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adpcm.c
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/*
* ADPCM codecs
* Copyright (c) 2001-2003 The ffmpeg Project
*
* This file is part of FFmpeg.
*
* FFmpeg 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.
*
* FFmpeg 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 FFmpeg; if not, write to the Free Software
* Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
*/
#include "avcodec.h"
#include "get_bits.h"
#include "put_bits.h"
#include "bytestream.h"
/**
* @file
* ADPCM codecs.
* First version by Francois Revol (revol@free.fr)
* Fringe ADPCM codecs (e.g., DK3, DK4, Westwood)
* by Mike Melanson (melanson@pcisys.net)
* CD-ROM XA ADPCM codec by BERO
* EA ADPCM decoder by Robin Kay (komadori@myrealbox.com)
* EA ADPCM R1/R2/R3 decoder by Peter Ross (pross@xvid.org)
* EA IMA EACS decoder by Peter Ross (pross@xvid.org)
* EA IMA SEAD decoder by Peter Ross (pross@xvid.org)
* EA ADPCM XAS decoder by Peter Ross (pross@xvid.org)
* MAXIS EA ADPCM decoder by Robert Marston (rmarston@gmail.com)
* THP ADPCM decoder by Marco Gerards (mgerards@xs4all.nl)
*
* Features and limitations:
*
* Reference documents:
* http://www.pcisys.net/~melanson/codecs/simpleaudio.html
* http://www.geocities.com/SiliconValley/8682/aud3.txt
* http://openquicktime.sourceforge.net/plugins.htm
* XAnim sources (xa_codec.c) http://www.rasnaimaging.com/people/lapus/download.html
* http://www.cs.ucla.edu/~leec/mediabench/applications.html
* SoX source code http://home.sprynet.com/~cbagwell/sox.html
*
* CD-ROM XA:
* http://ku-www.ss.titech.ac.jp/~yatsushi/xaadpcm.html
* vagpack & depack http://homepages.compuserve.de/bITmASTER32/psx-index.html
* readstr http://www.geocities.co.jp/Playtown/2004/
*/
#define BLKSIZE 1024
/* step_table[] and index_table[] are from the ADPCM reference source */
/* This is the index table: */
static const int index_table[16] = {
-1, -1, -1, -1, 2, 4, 6, 8,
-1, -1, -1, -1, 2, 4, 6, 8,
};
/**
* This is the step table. Note that many programs use slight deviations from
* this table, but such deviations are negligible:
*/
static const int step_table[89] = {
7, 8, 9, 10, 11, 12, 13, 14, 16, 17,
19, 21, 23, 25, 28, 31, 34, 37, 41, 45,
50, 55, 60, 66, 73, 80, 88, 97, 107, 118,
130, 143, 157, 173, 190, 209, 230, 253, 279, 307,
337, 371, 408, 449, 494, 544, 598, 658, 724, 796,
876, 963, 1060, 1166, 1282, 1411, 1552, 1707, 1878, 2066,
2272, 2499, 2749, 3024, 3327, 3660, 4026, 4428, 4871, 5358,
5894, 6484, 7132, 7845, 8630, 9493, 10442, 11487, 12635, 13899,
15289, 16818, 18500, 20350, 22385, 24623, 27086, 29794, 32767
};
/* These are for MS-ADPCM */
/* AdaptationTable[], AdaptCoeff1[], and AdaptCoeff2[] are from libsndfile */
static const int AdaptationTable[] = {
230, 230, 230, 230, 307, 409, 512, 614,
768, 614, 512, 409, 307, 230, 230, 230
};
/** Divided by 4 to fit in 8-bit integers */
static const uint8_t AdaptCoeff1[] = {
64, 128, 0, 48, 60, 115, 98
};
/** Divided by 4 to fit in 8-bit integers */
static const int8_t AdaptCoeff2[] = {
0, -64, 0, 16, 0, -52, -58
};
/* These are for CD-ROM XA ADPCM */
static const int xa_adpcm_table[5][2] = {
{ 0, 0 },
{ 60, 0 },
{ 115, -52 },
{ 98, -55 },
{ 122, -60 }
};
static const int ea_adpcm_table[] = {
0, 240, 460, 392, 0, 0, -208, -220, 0, 1,
3, 4, 7, 8, 10, 11, 0, -1, -3, -4
};
// padded to zero where table size is less then 16
static const int swf_index_tables[4][16] = {
/*2*/ { -1, 2 },
/*3*/ { -1, -1, 2, 4 },
/*4*/ { -1, -1, -1, -1, 2, 4, 6, 8 },
/*5*/ { -1, -1, -1, -1, -1, -1, -1, -1, 1, 2, 4, 6, 8, 10, 13, 16 }
};
static const int yamaha_indexscale[] = {
230, 230, 230, 230, 307, 409, 512, 614,
230, 230, 230, 230, 307, 409, 512, 614
};
static const int yamaha_difflookup[] = {
1, 3, 5, 7, 9, 11, 13, 15,
-1, -3, -5, -7, -9, -11, -13, -15
};
/* end of tables */
typedef struct ADPCMChannelStatus {
int predictor;
short int step_index;
int step;
/* for encoding */
int prev_sample;
/* MS version */
short sample1;
short sample2;
int coeff1;
int coeff2;
int idelta;
} ADPCMChannelStatus;
typedef struct TrellisPath {
int nibble;
int prev;
} TrellisPath;
typedef struct TrellisNode {
uint32_t ssd;
int path;
int sample1;
int sample2;
int step;
} TrellisNode;
typedef struct ADPCMContext {
ADPCMChannelStatus status[6];
TrellisPath *paths;
TrellisNode *node_buf;
TrellisNode **nodep_buf;
} ADPCMContext;
#define FREEZE_INTERVAL 128
/* XXX: implement encoding */
#if CONFIG_ENCODERS
static av_cold int adpcm_encode_init(AVCodecContext *avctx)
{
ADPCMContext *s = avctx->priv_data;
uint8_t *extradata;
int i;
if (avctx->channels > 2)
return -1; /* only stereo or mono =) */
if(avctx->trellis && (unsigned)avctx->trellis > 16U){
av_log(avctx, AV_LOG_ERROR, "invalid trellis size\n");
return -1;
}
if (avctx->trellis) {
int frontier = 1 << avctx->trellis;
int max_paths = frontier * FREEZE_INTERVAL;
FF_ALLOC_OR_GOTO(avctx, s->paths, max_paths * sizeof(*s->paths), error);
FF_ALLOC_OR_GOTO(avctx, s->node_buf, 2 * frontier * sizeof(*s->node_buf), error);
FF_ALLOC_OR_GOTO(avctx, s->nodep_buf, 2 * frontier * sizeof(*s->nodep_buf), error);
}
switch(avctx->codec->id) {
case CODEC_ID_ADPCM_IMA_WAV:
avctx->frame_size = (BLKSIZE - 4 * avctx->channels) * 8 / (4 * avctx->channels) + 1; /* each 16 bits sample gives one nibble */
/* and we have 4 bytes per channel overhead */
avctx->block_align = BLKSIZE;
/* seems frame_size isn't taken into account... have to buffer the samples :-( */
break;
case CODEC_ID_ADPCM_IMA_QT:
avctx->frame_size = 64;
avctx->block_align = 34 * avctx->channels;
break;
case CODEC_ID_ADPCM_MS:
avctx->frame_size = (BLKSIZE - 7 * avctx->channels) * 2 / avctx->channels + 2; /* each 16 bits sample gives one nibble */
/* and we have 7 bytes per channel overhead */
avctx->block_align = BLKSIZE;
avctx->extradata_size = 32;
extradata = avctx->extradata = av_malloc(avctx->extradata_size);
if (!extradata)
return AVERROR(ENOMEM);
bytestream_put_le16(&extradata, avctx->frame_size);
bytestream_put_le16(&extradata, 7); /* wNumCoef */
for (i = 0; i < 7; i++) {
bytestream_put_le16(&extradata, AdaptCoeff1[i] * 4);
bytestream_put_le16(&extradata, AdaptCoeff2[i] * 4);
}
break;
case CODEC_ID_ADPCM_YAMAHA:
avctx->frame_size = BLKSIZE * avctx->channels;
avctx->block_align = BLKSIZE;
break;
case CODEC_ID_ADPCM_SWF:
if (avctx->sample_rate != 11025 &&
avctx->sample_rate != 22050 &&
avctx->sample_rate != 44100) {
av_log(avctx, AV_LOG_ERROR, "Sample rate must be 11025, 22050 or 44100\n");
goto error;
}
avctx->frame_size = 512 * (avctx->sample_rate / 11025);
break;
default:
goto error;
}
avctx->coded_frame= avcodec_alloc_frame();
avctx->coded_frame->key_frame= 1;
return 0;
error:
av_freep(&s->paths);
av_freep(&s->node_buf);
av_freep(&s->nodep_buf);
return -1;
}
static av_cold int adpcm_encode_close(AVCodecContext *avctx)
{
ADPCMContext *s = avctx->priv_data;
av_freep(&avctx->coded_frame);
av_freep(&s->paths);
av_freep(&s->node_buf);
av_freep(&s->nodep_buf);
return 0;
}
static inline unsigned char adpcm_ima_compress_sample(ADPCMChannelStatus *c, short sample)
{
int delta = sample - c->prev_sample;
int nibble = FFMIN(7, abs(delta)*4/step_table[c->step_index]) + (delta<0)*8;
c->prev_sample += ((step_table[c->step_index] * yamaha_difflookup[nibble]) / 8);
c->prev_sample = av_clip_int16(c->prev_sample);
c->step_index = av_clip(c->step_index + index_table[nibble], 0, 88);
return nibble;
}
static inline unsigned char adpcm_ms_compress_sample(ADPCMChannelStatus *c, short sample)
{
int predictor, nibble, bias;
predictor = (((c->sample1) * (c->coeff1)) + ((c->sample2) * (c->coeff2))) / 64;
nibble= sample - predictor;
if(nibble>=0) bias= c->idelta/2;
else bias=-c->idelta/2;
nibble= (nibble + bias) / c->idelta;
nibble= av_clip(nibble, -8, 7)&0x0F;
predictor += (signed)((nibble & 0x08)?(nibble - 0x10):(nibble)) * c->idelta;
c->sample2 = c->sample1;
c->sample1 = av_clip_int16(predictor);
c->idelta = (AdaptationTable[(int)nibble] * c->idelta) >> 8;
if (c->idelta < 16) c->idelta = 16;
return nibble;
}
static inline unsigned char adpcm_yamaha_compress_sample(ADPCMChannelStatus *c, short sample)
{
int nibble, delta;
if(!c->step) {
c->predictor = 0;
c->step = 127;
}
delta = sample - c->predictor;
nibble = FFMIN(7, abs(delta)*4/c->step) + (delta<0)*8;
c->predictor += ((c->step * yamaha_difflookup[nibble]) / 8);
c->predictor = av_clip_int16(c->predictor);
c->step = (c->step * yamaha_indexscale[nibble]) >> 8;
c->step = av_clip(c->step, 127, 24567);
return nibble;
}
static void adpcm_compress_trellis(AVCodecContext *avctx, const short *samples,
uint8_t *dst, ADPCMChannelStatus *c, int n)
{
//FIXME 6% faster if frontier is a compile-time constant
ADPCMContext *s = avctx->priv_data;
const int frontier = 1 << avctx->trellis;
const int stride = avctx->channels;
const int version = avctx->codec->id;
TrellisPath *paths = s->paths, *p;
TrellisNode *node_buf = s->node_buf;
TrellisNode **nodep_buf = s->nodep_buf;
TrellisNode **nodes = nodep_buf; // nodes[] is always sorted by .ssd
TrellisNode **nodes_next = nodep_buf + frontier;
int pathn = 0, froze = -1, i, j, k;
memset(nodep_buf, 0, 2 * frontier * sizeof(*nodep_buf));
nodes[0] = node_buf + frontier;
nodes[0]->ssd = 0;
nodes[0]->path = 0;
nodes[0]->step = c->step_index;
nodes[0]->sample1 = c->sample1;
nodes[0]->sample2 = c->sample2;
if((version == CODEC_ID_ADPCM_IMA_WAV) || (version == CODEC_ID_ADPCM_IMA_QT) || (version == CODEC_ID_ADPCM_SWF))
nodes[0]->sample1 = c->prev_sample;
if(version == CODEC_ID_ADPCM_MS)
nodes[0]->step = c->idelta;
if(version == CODEC_ID_ADPCM_YAMAHA) {
if(c->step == 0) {
nodes[0]->step = 127;
nodes[0]->sample1 = 0;
} else {
nodes[0]->step = c->step;
nodes[0]->sample1 = c->predictor;
}
}
for(i=0; i<n; i++) {
TrellisNode *t = node_buf + frontier*(i&1);
TrellisNode **u;
int sample = samples[i*stride];
memset(nodes_next, 0, frontier*sizeof(TrellisNode*));
for(j=0; j<frontier && nodes[j]; j++) {
// higher j have higher ssd already, so they're unlikely to use a suboptimal next sample too
const int range = (j < frontier/2) ? 1 : 0;
const int step = nodes[j]->step;
int nidx;
if(version == CODEC_ID_ADPCM_MS) {
const int predictor = ((nodes[j]->sample1 * c->coeff1) + (nodes[j]->sample2 * c->coeff2)) / 64;
const int div = (sample - predictor) / step;
const int nmin = av_clip(div-range, -8, 6);
const int nmax = av_clip(div+range, -7, 7);
for(nidx=nmin; nidx<=nmax; nidx++) {
const int nibble = nidx & 0xf;
int dec_sample = predictor + nidx * step;
#define STORE_NODE(NAME, STEP_INDEX)\
int d;\
uint32_t ssd;\
dec_sample = av_clip_int16(dec_sample);\
d = sample - dec_sample;\
ssd = nodes[j]->ssd + d*d;\
if(nodes_next[frontier-1] && ssd >= nodes_next[frontier-1]->ssd)\
continue;\
/* Collapse any two states with the same previous sample value. \
* One could also distinguish states by step and by 2nd to last
* sample, but the effects of that are negligible. */\
for(k=0; k<frontier && nodes_next[k]; k++) {\
if(dec_sample == nodes_next[k]->sample1) {\
assert(ssd >= nodes_next[k]->ssd);\
goto next_##NAME;\
}\
}\
for(k=0; k<frontier; k++) {\
if(!nodes_next[k] || ssd < nodes_next[k]->ssd) {\
TrellisNode *u = nodes_next[frontier-1];\
if(!u) {\
assert(pathn < FREEZE_INTERVAL<<avctx->trellis);\
u = t++;\
u->path = pathn++;\
}\
u->ssd = ssd;\
u->step = STEP_INDEX;\
u->sample2 = nodes[j]->sample1;\
u->sample1 = dec_sample;\
paths[u->path].nibble = nibble;\
paths[u->path].prev = nodes[j]->path;\
memmove(&nodes_next[k+1], &nodes_next[k], (frontier-k-1)*sizeof(TrellisNode*));\
nodes_next[k] = u;\
break;\
}\
}\
next_##NAME:;
STORE_NODE(ms, FFMAX(16, (AdaptationTable[nibble] * step) >> 8));
}
} else if((version == CODEC_ID_ADPCM_IMA_WAV)|| (version == CODEC_ID_ADPCM_IMA_QT)|| (version == CODEC_ID_ADPCM_SWF)) {
#define LOOP_NODES(NAME, STEP_TABLE, STEP_INDEX)\
const int predictor = nodes[j]->sample1;\
const int div = (sample - predictor) * 4 / STEP_TABLE;\
int nmin = av_clip(div-range, -7, 6);\
int nmax = av_clip(div+range, -6, 7);\
if(nmin<=0) nmin--; /* distinguish -0 from +0 */\
if(nmax<0) nmax--;\
for(nidx=nmin; nidx<=nmax; nidx++) {\
const int nibble = nidx<0 ? 7-nidx : nidx;\
int dec_sample = predictor + (STEP_TABLE * yamaha_difflookup[nibble]) / 8;\
STORE_NODE(NAME, STEP_INDEX);\
}
LOOP_NODES(ima, step_table[step], av_clip(step + index_table[nibble], 0, 88));
} else { //CODEC_ID_ADPCM_YAMAHA
LOOP_NODES(yamaha, step, av_clip((step * yamaha_indexscale[nibble]) >> 8, 127, 24567));
#undef LOOP_NODES
#undef STORE_NODE
}
}
u = nodes;
nodes = nodes_next;
nodes_next = u;
// prevent overflow
if(nodes[0]->ssd > (1<<28)) {
for(j=1; j<frontier && nodes[j]; j++)
nodes[j]->ssd -= nodes[0]->ssd;
nodes[0]->ssd = 0;
}
// merge old paths to save memory
if(i == froze + FREEZE_INTERVAL) {
p = &paths[nodes[0]->path];
for(k=i; k>froze; k--) {
dst[k] = p->nibble;
p = &paths[p->prev];
}
froze = i;
pathn = 0;
// other nodes might use paths that don't coincide with the frozen one.
// checking which nodes do so is too slow, so just kill them all.
// this also slightly improves quality, but I don't know why.
memset(nodes+1, 0, (frontier-1)*sizeof(TrellisNode*));
}
}
p = &paths[nodes[0]->path];
for(i=n-1; i>froze; i--) {
dst[i] = p->nibble;
p = &paths[p->prev];
}
c->predictor = nodes[0]->sample1;
c->sample1 = nodes[0]->sample1;
c->sample2 = nodes[0]->sample2;
c->step_index = nodes[0]->step;
c->step = nodes[0]->step;
c->idelta = nodes[0]->step;
}
static int adpcm_encode_frame(AVCodecContext *avctx,
unsigned char *frame, int buf_size, void *data)
{
int n, i, st;
short *samples;
unsigned char *dst;
ADPCMContext *c = avctx->priv_data;
uint8_t *buf;
dst = frame;
samples = (short *)data;
st= avctx->channels == 2;
/* n = (BLKSIZE - 4 * avctx->channels) / (2 * 8 * avctx->channels); */
switch(avctx->codec->id) {
case CODEC_ID_ADPCM_IMA_WAV:
n = avctx->frame_size / 8;
c->status[0].prev_sample = (signed short)samples[0]; /* XXX */
/* c->status[0].step_index = 0; *//* XXX: not sure how to init the state machine */
bytestream_put_le16(&dst, c->status[0].prev_sample);
*dst++ = (unsigned char)c->status[0].step_index;
*dst++ = 0; /* unknown */
samples++;
if (avctx->channels == 2) {
c->status[1].prev_sample = (signed short)samples[0];
/* c->status[1].step_index = 0; */
bytestream_put_le16(&dst, c->status[1].prev_sample);
*dst++ = (unsigned char)c->status[1].step_index;
*dst++ = 0;
samples++;
}
/* stereo: 4 bytes (8 samples) for left, 4 bytes for right, 4 bytes left, ... */
if(avctx->trellis > 0) {
FF_ALLOC_OR_GOTO(avctx, buf, 2*n*8, error);
adpcm_compress_trellis(avctx, samples, buf, &c->status[0], n*8);
if(avctx->channels == 2)
adpcm_compress_trellis(avctx, samples+1, buf + n*8, &c->status[1], n*8);
for(i=0; i<n; i++) {
*dst++ = buf[8*i+0] | (buf[8*i+1] << 4);
*dst++ = buf[8*i+2] | (buf[8*i+3] << 4);
*dst++ = buf[8*i+4] | (buf[8*i+5] << 4);
*dst++ = buf[8*i+6] | (buf[8*i+7] << 4);
if (avctx->channels == 2) {
uint8_t *buf1 = buf + n*8;
*dst++ = buf1[8*i+0] | (buf1[8*i+1] << 4);
*dst++ = buf1[8*i+2] | (buf1[8*i+3] << 4);
*dst++ = buf1[8*i+4] | (buf1[8*i+5] << 4);
*dst++ = buf1[8*i+6] | (buf1[8*i+7] << 4);
}
}
av_free(buf);
} else
for (; n>0; n--) {
*dst = adpcm_ima_compress_sample(&c->status[0], samples[0]);
*dst |= adpcm_ima_compress_sample(&c->status[0], samples[avctx->channels]) << 4;
dst++;
*dst = adpcm_ima_compress_sample(&c->status[0], samples[avctx->channels * 2]);
*dst |= adpcm_ima_compress_sample(&c->status[0], samples[avctx->channels * 3]) << 4;
dst++;
*dst = adpcm_ima_compress_sample(&c->status[0], samples[avctx->channels * 4]);
*dst |= adpcm_ima_compress_sample(&c->status[0], samples[avctx->channels * 5]) << 4;
dst++;
*dst = adpcm_ima_compress_sample(&c->status[0], samples[avctx->channels * 6]);
*dst |= adpcm_ima_compress_sample(&c->status[0], samples[avctx->channels * 7]) << 4;
dst++;
/* right channel */
if (avctx->channels == 2) {
*dst = adpcm_ima_compress_sample(&c->status[1], samples[1]);
*dst |= adpcm_ima_compress_sample(&c->status[1], samples[3]) << 4;
dst++;
*dst = adpcm_ima_compress_sample(&c->status[1], samples[5]);
*dst |= adpcm_ima_compress_sample(&c->status[1], samples[7]) << 4;
dst++;
*dst = adpcm_ima_compress_sample(&c->status[1], samples[9]);
*dst |= adpcm_ima_compress_sample(&c->status[1], samples[11]) << 4;
dst++;
*dst = adpcm_ima_compress_sample(&c->status[1], samples[13]);
*dst |= adpcm_ima_compress_sample(&c->status[1], samples[15]) << 4;
dst++;
}
samples += 8 * avctx->channels;
}
break;
case CODEC_ID_ADPCM_IMA_QT:
{
int ch, i;
PutBitContext pb;
init_put_bits(&pb, dst, buf_size*8);
for(ch=0; ch<avctx->channels; ch++){
put_bits(&pb, 9, (c->status[ch].prev_sample + 0x10000) >> 7);
put_bits(&pb, 7, c->status[ch].step_index);
if(avctx->trellis > 0) {
uint8_t buf[64];
adpcm_compress_trellis(avctx, samples+ch, buf, &c->status[ch], 64);
for(i=0; i<64; i++)
put_bits(&pb, 4, buf[i^1]);
c->status[ch].prev_sample = c->status[ch].predictor & ~0x7F;
} else {
for (i=0; i<64; i+=2){
int t1, t2;
t1 = adpcm_ima_compress_sample(&c->status[ch], samples[avctx->channels*(i+0)+ch]);
t2 = adpcm_ima_compress_sample(&c->status[ch], samples[avctx->channels*(i+1)+ch]);
put_bits(&pb, 4, t2);
put_bits(&pb, 4, t1);
}
c->status[ch].prev_sample &= ~0x7F;
}
}
flush_put_bits(&pb);
dst += put_bits_count(&pb)>>3;
break;
}
case CODEC_ID_ADPCM_SWF:
{
int i;
PutBitContext pb;
init_put_bits(&pb, dst, buf_size*8);
n = avctx->frame_size-1;
//Store AdpcmCodeSize
put_bits(&pb, 2, 2); //Set 4bits flash adpcm format
//Init the encoder state
for(i=0; i<avctx->channels; i++){
c->status[i].step_index = av_clip(c->status[i].step_index, 0, 63); // clip step so it fits 6 bits
put_sbits(&pb, 16, samples[i]);
put_bits(&pb, 6, c->status[i].step_index);
c->status[i].prev_sample = (signed short)samples[i];
}
if(avctx->trellis > 0) {
FF_ALLOC_OR_GOTO(avctx, buf, 2*n, error);
adpcm_compress_trellis(avctx, samples+2, buf, &c->status[0], n);
if (avctx->channels == 2)
adpcm_compress_trellis(avctx, samples+3, buf+n, &c->status[1], n);
for(i=0; i<n; i++) {
put_bits(&pb, 4, buf[i]);
if (avctx->channels == 2)
put_bits(&pb, 4, buf[n+i]);
}
av_free(buf);
} else {
for (i=1; i<avctx->frame_size; i++) {
put_bits(&pb, 4, adpcm_ima_compress_sample(&c->status[0], samples[avctx->channels*i]));
if (avctx->channels == 2)
put_bits(&pb, 4, adpcm_ima_compress_sample(&c->status[1], samples[2*i+1]));
}
}
flush_put_bits(&pb);
dst += put_bits_count(&pb)>>3;
break;
}
case CODEC_ID_ADPCM_MS:
for(i=0; i<avctx->channels; i++){
int predictor=0;
*dst++ = predictor;
c->status[i].coeff1 = AdaptCoeff1[predictor];
c->status[i].coeff2 = AdaptCoeff2[predictor];
}
for(i=0; i<avctx->channels; i++){
if (c->status[i].idelta < 16)
c->status[i].idelta = 16;
bytestream_put_le16(&dst, c->status[i].idelta);
}
for(i=0; i<avctx->channels; i++){
c->status[i].sample2= *samples++;
}
for(i=0; i<avctx->channels; i++){
c->status[i].sample1= *samples++;
bytestream_put_le16(&dst, c->status[i].sample1);
}
for(i=0; i<avctx->channels; i++)
bytestream_put_le16(&dst, c->status[i].sample2);
if(avctx->trellis > 0) {
int n = avctx->block_align - 7*avctx->channels;
FF_ALLOC_OR_GOTO(avctx, buf, 2*n, error);
if(avctx->channels == 1) {
adpcm_compress_trellis(avctx, samples, buf, &c->status[0], n);
for(i=0; i<n; i+=2)
*dst++ = (buf[i] << 4) | buf[i+1];
} else {
adpcm_compress_trellis(avctx, samples, buf, &c->status[0], n);
adpcm_compress_trellis(avctx, samples+1, buf+n, &c->status[1], n);
for(i=0; i<n; i++)
*dst++ = (buf[i] << 4) | buf[n+i];
}
av_free(buf);
} else
for(i=7*avctx->channels; i<avctx->block_align; i++) {
int nibble;
nibble = adpcm_ms_compress_sample(&c->status[ 0], *samples++)<<4;
nibble|= adpcm_ms_compress_sample(&c->status[st], *samples++);
*dst++ = nibble;
}
break;
case CODEC_ID_ADPCM_YAMAHA:
n = avctx->frame_size / 2;
if(avctx->trellis > 0) {
FF_ALLOC_OR_GOTO(avctx, buf, 2*n*2, error);
n *= 2;
if(avctx->channels == 1) {
adpcm_compress_trellis(avctx, samples, buf, &c->status[0], n);
for(i=0; i<n; i+=2)
*dst++ = buf[i] | (buf[i+1] << 4);
} else {
adpcm_compress_trellis(avctx, samples, buf, &c->status[0], n);
adpcm_compress_trellis(avctx, samples+1, buf+n, &c->status[1], n);
for(i=0; i<n; i++)
*dst++ = buf[i] | (buf[n+i] << 4);
}
av_free(buf);
} else
for (n *= avctx->channels; n>0; n--) {
int nibble;
nibble = adpcm_yamaha_compress_sample(&c->status[ 0], *samples++);
nibble |= adpcm_yamaha_compress_sample(&c->status[st], *samples++) << 4;
*dst++ = nibble;
}
break;
default:
error:
return -1;
}
return dst - frame;
}
#endif //CONFIG_ENCODERS
static av_cold int adpcm_decode_init(AVCodecContext * avctx)
{
ADPCMContext *c = avctx->priv_data;
unsigned int max_channels = 2;
switch(avctx->codec->id) {
case CODEC_ID_ADPCM_EA_R1:
case CODEC_ID_ADPCM_EA_R2:
case CODEC_ID_ADPCM_EA_R3:
max_channels = 6;
break;
}
if(avctx->channels > max_channels){
return -1;
}
switch(avctx->codec->id) {
case CODEC_ID_ADPCM_CT:
c->status[0].step = c->status[1].step = 511;
break;
case CODEC_ID_ADPCM_IMA_WAV:
if (avctx->bits_per_coded_sample != 4) {
av_log(avctx, AV_LOG_ERROR, "Only 4-bit ADPCM IMA WAV files are supported\n");
return -1;
}
break;
case CODEC_ID_ADPCM_IMA_WS:
if (avctx->extradata && avctx->extradata_size == 2 * 4) {
c->status[0].predictor = AV_RL32(avctx->extradata);
c->status[1].predictor = AV_RL32(avctx->extradata + 4);
}
break;
default:
break;
}
avctx->sample_fmt = SAMPLE_FMT_S16;
return 0;
}
static inline short adpcm_ima_expand_nibble(ADPCMChannelStatus *c, char nibble, int shift)
{
int step_index;
int predictor;
int sign, delta, diff, step;
step = step_table[c->step_index];
step_index = c->step_index + index_table[(unsigned)nibble];
if (step_index < 0) step_index = 0;
else if (step_index > 88) step_index = 88;
sign = nibble & 8;
delta = nibble & 7;
/* perform direct multiplication instead of series of jumps proposed by
* the reference ADPCM implementation since modern CPUs can do the mults
* quickly enough */
diff = ((2 * delta + 1) * step) >> shift;
predictor = c->predictor;
if (sign) predictor -= diff;
else predictor += diff;
c->predictor = av_clip_int16(predictor);
c->step_index = step_index;
return (short)c->predictor;
}
static inline short adpcm_ms_expand_nibble(ADPCMChannelStatus *c, char nibble)
{
int predictor;
predictor = (((c->sample1) * (c->coeff1)) + ((c->sample2) * (c->coeff2))) / 64;
predictor += (signed)((nibble & 0x08)?(nibble - 0x10):(nibble)) * c->idelta;
c->sample2 = c->sample1;
c->sample1 = av_clip_int16(predictor);
c->idelta = (AdaptationTable[(int)nibble] * c->idelta) >> 8;
if (c->idelta < 16) c->idelta = 16;
return c->sample1;
}
static inline short adpcm_ct_expand_nibble(ADPCMChannelStatus *c, char nibble)
{
int sign, delta, diff;
int new_step;
sign = nibble & 8;
delta = nibble & 7;
/* perform direct multiplication instead of series of jumps proposed by
* the reference ADPCM implementation since modern CPUs can do the mults
* quickly enough */
diff = ((2 * delta + 1) * c->step) >> 3;
/* predictor update is not so trivial: predictor is multiplied on 254/256 before updating */
c->predictor = ((c->predictor * 254) >> 8) + (sign ? -diff : diff);
c->predictor = av_clip_int16(c->predictor);
/* calculate new step and clamp it to range 511..32767 */
new_step = (AdaptationTable[nibble & 7] * c->step) >> 8;
c->step = av_clip(new_step, 511, 32767);
return (short)c->predictor;
}
static inline short adpcm_sbpro_expand_nibble(ADPCMChannelStatus *c, char nibble, int size, int shift)
{
int sign, delta, diff;
sign = nibble & (1<<(size-1));
delta = nibble & ((1<<(size-1))-1);
diff = delta << (7 + c->step + shift);
/* clamp result */
c->predictor = av_clip(c->predictor + (sign ? -diff : diff), -16384,16256);
/* calculate new step */
if (delta >= (2*size - 3) && c->step < 3)
c->step++;
else if (delta == 0 && c->step > 0)
c->step--;
return (short) c->predictor;
}
static inline short adpcm_yamaha_expand_nibble(ADPCMChannelStatus *c, unsigned char nibble)
{
if(!c->step) {
c->predictor = 0;
c->step = 127;
}
c->predictor += (c->step * yamaha_difflookup[nibble]) / 8;
c->predictor = av_clip_int16(c->predictor);
c->step = (c->step * yamaha_indexscale[nibble]) >> 8;
c->step = av_clip(c->step, 127, 24567);
return c->predictor;
}
static void xa_decode(short *out, const unsigned char *in,
ADPCMChannelStatus *left, ADPCMChannelStatus *right, int inc)
{
int i, j;
int shift,filter,f0,f1;
int s_1,s_2;
int d,s,t;
for(i=0;i<4;i++) {
shift = 12 - (in[4+i*2] & 15);
filter = in[4+i*2] >> 4;
f0 = xa_adpcm_table[filter][0];
f1 = xa_adpcm_table[filter][1];
s_1 = left->sample1;
s_2 = left->sample2;
for(j=0;j<28;j++) {
d = in[16+i+j*4];
t = (signed char)(d<<4)>>4;
s = ( t<<shift ) + ((s_1*f0 + s_2*f1+32)>>6);
s_2 = s_1;
s_1 = av_clip_int16(s);
*out = s_1;
out += inc;
}
if (inc==2) { /* stereo */
left->sample1 = s_1;
left->sample2 = s_2;
s_1 = right->sample1;
s_2 = right->sample2;
out = out + 1 - 28*2;
}
shift = 12 - (in[5+i*2] & 15);
filter = in[5+i*2] >> 4;
f0 = xa_adpcm_table[filter][0];
f1 = xa_adpcm_table[filter][1];
for(j=0;j<28;j++) {
d = in[16+i+j*4];
t = (signed char)d >> 4;
s = ( t<<shift ) + ((s_1*f0 + s_2*f1+32)>>6);
s_2 = s_1;
s_1 = av_clip_int16(s);
*out = s_1;
out += inc;
}
if (inc==2) { /* stereo */
right->sample1 = s_1;
right->sample2 = s_2;
out -= 1;
} else {
left->sample1 = s_1;
left->sample2 = s_2;
}
}
}
/* DK3 ADPCM support macro */
#define DK3_GET_NEXT_NIBBLE() \
if (decode_top_nibble_next) \
{ \
nibble = last_byte >> 4; \
decode_top_nibble_next = 0; \
} \
else \
{ \
last_byte = *src++; \
if (src >= buf + buf_size) break; \
nibble = last_byte & 0x0F; \
decode_top_nibble_next = 1; \
}
static int adpcm_decode_frame(AVCodecContext *avctx,
void *data, int *data_size,
AVPacket *avpkt)
{
const uint8_t *buf = avpkt->data;
int buf_size = avpkt->size;
ADPCMContext *c = avctx->priv_data;
ADPCMChannelStatus *cs;
int n, m, channel, i;
int block_predictor[2];
short *samples;
short *samples_end;
const uint8_t *src;
int st; /* stereo */
/* DK3 ADPCM accounting variables */
unsigned char last_byte = 0;
unsigned char nibble;
int decode_top_nibble_next = 0;
int diff_channel;
/* EA ADPCM state variables */
uint32_t samples_in_chunk;
int32_t previous_left_sample, previous_right_sample;
int32_t current_left_sample, current_right_sample;
int32_t next_left_sample, next_right_sample;
int32_t coeff1l, coeff2l, coeff1r, coeff2r;
uint8_t shift_left, shift_right;
int count1, count2;
int coeff[2][2], shift[2];//used in EA MAXIS ADPCM
if (!buf_size)
return 0;
//should protect all 4bit ADPCM variants
//8 is needed for CODEC_ID_ADPCM_IMA_WAV with 2 channels
//
if(*data_size/4 < buf_size + 8)
return -1;
samples = data;
samples_end= samples + *data_size/2;
*data_size= 0;
src = buf;
st = avctx->channels == 2 ? 1 : 0;
switch(avctx->codec->id) {
case CODEC_ID_ADPCM_IMA_QT:
n = buf_size - 2*avctx->channels;
for (channel = 0; channel < avctx->channels; channel++) {
cs = &(c->status[channel]);
/* (pppppp) (piiiiiii) */
/* Bits 15-7 are the _top_ 9 bits of the 16-bit initial predictor value */
cs->predictor = (*src++) << 8;
cs->predictor |= (*src & 0x80);
cs->predictor &= 0xFF80;
/* sign extension */
if(cs->predictor & 0x8000)
cs->predictor -= 0x10000;
cs->predictor = av_clip_int16(cs->predictor);
cs->step_index = (*src++) & 0x7F;
if (cs->step_index > 88){
av_log(avctx, AV_LOG_ERROR, "ERROR: step_index = %i\n", cs->step_index);
cs->step_index = 88;
}
cs->step = step_table[cs->step_index];
samples = (short*)data + channel;
for(m=32; n>0 && m>0; n--, m--) { /* in QuickTime, IMA is encoded by chuncks of 34 bytes (=64 samples) */
*samples = adpcm_ima_expand_nibble(cs, src[0] & 0x0F, 3);
samples += avctx->channels;