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adpcm: split ADPCM encoders and decoders into separate files.

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Move shared tables to a separate file as well.
This commit is contained in:
Justin Ruggles 2011-09-07 18:34:09 -04:00
parent 57650c70e2
commit 826c56d16e
6 changed files with 861 additions and 729 deletions
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40
libavcodec/Makefile
View File

@ -483,10 +483,10 @@ OBJS-$(CONFIG_PCM_U32LE_ENCODER) += pcm.o
OBJS-$(CONFIG_PCM_ZORK_DECODER) += pcm.o OBJS-$(CONFIG_PCM_ZORK_DECODER) += pcm.o
OBJS-$(CONFIG_PCM_ZORK_ENCODER) += pcm.o OBJS-$(CONFIG_PCM_ZORK_ENCODER) += pcm.o
OBJS-$(CONFIG_ADPCM_4XM_DECODER) += adpcm.o OBJS-$(CONFIG_ADPCM_4XM_DECODER) += adpcm.o adpcm_data.o
OBJS-$(CONFIG_ADPCM_ADX_DECODER) += adxdec.o OBJS-$(CONFIG_ADPCM_ADX_DECODER) += adxdec.o
OBJS-$(CONFIG_ADPCM_ADX_ENCODER) += adxenc.o OBJS-$(CONFIG_ADPCM_ADX_ENCODER) += adxenc.o
OBJS-$(CONFIG_ADPCM_CT_DECODER) += adpcm.o OBJS-$(CONFIG_ADPCM_CT_DECODER) += adpcm.o adpcm_data.o
OBJS-$(CONFIG_ADPCM_EA_DECODER) += adpcm.o OBJS-$(CONFIG_ADPCM_EA_DECODER) += adpcm.o
OBJS-$(CONFIG_ADPCM_EA_MAXIS_XA_DECODER) += adpcm.o OBJS-$(CONFIG_ADPCM_EA_MAXIS_XA_DECODER) += adpcm.o
OBJS-$(CONFIG_ADPCM_EA_R1_DECODER) += adpcm.o OBJS-$(CONFIG_ADPCM_EA_R1_DECODER) += adpcm.o
@ -497,29 +497,29 @@ OBJS-$(CONFIG_ADPCM_G722_DECODER) += g722.o
OBJS-$(CONFIG_ADPCM_G722_ENCODER) += g722.o OBJS-$(CONFIG_ADPCM_G722_ENCODER) += g722.o
OBJS-$(CONFIG_ADPCM_G726_DECODER) += g726.o OBJS-$(CONFIG_ADPCM_G726_DECODER) += g726.o
OBJS-$(CONFIG_ADPCM_G726_ENCODER) += g726.o OBJS-$(CONFIG_ADPCM_G726_ENCODER) += g726.o
OBJS-$(CONFIG_ADPCM_IMA_AMV_DECODER) += adpcm.o OBJS-$(CONFIG_ADPCM_IMA_AMV_DECODER) += adpcm.o adpcm_data.o
OBJS-$(CONFIG_ADPCM_IMA_DK3_DECODER) += adpcm.o OBJS-$(CONFIG_ADPCM_IMA_DK3_DECODER) += adpcm.o adpcm_data.o
OBJS-$(CONFIG_ADPCM_IMA_DK4_DECODER) += adpcm.o OBJS-$(CONFIG_ADPCM_IMA_DK4_DECODER) += adpcm.o adpcm_data.o
OBJS-$(CONFIG_ADPCM_IMA_EA_EACS_DECODER) += adpcm.o OBJS-$(CONFIG_ADPCM_IMA_EA_EACS_DECODER) += adpcm.o adpcm_data.o
OBJS-$(CONFIG_ADPCM_IMA_EA_SEAD_DECODER) += adpcm.o OBJS-$(CONFIG_ADPCM_IMA_EA_SEAD_DECODER) += adpcm.o adpcm_data.o
OBJS-$(CONFIG_ADPCM_IMA_ISS_DECODER) += adpcm.o OBJS-$(CONFIG_ADPCM_IMA_ISS_DECODER) += adpcm.o adpcm_data.o
OBJS-$(CONFIG_ADPCM_IMA_QT_DECODER) += adpcm.o OBJS-$(CONFIG_ADPCM_IMA_QT_DECODER) += adpcm.o adpcm_data.o
OBJS-$(CONFIG_ADPCM_IMA_QT_ENCODER) += adpcm.o OBJS-$(CONFIG_ADPCM_IMA_QT_ENCODER) += adpcmenc.o adpcm_data.o
OBJS-$(CONFIG_ADPCM_IMA_SMJPEG_DECODER) += adpcm.o OBJS-$(CONFIG_ADPCM_IMA_SMJPEG_DECODER) += adpcm.o adpcm_data.o
OBJS-$(CONFIG_ADPCM_IMA_WAV_DECODER) += adpcm.o OBJS-$(CONFIG_ADPCM_IMA_WAV_DECODER) += adpcm.o adpcm_data.o
OBJS-$(CONFIG_ADPCM_IMA_WAV_ENCODER) += adpcm.o OBJS-$(CONFIG_ADPCM_IMA_WAV_ENCODER) += adpcmenc.o adpcm_data.o
OBJS-$(CONFIG_ADPCM_IMA_WS_DECODER) += adpcm.o OBJS-$(CONFIG_ADPCM_IMA_WS_DECODER) += adpcm.o adpcm_data.o
OBJS-$(CONFIG_ADPCM_MS_DECODER) += adpcm.o OBJS-$(CONFIG_ADPCM_MS_DECODER) += adpcm.o adpcm_data.o
OBJS-$(CONFIG_ADPCM_MS_ENCODER) += adpcm.o OBJS-$(CONFIG_ADPCM_MS_ENCODER) += adpcmenc.o adpcm_data.o
OBJS-$(CONFIG_ADPCM_SBPRO_2_DECODER) += adpcm.o OBJS-$(CONFIG_ADPCM_SBPRO_2_DECODER) += adpcm.o
OBJS-$(CONFIG_ADPCM_SBPRO_3_DECODER) += adpcm.o OBJS-$(CONFIG_ADPCM_SBPRO_3_DECODER) += adpcm.o
OBJS-$(CONFIG_ADPCM_SBPRO_4_DECODER) += adpcm.o OBJS-$(CONFIG_ADPCM_SBPRO_4_DECODER) += adpcm.o
OBJS-$(CONFIG_ADPCM_SWF_DECODER) += adpcm.o OBJS-$(CONFIG_ADPCM_SWF_DECODER) += adpcm.o adpcm_data.o
OBJS-$(CONFIG_ADPCM_SWF_ENCODER) += adpcm.o OBJS-$(CONFIG_ADPCM_SWF_ENCODER) += adpcmenc.o adpcm_data.o
OBJS-$(CONFIG_ADPCM_THP_DECODER) += adpcm.o OBJS-$(CONFIG_ADPCM_THP_DECODER) += adpcm.o
OBJS-$(CONFIG_ADPCM_XA_DECODER) += adpcm.o OBJS-$(CONFIG_ADPCM_XA_DECODER) += adpcm.o
OBJS-$(CONFIG_ADPCM_YAMAHA_DECODER) += adpcm.o OBJS-$(CONFIG_ADPCM_YAMAHA_DECODER) += adpcm.o adpcm_data.o
OBJS-$(CONFIG_ADPCM_YAMAHA_ENCODER) += adpcm.o OBJS-$(CONFIG_ADPCM_YAMAHA_ENCODER) += adpcmenc.o adpcm_data.o
# libavformat dependencies # libavformat dependencies
OBJS-$(CONFIG_ADTS_MUXER) += mpeg4audio.o OBJS-$(CONFIG_ADTS_MUXER) += mpeg4audio.o

734
libavcodec/adpcm.c
View File

@ -1,5 +1,4 @@
/* /*
* ADPCM codecs
* Copyright (c) 2001-2003 The ffmpeg Project * Copyright (c) 2001-2003 The ffmpeg Project
* *
* This file is part of Libav. * This file is part of Libav.
@ -22,10 +21,12 @@
#include "get_bits.h" #include "get_bits.h"
#include "put_bits.h" #include "put_bits.h"
#include "bytestream.h" #include "bytestream.h"
#include "adpcm.h"
#include "adpcm_data.h"
/** /**
* @file * @file
* ADPCM codecs. * ADPCM decoders
* First version by Francois Revol (revol@free.fr) * First version by Francois Revol (revol@free.fr)
* Fringe ADPCM codecs (e.g., DK3, DK4, Westwood) * Fringe ADPCM codecs (e.g., DK3, DK4, Westwood)
* by Mike Melanson (melanson@pcisys.net) * by Mike Melanson (melanson@pcisys.net)
@ -54,48 +55,6 @@
* readstr http://www.geocities.co.jp/Playtown/2004/ * 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 */ /* These are for CD-ROM XA ADPCM */
static const int xa_adpcm_table[5][2] = { static const int xa_adpcm_table[5][2] = {
{ 0, 0 }, { 0, 0 },
@ -118,632 +77,15 @@ static const int swf_index_tables[4][16] = {
/*5*/ { -1, -1, -1, -1, -1, -1, -1, -1, 1, 2, 4, 6, 8, 10, 13, 16 } /*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 */ /* end of tables */
typedef struct ADPCMChannelStatus { typedef struct ADPCMDecodeContext {
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]; ADPCMChannelStatus status[6];
TrellisPath *paths; } ADPCMDecodeContext;
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) static av_cold int adpcm_decode_init(AVCodecContext * avctx)
{ {
ADPCMContext *c = avctx->priv_data; ADPCMDecodeContext *c = avctx->priv_data;
unsigned int max_channels = 2; unsigned int max_channels = 2;
switch(avctx->codec->id) { switch(avctx->codec->id) {
@ -786,8 +128,8 @@ static inline short adpcm_ima_expand_nibble(ADPCMChannelStatus *c, char nibble,
int predictor; int predictor;
int sign, delta, diff, step; int sign, delta, diff, step;
step = step_table[c->step_index]; step = ff_adpcm_step_table[c->step_index];
step_index = c->step_index + index_table[(unsigned)nibble]; step_index = c->step_index + ff_adpcm_index_table[(unsigned)nibble];
if (step_index < 0) step_index = 0; if (step_index < 0) step_index = 0;
else if (step_index > 88) step_index = 88; else if (step_index > 88) step_index = 88;
@ -816,7 +158,7 @@ static inline short adpcm_ms_expand_nibble(ADPCMChannelStatus *c, char nibble)
c->sample2 = c->sample1; c->sample2 = c->sample1;
c->sample1 = av_clip_int16(predictor); c->sample1 = av_clip_int16(predictor);
c->idelta = (AdaptationTable[(int)nibble] * c->idelta) >> 8; c->idelta = (ff_adpcm_AdaptationTable[(int)nibble] * c->idelta) >> 8;
if (c->idelta < 16) c->idelta = 16; if (c->idelta < 16) c->idelta = 16;
return c->sample1; return c->sample1;
@ -837,7 +179,7 @@ static inline short adpcm_ct_expand_nibble(ADPCMChannelStatus *c, char nibble)
c->predictor = ((c->predictor * 254) >> 8) + (sign ? -diff : diff); c->predictor = ((c->predictor * 254) >> 8) + (sign ? -diff : diff);
c->predictor = av_clip_int16(c->predictor); c->predictor = av_clip_int16(c->predictor);
/* calculate new step and clamp it to range 511..32767 */ /* calculate new step and clamp it to range 511..32767 */
new_step = (AdaptationTable[nibble & 7] * c->step) >> 8; new_step = (ff_adpcm_AdaptationTable[nibble & 7] * c->step) >> 8;
c->step = av_clip(new_step, 511, 32767); c->step = av_clip(new_step, 511, 32767);
return (short)c->predictor; return (short)c->predictor;
@ -870,9 +212,9 @@ static inline short adpcm_yamaha_expand_nibble(ADPCMChannelStatus *c, unsigned c
c->step = 127; c->step = 127;
} }
c->predictor += (c->step * yamaha_difflookup[nibble]) / 8; c->predictor += (c->step * ff_adpcm_yamaha_difflookup[nibble]) / 8;
c->predictor = av_clip_int16(c->predictor); c->predictor = av_clip_int16(c->predictor);
c->step = (c->step * yamaha_indexscale[nibble]) >> 8; c->step = (c->step * ff_adpcm_yamaha_indexscale[nibble]) >> 8;
c->step = av_clip(c->step, 127, 24567); c->step = av_clip(c->step, 127, 24567);
return c->predictor; return c->predictor;
} }
@ -964,7 +306,7 @@ static int adpcm_decode_frame(AVCodecContext *avctx,
{ {
const uint8_t *buf = avpkt->data; const uint8_t *buf = avpkt->data;
int buf_size = avpkt->size; int buf_size = avpkt->size;
ADPCMContext *c = avctx->priv_data; ADPCMDecodeContext *c = avctx->priv_data;
ADPCMChannelStatus *cs; ADPCMChannelStatus *cs;
int n, m, channel, i; int n, m, channel, i;
int block_predictor[2]; int block_predictor[2];
@ -1030,7 +372,7 @@ static int adpcm_decode_frame(AVCodecContext *avctx,
cs->step_index = 88; cs->step_index = 88;
} }
cs->step = step_table[cs->step_index]; cs->step = ff_adpcm_step_table[cs->step_index];
samples = (short*)data + channel; samples = (short*)data + channel;
@ -1114,10 +456,10 @@ static int adpcm_decode_frame(AVCodecContext *avctx,
if (st){ if (st){
c->status[1].idelta = (int16_t)bytestream_get_le16(&src); c->status[1].idelta = (int16_t)bytestream_get_le16(&src);
} }
c->status[0].coeff1 = AdaptCoeff1[block_predictor[0]]; c->status[0].coeff1 = ff_adpcm_AdaptCoeff1[block_predictor[0]];
c->status[0].coeff2 = AdaptCoeff2[block_predictor[0]]; c->status[0].coeff2 = ff_adpcm_AdaptCoeff2[block_predictor[0]];
c->status[1].coeff1 = AdaptCoeff1[block_predictor[1]]; c->status[1].coeff1 = ff_adpcm_AdaptCoeff1[block_predictor[1]];
c->status[1].coeff2 = AdaptCoeff2[block_predictor[1]]; c->status[1].coeff2 = ff_adpcm_AdaptCoeff2[block_predictor[1]];
c->status[0].sample1 = bytestream_get_le16(&src); c->status[0].sample1 = bytestream_get_le16(&src);
if (st) c->status[1].sample1 = bytestream_get_le16(&src); if (st) c->status[1].sample1 = bytestream_get_le16(&src);
@ -1586,7 +928,7 @@ static int adpcm_decode_frame(AVCodecContext *avctx,
for (i = 0; i < avctx->channels; i++) { for (i = 0; i < avctx->channels; i++) {
// similar to IMA adpcm // similar to IMA adpcm
int delta = get_bits(&gb, nb_bits); int delta = get_bits(&gb, nb_bits);
int step = step_table[c->status[i].step_index]; int step = ff_adpcm_step_table[c->status[i].step_index];
long vpdiff = 0; // vpdiff = (delta+0.5)*step/4 long vpdiff = 0; // vpdiff = (delta+0.5)*step/4
int k = k0; int k = k0;
@ -1705,44 +1047,18 @@ static int adpcm_decode_frame(AVCodecContext *avctx,
} }
#if CONFIG_ENCODERS
#define ADPCM_ENCODER(id,name,long_name_) \
AVCodec ff_ ## 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_) \ #define ADPCM_DECODER(id,name,long_name_) \
AVCodec ff_ ## name ## _decoder = { \ AVCodec ff_ ## name ## _decoder = { \
#name, \ #name, \
AVMEDIA_TYPE_AUDIO, \ AVMEDIA_TYPE_AUDIO, \
id, \ id, \
sizeof(ADPCMContext), \ sizeof(ADPCMDecodeContext), \
adpcm_decode_init, \ adpcm_decode_init, \
NULL, \ NULL, \
NULL, \ NULL, \
adpcm_decode_frame, \ adpcm_decode_frame, \
.long_name = NULL_IF_CONFIG_SMALL(long_name_), \ .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. */ /* 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_4XM, adpcm_4xm, "ADPCM 4X Movie");
@ -1759,15 +1075,15 @@ 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_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_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_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_QT, adpcm_ima_qt, "ADPCM IMA QuickTime");
ADPCM_DECODER(CODEC_ID_ADPCM_IMA_SMJPEG, adpcm_ima_smjpeg, "ADPCM IMA Loki SDL MJPEG"); 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_WAV, adpcm_ima_wav, "ADPCM IMA WAV");
ADPCM_DECODER(CODEC_ID_ADPCM_IMA_WS, adpcm_ima_ws, "ADPCM IMA Westwood"); 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_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_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_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_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_SWF, adpcm_swf, "ADPCM Shockwave Flash");
ADPCM_DECODER(CODEC_ID_ADPCM_THP, adpcm_thp, "ADPCM Nintendo Gamecube THP"); ADPCM_DECODER(CODEC_ID_ADPCM_THP, adpcm_thp, "ADPCM Nintendo Gamecube THP");
ADPCM_DECODER(CODEC_ID_ADPCM_XA, adpcm_xa, "ADPCM CDROM XA"); ADPCM_DECODER(CODEC_ID_ADPCM_XA, adpcm_xa, "ADPCM CDROM XA");
ADPCM_CODEC (CODEC_ID_ADPCM_YAMAHA, adpcm_yamaha, "ADPCM Yamaha"); ADPCM_DECODER(CODEC_ID_ADPCM_YAMAHA, adpcm_yamaha, "ADPCM Yamaha");

46
libavcodec/adpcm.h Normal file
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@ -0,0 +1,46 @@
/*
* Copyright (c) 2001-2003 The ffmpeg Project
*
* This file is part of Libav.
*
* Libav 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.
*
* Libav 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 Libav; if not, write to the Free Software
* Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
*/
/**
* @file
* ADPCM encoder/decoder common header.
*/
#ifndef AVCODEC_ADPCM_H
#define AVCODEC_ADPCM_H
#define BLKSIZE 1024
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;
#endif /* AVCODEC_ADPCM_H */

78
libavcodec/adpcm_data.c Normal file
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@ -0,0 +1,78 @@
/*
* Copyright (c) 2001-2003 The ffmpeg Project
*
* This file is part of Libav.
*
* Libav 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.
*
* Libav 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 Libav; if not, write to the Free Software
* Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
*/
/**
* @file
* ADPCM tables
*/
#include <stdint.h>
/* ff_adpcm_step_table[] and ff_adpcm_index_table[] are from the ADPCM
reference source */
/* This is the index table: */
const int8_t ff_adpcm_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:
*/
const int16_t ff_adpcm_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 */
/* ff_adpcm_AdaptationTable[], ff_adpcm_AdaptCoeff1[], and
ff_adpcm_AdaptCoeff2[] are from libsndfile */
const int16_t ff_adpcm_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 */
const uint8_t ff_adpcm_AdaptCoeff1[] = {
64, 128, 0, 48, 60, 115, 98
};
/** Divided by 4 to fit in 8-bit integers */
const int8_t ff_adpcm_AdaptCoeff2[] = {
0, -64, 0, 16, 0, -52, -58
};
const int16_t ff_adpcm_yamaha_indexscale[] = {
230, 230, 230, 230, 307, 409, 512, 614,
230, 230, 230, 230, 307, 409, 512, 614
};
const int8_t ff_adpcm_yamaha_difflookup[] = {
1, 3, 5, 7, 9, 11, 13, 15,
-1, -3, -5, -7, -9, -11, -13, -15
};

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/*
* Copyright (c) 2001-2003 The ffmpeg Project
*
* This file is part of Libav.
*
* Libav 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.
*
* Libav 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 Libav; if not, write to the Free Software
* Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
*/
/**
* @file
* ADPCM tables
*/
#ifndef AVCODEC_ADPCM_DATA_H
#define AVCODEC_ADPCM_DATA_H
extern const int8_t ff_adpcm_index_table[16];
extern const int16_t ff_adpcm_step_table[89];
extern const int16_t ff_adpcm_AdaptationTable[];
extern const uint8_t ff_adpcm_AdaptCoeff1[];
extern const int8_t ff_adpcm_AdaptCoeff2[];
extern const int16_t ff_adpcm_yamaha_indexscale[];
extern const int8_t ff_adpcm_yamaha_difflookup[];
#endif /* AVCODEC_ADPCM_DATA_H */

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/*
* Copyright (c) 2001-2003 The ffmpeg Project
*
* This file is part of Libav.
*
* Libav 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.
*
* Libav 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 Libav; 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"
#include "adpcm.h"
#include "adpcm_data.h"
/**
* @file
* ADPCM encoders
* First version by Francois Revol (revol@free.fr)
* Fringe ADPCM codecs (e.g., DK3, DK4, Westwood)
* by Mike Melanson (melanson@pcisys.net)
*
* 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
*/
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 ADPCMEncodeContext {
ADPCMChannelStatus status[6];
TrellisPath *paths;
TrellisNode *node_buf;
TrellisNode **nodep_buf;
uint8_t *trellis_hash;
} ADPCMEncodeContext;
#define FREEZE_INTERVAL 128
static av_cold int adpcm_encode_init(AVCodecContext *avctx)
{
ADPCMEncodeContext *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, ff_adpcm_AdaptCoeff1[i] * 4);
bytestream_put_le16(&extradata, ff_adpcm_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)
{
ADPCMEncodeContext *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/ff_adpcm_step_table[c->step_index]) + (delta<0)*8;
c->prev_sample += ((ff_adpcm_step_table[c->step_index] * ff_adpcm_yamaha_difflookup[nibble]) / 8);
c->prev_sample = av_clip_int16(c->prev_sample);
c->step_index = av_clip(c->step_index + ff_adpcm_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 = (ff_adpcm_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 * ff_adpcm_yamaha_difflookup[nibble]) / 8);
c->predictor = av_clip_int16(c->predictor);
c->step = (c->step * ff_adpcm_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
ADPCMEncodeContext *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, (ff_adpcm_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 * ff_adpcm_yamaha_difflookup[nibble]) / 8;\
STORE_NODE(NAME, STEP_INDEX);\
}
LOOP_NODES(ima, ff_adpcm_step_table[step], av_clip(step + ff_adpcm_index_table[nibble], 0, 88));
} else { //CODEC_ID_ADPCM_YAMAHA
LOOP_NODES(yamaha, step, av_clip((step * ff_adpcm_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;
ADPCMEncodeContext *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 = ff_adpcm_AdaptCoeff1[predictor];
c->status[i].coeff2 = ff_adpcm_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;
}
#define ADPCM_ENCODER(id,name,long_name_) \
AVCodec ff_ ## name ## _encoder = { \
#name, \
AVMEDIA_TYPE_AUDIO, \
id, \
sizeof(ADPCMEncodeContext), \
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_), \
}
ADPCM_ENCODER(CODEC_ID_ADPCM_IMA_QT, adpcm_ima_qt, "ADPCM IMA QuickTime");
ADPCM_ENCODER(CODEC_ID_ADPCM_IMA_WAV, adpcm_ima_wav, "ADPCM IMA WAV");
ADPCM_ENCODER(CODEC_ID_ADPCM_MS, adpcm_ms, "ADPCM Microsoft");
ADPCM_ENCODER(CODEC_ID_ADPCM_SWF, adpcm_swf, "ADPCM Shockwave Flash");
ADPCM_ENCODER(CODEC_ID_ADPCM_YAMAHA, adpcm_yamaha, "ADPCM Yamaha");