generic-library/ffmpeg
1
0
Fork 0
Code Issues Pull Requests Releases Wiki Activity
fa178ed2a4
ffmpeg/libavcodec/adpcm.c
Vladimir Voroshilov 94d9633a32 Replace "signed short" typecast with "int16_t" as suggested by 
Mans Rullgard. Value at the right side is 16 bit length signed.
We can not know for sure that short is in fact 16 bits, even if
this virtually always is the case.

Originally committed as revision 10666 to svn://svn.ffmpeg.org/ffmpeg/trunk
2007-10-04 15:13:42 +00:00

1469 lines
52 KiB
C
Raw Blame History

/*
* 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
/* 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 += ((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))) / 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;
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;
}
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) || (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[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;\
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 < 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)|| (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;
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);
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_bits(&pb, 16, samples[i] & 0xFFFF);
put_bits(&pb, 6, c->status[i].step_index);
c->status[i].prev_sample = (signed short)samples[i];
}
if(avctx->trellis > 0) {
uint8_t buf[2][n];
adpcm_compress_trellis(avctx, samples+2, buf[0], &c->status[0], n);
if (avctx->channels == 2)
adpcm_compress_trellis(avctx, samples+3, buf[1], &c->status[1], n);
for(i=0; i<n; i++) {
put_bits(&pb, 4, buf[0][i]);
if (avctx->channels == 2)
put_bits(&pb, 4, buf[1][i]);
}
} else {
for (i=1; 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;
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))) / 256;
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 = (ct_adpcm_table[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) & 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;
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];
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;
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;
}
}
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) & 0x0F;
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) & 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, 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) & 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);
*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;
}
#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_4XM, adpcm_4xm);
ADPCM_CODEC(CODEC_ID_ADPCM_CT, adpcm_ct);
ADPCM_CODEC(CODEC_ID_ADPCM_EA, adpcm_ea);
ADPCM_CODEC(CODEC_ID_ADPCM_IMA_AMV, adpcm_ima_amv);
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_QT, adpcm_ima_qt);
ADPCM_CODEC(CODEC_ID_ADPCM_IMA_SMJPEG, adpcm_ima_smjpeg);
ADPCM_CODEC(CODEC_ID_ADPCM_IMA_WAV, adpcm_ima_wav);
ADPCM_CODEC(CODEC_ID_ADPCM_IMA_WS, adpcm_ima_ws);
ADPCM_CODEC(CODEC_ID_ADPCM_MS, adpcm_ms);
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_SWF, adpcm_swf);
ADPCM_CODEC(CODEC_ID_ADPCM_THP, adpcm_thp);
ADPCM_CODEC(CODEC_ID_ADPCM_XA, adpcm_xa);
ADPCM_CODEC(CODEC_ID_ADPCM_YAMAHA, adpcm_yamaha);
#undef ADPCM_CODEC
Reference in New Issue View Git Blame Copy Permalink