ref: a3c890b854952cb1e6d44948c5ad8e84515e97b5
dir: /lpc10/analys.c/
/* -- translated by f2c (version 19951025).
You must link the resulting object file with the libraries:
-lf2c -lm (in that order)
*/
#include "f2c.h"
int analys_(real *speech, integer *voice, integer *pitch, real *rms, real *rc, struct lpc10_encoder_state *st);
/* Common Block Declarations */
extern struct {
integer order, lframe;
logical corrp;
} contrl_;
#define contrl_1 contrl_
/* Table of constant values */
static integer c__10 = 10;
static integer c__181 = 181;
static integer c__720 = 720;
static integer c__3 = 3;
static integer c__90 = 90;
static integer c__156 = 156;
static integer c__307 = 307;
static integer c__462 = 462;
static integer c__312 = 312;
static integer c__60 = 60;
static integer c__1 = 1;
/* ****************************************************************** */
/* ANALYS Version 55 */
/* Revision 1.9 1996/05/23 19:41:07 jaf */
/* Commented out some unnecessary lines that were reading uninitialized */
/* values. */
/* Revision 1.8 1996/03/27 23:57:55 jaf */
/* Added some comments about which indices of the local buffers INBUF, */
/* LPBUF, etc., get read or modified by some of the subroutine calls. I */
/* just did this while trying to figure out the discrepancy between the */
/* embedded code compiled with all local variables implicitly saved, and */
/* without. */
/* I added some debugging write statements in hopes of finding a problem. */
/* None of them ever printed anything while running with the long input */
/* speech file dam9.spd provided in the distribution. */
/* Revision 1.7 1996/03/27 18:06:20 jaf */
/* Commented out access to MAXOSP, which is just a debugging variable */
/* that was defined in the COMMON block CONTRL in contrl.fh. */
/* Revision 1.6 1996/03/26 19:31:33 jaf */
/* Commented out trace statements. */
/* Revision 1.5 1996/03/21 15:19:35 jaf */
/* Added comments for ENTRY PITDEC. */
/* Revision 1.4 1996/03/19 20:54:27 jaf */
/* Added a line to INITANALYS. See comments there. */
/* Revision 1.3 1996/03/19 20:52:49 jaf */
/* Rearranged the order of the local variables quite a bit, to separate */
/* them into groups of "constants", "locals that don't need to be saved */
/* from one call to the next", and "local that do need to be saved from */
/* one call to the next". */
/* Several locals in the last set should have been given initial values, */
/* but weren't. I gave them all initial values of 0. */
/* Added a separate ENTRY INITANALYS that initializes all local state */
/* that should be, and also calls the corresponding entries of the */
/* subroutines called by ANALYS that also have local state. */
/* There used to be DATA statements in ANALYS. I got rid of most of */
/* them, and added a local logical variable FIRST that calls the entry */
/* INITANALYS on the first call to ANALYS. This is just so that one need */
/* not remember to call INITANALYS first in order for the state to be */
/* initialized. */
/* Revision 1.2 1996/03/11 23:29:32 jaf */
/* Added several comments with my own personal questions about the */
/* Fortran 77 meaning of the parameters passed to the subroutine PREEMP. */
/* Revision 1.1 1996/02/07 14:42:29 jaf */
/* Initial revision */
/* ****************************************************************** */
/* SUBROUTINE ANALYS */
/* Input: */
/* SPEECH */
/* Indices 1 through LFRAME read. */
/* Output: */
/* VOICE */
/* Indices 1 through 2 written. */
/* PITCH */
/* Written in subroutine DYPTRK, and then perhaps read and written */
/* some more. */
/* RMS */
/* Written. */
/* RC */
/* Indices 1 through ORDER written (ORDER defined in contrl.fh). */
/* This subroutine maintains local state from one call to the next. If */
/* you want to switch to using a new audio stream for this filter, or */
/* reinitialize its state for any other reason, call the ENTRY */
/* INITANALYS. */
/* ENTRY PITDEC */
/* Input: */
/* PITCH - Encoded pitch index */
/* Output: */
/* PTAU - Decoded pitch period */
/* This entry has no local state. It accesses a "constant" array */
/* declared in ANALYS. */
/* Subroutine */ int analys_(real *speech, integer *voice, integer
*pitch, real *rms, real *rc, struct lpc10_encoder_state *st)
{
/* Initialized data */
static integer tau[60] = { 20,21,22,23,24,25,26,27,28,29,30,31,32,33,34,
35,36,37,38,39,40,42,44,46,48,50,52,54,56,58,60,62,64,66,68,70,72,
74,76,78,80,84,88,92,96,100,104,108,112,116,120,124,128,132,136,
140,144,148,152,156 };
static integer buflim[4] = { 181,720,25,720 };
static real precoef = .9375f;
/* System generated locals */
integer i__1;
/* Local variables */
real amdf[60];
integer half;
real abuf[156];
real *bias;
extern /* Subroutine */ int tbdm_(real *, integer *, integer *, integer *,
real *, integer *, integer *, integer *);
integer *awin;
integer midx, ewin[6] /* was [2][3] */;
real ivrc[2], temp;
real *zpre;
integer *vwin;
integer i__, j, lanal;
extern /* Subroutine */ int rcchk_(integer *, real *, real *), mload_(
integer *, integer *, integer *, real *, real *, real *);
real *inbuf, *pebuf;
real *lpbuf, *ivbuf;
real *rcbuf;
integer *osbuf;
extern /* Subroutine */ int onset_(real *, integer *, integer *, integer *
, integer *, integer *, integer *, struct lpc10_encoder_state *);
integer *osptr;
extern int placea_(integer *, integer *
, integer *, integer *, integer *, integer *, integer *, integer *
, integer *), dcbias_(integer *, real *, real *), placev_(integer
*, integer *, integer *, integer *, integer *, integer *, integer
*, integer *, integer *, integer *, integer *);
integer ipitch;
integer *obound;
extern /* Subroutine */ int preemp_(real *, real *, integer *, real *,
real *), voicin_(integer *, real *, real *, integer *, integer *,
real *, real *, integer *, real *, integer *, integer *, integer *,
struct lpc10_encoder_state *);
integer *voibuf;
integer mintau;
real *rmsbuf;
extern /* Subroutine */ int lpfilt_(real *, real *, integer *, integer *),
ivfilt_(real *, real *, integer *, integer *, real *), energy_(
integer *, real *, real *), invert_(integer *, real *, real *,
real *);
integer minptr, maxptr;
extern /* Subroutine */ int dyptrk_(real *, integer *, integer *, integer
*, integer *, integer *, struct lpc10_encoder_state *);
real phi[100] /* was [10][10] */, psi[10];
/* 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 versio
n */
/* 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_ */