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FFmpeg/libavcodec/adpcmenc.c
Michael Niedermayer ee77140afa Merge commit 'b2bed9325dbd6be0da1d91ffed3f513c40274fd2' 
* commit 'b2bed9325dbd6be0da1d91ffed3f513c40274fd2':
  cosmetics: Group .name and .long_name together in codec/format declarations

Conflicts:
	libavcodec/8svx.c
	libavcodec/alac.c
	libavcodec/cljr.c
	libavcodec/dnxhddec.c
	libavcodec/dnxhdenc.c
	libavcodec/dpxenc.c
	libavcodec/dvdec.c
	libavcodec/dvdsubdec.c
	libavcodec/dvdsubenc.c
	libavcodec/ffv1dec.c
	libavcodec/flacdec.c
	libavcodec/flvdec.c
	libavcodec/fraps.c
	libavcodec/frwu.c
	libavcodec/g726.c
	libavcodec/gif.c
	libavcodec/gifdec.c
	libavcodec/h261dec.c
	libavcodec/h263dec.c
	libavcodec/iff.c
	libavcodec/imc.c
	libavcodec/libopencore-amr.c
	libavcodec/libopenjpegdec.c
	libavcodec/libopenjpegenc.c
	libavcodec/libspeexenc.c
	libavcodec/libvo-amrwbenc.c
	libavcodec/libvorbisenc.c
	libavcodec/libvpxenc.c
	libavcodec/libx264.c
	libavcodec/libxavs.c
	libavcodec/libxvid.c
	libavcodec/ljpegenc.c
	libavcodec/mjpegbdec.c
	libavcodec/mjpegdec.c
	libavcodec/mpeg12dec.c
	libavcodec/mpeg4videodec.c
	libavcodec/msmpeg4dec.c
	libavcodec/pgssubdec.c
	libavcodec/pngdec.c
	libavcodec/pngenc.c
	libavcodec/proresdec_lgpl.c
	libavcodec/proresenc_kostya.c
	libavcodec/ra144enc.c
	libavcodec/rawdec.c
	libavcodec/rv10.c
	libavcodec/sp5xdec.c
	libavcodec/takdec.c
	libavcodec/tta.c
	libavcodec/v210dec.c
	libavcodec/vp6.c
	libavcodec/wavpack.c
	libavcodec/xbmenc.c
	libavcodec/yop.c

Merged-by: Michael Niedermayer <michaelni@gmx.at>
2013-10-04 12:34:23 +02:00

726 lines
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/*
* Copyright (c) 2001-2003 The ffmpeg Project
*
* first version by Francois Revol (revol@free.fr)
* fringe ADPCM codecs (e.g., DK3, DK4, Westwood)
* by Mike Melanson (melanson@pcisys.net)
*
* 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 "put_bits.h"
#include "bytestream.h"
#include "adpcm.h"
#include "adpcm_data.h"
#include "internal.h"
/**
* @file
* ADPCM encoders
* See ADPCM decoder reference documents for codec information.
*/
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_close(AVCodecContext *avctx);
static av_cold int adpcm_encode_init(AVCodecContext *avctx)
{
ADPCMEncodeContext *s = avctx->priv_data;
uint8_t *extradata;
int i;
int ret = AVERROR(ENOMEM);
if (avctx->channels > 2) {
av_log(avctx, AV_LOG_ERROR, "only stereo or mono is supported\n");
return AVERROR(EINVAL);
}
if (avctx->trellis && (unsigned)avctx->trellis > 16U) {
av_log(avctx, AV_LOG_ERROR, "invalid trellis size\n");
return AVERROR(EINVAL);
}
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);
}
avctx->bits_per_coded_sample = av_get_bits_per_sample(avctx->codec->id);
switch (avctx->codec->id) {
case AV_CODEC_ID_ADPCM_IMA_WAV:
/* each 16 bits sample gives one nibble
and we have 4 bytes per channel overhead */
avctx->frame_size = (BLKSIZE - 4 * avctx->channels) * 8 /
(4 * avctx->channels) + 1;
/* seems frame_size isn't taken into account...
have to buffer the samples :-( */
avctx->block_align = BLKSIZE;
avctx->bits_per_coded_sample = 4;
break;
case AV_CODEC_ID_ADPCM_IMA_QT:
avctx->frame_size = 64;
avctx->block_align = 34 * avctx->channels;
break;
case AV_CODEC_ID_ADPCM_MS:
/* each 16 bits sample gives one nibble
and we have 7 bytes per channel overhead */
avctx->frame_size = (BLKSIZE - 7 * avctx->channels) * 2 / avctx->channels + 2;
avctx->bits_per_coded_sample = 4;
avctx->block_align = BLKSIZE;
if (!(avctx->extradata = av_malloc(32 + FF_INPUT_BUFFER_PADDING_SIZE)))
goto error;
avctx->extradata_size = 32;
extradata = avctx->extradata;
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 AV_CODEC_ID_ADPCM_YAMAHA:
avctx->frame_size = BLKSIZE * 2 / avctx->channels;
avctx->block_align = BLKSIZE;
break;
case AV_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");
ret = AVERROR(EINVAL);
goto error;
}
avctx->frame_size = 512 * (avctx->sample_rate / 11025);
break;
default:
ret = AVERROR(EINVAL);
goto error;
}
return 0;
error:
adpcm_encode_close(avctx);
return ret;
}
static av_cold int adpcm_encode_close(AVCodecContext *avctx)
{
ADPCMEncodeContext *s = avctx->priv_data;
av_freep(&s->paths);
av_freep(&s->node_buf);
av_freep(&s->nodep_buf);
av_freep(&s->trellis_hash);
return 0;
}
static inline uint8_t adpcm_ima_compress_sample(ADPCMChannelStatus *c,
int16_t 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 uint8_t adpcm_ima_qt_compress_sample(ADPCMChannelStatus *c,
int16_t sample)
{
int delta = sample - c->prev_sample;
int diff, step = ff_adpcm_step_table[c->step_index];
int nibble = 8*(delta < 0);
delta= abs(delta);
diff = delta + (step >> 3);
if (delta >= step) {
nibble |= 4;
delta -= step;
}
step >>= 1;
if (delta >= step) {
nibble |= 2;
delta -= step;
}
step >>= 1;
if (delta >= step) {
nibble |= 1;
delta -= step;
}
diff -= delta;
if (nibble & 8)
c->prev_sample -= diff;
else
c->prev_sample += diff;
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 uint8_t adpcm_ms_compress_sample(ADPCMChannelStatus *c,
int16_t 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 += ((nibble & 0x08) ? (nibble - 0x10) : nibble) * c->idelta;
c->sample2 = c->sample1;
c->sample1 = av_clip_int16(predictor);
c->idelta = (ff_adpcm_AdaptationTable[nibble] * c->idelta) >> 8;
if (c->idelta < 16)
c->idelta = 16;
return nibble;
}
static inline uint8_t adpcm_yamaha_compress_sample(ADPCMChannelStatus *c,
int16_t 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 int16_t *samples, uint8_t *dst,
ADPCMChannelStatus *c, int n, int stride)
{
//FIXME 6% faster if frontier is a compile-time constant
ADPCMEncodeContext *s = avctx->priv_data;
const int frontier = 1 << avctx->trellis;
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 == AV_CODEC_ID_ADPCM_IMA_WAV ||
version == AV_CODEC_ID_ADPCM_IMA_QT ||
version == AV_CODEC_ID_ADPCM_SWF)
nodes[0]->sample1 = c->prev_sample;
if (version == AV_CODEC_ID_ADPCM_MS)
nodes[0]->step = c->idelta;
if (version == AV_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 == AV_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) {\
av_assert1(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 == AV_CODEC_ID_ADPCM_IMA_WAV ||
version == AV_CODEC_ID_ADPCM_IMA_QT ||
version == AV_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 { //AV_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, AVPacket *avpkt,
const AVFrame *frame, int *got_packet_ptr)
{
int n, i, ch, st, pkt_size, ret;
const int16_t *samples;
int16_t **samples_p;
uint8_t *dst;
ADPCMEncodeContext *c = avctx->priv_data;
uint8_t *buf;
samples = (const int16_t *)frame->data[0];
samples_p = (int16_t **)frame->extended_data;
st = avctx->channels == 2;
if (avctx->codec_id == AV_CODEC_ID_ADPCM_SWF)
pkt_size = (2 + avctx->channels * (22 + 4 * (frame->nb_samples - 1)) + 7) / 8;
else
pkt_size = avctx->block_align;
if ((ret = ff_alloc_packet2(avctx, avpkt, pkt_size)) < 0)
return ret;
dst = avpkt->data;
switch(avctx->codec->id) {
case AV_CODEC_ID_ADPCM_IMA_WAV:
{
int blocks, j;
blocks = (frame->nb_samples - 1) / 8;
for (ch = 0; ch < avctx->channels; ch++) {
ADPCMChannelStatus *status = &c->status[ch];
status->prev_sample = samples_p[ch][0];
/* status->step_index = 0;
XXX: not sure how to init the state machine */
bytestream_put_le16(&dst, status->prev_sample);
*dst++ = status->step_index;
*dst++ = 0; /* unknown */
}
/* stereo: 4 bytes (8 samples) for left, 4 bytes for right */
if (avctx->trellis > 0) {
FF_ALLOC_OR_GOTO(avctx, buf, avctx->channels * blocks * 8, error);
for (ch = 0; ch < avctx->channels; ch++) {
adpcm_compress_trellis(avctx, &samples_p[ch][1],
buf + ch * blocks * 8, &c->status[ch],
blocks * 8, 1);
}
for (i = 0; i < blocks; i++) {
for (ch = 0; ch < avctx->channels; ch++) {
uint8_t *buf1 = buf + ch * blocks * 8 + i * 8;
for (j = 0; j < 8; j += 2)
*dst++ = buf1[j] | (buf1[j + 1] << 4);
}
}
av_free(buf);
} else {
for (i = 0; i < blocks; i++) {
for (ch = 0; ch < avctx->channels; ch++) {
ADPCMChannelStatus *status = &c->status[ch];
const int16_t *smp = &samples_p[ch][1 + i * 8];
for (j = 0; j < 8; j += 2) {
uint8_t v = adpcm_ima_compress_sample(status, smp[j ]);
v |= adpcm_ima_compress_sample(status, smp[j + 1]) << 4;
*dst++ = v;
}
}
}
}
break;
}
case AV_CODEC_ID_ADPCM_IMA_QT:
{
PutBitContext pb;
init_put_bits(&pb, dst, pkt_size * 8);
for (ch = 0; ch < avctx->channels; ch++) {
ADPCMChannelStatus *status = &c->status[ch];
put_bits(&pb, 9, (status->prev_sample & 0xFFFF) >> 7);
put_bits(&pb, 7, status->step_index);
if (avctx->trellis > 0) {
uint8_t buf[64];
adpcm_compress_trellis(avctx, &samples_p[ch][1], buf, status,
64, 1);
for (i = 0; i < 64; i++)
put_bits(&pb, 4, buf[i ^ 1]);
} else {
for (i = 0; i < 64; i += 2) {
int t1, t2;
t1 = adpcm_ima_qt_compress_sample(status, samples_p[ch][i ]);
t2 = adpcm_ima_qt_compress_sample(status, samples_p[ch][i + 1]);
put_bits(&pb, 4, t2);
put_bits(&pb, 4, t1);
}
}
}
flush_put_bits(&pb);
break;
}
case AV_CODEC_ID_ADPCM_SWF:
{
PutBitContext pb;
init_put_bits(&pb, dst, pkt_size * 8);
n = frame->nb_samples - 1;
// store AdpcmCodeSize
put_bits(&pb, 2, 2); // set 4-bit flash adpcm format
// init the encoder state
for (i = 0; i < avctx->channels; i++) {
// clip step so it fits 6 bits
c->status[i].step_index = av_clip(c->status[i].step_index, 0, 63);
put_sbits(&pb, 16, samples[i]);
put_bits(&pb, 6, c->status[i].step_index);
c->status[i].prev_sample = samples[i];
}
if (avctx->trellis > 0) {
FF_ALLOC_OR_GOTO(avctx, buf, 2 * n, error);
adpcm_compress_trellis(avctx, samples + avctx->channels, buf,
&c->status[0], n, avctx->channels);
if (avctx->channels == 2)
adpcm_compress_trellis(avctx, samples + avctx->channels + 1,
buf + n, &c->status[1], n,
avctx->channels);
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 < frame->nb_samples; 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);
break;
}
case AV_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) {
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,
avctx->channels);
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, avctx->channels);
adpcm_compress_trellis(avctx, samples + 1, buf + n,
&c->status[1], n, avctx->channels);
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 AV_CODEC_ID_ADPCM_YAMAHA:
n = frame->nb_samples / 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,
avctx->channels);
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, avctx->channels);
adpcm_compress_trellis(avctx, samples + 1, buf + n,
&c->status[1], n, avctx->channels);
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:
return AVERROR(EINVAL);
}
avpkt->size = pkt_size;
*got_packet_ptr = 1;
return 0;
error:
return AVERROR(ENOMEM);
}
static const enum AVSampleFormat sample_fmts[] = {
AV_SAMPLE_FMT_S16, AV_SAMPLE_FMT_NONE
};
static const enum AVSampleFormat sample_fmts_p[] = {
AV_SAMPLE_FMT_S16P, AV_SAMPLE_FMT_NONE
};
#define ADPCM_ENCODER(id_, name_, sample_fmts_, long_name_) \
AVCodec ff_ ## name_ ## _encoder = { \
.name = #name_, \
.long_name = NULL_IF_CONFIG_SMALL(long_name_), \
.type = AVMEDIA_TYPE_AUDIO, \
.id = id_, \
.priv_data_size = sizeof(ADPCMEncodeContext), \
.init = adpcm_encode_init, \
.encode2 = adpcm_encode_frame, \
.close = adpcm_encode_close, \
.sample_fmts = sample_fmts_, \
}
ADPCM_ENCODER(AV_CODEC_ID_ADPCM_IMA_QT, adpcm_ima_qt, sample_fmts_p, "ADPCM IMA QuickTime");
ADPCM_ENCODER(AV_CODEC_ID_ADPCM_IMA_WAV, adpcm_ima_wav, sample_fmts_p, "ADPCM IMA WAV");
ADPCM_ENCODER(AV_CODEC_ID_ADPCM_MS, adpcm_ms, sample_fmts, "ADPCM Microsoft");
ADPCM_ENCODER(AV_CODEC_ID_ADPCM_SWF, adpcm_swf, sample_fmts, "ADPCM Shockwave Flash");
ADPCM_ENCODER(AV_CODEC_ID_ADPCM_YAMAHA, adpcm_yamaha, sample_fmts, "ADPCM Yamaha");
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