/*
 * CDDL HEADER START
 *
 * The contents of this file are subject to the terms of the
 * Common Development and Distribution License, Version 1.0 only
 * (the "License").  You may not use this file except in compliance
 * with the License.
 *
 * You can obtain a copy of the license at usr/src/OPENSOLARIS.LICENSE
 * or http://www.opensolaris.org/os/licensing.
 * See the License for the specific language governing permissions
 * and limitations under the License.
 *
 * When distributing Covered Code, include this CDDL HEADER in each
 * file and include the License file at usr/src/OPENSOLARIS.LICENSE.
 * If applicable, add the following below this CDDL HEADER, with the
 * fields enclosed by brackets "[]" replaced with your own identifying
 * information: Portions Copyright [yyyy] [name of copyright owner]
 *
 * CDDL HEADER END
 */
/*
 * Copyright (c) 1992-2001 by Sun Microsystems, Inc.
 * All rights reserved.
 */

/*
 * Description:
 *
 * g723_init_state(), g723_encode(), g723_decode()
 *
 * These routines comprise an implementation of the CCITT G.723 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 look up tables and software 2's complement operations are
 * replaced with hardware 2's complement.
 *
 * The deviation (look up tables) from the bit level
 * specification, preserves the bit level performance specifications.
 *
 * As outlined in the G.723 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.
 *
 */
#include <stdlib.h>
#include <libaudio.h>

/*
 * g723_tables.c
 *
 * Description:
 *
 * This file contains statically defined lookup tables for
 * use with the G.723 coding routines.
 */

/*
 * Maps G.723 code word to reconstructed scale factor normalized log
 * magnitude values.
 */
static short    _dqlntab[8] = {-2048, 135, 273, 373, 373, 273, 135, -2048};

/* Maps G.723 code word to log of scale factor multiplier. */
static short    _witab[8] = {-128, 960, 4384, 18624, 18624, 4384, 960, -128};

/*
 * Maps G.723 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[8] = {0, 0x200, 0x400, 0xE00, 0xE00, 0x400, 0x200, 0};

/*
 * g723_init_state()
 *
 * Description:
 *
 * This routine initializes and/or resets the audio_encode_state structure
 * pointed to by 'state_ptr'.
 * All the state initial values are specified in the G.723 standard specs.
 */
void
g723_init_state(
        struct audio_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;
        state_ptr->leftover_cnt = 0;            /* no left over codes */
}

/*
 * _g723_fmult()
 *
 * returns the integer product of the "floating point" an and srn
 * by the lookup table _fmultwanmant[].
 *
 */
static int
_g723_fmult(
                int an,
                int srn)
{
        short   anmag, anexp, anmant;
        short   wanexp;

        if (an == 0) {
                return ((srn >= 0) ?
                    ((srn & 077) + 1) >> (18 - (srn >> 6)) :
                    -(((srn & 077) + 1) >> (2 - (srn >> 6))));
        } else if (an > 0) {
                anexp = _fmultanexp[an] - 12;
                anmant = ((anexp >= 0) ? an >> anexp : an << -anexp) & 07700;
                if (srn >= 0) {
                        wanexp = anexp + (srn >> 6) - 7;
                        return ((wanexp >= 0) ?
                            (_fmultwanmant[(srn & 077) + anmant] << wanexp)
                            & 0x7FFF :
                            _fmultwanmant[(srn & 077) + anmant] >> -wanexp);
                } else {
                        wanexp = anexp + (srn >> 6) - 0xFFF7;
                        return ((wanexp >= 0) ?
                            -((_fmultwanmant[(srn & 077) + anmant] << wanexp)
                            & 0x7FFF) :
                            -(_fmultwanmant[(srn & 077) + anmant] >> -wanexp));
                }
        } else {
                anmag = (-an) & 0x1FFF;
                anexp = _fmultanexp[anmag] - 12;
                anmant = ((anexp >= 0) ? anmag >> anexp : anmag << -anexp)
                    & 07700;
                if (srn >= 0) {
                        wanexp = anexp + (srn >> 6) - 7;
                        return ((wanexp >= 0) ?
                            -((_fmultwanmant[(srn & 077) + anmant] << wanexp)
                            & 0x7FFF) :
                            -(_fmultwanmant[(srn & 077) + anmant] >> -wanexp));
                } else {
                        wanexp = anexp + (srn >> 6) - 0xFFF7;
                        return ((wanexp >= 0) ?
                            (_fmultwanmant[(srn & 077) + anmant] << wanexp)
                            & 0x7FFF :
                            _fmultwanmant[(srn & 077) + anmant] >> -wanexp);
                }
        }

}

/*
 * _g723_update()
 *
 * updates the state variables for each output code
 *
 */
static void
_g723_update(
        int     y,
        int     i,
        int     dq,
        int     sr,
        int     pk0,
        struct audio_g72x_state *state_ptr,
        int     sigpk)
{
        int     cnt;
        long    fi;                     /* Adaptation speed control, FUNCTF */
        short   mag, exp;               /* Adaptive predictor, FLOAT A */
        short   a2p;                    /* LIMC */
        short   a1ul;                   /* UPA1 */
        short   pks1, fa1;              /* UPA2 */
        char    tr;                     /* tone/transition detector */
        short   thr2;

        mag = dq & 0x3FFF;
        /* TRANS */
        if (state_ptr->td == 0)
                tr = 0;
        else if (state_ptr->yl > 0x40000)
                tr = (mag <= 0x2F80) ? 0 : 1;
        else {
                thr2 = (0x20 + ((state_ptr->yl >> 10) & 0x1F)) <<
                    (state_ptr->yl >> 15);
                if (mag >= thr2)
                        tr = 1;
                else
                        tr = (mag <= (thr2 - (thr2 >> 2))) ? 0 : 1;
        }

        /*
         * Quantizer scale factor adaptation.
         */

        /* FUNCTW & FILTD & DELAY */
        state_ptr->yu = y + ((_witab[i] - y) >> 5);

        /* LIMB */
        if (state_ptr->yu < 544)
                state_ptr->yu = 544;
        else if (state_ptr->yu > 5120)
                state_ptr->yu = 5120;

        /* FILTE & DELAY */
        state_ptr->yl += state_ptr->yu + ((-state_ptr->yl) >> 6);

        /*
         * Adaptive predictor coefficients.
         */
        if (tr == 1) {
                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 {

                /* UPA2 */
                pks1 = pk0 ^ state_ptr->pk[0];

                a2p = state_ptr->a[1] - (state_ptr->a[1] >> 7);
                if (sigpk == 0) {
                        fa1 = (pks1) ? state_ptr->a[0] : -state_ptr->a[0];
                        if (fa1 < -8191)
                                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 */
                state_ptr->a[0] -= state_ptr->a[0] >> 8;
                if (sigpk == 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 of b's */
                for (cnt = 0; cnt < 6; cnt++) {
                        state_ptr->b[cnt] -= state_ptr->b[cnt] >> 8;
                        if (dq & 0x3FFF) {
                                /* 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 */
        if (mag == 0) {
                state_ptr->dq[0] = (dq >= 0) ? 0x20 : 0xFC20;
        } else {
                exp = _fmultanexp[mag];
                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 */
        if (sr == 0) {
                state_ptr->sr[0] = 0x20;
        } else if (sr > 0) {
                exp = _fmultanexp[sr];
                state_ptr->sr[0] = (exp << 6) + ((sr << 6) >> exp);
        } else {
                mag = -sr;
                exp = _fmultanexp[mag];
                state_ptr->sr[0] =  (exp << 6) + ((mag << 6) >> exp) - 0x400;
        }

        /* DELAY A */
        state_ptr->pk[1] = state_ptr->pk[0];
        state_ptr->pk[0] = pk0;

        /* TONE */
        if (tr == 1)
                state_ptr->td = 0;
        else if (a2p < -11776)
                state_ptr->td = 1;
        else
                state_ptr->td = 0;

        /*
         * Adaptation speed control.
         */
        fi = _fitab[i];                                         /* FUNCTF */
        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;
}

/*
 * _g723_quantize()
 *
 * Description:
 *
 * Given a raw sample, 'd', of the difference signal and a
 * quantization step size scale factor, 'y', this routine returns the
 * G.723 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 unsigned int
_g723_quantize(
        int     d,      /* Raw difference signal sample. */
        int     y)      /* Step size multiplier. */
{
        /* LOG */
        short   dqm;    /* Magnitude of 'd'. */
        short   exp;    /* Integer part of base 2 log of magnitude of 'd'. */
        short   mant;   /* Fractional part of base 2 log. */
        short   dl;     /* Log of magnitude of 'd'. */

        /* SUBTB */
        short   dln;    /* Step size scale factor normalized log. */

        /* QUAN */
        unsigned char   i;      /* G.723 codeword. */

        /*
         * LOG
         *
         * Compute base 2 log of 'd', and store in 'dln'.
         *
         */
        dqm = abs(d);
        exp = _fmultanexp[dqm >> 1];
        mant = ((dqm << 7) >> exp) & 0x7F;      /* Fractional portion. */
        dl = (exp << 7) + mant;

        /*
         * SUBTB
         *
         * "Divide" by step size multiplier.
         */
        dln = dl - (y >> 2);

        /*
         * QUAN
         *
         * Obtain codword for 'd'.
         */
        i = _g723quani[dln & 0xFFF];
        if (d < 0)
                i ^= 7;         /* Stuff in sign of 'd'. */
        else if (i == 0)
                i = 7;          /* New in 1988 revision */

        return (i);
}

/*
 * _g723_reconstr()
 *
 * Description:
 *
 * Returns reconstructed difference signal 'dq' obtained from
 * G.723 codeword 'i' and quantization step size scale factor 'y'.
 * Multiplication is performed in log base 2 domain as addition.
 */
static int
_g723_reconstr(
        int             i,      /* G.723 codeword. */
        unsigned long   y)      /* Step size multiplier. */
{
        /* ADD A */
        short   dql;    /* Log of 'dq' magnitude. */

        /* ANTILOG */
        short   dex;    /* Integer part of log. */
        short   dqt;
        short   dq;     /* Reconstructed difference signal sample. */


        dql = _dqlntab[i] + (y >> 2);   /* ADDA */

        if (dql < 0)
                dq = 0;
        else {                          /* ANTILOG */
                dex = (dql >> 7) & 15;
                dqt = 128 + (dql & 127);
                dq = (dqt << 7) >> (14 - dex);
        }
        if (i & 4)
                dq -= 0x8000;

        return (dq);
}

/*
 * _tandem_adjust(sr, se, y, i)
 *
 * Description:
 *
 * At the end of ADPCM decoding, it simulates an encoder which may be receiving
 * the output of this decoder as a tandem process. If the output of the
 * simulated encoder differs from the input to this decoder, the decoder output
 * is adjusted by one level of A-law or Mu-law codes.
 *
 * Input:
 *      sr      decoder output linear PCM sample,
 *      se      predictor estimate sample,
 *      y       quantizer step size,
 *      i       decoder input code
 *
 * Return:
 *      adjusted A-law or Mu-law compressed sample.
 */
static int
_tandem_adjust_alaw(
        int     sr,     /* decoder output linear PCM sample */
        int     se,     /* predictor estimate sample */
        int     y,      /* quantizer step size */
        int     i)      /* decoder input code */
{
        unsigned char   sp;     /* A-law compressed 8-bit code */
        short   dx;             /* prediction error */
        char    id;             /* quantized prediction error */
        int     sd;             /* adjusted A-law decoded sample value */
        int     im;             /* biased magnitude of i */
        int     imx;            /* biased magnitude of id */

        sp = audio_s2a((sr <= -0x2000)? -0x8000 :
            (sr < 0x1FFF)? sr << 2 : 0x7FFF); /* short to A-law compression */
        dx = (audio_a2s(sp) >> 2) - se;  /* 16-bit prediction error */
        id = _g723_quantize(dx, y);

        if (id == i)                    /* no adjustment on sp */
                return (sp);
        else {                          /* sp adjustment needed */
                im = i ^ 4;             /* 2's complement to biased unsigned */
                imx = id ^ 4;

                if (imx > im) {         /* sp adjusted to next lower value */
                        if (sp & 0x80)
                                sd = (sp == 0xD5)? 0x55 :
                                    ((sp ^ 0x55) - 1) ^ 0x55;
                        else
                                sd = (sp == 0x2A)? 0x2A :
                                    ((sp ^ 0x55) + 1) ^ 0x55;
                } else {        /* sp adjusted to next higher value */
                        if (sp & 0x80)
                                sd = (sp == 0xAA)? 0xAA :
                                    ((sp ^ 0x55) + 1) ^ 0x55;
                        else
                                sd = (sp == 0x55)? 0xD5 :
                                    ((sp ^ 0x55) - 1) ^ 0x55;
                }
                return (sd);
        }
}

static int
_tandem_adjust_ulaw(
        int     sr,             /* decoder output linear PCM sample */
        int     se,             /* predictor estimate sample */
        int     y,              /* quantizer step size */
        int     i)              /* decoder input code */
{
        unsigned char   sp;     /* A-law compressed 8-bit code */
        short   dx;             /* prediction error */
        char    id;             /* quantized prediction error */
        int     sd;             /* adjusted A-law decoded sample value */
        int     im;             /* biased magnitude of i */
        int     imx;            /* biased magnitude of id */

        sp = audio_s2u((sr <= -0x2000)? -0x8000 :
            (sr >= 0x1FFF)? 0x7FFF : sr << 2); /* short to u-law compression */
        dx = (audio_u2s(sp) >> 2) - se;  /* 16-bit prediction error */
        id = _g723_quantize(dx, y);
        if (id == i)
                return (sp);
        else {
                /* ADPCM codes : 8, 9, ... F, 0, 1, ... , 6, 7 */
                im = i ^ 4;             /* 2's complement to biased unsigned */
                imx = id ^ 4;

                /* u-law codes : 0, 1, ... 7E, 7F, FF, FE, ... 81, 80 */
                if (imx > im) {         /* sp adjusted to next lower value */
                        if (sp & 0x80)
                                sd = (sp == 0xFF)? 0x7E : sp + 1;
                        else
                                sd = (sp == 0)? 0 : sp - 1;

                } else {                /* sp adjusted to next higher value */
                        if (sp & 0x80)
                                sd = (sp == 0x80)? 0x80 : sp - 1;
                        else
                                sd = (sp == 0x7F)? 0xFE : sp + 1;
                }
                return (sd);
        }
}

static unsigned char
_encoder(
        int             sl,
        struct audio_g72x_state *state_ptr)
{
        short   sei, sezi, se, sez;     /* ACCUM */
        short   d;                      /* SUBTA */
        float   al;             /* use floating point for faster multiply */
        short   y, dif;                 /* MIX */
        short   sr;                     /* ADDB */
        short   pk0, sigpk, dqsez;      /* ADDC */
        short   dq, i;
        int     cnt;

        /* ACCUM */
        sezi = _g723_fmult(state_ptr->b[0] >> 2, state_ptr->dq[0]);
        for (cnt = 1; cnt < 6; cnt++)
                sezi = sezi + _g723_fmult(state_ptr->b[cnt] >> 2,
                    state_ptr->dq[cnt]);
        sei = sezi;
        for (cnt = 1; cnt > -1; cnt--)
                sei = sei + _g723_fmult(state_ptr->a[cnt] >> 2,
                    state_ptr->sr[cnt]);
        sez = sezi >> 1;
        se = sei >> 1;

        d = sl - se;                                    /* SUBTA */

        if (state_ptr->ap >= 256)
                y = state_ptr->yu;
        else {
                y = state_ptr->yl >> 6;
                dif = state_ptr->yu - y;
                al = state_ptr->ap >> 2;
                if (dif > 0)
                        y += ((int)(dif * al)) >> 6;
                else if (dif < 0)
                        y += ((int)(dif * al) + 0x3F) >> 6;
        }

        i = _g723_quantize(d, y);
        dq = _g723_reconstr(i, y);

        sr = (dq < 0) ? se - (dq & 0x3FFF) : se + dq;   /* ADDB */

        dqsez = sr + sez - se;                          /* ADDC */
        if (dqsez == 0) {
                pk0 = 0;
                sigpk = 1;
        } else {
                pk0 = (dqsez < 0) ? 1 : 0;
                sigpk = 0;
        }

        _g723_update(y, i, dq, sr, pk0, state_ptr, sigpk);

        return (i);
}

/*
 * g723_encode()
 *
 * Description:
 *
 * Encodes a buffer of linear PCM, A-law or Mu-law data pointed to by 'in_buf'
 * according the G.723 encoding algorithm and packs the resulting code words
 * into bytes. The bytes of codewords are written to a buffer
 * pointed to by 'out_buf'.
 *
 * Notes:
 *
 * In the event that the number packed codes is shorter than a sample unit,
 * the remainder is saved in the state stucture till next call.  It is then
 * packed into the new buffer on the next call.
 * The number of valid bytes in 'out_buf' is returned in *out_size.  Note that
 * this will not always be equal to 3/8 of 'data_size' on input. On the
 * final call to 'g723_encode()' the calling program might want to
 * check if any code bits was left over.  This can be
 * done by calling 'g723_encode()' with data_size = 0, which returns in
 * *out_size a* 0 if nothing was leftover and the number of bits left over in
 * the state structure which now is in out_buf[0].
 *
 * The 3 lower significant bits of an individual byte in the output byte
 * stream is packed with a G.723 code first.  Then the 3 higher order
 * bits are packed with the next code.
 */
int
g723_encode(
        void            *in_buf,
        int             data_size,
        Audio_hdr       *in_header,
        unsigned char   *out_buf,
        int             *out_size,
        struct audio_g72x_state *state_ptr)
{
        int             i;
        unsigned char   *out_ptr;
        unsigned char   *leftover;
        unsigned int    bits;
        unsigned int    codes;
        int             offset;
        short           *short_ptr;
        unsigned char   *char_ptr;

        /* Dereference the array pointer for faster access */
        leftover = &state_ptr->leftover[0];

        /* Return all cached leftovers */
        if (data_size == 0) {
                for (i = 0; state_ptr->leftover_cnt > 0; i++) {
                        *out_buf++ = leftover[i];
                        state_ptr->leftover_cnt -= 8;
                }
                if (i > 0) {
                        /* Round up to a complete sample unit */
                        for (; i < 3; i++)
                                *out_buf++ = 0;
                }
                *out_size = i;
                state_ptr->leftover_cnt = 0;
                return (AUDIO_SUCCESS);
        }

        /* XXX - if linear, it had better be 16-bit! */
        if (in_header->encoding == AUDIO_ENCODING_LINEAR) {
                if (data_size & 1) {
                        return (AUDIO_ERR_BADFRAME);
                } else {
                        data_size >>= 1;
                        short_ptr = (short *)in_buf;
                }
        } else {
                char_ptr = (unsigned char *)in_buf;
        }
        out_ptr = (unsigned char *)out_buf;

        offset = state_ptr->leftover_cnt / 8;
        bits = state_ptr->leftover_cnt % 8;
        codes = (bits > 0) ? leftover[offset] : 0;

        while (data_size--) {
                switch (in_header->encoding) {
                case AUDIO_ENCODING_LINEAR:
                        i = _encoder(*short_ptr++ >> 2, state_ptr);
                        break;
                case AUDIO_ENCODING_ALAW:
                        i = _encoder(audio_a2s(*char_ptr++) >> 2, state_ptr);
                        break;
                case AUDIO_ENCODING_ULAW:
                        i = _encoder(audio_u2s(*char_ptr++) >> 2, state_ptr);
                        break;
                default:
                        return (AUDIO_ERR_ENCODING);
                }
                /* pack the resulting code into leftover buffer */
                codes += i << bits;
                bits += 3;
                if (bits >= 8) {
                        leftover[offset] = codes & 0xff;
                        bits -= 8;
                        codes >>= 8;
                        offset++;
                }
                state_ptr->leftover_cnt += 3;

                /* got a whole sample unit so copy it out and reset */
                if (bits == 0) {
                        *out_ptr++ = leftover[0];
                        *out_ptr++ = leftover[1];
                        *out_ptr++ = leftover[2];
                        codes = 0;
                        state_ptr->leftover_cnt = 0;
                        offset = 0;
                }
        }
        /* If any residual bits, save them for the next call */
        if (bits > 0) {
                leftover[offset] = codes & 0xff;
                state_ptr->leftover_cnt += bits;
        }
        *out_size = (out_ptr - (unsigned char *)out_buf);
        return (AUDIO_SUCCESS);
}

/*
 * g723_decode()
 *
 * Description:
 *
 * Decodes a buffer of G.723 encoded data pointed to by 'in_buf' and
 * writes the resulting linear PCM, A-law or Mu-law words into a buffer
 * pointed to by 'out_buf'.
 *
 */
int
g723_decode(
        unsigned char   *in_buf,        /* Buffer of g723 encoded data. */
        int             data_size,      /* Size in bytes of in_buf. */
        Audio_hdr       *out_header,
        void            *out_buf,       /* Decoded data buffer. */
        int             *out_size,
        struct audio_g72x_state *state_ptr) /* the decoder's state structure. */
{
        unsigned char   *inbuf_end;
        unsigned char   *in_ptr, *out_ptr;
        short           *linear_ptr;
        unsigned int    codes;
        unsigned int    bits;
        int             cnt;

        short   sezi, sei, sez, se;             /* ACCUM */
        float   al;             /* use floating point for faster multiply */
        short   y, dif;                         /* MIX */
        short   sr;                             /* ADDB */
        char    pk0;                            /* ADDC */
        short   dq;
        char    sigpk;
        short   dqsez;
        unsigned char i;

        in_ptr = in_buf;
        inbuf_end = in_buf + data_size;
        out_ptr = (unsigned char *)out_buf;
        linear_ptr = (short *)out_buf;

        /* Leftovers in decoding are only up to 8 bits */
        bits = state_ptr->leftover_cnt;
        codes = (bits > 0) ? state_ptr->leftover[0] : 0;

        while ((bits >= 3) || (in_ptr < (unsigned char *)inbuf_end)) {
                if (bits < 3) {
                        codes += *in_ptr++ << bits;
                        bits += 8;
                }

                /* ACCUM */
                sezi = _g723_fmult(state_ptr->b[0] >> 2, state_ptr->dq[0]);
                for (cnt = 1; cnt < 6; cnt++)
                        sezi = sezi + _g723_fmult(state_ptr->b[cnt] >> 2,
                            state_ptr->dq[cnt]);
                sei = sezi;
                for (cnt = 1; cnt >= 0; cnt--)
                        sei = sei + _g723_fmult(state_ptr->a[cnt] >> 2,
                            state_ptr->sr[cnt]);

                sez = sezi >> 1;
                se = sei >> 1;
                if (state_ptr->ap >= 256)
                        y = state_ptr->yu;
                else {
                        y = state_ptr->yl >> 6;
                        dif = state_ptr->yu - y;
                        al = state_ptr->ap >> 2;
                        if (dif > 0)
                                y += ((int)(dif * al)) >> 6;
                        else if (dif < 0)
                                y += ((int)(dif * al) + 0x3F) >> 6;
                }

                i = codes & 7;
                dq = _g723_reconstr(i, y);
                /* ADDB */
                if (dq < 0)
                        sr = se - (dq & 0x3FFF);
                else
                        sr = se + dq;


                dqsez = sr - se + sez;                  /* ADDC */
                pk0 = (dqsez < 0) ? 1 : 0;
                sigpk = (dqsez) ? 0 : 1;

                _g723_update(y, i, dq, sr, pk0, state_ptr, sigpk);

                switch (out_header->encoding) {
                case AUDIO_ENCODING_LINEAR:
                        *linear_ptr++ = ((sr <= -0x2000) ? -0x8000 :
                            (sr >= 0x1FFF) ? 0x7FFF : sr << 2);
                        break;
                case AUDIO_ENCODING_ALAW:
                        *out_ptr++ = _tandem_adjust_alaw(sr, se, y, i);
                        break;
                case AUDIO_ENCODING_ULAW:
                        *out_ptr++ = _tandem_adjust_ulaw(sr, se, y, i);
                        break;
                default:
                        return (AUDIO_ERR_ENCODING);
                }
                codes >>= 3;
                bits -= 3;
        }
        state_ptr->leftover_cnt = bits;
        if (bits > 0)
                state_ptr->leftover[0] = codes;

        /* Calculate number of samples returned */
        if (out_header->encoding == AUDIO_ENCODING_LINEAR)
                *out_size = linear_ptr - (short *)out_buf;
        else
                *out_size = out_ptr - (unsigned char *)out_buf;

        return (AUDIO_SUCCESS);
}