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FFmpeg/libavcodec/adpcmenc.c
Andreas Rheinhardt 4243da4ff4 avcodec/codec_internal: Use union for FFCodec decode/encode callbacks 
This is possible, because every given FFCodec has to implement
exactly one of these. Doing so decreases sizeof(FFCodec) and
therefore decreases the size of the binary.
Notice that in case of position-independent code the decrease
is in .data.rel.ro, so that this translates to decreased
memory consumption.

Signed-off-by: Andreas Rheinhardt <andreas.rheinhardt@outlook.com>
2022-04-05 20:02:37 +02:00

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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 "config_components.h"
#include "libavutil/opt.h"
#include "avcodec.h"
#include "put_bits.h"
#include "bytestream.h"
#include "adpcm.h"
#include "adpcm_data.h"
#include "codec_internal.h"
#include "encode.h"
/**
* @file
* ADPCM encoders
* See ADPCM decoder reference documents for codec information.
*/
#define CASE_0(codec_id, ...)
#define CASE_1(codec_id, ...) \
case codec_id: \
{ __VA_ARGS__ } \
break;
#define CASE_2(enabled, codec_id, ...) \
CASE_ ## enabled(codec_id, __VA_ARGS__)
#define CASE_3(config, codec_id, ...) \
CASE_2(config, codec_id, __VA_ARGS__)
#define CASE(codec, ...) \
CASE_3(CONFIG_ ## codec ## _ENCODER, AV_CODEC_ID_ ## codec, __VA_ARGS__)
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 {
AVClass *class;
int block_size;
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;
int channels = avctx->ch_layout.nb_channels;
/*
* AMV's block size has to match that of the corresponding video
* stream. Relax the POT requirement.
*/
if (avctx->codec->id != AV_CODEC_ID_ADPCM_IMA_AMV &&
(s->block_size & (s->block_size - 1))) {
av_log(avctx, AV_LOG_ERROR, "block size must be power of 2\n");
return AVERROR(EINVAL);
}
if (avctx->trellis) {
int frontier, max_paths;
if ((unsigned)avctx->trellis > 16U) {
av_log(avctx, AV_LOG_ERROR, "invalid trellis size\n");
return AVERROR(EINVAL);
}
if (avctx->codec->id == AV_CODEC_ID_ADPCM_IMA_SSI ||
avctx->codec->id == AV_CODEC_ID_ADPCM_IMA_APM ||
avctx->codec->id == AV_CODEC_ID_ADPCM_ARGO ||
avctx->codec->id == AV_CODEC_ID_ADPCM_IMA_WS) {
/*
* The current trellis implementation doesn't work for extended
* runs of samples without periodic resets. Disallow it.
*/
av_log(avctx, AV_LOG_ERROR, "trellis not supported\n");
return AVERROR_PATCHWELCOME;
}
frontier = 1 << avctx->trellis;
max_paths = frontier * FREEZE_INTERVAL;
if (!FF_ALLOC_TYPED_ARRAY(s->paths, max_paths) ||
!FF_ALLOC_TYPED_ARRAY(s->node_buf, 2 * frontier) ||
!FF_ALLOC_TYPED_ARRAY(s->nodep_buf, 2 * frontier) ||
!FF_ALLOC_TYPED_ARRAY(s->trellis_hash, 65536))
return AVERROR(ENOMEM);
}
avctx->bits_per_coded_sample = av_get_bits_per_sample(avctx->codec->id);
switch (avctx->codec->id) {
CASE(ADPCM_IMA_WAV,
/* each 16 bits sample gives one nibble
and we have 4 bytes per channel overhead */
avctx->frame_size = (s->block_size - 4 * channels) * 8 /
(4 * channels) + 1;
/* seems frame_size isn't taken into account...
have to buffer the samples :-( */
avctx->block_align = s->block_size;
avctx->bits_per_coded_sample = 4;
) /* End of CASE */
CASE(ADPCM_IMA_QT,
avctx->frame_size = 64;
avctx->block_align = 34 * channels;
) /* End of CASE */
CASE(ADPCM_MS,
uint8_t *extradata;
/* each 16 bits sample gives one nibble
and we have 7 bytes per channel overhead */
avctx->frame_size = (s->block_size - 7 * channels) * 2 / channels + 2;
avctx->bits_per_coded_sample = 4;
avctx->block_align = s->block_size;
if (!(avctx->extradata = av_malloc(32 + AV_INPUT_BUFFER_PADDING_SIZE)))
return AVERROR(ENOMEM);
avctx->extradata_size = 32;
extradata = avctx->extradata;
bytestream_put_le16(&extradata, avctx->frame_size);
bytestream_put_le16(&extradata, 7); /* wNumCoef */
for (int i = 0; i < 7; i++) {
bytestream_put_le16(&extradata, ff_adpcm_AdaptCoeff1[i] * 4);
bytestream_put_le16(&extradata, ff_adpcm_AdaptCoeff2[i] * 4);
}
) /* End of CASE */
CASE(ADPCM_YAMAHA,
avctx->frame_size = s->block_size * 2 / channels;
avctx->block_align = s->block_size;
) /* End of CASE */
CASE(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 AVERROR(EINVAL);
}
avctx->frame_size = 4096; /* Hardcoded according to the SWF spec. */
avctx->block_align = (2 + channels * (22 + 4 * (avctx->frame_size - 1)) + 7) / 8;
) /* End of CASE */
case AV_CODEC_ID_ADPCM_IMA_SSI:
case AV_CODEC_ID_ADPCM_IMA_ALP:
avctx->frame_size = s->block_size * 2 / channels;
avctx->block_align = s->block_size;
break;
CASE(ADPCM_IMA_AMV,
if (avctx->sample_rate != 22050) {
av_log(avctx, AV_LOG_ERROR, "Sample rate must be 22050\n");
return AVERROR(EINVAL);
}
if (channels != 1) {
av_log(avctx, AV_LOG_ERROR, "Only mono is supported\n");
return AVERROR(EINVAL);
}
avctx->frame_size = s->block_size;
avctx->block_align = 8 + (FFALIGN(avctx->frame_size, 2) / 2);
) /* End of CASE */
CASE(ADPCM_IMA_APM,
avctx->frame_size = s->block_size * 2 / channels;
avctx->block_align = s->block_size;
if (!(avctx->extradata = av_mallocz(28 + AV_INPUT_BUFFER_PADDING_SIZE)))
return AVERROR(ENOMEM);
avctx->extradata_size = 28;
) /* End of CASE */
CASE(ADPCM_ARGO,
avctx->frame_size = 32;
avctx->block_align = 17 * channels;
) /* End of CASE */
CASE(ADPCM_IMA_WS,
/* each 16 bits sample gives one nibble */
avctx->frame_size = s->block_size * 2 / channels;
avctx->block_align = s->block_size;
) /* End of CASE */
default:
return AVERROR(EINVAL);
}
return 0;
}
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_alp_compress_sample(ADPCMChannelStatus *c, int16_t sample)
{
const int delta = sample - c->prev_sample;
const int step = ff_adpcm_step_table[c->step_index];
const int sign = (delta < 0) * 8;
int nibble = FFMIN(abs(delta) * 4 / step, 7);
int diff = (step * nibble) >> 2;
if (sign)
diff = -diff;
nibble = sign | nibble;
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_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_intp2(nibble, 3) & 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, 24576);
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_IMA_AMV ||
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*(unsigned)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_IMA_AMV ||
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, 24576));
#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;
}
#if CONFIG_ADPCM_ARGO_ENCODER
static inline int adpcm_argo_compress_nibble(const ADPCMChannelStatus *cs, int16_t s,
int shift, int flag)
{
int nibble;
if (flag)
nibble = 4 * s - 8 * cs->sample1 + 4 * cs->sample2;
else
nibble = 4 * s - 4 * cs->sample1;
return (nibble >> shift) & 0x0F;
}
static int64_t adpcm_argo_compress_block(ADPCMChannelStatus *cs, PutBitContext *pb,
const int16_t *samples, int nsamples,
int shift, int flag)
{
int64_t error = 0;
if (pb) {
put_bits(pb, 4, shift - 2);
put_bits(pb, 1, 0);
put_bits(pb, 1, !!flag);
put_bits(pb, 2, 0);
}
for (int n = 0; n < nsamples; n++) {
/* Compress the nibble, then expand it to see how much precision we've lost. */
int nibble = adpcm_argo_compress_nibble(cs, samples[n], shift, flag);
int16_t sample = ff_adpcm_argo_expand_nibble(cs, nibble, shift, flag);
error += abs(samples[n] - sample);
if (pb)
put_bits(pb, 4, nibble);
}
return error;
}
#endif
static int adpcm_encode_frame(AVCodecContext *avctx, AVPacket *avpkt,
const AVFrame *frame, int *got_packet_ptr)
{
int st, pkt_size, ret;
const int16_t *samples;
int16_t **samples_p;
uint8_t *dst;
ADPCMEncodeContext *c = avctx->priv_data;
int channels = avctx->ch_layout.nb_channels;
samples = (const int16_t *)frame->data[0];
samples_p = (int16_t **)frame->extended_data;
st = channels == 2;
if (avctx->codec_id == AV_CODEC_ID_ADPCM_IMA_SSI ||
avctx->codec_id == AV_CODEC_ID_ADPCM_IMA_ALP ||
avctx->codec_id == AV_CODEC_ID_ADPCM_IMA_APM ||
avctx->codec_id == AV_CODEC_ID_ADPCM_IMA_WS)
pkt_size = (frame->nb_samples * channels) / 2;
else
pkt_size = avctx->block_align;
if ((ret = ff_get_encode_buffer(avctx, avpkt, pkt_size, 0)) < 0)
return ret;
dst = avpkt->data;
switch(avctx->codec->id) {
CASE(ADPCM_IMA_WAV,
int blocks = (frame->nb_samples - 1) / 8;
for (int ch = 0; ch < 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) {
uint8_t *buf;
if (!FF_ALLOC_TYPED_ARRAY(buf, channels * blocks * 8))
return AVERROR(ENOMEM);
for (int ch = 0; ch < channels; ch++) {
adpcm_compress_trellis(avctx, &samples_p[ch][1],
buf + ch * blocks * 8, &c->status[ch],
blocks * 8, 1);
}
for (int i = 0; i < blocks; i++) {
for (int ch = 0; ch < channels; ch++) {
uint8_t *buf1 = buf + ch * blocks * 8 + i * 8;
for (int j = 0; j < 8; j += 2)
*dst++ = buf1[j] | (buf1[j + 1] << 4);
}
}
av_free(buf);
} else {
for (int i = 0; i < blocks; i++) {
for (int ch = 0; ch < channels; ch++) {
ADPCMChannelStatus *status = &c->status[ch];
const int16_t *smp = &samples_p[ch][1 + i * 8];
for (int 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;
}
}
}
}
) /* End of CASE */
CASE(ADPCM_IMA_QT,
PutBitContext pb;
init_put_bits(&pb, dst, pkt_size);
for (int ch = 0; ch < 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][0], buf, status,
64, 1);
for (int i = 0; i < 64; i++)
put_bits(&pb, 4, buf[i ^ 1]);
status->prev_sample = status->predictor;
} else {
for (int 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);
) /* End of CASE */
CASE(ADPCM_IMA_SSI,
PutBitContext pb;
init_put_bits(&pb, dst, pkt_size);
av_assert0(avctx->trellis == 0);
for (int i = 0; i < frame->nb_samples; i++) {
for (int ch = 0; ch < channels; ch++) {
put_bits(&pb, 4, adpcm_ima_qt_compress_sample(c->status + ch, *samples++));
}
}
flush_put_bits(&pb);
) /* End of CASE */
CASE(ADPCM_IMA_ALP,
PutBitContext pb;
init_put_bits(&pb, dst, pkt_size);
av_assert0(avctx->trellis == 0);
for (int n = frame->nb_samples / 2; n > 0; n--) {
for (int ch = 0; ch < channels; ch++) {
put_bits(&pb, 4, adpcm_ima_alp_compress_sample(c->status + ch, *samples++));
put_bits(&pb, 4, adpcm_ima_alp_compress_sample(c->status + ch, samples[st]));
}
samples += channels;
}
flush_put_bits(&pb);
) /* End of CASE */
CASE(ADPCM_SWF,
const int n = frame->nb_samples - 1;
PutBitContext pb;
init_put_bits(&pb, dst, pkt_size);
/* NB: This is safe as we don't have AV_CODEC_CAP_SMALL_LAST_FRAME. */
av_assert0(n == 4095);
// store AdpcmCodeSize
put_bits(&pb, 2, 2); // set 4-bit flash adpcm format
// init the encoder state
for (int i = 0; i < channels; i++) {
// clip step so it fits 6 bits
c->status[i].step_index = av_clip_uintp2(c->status[i].step_index, 6);
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) {
uint8_t buf[8190 /* = 2 * n */];
adpcm_compress_trellis(avctx, samples + channels, buf,
&c->status[0], n, channels);
if (channels == 2)
adpcm_compress_trellis(avctx, samples + channels + 1,
buf + n, &c->status[1], n,
channels);
for (int i = 0; i < n; i++) {
put_bits(&pb, 4, buf[i]);
if (channels == 2)
put_bits(&pb, 4, buf[n + i]);
}
} else {
for (int i = 1; i < frame->nb_samples; i++) {
put_bits(&pb, 4, adpcm_ima_compress_sample(&c->status[0],
samples[channels * i]));
if (channels == 2)
put_bits(&pb, 4, adpcm_ima_compress_sample(&c->status[1],
samples[2 * i + 1]));
}
}
flush_put_bits(&pb);
) /* End of CASE */
CASE(ADPCM_MS,
for (int i = 0; i < channels; i++) {
int predictor = 0;
*dst++ = predictor;
c->status[i].coeff1 = ff_adpcm_AdaptCoeff1[predictor];
c->status[i].coeff2 = ff_adpcm_AdaptCoeff2[predictor];
}
for (int i = 0; i < channels; i++) {
if (c->status[i].idelta < 16)
c->status[i].idelta = 16;
bytestream_put_le16(&dst, c->status[i].idelta);
}
for (int i = 0; i < channels; i++)
c->status[i].sample2= *samples++;
for (int i = 0; i < channels; i++) {
c->status[i].sample1 = *samples++;
bytestream_put_le16(&dst, c->status[i].sample1);
}
for (int i = 0; i < channels; i++)
bytestream_put_le16(&dst, c->status[i].sample2);
if (avctx->trellis > 0) {
const int n = avctx->block_align - 7 * channels;
uint8_t *buf = av_malloc(2 * n);
if (!buf)
return AVERROR(ENOMEM);
if (channels == 1) {
adpcm_compress_trellis(avctx, samples, buf, &c->status[0], n,
channels);
for (int i = 0; i < n; i += 2)
*dst++ = (buf[i] << 4) | buf[i + 1];
} else {
adpcm_compress_trellis(avctx, samples, buf,
&c->status[0], n, channels);
adpcm_compress_trellis(avctx, samples + 1, buf + n,
&c->status[1], n, channels);
for (int i = 0; i < n; i++)
*dst++ = (buf[i] << 4) | buf[n + i];
}
av_free(buf);
} else {
for (int i = 7 * 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;
}
}
) /* End of CASE */
CASE(ADPCM_YAMAHA,
int n = frame->nb_samples / 2;
if (avctx->trellis > 0) {
uint8_t *buf = av_malloc(2 * n * 2);
if (!buf)
return AVERROR(ENOMEM);
n *= 2;
if (channels == 1) {
adpcm_compress_trellis(avctx, samples, buf, &c->status[0], n,
channels);
for (int i = 0; i < n; i += 2)
*dst++ = buf[i] | (buf[i + 1] << 4);
} else {
adpcm_compress_trellis(avctx, samples, buf,
&c->status[0], n, channels);
adpcm_compress_trellis(avctx, samples + 1, buf + n,
&c->status[1], n, channels);
for (int i = 0; i < n; i++)
*dst++ = buf[i] | (buf[n + i] << 4);
}
av_free(buf);
} else
for (n *= 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;
}
) /* End of CASE */
CASE(ADPCM_IMA_APM,
PutBitContext pb;
init_put_bits(&pb, dst, pkt_size);
av_assert0(avctx->trellis == 0);
for (int n = frame->nb_samples / 2; n > 0; n--) {
for (int ch = 0; ch < channels; ch++) {
put_bits(&pb, 4, adpcm_ima_qt_compress_sample(c->status + ch, *samples++));
put_bits(&pb, 4, adpcm_ima_qt_compress_sample(c->status + ch, samples[st]));
}
samples += channels;
}
flush_put_bits(&pb);
) /* End of CASE */
CASE(ADPCM_IMA_AMV,
av_assert0(channels == 1);
c->status[0].prev_sample = *samples;
bytestream_put_le16(&dst, c->status[0].prev_sample);
bytestream_put_byte(&dst, c->status[0].step_index);
bytestream_put_byte(&dst, 0);
bytestream_put_le32(&dst, avctx->frame_size);
if (avctx->trellis > 0) {
const int n = frame->nb_samples >> 1;
uint8_t *buf = av_malloc(2 * n);
if (!buf)
return AVERROR(ENOMEM);
adpcm_compress_trellis(avctx, samples, buf, &c->status[0], 2 * n, channels);
for (int i = 0; i < n; i++)
bytestream_put_byte(&dst, (buf[2 * i] << 4) | buf[2 * i + 1]);
samples += 2 * n;
av_free(buf);
} else for (int n = frame->nb_samples >> 1; n > 0; n--) {
int nibble;
nibble = adpcm_ima_compress_sample(&c->status[0], *samples++) << 4;
nibble |= adpcm_ima_compress_sample(&c->status[0], *samples++) & 0x0F;
bytestream_put_byte(&dst, nibble);
}
if (avctx->frame_size & 1) {
int nibble = adpcm_ima_compress_sample(&c->status[0], *samples++) << 4;
bytestream_put_byte(&dst, nibble);
}
) /* End of CASE */
CASE(ADPCM_ARGO,
PutBitContext pb;
init_put_bits(&pb, dst, pkt_size);
av_assert0(frame->nb_samples == 32);
for (int ch = 0; ch < channels; ch++) {
int64_t error = INT64_MAX, tmperr = INT64_MAX;
int shift = 2, flag = 0;
int saved1 = c->status[ch].sample1;
int saved2 = c->status[ch].sample2;
/* Find the optimal coefficients, bail early if we find a perfect result. */
for (int s = 2; s < 18 && tmperr != 0; s++) {
for (int f = 0; f < 2 && tmperr != 0; f++) {
c->status[ch].sample1 = saved1;
c->status[ch].sample2 = saved2;
tmperr = adpcm_argo_compress_block(c->status + ch, NULL, samples_p[ch],
frame->nb_samples, s, f);
if (tmperr < error) {
shift = s;
flag = f;
error = tmperr;
}
}
}
/* Now actually do the encode. */
c->status[ch].sample1 = saved1;
c->status[ch].sample2 = saved2;
adpcm_argo_compress_block(c->status + ch, &pb, samples_p[ch],
frame->nb_samples, shift, flag);
}
flush_put_bits(&pb);
) /* End of CASE */
CASE(ADPCM_IMA_WS,
PutBitContext pb;
init_put_bits(&pb, dst, pkt_size);
av_assert0(avctx->trellis == 0);
for (int n = frame->nb_samples / 2; n > 0; n--) {
/* stereo: 1 byte (2 samples) for left, 1 byte for right */
for (int ch = 0; ch < channels; ch++) {
int t1, t2;
t1 = adpcm_ima_compress_sample(&c->status[ch], *samples++);
t2 = adpcm_ima_compress_sample(&c->status[ch], samples[st]);
put_bits(&pb, 4, t2);
put_bits(&pb, 4, t1);
}
samples += channels;
}
flush_put_bits(&pb);
) /* End of CASE */
default:
return AVERROR(EINVAL);
}
*got_packet_ptr = 1;
return 0;
}
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
};
static const AVChannelLayout ch_layouts[] = {
AV_CHANNEL_LAYOUT_MONO,
AV_CHANNEL_LAYOUT_STEREO,
{ 0 },
};
static const AVOption options[] = {
{
.name = "block_size",
.help = "set the block size",
.offset = offsetof(ADPCMEncodeContext, block_size),
.type = AV_OPT_TYPE_INT,
.default_val = {.i64 = 1024},
.min = 32,
.max = 8192, /* Is this a reasonable upper limit? */
.flags = AV_OPT_FLAG_ENCODING_PARAM | AV_OPT_FLAG_AUDIO_PARAM
},
{ NULL }
};
static const AVClass adpcm_encoder_class = {
.class_name = "ADPCM encoder",
.item_name = av_default_item_name,
.option = options,
.version = LIBAVUTIL_VERSION_INT,
};
#define ADPCM_ENCODER_0(id_, name_, sample_fmts_, capabilities_, long_name_)
#define ADPCM_ENCODER_1(id_, name_, sample_fmts_, capabilities_, long_name_) \
const FFCodec ff_ ## name_ ## _encoder = { \
.p.name = #name_, \
.p.long_name = NULL_IF_CONFIG_SMALL(long_name_), \
.p.type = AVMEDIA_TYPE_AUDIO, \
.p.id = id_, \
.p.sample_fmts = sample_fmts_, \
.p.ch_layouts = ch_layouts, \
.p.capabilities = capabilities_ | AV_CODEC_CAP_DR1, \
.p.priv_class = &adpcm_encoder_class, \
.priv_data_size = sizeof(ADPCMEncodeContext), \
.init = adpcm_encode_init, \
FF_CODEC_ENCODE_CB(adpcm_encode_frame), \
.close = adpcm_encode_close, \
.caps_internal = FF_CODEC_CAP_INIT_CLEANUP | FF_CODEC_CAP_INIT_THREADSAFE, \
};
#define ADPCM_ENCODER_2(enabled, codec_id, name, sample_fmts, capabilities, long_name) \
ADPCM_ENCODER_ ## enabled(codec_id, name, sample_fmts, capabilities, long_name)
#define ADPCM_ENCODER_3(config, codec_id, name, sample_fmts, capabilities, long_name) \
ADPCM_ENCODER_2(config, codec_id, name, sample_fmts, capabilities, long_name)
#define ADPCM_ENCODER(codec, name, sample_fmts, capabilities, long_name) \
ADPCM_ENCODER_3(CONFIG_ ## codec ## _ENCODER, AV_CODEC_ID_ ## codec, \
name, sample_fmts, capabilities, long_name)
ADPCM_ENCODER(ADPCM_ARGO, adpcm_argo, sample_fmts_p, 0, "ADPCM Argonaut Games")
ADPCM_ENCODER(ADPCM_IMA_AMV, adpcm_ima_amv, sample_fmts, 0, "ADPCM IMA AMV")
ADPCM_ENCODER(ADPCM_IMA_APM, adpcm_ima_apm, sample_fmts, AV_CODEC_CAP_SMALL_LAST_FRAME, "ADPCM IMA Ubisoft APM")
ADPCM_ENCODER(ADPCM_IMA_ALP, adpcm_ima_alp, sample_fmts, AV_CODEC_CAP_SMALL_LAST_FRAME, "ADPCM IMA High Voltage Software ALP")
ADPCM_ENCODER(ADPCM_IMA_QT, adpcm_ima_qt, sample_fmts_p, 0, "ADPCM IMA QuickTime")
ADPCM_ENCODER(ADPCM_IMA_SSI, adpcm_ima_ssi, sample_fmts, AV_CODEC_CAP_SMALL_LAST_FRAME, "ADPCM IMA Simon & Schuster Interactive")
ADPCM_ENCODER(ADPCM_IMA_WAV, adpcm_ima_wav, sample_fmts_p, 0, "ADPCM IMA WAV")
ADPCM_ENCODER(ADPCM_IMA_WS, adpcm_ima_ws, sample_fmts, AV_CODEC_CAP_SMALL_LAST_FRAME, "ADPCM IMA Westwood")
ADPCM_ENCODER(ADPCM_MS, adpcm_ms, sample_fmts, 0, "ADPCM Microsoft")
ADPCM_ENCODER(ADPCM_SWF, adpcm_swf, sample_fmts, 0, "ADPCM Shockwave Flash")
ADPCM_ENCODER(ADPCM_YAMAHA, adpcm_yamaha, sample_fmts, 0, "ADPCM Yamaha")
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