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ffmpeg/libavcodec/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 "bitstream.h"
#include "bytestream.h"
/**
* @file adpcm.c
* 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)
* 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
#define CLAMP_TO_SHORT(value) \
if (value > 32767) \
value = 32767; \
else if (value < -32768) \
value = -32768; \
/* 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
};
static const int AdaptCoeff1[] = {
256, 512, 0, 192, 240, 460, 392
};
static const int AdaptCoeff2[] = {
0, -256, 0, 64, 0, -208, -232
};
/* 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
};
static const int ct_adpcm_table[8] = {
0x00E6, 0x00E6, 0x00E6, 0x00E6,
0x0133, 0x0199, 0x0200, 0x0266
};
// 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 ADPCMContext {
int channel; /* for stereo MOVs, decode left, then decode right, then tell it's decoded */
ADPCMChannelStatus status[2];
short sample_buffer[32]; /* hold left samples while waiting for right samples */
} ADPCMContext;
/* XXX: implement encoding */
#ifdef CONFIG_ENCODERS
static int adpcm_encode_init(AVCodecContext *avctx)
{
if (avctx->channels > 2)
return -1; /* only stereo or mono =) */
switch(avctx->codec->id) {
case CODEC_ID_ADPCM_IMA_QT:
av_log(avctx, AV_LOG_ERROR, "ADPCM: codec adpcm_ima_qt unsupported for encoding !\n");
avctx->frame_size = 64; /* XXX: can multiple of avctx->channels * 64 (left and right blocks are interleaved) */
return -1;
break;
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_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;
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");
return -1;
}
avctx->frame_size = 512 * (avctx->sample_rate / 11025);
break;
default:
return -1;
break;
}
avctx->coded_frame= avcodec_alloc_frame();
avctx->coded_frame->key_frame= 1;
return 0;
}
static int adpcm_encode_close(AVCodecContext *avctx)
{
av_freep(&avctx->coded_frame);
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 = c->prev_sample + ((step_table[c->step_index] * yamaha_difflookup[nibble]) / 8);
CLAMP_TO_SHORT(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))) / 256;
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;
CLAMP_TO_SHORT(predictor);
c->sample2 = c->sample1;
c->sample1 = 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->predictor + ((c->step * yamaha_difflookup[nibble]) / 8);
CLAMP_TO_SHORT(c->predictor);
c->step = (c->step * yamaha_indexscale[nibble]) >> 8;
c->step = av_clip(c->step, 127, 24567);
return nibble;
}
typedef struct TrellisPath {
int nibble;
int prev;
} TrellisPath;
typedef struct TrellisNode {
uint32_t ssd;
int path;
int sample1;
int sample2;
int step;
} TrellisNode;
static void adpcm_compress_trellis(AVCodecContext *avctx, const short *samples,
uint8_t *dst, ADPCMChannelStatus *c, int n)
{
#define FREEZE_INTERVAL 128
//FIXME 6% faster if frontier is a compile-time constant
const int frontier = 1 << avctx->trellis;
const int stride = avctx->channels;
const int version = avctx->codec->id;
const int max_paths = frontier*FREEZE_INTERVAL;
TrellisPath paths[max_paths], *p;
TrellisNode node_buf[2][frontier];
TrellisNode *nodep_buf[2][frontier];
TrellisNode **nodes = nodep_buf[0]; // nodes[] is always sorted by .ssd
TrellisNode **nodes_next = nodep_buf[1];
int pathn = 0, froze = -1, i, j, k;
assert(!(max_paths&(max_paths-1)));
memset(nodep_buf, 0, sizeof(nodep_buf));
nodes[0] = &node_buf[1][0];
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)
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[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)) / 256;
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;\
CLAMP_TO_SHORT(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 < max_paths);\
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) {
#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;
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_QT: /* XXX: can't test until we get .mov writer */
break;
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[1];
/* 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) {
uint8_t buf[2][n*8];
adpcm_compress_trellis(avctx, samples, buf[0], &c->status[0], n*8);
if(avctx->channels == 2)
adpcm_compress_trellis(avctx, samples+1, buf[1], &c->status[1], n*8);
for(i=0; i<n; i++) {
*dst++ = buf[0][8*i+0] | (buf[0][8*i+1] << 4);
*dst++ = buf[0][8*i+2] | (buf[0][8*i+3] << 4);
*dst++ = buf[0][8*i+4] | (buf[0][8*i+5] << 4);
*dst++ = buf[0][8*i+6] | (buf[0][8*i+7] << 4);
if (avctx->channels == 2) {
*dst++ = buf[1][8*i+0] | (buf[1][8*i+1] << 4);
*dst++ = buf[1][8*i+2] | (buf[1][8*i+3] << 4);
*dst++ = buf[1][8*i+4] | (buf[1][8*i+5] << 4);
*dst++ = buf[1][8*i+6] | (buf[1][8*i+7] << 4);
}
}
} else
for (; n>0; n--) {
*dst = adpcm_ima_compress_sample(&c->status[0], samples[0]) & 0x0F;
*dst |= (adpcm_ima_compress_sample(&c->status[0], samples[avctx->channels]) << 4) & 0xF0;
dst++;
*dst = adpcm_ima_compress_sample(&c->status[0], samples[avctx->channels * 2]) & 0x0F;
*dst |= (adpcm_ima_compress_sample(&c->status[0], samples[avctx->channels * 3]) << 4) & 0xF0;
dst++;
*dst = adpcm_ima_compress_sample(&c->status[0], samples[avctx->channels * 4]) & 0x0F;
*dst |= (adpcm_ima_compress_sample(&c->status[0], samples[avctx->channels * 5]) << 4) & 0xF0;
dst++;
*dst = adpcm_ima_compress_sample(&c->status[0], samples[avctx->channels * 6]) & 0x0F;
*dst |= (adpcm_ima_compress_sample(&c->status[0], samples[avctx->channels * 7]) << 4) & 0xF0;
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_SWF:
{
int i;
PutBitContext pb;
init_put_bits(&pb, dst, buf_size*8);
//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_bits(&pb, 16, samples[i] & 0xFFFF);
put_bits(&pb, 6, c->status[i].step_index);
c->status[i].prev_sample = (signed short)samples[i];
}
for (i=0; i<avctx->frame_size; i++) {
put_bits(&pb, 4, adpcm_ima_compress_sample(&c->status[0], samples[avctx->channels*i]) & 0xF);
if (avctx->channels == 2)
put_bits(&pb, 4, adpcm_ima_compress_sample(&c->status[1], samples[2*i+1]) & 0xF);
}
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].sample1= *samples++;
bytestream_put_le16(&dst, c->status[i].sample1);
}
for(i=0; i<avctx->channels; i++){
c->status[i].sample2= *samples++;
bytestream_put_le16(&dst, c->status[i].sample2);
}
if(avctx->trellis > 0) {
int n = avctx->block_align - 7*avctx->channels;
uint8_t buf[2][n];
if(avctx->channels == 1) {
n *= 2;
adpcm_compress_trellis(avctx, samples, buf[0], &c->status[0], n);
for(i=0; i<n; i+=2)
*dst++ = (buf[0][i] << 4) | buf[0][i+1];
} else {
adpcm_compress_trellis(avctx, samples, buf[0], &c->status[0], n);
adpcm_compress_trellis(avctx, samples+1, buf[1], &c->status[1], n);
for(i=0; i<n; i++)
*dst++ = (buf[0][i] << 4) | buf[1][i];
}
} 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) {
uint8_t buf[2][n*2];
n *= 2;
if(avctx->channels == 1) {
adpcm_compress_trellis(avctx, samples, buf[0], &c->status[0], n);
for(i=0; i<n; i+=2)
*dst++ = buf[0][i] | (buf[0][i+1] << 4);
} else {
adpcm_compress_trellis(avctx, samples, buf[0], &c->status[0], n);
adpcm_compress_trellis(avctx, samples+1, buf[1], &c->status[1], n);
for(i=0; i<n; i++)
*dst++ = buf[0][i] | (buf[1][i] << 4);
}
} else
for (; n>0; n--) {
for(i = 0; i < avctx->channels; i++) {
int nibble;
nibble = adpcm_yamaha_compress_sample(&c->status[i], samples[i]);
nibble |= adpcm_yamaha_compress_sample(&c->status[i], samples[i+avctx->channels]) << 4;
*dst++ = nibble;
}
samples += 2 * avctx->channels;
}
break;
default:
return -1;
}
return dst - frame;
}
#endif //CONFIG_ENCODERS
static int adpcm_decode_init(AVCodecContext * avctx)
{
ADPCMContext *c = avctx->priv_data;
if(avctx->channels > 2U){
return -1;
}
c->channel = 0;
c->status[0].predictor = c->status[1].predictor = 0;
c->status[0].step_index = c->status[1].step_index = 0;
c->status[0].step = c->status[1].step = 0;
switch(avctx->codec->id) {
case CODEC_ID_ADPCM_CT:
c->status[0].step = c->status[1].step = 511;
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;
}
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;
CLAMP_TO_SHORT(predictor);
c->predictor = predictor;
c->step_index = step_index;
return (short)predictor;
}
static inline short adpcm_ms_expand_nibble(ADPCMChannelStatus *c, char nibble)
{
int predictor;
predictor = (((c->sample1) * (c->coeff1)) + ((c->sample2) * (c->coeff2))) / 256;
predictor += (signed)((nibble & 0x08)?(nibble - 0x10):(nibble)) * c->idelta;
CLAMP_TO_SHORT(predictor);
c->sample2 = c->sample1;
c->sample1 = predictor;
c->idelta = (AdaptationTable[(int)nibble] * c->idelta) >> 8;
if (c->idelta < 16) c->idelta = 16;
return (short)predictor;
}
static inline short adpcm_ct_expand_nibble(ADPCMChannelStatus *c, char nibble)
{
int predictor;
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 = c->predictor;
/* predictor update is not so trivial: predictor is multiplied on 254/256 before updating */
if(sign)
predictor = ((predictor * 254) >> 8) - diff;
else
predictor = ((predictor * 254) >> 8) + diff;
/* calculate new step and clamp it to range 511..32767 */
new_step = (ct_adpcm_table[nibble & 7] * c->step) >> 8;
c->step = new_step;
if(c->step < 511)
c->step = 511;
if(c->step > 32767)
c->step = 32767;
CLAMP_TO_SHORT(predictor);
c->predictor = predictor;
return (short)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);
if (sign)
c->predictor -= diff;
else
c->predictor += diff;
/* clamp result */
if (c->predictor > 16256)
c->predictor = 16256;
else if (c->predictor < -16384)
c->predictor = -16384;
/* 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;
CLAMP_TO_SHORT(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);
CLAMP_TO_SHORT(s);
*out = s;
out += inc;
s_2 = s_1;
s_1 = s;
}
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);
CLAMP_TO_SHORT(s);
*out = s;
out += inc;
s_2 = s_1;
s_1 = s;
}
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) & 0x0F; \
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,
uint8_t *buf, int buf_size)
{
ADPCMContext *c = avctx->priv_data;
ADPCMChannelStatus *cs;
int n, m, channel, i;
int block_predictor[2];
short *samples;
short *samples_end;
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;
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);/* >> 2*avctx->channels;*/
channel = c->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;
CLAMP_TO_SHORT(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];
if (st && channel)
samples++;
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) & 0x0F, 3);
samples += avctx->channels;
src ++;
}
if(st) { /* handle stereo interlacing */
c->channel = (channel + 1) % 2; /* we get one packet for left, then one for right data */
if(channel == 1) { /* wait for the other packet before outputing anything */
return src - buf;
}
}
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 = (int16_t)(src[0] + (src[1]<<8));
src+=2;
// XXX: is this correct ??: *samples++ = cs->predictor;
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)(src[0] + (src[1]<<8)); src+=2;
if(st){
c->status[1].predictor= (int16_t)(src[0] + (src[1]<<8)); src+=2;
}
c->status[0].step_index= (int16_t)(src[0] + (src[1]<<8)); src+=2;
if(st){
c->status[1].step_index= (int16_t)(src[0] + (src[1]<<8)); src+=2;
}
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, 7);
block_predictor[1] = 0;
if (st)
block_predictor[1] = av_clip(*src++, 0, 7);
c->status[0].idelta = (int16_t)((*src & 0xFF) | ((src[1] << 8) & 0xFF00));
src+=2;
if (st){
c->status[1].idelta = (int16_t)((*src & 0xFF) | ((src[1] << 8) & 0xFF00));
src+=2;
}
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 = ((*src & 0xFF) | ((src[1] << 8) & 0xFF00));
src+=2;
if (st) c->status[1].sample1 = ((*src & 0xFF) | ((src[1] << 8) & 0xFF00));
if (st) src+=2;
c->status[0].sample2 = ((*src & 0xFF) | ((src[1] << 8) & 0xFF00));
src+=2;
if (st) c->status[1].sample2 = ((*src & 0xFF) | ((src[1] << 8) & 0xFF00));
if (st) src+=2;
*samples++ = c->status[0].sample1;
if (st) *samples++ = c->status[1].sample1;
*samples++ = c->status[0].sample2;
if (st) *samples++ = c->status[1].sample2;
for(;n>0;n--) {
*samples++ = adpcm_ms_expand_nibble(&c->status[0], (src[0] >> 4) & 0x0F);
*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)(src[0] | (src[1] << 8));
c->status[0].step_index = src[2];
src += 4;
*samples++ = c->status[0].predictor;
if (st) {
c->status[1].predictor = (int16_t)(src[0] | (src[1] << 8));
c->status[1].step_index = src[2];
src += 4;
*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) & 0x0F, 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)(src[10] | (src[11] << 8));
c->status[1].predictor = (int16_t)(src[12] | (src[13] << 8));
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_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) & 0x0F, 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) & 0x0F, 3);
*samples++ = adpcm_ima_expand_nibble(&c->status[0],
src[0] & 0x0F, 3);
}
src++;
}
break;
case CODEC_ID_ADPCM_XA:
c->status[0].sample1 = c->status[0].sample2 =
c->status[1].sample1 = c->status[1].sample2 = 0;
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_EA:
samples_in_chunk = AV_RL32(src);
if (samples_in_chunk >= ((buf_size - 12) * 2)) {
src += buf_size;
break;
}
src += 4;
current_left_sample = (int16_t)AV_RL16(src);
src += 2;
previous_left_sample = (int16_t)AV_RL16(src);
src += 2;
current_right_sample = (int16_t)AV_RL16(src);
src += 2;
previous_right_sample = (int16_t)AV_RL16(src);
src += 2;
for (count1 = 0; count1 < samples_in_chunk/28;count1++) {
coeff1l = ea_adpcm_table[(*src >> 4) & 0x0F];
coeff2l = ea_adpcm_table[((*src >> 4) & 0x0F) + 4];
coeff1r = ea_adpcm_table[*src & 0x0F];
coeff2r = ea_adpcm_table[(*src & 0x0F) + 4];
src++;
shift_left = ((*src >> 4) & 0x0F) + 8;
shift_right = (*src & 0x0F) + 8;
src++;
for (count2 = 0; count2 < 28; count2++) {
next_left_sample = (((*src & 0xF0) << 24) >> shift_left);
next_right_sample = (((*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;
CLAMP_TO_SHORT(next_left_sample);
CLAMP_TO_SHORT(next_right_sample);
previous_left_sample = current_left_sample;
current_left_sample = next_left_sample;
previous_right_sample = current_right_sample;
current_right_sample = next_right_sample;
*samples++ = (unsigned short)current_left_sample;
*samples++ = (unsigned short)current_right_sample;
}
}
break;
case CODEC_ID_ADPCM_IMA_SMJPEG:
c->status[0].predictor = *src;
src += 2;
c->status[0].step_index = *src++;
src++; /* skip another byte before getting to the meat */
while (src < buf + buf_size) {
*samples++ = adpcm_ima_expand_nibble(&c->status[0],
*src & 0x0F, 3);
*samples++ = adpcm_ima_expand_nibble(&c->status[0],
(*src >> 4) & 0x0F, 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) & 0x0F);
*samples++ = adpcm_ct_expand_nibble(&c->status[1],
src[0] & 0x0F);
} else {
*samples++ = adpcm_ct_expand_nibble(&c->status[0],
(src[0] >> 4) & 0x0F);
*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) & 0x0F, 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) & 0x07, 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) & 0x03, 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;
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);
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);
}
while (get_bits_count(&gb) < size)
{
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(c->status[i].predictor, -32768, 32767);
*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) & 0x0F);
} else {
*samples++ = adpcm_yamaha_expand_nibble(&c->status[0],
src[0] & 0x0F);
*samples++ = adpcm_yamaha_expand_nibble(&c->status[0],
(src[0] >> 4) & 0x0F);
}
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);
CLAMP_TO_SHORT(sampledat);
*samples = 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;
}
#ifdef CONFIG_ENCODERS
#define ADPCM_ENCODER(id,name) \
AVCodec name ## _encoder = { \
#name, \
CODEC_TYPE_AUDIO, \
id, \
sizeof(ADPCMContext), \
adpcm_encode_init, \
adpcm_encode_frame, \
adpcm_encode_close, \
NULL, \
};
#else
#define ADPCM_ENCODER(id,name)
#endif
#ifdef CONFIG_DECODERS
#define ADPCM_DECODER(id,name) \
AVCodec name ## _decoder = { \
#name, \
CODEC_TYPE_AUDIO, \
id, \
sizeof(ADPCMContext), \
adpcm_decode_init, \
NULL, \
NULL, \
adpcm_decode_frame, \
};
#else
#define ADPCM_DECODER(id,name)
#endif
#define ADPCM_CODEC(id, name) \
ADPCM_ENCODER(id,name) ADPCM_DECODER(id,name)
ADPCM_CODEC(CODEC_ID_ADPCM_IMA_QT, adpcm_ima_qt);
ADPCM_CODEC(CODEC_ID_ADPCM_IMA_WAV, adpcm_ima_wav);
ADPCM_CODEC(CODEC_ID_ADPCM_IMA_DK3, adpcm_ima_dk3);
ADPCM_CODEC(CODEC_ID_ADPCM_IMA_DK4, adpcm_ima_dk4);
ADPCM_CODEC(CODEC_ID_ADPCM_IMA_WS, adpcm_ima_ws);
ADPCM_CODEC(CODEC_ID_ADPCM_IMA_SMJPEG, adpcm_ima_smjpeg);
ADPCM_CODEC(CODEC_ID_ADPCM_MS, adpcm_ms);
ADPCM_CODEC(CODEC_ID_ADPCM_4XM, adpcm_4xm);
ADPCM_CODEC(CODEC_ID_ADPCM_XA, adpcm_xa);
ADPCM_CODEC(CODEC_ID_ADPCM_EA, adpcm_ea);
ADPCM_CODEC(CODEC_ID_ADPCM_CT, adpcm_ct);
ADPCM_CODEC(CODEC_ID_ADPCM_SWF, adpcm_swf);
ADPCM_CODEC(CODEC_ID_ADPCM_YAMAHA, adpcm_yamaha);
ADPCM_CODEC(CODEC_ID_ADPCM_SBPRO_4, adpcm_sbpro_4);
ADPCM_CODEC(CODEC_ID_ADPCM_SBPRO_3, adpcm_sbpro_3);
ADPCM_CODEC(CODEC_ID_ADPCM_SBPRO_2, adpcm_sbpro_2);
ADPCM_CODEC(CODEC_ID_ADPCM_THP, adpcm_thp);
#undef ADPCM_CODEC
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