Libav
adpcmenc.c
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1 /*
2  * Copyright (c) 2001-2003 The FFmpeg project
3  *
4  * first version by Francois Revol (revol@free.fr)
5  * fringe ADPCM codecs (e.g., DK3, DK4, Westwood)
6  * by Mike Melanson (melanson@pcisys.net)
7  *
8  * This file is part of Libav.
9  *
10  * Libav is free software; you can redistribute it and/or
11  * modify it under the terms of the GNU Lesser General Public
12  * License as published by the Free Software Foundation; either
13  * version 2.1 of the License, or (at your option) any later version.
14  *
15  * Libav is distributed in the hope that it will be useful,
16  * but WITHOUT ANY WARRANTY; without even the implied warranty of
17  * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
18  * Lesser General Public License for more details.
19  *
20  * You should have received a copy of the GNU Lesser General Public
21  * License along with Libav; if not, write to the Free Software
22  * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
23  */
24 
25 #include "avcodec.h"
26 #include "put_bits.h"
27 #include "bytestream.h"
28 #include "adpcm.h"
29 #include "adpcm_data.h"
30 #include "internal.h"
31 
38 typedef struct TrellisPath {
39  int nibble;
40  int prev;
41 } TrellisPath;
42 
43 typedef struct TrellisNode {
44  uint32_t ssd;
45  int path;
46  int sample1;
47  int sample2;
48  int step;
49 } TrellisNode;
50 
51 typedef struct ADPCMEncodeContext {
52  ADPCMChannelStatus status[6];
58 
59 #define FREEZE_INTERVAL 128
60 
62 {
63  ADPCMEncodeContext *s = avctx->priv_data;
64  uint8_t *extradata;
65  int i;
66  int ret = AVERROR(ENOMEM);
67 
68  if (avctx->channels > 2) {
69  av_log(avctx, AV_LOG_ERROR, "only stereo or mono is supported\n");
70  return AVERROR(EINVAL);
71  }
72 
73  if (avctx->trellis && (unsigned)avctx->trellis > 16U) {
74  av_log(avctx, AV_LOG_ERROR, "invalid trellis size\n");
75  return AVERROR(EINVAL);
76  }
77 
78  if (avctx->trellis) {
79  int frontier = 1 << avctx->trellis;
80  int max_paths = frontier * FREEZE_INTERVAL;
81  FF_ALLOC_OR_GOTO(avctx, s->paths,
82  max_paths * sizeof(*s->paths), error);
83  FF_ALLOC_OR_GOTO(avctx, s->node_buf,
84  2 * frontier * sizeof(*s->node_buf), error);
85  FF_ALLOC_OR_GOTO(avctx, s->nodep_buf,
86  2 * frontier * sizeof(*s->nodep_buf), error);
88  65536 * sizeof(*s->trellis_hash), error);
89  }
90 
92 
93  switch (avctx->codec->id) {
95  /* each 16 bits sample gives one nibble
96  and we have 4 bytes per channel overhead */
97  avctx->frame_size = (BLKSIZE - 4 * avctx->channels) * 8 /
98  (4 * avctx->channels) + 1;
99  /* seems frame_size isn't taken into account...
100  have to buffer the samples :-( */
101  avctx->block_align = BLKSIZE;
102  break;
104  avctx->frame_size = 64;
105  avctx->block_align = 34 * avctx->channels;
106  break;
108  /* each 16 bits sample gives one nibble
109  and we have 7 bytes per channel overhead */
110  avctx->frame_size = (BLKSIZE - 7 * avctx->channels) * 2 /
111  avctx->channels + 2;
112  avctx->block_align = BLKSIZE;
114  goto error;
115  avctx->extradata_size = 32;
116  extradata = avctx->extradata;
117  bytestream_put_le16(&extradata, avctx->frame_size);
118  bytestream_put_le16(&extradata, 7); /* wNumCoef */
119  for (i = 0; i < 7; i++) {
120  bytestream_put_le16(&extradata, ff_adpcm_AdaptCoeff1[i] * 4);
121  bytestream_put_le16(&extradata, ff_adpcm_AdaptCoeff2[i] * 4);
122  }
123  break;
125  avctx->frame_size = BLKSIZE * 2 / avctx->channels;
126  avctx->block_align = BLKSIZE;
127  break;
129  if (avctx->sample_rate != 11025 &&
130  avctx->sample_rate != 22050 &&
131  avctx->sample_rate != 44100) {
132  av_log(avctx, AV_LOG_ERROR, "Sample rate must be 11025, "
133  "22050 or 44100\n");
134  ret = AVERROR(EINVAL);
135  goto error;
136  }
137  avctx->frame_size = 512 * (avctx->sample_rate / 11025);
138  break;
139  default:
140  ret = AVERROR(EINVAL);
141  goto error;
142  }
143 
144  return 0;
145 error:
146  av_freep(&s->paths);
147  av_freep(&s->node_buf);
148  av_freep(&s->nodep_buf);
149  av_freep(&s->trellis_hash);
150  return ret;
151 }
152 
154 {
155  ADPCMEncodeContext *s = avctx->priv_data;
156  av_freep(&s->paths);
157  av_freep(&s->node_buf);
158  av_freep(&s->nodep_buf);
159  av_freep(&s->trellis_hash);
160 
161  return 0;
162 }
163 
164 
166  int16_t sample)
167 {
168  int delta = sample - c->prev_sample;
169  int nibble = FFMIN(7, abs(delta) * 4 /
170  ff_adpcm_step_table[c->step_index]) + (delta < 0) * 8;
173  c->prev_sample = av_clip_int16(c->prev_sample);
174  c->step_index = av_clip(c->step_index + ff_adpcm_index_table[nibble], 0, 88);
175  return nibble;
176 }
177 
179  int16_t sample)
180 {
181  int delta = sample - c->prev_sample;
183  int diff = step >> 3;
184  int nibble = 0;
185 
186  if (delta < 0) {
187  nibble = 8;
188  delta = -delta;
189  }
190 
191  for (mask = 4; mask;) {
192  if (delta >= step) {
193  nibble |= mask;
194  delta -= step;
195  diff += step;
196  }
197  step >>= 1;
198  mask >>= 1;
199  }
200 
201  if (nibble & 8)
202  c->prev_sample -= diff;
203  else
204  c->prev_sample += diff;
205 
206  c->prev_sample = av_clip_int16(c->prev_sample);
207  c->step_index = av_clip(c->step_index + ff_adpcm_index_table[nibble], 0, 88);
208 
209  return nibble;
210 }
211 
213  int16_t sample)
214 {
215  int predictor, nibble, bias;
216 
217  predictor = (((c->sample1) * (c->coeff1)) +
218  (( c->sample2) * (c->coeff2))) / 64;
219 
220  nibble = sample - predictor;
221  if (nibble >= 0)
222  bias = c->idelta / 2;
223  else
224  bias = -c->idelta / 2;
225 
226  nibble = (nibble + bias) / c->idelta;
227  nibble = av_clip(nibble, -8, 7) & 0x0F;
228 
229  predictor += ((nibble & 0x08) ? (nibble - 0x10) : nibble) * c->idelta;
230 
231  c->sample2 = c->sample1;
232  c->sample1 = av_clip_int16(predictor);
233 
234  c->idelta = (ff_adpcm_AdaptationTable[nibble] * c->idelta) >> 8;
235  if (c->idelta < 16)
236  c->idelta = 16;
237 
238  return nibble;
239 }
240 
242  int16_t sample)
243 {
244  int nibble, delta;
245 
246  if (!c->step) {
247  c->predictor = 0;
248  c->step = 127;
249  }
250 
251  delta = sample - c->predictor;
252 
253  nibble = FFMIN(7, abs(delta) * 4 / c->step) + (delta < 0) * 8;
254 
255  c->predictor += ((c->step * ff_adpcm_yamaha_difflookup[nibble]) / 8);
256  c->predictor = av_clip_int16(c->predictor);
257  c->step = (c->step * ff_adpcm_yamaha_indexscale[nibble]) >> 8;
258  c->step = av_clip(c->step, 127, 24567);
259 
260  return nibble;
261 }
262 
264  const int16_t *samples, uint8_t *dst,
265  ADPCMChannelStatus *c, int n, int stride)
266 {
267  //FIXME 6% faster if frontier is a compile-time constant
268  ADPCMEncodeContext *s = avctx->priv_data;
269  const int frontier = 1 << avctx->trellis;
270  const int version = avctx->codec->id;
271  TrellisPath *paths = s->paths, *p;
272  TrellisNode *node_buf = s->node_buf;
273  TrellisNode **nodep_buf = s->nodep_buf;
274  TrellisNode **nodes = nodep_buf; // nodes[] is always sorted by .ssd
275  TrellisNode **nodes_next = nodep_buf + frontier;
276  int pathn = 0, froze = -1, i, j, k, generation = 0;
277  uint8_t *hash = s->trellis_hash;
278  memset(hash, 0xff, 65536 * sizeof(*hash));
279 
280  memset(nodep_buf, 0, 2 * frontier * sizeof(*nodep_buf));
281  nodes[0] = node_buf + frontier;
282  nodes[0]->ssd = 0;
283  nodes[0]->path = 0;
284  nodes[0]->step = c->step_index;
285  nodes[0]->sample1 = c->sample1;
286  nodes[0]->sample2 = c->sample2;
287  if (version == AV_CODEC_ID_ADPCM_IMA_WAV ||
288  version == AV_CODEC_ID_ADPCM_IMA_QT ||
289  version == AV_CODEC_ID_ADPCM_SWF)
290  nodes[0]->sample1 = c->prev_sample;
291  if (version == AV_CODEC_ID_ADPCM_MS)
292  nodes[0]->step = c->idelta;
293  if (version == AV_CODEC_ID_ADPCM_YAMAHA) {
294  if (c->step == 0) {
295  nodes[0]->step = 127;
296  nodes[0]->sample1 = 0;
297  } else {
298  nodes[0]->step = c->step;
299  nodes[0]->sample1 = c->predictor;
300  }
301  }
302 
303  for (i = 0; i < n; i++) {
304  TrellisNode *t = node_buf + frontier*(i&1);
305  TrellisNode **u;
306  int sample = samples[i * stride];
307  int heap_pos = 0;
308  memset(nodes_next, 0, frontier * sizeof(TrellisNode*));
309  for (j = 0; j < frontier && nodes[j]; j++) {
310  // higher j have higher ssd already, so they're likely
311  // to yield a suboptimal next sample too
312  const int range = (j < frontier / 2) ? 1 : 0;
313  const int step = nodes[j]->step;
314  int nidx;
315  if (version == AV_CODEC_ID_ADPCM_MS) {
316  const int predictor = ((nodes[j]->sample1 * c->coeff1) +
317  (nodes[j]->sample2 * c->coeff2)) / 64;
318  const int div = (sample - predictor) / step;
319  const int nmin = av_clip(div-range, -8, 6);
320  const int nmax = av_clip(div+range, -7, 7);
321  for (nidx = nmin; nidx <= nmax; nidx++) {
322  const int nibble = nidx & 0xf;
323  int dec_sample = predictor + nidx * step;
324 #define STORE_NODE(NAME, STEP_INDEX)\
325  int d;\
326  uint32_t ssd;\
327  int pos;\
328  TrellisNode *u;\
329  uint8_t *h;\
330  dec_sample = av_clip_int16(dec_sample);\
331  d = sample - dec_sample;\
332  ssd = nodes[j]->ssd + d*d;\
333  /* Check for wraparound, skip such samples completely. \
334  * Note, changing ssd to a 64 bit variable would be \
335  * simpler, avoiding this check, but it's slower on \
336  * x86 32 bit at the moment. */\
337  if (ssd < nodes[j]->ssd)\
338  goto next_##NAME;\
339  /* Collapse any two states with the same previous sample value. \
340  * One could also distinguish states by step and by 2nd to last
341  * sample, but the effects of that are negligible.
342  * Since nodes in the previous generation are iterated
343  * through a heap, they're roughly ordered from better to
344  * worse, but not strictly ordered. Therefore, an earlier
345  * node with the same sample value is better in most cases
346  * (and thus the current is skipped), but not strictly
347  * in all cases. Only skipping samples where ssd >=
348  * ssd of the earlier node with the same sample gives
349  * slightly worse quality, though, for some reason. */ \
350  h = &hash[(uint16_t) dec_sample];\
351  if (*h == generation)\
352  goto next_##NAME;\
353  if (heap_pos < frontier) {\
354  pos = heap_pos++;\
355  } else {\
356  /* Try to replace one of the leaf nodes with the new \
357  * one, but try a different slot each time. */\
358  pos = (frontier >> 1) +\
359  (heap_pos & ((frontier >> 1) - 1));\
360  if (ssd > nodes_next[pos]->ssd)\
361  goto next_##NAME;\
362  heap_pos++;\
363  }\
364  *h = generation;\
365  u = nodes_next[pos];\
366  if (!u) {\
367  assert(pathn < FREEZE_INTERVAL << avctx->trellis);\
368  u = t++;\
369  nodes_next[pos] = u;\
370  u->path = pathn++;\
371  }\
372  u->ssd = ssd;\
373  u->step = STEP_INDEX;\
374  u->sample2 = nodes[j]->sample1;\
375  u->sample1 = dec_sample;\
376  paths[u->path].nibble = nibble;\
377  paths[u->path].prev = nodes[j]->path;\
378  /* Sift the newly inserted node up in the heap to \
379  * restore the heap property. */\
380  while (pos > 0) {\
381  int parent = (pos - 1) >> 1;\
382  if (nodes_next[parent]->ssd <= ssd)\
383  break;\
384  FFSWAP(TrellisNode*, nodes_next[parent], nodes_next[pos]);\
385  pos = parent;\
386  }\
387  next_##NAME:;
388  STORE_NODE(ms, FFMAX(16,
389  (ff_adpcm_AdaptationTable[nibble] * step) >> 8));
390  }
391  } else if (version == AV_CODEC_ID_ADPCM_IMA_WAV ||
392  version == AV_CODEC_ID_ADPCM_IMA_QT ||
393  version == AV_CODEC_ID_ADPCM_SWF) {
394 #define LOOP_NODES(NAME, STEP_TABLE, STEP_INDEX)\
395  const int predictor = nodes[j]->sample1;\
396  const int div = (sample - predictor) * 4 / STEP_TABLE;\
397  int nmin = av_clip(div - range, -7, 6);\
398  int nmax = av_clip(div + range, -6, 7);\
399  if (nmin <= 0)\
400  nmin--; /* distinguish -0 from +0 */\
401  if (nmax < 0)\
402  nmax--;\
403  for (nidx = nmin; nidx <= nmax; nidx++) {\
404  const int nibble = nidx < 0 ? 7 - nidx : nidx;\
405  int dec_sample = predictor +\
406  (STEP_TABLE *\
407  ff_adpcm_yamaha_difflookup[nibble]) / 8;\
408  STORE_NODE(NAME, STEP_INDEX);\
409  }
410  LOOP_NODES(ima, ff_adpcm_step_table[step],
411  av_clip(step + ff_adpcm_index_table[nibble], 0, 88));
412  } else { //AV_CODEC_ID_ADPCM_YAMAHA
413  LOOP_NODES(yamaha, step,
414  av_clip((step * ff_adpcm_yamaha_indexscale[nibble]) >> 8,
415  127, 24567));
416 #undef LOOP_NODES
417 #undef STORE_NODE
418  }
419  }
420 
421  u = nodes;
422  nodes = nodes_next;
423  nodes_next = u;
424 
425  generation++;
426  if (generation == 255) {
427  memset(hash, 0xff, 65536 * sizeof(*hash));
428  generation = 0;
429  }
430 
431  // prevent overflow
432  if (nodes[0]->ssd > (1 << 28)) {
433  for (j = 1; j < frontier && nodes[j]; j++)
434  nodes[j]->ssd -= nodes[0]->ssd;
435  nodes[0]->ssd = 0;
436  }
437 
438  // merge old paths to save memory
439  if (i == froze + FREEZE_INTERVAL) {
440  p = &paths[nodes[0]->path];
441  for (k = i; k > froze; k--) {
442  dst[k] = p->nibble;
443  p = &paths[p->prev];
444  }
445  froze = i;
446  pathn = 0;
447  // other nodes might use paths that don't coincide with the frozen one.
448  // checking which nodes do so is too slow, so just kill them all.
449  // this also slightly improves quality, but I don't know why.
450  memset(nodes + 1, 0, (frontier - 1) * sizeof(TrellisNode*));
451  }
452  }
453 
454  p = &paths[nodes[0]->path];
455  for (i = n - 1; i > froze; i--) {
456  dst[i] = p->nibble;
457  p = &paths[p->prev];
458  }
459 
460  c->predictor = nodes[0]->sample1;
461  c->sample1 = nodes[0]->sample1;
462  c->sample2 = nodes[0]->sample2;
463  c->step_index = nodes[0]->step;
464  c->step = nodes[0]->step;
465  c->idelta = nodes[0]->step;
466 }
467 
468 static int adpcm_encode_frame(AVCodecContext *avctx, AVPacket *avpkt,
469  const AVFrame *frame, int *got_packet_ptr)
470 {
471  int n, i, ch, st, pkt_size, ret;
472  const int16_t *samples;
473  int16_t **samples_p;
474  uint8_t *dst;
475  ADPCMEncodeContext *c = avctx->priv_data;
476  uint8_t *buf;
477 
478  samples = (const int16_t *)frame->data[0];
479  samples_p = (int16_t **)frame->extended_data;
480  st = avctx->channels == 2;
481 
482  if (avctx->codec_id == AV_CODEC_ID_ADPCM_SWF)
483  pkt_size = (2 + avctx->channels * (22 + 4 * (frame->nb_samples - 1)) + 7) / 8;
484  else
485  pkt_size = avctx->block_align;
486  if ((ret = ff_alloc_packet(avpkt, pkt_size))) {
487  av_log(avctx, AV_LOG_ERROR, "Error getting output packet\n");
488  return ret;
489  }
490  dst = avpkt->data;
491 
492  switch(avctx->codec->id) {
494  {
495  int blocks, j;
496 
497  blocks = (frame->nb_samples - 1) / 8;
498 
499  for (ch = 0; ch < avctx->channels; ch++) {
500  ADPCMChannelStatus *status = &c->status[ch];
501  status->prev_sample = samples_p[ch][0];
502  /* status->step_index = 0;
503  XXX: not sure how to init the state machine */
504  bytestream_put_le16(&dst, status->prev_sample);
505  *dst++ = status->step_index;
506  *dst++ = 0; /* unknown */
507  }
508 
509  /* stereo: 4 bytes (8 samples) for left, 4 bytes for right */
510  if (avctx->trellis > 0) {
511  FF_ALLOC_OR_GOTO(avctx, buf, avctx->channels * blocks * 8, error);
512  for (ch = 0; ch < avctx->channels; ch++) {
513  adpcm_compress_trellis(avctx, &samples_p[ch][1],
514  buf + ch * blocks * 8, &c->status[ch],
515  blocks * 8, 1);
516  }
517  for (i = 0; i < blocks; i++) {
518  for (ch = 0; ch < avctx->channels; ch++) {
519  uint8_t *buf1 = buf + ch * blocks * 8 + i * 8;
520  for (j = 0; j < 8; j += 2)
521  *dst++ = buf1[j] | (buf1[j + 1] << 4);
522  }
523  }
524  av_free(buf);
525  } else {
526  for (i = 0; i < blocks; i++) {
527  for (ch = 0; ch < avctx->channels; ch++) {
528  ADPCMChannelStatus *status = &c->status[ch];
529  const int16_t *smp = &samples_p[ch][1 + i * 8];
530  for (j = 0; j < 8; j += 2) {
531  uint8_t v = adpcm_ima_compress_sample(status, smp[j ]);
532  v |= adpcm_ima_compress_sample(status, smp[j + 1]) << 4;
533  *dst++ = v;
534  }
535  }
536  }
537  }
538  break;
539  }
541  {
542  PutBitContext pb;
543  init_put_bits(&pb, dst, pkt_size * 8);
544 
545  for (ch = 0; ch < avctx->channels; ch++) {
546  ADPCMChannelStatus *status = &c->status[ch];
547  put_bits(&pb, 9, (status->prev_sample & 0xFFFF) >> 7);
548  put_bits(&pb, 7, status->step_index);
549  if (avctx->trellis > 0) {
550  uint8_t buf[64];
551  adpcm_compress_trellis(avctx, &samples_p[ch][0], buf, status,
552  64, 1);
553  for (i = 0; i < 64; i++)
554  put_bits(&pb, 4, buf[i ^ 1]);
555  status->prev_sample = status->predictor;
556  } else {
557  for (i = 0; i < 64; i += 2) {
558  int t1, t2;
559  t1 = adpcm_ima_qt_compress_sample(status, samples_p[ch][i ]);
560  t2 = adpcm_ima_qt_compress_sample(status, samples_p[ch][i + 1]);
561  put_bits(&pb, 4, t2);
562  put_bits(&pb, 4, t1);
563  }
564  }
565  }
566 
567  flush_put_bits(&pb);
568  break;
569  }
571  {
572  PutBitContext pb;
573  init_put_bits(&pb, dst, pkt_size * 8);
574 
575  n = frame->nb_samples - 1;
576 
577  // store AdpcmCodeSize
578  put_bits(&pb, 2, 2); // set 4-bit flash adpcm format
579 
580  // init the encoder state
581  for (i = 0; i < avctx->channels; i++) {
582  // clip step so it fits 6 bits
583  c->status[i].step_index = av_clip(c->status[i].step_index, 0, 63);
584  put_sbits(&pb, 16, samples[i]);
585  put_bits(&pb, 6, c->status[i].step_index);
586  c->status[i].prev_sample = samples[i];
587  }
588 
589  if (avctx->trellis > 0) {
590  FF_ALLOC_OR_GOTO(avctx, buf, 2 * n, error);
591  adpcm_compress_trellis(avctx, samples + avctx->channels, buf,
592  &c->status[0], n, avctx->channels);
593  if (avctx->channels == 2)
594  adpcm_compress_trellis(avctx, samples + avctx->channels + 1,
595  buf + n, &c->status[1], n,
596  avctx->channels);
597  for (i = 0; i < n; i++) {
598  put_bits(&pb, 4, buf[i]);
599  if (avctx->channels == 2)
600  put_bits(&pb, 4, buf[n + i]);
601  }
602  av_free(buf);
603  } else {
604  for (i = 1; i < frame->nb_samples; i++) {
606  samples[avctx->channels * i]));
607  if (avctx->channels == 2)
609  samples[2 * i + 1]));
610  }
611  }
612  flush_put_bits(&pb);
613  break;
614  }
616  for (i = 0; i < avctx->channels; i++) {
617  int predictor = 0;
618  *dst++ = predictor;
621  }
622  for (i = 0; i < avctx->channels; i++) {
623  if (c->status[i].idelta < 16)
624  c->status[i].idelta = 16;
625  bytestream_put_le16(&dst, c->status[i].idelta);
626  }
627  for (i = 0; i < avctx->channels; i++)
628  c->status[i].sample2= *samples++;
629  for (i = 0; i < avctx->channels; i++) {
630  c->status[i].sample1 = *samples++;
631  bytestream_put_le16(&dst, c->status[i].sample1);
632  }
633  for (i = 0; i < avctx->channels; i++)
634  bytestream_put_le16(&dst, c->status[i].sample2);
635 
636  if (avctx->trellis > 0) {
637  n = avctx->block_align - 7 * avctx->channels;
638  FF_ALLOC_OR_GOTO(avctx, buf, 2 * n, error);
639  if (avctx->channels == 1) {
640  adpcm_compress_trellis(avctx, samples, buf, &c->status[0], n,
641  avctx->channels);
642  for (i = 0; i < n; i += 2)
643  *dst++ = (buf[i] << 4) | buf[i + 1];
644  } else {
645  adpcm_compress_trellis(avctx, samples, buf,
646  &c->status[0], n, avctx->channels);
647  adpcm_compress_trellis(avctx, samples + 1, buf + n,
648  &c->status[1], n, avctx->channels);
649  for (i = 0; i < n; i++)
650  *dst++ = (buf[i] << 4) | buf[n + i];
651  }
652  av_free(buf);
653  } else {
654  for (i = 7 * avctx->channels; i < avctx->block_align; i++) {
655  int nibble;
656  nibble = adpcm_ms_compress_sample(&c->status[ 0], *samples++) << 4;
657  nibble |= adpcm_ms_compress_sample(&c->status[st], *samples++);
658  *dst++ = nibble;
659  }
660  }
661  break;
663  n = frame->nb_samples / 2;
664  if (avctx->trellis > 0) {
665  FF_ALLOC_OR_GOTO(avctx, buf, 2 * n * 2, error);
666  n *= 2;
667  if (avctx->channels == 1) {
668  adpcm_compress_trellis(avctx, samples, buf, &c->status[0], n,
669  avctx->channels);
670  for (i = 0; i < n; i += 2)
671  *dst++ = buf[i] | (buf[i + 1] << 4);
672  } else {
673  adpcm_compress_trellis(avctx, samples, buf,
674  &c->status[0], n, avctx->channels);
675  adpcm_compress_trellis(avctx, samples + 1, buf + n,
676  &c->status[1], n, avctx->channels);
677  for (i = 0; i < n; i++)
678  *dst++ = buf[i] | (buf[n + i] << 4);
679  }
680  av_free(buf);
681  } else
682  for (n *= avctx->channels; n > 0; n--) {
683  int nibble;
684  nibble = adpcm_yamaha_compress_sample(&c->status[ 0], *samples++);
685  nibble |= adpcm_yamaha_compress_sample(&c->status[st], *samples++) << 4;
686  *dst++ = nibble;
687  }
688  break;
689  default:
690  return AVERROR(EINVAL);
691  }
692 
693  avpkt->size = pkt_size;
694  *got_packet_ptr = 1;
695  return 0;
696 error:
697  return AVERROR(ENOMEM);
698 }
699 
700 static const enum AVSampleFormat sample_fmts[] = {
702 };
703 
704 static const enum AVSampleFormat sample_fmts_p[] = {
706 };
707 
708 #define ADPCM_ENCODER(id_, name_, sample_fmts_, long_name_) \
709 AVCodec ff_ ## name_ ## _encoder = { \
710  .name = #name_, \
711  .long_name = NULL_IF_CONFIG_SMALL(long_name_), \
712  .type = AVMEDIA_TYPE_AUDIO, \
713  .id = id_, \
714  .priv_data_size = sizeof(ADPCMEncodeContext), \
715  .init = adpcm_encode_init, \
716  .encode2 = adpcm_encode_frame, \
717  .close = adpcm_encode_close, \
718  .sample_fmts = sample_fmts_, \
719 }
720 
721 ADPCM_ENCODER(AV_CODEC_ID_ADPCM_IMA_QT, adpcm_ima_qt, sample_fmts_p, "ADPCM IMA QuickTime");
722 ADPCM_ENCODER(AV_CODEC_ID_ADPCM_IMA_WAV, adpcm_ima_wav, sample_fmts_p, "ADPCM IMA WAV");
723 ADPCM_ENCODER(AV_CODEC_ID_ADPCM_MS, adpcm_ms, sample_fmts, "ADPCM Microsoft");
724 ADPCM_ENCODER(AV_CODEC_ID_ADPCM_SWF, adpcm_swf, sample_fmts, "ADPCM Shockwave Flash");
725 ADPCM_ENCODER(AV_CODEC_ID_ADPCM_YAMAHA, adpcm_yamaha, sample_fmts, "ADPCM Yamaha");
const struct AVCodec * codec
Definition: avcodec.h:1418
void * av_malloc(size_t size)
Allocate a block of size bytes with alignment suitable for all memory accesses (including vectors if ...
Definition: mem.c:62
int sample1
Definition: adpcmenc.c:46
int path
Definition: adpcmenc.c:45
static av_cold int adpcm_encode_init(AVCodecContext *avctx)
Definition: adpcmenc.c:61
This structure describes decoded (raw) audio or video data.
Definition: frame.h:140
static void put_sbits(PutBitContext *pb, int n, int32_t value)
Definition: put_bits.h:172
Definition: vf_drawbox.c:37
static uint8_t adpcm_ms_compress_sample(ADPCMChannelStatus *c, int16_t sample)
Definition: adpcmenc.c:212
#define BLKSIZE
Definition: adpcm.h:31
int size
Definition: avcodec.h:1347
av_log(ac->avr, AV_LOG_TRACE, "%d samples - audio_convert: %s to %s (%s)\, len, av_get_sample_fmt_name(ac->in_fmt), av_get_sample_fmt_name(ac->out_fmt), use_generic ? ac->func_descr_generic :ac->func_descr)
static uint8_t adpcm_ima_qt_compress_sample(ADPCMChannelStatus *c, int16_t sample)
Definition: adpcmenc.c:178
static av_cold int adpcm_encode_close(AVCodecContext *avctx)
Definition: adpcmenc.c:153
#define sample
int stride
Definition: mace.c:144
int block_align
number of bytes per packet if constant and known or 0 Used by some WAV based audio codecs...
Definition: avcodec.h:2189
void av_freep(void *arg)
Free a memory block which has been allocated with av_malloc(z)() or av_realloc() and set the pointer ...
Definition: mem.c:202
uint8_t * trellis_hash
Definition: adpcmenc.c:56
static uint8_t adpcm_yamaha_compress_sample(ADPCMChannelStatus *c, int16_t sample)
Definition: adpcmenc.c:241
const uint8_t ff_adpcm_AdaptCoeff1[]
Divided by 4 to fit in 8-bit integers.
Definition: adpcm_data.c:61
uint8_t
#define av_cold
Definition: attributes.h:66
float delta
uint8_t * extradata
some codecs need / can use extradata like Huffman tables.
Definition: avcodec.h:1523
ADPCM tables.
uint8_t * data
Definition: avcodec.h:1346
int bits_per_coded_sample
bits per sample/pixel from the demuxer (needed for huffyuv).
Definition: avcodec.h:2769
uint8_t hash[HASH_SIZE]
Definition: movenc.c:57
static void predictor(uint8_t *src, int size)
Definition: exr.c:220
uint32_t ssd
Definition: adpcmenc.c:44
enum AVCodecID id
Definition: avcodec.h:3134
#define AV_LOG_ERROR
Something went wrong and cannot losslessly be recovered.
Definition: log.h:124
int av_get_bits_per_sample(enum AVCodecID codec_id)
Return codec bits per sample.
Definition: utils.c:2348
void av_free(void *ptr)
Free a memory block which has been allocated with av_malloc(z)() or av_realloc(). ...
Definition: mem.c:190
ADPCM encoder/decoder common header.
static const uint16_t mask[17]
Definition: lzw.c:38
#define AVERROR(e)
Definition: error.h:43
#define STORE_NODE(NAME, STEP_INDEX)
const int16_t ff_adpcm_step_table[89]
This is the step table.
Definition: adpcm_data.c:40
int16_t sample2
Definition: adpcm.h:42
static void put_bits(PutBitContext *s, int n, unsigned int value)
Write up to 31 bits into a bitstream.
Definition: put_bits.h:134
#define FFMAX(a, b)
Definition: common.h:64
const int8_t ff_adpcm_index_table[16]
Definition: adpcm_data.c:31
#define FREEZE_INTERVAL
Definition: adpcmenc.c:59
static uint8_t adpcm_ima_compress_sample(ADPCMChannelStatus *c, int16_t sample)
Definition: adpcmenc.c:165
#define FFMIN(a, b)
Definition: common.h:66
TrellisNode ** nodep_buf
Definition: adpcmenc.c:55
const int8_t ff_adpcm_AdaptCoeff2[]
Divided by 4 to fit in 8-bit integers.
Definition: adpcm_data.c:66
static void adpcm_compress_trellis(AVCodecContext *avctx, const int16_t *samples, uint8_t *dst, ADPCMChannelStatus *c, int n, int stride)
Definition: adpcmenc.c:263
int16_t sample1
Definition: adpcm.h:41
static int adpcm_encode_frame(AVCodecContext *avctx, AVPacket *avpkt, const AVFrame *frame, int *got_packet_ptr)
Definition: adpcmenc.c:468
int ff_alloc_packet(AVPacket *avpkt, int size)
Check AVPacket size and/or allocate data.
Definition: utils.c:1211
TrellisPath * paths
Definition: adpcmenc.c:53
int sample2
Definition: adpcmenc.c:47
if(ac->has_optimized_func)
TrellisNode * node_buf
Definition: adpcmenc.c:54
int frame_size
Number of samples per channel in an audio frame.
Definition: avcodec.h:2172
const int16_t ff_adpcm_AdaptationTable[]
Definition: adpcm_data.c:55
Libavcodec external API header.
version
Definition: ffv1enc.c:1091
AVSampleFormat
Audio Sample Formats.
Definition: samplefmt.h:60
enum AVCodecID codec_id
Definition: avcodec.h:1426
int sample_rate
samples per second
Definition: avcodec.h:2152
main external API structure.
Definition: avcodec.h:1409
int nibble
Definition: adpcmenc.c:39
int extradata_size
Definition: avcodec.h:1524
int step
Definition: adpcmenc.c:48
static int step
Definition: avplay.c:247
#define u(width,...)
#define ADPCM_ENCODER(id_, name_, sample_fmts_, long_name_)
Definition: adpcmenc.c:708
uint8_t * data[AV_NUM_DATA_POINTERS]
pointer to the picture/channel planes.
Definition: frame.h:146
const int8_t ff_adpcm_yamaha_difflookup[]
Definition: adpcm_data.c:75
common internal api header.
static void flush_put_bits(PutBitContext *s)
Pad the end of the output stream with zeros.
Definition: put_bits.h:83
const int16_t ff_adpcm_yamaha_indexscale[]
Definition: adpcm_data.c:70
#define FF_ALLOC_OR_GOTO(ctx, p, size, label)
Definition: internal.h:120
signed 16 bits
Definition: samplefmt.h:63
#define LOOP_NODES(NAME, STEP_TABLE, STEP_INDEX)
static void init_put_bits(PutBitContext *s, uint8_t *buffer, int buffer_size)
Initialize the PutBitContext s.
Definition: put_bits.h:48
int trellis
trellis RD quantization
Definition: avcodec.h:2486
#define AV_INPUT_BUFFER_PADDING_SIZE
Required number of additionally allocated bytes at the end of the input bitstream for decoding...
Definition: avcodec.h:638
void * priv_data
Definition: avcodec.h:1451
int channels
number of audio channels
Definition: avcodec.h:2153
static enum AVSampleFormat sample_fmts[]
Definition: adpcmenc.c:700
int16_t step_index
Definition: adpcm.h:35
signed 16 bits, planar
Definition: samplefmt.h:69
uint8_t ** extended_data
pointers to the data planes/channels.
Definition: frame.h:174
ADPCMChannelStatus status[6]
Definition: adpcmenc.c:52
This structure stores compressed data.
Definition: avcodec.h:1323
int nb_samples
number of audio samples (per channel) described by this frame
Definition: frame.h:184
for(j=16;j >0;--j)
static enum AVSampleFormat sample_fmts_p[]
Definition: adpcmenc.c:704
bitstream writer API