ref: 1d710d29c8cc45e8ad2f5c19c137bedca092707c
dir: /src/adpcm.c/
/* adpcm.c codex functions for MS_ADPCM data
* (hopefully) provides interoperability with
* Microsoft's ADPCM format, but, as usual,
* see LACK-OF-WARRANTY information below.
*
* Copyright (C) 1999 Stanley J. Brooks <stabro@megsinet.net>
*
* This library is free software; you can redistribute it and/or modify it
* under the terms of the GNU Lesser General Public License as published by
* the Free Software Foundation; either version 2.1 of the License, or (at
* your option) any later version.
*
* This library is distributed in the hope that it will be useful, but
* WITHOUT ANY WARRANTY; without even the implied warranty of
* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU Lesser
* General Public License for more details.
*
* You should have received a copy of the GNU Lesser General Public License
* along with this library; if not, write to the Free Software Foundation,
* Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
*
*/
/*
* November 22, 1999
* specs I've seen are unclear about ADPCM supporting more than 2 channels,
* but these routines support more channels in a manner which looks (IMHO)
* like the most natural extension.
*
* Remark: code still turbulent, encoding very new.
*
*/
#include "sox_i.h"
#include "adpcm.h"
#include <sys/types.h>
#include <stdio.h>
typedef struct {
sox_sample_t step; /* step size */
short lsx_ms_adpcm_i_coef[2];
} MsState_t;
#define lsbshortldi(x,p) { (x)=((short)((int)(p)[0] + ((int)(p)[1]<<8))); (p) += 2; }
/*
* Lookup tables for MS ADPCM format
*/
/* these are step-size adjust factors, where
* 1.0 is scaled to 0x100
*/
static const
sox_sample_t stepAdjustTable[] = {
230, 230, 230, 230, 307, 409, 512, 614,
768, 614, 512, 409, 307, 230, 230, 230
};
/* TODO : The first 7 lsx_ms_adpcm_i_coef sets are always hardcoded and must
appear in the actual WAVE file. They should be read in
in case a sound program added extras to the list. */
const short lsx_ms_adpcm_i_coef[7][2] = {
{ 256, 0},
{ 512,-256},
{ 0, 0},
{ 192, 64},
{ 240, 0},
{ 460,-208},
{ 392,-232}
};
static inline sox_sample_t AdpcmDecode(sox_sample_t c, MsState_t *state,
sox_sample_t sample1, sox_sample_t sample2)
{
sox_sample_t vlin;
sox_sample_t sample;
sox_sample_t step;
/** Compute next step value **/
step = state->step;
{
sox_sample_t nstep;
nstep = (stepAdjustTable[c] * step) >> 8;
state->step = (nstep < 16)? 16:nstep;
}
/** make linear prediction for next sample **/
vlin =
((sample1 * state->lsx_ms_adpcm_i_coef[0]) +
(sample2 * state->lsx_ms_adpcm_i_coef[1])) >> 8;
/** then add the code*step adjustment **/
c -= (c & 0x08) << 1;
sample = (c * step) + vlin;
if (sample > 0x7fff) sample = 0x7fff;
else if (sample < -0x8000) sample = -0x8000;
return (sample);
}
/* lsx_ms_adpcm_block_expand_i() outputs interleaved samples into one output buffer */
const char *lsx_ms_adpcm_block_expand_i(
unsigned chans, /* total channels */
int nCoef,
const short *lsx_ms_adpcm_i_coef,
const unsigned char *ibuff,/* input buffer[blockAlign] */
SAMPL *obuff, /* output samples, n*chans */
int n /* samples to decode PER channel */
)
{
const unsigned char *ip;
unsigned ch;
const char *errmsg = NULL;
MsState_t state[4]; /* One decompressor state for each channel */
/* Read the four-byte header for each channel */
ip = ibuff;
for (ch = 0; ch < chans; ch++) {
unsigned char bpred = *ip++;
if (bpred >= nCoef) {
errmsg = "MSADPCM bpred >= nCoef, arbitrarily using 0\n";
bpred = 0;
}
state[ch].lsx_ms_adpcm_i_coef[0] = lsx_ms_adpcm_i_coef[(int)bpred*2+0];
state[ch].lsx_ms_adpcm_i_coef[1] = lsx_ms_adpcm_i_coef[(int)bpred*2+1];
}
for (ch = 0; ch < chans; ch++)
lsbshortldi(state[ch].step, ip);
/* sample1's directly into obuff */
for (ch = 0; ch < chans; ch++)
lsbshortldi(obuff[chans+ch], ip);
/* sample2's directly into obuff */
for (ch = 0; ch < chans; ch++)
lsbshortldi(obuff[ch], ip);
{
unsigned ch;
unsigned char b;
short *op, *top, *tmp;
/* already have 1st 2 samples from block-header */
op = obuff + 2*chans;
top = obuff + n*chans;
ch = 0;
while (op < top) {
b = *ip++;
tmp = op;
*op++ = AdpcmDecode(b >> 4, state+ch, tmp[-chans], tmp[-(2*chans)]);
if (++ch == chans) ch = 0;
/* ch = ++ch % chans; */
tmp = op;
*op++ = AdpcmDecode(b&0x0f, state+ch, tmp[-chans], tmp[-(2*chans)]);
if (++ch == chans) ch = 0;
/* ch = ++ch % chans; */
}
}
return errmsg;
}
static int AdpcmMashS(
unsigned ch, /* channel number to encode, REQUIRE 0 <= ch < chans */
unsigned chans, /* total channels */
SAMPL v[2], /* values to use as starting 2 */
const short lsx_ms_adpcm_i_coef[2],/* lin predictor coeffs */
const SAMPL *ibuff, /* ibuff[] is interleaved input samples */
int n, /* samples to encode PER channel */
int *iostep, /* input/output step, REQUIRE 16 <= *st <= 0x7fff */
unsigned char *obuff /* output buffer[blockAlign], or NULL for no output */
)
{
const SAMPL *ip, *itop;
unsigned char *op;
int ox = 0; /* */
int i, d, v0, v1, step;
double d2; /* long long is okay also, speed abt the same */
ip = ibuff + ch; /* point ip to 1st input sample for this channel */
itop = ibuff + n*chans;
v0 = v[0];
v1 = v[1];
d = *ip - v1; ip += chans; /* 1st input sample for this channel */
d2 = d*d; /* d2 will be sum of squares of errors, given input v0 and *st */
d = *ip - v0; ip += chans; /* 2nd input sample for this channel */
d2 += d*d;
step = *iostep;
op = obuff; /* output pointer (or NULL) */
if (op) { /* NULL means don't output, just compute the rms error */
op += chans; /* skip bpred indices */
op += 2*ch; /* channel's stepsize */
op[0] = step; op[1] = step>>8;
op += 2*chans; /* skip to v0 */
op[0] = v0; op[1] = v0>>8;
op += 2*chans; /* skip to v1 */
op[0] = v1; op[1] = v1>>8;
op = obuff+7*chans; /* point to base of output nibbles */
ox = 4*ch;
}
for (i = 0; ip < itop; ip+=chans) {
int vlin,d,dp,c;
/* make linear prediction for next sample */
vlin = (v0 * lsx_ms_adpcm_i_coef[0] + v1 * lsx_ms_adpcm_i_coef[1]) >> 8;
d = *ip - vlin; /* difference between linear prediction and current sample */
dp = d + (step<<3) + (step>>1);
c = 0;
if (dp>0) {
c = dp/step;
if (c>15) c = 15;
}
c -= 8;
dp = c * step; /* quantized estimate of samp - vlin */
c &= 0x0f; /* mask to 4 bits */
v1 = v0; /* shift history */
v0 = vlin + dp;
if (v0<-0x8000) v0 = -0x8000;
else if (v0>0x7fff) v0 = 0x7fff;
d = *ip - v0;
d2 += d*d; /* update square-error */
if (op) { /* if we want output, put it in proper place */
op[ox>>3] |= (ox&4)? c:(c<<4);
ox += 4*chans;
sox_debug_more("%.1x",c);
}
/* Update the step for the next sample */
step = (stepAdjustTable[c] * step) >> 8;
if (step < 16) step = 16;
}
if (op) sox_debug_more("\n");
d2 /= n; /* be sure it's non-negative */
sox_debug_more("ch%d: st %d->%d, d %.1f\n", ch, *iostep, step, sqrt(d2));
*iostep = step;
return (int) sqrt(d2);
}
static inline void AdpcmMashChannel(
unsigned ch, /* channel number to encode, REQUIRE 0 <= ch < chans */
unsigned chans, /* total channels */
const SAMPL *ip, /* ip[] is interleaved input samples */
int n, /* samples to encode PER channel, REQUIRE */
int *st, /* input/output steps, 16<=st[i] */
unsigned char *obuff /* output buffer[blockAlign] */
)
{
SAMPL v[2];
int n0,s0,s1,ss,smin;
int d,dmin,k,kmin;
n0 = n/2; if (n0>32) n0=32;
if (*st<16) *st = 16;
v[1] = ip[ch];
v[0] = ip[ch+chans];
dmin = 0; kmin = 0; smin = 0;
/* for each of 7 standard coeff sets, we try compression
* beginning with last step-value, and with slightly
* forward-adjusted step-value, taking best of the 14
*/
for (k=0; k<7; k++) {
int d0,d1;
ss = s0 = *st;
d0=AdpcmMashS(ch, chans, v, lsx_ms_adpcm_i_coef[k], ip, n, &ss, NULL); /* with step s0 */
s1 = s0;
AdpcmMashS(ch, chans, v, lsx_ms_adpcm_i_coef[k], ip, n0, &s1, NULL);
sox_debug_more(" s32 %d\n",s1);
ss = s1 = (3*s0+s1)/4;
d1=AdpcmMashS(ch, chans, v, lsx_ms_adpcm_i_coef[k], ip, n, &ss, NULL); /* with step s1 */
if (!k || d0<dmin || d1<dmin) {
kmin = k;
if (d0<=d1) {
dmin = d0;
smin = s0;
}else{
dmin = d1;
smin = s1;
}
}
}
*st = smin;
sox_debug_more("kmin %d, smin %5d, ",kmin,smin);
d=AdpcmMashS(ch, chans, v, lsx_ms_adpcm_i_coef[kmin], ip, n, st, obuff);
obuff[ch] = kmin;
}
void lsx_ms_adpcm_block_mash_i(
unsigned chans, /* total channels */
const SAMPL *ip, /* ip[n*chans] is interleaved input samples */
int n, /* samples to encode PER channel */
int *st, /* input/output steps, 16<=st[i] */
unsigned char *obuff, /* output buffer[blockAlign] */
int blockAlign /* >= 7*chans + chans*(n-2)/2.0 */
)
{
unsigned ch;
unsigned char *p;
sox_debug_more("AdpcmMashI(chans %d, ip %p, n %d, st %p, obuff %p, bA %d)\n",
chans, (void *)ip, n, (void *)st, obuff, blockAlign);
for (p=obuff+7*chans; p<obuff+blockAlign; p++) *p=0;
for (ch=0; ch<chans; ch++)
AdpcmMashChannel(ch, chans, ip, n, st+ch, obuff);
}
/*
* lsx_ms_adpcm_samples_in(dataLen, chans, blockAlign, samplesPerBlock)
* returns the number of samples/channel which would be
* in the dataLen, given the other parameters ...
* if input samplesPerBlock is 0, then returns the max
* samplesPerBlock which would go into a block of size blockAlign
* Yes, it is confusing usage.
*/
size_t lsx_ms_adpcm_samples_in(
size_t dataLen,
size_t chans,
size_t blockAlign,
size_t samplesPerBlock
){
size_t m, n;
if (samplesPerBlock) {
n = (dataLen / blockAlign) * samplesPerBlock;
m = (dataLen % blockAlign);
} else {
n = 0;
m = blockAlign;
}
if (m >= (size_t)(7*chans)) {
m -= 7*chans; /* bytes beyond block-header */
m = (2*m)/chans + 2; /* nibbles/chans + 2 in header */
if (samplesPerBlock && m > samplesPerBlock) m = samplesPerBlock;
n += m;
}
return n;
}
size_t lsx_ms_adpcm_bytes_per_block(
size_t chans,
size_t samplesPerBlock
)
{
size_t n;
n = 7*chans; /* header */
if (samplesPerBlock > 2)
n += (((size_t)samplesPerBlock-2)*chans + 1)/2;
return n;
}