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📄 analys.c

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 * Revision 1.1  1996/08/19  22:29:08  jaf * Initial revision * *//* Revision 1.3  1996/03/29  22:05:55  jaf *//* Commented out the common block variables that are not needed by the *//* embedded version. *//* Revision 1.2  1996/03/26  19:34:50  jaf *//* Added comments indicating which constants are not needed in an *//* application that uses the LPC-10 coder. *//* Revision 1.1  1996/02/07  14:44:09  jaf *//* Initial revision *//*   LPC Processing control variables: *//* *** Read-only: initialized in setup *//*  Files for Speech, Parameter, and Bitstream Input & Output, *//*    and message and debug outputs. *//* Here are the only files which use these variables: *//* lpcsim.f setup.f trans.f error.f vqsetup.f *//* Many files which use fdebug are not listed, since it is only used in *//* those other files conditionally, to print trace statements. *//* 	integer fsi, fso, fpi, fpo, fbi, fbo, pbin, fmsg, fdebug *//*  LPC order, Frame size, Quantization rate, Bits per frame, *//*    Error correction *//* Subroutine SETUP is the only place where order is assigned a value, *//* and that value is 10.  It could increase efficiency 1% or so to *//* declare order as a constant (i.e., a Fortran PARAMETER) instead of as *//* a variable in a COMMON block, since it is used in many places in the *//* core of the coding and decoding routines.  Actually, I take that back. *//* At least when compiling with f2c, the upper bound of DO loops is *//* stored in a local variable before the DO loop begins, and then that is *//* compared against on each iteration. *//* Similarly for lframe, which is given a value of MAXFRM in SETUP. *//* Similarly for quant, which is given a value of 2400 in SETUP.  quant *//* is used in only a few places, and never in the core coding and *//* decoding routines, so it could be eliminated entirely. *//* nbits is similar to quant, and is given a value of 54 in SETUP. *//* corrp is given a value of .TRUE. in SETUP, and is only used in the *//* subroutines ENCODE and DECODE.  It doesn't affect the speed of the *//* coder significantly whether it is .TRUE. or .FALSE., or whether it is *//* a constant or a variable, since it is only examined once per frame. *//* Leaving it as a variable that is set to .TRUE.  seems like a good *//* idea, since it does enable some error-correction capability for *//* unvoiced frames, with no change in the coding rate, and no noticeable *//* quality difference in the decoded speech. *//* 	integer quant, nbits *//* *** Read/write: variables for debugging, not needed for LPC algorithm *//*  Current frame, Unstable frames, Output clip count, Max onset buffer, *//*    Debug listing detail level, Line count on listing page *//* nframe is not needed for an embedded LPC10 at all. *//* nunsfm is initialized to 0 in SETUP, and incremented in subroutine *//* ERROR, which is only called from RCCHK.  When LPC10 is embedded into *//* an application, I would recommend removing the call to ERROR in RCCHK, *//* and remove ERROR and nunsfm completely. *//* iclip is initialized to 0 in SETUP, and incremented in entry SWRITE in *//* sread.f.  When LPC10 is embedded into an application, one might want *//* to cause it to be incremented in a routine that takes the output of *//* SYNTHS and sends it to an audio device.  It could be optionally *//* displayed, for those that might want to know what it is. *//* maxosp is never initialized to 0 in SETUP, although it probably should *//* be, and it is updated in subroutine ANALYS.  I doubt that its value *//* would be of much interest to an application in which LPC10 is *//* embedded. *//* listl and lincnt are not needed for an embedded LPC10 at all. *//* 	integer nframe, nunsfm, iclip, maxosp, listl, lincnt *//* 	common /contrl/ fsi, fso, fpi, fpo, fbi, fbo, pbin, fmsg, fdebug *//* 	common /contrl/ quant, nbits *//* 	common /contrl/ nframe, nunsfm, iclip, maxosp, listl, lincnt *//*       Arguments to entry PITDEC (below) *//* 	Parameters/constants *//*  Constants *//*    NF =     Number of frames *//*    AF =     Frame in which analysis is done *//*    OSLEN =  Length of the onset buffer *//*    LTAU =   Number of pitch lags *//*    SBUFL, SBUFH =   Start and end index of speech buffers *//*    LBUFL, LBUFH =   Start and end index of LPF speech buffer *//*   MINWIN, MAXWIN = Min and Max length of voicing (and analysis) windows*//*    PWLEN, PWINH, PWINL = Length, upper and lower limits of pitch window *//*    DVWINL, DVWINH = Default lower and upper limits of voicing window *//*       The tables TAU and BUFLIM, and the variable PRECOEF, are not *//*       Fortran PARAMETER's, but they are initialized with DATA *//*       statements, and never modified.  Thus, they need not have SAVE *//*       statements for them to keep their values from one invocation to *//*       the next. *//*       Local variables that need not be saved *//*       Local state *//*  Data Buffers *//*    INBUF	Raw speech (with DC bias removed each frame) *//*    PEBUF	Preemphasized speech *//*    LPBUF	Low pass speech buffer *//*    IVBUF	Inverse filtered speech *//*    OSBUF	Indexes of onsets in speech buffers *//*    VWIN	Voicing window indices *//*    AWIN	Analysis window indices *//*    EWIN	Energy window indices *//*    VOIBUF	Voicing decisions on windows in VWIN *//*    RMSBUF	RMS energy *//*    RCBUF	Reflection Coefficients *//*  Pitch is handled separately from the above parameters. *//*  The following variables deal with pitch: *//*    MIDX	Encoded initial pitch estimate for analysis frame *//*    IPITCH	Initial pitch computed for frame AF (decoded from MIDX) *//*    PITCH 	The encoded pitch value (index into TAU) for the present *//* 		frame (delayed and smoothed by Dyptrack) */    /* Parameter adjustments */    if (speech) {	--speech;	}    if (voice) {	--voice;	}    if (rc) {	--rc;	}    /* Function Body *//*   Calculations are done on future frame due to requirements *//*   of the pitch tracker.  Delay RMS and RC's 2 frames to give *//*   current frame parameters on return. *//*   Update all buffers */    inbuf = &(st->inbuf[0]);    pebuf = &(st->pebuf[0]);    lpbuf = &(st->lpbuf[0]);    ivbuf = &(st->ivbuf[0]);    bias = &(st->bias);    osbuf = &(st->osbuf[0]);    osptr = &(st->osptr);    obound = &(st->obound[0]);    vwin = &(st->vwin[0]);    awin = &(st->awin[0]);    voibuf = &(st->voibuf[0]);    rmsbuf = &(st->rmsbuf[0]);    rcbuf = &(st->rcbuf[0]);    zpre = &(st->zpre);    i__1 = 720 - contrl_1.lframe;    for (i__ = 181; i__ <= i__1; ++i__) {	inbuf[i__ - 181] = inbuf[contrl_1.lframe + i__ - 181];	pebuf[i__ - 181] = pebuf[contrl_1.lframe + i__ - 181];    }    i__1 = 540 - contrl_1.lframe;    for (i__ = 229; i__ <= i__1; ++i__) {	ivbuf[i__ - 229] = ivbuf[contrl_1.lframe + i__ - 229];    }    i__1 = 720 - contrl_1.lframe;    for (i__ = 25; i__ <= i__1; ++i__) {	lpbuf[i__ - 25] = lpbuf[contrl_1.lframe + i__ - 25];    }    j = 1;    i__1 = (*osptr) - 1;    for (i__ = 1; i__ <= i__1; ++i__) {	if (osbuf[i__ - 1] > contrl_1.lframe) {	    osbuf[j - 1] = osbuf[i__ - 1] - contrl_1.lframe;	    ++j;	}    }    *osptr = j;    voibuf[0] = voibuf[2];    voibuf[1] = voibuf[3];    for (i__ = 1; i__ <= 2; ++i__) {	vwin[(i__ << 1) - 2] = vwin[((i__ + 1) << 1) - 2] - contrl_1.lframe;	vwin[(i__ << 1) - 1] = vwin[((i__ + 1) << 1) - 1] - contrl_1.lframe;	awin[(i__ << 1) - 2] = awin[((i__ + 1) << 1) - 2] - contrl_1.lframe;	awin[(i__ << 1) - 1] = awin[((i__ + 1) << 1) - 1] - contrl_1.lframe;/*       EWIN(*,J) is unused for J .NE. AF, so the following shift is *//*       unnecessary.  It also causes error messages when the C version *//*       of the code created from this by f2c is run with Purify.  It *//*       correctly complains that uninitialized memory is being read. *//* 	   EWIN(1,I) = EWIN(1,I+1) - LFRAME *//* 	   EWIN(2,I) = EWIN(2,I+1) - LFRAME */	obound[i__ - 1] = obound[i__];	voibuf[i__ * 2] = voibuf[(i__ + 1) * 2];	voibuf[(i__ << 1) + 1] = voibuf[((i__ + 1) << 1) + 1];	rmsbuf[i__ - 1] = rmsbuf[i__];	i__1 = contrl_1.order;	for (j = 1; j <= i__1; ++j) {	    rcbuf[j + i__ * 10 - 11] = rcbuf[j + (i__ + 1) * 10 - 11];	}    }/*   Copy input speech, scale to sign+12 bit integers *//*   Remove long term DC bias. *//*       If the average value in the frame was over 1/4096 (after current *//*       BIAS correction), then subtract that much more from samples in *//*       next frame.  If the average value in the frame was under *//*       -1/4096, add 1/4096 more to samples in next frame.  In all other *//*       cases, keep BIAS the same. */    temp = 0.f;    i__1 = contrl_1.lframe;    for (i__ = 1; i__ <= i__1; ++i__) {	inbuf[720 - contrl_1.lframe + i__ - 181] = speech[i__] * 4096.f - 		(*bias);	temp += inbuf[720 - contrl_1.lframe + i__ - 181];    }    if (temp > (real) contrl_1.lframe) {	*bias += 1;    }    if (temp < (real) (-contrl_1.lframe)) {	*bias += -1;    }/*   Place Voicing Window */    i__ = 721 - contrl_1.lframe;    preemp_(&inbuf[i__ - 181], &pebuf[i__ - 181], &contrl_1.lframe, &precoef, 	    zpre);    onset_(pebuf, osbuf, osptr, &c__10, &c__181, &c__720, &contrl_1.lframe, st);/*       MAXOSP is just a debugging variable. *//* 	MAXOSP = MAX( MAXOSP, OSPTR ) */    placev_(osbuf, osptr, &c__10, &obound[2], vwin, &c__3, &contrl_1.lframe, 	    &c__90, &c__156, &c__307, &c__462);/*        The Pitch Extraction algorithm estimates the pitch for a frame *//*   of speech by locating the minimum of the average magnitude difference *//*   function (AMDF).  The AMDF operates on low-pass, inverse filtered *//*   speech.  (The low-pass filter is an 800 Hz, 19 tap, equiripple, FIR *//*   filter and the inverse filter is a 2nd-order LPC filter.)  The pitch *//*   estimate is later refined by dynamic programming (DYPTRK).  However, *//*   since some of DYPTRK's parameters are a function of the voicing *//*  decisions, a voicing decision must precede the final pitch estimation.*//*   See subroutines LPFILT, IVFILT, and TBDM. *//*       LPFILT reads indices LBUFH-LFRAME-29 = 511 through LBUFH = 720 *//*       of INBUF, and writes indices LBUFH+1-LFRAME = 541 through LBUFH *//*       = 720 of LPBUF. */    lpfilt_(&inbuf[228], &lpbuf[384], &c__312, &contrl_1.lframe);/*       IVFILT reads indices (PWINH-LFRAME-7) = 353 through PWINH = 540 *//*       of LPBUF, and writes indices (PWINH-LFRAME+1) = 361 through *//*       PWINH = 540 of IVBUF. */    ivfilt_(&lpbuf[204], ivbuf, &c__312, &contrl_1.lframe, ivrc);/*       TBDM reads indices PWINL = 229 through *//*       (PWINL-1)+MAXWIN+(TAU(LTAU)-TAU(1))/2 = 452 of IVBUF, and writes *//*       indices 1 through LTAU = 60 of AMDF. */    tbdm_(ivbuf, &c__156, tau, &c__60, amdf, &minptr, &maxptr, &mintau);/*        Voicing decisions are made for each half frame of input speech. *//*   An initial voicing classification is made for each half of the *//*   analysis frame, and the voicing decisions for the present frame *//*   are finalized.  See subroutine VOICIN. *//*        The voicing detector (VOICIN) classifies the input signal as *//*   unvoiced (including silence) or voiced using the AMDF windowed *//*   maximum-to-minimum ratio, the zero crossing rate, energy measures, *//*   reflection coefficients, and prediction gains. *//*        The pitch and voicing rules apply smoothing and isolated *//*   corrections to the pitch and voicing estimates and, in the process, *//*   introduce two frames of delay into the corrected pitch estimates and *//*   voicing decisions. */    for (half = 1; half <= 2; ++half) {	voicin_(&vwin[4], inbuf, lpbuf, buflim, &half, &amdf[minptr - 1], &		amdf[maxptr - 1], &mintau, ivrc, obound, voibuf, &c__3, st);    }/*   Find the minimum cost pitch decision over several frames *//*   given the current voicing decision and the AMDF array */    dyptrk_(amdf, &c__60, &minptr, &voibuf[7], pitch, &midx, st);    ipitch = tau[midx - 1];/*   Place spectrum analysis and energy windows */    placea_(&ipitch, voibuf, &obound[2], &c__3, vwin, awin, ewin, &	    contrl_1.lframe, &c__156);/*  Remove short term DC bias over the analysis window, Put result in ABUF*/    lanal = awin[5] + 1 - awin[4];    dcbias_(&lanal, &pebuf[awin[4] - 181], abuf);/*       ABUF(1:LANAL) is now defined.  It is equal to *//*       PEBUF(AWIN(1,AF):AWIN(2,AF)) corrected for short term DC bias. *//*   Compute RMS over integer number of pitch periods within the *//*   analysis window. *//*   Note that in a hardware implementation this computation may be *//*   simplified by using diagonal elements of PHI computed by MLOAD. */    i__1 = ewin[5] - ewin[4] + 1;    energy_(&i__1, &abuf[ewin[4] - awin[4]], &rmsbuf[2]);/*   Matrix load and invert, check RC's for stability */    mload_(&contrl_1.order, &c__1, &lanal, abuf, phi, psi);    invert_(&contrl_1.order, phi, psi, &rcbuf[20]);    rcchk_(&contrl_1.order, &rcbuf[10], &rcbuf[20]);/*   Set return parameters */    voice[1] = voibuf[2];    voice[2] = voibuf[3];    *rms = rmsbuf[0];    i__1 = contrl_1.order;    for (i__ = 1; i__ <= i__1; ++i__) {	rc[i__] = rcbuf[i__ - 1];    }    return 0;} /* analys_ */

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