virtualenv/FFmpeg
1
0
Fork 0
mirror of https://github.com/FFmpeg/FFmpeg.git synced 2024-12-07 11:13:41 +02:00
Code Issues Releases Activity
bd737b5178
FFmpeg/libavcodec/adpcmenc.c
Martin Storsjö 0776e0ef6b adpcm: Write the proper predictor in trellis mode in IMA QT 
The actual predictor value, set by the trellis code, never
was written back into the variable that was written into
the block header. This was accidentally removed in b304244b.

This significantly improves the audio quality of the trellis
case, which was plain broken since b304244b.

Encoding IMA QT with trellis still actually gives a slightly
worse quality than without trellis, since the trellis encoder
doesn't use the exact same way of rounding as in
adpcm_ima_qt_compress_sample and adpcm_ima_qt_expand_nibble.

CC: libav-stable@libav.org
Signed-off-by: Martin Storsjö <martin@martin.st>
2014-06-10 16:28:47 +03:00

727 lines
27 KiB
C
Raw Blame History

/*
* 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 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"
#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_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;
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->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:
av_freep(&s->paths);
av_freep(&s->node_buf);
av_freep(&s->nodep_buf);
av_freep(&s->trellis_hash);
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 mask, step = ff_adpcm_step_table[c->step_index];
int diff = step >> 3;
int nibble = 0;
if (delta < 0) {
nibble = 8;
delta = -delta;
}
for (mask = 4; mask;) {
if (delta >= step) {
nibble |= mask;
delta -= step;
diff += step;
}
step >>= 1;
mask >>= 1;
}
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) {\
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 == 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_packet(avpkt, pkt_size))) {
av_log(avctx, AV_LOG_ERROR, "Error getting output packet\n");
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][0], buf, status,
64, 1);
for (i = 0; i < 64; i++)
put_bits(&pb, 4, buf[i ^ 1]);
status->prev_sample = status->predictor;
} 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");
Reference in New Issue View Git Blame Copy Permalink