📄 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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