/* * 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. */ #pragma ident "%Z%%M% %I% %E% SMI" /* * * Description: * * g721_encode(), g721_decode(), g721_set_law() * * 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 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.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. * */ #include #include /* * 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 long _witab[16] = {-384, 576, 1312, 2048, 3584, 6336, 11360, 35904, 35904, 11360, 6336, 3584, 2048, 1312, 576, -384}; /* * 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_init_state() * * Description: * * This routine initializes and/or resets the audio_g72x_state structure * pointed to by 'state_ptr'. * All the initial state values are specified in the G.721 standard specs. */ void g721_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 */ } /* * _g721_fmult() * * returns the integer product of the "floating point" an and srn * by the lookup table _fmultwanmant[]. * */ static int _g721_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); } } } /* * _g721_update() * * updates the state variables for each output code * */ static void _g721_update( int y, int i, int dq, int sr, int pk0, struct audio_g72x_state *state_ptr, int sigpk) { int cnt; long fi; /* FUNCTF */ short mag, exp; /* 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. */ 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; } /* * _g721_quantize() * * Description: * * Given a raw sample, 'd', of the difference signal and a * quantization step size scale factor, 'y', this routine returns the * G.721 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 _g721_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 */ char i; /* G.721 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 = _quani[dln & 0xFFF]; if (d < 0) i ^= 0xF; /* Stuff in sign of 'd'. */ else if (i == 0) i = 0xF; /* New in 1988 revision */ return (i); } /* * _g721_reconstr() * * Description: * * Returns reconstructed difference signal 'dq' obtained from * G.721 codeword 'i' and quantization step size scale factor 'y'. * Multiplication is performed in log base 2 domain as addition. */ static unsigned long _g721_reconstr( int i, /* G.721 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 & 8) dq -= 0x4000; 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 u-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 u-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)? 0x7FFF : sr << 2); /* short to A-law compression */ dx = (audio_a2s(sp) >> 2) - se; /* 16-bit prediction error */ id = _g721_quantize(dx, y); if (id == i) /* no adjustment on sp */ return (sp); else { /* sp adjustment needed */ /* ADPCM codes : 8, 9, ... F, 0, 1, ... , 6, 7 */ im = i ^ 8; /* 2's complement to biased unsigned */ imx = id ^ 8; 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 = _g721_quantize(dx, y); if (id == i) return (sp); else { /* ADPCM codes : 8, 9, ... F, 0, 1, ... , 6, 7 */ im = i ^ 8; /* 2's complement to biased unsigned */ imx = id ^ 8; if (imx > im) { /* sp adjusted to next lower value */ if (sp & 0x80) sd = (sp == 0xFF)? 0x7F : 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)? 0xFF : sp + 1; } return (sd); } } /* * g721_encode() * * Description: * * Encodes a buffer of linear PCM, A-law or u-law data pointed to by * 'in_buf' according * the G.721 encoding algorithm and packs the * resulting code words into bytes. The bytes of codeword pairs are * written to a buffer pointed to by 'out_buf'. * * Notes: * * In the event that the total number of codewords which have to be * written is odd, the last unpairable codeword is saved in the * state structure till the next call. It is then paired off and * packed with the first codeword of the new buffer. The number of * valid bytes in 'out_buf' is returned in *out_size. Note that * *out_size will not always be equal to half * of 'data_size' on input. * On the final call to 'g721_encode()' the calling program might want to * check if a codeword was left over. This can be * done by calling 'g721_encode()' with data_size = 0, which returns in * *out_size a 0 if nothing was leftover and 1 if a codeword was leftover * which now is in out_buf[0]. * * The 4 lower significant bits of an individual byte in the output byte * stream is packed with a G.721 codeword first. Then the 4 higher order * bits are packed with the next codeword. */ int g721_encode( void *in_buf, int data_size, Audio_hdr *in_header, unsigned char *out_buf, int *out_size, struct audio_g72x_state *state_ptr) { short sl; /* EXPAND */ 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, cnta; int out_leng; unsigned char *char_in; unsigned char *char_out; short *short_ptr; if (data_size == 0) { /* Actually, the leftover count will never be more than 4 */ for (i = 0; state_ptr->leftover_cnt > 0; i++) { *out_buf++ = state_ptr->leftover[i]; state_ptr->leftover_cnt -= 8; } *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; /* divide to get sample cnt */ short_ptr = (short *)in_buf; } } else { char_in = (unsigned char *)in_buf; } char_out = (unsigned char *)out_buf; if (state_ptr->leftover_cnt > 0) { *char_out = state_ptr->leftover[0]; state_ptr->leftover_cnt = 0; data_size += 1; cnta = 1; } else { cnta = 0; } out_leng = (data_size & ~0x01); /* clear low order bit */ for (; cnta < data_size; cnta++) { /* EXPAND */ switch (in_header->encoding) { case AUDIO_ENCODING_LINEAR: sl = *short_ptr++ >> 2; break; case AUDIO_ENCODING_ALAW: sl = audio_a2s(*char_in++) >> 2; break; case AUDIO_ENCODING_ULAW: sl = audio_u2s(*char_in++) >> 2; /* u-law to short */ break; default: return (AUDIO_ERR_ENCODING); } /* ACCUM */ sezi = _g721_fmult(state_ptr->b[0] >> 2, state_ptr->dq[0]); for (cnt = 1; cnt < 6; cnt++) sezi = sezi + _g721_fmult(state_ptr->b[cnt] >> 2, state_ptr->dq[cnt]); sei = sezi; for (cnt = 1; cnt > -1; cnt--) sei = sei + _g721_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 = _g721_quantize(d, y); dq = _g721_reconstr(i, y); /* ADDB */ sr = (dq < 0) ? se - (dq & 0x3FFF) : se + dq; if (cnta & 1) { *char_out++ += i << 4; } else if (cnta < out_leng) { *char_out = i; } else { /* * save the last codeword which can not be paired into * a byte in the state stucture and set leftover_flag. */ state_ptr->leftover[0] = i; state_ptr->leftover_cnt = 4; } dqsez = sr + sez - se; /* ADDC */ if (dqsez == 0) { pk0 = 0; sigpk = 1; } else { pk0 = (dqsez < 0) ? 1 : 0; sigpk = 0; } _g721_update(y, i, dq, sr, pk0, state_ptr, sigpk); } *out_size = cnta >> 1; return (AUDIO_SUCCESS); } /* * g721_decode() * * Description: * * Decodes a buffer of G.721 encoded data pointed to by 'in_buf' and * writes the resulting linear PCM, A-law or Mu-law bytes into a buffer * pointed to by 'out_buf'. */ int g721_decode( unsigned char *in_buf, /* Buffer of g721 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. */ { short sezi, sei, sez, se; /* ACCUM */ float al; /* use floating point for faster multiply */ short y, dif; /* MIX */ short sr; /* ADDB */ char pk0, i; /* ADDC */ short dq; char sigpk; short dqsez; unsigned char *char_in; unsigned char *char_out; int cnt, cnta; short *linear_out; *out_size = data_size << 1; char_in = (unsigned char *)in_buf; char_out = (unsigned char *)out_buf; linear_out = (short *)out_buf; for (cnta = 0; cnta < *out_size; cnta++) { if (cnta & 1) i = *char_in++ >> 4; else i = *char_in & 0xF; /* ACCUM */ sezi = _g721_fmult(state_ptr->b[0] >> 2, state_ptr->dq[0]); for (cnt = 1; cnt < 6; cnt++) sezi = sezi + _g721_fmult(state_ptr->b[cnt] >> 2, state_ptr->dq[cnt]); sei = sezi; for (cnt = 1; cnt >= 0; cnt--) sei = sei + _g721_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; } dq = _g721_reconstr(i, y); /* ADDB */ if (dq < 0) sr = se - (dq & 0x3FFF); else sr = se + dq; switch (out_header->encoding) { case AUDIO_ENCODING_LINEAR: *linear_out++ = ((sr <= -0x2000) ? -0x8000 : (sr >= 0x1FFF) ? 0x7FFF : sr << 2); break; case AUDIO_ENCODING_ALAW: *char_out++ = _tandem_adjust_alaw(sr, se, y, i); break; case AUDIO_ENCODING_ULAW: *char_out++ = _tandem_adjust_ulaw(sr, se, y, i); break; default: return (AUDIO_ERR_ENCODING); } /* ADDC */ dqsez = sr - se + sez; pk0 = (dqsez < 0) ? 1 : 0; sigpk = (dqsez) ? 0 : 1; _g721_update(y, i, dq, sr, pk0, state_ptr, sigpk); } *out_size = cnta; return (AUDIO_SUCCESS); }