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adpcmenc.c
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1 /*
2  * Copyright (c) 2001-2003 The ffmpeg Project
3  *
4  * This file is part of FFmpeg.
5  *
6  * FFmpeg is free software; you can redistribute it and/or
7  * modify it under the terms of the GNU Lesser General Public
8  * License as published by the Free Software Foundation; either
9  * version 2.1 of the License, or (at your option) any later version.
10  *
11  * FFmpeg is distributed in the hope that it will be useful,
12  * but WITHOUT ANY WARRANTY; without even the implied warranty of
13  * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
14  * Lesser General Public License for more details.
15  *
16  * You should have received a copy of the GNU Lesser General Public
17  * License along with FFmpeg; if not, write to the Free Software
18  * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
19  */
20 
21 #include "avcodec.h"
22 #include "get_bits.h"
23 #include "put_bits.h"
24 #include "bytestream.h"
25 #include "adpcm.h"
26 #include "adpcm_data.h"
27 #include "internal.h"
28 
29 /**
30  * @file
31  * ADPCM encoders
32  * First version by Francois Revol (revol@free.fr)
33  * Fringe ADPCM codecs (e.g., DK3, DK4, Westwood)
34  * by Mike Melanson (melanson@pcisys.net)
35  *
36  * See ADPCM decoder reference documents for codec information.
37  */
38 
39 typedef struct TrellisPath {
40  int nibble;
41  int prev;
42 } TrellisPath;
43 
44 typedef struct TrellisNode {
45  uint32_t ssd;
46  int path;
47  int sample1;
48  int sample2;
49  int step;
50 } TrellisNode;
51 
52 typedef struct ADPCMEncodeContext {
53  ADPCMChannelStatus status[6];
54  TrellisPath *paths;
55  TrellisNode *node_buf;
56  TrellisNode **nodep_buf;
57  uint8_t *trellis_hash;
58 } ADPCMEncodeContext;
59 
60 #define FREEZE_INTERVAL 128
61 
62 static av_cold int adpcm_encode_close(AVCodecContext *avctx);
63 
64 static av_cold int adpcm_encode_init(AVCodecContext *avctx)
65 {
66  ADPCMEncodeContext *s = avctx->priv_data;
67  uint8_t *extradata;
68  int i;
69  int ret = AVERROR(ENOMEM);
70 
71  if (avctx->channels > 2) {
72  av_log(avctx, AV_LOG_ERROR, "only stereo or mono is supported\n");
73  return AVERROR(EINVAL);
74  }
75 
76  if (avctx->trellis && (unsigned)avctx->trellis > 16U) {
77  av_log(avctx, AV_LOG_ERROR, "invalid trellis size\n");
78  return AVERROR(EINVAL);
79  }
80 
81  if (avctx->trellis) {
82  int frontier = 1 << avctx->trellis;
83  int max_paths = frontier * FREEZE_INTERVAL;
84  FF_ALLOC_OR_GOTO(avctx, s->paths,
85  max_paths * sizeof(*s->paths), error);
86  FF_ALLOC_OR_GOTO(avctx, s->node_buf,
87  2 * frontier * sizeof(*s->node_buf), error);
88  FF_ALLOC_OR_GOTO(avctx, s->nodep_buf,
89  2 * frontier * sizeof(*s->nodep_buf), error);
90  FF_ALLOC_OR_GOTO(avctx, s->trellis_hash,
91  65536 * sizeof(*s->trellis_hash), error);
92  }
93 
94  avctx->bits_per_coded_sample = av_get_bits_per_sample(avctx->codec->id);
95 
96  switch (avctx->codec->id) {
97  case AV_CODEC_ID_ADPCM_IMA_WAV:
98  /* each 16 bits sample gives one nibble
99  and we have 4 bytes per channel overhead */
100  avctx->frame_size = (BLKSIZE - 4 * avctx->channels) * 8 /
101  (4 * avctx->channels) + 1;
102  /* seems frame_size isn't taken into account...
103  have to buffer the samples :-( */
104  avctx->block_align = BLKSIZE;
105  avctx->bits_per_coded_sample = 4;
106  break;
107  case AV_CODEC_ID_ADPCM_IMA_QT:
108  avctx->frame_size = 64;
109  avctx->block_align = 34 * avctx->channels;
110  break;
111  case AV_CODEC_ID_ADPCM_MS:
112  /* each 16 bits sample gives one nibble
113  and we have 7 bytes per channel overhead */
114  avctx->frame_size = (BLKSIZE - 7 * avctx->channels) * 2 / avctx->channels + 2;
115  avctx->bits_per_coded_sample = 4;
116  avctx->block_align = BLKSIZE;
117  if (!(avctx->extradata = av_malloc(32 + FF_INPUT_BUFFER_PADDING_SIZE)))
118  goto error;
119  avctx->extradata_size = 32;
120  extradata = avctx->extradata;
121  bytestream_put_le16(&extradata, avctx->frame_size);
122  bytestream_put_le16(&extradata, 7); /* wNumCoef */
123  for (i = 0; i < 7; i++) {
124  bytestream_put_le16(&extradata, ff_adpcm_AdaptCoeff1[i] * 4);
125  bytestream_put_le16(&extradata, ff_adpcm_AdaptCoeff2[i] * 4);
126  }
127  break;
128  case AV_CODEC_ID_ADPCM_YAMAHA:
129  avctx->frame_size = BLKSIZE * 2 / avctx->channels;
130  avctx->block_align = BLKSIZE;
131  break;
132  case AV_CODEC_ID_ADPCM_SWF:
133  if (avctx->sample_rate != 11025 &&
134  avctx->sample_rate != 22050 &&
135  avctx->sample_rate != 44100) {
136  av_log(avctx, AV_LOG_ERROR, "Sample rate must be 11025, "
137  "22050 or 44100\n");
138  ret = AVERROR(EINVAL);
139  goto error;
140  }
141  avctx->frame_size = 512 * (avctx->sample_rate / 11025);
142  break;
143  default:
144  ret = AVERROR(EINVAL);
145  goto error;
146  }
147 
148 #if FF_API_OLD_ENCODE_AUDIO
149  if (!(avctx->coded_frame = avcodec_alloc_frame()))
150  goto error;
151 #endif
152 
153  return 0;
154 error:
155  adpcm_encode_close(avctx);
156  return ret;
157 }
158 
159 static av_cold int adpcm_encode_close(AVCodecContext *avctx)
160 {
161  ADPCMEncodeContext *s = avctx->priv_data;
162 #if FF_API_OLD_ENCODE_AUDIO
163  av_freep(&avctx->coded_frame);
164 #endif
165  av_freep(&s->paths);
166  av_freep(&s->node_buf);
167  av_freep(&s->nodep_buf);
168  av_freep(&s->trellis_hash);
169 
170  return 0;
171 }
172 
173 
174 static inline uint8_t adpcm_ima_compress_sample(ADPCMChannelStatus *c,
175  int16_t sample)
176 {
177  int delta = sample - c->prev_sample;
178  int nibble = FFMIN(7, abs(delta) * 4 /
179  ff_adpcm_step_table[c->step_index]) + (delta < 0) * 8;
180  c->prev_sample += ((ff_adpcm_step_table[c->step_index] *
181  ff_adpcm_yamaha_difflookup[nibble]) / 8);
182  c->prev_sample = av_clip_int16(c->prev_sample);
183  c->step_index = av_clip(c->step_index + ff_adpcm_index_table[nibble], 0, 88);
184  return nibble;
185 }
186 
187 static inline uint8_t adpcm_ima_qt_compress_sample(ADPCMChannelStatus *c,
188  int16_t sample)
189 {
190  int delta = sample - c->prev_sample;
191  int diff, step = ff_adpcm_step_table[c->step_index];
192  int nibble = 8*(delta < 0);
193 
194  delta= abs(delta);
195  diff = delta + (step >> 3);
196 
197  if (delta >= step) {
198  nibble |= 4;
199  delta -= step;
200  }
201  step >>= 1;
202  if (delta >= step) {
203  nibble |= 2;
204  delta -= step;
205  }
206  step >>= 1;
207  if (delta >= step) {
208  nibble |= 1;
209  delta -= step;
210  }
211  diff -= delta;
212 
213  if (nibble & 8)
214  c->prev_sample -= diff;
215  else
216  c->prev_sample += diff;
217 
218  c->prev_sample = av_clip_int16(c->prev_sample);
219  c->step_index = av_clip(c->step_index + ff_adpcm_index_table[nibble], 0, 88);
220 
221  return nibble;
222 }
223 
224 static inline uint8_t adpcm_ms_compress_sample(ADPCMChannelStatus *c,
225  int16_t sample)
226 {
227  int predictor, nibble, bias;
228 
229  predictor = (((c->sample1) * (c->coeff1)) +
230  (( c->sample2) * (c->coeff2))) / 64;
231 
232  nibble = sample - predictor;
233  if (nibble >= 0)
234  bias = c->idelta / 2;
235  else
236  bias = -c->idelta / 2;
237 
238  nibble = (nibble + bias) / c->idelta;
239  nibble = av_clip(nibble, -8, 7) & 0x0F;
240 
241  predictor += ((nibble & 0x08) ? (nibble - 0x10) : nibble) * c->idelta;
242 
243  c->sample2 = c->sample1;
244  c->sample1 = av_clip_int16(predictor);
245 
246  c->idelta = (ff_adpcm_AdaptationTable[nibble] * c->idelta) >> 8;
247  if (c->idelta < 16)
248  c->idelta = 16;
249 
250  return nibble;
251 }
252 
253 static inline uint8_t adpcm_yamaha_compress_sample(ADPCMChannelStatus *c,
254  int16_t sample)
255 {
256  int nibble, delta;
257 
258  if (!c->step) {
259  c->predictor = 0;
260  c->step = 127;
261  }
262 
263  delta = sample - c->predictor;
264 
265  nibble = FFMIN(7, abs(delta) * 4 / c->step) + (delta < 0) * 8;
266 
267  c->predictor += ((c->step * ff_adpcm_yamaha_difflookup[nibble]) / 8);
268  c->predictor = av_clip_int16(c->predictor);
269  c->step = (c->step * ff_adpcm_yamaha_indexscale[nibble]) >> 8;
270  c->step = av_clip(c->step, 127, 24567);
271 
272  return nibble;
273 }
274 
275 static void adpcm_compress_trellis(AVCodecContext *avctx,
276  const int16_t *samples, uint8_t *dst,
277  ADPCMChannelStatus *c, int n, int stride)
278 {
279  //FIXME 6% faster if frontier is a compile-time constant
280  ADPCMEncodeContext *s = avctx->priv_data;
281  const int frontier = 1 << avctx->trellis;
282  const int version = avctx->codec->id;
283  TrellisPath *paths = s->paths, *p;
284  TrellisNode *node_buf = s->node_buf;
285  TrellisNode **nodep_buf = s->nodep_buf;
286  TrellisNode **nodes = nodep_buf; // nodes[] is always sorted by .ssd
287  TrellisNode **nodes_next = nodep_buf + frontier;
288  int pathn = 0, froze = -1, i, j, k, generation = 0;
289  uint8_t *hash = s->trellis_hash;
290  memset(hash, 0xff, 65536 * sizeof(*hash));
291 
292  memset(nodep_buf, 0, 2 * frontier * sizeof(*nodep_buf));
293  nodes[0] = node_buf + frontier;
294  nodes[0]->ssd = 0;
295  nodes[0]->path = 0;
296  nodes[0]->step = c->step_index;
297  nodes[0]->sample1 = c->sample1;
298  nodes[0]->sample2 = c->sample2;
299  if (version == AV_CODEC_ID_ADPCM_IMA_WAV ||
300  version == AV_CODEC_ID_ADPCM_IMA_QT ||
301  version == AV_CODEC_ID_ADPCM_SWF)
302  nodes[0]->sample1 = c->prev_sample;
303  if (version == AV_CODEC_ID_ADPCM_MS)
304  nodes[0]->step = c->idelta;
305  if (version == AV_CODEC_ID_ADPCM_YAMAHA) {
306  if (c->step == 0) {
307  nodes[0]->step = 127;
308  nodes[0]->sample1 = 0;
309  } else {
310  nodes[0]->step = c->step;
311  nodes[0]->sample1 = c->predictor;
312  }
313  }
314 
315  for (i = 0; i < n; i++) {
316  TrellisNode *t = node_buf + frontier*(i&1);
317  TrellisNode **u;
318  int sample = samples[i * stride];
319  int heap_pos = 0;
320  memset(nodes_next, 0, frontier * sizeof(TrellisNode*));
321  for (j = 0; j < frontier && nodes[j]; j++) {
322  // higher j have higher ssd already, so they're likely
323  // to yield a suboptimal next sample too
324  const int range = (j < frontier / 2) ? 1 : 0;
325  const int step = nodes[j]->step;
326  int nidx;
327  if (version == AV_CODEC_ID_ADPCM_MS) {
328  const int predictor = ((nodes[j]->sample1 * c->coeff1) +
329  (nodes[j]->sample2 * c->coeff2)) / 64;
330  const int div = (sample - predictor) / step;
331  const int nmin = av_clip(div-range, -8, 6);
332  const int nmax = av_clip(div+range, -7, 7);
333  for (nidx = nmin; nidx <= nmax; nidx++) {
334  const int nibble = nidx & 0xf;
335  int dec_sample = predictor + nidx * step;
336 #define STORE_NODE(NAME, STEP_INDEX)\
337  int d;\
338  uint32_t ssd;\
339  int pos;\
340  TrellisNode *u;\
341  uint8_t *h;\
342  dec_sample = av_clip_int16(dec_sample);\
343  d = sample - dec_sample;\
344  ssd = nodes[j]->ssd + d*d;\
345  /* Check for wraparound, skip such samples completely. \
346  * Note, changing ssd to a 64 bit variable would be \
347  * simpler, avoiding this check, but it's slower on \
348  * x86 32 bit at the moment. */\
349  if (ssd < nodes[j]->ssd)\
350  goto next_##NAME;\
351  /* Collapse any two states with the same previous sample value. \
352  * One could also distinguish states by step and by 2nd to last
353  * sample, but the effects of that are negligible.
354  * Since nodes in the previous generation are iterated
355  * through a heap, they're roughly ordered from better to
356  * worse, but not strictly ordered. Therefore, an earlier
357  * node with the same sample value is better in most cases
358  * (and thus the current is skipped), but not strictly
359  * in all cases. Only skipping samples where ssd >=
360  * ssd of the earlier node with the same sample gives
361  * slightly worse quality, though, for some reason. */ \
362  h = &hash[(uint16_t) dec_sample];\
363  if (*h == generation)\
364  goto next_##NAME;\
365  if (heap_pos < frontier) {\
366  pos = heap_pos++;\
367  } else {\
368  /* Try to replace one of the leaf nodes with the new \
369  * one, but try a different slot each time. */\
370  pos = (frontier >> 1) +\
371  (heap_pos & ((frontier >> 1) - 1));\
372  if (ssd > nodes_next[pos]->ssd)\
373  goto next_##NAME;\
374  heap_pos++;\
375  }\
376  *h = generation;\
377  u = nodes_next[pos];\
378  if (!u) {\
379  av_assert1(pathn < FREEZE_INTERVAL << avctx->trellis);\
380  u = t++;\
381  nodes_next[pos] = u;\
382  u->path = pathn++;\
383  }\
384  u->ssd = ssd;\
385  u->step = STEP_INDEX;\
386  u->sample2 = nodes[j]->sample1;\
387  u->sample1 = dec_sample;\
388  paths[u->path].nibble = nibble;\
389  paths[u->path].prev = nodes[j]->path;\
390  /* Sift the newly inserted node up in the heap to \
391  * restore the heap property. */\
392  while (pos > 0) {\
393  int parent = (pos - 1) >> 1;\
394  if (nodes_next[parent]->ssd <= ssd)\
395  break;\
396  FFSWAP(TrellisNode*, nodes_next[parent], nodes_next[pos]);\
397  pos = parent;\
398  }\
399  next_##NAME:;
400  STORE_NODE(ms, FFMAX(16,
401  (ff_adpcm_AdaptationTable[nibble] * step) >> 8));
402  }
403  } else if (version == AV_CODEC_ID_ADPCM_IMA_WAV ||
404  version == AV_CODEC_ID_ADPCM_IMA_QT ||
405  version == AV_CODEC_ID_ADPCM_SWF) {
406 #define LOOP_NODES(NAME, STEP_TABLE, STEP_INDEX)\
407  const int predictor = nodes[j]->sample1;\
408  const int div = (sample - predictor) * 4 / STEP_TABLE;\
409  int nmin = av_clip(div - range, -7, 6);\
410  int nmax = av_clip(div + range, -6, 7);\
411  if (nmin <= 0)\
412  nmin--; /* distinguish -0 from +0 */\
413  if (nmax < 0)\
414  nmax--;\
415  for (nidx = nmin; nidx <= nmax; nidx++) {\
416  const int nibble = nidx < 0 ? 7 - nidx : nidx;\
417  int dec_sample = predictor +\
418  (STEP_TABLE *\
419  ff_adpcm_yamaha_difflookup[nibble]) / 8;\
420  STORE_NODE(NAME, STEP_INDEX);\
421  }
422  LOOP_NODES(ima, ff_adpcm_step_table[step],
423  av_clip(step + ff_adpcm_index_table[nibble], 0, 88));
424  } else { //AV_CODEC_ID_ADPCM_YAMAHA
425  LOOP_NODES(yamaha, step,
426  av_clip((step * ff_adpcm_yamaha_indexscale[nibble]) >> 8,
427  127, 24567));
428 #undef LOOP_NODES
429 #undef STORE_NODE
430  }
431  }
432 
433  u = nodes;
434  nodes = nodes_next;
435  nodes_next = u;
436 
437  generation++;
438  if (generation == 255) {
439  memset(hash, 0xff, 65536 * sizeof(*hash));
440  generation = 0;
441  }
442 
443  // prevent overflow
444  if (nodes[0]->ssd > (1 << 28)) {
445  for (j = 1; j < frontier && nodes[j]; j++)
446  nodes[j]->ssd -= nodes[0]->ssd;
447  nodes[0]->ssd = 0;
448  }
449 
450  // merge old paths to save memory
451  if (i == froze + FREEZE_INTERVAL) {
452  p = &paths[nodes[0]->path];
453  for (k = i; k > froze; k--) {
454  dst[k] = p->nibble;
455  p = &paths[p->prev];
456  }
457  froze = i;
458  pathn = 0;
459  // other nodes might use paths that don't coincide with the frozen one.
460  // checking which nodes do so is too slow, so just kill them all.
461  // this also slightly improves quality, but I don't know why.
462  memset(nodes + 1, 0, (frontier - 1) * sizeof(TrellisNode*));
463  }
464  }
465 
466  p = &paths[nodes[0]->path];
467  for (i = n - 1; i > froze; i--) {
468  dst[i] = p->nibble;
469  p = &paths[p->prev];
470  }
471 
472  c->predictor = nodes[0]->sample1;
473  c->sample1 = nodes[0]->sample1;
474  c->sample2 = nodes[0]->sample2;
475  c->step_index = nodes[0]->step;
476  c->step = nodes[0]->step;
477  c->idelta = nodes[0]->step;
478 }
479 
480 static int adpcm_encode_frame(AVCodecContext *avctx, AVPacket *avpkt,
481  const AVFrame *frame, int *got_packet_ptr)
482 {
483  int n, i, ch, st, pkt_size, ret;
484  const int16_t *samples;
485  int16_t **samples_p;
486  uint8_t *dst;
487  ADPCMEncodeContext *c = avctx->priv_data;
488  uint8_t *buf;
489 
490  samples = (const int16_t *)frame->data[0];
491  samples_p = (int16_t **)frame->extended_data;
492  st = avctx->channels == 2;
493 
494  if (avctx->codec_id == AV_CODEC_ID_ADPCM_SWF)
495  pkt_size = (2 + avctx->channels * (22 + 4 * (frame->nb_samples - 1)) + 7) / 8;
496  else
497  pkt_size = avctx->block_align;
498  if ((ret = ff_alloc_packet2(avctx, avpkt, pkt_size)))
499  return ret;
500  dst = avpkt->data;
501 
502  switch(avctx->codec->id) {
503  case AV_CODEC_ID_ADPCM_IMA_WAV:
504  {
505  int blocks, j;
506 
507  blocks = (frame->nb_samples - 1) / 8;
508 
509  for (ch = 0; ch < avctx->channels; ch++) {
510  ADPCMChannelStatus *status = &c->status[ch];
511  status->prev_sample = samples_p[ch][0];
512  /* status->step_index = 0;
513  XXX: not sure how to init the state machine */
514  bytestream_put_le16(&dst, status->prev_sample);
515  *dst++ = status->step_index;
516  *dst++ = 0; /* unknown */
517  }
518 
519  /* stereo: 4 bytes (8 samples) for left, 4 bytes for right */
520  if (avctx->trellis > 0) {
521  FF_ALLOC_OR_GOTO(avctx, buf, avctx->channels * blocks * 8, error);
522  for (ch = 0; ch < avctx->channels; ch++) {
523  adpcm_compress_trellis(avctx, &samples_p[ch][1],
524  buf + ch * blocks * 8, &c->status[ch],
525  blocks * 8, 1);
526  }
527  for (i = 0; i < blocks; i++) {
528  for (ch = 0; ch < avctx->channels; ch++) {
529  uint8_t *buf1 = buf + ch * blocks * 8 + i * 8;
530  for (j = 0; j < 8; j += 2)
531  *dst++ = buf1[j] | (buf1[j + 1] << 4);
532  }
533  }
534  av_free(buf);
535  } else {
536  for (i = 0; i < blocks; i++) {
537  for (ch = 0; ch < avctx->channels; ch++) {
538  ADPCMChannelStatus *status = &c->status[ch];
539  const int16_t *smp = &samples_p[ch][1 + i * 8];
540  for (j = 0; j < 8; j += 2) {
541  uint8_t v = adpcm_ima_compress_sample(status, smp[j ]);
542  v |= adpcm_ima_compress_sample(status, smp[j + 1]) << 4;
543  *dst++ = v;
544  }
545  }
546  }
547  }
548  break;
549  }
550  case AV_CODEC_ID_ADPCM_IMA_QT:
551  {
552  PutBitContext pb;
553  init_put_bits(&pb, dst, pkt_size * 8);
554 
555  for (ch = 0; ch < avctx->channels; ch++) {
556  ADPCMChannelStatus *status = &c->status[ch];
557  put_bits(&pb, 9, (status->prev_sample & 0xFFFF) >> 7);
558  put_bits(&pb, 7, status->step_index);
559  if (avctx->trellis > 0) {
560  uint8_t buf[64];
561  adpcm_compress_trellis(avctx, &samples_p[ch][1], buf, status,
562  64, 1);
563  for (i = 0; i < 64; i++)
564  put_bits(&pb, 4, buf[i ^ 1]);
565  } else {
566  for (i = 0; i < 64; i += 2) {
567  int t1, t2;
568  t1 = adpcm_ima_qt_compress_sample(status, samples_p[ch][i ]);
569  t2 = adpcm_ima_qt_compress_sample(status, samples_p[ch][i + 1]);
570  put_bits(&pb, 4, t2);
571  put_bits(&pb, 4, t1);
572  }
573  }
574  }
575 
576  flush_put_bits(&pb);
577  break;
578  }
579  case AV_CODEC_ID_ADPCM_SWF:
580  {
581  PutBitContext pb;
582  init_put_bits(&pb, dst, pkt_size * 8);
583 
584  n = frame->nb_samples - 1;
585 
586  // store AdpcmCodeSize
587  put_bits(&pb, 2, 2); // set 4-bit flash adpcm format
588 
589  // init the encoder state
590  for (i = 0; i < avctx->channels; i++) {
591  // clip step so it fits 6 bits
592  c->status[i].step_index = av_clip(c->status[i].step_index, 0, 63);
593  put_sbits(&pb, 16, samples[i]);
594  put_bits(&pb, 6, c->status[i].step_index);
595  c->status[i].prev_sample = samples[i];
596  }
597 
598  if (avctx->trellis > 0) {
599  FF_ALLOC_OR_GOTO(avctx, buf, 2 * n, error);
600  adpcm_compress_trellis(avctx, samples + avctx->channels, buf,
601  &c->status[0], n, avctx->channels);
602  if (avctx->channels == 2)
603  adpcm_compress_trellis(avctx, samples + avctx->channels + 1,
604  buf + n, &c->status[1], n,
605  avctx->channels);
606  for (i = 0; i < n; i++) {
607  put_bits(&pb, 4, buf[i]);
608  if (avctx->channels == 2)
609  put_bits(&pb, 4, buf[n + i]);
610  }
611  av_free(buf);
612  } else {
613  for (i = 1; i < frame->nb_samples; i++) {
614  put_bits(&pb, 4, adpcm_ima_compress_sample(&c->status[0],
615  samples[avctx->channels * i]));
616  if (avctx->channels == 2)
617  put_bits(&pb, 4, adpcm_ima_compress_sample(&c->status[1],
618  samples[2 * i + 1]));
619  }
620  }
621  flush_put_bits(&pb);
622  break;
623  }
624  case AV_CODEC_ID_ADPCM_MS:
625  for (i = 0; i < avctx->channels; i++) {
626  int predictor = 0;
627  *dst++ = predictor;
628  c->status[i].coeff1 = ff_adpcm_AdaptCoeff1[predictor];
629  c->status[i].coeff2 = ff_adpcm_AdaptCoeff2[predictor];
630  }
631  for (i = 0; i < avctx->channels; i++) {
632  if (c->status[i].idelta < 16)
633  c->status[i].idelta = 16;
634  bytestream_put_le16(&dst, c->status[i].idelta);
635  }
636  for (i = 0; i < avctx->channels; i++)
637  c->status[i].sample2= *samples++;
638  for (i = 0; i < avctx->channels; i++) {
639  c->status[i].sample1 = *samples++;
640  bytestream_put_le16(&dst, c->status[i].sample1);
641  }
642  for (i = 0; i < avctx->channels; i++)
643  bytestream_put_le16(&dst, c->status[i].sample2);
644 
645  if (avctx->trellis > 0) {
646  n = avctx->block_align - 7 * avctx->channels;
647  FF_ALLOC_OR_GOTO(avctx, buf, 2 * n, error);
648  if (avctx->channels == 1) {
649  adpcm_compress_trellis(avctx, samples, buf, &c->status[0], n,
650  avctx->channels);
651  for (i = 0; i < n; i += 2)
652  *dst++ = (buf[i] << 4) | buf[i + 1];
653  } else {
654  adpcm_compress_trellis(avctx, samples, buf,
655  &c->status[0], n, avctx->channels);
656  adpcm_compress_trellis(avctx, samples + 1, buf + n,
657  &c->status[1], n, avctx->channels);
658  for (i = 0; i < n; i++)
659  *dst++ = (buf[i] << 4) | buf[n + i];
660  }
661  av_free(buf);
662  } else {
663  for (i = 7 * avctx->channels; i < avctx->block_align; i++) {
664  int nibble;
665  nibble = adpcm_ms_compress_sample(&c->status[ 0], *samples++) << 4;
666  nibble |= adpcm_ms_compress_sample(&c->status[st], *samples++);
667  *dst++ = nibble;
668  }
669  }
670  break;
671  case AV_CODEC_ID_ADPCM_YAMAHA:
672  n = frame->nb_samples / 2;
673  if (avctx->trellis > 0) {
674  FF_ALLOC_OR_GOTO(avctx, buf, 2 * n * 2, error);
675  n *= 2;
676  if (avctx->channels == 1) {
677  adpcm_compress_trellis(avctx, samples, buf, &c->status[0], n,
678  avctx->channels);
679  for (i = 0; i < n; i += 2)
680  *dst++ = buf[i] | (buf[i + 1] << 4);
681  } else {
682  adpcm_compress_trellis(avctx, samples, buf,
683  &c->status[0], n, avctx->channels);
684  adpcm_compress_trellis(avctx, samples + 1, buf + n,
685  &c->status[1], n, avctx->channels);
686  for (i = 0; i < n; i++)
687  *dst++ = buf[i] | (buf[n + i] << 4);
688  }
689  av_free(buf);
690  } else
691  for (n *= avctx->channels; n > 0; n--) {
692  int nibble;
693  nibble = adpcm_yamaha_compress_sample(&c->status[ 0], *samples++);
694  nibble |= adpcm_yamaha_compress_sample(&c->status[st], *samples++) << 4;
695  *dst++ = nibble;
696  }
697  break;
698  default:
699  return AVERROR(EINVAL);
700  }
701 
702  avpkt->size = pkt_size;
703  *got_packet_ptr = 1;
704  return 0;
705 error:
706  return AVERROR(ENOMEM);
707 }
708 
709 static const enum AVSampleFormat sample_fmts[] = {
710  AV_SAMPLE_FMT_S16, AV_SAMPLE_FMT_NONE
711 };
712 
713 static const enum AVSampleFormat sample_fmts_p[] = {
714  AV_SAMPLE_FMT_S16P, AV_SAMPLE_FMT_NONE
715 };
716 
717 #define ADPCM_ENCODER(id_, name_, sample_fmts_, long_name_) \
718 AVCodec ff_ ## name_ ## _encoder = { \
719  .name = #name_, \
720  .type = AVMEDIA_TYPE_AUDIO, \
721  .id = id_, \
722  .priv_data_size = sizeof(ADPCMEncodeContext), \
723  .init = adpcm_encode_init, \
724  .encode2 = adpcm_encode_frame, \
725  .close = adpcm_encode_close, \
726  .sample_fmts = sample_fmts_, \
727  .long_name = NULL_IF_CONFIG_SMALL(long_name_), \
728 }
729 
730 ADPCM_ENCODER(AV_CODEC_ID_ADPCM_IMA_QT, adpcm_ima_qt, sample_fmts_p, "ADPCM IMA QuickTime");
731 ADPCM_ENCODER(AV_CODEC_ID_ADPCM_IMA_WAV, adpcm_ima_wav, sample_fmts_p, "ADPCM IMA WAV");
732 ADPCM_ENCODER(AV_CODEC_ID_ADPCM_MS, adpcm_ms, sample_fmts, "ADPCM Microsoft");
733 ADPCM_ENCODER(AV_CODEC_ID_ADPCM_SWF, adpcm_swf, sample_fmts, "ADPCM Shockwave Flash");
734 ADPCM_ENCODER(AV_CODEC_ID_ADPCM_YAMAHA, adpcm_yamaha, sample_fmts, "ADPCM Yamaha");
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