/* * usr.c * * Conversion pvf <--> USR GSM and ADPCM formats. * * $Id: usr.c,v 1.5 2001/12/22 19:44:09 marcs Exp $ * */ #include "../include/voice.h" /* Forward defs of the format-specific routines */ static int pvftousrgsm (FILE *fd_in, FILE *fd_out, pvf_header *header_in); static int pvftousradpcm (FILE *fd_in, FILE *fd_out, pvf_header *header_in); static int usrgsmtopvf (FILE *fd_in, FILE *fd_out, pvf_header *header_out); static int usradpcmtopvf (FILE *fd_in, FILE *fd_out, pvf_header *header_out); int pvftousr(FILE *fd_in, FILE *fd_out, int compression, pvf_header *header_in) { switch (compression) { case 1: return (pvftousrgsm(fd_in, fd_out, header_in)); case 4: return (pvftousradpcm(fd_in, fd_out, header_in)); default: return -1; } } int usrtopvf (FILE *fd_in, FILE *fd_out, int compression, pvf_header *header_out) { switch (compression) { case 1: return (usrgsmtopvf(fd_in, fd_out, header_out)); case 4: return (usradpcmtopvf(fd_in, fd_out, header_out)); default: return -1; } } /***************** ** GSM SECTION ** *****************/ #include "../libmgsm/gsm.h" /* USR's GSM data format consists of 38-byte frames of data where the * first two bytes of the frame (usually "0xFE 0xFE" for valid data and * "0xB6 0xB6" for silence, and 3 bytes of trailer ("0x0 0xA5 0xA5") can * be discarded, giving 33 bytes of useful data. * Newer models can also generate frames with raw data (without the * trailing and leading bytes). * The decoding function tries to detect the frame type and pass the * 33 bytes of data to a garden variety GSM decode process. * In my case, I used GSM 06.10 from Technische * Universitaet Berlin ftp://ftp.cs.tu-berlin.de/pub/local/kbs/tubmik/gsm/ * * The pvftousrgsm function just generates the old type of frame * since it can be played on both new and old models. */ unsigned char gsm_head[2] = { 0xfe, 0xfe }; unsigned char gsm_tail[3] = { 0x0, 0xa5, 0xa5 }; static int pvftousrgsm (FILE *fd_in, FILE *fd_out, pvf_header *header_in) { gsm r; gsm_signal s[ 160 ]; gsm_frame d; int opt_fast = 0; int opt_verbose = 0; int opt_ltp_cut = 0; int i; int sample = 0; if (!(r = gsm_create())) { perror("gsm_create"); return -1; } (void)gsm_option(r, GSM_OPT_FAST, &opt_fast); (void)gsm_option(r, GSM_OPT_VERBOSE, &opt_verbose); (void)gsm_option(r, GSM_OPT_LTP_CUT, &opt_ltp_cut); /* GSM operates on frames of 160 samples each, so we may run up against * the end of the file and be forced to backfill with zeroes. */ while (!feof(fd_in)) { for (i=0;i<160;i++) { sample = header_in->read_pvf_data(fd_in); if (feof(fd_in)) { memset((char *)(&s[i]), 0, sizeof(gsm_signal)*(160-i)); } else { sample >>= 8; if (sample > 0x7fff) sample = 0x7fff; if (sample < -0x8000) sample = -0x8000; s[i] = sample; } } gsm_encode(r, s, d); fwrite((char *)gsm_head, 2, 1, fd_out); fwrite((char *)d, sizeof(d), 1, fd_out); fwrite((char *)gsm_tail, 3, 1, fd_out); } gsm_destroy(r); return(OK); } static int usrgsmtopvf (FILE *fd_in, FILE *fd_out, pvf_header *header_out) { unsigned char inbuf[38]; gsm r; gsm_byte *s; gsm_signal d[ 160 ]; int opt_fast = 0; int opt_verbose = 0; int i, sample, bytes2read, chunksread; if (!(r = gsm_create())) { perror("gsm_create"); return -1; } (void)gsm_option(r, GSM_OPT_FAST, &opt_fast); (void)gsm_option(r, GSM_OPT_VERBOSE, &opt_verbose); /* * read the first frame to see if it has an * header or is raw data */ if ((chunksread=fread(inbuf, 33, 1, fd_in)) > 0) { if ((inbuf[0] == inbuf[1]) && ((inbuf[0] == 0xfe) || (inbuf[0] == 0xb6))) { /* * has an header */ fread(&inbuf[33], 5, 1, fd_in); s=&inbuf[2]; bytes2read=38; } else { /* * raw data */ s=&inbuf[0]; bytes2read=33; } } while (chunksread > 0) { /* --- MNI_p/JoSch ---> * I don'n know how this (now redundant to leave libmgsm untouched * -> see ../libmgsm/decode.c line 20) control for GSM_MAGIC * works with other USR- modems. Do they realy produce wrong * bytes so that it is necessary to control it? * For my US Robotics Vi 28.8 Faxmodem (gr) with Personal Voice * Mail (speed 33600) this seems to work. * May be there is a modem-setting I don't know (I don't know * any voice-settig-switches for this modem) that switches the * modem to a workable compression. At this time this patch works * for me until I get informations from the USR-Support. */ if ((((*s >> 4) & 0x0F) != GSM_MAGIC) || (((*s >> 4) & 0x0F) != 0)) *s |= (GSM_MAGIC << 4); /* <--- MNI_p/JoSch --- */ if (gsm_decode(r, s, d)) { fprintf(stderr, "%s: bad frame in input\n", program_name); gsm_destroy(r); return(ERROR); } for (i=0;i<160;i++) { sample = d[i]; if (sample > 0x7fff) { sample -= 0x10000; } header_out->write_pvf_data(fd_out, sample << 8); } chunksread=fread(inbuf, bytes2read, 1, fd_in); } return(OK); } /******************* ** ADPCM SECTION ** *******************/ /* This ADPCM code was released by Sun Microsystems, Inc. to the public domain * as a part of the CCITT (International Telegraph and Telephone Consultative * Committee) reference implementation of the G.711, G.721 and G.723 voice * compression standards. * * The following files from the original distribution have been concatenated * in this section: * * g72x.h header file for g721.c, g723_24.c and g723_40.c * g72x.c common denominator of G.721 and G.723 ADPCM codes * g721.c CCITT G.721 32Kbps ADPCM coder (with g72x.c) * * Below this, the pvf conversion routines follow. */ /* * This source code is a product of Sun Microsystems, Inc. and is provided * for unrestricted use. Users may copy or modify this source code without * charge. * * SUN SOURCE CODE IS PROVIDED AS IS WITH NO WARRANTIES OF ANY KIND INCLUDING * THE WARRANTIES OF DESIGN, MERCHANTIBILITY AND FITNESS FOR A PARTICULAR * PURPOSE, OR ARISING FROM A COURSE OF DEALING, USAGE OR TRADE PRACTICE. * * Sun source code is provided with no support and without any obligation on * the part of Sun Microsystems, Inc. to assist in its use, correction, * modification or enhancement. * * SUN MICROSYSTEMS, INC. SHALL HAVE NO LIABILITY WITH RESPECT TO THE * INFRINGEMENT OF COPYRIGHTS, TRADE SECRETS OR ANY PATENTS BY THIS SOFTWARE * OR ANY PART THEREOF. * * In no event will Sun Microsystems, Inc. be liable for any lost revenue * or profits or other special, indirect and consequential damages, even if * Sun has been advised of the possibility of such damages. * * Sun Microsystems, Inc. * 2550 Garcia Avenue * Mountain View, California 94043 */ /* * g72x.h * * Header file for CCITT conversion routines. * */ #ifndef _G72X_H #define _G72X_H #define AUDIO_ENCODING_LINEAR (3) /* PCM 2's-complement (0-center) */ /* * The following is the definition of the state structure * used by the G.721/G.723 encoder and decoder to preserve their internal * state between successive calls. The meanings of the majority * of the state structure fields are explained in detail in the * CCITT Recommendation G.721. The field names are essentially indentical * to variable names in the bit level description of the coding algorithm * included in this Recommendation. */ struct g72x_state { long yl; /* Locked or steady state step size multiplier. */ short yu; /* Unlocked or non-steady state step size multiplier. */ short dms; /* Short term energy estimate. */ short dml; /* Long term energy estimate. */ short ap; /* Linear weighting coefficient of 'yl' and 'yu'. */ short a[2]; /* Coefficients of pole portion of prediction filter. */ short b[6]; /* Coefficients of zero portion of prediction filter. */ short pk[2]; /* * Signs of previous two samples of a partially * reconstructed signal. */ short dq[6]; /* * Previous 6 samples of the quantized difference * signal represented in an internal floating point * format. */ short sr[2]; /* * Previous 2 samples of the quantized difference * signal represented in an internal floating point * format. */ char td; /* delayed tone detect, new in 1988 version */ }; #endif /* !_G72X_H */ /* * This source code is a product of Sun Microsystems, Inc. and is provided * for unrestricted use. Users may copy or modify this source code without * charge. * * SUN SOURCE CODE IS PROVIDED AS IS WITH NO WARRANTIES OF ANY KIND INCLUDING * THE WARRANTIES OF DESIGN, MERCHANTIBILITY AND FITNESS FOR A PARTICULAR * PURPOSE, OR ARISING FROM A COURSE OF DEALING, USAGE OR TRADE PRACTICE. * * Sun source code is provided with no support and without any obligation on * the part of Sun Microsystems, Inc. to assist in its use, correction, * modification or enhancement. * * SUN MICROSYSTEMS, INC. SHALL HAVE NO LIABILITY WITH RESPECT TO THE * INFRINGEMENT OF COPYRIGHTS, TRADE SECRETS OR ANY PATENTS BY THIS SOFTWARE * OR ANY PART THEREOF. * * In no event will Sun Microsystems, Inc. be liable for any lost revenue * or profits or other special, indirect and consequential damages, even if * Sun has been advised of the possibility of such damages. * * Sun Microsystems, Inc. * 2550 Garcia Avenue * Mountain View, California 94043 */ /* * g72x.c * * Common routines for G.721 and G.723 conversions. */ static short power2[15] = {1, 2, 4, 8, 0x10, 0x20, 0x40, 0x80, 0x100, 0x200, 0x400, 0x800, 0x1000, 0x2000, 0x4000}; /* * quan() * * quantizes the input val against the table of size short integers. * It returns i if table[i - 1] <= val < table[i]. * * Using linear search for simple coding. */ static int quan( int val, short *table, int size) { int i; for (i = 0; i < size; i++) if (val < *table++) break; return (i); } /* * fmult() * * returns the integer product of the 14-bit integer "an" and * "floating point" representation (4-bit exponent, 6-bit mantessa) "srn". */ static int fmult( int an, int srn) { short anmag, anexp, anmant; short wanexp, /* wanmag, */ wanmant; short retval; anmag = (an > 0) ? an : ((-an) & 0x1FFF); anexp = quan(anmag, power2, 15) - 6; anmant = (anmag == 0) ? 32 : (anexp >= 0) ? anmag >> anexp : anmag << -anexp; wanexp = anexp + ((srn >> 6) & 0xF) - 13; wanmant = (anmant * (srn & 077) + 0x30) >> 4; retval = (wanexp >= 0) ? ((wanmant << wanexp) & 0x7FFF) : (wanmant >> -wanexp); return (((an ^ srn) < 0) ? -retval : retval); } /* * g72x_init_state() * * This routine initializes and/or resets the g72x_state structure * pointed to by 'state_ptr'. * All the initial state values are specified in the CCITT G.721 document. */ static void g72x_init_state( struct g72x_state *state_ptr) { int cnta; state_ptr->yl = 34816; state_ptr->yu = 544; state_ptr->dms = 0; state_ptr->dml = 0; state_ptr->ap = 0; for (cnta = 0; cnta < 2; cnta++) { state_ptr->a[cnta] = 0; state_ptr->pk[cnta] = 0; state_ptr->sr[cnta] = 32; } for (cnta = 0; cnta < 6; cnta++) { state_ptr->b[cnta] = 0; state_ptr->dq[cnta] = 32; } state_ptr->td = 0; } /* * predictor_zero() * * computes the estimated signal from 6-zero predictor. * */ static int predictor_zero( struct g72x_state *state_ptr) { int i; int sezi; sezi = fmult(state_ptr->b[0] >> 2, state_ptr->dq[0]); for (i = 1; i < 6; i++) /* ACCUM */ sezi += fmult(state_ptr->b[i] >> 2, state_ptr->dq[i]); return (sezi); } /* * predictor_pole() * * computes the estimated signal from 2-pole predictor. * */ static int predictor_pole( struct g72x_state *state_ptr) { return (fmult(state_ptr->a[1] >> 2, state_ptr->sr[1]) + fmult(state_ptr->a[0] >> 2, state_ptr->sr[0])); } /* * step_size() * * computes the quantization step size of the adaptive quantizer. * */ static int step_size( struct g72x_state *state_ptr) { int y; int dif; int al; if (state_ptr->ap >= 256) return (state_ptr->yu); else { y = state_ptr->yl >> 6; dif = state_ptr->yu - y; al = state_ptr->ap >> 2; if (dif > 0) y += (dif * al) >> 6; else if (dif < 0) y += (dif * al + 0x3F) >> 6; return (y); } } /* * quantize() * * Given a raw sample, 'd', of the difference signal and a * quantization step size scale factor, 'y', this routine returns the * ADPCM codeword to which that sample gets quantized. The step * size scale factor division operation is done in the log base 2 domain * as a subtraction. */ static int quantize( int d, /* Raw difference signal sample */ int y, /* Step size multiplier */ short *table, /* quantization table */ int size) /* table size of short integers */ { short dqm; /* Magnitude of 'd' */ short exp; /* Integer part of base 2 log of 'd' */ short mant; /* Fractional part of base 2 log */ short dl; /* Log of magnitude of 'd' */ short dln; /* Step size scale factor normalized log */ int i; /* * LOG * * Compute base 2 log of 'd', and store in 'dl'. */ dqm = abs(d); exp = quan(dqm >> 1, power2, 15); mant = ((dqm << 7) >> exp) & 0x7F; /* Fractional portion. */ dl = (exp << 7) + mant; /* * SUBTB * * "Divide" by step size multiplier. */ dln = dl - (y >> 2); /* * QUAN * * Obtain codword i for 'd'. */ i = quan(dln, table, size); if (d < 0) /* take 1's complement of i */ return ((size << 1) + 1 - i); else if (i == 0) /* take 1's complement of 0 */ return ((size << 1) + 1); /* new in 1988 */ else return (i); } /* * reconstruct() * * Returns reconstructed difference signal 'dq' obtained from * codeword 'i' and quantization step size scale factor 'y'. * Multiplication is performed in log base 2 domain as addition. */ static int reconstruct( int sign, /* 0 for non-negative value */ int dqln, /* G.72x codeword */ int y) /* Step size multiplier */ { short dql; /* Log of 'dq' magnitude */ short dex; /* Integer part of log */ short dqt; short dq; /* Reconstructed difference signal sample */ dql = dqln + (y >> 2); /* ADDA */ if (dql < 0) { return ((sign) ? -0x8000 : 0); } else { /* ANTILOG */ dex = (dql >> 7) & 15; dqt = 128 + (dql & 127); dq = (dqt << 7) >> (14 - dex); return ((sign) ? (dq - 0x8000) : dq); } } /* * update() * * updates the state variables for each output code */ static void update( int code_size, /* distinguish 723_40 with others */ int y, /* quantizer step size */ int wi, /* scale factor multiplier */ int fi, /* for long/short term energies */ int dq, /* quantized prediction difference */ int sr, /* reconstructed signal */ int dqsez, /* difference from 2-pole predictor */ struct g72x_state *state_ptr) /* coder state pointer */ { int cnt; short mag, exp /* , mant */ ; /* Adaptive predictor, FLOAT A */ short a2p = 0; /* LIMC */ short a1ul; /* UPA1 */ short /* ua2, */ pks1; /* UPA2 */ short /* uga2a, */ fa1; /* short uga2b; */ char tr; /* tone/transition detector */ short ylint, thr2, dqthr; short ylfrac, thr1; short pk0; pk0 = (dqsez < 0) ? 1 : 0; /* needed in updating predictor poles */ mag = dq & 0x7FFF; /* prediction difference magnitude */ /* TRANS */ ylint = state_ptr->yl >> 15; /* exponent part of yl */ ylfrac = (state_ptr->yl >> 10) & 0x1F; /* fractional part of yl */ thr1 = (32 + ylfrac) << ylint; /* threshold */ thr2 = (ylint > 9) ? 31 << 10 : thr1; /* limit thr2 to 31 << 10 */ dqthr = (thr2 + (thr2 >> 1)) >> 1; /* dqthr = 0.75 * thr2 */ if (state_ptr->td == 0) /* signal supposed voice */ tr = 0; else if (mag <= dqthr) /* supposed data, but small mag */ tr = 0; /* treated as voice */ else /* signal is data (modem) */ tr = 1; /* * Quantizer scale factor adaptation. */ /* FUNCTW & FILTD & DELAY */ /* update non-steady state step size multiplier */ state_ptr->yu = y + ((wi - y) >> 5); /* LIMB */ if (state_ptr->yu < 544) /* 544 <= yu <= 5120 */ state_ptr->yu = 544; else if (state_ptr->yu > 5120) state_ptr->yu = 5120; /* FILTE & DELAY */ /* update steady state step size multiplier */ state_ptr->yl += state_ptr->yu + ((-state_ptr->yl) >> 6); /* * Adaptive predictor coefficients. */ if (tr == 1) { /* reset a's and b's for modem signal */ state_ptr->a[0] = 0; state_ptr->a[1] = 0; state_ptr->b[0] = 0; state_ptr->b[1] = 0; state_ptr->b[2] = 0; state_ptr->b[3] = 0; state_ptr->b[4] = 0; state_ptr->b[5] = 0; } else { /* update a's and b's */ pks1 = pk0 ^ state_ptr->pk[0]; /* UPA2 */ /* update predictor pole a[1] */ a2p = state_ptr->a[1] - (state_ptr->a[1] >> 7); if (dqsez != 0) { fa1 = (pks1) ? state_ptr->a[0] : -state_ptr->a[0]; if (fa1 < -8191) /* a2p = function of fa1 */ a2p -= 0x100; else if (fa1 > 8191) a2p += 0xFF; else a2p += fa1 >> 5; if (pk0 ^ state_ptr->pk[1]) /* LIMC */ if (a2p <= -12160) a2p = -12288; else if (a2p >= 12416) a2p = 12288; else a2p -= 0x80; else if (a2p <= -12416) a2p = -12288; else if (a2p >= 12160) a2p = 12288; else a2p += 0x80; } /* TRIGB & DELAY */ state_ptr->a[1] = a2p; /* UPA1 */ /* update predictor pole a[0] */ state_ptr->a[0] -= state_ptr->a[0] >> 8; if (dqsez != 0) { if (pks1 == 0) { state_ptr->a[0] += 192; } else { state_ptr->a[0] -= 192; } } /* LIMD */ a1ul = 15360 - a2p; if (state_ptr->a[0] < -a1ul) state_ptr->a[0] = -a1ul; else if (state_ptr->a[0] > a1ul) state_ptr->a[0] = a1ul; /* UPB : update predictor zeros b[6] */ for (cnt = 0; cnt < 6; cnt++) { if (code_size == 5) /* for 40Kbps G.723 */ state_ptr->b[cnt] -= state_ptr->b[cnt] >> 9; else /* for G.721 and 24Kbps G.723 */ state_ptr->b[cnt] -= state_ptr->b[cnt] >> 8; if (dq & 0x7FFF) { /* XOR */ if ((dq ^ state_ptr->dq[cnt]) >= 0) state_ptr->b[cnt] += 128; else state_ptr->b[cnt] -= 128; } } } for (cnt = 5; cnt > 0; cnt--) state_ptr->dq[cnt] = state_ptr->dq[cnt-1]; /* FLOAT A : convert dq[0] to 4-bit exp, 6-bit mantissa f.p. */ if (mag == 0) { state_ptr->dq[0] = (dq >= 0) ? 0x20 : 0xFC20; } else { exp = quan(mag, power2, 15); state_ptr->dq[0] = (dq >= 0) ? (exp << 6) + ((mag << 6) >> exp) : (exp << 6) + ((mag << 6) >> exp) - 0x400; } state_ptr->sr[1] = state_ptr->sr[0]; /* FLOAT B : convert sr to 4-bit exp., 6-bit mantissa f.p. */ if (sr == 0) { state_ptr->sr[0] = 0x20; } else if (sr > 0) { exp = quan(sr, power2, 15); state_ptr->sr[0] = (exp << 6) + ((sr << 6) >> exp); } else if (sr > -32768) { mag = -sr; exp = quan(mag, power2, 15); state_ptr->sr[0] = (exp << 6) + ((mag << 6) >> exp) - 0x400; } else state_ptr->sr[0] = (unsigned short) 0xFC20; /* DELAY A */ state_ptr->pk[1] = state_ptr->pk[0]; state_ptr->pk[0] = pk0; /* TONE */ if (tr == 1) /* this sample has been treated as data */ state_ptr->td = 0; /* next one will be treated as voice */ else if (a2p < -11776) /* small sample-to-sample correlation */ state_ptr->td = 1; /* signal may be data */ else /* signal is voice */ state_ptr->td = 0; /* * Adaptation speed control. */ state_ptr->dms += (fi - state_ptr->dms) >> 5; /* FILTA */ state_ptr->dml += (((fi << 2) - state_ptr->dml) >> 7); /* FILTB */ if (tr == 1) state_ptr->ap = 256; else if (y < 1536) /* SUBTC */ state_ptr->ap += (0x200 - state_ptr->ap) >> 4; else if (state_ptr->td == 1) state_ptr->ap += (0x200 - state_ptr->ap) >> 4; else if (abs((state_ptr->dms << 2) - state_ptr->dml) >= (state_ptr->dml >> 3)) state_ptr->ap += (0x200 - state_ptr->ap) >> 4; else state_ptr->ap += (-state_ptr->ap) >> 4; } /* * This source code is a product of Sun Microsystems, Inc. and is provided * for unrestricted use. Users may copy or modify this source code without * charge. * * SUN SOURCE CODE IS PROVIDED AS IS WITH NO WARRANTIES OF ANY KIND INCLUDING * THE WARRANTIES OF DESIGN, MERCHANTIBILITY AND FITNESS FOR A PARTICULAR * PURPOSE, OR ARISING FROM A COURSE OF DEALING, USAGE OR TRADE PRACTICE. * * Sun source code is provided with no support and without any obligation on * the part of Sun Microsystems, Inc. to assist in its use, correction, * modification or enhancement. * * SUN MICROSYSTEMS, INC. SHALL HAVE NO LIABILITY WITH RESPECT TO THE * INFRINGEMENT OF COPYRIGHTS, TRADE SECRETS OR ANY PATENTS BY THIS SOFTWARE * OR ANY PART THEREOF. * * In no event will Sun Microsystems, Inc. be liable for any lost revenue * or profits or other special, indirect and consequential damages, even if * Sun has been advised of the possibility of such damages. * * Sun Microsystems, Inc. * 2550 Garcia Avenue * Mountain View, California 94043 */ /* * g721.c * * Description: * * g721_encoder(), g721_decoder() * * These routines comprise an implementation of the CCITT G.721 ADPCM * coding algorithm. Essentially, this implementation is identical to * the bit level description except for a few deviations which * take advantage of work station attributes, such as hardware 2's * complement arithmetic and large memory. Specifically, certain time * consuming operations such as multiplications are replaced * with lookup tables and software 2's complement operations are * replaced with hardware 2's complement. * * The deviation from the bit level specification (lookup tables) * preserves the bit level performance specifications. * * As outlined in the G.721 Recommendation, the algorithm is broken * down into modules. Each section of code below is preceded by * the name of the module which it is implementing. * */ static short qtab_721[7] = {-124, 80, 178, 246, 300, 349, 400}; /* * Maps G.721 code word to reconstructed scale factor normalized log * magnitude values. */ static short _dqlntab[16] = {-2048, 4, 135, 213, 273, 323, 373, 425, 425, 373, 323, 273, 213, 135, 4, -2048}; /* Maps G.721 code word to log of scale factor multiplier. */ static short _witab[16] = {-12, 18, 41, 64, 112, 198, 355, 1122, 1122, 355, 198, 112, 64, 41, 18, -12}; /* * Maps G.721 code words to a set of values whose long and short * term averages are computed and then compared to give an indication * how stationary (steady state) the signal is. */ static short _fitab[16] = {0, 0, 0, 0x200, 0x200, 0x200, 0x600, 0xE00, 0xE00, 0x600, 0x200, 0x200, 0x200, 0, 0, 0}; /* * g721_encoder() * * Encodes the input vale of linear PCM, A-law or u-law data sl and returns * the resulting code. -1 is returned for unknown input coding value. */ static int g721_encoder( int sl, int in_coding, struct g72x_state *state_ptr) { short sezi, se, sez; /* ACCUM */ short d; /* SUBTA */ short sr; /* ADDB */ short y; /* MIX */ short dqsez; /* ADDC */ short dq, i; switch (in_coding) { /* linearize input sample to 14-bit PCM */ case AUDIO_ENCODING_LINEAR: sl >>= 2; /* 14-bit dynamic range */ break; default: return (-1); } sezi = predictor_zero(state_ptr); sez = sezi >> 1; se = (sezi + predictor_pole(state_ptr)) >> 1; /* estimated signal */ d = sl - se; /* estimation difference */ /* quantize the prediction difference */ y = step_size(state_ptr); /* quantizer step size */ i = quantize(d, y, qtab_721, 7); /* i = ADPCM code */ dq = reconstruct(i & 8, _dqlntab[i], y); /* quantized est diff */ sr = (dq < 0) ? se - (dq & 0x3FFF) : se + dq; /* reconst. signal */ dqsez = sr + sez - se; /* pole prediction diff. */ update(4, y, _witab[i] << 5, _fitab[i], dq, sr, dqsez, state_ptr); return (i); } /* * g721_decoder() * * Description: * * Decodes a 4-bit code of G.721 encoded data of i and * returns the resulting linear PCM, A-law or u-law value. * return -1 for unknown out_coding value. */ static int g721_decoder( int i, int out_coding, struct g72x_state *state_ptr) { short sezi, sei, sez, se; /* ACCUM */ short y; /* MIX */ short sr; /* ADDB */ short dq; short dqsez; i &= 0x0f; /* mask to get proper bits */ sezi = predictor_zero(state_ptr); sez = sezi >> 1; sei = sezi + predictor_pole(state_ptr); se = sei >> 1; /* se = estimated signal */ y = step_size(state_ptr); /* dynamic quantizer step size */ dq = reconstruct(i & 0x08, _dqlntab[i], y); /* quantized diff. */ sr = (dq < 0) ? (se - (dq & 0x3FFF)) : se + dq; /* reconst. signal */ dqsez = sr - se + sez; /* pole prediction diff. */ update(4, y, _witab[i] << 5, _fitab[i], dq, sr, dqsez, state_ptr); switch (out_coding) { case AUDIO_ENCODING_LINEAR: return (sr << 2); /* sr was 14-bit dynamic range */ default: return (-1); } } static int pack_output( unsigned code, int bits, FILE* fd_out) { static unsigned int out_buffer = 0; static int out_bits = 0; unsigned char out_byte; out_buffer |= (code << out_bits); out_bits += bits; if (out_bits >= 8) { out_byte = out_buffer & 0xff; out_bits -= 8; out_buffer >>= 8; fwrite(&out_byte, sizeof (char), 1, fd_out); } return (out_bits > 0); } static int unpack_input( unsigned char *code, int bits, FILE* fd_in) { static unsigned int in_buffer = 0; static int in_bits = 0; unsigned char in_byte; if (in_bits < bits) { if (fread(&in_byte, sizeof (char), 1, fd_in) != 1) { *code = 0; return (-1); } in_buffer |= (in_byte << in_bits); in_bits += 8; } *code = in_buffer & ((1 << bits) - 1); in_buffer >>= bits; in_bits -= bits; return (in_bits > 0); } int pvftousradpcm (FILE *fd_in, FILE *fd_out, pvf_header *header_in) { struct g72x_state state; int r = 0; int sample = 0; unsigned char code = 0; g72x_init_state(&state); /* GSM operates on frames of 160 samples each, so we may run up against * the end of the file and be forced to backfill with zeroes. */ while (!feof(fd_in)) { sample = header_in->read_pvf_data(fd_in); sample >>= 8; if (sample > 0x7fff) sample = 0x7fff; if (sample < -0x8000) sample = -0x8000; code = g721_encoder(sample, AUDIO_ENCODING_LINEAR, &state); r = pack_output(code, 4, fd_out); } while (r) { r = pack_output(0, 4, fd_out); } return(OK); } static int usradpcmtopvf (FILE *fd_in, FILE *fd_out, pvf_header *header_out) { int sample; struct g72x_state state; unsigned char code = 0; while (unpack_input(&code, 4, fd_in) >= 0) { sample = g721_decoder(code, AUDIO_ENCODING_LINEAR, &state); header_out->write_pvf_data(fd_out, sample << 8); } return(OK); }