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FFmpeg/libavcodec/adpcm.c
Martin Storsjö cfff297d98 adpcm: Skip samples whose ssd calculation has wrapped around 
Wraparound in ssd is mainly avoided by subtracting the ssd of the
best node from all the others once it has grown large enough.

If using very large trellis sizes (e.g. -trellis 15), the frontier
is so large that the difference between the best and the worst is
large enough to cause wraparound, even if the ssd of the best one
is subtracted regularly.

When using -trellis 10 on a 30 second sample, this causes only a slight
slowdown, from 61 to 64 seconds.

Originally committed as revision 25858 to svn://svn.ffmpeg.org/ffmpeg/trunk
2010-12-01 08:57:45 +00:00

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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;
uint8_t *trellis_hash;
} 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);
FF_ALLOC_OR_GOTO(avctx, s->trellis_hash, 65536 * sizeof(*s->trellis_hash), 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);
av_freep(&s->trellis_hash);
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);
av_freep(&s->trellis_hash);
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, generation = 0;
uint8_t *hash = s->trellis_hash;
memset(hash, 0xff, 65536 * sizeof(*hash));
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];
int heap_pos = 0;
memset(nodes_next, 0, frontier*sizeof(TrellisNode*));
for(j=0; j<frontier && nodes[j]; j++) {
// higher j have higher ssd already, so they're likely to yield 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;\
int pos;\
TrellisNode *u;\
uint8_t *h;\
dec_sample = av_clip_int16(dec_sample);\
d = sample - dec_sample;\
ssd = nodes[j]->ssd + d*d;\
/* Check for wraparound, skip such samples completely. \
* Note, changing ssd to a 64 bit variable would be \
* simpler, avoiding this check, but it's slower on \
* x86 32 bit at the moment. */\
if (ssd < nodes[j]->ssd)\
goto next_##NAME;\
/* 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.
* Since nodes in the previous generation are iterated
* through a heap, they're roughly ordered from better to
* worse, but not strictly ordered. Therefore, an earlier
* node with the same sample value is better in most cases
* (and thus the current is skipped), but not strictly
* in all cases. Only skipping samples where ssd >=
* ssd of the earlier node with the same sample gives
* slightly worse quality, though, for some reason. */ \
h = &hash[(uint16_t) dec_sample];\
if (*h == generation)\
goto next_##NAME;\
if (heap_pos < frontier) {\
pos = heap_pos++;\
} else {\
/* Try to replace one of the leaf nodes with the new \
* one, but try a different slot each time. */\
pos = (frontier >> 1) + (heap_pos & ((frontier >> 1) - 1));\
if (ssd > nodes_next[pos]->ssd)\
goto next_##NAME;\
heap_pos++;\
}\
*h = generation;\
u = nodes_next[pos];\
if(!u) {\
assert(pathn < FREEZE_INTERVAL<<avctx->trellis);\
u = t++;\
nodes_next[pos] = u;\
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;\
/* Sift the newly inserted node up in the heap to \
* restore the heap property. */\
while (pos > 0) {\
int parent = (pos - 1) >> 1;\
if (nodes_next[parent]->ssd <= ssd)\
break;\
FFSWAP(TrellisNode*, nodes_next[parent], nodes_next[pos]);\
pos = parent;\
}\
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;
generation++;
if (generation == 255) {
memset(hash, 0xff, 65536 * sizeof(*hash));
generation = 0;
}
// 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 = AV_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;
*samples = adpcm_ima_expand_nibble(cs, src[0] >> 4 , 3);
samples += avctx->channels;
src ++;
}
}
if (st)
samples--;
break;
case CODEC_ID_ADPCM_IMA_WAV:
if (avctx->block_align != 0 && buf_size > avctx->block_align)
buf_size = avctx->block_align;
// samples_per_block= (block_align-4*chanels)*8 / (bits_per_sample * chanels) + 1;
for(i=0; i<avctx->channels; i++){
cs = &(c->status[i]);
cs->predictor = *samples++ = (int16_t)bytestream_get_le16(&src);
cs->step_index = *src++;
if (cs->step_index > 88){
av_log(avctx, AV_LOG_ERROR, "ERROR: step_index = %i\n", cs->step_index);
cs->step_index = 88;
}
if (*src++) av_log(avctx, AV_LOG_ERROR, "unused byte should be null but is %d!!\n", src[-1]); /* unused */
}
while(src < buf + buf_size){
for(m=0; m<4; m++){
for(i=0; i<=st; i++)
*samples++ = adpcm_ima_expand_nibble(&c->status[i], src[4*i] & 0x0F, 3);
for(i=0; i<=st; i++)
*samples++ = adpcm_ima_expand_nibble(&c->status[i], src[4*i] >> 4 , 3);
src++;
}
src += 4*st;
}
break;
case CODEC_ID_ADPCM_4XM:
cs = &(c->status[0]);
c->status[0].predictor= (int16_t)bytestream_get_le16(&src);
if(st){
c->status[1].predictor= (int16_t)bytestream_get_le16(&src);
}
c->status[0].step_index= (int16_t)bytestream_get_le16(&src);
if(st){
c->status[1].step_index= (int16_t)bytestream_get_le16(&src);
}
if (cs->step_index < 0) cs->step_index = 0;
if (cs->step_index > 88) cs->step_index = 88;
m= (buf_size - (src - buf))>>st;
for(i=0; i<m; i++) {
*samples++ = adpcm_ima_expand_nibble(&c->status[0], src[i] & 0x0F, 4);
if (st)
*samples++ = adpcm_ima_expand_nibble(&c->status[1], src[i+m] & 0x0F, 4);
*samples++ = adpcm_ima_expand_nibble(&c->status[0], src[i] >> 4, 4);
if (st)
*samples++ = adpcm_ima_expand_nibble(&c->status[1], src[i+m] >> 4, 4);
}
src += m<<st;
break;
case CODEC_ID_ADPCM_MS:
if (avctx->block_align != 0 && buf_size > avctx->block_align)
buf_size = avctx->block_align;
n = buf_size - 7 * avctx->channels;
if (n < 0)
return -1;
block_predictor[0] = av_clip(*src++, 0, 6);
block_predictor[1] = 0;
if (st)
block_predictor[1] = av_clip(*src++, 0, 6);
c->status[0].idelta = (int16_t)bytestream_get_le16(&src);
if (st){
c->status[1].idelta = (int16_t)bytestream_get_le16(&src);
}
c->status[0].coeff1 = AdaptCoeff1[block_predictor[0]];
c->status[0].coeff2 = AdaptCoeff2[block_predictor[0]];
c->status[1].coeff1 = AdaptCoeff1[block_predictor[1]];
c->status[1].coeff2 = AdaptCoeff2[block_predictor[1]];
c->status[0].sample1 = bytestream_get_le16(&src);
if (st) c->status[1].sample1 = bytestream_get_le16(&src);
c->status[0].sample2 = bytestream_get_le16(&src);
if (st) c->status[1].sample2 = bytestream_get_le16(&src);
*samples++ = c->status[0].sample2;
if (st) *samples++ = c->status[1].sample2;
*samples++ = c->status[0].sample1;
if (st) *samples++ = c->status[1].sample1;
for(;n>0;n--) {
*samples++ = adpcm_ms_expand_nibble(&c->status[0 ], src[0] >> 4 );
*samples++ = adpcm_ms_expand_nibble(&c->status[st], src[0] & 0x0F);
src ++;
}
break;
case CODEC_ID_ADPCM_IMA_DK4:
if (avctx->block_align != 0 && buf_size > avctx->block_align)
buf_size = avctx->block_align;
c->status[0].predictor = (int16_t)bytestream_get_le16(&src);
c->status[0].step_index = *src++;
src++;
*samples++ = c->status[0].predictor;
if (st) {
c->status[1].predictor = (int16_t)bytestream_get_le16(&src);
c->status[1].step_index = *src++;
src++;
*samples++ = c->status[1].predictor;
}
while (src < buf + buf_size) {
/* take care of the top nibble (always left or mono channel) */
*samples++ = adpcm_ima_expand_nibble(&c->status[0],
src[0] >> 4, 3);
/* take care of the bottom nibble, which is right sample for
* stereo, or another mono sample */
if (st)
*samples++ = adpcm_ima_expand_nibble(&c->status[1],
src[0] & 0x0F, 3);
else
*samples++ = adpcm_ima_expand_nibble(&c->status[0],
src[0] & 0x0F, 3);
src++;
}
break;
case CODEC_ID_ADPCM_IMA_DK3:
if (avctx->block_align != 0 && buf_size > avctx->block_align)
buf_size = avctx->block_align;
if(buf_size + 16 > (samples_end - samples)*3/8)
return -1;
c->status[0].predictor = (int16_t)AV_RL16(src + 10);
c->status[1].predictor = (int16_t)AV_RL16(src + 12);
c->status[0].step_index = src[14];
c->status[1].step_index = src[15];
/* sign extend the predictors */
src += 16;
diff_channel = c->status[1].predictor;
/* the DK3_GET_NEXT_NIBBLE macro issues the break statement when
* the buffer is consumed */
while (1) {
/* for this algorithm, c->status[0] is the sum channel and
* c->status[1] is the diff channel */
/* process the first predictor of the sum channel */
DK3_GET_NEXT_NIBBLE();
adpcm_ima_expand_nibble(&c->status[0], nibble, 3);
/* process the diff channel predictor */
DK3_GET_NEXT_NIBBLE();
adpcm_ima_expand_nibble(&c->status[1], nibble, 3);
/* process the first pair of stereo PCM samples */
diff_channel = (diff_channel + c->status[1].predictor) / 2;
*samples++ = c->status[0].predictor + c->status[1].predictor;
*samples++ = c->status[0].predictor - c->status[1].predictor;
/* process the second predictor of the sum channel */
DK3_GET_NEXT_NIBBLE();
adpcm_ima_expand_nibble(&c->status[0], nibble, 3);
/* process the second pair of stereo PCM samples */
diff_channel = (diff_channel + c->status[1].predictor) / 2;
*samples++ = c->status[0].predictor + c->status[1].predictor;
*samples++ = c->status[0].predictor - c->status[1].predictor;
}
break;
case CODEC_ID_ADPCM_IMA_ISS:
c->status[0].predictor = (int16_t)AV_RL16(src + 0);
c->status[0].step_index = src[2];
src += 4;
if(st) {
c->status[1].predictor = (int16_t)AV_RL16(src + 0);
c->status[1].step_index = src[2];
src += 4;
}
while (src < buf + buf_size) {
if (st) {
*samples++ = adpcm_ima_expand_nibble(&c->status[0],
src[0] >> 4 , 3);
*samples++ = adpcm_ima_expand_nibble(&c->status[1],
src[0] & 0x0F, 3);
} else {
*samples++ = adpcm_ima_expand_nibble(&c->status[0],
src[0] & 0x0F, 3);
*samples++ = adpcm_ima_expand_nibble(&c->status[0],
src[0] >> 4 , 3);
}
src++;
}
break;
case CODEC_ID_ADPCM_IMA_WS:
/* no per-block initialization; just start decoding the data */
while (src < buf + buf_size) {
if (st) {
*samples++ = adpcm_ima_expand_nibble(&c->status[0],
src[0] >> 4 , 3);
*samples++ = adpcm_ima_expand_nibble(&c->status[1],
src[0] & 0x0F, 3);
} else {
*samples++ = adpcm_ima_expand_nibble(&c->status[0],
src[0] >> 4 , 3);
*samples++ = adpcm_ima_expand_nibble(&c->status[0],
src[0] & 0x0F, 3);
}
src++;
}
break;
case CODEC_ID_ADPCM_XA:
while (buf_size >= 128) {
xa_decode(samples, src, &c->status[0], &c->status[1],
avctx->channels);
src += 128;
samples += 28 * 8;
buf_size -= 128;
}
break;
case CODEC_ID_ADPCM_IMA_EA_EACS:
samples_in_chunk = bytestream_get_le32(&src) >> (1-st);
if (samples_in_chunk > buf_size-4-(8<<st)) {
src += buf_size - 4;
break;
}
for (i=0; i<=st; i++)
c->status[i].step_index = bytestream_get_le32(&src);
for (i=0; i<=st; i++)
c->status[i].predictor = bytestream_get_le32(&src);
for (; samples_in_chunk; samples_in_chunk--, src++) {
*samples++ = adpcm_ima_expand_nibble(&c->status[0], *src>>4, 3);
*samples++ = adpcm_ima_expand_nibble(&c->status[st], *src&0x0F, 3);
}
break;
case CODEC_ID_ADPCM_IMA_EA_SEAD:
for (; src < buf+buf_size; src++) {
*samples++ = adpcm_ima_expand_nibble(&c->status[0], src[0] >> 4, 6);
*samples++ = adpcm_ima_expand_nibble(&c->status[st],src[0]&0x0F, 6);
}
break;
case CODEC_ID_ADPCM_EA:
if (buf_size < 4 || AV_RL32(src) >= ((buf_size - 12) * 2)) {
src += buf_size;
break;
}
samples_in_chunk = AV_RL32(src);
src += 4;
current_left_sample = (int16_t)bytestream_get_le16(&src);
previous_left_sample = (int16_t)bytestream_get_le16(&src);
current_right_sample = (int16_t)bytestream_get_le16(&src);
previous_right_sample = (int16_t)bytestream_get_le16(&src);
for (count1 = 0; count1 < samples_in_chunk/28;count1++) {
coeff1l = ea_adpcm_table[ *src >> 4 ];
coeff2l = ea_adpcm_table[(*src >> 4 ) + 4];
coeff1r = ea_adpcm_table[*src & 0x0F];
coeff2r = ea_adpcm_table[(*src & 0x0F) + 4];
src++;
shift_left = (*src >> 4 ) + 8;
shift_right = (*src & 0x0F) + 8;
src++;
for (count2 = 0; count2 < 28; count2++) {
next_left_sample = (int32_t)((*src & 0xF0) << 24) >> shift_left;
next_right_sample = (int32_t)((*src & 0x0F) << 28) >> shift_right;
src++;
next_left_sample = (next_left_sample +
(current_left_sample * coeff1l) +
(previous_left_sample * coeff2l) + 0x80) >> 8;
next_right_sample = (next_right_sample +
(current_right_sample * coeff1r) +
(previous_right_sample * coeff2r) + 0x80) >> 8;
previous_left_sample = current_left_sample;
current_left_sample = av_clip_int16(next_left_sample);
previous_right_sample = current_right_sample;
current_right_sample = av_clip_int16(next_right_sample);
*samples++ = (unsigned short)current_left_sample;
*samples++ = (unsigned short)current_right_sample;
}
}
if (src - buf == buf_size - 2)
src += 2; // Skip terminating 0x0000
break;
case CODEC_ID_ADPCM_EA_MAXIS_XA:
for(channel = 0; channel < avctx->channels; channel++) {
for (i=0; i<2; i++)
coeff[channel][i] = ea_adpcm_table[(*src >> 4) + 4*i];
shift[channel] = (*src & 0x0F) + 8;
src++;
}
for (count1 = 0; count1 < (buf_size - avctx->channels) / avctx->channels; count1++) {
for(i = 4; i >= 0; i-=4) { /* Pairwise samples LL RR (st) or LL LL (mono) */
for(channel = 0; channel < avctx->channels; channel++) {
int32_t sample = (int32_t)(((*(src+channel) >> i) & 0x0F) << 0x1C) >> shift[channel];
sample = (sample +
c->status[channel].sample1 * coeff[channel][0] +
c->status[channel].sample2 * coeff[channel][1] + 0x80) >> 8;
c->status[channel].sample2 = c->status[channel].sample1;
c->status[channel].sample1 = av_clip_int16(sample);
*samples++ = c->status[channel].sample1;
}
}
src+=avctx->channels;
}
break;
case CODEC_ID_ADPCM_EA_R1:
case CODEC_ID_ADPCM_EA_R2:
case CODEC_ID_ADPCM_EA_R3: {
/* channel numbering
2chan: 0=fl, 1=fr
4chan: 0=fl, 1=rl, 2=fr, 3=rr
6chan: 0=fl, 1=c, 2=fr, 3=rl, 4=rr, 5=sub */
const int big_endian = avctx->codec->id == CODEC_ID_ADPCM_EA_R3;
int32_t previous_sample, current_sample, next_sample;
int32_t coeff1, coeff2;
uint8_t shift;
unsigned int channel;
uint16_t *samplesC;
const uint8_t *srcC;
const uint8_t *src_end = buf + buf_size;
samples_in_chunk = (big_endian ? bytestream_get_be32(&src)
: bytestream_get_le32(&src)) / 28;
if (samples_in_chunk > UINT32_MAX/(28*avctx->channels) ||
28*samples_in_chunk*avctx->channels > samples_end-samples) {
src += buf_size - 4;
break;
}
for (channel=0; channel<avctx->channels; channel++) {
int32_t offset = (big_endian ? bytestream_get_be32(&src)
: bytestream_get_le32(&src))
+ (avctx->channels-channel-1) * 4;
if ((offset < 0) || (offset >= src_end - src - 4)) break;
srcC = src + offset;
samplesC = samples + channel;
if (avctx->codec->id == CODEC_ID_ADPCM_EA_R1) {
current_sample = (int16_t)bytestream_get_le16(&srcC);
previous_sample = (int16_t)bytestream_get_le16(&srcC);
} else {
current_sample = c->status[channel].predictor;
previous_sample = c->status[channel].prev_sample;
}
for (count1=0; count1<samples_in_chunk; count1++) {
if (*srcC == 0xEE) { /* only seen in R2 and R3 */
srcC++;
if (srcC > src_end - 30*2) break;
current_sample = (int16_t)bytestream_get_be16(&srcC);
previous_sample = (int16_t)bytestream_get_be16(&srcC);
for (count2=0; count2<28; count2++) {
*samplesC = (int16_t)bytestream_get_be16(&srcC);
samplesC += avctx->channels;
}
} else {
coeff1 = ea_adpcm_table[ *srcC>>4 ];
coeff2 = ea_adpcm_table[(*srcC>>4) + 4];
shift = (*srcC++ & 0x0F) + 8;
if (srcC > src_end - 14) break;
for (count2=0; count2<28; count2++) {
if (count2 & 1)
next_sample = (int32_t)((*srcC++ & 0x0F) << 28) >> shift;
else
next_sample = (int32_t)((*srcC & 0xF0) << 24) >> shift;
next_sample += (current_sample * coeff1) +
(previous_sample * coeff2);
next_sample = av_clip_int16(next_sample >> 8);
previous_sample = current_sample;
current_sample = next_sample;
*samplesC = current_sample;
samplesC += avctx->channels;
}
}
}
if (avctx->codec->id != CODEC_ID_ADPCM_EA_R1) {
c->status[channel].predictor = current_sample;
c->status[channel].prev_sample = previous_sample;
}
}
src = src + buf_size - (4 + 4*avctx->channels);
samples += 28 * samples_in_chunk * avctx->channels;
break;
}
case CODEC_ID_ADPCM_EA_XAS:
if (samples_end-samples < 32*4*avctx->channels
|| buf_size < (4+15)*4*avctx->channels) {
src += buf_size;
break;
}
for (channel=0; channel<avctx->channels; channel++) {
int coeff[2][4], shift[4];
short *s2, *s = &samples[channel];
for (n=0; n<4; n++, s+=32*avctx->channels) {
for (i=0; i<2; i++)
coeff[i][n] = ea_adpcm_table[(src[0]&0x0F)+4*i];
shift[n] = (src[2]&0x0F) + 8;
for (s2=s, i=0; i<2; i++, src+=2, s2+=avctx->channels)
s2[0] = (src[0]&0xF0) + (src[1]<<8);
}
for (m=2; m<32; m+=2) {
s = &samples[m*avctx->channels + channel];
for (n=0; n<4; n++, src++, s+=32*avctx->channels) {
for (s2=s, i=0; i<8; i+=4, s2+=avctx->channels) {
int level = (int32_t)((*src & (0xF0>>i)) << (24+i)) >> shift[n];
int pred = s2[-1*avctx->channels] * coeff[0][n]
+ s2[-2*avctx->channels] * coeff[1][n];
s2[0] = av_clip_int16((level + pred + 0x80) >> 8);
}
}
}
}
samples += 32*4*avctx->channels;
break;
case CODEC_ID_ADPCM_IMA_AMV:
case CODEC_ID_ADPCM_IMA_SMJPEG:
c->status[0].predictor = (int16_t)bytestream_get_le16(&src);
c->status[0].step_index = bytestream_get_le16(&src);
if (avctx->codec->id == CODEC_ID_ADPCM_IMA_AMV)
src+=4;
while (src < buf + buf_size) {
char hi, lo;
lo = *src & 0x0F;
hi = *src >> 4;
if (avctx->codec->id == CODEC_ID_ADPCM_IMA_AMV)
FFSWAP(char, hi, lo);
*samples++ = adpcm_ima_expand_nibble(&c->status[0],
lo, 3);
*samples++ = adpcm_ima_expand_nibble(&c->status[0],
hi, 3);
src++;
}
break;
case CODEC_ID_ADPCM_CT:
while (src < buf + buf_size) {
if (st) {
*samples++ = adpcm_ct_expand_nibble(&c->status[0],
src[0] >> 4);
*samples++ = adpcm_ct_expand_nibble(&c->status[1],
src[0] & 0x0F);
} else {
*samples++ = adpcm_ct_expand_nibble(&c->status[0],
src[0] >> 4);
*samples++ = adpcm_ct_expand_nibble(&c->status[0],
src[0] & 0x0F);
}
src++;
}
break;
case CODEC_ID_ADPCM_SBPRO_4:
case CODEC_ID_ADPCM_SBPRO_3:
case CODEC_ID_ADPCM_SBPRO_2:
if (!c->status[0].step_index) {
/* the first byte is a raw sample */
*samples++ = 128 * (*src++ - 0x80);
if (st)
*samples++ = 128 * (*src++ - 0x80);
c->status[0].step_index = 1;
}
if (avctx->codec->id == CODEC_ID_ADPCM_SBPRO_4) {
while (src < buf + buf_size) {
*samples++ = adpcm_sbpro_expand_nibble(&c->status[0],
src[0] >> 4, 4, 0);
*samples++ = adpcm_sbpro_expand_nibble(&c->status[st],
src[0] & 0x0F, 4, 0);
src++;
}
} else if (avctx->codec->id == CODEC_ID_ADPCM_SBPRO_3) {
while (src < buf + buf_size && samples + 2 < samples_end) {
*samples++ = adpcm_sbpro_expand_nibble(&c->status[0],
src[0] >> 5 , 3, 0);
*samples++ = adpcm_sbpro_expand_nibble(&c->status[0],
(src[0] >> 2) & 0x07, 3, 0);
*samples++ = adpcm_sbpro_expand_nibble(&c->status[0],
src[0] & 0x03, 2, 0);
src++;
}
} else {
while (src < buf + buf_size && samples + 3 < samples_end) {
*samples++ = adpcm_sbpro_expand_nibble(&c->status[0],
src[0] >> 6 , 2, 2);
*samples++ = adpcm_sbpro_expand_nibble(&c->status[st],
(src[0] >> 4) & 0x03, 2, 2);
*samples++ = adpcm_sbpro_expand_nibble(&c->status[0],
(src[0] >> 2) & 0x03, 2, 2);
*samples++ = adpcm_sbpro_expand_nibble(&c->status[st],
src[0] & 0x03, 2, 2);
src++;
}
}
break;
case CODEC_ID_ADPCM_SWF:
{
GetBitContext gb;
const int *table;
int k0, signmask, nb_bits, count;
int size = buf_size*8;
init_get_bits(&gb, buf, size);
//read bits & initial values
nb_bits = get_bits(&gb, 2)+2;
//av_log(NULL,AV_LOG_INFO,"nb_bits: %d\n", nb_bits);
table = swf_index_tables[nb_bits-2];
k0 = 1 << (nb_bits-2);
signmask = 1 << (nb_bits-1);
while (get_bits_count(&gb) <= size - 22*avctx->channels) {
for (i = 0; i < avctx->channels; i++) {
*samples++ = c->status[i].predictor = get_sbits(&gb, 16);
c->status[i].step_index = get_bits(&gb, 6);
}
for (count = 0; get_bits_count(&gb) <= size - nb_bits*avctx->channels && count < 4095; count++) {
int i;
for (i = 0; i < avctx->channels; i++) {
// similar to IMA adpcm
int delta = get_bits(&gb, nb_bits);
int step = step_table[c->status[i].step_index];
long vpdiff = 0; // vpdiff = (delta+0.5)*step/4
int k = k0;
do {
if (delta & k)
vpdiff += step;
step >>= 1;
k >>= 1;
} while(k);
vpdiff += step;
if (delta & signmask)
c->status[i].predictor -= vpdiff;
else
c->status[i].predictor += vpdiff;
c->status[i].step_index += table[delta & (~signmask)];
c->status[i].step_index = av_clip(c->status[i].step_index, 0, 88);
c->status[i].predictor = av_clip_int16(c->status[i].predictor);
*samples++ = c->status[i].predictor;
if (samples >= samples_end) {
av_log(avctx, AV_LOG_ERROR, "allocated output buffer is too small\n");
return -1;
}
}
}
}
src += buf_size;
break;
}
case CODEC_ID_ADPCM_YAMAHA:
while (src < buf + buf_size) {
if (st) {
*samples++ = adpcm_yamaha_expand_nibble(&c->status[0],
src[0] & 0x0F);
*samples++ = adpcm_yamaha_expand_nibble(&c->status[1],
src[0] >> 4 );
} else {
*samples++ = adpcm_yamaha_expand_nibble(&c->status[0],
src[0] & 0x0F);
*samples++ = adpcm_yamaha_expand_nibble(&c->status[0],
src[0] >> 4 );
}
src++;
}
break;
case CODEC_ID_ADPCM_THP:
{
int table[2][16];
unsigned int samplecnt;
int prev[2][2];
int ch;
if (buf_size < 80) {
av_log(avctx, AV_LOG_ERROR, "frame too small\n");
return -1;
}
src+=4;
samplecnt = bytestream_get_be32(&src);
for (i = 0; i < 32; i++)
table[0][i] = (int16_t)bytestream_get_be16(&src);
/* Initialize the previous sample. */
for (i = 0; i < 4; i++)
prev[0][i] = (int16_t)bytestream_get_be16(&src);
if (samplecnt >= (samples_end - samples) / (st + 1)) {
av_log(avctx, AV_LOG_ERROR, "allocated output buffer is too small\n");
return -1;
}
for (ch = 0; ch <= st; ch++) {
samples = (unsigned short *) data + ch;
/* Read in every sample for this channel. */
for (i = 0; i < samplecnt / 14; i++) {
int index = (*src >> 4) & 7;
unsigned int exp = 28 - (*src++ & 15);
int factor1 = table[ch][index * 2];
int factor2 = table[ch][index * 2 + 1];
/* Decode 14 samples. */
for (n = 0; n < 14; n++) {
int32_t sampledat;
if(n&1) sampledat= *src++ <<28;
else sampledat= (*src&0xF0)<<24;
sampledat = ((prev[ch][0]*factor1
+ prev[ch][1]*factor2) >> 11) + (sampledat>>exp);
*samples = av_clip_int16(sampledat);
prev[ch][1] = prev[ch][0];
prev[ch][0] = *samples++;
/* In case of stereo, skip one sample, this sample
is for the other channel. */
samples += st;
}
}
}
/* In the previous loop, in case stereo is used, samples is
increased exactly one time too often. */
samples -= st;
break;
}
default:
return -1;
}
*data_size = (uint8_t *)samples - (uint8_t *)data;
return src - buf;
}
#if CONFIG_ENCODERS
#define ADPCM_ENCODER(id,name,long_name_) \
AVCodec name ## _encoder = { \
#name, \
AVMEDIA_TYPE_AUDIO, \
id, \
sizeof(ADPCMContext), \
adpcm_encode_init, \
adpcm_encode_frame, \
adpcm_encode_close, \
NULL, \
.sample_fmts = (const enum AVSampleFormat[]){AV_SAMPLE_FMT_S16,AV_SAMPLE_FMT_NONE}, \
.long_name = NULL_IF_CONFIG_SMALL(long_name_), \
};
#else
#define ADPCM_ENCODER(id,name,long_name_)
#endif
#if CONFIG_DECODERS
#define ADPCM_DECODER(id,name,long_name_) \
AVCodec name ## _decoder = { \
#name, \
AVMEDIA_TYPE_AUDIO, \
id, \
sizeof(ADPCMContext), \
adpcm_decode_init, \
NULL, \
NULL, \
adpcm_decode_frame, \
.long_name = NULL_IF_CONFIG_SMALL(long_name_), \
};
#else
#define ADPCM_DECODER(id,name,long_name_)
#endif
#define ADPCM_CODEC(id,name,long_name_) \
ADPCM_ENCODER(id,name,long_name_) ADPCM_DECODER(id,name,long_name_)
/* Note: Do not forget to add new entries to the Makefile as well. */
ADPCM_DECODER(CODEC_ID_ADPCM_4XM, adpcm_4xm, "ADPCM 4X Movie");
ADPCM_DECODER(CODEC_ID_ADPCM_CT, adpcm_ct, "ADPCM Creative Technology");
ADPCM_DECODER(CODEC_ID_ADPCM_EA, adpcm_ea, "ADPCM Electronic Arts");
ADPCM_DECODER(CODEC_ID_ADPCM_EA_MAXIS_XA, adpcm_ea_maxis_xa, "ADPCM Electronic Arts Maxis CDROM XA");
ADPCM_DECODER(CODEC_ID_ADPCM_EA_R1, adpcm_ea_r1, "ADPCM Electronic Arts R1");
ADPCM_DECODER(CODEC_ID_ADPCM_EA_R2, adpcm_ea_r2, "ADPCM Electronic Arts R2");
ADPCM_DECODER(CODEC_ID_ADPCM_EA_R3, adpcm_ea_r3, "ADPCM Electronic Arts R3");
ADPCM_DECODER(CODEC_ID_ADPCM_EA_XAS, adpcm_ea_xas, "ADPCM Electronic Arts XAS");
ADPCM_DECODER(CODEC_ID_ADPCM_IMA_AMV, adpcm_ima_amv, "ADPCM IMA AMV");
ADPCM_DECODER(CODEC_ID_ADPCM_IMA_DK3, adpcm_ima_dk3, "ADPCM IMA Duck DK3");
ADPCM_DECODER(CODEC_ID_ADPCM_IMA_DK4, adpcm_ima_dk4, "ADPCM IMA Duck DK4");
ADPCM_DECODER(CODEC_ID_ADPCM_IMA_EA_EACS, adpcm_ima_ea_eacs, "ADPCM IMA Electronic Arts EACS");
ADPCM_DECODER(CODEC_ID_ADPCM_IMA_EA_SEAD, adpcm_ima_ea_sead, "ADPCM IMA Electronic Arts SEAD");
ADPCM_DECODER(CODEC_ID_ADPCM_IMA_ISS, adpcm_ima_iss, "ADPCM IMA Funcom ISS");
ADPCM_CODEC (CODEC_ID_ADPCM_IMA_QT, adpcm_ima_qt, "ADPCM IMA QuickTime");
ADPCM_DECODER(CODEC_ID_ADPCM_IMA_SMJPEG, adpcm_ima_smjpeg, "ADPCM IMA Loki SDL MJPEG");
ADPCM_CODEC (CODEC_ID_ADPCM_IMA_WAV, adpcm_ima_wav, "ADPCM IMA WAV");
ADPCM_DECODER(CODEC_ID_ADPCM_IMA_WS, adpcm_ima_ws, "ADPCM IMA Westwood");
ADPCM_CODEC (CODEC_ID_ADPCM_MS, adpcm_ms, "ADPCM Microsoft");
ADPCM_DECODER(CODEC_ID_ADPCM_SBPRO_2, adpcm_sbpro_2, "ADPCM Sound Blaster Pro 2-bit");
ADPCM_DECODER(CODEC_ID_ADPCM_SBPRO_3, adpcm_sbpro_3, "ADPCM Sound Blaster Pro 2.6-bit");
ADPCM_DECODER(CODEC_ID_ADPCM_SBPRO_4, adpcm_sbpro_4, "ADPCM Sound Blaster Pro 4-bit");
ADPCM_CODEC (CODEC_ID_ADPCM_SWF, adpcm_swf, "ADPCM Shockwave Flash");
ADPCM_DECODER(CODEC_ID_ADPCM_THP, adpcm_thp, "ADPCM Nintendo Gamecube THP");
ADPCM_DECODER(CODEC_ID_ADPCM_XA, adpcm_xa, "ADPCM CDROM XA");
ADPCM_CODEC (CODEC_ID_ADPCM_YAMAHA, adpcm_yamaha, "ADPCM Yamaha");
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