software Foundation; either 
- * version 2.1 of the License, or (at your option) any later version. 
- *
- * The CDP System is distributed in the hope that it will be useful, 
- * but WITHOUT ANY WARRANTY; without even the implied warranty of 
- * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  
- * See the GNU Lesser General Public License for more details.
- * You should have received a copy of the GNU Lesser General Public
- * License along with the CDP System; if not, write to the Free Software
- * Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
- *
- */
-
-/* fastconv.cpp: */
-/*  self-contained prog for fast convolution (reverb, FIR filtering, bformat etc) 
- *  
-*/
-
-/* um, there is nothing C++ish about this code apart from use of new, delete! */
-
-/*TODO: control wet/dry mix with something... */
-/* auto-rescale arbitrary impulse responses? */
-/* Feb 2013: rebuilt with updated portsf */
-/* August 2013 epanded usage message re text file. */
-extern "C"
-{
-
-#include <portsf.h>
-
-}
-#include <stdio.h>
-#include <stdlib.h>
-#include <math.h>
-#include <string.h>
-#include <time.h>
-#include <sys/types.h>
-#include <sys/timeb.h>
-
-
-#ifdef _DEBUG
-#include <assert.h>
-#endif
-
-#ifdef unix
-#include <ctype.h>
-int stricmp(const char *a, const char *b);
-int strnicmp(const char *a, const char *b, const int length);
-#endif
-
-
-#include <rfftw.h>
-
-
-#ifndef WAVE_FORMAT_IEEE_FLOAT
-#define WAVE_FORMAT_IEEE_FLOAT (0x0003)
-#endif
-
-//#define FFTTEST
-
-/* convert from mono text impulse file */
-long genimpframe1(double *insbuf, double*** outbuf, long npoints, double scalefac);
-/* convert from multichan soundfile */
-long genimpframe2(int ifd,double*** outframe, double* rms,long imchans,double scalefac);
-void mc_split(double* inbuf,double** out,long insize,long chans);
-void mc_interl(double** in,double* out,long insize,long chans);
-void complexmult(double *frame,const double *impulse,long length);
-
-#ifdef FFTTEST
-extern "C"{
-void fft_(double *, double *,int,int,int,int);
-void fftmx(double *,double *,int,int,int,int,int,int *,double *,double *,double *,double *,int *,int[]);
-void reals_(double *,double *,int,int);
-}
-#endif
-
-#define DEFAULT_BUFLEN (32768)
-
-enum {ARG_NAME,ARG_INFILE,ARG_IMPFILE,ARG_OUTFILE,ARG_NARGS};
-
-void usage(const char *progname)
-{
-
-    printf("\n\n%s v1.2 RWD,CDP July 2010,2013",progname);
-	printf("\nMulti-channel FFT-based convolver");
-
-	printf("\nUsage: \n    %s [-aX][-f] infile impulsefile outfile [dry]\n"		               
-           "   -aX        : scale output amplitude by X\n"
-           "   -f         : write output as floats (no clipping)\n"
-           "  infile      : input soundfile to be processed.\n"
-           "  impulsefile : m/c soundfile or mono text file containing impulse response,\n"
-           "                  e.g. reverb or FIR filter.\n"
-           "                Text file name must have extension .txt.\n"
-		   "                  File must contain 1 column of floating point values,\n"
-		   "                  typically in the range -1.0 to 1.0.\n"
-           "              Supported channel combinations:\n"
-           "               (a) mono infile, N-channel impulsefile;\n"
-           "               (b) channels are the same;\n"
-           "               (c) mono impulsefile, N-channel infile.\n"
-           "  [dry]       :  set dry/wet mix (e.g. for reverb)\n"
-           "                 Range: 0.0 - 1.0,  default = 0.0\n"
-           "                 (uses sin/cos law for constant power mix)\n"
-           "Note: some recorded reverb impulses effectively include the direct signal.\n"
-           "In such cases  <dry>  need not be used\n"
-           "Where impulsefile is filter response (FIR), optimum length is power-of-two - 1.\n",progname
-		   );		   
-}
-
-
-double
-timer()
-{
-	struct timeb now;
-	double secs, ticks;	
-	ftime(&now);
-	ticks = (double)now.millitm/(1000.0);
-	secs = (double) now.time;
-
-	return secs + ticks;
-}
-
-
-void			
-stopwatch(int flag) 
-{
-	static double start;		    
-	long mins=0,hours=0;
-	double end, secs=0;
-
-	if(flag)
-		start = timer();	
-	else 
-	{
-		end    = timer(); 
-		secs   = end - start;
-		mins   = (long)(secs/60.0);
-		secs  -= mins*60.0; 
-		hours  = mins/60;
-		mins  -= hours*60;
-
-		fprintf(stderr,"\nElapsed time: ");
-		if(hours > 0)
-			fprintf(stderr,"%ld h ", hours);
-		if(mins > 0)
-			fprintf(stderr,"%ld m ", mins);
-		fprintf(stderr,"%2.3lf s\n\n",secs);
-	}
-}
-
-/* how do we want this to work?
-
-  (1)  multi-chan impulse:  EITHER: mono infile  OR infile with same chan count
-  (2)  mono impulse:  expanded to infile chancount
-     Therefore: need to open both infiles before deciding action
-*/
-#define PI2 (1.570796327)
-
-int main(int argc,char **argv)
-{
-
-	long fftlen = 1,imlen = 0,chans = 0,srate;
-	long buflen = 0;
-	double scalefac = 1.0f;
-	double ampfac = 1.0;
-    double Ninv = 1.0;
-	int insndfile = -1,inimpfile = -1,outsndfile = -1; 
-    int dorms = 0;
-    int nameoffset = 0;
-	PSF_PROPS inprops,outprops, impulseprops;
-    psf_format format;
-    PSF_CHPEAK  *fpeaks = NULL;
-	MYLONG peaktime;
-	int i,jj,minheader = 0,do_timer = 1;
-    int writefloats = 0;
-    long framesneeded;
-	double oneovrsr;
-    double *inmonobuf = 0;
-	double *insbuf=0,*outsbuf = 0;
-    double **inbuf = 0, **outbuf = 0;    
-	double **imbuf = 0;
-	double **overlapbuf = 0;
-    double rms = 0.0;
-    double dry = 0.0;
-    double wet = 1.0;
-#ifdef FFTTEST
-    double *anal = 0;
-#endif
-    rfftwnd_plan* forward_plan = 0;
-    rfftwnd_plan* inverse_plan = 0;
-	
-    if(argv[0][0]=='.' && argv[0][1]=='/'){
-        nameoffset  = 2;
-    }
-	
-    if(argc==2){
-        if(strcmp(argv[1],"--version")==0){
-            printf("1.2.0.\n");
-            return 0;
-        }
-    }
-
-	if(psf_init()){
-		puts("unable to start portsf\n");
-		return 1;
-	}
-
-
-	if(argc < ARG_NARGS){		 
-		usage(argv[0]+nameoffset);
-		return(1);
-	}
-
-    
-
-    while(argc > 1 && argv[1][0]=='-'){				
-		switch(argv[1][1]){
-        case 'a':
-            scalefac =  atof(&(argv[1][2]));
-            if(scalefac <=0.0){
-                printf("Error: scalefac must be positive!\n");
-                return 1;
-            }
-            break;        
-        case 'f':
-            writefloats = 1;
-            break;
-        default:
-            break;
-        }
-        argv++;
-        argc--;
-    }		
-	/* 2 legal possibilities: infile and outfile, or -I used with infile only */
-	if(argc< ARG_NARGS){
-		fprintf(stderr,"\nInsufficient arguments.");
-		usage(argv[0]+nameoffset);
-		return(1);
-	}
-    if(argc==ARG_NARGS+1){
-        double dryarg;
-        dryarg = atof(argv[ARG_NARGS]);
-        if(dryarg < 0.0 || dryarg > 1.0){
-            printf("dry value out of range.\n");
-            return 0;
-        }
-        if(dryarg==1.0){
-            printf("Warning: dry=1 copies input!\n");
-            wet = 0.0;
-        }
-        else{
-            dry = sin(PI2 * dryarg);
-            wet = cos(PI2 * dryarg);
-        }
-    }
-
-		
-
-/* TODO:  where -F[file] is combined with -A, it signifies create analysis file
-           compatible with impulse file (e.g 50% overlap, etc) */
-
-	
-	/* open infile, check props */	
-		
-	if((insndfile = psf_sndOpen(argv[ARG_INFILE],&inprops, 0)) < 0){
-		fprintf(stderr,"\nUnable to open input soundfile %s",argv[1]);
-        delete []imbuf;
-		return 1;
-	}
-    srate = inprops.srate;
-	if(srate <=0){
-		fprintf(stderr,"\nBad srate found: corrupted file?\n");
-        delete []imbuf;
-		return 1;
-	}
-	chans = inprops.chans;
-    framesneeded = psf_sndSize(insndfile);
-    if(framesneeded <= 0){
-        printf("Error in input file - no data!\n");
-        psf_sndClose(insndfile);
-        return 1;
-    }
-    /* open impulse file */
-    /* check for soundfile */
-      
-    format = psf_getFormatExt(argv[ARG_IMPFILE]);
-    if(format==PSF_FMT_UNKNOWN){  /* must be a text file */
-        FILE *fp = 0;
-        char tmp[80];
-        char* dot;
-        int size = 0,got = 0;
-        int read = 0;
-
-        dot = strrchr(argv[ARG_IMPFILE],'.');
-        if(dot==NULL || stricmp(dot,".txt") != 0){
-            fprintf(stderr,"Error: impulse text file must have .txt extension.\n");
-            return 1;
-        }
-        fp = fopen(argv[ARG_IMPFILE],"r");
-        if(fp==0){
-            printf("Cannot open impulse text file %s\n.",argv[ARG_IMPFILE]);
-            return 1;
-        }
-        /* count lines! */
-        while(fgets(tmp,80,fp) != NULL)
-            size++;
-        if(size==0){
-            printf("Impulse textfile %s has no data!.\n",argv[ARG_IMPFILE]);
-            return 1;
-        }
-        rewind(fp);
-            
-        inmonobuf = new double[size+1];
-        for(i=0;i < size;i++)  {
-            read = fscanf(fp,"%lf",&inmonobuf[i]);
-            if(read==0){
-                i--;
-                continue;
-            }
-            if(read==EOF)
-                break;
-            got++;
-        }
-        imlen = got;
-        impulseprops.chans = 1;
-        fclose(fp);
-    }
-    else{
-        if((inimpfile = psf_sndOpen(argv[ARG_IMPFILE],&impulseprops, 0))< 0){
-            fprintf(stderr,"\nUnable to open impulse file %s",argv[ARG_IMPFILE]);       
-            return 0;
-        }
-        //printf("impulse file sr = %d\n",impulseprops.srate);
-        if(srate != impulseprops.srate){
-            printf("Error: files must have same sample rate");
-            delete []imbuf;
-            return 1;
-        }
-        long  len = psf_sndSize(inimpfile);
-        if(len <= 0){
-            printf("Error in impulse soundfile - no data!\n");
-            return 1;
-        }
-        if(impulseprops.chans > 1){
-            if(! (chans == 1 || chans == impulseprops.chans)){
-                fprintf(stderr,"\nChannel mismatch.\n    Infile must be mono or have same channels as irfile.\n");
-                return(1);
-            }
-            chans = impulseprops.chans;
-        }    
-    }
-
-    imbuf = new double*[chans];
-
-    /* if impulse file is mono, we will need to copy data to the other buffers */
-    // allocate  impulseprops.chans buffers ; may need more
-
-    if(inimpfile >=0){
-	    if((imlen = genimpframe2(inimpfile,&imbuf,&rms,impulseprops.chans, scalefac)) == 0){
-	    	fprintf(stderr,"Error reading impulse file\n");
-	    	return(1);
-	    }
-    }
-    
-    else if(imlen){        
-        genimpframe1(inmonobuf,&imbuf,imlen,scalefac);       
-    }
-    printf("mean rms level = %.4lf (%.2lf dB)\n",rms, 20.0 * log10(rms));
-    
-
-    framesneeded +=  imlen;        /* can  close outfile, when this length reached */
-
-    while(fftlen < imlen*2)			/* convolution style - double-length */
-		fftlen <<= 1;
-    double norm = sqrt(2.0);       /* relative: rms of 0dBFS sine is 0.707 = 1/root2 */
-    Ninv = 1.0 / sqrt(imlen*2);
-    Ninv *= norm;
-    Ninv /= imlen;
-    // take simple avg of  rms for adjustment factor.
-    // may not adequately represent some responses, e.g. with strong attack/earlies, soft reverb tail */
-    double rmsdif2 =  (-20.0 * log10(rms)) * 0.5;
-    double rmsadjust = pow(10, rmsdif2/20.0);
-#ifdef _DEBUG
-    printf("rescaling factor = %.6e\n",Ninv);   
-    printf("rmsadjust = %.4lf\n",rmsadjust);
-#endif
-   Ninv /= rmsadjust; 
-
-
-    /* copy buffers if required */
-    for(i = impulseprops.chans;i < chans;i++){
-        imbuf[i] = new double[fftlen+2];
-        memcpy(imbuf[i],imbuf[0],(fftlen+2)* sizeof(double));
-    }         
-    oneovrsr = 1.0 / (double) srate;
-    
-    /*make sure buflen is at least fftlen */
-//	if(fftlen > buflen)
-//		buflen = fftlen;
-
-    
-	/* main i/o buffers */
-	
-	insbuf     = new double [fftlen  * chans];
-    outsbuf    = new double [fftlen  * chans];
-	inbuf      = new double*[chans];
-    outbuf     = new double*[chans];  /* overlap-add buffers */
-    overlapbuf = new double*[chans];
-    
-    for(i=0;i < chans;i++){
-        inbuf[i]  = new double[fftlen+2];   // main in-place buf 
-        outbuf[i] = new double[fftlen+2];        
-	    /* channel buffers */	
-        overlapbuf[i] = new double[fftlen/2];
-        memset(overlapbuf[i],0,sizeof(double)*(fftlen/2));        
-    }
-#ifdef FFTTEST
-    anal = new double[fftlen+2];
-#endif
-    
-    
-    forward_plan = new rfftwnd_plan[chans];
-    inverse_plan = new rfftwnd_plan[chans];
-    int insize = (int) fftlen;
-    for(i=0;i < chans;i++){		    
-	    memset(inbuf[i], 0, sizeof(double) * (fftlen+2));
-        memset(outbuf[i],0,sizeof(double) * (fftlen+2));
-        
-        forward_plan[i] = rfftwnd_create_plan_specific(1,&insize, 
-		FFTW_REAL_TO_COMPLEX, FFTW_ESTIMATE | FFTW_IN_PLACE,
-		inbuf[i],1,NULL,1);
-        inverse_plan[i] = rfftwnd_create_plan_specific(1,&insize, 
-		FFTW_COMPLEX_TO_REAL, FFTW_ESTIMATE | FFTW_IN_PLACE,
-		inbuf[i],1,NULL,1);
-    }
-
-    
-	/* use generic init function */
-/*bool phasevocoder::init(long outsrate,long fftlen,long winlen,long decfac,float scalefac,
-						pvoc_scaletype stype,pvoc_windowtype wtype,pvocmode mode)
-*/
-	
-	/*create output sfile */
-	psf_stype smptype;
-	psf_format outformat;
-	/* will it be aiff, wav, etc? */
-	outformat = psf_getFormatExt(argv[ARG_OUTFILE]);
-	if(outformat==PSF_FMT_UNKNOWN){
-		fprintf(stderr,"\nOutfile name has unrecognized extension.");
-        delete []imbuf;
-		return(1);
-	}
-		
-		
-	smptype = inprops.samptype;
-    if(writefloats)
-        smptype= PSF_SAMP_IEEE_FLOAT;
-
-	/*the one gotcha: if output is floats and format is aiff, change to aifc */
-	if(smptype==PSF_SAMP_IEEE_FLOAT && outformat==PSF_AIFF){
-		fprintf(stderr,"Warning: AIFF output written as AIFC for float samples\n");
-		outformat = PSF_AIFC;
-	}
-
-	outprops          = inprops;
-	outprops.chans    = chans;
-	outprops.srate    = srate;
-	outprops.format   = outformat;
-	outprops.samptype = smptype;
-	outprops.chformat = STDWAVE;
-    /* if irfile is BFormat, need to set outfile fmt likewise */
-    if(impulseprops.chformat==MC_BFMT)
-        outprops.chformat = MC_BFMT;
-    /* I suppose people will want automatic decoding too...grrr */
-
-    fpeaks = new PSF_CHPEAK[chans];
-	if(fpeaks==NULL){
-		puts("no memory for fpeak data buffer\n");
-		return 1;
-	}
-
-	/*last arg: no clipping of f/p samples: we use PEAK chunk */
-	if((outsndfile = psf_sndCreate(argv[ARG_OUTFILE],&outprops,0,minheader,PSF_CREATE_RDWR)) <0 ){
-		fprintf(stderr,"\nUnable to open outfile %s",argv[ARG_OUTFILE]);
-        delete []imbuf;
-		return(1);
-	}
-				
-
-	
-	printf("\n");
-	
-	
-	stopwatch(1);
-
-	long written,thisblock,framesread;
-    long frameswritten = 0;
-	double intime= 0.0,outtime = 0.0;
-    int last = 0;
-				
-	while((framesread = psf_sndReadDoubleFrames(insndfile,insbuf,imlen)) > 0){        
-		written = thisblock =  0;
-
-		for(i = framesread * inprops.chans; i< fftlen * inprops.chans; i++)
-			insbuf[i] = 0.0f;
-        framesread = imlen;
-		memset(inbuf[0],0,(fftlen+2) * sizeof(double));
-        if(chans == inprops.chans)  {
-            /* must clean buffers! */
-		    for(i=0;i < chans;i++)
-                memset(inbuf[i],0,(fftlen+2) * sizeof(double));
-            mc_split(insbuf,inbuf,imlen * chans, chans);
-
-        }
-        else{
-            for(i=0;i < chans;i++) {
-                memset(inbuf[i],0,(fftlen+2) * sizeof(double));
-                memcpy(inbuf[i],insbuf,imlen * sizeof(double));
-                memset(outbuf[i],0,sizeof(double) * (fftlen+2));
-
-            }
-        }
-        if(impulseprops.chans==1){
-            for(jj = 0; jj < chans;jj++){
-#ifdef FFTTEST
-                int zz;
-                memcpy(anal,inbuf[jj],(fftlen+2) * sizeof(double));
-                fft_(anal,anal+1,1,fftlen/2,1,-2);
-	            reals_(anal,anal+1,fftlen/2,-2);
-                for(zz=0;zz < fftlen+2;zz++)
-                    anal[zz] *= 0.001;
-#endif
-                rfftwnd_one_real_to_complex(forward_plan[jj],inbuf[jj],NULL);			    				
-                complexmult(inbuf[jj],imbuf[0],fftlen+2);                
-			    rfftwnd_one_complex_to_real(inverse_plan[jj],(fftw_complex * )inbuf[jj],NULL);				
-            }           
-        }
-        else{
-            for(jj = 0; jj < chans;jj++){
-				rfftwnd_one_real_to_complex(forward_plan[jj],inbuf[jj],NULL);                   				
-                complexmult(inbuf[jj],imbuf[jj],fftlen+2);                
-				rfftwnd_one_complex_to_real(inverse_plan[jj],(fftw_complex * )inbuf[jj],NULL);    				    			    
-            }            
-        }
-        
-        /* overlap-add  for each channel */
-        /* TODO: verify  use of imlen here - should it be fftlen/2 -1 ? */
-        for(jj=0;jj < chans;jj++){
-            for(i=0;i < imlen;i++) {
-                outbuf[jj][i] = inbuf[jj][i] + overlapbuf[jj][i];
-                overlapbuf[jj][i] = inbuf[jj][i+imlen];
-            }
-        }
-		mc_interl(outbuf,outsbuf,imlen,chans);
-        if(inprops.chans == chans){
-            for(i=0;i < framesread; i++) {
-                for(jj=0;jj < chans;jj++){
-                    long  outindex = i*chans + jj;            
-                    outsbuf[outindex] *= Ninv;
-                    outsbuf[outindex] *= wet;
-                    outsbuf[outindex] += dry * insbuf[outindex];
-                }
-            }
-        }
-        /* elso mono input */
-        else {
-            for(i=0;i < framesread; i++) {
-                for(jj=0;jj < chans;jj++){
-                    long  outindex = i*chans + jj;
-                    double inval = dry *  insbuf[i];
-                    outsbuf[outindex] *= Ninv;
-                    outsbuf[outindex] *= wet;
-                    outsbuf[outindex] += inval;
-                }
-            }
-        }
-
-        if((written = psf_sndWriteDoubleFrames(outsndfile,outsbuf,framesread)) != framesread){
-		    fprintf(stderr,"\nerror writing to outfile");
-		    return(1);		               
-        }
-        frameswritten += written;
-
-		if(do_timer){
-			intime += (double)framesread * oneovrsr;
-			printf("Input time: %.2lf\r",intime);
-		}    
-    }
-    /* complete tail */
-
-    if(frameswritten < framesneeded){
-        // TODO: imlen? see above
-        mc_interl(overlapbuf,outsbuf,imlen,chans);
-        long towrite = framesneeded - frameswritten; 
-        for(i=0;i < towrite * chans; i++) {
-            outsbuf[i] *= Ninv;
-            outsbuf[i] *= wet;
-        }
-        if((written = psf_sndWriteDoubleFrames(outsndfile,outsbuf,towrite)) != towrite){
-	        fprintf(stderr,"\nerror writing to outfile");
-	        return(1);
-	    }
-    }
-
-	stopwatch(0);
-    if(psf_sndReadPeaks( outsndfile,fpeaks,&peaktime)){
-        double peakmax = 0.0;
-		printf("PEAK values:\n");
-		for(i=0; i < chans; i++)	{
-            peakmax = fpeaks[i].val > peakmax ? fpeaks[i].val : peakmax;
-			if(fpeaks[i].val < 0.0001f)
-				printf("CH %d:\t%e (%.2lf dB)\tat frame %10lu :\t%.4lf secs\n",i+1,
-				fpeaks[i].val,20.0*log10(fpeaks[i].val),fpeaks[i].pos,(double) (fpeaks[i].pos) / (double)srate);
-			else
-				printf("CH %d:\t%.4lf (%.2lf dB)\tat frame %10lu :\t%.4lf secs\n",i+1,
-				fpeaks[i].val,20.0 * log10(fpeaks[i].val),fpeaks[i].pos,(double) (fpeaks[i].pos) / (double)srate);		
-		}
-        if(peakmax > 1.0)
-            printf("Overflows!  Rescale by %.10lf\n",1.0 / peakmax);
-	}
-
-	if(insndfile >=0)
-		psf_sndClose(insndfile);
-	if(outsndfile >=0)
-		psf_sndClose(outsndfile);
-    if(inimpfile >=0)
-	    psf_sndClose(inimpfile);
-	psf_finish();
-    delete [] fpeaks;
-	if(insbuf)
-		delete [] insbuf;
-	if(outsbuf)
-		delete [] outsbuf;
-    
-
-    for(i=0;i < chans;i++){
-        if(inbuf){       
-		    delete [] inbuf[i];
-        }           
-        if(outbuf){
-		    delete [] outbuf[i];
-        }       	
-        if(imbuf){        
-	        delete [] imbuf[i];
-        }
-        if(overlapbuf)
-            delete [] overlapbuf[i];
-        if(forward_plan)
-            rfftwnd_destroy_plan(forward_plan[i]);
-        if(inverse_plan)
-            rfftwnd_destroy_plan(inverse_plan[i]);
-    }
-    delete [] outbuf;
-    delete [] inbuf;
-    delete [] imbuf;
-    delete [] overlapbuf;
-	return 0;
-}
-
-
-// insize is raw samplecount,buflen is insize/chans
-void mc_split(double* inbuf,double** out,long insize,long chans)
-{
-    long i,j,buflen = insize/chans;
-    double* pinbuf;
-
-    
-    for(j=0;j < chans;j++){
-        pinbuf = inbuf+j;
-        for(i=0;i < buflen;i++){
-            out[j][i] = *pinbuf;
-            pinbuf += chans;
-        }
-    }
-}
-
-
-/* insize is m/c frame count */
-void mc_interl(double** in,double* out,long insize,long chans)
-{
-    long i,j;
-    double* poutbuf;
-
-    for(j = 0;j < chans;j++){
-        poutbuf = out+j;
-        for(i=0;i < insize;i++){
-            *poutbuf = in[j][i];
-            poutbuf += chans;
-        }
-    }
-}
-
-/* OR:  apply scalefac to impulse responses */
-void complexmult(double *frame,const double *impulse,long length)
-{
-	double re,im;
-	
-	int i,j;
-	
-
-	for(i=0,j = 1;i < length;i+=2,j+=2){
-		re = frame[i] * impulse[i] - frame[j] * impulse[j];
-		im = frame[i] * impulse[j] + frame[j]* impulse[i];
-		frame[i] = re;
-		frame[j] = im;
-	}
-}
-
-#ifdef unix
-int stricmp(const char *a, const char *b)
-{
-	while(*a != '\0' && *b != '\0') {
-		int ca = islower(*a) ? toupper(*a) : *a;
-		int cb = islower(*b) ? toupper(*b) : *b;
-        
-		if(ca < cb)
-			return -1;
-		if(ca > cb)
-			return 1;
-        
-		a++;
-		b++;
-	}
-	if(*a == '\0' && *b == '\0')
-		return 0;
-	if(*a != '\0')
-		return 1;
-	return -1;
-}
-
-int
-strnicmp(const char *a, const char *b, const int length)
-{
-	int len = length;
-    
-	while(*a != '\0' && *b != '\0') {
-		int ca = islower(*a) ? toupper(*a) : *a;
-		int cb = islower(*b) ? toupper(*b) : *b;
-        
-		if(len-- < 1)
-			return 0;
-        
-		if(ca < cb)
-			return -1;
-		if(ca > cb)
-			return 1;
-        
-		a++;
-		b++;
-	}
-	if(*a == '\0' && *b == '\0')
-		return 0;
-	if(*a != '\0')
-		return 1;
-	return -1;
-}
-#endif
-
-
-
-
-
-

dev/externals/fastconv/fconv.cpp

@@ -1,752 +0,0 @@
-/*
- * Copyright (c) 1983-2013 Richard Dobson and Composers Desktop Project Ltd
- * http://people.bath.ac.uk/masrwd
- * http://www.composersdesktop.com
- * This file is part of the CDP System.
- * The CDP System is free software; you can redistribute it
- * and/or modify it under the terms of the GNU Lesser General Public 
- * License as published by the Free Software Foundation; either 
- * version 2.1 of the License, or (at your option) any later version. 
- *
- * The CDP System is distributed in the hope that it will be useful, 
- * but WITHOUT ANY WARRANTY; without even the implied warranty of 
- * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  
- * See the GNU Lesser General Public License for more details.
- * You should have received a copy of the GNU Lesser General Public
- * License along with the CDP System; if not, write to the Free Software
- * Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
- *
- */
-
-/* fastconv.cpp: */
-/*  self-contained prog for fast convolution (reverb, FIR filtering, bformat etc) 
- *  
-*/
-
-/* um, there is nothing C++ish about this code apart from use of new, delete! */
-
-/*TODO: control wet/dry mix with something... */
-/* auto-rescale arbitrary impulse responses? */
-/* Feb 2013: rebuilt with updated portsf */
-/* August 2013 epanded usage message re text file. */
-extern "C"
-{
-
-#include <portsf.h>
-
-}
-#include <stdio.h>
-#include <stdlib.h>
-#include <math.h>
-#include <string.h>
-#include <time.h>
-#include <sys/types.h>
-#include <sys/timeb.h>
-
-
-#ifdef _DEBUG
-#include <assert.h>
-#endif
-
-#ifdef unix
-#include <ctype.h>
-int stricmp(const char *a, const char *b);
-int strnicmp(const char *a, const char *b, const int length);
-#endif
-
-#ifndef WAVE_FORMAT_IEEE_FLOAT
-#define WAVE_FORMAT_IEEE_FLOAT (0x0003)
-#endif
-
-/* convert from mono text impulse file */
-int genimpframe1(double *insbuf, double*** outbuf, int npoints, double scalefac);
-/* convert from multichan soundfile */
-int genimpframe2(int ifd,double*** outframe, double* rms,int imchans,double scalefac);
-void mc_split(double* inbuf,double** out,int insize,int chans);
-void mc_interl(double** in,double* out,int insize,int chans);
-void complexmult(double *frame,const double *impulse,int length);
-
-
-extern "C"{
-void fft_(double *, double *,int,int,int,int);
-void fftmx(double *,double *,int,int,int,int,int,int *,double *,double *,double *,double *,int *,int[]);
-void reals_(double *,double *,int,int);
-}
-
-
-#define DEFAULT_BUFLEN (32768)
-
-enum {ARG_NAME,ARG_INFILE,ARG_IMPFILE,ARG_OUTFILE,ARG_NARGS};
-
-void usage(const char *progname)
-{
-
-    printf("\n\n%s v1.2 RWD,CDP July 2010,2013",progname);
-	printf("\nMulti-channel FFT-based convolver");
-
-	printf("\nUsage: \n    %s [-aX][-f] infile impulsefile outfile [dry]\n"		               
-           "   -aX        : scale output amplitude by X\n"
-           "   -f         : write output as floats (no clipping)\n"
-           "  infile      : input soundfile to be processed.\n"
-           "  impulsefile : m/c soundfile or mono text file containing impulse response,\n"
-           "                  e.g. reverb or FIR filter.\n"
-           "                Text file name must have extension .txt.\n"
-		   "                  File must contain 1 column of floating point values,\n"
-		   "                  typically in the range -1.0 to 1.0.\n"
-           "              Supported channel combinations:\n"
-           "               (a) mono infile, N-channel impulsefile;\n"
-           "               (b) channels are the same;\n"
-           "               (c) mono impulsefile, N-channel infile.\n"
-           "  [dry]       :  set dry/wet mix (e.g. for reverb)\n"
-           "                 Range: 0.0 - 1.0,  default = 0.0\n"
-           "                 (uses sin/cos law for constant power mix)\n"
-           "Note: some recorded reverb impulses effectively include the direct signal.\n"
-           "In such cases  <dry>  need not be used\n"
-           "Where impulsefile is filter response (FIR), optimum length is power-of-two - 1.\n",progname
-		   );		   
-}
-
-
-double
-timer()
-{
-	struct timeb now;
-	double secs, ticks;	
-	ftime(&now);
-	ticks = (double)now.millitm/(1000.0);
-	secs = (double) now.time;
-
-	return secs + ticks;
-}
-
-
-void			
-stopwatch(int flag) 
-{
-	static double start;		    
-	int mins=0,hours=0;
-	double end, secs=0;
-
-	if(flag)
-		start = timer();	
-	else 
-	{
-		end    = timer(); 
-		secs   = end - start;
-		mins   = (int)(secs/60.0);
-		secs  -= mins*60.0; 
-		hours  = mins/60;
-		mins  -= hours*60;
-
-		fprintf(stderr,"\nElapsed time: ");
-		if(hours > 0)
-			fprintf(stderr,"%d h ", hours);
-		if(mins > 0)
-			fprintf(stderr,"%d m ", mins);
-		fprintf(stderr,"%2.3lf s\n\n",secs);
-	}
-}
-
-/* how do we want this to work?
-
-  (1)  multi-chan impulse:  EITHER: mono infile  OR infile with same chan count
-  (2)  mono impulse:  expanded to infile chancount
-     Therefore: need to open both infiles before deciding action
-*/
-#define PI2 (1.570796327)
-
-int main(int argc,char **argv)
-{
-
-	int fftlen = 1,imlen = 0,chans = 0,srate;
-	double scalefac = 1.0f;
-    double Ninv = 1.0;
-	int insndfile = -1,inimpfile = -1,outsndfile = -1; 
-    int nameoffset = 0;
-	PSF_PROPS inprops,outprops, impulseprops;
-    psf_format format;
-    PSF_CHPEAK  *fpeaks = NULL;
-	MYLONG peaktime;
-	int i,jj,minheader = 0,do_timer = 1;
-    int writefloats = 0;
-    int framesneeded;
-	double oneovrsr;
-    double *inmonobuf = 0;
-	double *insbuf=0,*outsbuf = 0;
-    double **inbuf = 0, **outbuf = 0;    
-	double **imbuf = 0;
-	double **overlapbuf = 0;
-    double rms = 0.0;
-    double dry = 0.0;
-    double wet = 1.0;
-#ifdef FFTTEST
-    double *anal = 0;
-#endif
-    	
-    if(argv[0][0]=='.' && argv[0][1]=='/'){
-        nameoffset  = 2;
-    }
-	
-    if(argc==2){
-        if(strcmp(argv[1],"--version")==0){
-            printf("1.2.0.\n");
-            return 0;
-        }
-    }
-
-	if(psf_init()){
-		puts("unable to start portsf\n");
-		return 1;
-	}
-
-
-	if(argc < ARG_NARGS){		 
-		usage(argv[0]+nameoffset);
-		return(1);
-	}
-
-    
-
-    while(argc > 1 && argv[1][0]=='-'){				
-		switch(argv[1][1]){
-        case 'a':
-            scalefac =  atof(&(argv[1][2]));
-            if(scalefac <=0.0){
-                printf("Error: scalefac must be positive!\n");
-                return 1;
-            }
-            break;        
-        case 'f':
-            writefloats = 1;
-            break;
-        default:
-            break;
-        }
-        argv++;
-        argc--;
-    }		
-	/* 2 legal possibilities: infile and outfile, or -I used with infile only */
-	if(argc< ARG_NARGS){
-		fprintf(stderr,"\nInsufficient arguments.");
-		usage(argv[0]+nameoffset);
-		return(1);
-	}
-    if(argc==ARG_NARGS+1){
-        double dryarg;
-        dryarg = atof(argv[ARG_NARGS]);
-        if(dryarg < 0.0 || dryarg > 1.0){
-            printf("dry value out of range.\n");
-            return 0;
-        }
-        if(dryarg==1.0){
-            printf("Warning: dry=1 copies input!\n");
-            wet = 0.0;
-        }
-        else{
-            dry = sin(PI2 * dryarg);
-            wet = cos(PI2 * dryarg);
-        }
-    }
-
-		
-
-/* TODO:  where -F[file] is combined with -A, it signifies create analysis file
-           compatible with impulse file (e.g 50% overlap, etc) */
-
-	
-	/* open infile, check props */	
-		
-	if((insndfile = psf_sndOpen(argv[ARG_INFILE],&inprops, 0)) < 0){
-		fprintf(stderr,"\nUnable to open input soundfile %s",argv[1]);
-        delete []imbuf;
-		return 1;
-	}
-    srate = inprops.srate;
-	if(srate <=0){
-		fprintf(stderr,"\nBad srate found: corrupted file?\n");
-        delete []imbuf;
-		return 1;
-	}
-	chans = inprops.chans;
-    framesneeded = psf_sndSize(insndfile);
-    if(framesneeded <= 0){
-        printf("Error in input file - no data!\n");
-        psf_sndClose(insndfile);
-        return 1;
-    }
-    /* open impulse file */
-    /* check for soundfile */
-      
-    format = psf_getFormatExt(argv[ARG_IMPFILE]);
-    if(format==PSF_FMT_UNKNOWN){  /* must be a text file */
-        FILE *fp = 0;
-        char tmp[80];
-        char* dot;
-        int size = 0,got = 0;
-        int read = 0;
-
-        dot = strrchr(argv[ARG_IMPFILE],'.');
-        if(dot==NULL || stricmp(dot,".txt") != 0){
-            fprintf(stderr,"Error: impulse text file must have .txt extension.\n");
-            return 1;
-        }
-        fp = fopen(argv[ARG_IMPFILE],"r");
-        if(fp==0){
-            printf("Cannot open impulse text file %s\n.",argv[ARG_IMPFILE]);
-            return 1;
-        }
-        /* count lines! */
-        while(fgets(tmp,80,fp) != NULL)
-            size++;
-        if(size==0){
-            printf("Impulse textfile %s has no data!.\n",argv[ARG_IMPFILE]);
-            return 1;
-        }
-        rewind(fp);
-            
-        inmonobuf = new double[size+1];
-        for(i=0;i < size;i++)  {
-            read = fscanf(fp,"%lf",&inmonobuf[i]);
-            if(read==0){
-                i--;
-                continue;
-            }
-            if(read==EOF)
-                break;
-            got++;
-        }
-        imlen = got;
-        impulseprops.chans = 1;
-        fclose(fp);
-    }
-    else{
-        if((inimpfile = psf_sndOpen(argv[ARG_IMPFILE],&impulseprops, 0))< 0){
-            fprintf(stderr,"\nUnable to open impulse file %s",argv[ARG_IMPFILE]);       
-            return 0;
-        }
-        //printf("impulse file sr = %d\n",impulseprops.srate);
-        if(srate != impulseprops.srate){
-            printf("Error: files must have same sample rate");
-            delete []imbuf;
-            return 1;
-        }
-        int  len = psf_sndSize(inimpfile);
-        if(len <= 0){
-            printf("Error in impulse soundfile - no data!\n");
-            return 1;
-        }
-        if(impulseprops.chans > 1){
-            if(! (chans == 1 || chans == impulseprops.chans)){
-                fprintf(stderr,"\nChannel mismatch.\n    Infile must be mono or have same channels as irfile.\n");
-                return(1);
-            }
-            chans = impulseprops.chans;
-        }    
-    }
-
-    imbuf = new double*[chans];
-
-    /* if impulse file is mono, we will need to copy data to the other buffers */
-    // allocate  impulseprops.chans buffers ; may need more
-
-    if(inimpfile >=0){
-	    if((imlen = genimpframe2(inimpfile,&imbuf,&rms,impulseprops.chans, scalefac)) == 0){
-	    	fprintf(stderr,"Error reading impulse file\n");
-	    	return(1);
-	    }
-    }
-    
-    else if(imlen){        
-        genimpframe1(inmonobuf,&imbuf,imlen,scalefac);       
-    }
-    printf("mean rms level = %.4lf (%.2lf dB)\n",rms, 20.0 * log10(rms));
-    
-
-    framesneeded +=  imlen;        /* can  close outfile, when this length reached */
-
-    while(fftlen < imlen*2)			/* convolution style - double-length */
-		fftlen <<= 1;
-    double norm = sqrt(2.0);       /* relative: rms of 0dBFS sine is 0.707 = 1/root2 */
-    // scale factor: most of this sheer guesswork!
-    Ninv = fftlen;
-    Ninv /= sqrt(imlen*2);
-    Ninv *= norm;
-    Ninv /= imlen;
-    
-    // take simple avg of  rms for adjustment factor.
-    // may not adequately represent some responses, e.g. with strong attack/earlies, soft reverb tail */
-    double rmsdif2 =  (-20.0 * log10(rms)) * 0.5;
-    double rmsadjust = pow(10, rmsdif2/20.0);
-#ifdef _DEBUG
-    printf("rescaling factor = %.6e\n",Ninv);   
-    printf("rmsadjust = %.4lf\n",rmsadjust);
-#endif
-   Ninv /= rmsadjust; 
-
-
-    /* copy buffers if required */
-    for(i = impulseprops.chans;i < chans;i++){
-        imbuf[i] = new double[fftlen+2];
-        memcpy(imbuf[i],imbuf[0],(fftlen+2)* sizeof(double));
-    }         
-    oneovrsr = 1.0 / (double) srate;
-    
-   
-	/* main i/o buffers */
-	
-	insbuf     = new double [fftlen  * chans];
-    outsbuf    = new double [fftlen  * chans];
-	inbuf      = new double*[chans];
-    outbuf     = new double*[chans];  /* overlap-add buffers */
-    overlapbuf = new double*[chans];
-    
-    for(i=0;i < chans;i++){
-        inbuf[i]  = new double[fftlen+2];   // main in-place buf 
-        outbuf[i] = new double[fftlen+2];        
-	    /* channel buffers */	
-        overlapbuf[i] = new double[fftlen/2];
-        memset(overlapbuf[i],0,sizeof(double)*(fftlen/2));        
-    }
-#ifdef FFTTEST
-    anal = new double[fftlen+2];
-#endif
-    
-    for(i=0;i < chans;i++){		    
-	    memset(inbuf[i], 0, sizeof(double) * (fftlen+2));
-        memset(outbuf[i],0,sizeof(double) * (fftlen+2));
-    }
-
-    
-	/* use generic init function */
-/*bool phasevocoder::init(int outsrate,int fftlen,int winlen,int decfac,float scalefac,
-						pvoc_scaletype stype,pvoc_windowtype wtype,pvocmode mode)
-*/
-	
-	/*create output sfile */
-	psf_stype smptype;
-	psf_format outformat;
-	/* will it be aiff, wav, etc? */
-	outformat = psf_getFormatExt(argv[ARG_OUTFILE]);
-	if(outformat==PSF_FMT_UNKNOWN){
-		fprintf(stderr,"\nOutfile name has unrecognized extension.");
-        delete []imbuf;
-		return(1);
-	}
-		
-		
-	smptype = inprops.samptype;
-    if(writefloats)
-        smptype= PSF_SAMP_IEEE_FLOAT;
-
-	/*the one gotcha: if output is floats and format is aiff, change to aifc */
-	if(smptype==PSF_SAMP_IEEE_FLOAT && outformat==PSF_AIFF){
-		fprintf(stderr,"Warning: AIFF output written as AIFC for float samples\n");
-		outformat = PSF_AIFC;
-	}
-
-	outprops          = inprops;
-	outprops.chans    = chans;
-	outprops.srate    = srate;
-	outprops.format   = outformat;
-	outprops.samptype = smptype;
-	outprops.chformat = STDWAVE;
-    /* if irfile is BFormat, need to set outfile fmt likewise */
-    if(impulseprops.chformat==MC_BFMT)
-        outprops.chformat = MC_BFMT;
-    /* I suppose people will want automatic decoding too...grrr */
-
-    fpeaks = new PSF_CHPEAK[chans];
-	if(fpeaks==NULL){
-		puts("no memory for fpeak data buffer\n");
-		return 1;
-	}
-
-	/*last arg: no clipping of f/p samples: we use PEAK chunk */
-	if((outsndfile = psf_sndCreate(argv[ARG_OUTFILE],&outprops,0,minheader,PSF_CREATE_RDWR)) <0 ){
-		fprintf(stderr,"\nUnable to open outfile %s",argv[ARG_OUTFILE]);
-        delete []imbuf;
-		return(1);
-	}
-				
-
-	
-	printf("\n");
-	
-	
-	stopwatch(1);
-
-	int written,thisblock,framesread;
-    int frameswritten = 0;
-	double intime= 0.0;
-				
-	while((framesread = psf_sndReadDoubleFrames(insndfile,insbuf,imlen)) > 0){        
-		written = thisblock =  0;
-
-		for(i = framesread * inprops.chans; i< fftlen * inprops.chans; i++)
-			insbuf[i] = 0.0f;
-        framesread = imlen;
-		memset(inbuf[0],0,(fftlen+2) * sizeof(double));
-        if(chans == inprops.chans)  {
-            /* must clean buffers! */
-		    for(i=0;i < chans;i++)
-                memset(inbuf[i],0,(fftlen+2) * sizeof(double));
-            mc_split(insbuf,inbuf,imlen * chans, chans);
-
-        }
-        else{
-            for(i=0;i < chans;i++) {
-                memset(inbuf[i],0,(fftlen+2) * sizeof(double));
-                memcpy(inbuf[i],insbuf,imlen * sizeof(double));
-                memset(outbuf[i],0,sizeof(double) * (fftlen+2));
-
-            }
-        }
-        if(impulseprops.chans==1){
-			
-            for(jj = 0; jj < chans;jj++){
-#ifdef FFTTEST
-                int zz;
-                memcpy(anal,inbuf[jj],(fftlen+2) * sizeof(double));
-                fft_(anal,anal+1,1,fftlen/2,1,-2);
-	            reals_(anal,anal+1,fftlen/2,-2);
-                for(zz=0;zz < fftlen+2;zz++)
-                    anal[zz] *= 0.001;
-#endif
-                //rfftwnd_one_real_to_complex(forward_plan[jj],inbuf[jj],NULL);	
-				double *danal = inbuf[jj];
-				fft_(danal,danal+1,1,fftlen/2,1,-2);
-				reals_(danal,danal+1,fftlen/2,-2);
-                complexmult(inbuf[jj],imbuf[0],fftlen+2);                
-			    //rfftwnd_one_complex_to_real(inverse_plan[jj],(fftw_complex * )inbuf[jj],NULL);
-				reals_(danal,danal+1,fftlen/2,2);
-				fft_(danal,danal+1,1,fftlen/2,1,2);
-            }           
-        }
-        else{
-            for(jj = 0; jj < chans;jj++){
-				//rfftwnd_one_real_to_complex(forward_plan[jj],inbuf[jj],NULL);                   				
-                double *danal = inbuf[jj];
-				fft_(danal,danal+1,1,fftlen/2,1,-2);
-				reals_(danal,danal+1,fftlen/2,-2);
-				complexmult(inbuf[jj],imbuf[jj],fftlen+2);                
-				//rfftwnd_one_complex_to_real(inverse_plan[jj],(fftw_complex * )inbuf[jj],NULL); 
-				reals_(danal,danal+1,fftlen/2,2);
-				fft_(danal,danal+1,1,fftlen/2,1,2);
-            }            
-        }
-        
-        /* overlap-add  for each channel */
-        /* TODO: verify  use of imlen here - should it be fftlen/2 -1 ? */
-        for(jj=0;jj < chans;jj++){
-            for(i=0;i < imlen;i++) {
-                outbuf[jj][i] = inbuf[jj][i] + overlapbuf[jj][i];
-                overlapbuf[jj][i] = inbuf[jj][i+imlen];
-            }
-        }
-		mc_interl(outbuf,outsbuf,imlen,chans);
-
-        if(inprops.chans == chans){
-            for(i=0;i < framesread; i++) {
-                for(jj=0;jj < chans;jj++){
-                    int  outindex = i*chans + jj;            
-                    outsbuf[outindex] *= Ninv;
-                    outsbuf[outindex] *= wet;
-                    outsbuf[outindex] += dry * insbuf[outindex];
-                }
-            }
-        }
-        /* elso mono input */
-        else {
-            for(i=0;i < framesread; i++) {
-                for(jj=0;jj < chans;jj++){
-                    int  outindex = i*chans + jj;
-                    double inval = dry *  insbuf[i];
-                    outsbuf[outindex] *= Ninv;
-                    outsbuf[outindex] *= wet;
-                    outsbuf[outindex] += inval;
-                }
-            }
-        }
-
-        if((written = psf_sndWriteDoubleFrames(outsndfile,outsbuf,framesread)) != framesread){
-		    fprintf(stderr,"\nerror writing to outfile");
-		    return(1);		               
-        }
-        frameswritten += written;
-
-		if(do_timer){
-			intime += (double)framesread * oneovrsr;
-			printf("Input time: %.2lf\r",intime);
-		}    
-    }
-    /* complete tail */
-
-    if(frameswritten < framesneeded){
-        // TODO: imlen? see above
-        mc_interl(overlapbuf,outsbuf,imlen,chans);
-        int towrite = framesneeded - frameswritten; 
-        for(i=0;i < towrite * chans; i++) {
-            outsbuf[i] *= Ninv;
-            outsbuf[i] *= wet;
-        }
-        if((written = psf_sndWriteDoubleFrames(outsndfile,outsbuf,towrite)) != towrite){
-	        fprintf(stderr,"\nerror writing to outfile");
-	        return(1);
-	    }
-    }
-
-	stopwatch(0);
-    if(psf_sndReadPeaks( outsndfile,fpeaks,&peaktime)){
-        double peakmax = 0.0;
-		printf("PEAK values:\n");
-		for(i=0; i < chans; i++)	{
-            peakmax = fpeaks[i].val > peakmax ? fpeaks[i].val : peakmax;
-			if(fpeaks[i].val < 0.0001f)
-				printf("CH %d:\t%e (%.2lf dB)\tat frame %10u :\t%.4lf secs\n",i+1,
-				fpeaks[i].val,20.0*log10(fpeaks[i].val),fpeaks[i].pos,(double) (fpeaks[i].pos) / (double)srate);
-			else
-				printf("CH %d:\t%.4lf (%.2lf dB)\tat frame %10u :\t%.4lf secs\n",i+1,
-				fpeaks[i].val,20.0 * log10(fpeaks[i].val),fpeaks[i].pos,(double) (fpeaks[i].pos) / (double)srate);		
-		}
-        if(peakmax > 1.0)
-            printf("Overflows!  Rescale by %.10lf\n",1.0 / peakmax);
-	}
-
-	if(insndfile >=0)
-		psf_sndClose(insndfile);
-	if(outsndfile >=0)
-		psf_sndClose(outsndfile);
-    if(inimpfile >=0)
-	    psf_sndClose(inimpfile);
-	psf_finish();
-    delete [] fpeaks;
-	if(insbuf)
-		delete [] insbuf;
-	if(outsbuf)
-		delete [] outsbuf;
-    
-
-    for(i=0;i < chans;i++){
-        if(inbuf){       
-		    delete [] inbuf[i];
-        }           
-        if(outbuf){
-		    delete [] outbuf[i];
-        }       	
-        if(imbuf){        
-	        delete [] imbuf[i];
-        }
-        if(overlapbuf)
-            delete [] overlapbuf[i];
-    }
-    delete [] outbuf;
-    delete [] inbuf;
-    delete [] imbuf;
-    delete [] overlapbuf;
-	return 0;
-}
-
-
-// insize is raw samplecount,buflen is insize/chans
-void mc_split(double* inbuf,double** out,int insize,int chans)
-{
-    int i,j,buflen = insize/chans;
-    double* pinbuf;
-
-    
-    for(j=0;j < chans;j++){
-        pinbuf = inbuf+j;
-        for(i=0;i < buflen;i++){
-            out[j][i] = *pinbuf;
-            pinbuf += chans;
-        }
-    }
-}
-
-
-/* insize is m/c frame count */
-void mc_interl(double** in,double* out,int insize,int chans)
-{
-    int i,j;
-    double* poutbuf;
-
-    for(j = 0;j < chans;j++){
-        poutbuf = out+j;
-        for(i=0;i < insize;i++){
-            *poutbuf = in[j][i];
-            poutbuf += chans;
-        }
-    }
-}
-
-/* OR:  apply scalefac to impulse responses */
-void complexmult(double *frame,const double *impulse,int length)
-{
-	double re,im;
-	
-	int i,j;
-	
-
-	for(i=0,j = 1;i < length;i+=2,j+=2){
-		re = frame[i] * impulse[i] - frame[j] * impulse[j];
-		im = frame[i] * impulse[j] + frame[j]* impulse[i];
-		frame[i] = re;
-		frame[j] = im;
-	}
-}
-
-#ifdef unix
-int stricmp(const char *a, const char *b)
-{
-	while(*a != '\0' && *b != '\0') {
-		int ca = islower(*a) ? toupper(*a) : *a;
-		int cb = islower(*b) ? toupper(*b) : *b;
-        
-		if(ca < cb)
-			return -1;
-		if(ca > cb)
-			return 1;
-        
-		a++;
-		b++;
-	}
-	if(*a == '\0' && *b == '\0')
-		return 0;
-	if(*a != '\0')
-		return 1;
-	return -1;
-}
-
-int
-strnicmp(const char *a, const char *b, const int length)
-{
-	int len = length;
-    
-	while(*a != '\0' && *b != '\0') {
-		int ca = islower(*a) ? toupper(*a) : *a;
-		int cb = islower(*b) ? toupper(*b) : *b;
-        
-		if(len-- < 1)
-			return 0;
-        
-		if(ca < cb)
-			return -1;
-		if(ca > cb)
-			return 1;
-        
-		a++;
-		b++;
-	}
-	if(*a == '\0' && *b == '\0')
-		return 0;
-	if(*a != '\0')
-		return 1;
-	return -1;
-}
-#endif
-
-
-
-
-
-

dev/externals/fastconv/genrespframe2.cpp

@@ -1,235 +0,0 @@
-/*
- * Copyright (c) 1983-2013 Richard Dobson and Composers Desktop Project Ltd
- * http://people.bath.ac.uk/masrwd
- * http://www.composersdesktop.com
- * This file is part of the CDP System.
- * The CDP System is free software; you can redistribute it
- * and/or modify it under the terms of the GNU Lesser General Public 
- * License as published by the Free Software Foundation; either 
- * version 2.1 of the License, or (at your option) any later version. 
- *
- * The CDP System is distributed in the hope that it will be useful, 
- * but WITHOUT ANY WARRANTY; without even the implied warranty of 
- * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  
- * See the GNU Lesser General Public License for more details.
- * You should have received a copy of the GNU Lesser General Public
- * License along with the CDP System; if not, write to the Free Software
- * Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
- *
- */
- 
-/* genrespframe2.cpp */
-/* generate m/c pvoc frames containing impulse response */
-extern "C"
-{
-#include <portsf.h>
-}
-#include <stdio.h>
-#include <stdlib.h>
-#include <math.h>
-#include <string.h>
-
-
-
-#ifdef _DEBUG
-#include <assert.h>
-#endif
-
-/* deinterleave input buffer of insize samps to array of chans buffers */
-void mc_split(double* inbuf,double** out,int insize,int chans);
-extern "C"{
-	void fft_(double *, double *,int,int,int,int);
-	void fftmx(double *,double *,int,int,int,int,int,int *,double *,double *,double *,double *,int *,int[]);
-	void reals_(double *,double *,int,int);
-}
-
-/*  read impulse data from soundfile */
-/* return 0 for error, or size of impulse */
-/*we assume pvsys initialized by caller */
-int genimpframe2(int ifd, double*** outbufs, double* rms,int imchans, double scalefac)
-{
-	int i,j;	
-	double *insbuf = 0;
-    double **inbuf = *outbufs;
-	int fftsize = 1;
-    int inframes;
-    double*  rmsfac = 0;
-    double* ampsum = 0;
-    double  maxrmsfac = 0.0;
-#ifdef _DEBUG
-    double *re,*im,*mag;
-#endif
-	inframes = psf_sndSize(ifd);  // m/c frames
-	if(inframes <= 0)
-        return 0;
-	while(fftsize < inframes*2)			/* convolution style - double-length */
-		fftsize <<= 1;
-	printf("impulse length = %d frames, fftsize = %d\n",inframes,fftsize);
-    insbuf = new double[inframes * imchans];
-#ifdef _DEBUG
-    re = new double[fftsize/2+1];
-    im = new double[fftsize/2+1];
-    mag = new double[fftsize/2+1];
-#endif
-    for(i=0;i < imchans;i++){	
-	    inbuf[i] = new double[fftsize+2];
-	    if(inbuf[i]==NULL){
-		    puts("no memory for file buffer!\n");                    
-		    return 0;
-	    }
-	    memset(inbuf[i],0,(fftsize+2)* sizeof(double));
-    }
-
-
-	if(inframes != psf_sndReadDoubleFrames(ifd,insbuf,inframes)){
-		fprintf(stderr,"Error reading impulse data\n");
-		
-		delete [] insbuf;
-        // and do other cleanup!
-        
-		return 0;
-	}
-    rmsfac = new double[imchans];
-    ampsum = new double[imchans];
-    for(i=0;i< imchans;i++) {
-        rmsfac[i] = 0.0;
-        ampsum[i] = 0.0;
-    }
-    /* do user scaling first */
-    for(i = 0;i < inframes * imchans;i++)
-        insbuf[i] *= scalefac;
-    /* now try to adjust rms  */
-
-    for(i = 0;i < inframes;i++) {
-        for(j=0;j < imchans;j++){
-            double val = insbuf[i*imchans + j];
-            ampsum[j] += fabs(val);
-            rmsfac[j] += val*val; 
-        }
-    }
-    for(j=0;j < imchans;j++){
-        //    rmsfac = sqrt(rmsfac);
-        rmsfac[j] /= inframes;
-        rmsfac[j] = sqrt(rmsfac[j]);
-        ampsum[j] /= inframes;
-        if(rmsfac[j] > maxrmsfac)
-            maxrmsfac = rmsfac[j];
-#ifdef _DEBUG
-        if(ampsum[j] > maxampsum)
-            maxampsum = ampsum[j];
-#endif
-    }
-    /* do the rescaling! */
-          
-#ifdef _DEBUG        
-        printf("ampsum = %.4f\n",maxampsum);
-#endif
-    
-    // now deinterleave to each inbuf
-    mc_split(insbuf,inbuf,inframes * imchans,imchans);
-    for(i=0;i < imchans;i++){
-		double *anal = inbuf[i];
-	    //rfftwnd_one_real_to_complex(forward_plan[i],inbuf[i],NULL);
-		fft_(anal,anal+1,1,fftsize/2,1,-2);
-		reals_(anal,anal+1,fftsize/2,-2);
-    }
-#ifdef _DEBUG
-    /* in order to look at it all */
-    double  magmax = 0.0;
-    double magsum = 0.0;
-    for(i=0;i < fftsize/2+1;i++){
-        double thisre, thisim;
-        thisre =inbuf[0][i*2];        
-        thisim = inbuf[0][i*2+1]; 
-        re[i] = thisre;
-        im[i] = thisim;
-        mag[i] = sqrt(thisre*thisre + thisim*thisim); 
-        magsum += (mag[i] * mag[i]);
-        if(mag[i] > magmax)
-            magmax = mag[i];
-    }
-    magsum = sqrt(magsum);
-    printf("maxamp of FFT = %.4f\n",magmax);
-    printf("mean level of FFT = %.4f\n",magsum / (fftsize/2+1));
-#endif	
-    
-    delete [] rmsfac;
-    delete [] ampsum;
-
-#ifdef _DEBUG
-    delete [] re;
-    delete [] im;
-    delete [] mag;
-#endif
-    *rms = maxrmsfac;
-	return inframes;
-}
-/* convert from input double buffer (read from mono text impulse file) */
-int genimpframe1(double *insbuf, double*** outbuf, int npoints, double scalefac)
-{
-	int i;	
-    double **inbuf = *outbuf;
-	int fftsize = 1;
-    int insamps;
-#ifdef _DEBUG
-    double *re,*im,*mag;
-#endif
-	insamps = npoints;  // m/c frames
-	if(insamps <= 0)
-        return 0;
-	while(fftsize < insamps*2)			/* convolution style - double-length */
-		fftsize <<= 1;
-	printf("infile size = %d,impulse framesize = %d\n",insamps,fftsize);
-    
-#ifdef _DEBUG
-    re = new double[fftsize/2+1];
-    im = new double[fftsize/2+1];
-    mag = new double[fftsize/2+1];
-#endif
-    	
-	inbuf[0] = new double[fftsize+2];
-	if(inbuf[0]==NULL){
-	    puts("no memory for file buffer!\n");                    
-	    return 0;
-	}
-	memset(inbuf[0],0,(fftsize+2)* sizeof(double)); 
-#ifdef _DEBUG
-    double ampsum = 0.0;
-#endif
-    for(i = 0;i < insamps;i++) {
-#ifdef _DEBUG
-        ampsum += fabs(insbuf[i]);
-#endif        
-        insbuf[i] *= scalefac;
-    }
-#ifdef _DEBUG
-    printf("amplitude sum of impulse file = %f\n",ampsum);
-#endif
-    memcpy(inbuf[0],insbuf,npoints* sizeof(double));
-	//rfftwnd_one_real_to_complex(forward_plan,inbuf[0],NULL);
-	double *anal = inbuf[0];
-	fft_(anal,anal+1,1,fftsize/2,1,-2);
-	reals_(anal,anal+1,fftsize/2,-2);
-    
-#ifdef _DEBUG
-    /* in order to look at it all */
-    for(i=0;i < fftsize/2+1;i++){
-        double thisre, thisim;
-        thisre =inbuf[0][i*2];        
-        thisim = inbuf[0][i*2+1]; 
-        re[i] = thisre;
-        im[i] = thisim;
-        mag[i] = sqrt(thisre*thisre + thisim*thisim);
-    }
-#endif	
-    
-    
-#ifdef _DEBUG
-    delete [] re;
-    delete [] im;
-    delete [] mag;
-#endif
-   
-
-	return insamps;
-}

dev/externals/fastconv/mxfftd.c

@@ -1,883 +0,0 @@
-/*  
-    mxfftd.c: double precision version of mxfft.c
-	 
-	This file is part of the CDP System.
-
-    The CDP System is free software; you can redistribute it
-    and/or modify it under the terms of the GNU Lesser General Public
-    License as published by the Free Software Foundation; either
-    version 2.1 of the License, or (at your option) any later version.
-
-    The CDP System is distributed in the hope that it will be useful,
-    but WITHOUT ANY WARRANTY; without even the implied warranty of
-    MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the
-    GNU Lesser General Public License for more details.
-
-    You should have received a copy of the GNU Lesser General Public
-    License along with the CDP System; if not, write to the Free Software
-    Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA
-    02111-1307 USA
-
-*/
-
-/* This program converted from the FORTRAN routines by Singleton in
- * Section 1.4 of  "Programs for Digital Signal Processing", IEEE Press, 1979.
- *  Conversion by Trevor Wishart and Keith Henderson, York Univ.
- */
-
-
-
-
-#include <stdio.h>
-#include <stdlib.h>
-#include <math.h>
-
-
-void fft_(double *, double *, int, int, int, int);
-void fftmx(double *, double *, int, int, int, int, int,
-      int*, double *, double *, double *, double *, int *, int[]);
-void reals_(double *, double *, int, int);
-
-/*
- *-----------------------------------------------------------------------
- * subroutine:  fft
- * multivariate complex fourier transform, computed in place
- * using mixed-radix fast fourier transform algorithm.
- *-----------------------------------------------------------------------
- *
- *      this is the call from C:
- *              fft_(anal,banal,&one,&N2,&one,&mtwo);
- *      CHANGED TO:-
- *              fft_(anal,banal,one,N2,one,mtwo);
- */
-
-void
-fft_(double *a, double *b, int nseg, int n, int nspn, int isn)
-  /*    *a,       pointer to array 'anal'  */
-  /*    *b;       pointer to array 'banal' */
-{
-    int nfac[16];               /*  These are one bigger than needed   */
-                                /*  because wish to use Fortran array  */
-                                /* index which runs 1 to n, not 0 to n */
-
-    int         m = 0,
-                nf,
-                k,
-                kt,
-                ntot,
-                j,
-                jj,
-                maxf, maxp=-1;
-
-/* work space pointers */
-    double       *at, *ck, *bt, *sk;
-    int *np;
-
-
-/* reduce the pointers to input arrays - by doing this, FFT uses FORTRAN
-   indexing but retains compatibility with C arrays */
-    a--;        b--;
-
-/*
- * determine the factors of n
- */
-    k=nf=abs(n);
-    if (nf==1)
-      return;
-
-    nspn=abs(nf*nspn);
-    ntot=abs(nspn*nseg);
-
-    if (isn*ntot == 0) {
-      printf("\nerror - zero in fft parameters %d %d %d %d",
-              nseg, n, nspn, isn);
-      return;
-    }
-    for (m=0; !(k%16); nfac[++m]=4,k/=16);
-    for (j=3,jj=9; jj<=k; j+=2,jj=j*j)
-      for (; !(k%jj); nfac[++m]=j,k/=jj);
-
-    if (k<=4) {
-      kt = m;
-      nfac[m+1] = k;
-      if (k != 1)
-        m++;
-    }
-    else {
-      if (k%4==0) {
-        nfac[++m]=2;
-        k/=4;
-      }
-
-      kt = m;
-      maxp = (kt+kt+2 > k-1 ? kt+kt+2 : k-1);
-      for (j=2; j<=k; j=1+((j+1)/2)*2)
-        if (k%j==0) {
-          nfac[++m]=j;
-          k/=j;
-        }
-    }
-    if (m <= kt+1)
-      maxp = m + kt + 1;
-    if (m+kt > 15) {
-      printf("\nerror - fft parameter n has more than 15 factors : %d", n);
-      return;
-    }
-    if (kt!=0) {
-      j = kt;
-      while (j)
-        nfac[++m]=nfac[j--];
-    }
-    maxf = nfac[m-kt];
-    if (kt > 0 && maxf <nfac[kt])
-      maxf = nfac[kt];
-
-/*  allocate workspace - assume no errors! */
-    at = (double *) calloc(maxf,sizeof(double));
-    ck = (double *) calloc(maxf,sizeof(double));
-    bt = (double *) calloc(maxf,sizeof(double));
-    sk = (double *) calloc(maxf,sizeof(double));
-    np = (int *) calloc(maxp,sizeof(int));
-
-/* decrement pointers to allow FORTRAN type usage in fftmx */
-    at--;       bt--;   ck--;   sk--;   np--;
-
-/* call fft driver */
-
-    fftmx(a,b,ntot,nf,nspn,isn,m,&kt,at,ck,bt,sk,np,nfac);
-
-/* restore pointers before releasing */
-    at++;       bt++;   ck++;   sk++;   np++;
-
-/* release working storage before returning - assume no problems */
-    free(at);
-    free(sk);
-    free(bt);
-    free(ck);
-    free(np);
-    return;
-}
-
-/*
- *-----------------------------------------------------------------------
- * subroutine:  fftmx
- * called by subroutine 'fft' to compute mixed-radix fourier transform
- *-----------------------------------------------------------------------
- */
-void
-fftmx(double *a, double *b, int ntot, int n, int nspan, int isn, int m,
-      int *kt, double *at, double *ck, double *bt, double *sk, int *np, int nfac[])
-{
-    int i,inc,
-      j,jc,jf, jj,
-      k, k1, k2, k3, k4,
-      kk,klim,ks,kspan, kspnn,
-      lim,
-      maxf,mm,
-      nn,nt;
-    double  aa, aj, ajm, ajp, ak, akm, akp,
-      bb, bj, bjm, bjp, bk, bkm, bkp,
-      c1, c2, c3, c72, cd,
-      dr,
-      rad,
-      sd, s1, s2, s3, s72, s120;
-
-    double      xx;     /****** ADDED APRIL 1991 *********/
-    inc=abs(isn);
-    nt = inc*ntot;
-    ks = inc*nspan;
-/******************* REPLACED MARCH 29: ***********************
-                                        rad = atan((double)1.0);
-**************************************************************/
-    rad = 0.785398163397448278900;
-/******************* REPLACED MARCH 29: ***********************
-                                        s72 = rad/0.625;
-                                        c72 = cos(s72);
-                                        s72 = sin(s72);
-**************************************************************/
-    c72 = 0.309016994374947451270;
-    s72 = 0.951056516295153531190;
-/******************* REPLACED MARCH 29: ***********************
-                                        s120 = sqrt((double)0.75);
-**************************************************************/
-    s120 = 0.866025403784438707600;
-
-/* scale by 1/n for isn > 0 ( reverse transform ) */
-
-    if (isn < 0) {
-      s72 = -s72;
-      s120 = -s120;
-      rad = -rad;}
-    else {
-      ak = 1.0/(double)n;
-      for (j=1; j<=nt;j += inc) {
-        a[j] *= (double)ak;
-        b[j] *= (double)ak;
-      }
-    }
-    kspan = ks;
-    nn = nt - inc;
-    jc = ks/n;
-
-/* sin, cos values are re-initialised each lim steps  */
-
-    lim = 32;
-    klim = lim * jc;
-    i = 0;
-    jf = 0;
-    maxf = m - (*kt);
-    maxf = nfac[maxf];
-    if ((*kt) > 0 && maxf < nfac[*kt])
-      maxf = nfac[*kt];
-
-/*
- * compute fourier transform
- */
-
- lbl40:
-    dr = (8.0 * (double)jc)/((double)kspan);
-/*************************** APRIL 1991 POW & POW2 not WORKING.. REPLACE *******
-                    cd = 2.0 * (pow2 ( sin((double)0.5 * dr * rad)) );
-*******************************************************************************/
-    xx =  sin((double)0.5 * dr * rad);
-    cd = 2.0 * xx * xx;
-    sd = sin(dr * rad);
-    kk = 1;
-    if (nfac[++i]!=2) goto lbl110;
-/*
- * transform for factor of 2 (including rotation factor)
- */
-    kspan /= 2;
-    k1 = kspan + 2;
-    do {
-      do {
-        k2 = kk + kspan;
-        ak = a[k2];
-        bk = b[k2];
-        a[k2] = (a[kk]) - (double)ak;
-        b[k2] = (b[kk]) - (double)bk;
-        a[kk] = (a[kk]) + (double)ak;
-        b[kk] = (b[kk]) + (double)bk;
-        kk = k2 + kspan;
-      } while (kk <= nn);
-      kk -= nn;
-    } while (kk <= jc);
-    if (kk > kspan) goto lbl350;
- lbl60:
-    c1 = 1.0 - cd;
-    s1 = sd;
-    mm = (k1/2 < klim ? k1/2 :klim);
-    goto lbl80;
- lbl70:
-    ak = c1 - ((cd*c1)+(sd*s1));
-    s1 = ((sd*c1)-(cd*s1)) + s1;
-    c1 = ak;
-lbl80:
-    do {
-      do {
-        k2 = kk + kspan;
-        ak = a[kk] - a[k2];
-        bk = b[kk] - b[k2];
-        a[kk] = a[kk] + a[k2];
-        b[kk] = b[kk] + b[k2];
-        a[k2] = (double)((c1 * ak) - (s1 * bk));
-        b[k2] = (double)((s1 * ak) + (c1 * bk));
-        kk = k2 + kspan;
-      } while (kk < nt);
-      k2 = kk - nt;
-      c1 = -c1;
-      kk = k1 - k2;
-    } while (kk > k2);
-    kk += jc;
-    if (kk <= mm) goto lbl70;
-    if (kk < k2)  goto lbl90;
-    k1 += (inc + inc);
-    kk = ((k1-kspan)/2) + jc;
-    if (kk <= (jc+jc)) goto lbl60;
-    goto lbl40;
- lbl90:
-    s1 = ((double)((kk-1)/jc)) * dr * rad;
-    c1 = cos(s1);
-    s1 = sin(s1);
-    mm = (k1/2 < mm+klim ? k1/2 : mm+klim);
-    goto lbl80;
-/*
- * transform for factor of 3 (optional code)
- */
-
-
- lbl100:
-    k1 = kk + kspan;
-    k2 = k1 + kspan;
-    ak = a[kk];
-    bk = b[kk];
-    aj = a[k1] + a[k2];
-    bj = b[k1] + b[k2];
-    a[kk] = (double)(ak + aj);
-    b[kk] = (double)(bk + bj);
-    ak += (-0.5 * aj);
-    bk += (-0.5 * bj);
-    aj = (a[k1] - a[k2]) * s120;
-    bj = (b[k1] - b[k2]) * s120;
-    a[k1] = (double)(ak - bj);
-    b[k1] = (double)(bk + aj);
-    a[k2] = (double)(ak + bj);
-    b[k2] = (double)(bk - aj);
-    kk = k2 + kspan;
-    if (kk < nn)     goto lbl100;
-    kk -= nn;
-    if (kk <= kspan) goto lbl100;
-    goto lbl290;
-
-/*
- * transform for factor of 4
- */
-
- lbl110:
-    if (nfac[i] != 4) goto lbl230;
-    kspnn = kspan;
-    kspan = kspan/4;
- lbl120:
-    c1 = 1.0;
-    s1 = 0;
-    mm = (kspan < klim ? kspan : klim);
-    goto lbl150;
- lbl130:
-    c2 = c1 - ((cd*c1)+(sd*s1));
-    s1 = ((sd*c1)-(cd*s1)) + s1;
-/*
- * the following three statements compensate for truncation
- * error.  if rounded arithmetic is used, substitute
- * c1=c2
- *
- * c1 = (0.5/(pow2(c2)+pow2(s1))) + 0.5;
- * s1 = c1*s1;
- * c1 = c1*c2;
- */
-    c1 = c2;
- lbl140:
-    c2 = (c1 * c1) - (s1 * s1);
-    s2 = c1 * s1 * 2.0;
-    c3 = (c2 * c1) - (s2 * s1);
-    s3 = (c2 * s1) + (s2 * c1);
- lbl150:
-    k1 = kk + kspan;
-    k2 = k1 + kspan;
-    k3 = k2 + kspan;
-    akp = a[kk] + a[k2];
-    akm = a[kk] - a[k2];
-    ajp = a[k1] + a[k3];
-    ajm = a[k1] - a[k3];
-    a[kk] = (double)(akp + ajp);
-    ajp = akp - ajp;
-    bkp = b[kk] + b[k2];
-    bkm = b[kk] - b[k2];
-    bjp = b[k1] + b[k3];
-    bjm = b[k1] - b[k3];
-    b[kk] = (double)(bkp + bjp);
-    bjp = bkp - bjp;
-    if (isn < 0) goto lbl180;
-    akp = akm - bjm;
-    akm = akm + bjm;
-    bkp = bkm + ajm;
-    bkm = bkm - ajm;
-    if (s1 == 0.0) goto lbl190;
- lbl160:
-    a[k1] = (double)((akp*c1) - (bkp*s1));
-    b[k1] = (double)((akp*s1) + (bkp*c1));
-    a[k2] = (double)((ajp*c2) - (bjp*s2));
-    b[k2] = (double)((ajp*s2) + (bjp*c2));
-    a[k3] = (double)((akm*c3) - (bkm*s3));
-    b[k3] = (double)((akm*s3) + (bkm*c3));
-    kk = k3 + kspan;
-    if (kk <= nt)   goto lbl150;
- lbl170:
-    kk -= (nt - jc);
-    if (kk <= mm)   goto lbl130;
-    if (kk < kspan) goto lbl200;
-    kk -= (kspan - inc);
-    if (kk <= jc)   goto lbl120;
-    if (kspan==jc)  goto lbl350;
-    goto lbl40;
-lbl180:
-    akp = akm + bjm;
-    akm = akm - bjm;
-    bkp = bkm - ajm;
-    bkm = bkm + ajm;
-    if (s1 != 0.0)  goto lbl160;
- lbl190:
-    a[k1] = (double)akp;
-    b[k1] = (double)bkp;
-    a[k2] = (double)ajp;
-    b[k2] = (double)bjp;
-    a[k3] = (double)akm;
-    b[k3] = (double)bkm;
-    kk = k3 + kspan;
-    if (kk <= nt) goto lbl150;
-    goto lbl170;
- lbl200:
-    s1 = ((double)((kk-1)/jc)) * dr * rad;
-    c1 = cos(s1);
-    s1 = sin(s1);
-    mm = (kspan < mm+klim ? kspan : mm+klim);
-    goto lbl140;
-
-/*
- * transform for factor of 5 (optional code)
- */
-
- lbl210:
-    c2 = (c72*c72) - (s72*s72);
-    s2 = 2.0 * c72 * s72;
- lbl220:
-    k1 = kk + kspan;
-    k2 = k1 + kspan;
-    k3 = k2 + kspan;
-    k4 = k3 + kspan;
-    akp = a[k1] + a[k4];
-    akm = a[k1] - a[k4];
-    bkp = b[k1] + b[k4];
-    bkm = b[k1] - b[k4];
-    ajp = a[k2] + a[k3];
-    ajm = a[k2] - a[k3];
-    bjp = b[k2] + b[k3];
-    bjm = b[k2] - b[k3];
-    aa = a[kk];
-    bb = b[kk];
-    a[kk] = (double)(aa + akp + ajp);
-    b[kk] = (double)(bb + bkp + bjp);
-    ak = (akp*c72) + (ajp*c2) + aa;
-    bk = (bkp*c72) + (bjp*c2) + bb;
-    aj = (akm*s72) + (ajm*s2);
-    bj = (bkm*s72) + (bjm*s2);
-    a[k1] = (double)(ak - bj);
-    a[k4] = (double)(ak + bj);
-    b[k1] = (double)(bk + aj);
-    b[k4] = (double)(bk - aj);
-    ak = (akp*c2) + (ajp*c72) + aa;
-    bk = (bkp*c2) + (bjp*c72) + bb;
-    aj = (akm*s2) - (ajm*s72);
-    bj = (bkm*s2) - (bjm*s72);
-    a[k2] = (double)(ak - bj);
-    a[k3] = (double)(ak + bj);
-    b[k2] = (double)(bk + aj);
-    b[k3] = (double)(bk - aj);
-    kk = k4 + kspan;
-    if (kk < nn)     goto lbl220;
-    kk -= nn;
-    if (kk <= kspan) goto lbl220;
-    goto lbl290;
-
-/*
- * transform for odd factors
- */
-
- lbl230:
-    k = nfac[i];
-    kspnn = kspan;
-    kspan /= k;
-    if (k==3)   goto lbl100;
-    if (k==5)   goto lbl210;
-    if (k==jf)  goto lbl250;
-    jf = k;
-    s1 = rad/(((double)(k))/8.0);
-    c1 = cos(s1);
-    s1 = sin(s1);
-    ck[jf] = 1.0;
-    sk[jf] = 0.0;
-    for (j=1; j<k ; j++) {
-      ck[j] = (double)((ck[k])*c1 + (sk[k])*s1);
-      sk[j] = (double)((ck[k])*s1 - (sk[k])*c1);
-      k--;
-      ck[k] = ck[j];
-      sk[k] = -(sk[j]);
-    }
- lbl250:
-    k1 = kk;
-    k2 = kk + kspnn;
-    aa = a[kk];
-    bb = b[kk];
-    ak = aa;
-    bk = bb;
-    j = 1;
-    k1 += kspan;
-    do {
-      k2 -= kspan;
-      j++;
-      at[j] = a[k1] + a[k2];
-      ak = at[j] + ak;
-      bt[j] = b[k1] + b[k2];
-      bk = bt[j] + bk;
-      j++;
-      at[j] = a[k1] - a[k2];
-      bt[j] = b[k1] - b[k2];
-      k1 += kspan;
-    } while (k1 < k2);
-    a[kk] = (double)ak;
-    b[kk] = (double)bk;
-    k1 = kk;
-    k2 = kk + kspnn;
-    j = 1;
- lbl270:
-    k1 += kspan;
-    k2 -= kspan;
-    jj = j;
-    ak = aa;
-    bk = bb;
-    aj = 0.0;
-    bj = 0.0;
-    k = 1;
-    do {
-      k++;
-      ak = (at[k] * ck[jj]) + ak;
-      bk = (bt[k] * ck[jj]) + bk;
-      k++;
-      aj = (at[k] * sk[jj]) + aj;
-      bj = (bt[k] * sk[jj]) + bj;
-      jj += j;
-      if (jj > jf)
-        jj -= jf;
-    } while (k < jf);
-    k = jf - j;
-    a[k1] = (double)(ak - bj);
-    b[k1] = (double)(bk + aj);
-    a[k2] = (double)(ak + bj);
-    b[k2] = (double)(bk - aj);
-    j++;
-    if (j < k)     goto lbl270;
-    kk += kspnn;
-    if (kk <= nn)  goto lbl250;
-    kk -= nn;
-    if (kk<=kspan) goto lbl250;
-
-/*
- * multiply by rotation factor (except for factors of 2 and 4)
- */
-
- lbl290:
-    if (i==m) goto lbl350;
-    kk = jc + 1;
- lbl300:
-    c2 = 1.0 - cd;
-    s1 = sd;
-    mm = (kspan < klim ? kspan : klim);
-    goto lbl320;
- lbl310:
-    c2 = c1 - ((cd*c1) + (sd*s1));
-    s1 = s1 + ((sd*c1) - (cd*s1));
- lbl320:
-    c1 = c2;
-    s2 = s1;
-    kk += kspan;
- lbl330:
-    ak = a[kk];
-    a[kk] = (double)((c2*ak) - (s2 * b[kk]));
-    b[kk] = (double)((s2*ak) + (c2 * b[kk]));
-    kk += kspnn;
-    if (kk <= nt) goto lbl330;
-    ak = s1*s2;
-    s2 = (s1*c2) + (c1*s2);
-    c2 = (c1*c2) - ak;
-    kk -= (nt - kspan);
-    if (kk <= kspnn) goto lbl330;
-    kk -= (kspnn - jc);
-    if (kk <= mm)   goto lbl310;
-    if (kk < kspan) goto lbl340;
-    kk -= (kspan - jc - inc);
-    if (kk <= (jc+jc)) goto lbl300;
-    goto lbl40;
- lbl340:
-    s1 = ((double)((kk-1)/jc)) * dr * rad;
-    c2 = cos(s1);
-    s1 = sin(s1);
-    mm = (kspan < mm+klim ?  kspan :mm+klim);
-    goto lbl320;
-
-/*
- * permute the results to normal order---done in two stages
- * permutation for square factors of n
- */
-
- lbl350:
-    np[1] = ks;
-    if (!(*kt)) goto lbl440;
-    k = *kt + *kt + 1;
-    if (m < k)
-      k--;
-    np[k+1] = jc;
-    for (j=1; j < k; j++,k--) {
-      np[j+1] = np[j] / nfac[j];
-      np[k] = np[k+1] * nfac[j];
-    }
-    k3 = np[k+1];
-    kspan = np[2];
-    kk = jc + 1;
-    k2 = kspan + 1;
-    j = 1;
-    if (n != ntot) goto lbl400;
-/*
- * permutation for single-variate transform (optional code)
- */
- lbl370:
-    do {
-      ak = a[kk];
-      a[kk] = a[k2];
-      a[k2] = (double)ak;
-      bk = b[kk];
-      b[kk] = b[k2];
-      b[k2] = (double)bk;
-      kk += inc;
-      k2 += kspan;
-    } while (k2 < ks);
-lbl380:
-    do {
-      k2 -= np[j++];
-      k2 += np[j+1];
-    } while (k2 > np[j]);
-    j = 1;
- lbl390:
-    if (kk < k2) {
-      goto lbl370;
-    }
-    kk += inc;
-    k2 += kspan;
-    if (k2 < ks) goto lbl390;
-    if (kk < ks) goto lbl380;
-    jc = k3;
-    goto lbl440;
-/*
- * permutation for multivariate transform
- */
- lbl400:
-    do {
-      do {
-        k = kk + jc;
-        do {
-          ak = a[kk];
-          a[kk] = a[k2];
-          a[k2] = (double)ak;
-          bk = b[kk];
-          b[kk] = b[k2];
-          b[k2] = (double)bk;
-          kk += inc;
-          k2 += inc;
-        } while (kk < k);
-        kk += (ks - jc);
-        k2 += (ks - jc);
-      } while (kk < nt);
-      k2 -= (nt - kspan);
-      kk -= (nt - jc);
-    } while (k2 < ks);
- lbl420:
-    do {
-      k2 -= np[j++];
-      k2 += np[j+1];
-    } while (k2 > np[j]);
-    j = 1;
- lbl430:
-    if (kk < k2)         goto lbl400;
-    kk += jc;
-    k2 += kspan;
-    if (k2 < ks)      goto lbl430;
-    if (kk < ks)      goto lbl420;
-    jc = k3;
- lbl440:
-    if ((2*(*kt))+1 >= m)
-      return;
-
-    kspnn = *(np + *(kt) + 1);
-    j = m - *kt;
-    nfac[j+1] = 1;
- lbl450:
-    nfac[j] = nfac[j] * nfac[j+1];
-    j--;
-    if (j != *kt) goto lbl450;
-    *kt = *(kt) + 1;
-    nn = nfac[*kt] - 1;
-    jj = 0;
-    j = 0;
-    goto lbl480;
- lbl460:
-    jj -= k2;
-    k2 = kk;
-    kk = nfac[++k];
- lbl470:
-    jj += kk;
-    if (jj >= k2) goto lbl460;
-    np[j] = jj;
- lbl480:
-    k2 = nfac[*kt];
-    k = *kt + 1;
-    kk = nfac[k];
-    j++;
-    if (j <= nn) goto lbl470;
-/* Determine permutation cycles of length greater than 1 */
-    j = 0;
-    goto lbl500;
- lbl490:
-    k = kk;
-    kk = np[k];
-    np[k] = -kk;
-    if (kk != j) goto lbl490;
-    k3 = kk;
- lbl500:
-    kk = np[++j];
-    if (kk < 0)  goto lbl500;
-    if (kk != j) goto lbl490;
-    np[j] = -j;
-    if (j != nn) goto lbl500;
-    maxf *= inc;
-    /* Perform reordering following permutation cycles */
-    goto lbl570;
- lbl510:
-    j--;
-    if (np[j] < 0) goto lbl510;
-    jj = jc;
- lbl520:
-    kspan = jj;
-    if (jj > maxf)
-      kspan = maxf;
-    jj -= kspan;
-    k = np[j];
-    kk = (jc*k) + i + jj;
-    k1 = kk + kspan;
-    k2 = 0;
- lbl530:
-    k2++;
-    at[k2] = a[k1];
-    bt[k2] = b[k1];
-    k1 -= inc;
-    if (k1 != kk) goto lbl530;
- lbl540:
-    k1 = kk + kspan;
-    k2 = k1 - (jc * (k + np[k]));
-    k = -(np[k]);
- lbl550:
-    a[k1] = a[k2];
-    b[k1] = b[k2];
-    k1 -= inc;
-    k2 -= inc;
-    if (k1 != kk) goto lbl550;
-    kk = k2;
-    if (k != j)   goto lbl540;
-    k1 = kk + kspan;
-    k2 = 0;
- lbl560:
-    k2++;
-    a[k1] = at[k2];
-    b[k1] = bt[k2];
-    k1 -= inc;
-    if (k1 != kk) goto lbl560;
-    if (jj)       goto lbl520;
-    if (j  != 1)  goto lbl510;
-lbl570:
-    j = k3 + 1;
-    nt -= kspnn;
-    i = nt - inc + 1;
-    if (nt >= 0)  goto lbl510;
-    return;
-}
-
-
-/*
- *-----------------------------------------------------------------------
- * subroutine:
-  reals
- * used with 'fft' to compute fourier transform or inverse for real data
- *-----------------------------------------------------------------------
- *      this is the call from C:
-
- *              reals_(anal,banal,N2,mtwo);
- *      which has been changed from CARL call
- *              reals_(anal,banal,&N2,&mtwo);
- */
-
-void
-reals_(double *a, double *b, int n, int isn)
-
-  /*    *a,       a refers to an array of floats 'anal'   */
-  /*    *b;       b refers to an array of floats 'banal'  */
-/* See IEEE book for a long comment here on usage */
-
-{
-    int inc,
-      j,
-      k,
-      lim,
-      mm,ml,
-      nf,nk,nh;
-
-    double      aa,ab,
-      ba,bb,
-      cd,cn,
-      dr,
-      em,
-      rad,re,
-      sd,sn;
-    double      xx;     /******* ADDED APRIL 1991 ******/
-    /* adjust  input array pointers (called from C) */
-    a--;        b--;
-    inc=abs(isn);
-    nf=abs(n);
-    if (nf*isn==0) {
-      printf("\nerror - zero in reals parameters : %d : %d ",n,isn);
-      return;
-    }
-    nk = (nf*inc) + 2;
-    nh = nk/2;
-/*****************************
-        rad  = atan((double)1.0);
-******************************/
-    rad = 0.785398163397448278900;
-    dr = -4.0/(double)(nf);
-/********************************** POW2 REMOVED APRIL 1991 *****************
-                                cd = 2.0 * (pow2(sin((double)0.5 * dr * rad)));
-*****************************************************************************/
-    xx = sin((double)0.5 * dr * rad);
-    cd = 2.0 * xx * xx;
-    sd = sin(dr * rad);
-/*
- * sin,cos values are re-initialised each lim steps
- */
-    lim = 32;
-    mm = lim;
-    ml = 0;
-    sn = 0.0;
-    if (isn<0) {
-      cn = 1.0;
-      a[nk-1] = a[1];
-      b[nk-1] = b[1]; }
-    else {
-      cn = -1.0;
-      sd = -sd;
-    }
-    for (j=1;j<=nh;j+=inc)      {
-      k = nk - j;
-      aa = a[j] + a[k];
-      ab = a[j] - a[k];
-      ba = b[j] + b[k];
-      bb = b[j] - b[k];
-      re = (cn*ba) + (sn*ab);
-      em = (sn*ba) - (cn*ab);
-      b[k] = (double)((em-bb)*0.5);
-      b[j] = (double)((em+bb)*0.5);
-      a[k] = (double)((aa-re)*0.5);
-      a[j] = (double)((aa+re)*0.5);
-      ml++;
-      if (ml!=mm) {
-        aa = cn - ((cd*cn)+(sd*sn));
-        sn = ((sd*cn) - (cd*sn)) + sn;
-        cn = aa;}
-      else {
-        mm +=lim;
-        sn = ((double)ml) * dr * rad;
-        cn = cos(sn);
-        if (isn>0)
-          cn = -cn;
-        sn = sin(sn);
-      }
-    }
-    return;
-}
-
-

dev/externals/fastconv/readme.txt

@@ -1,110 +0,0 @@
-Preliminary documentation  for fastconv.
-
-
-FASTCONV:  multi-channel fast convolution (using FFTs)
-
-version: 1.0 2010
-
-usage message: 
-fastconv [-aX][-f] infile impulsefile outfile [dry]
-   -aX        : scale output amplitude by X
-   -f         : write output as floats (no clipping)
-  infile      : input soundfile to be processed.
-  impulsefile : soundfile or text file containing impulse response,
-                  e.g. reverb or FIR filter.
-                (text file name must have extension .txt)
-              Supported channel combinations:
-               (a) mono infile, N-channel impulsefile;
-               (b) channels are the same;
-               (c) mono impulsefile, N-channel infile.
-  [dry]       :  set dry/wet mix (e.g. for reverb)
-                 Range: 0.0 - 1.0,  default = 0.0
-                 (uses sin/cos law for constant power mix)
-Note: some recorded reverb impulses effectively include the direct signal.
-In such cases  <dry>  need not be used
-Where impulsefile is a filter response (FIR), optimum length is power-of-two - 1.
-
-
-The primary application of fastconv is convolution reverberation using a 
-sampled impulse response of a building or other responsive space. The term "fast"  
-refers to the use of the Fast Fourier transform (FFT) to perfume the convolution. 
-The program can also be used more experimentally, as the impulse response input can 
-be any mono or multi-channel file (see details of available channel combinations below); 
-a file can also be convolved with itself. 
-The program uses double-precision processing throughout, and files of considerable 
-length can be processed cleanly.
-Note however that the FFT of the impulse response is stored in main memory, 
-so very large files may raise memory demands to critical levels.
-
-
-More on channel options:
-
-  Impulse soundfile can be multi-channel.
-  The .amb format is supported, also Logic Pro SDIR files for reading (change the file extension to .aifc).
-
-  When the infile is multi-channel, impulse file must be mono, or the same number of channels.
-        Where the impulse file is mono, data is duplicated for all input channels.
-		Typical usage: linear-phase filtering. Output should be 100% "wet".
-	The optimum length for a filter impulse response is power-of-two-1, 
-	e.g. 127,255, 511 etc. Most filter creation tools will output a file of this size.
-
-	(More tools to support linear-phase filtering are in preparation!).
-
-
-  When infile is mono, impulse soundfile can be multi-channel, outfile has channel count of
-      impulse response.
-    Typical usage: reverb convolution for spatialization, B-Format convolution
-	It will be usual to supply a non-zero value for "dry", e.g. 0.5. Note however that some recorded 
-       or synthetic impulse responses  may already include a "direct" component. In such cases, 
-	a "dry" value may not be needed.
-
-  The program employs an rms-based gain scaling algorithm which attempts to ensure all outputs are
- approximately at the same level as the input. In normal use (e.g. a naturally decaying reverb impulse response),
- the -a flag should not be needed.
-When the -f flag is used, output is forced to floats, with no clipping. The program reports 
-the output level together with a suggested corrective gain factor. This will be of particular relevance to more
-experimental procedures, such as convolving a soundfile with itself or with some other arbitrary source.
-
-
-Note for advanced users - use for FIR linear-phase filtering.
-
-Convolution implements a filtering process - equivalent to multiplication of the spectra of the two inputs. 
-The most common filter in audio work is recursive - it recycles output samples back into the input. The output 
-continues in principle for ever, hence the term "infinite impulse response" (IIR). The advantage of this technique 
-(as employed for example in CDP's "filter' package) is that even a low-order filter (i.e. using a small 
-number of delayed inputs and outputs) can be very powerful in its effects. A disadvantage in some applications 
-can be that an IIR filter changes the phase of components in the input (frequency components are 
-delayed by different amounts). This means among the things that the waveform shape of the input is not 
-preserved. The timbre of the sound (even in regions  not directly boosted 
-or attenuated by the filter) will therefore be changed. In common parlance, such a filter "colours" the signal.
-
-The alternative is a linear phase filter, which preserves all phase relationships in the signal. 
-All frequency components are delayed equally. To achieve this the impulse response must have a symmetrical 
-shape (see illustration). The response decays identically either side of the central peak.  
-
-This requires that there be no recirculated outputs reinjected into the filter. 
-Such a filter has a Finite Impulse Response (FIR). The impulse response data now comprises literally the 
-response of the filter to a single input sample (impulse). An impulse response of 31 samples means 
-that the filter generates and adds together 31 delayed copies of the input, to create each output sample. 
-While IIR filter coefficients may involve only two delayed samples ("a "second-order" filter), FIR responses 
-need to employ many more samples to achieve similar effects. It would not be unusual to use a 511-th order FIR filter. 
-This also means that the overall delay ("latency") of a FIR filter is much longer than that of an IIR filter. 
-
-A FIR filter cannot resonate as an IIR filter can. By computing only delayed inputs, It is unconditionally 
-"stable" - whereas a badly designed IIR filter can "blow up" with output values rapidly exceeding the sample limit.
-
-Fastconv supports the use of FIR coefficient files either in the form of either a short soundfile, or a 
-plain text file containing (in a single column)  the list of coefficients as floating-point numbers within 
-the "normalised" range -1.0 to 1.0. For orthodox filtering purposes a mono soundfile should be used, to process 
-all channels identically. FIR coefficient text files are generated by many engineering-oriented filter design 
-applications.  User may be tempted to  write response files by hand; this can be done, but the results will be 
-virtually impossible to predict or control. 
-
-For maximum efficiency, such files should ideally have a size that is a power-of-two less one: e.g. 255, 511, 1023, etc.
-
-For more information about FIR filters, see :
-
-http://www.labbookpages.co.uk/audio/firWindowing.html
-
-
-August 23 2010

dev/externals/include/portsf.h

@@ -1,258 +0,0 @@
-/* Copyright (c) 2009, 2014 Richard Dobson
-
-Permission is hereby granted, free of charge, to any person
-obtaining a copy of this software and associated documentation
-files (the "Software"), to deal in the Software without
-restriction, including without limitation the rights to use,
-copy, modify, merge, publish, distribute, sublicense, and/or sell
-copies of the Software, and to permit persons to whom the
-Software is furnished to do so, subject to the following
-conditions:
-
-The above copyright notice and this permission notice shall be
-included in all copies or substantial portions of the Software.
-
-THE SOFTWARE IS PROVIDED "AS IS", WITHOUT WARRANTY OF ANY KIND,
-EXPRESS OR IMPLIED, INCLUDING BUT NOT LIMITED TO THE WARRANTIES
-OF MERCHANTABILITY, FITNESS FOR A PARTICULAR PURPOSE AND
-NONINFRINGEMENT. IN NO EVENT SHALL THE AUTHORS OR COPYRIGHT
-HOLDERS BE LIABLE FOR ANY CLAIM, DAMAGES OR OTHER LIABILITY,
-WHETHER IN AN ACTION OF CONTRACT, TORT OR OTHERWISE, ARISING
-FROM, OUT OF OR IN CONNECTION WITH THE SOFTWARE OR THE USE OR
-OTHER DEALINGS IN THE SOFTWARE.
-*/
-
-/* POST_BOOK!*/
-/* RWD Oct 2009: added MC_CUBE to listy of supported speaker layouts */
-/* corrected 7.1 speaker value (hex) */
-/* Aug 2012 corrected SPKRS_MONO value */
-/* Nov 2013: added SPKRS_6_1 and MC_SURR_6_1 to list */
-
-#ifndef __RIFFWAV_H_INCLUDED
-#define __RIFFWAV_H_INCLUDED
-
-/* Revision 16th September 2003: added TPDF dither support */
-/* revision Dec 2005: support new B-Format extensions */
-/* RWD Nov 1 2006: extended with double sread/write for research purposes! */
-/* July 2009: attempt portability with 64bit platforms */
-/* unable to test Win64 yet - but the symbol to check is simply _WIN64  */
-#ifdef __GNUC__ 
-# ifdef __LP64__
-#   define CPLONG64
-# endif
-#endif
-/* failing that, manually define CPLONG64 in makefile or project setting */
-#ifdef CPLONG64
-// all this effort is for readPeaks() and ctime only!
-# define MYLONG int
-#else
-# define MYLONG long
-#endif
-
-#ifdef __cplusplus
-extern "C" {	
-#endif	   
-/* compatible with <windows.h> */
-#ifndef DWORD
-typedef unsigned MYLONG DWORD;
-typedef unsigned short WORD;
-#endif
-/* NB: AIFF spec always illustrates chunksize as (signed) long; 
-   even though nFrames is always unsigned long!
-   So we make everything DWORD here.
- */
-
-/* the file sample formats we could support */
-
-typedef enum {
-	PSF_SAMP_UNKNOWN	=	0,
-	PSF_SAMP_8,				   /* not yet supported! */
-	PSF_SAMP_16,
-	PSF_SAMP_24,
-	PSF_SAMP_32,
-	PSF_SAMP_IEEE_FLOAT
-} psf_stype;
-
-/* the file format */
-/* currently based only on file extension. 
-   To be friendly, we should parse the header to get the format.
-*/
-typedef enum {
-	PSF_FMT_UNKNOWN = 0,		/* e.g if no extension given. This could also signify 'raw' */
-	PSF_STDWAVE,
-	PSF_WAVE_EX,
-	PSF_AIFF,
-	PSF_AIFC
-} psf_format;
-
-/* provisional stab at error codes */
-enum {
-	PSF_E_NOERROR		= 0,
-	PSF_E_CANT_OPEN		= -1,
-	PSF_E_CANT_CLOSE	= -2,
-	PSF_E_CANT_WRITE	= -3,
-	PSF_E_CANT_READ		= -4,
-	PSF_E_NOT_WAVE		= -5,
-	PSF_E_BAD_TYPE		= -6,
-	PSF_E_BAD_FORMAT	= -7,
-	PSF_E_UNSUPPORTED	= -8,
-	PSF_E_NOMEM			= -9,
-	PSF_E_BADARG		= -10,
-	PSF_E_CANT_SEEK		= -11,
-	PSF_E_TOOMANYFILES  = -12,
-	PSF_E_FILE_READONLY = -13,
-	PSF_E_SEEK_BEYOND_EOF = -14
-};
-
-#define NUM_SPEAKER_POSITIONS (18)
-
-#define SPEAKER_FRONT_LEFT				0x1
-#define SPEAKER_FRONT_RIGHT				0x2
-#define SPEAKER_FRONT_CENTER			0x4
-#define SPEAKER_LOW_FREQUENCY			0x8
-#define SPEAKER_BACK_LEFT				0x10
-#define SPEAKER_BACK_RIGHT				0x20
-#define SPEAKER_FRONT_LEFT_OF_CENTER	0x40
-#define SPEAKER_FRONT_RIGHT_OF_CENTER	0x80
-#define SPEAKER_BACK_CENTER				0x100
-#define SPEAKER_SIDE_LEFT				0x200
-#define SPEAKER_SIDE_RIGHT				0x400
-#define SPEAKER_TOP_CENTER				0x800
-#define SPEAKER_TOP_FRONT_LEFT			0x1000
-#define SPEAKER_TOP_FRONT_CENTER		0x2000
-#define SPEAKER_TOP_FRONT_RIGHT			0x4000
-#define SPEAKER_TOP_BACK_LEFT			0x8000
-#define SPEAKER_TOP_BACK_CENTER			0x10000
-#define SPEAKER_TOP_BACK_RIGHT			0x20000
-#define SPEAKER_RESERVED      			0x80000000
-
-/* my extras*/
-#define SPKRS_UNASSIGNED	(0)
-#define SPKRS_MONO			(0x00000004)
-#define SPKRS_STEREO		(0x00000003)
-#define SPKRS_GENERIC_QUAD	(0x00000033)
-#define SPKRS_SURROUND_LCRS	(0x00000107)
-#define SPKRS_SURR_5_0      (0x00000037)
-#define SPKRS_DOLBY5_1		(0x0000003f)
-#define SPKRS_6_1			(0x0000013f)
-#define SPKRS_7_1           (0x000000ff)
-#define SPKRS_CUBE          (SPKRS_GENERIC_QUAD | SPEAKER_TOP_FRONT_LEFT | SPEAKER_TOP_FRONT_RIGHT | SPEAKER_TOP_BACK_LEFT | SPEAKER_TOP_BACK_RIGHT)
-#define SPKRS_ACCEPT_ALL	(0xffffffff)	 /*???? no use for a file*/
-
-
-/* support for the PEAK chunk */
-typedef struct psf_chpeak {	
-	float val;
-	DWORD pos;   /* OK for all WAVE and AIFF <= 4GB */
-} PSF_CHPEAK;
-
-/* second two are speculative at present! */
-typedef enum  {PSF_CREATE_RDWR,PSF_CREATE_TEMPORARY,PSF_CREATE_WRONLY} psf_create_mode;
-/* the speakerfeed format */
-/* MC_WAVE_EX is a possibly temporary one to cover abstruse infile formats! */
-typedef enum { STDWAVE,MC_STD,MC_MONO,MC_STEREO,MC_QUAD,MC_LCRS,MC_BFMT,MC_DOLBY_5_1,
-				MC_SURR_5_0,MC_SURR_6_1,MC_SURR_7_1,MC_CUBE,MC_WAVE_EX } psf_channelformat;
-
-/* read access support */
-/* for psf_sndSeek(); ~should~ map directly to fseek mode flags*/
-enum { PSF_SEEK_SET=0,PSF_SEEK_CUR,PSF_SEEK_END};
-
-enum {PSF_DITHER_OFF,PSF_DITHER_TPDF};
-
-/* main structure to define a soundile. Extended props must be asked for separately */
-typedef struct psf_props 
-{
-	int		srate;
-	int		chans;	
-	psf_stype	samptype;		
-	psf_format	format;			
-	psf_channelformat chformat;	
-	/* probably add more stuff...esp for full WAVE-EX support */
-} PSF_PROPS;
-
-
-/*************** PUBLIC FUNCS */
-
-/* init sfs system. return 0 for success */
-int psf_init(void);
-/* close sfs-system. Does auto cleanup of open files, etc. return 0 for success */
-int psf_finish(void);
-/* Create soundfile from props.
-   Supports clipping or non-clipping of floats to 0dbFS, 
-   set minimum header (or use PEAK)
-   returns Sf descriptor >= 0, or some PSF_E_*  on error.
-*/
-/* using WIN32, it is possible to share for reading, but not under ANSI */
-/* maybe we just abandon all that */ 
-/* TODO (?): enforce non-destructive creation */
-int psf_sndCreate(const char *path,const PSF_PROPS *props, int clip_floats,int minheader,int mode);
-
-
-/* open existing soundfile. Receive format info in props. Supports auto rescale from PEAK
-   data, with floats files. Only RDONLY access supported.
-   Return sf descriptor >= 0, or some PSF_E_+ on error.
-  */
-int psf_sndOpen(const char *path,PSF_PROPS *props, int rescale);
-/*snd close. Updates PEAK data if used. return 0 for success, or some PSF_E_+ on error.*/
-int psf_sndClose(int sfd);
-
-/* all data read/write is counted in multi-channel sample 'frames', NOT in raw samples.*/
-
-/* get size of file, in m/c frames. Return size, or PSF_E_BADARG on bad sfd  */
-int psf_sndSize(int sfd);
-
-/* write m/c frames of floats. this updates internal PEAK data automatically.
-   return num frames written, or  some PSF_E_* on error. 
-  */
-int psf_sndWriteFloatFrames(int sfd, const float *buf, DWORD nFrames);
-int psf_sndWriteDoubleFrames(int sfd, const double *buf, DWORD nFrames);
-/* as above, with 16bit data */
-int psf_sndWriteShortFrames(int sfd, const short *buf, DWORD nFrames);
-
-/* 	get current m/c frame position in file, or PSF_E_BADARG with bad sfd*/    
-int psf_sndTell(int sfd);
-
-/* m/c frame wrapper for stdio fseek. return 0 for success. Offset counted in m/c frames.
- seekmode must be one of PSF_SEEK_* options, to ensure correct mapping to SEEK_CUR, etc */
-int psf_sndSeek(int sfd,int offset,int seekmode);
-
-/* read m/c sample frames into floats buffer. return nFrames, or some PSF_E_*.
-    if file opened with rescale = 1, over-range floats data, as indicated by PEAK chunk,
-	is automatically scaled to 0dbFS.
-	NB we could add a facility to define 'headroom', as some level below 0dBFS.
- */
-int psf_sndReadFloatFrames(int sfd, float *buf, DWORD nFrames);
-int psf_sndReadDoubleFrames(int sfd, double *buf, DWORD nFrames);
-/* can add ReadShortFrames if we really need it! */
-
-int psf_sndReadPeaks(int sfd,PSF_CHPEAK peakdata[],MYLONG *peaktime);
-
-/* find the soundfile format from the filename extension */
-/* currently supported: .wav, .aif, .aiff,.aifc,.afc, .amb */
-psf_format psf_getFormatExt(const char *path);
-
-
-/* set/unset dither. 
-   Returns 0 on success, -1 if error (unrecognised type, or read-only)
-   no-op for input files
-*/
-
-int psf_sndSetDither(int sfd,unsigned int dtype);
-/* get current dither setting */
-int psf_sndGetDither(int sfd);
-
-/* RWD NEW Nov 2009 */
-int psf_speakermask(int sfd);
-
-/* Feb 2010 */
-int psf_getWarning(int sfd,const char** warnstring);
-
-/* Sept 2010  return 1 for true, 0 for false */
-    /* NB tests up to 32bit size for now! */
-int is_legalsize(unsigned long nFrames, const PSF_PROPS *props);
-    
-#ifdef __cplusplus
-}
-#endif
-
-#endif

dev/externals/mctools/CMakeLists.txt

@@ -1,81 +0,0 @@
-if(APPLE)
-  set(CMAKE_C_FLAGS "-O2 -Wall -mmacosx-version-min=10.8 -Dunix -fomit-frame-pointer -funroll-loops")
-  set(CMAKE_CXX_FLAGS "-O2 -Wall -mmacosx-version-min=10.8 -Dunix -fomit-frame-pointer -funroll-loops -std=c++11 -stdlib=libc++")
-#  SET(CMAKE_EXE_LINKER_FLAGS "-static")
-else()
-  if(MINGW)
-    set(CMAKE_C_FLAGS "-O3 -DWIN32 -D_WIN32 -fomit-frame-pointer  -funroll-loops")
-  else()
-    set(CMAKE_C_FLAGS "-O3 -Wall -Dlinux -Dunix -fomit-frame-pointer -funroll-loops")
-  endif()
-endif()
-
-link_directories(../lib)
-
-include_directories(../../../include ../include)
-
-add_executable(abfpan abfpan.cpp)
-target_link_libraries(abfpan portsf sfsys ${EXTRA_LIBRARIES})
-
-my_install(abfpan)
-
-add_executable(abfpan2 abfpan2.cpp)
-target_link_libraries(abfpan2 portsf sfsys ${EXTRA_LIBRARIES})
-
-my_install(abfpan2)
-
-add_executable(abfdcode abfdcode.cpp)
-target_link_libraries(abfdcode portsf sfsys ${EXTRA_LIBRARIES})
-
-my_install(abfdcode)
-
-add_executable(fmdcode fmdcode.c fmhfuncs.c)
-target_link_libraries(fmdcode portsf sfsys ${EXTRA_LIBRARIES})
-
-my_install(fmdcode)
-
-add_executable(channelx channel.c)
-target_link_libraries(channelx portsf sfsys m ${EXTRA_LIBRARIES})
-
-my_install(channelx)
-
-add_executable(chorder chorder.c)
-target_link_libraries(chorder portsf sfsys ${EXTRA_LIBRARIES})
-
-my_install(chorder)
-
-add_executable(chxformat chxformat.c)
-target_link_libraries(chxformat portsf sfsys ${EXTRA_LIBRARIES})
-
-my_install(chxformat)
-
-add_executable(copysfx copysf.c)
-target_link_libraries(copysfx portsf sfsys ${EXTRA_LIBRARIES})
-
-my_install(copysfx)
-
-add_executable(interlx interlx.c)
-target_link_libraries(interlx portsf sfsys ${EXTRA_LIBRARIES})
-
-my_install(interlx)
-
-add_executable(nmix nmix.c)
-target_link_libraries(nmix portsf sfsys ${EXTRA_LIBRARIES})
-
-my_install(nmix)
-
-add_executable(njoin njoin.c)
-target_link_libraries(njoin portsf sfsys ${EXTRA_LIBRARIES})
-
-my_install(njoin)
-
-add_executable(rmsinfo rmsinfo.cpp)
-target_link_libraries(rmsinfo portsf sfsys ${EXTRA_LIBRARIES})
-
-my_install(rmsinfo)
-
-add_executable(sfprops sfprops.c)
-target_link_libraries(sfprops portsf sfsys ${EXTRA_LIBRARIES})
-
-my_install(sfprops)
-

dev/externals/mctools/abfdcode.cpp

@@ -1,189 +0,0 @@
-/*
- * Copyright (c) 1983-2013 Richard Dobson and Composers Desktop Project Ltd
- * http://people.bath.ac.uk/masrwd
- * http://www.composersdesktop.com
- * This file is part of the CDP System.
- * The CDP System is free software; you can redistribute it
- * and/or modify it under the terms of the GNU Lesser General Public 
- * License as published by the Free Software Foundation; either 
- * version 2.1 of the License, or (at your option) any later version. 
- *
- * The CDP System is distributed in the hope that it will be useful, 
- * but WITHOUT ANY WARRANTY; without even the implied warranty of 
- * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  
- * See the GNU Lesser General Public License for more details.
- * You should have received a copy of the GNU Lesser General Public
- * License along with the CDP System; if not, write to the Free Software
- * Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
- *
- */
-
-//abfdcode.cpp  : write mono wave file into ambisonic B-format
-#include <stdio.h>
-#include <stdlib.h>
-#include <string.h>
-#include <math.h>
-#include "portsf.h"
-
-#ifdef unix
-/* in portsf.lib */
-extern  "C" {
-    int stricmp(const char *a, const char *b);
-}
-#endif
-
-typedef struct abf_samp {
-	float W;
-	float X;
-	float Y;
-	float Z;	
-} ABFSAMPLE;
-
-int main(int argc,char *argv[])
-{
-	int i,ifd, ofd;
-	long outchans = 4;
-	long outsize;
-	char *sfname;
-	bool write_wavex = false;
-	ABFSAMPLE abfsample;
-	float frame[4];
-	PSF_PROPS props;
-	PSF_CHPEAK peaks[4];
-	long inchans;
-    MYLONG peaktime;
-
-    /* CDP version number */
-    if(argc==2 && (stricmp(argv[1],"--version")==0)){
-        printf("1.2.1\n");
-        return 0;
-    }
-    
-	if(argc < 3){
-		printf("\nCDP MCTOOLS: ABFDCODE v 1.2.1: CDP 1999,2004,2005\n"
-				"Horizontal B Format Decoder\n"
-				"usage: abfdcode [-x] infile outfile\n"
-				"       -x    :   write WAVE_EX (Quad) outfile format\n"
-				"                (requires .wav extension)\n"
-				"NB: infile must have 3 or 4 channels\n"
-                "UPDATE 2009: This program is now replaced by FMDCODE, \n"
-                "and may be omitted in future updates to the Toolkit.\n"
-                "For this task use fmdcode with e.g. layout 4.\n"
-				);
-		return 1;
-	}
-	while(argv[1][0] =='-'){		
-		switch(argv[1][1]){
-		case('x'):
-			write_wavex = true;
-			break;
-		default:
-			fprintf(stderr,"abfdecode: error: illegal flag option %s\n",argv[1]);
-			return 1;
-		}
-		argc--; argv++;
-	}
-	if(argc < 3){
-		fprintf(stderr,"CDP MCTOOLS: ABFDCODE.EXE: CDP 1999,2005\nHorizontal B Format Decoder\nusage: abfdcode [-x] infile outfile\n");
-		return 1;
-	}
-	if(psf_init()){
-		fprintf(stderr,"abfdcode: startup failure.\n");
-		return 1;
-	}
-	
-	sfname = argv[2];
-	
-	ifd = psf_sndOpen(argv[1],&props,0);
-	if(ifd < 0){
-		fprintf(stderr,"unable toopen infile %s\n",argv[1]);
-		return 1;
-	}
-
-	inchans = props.chans;
-	if(!(inchans == 3 || inchans == 4)){
-		fprintf(stderr,"Sorry: infile is not first-order\n");
-		psf_sndClose(ifd);
-		return 1;
-	}
-
-	outsize = psf_sndSize(ifd);
-	if(outsize <= 0){
-		fprintf(stderr,"infile is empty!\n");
-		psf_sndClose(ifd);
-		return 1;
-	}
-	
-	props.chformat = STDWAVE;
-	props.chans = 4;
-    if(!is_legalsize(outsize,&props)){
-        fprintf(stderr,"error: outfile size exceeds capacity of format.\n");
-        return 1;
-    }
-    
-    
-	if(write_wavex){
-		props.chformat = MC_QUAD;
-		props.format = PSF_WAVE_EX;
-	}
-	//ofd = sndcreat_formatted(sfname,outsize * outchans,stype,outchans,srate,CDP_CREATE_NORMAL);
-	ofd = psf_sndCreate(sfname,&props,0,0,PSF_CREATE_RDWR);
-	if(ofd < 0){
-		fprintf(stderr,"can't create outfile %s.\n",sfname);
-		psf_sndClose(ifd);
-		return 1;
-	}
-	
-
-	int got, halfsec = props.srate / 2;
-	unsigned int framepos = 0;
-	printf("\ndoing b-format decoding:\n");
-	while((got = psf_sndReadFloatFrames(ifd,(float *) &abfsample,1))==1){
-		//this_samp	= 0.5 * p_iosc->tick(1000);
-		float aw = abfsample.W;
-		float ax = abfsample.X * 0.707f;
-		float ay = abfsample.Y * 0.707f;
-		frame[0] = 0.3333f * (aw + ax + ay);
-		frame[1] = 0.3333f * (aw + ax - ay);
-		frame[2] = 0.3333f * (aw - ax + ay);
-		frame[3] = 0.3333f * (aw - ax - ay);
-
-		if(0 > psf_sndWriteFloatFrames(ofd,frame,1)){
-			fprintf(stderr,"error writing sample block  %ld\n",got * outchans);
-			psf_sndClose(ifd);
-			psf_sndClose(ofd);
-			return 1;
-		}
-		if(framepos % halfsec==0) {
-			printf("%.2lf secs\r",(double)framepos / (double) props.srate);
-            fflush(stdout);
-        }
-		framepos++;
-	}
-
-	if(got != 0){
-		fprintf(stderr,"warning: not all data was read\n");
-	}
-	printf("\n%.4lf secs\nWritten %d quad frames to %s\n",(double)framepos / (double) props.srate,framepos,sfname);
-    if(psf_sndReadPeaks( ofd,peaks,&peaktime)){
-        printf("PEAK values:\n");
-        for(i=0; i < outchans; i++){
-            double val, dbval;
-            val = (double) peaks[i].val;
-            if(val > 0.0){
-                dbval = 20.0 * log10(val);
-                printf("CH %d: %.6f (%.2lfdB) at frame %u:\t%.4f secs\n",i,
-                        val,dbval,(unsigned int) peaks[i].pos,(double)peaks[i].pos / (double) props.srate);
-            }
-            else{
-                printf("CH %d: %.6f (-infdB) at frame %u:\t%.4f secs\n",i,
-                    val,(unsigned int) peaks[i].pos,(double)peaks[i].pos / (double) props.srate); 
-            }
-        }
-              
-    }
-    psf_sndClose(ifd);
-	psf_sndClose(ofd);
-	psf_finish();
-	return 0;
-}

dev/externals/mctools/abfdcode2.cpp

@@ -1,180 +0,0 @@
-/*
- * Copyright (c) 1983-2013 Richard Dobson and Composers Desktop Project Ltd
- * http://people.bath.ac.uk/masrwd
- * http://www.composersdesktop.com
- * This file is part of the CDP System.
- * The CDP System is free software; you can redistribute it
- * and/or modify it under the terms of the GNU Lesser General Public 
- * License as published by the Free Software Foundation; either 
- * version 2.1 of the License, or (at your option) any later version. 
- *
- * The CDP System is distributed in the hope that it will be useful, 
- * but WITHOUT ANY WARRANTY; without even the implied warranty of 
- * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  
- * See the GNU Lesser General Public License for more details.
- * You should have received a copy of the GNU Lesser General Public
- * License along with the CDP System; if not, write to the Free Software
- * Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
- *
- */
-
-//abfdcode2.cpp  : decode to std wavex speaker positions
-#include <stdio.h>
-#include <stdlib.h>
-#include <math.h>
-
-extern "C"{
-#include <sfsys.h>
-
-}
-
-typedef struct abf_samp {
-	float W;
-	float X;
-	float Y;
-	float Z;	
-} ABFSAMPLE;
-
-int main(int argc,char *argv[])
-{
-	int ifd, ofd;
-	long outchans = 4;
-	long outsize;
-	char *sfname;
-	bool write_wavex = false;
-	ABFSAMPLE abfsample;
-	float frame[4];
-	SFPROPS props;
-	CHPEAK peaks[4];
-	long inchans;
-
-	if(argc < 3){
-		printf("\nCDP MCTOOLS: ABFDCODE2 v 1.0: RWD,CDP 2009\n"
-				"Horizontal B Format Decoder\n"
-				"usage: abfdcode2 infile outfile layout\n"
-				"       (wavex layouts require .wav extension)\n"
-				
-				);
-		return 1;
-	}
-	while(argv[1][0] =='-'){		
-		switch(argv[1][1]){
-		case('x'):
-			write_wavex = true;
-			break;
-		default:
-			fprintf(stderr,"abfdecode: error: illegal flag option %s\n",argv[1]);
-			return 1;
-		}
-		argc--; argv++;
-	}
-	if(argc < 3){
-		fprintf(stderr,"CDP MCTOOLS: ABFDCODE.EXE: CDP 1999,2005\nHorizontal B Format Decoder\nusage: abfdcode [-x] infile outfile\n");
-		return 1;
-	}
-	if(sflinit("abfdcode")){
-		fprintf(stderr,"unable to initialize sfsys\n");
-		return 1;
-	}
-	
-	sfname = argv[2];
-	
-	ifd = sndopenEx(argv[1],0,CDP_OPEN_RDONLY);
-	if(ifd < 0){
-		fprintf(stderr,"unable toopen infile %s\n",argv[1]);
-		return 1;
-	}
-
-	if(!snd_headread(ifd,&props)){
-		fprintf(stderr,"unable to read infile header data\n");
-		sndcloseEx(ifd);
-		return 1;
-	}
-	if(!props.type==wt_wave){
-		fprintf(stderr,"infile is not a soundfile\n");
-		sndcloseEx(ifd);
-		return 1;
-	}
-	inchans = props.chans;
-	if(!(inchans == 3 || inchans == 4)){
-		fprintf(stderr,"Sorry: infile is not first-order\n");
-		sndcloseEx(ifd);
-		return 1;
-	}
-
-	outsize = sndsizeEx(ifd) / inchans;
-	if(outsize <= 0){
-		fprintf(stderr,"infile is empty!\n");
-		sndcloseEx(ifd);
-		return 1;
-	}
-	
-
-	//srate = props.srate;
-	//stype = props.samptype == FLOAT32 ? SAMP_FLOAT : SAMP_SHORT;
-	props.chformat = STDWAVE;
-	props.chans = 4;
-	if(write_wavex){
-		props.chformat = MC_QUAD;
-		props.format = WAVE_EX;
-	}
-	//ofd = sndcreat_formatted(sfname,outsize * outchans,stype,outchans,srate,CDP_CREATE_NORMAL);
-	ofd = sndcreat_ex(sfname,outsize * outchans,&props,SFILE_CDP,CDP_CREATE_NORMAL);
-	if(ofd < 0){
-		fprintf(stderr,"can't create outfile %s : %s\n",sfname,sferrstr());
-		//delete p_iosc;
-		//delete p_sintable;
-		sndcloseEx(ifd);
-		return 1;
-	}
-	peaks[0].value = peaks[1].value = peaks[2].value = peaks[3].value = 0.0f;
-	peaks[0].position = peaks[1].position = peaks[2].position = peaks[3].position = 0;
-
-	int got,quartersec = props.srate / 4;
-	unsigned int framepos = 0;
-	printf("\ndoing b-format decoding:\n");
-	while((got = fgetfbufEx((float *) &abfsample,inchans,ifd,0))==inchans){
-		int i;
-		//this_samp	= 0.5 * p_iosc->tick(1000);
-		float aw = abfsample.W;
-		float ax = abfsample.X * 0.707f;
-		float ay = abfsample.Y * 0.707f;
-		frame[0] = 0.3333f * (aw + ax + ay);
-		frame[1] = 0.3333f * (aw + ax - ay);
-		frame[2] = 0.3333f * (aw - ax + ay);
-		frame[3] = 0.3333f * (aw - ax - ay);
-
-		if(0 > fputfbufEx(frame,outchans,ofd)){
-			fprintf(stderr,"error writing sample block  %ld\n",got * outchans);
-			sndcloseEx(ifd);
-			sndcloseEx(ofd);
-			return 1;
-		}
-		for(i=0;i < 4;i++){
-			float val;
-			val= (float)fabs(frame[i]);
-			if(val > peaks[i].value) {
-				peaks[i].value = val;
-				peaks[i].position = framepos;
-			}
-		}
-		if(framepos % quartersec==0)
-			printf("%.2lf secs\r",(double)framepos / (double) props.srate);
-		framepos++;
-	}
-
-	if(got != 0){
-		fprintf(stderr,"warning: not all data was read\n");
-	}
-	printf("\n%.4lf secs\nWritten %d quad frames to %s\n",(double)framepos / (double) props.srate,framepos,sfname);
-	printf("PEAK values:\n");
-	printf("CH 1: %.4f at frame %ld:\t%.4f secs\n",peaks[0].value,peaks[0].position,(double)peaks[0].position / (double) props.srate);
-	printf("CH 2: %.4f at frame %ld:\t%.4f secs\n",peaks[1].value,peaks[1].position,(double)peaks[1].position / (double) props.srate);
-	printf("CH 3: %.4f at frame %ld:\t%.4f secs\n",peaks[2].value,peaks[2].position,(double)peaks[2].position / (double) props.srate);
-	printf("CH 4: %.4f at frame %ld:\t%.4f secs\n",peaks[3].value,peaks[3].position,(double)peaks[3].position / (double) props.srate);
-	sndputpeaks(ofd,4,peaks);
-	sndcloseEx(ifd);
-	sndcloseEx(ofd);
-	sffinish();
-	return 0;
-}

dev/externals/mctools/abfpan.cpp

@@ -1,276 +0,0 @@
-/*
- * Copyright (c) 1983-2013 Richard Dobson and Composers Desktop Project Ltd
- * http://people.bath.ac.uk/masrwd
- * http://www.composersdesktop.com
- * This file is part of the CDP System.
- * The CDP System is free software; you can redistribute it
- * and/or modify it under the terms of the GNU Lesser General Public 
- * License as published by the Free Software Foundation; either 
- * version 2.1 of the License, or (at your option) any later version. 
- *
- * The CDP System is distributed in the hope that it will be useful, 
- * but WITHOUT ANY WARRANTY; without even the implied warranty of 
- * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  
- * See the GNU Lesser General Public License for more details.
- * You should have received a copy of the GNU Lesser General Public
- * License along with the CDP System; if not, write to the Free Software
- * Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
- *
- */
-
-//abfpan.cpp  : write mono wave file into ambisonic B-format
-// Dec 2005 support .amb extension, and 3ch output
-// Jan 2010: corrected usage message about -b and -x
-// Nov 2013: updated usage message formatting, infile must be mono
-#include <stdio.h>
-#include <stdlib.h>
-#include <string.h>
-#include <math.h>
-#include "portsf.h"
-
-#ifdef unix
-/* in portsf.lib */
-extern "C" {
-    int stricmp(const char *a, const char *b);
-}
-#endif
-
-#ifndef TWOPI
-#define TWOPI	(6.283185307)
-#endif
-typedef struct abf_samp {
-	float W;
-	float X;
-	float Y;
-	float Z;	
-} ABFSAMPLE;
-
-
-enum {ARG_PROGNAME, ARG_INFILE, ARG_OUTFILE,ARG_STARTPOS, ARG_ENDPOS,ARG_NARGS};
-void usage()
-{
-	fprintf(stderr,"CDP MCTOOLS V1.5.3 (c) RWD,CDP 2009,2010,2013\n"
-				   "ABFPAN:  apply fixed or orbiting 1st-order B-Format pan to infile\n");
-	fprintf(stderr,"usage : abfpan [-b][-x][-oN] infile outfile startpos endpos\n");
-	fprintf(stderr,"    infile:   input soundfile, must be mono\n");
-	fprintf(stderr,"    startpos: 0.0 <= startpos <= 1.0 (0.0 and 1.0 = Centre Front)\n");
-	fprintf(stderr,"    endpos:   endpos < 0.0 gives anticlockwise rotation,\n"
-				   "                endpos > 0.0 gives clockwise rotation.\n"
-				   "                Units give number of revolutions, fraction gives final position\n"
-				   "                Set endpos = startpos for fixed pan\n");
-	fprintf(stderr,"    -b:       write output as horizontal Ambisonic B-format (.amb),\n"
-                   "                (i.e. bypass B-Format decoding).\n"
-			       "    -x:       write (WAVE_EX) format file (.wav). \n"			   
-				   "                Default: write standard m/c soundfile (WAVE,AIFF).\n"
-                   "                NB: -b overrides -x - no meaning in using both.\n"
-				   "    -oN:      number of B-Format output channels: N = 3 or 4 only.\n"
-				   "                Default: write 4-ch file.\n"
-                   "    See also ABFPAN2 for 2nd order version with fixed height.\n"
-	);
-    fprintf(stderr,"\n");
-}
-
-
-
-#define SYN_BUFLEN	(32768)
-#define TABLEN (1024)
-int main(int argc,char *argv[])
-{
-	int ifd, ofd;
-	int srate = 44100;
-	int outchans = 4;
-    MYLONG peaktime;
-	long outsize;	
-	float this_samp;
-	char *sfname;
-	double angle_incr;
-	double start_angle = 0.0;
-	double startpos,endpos;
-	
-	ABFSAMPLE abfsample;
-	PSF_PROPS props;
-	PSF_CHPEAK peaks[4];
-
-	bool write_bformat = false;
-	bool write_wavex = false;
-	int iarg = outchans;
-
-    /* CDP version number */
-    if(argc==2 && (stricmp(argv[1],"--version")==0)){
-        printf("1.5.2.\n");
-        return 0;
-    }
-    
-	if(argc < 5){
-		usage();
-		return 1;
-	}
-
-	if(psf_init()){
-		fprintf(stderr,"abfpan2: Startup failure.\n");
-		return 1;
-	}
-	
-	while(argv[1][0] =='-'){
-		
-		switch(argv[1][1]){
-		case('b'):
-			write_bformat = true;
-			break;
-
-		case('x'):
-			write_wavex = true;
-			break;
-		case('o'):
-			if(argv[1][2]=='\0'){
-				fprintf(stderr,"abfpan: error: -o flag requires channel value\n");
-				return 1;
-			}
-			iarg = atoi(&(argv[1][2]));
-			if(!(iarg == 3 || iarg == 4)){
-				fprintf(stderr,"abfpan: error: invalid channel value for -o flag\n");
-				return 1;
-			}
-
-			break;
-		default:
-			fprintf(stderr,"abfpan: error: illegal flag option %s.\n",argv[1]);
-			return 1;
-		}
-
-		argc--; argv++;
-	}
-
-	if(argc != 5){
-		usage();
-		return 1;
-	}
-
-	if(write_bformat)
-		outchans = iarg;
-    
-	sfname = argv[ARG_INFILE];
-	startpos = atof(argv[ARG_STARTPOS]);
-	if(startpos < 0.0 || startpos > 1.0){
-		fprintf(stderr,"abfpan: startpos %.4lf out of range: must be between 0.0 and 1.0.\n",startpos);
-		return 1;
-	}
-
-	endpos = atof(argv[ARG_ENDPOS]);
-
-
-
-	ifd = psf_sndOpen(sfname,&props,0);
-	if(ifd < 0){
-		fprintf(stderr,"abfpan: unable to open infile %s.\n",argv[1]);
-		return 1;
-	}
-	if(props.chans != 1){
-		fprintf(stderr,"abfpan: infile must be mono.\n");
-		psf_sndClose(ifd);
-		return 1;
-	}
-
-	outsize = psf_sndSize(ifd);
-	if(outsize <= 0){
-		fprintf(stderr,"abfpan: infile is empty!\n");
-		psf_sndClose(ifd);
-		return 1;
-	}
-    srate = props.srate;
-	props.chans = outchans;
-    if(!is_legalsize(outsize,&props)){
-        fprintf(stderr,"error: outfile size exceeds capacity of format.\n");
-        return 1;
-    }
-                
-	start_angle = - (TWOPI * startpos);	   //we think of positive as clockwise at cmdline!
-	angle_incr = TWOPI / outsize;
-	angle_incr *= (endpos - startpos);
-	
-	if(write_wavex){
-		props.format = PSF_WAVE_EX;
-		props.chformat = MC_QUAD;
-	}
-	if(write_bformat)
-		props.chformat = MC_BFMT;
-	ofd = psf_sndCreate(argv[ARG_OUTFILE],&props,0,0,PSF_CREATE_RDWR);
-	if(ofd < 0){
-		fprintf(stderr,"abfpan: can't create outfile %s.\n",argv[ARG_OUTFILE]);	
-		return 1;
-	}
-
-	int i,got;
-	int half_sec = srate / 4;
-	long total_frames = 0;
-	double d_srate = (double)srate;
-
-	abfsample.Z = 0.0f;
-	for(i=0; i < outchans; i++){
-		peaks[i].val = 0.0f;
-		peaks[i].pos = 0;
-	}
-	//should make one 360deg rotate over duration
-	//TODO: make lookup_sin and lookup_cos ugens, taking angle arg
-	if(write_bformat)
-		printf("\nWriting B-Format file:\n");
-	else
-		printf("\n");
-	while((got = psf_sndReadFloatFrames(ifd,&this_samp,1))==1){		
-		abfsample.W = (float) (this_samp * 0.707);
-        abfsample.X = (float) (this_samp * cos(start_angle));
-		abfsample.Y	= (float) (this_samp * sin(start_angle));
-
-		//set this for periphonic; otherwise = 0.0 for horizontal_only
-		//abfsample.Z = (float) (this_samp*0.25);
-		if(!write_bformat){
-			//decode into quad file
-			float aw = abfsample.W;
-			float ax = abfsample.X * 0.707f;
-			float ay = abfsample.Y * 0.707f;
-            /* ignore wxyz names  - this is now handy container for quad speaker feeds*/
-			abfsample.W = 0.33333f * (aw + ax + ay); /* front L */
-			abfsample.X = 0.33333f * (aw + ax - ay); /* front R */
-			abfsample.Y = 0.33333f * (aw - ax + ay); /* rear  L */
-			abfsample.Z = 0.33333f * (aw - ax - ay); /* rear  R */
-		}
-		if(0 > psf_sndWriteFloatFrames(ofd,(float*)  &abfsample,1)){
-			fprintf(stderr,"error writing abf sample frame  %ld\n",total_frames);
-			return 1;
-		}
-		start_angle -= angle_incr;
-		total_frames++;		
-		if(total_frames % half_sec ==0) {
-			printf("%.2lf secs\r",(double)total_frames / d_srate);
-            fflush(stdout);
-        }
-	}
-
-	if(got != 0){
-		fprintf(stderr,"abfpan: warning: not all data was read.\n");
-	}
-
-	printf("%.4lf secs\nwritten %ld %d-ch sample frames to %s\n",(double)outsize / d_srate,outsize,outchans, argv[ARG_OUTFILE]);
-	if(psf_sndReadPeaks( ofd,peaks,&peaktime)){
-		printf("PEAK values:\n");
-		for(i=0; i < outchans; i++){
-            double val, dbval;
-            val = (double) peaks[i].val;
-            if(val > 0.0){
-                dbval = 20.0 * log10(val);
-                printf("CH %d: %.6f (%.2lfdB) at frame %u:\t%.4f secs\n",i,
-                       val,dbval,(unsigned int) peaks[i].pos,(double)peaks[i].pos / (double) props.srate);
-            }
-            else{
-                printf("CH %d: %.6f (-infdB) at frame %u:\t%.4f secs\n",i,
-                        val, (unsigned int) peaks[i].pos,(double)peaks[i].pos / (double) props.srate); 
-            }
-        }
-        
-	}
-	printf("\n");
-	psf_sndClose(ifd);
-	psf_sndClose(ofd);
-	psf_finish();
-	return 0;
-}

dev/externals/mctools/abfpan2.cpp

@@ -1,296 +0,0 @@
-/*
- * Copyright (c) 1983-2013 Richard Dobson and Composers Desktop Project Ltd
- * http://people.bath.ac.uk/masrwd
- * http://www.composersdesktop.com
- * This file is part of the CDP System.
- * The CDP System is free software; you can redistribute it
- * and/or modify it under the terms of the GNU Lesser General Public 
- * License as published by the Free Software Foundation; either 
- * version 2.1 of the License, or (at your option) any later version. 
- *
- * The CDP System is distributed in the hope that it will be useful, 
- * but WITHOUT ANY WARRANTY; without even the implied warranty of 
- * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  
- * See the GNU Lesser General Public License for more details.
- * You should have received a copy of the GNU Lesser General Public
- * License along with the CDP System; if not, write to the Free Software
- * Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
- *
- */
-
-//abfpan2.cpp  : write mono wave file into ambisonic B-format
-// Dec 2005 support .amb extension, and 3ch output
-// OCT 2009 portsf version, with 2nd-order encoding
-#include <stdio.h>
-#include <stdlib.h>
-#include <string.h>
-#include <math.h>
-#include <memory.h>
-#include "portsf.h"
-
-#ifndef M_PI
-#define M_PI (3.141592654)
-#endif
-#ifndef TWOPI
-#define TWOPI	(6.283185307)
-#endif
-
-#ifdef unix
-/* in portsf.lib */
-extern "C" {
-    int stricmp(const char *a, const char *b);
-}
-#endif
-
-enum {W,X,Y,Z,R,S,T,U,V};
-
-
-enum {ARG_PROGNAME, ARG_INFILE, ARG_OUTFILE,ARG_STARTPOS,ARG_ENDPOS,ARG_NARGS};
-
-void usage()
-{
-	fprintf(stderr,"\nCDP MCTOOLS V1.0.1 beta (c) RWD,CDP 2010"
-				   "\n\nABFPAN2:  apply fixed or orbiting 2nd-order B-Format pan to infile\n");
-	fprintf(stderr,"usage : abfpan2 [-gGAIN][-w] [-p[DEG]] infile outfile startpos endpos\n"
-                   "    infile : mono source.\n"
-                   "    outfile: 2nd order B-format output.\n"        
-	               "        0.0 <= startpos <= 1.0 (0.0 and 1.0 = Centre Front).\n");
-	fprintf(stderr,"    endpos : endpos < 0.0 gives anticlockwise rotation,\n"
-				   "            endpos > 0.0 gives clockwise rotation.\n"
-				   "            Units give number of revolutions, fraction gives final position\n"
-				   "            Set endpos = startpos for fixed pan\n"
-                   "    -gGAIN : scale infile amplitude by GAIN (GAIN > 0).\n"
-                   "    -w     : Write standard soundfile (wave, aiff)\n"
-                   "            Default: WAVEX B-Format; use .amb extension.\n"
-                   "    -p[DEG]: write full 9-channel (periphonic) B-Format file.\n"
-                   "        Default: write 5-channel (2nd-order horizontal) file.\n"
-                   "        DEG:  optional fixed height argument (degrees).\n"
-                   "             Range = -180  to +180,\n"
-                   "             where  -90 = nadir, +90 = zenith (directly above).\n"
-                   "             Default: DEG=0;  height channels (Z,R,S,T)  will be empty.\n"
-                   "    NB: this program does not create a decoded output.\n"
-                   "    Use FMDCODE to decode to choice of speaker layouts.\n"
-				   );
-    fprintf(stderr,"\n");
-}
-
-int main(int argc,char *argv[])
-{
-	int i,got,ifd, ofd;
-	long srate = 44100;
-	long outchans = 5;
-    long do_peri = 0;
-    int write_wav = 1;
-	MYLONG peaktime;
-	long outsize;	
-	float this_samp;
-	int half_sec;
-	long total_frames;
-	double d_srate;
-	char *sfname;
-	double angle_incr;
-	double start_angle = 0.0;
-	double startpos,endpos;
-    double gain = 1.0;
-    double degree=0.0,elevation = 0.0;  
-	
-	float abfsample[9];
-    float outframe[5];
-    float *p_frame;
-    
-	PSF_PROPS props;
-	PSF_CHPEAK *peaks = NULL;
-	
-    /* CDP version number */
-    if(argc==2 && (stricmp(argv[1],"--version")==0)){
-        printf("1.0.1b\n");
-        return 0;
-    }
-    
-	if(argc < 5){
-		usage();
-		return 1;
-	}
-
-	if(psf_init()){
-		fprintf(stderr,"\nabfpan2: Startup failure");
-		return 1;
-	}
-	
-	while(argv[1][0] =='-'){
-		switch(argv[1][1]){
-        case 'g':
-            if(argv[1][2] == '\0'){
-                fprintf(stderr,"abfpan2 Error: -g flag requires a value.\n");
-                return 1;
-            }
-            gain = atof(&argv[1][2]);
-            if(gain <= 0.0){
-                printf("abfpan2: gain value must be positive!\n");
-                return 1;
-            }
-            break;
-        case 'p':
-            if(argv[1][2] != '\0'){
-                degree = atof(&argv[1][2]);
-                if(degree < -180.0 || degree > 180.0){
-                    fprintf(stderr,"-p: degree value out of range.\n");
-                    return 1;
-                }
-                elevation = degree * (M_PI  / 180.0);
-            }
-            outchans = 9;
-            do_peri = 1;
-            break;
-        case 'w':
-            write_wav = 0;
-            break;
-		default:
-			fprintf(stderr,"\nabfpan: error: illegal flag option %s",argv[1]);
-			return 1;
-		}
-		argc--; argv++;
-	}
-
-	if(argc < ARG_NARGS ){
-		usage();
-		return 1;
-	}
-
-	sfname = argv[ARG_INFILE];
-	startpos = atof(argv[ARG_STARTPOS]);
-	if(startpos < 0.0 || startpos > 1.0){
-		fprintf(stderr,"abfpan2: startpos %.4lf out of range: must be between 0.0 and 1.0\n",startpos);
-		return 1;
-	}
-
-	endpos = atof(argv[ARG_ENDPOS]);
-	ifd = psf_sndOpen(sfname,&props,0);
-	if(ifd < 0){
-		fprintf(stderr,"abfpan2: unable toopen infile %s.\n",sfname);
-		return 1;
-	}
-
-    if(props.chans != 1){
-		fprintf(stderr,"abfpan2: infile must be mono.\n");
-		psf_sndClose(ifd);
-		return 1;
-	}
-
-	outsize = psf_sndSize(ifd);
-	if(outsize <= 0){
-		fprintf(stderr,"abfpan2: infile is empty!\n");
-		psf_sndClose(ifd);
-		return 1;
-	}
-    srate = props.srate;
-	props.chans = outchans;
-    if(!is_legalsize(outsize,&props)){
-        fprintf(stderr,"error: outfile size exceeds capacity of format.\n");
-        return 1;
-    }
-    
-	start_angle = - (TWOPI * startpos);	   //we think of positive as clockwise at cmdline!
-	angle_incr = TWOPI / outsize;
-	angle_incr *= (endpos - startpos);
-    
-	
-    if(write_wav == 0)
-        props.format = PSF_STDWAVE;
-    else {
-        printf("Writing B-Format file.\n");
-	    props.format = PSF_WAVE_EX;
-        props.chformat = MC_BFMT;
-    }
-    
-    peaks = (PSF_CHPEAK*)  malloc(sizeof(PSF_CHPEAK) * outchans);
-    memset(peaks,0,sizeof(PSF_CHPEAK) * outchans);
-    
-    
-	ofd = psf_sndCreate(argv[ARG_OUTFILE],&props,0,0,PSF_CREATE_RDWR);
-	if(ofd < 0){
-		fprintf(stderr,"abfpan2: can't create outfile %s.\n",sfname);	
-		return 1;
-	}
-
-	half_sec = srate / 2;
-	total_frames = 0;
-	d_srate = (double)srate;
-	
-	//should make one 360deg rotate over duration
-	//TODO: make lookup_sin and lookup_cos ugens, taking angle arg
-	
-    if(do_peri)
-        p_frame = abfsample;
-    else
-        p_frame = outframe;
-    
-	while((got = psf_sndReadFloatFrames(ifd,&this_samp,1))==1){
-		double x,y,z,xx,yy,zz;
-        x = cos(start_angle);
-        y = sin(start_angle);
-        if(elevation ==0.0)
-            z = 0.0;
-        else
-            z = sin(elevation);
-        xx = x * x;
-        yy = y * y;
-        zz = z * z;
-        this_samp *= gain;
-        if(do_peri) {
-            abfsample[W] = (float) (this_samp * 0.7071);
-            abfsample[X] = (float) (this_samp * x);
-            abfsample[Y] = (float) (this_samp * y);
-            abfsample[Z] = (float) (this_samp * z);
-            abfsample[R] = (float) (this_samp * (1.5 * zz - 0.5));  /// ????
-            abfsample[S] = (float) (this_samp * (2.0 * z * x));
-            abfsample[T] = (float) (this_samp * (2.0 * y * z));
-            abfsample[U] = (float) (this_samp * (xx-yy));
-            abfsample[V] = (float) (this_samp * (2.0 * x * y));
-        }
-        else{
-            outframe[0]  = (float) (this_samp * 0.7071);
-            outframe[1]  = (float) (this_samp * x);
-            outframe[2]  = (float) (this_samp * y);
-            outframe[3] = (float) (this_samp * (xx-yy));
-            outframe[4] = (float) (this_samp * (2.0 * x * y));
-        }
-            
-		if(0 > psf_sndWriteFloatFrames(ofd, p_frame,1)){
-			fprintf(stderr,"abfpan2: error writing frame %ld\n",total_frames);
-			return 1;
-		}
-		start_angle -= angle_incr;
-		total_frames++;		
-		if(total_frames % half_sec ==0) {
-			printf("%.2lf secs\r",(double)total_frames / d_srate);
-            fflush(stdout);
-        }
-	}
-	if(got != 0){
-		fprintf(stderr,"abfpan2: warning: not all data was read.\n");
-	}
-	printf("%.4lf secs\nWritten %ld frames to %s.\n",(double)total_frames / d_srate,total_frames, argv[ARG_OUTFILE]);
-	if(psf_sndReadPeaks( ofd,peaks,&peaktime)){
-		printf("PEAK values:\n");
-		for(i=0; i < outchans; i++){
-            double val, dbval;
-            val = (double) peaks[i].val;
-            if(val > 0.0){
-                dbval = 20.0 * log10(val);
-                printf("CH %d: %.6f (%.2lfdB) at frame %u:\t%.4f secs\n",i,
-                       val,dbval,(unsigned int) peaks[i].pos,(double)peaks[i].pos / (double) props.srate);
-            }
-            else{
-                printf("CH %d: %.6f (-infdB) at frame %u:\t%.4f secs\n",i,
-                       val,(unsigned int)peaks[i].pos,(double)peaks[i].pos / (double) props.srate); 
-            }
-        }
-        
-	}
-
-	psf_sndClose(ifd);
-	psf_sndClose(ofd);
-	psf_finish();
-	return 0;
-}

dev/externals/mctools/channel.c

@@ -1,302 +0,0 @@
-/*
- * Copyright (c) 1983-2013 Richard Dobson and Composers Desktop Project Ltd
- * http://people.bath.ac.uk/masrwd
- * http://www.composersdesktop.com
- * This file is part of the CDP System.
- * The CDP System is free software; you can redistribute it
- * and/or modify it under the terms of the GNU Lesser General Public
- * License as published by the Free Software Foundation; either
- * version 2.1 of the License, or (at your option) any later version.
- *
- * The CDP System is distributed in the hope that it will be useful,
- * but WITHOUT ANY WARRANTY; without even the implied warranty of
- * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.
- * See the GNU Lesser General Public License for more details.
- * You should have received a copy of the GNU Lesser General Public
- * License along with the CDP System; if not, write to the Free Software
- * Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
- *
- */
-
-/* channelx.c */
-//RWD MCTOOLS, portsf version Oct 2009
-// nothing special, but use PEAK chunk (reading it where available)
-
-#include <stdio.h>
-#include <stdlib.h>
-#include <string.h>
-#include <math.h>
-#include <memory.h>
-#include <ctype.h>
-#include <portsf.h>
-
-#ifndef _MAX_PATH
-#define _MAX_PATH (1024)
-#endif
-#define MAXFILES 16
-
-#ifdef unix
-/* in portsf.lib */
-extern int stricmp(const char *a, const char *b);
-#endif
-
-
-void usage(void);
-//RWD.7.99 extended to get file extension from base name
-// so can be used without CDP_SOUND_EXT
-static void getbody(char*,char*,char *);
-
-void
-usage(void)
-{
-    fprintf(stderr,"\nCDP MCTOOLS: CHANNELX V1.6 (c) RWD, CDP 2010\n");
-    fprintf(stderr,
-            "Extract all channels from m/c file\n"
-            "usage: channelx [-oBASENAME] infile chan_no [chan_no...]\n"
-            //"\nusage:  channelx [-oBASENAME] infile"
-            "  -oBASENAME = base name (with extension) of outfiles (appended *_cN for ch N)\n"
-            "(NB: Channels of WAVE-EX files are written as standard soundfiles)\n"
-            );
-}
-
-#if defined(unix)
-#define SEP     '/'
-#else
-#define SEP     '\\'
-#endif
-
-void getbody(char *filename,char *body,char *ext)
-{
-    char *fn, *sl;
-
-    if((sl = strrchr(filename, SEP)) != 0)
-        sl++;
-    else
-        sl = filename;
-
-    for(fn = sl; *fn != '.' && *fn != '\0'; ) {
-        *body++ = *fn++;
-    }
-    *body++ = '\0';
-    //save extension
-    while(*fn != '\0')
-        *ext++ = *fn++;
-    *ext = '\0';
-}
-
-
-
-
-
-int main(int argc,char *argv[])
-{
-
-    int ifd = 0;                            /* Infile descriptor */
-    int ofd[MAXFILES];                      /* Outfile descriptors */
-    int i;                      /* Assorted indexes */
-    int inchans=1;                          /* Infile channels */
-    char filename[_MAX_PATH];       /* Infilename */
-    int select[MAXFILES];           /* Flags for each channel to MAXFILES */
-    int lastchan = 0;                       /* Maximum Requested channels */
-    int numchans = 0;                       /* number of channels extracted */
-    char body[100];                         /* Infile name prefix */
-    char ext[8];
-    char outext[8];                         /* output extension: to change .amb to .wav */
-    char *p_dot;
-    char ofile[_MAX_PATH];          /* Raw outfile name */
-    PSF_PROPS inprops;
-    int nframes =0, total_frames = 0;
-    float *frame = NULL;
-    int writefiles = 0;                     /* try to keep working even if one file fails...*/
-
-
-    /* CDP version number */
-    if(argc==2 && (stricmp(argv[1],"--version")==0)){
-        printf("1.6.\n");
-        return 0;
-    }
-    if(argc<2) {
-        usage();
-        return 1;
-    }
-
-    if(psf_init()) {
-        fprintf(stderr,"Startup failure.\n");
-        return 1;
-    }
-
-    body[0] = '\0';
-    ext[0] = '\0';
-
-    while(argv[1] != 0 && argv[1][0] == '-') {
-        if(strncmp(argv[1], "-o", 2) == 0) {
-            if(body[0] != '\0')
-                usage();
-            strcpy(body, &argv[1][2]);
-        }
-        argv++;
-        argc--;
-    }
-    if(argc < 2) {
-        usage();
-        return 1;
-    }
-
-    if(argv[1] != 0) {
-        sprintf(filename,"%s",argv[1]);
-        if((ifd = psf_sndOpen(filename,&inprops,0)) < 0) {
-            fprintf(stderr,"Channelx: Cannot open input file %s\n",argv[1]);
-            return 1;
-        }
-    }
-    argv++;
-
-    if(inprops.chans > MAXFILES){
-        fprintf(stderr,"Channelx: infile has too many channels!\n");
-        psf_sndClose(ifd);
-        return 1;
-    }
-    inchans = inprops.chans;
-    nframes = psf_sndSize(ifd);
-    if(nframes < 0){
-        fprintf(stderr,"Channelx: error reading infile size\n");
-        psf_sndClose(ifd);
-        return 1;
-    }
-    if(nframes==0){
-        fprintf(stderr,"Channelx: infile contains no data!\n");
-        psf_sndClose(ifd);
-        return 1;
-    }
-    nframes /= inchans;
-    //always create a standard file!
-    if(inprops.format == PSF_WAVE_EX) {
-        inprops.format = PSF_STDWAVE;
-        inprops.chformat = STDWAVE;
-    }
-    //that's all we need to know, now set one output channel
-    inprops.chans = 1;
-
-    frame = calloc(inchans,sizeof(float));
-    if(frame == NULL){
-        puts("Channelx: no memory for internal sample data\n");
-        psf_sndClose(ifd);
-        return 1;
-    }
-
-    for(i=0;i<MAXFILES;i++){
-        select[i] = 0;
-        ofd[i] = -1;
-    }
-
-
-    while(argv[1] != 0) {
-        int chn;
-
-        if(sscanf(argv[1],"%d",&chn) < 1){
-            fprintf(stderr,
-                    "Channelx: illegal channel argument %s\n\n",argv[1]);
-            goto cleanup;
-        }
-
-        if((chn > inchans) || (chn < 1)){
-            fprintf(stderr,"Channelx: channel argument out of range\n\n");
-            goto cleanup;
-        }
-        //check for duplicates
-        if(select[chn-1]==0){
-            select[chn-1] = 1;
-            if(chn > lastchan)
-                lastchan = chn;
-            numchans++;
-        }
-        else
-            fprintf(stderr,"Channelx: WARNING: duplicate channel number %d\n",chn);
-
-        argv++;
-        argc--;
-    }
-    if(numchans < 1) {              //RWD
-        fprintf(stderr, "channelx: must extract at least one channel\n");
-        goto cleanup;
-    }
-
-    writefiles = numchans;
-
-    if(body[0] == '\0')
-        getbody(filename,body,ext);
-
-    else if((p_dot = strrchr(body,'.')) != NULL){
-        //extension set in body arg
-        *p_dot++ = '\0';
-        sprintf(ext,".%s",p_dot);
-    }
-    //trap ambisonic file; single channels must be wav
-    strcpy(outext,ext);
-    if(stricmp(outext,".amb")==0)
-        strcpy(outext,".wav");
-    for(i=0;i<inchans;i++){
-        if(select[i]){
-            sprintf(ofile, "%s_c%d%s", body, i+1,outext);
-
-
-            if((ofd[i] = psf_sndCreate(ofile,&inprops,0,0,PSF_CREATE_RDWR)) < 0) {
-                fprintf(stderr,"Channelx: Failed to create %s\n\n",ofile);
-                goto cleanup;
-            }
-        }
-    }
-
-    printf("Extracting %d channels from '%s':\n%d sample frames\n",numchans,filename,nframes);
-
-    //stopwatch(1);
-    for(i=0;i < nframes; i++){
-        int got;
-        if(writefiles==0)
-            goto cleanup;
-
-        if((got = psf_sndReadFloatFrames(ifd,frame,1)) < 1){
-            if(got < 0)
-                fprintf(stderr,"error reading data from infile, after %d frames\n",i);
-            //sndunlink all outfiles?
-            else
-                break;
-            //goto cleanup;
-        }
-
-        for(i=1;i<lastchan+1;i++){
-            if(select[i-1]){
-                if((psf_sndWriteFloatFrames(ofd[i-1],frame +(i-1),1))<0){
-                    fprintf(stderr,"Channelx: Write [channel %d] failed.\n\n",i);
-                    //RWD
-                    //sndunlink(ofd[i-1]);
-                    writefiles--;
-                    if(writefiles==0)
-                        break;
-                    else
-                        continue;       //may be other files we can still write to...
-                }
-            }
-        }
-        total_frames++;
-        if((total_frames % (inprops.srate/2)) == 0){
-            printf("%.2lf secs\r",(double)total_frames / (double)inprops.srate);
-            fflush(stdout);
-        }
-    }
-    printf("%.4lf secs\n",(double)total_frames / (double)inprops.srate);
-    printf("%d files created.\n",writefiles);
-
- cleanup:
-    psf_sndClose(ifd);
-
-    for(i=0;i < MAXFILES; i++){
-        if(ofd[i] >= 0)
-            psf_sndClose(ofd[i]);
-    }
-
-
-    free(frame);
-    psf_finish();
-    return 0;
-}

dev/externals/mctools/chorder.c

@@ -1,297 +0,0 @@
-/*
- * Copyright (c) 1983-2013 Richard Dobson and Composers Desktop Project Ltd
- * http://people.bath.ac.uk/masrwd
- * http://www.composersdesktop.com
- * This file is part of the CDP System.
- * The CDP System is free software; you can redistribute it
- * and/or modify it under the terms of the GNU Lesser General Public 
- * License as published by the Free Software Foundation; either 
- * version 2.1 of the License, or (at your option) any later version. 
- *
- * The CDP System is distributed in the hope that it will be useful, 
- * but WITHOUT ANY WARRANTY; without even the implied warranty of 
- * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  
- * See the GNU Lesser General Public License for more details.
- * You should have received a copy of the GNU Lesser General Public
- * License along with the CDP System; if not, write to the Free Software
- * Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
- *
- */
- 
-/* chorder :  reorder channels in m/c file */
-/* Jan 2010: corrected behaviour with out of range order chars */
-/* July 2010 removed spurious premature call to usage() */
-#include <stdio.h>
-#include <stdlib.h>
-#include <string.h>
-#include <math.h>
-#include <portsf.h>
-
-/* set size of multi-channel frame-buffer */
-#define NFRAMES (1024)
-
-/* TODO define program argument list, excluding flags */
-enum {ARG_PROGNAME,ARG_INFILE,ARG_OUTFILE,ARG_ORDERSTRING,ARG_NARGS};
-#define N_BFORMATS (10)
-static const int bformats[N_BFORMATS] = {2,3,4,5,6,7,8,9,11,16};
-
-#ifdef unix
-/* in portsf.lib */
-extern int stricmp(const char *a, const char *b);
-#endif
-
-
-void usage(void)
-{
-    fprintf(stderr,"\nCDP MCTOOLS: CHORDER V1.2 (c) RWD,CDP 2009,2010\n");	
-    fprintf(stderr,
-            "Reorder soundfile channels.\n"
-            "Usage:  chorder infile outfile orderstring\n"
-            "  orderstring = any combination of characters a-z inclusive.\n"
-            "  Infile channels are mapped in order as a=1,b=2...z=26\n"
-            "  (For example: channels in a 4-channel file are represented by the\n"
-            "    characters abcd; any other character is an error).\n"
-            "  Characters must be lower case, and may be used more than once.\n"
-            "  Duplicate characters duplicate the corresponding input channel.\n"
-            "  The zero character (0) may be used to set a silent channel.\n"
-            "  A maximum of 26 channels is supported for both input and output.\n"
-            "  NB: infile (WAVEX) speaker positions are discarded.\n"
-            "  The .amb extension is supported for the outfile.\n\n"
-            );
-}
-
-
-int main(int argc, char* argv[])
-{
-	PSF_PROPS inprops,outprops;									
-	long framesread;	
-	/* init all dynamic resources to default states */
-	unsigned int i;
-    int halfsec;
-	unsigned int framepos;
-    long outsize;
-    int ifd = -1,ofd = -1;
-	int error = 0;
-	PSF_CHPEAK* peaks = NULL;	
-	psf_format outformat =  PSF_FMT_UNKNOWN;
-	unsigned long nframes = 1;
-	float* inframe = NULL;
-	float* outframe = NULL;
-    float* orderptr[26];
-    char* argstring = NULL;
-    unsigned int rootchar = 'a';
-    unsigned int maxchar = 'z';
-    unsigned int  nchars,nzeros = 0;
-    unsigned int max_inchar;
-    MYLONG peaktime;
-	
-    /* CDP version number */
-    if(argc==2 && (stricmp(argv[1],"--version")==0)){
-        printf("1.2.\n");
-        return 0;
-    }
-
-	/* process any optional flags: remove this block if none used! */
-	if(argc > 1){
-		char flag;
-		while(argv[1][0] == '-') {
-			flag = argv[1][1];
-			switch(flag){
-			/*TODO: handle any  flag arguments here */
-			case('\0'):
-				printf("Error: missing flag name\n");
-				return 1;
-			default:
-				break;
-			}
-			argc--;
-			argv++;
-		}
-	}
-
-	if(argc < ARG_NARGS){
-		printf("insufficient arguments.\n");
-		usage();
-		return 1;
-	}
-    
-    /* initial check of charstring */
-    argstring = argv[ARG_ORDERSTRING];
-    nchars = strlen(argstring);
-    if(nchars > 26) {
-        printf("error: order string too long.\n");
-        return 1;
-    }
-    
-	/*  always startup portsf */
-	if(psf_init()){
-		printf("unable to start portsf\n");
-		return 1;
-	}
-																							
-	ifd = psf_sndOpen(argv[ARG_INFILE],&inprops,0);															  
-	if(ifd < 0){
-		printf("Error: unable to open infile %s\n",argv[ARG_INFILE]);
-		error++;
-		goto exit;
-	}
-    outsize = psf_sndSize(ifd);
-	if(outsize <= 0){
-		fprintf(stderr,"chorder: infile is empty!\n");
-		psf_sndClose(ifd);
-		return 1;
-	}
-    inframe = (float*) malloc(nframes * inprops.chans * sizeof(float));
-	if(inframe==NULL){
-		puts("No memory!\n");
-		error++;
-		goto exit;
-	}
-    /* final validate and parse of charstring */
-    max_inchar = rootchar;
-    for(i=0;i < nchars;i++){
-        unsigned int thischar = argstring[i];
-//        printf("reading char %c (%d)\n",thischar,thischar);
-        unsigned int chindex;
-        if(thischar != '0' && (thischar < rootchar || thischar > maxchar)){
-            printf("illegal character in order string: %c\n",thischar);
-            goto exit;
-        }
-        if(thischar =='0'){
-//            printf("setting channel %d to zero.\n",i);
-            orderptr[i] = NULL;
-            nzeros++;
-        }
-        else{
-            if(thischar > max_inchar)
-                max_inchar = thischar;
-            chindex = thischar - rootchar;
-            orderptr[i] = &inframe[chindex];
-        }
-    }
-    if(nzeros==nchars)
-        printf("Warning: order string is all zeros - a silent file will be made!\n");
-    else{    
-        /* count inclusively! */
-        if(inprops.chans < (max_inchar - rootchar + 1)){
-            printf("File has %d channels; order string defines non-existent channels.\n",inprops.chans);
-            printf("For this file, maximum character is %c\n",rootchar+inprops.chans-1);
-            goto exit;
-        }
-    }
-	/* check file extension of outfile name, so we use correct output file format*/
-	outformat = psf_getFormatExt(argv[ARG_OUTFILE]);
-	if(outformat == PSF_FMT_UNKNOWN){
-		printf("outfile name %s has unsupported extension.\n"
-			"Use any of .wav, .aiff, .aif, .afc, .aifc, .amb\n",argv[ARG_OUTFILE]);
-		error++;
-		goto exit;
-	}
-	inprops.format = outformat;
-	outprops = inprops;
-	outprops.chans = nchars;
-    if(!is_legalsize(outsize,&outprops)){
-        fprintf(stderr,"error: outfile size exceeds capacity of format.\n");
-        return 1;
-    }
-    /* any speaker assignment etc invalidated! */
-    if(outformat == PSF_STDWAVE) {
-        outprops.chformat = MC_STD;
-        outprops.format =  PSF_WAVE_EX;
-    }
-    
-    if(outformat==PSF_WAVE_EX){
-        int matched = 0;
-        for(i=0;i < N_BFORMATS;i++)	{
-            if(inprops.chans == bformats[i]){
-                matched = 1;
-                break;
-            }
-        }
-        if(!matched){
-            printf("WARNING: No B-format definition for %d-channel file.\n",outprops.chans);
-        }
-        outprops.chformat = MC_BFMT;
-        outprops.format =  PSF_WAVE_EX;
-    }
-    
-    outframe = malloc(sizeof(float) * nchars);
-    
-	peaks  =  (PSF_CHPEAK*) malloc(outprops.chans * sizeof(PSF_CHPEAK));
-	if(peaks == NULL){
-		puts("No memory!\n");
-		error++;
-		goto exit;
-	}
-	ofd = psf_sndCreate(argv[ARG_OUTFILE],&outprops,0,0,PSF_CREATE_RDWR);
-	if(ofd < 0){
-		printf("Error: unable to create outfile %s\n",argv[ARG_OUTFILE]);
-		error++;
-		goto exit;
-	}
-	
-    halfsec = inprops.srate / 2;
-	framepos = 0;
-	printf("processing....\n");									
-	
-	while ((framesread = psf_sndReadFloatFrames(ifd,inframe,1)) > 0){
-        float val;
-        for(i=0;i < nchars;i++){
-            if(orderptr[i] == NULL)
-                val = 0.0f;
-            else
-                val = *orderptr[i];
-            outframe[i] = val;
-        }
-		
-
-		if(psf_sndWriteFloatFrames(ofd,outframe,1) != 1){
-			printf("Error writing to outfile\n");
-			error++;
-			break;
-		}
-        if((framepos % halfsec) == 0){
-			printf("%.2lf secs\r",(double) framepos / (double) outprops.srate);
-            fflush(stdout);
-        }
-		framepos++;
-	}
-
-	if(framesread < 0)	{
-		printf("Error reading infile. Outfile is incomplete.\n");
-		error++;
-	}
-	printf("\n%.4lf secs\nWritten %d frames to %s\n",(double)framepos / (double) outprops.srate,framepos,argv[ARG_OUTFILE]);
-		
-	if(psf_sndReadPeaks( ofd,peaks,&peaktime)){
-		printf("PEAK values:\n");
-		for(i=0; i < outprops.chans; i++){
-            double val, dbval;
-            val = (double) peaks[i].val;
-            if(val > 0.0){
-                dbval = 20.0 * log10(val);
-                printf("CH %d: %.6f (%.2lfdB) at frame %u:\t%.4f secs\n",i,
-                       val,dbval,(unsigned int) peaks[i].pos,(double)peaks[i].pos / (double) outprops.srate);
-            }
-            else{
-                printf("CH %d: %.6f (-infdB) at frame %u:\t%.4f secs\n",i,
-                       val,(unsigned int) peaks[i].pos,(double)peaks[i].pos / (double) outprops.srate); 
-            }
-        }
-	}
-	printf("\n");
-exit:	 	
-	if(ifd >= 0)
-		psf_sndClose(ifd);
-	if(ofd >= 0)
-		psf_sndClose(ofd);
-	if(inframe)
-		free(inframe);
-    if(outframe)
-        free(outframe);
-	if(peaks)
-		free(peaks);
-
-	psf_finish();
-	return error;
-}

dev/externals/mctools/chxformat.c

@@ -1,760 +0,0 @@
-/*
- * Copyright (c) 1983-2013 Richard Dobson and Composers Desktop Project Ltd
- * http://people.bath.ac.uk/masrwd
- * http://www.composersdesktop.com
- * This file is part of the CDP System.
- * The CDP System is free software; you can redistribute it
- * and/or modify it under the terms of the GNU Lesser General Public 
- * License as published by the Free Software Foundation; either 
- * version 2.1 of the License, or (at your option) any later version. 
- *
- * The CDP System is distributed in the hope that it will be useful, 
- * but WITHOUT ANY WARRANTY; without even the implied warranty of 
- * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  
- * See the GNU Lesser General Public License for more details.
- * You should have received a copy of the GNU Lesser General Public
- * License along with the CDP System; if not, write to the Free Software
- * Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
- *
- */
- 
-/* chxformat.c */
-#include <stdio.h>
-#include <stdlib.h>
-#include <string.h>
-#include <memory.h>
-#include <limits.h>
-#include <portsf.h>
-
-#ifdef unix
-/* in portsf.lib */
-extern int stricmp(const char *a, const char *b);
-#endif
-
-enum {ARG_PROGNAME, ARG_INFILE,ARG_NARGS};
-enum {GUID_PCM,GUID_IEEE,GUID_AMB_PCM,GUID_AMB_IEEE};
-
-char* guidnames[] = {"PCM","PCM FLOAT","AMB PCM","AMB FLOAT"};
-
-
-#define REVDWBYTES(t)	( (((t)&0xff) << 24) | (((t)&0xff00) << 8) | (((t)&0xff0000) >> 8) | (((t)>>24) & 0xff) )
-#define REVWBYTES(t)	( (((t)&0xff) << 8) | (((t)>>8) &0xff) )
-#define TAG(a,b,c,d)	( ((a)<<24) | ((b)<<16) | ((c)<<8) | (d) )
-
-#ifdef linux
-#define POS64(x) (x.__pos)
-#else
-#define POS64(x) (x)
-#endif
-
-#define WAVE_FORMAT_PCM			(0x0001)
-#define sizeof_WFMTEX  (40)
-typedef struct _GUID 
-{ 
-    unsigned int        Data1; 
-    unsigned short       Data2; 
-    unsigned short       Data3; 
-    unsigned char        Data4[8]; 
-} GUID; 
-
-
-typedef struct  {
-	WORD  wFormatTag; 
-    WORD  nChannels; 
-    DWORD nSamplesPerSec; 
-    DWORD nAvgBytesPerSec; 
-    WORD  nBlockAlign; 
-    WORD  wBitsPerSample; 
-
-} WAVEFORMAT;
-
-
-typedef struct { 
-    WORD  wFormatTag; 
-    WORD  nChannels; 
-    DWORD nSamplesPerSec; 
-    DWORD nAvgBytesPerSec; 
-    WORD  nBlockAlign; 
-    WORD  wBitsPerSample; 
-    WORD  cbSize; 
-} WAVEFORMATEX; 
-
-typedef struct {
-    WAVEFORMATEX    Format;				/* 18 bytes */
-    union {
-        WORD wValidBitsPerSample;       /* bits of precision  */
-        WORD wSamplesPerBlock;          /* valid if wBitsPerSample==0 */
-        WORD wReserved;                 /* If neither applies, set to */
-                                        /* zero. */
-    } Samples;
-    DWORD    dwChannelMask;				/* which channels are */
-                                        /* present in stream  */
-    GUID     SubFormat;
-} WAVEFORMATEXTENSIBLE;
-
-static const GUID  KSDATAFORMAT_SUBTYPE_PCM = {0x00000001,0x0000,0x0010,
-								{0x80,
-								0x00,
-								0x00,
-								0xaa,
-								0x00,
-								0x38,
-								0x9b,
-								0x71}};
-
-static const GUID  KSDATAFORMAT_SUBTYPE_IEEE_FLOAT = {0x00000003,0x0000,0x0010,
-								{0x80,
-								0x00,
-								0x00,
-								0xaa,
-								0x00,
-								0x38,
-								0x9b,
-								0x71}};
-
-static const GUID SUBTYPE_AMBISONIC_B_FORMAT_PCM = { 0x00000001, 0x0721, 0x11d3, 
-												{ 0x86,
-												0x44,
-												0xc8,
-												0xc1,
-												0xca,
-												0x0,
-												0x0,
-												0x0 } };
-
-
-static const GUID SUBTYPE_AMBISONIC_B_FORMAT_IEEE_FLOAT = { 0x00000003, 0x0721, 0x11d3, 
-												{ 0x86,
-												0x44,
-												0xc8,
-												0xc1,
-												0xca,
-												0x0,
-												0x0,
-												0x0 } };
-
-
-#define WAVE_FORMAT_IEEE_FLOAT	(0x0003)
-#define WAVE_FORMAT_EXTENSIBLE	(0xfffe)
-
-static int compare_guids(const GUID *gleft, const GUID *gright)
-{
-	const char *left = (const char *) gleft, *right = (const char *) gright;
-	return !memcmp(left,right,sizeof(GUID));
-}
-
-
-static int wavDoRead(FILE* fp, void* buf, DWORD nBytes)
-{
-	DWORD got = 0;	
-	if((got = fread(buf,sizeof(char),nBytes,fp)) != nBytes) {
-        return 1;
-    }
-	return 0;
-}
-
-static int byte_order()					
-{						    
-  int   one = 1;
-  char* endptr = (char *) &one;
-  return (*endptr);
-}
-
-static void fmtSwapBytes(WAVEFORMATEX  *pfmt)
-{	
-	pfmt->wFormatTag	= (WORD) REVWBYTES(pfmt->wFormatTag);
-	pfmt->nChannels		= (WORD) REVWBYTES(pfmt->nChannels);
-	pfmt->nSamplesPerSec	= REVDWBYTES(pfmt->nSamplesPerSec);
-	pfmt->nAvgBytesPerSec	= REVDWBYTES(pfmt->nAvgBytesPerSec);
-	pfmt->nBlockAlign	= (WORD) REVWBYTES(pfmt->nBlockAlign);
-	pfmt->wBitsPerSample	= (WORD) REVWBYTES(pfmt->wBitsPerSample);	
-}
-
-
-
-void printmaskinfo()
-{
-    printf("WAVEX Speaker positions:\n");
-
-    
-printf(
-"FRONT LEFT             0x1\n"
-"FRONT RIGHT            0x2\n"
-"FRONT CENTER           0x4\n"
-"LOW FREQUENCY          0x8\n"
-"BACK LEFT              0x10\n"
-"BACK RIGHT             0x20\n"
-"FRONT LEFT OF CENTER   0x40\n"
-"FRONT RIGHT OF CENTER  0x80\n"
-"BACK CENTER            0x100\n"
-"SIDE LEFT              0x200\n"
-"SIDE RIGHT             0x400\n"
-"TOP CENTER             0x800\n"
-"TOP FRONT LEFT         0x1000\n"
-"TOP FRONT CENTER       0x2000\n"
-"TOP FRONT RIGHT        0x4000\n"
-"TOP BACK LEFT          0x8000\n"
-"TOP BACK CENTER        0x10000\n"
-"TOP BACK RIGHT         0x20000\n"
-       );
-printf("The value 0x80000000 is reserved.\n");
-    
-printf("Standard layouts:\n"
-
-"Mono       0x40        (64)\n"
-"Stereo     0x03         (3)\n"
-"Quad       0x33        (51)\n"
-"LCRS       0x107      (263)\n"
-"5.0        0x37        (55)\n"
-"5.1        0x3f        (63)\n"
-"7.1        0xff       (255)\n"
-"Cube       0x2d033 (184371)\n"
-       ); 
-}
-
-void usage(void)
-{
-    printf("CDP MCTOOLS: CHXFORMAT v1.0.1beta (c) RWD,CDP 2009\n");
-    printf("change GUID and/or speaker mask in WAVEX header of infile.\n");
-    printf("usage: chxformat [-m | [[-t] [-gGUID] [-sMASK]] filename\n");
-    printf(
-           " -gGUID : change GUID type between PCM and AMB.\n"
-           "          Plain WAVEX: GUID = 1\n"
-           "          AMB:         GUID = 2\n"
-           " -sMASK : change speaker position mask.\n"
-           "          MASK = 0:  unset channel mask\n"
-           "             else set to MASK\n"
-           "          MASK may be given in decimal or hex (prefix '0x').\n"
-           " -m     : (not in combination with other options)\n"
-           " NOTE: speaker positions are only supported for WAVEX PCM files.\n"
-           "       If GUID is set to 2, the -s option should not be used. Any existing\n"
-           "       speaker positions will be set to zero.\n"
-           " Type chxformat -m to see list of WAVEX mask values.\n"
-           " -t     : Test only: do not modify file.\n"
-           "  If only infile given, program prints current GUID type and speaker mask.\n"
-           "  In test mode, program checks file, prints current channel mask as binary,\n"
-           "   and new mask in binary, if -s option set.\n"
-           "  Otherwise: program modifies infile (\"destructively\") - use with care!\n"
-           );
-    
-}
-
-void binprintf(int val,int skipzeroes)
-{
-    unsigned int nbits = sizeof(int) * CHAR_BIT;
-    unsigned int i = 0;
-    unsigned int mask = 1 << (nbits-1); /* i.e. 0x80000000; */
-    // skip leading zeroes
-    if(skipzeroes){
-        for(;i < nbits;i++){
-            if(val&mask)
-                break;
-            mask>>=1;
-        }
-    }
-    for(;i < nbits;i++){
-        if(i > 0 && i%8 == 0)
-            printf(" ");
-        printf("%d",(val & mask) ? 1 : 0);
-        mask >>= 1;
-    }
-}
-
-int countbits(int val)
-{
-    unsigned int nbits = sizeof(int) * CHAR_BIT;
-    unsigned int i = 0;
-    int count = 0;
-    unsigned int mask = 1 << (nbits-1); /* i.e. 0x80000000; */
-    for(;i< nbits;i++){
-        if(val & mask)
-            count++;
-        mask >>= 1;
-    }
-    return count;
-}
-
-int isdec(int ch){
-    if(ch >= '0' && ch <= '9')
-        return 1;
-//    printf("isdec fail: %d\n",ch);
-    return 0;
-}
-
-int ishex(int ch){
-    
-    if((ch >='A' && ch <= 'F') || (ch >= 'a' && ch <= 'f'))
-        return 1;
-//     printf("ishex fail: %c\n",ch);
-    return 0;
-}
-
-int validate_mask(const char* mask)
-{
-    int is_dec = 1,is_hex = 1;
-    size_t i,len;
-    char* pch;
-    if(mask==NULL)
-        return 0;
-    len = strlen(mask);
-    if(len==0)
-        return 0;
-    
-    pch = (char*) mask;
-    /*try hex */
-    if(len > 2 && *pch =='0'){
-        pch++;
-        if(*pch=='x' || *pch =='X'){ // prefix found
-            pch++;
-//            printf("got hex prefix\n");
-            for(i = 0;i < len-2;i++){
-//                printf("testing %d \n",pch[i]);
-                if(!(isdec(pch[i]) || ishex(pch[i]))){
-                    is_hex = 0;
-                    break;
-                }
-            }
-            if(is_hex)
-                return 1;
-        }
-        // not hex, maybe decimal
-    }
-    pch = (char*) mask;
-    for(i=0;i < len;i++){
-        if(!isdec(pch[i])){
-            is_dec = 0;
-            break;
-        }
-    }
-    return is_dec;
-}
-
-
-
-int main (int argc, char**argv)
-{
-    int do_changeguid = 0;
-    int in_guidtype =0;
-    int guidtoset = 0;
-    int do_changemask = 0;
-    unsigned int speakermask = 0;
-    int fmtfound = 0;
-    int test = 0;
-    int src_is_amb = 0;
-    FILE* fp  = NULL;
-    char* maskstring = NULL;
-    psf_format outformat;
-    psf_format new_outtype;
-    DWORD tag;
-    DWORD size;
-    fpos_t bytepos;
-    DWORD fmtoffset = 0;
-    DWORD guidoffset = 0;
-    DWORD maskoffset = 0;
-	WORD cbSize;
-    WORD validbits;
-    DWORD chmask;
-    WAVEFORMATEXTENSIBLE fmt;
-    int is_little_endian = byte_order();
-    
-    /* CDP version number */
-    if(argc==2 && (stricmp(argv[1],"--version")==0)){
-        printf("1.0.1b\n");
-        return 0;
-    }
-    
-    if(argc<2){
-        usage();
-        return 1;
-    }
-    while(argv[1][0]=='-'){
-		switch(argv[1][1]){
-            case 'm':
-                if(argc > 2){
-                    printf("cannot use -m with other options\n");
-                    return 1;
-                }
-                else{
-                    printmaskinfo();
-                    return 0;
-                }
-                break;
-            case 'g':
-                if(do_changeguid){
-                    printf("Cannot use -g option more than once!\n");
-                    return 1;
-                }
-                guidtoset = atoi(&argv[1][2]);
-                if(guidtoset < 1 || guidtoset > 2) {
-                    printf("bad value for -g option - use 1 or 2 only.\n");
-                    return 1;
-                }
-                do_changeguid = 1;
-                break;
-            case 's':
-                if(do_changemask){
-                    printf("Cannot use -s option more than once!\n");
-                    return 1;
-                }
-                maskstring =  &argv[1][2];
-                if(validate_mask(maskstring)==0){
-                    printf("Bad format for mask argument.\n"
-                           "Value must be decimal, or hex with 0x prefix.\n");
-                    return 1;
-                }
-                    
-                do_changemask = 1;
-                break;
-            case 't':
-                if(test==1){
-                    printf("You only need to set -t once!\n");
-                }
-                else
-                    printf("Test mode: no changes will be made.\n");
-                test = 1;
-                
-                break;
-            default:
-                printf("illegal option %s\n",argv[1]);
-                return 1;
-        }
-        argc--;
-        argv++;
-    }
-    if(argc != 2){
-        printf("infile argument required.\n");
-        usage();
-        return 1;
-    }
-    psf_init();
-    /* nitty-gritty to read header */
-    outformat = psf_getFormatExt(argv[ARG_INFILE]);
-	if(!(outformat == PSF_STDWAVE|| outformat==PSF_WAVE_EX)){
-		printf("file mustbe WAVEX format with .wav or .amb extension.\n");
-        return 1;
-	}
-    
-    fp = fopen(argv[ARG_INFILE],"rb+");
-    if(fp==NULL){
-        printf("unable to open infile %s\n",argv[ARG_INFILE]);
-        return 1;
-    }
-    if(wavDoRead(fp,(char *)&tag,sizeof(DWORD))
-		|| wavDoRead(fp,(char *) &size,sizeof(DWORD))) {
-		printf("read error 1\n");
-        return 1;
-    }
-	if(!is_little_endian)
-		size = REVDWBYTES(size);
-	else
-		tag = REVDWBYTES(tag);
-	if(tag != TAG('R','I','F','F')){
-		printf("not a RIFF file\n");
-        return 1;
-    }
-	if(size < (sizeof(WAVEFORMAT) + 3 * sizeof(WORD))){
-		printf("file has bad header.\n");
-        return 1;
-    }
-
-	if(wavDoRead(fp,(char *)&tag,sizeof(DWORD))) {
-		printf("read error 2\n");
-        return 1;
-    }
-	if(is_little_endian)
-		tag = REVDWBYTES(tag);
-	if(tag != TAG('W','A','V','E')){
-		printf("Not a WAVE file.\n");
-        return 1;    
-    }
-	for(;;){
-        if(fmtfound) 
-            break;
-		if(wavDoRead(fp,(char *)&tag,sizeof(DWORD))
-				|| wavDoRead(fp,(char *) &size,sizeof(DWORD))) {
-			printf("read error 3\n");
-            return 1;
-        }
-		if(!is_little_endian)
-			size = REVDWBYTES(size);
-		else
-			tag = REVDWBYTES(tag);
-		switch(tag){
-		case(TAG('f','m','t',' ')):
-			if( size < sizeof(WAVEFORMAT)){
-				printf("file has bad format.\n");
-                return 1;
-            }
-            
-			if(size > sizeof_WFMTEX) {
-				printf("file has unsupported WAVE format.\n");
-                return 1;
-            }
-			if(fgetpos(fp,&bytepos)) {
-			    printf("seek error\n");
-                return 1;
-            }
-			fmtoffset = (DWORD) POS64(bytepos);
-			if(wavDoRead(fp,(char *) &fmt.Format,sizeof(WAVEFORMAT))){				
-				printf("read error 4\n");
-                return 1;
-			}
-			if(!is_little_endian)		
-				fmtSwapBytes(&fmt.Format);	
-			/* calling function decides if format is supported*/
-			if(size > sizeof(WAVEFORMAT)) {
-				if(wavDoRead(fp,(char*)&cbSize,sizeof(WORD))){
-					printf("read error 5\n");
-                    return 1;
-                }
-				if(!is_little_endian)		
-					cbSize = (WORD) REVWBYTES(cbSize);
-                if(cbSize==0){
-                    printf("file is plain WAVE format.\n");
-                    return 1;
-                }
-				if(fmt.Format.wFormatTag == WAVE_FORMAT_EXTENSIBLE){
-					if(cbSize != 22) {
-						printf("not a recognized WAVEX file.\n");
-                        return 1;
-					}
-				}				
-				if(wavDoRead(fp,(char *) &validbits,sizeof(WORD))){
-					printf("read error 6\n");
-                    return 1;
-                }
-                
-				if(!is_little_endian)
-                    validbits = REVWBYTES(validbits);
-                fmt.Samples.wValidBitsPerSample = (WORD) validbits;
-				if(wavDoRead(fp,(char *) &chmask,sizeof(DWORD))) {
-					printf("read error 7\n");
-                    return 1;
-                }
-				if(!is_little_endian)
-                    chmask = REVDWBYTES(chmask);
-                fmt.dwChannelMask = chmask;
-				if(wavDoRead(fp,(char *) &(fmt.SubFormat),sizeof(GUID))) {
-					printf("read error 8 \n");
-                    return 1;
-                }
-				if(!is_little_endian){
-					fmt.SubFormat.Data1 = REVDWBYTES(fmt.SubFormat.Data1);
-					fmt.SubFormat.Data2 = (WORD) REVWBYTES(fmt.SubFormat.Data2);
-					fmt.SubFormat.Data3 = (WORD) REVWBYTES(fmt.SubFormat.Data3);	
-				}
-				/* if we get a good GUID, we are ready to make changes! */
-                
-				if(compare_guids(&(fmt.SubFormat),&(KSDATAFORMAT_SUBTYPE_PCM))) {
-                    in_guidtype = GUID_PCM;
-                    if(test)
-                        printf("Current GUID: KSDATAFORMAT_SUBTYPE_PCM.\n");
-                }
-                else if(compare_guids(&(fmt.SubFormat),&(KSDATAFORMAT_SUBTYPE_IEEE_FLOAT))) {
-                    in_guidtype = GUID_IEEE;
-                    if(test)
-                        printf("Current GUID: KSDATAFORMAT_SUBTYPE_IEEE_FLOAT.\n");
-                }
-                else if(compare_guids(&(fmt.SubFormat),&(SUBTYPE_AMBISONIC_B_FORMAT_PCM))) {
-                    in_guidtype = GUID_AMB_PCM;
-                    src_is_amb = 1;
-                    if(test)
-                        printf("Current GUID: SUBTYPE_AMBISONIC_B_FORMAT_PCM.\n");
-                }
-                else if(compare_guids(&(fmt.SubFormat),&(SUBTYPE_AMBISONIC_B_FORMAT_IEEE_FLOAT))){
-                    in_guidtype = GUID_AMB_IEEE;
-                    src_is_amb = 1;
-                    if(test)
-                        printf("Current GUID: SUBTYPE_AMBISONIC_B_FORMAT_IEEE_FLOAT.\n");
-                }
-                else {
-                    printf("unrecognized WAVE_EX GUID.\n");
-                    return 1;
-                }				
-			}
-            else {
-                printf("WAVEX format required.\n"
-                       "Use copysfx to convert to WAVEX format.\n");
-                return 1;
-            }
-            fmtfound = 1;			
-			break;
-		case TAG('d','a','t','a'):
-            if(!fmtfound){
-                printf("bad WAVE file: no fmt chunk found!\n");
-                return 1;
-            }                 
-        default:
-			/* unknown chunk - skip */
-            if(!fmtfound) {
-                if(fseek(fp,size,SEEK_CUR)){
-                    printf("seek error\n");
-                    return 1;
-                }
-            }
-			break;
-		}
-	}
-    if(!fmtfound){
-         printf("no fmt chunk found!\n");
-         return 1;
-    }
-    
-    maskoffset = fmtoffset + sizeof(WAVEFORMAT) + sizeof(WORD)*2;
-    guidoffset = maskoffset + sizeof(DWORD);
-    
-    if(!(do_changeguid || do_changemask)){
-        /* display what we know */
-        printf("Wordsize     : %d\n",validbits);
-        printf("Channels     : %d\n",fmt.Format.nChannels);
-        printf("GUID         : %s\n",guidnames[in_guidtype]);
-        if(chmask==0)
-            printf("Channel mask : 0\n");
-        else
-            printf("Channel mask : 0x%06x (%d)\n",(unsigned int) chmask,(int) chmask);
-        if(chmask){
-            printf("Channel mask (binary): ");
-            binprintf(chmask,1);
-            printf("There are %d speaker positions set.\n",countbits(chmask));
-            
-        }
-        printf("\n");
-    }
-    else {
-        int bits,is_same = 0;
-
-        /* check all settings before making any changes */
-        if(do_changemask){
-            // be strict!
-            if(src_is_amb && guidtoset != 1){
-                printf("-s flag only supported for PCM files (AMB speaker mask must be zero).\n"
-                       "Exiting.\n");
-                return 1;
-            }
-            if(strncmp(maskstring,"0x",2)==0)
-                sscanf(maskstring,"%x", &speakermask);
-            else
-                sscanf(maskstring,"%u", &speakermask);
-            bits = countbits(speakermask);
-            /* can't do much if a silly huge value given */
-            if(speakermask < 0 || speakermask >= 0x40000){
-                printf("Mask value out of range!\n");
-                return 1;
-            }
-            if(bits > fmt.Format.nChannels){
-                printf("Error: %d positions requested for %d channels: \n",bits,fmt.Format.nChannels);
-                binprintf(speakermask,1);
-                printf("\n");
-                return 1;
-            }
-            if(bits && bits < fmt.Format.nChannels){
-                printf("Warning: %d requested positions less than channels in file.\n",bits);
-            }
-        }
-                
-        if(do_changeguid){
-            GUID nuguid;
-            
-            // set float or pcm GUID as required.
-            // no copy to same: only convert if types are different
-            switch(guidtoset){
-            case 1:       // plain wavex - the easy iption
-                if(in_guidtype == GUID_AMB_PCM){
-                    nuguid = KSDATAFORMAT_SUBTYPE_PCM;
-                }
-                else if(in_guidtype == GUID_AMB_IEEE) {
-                    nuguid = KSDATAFORMAT_SUBTYPE_IEEE_FLOAT;
-                }
-                else
-                    is_same = 1;
-                new_outtype = PSF_STDWAVE;
-                break;
-            case 2:  // convert to amb - must zero speaker flags
-                if(in_guidtype == GUID_PCM)
-                    nuguid = SUBTYPE_AMBISONIC_B_FORMAT_PCM;
-                else if(in_guidtype == GUID_IEEE)
-                    nuguid = SUBTYPE_AMBISONIC_B_FORMAT_IEEE_FLOAT;
-                else
-                    is_same = 1;
-                // set mask to zero if setting AMB format
-                if(is_same==0) {
-                    do_changemask = 1;
-                    maskstring = "0";
-                    speakermask = 0;
-                }
-                new_outtype = PSF_WAVE_EX;    
-                break;
-            }
-            if(is_same){
-                printf("new GUID identical - no change made.\n");
-            }
-            else {
-                int err;
-                //make the change!
-                if(!is_little_endian){
-                    nuguid.Data1 = REVDWBYTES(nuguid.Data1);
-					nuguid.Data2 = (WORD) REVWBYTES(nuguid.Data2);
-					nuguid.Data3 = (WORD) REVWBYTES(nuguid.Data3);
-                }
-                if(!test){
-                    if(fseek(fp,guidoffset,SEEK_SET) !=0){
-                        printf("seek error updating channelmask. exiting.\n");
-                        return 1;
-                    }
-                    err = fwrite((char*) &nuguid,1,sizeof(GUID),fp);
-                    if(err != sizeof(GUID)){
-                        printf("Error updating GUID. File may have inconsistent header.\n");
-                        return 1;
-                    }
-                    printf("new GUID set.\n");
-                    
-                }
-            }
-            
-        }
-        if(do_changemask){
-            /* read user mask value */
-            /* TODO: full parse, trap bad characters */
-            if(chmask==speakermask){
-                if(speakermask > 0)
-                    printf("Requested mask is already set. Data not modified.\n");
-            }
-            else {
-                DWORD writemask;
-                int err;
-                
-                if(speakermask > 0){
-                    printf("new mask = %d (",speakermask);
-                    binprintf(speakermask,1);
-                    printf(")\n");
-                }
-                if(fseek(fp,maskoffset,SEEK_SET) !=0){
-                    printf("seek error updating channelmask. exiting.\n");
-                    return 1;
-                }
-                writemask = speakermask;
-                if(!is_little_endian)
-                    writemask = REVDWBYTES(speakermask);
-                if(!test){
-                    err = fwrite((char*) &writemask,1,sizeof(DWORD),fp);
-                    if(err != sizeof(DWORD)){
-                        printf("Error updating mask. File may have inconsistent header.\n");
-                        return 1;
-                    }
-                    printf("New mask set.\n");
-                }
-            }
-                
-        }
-        if(outformat == PSF_STDWAVE && new_outtype==PSF_WAVE_EX)
-            printf("File extension should be changed to amb.\n");
-        else if(outformat == PSF_WAVE_EX && new_outtype == PSF_STDWAVE)
-            printf("File extension should be changed to wav.\n");
-    }
-    fseek(fp,0,SEEK_END);
-    fclose(fp);
-    return 0;
-}
-
-
-    

dev/externals/mctools/copysf.c

@@ -1,604 +0,0 @@
-/*
- * Copyright (c) 1983-2013 Richard Dobson and Composers Desktop Project Ltd
- * http://people.bath.ac.uk/masrwd
- * http://www.composersdesktop.com
- * This file is part of the CDP System.
- * The CDP System is free software; you can redistribute it
- * and/or modify it under the terms of the GNU Lesser General Public 
- * License as published by the Free Software Foundation; either 
- * version 2.1 of the License, or (at your option) any later version. 
- *
- * The CDP System is distributed in the hope that it will be useful, 
- * but WITHOUT ANY WARRANTY; without even the implied warranty of 
- * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  
- * See the GNU Lesser General Public License for more details.
- * You should have received a copy of the GNU Lesser General Public
- * License along with the CDP System; if not, write to the Free Software
- * Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
- *
- */
- 
-/*
- *	copysfx: version of copysfx using portsf
- */
-/* Oct 2009: added cube */
-/* Dec 2009 1.9.1. fixed horrible bug failing to convert to WAVE_EX! */
-/* March 2012 corrected wave type arg parsing  */
-/* August 2012 updated portsf to correct SPKRS_MONO */
-/* Nov 2013 added MC_SURR_6_1 */
-#include <stdio.h>
-#include <stdlib.h>
-#include <ctype.h>
-#ifdef _WIN32
-#include <malloc.h>			//RWD.6.5.99
-#endif
-#include <string.h>
-#include <math.h>
-#include <time.h>
-#include <sys/types.h>
-#include <sys/timeb.h>
-#include <portsf.h>
-
-#define VERSION "Revision: 2.1.1 2020 "
-
-#ifdef unix
-/* in portsf.lib */
-extern int stricmp(const char *a, const char *b);
-#endif
-
-enum {OPT_STD_WAVE,OPT_WAVEX_GENERIC,OPT_WAVEX,OPT_WAVEX_LCRS,OPT_WAVEX_SURROUND,OPT_WAVEX_BFORMAT,OPT_WAVEX_5_0,OPT_WAVEX_7_1,OPT_WAVEX_CUBE,OPT_WAVEX_6_1,};
-/*
-The number of channels defines the order of the soundfield:
- 3 channel = h   = 1st order horizontal
- 4 channel = f   = 1st order 3-D
- 5 channel = hh  = 2nd order horizontal
- 6 channel = fh  = 2nd order horizontal + 1st order height (formerly
-called 2.5 order)
- 7 channel = hhh = 3rd order horizontal
- 8 channel = fhh = 3rd order horizontal + 1st order height
- 9 channel = ff  = 2nd order 3-D
-11 channel = ffh = 3rd order horizontal + 2nd order height
-16 channel = fff = 3rd order 3-D
-*/
-#define N_BFORMATS (10)
-static const int bformats[N_BFORMATS] = {2,3,4,5,6,7,8,9,11,16};
-
-int get_sampsize(psf_stype type)
-{
-    int size = 0;
-    switch(type){
-		case PSF_SAMP_8:
-			size = 1;
-			break;
-		case PSF_SAMP_16:
-			size = 2;
-			break;
-		case PSF_SAMP_24:
-			size = 3;
-			break;
-		case PSF_SAMP_32:
-		case PSF_SAMP_IEEE_FLOAT:
-			size = 4;
-			break;
-		default:
-			break;
-    }
-    return size;
-}
-
-
-
-void
-usage()
-{
-	fprintf(stderr,"\n\nCDP MCTOOLS: COPYSFX  (c) RWD,CDP %s\n",VERSION);
-	fprintf(stderr,"\tcopy/convert soundfiles\n");
-	fprintf(stderr,"usage: copysfx [-d][-h][-sN] [-tN] infile outfile\n");
-	fprintf(stderr,"-d   : apply TPDF dither to 16bit outfile.\n"
-                   "-s   : force output sample type to type N.\n");
-	fprintf(stderr,"      Available sample types:\n"
-                   "      1  : 16bit integers (shorts)\n"
-                   "      2  : 32bit integer  (longs)\n"
-                   "      3  : 32bit floating-point\n"
-                   "      4  : 24bit integer 'packed' \n"
-                   "      Default: format of infile.\n");
-				   
-	fprintf(stderr,"-h   : write mimimum header (no extra data).\n"
-				   "       Default: include PEAK data.\n"
-            );
-/*	fprintf(stderr,"-i   : interpret 32bit files as floats (use with care!)\n"); */
-	fprintf(stderr,"-tN  : write outfile format as type N\n");
-	fprintf(stderr,"Possible formats:\n"
-					"0   : standard soundfile (.wav, .aif, .afc, .aifc)\n"
-					"1   : generic WAVE_EX (no speaker assignments)\n"
-					"2   : WAVE_EX mono/stereo/quad(LF,RF,LR,RR) - infile nchans must match\n"
-					"3   : WAVE_EX quad surround (L,C,R,S)       - infile must be quad\n"
-					"4   : WAVE_EX 5.1 format surround           - infile must be 6-channel\n"
-					"5   : WAVE_EX Ambisonic B-format (W,X,Y,Z...) - file extension .amb supported \n"
-                    "6   : WAVE_EX 5.0 Surround                  - infile must be 5-channel\n"
-                    "7   : WAVE_EX 7.1 Surround                  - infile must be 8-channel\n"
-                    "8   : WAVE_EX Cube Surround                 - infile must be 8-channel\n"
-                    "9   : WAVE_EX 6.1 Surround  (new in v2.1.0) - infile must be 7-channel\n"
-					"default in all cases: outfile has format of infile\n"
-					"NB: types 1 to 9 are for WAV format only\n"
-                    "See also CHXFORMAT: destructively change WAVE_EX GUID and/or speaker mask.\n"
-					);	
-}
-
-
-
-int
-main(int argc, char *argv[])
-{
-	unsigned long size,i;
-	unsigned long channels,srate;
-	//RWD.6.5.99
-	PSF_PROPS  inprops,outprops; 
-	int force_wave = 0,rc = 0;
-    psf_stype force_stype = 0;
-	int opt_wave_type = -1;
-	int min_header = 0;
-	int interpret_floats = 0;
-	int res;	
-	char *create_msg = NULL;
-    int do_dither = 0;
-    psf_format  outformat = PSF_FMT_UNKNOWN;
-	unsigned long update_size;
-	PSF_CHPEAK  *peaks = NULL;
-	MYLONG peaktime;
-	char* ext = NULL;
-	int have_amb_ext = 0;
-    int ifd;		/* the input file */
-    int ofd;		/* the output file */
-    float *sampleframe = NULL;
-	double outsize_bytes;
-	double maxsize = pow(2.0,32.0) - 200.0;  /* some safety margin! */
-	
-    /* CDP version number */
-    if(argc==2 && (stricmp(argv[1],"--version")==0)){
-        printf("2.0.3\n");
-        return 0;
-    }
-
-	if(psf_init()) {		
-		printf("failed to init psfsys\n");
-		exit(1);
-	}
-
-	if(argc < 3){
-		usage();
-		exit(1);
-	}
-
-	printf("\nCDP MCTOOLS: COPYSFX v2.0.3 (c) RWD,CDP 2011\n");
-
-	while(argv[1][0]=='-'){
-		int sflag = 0;
-		switch(argv[1][1]){
-        case 'd':
-            do_dither = 1;
-            break;
-		case('t'):
-			if(argv[1][2]=='\0'){
-				fprintf(stderr,"-t flag requires parameter\n");
-				usage();
-				exit(1);
-			}
-			opt_wave_type = atoi(&(argv[1][2]));
-			if((opt_wave_type < 0) || opt_wave_type > OPT_WAVEX_CUBE){
-				fprintf(stderr,"wave type out of range\n");
-				usage();
-				exit(1);
-			}
-			force_wave = 1;
-			break;		
-		case('s'):
-			if(argv[1][2]=='\0'){
-				fprintf(stderr,"-s flag requires parameter\n");
-				usage();
-				exit(1);
-			}
-			if(force_stype >0){
-				fprintf(stderr,"copysfx: error: -s flag may only be used once!\n");
-				usage();
-				exit(1);
-			}
-			sflag = atoi(&(argv[1][2]));
-			if(sflag < 1 || sflag > 6){
-				fprintf(stderr,"-s parameter out of range\n");
-				usage();
-				exit(1);
-			}
-			switch(sflag){
-			case(1):
-				force_stype = PSF_SAMP_16;
-				break;
-			case(2):
-				force_stype = PSF_SAMP_32;
-				break;
-			case(3):
-				force_stype = PSF_SAMP_IEEE_FLOAT;
-				break;
-			case(4):
-				force_stype = PSF_SAMP_24;
-				break;			
-			default:
-				fprintf(stderr,"internal error: unmatched sampletype parameter!\n");
-				break;
-			}
-			break;
-		case('h'):
-            min_header = 1;    
-			break;
-#ifdef NOTDEF
-		case('i'):
-			interpret_floats = 1;
-			break;
-#endif
-		default:
-			fprintf(stderr,"unknown flag option %s\n",argv[1]);
-			usage();
-			exit(1);
-			
-		}
-		argc--; argv++;
-	}
-
-	if(argc < 3){
-		usage();
-		exit(1);
-	}
-
-	if((ifd = psf_sndOpen(argv[1],&inprops,0)) < 0) {
-		fprintf(stderr, "copysfx: can't open input SFfile %s:\n\t",
-				argv[1]);
-		exit(1);
-	}
-		
-	size = psf_sndSize(ifd);
-	channels = inprops.chans;
-	sampleframe = (float *) malloc(channels * sizeof(float));
-	if(sampleframe==NULL){
-		psf_sndClose(ifd);
-		puts("no memory for sample buffer\n");
-		exit(1);
-	}
-	
-	srate = inprops.srate;
-	update_size = (unsigned long)((double)srate * 0.5);
-	outprops = inprops;
-	/* if -i flag used, infile is floats, therefore we write outfile as floats,
-	 * UNLESS we also have outtype coercion!
-	 */
-	if(interpret_floats)
-		outprops.samptype = PSF_SAMP_IEEE_FLOAT;
-
-	if(inprops.format== PSF_WAVE_EX){		
-		printf("opened WAVE-EX file:\n");
-		switch(inprops.chformat){
-		
-		case(MC_STD):
-			printf("\tno speaker assignments defined\n");		
-			break;
-		case(MC_MONO):
-			printf("\tMONO\n");
-			break;
-		case(MC_STEREO):
-			printf("\tSTEREO\n");
-			break;		
-		case(MC_QUAD):
-			printf("\tGeneric quad format: LF-RF-LR-RR\n");			
-			break;
-		case(MC_LCRS):
-			printf("\tQuad Surround: L-C-R-S format\n");			
-			break;
-		case(MC_BFMT):
-			printf("\tAmbisonic B-Format\n");
-			break;
-		case(MC_DOLBY_5_1):
-			printf("\t5.1 surround\n");
-			break;
-        case(MC_SURR_5_0):
-			printf("\t5.0 surround\n");
-			break;
-		case(MC_SURR_6_1):
-            printf("\t6.1 Surround\n");
-            break;
-        case(MC_SURR_7_1):
-            printf("\t7.1 Surround\n");
-            break;
-        case(MC_CUBE):
-            printf("\tCube Surround\n");
-            break;
-		default:
-			printf("\tunrecognized speaker positions\n");
-			break;						
-		}
-		if(min_header==1)
-			printf("WARNING: minimum header is not recommended for WAVE_EX files\n");
-	}
-
-	peaks = (PSF_CHPEAK *) calloc(channels,sizeof(PSF_CHPEAK));
-	if(peaks==NULL){
-		puts("no memory for fpeak data buffer\n");
-		psf_sndClose(ifd);
-		exit(1);
-	}
-
-	//read infile peak data, if it exists, and report to user	
-	res = psf_sndReadPeaks(ifd,peaks,  &peaktime);
-	if(res ==0)	{
-		printf("no peak data in this infile\n");
-	}
-	else if(res < 0){
-		fprintf(stderr,"error reading infile peak data\n");
-		psf_sndClose(ifd);
-		exit(1);
-	}
-	else {
-		time_t t_peaktime = (time_t) peaktime;
-		printf("Infile PEAK data:\n\tcreation time: %s\n", ctime(&t_peaktime));
-		for(i=0;i < channels; i++)
-#ifdef CPLONG64
-			printf("CH %ld: %.4lf at frame %u:\t%.4lf secs\n",
-#else
-            printf("(32) CH %ld: %.4lf at frame %lu:\t%.4lf secs\n",
-#endif
-				i,peaks[i].val,peaks[i].pos,(double) (peaks[i].pos) / (double)srate);
-	}
-	
-	if(force_stype > PSF_SAMP_UNKNOWN)		
-		outprops.samptype = force_stype;
-
-	outsize_bytes = (double) size * outprops.chans * get_sampsize(outprops.samptype);
-    if(outsize_bytes > maxsize){
-        printf("output file size %.0f MB exceeds 4GB: cannot proceed.\n",outsize_bytes / 1048576.0);
-        exit(1);
-    }
-	
-	if(force_wave){
-		int i,matched;
-		//check file extension...
-		switch(opt_wave_type){
-		case(OPT_STD_WAVE):
-			outprops.chformat = STDWAVE;
-			outprops.format = PSF_STDWAVE;
-			create_msg = "creating standard WAVE file\n";
-			break;
-		case(OPT_WAVEX_GENERIC):
-			inprops.chformat = MC_STD;
-			outprops.format = PSF_WAVE_EX;
-			create_msg = "creating STD WAVE_EX file\n";
-			break;
-		case(OPT_WAVEX):
-			switch(inprops.chans){
-			case(1):
-				outprops.chformat = MC_MONO;				
-				outprops.format = PSF_WAVE_EX;
-				create_msg = "creating MONO WAVE_EX file\n";
-				break;
-			case(2):
-				outprops.chformat = MC_STEREO;				
-				outprops.format = PSF_WAVE_EX;
-				create_msg = "creating STEREO WAVE_EX file\n";
-				break;
-			case(4):
-				outprops.chformat = MC_QUAD;			
-				outprops.format = PSF_WAVE_EX;
-				create_msg = "creating QUAD WAVE_EX file\n";
-				break;
-			default:
-				fprintf(stderr,"infile nchans incompatible with requested WAVE-EX format\n");
-				usage();
-				psf_sndClose(ifd);
-				if(peaks)
-					free(peaks);
-				exit(1);				
-			}
-			break;
-		case(OPT_WAVEX_LCRS):
-			if(inprops.chans != 4){
-				fprintf(stderr,"infile must have four channels\n");
-				usage();
-				psf_sndClose(ifd);
-				if(peaks)
-					free(peaks);
-				exit(1);
-			}
-			outprops.chformat = MC_LCRS;
-			outprops.format = PSF_WAVE_EX;
-			create_msg = "creating LCRS-surround WAVE_EX file\n";
-			break;
-		case(OPT_WAVEX_SURROUND):
-			if(inprops.chans != 6){
-				fprintf(stderr,"infile must have six channels\n");
-				usage();
-				psf_sndClose(ifd);
-				if(peaks)
-					free(peaks);
-				exit(1);
-			}
-			outprops.chformat = MC_DOLBY_5_1;			
-			outprops.format = PSF_WAVE_EX;
-			create_msg = "creating 5.1 surround WAVE_EX file\n";
-			break;			
-		case(OPT_WAVEX_BFORMAT):
-			matched = 0;
-			for(i=0;i < N_BFORMATS;i++)	{
-				if(inprops.chans == bformats[i]){
-					matched = 1;
-					break;
-				}
-			}
-			if(!matched){
-				printf("WARNING: No Bformat definition for %d-channel file.\n",inprops.chans);
-			}
-			outprops.chformat = MC_BFMT;			
-			outprops.format = inprops.format = PSF_WAVE_EX;
-			create_msg = "creating AMBISONIC B-FORMAT WAVE_EX file\n";
-			break;
-        case(OPT_WAVEX_5_0):
-            if(inprops.chans != 5){
-				fprintf(stderr,"infile must have five channels\n");
-				usage();
-				psf_sndClose(ifd);
-				if(peaks)
-					free(peaks);
-				exit(1);
-			}
-			outprops.chformat = MC_SURR_5_0;			
-			outprops.format = PSF_WAVE_EX;
-			create_msg = "creating 5.0 surround WAVE_EX file\n";
-			break;
-				
-        case(OPT_WAVEX_7_1):
-            if(inprops.chans != 8){
-				fprintf(stderr,"infile must have eight channels\n");
-				usage();
-				psf_sndClose(ifd);
-				if(peaks)
-					free(peaks);
-				exit(1);
-			}
-			outprops.chformat = MC_SURR_7_1;			
-			outprops.format = PSF_WAVE_EX;
-			create_msg = "creating 7.1 surround WAVE_EX file\n";
-			break;
-        case OPT_WAVEX_CUBE:
-            if(inprops.chans != 8){
-				fprintf(stderr,"infile must have eight channels\n");
-				usage();
-				psf_sndClose(ifd);
-				if(peaks)
-					free(peaks);
-				exit(1);
-			}
-			outprops.chformat = MC_CUBE;			
-			outprops.format = PSF_WAVE_EX;
-			create_msg = "creating 5.0 surround WAVE_EX file\n";
-			break;			
-        case(OPT_WAVEX_6_1):
-            if(inprops.chans != 7){
-				fprintf(stderr,"infile must have seven channels\n");
-				usage();
-				psf_sndClose(ifd);
-				if(peaks)
-					free(peaks);
-				exit(1);
-			}
-			outprops.chformat = MC_SURR_6_1;			
-			outprops.format = PSF_WAVE_EX;
-			create_msg = "creating 6.1 surround WAVE_EX file\n";
-			break;	    
-		default:
-			 printf("copysfx: Program error: impossible wave_ex type\n");
-			 psf_sndClose(ifd);
-				if(peaks)
-					free(peaks);				
-			 exit(1);
-			 
-		}
-	}
-	//ignore all that if user wants aiff!
-	//informat = inprops.format;
-	ext = strrchr(argv[2],'.');
-	if(ext && stricmp(ext,".amb")==0)
-		have_amb_ext = 1;
-
-	if(have_amb_ext){
-		if(!(outprops.format == PSF_WAVE_EX && outprops.chformat == MC_BFMT)){
-			fprintf(stderr,"Error: .amb extension only allowed for WAVE_EX B-Format file.\n");
-			exit(1);			
-		} 
-	}
-    outformat  = psf_getFormatExt(argv[2]);
-
-	if((ofd = psf_sndCreate(argv[2],&outprops,0,min_header,PSF_CREATE_RDWR)) < 0){	
-		fprintf(stderr, "copysfx: can't create output file %s:\n\t",argv[2]);		
-		psf_sndClose(ifd);
-		fprintf(stderr,"\n");
-		if(peaks)
-			free(peaks);				
-		exit(1);
-	}
-    if(do_dither)
-        psf_sndSetDither(ofd,PSF_DITHER_TPDF);
-
-	if(force_wave){
-		if(outprops.format==PSF_WAVE_EX){
-			if(outformat > PSF_WAVE_EX)
-                printf("WARNING: extended formats require .wav file format:\n\t - creating standard file\n");		
-			else
-			    printf("%s\n",create_msg);	
-		}
-	}	
-	printf("copying...\n");    
-    for(i=0;i < size;i++){
-        /* salve to CEP users: need interpret_floats somewhere...? */
-        if((psf_sndReadFloatFrames(ifd, sampleframe, 1)) != 1){
-            
-            fprintf(stderr,"copysfx: error reading from infile\n");
-            psf_sndClose(ifd);
-            psf_sndClose(ofd);
-            free(sampleframe);
-            if(peaks)
-                free(peaks);
-            exit(1);
-        }
-        if(psf_sndWriteFloatFrames(ofd, sampleframe,1)!=1){
-            fprintf(stderr,"copysfx: error writing to outfile\n");
-            psf_sndClose(ifd);
-            psf_sndClose(ofd);
-            free(sampleframe);
-            if(peaks)
-                free(peaks);
-            exit(1);			
-        }
-        if((i / channels) % update_size == 0) {	 
-            printf("%.2lf secs\r",(double) (i / channels) / (double) srate);
-            fflush(stdout);
-        }
-    }
-    printf("%.4lf secs\r",(double) (i / channels) / (double) srate);
-    	
-	if(psf_sndReadPeaks( ofd,peaks,&peaktime)){
-		printf("Outfile PEAK values:\n");
-		for(i=0; i < (unsigned long) inprops.chans; i++){
-            double val, dbval;
-            val = (double) peaks[i].val;
-            if(val > 0.0){
-                dbval = 20.0 * log10(val);
-#ifdef CPLONG64
-                printf("CH %ld: %.6f (%.2lfdB) at frame %u:\t%.4f secs\n",i,
-#else
-                printf("CH %ld: %.6f (%.2lfdB) at frame %lu:\t%.4f secs\n",i,
-#endif
-                       val,dbval,peaks[i].pos,(double)peaks[i].pos / (double) inprops.srate);
-            }
-            else{
-#ifdef CPLONG64
-                printf("CH %ld: %.6f (-infdB) at frame %u:\t%.4f secs\n",i,
-#else        
-                printf("CH %ld: %.6f (-infdB) at frame %lu:\t%.4f secs\n",i,
-#endif
-                       val,peaks[i].pos,(double)peaks[i].pos / (double) inprops.srate); 
-            }
-        }
-	}
-	if(psf_sndClose(ifd) < 0) {		
-		rc++;
-	}
-
-	if(psf_sndClose(ofd) < 0) {		
-		rc++;
-	}
-    if(sampleframe)
-		free(sampleframe);
-	if(peaks)
-		free(peaks);	
-	psf_finish();
-	return rc;
-}
-
-
-

dev/externals/mctools/fmdcode.c

@@ -1,400 +0,0 @@
-/*
- * Copyright (c) 1983-2013 Richard Dobson and Composers Desktop Project Ltd
- * http://people.bath.ac.uk/masrwd
- * http://www.composersdesktop.com
- * This file is part of the CDP System.
- * The CDP System is free software; you can redistribute it
- * and/or modify it under the terms of the GNU Lesser General Public 
- * License as published by the Free Software Foundation; either 
- * version 2.1 of the License, or (at your option) any later version. 
- *
- * The CDP System is distributed in the hope that it will be useful, 
- * but WITHOUT ANY WARRANTY; without even the implied warranty of 
- * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  
- * See the GNU Lesser General Public License for more details.
- * You should have received a copy of the GNU Lesser General Public
- * License along with the CDP System; if not, write to the Free Software
- * Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
- *
- */
- 
-//fmdcode.c  : decode .amb file to various speaker layouts
-#include <stdio.h>
-#include <stdlib.h>
-#include <string.h>
-#include <math.h>
-#include <memory.h>
-#include <sys/types.h>
-#include <sys/timeb.h>
-#include <portsf.h>
-#include "fmdcode.h"
-
-#ifdef unix
-/* in portsf.lib */
-extern int stricmp(const char *a, const char *b);
-#endif
-
-/*
- Channel order is WXYZ,RSTUV,KLMNOPQ
- 
- The number of channels defines the order of the soundfield:
- 2 channel = W+Y =  "Mid-Side" 
- 3 channel = h   = 1st order horizontal
- 4 channel = f   = 1st order 3-D
- 5 channel = hh  = 2nd order horizontal
- 6 channel = fh  = 2nd order horizontal + 1st order height (formerly
-                                                            called 2.5 order)
- 7 channel = hhh = 3rd order horizontal
- 8 channel = fhh = 3rd order horizontal + 1st order height
- 9 channel = ff  = 2nd order 3-D
- 11 channel = ffh = 3rd order horizontal + 2nd order height
- 16 channel = fff = 3rd order 3-D
- 
- 
- Horizontal   Height  Soundfield   Number of    Channels
- order 	      order 	  type      channels 	
- 1 	         0 	       horizontal 	  3 	    WXY
- 1 	         1 	      full-sphere 	  4 	    WXYZ
- 2 	         0 	       horizontal 	  5 	    WXY....UV
- 2 	         1 	       mixed-order    6 	    WXYZ...UV
- 2 	         2 	      full-sphere     9 	    WXYZRSTUV
- 3           0 	       horizontal 	  7 	    WXY....UV.....PQ
- 3 	         1         mixed-order 	  8 	    WXYZ...UV.....PQ
- 3 	         2 	       mixed-order 	 11 	    WXYZRSTUV.....PQ
- 3 	         3 	      full-sphere 	 16 	    WXYZRSTUVKLMNOPQ
-
- */
-
-
-enum {ARG_PROGNAME, ARG_INFILE,ARG_OUTFILE, ARG_LAYOUT,ARG_NARGS};
-#define N_BFORMATS (10)
-enum {FM_MONO,FM_STEREO,FM_SQUARE,FM_QUAD,FM_PENT,DM_5_0,DM_5_1,FM_HEX,FM_OCT1,FM_OCT2,FM_CUBE,FM_QUADCUBE,FM_NLAYOUTS};
-
-
-//static const int bformats[N_BFORMATS] = {2,3,4,5,6,7,8,9,11,16};
-static const int layout_chans[] = {1,2,4,4,5,5,6,6,8,8,8,8};
-
-void usage(void)
-{
-   printf(
-    "usage: fmdcode [-x][-w] infile outfile layout\n"
-	"       -w    :   write plain WAVE outfile format\n"
-    "                (.wav default - use generic wavex format).\n"
-    "       -x    : write std WAVEX speaker positions to header\n"
-    "               (applies to compatible layouts only; requires .wav extension).\n"
-	"   layout    : one of the choices below.\n"
-    "       Output channel order is anticlockwise from centre front\n"
-    "            except where indicated.\n"
-    "   Layouts indicated with * are compatible with WAVEX speaker position order. \n"
-    "   Available speaker layouts:\n"
-    "   1    :  *  mono (= W signal only)\n"
-    "   2    :  *  stereo (quasi mid/side, = W +- Y)\n"
-    "   3    :     square\n"
-    "   4    :  *  quad FL,FR,RL,RR order\n"
-    "   5    :     pentagon\n"
-    "   6    :  *  5.0 surround (WAVEX order)\n"
-    "   7    :  *  5.1 surround (WAVEX order, silent LFE)\n"
-    "   8    :     hexagon\n"
-    "   9    :     octagon 1 (front pair, 45deg)\n"
-    "   10   :     octagon 2 (front centre speaker)\n"
-    "   11   :     cube (as 3, low-high interleaved. Csound-compatible.)\n"
-    "   12   :  *  cube (as 4, low quad followed by high quad).\n"
-    " NOTE: no shelf filters or NF compensation used.\n");
-}
-
-int main(int argc,char *argv[])
-{
-	int i,ifd, ofd;
-	int layout,inorder = 1;
-    int got,halfsec;
-	unsigned int framepos;
-    int inchans,outchans;
-    int outsize;
-    int write_speakerpositions = 0;
-    MYLONG peaktime;
-    psf_channelformat chformat = MC_STD;
-    psf_format  outformat;
-	char *sfname;
-    float *frame = NULL;
-    fmhcopyfunc copyfunc;
-    fmhdecodefunc decodefunc = NULL;
-	int write_wavex = 1;
-	ABFSAMPLE abfsample;
-	PSF_PROPS props;
-	PSF_CHPEAK *peaks = NULL;
-	float abfframe[16];
-
-    /* CDP version number */
-    if(argc==2 && (stricmp(argv[1],"--version")==0)){
-        printf("1.0b\n");
-        return 0;
-    }
-    
-	if(argc < 3){
-		printf("\nCDP MCTOOLS: FMDCODE v 1.0beta: RWD,CDP 2009\n"
-				"Plain multi-layout decoder for .amb files.\n"
-                "Regular layouts use standard Furse-Malham in-phase coefficients.\n" 
-                "5.x surround coefficients (maxre) from David Moore.\n");
-        usage();
-		return 1;
-	}
-	while(argv[1][0] =='-'){		
-		switch(argv[1][1]){
-		case('w'):
-			write_wavex = 0;
-			break;
-        case 'x':
-            write_speakerpositions = 1;
-            break;
-		default:
-			fprintf(stderr,"fmdcode: error: illegal flag option %s\n",argv[1]);
-			return 1;
-		}
-		argc--; argv++;
-	}
-	if(argc < ARG_NARGS){
-        
-		usage();
-		return 1;
-	}
-	if(psf_init()) {		
-		printf("failed to init psfsys\n");
-		exit(1);
-	}
-	
-	sfname = argv[ARG_OUTFILE];
-	layout = atoi(argv[ARG_LAYOUT]);
-    if(layout < 1 || layout > FM_NLAYOUTS+1){
-        printf("Unsupported layout type.\n");
-        return 1;
-    }
-    
-    
-	ifd = psf_sndOpen(argv[ARG_INFILE],&props,0);
-	if(ifd < 0){
-		fprintf(stderr,"unable toopen infile %s\n",argv[ARG_INFILE]);
-		return 1;
-	}
-    inchans = props.chans;
-    if(inchans > 4) {
-        inorder = 2;
-        printf("%d-channel input: performing 2nd-order decode.\n",inchans);
-    }
-    outsize = psf_sndSize(ifd);
-	if(outsize <= 0){
-		fprintf(stderr,"fmdcode: infile is empty!\n");
-		psf_sndClose(ifd);
-		return 1;
-	}
-    
-    switch(inchans){
-	case 3:
-        copyfunc = fmhcopy_3;
-        break;
-    case 4:
-        copyfunc = fmhcopy_4;       
-        break;
-    case 5:
-        copyfunc = fmhcopy_5;
-        break;
-    case 6:
-        copyfunc = fmhcopy_6;
-        break;
-    case 7:
-        copyfunc = fmhcopy_7;
-        break;
-    case 8:
-        copyfunc = fmhcopy_8;
-        break;
-    case 9:
-        copyfunc = fmhcopy_9;
-        break;
-    case 11:
-        copyfunc = fmhcopy_11;
-        break;
-    case 16:
-        copyfunc = fmhcopy_16;
-        break;
-    default:
-        printf("file has unsupported number of channels (%d)\n",inchans);
-        psf_sndClose(ifd);
-        return 1;
-	}
-    //FM_MONO,FM_STEREO,FM_SQUARE,FM_PENT,FM_SURR,FM_SURR6,FM_HEX,FM_OCT1,FM_OCT2,FM_CUBE
-    switch(layout-1){
-        case FM_MONO:
-            printf("Decoding to Mono\n");
-            decodefunc = fm_i1_mono;
-            if(write_wavex && write_speakerpositions)
-                chformat = MC_MONO;
-            break;
-        case FM_STEREO:
-            printf("Decoding to Stereo\n");
-            decodefunc = fm_i1_stereo;
-            if(write_wavex && write_speakerpositions)
-                chformat = MC_STEREO;
-            break;
-        case FM_SQUARE:
-            printf("Decoding to Square\n");
-            if(inorder == 1)
-               decodefunc = fm_i1_square;
-            else
-               decodefunc = fm_i2_square; 
-            break;
-        case FM_QUAD:
-            printf("Decoding to quad surround (WAVEX order)\n");
-            if(inorder == 1)
-                decodefunc = fm_i1_quad;
-            else
-                decodefunc = fm_i2_quad;
-            if(write_wavex && write_speakerpositions)
-                chformat = MC_QUAD;
-            break;
-        case FM_PENT:
-            printf("Decoding to pentagon\n");
-            if(inorder==1)
-                decodefunc = fm_i1_pent;
-            else
-                decodefunc = fm_i2_pent;
-            break;
-
-        case DM_5_0:
-            printf("Decoding to 5.0 surround (David Moore)\n");
-            if(inorder==1)
-                decodefunc = dm_i1_surr;
-            else
-                decodefunc = dm_i2_surr;
-            if(write_wavex && write_speakerpositions)
-                chformat = MC_SURR_5_0;
-            break;
-        case DM_5_1:
-            printf("Decoding to  5.1 surround (David Moore)\n");
-            if(inorder==1)
-                decodefunc = dm_i1_surr6;
-            else
-                decodefunc = dm_i2_surr6;
-            if(write_wavex && write_speakerpositions)
-                chformat = MC_DOLBY_5_1;
-            break;
-        case FM_HEX:
-            printf("Decoding to Hexagon\n");
-            if(inorder==1)
-                decodefunc = fm_i1_hex;
-            else
-                decodefunc = fm_i2_hex;
-            break;
-        case FM_OCT1:
-            printf("Decoding to Octagon 1\n");
-            if(inorder==1)
-                decodefunc = fm_i1_oct1;
-            else
-                decodefunc = fm_i2_oct1;
-            break;
-        case FM_OCT2:
-            printf("Decoding to Octagon 2\n");
-            if(inorder==1)
-                decodefunc = fm_i1_oct2;
-            else
-                decodefunc = fm_i2_oct2;
-            break;
-        case FM_CUBE:
-            printf("Decoding to Cube (FM interleaved)\n");
-            if(inorder==1)
-                decodefunc = fm_i1_cube;
-            else
-                decodefunc = fm_i2_cube;
-            break; 
-        case FM_QUADCUBE:
-            printf("Decoding to Octagon 1 (WAVEX order)\n");
-            if(inorder==1)
-                decodefunc = fm_i1_cubex;
-            else
-                decodefunc = fm_i2_cubex;
-            if(write_wavex && write_speakerpositions)
-                chformat = MC_CUBE;
-            break;
-    }
-    outformat = psf_getFormatExt(sfname);
-    if(outformat >= PSF_AIFF){
-        if(write_speakerpositions)
-            printf("Warning: -x requires .wav format\n");
-    }
-    outchans = layout_chans[layout-1];
-    frame = malloc(sizeof(float) * outchans);
-    if(frame==NULL){
-        puts("No Memory!\n");
-        return 1;
-    }
-	props.chformat = STDWAVE;
-	props.chans = outchans;
-    if(!is_legalsize(outsize,&props)){
-        fprintf(stderr,"error: outfile size exceeds capacity of format.\n");
-        return 1;
-    }
-    
-    /*TODO: set speaker pos when we can */
-	if(write_wavex){
-		props.chformat = chformat;
-		props.format = PSF_WAVE_EX;
-	}
-
-	ofd = psf_sndCreate(sfname,&props,0,0,PSF_CREATE_RDWR);
-	if(ofd < 0){
-		fprintf(stderr,"can't create outfile %s\n",sfname);
-		psf_sndClose(ifd);
-		return 1;
-	}
-    peaks = (PSF_CHPEAK*)  malloc(sizeof(PSF_CHPEAK) * outchans);
-    memset(peaks,0,sizeof(PSF_CHPEAK) * outchans);
-
-	halfsec = props.srate / 2;
-	framepos = 0;
-	printf("\ndecoding:\n");
-	while((got = psf_sndReadFloatFrames(ifd,abfframe,1))==1){
-        memset(&abfsample,0,sizeof(ABFSAMPLE));
-        copyfunc(&abfsample,abfframe);
-        decodefunc(&abfsample,frame,1);            
-		if(0 > psf_sndWriteFloatFrames(ofd,frame,1)){
-			fprintf(stderr,"error writing to outfile\n");
-			psf_sndClose(ifd);
-			psf_sndClose(ofd);
-			return 1;
-		}
-
-		if((framepos % halfsec) == 0){
-			printf("%.2lf secs\r",(double) framepos / (double) props.srate);
-            fflush(stdout);
-        }
-		framepos++;
-	}
-
-	if(got != 0){
-		fprintf(stderr,"warning: not all data was read\n");
-	}
-	printf("\n%.4lf secs\nWritten %d frames to %s\n",(double)framepos / (double) props.srate,framepos,sfname);
-	
-    if(psf_sndReadPeaks( ofd,peaks,&peaktime)){
-        printf("PEAK values:\n");
-        for(i=0; i < outchans; i++){
-            double val, dbval;
-            val = (double) peaks[i].val;
-            
-            if(val > 0.0){
-                dbval = 20.0 * log10(val);
-                printf("CH %d: %.6f (%.2lfdB) at frame %u:\t%.4f secs\n",i,
-                       val,dbval,(unsigned int)peaks[i].pos,(double)peaks[i].pos / (double) props.srate);
-            }
-            else{
-                printf("CH %d: %.6f (-infdB) at frame %u:\t%.4f secs\n",i,
-                       val,(unsigned int)peaks[i].pos,(double)peaks[i].pos / (double) props.srate); 
-            }
-        }
-    }
-    printf("\n");
-	psf_sndClose(ifd);
-	psf_sndClose(ofd);
-    if(peaks)
-        free(peaks);
-	psf_finish();
-	return 0;
-}

dev/externals/mctools/fmdcode.h

@@ -1,106 +0,0 @@
-/*
- * Copyright (c) 1983-2013 Richard Dobson and Composers Desktop Project Ltd
- * http://people.bath.ac.uk/masrwd
- * http://www.composersdesktop.com
- * This file is part of the CDP System.
- * The CDP System is free software; you can redistribute it
- * and/or modify it under the terms of the GNU Lesser General Public 
- * License as published by the Free Software Foundation; either 
- * version 2.1 of the License, or (at your option) any later version. 
- *
- * The CDP System is distributed in the hope that it will be useful, 
- * but WITHOUT ANY WARRANTY; without even the implied warranty of 
- * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  
- * See the GNU Lesser General Public License for more details.
- * You should have received a copy of the GNU Lesser General Public
- * License along with the CDP System; if not, write to the Free Software
- * Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
- *
- */
- 
-/* fmdcode.h */
-
-/*
- Channel order is WXYZ,RSTUV,KLMNOPQ
- 
- The number of channels defines the order of the soundfield:
- 2 channel = UHJ 
- 3 channel = h   = 1st order horizontal
- 4 channel = f   = 1st order 3-D
- 5 channel = hh  = 2nd order horizontal
- 6 channel = fh  = 2nd order horizontal + 1st order height (formerly
-                                                            called 2.5 order)
- 7 channel = hhh = 3rd order horizontal
- 8 channel = fhh = 3rd order horizontal + 1st order height
- 9 channel = ff  = 2nd order 3-D
- 11 channel = ffh = 3rd order horizontal + 2nd order height
- 16 channel = fff = 3rd order 3-D
- 
- 
- Horizontal   Height  Soundfield   Number of    Channels
- order 	      order 	  type      channels 	
- 1 	         0 	       horizontal 	  3 	    WXY
- 1 	         1 	      full-sphere 	  4 	    WXYZ
- 2 	         0 	       horizontal 	  5 	    WXY....UV
- 2 	         1 	       mixed-order    6 	    WXYZ...UV
- 2 	         2 	      full-sphere     9 	    WXYZRSTUV
- 3           0 	       horizontal 	  7 	    WXY....UV.....PQ
- 3 	         1         mixed-order 	  8 	    WXYZ...UV.....PQ
- 3 	         2 	       mixed-order 	 11 	    WXYZRSTUV.....PQ
- 3 	         3 	      full-sphere 	 16 	    WXYZRSTUVKLMNOPQ
- */
-
-typedef struct abf_samp {
-	float W;
-	float X;
-	float Y;
-	float Z;
-    float R;
-    float S;
-    float T;
-	float U;
-    float V;
-} ABFSAMPLE;
-
-typedef void (*fmhcopyfunc)(ABFSAMPLE*,const float*);
-
-typedef void (*fmhdecodefunc)(const ABFSAMPLE*, float*,unsigned int);
-//void bfdcode4(float *inbuf,long numframes);
-void fmhcopy_3(ABFSAMPLE* abf,const float*buf);
-void fmhcopy_4(ABFSAMPLE* abf,const float*buf);
-void fmhcopy_5(ABFSAMPLE* abf,const float*buf);
-void fmhcopy_6(ABFSAMPLE* abf,const float*buf);
-void fmhcopy_7(ABFSAMPLE* abf,const float*buf);
-void fmhcopy_8(ABFSAMPLE* abf,const float*buf);
-void fmhcopy_9(ABFSAMPLE* abf,const float*buf);
-void fmhcopy_11(ABFSAMPLE* abf,const float*buf);
-void fmhcopy_16(ABFSAMPLE* abf,const float*buf);
-
-
-// i1 = inphase 1st order, etc
-void fm_i1_mono(const ABFSAMPLE *inbuf,float *outbuf,unsigned int numframes);
-void fm_i1_stereo(const ABFSAMPLE *inbuf,float *outbuf,unsigned int numframes);
-void fm_i1_square(const ABFSAMPLE *inbuf,float *outbuf,unsigned int numframes);
-void fm_i2_square(const ABFSAMPLE *inbuf,float *outbuf,unsigned int numframes);
-void fm_i1_quad(const ABFSAMPLE *inbuf,float *outbuf,unsigned int numframes);
-void fm_i2_quad(const ABFSAMPLE *inbuf,float *outbuf,unsigned int numframes);
-void fm_i1_pent(const ABFSAMPLE *inbuf,float *outbuf,unsigned int numframes);
-void fm_i2_pent(const ABFSAMPLE *inbuf,float *outbuf,unsigned int numframes);
-void fm_i1_surr(const ABFSAMPLE *inbuf,float *outbuf,unsigned int numframes);
-void fm_i2_surr(const ABFSAMPLE *inbuf,float*outbuf,unsigned int numframes);
-void fm_i1_surr6(const ABFSAMPLE *inbuf,float *outbuf,unsigned int numframes);
-void fm_i2_surr6(const ABFSAMPLE *inbuf,float*outbuf,unsigned int numframes);
-void dm_i1_surr(const ABFSAMPLE *inbuf,float*outbuf,unsigned int numframes);
-void dm_i1_surr6(const ABFSAMPLE *inbuf,float*outbuf,unsigned int numframes);
-void dm_i2_surr(const ABFSAMPLE *inbuf,float*outbuf,unsigned int numframes);
-void dm_i2_surr6(const ABFSAMPLE *inbuf,float*outbuf,unsigned int numframes);
-void fm_i1_hex(const ABFSAMPLE *inbuf,float *outbuf,unsigned int numframes);
-void fm_i2_hex(const ABFSAMPLE *inbuf,float *outbuf,unsigned int numframes);
-void fm_i1_oct1(const ABFSAMPLE *inbuf,float *outbuf,unsigned int numframes);
-void fm_i2_oct1(const ABFSAMPLE *inbuf,float *outbuf,unsigned int numframes);
-void fm_i1_oct2(const ABFSAMPLE *inbuf,float *outbuf,unsigned int numframes);
-void fm_i2_oct2(const ABFSAMPLE *inbuf,float *outbuf,unsigned int numframes);
-void fm_i1_cube(const ABFSAMPLE *inbuf,float *outbuf,unsigned int numframes);
-void fm_i2_cube(const ABFSAMPLE *inbuf,float *outbuf,unsigned int numframes);
-void fm_i1_cubex(const ABFSAMPLE *inbuf,float *outbuf,unsigned int numframes);
-void fm_i2_cubex(const ABFSAMPLE *inbuf,float *outbuf,unsigned int numframes);

dev/externals/mctools/fmhcube1.txt

@@ -1,32 +0,0 @@
-f left low	<0.5774,0.5774,-0.5774>		0.1768	0.1140	0.1140	-0.1140	0.0000	-0.0369	-0.0369	0.0000	0.0369
-f right	low 	<0.5774,-0.5774,-0.5774>	0.1768	0.1140	-0.1140	-0.1140	0.0000	-0.0369	0.0369	0.0000	-0.0369
-r right low	<-0.5774,-0.5774,-0.5774>	0.1768	-0.1140	-0.1140	-0.1140	0.0000	0.0369	0.0369	0.0000	0.0369
-r left low	<-0.5774,0.5774,-0.5774>	0.1768	-0.1140	0.1140	-0.1140	0.0000	0.0369	-0.0369	0.0000	-0.0369
-f left high	<0.5774,0.5774,0.5774>		0.1768	0.1140	0.1140	0.1140	0.0000	0.0369	0.0369	0.0000	0.0369
-f right high	<0.5774,-0.5774,0.5774>		0.1768	0.1140	-0.1140	0.1140	0.0000	0.0369	-0.0369	0.0000	-0.0369
-r right high	<-0.5774,-0.5774,0.5774>	0.1768	-0.1140	-0.1140	0.1140	0.0000	-0.0369	-0.0369	0.0000	0.0369
-r left high	<-0.5774,0.5774,0.5774>		0.1768	-0.1140	0.1140	0.1140	0.0000	-0.0369	0.0369	0.0000	-0.0369
-
-
-Csound order					W	  X	  Y       Z      R         S      T        U       V
-f left low	<0.5774,0.5774,-0.5774>		0.1768	 0.1140	 0.1140	-0.1140	0.0000	-0.0369	-0.0369	0.0000	 0.0369
-f left high	<0.5774,0.5774,0.5774>		0.1768	 0.1140	 0.1140	 0.1140	0.0000	 0.0369	 0.0369	0.0000	 0.0369
-r left low	<-0.5774,0.5774,-0.5774>	0.1768	-0.1140	 0.1140	-0.1140	0.0000	 0.0369	-0.0369	0.0000	-0.0369
-r left high	<-0.5774,0.5774,0.5774>		0.1768	-0.1140	 0.1140	 0.1140	0.0000	-0.0369	 0.0369	0.0000	-0.0369
-r right low	<-0.5774,-0.5774,-0.5774>	0.1768	-0.1140	-0.1140	-0.1140	0.0000	 0.0369	 0.0369	0.0000	 0.0369
-r right high	<-0.5774,-0.5774,0.5774>	0.1768	-0.1140	-0.1140	 0.1140	0.0000	-0.0369	-0.0369	0.0000	 0.0369
-f right	low 	<0.5774,-0.5774,-0.5774>	0.1768	 0.1140	-0.1140	-0.1140	0.0000	-0.0369	 0.0369	0.0000	-0.0369
-f right high	<0.5774,-0.5774,0.5774>		0.1768	 0.1140	-0.1140	 0.1140	0.0000	 0.0369	-0.0369	0.0000	-0.0369
-
-WAVEX ORDER, LOW + HIGH
-
-f left low	<0.5774,0.5774,-0.5774>		0.1768	 0.1140	 0.1140	-0.1140	0.0000	-0.0369	-0.0369	0.0000	 0.0369
-f right	low 	<0.5774,-0.5774,-0.5774>	0.1768	 0.1140	-0.1140	-0.1140	0.0000	-0.0369	 0.0369	0.0000	-0.0369
-r left low	<-0.5774,0.5774,-0.5774>	0.1768	-0.1140	 0.1140	-0.1140	0.0000	 0.0369	-0.0369	0.0000	-0.0369
-r right low	<-0.5774,-0.5774,-0.5774>	0.1768	-0.1140	-0.1140	-0.1140	0.0000	 0.0369	 0.0369	0.0000	 0.0369
-
-f left high	<0.5774,0.5774,0.5774>		0.1768	 0.1140	 0.1140	 0.1140	0.0000	 0.0369	 0.0369	0.0000	 0.0369
-f right high	<0.5774,-0.5774,0.5774>		0.1768	 0.1140	-0.1140	 0.1140	0.0000	 0.0369	-0.0369	0.0000	-0.0369
-r left high	<-0.5774,0.5774,0.5774>		0.1768	-0.1140	 0.1140	 0.1140	0.0000	-0.0369	 0.0369	0.0000	-0.0369
-r right high	<-0.5774,-0.5774,0.5774>	0.1768	-0.1140	-0.1140	 0.1140	0.0000	-0.0369	-0.0369	0.0000	 0.0369
-

dev/externals/mctools/fmhfuncs.c

@@ -1,712 +0,0 @@
-/*
- * Copyright (c) 1983-2013 Richard Dobson and Composers Desktop Project Ltd
- * http://people.bath.ac.uk/masrwd
- * http://www.composersdesktop.com
- * This file is part of the CDP System.
- * The CDP System is free software; you can redistribute it
- * and/or modify it under the terms of the GNU Lesser General Public 
- * License as published by the Free Software Foundation; either 
- * version 2.1 of the License, or (at your option) any later version. 
- *
- * The CDP System is distributed in the hope that it will be useful, 
- * but WITHOUT ANY WARRANTY; without even the implied warranty of 
- * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  
- * See the GNU Lesser General Public License for more details.
- * You should have received a copy of the GNU Lesser General Public
- * License along with the CDP System; if not, write to the Free Software
- * Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
- *
- */
- 
-/* fmhfuncs.c */
-#include "fmdcode.h"
-
-/* TODO: expand to handle numframes frames? */
-
-void fmhcopy_3(ABFSAMPLE* abf,const float*buf)
-{
-    abf->W = *buf++;
-    abf->X = *buf++;
-    abf->Y = *buf;
-}
-
-void fmhcopy_4(ABFSAMPLE* abf,const float*buf)
-{
-    abf->W = *buf++;
-    abf->X = *buf++;
-    abf->Y = *buf++;
-    abf->Z = *buf;
-}
-
-void fmhcopy_5(ABFSAMPLE* abf,const float*buf)
-{
-    abf->W = *buf++;
-    abf->X = *buf++;
-    abf->Y = *buf++;
-    abf->U = *buf++;
-    abf->V = *buf;
-}
-
-void fmhcopy_6(ABFSAMPLE* abf,const float*buf)
-{
-    abf->W = *buf++;
-    abf->X = *buf++;
-    abf->Y = *buf++;
-    abf->Z = *buf++;
-    abf->U = *buf++;
-    abf->V = *buf;
-}
-// following discard 3rd order chans
-void fmhcopy_7(ABFSAMPLE* abf,const float*buf)
-{
-    abf->W = *buf++;
-    abf->X = *buf++;
-    abf->Y = *buf++;
-    abf->U = *buf++;
-    abf->V = *buf;
-}
-
-void fmhcopy_8(ABFSAMPLE* abf,const float*buf)
-{
-    abf->W = *buf++;
-    abf->X = *buf++;
-    abf->Y = *buf++;
-    abf->Z = *buf++;
-    abf->U = *buf++;
-    abf->V = *buf;
-}
-// these identical for 2nd order horiz max, but may be expanded later!
-void fmhcopy_9(ABFSAMPLE* abf,const float*buf)
-{
-    abf->W = *buf++;
-    abf->X = *buf++;
-    abf->Y = *buf++;
-    abf->Z = *buf++;
-    abf->R = *buf++;
-    abf->S = *buf++;
-    abf->T = *buf++;
-    abf->U = *buf++;
-    abf->V = *buf;
-}
-
-void fmhcopy_11(ABFSAMPLE* abf,const float*buf)
-{
-    abf->W = *buf++;
-    abf->X = *buf++;
-    abf->Y = *buf++;
-    abf->Z = *buf++;
-    abf->R = *buf++;
-    abf->S = *buf++;
-    abf->T = *buf++;
-    abf->U = *buf++;
-    abf->V = *buf;
-}
-void fmhcopy_16(ABFSAMPLE* abf,const float*buf)
-{
-    abf->W = *buf++;
-    abf->X = *buf++;
-    abf->Y = *buf++;
-    abf->Z = *buf++;
-    abf->R = *buf++;
-    abf->S = *buf++;
-    abf->T = *buf++;
-    abf->U = *buf++;
-    abf->V = *buf;
-}
-
-/********** DECODE FUNCS *************/
-/* TODO: complete support for numframes > 1 */
-
-void fm_i1_mono(const ABFSAMPLE *inbuf,float *outbuf,unsigned int numframes)
-{
-    int i;	
-	float *p_out = outbuf;
-	double aw;
-    for(i=0;i < numframes; i++){
-		aw = (double) inbuf->W * 0.7071;
-		*p_out++ = (float) aw;  		  
-	}
-}
-
-void fm_i1_stereo(const ABFSAMPLE *inbuf,float *outbuf,unsigned int numframes)
-{
-    int i;	
-	float *p_out = outbuf;
-	double aw,ay;
-    for(i=0;i < numframes; i++){
-		aw = (double) inbuf->W * 0.7071;
-		ay = (double) inbuf->Y * 0.5;		
-		
-		*p_out++ = (float) (aw +  ay);  
-		*p_out++ = (float) (aw  - ay);  
-	}
-}
-
-void fm_i1_square(const ABFSAMPLE *inbuf,float *outbuf,unsigned int numframes)
-{
-	int i;	
-	float *p_out = outbuf;
-	double aw,ax,ay;
-	
-	for(i=0;i < numframes; i++){
-		aw = (double) inbuf->W * 0.35355;
-		ax = (double) inbuf->X * 0.17677;
-		ay = (double) inbuf->Y * 0.17677;		
-		
-		*p_out++ = (float) (aw + ax + ay);  //FL
-		*p_out++ = (float) (aw - ax + ay);  //RL
-		*p_out++ = (float) (aw - ax - ay);  //RR
-		*p_out++ = (float) (aw + ax - ay);  //FR
-	}
-}
-void fm_i2_square(const ABFSAMPLE *inbuf,float*outbuf,unsigned int numframes)
-{
-	int i;		
-	float *p_out = outbuf;	
-	double aw,ax,ay,av;
-	
-	for(i=0;i < numframes; i++){
-		aw = (double) inbuf->W * 0.3536;
-		ax = (double) inbuf->X * 0.2434;
-		ay = (double) inbuf->Y * 0.2434;		
-		av = (double) inbuf->V * 0.0964;
-		*p_out++ = (float) (aw + ax + ay + av);  //FL
-		*p_out++ = (float) (aw - ax + ay - av ); //RL
-		*p_out++ = (float) (aw - ax - ay + av);  //RR
-		*p_out++ = (float) (aw + ax - ay - av);  //FR
-	}
-}
-/* ditto, RLRL layout for WAVEX */
-void fm_i1_quad(const ABFSAMPLE *inbuf,float *outbuf,unsigned int numframes)
-{
-	int i;	
-	float *p_out = outbuf;
-	double aw,ax,ay;
-	
-	for(i=0;i < numframes; i++){
-		aw = (double) inbuf->W * 0.35355;
-		ax = (double) inbuf->X * 0.17677;
-		ay = (double) inbuf->Y * 0.17677;		
-		
-		*p_out++ = (float) (aw + ax + ay);  //FL
-        *p_out++ = (float) (aw + ax - ay);  //FR
-		*p_out++ = (float) (aw - ax + ay);  //RL
-		*p_out++ = (float) (aw - ax - ay);  //RR
-		
-	}
-}
-void fm_i2_quad(const ABFSAMPLE *inbuf,float*outbuf,unsigned int numframes)
-{
-	int i;		
-	float *p_out = outbuf;	
-	double aw,ax,ay,av;
-	
-	for(i=0;i < numframes; i++){
-		aw = (double) inbuf->W * 0.3536;
-		ax = (double) inbuf->X * 0.2434;
-		ay = (double) inbuf->Y * 0.2434;		
-		av = (double) inbuf->V * 0.0964;
-		*p_out++ = (float) (aw + ax + ay + av);  //FL
-        *p_out++ = (float) (aw + ax - ay - av);  //FR
-		*p_out++ = (float) (aw - ax + ay - av ); //RL
-		*p_out++ = (float) (aw - ax - ay + av);  //RR
-		
-	}
-}
-
-
-//front pair angle 72deg
-void fm_i1_pent(const ABFSAMPLE *inbuf,float*outbuf,unsigned int numframes)
-{
-	int i;	
-	float *p_out = outbuf;
-	double aw,ax,ay;
-	
-	for(i=0;i < numframes; i++){
-		aw = (double) inbuf->W * 0.2828;
-		ax = (double) inbuf->X;
-		ay = (double) inbuf->Y;		
-		
-		*p_out++ = (float) (aw + (ax*0.1618) + (ay*0.1176));  //FL
-		*p_out++ = (float) (aw - (ax*0.0618) + (ay*0.1902));  
-		*p_out++ = (float) (aw - (ax*0.2));  
-		*p_out++ = (float) (aw - (ax*0.0618) - (ay*0.1902));  
-        *p_out++ = (float) (aw + (ax*0.1618) - (ay*0.1176)); //FR
-	}
-}
-
-void fm_i2_pent(const ABFSAMPLE *inbuf,float*outbuf,unsigned int numframes)
-{
-	int i;	
-	float *p_out = outbuf;	
-	double aw,ax,ay,au,av;
-	
-	for(i=0;i < numframes; i++){
-		aw = (double) inbuf->W * 0.2828;
-		ax = (double) inbuf->X;
-		ay = (double) inbuf->Y;	
-        au = (double) inbuf->U;
-        av = (double) inbuf->V;
-		
-		*p_out++ = (float) (aw + (ax*0.2227) + (ay*0.1618) + (au*0.0238) + (av * 0.0733));  
-		*p_out++ = (float) (aw - (ax*0.0851) + (ay*0.2619) - (au*0.0624) - (av * 0.0453));  
-		*p_out++ = (float) (aw - (ax*0.2753)               + (au * 0.0771)              );  
-		*p_out++ = (float) (aw - (ax*0.0851) - (ay*0.2619) - (au*0.0624) + (av * 0.0453));  
-        *p_out++ = (float) (aw + (ax*0.2227) - (ay*0.1618) + (au*0.0238) - (av * 0.0733));
-	}
-}
-
-/* FMH only defines 1st order decode */ 
-void fm_i1_surr(const ABFSAMPLE *inbuf,float*outbuf,unsigned int numframes)
-{
-	int i;	
-	float *p_out = outbuf;
-	double aw,ax,ay;
-	
-	for(i=0;i < numframes; i++){
-		aw = (double) inbuf->W;
-		ax = (double) inbuf->X;
-		ay = (double) inbuf->Y;		
-		/* TODO: fix this order! */
-		*p_out++ = (float) ((aw * 0.169)  + (ax*0.0797) + (ay * 0.0891));   //L
-		*p_out++ = (float) ((aw * 0.1635) + (ax*0.0923));                   //C    ///???
-		*p_out++ = (float) ((aw * 0.169)  - (ax*0.0797) - (ay * 0.0891));   //R    ///????
-		*p_out++ = (float) ((aw * 0.4563) - (ax*0.1259) + (ay * 0.1543));   //LS
-        *p_out++ = (float) ((aw * 0.4563) - (ax*0.1259) - (ay * 0.1543));   //RS
-	}
-}
-/* from Bruce Wiggins via Csound */
-void fm_i2_surr(const ABFSAMPLE *inbuf,float*outbuf,unsigned int numframes)
-{
-	int i;	
-	float *p_out = outbuf;
-	double aw,ax,ay,au,av;
-	
-	for(i=0;i < numframes; i++){
-		aw = (double) inbuf->W;
-		ax = (double) inbuf->X;
-		ay = (double) inbuf->Y;		
-		au = (double) inbuf->U;
-        av = (double) inbuf->V;
-        
-		*p_out++ = (float) ((aw * 0.405) + (ax*0.32) + (ay * 0.31)  + (au * 0.085) + (av * 0.125));   //L
-        *p_out++ = (float) ((aw * 0.405) + (ax*0.32) - (ay * 0.31)  + (au * 0.085) - (av * 0.125));   //R
-		*p_out++ = (float) ((aw * 0.085) + (ax*0.04)                + (au * 0.045)               );   //C
-		*p_out++ = (float) ((aw * 0.635) - (ax*0.335) + (ay * 0.28) - (au * 0.08)  + (av * 0.08));    //LS
-        *p_out++ = (float) ((aw * 0.635) - (ax*0.335) - (ay * 0.28) - (au * 0.08)  - (av * 0.08));    //RS
-	}
-}
-
-/* 5.1 versions - silent LFE */
-/* FMH only defines 1st order decode */ 
-void fm_i1_surr6(const ABFSAMPLE *inbuf,float*outbuf,unsigned int numframes)
-{
-	int i;	
-	float *p_out = outbuf;
-	double aw,ax,ay;
-	
-	for(i=0;i < numframes; i++){
-		aw = (double) inbuf->W;
-		ax = (double) inbuf->X;
-		ay = (double) inbuf->Y;		
-		
-		*p_out++ = (float) ((aw * 0.169)  + (ax*0.0797) + (ay * 0.0891));   //L
-		*p_out++ = (float) ((aw * 0.1635) + (ax*0.0923));                   //C
-		*p_out++ = (float) ((aw * 0.169)  - (ax*0.0797) - (ay * 0.0891));   //R
-        *p_out++ = 0.0f;                                                    //LFE
-		*p_out++ = (float) ((aw * 0.4563) - (ax*0.1259) + (ay * 0.1543));   //LS
-        *p_out++ = (float) ((aw * 0.4563) - (ax*0.1259) - (ay * 0.1543));   //RS
-	}
-}
-/* from Bruce Wiggins via Csound */
-void fm_i2_surr6(const ABFSAMPLE *inbuf,float*outbuf,unsigned int numframes)
-{
-	int i;	
-	float *p_out = outbuf;
-	double aw,ax,ay,au,av;
-	
-	for(i=0;i < numframes; i++){
-		aw = (double) inbuf->W;
-		ax = (double) inbuf->X;
-		ay = (double) inbuf->Y;		
-		au = (double) inbuf->U;
-        av = (double) inbuf->V;
-        
-		*p_out++ = (float) ((aw * 0.405) + (ax*0.32) + (ay * 0.31)  + (au * 0.085) + (av * 0.125));   //L
-        *p_out++ = (float) ((aw * 0.405) + (ax*0.32) - (ay * 0.31)  + (au * 0.085) - (av * 0.125));   //R
-		*p_out++ = (float) ((aw * 0.085) + (ax*0.04)                + (au * 0.045)               );   //C
-        *p_out++ = 0.0f;                                                                              //LFE
-		*p_out++ = (float) ((aw * 0.635) - (ax*0.335) + (ay * 0.28) - (au * 0.08)  + (av * 0.08));    //LS
-        *p_out++ = (float) ((aw * 0.635) - (ax*0.335) - (ay * 0.28) - (au * 0.08)  - (av * 0.08));    //RS
-	}
-}
-
-// 1st order 5.0
-void dm_i1_surr(const ABFSAMPLE *inbuf,float*outbuf,unsigned int numframes)
-{
-	int i;	
-	float *p_out = outbuf;
-	double aw,ax,ay;
-	
-	for(i=0;i < numframes; i++){
-		aw = (double) inbuf->W;
-		ax = (double) inbuf->X;
-		ay = (double) inbuf->Y;		
-		
-        
-		*p_out++ = (float) ((aw * 0.4597)  + (ax*0.4536) + (ay * 0.3591));   //L
-        *p_out++ = (float)  ((aw * 0.4597)  + (ax*0.4536) - (ay * 0.3591));  //R 
-		*p_out++ = 0.0f;                                                     //C
-		
-		*p_out++ = (float) ((aw * 0.5662) - (ax*0.3681) + (ay * 0.4606));    //LS
-        *p_out++ = (float) ((aw * 0.5662) - (ax*0.3681) - (ay * 0.4606));    //RS
-    }
-}
-//1st order 5.1
-void dm_i1_surr6(const ABFSAMPLE *inbuf,float*outbuf,unsigned int numframes)
-{
-	int i;	
-	float *p_out = outbuf;
-	double aw,ax,ay;
-	
-	for(i=0;i < numframes; i++){
-		aw = (double) inbuf->W;
-		ax = (double) inbuf->X;
-		ay = (double) inbuf->Y;		
-		
-        
-		*p_out++ = (float) ((aw * 0.4597)  + (ax*0.4536) + (ay * 0.3591));   //L
-        *p_out++ = (float)  ((aw * 0.4597)  + (ax*0.4536) - (ay * 0.3591));  //R 
-		*p_out++ = 0.0f;                                                     //C
-		*p_out++ = 0.0f;                                                     //LFE
-		*p_out++ = (float) ((aw * 0.5662) - (ax*0.3681) + (ay * 0.4606));    //LS
-        *p_out++ = (float) ((aw * 0.5662) - (ax*0.3681) - (ay * 0.4606));    //RS
-    }
-}
-// 2nd order 5.0
-void dm_i2_surr(const ABFSAMPLE *inbuf,float*outbuf,unsigned int numframes)
-{
-	int i;	
-	float *p_out = outbuf;
-	double aw,ax,ay,au,av;
-	
-	for(i=0;i < numframes; i++){
-		aw = (double) inbuf->W;
-		ax = (double) inbuf->X;
-		ay = (double) inbuf->Y;		
-		au = (double) inbuf->U;
-        av = (double) inbuf->V;
-        
-		*p_out++ = (float) ((aw * 0.3314)  + (ax*0.4097) + (ay * 0.3487) + (au * 0.0828) + (av*0.1489));  //L
-        *p_out++ = (float) ((aw * 0.3314)  + (ax*0.4097) - (ay * 0.3487) + (au * 0.0828) - (av*0.1489));  //R 
-		*p_out++ = (float) ((aw * 0.0804)  + (ax * 0.1327));                                              //C
-		*p_out++ = (float) ((aw * 0.6025) - (ax*0.3627) + (ay * 0.4089) - (au * 0.0567));                 //LS
-        *p_out++ = (float) ((aw * 0.6025) - (ax*0.3627) - (ay * 0.4089) - (au * 0.0567));                 //RS
-    }
-}
-// 2nd order 5.1
-void dm_i2_surr6(const ABFSAMPLE *inbuf,float*outbuf,unsigned int numframes)
-{
-	int i;	
-	float *p_out = outbuf;
-	double aw,ax,ay,au,av;
-	
-	for(i=0;i < numframes; i++){
-		aw = (double) inbuf->W;
-		ax = (double) inbuf->X;
-		ay = (double) inbuf->Y;		
-		au = (double) inbuf->U;
-        av = (double) inbuf->V;
-        
-		*p_out++ = (float) ((aw * 0.3314)  + (ax*0.4097) + (ay * 0.3487) + (au * 0.0828) + (av*0.1489));  //L
-        *p_out++ = (float) ((aw * 0.3314)  + (ax*0.4097) - (ay * 0.3487) + (au * 0.0828) - (av*0.1489));  //R 
-		*p_out++ = (float) ((aw * 0.0804)  + (ax * 0.1327));                                              //C
-		*p_out++ = 0.0f;                                                                                  //LFE
-		*p_out++ = (float) ((aw * 0.6025) - (ax*0.3627) + (ay * 0.4089) - (au * 0.0567));                 //LS
-        *p_out++ = (float) ((aw * 0.6025) - (ax*0.3627) - (ay * 0.4089) - (au * 0.0567));                 //RS
-    }
-}
-
-
-void fm_i1_hex(const ABFSAMPLE *inbuf,float*outbuf,unsigned int numframes)
-{
-	int i;	
-	float *p_out = outbuf;
-	double aw,ax,ay;
-	
-	for(i=0;i < numframes; i++){
-		aw = (double) inbuf->W * 0.2357;
-		ax = (double) inbuf->X * 0.1443;
-		ay = (double) inbuf->Y;		
-		
-		*p_out++ = (float) (aw + ax + (ay * 0.0833));  //FL
-		*p_out++ = (float) (aw      + (ay * 0.1667));  //SL
-		*p_out++ = (float) (aw - ax + (ay * 0.0833));  //RL
-		*p_out++ = (float) (aw - ax - (ay * 0.0833));  //RR
-        *p_out++ = (float) (aw      - (ay * 0.1667));  //SR
-        *p_out++ = (float) (aw + ax - (ay * 0.0833));  //FR
-	}
-}
-void fm_i2_hex(const ABFSAMPLE *inbuf,float*outbuf,unsigned int numframes)
-{
-	int i;	
-	float *p_out = outbuf;	
-	double aw,ax,ay,au,av;
-	
-	for(i=0;i < numframes; i++){
-		aw = (double) inbuf->W * 0.2357;
-		ax = (double) inbuf->X * 0.1987;
-		ay = (double) inbuf->Y;
-		au = (double) inbuf->U;
-        av = (double) inbuf->V * 0.0556;
-		
-		*p_out++ = (float) (aw + ax + (ay * 0.1147) + (au * 0.0321) + av);  //FL
-		*p_out++ = (float) (aw      + (ay * 0.2294) - (au * 0.0643)     );  //SL
-		*p_out++ = (float) (aw - ax + (ay * 0.1147) + (au * 0.0321) - av);  //RL
-		*p_out++ = (float) (aw - ax - (ay * 0.1147) + (au * 0.0321) + av);  //RR
-        *p_out++ = (float) (aw      - (ay * 0.2294) - (au * 0.0643)     );  //SR
-        *p_out++ = (float) (aw + ax - (ay * 0.1147) + (au * 0.0321) - av);  //FR
-	}
-}
-
-void fm_i1_oct1(const ABFSAMPLE *inbuf,float*outbuf,unsigned int numframes)
-{
-	int i;	
-	float *p_out = outbuf;
-	double aw,ax,ay;
-	
-	for(i=0;i < numframes; i++){
-		aw = (double) inbuf->W * 0.1768;
-		ax = (double) inbuf->X;
-		ay = (double) inbuf->Y;		
-		
-		*p_out++ = (float) (aw + (ax * 0.1155) + (ay * 0.0478));  
-		*p_out++ = (float) (aw + (ax * 0.0478) + (ay * 0.1155));  
-		*p_out++ = (float) (aw - (ax * 0.0478) + (ay * 0.1155));  
-		*p_out++ = (float) (aw - (ax * 0.1155) + (ay * 0.0478));  
-        *p_out++ = (float) (aw - (ax * 0.231)  - (ay * 0.0957));  
-        *p_out++ = (float) (aw - (ax * 0.0478) - (ay * 0.1155));  
-        *p_out++ = (float) (aw + (ax * 0.0478) - (ay * 0.1155));  
-        *p_out++   = (float) (aw + (ax * 0.1155) - (ay * 0.0478));  
-	}
-}
-void fm_i2_oct1(const ABFSAMPLE *inbuf,float *outbuf,unsigned int numframes)
-{
-	int i;	
-	float *p_out = outbuf;	
-	double aw,ax,ay,au,av;
-	
-	for(i=0;i < numframes; i++){
-		aw = (double) inbuf->W * 0.17677;
-		ax = (double) inbuf->X;
-		ay = (double) inbuf->Y;
-		au = (double) inbuf->U * 0.03417;
-		av = (double) inbuf->V * 0.03417;
-        
-		*p_out++ = (float) (aw + (ax * 0.15906) + (ay * 0.06588) + au + av);  
-		*p_out++ = (float) (aw + (ax * 0.06588) + (ay * 0.15906) - au + av);  
-		*p_out++ = (float) (aw - (ax * 0.06588) + (ay * 0.15906) - au - av);  
-		*p_out++ = (float) (aw - (ax * 0.15906) + (ay * 0.06588) + au - av);  
-        *p_out++ = (float) (aw - (ax * 0.15906) - (ay * 0.06588) + au + av);  
-        *p_out++ = (float) (aw - (ax * 0.06588) - (ay * 0.15906) - au + av);  
-        *p_out++ = (float) (aw + (ax * 0.06588) - (ay * 0.15906) - au - av);  
-        *p_out++ = (float) (aw + (ax * 0.15906) - (ay * 0.06588) + au - av);  
-	}
-}
-
-void fm_i1_oct2(const ABFSAMPLE *inbuf,float *outbuf,unsigned int numframes)
-{
-	int i;	
-	float *p_out = outbuf;
-	double aw,ax,ay;
-	
-	for(i=0;i < numframes; i++){
-		aw = (double) inbuf->W * 0.1768;
-		ax = (double) inbuf->X;
-		ay = (double) inbuf->Y;		
-		
-		*p_out++ = (float) (aw + (ax * 0.125)                 );  
-		*p_out++ = (float) (aw + (ax * 0.0884) + (ay * 0.0884));  
-		*p_out++ = (float) (aw                 + (ay * 0.125) );  
-		*p_out++ = (float) (aw - (ax * 0.0884) + (ay * 0.0884));  
-        *p_out++ = (float) (aw - (ax * 0.125)                 );  
-        *p_out++ = (float) (aw - (ax * 0.0884) - (ay * 0.0884));  
-        *p_out++ = (float) (aw                 - (ay * 0.125) );  
-        *p_out++ = (float) (aw + (ax * 0.0884) - (ay * 0.0884));  
-	}
-}
-void fm_i2_oct2(const ABFSAMPLE *inbuf,float *outbuf,unsigned int numframes)
-{
-	int i;	
-	float *p_out = outbuf;	
-	double aw,ax,ay,au,av;
-	
-	for(i=0;i < numframes; i++){
-		aw = (double) inbuf->W * 0.1768;
-		ax = (double) inbuf->X;
-		ay = (double) inbuf->Y;
-		au = (double) inbuf->U * 0.0482;
-		av = (double) inbuf->V * 0.0482;
-        
-		*p_out++ = (float) (aw + (ax * 0.1721)                 + au     );  
-		*p_out++ = (float) (aw + (ax * 0.1217) + (ay * 0.1217)      + av);  
-		*p_out++ = (float) (aw                 + (ay * 0.1721) - au     );  
-		*p_out++ = (float) (aw - (ax * 0.1217) + (ay * 0.1217)      - av);  
-        *p_out++ = (float) (aw - (ax * 0.1721)                 + au     );  
-        *p_out++ = (float) (aw - (ax * 0.1217) - (ay * 0.1217)      + av);  
-        *p_out++ = (float) (aw                 - (ay * 0.1721) - au     );  
-        *p_out++ = (float) (aw + (ax * 0.1217) - (ay * 0.1217)      - av);  
-	}
-}
-
-/* csound order; low/high anti-clockwise. 
-FMH page order, 4 low folowed by 4 high , clockwise! */
-void fm_i1_cube(const ABFSAMPLE *inbuf,float *outbuf,unsigned int numframes)
-{
-	int i;	
-	float *p_out = outbuf;
-	double aw,ax,ay,az;
-	
-	for(i=0;i < numframes; i++){
-		aw = (double) inbuf->W * 0.17677;
-		ax = (double) inbuf->X * 0.07216;
-		ay = (double) inbuf->Y * 0.07216;
-		az = (double) inbuf->Z * 0.07216;
-		
-		*p_out++ = (float) (aw + ax + ay - az);  // FL low
-		*p_out++ = (float) (aw + ax + ay + az);  // FL hi
-        
-		*p_out++ = (float) (aw - ax + ay - az);  // RL low
-		*p_out++ = (float) (aw - ax + ay + az);  //    hi
-        
-        *p_out++ = (float) (aw - ax - ay - az);  // RR low
-        *p_out++ = (float) (aw - ax - ay + az);  //   hi
-        
-        *p_out++ = (float) (aw + ax - ay - az);  // FR low
-        *p_out++ = (float) (aw + ax - ay + az);  //    hi
-	}
-}
-void fm_i2_cube(const ABFSAMPLE *inbuf,float *outbuf,unsigned int numframes)
-{
-	int i;	
-	float *p_out = outbuf;	
-	double aw,ax,ay,az,as,at,av;
-	
-	for(i=0;i < numframes; i++){
-		aw = (double) inbuf->W * 0.1768;
-		ax = (double) inbuf->X * 0.114;
-		ay = (double) inbuf->Y * 0.114;
-        az = (double) inbuf->Z * 0.114;
-        as = (double) inbuf->S * 0.0369;
-        at = (double) inbuf->T * 0.0369;
-		av = (double) inbuf->V * 0.0369;
-        
-		*p_out++ = (float) (aw + ax + ay - az - as - at + av); //FL low 
-		*p_out++ = (float) (aw + ax + ay + az + as + at + av); //   hi 
-		
-        *p_out++ = (float) (aw - ax + ay - az + as - at - av); //RL low
-		*p_out++ = (float) (aw - ax + ay + az - as + at - av);  
-        
-        *p_out++ = (float) (aw - ax - ay - az + as + at + av); // RR low
-        *p_out++ = (float) (aw - ax - ay + az - as - at + av);  
-        
-        *p_out++ = (float) (aw + ax - ay - az - as + at - av);  // FR low
-        *p_out++ = (float) (aw + ax - ay + az + as - at - av);   
-	}
-}
-/* ditto, wavex order */
-/* Front L, front R, Back L, Back R; top Front L, Top Fr R, Top Back L, Top back R */
-void fm_i1_cubex(const ABFSAMPLE *inbuf,float *outbuf,unsigned int numframes)
-{
-	int i;	
-	float *p_out = outbuf;
-	double aw,ax,ay,az;
-	
-	for(i=0;i < numframes; i++){
-		aw = (double) inbuf->W * 0.17677;
-		ax = (double) inbuf->X * 0.07216;
-		ay = (double) inbuf->Y * 0.07216;
-		az = (double) inbuf->Z * 0.07216;
-		
-		*p_out++ = (float) (aw + ax + ay - az);  // FL low
-        *p_out++ = (float) (aw + ax - ay - az);  // FR low
-        *p_out++ = (float) (aw - ax + ay - az);  // RL low
-        *p_out++ = (float) (aw - ax - ay - az);  // RR low
-        
-		*p_out++ = (float) (aw + ax + ay + az);  // FL hi
-        *p_out++ = (float) (aw + ax - ay + az);  // FR hi
-		*p_out++ = (float) (aw - ax + ay + az);  // RL hi
-        *p_out++ = (float) (aw - ax - ay + az);  // RR hi
-	}
-}
-void fm_i2_cubex(const ABFSAMPLE *inbuf,float *outbuf,unsigned int numframes)
-{
-	int i;	
-	float *p_out = outbuf;	
-	double aw,ax,ay,az,as,at,av;
-	
-	for(i=0;i < numframes; i++){
-		aw = (double) inbuf->W * 0.1768;
-		ax = (double) inbuf->X * 0.114;
-		ay = (double) inbuf->Y * 0.114;
-        az = (double) inbuf->Z * 0.114;
-        as = (double) inbuf->S * 0.0369;
-        at = (double) inbuf->T * 0.0369;
-		av = (double) inbuf->V * 0.0369;
-        
-		*p_out++ = (float) (aw + ax + ay - az - as - at + av);  // FL low
-        *p_out++ = (float) (aw + ax - ay - az - as + at - av);  // FR low
-        *p_out++ = (float) (aw - ax + ay - az + as - at - av);  // RL low
-        *p_out++ = (float) (aw - ax - ay - az + as + at + av);  // RR low
-		
-        *p_out++ = (float) (aw + ax + ay + az + as + at + av);  // FL  hi 
-		*p_out++ = (float) (aw + ax - ay + az + as - at - av);  // FR  hi
-		*p_out++ = (float) (aw - ax + ay + az - as + at - av);  // RL  hi 
-        *p_out++ = (float) (aw - ax - ay + az - as - at + av);  // RR  hi 
-	}
-}
-
-
-#ifdef NOTDEF
-void bfdcode4(float *inbuf,long numframes)
-{
-	int i;	
-	float *p_buf = inbuf;	
-	double aw,ax,ay;
-	
-	for(i=0;i < numframes; i++){
-		aw = (double)(p_buf[0]) ;
-		ax = (double)(p_buf[1]) * 0.707;
-		ay = (double)(p_buf[2]) * 0.707;		
-		//decode frame
-		*p_buf++ = (float)(0.3333 * (aw + ax + ay)); //FL
-		*p_buf++ = (float)(0.3333 * (aw + ax - ay)); //FR
-		*p_buf++ = (float)(0.3333 * (aw - ax + ay)); //RL
-		*p_buf++ = (float)(0.3333 * (aw - ax - ay));  //RR
-	}
-}
-
-
-/* handle 3ch in to 4ch out! */
-
-void bfdcode324(float *inbuf,float*outbuf,long numframes)
-{
-	int i;	
-	float *p_buf = inbuf;
-	float * p_outbuf = outbuf;
-	double aw,ax,ay;
-	
-	for(i=0;i < numframes; i++){
-		int j;
-		aw = (double)(*p_buf++) ;
-		ax = (double)(*p_buf++) * 0.707;
-		ay = (double)(*p_buf++) * 0.707;
-        
-		//decode frame
-		*p_outbuf++ = (float)(0.3333 * (aw + ax + ay));
-		*p_outbuf++ = (float)(0.3333 * (aw + ax - ay));
-		*p_outbuf++ = (float)(0.3333 * (aw - ax + ay));
-		*p_outbuf++ = (float)(0.3333 * (aw - ax - ay));
-	}
-}
-#endif

dev/externals/mctools/interlx.c

@@ -1,504 +0,0 @@
-/*
- * Copyright (c) 1983-2013 Richard Dobson and Composers Desktop Project Ltd
- * http://people.bath.ac.uk/masrwd
- * http://www.composersdesktop.com
- * This file is part of the CDP System.
- * The CDP System is free software; you can redistribute it
- * and/or modify it under the terms of the GNU Lesser General Public 
- * License as published by the Free Software Foundation; either 
- * version 2.1 of the License, or (at your option) any later version. 
- *
- * The CDP System is distributed in the hope that it will be useful, 
- * but WITHOUT ANY WARRANTY; without even the implied warranty of 
- * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  
- * See the GNU Lesser General Public License for more details.
- * You should have received a copy of the GNU Lesser General Public
- * License along with the CDP System; if not, write to the Free Software
- * Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
- *
- */
- 
-/*TODO: fix bug writing header when using -x with .amb extension (omits cbSize setting) */
-
-/*interlx.c */
-/* v1.3. nov 2005; support placeholder arg for silent channel */
-/* v 1.7 beta ; added surr 5.0; updated sfsys with bit-correct 24bit copy */
-/* v 1.8 March 2009 updated sfsys  for AIFC int24 suport */
-
-/* OCT 2009 portsf(64) version. int24 aifc supported for reading. */
-/* v2.0.1 Jan 2010: Corrected usage message to refer to outfile.  */
-/* Nov 2013 added MC_SURR_6_1 */
-#include <stdio.h>
-#include <stdlib.h>
-#include <math.h>
-#include <memory.h>
-#include <string.h>
-#include <ctype.h>
-#include "portsf.h"
-
-/*
-The number of channels defines the order of the soundfield:
- 3 channel = h   = 1st order horizontal
- 4 channel = f   = 1st order 3-D
- 5 channel = hh  = 2nd order horizontal
- 6 channel = fh  = 2nd order horizontal + 1st order height (formerly called 2.5 order)
- 7 channel = hhh = 3rd order horizontal
- 8 channel = fhh = 3rd order horizontal + 1st order height
- 9 channel = ff  = 2nd order 3-D
-11 channel = ffh = 3rd order horizontal + 2nd order height
-16 channel = fff = 3rd order 3-D
-*/
-
-#define N_BFORMATS (10)
-static const int bformats[N_BFORMATS] = {2,3,4,5,6,7,8,9,11,16};
-#define MAX_INFILES (16)	//should keep most people happy! Won't get much more on a cmdline anyway
-
-enum {OPT_STD_WAVE,OPT_WAVEX_GENERIC,OPT_WAVEX,OPT_WAVEX_LCRS,OPT_WAVEX_SURROUND,OPT_WAVEX_BFORMAT,
-   OPT_SURR_5_0,OPT_WAVEX_7_1,OPT_WAVEX_CUBE,OPT_WAVEX_6_1, OPT_MAXOPTS};
-
-#ifdef unix
-/* in portsf.lib */
-extern int stricmp(const char *a, const char *b);
-#endif
-
-#ifndef max
-#define max(x,y) ((x) > (y) ? (x) : (y))
-#endif
-
-void
-usage()
-{
-	fprintf(stderr,"\nCDP MCTOOLS: INTERLX v2.1.0 (c) RWD,CDP 2009,2013\n"
-			"Interleave mono or stereo files into a multi-channel file\n"
-			"Usage: interlx [-tN] outfile infile1 infile2 [infile3...]\n"
-			
-			"       Up to %d files may be interleaved.\n"						
-			"       Output format is taken from infile1.\n"
-			"       Files must match sample rate and number of channels,\n"
-			"          but can have different sample types.\n"
-			"       To create a silent channel, for infile2 onwards,\n"
-			"          use 0 (zero) as filename. Infile1 must be a soundfile.\n"
-			"       NB: Speaker-positions in WAVE_EX infiles are ignored\n"
-			"       Note that the same infile can be listed multiple times,\n"
-			"          for example, to write a mono file as stereo, quad, etc.\n"
-			"       The .amb B-Format extension is supported: \n"
-			"          the program warns if channel count is anomalous.\n"
-			"       recognised Bformat channel counts: 3,4,5,6,7,8,9,11,16.\n"
-            "   -tN  : write outfile format as type N\n"
-	        "    Available formats:\n"
-			"    0   : (default) standard soundfile (.wav, .aif, .afc, .aifc)\n"
-			"    1   : generic WAVE_EX (no speaker assignments)\n"
-			"    2   : WAVE_EX mono/stereo/quad(LF,RF,LR,RR)   - total chans must match.\n"
-			"    3   : WAVE_EX quad surround (L,C,R,S)         - total chans must be 4.\n"
-			"    4   : WAVE_EX 5.1 format surround             - total chans must be 6.\n"
-			"    5   : WAVE_EX Ambisonic B-format (W,X,Y,Z...) - extension .amb recommended.\n"
-            "    6   : WAVE_EX 5.0 surround                    - total chans must be 5.\n"
-            "    7   : WAVE_EX 7.1 Surround                    - total chans must be 8.\n"
-            "    8   : WAVE_EX Cube Surround                   - total chans must be 8.\n"
-            "    9   : WAVE_EX 6.1 surround (new in v 2.1.0)   - total chans must be 7.\n"
-			"          in all cases: outfile has sample format of infile1\n"
-			"          NB: types 1 to %d are for WAV format only\n"			
-			,MAX_INFILES,OPT_MAXOPTS-1);
-	
-}
-	
-
-void cleanup(int *sflist)
-{
-
-	int i;
-	for(i=0;i < MAX_INFILES; i++)
-		if(sflist[i] >= 0)
-			psf_sndClose(sflist[i]);
-	psf_finish();
-}
-
-int main(int argc, char *argv[])
-{
-
-	long outframesize,thissize;
-	int i,ofd;
-	int ifdlist[MAX_INFILES];
-	//int force_stype = -1,out_stype;
-    int force_wave = 0;
-	int infilearg,inchans;
-	int num_infiles = 0;
-	int halfsec;
-    MYLONG peaktime;
-	float *outframe	= NULL,*inframe = NULL;
-	PSF_PROPS firstinprops,inprops;
-	PSF_CHPEAK *fpeaks = NULL;
-    int wave_type = -1;
-    char *create_msg = NULL;
-    psf_format informat = PSF_STDWAVE;
-	char* p_dot = NULL;		/* to find extension of outfile */
-	
-	for(i=0;i < MAX_INFILES;i++)
-		ifdlist[i] = -1;
-
-    /* CDP version number */
-    if(argc==2 && (stricmp(argv[1],"--version")==0)){
-        printf("2.0.1.\n");
-        return 0;
-    }
-    
-	if(argc < 4) {
-		fprintf(stderr,"interlx: insufficient arguments\n");
-		usage();
-		exit(1);
-	}
-
-	while(argv[1][0] =='-'){
-		
-		switch(argv[1][1]){
-        case('t'):
-			if(argv[1][2]=='\0'){
-				fprintf(stderr,"-t flag requires parameter\n");
-				usage();
-				exit(1);
-			}
-			wave_type = atoi(&(argv[1][2]));
-			if((wave_type < 0) || wave_type >= OPT_MAXOPTS){
-				fprintf(stderr,"wave type out of range\n");
-				usage();
-				exit(1);
-			}
-			force_wave = 1;
-			break;
-		default:
-			fprintf(stderr,"\nabfpan: error: illegal flag option %s",argv[1]);
-			exit(1);
-		}
-
-		argc--; argv++;
-	}
-
-	if(argc < 4){
-		fprintf(stderr,"interlx error: at least two infiles required!\n");
-		usage();
-		exit(1);
-	}
-
-	if(psf_init()){
-		printf("Startup failure.\n");
-		return  1;
-	}
-	
-	//open first infile and get properties
-	ifdlist[0] = psf_sndOpen(argv[2],&firstinprops,0);
-	if(ifdlist[0] < 0){
-		fprintf(stderr,"unable to open infile %s\n",argv[2]);
-		cleanup(ifdlist);
-		return 1;
-	}
-
-/* we don't know how to deal with speaker positions yet, so disregard these
-	if(firstinprops.chformat > MC_STD){
-		printf(stderr,"Warning,interlx: ignoring source file speaker positions\n");
-	}
-*/
-	outframesize = psf_sndSize(ifdlist[0]);
-	if(outframesize < 0){
-		fprintf(stderr,"unable to read size of infile %s\n",argv[2]);
-		cleanup(ifdlist);
-		return 1;
-	}
-	inchans = firstinprops.chans;
-	/*we can always allow more channels if someone really needs it! */
-	if(!(inchans==1 || inchans==2)){
-		fprintf(stderr,"interlx: error: infile %s has %d channels\n"
-				"\t(only mono and stereo files can be used)\n",argv[2],inchans);
-		cleanup(ifdlist);
-		return 1;
-	}
-	
-	num_infiles = 1;
-	printf("interleaving %d-channel files,sample rate = %d\n",inchans,firstinprops.srate);
-
-	infilearg = 3;
-	while(argv[infilearg] != NULL){
-		if(strcmp(argv[infilearg],"0")==0){
-			ifdlist[num_infiles] = -1;           // mark silent channel
-		}
-		else{
-
-			if((ifdlist[num_infiles] = psf_sndOpen(argv[infilearg],&inprops,0)) < 0){
-				fprintf(stderr,"cannot open infile %s\n",argv[infilearg]);
-				cleanup(ifdlist);
-				return 1;
-			}
-			if(inprops.chans != firstinprops.chans){
-				fprintf(stderr,"interlx: error: channel mismatch from infile %s\n",argv[infilearg]);
-				cleanup(ifdlist);
-				return 1;
-			}
-
-			if(inprops.srate != firstinprops.srate){
-			   fprintf(stderr,"interlx: error: sample rate mismatch from infile %s\n",argv[infilearg]);
-				cleanup(ifdlist);
-				return 1;
-			}
-
-			thissize = psf_sndSize(ifdlist[num_infiles]);
-			if(thissize < 0){
-				fprintf(stderr,"unable to read size of infile %s\n",argv[infilearg]);
-				cleanup(ifdlist);
-				return 1;
-			}			
-			outframesize = max(outframesize,thissize);
-		}
-		
-		infilearg++;
-		num_infiles++;
-		if(num_infiles > MAX_INFILES){
-			fprintf(stderr,"Sorry! too many infiles. Maximum accepted is %d.\n",MAX_INFILES);
-			cleanup(ifdlist);
-			exit(1);
-		}
-
-	}
-
-	
-	inframe = malloc(inchans * sizeof(float));
-	if(inframe==NULL){
-		puts("interlx: error: no memory for input buffer!\n");
-		cleanup(ifdlist);
-		return 1;
-	}
-
-	firstinprops.chans *= num_infiles;
-	outframe = (float *) malloc(firstinprops.chans * sizeof(float));
-	if(outframe==NULL){
-		puts("\ninterlx: error: no memory for output buffer!\n");
-		cleanup(ifdlist);
-		return 1;
-	}
-
-	fpeaks = (PSF_CHPEAK *) calloc(firstinprops.chans,sizeof(PSF_CHPEAK));
-	if(fpeaks==NULL){
-		puts("interlx: error: no memory for internal PEAK buffer\n");
-		cleanup(ifdlist);
-		return 1;
-	}
-    if(force_wave){
-		int i,matched;
-		
-		switch(wave_type){
-        case(OPT_WAVEX_GENERIC):
-			inprops.chformat = MC_STD;
-			informat = PSF_WAVE_EX;
-			create_msg = "creating STD WAVE_EX file";
-			break;
-		case(OPT_WAVEX):
-			switch(firstinprops.chans){
-			case(1):
-				firstinprops.chformat = MC_MONO;
-				informat = PSF_WAVE_EX;
-				create_msg = "creating MONO WAVE_EX file";
-				break;
-			case(2):
-				firstinprops.chformat = MC_STEREO;
-				informat = PSF_WAVE_EX;
-				create_msg = "creating STEREO WAVE_EX file";
-				break;
-			case(4):
-				firstinprops.chformat = MC_QUAD;
-				informat = PSF_WAVE_EX;
-				create_msg = "creating QUAD WAVE_EX file";
-				break;
-			default:
-				fprintf(stderr,"infile nchans incompatible with requested WAVE-EX format\n");
-				usage();
-				cleanup(ifdlist);
-				return 1;				
-			}
-			break;
-		case(OPT_WAVEX_LCRS):
-			if(firstinprops.chans != 4){
-				fprintf(stderr,"result must have four channels\n");
-				usage();
-				cleanup(ifdlist);
-				return 1;
-			}
-			firstinprops.chformat = MC_LCRS;
-			informat = PSF_WAVE_EX;
-			create_msg = "creating LCRS-surround WAVE_EX file";
-			break;
-		case(OPT_WAVEX_SURROUND):
-			if(firstinprops.chans != 6){
-				fprintf(stderr,"result must have six channels\n");
-				usage();
-				cleanup(ifdlist);
-				exit(1);
-			}
-			firstinprops.chformat = MC_DOLBY_5_1;
-			informat = PSF_WAVE_EX;
-			create_msg = "creating 5.1 surround WAVE_EX file";
-			break;
-            
-        case(OPT_SURR_5_0):
-			if(firstinprops.chans != 5){
-				fprintf(stderr,"result must have five channels.\n");
-				usage();
-				cleanup(ifdlist);
-				return 1;
-			}
-			firstinprops.chformat = MC_SURR_5_0;
-			informat = PSF_WAVE_EX;
-			create_msg = "creating 5.0 surround WAVE_EX file";
-			break;
-
-		case(OPT_WAVEX_BFORMAT):
-			matched = 0;
-			for(i=0;i < N_BFORMATS;i++)	{
-				if(firstinprops.chans == bformats[i]){
-					matched = 1;
-					break;
-				}
-			}
-			if(!matched){
-				printf("WARNING: No Bformat definition for %d-channel file.\n",inprops.chans);
-			}
-			firstinprops.chformat = MC_BFMT;
-			informat = PSF_WAVE_EX;
-			create_msg = "creating AMBISONIC B-FORMAT WAVE_EX file";
-			break;
-        case OPT_WAVEX_7_1:
-            if(firstinprops.chans != 8){
-				fprintf(stderr,"result must have  channels\n");
-				usage();
-				cleanup(ifdlist);
-				return 1;
-			}
-			firstinprops.chformat = MC_SURR_7_1;
-			informat = PSF_WAVE_EX;
-			create_msg = "creating 7.1 surround WAVE_EX file";
-			break;
-        case OPT_WAVEX_CUBE:
-            if(firstinprops.chans != 8){
-				fprintf(stderr,"result must have  channels\n");
-				usage();
-				cleanup(ifdlist);
-				return 1;
-			}
-			firstinprops.chformat = MC_CUBE;
-			informat = PSF_WAVE_EX;
-			create_msg = "creating cube surround WAVE_EX file";
-			break;
-		case OPT_WAVEX_6_1:
-            if(firstinprops.chans != 7){
-				fprintf(stderr,"result must have  channels\n");
-				usage();
-				cleanup(ifdlist);
-				return 1;
-			}
-			firstinprops.chformat = MC_SURR_6_1;
-			informat = PSF_WAVE_EX;
-			create_msg = "creating 6.1 surround WAVE_EX file";
-			break;
-		default:
-            inprops.chformat = STDWAVE;
-			informat = PSF_STDWAVE;
-			create_msg = "creating plain sound file";
-			break;
-		}		
-	}
-	
-	/*  want to avoid WAVE_EX if just plain WAVE possible */
-	firstinprops.format = informat;
-		/*firstinprops.chformat = MC_STD;*/		/* RWD April 2006 do this here? */
-	
-	p_dot = strrchr(argv[1],'.');
-	if(stricmp(++p_dot,"amb")==0) {
-		int i;
-		int matched = 0;
-		firstinprops.chformat = MC_BFMT;
-		for(i=0;i < N_BFORMATS;i++)	{
-			if(firstinprops.chans == bformats[i]){
-				matched = 1;
-				break;
-			}
-		}
-		if(!matched)
-			printf("\nWARNING: channel count %d unknown for BFormat.\n",firstinprops.chans);
-
-	}
-    if(!is_legalsize(outframesize,&firstinprops)){
-        fprintf(stderr,"error: outfile size %ld exceeds capacity of format.\n",outframesize);
-        return 1;
-    }
-    printf("\n%s: %d channels, %ld frames.\n",create_msg,firstinprops.chans,outframesize);
-	ofd = psf_sndCreate(argv[1],&firstinprops,0,0,PSF_CREATE_RDWR);
-	
-	
-	if(ofd < 0){
-		fprintf(stderr,"interlx: error: unable to create outfile %s.\n",argv[1]);
-		cleanup(ifdlist);
-		return 1;
-	}
-
-	halfsec = firstinprops.srate / 2;
-	for(i=0;i < outframesize; i++){		// frame loop
-		float *p_framesamp,*p_filesamp;
-		int j;
-		
-		p_framesamp = outframe;
-		memset((char *)outframe,0,firstinprops.chans * sizeof(float));
-
-		for(j=0;j < num_infiles;j++) {	//file loop
-			int k,got;
-            
-			memset((char *)inframe,0,inchans * sizeof(float));
-			if(ifdlist[j] < 0){
-				got = inchans; // placeholder - write silent channel
-			}
-			else{
-				if((got = psf_sndReadFloatFrames(ifdlist[j],inframe,1)) < 0){
-					fprintf(stderr,"interlx: error reading from infile %s\n",argv[2+j]);
-					psf_sndClose(ofd);					
-					cleanup(ifdlist);
-					return 1;
-				}
-			}
-			if(got==1){
-				p_filesamp = inframe;
-				for(k=0;k < inchans;k++)	//channel loop
-					*p_framesamp++	= *p_filesamp++;
-			}
-			else
-				p_framesamp += inchans;
-		}
-		
-		if(psf_sndWriteFloatFrames(ofd,outframe,1) < 0){
-			fprintf(stderr,"interlx: error writing to outfile\n");
-			psf_sndClose(ofd);            
-            cleanup(ifdlist);
-            return 1;
-		}
-        if(i % halfsec==0) {
-			printf("%.2lf secs\r",(double)i / (double) firstinprops.srate);
-            fflush(stdout);
-        }
-	}
-	printf("%.4lf secs\nWritten %ld sample frames to %s\n",(double)outframesize / (double)firstinprops.srate,outframesize,argv[1]);
-	if(psf_sndReadPeaks( ofd,fpeaks,&peaktime)){
-        printf("PEAK data:\n");
-        for(i = 0; i < firstinprops.chans; i++) {
-            double val, dbval;
-            val = (double) fpeaks[i].val;
-            dbval = 20.0 * log10(val);
-            printf("CH %d: %.6f (%.2lfdB) at frame %u:\t%.4f secs\n",i,
-                   val,dbval,(unsigned int) fpeaks[i].pos,(double)fpeaks[i].pos / (double) firstinprops.srate);
-        }
-    }
-	
-	printf("\n");
-	psf_sndClose(ofd);
-	free(inframe);
-	free(outframe);
-    if(fpeaks)
-        free(fpeaks);
-	cleanup(ifdlist);
-    
-	return 0;
-}

dev/externals/mctools/njoin.c

@@ -1,698 +0,0 @@
-/*
- * Copyright (c) 1983-2013 Richard Dobson and Composers Desktop Project Ltd
- * http://people.bath.ac.uk/masrwd
- * http://www.composersdesktop.com
- * This file is part of the CDP System.
- * The CDP System is free software; you can redistribute it
- * and/or modify it under the terms of the GNU Lesser General Public 
- * License as published by the Free Software Foundation; either 
- * version 2.1 of the License, or (at your option) any later version. 
- *
- * The CDP System is distributed in the hope that it will be useful, 
- * but WITHOUT ANY WARRANTY; without even the implied warranty of 
- * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  
- * See the GNU Lesser General Public License for more details.
- * You should have received a copy of the GNU Lesser General Public
- * License along with the CDP System; if not, write to the Free Software
- * Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
- *
- */
- 
-/* njoin.c: concantenate files with optional spacing */
-/* 12 Dec 2006 v 0.7: fixed bug in read_filelist: trap leading spaces on line */
-/* OCT 2009: v1.0 support tilde-prefixed path under unix */
-/* Jan 2010 v1.0.1 fixed bug processing lots of files and running out of portsf slots */
-/* Nov 2013 recognise MC_SURR_6_1 */
-/* Mar 2021  fix small file format type mismatch error (line 543 */
-#include <stdio.h>
-#include <stdlib.h>
-#include <math.h>
-#include <memory.h>
-#include <string.h>
-#include <assert.h>
-#ifdef unix
-#include <glob.h>
-#endif
-#include "portsf.h"
-
-
-#ifdef unix
-/* in portsf.lib */
-extern int stricmp(const char *a, const char *b);
-#endif
-
-#define F_MAXLEN (1024) 
-/* For CERN will need to increase this a lot! */
-#define MAXFILES (512)
-
-
-enum {ARG_PROGNAME, ARG_FLIST, ARG_NARGS};
-enum {ARG_OUTFILE = ARG_NARGS};
-
-int read_filelist(FILE* ifp, char* p_flist[]);
-void cleanup(int ifd, int ofd,char* flist[], int nfiles);
-
-#ifdef unix
-char* CreatePathByExpandingTildePath(char* path)
-{
-    glob_t globbuf;
-    char **v;
-    char *expandedPath = NULL, *result = NULL;
-    
-    assert(path != NULL);
-    
-    if (glob(path, GLOB_TILDE, NULL, &globbuf) == 0) //success
-    {
-        v = globbuf.gl_pathv; //list of matched pathnames
-        expandedPath = v[0]; //number of matched pathnames, gl_pathc == 1
-        
-        result = (char*)calloc( strlen(expandedPath) + 1,sizeof(char)); //the extra char is for the null-termination
-        if(result)
-            strncpy(result, expandedPath, strlen(expandedPath) + 1); //copy the null-termination as well
-        
-        globfree(&globbuf);
-    }
-    
-    return result;
-}
-#endif
-
-void
-usage()
-{
-	fprintf(stderr,"\nCDP MCTOOLS: NJOIN v1.1.1 (c) RWD,CDP 2006,2010,2013,2021\n"
-			"concatenate multiple files into a single file\n"
-			"Usage: njoin [-sSECS | -SSECS][-cCUEFILE][-x] filelist.txt [outfile] \n"						
-			"       filelist.txt: text file containing list of sfiles\n"
-			"                     in order. One file per line. \n"
-			"                     Channel spec (if present) must be the same,\n"
-			"                     but files with no spec assumed compatible.\n"
-			"       -cCUEFILE   : if outfile used, generate cue textfile as CUEFILE.\n"
-			"       -sSECS      : separate files with silence of SECS seconds\n"
-			"       -SSECS      : as above, but no silence before first file.\n"
-			"                     Default: files are joined with no gap.\n"
-            "      -x           : strict: allow only CD-compatible files:\n"
-            "                     Must use sr=44100, minimum duration 4 secs.\n"
-			"       NB: Files must match sample rate and number of channels,\n"
-			"        but can have different sample types.\n"
-			"       Output sample format taken from file with highest precision.\n"
-			"       If no outfile given: program scans files and prints report.\n"
-#ifdef unix
-            "       Unix systems:  ~/ notation for home dir supported for file paths.\n"
-#endif
-			);	
-}
-
-void cleanup(int ifd, int ofd,char* flist[], int nfiles)
-{
-	int i;
-	for(i=0;i < nfiles; i++) {		
-		if(flist[i])
-			free(flist[i]);
-	}
-	if(ifd >=0)
-		psf_sndClose(ifd);
-	if(ofd >=0)
-		psf_sndClose(ofd);	
-}
-/*SHORT8,SHORT16,FLOAT32,INT_32,INT2424,INT2432,INT2024,INT_MASKED*/
-static int wordsize[] = {1,2,4,4,3,4,3};
-
-/*return 0 for props2 same or less, 1 for props2 higher*/
-int compare_precision(const PSF_PROPS* props1,const PSF_PROPS* props2)
-{
-	int retval = 0;
-	switch(props1->samptype){
-	case(PSF_SAMP_8):
-		if(props2->samptype != props1->samptype)
-			retval = 1;
-		break;
-	case(PSF_SAMP_16):
-		if(props2->samptype > props1->samptype)
-			retval = 1;
-		break;
-	case(PSF_SAMP_IEEE_FLOAT):
-		/* only higher prec is 32bit int */
-		if(props2->samptype== PSF_SAMP_32)
-			retval = 1;
-		break;
-	case(PSF_SAMP_32):
-		/* nothing higher than this!*/
-		break;
-	case(PSF_SAMP_24):
-	//case(INT2432): // NB illegal for almost all formats!
-	//case(INT2024):
-		if(props2->samptype == PSF_SAMP_IEEE_FLOAT || props2->samptype == PSF_SAMP_32)
-			retval = 1;
-		break;
-	default:  
-		break;
-	}
-	return retval;
-}
-
-const char* stype_as_string(const PSF_PROPS* props)
-{
-	const char* msg;
-	switch(props->samptype){
-	case(PSF_SAMP_8):
-		msg = "8-bit";
-		break;
-	case(PSF_SAMP_16):
-		msg = "16-bit";
-		break;
-	case(PSF_SAMP_IEEE_FLOAT):
-		msg = "32-bit floats";
-		break;
-	case(PSF_SAMP_32):
-		msg = "32-bit integer";
-		break;
-	case(PSF_SAMP_24):
-		msg = "24-bit";
-		break;
-	default:
-		msg = "unknown WAVE_EX sample size!";
-		break;
-	}
-	return msg;
-}
-//STDWAVE,MC_STD,MC_MONO,MC_STEREO,MC_QUAD,MC_LCRS,MC_BFMT,MC_DOLBY_5_1,MC_WAVE_EX 
-const char* chformat_as_string(const PSF_PROPS* props)
-{
-	const char* msg;
-	switch(props->chformat){
-	case MC_STD:
-		msg = "Generic WAVE-EX";
-		break;
-	case MC_MONO:
-		msg = "WAVE_EX Mono";
-		break;
-	case MC_STEREO:
-		msg = "WAVE_EX Stereo";
-		break;
-	case MC_QUAD:
-		msg = "WAVE_EX Quad";
-		break;
-	case MC_LCRS:
-		msg = "WAVE_EX LCRS Surround";
-		break;
-	case MC_BFMT:
-		msg = "WAVE_EX B-Format";
-		break;
-	case MC_DOLBY_5_1:
-		msg = "WAVE_EX Dolby 5.1";
-		break;
-	case MC_SURR_6_1:
-	    msg =  "6.1 surround";
-	    break;
-    case MC_SURR_5_0:
-        msg = "5.0 surround";
-        break;
-    case MC_SURR_7_1:
-        msg = "7.1 Surround";
-        break;
-    case MC_CUBE:
-        msg = "Cube Surround";
-        break;
-	case MC_WAVE_EX:
-		msg = "WAVE-EX Custom Multi-Channel";
-		break;
-    default:  // STDWAVE
-        msg = "Standard soundfile";
-        break;
-	}
-
-	return msg;
-}
-
-int read_filelist(FILE* ifp, char* p_flist[])
-{
-	char buf[F_MAXLEN];
-	long len;
-	int nfiles = 0;
-#ifdef _DEBUG
-	assert(ifp);
-	assert(p_flist);
-#endif
-	if(ifp==NULL || p_flist == NULL)
-		return -1;
-	while (fgets(buf,F_MAXLEN-1,ifp)){
-		char* pbuf = buf;
-		while(*pbuf == ' ')
-			pbuf++;
-		len = strlen(pbuf);
-		if(len > 1){	// line has at least a eol byte
-			p_flist[nfiles] = malloc(len+1);
-			strcpy(p_flist[nfiles],pbuf);
-			if(p_flist[nfiles][len-1] == 0x0A)
-				p_flist[nfiles][len-1] = '\0';
-			nfiles++;
-		}
-		if(feof(ifp))
-			break;
-		if(ferror(ifp))
-			return -1;
-	}	
-	return nfiles;
-}
- 
-int strict_check(PSF_PROPS* props,unsigned long dur)
-{
-    int ret = 0;
-    unsigned long mindur = 4 * 44100;
-    
-    if(props->srate==44100 && dur >= mindur && props->chans==2)
-        ret = 1;
-    return ret;
-}
-
-int main(int argc, char* argv[])
-{
-	int i = 0,j,ofd = -1;
-	int ifd = -1;
-	char* flist[MAXFILES];
-	char* cuefilename = NULL;
-	FILE* cuefp = NULL;
-	int num_infiles = 0;
-	float *inframe = NULL;
-    float*space_frame = NULL;
-	PSF_PROPS inprops, thisinprops,outprops;
-	PSF_CHPEAK *fpeaks = NULL;
-	double space_secs = 0.0;
-	long space_frames = 0;
-    long thisdur = 0;
-	double totaldur = 0.0;
-	FILE* fp = NULL;
-	int formatsrc = 0;
-	unsigned int max_datachunk = 0xFFFFFFFFU - 1024U;   /* check Ok for PEAK chunk */
-	double maxdur;
-	double blockdur = 0.25;  /* match buffersize to srate, so we get tidy upodate msgs */
-	long buflen,block_frames;
-	unsigned long written;
-	int error = 0;
-	int do_process = 1;
-	int have_s = 0, have_S = 0;
-    int strict  =0;
-    int strict_failures = 0;
-    char* fname;
-
-    /* CDP version number */
-    if(argc==2 && (stricmp(argv[1],"--version")==0)){
-        printf("1.1.0\n");
-        return 0;
-    }
-    
-	if(argc < ARG_NARGS){
-		fprintf(stderr,"njoin: insufficient arguemnts\n");
-		usage();
-		return 1;
-	}
-	while(argv[1][0] =='-'){	
-		switch(argv[1][1]){
-		case('c'):
-			if(argv[1][2]== '\0'){
-				fprintf(stderr,"-c flag needs filename.\n");
-				return 1;
-			}
-			cuefilename = &(argv[1][2]);
-			break;
-		case('s'):
-			if(have_S){
-				fprintf(stderr,"njoin: cannot have both -s and -S.\n");
-				return 1;
-			}
-			space_secs = atof(&argv[1][2]);
-			if(space_secs < 0.0){
-				fprintf(stderr,"njoin: -tSECS cannot be negative!\n");
-				return 1;
-			}
-			have_s = 1;
-			break;
-		case('S'):
-			if(have_s){
-				fprintf(stderr,"njoin: cannot have both -s and -S.\n");
-				return 1;
-			}
-			space_secs = atof(&argv[1][2]);
-			if(space_secs < 0.0){
-				fprintf(stderr,"njoin: -tSECS cannot be negative!\n");
-				return 1;
-			}
-			have_S = 1;
-			break;
-        case 'x':
-            strict = 1;
-            break;
-		default:
-			fprintf(stderr,"\nnjoin: illegal flag option %s",argv[1]);
-			exit(1);
-		}
-		argc--; argv++;
-	}
-
-	if(argc < ARG_NARGS){
-		fprintf(stderr,"njoin: insufficient arguemnts\n");
-		usage();
-		return 1;
-	}
-	
-	if(argc==ARG_NARGS)
-		do_process = 0;
-	/********** READ filelist *********/
-
-	fp = fopen(argv[ARG_FLIST],"r");
-	if(fp==NULL){
-		fprintf(stderr,"njoin: Unable to open input file %s\n",argv[ARG_FLIST]);
-		return 1;
-	}
-	
-	memset(flist,0,MAXFILES * sizeof(char*));
-	num_infiles = read_filelist(fp, flist);
-	if(num_infiles < 0){
-		fprintf(stderr,"njoin: file error reading filelist %s\n",argv[ARG_FLIST]);
-		fclose(fp);
-		return 1;
-	}
-	if(num_infiles ==0){
-		fprintf(stderr,"njoin: filelist is empty!\n");
-		fclose(fp);
-		return 1;
-	}
-	if(num_infiles ==1){
-		fprintf(stderr,"njoin: only one file listed - nothing to do!\n");
-		fclose(fp);
-		return 1;
-	}
-	fclose(fp); fp = NULL;
-
-#ifdef _DEBUG
-	fprintf(stderr, "file list contains %d files: \n",num_infiles);	
-	for(i=0;i < num_infiles; i++)
-		fprintf(stderr,"%s\n",flist[i]);
-#endif
-
-	/********* open and check all soundfiles ***********/
-	fprintf(stderr,"checking files...\n");
-	
-	if(psf_init()){
-		fprintf(stderr,"njoin: startup failure.\n");
-		return 1;
-	}
-    i = 0;
-#ifdef unix
-    fname = CreatePathByExpandingTildePath(flist[i]);
-    /* must free pointer later */
-#else
-    fname = flist[i];
-#endif
-	//open first infile and get properties
-	ifd = psf_sndOpen(fname,&inprops,0);
-	if(ifd < 0){
-		fprintf(stderr,"unable to open infile %s.\n",fname);
-		cleanup(ifd,ofd,flist,num_infiles);
-		return 1;
-	}
-	thisdur = psf_sndSize(ifd);
-    if(strict){
-        if(!strict_check(&inprops,thisdur)){
-            fprintf(stderr,"Strict: file %s is not CD-compatible.\n",fname);
-            strict_failures++;
-        }
-    }
-	if(thisdur==0){
-		fprintf(stderr,"WARNING: file 1 empty: %s\n",fname);
-	}
-	else  {	
-		totaldur += (double) thisdur / inprops.srate;
-	}
-	psf_sndClose(ifd);
-#ifdef unix
-    free(fname);
-#endif
-	ifd = -1;
-	/* scan firther files, find max precision */
-	/* drop out if channel formats different */
-	for(i=1; i <	num_infiles; i++){
-#ifdef unix
-        fname = CreatePathByExpandingTildePath(flist[i]);
-        /* must free pointer later */
-#else
-        fname = flist[i];
-#endif
-        
-        
-		ifd = psf_sndOpen(fname,&thisinprops,0);
-		if(ifd < 0){
-			fprintf(stderr,"unable to open infile %s.\n",fname);
-			cleanup(ifd,ofd,flist,num_infiles);
-			exit(1);
-		}
-        thisdur = psf_sndSize(ifd);
-        if(strict){
-            if(!strict_check(&thisinprops,thisdur)){
-                fprintf(stderr,"Strict: file %s is not CD-compatible.\n",fname);
-                strict_failures++;
-            }
-        }
-		if(inprops.chans != thisinprops.chans){
-			fprintf(stderr,"njoin: channel mismatch in file %s",fname);
-			cleanup(ifd,ofd,flist,num_infiles);
-#ifdef unix
-            free(fname);
-#endif
-			return 1;
-		}
-		if(inprops.srate != thisinprops.srate){
-			fprintf(stderr,"njoin: sample rate mismatch in file %s",fname);
-			cleanup(ifd,ofd,flist,num_infiles);
-#ifdef unix
-            free(fname);
-#endif
-			return 1;
-		}
-		/* allow old multichannel files to be compatible with everything! */
-		if(! (inprops.chformat==(psf_channelformat)PSF_STDWAVE || thisinprops.chformat==(psf_channelformat)PSF_STDWAVE)){
-			if(inprops.chformat != thisinprops.chformat){
-				fprintf(stderr,"njoin: channel format mismatch in file %s",fname);
-				cleanup(ifd,ofd,flist,num_infiles);
-#ifdef unix
-                free(fname);
-#endif
-				return 1;
-			}
-		}
-		else {
-			/* one file is generic: promote format if possible*/
-			if(thisinprops.chformat > inprops.chformat)
-				inprops.chformat = thisinprops.chformat;
-		}
-
-		/* compare wordlength precision */
-		if(compare_precision(&inprops,&thisinprops)) {
-			inprops = thisinprops;
-			formatsrc = i;
-		}
-		thisdur = psf_sndSize(ifd);
-		if(thisdur==0){
-			fprintf(stderr,"WARNING: file %d empty: %s\n",i+1,fname);
-		}
-		else  {	
-			totaldur += (double) thisdur / thisinprops.srate;
-		}
-		psf_sndClose(ifd);
-		ifd = -1;
-#ifdef unix
-        free(fname);
-#endif
-	}
-    if(strict_failures){ 
-        fprintf(stderr,"Strict: %d files are CD-incompatible. Exiting.\n",strict_failures);
-        return 1;
-    }
-    
-	fprintf(stderr, "output format taken from file %d:\n\t%s\n",formatsrc+1,flist[formatsrc]);
-	fprintf(stderr, "sample type: %s\n",stype_as_string(&inprops));  
-	fprintf(stderr,"channel format: %s\n",chformat_as_string(&inprops));
-
-	maxdur = (double)(max_datachunk/ inprops.chans / wordsize[inprops.samptype]) / inprops.srate;
-	/*TODO: make sure we allow for size of PEAK chunk */
-	fprintf(stderr, "Max duration available for this format: %f secs.\n",maxdur);
-	if(have_S)
-		fprintf(stderr,"Total outfile length including spacing: %f secs\n", totaldur + (num_infiles-1) * space_secs);
-	else
-		fprintf(stderr,"Total outfile length including spacing: %f secs\n", totaldur + (num_infiles) * space_secs);
-
-	if(do_process == 0){
-		if(totaldur > maxdur)
-			fprintf(stderr, "Error: total duration exceeds capacity of file format.\n");
-		cleanup(ifd,ofd,flist,num_infiles);
-		return 0;
-	}
-
-	if(totaldur > maxdur){
-		fprintf(stderr, "Sorry! Total duration exceeds capacity of file format.\nProcess aborted.\n");
-		cleanup(ifd,ofd,flist,num_infiles);
-		return 1;
-	}
-    
-	/* if here, OK! We can make the file */
-    printf("processing files...\n");
-	outprops = inprops;
-	/* try to make a legal wave file! */
-	if((outprops.chans > 2 || outprops.samptype > PSF_SAMP_IEEE_FLOAT)
-		 && (outprops.format==PSF_STDWAVE))
-        outprops.chformat = /* PSF_WAVE_EX */ MC_STD;       //RWD 10:03:21
-    block_frames = (long)(blockdur * outprops.srate);
-	buflen = block_frames * outprops.chans;
-	inframe = malloc(buflen * sizeof(float));
-	if(inframe==NULL){
-		puts("No memory!\n");
-		cleanup(ifd,ofd,flist,num_infiles);
-		return 1;
-	}
-	//setup PEAK data
-	fpeaks = (PSF_CHPEAK *) calloc(outprops.chans,sizeof(PSF_CHPEAK));
-	if(fpeaks==NULL){
-		puts("njoin: error: no memory for internal PEAK buffer\n");
-		cleanup(ifd,ofd,flist,num_infiles);
-		return 1;
-	}
-
-	if(cuefilename){
-		cuefp = fopen(cuefilename, "w");
-		if(cuefp == NULL){
-			fprintf(stderr, "WARNING: unable to create cue file %s.\n",cuefilename);
-			cuefilename = NULL;
-		}
-	}
-
-    space_frame = calloc(sizeof(float),outprops.chans);
-    
-	ofd = psf_sndCreate(argv[ARG_OUTFILE],&outprops,0,0,PSF_CREATE_RDWR);
-	if(ofd < 0){
-		fprintf(stderr,"njoin: Cannot create outfile %s\n",argv[ARG_OUTFILE]);
-		cleanup(ifd,ofd,flist,num_infiles);
-		return 1;
-	}
-	fprintf(stderr, "generating outfile...\n");
-	written = 0;
-	if(cuefp) {
-		//fprintf(cuefp,"FILE %s WAVE\n",snd_getfilename(ofd));
-        fprintf(cuefp,"FILE %s WAVE\n",argv[ARG_OUTFILE]);
-		fprintf(cuefp,"\tTRACK 01 AUDIO\n");
-				fprintf(cuefp,"\t\tINDEX 01 00:00:00\n");
-	}
-	space_frames = (long) (space_secs * outprops.srate + 0.5);
-	/* add leading space ? */
-	if(have_s && space_frames > 0){
-		for(j=0; j < space_frames; j++) {
-			if(psf_sndWriteFloatFrames(ofd,space_frame,1) < 0){
-				fprintf(stderr,"njoin: error writing outfile\n");
-				cleanup(ifd,ofd,flist,num_infiles);
-				free(inframe);
-				free(fpeaks);
-				return 1;
-			}
-		}
-		written += space_frames;
-	}
-	for(i=0; i < num_infiles; i++){
-		long got, put;
-        PSF_PROPS fprops; // dummy - not needed 
-#ifdef unix
-        fname = CreatePathByExpandingTildePath(flist[i]);
-        /* must free pointer later */
-        /*RWD TODO: may fail with null return if file does not exist */
-#else
-        
-        fname = flist[i];
-#endif        
-		ifd = psf_sndOpen(fname,&fprops,0);
-		if(ifd < 0){
-			fprintf(stderr,"unable to open infile %s.\n",fname);
-			error++;
-			break;
-		} 
-        
-		do {			
-			got = psf_sndReadFloatFrames(ifd,inframe,block_frames);
-			if(got < 0){
-				fprintf(stderr,"njoin: error reading file %s\n",fname);
-				error++;
-				break;
-			}
-			
-			put = psf_sndWriteFloatFrames(ofd,inframe,got);
-			if(put != got){
-				fprintf(stderr,"njoin: error writing outfile\n");
-				error++;
-				break;
-			}
-			written += got;
-			fprintf(stderr,"%.2f\r",(double) written / outprops.srate);
-		} while (got > 0);
-		if(error)
-			break;
-		
-		/* add space */
-		if(i < num_infiles - 1){
-			/* update cue file */
-			if(cuefp){
-				double pos = (double)(psf_sndTell(ofd)) / outprops.srate;
-				int mins = (int)(pos / 60.0);
-				int secs = (int)(pos - mins);
-				int frames = (int) ((pos - (60.0 * mins + secs)) * 75.0);
-
-				fprintf(cuefp,"\tTRACK %.2d AUDIO\n",i+2);
-				fprintf(cuefp,"\t\tINDEX 01 %.2d:%.2d:%.2d\n",mins,secs,frames); 
-			}
-			
-			for(j=0; j < space_frames ; j++) {
-				if(psf_sndWriteFloatFrames(ofd,space_frame,1) < 0){
-					fprintf(stderr,"njoin: error writing outfile\n");
-					error++;
-					break;
-				}
-			}
-		}
-		if(error)
-			break;
-		written += space_frames;
-#ifdef unix
-        free(fname);
-#endif
-        /*RWD Jan 2010 MUST close the file or we may run out of portsf slots! */
-        psf_sndClose(ifd);
-        ifd = -1;
-	}
-
-	if(error){
-		fprintf(stderr,"Error: Outfile incomplete.\n");
-//		sndunlink(ofd);
-	}
-	else {
-		fprintf(stderr, "Done.\nWritten %ld frames to outfile\n",written);
-		if(psf_sndReadPeaks(ofd,fpeaks,NULL) > 0){
-            long i;
-            double peaktime;
-            printf("PEAK information:\n");	
-			for(i=0;i < inprops.chans;i++){
-                double val = fpeaks[i].val;
-				peaktime = (double) fpeaks[i].pos / (double) inprops.srate;
-                if(val==0.0)
-                    printf("CH %ld:\t%.4f           at %.4f secs\n", i+1, val,peaktime);
-                else
-                    printf("CH %ld:\t%.4f (%.2fdB) at %.4f secs\n", i+1, val, 20.0*log10(val),peaktime);
-			}
-        }
-	}
-	if(inframe)
-		free(inframe);
-	if(fpeaks)
-		free(fpeaks);
-	if(cuefp)
-		fclose(cuefp);
-	cleanup(ifd,ofd,flist,num_infiles);
-	psf_finish();
-	return 0;
-}

dev/externals/mctools/nmix.c

@@ -1,333 +0,0 @@
-/*
- * Copyright (c) 1983-2013 Richard Dobson and Composers Desktop Project Ltd
- * http://people.bath.ac.uk/masrwd
- * http://www.composersdesktop.com
- * This file is part of the CDP System.
- * The CDP System is free software; you can redistribute it
- * and/or modify it under the terms of the GNU Lesser General Public
- * License as published by the Free Software Foundation; either
- * version 2.1 of the License, or (at your option) any later version.
- *
- * The CDP System is distributed in the hope that it will be useful,
- * but WITHOUT ANY WARRANTY; without even the implied warranty of
- * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.
- * See the GNU Lesser General Public License for more details.
- * You should have received a copy of the GNU Lesser General Public
- * License along with the CDP System; if not, write to the Free Software
- * Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
- *
- */
-
-//nmix.c
-//Oct 2009 updated to use portsf
-#include <stdio.h>
-#include <stdlib.h>
-#include <string.h>
-#include <math.h>
-#include <memory.h>
-#include <sys/timeb.h>
-#include "portsf.h"
-
-#ifdef unix
-/* in portsf.lib */
-extern int stricmp(const char *a, const char *b);
-#endif
-
-//usage: nmix [-oOFFSET]  [-f] infile1 infile2 outfile
-void usage() {
-    printf("\nCDP MCTOOLS: NMIX V2.0.1 (c) RWD,CDP 1999,2009\n"
-           "mix two multi-channel files\n"
-           "usage:\n"
-           "nmix [-d][-f][-oOFFSET] infile1 infile2 outfile\n"
-           "     -d      : apply TPDF dither (16bit format output only).\n"
-           "     -f      : set output sample type to floats.\n"
-           "                Default: outfile type is that of infile 1\n"
-           "     -oOFFSET:  start infile2 at OFFSET seconds.\n"
-           "Files must have the same channel count.\n"
-           "WAVE-EX files must have the same speaker layout\n");
-
-    exit(0);
-}
-
-#define BUFLEN (1024)
-
-int main(int argc, char *argv[])
-{
-    int i,ifd1 = -1,ifd2=-1,ofd=-1;
-    int force_floats = 0;
-    int do_dither = 0;
-    float ampfac = 0.0;
-    long len_in1, len_in2,outlen;
-    double offset = 0.0;
-    long offsetframes = 0,framecount = 0;
-    long chans;
-    long sr;
-    long in1_sofar,in2_sofar;
-    int halfsec;
-    MYLONG peaktime;
-    PSF_PROPS props_in1,props_in2;
-    float *frame_in1 = NULL, *frame_in2 = NULL, *outframe = NULL;
-    PSF_CHPEAK *peaks = NULL;
-
-    /* CDP version number */
-    if(argc==2 && (stricmp(argv[1],"--version")==0)){
-        printf("2.0.1\n");
-        return 0;
-    }
-
-    if(argc < 4)
-        usage();
-
-    while(argv[1][0]=='-'){
-        char *valstr;
-        double val;
-        switch(argv[1][1]){
-        case 'd':
-            do_dither = 1;
-            break;
-        case('f'):
-            force_floats = 1;
-            break;
-        case('o'):
-            if(argv[1][2]=='\0'){
-                fprintf(stderr,"-o flag needs a number\n");
-                exit(1);
-            }
-            valstr = &argv[1][2];
-            val=atof(valstr);
-            if(val < 0.0){
-
-                fprintf(stderr,"offset must be >= 0\n");
-                exit(1);
-            }
-            if(val > 0.0)
-                offset = val;
-            break;
-
-        default:
-            fprintf(stderr,"incorrect flag option %c\n",argv[1][1]);
-            exit(1);
-        }
-        argc--; argv++;
-    }
-
-    if(argc < 4)
-        usage();
-    if(force_floats && do_dither)
-        fprintf(stderr,"Warning: dither option ignored for floats format.\n");
-
-    psf_init();
-
-    //open soundfiles: no auto-rescaling
-    if((ifd1 = psf_sndOpen(argv[1],&props_in1,0)) < 0){
-        fprintf(stderr,"Cannot open soundfile %s\n",argv[1]);
-        goto cleanup;
-    }
-
-    if((ifd2 = psf_sndOpen(argv[2],&props_in2,0)) < 0){
-        fprintf(stderr,"Cannot open soundfile %s \n",argv[2]);
-        goto cleanup;
-    }
-
-    if(props_in1.chans != props_in2.chans){
-        fprintf(stderr,"files do not have the same number of channels\n");
-        goto cleanup;
-    }
-
-    if(props_in1.srate != props_in2.srate){
-        fprintf(stderr,"files do not have the same sample rate\n");
-        goto cleanup;
-
-    }
-
-    if(props_in1.chformat != props_in2.chformat){
-        fprintf(stderr,"files do not have the same channel format\n");
-        goto cleanup;
-    }
-
-
-    sr = props_in1.srate;
-    chans  = props_in1.chans;
-    ampfac = 0.5f;                                  //just mixing 2 files...
-    if(offset > 0.0){
-        offsetframes = (long) (offset * (double) sr);
-    }
-    /* now we can set up frame arrays */
-
-    frame_in1 = (float *) calloc(chans, sizeof(float));
-    frame_in2 = (float *) calloc(chans, sizeof(float));
-    outframe = (float *) calloc(chans, sizeof(float));
-
-    if(frame_in1==NULL || frame_in2==NULL || outframe==NULL ){
-        puts("\nno memory for frame buffers");
-        goto cleanup;
-    }
-
-    len_in1 = psf_sndSize(ifd1);
-    if(len_in1==0){
-        fprintf(stderr,"infile %s is empty!\n",argv[1]);
-        goto cleanup;
-    }
-    len_in2 = psf_sndSize(ifd2);
-    if(len_in2==0){
-        fprintf(stderr,"infile %s is empty!\n",argv[2]);
-        goto cleanup;
-    }
-    if(len_in1 < 0){
-        fprintf(stderr,"system problem: cannot read size of infile %s\n",argv[1]);
-        goto cleanup;
-    }
-
-    if(len_in2 < 0){
-        fprintf(stderr,"system problem: cannot read size of infile %s\n",argv[2]);
-        goto cleanup;
-    }
-
-
-
-    outlen = len_in2 + offsetframes;
-    if(len_in1 > outlen)
-        outlen = len_in1;
-
-#ifdef _DEBUG
-    printf("DEBUG: outfile size expected to be %d frames\n",outlen);
-#endif
-    //setup PEAK data
-    peaks = (PSF_CHPEAK *) calloc(props_in1.chans,sizeof(PSF_CHPEAK));
-    if(peaks==NULL){
-        puts("nmix: error: no memory for internal PEAK buffer\n");
-        goto cleanup;
-    }
-    if(force_floats)
-        props_in1.samptype = PSF_SAMP_IEEE_FLOAT;
-
-
-    if(!is_legalsize(outlen,&props_in1)){
-        fprintf(stderr,"error: outfile size exceeds capacity of format.\n");
-        return 1;
-    }
-
-
-    if((ofd = psf_sndCreate(argv[3],&props_in1,0,0,PSF_CREATE_RDWR)) < 0){
-        fprintf(stderr,"unable to create outfile %s\n",argv[3]);
-        goto cleanup;
-    }
-    if(do_dither)
-        psf_sndSetDither(ofd,PSF_DITHER_TPDF);
-    halfsec = sr / 2;
-    //OK now we can do it....
-    printf("\nmixing....\n");
-
-    in1_sofar = in2_sofar = 0;
-    if(offsetframes > 0){
-        for(i=0;i < offsetframes; i++){
-            int got,j;
-            got = psf_sndReadFloatFrames(ifd1,frame_in1,1);
-            if(got != 1){
-                fprintf(stderr,"error reading from infile 1\n");
-                goto cleanup;
-            }
-            for(j=0;j < chans; j++) {
-                frame_in1[j] *= ampfac;
-            }
-            if((psf_sndWriteFloatFrames(ofd,frame_in1,1)) !=1){
-                fprintf(stderr,"\nerror writing to outfile\n");
-                goto cleanup;
-            }
-            if(framecount % halfsec==0)
-                printf("%.2lf secs\r",(double)framecount / (double) sr);
-
-            framecount++;
-        }
-    }
-    in1_sofar = offsetframes;
-    //now we are mixing two files...
-    for(i= offsetframes; i < outlen; i++){
-        int got, j;
-        //clear frame blocks
-        memset(frame_in1,0, chans * sizeof(float));
-        memset(frame_in2,0, chans * sizeof(float));
-        //if we  have data, fill frame and scale
-        if(in1_sofar < len_in1){
-            got = psf_sndReadFloatFrames(ifd1,frame_in1,1);
-            if(got != 1){
-                fprintf(stderr,"\nerror reading from infile 1\n");
-                goto cleanup;
-            }
-            in1_sofar++;
-            for(j=0;j < chans; j++)
-                frame_in1[j] *= ampfac;
-        }
-
-
-        if(in2_sofar < len_in2){
-            got = psf_sndReadFloatFrames(ifd2,frame_in2,1);
-            if(got != 1){
-                fprintf(stderr,"\nerror reading from infile 1\n");
-                goto cleanup;
-            }
-
-            in2_sofar++;
-
-            for(j=0;j < chans; j++)
-                frame_in2[j] *= ampfac;
-
-        }
-
-        //mix and write
-        for(j=0;j < chans; j++) {
-            outframe[j] = frame_in1[j] + frame_in2[j];
-
-        }
-        if(psf_sndWriteFloatFrames(ofd,outframe,1) < 0){
-            fprintf(stderr,"\nerror writing to outfile\n");
-            goto cleanup;
-        }
-        if(framecount % halfsec==0) {
-            printf("%.2lf secs\r",(double)framecount / (double) sr);
-            fflush(stdout);
-        }
-        framecount++;
-    }
-    printf("%.4lf secs\nWritten %ld sample frames to %s\n",(double)framecount / (double)sr,framecount,argv[3]);
-    if(psf_sndReadPeaks( ofd,peaks,&peaktime)){
-        printf("PEAK values:\n");
-        for(i=0; i < chans; i++){
-            double val, dbval;
-            val = (double) peaks[i].val;
-
-            if(val > 0.0){
-                dbval = 20.0 * log10(val);
-                printf("CH %d: %.6f (%.2lfdB) at frame %u:\t%.4f secs\n",i,
-                       val,dbval,(unsigned int) peaks[i].pos,(double)peaks[i].pos / (double) props_in1.srate);
-            }
-            else{
-                printf("CH %d: %.6f (-infdB) at frame %u:\t%.4f secs\n",i,
-                       val,(unsigned int) peaks[i].pos,(double)peaks[i].pos / (double) props_in1.srate);
-            }
-        }
-    }
-
-
-
- cleanup:
-    if(ifd1 >=0)
-        psf_sndClose(ifd1);
-    if(ifd2 >=0)
-        psf_sndClose(ifd2);
-    if(ofd >=0)
-        psf_sndClose(ofd);
-    if(peaks)
-        free(peaks);
-    if(frame_in1 != NULL)
-        free(frame_in1);
-
-    if(frame_in2 != NULL)
-        free(frame_in2);
-    if(outframe != NULL)
-        free(outframe);
-
-
-    psf_finish();
-    return 0;
-}

dev/externals/mctools/rmsinfo.cpp

@@ -1,320 +0,0 @@
-/*
- * Copyright (c) 1983-2013 Richard Dobson and Composers Desktop Project Ltd
- * http://people.bath.ac.uk/masrwd
- * http://www.composersdesktop.com
- * This file is part of the CDP System.
- * The CDP System is free software; you can redistribute it
- * and/or modify it under the terms of the GNU Lesser General Public 
- * License as published by the Free Software Foundation; either 
- * version 2.1 of the License, or (at your option) any later version. 
- *
- * The CDP System is distributed in the hope that it will be useful, 
- * but WITHOUT ANY WARRANTY; without even the implied warranty of 
- * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  
- * See the GNU Lesser General Public License for more details.
- * You should have received a copy of the GNU Lesser General Public
- * License along with the CDP System; if not, write to the Free Software
- * Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
- *
- */
-
-// rmsinfo
-#include <stdio.h>
-#include <stdlib.h>
-#include <string.h>
-#include <portsf.h>
-#include <math.h>
-#include <signal.h>
-#ifdef WIN32
-# if _MSC_VER  && _MSC_VER <= 1200
-# include <new.h>
-# endif
-#endif
-
-
-#ifdef unix
-/* in portsf.lib */
-extern "C" {
-    int stricmp(const char *a, const char *b);
-}
-#endif
-
-enum {ARG_NAME, ARG_INFILE,ARG_NARGS};
-
-enum {CH_RMS_POWER,CH_AMPSUM,CH_BISUM,CH_NORM_RMS,CH_NORM_AMPSUM};
-
-typedef struct {
-    double rmspower;
-    double abssum;
-    double bisum;
-    double norm_rms;
-    double norm_abssum;
-} CH_RMSINFO;
-
-#define BUFLEN (1024)
-
-static int scanning(1);
-
-void runhandler(int sig)
-{
-	if(sig == SIGINT){
-		scanning = 0;
-    }
-}
-
-void usage(){
-    printf("\nCDP MCTOOLS: RMSINFO v1.0.1 (c) RWD, CDP 2009\n");
-    printf("Scans infile and reports rms and average loudness (power) of infile,\n"
-           " relative to digital peak (0dBFS)\n"
-           );
-    printf("Usage: rmsinfo [-n] infile [startpos [endpos]]\n"
-           "Standard output shows:\n"
-           "     RMS level\n"
-           "     Average level\n"
-           "     DC level (bipolar average)\n"
-           "-n : Include equivalent 0dBFS-normalised RMS and AVG levels.\n"
-           );
-    printf("Optional arguments:\n"
-           "startpos  :  start file scan from <startpos> seconds.\n"
-           "endpos    :  finish file scan at <endpos> seconds.\n"
-           );
-    printf("To stop a scan early, use CTRL-C.\n"
-           "Program will report levels up to that point.\n\n"
-           );
-}
-#ifdef WIN32
-#if _MSC_VER  && _MSC_VER <= 1200
-
- int newhandler(size_t size);
- class bad_alloc{};
- int newhandler(size_t size)
- {
-     throw bad_alloc();
- }
-#endif
-#endif
-
-
-
-int main(int argc, char**argv)  {
-    long inframes,framesread=0,total_framesread;
-    double maxsamp = 0.0,startpos = 0.0,endpos;
-    long startframe = 0,endframe;
-    long halfsecframes = 0;
-    int i,j,ifd = -1;
-    int chans;
-    int do_norm = 0;
-	PSF_PROPS inprops;
-    CH_RMSINFO* rmsinfo = NULL;
-    double* rmsfac  = 0;
-    double* ampsum  = 0;
-    double* ampsumb = 0;
-    double* inbuf   = 0;
-    double* nrmsfac = 0;
-    double* nampsum = 0;
-    
-    
-    /* CDP version number */
-    if(argc==2 && (stricmp(argv[1],"--version")==0)){
-        printf("1.0.1\n");
-        return 0;
-    }
-#ifdef WIN32    
-# if _MSC_VER  && _MSC_VER <= 1200
-     _set_new_handler( newhandler );
-# endif
-#endif
-    if(psf_init()){
-		puts("unable to start portsf\n");
-		return 1;
-	}
-    if(argc < 2){		 
-		usage();
-		return(1);
-	}
-    while(argv[1][0]=='-'){
-		switch(argv[1][1]){
-        case 'n':
-            do_norm = 1;
-            break;
-        default:
-            fprintf(stderr, "Unrecognised flag option %s\n",argv[1]);
-            return 1;
-        }
-        argc--; argv++;
-    }
-    if(argc < 2){		 
-		usage();
-		return(1);
-	}
-    if((ifd = psf_sndOpen(argv[ARG_INFILE],&inprops, 0)) < 0){
-		fprintf(stderr,"\nUnable to open input soundfile %s",argv[ARG_INFILE]);
-		return(1);
-	}
-	inframes = psf_sndSize(ifd);  // m/c frames
-    endframe = inframes;
-    
-	if(inframes <= 0)
-        return 0;
-	if(argc >= 3) {
-        long lpos;
-        startpos = atof(argv[ARG_INFILE+1]);
-        if(startpos < 0.0){
-            fprintf(stderr,"Error: startpos must be positive\n");
-            return 1;
-        }
-        lpos = (long)( startpos * inprops.srate);
-        if(lpos > inframes){
-            fprintf(stderr,"Error: startpos value beyond end of file.\n");
-            return 1;
-        }
-        startframe = lpos;
-    }
-    if(argc >= 4) {
-        long lpos;
-        endpos = atof(argv[ARG_INFILE+2]);
-        
-        lpos = (long)(endpos * inprops.srate);
-        if(lpos > inframes){
-            fprintf(stderr,"Warning: endpos value too large - reset to end of file.\n");
-            return 1;
-        }
-        endframe = lpos;
-        if(!(endframe > startframe)){
-            fprintf(stderr,"Error: endpos must be beyond startpos.\n");
-            return 1;
-        }
-    }
-    if(startframe)
-        printf("Starting at frame %ld, ending at frame %ld\n",startframe, endframe);
-    
-    chans   = inprops.chans;
-    try {
-        inbuf   = new double[BUFLEN * chans];
-        rmsinfo = new CH_RMSINFO[chans];
-        rmsfac  = new double[chans];
-        ampsum  = new double[chans];
-        ampsumb = new double[chans];
-        nrmsfac = new double[chans];
-        nampsum = new double[chans];
-    }
-    catch(...){
-        fputs("no memory!\n",stderr);
-        return 1;
-    }
-    for(i=0; i < chans; i++){
-        rmsfac[i]  = 0.0;
-        ampsum[i]  = 0.0;
-        ampsumb[i] = 0.0;
-        nrmsfac[i] = 0.0;
-        nampsum[i] = 0.0;
-    }
-    halfsecframes = inprops.srate / 2; 
-    signal(SIGINT,runhandler);
-    long wanted = endframe - startframe;
-    printf("Scanning %ld frames (%.3lf secs):\n",wanted, (double)wanted / inprops.srate);
-    total_framesread = 0;
-    if(startframe) {
-        if(psf_sndSeek(ifd,startframe,PSF_SEEK_SET)){
-            fprintf(stderr,"File Seek error.\n");
-            return 1;
-        }
-    }
-        
-	while((framesread = psf_sndReadDoubleFrames(ifd,inbuf,BUFLEN)) > 0){
-        double fval;
-        for(i = 0;i < framesread;i++) {
-            for(j = 0; j < chans; j++){
-                double val = inbuf[i*chans + j];
-                fval = fabs(val);
-                maxsamp = fval >  maxsamp ? fval : maxsamp;
-                ampsum[j]   += fval;
-                rmsfac[j]   += val*val; 
-                ampsumb[j]  += val;
-            }
-            total_framesread++;
-            if(scanning==0)
-                break;
-            if(total_framesread == wanted) 
-                break;
-            if((total_framesread % halfsecframes) == 0){
-                printf("%.2lf\r",total_framesread / (double) inprops.srate);
-                fflush(stdout);
-            }
-        }
-        if(total_framesread == wanted) {
-            break;
-        }
-    }
-    if(framesread < 0){
-        fprintf(stderr,"Error reading file.\n");
-        return 1;
-    }
-    for(i=0;i < chans;i++){
-        rmsfac[i]  /= total_framesread;
-        rmsfac[i]   = sqrt(rmsfac[i]);
-        ampsum[i]  /= total_framesread;
-        ampsumb[i] /= total_framesread;
-    }
-
-    double normfac = 1.0 / maxsamp;
-    if(scanning==0)
-        printf("\nScan stopped.\n");
-    if(total_framesread < inframes){
-        printf("Scanned %ld frames (%.2lf secs).\n",total_framesread,total_framesread / (double)inprops.srate);
-    }
-    
-    printf("Maximum sample = %lf (%.2lfdB)\n",maxsamp,20.0 * log10(maxsamp));
-    printf("Maximum normalisation factor = %.4f\n",normfac);
-    
-    for(i=0;i < chans;i++){
-        rmsinfo[i].rmspower    = rmsfac[i];
-        rmsinfo[i].abssum      = ampsum[i];
-        rmsinfo[i].bisum       = ampsumb[i];
-        rmsinfo[i].norm_rms    = normfac * rmsfac[i];
-        rmsinfo[i].norm_abssum = normfac * ampsum[i];
-    }
-    if(do_norm){
-        printf("\t   RMS LEVEL\t    AVG  \t   NET DC\t   NORM RMS\t  NORM AVG\n");
-        printf("CH\t AMP\t DB\t AMP\t DB\t AMP\t DB\t AMP\t DB\t AMP\t DB     \n");
-    }
-    else{
-        printf("\t   RMS LEVEL\t    AVG  \t   NET DC\n");
-        printf("CH\t AMP\t DB\t AMP\t DB\t AMP\t DB\n"); 
-    }
-    for(i=0;i < chans;i++){
-        double d1,d2,d3,d4,d5;
-        
-        d1 = 20*log10(rmsfac[i]);
-        d2 = 20*log10(ampsum[i]);
-        d3 = 20*log10(fabs(ampsumb[i]));
-        d4 = 20*log10(normfac * rmsfac[i]);
-        d5 = 20*log10(normfac * ampsum[i]);
-        if(do_norm){
-            printf("%d\t%.5lf\t%.2lf\t%.5lf\t%.2lf\t%+.4lf\t%.2lf\t%.5lf\t%.2lf\t%.5lf\t%.2lf\n",i+1,
-               rmsinfo[i].rmspower,d1,
-               rmsinfo[i].abssum,d2,
-               rmsinfo[i].bisum,d3,
-               rmsinfo[i].norm_rms,d4,
-               rmsinfo[i].norm_abssum,d5
-            );
-        }
-        else {
-            printf("%d\t%.5lf\t%.2lf\t%.5lf\t%.2lf\t%+.4lf\t%.2lf\n",i+1,
-                   rmsinfo[i].rmspower,d1,
-                   rmsinfo[i].abssum,d2,
-                   rmsinfo[i].bisum,d3
-            );
-        }
-    }
-        
-    delete [] inbuf;
-    delete [] rmsfac;
-    delete [] ampsum;
-    delete [] ampsumb;
-    delete [] rmsinfo;
-    psf_sndClose(ifd);
-    psf_finish();
-
-	return 0;
-}

dev/externals/mctools/sfprops.c

@@ -1,378 +0,0 @@
-/*
- * Copyright (c) 1983-2013 Richard Dobson and Composers Desktop Project Ltd
- * http://people.bath.ac.uk/masrwd
- * http://www.composersdesktop.com
- * This file is part of the CDP System.
- * The CDP System is free software; you can redistribute it
- * and/or modify it under the terms of the GNU Lesser General Public 
- * License as published by the Free Software Foundation; either 
- * version 2.1 of the License, or (at your option) any later version. 
- *
- * The CDP System is distributed in the hope that it will be useful, 
- * but WITHOUT ANY WARRANTY; without even the implied warranty of 
- * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  
- * See the GNU Lesser General Public License for more details.
- * You should have received a copy of the GNU Lesser General Public
- * License along with the CDP System; if not, write to the Free Software
- * Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
- *
- */
- 
-/* sfprops.c: display primary soundfile properties 
- * uses sfsysEx.lib: recognizes WAVE-EX formats
- * part of the CDP MCTOOLS suite
- */
-//RWD.6.99 use new funcs to report 24bit formats, etc
-//last update: 22.6.99 enumerate PEAK chans from 1, link with latest sfsysEx
-/* Nov28 2001: rebuild with sfsysEx recognizing stupid QuickTime AIFC floats with 16bit size! */
-/*Dec 2005 support .amb extension */
-/*April 2006: build with updated sfsys to read PEAK chunk after data chunk (Thanks to Sony!)*/
-/* OCt 2009 TODO: sort out 64bit platform issues (peaktime etc) */
-/* FEB 2010: decl filesize etc as unsigned to read huge file sizes! */
-/* Nov 2013 added MC_SURR_6_1 */
-#include <stdio.h>
-#include <stdlib.h>
-#include <string.h>
-#include <math.h>
-#include <ctype.h>
-#include <time.h>
-#include <sfsys.h>
-#include <chanmask.h>
-
-#ifdef unix
-
-int stricmp(const char *a, const char *b)
-{
-    while(*a != '\0' && *b != '\0') {
-        int ca = islower(*a) ? toupper(*a) : *a;
-        int cb = islower(*b) ? toupper(*b) : *b;
-        
-        if(ca < cb)
-            return -1;
-        if(ca > cb)
-            return 1;
-        
-        a++;
-        b++;
-    }
-    if(*a == '\0' && *b == '\0')
-        return 0;
-    if(*a != '\0')
-        return 1;
-    return -1;
-}
-#endif
-
-typedef struct wave_ex_speaker {
-    int mask;
-    char name[28];
-} WAVE_EX_SPEAKER;
-
-
-static const WAVE_EX_SPEAKER speakers[NUM_SPEAKER_POSITIONS] = {
-    {SPEAKER_FRONT_LEFT,"Front Left"},
-    {SPEAKER_FRONT_RIGHT,"Front Right"},
-    {SPEAKER_FRONT_CENTER,"Front Centre"},
-    {SPEAKER_LOW_FREQUENCY,"Low Frequency"},
-    {SPEAKER_BACK_LEFT,"Back Left"},
-    {SPEAKER_BACK_RIGHT,"Back Right"},
-    {SPEAKER_FRONT_LEFT_OF_CENTER,"Front Centre-Left"},
-    {SPEAKER_FRONT_RIGHT_OF_CENTER,"Front Centre-Right"},
-    {SPEAKER_BACK_CENTER,"Back Centre"},
-    {SPEAKER_SIDE_LEFT,"Side Left"},
-    {SPEAKER_SIDE_RIGHT,"Side Right"},
-    {SPEAKER_TOP_CENTER,"Top Centre"},
-    {SPEAKER_TOP_FRONT_LEFT,"Front Top Left"},
-    {SPEAKER_TOP_FRONT_CENTER,"Front Top Centre"},
-    {SPEAKER_TOP_FRONT_RIGHT,"Front Top Right"},
-    {SPEAKER_TOP_BACK_LEFT,"Back Top Left"},
-    {SPEAKER_TOP_BACK_CENTER,"Back Top Centre"},
-    {SPEAKER_TOP_BACK_RIGHT,"Back Top Right"}
-};
-
-void usage(){
-    fprintf(stderr,"CDP MCTOOLS: SFPROPS v2.2.0 (c) RWD,CDP,1999,2009,2010,2013\n"
-                    "Display soundfile details, with WAVE-EX speaker positions\n"
-                    "\n\tusage: sfprops infile\n");
-}
-
- 
-
-int main(int argc, char **argv)
-{
-
-    int i,ifd;
-    unsigned int nframes,filesize; /* FEB 2010 */
-    double srate;
-    SFPROPS props = {0};
-    const char *name = NULL;
-    CHPEAK *peaks = NULL;
-    int res,peaktime,speakermask = 0;
-    float fmaxamp;
-    int lmaxamp;
-    
-    /* CDP version number */
-    if(argc==2 && (stricmp(argv[1],"--version")==0)){
-        printf("2.2.0.\n");
-        return 0;
-    }
-    
-    if(argc< 2){
-        usage();
-        exit(1);
-    }
-    if(sflinit("sfprops")){
-        fprintf(stderr,"SFPROPS: unable to initialize CDP Sound Filing System\n");
-        exit(1);
-    }
-    if((ifd = sndopenEx(argv[1],0,CDP_OPEN_RDONLY)) < 0){
-        fprintf(stderr,"SFPROPS: unable to open infile %s: %s\n",argv[1], sferrstr());
-        exit(1);
-    }
-
-    if(!snd_headread(ifd,&props)){
-        fprintf(stderr,"SFPROPS: unable to read infile header\n");
-        exit(1);
-    }
-
-    srate = (double) props.srate;
-    filesize = sndsizeEx(ifd);
-    nframes = filesize/ props.chans;        /*assume soundfile for now */
-    name = snd_getfilename(ifd);
-    printf("Properties of %s:\n",name);
-    printf("File type:  ");
-    switch(props.type){
-    case(wt_wave):
-        printf("soundfile\nFormat         :  ");
-
-        switch(props.format){
-        case(WAVE):
-            printf("Standard WAVE format\n");
-            break;
-        case(WAVE_EX):
-            printf("MS WAVE-FORMAT-EXTENSIBLE\n");
-            printf("SPEAKER CONFIGURATION:\t");
-
-            switch(props.chformat){
-            //case(MC_STD):
-            //  printf("unassigned (generic format)");
-            //  break;
-            case(MC_MONO):
-                printf("Mono\n");
-                break;
-            case(MC_STEREO):
-                printf("Stereo\n");
-                break;
-            case(MC_QUAD):
-                printf("Rectangular Quad\n");
-                break;
-            case(MC_LCRS):
-                printf("Quad Surround\n");
-                break;
-            case(MC_BFMT):
-                printf("Ambisonic B-Format\n");
-                break;
-            case(MC_DOLBY_5_1):
-                printf("5.1 (Dolby) Surround\n");
-                break;
-            case(MC_SURR_6_1):
-                printf("6.1 Surround\n");
-                break;
-            case(MC_SURR_7_1):
-                printf("7.1 Surround\n");
-                break;
-            case(MC_CUBE):
-                printf("Cube Surround\n");
-                break;
-            default:
-                printf("Special speaker assignments\n");
-                break;
-            }
-            speakermask = sndgetchanmask(ifd);
-            if(speakermask < 0)
-                fprintf(stderr,"Unable to read speakermask from WAVE_EX header\n");
-            else {
-                //try to figure out the mask
-                int assigned = 0;
-                int shift = 0;
-                int this_channel = 1;
-                printf("Speaker Mask = %d (0x%x)\n",speakermask,(unsigned int) speakermask);
-                while(assigned < props.chans){
-                    if(speakers[shift].mask & speakermask){
-                        printf("Channel %d: %s\n",this_channel++,speakers[shift].name);
-                        assigned++;
-                    }
-                    if(++shift == NUM_SPEAKER_POSITIONS)
-                        break;
-                }
-                if(assigned < props.chans)
-                    printf("Remaining channels not assigned speaker locations.\n");
-
-            }
-            break;
-                
-        case(AIFF):
-            printf("AIFF format\n");
-            break;
-        case(AIFC):
-            printf("AIFC format\n");
-            break;
-        default:
-            printf("unknown format\n");
-            break;
-        }
-        printf("Sample Rate    :  %d\n",props.srate);
-        printf("Channels       :  %d\n",props.chans);
-        printf("Sample Frames  :  %d\n",nframes);
-        printf("sample type:   :  ");
-        switch(props.samptype){
-        case(SHORT8):
-            printf("8-bit\n");
-            break;
-        case(SHORT16):
-            printf("16-bit\n");
-            break;
-        case(FLOAT32):
-            printf("32bit floats\n");
-            break;
-        case(INT_32):
-            printf("32bit (integer)\n");
-            break;
-        case(INT2424):
-            printf("24bit (packed)\n");
-            break;
-        case(INT2432):
-            printf("24bit in 32bit words\n");
-            break;
-        case(INT2024):
-            printf("20bit in 24bit words\n");
-            break;
-        case(INT_MASKED):
-            printf("non-standard WAVE_EX format\n");
-            //fet more info....
-            break;
-        default:
-            printf("sorry: don't recognize word format!\n");
-            break;
-
-        }
-        printf("duration       :  %.4lf secs\n",(double)nframes / srate);
-        //this is the problem: we never decided what type maxamp is, officially!
-        if(props.samptype==FLOAT32){
-            if(sndgetprop(ifd,"maxamp",(char *)&fmaxamp,sizeof(float)) == sizeof(float)){
-                printf("CDP maxamp     :  %.4lf\n",fmaxamp);    
-            }
-        }
-        else{
-            if(sndgetprop(ifd,"maxamp",(char *)&lmaxamp,sizeof(int)) == sizeof(int)){
-                printf("CDP maxamp     :  %d\n",lmaxamp);   
-            }
-        }
-        peaks = (CHPEAK *) calloc(props.chans,sizeof(CHPEAK));
-        if(peaks==NULL){
-            puts("sfprops: no memory for fpeak data buffer\n");
-            exit(1);
-        }
-        else{
-            time_t thistime;
-            res = sndreadpeaks(ifd,props.chans,peaks, (int *) &peaktime);
-            thistime = (time_t) peaktime;
-            if(res ==0)
-                printf("no peak data in this infile\n");
-            else if(res < 0){
-                fprintf(stderr,"sfprops: WARNING: error reading infile peak data\n");
-                
-            }
-            else {      
-                printf("PEAK data:\n\tcreation time: %s\n", ctime(&thistime));
-                //for(i=0;i < props.chans; i++)
-                //  printf("CH %d: %.4lf at frame %d:\t%.4lf secs\n",
-                //  i+1,peaks[i].value,peaks[i].position,(double) (peaks[i].position) / srate);
-                for(i=0; i < props.chans; i++){
-                    double val, dbval;
-                    val = (double) peaks[i].value;
-                    
-                    if(val > 0.0){
-                        dbval = 20.0 * log10(val);
-                        printf("CH %d: %.6f (%.2lfdB) at frame %9d:\t%.4f secs\n",i,
-                               val,dbval,peaks[i].position,(double)(peaks[i].position / (double) srate));
-                    }
-                    else{
-                        printf("CH %d: %.6f (-infdB)  \t\t:\t%.4f secs\n",
-                               i,
-                               val, (double)(peaks[i].position / (double) srate)); 
-                    }
-                }
-            }
-        }
-        
-        break;
-    case(wt_analysis):
-        printf("CDP pvoc analysis file.\n");
-        printf("Channel Format:       Amplitude,Frequency\n");
-        printf("Orig rate:            %d\n",props.origrate);
-        printf("Analysis Window Size: %d\n",props.winlen);
-        printf("Analysis channels:    %d\n",props.chans/2);
-        printf("Window Overlap:       %d\n",props.decfac);
-        printf("Analysis rate:        %.4f\n",props.arate);
-        printf("Frame count:          %d\n",filesize / (props.chans));
-        printf("Data size (floats):   %d\n",filesize);
-        printf("Duration (secs):      %.3lf\n",
-            ((filesize / props.chans) * props.decfac) / (double) props.origrate);
-        break;
-    case(wt_formant):
-        printf("CDP formant data\n");
-        printf("Specenvcnt:           %d\n",props.specenvcnt);
-        printf("Analysis Window Size: %d\n",props.winlen);
-        printf("Channels:             %d\n",props.chans);
-        printf("Frame count:          %d\n",filesize);
-        printf("Orig rate:            %d\n",props.origrate);        
-        printf("Window Overlap:       %d\n",props.decfac);
-        printf("Analysis rate:        %.4f\n",props.arate);
-        printf("Orig Chans:          %d\n",props.origchans);        
-        printf("Duration (secs):      %.3lf\n",(filesize /props.specenvcnt)/ props.arate );
-        break;
-    case(wt_transposition):
-        printf("CDP transposition data\n");
-        printf("Orig sample rate:     %d\n",props.origrate);
-        printf("Analysis Window Size: %d\n",props.winlen);
-        printf("Channels:             %d\n",props.chans);
-        printf("Window Overlap:       %d\n",props.decfac);
-        printf("Analysis rate:        %.4f\n",props.arate);
-        printf("Orig Chans:          %d\n",props.origchans);
-        printf("Data size (floats):   %d\n",filesize);
-        printf("Duration (secs):      %.3lf\n",
-            ((filesize / props.chans) * props.decfac) / (double) props.origrate);
-        break;
-    case(wt_pitch):
-        printf("CDP pitch data\n");
-        printf("Orig sample rate:     %d\n",props.origrate);
-        printf("Analysis Window Size: %d\n",props.winlen);
-        printf("Channels:             %d\n",props.chans);
-        printf("Window Overlap:       %d\n",props.decfac);
-        printf("Analysis rate:        %.4f\n",props.arate);
-        printf("Orig Chans:          %d\n",props.origchans);
-        printf("Data size (floats):   %d\n",filesize);
-        printf("Duration (secs):      %.3lf\n",
-            ((filesize / props.chans) * props.decfac) / (double) props.origrate);
-        break;
-    case(wt_binenv):
-        printf("CDP binary envelope data\n");
-        printf("envelope window size: %f msecs\n",props.window_size);
-        printf("sample rate:          %d\n",props.srate);
-        printf("Channels:             %d\n",props.chans);
-        printf("Points:               %d\n",filesize);
-        printf("Duration (secs):      %.3f\n",filesize * (props.window_size * 0.001));
-        break;
-    default:
-        printf("internal error! unlisted soundfile type\n");
-        break;
-    }
-
-    sndcloseEx(ifd);
-//  sffinish();
-    return 0;
-}
-
-
-

dev/externals/paprogs/CMakeLists.txt

@@ -1,5 +0,0 @@
-
-add_subdirectory(listaudevs)
-add_subdirectory(paplay)
-add_subdirectory(pvplay)
-add_subdirectory(recsf)

dev/externals/paprogs/listaudevs/CMakeLists.txt

@@ -1,26 +0,0 @@
-if(APPLE)
-  set(CMAKE_C_FLAGS "-O2 -Wall -mmacosx-version-min=10.5 -Dunix -fomit-frame-pointer -funroll-loops")
-  include_directories ( /Developer/Headers/FlatCarbon )
-  find_library(COREAUDIOLIB CoreAudio)
-  find_library(AUDIOTOOLBOX AudioToolbox)
-  find_library(AULIB AudioUnit)
-  find_library(CARBONLIB Carbon)
-  set(EXTRA_LIBRARIES1 ${COREAUDIOLIB} ${AUDIOTOOLBOX} ${AULIB} ${CARBONLIB} ${EXTRA_LIBRARIES})
-else()
-  if(MINGW)
-    set(CMAKE_C_FLAGS "-O3 -DWIN32 -D_WIN32 -fomit-frame-pointer  -funroll-loops")
-    set(EXTRA_LIBRARIES1 winmm dsound winspool ${EXTRA_LIBRARIES})
-  else()
-    set(CMAKE_C_FLAGS "-O3 -Wall -Dlinux -Dunix -fomit-frame-pointer -funroll-loops")
-    set(EXTRA_LIBRARIES1 jack asound portsf pthread ${EXTRA_LIBRARIES})
-  endif()
-endif()
-
-link_directories(../../include ../portaudio/lib/.libs)
-
-include_directories(../../../include ../include ../portaudio/include ../portaudio/src/common )
-
-add_executable(listaudevs devs.c)
-target_link_libraries(listaudevs portaudio.a  ${EXTRA_LIBRARIES1})
-
-my_install(listaudevs)

dev/externals/paprogs/listaudevs/devs.c

@@ -1,69 +0,0 @@
-/*
- * Copyright (c) 1983-2020 Richard Dobson and Composers Desktop Project Ltd
- * http://www.rwdobson.com
- * http://www.composersdesktop.com
- * This file is part of the CDP System.
- * The CDP System is free software; you can redistribute it
- * and/or modify it under the terms of the GNU Lesser General Public 
- * License as published by the Free Software Foundation; either 
- * version 2.1 of the License, or (at your option) any later version. 
- *
- * The CDP System is distributed in the hope that it will be useful, 
- * but WITHOUT ANY WARRANTY; without even the implied warranty of 
- * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  
- * See the GNU Lesser General Public License for more details.
- * You should have received a copy of the GNU Lesser General Public
- * License along with the CDP System; if not, write to the Free Software
- * Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
- *
- */
-
-/* devs.c : display list of installed audio devices */  
-#include <stdio.h>
-#include <stdlib.h>
-#ifdef _WIN32
-# include <windows.h>
-#endif
-#include <portaudio.h>
-
-int show_devices(void);
-
-int main(void)
-{
-    return show_devices();
-}
-
-int show_devices(void)
-{
-        PaDeviceIndex numDevices,p;
-        const    PaDeviceInfo *pdi;
-        PaError  err;
-        int nOutputDevices = 0;
-
-        Pa_Initialize();
-        numDevices =  Pa_GetDeviceCount();
-        if( numDevices < 0 )
-        {
-            printf("ERROR: Pa_CountDevices returned 0x%x\n", numDevices );
-            err = numDevices;
-            return err;
-        }
-        //printf("Number of devices = %d\n", numDevices );
-        printf("Device\tInput\tOutput\tName\n");
-        
-        for( p=0; p<numDevices; p++ )
-        {
-            pdi = Pa_GetDeviceInfo( p );    
-            nOutputDevices++;
-            if( p == Pa_GetDefaultOutputDevice() ) 
-                printf("*");
-            else
-                printf(" ");
-            printf("%d\t%d\t%d\t%s\n",p,
-                pdi->maxInputChannels,
-                pdi->maxOutputChannels,
-                pdi->name);         
-        }
-        Pa_Terminate();
-        return 0;
-}

dev/externals/paprogs/listaudevs/orig_Makefile

@@ -1,32 +0,0 @@
-# OSX makefile for listaudevs © Richard Dobson Feb 1 2014
-
-CC = gcc
-OPT = -O3
-#OPT = -g
-# change this path as needed
-PADIR = ../mctools/portaudio
-PAINC = $(PADIR)/include
-PACOM = $(PADIR)/src/common
-
-PALIB = $(PADIR)/lib/libportaudio.a
-CFLAGS = $(OPT)  -g -mmacosx-version-min=10.5 -arch i386 -Dunix -DMAC  -DPA_USE_COREAUDIO -I$(PAINC)
-IOLINK =  -framework CoreAudio -framework AudioToolbox -framework AudioUnit -framework Carbon
-PROG=listaudevs  
-
-.c.o:
-	$(CC) $(CFLAGS) -c $<
-
-	 
-$(PROG):	devs.o 
-	$(CC) $(CFLAGS) devs.o -o $(PROG) $(IOLINK)  $(PALIB)
-
-
-clean: 
-	rm  devs.o
-veryclean:	clean
-	rm $(PROG)
-
-
-
-
-

dev/externals/paprogs/palinuxbuild.txt

@@ -1,37 +0,0 @@
-palinuxbuild.txt:
-
-Portaudio is required by the audio programs (paplay, pvplay, recsf, listaudevs).
-If the required libportaudio.a is not detected, the top level build script 
-(makeprograms.sh)  will skip this stage.
-
-The audio program makefiles assume availability and use of both alsa and jack.
-
-The supplied source archive is the latest v19 stable release 2014.
-
-To build, unpack pa--tgz in situ; this will create a portaudio directory. 
-Run configure with these options:
-
-./configure --with-alsa --with-jack
-
-Then run make as usual to build, 
-
-Do not perform run 'make install' unless you are happy for it to replace your current system portaudio libs.
-
-The CDP makefiles look for the local lib/.libs folder for libportaudio.a
-
-To build the audio programs, either:
-
-run the Makefile in each program directory:
-
-make install -f Makefile.linux
-
-or run the top level makeprograms.sh script.
-
-NB "install" above merely copies the binaries into the local Release directory under dev/.
-
-
-
-
-
-
-

dev/externals/paprogs/pamacbuild.txt

@@ -1,44 +0,0 @@
-pamacbuild.txt:
-
-Use this combination command to configure portaudio to build as a (single architecture) 
-static library on OS X, required by the audio programs (paplay, pvplay, recsf, listaudevs).
-
-The CDP programs build only for the 32bit architecture (-arch i386), 
-as they have not yet been fully tested for correctness with a 64bit architecture.
-
-Portaudio tends to assume all architectures are available, but this is not always the case 
-from Xcode 4.x onwards (e.g. no ppc support).
-
-You may need to edit the Makefile (or configure.in) by hand to add or exclude architectures. 
-
-You will likely need to type the commands below directly, in Terminal. Cutting and pasting from a text
- editor (possibly using Unicode) may result in a config error.
- 
-On a 32bit OS type:
-
-CFLAGS=$CFLAGS -mmacosx-version-min=10.4  ./configure --enable-shared=no --enable-mac-universal=no 
-
-If building on a 64bit machine, omit --enable-mac-universal=no and  make sure to add -arch i386 
-to the CFLAGS if not present, removing any which are not supported or needed, e.g:
-
-CFLAGS="$CFLAGS -arch i386 -arch x86_64"
-
-(the quotation marks are required if the CFLAGS string contains multiple options separated by spaces). 
-
-Then run make as usual to build. 
-
-The CDP makefiles look for the local lib/.libs folder for libportaudio.a
-There is no need to run sudo make install
-
-This is to avoid a couple of warnings when compiling pa_mac-core.c.
-The macosx version setting is to prevent warnings  about deprecated APIs.
-The configure script specifies -Werror ("treat warnings as errors"), 
-which means that any such warning stops the build.
-
-The command --enable-shared=no also seems to be needed, otherwise (for some reason) 
-the linker fails to find the Pa_Util functions.
-
-Newer builds of portaudio can of course be used; they may obviate some or all of the above issues; but may introduce new ones too!
-
-
-

dev/externals/paprogs/pamingwbuild.txt

@@ -1,74 +0,0 @@
-How to buld Portaudio static library for Windows, using MinGW. 
-
-This is required by the audio programs (paplay, pvplay, recsf, listaudevs)
-
-This assumes MinGW is already installed. This has been developed using the standard MinGW distribution; it has not been tested with the 
-Either or both ASIO and Directx support is needed for building the programs.
-
-An unpacked portaudio folder is provided, based on the latest release
-(see http://www.portaudio.com). It has a modified configure.in file to plug gaps in the default 
-portaudio configure setup.
-
-
-1. If ASIO support is desired, obtain the SDK from the Steinberg site (we are not allowed to distribute it ourselves).
-See: http://www.steinberg.net/en/company/developer.html
-
-It is convenient to put the unpacked directory ASIOSDK2 inside the portaudio folder, for ease of reference,
-and especially if it is shared with other projects, but other locations are possible, including /usr/local. 
-See the web page referenced at the bottom of this document.
-
-2. DirectSound support
-
-Unzip dx9mgw.zip into (e.g.) your MinGW home directory. This is the location assumed in the following instructions.
-
-Source of this sdk: http://alleg.sourceforge.net/files/dx9mgw.zip
-(see https://www.allegro.cc/forums/thread/610763)
-
-These examples assume the user name is "Richard". of course, replace this as needed with your MinGW user name
- 
-Portaudio configure has to be told which APIs to activate, and the paths to the SDKs for those APIs.
-
-cd to the portaudio directory (inside dev/externals/paprogs)
-
-ASIO only configure (NB long lines here):
-
-$ ./configure --with-host_os=mingw --with-winapi=asio --without-jack --enable-shared=no --enable-debug-output=no --with-asiodir=./ASIOSDK2
-
-
-dsound only configure:
-
-$ ./configure --with-host_os=mingw --with-winapi=directx --without-jack --enable-shared=no --enable-debug-output=no --with-dxdir=/home/Richard/dx9mgw 
-
-
-Both (recommended):
-
-$ ./configure --with-host_os=mingw --with-winapi=directx,asio --without-jack --enable-shared=no --enable-debug-output=no --with-dxdir=/home/Richard/dx9mgw --with-asiodir=./ASIOSDK2
-
-Thjis generates the Makefile which we will use to build portaudio
-
-3. Modify the Makefile
-
-Add the define -DPAWIN_USE_WDMKS_DEVICE_INFO to CFLAGS and CXXFLAGS
-
-Add the object file src/os/win/pa_win_wdmks_utils.o  to the OTHER_OJBJS list
-
-
-The standard configure does not set up the exact combination we need, for multichannel device support.
-
-
-
-And finally:
-
-$ make
-
-Note: only the static library libportaudio.a is built.
-The makefiles for the audio programs look for this in the local hidden directory (lib/.libs).
-There is no need to run [sudo] make install, which would overwrite any existing copy.
-
-
-See also here, for an extended discussion of how to build portaudio with MinGW:
-
-https://www.assembla.com/spaces/portaudio/wiki/Notes_about_building_PortAudio_with_MinGW
-
-Richard Dobson Feb 2014
-

dev/externals/paprogs/paplay/CMakeLists.txt

@@ -1,30 +0,0 @@
-if(APPLE)
-  set(CMAKE_C_FLAGS "-O2 -Wall -mmacosx-version-min=10.5 -Dunix -fomit-frame-pointer -funroll-loops")
-  include_directories ( /Developer/Headers/FlatCarbon )
-  find_library(COREAUDIOLIB CoreAudio)
-  find_library(AUDIOTOOLBOX AudioToolbox)
-  find_library(AULIB AudioUnit)
-  find_library(CARBONLIB Carbon)
-  find_library(AAIOLIB names libaaio.a paths /usr/local/lib)
-  set(EXTRA_LIBRARIES1 portsf pthread ${AAIOLIB} ${COREAUDIOLIB} ${AUDIOTOOLBOX} ${AULIB} ${CARBONLIB} ${EXTRA_LIBRARIES})
-  
-else()
-  if(MINGW)
-    set(CMAKE_C_FLAGS "-O3 -DWIN32 -D_WIN32 -DUSE_ASIO -fomit-frame-pointer -funroll-loops")
-    set(EXTRA_LIBRARIES1 winmm dsound winspool ${EXTRA_LIBRARIES})
-  else()
-    set(CMAKE_C_FLAGS "-O3 -Wall -Dlinux -Dunix -fomit-frame-pointer -funroll-loops")
-    set(EXTRA_LIBRARIES1 jack asound portsf pthread ${EXTRA_LIBRARIES})
-  endif()
-endif()
-
-
-
-link_directories (../../lib ../portaudio/lib/.libs  /usr/local/lib)
-
-include_directories(../../include ../include ../portaudio/include ../portaudio/src/common /usr/local/include)
-
-add_executable(paplay paplay.c fmhfuncs.c)
-target_link_libraries(paplay portaudio.a  ${EXTRA_LIBRARIES1})
-
-my_install(paplay)

dev/externals/paprogs/paplay/fmdcode.h

@@ -1,106 +0,0 @@
-/*
- * Copyright (c) 1983-2013 Richard Dobson and Composers Desktop Project Ltd
- * http://people.bath.ac.uk/masrwd
- * http://www.composersdesktop.com
- * This file is part of the CDP System.
- * The CDP System is free software; you can redistribute it
- * and/or modify it under the terms of the GNU Lesser General Public 
- * License as published by the Free Software Foundation; either 
- * version 2.1 of the License, or (at your option) any later version. 
- *
- * The CDP System is distributed in the hope that it will be useful, 
- * but WITHOUT ANY WARRANTY; without even the implied warranty of 
- * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  
- * See the GNU Lesser General Public License for more details.
- * You should have received a copy of the GNU Lesser General Public
- * License along with the CDP System; if not, write to the Free Software
- * Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
- *
- */
- 
-/* fmdcode.h */
-
-/*
- Channel order is WXYZ,RSTUV,KLMNOPQ
- 
- The number of channels defines the order of the soundfield:
- 2 channel = UHJ 
- 3 channel = h   = 1st order horizontal
- 4 channel = f   = 1st order 3-D
- 5 channel = hh  = 2nd order horizontal
- 6 channel = fh  = 2nd order horizontal + 1st order height (formerly
-                                                            called 2.5 order)
- 7 channel = hhh = 3rd order horizontal
- 8 channel = fhh = 3rd order horizontal + 1st order height
- 9 channel = ff  = 2nd order 3-D
- 11 channel = ffh = 3rd order horizontal + 2nd order height
- 16 channel = fff = 3rd order 3-D
- 
- 
- Horizontal   Height  Soundfield   Number of    Channels
- order 	      order 	  type      channels 	
- 1 	         0 	       horizontal 	  3 	    WXY
- 1 	         1 	      full-sphere 	  4 	    WXYZ
- 2 	         0 	       horizontal 	  5 	    WXY....UV
- 2 	         1 	       mixed-order    6 	    WXYZ...UV
- 2 	         2 	      full-sphere     9 	    WXYZRSTUV
- 3           0 	       horizontal 	  7 	    WXY....UV.....PQ
- 3 	         1         mixed-order 	  8 	    WXYZ...UV.....PQ
- 3 	         2 	       mixed-order 	 11 	    WXYZRSTUV.....PQ
- 3 	         3 	      full-sphere 	 16 	    WXYZRSTUVKLMNOPQ
- */
-
-typedef struct abf_samp {
-	float W;
-	float X;
-	float Y;
-	float Z;
-    float R;
-    float S;
-    float T;
-	float U;
-    float V;
-} ABFSAMPLE;
-
-typedef void (*fmhcopyfunc)(ABFSAMPLE*,const float*);
-
-typedef void (*fmhdecodefunc)(const ABFSAMPLE*, float*,unsigned int);
-//void bfdcode4(float *inbuf,long numframes);
-void fmhcopy_3(ABFSAMPLE* abf,const float*buf);
-void fmhcopy_4(ABFSAMPLE* abf,const float*buf);
-void fmhcopy_5(ABFSAMPLE* abf,const float*buf);
-void fmhcopy_6(ABFSAMPLE* abf,const float*buf);
-void fmhcopy_7(ABFSAMPLE* abf,const float*buf);
-void fmhcopy_8(ABFSAMPLE* abf,const float*buf);
-void fmhcopy_9(ABFSAMPLE* abf,const float*buf);
-void fmhcopy_11(ABFSAMPLE* abf,const float*buf);
-void fmhcopy_16(ABFSAMPLE* abf,const float*buf);
-
-
-// i1 = inphase 1st order, etc
-void fm_i1_mono(const ABFSAMPLE *inbuf,float *outbuf,unsigned int numframes);
-void fm_i1_stereo(const ABFSAMPLE *inbuf,float *outbuf,unsigned int numframes);
-void fm_i1_square(const ABFSAMPLE *inbuf,float *outbuf,unsigned int numframes);
-void fm_i2_square(const ABFSAMPLE *inbuf,float *outbuf,unsigned int numframes);
-void fm_i1_quad(const ABFSAMPLE *inbuf,float *outbuf,unsigned int numframes);
-void fm_i2_quad(const ABFSAMPLE *inbuf,float *outbuf,unsigned int numframes);
-void fm_i1_pent(const ABFSAMPLE *inbuf,float *outbuf,unsigned int numframes);
-void fm_i2_pent(const ABFSAMPLE *inbuf,float *outbuf,unsigned int numframes);
-void fm_i1_surr(const ABFSAMPLE *inbuf,float *outbuf,unsigned int numframes);
-void fm_i2_surr(const ABFSAMPLE *inbuf,float*outbuf,unsigned int numframes);
-void fm_i1_surr6(const ABFSAMPLE *inbuf,float *outbuf,unsigned int numframes);
-void fm_i2_surr6(const ABFSAMPLE *inbuf,float*outbuf,unsigned int numframes);
-void dm_i1_surr(const ABFSAMPLE *inbuf,float*outbuf,unsigned int numframes);
-void dm_i1_surr6(const ABFSAMPLE *inbuf,float*outbuf,unsigned int numframes);
-void dm_i2_surr(const ABFSAMPLE *inbuf,float*outbuf,unsigned int numframes);
-void dm_i2_surr6(const ABFSAMPLE *inbuf,float*outbuf,unsigned int numframes);
-void fm_i1_hex(const ABFSAMPLE *inbuf,float *outbuf,unsigned int numframes);
-void fm_i2_hex(const ABFSAMPLE *inbuf,float *outbuf,unsigned int numframes);
-void fm_i1_oct1(const ABFSAMPLE *inbuf,float *outbuf,unsigned int numframes);
-void fm_i2_oct1(const ABFSAMPLE *inbuf,float *outbuf,unsigned int numframes);
-void fm_i1_oct2(const ABFSAMPLE *inbuf,float *outbuf,unsigned int numframes);
-void fm_i2_oct2(const ABFSAMPLE *inbuf,float *outbuf,unsigned int numframes);
-void fm_i1_cube(const ABFSAMPLE *inbuf,float *outbuf,unsigned int numframes);
-void fm_i2_cube(const ABFSAMPLE *inbuf,float *outbuf,unsigned int numframes);
-void fm_i1_cubex(const ABFSAMPLE *inbuf,float *outbuf,unsigned int numframes);
-void fm_i2_cubex(const ABFSAMPLE *inbuf,float *outbuf,unsigned int numframes);

dev/externals/paprogs/paplay/fmhfuncs.c

@@ -1,712 +0,0 @@
-/*
- * Copyright (c) 1983-2013 Richard Dobson and Composers Desktop Project Ltd
- * http://people.bath.ac.uk/masrwd
- * http://www.composersdesktop.com
- * This file is part of the CDP System.
- * The CDP System is free software; you can redistribute it
- * and/or modify it under the terms of the GNU Lesser General Public 
- * License as published by the Free Software Foundation; either 
- * version 2.1 of the License, or (at your option) any later version. 
- *
- * The CDP System is distributed in the hope that it will be useful, 
- * but WITHOUT ANY WARRANTY; without even the implied warranty of 
- * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  
- * See the GNU Lesser General Public License for more details.
- * You should have received a copy of the GNU Lesser General Public
- * License along with the CDP System; if not, write to the Free Software
- * Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
- *
- */
- 
-/* fmhfuncs.c */
-#include "fmdcode.h"
-
-/* TODO: expand to handle numframes frames? */
-
-void fmhcopy_3(ABFSAMPLE* abf,const float *buf)
-{
-    abf->W = *buf++;
-    abf->X = *buf++;
-    abf->Y = *buf;
-}
-
-void fmhcopy_4(ABFSAMPLE* abf,const float *buf)
-{
-    abf->W = *buf++;
-    abf->X = *buf++;
-    abf->Y = *buf++;
-    abf->Z = *buf;
-}
-
-void fmhcopy_5(ABFSAMPLE* abf,const float*buf)
-{
-    abf->W = *buf++;
-    abf->X = *buf++;
-    abf->Y = *buf++;
-    abf->U = *buf++;
-    abf->V = *buf;
-}
-
-void fmhcopy_6(ABFSAMPLE* abf,const float*buf)
-{
-    abf->W = *buf++;
-    abf->X = *buf++;
-    abf->Y = *buf++;
-    abf->Z = *buf++;
-    abf->U = *buf++;
-    abf->V = *buf;
-}
-// following discard 3rd order chans
-void fmhcopy_7(ABFSAMPLE* abf,const float*buf)
-{
-    abf->W = *buf++;
-    abf->X = *buf++;
-    abf->Y = *buf++;
-    abf->U = *buf++;
-    abf->V = *buf;
-}
-
-void fmhcopy_8(ABFSAMPLE* abf,const float*buf)
-{
-    abf->W = *buf++;
-    abf->X = *buf++;
-    abf->Y = *buf++;
-    abf->Z = *buf++;
-    abf->U = *buf++;
-    abf->V = *buf;
-}
-// these identical for 2nd order horiz max, but may be expanded later!
-void fmhcopy_9(ABFSAMPLE* abf,const float*buf)
-{
-    abf->W = *buf++;
-    abf->X = *buf++;
-    abf->Y = *buf++;
-    abf->Z = *buf++;
-    abf->R = *buf++;
-    abf->S = *buf++;
-    abf->T = *buf++;
-    abf->U = *buf++;
-    abf->V = *buf;
-}
-
-void fmhcopy_11(ABFSAMPLE* abf,const float*buf)
-{
-    abf->W = *buf++;
-    abf->X = *buf++;
-    abf->Y = *buf++;
-    abf->Z = *buf++;
-    abf->R = *buf++;
-    abf->S = *buf++;
-    abf->T = *buf++;
-    abf->U = *buf++;
-    abf->V = *buf;
-}
-void fmhcopy_16(ABFSAMPLE* abf,const float*buf)
-{
-    abf->W = *buf++;
-    abf->X = *buf++;
-    abf->Y = *buf++;
-    abf->Z = *buf++;
-    abf->R = *buf++;
-    abf->S = *buf++;
-    abf->T = *buf++;
-    abf->U = *buf++;
-    abf->V = *buf;
-}
-
-/********** DECODE FUNCS *************/
-/* TODO: complete support for numframes > 1 */
-
-void fm_i1_mono(const ABFSAMPLE *inbuf,float *outbuf,unsigned int numframes)
-{
-    unsigned int i;	
-	float *p_out = outbuf;
-	double aw;
-    for(i=0;i < numframes; i++){
-		aw = (double) inbuf->W * 0.7071;
-		*p_out++ = (float) aw;  		  
-	}
-}
-
-void fm_i1_stereo(const ABFSAMPLE *inbuf,float *outbuf,unsigned int numframes)
-{
-    unsigned int i;	
-	float *p_out = outbuf;
-	double aw,ay;
-    for(i=0;i < numframes; i++){
-		aw = (double) inbuf->W * 0.7071;
-		ay = (double) inbuf->Y * 0.5;		
-		
-		*p_out++ = (float) (aw +  ay);  
-		*p_out++ = (float) (aw  - ay);  
-	}
-}
-
-void fm_i1_square(const ABFSAMPLE *inbuf,float *outbuf,unsigned int numframes)
-{
-	unsigned int i;	
-	float *p_out = outbuf;
-	double aw,ax,ay;
-	
-	for(i=0;i < numframes; i++){
-		aw = (double) inbuf->W * 0.35355;
-		ax = (double) inbuf->X * 0.17677;
-		ay = (double) inbuf->Y * 0.17677;		
-		
-		*p_out++ = (float) (aw + ax + ay);  //FL
-		*p_out++ = (float) (aw - ax + ay);  //RL
-		*p_out++ = (float) (aw - ax - ay);  //RR
-		*p_out++ = (float) (aw + ax - ay);  //FR
-	}
-}
-void fm_i2_square(const ABFSAMPLE *inbuf,float*outbuf,unsigned int numframes)
-{
-	unsigned int i;		
-	float *p_out = outbuf;	
-	double aw,ax,ay,av;
-	
-	for(i=0;i < numframes; i++){
-		aw = (double) inbuf->W * 0.3536;
-		ax = (double) inbuf->X * 0.2434;
-		ay = (double) inbuf->Y * 0.2434;		
-		av = (double) inbuf->V * 0.0964;
-		*p_out++ = (float) (aw + ax + ay + av);  //FL
-		*p_out++ = (float) (aw - ax + ay - av ); //RL
-		*p_out++ = (float) (aw - ax - ay + av);  //RR
-		*p_out++ = (float) (aw + ax - ay - av);  //FR
-	}
-}
-/* ditto, RLRL layout for WAVEX */
-void fm_i1_quad(const ABFSAMPLE *inbuf,float *outbuf,unsigned int numframes)
-{
-	unsigned int i;	
-	float *p_out = outbuf;
-	double aw,ax,ay;
-	
-	for(i=0;i < numframes; i++){
-		aw = (double) inbuf->W * 0.35355;
-		ax = (double) inbuf->X * 0.17677;
-		ay = (double) inbuf->Y * 0.17677;		
-		
-		*p_out++ = (float) (aw + ax + ay);  //FL
-        *p_out++ = (float) (aw + ax - ay);  //FR
-		*p_out++ = (float) (aw - ax + ay);  //RL
-		*p_out++ = (float) (aw - ax - ay);  //RR
-		
-	}
-}
-void fm_i2_quad(const ABFSAMPLE *inbuf,float*outbuf,unsigned int numframes)
-{
-	unsigned int i;		
-	float *p_out = outbuf;	
-	double aw,ax,ay,av;
-	
-	for(i=0;i < numframes; i++){
-		aw = (double) inbuf->W * 0.3536;
-		ax = (double) inbuf->X * 0.2434;
-		ay = (double) inbuf->Y * 0.2434;		
-		av = (double) inbuf->V * 0.0964;
-		*p_out++ = (float) (aw + ax + ay + av);  //FL
-        *p_out++ = (float) (aw + ax - ay - av);  //FR
-		*p_out++ = (float) (aw - ax + ay - av ); //RL
-		*p_out++ = (float) (aw - ax - ay + av);  //RR
-		
-	}
-}
-
-
-//front pair angle 72deg
-void fm_i1_pent(const ABFSAMPLE *inbuf,float*outbuf,unsigned int numframes)
-{
-	unsigned int i;	
-	float *p_out = outbuf;
-	double aw,ax,ay;
-	
-	for(i=0;i < numframes; i++){
-		aw = (double) inbuf->W * 0.2828;
-		ax = (double) inbuf->X;
-		ay = (double) inbuf->Y;		
-		
-		*p_out++ = (float) (aw + (ax*0.1618) + (ay*0.1176));  //FL
-		*p_out++ = (float) (aw - (ax*0.0618) + (ay*0.1902));  
-		*p_out++ = (float) (aw - (ax*0.2));  
-		*p_out++ = (float) (aw - (ax*0.0618) - (ay*0.1902));  
-        *p_out++ = (float) (aw + (ax*0.1618) - (ay*0.1176)); //FR
-	}
-}
-
-void fm_i2_pent(const ABFSAMPLE *inbuf,float*outbuf,unsigned int numframes)
-{
-	unsigned int i;	
-	float *p_out = outbuf;	
-	double aw,ax,ay,au,av;
-	
-	for(i=0;i < numframes; i++){
-		aw = (double) inbuf->W * 0.2828;
-		ax = (double) inbuf->X;
-		ay = (double) inbuf->Y;	
-        au = (double) inbuf->U;
-        av = (double) inbuf->V;
-		
-		*p_out++ = (float) (aw + (ax*0.2227) + (ay*0.1618) + (au*0.0238) + (av * 0.0733));  
-		*p_out++ = (float) (aw - (ax*0.0851) + (ay*0.2619) - (au*0.0624) - (av * 0.0453));  
-		*p_out++ = (float) (aw - (ax*0.2753)               + (au * 0.0771)              );  
-		*p_out++ = (float) (aw - (ax*0.0851) - (ay*0.2619) - (au*0.0624) + (av * 0.0453));  
-        *p_out++ = (float) (aw + (ax*0.2227) - (ay*0.1618) + (au*0.0238) - (av * 0.0733));
-	}
-}
-
-/* FMH only defines 1st order decode */ 
-void fm_i1_surr(const ABFSAMPLE *inbuf,float*outbuf,unsigned int numframes)
-{
-	unsigned int i;	
-	float *p_out = outbuf;
-	double aw,ax,ay;
-	
-	for(i=0;i < numframes; i++){
-		aw = (double) inbuf->W;
-		ax = (double) inbuf->X;
-		ay = (double) inbuf->Y;		
-		/* TODO: fix this order! */
-		*p_out++ = (float) ((aw * 0.169)  + (ax*0.0797) + (ay * 0.0891));   //L
-		*p_out++ = (float) ((aw * 0.1635) + (ax*0.0923));                   //C    ///???
-		*p_out++ = (float) ((aw * 0.169)  - (ax*0.0797) - (ay * 0.0891));   //R    ///????
-		*p_out++ = (float) ((aw * 0.4563) - (ax*0.1259) + (ay * 0.1543));   //LS
-        *p_out++ = (float) ((aw * 0.4563) - (ax*0.1259) - (ay * 0.1543));   //RS
-	}
-}
-/* from Bruce Wiggins via Csound */
-void fm_i2_surr(const ABFSAMPLE *inbuf,float*outbuf,unsigned int numframes)
-{
-	unsigned int i;	
-	float *p_out = outbuf;
-	double aw,ax,ay,au,av;
-	
-	for(i=0;i < numframes; i++){
-		aw = (double) inbuf->W;
-		ax = (double) inbuf->X;
-		ay = (double) inbuf->Y;		
-		au = (double) inbuf->U;
-        av = (double) inbuf->V;
-        
-		*p_out++ = (float) ((aw * 0.405) + (ax*0.32) + (ay * 0.31)  + (au * 0.085) + (av * 0.125));   //L
-        *p_out++ = (float) ((aw * 0.405) + (ax*0.32) - (ay * 0.31)  + (au * 0.085) - (av * 0.125));   //R
-		*p_out++ = (float) ((aw * 0.085) + (ax*0.04)                + (au * 0.045)               );   //C
-		*p_out++ = (float) ((aw * 0.635) - (ax*0.335) + (ay * 0.28) - (au * 0.08)  + (av * 0.08));    //LS
-        *p_out++ = (float) ((aw * 0.635) - (ax*0.335) - (ay * 0.28) - (au * 0.08)  - (av * 0.08));    //RS
-	}
-}
-
-/* 5.1 versions - silent LFE */
-/* FMH only defines 1st order decode */ 
-void fm_i1_surr6(const ABFSAMPLE *inbuf,float*outbuf,unsigned int numframes)
-{
-	unsigned int i;	
-	float *p_out = outbuf;
-	double aw,ax,ay;
-	
-	for(i=0;i < numframes; i++){
-		aw = (double) inbuf->W;
-		ax = (double) inbuf->X;
-		ay = (double) inbuf->Y;		
-		
-		*p_out++ = (float) ((aw * 0.169)  + (ax*0.0797) + (ay * 0.0891));   //L
-		*p_out++ = (float) ((aw * 0.1635) + (ax*0.0923));                   //C
-		*p_out++ = (float) ((aw * 0.169)  - (ax*0.0797) - (ay * 0.0891));   //R
-        *p_out++ = 0.0f;                                                    //LFE
-		*p_out++ = (float) ((aw * 0.4563) - (ax*0.1259) + (ay * 0.1543));   //LS
-        *p_out++ = (float) ((aw * 0.4563) - (ax*0.1259) - (ay * 0.1543));   //RS
-	}
-}
-/* from Bruce Wiggins via Csound */
-void fm_i2_surr6(const ABFSAMPLE *inbuf,float*outbuf,unsigned int numframes)
-{
-	unsigned int i;	
-	float *p_out = outbuf;
-	double aw,ax,ay,au,av;
-	
-	for(i=0;i < numframes; i++){
-		aw = (double) inbuf->W;
-		ax = (double) inbuf->X;
-		ay = (double) inbuf->Y;		
-		au = (double) inbuf->U;
-        av = (double) inbuf->V;
-        
-		*p_out++ = (float) ((aw * 0.405) + (ax*0.32) + (ay * 0.31)  + (au * 0.085) + (av * 0.125));   //L
-        *p_out++ = (float) ((aw * 0.405) + (ax*0.32) - (ay * 0.31)  + (au * 0.085) - (av * 0.125));   //R
-		*p_out++ = (float) ((aw * 0.085) + (ax*0.04)                + (au * 0.045)               );   //C
-        *p_out++ = 0.0f;                                                                              //LFE
-		*p_out++ = (float) ((aw * 0.635) - (ax*0.335) + (ay * 0.28) - (au * 0.08)  + (av * 0.08));    //LS
-        *p_out++ = (float) ((aw * 0.635) - (ax*0.335) - (ay * 0.28) - (au * 0.08)  - (av * 0.08));    //RS
-	}
-}
-
-// 1st order 5.0
-void dm_i1_surr(const ABFSAMPLE *inbuf,float*outbuf,unsigned int numframes)
-{
-	unsigned int i;	
-	float *p_out = outbuf;
-	double aw,ax,ay;
-	
-	for(i=0;i < numframes; i++){
-		aw = (double) inbuf->W;
-		ax = (double) inbuf->X;
-		ay = (double) inbuf->Y;		
-		
-        
-		*p_out++ = (float) ((aw * 0.4597)  + (ax*0.4536) + (ay * 0.3591));   //L
-        *p_out++ = (float)  ((aw * 0.4597)  + (ax*0.4536) - (ay * 0.3591));  //R 
-		*p_out++ = 0.0f;                                                     //C
-		
-		*p_out++ = (float) ((aw * 0.5662) - (ax*0.3681) + (ay * 0.4606));    //LS
-        *p_out++ = (float) ((aw * 0.5662) - (ax*0.3681) - (ay * 0.4606));    //RS
-    }
-}
-//1st order 5.1
-void dm_i1_surr6(const ABFSAMPLE *inbuf,float*outbuf,unsigned int numframes)
-{
-	unsigned int i;	
-	float *p_out = outbuf;
-	double aw,ax,ay;
-	
-	for(i=0;i < numframes; i++){
-		aw = (double) inbuf->W;
-		ax = (double) inbuf->X;
-		ay = (double) inbuf->Y;		
-		
-        
-		*p_out++ = (float) ((aw * 0.4597)  + (ax*0.4536) + (ay * 0.3591));   //L
-        *p_out++ = (float)  ((aw * 0.4597)  + (ax*0.4536) - (ay * 0.3591));  //R 
-		*p_out++ = 0.0f;                                                     //C
-		*p_out++ = 0.0f;                                                     //LFE
-		*p_out++ = (float) ((aw * 0.5662) - (ax*0.3681) + (ay * 0.4606));    //LS
-        *p_out++ = (float) ((aw * 0.5662) - (ax*0.3681) - (ay * 0.4606));    //RS
-    }
-}
-// 2nd order 5.0
-void dm_i2_surr(const ABFSAMPLE *inbuf,float*outbuf,unsigned int numframes)
-{
-	unsigned int i;	
-	float *p_out = outbuf;
-	double aw,ax,ay,au,av;
-	
-	for(i=0;i < numframes; i++){
-		aw = (double) inbuf->W;
-		ax = (double) inbuf->X;
-		ay = (double) inbuf->Y;		
-		au = (double) inbuf->U;
-        av = (double) inbuf->V;
-        
-		*p_out++ = (float) ((aw * 0.3314)  + (ax*0.4097) + (ay * 0.3487) + (au * 0.0828) + (av*0.1489));  //L
-        *p_out++ = (float) ((aw * 0.3314)  + (ax*0.4097) - (ay * 0.3487) + (au * 0.0828) - (av*0.1489));  //R 
-		*p_out++ = (float) ((aw * 0.0804)  + (ax * 0.1327));                                              //C
-		*p_out++ = (float) ((aw * 0.6025) - (ax*0.3627) + (ay * 0.4089) - (au * 0.0567));                 //LS
-        *p_out++ = (float) ((aw * 0.6025) - (ax*0.3627) - (ay * 0.4089) - (au * 0.0567));                 //RS
-    }
-}
-// 2nd order 5.1
-void dm_i2_surr6(const ABFSAMPLE *inbuf,float*outbuf,unsigned int numframes)
-{
-	unsigned int i;	
-	float *p_out = outbuf;
-	double aw,ax,ay,au,av;
-	
-	for(i=0;i < numframes; i++){
-		aw = (double) inbuf->W;
-		ax = (double) inbuf->X;
-		ay = (double) inbuf->Y;		
-		au = (double) inbuf->U;
-        av = (double) inbuf->V;
-        
-		*p_out++ = (float) ((aw * 0.3314)  + (ax*0.4097) + (ay * 0.3487) + (au * 0.0828) + (av*0.1489));  //L
-        *p_out++ = (float) ((aw * 0.3314)  + (ax*0.4097) - (ay * 0.3487) + (au * 0.0828) - (av*0.1489));  //R 
-		*p_out++ = (float) ((aw * 0.0804)  + (ax * 0.1327));                                              //C
-		*p_out++ = 0.0f;                                                                                  //LFE
-		*p_out++ = (float) ((aw * 0.6025) - (ax*0.3627) + (ay * 0.4089) - (au * 0.0567));                 //LS
-        *p_out++ = (float) ((aw * 0.6025) - (ax*0.3627) - (ay * 0.4089) - (au * 0.0567));                 //RS
-    }
-}
-
-
-void fm_i1_hex(const ABFSAMPLE *inbuf,float*outbuf,unsigned int numframes)
-{
-	unsigned int i;	
-	float *p_out = outbuf;
-	double aw,ax,ay;
-	
-	for(i=0;i < numframes; i++){
-		aw = (double) inbuf->W * 0.2357;
-		ax = (double) inbuf->X * 0.1443;
-		ay = (double) inbuf->Y;		
-		
-		*p_out++ = (float) (aw + ax + (ay * 0.0833));  //FL
-		*p_out++ = (float) (aw      + (ay * 0.1667));  //SL
-		*p_out++ = (float) (aw - ax + (ay * 0.0833));  //RL
-		*p_out++ = (float) (aw - ax - (ay * 0.0833));  //RR
-        *p_out++ = (float) (aw      - (ay * 0.1667));  //SR
-        *p_out++ = (float) (aw + ax - (ay * 0.0833));  //FR
-	}
-}
-void fm_i2_hex(const ABFSAMPLE *inbuf,float*outbuf,unsigned int numframes)
-{
-	unsigned int i;	
-	float *p_out = outbuf;	
-	double aw,ax,ay,au,av;
-	
-	for(i=0;i < numframes; i++){
-		aw = (double) inbuf->W * 0.2357;
-		ax = (double) inbuf->X * 0.1987;
-		ay = (double) inbuf->Y;
-		au = (double) inbuf->U;
-        av = (double) inbuf->V * 0.0556;
-		
-		*p_out++ = (float) (aw + ax + (ay * 0.1147) + (au * 0.0321) + av);  //FL
-		*p_out++ = (float) (aw      + (ay * 0.2294) - (au * 0.0643)     );  //SL
-		*p_out++ = (float) (aw - ax + (ay * 0.1147) + (au * 0.0321) - av);  //RL
-		*p_out++ = (float) (aw - ax - (ay * 0.1147) + (au * 0.0321) + av);  //RR
-        *p_out++ = (float) (aw      - (ay * 0.2294) - (au * 0.0643)     );  //SR
-        *p_out++ = (float) (aw + ax - (ay * 0.1147) + (au * 0.0321) - av);  //FR
-	}
-}
-
-void fm_i1_oct1(const ABFSAMPLE *inbuf,float*outbuf,unsigned int numframes)
-{
-	unsigned int i;	
-	float *p_out = outbuf;
-	double aw,ax,ay;
-	
-	for(i=0;i < numframes; i++){
-		aw = (double) inbuf->W * 0.1768;
-		ax = (double) inbuf->X;
-		ay = (double) inbuf->Y;		
-		
-		*p_out++ = (float) (aw + (ax * 0.1155) + (ay * 0.0478));  
-		*p_out++ = (float) (aw + (ax * 0.0478) + (ay * 0.1155));  
-		*p_out++ = (float) (aw - (ax * 0.0478) + (ay * 0.1155));  
-		*p_out++ = (float) (aw - (ax * 0.1155) + (ay * 0.0478));  
-        *p_out++ = (float) (aw - (ax * 0.231)  - (ay * 0.0957));  
-        *p_out++ = (float) (aw - (ax * 0.0478) - (ay * 0.1155));  
-        *p_out++ = (float) (aw + (ax * 0.0478) - (ay * 0.1155));  
-        *p_out++   = (float) (aw + (ax * 0.1155) - (ay * 0.0478));  
-	}
-}
-void fm_i2_oct1(const ABFSAMPLE *inbuf,float *outbuf,unsigned int numframes)
-{
-	unsigned int i;	
-	float *p_out = outbuf;	
-	double aw,ax,ay,au,av;
-	
-	for(i=0;i < numframes; i++){
-		aw = (double) inbuf->W * 0.17677;
-		ax = (double) inbuf->X;
-		ay = (double) inbuf->Y;
-		au = (double) inbuf->U * 0.03417;
-		av = (double) inbuf->V * 0.03417;
-        
-		*p_out++ = (float) (aw + (ax * 0.15906) + (ay * 0.06588) + au + av);  
-		*p_out++ = (float) (aw + (ax * 0.06588) + (ay * 0.15906) - au + av);  
-		*p_out++ = (float) (aw - (ax * 0.06588) + (ay * 0.15906) - au - av);  
-		*p_out++ = (float) (aw - (ax * 0.15906) + (ay * 0.06588) + au - av);  
-        *p_out++ = (float) (aw - (ax * 0.15906) - (ay * 0.06588) + au + av);  
-        *p_out++ = (float) (aw - (ax * 0.06588) - (ay * 0.15906) - au + av);  
-        *p_out++ = (float) (aw + (ax * 0.06588) - (ay * 0.15906) - au - av);  
-        *p_out++ = (float) (aw + (ax * 0.15906) - (ay * 0.06588) + au - av);  
-	}
-}
-
-void fm_i1_oct2(const ABFSAMPLE *inbuf,float *outbuf,unsigned int numframes)
-{
-	unsigned int i;	
-	float *p_out = outbuf;
-	double aw,ax,ay;
-	
-	for(i=0;i < numframes; i++){
-		aw = (double) inbuf->W * 0.1768;
-		ax = (double) inbuf->X;
-		ay = (double) inbuf->Y;		
-		
-		*p_out++ = (float) (aw + (ax * 0.125)                 );  
-		*p_out++ = (float) (aw + (ax * 0.0884) + (ay * 0.0884));  
-		*p_out++ = (float) (aw                 + (ay * 0.125) );  
-		*p_out++ = (float) (aw - (ax * 0.0884) + (ay * 0.0884));  
-        *p_out++ = (float) (aw - (ax * 0.125)                 );  
-        *p_out++ = (float) (aw - (ax * 0.0884) - (ay * 0.0884));  
-        *p_out++ = (float) (aw                 - (ay * 0.125) );  
-        *p_out++ = (float) (aw + (ax * 0.0884) - (ay * 0.0884));  
-	}
-}
-void fm_i2_oct2(const ABFSAMPLE *inbuf,float *outbuf,unsigned int numframes)
-{
-	unsigned int i;	
-	float *p_out = outbuf;	
-	double aw,ax,ay,au,av;
-	
-	for(i=0;i < numframes; i++){
-		aw = (double) inbuf->W * 0.1768;
-		ax = (double) inbuf->X;
-		ay = (double) inbuf->Y;
-		au = (double) inbuf->U * 0.0482;
-		av = (double) inbuf->V * 0.0482;
-        
-		*p_out++ = (float) (aw + (ax * 0.1721)                 + au     );  
-		*p_out++ = (float) (aw + (ax * 0.1217) + (ay * 0.1217)      + av);  
-		*p_out++ = (float) (aw                 + (ay * 0.1721) - au     );  
-		*p_out++ = (float) (aw - (ax * 0.1217) + (ay * 0.1217)      - av);  
-        *p_out++ = (float) (aw - (ax * 0.1721)                 + au     );  
-        *p_out++ = (float) (aw - (ax * 0.1217) - (ay * 0.1217)      + av);  
-        *p_out++ = (float) (aw                 - (ay * 0.1721) - au     );  
-        *p_out++ = (float) (aw + (ax * 0.1217) - (ay * 0.1217)      - av);  
-	}
-}
-
-/* csound order; low/high anti-clockwise. 
-FMH page order, 4 low folowed by 4 high , clockwise! */
-void fm_i1_cube(const ABFSAMPLE *inbuf,float *outbuf,unsigned int numframes)
-{
-	unsigned int i;	
-	float *p_out = outbuf;
-	double aw,ax,ay,az;
-	
-	for(i=0;i < numframes; i++){
-		aw = (double) inbuf->W * 0.17677;
-		ax = (double) inbuf->X * 0.07216;
-		ay = (double) inbuf->Y * 0.07216;
-		az = (double) inbuf->Z * 0.07216;
-		
-		*p_out++ = (float) (aw + ax + ay - az);  // FL low
-		*p_out++ = (float) (aw + ax + ay + az);  // FL hi
-        
-		*p_out++ = (float) (aw - ax + ay - az);  // RL low
-		*p_out++ = (float) (aw - ax + ay + az);  //    hi
-        
-        *p_out++ = (float) (aw - ax - ay - az);  // RR low
-        *p_out++ = (float) (aw - ax - ay + az);  //   hi
-        
-        *p_out++ = (float) (aw + ax - ay - az);  // FR low
-        *p_out++ = (float) (aw + ax - ay + az);  //    hi
-	}
-}
-void fm_i2_cube(const ABFSAMPLE *inbuf,float *outbuf,unsigned int numframes)
-{
-	unsigned int i;	
-	float *p_out = outbuf;	
-	double aw,ax,ay,az,as,at,av;
-	
-	for(i=0;i < numframes; i++){
-		aw = (double) inbuf->W * 0.1768;
-		ax = (double) inbuf->X * 0.114;
-		ay = (double) inbuf->Y * 0.114;
-        az = (double) inbuf->Z * 0.114;
-        as = (double) inbuf->S * 0.0369;
-        at = (double) inbuf->T * 0.0369;
-		av = (double) inbuf->V * 0.0369;
-        
-		*p_out++ = (float) (aw + ax + ay - az - as - at + av); //FL low 
-		*p_out++ = (float) (aw + ax + ay + az + as + at + av); //   hi 
-		
-        *p_out++ = (float) (aw - ax + ay - az + as - at - av); //RL low
-		*p_out++ = (float) (aw - ax + ay + az - as + at - av);  
-        
-        *p_out++ = (float) (aw - ax - ay - az + as + at + av); // RR low
-        *p_out++ = (float) (aw - ax - ay + az - as - at + av);  
-        
-        *p_out++ = (float) (aw + ax - ay - az - as + at - av);  // FR low
-        *p_out++ = (float) (aw + ax - ay + az + as - at - av);   
-	}
-}
-/* ditto, wavex order */
-/* Front L, front R, Back L, Back R; top Front L, Top Fr R, Top Back L, Top back R */
-void fm_i1_cubex(const ABFSAMPLE *inbuf,float *outbuf,unsigned int numframes)
-{
-	unsigned int i;	
-	float *p_out = outbuf;
-	double aw,ax,ay,az;
-	
-	for(i=0;i < numframes; i++){
-		aw = (double) inbuf->W * 0.17677;
-		ax = (double) inbuf->X * 0.07216;
-		ay = (double) inbuf->Y * 0.07216;
-		az = (double) inbuf->Z * 0.07216;
-		
-		*p_out++ = (float) (aw + ax + ay - az);  // FL low
-        *p_out++ = (float) (aw + ax - ay - az);  // FR low
-        *p_out++ = (float) (aw - ax + ay - az);  // RL low
-        *p_out++ = (float) (aw - ax - ay - az);  // RR low
-        
-		*p_out++ = (float) (aw + ax + ay + az);  // FL hi
-        *p_out++ = (float) (aw + ax - ay + az);  // FR hi
-		*p_out++ = (float) (aw - ax + ay + az);  // RL hi
-        *p_out++ = (float) (aw - ax - ay + az);  // RR hi
-	}
-}
-void fm_i2_cubex(const ABFSAMPLE *inbuf,float *outbuf,unsigned int numframes)
-{
-	unsigned int i;	
-	float *p_out = outbuf;	
-	double aw,ax,ay,az,as,at,av;
-	
-	for(i=0;i < numframes; i++){
-		aw = (double) inbuf->W * 0.1768;
-		ax = (double) inbuf->X * 0.114;
-		ay = (double) inbuf->Y * 0.114;
-        az = (double) inbuf->Z * 0.114;
-        as = (double) inbuf->S * 0.0369;
-        at = (double) inbuf->T * 0.0369;
-		av = (double) inbuf->V * 0.0369;
-        
-		*p_out++ = (float) (aw + ax + ay - az - as - at + av);  // FL low
-        *p_out++ = (float) (aw + ax - ay - az - as + at - av);  // FR low
-        *p_out++ = (float) (aw - ax + ay - az + as - at - av);  // RL low
-        *p_out++ = (float) (aw - ax - ay - az + as + at + av);  // RR low
-		
-        *p_out++ = (float) (aw + ax + ay + az + as + at + av);  // FL  hi 
-		*p_out++ = (float) (aw + ax - ay + az + as - at - av);  // FR  hi
-		*p_out++ = (float) (aw - ax + ay + az - as + at - av);  // RL  hi 
-        *p_out++ = (float) (aw - ax - ay + az - as - at + av);  // RR  hi 
-	}
-}
-
-
-#ifdef NOTDEF
-void bfdcode4(float *inbuf,long numframes)
-{
-	int i;	
-	float *p_buf = inbuf;	
-	double aw,ax,ay;
-	
-	for(i=0;i < numframes; i++){
-		aw = (double)(p_buf[0]) ;
-		ax = (double)(p_buf[1]) * 0.707;
-		ay = (double)(p_buf[2]) * 0.707;		
-		//decode frame
-		*p_buf++ = (float)(0.3333 * (aw + ax + ay)); //FL
-		*p_buf++ = (float)(0.3333 * (aw + ax - ay)); //FR
-		*p_buf++ = (float)(0.3333 * (aw - ax + ay)); //RL
-		*p_buf++ = (float)(0.3333 * (aw - ax - ay));  //RR
-	}
-}
-
-
-/* handle 3ch in to 4ch out! */
-
-void bfdcode324(float *inbuf,float*outbuf,long numframes)
-{
-	int i;	
-	float *p_buf = inbuf;
-	float * p_outbuf = outbuf;
-	double aw,ax,ay;
-	
-	for(i=0;i < numframes; i++){
-		int j;
-		aw = (double)(*p_buf++) ;
-		ax = (double)(*p_buf++) * 0.707;
-		ay = (double)(*p_buf++) * 0.707;
-        
-		//decode frame
-		*p_outbuf++ = (float)(0.3333 * (aw + ax + ay));
-		*p_outbuf++ = (float)(0.3333 * (aw + ax - ay));
-		*p_outbuf++ = (float)(0.3333 * (aw - ax + ay));
-		*p_outbuf++ = (float)(0.3333 * (aw - ax - ay));
-	}
-}
-#endif

dev/externals/paprogs/paplay/paplay.c

@@ -1,1593 +0,0 @@
-/*
- * Copyright (c) 1983-2013 Richard Dobson and Composers Desktop Project Ltd
- * http://people.bath.ac.uk/masrwd
- * http://www.composersdesktop.com
- * This file is part of the CDP System.
- * The CDP System is free software; you can redistribute it
- * and/or modify it under the terms of the GNU Lesser General Public
- * License as published by the Free Software Foundation; either
- * version 2.1 of the License, or (at your option) any later version.
- *
- * The CDP System is distributed in the hope that it will be useful,
- * but WITHOUT ANY WARRANTY; without even the implied warranty of
- * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.
- * See the GNU Lesser General Public License for more details.
- * You should have received a copy of the GNU Lesser General Public
- * License along with the CDP System; if not, write to the Free Software
- * Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
- *
- */
-
-/* paplay.c
- * Play a soundfile using the Portable Audio api for Max OS X using Core Audio.
- *
- • new version using threads and ring buffer, with thanks to Robert Bielik
-*/
-
-/* Dec 2005 support 3ch B-Format, b-format files via portsf */
-/* OCT 2009 much revised, extended b-format decoding options */
-/*     set ch mask to zero on Windows mme/ds.
-       TODO: decide whether/how to enable user to pass chmask of file to driver!
-*/
-/* Apr 2011 build with new portaudio stable v 19 */
-/* July 2012 added "to" option, memory playback mode */
-/* August 2012  restored -x flag */
-
-#include <stdio.h>
-#include <stdlib.h>
-#include <memory.h>
-#include <assert.h>
-#include <time.h>
-
-#ifdef _WIN32
-#define _WIN32_WINNT 0x0500
-#include <conio.h>
-#include <windows.h>
-#include <process.h>
-// NB: need to have PAWIN_USE_WDMKS_DEVICE_INFO defined
-#include "pa_win_ds.h"
-#endif
-
-#ifdef unix
-
-#include <sys/types.h>
-#include <sys/timeb.h>
-#endif
-
-#include <signal.h>
-
-#ifdef unix
-#include <aaio.h>
-#endif
-
-#ifdef MAC
-#include <libkern/OSAtomic.h>
-#endif
-
-#ifdef unix
-#include <sys/time.h>
-#include <pthread.h>
-/* in portsf.lib */
-extern int stricmp(const char *a, const char *b);
-#endif
-
-#include "portaudio.h"
-#include "pa_ringbuffer.h"
-#include "pa_util.h"
-#include "portsf.h"
-#include "fmdcode.h"
-
-
-
-#ifndef min
-#define min(x,y)  ((x) < (y) ? (x) : (y))
-#endif
-
-// TODO: scale by sample rate
-
-
-#define FRAMES_PER_BUFFER (4096)
-#define RINGBUF_NFRAMES    (32768)
-#define NUM_WRITES_PER_BUFFER   (4)
-
-
-enum {FM_MONO,FM_STEREO,FM_SQUARE,FM_QUAD,FM_PENT,DM_5_0,DM_5_1,FM_HEX,FM_OCT1,FM_OCT2,FM_CUBE,FM_QUADCUBE,FM_NLAYOUTS};
-
-#ifdef _WIN32
-HANDLE ghEvent;
-#endif
-
-
-typedef struct {
-    PaUtilRingBuffer ringbuf;
-    PaStream *stream;
-    PaTime startTime;
-    PaTime lastTime;
-    float *ringbufData;
-    float *inbuf;   //for decoding and other tx; used as temp working buffer for read thread func.
-    /* for memory read func */
-    float *membuf;
-    float** orderptr; // for chan mapping;
-#ifdef _WIN32
-    void *hThread;
-    HANDLE hTimer;
-    HANDLE hTimerCount;
-#else
-#ifdef unix
-    pthread_t hThread;
-#endif
-#endif
-    fmhdecodefunc decodefunc;
-    fmhcopyfunc copyfunc;
-    double gain;
-    unsigned long frames_played;
-    unsigned long current_frame;
-    unsigned long from_frame;
-    unsigned long to_frame;
-    unsigned long memFramelength;
-    unsigned long memFramePos;
-    unsigned long inbuflen;
-    int srate;
-    int inchans;
-    int outchans;
-    int flag;
-    int ifd;
-#ifdef DEBUG
-    int ofd;
-#endif
-    int do_decode;
-    int do_mapping;
-    int play_looped;
-    int finished;
-} psfdata;
-
-static int file_playing;
-static psfdata *g_pdata = NULL;  // for timer interrupt routine
-
-void playhandler(int sig)
-{
-    if(sig == SIGINT)
-        file_playing = 0;
-}
-
-
-#ifdef unix
-void alarm_wakeup (int i)
-{
-    struct itimerval tout_val;
-
-    signal(SIGALRM,alarm_wakeup);
-
-
-    if(file_playing && g_pdata->stream) {
-        //printf("%.4f secs\r",(float)(g_pdata->frames_played /(double) g_pdata->srate));
-        g_pdata->lastTime = Pa_GetStreamTime(g_pdata->stream ) - g_pdata->startTime;
-        printf("%.2f secs\r", g_pdata->lastTime);
-        fflush(stdout);
-    }
-    tout_val.it_interval.tv_sec = 0;
-    tout_val.it_interval.tv_usec = 0;
-    tout_val.it_value.tv_sec = 0;
-    tout_val.it_value.tv_usec = 200000;
-
-    setitimer(ITIMER_REAL, &tout_val,0);
-
-}
-#endif
-
-
-void finishedCallback(void *userData)
-{
-    psfdata *pdata = (psfdata*) userData;
-    //    printf("stream finished!\n");
-    pdata->finished = 1;
-    file_playing = 0;
-
-
-}
-
-
-#ifdef _WIN32
-
-VOID CALLBACK TimerCallback(PVOID lpParam, BOOLEAN TimerOrWaitFired)
-{
-    psfdata *pdata = (psfdata*) lpParam;
-
-    if(file_playing && pdata->stream) {
-        //printf("%.4f secs\r",(float)(g_pdata->frames_played /(double) g_pdata->srate));
-        pdata->lastTime = Pa_GetStreamTime(pdata->stream ) - pdata->startTime;
-        printf("%.2f secs\r", pdata->lastTime);
-        fflush(stdout);
-    }
-
-    SetEvent(ghEvent);
-}
-
-#endif
-/* thread func for file playback */
-/* writes decoded data to ring buffer */
-/* size of element is full m/c frame */
-
-// TODO: unix thread should return void*
-#ifdef unix
-static  int  threadFunctionReadFromRawFile(void* ptr)
-#else
-    static unsigned int __stdcall threadFunctionReadFromRawFile(void* ptr)
-#endif
-{
-    psfdata* pdata = (psfdata*)ptr;
-
-    /* Mark thread started */
-    pdata->flag = 0;
-
-    while (1) {
-        ring_buffer_size_t nElementsProcessed = 0;
-        ring_buffer_size_t elementsInBuffer = PaUtil_GetRingBufferWriteAvailable(&pdata->ringbuf);
-
-        //memset(pdata->inbuf,0,pdata->inbuflen * sizeof(float) * pdata->inchans);
-
-        if (elementsInBuffer >= pdata->ringbuf.bufferSize / 4) {
-            void* ptr[2] = {NULL};
-            ring_buffer_size_t sizes[2] = {0};
-
-            /* By using PaUtil_GetRingBufferWriteRegions, we can write directly into the ring buffer */
-            PaUtil_GetRingBufferWriteRegions(&pdata->ringbuf, elementsInBuffer, &ptr[0], &sizes[0], &ptr[1], &sizes[1]);
-
-            if (file_playing)  {
-                ring_buffer_size_t itemsReadFromFile;
-                int i,j;
-                int framesread = 0;
-                // we work with frames, = constant across inchans and outchans
-                itemsReadFromFile = 0;
-                for(i = 0; i < 2 && (ptr[i] != NULL); i++) {
-                    // NB ringbuf is sized by m/c frames
-                    int frameswanted = sizes[i];
-
-                    pdata->current_frame = psf_sndTell(pdata->ifd);
-                    // read up to end frame if requested
-                    if(pdata->to_frame < pdata->current_frame + frameswanted)
-                        frameswanted = pdata->to_frame - pdata->current_frame;
-
-                    if(frameswanted > 0){
-                        framesread = psf_sndReadFloatFrames(pdata->ifd,pdata->inbuf,frameswanted);
-                        if(framesread < 0){ // read error!
-                            printf("Error reading soundfile\n");
-                            pdata->flag = 1;
-                            file_playing = 0;
-                            break;  // just out of for loop - need to return instead?
-                        }
-                    }
-                    if(framesread == 0){
-                        /* EOF. EITHER: finish, or rewind if looping playback*/
-                        if(pdata->play_looped){
-                            if(psf_sndSeek(pdata->ifd,pdata->from_frame,PSF_SEEK_SET)){
-                                printf("Error looping soundfile\n");
-                                pdata->flag = 1;
-                                file_playing = 0;
-                                break;
-                            }
-                            // sizes[1] especially may well = 0
-                            if(frameswanted==0)
-                                break;
-                            framesread = psf_sndReadFloatFrames(pdata->ifd,pdata->inbuf,frameswanted);
-                            if(framesread < 0){ // read error!
-                                printf("Error reading soundfile\n");
-                                pdata->flag = 1;
-                                file_playing = 0;
-                                break;
-                            }
-                        }
-                        else {
-                            // we must watch the ring buffer to make sure all data has been rendered,
-                            // over several callback blocks
-                            //printf("End of data. playing = %d:\n", file_playing);
-                            //printf("\telements remaining = %d\n",elementsInBuffer);
-                            if(elementsInBuffer == pdata->ringbuf.bufferSize)  {
-                                pdata->flag = 1;
-                                break;
-                                //return 0;
-                            }
-                        }
-                    }
-                    else {
-                        // ringbuf calcs always relative to outchans
-                        itemsReadFromFile += framesread;
-
-                        // now ready to apply decoding or other processing
-
-                        if(pdata->gain != 1.0){
-                            for(j = 0; j < framesread * pdata->inchans; j++)
-                                pdata->inbuf[j]  = (float)(pdata->inbuf[j] * pdata->gain);
-                        }
-                        if(pdata->do_mapping) {
-                            int k;
-                            float val;
-                            float *buf = ptr[i];
-                            float *pchan;
-                            for(j=0; j < framesread; j++){
-                                for(k=0; k < pdata->outchans; k++){
-                                    pchan = pdata->orderptr[k];
-                                    if(pchan == NULL)
-                                        val = 0.0f;
-                                    else
-                                        val = *(pchan + j * pdata->inchans);
-                                    buf[j*pdata->outchans + k] = val;
-                                }
-                            }
-                        }
-                        else if(pdata->do_decode) {
-                            ABFSAMPLE abfsamp;
-
-                            for(j=0; j < framesread; j++){
-                                // single frame only
-                                pdata->copyfunc(&abfsamp,pdata->inbuf + (j * pdata->inchans));
-                                /* BIG TODO: update funcs to process large frame buffer! */
-                                /* NB: ring buffer is effectively defined as raw bytes */
-#ifdef unix
-                                pdata->decodefunc(&abfsamp,ptr[i] + (j * pdata->outchans * sizeof(float)), 1);
-#else
-                                // suppress VC++ complaint of unknown size
-                                pdata->decodefunc(&abfsamp, (float*)(ptr[i]) + (j * pdata->outchans), 1);
-
-#endif
-                                nElementsProcessed++;
-                            }
-                        }
-                        else {  // inchans = outchans
-                            memcpy(ptr[i],pdata->inbuf,framesread * sizeof(float) * pdata->inchans);
-                            nElementsProcessed += framesread;
-                        }
-                    }
-                }
-                //                assert(nElementsProcessed == itemsReadFromFile);
-                if(framesread)
-                    PaUtil_AdvanceRingBufferWriteIndex(&pdata->ringbuf, itemsReadFromFile);
-
-
-            }
-            else {
-                // this code is activated on Ctrl-C. Can do immediate finish by setting flag
-                //               printf("file done\n");
-                pdata->flag = 1;
-                break;
-            }
-        }
-        //  else {
-        //      printf("ringbuf size = %d, elements remaining = %d, playing = %d\n",pdata->ringbuf.bufferSize,elementsInBuffer,file_playing);
-        //  }
-
-        /* Sleep a little while... */
-        Pa_Sleep(10);
-    }
-    return 0;
-}
-
-
-
-#ifdef unix
-// TODO: unix return type should be void*
-typedef int (*threadfunc)(void*);
-#endif
-#ifdef _WIN32
-typedef unsigned int (__stdcall *threadfunc)(void*);
-#endif
-
-
-/* Start up a new thread for given function */
-static PaError startThread( psfdata* pdata, threadfunc fn )
-{
-    pdata->flag = 1;
-#ifdef _WIN32
-    pdata->hThread = (void*)_beginthreadex(NULL, 0, fn, pdata, 0, NULL);
-    if (pdata->hThread == NULL)
-        return paUnanticipatedHostError;
-    /* Wait for thread to startup */
-    while (pdata->flag) {
-        Pa_Sleep(10);
-    }
-    /* Set file thread to a little higher prio than normal */
-    SetThreadPriority(pdata->hThread, THREAD_PRIORITY_ABOVE_NORMAL);
-#else
-
-#if defined(__APPLE__) || defined(__GNUC__)
-    if(pthread_create(&pdata->hThread,NULL,(void*) fn,pdata))
-        return -1;
-    /* Wait for thread to startup */
-    while (pdata->flag) {
-        Pa_Sleep(10);
-    }
-#endif
-#endif
-
-    return paNoError;
-}
-
-#if 0
-static int stopThread( psfdata* pdata )
-{
-    // RWD: just called when all data played; must be called before StopStream
-    //   pdata->flag = 1;
-    /* Wait for thread to stop */
-    //   while (pdata->flag) {
-    //       Pa_Sleep(10);
-    //   }
-#ifdef _WIN32
-    CloseHandle(pdata->hThread);
-    pdata->hThread = 0;
-#else
-
-#if defined(__APPLE__) || defined(__GNUC__)
-    pthread_cancel(pdata->hThread);
-#endif
-#endif
-    return paNoError;
-}
-#endif
-
-static int paplayCallback(const void *inputBuffer, void *outputBuffer,
-                          unsigned long framesPerBuffer,
-                          const PaStreamCallbackTimeInfo* timeInfo,
-                          PaStreamCallbackFlags statusFlags,
-                          void *userData )
-{
-    psfdata *data = (psfdata *)userData;
-    ring_buffer_size_t sampsAvailable = PaUtil_GetRingBufferReadAvailable(&data->ringbuf) * data->outchans;
-    ring_buffer_size_t sampsToPlay = min(sampsAvailable, (ring_buffer_size_t)(framesPerBuffer * data->outchans));
-    float *out = (float*) outputBuffer;
-    int framesToPlay = sampsToPlay/data->outchans;
-    int played;
-    /* outbuf may be NULL on initial startup */
-    if(out == NULL)
-        return paContinue;
-
-    // if framestoplay < framesPerBuffer, need to clean up the remainder
-    memset(out,0,framesPerBuffer * data->outchans * sizeof(float));
-    played = PaUtil_ReadRingBuffer(&data->ringbuf, out, framesToPlay);
-
-    data->frames_played += played;
-    if(data->flag) {
-        //printf("callback - complete\n");
-        return paComplete;
-    }
-    return paContinue;
-}
-
-
-
-static int MemCallback(const    void *inputBuffer, void *outputBuffer,
-                       unsigned long framesPerBuffer,
-                       const PaStreamCallbackTimeInfo* timeInfo,
-                       PaStreamCallbackFlags statusFlags,
-                       void *userData )
-{
-    psfdata *pdata = (psfdata *)userData;
-
-    float *out = (float*) outputBuffer;
-    unsigned long inSamps,outSamps, inSampPos;
-    unsigned long i;
-    unsigned long framesToPlay = framesPerBuffer;
-
-    /* outbuf may be NULL on initial startup */
-    if(out == NULL)
-        return paContinue;
-
-    // if framesToPlay < framesPerBuffer, need to clean up the remainder
-    memset(out,0,framesPerBuffer * pdata->outchans * sizeof(float));
-    inSamps = pdata->memFramelength * pdata->inchans;
-    outSamps = framesPerBuffer * pdata->inchans;
-    inSampPos = pdata->memFramePos * pdata->inchans;
-
-    if(pdata->play_looped){
-        for(i=0;i < outSamps; i++){
-            pdata->inbuf[i] = pdata->membuf[inSampPos++] * pdata->gain;
-            if(inSampPos == inSamps)
-                inSampPos = 0;
-        }
-    }
-    else {
-        for(i=0;i < outSamps; i++){
-            pdata->inbuf[i] = pdata->membuf[inSampPos++] * pdata->gain;
-            if(inSampPos == inSamps)
-                break;
-        }
-        framesToPlay = i / pdata->inchans;
-        for(; i < outSamps;i++){
-            pdata->inbuf[i] = 0.0;
-        }
-
-    }
-    // write to output
-    // everything now in inbuf
-    if(pdata->do_decode) {
-        ABFSAMPLE abfsamp;
-        int j;
-        for(j=0; j < framesToPlay; j++){
-            pdata->copyfunc(&abfsamp,pdata->inbuf + (j * pdata->inchans));
-            pdata->decodefunc(&abfsamp,out + (j * pdata->outchans), 1);
-        }
-    }
-    else {  // inchans = outchans
-        memcpy(out,pdata->inbuf,framesToPlay * sizeof(float) * pdata->inchans);
-    }
-    pdata->memFramePos = inSampPos / pdata->inchans;
-    pdata->frames_played += framesToPlay;
-    if(framesToPlay < framesPerBuffer)
-        pdata->flag = 1;           // end of stream
-    if(pdata->flag) {
-        //printf("\nnewcallback - complete\n");
-        //fflush(stdout);
-        return paComplete;
-    }
-    return paContinue;
-}
-
-
-
-
-static unsigned NextPowerOf2(unsigned val)
-{
-    val--;
-    val = (val >> 1) | val;
-    val = (val >> 2) | val;
-    val = (val >> 4) | val;
-    val = (val >> 8) | val;
-    val = (val >> 16) | val;
-    return ++val;
-}
-
-
-int show_devices(void);
-
-#define N_BFORMATS (10)
-//static const int bformats[N_BFORMATS] = {2,3,4,5,6,7,8,9,11,16};
-//static const int layout_chans[] = {1,2,4,4,5,5,6,6,8,8,8,8};
-
-enum {FLAG_B = 0, FLAG_BM, FLAG_D, FLAG_G, FLAG_H, FLAG_I, FLAG_L, FLAG_M, FLAG_NFLAGS};
-
-
-void usage(void){
-#ifdef WIN32
-    printf("usage:\n  paplay [-BN][-dN][-gN]-hN][-i][-l][-b[N]|-m[S]][-u][-x] infile [from] [to] \n"
-#else
-           printf("usage:\n  paplay [-BN][-dN][-gN]-hN][-i][-l][-b[N]|-m[S]][-u] infile [from] [to]\n"
-#endif
-                  "       -dN  : use output Device N.\n"
-                  "       -gN  : apply gain factor N to input.\n"
-                  "       -BN  : set memory buffer size to N frames (default: %d).\n"
-                  "                N must be a power of 2 (e.g 4096, 8192, 16384 etc).\n"
-                  "       -hN  : set hardware blocksize to N frames (32 < N <= BN/4).\n"
-                  "               (N recommended to be a power of two size. Default: from soundcard).\n"
-                  "               Where set, buffer sizes are doubled internally for sample rates > 48KHz.\n"
-                  "       -i   : play immediately (do not wait for keypress).\n"
-                  "       -l   : loop file continuously, from start-time to end-time.\n"
-                  "       -m[S]: render using channel map string S.\n"
-                  "                Use -m without parameter for usage.\n"
-                  "       -u   : suppress elapsed time updates\n"
-#ifdef WIN32
-                  "       -x   : Apply WAVE_EX infile channel mask to DS audio stream\n"
-                  "              (ignored if -m or -b used)\n"
-#endif
-                  "       from : set start time (secs) for playback and looping. Default: 0.\n"
-                  "       to   : set end time (secs) for playback and looping. Default: EOF.\n"
-                  "       -b[N]: apply 1st-order B-Format decoding to standard soundfile.\n"
-                  "              (file must have at least 3 channels)\n"
-                  "               B-Format files (.amb) will be decoded automatically.\n"
-                  "               N sets speaker layout to one of the following:\n"
-                  "               1    :  *  mono (= W signal only)\n"
-                  "               2    :  *  stereo (quasi mid/side, = W +- Y)\n"
-                  "               3    :     square\n"
-                  "               4    :  *  quad FL,FR,RL,RR order   (default)\n"
-                  "               5    :     pentagon\n"
-                  "               6    :  *  5.0 surround (WAVEX order)\n"
-                  "               7    :  *  5.1 surround (WAVEX order, silent LFE)\n"
-                  "               8    :     hexagon\n"
-                  "               9    :     octagon 1 (front pair, 45deg)\n"
-                  "              10    :     octagon 2 (front centre speaker)\n"
-                  "              11    :     cube (as 3, low-high interleaved. Csound-compatible.)\n"
-                  "              12    :  *  cube (as 4, low quad followed by high quad).\n"
-                  "              Default decode layout is 4 (quad).\n"
-                  "              NB: no decoding is performed if -m flag used.\n"
-                  "paplay reads PEAK chunk if present to rescale over-range floatsam files.\n\n"
-                  ,RINGBUF_NFRAMES);
-           }
-
-        void mapusage(void)
-        {
-            printf( "  PAPLAY channel map mode -mS: Map arbitrary infile channels to output channels\n\n"
-                    "  Order string S = any combination of characters a-z inclusive.\n"
-                    "  Infile channels are mapped in order as a=1,b=2...z=26\n"
-                    "  (For example: channels in a 4-channel file are represented by the\n"
-                    "    characters abcd; any other character is an error).\n"
-                    "  Characters must be lower case, and may be used more than once.\n"
-                    "  Duplicate characters duplicate the corresponding input channel.\n"
-                    "  The zero character (0) may be used to set a silent channel.\n"
-                    "  A maximum of 26 channels is supported for both input and output.\n"
-                    "  The number of characters in S sets the number of output channels,\n"
-                    "    which must be supported by the selected device.\n"
-                    "  NB: -m cannot be combined with -b (B-Format decoding).\n"
-                    "  If channel mapping is used, a B-format file (AMB) \n"
-                    "    will be rendered without decoding.\n"
-                    );
-        }
-
-    /*******************************************************************/
-    int main(int argc,char **argv)
-    {
-        PaStreamParameters outputParameters;
-#ifdef _WIN32
-        /* portaudio sets default channel masks we can't use; we must do all this to set default mask = 0! */
-        PaWinDirectSoundStreamInfo directSoundStreamInfo;
-        //    PaWinMmeStreamInfo winmmeStreamInfo;
-
-#endif
-        PaDeviceInfo *devinfo = NULL;
-        PaStream *stream = NULL;
-        PaStreamCallback *playcallback = paplayCallback;
-        PaError err = paNoError;
-        psfdata sfdata;
-        PSF_CHPEAK *fpeaks = NULL;
-        PSF_PROPS props;
-        // ABFSAMPLE abfsample;
-        // ABFSAMPLE *abf_frame = NULL;
-        int res,inorder = 1;
-        time_t in_peaktime = 0;
-        MYLONG lpeaktime;
-        int waitkey = 1;
-        //int play_looped = 0;
-        double fromdur = 0.0;
-        double totime  = 0.0;
-        int ifd = -1,from_frame = 0, to_frame = 0;
-        unsigned int i=0;
-        PaDeviceIndex device;
-        long filesize;
-        //long framesize = 0;   // Apr 2010 to be scaled by sample rate
-        unsigned long ringframelen = RINGBUF_NFRAMES;  // length of ring buffer in m/c frames
-        unsigned long frames_per_buffer = 0;
-        unsigned long nFramesToPlay = 0;
-        double gain = 1.0;
-        unsigned int inchans,outchans;
-        int layout = 4; // default, quad decode */
-        fmhcopyfunc copyfunc = NULL;
-        fmhdecodefunc decodefunc = NULL;
-        // vars for chorder facilit
-        char* argstring = NULL;
-        unsigned int rootchar = 'a';
-        unsigned int maxchar = 'z';
-        unsigned int  nchars=0,nzeros = 0;
-        unsigned int max_inchar = 0;
-        unsigned int flags[FLAG_NFLAGS] = {0};
-#ifdef WIN32
-        int do_speakermask = 0;
-        int speakermask = 0;
-#endif
-        int do_updatemessages = 1;
-#ifdef unix
-        struct itimerval tout_val;
-        tout_val.it_interval.tv_sec = 0;
-        tout_val.it_interval.tv_usec = 0;
-        tout_val.it_value.tv_sec = 0;
-        tout_val.it_value.tv_usec = 200000;
-#endif
-
-#ifdef _WIN32
-        HANDLE hTimerQueue;
-        ghEvent = CreateEvent(NULL,TRUE,FALSE,NULL);
-        if(ghEvent==NULL){
-            printf("Failed to start internal timer (1).\n");
-            return 1;
-        }
-        hTimerQueue = CreateTimerQueue();
-        if(hTimerQueue == NULL){
-            printf("Failed to start internal timer (2).\n");
-            return 1;
-        }
-#endif
-        /* CDP version number */
-        if(argc==2 && (stricmp(argv[1],"--version")==0)){
-            printf("3.0.1\n");
-            return 0;
-        }
-
-        signal(SIGINT,playhandler);
-
-        sfdata.inbuf = NULL;
-        sfdata.membuf = NULL;
-        sfdata.memFramelength = 0;
-        sfdata.memFramePos = 0;
-        sfdata.ringbufData = NULL;
-        sfdata.orderptr = NULL;
-        sfdata.do_mapping = 0;
-
-        printf("PAPLAY: play multi-channel soundfile. V3.0.1 (c)  RWD,CDP 2012, 2013\n");
-        file_playing = 0;
-        err = Pa_Initialize();
-        if( err != paNoError ) {
-            printf("Failed to initialize Portaudio.\n");
-            return 1;
-        }
-        device =  Pa_GetDefaultOutputDevice();
-
-        if(argc < 2){
-            usage();
-            show_devices();
-            Pa_Terminate();
-            return 1;
-        }
-
-        while(argv[1][0]=='-'){
-            unsigned long userbuflen;
-            switch(argv[1][1]){
-            case('b'):
-                if(flags[FLAG_B]){
-                    printf("error: multiple -b flags\n");
-                    Pa_Terminate();
-                    return 1;
-                }
-                if(argv[1][2] != '\0'){
-                    layout = atoi(&argv[1][2]);
-                    if(layout < 1 || layout > 12){
-                        fprintf(stderr,"value for -b flag out of range.\n");
-                        Pa_Terminate();
-                        return 1;
-                    }
-                }
-                flags[FLAG_B] = 1;
-                break;
-            case('d'):
-                if(flags[FLAG_D]){
-                    printf("error: multiple -d flags\n");
-                    Pa_Terminate();
-                    return 1;
-                }
-                if(argv[1][2]=='\0'){
-                    printf("-d flag requires parameter\n");
-                    Pa_Terminate();
-                    return 1;
-                }
-                device = atoi(&(argv[1][2]));
-                flags[FLAG_D] = 1;
-                break;
-            case('g'):
-                if(flags[FLAG_G]){
-                    printf("error: multiple -g flags\n");
-                    Pa_Terminate();
-                    return 1;
-                }
-                if(argv[1][2]=='\0'){
-                    printf("-g flag requires parameter\n");
-                    Pa_Terminate();
-                    return 1;
-                }
-                gain = atof(&(argv[1][2]));
-                if(gain <= 0.0){
-                    fprintf(stderr,"gain value must be positive.\n");
-                    Pa_Terminate();
-                    return 1;
-                }
-                flags[FLAG_G] = 1;
-                break;
-            case 'h':
-                if(flags[FLAG_H]){
-                    printf("error: multiple -h flags\n");
-                    Pa_Terminate();
-                    return 1;
-                }
-                if(argv[1][2]=='\0'){
-                    printf("-h flag requires parameter\n");
-                    Pa_Terminate();
-                    return 1;
-                }
-                frames_per_buffer = atoi(&(argv[1][2]));
-                if((frames_per_buffer > 0) && (frames_per_buffer < 32)){
-                    printf("-h value too small: must be at least 32.\n");
-                    Pa_Terminate();
-                    return 1;
-                }
-                flags[FLAG_H] = 1;
-                break;
-            case('i'):
-                if(flags[FLAG_I]){
-                    printf("error: multiple -i flags\n");
-                    Pa_Terminate();
-                    return 1;
-                }
-                flags[FLAG_I] = 1;
-                waitkey = 0;
-                break;
-            case ('l'):
-                flags[FLAG_L] = 1;
-                break;
-            case 'm':
-                if(flags[FLAG_M]){
-                    printf("error: multiple -m flags\n");
-                    Pa_Terminate();
-                    return 1;
-                }
-                if(argv[1][2]=='\0'){
-                    mapusage();
-                    Pa_Terminate();
-                    return 1;
-                }
-                argstring = &argv[1][2];
-                nchars = strlen(argstring);
-                if(nchars > 26) {
-                    printf("error: -m order list too long.\n");
-                    Pa_Terminate();
-                    return 1;
-                }
-                if(nchars < 1){
-                    printf("error: -m must specify at least one channel.\n");
-                    Pa_Terminate();
-                    return 1;
-
-                }
-                flags[FLAG_M] = 1;
-                break;
-
-                // RWD Feb 2011: need to support dummy -u flag for compatibility with pvplay
-                // will we actually need to implement it?
-            case 'u':
-                do_updatemessages = 0;
-                break;
-#ifdef WIN32
-            case 'x':
-                do_speakermask = 1;
-                break;
-#endif
-            case 'B':
-                if(flags[FLAG_BM]){
-                    printf("error: multiple -B flags\n");
-                    Pa_Terminate();
-                    return 1;
-                }
-                if(argv[1][2]=='\0'){
-                    printf("-B flag requires parameter\n");
-                    Pa_Terminate();
-                    return 1;
-                }
-
-                ringframelen = atoi(&(argv[1][2]));
-                if(ringframelen <= 1024){
-                    printf("-B: framesize value must be >=1024!n");
-                    Pa_Terminate();
-                    return 1;
-                }
-                userbuflen = NextPowerOf2(ringframelen);
-                if(userbuflen != ringframelen){
-                    printf("-B: framesize value must be power of 2 size\n");
-                    Pa_Terminate();
-                    return 1;
-                }
-                flags[FLAG_BM] = 1;
-                break;
-            default:
-                printf("unrecognised flag option\n");
-                Pa_Terminate();
-                return 1;
-            }
-            argv++;  argc--;
-        }
-        // this just cheks flag usage. We render amb files undecoded, if using channel mapping
-        if(flags[FLAG_B] && flags[FLAG_M]){
-            printf("Sorry: cannot use -m with -b\n");
-            Pa_Terminate();
-            return 1;
-        }
-        if(frames_per_buffer > ringframelen/4){
-            printf(" hardware (-h) framesize %lu too large for file buffer (-B) size %lu\n",
-                   frames_per_buffer,ringframelen);
-            Pa_Terminate();
-            return 1;
-        }
-
-        if(argc < 2 || argc > 4){
-            usage();
-            show_devices();
-            Pa_Terminate();
-            return 1;
-        }
-
-        if(argc>=3){
-            fromdur = atof(argv[2]);
-            if(fromdur < 0.0){
-                printf("Error: start position must be positive\n");
-                Pa_Terminate();
-                return 1;
-            }
-        }
-        if(argc==4){
-            totime = atof(argv[3]);
-            if(totime <= fromdur){
-                printf("Error: end time must be after from time\n");
-                Pa_Terminate();
-                return 1;
-            }
-
-
-        }
-        if(psf_init()){
-            printf("Error initializing psfsys\n");
-            goto error;
-        }
-        /* allow auto rescaling of overrange floats via PEAK chunk */
-        ifd = psf_sndOpen(argv[1],&props,1);
-        if(ifd < 0){
-            printf("unable to open soundfile %s\n",argv[1]);
-            goto error;
-        }
-        filesize = psf_sndSize(ifd);
-        if(filesize ==0){
-            printf("Soundfile is empty!\n");
-            goto error;
-        }
-
-#if WIN32
-        if(props.format == PSF_WAVE_EX){
-            if(do_speakermask){
-                if(flags[FLAG_B] || flags[FLAG_M]){
-                    printf("-x flag ignored if -B or -m used\n");
-                }
-                else {
-                    int mask = psf_speakermask(ifd);
-                    if(mask < 0){
-                        printf("could not read speaker mask. Using 0\n");
-                        speakermask = 0;
-                    }
-                    else
-                        speakermask = mask;
-                }
-            }
-        }
-#endif
-        from_frame = (long)(fromdur * props.srate);
-        if(from_frame >= filesize){
-            printf("Error: start is beyond end of file\n");
-            goto error;
-        }
-
-        nFramesToPlay = filesize - from_frame;
-        to_frame = filesize -1;
-        if(totime > 0.0){
-            to_frame = (long)(totime * props.srate);
-            if(to_frame >= filesize){
-                printf("Error: end time is beyond end of file\n");
-                goto error;
-            }
-            //printf("fromframe = %d, toframe = %d\n",(int)from_frame,(int) to_frame);
-            if(to_frame <= from_frame){
-                printf("end time must be later than from time\n");
-                goto error;
-            }
-            nFramesToPlay = to_frame - from_frame;
-            //printf("playing %d frames\n",(int) nFramesToPlay);
-        }
-
-        outchans = inchans = props.chans;
-        /* if using order string, outchans may change arbitrarily;
-           later on we set inchans to outchans,
-           to emulate a plain outchans-sized file
-        */
-        if(flags[FLAG_M]){
-            outchans = nchars;
-        }
-
-        if(props.chformat==MC_BFMT){
-            if(flags[FLAG_M]){
-                printf("-m specified: not decoding B-Format file.\n");
-                flags[FLAG_B] = 0;
-            }
-            else
-                flags[FLAG_B] = 1;
-        }
-
-        // we will not have both do_channel_mapping and do_decode toigether
-        if(flags[FLAG_B] == 1){
-            if(inchans > 4) {
-                inorder = 2;
-                printf("%u-channel input: performing 2nd-order decode.\n",inchans);
-            }
-            switch(inchans){
-            case 3:
-                copyfunc = fmhcopy_3;
-                break;
-            case 4:
-                copyfunc = fmhcopy_4;
-                break;
-            case 5:
-                copyfunc = fmhcopy_5;
-                break;
-            case 6:
-                copyfunc = fmhcopy_6;
-                break;
-            case 9:
-                copyfunc = fmhcopy_9;
-                break;
-            case 11:
-                copyfunc = fmhcopy_11;
-                break;
-            case 16:
-                copyfunc = fmhcopy_16;
-                break;
-            default:
-                fprintf(stderr,"unsupported channel count (%u) for B-format file.\n",inchans);
-                return 1;
-            }
-
-            switch(layout-1){
-            case FM_MONO:
-                printf("Decoding to Mono\n");
-                decodefunc = fm_i1_mono;
-                outchans = 1;
-                break;
-            case FM_STEREO:
-                printf("Decoding to Stereo\n");
-                decodefunc = fm_i1_stereo;
-                outchans = 2;
-                break;
-            case FM_SQUARE:
-                printf("Decoding to Square\n");
-                if(inorder == 1)
-                    decodefunc = fm_i1_square;
-                else
-                    decodefunc = fm_i2_square;
-                outchans = 4;
-                break;
-            case FM_PENT:
-                printf("Decoding to pentagon\n");
-                if(inorder==1)
-                    decodefunc = fm_i1_pent;
-                else
-                    decodefunc = fm_i2_pent;
-                outchans = 5;
-                break;
-            case DM_5_0:
-                printf("Decoding to 5.0 surround (David Moore)\n");
-                if(inorder==1)
-                    decodefunc = dm_i1_surr;
-                else
-                    decodefunc = dm_i2_surr;
-                outchans = 5;
-                break;
-            case DM_5_1:
-                printf("Decoding to  5.1 surround (David Moore)\n");
-                if(inorder==1)
-                    decodefunc = dm_i1_surr6;
-                else
-                    decodefunc = dm_i2_surr6;
-                outchans = 6;
-                break;
-            case FM_HEX:
-                printf("Decoding to Hexagon\n");
-                if(inorder==1)
-                    decodefunc = fm_i1_hex;
-                else
-                    decodefunc = fm_i2_hex;
-                outchans = 6;
-                break;
-            case FM_OCT1:
-                printf("Decoding to Octagon 1\n");
-                if(inorder==1)
-                    decodefunc = fm_i1_oct1;
-                else
-                    decodefunc = fm_i2_oct1;
-                outchans = 8;
-                break;
-            case FM_OCT2:
-                printf("Decoding to Octagon 2\n");
-                if(inorder==1)
-                    decodefunc = fm_i1_oct2;
-                else
-                    decodefunc = fm_i2_oct2;
-                outchans = 8;
-                break;
-            case FM_CUBE:
-                printf("Decoding to Cube (FM interleaved)\n");
-                if(inorder==1)
-                    decodefunc = fm_i1_cube;
-                else
-                    decodefunc = fm_i2_cube;
-                outchans = 8;
-                break;
-            case FM_QUADCUBE:
-                printf("Decoding to Octagon 1 (WAVEX order)\n");
-                if(inorder==1)
-                    decodefunc = fm_i1_cubex;
-                else
-                    decodefunc = fm_i2_cubex;
-                outchans = 8;
-                break;
-            default:
-                //quad
-                printf("Decoding to quad surround (WAVEX order)\n");
-                if(inorder == 1)
-                    decodefunc = fm_i1_quad;
-                else
-                    decodefunc = fm_i2_quad;
-                outchans = 4;
-                break;
-            }
-        }
-
-
-
-        printf("opened %s: %ld frames, %u channels\n",argv[1],filesize,inchans);
-        fpeaks = (PSF_CHPEAK *) calloc(inchans,sizeof(PSF_CHPEAK));
-        if(fpeaks==NULL){
-            puts("no memory for PEAK data\n");
-            goto error;
-        }
-        //lpeaktime = (int) in_peaktime;
-        res = psf_sndReadPeaks(ifd,fpeaks,(MYLONG *) &lpeaktime);
-        if(res==0) {
-            printf("no PEAK data in this soundfile\n");
-        }
-        else if(res < 0){
-            printf("Error reading PEAK data\n");
-            goto error;
-        }
-        else{
-            in_peaktime = (time_t) lpeaktime;
-            printf("PEAK data:\ncreation time: %s",ctime(&in_peaktime));
-
-            for(i=0;i < inchans;i++){
-                printf("CH %d: %.4f at frame %lu: \t%.4f secs\n",
-                       i,fpeaks[i].val,fpeaks[i].pos,(double)(fpeaks[i].pos / (double) props.srate));
-            }
-        }
-        // if doing channel map,  precompute into mem buffer
-        if(nFramesToPlay <= ringframelen){
-            sfdata.membuf =  (float *) PaUtil_AllocateMemory(nFramesToPlay * sizeof(float) * /*inchans*/ outchans);
-            if( sfdata.membuf == NULL )   {
-                puts("Could not allocate memory play buffer.\n");
-                goto error;
-            }
-            sfdata.memFramelength = nFramesToPlay;
-            sfdata.memFramePos = 0;
-            playcallback = MemCallback;
-            printf("RAM block size =  %lu\n",nFramesToPlay);
-
-        }
-        else {
-            // set up ring buffer  NB must be power of 2 size
-            if(props.srate > 48000)
-                ringframelen <<= 1;
-
-            // NB ring buffer sized for decoded data, hence outchans here; otherwise inchans = outchans
-            sfdata.ringbufData = (float *) PaUtil_AllocateMemory( ringframelen * sizeof(float) * outchans); /* From now on, recordedSamples is initialised. */
-            if( sfdata.ringbufData == NULL )   {
-                puts("Could not allocate play buffer.\n");
-                goto error;
-            }
-
-            // number of elements has to be a power of two, so each element has to be a full m/c frame
-            if (PaUtil_InitializeRingBuffer(&sfdata.ringbuf, sizeof(float) * outchans, ringframelen , sfdata.ringbufData) < 0) {
-                puts("Could not initialise play buffer.\n");
-                goto error;
-            }
-            printf("File buffer size = %ld\n",ringframelen);
-
-        }
-
-        // worst case, ring buffer is empty! So need enough space
-        // NB inchans may well be > outchans
-        sfdata.inbuf = (float *) PaUtil_AllocateMemory(ringframelen * sizeof(float) * inchans);
-        if(sfdata.inbuf==NULL){
-            puts("No memory for read buffer\n");
-            goto error;
-        }
-        sfdata.inbuflen = ringframelen;
-
-        if(flags[FLAG_M]) {
-            max_inchar = rootchar;
-
-            sfdata.orderptr = malloc(outchans * sizeof(float*));
-            for(i=0;i < nchars;i++){
-                unsigned int thischar = argstring[i];
-                //          printf("reading char %c (%d)\n",thischar,thischar);
-                unsigned int chindex;
-                if(thischar != '0' && (thischar < rootchar || thischar > maxchar)){
-                    printf("illegal character in order string: %c\n",thischar);
-                    goto error;
-                }
-                if(thischar =='0'){
-                    //            printf("setting channel %d to zero.\n",i);
-                    sfdata.orderptr[i] = NULL;
-                    nzeros++;
-                }
-                else{
-                    if(thischar > max_inchar)
-                        max_inchar = thischar;
-                    chindex = thischar - rootchar;
-                    sfdata.orderptr[i] = sfdata.inbuf + chindex;
-                }
-            }
-            if(nzeros==nchars)
-                printf("Warning: -m order string is all zeros - a silent file will be made!\n");
-            else{
-                /* count inclusively! */
-                if(props.chans < (int)(max_inchar - rootchar + 1)){
-                    printf("File has %d channels; order string defines non-existent channels.\n",props.chans);
-                    printf("For this file, maximum character is %c\n",rootchar + props.chans -1);
-                    goto error;
-                }
-            }
-            outchans = nchars;
-            sfdata.do_mapping = 1;
-
-            // final alloc when infile is read, below
-        }
-
-        if(props.srate > 48000) {
-            frames_per_buffer *= 2;
-            if(frames_per_buffer)
-                printf("Audio buffer size = %lu\n",frames_per_buffer);
-        }
-        sfdata.ifd          = ifd;
-        sfdata.inchans      = inchans;
-        sfdata.outchans     = outchans;
-        sfdata.frames_played  = 0;
-        sfdata.gain         = gain;
-        sfdata.copyfunc     = copyfunc;
-        sfdata.decodefunc   = decodefunc;
-        sfdata.do_decode    = flags[FLAG_B];
-        sfdata.from_frame   = from_frame;
-        if(to_frame > 0)
-            sfdata.to_frame = to_frame;
-        else {
-            sfdata.to_frame = 0;
-        }
-        sfdata.current_frame = 0;
-        sfdata.play_looped  = flags[FLAG_L];
-
-        sfdata.srate = props.srate;
-
-        sfdata.finished = 0;
-        g_pdata = &sfdata;
-
-#ifdef DEBUG
-        {
-            PSF_PROPS outprops = props;
-            outprops.chans = outchans;
-            sfdata.ofd = psf_sndCreate("debug.wav",&outprops,0,0,PSF_CREATE_RDWR);
-            if(sfdata.ofd == -1){
-                printf("Sorry - unable to create debug outfile\n");
-            }
-        }
-#endif
-
-        outputParameters.device = device;   /*Pa_GetDefaultOutputDevice(); */ /* default output device */
-        outputParameters.channelCount = outchans;
-        outputParameters.sampleFormat = paFloat32;             /* 32 bit floating point output */
-        outputParameters.suggestedLatency = Pa_GetDeviceInfo( outputParameters.device )->defaultLowOutputLatency;
-        outputParameters.hostApiSpecificStreamInfo = NULL;
-
-        devinfo = (PaDeviceInfo *) Pa_GetDeviceInfo(device);
-#if defined MAC || defined unix
-        if(devinfo){
-            printf("Using device %d: %s\n",device,devinfo->name);
-        }
-#endif
-
-#ifdef _WIN32
-
-#ifdef NOTDEF
-        if(devinfo){
-            if(apiinfo->type  == paMME ){
-                winmmeStreamInfo.size = sizeof(winmmeStreamInfo);
-                winmmeStreamInfo.hostApiType = paMME;
-                winmmeStreamInfo.version = 1;
-                winmmeStreamInfo.flags = paWinMmeUseChannelMask;
-                winmmeStreamInfo.channelMask = 0;
-                outputParameters.hostApiSpecificStreamInfo = &winmmeStreamInfo;
-# ifdef _DEBUG
-                printf("WIN DEBUG: WinMME device channel mask set to 0.\n");
-# endif
-            }
-        }
-#endif
-
-        if(devinfo) {
-            int apitype = devinfo->hostApi;
-            const PaHostApiInfo *apiinfo = Pa_GetHostApiInfo(apitype);
-            //  if(do_speakermask)
-            //      printf("Using file channel format %d\n",speakermask);
-            printf("Using device %d: %s ",device,devinfo->name);
-
-            if(apiinfo->type  == paDirectSound ){
-                printf("(DS)\n");
-                /* set this IF we are using Dsound device. */
-                directSoundStreamInfo.size = sizeof(PaWinDirectSoundStreamInfo);
-                directSoundStreamInfo.hostApiType = paDirectSound;
-                directSoundStreamInfo.version = 2;
-                directSoundStreamInfo.flags = paWinDirectSoundUseChannelMask;
-                directSoundStreamInfo.channelMask = speakermask;
-                outputParameters.hostApiSpecificStreamInfo = &directSoundStreamInfo;
-            }
-            else if(apiinfo->type == paASIO)
-                printf("(ASIO)\n");
-            // else
-            //    printf("API unknown!);
-            else
-                printf("\n");
-        }
-#endif
-
-        err = Pa_OpenStream(
-                            &stream,
-                            NULL,  /* No input */
-                            &outputParameters, /* As above. */
-                            props.srate,
-                            frames_per_buffer,
-                            paClipOff,      /* we won't output out of range samples so don't bother clipping them */
-                            playcallback,
-                            &sfdata );
-
-
-
-        if( err != paNoError ) {
-            printf("Unable to open output device for %u-channel file.\n",outchans);
-            goto error;
-        }
-        err =  Pa_SetStreamFinishedCallback( stream, finishedCallback );
-        if( err != paNoError ) {
-            printf("Unable to set finish callback\n");
-            goto error;
-        }
-        sfdata.stream = stream;
-
-        file_playing = 1; // need this for c/l test below!
-
-        if(waitkey){
-            printf("Press any key to start:\n");
-            while (!kbhit()){
-                if(!file_playing)        //check for instant CTRL-C
-                    goto error;
-            };
-#ifdef _WIN32
-            if(kbhit())
-                _getch();                        //prevent display of char
-#endif
-        }
-
-        // should this go in read thread func too?
-        if(from_frame > 0){
-            if(psf_sndSeek(sfdata.ifd,from_frame,PSF_SEEK_SET))     {
-                printf("Error setting start position\n");
-                goto error;
-            }
-            sfdata.current_frame = from_frame;
-        }
-
-        /* if small block, read it all into memory
-         * if doing channel map, do it here rather in callback; we must set inchans to outchans */
-        if(sfdata.memFramelength > 0){
-            if(flags[FLAG_M]) {
-                unsigned int i,j;
-                int k;
-
-                float **orderptr = NULL;
-                float val;
-                float *tempframebuf = malloc(sizeof(float) * sfdata.memFramelength * props.chans);
-                nzeros = 0;
-                if(tempframebuf==NULL){
-                    puts("no memory for audio data\n");
-                    goto error;
-                }
-                orderptr = malloc(outchans * sizeof(float*));
-                for(i=0;i < nchars;i++){
-                    unsigned int thischar = argstring[i];
-                    unsigned int chindex;
-                    if(thischar != '0' && (thischar < rootchar || thischar > maxchar)){
-                        printf("illegal character in order string: %c\n",thischar);
-                        goto error;
-                    }
-                    if(thischar =='0'){
-                        orderptr[i] = NULL;
-                        nzeros++;
-                    }
-                    else{
-                        if(thischar > max_inchar)
-                            max_inchar = thischar;
-                        chindex = thischar - rootchar;
-                        orderptr[i] = tempframebuf + chindex;
-                    }
-                }
-                if(nzeros==nchars)
-                    printf("Warning: -m order string is all zeros - a silent file will be made!\n");
-                else {
-                    /* count inclusively! */
-                    if(props.chans < (int)(max_inchar - rootchar + 1)){
-                        printf("File has %d channels; order string defines non-existent channels.\n",props.chans);
-                        printf("For this file, maximum character is %c\n",rootchar + props.chans -1);
-                        goto error;
-                    }
-                }
-                if(psf_sndReadFloatFrames(sfdata.ifd,tempframebuf,sfdata.memFramelength) != sfdata.memFramelength) {
-                    printf("Error reading soundfile into memory\n");
-                    goto error;
-                }
-
-
-                for(j=0;j < sfdata.memFramelength; j++) {
-                    for(k = 0; k < sfdata.outchans; k++) {
-                        // find requested input chan
-                        float *pchan = orderptr[k];
-                        if(pchan == NULL)
-                            val = 0.0f;
-                        else
-                            val = *(pchan + j * props.chans);
-                        // and write to stream memory buffer
-                        sfdata.membuf[j * sfdata.outchans + k] = val;
-                    }
-                }
-                // done with these now
-                free(tempframebuf);
-                free(orderptr);
-                sfdata.inchans = sfdata.outchans;
-            }
-            else {  // no fancy stuff, just read straight into mem buffer
-                if(psf_sndReadFloatFrames(sfdata.ifd,sfdata.membuf,sfdata.memFramelength)
-                   != sfdata.memFramelength) {
-                    printf("Error reading soundfile into memory\n");
-                    goto error;
-                }
-            }
-        }
-
-        // set up timer
-#ifdef unix
-        if(do_updatemessages) {
-            setitimer(ITIMER_REAL, &tout_val,0);
-            signal(SIGALRM,alarm_wakeup); /* set the Alarm signal capture */
-        }
-#endif
-
-#ifdef _WIN32
-        // not sure of the best position for this
-        //if(!CreateTimerQueueTimer(&sfdata.hTimer, hTimerQueue,
-        //    (WAITORTIMERCALLBACK) TimerCallback, &sfdata,250,250,0)) {
-        //    printf("failed to start timer (3).\n");
-        //    return 1;
-        // }
-#endif
-        /* Start the file reading thread */
-        sfdata.startTime = Pa_GetStreamTime(stream );
-
-        if(sfdata.memFramelength == 0){
-
-            err = startThread(&sfdata, threadFunctionReadFromRawFile);
-            if( err != paNoError ) goto error;
-        }
-        else {
-            sfdata.flag = 0;
-        }
-
-
-#ifdef _WIN32
-        if(do_updatemessages){
-            if(!CreateTimerQueueTimer(&sfdata.hTimer, hTimerQueue,
-                                      (WAITORTIMERCALLBACK) TimerCallback, &sfdata,200,200,0)) {
-                printf("failed to start timer (3).\n");
-                return 1;
-            }
-        }
-#endif
-        err = Pa_StartStream( stream );
-        if( err != paNoError )
-            goto error;
-
-        printf("Hit CTRL-C to stop.\n");
-        while(!sfdata.finished && file_playing){
-            // nothing to do!
-            Pa_Sleep(10);
-        }
-        // note to programmer: any bug in audio buffer arithmetic will likely cause crash here!
-        err = Pa_StopStream( stream );
-        if( err != paNoError ) {
-            printf("Error stopping stream\n");
-            goto error;
-        }
-
-        /*  read thread should exit always, no need to Stop it here */
-
-
-        err = Pa_CloseStream( stream );
-        if( err != paNoError ) {
-            printf("Error closing stream\n");
-            goto error;
-        }
-        printf("\r%.2f secs\n",(float)(sfdata.lastTime));
-        fflush(stdout);
-        printf("Playback finished.\n");
-    error:
-        Pa_Terminate();
-
-#ifdef _WIN32
-        CloseHandle(ghEvent);
-        DeleteTimerQueue(hTimerQueue);
-#endif
-        if( sfdata.ringbufData )
-            PaUtil_FreeMemory(sfdata.ringbufData);
-        if(sfdata.inbuf)
-            PaUtil_FreeMemory(sfdata.inbuf);
-        if(sfdata.membuf)
-            PaUtil_FreeMemory(sfdata.membuf);
-        if(sfdata.orderptr)
-            free(sfdata.orderptr);
-        if(ifd >=0)
-            psf_sndClose(ifd);
-        if(fpeaks)
-            free(fpeaks);
-#ifdef DEBUG
-        if(sfdata.ofd >=0)
-            psf_sndClose(sfdata.ofd);
-#endif
-        psf_finish();
-
-        return 0;
-    }
-
-    int show_devices(void)
-    {
-        PaDeviceIndex numDevices,p;
-        const    PaDeviceInfo *pdi;
-#ifdef _WIN32
-        const PaHostApiInfo* api_info;
-        const char *apiname;
-#endif
-        PaError  err;
-        int nOutputDevices = 0;
-
-#ifdef USE_ASIO
-        printf("For ASIO multi-channel, you may need to select the highest device no.\n");
-#endif
-        /*Pa_Initialize();*/
-        numDevices =  Pa_GetDeviceCount();
-        if( numDevices < 0 )
-            {
-                printf("ERROR: Pa_CountDevices returned 0x%x\n", numDevices );
-                err = numDevices;
-                return err;
-            }
-#ifdef WIN32
-        printf("Driver\tDevice\tInput\tOutput\tName\n");
-#else
-        printf("Device\tInput\tOutput\tName\n");
-#endif
-        //printf("Number of devices = %d\n", numDevices );
-        for( p = 0; p < numDevices; p++ )
-            {
-                pdi = Pa_GetDeviceInfo( p );
-
-                //#ifdef _WIN32
-                //                      /*RWD: skip, info on inputs */
-                //                      if(pdi->maxOutputChannels == 0)
-                //                              continue;
-                //#endif
-                nOutputDevices++;
-
-                if( p == Pa_GetDefaultOutputDevice() )
-                    printf("*");
-                else
-                    printf(" ");
-#ifdef WIN32
-                api_info = Pa_GetHostApiInfo(pdi->hostApi);
-                apiname = api_info->name;
-                if(strcmp(apiname,"Windows DirectSound")==0)
-                    apiname = "DS ";
-                printf("(%s)\t%d\t%d\t%d\t%s\n",apiname,p,
-                       pdi->maxInputChannels,
-                       pdi->maxOutputChannels,
-                       pdi->name);
-#else
-                printf("%d\t%d\t%d\t%s\n",p,
-                       pdi->maxInputChannels,
-                       pdi->maxOutputChannels,
-                       pdi->name);
-
-
-#endif
-            }
-        /*Pa_Terminate();*/
-        return 0;
-    }

dev/externals/paprogs/pvplay/.gdb_history

@@ -1,248 +0,0 @@
-break pvplay2.cpp:467
-run sh.ana
- p a  pv
- p /a  pv
-pv
- p /h  pv
- p /x  pv
-n
-print props.origrate
-print pv
-print anal_framesize
-print props.winlen
-print props.decfac
-print winlen
-c
-q
-break pvplay2.cpp:443
-run sh.ana
-q
-break pvplay2.cpp:443
-run sh.ana
-n
-n
-n
-print props.chans
-print props.origrate
-q
-break pvplay2.cpp:443
-run sh.ana
-ni
-ni
-ni
-s
-s
-s
-s
-s
-s
-s
-s
-s
-s
-s
-s
-s
-s
-s
-s
-s
-s
-s
-s
-s
-s
-s
-s
-s
-s
-s
-s
-s
-s
-s
-s
-s
-s
-s
-s
-s
-s
-s
-s
-s
-s
-s
-s
-s
-s
-s
-s
-s
-s
-s
-s
-s
-s
-s
-s
-s
-s
-s
-s
-s
-s
-s
-s
-s
-s
-s
-s
-s
-s
-s
-s
-s
-s
-s
-s
-s
-s
-s
-s
-s
-s
-s
-s
-s
-s
-s
-s
-s
-s
-s
-s
-s
-s
-s
-s
-s
-s
-s
-s
-s
-s
-s
-s
-s
-s
-s
-s
-s
-s
-s
-s
-s
-s
-s
-s
-s
-s
-s
-s
-s
-s
-s
-s
-s
-s
-s
-s
-s
-s
-s
-s
-s
-s
-s
-s
-s
-s
-s
-s
-s
-s
-s
-s
-s
-s
-s
-s
-s
-s
-s
-s
-s
-s
-s
-s
-s
-s
-s
-s
-s
-s
-s
-s
-n
-n
-n
-n
-n
-n
-n
-n
-n
-n
-n
-n
-n
-n
-n
-n
-n
-n
-n
-n
-n
-q
-break snd.c:448
-run sh.ana
-q
-break snd.c:448
-run sh.ana
-break pvplay2.cpp:443
-run sh.ana
-s
-n
-n
-n
-n
-n
-n
-n
-n
-n
-n
-n
-s
-break 367
-c
-n
-n
-p origsize
-print origsize
-print origrate
-n
-print origrate
-q

dev/externals/paprogs/pvplay/CMakeLists.txt

@@ -1,30 +0,0 @@
-if(APPLE)
-  set(CMAKE_C_FLAGS "-O2 -Wall -mmacosx-version-min=10.8 -Dunix  -fomit-frame-pointer -funroll-loops")
-  set(CMAKE_CXX_FLAGS "-O2 -Wall -mmacosx-version-min=10.8  -Dunix -fomit-frame-pointer -funroll-loops -std=c++11 -stdlib=libc++")
-  include_directories ( /Developer/Headers/FlatCarbon )
-  find_library(COREAUDIOLIB CoreAudio)
-  find_library(AUDIOTOOLBOX AudioToolbox)
-  find_library(AULIB AudioUnit)
-  find_library(CARBONLIB Carbon)
-  #RWD only need this for APPLE?
-  find_library(AAIOLIB names libaaio.a paths /usr/local/lib)
-  set(EXTRA_LIBRARIES1 pthread ${AAIOLIB} ${COREAUDIOLIB} ${AUDIOTOOLBOX} ${AULIB} ${CARBONLIB} ${EXTRA_LIBRARIES})
-else()
-  if(MINGW)
-    set(CMAKE_C_FLAGS "-O3 -DWIN32 -D_WIN32 -DUSE_ASIO -fomit-frame-pointer -funroll-loops")
-    set(CMAKE_CXX_FLAGS "-O3 -DWIN32 -D_WIN32 -DUSE_ASIO -fomit-frame-pointer -funroll-loops -static-libgcc -static-libstdc++")
-    set(EXTRA_LIBRARIES1 ${EXTRA_LIBRARIES})
-  else()
-    set(CMAKE_C_FLAGS "-O3 -Wall -Dlinux -Dunix -fomit-frame-pointer -funroll-loops")
-    set(EXTRA_LIBRARIES1 asound jack pthread ${EXTRA_LIBRARIES})
-  endif()
-endif()
-
-link_directories(../../lib ../portaudio/lib/.libs ../../../sfsys)
-
-include_directories(../../../include ../include ../portaudio/include ../portaudio/src/common /usr/local/include)
-
-add_executable(pvplay pvplay.cpp pvoc2.cpp fmhfuncs.c pvthreads.cpp pvfileio.c mxfft.c)
-target_link_libraries(pvplay portaudio.a sfsys ${EXTRA_LIBRARIES1})
-
-my_install(pvplay)

dev/externals/paprogs/pvplay/fmdcode.h

@@ -1,106 +0,0 @@
-/*
- * Copyright (c) 1983-2013 Richard Dobson and Composers Desktop Project Ltd
- * http://people.bath.ac.uk/masrwd
- * http://www.composersdesktop.com
- * This file is part of the CDP System.
- * The CDP System is free software; you can redistribute it
- * and/or modify it under the terms of the GNU Lesser General Public 
- * License as published by the Free Software Foundation; either 
- * version 2.1 of the License, or (at your option) any later version. 
- *
- * The CDP System is distributed in the hope that it will be useful, 
- * but WITHOUT ANY WARRANTY; without even the implied warranty of 
- * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  
- * See the GNU Lesser General Public License for more details.
- * You should have received a copy of the GNU Lesser General Public
- * License along with the CDP System; if not, write to the Free Software
- * Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
- *
- */
- 
-/* fmdcode.h FMH decode functions */
-
-/*
- Channel order is WXYZ,RSTUV,KLMNOPQ
- 
- The number of channels defines the order of the soundfield:
- 2 channel = UHJ 
- 3 channel = h   = 1st order horizontal
- 4 channel = f   = 1st order 3-D
- 5 channel = hh  = 2nd order horizontal
- 6 channel = fh  = 2nd order horizontal + 1st order height (formerly
-                                                            called 2.5 order)
- 7 channel = hhh = 3rd order horizontal
- 8 channel = fhh = 3rd order horizontal + 1st order height
- 9 channel = ff  = 2nd order 3-D
- 11 channel = ffh = 3rd order horizontal + 2nd order height
- 16 channel = fff = 3rd order 3-D
- 
- 
- Horizontal   Height  Soundfield   Number of    Channels
- order 	      order 	  type      channels 	
- 1 	         0 	       horizontal 	  3 	    WXY
- 1 	         1 	      full-sphere 	  4 	    WXYZ
- 2 	         0 	       horizontal 	  5 	    WXY....UV
- 2 	         1 	       mixed-order    6 	    WXYZ...UV
- 2 	         2 	      full-sphere     9 	    WXYZRSTUV
- 3           0 	       horizontal 	  7 	    WXY....UV.....PQ
- 3 	         1         mixed-order 	  8 	    WXYZ...UV.....PQ
- 3 	         2 	       mixed-order 	 11 	    WXYZRSTUV.....PQ
- 3 	         3 	      full-sphere 	 16 	    WXYZRSTUVKLMNOPQ
- */
-
-typedef struct abf_samp {
-	float W;
-	float X;
-	float Y;
-	float Z;
-    float R;
-    float S;
-    float T;
-	float U;
-    float V;
-} ABFSAMPLE;
-
-typedef void (*fmhcopyfunc)(ABFSAMPLE*,const float*);
-
-typedef void (*fmhdecodefunc)(const ABFSAMPLE*, float*,unsigned int);
-//void bfdcode4(float *inbuf,long numframes);
-void fmhcopy_3(ABFSAMPLE* abf,const float*buf);
-void fmhcopy_4(ABFSAMPLE* abf,const float*buf);
-void fmhcopy_5(ABFSAMPLE* abf,const float*buf);
-void fmhcopy_6(ABFSAMPLE* abf,const float*buf);
-void fmhcopy_7(ABFSAMPLE* abf,const float*buf);
-void fmhcopy_8(ABFSAMPLE* abf,const float*buf);
-void fmhcopy_9(ABFSAMPLE* abf,const float*buf);
-void fmhcopy_11(ABFSAMPLE* abf,const float*buf);
-void fmhcopy_16(ABFSAMPLE* abf,const float*buf);
-
-
-// i1 = inphase 1st order, etc
-void fm_i1_mono(const ABFSAMPLE *inbuf,float *outbuf,unsigned int numframes);
-void fm_i1_stereo(const ABFSAMPLE *inbuf,float *outbuf,unsigned int numframes);
-void fm_i1_square(const ABFSAMPLE *inbuf,float *outbuf,unsigned int numframes);
-void fm_i2_square(const ABFSAMPLE *inbuf,float *outbuf,unsigned int numframes);
-void fm_i1_quad(const ABFSAMPLE *inbuf,float *outbuf,unsigned int numframes);
-void fm_i2_quad(const ABFSAMPLE *inbuf,float *outbuf,unsigned int numframes);
-void fm_i1_pent(const ABFSAMPLE *inbuf,float *outbuf,unsigned int numframes);
-void fm_i2_pent(const ABFSAMPLE *inbuf,float *outbuf,unsigned int numframes);
-void fm_i1_surr(const ABFSAMPLE *inbuf,float *outbuf,unsigned int numframes);
-void fm_i2_surr(const ABFSAMPLE *inbuf,float*outbuf,unsigned int numframes);
-void fm_i1_surr6(const ABFSAMPLE *inbuf,float *outbuf,unsigned int numframes);
-void fm_i2_surr6(const ABFSAMPLE *inbuf,float*outbuf,unsigned int numframes);
-void dm_i1_surr(const ABFSAMPLE *inbuf,float*outbuf,unsigned int numframes);
-void dm_i1_surr6(const ABFSAMPLE *inbuf,float*outbuf,unsigned int numframes);
-void dm_i2_surr(const ABFSAMPLE *inbuf,float*outbuf,unsigned int numframes);
-void dm_i2_surr6(const ABFSAMPLE *inbuf,float*outbuf,unsigned int numframes);
-void fm_i1_hex(const ABFSAMPLE *inbuf,float *outbuf,unsigned int numframes);
-void fm_i2_hex(const ABFSAMPLE *inbuf,float *outbuf,unsigned int numframes);
-void fm_i1_oct1(const ABFSAMPLE *inbuf,float *outbuf,unsigned int numframes);
-void fm_i2_oct1(const ABFSAMPLE *inbuf,float *outbuf,unsigned int numframes);
-void fm_i1_oct2(const ABFSAMPLE *inbuf,float *outbuf,unsigned int numframes);
-void fm_i2_oct2(const ABFSAMPLE *inbuf,float *outbuf,unsigned int numframes);
-void fm_i1_cube(const ABFSAMPLE *inbuf,float *outbuf,unsigned int numframes);
-void fm_i2_cube(const ABFSAMPLE *inbuf,float *outbuf,unsigned int numframes);
-void fm_i1_cubex(const ABFSAMPLE *inbuf,float *outbuf,unsigned int numframes);
-void fm_i2_cubex(const ABFSAMPLE *inbuf,float *outbuf,unsigned int numframes);

dev/externals/paprogs/pvplay/fmhfuncs.c

@@ -1,667 +0,0 @@
-/*
- * Copyright (c) 1983-2013 Richard Dobson and Composers Desktop Project Ltd
- * http://people.bath.ac.uk/masrwd
- * http://www.composersdesktop.com
- * This file is part of the CDP System.
- * The CDP System is free software; you can redistribute it
- * and/or modify it under the terms of the GNU Lesser General Public 
- * License as published by the Free Software Foundation; either 
- * version 2.1 of the License, or (at your option) any later version. 
- *
- * The CDP System is distributed in the hope that it will be useful, 
- * but WITHOUT ANY WARRANTY; without even the implied warranty of 
- * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  
- * See the GNU Lesser General Public License for more details.
- * You should have received a copy of the GNU Lesser General Public
- * License along with the CDP System; if not, write to the Free Software
- * Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
- *
- */
- 
-/* fmhfuncs.c : FMH decode funcs */
-#include "fmdcode.h"
-
-/* TODO: expand to handle numframes frames? */
-
-void fmhcopy_3(ABFSAMPLE* abf,const float*buf)
-{
-    abf->W = *buf++;
-    abf->X = *buf++;
-    abf->Y = *buf;
-}
-
-void fmhcopy_4(ABFSAMPLE* abf,const float*buf)
-{
-    abf->W = *buf++;
-    abf->X = *buf++;
-    abf->Y = *buf++;
-    abf->Z = *buf;
-}
-
-void fmhcopy_5(ABFSAMPLE* abf,const float*buf)
-{
-    abf->W = *buf++;
-    abf->X = *buf++;
-    abf->Y = *buf++;
-    abf->U = *buf++;
-    abf->V = *buf;
-}
-
-void fmhcopy_6(ABFSAMPLE* abf,const float*buf)
-{
-    abf->W = *buf++;
-    abf->X = *buf++;
-    abf->Y = *buf++;
-    abf->Z = *buf++;
-    abf->U = *buf++;
-    abf->V = *buf;
-}
-// following discard 3rd order chans
-void fmhcopy_7(ABFSAMPLE* abf,const float*buf)
-{
-    abf->W = *buf++;
-    abf->X = *buf++;
-    abf->Y = *buf++;
-    abf->U = *buf++;
-    abf->V = *buf;
-}
-
-void fmhcopy_8(ABFSAMPLE* abf,const float*buf)
-{
-    abf->W = *buf++;
-    abf->X = *buf++;
-    abf->Y = *buf++;
-    abf->Z = *buf++;
-    abf->U = *buf++;
-    abf->V = *buf;
-}
-// these identical for 2nd order horiz max, but may be expanded later!
-void fmhcopy_9(ABFSAMPLE* abf,const float*buf)
-{
-    abf->W = *buf++;
-    abf->X = *buf++;
-    abf->Y = *buf++;
-    abf->Z = *buf++;
-    abf->R = *buf++;
-    abf->S = *buf++;
-    abf->T = *buf++;
-    abf->U = *buf++;
-    abf->V = *buf;
-}
-
-void fmhcopy_11(ABFSAMPLE* abf,const float*buf)
-{
-    abf->W = *buf++;
-    abf->X = *buf++;
-    abf->Y = *buf++;
-    abf->Z = *buf++;
-    abf->R = *buf++;
-    abf->S = *buf++;
-    abf->T = *buf++;
-    abf->U = *buf++;
-    abf->V = *buf;
-}
-void fmhcopy_16(ABFSAMPLE* abf,const float*buf)
-{
-    abf->W = *buf++;
-    abf->X = *buf++;
-    abf->Y = *buf++;
-    abf->Z = *buf++;
-    abf->R = *buf++;
-    abf->S = *buf++;
-    abf->T = *buf++;
-    abf->U = *buf++;
-    abf->V = *buf;
-}
-
-/********** DECODE FUNCS *************/
-/* TODO: complete support for numframes > 1 */
-
-void fm_i1_mono(const ABFSAMPLE *inbuf,float *outbuf,unsigned int numframes)
-{
-    unsigned int i;	
-	float *p_out = outbuf;
-	double aw;
-    for(i=0;i < numframes; i++){
-		aw = (double) inbuf->W * 0.7071;
-		*p_out++ = (float) aw;  		  
-	}
-}
-
-void fm_i1_stereo(const ABFSAMPLE *inbuf,float *outbuf,unsigned int numframes)
-{
-    unsigned int i;	
-	float *p_out = outbuf;
-	double aw,ay;
-    for(i=0;i < numframes; i++){
-		aw = (double) inbuf->W * 0.7071;
-		ay = (double) inbuf->Y * 0.5;		
-		
-		*p_out++ = (float) (aw +  ay);  
-		*p_out++ = (float) (aw  - ay);  
-	}
-}
-
-void fm_i1_square(const ABFSAMPLE *inbuf,float *outbuf,unsigned int numframes)
-{
-	unsigned int i;	
-	float *p_out = outbuf;
-	double aw,ax,ay;
-	
-	for(i=0;i < numframes; i++){
-		aw = (double) inbuf->W * 0.35355;
-		ax = (double) inbuf->X * 0.17677;
-		ay = (double) inbuf->Y * 0.17677;		
-		
-		*p_out++ = (float) (aw + ax + ay);  //FL
-		*p_out++ = (float) (aw - ax + ay);  //RL
-		*p_out++ = (float) (aw - ax - ay);  //RR
-		*p_out++ = (float) (aw + ax - ay);  //FR
-	}
-}
-void fm_i2_square(const ABFSAMPLE *inbuf,float*outbuf,unsigned int numframes)
-{
-	unsigned int i;		
-	float *p_out = outbuf;	
-	double aw,ax,ay,av;
-	
-	for(i=0;i < numframes; i++){
-		aw = (double) inbuf->W * 0.3536;
-		ax = (double) inbuf->X * 0.2434;
-		ay = (double) inbuf->Y * 0.2434;		
-		av = (double) inbuf->V * 0.0964;
-		*p_out++ = (float) (aw + ax + ay + av);  //FL
-		*p_out++ = (float) (aw - ax + ay - av ); //RL
-		*p_out++ = (float) (aw - ax - ay + av);  //RR
-		*p_out++ = (float) (aw + ax - ay - av);  //FR
-	}
-}
-/* ditto, RLRL layout for WAVEX */
-void fm_i1_quad(const ABFSAMPLE *inbuf,float *outbuf,unsigned int numframes)
-{
-	unsigned int i;	
-	float *p_out = outbuf;
-	double aw,ax,ay;
-	
-	for(i=0;i < numframes; i++){
-		aw = (double) inbuf->W * 0.35355;
-		ax = (double) inbuf->X * 0.17677;
-		ay = (double) inbuf->Y * 0.17677;		
-		
-		*p_out++ = (float) (aw + ax + ay);  //FL
-        *p_out++ = (float) (aw + ax - ay);  //FR
-		*p_out++ = (float) (aw - ax + ay);  //RL
-		*p_out = (float) (aw - ax - ay);  //RR
-		
-	}
-}
-void fm_i2_quad(const ABFSAMPLE *inbuf,float*outbuf,unsigned int numframes)
-{
-	unsigned int i;		
-	float *p_out = outbuf;	
-	double aw,ax,ay,av;
-	
-	for(i=0;i < numframes; i++){
-		aw = (double) inbuf->W * 0.3536;
-		ax = (double) inbuf->X * 0.2434;
-		ay = (double) inbuf->Y * 0.2434;		
-		av = (double) inbuf->V * 0.0964;
-		*p_out++ = (float) (aw + ax + ay + av);  //FL
-        *p_out++ = (float) (aw + ax - ay - av);  //FR
-		*p_out++ = (float) (aw - ax + ay - av ); //RL
-		*p_out = (float) (aw - ax - ay + av);  //RR
-		
-	}
-}
-
-
-//front pair angle 72deg
-void fm_i1_pent(const ABFSAMPLE *inbuf,float*outbuf,unsigned int numframes)
-{
-	unsigned int i;	
-	float *p_out = outbuf;
-	double aw,ax,ay;
-	
-	for(i=0;i < numframes; i++){
-		aw = (double) inbuf->W * 0.2828;
-		ax = (double) inbuf->X;
-		ay = (double) inbuf->Y;		
-		
-		*p_out++ = (float) (aw + (ax*0.1618) + (ay*0.1176));  //FL
-		*p_out++ = (float) (aw - (ax*0.0618) + (ay*0.1902));  
-		*p_out++ = (float) (aw - (ax*0.2));  
-		*p_out++ = (float) (aw - (ax*0.0618) - (ay*0.1902));  
-        *p_out++ = (float) (aw + (ax*0.1618) - (ay*0.1176)); //FR
-	}
-}
-
-void fm_i2_pent(const ABFSAMPLE *inbuf,float*outbuf,unsigned int numframes)
-{
-	unsigned int i;	
-	float *p_out = outbuf;	
-	double aw,ax,ay,au,av;
-	
-	for(i=0;i < numframes; i++){
-		aw = (double) inbuf->W * 0.2828;
-		ax = (double) inbuf->X;
-		ay = (double) inbuf->Y;	
-        au = (double) inbuf->U;
-        av = (double) inbuf->V;
-		
-		*p_out++ = (float) (aw + (ax*0.2227) + (ay*0.1618) + (au*0.0238) + (av * 0.0733));  
-		*p_out++ = (float) (aw - (ax*0.0851) + (ay*0.2619) - (au*0.0624) - (av * 0.0453));  
-		*p_out++ = (float) (aw - (ax*0.2753)               + (au * 0.0771)              );  
-		*p_out++ = (float) (aw - (ax*0.0851) - (ay*0.2619) - (au*0.0624) + (av * 0.0453));  
-        *p_out++ = (float) (aw + (ax*0.2227) - (ay*0.1618) + (au*0.0238) - (av * 0.0733));
-	}
-}
-
-/* FMH only defines 1st order decode */ 
-void fm_i1_surr(const ABFSAMPLE *inbuf,float*outbuf,unsigned int numframes)
-{
-	unsigned int i;	
-	float *p_out = outbuf;
-	double aw,ax,ay;
-	
-	for(i=0;i < numframes; i++){
-		aw = (double) inbuf->W;
-		ax = (double) inbuf->X;
-		ay = (double) inbuf->Y;		
-		/* TODO: fix this order! */
-		*p_out++ = (float) ((aw * 0.169)  + (ax*0.0797) + (ay * 0.0891));   //L
-		*p_out++ = (float) ((aw * 0.1635) + (ax*0.0923));                   //C    ///???
-		*p_out++ = (float) ((aw * 0.169)  - (ax*0.0797) - (ay * 0.0891));   //R    ///????
-		*p_out++ = (float) ((aw * 0.4563) - (ax*0.1259) + (ay * 0.1543));   //LS
-        *p_out++ = (float) ((aw * 0.4563) - (ax*0.1259) - (ay * 0.1543));   //RS
-	}
-}
-/* from Bruce Wiggins via Csound */
-void fm_i2_surr(const ABFSAMPLE *inbuf,float*outbuf,unsigned int numframes)
-{
-	unsigned int i;	
-	float *p_out = outbuf;
-	double aw,ax,ay,au,av;
-	
-	for(i=0;i < numframes; i++){
-		aw = (double) inbuf->W;
-		ax = (double) inbuf->X;
-		ay = (double) inbuf->Y;		
-		au = (double) inbuf->U;
-        av = (double) inbuf->V;
-        
-		*p_out++ = (float) ((aw * 0.405) + (ax*0.32) + (ay * 0.31)  + (au * 0.085) + (av * 0.125));   //L
-        *p_out++ = (float) ((aw * 0.405) + (ax*0.32) - (ay * 0.31)  + (au * 0.085) - (av * 0.125));   //R
-		*p_out++ = (float) ((aw * 0.085) + (ax*0.04)                + (au * 0.045)               );   //C
-		*p_out++ = (float) ((aw * 0.635) - (ax*0.335) + (ay * 0.28) - (au * 0.08)  + (av * 0.08));    //LS
-        *p_out++ = (float) ((aw * 0.635) - (ax*0.335) - (ay * 0.28) - (au * 0.08)  - (av * 0.08));    //RS
-	}
-}
-
-/* 5.1 versions - silent LFE */
-/* FMH only defines 1st order decode */ 
-void fm_i1_surr6(const ABFSAMPLE *inbuf,float*outbuf,unsigned int numframes)
-{
-	unsigned int i;	
-	float *p_out = outbuf;
-	double aw,ax,ay;
-	
-	for(i=0;i < numframes; i++){
-		aw = (double) inbuf->W;
-		ax = (double) inbuf->X;
-		ay = (double) inbuf->Y;		
-		
-		*p_out++ = (float) ((aw * 0.169)  + (ax*0.0797) + (ay * 0.0891));   //L
-		*p_out++ = (float) ((aw * 0.1635) + (ax*0.0923));                   //C
-		*p_out++ = (float) ((aw * 0.169)  - (ax*0.0797) - (ay * 0.0891));   //R
-        *p_out++ = 0.0f;                                                    //LFE
-		*p_out++ = (float) ((aw * 0.4563) - (ax*0.1259) + (ay * 0.1543));   //LS
-        *p_out++ = (float) ((aw * 0.4563) - (ax*0.1259) - (ay * 0.1543));   //RS
-	}
-}
-/* from Bruce Wiggins via Csound */
-void fm_i2_surr6(const ABFSAMPLE *inbuf,float*outbuf,unsigned int numframes)
-{
-	unsigned int i;	
-	float *p_out = outbuf;
-	double aw,ax,ay,au,av;
-	
-	for(i=0;i < numframes; i++){
-		aw = (double) inbuf->W;
-		ax = (double) inbuf->X;
-		ay = (double) inbuf->Y;		
-		au = (double) inbuf->U;
-        av = (double) inbuf->V;
-        
-		*p_out++ = (float) ((aw * 0.405) + (ax*0.32) + (ay * 0.31)  + (au * 0.085) + (av * 0.125));   //L
-        *p_out++ = (float) ((aw * 0.405) + (ax*0.32) - (ay * 0.31)  + (au * 0.085) - (av * 0.125));   //R
-		*p_out++ = (float) ((aw * 0.085) + (ax*0.04)                + (au * 0.045)               );   //C
-        *p_out++ = 0.0f;                                                                              //LFE
-		*p_out++ = (float) ((aw * 0.635) - (ax*0.335) + (ay * 0.28) - (au * 0.08)  + (av * 0.08));    //LS
-        *p_out++ = (float) ((aw * 0.635) - (ax*0.335) - (ay * 0.28) - (au * 0.08)  - (av * 0.08));    //RS
-	}
-}
-
-// 1st order 5.0
-void dm_i1_surr(const ABFSAMPLE *inbuf,float*outbuf,unsigned int numframes)
-{
-	unsigned int i;	
-	float *p_out = outbuf;
-	double aw,ax,ay;
-	
-	for(i=0;i < numframes; i++){
-		aw = (double) inbuf->W;
-		ax = (double) inbuf->X;
-		ay = (double) inbuf->Y;		
-		
-        
-		*p_out++ = (float) ((aw * 0.4597)  + (ax*0.4536) + (ay * 0.3591));   //L
-        *p_out++ = (float)  ((aw * 0.4597)  + (ax*0.4536) - (ay * 0.3591));  //R 
-		*p_out++ = 0.0f;                                                     //C
-		
-		*p_out++ = (float) ((aw * 0.5662) - (ax*0.3681) + (ay * 0.4606));    //LS
-        *p_out++ = (float) ((aw * 0.5662) - (ax*0.3681) - (ay * 0.4606));    //RS
-    }
-}
-//1st order 5.1
-void dm_i1_surr6(const ABFSAMPLE *inbuf,float*outbuf,unsigned int numframes)
-{
-	unsigned int i;	
-	float *p_out = outbuf;
-	double aw,ax,ay;
-	
-	for(i=0;i < numframes; i++){
-		aw = (double) inbuf->W;
-		ax = (double) inbuf->X;
-		ay = (double) inbuf->Y;		
-		
-        
-		*p_out++ = (float) ((aw * 0.4597)  + (ax*0.4536) + (ay * 0.3591));   //L
-        *p_out++ = (float)  ((aw * 0.4597)  + (ax*0.4536) - (ay * 0.3591));  //R 
-		*p_out++ = 0.0f;                                                     //C
-		*p_out++ = 0.0f;                                                     //LFE
-		*p_out++ = (float) ((aw * 0.5662) - (ax*0.3681) + (ay * 0.4606));    //LS
-        *p_out++ = (float) ((aw * 0.5662) - (ax*0.3681) - (ay * 0.4606));    //RS
-    }
-}
-// 2nd order 5.0
-void dm_i2_surr(const ABFSAMPLE *inbuf,float*outbuf,unsigned int numframes)
-{
-	unsigned int i;	
-	float *p_out = outbuf;
-	double aw,ax,ay,au,av;
-	
-	for(i=0;i < numframes; i++){
-		aw = (double) inbuf->W;
-		ax = (double) inbuf->X;
-		ay = (double) inbuf->Y;		
-		au = (double) inbuf->U;
-        av = (double) inbuf->V;
-        
-		*p_out++ = (float) ((aw * 0.3314)  + (ax*0.4097) + (ay * 0.3487) + (au * 0.0828) + (av*0.1489));  //L
-        *p_out++ = (float) ((aw * 0.3314)  + (ax*0.4097) - (ay * 0.3487) + (au * 0.0828) - (av*0.1489));  //R 
-		*p_out++ = (float) ((aw * 0.0804)  + (ax * 0.1327));                                              //C
-		*p_out++ = (float) ((aw * 0.6025) - (ax*0.3627) + (ay * 0.4089) - (au * 0.0567));                 //LS
-        *p_out++ = (float) ((aw * 0.6025) - (ax*0.3627) - (ay * 0.4089) - (au * 0.0567));                 //RS
-    }
-}
-// 2nd order 5.1
-void dm_i2_surr6(const ABFSAMPLE *inbuf,float*outbuf,unsigned int numframes)
-{
-	unsigned int i;	
-	float *p_out = outbuf;
-	double aw,ax,ay,au,av;
-	
-	for(i=0;i < numframes; i++){
-		aw = (double) inbuf->W;
-		ax = (double) inbuf->X;
-		ay = (double) inbuf->Y;		
-		au = (double) inbuf->U;
-        av = (double) inbuf->V;
-        
-		*p_out++ = (float) ((aw * 0.3314)  + (ax*0.4097) + (ay * 0.3487) + (au * 0.0828) + (av*0.1489));  //L
-        *p_out++ = (float) ((aw * 0.3314)  + (ax*0.4097) - (ay * 0.3487) + (au * 0.0828) - (av*0.1489));  //R 
-		*p_out++ = (float) ((aw * 0.0804)  + (ax * 0.1327));                                              //C
-		*p_out++ = 0.0f;                                                                                  //LFE
-		*p_out++ = (float) ((aw * 0.6025) - (ax*0.3627) + (ay * 0.4089) - (au * 0.0567));                 //LS
-        *p_out++ = (float) ((aw * 0.6025) - (ax*0.3627) - (ay * 0.4089) - (au * 0.0567));                 //RS
-    }
-}
-
-
-void fm_i1_hex(const ABFSAMPLE *inbuf,float*outbuf,unsigned int numframes)
-{
-	unsigned int i;	
-	float *p_out = outbuf;
-	double aw,ax,ay;
-	
-	for(i=0;i < numframes; i++){
-		aw = (double) inbuf->W * 0.2357;
-		ax = (double) inbuf->X * 0.1443;
-		ay = (double) inbuf->Y;		
-		
-		*p_out++ = (float) (aw + ax + (ay * 0.0833));  //FL
-		*p_out++ = (float) (aw      + (ay * 0.1667));  //SL
-		*p_out++ = (float) (aw - ax + (ay * 0.0833));  //RL
-		*p_out++ = (float) (aw - ax - (ay * 0.0833));  //RR
-        *p_out++ = (float) (aw      - (ay * 0.1667));  //SR
-        *p_out++ = (float) (aw + ax - (ay * 0.0833));  //FR
-	}
-}
-void fm_i2_hex(const ABFSAMPLE *inbuf,float*outbuf,unsigned int numframes)
-{
-	unsigned int i;	
-	float *p_out = outbuf;	
-	double aw,ax,ay,au,av;
-	
-	for(i=0;i < numframes; i++){
-		aw = (double) inbuf->W * 0.2357;
-		ax = (double) inbuf->X * 0.1987;
-		ay = (double) inbuf->Y;
-		au = (double) inbuf->U;
-        av = (double) inbuf->V * 0.0556;
-		
-		*p_out++ = (float) (aw + ax + (ay * 0.1147) + (au * 0.0321) + av);  //FL
-		*p_out++ = (float) (aw      + (ay * 0.2294) - (au * 0.0643)     );  //SL
-		*p_out++ = (float) (aw - ax + (ay * 0.1147) + (au * 0.0321) - av);  //RL
-		*p_out++ = (float) (aw - ax - (ay * 0.1147) + (au * 0.0321) + av);  //RR
-        *p_out++ = (float) (aw      - (ay * 0.2294) - (au * 0.0643)     );  //SR
-        *p_out++ = (float) (aw + ax - (ay * 0.1147) + (au * 0.0321) - av);  //FR
-	}
-}
-
-void fm_i1_oct1(const ABFSAMPLE *inbuf,float*outbuf,unsigned int numframes)
-{
-	unsigned int i;	
-	float *p_out = outbuf;
-	double aw,ax,ay;
-	
-	for(i=0;i < numframes; i++){
-		aw = (double) inbuf->W * 0.1768;
-		ax = (double) inbuf->X;
-		ay = (double) inbuf->Y;		
-		
-		*p_out++ = (float) (aw + (ax * 0.1155) + (ay * 0.0478));  
-		*p_out++ = (float) (aw + (ax * 0.0478) + (ay * 0.1155));  
-		*p_out++ = (float) (aw - (ax * 0.0478) + (ay * 0.1155));  
-		*p_out++ = (float) (aw - (ax * 0.1155) + (ay * 0.0478));  
-        *p_out++ = (float) (aw - (ax * 0.231)  - (ay * 0.0957));  
-        *p_out++ = (float) (aw - (ax * 0.0478) - (ay * 0.1155));  
-        *p_out++ = (float) (aw + (ax * 0.0478) - (ay * 0.1155));  
-        *p_out++   = (float) (aw + (ax * 0.1155) - (ay * 0.0478));  
-	}
-}
-void fm_i2_oct1(const ABFSAMPLE *inbuf,float *outbuf,unsigned int numframes)
-{
-	unsigned int i;	
-	float *p_out = outbuf;	
-	double aw,ax,ay,au,av;
-	
-	for(i=0;i < numframes; i++){
-		aw = (double) inbuf->W * 0.17677;
-		ax = (double) inbuf->X;
-		ay = (double) inbuf->Y;
-		au = (double) inbuf->U * 0.03417;
-		av = (double) inbuf->V * 0.03417;
-        
-		*p_out++ = (float) (aw + (ax * 0.15906) + (ay * 0.06588) + au + av);  
-		*p_out++ = (float) (aw + (ax * 0.06588) + (ay * 0.15906) - au + av);  
-		*p_out++ = (float) (aw - (ax * 0.06588) + (ay * 0.15906) - au - av);  
-		*p_out++ = (float) (aw - (ax * 0.15906) + (ay * 0.06588) + au - av);  
-        *p_out++ = (float) (aw - (ax * 0.15906) - (ay * 0.06588) + au + av);  
-        *p_out++ = (float) (aw - (ax * 0.06588) - (ay * 0.15906) - au + av);  
-        *p_out++ = (float) (aw + (ax * 0.06588) - (ay * 0.15906) - au - av);  
-        *p_out++ = (float) (aw + (ax * 0.15906) - (ay * 0.06588) + au - av);  
-	}
-}
-
-void fm_i1_oct2(const ABFSAMPLE *inbuf,float *outbuf,unsigned int numframes)
-{
-	unsigned int i;	
-	float *p_out = outbuf;
-	double aw,ax,ay;
-	
-	for(i=0;i < numframes; i++){
-		aw = (double) inbuf->W * 0.1768;
-		ax = (double) inbuf->X;
-		ay = (double) inbuf->Y;		
-		
-		*p_out++ = (float) (aw + (ax * 0.125)                 );  
-		*p_out++ = (float) (aw + (ax * 0.0884) + (ay * 0.0884));  
-		*p_out++ = (float) (aw                 + (ay * 0.125) );  
-		*p_out++ = (float) (aw - (ax * 0.0884) + (ay * 0.0884));  
-        *p_out++ = (float) (aw - (ax * 0.125)                 );  
-        *p_out++ = (float) (aw - (ax * 0.0884) - (ay * 0.0884));  
-        *p_out++ = (float) (aw                 - (ay * 0.125) );  
-        *p_out++ = (float) (aw + (ax * 0.0884) - (ay * 0.0884));  
-	}
-}
-void fm_i2_oct2(const ABFSAMPLE *inbuf,float *outbuf,unsigned int numframes)
-{
-	unsigned int i;	
-	float *p_out = outbuf;	
-	double aw,ax,ay,au,av;
-	
-	for(i=0;i < numframes; i++){
-		aw = (double) inbuf->W * 0.1768;
-		ax = (double) inbuf->X;
-		ay = (double) inbuf->Y;
-		au = (double) inbuf->U * 0.0482;
-		av = (double) inbuf->V * 0.0482;
-        
-		*p_out++ = (float) (aw + (ax * 0.1721)                 + au     );  
-		*p_out++ = (float) (aw + (ax * 0.1217) + (ay * 0.1217)      + av);  
-		*p_out++ = (float) (aw                 + (ay * 0.1721) - au     );  
-		*p_out++ = (float) (aw - (ax * 0.1217) + (ay * 0.1217)      - av);  
-        *p_out++ = (float) (aw - (ax * 0.1721)                 + au     );  
-        *p_out++ = (float) (aw - (ax * 0.1217) - (ay * 0.1217)      + av);  
-        *p_out++ = (float) (aw                 - (ay * 0.1721) - au     );  
-        *p_out++ = (float) (aw + (ax * 0.1217) - (ay * 0.1217)      - av);  
-	}
-}
-
-/* csound order; low/high anti-clockwise. 
-FMH page order, 4 low folowed by 4 high , clockwise! */
-void fm_i1_cube(const ABFSAMPLE *inbuf,float *outbuf,unsigned int numframes)
-{
-	unsigned int i;	
-	float *p_out = outbuf;
-	double aw,ax,ay,az;
-	
-	for(i=0;i < numframes; i++){
-		aw = (double) inbuf->W * 0.17677;
-		ax = (double) inbuf->X * 0.07216;
-		ay = (double) inbuf->Y * 0.07216;
-		az = (double) inbuf->Z * 0.07216;
-		
-		*p_out++ = (float) (aw + ax + ay - az);  // FL low
-		*p_out++ = (float) (aw + ax + ay + az);  // FL hi
-        
-		*p_out++ = (float) (aw - ax + ay - az);  // RL low
-		*p_out++ = (float) (aw - ax + ay + az);  //    hi
-        
-        *p_out++ = (float) (aw - ax - ay - az);  // RR low
-        *p_out++ = (float) (aw - ax - ay + az);  //   hi
-        
-        *p_out++ = (float) (aw + ax - ay - az);  // FR low
-        *p_out++ = (float) (aw + ax - ay + az);  //    hi
-	}
-}
-void fm_i2_cube(const ABFSAMPLE *inbuf,float *outbuf,unsigned int numframes)
-{
-	unsigned int i;	
-	float *p_out = outbuf;	
-	double aw,ax,ay,az,as,at,av;
-	
-	for(i=0;i < numframes; i++){
-		aw = (double) inbuf->W * 0.1768;
-		ax = (double) inbuf->X * 0.114;
-		ay = (double) inbuf->Y * 0.114;
-        az = (double) inbuf->Z * 0.114;
-        as = (double) inbuf->S * 0.0369;
-        at = (double) inbuf->T * 0.0369;
-		av = (double) inbuf->V * 0.0369;
-        
-		*p_out++ = (float) (aw + ax + ay - az - as - at + av); //FL low 
-		*p_out++ = (float) (aw + ax + ay + az + as + at + av); //   hi 
-		
-        *p_out++ = (float) (aw - ax + ay - az + as - at - av); //RL low
-		*p_out++ = (float) (aw - ax + ay + az - as + at - av);  
-        
-        *p_out++ = (float) (aw - ax - ay - az + as + at + av); // RR low
-        *p_out++ = (float) (aw - ax - ay + az - as - at + av);  
-        
-        *p_out++ = (float) (aw + ax - ay - az - as + at - av);  // FR low
-        *p_out++ = (float) (aw + ax - ay + az + as - at - av);   
-	}
-}
-/* ditto, wavex order */
-/* Front L, front R, Back L, Back R; top Front L, Top Fr R, Top Back L, Top back R */
-void fm_i1_cubex(const ABFSAMPLE *inbuf,float *outbuf,unsigned int numframes)
-{
-	unsigned int i;	
-	float *p_out = outbuf;
-	double aw,ax,ay,az;
-	
-	for(i=0;i < numframes; i++){
-		aw = (double) inbuf->W * 0.17677;
-		ax = (double) inbuf->X * 0.07216;
-		ay = (double) inbuf->Y * 0.07216;
-		az = (double) inbuf->Z * 0.07216;
-		
-		*p_out++ = (float) (aw + ax + ay - az);  // FL low
-        *p_out++ = (float) (aw + ax - ay - az);  // FR low
-        *p_out++ = (float) (aw - ax + ay - az);  // RL low
-        *p_out++ = (float) (aw - ax - ay - az);  // RR low
-        
-		*p_out++ = (float) (aw + ax + ay + az);  // FL hi
-        *p_out++ = (float) (aw + ax - ay + az);  // FR hi
-		*p_out++ = (float) (aw - ax + ay + az);  // RL hi
-        *p_out++ = (float) (aw - ax - ay + az);  // RR hi
-	}
-}
-void fm_i2_cubex(const ABFSAMPLE *inbuf,float *outbuf,unsigned int numframes)
-{
-	unsigned int i;	
-	float *p_out = outbuf;	
-	double aw,ax,ay,az,as,at,av;
-	
-	for(i=0;i < numframes; i++){
-		aw = (double) inbuf->W * 0.1768;
-		ax = (double) inbuf->X * 0.114;
-		ay = (double) inbuf->Y * 0.114;
-        az = (double) inbuf->Z * 0.114;
-        as = (double) inbuf->S * 0.0369;
-        at = (double) inbuf->T * 0.0369;
-		av = (double) inbuf->V * 0.0369;
-        
-		*p_out++ = (float) (aw + ax + ay - az - as - at + av);  // FL low
-        *p_out++ = (float) (aw + ax - ay - az - as + at - av);  // FR low
-        *p_out++ = (float) (aw - ax + ay - az + as - at - av);  // RL low
-        *p_out++ = (float) (aw - ax - ay - az + as + at + av);  // RR low
-		
-        *p_out++ = (float) (aw + ax + ay + az + as + at + av);  // FL  hi 
-		*p_out++ = (float) (aw + ax - ay + az + as - at - av);  // FR  hi
-		*p_out++ = (float) (aw - ax + ay + az - as + at - av);  // RL  hi 
-        *p_out++ = (float) (aw - ax - ay + az - as - at + av);  // RR  hi 
-	}
-}

dev/externals/paprogs/pvplay/mxfft.c

@@ -1,831 +0,0 @@
-/*
- * Copyright (c) 1983-2013 Composers Desktop Project Ltd
- * http://www.composersdesktop.com
- * This file is part of the CDP System.
- * The CDP System is free software; you can redistribute it
- * and/or modify it under the terms of the GNU Lesser General Public 
- * License as published by the Free Software Foundation; either 
- * version 2.1 of the License, or (at your option) any later version. 
- *
- * The CDP System is distributed in the hope that it will be useful, 
- * but WITHOUT ANY WARRANTY; without even the implied warranty of 
- * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  
- * See the GNU Lesser General Public License for more details.
- * You should have received a copy of the GNU Lesser General Public
- * License along with the CDP System; if not, write to the Free Software
- * Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
- *
- */
- 
-/* This program converted from the FORTRAN routines by Singleton in
- * Section 1.4 of  "Programs for Digital Signal Processing", IEEE Press, 1979.
- *  Conversion by Trevor Wishart and Keith Henderson, York Univ.
- */
-
-static char *rcsid = "$Id: mxfft.c%v 3.4 1994/10/31 17:37:28 martin Exp $";
-/*
- *	$Log: mxfft.c%v $
- * Revision 3.4  1994/10/31  17:37:28  martin
- * Starting with rcs
- *
- */
-#include <stdio.h>
-#include <stdlib.h>
-#include <math.h>
-#include "pvoc.h"
-
-void fft_(),fftmx(),reals_();
-
-/*
- *-----------------------------------------------------------------------
- * subroutine:  fft
- * multivariate complex fourier transform, computed in place
- * using mixed-radix fast fourier transform algorithm.
- *-----------------------------------------------------------------------
- *
- *	this is the call from C:
- *		fft_(anal,banal,&one,&N2,&one,&mtwo);
- *	CHANGED TO:-
- *		fft_(anal,banal,one,N2,one,mtwo);
- */
-
-void
-fft_(a, b, nseg, n, nspn, isn)
-float 	*a,	/*  pointer to array 'anal'  */
-	*b;	/*  pointer to array 'banal' */
-
-int	nseg,
-	n,
-	nspn,
-	isn;
-      
-{	int nfac[16];		/*  These are one bigger than needed   */
-				/*  because wish to use Fortran array  */
-				/* index which runs 1 to n, not 0 to n */
-
-	int 	m = 0,
-		nf,
-		k,
-		kt,
-		ntot,
-		j, 
-		jj,
-		maxf, maxp;
-
-/* work space pointers */
-	float	*at, *ck, *bt, *sk;
-	int	*np;
-
-
-/* reduce the pointers to input arrays - by doing this, FFT uses FORTRAN
-   indexing but retains compatibility with C arrays */
-	a--;	b--;
-
-/*	
- * determine the factors of n
- */
-	k=nf=abs(n);
-      	if(nf==1) 
-		return;
-
-	nspn=abs(nf*nspn);
-	ntot=abs(nspn*nseg);
-
-	if(isn*ntot == 0){
-		fprintf(stderr,"\nerror - zero in fft parameters %d %d %d %d",
-			nseg, n, nspn, isn);
-		return;
-	}
-	for (m=0; !(k%16); nfac[++m]=4,k/=16);
-	for (j=3,jj=9; jj<=k; j+=2,jj=j*j)
-		for (; !(k%jj); nfac[++m]=j,k/=jj);
-
-      	if (k<=4){
-	      	kt = m;
-	      	nfac[m+1] = k;
-	      	if(k != 1) 
-			m++;
-	}
-	else{
-		if(k%4==0){
-		nfac[++m]=2;
-		k/=4;
-		}
-
-	 	kt = m;
-	       	maxp = max((kt+kt+2),(k-1));
-		for(j=2; j<=k; j=1+((j+1)/2)*2)
-			if(k%j==0){
-				nfac[++m]=j;
-				k/=j;
-			}
-	}
-	if(m <= kt+1) 
-		maxp = m + kt + 1;
-      	if(m+kt > 15) {
-		fprintf(stderr,
-		"\nerror - fft parameter n has more than 15 factors : %d", n);
-	    	return;
-	}
-      	if(kt!=0){
-	      	j = kt;
-		while(j)
-			nfac[++m]=nfac[j--];
-	}
-	maxf = nfac[m-kt];
-      	if(kt > 0) 
-		maxf = max(nfac[kt],maxf);
-
-/*  allocate workspace - assume no errors! */
-	at = (float *) calloc(maxf,sizeof(float));
-	ck = (float *) calloc(maxf,sizeof(float));
-	bt = (float *) calloc(maxf,sizeof(float));
-	sk = (float *) calloc(maxf,sizeof(float));
-	np = (int *) calloc(maxp,sizeof(int));
-
-/* decrement pointers to allow FORTRAN type usage in fftmx */
-	at--;	bt--;	ck--;	sk--;	np--;
-
-/* call fft driver */
-
-	fftmx(a,b,ntot,nf,nspn,isn,m,&kt,at,ck,bt,sk,np,nfac);
-
-/* restore pointers before releasing */
-	at++;	bt++;	ck++;	sk++;	np++;
-
-/* release working storage before returning - assume no problems */
-	(void) free(at);
-	(void) free(sk);
-	(void) free(bt);
-	(void) free(ck);
-	(void) free(np);
-      	return;
-}
-
-/*
- *-----------------------------------------------------------------------
- * subroutine:  fftmx
- * called by subroutine 'fft' to compute mixed-radix fourier transform
- *-----------------------------------------------------------------------
- */
-void
-fftmx(a,b,ntot,n,nspan,isn,m,kt,at,ck,bt,sk,np,nfac)
-float	*a,*b,*at,*ck,*bt,*sk;
-int	*np;
-int ntot, n, nspan, isn, m;
-int *kt;
-int nfac[];
-{	
-	int	i,inc,
-		j,jc,jf, jj,
-		k, k1, k2, k3, k4,
-		kk,klim,ks,kspan, kspnn,
-		lim,
-		maxf,mm,
-		nn,nt;
-	double  aa, aj, ajm, ajp, ak, akm, akp,
-		bb, bj, bjm, bjp, bk, bkm, bkp,
-		c1, c2, c3, c72, cd,
-		dr,
-		rad, 
-		sd, s1, s2, s3, s72, s120;
-
-	double	xx;	/****** ADDED APRIL 1991 *********/
-	inc=abs(isn);
-	nt = inc*ntot;
-      	ks = inc*nspan;
-/******************* REPLACED MARCH 29: ***********************
-					rad = atan((double)1.0);
-**************************************************************/
-	rad = 0.785398163397448278900;
-/******************* REPLACED MARCH 29: ***********************
-			      		s72 = rad/0.625;
-			      		c72 = cos(s72);
-				      	s72 = sin(s72);
-**************************************************************/
-	c72 = 0.309016994374947451270;
-	s72 = 0.951056516295153531190;
-/******************* REPLACED MARCH 29: ***********************
-			      		s120 = sqrt((double)0.75);
-**************************************************************/
-      	s120 = 0.866025403784438707600;
-
-/* scale by 1/n for isn > 0 ( reverse transform ) */
-
-      	if (isn < 0){
-	      	s72 = -s72;
-	      	s120 = -s120;
-	      	rad = -rad;}
-	else{	ak = 1.0/(double)n;
-		for(j=1; j<=nt;j += inc){
-	        	a[j] *= ak;
-	        	b[j] *= ak;
-		}
-	}
-	kspan = ks;
-      	nn = nt - inc;
-      	jc = ks/n;
-
-/* sin, cos values are re-initialized each lim steps  */
-
-      	lim = 32;
-      	klim = lim * jc;
-      	i = 0;
-      	jf = 0;
-      	maxf = m - (*kt);
-      	maxf = nfac[maxf];
-      	if((*kt) > 0) 
-		maxf = max(nfac[*kt],maxf);
-
-/*
- * compute fourier transform
- */
-
-lbl40:
-	dr = (8.0 * (double)jc)/((double)kspan);
-/*************************** APRIL 1991 POW & POW2 not WORKING.. REPLACE *******
-	  	    cd = 2.0 * (pow2 ( sin((double)0.5 * dr * rad)) );
-*******************************************************************************/
-	xx =  sin((double)0.5 * dr * rad);
-      	cd = 2.0 * xx * xx;
-      	sd = sin(dr * rad);
-      	kk = 1;
-      	if(nfac[++i]!=2) goto lbl110;
-/*
- * transform for factor of 2 (including rotation factor)
- */
-      	kspan /= 2;
-      	k1 = kspan + 2;
-lbl50:	do{	do{	k2 = kk + kspan;
-		      	ak = a[k2];
-		      	bk = b[k2];
-		      	a[k2] = (a[kk]) - ak;
-		      	b[k2] = (b[kk]) - bk;
-		      	a[kk] = (a[kk]) + ak;
-		      	b[kk] = (b[kk]) + bk;
-		      	kk = k2 + kspan;
-		} while(kk <= nn);
-	      	kk -= nn;
-	}while(kk <= jc);
-      	if(kk > kspan) goto lbl350;
-lbl60:  c1 = 1.0 - cd;
-      	s1 = sd;
-      	mm = min((k1/2),klim);
-      	goto lbl80;
-lbl70:	ak = c1 - ((cd*c1)+(sd*s1));
-      	s1 = ((sd*c1)-(cd*s1)) + s1;
-      	c1 = ak;
-lbl80:	do{	do{	k2 = kk + kspan;
-		      	ak = a[kk] - a[k2];
-		      	bk = b[kk] - b[k2];
-		      	a[kk] = a[kk] + a[k2];
-		      	b[kk] = b[kk] + b[k2];
-		      	a[k2] = (c1 * ak) - (s1 * bk);
-		      	b[k2] = (s1 * ak) + (c1 * bk);
-		      	kk = k2 + kspan;
-		}while(kk < nt);
-	      	k2 = kk - nt;
-	      	c1 = -c1;
-	      	kk = k1 - k2;
-	}while(kk > k2);
-      	kk += jc;
-      	if(kk <= mm) goto lbl70;
-      	if(kk < k2)  goto lbl90;
-      	k1 += (inc + inc);
-      	kk = ((k1-kspan)/2) + jc;
-      	if(kk <= (jc+jc)) goto lbl60;
-      	goto lbl40;
-lbl90: 	s1 = ((double)((kk-1)/jc)) * dr * rad;
-      	c1 = cos(s1);
-      	s1 = sin(s1);
-      	mm = min( k1/2, mm+klim);
-      	goto lbl80;
-/*
- * transform for factor of 3 (optional code)
- */
-
-
-lbl100:	k1 = kk + kspan;
-	k2 = k1 + kspan;
-	ak = a[kk];
-	bk = b[kk];
-      	aj = a[k1] + a[k2];
-      	bj = b[k1] + b[k2];
-      	a[kk] = ak + aj;
-      	b[kk] = bk + bj;
-      	ak += (-0.5 * aj);
-      	bk += (-0.5 * bj);
-      	aj = (a[k1] - a[k2]) * s120;
-      	bj = (b[k1] - b[k2]) * s120;
-      	a[k1] = ak - bj;
-      	b[k1] = bk + aj;
-      	a[k2] = ak + bj;
-      	b[k2] = bk - aj;
-      	kk = k2 + kspan;
-      	if(kk < nn)     goto lbl100;
-      	kk -= nn;
-      	if(kk <= kspan) goto lbl100;
-      	goto lbl290;
-
-/*
- * transform for factor of 4
- */
-
-lbl110:	if(nfac[i] != 4) goto lbl230;
-      	kspnn = kspan;
-      	kspan = kspan/4;
-lbl120:	c1 = 1.0;
-      	s1 = 0;
-      	mm = min( kspan, klim);
-      	goto lbl150;
-lbl130:	c2 = c1 - ((cd*c1)+(sd*s1));
-      	s1 = ((sd*c1)-(cd*s1)) + s1;
-/*
- * the following three statements compensate for truncation
- * error.  if rounded arithmetic is used, substitute
- * c1=c2
- *
- * c1 = (0.5/(pow2(c2)+pow2(s1))) + 0.5;
- * s1 = c1*s1;
- * c1 = c1*c2;
- */
-      	c1 = c2;
-lbl140:	c2 = (c1 * c1) - (s1 * s1);
-      	s2 = c1 * s1 * 2.0;
-      	c3 = (c2 * c1) - (s2 * s1);
-      	s3 = (c2 * s1) + (s2 * c1);
-lbl150:	k1 = kk + kspan;
-      	k2 = k1 + kspan;
-      	k3 = k2 + kspan;
-      	akp = a[kk] + a[k2];
-      	akm = a[kk] - a[k2];
-      	ajp = a[k1] + a[k3];
-      	ajm = a[k1] - a[k3];
-      	a[kk] = akp + ajp;
-       	ajp = akp - ajp;
-      	bkp = b[kk] + b[k2];
-      	bkm = b[kk] - b[k2];
-      	bjp = b[k1] + b[k3];
-      	bjm = b[k1] - b[k3];
-      	b[kk] = bkp + bjp;
-      	bjp = bkp - bjp;
-      	if(isn < 0) goto lbl180;
-      	akp = akm - bjm;
-      	akm = akm + bjm;
-      	bkp = bkm + ajm;
-      	bkm = bkm - ajm;
-      	if(s1 == 0.0) goto lbl190;
-lbl160:	a[k1] = (akp*c1) - (bkp*s1);
-      	b[k1] = (akp*s1) + (bkp*c1);
-      	a[k2] = (ajp*c2) - (bjp*s2);
-      	b[k2] = (ajp*s2) + (bjp*c2);
-      	a[k3] = (akm*c3) - (bkm*s3);
-      	b[k3] = (akm*s3) + (bkm*c3);
-      	kk = k3 + kspan;
-      	if(kk <= nt)   goto lbl150;
-lbl170: kk -= (nt - jc);
-      	if(kk <= mm)   goto lbl130;
-      	if(kk < kspan) goto lbl200;
-      	kk -= (kspan - inc);
-      	if(kk <= jc)   goto lbl120;
-      	if(kspan==jc)  goto lbl350;
-      	goto lbl40;
-lbl180:	akp = akm + bjm;
-      	akm = akm - bjm;
-      	bkp = bkm - ajm;
-      	bkm = bkm + ajm;
-      	if(s1 != 0.0)  goto lbl160;
-lbl190:	a[k1] = akp;
-      	b[k1] = bkp;
-      	a[k2] = ajp;
-      	b[k2] = bjp;
-      	a[k3] = akm;
-      	b[k3] = bkm;
-      	kk = k3 + kspan;
-      	if(kk <= nt) goto lbl150;
-      	goto lbl170;
-lbl200: s1 = ((double)((kk-1)/jc)) * dr * rad;
-      	c1 = cos(s1);
-      	s1 = sin(s1);
-      	mm = min( kspan, mm+klim);
-      	goto lbl140;
-
-/*
- * transform for factor of 5 (optional code)
- */
-
-lbl210:	c2 = (c72*c72) - (s72*s72);
-      	s2 = 2.0 * c72 * s72;
-lbl220:	k1 = kk + kspan;
-      	k2 = k1 + kspan;
-      	k3 = k2 + kspan;
-      	k4 = k3 + kspan;
-      	akp = a[k1] + a[k4];
-      	akm = a[k1] - a[k4];
-      	bkp = b[k1] + b[k4];
-      	bkm = b[k1] - b[k4];
-      	ajp = a[k2] + a[k3];
-      	ajm = a[k2] - a[k3];
-      	bjp = b[k2] + b[k3];
-      	bjm = b[k2] - b[k3];
-      	aa = a[kk];
-      	bb = b[kk];
-      	a[kk] = aa + akp + ajp;
-      	b[kk] = bb + bkp + bjp;
-      	ak = (akp*c72) + (ajp*c2) + aa;
-      	bk = (bkp*c72) + (bjp*c2) + bb;
-      	aj = (akm*s72) + (ajm*s2);
-      	bj = (bkm*s72) + (bjm*s2);
-      	a[k1] = ak - bj;
-      	a[k4] = ak + bj;
-      	b[k1] = bk + aj;
-      	b[k4] = bk - aj;
-      	ak = (akp*c2) + (ajp*c72) + aa;
-      	bk = (bkp*c2) + (bjp*c72) + bb;
-      	aj = (akm*s2) - (ajm*s72);
-	bj = (bkm*s2) - (bjm*s72);
-	a[k2] = ak - bj;
-      	a[k3] = ak + bj;
-      	b[k2] = bk + aj;
-      	b[k3] = bk - aj;
-      	kk = k4 + kspan;
-      	if(kk < nn)     goto lbl220;
-      	kk -= nn;
-      	if(kk <= kspan) goto lbl220;
-      	goto lbl290;
-
-/*
- * transform for odd factors
- */
-
-lbl230:	k = nfac[i];
-	kspnn = kspan;
-	kspan /= k;
-	if(k==3)   goto lbl100;
-	if(k==5)   goto lbl210;
-	if(k==jf)  goto lbl250;
-      	jf = k;
-      	s1 = rad/(((double)(k))/8.0);
-      	c1 = cos(s1);
-      	s1 = sin(s1);
-      	ck[jf] = 1.0;
-	sk[jf] = 0.0;
-	for(j=1; j<k ; j++){
-		ck[j] = (ck[k])*c1 + (sk[k])*s1;
-	      	sk[j] = (ck[k])*s1 - (sk[k])*c1;
-	      	k--;
-	      	ck[k] = ck[j];
-	      	sk[k] = -(sk[j]);
-	}
-lbl250:	k1 = kk;
-      	k2 = kk + kspnn;
-	aa = a[kk];
-	bb = b[kk];
-      	ak = aa;
-      	bk = bb;
-      	j = 1;
-      	k1 += kspan;
-lbl260:	do{	k2 -= kspan;
-	      	j++;
-	      	at[j] = a[k1] + a[k2];
-	      	ak = at[j] + ak;	
-	      	bt[j] = b[k1] + b[k2];
-	      	bk = bt[j] + bk;	
-	      	j++;
-	      	at[j] = a[k1] - a[k2];
-	      	bt[j] = b[k1] - b[k2];
-	      	k1 += kspan;
-	}while(k1 < k2);
-      	a[kk] = ak;
-      	b[kk] = bk;
-      	k1 = kk;
-      	k2 = kk + kspnn;
-      	j = 1;
-lbl270:	k1 += kspan;
-      	k2 -= kspan;
-      	jj = j;
-      	ak = aa;
-      	bk = bb;
-      	aj = 0.0;
-      	bj = 0.0;
-      	k = 1;
-lbl280:	do{	k++;
-	      	ak = (at[k] * ck[jj]) + ak;
-	      	bk = (bt[k] * ck[jj]) + bk;	
-	      	k++;
-	      	aj = (at[k] * sk[jj]) + aj;
-	      	bj = (bt[k] * sk[jj]) + bj;
-	      	jj += j;
-	      	if (jj > jf) 
-			jj -= jf;
-	}while(k < jf);
-      	k = jf - j;
-      	a[k1] = ak - bj;
-      	b[k1] = bk + aj;
-      	a[k2] = ak + bj;
-      	b[k2] = bk - aj;
-      	j++;
-      	if(j < k)     goto lbl270;
-      	kk += kspnn;
-      	if(kk <= nn)  goto lbl250;
-      	kk -= nn;
-      	if(kk<=kspan) goto lbl250;
-
-/*
- * multiply by rotation factor (except for factors of 2 and 4)
- */
-
-lbl290:	if(i==m) goto lbl350;
-      	kk = jc + 1;
-lbl300:	c2 = 1.0 - cd;
-      	s1 = sd;
-      	mm = min( kspan, klim);
-      	goto lbl320;
-lbl310:	c2 = c1 - ((cd*c1) + (sd*s1));
-      	s1 = s1 + ((sd*c1) - (cd*s1));
-lbl320:	c1 = c2;
-      	s2 = s1;
-      	kk += kspan;
-lbl330:	ak = a[kk];
-      	a[kk] = (c2*ak) - (s2 * b[kk]);
-      	b[kk] = (s2*ak) + (c2 * b[kk]);
-      	kk += kspnn;
-      	if(kk <= nt) goto lbl330;
-      	ak = s1*s2;
-      	s2 = (s1*c2) + (c1*s2);
-      	c2 = (c1*c2) - ak;
-      	kk -= (nt - kspan);
-     	if(kk <= kspnn) goto lbl330;
-      	kk -= (kspnn - jc);
-      	if(kk <= mm)   goto lbl310;
-      	if(kk < kspan) goto lbl340;
-      	kk -= (kspan - jc - inc);
-      	if(kk <= (jc+jc)) goto lbl300;
-      	goto lbl40;
-lbl340:	s1 = ((double)((kk-1)/jc)) * dr * rad;
-      	c2 = cos(s1);
-      	s1 = sin(s1);
-     	mm = min( kspan, mm+klim);
-      	goto lbl320;
-
-/*
- * permute the results to normal order---done in two stages
- * permutation for square factors of n
- */
-
-lbl350:	np[1] = ks;
-      	if (!(*kt)) goto lbl440;
-      	k = *kt + *kt + 1;
-      	if(m < k) 
-		k--;
-	np[k+1] = jc;
-lbl360:	for(j=1; j < k; j++,k--){
-		np[j+1] = np[j] / nfac[j];
-	      	np[k] = np[k+1] * nfac[j];
-	}
-      	k3 = np[k+1];
-      	kspan = np[2];
-      	kk = jc + 1;
-      	k2 = kspan + 1;
-      	j = 1;
-      	if(n != ntot) goto lbl400;
-/*
- * permutation for single-variate transform (optional code)
- */
-lbl370:	do{	ak = a[kk];
-	      	a[kk] = a[k2];
-	      	a[k2] = ak;
-	      	bk = b[kk];
-	      	b[kk] = b[k2];
-	      	b[k2] = bk;
-	      	kk += inc;
-	      	k2 += kspan;
-	}while(k2 < ks);
-lbl380:	do{	k2 -= np[j++];
-	      	k2 += np[j+1];
-	}while(k2 > np[j]);
-      	j = 1;
-lbl390:	if(kk < k2){
-		goto lbl370;
-	}
-      	kk += inc;
-      	k2 += kspan;
-      	if(k2 < ks) goto lbl390;
-      	if(kk < ks) goto lbl380;
-      	jc = k3;
-      	goto lbl440;
-/*
- * permutation for multivariate transform
- */
-lbl400:	do{	do{	k = kk + jc;
-lbl410:			do{	ak = a[kk];
-			      	a[kk] = a[k2];
-			      	a[k2] = ak;
-			      	bk = b[kk];
-			      	b[kk] = b[k2];
-			      	b[k2] = bk;
-			      	kk += inc;
-			      	k2 += inc;
-			}while(kk < k);
-		      	kk += (ks - jc);
-		      	k2 += (ks - jc);
-		}while(kk < nt);
-	      	k2 -= (nt - kspan);
-	      	kk -= (nt - jc);
-	}while(k2 < ks);
-lbl420:	do{	k2 -= np[j++];
-	      	k2 += np[j+1];
-	}while(k2 > np[j]);
-      	j = 1;
-lbl430:	if(kk < k2) 	 goto lbl400;
-      	kk += jc;
-      	k2 += kspan;
-      	if(k2 < ks)      goto lbl430;
-      	if(kk < ks)      goto lbl420;
-      	jc = k3;
-lbl440:	if((2*(*kt))+1 >= m)
-		return;
-
-      	kspnn = *(np + *(kt) + 1);
-      	j = m - *kt;		
-      	nfac[j+1] = 1;
-lbl450:	nfac[j] = nfac[j] * nfac[j+1];
-      	j--;
-      	if(j != *kt) goto lbl450;
-      	*kt = *(kt) + 1;
-      	nn = nfac[*kt] - 1;
-      	jj = 0;
-      	j = 0;
-      	goto lbl480;
-lbl460:	jj -= k2;
-      	k2 = kk;
-      	kk = nfac[++k];
-lbl470:	jj += kk;
-      	if(jj >= k2) goto lbl460;
-      	np[j] = jj;
-lbl480:	k2 = nfac[*kt];
-      	k = *kt + 1;	
-      	kk = nfac[k];
-      	j++;
-      	if(j <= nn) goto lbl470;
-/* Determine permutation cycles of length greater than 1 */
-      	j = 0;
-      	goto lbl500;
-lbl490:	k = kk;
-     	kk = np[k];	
-      	np[k] = -kk;	
-      	if(kk != j) goto lbl490;
-      	k3 = kk;
-lbl500:	kk = np[++j];	
-      	if(kk < 0)  goto lbl500;
-      	if(kk != j) goto lbl490;
-      	np[j] = -j;
-      	if(j != nn) goto lbl500;
-      	maxf *= inc;
-/* Perform reordering following permutation cycles */
-      	goto lbl570;
-lbl510:	j--;
-      	if (np[j] < 0) goto lbl510;
-      	jj = jc;
-lbl520:	kspan = jj;
-      	if(jj > maxf) 
-		kspan = maxf;
-      	jj -= kspan;
-      	k = np[j];	
-      	kk = (jc*k) + i + jj;
-      	k1 = kk + kspan;
-      	k2 = 0;
-lbl530:	k2++;
-      	at[k2] = a[k1];
-      	bt[k2] = b[k1];
-      	k1 -= inc;
-      	if(k1 != kk) goto lbl530;
-lbl540:	k1 = kk + kspan;
-      	k2 = k1 - (jc * (k + np[k]));
-      	k = -(np[k]);
-lbl550:	a[k1] = a[k2];
-      	b[k1] = b[k2];
-      	k1 -= inc;
-      	k2 -= inc;
-      	if(k1 != kk) goto lbl550;
-      	kk = k2;
-      	if(k != j)   goto lbl540;
-      	k1 = kk + kspan;
-      	k2 = 0;
-lbl560:	k2++;
-      	a[k1] = at[k2];
-      	b[k1] = bt[k2];
-      	k1 -= inc;
-      	if(k1 != kk) goto lbl560;
-      	if(jj)       goto lbl520;
-      	if(j  != 1)  goto lbl510;
-lbl570:	j = k3 + 1;
-      	nt -= kspnn;
-      	i = nt - inc + 1;
-      	if(nt >= 0)  goto lbl510;
-      	return; 
-}
-
-
-/*
- *-----------------------------------------------------------------------
- * subroutine:  reals
- * used with 'fft' to compute fourier transform or inverse for real data
- *-----------------------------------------------------------------------
- *	this is the call from C:
- *		reals_(anal,banal,N2,mtwo);
- *	which has been changed from CARL call
- *		reals_(anal,banal,&N2,&mtwo);
- */
-
-void
-reals_(a, b, n, isn)
-
-float 	*a,	/* a refers to an array of floats 'anal'   */
-	*b;	/* b refers to an array of floats 'banal'  */
-int	n,
-	isn;
-
-/* See IEEE book for a long comment here on usage */
-
-{	int	inc,
-		j,
-		k,
-		lim,
-		mm,ml,
-		nf,nk,nh;
- 
-	double	aa,ab,
-		ba,bb,
-		cd,cn,
-		dr,
-		em,
-		rad,re,
-		sd,sn;
-	double	xx;	/******* ADDED APRIL 1991 ******/
-/* adjust  input array pointers (called from C) */
-	a--;	b--;
-	inc=abs(isn);
-	nf=abs(n);
-      	if(nf*isn==0){
-		fprintf(stderr,"\nerror - zero in reals parameters : %d : %d ",n,isn);
-	       	return;
-	}
-lbl10: 	nk = (nf*inc) + 2;
-      	nh = nk/2;
-/*****************************
-    	rad  = atan((double)1.0);
-******************************/
-	rad = 0.785398163397448278900;
-      	dr = -4.0/(double)(nf);
-/********************************** POW2 REMOVED APRIL 1991 *****************
-			      	cd = 2.0 * (pow2(sin((double)0.5 * dr * rad)));
-*****************************************************************************/
-	xx = sin((double)0.5 * dr * rad);
-      	cd = 2.0 * xx * xx;
-      	sd = sin(dr * rad);
-/*
- * sin,cos values are re-initialized each lim steps
- */
-      	lim = 32;
-      	mm = lim;
-      	ml = 0;
-      	sn = 0.0;
-	if(isn<0){
-		cn = 1.0;
-		a[nk-1] = a[1];
-		b[nk-1] = b[1]; }
-	else {
-		cn = -1.0;
-		sd = -sd;
-	}
-lbl20: 	for(j=1;j<=nh;j+=inc)	{
-        	k = nk - j;
-        	aa = a[j] + a[k];
-        	ab = a[j] - a[k];
-        	ba = b[j] + b[k];
-        	bb = b[j] - b[k];
-        	re = (cn*ba) + (sn*ab);
-        	em = (sn*ba) - (cn*ab);
-        	b[k] = (em-bb)*0.5;
-        	b[j] = (em+bb)*0.5;
-        	a[k] = (aa-re)*0.5;
-		a[j] = (aa+re)*0.5;
-        	ml++;
-		if(ml!=mm){
-			aa = cn - ((cd*cn)+(sd*sn));
-			sn = ((sd*cn) - (cd*sn)) + sn;
-			cn = aa;}
-		else {
-			mm +=lim;
-			sn = ((float)ml) * dr * rad;
-			cn = cos(sn);
-			if(isn>0)
-				cn = -cn;
-			sn = sin(sn);
-		}
-	}
-	return;
-}
-
-

dev/externals/paprogs/pvplay/pvdefs.h

@@ -1,43 +0,0 @@
-/* pvdefs.h */
-#ifndef __PVDEFS_H_INCLUDED
-#define __PVDEFS_H_INCLUDED
-
-#ifdef __cplusplus
-extern "C" {
-#endif
-
-
-#define VERY_TINY_VAL (1e-20)
-
-#define ODD(x)	  			((x)&1)
-#define EVEN(x)	  			(!ODD(x))
-#define CHAN_SRCHRANGE_F	(4)
-
-
-#ifndef max
-#define max(a, b)  (((a) > (b)) ? (a) : (b)) 
-#endif
-#ifndef min
-#define min(a, b)  (((a) < (b)) ? (a) : (b)) 
-#endif
-
-
-/* for future reference: IEEE_DOUBLE not implemented yet for PVOCEX */
-enum pvoc_wordformat { PVOC_IEEE_FLOAT, PVOC_IEEE_DOUBLE};
-typedef enum pvoc_mode {  PVPP_NOT_SET,PVPP_OFFLINE,PVPP_STREAMING} pvocmode;
-/* the old CARL pvoc flags */
-typedef enum pvoc_wfac {PVOC_O_W0,PVOC_O_W1,PVOC_O_W2,PVOC_O_W3,PVOC_O_DEFAULT} pvoc_overlapfac;
-typedef enum pvoc_scaletype {PVOC_S_TIME,PVOC_S_PITCH,PVOC_S_NONE} pv_scaletype;
-
-typedef enum pvoc_frametype { PVOC_AMP_FREQ,PVOC_AMP_PHASE,PVOC_COMPLEX } pv_frametype;
-typedef enum pvoc_windowtype {PVOC_DEFAULT,
-								PVOC_HAMMING,
-								PVOC_HANN,
-								PVOC_KAISER,
-								PVOC_RECT,
-								PVOC_CUSTOM} pv_wtype;
-
-#ifdef __cplusplus
-}
-#endif
-#endif

dev/externals/paprogs/pvplay/pvfileio.c

@@ -1,1300 +0,0 @@
-/* pvfileio.c */
-/* pvocex format test routines*/
-/* Initial version RWD May 2000.
- * All rights reserved: work in progress!
- * Manifestly not a complete API yet!
- * In particular, error returns are kept very simplistic at the moment.
- * (and are not even very consistent yet...)
- * In time, a full set of error values and messages will be developed.
- *
- * NB: the RIFF<WAVE> functions only look for, and accept, a PVOCEX format file.
- * NB also: if windows.h is included anywhere (should be no need in this file,
- *          or in pvfileio.h),
- *          the WAVE_FORMAT~ symbols may need to be #ifndef-ed.
- */
-
-/*   very simple CUSTOM window chunk:
- *
- *  <PVXW><size><data>
- *
- *  where size as usual gives the size of the data in bytes.
- *  the size in samples much match dwWinlen (which may not be the same as N (fft length)
- *  the sample type must be the same as that of the pvoc data itself
- *  (only floatsams supported so far)
- *  values must be normalized to peak of 1.0
- */
-//#ifdef WINDOWS
-//#include "stdafx.h"
-//#endif
-
-#include <stdio.h>
-#include <stdlib.h>
-#include <memory.h>
-#include <string.h>
-#include <fcntl.h>
-#include <sys/types.h>
-#include <sys/stat.h>
-#ifdef _WIN32
-#include <io.h>
-#endif
-#ifdef unix
-#include <unistd.h>
-#define O_BINARY (0)
-#define _S_IWRITE S_IWRITE
-#define _S_IREAD  S_IREAD
-#endif
-
-#ifdef _DEBUG
-#include <assert.h>
-#endif
-#include "pvdefs.h"
-#include "pvfileio.h"
-
-/* NB probably no good on 64bit platforms */
-#define REVDWBYTES(t)   ( (((t)&0xff) << 24) | (((t)&0xff00) << 8) | (((t)&0xff0000) >> 8) | (((t)>>24) & 0xff) )
-#define REVWBYTES(t)    ( (((t)&0xff) << 8) | (((t)>>8) &0xff) )
-#define TAG(a,b,c,d)    ( ((a)<<24) | ((b)<<16) | ((c)<<8) | (d) )
-#define WAVE_FORMAT_EXTENSIBLE (0xFFFE)
-#ifndef WAVE_FORMAT_PCM
-#define WAVE_FORMAT_PCM (0x0001)
-#endif
-#define WAVE_FORMAT_IEEE_FLOAT (0x0003)
-
-
-const GUID KSDATAFORMAT_SUBTYPE_PVOC = {
-                    0x8312b9c2,
-                    0x2e6e,
-                    0x11d4,
-                    { 0xa8, 0x24, 0xde, 0x5b, 0x96, 0xc3, 0xab, 0x21 }
-};
-
-
-static  char *pv_errstr = "";
-
-#define MAXFILES (16)
-/* or any desired larger number: will be dynamically allocated one day */
-
-
-typedef struct pvoc_file {
-    WAVEFORMATEX fmtdata;
-    PVOCDATA pvdata;
-    long datachunkoffset;
-    long nFrames;   /* no of frames in file */
-    long FramePos;  /* where we are in file */
-    long curpos;
-    int fd;
-    int to_delete;
-    int readonly;
-    int do_byte_reverse;
-    char *name;
-    float *customWindow;
-} PVOCFILE;
-
-static PVOCFILE *files[MAXFILES];
-
-static int pvoc_writeheader(int ofd);
-static int pvoc_readheader(int ifd,WAVEFORMATPVOCEX *pWfpx);
-
-
-static int write_guid(int fd,int byterev,const GUID *pGuid)
-{
-    long written;
-#ifdef _DEBUG
-    assert(fd >= 0);
-    assert(pGuid);
-#endif
-
-    if(byterev){
-        GUID guid;
-        guid.Data1 = REVDWBYTES(pGuid->Data1);
-        guid.Data2 = REVWBYTES(pGuid->Data2);
-        guid.Data3 = REVWBYTES(pGuid->Data3);
-        memcpy((char *) (guid.Data4),(char *) (pGuid->Data4),8);
-        written = write(fd,(char *) &guid,sizeof(GUID));
-    }
-    else
-        written = write(fd,(char *) pGuid,sizeof(GUID));
-
-    return written;
-
-}
-
-static int compare_guids(const GUID *gleft, const GUID *gright)
-{
-    const char *left = (const char *) gleft, *right = (const char *) gright;
-    return !memcmp(left,right,sizeof(GUID));
-}
-
-
-static int write_pvocdata(int fd,int byterev,const PVOCDATA *pData)
-{
-    long written;
-    long dwval;
-#ifdef _DEBUG
-    assert(fd >= 0);
-    assert(pData);
-#endif
-
-
-    if(byterev){
-        PVOCDATA data;
-        data.wWordFormat = REVWBYTES(pData->wWordFormat);
-        data.wAnalFormat = REVWBYTES(pData->wAnalFormat);
-        data.wSourceFormat = REVWBYTES(pData->wSourceFormat);
-        data.wWindowType = REVWBYTES(pData->wWindowType);
-        data.nAnalysisBins = REVDWBYTES(pData->nAnalysisBins);
-        data.dwWinlen   = REVDWBYTES(pData->dwWinlen);
-        data.dwOverlap   = REVDWBYTES(pData->dwOverlap);
-        data.dwFrameAlign = REVDWBYTES(pData->dwFrameAlign);
-        
-        dwval = * (long *) &(pData->fAnalysisRate);
-        dwval = REVDWBYTES(dwval);
-        data.fAnalysisRate = * (float *) &dwval;
-
-        dwval = * (long *) &(pData->fWindowParam);
-        dwval = REVDWBYTES(dwval);
-        data.fWindowParam = * (float *) &dwval;
-
-        written = write(fd,(char *) &data,sizeof(PVOCDATA));
-    }
-    else
-        written = write(fd,(char *) pData,sizeof(PVOCDATA));
-
-    return written;
-
-}
-
-static int write_fmt(int fd, int byterev,const WAVEFORMATEX *pfmt)
-{   
-    /*NB have to write out each element, as not guaranteed alignment othewise.
-     *  Consider it documentation. */
-
-#ifdef _DEBUG
-    assert(fd >=0);
-    assert(pfmt);
-#endif
-
-    if(byterev){
-        WAVEFORMATEX fmt;
-        fmt.wFormatTag      = REVWBYTES(pfmt->wFormatTag);
-        fmt.nChannels       = REVWBYTES(pfmt->nChannels);
-        fmt.nSamplesPerSec  = REVDWBYTES(pfmt->nSamplesPerSec);
-        fmt.nAvgBytesPerSec = REVDWBYTES(pfmt->nAvgBytesPerSec);
-        fmt.nBlockAlign     = REVWBYTES(pfmt->nBlockAlign);
-        fmt.wBitsPerSample  = REVWBYTES(pfmt->wBitsPerSample);
-        fmt.cbSize          = REVWBYTES(pfmt->cbSize);
-
-        if(write(fd,(char *) &(fmt.wFormatTag),sizeof(WORD)) != sizeof(WORD)
-            || write(fd,(char *) &(fmt.nChannels),sizeof(WORD)) != sizeof(WORD)
-            || write(fd,(char *) &(fmt.nSamplesPerSec),sizeof(DWORD)) != sizeof(DWORD)
-            || write(fd,(char *) &(fmt.nAvgBytesPerSec),sizeof(DWORD)) != sizeof(DWORD)
-            || write(fd,(char *) &(fmt.nBlockAlign),sizeof(WORD)) != sizeof(WORD)
-            || write(fd,(char *) &(fmt.wBitsPerSample),sizeof(WORD)) != sizeof(WORD)
-            || write(fd,(char *) &(fmt.cbSize),sizeof(WORD)) != sizeof(WORD))
-
-        return 0;
-
-    }
-    else {
-       if(write(fd,(char *) &(pfmt->wFormatTag),sizeof(WORD)) != sizeof(WORD)
-            || write(fd,(char *) &(pfmt->nChannels),sizeof(WORD)) != sizeof(WORD)
-            || write(fd,(char *) &(pfmt->nSamplesPerSec),sizeof(DWORD)) != sizeof(DWORD)
-            || write(fd,(char *) &(pfmt->nAvgBytesPerSec),sizeof(DWORD)) != sizeof(DWORD)
-            || write(fd,(char *) &(pfmt->nBlockAlign),sizeof(WORD)) != sizeof(WORD)
-            || write(fd,(char *) &(pfmt->wBitsPerSample),sizeof(WORD)) != sizeof(WORD)
-            || write(fd,(char *) &(pfmt->cbSize),sizeof(WORD)) != sizeof(WORD))
-
-        return 0;
-    }
-
-    return SIZEOF_WFMTEX;
-}
-
-static int pvoc_writeWindow(int fd,int byterev,float *window,DWORD length)
-{
-    if(byterev){
-        /* don't corrupt source array! */
-        DWORD i;
-        long lval, *lp = (long *) window;
-
-        for(i=0;i < length; i++){
-            lval = *lp++;
-            lval = REVDWBYTES(lval);
-            if(write(fd,(char *)&lval,sizeof(long)) != sizeof(long))
-                return 0;
-        }
-    }
-    else{
-        if(write(fd,(char *) window,length * sizeof(float)) != (int)(length*sizeof(float)))
-            return 0;
-    }
-
-
-    return length * sizeof(float);
-}
-
-static int pvoc_readWindow(int fd, int byterev, float *window,DWORD length)
-{
-    if(byterev){
-        DWORD i;
-        long got,oval,lval, *lp = (long *) window;
-#ifdef SINGLE_FLOAT 
-        for(i=0;i < length;i++){
-            if(read(fd,(char *)&lval,sizeof(long)) != sizeof(long))
-                return 0;
-            oval = REVDWBYTES(lval);
-            *lp++ = oval;
-        }
-#else
-        /* read whole block then swap...should be faster */
-        got = read(fd,(char *) window,length * sizeof(float));
-        if(got != (int)(length * sizeof(float)))
-            return 0;
-        /* then byterev */
-        for(i=0;i < length;i++){
-            lval = *lp;
-            oval = REVDWBYTES(lval);
-            *lp++ = oval;
-        }
-
-#endif
-    }
-    else{
-        if(read(fd,(char *) window,length * sizeof(float)) != (int)(length * sizeof(float)))
-            return 0;
-    }
-
-    return length * sizeof(float);
-
-}
-
-
-
-const char *pvoc_errorstr()
-{
-    return (const char *) pv_errstr;
-}
-
-
-
-/* thanks to the SNDAN programmers for this! */
-/* return 0 for big-endian machine, 1 for little-endian machine*/
-/* probably no good for 16bit swapping though */
-static int byte_order()
-{
-  int   one = 1;
-  char* endptr = (char *) &one;
-  return (*endptr);
-}
-
-/***** loosely modelled on CDP sfsys ******
- *  This is a static array, but coul be made dynamic in an OOP sort of way.
- *  The idea is that all low-level operations and data 
- *  are completely hidden from the user, so that internal format changes can be made
- * with little or no disruption to the public functions.
- * But avoiding the full monty of a C++ implementation.
-
- *******************************************/
-
-int init_pvsys(void)
-{
-    int i;
-
-    if(files[0] != NULL) {
-        pv_errstr = "\npvsys: already initialized";
-        return 0;
-    }
-    for(i = 0;i < MAXFILES;i++)
-        files[i] = NULL;
-
-    return 1;
-}
-
-static void prepare_pvfmt(WAVEFORMATEX *pfmt,unsigned long chans, unsigned long srate, 
-                          pv_stype stype)
-{
-
-#ifdef _DEBUG
-    assert(pfmt);
-#endif
-
-
-    pfmt->wFormatTag        = WAVE_FORMAT_EXTENSIBLE;
-    pfmt->nChannels         = (WORD) chans;
-    pfmt->nSamplesPerSec    = srate;
-    switch(stype){
-    case(STYPE_16):
-        pfmt->wBitsPerSample = (WORD)16;
-        pfmt->nBlockAlign    = (WORD)(chans * 2 *  sizeof(char));
-        
-        break;
-    case(STYPE_24):
-        pfmt->wBitsPerSample = (WORD) 24;
-        pfmt->nBlockAlign    = (WORD)(chans *  3 * sizeof(char));
-        break;
-    case(STYPE_32):
-    case(STYPE_IEEE_FLOAT):
-        pfmt->wBitsPerSample = (WORD) 32;
-        pfmt->nBlockAlign    = (WORD)(chans *  4 * sizeof(char));
-        break;
-    default:
-#ifdef _DEBUG
-        assert(0);
-#endif
-        break;
-    }
-
-    pfmt->nAvgBytesPerSec   = pfmt->nBlockAlign * srate;
-    /* if we have extended WindowParam fields, or something ,will need to adjust this */
-    pfmt->cbSize            = 62;
-
-}
-
-
-/* lots of different ways of doing this! */
-/* we will need  one in the form:
- * in pvoc_fmtcreate(const char *fname,PVOCDATA *p_pvfmt, WAVEFORMATEX *p_wvfmt);
- */
-
-/* a simple minimalist function to begin with!*/
-/*set D to 0, and/or dwWinlen to 0, to use internal default */
-/* fWindow points to userdef window, or is NULL */
-/* NB currently this does not enforce a soundfile extension; probably it should... */
-int  pvoc_createfile(const char *filename, 
-                     DWORD fftlen,DWORD overlap,DWORD chans,
-                     DWORD format,long srate, 
-                     pv_stype stype,pv_wtype wtype,
-                     float wparam,float *fWindow,DWORD dwWinlen)
-{
-    
-    int i;
-    long N,D;
-    char *pname;
-    PVOCFILE *pfile = NULL;
-    float winparam = 0.0f;
-
-    N = fftlen;                   /* keep the CARL varnames for now */
-    D = overlap;
-
-    if(N == 0 || chans <=0 || filename==NULL || D > N) {
-        pv_errstr = "\npvsys: bad arguments";
-        return -1;
-    }
-    if(/*format < PVOC_AMP_FREQ ||*/ format > PVOC_COMPLEX) {    /* RWD unsigned, so nothing < 0 possible */
-        pv_errstr = "\npvsys: bad format parameter";
-        return -1;
-    }
-
-    if(!(wtype >= PVOC_DEFAULT && wtype <= PVOC_CUSTOM)){
-        pv_errstr = "\npvsys: bad window type";
-        return -1;
-    }
-
-    /* load it, but ca't write until we have a PVXW chunk definition...*/
-    if(wtype==PVOC_CUSTOM){
-
-    }
-
-    if(wtype==PVOC_DEFAULT)
-        wtype = PVOC_HAMMING;
-
-    if(wtype==PVOC_KAISER)
-        if(wparam != 0.0f)
-            winparam = wparam;
-    /*will need an internal default for window parameters...*/
-
-    for(i=0;i < MAXFILES;i++)
-        if(files[i]==NULL)
-            break;
-    if(i==MAXFILES) {
-        pv_errstr = "\npvsys: too many files open";
-        return -1;
-    }
-    pfile = (PVOCFILE *) malloc(sizeof(PVOCFILE));
-
-    if(pfile==NULL){
-        pv_errstr = "\npvsys: no memory";
-        return -1;
-    }
-    pname = (char *) malloc(strlen(filename)+1);
-    if(pname == NULL){
-        free(pfile);
-        pv_errstr = "\npvsys: no memory";
-        return -1;
-    }
-    pfile->customWindow = NULL;
-    /* setup rendering inforamtion */
-    prepare_pvfmt(&pfile->fmtdata,chans,srate,stype);
-
-    strcpy(pname,filename);
-
-    pfile->pvdata.wWordFormat = PVOC_IEEE_FLOAT;
-    pfile->pvdata.wAnalFormat = (WORD) format;
-    pfile->pvdata.wSourceFormat =  (stype == STYPE_IEEE_FLOAT ? WAVE_FORMAT_IEEE_FLOAT : WAVE_FORMAT_PCM);
-    pfile->pvdata.wWindowType = wtype;
-    pfile->pvdata.nAnalysisBins = (N>>1) + 1;
-    if(dwWinlen==0)
-        pfile->pvdata.dwWinlen      = N;
-    else
-        pfile->pvdata.dwWinlen  = dwWinlen;
-    if(D==0)
-        pfile->pvdata.dwOverlap = N/8;
-    else
-        pfile->pvdata.dwOverlap = D;
-    pfile->pvdata.dwFrameAlign = pfile->pvdata.nAnalysisBins * 2 * sizeof(float);
-    pfile->pvdata.fAnalysisRate = (float)srate / (float) pfile->pvdata.dwOverlap;
-    pfile->pvdata.fWindowParam = winparam;
-    pfile->to_delete = 0;
-    pfile->readonly = 0;
-    if(fWindow!= NULL){
-        pfile->customWindow = malloc(dwWinlen * sizeof(float));
-        if(pfile->customWindow==NULL){
-            pv_errstr = "\npvsys: no memory for custom window";
-            return -1;
-        }
-        memcpy((char *)(pfile->customWindow),(char *)fWindow,dwWinlen * sizeof(float));
-    }
-
-
-    pfile->fd = open(filename,O_BINARY | O_CREAT | O_RDWR | O_TRUNC,_S_IWRITE | _S_IREAD);
-    if(pfile->fd < 0){
-        free(pname);        
-        if(pfile->customWindow)
-            free(pfile->customWindow);
-        free(pfile);
-
-        pv_errstr = "\npvsys: unable to create file";
-        return -1;
-    }
-
-    pfile->datachunkoffset = 0;
-    pfile->nFrames = 0;
-    pfile->FramePos = 0;
-    pfile->curpos = 0;
-    pfile->name = pname;
-    pfile->do_byte_reverse = !byte_order(); 
-    files[i] = pfile;
-
-    if(!pvoc_writeheader(i)) {
-        close(pfile->fd);
-        remove(pfile->name);
-        free(pfile->name);
-        if(pfile->customWindow)
-            free(pfile->customWindow);
-        free(pfile);
-        files[i] = NULL;
-        return -1;
-    }
-
-    return i;
-}
-
-int pvoc_openfile(const char *filename,PVOCDATA *data,WAVEFORMATEX *fmt)
-{
-    int i;
-    WAVEFORMATPVOCEX wfpx;
-    char *pname;
-    PVOCFILE *pfile = NULL;
-//  long size = sizeof(WAVEFORMATPVOCEX);
-    
-    if(data==NULL || fmt==NULL){
-        pv_errstr = "\npvsys: Internal error: NULL data arrays";
-        return -1;
-    }
-
-    for(i=0;i < MAXFILES;i++)
-        if(files[i]==NULL)
-            break;
-    if(i==MAXFILES){
-        pv_errstr = "\npvsys: too many files open";
-        return -1;
-    }
-
-    pfile = (PVOCFILE *) malloc(sizeof(PVOCFILE));
-    if(pfile==NULL){
-        pv_errstr = "\npvsys: no memory for file data";
-        return -1;
-    }
-    pfile->customWindow = NULL;
-    pname = (char *) malloc(strlen(filename)+1);
-    if(pname == NULL){
-        free(pfile);
-        pv_errstr = "\npvsys: no memory";
-        return -1;
-    }
-    pfile->fd = open(filename,O_BINARY | O_RDONLY,_S_IREAD);
-    if(pfile->fd < 0){
-        free(pname);
-        free(pfile);
-        pv_errstr = "\npvsys: unable to create file";
-        return -1;
-    }
-    strcpy(pname,filename);
-    pfile->datachunkoffset = 0;
-    pfile->nFrames = 0;
-    pfile->curpos = 0;
-    pfile->FramePos = 0;
-    pfile->name = pname;
-    pfile->do_byte_reverse = !byte_order(); 
-    pfile->readonly = 1;
-    files[i] = pfile;
-
-    if(!pvoc_readheader(i,&wfpx)){
-        close(pfile->fd);
-        free(pfile->name);
-        if(pfile->customWindow)
-            free(pfile->customWindow);
-        free(pfile);
-        files[i] = NULL;
-        return -1;
-    }
-
-    memcpy((char *)data, (char *)&(wfpx.data),sizeof(PVOCDATA));
-    memcpy((char *)fmt,(char *)&(wfpx.wxFormat.Format),SIZEOF_WFMTEX);
-
-    files[i] = pfile;
-
-    return i;
-
-}
-/*RWD TODO: add byterev stuff*/
-static int pvoc_readfmt(int fd,int byterev,WAVEFORMATPVOCEX *pWfpx)
-{
-    unsigned long dword;
-    unsigned short word;
-
-#ifdef _DEBUG
-    assert(fd >= 0);
-    assert(pWfpx);
-#endif
-
-    if(read(fd,(char *) &(pWfpx->wxFormat.Format.wFormatTag),sizeof(WORD)) != sizeof(WORD)
-        || read(fd,(char *) &(pWfpx->wxFormat.Format.nChannels),sizeof(WORD)) != sizeof(WORD)
-        || read(fd,(char *) &(pWfpx->wxFormat.Format.nSamplesPerSec),sizeof(DWORD)) != sizeof(DWORD)
-        || read(fd,(char *) &(pWfpx->wxFormat.Format.nAvgBytesPerSec),sizeof(DWORD)) != sizeof(DWORD)
-        || read(fd,(char *) &(pWfpx->wxFormat.Format.nBlockAlign),sizeof(WORD)) != sizeof(WORD)
-        || read(fd,(char *) &(pWfpx->wxFormat.Format.wBitsPerSample),sizeof(WORD)) != sizeof(WORD)
-        || read(fd,(char *) &(pWfpx->wxFormat.Format.cbSize),sizeof(WORD)) != sizeof(WORD)){
-        pv_errstr = "\npvsys: error reading Source format data";
-        return 0;
-    }
-
-    if(byterev){
-        word = pWfpx->wxFormat.Format.wFormatTag;
-        pWfpx->wxFormat.Format.wFormatTag= REVWBYTES(word);
-        word = pWfpx->wxFormat.Format.nChannels;
-        pWfpx->wxFormat.Format.nChannels = REVWBYTES(word);
-        dword = pWfpx->wxFormat.Format.nSamplesPerSec;
-        pWfpx->wxFormat.Format.nSamplesPerSec = REVDWBYTES(dword);
-        dword = pWfpx->wxFormat.Format.nAvgBytesPerSec;
-        pWfpx->wxFormat.Format.nAvgBytesPerSec = REVDWBYTES(dword);
-        word = pWfpx->wxFormat.Format.nBlockAlign;
-        pWfpx->wxFormat.Format.nBlockAlign = REVWBYTES(word);
-        word = pWfpx->wxFormat.Format.wBitsPerSample;
-        pWfpx->wxFormat.Format.wBitsPerSample = REVWBYTES(word);
-        word = pWfpx->wxFormat.Format.cbSize;
-        pWfpx->wxFormat.Format.cbSize = REVWBYTES(word);
-
-    }
-
-    /* the first clues this is pvx format...*/
-    if(pWfpx->wxFormat.Format.wFormatTag != WAVE_FORMAT_EXTENSIBLE){
-        pv_errstr = "\npvsys: not a WAVE_EX file";
-        return 0;
-    }
-
-    if(pWfpx->wxFormat.Format.cbSize != 62){
-        pv_errstr = "\npvsys: bad size for fmt chunk";
-        return 0;
-    }
-
-    if(read(fd,(char *) &(pWfpx->wxFormat.Samples.wValidBitsPerSample),sizeof(WORD)) != sizeof(WORD)
-        || read(fd,(char *) &(pWfpx->wxFormat.dwChannelMask),sizeof(DWORD)) != sizeof(DWORD)
-        || read(fd,(char *) &(pWfpx->wxFormat.SubFormat),sizeof(GUID)) != sizeof(GUID)){
-        pv_errstr = "\npvsys: error reading Extended format data";
-        return 0;
-    }
-
-    if(byterev){
-        word = pWfpx->wxFormat.Samples.wValidBitsPerSample;
-        pWfpx->wxFormat.Samples.wValidBitsPerSample = REVWBYTES(word);
-        dword = pWfpx->wxFormat.dwChannelMask;
-        pWfpx->wxFormat.dwChannelMask = REVDWBYTES(dword);
-
-        dword = pWfpx->wxFormat.SubFormat.Data1;
-        pWfpx->wxFormat.SubFormat.Data1 = REVDWBYTES(dword);
-        word = pWfpx->wxFormat.SubFormat.Data2;
-        pWfpx->wxFormat.SubFormat.Data2 = REVWBYTES(word);
-        word = pWfpx->wxFormat.SubFormat.Data3;
-        pWfpx->wxFormat.SubFormat.Data3 = REVWBYTES(word);
-        /* don't need to reverse the char array */
-    }
-
-    /* ... but this is the clincher */
-    if(!compare_guids(&(pWfpx->wxFormat.SubFormat),&KSDATAFORMAT_SUBTYPE_PVOC)){
-        pv_errstr = "\npvsys: not a PVOCEX file";
-        return 0;
-    }
-
-    if(read(fd,(char *) &(pWfpx->dwVersion),sizeof(DWORD)) != sizeof(DWORD)
-        || read(fd,(char *) &(pWfpx->dwDataSize),sizeof(DWORD)) != sizeof(DWORD)
-        || read(fd,(char *) &(pWfpx->data),sizeof(PVOCDATA)) != sizeof(PVOCDATA)){
-        pv_errstr = "\npvsys: error reading Extended pvoc format data";
-        return 0;
-    }
-
-    if(byterev){
-        dword = pWfpx->dwVersion;
-        pWfpx->dwVersion = REVDWBYTES(dword);
-
-        /* check it now! */
-        if(pWfpx->dwVersion != PVX_VERSION){
-            pv_errstr = "\npvsys: unknown pvocex Version";
-            return 0;
-        }
-
-        dword = pWfpx->dwDataSize;
-        pWfpx->dwDataSize = REVDWBYTES(dword);
-
-        word = pWfpx->data.wWordFormat;
-        pWfpx->data.wWordFormat= REVWBYTES(word);
-        word = pWfpx->data.wAnalFormat;
-        pWfpx->data.wAnalFormat= REVWBYTES(word);
-        word = pWfpx->data.wSourceFormat;
-        pWfpx->data.wSourceFormat= REVWBYTES(word);
-        word = pWfpx->data.wWindowType;
-        pWfpx->data.wWindowType= REVWBYTES(word);
-
-        dword = pWfpx->data.nAnalysisBins;
-        pWfpx->data.nAnalysisBins = REVDWBYTES(dword);
-        dword = pWfpx->data.dwWinlen;
-        pWfpx->data.dwWinlen = REVDWBYTES(dword);
-        dword = pWfpx->data.dwOverlap;
-        pWfpx->data.dwOverlap = REVDWBYTES(dword);
-        dword = pWfpx->data.dwFrameAlign;
-        pWfpx->data.dwFrameAlign = REVDWBYTES(dword);
-
-        dword = * (DWORD *)&(pWfpx->data.fAnalysisRate);
-        dword = REVDWBYTES(dword);
-        pWfpx->data.fAnalysisRate = *(float *)&dword;
-        dword = * (DWORD *)&(pWfpx->data.fWindowParam);
-        dword = REVDWBYTES(dword);
-        pWfpx->data.fWindowParam = *(float *)&dword;
-
-    }
-    if(pWfpx->dwVersion != PVX_VERSION){
-        pv_errstr = "\npvsys: unknown pvocex Version";
-        return 0;
-    }
-
-    return 1;
-
-
-}
-
-
-static int pvoc_readheader(int ifd,WAVEFORMATPVOCEX *pWfpx)
-{
-    DWORD tag, size,riffsize;
-    int fmtseen = 0, /* dataseen = 0,*/ windowseen = 0;
-//  DWORD windowlength = 0;
-
-#ifdef _DEBUG
-    assert(pWfpx);
-    assert(files[ifd]);
-    assert(files[ifd]->fd >= 0);
-    size = sizeof(WAVEFORMATEXTENSIBLE);
-    size += 2 * sizeof(DWORD);
-    size += sizeof(PVOCDATA);
-#endif
-
-    if(read(files[ifd]->fd,(char *) &tag,sizeof(DWORD)) != sizeof(DWORD)
-        || read(files[ifd]->fd,(char *) &size,sizeof(DWORD)) != sizeof(DWORD)){
-        pv_errstr = "\npvsys: error reading header";
-        return 0;
-    }
-    if(files[ifd]->do_byte_reverse)
-        size = REVDWBYTES(size);
-    else
-        tag = REVDWBYTES(tag);
-
-    if(tag != TAG('R','I','F','F')){
-        pv_errstr = "\npvsys: not a RIFF file";
-        return 0;
-    }
-    if(size < 24 * sizeof(DWORD) + SIZEOF_FMTPVOCEX){
-        pv_errstr = "\npvsys: file too small";
-        return 0;
-    }
-    riffsize = size;
-    if(read(files[ifd]->fd,(char *) &tag,sizeof(DWORD)) != sizeof(DWORD)){
-        pv_errstr = "\npvsys: error reading header";
-        return 0;
-    }
-
-    if(!files[ifd]->do_byte_reverse)
-        tag = REVDWBYTES(tag);
-
-    if(tag != TAG('W','A','V','E')){
-        pv_errstr = "\npvsys: not a WAVE file";
-        return 0;
-    }
-    riffsize -= sizeof(DWORD);
-    /*loop for chunks */
-    while(riffsize > 0){
-        if(read(files[ifd]->fd,(char *) &tag,sizeof(DWORD)) != sizeof(DWORD)
-            || read(files[ifd]->fd,(char *) &size,sizeof(DWORD)) != sizeof(DWORD)){
-            pv_errstr = "\npvsys: error reading header";
-            return 0;
-        }
-        if(files[ifd]->do_byte_reverse)
-            size = REVDWBYTES(size);
-        else
-            tag = REVDWBYTES(tag);
-        riffsize -= 2 * sizeof(DWORD);
-        switch(tag){
-        case TAG('f','m','t',' '):
-            /* bail out if not a pvoc file: not trying to read all WAVE formats!*/
-            if(size < SIZEOF_FMTPVOCEX){
-                pv_errstr = "\npvsys:   not a PVOC-EX file";
-                return 0;
-            }
-            if(!pvoc_readfmt(files[ifd]->fd,files[ifd]->do_byte_reverse,pWfpx)){
-                pv_errstr = "\npvsys: error reading format chunk";
-                return 0;
-            }
-            riffsize -=  SIZEOF_FMTPVOCEX;
-            fmtseen = 1;
-            memcpy((char *)&(files[ifd]->fmtdata),(char *)&(pWfpx->wxFormat),SIZEOF_WFMTEX);
-            memcpy((char *)&(files[ifd]->pvdata),(char *)&(pWfpx->data),sizeof(PVOCDATA));
-            break;
-        case TAG('P','V','X','W'):
-            if(!fmtseen){
-                pv_errstr = "\npvsys: PVXW chunk found before fmt chunk.";
-                return 0;
-            }
-            if(files[ifd]->pvdata.wWindowType!=PVOC_CUSTOM){
-                /*whaddayado? can you warn the user and continue?*/
-                pv_errstr = "\npvsys: PVXW chunk found but custom window not specified";
-                return 0;
-            }
-            files[ifd]->customWindow = malloc(files[ifd]->pvdata.dwWinlen * sizeof(float));
-            if(files[ifd]->customWindow == NULL){
-                pv_errstr = "\npvsys: no memory for custom window data.";
-                return 0;
-            }
-            if(pvoc_readWindow(files[ifd]->fd,files[ifd]->do_byte_reverse,
-                files[ifd]->customWindow,files[ifd]->pvdata.dwWinlen)
-                != (int)(files[ifd]->pvdata.dwWinlen * sizeof(float))){
-                pv_errstr = "\npvsys: error reading window data.";
-                return 0;
-            }
-            windowseen = 1;
-            break;
-        case TAG('d','a','t','a'):
-            if(riffsize - size  != 0){
-                pv_errstr = "\npvsys: bad RIFF file";
-                return 0;
-            }
-            
-            if(!fmtseen){
-                pv_errstr = "\npvsys: bad format, data chunk before fmt chunk";
-                return 0;
-            }
-
-            if(files[ifd]->pvdata.wWindowType==PVOC_CUSTOM)
-                if(!windowseen){
-                    pv_errstr = "\npvsys: custom window chunk PVXW not found";
-                    return 0;
-                }
-
-            files[ifd]->datachunkoffset = lseek(files[ifd]->fd,0,SEEK_CUR);
-            files[ifd]->curpos = files[ifd]->datachunkoffset;
-            /* not m/c frames, for now */
-            files[ifd]->nFrames = size / files[ifd]->pvdata.dwFrameAlign;
-
-            return 1;
-            break;
-        default:
-            /* skip any onknown chunks */
-            riffsize -= 2 * sizeof(DWORD);
-            if(lseek(files[ifd]->fd,size,SEEK_CUR) < 0){
-                pv_errstr = "\npvsys: error skipping unknown WAVE chunk";
-                return 0;
-            }
-            riffsize -= size;
-            break;
-        }
-
-    }
-    /* if here, something very wrong!*/
-    pv_errstr = "\npvsys: bad format in RIFF file";
-    return 0;
-
-}
-
-static int pvoc_writeheader(int ofd)
-{
-    long tag,size,version;
-    WORD validbits;
-    
-    const GUID *pGuid =  &KSDATAFORMAT_SUBTYPE_PVOC;
-
-#ifdef _DEBUG
-    assert(files[ofd] != NULL);
-    assert(files[ofd]->fd >=0);
-#endif
-
-
-    tag = TAG('R','I','F','F');
-    size = 0;
-    if(files[ofd]->do_byte_reverse)
-        size = REVDWBYTES(size);
-    if(!files[ofd]->do_byte_reverse)
-        tag = REVDWBYTES(tag);
-
-    if(write(files[ofd]->fd,&tag,sizeof(long)) != sizeof(long)
-        || write(files[ofd]->fd,&size,sizeof(long)) != sizeof(long)) {
-        pv_errstr = "\npvsys: error writing header";
-        return 0;
-    }
-
-    tag = TAG('W','A','V','E');
-    if(!files[ofd]->do_byte_reverse)
-        tag = REVDWBYTES(tag);
-    if(write(files[ofd]->fd,&tag,sizeof(long)) != sizeof(long)){
-        pv_errstr = "\npvsys: error writing header";
-        return 0;
-    }
-
-    tag = TAG('f','m','t',' ');
-    size =  SIZEOF_WFMTEX + sizeof(WORD) + 
-            sizeof(DWORD) 
-            + sizeof(GUID) 
-            + 2*sizeof(DWORD)
-            + sizeof(PVOCDATA);
-    if(files[ofd]->do_byte_reverse)
-        size = REVDWBYTES(size);
-    if(!files[ofd]->do_byte_reverse)
-        tag = REVDWBYTES(tag);
-    if(write(files[ofd]->fd,(char *)&tag,sizeof(long)) != sizeof(long)
-        || write(files[ofd]->fd,(char *)&size,sizeof(long)) != sizeof(long)) {
-        pv_errstr = "\npvsys: error writing header";
-        return 0;
-    }
-    
-    if(write_fmt(files[ofd]->fd,files[ofd]->do_byte_reverse,&(files[ofd]->fmtdata)) != SIZEOF_WFMTEX){
-        pv_errstr = "\npvsys: error writing fmt chunk";
-        return 0;
-    }
-
-    validbits = files[ofd]->fmtdata.wBitsPerSample;  /*nothing fancy here */
-    if(files[ofd]->do_byte_reverse)
-        validbits = REVWBYTES(validbits);
-
-    if(write(files[ofd]->fd,(char *) &validbits,sizeof(WORD)) != sizeof(WORD)){
-        pv_errstr = "\npvsys: error writing fmt chunk";
-        return 0;
-    }
-    /* we will take this from a WAVE_EX file, in due course */
-    size = 0;   /*dwChannelMask*/
-    if(write(files[ofd]->fd,(char *)&size,sizeof(long)) != sizeof(DWORD)){
-        pv_errstr = "\npvsys: error writing fmt chunk";
-        return 0;
-    }
-
-    if(write_guid(files[ofd]->fd,files[ofd]->do_byte_reverse,pGuid) != sizeof(GUID)){
-        pv_errstr = "\npvsys: error writing fmt chunk";
-        return 0;
-    }
-    version  = 1;
-    size = sizeof(PVOCDATA);
-    if(files[ofd]->do_byte_reverse){
-        version = REVDWBYTES(version);
-        size = REVDWBYTES(size);
-    }
-
-    if(write(files[ofd]->fd,&version,sizeof(long)) != sizeof(long)
-        || write(files[ofd]->fd,&size,sizeof(long)) != sizeof(long)){
-        pv_errstr = "\npvsys: error writing fmt chunk";
-        return 0;
-    }
-
-
-    if(write_pvocdata(files[ofd]->fd,files[ofd]->do_byte_reverse,&(files[ofd]->pvdata)) != sizeof(PVOCDATA)){
-        pv_errstr = "\npvsys: error writing fmt chunk";
-        return 0;
-
-    }
-
-    /* VERY experimental; may not even be a good idea...*/
-
-    if(files[ofd]->customWindow){
-        tag = TAG('P','V','X','W');
-        size = files[ofd]->pvdata.dwWinlen * sizeof(float);
-        if(files[ofd]->do_byte_reverse)
-            size = REVDWBYTES(size);
-        else
-            tag = REVDWBYTES(tag);
-        if(write(files[ofd]->fd,(char *)&tag,sizeof(long)) != sizeof(long)
-            || write(files[ofd]->fd,(char *)&size,sizeof(long)) != sizeof(long)) {
-            pv_errstr = "\npvsys: error writing header";
-            return 0;
-        }
-        if(pvoc_writeWindow(files[ofd]->fd,
-                            files[ofd]->do_byte_reverse,
-                            files[ofd]->customWindow,
-                            files[ofd]->pvdata.dwWinlen)!= (int)(files[ofd]->pvdata.dwWinlen * sizeof(float))){
-            pv_errstr = "\npvsys: error writing window data.";
-            return 0;
-        }
-    }
-
-    /* no other chunks to write yet! */
-    tag = TAG('d','a','t','a');
-    if(!files[ofd]->do_byte_reverse)
-        tag = REVDWBYTES(tag);
-    if(write(files[ofd]->fd,&tag,sizeof(long)) != sizeof(long)){
-        pv_errstr = "\npvsys: error writing header";
-        return 0;
-    }
-
-    /* we need to update size later on...*/
-
-    size = 0;
-    if(write(files[ofd]->fd,&size,sizeof(long)) != sizeof(long)){
-        pv_errstr = "\npvsys: error writing header";
-        return 0;
-    }
-    files[ofd]->datachunkoffset = lseek(files[ofd]->fd,0,SEEK_CUR);
-
-    files[ofd]->curpos = files[ofd]->datachunkoffset;
-    return 1;
-}
-
-
-static int pvoc_updateheader(int ofd)
-{
-    long riffsize,datasize;
-    unsigned long pos;
-
-#ifdef _DEBUG   
-    assert(files[ofd]);
-    assert(files[ofd]->fd >= 0);
-    assert(files[ofd]->curpos == lseek(files[ofd]->fd,0,SEEK_CUR));
-#endif
-
-    datasize = files[ofd]->curpos - files[ofd]->datachunkoffset;
-    pos = lseek(files[ofd]->fd,files[ofd]->datachunkoffset-sizeof(DWORD),SEEK_SET);
-    if(pos != files[ofd]->datachunkoffset-sizeof(DWORD)){
-        pv_errstr = "\npvsys: error updating data chunk";
-        return 0;
-    }
-
-    if(files[ofd]->do_byte_reverse)
-        datasize = REVDWBYTES(datasize);
-    if(write(files[ofd]->fd,(char *) &datasize,sizeof(DWORD)) != sizeof(DWORD)){
-        pv_errstr = "\npvsys: error updating data chunk";
-        return 0;
-    }
-
-
-    riffsize = files[ofd]->curpos - 2* sizeof(DWORD);
-    if(files[ofd]->do_byte_reverse)
-        riffsize = REVDWBYTES(riffsize);
-    pos = lseek(files[ofd]->fd,sizeof(DWORD),SEEK_SET);
-    if(pos != sizeof(DWORD)){
-        pv_errstr = "\npvsys: error updating data chunk";
-        return 0;
-    }
-    if(write(files[ofd]->fd,(char *) &riffsize,sizeof(DWORD)) != sizeof(DWORD)){
-        pv_errstr = "\npvsys: error updating riff chunk";
-        return 0;
-    }
-
-    pos = lseek(files[ofd]->fd,0,SEEK_END);
-    if(pos < 0){
-        pv_errstr = "\npvsys: error seeking to end of file";
-        return 0;
-    }
-    return 1;
-}
-
-
-
-
-int pvoc_closefile(int ofd)
-{
-    if(files[ofd]==NULL){
-        pv_errstr = "\npvsys: file does not exist";
-        return 0;
-    }
-
-    if(files[ofd]->fd < 0){
-        pv_errstr = "\npvsys: file not open";
-        return 0;
-    }
-    if(!files[ofd]->readonly)
-        if(!pvoc_updateheader(ofd))
-            return 0;
-    
-    close(files[ofd]->fd);
-    if(files[ofd]->to_delete && !(files[ofd]->readonly))
-        remove(files[ofd]->name);
-
-    free(files[ofd]->name);
-    free(files[ofd]);
-    files[ofd] = NULL;
-
-    return 1;
-}
-
-/* does not directly address m/c streams, or alternative numeric formats, yet
- * so for m/c files, write each frame in turn, for each channel.
- * The format requires multi-channel frames to be interleaved in the usual way:
- * if nChannels= 4, the file will contain:
- * frame[0][0],frame[0][1],frame[0][2],frame[0][3],frme[1][0],frame[1][1].....
- *
- * The idea is to offer e.g. a floats version and a longs version ONLY, but
- * independently of the underlying representation, so that the user can write a floats
- * block, even though the underlying format might be longs or doubles. Most importantly,
- * the user does not have to deal with byte-reversal, which would otherwise always be the case
- * it the user had direct access to the file.
- * 
- * So these functions are the most likely to change over time!.
- *
- * return 0 for error, 1 for success. This could change....
-
-
- */
-int pvoc_putframes(int ofd,const float *frame,long numframes)
-{
-    DWORD i;
-    DWORD towrite;  /* count in 'words' */
-    long temp,*lfp;
-    
-
-    if(files[ofd]==NULL){
-        pv_errstr = "\npvsys: bad file descriptor";
-        return 0;
-    }
-    if(files[ofd]->fd < 0){
-        pv_errstr = "\npvsys: file not open";
-        return 0;
-    }
-    /* NB doubles not supported yet */
-    
-    towrite = files[ofd]->pvdata.nAnalysisBins * 2 * numframes;
-    
-    if(files[ofd]->do_byte_reverse){
-        /* do this without overwriting source data! */
-        lfp = (long *) frame;
-        for(i=0;i < towrite; i++){
-            temp = *lfp++;
-            temp = REVDWBYTES(temp);
-            if(write(files[ofd]->fd,(char *) &temp,sizeof(long)) != sizeof(long)){
-                pv_errstr = "\npvsys: error writing data";
-                return 0;
-            }
-
-        }
-    }
-    else {
-        if(write(files[ofd]->fd,(char *) frame,towrite * sizeof(float)) != (int)(towrite * sizeof(float))){
-            pv_errstr = "\npvsys: error writing data";
-            return 0;
-        }
-    }
-
-    files[ofd]->FramePos += numframes;
-    files[ofd]->curpos += towrite * sizeof(float);
-    return 1;
-}
-
-/* Simplistic read function
- * best practice here is to read nChannels frames *
- * return -1 for error, 0 for EOF, else numframes read
- */
-int pvoc_getframes(int ifd,float *frames,unsigned long nframes)
-{
-    long i;
-    long toread;
-    long oval,temp,*lfp;
-    long got;
-    int rc = -1;
-    if(files[ifd]==NULL){
-        pv_errstr = "\npvsys: bad file descriptor";
-        return rc;
-    }
-    if(files[ifd]->fd < 0){
-        pv_errstr = "\npvsys: file not open";
-        return rc;
-    }
-
-    toread = files[ifd]->pvdata.nAnalysisBins * 2 * nframes;
-
-    if(files[ifd]->do_byte_reverse){
-        lfp = (long *) frames;
-#ifdef SINGLE_FLOAT
-        for(i=0;i < toread;i++){
-            if((got=read(files[ifd]->fd,(char *) &temp,sizeof(long))) <0){
-                pv_errstr = "\npvsys: error reading data";
-                return rc;
-            }
-            if(got < sizeof(long)){
-                /* file size incorrect? */
-                return 0;           /* assume EOF */
-            }
-            temp = REVDWBYTES(temp);
-            *lfp++ = temp;
-        }
-#else
-        /* much faster on G4!!! */
-        got = read(files[ifd]->fd,(char *)frames,toread * sizeof(float));
-        if(got < 0){
-            pv_errstr = "\npvsys: error reading data";
-            return rc;
-        }
-        for(i=0;i < got / sizeof(float);i++){
-            temp = *lfp;
-            oval = REVDWBYTES(temp);
-            *lfp++ = oval;
-        }
-        if(got < (int)(toread * sizeof(float))){
-            /* some (user?) error in file size: return integral number of frames read*/
-            toread  = got / (files[ifd]->pvdata.nAnalysisBins * 2 * sizeof(float));
-            /* RWD 4:2002 need to revise this too */
-            nframes = toread;
-        }
-            
-#endif
-        rc = nframes;   /*RWD 4:2002 */
-    }
-    else{
-        if((got = read(files[ifd]->fd,(char *)frames,toread * sizeof(float))) < (int)(toread * sizeof(float))){
-            if(got < 0){
-                pv_errstr = "\npvsys: error reading data";
-                return rc;
-            }
-            else if(got < (int)(toread * sizeof(float))){
-                /* some (user?) error in file size: return integral number of frames read*/
-                toread  = got / (files[ifd]->pvdata.nAnalysisBins * 2 * sizeof(float));
-                rc = toread;
-                /* RWD 4:2002 need to revise this too */
-                nframes = toread;
-            }
-        }
-        else
-            rc = nframes;
-    }
-    /*files[ifd]->curpos += (toread * sizeof(float));*/
-    files[ifd]->curpos += got;  /* RWD 4:2002 */
-    files[ifd]->FramePos += nframes;
-
-    return rc;
-}
-
-int pvoc_rewind(int ifd,int skip_first_frame)
-{
-    int rc = -1;
-    int fd;
-    long pos;
-    long skipsize = 0;
-    long skipframes = 0;
-
-    if(files[ifd]==NULL){
-        pv_errstr = "\npvsys: bad file descriptor";
-        return rc;
-    }
-    if(files[ifd]->fd < 0){
-        pv_errstr = "\npvsys: file not open";
-        return rc;
-    }
-    skipsize =  files[ifd]->pvdata.dwFrameAlign * files[ifd]->fmtdata.nChannels;
-    skipframes = files[ifd]->fmtdata.nChannels;
-    
-    fd = files[ifd]->fd;
-    pos = files[ifd]->datachunkoffset;
-    if(skip_first_frame){
-        skipsize =  files[ifd]->pvdata.dwFrameAlign * skipframes;
-        pos += skipsize;
-    }
-    if(lseek(fd,pos,SEEK_SET) != pos ) {
-        pv_errstr = "\npvsys: error rewinding file";
-        return rc;
-    }
-    files[ifd]->curpos = files[ifd]->datachunkoffset + skipsize;
-    files[ifd]->FramePos = skipframes;
-
-    return 0;
-
-}
-
-/* may be more to do in here later on */
-int pvsys_release(void)
-{
-    int i;
-    
-
-    for(i=0;i < MAXFILES;i++){
-        if(files[i]) {
-#ifdef _DEBUG
-            fprintf(stderr,"\nDEBUG WARNING: files still open!\n");
-#endif
-            if(!pvoc_closefile(i)){             
-                pv_errstr = "\npvsys: unable to close file on termination";
-                return 0;
-            }
-        }
-    }
-    return 1;
-}
-
-/*return raw framecount:  channel-agnostic for now */
-int pvoc_framecount(int ifd)
-{
-    if(files[ifd]==NULL)
-        return -1;
-
-    return files[ifd]->nFrames;
-}
-/* RWD Jan 2014   */
-int pvoc_framepos(int ifd)
-{
-    if(files[ifd]==NULL)
-        return -1;
-        
-    return files[ifd]->FramePos;
-}
-
-int pvoc_seek_mcframe(int ifd, long offset, int mode)
-{
-    unsigned long mcframealign;
-    unsigned long newpos;
-    int rc = -1;
-    if(files[ifd]==NULL)
-        return -1;
-    mcframealign =  files[ifd]->pvdata.dwFrameAlign * files[ifd]->fmtdata.nChannels;
-    switch(mode){
-    case SEEK_SET:
-        newpos = mcframealign * offset;
-        if(offset >= files[ifd]->nFrames / files[ifd]->fmtdata.nChannels){
-            pv_errstr = "\npvsys: seek target beyond end of file";
-            break;
-        } 
-        newpos += files[ifd]->datachunkoffset;
-        if(lseek(files[ifd]->fd,newpos,SEEK_SET) != newpos ) {
-            pv_errstr = "\npvsys: seek error";
-            return -1;
-        }
-        files[ifd]->curpos = newpos;
-        files[ifd]->FramePos = offset * files[ifd]->fmtdata.nChannels;
-        rc = 0;
-        break;
-    case SEEK_END:
-    case SEEK_CUR:
-        pv_errstr = "\npvsys: seek mode not supported yet!";
-        break;
-    }
-    return rc;
-}

dev/externals/paprogs/pvplay/pvfileio.h

@@ -1,131 +0,0 @@
-/*pvfileio.h: header file for PVOC_EX file format */
-/* Initial Version 0.1 RWD 25:5:2000 all rights reserved: work in progress! */
-/* NB a special version of this file is used in Csound - do not auto-replace! */ 
-#ifndef __PVFILEIO_H_INCLUDED
-#define __PVFILEIO_H_INCLUDED
-
-//#ifndef WORD
-//#define WORD unsigned short
-//#endif
-//#ifndef DWORD
-//#define DWORD unsigned long
-//#endif
-#ifdef __cplusplus
-extern "C" {
-#endif
-
-
-#ifdef _WINDOWS
-#include <windows.h>
-#endif
-
-#ifndef _WINDOWS
-#ifndef WORD
-#define WORD unsigned short
-#endif
-#ifndef DWORD
-#define DWORD unsigned long
-#endif
-
-typedef struct _GUID 
-{ 
-    DWORD			Data1; 
-    WORD			Data2; 
-    WORD			Data3; 
-    unsigned char	Data4[8]; 
-} GUID;
-
-
-typedef struct /*waveformatex */{ 
-    WORD  wFormatTag; 
-    WORD  nChannels; 
-    DWORD nSamplesPerSec; 
-    DWORD nAvgBytesPerSec; 
-    WORD  nBlockAlign; 
-    WORD  wBitsPerSample; 
-    WORD  cbSize; 
-} WAVEFORMATEX; 
-#endif
-
-
-#include "pvdefs.h"
-
-
-/* Renderer information: source is presumed to be of this type */
-typedef enum pvoc_sampletype {STYPE_16,STYPE_24,STYPE_32,STYPE_IEEE_FLOAT} pv_stype;
-
-
-
-typedef struct pvoc_data {				/* 32 bytes*/
-	WORD	wWordFormat;				/* pvoc_wordformat */
-	WORD	wAnalFormat;				/* pvoc_frametype */
-	WORD	wSourceFormat;				/* WAVE_FORMAT_PCM or WAVE_FORMAT_IEEE_FLOAT*/
-	WORD	wWindowType;				/* pvoc_windowtype */
-	DWORD	nAnalysisBins;				/* implicit FFT size = (nAnalysisBins-1) * 2*/
-	DWORD	dwWinlen;					/* analysis winlen,samples, NB may be <> FFT size */
-	DWORD	dwOverlap;					/* samples */
-	DWORD	dwFrameAlign;				/* usually nAnalysisBins * 2 * sizeof(float) */
-	float	fAnalysisRate;
-	float	fWindowParam;				/* default 0.0f unless needed */
-} PVOCDATA;
-
-
-typedef struct {
-    WAVEFORMATEX    Format;				 /* 18 bytes:  info for renderer as well as for pvoc*/
-    union {								 /* 2 bytes*/
-        WORD wValidBitsPerSample;        /*  as per standard WAVE_EX: applies to renderer*/
-        WORD wSamplesPerBlock;          
-        WORD wReserved;                  
-                                        
-    } Samples;
-    DWORD           dwChannelMask;      /*  4 bytes : can be used as in standrad WAVE_EX */
-                                        
-    GUID            SubFormat;			/* 16 bytes*/
-} WAVEFORMATEXTENSIBLE, *PWAVEFORMATEXTENSIBLE;
-
-
-typedef struct {
-	WAVEFORMATEXTENSIBLE wxFormat;		 /* 40 bytes*/
-	DWORD dwVersion;					 /* 4 bytes*/
-	DWORD dwDataSize;					 /* 4 bytes: sizeof PVOCDATA data block */ 
-	PVOCDATA data;						 /* 32 bytes*/
-} WAVEFORMATPVOCEX;						 /* total 80 bytes*/
-
-
-/* at least VC++ will give 84 for sizeof(WAVEFORMATPVOCEX), so we need our own version*/
-#define SIZEOF_FMTPVOCEX (80)
-/* for the same reason:*/
-#define SIZEOF_WFMTEX (18)
-#define PVX_VERSION		(1)
-/******* the all-important PVOC GUID 
-
- {8312B9C2-2E6E-11d4-A824-DE5B96C3AB21}
-
-**************/
-
-extern 	const GUID KSDATAFORMAT_SUBTYPE_PVOC;
-
-
-/* pvoc file handling functions */
-
-const char *pvoc_errorstr();
-int init_pvsys(void);
-int  pvoc_createfile(const char *filename, 
-					 unsigned long fftlen,unsigned long overlap, unsigned long chans,
-					 unsigned long format,long srate, 
-					 pv_stype stype,pv_wtype wtype,float wparam,float *fWindow,DWORD dwWinlen);
-int pvoc_openfile(const char *filename,PVOCDATA *data,WAVEFORMATEX *fmt);	
-int pvoc_closefile(int ofd);
-int pvoc_putframes(int ofd,const float *frame,long numframes);
-int pvoc_getframes(int ifd,float *frames,unsigned long nframes);
-int pvoc_framecount(int ifd);
-int pvoc_rewind(int ifd,int skip_first_frame);		/* RWD 14:4:2001 */
-int pvoc_framepos(int ifd);							/* RWD Jan 2014 */
-int pvoc_seek_mcframe(int ifd, long offset, int mode);
-int pvsys_release(void);
-
-#ifdef __cplusplus
-}
-#endif
-
-#endif

dev/externals/paprogs/pvplay/pvoc.h

@@ -1,43 +0,0 @@
-/*
- * Copyright (c) 1983-2020  Composers Desktop Project Ltd
- * http://www.composersdesktop.com
- * This file is part of the CDP System.
- * The CDP System is free software; you can redistribute it
- * and/or modify it under the terms of the GNU Lesser General Public 
- * License as published by the Free Software Foundation; either 
- * version 2.1 of the License, or (at your option) any later version. 
- *
- * The CDP System is distributed in the hope that it will be useful, 
- * but WITHOUT ANY WARRANTY; without even the implied warranty of 
- * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  
- * See the GNU Lesser General Public License for more details.
- * You should have received a copy of the GNU Lesser General Public
- * License along with the CDP System; if not, write to the Free Software
- * Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
- *
- */
- 
-/* Functions in PVOC.C */
-
-void 	logo(void);
-void	warpse(float*,float*,int,double);
-void	usage(void);
-void	hamming(float*,int,int);
-float	*float_array(int);
-void	malerr(char*,int);
-void	kaiser_(int*,float*,int*,int*,float*);
-
-
-/* Functions in MXFFT.C */
-#ifndef USE_FFTW
-void fft_(float *, float *,int,int,int,int);
-void fftmx(float *,float *,int,int,int,int,int,int *,float *,float *,float *,float *,int *,int[]);
-void reals_(float *,float *,int,int);
-#endif
-
-#ifndef min
-#define min(a,b)	(((a)<(b)) ? (a) : (b))
-#endif
-#ifndef max
-#define max(a,b)	(((a)>(b)) ? (a) : (b))
-#endif

dev/externals/paprogs/pvplay/pvoc2.cpp

@@ -1,1541 +0,0 @@
-/*
- * Copyright (c) 1983-2020 Richard Dobson and Composers Desktop Project Ltd
- * http://www.rwdobson.com
- * http://www.composersdesktop.com
- * This file is part of the CDP System.
- * The CDP System is free software; you can redistribute it
- * and/or modify it under the terms of the GNU Lesser General Public 
- * License as published by the Free Software Foundation; either 
- * version 2.1 of the License, or (at your option) any later version. 
- *
- * The CDP System is distributed in the hope that it will be useful, 
- * but WITHOUT ANY WARRANTY; without even the implied warranty of 
- * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  
- * See the GNU Lesser General Public License for more details.
- * You should have received a copy of the GNU Lesser General Public
- * License along with the CDP System; if not, write to the Free Software
- * Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
- *
- */
-
-
-/*pvoc.cpp*/
-/*  simple class wrapper for (modified) CARL pvoc; supports realtime streaming. (c) Richard Dobson 2000,2014 */
-
-/* includes correction to sinc function code from Dan Timis, June 2000 */
-/* RWD 12:2000: define PVOC_NORM_PHASE to use slower normalized phase calcs */
-/* NB: optional FFTW code is for v 2.1.5 */
-
-#include <math.h>
-#include <stdlib.h>
-#include <memory.h>
-extern "C" {
-#ifdef USE_FFTW
-# include <rfftw.h>
-# else
-void fft_(float *, float *,int,int,int,int);
-void fftmx(float *,float *,int,int,int,int,int,int *,float *,float *,float *,float *,int *,int[]);
-void reals_(float *,float *,int,int);
-#endif
-}
-#include "pvpp.h"
-
-#ifndef PI
-#define PI (3.141592653589793)
-#endif
-#define TWOPI (2.0 * PI)
-
-#if defined _WIN32 && defined _MSC_VER
-#pragma message ("using assembler round()") 
-__inline static int round(double fval)
-{
-    int result;
-    _asm{
-        fld     fval
-        fistp   result
-        mov     eax,result
-    }
-    return result;
-}
-
-#else
-#define round(x) lround((x))
-#endif
-
-
-phasevocoder::phasevocoder()
-{
-
-    input       =  NULL;
-    output      = NULL;
-    anal        = NULL;
-    syn         =  NULL;        /* pointer to start of synthesis buffer */
-    banal       =  NULL;        /* pointer to anal[1] (for FFT calls) */
-    bsyn        =  NULL;        /* pointer to syn[1]  (for FFT calls) */
-    nextIn      =  NULL;    /* pointer to next empty word in input */
-    nextOut     =  NULL;    /* pointer to next empty word in output */
-    analWindow  =  NULL;    /* pointer to center of analysis window */
-    synWindow   =  NULL;    /* pointer to center of synthesis window */
-    maxAmp      =  NULL;    /* pointer to start of max amp buffer */
-    avgAmp      =  NULL;    /* pointer to start of avg amp buffer */
-    avgFrq      =  NULL;    /* pointer to start of avg frq buffer */
-    env     =  NULL;        /* pointer to start of spectral envelope */
-    i0      =  NULL;        /* pointer to amplitude channels */
-    i1      =  NULL;        /* pointer to frequency channels */
-    oi      =  NULL;        /* pointer to old phase channels */
-    oldInPhase      =  NULL;    /* pointer to start of input phase buffer */
-    oldOutPhase     =  NULL;    /* pointer to start of output phase buffer */
-    m = n = 0;
-
-    N = 0;      /* number of phase vocoder channels (bands) */
-    M = 0;      /* length of analWindow impulse response */
-    L = 0;      /* length of synWindow impulse response */
-    D = 0;      /* decimation factor (default will be M/8) */
-    I = 0;      /* interpolation factor (default will be I=D)*/
-    W = -1;     /* filter overlap factor (determines M, L) */
-    //F = 0;        /* fundamental frequency (determines N) */
-    //F2 = 0;       /* F/2 */
-    /*RWD */
-    Fexact = 0.0f;
-    analWinLen = 0, /* half-length of analysis window */
-    synWinLen = 0;  /* half-length of synthesis window */
-
-    sampsize = 0;   /* sample size for output file */
-    outCount = 0;   /* number of samples written to output */
-    ibuflen= 0; /* length of input buffer */
-    obuflen= 0; /* length of output buffer */
-    nI = 0;     /* current input (analysis) sample */
-    nO= 0;      /* current output (synthesis) sample */
-    nMaxOut= 0; /* last output (synthesis) sample */
-    nMin = 0;   /* first input (analysis) sample */
-    nMax = 100000000;   /* last input sample (unless EOF) */
-/***************************** 6:2:91  OLD CODE **************
-                        long    origsize;
-*******************************NEW CODE **********************/
-    origsize = 0;   /* sample type of file analysed */
-    ch = 0;
-    ifd =  ofd = -1;
-    beta = 6.8f;    /* parameter for Kaiser window */
-    real = 0.0f;        /* real part of analysis data */
-    imag= 0.0f;     /* imaginary part of analysis data */
-    mag= 0.0f;      /* magnitude of analysis data */
-    phase= 0.0f;        /* phase of analysis data */
-    angleDif= 0.0f; /* angle difference */
-    RoverTwoPi= 0.0f;   /* R/D divided by 2*Pi */
-    TwoPioverR= 0.0f;   /* 2*Pi divided by R/I */
-    sum= 0.0f;      /* scale factor for renormalizing windows */
-    ftot = 0.0f,    /* scale factor for calculating statistics */
-    rIn= 0.0f;      /* decimated sampling rate */
-    rOut= 0.0f;     /* pre-interpolated sampling rate */
-    invR= 0.0f;     /* 1. / srate */
-    time= 0.0f;     /* nI / srate */
-    tvx0 = 0.0f;
-    tvx1 = 0.0f;
-    tvdx = 0.0f;
-    tvy0 = tvy1 = 0.0f;
-    tvdy = 0.0f;
-    frac = 0.0f;
-    warp = 0.0f;    /* spectral envelope warp factor */
-    R = 0.0f;       /* input sampling rate */
-    P = 1.0f;       /* pitch scale factor */
-    Pinv= 0.0f;     /* 1. / P */
-    T = 1.0f;       /* time scale factor ( >1 to expand)*/
-    //Mlen,
-    Mf = 0;     /* flag for even M */
-    Lf = 0;     /* flag for even L */
-    //Dfac,
-    flag = 0;       /* end-of-input flag */
-    C = 0;      /* flag for resynthesizing even or odd chans */
-    Ci = 0;     /* flag for resynthesizing chans i to j only */
-    Cj = 0;     /* flag for resynthesizing chans i to j only */
-    CC = 0;     /* flag for selected channel resynthesis */
-    X = 0;      /* flag for magnitude output */
-    E = 0;      /* flag for spectral envelope output */
-    tvflg = 0;  /* flag for time-varying time-scaling */
-    tvnxt = 0;
-    tvlen = 0;
-    timecheckf= 0.0f;
-    K = H = 0;  /* default window is Hamming */
-    Nchans = 0;
-    NO2 = 0;
-    vH = 0;                     /* RWD set to 1 to set von Hann window */
-    bin_index = 0;
-    m_mode = PVPP_NOT_SET;
-    synWindow_base = NULL;
-    analWindow_base = NULL;
-};
-
-/* use when decfac needs specifying directly: cuts out other options */
-bool phasevocoder::init(long outsrate,long fftlen,long winlen,long decfac,float scalefac,
-                        pvoc_scaletype stype,pvoc_windowtype wtype,pvocmode mode)
-{       
-    N = fftlen;  
-    M = N*2;          /* RWD make double-window the default  */
-    D = decfac;
-    if(scalefac <=0.0)
-        return false;
-    
-    switch(stype){
-    case PVOC_S_TIME:
-        T = scalefac;
-        P = 1.0f;
-        break;
-    case PVOC_S_PITCH:
-        T = P = scalefac;
-        break;
-    default:
-        T = P = 1.0f;
-        break;
-    }
-    switch(wtype){
-    case PVOC_HANN:
-        H = 1;
-        break;
-    case PVOC_KAISER:
-        K = 1;
-        break;
-    default:
-        /* for now, anything else just sets Hamming window! */
-        break;
-    }
-    if(N <=0)
-        return false;
-    if(D < 0)
-        return false;
-    if(M < 0)
-        return false;
-
-    /*for now */
-    if(!(mode == PVPP_OFFLINE || mode == PVPP_STREAMING) )
-        return false;
-    m_mode  = mode;
-
-    isr = outsrate;
-    R       = srate = (float) outsrate;
-    N       = N  + N%2; /* Make N even */
-    N2      = N / 2;
-//  if (N2 > 16384){
-//      return false;
-//  }
-
-//  F       = (int)((float) R / N);
-    Fexact  = (float)R / (float)N;      /* RWD */
-//  F2      = F / 2;
-    if(winlen > 0)
-        M   = winlen;
-    Mf      = 1 - M%2;
-
-    L       =  (L != 0 ? L : M);
-    Lf      = 1 - L%2;
-    ibuflen = 4 * M;
-    obuflen = 4 * L;
-
-
-    if (W == -1)
-        W = (int)(3.322 * log10((double)(4. * N) / M));/* cosmetic */
-
-    if (Cj == 0)
-        Cj = N2;
-    if (Cj > N2)
-        Cj = N2;
-    if (Ci < 0)
-        Ci = 0;
-    D = (int)((D != 0 ? D : M/(8.0*(T > 1.0 ? T : 1.0))));
-
-    if (D == 0){
-        //fprintf(stderr,"pvoc: warning - T greater than M/8 \n");
-        D = 1;
-    }
-
-    I = (int)(I != 0 ? I : (float) T*D );
-
-    T = ((float) I / D);
-
-    if (P != 1.)
-        P = T;
-
-    NO = (int)((float) N / P);  /* synthesis transform will be NO points */
-    NO = NO + NO%2;             /* make NO even */
-
-    NO2 = NO / 2;
-
-    P = ((float) N / NO);       /* ideally, N / NO = I / D = pitch change */
-    Pinv = (float)(1.0/ P);
-
-    if (warp == -1.)
-        warp = P;
-    if ((E == 1) && (warp == 0.))
-        warp = 1.0f;
-
-
-    //if ((P != 1.) && (P != T))
-    //   fprintf(stderr,"pvoc: warning P=%f not equal to T=%f\n",P,T);
-
-    IO = (int)((float) I / P);
-
-    nMax -= nMin;
-
-    /*RWD need this to get sum setup for synth window! */
-    /* NB vonHann window also available now */
-
-    /* set up analysis window: The window is assumed to be symmetric
-        with M total points.  After the initial memory allocation,
-        analWindow always points to the midpoint of the window
-        (or one half sample to the right, if M is even); analWinLen
-        is half the true window length (rounded down). Any low pass
-        window will work; a Hamming window is generally fine,
-        but a Kaiser is also available.  If the window duration is
-        longer than the transform (M > N), then the window is
-        multiplied by a sin(x)/x function to meet the condition:
-        analWindow[Ni] = 0 for i != 0.  In either case, the
-        window is renormalized so that the phase vocoder amplitude
-        estimates are properly scaled.  The maximum allowable
-        window duration is ibuflen/2. */
-
-
-    analWindow_base = new float[M+Mf];
-    analWindow = analWindow_base + (analWinLen = M/2);
-#ifdef USE_FFTW 
-    in_fftw_size = N;
-    out_fftw_size = NO;
-    forward_plan = rfftwnd_create_plan_specific(1,&in_fftw_size, 
-        FFTW_REAL_TO_COMPLEX, FFTW_ESTIMATE | FFTW_IN_PLACE,
-        (fftw_real*) analWindow_base,1,NULL,1);
-    inverse_plan = rfftwnd_create_plan(1,&out_fftw_size, FFTW_COMPLEX_TO_REAL, FFTW_ESTIMATE | FFTW_IN_PLACE);
-#endif
-
-    if (K)  
-        kaiser(analWindow_base,M+Mf,beta);          /* ??? or just M?*/
-    else if (vH)
-        vonhann(analWindow,analWinLen,Mf);
-    else
-        hamming(analWindow,analWinLen,Mf);
-
-
-    for (i = 1; i <= analWinLen; i++)
-        *(analWindow - i) = *(analWindow + i - Mf);
-
-    if (M > N) {
-        if (Mf)
-        *analWindow *=(float)( (double)N * sin((double)PI*.5/N) /(double)( PI*.5));
-        for (i = 1; i <= analWinLen; i++) 
-            *(analWindow + i) *=(float)
-            ((double)N * sin((double) (PI*(i+.5*Mf)/N)) / (PI*(i+.5*Mf)));  /* D.T. 2000*/
-        for (i = 1; i <= analWinLen; i++)
-            *(analWindow - i) = *(analWindow + i - Mf);
-    }
-
-    sum = 0.0f;
-    for (i = -analWinLen; i <= analWinLen; i++)
-        sum += *(analWindow + i);
-
-    sum = (float)(2.0 / sum);       /*factor of 2 comes in later in trig identity*/
-
-    for (i = -analWinLen; i <= analWinLen; i++)
-        *(analWindow + i) *= sum;
-
-    /* set up synthesis window:  For the minimal mean-square-error
-        formulation (valid for N >= M), the synthesis window
-        is identical to the analysis window (except for a
-        scale factor), and both are even in length.  If N < M,
-        then an interpolating synthesis window is used. */
-
-    synWindow_base = new float[L+Lf];
-    synWindow = synWindow_base + (synWinLen = L/2);
-#ifdef USE_FFTW
-    Ninv = (float) (1.0 / N);
-#endif
-    if (M <= N){
-        
-        if(K)
-            kaiser(synWindow_base,L+Lf,beta);
-        else if(vH)
-            vonhann(synWindow,synWinLen,Lf);
-        else
-            hamming(synWindow,synWinLen,Lf);
-        for (i = 1; i <= synWinLen; i++)
-            *(synWindow - i) = *(synWindow + i - Lf);
-
-        for (i = -synWinLen; i <= synWinLen; i++)
-            *(synWindow + i) *= sum;
-
-        sum = 0.0f;
-        for (i = -synWinLen; i <= synWinLen; i+=I)
-            sum += *(synWindow + i) * *(synWindow + i);
-
-        sum = (float)(1.0/ sum);
-#ifdef USE_FFTW
-        sum *= Ninv;
-#endif
-        for (i = -synWinLen; i <= synWinLen; i++)
-            *(synWindow + i) *= sum;
-    }
-    else {
-        if(K)
-            kaiser(synWindow_base,L+Lf,beta);
-        else if(vH)
-            vonhann(synWindow,synWinLen,Lf);
-        else
-            hamming(synWindow,synWinLen,Lf);
-        for (i = 1; i <= synWinLen; i++)
-            *(synWindow - i) = *(synWindow + i - Lf);
-
-        if (Lf)
-            *synWindow *= (float)((double)IO * sin((double) (PI*.5/IO)) / (double)(PI*.5));
-        for (i = 1; i <= synWinLen; i++) 
-                *(synWindow + i) *=(float)
-                ((double)IO * sin((double) (PI*(i+.5*Lf)/IO)) /(double) (PI*(i+.5*Lf)));
-        for (i = 1; i <= synWinLen; i++)
-            *(synWindow - i) = *(synWindow + i - Lf);
-
-        sum = (float)(1.0/sum);
-#ifdef USE_FFTW
-        sum *= Ninv;
-#endif
-        for (i = -synWinLen; i <= synWinLen; i++)
-            *(synWindow + i) *= sum;
-    }
-
-
-    
-    try{
-
-        /* set up input buffer:  nextIn always points to the next empty
-        word in the input buffer (i.e., the sample following
-        sample number (n + analWinLen)).  If the buffer is full,
-        then nextIn jumps back to the beginning, and the old
-        values are written over. */
-
-        input = new float[ibuflen];
-
-        nextIn = input;
-
-        /* set up output buffer:  nextOut always points to the next word
-        to be shifted out.  The shift is simulated by writing the
-        value to the standard output and then setting that word
-        of the buffer to zero.  When nextOut reaches the end of
-        the buffer, it jumps back to the beginning.  */
-        output =    new float [obuflen];
-
-        nextOut =   output;
-
-
-        /* set up analysis buffer for (N/2 + 1) channels: The input is real,
-        so the other channels are redundant. oldInPhase is used
-        in the conversion to remember the previous phase when
-        calculating phase difference between successive samples. */
-
-        anal    =   new float[N+2];
-        banal   =   anal + 1;
-
-        oldInPhase =    new float[N2+1];
-        maxAmp =    new float[N2+1];
-        avgAmp =    new float[N2+1];
-        avgFrq =    new float[N2+1];
-        env =       new float[N2+1];
-
-        /* set up synthesis buffer for (N/2 + 1) channels: (This is included
-        only for clarity.)  oldOutPhase is used in the re-
-        conversion to accumulate angle differences (actually angle
-        difference per second). */
-
-        syn =       new float[NO+2];
-        bsyn = syn + 1;
-        oldOutPhase =   new float[NO2+1];
-    }
-    catch(...){
-        if(synWindow_base){
-            delete [] synWindow_base;
-            synWindow_base = 0;
-        }
-        if(analWindow_base){
-            delete [] analWindow_base;
-            analWindow_base = 0;
-
-        }
-        if(input) {
-            delete [] input;
-            input = 0;
-        }
-
-        if(output) {
-            delete [] output;
-            output = 0;
-        }
-        if(anal) {
-            delete [] anal;
-            anal = 0;
-        }
-        if(oldInPhase) {
-            delete [] oldInPhase;
-            oldInPhase = 0;
-        }
-        if(maxAmp){
-            delete [] maxAmp;
-            maxAmp = 0;
-        }
-        if(avgAmp) {
-            delete [] avgAmp;
-            avgAmp = 0;
-        }
-        if(avgFrq) {
-            delete [] avgFrq;
-            avgFrq = 0;
-        }
-        if(env){
-            delete [] env;
-            env= 0;
-        }
-        if(syn){
-            delete [] syn;
-            syn = 0;
-        }
-        if(oldOutPhase){
-            delete [] oldOutPhase;
-            oldOutPhase = 0;
-        }
-        return false;
-    }
-
-    outCount = 0;
-    rIn = ((float) R / D);
-    rOut = ((float) R / I);
-    invR =((float) 1. / R);
-    RoverTwoPi = (float)(rIn / TWOPI);
-    TwoPioverR = (float)(TWOPI / rOut);
-    nI = -(analWinLen / D) * D; /* input time (in samples) */
-    nO = (long)((float) T/P * nI);  /* output time (in samples) */
-    Dd = analWinLen + nI + 1;   /* number of new inputs to read */
-    Ii = 0;             /* number of new outputs to write */
-    IOi = 0;
-    flag = 1;
-
-    for(i=0;i < ibuflen;i++) {
-        input[i] = 0.0f;
-        output[i] = 0.0f;
-    }
-    for(i=0;i < NO+2;i++)
-        syn[i] = 0.0f;
-    for(i=0;i < N+2;i++)
-        anal[i] = 0.0f;
-    for(i=0;i < NO2+1;i++)
-        oldOutPhase[i] = 0.0f;
-    for(i=0;i < N2+1;i++)
-        oldInPhase[i] = maxAmp[i] = avgAmp[i] = avgFrq[i] = env[i] = 0.0f;
-    return true;
-}
-
-/* closer to PVOC command-line format; we will need a full array of settings ere long */
-/* e.g to install a custom window...*/
-bool phasevocoder::init(long outsrate,long fftlen,pvoc_overlapfac ofac,float scalefac,
-                        pvoc_scaletype stype,pvoc_windowtype wtype,pvocmode mode)
-{
-    N   = fftlen;
-    D   = 0;
-    //D = N/4;      /* one problem - when M is larger, overlap is larger too */
-    switch(ofac){   
-    case PVOC_O_W0:
-        W = 0;
-        break;
-    case PVOC_O_W1:
-        W = 1;
-        break;
-    case PVOC_O_W2:
-        W = 2;
-        break;
-    case PVOC_O_W3:
-        W = 3;
-        break;
-    default:
-        W = 1;       /* double-window option */
-        break;
-    }
-    switch(wtype){
-    case PVOC_HANN:
-        H = 1;      
-        break;
-    case PVOC_KAISER:
-        K = 1;      
-        break;
-    default:
-        /* for now, anything else just sets Hamming window! */
-        break;
-    }
-
-
-    if(scalefac <=0.0)
-        return false;
-    switch(stype){
-    case PVOC_S_TIME:
-        T = scalefac;
-        P = 1.0f;
-        break;
-    case PVOC_S_PITCH:
-        T = P = scalefac;
-        break;
-    default:
-        T = P = 1.0f;
-        break;
-    }
-
-
-    if(N <=0)
-        return false;
-    if(D < 0)
-        return false;
-    /*for now */
-    if(!(mode == PVPP_OFFLINE || mode == PVPP_STREAMING) )
-        return false;
-    m_mode  = mode;
-
-    isr = outsrate;
-    R       = srate = (float) outsrate;
-    N       = N  + N%2; /* Make N even */
-    N2      = N / 2;
-//  if (N2 > 16384){
-//      return false;
-//  }
-
-//  F       = (int)((float) R / N);
-    Fexact  = (float)R / (float)N;      /* RWD */
-//  F2      = F / 2;
-    if (W != -1)
-        /* cannot be used with M; but we don't touch M anyway here */
-        switch(W){
-        case 0: M = 4*N;
-            break;
-        case 1: M = 2*N;        /* RWD: we want this one, most of the time */
-            break;
-        case 2: M = N;
-            break;
-        case 3: M = N2;
-            break;
-        default:
-            break;
-        }
-
-    M       = (M != 0 ? M : N*2);      /* RWD double-window as default */
-    Mf      = 1 - M%2;
-
-    L       =  M;
-    Lf      = 1 - L%2;
-    ibuflen = 4 * M;
-    obuflen = 4 * L;
-
-    if (W == -1)
-        W = (int)(3.322 * log10((double)(4. * N) / M));/* cosmetic */
-    
-    if (Cj == 0)
-        Cj = N2;
-    if (Cj > N2)
-        Cj = N2;
-    if (Ci < 0)
-        Ci = 0;
-    D = (int)((D != 0 ? D : M/(8.0*(T > 1.0 ? T : 1.0))));
-
-    if (D == 0){
-        //fprintf(stderr,"pvoc: warning - T greater than M/8 \n");
-        D = 1;
-    }
-
-    I = (int)(I != 0 ? I : (float) T*D );
-
-    T = ((float) I / D);
-
-    if (P != 1.)
-        P = T;
-
-    NO = (int)((float) N / P);  /* synthesis transform will be NO points */
-    NO = NO + NO%2;             /* make NO even */
-
-    NO2 = NO / 2;
-
-    P = ((float) N / NO);       /* ideally, N / NO = I / D = pitch change */
-    Pinv = (float)(1.0/ P);
-
-    if (warp == -1.)
-        warp = P;
-    if ((E == 1) && (warp == 0.))
-        warp = 1.0f;
-
-
-    //if ((P != 1.) && (P != T))
-    //   fprintf(stderr,"pvoc: warning P=%f not equal to T=%f\n",P,T);
-
-    IO = (int)((float) I / P);
-
-    nMax -= nMin;
-    /*RWD need this to get sum setup for synth window! */
-    /* NB vonHann window also available now */
-
-    /* set up analysis window: The window is assumed to be symmetric
-        with M total points.  After the initial memory allocation,
-        analWindow always points to the midpoint of the window
-        (or one half sample to the right, if M is even); analWinLen
-        is half the true window length (rounded down). Any low pass
-        window will work; a Hamming window is generally fine,
-        but a Kaiser is also available.  If the window duration is
-        longer than the transform (M > N), then the window is
-        multiplied by a sin(x)/x function to meet the condition:
-        analWindow[Ni] = 0 for i != 0.  In either case, the
-        window is renormalized so that the phase vocoder amplitude
-        estimates are properly scaled.  The maximum allowable
-        window duration is ibuflen/2. */
-
-
-    analWindow_base = new float[M+Mf];
-    analWindow = analWindow_base + (analWinLen = M/2);
-
-
-    if (K)  
-        kaiser(analWindow_base,M+Mf,beta);          /* ??? or just M?*/
-    else if (H)
-        vonhann(analWindow,analWinLen,Mf);
-    else
-        hamming(analWindow,analWinLen,Mf);
-
-    for (i = 1; i <= analWinLen; i++)
-        *(analWindow - i) = *(analWindow + i - Mf);
-
-    if (M > N) {
-        if (Mf)
-        *analWindow *=(float)( (double)N * sin((double)PI*.5/N) /(double)( PI*.5));
-        for (i = 1; i <= analWinLen; i++) 
-            *(analWindow + i) *=(float)
-            ((double)N * sin((double) (PI*(i+.5*Mf)/N)) / (PI*(i+.5*Mf)));  /* D.T 2000*/
-        for (i = 1; i <= analWinLen; i++)
-            *(analWindow - i) = *(analWindow + i - Mf);
-    }
-
-    sum = 0.0f;
-    for (i = -analWinLen; i <= analWinLen; i++)
-        sum += *(analWindow + i);
-
-    sum = (float)(2.0 / sum);       /*factor of 2 comes in later in trig identity*/
-
-    for (i = -analWinLen; i <= analWinLen; i++)
-        *(analWindow + i) *= sum;
-
-    /* set up synthesis window:  For the minimal mean-square-error
-        formulation (valid for N >= M), the synthesis window
-        is identical to the analysis window (except for a
-        scale factor), and both are even in length.  If N < M,
-        then an interpolating synthesis window is used. */
-
-    synWindow_base = new float[L+Lf];
-    synWindow = synWindow_base + (synWinLen = L/2);
-#ifdef USE_FFTW
-    Ninv = (float) (1.0 / N);
-#endif
-
-    if (M <= N){
-        if(K)
-            kaiser(synWindow_base,L+Lf,beta);
-        else if(H)
-            vonhann(synWindow,synWinLen,Lf);
-        else
-            hamming(synWindow,synWinLen,Lf);
-
-        for (i = 1; i <= synWinLen; i++)
-            *(synWindow - i) = *(synWindow + i - Lf);
-
-        for (i = -synWinLen; i <= synWinLen; i++)
-            *(synWindow + i) *= sum;
-
-        sum = 0.0f;
-        /* RWD ??? this jumps by overlap samps thru table*/
-        for (i = -synWinLen; i <= synWinLen; i+=I)
-            sum += *(synWindow + i) * *(synWindow + i);
-
-        sum = (float)(1.0/ sum);
-#ifdef USE_FFTW
-        sum *= Ninv;
-#endif
-        for (i = -synWinLen; i <= synWinLen; i++)
-            *(synWindow + i) *= sum;
-    }
-    else {
-        if(K)
-            kaiser(synWindow_base,L+Lf,beta);
-        else if(H)
-            vonhann(synWindow,synWinLen,Lf);
-        else
-            hamming(synWindow,synWinLen,Lf);
-        for (i = 1; i <= synWinLen; i++)
-            *(synWindow - i) = *(synWindow + i - Lf);
-
-        if (Lf)
-            *synWindow *= (float)((double)IO * sin((double) (PI*.5/IO)) / (double)(PI*.5));
-        for (i = 1; i <= synWinLen; i++) 
-                *(synWindow + i) *=(float)
-                ((double)IO * sin((double) (PI*(i+.5*Lf)/IO)) /(double) (PI*(i+.5*Lf)));
-        for (i = 1; i <= synWinLen; i++)
-            *(synWindow - i) = *(synWindow + i - Lf);
-
-        sum = (float)(1.0/sum);
-#ifdef USE_FFTW
-        sum *= Ninv;
-#endif
-        for (i = -synWinLen; i <= synWinLen; i++)
-            *(synWindow + i) *= sum;
-    }
-
-#ifdef USE_FFTW 
-    in_fftw_size = N;
-    out_fftw_size = NO;
-    forward_plan = rfftwnd_create_plan_specific(1,&in_fftw_size, 
-        FFTW_REAL_TO_COMPLEX, FFTW_ESTIMATE | FFTW_IN_PLACE,
-        (fftw_real*) analWindow_base,1,NULL,1);
-    inverse_plan = rfftwnd_create_plan_specific(1,&out_fftw_size, 
-        FFTW_COMPLEX_TO_REAL, FFTW_ESTIMATE | FFTW_IN_PLACE,
-        (fftw_real*) synWindow_base,1,NULL,1);
-
-#endif
-
-    try{
-
-        /* set up input buffer:  nextIn always points to the next empty
-        word in the input buffer (i.e., the sample following
-        sample number (n + analWinLen)).  If the buffer is full,
-        then nextIn jumps back to the beginning, and the old
-        values are written over. */
-
-        input = new float[ibuflen];
-
-        nextIn = input;
-
-        /* set up output buffer:  nextOut always points to the next word
-        to be shifted out.  The shift is simulated by writing the
-        value to the standard output and then setting that word
-        of the buffer to zero.  When nextOut reaches the end of
-        the buffer, it jumps back to the beginning.  */
-        output =    new float [obuflen];
-
-        nextOut =   output;
-
-
-        /* set up analysis buffer for (N/2 + 1) channels: The input is real,
-        so the other channels are redundant. oldInPhase is used
-        in the conversion to remember the previous phase when
-        calculating phase difference between successive samples. */
-
-        anal    =   new float[N+2];
-        banal   =   anal + 1;
-
-        oldInPhase =    new float[N2+1];
-        maxAmp =    new float[N2+1];
-        avgAmp =    new float[N2+1];
-        avgFrq =    new float[N2+1];
-        env =       new float[N2+1];
-
-        /* set up synthesis buffer for (N/2 + 1) channels: (This is included
-        only for clarity.)  oldOutPhase is used in the re-
-        conversion to accumulate angle differences (actually angle
-        difference per second). */
-
-        syn =       new float[NO+2];
-        bsyn = syn + 1;
-        oldOutPhase =   new float[NO2+1];
-    }
-    catch(...){
-        if(synWindow_base){
-            delete [] synWindow_base;
-            synWindow_base = 0;
-        }
-        if(analWindow_base){
-            delete [] analWindow_base;
-            analWindow_base = 0;
-
-        }
-        if(input) {
-            delete [] input;
-            input = 0;
-        }
-
-        if(output) {
-            delete [] output;
-            output = 0;
-        }
-        if(anal) {
-            delete [] anal;
-            anal = 0;
-        }
-        if(oldInPhase) {
-            delete [] oldInPhase;
-            oldInPhase = 0;
-        }
-        if(maxAmp){
-            delete [] maxAmp;
-            maxAmp = 0;
-        }
-        if(avgAmp) {
-            delete [] avgAmp;
-            avgAmp = 0;
-        }
-        if(avgFrq) {
-            delete [] avgFrq;
-            avgFrq = 0;
-        }
-        if(env){
-            delete [] env;
-            env= 0;
-        }
-        if(syn){
-            delete [] syn;
-            syn = 0;
-        }
-        if(oldOutPhase){
-            delete [] oldOutPhase;
-            oldOutPhase = 0;
-        }
-        return false;
-    }
-
-    outCount = 0;
-    rIn = ((float) R / D);
-    rOut = ((float) R / I);
-    invR =((float) 1. / R);
-    RoverTwoPi = (float)(rIn / TWOPI);
-    TwoPioverR = (float)(TWOPI / rOut);
-    nI = -(analWinLen / D) * D; /* input time (in samples) */
-    nO = (long)((float) T/P * nI);  /* output time (in samples) */
-    Dd = analWinLen + nI + 1;   /* number of new inputs to read */
-    Ii = 0;             /* number of new outputs to write */
-    IOi = 0;
-    flag = 1;
-
-    for(i=0;i < ibuflen;i++) {
-        input[i] = 0.0f;
-        output[i] = 0.0f;
-    }
-    for(i=0;i < NO+2;i++)
-        syn[i] = 0.0f;
-    for(i=0;i < N+2;i++)
-        anal[i] = 0.0f;
-    for(i=0;i < NO2+1;i++)
-        oldOutPhase[i] = 0.0f;
-    for(i=0;i < N2+1;i++)
-        oldInPhase[i] = maxAmp[i] = avgAmp[i] = avgFrq[i] = env[i] = 0.0f;
-    return true;
-}
-
-
-
-
-
-phasevocoder::~phasevocoder()
-{
-    if(synWindow_base)
-        delete [] synWindow_base;
-    if(analWindow_base)
-        delete [] analWindow_base;
-    if(input)
-        delete [] input;
-    if(output)
-        delete [] output;
-    if(anal)
-        delete [] anal;
-    if(oldInPhase)
-        delete [] oldInPhase;
-    if(maxAmp)
-        delete [] maxAmp;
-    if(avgAmp)
-        delete [] avgAmp;
-    if(avgFrq)
-        delete [] avgFrq;
-    if(env)
-        delete [] env;
-    if(syn)
-        delete [] syn;  
-    if(oldOutPhase)
-        delete [] oldOutPhase;
-
-#ifdef USE_FFTW
-    rfftwnd_destroy_plan(forward_plan);
-    rfftwnd_destroy_plan(inverse_plan);
-#endif
-
-}
-
-void phasevocoder::hamming(float *win,int winLen,int even)
-{
-    double Pi,ftmp;
-    int i;
-
-/***********************************************************
-                    Pi = (double)((double)4.*atan((double)1.));
-***********************************************************/
-    Pi = (double)PI;
-    ftmp = Pi/winLen;
-
-    if (even) {
-        for (i=0; i<winLen; i++)
-        *(win+i) = (float)(.54 + .46*cos(ftmp*((double)i+.5)));
-        *(win+winLen) = 0.0;}
-    else{   
-        *(win) = 1.0;
-        for (i=1; i<=winLen; i++)
-        *(win+i) = (float)(.54 + .46*cos(ftmp*(double)i));
-    }
-
-}
-
-
-double phasevocoder::besseli( double x)
-{
-    double ax, ans;
-    double y;
-
-    if (( ax = fabs( x)) < 3.75)     {
-    y = x / 3.75;
-    y *= y;
-    ans = (1.0 + y * ( 3.5156229 +
-              y * ( 3.0899424 +
-                y * ( 1.2067492 +
-                      y * ( 0.2659732 +
-                        y * ( 0.360768e-1 +
-                          y * 0.45813e-2))))));
-    }
-    else {
-    y = 3.75 / ax;
-    ans = ((exp ( ax) / sqrt(ax))
-        * (0.39894228 +
-           y * (0.1328592e-1 +
-            y * (0.225319e-2 +
-             y * (-0.157565e-2 +
-                  y * (0.916281e-2 +
-                   y * (-0.2057706e-1 +
-                    y * (0.2635537e-1 +
-                         y * (-0.1647633e-1 +
-                          y * 0.392377e-2)))))))));
-    }
-    return ans;
-}
-
-//courtesy of Csound....
-
-void phasevocoder::kaiser(float *win,int len,double Beta)
-{
-    float *ft = win;
-    double i,xarg = 1.0;    //xarg = amp scalefactor
-    for (i = -len/2.0 + .1 ; i < len/2.0 ; i++)
-        *ft++ = (float) (xarg *
-              besseli((Beta * sqrt(1.0-pow((2.0*i/(len - 1)), 2.0)))) /
-              besseli( Beta));
-   // assymetrical hack: sort out first value!
-   win[0] = win[len-1];
-}
-
-void phasevocoder::vonhann(float *win,int winLen,int even)
-{
-    float Pi,ftmp;
-    int i;
-
-    Pi = (float)PI;
-    ftmp = Pi/winLen;
-
-    if (even) {
-        for (i=0; i<winLen; i++)
-        *(win+i) = (float)(.5 + .5 *cos(ftmp*((double)i+.5)));
-        *(win+winLen) = 0.0f;
-    }
-    else{   *(win) = 1.0f;
-        for (i=1; i<=winLen; i++)
-        *(win+i) =(float)(.5 + .5 *cos(ftmp*(double)i));
-    }   
-}
-
-
-
-long phasevocoder::process_frame(float *anal,float *outbuf,pvoc_frametype frametype)
-{
-
-    /*RWD vars */
-    int n;
-    long written;
-    float *obufptr;
-
-    /* reconversion: The magnitude and angle-difference-per-second in syn
-        (assuming an intermediate sampling rate of rOut) are
-        converted to real and imaginary values and are returned in syn.
-        This automatically incorporates the proper phase scaling for
-        time modifications. */
-    
-    if (NO <= N){
-        for (i = 0; i < NO+2; i++)
-            *(syn+i) = *(anal+i);
-    }
-    else {
-        for (i = 0; i <= N+1; i++)
-            *(syn+i) = *(anal+i);
-        for (i = N+2; i < NO+2; i++)
-            *(syn+i) = 0.0f;
-    }
-
-    if(frametype==PVOC_AMP_PHASE){
-        for(i=0, i0=syn, i1=syn+1; i<= NO2; i++, i0+=2,  i1+=2){
-            mag = *i0;
-            phase = *i1;
-            *i0 = (float)((double)mag * cos((double)phase));
-            *i1 = (float)((double)mag * sin((double)phase));
-        }
-    }
-    else if(frametype == PVOC_AMP_FREQ){
-        for(i=0, i0=syn, i1=syn+1; i<= NO2; i++, i0+=2,  i1+=2){
-            mag = *i0;
-            /* keep phase wrapped within +- TWOPI */
-            /* this could possibly be spread across several frame cycles, as the problem does not
-                develop for a while */
-            double angledif, the_phase;
-            angledif = TwoPioverR * (*i1  - ((float) i * /*F*/Fexact));
-            the_phase = *(oldOutPhase + i) +angledif;
-            if(i== bin_index)
-                the_phase = (float) fmod(the_phase,TWOPI);
-            *(oldOutPhase + i) = (float) the_phase;
-            phase = (float) the_phase;
-
-
-            *i0 = (float)((double)mag * cos((double)phase));
-            *i1 = (float)((double)mag * sin((double)phase));
-        }
-    }
-#ifdef PVOC_NORM_PHASE
-    if(++bin_index == NO2+1)
-        bin_index = 0;
-#endif
-
-    /* else it must be PVOC_COMPLEX */
-    if (P != 1.)
-        for (i = 0; i < NO+2; i++)
-            *(syn+i) *= Pinv;
-    
-    /* synthesis: The synthesis subroutine uses the Weighted Overlap-Add
-            technique to reconstruct the time-domain signal.  The (N/2 + 1)
-            phase vocoder channel outputs at time n are inverse Fourier
-            transformed, windowed, and added into the output array.  The
-            subroutine thinks of output as a shift register in which
-            locations are referenced modulo obuflen.  Therefore, the main
-            program must take care to zero each location which it "shifts"
-            out (to standard output). The subroutines reals and fft
-            together perform an efficient inverse FFT.  */
-
-
-# ifdef USE_FFTW
-    rfftwnd_one_complex_to_real(inverse_plan,(fftw_complex * )syn,NULL);
-# else
-    reals_(syn,bsyn,NO2,2);
-    fft_(syn,bsyn,1,NO2,1,2);
-#endif
-    j = nO - synWinLen - 1;
-    while (j < 0)
-        j += obuflen;
-    j = j % obuflen;
-
-    k = nO - synWinLen - 1;
-    while (k < 0)
-        k += NO;
-    k = k % NO;
-
-    for (i = -synWinLen; i <= synWinLen; i++) { /*overlap-add*/
-        if (++j >= obuflen)
-            j -= obuflen;
-        if (++k >= NO)
-            k -= NO;
-        *(output + j) += *(syn + k) * *(synWindow + i);
-    }
-
-    obufptr = outbuf;   /*RWD */
-    written = 0;
-    for (i = 0; i < IOi;){  /* shift out next IOi values */
-        int j;
-        int todo = MIN(IOi-i, output+obuflen-nextOut);
-        /*outfloats(nextOut, todo, ofd);*/
-        /*copy data to external buffer */
-        for(n=0;n < todo;n++)
-            *obufptr++ = nextOut[n];
-        written += todo;
-
-        i += todo;
-        outCount += todo;
-        for(j = 0; j < todo; j++)
-            *nextOut++ = 0.0f;
-        if (nextOut >= (output + obuflen))
-            nextOut -= obuflen;
-    }
-
-    if (flag)   /* flag means do this operation only once */
-        if ((nI > 0) && (Dd < D)){  /* EOF detected */
-            flag = 0;
-            nMax = nI + analWinLen - (D - Dd);
-        }
-
-
-    /*  D = some_function(nI);      for variable time-scaling */
-    /*  rIn = ((float) R / D);      for variable time-scaling */
-    /*  RoverTwoPi =  rIn / TwoPi;  for variable time-scaling */
-
-    nI += D;                /* increment time */
-    nO += IO;
-
-    /* Dd = D except when the end of the sample stream intervenes */
-    /* RWD handle offline and streaming separately - 
-        can't count an infinite number of real-time samples! */
-    if(m_mode == PVPP_OFFLINE)
-        Dd = MIN(D, MAX(0L, D+nMax-nI-analWinLen));   /*  CARL */
-    else
-        Dd = D;                                       /* RWD */
-
-    if (nO > (synWinLen + I))
-        Ii = I;
-    else
-        if (nO > synWinLen)
-            Ii = nO - synWinLen;
-        else {
-            Ii = 0;
-            for (i=nO+synWinLen; i<obuflen; i++)
-                if (i > 0)
-                    *(output+i) = 0.0f;
-        }
-    IOi = (int)((float) Ii / P);
-
-
-
-    return written;
-
-}
-
-
-/*RWD arrgh! */
-long phasevocoder::process_frame(float *anal,short *outbuf,pvoc_frametype frametype)
-{
-
-    /*RWD vars */
-    int n;
-    long written;
-    short *obufptr;
-
-        /* reconversion: The magnitude and angle-difference-per-second in syn
-        (assuming an intermediate sampling rate of rOut) are
-        converted to real and imaginary values and are returned in syn.
-        This automatically incorporates the proper phase scaling for
-        time modifications. */
-
-    if (NO <= N){
-        for (i = 0; i < NO+2; i++)
-            *(syn+i) = *(anal+i);
-    }
-    else {
-        for (i = 0; i <= N+1; i++) 
-            *(syn+i) = *(anal+i);
-        for (i = N+2; i < NO+2; i++) 
-            *(syn+i) = 0.0f;
-    }
-    if(frametype != PVOC_COMPLEX){      /* assume AMP_FREQ otherwise, for now */
-        for(i=0, i0=syn, i1=syn+1; i<= NO2; i++, i0+=2,  i1+=2){
-            mag = *i0;
-            *(oldOutPhase + i) += *i1 - ((float) i * /*F*/ Fexact);
-            phase = *(oldOutPhase + i) * TwoPioverR;
-            *i0 = (float)((double)mag * cos((double)phase));
-            *i1 = (float)((double)mag * sin((double)phase));
-        }
-    }
-    if (P != 1.)
-        for (i = 0; i < NO+2; i++)
-            *(syn+i) *= Pinv;
-
-    /* synthesis: The synthesis subroutine uses the Weighted Overlap-Add
-            technique to reconstruct the time-domain signal.  The (N/2 + 1)
-            phase vocoder channel outputs at time n are inverse Fourier
-            transformed, windowed, and added into the output array.  The
-            subroutine thinks of output as a shift register in which 
-            locations are referenced modulo obuflen.  Therefore, the main
-            program must take care to zero each location which it "shifts"
-            out (to standard output). The subroutines reals and fft
-            together perform an efficient inverse FFT.  */
-
-
-#ifdef USE_FFTW
-    rfftwnd_one_complex_to_real(inverse_plan,(fftw_complex * )syn,NULL);
-#else
-    reals_(syn,bsyn,NO2,2);
-    fft_(syn,bsyn,1,NO2,1,2);
-#endif
-    j = nO - synWinLen - 1;
-    while (j < 0)
-        j += obuflen;
-    j = j % obuflen;
-
-    k = nO - synWinLen - 1;
-    while (k < 0)
-        k += NO;
-    k = k % NO;
-
-    for (i = -synWinLen; i <= synWinLen; i++) { /*overlap-add*/
-        if (++j >= obuflen)
-            j -= obuflen;
-        if (++k >= NO)
-            k -= NO;
-        *(output + j) += *(syn + k) * *(synWindow + i);
-    }
-
-    obufptr = outbuf;   /*RWD */
-    written = 0;
-    for (i = 0; i < IOi;){  /* shift out next IOi values */
-        int j;
-        int todo = MIN(IOi-i, output+obuflen-nextOut);
-        
-        /*copy data to external buffer */
-        for(n=0;n < todo;n++)
-            *obufptr++ = (short) round(32767.0 * nextOut[n]);
-        written += todo;
-
-        i += todo;
-        outCount += todo;
-        for(j = 0; j < todo; j++)
-            *nextOut++ = 0.0f;
-        if (nextOut >= (output + obuflen))
-            nextOut -= obuflen;
-    }
-                
-    if (flag)   /* flag means do this operation only once */
-        if ((nI > 0) && (Dd < D)){  /* EOF detected */
-            flag = 0;
-            nMax = nI + analWinLen - (D - Dd);
-        }
-
-
-    /*  D = some_function(nI);      for variable time-scaling */
-    /*  rIn = ((float) R / D);      for variable time-scaling */
-    /*  RoverTwoPi =  rIn / TwoPi;  for variable time-scaling */
-
-    nI += D;                /* increment time */
-    nO += IO;
-
-    /* Dd = D except when the end of the sample stream intervenes */
-
-    Dd = MIN(D, MAX(0L, D+nMax-nI-analWinLen));
-
-
-    if (nO > (synWinLen + I))
-        Ii = I;
-    else
-        if (nO > synWinLen)
-            Ii = nO - synWinLen;
-        else {
-            Ii = 0;
-            for (i=nO+synWinLen; i<obuflen; i++)
-                if (i > 0)
-                    *(output+i) = 0.0f;
-        }
-    IOi = (int)((float) Ii / P);
-
-
-
-    return written;
-
-}
-/* trying to get clean playback when looping! */
-void phasevocoder::reset_phases(void)
-{
-    int i;
-    if(oldInPhase){
-        for(i=0;i < N2+1;i++)
-            oldInPhase[i] = 0.0f;
-    }
-    if(oldOutPhase){
-        for(i=0;i < NO2+1;i++)
-            oldOutPhase[i] = 0.0f;
-    }
-    if(syn){
-        for(i=0;i < NO+2;i++)
-            syn[i] = 0.0f;
-    }
-    
-    
-#ifdef _DEBUG
-    //printf("\nnI = %d; nO = %d; j = %d; k = %d\n",nI,nO,j,k);
-#endif
-
-    /* these seem definitely necessary, but don't solve everything... */
-    nI = -(analWinLen / D) * D; /* input time (in samples) */
-    nO = (long)((float) T/P * nI);  /* output time (in samples) */
-
-}
-
-
-/* we don't read in a single sample, a la pvoc, but just output an empty first frame*/
-/* we assume final block zero-padded if necessary */
-long phasevocoder::generate_frame(float *fbuf,float *outanal,long samps,pvoc_frametype frametype)
-{
-    
-    /*sblen = decfac = D */
-    //static int sblen = 0;
-    int got, tocp;
-    float *fp,*ofp;
-
-    got = samps;     /*always assume */
-    if(got < Dd)
-        Dd = got;
-
-    fp = fbuf;
-    
-
-    tocp = MIN(got, input+ibuflen-nextIn);
-    got -= tocp;
-    while(tocp-- > 0)
-        *nextIn++ = *fp++; 
-
-    if(got > 0) {
-        nextIn -= ibuflen;
-        while(got-- > 0)
-            *nextIn++ = *fp++;
-    }
-    if (nextIn >= (input + ibuflen))
-        nextIn -= ibuflen;
-
-    if (nI > 0)
-        for (i = Dd; i < D; i++){   /* zero fill at EOF */
-            *(nextIn++) = 0.0f;
-            if (nextIn >= (input + ibuflen))
-                nextIn -= ibuflen;
-        }
-
-    /* analysis: The analysis subroutine computes the complex output at
-        time n of (N/2 + 1) of the phase vocoder channels.  It operates
-        on input samples (n - analWinLen) thru (n + analWinLen) and
-        expects to find these in input[(n +- analWinLen) mod ibuflen].
-        It expects analWindow to point to the center of a
-        symmetric window of length (2 * analWinLen +1).  It is the
-        responsibility of the main program to ensure that these values
-        are correct!  The results are returned in anal as succesive
-        pairs of real and imaginary values for the lowest (N/2 + 1)
-        channels.   The subroutines fft and reals together implement
-        one efficient FFT call for a real input sequence.  */
-
-
-    for (i = 0; i < N+2; i++)
-        *(anal + i) = 0.0f; /*initialize*/
-
-    j = (nI - analWinLen - 1 + ibuflen) % ibuflen;  /*input pntr*/
-
-    k = nI - analWinLen - 1;            /*time shift*/
-    while (k < 0)
-        k += N;
-    k = k % N;
-    for (i = -analWinLen; i <= analWinLen; i++) {
-        if (++j >= ibuflen)
-            j -= ibuflen;
-        if (++k >= N)
-            k -= N;
-        *(anal + k) += *(analWindow + i) * *(input + j);
-    }
-#ifdef USE_FFTW
-    rfftwnd_one_real_to_complex(forward_plan,(fftw_real*) anal,NULL);
-            //reals_(anal,banal,N2,-2);
-#else
-    fft_(anal,banal,1,N2,1,-2);
-    reals_(anal,banal,N2,-2);
-#endif
-    /* conversion: The real and imaginary values in anal are converted to
-        magnitude and angle-difference-per-second (assuming an 
-        intermediate sampling rate of rIn) and are returned in
-        anal. */
-    if(frametype == PVOC_AMP_PHASE){
-        /* PVOCEX uses plain (wrapped) phase format, ref Soundhack */
-        for(i=0,i0=anal,i1=anal+1,oi=oldInPhase; i <= N2; i++,i0+=2,i1+=2, oi++){
-            real = *i0;
-            imag = *i1;
-            *i0 =(float) sqrt((double)(real * real + imag * imag));
-            /* phase unwrapping */
-            /*if (*i0 == 0.)*/
-            if(*i0 < 1.0E-10f)
-                //angleDif = 0.0f;
-                phase = 0.0f;
-
-            else {
-#ifdef OLDCALC
-                rratio = atan((double)imag/(double)real);
-                if(real<0.0) {
-                    if(imag<0.0)
-                        rratio -= PI;
-                    else
-                        rratio += PI;
-                }
-#else
-                rratio = atan2((double)imag,(double)real);
-#endif
-                /*angleDif  = (phase = (float)rratio) - *oi;
-                *oi = phase;
-                */
-                phase = (float)rratio;
-            }
-
-            *i1 = phase;
-        }
-    }
-    if(frametype==PVOC_AMP_FREQ){
-        for(i=0,i0=anal,i1=anal+1,oi=oldInPhase; i <= N2; i++,i0+=2,i1+=2, oi++){
-            real = *i0;
-            imag = *i1;
-            *i0 =(float) sqrt((double)(real * real + imag * imag));
-            /* phase unwrapping */
-            /*if (*i0 == 0.)*/
-            if(*i0 < 1.0E-10f)
-                angleDif = 0.0f;
-
-            else {
-#ifdef OLDCALC
-/*RWD 12.99: slower with Pentium Pro build, but otherwise faster??!?? */
-                rratio = atan((double)imag/(double)real);
-                if(real<0.0) {
-                    if(imag<0.0)
-                        rratio -= PI;
-                    else
-                        rratio += PI;
-                }
-#else
-/*RWD98*/       rratio = atan2((double)imag,(double)real);
-#endif
-                angleDif  = (phase = (float)rratio) - *oi;
-                *oi = phase;
-            }
-            if (angleDif > PI)
-                angleDif = (float)(angleDif - TWOPI);
-            if (angleDif < -PI)
-                angleDif = (float)(angleDif + TWOPI);
-
-            /* add in filter center freq.*/
-            *i1 = angleDif * RoverTwoPi + ((float) i * /*F*/Fexact);
-        }
-    }
-    /* else must be PVOC_COMPLEX */
-    fp = anal;
-    ofp = outanal;
-    for(i=0;i < N+2;i++)
-        *ofp++ = *fp++;
-
-    nI += D;                /* increment time */
-    nO += IO;
-    /* deal with offline and streaming differently */
-    if(m_mode== PVPP_OFFLINE)
-        Dd = MIN(D, MAX(0L, D+nMax-nI-analWinLen));     /* CARL */
-    else
-        Dd = D;                         /* RWD */
-
-    if (nO > (synWinLen + I))
-        Ii = I;
-    else
-        if (nO > synWinLen)
-            Ii = nO - synWinLen;
-        else {
-            Ii = 0;
-            for (i=nO+synWinLen; i<obuflen; i++)
-                if (i > 0)
-                    *(output+i) = 0.0f;
-        }
-    IOi = (int)((float) Ii / P);
-
-
-    return D;
-}
-

dev/externals/paprogs/pvplay/pvplay.cpp

@@ -1,1435 +0,0 @@
-/*
- * Copyright (c) 1983-2020 Richard Dobson and Composers Desktop Project Ltd
- * http://www.rwdobson.com
- * http://www.composersdesktop.com
- * This file is part of the CDP System.
- * The CDP System is free software; you can redistribute it
- * and/or modify it under the terms of the GNU Lesser General Public 
- * License as published by the Free Software Foundation; either 
- * version 2.1 of the License, or (at your option) any later version. 
- *
- * The CDP System is distributed in the hope that it will be useful, 
- * but WITHOUT ANY WARRANTY; without even the implied warranty of 
- * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  
- * See the GNU Lesser General Public License for more details.
- * You should have received a copy of the GNU Lesser General Public
- * License along with the CDP System; if not, write to the Free Software
- * Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
- *
- */
-
-/* pvplay.cpp
- * Play a soundfile or analysis file using PortAudio
- *
- * Yes yes, I know, this is iffy C++, and there are some gotos to boot. Work in progress...
- *  and I may yet redo the whole thing in plain C anyway, like paplay.
- */ 
-/* OCT 2009 rebuilt with updated pvfileio.c for faster performance on G4 */
-/* Feb 2010 rebuilt with new stable portaudio */
-/* June 2012  new portaudio, settable audio block size */
-/* Jan 2013 changed (as also paplay), at last, to use threaded portaudio ringbuffer code (etc) */
-/* Jan 2014: completed pvx support */
-
-#include "pvplay.h"
-
-typedef enum  {PLAY_SFILE,PLAY_ANAFILE,PLAY_PVXFILE} PLAYTYPE;
-enum {FM_MONO,FM_STEREO,FM_SQUARE,FM_QUAD,FM_PENT,DM_5_0,DM_5_1,FM_HEX,FM_OCT1,FM_OCT2,FM_CUBE,FM_QUADCUBE,FM_NLAYOUTS};
-
-#define NUM_ANALFRAMES (1)
-#define RINGBUF_NFRAMES    (32768)
-#define NUM_WRITES_PER_BUFFER   (4)
-
-#ifndef min
-#define min(x,y)  ((x) < (y) ? (x) : (y))
-#endif
-
-#ifdef unix
-#include <ctype.h>
-int stricmp(const char *a, const char *b);
-int strnicmp(const char *a, const char *b, const int length);
-#endif
-
-#ifdef WIN32
-HANDLE ghEvent;
-#endif
-
-int file_playing;
-psfdata *g_pdata = NULL;
-
-void playhandler(int sig)
-{
-    if(sig == SIGINT) {
-        if(file_playing) {
-            file_playing = 0;
-            g_pdata->flag = 1;
-            Pa_Sleep(5);
-        }
-    }
-}
-
-#ifdef unix
-void alarm_wakeup (int i)
-{
-    struct itimerval tout_val;
-    
-    signal(SIGALRM,alarm_wakeup);
-    if(file_playing && g_pdata->stream) {
-        //printf("%.4f secs\r",(float)(g_pdata->frames_played /(double) g_pdata->srate));
-        g_pdata->lastTime = Pa_GetStreamTime(g_pdata->stream ) - g_pdata->startTime;
-        printf("%.2f secs\r", g_pdata->lastTime); 
-        fflush(stdout);
-    }
-    tout_val.it_interval.tv_sec = 0;
-    tout_val.it_interval.tv_usec = 0;
-    tout_val.it_value.tv_sec = 0; 
-    tout_val.it_value.tv_usec = 200000; 
-    setitimer(ITIMER_REAL, &tout_val,0); 
-}
-#endif
-
-void finishedCallback(void *userData)
-{
-    psfdata *pdata = (psfdata*) userData;
-    //printf("stream finished!\n");
-    pdata->finished = 1;
-    file_playing = 0;
-}
-
-#ifdef WIN32
-VOID CALLBACK TimerCallback(PVOID lpParam, BOOLEAN TimerOrWaitFired)
-{
-    psfdata *pdata = (psfdata*) lpParam;
-    
-    if(file_playing && pdata->stream) {
-        //printf("%.4f secs\r",(float)(g_pdata->frames_played /(double) g_pdata->srate));
-        pdata->lastTime = Pa_GetStreamTime(pdata->stream ) - pdata->startTime;
-        printf("%.2f secs\r", pdata->lastTime); 
-        fflush(stdout);
-    }
-    SetEvent(ghEvent);
-}
-#endif
-
-/* This routine will be called by the PortAudio engine when audio is needed.
-** It may called at interrupt level on some machines so don't do anything
-** that could mess up the system like calling malloc() or free().
-*/
-
-#ifdef NOTDEF
-// need this?
-static void porttimeCallback(PtTimestamp timestamp, void *userData)
-{
-    PtTimestamp curtime =  Pt_Time();
-    double timesecs = (double) curtime * 0.001;
-    printf("%.3lf\r",timesecs);
-}
-
-#endif
-
-/* Start up a new thread for given function */
-static PaError startThread( psfdata* pdata, threadfunc fn )
-{
-    pdata->flag = 1;
-#ifdef WIN32
-    pdata->hThread = (void*)_beginthreadex(NULL, 0, fn, pdata, 0, NULL);
-    if (pdata->hThread == NULL) 
-        return paUnanticipatedHostError;
-    /* Wait for thread to startup */
-    while (pdata->flag) {
-        Pa_Sleep(10);
-    }
-    /* Set file thread to a little higher priority than normal */
-    SetThreadPriority(pdata->hThread, THREAD_PRIORITY_ABOVE_NORMAL);
-//#endif
-    
-#else
-#if defined(__APPLE__) || defined(__GNUC__)
-    if(pthread_create(&pdata->hThread,NULL,/*(void*)*/ fn,pdata))
-        return -1;
-    /* Wait for thread to startup */
-    while (pdata->flag) {
-        Pa_Sleep(10);
-    } 
-#endif
-#endif
-    
-    return paNoError;
-}
-#if 0
-static int stopThread( psfdata* pdata )
-{
-    // just called when all data played; must be called before StopStream
-    //   pdata->flag = 1;
-    /* Wait for thread to stop */
-       while (pdata->flag==0) {
-           Pa_Sleep(10);
-       }
-#ifdef WIN32
-    CloseHandle(pdata->hThread);
-    pdata->hThread = 0;
-#else
-#if defined(__APPLE__) || defined(__GNUC__)
-    pthread_cancel(pdata->hThread);
-#endif
-#endif
-    return paNoError;
-}
-#endif
-
-static int paplayCallback(const void *inputBuffer, void *outputBuffer,
-                          unsigned long framesPerBuffer,
-                          const PaStreamCallbackTimeInfo* timeInfo,
-                          PaStreamCallbackFlags statusFlags,
-                          void *userData )
-{
-    psfdata *data = (psfdata *)userData;
-    ring_buffer_size_t sampsAvailable = PaUtil_GetRingBufferReadAvailable(&data->ringbuf) * data->outchans;
-    ring_buffer_size_t sampsToPlay = min(sampsAvailable, (ring_buffer_size_t)(framesPerBuffer * data->outchans));
-    float *out = (float*) outputBuffer;
-    int framesToPlay = sampsToPlay / data->outchans;
-    int played;
-    
-    // if framestoplay < framesPerBuffer, need to clean up the remainder
-    memset(out,0,framesPerBuffer * data->outchans * sizeof(float));
-    played = PaUtil_ReadRingBuffer(&data->ringbuf, out, framesToPlay);
-    
-    data->frames_played += played;      
-    if(data->flag) {
-        //printf("callback - complete\n");
-        return paComplete;
-    }
-    return paContinue;
-}
-
-static int MemCallback(const    void *inputBuffer, void *outputBuffer,
-                       unsigned long framesPerBuffer,
-                       const PaStreamCallbackTimeInfo* timeInfo,
-                       PaStreamCallbackFlags statusFlags,
-                       void *userData )
-{
-    psfdata *pdata = (psfdata *)userData;
-    
-    float *out = (float*) outputBuffer;
-    unsigned long inSamps,outSamps, inSampPos;
-    unsigned long i;
-    unsigned long framesToPlay = framesPerBuffer;
-    
-    // if framesToPlay < framesPerBuffer, need to clean up the remainder
-    memset(out,0,framesPerBuffer * pdata->outchans * sizeof(float));
-    inSamps = pdata->memFramelength * pdata->inchans;
-    outSamps = framesPerBuffer * pdata->inchans;
-    inSampPos = pdata->memFramePos * pdata->inchans;
-    
-    if(pdata->play_looped){
-        for(i=0;i < outSamps; i++){
-            pdata->inbuf[i] = pdata->membuf[inSampPos++] * pdata->gain;
-            if(inSampPos == inSamps)
-                inSampPos = 0;
-        }
-    }
-    else {
-        for(i=0;i < outSamps; i++){
-            pdata->inbuf[i] = pdata->membuf[inSampPos++] * pdata->gain;
-            if(inSampPos == inSamps)
-                break;
-        }
-        framesToPlay = i / pdata->inchans;
-        for(; i < outSamps;i++){
-            pdata->inbuf[i] = 0.0;
-        }
-        
-    }
-    // write to output
-    // everything now in inbuf
-    if(pdata->do_decode) {
-        ABFSAMPLE abfsamp;
-        int j;    
-        for(j=0; j < framesToPlay; j++){
-            pdata->copyfunc(&abfsamp,pdata->inbuf + (j * pdata->inchans));
-            pdata->decodefunc(&abfsamp,out + (j * pdata->outchans), 1);
-        }
-    }
-    else {  // inchans = outchans
-        memcpy(out,pdata->inbuf,framesToPlay * sizeof(float) * pdata->inchans);  
-    }
-    pdata->memFramePos = inSampPos / pdata->inchans;
-    pdata->frames_played += framesToPlay;
-    if(framesToPlay < framesPerBuffer)
-        pdata->flag = 1;           // end of stream
-    if(pdata->flag) {
-        return paComplete;
-    }
-    return paContinue;
-}
-
-static unsigned NextPowerOf2(unsigned val)
-{
-    val--;
-    val = (val >> 1) | val;
-    val = (val >> 2) | val;
-    val = (val >> 4) | val;
-    val = (val >> 8) | val;
-    val = (val >> 16) | val;
-    return ++val;
-}
-
-#define N_BFORMATS (10)
-// static const int bformats[N_BFORMATS] = {2,3,4,5,6,7,8,9,11,16};
-// static const int layout_chans[] = {1,2,4,4,5,5,6,6,8,8,8,8};
-/* no M flag here */
-enum {FLAG_B = 0, FLAG_BM, FLAG_D, FLAG_G, FLAG_H, FLAG_I, FLAG_L, FLAG_NFLAGS};
-
-void usage(void){
-#ifdef WIN32
-    printf("usage:\n  pvplay [-BN][-dN][-gN]-hN][-i][-l][-b[N]][-u][-x] infile [from] [to] \n"
-#else
-    printf("usage:\n  pvplay [-BN][-dN][-gN]-hN][-i][-l][-b[N]][-u] infile [from] [to]\n"
-#endif
-                  "       -dN  : use output Device N.\n"
-                  "       -gN  : apply gain factor N to input.\n"
-                  "       -BN  : set memory buffer size to N frames (default: %d).\n"
-                  "                N must be a power of 2 (e.g 4096, 8192, 16384 etc).\n"
-                  "       -hN  : set hardware blocksize to N frames (32 < N <= BN/4), default is set internally.\n"
-                  "               (N recommended to be a power of two size).\n"
-                  "            : NB: for sample rates > 48KHz, buffer sizes are doubled internally.\n"
-                  "       -i   : play immediately (do not wait for keypress).\n"
-                  "       -l   : loop file continuously, from start-time to end-time.\n"
-                  "       -u   : suppress elapsed time updates\n"
-#ifdef WIN32
-                  "       -x   : Apply WAVE_EX infile channel mask to DS audio stream\n"
-                  "              (ignored if -m or -b used)\n"
-#endif
-                  "       from : set start time (secs) for playback and looping. Default: 0.\n"
-                  "       to   : set end time (secs) for playback and looping. Default: EOF.\n"
-                  "       -b[N]: apply 1st-order B-Format decoding to standard soundfile.\n"
-                  "              (file must have at least 3 channels)\n"
-                  "               B-Format files (.amb) will be decoded automatically.\n"
-                  "               N sets speaker layout to one of the following:\n"
-                  "               1    :  *  mono (= W signal only)\n"
-                  "               2    :  *  stereo (quasi mid/side, = W +- Y)\n"
-                  "               3    :     square\n"
-                  "               4    :  *  quad FL,FR,RL,RR order   (default)\n"
-                  "               5    :     pentagon\n"
-                  "               6    :  *  5.0 surround (WAVEX order)\n"
-                  "               7    :  *  5.1 surround (WAVEX order, silent LFE)\n"
-                  "               8    :     hexagon\n"
-                  "               9    :     octagon 1 (front pair, 45deg)\n"
-                  "              10    :     octagon 2 (front centre speaker)\n"
-                  "              11    :     cube (as 3, low-high interleaved. Csound-compatible.)\n"
-                  "              12    :  *  cube (as 4, low quad followed by high quad).\n"
-                  "              Default decode layout is 4 (quad).\n"
-                  "              NB: no decoding is performed if -m flag used.\n"
-                  "  pvplay reads PEAK chunk if present to rescale over-range floatsam files.\n\n"
-                  ,RINGBUF_NFRAMES);
-}
-           
-
-/*******************************************************************/
-int main(int argc,char **argv)
-{
-    PaStreamParameters outputParameters;
-#ifdef WIN32
-    /* portaudio sets default channel masks we can't use; we must do all this to set default mask = 0! */
-    PaWinDirectSoundStreamInfo directSoundStreamInfo;
-//    PaWinMmeStreamInfo winmmeStreamInfo;
-#endif
-    PaDeviceInfo *devinfo = NULL;
-    PaStream *stream = NULL;
-    PaStreamCallback *playcallback = paplayCallback;
-    PaError err = paNoError;
-    SFPROPS props; 
-    psfdata sfdata;
-    CHPEAK *fpeaks = NULL;
-    //ABFSAMPLE *abf_frame = NULL;
-    int res,inorder = 1;
-    time_t in_peaktime = 0;
-    int waitkey = 1;
-    //int play_looped = 0;
-    double fromdur = 0.0;
-    double totime  = 0.0;
-    int from_frame = 0, to_frame = 0;
-    unsigned int i=0;
-    long anal_nframes;
-    PaDeviceIndex device;
-    //long framesread;
-    long filesize;
-    //long framesize = 0;
-    unsigned long ringframelen = RINGBUF_NFRAMES;   // length of ring buffer in m/c frames
-    unsigned long frames_per_buffer = 0;            // default is to let portaudio decide
-    unsigned long nFramesToPlay = 0;
-    double gain = 1.0;
-    unsigned int inchans = 0,outchans = 0;          // may be different if decoding B-Format
-    int layout = 4;                                 // default, quad decode */
-    fmhcopyfunc copyfunc = NULL;
-    fmhdecodefunc decodefunc = NULL;
-    /* chorder facility not included in pvplay - use paplay! */
-    unsigned int flags[FLAG_NFLAGS] = {0};
-    int speakermask = 0;
-    int do_speakermask = 0;
-    int do_updatemessages = 1;
-#ifdef unix
-    struct itimerval tout_val;    
-    tout_val.it_interval.tv_sec = 0;
-    tout_val.it_interval.tv_usec = 0;
-    tout_val.it_value.tv_sec = 0; 
-    tout_val.it_value.tv_usec = 200000;
-#endif
-    char* ext = 0;
-    PVOCDATA *p_pvdata = NULL;
-    WAVEFORMATEX *p_wfx = NULL;
-    //int pvfile = -1;
-    //pvoc_frametype inframetype;
-    pvoc_windowtype wtype;
-    
-    //double oneovrsr;
-    int anal_buflen,overlap,winlen;
-    // for sfiles only, for now 
-    //int framesize_factor = 0;
-    //phasevocoder  *pv = NULL, *pv_r = NULL;
-    PLAYTYPE playtype = PLAY_SFILE;
-    
-    
-#ifdef WIN32
-    HANDLE hTimerQueue;
-    ghEvent = CreateEvent(NULL,TRUE,FALSE,NULL);
-    if(ghEvent==NULL){
-        printf("Failed to start internal timer (1).\n");
-        return 1;
-    }
-    hTimerQueue = CreateTimerQueue();
-    if(hTimerQueue == NULL){
-        printf("Failed to start internal timer (2).\n");
-        return 1;
-    }
-#endif
-    
-    
-    /* CDP version number; now set for release 7 */
-    if(argc==2 && (stricmp(argv[1],"--version")==0)){
-        printf("7.1.0\n");
-        return 0;
-    }
-    
-    signal(SIGINT,playhandler);
-    //sf_frames = FRAMESIZE;  // need this for pvoc playback reporting
-    sfdata.inbuf = NULL;
-    sfdata.membuf = NULL;
-    sfdata.memFramelength = 0;
-    sfdata.memFramePos = 0;
-    sfdata.current_frame = 0;
-    sfdata.ringbufData = NULL;
-    sfdata.outbuf_l = NULL;
-    sfdata.outbuf_r = NULL;
-    sfdata.pvfile = -1;
-    
-    printf("PvPlay: play multi-channel PCM and analysis files (.ana, .pvx). V7.0.0 RWD,CDP 2014\n");
-    file_playing = 0;
-    err = Pa_Initialize();
-    if( err != paNoError ) {
-        fprintf(stderr,"Portaudio startup error.\n");
-        return 1;
-    }
-    device =  Pa_GetDefaultOutputDevice();
-    if(device < 0){
-        fprintf(stderr,"Cannot find any audio output device\n");
-        return 1;
-    }
-
-    if(argc < 2){
-        usage();
-        show_devices();
-        Pa_Terminate();
-            return 1;
-    }
-
-    while(argv[1][0]=='-'){
-        unsigned long userbuflen;
-        switch(argv[1][1]){
-            case('b'):
-                if(flags[FLAG_B]){
-                    printf("error: multiple -b flags\n");
-                    Pa_Terminate();
-                    return 1;
-                }
-                if(argv[1][2] != '\0'){
-                    layout = atoi(&argv[1][2]);
-                    if(layout < 1 || layout > 12){
-                        fprintf(stderr,"value for -b flag out of range.\n");
-                        Pa_Terminate();
-                        return 1;
-                    }
-                }
-                flags[FLAG_B] = 1;
-                break;
-            case('d'):
-                if(flags[FLAG_D]){
-                    printf("error: multiple -d flags\n");
-                    Pa_Terminate();
-                    return 1;
-                }
-                if(argv[1][2]=='\0'){
-                    printf("-d flag requires parameter\n");
-                    Pa_Terminate();
-                    return 1;
-                }
-                device = atoi(&(argv[1][2]));
-                flags[FLAG_D] = 1;
-                break;
-            case('g'):
-                if(flags[FLAG_G]){
-                    printf("error: multiple -g flags\n");
-                    Pa_Terminate();
-                    return 1;
-                }
-                if(argv[1][2]=='\0'){
-                    printf("-g flag requires parameter\n");
-                    Pa_Terminate();
-                    return 1;
-                }
-                gain = atof(&(argv[1][2]));
-                if(gain <= 0.0){
-                    fprintf(stderr,"gain value must be positive.\n");
-                    Pa_Terminate();
-                    return 1;
-                }
-                flags[FLAG_G] = 1;
-                break;
-            case 'h':
-                if(flags[FLAG_H]){
-                    printf("error: multiple -h flags\n");
-                    Pa_Terminate();
-                    return 1;
-                }
-                if(argv[1][2]=='\0'){
-                    printf("-h flag requires parameter\n");
-                    Pa_Terminate();
-                    return 1;
-                }
-                frames_per_buffer = atoi(&(argv[1][2]));
-                if((frames_per_buffer > 0) && (frames_per_buffer < 32)){
-                    printf("-h value too small: must be at least 32.\n");
-                    Pa_Terminate();
-                    return 1;
-                }
-                flags[FLAG_H] = 1;
-                break;
-            case('i'):
-                if(flags[FLAG_I]){
-                    printf("error: multiple -i flags\n");
-                    Pa_Terminate();
-                    return 1;
-                }
-                flags[FLAG_I] = 1;
-                waitkey = 0;
-                break;
-            case ('l'):
-                flags[FLAG_L] = 1;
-                break;
-            case 'u':
-                do_updatemessages = 0;
-                break;
-            case 'x':
-                do_speakermask = 1;
-                break;
-            case 'B':
-                if(flags[FLAG_BM]){
-                    printf("error: multiple -B flags\n");
-                    Pa_Terminate();
-                    return 1;
-                }
-                if(argv[1][2]=='\0'){
-                    printf("-B flag requires parameter\n");
-                    Pa_Terminate();
-                    return 1;
-                }
-                
-                ringframelen = atoi(&(argv[1][2]));
-                if(ringframelen <= 1024){
-                    printf("-B: framesize value must be >=1024!n");
-                    Pa_Terminate();
-                    return 1;
-                }
-                userbuflen = NextPowerOf2(ringframelen);
-                if(userbuflen != ringframelen){
-                    printf("-B: framesize value must be power of 2 size\n");
-                    Pa_Terminate();
-                    return 1;
-                }
-                flags[FLAG_BM] = 1;
-                break;
-            default:
-                printf("unrecognised flag option\n");
-                Pa_Terminate();
-                return 1;
-        }
-        argv++;  argc--;
-    }
-    
-    if(frames_per_buffer > ringframelen/4){
-        printf(" hardware (-h) framesize %lu too large for file buffer (-B) size %lu\n",
-               frames_per_buffer,ringframelen);
-        Pa_Terminate();
-        return 1;
-    }
-    
-    if(argc < 2 || argc > 4){
-        usage();
-        show_devices();
-        Pa_Terminate();
-        return 1;
-    }
-    
-    if(argc>=3){        
-        fromdur = atof(argv[2]);
-        if(fromdur < 0.0){
-            printf("Error: start position must be positive\n");
-            Pa_Terminate();
-            return 1;
-        }
-    }
-    if(argc==4){
-        totime = atof(argv[3]);
-        if(totime <= fromdur){
-            printf("Error: end time must be after from time\n");
-            Pa_Terminate();
-            return 1;
-        }
-    }
-    
-    if(!init_pvsys()){
-        puts("\nUnable to start pvsys.");
-        return 1;
-    }
-    if(sflinit("pvplay")){
-        puts("\nUnable to start sfsys.");
-        return 1;
-    }   
-    
-    /* get pvocex filetype from extension */
-    ext = strrchr(argv[1],'.');   
-    if(ext && stricmp(ext,".pvx")==0){
-        p_pvdata =  new PVOCDATA;
-        if(p_pvdata==NULL){
-            puts("\nNo Memory!");
-            return 1;
-        }
-        p_wfx =  new WAVEFORMATEX;
-        if(p_wfx==NULL){
-            puts("\nNo Memory!");
-            return 1;
-        }
-
-        sfdata.pvfile   = pvoc_openfile(argv[1],p_pvdata,p_wfx);
-        if(sfdata.pvfile< 0){
-            fprintf(stderr,"\npvplay: unable to open pvx file: %s",pvoc_errorstr());
-            return 1;
-        }
-        props.srate = p_wfx->nSamplesPerSec;
-        if(props.srate <=0){
-            fprintf(stderr,"\nbad srate found: corrupted file?\n");
-            return 1;
-        }
-        sfdata.srate = props.srate;
-        /*stick to mono/stereo for now */
-        if(p_wfx->nChannels > 2){
-            fprintf(stderr,"\nSorry: can only read mono or stereo pvx files.");
-            return 1;
-        }
-        inchans = p_wfx->nChannels;
-        props.chans = inchans;
-        outchans = inchans;
-        /* calc size in m/c analysis frames of this analysis file */
-        filesize = pvoc_framecount(sfdata.pvfile) / inchans;
-        double dur = (double) filesize  / (double)(p_pvdata->fAnalysisRate);
-        filesize = (long)(dur * props.srate);
-
-        sfdata.anal_framesize = (p_pvdata->nAnalysisBins/*- 1*/) * 2; 
-        sfdata.fftsize = sfdata.anal_framesize-2;
-        winlen = p_pvdata->dwWinlen;
-        overlap = p_pvdata->dwOverlap;
-        sfdata.inframetype = (pvoc_frametype) p_pvdata->wAnalFormat;
-        
-//        oneovrsr = 1.0 / (double) props.srate;
-        wtype = (pvoc_windowtype) p_pvdata->wWindowType;        
-        
-        /* adjust buffer size to multiple of overlap   */
-        anal_buflen = sfdata.anal_framesize * NUM_ANALFRAMES;
-        anal_buflen = (anal_buflen / overlap) * overlap;
-        /* mono */
-        sfdata.analframe1 = new float[sfdata.anal_framesize];
-        memset(sfdata.analframe1,0,sfdata.anal_framesize * sizeof(float));
-
-        sfdata.pv_l = new phasevocoder();
-        if(!sfdata.pv_l->init(props.srate,sfdata.anal_framesize-2,winlen,overlap,1.0,PVOC_S_NONE,wtype,PVPP_STREAMING)){
-            fprintf(stderr,"\nUnable to initialize pvoc Channel 1.");
-            //delete sfdata.pv_l;
-            //delete sfdata.analframe1;
-            return 1;
-        }
-        if(inchans==2){
-            /* need two intermediate buffers, to be interleaved into sampbuf */
-            sfdata.analframe2 = new float[sfdata.anal_framesize];
-            memset(sfdata.analframe2,0,sfdata.anal_framesize * sizeof(float));
-            sfdata.pv_r = new phasevocoder();
-            if(!sfdata.pv_r->init(props.srate,sfdata.anal_framesize-2,winlen,overlap,1.0,PVOC_S_NONE,wtype,PVPP_STREAMING)){
-                fprintf(stderr,"\nUnable to initialize pvoc Channel 2.");
-                //delete sfdata.pv_l;
-                //delete sfdata.pv_r;
-                //delete sfdata.outbuf_l;
-                //delete sfdata.outbuf_r;
-                //delete sfdata.analframe1;
-                //delete sfdata.analframe2;
-                return 1;
-            }
-        }
-        to_frame = (pvoc_framecount(sfdata.pvfile) / inchans) - 1;
-        if(fromdur > 0.0){
-            //printf("SORRY: from parameter ignored: not yet supported for pvx files\n");
-            from_frame = (long)(fromdur * (props.srate/overlap));
-            if(from_frame >= to_frame){
-                printf("Error: start frame %d is beyond end of file\n", from_frame);
-                goto error;
-            }
-            if(pvoc_seek_mcframe(sfdata.pvfile,from_frame,SEEK_SET)){
-                printf("Error: from: failed to seek to frame %d\n",from_frame);
-                goto error;
-            }
-        } 
-                
-        if(totime > 0.0){
-            long targetframe = (long)(totime * (props.srate/overlap));
-            if(targetframe >= to_frame){
-                printf("Error: end time is beyond end of file\n");
-                goto error;
-            }
-                    
-            if(targetframe <= from_frame){
-                printf("Start time must be less than end time.\n");
-                goto error;
-            }
-            to_frame = targetframe;       
-            if(to_frame - from_frame < 1){
-                to_frame = from_frame+1;   // maybe we can in fact freeze on a single frame!
-            // but we will not use in-memory system in this case
-            }
-            nFramesToPlay = to_frame - from_frame;
-                    
-            printf("Playing %d analysis frames\n",(int) nFramesToPlay);
-        } 
-        playtype = PLAY_PVXFILE; 
-    }
-
-    /* soundfile or anafile*/
-
-    else {
-        /* allow auto rescaling of overrange floats via PEAK chunk */
-        sfdata.ifd = sndopenEx(argv[1],1,CDP_OPEN_RDONLY);
-        if(sfdata.ifd < 0) {
-            printf("unable to open soundfile %s : %s\n",argv[1],sferrstr());
-           return 1;
-        }
-        if(!snd_headread(sfdata.ifd,&props)) {
-            fprintf(stderr,"\nError reading sfile header");
-            fprintf(stderr,"\n%s",props_errstr);
-            return 1;
-        }
-        inchans = props.chans;
-        filesize = sndsizeEx(sfdata.ifd);  // get size in frames
-        if(filesize == 0){
-            printf("File is empty!\n");
-            return 1;
-        }
-        
-        bool adjusted = false; // in case we have to increase ringframelen for huge FFT size
-        switch(props.type){
-            case wt_analysis:
-                sfdata.anal_framesize = props.chans;
-                outchans = inchans = props.chans = 1;
-                props.srate = props.origrate;
-                anal_nframes =  filesize / sfdata.anal_framesize ;   // * chans ?
-                if(anal_nframes <= 1){
-                    fprintf(stderr,"Bad analysis file - not enough frames\n");
-                    return 1;
-                }
-                sfdata.analframe1  = new float[sfdata.anal_framesize];                                      
-                winlen = props.winlen;
-                /* TODO: match ringbuf length to multiple of FFT size */
-                sfdata.fftsize = sfdata.anal_framesize - 2;
-                        
-                sfdata.pv_l = new phasevocoder;
-                if(!sfdata.pv_l->init(props.origrate,sfdata.anal_framesize-2,props.winlen,props.decfac,1.0,
-                        PVOC_S_NONE,PVOC_HAMMING,PVPP_STREAMING)){
-                    fprintf(stderr,"Error: unable to initialize pvoc engine.\n");           
-                    return 1;
-                }
-                overlap = sfdata.pv_l->anal_overlap();
-                //anal_buflen = /*FRAMESIZE * NUM_ANALFRAMES*/ anal_framesize;  // always mono
-                //anal_buflen = (anal_buflen / overlap) * overlap;
-                // buflen = anal_buflen;
-                from_frame = (long)(fromdur * props.arate);
-                if(from_frame >= anal_nframes){
-                    printf("Error: start is beyond end of file\n");
-                    goto error;
-                }
-                
-                to_frame = anal_nframes -1;
-                
-                if(totime > 0.0){
-                    to_frame = (long)(totime * props.arate + 0.5);
-                    if(to_frame >= anal_nframes){
-                        printf("Error: end time is beyond end of file\n");
-                        goto error;
-                    }
-                    
-                    if(to_frame <= from_frame){
-                        printf("Start time must be less than end time.\n");
-                        goto error;
-                    }
-                    
-                    if(to_frame - from_frame < 1){
-                        to_frame = from_frame +1;   // maybe we can in fact freeze on a single frame!
-                        // but we will not use in-memory system in this case
-                    }
-                    nFramesToPlay = to_frame - from_frame;
-                    
-                    printf("Playing %d analysis frames\n",(int) nFramesToPlay);
-                }
-               if(from_frame > 0){
-                    if(sndseekEx(sfdata.ifd,from_frame * sfdata.anal_framesize,SEEK_SET) < 0)   {
-                        printf("Error setting start position to frame %d\n", from_frame);
-                        printf("%s", sferrstr());
-                        goto error;
-                    }
-                    sfdata.current_frame = from_frame;
-                }
-                // ensure ring buffer is large enough even for huge FFT lengths!
-                adjusted = false;
-                while(ringframelen < (sfdata.fftsize * 4)) {
-                    ringframelen <<= 1;
-                    adjusted = true;
-                }
-                    
-                //printf("from = %d, to = %d\n",from_frame, to_frame);
-                playtype = PLAY_ANAFILE; 
-                printf("Opened %s: %ld frames, %d channels\n",argv[1],anal_nframes,props.chans);
-                if(adjusted)
-                    printf("FFT frame size = %d, expanding buffer to suit...\n", sfdata.fftsize);
-                break;
-            case wt_wave:
-                playtype = PLAY_SFILE;
-                printf("Opened %s: %ld frames, %d channels\n",argv[1],filesize,props.chans);
-                //framesize_factor = (long) (48000/ props.srate);
-                //framesize_factor += 1;
-                filesize /= inchans;
-                                
-                if(props.format == WAVE_EX){
-                    if(do_speakermask){
-                        int mask;
-                        if(flags[FLAG_B]){
-                            printf("-x flag ignored if -B used\n");
-                        }
-                        else {
-                            mask = sndgetchanmask(sfdata.ifd);
-                            if(mask < 0){
-                                printf("Could not read speaker mask. Using 0\n");
-                            }
-                            else 
-                                speakermask = mask;
-                        }
-                    }
-                }
-                from_frame = (long)(fromdur * props.srate);
-                if(from_frame >= filesize){
-                    printf("Error: start is beyond end of file\n");
-                    goto error;
-                }
-                nFramesToPlay = filesize - from_frame;
-                to_frame = filesize -1;
-                if(totime > 0.0){
-                    to_frame = (long)(totime * props.srate);
-                    if(to_frame >= filesize){
-                        printf("Error: end time is beyond end of file\n");
-                        goto error;
-                    }
-                    //printf("fromframe = %d, toframe = %d\n",(int)from_frame,(int) to_frame);
-                    if(to_frame <= from_frame){
-                        printf("End time must be later than from time\n");
-                        goto error;
-                    }
-                    nFramesToPlay = to_frame - from_frame;
-                    printf("Playing %d frames\n",(int) nFramesToPlay);
-                }
-                if(from_frame > 0){
-                    if(sndseekEx(sfdata.ifd,from_frame * inchans,SEEK_SET) < 0) {
-                        printf("Error setting start position\n");
-                        goto error;
-                    }
-                    sfdata.current_frame = from_frame;
-                }
-                
-                
-                outchans = inchans = props.chans;
-                if(props.chformat==MC_BFMT){
-                    flags[FLAG_B] = 1;
-                    if(inchans > 4) {
-                        inorder = 2;
-                        printf("%u-channel input: performing 2nd-order decode.\n",inchans);
-                    }
-                    switch(inchans){
-                    case 3:
-                       copyfunc = fmhcopy_3;
-                        break;
-                    case 4:
-                        copyfunc = fmhcopy_4;
-                        break;
-                    case 5:
-                        copyfunc = fmhcopy_5;
-                        break;
-                    case 6:
-                        copyfunc = fmhcopy_6;
-                        break;
-                    case 9:
-                        copyfunc = fmhcopy_9;
-                        break;
-                    case 11:
-                        copyfunc = fmhcopy_11;
-                        break;
-                    case 16:
-                        copyfunc = fmhcopy_16;
-                        break;
-                    default:
-                        fprintf(stderr,"Unsupported channel count (%u) for B-format file.\n",inchans);
-                        return 1;
-                    }
-                    
-                    switch(layout-1) {
-                    case FM_MONO:
-                        printf("Decoding to Mono\n");
-                        decodefunc = fm_i1_mono;
-                        outchans = 1;
-                        break;
-                    case FM_STEREO:
-                        printf("Decoding to Stereo\n");
-                        decodefunc = fm_i1_stereo;
-                        outchans = 2;
-                        break;
-                    case FM_SQUARE:
-                        printf("Decoding to Square\n");
-                        if(inorder == 1)
-                            decodefunc = fm_i1_square;
-                        else
-                            decodefunc = fm_i2_square;
-                        outchans = 4;
-                        break;
-                    case FM_PENT:
-                        printf("Decoding to pentagon\n");
-                        if(inorder==1)
-                            decodefunc = fm_i1_pent;
-                        else
-                            decodefunc = fm_i2_pent;
-                        outchans = 5;
-                        break;
-                    case DM_5_0:
-                        printf("Decoding to 5.0 surround (David Moore)\n");
-                        if(inorder==1)
-                            decodefunc = dm_i1_surr;
-                        else
-                            decodefunc = dm_i2_surr;
-                        outchans = 5;
-                        break;
-                    case DM_5_1:
-                        printf("Decoding to  5.1 surround (David Moore)\n");
-                        if(inorder==1)
-                            decodefunc = dm_i1_surr6;
-                        else
-                            decodefunc = dm_i2_surr6;
-                        outchans = 6;
-                        break;
-                    case FM_HEX:
-                        printf("Decoding to Hexagon\n");
-                        if(inorder==1)
-                            decodefunc = fm_i1_hex;
-                        else
-                            decodefunc = fm_i2_hex;
-                        outchans = 6;
-                        break;
-                    case FM_OCT1:
-                        printf("Decoding to Octagon 1\n");
-                        if(inorder==1)
-                            decodefunc = fm_i1_oct1;
-                        else
-                            decodefunc = fm_i2_oct1;
-                        outchans = 8;
-                        break;
-                    case FM_OCT2:
-                        printf("Decoding to Octagon 2\n");
-                        if(inorder==1)
-                            decodefunc = fm_i1_oct2;
-                        else
-                            decodefunc = fm_i2_oct2;
-                        outchans = 8;
-                        break;
-                    case FM_CUBE:
-                        printf("Decoding to Cube (FM interleaved)\n");
-                        if(inorder==1)
-                            decodefunc = fm_i1_cube;
-                        else
-                            decodefunc = fm_i2_cube;
-                        outchans = 8;
-                        break; 
-                    case FM_QUADCUBE:
-                        printf("Decoding to Octagon 1 (WAVEX order)\n");
-                        if(inorder==1)
-                            decodefunc = fm_i1_cubex;
-                        else
-                            decodefunc = fm_i2_cubex;
-                        outchans = 8;
-                        break;
-                    default:
-                        //quad
-                        printf("Decoding to quad surround (WAVEX order)\n");
-                        if(inorder == 1)
-                            decodefunc = fm_i1_quad;
-                        else
-                            decodefunc = fm_i2_quad;
-                        outchans = 4;
-                        break;
-                    }
-                }  // MC_BFMT            
-                
-                /* just read peaks for soundfile */
-                fpeaks = (CHPEAK *) calloc(inchans,sizeof(CHPEAK));
-                if(fpeaks==NULL){
-                    puts("No memory for PEAK data\n");
-                    goto error;
-                }
-                int thispeaktime;
-                res = sndreadpeaks(sfdata.ifd,inchans,fpeaks, &thispeaktime);
-                in_peaktime = (time_t) thispeaktime;
-                if(res==0){
-                    printf("No PEAK data in this file\n");
-                }
-                if(res > 0) {
-                    printf("PEAK data:\ncreation time: %s",ctime( &in_peaktime));
-                    
-                    for(i=0;i < inchans;i++){
-                        printf("CH %d: %.4f at frame %u: \t%.4f secs\n",
-                               i,fpeaks[i].value,fpeaks[i].position,(double)(fpeaks[i].position / (double) props.srate));
-                    }
-                }        
-                break;
-            default:
-                fprintf(stderr,"\nSorry: Pvplay can only play soundfiles and analysis files.");
-                return 1;
-        }
-    }
-
-    /* memory playback mode? */
-    if(playtype == PLAY_SFILE && nFramesToPlay <= ringframelen){
-        sfdata.membuf =  (float *) PaUtil_AllocateMemory(nFramesToPlay * sizeof(float) * /*inchans*/ outchans);
-        if( sfdata.membuf == NULL )   {
-            puts("Could not allocate memory play buffer.\n");
-            goto error;
-        }
-        sfdata.memFramelength = nFramesToPlay;
-        sfdata.memFramePos = 0;
-        playcallback = MemCallback;
-        printf("RAM block size =  %lu\n",nFramesToPlay);
-                    
-    }
-    else {
-        // set up pcm ring buffer  NB must be power of 2 size
-        if(props.srate > 48000)
-            ringframelen <<= 1;
-        printf("File buffer size set to %ld sample frames\n",ringframelen);
-        // NB ring buffer sized for decoded data, hence outchans here; otherwise inchans = outchans
-        sfdata.ringbufData = (float *) PaUtil_AllocateMemory( ringframelen * sizeof(float) * outchans); /* From now on, recordedSamples is initialised. */
-        if( sfdata.ringbufData == NULL )   {
-            puts("Could not allocate play buffer.\n");
-            goto error;
-        }                 
-        // number of elements has to be a power of two, so each element has to be a full m/c frame
-        if (PaUtil_InitializeRingBuffer(&sfdata.ringbuf, sizeof(float) * outchans, ringframelen , sfdata.ringbufData) < 0) {
-            puts("Could not initialise play buffer.\n");
-            goto error;
-        }            
-    } 
-    
-    
-    /* create input side workspace buffer, used for both soundfiles and analysis files! */
-    sfdata.inbuf = (float *) PaUtil_AllocateMemory(ringframelen * sizeof(float) * inchans);
-    if(sfdata.inbuf==NULL){
-        puts("No memory for read buffer\n");
-        goto error;
-    }
-    sfdata.inbuflen = ringframelen;
-    
-    if(playtype == PLAY_PVXFILE && (inchans==2)){
-        sfdata.outbuf_l = (float *) PaUtil_AllocateMemory(ringframelen * sizeof(float) );
-        sfdata.outbuf_r = (float *) PaUtil_AllocateMemory(ringframelen * sizeof(float) );
-        
-    }
-    
-    if(props.srate > 48000)
-        frames_per_buffer *= 2;
-    if(frames_per_buffer > 0)
-        printf("Audio buffer size = %lu\n",frames_per_buffer);
-    
-    sfdata.inchans      = inchans;
-    sfdata.outchans     = outchans;
-    sfdata.frames_played  = 0;
-    sfdata.gain         = gain;
-    sfdata.copyfunc     = copyfunc;
-    sfdata.decodefunc   = decodefunc;
-    sfdata.do_decode    = flags[FLAG_B];
-    sfdata.from_frame   = from_frame;  // interpreted according to Play Type
-    sfdata.to_frame     = to_frame;
-    sfdata.play_looped  = flags[FLAG_L];
-    sfdata.srate = props.srate;
-    sfdata.finished = 0;
-    g_pdata = &sfdata;
-    
-    outputParameters.device = device;   /*Pa_GetDefaultOutputDevice(); */ /* default output device */
-    outputParameters.channelCount = outchans;                     
-    outputParameters.sampleFormat = paFloat32;             /* 32 bit floating point output */
-    outputParameters.suggestedLatency = Pa_GetDeviceInfo( outputParameters.device )->defaultLowOutputLatency;
-    outputParameters.hostApiSpecificStreamInfo = NULL;
-    
-    devinfo = (PaDeviceInfo *) Pa_GetDeviceInfo(device);
-#if defined MAC || defined unix
-    if(devinfo){
-        printf("Using device %d: %s\n",device,devinfo->name);
-    }
-#endif
-    
-#ifdef WIN32
-    
-# ifdef NOTDEF
-    if(devinfo){
-        if(apiinfo->type  == paMME ){
-            winmmeStreamInfo.size = sizeof(winmmeStreamInfo);
-            winmmeStreamInfo.hostApiType = paMME; 
-            winmmeStreamInfo.version = 1;
-            winmmeStreamInfo.flags = paWinMmeUseChannelMask;
-            winmmeStreamInfo.channelMask = 0;
-            outputParameters.hostApiSpecificStreamInfo = &winmmeStreamInfo; 
-#  ifdef _DEBUG
-            printf("WIN DEBUG: WinMME device channel mask set to 0.\n");
-#  endif
-        }
-    }
-# endif
-    /* change val to 1 if also using MME */
-    if(devinfo){
-        int apitype = devinfo->hostApi; // get index of api type
-        const PaHostApiInfo *apiinfo =  Pa_GetHostApiInfo( apitype );
-        printf("Using device %d: %s:",device,devinfo->name);
-        if(apiinfo->type  == paDirectSound) {
-            printf ("(DS)\n");
-            /* set this IF we are using Dsound device. */
-            directSoundStreamInfo.size = sizeof(PaWinDirectSoundStreamInfo);
-            directSoundStreamInfo.hostApiType = paDirectSound; 
-            directSoundStreamInfo.version = 2;
-            directSoundStreamInfo.flags = paWinDirectSoundUseChannelMask;
-            directSoundStreamInfo.channelMask = speakermask;
-            outputParameters.hostApiSpecificStreamInfo = &directSoundStreamInfo;
-        }
-        else if(apiinfo->type == paASIO)
-            printf("(ASIO)\n");
-        // else
-        //    printf("API unknown!);
-    }
-#endif    
-    
-    
-    err = Pa_OpenStream(
-            &stream,
-            NULL,  /* No input */
-            &outputParameters, /* As above. */
-            props.srate,
-            frames_per_buffer,          /* Frames per buffer. */
-            paClipOff,      /* we won't output out of range samples so don't bother clipping them */
-            playcallback,
-            &sfdata );
-    
-    if( err != paNoError ){
-        fprintf(stderr,"Unable to open audio device: err = %d\n", err);   
-        exit(1);
-    }
-    err =  Pa_SetStreamFinishedCallback( stream, finishedCallback );
-    if( err != paNoError ) {
-        printf("Unable to set finish callback\n");
-        goto error;
-    }
-    sfdata.stream = stream;
-    file_playing = 1;
-
-    if(waitkey){
-        printf("Press any key to start:\n");
-        printf("Hit CTRL-C to stop.\n");
-        fflush(stdout);
-        while (!kbhit()){   
-            if(!file_playing)    //check for instant CTRL-C
-                goto error;     
-            };
-#ifdef _WIN32
-        if(kbhit())
-            _getch();            //prevent display of char
-#endif
-    }
-
-    // should this go in read thread func too?
-    
-    
-#ifdef NOTDEF
-    // TODO: any benefits in using this?
-    pterror = Pt_Start(200, porttimeCallback, NULL);
-    if(pterror != ptNoError){
-        printf("porttime timer failed to initialise!\n");
-        return 1;
-    }
-#endif
-    
-    /* if small block, read it all into memory
-     NB not doing paplay channel mapping  */
-    if(sfdata.memFramelength > 0){
-        //if(psf_sndReadFloatFrames(sfdata.ifd,sfdata.membuf,sfdata.memFramelength) 
-        if(fgetfbufEx(sfdata.membuf,sfdata.memFramelength*sfdata.inchans,sfdata.ifd,0)
-           != (int) sfdata.memFramelength*sfdata.inchans) {
-            printf("Error reading soundfile into memory\n");
-            goto error;
-        }
-    }
-    
-    // set up timer
-#ifdef unix
-    if(do_updatemessages) {
-        setitimer(ITIMER_REAL, &tout_val,0);
-        signal(SIGALRM,alarm_wakeup); /* set the Alarm signal capture */
-    }
-#endif
-//#ifdef _WIN32
-    // not sure of the best position for this
-    //if(!CreateTimerQueueTimer(&sfdata.hTimer, hTimerQueue,
-    //    (WAITORTIMERCALLBACK) TimerCallback, &sfdata,250,250,0)) {
-    //    printf("failed to start timer (3).\n");
-    //    return 1;
-    // }
-//#endif
-    /* Start the file reading thread */
-    
-    sfdata.startTime = Pa_GetStreamTime(stream );
-    
-    switch(playtype){
-        case PLAY_SFILE:    
-            if(sfdata.memFramelength == 0){
-                err = startThread(&sfdata, threadFunctionReadFromRawFile);
-                if( err != paNoError ) goto error;
-            }
-            else {
-                sfdata.flag = 0;
-            }
-            break;
-        case PLAY_ANAFILE:
-            err = startThread(&sfdata, threadFunctionReadFromAnalFile);
-            if( err != paNoError ) goto error;
-            break;
-        case PLAY_PVXFILE:
-            err = startThread(&sfdata, threadFunctionReadFromPVXFile);
-            if( err != paNoError ) goto error;
-            break;
-        default:
-            printf("Internal error: no playback file type!\n");
-            return 1;
-            
-    }
-    
-#ifdef _WIN32
-    if(do_updatemessages){
-        if(!CreateTimerQueueTimer(&sfdata.hTimer, hTimerQueue,
-                                  (WAITORTIMERCALLBACK) TimerCallback, &sfdata,200,200,0)) {
-            printf("failed to start timer (3).\n");
-            return 1;
-        }
-    }
-#endif
-    err = Pa_StartStream( stream );
-    if( err != paNoError ) {
-        fprintf(stderr,"Unable to start playback.\n");
-        goto error;
-    }
-    
-    while((!sfdata.finished) && file_playing){
-        // nothing to do!
-        Pa_Sleep(20);
-    }
-    // note to the curious: any bug in audio buffer arithmetic will likely cause crash here!
-    err = Pa_StopStream( stream );
-    if( err != paNoError ) {
-        printf("Error stopping stream\n");
-        goto error;
-    }
-    /*  read thread should exit always, no need to Stop it here */
-    
-    err = Pa_CloseStream( stream ); 
-    if( err != paNoError ) {
-        printf("Error closing stream\n");
-        goto error;
-    }
-    printf("%.2f secs\n",(float)(sfdata.lastTime));
-    fflush(stdout);
-    printf("Playback finished.\n");
-error:
-    Pa_Terminate();
-#ifdef _WIN32
-    CloseHandle(ghEvent);
-    DeleteTimerQueue(hTimerQueue);
-#endif
-    if( sfdata.ringbufData )       
-        PaUtil_FreeMemory(sfdata.ringbufData);
-    
-    if(sfdata.inbuf)
-        PaUtil_FreeMemory(sfdata.inbuf);
-    
-    if(sfdata.membuf)
-        PaUtil_FreeMemory(sfdata.membuf);
-    
-    if(sfdata.outbuf_l)
-        PaUtil_FreeMemory(sfdata.outbuf_l);
-    
-    if(sfdata.outbuf_r)
-        PaUtil_FreeMemory(sfdata.outbuf_r); 
-    
-    if(sfdata.ifd >= 0)
-        sndcloseEx(sfdata.ifd);
-    
-    if(sfdata.pvfile >= 0)
-        pvoc_closefile(sfdata.pvfile);
-        
-    if(fpeaks)
-        free(fpeaks);
-    
-//    sffinish();
-    
-    pvsys_release();
-    
-    if(err != paNoError){
-        fprintf( stderr, "An error occured while using the portaudio stream\n" ); 
-        fprintf( stderr, "Error number: %d\n", err );
-        fprintf( stderr, "Error message: %s\n", Pa_GetErrorText( err ) );
-        return err;
-    }
-    return 0;
-}
-
-int show_devices(void)
-{
-//      int i,j;
-        PaDeviceIndex numDevices,p;
-        const    PaDeviceInfo *pdi;
-#ifdef _WIN32
-        const PaHostApiInfo* api_info;
-#endif
-        PaError  err;
-        int nOutputDevices = 0;
-        
-#ifdef USE_ASIO
-        printf("For ASIO multi-channel, you may need to select the highest device no.\n");      
-#endif
-        /*Pa_Initialize();*/
-        numDevices =  Pa_GetDeviceCount();
-        if( numDevices < 0 )
-        {
-            printf("ERROR: Pa_CountDevices returned 0x%x\n", numDevices );
-            err = numDevices;
-            return err;
-        }
-#ifdef WIN32
-        printf("Driver\tDevice\tInput\tOutput\tName\n");
-#else
-        printf("Device\tInput\tOutput\tName\n");
-#endif
-        //printf("Number of devices = %d\n", numDevices );
-        for( p=0; p<numDevices; p++ )
-        {
-            pdi = Pa_GetDeviceInfo( p );
-            nOutputDevices++;
-            
-            if( p == Pa_GetDefaultOutputDevice() ) 
-                printf("*");
-            else
-                printf(" ");
-#ifdef WIN32
-            api_info = Pa_GetHostApiInfo(pdi->hostApi);
-            const char *apiname = api_info->name;
-            if(strcmp(apiname,"Windows DirectSound")==0)
-                apiname = "DS ";
-            printf("(%s)\t%d\t%d\t%d\t%s\n",apiname,p,
-                pdi->maxInputChannels,
-                pdi->maxOutputChannels,
-                pdi->name); 
-#else
-            printf("%d\t%d\t%d\t%s\n",p,
-                pdi->maxInputChannels,
-                pdi->maxOutputChannels,
-                pdi->name);
-
-
-#endif
-        }
-        /*Pa_Terminate();*/
-        return 0;
-}
-
-#ifdef unix
-int stricmp(const char *a, const char *b)
-{
-    while(*a != '\0' && *b != '\0') {
-        int ca = islower(*a) ? toupper(*a) : *a;
-        int cb = islower(*b) ? toupper(*b) : *b;
-        
-        if(ca < cb)
-            return -1;
-        if(ca > cb)
-            return 1;
-        
-        a++;
-        b++;
-    }
-    if(*a == '\0' && *b == '\0')
-        return 0;
-    if(*a != '\0')
-        return 1;
-    return -1;
-}
-
-int
-strnicmp(const char *a, const char *b, const int length)
-{
-    int len = length;
-    
-    while(*a != '\0' && *b != '\0') {
-        int ca = islower(*a) ? toupper(*a) : *a;
-        int cb = islower(*b) ? toupper(*b) : *b;
-        
-        if(len-- < 1)
-            return 0;
-        
-        if(ca < cb)
-            return -1;
-        if(ca > cb)
-            return 1;
-        
-        a++;
-        b++;
-    }
-    if(*a == '\0' && *b == '\0')
-        return 0;
-    if(*a != '\0')
-        return 1;
-    return -1;
-}
-#endif
-
-
-

dev/externals/paprogs/pvplay/pvplay.h

@@ -1,233 +0,0 @@
-/*
- * Copyright (c) 1983-2020 Richard Dobson and Composers Desktop Project Ltd
- * http://www.rwdobson.com
- * http://www.composersdesktop.com
- * This file is part of the CDP System.
- * The CDP System is free software; you can redistribute it
- * and/or modify it under the terms of the GNU Lesser General Public 
- * License as published by the Free Software Foundation; either 
- * version 2.1 of the License, or (at your option) any later version. 
- *
- * The CDP System is distributed in the hope that it will be useful, 
- * but WITHOUT ANY WARRANTY; without even the implied warranty of 
- * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  
- * See the GNU Lesser General Public License for more details.
- * You should have received a copy of the GNU Lesser General Public
- * License along with the CDP System; if not, write to the Free Software
- * Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
- *
- */
- 
-//
-//  pvplay.h
-//  
-//
-//  Created by Archer on 28/01/2013.
-//  Copyright (c) 2013 __MyCompanyName__. All rights reserved.
-//
-
-#ifndef _pvthreads_h
-#define _pvthreads_h
-
-#include <stdio.h>
-#include <stdlib.h>
-#include <memory.h>
-#include <time.h>
-#include <signal.h>
-#include <assert.h>
-
-#include "portaudio.h"
-//#include "porttime.h"
-#include "pa_ringbuffer.h"
-#include "pa_util.h"
-
-#ifdef _WIN32
-#define _WIN32_WINNT 0x0500
-#include <windows.h>
-#include <conio.h>
-#include <process.h>
-#include "pa_win_wmme.h"
-#include "pa_win_ds.h"
-#endif
-
-
-#ifdef unix
-#include <sys/types.h>
-#include <sys/timeb.h>
-#include <sys/time.h>
-#include <pthread.h>
-#include <ctype.h>
-/* in portsf.lib */
-int stricmp(const char *a, const char *b);
-int strnicmp(const char *a, const char *b, const int length);
-
-#endif
-
-#ifdef CPLONG64
-
-#define MYLONG int
-#else
-#define MYLONG long
-#endif
-
-
-/* need this to avoid clashes of GUID defs, etc */
-#ifdef _WIN32
-#ifndef _WINDOWS
-#define _WINDOWS
-#endif
-#endif
-
-
-
-/* sfsys just for analysis files! */
-#ifdef __cplusplus
-extern "C" {	
-#endif
-#include "sfsys.h"
-#include "fmdcode.h"
-#include "pvfileio.h"
-#ifdef __cplusplus
-}
-#endif
-
-#include "pvpp.h"
-
-#if defined MAC || defined unix
-#include <aaio.h>
-#endif
-
-#ifdef MAC
-#include <libkern/OSAtomic.h>
-#endif
-
-#ifndef min
-#define min(x,y)  ((x) < (y) ? (x) : (y))
-#endif
-
-#ifdef _WIN32
-extern HANDLE ghEvent;
-#endif
-
-// paplayt
-//typedef struct {
-//	int sfd;
-//    int chans;
-//	psf_buffer *outbuf;
-//	unsigned long frames_played;
-//	int last_buffer; /* init to 1; at detected eof send one extra buffer to flush, and set this to 0 */
-//} psfdata;
-
-
-typedef struct {
-	float *buffer;
-	long nframes;
-	MYLONG used;
-}
-psf_buffer;
-
-typedef struct {
-    PaUtilRingBuffer ringbuf;
-    PaStream *stream;
-    PaTime startTime;
-    PaTime lastTime;
-	float *ringbufData;
-    float *inbuf;   //for decoding and other tx; used as temp working buffer for read thread func.
-    /* for memory read func */
-    float *membuf;
-/* analysis file vars */
-    // pv objects
-    phasevocoder *pv_l;
-    phasevocoder *pv_r;
-    float *analframe1;
-    float *analframe2;
-    int anal_framesize; // in floats
-    int fftsize;        // sample frames from resynthesis of analysis frame(s)
-    float *outbuf_l;    // stereo pvx files - needed?
-    float *outbuf_r;
-    
-//    float** orderptr; // for chan mapping;
-#ifdef _WIN32
-    void *hThread;
-    HANDLE hTimer;
-    HANDLE hTimerCount;
-#endif
-  
-#ifdef unix
-    pthread_t hThread;
-#endif
-    fmhdecodefunc decodefunc;
-    fmhcopyfunc copyfunc;
-    double gain;
-	unsigned long frames_played;  // for .ana, .pvx files, these are analysis frames
-    unsigned long from_frame;
-    unsigned long to_frame;
-    unsigned long memFramelength;
-    unsigned long memFramePos;
-    unsigned long inbuflen;
-    unsigned long current_frame;
-    int srate;
-    int inchans;
-    int outchans;
-    int flag;
-	int ifd; // soundfiles, .ana files
-    int pvfile; // pvx file
-    pvoc_frametype inframetype;
-#ifdef DEBUG
-    int ofd;
-#endif
-    int do_decode;
-//    int do_mapping;
-    int play_looped;
-    int finished;
-} psfdata;
-
-#ifdef NOTDEF
-typedef struct {
-    PaStream *stream;
-    PaTime starttime /* = Pa_GetStreamTime(stream)*/;
-} ptimedata;
-#endif
-
-//extern int file_playing;
-//extern psfdata *g_pdata = NULL;  // for timer interrupt routine
-
-void stereo_interls(const float *in_l,const float *in_r,float *out,long insize);
-int show_devices(void);
-
-
-/* handlers */
-void playhandler(int sig);
-void finishedCallback(void *userData);
-#ifdef unix
-void alarm_wakeup (int i);
-#endif
-#ifdef _WIN32
-VOID CALLBACK TimerCallback(PVOID lpParam, BOOLEAN TimerOrWaitFired);
-#endif
-
-
-
-
-/* playback thread functions */
-
-
-#ifdef unix
-// TODO: unix return type should be void*
-typedef void* (*threadfunc)(void*);
-void*  threadFunctionReadFromRawFile(void* ptr);
-void*  threadFunctionReadFromAnalFile(void* ptr);
-void*  threadFunctionReadFromPVXFile(void* ptr);
-#endif
-
-
-#ifdef _WIN32
-typedef unsigned int (__stdcall *threadfunc)(void*);
-// soundfile
-unsigned int __stdcall threadFunctionReadFromRawFile(void* ptr);
-unsigned int __stdcall threadFunctionReadFromAnalFile(void* ptr);
-unsigned int __stdcall threadFunctionReadFromPVXFile(void* ptr);
-#endif
-
-
-#endif

dev/externals/paprogs/pvplay/pvpp.h

@@ -1,247 +0,0 @@
-/*
- * Copyright (c) 1983-2020 Richard Dobson and Composers Desktop Project Ltd
- * http://www.rwdobson.com
- * http://www.composersdesktop.com
- * This file is part of the CDP System.
- * The CDP System is free software; you can redistribute it
- * and/or modify it under the terms of the GNU Lesser General Public 
- * License as published by the Free Software Foundation; either 
- * version 2.1 of the License, or (at your option) any later version. 
- *
- * The CDP System is distributed in the hope that it will be useful, 
- * but WITHOUT ANY WARRANTY; without even the implied warranty of 
- * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  
- * See the GNU Lesser General Public License for more details.
- * You should have received a copy of the GNU Lesser General Public
- * License along with the CDP System; if not, write to the Free Software
- * Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
- *
- */
- 
-/*pvpp.h*/
-/* class def for pvoc wrapper */
-#ifndef __PVPP_H_INCLUDED__
-#define __PVPP_H_INCLUDED__
-
-#ifndef MIN
-#define MIN(x,y)    ( ((x)>(y)) ? (y) : (x) )
-#endif
-
-#ifndef MAX
-#define MAX(x,y)    ( ((x)>(y)) ? (x) : (y) )
-#endif
-
-
-#ifdef __cplusplus
-extern "C" {
-//#include <pvfileio.h>
-#endif
-    
-#include "pvdefs.h"
-#ifdef USE_FFTW
-#include <rfftw.h>
-#endif
-    
-#ifdef __cplusplus
-}
-#endif
-
-#ifdef MAC
-#include <sys/param.h>
-#include <sys/sysctl.h>
-#include <CoreServices/CoreServices.h>
-#include <Accelerate/Accelerate.h>
-#endif      
-    
-//}
-
-
-// must have -faltivec in C compiler flags
-#if defined(__VEC__)
-Boolean HasAltiVec(void);
-void TurnJavaModeOffOnG4(void);
-void RestoreJavaModeOnG4(void);
-#endif
-
-#ifdef NOTDEF
-typedef enum pvocmode {  PVPP_NOT_SET,PVPP_OFFLINE,PVPP_STREAMING};
-
-typedef enum pvoc_overlapfac {PVOC_O_W0,PVOC_O_W1,PVOC_O_W2,PVOC_O_W3,PVOC_O_DEFAULT};
-typedef enum pvoc_scaletype {PVOC_S_TIME,PVOC_S_PITCH,PVOC_S_NONE};
-
-typedef enum pvoc_frametype { PVOC_AMP_FREQ,PVOC_AMP_PHASE,PVOC_COMPLEX };
-typedef enum pvoc_windowtype {PVOC_DEFAULT,
-                                PVOC_HAMMING,
-                                PVOC_HANN,
-                                PVOC_KAISER,
-                                PVOC_RECT,
-                                PVOC_CUSTOM};
-#endif
-class phasevocoder
-{
-/* some may be made protected in due course, for derived classes */
-private:
-    double  rratio;
-    float   *input,     /* pointer to start of input buffer */
-        *output,        /* pointer to start of output buffer */
-        *anal,          /* pointer to start of analysis buffer */
-        *syn,           /* pointer to start of synthesis buffer */
-        *banal,         /* pointer to anal[1] (for FFT calls) */
-        *bsyn,          /* pointer to syn[1]  (for FFT calls) */
-        *nextIn,        /* pointer to next empty word in input */
-        *nextOut,       /* pointer to next empty word in output */
-        *analWindow,    /* pointer to center of analysis window */
-        *synWindow,     /* pointer to center of synthesis window */
-        *maxAmp,        /* pointer to start of max amp buffer */
-        *avgAmp,        /* pointer to start of avg amp buffer */
-        *avgFrq,        /* pointer to start of avg frq buffer */
-        *env,           /* pointer to start of spectral envelope */
-        *i0,            /* pointer to amplitude channels */
-        *i1,            /* pointer to frequency channels */
-        *oi,            /* pointer to old phase channels */
-        *oldInPhase,    /* pointer to start of input phase buffer */
-        *oldOutPhase;   /* pointer to start of output phase buffer */
-
-    int   m, n;
-
-    int N ,         /* number of phase vocoder channels (bands) */
-        M,          /* length of analWindow impulse response */
-        L,          /* length of synWindow impulse response */
-        D,          /* decimation factor (default will be M/8) */
-        I,          /* interpolation factor (default will be I=D)*/
-        W ,         /* filter overlap factor (determines M, L) */
-        /*RWD: want EXACT frequency! */
-    //  F,          /* fundamental frequency (determines N) */
-    //  F2,         /* F/2 */    /* RWD NOT USED */
-        
-        K,          /* set kaiser window*/
-        H,          /* set vonHann window */
-        analWinLen,     /* half-length of analysis window */
-        synWinLen;      /* half-length of synthesis window */
-    /* RWD see above */
-    float Fexact;
-
-    long // tmprate,    /* temporary variable */
-        sampsize,       /* sample size for output file */
-        //origrate,     /* sample rate of file analysed */
-        outCount,       /* number of samples written to output */
-        ibuflen,        /* length of input buffer */
-        obuflen,        /* length of output buffer */
-        nI,         /* current input (analysis) sample */
-        nO,             /* current output (synthesis) sample */
-        nMaxOut,        /* last output (synthesis) sample */
-        nMin,       /* first input (analysis) sample */
-        nMax;   /* last input sample (unless EOF) */
-/***************************** 6:2:91  OLD CODE **************
-                        long    origsize;
-*******************************NEW CODE **********************/
-    long    origsize;   /* sample type of file analysed */
-
-    char    ch;     /* needed for crack (commandline interpreter)*/
-    
-    int ifd, ofd;   /* CDP sound file handles */
-    float   beta ,  /* parameter for Kaiser window */
-        real,       /* real part of analysis data */
-        imag,       /* imaginary part of analysis data */
-        mag,        /* magnitude of analysis data */
-        phase,      /* phase of analysis data */
-        angleDif,   /* angle difference */
-        RoverTwoPi, /* R/D divided by 2*Pi */
-        TwoPioverR, /* 2*Pi divided by R/I */
-        sum,        /* scale factor for renormalizing windows */
-        ftot,   /* scale factor for calculating statistics */
-        rIn,        /* decimated sampling rate */
-        rOut,       /* pre-interpolated sampling rate */
-        invR,       /* 1. / srate */
-        time,       /* nI / srate */
-        tvx0,       /* current x value of time-var function */
-        tvx1,       /* next x value of time-var function */
-        tvdx,       /* tvx1 - tvx0 */
-        tvy0,       /* current y value of time-var function */
-        tvy1,       /* next y value of time-var function */
-        tvdy,       /* tvy1 - tvy0 */
-        frac,       /* tvdy / tvdx */
-        warp,   /* spectral envelope warp factor */
-        R ,     /* input sampling rate */
-        P ,     /* pitch scale factor */
-        Pinv,       /* 1. / P */
-        T;      /* time scale factor ( >1 to expand)*/
-
-    int i,j,k,      /* index variables */
-        Dd,     /* number of new inputs to read (Dd <= D) */
-        Ii,     /* number of new outputs to write (Ii <= I) */
-        N2,     /* N/2 */
-        NO,     /* synthesis NO = N / P */
-        NO2,        /* NO/2 */
-        IO,     /* synthesis IO = I / P */
-        IOi,        /* synthesis IOi = Ii / P */
-        //Mlen,
-        Mf,     /* flag for even M */
-        Lf,     /* flag for even L */
-        //Dfac,
-        flag,       /* end-of-input flag */
-        C,      /* flag for resynthesizing even or odd chans */
-        Ci,     /* flag for resynthesizing chans i to j only */
-        Cj,     /* flag for resynthesizing chans i to j only */
-        CC,     /* flag for selected channel resynthesis */
-        X,      /* flag for magnitude output */
-        E,      /* flag for spectral envelope output */ 
-        tvflg,  /* flag for time-varying time-scaling */
-        tvnxt,      /* counter for stepping thru time-var func */
-        tvlen;      /* length of time-varying function */
-    
-    float   srate;      /* sample rate from header on stdin */
-        
-    float   timecheckf;
-    long    isr,            /* sampling rate */
-    Nchans;         /* no of chans */
-
-    /* my vars */
-    int vH;             /* von Hann window */
-    pvocmode m_mode;
-    long  bin_index;    /* index into oldOutPhase to do fast norm_phase */ 
-    float *synWindow_base;
-    float *analWindow_base;
-#ifdef USE_FFTW
-    rfftwnd_plan forward_plan, inverse_plan;
-    int in_fftw_size,out_fftw_size;
-    float Ninv;
-#endif
-protected:
-    double besseli( double x);           /* from Csound*/
-    void hamming(float *win,int winLen,int even);
-    void kaiser(float *win,int len,double Beta);     /* from Csound */
-    void vonhann(float *win,int winLen,int even);
-
-public:
-    phasevocoder();
-    virtual ~phasevocoder();
-    
-
-    /* neither init func inputs all possible pvoc parameters - these are two possible subsets */
-    bool init(long outsrate,long fftlen,long winlen,long decfac,float scalefac,
-                        pvoc_scaletype stype,pvoc_windowtype wtype,pvocmode mode);
-    /* when pvoc -W flag used (ofac), it sets anal window size, decimation, etc */
-    bool init(long outsrate,long fftlen,pvoc_overlapfac ofac,float scalefac,
-                        pvoc_scaletype stype,pvoc_windowtype wtype,pvocmode mode);
-    /* to come: full init using PVOCDATA etc, or directly from another phasevocoder */
-    long anal_overlap(void) { return D;}
-    long syn_interp(void)   { 
-                    if(P != 1.0f) 
-                        return D;
-                    else 
-                        return I;}
-    long fftlen(void)  {return N;}
-    long winlen(void) { return M;}
-    /*retval gives numsamps written to outbuf, -1 for error */
-    long process_frame(float *frame,float *outbuf,pvoc_frametype frametype);    
-    long process_frame(float *frame,short *outbuf,pvoc_frametype frametype);
-    /* returns (variable) numsamps written to outbuf*/
-    long generate_frame(float *fbuf,float *outanal,long samps,pvoc_frametype frametype);
-    void reset_phases(void);
-#ifdef USE_CDP
-    void setprops(SFPROPS *props);
-#endif
-};
-
-#endif

dev/externals/paprogs/pvplay/pvthreads.cpp

@@ -1,487 +0,0 @@
-
-/*
- * Copyright (c) 1983-2020 Richard Dobson and Composers Desktop Project Ltd
- * http://www.rwdobson.com
- * http://www.composersdesktop.com
- * This file is part of the CDP System.
- * The CDP System is free software; you can redistribute it
- * and/or modify it under the terms of the GNU Lesser General Public 
- * License as published by the Free Software Foundation; either 
- * version 2.1 of the License, or (at your option) any later version. 
- *
- * The CDP System is distributed in the hope that it will be useful, 
- * but WITHOUT ANY WARRANTY; without even the implied warranty of 
- * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  
- * See the GNU Lesser General Public License for more details.
- * You should have received a copy of the GNU Lesser General Public
- * License along with the CDP System; if not, write to the Free Software
- * Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
- *
- */
-
-/*  pvthreads.c: thread functions for pvplay */
-/* Jan 2014: completed pvx support */
-
-#include "pvplay.h"
-#include <assert.h>
-
-extern int file_playing;
-extern psfdata *g_pdata;
-
-__inline void stereo_interls(const float *in_l,const float *in_r,float *out,long insize)
-{
-    long i;
-    const float *pfl_l,*pfl_r;
-    float*pfl_o;
-    pfl_o = out;
-    pfl_l = in_l;
-    pfl_r = in_r;
-
-    for(i=insize;i;--i){
-        *pfl_o++ = *pfl_l++;
-        *pfl_o++ = *pfl_r++;
-    }
-}
-
-/* soundfile */
-
-/* writes decoded data to ring buffer */
-/* size of element is full m/c frame */
-// TODO: unix thread should return void*
-#ifdef unix
-/*int*/void*  threadFunctionReadFromRawFile(void* ptr)
-#else
-unsigned int __stdcall threadFunctionReadFromRawFile(void* ptr)
-#endif
-{
-    psfdata* pdata = (psfdata*)ptr;
-    
-    /* Mark thread started */
-    pdata->flag = 0;
-    //printf("thread: from_frame = %d, to_frame = %d, current_Frame = %d\n",pdata->from_frame, pdata->to_frame, pdata->current_frame);
-    while (1) {
-        ring_buffer_size_t nElementsProcessed = 0;
-        ring_buffer_size_t elementsInBuffer = PaUtil_GetRingBufferWriteAvailable(&pdata->ringbuf);
-           
-        if (elementsInBuffer >= pdata->ringbuf.bufferSize / 4) {
-            void* ptr[2] = {NULL};
-            ring_buffer_size_t sizes[2] = {0};
-            
-            /* By using PaUtil_GetRingBufferWriteRegions, we can write directly into the ring buffer */
-            PaUtil_GetRingBufferWriteRegions(&pdata->ringbuf, elementsInBuffer, &ptr[0], &sizes[0], &ptr[1], &sizes[1]);
-            
-            if (file_playing)  {
-                ring_buffer_size_t itemsReadFromFile;
-                int i,j;
-                int framesread = 0;
-                int sampswanted = 0;
-                int sampsread = 0;
-                // we work with frames, = constant across inchans and outchans
-                itemsReadFromFile = 0;
-                
-                for(i = 0; i < 2 && (ptr[i] != NULL); i++) {
-                    // NB ringbuf is sized by m/c frames
-                    int frameswanted = sizes[i];
-                    
-                    pdata->current_frame = sndtellEx(pdata->ifd) / pdata->inchans;
-                    // read up to end frame if requested
-                    if(pdata->to_frame < pdata->current_frame + frameswanted)
-                        frameswanted = pdata->to_frame - pdata->current_frame;
-                    
-                    if(frameswanted > 0){
-                        sampswanted = frameswanted * pdata->inchans;
-                        sampsread = fgetfbufEx(pdata->inbuf,sampswanted,pdata->ifd,0);
-                        framesread = sampsread / pdata->inchans;
-                        
-                        if(framesread < 0){ // read error!
-                            printf("Error reading soundfile: %s ifd = %d\n", sferrstr(),pdata->ifd);
-                            pdata->flag = 1;
-                            file_playing = 0;
-                            break;  // just out of for loop - need to return instead?
-                        }
-                    }
-                    if(framesread == 0){
-                        /* EOF. EITHER: finish, or rewind if looping playback*/
-                        if(pdata->play_looped){
-                            if(sndseekEx(pdata->ifd,pdata->from_frame * pdata->inchans,SEEK_SET) < 0){
-                                printf("Error looping soundfile\n");
-                                pdata->flag = 1;
-                                file_playing = 0;
-                                break;
-                            }
-                            // sizes[1] especially may well = 0
-                            if(frameswanted==0)
-                                break;
-                            sampswanted = frameswanted * pdata->inchans;
-                            sampsread = fgetfbufEx(pdata->inbuf,sampswanted,pdata->ifd,0);
-                            framesread = sampsread / pdata->inchans;
-                            if(framesread < 0){ // read error!
-                                printf("Error reading soundfile\n");
-                                pdata->flag = 1;
-                                file_playing = 0;
-                                break;
-                            }
-                        }
-                        else {
-                            // we must watch the ring buffer to make sure all data has been rendered,
-                            // over several callback blocks
-                            //printf("End of data. playing = %d:\n", file_playing);
-                            //printf("\telements remaining = %d\n",elementsInBuffer);
-                            if(elementsInBuffer == pdata->ringbuf.bufferSize)  {
-                                pdata->flag = 1;
-                                break;
-                            }
-                        }
-                    }
-                    else {
-                        // ringbuf calcs always relative to outchans
-                        itemsReadFromFile += framesread;
-                        
-                        // now ready to apply decoding or other processing
-                        if(pdata->gain != 1.0){
-                            for(j = 0; j < framesread * pdata->inchans; j++)
-                                pdata->inbuf[j]  = (float)(pdata->inbuf[j] * pdata->gain);
-                        }
-                        // no channel mapping in pvplay
-                        if(pdata->do_decode) {
-                            ABFSAMPLE abfsamp;
-                            
-                            for(j=0; j < framesread; j++){
-                                // single frame only
-                                pdata->copyfunc(&abfsamp,pdata->inbuf + (j * pdata->inchans));
-                                /* BIG TODO: update funcs to process large frame buffer! */
-                                /* NB: ring buffer is effectively defined as raw bytes */
-                                pdata->decodefunc(&abfsamp, (float*)(ptr[i]) + (j * pdata->outchans), 1);
-                                nElementsProcessed++;
-                            }
-                        }
-                        else {  // inchans = outchans
-                            memcpy(ptr[i],pdata->inbuf,framesread * sizeof(float) * pdata->inchans);
-                            nElementsProcessed += framesread;
-                        }
-                    }
-                }
-                PaUtil_AdvanceRingBufferWriteIndex(&pdata->ringbuf, itemsReadFromFile);
-            }
-            else {
-                // this code is activated on Ctrl-C. Can do immediate finish by setting flag
-                //    printf("file done\n");
-                pdata->flag = 1;  
-                break;
-            }
-        }
-        //  else {
-        //      printf("ringbuf size = %d, elements remaining = %d, playing = %d\n",pdata->ringbuf.bufferSize,elementsInBuffer,file_playing);
-        //  }
-
-        /* Sleep a little while...! */
-        Pa_Sleep(10);
-    }
-    return 0;
-}
-
-/*  versions for .ana file (always mono only) */
-
-#ifdef unix
-/*int*/void*  threadFunctionReadFromAnalFile(void* ptr)
-#else
-unsigned int __stdcall threadFunctionReadFromAnalFile(void* ptr)
-#endif
-{
-    psfdata* pdata = (psfdata*)ptr;
-    
-    /* Mark thread started */
-    pdata->flag = 0;
-    
-    while (1) {
-        ring_buffer_size_t nElementsProcessed = 0;
-        ring_buffer_size_t elementsInBuffer = PaUtil_GetRingBufferWriteAvailable(&pdata->ringbuf);
-        
-        if (elementsInBuffer >= pdata->ringbuf.bufferSize / 4) {
-            void* ptr[2] = {NULL};
-            ring_buffer_size_t sizes[2] = {0};
-            
-            //read and analyse  as many frames as resynth space is available for, into inbuf
-            // then distribute to ringbuf sections as necessary
-            //printf("from_frame = %d,to_frame = %d\n",pdata->from_frame,pdata->to_frame);
-            int framestoget = elementsInBuffer / pdata->fftsize;
-            //assert(framestoget); // must be at least one, or ring buffer is too small!
-            if(framestoget==0)
-                break;
-            float *pbuf = pdata->inbuf;
-            int i, got = 0;
-            int elementstowrite = 0;
-            
-            if (file_playing)  {
-                /* find out where we are ... */
-                long framepos1 = sndtellEx(pdata->ifd) / pdata->anal_framesize;
-                
-                for(i = 0;i < framestoget;i++){ 
-                    /* read one frame*/
-                    got = fgetfbufEx(pdata->analframe1,pdata->anal_framesize,pdata->ifd,0);
-                    if(got == -1) {
-                        fprintf(stderr,"\nError reading from sfile\n");
-                        file_playing  = 0;
-                        break;
-                    }
-                    if(got != pdata->anal_framesize) {
-                        if(got > 0) {
-                            fprintf(stderr,"Infile error: incomplete analysis frame encountered\n");
-                            file_playing  = 0;
-                            break;
-                        }
-                    }
-                    framepos1++;
-                    
-                    if(got && (framepos1 <= pdata->to_frame)) {
-                        long samps = pdata->pv_l->process_frame(pdata->analframe1,pbuf,PVOC_AMP_FREQ);
-                        elementstowrite += samps;
-                        pbuf += samps;
-                    }
-                    
-                    if (got==0 || framepos1 >= pdata->to_frame){
-                        if(pdata->play_looped) {
-                            long pos;
-                            pos = sndseekEx(pdata->ifd,pdata->from_frame * pdata->anal_framesize,SEEK_SET);
-                            if(pos < 0){
-                                fprintf(stderr,"Error rewinding frame loop.\n");
-                                file_playing  = 0;
-                                break;
-                            }
-                        }
-                        else if(elementsInBuffer == pdata->ringbuf.bufferSize)  {
-                            //printf("buffer empty!\n");
-                            pdata->flag = 1;
-                            break;
-                        }
-                    }  
-                }
-                
-                //apply gain to inbuf first, then copy to ringbuf
-                if(pdata->gain != 1.0) {
-                    for(i=0;i < elementstowrite;i++)
-                        pdata->inbuf[i]  = (float)(pdata->inbuf[i] * pdata->gain);
-                }
-                /* By using PaUtil_GetRingBufferWriteRegions, we can write directly into the ring buffer */
-                PaUtil_GetRingBufferWriteRegions(&pdata->ringbuf, elementsInBuffer, &ptr[0], &sizes[0], &ptr[1], &sizes[1]);
-                pbuf = pdata->inbuf;
-                //printf("ring buffer sizes : %d:%d",(int)sizes[0], (int)sizes[1]);
-                for(i = 0; i < 2; i++) {
-                    if(ptr[i]) {
-                        int frameswanted = sizes[i];
-                        memcpy(ptr[i],pbuf,frameswanted * sizeof(float) * pdata->inchans);
-                        pbuf += frameswanted * pdata->inchans;
-                        nElementsProcessed += frameswanted;
-                    }
-                }
-                if(elementstowrite)
-                    PaUtil_AdvanceRingBufferWriteIndex(&pdata->ringbuf, elementstowrite);
-            } // file_playing
-            else {
-                // this code is activated on Ctrl-C. Can do immediate finish by setting flag
-                //               printf("file done\n");
-                pdata->flag = 1;  
-                break;
-            }
-        }
-        /* Sleep a little while... */
-        Pa_Sleep(5);
-    }    
-    return 0;
-}
-
-// version for pvx format, just mono and stereo supported for now
-#ifdef unix
-/*int*/void*  threadFunctionReadFromPVXFile(void* ptr)
-#else
-unsigned int __stdcall threadFunctionReadFromPVXFile(void* ptr)
-#endif
-{
-    psfdata* pdata = (psfdata*)ptr;
-
-    /* Mark thread started */
-    pdata->flag = 0;
-
-    while (1) {
-        ring_buffer_size_t nElementsProcessed = 0;
-        ring_buffer_size_t elementsInBuffer = PaUtil_GetRingBufferWriteAvailable(&pdata->ringbuf);
-
-        // we remember that an element is a frame (mono or stereo)
-        if (elementsInBuffer >= pdata->ringbuf.bufferSize / 4) {
-            void* ptr[2] = {NULL};
-            ring_buffer_size_t sizes[2] = {0};
-
-            //read and analyse  as many frames as resynth space is available for, into inbuf
-            // then distribute to ringbuf sections as necessary
-            //printf("from_frame = %d,to_frame = %d\n",pdata->from_frame,pdata->to_frame);
-            /* pvx : file framecount is of single frames - need enough space to synth a single mc analysis frame */
-            /* but to_frame and from_frame are m/c frame count */
-            int framestoget = (elementsInBuffer  / pdata->fftsize);
-
-            if(framestoget==0)
-                break;
-
-            int i, got = 0;
-            int elementstowrite = 0;
-
-            if(pdata->inchans==1) {
-                float *pbuf = pdata->inbuf;
-                if (file_playing)  {
-                    /* find out where we are ...multi-chan frame count */
-                    long framepos = pvoc_framepos(pdata->pvfile) / pdata->inchans; // NB
-                    if(framepos < 0){
-                        fprintf(stderr,"\nError reading file frame position\n");
-                        file_playing  = 0;
-                        break;  
-                    }
-                    for(i = 0;i < framestoget;i++){ 
-                        /* read one frame*/
-                        got = pvoc_getframes(pdata->pvfile,pdata->analframe1,1);
-                        if(got == -1) {
-                            fprintf(stderr,"\nError reading from sfile\n");
-                            file_playing  = 0;
-                            break;
-                        }
-
-                        framepos++;
-
-                        if(got && (framepos <= pdata->to_frame)) {
-                            long samps = pdata->pv_l->process_frame(pdata->analframe1,pbuf,PVOC_AMP_FREQ);
-                            elementstowrite += samps;
-                            pbuf += samps;
-                        }
-
-                        if (got==0 || framepos >= pdata->to_frame){
-                            if(pdata->play_looped) {
-                                if(pvoc_seek_mcframe(pdata->pvfile,pdata->from_frame,SEEK_SET)) {
-                                    fprintf(stderr,"Error rewinding frame loop.\n");
-                                    file_playing  = 0;
-                                    break;
-                                }
-                            }
-                            else if(elementsInBuffer == pdata->ringbuf.bufferSize)  {
-                                //printf("buffer empty!\n");  
-                                pdata->flag = 1;
-                                break;
-                            }
-                        }
-                    }  // framestoget
-
-                    //apply gain to inbuf first, then copy to ringbuf
-                    if(pdata->gain != 1.0) {
-                        for(i=0;i < elementstowrite;i++)
-                            pdata->inbuf[i]  = (float)(pdata->inbuf[i] * pdata->gain);
-                    }
-                    /* By using PaUtil_GetRingBufferWriteRegions, we can write directly into the ring buffer */
-                    PaUtil_GetRingBufferWriteRegions(&pdata->ringbuf, elementsInBuffer, &ptr[0], &sizes[0], &ptr[1], &sizes[1]);
-                    pbuf = pdata->inbuf;
-                    //printf("ring buffer sizes : %d:%d",(int)sizes[0], (int)sizes[1]);
-                    for(i = 0; i < 2; i++) {        
-                        if(ptr[i]) {
-                            int frameswanted = sizes[i];
-                            memcpy(ptr[i],pbuf,frameswanted * sizeof(float) * pdata->inchans);
-                            pbuf += frameswanted * pdata->inchans;
-                            nElementsProcessed += frameswanted;
-                        }
-                    }
-                    if(elementstowrite)
-                        PaUtil_AdvanceRingBufferWriteIndex(&pdata->ringbuf, elementstowrite);
-                } // file_playing
-                else {
-                    // this code is activated on Ctrl-C. Can do immediate finish by setting flag
-                    //               printf("file done\n");
-                    pdata->flag = 1;  
-                    break;
-                }
-            }  //inchans==1
-            else if (pdata->inchans==2) {
-                float *pbuf = pdata->outbuf_l;
-                float *pbuf_r = pdata->outbuf_r;
-                // framestoget is count of m/c analysis frames to read
-                if(file_playing){
-                    /* find out where we are ...multi-chan frame count */
-                    long framepos = pvoc_framepos(pdata->pvfile) / pdata->inchans; // NB
-                    if(framepos < 0){
-                        fprintf(stderr,"\nError reading file frame position\n");
-                        file_playing  = 0;
-                        break;
-                    }
-                    for(i = 0;i < framestoget;i++){
-                        /* read one stereo frame*/
-                        got = pvoc_getframes(pdata->pvfile,pdata->analframe1,1);
-                        if(got == -1) {
-                            fprintf(stderr,"\nError reading ch 1 from pvx file\n");
-                            file_playing  = 0;
-                            break;
-                        }
-                        if(got){
-                            got = pvoc_getframes(pdata->pvfile,pdata->analframe2,1);
-                            if(got == -1) {
-                                fprintf(stderr,"\nError reading ch 2 from pvx file, frame %lu\n",framepos);
-                                file_playing  = 0;
-                                break;
-                            }
-                        }
-                        framepos++;
-                        if(got && (framepos <= pdata->to_frame)) {
-                            // each call returns <overlap> samples
-                            long samps = pdata->pv_l->process_frame(pdata->analframe1,pbuf,PVOC_AMP_FREQ);
-                            pbuf += samps;
-                            samps = pdata->pv_r->process_frame(pdata->analframe2,pbuf_r,PVOC_AMP_FREQ);
-                            pbuf_r += samps;
-                            
-                            elementstowrite += samps; // element is a stereo sample frame
-                        }
-                        if (got==0 || framepos >= pdata->to_frame){
-                            if(pdata->play_looped) {
-                                //pos = sndseekEx(pdata->ifd,pdata->from_frame * pdata->anal_framesize,SEEK_SET);
-                                //if(pvoc_rewind(pdata->pvfile,1)){  /* 1 = skip empty frame */
-                                if(pvoc_seek_mcframe(pdata->pvfile,pdata->from_frame,SEEK_SET)) {
-                                    fprintf(stderr,"Error rewinding frame loop.\n");
-                                    file_playing  = 0;
-                                    break;
-                                }
-                            }
-                            else if(elementsInBuffer == pdata->ringbuf.bufferSize)  {
-                                //printf("buffer empty!\n");
-                                pdata->flag = 1;
-                                break;
-                            }
-                        }
-                    }  // framestogeet
-                    assert(elementstowrite <= elementsInBuffer);
-                    //apply gain to inbuf first, then copy to ringbuf
-                    pbuf = pdata->outbuf_l;
-                    pbuf_r = pdata->outbuf_r;
-                    stereo_interls(pbuf,pbuf_r,pdata->inbuf,elementstowrite);
-                    
-                    if(pdata->gain != 1.0) {
-                        for(i=0;i < elementstowrite * pdata->inchans;i++)
-                            pdata->inbuf[i]  = (float)(pdata->inbuf[i] * pdata->gain);
-                    }
-                    /* By using PaUtil_GetRingBufferWriteRegions, we can write directly into the ring buffer */
-                    PaUtil_GetRingBufferWriteRegions(&pdata->ringbuf, elementsInBuffer, &ptr[0], &sizes[0], &ptr[1], &sizes[1]);
-                    pbuf = pdata->inbuf;
-                    //printf("ring buffer sizes : %d:%d",(int)sizes[0], (int)sizes[1]);
-                    for(i = 0; i < 2; i++) {
-                        if(ptr[i]) {
-                            int frameswanted = sizes[i];
-                            memcpy(ptr[i],pbuf,frameswanted * sizeof(float) * pdata->inchans);
-                            pbuf += frameswanted * pdata->inchans;
-                            nElementsProcessed += frameswanted;
-                        }
-                    }
-                    if(elementstowrite)
-                        PaUtil_AdvanceRingBufferWriteIndex(&pdata->ringbuf, elementstowrite);
-                } // file_playing
-                else {
-                    pdata->flag = 1;
-                    break;
-                }
-            }
-        }
-        /* Sleep a little while... */
-        Pa_Sleep(5);
-    }
-    return 0;
-}

dev/externals/paprogs/recsf/CMakeLists.txt

@@ -1,27 +0,0 @@
-if(APPLE)
-  set(CMAKE_C_FLAGS "-O2 -Wall -mmacosx-version-min=10.5 -Dunix  -fomit-frame-pointer -funroll-loops")
-  include_directories ( /Developer/Headers/FlatCarbon )
-  find_library(COREAUDIOLIB CoreAudio)
-  find_library(AUDIOTOOLBOX AudioToolbox)
-  find_library(AULIB AudioUnit)
-  find_library(CARBONLIB Carbon)
-  find_library(AAIOLIB names libaaio.a paths /usr/local/lib)
-  set(EXTRA_LIBRARIES1 portsf pthread ${AAIOLIB} ${COREAUDIOLIB} ${AUDIOTOOLBOX} ${AULIB} ${CARBONLIB} ${EXTRA_LIBRARIES})
-else()
-  if(MINGW)
-    set(CMAKE_C_FLAGS "-O3 -DWIN32 -D_WIN32 -DUSE_ASIO -fomit-frame-pointer -funroll-loops")
-    set(EXTRA_LIBRARIES1 ${EXTRA_LIBARIES})
-  else()
-    set(CMAKE_C_FLAGS "-O3 -Wall -Dlinux -Dunix -fomit-frame-pointer -funroll-loops")
-    set(EXTRA_LIBRARIES1 asound jack pthread m ${EXTRA_LIBARIES})
-  endif()
-endif()
-
-link_directories(../../lib ../portaudio/lib/.libs)
-
-include_directories(../../include ../portaudio/include ../portaudio/src/common /usr/local/include)
-
-add_executable(recsf recsf.c)
-target_link_libraries(recsf portaudio.a portsf ${EXTRA_LIBRARIES1})
-
-my_install(recsf)

dev/externals/paprogs/recsf/recsf.c

@@ -1,974 +0,0 @@
-/*
- * Copyright (c) 1983-2013 Richard Dobson and Composers Desktop Project Ltd
- * http://people.bath.ac.uk/masrwd
- * http://www.composersdesktop.com
- * This file is part of the CDP System.
- * The CDP System is free software; you can redistribute it
- * and/or modify it under the terms of the GNU Lesser General Public 
- * License as published by the Free Software Foundation; either 
- * version 2.1 of the License, or (at your option) any later version. 
- *
- * The CDP System is distributed in the hope that it will be useful, 
- * but WITHOUT ANY WARRANTY; without even the implied warranty of 
- * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  
- * See the GNU Lesser General Public License for more details.
- * You should have received a copy of the GNU Lesser General Public
- * License along with the CDP System; if not, write to the Free Software
- * Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
- *
- */
- 
-/*
- *  recsf.c
- *
- */
-
-#include <stdio.h>
-#include <stdlib.h>
-#include <memory.h>
-#include <math.h>
-#include <assert.h>
-#include <time.h>
-#include <string.h>
-
-#ifdef _WIN32
-#define _WIN32_WINNT 0x0500
-#include <conio.h>
-#include <windows.h>
-#include <process.h>
-// NB: need to have PAWIN_USE_WDMKS_DEVICE_INFO defined
-#include "pa_win_ds.h"
-#endif
-
-#ifdef unix
-
-#include <sys/types.h>
-#include <sys/timeb.h>
-#endif
-
-#include <signal.h>
-
-#ifdef MAC
-
-#include <aaio.h>
-#include <libkern/OSAtomic.h>
-#endif
-
-#ifdef unix
-#include <sys/time.h>
-#include <pthread.h>
-/* in portsf.lib */
-extern int stricmp(const char *a, const char *b);
-#endif
-
-#include "portaudio.h"
-#include "pa_ringbuffer.h"
-#include "pa_util.h"
-#include "portsf.h"
-
-#ifndef min
-#define min(x,y)  ((x) < (y) ? (x) : (y))
-#endif
-
-#define N_BFORMATS (10)
-static const int bformats[N_BFORMATS] = {2,3,4,5,6,7,8,9,11,16};
-
-
-//#define FRAMES_PER_BUFFER (4096)
-// want nice big buffer for recording!
-#define RINGBUF_NFRAMES    (32768)
-#define NUM_WRITES_PER_BUFFER   (4)
-#define DEFAULT_SRATE (44100)
-#define DEFAULT_STYPE (1)
-#define DEFAULT_CHANS (2)
-#define DBRANGE (64)
-
-enum { ARG_PROGNAME, ARG_OUTFILE, ARG_DUR };
-
-#ifdef _WIN32
-HANDLE ghEvent;
-#endif
-
-
-typedef struct {
-    PaUtilRingBuffer ringbuf;
-    PaStream *stream;
-	float *ringbufData;
-    char  peakstr[(DBRANGE/2)+1];
-    PaTime startTime;
-    PaTime lastTime;
-#ifdef WIN32
-    void *hThread;
-    HANDLE hTimer;
-    HANDLE hTimerCount;
-#endif
-#ifdef unix
-    pthread_t hThread;
-#endif
-    double peak;
-	unsigned long frames_written;
-    unsigned long frames_to_write; // for optional duration arg
-    unsigned long current_frame;
-    int srate;
-    int chans;
-    int flag;
-    int ofd;
-    int finished;
-    int showlevels;
-} psfdata;
-
-static int file_recording;
-static psfdata *g_pdata = NULL;  // for timer interrupt routine
-
-
-int show_devices(void);
-
-void playhandler(int sig)
-{
-	if(sig == SIGINT)
-		file_recording = 0;
-}
-
-
-#ifdef unix
-void alarm_wakeup (int i)
-{
-    struct itimerval tout_val;
-    
-    signal(SIGALRM,alarm_wakeup);
-    
-    
-    if(file_recording && g_pdata->stream) {
-        double dBpeak  = (int)( 20.0 * log10(sqrt(g_pdata->peak)));
-        int dBmin = -DBRANGE;
-        int dBval = dBmin;
-        int i;
-        for(i=0;i < DBRANGE/2;i++){
-            g_pdata->peakstr[i] = dBpeak > dBmin+(i*2) ? '*' : '.';    
-        }
-        //printf("%.4f secs\r",(float)(g_pdata->frames_played /(double) g_pdata->srate));
-        g_pdata->lastTime = Pa_GetStreamTime(g_pdata->stream ) - g_pdata->startTime;
-        //printf("%.2f secs\r", g_pdata->lastTime);
-        if(g_pdata->showlevels)
-            printf("%.2f secs\t\t\%s\r", g_pdata->lastTime,g_pdata->peakstr );
-        else 
-            printf("%.2f secs\r", g_pdata->lastTime);
-        fflush(stdout);
-    }
-    tout_val.it_interval.tv_sec = 0;
-    tout_val.it_interval.tv_usec = 0;
-    tout_val.it_value.tv_sec = 0; 
-    tout_val.it_value.tv_usec = 250000;
-    
-    setitimer(ITIMER_REAL, &tout_val,0);
-    
-}
-#endif
-
-
-void finishedCallback(void *userData)
-{
-    psfdata *pdata = (psfdata*) userData;
-    //printf("stream finished!\n");
-    pdata->finished = 1;
-    file_recording = 0;    
-}
-
-
-#ifdef WIN32
-
-VOID CALLBACK TimerCallback(PVOID lpParam, BOOLEAN TimerOrWaitFired)
-{
-    psfdata *pdata = (psfdata*) lpParam;
-    
-    
-    if(file_recording && pdata->stream) {
-        //printf("%.4f secs\r",(float)(g_pdata->frames_played /(double) g_pdata->srate));
-        double dBpeak  = (int)( 20.0 * log10(sqrt(pdata->peak)));
-        int dBmin = -DBRANGE;
-        int dBval = dBmin;
-        int i;
-        for(i=0;i < DBRANGE/2;i++) {
-            pdata->peakstr[i] = dBpeak > dBmin+(i*2) ? '*' : '.';    
-        }
-        
-        pdata->lastTime = Pa_GetStreamTime(pdata->stream ) - pdata->startTime;
-        
-        printf("%.2f secs\t\t%s\r", pdata->lastTime,pdata->peakstr ); 
-        fflush(stdout);
-        SetEvent(ghEvent);
-    }
-    else
-        printf("\n");
-}
-
-#endif
-
-// TODO: implement optional duration arg
-
-
-#ifdef unix
-static int threadFunctionWriteFile(void* ptr)
-#else
-static unsigned int __stdcall threadFunctionWriteFile(void* ptr)
-#endif
-{
-    psfdata* pData = (psfdata*)ptr;
-    
-    /* Mark thread started */
-    pData->flag = 0;
-    
-    while (1) {
-        ring_buffer_size_t elementsInBuffer = PaUtil_GetRingBufferReadAvailable(&pData->ringbuf);
-        if(file_recording == 0){
-            //write out whatever remains in ring buffer
-            void* ptr[2] = {0};
-            ring_buffer_size_t sizes[2] = {0};
-            
-            //printf("flushing ring buffer...\n");
-            
-            ring_buffer_size_t elementsRead = PaUtil_GetRingBufferReadRegions(&pData->ringbuf, elementsInBuffer, ptr + 0, sizes + 0, ptr + 1, sizes + 1);
-            if (elementsRead > 0) {       
-                int i;
-                
-                for (i = 0; i < 2 && ptr[i] != NULL; ++i)  {
-                    unsigned long towrite = sizes[i];
-                    if(pData->frames_to_write){  
-                        if(pData->frames_written + towrite > pData->frames_to_write)
-                            towrite = pData->frames_to_write - pData->frames_written;
-                    }
-                    if(psf_sndWriteFloatFrames(pData->ofd,(float*) ptr[i],towrite) != towrite) {
-                        printf("File %d write error\n",pData->ofd);
-                        pData->flag = 0;
-                        break;
-                    }
-                    pData->frames_written += towrite;
-                    if(pData->frames_written == pData->frames_to_write){
-                        //printf("recording done\n");
-                        pData->flag = 1;
-                        file_recording = 0;
-                        break;
-                    }
-                }
-                PaUtil_AdvanceRingBufferReadIndex(&pData->ringbuf, elementsRead);
-            }
-
-            break;
-        }
-        if ( (elementsInBuffer >= pData->ringbuf.bufferSize / NUM_WRITES_PER_BUFFER) || pData->flag ) {
-            void* ptr[2] = {0};
-            ring_buffer_size_t sizes[2] = {0};
-            
-            /* By using PaUtil_GetRingBufferReadRegions, we can read directly from the ring buffer */
-            ring_buffer_size_t elementsRead = PaUtil_GetRingBufferReadRegions(&pData->ringbuf, elementsInBuffer, ptr + 0, sizes + 0, ptr + 1, sizes + 1);
-            if (elementsRead > 0) {       
-                int i;
-                
-                for (i = 0; i < 2 && ptr[i] != NULL; ++i)  {
-                    unsigned long towrite = sizes[i];
-                    if(pData->frames_to_write){  
-                        if(pData->frames_written + towrite > pData->frames_to_write)
-                            towrite = pData->frames_to_write - pData->frames_written;
-                    }
-                    if(psf_sndWriteFloatFrames(pData->ofd,(float*) ptr[i],towrite) != towrite) {
-                        printf("File %d write error\n",pData->ofd);
-                        pData->flag = 0;
-                        break;
-                    }
-                    pData->frames_written += towrite;
-                    if(pData->frames_written == pData->frames_to_write){
-                        //printf("recording done\n");
-                        pData->flag = 1;
-                        file_recording = 0;
-                        break;
-                    }
-                }
-                PaUtil_AdvanceRingBufferReadIndex(&pData->ringbuf, elementsRead);
-            }
-            
-            if (pData->flag) {
-                break;
-            }
-        }
-        
-        /* Sleep a little while... */
-        Pa_Sleep(10);
-    }
-       
-    return 0;
-}
-
-#ifdef unix
-// TODO: unix return type should be void*
-typedef int (*threadfunc)(void*);
-#endif
-#ifdef WIN32
-typedef unsigned int (__stdcall *threadfunc)(void*);
-#endif
-
-
-/* Start up a new thread for given function */
-static PaError startThread( psfdata* pdata, threadfunc fn )
-{
-    pdata->flag = 1;
-#ifdef _WIN32
-    pdata->hThread = (void*)_beginthreadex(NULL, 0, fn, pdata, 0, NULL);
-    if (pdata->hThread == NULL) 
-        return paUnanticipatedHostError;
-    /* Wait for thread to startup */
-    while (pdata->flag) {
-        Pa_Sleep(10);
-    }
-    /* Set file thread to a little higher prio than normal */
-    SetThreadPriority(pdata->hThread, THREAD_PRIORITY_ABOVE_NORMAL);
-#else
-    
-#if defined(__APPLE__) || defined(__GNUC__)
-    if(pthread_create(&pdata->hThread,NULL,(void*) fn,pdata))
-        return -1;
-    /* Wait for thread to startup */
-    while (pdata->flag) {
-        Pa_Sleep(10);
-    } 
-#endif
-#endif   
-    return paNoError;
-}
-// for sake of completion - curently NOT USED 
-static int stopThread( psfdata* pdata )
-{
-    // RWD: just called when all data played; must be called before StopStream
-    //   pdata->flag = 1;
-    /* Wait for thread to stop */
-       while (pdata->flag == 0) {
-           Pa_Sleep(10);
-       }
-#ifdef _WIN32
-    CloseHandle(pdata->hThread);
-    pdata->hThread = 0;
-#else
-    
-#if defined(__APPLE__) || defined(__GNUC__)
-    pthread_cancel(pdata->hThread);
-#endif
-#endif
-    return paNoError;
-}
-
-static int recordCallback( const void *inputBuffer, void *outputBuffer,
-                          unsigned long framesPerBuffer,
-                          const PaStreamCallbackTimeInfo* timeInfo,
-                          PaStreamCallbackFlags statusFlags,
-                          void *userData )
-{
-    psfdata *data = (psfdata*) userData;
-    const float *rptr = (const float*) inputBuffer;
-    double peak = 0.0, val;
-    unsigned int i;
-    
-    (void) outputBuffer; /* Prevent unused variable warnings. */
-    (void) timeInfo;
-    (void) statusFlags;
-    (void) userData;
-    
-    data->current_frame += PaUtil_WriteRingBuffer(&data->ringbuf, rptr, framesPerBuffer);
-    
-    // simple level meter!
-    val = rptr[0];
-    val  =  val * val;
-    peak = val;
-    for(i=0;i < framesPerBuffer * data->chans ;i++){
-        val = rptr[i];
-        val  =  val * val;
-        peak =  peak < val ? val : peak;
-    }
-    data->peak = peak;
-    if(data->flag) {
-        return paComplete; // or paAbort?
-    }
-    return paContinue;
-}
-
-void usage(void)
-{
-        printf("usage: recsf [-BN][-cN][-dN][-hN][-i][-p][-rN][-tN] outfile [dur]\n"
-               "outfile:   output file in WAVE,AIFF or AIFC formats,\n"
-               "           determined by the file extension.\n"
-               "           use extension .amb to create a B-Format file.\n"
-               "-BN    :   set memory buffer size to N frames (default: %d).\n"
-               "              N must be a power of 2 (e.g 4096, 8192, 16384 etc).\n"
-               "-cN    :   set channels to N (default: 2).\n"
-               "-hN    :   set hardware buffer size to N frames (default: set by device)\n"
-               "-p     :   suppress running peak level indicator\n"
-               "-rN    :   set sample rate to N Hz (default: 44100)\n"
-               "-tN    :   set sample type to N (default: 1)\n"
-               "           0 :  16 bits\n"
-               "           1 :  24 bits\n"
-               "           2 :  32bit floats\n"
-               "dur    :   optional fixed duration for outfile\n"
-               "             (overriden by Ctrl-C)\n"
-               "           Otherwise, use Ctrl_C to terminate recording.\n"
-               "-dN    :   use input device N\n"
-               "-i     :   start recording immediately (default: wait for keypress)\n"
-               ,RINGBUF_NFRAMES);
-}
-
-static unsigned NextPowerOf2(unsigned val)
-{
-    val--;
-    val = (val >> 1) | val;
-    val = (val >> 2) | val;
-    val = (val >> 4) | val;
-    val = (val >> 8) | val;
-    val = (val >> 16) | val;
-    return ++val;
-}
-
-int main(int argc,char **argv)
-{
-	PaStreamParameters inputParameters;
-#ifdef _WIN32
-    /* portaudio sets default channel masks we can't use; we must do all this to set default mask = 0! */
-    PaWinDirectSoundStreamInfo directSoundStreamInfo;
-    //    PaWinMmeStreamInfo winmmeStreamInfo;
-    
-#endif
-    PaDeviceInfo *devinfo = NULL;
-    PaStream *stream = NULL;
-    PaStreamCallback *callback = recordCallback; 
-	PaError err = paNoError;
-	psfdata sfdata;
-	PSF_CHPEAK *fpeaks = NULL;
-    PSF_PROPS props;
-    const char* outfilename = NULL;
-    time_t in_peaktime = 0;
-    MYLONG lpeaktime;
-	int waitkey = 1;
-    int showlevels = 1;
-    double duration = 0.0;
-    double framedurms;
-    int i,j= 0;
-    int res;
-    int samptype = DEFAULT_STYPE;
-    char* ext = NULL;
-	PaDeviceIndex device;
-    long framesize = 0;   // Apr 2010 to be scaled by sample rate
-    unsigned long ringframelen = RINGBUF_NFRAMES;  // length of ring buffer in m/c frames
-    unsigned long frames_per_buffer = 0;
-    unsigned int frameBlocksize = 0;
-    double maxdurFrames = 0.0;
-    unsigned long LmaxdurFrames = 0;
-    unsigned long LmaxdurSecs = 0;
-    unsigned long LmaxdurMins = 0;
-    
-#ifdef unix
-    struct itimerval tout_val;    
-    tout_val.it_interval.tv_sec = 0;
-    tout_val.it_interval.tv_usec = 0;
-    tout_val.it_value.tv_sec = 0; 
-    tout_val.it_value.tv_usec = 200000;
-#endif
-    
-#ifdef _WIN32
-    HANDLE hTimerQueue;
-    ghEvent = CreateEvent(NULL,TRUE,FALSE,NULL);
-    if(ghEvent==NULL){
-        printf("Failed to start internal timer (1).\n");
-        return 1;
-    }
-    hTimerQueue = CreateTimerQueue();
-    if(hTimerQueue == NULL){
-        printf("Failed to start internal timer (2).\n");
-        return 1;
-    }
-#endif
-    
-    
-    props.chans = DEFAULT_CHANS;
-    props.srate = DEFAULT_SRATE;
-    props.samptype = samptype;
-    
-    
-	signal(SIGINT,playhandler);
-    
-	sfdata.ringbufData = NULL;
-    sfdata.frames_to_write = 0;
-    printf("RECSF: multi-channel record to file. v 1.1.0 RWD,CDP 2013\n");
-    file_recording = 0;
-    err = Pa_Initialize();
-	if( err != paNoError ) {
-		printf("Failed to initialize Portaudio.\n");
-        Pa_Terminate();
-        return 1;
-    }
-	device =  Pa_GetDefaultInputDevice();
-    
-    /* CDP version number */
-    if(argc==2 && (stricmp(argv[1],"--version")==0)){
-        printf("1.1.0\n");
-        return 0;
-    }
-    
-    if(argc < ARG_DUR) {
-        printf("insufficient args\n");
-        usage();
-        show_devices();
-		Pa_Terminate();
-        return 1;
-    }
-    
-    while(argv[1][0]=='-'){
-        int err = 0;
-        unsigned long userbuflen = 0;
-		switch(argv[1][1]){
-            case 'c':
-                if(argv[1][2]=='\0'){
-                    printf("-c flag requires parameter\n");
-                    err++;
-                    break;
-                }
-                props.chans = atoi(&(argv[1][2]));
-                if(props.chans < 1){
-                    printf("bad value %d for channels\n",props.chans);
-                    err++;
-                }
-                break;
-            case 'd':
-                if(argv[1][2]=='\0'){
-                    printf("-d flag requires parameter\n");
-                    err++;
-                    break;
-                }
-                device = atoi(&(argv[1][2]));
-                break;
-            case 'h':
-                if(argv[1][2]=='\0'){
-                    printf("-h flag requires parameter\n");
-                    err++;
-                    break;
-                }
-                frames_per_buffer = atoi(&(argv[1][2]));
-                if(frames_per_buffer < 32){
-                    printf("-h value too small; must be >= 32\n");
-                    err++;
-                }
-                break;
-            case 'i':
-                waitkey = 0;
-                break;
-            case 'p':
-                showlevels = 0;
-                break;
-            case 'r':
-                if(argv[1][2]=='\0'){
-                    printf("-r flag requires parameter\n");
-                    err++;
-                    break;
-                }
-                props.srate = atoi(&(argv[1][2]));
-                if(props.srate <= 0){
-                    printf("bad value %d for srate\n",props.srate);
-                    err++;
-                }
-                break;
-            case 't':
-                if(argv[1][2]=='\0'){
-                    printf("-t flag requires parameter\n");
-                    err++;
-                    break;
-                }
-                samptype = atoi(&(argv[1][2]));
-                if(samptype < 0 || samptype > 2){
-                    printf("bad value %d for sample type\n",samptype);
-                    err++;
-                }
-                break;
-            case 'B':
-                if(argv[1][2]=='\0'){
-                    printf("-B flag requires parameter\n");
-                    err++;
-                    break;
-                }
-                ringframelen = atoi(&(argv[1][2]));
-                if(ringframelen < 1024){
-                    printf("-B: buffer size must be >=1024\n");
-                    err++;
-                }
-                userbuflen = NextPowerOf2(ringframelen);
-                if(userbuflen != ringframelen){
-                    printf("-B: buffer size must be power of 2 size\n");
-                    Pa_Terminate();
-                    return 1;
-                }
-                
-                break;
-            default:
-                printf("unrecognised flag option\n");
-                err++;
-                break;
-		}
-        if(err){
-            Pa_Terminate();
-            return 1;
-        }
-		argv++;	 argc--;
-	}
-
-    if(argc < ARG_DUR || argc > ARG_DUR+1) {
-        usage();
-        show_devices();
-		Pa_Terminate();
-        return 1;
-    }
-    outfilename = argv[ARG_OUTFILE];
-    
-    switch(samptype){
-        case 0:
-            props.samptype = PSF_SAMP_16;
-            frameBlocksize = 2;
-            break;
-        case 1:
-            props.samptype = PSF_SAMP_24;
-            frameBlocksize = 3;
-            break;
-        case 2:
-            props.samptype = PSF_SAMP_IEEE_FLOAT;
-            frameBlocksize = 4;
-            break;
-        default:
-            printf("stype must be 0,1 or 2\n");
-            Pa_Terminate();
-            return 1;
-    }
-    
-    // find max recording time, with safety margin
-    frameBlocksize *= props.chans; 
-    maxdurFrames = (pow(2.0,32.0) / frameBlocksize) - 1000;
-    LmaxdurFrames = (unsigned long) maxdurFrames;
-    LmaxdurSecs =  LmaxdurFrames / props.srate;
-    LmaxdurMins =  LmaxdurSecs / 60;
-    printf("Max recording time: %lu mins, %lu secs\n",LmaxdurMins,LmaxdurSecs - (LmaxdurMins * 60));
-    
-    if(argc==ARG_DUR+1){
-        duration = atof(argv[ARG_DUR]);
-        if(duration <=0.0){
-            printf("duration must be positive!\n");
-            Pa_Terminate();
-            return 1;
-        }
-        if(duration > (double) LmaxdurSecs){
-            printf("specified duration too long for file format.\n");
-            Pa_Terminate();
-            return 1;
-        }
-        sfdata.frames_to_write = (unsigned long) (duration * props.srate);
-    }
-    
-    ext = strrchr(outfilename,'.');
-    if(ext && stricmp(ext,".amb")==0) {
-        int matched = 0;
-		for(i=0;i < N_BFORMATS;i++)	{
-			if(props.chans == bformats[i]){
-				matched = 1;
-				break;
-			}
-		}
-		if(!matched){
-			printf("WARNING: No Bformat definition for %d-channel file.\n",props.chans);
-		}
-		
-        props.format = PSF_WAVE_EX;
-        props.chformat = MC_BFMT;
-    }
-    else {
-        // we must be strictly correct with WAVE formats!
-        props.chformat = props.chans > 2 ? MC_STD : STDWAVE;
-        props.format = psf_getFormatExt(outfilename);
-        if(props.chans > 2 || props.samptype > PSF_SAMP_16 || props.srate > 48000)
-            props.format = PSF_WAVE_EX;
-            
-    }
-    sfdata.ofd = psf_sndCreate(outfilename,&props,0,0,PSF_CREATE_RDWR);
-    if(sfdata.ofd < 0){
-        printf("Sorry - unable to create outfile\n");
-        goto error;
-    } 
-    
-    fpeaks = (PSF_CHPEAK *) calloc(props.chans,sizeof(PSF_CHPEAK));
-	if(fpeaks==NULL){
-		puts("no memory for PEAK data\n");
-		goto error;
-	}
-    
-    if(props.srate > 48000)
-        ringframelen <<= 1;
-    printf("File buffer size = %ld\n",ringframelen);
-    // NB ring buffer sized for decoded data, hence outchans here; otherwise inchans = outchans
-    sfdata.ringbufData = (float *) PaUtil_AllocateMemory( ringframelen * sizeof(float) * props.chans); /* From now on, recordedSamples is initialised. */
-    if( sfdata.ringbufData == NULL )   {
-        puts("Could not allocate play buffer.\n");
-        goto error;
-    }
-    // number of elements has to be a power of two, so each element has to be a full m/c frame
-    if (PaUtil_InitializeRingBuffer(&sfdata.ringbuf, sizeof(float) * props.chans, ringframelen , sfdata.ringbufData) < 0) {
-        puts("Could not initialise play buffer.\n");
-        goto error;
-    }
-    // scale frame buf to sample rate
-    if(props.srate > 48000)
-         frames_per_buffer *= 2;
-    if(frames_per_buffer > 0)
-        printf("Audio buffer size = %lu frames\n",frames_per_buffer);
-
-	sfdata.chans      = props.chans;
-	sfdata.frames_written  = 0;
-    sfdata.current_frame = 0;
-    sfdata.srate = props.srate;
-    sfdata.peakstr[DBRANGE/2] = '\0';
-    
-    sfdata.finished = 0;
-    sfdata.showlevels = showlevels;
-    g_pdata = &sfdata;
-    
-    inputParameters.device = device;   /*Pa_GetDefaultOutputDevice(); */ /* default output device */
-    inputParameters.channelCount = props.chans;                     
-    inputParameters.sampleFormat = paFloat32;             /* 32 bit floating point output */
-    inputParameters.suggestedLatency = Pa_GetDeviceInfo( inputParameters.device )->defaultHighInputLatency;
-    inputParameters.hostApiSpecificStreamInfo = NULL;
-    
-    devinfo = (PaDeviceInfo *) Pa_GetDeviceInfo(device);
-#ifdef MAC
-    if(devinfo){
-        printf("Using device %d: %s\n",device,devinfo->name);
-    }
-#endif
-    
-#ifdef WIN32
-    if(devinfo) {
-        int apitype = devinfo->hostApi;
-        const PaHostApiInfo *apiinfo = Pa_GetHostApiInfo(apitype);
-        printf("Using device %d: %s:\n",device,devinfo->name);
-        
-        if(apiinfo->type  == paDirectSound ){
-            printf("(DS)\n");
-            /* set this IF we are using Dsound device. */
-            directSoundStreamInfo.size = sizeof(PaWinDirectSoundStreamInfo);
-            directSoundStreamInfo.hostApiType = paDirectSound; 
-            directSoundStreamInfo.version = 2;
-            directSoundStreamInfo.flags = paWinDirectSoundUseChannelMask;
-            directSoundStreamInfo.channelMask = 0;
-            inputParameters.hostApiSpecificStreamInfo = &directSoundStreamInfo; 
-        }
-        else if(apiinfo->type == paASIO)
-            printf("(ASIO)\n");
-        // else
-        //    printf("API unknown!);
-    }
-    
-#endif  
-    // TODO; move this up to before file is created?
-    err = Pa_IsFormatSupported(&inputParameters, NULL,props.srate);
-    if(err != paNoError){
-        printf("Selected device does not support this format.\n");
-        goto error;
-    }
-
-    err = Pa_OpenStream(
-                        &stream,
-                        &inputParameters, 
-                        NULL,  /* No output */
-                        props.srate,
-                        frames_per_buffer,			
-                        paClipOff,  
-                        recordCallback,
-                        &sfdata );
-    
-    
-	if( err != paNoError ) {
-        printf("Unable to open output device for %d-channel file.\n",props.chans);
-        goto error;
-    }
-    err =  Pa_SetStreamFinishedCallback( stream, finishedCallback );
-    if( err != paNoError ) {
-        printf("Internal error: unable to set finish callback\n");
-        goto error;
-    }
-	sfdata.stream = stream;
-    
-    
- 
-    file_recording = 1;
-    if(waitkey){
-		printf("Press any key to start:\n");	
-		while (!kbhit()){	
-			if(!file_recording)	 //check for instant CTRL-C
-                goto error;		
-        };
-#ifdef WIN32
-		if(kbhit())
-			_getch();			 //prevent display of char
-#endif
-	}
-    
-    // set up timer
-#ifdef unix   
-    setitimer(ITIMER_REAL, &tout_val,0);
-    signal(SIGALRM,alarm_wakeup); /* set the Alarm signal capture */
-#endif
-    /* Start the file reading thread */
-    sfdata.startTime = Pa_GetStreamTime(stream );
-    err = startThread(&sfdata, threadFunctionWriteFile);
-    if( err != paNoError ) goto error;
-    
-#ifdef WIN32
-    if(!CreateTimerQueueTimer(&sfdata.hTimer, hTimerQueue,
-                              (WAITORTIMERCALLBACK) TimerCallback, &sfdata,200,200,0)) {
-        printf("failed to start timer (3).\n");
-        return 1;
-    }
-#endif
-	err = Pa_StartStream( stream );
-	if( err != paNoError ) 
-		goto error;
-    
-	printf("Hit CTRL-C to stop.\n");
-    
-    while((!sfdata.finished) && file_recording){
-        // nothing to do!
-		Pa_Sleep(10);
-	}
-    // note to programmer: any bug in audio buffer arithmetic will likely cause crash here!
-    err = Pa_StopStream( stream );
-    if( err != paNoError ) {
-        printf("Error stopping stream\n");
-		goto error;
-    }
-    
-    // need to stop thread explicitly?
-    
-    err = Pa_CloseStream( stream ); 
-	if( err != paNoError ) {
-        printf("Error closing stream\n");
-		goto error;
-    }
-#ifdef WIN32
-    CloseHandle(ghEvent);
-    DeleteTimerQueue(hTimerQueue);
-#endif
-    printf("%.2f secs\n",(float)(sfdata.lastTime));
-	fflush(stdout);
-	printf("Recording finished.\n");
-    res = psf_sndReadPeaks(sfdata.ofd,fpeaks,(MYLONG *) &lpeaktime);
-	if(res==0) {
-		printf("no PEAK data in this soundfile\n");
-	}
-	else if(res < 0){
-		printf("Error reading PEAK data\n");
-		goto error;
-	}
-	else{
-        // creation time not available until file closed; don't need it here!
-		printf("PEAK data:\n");
-		for(i=0;i < sfdata.chans;i++){
-            if(fpeaks[i].val > 0.0){
-                double dBval = 20.0 * log10(fpeaks[i].val);
-			    printf("CH %d: %.4f (%.2lfdB) at frame %u: \t%.4f secs\n",
-                   i,fpeaks[i].val,dBval,fpeaks[i].pos,(double)(fpeaks[i].pos / (double) props.srate));
-            }
-            else {
-                printf("CH %d: %.4f (-infdB)at frame %u: \t%.4f secs\n",
-                   i,fpeaks[i].val,fpeaks[i].pos,(double)(fpeaks[i].pos / (double) props.srate));
-
-            }
-		}
-	}
-    
-error:
-	Pa_Terminate();
-    
-//#ifdef _WIN32
-//    CloseHandle(ghEvent);
-//    DeleteTimerQueue(hTimerQueue);
-//#endif
-    if( sfdata.ringbufData )       
-        PaUtil_FreeMemory(sfdata.ringbufData);
-    
-	if(sfdata.ofd >=0)
-		psf_sndClose(sfdata.ofd);
-	if(fpeaks)
-		free(fpeaks);
-
-	psf_finish();
-	
-	return 0;    
-    
-}
-
-
-
-int show_devices(void)
-{
-    int i,j;
-    PaDeviceIndex numDevices,p;
-    const    PaDeviceInfo *pdi;
-#ifdef _WIN32
-    const PaHostApiInfo* api_info;
-    const char *apiname;
-#endif
-    PaError  err;
-    int nInputDevices = 0;
-    
-#ifdef USE_ASIO
-    printf("For ASIO multi-channel, you may need to select the highest device no.\n");		
-#endif
-    /*Pa_Initialize();*/
-    numDevices =  Pa_GetDeviceCount();
-    if( numDevices < 0 )
-    {
-        printf("ERROR: Pa_CountDevices returned 0x%x\n", numDevices );
-        err = numDevices;
-        return err;
-    }
-#ifdef WIN32
-    printf("Driver\tDevice\tInput\tOutput\tName\n");
-#else
-    printf("Device\tInput\tOutput\tName\n");
-#endif
-    //printf("Number of devices = %d\n", numDevices );
-    for( p=0; p<numDevices; p++ )
-    {
-        pdi = Pa_GetDeviceInfo( p );
-        
-        //#ifdef _WIN32
-        //			/*RWD: skip, info on inputs */
-        //			if(pdi->maxOutputChannels == 0)
-        //				continue;
-        //#endif
-        nInputDevices++;
-        
-        if( p == Pa_GetDefaultInputDevice() ) 
-            printf("*");
-        else
-            printf(" ");
-#ifdef WIN32
-        api_info = Pa_GetHostApiInfo(pdi->hostApi);
-        apiname = api_info->name;
-        if(strcmp(apiname,"Windows DirectSound")==0)
-            apiname = "DS ";
-        printf("(%s)\t%d\t%d\t%d\t%s\n",apiname,p,
-               pdi->maxInputChannels,
-               pdi->maxOutputChannels,
-               pdi->name);	
-#else
-        printf("%d\t%d\t%d\t%s\n",p,
-               pdi->maxInputChannels,
-               pdi->maxOutputChannels,
-               pdi->name);
-        
-        
-#endif
-    }
-    
-    return 0;
-}
-

dev/externals/portsf/CMakeLists.txt

@@ -1,14 +0,0 @@
-if(APPLE)
-  set(CMAKE_C_FLAGS "-O2 -Wall -mmacosx-version-min=10.5 -arch i386 -arch x86_64 -Dunix")
-else()
-  if(MINGW)
-    set(CMAKE_C_FLAGS "-O2")
-  else()
-    set(CMAKE_C_FLAGS "-O2 -Dunix -Dlinux")
-  endif()
-endif()
-
-include_directories(../include)
-set(PORTSF_SRCS portsf.c ieee80.c)
-
-add_library(portsf ${PORTSF_SRCS})

dev/externals/portsf/ieee80.c

@@ -1,149 +0,0 @@
-/***************************************************************\
-*   IEEE80.c							*
-*   Convert between "double" and IEEE 80 bit format  		*
-*   in machine independent manner.				*
-*   Assumes array of char is a continuous region of 8 bit frames*
-*   Assumes (unsigned long) has 32 useable bits			*
-*   billg, dpwe @media.mit.edu					*
-*   01aug91							*
-*   19aug91  aldel/dpwe  workaround top bit problem in Ultrix   *
-*                        cc's double->ulong cast		*
-*   05feb92  dpwe/billg  workaround top bit problem in 		*
-*                        THINKC4 + 68881 casting		*
-\***************************************************************/
-#include <stdio.h>
-#include <stdlib.h>
-#include 	<math.h>
-
-#include	"ieee80.h"
-
-/* #define MAIN	1	 to compile test routines */
-
-#define ULPOW2TO31	((unsigned long)0x80000000L)
-#define DPOW2TO31	((double)2147483648.0)	/* 2^31 */
-
-/* have to deal with ulong's 32nd bit conditionally as double<->ulong casts 
-   don't work in some C compilers */
-
-static double   myUlongToDouble (unsigned long ul);
-static unsigned long myDoubleToUlong (double val);
-
-static double myUlongToDouble(unsigned long ul)
-{
-	double val;
-	
-	/* in THINK_C, ulong -> double apparently goes via long, so can only 
-	   apply to 31 bit numbers.  If 32nd bit is set, explicitly add on its
-	   value */
-	if(ul & ULPOW2TO31)
-		val = DPOW2TO31 + (ul & (~ULPOW2TO31));
-	else
-		val = ul;
-	return val;
-}
-
-static unsigned long myDoubleToUlong(double	val)
-{
-	unsigned long ul;
-	
-	/* cannot cast negative numbers into unsigned longs */
-	if(val < 0)	
-	{ 
-		fprintf(stderr,"IEEE80:DoubleToUlong: val < 0\n"); 
-	}
-	
-	/* in ultrix 4.1's cc, double -> unsigned long loses the top bit, 
-	   so we do the conversion only on the bottom 31 bits and set the 
-	   last one by hand, if val is truly that big */
-	/* should maybe test for val > (double)(unsigned long)0xFFFFFFFF ? */
-	if(val < DPOW2TO31)
-		ul = (unsigned long)val;
-	else
-		ul = ULPOW2TO31 | (unsigned long)(val-DPOW2TO31);
-	return ul;
-}
-
-
-/*
-* Convert IEEE 80 bit floating point to double.
-* Should be portable to all C compilers.
-*/
-double ieee_80_to_double(unsigned char *p)
-{
-	char sign;
-	short exp = 0;
-	unsigned long mant1 = 0;
-	unsigned long 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++;
-	
-	/* special test for all bits zero meaning zero 
-	   - else pow(2,-16383) bombs */
-	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;
-	}
-}
-
-/*
-* Convert double to IEEE 80 bit floating point
-* Should be portable to all C compilers.
-* 19aug91 aldel/dpwe  covered for MSB bug in Ultrix 'cc'
-*/
-
-void double_to_ieee_80(double val,unsigned char *p)
-{
-	char sign = 0;
-	short exp = 0;
-	unsigned long mant1 = 0;
-	unsigned long 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++ = (char)(0xFF & (mant1>>24));
-	*p++ = (char)(0xFF & (mant1>>16));
-	*p++ = (char)(0xFF & (mant1>> 8));
-	*p++ = (char)(0xFF & (mant1));
-	*p++ = (char)(0xFF & (mant0>>24));
-	*p++ = (char)(0xFF & (mant0>>16));
-	*p++ = (char)(0xFF & (mant0>> 8));
-	*p++ = (char)(0xFF & (mant0));
-	
-}

dev/externals/portsf/ieee80.h

@@ -1,17 +0,0 @@
-
-/***************************************************************\
-*   IEEE80.h							*
-*   Convert between "double" and IEEE 80 bit format  		*
-*   in machine independent manner.				*
-*   Assumes array of char is a continuous region of 8 bit frames*
-*   Assumes (unsigned long) has 32 useable bits			*
-*   billg, dpwe @media.mit.edu					*
-*   01aug91							*
-*   19aug91  aldel/dpwe  workaround top bit problem in Ultrix   *
-*                        cc's double->ulong cast		*
-*   05feb92  dpwe/billg  workaround top bit problem in 		*
-*                        THINKC4 + 68881 casting		*
-\***************************************************************/
-
-double ieee_80_to_double(unsigned char *p);
-void   double_to_ieee_80(double val, unsigned char *p);

dev/externals/portsf/portsf.c

@@ -1,3573 +0,0 @@
-/* Copyright (c) 1999-2009 Richard Dobson
- 
- Permission is hereby granted, free of charge, to any person
- obtaining a copy of this software and associated documentation
- files (the "Software"), to deal in the Software without
- restriction, including without limitation the rights to use,
- copy, modify, merge, publish, distribute, sublicense, and/or sell
- copies of the Software, and to permit persons to whom the
- Software is furnished to do so, subject to the following
- conditions:
- 
- The above copyright notice and this permission notice shall be
- included in all copies or substantial portions of the Software.
- 
- THE SOFTWARE IS PROVIDED "AS IS", WITHOUT WARRANTY OF ANY KIND,
- EXPRESS OR IMPLIED, INCLUDING BUT NOT LIMITED TO THE WARRANTIES
- OF MERCHANTABILITY, FITNESS FOR A PARTICULAR PURPOSE AND
- NONINFRINGEMENT. IN NO EVENT SHALL THE AUTHORS OR COPYRIGHT
- HOLDERS BE LIABLE FOR ANY CLAIM, DAMAGES OR OTHER LIABILITY,
- WHETHER IN AN ACTION OF CONTRACT, TORT OR OTHERWISE, ARISING
- FROM, OUT OF OR IN CONNECTION WITH THE SOFTWARE OR THE USE OR
- OTHER DEALINGS IN THE SOFTWARE.
- */
-
-/* Changes Oct 28 2002: support overwrite mode (seek-back->read->wrtite) */
-/* fix bug in floats WAVE header reading! */
-/* more comprehensive use of assert in _DEBUG mode (still not complete ....) */
-/* sundry tidy-ups inspired by Intel compiler messages! */
-/* changes Apr 27 2003: fix reversal bug in wave format writing on big-endian platforms */
-/* changes Aug 2003: added basic TPDF dither to 16bit output */
-/* Nov 2003: fixed aiffsize bug in aiff write func: 8 bytes too big! */
-/* Jan 2004 fixed error reading from WAVE_EX file */
-
-/*Oct 19 2006: cleared up some memory leaks on SFFILE (psf_sndOpen etc)*/
-/* July 2009: revised to use fget/setpos, 4GB file support, 64bit platforms
- Move to 2LSB TPDF dither
- */
-
-/* POST BOOK!*/
-/* OCT 2009 ADDED MC_CUBE, completed bformat support */
-/* fixed bad omission in byteswaaping wavex elements! */
-/* added recognition of speaker layouts */
-/* corrected absent assignment of chmask on little-endian */
-
-/* Apr 2010 upped MAXFILES to 256! */
-/* Aug 2010, and now to 512! */
-
-/* Aug 2012 TODO (?):  add running check for 4GB limit to writeFrame funcs */
-/* Aug 2012 corrected SPKRS_MONO value (portsf.h) */
-/* Nov 2013: added SPKRS_6_1 to list */
-/* Mar 2020: fix fault parsing floats amb files! */
-
-#include <stdio.h>
-#ifdef unix
-#include <unistd.h>
-#endif
-#include <stdlib.h>
-#include <memory.h>
-#include <string.h>
-#include <math.h>
-#include <time.h>
-#include "ieee80.h"
-#ifdef _DEBUG
-#include <assert.h>
-#endif
-
-#include <portsf.h>
-
-#ifndef DBGFPRINTF
-# ifdef _DEBUG
-# define DBGFPRINTF(a) fprintf a
-# else
-# define DBGFPRINTF(a)
-# endif
-#endif
-
-#ifndef max
-#define max(x,y) ((x) > (y) ? (x) : (y))
-#endif
-#ifndef min
-#define min(x,y) ((x) < (y) ? (x) : (y))
-#endif
-
-#ifndef BITS_PER_BYTE
-#define BITS_PER_BYTE (8)
-#endif
-#ifndef WIN32
-#include <ctype.h>
-int stricmp(const char *a, const char *b);
-int strnicmp(const char *a, const char *b, const int length);
-#endif
-
-#ifdef linux
-#define POS64(x) (x.__pos)
-#else
-#define POS64(x) (x)
-#endif
-
-
-/* probably no good for 64bit platforms */
-#define REVDWBYTES(t)   ( (((t)&0xff) << 24) | (((t)&0xff00) << 8) | (((t)&0xff0000) >> 8) | (((t)>>24) & 0xff) )
-#define REVWBYTES(t)    ( (((t)&0xff) << 8) | (((t)>>8) &0xff) )
-#define TAG(a,b,c,d)    ( ((a)<<24) | ((b)<<16) | ((c)<<8) | (d) )
-/* RW changed 15:10:2002 - one less! */
-/* RWD Dec 2009 changed back again for bit-accurate scaling */
-#define MAX_16BIT  (32768.0)
-#define MAX_32BIT  (2147483648.0)
-
-#define AIFC_VERSION_1 (0xA2805140)
-/*pstring for AIFC  - includes the pad byte*/
-static const char aifc_floatstring[10] = { 0x08,'F','l','o','a','t',0x20,'3','2',0x00};
-static const char aifc_notcompressed[16] = {0x0e,'n','o','t',0x20,'c','o','m','p','r','e','s','s','e','d',0x00};
-
-static float trirand();
-static psf_channelformat get_speakerlayout(DWORD chmask,DWORD chans);
-static double inv_randmax  = 1.0 / RAND_MAX;
-static const float dclip16 = (float)(32767.0/32768.0);
-static const float dclip24 = (float)(8388607.0/8388608.0);
-static const float dclip32 = (float)(2147483647.0/2147483648.0);
-
-/* we need the standard Windows defs, when compiling on other platforms.
- <windows.h> defines _INC_WINDOWS
- */
-#ifndef _INC_WINDOWS
-
-#define WAVE_FORMAT_PCM         (0x0001)
-
-typedef union {
-    int lsamp;
-    float fsamp;
-    unsigned char bytes[4];
-} SND_SAMP;
-
-typedef struct _GUID 
-{ 
-    unsigned int      Data1;
-    unsigned short    Data2;
-    unsigned short    Data3;
-    unsigned char     Data4[8];
-} GUID; 
-
-
-typedef struct  {
-    WORD  wFormatTag;
-    WORD  nChannels;
-    DWORD nSamplesPerSec;
-    DWORD nAvgBytesPerSec;
-    WORD  nBlockAlign;
-    WORD  wBitsPerSample;
-    
-} WAVEFORMAT;
-
-
-typedef struct { 
-    WORD  wFormatTag;
-    WORD  nChannels;
-    DWORD nSamplesPerSec;
-    DWORD nAvgBytesPerSec;
-    WORD  nBlockAlign;
-    WORD  wBitsPerSample;
-    WORD  cbSize;
-} WAVEFORMATEX; 
-#endif
-
-
-/* basic support for WAVE_FORMAT_EXTENSIBLE */
-
-typedef struct {
-    WAVEFORMATEX    Format;             /* 18 bytes */
-    union {
-        WORD wValidBitsPerSample;       /* bits of precision  */
-        WORD wSamplesPerBlock;          /* valid if wBitsPerSample==0 */
-        WORD wReserved;                 /* If neither applies, set to */
-        /* zero. */
-    } Samples;
-    DWORD    dwChannelMask;             /* which channels are */
-    /* present in stream  */
-    GUID     SubFormat;
-} WAVEFORMATEXTENSIBLE;
-
-/* sizeof(WAVEFORMATEXTENSIBLE) gives size plus alignment padding; not good here */
-/* size = 18 + 2 + 4 + 16 */
-#define sizeof_WFMTEX  (40)
-
-/* std WAVE-EX GUIDS from <ksmedia.h> */
-static const GUID  KSDATAFORMAT_SUBTYPE_PCM = {0x00000001,0x0000,0x0010,
-    {0x80,
-        0x00,
-        0x00,
-        0xaa,
-        0x00,
-        0x38,
-        0x9b,
-        0x71}};
-
-static const GUID  KSDATAFORMAT_SUBTYPE_IEEE_FLOAT = {0x00000003,0x0000,0x0010,
-    {0x80,
-        0x00,
-        0x00,
-        0xaa,
-        0x00,
-        0x38,
-        0x9b,
-        0x71}};
-
-static const GUID SUBTYPE_AMBISONIC_B_FORMAT_PCM = { 0x00000001, 0x0721, 0x11d3, 
-    { 0x86,
-        0x44,
-        0xc8,
-        0xc1,
-        0xca,
-        0x0,
-        0x0,
-        0x0 } };
-
-
-static const GUID SUBTYPE_AMBISONIC_B_FORMAT_IEEE_FLOAT = { 0x00000003, 0x0721, 0x11d3, 
-    { 0x86,
-        0x44,
-        0xc8,
-        0xc1,
-        0xca,
-        0x0,
-        0x0,
-        0x0 } };
-
-#ifndef WAVE_FORMAT_IEEE_FLOAT
-#define WAVE_FORMAT_IEEE_FLOAT  (0x0003)
-#endif
-#ifndef WAVE_FORMAT_EXTENSIBLE
-#define WAVE_FORMAT_EXTENSIBLE  (0xfffe)
-#endif
-/*RWD Feb 2010 */
-#define WARNSTRING_SIZE (64)
-/******** the private structure holding all sfile stuff */
-enum lastop {PSF_OP_READ,PSF_OP_WRITE};
-
-typedef struct psffile {
-    FILE            *file;
-    char            *filename;
-    DWORD           curframepos;    /* for read operations */
-    DWORD           nFrames;        /* multi-channel sample frames */
-    int             isRead;         /* how we are using it */
-    int             clip_floats;
-    int             rescale;
-    float           rescale_fac;
-    psf_format      riff_format;
-    /*int               isSeekable;*/    /* any use ? */
-    int             is_little_endian;
-    psf_stype       samptype;       /* = nBlockAlign / nChannels */
-    fpos_t          dataoffset;     /* = sizeof(header) */
-    fpos_t          fmtoffset;
-    fpos_t          peakoffset;
-    WAVEFORMATEXTENSIBLE fmt;           /* serves all WAVE,AIFF support.*/
-    psf_channelformat chformat;
-    PSF_CHPEAK      *pPeaks;
-    time_t          peaktime;
-    fpos_t          lastwritepos;
-    int             lastop;         /* last op was read or write? */
-    int             dithertype;
-    /* RWD Feb 2010 ; to warn user of ill-formed files*/
-    int             illformed;
-    char            warnstring[WARNSTRING_SIZE];
-} PSFFILE;
-
-
-static int compare_guids(const GUID *gleft, const GUID *gright)
-{
-    const char *left = (const char *) gleft, *right = (const char *) gright;
-    return !memcmp(left,right,sizeof(GUID));
-}
-
-
-
-#define psf_maxfiles (512)
-
-/* could make this dynamically allocated, via psf_init, one day, if it matters. */
-
-static PSFFILE *psf_files[psf_maxfiles];
-
-/* return 0 for success, non-zero for error */
-int psf_init(void)
-{
-    int i;
-    
-    for(i=0;i < psf_maxfiles;i++)
-        psf_files[i] = NULL;
-    /* do any other inits we need.... */
-    return 0;
-}
-
-/* return zero for success, non-zero for error*/
-static int psf_release_file(PSFFILE *psff)
-{
-    int rc = 0;
-#ifdef _DEBUG
-    assert(psff);
-#endif
-    
-    if(psff->file){
-        rc = fclose(psff->file);
-        if(rc)
-            return rc;
-        psff->file = NULL;
-    }
-    if(psff->filename){
-        free(psff->filename);
-        psff->filename = NULL;
-    }
-    if(psff->pPeaks) {
-        free(psff->pPeaks);
-        psff->pPeaks = NULL;
-    }
-    return rc;
-}
-
-/* return zero for success, non-zero for error*/
-int psf_finish(void)
-{
-    int i,rc = 0;
-    for(i=0;i < psf_maxfiles;i++) {
-        if(psf_files[i]!= NULL){
-#ifdef _DEBUG
-            printf("sfile %s not closed: closing.\n",psf_files[i]->filename);
-#endif
-            rc = psf_release_file(psf_files[i]);
-            /* an alternative is to continue, and write error info to a logfile */
-            if(rc)
-                return rc;
-            
-        }
-        free(psf_files[i]);
-        psf_files[i] = NULL;
-    }
-    return rc;
-}
-
-
-/* thanks to the SNDAN programmers for this! */
-/* return 0 for big-endian machine, 1 for little-endian machine*/
-/* probably no good for 16bit swapping though */
-static int byte_order()                 
-{                           
-    int   one = 1;
-    char* endptr = (char *) &one;
-    return (*endptr);
-}
-
-
-static void fmtSwapBytes(PSFFILE *sfdat)
-{
-    WAVEFORMATEX  *pfmt = (WAVEFORMATEX *) &(sfdat->fmt.Format);
-    
-    pfmt->wFormatTag    = (WORD) REVWBYTES(pfmt->wFormatTag);
-    pfmt->nChannels     = (WORD) REVWBYTES(pfmt->nChannels);
-    pfmt->nSamplesPerSec    = REVDWBYTES(pfmt->nSamplesPerSec);
-    pfmt->nAvgBytesPerSec   = REVDWBYTES(pfmt->nAvgBytesPerSec);
-    pfmt->nBlockAlign   = (WORD) REVWBYTES(pfmt->nBlockAlign);
-    pfmt->wBitsPerSample    = (WORD) REVWBYTES(pfmt->wBitsPerSample);
-}
-
-static void fmtExSwapBytes(PSFFILE *sfdat)
-{
-    WAVEFORMATEXTENSIBLE  *pfmtEx =  &(sfdat->fmt);
-    WAVEFORMATEX          *pfmt   = &(pfmtEx->Format);
-    
-    pfmt->wFormatTag    = (WORD) REVWBYTES(pfmt->wFormatTag);
-    pfmt->nChannels     = (WORD) REVWBYTES(pfmt->nChannels);
-    pfmt->nSamplesPerSec    = REVDWBYTES(pfmt->nSamplesPerSec);
-    pfmt->nAvgBytesPerSec   = REVDWBYTES(pfmt->nAvgBytesPerSec);
-    pfmt->nBlockAlign   = (WORD) REVWBYTES(pfmt->nBlockAlign);
-    pfmt->wBitsPerSample    = (WORD) REVWBYTES(pfmt->wBitsPerSample);
-    pfmt->cbSize            = (WORD) REVWBYTES(pfmt->cbSize);
-    // OCT 09: missing!
-    pfmtEx->Samples.wValidBitsPerSample = (WORD) REVWBYTES(pfmtEx->Samples.wValidBitsPerSample);
-    pfmtEx->dwChannelMask     = (DWORD) REVDWBYTES(pfmtEx->dwChannelMask);
-    /* we swap numeric fields of GUID, but not the char string */
-    pfmtEx->SubFormat.Data1 = REVDWBYTES(pfmtEx->SubFormat.Data1);
-    pfmtEx->SubFormat.Data2 = (WORD) REVWBYTES(pfmtEx->SubFormat.Data2);
-    pfmtEx->SubFormat.Data3 = (WORD) REVWBYTES(pfmtEx->SubFormat.Data3);
-}
-
-static int check_guid(PSFFILE *sfdat)
-{
-    /* expects a GUID to be loaded already into sfdat.*/
-    if(sfdat->riff_format != PSF_WAVE_EX)
-        return 1;
-    
-    if(compare_guids(&(sfdat->fmt.SubFormat),&(KSDATAFORMAT_SUBTYPE_PCM))){
-        switch(sfdat->fmt.Format.wBitsPerSample){
-            case(16):
-                sfdat->samptype = PSF_SAMP_16;
-                break;
-            case(24):
-                /* only support packed format for now */
-                if((sfdat->fmt.Format.nBlockAlign / sfdat->fmt.Format.nChannels) != 3){
-                    sfdat->samptype = PSF_SAMP_UNKNOWN;
-                    return 1;
-                }
-                sfdat->samptype = PSF_SAMP_24;
-                break;
-            case(32):
-                sfdat->samptype = PSF_SAMP_32;
-                break;
-            default:
-                sfdat->samptype = PSF_SAMP_UNKNOWN;
-                return 1;
-        }
-        return 0;
-    }
-    if(compare_guids(&(sfdat->fmt.SubFormat),&(KSDATAFORMAT_SUBTYPE_IEEE_FLOAT)))
-        if(sfdat->fmt.Format.wBitsPerSample == 32) {
-            sfdat->samptype = PSF_SAMP_IEEE_FLOAT;
-            return 0;
-        }
-    /* add other recognised GUIDs here... */
-    if(compare_guids(&(sfdat->fmt.SubFormat),&(SUBTYPE_AMBISONIC_B_FORMAT_IEEE_FLOAT)))
-        if(sfdat->fmt.Format.wBitsPerSample == 32) {
-            sfdat->samptype = PSF_SAMP_IEEE_FLOAT;
-            sfdat->chformat = MC_BFMT;
-            return 0;
-        }
-    if(compare_guids(&(sfdat->fmt.SubFormat),&(SUBTYPE_AMBISONIC_B_FORMAT_PCM))) {
-        switch(sfdat->fmt.Format.wBitsPerSample){
-            case(16):
-                sfdat->samptype = PSF_SAMP_16;
-                break;
-            case(24):
-                /* only support packed format for now */
-                if((sfdat->fmt.Format.nBlockAlign / sfdat->fmt.Format.nChannels) != 3){
-                    sfdat->samptype = PSF_SAMP_UNKNOWN;
-                    return 1;
-                }
-                sfdat->samptype = PSF_SAMP_24;
-                break;
-            case(32):
-                sfdat->samptype = PSF_SAMP_32;
-                break;
-            default:
-                sfdat->samptype = PSF_SAMP_UNKNOWN;
-                return 1;
-        }
-        sfdat->chformat = MC_BFMT;
-        return 0;
-    }
-    return 1;
-}
-/* return actual validbits */
-static int psf_bitsize(psf_stype type)
-{
-    int size = 0;
-    switch(type){
-        case(PSF_SAMP_16):
-            size = 16;
-            break;
-        case (PSF_SAMP_24):
-            size = 24;
-            break;
-        case(PSF_SAMP_32):
-        case(PSF_SAMP_IEEE_FLOAT):
-            size = 32;
-            break;
-        default:
-            break;
-    }
-    return size;
-}
-/* return sample size in bytes */
-static int psf_wordsize(psf_stype type)
-{
-    int size = 0;
-    switch(type){
-        case(PSF_SAMP_16):
-            size = 2;
-            break;
-        case (PSF_SAMP_24):
-            size = 3;
-            break;
-        case(PSF_SAMP_32):
-        case(PSF_SAMP_IEEE_FLOAT):
-            size = 4;
-            break;
-        default:
-            break;
-    }
-    return size;
-    
-}
-
-
-#if defined _WIN32 && defined _MSC_VER
-/* fast convergent rounding */
-__inline long psf_round(double fval)
-{
-    int result;
-    _asm{
-        fld fval
-        fistp   result
-        mov eax,result
-    }
-    return result;
-}
-
-#else
-/* slow convergent rounding ! */
-/* TODO: implement IEEE round-to-even */
-long psf_round(double val);
-
-long psf_round(double val)
-{
-    long k;
-    k = (long)(fabs(val)+0.5);
-    if(val < 0.0)
-        k = -k;
-    return k;
-}
-#endif
-
-#ifndef WIN32
-int stricmp(const char *a, const char *b)
-{
-    while(*a != '\0' && *b != '\0') {
-        int ca = islower(*a) ? toupper(*a) : *a;
-        int cb = islower(*b) ? toupper(*b) : *b;
-        
-        if(ca < cb)
-            return -1;
-        if(ca > cb)
-            return 1;
-        
-        a++;
-        b++;
-    }
-    if(*a == '\0' && *b == '\0')
-        return 0;
-    if(*a != '\0')
-        return 1;
-    return -1;
-}
-
-int
-strnicmp(const char *a, const char *b, const int length)
-{
-    int len = length;
-    
-    while(*a != '\0' && *b != '\0') {
-        int ca = islower(*a) ? toupper(*a) : *a;
-        int cb = islower(*b) ? toupper(*b) : *b;
-        
-        if(len-- < 1)
-            return 0;
-        
-        if(ca < cb)
-            return -1;
-        if(ca > cb)
-            return 1;
-        
-        a++;
-        b++;
-    }
-    if(*a == '\0' && *b == '\0')
-        return 0;
-    if(*a != '\0')
-        return 1;
-    return -1;
-}
-#endif
-
-/* create a new soundfile, from input props, or with default format  if props==NULL */
-/* current default = sr 44100, ch 1, WAVE, 16bit */
-/* could have func to define a new default format...*/
-
-static PSFFILE *psf_newFile(const PSF_PROPS *props)
-{
-    PSFFILE *sfdat;
-    
-    if(props){
-        if(props->srate <=0)
-            return NULL;
-        if(props->chans <=0)
-            return NULL;
-        /* NO support for PSF_SAMP_8 yet...*/
-        if(props->samptype < PSF_SAMP_16 || props->samptype > PSF_SAMP_IEEE_FLOAT)
-            return NULL;
-        if(props->format    <= PSF_FMT_UNKNOWN || props->format > PSF_AIFC)
-            return NULL;
-        if(props->chformat < STDWAVE || props->chformat > MC_WAVE_EX)
-            return NULL;
-    }
-    
-    
-    sfdat = (PSFFILE *) malloc(sizeof(PSFFILE));
-    if(sfdat==NULL)
-        return sfdat;
-    
-    POS64(sfdat->lastwritepos)      = 0;
-    sfdat->file         = NULL;
-    sfdat->filename         = NULL;
-    sfdat->nFrames          = 0;
-    sfdat->curframepos      = 0;
-    sfdat->isRead           = 1;                /* OK. who knows?    */
-    /* or use platform default format.... */
-    sfdat->riff_format      = props ? props->format : PSF_STDWAVE;      /* almost certainly! */
-    /*sfdat->isSeekable     = 1;*/
-    sfdat->clip_floats      = 1;
-    sfdat->rescale          = 0;
-    sfdat->rescale_fac      = 1.0f;
-    sfdat->is_little_endian = byte_order();
-    sfdat->samptype         = props ? props->samptype : PSF_SAMP_16;        /* reasonable...     */
-    POS64(sfdat->dataoffset)        = 0;
-    POS64(sfdat->fmtoffset)     = 0;
-    POS64(sfdat->peakoffset)        = 0;
-    sfdat->chformat         = props ? props->chformat : STDWAVE;
-    /*setup Format */
-    if(props)
-        sfdat->fmt.Format.wFormatTag  = (WORD) (props->samptype == PSF_SAMP_IEEE_FLOAT ?  WAVE_FORMAT_IEEE_FLOAT : WAVE_FORMAT_PCM);
-    else
-        sfdat->fmt.Format.wFormatTag  = WAVE_FORMAT_PCM;
-    sfdat->fmt.Format.nChannels       = (WORD) (props ?  props->chans : 1);
-    sfdat->fmt.Format.nSamplesPerSec  = props ? props->srate : 44100;
-    sfdat->fmt.Format.nBlockAlign     = (WORD) (props  ?  sfdat->fmt.Format.nChannels * psf_wordsize(props->samptype) : sfdat->fmt.Format.nChannels * sizeof(short));
-    sfdat->fmt.Format.wBitsPerSample  = (WORD) (props ?  psf_bitsize(props->samptype)  : sizeof(short) * BITS_PER_BYTE);
-    sfdat->fmt.Format.nAvgBytesPerSec = sfdat->fmt.Format.nSamplesPerSec
-    *sfdat->fmt.Format.nChannels
-    * (sfdat->fmt.Format.wBitsPerSample / BITS_PER_BYTE);
-    sfdat->pPeaks           = NULL;
-    sfdat->peaktime         = 0;
-    sfdat->fmt.Format.cbSize = 0;
-    /* set initial defaults for WAVE-EX stuff; may change */
-    /* but nobody should look at these fields unless we have a real WAVE-EX file anyway... */
-    sfdat->fmt.dwChannelMask = SPKRS_UNASSIGNED;
-    sfdat->fmt.Samples.wValidBitsPerSample  = sfdat->fmt.Format.wBitsPerSample;
-    /* 0 should be a guaranteed non-valid GUID! */
-    memset((char *) &(sfdat->fmt.SubFormat),0,sizeof(GUID));
-    
-    if(props && (props->format == PSF_WAVE_EX)) {
-        sfdat->fmt.Format.cbSize = 22;
-        /* NB we will set the GUID from wFormatTag in waveExWriteHeader() */
-        /* should really flag an error if user sets this */
-        if(sfdat->chformat==STDWAVE)
-            sfdat->chformat = MC_STD;
-        
-        /* set wavex speaker mask */
-        /* TODO: support custom speaker masks, wordsizes, etc */
-        switch(sfdat->chformat){
-            case MC_MONO:
-                if(props->chans != 1){
-                    //rsferrstr = "conflicting channel configuration for WAVE-EX file";
-                    free(sfdat);
-                    return NULL;
-                }
-                sfdat->fmt.dwChannelMask = SPKRS_MONO;
-                break;
-            case MC_STEREO:
-                if(props->chans != 2){
-                    //rsferrstr = "conflicting channel configuration for WAVE-EX file";
-                    free(sfdat);
-                    return NULL;
-                }
-                sfdat->fmt.dwChannelMask = SPKRS_STEREO;
-                break;
-            case MC_QUAD:
-                if(props->chans != 4){
-                    //rsferrstr = "conflicting channel configuration for WAVE-EX file";
-                    free(sfdat);
-                    return NULL;
-                }
-                sfdat->fmt.dwChannelMask = SPKRS_GENERIC_QUAD;
-                break;
-            case MC_LCRS:
-                if(props->chans != 4){
-                    free(sfdat);
-                    return NULL;
-                }
-                sfdat->fmt.dwChannelMask = SPKRS_SURROUND_LCRS;
-                break;
-                
-            case MC_DOLBY_5_1:
-                if(props->chans != 6){
-                    //rsferrstr = "conflicting channel configuration for WAVE-EX file";
-                    free(sfdat);
-                    return NULL;
-                }
-                sfdat->fmt.dwChannelMask = SPKRS_DOLBY5_1;
-                break;
-            case MC_SURR_5_0:
-                if(props->chans != 5){
-                    //rsferrstr = "conflicting channel configuration for WAVE-EX file";
-                    free(sfdat);
-                    return NULL;
-                }
-                sfdat->fmt.dwChannelMask = SPKRS_SURR_5_0;
-                break;
-            case MC_SURR_6_1:
-                if(props->chans != 7){
-                    //rsferrstr = "conflicting channel configuration for WAVE-EX file";
-                    free(sfdat);
-                    return NULL;
-                }
-                sfdat->fmt.dwChannelMask = SPKRS_6_1;
-                break;
-            case MC_SURR_7_1:
-                if(props->chans != 8){
-                    //rsferrstr = "conflicting channel configuration for WAVE-EX file";
-                    free(sfdat);
-                    return NULL;
-                }
-                sfdat->fmt.dwChannelMask = SPKRS_7_1;
-                break;
-            case MC_CUBE:
-                if(props->chans != 8){
-                    //rsferrstr = "conflicting channel configuration for WAVE-EX file";
-                    free(sfdat);
-                    return NULL;
-                }
-                sfdat->fmt.dwChannelMask = SPKRS_CUBE;
-                break;
-            default:
-                /*MC_STD, MC_BFMT */
-                sfdat->fmt.dwChannelMask = SPKRS_UNASSIGNED;
-                break;
-        }
-    }
-    /* no dither, by default */
-    sfdat->dithertype = PSF_DITHER_OFF;
-    sfdat->illformed = 0;                   /*RWD Feb 2010 */
-    memset(sfdat->warnstring,0,WARNSTRING_SIZE);
-    return sfdat;
-}
-
-int psf_getWarning(int sfd,const char** warnstring)
-{
-    int retval = 0;
-    PSFFILE *sfdat;
-    if(sfd < 0 || sfd > psf_maxfiles)
-        retval = PSF_E_BADARG;
-    sfdat  = psf_files[sfd];
-    
-    if(sfdat->illformed==0)
-        retval = 0;
-    else{
-        *warnstring = sfdat->warnstring;
-        retval =  1;
-    }
-    return retval;
-}
-/* complete header before closing file; return PSF_E_NOERROR[= 0] on success */
-static int wavUpdate(PSFFILE *sfdat)
-{
-    DWORD riffsize,datasize;
-    fpos_t bytepos;
-#ifdef _DEBUG
-    assert(sfdat);
-    assert(sfdat->file);
-    assert(POS64(sfdat->dataoffset) != 0);
-#endif      
-    POS64(bytepos) = sizeof(int);
-    if((fsetpos(sfdat->file,&bytepos))==0) {
-        riffsize = (sfdat->nFrames * sfdat->fmt.Format.nBlockAlign) +  (MYLONG) POS64(sfdat->dataoffset);
-        riffsize -= 2 * sizeof(DWORD);
-        if(!sfdat->is_little_endian)
-            riffsize = REVDWBYTES(riffsize);
-        if(fwrite((char *) &riffsize,sizeof(int),1,sfdat->file) != 1)
-            return PSF_E_CANT_WRITE;
-    }
-    else
-        return PSF_E_CANT_SEEK;
-    if(sfdat->pPeaks){
-        if(POS64(sfdat->peakoffset)==0)
-            return PSF_E_BADARG;
-        
-        /*do byterev if necessary...*/
-        if((fsetpos(sfdat->file,&sfdat->peakoffset))==0){
-            /*set current time*/
-            DWORD *pblock;
-            int i;
-            time_t now = time(0);
-            if(!sfdat->is_little_endian){
-                now = REVDWBYTES(now);
-                pblock = (DWORD *) (sfdat->pPeaks);
-                for(i=0;i < sfdat->fmt.Format.nChannels * 2; i++)
-                    pblock[i] = REVDWBYTES(pblock[i]);
-            }
-            if((fwrite((char*)&now,sizeof(DWORD),1,sfdat->file)) != 1)
-                return PSF_E_CANT_WRITE;
-            
-            if((fwrite((char *) (sfdat->pPeaks),sizeof(PSF_CHPEAK),sfdat->fmt.Format.nChannels,sfdat->file))
-               != sfdat->fmt.Format.nChannels )
-                return PSF_E_CANT_WRITE;
-        }
-        else
-            return PSF_E_CANT_SEEK;
-    }
-    POS64(bytepos) = POS64(sfdat->dataoffset) -  sizeof(int);
-    if((fsetpos(sfdat->file,&bytepos))==0) {
-        datasize = sfdat->nFrames * sfdat->fmt.Format.nBlockAlign;
-        if(!sfdat->is_little_endian)
-            datasize = REVDWBYTES(datasize);
-        if(fwrite((char *) & datasize,sizeof(DWORD),1,sfdat->file) != 1)
-            return PSF_E_CANT_WRITE;
-    }
-    if(fseek(sfdat->file,0,SEEK_END)){
-        /*DBGFPRINTF((stderr,"wavUpdate: error reseeking to end of file\n"));*/
-        return PSF_E_CANT_SEEK;
-    }
-    
-    return PSF_E_NOERROR;
-}
-
-/* ditto for AIFF... */
-
-/* NB: the AIFF spec is unclear on type of size field. We decide on unsigned long (DWORD) here;
- on the principle that a COMM chunk with an unsigned long nSampleFrames really needs the
- chunk size to be unsigned long too!.
- 
- */
-static int aiffUpdate(PSFFILE *sfdat)
-{
-    DWORD aiffsize,datasize,rev_datasize,frames;
-    fpos_t bytepos,filesize;
-    unsigned char pad = 0x00;
-    
-    if(sfdat==NULL || sfdat->file== NULL)
-        return PSF_E_BADARG;
-    
-    if(POS64(sfdat->dataoffset)  == 0)
-        return PSF_E_BADARG;
-    POS64(bytepos) = sizeof(int);
-    if((fsetpos(sfdat->file,&bytepos))==0) {
-        /* RWD 26:10:2002 */
-        aiffsize = (sfdat->nFrames * sfdat->fmt.Format.nBlockAlign)
-        + (MYLONG) POS64(sfdat->dataoffset);
-        // need to count any needed pad byte
-        aiffsize += (aiffsize % 2);
-        // deduct 8 bytes for FORM<size>
-        aiffsize -= 2 * sizeof(DWORD);
-        if(sfdat->is_little_endian)
-            aiffsize = REVDWBYTES(aiffsize);
-        if(fwrite((char *) &aiffsize,sizeof(DWORD),1,sfdat->file) != 1)
-            return PSF_E_CANT_WRITE;
-    }
-    else
-        return PSF_E_CANT_SEEK;
-    POS64(bytepos)  = POS64(sfdat->fmtoffset) + sizeof(WORD);
-    if((fsetpos(sfdat->file,&bytepos))==0) {
-        frames = sfdat->nFrames;
-        if(sfdat->is_little_endian)
-            frames = REVDWBYTES(frames);
-        if(fwrite((char *) &frames,sizeof(DWORD),1,sfdat->file) != 1)
-            return PSF_E_CANT_WRITE;
-    }
-    else
-        return PSF_E_CANT_SEEK;
-    if(sfdat->pPeaks){
-        if(POS64(sfdat->peakoffset)==0)
-            return PSF_E_BADARG;
-        
-        /*do byterev if necessary...*/
-        if((fsetpos(sfdat->file,&sfdat->peakoffset))==0){
-            /*set current time*/
-            DWORD *pblock;
-            int i;
-            time_t now = time(0);
-            if(sfdat->is_little_endian){
-                now = REVDWBYTES(now);
-                pblock = (DWORD *) (sfdat->pPeaks);
-                for(i=0;i < sfdat->fmt.Format.nChannels * 2; i++)
-                    pblock[i] = REVDWBYTES(pblock[i]);
-            }
-            if((fwrite((char*)&now,sizeof(DWORD),1,sfdat->file)) != 1)
-                return PSF_E_CANT_WRITE;
-            
-            if((fwrite((char *) (sfdat->pPeaks),sizeof(PSF_CHPEAK),sfdat->fmt.Format.nChannels,sfdat->file))
-               != sfdat->fmt.Format.nChannels )
-                return PSF_E_CANT_WRITE;
-        }
-        else
-            return PSF_E_CANT_SEEK;
-    }
-    POS64(bytepos) = POS64(sfdat->dataoffset) - (3 * sizeof(int));
-    if((fsetpos(sfdat->file,&bytepos))==0) {
-        datasize = sfdat->nFrames * sfdat->fmt.Format.nBlockAlign;
-        datasize += 2* sizeof(DWORD);   /* add offset and blocksize fields */
-        rev_datasize = datasize; /* preserve this for the seek later on */
-        if(sfdat->is_little_endian)
-            rev_datasize = REVDWBYTES(datasize);
-        if(fwrite((char *) & rev_datasize,sizeof(DWORD),1,sfdat->file) != 1)
-            return PSF_E_CANT_WRITE;
-    }
-    else
-        return PSF_E_CANT_SEEK;
-    /* datachunk needs added pad byte if odd, not included in saved chunksize*/
-    POS64(bytepos) = POS64(sfdat->dataoffset) + datasize;
-    if((fsetpos(sfdat->file,&bytepos))){
-        return PSF_E_CANT_SEEK;
-    }
-    if(fgetpos(sfdat->file,&filesize))
-        return PSF_E_CANT_SEEK;
-#ifdef _DEBUG   
-    assert(POS64(filesize) == POS64(bytepos));
-#endif  
-    if(POS64(filesize) % 2)
-        if(fwrite(&pad,sizeof(unsigned char),1,sfdat->file) != 1)
-            return PSF_E_CANT_WRITE;
-    
-    return PSF_E_NOERROR;
-}
-
-
-/* internal write func: return 0 for success */
-static int wavDoWrite(PSFFILE *sfdat, const void* buf, DWORD nBytes)
-{
-    
-    DWORD written = 0;
-    if(sfdat==NULL || buf==NULL)
-        return PSF_E_BADARG;
-    
-    if(sfdat->file==NULL)
-        return PSF_E_CANT_WRITE;
-    
-    if((written = fwrite(buf,sizeof(char),nBytes,sfdat->file)) != nBytes) {
-        DBGFPRINTF((stderr, "wavDoWrite: wanted %d got %d.\n",
-                    (int) nBytes,(int) written));
-        return PSF_E_CANT_WRITE;
-    }
-    sfdat->lastop  = PSF_OP_WRITE;
-    return PSF_E_NOERROR;
-}
-
-static int wavDoRead(PSFFILE *sfdat, void* buf, DWORD nBytes)
-{
-    
-    DWORD got = 0;
-    if(sfdat==NULL || buf==NULL)
-        return PSF_E_BADARG;
-    
-    if(sfdat->file==NULL)
-        return PSF_E_CANT_READ;
-    
-    if((got = fread(buf,sizeof(char),nBytes,sfdat->file)) != nBytes) {
-        DBGFPRINTF((stderr, "wavDoRead: wanted %d got %d.\n",
-                    (int) nBytes,(int) got));
-        return PSF_E_CANT_READ;
-    }
-    sfdat->lastop = PSF_OP_READ;
-    return PSF_E_NOERROR;
-    
-}
-
-/* write PEAK chunk if we have the data */
-static int wavWriteHeader(PSFFILE *sfdat)
-{
-    DWORD tag,size;
-    WORD cbSize = 0;
-    WAVEFORMATEX *pfmt;
-    PSF_CHPEAK *peaks;
-    fpos_t bytepos;
-#ifdef _DEBUG
-    assert(sfdat);
-    assert(sfdat->file);
-    assert(sfdat->riff_format == PSF_STDWAVE);
-    assert(sfdat->nFrames == 0);
-    assert(!sfdat->isRead);
-    assert(sfdat->fmt.Format.nChannels != 0);
-#endif
-    
-    /*clear pPeaks array*/
-    if(sfdat->pPeaks)
-        memset((char *)sfdat->pPeaks,0,sizeof(PSF_CHPEAK) * sfdat->fmt.Format.nChannels);
-    
-    
-    tag = TAG('R','I','F','F');
-    size = 0;
-    if(!sfdat->is_little_endian)
-        size = REVDWBYTES(size);
-    else
-        tag = REVDWBYTES(tag);
-    
-    
-    if(wavDoWrite(sfdat,(char *)&tag,sizeof(DWORD))
-       || wavDoWrite(sfdat,(char *)&size,sizeof(DWORD)))
-        return PSF_E_CANT_WRITE;
-    
-    tag = TAG('W','A','V','E');
-    if(sfdat->is_little_endian)
-        tag = REVDWBYTES(tag);
-    if(wavDoWrite(sfdat,(char *)&tag,sizeof(DWORD)))
-        return PSF_E_CANT_WRITE;
-    
-    pfmt = &(sfdat->fmt.Format);
-    
-    tag = TAG('f','m','t',' ');
-    size = sizeof(WAVEFORMAT);
-    if(sfdat->samptype==PSF_SAMP_IEEE_FLOAT)
-        size += sizeof(WORD);         /* for cbSize: WAVEOFRMATEX */
-    if(!sfdat->is_little_endian){
-        size = REVDWBYTES(size);
-        fmtSwapBytes(sfdat);
-    }
-    else
-        tag = REVDWBYTES(tag);
-    if(wavDoWrite(sfdat,(char *)&tag,sizeof(DWORD))
-       || wavDoWrite(sfdat,(char *)&size,sizeof(DWORD)))
-        return PSF_E_CANT_WRITE;
-    if(fgetpos(sfdat->file,&bytepos))
-        return PSF_E_CANT_SEEK;
-    sfdat->fmtoffset = bytepos;
-    
-    if(wavDoWrite(sfdat,(char *)pfmt,sizeof(WAVEFORMAT)))
-        return PSF_E_CANT_WRITE;
-    /*add cbSize if floatsams */
-    if(sfdat->samptype==PSF_SAMP_IEEE_FLOAT)
-        if(wavDoWrite(sfdat,(char *)&cbSize,sizeof(WORD)))
-            return PSF_E_CANT_WRITE;
-    /* reswap it all */
-    if(!sfdat->is_little_endian){
-        fmtSwapBytes(sfdat);
-    }
-    
-    if(sfdat->pPeaks){
-        DWORD version = 1, now = 0;
-        peaks = sfdat->pPeaks;
-        
-        tag = TAG('P','E','A','K');
-        size = 2 * sizeof(DWORD) + sizeof(PSF_CHPEAK) * pfmt->nChannels;
-        if(!sfdat->is_little_endian){
-            size = REVDWBYTES(size);
-            version  = REVDWBYTES(version);
-        }
-        else
-            tag = REVDWBYTES(tag);
-        
-        if(wavDoWrite(sfdat,(char *)&tag,sizeof(DWORD))
-           || wavDoWrite(sfdat,(char *)&size,sizeof(DWORD))
-           || wavDoWrite(sfdat,(char *)&version,sizeof(DWORD)))
-            return PSF_E_CANT_WRITE;
-        if(fgetpos(sfdat->file,&bytepos))
-            return PSF_E_CANT_SEEK;
-        sfdat->peakoffset = bytepos;  /*we need to update time*/
-        
-        if(wavDoWrite(sfdat,(char *) &now,sizeof(DWORD))
-           || wavDoWrite(sfdat,(char *) peaks, sizeof(PSF_CHPEAK) * sfdat->fmt.Format.nChannels))
-            return PSF_E_CANT_WRITE;
-    }
-    tag = TAG('d','a','t','a');
-    size = 0;
-    if(sfdat->is_little_endian)
-        tag = REVDWBYTES(tag);
-    if(wavDoWrite(sfdat,(char *)&tag,sizeof(DWORD))
-       || wavDoWrite(sfdat,(char *)&size,sizeof(DWORD)))
-        return PSF_E_CANT_WRITE;
-    if(fgetpos(sfdat->file,&bytepos))
-        return PSF_E_CANT_SEEK;
-    sfdat->dataoffset = bytepos;
-    return PSF_E_NOERROR;
-}
-
-
-static int waveExWriteHeader(PSFFILE *sfdat)
-{
-    DWORD tag,size;
-    WAVEFORMATEXTENSIBLE *pfmt;
-    PSF_CHPEAK *peaks;
-    GUID *pGuid = NULL;
-    fpos_t bytepos;
-#ifdef _DEBUG
-    assert(sfdat);
-    assert(sfdat->file);
-    assert(sfdat->chformat > STDWAVE);
-    assert(sfdat->nFrames==0);
-    assert(!sfdat->isRead);
-    assert(sfdat->fmt.Format.nChannels != 0);
-#endif
-    /*clear pPeaks array*/
-    if(sfdat->pPeaks)
-        memset((char *)sfdat->pPeaks,0,sizeof(PSF_CHPEAK) * sfdat->fmt.Format.nChannels);
-    /* complete WAVE-EX format fields: */
-    if(sfdat->chformat==MC_BFMT){
-        if(sfdat->samptype== PSF_SAMP_IEEE_FLOAT){
-            pGuid = (GUID *)  &SUBTYPE_AMBISONIC_B_FORMAT_IEEE_FLOAT;
-        }
-        else{
-            pGuid =(GUID *) &SUBTYPE_AMBISONIC_B_FORMAT_PCM;
-        }
-        
-    }else {
-        if(sfdat->fmt.Format.wFormatTag==  WAVE_FORMAT_IEEE_FLOAT){
-            pGuid = (GUID *)  &KSDATAFORMAT_SUBTYPE_IEEE_FLOAT;
-        }
-        else{
-            pGuid =(GUID *) &KSDATAFORMAT_SUBTYPE_PCM;
-        }
-    }
-    sfdat->fmt.Format.wFormatTag = WAVE_FORMAT_EXTENSIBLE;
-    memcpy((char *) &(sfdat->fmt.SubFormat),(char *)pGuid,sizeof(GUID));
-    tag = TAG('R','I','F','F');
-    size = 0;
-    if(!sfdat->is_little_endian)
-        size = REVDWBYTES(size);
-    else
-        tag = REVDWBYTES(tag);
-    if(wavDoWrite(sfdat,(char *)&tag,sizeof(DWORD))
-       || wavDoWrite(sfdat,(char *)&size,sizeof(DWORD)))
-        return PSF_E_CANT_WRITE;
-    tag = TAG('W','A','V','E');
-    if(sfdat->is_little_endian)
-        tag = REVDWBYTES(tag);
-    if(wavDoWrite(sfdat,(char *)&tag,sizeof(DWORD)))
-        return PSF_E_CANT_WRITE;
-    pfmt = &(sfdat->fmt);
-    tag = TAG('f','m','t',' ');
-    size = sizeof_WFMTEX;
-    if(!sfdat->is_little_endian){
-        size = REVDWBYTES(size);
-        fmtExSwapBytes(sfdat);
-    }
-    else
-        tag = REVDWBYTES(tag);
-    if(wavDoWrite(sfdat,(char *)&tag,sizeof(DWORD))
-       || wavDoWrite(sfdat,(char *)&size,sizeof(DWORD)))
-        return PSF_E_CANT_WRITE;
-    if(fgetpos(sfdat->file,&bytepos))
-        return PSF_E_CANT_SEEK;
-    sfdat->fmtoffset = bytepos;
-    /* write fmt elementwise, to avoid C alignment traps with WORD */
-    /* 16byte format...*/
-    if(wavDoWrite(sfdat,(char *)pfmt,sizeof(WAVEFORMAT))
-       /** cbSize... */
-       ||wavDoWrite(sfdat,(char *) &(pfmt->Format.cbSize),sizeof(WORD))
-       /* validbits... */
-       ||wavDoWrite(sfdat,(char *) &(pfmt->Samples.wValidBitsPerSample),sizeof(WORD))
-       /* ChannelMask .... */
-       ||wavDoWrite(sfdat,(char *) &(pfmt->dwChannelMask),sizeof(DWORD))
-       /*  and the GUID */
-       ||wavDoWrite(sfdat,(char *) &(pfmt->SubFormat),sizeof(GUID)))
-        return PSF_E_CANT_WRITE;
-    /* reswap it all */
-    if(!sfdat->is_little_endian){
-        fmtExSwapBytes(sfdat);
-    }
-    if(sfdat->pPeaks){
-        DWORD version = 1, now = 0;
-        
-        peaks = sfdat->pPeaks;
-        tag = TAG('P','E','A','K');
-        size = 2 * sizeof(DWORD) + sizeof(PSF_CHPEAK) * pfmt->Format.nChannels;
-        if(!sfdat->is_little_endian){
-            size = REVDWBYTES(size);
-            version  = REVDWBYTES(version);
-        }
-        else
-            tag = REVDWBYTES(tag);
-        if(wavDoWrite(sfdat,(char *)&tag,sizeof(DWORD))
-           || wavDoWrite(sfdat,(char *)&size,sizeof(DWORD))
-           || wavDoWrite(sfdat,(char *)&version,sizeof(DWORD)))
-            return PSF_E_CANT_WRITE;
-        if(fgetpos(sfdat->file,&bytepos))
-            return PSF_E_CANT_SEEK;
-        sfdat->peakoffset = bytepos;  /*we need to update time*/
-        if(wavDoWrite(sfdat,(char *) &now,sizeof(DWORD))
-           || wavDoWrite(sfdat,(char *) peaks, sizeof(PSF_CHPEAK) * sfdat->fmt.Format.nChannels))
-            return PSF_E_CANT_WRITE;
-    }
-    
-    tag = TAG('d','a','t','a');
-    size = 0;
-    if(sfdat->is_little_endian)
-        tag = REVDWBYTES(tag);
-    if(wavDoWrite(sfdat,(char *)&tag,sizeof(DWORD))
-       || wavDoWrite(sfdat,(char *)&size,sizeof(DWORD)))
-        return PSF_E_CANT_WRITE;
-    if(fgetpos(sfdat->file,&bytepos))
-        return PSF_E_CANT_SEEK;
-    sfdat->dataoffset = bytepos;
-    return PSF_E_NOERROR;
-}
-
-static int aiffWriteHeader(PSFFILE *sfdat)
-{
-    DWORD tag,size;
-    PSF_CHPEAK *peaks;
-    DWORD dwData,offset,blocksize;
-    unsigned char ieee[10];
-    WORD wData;
-    fpos_t bytepos;
-#ifdef _DEBUG
-    assert(sfdat);
-    assert(sfdat->file);
-    assert(sfdat->chformat == STDWAVE);
-    assert(sfdat->nFrames==0);
-    assert(!sfdat->isRead);
-    assert(sfdat->riff_format == PSF_AIFF);
-    assert(sfdat->fmt.Format.nChannels != 0);
-#endif
-    
-    /*clear pPeaks array*/
-    if(sfdat->pPeaks)
-        memset((char *)sfdat->pPeaks,0,sizeof(PSF_CHPEAK) * sfdat->fmt.Format.nChannels);
-    
-    tag = TAG('F','O','R','M');
-    size = 0;
-    
-    if(sfdat->is_little_endian) {
-        size = REVDWBYTES(size);
-        tag = REVDWBYTES(tag);
-    }
-    if(wavDoWrite(sfdat,(char *)&tag,sizeof(DWORD))
-       || wavDoWrite(sfdat,(char *)&size,sizeof(DWORD)))
-        return PSF_E_CANT_WRITE;
-    tag = TAG('A','I','F','F');
-    if(sfdat->is_little_endian)
-        tag = REVDWBYTES(tag);
-    if(wavDoWrite(sfdat,(char *)&tag,sizeof(DWORD)))
-        return PSF_E_CANT_WRITE;
-    
-    tag = TAG('C','O','M','M');
-    size = 18;
-    if(sfdat->is_little_endian){
-        size = REVDWBYTES(size);
-        tag = REVDWBYTES(tag);
-    }
-    if(wavDoWrite(sfdat,(char *)&tag,sizeof(DWORD))
-       || wavDoWrite(sfdat,(char *)&size,sizeof(DWORD)))
-        return PSF_E_CANT_WRITE;
-    if(fgetpos(sfdat->file,&bytepos))
-        return PSF_E_CANT_SEEK;
-    sfdat->fmtoffset = bytepos;
-    
-    wData = sfdat->fmt.Format.nChannels;
-    if(sfdat->is_little_endian)
-        wData = (WORD) REVWBYTES(wData);
-    if(wavDoWrite(sfdat,(char *)&wData,sizeof(WORD)))
-        return PSF_E_CANT_WRITE;
-    
-    dwData = 0;         /* nFrames */
-    if(wavDoWrite(sfdat,(char *)&dwData,sizeof(DWORD)))
-        return PSF_E_CANT_WRITE;
-    
-    wData = sfdat->fmt.Format.wBitsPerSample;
-    if(sfdat->is_little_endian)
-        wData = (WORD) REVWBYTES(wData);
-    if(wavDoWrite(sfdat,(char *)&wData,sizeof(WORD)))
-        return PSF_E_CANT_WRITE;
-    double_to_ieee_80((double)sfdat->fmt.Format.nSamplesPerSec,ieee);
-    
-    if(wavDoWrite(sfdat,ieee,10))
-        return PSF_E_CANT_WRITE;
-    if(sfdat->pPeaks){
-        DWORD version = 1, now = 0;
-        peaks = sfdat->pPeaks;
-        
-        tag = TAG('P','E','A','K');
-        size = 2 * sizeof(DWORD) + sizeof(PSF_CHPEAK) * sfdat->fmt.Format.nChannels;
-        if(sfdat->is_little_endian){
-            size = REVDWBYTES(size);
-            version  = REVDWBYTES(version);
-            tag = REVDWBYTES(tag);
-        }
-        if(wavDoWrite(sfdat,(char *)&tag,sizeof(DWORD))
-           || wavDoWrite(sfdat,(char *)&size,sizeof(DWORD))
-           || wavDoWrite(sfdat,(char *)&version,sizeof(DWORD)))
-            return PSF_E_CANT_WRITE;
-        if(fgetpos(sfdat->file,&bytepos))
-            return PSF_E_CANT_SEEK;
-        sfdat->peakoffset = bytepos;  /*we need to update time*/
-        
-        if(wavDoWrite(sfdat,(char *) &now,sizeof(DWORD))
-           || wavDoWrite(sfdat,(char *) peaks, sizeof(PSF_CHPEAK) * sfdat->fmt.Format.nChannels))
-            return PSF_E_CANT_WRITE;
-    }
-    
-    tag = TAG('S','S','N','D');
-    size = offset = blocksize = 0;
-    if(sfdat->is_little_endian)
-        tag = REVDWBYTES(tag);
-    if(wavDoWrite(sfdat,(char *)&tag,sizeof(DWORD))
-       || wavDoWrite(sfdat,(char *)&size,sizeof(DWORD))
-       || wavDoWrite(sfdat,(char *)&offset,sizeof(DWORD))
-       || wavDoWrite(sfdat,(char *)&blocksize,sizeof(DWORD)))
-        return PSF_E_CANT_WRITE;
-    if(fgetpos(sfdat->file,&bytepos))
-        return PSF_E_CANT_SEEK;
-    sfdat->dataoffset = bytepos;
-    return PSF_E_NOERROR;
-}
-
-
-static int aifcWriteHeader(PSFFILE *sfdat)
-{
-    DWORD tag,size;
-    PSF_CHPEAK *peaks;
-    DWORD dwData,offset,blocksize,aifcver = AIFC_VERSION_1,ID_compression;
-    /*assume 32bit floats, but we may be asked to use aifc for integer formats too*/
-    char *str_compressed = (char *) aifc_floatstring;
-    int pstring_size = 10;
-    unsigned char ieee[10];
-    WORD wData;
-    fpos_t bytepos;
-    
-#ifdef _DEBUG
-    assert(sfdat);
-    assert(sfdat->file);
-    assert(sfdat->nFrames==0);
-    assert(!sfdat->isRead);
-    assert(sfdat->riff_format == PSF_AIFC);
-    assert(sfdat->fmt.Format.nChannels != 0);
-#endif
-    
-    if(sfdat->samptype==PSF_SAMP_IEEE_FLOAT)
-        ID_compression = TAG('f','l','3','2');
-    else {
-        ID_compression = TAG('N','O','N','E');
-        pstring_size = 16;
-        str_compressed = (char *) aifc_notcompressed;
-    }
-    /*clear pPeaks array*/
-    if(sfdat->pPeaks)
-        memset((char *)sfdat->pPeaks,0,sizeof(PSF_CHPEAK) * sfdat->fmt.Format.nChannels);
-    
-    tag = TAG('F','O','R','M');
-    size = 0;
-    
-    if(sfdat->is_little_endian) {
-        size = REVDWBYTES(size);
-        tag = REVDWBYTES(tag);
-    }
-    if(wavDoWrite(sfdat,(char *)&tag,sizeof(DWORD))
-       || wavDoWrite(sfdat,(char *)&size,sizeof(DWORD)))
-        return PSF_E_CANT_WRITE;
-    tag = TAG('A','I','F','C');
-    if(sfdat->is_little_endian)
-        tag = REVDWBYTES(tag);
-    if(wavDoWrite(sfdat,(char *)&tag,sizeof(DWORD)))
-        return PSF_E_CANT_WRITE;
-    
-    tag = TAG('F','V','E','R');
-    size = sizeof(DWORD);
-    if(sfdat->is_little_endian){
-        size = REVDWBYTES(size);
-        tag = REVDWBYTES(tag);
-        aifcver = REVDWBYTES(aifcver);
-    }
-    if(wavDoWrite(sfdat,(char *)&tag,sizeof(DWORD))
-       || wavDoWrite(sfdat,(char *)&size,sizeof(DWORD))
-       || wavDoWrite(sfdat,(char *)&aifcver,sizeof(DWORD)))
-        return PSF_E_CANT_WRITE;
-    
-    tag = TAG('C','O','M','M');
-    size = 22 + pstring_size;
-    if(sfdat->is_little_endian){
-        size = REVDWBYTES(size);
-        tag = REVDWBYTES(tag);
-    }
-    if(wavDoWrite(sfdat,(char *)&tag,sizeof(DWORD))
-       || wavDoWrite(sfdat,(char *)&size,sizeof(DWORD)))
-        return PSF_E_CANT_WRITE;
-    if(fgetpos(sfdat->file,&bytepos))
-        return PSF_E_CANT_SEEK;
-    sfdat->fmtoffset = bytepos;
-    
-    wData = sfdat->fmt.Format.nChannels;
-    if(sfdat->is_little_endian)
-        wData = (WORD) REVWBYTES(wData);
-    if(wavDoWrite(sfdat,(char *)&wData,sizeof(WORD)))
-        return PSF_E_CANT_WRITE;
-    
-    dwData = 0;         /* nFrames */
-    if(wavDoWrite(sfdat,(char *)&dwData,sizeof(DWORD)))
-        return PSF_E_CANT_WRITE;
-    
-    wData = sfdat->fmt.Format.wBitsPerSample;
-    if(sfdat->is_little_endian)
-        wData = (WORD) REVWBYTES(wData);
-    if(wavDoWrite(sfdat,(char *)&wData,sizeof(WORD)))
-        return PSF_E_CANT_WRITE;
-    double_to_ieee_80((double)sfdat->fmt.Format.nSamplesPerSec,ieee);
-    
-    if(wavDoWrite(sfdat,ieee,10))
-        return PSF_E_CANT_WRITE;
-    /*AIFC bits */
-    if(sfdat->is_little_endian)
-        ID_compression = REVDWBYTES(ID_compression);
-    if(wavDoWrite(sfdat,(char *)&ID_compression,sizeof(DWORD)))
-        return PSF_E_CANT_WRITE;
-    if(wavDoWrite(sfdat,str_compressed,pstring_size))
-        return PSF_E_CANT_WRITE;
-    
-    if(sfdat->pPeaks){
-        DWORD version = 1, now = 0;
-        
-        peaks = sfdat->pPeaks;
-        tag = TAG('P','E','A','K');
-        size = 2 * sizeof(DWORD) + sizeof(PSF_CHPEAK) * sfdat->fmt.Format.nChannels;
-        if(sfdat->is_little_endian){
-            size = REVDWBYTES(size);
-            version  = REVDWBYTES(version);
-            tag = REVDWBYTES(tag);
-        }
-        if(wavDoWrite(sfdat,(char *)&tag,sizeof(DWORD))
-           || wavDoWrite(sfdat,(char *)&size,sizeof(DWORD))
-           || wavDoWrite(sfdat,(char *)&version,sizeof(DWORD)))
-            return PSF_E_CANT_WRITE;
-        if(fgetpos(sfdat->file,&bytepos))
-            return PSF_E_CANT_SEEK;
-        sfdat->peakoffset = bytepos;  /*we need to update time*/
-        
-        if(wavDoWrite(sfdat,(char *) &now,sizeof(DWORD))
-           || wavDoWrite(sfdat,(char *) peaks, sizeof(PSF_CHPEAK) * sfdat->fmt.Format.nChannels))
-            return PSF_E_CANT_WRITE;
-    }
-    
-    tag = TAG('S','S','N','D');
-    size = offset = blocksize = 0;
-    if(sfdat->is_little_endian)
-        tag = REVDWBYTES(tag);
-    if(wavDoWrite(sfdat,(char *)&tag,sizeof(DWORD))
-       || wavDoWrite(sfdat,(char *)&size,sizeof(DWORD))
-       || wavDoWrite(sfdat,(char *)&offset,sizeof(DWORD))
-       || wavDoWrite(sfdat,(char *)&blocksize,sizeof(DWORD)))
-        return PSF_E_CANT_WRITE;
-    if(fgetpos(sfdat->file,&bytepos))
-        return PSF_E_CANT_SEEK;
-    sfdat->dataoffset = bytepos;
-    return PSF_E_NOERROR;
-}
-
-/* create soundfile. return descriptor, or some PSF_error value < 0 */
-/* supported clipping or non-clipping of floats to 0dbFS, 
- minimum header (or PEAK), and RDWR or RDONLY (but last not implemented yet!) */
-/* we expect full format info to be set in props */
-/* I want to offer share-read access (easy with WIN32), but can't with  ANSI! */
-/* possible TODO:  enforce non-destructive by e.g. rejecting create on existing file */
-int psf_sndCreate(const char *path,const PSF_PROPS *props,int clip_floats,int minheader, int mode)
-{       
-    int i,rc = PSF_E_UNSUPPORTED;
-    psf_format fmt;
-    PSFFILE *sfdat;
-    char *fmtstr = "wb+";   /* default is READ+WRITE */
-    /*  disallow props = NULL here, until/unless I can offer mechanism to set default props via psf_init() */
-    if(path == NULL || props == NULL)
-        return PSF_E_BADARG;
-    
-    for(i=0; i < psf_maxfiles; i++) {
-        if(psf_files[i] == NULL)
-            break;
-    }
-    if(i==psf_maxfiles)
-        return PSF_E_TOOMANYFILES;
-    
-    sfdat = psf_newFile(props);
-    if(sfdat == NULL)
-        return PSF_E_NOMEM;
-    
-    sfdat->clip_floats = clip_floats;
-    fmt = psf_getFormatExt(path);
-    if(fmt==PSF_FMT_UNKNOWN)
-        return PSF_E_UNSUPPORTED;
-    if(sfdat->samptype == PSF_SAMP_UNKNOWN)
-        return PSF_E_BADARG;
-    
-    sfdat->filename = (char *) malloc(strlen(path)+1);
-    if(sfdat->filename==NULL) {
-        DBGFPRINTF((stderr, "wavOpenWrite: no memory for filename\n"));
-        return PSF_E_NOMEM;
-    }
-    if(!minheader){
-        sfdat->pPeaks = (PSF_CHPEAK *) malloc(sizeof(PSF_CHPEAK) * sfdat->fmt.Format.nChannels);
-        if(sfdat->pPeaks==NULL){
-            DBGFPRINTF((stderr, "wavOpenWrite: no memory for peak data\n"));
-            return PSF_E_NOMEM;
-        }
-    }
-    /*switch (mode).... */
-    if(mode==PSF_CREATE_WRONLY)
-        fmtstr = "wb";
-    /* deal with CREATE_TEMPORARY later on! */
-    if((sfdat->file = fopen(path,fmtstr))  == NULL) {
-        DBGFPRINTF((stderr, "wavOpenWrite: cannot create '%s'\n", path));
-        return PSF_E_CANT_OPEN;
-    }
-    
-    strcpy(sfdat->filename, path);
-    sfdat->isRead = 0;
-    sfdat->nFrames = 0;
-    /* force aif f/p data to go to aifc format */
-    if(sfdat->samptype==PSF_SAMP_IEEE_FLOAT && fmt==PSF_AIFF){
-        DBGFPRINTF((stderr, "Warning: writing floating point data in AIFC format\n"));
-        fmt= PSF_AIFC;
-    }
-    /* .wav extension can be either std WAVE or WAVE-EX */
-    if(fmt==PSF_STDWAVE){
-        if(props->format==PSF_WAVE_EX)
-            fmt = PSF_WAVE_EX;
-    }
-    sfdat->riff_format = fmt;
-    
-    switch(fmt){
-        case(PSF_STDWAVE):
-            rc = wavWriteHeader(sfdat);
-            break;
-        case(PSF_AIFF):
-            
-            rc = aiffWriteHeader(sfdat);
-            break;
-        case(PSF_AIFC):
-            
-            rc = aifcWriteHeader(sfdat);
-            break;
-        case (PSF_WAVE_EX):
-            rc = waveExWriteHeader(sfdat);
-            break;
-        default:
-            sfdat->riff_format = PSF_FMT_UNKNOWN;
-            /* RAW? */
-            break;
-    }
-    if(rc < PSF_E_NOERROR)
-        return rc;
-    psf_files[i] = sfdat;
-    return i;
-}
-
-/* snd close:  automatically completes PEAK data when writing */
-/* return 0 for success */
-int psf_sndClose(int sfd)
-{
-    int rc = PSF_E_NOERROR;
-    PSFFILE *sfdat;
-    
-    if(sfd < 0 || sfd > psf_maxfiles)
-        return PSF_E_BADARG;
-    sfdat  = psf_files[sfd];
-#ifdef _DEBUG       
-    assert(sfdat->file);
-    assert(sfdat->filename);
-#endif
-    if(sfdat==NULL || sfdat->file==NULL)
-        return PSF_E_BADARG;
-    if(!sfdat->isRead){
-        switch(sfdat->riff_format){
-            case(PSF_STDWAVE):
-            case(PSF_WAVE_EX):
-                rc = wavUpdate(sfdat);
-                break;
-            case(PSF_AIFF):
-            case(PSF_AIFC):
-                rc = aiffUpdate(sfdat);
-                break;
-            default:
-                rc = PSF_E_CANT_CLOSE;
-                break;
-        }
-    }
-    if(psf_release_file(sfdat))
-        rc = PSF_E_CANT_CLOSE;
-    else {
-        free(sfdat);
-        psf_files[sfd]= NULL;
-    }
-    return rc;
-}
-
-/* write floats (multi-channel) framebuf to whichever target format. tracks PEAK data.*/ 
-/* bend over backwards not to modify source data */
-/* returns nFrames, or errval < 0 */
-int psf_sndWriteFloatFrames(int sfd, const float *buf, DWORD nFrames)
-{
-    int chans,lsamp;
-    DWORD i;
-    int j,do_reverse;
-    const float *pbuf = buf;
-    float fsamp,absfsamp;
-    int do_shift = 1;
-    PSFFILE *sfdat;
-    SND_SAMP s_samp;
-    
-    if(sfd < 0 || sfd > psf_maxfiles)
-        return PSF_E_BADARG;
-    
-    sfdat  = psf_files[sfd];
-    
-#ifdef _DEBUG       
-    assert(sfdat->file);
-    assert(sfdat->filename);
-#endif
-    
-    if(buf==NULL)
-        return PSF_E_BADARG;
-    if(nFrames == 0)
-        return nFrames;
-    if(sfdat->isRead)
-        return PSF_E_FILE_READONLY;
-    chans = sfdat->fmt.Format.nChannels;
-    
-    switch(sfdat->riff_format){
-        case(PSF_STDWAVE):
-        case(PSF_WAVE_EX):
-            do_reverse = (sfdat->is_little_endian ? 0 : 1 );
-            do_shift = 1;
-            break;
-        case(PSF_AIFF):
-        case(PSF_AIFC):
-            do_reverse = (sfdat->is_little_endian ? 1 : 0 );
-            do_shift = 0;
-            break;
-        default:
-            return PSF_E_UNSUPPORTED;
-    }
-    if(sfdat->lastop  == PSF_OP_READ)
-        fflush(sfdat->file);
-    switch(sfdat->samptype){
-        case(PSF_SAMP_IEEE_FLOAT):
-            if(do_reverse){
-                for(i=0; i < nFrames; i++){
-                    for(j=0;j < chans; j++) {
-                        fsamp = *pbuf++;
-                        if(sfdat->clip_floats){
-                            fsamp = min(fsamp,1.0f);
-                            fsamp = max(fsamp,-1.0f);
-                        }
-                        absfsamp = (float) fabs((double)fsamp);
-                        if(sfdat->pPeaks && (sfdat->pPeaks[j].val < absfsamp)){
-                            sfdat->pPeaks[j].pos = sfdat->nFrames + i;
-                            sfdat->pPeaks[j].val = absfsamp;
-                        }
-                        // lsamp = * (int *) pbuf++;
-                        s_samp.fsamp = fsamp;
-                        lsamp = s_samp.lsamp;
-                        lsamp = REVDWBYTES(lsamp);
-                        if(wavDoWrite(sfdat,(char *) &lsamp,sizeof(int))){
-                            DBGFPRINTF((stderr, "wavOpenWrite: write error\n"));
-                            return PSF_E_CANT_WRITE;
-                        }
-                    }
-                }
-            }
-            else {
-                for(i=0; i < nFrames; i++, pbuf += chans){
-                    for(j=0;j < chans; j++) {
-                        fsamp = pbuf[j];
-                        if(sfdat->clip_floats){
-                            fsamp = min(fsamp,1.0f);
-                            fsamp = max(fsamp,-1.0f);
-                        }
-                        absfsamp = (float)fabs((double)fsamp);
-                        if(sfdat->pPeaks && (sfdat->pPeaks[j].val < absfsamp)){
-                            sfdat->pPeaks[j].pos = sfdat->nFrames + i;
-                            sfdat->pPeaks[j].val = absfsamp;
-                        }
-                    }
-                }
-                if(wavDoWrite(sfdat,(char *)buf,nFrames * chans * sizeof(float))){
-                    DBGFPRINTF((stderr, "wavOpenWrite: write error\n"));
-                    return PSF_E_CANT_WRITE;
-                }
-            }
-            break;
-        case(PSF_SAMP_16):
-            /* TODO: optimise all this with func pointers etc */
-            if(do_reverse){
-                short ssamp;
-                for(i=0; i < nFrames; i++){
-                    for(j=0;j < chans; j++) {
-                        fsamp = *buf++;
-                        /* clip now! we may have a flag to rescale first...one day */
-                        fsamp = min(fsamp,dclip16);
-                        fsamp = max(fsamp,-dclip16);
-                        absfsamp = (float) fabs((double)fsamp);
-                        if(sfdat->pPeaks && (sfdat->pPeaks[j].val < absfsamp)){
-                            sfdat->pPeaks[j].pos = sfdat->nFrames + i;
-                            sfdat->pPeaks[j].val = absfsamp;
-                        }
-                        if(sfdat->dithertype == PSF_DITHER_TPDF)
-                            ssamp = (short) psf_round(fsamp * 32766.0 + 2.0 * trirand());
-                        else
-                            ssamp = (short) psf_round(fsamp * MAX_16BIT);
-                        ssamp = (short) REVWBYTES(ssamp);
-                        if( wavDoWrite(sfdat,(char *) &ssamp,sizeof(short))){
-                            DBGFPRINTF((stderr, "wavOpenWrite: write error\n"));
-                            return PSF_E_CANT_WRITE;
-                        }
-                    }
-                }
-            }
-            else {
-                short ssamp;
-                for(i=0; i < nFrames; i++, buf += chans){
-                    for(j=0;j < chans; j++) {
-                        fsamp = buf[j];
-                        /* clip now! we may have a flag to rescale first...one day */
-                        fsamp = min(fsamp,dclip16);
-                        fsamp = max(fsamp,-dclip16);
-                        absfsamp = (float) fabs((double)fsamp);
-                        if(sfdat->pPeaks && (sfdat->pPeaks[j].val < absfsamp)){
-                            sfdat->pPeaks[j].pos = sfdat->nFrames + i;
-                            sfdat->pPeaks[j].val = absfsamp;
-                        }
-                        if(sfdat->dithertype == PSF_DITHER_TPDF)
-                            ssamp = (short) psf_round(fsamp * 32766.0 + 2.0 * trirand());
-                        else
-                            ssamp = (short) psf_round(fsamp * MAX_16BIT);
-                        
-                        if(wavDoWrite(sfdat,(char *) &ssamp,sizeof(short))){
-                            DBGFPRINTF((stderr, "wavOpenWrite: write error\n"));
-                            return PSF_E_CANT_WRITE;
-                        }
-                    }
-                }
-            }
-            break;
-        case(PSF_SAMP_24):
-            if(do_reverse){
-                for(i=0; i < nFrames; i++){
-                    for(j=0;j < chans; j++) {
-                        fsamp = *buf++;
-                        /* clip now! we may have a flag to rescale first...one day */
-                        fsamp = min(fsamp,dclip24);
-                        fsamp = max(fsamp,-dclip24);
-                        absfsamp = (float) fabs((double)fsamp);
-                        if(sfdat->pPeaks && (sfdat->pPeaks[j].val < absfsamp)){
-                            sfdat->pPeaks[j].pos = sfdat->nFrames + i;
-                            sfdat->pPeaks[j].val = absfsamp;
-                        }
-                        lsamp = psf_round(fsamp * (MAX_32BIT));
-                        lsamp = REVDWBYTES(lsamp);
-                        if(do_shift){
-                            if(sfdat->is_little_endian)
-                                lsamp >>= 8;
-                            else
-                                lsamp <<= 8;
-                        }
-                        if( wavDoWrite(sfdat,(char *) &lsamp,3)){
-                            DBGFPRINTF((stderr, "wavOpenWrite: write error\n"));
-                            return PSF_E_CANT_WRITE;
-                        }
-                    }
-                }
-            }
-            else {
-                for(i=0; i < nFrames; i++, buf += chans){
-                    for(j=0;j < chans; j++) {
-                        fsamp = buf[j];
-                        /* clip now! we may have a flag to rescale first...one day */
-                        fsamp = min(fsamp,dclip24);
-                        fsamp = max(fsamp,-dclip24);
-                        absfsamp = (float) fabs((double)fsamp);
-                        if(sfdat->pPeaks && (sfdat->pPeaks[j].val < absfsamp)){
-                            sfdat->pPeaks[j].pos = sfdat->nFrames + i;
-                            sfdat->pPeaks[j].val = absfsamp;
-                        }
-                        lsamp = psf_round(fsamp * (MAX_32BIT));
-                        if(do_shift){
-                            if(sfdat->is_little_endian)
-                                lsamp >>= 8;
-                            else
-                                lsamp <<= 8;
-                        }
-                        if(wavDoWrite(sfdat,(char *) &lsamp,3)){
-                            DBGFPRINTF((stderr, "wavOpenWrite: write error\n"));
-                            return PSF_E_CANT_WRITE;
-                        }
-                    }
-                }
-            }
-            break;
-        case(PSF_SAMP_32):
-            if(do_reverse){
-                for(i=0; i < nFrames; i++){
-                    for(j=0;j < chans; j++) {
-                        fsamp = *buf++;
-                        /* clip now! we may have a flag to rescale first...one day */
-                        fsamp = min(fsamp,dclip32);
-                        fsamp = max(fsamp,-dclip32);
-                        absfsamp = (float) fabs((double)fsamp);
-                        if(sfdat->pPeaks && (sfdat->pPeaks[j].val < absfsamp)){
-                            sfdat->pPeaks[j].pos = sfdat->nFrames + i;
-                            sfdat->pPeaks[j].val = absfsamp;
-                        }
-                        lsamp = psf_round(fsamp * (MAX_32BIT));
-                        lsamp = REVDWBYTES(lsamp);
-                        if( wavDoWrite(sfdat,(char *) &lsamp,sizeof(int))){
-                            DBGFPRINTF((stderr, "wavOpenWrite: write error\n"));
-                            return PSF_E_CANT_WRITE;
-                        }
-                    }
-                }
-            }
-            else {
-                for(i=0; i < nFrames; i++, buf += chans){
-                    for(j=0;j < chans; j++) {
-                        fsamp = buf[j];
-                        /* clip now! we may have a flag to rescale first...one day */
-                        fsamp = min(fsamp,dclip32);
-                        fsamp = max(fsamp,-dclip32);
-                        absfsamp = (float) fabs((double)fsamp);
-                        if(sfdat->pPeaks && (sfdat->pPeaks[j].val < absfsamp)){
-                            sfdat->pPeaks[j].pos = sfdat->nFrames + i;
-                            sfdat->pPeaks[j].val = absfsamp;
-                        }
-                        lsamp = psf_round(fsamp * (MAX_32BIT));
-                        
-                        if(wavDoWrite(sfdat,(char *) &lsamp,sizeof(int))){
-                            DBGFPRINTF((stderr, "wavOpenWrite: write error\n"));
-                            return PSF_E_CANT_WRITE;
-                        }
-                    }
-                }
-            }
-            break;
-        default:
-            DBGFPRINTF((stderr, "wavOpenWrite: unsupported sample format\n"));
-            return PSF_E_UNSUPPORTED;
-            
-    }
-    POS64(sfdat->lastwritepos) += nFrames;
-    sfdat->curframepos = (MYLONG) POS64(sfdat->lastwritepos);
-    sfdat->nFrames = max(sfdat->nFrames,(DWORD) POS64(sfdat->lastwritepos));
-    /*  fflush(sfdat->file); */ /* ? may need this if reading/seeking as well as  write, etc */
-    return nFrames;
-    
-}
-
-int psf_sndWriteDoubleFrames(int sfd, const double *buf, DWORD nFrames)
-{
-    int chans,lsamp;
-    DWORD i;
-    int j,do_reverse;
-    const double *pbuf = buf;
-    float fsamp,absfsamp;
-    PSFFILE *sfdat;
-    int do_shift = 1;
-    SND_SAMP s_samp;
-    
-    if(sfd < 0 || sfd > psf_maxfiles)
-        return PSF_E_BADARG;
-    
-    sfdat  = psf_files[sfd];
-#ifdef _DEBUG       
-    assert(sfdat->file);
-    assert(sfdat->filename);
-#endif
-    if(buf==NULL)
-        return PSF_E_BADARG;
-    if(nFrames == 0)
-        return nFrames;
-    if(sfdat->isRead)
-        return PSF_E_FILE_READONLY;
-    chans = sfdat->fmt.Format.nChannels;
-    
-    switch(sfdat->riff_format){
-        case(PSF_STDWAVE):
-        case(PSF_WAVE_EX):
-            do_reverse = (sfdat->is_little_endian ? 0 : 1 );
-            do_shift = 1;
-            break;
-        case(PSF_AIFF):
-        case(PSF_AIFC):
-            do_reverse = (sfdat->is_little_endian ? 1 : 0 );
-            do_shift = 0;
-            break;
-        default:
-            return PSF_E_UNSUPPORTED;
-    }
-    if(sfdat->lastop  == PSF_OP_READ)
-        fflush(sfdat->file);
-    switch(sfdat->samptype){
-        case(PSF_SAMP_IEEE_FLOAT):
-            if(do_reverse){
-                for(i=0; i < nFrames; i++){
-                    for(j=0;j < chans; j++) {
-                        fsamp = (float) *pbuf++;
-                        if(sfdat->clip_floats){
-                            fsamp = min(fsamp,1.0f);
-                            fsamp = max(fsamp,-1.0f);
-                        }
-                        absfsamp = (float) fabs((double)fsamp);
-                        if(sfdat->pPeaks && (sfdat->pPeaks[j].val < absfsamp)){
-                            sfdat->pPeaks[j].pos = sfdat->nFrames + i;
-                            sfdat->pPeaks[j].val = absfsamp;
-                        }
-                        s_samp.fsamp = fsamp;
-                        lsamp = s_samp.lsamp;
-                        //   lsamp = * (int *) &fsamp;
-                        lsamp = REVDWBYTES(lsamp);
-                        if(wavDoWrite(sfdat,(char *) &lsamp,sizeof(int))){
-                            DBGFPRINTF((stderr, "wavOpenWrite: write error\n"));
-                            return PSF_E_CANT_WRITE;
-                        }
-                    }
-                }
-            }
-            else {
-                for(i=0; i < nFrames; i++, pbuf += chans){
-                    for(j=0;j < chans; j++) {
-                        fsamp = (float) pbuf[j];
-                        if(sfdat->clip_floats){
-                            fsamp = min(fsamp,1.0f);
-                            fsamp = max(fsamp,-1.0f);
-                        }
-                        if(wavDoWrite(sfdat,(char*)&fsamp,sizeof(float))){
-                            DBGFPRINTF((stderr, "wavOpenWrite: write error\n"));
-                            return PSF_E_CANT_WRITE;
-                        }
-                        absfsamp = (float)fabs((double)fsamp);
-                        if(sfdat->pPeaks && (sfdat->pPeaks[j].val < absfsamp)){
-                            sfdat->pPeaks[j].pos = sfdat->nFrames + i;
-                            sfdat->pPeaks[j].val = absfsamp;
-                        }
-                    }
-                }
-            }
-            break;
-        case(PSF_SAMP_16):
-            /* TODO: optimise all this with func pointers etc */
-            if(do_reverse){
-                short ssamp;
-                for(i=0; i < nFrames; i++){
-                    for(j=0;j < chans; j++) {
-                        fsamp = (float)  *buf++;
-                        /* clip now! we may have a flag to rescale first...one day */
-                        fsamp = min(fsamp,dclip16);
-                        fsamp = max(fsamp,-dclip16);
-                        absfsamp = (float) fabs((double)fsamp);
-                        if(sfdat->pPeaks && (sfdat->pPeaks[j].val < absfsamp)){
-                            sfdat->pPeaks[j].pos = sfdat->nFrames + i;
-                            sfdat->pPeaks[j].val = absfsamp;
-                        }
-                        if(sfdat->dithertype == PSF_DITHER_TPDF)
-                            ssamp = (short) psf_round(fsamp * 32766.0 + 2.0 * trirand());
-                        else
-                            ssamp = (short) psf_round(fsamp * MAX_16BIT);
-                        ssamp = (short) REVWBYTES(ssamp);
-                        if( wavDoWrite(sfdat,(char *) &ssamp,sizeof(short))){
-                            DBGFPRINTF((stderr, "wavOpenWrite: write error\n"));
-                            return PSF_E_CANT_WRITE;
-                        }
-                    }
-                }
-            }
-            else {
-                short ssamp;
-                for(i=0; i < nFrames; i++, buf += chans){
-                    for(j=0;j < chans; j++) {
-                        fsamp = (float) buf[j];
-                        /* clip now! we may have a flag to rescale first...one day */
-                        fsamp = min(fsamp,dclip16);
-                        fsamp = max(fsamp,-dclip16);
-                        absfsamp = (float) fabs((double)fsamp);
-                        if(sfdat->pPeaks && (sfdat->pPeaks[j].val < absfsamp)){
-                            sfdat->pPeaks[j].pos = sfdat->nFrames + i;
-                            sfdat->pPeaks[j].val = absfsamp;
-                        }
-                        if(sfdat->dithertype == PSF_DITHER_TPDF)
-                            ssamp = (short) psf_round(fsamp * 32766.0 + 2.0 * trirand());
-                        else
-                            ssamp = (short) psf_round(fsamp * MAX_16BIT);
-                        
-                        if(wavDoWrite(sfdat,(char *) &ssamp,sizeof(short))){
-                            DBGFPRINTF((stderr, "wavOpenWrite: write error\n"));
-                            return PSF_E_CANT_WRITE;
-                        }
-                    }
-                }
-            }
-            break;
-        case(PSF_SAMP_24):
-            if(do_reverse){
-                for(i=0; i < nFrames; i++){
-                    for(j=0;j < chans; j++) {
-                        fsamp =(float)  *buf++;
-                        /* clip now! we may have a flag to rescale first...one day */
-                        fsamp = min(fsamp,dclip24);
-                        fsamp = max(fsamp,-dclip24);
-                        absfsamp = (float) fabs((double)fsamp);
-                        if(sfdat->pPeaks && (sfdat->pPeaks[j].val < absfsamp)){
-                            sfdat->pPeaks[j].pos = sfdat->nFrames + i;
-                            sfdat->pPeaks[j].val = absfsamp;
-                        }
-                        lsamp = psf_round(fsamp * MAX_32BIT);
-                        lsamp = REVDWBYTES(lsamp);
-                        if(do_shift){
-                            if(sfdat->is_little_endian)
-                                lsamp >>= 8;
-                            else
-                                lsamp <<= 8;
-                        }
-                        if( wavDoWrite(sfdat,(char *) &lsamp,3)){
-                            DBGFPRINTF((stderr, "wavOpenWrite: write error\n"));
-                            return PSF_E_CANT_WRITE;
-                        }
-                    }
-                }
-            }
-            else {
-                for(i=0; i < nFrames; i++, buf += chans){
-                    for(j=0;j < chans; j++) {
-                        fsamp = (float)  buf[j];
-                        /* clip now! we may have a flag to rescale first...one day */
-                        fsamp = min(fsamp,dclip24);
-                        fsamp = max(fsamp,-dclip24);
-                        absfsamp = (float) fabs((double)fsamp);
-                        if(sfdat->pPeaks && (sfdat->pPeaks[j].val < absfsamp)){
-                            sfdat->pPeaks[j].pos = sfdat->nFrames + i;
-                            sfdat->pPeaks[j].val = absfsamp;
-                        }
-                        lsamp = psf_round(fsamp * MAX_32BIT);
-                        if(do_shift){
-                            if(sfdat->is_little_endian)
-                                lsamp >>= 8;
-                            else
-                                lsamp <<= 8;
-                        }
-                        if(wavDoWrite(sfdat,(char *) &lsamp,3)){
-                            DBGFPRINTF((stderr, "wavOpenWrite: write error\n"));
-                            return PSF_E_CANT_WRITE;
-                        }
-                    }
-                }
-            }
-            break;
-        case(PSF_SAMP_32):
-            if(do_reverse){
-                for(i=0; i < nFrames; i++){
-                    for(j=0;j < chans; j++) {
-                        fsamp = (float) *buf++;
-                        /* clip now! we may have a flag to rescale first...one day */
-                        fsamp = min(fsamp,dclip32);
-                        fsamp = max(fsamp,-dclip32);
-                        absfsamp = (float) fabs((double)fsamp);
-                        if(sfdat->pPeaks && (sfdat->pPeaks[j].val < absfsamp)){
-                            sfdat->pPeaks[j].pos = sfdat->nFrames + i;
-                            sfdat->pPeaks[j].val = absfsamp;
-                        }
-                        lsamp = psf_round(fsamp * MAX_32BIT);
-                        lsamp = REVDWBYTES(lsamp);
-                        if( wavDoWrite(sfdat,(char *) &lsamp,sizeof(int))){
-                            DBGFPRINTF((stderr, "wavOpenWrite: write error\n"));
-                            return PSF_E_CANT_WRITE;
-                        }
-                    }
-                }
-            }
-            else {
-                for(i=0; i < nFrames; i++, buf += chans){
-                    for(j=0;j < chans; j++) {
-                        fsamp = (float) buf[j];
-                        /* clip now! we may have a flag to rescale first...one day */
-                        fsamp = min(fsamp,dclip32);
-                        fsamp = max(fsamp,-dclip32);
-                        absfsamp = (float) fabs((double)fsamp);
-                        if(sfdat->pPeaks && (sfdat->pPeaks[j].val < absfsamp)){
-                            sfdat->pPeaks[j].pos = sfdat->nFrames + i;
-                            sfdat->pPeaks[j].val = absfsamp;
-                        }
-                        lsamp = psf_round(fsamp * MAX_32BIT);
-                        
-                        if(wavDoWrite(sfdat,(char *) &lsamp,sizeof(int))){
-                            DBGFPRINTF((stderr, "wavOpenWrite: write error\n"));
-                            return PSF_E_CANT_WRITE;
-                        }
-                    }
-                }
-            }
-            break;
-        default:
-            DBGFPRINTF((stderr, "wavOpenWrite: unsupported sample format\n"));
-            return PSF_E_UNSUPPORTED;
-            
-    }
-    POS64(sfdat->lastwritepos) += nFrames;
-    /* keep this as is for now, don't optimize, work in progress, etc */
-    sfdat->curframepos =  (DWORD) POS64(sfdat->lastwritepos);
-    sfdat->nFrames = max(sfdat->nFrames, ((DWORD) POS64(sfdat->lastwritepos)));
-    /*  fflush(sfdat->file);*/  /* ? need this if reading/seeking as well as  write, etc */
-    return nFrames;
-    
-}
-
-
-/* deprecated! Do not use. */
-
-int psf_sndWriteShortFrames(int sfd, const short *buf, DWORD nFrames)
-{
-    int chans;
-    DWORD i;
-    int j;
-    PSFFILE *sfdat;
-    
-    if(sfd < 0 || sfd > psf_maxfiles)
-        return PSF_E_BADARG;
-    
-    sfdat  = psf_files[sfd];
-    
-#ifdef _DEBUG       
-    assert(sfdat->file);
-    assert(sfdat->filename);
-#endif
-    
-    if(buf==NULL)
-        return PSF_E_BADARG;
-    if(nFrames == 0)
-        return nFrames;
-    if(sfdat->isRead)
-        return PSF_E_FILE_READONLY;
-    chans = sfdat->fmt.Format.nChannels;
-    
-    /* well, it can't be ~less~ efficient than converting twice! */
-    if(!sfdat->is_little_endian){
-        short ssamp;
-        double fval;
-        for(i=0; i < nFrames; i++){
-            for(j=0;j < chans; j++) {
-                ssamp = *buf++;
-                fval = ((double) ssamp / MAX_16BIT);
-                if(sfdat->pPeaks && (sfdat->pPeaks[j].val < (float)(fabs(fval)))){
-                    sfdat->pPeaks[j].pos = sfdat->nFrames + i;
-                    sfdat->pPeaks[j].val = (float)fval;
-                }
-                
-                ssamp = (short) REVWBYTES(ssamp);
-                if(wavDoWrite(sfdat,(char *) &ssamp,sizeof(short))){
-                    DBGFPRINTF((stderr, "wavOpenWrite: write error\n"));
-                    return PSF_E_CANT_WRITE;
-                }
-            }
-        }
-    }
-    else {
-        short ssamp;
-        double fval;
-        for(i=0; i < nFrames; i++){
-            for(j=0;j < chans; j++) {
-                ssamp = *buf++;
-                fval = ((double) ssamp / MAX_16BIT);
-                if(sfdat->pPeaks && (sfdat->pPeaks[j].val < (float)(fabs(fval)))){
-                    sfdat->pPeaks[j].pos = sfdat->nFrames + i;
-                    sfdat->pPeaks[j].val = (float)fval;
-                }
-                
-                if(wavDoWrite(sfdat,(char *) &ssamp,sizeof(short))){
-                    DBGFPRINTF((stderr, "wavOpenWrite: write error\n"));
-                    return PSF_E_CANT_WRITE;
-                }
-            }
-        }
-    }
-    POS64(sfdat->lastwritepos) += nFrames;
-    sfdat->nFrames = max(sfdat->nFrames, ((DWORD) POS64(sfdat->lastwritepos)));
-    fflush(sfdat->file);
-    return nFrames;
-}
-
-/******** READ ***********/
-static int wavReadHeader(PSFFILE *sfdat)
-{
-    DWORD tag,version,peaktime;
-    DWORD size;
-    WORD cbSize;
-    fpos_t bytepos;
-    
-    if(sfdat==NULL || sfdat->file == NULL)
-        return PSF_E_BADARG;
-    
-    if(wavDoRead(sfdat,(char *)&tag,sizeof(DWORD))
-       || wavDoRead(sfdat,(char *) &size,sizeof(DWORD)))
-        return PSF_E_CANT_READ;
-    if(!sfdat->is_little_endian)
-        size = REVDWBYTES(size);
-    else
-        tag = REVDWBYTES(tag);
-    if(tag != TAG('R','I','F','F'))
-        return PSF_E_NOT_WAVE;
-    if(size < (sizeof(WAVEFORMAT) + 3 * sizeof(WORD)))
-        return PSF_E_BAD_FORMAT;
-    
-    if(wavDoRead(sfdat,(char *)&tag,sizeof(DWORD)))
-        return PSF_E_CANT_READ;
-    if(sfdat->is_little_endian)
-        tag = REVDWBYTES(tag);
-    if(tag != TAG('W','A','V','E'))
-        return PSF_E_NOT_WAVE;
-    for(;;){
-        if(wavDoRead(sfdat,(char *)&tag,sizeof(DWORD))
-           || wavDoRead(sfdat,(char *) &size,sizeof(DWORD)))
-            return PSF_E_CANT_READ;
-        if(!sfdat->is_little_endian)
-            size = REVDWBYTES(size);
-        else
-            tag = REVDWBYTES(tag);
-        switch(tag){
-            case(TAG('f','m','t',' ')):
-                if( size < sizeof(WAVEFORMAT))
-                    return PSF_E_BAD_FORMAT;
-                if(size > sizeof_WFMTEX)
-                    return PSF_E_UNSUPPORTED;
-                if(fgetpos(sfdat->file,&bytepos))
-                    return PSF_E_CANT_SEEK;
-                sfdat->fmtoffset = bytepos;
-                if(wavDoRead(sfdat,(char *)&(sfdat->fmt.Format),sizeof(WAVEFORMAT))){
-                    return PSF_E_CANT_READ;
-                }
-                if(!sfdat->is_little_endian)
-                    fmtSwapBytes(sfdat);
-                /* calling function decides if format is supported*/
-                if(size > sizeof(WAVEFORMAT)) {
-                    if(wavDoRead(sfdat,(char*)&cbSize,sizeof(WORD)))
-                        return PSF_E_CANT_READ;
-                    if(!sfdat->is_little_endian)
-                        cbSize = (WORD) REVWBYTES(cbSize);
-                    if(cbSize != (WORD)0) {
-                        if(sfdat->fmt.Format.wFormatTag == WAVE_FORMAT_EXTENSIBLE){
-                            if(cbSize != 22)
-                                return PSF_E_BAD_FORMAT;
-                            sfdat->riff_format = PSF_WAVE_EX;
-                        }
-                    }
-                    else {
-                        int fmtsize = 18;
-                        /* cbSize = 0: has to be 18-byte WAVEFORMATEX */
-                        if((sfdat->fmt.Format.wFormatTag == WAVE_FORMAT_PCM
-                            || sfdat->fmt.Format.wFormatTag == WAVE_FORMAT_IEEE_FLOAT))
-                            sfdat->riff_format = PSF_STDWAVE;
-                        else  /* some horribly mangled format! */
-                            return PSF_E_BAD_FORMAT;
-                        /* Feb 2010: hack to handle bad files with overlarge fmt chunk! */
-                        while (size > fmtsize){
-                            char dummy;
-                            if(wavDoRead(sfdat,(char*)&dummy,sizeof(char)))
-                                return PSF_E_CANT_READ;
-                            fmtsize++;
-                            strcpy(sfdat->warnstring,"fmt chunk too large for format");
-                            sfdat->illformed = 1;
-                        }
-                        
-                    }
-                    sfdat->fmt.Format.cbSize = cbSize;
-                    /* fill in as if basic Format; may change later from WAVE-EX */
-                    sfdat->fmt.Samples.wValidBitsPerSample = sfdat->fmt.Format.wBitsPerSample;
-                    sfdat->fmt.dwChannelMask = 0;
-                    /* get rest of WAVE-EX, if we have it */
-                    if(sfdat->fmt.Format.wFormatTag == WAVE_FORMAT_EXTENSIBLE){
-                        WORD validbits;
-                        DWORD chmask;
-                        if(wavDoRead(sfdat,(char *) &validbits,sizeof(WORD)))
-                            return PSF_E_CANT_READ;
-                        if(!sfdat->is_little_endian)
-                            sfdat->fmt.Samples.wValidBitsPerSample = (WORD) REVWBYTES(validbits);
-                        if(wavDoRead(sfdat,(char *) &chmask,sizeof(DWORD)))
-                            return PSF_E_CANT_READ;
-                        sfdat->fmt.dwChannelMask = chmask;      /* RWD OCT 2009 ! */
-                        if(!sfdat->is_little_endian)
-                            sfdat->fmt.dwChannelMask = REVDWBYTES(chmask);
-                        /* OCT 2009 identify speaker layout */
-                        sfdat->chformat = get_speakerlayout(sfdat->fmt.dwChannelMask,sfdat->fmt.Format.nChannels);
-                        if(wavDoRead(sfdat,(char *) &(sfdat->fmt.SubFormat),sizeof(GUID)))
-                            return PSF_E_CANT_READ;
-                        if(!sfdat->is_little_endian){
-                            sfdat->fmt.SubFormat.Data1 = REVDWBYTES(sfdat->fmt.SubFormat.Data1);
-                            sfdat->fmt.SubFormat.Data2 = (WORD) REVWBYTES(sfdat->fmt.SubFormat.Data2);
-                            sfdat->fmt.SubFormat.Data3 = (WORD) REVWBYTES(sfdat->fmt.SubFormat.Data3);
-                        }
-                        /* if we get a good GUID, this sets up sfdat with samplesize info */
-                        if(check_guid(sfdat))
-                            return PSF_E_UNSUPPORTED;
-                    }
-                }
-                break;
-            case(TAG('P','E','A','K')):
-                /* I SHOULD  report an error if this is after data chunk;
-                 but I suppose innocent users (e.g. of Cubase ) will grumble... */
-                if(wavDoRead(sfdat,(char  *) &version,sizeof(DWORD)))
-                    return PSF_E_CANT_READ;
-                if(!sfdat->is_little_endian)
-                    version = REVDWBYTES(version);
-                if(version != 1) {
-                    DBGFPRINTF((stderr, "Unexpected version level for PEAK chunk!\n"));
-                    return PSF_E_UNSUPPORTED;
-                }
-                if(size != (2 * sizeof(DWORD) + sizeof(PSF_CHPEAK) * sfdat->fmt.Format.nChannels)) {
-                    DBGFPRINTF((stderr, "\nBad size for PEAK chunk\n"));
-                    return PSF_E_BAD_FORMAT;
-                }
-                if(fgetpos(sfdat->file,&bytepos))
-                    return PSF_E_CANT_SEEK;
-                sfdat->peakoffset = bytepos;
-                if(wavDoRead(sfdat,(char *) &peaktime,sizeof(DWORD))){
-                    DBGFPRINTF((stderr,"Error reading PEAK time\n"));
-                    return PSF_E_CANT_READ;
-                }
-                if(!sfdat->is_little_endian)
-                    peaktime = REVDWBYTES(peaktime);
-                sfdat->peaktime = (time_t) peaktime;
-                sfdat->pPeaks = (PSF_CHPEAK *) malloc(sizeof(PSF_CHPEAK) * sfdat->fmt.Format.nChannels);
-                if(sfdat->pPeaks==NULL){
-                    DBGFPRINTF((stderr, "wavOpenWrite: no memory for peak data\n"));
-                    return PSF_E_NOMEM;
-                }
-                if(wavDoRead(sfdat,(char *) sfdat->pPeaks,sizeof(PSF_CHPEAK) * sfdat->fmt.Format.nChannels)) {
-                    DBGFPRINTF((stderr,"Error reading PEAK peak data\n"));
-                    return PSF_E_CANT_READ;
-                }
-                if(!sfdat->is_little_endian){
-                    DWORD *pBlock;
-                    int i;
-                    pBlock = (DWORD *) (sfdat->pPeaks);
-                    for(i=0;i < sfdat->fmt.Format.nChannels * 2; i++)
-                        pBlock[i] = REVDWBYTES(pBlock[i]);
-                }
-                break;
-            case(TAG('d','a','t','a')):
-                if(fgetpos(sfdat->file,&bytepos))
-                    return PSF_E_CANT_SEEK;
-                sfdat->dataoffset = bytepos;
-                if(POS64(sfdat->fmtoffset)==0)
-                    return PSF_E_BAD_FORMAT;
-                sfdat->nFrames = size / sfdat->fmt.Format.nBlockAlign;
-                /* get rescale factor if available */
-                /* NB in correct format, val is always >= 0.0 */
-                if(sfdat->pPeaks && POS64(sfdat->peakoffset) != 0){
-                    float fac = 0.0f;
-                    int i;
-                    for(i=0;i < sfdat->fmt.Format.nChannels; i++)
-                        fac = max(fac,sfdat->pPeaks[i].val);
-                    if(fac > 1.0f)
-                        sfdat->rescale_fac = 1.0f / fac;
-                }
-                /* set sampletype */
-                switch(sfdat->fmt.Format.wFormatTag){
-                    case(WAVE_FORMAT_IEEE_FLOAT):
-                        sfdat->samptype = PSF_SAMP_IEEE_FLOAT;
-                        if(sfdat->fmt.Format.wBitsPerSample != 32) {
-                            /*return PSF_E_BAD_FORMAT;*/
-                            if(sfdat->fmt.Format.wBitsPerSample == sizeof(double))
-                                return PSF_E_UNSUPPORTED;           /*RWD 26:10:2002 allow possibility of doubles! */
-                            else
-                                return PSF_E_BAD_FORMAT;
-                        }
-                        sfdat->lastop  = PSF_OP_READ;
-                        /* RWD 26:10:2002 IMPORTANT BUGFIX! */
-                        return PSF_E_NOERROR;
-                    case(WAVE_FORMAT_PCM):
-                    case(WAVE_FORMAT_EXTENSIBLE):
-                        switch(sfdat->fmt.Format.wBitsPerSample){
-                            case(8):
-                                sfdat->samptype = PSF_SAMP_8;
-                                break;
-                            case(16):
-                                sfdat->samptype = PSF_SAMP_16;
-                                break;
-                            case(24):
-                                sfdat->samptype = PSF_SAMP_24;
-                                break;
-                            case(32):
-                                sfdat->samptype = PSF_SAMP_32;
-                                /* RWD 03/20: fix fault parsing floats amb files */
-                                if(sfdat->chformat == MC_BFMT) {
-                                    if(compare_guids(&(sfdat->fmt.SubFormat),&SUBTYPE_AMBISONIC_B_FORMAT_IEEE_FLOAT)) {
-                                        sfdat->samptype = PSF_SAMP_IEEE_FLOAT;
-                                    }
-                                    else if(!compare_guids(&(sfdat->fmt.SubFormat),&SUBTYPE_AMBISONIC_B_FORMAT_PCM)) {
-                                        sfdat->samptype = PSF_SAMP_UNKNOWN;
-                                    }
-                                }
-                                break;
-                            default:
-                                sfdat->samptype = PSF_SAMP_UNKNOWN;
-                                break;
-                        }
-                        if(sfdat->samptype == PSF_SAMP_UNKNOWN)
-                            return PSF_E_UNSUPPORTED;
-                        sfdat->lastop  = PSF_OP_READ;
-                        return PSF_E_NOERROR;
-                }
-            default:
-                //RWD Apr 2019
-                // handle odd chunk sizes
-                size = (size+1)&~1;
-                /* unknown chunk - skip */
-                if(fseek(sfdat->file,size,SEEK_CUR))
-                    return PSF_E_CANT_READ;
-                break;
-        }
-    }
-}
-
-static int aiffReadHeader(PSFFILE *sfdat)
-{
-    DWORD tag,version,peaktime,remain,offset,blocksize;
-    int have_comm =0,have_ssnd =0;
-    DWORD dwData,size;
-    unsigned char ieee[10];
-    WORD wData;
-    fpos_t bytepos;
-    
-    if(sfdat==NULL || sfdat->file == NULL)
-        return PSF_E_BADARG;
-    
-    if(wavDoRead(sfdat,(char *)&tag,sizeof(DWORD))
-       || wavDoRead(sfdat,(char *) &remain,sizeof(DWORD)))
-        return PSF_E_CANT_READ;
-    if(sfdat->is_little_endian) {
-        remain = REVDWBYTES(remain);
-        tag = REVDWBYTES(tag);
-    }
-    if(tag != TAG('F','O','R','M')){
-        DBGFPRINTF((stderr, "file is not AIFF: no PSF chunk\n"));
-        return PSF_E_BADARG;
-    }
-    
-    if(wavDoRead(sfdat,(char *)&tag,sizeof(DWORD)))
-        return PSF_E_CANT_READ;
-    if(sfdat->is_little_endian)
-        tag = REVDWBYTES(tag);
-    if(tag != TAG('A','I','F','F')){
-        DBGFPRINTF((stderr, "file is not AIFF: no AIFF chunk\n"));
-        return PSF_E_BADARG;
-    }
-    remain -= sizeof(int);
-    
-    
-    while(remain > 0){
-        if(wavDoRead(sfdat,(char *)&tag,sizeof(DWORD))
-           || wavDoRead(sfdat,(char *) &size,sizeof(DWORD)))
-            return PSF_E_CANT_READ;
-        if(sfdat->is_little_endian) {
-            size = REVDWBYTES(size);
-            tag = REVDWBYTES(tag);
-        }
-        remain -=(int)( 2 * sizeof(DWORD));
-        switch(tag){
-            case(TAG('C','O','M','M')):
-                if(size != 18){
-                    DBGFPRINTF((stderr,"AIFF file has bad size for COMM chunk\n"));
-                    return PSF_E_BAD_FORMAT;
-                }
-                if(fgetpos(sfdat->file,&bytepos))
-                    return PSF_E_CANT_SEEK;
-                sfdat->fmtoffset = bytepos;
-                if(wavDoRead(sfdat,(char *)&wData,sizeof(WORD)))
-                    return PSF_E_CANT_READ;
-                if(sfdat->is_little_endian)
-                    wData = (WORD) REVWBYTES(wData);
-                sfdat->fmt.Format.nChannels = wData;
-                if(wavDoRead(sfdat,(char *)&dwData,sizeof(DWORD)))
-                    return PSF_E_CANT_READ;
-                if(sfdat->is_little_endian)
-                    dwData = REVDWBYTES(dwData);
-                sfdat->nFrames = dwData;
-                if(wavDoRead(sfdat,(char *)&wData,sizeof(WORD)))
-                    return PSF_E_CANT_READ;
-                if(sfdat->is_little_endian)
-                    wData = (WORD) REVWBYTES(wData);
-                sfdat->fmt.Format.wBitsPerSample = wData;
-                if(wavDoRead(sfdat,ieee,10))
-                    return PSF_E_CANT_READ;
-                sfdat->fmt.Format.nSamplesPerSec = (DWORD)(ieee_80_to_double(ieee));
-                /*we have to deduce blockalign, and hence containersize*/
-                /* no support (yet) for strange wordsizes such as 20 in 24 */
-                switch(sfdat->fmt.Format.wBitsPerSample){
-                    case(32):
-                        sfdat->fmt.Format.nBlockAlign = sizeof(int);
-                        sfdat->samptype = PSF_SAMP_32;
-                        break;
-                    case(24):
-                        sfdat->fmt.Format.nBlockAlign = 3;
-                        sfdat->samptype = PSF_SAMP_24;
-                        break;
-                    case(16):
-                        sfdat->fmt.Format.nBlockAlign = sizeof(short);
-                        sfdat->samptype = PSF_SAMP_16;
-                        break;
-                    default:
-                        DBGFPRINTF((stderr, "unsupported sample format for AIFF file\n"));
-                        return PSF_E_UNSUPPORTED;
-                }
-                sfdat->fmt.Format.nBlockAlign = (WORD) (sfdat->fmt.Format.nBlockAlign * sfdat->fmt.Format.nChannels);
-                remain -= 18;
-                have_comm = 1;
-                break;
-            case (TAG('P','E','A','K')):
-                if(!have_comm){
-                    DBGFPRINTF((stderr, "AIFF file: found PEAK chunk before COMM chunk!\n"));
-                    return PSF_E_BAD_FORMAT;
-                }
-                if(size != (2 * sizeof(DWORD) + sizeof(PSF_CHPEAK) * sfdat->fmt.Format.nChannels)){
-                    DBGFPRINTF((stderr, "AIFF file has bad size for PEAK chunk\n"));
-                    return PSF_E_BAD_FORMAT;
-                }
-                if(wavDoRead(sfdat,(char *)&version,sizeof(DWORD))) {
-                    DBGFPRINTF((stderr,"Error reading PEAK version\n"));
-                    return PSF_E_CANT_READ;
-                }
-                if(sfdat->is_little_endian)
-                    version  = REVDWBYTES(version);
-                if(version != 1){
-                    DBGFPRINTF((stderr, "AIFF file has unexpected version level for PEAK chunk!\n"));
-                    return PSF_E_UNSUPPORTED;
-                }
-                if(fgetpos(sfdat->file,&bytepos))
-                    return PSF_E_CANT_SEEK;
-                sfdat->peakoffset = bytepos;
-                if(wavDoRead(sfdat,(char *) &peaktime,sizeof(DWORD))) {
-                    DBGFPRINTF((stderr,"Error reading PEAK time\n"));
-                    return PSF_E_CANT_READ;
-                }
-                if(sfdat->is_little_endian)
-                    peaktime = REVDWBYTES(peaktime);
-                sfdat->peaktime = (time_t) peaktime;
-                sfdat->pPeaks = (PSF_CHPEAK *)malloc(sizeof(PSF_CHPEAK) * sfdat->fmt.Format.nChannels);
-                if(sfdat->pPeaks==NULL){
-                    DBGFPRINTF((stderr, "wavOpenWrite: no memory for peak data\n"));
-                    return PSF_E_NOMEM;
-                }
-                if(wavDoRead(sfdat,(char *)(sfdat->pPeaks),sizeof(PSF_CHPEAK) * sfdat->fmt.Format.nChannels)){
-                    DBGFPRINTF((stderr,"Error reading PEAK peak data\n"));
-                    return PSF_E_CANT_READ;
-                }
-                if(sfdat->is_little_endian){
-                    DWORD *pBlock;
-                    int i;
-                    pBlock = (DWORD *) (sfdat->pPeaks);
-                    for(i=0;i < sfdat->fmt.Format.nChannels * 2; i++)
-                        pBlock[i] = REVDWBYTES(pBlock[i]);
-                }
-                remain -= size;
-                break;
-            case(TAG('S','S','N','D')):
-                if(wavDoRead(sfdat,(char *)&offset,sizeof(DWORD))
-                   || wavDoRead(sfdat,(char *) &blocksize,sizeof(DWORD)))
-                    return PSF_E_CANT_READ;
-                if(sfdat->is_little_endian){
-                    offset  = REVDWBYTES(offset);
-                    blocksize = REVDWBYTES(blocksize);
-                }
-                if(fgetpos(sfdat->file,&bytepos))
-                    return PSF_E_CANT_SEEK;
-                sfdat->dataoffset = bytepos;
-                POS64(sfdat->dataoffset) += offset;
-                sfdat->nFrames = (size - 2* sizeof(DWORD))/ sfdat->fmt.Format.nBlockAlign;
-                /* NB for seek: we used up 8 bytes with offset and blocksize */
-                /* if we already have COMM, we could finish here! */
-                /* TODO! this is still a signed int seek, no good for 4GB */
-                if(fseek(sfdat->file,((size - 2* sizeof(DWORD))+1)&~1,SEEK_CUR))
-                    return PSF_E_CANT_SEEK;
-                have_ssnd = 1;
-                remain -= (size+1)&~1;
-                break;
-                /*HARSH! traps linux sox error, for example */
-            case(0):
-                DBGFPRINTF((stderr, "AIFF file has bad main chunksize\n"));
-                return PSF_E_BAD_FORMAT;
-            default:
-                /* skip all unknown chunks */
-                if(fseek(sfdat->file,(size+1)&~1,SEEK_CUR))
-                    return PSF_E_CANT_SEEK;
-                remain -= (size+1)&~1;
-                break;
-        }
-    }
-    if(!(have_ssnd && have_comm)){
-        DBGFPRINTF((stderr, "AIFF file has missing chunks\n"));
-        return PSF_E_BAD_FORMAT;
-    }
-    /* we have seeked to EOF, so rewind to start of data */
-    if(fsetpos(sfdat->file,&sfdat->dataoffset))
-        return PSF_E_CANT_SEEK;
-    sfdat->curframepos = 0;
-    sfdat->riff_format = PSF_AIFF;
-    /* get rescale factor if available */
-    /* NB in correct format, val is always >= 0.0 */
-    if(sfdat->pPeaks &&  POS64(sfdat->peakoffset) != 0){
-        float fac = 0.0f;
-        int i;
-        for(i=0;i < sfdat->fmt.Format.nChannels; i++)
-            fac = max(fac,sfdat->pPeaks[i].val);
-        if(fac > 1.0f)
-            sfdat->rescale_fac = 1.0f / fac;
-    }
-    sfdat->lastop  = PSF_OP_READ;
-    return PSF_E_NOERROR;
-}
-
-static int aifcReadHeader(PSFFILE *sfdat)
-{
-    DWORD tag,version,peaktime,remain,offset,blocksize;
-    int have_comm =0,have_ssnd =0,have_fver = 0;
-    DWORD dwData,size,aifcver,ID_compression;
-    unsigned char ieee[10];
-    WORD wData;
-    fpos_t bytepos;
-    
-    if(sfdat==NULL || sfdat->file == NULL)
-        return PSF_E_BADARG;
-    
-    if(wavDoRead(sfdat,(char *)&tag,sizeof(DWORD))
-       || wavDoRead(sfdat,(char *) &remain,sizeof(DWORD)))
-        return PSF_E_CANT_READ;
-    if(sfdat->is_little_endian) {
-        remain = REVDWBYTES(remain);
-        tag = REVDWBYTES(tag);
-    }
-    if(tag != TAG('F','O','R','M')){
-        DBGFPRINTF((stderr, "file is not AIFC: no FORM chunk\n"));
-        return PSF_E_BADARG;
-    }
-    
-    if(wavDoRead(sfdat,(char *)&tag,sizeof(DWORD)))
-        return PSF_E_CANT_READ;
-    if(sfdat->is_little_endian)
-        tag = REVDWBYTES(tag);
-    if(tag != TAG('A','I','F','C')){
-        DBGFPRINTF((stderr, "file is not AIFC: no AIFC chunk\n"));
-        return PSF_E_BADARG;
-    }
-    remain -= sizeof(DWORD);
-    
-    
-    while(remain > 0){
-        if(wavDoRead(sfdat,(char *)&tag,sizeof(DWORD))
-           || wavDoRead(sfdat,(char *) &size,sizeof(DWORD)))
-            return PSF_E_CANT_READ;
-        if(sfdat->is_little_endian) {
-            size = REVDWBYTES(size);
-            tag = REVDWBYTES(tag);
-        }
-        remain -= 2 * sizeof(DWORD);
-        switch(tag){
-            case(TAG('F','V','E','R')):
-                if(size != sizeof(DWORD)){
-                    DBGFPRINTF((stderr, "AIFC file has bad size for FVER chunk\n"));
-                    return PSF_E_BAD_FORMAT;
-                }
-                if(wavDoRead(sfdat,(char *) &aifcver,sizeof(DWORD)))
-                    return PSF_E_CANT_READ;
-                if(sfdat->is_little_endian)
-                    aifcver = REVDWBYTES(aifcver);
-                remain-= sizeof(DWORD);
-                if(aifcver != AIFC_VERSION_1)
-                    return PSF_E_UNSUPPORTED;
-                have_fver = 1;
-                break;
-            case(TAG('C','O','M','M')):
-                if(size < 22) {
-                    DBGFPRINTF((stderr, "AIFC file has bad size for COMM chunk\n"));
-                    return PSF_E_BAD_FORMAT;
-                }
-                if(fgetpos(sfdat->file,&sfdat->fmtoffset))
-                    return PSF_E_CANT_SEEK;
-                if(wavDoRead(sfdat,(char *)&wData,sizeof(WORD)))
-                    return PSF_E_CANT_READ;
-                if(sfdat->is_little_endian)
-                    wData = (WORD) REVWBYTES(wData);
-                sfdat->fmt.Format.nChannels = wData;
-                if(wavDoRead(sfdat,(char *)&dwData,sizeof(DWORD)))
-                    return PSF_E_CANT_READ;
-                if(sfdat->is_little_endian)
-                    dwData = REVDWBYTES(dwData);
-                sfdat->nFrames = dwData;
-                if(wavDoRead(sfdat,(char *)&wData,sizeof(WORD)))
-                    return PSF_E_CANT_READ;
-                if(sfdat->is_little_endian)
-                    wData = (WORD) REVWBYTES(wData);
-                sfdat->fmt.Format.wBitsPerSample = wData;
-                if(wavDoRead(sfdat,ieee,10))
-                    return PSF_E_CANT_READ;
-                sfdat->fmt.Format.nSamplesPerSec = (DWORD)(ieee_80_to_double(ieee));
-                /*we have to deduce blockalign, and hence containersize*/
-                /* no support for strange wordsizes such as 20 in 24 */
-                switch(sfdat->fmt.Format.wBitsPerSample){
-                    case(32):
-                        sfdat->fmt.Format.nBlockAlign = sizeof(DWORD);
-                        sfdat->samptype = PSF_SAMP_32;
-                        break;
-                    case(24):
-                        sfdat->fmt.Format.nBlockAlign = 3;
-                        sfdat->samptype = PSF_SAMP_24;
-                        break;
-                    case(16):
-                        sfdat->fmt.Format.nBlockAlign = sizeof(short);
-                        sfdat->samptype = PSF_SAMP_16;
-                        break;
-                    default:
-                        DBGFPRINTF((stderr, "unsupported sample format for AIFC file\n"));
-                        return PSF_E_UNSUPPORTED;
-                }
-                sfdat->fmt.Format.nBlockAlign = (WORD) (sfdat->fmt.Format.nBlockAlign * sfdat->fmt.Format.nChannels);
-                if(wavDoRead(sfdat,(char *)&ID_compression,sizeof(DWORD))){
-                    return PSF_E_CANT_READ;
-                }
-                if(sfdat->is_little_endian)
-                    ID_compression = REVDWBYTES(ID_compression);
-                if(!(    (ID_compression == TAG('N','O','N','E'))
-                     || (ID_compression == TAG('F','L','3','2'))
-                     || (ID_compression == TAG('f','l','3','2'))
-                     || (ID_compression == TAG('i','n','2','4')))
-                   ){
-                    DBGFPRINTF((stderr, "AIFC file: unsupported compression format\n"));
-                    return PSF_E_UNSUPPORTED;
-                }
-                /*set stype info */
-                if((ID_compression== TAG('F','L','3','2'))
-                   || ID_compression == TAG('f','l','3','2')){
-                    if(sfdat->fmt.Format.wBitsPerSample != 32){
-                        DBGFPRINTF((stderr, "AIFC file: samples not 32bit in floats file\n"));
-                        return PSF_E_BAD_FORMAT;
-                    }
-                    else {
-                        sfdat->fmt.Format.wFormatTag = WAVE_FORMAT_IEEE_FLOAT;
-                        sfdat->samptype = PSF_SAMP_IEEE_FLOAT;
-                    }
-                }
-                /* yes, lazy! skip past pascal string*/
-                if(fseek(sfdat->file,((size-22)+1)&~1,SEEK_CUR))    /*written for documentation, not terseness!*/
-                    return PSF_E_CANT_SEEK;
-                remain -= (size+1)&~1;
-                have_comm = 1;
-                break;
-            case (TAG('P','E','A','K')):
-                if(!have_comm){
-                    DBGFPRINTF((stderr, "\nAIFC file: found PEAK chunk before COMM chunk!\n"));
-                    return PSF_E_BAD_FORMAT;
-                }
-                if(size != (2 * sizeof(DWORD) + sizeof(PSF_CHPEAK) * sfdat->fmt.Format.nChannels)){
-                    DBGFPRINTF((stderr, "\nBad size for PEAK chunk\n"));
-                    return PSF_E_BAD_FORMAT;
-                }
-                if(wavDoRead(sfdat,(char *)&version,sizeof(DWORD)))
-                    return PSF_E_CANT_READ;
-                if(sfdat->is_little_endian)
-                    version  = REVDWBYTES(version);
-                if(version != 1) {
-                    DBGFPRINTF((stderr, "Unexpected version level for PEAK chunk!\n"));
-                    return PSF_E_UNSUPPORTED;
-                }
-                if(fgetpos(sfdat->file,&bytepos))
-                    return PSF_E_CANT_SEEK;
-                sfdat->peakoffset = bytepos;
-                if(wavDoRead(sfdat,(char *) &peaktime,sizeof(DWORD))){
-                    DBGFPRINTF((stderr,"Error reading PEAK time\n"));
-                    return PSF_E_CANT_READ;
-                }
-                if(sfdat->is_little_endian)
-                    peaktime = REVDWBYTES(peaktime);
-                sfdat->peaktime = (time_t) peaktime;
-                sfdat->pPeaks = (PSF_CHPEAK *)malloc(sizeof(PSF_CHPEAK) * sfdat->fmt.Format.nChannels);
-                if(sfdat->pPeaks==NULL) {
-                    DBGFPRINTF((stderr, "wavOpenWrite: no memory for peak data\n"));
-                    return PSF_E_NOMEM;
-                }
-                if(wavDoRead(sfdat,(char *)(sfdat->pPeaks),sizeof(PSF_CHPEAK) * sfdat->fmt.Format.nChannels)) {
-                    DBGFPRINTF((stderr,"Error reading PEAK peak data\n"));
-                    return PSF_E_CANT_READ;
-                }
-                if(sfdat->is_little_endian){
-                    DWORD *pBlock;
-                    int i;
-                    pBlock = (DWORD *) (sfdat->pPeaks);
-                    for(i=0;i < sfdat->fmt.Format.nChannels * 2; i++)
-                        pBlock[i] = REVDWBYTES(pBlock[i]);
-                }
-                remain -= (size+1)&~1;
-                break;
-            case(TAG('S','S','N','D')):
-                if(wavDoRead(sfdat,(char *)&offset,sizeof(DWORD))
-                   || wavDoRead(sfdat,(char *) &blocksize,sizeof(DWORD)))
-                    return PSF_E_CANT_READ;
-                if(sfdat->is_little_endian){
-                    offset  = REVDWBYTES(offset);
-                    blocksize = REVDWBYTES(blocksize);
-                }
-                if(fgetpos(sfdat->file,&bytepos))
-                    return PSF_E_CANT_SEEK;
-                sfdat->dataoffset = bytepos;
-                POS64(sfdat->dataoffset) += offset;
-                sfdat->nFrames = (size - 2* sizeof(DWORD))/ sfdat->fmt.Format.nBlockAlign;
-                if(fseek(sfdat->file,((size - 2* sizeof(DWORD))+1)&~1,SEEK_CUR))
-                    return PSF_E_CANT_SEEK;
-                have_ssnd = 1;
-                remain -= (size+1)&~1;
-                break;
-                /* HARSH! as above */
-            case(0):
-                DBGFPRINTF((stderr, "AIFC file has bad main chunksize\n"));
-                return PSF_E_BAD_FORMAT;
-            default:
-                /* skip all unknown chunks */
-                if(fseek(sfdat->file,(size+1)&~1,SEEK_CUR))
-                    return PSF_E_CANT_SEEK;
-                remain -= (size+1)&~1;
-                break;
-        }
-    }
-    if(!(have_ssnd && have_comm && have_fver)){
-        DBGFPRINTF((stderr, "AIFC file has bad format\n"));
-        return PSF_E_BAD_FORMAT;
-    }
-    /* we have seeked (ugh) to EOF, so rewind to start of data */
-    if(fsetpos(sfdat->file,&sfdat->dataoffset))
-        return PSF_E_CANT_SEEK;
-    sfdat->curframepos = 0;
-    sfdat->riff_format = PSF_AIFC;
-    /* get rescale factor if available */
-    /* NB in correct format, val is always >= 0.0 */
-    if(sfdat->pPeaks &&  POS64(sfdat->peakoffset) != 0){
-        float fac = 0.0f;
-        int i;
-        for(i=0;i < sfdat->fmt.Format.nChannels; i++)
-            fac = max(fac,sfdat->pPeaks[i].val);
-        if(fac > 1.0f)
-            sfdat->rescale_fac = 1.0f / fac;
-    }
-    sfdat->lastop  = PSF_OP_READ;
-    return PSF_E_NOERROR;
-}
-/* only RDONLY access supported */
-int psf_sndOpen(const char *path,PSF_PROPS *props, int rescale)
-{
-    int i,rc = 0;
-    PSFFILE *sfdat;
-    psf_format fmt;
-    
-    /* RWD interesting syntax issue: I need the curlies, or break doesn't work properly */
-    for(i=0;i < psf_maxfiles;i++) {
-        if(psf_files[i]==NULL)
-            break;
-    }
-    if(i==psf_maxfiles){
-        
-        return PSF_E_TOOMANYFILES;
-    }
-    
-    sfdat = psf_newFile(NULL);
-    if(sfdat==NULL){
-        return PSF_E_NOMEM;
-    }
-    sfdat->rescale = rescale;
-    sfdat->is_little_endian = byte_order();
-    fmt = psf_getFormatExt(path);
-    if(!(fmt==PSF_STDWAVE || fmt==PSF_WAVE_EX || fmt==PSF_AIFF || fmt==PSF_AIFC))
-        return PSF_E_BADARG;
-    
-    if((sfdat->file = fopen(path,"rb"))  == NULL) {
-        DBGFPRINTF((stderr, "psf_sndOpen: cannot open '%s'\n", path));
-        return PSF_E_CANT_OPEN;
-    }
-    sfdat->filename = (char *) malloc(strlen(path)+1);
-    if(sfdat->filename==NULL) {
-        return PSF_E_NOMEM;
-    }
-    strcpy(sfdat->filename, path);
-    sfdat->isRead =  1;
-    sfdat->nFrames = 0;
-    /* no need to calc header sizes */
-    switch(fmt){
-        case(PSF_STDWAVE):
-        case(PSF_WAVE_EX):
-            rc =  wavReadHeader(sfdat);
-            break;
-        case(PSF_AIFF):
-            /* some .aiff files may actually be aifc - esp if floats! */
-        case(PSF_AIFC):
-            rc = aiffReadHeader(sfdat);
-            /* try AIFC if AIFF fails */
-            if(rc < PSF_E_NOERROR) {
-                rewind(sfdat->file);
-                rc =  aifcReadHeader(sfdat);
-            }
-            break;
-        default:
-            DBGFPRINTF((stderr, "psf_sndOpen: unsupported file format\n"));
-            rc =  PSF_E_UNSUPPORTED;
-    }
-    if(rc < PSF_E_NOERROR)
-        return rc;
-    /* fill props info*/
-    props->srate    = sfdat->fmt.Format.nSamplesPerSec;
-    props->chans    = sfdat->fmt.Format.nChannels;
-    props->samptype = sfdat->samptype;
-    props->chformat = sfdat->chformat;
-    props->format      =  fmt;
-    
-    if(fmt==PSF_STDWAVE && (sfdat->riff_format == PSF_WAVE_EX))
-        props->format = PSF_WAVE_EX;
-    
-    psf_files[i] = sfdat;
-    return i;
-}
-
-
-int psf_sndReadFloatFrames(int sfd, float *buf, DWORD nFrames)
-{
-    int chans;
-    DWORD framesread;
-    int blocksize,lsamp;
-    int temp;
-    int i,do_reverse;
-    short ssamp;
-    float *pbuf = buf;
-    float fsamp;
-    PSFFILE *sfdat;
-    int do_shift;
-    SND_SAMP s_samp;
-#ifdef _DEBUG
-    static int debug = 1;
-#endif
-    if(sfd < 0 || sfd > psf_maxfiles)
-        return PSF_E_BADARG;
-    if(buf==NULL)
-        return PSF_E_BADARG;
-    if(nFrames == 0)
-        return nFrames;
-    sfdat  = psf_files[sfd];
-#ifdef _DEBUG
-    assert(sfdat);
-    assert(sfdat->file);
-    assert(sfdat->filename);
-    /* must check our calcs! */
-    assert(sfdat->curframepos <= sfdat->nFrames);
-#endif
-    /* how much do we have left? return immediately if none! */
-    chans = sfdat->fmt.Format.nChannels;
-    framesread = min(sfdat->nFrames - sfdat->curframepos,nFrames);
-    if(framesread==0)
-        return (long) framesread;
-    
-    blocksize =  framesread * chans;
-    switch(sfdat->riff_format){
-        case(PSF_STDWAVE):
-        case(PSF_WAVE_EX):
-            do_reverse = (sfdat->is_little_endian ? 0 : 1 );
-            do_shift = 1;
-            break;
-        case(PSF_AIFF):
-        case(PSF_AIFC):
-            do_reverse = (sfdat->is_little_endian ? 1 : 0 );
-            do_shift = 0;
-            break;
-        default:
-            return PSF_E_UNSUPPORTED;
-    }
-    if(sfdat->lastop == PSF_OP_WRITE)
-        fflush(sfdat->file);
-    switch(sfdat->samptype){
-        case(PSF_SAMP_IEEE_FLOAT):
-            
-            if(do_reverse){
-                for(i=0;i < blocksize;i ++){
-                    if(wavDoRead(sfdat,(char *)&lsamp,sizeof(int)))
-                        return PSF_E_CANT_READ;
-                    lsamp = REVDWBYTES(lsamp);
-                    //    fsamp = * (float *)&lsamp;
-                    s_samp.lsamp = lsamp;
-                    fsamp = s_samp.fsamp;
-                    if(sfdat->rescale)
-                        fsamp *= sfdat->rescale_fac;
-                    *pbuf++ = fsamp;
-                }
-            }
-            else{
-                if(wavDoRead(sfdat,(char *) buf,blocksize * sizeof(float)))
-                    return PSF_E_CANT_READ;
-                if(sfdat->rescale){
-                    pbuf = buf;
-                    for(i=0;i < blocksize; i++)
-                        *pbuf++  *= sfdat->rescale_fac;
-                }
-            }
-            break;
-        case(PSF_SAMP_16):
-            if(do_reverse){
-                for(i = 0; i < blocksize; i++){
-                    if(wavDoRead(sfdat,(char *)&ssamp,sizeof(short)))
-                        return PSF_E_CANT_READ;
-                    ssamp = (short) REVWBYTES(ssamp);
-                    fsamp = (float)((double) ssamp / MAX_16BIT);
-                    *pbuf++ = fsamp;
-                }
-            }
-            else{
-                for(i = 0; i < blocksize; i++){
-                    if(wavDoRead(sfdat,(char *)&ssamp,sizeof(short)))
-                        return PSF_E_CANT_READ;
-                    fsamp = (float)((double) ssamp / MAX_16BIT);
-                    *pbuf++ = fsamp;
-                }
-            }
-            break;
-        case(PSF_SAMP_24):
-            if(do_reverse){
-#ifdef _DEBUG
-                if(debug){
-                    printf("do_reverse: riffformat=%d do_shift = %d little_endian = %d\n",
-                           sfdat->riff_format,do_shift,sfdat->is_little_endian);
-                    debug = 0;
-                }
-#endif            
-                for(i=0;i < blocksize;i++){
-                    temp = 0;
-                    if(wavDoRead(sfdat,(char *)&temp,3))
-                        return PSF_E_CANT_READ;
-                    lsamp = REVDWBYTES(temp);
-                    if(do_shift)
-                        lsamp <<= 8;
-                    fsamp = (float)((double)(lsamp) / MAX_32BIT);
-                    *pbuf++ = fsamp;
-                }
-            }
-            else{
-                for(i=0;i < blocksize;i++){
-                    lsamp = 0;
-                    if(wavDoRead(sfdat,(char *)&lsamp,3))
-                        return PSF_E_CANT_READ;
-                    if(do_shift)
-                        lsamp <<= 8;
-                    fsamp = (float)((double)(lsamp) / MAX_32BIT);
-                    *pbuf++ = fsamp;
-                }
-            }
-            break;
-        case(PSF_SAMP_32):
-            if(do_reverse){
-                for(i=0;i < blocksize;i++){
-                    if(wavDoRead(sfdat,(char *)&lsamp,sizeof(int)))
-                        return PSF_E_CANT_READ;
-                    lsamp = REVDWBYTES(lsamp);
-                    fsamp = (float)((double)(lsamp) / MAX_32BIT);
-                    *pbuf++ = fsamp;
-                }
-            }
-            else{
-                for(i=0;i < blocksize;i++){
-                    lsamp = 0L;
-                    if(wavDoRead(sfdat,(char *)&lsamp,sizeof(int)))
-                        return PSF_E_CANT_READ;
-                    fsamp = (float)((double)(lsamp) / MAX_32BIT);
-                    *pbuf++ = fsamp;
-                }
-            }
-            break;
-        default:
-            DBGFPRINTF((stderr, "psf_sndOpen: unsupported sample format\n"));
-            return PSF_E_UNSUPPORTED;
-    }
-    sfdat->curframepos += framesread;
-    
-    return framesread;
-}
-
-
-/* read doubles version! */
-int psf_sndReadDoubleFrames(int sfd, double *buf, DWORD nFrames)
-{
-    int chans;
-    DWORD framesread;
-    int blocksize,lsamp;
-    int i,do_reverse;
-    short ssamp;
-    double *pbuf = buf;
-    float fsamp;
-    PSFFILE *sfdat;
-    SND_SAMP s_samp;
-    int do_shift;
-    
-    if(sfd < 0 || sfd > psf_maxfiles)
-        return PSF_E_BADARG;
-    if(buf==NULL)
-        return PSF_E_BADARG;
-    if(nFrames == 0)
-        return nFrames;
-    sfdat  = psf_files[sfd];
-#ifdef _DEBUG
-    assert(sfdat);
-    assert(sfdat->file);
-    assert(sfdat->filename);
-    /* must check our calcs! */
-    assert(sfdat->curframepos <= sfdat->nFrames);
-#endif
-    /* how much do we have left? return immediately if none! */
-    chans = sfdat->fmt.Format.nChannels;
-    framesread = min(sfdat->nFrames - sfdat->curframepos,nFrames);
-    if(framesread==0)
-        return (long) framesread;
-    
-    blocksize =  framesread * chans;
-    switch(sfdat->riff_format){
-        case(PSF_STDWAVE):
-        case(PSF_WAVE_EX):
-            do_reverse = (sfdat->is_little_endian ? 0 : 1 );
-            do_shift = 1;
-            break;
-        case(PSF_AIFF):
-        case(PSF_AIFC):
-            do_reverse = (sfdat->is_little_endian ? 1 : 0 );
-            do_shift = 0;
-            break;
-        default:
-            return PSF_E_UNSUPPORTED;
-    }
-    if(sfdat->lastop == PSF_OP_WRITE)
-        fflush(sfdat->file);
-    switch(sfdat->samptype){
-        case(PSF_SAMP_IEEE_FLOAT):
-            if(do_reverse){
-                for(i=0;i < blocksize;i ++){
-                    if(wavDoRead(sfdat,(char *)&lsamp,sizeof(int)))
-                        return PSF_E_CANT_READ;
-                    lsamp = REVDWBYTES(lsamp);
-                    //    fsamp = * (float *)&lsamp;
-                    s_samp.lsamp = lsamp;
-                    fsamp = s_samp.fsamp;
-                    if(sfdat->rescale)
-                        fsamp *= sfdat->rescale_fac;
-                    *pbuf++ = (double) fsamp;
-                }
-            }
-            else{
-                for(i=0;i < blocksize;i++){
-                    if(wavDoRead(sfdat,(char *) &fsamp, sizeof(float)))
-                        return PSF_E_CANT_READ;
-                    *pbuf++ = (double) fsamp;
-                }
-                if(sfdat->rescale){
-                    pbuf = buf;
-                    for(i=0;i < blocksize; i++)
-                        *pbuf++  *= sfdat->rescale_fac;
-                }
-            }
-            break;
-        case(PSF_SAMP_16):
-            if(do_reverse){
-                for(i = 0; i < blocksize; i++){
-                    if(wavDoRead(sfdat,(char *)&ssamp,sizeof(short)))
-                        return PSF_E_CANT_READ;
-                    ssamp = (short) REVWBYTES(ssamp);
-                    fsamp = (float)((double) ssamp / MAX_16BIT);
-                    *pbuf++ = (double) fsamp;
-                }
-            }
-            else{
-                for(i = 0; i < blocksize; i++){
-                    if(wavDoRead(sfdat,(char *)&ssamp,sizeof(short)))
-                        return PSF_E_CANT_READ;
-                    fsamp = (float)((double) ssamp / MAX_16BIT);
-                    *pbuf++ = (double) fsamp;
-                }
-            }
-            break;
-        case(PSF_SAMP_24):
-            if(do_reverse){
-                for(i=0;i < blocksize;i++){
-                    if(wavDoRead(sfdat,(char *)&lsamp,3))
-                        return PSF_E_CANT_READ;
-                    lsamp = REVDWBYTES(lsamp);
-                    if(do_shift)
-                        lsamp <<= 8;
-                    fsamp = (float)((double)(lsamp) / MAX_32BIT);
-                    *pbuf++ = (double) fsamp;
-                }
-            }
-            else{
-                for(i=0;i < blocksize;i++){
-                    lsamp = 0L;
-                    if(wavDoRead(sfdat,(char *)&lsamp,3))
-                        return PSF_E_CANT_READ;
-                    if(do_shift)
-                        lsamp <<= 8;
-                    fsamp = (float)((double)(lsamp) / MAX_32BIT);
-                    *pbuf++ = (double) fsamp;
-                }
-            }
-            break;
-        case(PSF_SAMP_32):
-            if(do_reverse){
-                for(i=0;i < blocksize;i++){
-                    if(wavDoRead(sfdat,(char *)&lsamp,sizeof(int)))
-                        return PSF_E_CANT_READ;
-                    lsamp = REVDWBYTES(lsamp);
-                    fsamp = (float)((double)(lsamp) / MAX_32BIT);
-                    *pbuf++ = (double) fsamp;
-                }
-            }
-            else{
-                for(i=0;i < blocksize;i++){
-                    lsamp = 0L;
-                    if(wavDoRead(sfdat,(char *)&lsamp,sizeof(int)))
-                        return PSF_E_CANT_READ;
-                    fsamp = (float)((double)(lsamp) / MAX_32BIT);
-                    *pbuf++ = (double) fsamp;
-                }
-            }
-            break;
-        default:
-            DBGFPRINTF((stderr, "psf_sndOpen: unsupported sample format\n"));
-            return PSF_E_UNSUPPORTED;
-    }
-    sfdat->curframepos += framesread;
-    
-    return framesread;
-}
-
-
-#ifdef _DEBUG
-/* private test func to get raw file size */
-/* verify sfdat->nFrames always <= true EOF position */
-/* NB: size value is wrong if there is junk after data chunk! */
-static fpos_t getsize(FILE* fp){
-    fpos_t curpos;
-    fpos_t size;
-    if(fgetpos(fp,&curpos))
-        return -1;
-    
-    if(fseek(fp,0,SEEK_END))
-        return -1;
-    
-    if(fgetpos(fp,&size))
-        return -1;
-    
-    if(fsetpos(fp,&curpos))
-        return -1;
-    
-    return  size;
-}
-#endif
-
-/* return size in m/c frames */
-/* signed long because we want error return */
-/* 64 bit: this would have to return a 64bit type (or unsigned 32bit int) IF we want to support 4GB 8bit sfiles! */
-/* (which we don't) */
-int psf_sndSize(int sfd)
-{
-    PSFFILE *sfdat;
-#ifdef _DEBUG
-    fpos_t size;
-    DWORD framesize;
-#endif
-    if(sfd < 0 || sfd > psf_maxfiles)
-        return PSF_E_BADARG;
-    
-    sfdat  = psf_files[sfd];
-    if(sfdat==NULL)
-        return PSF_E_BADARG;
-#ifdef _DEBUG       
-    assert(sfdat->file);
-    assert(sfdat->filename);
-    if((size = getsize(sfdat->file)) < 0)   {
-        DBGFPRINTF((stderr, "getsize() error in psf_sndSize().\n"));
-        return -1;
-    }
-    /* NB deceived if any pad or other junk after data chunk! */
-    framesize = (DWORD)((POS64(size) - (MYLONG)(sfdat->dataoffset)) / sfdat->fmt.Format.nBlockAlign);
-    assert(framesize >= (DWORD) sfdat->nFrames);
-    
-#endif
-    
-    return sfdat->nFrames;
-    
-}
-
-/* returns multi-channel (frame)  position */
-/* 64bit see above! */
-int psf_sndTell(int sfd)
-{
-    fpos_t pos;
-    PSFFILE *sfdat;
-    
-    if(sfd < 0 || sfd > psf_maxfiles)
-        return PSF_E_BADARG;
-    
-    sfdat  = psf_files[sfd];
-    if(sfdat==NULL)
-        return PSF_E_BADARG;
-#ifdef _DEBUG       
-    assert(sfdat->file);
-    assert(sfdat->filename);
-#endif
-    
-    if(fgetpos(sfdat->file,&pos))
-        return PSF_E_CANT_SEEK;
-    POS64(pos) -= POS64(sfdat->dataoffset);
-    POS64(pos) /= sfdat->fmt.Format.nBlockAlign;
-#ifdef _DEBUG
-    if(sfdat->lastop == PSF_OP_READ)
-        assert(pos== sfdat->curframepos);
-    else if(sfdat->lastop == PSF_OP_WRITE)
-    /* RWD this will be out (but == curframepos) if lastop was a read . so maybe say >=, or test for lastop ? */
-        assert(pos == sfdat->lastwritepos);
-#endif
-    return (int) POS64(pos);
-}
-
-
-int psf_sndSeek(int sfd,int offset, int mode)
-{
-    long byteoffset;    /* can be negative - limited to 2GB moves*/
-    fpos_t data_end,pos_target,cur_pos;
-    PSFFILE *sfdat;
-    
-    
-    if(sfd < 0 || sfd > psf_maxfiles)
-        return PSF_E_BADARG;
-    
-    sfdat  = psf_files[sfd];
-    if(sfdat==NULL)
-        return PSF_E_BADARG;
-#ifdef _DEBUG       
-    assert(sfdat->file);
-    assert(sfdat->filename);
-#endif
-    /* RWD NB:dataoffset test only valid for files with headers! */
-    if(POS64(sfdat->dataoffset)==0)
-        return PSF_E_BADARG;
-    /* or, it indicates a RAW file.... */
-    
-    byteoffset =  offset *  sfdat->fmt.Format.nBlockAlign;
-    POS64(data_end) = POS64(sfdat->dataoffset) + (sfdat->nFrames * sfdat->fmt.Format.nBlockAlign);
-    switch(mode){
-        case PSF_SEEK_SET:
-            POS64(pos_target) =  POS64(sfdat->dataoffset) + byteoffset;
-            if(fsetpos(sfdat->file,&pos_target))
-                return PSF_E_CANT_SEEK;
-            break;
-        case PSF_SEEK_END:
-            /* NB can't just seek to end of file as there may be junk after data chunk! */
-            POS64(pos_target) = POS64(data_end) + byteoffset;
-            if(fsetpos(sfdat->file,&pos_target))
-                return PSF_E_CANT_SEEK;
-            break;
-        case PSF_SEEK_CUR:
-            /* should be safe using fseek for SEEK_END */
-            /* Currently UNDECIDED whether to allow seeks beyond end of file! */
-            if(fseek(sfdat->file,byteoffset,SEEK_CUR))
-                return PSF_E_CANT_SEEK;
-            break;
-    }
-    if(fgetpos(sfdat->file,&cur_pos))
-        return PSF_E_CANT_SEEK;
-    if(POS64(cur_pos) >= POS64(sfdat->dataoffset)){
-        sfdat->curframepos = (DWORD)(POS64(cur_pos) -  POS64(sfdat->dataoffset))  / sfdat->fmt.Format.nBlockAlign;
-        if(!sfdat->isRead)  {       /*RWD NEW*/
-            /* we are rewinding a file open for writing */
-            POS64(sfdat->lastwritepos) = sfdat->curframepos;
-        }
-        return PSF_E_NOERROR;
-    }
-    else
-        return PSF_E_CANT_SEEK;
-}
-
-
-/* decide sfile format from the filename extension */
-/* TODO: add func to get format from file header */
-psf_format psf_getFormatExt(const char *path)
-{
-    char *lastdot;
-    if(path==NULL || (strlen(path) < 4))
-        return PSF_FMT_UNKNOWN;             /* TODO: support RAW data... */
-    lastdot = strrchr(path,'.');
-    if(lastdot==NULL)
-        return PSF_FMT_UNKNOWN;
-    
-    if(stricmp(lastdot,".wav")==0)
-        return PSF_STDWAVE;
-    else if((stricmp(lastdot,".aif")==0) || stricmp(lastdot,".aiff")==0)
-        return PSF_AIFF;
-    else if((stricmp(lastdot,".afc")==0) || stricmp(lastdot,".aifc")==0)
-        return PSF_AIFC;
-    /* Ambisonic b-format files */
-    else if(stricmp(lastdot,".wxyz")==0)
-        return PSF_STDWAVE;
-    else if(stricmp(lastdot,".amb")==0)
-        return PSF_WAVE_EX;
-    else
-        return PSF_FMT_UNKNOWN;
-    
-}
-
-/* return 0 for no PEAK data, 1 for success */
-/* NB: we read PEAK data from sfdat, so we can read peaks while writing the file, before closing */
-int psf_sndReadPeaks(int sfd,PSF_CHPEAK peakdata[],MYLONG *peaktime)
-{
-    int i,nchans;
-    PSFFILE *sfdat;
-    
-    if(sfd < 0 || sfd > psf_maxfiles)
-        return PSF_E_BADARG;
-    
-    sfdat  = psf_files[sfd];
-    if(sfdat==NULL)
-        return PSF_E_BADARG;
-#ifdef _DEBUG       
-    assert(sfdat->file);
-    assert(sfdat->filename);
-#endif
-    /* TODO: we may want to have this, for RAW files, even though we won't write it */
-    if(sfdat->pPeaks==NULL){           /*NOT an error: just don't have the chunk*/
-        if(peaktime!= NULL)
-            *peaktime = 0;
-        return 0;
-        
-    }
-    if(peaktime != NULL)
-        *peaktime = (int) sfdat->peaktime;
-    nchans = sfdat->fmt.Format.nChannels;
-    for(i=0;i < nchans;i++){
-        peakdata[i].val = sfdat->pPeaks[i].val;
-        peakdata[i].pos = sfdat->pPeaks[i].pos;
-    }
-    return 1;
-}
-
-static float trirand(void)
-{
-    double r1,r2;
-    r1 = (double) rand() * inv_randmax;
-    r2 = (double) rand() * inv_randmax;
-    
-    return (float)((r1 + r2) - 1.0);
-    
-    
-}
-
-int psf_sndSetDither(int sfd,unsigned int dtype)
-{
-    PSFFILE *sfdat;;
-    
-    if(sfd < 0 || sfd > psf_maxfiles)
-        return PSF_E_BADARG;
-    
-    sfdat  = psf_files[sfd];
-    if(sfdat==NULL)
-        return PSF_E_BADARG;
-#ifdef _DEBUG       
-    assert(sfdat->file);
-    assert(sfdat->filename);
-#endif
-    if(dtype < PSF_DITHER_OFF || dtype > PSF_DITHER_TPDF || sfdat->isRead)
-        return PSF_E_BADARG;
-    
-    sfdat->dithertype = dtype;
-    
-    return PSF_E_NOERROR;
-    
-}
-/* get current dither setting */
-int psf_sndGetDither(int sfd)
-{
-    PSFFILE *sfdat;
-    
-    if(sfd < 0 || sfd > psf_maxfiles)
-        return PSF_E_BADARG;
-    
-    sfdat  = psf_files[sfd];
-    if(sfdat==NULL)
-        return PSF_E_BADARG;
-#ifdef _DEBUG       
-    assert(sfdat->file);
-    assert(sfdat->filename);
-#endif
-    
-    return sfdat->dithertype;
-    
-}
-
-/* OCT 2009 */
-psf_channelformat get_speakerlayout(DWORD chmask,DWORD chans)
-{
-    psf_channelformat chformat = MC_WAVE_EX;    // default is some weird format!
-    
-    /* accept chancount > numbits set in speakermask */
-    switch(chmask){
-            /*check all cross-platform formats first...*/
-        case(SPKRS_UNASSIGNED):
-            chformat = MC_STD;
-            break;
-        case(SPKRS_MONO):
-            if(chans==1)
-                chformat = MC_MONO;
-            break;
-        case(SPKRS_STEREO):
-            if(chans==2)
-                chformat = MC_STEREO;
-            break;
-        case(SPKRS_GENERIC_QUAD):
-            if(chans==4)
-                chformat = MC_QUAD;
-            break;
-        case(SPKRS_SURROUND_LCRS):
-            if(chans==4)
-                chformat = MC_LCRS;
-            break;
-        case(SPKRS_DOLBY5_1):
-            if(chans==6)
-                chformat = MC_DOLBY_5_1;
-            break;
-        case(SPKRS_SURR_5_0):                   /* March 2008 */
-            if(chans==5)
-                chformat = MC_SURR_5_0;
-            break;
-        case(SPKRS_6_1):
-            if(chans==7)
-                chformat = MC_SURR_6_1;
-            break;
-        case(SPKRS_7_1):
-            if(chans==8)
-                chformat = MC_SURR_7_1;
-            break;
-        case SPKRS_CUBE:
-            if(chans==8)
-                chformat = MC_CUBE;
-            break;
-        default:
-            break;
-    }
-    return chformat;
-}
-
-/* NOV 2009 get raw speaker mask */
-
-int psf_speakermask(int sfd)
-{
-    PSFFILE *sfdat;
-    
-    if(sfd < 0 || sfd > psf_maxfiles)
-        return PSF_E_BADARG;
-    
-    sfdat  = psf_files[sfd];
-    if(sfdat==NULL)
-        return PSF_E_BADARG;
-    
-    return (int) sfdat->fmt.dwChannelMask;
-}
-
-/* September 2010 to check output file sizes */
-#define PSF_SIZE_SAFETY (512)
-
-int is_legalsize(unsigned long nFrames, const PSF_PROPS *props)
-{
-    int samplesize = 0;
-    int result  = 0;
-    int blocksize;
-    unsigned long long bytesize;
-    
-    switch(props->samptype){
-        case PSF_SAMP_8:
-            samplesize = 1;
-            break;
-        case PSF_SAMP_16:
-            samplesize = 2;
-            break;
-        case PSF_SAMP_24:
-            samplesize = 3;
-            break;
-        case PSF_SAMP_32:
-        case PSF_SAMP_IEEE_FLOAT:
-            samplesize = 4;
-            break;
-        default:
-            return result;
-    }
-    
-    blocksize = props->chans * samplesize;
-    bytesize = nFrames * blocksize;
-    if( bytesize <=  0xffffffffLL - PSF_SIZE_SAFETY)
-        result = 1;
-    return result;
-}
-
-/* TODO: define a psf_writePeak function; probably to a single nominated channel. 
- This would be needed as soon as write is performed with random over-write activity.
- This is probably something to discourage, however!
- */
-
-

dev/externals/reverb/CMakeLists.txt

@@ -1,31 +0,0 @@
-if(APPLE)
-  set(CMAKE_C_FLAGS "-O2 -Wall -mmacosx-version-min=10.8  -Dunix -fomit-frame-pointer -funroll-loops")
-  set(CMAKE_CXX_FLAGS "-O2 -Wall -mmacosx-version-min=10.8  -Dunix -fomit-frame-pointer -funroll-loops -std=c++11 -stdlib=libc++")
-else()
-  if(MINGW)
-    set(CMAKE_C_FLAGS "-O3 -DWIN32 -D_WIN32 -fomit-frame-pointer  -funroll-loops")
-    set(CMAKE_CXX_FLAGS "-O3 -DWIN32 -D_WIN32 -fomit-frame-pointer  -funroll-loops -static-libgcc -static-libstdc++")
-  else()
-    set(CMAKE_C_FLAGS "-O3 -Wall -Dlinux -Dunix -fomit-frame-pointer -funroll-loops")
-  endif()
-endif()
-
-link_directories(../lib)
-
-include_directories(../../../include ../include)
-  
-add_executable(rmverb rmverb.cpp reverberator.cpp wavetable.cpp)
-target_link_libraries(rmverb portsf sfsys ${EXTRA_LIBRARIES})
-my_install(rmverb)
-
-add_executable(reverb reverb.cpp reverberator.cpp wavetable.cpp)
-target_link_libraries(reverb portsf sfsys ${EXTRA_LIBRARIES})
-my_install(reverb)
-
-add_executable(rmresp roomresp.cpp)
-target_link_libraries(rmresp portsf sfsys ${EXTRA_LIBRARIES})
-my_install(rmresp)
-
-add_executable(tapdelay tdelaymain.cpp reverberator.cpp wavetable.cpp)
-target_link_libraries(tapdelay portsf sfsys ${EXTRA_LIBRARIES})
-my_install(tapdelay)

dev/externals/reverb/LARGERM.TXT

@@ -1,7 +0,0 @@
-1.0               # Input gain for data file (float)
-0                 # Open Path = 0 Closed Path = 1 (int)
-43    31    19       # Room Dimensions (X,Y,Z) (float)
-0.99              # Reflectivity of walls (float)
-5                 # Number of rooms per dimension, 1 or greater
-20    15    2      # Listener coord (X,Y,Z) (float)
-7    15    2   0 # Source coord and time at coord (X,Y,Z,t) (float)

dev/externals/reverb/MEDIUMRM.TXT

@@ -1,7 +0,0 @@
-1.0               # Input gain for data file (float)
-0                 # Open Path = 0 Closed Path = 1 (int)
-21   16    5       # Room Dimensions (X,Y,Z) (float)
-0.9              # Reflectivity of walls (float)
-5                 # Number of rooms per dimension, 1 or greater
-12    8    2      # Listener coord (X,Y,Z) (float)
-5    8    2   0 # Source coord and time at coord (X,Y,Z,t) (float)

dev/externals/reverb/POS.TXT

@@ -1,7 +0,0 @@
-1.0               # Input gain for data file (float)
-0                 # Open Path = 0 Closed Path = 1 (int)
-20    10    6       # Room Dimensions (X,Y,Z) (float)
-0.5              # Reflectivity of walls (float)
-5                 # Number of rooms per dimension, 1 or greater
-15    5    2      # Listener coord (X,Y,Z) (float)
-5    4    2   0 # Source coord and time at coord (X,Y,Z,t) (float)

dev/externals/reverb/allpass.c

@@ -1,252 +0,0 @@
-/*
- * Copyright (c) 1983-2013  Composers Desktop Project Ltd
- * http://www.composersdesktop.com
- * This file is part of the CDP System.
- * The CDP System is free software; you can redistribute it
- * and/or modify it under the terms of the GNU Lesser General Public 
- * License as published by the Free Software Foundation; either 
- * version 2.1 of the License, or (at your option) any later version. 
- *
- * The CDP System is distributed in the hope that it will be useful, 
- * but WITHOUT ANY WARRANTY; without even the implied warranty of 
- * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  
- * See the GNU Lesser General Public License for more details.
- * You should have received a copy of the GNU Lesser General Public
- * License along with the CDP System; if not, write to the Free Software
- * Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
- *
- */
- 
-/*	allpass.c
- *	allpass filter
- *	A. Bentley, Composers' Desktop Project, Version Nov 1987
- */
-
-
-/*
- *	$Log: allpass.c%v $
- * Revision 3.3  1994/10/31  22:08:15  martin
- * first rcs version
- *
- */
-
-#include <stdio.h>
-#include <stdlib.h>	
-#include <math.h>
-#include <sfsys.h>
-#include <cdplib.h>
-static char *sfsys_h_rcsid = SFSYS_H_RCSID;
-static char *cdplib_h_rcsid = CDPLIB_H_RCSID;
-
-void usage();
-void allpass_long(long,long,long,long,const short*,short*,long*,short *);
-
-#define BUFLEN	(32768)
-#define FP_ONE  (65536.0)
-#define LOG_PS  (16)
-
-PROGRAM_NUMBER(0xf93c2700);
-
-void
-main(argc,argv)
-int argc;
-char *argv[];
-{
-	int ifd, ofd;
-	int sampsize;
-	int quickflag = 1;	/* default is fast fixed point calc */
-	register int rv;
-	float	ip, op;
-#ifdef ORIGINAL_VERSION
-	long	lip, lop;
-	long	i;
-#endif
-	float 	*delbuf1;
-	float   *delbuf2;
-	long	rdsamps;
-	long	*ldelbuf1;
-	short	*sdelbuf2;
-	short	*sinbuf;
-	short	*soutbuf;
-	double  delay; 
-	float	gain, prescale = 1.0f;
-	long	lgain, lprescale;
-	long 	delsamps;
-	long	dbp1 = 0, dbp2 = 0;
-	long	chans;
-	long	isr;
-	double	sr;
-
-	if(sflinit("allpass")){
-		sfperror("Allpass: initialisation");
-		exit(1);
-	}
-    	while(argv[1] != 0 && argv[1][0] == '-') {
-	      switch(argv[1][1]) {
-	      case 'f':	
-		 quickflag--;
-		 argv++;
-		 argc--;
-		 break;
-	      default:
-	         usage();
-	         break;
-	      }
-	}
-	if(argc < 5 || argc > 6) usage();
-
-	if(argv[1] != 0) {
-	      	if((ifd = sndopen(argv[1])) < 0) {
-		   fprintf(stderr,"Allpass: cannot open input file %s\n", argv[1]);
-		   exit(1);
-		}
-	}
-	if(sndgetprop(ifd,"sample rate",(char *) &isr, sizeof(long)) < 0){
-		fprintf(stderr,
-		  "\nAllpass: failed to read sample rate for file %s",argv[1]);
-		exit(1);
-	}
-	sr = (double) (isr);
-	if(sndgetprop(ifd,"channels",(char *) &chans, sizeof(long)) < 0){
-		fprintf(stderr,
-		  "\nAllpass: failed to read channel data for file %s", argv[1]);
-		exit(1);
-	}
-	if(chans != 1){
-		fprintf(stderr,"\nAllpass works only on mono files");
-		exit(1);
-	}
-	if(sndgetprop(ifd,"sample type",(char *) &sampsize, sizeof(long)) < 0){
-		fprintf(stderr,
-		  "\nAllpass: failed to read sample type for file %s", argv[1]);
-		exit(1);
-	}
-	/* get command line args */
-	delay = abexpr(argv[3],sr);
-	if(delay < 0.0){
-		fprintf(stderr,"\nAllpass: invalid delay parameter");
-		exit(1);
-	}
-	gain = (float)abexpr(argv[4],sr);
-	if((gain < -1.0) || (gain > 1.0)){
-		fprintf(stderr,"\nAllpass: gain out of range");
-		exit(1);
-	}
-
-	if((ofd = sndcreat(argv[2],-1,sampsize)) < 0) {
-		  fprintf(stderr,"Cannot open output file %s\n",argv[2]);
-		  exit(1);
-	}
-
-	lgain = (long) (gain * FP_ONE);
-	if(argc > 5) {
-		prescale = (float)abexpr(argv[5],sr);
-		if((prescale < -1.0) || (prescale > 1.0))
-		   fprintf(stderr,
-			"Allpass: warning - prescale exceeds +/-1\n");
-	}
-	lprescale = (long) (prescale * FP_ONE);		
-	/* allocate buffer for delay line */
-	delsamps = (long) (delay * sr);
-	if(quickflag) {
-	   if(( sinbuf = (short *) malloc(BUFLEN * sizeof(short))) == NULL) {
-		fprintf(stderr,"\nNot enough memory for input buffer");
-		exit(1);
-	   }
-	   if(( soutbuf = (short *) malloc(BUFLEN * sizeof(short))) == NULL) {
-		fprintf(stderr,"\nNot enough memory for output buffer");
-		exit(1);
-	   }
-	   if(( ldelbuf1 = (long *) calloc(delsamps, sizeof(long))) == NULL) {
-		fprintf(stderr,"\nNot enough memory for delay buffer 1");
-		exit(1);
-	   }
-	   if(( sdelbuf2 = (short *) calloc(delsamps, sizeof(short))) == NULL) {
-		fprintf(stderr,"\nNot enough memory for delay buffer 2");
-		exit(1);
-	   }
-	}else{
-	   if(( delbuf1 = (float *) calloc(delsamps, sizeof(float))) == NULL) {
-		fprintf(stderr,"\nNot enough memory for delay buffer 1");
-		exit(1);
-	   }
-	   if(( delbuf2 = (float *) calloc(delsamps, sizeof(float))) == NULL) {
-		fprintf(stderr,"\nNot enough memory for delay buffer 2");
-		exit(1);
-	   }
-	}
-	stopwatch(1);
-    
-	for(;;){
-		if((rv = fgetfloat(&ip,ifd)) < 0){
-			fprintf(stderr,"\nError in reading file"); 
-			exit(1);
-		}
-		if(!rv) break;
-			ip *= prescale;
-		op = (-gain) * ip;
-		op += delbuf1[dbp1];
-		op += gain * delbuf2[dbp2];
-			delbuf1[dbp1++] = ip;
-		delbuf2[dbp2++] = op;
-
-		if(dbp1 >= delsamps) dbp1 = 0;
-		if(dbp2 >= delsamps) dbp2 = 0;
-
-   		if(fputfloat(&op,ofd) < 1){
-			fprintf(stderr,"\nError writing to output file"); 
-			exit(1);
-		}	
-    }
-	stopwatch(0);
-
-	sndclose(ifd);
-	if(sndputprop(ofd,"sample rate", (char *) &isr, sizeof(long)) < 0){
-		fprintf(stderr,"\nAllpass: failed to write sample rate");
-	}
-	if(sndputprop(ofd,"channels",(char *) &chans, sizeof(long)) < 0){
-		fprintf(stderr,"\nAllpass: failed to write channel data");
-	}
-	sndclose(ofd);
-}
-
-void
-usage()
-{
-	fprintf(stderr,"\nAllpass - A Groucho Program: $Revision: 3.3 $");
-	fprintf(stderr,"\n%s\n",
-	"usage: allpass [-f] infile outfile delay gain [prescale]");
-	exit(0);
-}
-
-/*RWD: my allpass functions */
-//TODO: optimize using ptr arithmetic
-void allpass_long(long prescale, 
-				   long buflen, 
-				   long l_gain,
-				   long l_delsamps,
-				   const short *s_inbuf,
-				   short *s_outbuf,
-				   long *l_delbuf1,
-				   short *s_delbuf2)
-{
-	int i; 
-	static long	db_p1 = 0;
-	static long db_p2 = 0;
-	//do allpass on block of samples
-	for(i = 0; i < buflen; i++){
-		   long lip,lop;
-		   lip = s_inbuf[i] * prescale;
-		   lop = (-l_gain) * (lip >> LOG_PS);
-		   lop += l_delbuf1[db_p1];
-		   lop += l_gain * s_delbuf2[db_p2];
-
-		   l_delbuf1[db_p1++] = lip;
-		   s_outbuf[i] = s_delbuf2[db_p2++] = (short) (lop >> LOG_PS);
-
-		   if(db_p1 >= l_delsamps) db_p1 = 0;
-		   if(db_p2 >= l_delsamps) db_p2 = 0;
-    }
-	   //block done
-	//return &soutbuf[0];			//if we return anything?
-}

dev/externals/reverb/delay.c

@@ -1,546 +0,0 @@
-/*
- * Copyright (c) 1983-2013  Composers Desktop Project Ltd
- * http://www.composersdesktop.com
- * This file is part of the CDP System.
- * The CDP System is free software; you can redistribute it
- * and/or modify it under the terms of the GNU Lesser General Public 
- * License as published by the Free Software Foundation; either 
- * version 2.1 of the License, or (at your option) any later version. 
- *
- * The CDP System is distributed in the hope that it will be useful, 
- * but WITHOUT ANY WARRANTY; without even the implied warranty of 
- * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  
- * See the GNU Lesser General Public License for more details.
- * You should have received a copy of the GNU Lesser General Public
- * License along with the CDP System; if not, write to the Free Software
- * Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
- *
- */
- 
-/* delay.c  
- * generalised delay line featuring
- * feedthrough, feedforward and feedback
- * A. Bentley, Composers' Desktop Project, Nov 1987.
- * Revised and extended to stereo and command-line mode 
- * by Richard Dobson Oct 1993.
- */ 
-
-static char *rcsid = "$Id: delay.c%v 3.3 1994/10/31 22:30:12 martin Exp $";
-/*
- *	$Log: delay.c%v $
- * Revision 3.3  1994/10/31  22:30:12  martin
- * first rcs version
- *
- */
-
-#include <stdio.h>
-#include <stdlib.h>
-#include <math.h>
-#include "sfsys.h"
-#include "cdplib.h"
-static char *sfsys_h_rcsid = SFSYS_H_RCSID;
-static char *cdplib_h_rcsid = CDPLIB_H_RCSID;
-
-#define remainder	my_remainder
-
-#define BUFLEN	  (4096)
-#define	DELBUFSIZ (176400)	/* allows 2sec delay in stereo */
-#define ONE_FLT   (65536.0)
-
-void domono();
-void dostereo();
-void usage();
-void delinit(int cmndline,char *argv[]);
-void initfbuf();
-void initlbuf();
-
-int 	Funcflag = 0;
-int 	Quickflag = 1;
-long 	Nchans;
-long	Delaysamps;
-long    bufcnt;
-long	remainder;
-float 	*Fdelbuf;
-long	*Ldelbuf;
-double 	Sr;
-double	Delay;
-float	Feedforgain;
-float	Feedbckgain;
-float	Delaygain;
-float	Prescale = 1.0F;
-double	Invert =1.0;
-double  Rescale;
-long	Lfeedforgain;
-long	Lfeedbckgain;
-long	Ldelaygain;
-long	Lprescale;
-double  Trailtime = 0.;
-long	Trailsamps = 0;
-int	ifd;	 	/* input file descriptor */
-int	ofd;		/* output file descriptor */
-long 	isr;
-long	sampsize;
-short  	*inbuf;
-short	*outbuf;
-
-static char version[] =
-	"~CDPVER~GROUCHO: DELAY Portable version 3.0, REVISED BY R. DOBSON OCT 1993.";
- 
-PROGRAM_NUMBER(0xf93c2705);
-
-void
-main(argc,argv)
-int argc;
-char *argv[];
-{
-int cmndline = 0;
-char *ptr;
-	
-	if(sflinit("delay")) {
-		sfperror("Delay: initialisation");
-		exit(1);
-	}
-    	while(argv[1] != 0 && argv[1][0] == '-') {
-	      switch(argv[1][1]) {
-	          case 'f':	
-		 	Quickflag--;
-		 	break;
-		  case 'p':
-			ptr = (char *) &(argv[1][2]);
-			if(*ptr==0)
-			{
-				usage();
-			}
-			Prescale = (float)atof(ptr);
-			if(Prescale==0.0)
-			{
-			    printf("\ndelay: -p error, prescale = 0.0!\n");
-			    usage();
-			}
-			break;
-		   case 'i':
-			Invert = -1.0;
-			break;
-	          default:
-	         	usage();
-	         	break;
-	      }
-	argv++; argc--;
-	}
-	if(argc < 3) usage();
-	if(argc>4)
-	{
-	    if(argc<7)
-		{
-		fprintf(stderr,"\ndelay: incorrect no of parameters");
-		usage();
-		}
-	    else
-		{
-		cmndline=1;
-		}
-	}
-	if((inbuf = 
-		(short *) malloc((BUFLEN) * sizeof(short))) == NULL) {
-		fprintf(stderr,"\ndelay: not enough memory for input buffer");
-		exit(1);
-	}
-	if((outbuf = 
-		(short *) malloc(BUFLEN * sizeof(short))) == NULL) {
-		fprintf(stderr,"\ndelay: not enough memory for output buffer");
-		exit(1);
-	}
-      	if((ifd = sndopen(argv[1])) < 0) {
-	  fprintf(stderr,"\ndelay: unable to open input file\n");
- 	  exit(1);
-	} 
-	if(sndgetprop(ifd,"sample rate",(char *) &isr, sizeof(long)) < 0) {
-		fprintf(stderr,"\ndelay: failed to get sample rate");
-		exit(1);
-	}
-	if(sndgetprop(ifd,"channels",(char *) &Nchans, sizeof(long)) < 0) {
-		fprintf(stderr,"\ndelay: failed to get channel data");
-		exit(1);
-	}
-	if(sndgetprop(ifd,"sample type",(char *) &sampsize, sizeof(long)) < 0){
-		fprintf(stderr,"\ndelay: failed to get sample type");
-		exit(1);
-	}
-	if(Nchans > 2) {
-	  fprintf(stderr,"\ndelay: too many channels! Mono or stereo only.\n");
-	  exit(1);
-	}
-
-	Sr = (double) isr;
-
-	delinit(cmndline,argv);
-
-	if((ofd = sndcreat(argv[2],-1,sampsize)) < 0) {
-	  fprintf(stderr,"\ndelay: unable to open output file\n");
-	  exit(1);
-	}
-
-	if(!Quickflag) 
-		initfbuf();
-	else
-		initlbuf();
-	stopwatch(1);
-	if(Nchans==1) {domono();} else {dostereo();}
-	stopwatch(0);
-	sndclose(ifd);
-   	if(sndputprop(ofd,"sample rate",(char *) &isr, sizeof(long)) < 0){
-	fprintf(stderr,"\ndelay: failed to write sample rate");
-   	}
-   	if(sndputprop(ofd,"channels", (char *) &Nchans, sizeof(long)) < 0){
-	fprintf(stderr,"\ndelay: failed to write channel data");
-   	}
-	free(inbuf);
-	free(outbuf);
-	if(!Quickflag) {free(Fdelbuf);} else {free(Ldelbuf);}
-   	sndclose(ofd);
-	sffinish();
-}
-
-
-void
-domono()
-{
-	int 	i;	
-	long	rdsamps;
-	float	input,output;
-	long	linput,loutput;
-	long	ltemp1,ltemp2;
-	long	ipptr=0,opptr=0;
-
-	do		/* while rdsamps */
-	{
-	   	if((rdsamps = fgetsbuf(inbuf, BUFLEN, ifd)) < 0) {
-        	fprintf(stderr,"\ndelay: failure to read input file\n");
-          	exit(1);
-	    }
-	    
-	    for(i=0;i<rdsamps;i++)
-		{
-			
-				/* delay line */
-			input = inbuf[i] * Prescale;
-			output = (input * Feedforgain) +		  //dry signal
-				(Fdelbuf[opptr] * Delaygain);
-			Fdelbuf[ipptr] = Fdelbuf[opptr++] * Feedbckgain; 
-			Fdelbuf[ipptr++] += input ;
-			outbuf[i] = (short) output;				  //dry + wet
-
-			if(ipptr >= Delaysamps) ipptr -= Delaysamps;
-			if(opptr >= Delaysamps) opptr -= Delaysamps;
-			 
-			
-			if(ipptr < 0 || opptr < 0) {			
-				 printf(
-			  "\ninternal error, ipptr=%d,opptr=%d\n",ipptr,opptr);
-			}	
-		}	
-        if(fputsbuf(outbuf,rdsamps,ofd) < rdsamps) {
-		     fprintf(stderr,
-				"\ndelay: failure in writing to output file\n");
-				exit(1);
-		}
-		inform(rdsamps,Sr);
-	} while(rdsamps > 0);
-
-	    	/* now do trailer	*/
-    rdsamps=BUFLEN;
-	if (Trailsamps>0) {
-		do	{	/* while bufcnt */
-		    
-		  	if(!bufcnt) 
-				rdsamps = remainder;
-					    	
-			for(i=0;i<rdsamps;i++) {
-		       	output=(Fdelbuf[opptr] * Delaygain);
-		       	Fdelbuf[ipptr++]  = 
-				Fdelbuf[opptr++] *Feedbckgain;
-		       	outbuf[i] = (short) output;
-		       	if(ipptr >= Delaysamps) ipptr -= Delaysamps;
-		       	if(opptr >= Delaysamps) opptr -= Delaysamps;
-		    }
-					  		
-        	if(fputsbuf(outbuf,rdsamps,ofd) < rdsamps){
-				fprintf(stderr,
-					"\ndelay: failure in writing to output file\n");
-				exit(1);
-		  	}
-    		inform(BUFLEN,Sr);
-			bufcnt -=1;
-    	} while(bufcnt>=0);
-    }				/* end if Trailsamps */
-}				/* end domono	*/
-
-void
-dostereo()			/* do stereo delay */
-{
-int 	i;	
-long	rdsamps;
-float	Linput,Rinput;
-float 	Loutput,Routput;
-long	Llinput,Rlinput;
-long	Lloutput,Rloutput;
-long	Ltemp1,Ltemp2;
-long	Rtemp1,Rtemp2;
-long	Lipptr = 0,Ripptr = 1;
-long	Lopptr = 0,Ropptr = 1;
-
-    do			/* while rdsamps */
-    {
-	if((rdsamps = fgetsbuf(inbuf,BUFLEN,ifd)) < 0)
-	{
-		fprintf(stderr,"\ndelay: failure to read input file\n");
-		exit(1);
-	}
-	    for(i=0;i<(rdsamps-1);i+=2)
-	    {
-		if(!Quickflag)
-		{
-			Linput=inbuf[i]*Prescale;
-			Rinput=inbuf[i+1]*Prescale;
-			Loutput=(Linput*Feedforgain) + 
-				(Fdelbuf[Lopptr]*Delaygain);
-			Routput=(Rinput*Feedforgain) + 
-				(Fdelbuf[Ropptr]*Delaygain);
-			Fdelbuf[Lipptr] = Fdelbuf[Lopptr]*Feedbckgain;
-			Lopptr+=2;
-			Fdelbuf[Ripptr] = Fdelbuf[Ropptr]*Feedbckgain;
-			Ropptr+=2;
-			Fdelbuf[Lipptr]+=Linput;
-			Fdelbuf[Ripptr]+=Rinput;
-			Lipptr+=2;
-			Ripptr+=2;
-			outbuf[i] = (short) Loutput;
-			outbuf[i+1] = (short) Routput;
-
-			if(Lipptr >= Delaysamps) Lipptr -= Delaysamps;
-			if(Ripptr >= Delaysamps) Ripptr -= Delaysamps;
-			if(Lopptr >= Delaysamps) Lopptr -= Delaysamps;
-			if(Ropptr >= Delaysamps) Ropptr -= Delaysamps;
-			}
-		else
-		{
-			Llinput = inbuf[i] * Lprescale;
-			Rlinput = inbuf[i+1] * Lprescale;
-			Ltemp1 = Llinput >> 16;
-			Rtemp1 = Rlinput >> 16;
-			Ltemp2 = Ldelbuf[Lopptr] >> 16;
-			Rtemp2 = Ldelbuf[Ropptr] >> 16;
-				Lopptr += 2;
-				Ropptr += 2;
-			Lloutput = (Ltemp1 * Lfeedforgain) + 
-				(Ltemp2 * Ldelaygain);
-			Rloutput = (Rtemp1 * Lfeedforgain) +
-				(Rtemp2 * Ldelaygain);
-			Ldelbuf[Lipptr] = Ltemp2 * Lfeedbckgain;
-			Ldelbuf[Ripptr] = Rtemp2 * Lfeedbckgain;
-			Ldelbuf[Lipptr] += Llinput;
-			Ldelbuf[Ripptr] += Rlinput;
-				Lipptr += 2;
-				Ripptr += 2;
-			outbuf[i] = (short)(Lloutput>>16);
-			outbuf[i+1] = (short)(Rloutput>>16);
-
-			if(Lipptr >= Delaysamps) Lipptr -= Delaysamps;    
-			if(Ripptr >= Delaysamps) Ripptr -= Delaysamps;
-			if(Lopptr >= Delaysamps) Lopptr -= Delaysamps;
-			if(Ropptr >= Delaysamps) Ropptr -= Delaysamps;
-		}
-	}
-	if(fputsbuf(outbuf,rdsamps,ofd) < rdsamps) {
-		fprintf(stderr,"\ndelay: failure in writing to output file\n");
-		exit(1);
-	}
-	inform(rdsamps/2,Sr);
-   }while(rdsamps > 0);		/* end do */
-
-/* now do trailer */
-
-    rdsamps=BUFLEN;
-    if(Trailsamps>0)
-    {
-	do
-	{
-		if(!bufcnt) rdsamps = remainder;
-		if(!Quickflag)
-		    {
-		    for(i=0;i<(rdsamps-1);i+=2)
-			{
-			Loutput = Fdelbuf[Lopptr]*Delaygain;
-			Routput = Fdelbuf[Ropptr]*Delaygain;
-			Fdelbuf[Lipptr] = Fdelbuf[Lopptr]*Feedbckgain;
-			Fdelbuf[Ripptr] = Fdelbuf[Ropptr]*Feedbckgain;
-			Lopptr+=2;
-			Ropptr+=2;
-			Lipptr+=2;
-			Ripptr+=2;
-			outbuf[i] = (short) Loutput;
-			outbuf[i+1] = (short) Routput;
-			if(Lipptr >= Delaysamps) Lipptr -= Delaysamps;
-			if(Ripptr >= Delaysamps) Ripptr -= Delaysamps;
-			if(Lopptr >= Delaysamps) Lopptr -= Delaysamps;
-			if(Ropptr >= Delaysamps) Ropptr -= Delaysamps;
-			}
-		    }
-		else
-		    {
-		    for(i=0;i<(rdsamps-1);i+=2)
-		    	{
-			Ltemp2 = Ldelbuf[Lopptr] >> 16;
-				Lopptr+=2;
-			Rtemp2 = Ldelbuf[Ropptr] >> 16;
-				Ropptr+=2;
-			Lloutput = Ltemp2 * Ldelaygain;
-			Rloutput = Rtemp2 * Ldelaygain;
-			Ldelbuf[Lipptr] = Ltemp2*Lfeedbckgain;
-			Ldelbuf[Ripptr] = Rtemp2*Lfeedbckgain;
-			Lipptr +=2;
-			Ripptr +=2;			
-			outbuf[i] = (short)(Lloutput>>16);
-			outbuf[i+1] = (short)(Rloutput>>16);
-			if(Lipptr >= Delaysamps) Lipptr -= Delaysamps;
-			if(Ripptr >= Delaysamps) Ripptr -= Delaysamps;
-			if(Lopptr >= Delaysamps) Lopptr -= Delaysamps;
-			if(Ropptr >= Delaysamps) Ropptr -= Delaysamps;
-			}
-		    }
-		if(fputsbuf(outbuf,rdsamps,ofd) < rdsamps) {
-		fprintf(stderr,"\ndelay: failure in writing to output file\n");
-		exit(1);
-		}
-		inform(rdsamps/2,Sr);
-		bufcnt -=1;
-	    } while (bufcnt>=0);
-	}				/* end if */
-   }					/* end dostereo */		
- 
-void
-usage()
-{
-	fprintf(stderr,"delay - A Groucho Program: $Revision: 3.3 $\n");
-	fprintf(stderr,
-"usage: delay [-f][-pN][-i] infile outfile [4 cmndline parameters]\n");
-	fprintf(stderr,
-"	where -f sets output in floatsams, -p sets prescale = N for infile,\n");
-	fprintf(stderr,
-"	and -i inverts dry signal (for allpass filters).\n");
-	fprintf(stderr,"\n%s\n%s\n%s\n%s\n%s\n%s\n%s\n%s\n%s\n",
-"Cmndline option: after [flags] filenames, enter all parameters in this order:",
-"	delaytime mix feedbackgain trailertime",
-"	 0.0  	< delaytime (msecs) 	<= maxdelay",
-"	 0.0 	<= mix		 	<= 1.0",
-"	-1.0 	<= feedbackgain    	<= 1.0",
-"	 0.0 	<= trailertime (secs)	<= 30.0",
-"where maxdelay =  8000ms @22050 mono",
-"               =  4000ms @22050 stereo, and 44100 mono",
-"               =  2000ms @44100 stereo"); 
-	exit(0);
-}
-
-void
-delinit(int cmndlin, char *argv[])
-{
-	double temp,maxdelay,mix;
-	char msg[80];
-	/*	Some jiggery pokery is needed in setting the
-	 *	fixed point variables to avoid overloading
-	 *	the input to the delay line. The input is
-	 *	divided by two in conjunction with prescaling
-	 *	and the feedthrough and delay outputs are
-	 *	raised by two to compensate.
-	 */
-	maxdelay = ((double) (DELBUFSIZ) / Sr) * 1000.0; /* milliseconds */
-	if(Nchans==2) {maxdelay /= 2;}
-	if(cmndlin) 
-	{
-		temp = atof(argv[3]);
-		if (temp > maxdelay)
-		{
-			printf("delay: delay time too long\n");
-			exit(1);
-		}
-		Delaysamps = (long) ((Sr *temp) / 1000.0);
-		mix = atof(argv[4]);
-		if((mix < 0.0) || (mix > 1.0))
-		{
-			printf("delay: mix value out of range\n");
-			exit(1);
-		}
-		Feedbckgain = (float)atof(argv[5]);
-		if((Feedbckgain < -1.0) || (Feedbckgain > 1.0))
-		{
-			printf("delay: feedbackgain out of range\n");
-			exit(1);
-		}
-		Trailtime = atof(argv[6]);
-		if((Trailtime < 0.0) || (Trailtime > 1000.0))
-		{
-			printf("delay: trailtime out of range\n");
-			exit(1);
-		}
-				
-	}
-	else		/* ask for params from terminal */
-	{
-		printf("max delay in msec = %f\n",maxdelay);
-		sprintf(msg,
-"Give delay time in milliseconds			:");
-		temp  = accexpr(msg,0.0,maxdelay,0.0,0,Sr);
-		if(temp==0.0)
-		{
-	printf("\nzero delay time!!! No point in running program. Bye!\n");
-	exit(1);
-		}
-		Delaysamps = (long) ((Sr * temp) / 1000.0);
-		printf("\ndelay length in samples = %ld \n",Delaysamps);
-		sprintf(msg,
-"Give mix proportion (0.0 <= level <= 1.0)		:");
-		mix = accexpr(msg,0.0,1.0,0.0,0,Sr);
-		sprintf(msg,
-"Give feedback gain (-1.0 <= level <= 1.0)		:");
-		Feedbckgain = (float)accexpr(msg,-1.0,1.0,0.0,0,Sr);
-		sprintf(msg,
-"Give trailer time (default = 0 <= trailtime <= 30.0)	:");
-		Trailtime = accexpr(msg, 0.0,30.0,0.0,1,Sr);
-	}
-/* now massage parameters 			*/
-/* nb delaysamps is per channel here 	*/
- 
-	Feedforgain = (float)((1.0 - mix)*Invert);	
-	Lfeedbckgain = (long) (Feedbckgain * ONE_FLT);
-	Delaygain = (float)mix;
-	Ldelaygain = (long) (Delaygain * ONE_FLT * 2.0);	
-	Rescale = (1.0 / (Feedbckgain + 1.0)); /* i/p compensation */
-	Prescale *= (float)Rescale;
-	Lprescale= (long) (Prescale * ONE_FLT * .5);
-	Feedforgain /= (float)Rescale;
-	Lfeedforgain = (long) (Feedforgain * ONE_FLT * 2.0);	
-	Trailsamps = (long) (Trailtime * Sr);
-	if(Nchans==2) Trailsamps *= 2;
-	bufcnt = Trailsamps/BUFLEN;
-	remainder = Trailsamps % BUFLEN;
-}
-
-void
-initfbuf()
-{
-	Delaysamps*=Nchans;
-	if((Fdelbuf = (float *) calloc(Delaysamps+2, sizeof(float))) == NULL){
-	   fprintf(stderr,"\ndelay: not enough memory for main delay buffer");
-	   exit(1);
-	}
-}
-
-void
-initlbuf()
-{
-	Delaysamps*=Nchans;
-	if((Ldelbuf = (long *) calloc(Delaysamps+4, sizeof(long))) == NULL){
-	    fprintf(stderr,"\ndelay: not enough memory for main delay buffer");
-	    exit(1);
-	}
-}

dev/externals/reverb/diffuse.h

@@ -1,62 +0,0 @@
-/*
- * Copyright (c) 1983-2013 Richard Dobson and Composers Desktop Project Ltd
- * http://people.bath.ac.uk/masrwd
- * http://www.composersdesktop.com
- * This file is part of the CDP System.
- * The CDP System is free software; you can redistribute it
- * and/or modify it under the terms of the GNU Lesser General Public 
- * License as published by the Free Software Foundation; either 
- * version 2.1 of the License, or (at your option) any later version. 
- *
- * The CDP System is distributed in the hope that it will be useful, 
- * but WITHOUT ANY WARRANTY; without even the implied warranty of 
- * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  
- * See the GNU Lesser General Public License for more details.
- * You should have received a copy of the GNU Lesser General Public
- * License along with the CDP System; if not, write to the Free Software
- * Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
- *
- */
- 
-//diffuse.h
-//definition of classes for small, meduim large dense reverberators
-// based on gardner, using nested allpasses
-#ifndef __DIFFUSE_INCLUDED__
-#define __DIFFUSE_INCLUDED__
-
-#include "reverberator.h"
-#include "tapdelay.h"
-
-typedef struct nested_allpass_data{
-	unsigned int time1;
-	unsigned int time2;
-	unsigned int time3;
-	double gain1;
-	double gain2;
-	double gain3;
-}
-NESTDATA;
-
-
-class small_diffuser {
-public:
-	small_diffuser(unsigned int pre_delay,
-					const NESTDATA *apdata1,
-					const NESTDATA *apdata2,double lp_gain,double lp_coeff);
-	virtual ~small_diffuser();
-	bool create(void);
-	float tick(float insam);
-	//void clear(void);
-
-private:
-	NESTDATA ap1_data,ap2_data;
-	nested_allpass *ap1,*ap2;
-	delay *predelay;
-	lowpass1 *lp1;
-	unsigned int predelay_time;
-	double lpgain,lpcoeff;
-	float out1,out2;
-};
-
-
-#endif

dev/externals/reverb/lpcomb.cpp

@@ -1,169 +0,0 @@
-/*
- * Copyright (c) 1983-2013 Richard Dobson and Composers Desktop Project Ltd
- * http://people.bath.ac.uk/masrwd
- * http://www.composersdesktop.com
- * This file is part of the CDP System.
- * The CDP System is free software; you can redistribute it
- * and/or modify it under the terms of the GNU Lesser General Public 
- * License as published by the Free Software Foundation; either 
- * version 2.1 of the License, or (at your option) any later version. 
- *
- * The CDP System is distributed in the hope that it will be useful, 
- * but WITHOUT ANY WARRANTY; without even the implied warranty of 
- * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  
- * See the GNU Lesser General Public License for more details.
- * You should have received a copy of the GNU Lesser General Public
- * License along with the CDP System; if not, write to the Free Software
- * Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
- *
- */
- 
-#include <stdio.h>
-#include <stdlib.h>
-#include <math.h>
-
-//lpcomb.cpp
-extern "C" {
-#include <sfsys.h>
-#include <cdplib.h>	   //NB requires stdio.h etc - time to change this?
-}
-
-#include "reverberator.h"
-
-//straight from Csound: nreverb (vdelay.c)
-static bool prime( int val )
-{
-    int i, last;
-	
-    last = (int)sqrt( (double)val );
-    for ( i = 3; i <= last; i+=2 ) {
-      if ( (val % i) == 0 ) return false;
-    }
-    return true;
-}
-
-static int to_prime(double dur,double sr)
-{
-		int time = (int) (dur * sr);
-		if(time % 2== 0) time += 1;
-		while(!prime(time)) 
-			time +=2;
-		return time;
-}
-
-void
-usage()
-{
-
-	printf("\nusage: lpcomb  infile outfile looptime rvbtime filtgain");
-	exit(0);
-}
-
-void
-main(int argc,char *argv[])
-
-{
-	int		ifd, ofd;	
-	//double  delaytime,gain;
-	double	rvbtime,gain2,filtgain,dloop; 	
-	long	isr,chans,sampsize,ilooptime;
-	double	sr;
-	lpcomb	*comb1 = 0;
-
-
-	if(sflinit("lpcomb")){
-		sfperror("lpcomb: initialisation");
-		exit(1);
-	}
-
-
-	if(argc < 6)
-		usage();
-	dloop = atof(argv[3]);
-	if(dloop <= 0.0){
-		fprintf(stderr,"\nlpcomb: invalid looptime parameter");
-		exit(1);
-	}
-
-	rvbtime = atof(argv[4]);
-	if(rvbtime <= 0.0){
-		fprintf(stderr,"\nlpcomb: invalid delay parameter");
-		exit(1);
-	}
-	filtgain = atof(argv[5]);
-	if((filtgain < 0.0) || (filtgain > 1.0)){
-		fprintf(stderr,"\nlpcomb: filter gain %lf out of range",filtgain);
-		exit(1);
-	}
-
-	if(argv[1] != 0) {
-	    if((ifd = sndopen(argv[1])) < 0) {
-		   fprintf(stderr,"Allpass: cannot open input file %s\n", argv[1]);
-		   exit(1);
-		}
-	}
-	if(sndgetprop(ifd,"sample rate",(char *) &isr, sizeof(long)) < 0){
-		fprintf(stderr,
-		  "\nAllpass: failed to read sample rate for file %s",argv[1]);
-		exit(1);
-	}
-	sr = (double) (isr);
-	if(sndgetprop(ifd,"channels",(char *) &chans, sizeof(long)) < 0){
-		fprintf(stderr,
-		  "\nAllpass: failed to read channel data for file %s", argv[1]);
-		exit(1);
-	}
-	if(chans != 1){
-		fprintf(stderr,"\nAllpass works only on mono files");
-		exit(1);
-	}
-	if(sndgetprop(ifd,"sample type",(char *) &sampsize, sizeof(long)) < 0){
-		fprintf(stderr,
-		  "\nAllpass: failed to read sample type for file %s", argv[1]);
-		exit(1);
-	}
-
-
-	if((ofd = sndcreat(argv[2],-1,sampsize)) < 0) {
-		  fprintf(stderr,"Cannot open output file %s\n",argv[2]);
-		  exit(1);
-	}
-	ilooptime  = to_prime(dloop, sr);	
-	gain2 = exp( log(0.001)* (dloop/rvbtime)) * (1. - filtgain) ;
-	printf("\ngain2 = %lf\tfiltgain = %lf",gain2,filtgain);
-	comb1 = new lpcomb(ilooptime,gain2,filtgain,1.0);
-	if(!comb1->create()){
-		printf("\nFailed to create comb filter");
-		exit(1);
-	}
-
-	stopwatch(1);
-
-	for(;;){
-		int rv;
-		float ip,op;
-
-		if((rv = fgetfloat(&ip,ifd)) < 0){
-			fprintf(stderr,"\nError in reading file"); 
-			exit(1);
-		}
-		if(!rv)
-			break;
-		op = comb1->tick(ip);
-		if(fputfloat(&op,ofd) < 1){
-			fprintf(stderr,"\nError writing to output file"); 
-			exit(1);
-		}
-	}
-	stopwatch(0);
-	sndclose(ifd);
-	if(sndputprop(ofd,"sample rate", (char *) &isr, sizeof(long)) < 0){
-		fprintf(stderr,"\nAllpass: failed to write sample rate");
-	}
-	if(sndputprop(ofd,"channels",(char *) &chans, sizeof(long)) < 0){
-		fprintf(stderr,"\nAllpass: failed to write channel data");
-	}
-	sndclose(ofd);
-	delete comb1;
-	sffinish();
-}

dev/externals/reverb/readme.txt

@@ -1,121 +0,0 @@
-readme file for CDP reverb programs
-
-ROOMRESP
-
-simple utility program to generate early reflections for reverbs, using "mirror room" model.
-
-The format for early reflection data is a series of lines containing
-<time> <value> pairs. The first time is non-zero (since the direct sound
-is presumed to be there):
-
-0.0082 0.500000
-0.0121 0.204198
-0.0134 0.169094
-0.0145 0.143638
-0.0170 0.093391
-0.0180 0.084478
-0.0200 0.067530
-.. etc
-
-Data obtained from roomresp is used internally for the early reflections in both "rmverb" and "reverb". 
-Roomresp usually generates many more taps than are necessary or efficient; typically the first
-19 taps (say) will be used. The two reverb programs offer the facility to import reflection data from
-roomresp directly into the reverb (see the -e flag option).  As a FIR filter is used to process
-these, the CPU load of the program is directly affected by the number of reflections used.
-
-the choice of early reflections has an influence on the character and tonality of the reverb 
-- experimentation is recommended here. Hand-written reflection data is especially good for "creative" reverbs
-with heavy ringing and colour!
-
-RMVERB
-
-The program "rmverb" uses nested allpass filters based on Bill Gardner's published models; hence three 
-room models are offered, "small", "medium" and "large".
- 
-The "large room" model is modified with smaller delays (so really "large-ish"), to reduce the 
-obviousness of the slap-back echoes generated by the model. 
-
-Note that this relies on the fact that each reverb element is implemented in a self-contained manner, 
-unit-generator style (e.g. including plain delays between elements). If it is required to optimise 
-the architecture by having all elements share one delay line,  this change will not work - the outer 
-delay will need to be smaller than the  inner ones.
-
-Two allpass filters per channel are aded to the output, to obtain
-a stereo effect from the internal mono reverb. According to Gardner, his concept was to use one "diffuser" 
-(configuration of nested allpass filters) per speaker; this suggests that any thinness in the output can 
-be overcome (at the cost of processing time) by paralleling several diffusers per channel, with slightly 
-different parameters.
-
-Additionally (NEW from CDP!) each nested allpass filter has been extended with the addition of a first-order
-filter in the outer feedback loop. This acts in addition to the global feedback loop defined by Gardner.
-This is indicated in "rmverb" by the -d flag ("double damping"). This seems to reduce the metallic 
-ringing at the very end of the reverb decay, at the cost of shortening the overall decay time. It would
-easily be possible, with this change, to eliminate the global feedback filter and rely entirely on the
-filter inside the nested allpasses, to recover some of the decay time. This feature is experimental.
-
-As Gardner notes, pushing the feedback amount to obtain long decay times is not recommended, as the timbre
-deteriorates. The program is implemented in a relatively "raw" state; with a raw feedback level parameter 
-rather than a "reverb time"; some mapping will need to be computed for this!
-
-The programs also allow the delay between direct sound and first reflection (as defined in the reflection data)
-to be modified - see the -p flag option.	  Standard filters are provided to filter the input signal itself,
-and the signal entering the reverb. The latter is usual, the former is rarely necessary. 
-Some filters are second-order, offering a 12dB slope (currently fixed in the program). It would be easy to 
-make the slope variable as another user parameter.	 The sources include a library of low and high pass filters,
-first order and second order (shelving).
-
-
- REVERB
-
-The program "reverb" uses a standard moorer/schroeder comb+allpass structure, with	6 comb filters 
-(with lp filters in the fedback loop) feeding 4 allpass filters. A conventional,"reverb time" is provided, but
-no "room size". This wouild have to be obtained via more or less extensive early reflections. 
-
-
-Not shown in the programs usage message is a possble extra argument which gives the name of a text data file
-containing delay times (in msecs) for the comb and allpass filters.	 This will be a plain text file containing
-10 positive floating-point values. The internal preset times corresponds to this data file:
-
-50 56 61 68 72 78 20 14 9 6
-
- - which gives the canonical Moorer values.
-
-
-NOTE one importnat difference in the "absorb" argument to the programs.
-Reverb:  range 0 - 1, large values = more absorption.
-Rmverb:  value in Hzf for feedback filter (See Gardner for details). He suggests:
-	small room: 4200 Hz
-	medium:  2500 Hz
-	large:  2600 Hz
-
-Rmverb offers direct setting of this value. This then applies both the the global feedback loop
-and to the internal nested-allpass loop as described above.
-
-In both reverb programs, ~all~ delay times are adjusted to give prime number integral lengths in samples. This means
-that performance will not be ~identical~ at all sample rates. Naturally, if the sample rate is to be fixed, in a
-dsp implementation, the possibility exists to fine-tune delays for that sample rate.
-
-
-Both programs generate multi-channel (surround) signals by using a separate allpass filter on each channel for
-de-correlation. The direct sound is fed only to the front stereo pair.
-
-
-Note that these are command-line programs, and the various dsp elements are not fully re-entrant in terms of 
-running re-initializations.	 This would not be hard to do, howewver, especially given the goal is a 
-re-implementation on a DSP architecture.
-
-CDP plans to extend these algorithms with variable delay lines, and also to increase density by addition of 
-elements.  To say nothing of real-time versions in plugin form!
-
- Richard Dobson
- CDP
- September 14th 2004
-
-
-
-
-
-
-
-
-

dev/externals/reverb/reflect.cpp

@@ -1,189 +0,0 @@
-/*
- * Copyright (c) 1983-2013 Richard Dobson and Composers Desktop Project Ltd
- * http://people.bath.ac.uk/masrwd
- * http://www.composersdesktop.com
- * This file is part of the CDP System.
- * The CDP System is free software; you can redistribute it
- * and/or modify it under the terms of the GNU Lesser General Public 
- * License as published by the Free Software Foundation; either 
- * version 2.1 of the License, or (at your option) any later version. 
- *
- * The CDP System is distributed in the hope that it will be useful, 
- * but WITHOUT ANY WARRANTY; without even the implied warranty of 
- * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  
- * See the GNU Lesser General Public License for more details.
- * You should have received a copy of the GNU Lesser General Public
- * License along with the CDP System; if not, write to the Free Software
- * Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
- *
- */
- 
-//NB it is most important to avoid periodicities in these
-// using rmresp, best to place source and listener near opposite corners, or both near one corner 
-//preset early reflections - could get these from a brkfile, of course
-//except small room ; seems best to have src and listener same end of room, to get even earlier reflections
-//11 8 3.3
-//0.9
-//5
-//9 3 1.75
-//9 5.7 1.29
-//normed to 0.5
-
-deltap smalltaps[] = {
-	{0.0082,0.500000},
-	{0.0121,0.204198},
-	{0.0134,0.169094},
-	{0.0145,0.143638},
-	{0.0170,0.093391},
-	{0.0180,0.084478},
-	{0.0200,0.067530},
-	{0.0219,0.063096},
-	{0.0226,0.053176},
-	{0.0233,0.044833},
-	{0.0237,0.048586},
-	{0.0243,0.046058},
-	{0.0250,0.043713},
-	{0.0256,0.037388},
-	{0.0261,0.044475},
-	{0.0265,0.034818},
-	{0.0271,0.033360},
-	{0.0276,0.035772},
-	{0.0281,0.034382},
-	{0.0285,0.030051},
-	{0.0287,0.033059},
-	{0.0301,0.027091},
-	{0.0304,0.026513},
-	{0.0306,0.026200},
-	{0.0309,0.023001},
-	{0.0319,0.024114},
-	{0.0326,0.020657},
-	{0.0335,0.021782},
-	{0.0340,0.019019},
-	{0.0356,0.017387},
-	{0.0398,0.017155},
-	{0.0408,0.014690},
-	{0.0412,0.014424},
-	{0.0416,0.014160},
-	{0.0425,0.012175},
-	{0.0429,0.011971},
-	{0.0439,0.011468},
-	{0.0451,0.010854}
-	};
-
-//dim: 21 16 5
-// refl: 0.95
-//nrooms 5
-//listenerpos
-//17 8 2
-//srcpos:
-//6 9.35 2
-//RWD NB does sound worse with with only 32 taps, or refl-order 3
-// this is the 'no-compromise' reverb!
-deltap mediumtaps[] = 
-{
-	{0.0332,0.993303},
-	{0.0353,0.834879},
-	{0.0377,0.729745},
-	{0.0447,0.494138},
-	{0.0548,0.345328},
-	{0.0561,0.313134},
-	{0.0570,0.319440},
-	{0.0577,0.296282},
-	{0.0583,0.290651},
-	{0.0598,0.276075},
-	{0.0615,0.274630},
-	{0.0625,0.240116},
-	{0.0627,0.251368},
-	{0.0641,0.240392},
-	{0.0644,0.226004},
-	{0.0684,0.200374},
-	{0.0690,0.218340},
-	{0.0700,0.201354},
-	{0.0713,0.194250},
-	{0.0718,0.191280},
-	{0.0728,0.176802},
-	{0.0740,0.171020},
-	{0.0752,0.165815},
-	{0.0770,0.166316},
-	{0.0778,0.147079},
-	{0.0780,0.154272},
-	{0.0791,0.149852},
-	{0.0816,0.148066},
-	{0.0825,0.137700},
-	{0.0826,0.130403},
-	{0.0836,0.134167},
-	{0.0862,0.132653} ,
-	{0.0870,0.117789},
-	{0.0871,0.123637},
-	{0.0881,0.120782},
-	{0.0913,0.106847},
-	{0.0929,0.114356},
-	{0.0937,0.106863},
-	{0.0946,0.104723},
-    {0.0975,0.103831}
-	
-};
-
-//normed to 0.5:
-//43 31 19
-//0.95
-//7
-//7.6 15 2
-//30 23.7 6
-
-deltap largetaps[] = 
-{
-	{0.0729,0.500000},
-	{0.0758,0.439456},
-	{0.0975,0.265770},
-	{0.0997,0.241550},
-	{0.1151,0.190794},
-	{0.1162,0.187246},
-	{0.1180,0.172387},
-	{0.1247,0.154486},
-	{0.1320,0.137708},
-	{0.1330,0.135754},
-	{0.1344,0.139864},
-	{0.1346,0.125902},
-	{0.1360,0.129780},
-	{0.1404,0.115604},
-	{0.1449,0.114371},
-	{0.1464,0.112074},
-	{0.1477,0.115801},
-	{0.1492,0.107884},
-	{0.1540,0.096167}
-};
-
-//Moorer
-deltap longtaps[18] = {
-					{0.0043,0.841},
-					{0.0215,0.504},
-					{0.0225,0.491},
-					{0.0268,0.379},
-					{0.0270,0.380},
-					{0.0298,0.346},
-					{0.0458,0.289},
-					{0.0485,0.272},
-					{0.0572,0.192},
-					{0.0587,0.193},
-					{0.0595,0.217},
-					{0.0612,0.181},
-					{0.0707,0.180},
-					{0.0708,0.181},
-					{0.0726,0.176},
-					{0.0741,0.142},
-					{0.0753,0.167},
-					{0.0797,0.134}
-};
-
-
-deltap shorttaps[] = 
-{
-					{0.0099,1.02},			  //my times!
-					{0.0154,0.818},
-					{0.0179,0.635},
-					{0.0214,0.719},
-					{0.0309,0.267},
-					{0.0596,0.242}
-					};
-

dev/externals/reverb/reflect.h

@@ -1,33 +0,0 @@
-/*
- * Copyright (c) 1983-2013 Richard Dobson and Composers Desktop Project Ltd
- * http://people.bath.ac.uk/masrwd
- * http://www.composersdesktop.com
- * This file is part of the CDP System.
- * The CDP System is free software; you can redistribute it
- * and/or modify it under the terms of the GNU Lesser General Public 
- * License as published by the Free Software Foundation; either 
- * version 2.1 of the License, or (at your option) any later version. 
- *
- * The CDP System is distributed in the hope that it will be useful, 
- * but WITHOUT ANY WARRANTY; without even the implied warranty of 
- * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  
- * See the GNU Lesser General Public License for more details.
- * You should have received a copy of the GNU Lesser General Public
- * License along with the CDP System; if not, write to the Free Software
- * Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
- *
- */
- 
-//reflect.h
-#ifndef __REFLECT_INCLUDED
-#define __REFLECT_INCLUDED
-#include "reverberator.h"
-
-
-extern deltap smalltaps[];
-extern deltap mediumtaps[];
-extern deltap largetaps[];
-extern deltap longtaps[];
-extern deltap shorttaps[];
-#endif
-

dev/externals/reverb/reverb.cpp

@@ -1,745 +0,0 @@
-/*
- * Copyright (c) 1983-2013 Richard Dobson and Composers Desktop Project Ltd
- * http://people.bath.ac.uk/masrwd
- * http://www.composersdesktop.com
- * This file is part of the CDP System.
- * The CDP System is free software; you can redistribute it
- * and/or modify it under the terms of the GNU Lesser General Public 
- * License as published by the Free Software Foundation; either 
- * version 2.1 of the License, or (at your option) any later version. 
- *
- * The CDP System is distributed in the hope that it will be useful, 
- * but WITHOUT ANY WARRANTY; without even the implied warranty of 
- * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  
- * See the GNU Lesser General Public License for more details.
- * You should have received a copy of the GNU Lesser General Public
- * License along with the CDP System; if not, write to the Free Software
- * Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
- *
- */
- 
-/* MODULE FOR REVERB.EXE */
-#include <stdio.h>
-#include <stdlib.h>
-#include <math.h>
-#ifdef _DEBUG
-#include <assert.h>
-#endif
-#include <algorithm>
-#include <vector>
-using namespace std;
-/***** FORMULA FOR LOOP GAIN *****
-
-  g = pow(10, - ( ((60*ilooptime)/reverbtime) / 20))
-
-  derived from Moore:
-
-	reverbtime = 60/(-20.log10(g)) * ilooptime
-
-  then we set igain = g(1 - filtgain), officially...
-
-  */
-extern "C" {
-#include <sfsys.h>
-#include <osbind.h>
-#include <cdplib.h>	   //NB requires stdio.h etc - time to change this?
-}
-#include "reverberator.h"
-#include <string.h>
-void usage(void);
-long readtaps(FILE *fp, deltap **taps,double sr);
-
-//my library of early reflections...
-//by doing it this way I can use the sizeof() operator on the array names
-#include "reflect.cpp"
-
-enum cmdargs {
-	PROGNAME,
-	INFILE,
-	OUTFILE,
-	RGAIN,
-	MIX,
-	REVERBTIME,	
-	DAMPING,
-	LPFREQ,
-	TRTIME,
-	CONFIG
-};
-
-const char* cdp_version = "5.0.0";
-
-///////////// Moorer reverb
-int
-main(int argc,char *argv[])
-
-{
-	int		ifd=-1, ofd=-1,i;	
-	//double  delaytime,gain;
-	double	rvbtime, damping, trailertime = 0.0; 	
-	long	chans,outchans = 2;
-	double	sr;
-	//all this stuff from Brandt  Csound implementation of posh Moorer
-	MOORERDATA mrdata = {0.050,0.056,0.061,0.068,0.072,0.078,0.02,0.014,0.009,0.006};	
-	double predelay = 0.0;
-	moorer *moorerverb = 0;
-	tapdelay *tdelay = 0;
-	deltap *ptaps = 0;	
-	allpassfilter **chan_ap = 0;		//ptr to array of allpassfilter ptrs
-	tonecontrol *tc_l_lowcut = 0, *tc_r_lowcut = 0;
-	// could be replaced with a tonecontrol for deeper slope
-	onepole *l_highcut = 0, *r_highcut = 0, *lp = 0;
-	FILE *fp_earlies = 0; 
-	long ntaps;
-	bool want_mono = false;
-	bool want_floats = false;
-	bool usertaps = false;
-	double lowcut_freq = 0.0,highcut_freq = 0.0,lp_freq = 0.0;
-	float dry_gain,wet_gain,diff_gain;
-	CHPEAK *peaks;
-	SFPROPS props;
-	// test vcomb with variable delay taps 
-	//vmtcomb4 vcomb;
-	//fastlfo lfo1,lfo2,lfo3,lfo4;
-	if((argc==2) && strcmp(argv[1],"--version")==0) {
-		fprintf(stdout,"%s\n",cdp_version);
-		fflush(stdout);
-		return 0;
-	}
-
-	if(sflinit("reverb")){
-		sfperror("reverb: initialisation");
-		exit(1);
-	}
-	if(argc < CONFIG) usage();
-
-	while((argc > 1) && argv[1][0]=='-'){
-		char *arg = argv[1];
-		switch(*(++arg)){
-		case('p'):
-			if(*(++arg) =='\0'){
-				fprintf(stderr,"\npredelay requires a parameter");
-				usage();
-			}			
-			predelay = atof(arg);
-			if(predelay <0.0){
-				fprintf(stderr,"\npredelay must be >= 0.0");
-				usage();
-			}
-			predelay *= 0.001;			//arg is in msecs
-			break;
-		case('c'):			
-			if(*(++arg) == '\0') {
-				fprintf(stderr,"\n:reverb: -c flag requires a value");
-				usage();				
-			}
-			outchans = atoi(arg);
-			if(outchans < 1 || outchans > 16){
-				fprintf(stderr,"\nreverb: impossible channel value requested");
-				usage();
-			}
-			if(outchans==1)
-				want_mono = true;
-			break;
-		case('e'):
-			if(*(++arg)=='\0'){
-				fprintf(stderr,"\nreverb: -e flag needs a filename");
-				usage();
-			}
-			if((fp_earlies = fopen(arg,"r")) ==0){
-				fprintf(stderr,"\nreverb: unable to open breakpoint file %s",arg);
-				exit(1);
-			}
-			usertaps = true;
-			break;
-		case('f'):
-			want_floats = true;
-			if(*(++arg) != '\0')
-				fprintf(stderr,"\nreverb: WARNING: -f flag does not take a parameter");
-			break;
-		case('L'):
-			if(*(++arg) == '\0'){
-				fprintf(stderr,"\nreverb: Lowcut flag needs frequency argument");
-				usage();
-			}			
-			lowcut_freq = atof(arg);
-			if(lowcut_freq <= 0.0){
-				fprintf(stderr,"\nreverb: Lowcut freq must be greater than zero");
-				usage();
-			}
-			break;
-		case('H'):
-			if(*(++arg) == '\0'){
-				fprintf(stderr,"\nreverb: Highcut flag needs frequency argument");
-				usage();
-			}			
-			highcut_freq = atof(arg);
-			if(highcut_freq <= 0.0){
-				fprintf(stderr,"\nreverb: Highcut freq must be greater than zero");
-				usage();
-			}
-			break;
-		default:
-			fprintf(stderr,"\nreverb: illegal flag option %s",argv[1]);
-			usage();
-			break;
-		}
-		argc--;
-		argv++;
-	}
-	peaks = (CHPEAK *) malloc(sizeof(CHPEAK) * outchans);
-	if(peaks==NULL){
-		fputs("\nNo memory for PEAK data\n",stderr);
-		exit(1);
-	}
-	chan_ap = new allpassfilter*[outchans];
-	if(chan_ap==0){
-		fputs("\nreverb: no memory for multi-channel diffusion",stderr);
-		exit(1);
-	}
-
-
-	if(argc < CONFIG)
-		usage();
-
-	if(argv[INFILE] != 0) {
-	      	if((ifd = sndopenEx(argv[INFILE],0,CDP_OPEN_RDONLY)) < 0) {
-		   fprintf(stderr,"\nreverb: cannot open input file %s\n", argv[INFILE]);
-		   exit(1);
-		}
-	}
-	if(!snd_headread(ifd,&props)){
-		fprintf(stderr,"\nerror reading infile header: %s\n",rsferrstr);
-		sndcloseEx(ifd);
-		exit(1);
-	}
-	if(props.type != wt_wave){
-		fprintf(stderr,"\ninfile is not a waveform file");
-		sndcloseEx(ifd);
-		exit(1);
-	}
-	
-	sr = (double) props.srate;
-	chans = props.chans;
-
-	if(chans > 2){
-		fprintf(stderr,"\nreverb works only on mono or stereo files");
-		exit(1);
-	}
-	
-	
-	diff_gain = atof(argv[RGAIN]);
-	if(diff_gain < 0.0 || diff_gain > 1.0 ){
-		printf("\nreverb: rgain must be >= 0.0 and  <= 1.0");
-		usage();
-	}
-	dry_gain = (float) atof(argv[MIX]);			  //global output gain from diffuser
-	if(dry_gain < 0.0 || dry_gain >= 1.0 ){
-		printf("\nreverb: mix must be  between 0.0 and 1.0");
-		usage();
-	}		
-	wet_gain = 1.0f - dry_gain;
-	//probably not very scientific, but it works intuitively...
-	//wet_gain *= 2;		  
-	
-
-	rvbtime = atof(argv[REVERBTIME]);	  
-	if(rvbtime <= 0.0){
-	   fprintf(stderr,
-		"\nreverb: rvbtime must be > 0\n");
-		exit(1);
-	}
-
-	damping = atof(argv[DAMPING]);
-	if(damping < 0.0 || damping > 1.0){
-		fprintf(stderr,"\nreverb: absorb must be in the range 0.0 - 1.0\n");
-		exit(1);
-	}
-	double filt_limit = sr / 3.0;
-	lp_freq = atof(argv[LPFREQ]);
-	if(lp_freq < 0.0 || lp_freq > filt_limit){
-		printf("\napass: lp_freq must be within 0.0 to %.4lfHz",filt_limit);
-		usage();
-	}
-
-	trailertime = atof(argv[TRTIME]);
-	if(trailertime < 0.0)
-		trailertime = 0.0;
-	
-	if(argc==CONFIG+1){
-		int got = 0;
-		double dtime1,dtime2,dtime3,dtime4,dtime5,dtime6,
-			adtime1,adtime2,adtime3,adtime4;
-		FILE *fp = fopen(argv[CONFIG],"r");
-		if(!fp)
-			printf("\nreverb: can't open datafile %s: using presets",argv[CONFIG]);
-		else{
-			got = fscanf(fp,"%lf%lf%lf%lf%lf%lf%lf%lf%lf%lf",
-				&dtime1,
-				&dtime2,
-				&dtime3,
-				&dtime4,
-				&dtime5,
-				&dtime6,
-				&adtime1,			 //allpasses
-				&adtime2,
-				&adtime3,
-				&adtime4				
-				);
-			fclose(fp);
-			if(got != 10)
-				printf("\nreverb: error reading looptime values");
-			else {
-				//create prime delay times from the msec vals
-				//get this in a loop later on...				
-				//RWD TODO: check times are not too huge!
-				mrdata.ctime1 = dtime1 * 0.001;
-				mrdata.ctime2 = dtime2 * 0.001;
-				mrdata.ctime3 = dtime3 * 0.001;
-				mrdata.ctime4 = dtime4 * 0.001;
-				mrdata.ctime5 = dtime5 * 0.001;
-				mrdata.ctime6 = dtime6 * 0.001;
-				mrdata.atime1 = adtime1 * 0.001;
-				mrdata.atime2 = adtime2 * 0.001;
-				mrdata.atime3 = adtime3 * 0.001;
-				mrdata.atime4 = adtime4 * 0.001;
-			}
-		}
-	}
-
-	//create input low/high filters, if wanted
-	if(lowcut_freq > 0.0){
-		if(highcut_freq > 0.0 && highcut_freq <= lowcut_freq){
-			fprintf(stderr,"\nreverb: Highcut frequency	 must be higher than Lowcut frequency");
-			usage();
-		}
-		//lowcut is based on low shelving eq filter; here fixed at 12dB
-		tc_l_lowcut = new tonecontrol(lowcut_freq,-12.0,LOW_SHELVE,sr);
-		if(!tc_l_lowcut->create()){
-			fprintf(stderr,"\nreverb: unable to create Lowcut filter");
-			exit(1);
-		}
-		if(chans==2){
-			tc_r_lowcut = new tonecontrol(lowcut_freq,-12.0,LOW_SHELVE,sr);
-			if(!tc_r_lowcut->create()){
-				fprintf(stderr,"\nreverb: unable to create Lowcut filter");
-				exit(1);
-			}
-		}
-	}
-
-	if(highcut_freq > 0.0){		
-		l_highcut = new onepole(highcut_freq,sr,LOW_PASS);
-		if(chans==2)
-			r_highcut = new onepole(highcut_freq,sr,LOW_PASS);
-	}
-
-	if(lp_freq > 0.0)
-		lp		= new onepole(lp_freq,sr,LOW_PASS);
-
-
-	moorerverb =  new moorer(&mrdata,rvbtime,damping,sr);
-	//further external allpasses will be used to create multi-channel diffusion
-	
-	for(i = 0; i < outchans; i++){
-		chan_ap[i] = new allpassfilter(sr,to_prime(0.005 + 0.00025 * (double)i,sr),0.4 ,0);
-		if(chan_ap[i] == 0) {
-			fprintf(stderr,"\nreverb: no memory for output diffusers");
-			exit(1);
-		}
-		if(!chan_ap[i]->create()){
-			fprintf(stderr,"\nreverb: no memory for output diffusers");
-			exit(1);
-		}
-	}
-	if(usertaps){
-#ifdef _DEBUG
-		assert(fp_earlies);
-#endif
-		ntaps = readtaps(fp_earlies,&ptaps,sr);
-		if(ntaps==0){
-			fprintf(stderr,"\nreverb: error reading tapfile");
-			exit(1);
-		}
-		printf("\nloaded %ld early reflections from tapfile",ntaps);
-		fclose(fp_earlies);
-		fp_earlies = 0;
-	}
-	else {
-		if(rvbtime <= 0.6){
-			ntaps = sizeof(smalltaps) / sizeof(deltap);
-			ptaps = smalltaps;
-			printf("\nsmall-room: using %ld early reflections",ntaps);
-		}
-		else if(rvbtime > 0.6 && rvbtime <= 1.3){
-			ntaps = sizeof(mediumtaps) / sizeof(deltap);
-			ptaps = mediumtaps;
-			printf("\nmedium-room: using %ld early reflections",ntaps);
-		}
-		else{
-			ntaps = sizeof(largetaps) / sizeof(deltap);
-			ptaps = largetaps;
-			printf("\nlarge-room: using %ld early reflections",ntaps);
-		}
-	}
-	tdelay = new tapdelay(ptaps,ntaps,sr);
-
-	//if user prescribes a predelay, adjust all taptimes
-	if(predelay > 0.0){
-		double current_delay = ptaps[0].pos;
-		double adjust = predelay - current_delay;
-		for(int i = 0; i < ntaps; i++)
-			ptaps[i].pos += adjust;
-	}
-	
-	if(!moorerverb->create()){
-		fprintf(stderr,"\nreverb: unable to create reverberator");
-		exit(1);
-	}
-	if(!tdelay->create()){
-		fprintf(stderr,"\nreverb: unable to create early reflections");
-		exit(1);
-	}
-	
-	//double maxgain = tdelay->get_maxgain();
-	//if(maxgain > 1.0)
-	//	tdelay->set_gain(maxgain * early_gain);
-	printf("\npredelay = %.4lf secs",ptaps[0].pos);
-	//wet_gain *= floor(maxgain);
-	if(want_floats)
-		props.samptype =FLOAT32;
-	props.chans = outchans;
-
-	if((ofd = sndcreat_ex(argv[OUTFILE],-1,&props,SFILE_CDP,CDP_CREATE_NORMAL)) < 0) {
-		  fprintf(stderr,"\nreverb: cannot open output file %s\n",argv[OUTFILE]);
-		  exit(1);
-	}
-
-
-	printf("\n");
-	long outsamps = 0;
-	long step = (long) (0.25 * sr);
-#ifdef NOTDEF
-	// vcomb test
-	long seed = lfo1.init((double)inprops.srate,0.0,seed);
-	lfo1.set_WaveType(LFO_RANDOM);
-	lfo1.set_freq(0.5);
-	lfo1.set_mod(1.0); /* no change - we control modf range externally */
-	lfo2.init((double)inprops.srate,0.0,seed);
-	lfo2.set_WaveType(LFO_RANDOM);
-	lfo2.set_freq(0.55);
-	lfo2.set_mod(1.0);
-	lfo3.init((double)inprops.srate,0.0,seed);
-	lfo3.set_WaveType(LFO_RANDOM);
-	lfo3.set_freq(0.61);
-	lfo3.set_mod(1.0);
-	lfo4.init((double)inprops.srate,0.0,seed);
-	lfo4.set_WaveType(LFO_RANDOM);
-	lfo4.set_freq(0.66);
-	lfo4.set_mod(1.0);
-	vcomb.init(double)srate,0.8);
-	double delay1,delay2,delay3,delay4;
-	double delaymod = 0.05;
-	delay1 = 0.05;
-	delay2 = 0.056;
-	delay3 = 0,061;
-	delay4 = 0.068;
-	vcomb.init(double)srate,delay4 + (delay4 * delaymod));
-	vcomb.setfgains(
-#endif
-	//stopwatch(1);
-	//run main single-sample loop
-	float l_ip,r_ip,l_out,r_out,l_direct,r_direct,mono_direct;
-	for(;;){
-		int rv;
-		float out,earlies;
-		//read mon/left channnel
-		if((rv = fgetfloatEx(&l_ip,ifd,0)) < 0){
-			fprintf(stderr,"\nreverb: error reading infile"); 
-			exit(1);
-		}
-		if(!rv) break;	   // end of inputfile	- handle any trailertime
-		//apply any conditioning to direct signal
-		if(tc_l_lowcut)
-			l_ip = (float) tc_l_lowcut->tick((double)l_ip);
-		if(l_highcut)
-			l_ip = (float) l_highcut->tick((double)l_ip);
-		l_direct = l_ip;
-		mono_direct = l_direct;
-		r_direct = l_direct;
-		//handle R channnel if active
-		if(chans==2){
-			if((rv = fgetfloatEx(&r_ip,ifd,0)) < 0){
-				fprintf(stderr,"\nError in reading file"); 
-				exit(1);			
-			}		
-			if(!rv) break;					
-			//apply any conditioning to direct signal
-			if(tc_r_lowcut)
-				r_ip = (float) tc_r_lowcut->tick((double)r_ip);
-			if(r_highcut)
-				r_ip = (float) r_highcut->tick((double)r_ip);
-			r_direct = r_ip;
-			if(want_mono)
-				mono_direct = (l_direct + r_direct) * 0.5f;			
-		}
-		//mono_direct = (mixed) input to reverberator
-		//possibly also filter it...
-		if(lp)
-			mono_direct = (float) lp->tick(mono_direct);					//input lowpass
-		earlies = tdelay->tick(mono_direct);				// ---> early reflections 	
-		//send (filtered) mono output from reverberator
-		out = earlies + moorerverb->tick(earlies + mono_direct) * diff_gain;
-		//and send thru multi_channel diffusers
-		//NB must handle first 2 chans as special cases, then loop thru the rest
-		//l_out = want_mono ? mono_direct : l_direct;
-		//l_out *= dry_gain;
-		l_out = chan_ap[0]->tick(out) * wet_gain;
-		if(want_mono)			
-			l_out += (mono_direct * dry_gain);
-		else 
-			l_out += l_direct * dry_gain;
-		
-		
-		if((float)fabs((double)l_out) > peaks[0].value) {
-			peaks[0].value = (float)fabs((double)l_out);
-			peaks[0].position = outsamps;
-		}
-		if(fputfloatEx(&l_out,ofd) < 1){
-			fprintf(stderr,"\nreverb: error writing output file"); 
-			exit(1);
-		}
- 
-		if(outchans>=2){
-			r_out = r_direct * dry_gain;
-			r_out += chan_ap[1]->tick(out) * wet_gain;			
-			if((float)fabs((double)r_out) > peaks[1].value) {
-				peaks[1].value = (float)fabs((double)r_out);
-				peaks[1].position = outsamps;
-			}
-			if(fputfloatEx(&r_out,ofd) < 1){
-				fprintf(stderr,"\nreverb: error writing output file"); 
-				exit(1);
-			}
-		}
-
-		//now any further channels
-		for(i=2;i < outchans; i++){						
-			l_out = chan_ap[i]->tick(out) * wet_gain;		   			
-			if((float)fabs((double)l_out) > peaks[i].value) {
-				peaks[i].value = (float)fabs((double)l_out);
-				peaks[i].position = outsamps;
-			}
-			if(fputfloatEx(&out,ofd) < 1){
-				fprintf(stderr,"\nreverb: error writing output file"); 
-				exit(1);
-			}
-		}
-		outsamps++;
-		if((outsamps % step) == 0)
-			//inform(step,sr);
-            fprintf(stdout,"%.2f\r",(double)outsamps/(double)sr);
-
-	}
-	//end of infile; play out reverb tail as much as we're allowed
-	int trtime  = (int)( trailertime * sr);	
-	for(i = 0; i < 	trtime; i++){
-		float tr_l_ip,tr_r_ip = 0.0f, tr_out,tr_earlies,tr_l_direct,tr_mono_direct,tr_r_direct,
-			tr_l_out,tr_r_out;
-		tr_l_ip = 0.0;
-		if(tc_l_lowcut)
-			tr_l_ip = (float) tc_l_lowcut->tick(tr_l_ip);
-		if(l_highcut)
-			tr_l_ip = (float) l_highcut->tick(tr_l_ip);
-		tr_l_direct = tr_l_ip;
-		tr_mono_direct = tr_l_direct;
-		tr_r_direct = tr_l_direct;
-		//handle R channnel if active
-		if(chans==2){
-			tr_r_ip = 0.0;	   // inout is zero, except if using input filters
-			// get any samples from input conditioning filters
-			if(tc_r_lowcut)
-				tr_r_ip = (float) tc_r_lowcut->tick(tr_r_ip);						   
-			if(r_highcut)
-				tr_r_ip = (float) r_highcut->tick(tr_r_ip);
-			tr_r_direct = tr_r_ip;
-			if(want_mono)
-				tr_mono_direct = (tr_l_direct + tr_r_direct) *0.5f;			
-		}
-		if(lp)
-			tr_mono_direct = (float) lp->tick(tr_mono_direct);
-		tr_earlies = tdelay->tick(tr_mono_direct);
-		//send (filtered) mono output from reverberator
-		tr_out = tr_earlies +  moorerverb->tick(tr_earlies * tr_mono_direct) * diff_gain;
-		tr_l_out = chan_ap[0]->tick(tr_out) * wet_gain;
-		if(want_mono)			
-			tr_l_out += (tr_mono_direct * dry_gain);
-		else 
-			tr_l_out += tr_l_direct * dry_gain;
-		if((float)fabs((double)tr_l_out) > peaks[0].value) {
-			peaks[0].value = (float)fabs((double)tr_l_out);
-			peaks[0].position = outsamps;
-		}
-		if(fputfloatEx(&tr_l_out,ofd) < 1){
-			fprintf(stderr,"\nreverb: error writing to output file"); 
-			exit(1);
-		}
-		if(outchans>=2){
-			tr_r_out = tr_r_direct * dry_gain;
-			tr_r_out += chan_ap[1]->tick(tr_out) * wet_gain;	
-			if((float)fabs((double)tr_r_out) > peaks[1].value) {
-				peaks[1].value = (float)fabs((double)tr_r_out);
-				peaks[1].position = outsamps;
-			}
-			if(fputfloatEx(&tr_r_out,ofd) < 1){
-				fprintf(stderr,"\nreverb: error writing to output file"); 
-				exit(1);
-			}
-		}
-
-		for(int j = 2;j < outchans;j++){
-			float ch_out;
-			ch_out = chan_ap[j]->tick(tr_out)  * wet_gain;
-			if((float)fabs((double)ch_out) > peaks[j].value) {
-				peaks[j].value = (float)fabs((double)ch_out);
-				peaks[j].position = outsamps;
-			}
-			if(fputfloatEx(&ch_out,ofd) < 1){
-				fprintf(stderr,"\nreverb: error writing to output file"); 
-				exit(1);
-			}
-		}
-		outsamps++;
-		if((outsamps % step) ==0)
-			//inform(step,sr);
-            fprintf(stdout,"%.2f\r",(double)outsamps/(double)sr);
-	}
-	//stopwatch(0);
-	printf("\nPEAK data:\n");
-	for(i=0;i < outchans;i++)
-		printf("Channel %d: %.4f at frame %d: %.4lf secs\n",i+1,
-			peaks[i].value,peaks[i].position, (double) peaks[i].position / (double)props.srate);
-	sndputpeaks(ofd,outchans,peaks);
-	sndcloseEx(ifd);
-	
-	if(sndcloseEx(ofd) < 0)
-		fprintf(stderr,"\nwarning: error closing outfile");
-	delete [] peaks;
-	if(moorerverb)
-		delete moorerverb;
-	for(i=0;i < outchans;i++)
-		delete chan_ap[i];
-	delete [] chan_ap;
-	delete tdelay;
-	if(usertaps)
-		delete [] ptaps;
-	return 0;
-}
-
-
-
-void
-usage(void)
-{
-	fprintf(stderr,"\nreverb: Multi-channel reverberator\n\tVersion 1.1 Release 4 CDP 1998,1999\n");
-	fprintf(stderr,
-	"usage: reverb [flags] infile outfile rgain mix rvbtime absorb\n"
-	"                                 lpfreq trailertime [times.txt]\n"
-	"flags    : any or all of the following:\n"
-	"   -cN   : create outfile with N channels (1 <= N <= 16 : default =  2)\n"
-	"   -eFILE: read early reflections from breakpoint textfile FILE\n"
-	"   -f    : force floating-point output (default: infile format)\n"
-	"   -HN   : apply Highcut filter with cutoff freq N Hz to infile (6db/oct)\n"
-	"   -LN   : apply Lowcut filter with cutoff freq N Hz to infile	(12dB/oct)\n"
-	"   -pN   : force reverb predelay to N msecs (shifts early reflections)\n"
-	"rgain    : set level of dense reverb (0.0 <= rgain <= 1.0)\n"    
-	"mix      : dry/wet balance (source and reverb): 1.0<-- mix -->= 0.0\n"
-	"rvbtime  : reverb decay time (to -60dB) in seconds\n"
-	"absorb   : degree of hf damping to suggest air absorption: 0.0 <= absorb <= 1.0\n"
-	"lpfreq   : lowpass filter cutoff freq (Hz) applied at input to reverb\n" 
-	"           to disable either filter (absorb, lpfreq), use the value 0\n" 
-	"trtime   : trailer time added to outfile for reverb tail (secs)\n"
-	"times.txt: list of delay times (msecs) for 6 comb and 4 allpass filters\n"
-	);
-	exit(0);
-}
-
-long readtaps(FILE *fp, deltap **taps,double sr)
-{
-	vector<deltap> vtaps;
-	deltap thistap;
-	char line[256];
-	int ntaps = 0;
-	int linecount = 1;
-	bool error = false;
-	double time= 0.0, current_time = 0.0, val = 0.0;
-
-	double sample_duration = 1.0 / sr;
-	if(fp==0 || sr <= 0.0){
-		*taps = 0;
-		return 0;
-	}
-	while(fgets(line,256,fp))	{
-		int got;
-		if(line[0] == '\n' || line[0]== ';'){	//allow comment lines; this does not check for leading white-space...
-			linecount++;
-			continue;
-		}
-
-		if((got = sscanf(line,"%lf%lf",&time,&val))<2){			
-			fprintf(stderr,"\nerror in reflections file: line %d",linecount);
-			error =true;
-			break;
-		}
-		if(time < 0.0){
-			fprintf(stderr,"\nerror in tapfile: line %d: bad time value",linecount);
-			error = true;
-			break;
-		}
-		//if (first) val = 0.0, or VERY close
-		if(time < sample_duration)	{ //non-zero time must be at least one sample on!			
-			fprintf(stderr,"\nWARNING: very small taptime %.4lf treated as zero",time);
-			time = 0.0;			
-		}
-
-		if(current_time != 0.0 && time==current_time){
-			fprintf(stderr,"\nWARNING: duplicate time in line %d: ignored",linecount);
-			linecount++;
-			continue;
-		}
-		if(time < current_time){
-			fprintf(stderr,"\nerror in tapfile: time out of sequence: line %d",linecount);
-			error = true;
-			break;
-		}
-		current_time = time;
-		thistap.pos = time;
-		if(val <=0.0 || val > 1.0){
-			fprintf(stderr,"\nerror: bad amplitude in tapfile: line %d",linecount);
-			error = true;
-			break;
-		}
-		thistap.val = val;
-		vtaps.push_back(thistap);
-		linecount++;
-	}
-	
-	ntaps = vtaps.size();
-	if(ntaps==0){
-		fprintf(stderr,"\ntapfile contains no data!");
-		error = true;
-	}
-	if(error){
-		*taps = 0;
-		return 0;
-	}
-	*taps = new deltap[ntaps];
-	if(taps==0)
-		return 0;
-	vector<deltap>::iterator I =  vtaps.begin(); 
-	for(int i = 0; I != vtaps.end();i++, I++){
-		(*taps)[i].pos = I->pos;
-		(*taps)[i].val = I->val;
-	}
-
-	return ntaps;
-}

dev/externals/reverb/reverberator.cpp

@@ -1,1488 +0,0 @@
-/*
- * Copyright (c) 1983-2013 Richard Dobson and Composers Desktop Project Ltd
- * http://people.bath.ac.uk/masrwd
- * http://www.composersdesktop.com
- * This file is part of the CDP System.
- * The CDP System is free software; you can redistribute it
- * and/or modify it under the terms of the GNU Lesser General Public 
- * License as published by the Free Software Foundation; either 
- * version 2.1 of the License, or (at your option) any later version. 
- *
- * The CDP System is distributed in the hope that it will be useful, 
- * but WITHOUT ANY WARRANTY; without even the implied warranty of 
- * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  
- * See the GNU Lesser General Public License for more details.
- * You should have received a copy of the GNU Lesser General Public
- * License along with the CDP System; if not, write to the Free Software
- * Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
- *
- */
- 
-// reverberator.cpp: implementation of the reverberator class.
-// see also tapdelay.cpp
-//TODO: simple delay for use with tapped delayline
-// NB blame VC6 for horrible code, new returns null...
-//////////////////////////////////////////////////////////////////////
-#include <stdio.h>
-#include <stdlib.h>
-#include <memory.h>
-#include <math.h>
-#ifdef _DEBUG
-#include <assert.h>
-#endif
-#include "reverberator.h"
-#ifndef PI
-# define PI 3.141592653589793238462643
-#endif
-#ifndef TWOPI
-#define TWOPI (2.0 * PI)
-#endif
-
-#define ROOT2O2 (0.7071067811965475244)
-#define SPN 1.65436e-24			   //RWD: use a proper MSVC version...?
-
-//////////////////////////////////////////////////////////////////////
-// Construction/Destruction
-//////////////////////////////////////////////////////////////////////
-//for example: 
-//const deltap taps[4] = {{0.1,0.9},{0.25,.75},{0.5,.5},{0.618,.25}};
-//NB this accepts a zero taptime
-
-tapdelay::tapdelay(const deltap *taps,unsigned int ctaps,double sr) : dtaps(taps)
-{
-	tapbuf = 0;
-	ntaps = ctaps;
-	buflen = 0;
-	reader = 0;
-	tapptr = 0;
-	tapcoeff = 0;
-	srate = sr;
-	output_gain = 1.0;
-	max_gain = 1.0;
-}
-#define SAFETY (2)
-bool tapdelay::create(void)
-{
-#ifdef _DEBUG
-	assert(tapbuf==0);
-	assert(ntaps > 0);
-#endif
-	if(tapbuf)
-		return false;				//or some informative errorval...
-	if(ntaps <=0)
-		return false;
-	buflen = (unsigned int)((dtaps[ntaps-1].pos * srate) + SAFETY);
-	tapbuf = new double[buflen];
-	if(!tapbuf)
-		return false;
-	memset(tapbuf,0,buflen*sizeof(double));
-	
-	tapptr = new unsigned int[ntaps];
-	if(!tapptr){
-		delete [] tapbuf; tapbuf = 0;
-		return false;
-	}
-	tapcoeff = new double[ntaps];	
-	if(!tapcoeff) {
-		delete [] tapbuf; tapbuf = 0;
-		delete [] tapptr; tapptr = 0;
-		return false;
-	}
-	//copy taps data, and get scalefac from total amps
-	double sum = 0.0;
-	for(unsigned int i = 0;i < ntaps;i++){
-		//RWD.10.98 allow zero taptime, for input mix control...
-		if(dtaps[i].pos == 0.0)
-			tapptr[i] = 0;
-		else
-			tapptr[i] = buflen - (int)(dtaps[i].pos * srate);	
-			
-		tapcoeff[i] = dtaps[i].val;
-		sum += fabs(tapcoeff[i]);
-#ifdef _DEBUG
-		assert(tapptr[i] < buflen);
-#endif
-	}
-	//some idiot might give us empty taps! */
-	if(sum == 0.0){
-		delete [] tapbuf; tapbuf = 0;
-		delete [] tapptr; tapptr = 0;
-		delete [] tapcoeff; tapcoeff = 0;
-		return false;
-	}
-	max_gain = (double)ntaps / sum;
-	one_over_ntaps = 1.0 / (double)ntaps;
-	return true;
-}
-
-double tapdelay::get_maxgain(void)
-{
-	return max_gain;
-}
-
-void tapdelay::set_gain(double gain)
-{
-	output_gain = gain;
-}
-
-double tapdelay::tick(double insam)
-{
-	double val = 0.0;
-	tapbuf[reader++] = insam;
-	if(reader == buflen)
-		reader = 0;
-	for(unsigned int i = 0;i < ntaps;i++) {
-		val += 	((tapbuf[tapptr[i]++]) * (tapcoeff[i]));
-		if(tapptr[i] == buflen)
-			tapptr[i] = 0;
-	}
-	val *= one_over_ntaps;
-	return val * output_gain;
-
-}
-
-tapdelay::~tapdelay()
-{
-	delete [] tapbuf;
-	delete [] tapptr;
-	delete [] tapcoeff;
-}
-
-
-delay::delay(unsigned int len, double gain)
-{
-	delbuf= 0;
-	dgain = gain;
-	ptr = 0;
-	buflen = len;
-}
-
-delay::~delay()
-{
-	delete delbuf;
-}
-
-bool delay::create(void)
-{
-	 if(buflen <= 0)
-		 return false;
-	 if(delbuf)
-		 return false;
-
-	 delbuf = new double[buflen];
-	 if(!delbuf) {		
-		 return false;
-	 }
-	 memset(delbuf,0,buflen*sizeof(double));
-	 return true;
-}
-
-double delay::tick(double insam)
-{
-	double val;
-	val = delbuf[ptr];
-	delbuf[ptr++] = insam;
-	if(ptr==buflen)
-		ptr = 0;
-	return  val*dgain;
-}
-
-
-
-//////////////////////////////////////////////////////////////////////
-// allpassfilter 
-//////////////////////////////////////////////////////////////////////
-//global helper functions: really belong in a new cdplib...except it's nice to have proper bool retval
-//straight from Csound: nreverb (vdelay.c)
-static bool prime( int val )
-{
-    int i, last;
-	
-    last = (int)sqrt( (double)val );
-    for ( i = 3; i <= last; i+=2 ) {
-      if ( (val % i) == 0 ) return false;
-    }
-    return true;
-}
-
-unsigned int to_prime(double dur,double sr)
-{
-		unsigned int time = (unsigned int) (dur * sr);
-		if(time % 2== 0) time += 1;
-		while(!prime(time)) 
-			time +=2;
-		return time;
-}
-#define VMODMAX (0.5)
-//standard allpass, with optional pre_delay
-allpassfilter::allpassfilter(long sr,unsigned int buflen, double ingain,unsigned int predelay = 0)
-{
-    vmod                = 0.1;
-    vfreq               = 0.5; 
-	rvbuf1				= 0;
-	pre_dline			= 0;
-	writer1 = reader1	= 0;
-    sinelfo             = 0;
-    noiselfo            = 0;
-    srate               = sr;
-	rvblen				=  buflen + (unsigned int)(buflen * VMODMAX);
-	gain				= ingain;
-	pre_delay			= predelay;
-    
-}
-
-bool allpassfilter::create(void)
-{
-	if((gain < -1.0) || (gain > 1.0))		//do I want negative gain vals?
-		return false;
-	rvbuf1 = new double[rvblen];	
-	if(!rvbuf1 ){
-#ifdef _DEBUG
-		printf("\nallpass: failed to allocate buffer size %d",rvblen);
-#endif		
-		return false;
-	}
-				
-	if(pre_delay > 0){
-		pre_dline = new delay(pre_delay,1.0);
-		if(!pre_dline->create()){
-#ifdef _DEBUG
-		    printf("\nallpass: can't create pre_dline" );
-#endif		
-		    delete []rvbuf1; rvbuf1 = 0; 
-		    return false;
-	    }
-	}
-	memset(rvbuf1,0,rvblen*sizeof(double));
-    sinelfo = new fastlfo();
-    sinelfo->init((double) srate,0.0,0,1);
-    sinelfo->set_WaveType(LFO_SINE);
-    sinelfo->set_freq(vfreq);
-    sinelfo->set_mod(1.0);
-    noiselfo = new fastlfo();
-    noiselfo->init((double) srate,0.0,0,1);
-    noiselfo->set_WaveType(LFO_RAND_GAUSS);
-    noiselfo->set_freq(vfreq * 2.5);
-    noiselfo->set_mod(1.0);
-
-	return true;
-}
-
-void allpassfilter::set_vmod(double amount)
-{
-    double numod = VMODMAX;
-    if(amount < numod)
-        numod = amount;
-    vmod = numod;
-}
-
-void allpassfilter::set_vfreq(double freq)
-{
-    vfreq = freq;
-    sinelfo->set_freq(vfreq);
-    noiselfo->set_freq(vfreq * 2.5);
-}
-
-/* TODO: optimize this:  separate  funcs for pre-dline and plain */
-double allpassfilter::tick(double insamp)
-{
-	double output, input; 
-	input = insamp;
-	output = (-gain) * input;
-	output += rvbuf1[reader1++];
-	input += gain * output;
-
-	if(pre_dline)
-		rvbuf1[writer1++] = pre_dline->tick(input);
-	else
-		rvbuf1[writer1++] = input;
-	if(reader1 == rvblen)
-		reader1 = 0;
-	if(writer1 == rvblen)
-		writer1 = 0;
-
-	return output;
-}
-/* TODO: implement this! */
-double allpassfilter::vtick(double insamp)
-{
-	double output, input; 
-	input = insamp;
-	output = (-gain) * input;
-	output += rvbuf1[reader1++];
-	input += gain * output;
-
-	if(pre_dline)
-		rvbuf1[writer1++] = pre_dline->tick(input);
-	else
-		rvbuf1[writer1++] = input;
-	if(reader1 == rvblen)
-		reader1 = 0;
-	if(writer1 == rvblen)
-		writer1 = 0;
-
-	return output;
-}
-
-
-
-allpassfilter::~allpassfilter()
-{
-	delete [] rvbuf1;
-	delete pre_dline;
-}
-
-//////////////////////////////////////////////////////////////////////
-//// nested_allpass
-//overall delay = ap1.pre_delay + ap2.pre_deay + ap1.length + ap2.length + post_delay
-nested_allpass::nested_allpass(double srate,double lpfreq, unsigned int outertime, unsigned int time1, unsigned int time2,
-							   double gain, double gain1, double gain2, 
-							   unsigned int delay1 = 0, unsigned int delay2 = 0)
-							   
-{
-	outer_gain		= gain;
-	outer_time		= outertime;
-	ap1_gain		= gain1;
-	ap2_gain		= gain2;	
-	ap1_length		= time1;
-	ap2_length		= time2;
-	ap1_delay		= delay1;
-	ap2_delay		= delay2;
-	buf				= 0;
-	if(ap2_gain == 0.0)
-		ap2_length = 0;
-	ap1				= ap2	= 0;		
-	writer = reader = 0;	
-	lp = 0;
-	lpfreq_ = lpfreq;
-	sr_ = srate;
-
-}
-
-nested_allpass::~nested_allpass()
-{	
-	delete [] buf;
-	delete ap1;
-	delete ap2;
-	//RWD
-	delete lp;
-}
-
-bool nested_allpass::create(void)
-{
-	if(outer_gain < -1.0 || outer_gain > 1.0)
-		return false;	
-	if(outer_time <= 0)
-		return false;
-	buf = new double[outer_time];
-	if(!buf)
-		return false;
-	memset(buf,0,outer_time * sizeof(double));
-	if(ap1_length > 0 && ap1_gain > 0.0){
-		ap1 = new allpassfilter((long)sr_,ap1_length,ap1_gain,ap1_delay);
-		if(!ap1->create()){
-			delete [] buf; buf = 0;
-			return false;
-		}
-	}
-	
-	if(ap2_length >0 && ap2_gain > 0.0){
-		ap2 = new allpassfilter((long)sr_,ap2_length,ap2_gain,ap2_delay);
-		if(!ap2->create()){
-			delete ap1;	 ap1 = 0;
-			delete [] buf;  buf = 0;
-			return false;
-		}
-	}
-	// RWD experiment: internal absorption damping
-	lp = new onepole(lpfreq_,sr_,LOW_PASS);
-	if(!lp){
-		delete ap1;	 ap1 = 0;
-		delete ap2;	 ap2 = 0;
-		delete [] buf;  buf = 0;
-		return false;
-	}
-	damping_ = false;
-	return true;
-}
-
-//messy casts - sort this out later...
-double nested_allpass::tick(double insam)
-{
-
-	double output, input,nest_out; 
-	input = insam;
-	output = (-outer_gain) * input;
-	output += buf[reader++];
-	input += outer_gain * output;
-	//input is to two allpasses, and the post_delay, which acts as the overall allpass
-	if(!ap2) {	//no second allpass, may be one or no first allpass
-		if(ap1)			
-			nest_out =  ap1->tick(input);
-		else
-			//behave as normal allpass
-			nest_out =  input;
-	}
-	else		
-		nest_out = 	 ap2->tick(ap1->tick(input));
-
-	//RWD experiment!
-	if(damping_)
-	   buf[writer++] = lp->tick(nest_out);
-	else
-		buf[writer++] = nest_out;
-	//wrap indices
-	if(writer == outer_time)
-		writer = 0;
-	if(reader == outer_time)
-		reader = 0;
-			
-	return output;
-}
-
-
-//////////////////////////////////////////////////////////////////////
-// lpcomb 
-//////////////////////////////////////////////////////////////////////
-
-lpcomb::lpcomb(unsigned int buflen, double fbgain, double lpcoeff, double pre = 1.0)
-{
-	combbuf = 0;
-	lp = 0;
-	gain = fbgain;
-	lpgain = lpcoeff;
-	rvblen = buflen;
-	prescale  = pre;
-	reader1 = writer1 = 0;
-}
-
-bool lpcomb::create(void)
-{
-	if(prescale <= 0.0) 
-		return false;
-	if((gain < -1.0 ) || (gain > 1.0))		//do I want negative gain vals?
-		return false;
-
-	combbuf = new double[rvblen];
-	if(!combbuf)
-		return false;
-	lp = new lowpass1(lpgain);
-	if(!lp){
-		delete [] combbuf; combbuf = 0;
-		return false;
-	}
-	memset(combbuf,0,rvblen*sizeof(double));
-	return true;
-}
-
- //wet output only, unscaled
-double lpcomb::tick(double insam)
-{
-	double input,output,lpsig;
-	input = insam * prescale;
-	
-	output = combbuf[reader1++];		 //do gain scaling outside
-	lpsig = lp->tick(output);
-
-	combbuf[writer1++] = input + (gain * lpsig) ;	//feedback scaled output
-	
-
-	if(writer1 == rvblen)
-		writer1 = 0;
-	if(reader1 == rvblen)
-		reader1 = 0;
-	return  output;
-}
-
-lpcomb::~lpcomb()
-{
-	delete [] combbuf;
-	delete lp;
-}
- 
- //////////////////////////////////////////////////////////////////////
-// lowpass 	: perhaps a bit ott as a class, but it allows fairly independent fiddling
-//////////////////////////////////////////////////////////////////////
-lowpass1::lowpass1(double filtgain)
-{
-	temp = output = 0.0;
-	gain = filtgain;
-}
-
-lowpass1::~lowpass1()
-{
-}
- //IIR lowpass
-inline double lowpass1::tick(double insam)
-{	 
-	output  = temp;
-	temp = insam + (output * gain);
-	return output;
-
-}
-// NB allows zero frequency: straight pass-through
-onepole::onepole(double freq,double srate, int low_high)
-{
-	if(freq == 0.0){
-		a1 = 1.0;
-		a2 = 0.0;
-	}
-	else {
-		b = 2.0 - cos(TWOPI * (freq/srate));
-		a2 = sqrt((b*b) - 1.0) - b;
-		a1 = 1.0 + a2;
-		if(low_high==HIGH_PASS)
-			a2 = -a2;
-	}
-	lastout = 0.0;
-}
-
-onepole::~onepole()
-{
-	 //nothing to do
-}
-
-double onepole::tick(double input)
-{
-	double output;
-	output  =  a1*input - a2*lastout;
-	lastout =  output;
-	return output;
-}
-
-//uses code from CDP eq.c
-
-tonecontrol::tonecontrol(double frq,double dbfac,int type,double srate)
-{
-	freq = frq;
-	boost = dbfac;
-	sr = srate;
-	tc_type = type;
-	x1 = x2 = y1 = y2 = 0.0;
-	a0 = a1 = a2 = b1 = b2 = 0.0;
-}
-
-tonecontrol::~tonecontrol()
-{
-   //nothing to do;
-}
-
-bool tonecontrol::create(void)
-{
-	if(sr <= 0.0)
-		return false;
-	if(freq <= 0.0)
-		return false;
-	//normalize freq
-	freq /= sr;
-	switch(tc_type){
-	case(LOW_SHELVE):
-		lshelve();
-		break;
-	case(HIGH_SHELVE):
-		hshelve();
-		break;
-	default:
-		return false;
-		break;
-	}
-	return true;
-}
-
-double tonecontrol::tick(double input)
-{
-	double out;
-
-	if(boost== 0.0)
-		return input;
-
-	out = ((a0 * input) + (a1 * x1) + (a2 * x2) - (b1 * y1) - (b2 * y2));
-	x2 = x1;
-	x1 = input;
-	y2 = y1;
-	y1 = out;
-	return out;
-}
-
-void tonecontrol::lshelve(void)
-{
-	double a, A, F, tmp, b0, recipb0, asq, F2, gamma2, siggam2, gam2p1;
-	double gamman, gammad, ta0, ta1, ta2, tb0, tb1, tb2, aa1, ab1;
-
-	a = tan(PI * (freq - 0.25));	/* Warp factor */ 
-	asq = a * a;
-	A = pow(10.0, boost/20.0);	/* Cvt dB to factor */
-	if((boost < 6.0) && (boost > -6.0)) F = sqrt(A);
-	else if (A > 1.0) F = A/sqrt(2.0);
-	else F = A * sqrt(2.0);
-	  /* If |boost/cut| < 6dB, then doesn't make sense to use 3dB point.
-	     Use of root makes bandedge at half the boost/cut amount
-	  */
-
-	F2 = F * F;
-	tmp = A * A - F2;
-	if(fabs(tmp) <= SPN) gammad = 1;
-	else gammad = pow( (F2 - 1)/ tmp, 0.25);	/* Fourth root */
-	gamman = sqrt(A) * gammad;
-
-   /* Once for the numerator */
-
-	gamma2 = gamman * gamman;
-	gam2p1 = 1 + gamma2;
-	siggam2 = 2 * ROOT2O2 * gamman;
-	
-	ta0 = gam2p1 + siggam2;
-	ta1 = -2 * (1 - gamma2);
-	ta2 = gam2p1 - siggam2;
-
-   /* And again for the denominator */
-
-	gamma2 = gammad * gammad;
-	gam2p1 = 1 + gamma2;
-	siggam2 = 2 * ROOT2O2 * gammad;
-	
-	tb0 = gam2p1 + siggam2;
-	tb1 = -2 * (1 - gamma2);
-	tb2 = gam2p1 - siggam2;
-
-   /* Now bilinear transform to proper centre frequency */
-
-	aa1 = a * ta1;
-	a0 = ta0 + aa1 + asq * ta2;
-	a1 = 2 * a * (ta0 + ta2) + (1 + asq) * ta1;
-	a2 = asq * ta0 + aa1 + ta2;
-
-	ab1 = a * tb1;
-	b0 = tb0 + ab1 + asq * tb2;
-	b1 = 2 * a * (tb0 + tb2) + (1 + asq) * tb1;
-	b2 = asq * tb0 + ab1 + tb2;
-	
-   /* Normalise b0 to 1.0 for realisability */
-
-	recipb0 = 1 / b0;
-	a0 *= recipb0;
-	a1 *= recipb0;
-	a2 *= recipb0;
-	b1 *= recipb0;
-	b2 *= recipb0;
-
-}
-
-void tonecontrol::hshelve(void)
-{
-	double a, A, F, tmp, b0, recipb0, asq, F2, gamma2, siggam2, gam2p1;
-	double gamman, gammad, ta0, ta1, ta2, tb0, tb1, tb2, aa1, ab1;
-
-	a = tan(PI * (freq - 0.25));	/* Warp factor */ 
-	asq = a * a;
-	A = pow(10.0, boost/20.0);	/* Cvt dB to factor */
-	if(boost < 6.0 && boost > -6.0) F = sqrt(A);
-	else if (A > 1.0) F = A/sqrt(2.0);
-	else F = A * sqrt(2.0);
-	  /* If |boost/cut| < 6dB, then doesn't make sense to use 3dB point.
-	     Use of root makes bandedge at half the boost/cut amount
-	  */
-
-	F2 = F * F;
-	tmp = A * A - F2;
-	if(fabs(tmp) <= SPN) gammad = 1;
-	else gammad = pow( (F2 - 1)/ tmp, 0.25);	/* Fourth root */
-	gamman = sqrt(A) * gammad;
-
-   /* Once for the numerator */
-
-	gamma2 = gamman * gamman;
-	gam2p1 = 1 + gamma2;
-	siggam2 = 2 * ROOT2O2 * gamman;
-	
-	ta0 = gam2p1 + siggam2;
-	ta1 = -2 * (1 - gamma2);
-	ta2 = gam2p1 - siggam2;
-
-	ta1 = -ta1;
-	
-   /* And again for the denominator */
-
-	gamma2 = gammad * gammad;
-	gam2p1 = 1 + gamma2;
-	siggam2 = 2 * ROOT2O2 * gammad;
-	
-	tb0 = gam2p1 + siggam2;
-	tb1 = -2 * (1 - gamma2);
-	tb2 = gam2p1 - siggam2;
-
-	tb1 = -tb1;
-
-   /* Now bilinear transform to proper centre frequency */
-
-	aa1 = a * ta1;
-	a0 = ta0 + aa1 + asq * ta2;
-	a1 = 2 * a * (ta0 + ta2) + (1 + asq) * ta1;
-	a2 = asq * ta0 + aa1 + ta2;
-
-	ab1 = a * tb1;
-	b0 = tb0 + ab1 + asq * tb2;
-	b1 = 2 * a * (tb0 + tb2) + (1 + asq) * tb1;
-	b2 = asq * tb0 + ab1 + tb2;
-	
-   /* Normalise b0 to 1.0 for realisability */
-
-	recipb0 = 1 / b0;
-	a0 *= recipb0;
-	a1 *= recipb0;
-	a2 *= recipb0;
-	b1 *= recipb0;
-	b2 *= recipb0;
-
-}
-
-
-/***************** GARDNER DIFFUSERS ***************/
-
-// NOTE: the published Gardner designs are here modified by adding a onepole lp filter
-// in the internal outer feedback loop of each nested allpass filter. 
-// When combined with the overall feedback filter, this will of course reduce the reverb decay time.
-// The option exists to bypass this overall feedback filter, and rely solely on the internal ones.
-
-
-// NOTE: to get an infinite reverb, all feedback filters have to be bypassed by setting lp_freq to 0.0
-// otherwise, the output will still decay eventually !
-
-small_diffuser::small_diffuser(unsigned int pre_delay, 
-							   const NESTDATA *apdata1,
-							   const NESTDATA *apdata2,double fb_gain,double lp_freq,double srate){
-
-	  ap1_data = *apdata1;
-	  ap2_data = *apdata2;
-	  predelay_time = pre_delay;
-	  lpgain = fb_gain;
-	  lpfreq = lp_freq;
-	  ap1 = ap2 = 0;
-	  lp1 = 0;
-	  predelay = 0;
-	  out1 = out2 = 0.0;
-	  sr = srate;
-	  damping_ = false;
-}
-
-small_diffuser::~small_diffuser()
-{
-	delete ap1;
-	delete ap2;
-	delete predelay;	
-	delete lp1;
-}
-
-bool small_diffuser::create(void)
-{
-	if(ap1_data.time1 == 0 || ap1_data.gain1 <= 0.0){
-#ifdef _DEBUG
-		printf("\ndiffuser: bad parameters(1)");
-#endif		
-		return false;
-	}
-	if(ap1_data.time2 ==0 || ap1_data.gain2 <= 0.0){
-#ifdef _DEBUG
-		printf("\ndiffuser: bad parameters(2)");
-#endif		
-		return false;
-	}
-	if(ap2_data.time1 == 0 || ap2_data.gain1 < 0.0){
-#ifdef _DEBUG
-		printf("\ndiffuser: bad parameters(3)");
-#endif		
-		return false;
-	}
-	if(ap2_data.time2 ==0 || ap2_data.gain2 < 0.0){
-#ifdef _DEBUG
-		printf("\ndiffuser: bad parameters(4)");
-#endif		
-		return false;
-	}
-
-	if(sr <=0.0){
-#ifdef _DEBUG
-		printf("\ndiffuser: bad srate parameter)");
-#endif		
-		return false;
-	}
-	if(lpfreq <0.0){
-#ifdef _DEBUG
-		printf("\ndiffuser: bad freq parameter)");
-#endif		
-		return false;
-	}
-
-	ap1 = new nested_allpass(sr,lpfreq,ap1_data.time1,ap1_data.time2,ap1_data.time3,
-							ap1_data.gain1,ap1_data.gain2,ap1_data.gain3);
-	if(!ap1->create()){
-#ifdef _DEBUG
-		printf("\ndiffuser: can't create first allpass");
-#endif		
-		return false;
-	}
-	if(ap2_data.gain1 != 0.0){	 //allow ap to eliminate second block
-		ap2 = new nested_allpass(sr,lpfreq,ap2_data.time1,ap2_data.time2,ap2_data.time3,
-							ap2_data.gain1,ap2_data.gain2,ap2_data.gain3);
-		if(!ap2->create()){
-#ifdef _DEBUG
-			printf("\ndiffuser: can't create second allpass");
-#endif		
-			delete ap1; ap1 = 0;
-			return false;
-		}
-	}
-	if(predelay_time > 0){
-		predelay = new delay(predelay_time,1.0);
-		if(!predelay->create()){
-#ifdef _DEBUG
-			printf("\ndiffuser: can't create predelay");
-#endif		
-			delete ap1;	ap1 = 0;
-			delete ap2;	ap2 = 0;
-			return false;
-		}
-	}
-	// NB if lpfreq == 0, onepole is no-op - returns input
-	lp1 = new onepole(lpfreq,sr,LOW_PASS);
-	if(lp1 == 0){
-		delete [] predelay; predelay = 0;
-		delete ap1;	ap1 = 0;
-		delete ap2;	ap2 = 0;
-		return false;
-	}
-	ap1->set_damping(damping_);
-	ap2->set_damping(damping_);
-	return true;
-}
-
-
-double small_diffuser::tick(double insam) 
-{
-	double filter_out;	
-	double lp_in;	
-	double output;
-	double ip;
-	// may only contain one nested allpass filter:
-	// input to filter is either out1 or out2
-	if(ap2)
-		lp_in = out2;	   //= both in series
-	else
-		lp_in = out1;
-	// RWD NOTE: if the damping_ mechanism is preferred, this filter can be omitted and ...
-	filter_out =  lp1->tick(lp_in);
-	// ... just use this				 
-	//	filter_out = lp_in ;
-	
-	ip = insam + lpgain * filter_out;	
-	if(predelay)
-		ip = predelay->tick(ip);
-	out1 = ap1->tick( ip);
-	if(ap2){
-		out2 = ap2->tick(out1);
-		output =  (out1 + out2)  * 0.5;
-	}
-	else
-		output =  out1;
-	return  output;
-}
-
-//post-delay almost certainly doe not need to be prime...
-medium_diffuser::medium_diffuser(double post_delay,const NESTDATA *apdata1,
-					const NESTDATA *apdata2,
-					double gain,
-					double lp_freq,
-					double srate)
-{
-	ap1_data = *apdata1;
-	ap2_data = *apdata2;
-	postdelay_time = to_prime(post_delay,srate);
-	md_gain = gain;
-	lpfreq = lp_freq;
-	ap1 = ap2 = 0;
-	ap3 = 0;
-	lp1 = 0;
-	delay1 = delay2 = postdelay = 0;
-	sr = srate;
-	out1 = out2 =out3 = 0.0;
-	damping_  = false;
-}
-
-medium_diffuser::~medium_diffuser()
-{	
-		delete ap1;	
-		delete ap2;	
-		delete ap3;	
-		delete lp1;	
-		delete delay1;	
-		delete delay2;
-}
-
-
-bool medium_diffuser::create(void)
-{
-	if(ap1_data.time1 == 0 || ap1_data.gain1 <= 0.0){
-#ifdef _DEBUG
-		printf("\nmedium_diffuser: bad parameters(1)");
-#endif		
-		return false;
-	}
-	if(ap1_data.time2 ==0 || ap1_data.gain2 <= 0.0){
-#ifdef _DEBUG
-		printf("\nmedium diffuser: bad parameters(2)");
-#endif		
-		return false;
-	}
-	if(ap2_data.time1 == 0 || ap2_data.gain1 < 0.0){
-#ifdef _DEBUG
-		printf("\nmedium diffuser: bad parameters(3)");
-#endif		
-		return false;
-	}
-	if(ap2_data.time2 ==0 || ap2_data.gain2 < 0.0){
-#ifdef _DEBUG
-		printf("\nmedium diffuser: bad parameters(4)");
-#endif		
-		return false;
-	}
-	if(sr <=0.0){
-#ifdef _DEBUG
-		printf("\nmedium diffuser: bad srate parameter)");
-#endif		
-		return false;
-	}
-	if(lpfreq <0.0){
-#ifdef _DEBUG
-		printf("\nmedium diffuser: bad freq parameter)");
-#endif		
-		return false;
-	}
-
-	ap1 = new nested_allpass(sr,lpfreq,ap1_data.time1,ap1_data.time2,ap1_data.time3,
-							ap1_data.gain1,ap1_data.gain2,ap1_data.gain3);
-	if(!ap1->create()){
-#ifdef _DEBUG
-		printf("\nmedium diffuser: can't create first diffuser");
-#endif		
-		return false;
-	}
-	ap2 = new nested_allpass(sr,lpfreq,ap2_data.time1,ap2_data.time2,ap2_data.time3,
-							ap2_data.gain1,ap2_data.gain2,ap2_data.gain3);
-	if(!ap2->create()){
-#ifdef _DEBUG
-		printf("\nmedium diffuser: can't create second diffuser");
-#endif	
-		delete ap1; ap1 = 0;
-		return false;
-	}
-
-	ap3 = new allpassfilter((long)sr,to_prime(0.03,sr),0.5,to_prime(0.005,sr));
-	if(!ap3->create()){
-#ifdef _DEBUG
-		printf("\nmedium diffuser: can't create internal allpass filter");
-#endif
-		delete ap1; ap1 = 0;
-		delete ap2; ap2 = 0;
-		return false;
-	}
-
-
-	if(postdelay_time > 0){
-		postdelay = new delay(postdelay_time,1.0);
-		if(!postdelay->create()){
-#ifdef _DEBUG
-			printf("\nmedium diffuser: can't create postdelay");
-#endif		
-
-			delete ap1; ap1 = 0;
-			delete ap2; ap2 = 0;
-			delete ap3; ap3 = 0;
-			return false;
-		}
-	}
-
-	//internal fixed delays
-	delay1 = new delay(to_prime(/*0.007*/0.067,sr),1.0);
-	if(!delay1->create()){
-#ifdef _DEBUG
-		printf("\nmedium diffuser: can't create internal delay1");
-#endif
-		delete ap1; ap1 = 0;
-		delete ap2; ap2 = 0;
-		delete ap3; ap3 = 0;
-		delete [] postdelay; postdelay = 0;
-		return false;
-	}
-		delay2 = new delay(to_prime(0.015,sr),1.0);
-	if(!delay2->create()){
-#ifdef _DEBUG
-		printf("\nmedium diffuser: can't create internal delay2");
-#endif
-		delete ap1; ap1 = 0;
-		delete ap2; ap2 = 0;
-		delete ap3; ap3 = 0;
-		delete [] postdelay; postdelay = 0;
-		delete delay1; delay1 = 0;
-		return false;
-	}
-	
-	lp1 = new onepole(lpfreq,sr,LOW_PASS);
-	if(lp1 == 0){
-		delete ap1; ap1 = 0;
-		delete ap2; ap2 = 0;
-		delete ap3; ap3 = 0;
-		delete [] postdelay; postdelay = 0;
-		delete delay1; delay1 = 0;
-		delete delay2; delay2 = 0;
-		return false;
-	}
-	if(lpfreq == 0.0)
-		damping_ = false;
-	ap1->set_damping(damping_);
-	ap2->set_damping(damping_);
-	return true;
-}
-
-double medium_diffuser::tick(double insam)
-{
-	double filter_out;
-	double output;
-
-	//feedback from postdelay, thru lp filter
-	// RWD NOTE: if the damping_ mechanism is preferred, this filter can be omitted and...
-	filter_out = md_gain * lp1->tick(postdelay->tick(out3));
-	// ... just use this
-	//filter_out = (float)(md_gain * postdelay->tick(out3));
-	
-	//first nested-allpass, takes input plus feedback
-	out1 = ap1->tick(insam + filter_out);
-	//inner allpass with predelay, followed by plain delay
-	out2 = delay1->tick(ap3->tick(out1));
-	//second nested-allpass, takes direct input
-	out3 = ap2->tick(insam + md_gain * delay2->tick(out2));
-	output = 0.5 * (out1 + out2 + out3);
-	return output;
-}
-
-
-large_diffuser::large_diffuser(const NESTDATA *apdata1,
-					const NESTDATA *apdata2,
-					double gain,
-					double lp_freq,
-					double srate)
-{
-	ap1_data = *apdata1;
-	ap2_data = *apdata2;
-	ld_gain = gain;
-	lpfreq = lp_freq;
-	ap1 = ap2 = 0;
-	ap3 = ap4 = 0;
-	lp1 = 0;
-	delay1 = delay2 = delay3 = delay4 = 0;
-	sr = srate;
-	out1 = out2 =out3 = 0.0;
-	damping_ = false;
-}
-
-large_diffuser::~large_diffuser()
-{	
-	delete ap1;	
-	delete ap2;
-	delete ap3;
-	delete ap4;
- 	delete lp1;
-	delete delay1;
-	delete delay2;
-	delete delay3;
-	delete delay4;
-}
-
-bool large_diffuser::create(void)
-{
-	if(ap1_data.time1 == 0 || ap1_data.gain1 <= 0.0){
-#ifdef _DEBUG
-		printf("\nmedium_diffuser: bad parameters(1)");
-#endif		
-		return false;
-	}
-	if(ap1_data.time2 ==0 || ap1_data.gain2 <= 0.0){
-#ifdef _DEBUG
-		printf("\nmedium diffuser: bad parameters(2)");
-#endif		
-		return false;
-	}
-	if(ap2_data.time1 == 0 || ap2_data.gain1 < 0.0){
-#ifdef _DEBUG
-		printf("\nmedium diffuser: bad parameters(3)");
-#endif		
-		return false;
-	}
-	if(ap2_data.time2 ==0 || ap2_data.gain2 < 0.0){
-#ifdef _DEBUG
-		printf("\nmedium diffuser: bad parameters(4)");
-#endif		
-		return false;
-	}
-	if(sr <=0.0){
-#ifdef _DEBUG
-		printf("\nmedium diffuser: bad srate parameter)");
-#endif		
-		return false;
-	}
-	if(lpfreq <0.0){
-#ifdef _DEBUG
-		printf("\nmedium diffuser: bad freq parameter)");
-#endif		
-		return false;
-	}
-	ap1 = new nested_allpass(sr,lpfreq,ap1_data.time1,ap1_data.time2,ap1_data.time3,
-							ap1_data.gain1,ap1_data.gain2,ap1_data.gain3);
-	if(!ap1->create()){
-#ifdef _DEBUG
-		printf("\nlarge diffuser: can't create first nested_allpass");
-#endif		
-		return false;
-	}
-	ap2 = new nested_allpass(sr,lpfreq,ap2_data.time1,ap2_data.time2,ap2_data.time3,
-							ap2_data.gain1,ap2_data.gain2,ap2_data.gain3);
-	if(!ap2->create()){
-#ifdef _DEBUG
-		printf("\nlarge diffuser: can't create second netsed_allpass");
-#endif	
-		cleanup();
-		return false;
-	}
-
-	ap3 = new allpassfilter((long) sr,to_prime(0.008,sr),0.3);
-	if(!ap3->create()){
-#ifdef _DEBUG
-		printf("\nlarge diffuser: can't create first internal allpass filter");
-#endif
-		cleanup();
-		return false;
-	}
-
-	ap4 = new allpassfilter((long)sr,to_prime(0.012,sr),0.3);
-	if(!ap4->create()){
-#ifdef _DEBUG
-		printf("\nlarge diffuser: can't create second internal allpass filter");
-#endif
-		cleanup();
-		return false;
-	}
-	//internal fixed delays
-	delay1 = new delay((unsigned int)(0.004 * sr),1.0);
-	if(!delay1->create()){
-#ifdef _DEBUG
-		printf("\nlarge diffuser: can't create internal delay1");
-#endif
-		cleanup();
-		return false;
-	}
-		delay2 = new delay((unsigned int)(0.017 * sr),1.0);
-	if(!delay2->create()){
-#ifdef _DEBUG
-		printf("\nlarge diffuser: can't create internal delay2");
-#endif
-		cleanup();
-		return false;
-	}
-	delay3 = new delay((unsigned int)(0.031 * sr),1.0);
-	if(!delay3->create()){
-#ifdef _DEBUG
-		printf("\nlarge diffuser: can't create internal delay3");
-#endif
-		cleanup();
-		return false;
-	}
-	delay4 = new delay((unsigned int)(0.003 * sr),1.0);
-	if(!delay4->create()){
-#ifdef _DEBUG
-		printf("\nlarge diffuser: can't create internal delay4");
-#endif
-		cleanup();
-		return false;
-	}
-	
-	lp1 = new onepole(lpfreq,sr,LOW_PASS);
-	if(lp1 == 0){
-		cleanup();
-		return false;
-	}
-	if(lpfreq == 0.0)
-		damping_ = false;
-	
-	ap1->set_damping(damping_);
-	ap2->set_damping(damping_);
-	return true;
-}
-void  large_diffuser::cleanup()
-{
-	delete lp1; lp1 = 0;
-	delete delay4; delay4 = 0;
-	delete delay3; delay3 = 0;
-	delete delay2; delay2 = 0;
-	delete delay1; delay1 = 0;
-	delete ap4; ap4 = 0;
-	delete ap3; ap3 = 0;
-	delete ap2; ap2 = 0;
-	delete ap1; ap1 = 0;
-}
-
-// NOTE: the three output taps from the diffuser can lead to discrete slap-back echoes in the output.
-// possibly authentic, but may not suit all purposes. See rmverb.cpp: some delay lengths reduced from Gardner.
-// This is possible because delays are discrete: they do not share single delay line buffer.
-// true also of medium diffuser, but not so apparent as delays are smaller
-double large_diffuser::tick(double insam)
-{
-	double filter_out;
-	double output;
-	//feedback from lpfilter
-	// RWD NOTE: if the damping_ mechanism is preferred, this filter can be omitted and ...
-	filter_out = ld_gain * lp1->tick(out3);		
-	// ... just use this
-	//filter_out = (float)(ld_gain * out3);			  
-
-	//first tap, from two plain allpasses, and trailing delay
-	out1 = delay1->tick(ap4->tick(ap3->tick(insam + filter_out)));
-	//second tap, from first nested allpass and delays at entrance and exit
-	out2 = delay3->tick(ap1->tick(delay2->tick(out1)));
-	//third tap, from second nested-allpass with leading delay
-	out3 = ap2->tick(delay4->tick(out2));
-	//out3 also goes into lpfilter... see above
-	//prescribed scale factors from Gardner
-	output = (out1 * 0.34) + (out2 * 0.14) + (out3 * 0.14);
-	// this seems to give relatively soft reverb tail:  could try this:
-	//output = (float)((out1 * 0.7) + (out2 * 0.25) + (out3 * 0.25));
-	return  output;
-}
-
-
-moorer::moorer(const MOORERDATA *pdata,double reverbtime,double damping,double srate)
-{
-	mdata		= *pdata;
-	reverb_time = reverbtime;
-	dampfac		= damping;
-	sr			= srate;
-	comb1 = comb2 = comb3 = comb4 = comb5 = comb6 = 0;
-	ap1 = ap2 = ap3 =  ap4  = 0;
-	out = 0.0;
-}
-
-moorer::~moorer()
-{
-	delete comb1;
-	delete comb2;
-	delete comb3;
-	delete comb4;
-	delete comb5;
-	delete comb6;
-	delete ap1;
-	delete ap2;		
-	delete ap3;
-	delete ap4;
-}
-
-bool moorer::create(void)
-{
-	fgain1 = 0.42 * dampfac;
-	fgain2 = 0.46 * dampfac;
-	fgain3 = 0.48 * dampfac;
-	fgain4 = 0.49 * dampfac;
-	fgain5 = 0.50 * dampfac;
-	fgain6 = 0.52 * dampfac;
-	scalefac = (1.0 / 6.0);	
-	cgain1 = exp(log(0.001)*(mdata.ctime1/reverb_time)) * (1. - fgain1);
-	cgain2 = exp(log(0.001)*(mdata.ctime2/reverb_time)) * (1. - fgain2);
-	cgain3 = exp(log(0.001)*(mdata.ctime3/reverb_time)) * (1. - fgain3);
-	cgain4 = exp(log(0.001)*(mdata.ctime4/reverb_time)) * (1. - fgain4);
-	cgain5 = exp(log(0.001)*(mdata.ctime5/reverb_time)) * (1. - fgain5);
-	cgain6 = exp(log(0.001)*(mdata.ctime6/reverb_time)) * (1. - fgain6);
-
-	comb1 = new lpcomb(to_prime(mdata.ctime1,sr),cgain1,fgain1,1.0);
-	comb2 = new lpcomb(to_prime(mdata.ctime2,sr),cgain2,fgain2,1.0);
-	comb3 = new lpcomb(to_prime(mdata.ctime3,sr),cgain3,fgain3,1.0);
-	comb4 = new lpcomb(to_prime(mdata.ctime4,sr),cgain4,fgain4,1.0);
-	comb5 = new lpcomb(to_prime(mdata.ctime5,sr),cgain5,fgain5,1.0);
-	comb6 = new lpcomb(to_prime(mdata.ctime6,sr),cgain6,fgain6,1.0);
-	ap1		= new allpassfilter((long)sr,to_prime(mdata.atime1,sr),-0.6,0);
-	ap2		= new allpassfilter((long)sr,to_prime(mdata.atime2,sr),0.4,0);
-	ap3		= new allpassfilter((long)sr,to_prime(mdata.atime3,sr),-0.61,0);
-	ap4		= new allpassfilter((long)sr,to_prime(mdata.atime4,sr),0.39,0);
-
-	if(!
-		(comb1->create()
-		&& comb2->create()
-		&& comb3->create()
-		&& comb4->create()
-		&& comb5->create()
-		&& comb6->create()
-		&& ap1->create()
-		&& ap2->create()
-		&& ap3->create()
-		&& ap4->create()
-		)) {
-		cleanup();
-		return false;
-	}
-	return true;
-}
-
-double moorer::tick(double insam)
-{
-	out =  comb1->tick(insam) +
-		  comb2->tick(insam) +
-		  comb3->tick(insam) +
-		  comb4->tick(insam) +
-		  comb5->tick(insam) +
-		  comb6->tick(insam)
-		  ;
-	out *= scalefac;
-	out = ap4->tick(ap3->tick(ap2->tick(ap1->tick(out))));
-	return out;
-}
-
-void moorer::cleanup()
-{
-	delete comb6; comb6 = 0;
-	delete comb5; comb5 = 0;
-	delete comb4; comb4 = 0;
-	delete comb3; comb3 = 0;
-	delete comb2; comb2 = 0;
-	delete comb1; comb1 = 0;
-	delete ap4; ap4 = 0;
-	delete ap3; ap3 = 0;
-	delete ap2; ap2 = 0;
-	delete ap1; ap1 = 0;
-}
-
-
-/* vmtdelay *******************************/
-vcomb4::vcomb4()
-{
-	dl_buf = 0;
-	dl_length = 0;
-	dl_input_index = 0;
-	dl_srate = 0.0;
-	
-}
-
-vcomb4::~vcomb4()
-{
-	if(dl_buf && dl_length > 0)
-		delete [] dl_buf;
-		
-}
-
-bool vcomb4::init(long srate,double length_secs)
-{
-	unsigned long len_frames = (unsigned long)(srate * length_secs );	
-	if(len_frames == 0)
-		return false;
-	/*  round upwards to allow for interpolation */
-	len_frames++;
-	/* reject init  if already created*/
-	/* TODO: more sexy error checking/reporting... */
-	if(dl_buf) 
-		return false;
-	try {
-		dl_buf = new double[len_frames];
-	}
-	catch(...){
-		return false;
-	}
-	/* for VC 6 */
-	if(dl_buf == 0)
-		return false;
-	for(unsigned long i = 0; i < len_frames;i++)
-		dl_buf[i] = 0.0;
-
-	dl_length = len_frames;
-	dl_input_index = 0;
-	dl_srate = srate;
-	gain = 0.5;
-	return true;
-}
-
-bool vcomb4::init(const MOORERDATA *p_mdata,double reverbtime,double damping,double sr)
-{
-	if(!init((long) sr,reverbtime))
-		return false;
-    return true;
-}
-
-double vcomb4::tick(double vdelay_1,double vdelay_2,double vdelay_3,double vdelay_4,double feedback,double input)
-{
-	unsigned long base_readpos1,base_readpos2,base_readpos3,base_readpos4; 
-	unsigned long next_index1,next_index2,next_index3,next_index4; 
-	double vlength,dlength,frac;
-	double readpos1,readpos2,readpos3,readpos4;
-	double* buf = dl_buf;
-	
-	dlength = (double) dl_length;					/* get maxlen, save a cast later on */
-	/* read pointer is vlength ~behind~ write pointer */
-	/* tap1*/
-	vlength = dlength -  (vdelay_1  * dl_srate);
-	vlength = vlength < dlength ? vlength : dlength;	  /* clip vdelay to max del length */
-	readpos1 = dl_input_index + vlength;
-	base_readpos1 = (unsigned long) ( readpos1);				  
-	if(base_readpos1 >= dl_length)					  /* wrap dl indices */
-		base_readpos1 -= dl_length;
-	next_index1 = base_readpos1 + 1;
-	if(next_index1 >= dl_length)
-		next_index1 -= dl_length;
-	/*tap2*/
-	vlength = dlength -  (vdelay_2  * dl_srate);
-	vlength = vlength < dlength ? vlength : dlength;
-	readpos2 = dl_input_index + vlength;
-	base_readpos2 = (unsigned long) ( readpos2);				  
-	if(base_readpos2 >= dl_length)					  /* wrap dl indices */
-		base_readpos2 -= dl_length;
-	next_index2 = base_readpos2 + 1;
-	if(next_index2 >= dl_length)
-		next_index2 -= dl_length;
-	/*tap3*/
-	vlength = dlength -  (vdelay_3  * dl_srate);
-	vlength = vlength < dlength ? vlength : dlength;
-	readpos3 = dl_input_index + vlength;
-	base_readpos3 = (unsigned long) ( readpos3);				  
-	if(base_readpos3 >= dl_length)					  /* wrap dl indices */
-		base_readpos3 -= dl_length;
-	next_index3 = base_readpos3 + 1;
-	if(next_index3 >= dl_length)
-		next_index3 -= dl_length;
-	/* tap4 */
-	vlength = dlength -  (vdelay_4  * dl_srate);
-	vlength = vlength < dlength ? vlength : dlength;
-	readpos4 = dl_input_index + vlength;
-	base_readpos4 = (unsigned long) ( readpos4);				  
-	if(base_readpos4 >= dl_length)					  /* wrap dl indices */
-		base_readpos4 -= dl_length;
-	next_index4 = base_readpos4 + 1;
-	if(next_index4 >= dl_length)
-		next_index4 -= dl_length;
-
-
-	double outsum = 0.0;
-	fb1 *= feedback;
-	fb2 *= feedback;
-	fb3 *= feedback;
-	fb4 *= feedback;
-	/* basic interp of variable delay pos */
-	frac = readpos1 - (int) readpos1;	
-	outsum += fb1 * (buf[base_readpos1]+((buf[next_index1] - buf[base_readpos1]) * frac));
-	frac = readpos2 - (int) readpos2;	
-	outsum += fb2 * (buf[base_readpos2]+((buf[next_index2] - buf[base_readpos2]) * frac));
-	frac = readpos3 - (int) readpos3;	
-	outsum += fb3 * (buf[base_readpos3]+((buf[next_index3] - buf[base_readpos3]) * frac));
-	frac = readpos4 - (int) readpos4;	
-	outsum += fb4 * (buf[base_readpos4]+((buf[next_index4] - buf[base_readpos4]) * frac));
-	/* how do we scale all this?*/
-	input += gain * (outsum);	
-	/* add in new sample + fraction of ouput, unscaled, for minimum decay at max feedback */
-	buf[dl_input_index++] = input   + outsum * 0.25;
-	if(dl_input_index == dl_length)
-		dl_input_index = 0;
-	return  input + outsum * 0.25;
-
-}

dev/externals/reverb/reverberator.h

@@ -1,360 +0,0 @@
-/*
- * Copyright (c) 1983-2013 Richard Dobson and Composers Desktop Project Ltd
- * http://people.bath.ac.uk/masrwd
- * http://www.composersdesktop.com
- * This file is part of the CDP System.
- * The CDP System is free software; you can redistribute it
- * and/or modify it under the terms of the GNU Lesser General Public 
- * License as published by the Free Software Foundation; either 
- * version 2.1 of the License, or (at your option) any later version. 
- *
- * The CDP System is distributed in the hope that it will be useful, 
- * but WITHOUT ANY WARRANTY; without even the implied warranty of 
- * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  
- * See the GNU Lesser General Public License for more details.
- * You should have received a copy of the GNU Lesser General Public
- * License along with the CDP System; if not, write to the Free Software
- * Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
- *
- */
- 
-
-// reverberator.h: interface for the reverberator class.
-// RWD.9.98 now contains all my delay/allpass etc  ugs
-//RWD 03.20  corrected def of tick function to double everywhere
-//////////////////////////////////////////////////////////////////////
-
-#if !defined(AFX_REVERBERATOR_H__9DE45E43_83BA_11D1_96D4_444553540000__INCLUDED_)
-#define AFX_REVERBERATOR_H__9DE45E43_83BA_11D1_96D4_444553540000__INCLUDED_
-
-#if _MSC_VER >= 1000
-#pragma once
-#endif // _MSC_VER >= 1000
-
-#include "wavetable.h"
-
-enum {LOW_PASS,HIGH_PASS};
-enum {LOW_SHELVE,HIGH_SHELVE};
-
-unsigned int to_prime(double dur,double sr);
-
-
-
-typedef struct __tap
-{
-	double pos;
-	double val;
-}
-deltap;
-
-typedef struct nested_allpass_data{
-	unsigned int time1;
-	unsigned int time2;
-	unsigned int time3;
-	unsigned int delay1;
-	unsigned int delay2;
-	double gain1;
-	double gain2;
-	double gain3;
-}
-NESTDATA;
-
-typedef struct moorer_data{
-	double ctime1;
-	double ctime2;
-	double ctime3;
-	double ctime4;
-	double ctime5;
-	double ctime6;
-	double atime1;
-	double atime2;
-	double atime3;
-	double atime4;
-}
-MOORERDATA;
-//abstract base class for all processors
-
-class cdp_processor 
-{
-public:
-	cdp_processor() {}
-	virtual ~cdp_processor(){}
-	//not all processes will need to use this
-	virtual bool create() {return true;}
-//	virtual float tick(float insam)	{return insam;}
-    virtual double tick(double insam) = 0;
-};
-
-
-//could have a method create_taps(uint ntaps, float maxtime)  ... ?
-class tapdelay 
-{
-public:
-	tapdelay(const deltap *taps,unsigned int ctaps,double sr);
-	bool	create(void);
-	double  get_maxgain(void);
-	void set_gain(double gain);
-	double	tick(double insam);
-	virtual ~tapdelay();
-private:
-	double *tapbuf;
-	double srate;
-	double one_over_ntaps;
-	double max_gain;		/* get rms value of whole array */
-	double output_gain;
-	unsigned int ntaps;
-	unsigned int buflen;
-	unsigned int *tapptr;
-	double *tapcoeff;
-	unsigned int reader;
-	const deltap *dtaps;
-};
-
-//basic feedthrough delay (independent of srate) - could be base class?
-class delay
-{
-public:
-	delay(unsigned int len, double gain);
-	virtual ~delay();
-	bool create(void);
-	double tick(double insam);
-private:
-	double *delbuf;
-	unsigned int buflen;
-	double dgain;
-	unsigned int ptr;
-};
-
-
-
-class allpassfilter  
-{
-public:
-	allpassfilter(long sr,unsigned int time, double ingain,unsigned int predelay);
-	virtual ~allpassfilter();
-	bool create(void);    
-    void set_vfreq(double freq);
-    void set_vmod(double amount);
-	double tick(double insam);
-    double vtick(double insam);
-
-private:
-	double gain;
-	double *rvbuf1;			 //CDP allpass uses 2 buffers!
-    double vfreq,vmod;
-    fastlfo *sinelfo;
-    fastlfo *noiselfo;
-	unsigned int rvblen, writer1, reader1;
-	delay *pre_dline;
-	unsigned int pre_delay;
-    long srate;
-};
-
-///////////////// DOUBLE-NESTED ALLPASS
-///  set either or both gain2, gain3 to zero to disable the respective allpass
-////	
-class onepole;
-
-class  nested_allpass
-{
-public:
-	nested_allpass(double srate,double lpfreq,unsigned int outertime, 
-					unsigned int time1,
-					unsigned int time2,
-					double gain, 
-					double gain1,
-					double gain2, 
-					unsigned int delay1, 
-					unsigned int delay2);
-	virtual ~nested_allpass();
-	bool create(void);
-	double tick(double insam);
-    void  set_vfreq(double freq);
-    void set_vmod(double amount);
-    double vtick(double  insam);
-	void set_damping(bool damping) { damping_ = damping;}
-	bool get_damping() const { return damping_;}
-private:
-	
-	double srate_, outer_gain, ap1_gain,ap2_gain;
-	unsigned int ap1_length,ap2_length;
-	unsigned int outer_time,ap1_delay,ap2_delay;
-	allpassfilter *ap1, *ap2;
-	double *buf;
-	unsigned int writer,reader;
-	// RWD experimental - internal lf damping
-	onepole* lp; 
-	bool damping_;
-	double sr_,lpfreq_;
-
-};
-
-
-//TODO: comb filter (with lopass), and tapped delay
-
-//basic lowpass for combs
-
-class lowpass1
-{
-public:
-	lowpass1(double filtgain);
-	virtual ~lowpass1();
-	double tick(double insam);
-private:
-	double temp,gain,output;	
-};
-
-
-class onepole
-{
-public:
-	onepole(double freq,double srate,int low_high);
-	virtual ~onepole();
-	double tick(double input);
-private:
-	double a1,a2,b,lastout;
-};
-
-class tonecontrol
-{
-public:
-	tonecontrol(double frq,double dbfac,int type,double srate);
-	virtual ~tonecontrol();
-	bool create(void);
-	double tick(double input);
-private:
-	double freq, boost, sr,a0, a1, a2, b1, b2;
-	double x1,x2,y1,y2;
-	int tc_type;
-	void lshelve(void);
-	void hshelve(void);
-};
-
-class lpcomb
-{
-public:
-	lpcomb(unsigned int time, double ingain, double lpcoeff, double prescale);
-	virtual ~lpcomb();
-	bool create(void);
-	double tick(double insam);
-private:
-	double prescale, gain, lpgain;
-	double *combbuf;
-	unsigned int rvblen, reader1,writer1;
-	lowpass1 *lp;
-
-};
-
-
-class small_diffuser : public cdp_processor
-{
-public:
-	small_diffuser(unsigned int pre_delay,
-					const NESTDATA *apdata1,
-					const NESTDATA *apdata2,double fb_gain,double lp_freq,double srate);
-	virtual ~small_diffuser();
-	bool create(void);
-	double tick(double insam);
-	void set_damping(bool damping) { damping_ = damping; }
-private:
-	NESTDATA ap1_data,ap2_data;
-	nested_allpass *ap1,*ap2;
-	delay *predelay;
-	//lowpass1 *lp1;
-	onepole *lp1;
-	unsigned int predelay_time;
-	double lpgain,lpfreq,sr;
-	double out1,out2;
-	bool damping_;
-};
-
-//no public output from postdelay, so internal only
-class medium_diffuser : public cdp_processor
-{
-public:
-	medium_diffuser(double post_delay,const NESTDATA *apdata1,
-					const NESTDATA *apdata2,
-					double gain,
-					double lp_freq,
-					double srate);
-	virtual ~medium_diffuser();
-	bool create(void);
-	double tick(double insam);
-	void set_damping(bool damping) { damping_ = damping;	}
-private:
-	NESTDATA ap1_data,ap2_data;
-	nested_allpass *ap1,*ap2;
-	allpassfilter *ap3;				//no public access to this, for now...
-	onepole *lp1;
-	delay *delay1,*delay2,*postdelay;			
-	double md_gain,lpfreq,sr;
-	double out1,out2,out3;
-	unsigned int postdelay_time;
-	bool damping_;
-};
-
-
-class large_diffuser : public cdp_processor
-{
-public:
-	large_diffuser(const NESTDATA *apdata1,
-					const NESTDATA *apdata2,
-					double gain,
-					double lp_freq,
-					double srate);
-	virtual ~large_diffuser();
-	bool create(void);
-	double tick(double insam);
-	void set_damping(bool damping) { damping_ = damping; }
-private:
-	void cleanup();
-	NESTDATA ap1_data,ap2_data;
-	nested_allpass *ap1,*ap2;
-	allpassfilter *ap3,*ap4;				//no public access to this, for now...
-	onepole *lp1;
-	delay *delay1,*delay2,*delay3,*delay4;			
-	double ld_gain,lpfreq,sr;
-	double out1,out2,out3;
-	bool damping_;
-};
-
-class moorer : public cdp_processor
-{
-public:
-	moorer(const MOORERDATA *p_mdata,double reverbtime,double damping,double sr);
-	virtual ~moorer();
-	bool create(void);
-	double tick(double insam);
-private:
-	void cleanup();
-	lpcomb *comb1,*comb2,*comb3,*comb4,*comb5,*comb6;
-	allpassfilter *ap1,*ap2,*ap3,*ap4;
-	MOORERDATA mdata;
-	double cgain1,cgain2,cgain3,cgain4,cgain5,cgain6;
-	double fgain1,fgain2,fgain3,fgain4,fgain5,fgain6;	
-	double reverb_time,dampfac,sr;
-	double out,scalefac;
-};
-
-/* vmtdelay ****************/
-class vcomb4 
-{
-public:
-	vcomb4();
-	virtual ~vcomb4();
-	bool init(long srate,double length_secs);
-	bool init(const MOORERDATA *p_mdata,double reverbtime,double damping,double sr);
-	double tick(double vdelay_1,double vdelay_2,double vdelay_3,double vdelay_4,double feedback,double input);
-	void setfgains(double fg1,double fg2,double fg3,double fg4) {fb1 = fg1;fb2 = fg2;fb3 = fg3;fb4 = fg4;}
-private:
-	double*			dl_buf;
-	unsigned long	dl_length;
-	unsigned  long	dl_input_index;
-	double			dl_srate;
-	double			fb1,fb2,fb3,fb4;
-//	lowpass1		lp1,lp2,lp3,lp4;
-	double		gain;
-};
-
-
-#endif // !defined(AFX_REVERBERATOR_H__9DE45E43_83BA_11D1_96D4_444553540000__INCLUDED_)

dev/externals/reverb/rmverb.cpp

@@ -1,949 +0,0 @@
-/*
- * Copyright (c) 1983-2013 Richard Dobson and Composers Desktop Project Ltd
- * http://people.bath.ac.uk/masrwd
- * http://www.composersdesktop.com
- * This file is part of the CDP System.
- * The CDP System is free software; you can redistribute it
- * and/or modify it under the terms of the GNU Lesser General Public 
- * License as published by the Free Software Foundation; either 
- * version 2.1 of the License, or (at your option) any later version. 
- *
- * The CDP System is distributed in the hope that it will be useful, 
- * but WITHOUT ANY WARRANTY; without even the implied warranty of 
- * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  
- * See the GNU Lesser General Public License for more details.
- * You should have received a copy of the GNU Lesser General Public
- * License along with the CDP System; if not, write to the Free Software
- * Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
- *
- */
- 
-#include <stdio.h>
-#include <stdlib.h>
-#include <math.h>
-#include <string.h>
-#ifdef _DEBUG
-#include <assert.h>
-#endif
-#include <algorithm>
-#include <vector>
-
-using namespace std;
-// TODO: clean up "new" handling, move to VC7 and get to use exceptions!
-enum cmdargs {
-		PROGNAME,
-		INFILE,
-		OUTFILE,
-		REVERBTYPE,
-		RGAIN,
-		MIX,
-		FEEDBACK,		
-		LPABSORB,
-		LPFREQ,
-		TRTIME,
-		CONFIG
-};
-
-enum reverb_type {
-		DIFF_SMALL,
-		DIFF_MEDIUM,
-		DIFF_LARGE
-};
-
-extern "C" {
-#include "portsf.h"
-	   //NB requires stdio.h etc - time to change this?
-}
-
-#include "reverberator.h"
-//RWD.10.98 TODO: for final versions, MUST limit input delay times! can't have 108sec delays.....
-//get all reflection data
-#include "reflect.cpp"
-
-
-long readtaps(FILE *fp, deltap **taps,double sr);		
-
-const char* cdp_version = "6.0.0";
-
-void usage(void);
-
-int
-main(int argc,char *argv[])
-
-{
-	int i,ifd, ofd,reverbtype = DIFF_SMALL;
-	FILE *fp = 0;
-	long	chans,outchans = 2;
-	double	sr,trailertime;
-	cdp_processor *cdpdiff = 0;
-	NESTDATA data1,data2;
-	PSF_CHPEAK *peaks;
-	PSF_PROPS props;
-	unsigned int time1a,time1b,time1c,time2a,time2b,time2c;
-	double feedback,fb_lpfreq, lp_freq,predelay = -1.0;
-	double nyquist,pseudo_nyquist;
-	double dry_gain,wet_gain;
-	//double early_gain = 1.0;
-	double diff_gain = 1.0;
-	// nested-allpass diffusers
-	small_diffuser *s_diffuser = 0;
-	medium_diffuser *m_diffuser = 0;
-	large_diffuser *l_diffuser = 0;
-	// traioling plain allpasses for front pair de-correlation
-	allpassfilter *ap1_l = 0,*ap2_l = 0;		
-	allpassfilter *ap1_r = 0,*ap2_r = 0;
-	// decorrelation for other channels (surround)
-	allpassfilter **chan_ap = 0;
-	onepole *lp = 0;
-	// tonecontrols max -12dB slope, variable; 
-	tonecontrol *tc_l_lowcut = 0, *tc_r_lowcut = 0;	
-	tonecontrol *l_highcut = 0, *r_highcut = 0;
-	double lowcut_freq = 0.0,highcut_freq = 0.0;
-	tapdelay *tdelay = 0;	
-	long ntaps = 0;
-	deltap *ptaps = 0;
-	bool want_mono = false;
-	bool want_floats = false;
-	bool usertaps = false;
-	bool double_damping = false;
-
-	if((argc==2) && strcmp(argv[1],"--version")==0) {
-		fprintf(stdout,"%s\n",cdp_version);
-		fflush(stdout);
-		return 0;
-	}
-	if(psf_init()){
-		fprintf(stderr,"rmverb: initialisation failed\n");
-		return 1;
-	}
-    	
-	if(argc < CONFIG){
-		if(argc > 1)
-			fprintf(stderr,"rmverb: insufficient arguments\n");
-		usage();
-	}
-	
-	while((argc > 1) && argv[1][0]=='-'){
-		char *arg = argv[1];
-		switch(*(++arg)){
-		case('p'):
-			if(*(++arg) =='\0'){
-				fprintf(stderr,"rmverb: predelay requires a parameter\n");
-				usage();
-			}
-			
-			predelay = atof(arg);
-			if(predelay <0.0){
-				fprintf(stderr,"rmverb: predelay must be >= 0.0\n");
-				usage();
-			}
-			predelay *= 0.001;			//get msecs from user
-			break;
-		case('c'):
-			if(*(++arg) == '\0') {
-				fprintf(stderr,"rmverb: -c flag requires a value\n");
-				usage();
-			}
-			outchans = atoi(arg);
-			if(outchans < 1 || outchans > 16){
-				fprintf(stderr,"rmverb: impossible channel value requested\n");
-				usage();
-			}
-			if(outchans==1)
-				want_mono = true;			
-			break;
-		case('d'):
-			double_damping = true;
-			break;
-		case('e'):
-			if(*(++arg)=='\0'){
-				fprintf(stderr,"rmverb: -e flag needs a filename\n");
-				usage();
-			}
-			if((fp = fopen(arg,"r")) ==0){
-				fprintf(stderr,"rmverb: unable to open breakpoint file %s\n",arg);
-				exit(1);
-			}
-			usertaps = true;
-			break;
-			// -f is CDP-specific: read old 32bit int format as floats
-		case('f'):
-			want_floats = true;
-			if(*(++arg) != '\0')
-				fprintf(stderr,"rmverb WARNING: -f flag does not take a parameter\n");
-			break;
-		case('L'):
-			if(*(++arg) == '\0'){
-				fprintf(stderr,"rmverb: -L flag needs frequency argument\n");
-				usage();
-			}			
-			lowcut_freq = atof(arg);
-			if(lowcut_freq <= 0.0){
-				fprintf(stderr,"rmverb: Lowcut freq must be greater than zero\n");
-				usage();
-			}
-			break;
-		case('H'):
-			if(*(++arg) == '\0'){
-				fprintf(stderr,"rmverb: -H flag needs frequency argument\n");
-				usage();
-			}			
-			highcut_freq = atof(arg);
-			if(highcut_freq <= 0.0){
-				fprintf(stderr,"rmverb: Highcut freq must be greater than zero\n");
-				usage();
-			}
-			break;
-		default:
-			fprintf(stderr,"rmverb: illegal flag option %s\n",argv[1]);
-			usage();
-			break;
-		}
-		argc--;
-		argv++;
-	}
-
-	if(argc < CONFIG){
-		printf("rmverb: insufficient arguments\n");
-		usage();
-	}
-#ifdef _DEBUG
-	if(predelay > 0.0)
-		printf("got predelay = %.4lf\n",predelay);
-#endif
-
-	chan_ap = new allpassfilter*[outchans];
-	if(chan_ap==0){
-		fprintf(stderr,"rmverb: no memory for multi-channel diffusion\n");
-		exit(1);
-	}
-
-	if((ifd = psf_sndOpen(argv[INFILE],&props,0)) < 0) {
-				fprintf(stderr,"\nrmverb: cannot open input file %s", argv[INFILE]);
-				exit(1);
-	}
-	
-	if(props.format  <= PSF_FMT_UNKNOWN || props.format > PSF_AIFC){
-		fprintf(stderr,"infile is not a recognised soundfile\n");
-		psf_sndClose(ifd);
-		return 1;
-	}
-	
-	sr = (double) props.srate;
-	nyquist = sr / 2.0;
-	pseudo_nyquist = nyquist * 0.7; // filters not reliable close to nyquist
-
-	chans = props.chans;
-	if(chans > 2){
-		fprintf(stderr,"rmverb can only accept mono or stereo files\n");
-		exit(1);
-	}
-		
-	
-	reverbtype =  atoi(argv[REVERBTYPE]) - 1;
-	if((reverbtype < DIFF_SMALL) || (reverbtype > DIFF_LARGE)){
-		fprintf(stderr,"rmverb: rmsize must be 1,2 or 3\n");
-		exit(1);
-	}
-	
-	diff_gain = atof(argv[RGAIN]);
-	if(diff_gain <= 0.0 || diff_gain > 1.0){
-		fprintf(stderr,"rgain must be > 0 and <= 1.0\n");
-		exit(1);
-	}
-
-	dry_gain =  atof(argv[MIX]);			  //global output gain from diffuser
-	if(dry_gain < 0.0 || dry_gain > 1.0 ){
-		printf("reverb: mix must be  between 0.0 and 1.0\n");
-		usage();
-	}
-	wet_gain = 1.0f - dry_gain;
-	// some odd people like to have 100% wet signal...
-	wet_gain *= 2.0;					//not v scientific, but works...
-	
-	feedback = atof(argv[FEEDBACK]);			  //feedback in diffuser
-	if(feedback < 0.0 || feedback >1.0){
-		printf("rmverb: feedback must be within 0.0 - 1.0\n");
-		usage();
-	}
-	
-	fb_lpfreq = atof(argv[LPABSORB]);			 //feedback lp-filter in diffuser
-	if(fb_lpfreq < 0.0 || fb_lpfreq > pseudo_nyquist){
-		printf("rmverb: absorb must be within 0 to %f\n",pseudo_nyquist);
-		usage();
-	}
-
-	lp_freq = atof(argv[LPFREQ]);
-	if(lp_freq < 0.0 || lp_freq > pseudo_nyquist){
-		printf("rmverb: lpfreq must be within 0.0 to %.4lfHz\n",pseudo_nyquist);
-		usage();
-	}
-	trailertime = atof(argv[TRTIME]);
-	if(trailertime < 0.0){
-		fprintf(stderr,"rmverb: trtime must be >= 0.0\n");
-		usage();
-	}
-	//from -m flag; will override a stereo input too
-	if(want_mono)
-		outchans =  1;
-	// NB: now specifying two trailing allpasses for diffuser
-	// handles front pair; currently using chan_ap[] only for remaining channels
-	// values are experimental!
-	double apdelay = 0.02;
-	double apfac = 0.02 / outchans;
-	for(i = 0; i < outchans; i++){						// or use a random modifier?
-		chan_ap[i] =  new allpassfilter(sr,to_prime(apdelay - apfac * (double)i,sr),0.4,0);
-		if(chan_ap[i] ==0){
-			fprintf(stderr,"rmverb: no memory for output diffusers\n");
-			exit(1);
-		}
-		if(!chan_ap[i]->create()){
-			fprintf(stderr,"rmverb: no memory for output diffusers\n");
-			exit(1);
-		}
-	}
-    // custom early reflections from time/value text-file
-	if(usertaps){
-#ifdef _DEBUG
-		assert(fp);
-#endif
-		ntaps = readtaps(fp,&ptaps,sr);
-		if(ntaps==0){
-			fprintf(stderr,"rmverb: error reading tapfile\n");
-			exit(1);
-		}
-		printf("rmverb: loaded %ld early reflections from tapfile\n",ntaps);
-		fclose(fp);
-		fp = 0;
-	}
-
-
-	//create input low/high filters, if wanted
-	if(lowcut_freq > 0.0){
-		if(highcut_freq > 0.0 && highcut_freq <= lowcut_freq){
-			fprintf(stderr,"rmverb: Highcut frequency	 must be higher than Lowcut frequency\n");
-			usage();
-		}
-		//lowcut is based on low shelving eq filter; here fixed at 12dB
-		// but can easily add this as user param
-		tc_l_lowcut = new tonecontrol(lowcut_freq,-12.0,LOW_SHELVE,sr);
-		if(!tc_l_lowcut->create()){
-			fprintf(stderr,"rmverb: unable to create Lowcut filter\n");
-			exit(1);
-		}
-		if(chans==2){
-			tc_r_lowcut = new tonecontrol(lowcut_freq,-12.0,LOW_SHELVE,sr);
-			if(!tc_r_lowcut->create()){
-				fprintf(stderr,"rmverb: unable to create Lowcut filter\n");
-				exit(1);
-			}
-		}
-	}
-
-	if(highcut_freq > 0.0){		
-		l_highcut = new tonecontrol(highcut_freq,-12.0,HIGH_SHELVE,sr);
-		if(!l_highcut->create()){
-			fprintf(stderr,"rmverb: unable to create Highcut filter\n");
-				exit(1);
-		}
-		if(chans==2){
-			r_highcut = new tonecontrol(highcut_freq,-12.0,HIGH_SHELVE,sr);
-			if(!r_highcut->create()){
-				fprintf(stderr,"\nrmverb: unable to create Highcut filter");
-				exit(1);
-			}
-		}
-	}
-
-	//now create diffuser
-
-	switch(reverbtype){
-		case(DIFF_SMALL):
-			time1a = to_prime(0.0047, sr);
-			time1b = to_prime(0.022, sr);
-			time1c = to_prime(0.0083,sr);
-
-			time2a = to_prime(0.036,sr);
-			time2b = to_prime(0.03,sr);
-			time2c = 0;//dummy
-			data1.time1 = time1a;
-			data1.time2 = time1b;
-			data1.time3 = time1c;
-			data1.gain1 = 0.3;
-			data1.gain2 = 0.4;
-			data1.gain3 = 0.6;
-			data1.delay1 = 0;
-			data1.delay2 = 0;
-			data2.time1 = time2a;
-			data2.time2 = time2b;
-			data2.time3 = time2c;
-			data2.gain1 = 0.1;
-			data2.gain2 = 0.4;
-			data2.gain3 = 0.0;			//not used by gardner
-			data2.delay1 = 0;
-			data2.delay2 = 0;
-
-
-			break;
-		case(DIFF_MEDIUM):
-			time1a = to_prime(/*0.0047*/0.035, sr);
-			time1b = to_prime(0.0083, sr);
-			time1c = to_prime(0.022,sr);
-
-			time2a = to_prime(/*0.0292*/0.039,sr);
-			time2b = to_prime(0.0098,sr);
-			time2c = 0;//dummy
-			data1.time1 = time1a;
-			data1.time2 = time1b;
-			data1.time3 = time1c;
-			data1.gain1 = 0.3;
-			data1.gain2 = 0.7;
-			data1.gain3 = 0.5;
-			data1.delay1 = 0;
-			data1.delay2 = 0;
-			data2.time1 = time2a;
-			data2.time2 = time2b;
-			data2.time3 = time2c;
-			data2.gain1 = 0.3;
-			data2.gain2 = 0.6;
-			data2.gain3 = 0.0;			//not used by gardner
-			data2.delay1 = 0;
-			data2.delay2 = 0;
-
-
-
-			break;
-		case(DIFF_LARGE):
-			// VERY LARGE!
-			// NB: orig outer delays from Gardner generate very strong discrete echoes,
-			// and hence also some pulsing in the dense reverb
-			// for really large spaces (caverns) this is OK, otherwise we ant to control this
-			// via predelay, and earlies
-			// NOTE: this will NOT WORK when sharing a single delay line!
-			// in that case, outer delay MUST be > internal ones
-			// TODO: devise user param to control a range of delay values
-			// (maybe derived from feedback level?), to further scale room size
-			time1a = to_prime(0.03 /*0.087*/, sr);
-			time1b = to_prime(0.062, sr);
-			time1c = to_prime(0.022,sr);	//dummy: not used
-
-			time2a = to_prime(0.018/*0.12*/,sr);
-			time2b = to_prime(0.076,sr);
-			time2c = to_prime(0.03,sr);		
-			data1.time1 = time1a;
-			data1.time2 = time1b;
-			data1.time3 = time1c;
-			data1.gain1 = 0.5;
-			data1.gain2 = 0.25;
-			data1.gain3 = 0.0;			//not used
-			data1.delay1 = 0;
-			data1.delay2 = 0;
-			data2.time1 = time2a;
-			data2.time2 = time2b;
-			data2.time3 = time2c;
-			data2.gain1 = 0.5;
-			data2.gain2 = 0.25;
-			data2.gain3 = 0.25;		
-			data2.delay1 = 0;
-			data2.delay2 = 0;
-			break;
-		default:
-#ifdef _DEBUG
-			assert(false);
-#endif
-			break;
-	}
-#ifdef _DEBUG
-	//RWD: development only!
-	if(argc==CONFIG+1){
-		int got = 0;
-		double time1a,time1b,time1c,time2a,time2b,time2c,gain1a,gain1b,gain1c,gain2a,gain2b,gain2c;
-		FILE *fp = fopen(argv[CONFIG],"r");
-		if(!fp)
-			printf("rmverb: can't open datafile %s, using presets\n",argv[CONFIG]);
-		else {
-		   // time1a,gain1a,time2a,gain2a,time3a,gain3a...
-			printf("loading diffusion data...\n");
-			got = fscanf(fp,"%lf%lf%lf%lf%lf%lf%lf%lf%lf%lf%lf%lf",
-			  &time1a,&gain1a,
-			  &time1b,&gain1b,
-			  &time1c,&gain1c,
-			  &time2a,&gain2a,
-			  &time2b,&gain2b,
-			  &time2c,&gain2c);
-			fclose(fp);
-			if(got != 12)
-				printf("rmverb: error reading data values\n");
-			else {
-				data1.time1 = to_prime(time1a,sr);
-				data1.time2 = to_prime(time1b,sr);
-				data1.time3 = to_prime(time1c,sr);
-				data1.gain1 = gain1a;
-				data1.gain2 = gain1b;
-				data1.gain3 = gain1c;
-
-				data2.time1 = to_prime(time2a,sr);
-				data2.time2 = to_prime(time2b,sr);
-				data2.time3 = to_prime(time2c,sr);
-				data2.gain1 = gain2a;
-				data2.gain2 = gain2b;
-				data2.gain3 = gain2c;
-			}
-		}
-	}
-	
-	else{	
-		printf("using preset parameters:\n");
-		printf("DATA 1: time1 = %u\n\ttime2 = %u\n\ttime3 = %u\n",data1.time1,data1.time2,data1.time3);
-		printf("DATA 2: time1 = %u\n\ttime2 = %u\n\ttime3 = %u\n",data2.time1,data2.time2,data2.time3);
-		printf("feedback = %.4lf,fb_lpfreq = %.4lf\n",feedback,fb_lpfreq);
-		printf("dry gain = %.4lf, wet gain = %.4lf\n",dry_gain,wet_gain);
-	}
-#endif	
-
-	
-	switch(reverbtype){
-	case(DIFF_SMALL):	
-		printf("rmverb: using small room model\n");
-		s_diffuser = new small_diffuser((unsigned int)(0.024 * sr),&data1,&data2,feedback,fb_lpfreq,sr);
-		s_diffuser->set_damping(double_damping);
-		cdpdiff =  dynamic_cast<cdp_processor *>(s_diffuser);
-		if(!usertaps){
-			ptaps = smalltaps;
-			ntaps = sizeof(smalltaps) / sizeof(deltap);
-		}
-		break;
-	case(DIFF_MEDIUM):
-		printf("rmverb: using medium room model\n");
-		m_diffuser = new medium_diffuser(0.108,&data1,&data2,feedback,fb_lpfreq,sr);
-		m_diffuser->set_damping(double_damping);
-		cdpdiff =  dynamic_cast<cdp_processor *>(m_diffuser);
-		if(!usertaps){
-			ptaps = mediumtaps;
-			ntaps = sizeof(mediumtaps) / sizeof(deltap);
-		}
-		//ptaps = longtaps;
-		//ntaps = sizeof(longtaps) / sizeof(deltap);
-
-		break;
-	case(DIFF_LARGE):
-		printf("using large room model\n");
-		//fb_lpfreq = 2600.0; // Gardner fixed val
-		l_diffuser = new large_diffuser(&data1,&data2,feedback,fb_lpfreq,sr);
-		l_diffuser->set_damping(double_damping);
-		cdpdiff =  dynamic_cast<cdp_processor *>(l_diffuser);
-		if(!usertaps){
-			ptaps = largetaps;
-			ntaps = sizeof(largetaps) / sizeof(deltap);
-		}
-		break;
-	default:
-#ifdef _DEBUG
-		assert(false);
-#endif
-		break;
-	}
-
-	if(!cdpdiff){
-		puts("rmverb: no memory for reverberator\n");
-		exit(1);
-	}
-	if(!cdpdiff->create()){
-		puts("rmverb: no memory for reverberator buffers\n");
-		exit(1);
-	}
-
-
-	/*********** GLOBAL COMPONENTS *************/
-
-	if(lp_freq > 0.0)
-		lp		= new onepole(lp_freq,sr,LOW_PASS);
-	
-	//if user prescribes a predelay, adjust all taptimes
-	if(predelay >= 0.0){
-		double current_delay = ptaps[0].pos;
-		double adjust = predelay - current_delay;
-
-		for(int i = 0; i < ntaps; i++)
-			ptaps[i].pos += adjust;
-
-	}
-	else
-		predelay = ptaps[0].pos;	//to report to user
-	tdelay = new tapdelay(ptaps,ntaps,sr);
-
-	if(!tdelay->create()) {
-		fputs("rmverb: no memory for early reflections\n",stderr);
-		return 1;
-	}
-
-	//max_gain = tdelay->get_maxgain();
-	//if(max_gain > 1.0)
-	//	tdelay->set_gain(max_gain * early_gain);
-
-
-	// final allpass chain: 0.02,0.014,0.009,0.006
-	ap1_l		= new allpassfilter(sr,to_prime(0.02,sr),-0.6,0);
-	ap2_l		= new allpassfilter(sr,to_prime(0.014,sr),0.4,0);
-//	ap3_l		= new allpassfilter(to_prime(0.009,sr),-0.6,0);
-//	ap4_l		= new allpassfilter(to_prime(0.006,sr),0.4,0);
-	if(!(ap1_l->create()
-		&& ap2_l->create()
-//		&& ap3_l->create()
-//		&& ap4_l->create()
-		)){
-		fprintf(stderr,"rmverb: no memory for allpass chain\n");
-		exit(1);
-	}
-	if(outchans >=2){
-		// slightly different params to make stereo!
-		// tune front pair precisely; further chans use formula
-		ap1_r		= new allpassfilter(sr,to_prime(0.025,sr),-0.6,0);
-		ap2_r		= new allpassfilter(sr,to_prime(0.0145,sr),0.4,0);
-//		ap3_r		= new allpassfilter(to_prime(0.0095,sr),-0.6,0);
-//		ap4_r		= new allpassfilter(to_prime(0.0065,sr),0.4,0);
-		if(!(ap1_r->create()
-			&& ap2_r->create()
-//			&& ap3_r->create()
-//			&& ap4_r->create()
-			)){
-			fprintf(stderr,"rmverb: no memory for allpass chain\n");
-			exit(1);
-		}
-	}
-
-	
-
-	printf("using %ld early reflections: predelay = %.2lf msecs\n",ntaps,predelay * 1000.0);
-    int clipfloats = 1;
-    int minheader = 0;
-	if(want_floats) {
-		props.samptype =PSF_SAMP_IEEE_FLOAT;
-        clipfloats = 0;        
-    }
-	props.chans = outchans;
-    
-	if((ofd = psf_sndCreate(argv[OUTFILE],&props,clipfloats,minheader,PSF_CREATE_RDWR)) < 0) {
-        fprintf(stderr,"rmverb: cannot open output file %s\n",argv[OUTFILE]);
-        return 1;
-	}
-
-/****
- 																		|wet
- *****   in-> ---o-->pre_lp -->earlies-o--->diffuser-->*diffgain-->--+--*---+----out
-				 |				       |							 |		|
-				 |					   |>--------*-(earlies)->-------|		*--dry
-				 |							     ^earlygain				    |
-				 |__________________________________________________________|
-
- ****/
-	printf("\ninfile chans = %ld, outfile chans = %ld",chans,outchans);
-	printf("\n");
-	long outframes = 0;
-	long step = (long)(0.25 * sr);
-    float insamps[2], outsamps[16];
-	double l_out,r_out;
-//    float  finsamp, foutsamp;
-	for(;;){
-		int rv;
-		double l_ip,r_ip, out,l_direct,r_direct,mono_direct,earlies = 0.0f;
-        
-		//read mono/left
-		if((rv = psf_sndReadFloatFrames(ifd,insamps,1)) < 0){
-			fprintf(stderr,"rmverb: error reading file\n"); 
-			exit(1);			
-		}		
-		if(rv==0) break;		// end of inputfile	- handle any trailertime
-		l_ip = (double) insamps[0];
-		//apply any conditioning to direct signal
-		if(tc_l_lowcut)
-			l_ip =  tc_l_lowcut->tick(l_ip);
-		if(l_highcut)
-			l_ip =  l_highcut->tick(l_ip);
-		l_direct = l_ip;
-		mono_direct = l_direct;
-		r_direct = l_direct;
-		//handle R channnel if active
-		if(chans==2){
-            r_ip = (double) insamps[1];
-			//apply any conditioning to direct signal			
-			if(tc_r_lowcut)
-				r_ip =  tc_r_lowcut->tick(r_ip);
-			if(r_highcut)
-				r_ip =  r_highcut->tick(r_ip);
-			r_direct = r_ip;
-			if(want_mono)	//input merged sig to reverbreator
-				mono_direct = (l_direct + r_direct) * 0.5;			 
-		}
-		// mono_direct = (mixed) mono input to  reverb		
-		//possibly also filter it...
-		if(lp)
-			mono_direct =  lp->tick(mono_direct);					//input lowpass
-		earlies = tdelay->tick(mono_direct);				//early reflections	
-		//send (filtered) mono output from reverberator
-
-		// TODO: find formula for diffgain...
-		out = earlies + (cdpdiff->tick(earlies + mono_direct) * diff_gain * 0.707);		 //the dense reverberation		
-		// output:: use 2 allpasses per chan to generate stereo reverb, wider diffusion
-		l_out = wet_gain *  ap2_l->tick(ap1_l->tick(out));
-
-		// old method - 1 allpass per channel
-		//l_out = chan_ap[0]->tick(out) * wet_gain;
-
-		if(outchans == 1)			
-			l_out += (mono_direct * dry_gain);
-		else 
-			l_out += l_direct * dry_gain;
-        outsamps[0] = (float) l_out;
-		
-		if(outchans>=2){
-			r_out = wet_gain *  ap2_r->tick(ap1_r->tick(out));			
-			// old version
-			//r_out = chan_ap[1]->tick(out) * wet_gain;
-			r_out += r_direct * dry_gain;
-			outsamps[1] = (float) r_out;
-		}
-
-		//now any further channels; reduced level of direct sig
-		for(i=2;i < outchans; i++){
-            out = earlies + (cdpdiff->tick(earlies + (mono_direct * 0.3)) * diff_gain * 0.707);
-			l_out = wet_gain * chan_ap[i]->tick(out);
-            outsamps[i] = (float) l_out;
-			
-		}
-        if(psf_sndWriteFloatFrames(ofd,outsamps,1) != 1){
-            fprintf(stderr,"rmverb: error writing output file\n"); 
-			return 1;
-        }
-		outframes++;
-		if((outframes % step) == 0)
-			fprintf(stdout,"%.2f\r",(double)outframes/(double)sr);
-	}
-	
-	int trtime  = (int)( trailertime * sr);
-	for(i = 0; i < 	trtime; i++){		
-		double tr_l_ip,tr_r_ip = 0.0f,tr_out,tr_l_direct,tr_r_direct,
-			tr_mono_direct,tr_earlies = 0.0f,tr_l_out,tr_r_out;
-		//need last active samps from input filter(s)
-		tr_l_ip = 0.0;
-		if(tc_l_lowcut)
-			tr_l_ip =  tc_l_lowcut->tick(tr_l_ip);
-		if(l_highcut)
-			tr_l_ip =  l_highcut->tick(tr_l_ip);
-		tr_l_direct = tr_l_ip;
-		tr_mono_direct = tr_l_direct;
-		tr_r_direct = tr_l_direct;
-		//handle R channnel if active
-		if(chans==2){
-			tr_r_ip = 0.0;	   // inout is zero, except if using input filters
-			// get any samples from input conditioning filters
-			if(tc_r_lowcut)
-				tr_r_ip =  tc_r_lowcut->tick(0.0);						   
-			if(r_highcut)
-				tr_r_ip =  r_highcut->tick(tr_r_ip);
-			tr_r_direct = tr_r_ip;
-			if(want_mono)
-				tr_mono_direct = (tr_l_direct + tr_r_direct) *0.5f;			
-		}
-
-		// mono_direct = (mixed) mono input to  reverb
-		//possibly also filter it...
-		if(lp)
-			tr_mono_direct =  lp->tick(tr_mono_direct);		
-		tr_earlies = tdelay->tick(tr_mono_direct);
-		//send (filtered) mono output from reverberator
-		tr_out = tr_earlies + (cdpdiff->tick(tr_earlies + tr_mono_direct) * diff_gain * 0.707);		
-		// output:: use 2 allpasses to generate stereo reverb, further diffusion
-
-		// new version
-		tr_l_out = wet_gain *  ap2_l->tick(ap1_l->tick(tr_out));
-		// old version		
-		//tr_l_out = chan_ap[0]->tick(tr_out) * wet_gain;
-
-		if(want_mono)
-			tr_l_out += (tr_mono_direct * dry_gain);
-		else			
-			tr_l_out += tr_l_direct * dry_gain;
-
-        outsamps[0] = (float) tr_l_out;
-		
-		if(outchans>=2){						
-			tr_r_out = wet_gain * ap2_r->tick(ap1_r->tick(tr_out));
-			// old version:
-			//tr_r_out = chan_ap[1]->tick(tr_out) * wet_gain;
-
-			tr_r_out += tr_r_direct * dry_gain;	   // still need this cos of input filters
-            outsamps[1] = (float) tr_r_out;
-		}
-		for(int j = 2; j < outchans; j++){			
-			tr_r_out = chan_ap[j]->tick(tr_out) * wet_gain;			
-            outsamps[j] = (float) tr_r_out;
-			
-		}
-        if(psf_sndWriteFloatFrames(ofd,outsamps,1) != 1){
-            fprintf(stderr,"rmverb: error writing output file\n"); 
-			return 1;
-        }
-		outframes++;
-		if((outframes % step)==0)
-			//inform(step,sr);
-			fprintf(stdout,"%.2f\r",(double)outframes/(double)sr);
-	}
-	fprintf(stdout,"%.2f\n",(double)outframes/(double)sr);
-	//stopwatch(0);
-    
-	peaks = (PSF_CHPEAK *) malloc(sizeof(PSF_CHPEAK) * outchans);
-    if(peaks && psf_sndReadPeaks(ofd,peaks,NULL)) {
-        printf("\nPEAK data:\n");
-        for(i=0;i < outchans;i++)
-            printf("Channel %d: %.4f at frame %u: %.4lf secs\n",i+1,
-                   peaks[i].val,peaks[i].pos, (double) peaks[i].pos / (double)props.srate);
-	}
-	else {
-        fputs("rmverb: warning: unable to read PEAK data\n",stderr);
-    }
-	
-	psf_sndClose(ifd);
-	
-	if(psf_sndClose(ofd) < 0)
-		fprintf(stderr,"rmverb: error closing outfile\n");
-    printf("\n");
-    
-    psf_finish();
-	free(peaks);
-	
-	delete tdelay;
-	
-	if(usertaps)
-		delete [] ptaps;
-	//can I call delete on base-class ptr?
-	if(s_diffuser)
-		delete s_diffuser;
-	if(m_diffuser)
-		delete m_diffuser;
-	if(l_diffuser)
-		delete l_diffuser;
-	if(lp)
-		delete lp;
-	if(outchans > 1){
-		for(i=0;i < outchans; i++)
-			delete chan_ap[i];
-		delete [] chan_ap;
-	}	
-	delete tc_l_lowcut;	
-	delete tc_r_lowcut;	
-	delete l_highcut;
-	delete r_highcut;
-	return 0;
-}
-
-
-void
-usage(void)
-{
-	fprintf(stderr,"\nrmverb: Multi-channel reverb with room simulation\n\tVersion 1.3 Release 5 CDP 1998,1999,2011");
-	fprintf(stderr,
-	"\nusage:\nrmverb [flags] infile outfile rmsize rgain mix fback absorb lpfreq trtime"
-	"\nflags    : any or all of the following"	
-	"\n   -cN   : create outfile with N channels (1 <= N <= 16 : default =2)"
-	"\n   -d    : apply double lowpass damping (internal to nested allpasses)"
-	"\n            (see <absorb>. NB: reduces reverb time: "
-	"\n                  increase feedback to compensate)"
-	"\n   -eFILE: read early reflections data from breakpoint textfile FILE"
-	"\n   -f    : force floating-point output (default: infile format)"	
-	"\n   -HN   : apply 12dB Highcut filter with cutoff freq N Hz) to infile"
-	"\n   -LN   : apply 12dB Lowcut filter with cutoff freq N Hz to infile"
-	"\n           (NB: currently fixed at 12dB/oct - could be made variable)\n"
-	"\n   -pN   : force reverb predelay to N msecs (shifts early reflections)"
-	"\nrmsize   : 1,2 or 3: small, medium or large"
-	"\nrgain    : set level of dense reverb (0.0 < egain <= 1.0)"
-	"\nmix      : dry/wet balance (source and reverb). 1.0<-- mix -->=0.0"
-	"\nfback    : reverb feedback level: controls decay time. 0.0 <= fback <= 1.0"
-	"\nabsorb   : cutoff freq (Hz) of internal lowpass filters (models air absorption)\n"
-	"                (typ: 2500 for large room --- 4200 for small room "
-	"\nlpfreq   : lowpass filter cutoff freq(Hz) applied at input to reverb"	
-	"\n           NB: to disable either filter (absorb,lpfreq), use the value 0"
-	"\ntrtime   : trailer time added to outfile for reverb tail (secs)\n");	
-	exit(0);
-}
-
-long readtaps(FILE *fp, deltap **taps,double sr)
-{
-	vector<deltap> vtaps;
-	deltap thistap;
-	char line[256];
-	int ntaps = 0;
-	int linecount = 1;
-	bool error = false;
-	double time= 0.0, current_time = 0.0, val = 0.0;
-
-	double sample_duration = 1.0 / sr;
-	if(fp==0 || sr <= 0.0){
-		*taps = 0;
-		return 0;
-	}
-	while(fgets(line,256,fp))	{
-		int got;
-		if(line[0] == '\n' || line[0]== ';'){	//allow comment lines; this does not check for leading white-space...
-			linecount++;
-			continue;
-		}
-
-		if((got = sscanf(line,"%lf%lf",&time,&val))<2){			
-			fprintf(stderr,"\nerror in reflections file: line %d",linecount);
-			error =true;
-			break;
-		}
-		if(time < 0.0){
-			fprintf(stderr,"\nerror in tapfile: line %d: bad time value",linecount);
-			error = true;
-			break;
-		}
-		//if (first) val = 0.0, or VERY close
-		if(time < sample_duration)	{ //non-zero time must be at least one sample on!			
-			fprintf(stderr,"\nWARNING: very small taptime %.4lf treated as zero",time);
-			time = 0.0;			
-		}
-
-		if(current_time != 0.0 && time==current_time){
-			fprintf(stderr,"\nWARNING: duplicate time in line %d: ignored",linecount);
-			linecount++;
-			continue;
-		}
-		if(time < current_time){
-			fprintf(stderr,"\nerror in tapfile: time out of sequence: line %d",linecount);
-			error = true;
-			break;
-		}
-		current_time = time;
-		thistap.pos = time;
-		if(val <=0.0 || val > 1.0){
-			fprintf(stderr,"\nerror: bad amplitude in tapfile: line %d",linecount);
-			error = true;
-			break;
-		}
-		thistap.val = val;
-		vtaps.push_back(thistap);
-		linecount++;
-	}
-	
-	ntaps = vtaps.size();
-	if(ntaps==0){
-		fprintf(stderr,"\ntapfile contains no data!");
-		error = true;
-	}
-	if(error){
-		*taps = 0;
-		return 0;
-	}
-	*taps = new deltap[ntaps];
-	if(taps==0)
-		return 0;
-	vector<deltap>::iterator I =  vtaps.begin(); 
-	for(int i = 0; I != vtaps.end();i++, I++){
-		(*taps)[i].pos = I->pos;
-		(*taps)[i].val = I->val;
-	}
-
-	return ntaps;
-}

dev/externals/reverb/roomresp.cpp

@@ -1,713 +0,0 @@
-/*
- * Copyright (c) 1983-2013 Richard Dobson and Composers Desktop Project Ltd
- * http://people.bath.ac.uk/masrwd
- * http://www.composersdesktop.com
- * This file is part of the CDP System.
- * The CDP System is free software; you can redistribute it
- * and/or modify it under the terms of the GNU Lesser General Public 
- * License as published by the Free Software Foundation; either 
- * version 2.1 of the License, or (at your option) any later version. 
- *
- * The CDP System is distributed in the hope that it will be useful, 
- * but WITHOUT ANY WARRANTY; without even the implied warranty of 
- * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  
- * See the GNU Lesser General Public License for more details.
- * You should have received a copy of the GNU Lesser General Public
- * License along with the CDP System; if not, write to the Free Software
- * Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
- *
- */
- 
-/***********************************************************************
-Motorola
-
-Motorola does not assume any liability ...
-Author:   Tom Zudock ([email protected])
-
-see:
- http://collaboration.cmc.ec.gc.ca/science/rpn/biblio/ddj/Website/articles/DDJ/1996/9612/9612d/9612d.htm
- 
-Presented in Dr Dobbs Journal Vol21:12, December 1996.
- 
-Filename: roomresp.cpp
-
-This program calculates the room response for a simple room.  The output is not
-particularly delightful to listen to (it could sound phasey and
-have ringing due to the simple model used), but this program is the
-foundation for more complex virtual audio models which will eliminate
-these anomalies.
-
-Example form of the InputParametersTextFile:
-
-0.5               # Input gain for data file (float)
-1                 # Open Path = 0 Closed Path = 1 (int)
-6    5    4       # Room Dimensions (X,Y,Z) (float)
-0.85              # Reflectivity of walls (float)
-5                 # Number of rooms per dimension, 1 or greater
-2    3    2       # Listener coord (X,Y,Z) (float)
-2    2    2    1  # Source coord and time at coord (X,Y,Z,t) (float)
-3    2    2    1  # Source coord and time at coord (X,Y,Z,t) (float)
-4    3    2    1  # Source coord and time at coord (X,Y,Z,t) (float)
-3    4    2    1  # Source coord and time at coord (X,Y,Z,t) (float)
-2    4    2    1  # Source coord and time at coord (X,Y,Z,t) (float)
-
-***********************************************************************/
-
-// included header files
-#include <stdio.h>
-#include <stdlib.h>
-#include <math.h>
-
-//#include <process.h>
-#include <string.h> 
-#ifdef _DEBUG
-#include <assert.h>
-#endif
-#include <vector>
-#include <algorithm>
-using namespace std;
-
-#ifndef _MAX
-#define _MAX(x,y) ((x) > (y) ? (x) : (y))
-#endif
-
-// macro defined constants
-//#define Vs 334.0
-static const double Vs = 334.0;
-static const double srate = 20000;
-#define PI 3.14159265359
-#define RefDist 1.0
-enum args {
-		REFLECTIVITY,
-		REFLECTIONS,
-		R_X,
-		R_Y,
-		R_Z,
-		S_X,
-		S_Y,
-		S_Z,
-		L_X,
-		L_Y,
-		L_Z
-};
-
-// structure for a point in 3 dimensional space
-struct point {
-	double X;
-	double Y;
-	double Z;
-	point(){
-		X = Y = Z = 0.0;
-	}
-	virtual ~point() {}
-};
-
-// structure for a point paired with a time duration
-struct position {
-	point Coord;
-	double Time;
-	position *NextSource;
-};
-
-const char* cdp_version = "5.0.0";
-
-bool input_cmds(char *args[],int argc,point &room,point &source,point &listener,double &reflcoef,int &NR);
-
-void
-CalcImpResp(
-    long *,
-    double *,
-    point &,
-    point &,
-    point &,
-    double,
-    double,
-    int);
-
-position *
-InputParameters(
-    FILE *,
-    point &,
- //   point &,
-    point &,
-    double &,
-    double &,
-   int &);
-
-void
-InputParamsError(
-    int);
-
-//RWD stuff
-typedef struct deltap {
-		long time;
-		double val;
-	} DELTAP;
-
-bool operator<(const DELTAP &a,const DELTAP &b)
-{
-	return (a.time < b.time);
-}
-
-void usage()					//RWD.1..12.98 left out flag -nNORM for now...
-{
-	printf("\n*********  ROOMRESP.EXE: by Tom Zudock  CDP build 1998,1999 ******"
-		   "\n  GENERATE ROOM-RESPONSE EARLY REFLECTIONS DATAFILE\n"
-		   "\nusage: roomresp [-aMAXAMP][-rRES] txtout.dat liveness nrefs "
-		   "\n                          roomL roomW roomH"
-		   "\n                             sourceL sourceW sourceH"
-		   "\n                                listenerL listenerW listenerH");
-	printf("\ntxtout.dat: text outfile containing early reflections in brkpoint format"
-			"\n             suitable for input to rmverb, reverb or tdelay"
-			"\nMAXAMP   : peak amp for data (0.0 < MAXAMP <= 1.0; default 1.0)"
-			"\nRES      : time resolution for reflections in msecs"
-			"\n             (0.1 <= RES <= 2; default 0.1)"
-			"\nliveness : degree of reflection from each surface (0 to 1; typ: 0.95)"
-			"\nnrefs    : number of reflections from each surface(nrefs > 0; typ: 2 to 5)"
-			"\n           (WARNING: high values will create extremely long data files!)"
-			"\nROOM PARAMETERS:"
-			"\nroomL roomW roomH            : room size: (Length, Width, Height)"
-			"\nsourceL sourceW ourceH       : position of sound source (as above)"
-			"\nlistenerL listenerW listenerH: position of listener (as above)"
-			"\n                               all dimensions are in metres"
-			"\nNB: first output time is non-zero."
-			);
-}
-
-
-/***********************************************************************
-main
-
-The main program is responsible for the primary program flow.
-It contains the primary processing loop which calls subroutines
-to handle the specialized functionality.
-***********************************************************************/
-int main (int argc, char **argv) {
-	
-//	FILE        *PositionFile = 0;  // input text parameters file
-	//RWD to print earlies as text
-	FILE		*earliesFile = 0;
-#ifdef TESTING
-	FILE		*rawfile = 0;				///just for testing...raw output from impresp funtionm
-#endif
-	int			i,NR = 0;							// num of mirrored rooms per dimension
-	long		*TapDelay = 0;			// pointer to array of tap delays	
-	double		Fs = 0.0;							// the sampling frequency
-	double		*TapGain = 0;				// pointer to array of tap gains
-	double		ReflCoef = 0.0;				// the reflectivity of the walls
-//	position	*CurrentSource = 0;	// current source in positions list 
-	point		Room;						// coords for the room size
-	point		Source;					// coords of current sound source
-	point		Listener;				// coords of the listener
-#ifdef _DEBUG
-	bool		write_c_struct = false;
-#endif
-	bool		do_normalize = true;
-	double		normfac = 1.0;
-	double      min_separation = 0.0001;
-
-	if((argc==2) && strcmp(argv[1],"--version")==0) {
-		fprintf(stdout,"%s\n",cdp_version);
-		fflush(stdout);
-		return 0;
-	}
-	if(argc==1){
-		usage();
-		exit(0);
-	}
-	if (argc < 13){
-		fprintf(stderr,"\ninsufficient arguments\n");
-  		usage();
-		exit(-1);
-	}
-  
-	while(argc > 0 && argv[1][0]=='-'){
-		double val;
-		switch(argv[1][1]){
-#ifdef _DEBUG
-//my private facility...
-		case('c'):
-			write_c_struct = true;
-			printf("\nwriting C-format file");
-			break;
-#endif
-		case('a'):
-			if(argv[1][2]=='\0'){
-				fprintf(stderr,"\n-a flag requires a value\n");
-				usage();
-				exit(1);
-			}
-			val = atof(&(argv[1][2]));
-#ifdef _DEBUG
-			if(val <0.0 || val > 1.0){
-#else
-			if(val <=0.0 || val > 1.0){
-#endif
-				fprintf(stderr,"\n normalization value %lf out of range\n",val);
-				usage();
-				exit(1);
-			}
-#ifdef _DEBUG
-			if(val==0.0) {
-				do_normalize = false;
-				normfac = 1.0;
-				printf("\nskipping normalization");
-			}
-			else
-#endif
-				normfac = val;
-			break;
-		case('r'):
-			if(argv[1][2]=='\0'){
-				fprintf(stderr,"\n-r flag requires a value\n");
-				usage();
-				exit(1);
-			}
-			val = atof(&(argv[1][2]));			
-			if(val < 0.1 || val > 2.0) {
-				fprintf(stderr,"\nvalue %.4lf for RES out of range\n",val);
-				usage();
-				exit(1);				
-			}
-			else
-				min_separation = val * 0.001;
-			break;
-
-
-
-		default:		
-			fprintf(stderr,"\nbad flag option\n");
-			exit(1);
-			break;
-		}
-		argc--;
-		argv++;
-	}
-
-	//step along arglist:
-	argc--; argv++;
-
-	
-	if(argc != 12){
-		fprintf(stderr,"\nincorrect number of numeric arguments\n");
-		usage();
-		exit(1);
-	}	
-	if((earliesFile = fopen(argv[0],"w"))==NULL){
-		printf ("\nUnable to open %s for output\n",argv[2]);
-		exit(-1);
-	}
-#ifdef TESTING
-	if((rawfile = fopen("rawdata.txt","w"))==NULL){
-		printf ("Unable to open rawdata.txt for output\n");
-		exit(-1);
-	}
-#endif
-	//step along again...
-	argc--; argv++;
-
-	//NB this sets the efective time-resolution for each tap, ie 1.0 / fs
-	//so 1000 is a minimum, for 1msec resolution
-
-	Fs = srate;
-
-	if(!input_cmds(argv,argc,Room,Source,Listener,ReflCoef,NR))
-		exit(1);	//already got the errmsg
-  
-	if((TapDelay = new long[NR*NR*NR])==NULL) {
-    	printf("Error: Unable to allocate memory\n");
-    	exit(-1);
-    }
-    if((TapGain = new double[NR*NR*NR])==NULL) {
-    	printf("Error: Unable to allocate memory\n");
-    	exit(-1);
-    }
-	
-  
-	CalcImpResp(
-	    TapDelay,
-		TapGain,
-		Room,
-		/*CurrentSource->Coord,*/Source,
-		Listener,
-		ReflCoef,
-		Fs,
-		NR);
-
-	//sort deltaps, eliminate duplicates,  write to file
-	vector<DELTAP> vtaps;
-	DELTAP thistap,prevtap;
-	prevtap.time = 0;
-	prevtap.val = 0.0;
-	for(i = 0; i < NR*NR*NR;i++) {
-		thistap.time = TapDelay[i];
-		thistap.val  =TapGain[i];
-#ifdef TESTING
-		fprintf(rawfile,"%ld\t%.6lf\n",thistap.time,thistap.val);
-#endif
-		//exclude tapval of 1.0: = direct sig at listening point
-		if(thistap.val > 0.0 && thistap.val < 1.0)			// get a problem in release build otherwise....
-			vtaps.push_back(thistap);	
-	}
-
-	//now we can sort them
-
-	sort(vtaps.begin(),vtaps.end());
-	
-	vector<DELTAP>::iterator it;
-	it = vtaps.begin();
-	prevtap.time = it->time;
-	prevtap.val =  it->val;
-
-	//print all taps as we got them, first!
-
-	double maxval = 0.0;
-	for(;it != vtaps.end();it++){
-#ifdef NOTDEF		
-		fprintf(earliesFile,"\n\t%.4lf\t%.6lf",(double)it->time / Fs,it->val);
-#endif
-		maxval = _MAX(it->val,maxval);
-		
-	}
-#ifdef _DEBUG
-	printf("\noverall maxamp = %.4lf",maxval);
-#endif
-
-	//print first line, formatted for code
-	//normalize all vals to max amplitude!
-	
-	if(do_normalize){
-		normfac  /= prevtap.val;
-#ifdef _DEBUG
-		printf("\nnormalizing by %.4lf",normfac);
-#endif
-	}
-	it = vtaps.begin();
-	
-	
-	//normfac= floor(normfac);
-#ifdef _DEBUG
-	printf("\ninitial time=  %ld, initial amp = %.4lf",it->time,it->val);
-
-	//RWD private facility!
-	if(write_c_struct){
-		fprintf(earliesFile,"\n\n{\n\t{%.4lf,%.6lf}",(double)prevtap.time / Fs,prevtap.val * normfac);
-		int count = 1;
-	
-		for(;it != vtaps.end();it++){
-			//get maxamp of any taps sharing a time (to 5 sig figs) - or should I add them somehow?
-			if(fabs(it->time - prevtap.time) < min_separation){
-				it->val = _MAX(it->val,prevtap.val);
-				continue;
-			}
-			fprintf(earliesFile,",\n\t{%.5lf,%.6lf}",(double)it->time / Fs,it->val * normfac);
-			prevtap = *it;
-			count++;
-		}
-		fprintf(earliesFile,"\n\t};\n\nNUMTAPS = %d\n",count);
-	}
-	else {
-#endif
-		int count = 1;
-		fprintf(earliesFile,"%.5lf\t%.6lf\n",(double)prevtap.time / Fs, prevtap.val * normfac);
-		for(;it != vtaps.end();it++){
-			if(fabs(it->time - prevtap.time) < min_separation){
-				it->val = _MAX(it->val,prevtap.val);
-				continue;
-			}
-			fprintf(earliesFile,"%.5lf\t%.6lf\n",(double)it->time / Fs, it->val * normfac);
-			prevtap = *it;
-			count++;
-		}
-		printf("\n\nNUMTAPS = %d\n",count);
-#ifdef _DEBUG
-	}
-#endif
-	fclose(earliesFile);
-#ifdef TESTING
-	fclose(rawfile);
-#endif
-	delete TapDelay,
-	delete TapGain,
-	printf ("done\n");
-	return(0);
-}               
-
-/***********************************************************************
-CalcImpResp
-
-This subroutine calculates the time delays (in samples) and
-attenuations for the early reflections in the room.
-***********************************************************************/
-void
-CalcImpResp(
-  long * TapDelay,
-  double * TapGain,
-  point & Room,
-  point & Source,
-  point & Listener,
-  double ReflCoef,
-  double Fs,
-  int NR)						
-{
-	double Dist = 0.0;                       // distance travelled by sound ray
-	double ReflOrd = 0.0;                    // reflection order of sound ray
-	//RWD.11.98 need volatile, or VC++ optimizer does more bad things!
-	volatile point MirrSource;                  // mirrored source x,y,z coords
-	//RWD
-	int NRovr2 = NR / 2;
-	int index = 0;
-  for (int i=-NRovr2;i<=NRovr2;i++) {    // loop through all 3 dimensions
-    for (int j=-NRovr2;j<=NRovr2;j++) {
-      for (int k=-NRovr2;k<=NRovr2;k++) {
-        // calc x,y,z sound source coords in mirrored room		  
-        MirrSource.X = (i)*Room.X
-					   +fabs(((i)%2)*Room.X)
-                       +pow(-1,(i))*Source.X;                       
-        MirrSource.Y = (j)*Room.Y
-                       +fabs(((j)%2)*Room.Y)
-                       +pow(-1,(j))*Source.Y;
-        MirrSource.Z = (k)*Room.Z
-                       +fabs(((k)%2)*Room.Z)
-                       +pow(-1,(k))*Source.Z;
-        // calculate distance to listener
-        Dist = sqrt(pow(MirrSource.X-Listener.X,2)
-                    +pow(MirrSource.Y-Listener.Y,2)
-                    +pow(MirrSource.Z-Listener.Z,2));
-        // calculate delayed arrival time of reflection
-		//RWD problem with release build...
-		
-		index = (i+NRovr2)*NR*NR+(j+NRovr2)*NR+(k+NRovr2);
-		//this also stopped the optimizer doing bad things....
-		//if(index%NR==0)
-		//	printf("\nindex = %d\tX= %.4lf\tY=%.4lf\tZ=%.4lf\tDist = %.4lf",
-		//		index,MirrSource.X,MirrSource.Y,MirrSource.Z,Dist);
-																					
-        TapDelay[index]=Dist/Vs*Fs;		
-        ReflOrd = abs(i)+abs(j)+abs(k);
-        // calculate attenuation for the reflection
-		//RWD div/zero avoidance:
-		if(Dist==0.0)
-			TapGain[index] = 1.0;
-		else
-			TapGain[index]= pow(RefDist/Dist,2.0)*pow(ReflCoef,ReflOrd);		
-      }
-    }
-  }
-}
-
-/***********************************************************************
-InputParameters
-
-This subroutine inputs parameters from a text file.  An example of
-the format of the parameters file is below.  It also checks the
-range of some of the parameters and creates a linked list representing
-how the sound source changes position in the modeled room.
-
-0.5               # Input gain for data file (float)
-1                 # Open Path = 0 Closed Path = 1 (int)
-6    5    4       # Room Dimensions (X,Y,Z) (float)
-0.85              # Reflectivity of walls (float)
-5                 # Number of rooms per dimension, 1 or greater
-2    3    2       # Listener coord (X,Y,Z) (float)
-2    2    2    1  # Source coord and time at coord (X,Y,Z,t) (float)
-3    2    2    1  # Source coord and time at coord (X,Y,Z,t) (float)
-4    3    2    1  # Source coord and time at coord (X,Y,Z,t) (float)
-3    4    2    1  # Source coord and time at coord (X,Y,Z,t) (float)
-2    4    2    1  # Source coord and time at coord (X,Y,Z,t) (float)
-
-***********************************************************************/
-position*
-InputParameters(
-  FILE *PositionFile,
-  point &Room,
-//  point &Source,
-  point &Listener,
-  double &ReflCoef,
-  double &InGain,
-  int &NR)
-{
-	position *First = new position,*Previous=First,*Next=Previous;
-	int ClosedPath,LineCount=1;
-	char StrLine[256];
-
-	#define ReadParms(arg1)                           \
-	if ((fgets(StrLine,256,PositionFile)) != NULL) {  \
-		if (arg1)                                       \
-			LineCount++;                                  \
-		else                                            \
-			InputParamsError(LineCount);                  \
-	}                                                 \
-	else                                              \
-			InputParamsError(0);
-		
-	
-#ifdef _DEBUG
-	assert(PositionFile);
-#endif
-	ReadParms(sscanf(StrLine,"%le",&InGain)==1)
-	ReadParms(sscanf(StrLine,"%d",&ClosedPath)==1)
-	ReadParms(sscanf(StrLine,"%le %le %le",&Room.X,&Room.Y,&Room.Z)==3)
-	ReadParms(sscanf(StrLine,"%le",&ReflCoef)==1)
-	ReadParms(sscanf(StrLine,"%d",&NR)==1)
-	ReadParms(sscanf(StrLine,"%le %le %le",&Listener.X,&Listener.Y,&Listener.Z)==3)
-
-	if (NR%2 != 1) {
-		printf ("Error: The number of reflected rooms per dimension must be odd (for symmetry)\n");
-		exit(-1);
-	}
-	//RWD!!! was bitwise OR:  '|'
-	if ((Listener.Y > Room.Y)||(Listener.X > Room.X)) { 					// Check listener position
-		printf ("Error: Listener position puts listener outside of room \n");
-		exit(-1);
-	}		
-  // Choose a default position for listener
-	//RWD check this! am I always getting these setings????
-	First->Coord.X = Room.X/2;
-	First->Coord.Y = Room.Y/2;
-	First->Coord.Z = Room.Z/2;
-	First->Time = 0;
-	First->NextSource=NULL;
-	while (feof(PositionFile)==0) {                                              
-		if ((fgets(StrLine,256,PositionFile)) != NULL) {
-			if(sscanf(StrLine,"%le %le %le %le",&(Next->Coord.X),&(Next->Coord.Y),&(Next->Coord.Z),&(Next->Time))==4) {
-				Next->NextSource = new position;
-				Previous = Next;
-				Next = Next->NextSource;
-				LineCount++;
-			}
-			else {
-				InputParamsError(LineCount);
-			}
-		}
-	}
-	delete (Previous->NextSource);
-	if (ClosedPath)
-		Previous->NextSource=First;
-	else
-		Previous->NextSource=NULL;
-	return(First);
-}
-
-//RWD get params from cmdline: refl,nrefs,rX,rY,rZ,sX,sY,sZ,lX,lY,lZ
-//fix ingain at 1.0
-bool input_cmds(char *args[],int argc,point &room,point &source,point &listener,double &reflcoef,int &NR)
-{
-	double dval;
-	int ival;
-#ifdef _DEBUG
-	assert(argc==11);
-#endif
-	if(argc != 11)
-		return false;
-
-	dval = atof(args[REFLECTIVITY]);
-	if(dval <= 0.0 || dval > 1.0){
-		fprintf(stderr,"\nbad value for liveliness\n");
-		return false;
-	}
-	reflcoef = dval;
-
-	ival = atoi(args[REFLECTIONS]);
-	if(ival < 1){
-		fprintf(stderr,"\nbad value for nrefs\n");
-		return false;
-	}
-
-	NR = 1 + (2* ival);
-
-	room.X = atof(args[R_X]);
-	if(room.X <= 0.0){
-		fprintf(stderr,"\nbad size for roomX\n");
-		return false;
-	}
-	room.Y = atof(args[R_Y]);
-	if(room.Y <= 0.0){
-		fprintf(stderr,"\nbad size for roomY\n");
-		return false;
-	}
-	room.Z = atof(args[R_Z]);
-	if(room.Z <= 0.0){
-		fprintf(stderr,"\nbad size for roomZ\n");
-		return false;
-	}
-	listener.X = atof(args[L_X]);
-	if(listener.X < 0.0){
-		fprintf(stderr,"\nbad size for listenerX\n");
-		return false;
-	}
-	listener.Y = atof(args[L_Y]);
-	if(listener.Y < 0.0){
-		fprintf(stderr,"\nbad size for listenerY\n");
-		return false;
-	}
-	listener.Z = atof(args[L_Z]);
-	if(listener.Z < 0.0){
-		fprintf(stderr,"\nbad size for listenerZ\n");
-		return false;
-	}
-
-	if ((listener.Y > room.Y)||(listener.X > room.X)|| (listener.Z > room.Z)) { // Check listener position
-		printf ("\nError: Listener position puts listener outside room\n");
-		return false;
-	}
-
-	source.X = atof(args[S_X]);
-	if(source.X <= 0.0){
-		fprintf(stderr,"\nbad size for sourceX\n");
-		return false;
-	}
-
-	source.Y = atof(args[S_Y]);
-	if(source.X <= 0.0){
-		fprintf(stderr,"\nbad size for sourceY\n");
-		return false;
-	}
-
-	source.Z = atof(args[S_Z]);
-	if(source.X <= 0.0){
-		fprintf(stderr,"\nbad size for sourceX\n");
-		return false;
-	}
-
-	if ((source.Y > room.Y)||(source.X > room.X)|| (source.Z > room.Z)) { // Check source position
-		printf ("\nError: Source position puts source outside room\n");
-		return false;
-	}
-
-
-	return true;
-}
-
-
-
-/***********************************************************************
-InputParamsError
-
-This subroutine outputs an text error message to standard output
-when an error is encountered in the InputParams subroutine.  The 
-error text that is output is selected by the line number of the
-input parameters file where the error occured.
-***********************************************************************/
-void
-InputParamsError(
-  int ErrNum)
-{
-	char ErrorStr[][256] = {
-	"Error: EOF reached before all parameters input\n",
-	"Error: Line %d requires 1 parameter the gain for the input file\n",
-	"Error: Line %d requires 1 parameter (0 or 1): 0=closed path, 1=open path\n",
-	"Error: Line %d requires 3 parameters (X,Y,Z): the room dimensions\n",
-	"Error: Line %d requires 1 parameter the wall refelctivity\n",
-	"Error: Line %d requires 1 parameter the number of mirrored rooms per dimension\n",
-	"Error: Line %d requires 3 parameters (X,Y,Z): the listeners position\n",
-	"Error: Line %d requires 4 parameters (X,Y,Z,t): sound position position and time\n"};
-	if (ErrNum < 7)
-		printf(ErrorStr[ErrNum],ErrNum);
-	else
-		printf(ErrorStr[7],ErrNum);
-	exit(-1);
-
-}
-
- 

dev/externals/reverb/smalldiff.cpp

@@ -1,95 +0,0 @@
-/*
- * Copyright (c) 1983-2013 Richard Dobson and Composers Desktop Project Ltd
- * http://people.bath.ac.uk/masrwd
- * http://www.composersdesktop.com
- * This file is part of the CDP System.
- * The CDP System is free software; you can redistribute it
- * and/or modify it under the terms of the GNU Lesser General Public 
- * License as published by the Free Software Foundation; either 
- * version 2.1 of the License, or (at your option) any later version. 
- *
- * The CDP System is distributed in the hope that it will be useful, 
- * but WITHOUT ANY WARRANTY; without even the implied warranty of 
- * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  
- * See the GNU Lesser General Public License for more details.
- * You should have received a copy of the GNU Lesser General Public
- * License along with the CDP System; if not, write to the Free Software
- * Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
- *
- */
- 
-//smalldiff.cpp
-//implements diffuse reverberator for small rooms, based on gardner nested allpasses
-#include "reverberator.h"
-
-small_diffuser::small_diffuser(unsigned int pre_delay, 
-							   const NESTDATA *apdata1,
-							   const NESTDATA *apdata2,double fb_gain,double lp_coeff){
-
-	  ap1_data = *apdata1;
-	  ap2_data = *apdata2;
-	  predelay_time = pre_delay;
-	  lpgain = fb_gain;
-	  lpcoeff = lp_coeff;
-	  ap1 = ap2 = 0;
-	  lp1 = 0;
-	  predelay = 0;
-	  out1 = out2 = 0.0f;
-}
-
-small_diffuser::~small_diffuser()
-{
-	delete ap1;
-	delete ap2;
-	if(predelay)
-		delete predelay;
-	delete lp1;
-}
-
-bool small_diffuser::create(void)
-{
-	if(ap1_data.time1 == 0 || ap1_data.gain1 <= 0.0)
-		return false;
-	if(ap1_data.time2 ==0 || ap1_data.gain2 <= 0.0)
-		return false;
-	if(ap2_data.time1 == 0 || ap2_data.gain1 < 0.0)
-		return false;
-	if(ap2_data.time2 ==0 || ap2_data.gain2 < 0.0)
-		return false;
-	ap1 = new nested_allpass(ap1_data.time1,ap1_data.time2,ap1_data.time3,
-							ap1_data.gain1,ap1_data.gain2,ap1_data.gain3,ap1_data.delay1,ap1_data.delay2);
-	if(!ap1->create())
-		return false;
-	if(ap2_data.gain1 != 0.0){	 //allow ap to eliminate second block
-		ap2 = new nested_allpass(ap2_data.time1,ap2_data.time2,ap2_data.time3,
-							ap2_data.gain1,ap2_data.gain2,ap2_data.gain3,ap2_data.delay1,ap2_data.delay2);
-		if(!ap2->create())
-			return false;
-	}
-	if(predelay_time > 0){
-		predelay = new delay(predelay_time,1.0);
-		if(!predelay->create())
-			return false;
-	}
-	if(lpcoeff > 0.0)
-		lp1 = new lowpass1(lpcoeff);
-	return true;
-}
-
-
-float small_diffuser::tick(float insam) 
-{
-	float ip = insam;
-	double temp;
-	if(lp1)
-		temp = lp1->tick((out2 + out1)*0.5f);
-	else
-		temp = (out2 + out1)*0.5f;
-	ip += (float)(lpgain * temp);	
-	if(predelay)
-		ip = predelay->tick(ip);
-	out1 = ap1->tick(ip);
-	if(ap2)
-		out2 = ap2->tick(out1);
-	return (out1 + out2)  /** 0.5f*/;
-}

dev/externals/reverb/tdelaymain.cpp

@@ -1,688 +0,0 @@
-/*
- * Copyright (c) 1983-2013 Richard Dobson and Composers Desktop Project Ltd
- * http://people.bath.ac.uk/masrwd
- * http://www.composersdesktop.com
- * This file is part of the CDP System.
- * The CDP System is free software; you can redistribute it
- * and/or modify it under the terms of the GNU Lesser General Public 
- * License as published by the Free Software Foundation; either 
- * version 2.1 of the License, or (at your option) any later version. 
- *
- * The CDP System is distributed in the hope that it will be useful, 
- * but WITHOUT ANY WARRANTY; without even the implied warranty of 
- * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  
- * See the GNU Lesser General Public License for more details.
- * You should have received a copy of the GNU Lesser General Public
- * License along with the CDP System; if not, write to the Free Software
- * Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
- *
- */
- 
-//tdelay.cpp
-//mono/stereo tapped delay line
-//RWD,CDP 1998
-//VERSION: initial, 0.01
-//usage: tdelay [-f] infile outfile feedback tapfile.txt [trailertime]
-//		-f : floatsam output
-//		feedback (double) - obvious...
-// tapfile.txt	lines of at least two values: time  amplitude  [pan]
-// times must be increasing, non-identical
-// amplitude +- 1.0
-// pan -1.0--1.0: -1 = hard Left, 1= hard Right, 0.0 = Centre
-#include <stdio.h>
-#include <stdlib.h>
-#include <math.h>
-#include <string.h>
-extern "C" {
-#include <sfsys.h>
-}
-#include <props.h>
-#include "reverberator.h"
-#include <vector>
-#include <algorithm>
-using namespace std;
-
-typedef struct stereo_tap {
-	double time;
-	double amp;
-	double pan;
-} STAP ;
-
-
-void usage(void);
-#define ROOT2	(1.4142136)
-enum pan_direction {SIGNAL_TO_LEFT,SIGNAL_TO_RIGHT};
-
-enum args {INFILE=1,OUTFILE,TAPGAIN,FEEDBACK,MIX,TAPFILE,TRTIME};
-
-void pancalc(double position,double *leftgain,double *rightgain);
-
-const char* cdp_version = "5.0.0";
-
-
-int main(int argc, char **argv)
-{
-	int ifd,ofd,inchans;
-	float feedback = 0.0f;
-	float dryfac = 0.0f,wetfac = 1.0f;
-	double trailertime = 0.0;
-	double current_time = 0.0;
-	deltap *l_taps = 0,*r_taps = 0;
-	tapdelay *l_dline = 0, *r_dline = 0;
-	unsigned int i,nchans = 1,ntaps = 0,trailersamps = 0;
-	bool is_stereo = false;
-	bool floatsams_wanted = false;
-	bool mix_input = true;
-	SFPROPS props;
-	FILE *tapfile = 0;
-	double sample_duration,tapgain;
-	double max_lgain = 1.0,max_rgain= 1.0;
-	float max_gain = 1.0f;
-	double sr; 
-	CHPEAK *peaks;
-	
-	if((argc==2) && strcmp(argv[1],"--version")==0) {
-		fprintf(stdout,"%s\n",cdp_version);
-		fflush(stdout);
-		return 0;
-	}
-
-	if(argc < TRTIME){
-		usage();
-		exit(1);
-	}
-	if(sflinit("tdelay")){
-		sfperror("tdelay: initialisation");
-		exit(1);
-	}
-	
-	while(argv[1][0] == '-'){
-		switch(argv[1][1]){
-		//case('s'):
-		//	is_stereo = true;
-		//	break;
-		case('f'):
-			floatsams_wanted = true;
-			break;
-		default:
-			fprintf(stderr,"\nillegal flag option %s",argv[1]);
-			usage();
-			exit(1);
-			break;
-		}
-		argv++;
-		argc--;
-	}
-	if(argc < TRTIME){
-		usage();
-		exit(1);
-	}
-
-	if((ifd = sndopenEx(argv[INFILE],0,CDP_OPEN_RDONLY)) < 0) {
-		fprintf(stderr,"reverb: cannot open input file %s: %s\n", argv[1],rsferrstr);
-		exit(1);
-	}
-	if(!snd_headread(ifd,&props)){
-		fprintf(stderr,"\nerror reading infile header: %s\n",rsferrstr);
-		sndcloseEx(ifd);
-		exit(1);
-	}
-	if(props.type != wt_wave){
-		fprintf(stderr,"\ninfile is not a waveform file");
-		sndcloseEx(ifd);
-		exit(1);
-	}
-
-	inchans = props.chans;
-	if(inchans > 2){
-		fprintf(stderr,"\nSorry - cannot process files with more than two channels!");
-		sndcloseEx(ifd);
-		exit(1);
-	}
-	sample_duration = 1.0 / props.srate;
-
-	if(floatsams_wanted)
-		props.samptype =  FLOAT32;
-	
-
-	tapgain = atof(argv[TAPGAIN]);
-	if(tapgain <= 0.0){
-		fprintf(stderr,"\nError: tapgain must be > 0.0\n");
-		exit(1);
-	}
-
-
-	feedback = (float) atof(argv[FEEDBACK]);
-	if(feedback < -1.0f || feedback > 1.0f){
-		fprintf(stderr,"\nfeedback out of range	-1.0 to +1.0");
-		usage();
-		sndcloseEx(ifd);
-		exit(1);
-	}
-
-	dryfac = (float) atof(argv[MIX]);
-	if(dryfac < 0.0f || dryfac >= 1.0f){
-		fprintf(stderr,"\nmix value %s out of range",argv[4]);
-		usage();
-		sndcloseEx(ifd);
-		exit(1);
-	}
-	wetfac = 1.0f - dryfac;
-
-	if(argc==TRTIME+1){
-		trailertime = atof(argv[TRTIME]);
-		if(trailertime < 0.0){
-			fprintf(stderr,"\ntrailertime must be >= 0.0");
-			exit(1);
-		}
-		trailersamps = (unsigned int) (trailertime * props.srate);
-	}
-	tapfile = fopen(argv[TAPFILE],"r");
-	if(!tapfile){
-		fprintf(stderr,"\nunable to open tapfile %s",argv[TAPFILE]);
-		usage();
-		sndcloseEx(ifd);
-		exit(1);
-	}
-
-	//read all lines, into taps array
-	vector<STAP> ltaps;
-	STAP thistap;
-	double this_time,this_amp,this_pan;
-	char line[255];
-
-	int linecount = 1;
-	this_time = 0.0; this_amp = 0.0;
-	bool read_error = false;
-	vector<STAP>::iterator I;
-
-	//RWD TODO: take a tap at time = 0.0 as spec of direct sound : amp and pan
-	
-	while(fgets(line,255,tapfile) != NULL){
-		this_pan = 0.0;
-		int got;
-		if(line[0] == '\n' || line[0]== ';'){	//allow comment lines; this does not check for leading white-space...
-			linecount++;
-			continue;
-		}
-		if((got = sscanf(line,"%lf%lf%lf",&this_time,&this_amp,&this_pan)) < 2){
-			fprintf(stderr,"\nerror in tapfile: line %d: insufficient parameters",linecount);
-			read_error = true;
-			break;
-		}
-		//time= 0 no good for a taptime!
-		if(this_time < 0.0){
-			fprintf(stderr,"\nerror in tapfile: line %d: bad time value",linecount);
-			read_error = true;
-			break;
-		}
-		if(this_time < sample_duration)	{ //non-zero time must be at least one sample on!			
-			fprintf(stderr,"\nWARNING: very small taptime %.4lf treated as zero - ignoring mix value",this_time);
-			this_time = 0.0;
-			mix_input = false;
-		}
-		if(current_time != 0.0 && this_time==current_time){
-			fprintf(stderr,"\nWARNING: duplicate time in line %d: ignored",linecount);
-			linecount++;
-			continue;
-		}		
-		if(this_time < current_time){
-			fprintf(stderr,"\nerror in tapfile: time out of sequence: line %d",linecount);
-			read_error = true;
-			break;
-		}
-		current_time = this_time;
-		thistap.time = this_time;
-		if(this_amp < 0.0 || this_amp > 1.0){
-			fprintf(stderr,"\nerror in tapfile: line %d: bad amplitude value",linecount);
-			read_error = true;
-			break;
-		}
-		
-		thistap.amp = this_amp;
-		if(got==3){
-			thistap.pan = this_pan;
-			is_stereo = true;
-			nchans = 2;
-		}
-		else
-			thistap.pan = 0.0;
-		ltaps.push_back(thistap);
-		linecount++;
-	}
-
-	if(read_error){
-		sndcloseEx(ifd);
-		exit(1);
-	}
-	ntaps = ltaps.size();
-	if(ntaps==0){
-		printf("\nfile contains no data!");
-		sndcloseEx(ifd);
-		exit(1);
-	}
-	
-	if(!mix_input)
-		printf("\nzero taptime used: taking input mix control from tapfile");
-#ifdef _DEBUG
-	printf("\nread %d taps",ntaps);
-	
-	for(I = ltaps.begin(); I != ltaps.end();I++)
-		printf("\n%.4lf\t%.4lf\t%.4lf",I->time,I->amp,I->pan);	//or whatever.....
-
-	printf("\nfeedback =  %.4f",feedback);
-	printf("\ndryfac = %.4f, wetfac = %.4f",dryfac,wetfac);
-
-#endif
-	if(is_stereo)
-		printf("\ncreating stereo outfile");
-	else
-		printf("\ncreating mono outfile");
-	
-	
-	l_taps = new deltap[ntaps];
-	
-	if(is_stereo){
-		r_taps = new deltap[ntaps];
-		//and create stereo delay-lines
-		int j;
-		for(j = 0,I = ltaps.begin(); I != ltaps.end();j++,I++){
-		//printf("\n%.4lf\t%.4lf\t%.4lf",I->time,I->amp,I->pan);	//or whatever.....
-			double l_gain,r_gain;
-			pancalc(I->pan,&l_gain,&r_gain);
-			l_taps[j].pos = r_taps[j].pos = I->time;
-			l_taps[j].val = I->amp * l_gain;
-			r_taps[j].val = I->amp * r_gain;
-		}
-	}
-	//else create a mono delayline
-	else {
-		int j;
-		for(j = 0,I = ltaps.begin(); I != ltaps.end();j++,I++){				
-			l_taps[j].pos = I->time;
-			l_taps[j].val = I->amp;
-		}
-	}
-
-
-	l_dline  = new tapdelay(l_taps,ntaps,props.srate);
-	if(!l_dline->create()){
-		fputs("\nunable to  create delay line - probably no memory",stderr);
-		delete [] l_taps;
-		sndcloseEx(ifd);
-		exit(1);
-	}
-	max_lgain = l_dline->get_maxgain();
-	if(is_stereo){
-		r_dline = new tapdelay(r_taps,ntaps,props.srate);
-		if(!r_dline->create()){
-			fputs("\nunable to  create stereo delay line - probably no memory",stderr);
-			delete [] l_taps;
-			delete [] r_taps;
-			delete l_dline;
-			sndcloseEx(ifd);
-			exit(1);
-		}
-		max_rgain = r_dline->get_maxgain();
-	}
-	if(is_stereo)
-		max_gain = (float) min(max_lgain,max_rgain);
-	else
-		max_gain = (float) max_lgain;
-
-	/* recale max_gain by user param, eg:*/
-	max_gain *= (float) tapgain;
-
-
-	l_dline->set_gain((double)max_gain);
-	if(is_stereo)
-		r_dline->set_gain((double)max_gain);
-
-
-	props.chans = nchans;
-	peaks = (CHPEAK *) malloc(sizeof(CHPEAK) * nchans);
-	if(peaks==NULL){
-		fputs("No memory for PEAK data",stderr);
-		exit(1);
-	}
-	for(i=0; i < nchans; i++){
-		peaks[i].value = 0.0f;
-		peaks[i].position = 0;
-	}
-	//OK, now open outfile!
-
-	ofd = sndcreat_ex(argv[OUTFILE], -1,&props,SFILE_CDP,CDP_CREATE_NORMAL); 
-	//ofd = sndcreat_formatted(argv[OUTFILE],-1,outsamp_type,nchans,props.srate,CDP_CREATE_NORMAL);
-	if(ofd < 0){
-		printf("\nunable to open outfile %s",argv[OUTFILE]);
-		delete [] l_taps;
-		delete l_dline;
-		if(r_taps)
-			delete [] r_taps;
-		sndcloseEx(ifd);
-		exit(1);
-	}
-	printf("\n");
-
-	float l_op = 0.0f;		  
-	float r_op = 0.0f;
-	float l_out = 0.0f;
-	float r_out = 0.0f;
-	long outsamps = 0;
-	long step = (long) (0.25 * props.srate);
-	//RWD.10.98 BIG TODO: optimize all this!
-	sr = (double) props.srate;
-	if(is_stereo){
-		for(;;){
-			int rv;
-			float ip,l_ip,r_ip;										
-			if((rv = fgetfloatEx(&l_ip,ifd,0)) < 0){ 
-					fprintf(stderr,"\nerror reading infile data");
-					delete [] l_taps;
-					delete l_dline;
-					if(r_taps)
-						delete [] r_taps;
-					sndcloseEx(ifd);
-					exit(1);				
-			}
-			if(!rv) 
-				break;
-
-			ip = l_ip;
-			if(inchans==2){
-				//mix stereo ip to mono for delay purposers
-				if((rv = fgetfloatEx(&r_ip,ifd,0)) < 0){ 
-					fprintf(stderr,"\nerror reading infile data");
-					delete [] l_taps;
-					delete l_dline;
-					if(r_taps)
-						delete [] r_taps;
-					sndcloseEx(ifd);
-					exit(1);				
-				}
-				if(!rv) 
-				break;
-				ip = (l_ip + r_ip) * 0.5f;
-			}
-			
-		
-			
-			if(feedback != 0.0f) {
-				l_op = l_dline->tick(ip + (l_op * feedback));
-				r_op = r_dline->tick(ip + (r_op * feedback));
-			}
-			else{
-				l_op = l_dline->tick(ip);
-				r_op = r_dline->tick(ip);
-			}
-
-			//mix mono or stereo input with panned output:
-			if(mix_input){
-				l_out = l_op * wetfac + l_ip * dryfac;
-				r_out = r_op * wetfac;
-				if(inchans==2)
-					r_out  += r_ip * dryfac;
-				else
-					r_out += l_ip * dryfac;
-			}
-			else{
-				l_out = l_op;
-				r_out = r_op;
-			}
-			/* rescale */
-			l_out *=  max_gain;
-			r_out *=  max_gain;
-			if((float)fabs((double)l_out) > peaks[0].value) {
-				peaks[0].value = (float)fabs((double)l_out);
-				peaks[0].position = outsamps;
-			}
-			if((float)fabs((double)r_out) > peaks[1].value) {
-				peaks[1].value = (float)fabs((double)r_out);
-				peaks[1].position = outsamps;
-			}
-
-
-			if(fputfloatEx(&l_out,ofd) < 1){
-				fprintf(stderr,"\nerror writing to outfile: %s",sferrstr());
-				delete [] l_taps;
-				delete l_dline;
-				if(r_taps)
-					delete [] r_taps;
-				sndcloseEx(ifd);
-				exit(1);
-			}
-			if(fputfloatEx(&r_out,ofd) < 1){
-				fprintf(stderr,"\nerror writing to outfile: %s",sferrstr());
-				delete [] l_taps;
-				delete l_dline;
-				if(r_taps)
-					delete [] r_taps;
-				sndcloseEx(ifd);
-				exit(1);
-			}
-			outsamps++;
-			if((outsamps % step) == 0)
-				//inform(step,props.srate);
-				fprintf(stdout,"%.2f\r",(double)outsamps/sr);
-		}
-		for(i = 0; i < trailersamps;i++){
-			if(feedback != 0.0f){
-				l_op = l_dline->tick(feedback * l_op);
-				r_op = r_dline->tick(feedback * r_op);
-			}
-			else {
-				l_op = l_dline->tick(0.0f);
-				r_op = r_dline->tick(0.0f);
-			}
-			if(mix_input){			
-				l_out = l_op * wetfac;
-				r_out = r_op * wetfac;
-			}
-			else {
-				l_out = l_op;
-				r_out = r_op;
-			}
-			/* rescale */
-			l_out *=  max_gain;
-			r_out *=  max_gain;
-			if((float)fabs((double)l_out) > peaks[0].value) {
-				peaks[0].value = (float)fabs((double)l_out);
-				peaks[0].position = outsamps;
-			}
-			if((float)fabs((double)r_out) > peaks[1].value) {
-				peaks[1].value = (float)fabs((double)r_out);
-				peaks[1].position = outsamps;
-			}
-			if(fputfloatEx(&l_out,ofd) < 1){
-				fprintf(stderr,"\nerror writing to outfile: %s",sferrstr());
-				delete [] l_taps;
-				delete l_dline;
-				if(r_taps)
-					delete [] r_taps;
-				sndcloseEx(ifd);
-				exit(1);
-			}
-			if(fputfloatEx(&r_out,ofd) < 1){
-				fprintf(stderr,"\nerror writing to outfile: %s",sferrstr());
-				delete [] l_taps;
-				delete l_dline;
-				if(r_taps)
-					delete [] r_taps;
-				sndcloseEx(ifd);
-				exit(1);
-			}
-			outsamps++;
-			if((outsamps % step) == 0)
-				//inform(step,props.srate);
-				fprintf(stdout,"%.2f\r",(double)outsamps/sr);
-		}
-	}
-
-	else{
-		for(;;){
-			int rv;
-			float ip,l_ip,r_ip;			
-			if((rv = fgetfloatEx(&l_ip,ifd,0)) < 0){ 
-				fprintf(stderr,"\nerror reading infile data");
-				delete [] l_taps;
-				delete l_dline;
-				sndcloseEx(ifd);
-				exit(1);
-
-			}
-			if(!rv) 
-				break;
-			if(inchans==2){
-				//mix stereo ip to mono for delay purposers
-				if((rv = fgetfloatEx(&r_ip,ifd,0)) < 0){ 
-					fprintf(stderr,"\nerror reading infile data");
-					delete [] l_taps;
-					delete l_dline;
-					sndcloseEx(ifd);
-					exit(1);				
-				}
-				if(!rv) 
-				break;
-				ip = (l_ip + r_ip) * 0.5f;
-			}
-			else
-				ip = l_ip;
-			if(feedback != 0.0f)
-				l_op = l_dline->tick(ip + l_op * feedback);
-			else
-				l_op = l_dline->tick(ip);
-
-			if(mix_input)
-				l_out = l_op * wetfac + l_ip * dryfac;
-			else
-				l_out = l_op;
-			/* rescale */
-			l_out *=  max_gain;
-			if((float)fabs((double)l_out) > peaks[0].value) {
-				peaks[0].value = (float)fabs((double)l_out);
-				peaks[0].position = outsamps;
-			}
-			
-			if(fputfloatEx(&l_out,ofd) < 1){
-				fprintf(stderr,"\nerror writing to outfile");
-				delete [] l_taps;
-				delete l_dline;
-				sndcloseEx(ifd);
-				exit(1);
-			}
-			outsamps++;
-			if((outsamps % step) == 0)
-				//inform(step,props.srate);
-				fprintf(stdout,"%.2f\r",(double)outsamps/sr);
-		}
-		for(i = 0; i < trailersamps;i++){						
-			l_op = l_dline->tick(feedback * l_op);
-			if(mix_input)
-				l_out = l_op * wetfac;
-			else
-				l_out = l_op;
-			if((float)fabs((double)l_out) > peaks[0].value) {
-				peaks[0].value = (float)fabs((double)l_out);
-				peaks[0].position = outsamps;
-			}
-			if(fputfloatEx(&l_out,ofd) < 1){
-				fprintf(stderr,"\nerror writing to outfile");
-				delete [] l_taps;
-				delete l_dline;
-				sndcloseEx(ifd);
-				exit(1);
-			}
-			outsamps++;
-			if((outsamps % step) == 0)
-				//inform(step,props.srate);
-				fprintf(stdout,"%.2f\r",(double)outsamps/sr);
-		}
-	}
-	fprintf(stdout,"%.2f\n",(double)outsamps/sr);
-	printf("\nPEAK data:\n");
-	for(i=0;i < nchans;i++)
-		printf("Channel %u: %.4f at frame %d: %.4lf secs\n",i+1,
-			peaks[i].value,peaks[i].position, (double) peaks[i].position / (double)props.srate);
-	sndputpeaks(ofd,nchans,peaks);
-	if(sndcloseEx(ofd) < 0){
-		fprintf(stderr,"\nwarning: error closing outfile");
-
-	}
-	delete [] l_taps;
-	delete l_dline;
-	if(is_stereo) {
-		delete [] r_taps;
-		delete r_dline;
-	}
-	
-	sndcloseEx(ifd);
-	return 0;
-}
-
-	
-void usage(void)
-{
-	printf("\n*******  STEREO MULTI-TAPPED DELAY WITH PANNING v1.0 1998 : CDP Release 4 ******\n");
-	printf("\nusage:\n\ttapdelay [-f] infile outfile tapgain feedback mix taps.txt [trailtime]\n");
-	printf("\n\t-f\t\twrite floating-point outfile");
-	printf("\n\t\t\t(default: outfile has same format as infile)");
-	printf("\n\ttapgain\t\tgain factor applied to output from delayline");
-	printf("\n\t\t\t(tapgain > 0.0, typ 0.25)");
-	printf("\n\tfeedback\trange: -1 <= feeedback <= 1.0");
-	printf("\n\tmix\t\tproportion of source mixed with delay output.\n\t\t\trange: 0.0 <= mix < 1.0");
-	printf("\n\ttrailtime\textra time in seconds (beyond infile dur) ");
-	printf("\n\t\t\t  for delays to play out.");
-	printf("\n\tStereo inputs are mixed to mono at input to the delay line.\n");
-	printf("\n\ttaps.txt consists of list of taps in the format:");
-	printf("\n\ttime amp [pan]");
-	printf("\n\n\tAll values are floating-point, one tap per line.");
-	printf("\n\tTimes (seconds) must be increasing. Duplicate times are ignored.");
-	printf("\n\tA zero time (no delay) overrides the mix parameter,");
-	printf("\n\t  and determines the level and pan of the (mono) input.");	
-	printf("\n\tAmp values must be in the range 0.0 to 1.0");
-	printf("\n\tEmpty lines are permitted, as are lines starting with  a semni-colon,");	
-	printf("\n\t  which may be used for comments.");
-	printf("\n\tIf a pan value is used in any line, outfile will be stereo.");
-	printf("\n\tPan values are nominally in the range -1 to +1: 0 = centre (mono)");
-	printf("\n\t within which constant-power panning is used.");
-	printf("\n\tValues beyond these limits result in attenuation according to the");
-	printf("\n\t   inverse-square law, to suggest distance beyond the speakers.\n");
-	
-}
-
-//TW's constant-power pan routine, even does inv-square reductions beyond the speakers
-void pancalc(double position,double *leftgain,double *rightgain)
-{
-	int dirflag;
-	double temp;
-	double relpos;
-	double reldist, invsquare;
-
-	if(position < 0.0)
-		dirflag = SIGNAL_TO_LEFT;		/* signal on left */
-	else
-		dirflag = SIGNAL_TO_RIGHT;
-
-	if(position < 0) 
-		relpos = -position;
-	else 
-		relpos = position;
-	if(relpos <= 1.0){		/* between the speakers */
-		temp = 1.0 + (relpos * relpos);
-		reldist = ROOT2 / sqrt(temp);
-		temp = (position + 1.0) / 2.0;
-		*rightgain = temp * reldist;
-		*leftgain = (1.0 - temp ) * reldist;
-	} else {				/* outside the speakers */
-		temp = (relpos * relpos) + 1.0;			 /*RWD: NB same in both cases! */
-		reldist  = sqrt(temp) / ROOT2;   /* relative distance to source */
-		invsquare = 1.0 / (reldist * reldist);
-		if(dirflag == SIGNAL_TO_LEFT){
-			*leftgain = invsquare;
-			*rightgain = 0.0;
-		} else {   /* SIGNAL_TO_RIGHT */
-			*rightgain = invsquare;
-			*leftgain = 0;
-		}
-	}
-}

dev/externals/reverb/wavetable.cpp

@@ -1,809 +0,0 @@
-/*
- * Copyright (c) 1983-2013 Richard Dobson and Composers Desktop Project Ltd
- * http://people.bath.ac.uk/masrwd
- * http://www.composersdesktop.com
- * This file is part of the CDP System.
- * The CDP System is free software; you can redistribute it
- * and/or modify it under the terms of the GNU Lesser General Public 
- * License as published by the Free Software Foundation; either 
- * version 2.1 of the License, or (at your option) any later version. 
- *
- * The CDP System is distributed in the hope that it will be useful, 
- * but WITHOUT ANY WARRANTY; without even the implied warranty of 
- * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  
- * See the GNU Lesser General Public License for more details.
- * You should have received a copy of the GNU Lesser General Public
- * License along with the CDP System; if not, write to the Free Software
- * Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
- *
- */
- 
-// Jan 2009  extended random lfo to support gaussian distribution
-
-
-//////////////////////////////////////////////////////////////////////
-
-#include <math.h>
-#include <memory.h>
-#include <time.h>
-#include "wavetable.h"
-#include <assert.h>
-
-//////////////////////////////////////////////////////////////////////
-// Construction/Destruction
-//////////////////////////////////////////////////////////////////////
-#ifndef PI
-#define PI (3.141592634)
-#endif
-
-#ifndef TWOPI
-#define TWOPI (2.0 * PI)
-#endif
-
-#define LAMBDA (10.0)
-
-
-/* rand stuff from Csound*/
-static const long RANDINT_MAX = 2147483647L;        // 2^^31 - 1 */
-static const long BIPOLAR = 0x4fffffff;
-static const long RIA = 16807;
-static const long RIM = 0x7fffffffL;
-static const double dv2_31 = 1.0 / 2147483648.0;
-static const long MAXLEN = 0x1000000L;
-static const long PHMASK = 0x0ffffffL;
-static const double FMAXLEN = (double)(0x1000000L);
-// for tpdf - bit of a hack, but seems close enough!
-static const long TPDFSHIFT = 0x18000000;
-// for gauss, trial and error!
-static const long BISHIFT = 0x16000000;
-
-
-
-fastlfo::fastlfo()
-{
-	param.freq = 0.0;
-	param.modrange = 0.0;
-	m_cur_WaveType = LFO_SINE;
-	curfreq = param.freq = 0.0;
-	curphase = 0.0;
-	incr = 0.0;
-	tf = 0;
-	kicvt = 0;
-	offset= 0.0;
-	b_sync = false;
-	tpdf = false;
-    gauss = false;
-    /* for csrand */
-    csseed = 12345;
-}
-
-double      fastlfo::csrand()  
-{ 
-    long rval;
-    rval =  randint31(csseed);
-    csseed = rval;
-    return (double) rval / (double) RANDINT_MAX;
-}
-
-
-long fastlfo::init(double srate, double normphase, long seedval, unsigned long ksmps)
-{
-	long seed;
-
-	m_srate = srate / ksmps;
-	twopiovrsr = TWOPI / m_srate;	
-	m_inv_srate = 1.0 / m_srate;
-	set_WaveType(LFO_SINE);
-	phs = (long)(MAXLEN * normphase);
-	curphase = TWOPI * normphase;
-	if(seedval == 0)
-		seed = (long) time(NULL);
-	else
-		seed = seedval;
-	rand = randint31(seed);
-	rand = randint31(rand);	
-    num1 = 0;
-    // get 2nd randon val for diff calc
-    // TODO: check this! why << 1 ?
-	rand = randint31(rand);
-	//num2 = (double)(rand<<1) * dv2_31;
-    num2 = (double)((long)((unsigned)rand<<1)-BIPOLAR) * dv2_31;
-
-	dfdmax = (num2 - num1) / FMAXLEN;
-	kicvt = (long)((double)MAXLEN / m_srate);
-	lastval = 0.0;
-	ksamps = ksmps;
-	if(ksamps==0)
-		ksamps++;
-	kcount = 1;
-	b_sync = false;
-	curphs = (long)(normphase * MAXLEN);
-	return seed;
-}
-
-
-// we keep current wavetype; no reset of random values ;
-/* phase ranged 0 -1 */
-void fastlfo::reset(double phase)
-{ 
-	curphase  = TWOPI * phase;
-	phs =  (long)(MAXLEN * phase);
-	curphs = phs; // assume phase offset = 0 here? 
-	lastval = 0.0;
-	b_sync = false;
-	kcount = 1;
-}
-
-
-
-#define OSC_WRAPPHASE  if(curphase >= TWOPI) curphase -= TWOPI;	\
-						if(curphase < 0.0) curphase += TWOPI
-// input range 0-1
-void fastlfo::sync_phase(double phase,double phaseoffset,double phaseincr)
-{
-	if(b_sync) {
-		curphase = phase * TWOPI;
-		offset = phaseoffset * TWOPI;
-		incr = phaseincr * TWOPI;
-		phs_incr = (long)(phaseincr*ksamps * MAXLEN);
-		phs = (long)(phase * MAXLEN);
-		phs &= PHMASK;
-		phs_offset = (long)(phaseoffset * MAXLEN);
-		curphs = phs + phs_offset;
-		curphs &= PHMASK;
-	}
-}
-
-inline double fastlfo::sinetick(void)
-{
-	double val;
-	double thisphase = curphase + offset;	
-	while(thisphase >= TWOPI)
-		thisphase -= TWOPI;
-	if(kcount==ksamps){
-		if(!b_sync){
-			val = sin(thisphase);
-			if(curfreq != param.freq){
-				curfreq = param.freq;
-				incr = twopiovrsr * param.freq;		
-			}
-		}
-		else
-			val = sin(thisphase);
-		curphase += incr;
-		//OSC_WRAPPHASE;
-		if(curphase >= TWOPI) 
-			curphase -= TWOPI;	
-		if(curphase < 0.0) 
-			curphase += TWOPI;
-		lastval = val * param.modrange;
-		kcount = 1;
-	}
-	else
-		kcount++;
-	return lastval;
-}
-
-inline double fastlfo::tritick(void)
-{
-	double val;
-	double thisphase = curphase + offset;	
-	while(thisphase >= TWOPI)
-		thisphase -= TWOPI;
-	if(kcount==ksamps){
-		if(!b_sync){
-			if(curfreq != param.freq){
-				curfreq = param.freq;
-				incr = twopiovrsr * param.freq;		
-			}
-		}
-		if(thisphase <= PI)	{
-			val =   (4.0 * (thisphase * (1.0 / TWOPI) )) - 1.0;
-		}
-		else {
-			val =  3.0 - 4.0 * (thisphase * (1.0 / TWOPI) );
-		}
-		curphase += incr;
-		OSC_WRAPPHASE;
-		lastval = val * param.modrange;
-		kcount = 1;
-	}
-	else
-		kcount++;
-	return 	lastval;
-	
-}
-
-inline double fastlfo::squaretick(void)
-{
-	double val;
-	double thisphase = curphase + offset;	
-	while(thisphase >= TWOPI)
-		thisphase -= TWOPI;
-	if(kcount==ksamps){
-		if(!b_sync){
-			if(curfreq != param.freq){
-				curfreq = param.freq;
-				incr = twopiovrsr * param.freq;		
-			}
-		}
-		if(thisphase <= PI)
-			val = 1.0;
-		else
-			val = -1;
-		curphase += incr;
-		OSC_WRAPPHASE;
-		lastval = val * param.modrange;
-		kcount = 1;
-	}
-	else
-		kcount++;
-	return 	lastval;
-	
-}
-
-double fastlfo::sawuptick(void)
-{
-	double val;
-	double thisphase = curphase + offset;	
-	while(thisphase >= TWOPI)
-		thisphase -= TWOPI;
-
-	if(kcount==ksamps){
-		if(!b_sync){
-			if(curfreq != param.freq){
-				curfreq = param.freq;
-				incr = twopiovrsr * param.freq;		
-			}
-		}
-		val =  (2.0 * (thisphase * (1.0 / TWOPI) )) - 1.0;
-		curphase += incr;
-		OSC_WRAPPHASE;
-		lastval = val * param.modrange;
-		kcount = 1;
-	}
-	else
-		kcount++;
-	return 	lastval;
-	
-}
-
-inline double fastlfo::sawdowntick(void)
-{
-	double val;
-	double thisphase = curphase + offset;	
-	while(thisphase >= TWOPI)
-		thisphase -= TWOPI;
-
-	if(kcount==ksamps){
-		if(!b_sync){
-			if(curfreq != param.freq){
-				curfreq = param.freq;
-				incr = twopiovrsr * param.freq;		
-			}
-		}
-		val =  1.0 - 2.0 * (thisphase * (1.0 / TWOPI) );
-		curphase += incr;
-		OSC_WRAPPHASE;
-		lastval = val * param.modrange;
-		kcount = 1;
-	}
-	else
-		kcount++;
-	return 	lastval;
-}
-
-
-// some redundant code with phs and curphs here, but keep both for now!
-
-
-inline double fastlfo::randomtick(void)
-{
-	double val;
-	double dval = 0.0;
-    int i;
-
-	if(kcount==ksamps){
-		if(b_sync){
-			val = num1 +(double)(curphs * dfdmax);
-			phs += phs_incr;
-			curphs += phs_incr;
-			if(curphs >= MAXLEN){
-				long r =  randint31(rand);
-				if(tpdf){
-					long rr = randint31(r);
-                    long ri;
-                    unsigned long sum = r + rr;
-                    ri = (long)(sum/2);                       
-                    r =  ri - TPDFSHIFT;
-				}
-                else if(gauss){
-                    int nrands = 16;
-                    long ri = 0;
-                    long rr[16];
-                    rr[0] = rand;
-                    for(i = 1; i < nrands; i++)
-                        rr[i] = randint31(rr[i-1]);
-                    for(i=0;i < nrands;i++)
-                        ri += rr[i]>>4;                        
-                    r = ri - BISHIFT;    
-                }
-                else if(biexp){
-                    long rr = r;
-                    double dbiexp = 0.0;
-                                    
-                    dbiexp = 2.0 * (double)rr * dv2_31;
-                    while(dbiexp==0.0 || dbiexp == 2.0){                    
-                        rr = randint31(rr);                                            
-                    }                
-                    if(dbiexp > 1.0)
-                        dval = -log(2.0-dbiexp) / LAMBDA;
-                    if(dbiexp < 1.0)
-                        dval = log(dbiexp) / LAMBDA;
-                    if( dval > 0.9)
-                        dval = 0.9;
-                    if(dval < -0.9)
-                        dval = -0.9;                                        
-                    r =   ((unsigned long)(dval/dv2_31) +BIPOLAR)>>1;                                       
-                }
-				curphs &= PHMASK;
-				rand = r;
-				num1 = num2;
-				num2 = (double)((long)((unsigned)r<<1)-BIPOLAR) * dv2_31;
-				dfdmax = (num2-num1) / FMAXLEN;
-			}
-			if(phs >= MAXLEN)
-				phs &= PHMASK;
-		}
-		else{           
-            val = num1 +(double)(phs * dfdmax);
-            phs +=(long) (param.freq * kicvt);
-            if(phs >= MAXLEN){
-                long r =  randint31(rand);
-                assert(r>0);
-                if(tpdf){
-                    long rr = randint31(r);
-                    long ri;
-                    unsigned long sum = r + rr;
-                    ri = (long)(sum/2);                   
-                    r =  ri - TPDFSHIFT;                     
-                }
-                else if(gauss){
-                    int nrands = 16;
-                    long ri = 0;
-                    long rr[16];
-                    rr[0] = rand;
-                    for(i = 1;i < nrands;i++)
-                        rr[i]= randint31(rr[i-1]);
-                    for(i=0;i < nrands;i++)
-                        ri += rr[i]>>4;                        
-                    r = ri - BISHIFT;                        
-                }
-                else if(biexp){
-                    long rr = r;
-                    double dbiexp = 0.0;
-                                    
-                    dbiexp = 2.0 * (double)rr * dv2_31;
-                    while(dbiexp==0.0 || dbiexp == 2.0){                    
-                        rr = randint31(rr);                                            
-                    }                
-                    if(dbiexp > 1.0)
-                        dval = -log(2.0-dbiexp) / LAMBDA;
-                    if(dbiexp < 1.0)
-                        dval = log(dbiexp) / LAMBDA;
-                    if( dval > 0.9)
-                        dval = 0.9;
-                    if(dval < -0.9)
-                        dval = -0.9;                                        
-                    r =   ((unsigned long)(dval/dv2_31) +BIPOLAR)>>1;
-                    /* assert(r>0);
-                      assert(r < RANDINT_MAX);
-                    */
-                }            
-                phs &= PHMASK;
-                rand = r;
-                num1 = num2;
-                num2 = (double)((long)((unsigned)r<<1)-BIPOLAR) * dv2_31;
-                /* NB makes uniform rand not really uniform, more a sort of rounded distr
-                 * so maybe not ideal for strict plain uniform distr - still while though!
-                */
-                dfdmax = (num2-num1) / FMAXLEN;                            
-            }
-		}
-        if(gauss)
-            val *= 1.85;
-		lastval = val * param.modrange;
-		kcount = 1;
-	}
-	else
-		kcount++;
-	return 	lastval;
-}
-
-inline double fastlfo::rand_tpdf_tick(void)
-{
-	double val;
-
-	if(kcount==ksamps){
-		if(b_sync){
-			val = num1 +(double)(curphs * dfdmax);
-			phs += phs_incr;
-			curphs += phs_incr;
-			if(curphs >= MAXLEN){
-				long r =  randint31(rand);
-				
-				long rr = randint31(r);
-                long ri;
-                long sum = r + rr;
-                ri = (long)(sum/2);                       
-                r =  ri - TPDFSHIFT;
-				                              
-				curphs &= PHMASK;
-				rand = r;
-				num1 = num2;
-				num2 = (double)((long)((unsigned)r<<1)-BIPOLAR) * dv2_31;
-				dfdmax = (num2-num1) / FMAXLEN;
-			}
-			if(phs >= MAXLEN)
-				phs &= PHMASK;
-		}
-		else{           
-            val = num1 +(double)(phs * dfdmax);
-            /* need this test for audiorate noise; but is it the right test for everywhere? */
-            /* time-varying noise rate? */
-            if(ksamps > 1)
-                phs +=(long) (param.freq * kicvt + 0.5);
-            else
-                phs = MAXLEN;
-            if(phs >= MAXLEN){
-                long r =  randint31(rand);                                
-                long rr = randint31(r);
-                long rsum;
-#ifdef _DEBUG
-                double d1,d2,dtpdf;
-                d1 = ((2.0 *(double)r)  / (double) RANDINT_MAX) - 1.0;
-                d2 = ((2.0 *(double)rr) / (double) RANDINT_MAX) - 1.0;
-                dtpdf = d1 + d2;
-                dtpdf *= 0.5;
-#endif
-                rand = rr;
-                /* bipolar value  =  2*r - 1  or 2* (r - 0.5)*/
-                long bipolard2 = 0x40000000;
-                r  = (r - bipolard2);
-                rr = (rr- bipolard2);
-                rsum = r + rr;
-                double dsum = (double)rsum  / (double) RANDINT_MAX ;                                                                                          
-                phs &= PHMASK; 
-                num1 = num2;
-                num2 = dsum;
-                dfdmax = (num2-num1) / FMAXLEN;                            
-            }
-		}        
-		lastval = val * param.modrange;
-		kcount = 1;
-	}
-	else
-		kcount++;
-	return 	lastval;
-}
-
-inline double fastlfo::rand_gauss_tick(void)
-{
-	double val;
-    int i;
-
-	if(kcount==ksamps){
-		if(b_sync){
-			val = num1 +(double)(curphs * dfdmax);
-			phs += phs_incr;
-			curphs += phs_incr;
-			if(curphs >= MAXLEN){
-				long r =  randint31(rand);				                
-                int nrands = 16;
-                long ri = 0;
-                long rr[16];
-
-                rr[0] = rand;
-                for(i = 1; i < nrands; i++)
-                    rr[i] = randint31(rr[i-1]);
-                for(i=0;i < nrands;i++)
-                    ri += rr[i]>>4;                        
-                r = ri - BISHIFT;    
-                                
-				curphs &= PHMASK;
-				rand = r;
-				num1 = num2;
-				num2 = (double)((long)((unsigned)r<<1)-BIPOLAR) * dv2_31;
-				dfdmax = (num2-num1) / FMAXLEN;
-			}
-			if(phs >= MAXLEN)
-				phs &= PHMASK;
-		}
-		else{           
-            val = num1 +(double)(phs * dfdmax);
-            if(ksamps > 1)
-                phs +=(long) (param.freq * kicvt + 0.5);
-            else
-                phs = MAXLEN;
-            
-            if(phs >= MAXLEN){
-                long r =  randint31(rand);                                                
-                int nrands = 16;
-                long ri = 0;
-                long rr[16];
-
-                rr[0] = rand;
-                for(i = 1;i < nrands;i++)
-                    rr[i]= randint31(rr[i-1]);
-                for(i=0;i < nrands;i++)
-                    ri += rr[i]>>4;                        
-                r = ri - BISHIFT;                        
-                                         
-                phs &= PHMASK;
-                rand = r;
-                num1 = num2;
-                num2 = (double)((long)((unsigned)r<<1)-BIPOLAR) * dv2_31;                
-                dfdmax = (num2-num1) / FMAXLEN;                            
-            }
-		}
-        
-        val *= 1.85;
-		lastval = val * param.modrange;
-		kcount = 1;
-	}
-	else
-		kcount++;
-	return 	lastval;
-}
-
-inline double fastlfo::rand_exp_tick(void)
-{
-	double val;
-	double dval = 0.0;
-
-	if(kcount==ksamps){
-		if(b_sync){
-			val = num1 +(double)(curphs * dfdmax);
-			phs += phs_incr;
-			curphs += phs_incr;
-			if(curphs >= MAXLEN){
-				long r =  randint31(rand);				                
-                long rr = r;
-                double dbiexp = 0.0;
-                                    
-                dbiexp = 2.0 * (double)rr * dv2_31;
-                while(dbiexp==0.0 || dbiexp == 2.0){                    
-                    rr = randint31(rr);                                            
-                }                
-                if(dbiexp > 1.0)
-                    dval = -log(2.0-dbiexp) / LAMBDA;
-                if(dbiexp < 1.0)
-                    dval = log(dbiexp) / LAMBDA;
-                if( dval > 0.9)
-                    dval = 0.9;
-                if(dval < -0.9)
-                    dval = -0.9;                                        
-                r =   ((unsigned long)(dval/dv2_31) +BIPOLAR)>>1;                                       
-                
-				curphs &= PHMASK;
-				rand = r;
-				num1 = num2;
-				num2 = (double)((long)((unsigned)r<<1)-BIPOLAR) * dv2_31;
-				dfdmax = (num2-num1) / FMAXLEN;
-			}
-			if(phs >= MAXLEN)
-				phs &= PHMASK;
-		}
-		else{           
-            val = num1 +(double)(phs * dfdmax);
-            if(ksamps > 1)
-                phs +=(long) (param.freq * kicvt + 0.5);
-            else
-                phs = MAXLEN;            
-            if(phs >= MAXLEN){
-                long r =  randint31(rand);                                
-                long rr = r;
-                double dbiexp = 0.0;
-                                    
-                dbiexp = 2.0 * (double)rr * dv2_31;
-                while(dbiexp==0.0 || dbiexp == 2.0){                    
-                    rr = randint31(rr);                                            
-                }                
-                if(dbiexp > 1.0)
-                    dval = -log(2.0-dbiexp) / LAMBDA;
-                if(dbiexp < 1.0)
-                    dval = log(dbiexp) / LAMBDA;
-                if( dval > 0.9)
-                    dval = 0.9;
-                if(dval < -0.9)
-                    dval = -0.9;                                        
-                r =   ((unsigned long)(dval/dv2_31) +BIPOLAR)>>1;
-                                                                  
-                phs &= PHMASK;
-                rand = r;
-                num1 = num2;
-                num2 = (double)((long)((unsigned)r<<1)-BIPOLAR) * dv2_31;
-                
-                dfdmax = (num2-num1) / FMAXLEN;                            
-            }
-		}        
-		lastval = val * param.modrange;
-		kcount = 1;
-	}
-	else
-		kcount++;
-	return 	lastval;
-}
-
-
-inline double fastlfo::randhtick(void)
-{
-	double val;
-	
-	if(kcount==ksamps){
-		if(b_sync){
-			val = num1;
-			phs += phs_incr;
-			curphs += phs_incr;
-			if(curphs >= MAXLEN){
-				long r = randint31(rand);
-				if(tpdf){
-					long rr = randint31(r);
-                    long ri;
-                    unsigned long sum = r + rr;
-                    ri = (long)(sum/2);                       
-                    r =  ri - TPDFSHIFT;
-				}
-				rand = r;
-				num1 = (double)((long)((unsigned)r<<1) - BIPOLAR) * dv2_31;
-				curphs &= PHMASK;
-			}
-			if(phs >= MAXLEN)
-				phs &= PHMASK;
-		}
-		else {
-			val = num1;
-            if(ksamps > 1)
-                phs +=(long) (param.freq * kicvt + 0.5);
-            else
-                phs = MAXLEN;
-			
-			if(phs >= MAXLEN){
-				long r = randint31(rand);
-				if(tpdf){
-					long rr = randint31(r);
-                    long ri;
-                    unsigned long sum = r + rr;
-                    ri = (long)(sum/2);                       
-                    r =  ri - TPDFSHIFT;
-				}
-				phs &= PHMASK;
-				rand = r;
-				num1 = (double)((long)((unsigned)r<<1) - BIPOLAR) * dv2_31;
-			}
-		}
-		lastval = val * param.modrange;
-		kcount = 1;
-	}
-	else
-		kcount++;
-	return 	lastval;
-}
-
-
-inline double fastlfo::randh_tpdf_tick(void)
-{
-	double val;
-	
-	if(kcount==ksamps){
-		if(b_sync){
-			val = num1;
-			phs += phs_incr;
-			curphs += phs_incr;
-			if(curphs >= MAXLEN){
-				long r = randint31(rand);
-				
-				long rr = randint31(r);
-                long ri;
-                unsigned long sum = r + rr;
-                ri = (long)(sum/2);                       
-                r =  ri - TPDFSHIFT;
-				
-				rand = r;
-				num1 = (double)((long)((unsigned)r<<1) - BIPOLAR) * dv2_31;
-				curphs &= PHMASK;
-			}
-			if(phs >= MAXLEN)
-				phs &= PHMASK;
-		}
-		else {
-			val = num1;			
-			if(ksamps > 1)
-                phs +=(long) (param.freq * kicvt + 0.5);
-            else
-                phs = MAXLEN;
-			if(phs >= MAXLEN){
-				long r = randint31(rand);
-				
-				long rr = randint31(r);
-                long ri;
-                unsigned long sum = r + rr;
-                ri = (long)(sum/2);                       
-                r =  ri - TPDFSHIFT;
-				
-				phs &= PHMASK;
-				rand = r;
-				num1 = (double)((long)((unsigned)r<<1) - BIPOLAR) * dv2_31;
-			}
-		}
-		lastval = val * param.modrange;
-		kcount = 1;
-	}
-	else
-		kcount++;
-	return 	lastval;
-}
-
-/* return positive value between 0 and RANDINT_MAX */
-inline long fastlfo::randint31(long seed31)
-{
-	unsigned long rilo, rihi;
-
-    rilo = RIA * (long)(seed31 & 0xFFFF);
-    rihi = RIA * (long)((unsigned long)seed31 >> 16);
-    rilo += (rihi & 0x7FFF) << 16;
-    if (rilo > (unsigned long) RIM) {
-      rilo &= RIM;
-      ++rilo;
-    }
-    rilo += rihi >> 15;
-    if (rilo > (unsigned long) RIM) {
-      rilo &= RIM;
-      ++rilo;
-    }
-    return (long)rilo;
-}
-
-bool fastlfo::set_WaveType(LfoWaveType type)		
-{	
-	switch(type){
-	case(LFO_SINE):
-            tf = &fastlfo::sinetick;
-		break;
-	case(LFO_TRIANGLE):
-		tf = &fastlfo::tritick;
-		break;
-	case(LFO_SQUARE):
-		tf = &fastlfo::squaretick;
-		break;
-	case(LFO_SAWUP):
-		tf = &fastlfo::sawuptick;
-		break;
-	case(LFO_SAWDOWN):
-		tf = &fastlfo::sawdowntick;
-		break;
-	case(LFO_RANDOM):
-		tf = &fastlfo::randomtick;
-		break;
-       case(LFO_RAND_TPDF):
-		tf = &fastlfo::rand_tpdf_tick;
-		break;
-       case(LFO_RAND_GAUSS):
-		tf = &fastlfo::rand_gauss_tick;
-		break;
-       case(LFO_RAND_EXP):
-		tf = &fastlfo::rand_exp_tick;
-		break;
-	case(LFO_RND_SH):
-		tf = &fastlfo::randhtick;
-		break;
-       case(LFO_RANDH_TPDF):
-		tf = &fastlfo::randh_tpdf_tick;
-		break;
-	default:
-		return false;
-	}
-	m_cur_WaveType = type;	
-	return true;
-}
-

dev/externals/reverb/wavetable.h

@@ -1,126 +0,0 @@
-/*
- * Copyright (c) 1983-2013 Richard Dobson and Composers Desktop Project Ltd
- * http://people.bath.ac.uk/masrwd
- * http://www.composersdesktop.com
- * This file is part of the CDP System.
- * The CDP System is free software; you can redistribute it
- * and/or modify it under the terms of the GNU Lesser General Public 
- * License as published by the Free Software Foundation; either 
- * version 2.1 of the License, or (at your option) any later version. 
- *
- * The CDP System is distributed in the hope that it will be useful, 
- * but WITHOUT ANY WARRANTY; without even the implied warranty of 
- * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  
- * See the GNU Lesser General Public License for more details.
- * You should have received a copy of the GNU Lesser General Public
- * License along with the CDP System; if not, write to the Free Software
- * Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
- *
- */
- 
-
-// wavetable.h: interface for the fast_lfo class.
-// Feb 2009 added gaussian option to random
-//FWD 3.20: eliminate warning of unused private var (phsmask)
-
-//////////////////////////////////////////////////////////////////////
-
-
-#if !defined(AFX_WAVETABLE_H__2F961225_3FF4_11D2_96D4_444553540000__INCLUDED_)
-#define AFX_WAVETABLE_H__2F961225_3FF4_11D2_96D4_444553540000__INCLUDED_
-
-#if _MSC_VER >= 1000
-#pragma once
-#endif // _MSC_VER >= 1000
-
-typedef enum lfowavetype {LFO_SINE,LFO_TRIANGLE,LFO_SAWUP,LFO_SAWDOWN,LFO_SQUARE,LFO_RANDOM,LFO_RND_SH,
-LFO_RAND_TPDF,LFO_RAND_GAUSS,LFO_RAND_EXP, LFO_RANDH_TPDF,
-/*LFO_RANDH_GAUSS,LFO_RANDH_EXP, */
-LFO_NUM_WAVES } LfoWaveType;
-
-enum {PVS_LFO_FREQ_LORANGE, PVS_LFO_FREQ_HIRANGE};
-
-
-typedef struct lfoparam
-{
-	double freq;
-	double modrange;
-} LfoParam;
-
-
-
-class fastlfo {
-	typedef double (fastlfo:: *tickfunc)(void);
-public:
-
-	fastlfo();
-	virtual ~fastlfo() {}
-	long init(double srate, double normphase = 0.0,long seedval = 0,unsigned long ksmps = 1);
-	double		tick(void)							{ return (this->*tf)(); }
-	tickfunc	tf;
-	void		set_freq(double freq)				{ param.freq = freq;	}
-	double		get_freq(void)				const	{ return param.freq;	}
-	void		set_mod(double mod)					{ param.modrange = mod;	}
-	double		get_mod(void)				const	{ return param.modrange;}
-	void		reset(double phase);
-	bool		set_WaveType(LfoWaveType type);
-	void		set_tpdf()              		    { tpdf = true; gauss = biexp = false;}
-    void        set_white()                         {  tpdf = gauss = biexp = false;}
-    void        set_biexp()                         {  tpdf = gauss = false; biexp = true;}
-    void        set_flat()                          { tpdf = gauss = biexp = false;}
-    void        set_gaussian()                      { tpdf = biexp = false; gauss = true;}
-	void		sync_phase(double phase, double offset, double phaseincr);
-	void		set_sync(bool onoff)				{ b_sync = onoff;}
-    double      csrand();                   
-private:
-	double	sinetick(void);
-	double	tritick(void);
-	double	squaretick(void);
-	double	sawuptick(void);
-	double	sawdowntick(void);
-	double	randomtick(void);
-	double	randhtick(void);
-    double  rand_tpdf_tick(void);
-    double  rand_gauss_tick(void);
-    double  rand_exp_tick(void);
-    double  randh_tpdf_tick(void);
- //   double  randh_gauss_tick(void);
- //   double  randh_exp_tick(void);
-
-	long	randint31(long seed31);
-
-	double	m_srate, m_inv_srate;
-	double twopiovrsr;
-	double curfreq;
-	double curphase;
-	double incr;
-	LfoParam	param;
-	LfoWaveType  m_cur_WaveType;  
-	/* vars for random oscs, with a little help from Csound!*/
-	long		phs;
-	long		kicvt;
-//	long		phsmask;   // currently not used
-	long		rand;
-	double		num1,num2;
-	double		dfdmax;
-	/* for krate output */
-	unsigned long		ksamps;
-	unsigned long		kcount;
-	double		lastval;
-	bool	tpdf;
-    bool gauss;
-    bool biexp;
-	// for lfo sync
-	bool b_sync;
-	double		offset;
-	// for random oscs
-	long curphs;
-	long phs_incr;
-	long phs_offset;
-    /* for csrand */
-    long csseed;
-};
-
-#endif // !defined(AFX_WAVETABLE_H__2F961225_3FF4_11D2_96D4_444553540000__INCLUDED_)
-
-