avformat/mpeg: demux ivtv captions
[ffmpeg.git] / libavcodec / adpcmenc.c
blob4d7534e47fc60ddf069e62fb964fc683a8977746
1 /*
2 * Copyright (c) 2001-2003 The FFmpeg project
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)
8 * This file is part of FFmpeg.
10 * FFmpeg 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.
15 * FFmpeg 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.
20 * You should have received a copy of the GNU Lesser General Public
21 * License along with FFmpeg; if not, write to the Free Software
22 * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
25 #include "config_components.h"
27 #include "libavutil/mem.h"
28 #include "libavutil/opt.h"
30 #include "avcodec.h"
31 #include "put_bits.h"
32 #include "bytestream.h"
33 #include "adpcm.h"
34 #include "adpcm_data.h"
35 #include "codec_internal.h"
36 #include "encode.h"
38 /**
39 * @file
40 * ADPCM encoders
41 * See ADPCM decoder reference documents for codec information.
44 #define CASE_0(codec_id, ...)
45 #define CASE_1(codec_id, ...) \
46 case codec_id: \
47 { __VA_ARGS__ } \
48 break;
49 #define CASE_2(enabled, codec_id, ...) \
50 CASE_ ## enabled(codec_id, __VA_ARGS__)
51 #define CASE_3(config, codec_id, ...) \
52 CASE_2(config, codec_id, __VA_ARGS__)
53 #define CASE(codec, ...) \
54 CASE_3(CONFIG_ ## codec ## _ENCODER, AV_CODEC_ID_ ## codec, __VA_ARGS__)
56 typedef struct TrellisPath {
57 int nibble;
58 int prev;
59 } TrellisPath;
61 typedef struct TrellisNode {
62 uint32_t ssd;
63 int path;
64 int sample1;
65 int sample2;
66 int step;
67 } TrellisNode;
69 typedef struct ADPCMEncodeContext {
70 AVClass *class;
71 int block_size;
73 ADPCMChannelStatus status[6];
74 TrellisPath *paths;
75 TrellisNode *node_buf;
76 TrellisNode **nodep_buf;
77 uint8_t *trellis_hash;
78 } ADPCMEncodeContext;
80 #define FREEZE_INTERVAL 128
82 static av_cold int adpcm_encode_init(AVCodecContext *avctx)
84 ADPCMEncodeContext *s = avctx->priv_data;
85 int channels = avctx->ch_layout.nb_channels;
88 * AMV's block size has to match that of the corresponding video
89 * stream. Relax the POT requirement.
91 if (avctx->codec->id != AV_CODEC_ID_ADPCM_IMA_AMV &&
92 (s->block_size & (s->block_size - 1))) {
93 av_log(avctx, AV_LOG_ERROR, "block size must be power of 2\n");
94 return AVERROR(EINVAL);
97 if (avctx->trellis) {
98 int frontier, max_paths;
100 if ((unsigned)avctx->trellis > 16U) {
101 av_log(avctx, AV_LOG_ERROR, "invalid trellis size\n");
102 return AVERROR(EINVAL);
105 if (avctx->codec->id == AV_CODEC_ID_ADPCM_IMA_SSI ||
106 avctx->codec->id == AV_CODEC_ID_ADPCM_IMA_APM ||
107 avctx->codec->id == AV_CODEC_ID_ADPCM_ARGO ||
108 avctx->codec->id == AV_CODEC_ID_ADPCM_IMA_WS) {
110 * The current trellis implementation doesn't work for extended
111 * runs of samples without periodic resets. Disallow it.
113 av_log(avctx, AV_LOG_ERROR, "trellis not supported\n");
114 return AVERROR_PATCHWELCOME;
117 frontier = 1 << avctx->trellis;
118 max_paths = frontier * FREEZE_INTERVAL;
119 if (!FF_ALLOC_TYPED_ARRAY(s->paths, max_paths) ||
120 !FF_ALLOC_TYPED_ARRAY(s->node_buf, 2 * frontier) ||
121 !FF_ALLOC_TYPED_ARRAY(s->nodep_buf, 2 * frontier) ||
122 !FF_ALLOC_TYPED_ARRAY(s->trellis_hash, 65536))
123 return AVERROR(ENOMEM);
126 avctx->bits_per_coded_sample = av_get_bits_per_sample(avctx->codec->id);
128 switch (avctx->codec->id) {
129 CASE(ADPCM_IMA_WAV,
130 /* each 16 bits sample gives one nibble
131 and we have 4 bytes per channel overhead */
132 avctx->frame_size = (s->block_size - 4 * channels) * 8 /
133 (4 * channels) + 1;
134 /* seems frame_size isn't taken into account...
135 have to buffer the samples :-( */
136 avctx->block_align = s->block_size;
137 avctx->bits_per_coded_sample = 4;
138 ) /* End of CASE */
139 CASE(ADPCM_IMA_QT,
140 avctx->frame_size = 64;
141 avctx->block_align = 34 * channels;
142 ) /* End of CASE */
143 CASE(ADPCM_MS,
144 uint8_t *extradata;
145 /* each 16 bits sample gives one nibble
146 and we have 7 bytes per channel overhead */
147 avctx->frame_size = (s->block_size - 7 * channels) * 2 / channels + 2;
148 avctx->bits_per_coded_sample = 4;
149 avctx->block_align = s->block_size;
150 if (!(avctx->extradata = av_malloc(32 + AV_INPUT_BUFFER_PADDING_SIZE)))
151 return AVERROR(ENOMEM);
152 avctx->extradata_size = 32;
153 extradata = avctx->extradata;
154 bytestream_put_le16(&extradata, avctx->frame_size);
155 bytestream_put_le16(&extradata, 7); /* wNumCoef */
156 for (int i = 0; i < 7; i++) {
157 bytestream_put_le16(&extradata, ff_adpcm_AdaptCoeff1[i] * 4);
158 bytestream_put_le16(&extradata, ff_adpcm_AdaptCoeff2[i] * 4);
160 ) /* End of CASE */
161 CASE(ADPCM_YAMAHA,
162 avctx->frame_size = s->block_size * 2 / channels;
163 avctx->block_align = s->block_size;
164 ) /* End of CASE */
165 CASE(ADPCM_SWF,
166 if (avctx->sample_rate != 11025 &&
167 avctx->sample_rate != 22050 &&
168 avctx->sample_rate != 44100) {
169 av_log(avctx, AV_LOG_ERROR, "Sample rate must be 11025, "
170 "22050 or 44100\n");
171 return AVERROR(EINVAL);
173 avctx->frame_size = 4096; /* Hardcoded according to the SWF spec. */
174 avctx->block_align = (2 + channels * (22 + 4 * (avctx->frame_size - 1)) + 7) / 8;
175 ) /* End of CASE */
176 case AV_CODEC_ID_ADPCM_IMA_SSI:
177 case AV_CODEC_ID_ADPCM_IMA_ALP:
178 avctx->frame_size = s->block_size * 2 / channels;
179 avctx->block_align = s->block_size;
180 break;
181 CASE(ADPCM_IMA_AMV,
182 if (avctx->sample_rate != 22050) {
183 av_log(avctx, AV_LOG_ERROR, "Sample rate must be 22050\n");
184 return AVERROR(EINVAL);
187 if (channels != 1) {
188 av_log(avctx, AV_LOG_ERROR, "Only mono is supported\n");
189 return AVERROR(EINVAL);
192 avctx->frame_size = s->block_size;
193 avctx->block_align = 8 + (FFALIGN(avctx->frame_size, 2) / 2);
194 ) /* End of CASE */
195 CASE(ADPCM_IMA_APM,
196 avctx->frame_size = s->block_size * 2 / channels;
197 avctx->block_align = s->block_size;
199 if (!(avctx->extradata = av_mallocz(28 + AV_INPUT_BUFFER_PADDING_SIZE)))
200 return AVERROR(ENOMEM);
201 avctx->extradata_size = 28;
202 ) /* End of CASE */
203 CASE(ADPCM_ARGO,
204 avctx->frame_size = 32;
205 avctx->block_align = 17 * channels;
206 ) /* End of CASE */
207 CASE(ADPCM_IMA_WS,
208 /* each 16 bits sample gives one nibble */
209 avctx->frame_size = s->block_size * 2 / channels;
210 avctx->block_align = s->block_size;
211 ) /* End of CASE */
212 default:
213 return AVERROR(EINVAL);
216 return 0;
219 static av_cold int adpcm_encode_close(AVCodecContext *avctx)
221 ADPCMEncodeContext *s = avctx->priv_data;
222 av_freep(&s->paths);
223 av_freep(&s->node_buf);
224 av_freep(&s->nodep_buf);
225 av_freep(&s->trellis_hash);
227 return 0;
231 static inline uint8_t adpcm_ima_compress_sample(ADPCMChannelStatus *c,
232 int16_t sample)
234 int delta = sample - c->prev_sample;
235 int nibble = FFMIN(7, abs(delta) * 4 /
236 ff_adpcm_step_table[c->step_index]) + (delta < 0) * 8;
237 c->prev_sample += ((ff_adpcm_step_table[c->step_index] *
238 ff_adpcm_yamaha_difflookup[nibble]) / 8);
239 c->prev_sample = av_clip_int16(c->prev_sample);
240 c->step_index = av_clip(c->step_index + ff_adpcm_index_table[nibble], 0, 88);
241 return nibble;
244 static inline uint8_t adpcm_ima_alp_compress_sample(ADPCMChannelStatus *c, int16_t sample)
246 const int delta = sample - c->prev_sample;
247 const int step = ff_adpcm_step_table[c->step_index];
248 const int sign = (delta < 0) * 8;
250 int nibble = FFMIN(abs(delta) * 4 / step, 7);
251 int diff = (step * nibble) >> 2;
252 if (sign)
253 diff = -diff;
255 nibble = sign | nibble;
257 c->prev_sample += diff;
258 c->prev_sample = av_clip_int16(c->prev_sample);
259 c->step_index = av_clip(c->step_index + ff_adpcm_index_table[nibble], 0, 88);
260 return nibble;
263 static inline uint8_t adpcm_ima_qt_compress_sample(ADPCMChannelStatus *c,
264 int16_t sample)
266 int delta = sample - c->prev_sample;
267 int diff, step = ff_adpcm_step_table[c->step_index];
268 int nibble = 8*(delta < 0);
270 delta= abs(delta);
271 diff = delta + (step >> 3);
273 if (delta >= step) {
274 nibble |= 4;
275 delta -= step;
277 step >>= 1;
278 if (delta >= step) {
279 nibble |= 2;
280 delta -= step;
282 step >>= 1;
283 if (delta >= step) {
284 nibble |= 1;
285 delta -= step;
287 diff -= delta;
289 if (nibble & 8)
290 c->prev_sample -= diff;
291 else
292 c->prev_sample += diff;
294 c->prev_sample = av_clip_int16(c->prev_sample);
295 c->step_index = av_clip(c->step_index + ff_adpcm_index_table[nibble], 0, 88);
297 return nibble;
300 static inline uint8_t adpcm_ms_compress_sample(ADPCMChannelStatus *c,
301 int16_t sample)
303 int predictor, nibble, bias;
305 predictor = (((c->sample1) * (c->coeff1)) +
306 (( c->sample2) * (c->coeff2))) / 64;
308 nibble = sample - predictor;
309 if (nibble >= 0)
310 bias = c->idelta / 2;
311 else
312 bias = -c->idelta / 2;
314 nibble = (nibble + bias) / c->idelta;
315 nibble = av_clip_intp2(nibble, 3) & 0x0F;
317 predictor += ((nibble & 0x08) ? (nibble - 0x10) : nibble) * c->idelta;
319 c->sample2 = c->sample1;
320 c->sample1 = av_clip_int16(predictor);
322 c->idelta = (ff_adpcm_AdaptationTable[nibble] * c->idelta) >> 8;
323 if (c->idelta < 16)
324 c->idelta = 16;
326 return nibble;
329 static inline uint8_t adpcm_yamaha_compress_sample(ADPCMChannelStatus *c,
330 int16_t sample)
332 int nibble, delta;
334 if (!c->step) {
335 c->predictor = 0;
336 c->step = 127;
339 delta = sample - c->predictor;
341 nibble = FFMIN(7, abs(delta) * 4 / c->step) + (delta < 0) * 8;
343 c->predictor += ((c->step * ff_adpcm_yamaha_difflookup[nibble]) / 8);
344 c->predictor = av_clip_int16(c->predictor);
345 c->step = (c->step * ff_adpcm_yamaha_indexscale[nibble]) >> 8;
346 c->step = av_clip(c->step, 127, 24576);
348 return nibble;
351 static void adpcm_compress_trellis(AVCodecContext *avctx,
352 const int16_t *samples, uint8_t *dst,
353 ADPCMChannelStatus *c, int n, int stride)
355 //FIXME 6% faster if frontier is a compile-time constant
356 ADPCMEncodeContext *s = avctx->priv_data;
357 const int frontier = 1 << avctx->trellis;
358 const int version = avctx->codec->id;
359 TrellisPath *paths = s->paths, *p;
360 TrellisNode *node_buf = s->node_buf;
361 TrellisNode **nodep_buf = s->nodep_buf;
362 TrellisNode **nodes = nodep_buf; // nodes[] is always sorted by .ssd
363 TrellisNode **nodes_next = nodep_buf + frontier;
364 int pathn = 0, froze = -1, i, j, k, generation = 0;
365 uint8_t *hash = s->trellis_hash;
366 memset(hash, 0xff, 65536 * sizeof(*hash));
368 memset(nodep_buf, 0, 2 * frontier * sizeof(*nodep_buf));
369 nodes[0] = node_buf + frontier;
370 nodes[0]->ssd = 0;
371 nodes[0]->path = 0;
372 nodes[0]->step = c->step_index;
373 nodes[0]->sample1 = c->sample1;
374 nodes[0]->sample2 = c->sample2;
375 if (version == AV_CODEC_ID_ADPCM_IMA_WAV ||
376 version == AV_CODEC_ID_ADPCM_IMA_QT ||
377 version == AV_CODEC_ID_ADPCM_IMA_AMV ||
378 version == AV_CODEC_ID_ADPCM_SWF)
379 nodes[0]->sample1 = c->prev_sample;
380 if (version == AV_CODEC_ID_ADPCM_MS)
381 nodes[0]->step = c->idelta;
382 if (version == AV_CODEC_ID_ADPCM_YAMAHA) {
383 if (c->step == 0) {
384 nodes[0]->step = 127;
385 nodes[0]->sample1 = 0;
386 } else {
387 nodes[0]->step = c->step;
388 nodes[0]->sample1 = c->predictor;
392 for (i = 0; i < n; i++) {
393 TrellisNode *t = node_buf + frontier*(i&1);
394 TrellisNode **u;
395 int sample = samples[i * stride];
396 int heap_pos = 0;
397 memset(nodes_next, 0, frontier * sizeof(TrellisNode*));
398 for (j = 0; j < frontier && nodes[j]; j++) {
399 // higher j have higher ssd already, so they're likely
400 // to yield a suboptimal next sample too
401 const int range = (j < frontier / 2) ? 1 : 0;
402 const int step = nodes[j]->step;
403 int nidx;
404 if (version == AV_CODEC_ID_ADPCM_MS) {
405 const int predictor = ((nodes[j]->sample1 * c->coeff1) +
406 (nodes[j]->sample2 * c->coeff2)) / 64;
407 const int div = (sample - predictor) / step;
408 const int nmin = av_clip(div-range, -8, 6);
409 const int nmax = av_clip(div+range, -7, 7);
410 for (nidx = nmin; nidx <= nmax; nidx++) {
411 const int nibble = nidx & 0xf;
412 int dec_sample = predictor + nidx * step;
413 #define STORE_NODE(NAME, STEP_INDEX)\
414 int d;\
415 uint32_t ssd;\
416 int pos;\
417 TrellisNode *u;\
418 uint8_t *h;\
419 dec_sample = av_clip_int16(dec_sample);\
420 d = sample - dec_sample;\
421 ssd = nodes[j]->ssd + d*(unsigned)d;\
422 /* Check for wraparound, skip such samples completely. \
423 * Note, changing ssd to a 64 bit variable would be \
424 * simpler, avoiding this check, but it's slower on \
425 * x86 32 bit at the moment. */\
426 if (ssd < nodes[j]->ssd)\
427 goto next_##NAME;\
428 /* Collapse any two states with the same previous sample value. \
429 * One could also distinguish states by step and by 2nd to last
430 * sample, but the effects of that are negligible.
431 * Since nodes in the previous generation are iterated
432 * through a heap, they're roughly ordered from better to
433 * worse, but not strictly ordered. Therefore, an earlier
434 * node with the same sample value is better in most cases
435 * (and thus the current is skipped), but not strictly
436 * in all cases. Only skipping samples where ssd >=
437 * ssd of the earlier node with the same sample gives
438 * slightly worse quality, though, for some reason. */ \
439 h = &hash[(uint16_t) dec_sample];\
440 if (*h == generation)\
441 goto next_##NAME;\
442 if (heap_pos < frontier) {\
443 pos = heap_pos++;\
444 } else {\
445 /* Try to replace one of the leaf nodes with the new \
446 * one, but try a different slot each time. */\
447 pos = (frontier >> 1) +\
448 (heap_pos & ((frontier >> 1) - 1));\
449 if (ssd > nodes_next[pos]->ssd)\
450 goto next_##NAME;\
451 heap_pos++;\
453 *h = generation;\
454 u = nodes_next[pos];\
455 if (!u) {\
456 av_assert1(pathn < FREEZE_INTERVAL << avctx->trellis);\
457 u = t++;\
458 nodes_next[pos] = u;\
459 u->path = pathn++;\
461 u->ssd = ssd;\
462 u->step = STEP_INDEX;\
463 u->sample2 = nodes[j]->sample1;\
464 u->sample1 = dec_sample;\
465 paths[u->path].nibble = nibble;\
466 paths[u->path].prev = nodes[j]->path;\
467 /* Sift the newly inserted node up in the heap to \
468 * restore the heap property. */\
469 while (pos > 0) {\
470 int parent = (pos - 1) >> 1;\
471 if (nodes_next[parent]->ssd <= ssd)\
472 break;\
473 FFSWAP(TrellisNode*, nodes_next[parent], nodes_next[pos]);\
474 pos = parent;\
476 next_##NAME:;
477 STORE_NODE(ms, FFMAX(16,
478 (ff_adpcm_AdaptationTable[nibble] * step) >> 8));
480 } else if (version == AV_CODEC_ID_ADPCM_IMA_WAV ||
481 version == AV_CODEC_ID_ADPCM_IMA_QT ||
482 version == AV_CODEC_ID_ADPCM_IMA_AMV ||
483 version == AV_CODEC_ID_ADPCM_SWF) {
484 #define LOOP_NODES(NAME, STEP_TABLE, STEP_INDEX)\
485 const int predictor = nodes[j]->sample1;\
486 const int div = (sample - predictor) * 4 / STEP_TABLE;\
487 int nmin = av_clip(div - range, -7, 6);\
488 int nmax = av_clip(div + range, -6, 7);\
489 if (nmin <= 0)\
490 nmin--; /* distinguish -0 from +0 */\
491 if (nmax < 0)\
492 nmax--;\
493 for (nidx = nmin; nidx <= nmax; nidx++) {\
494 const int nibble = nidx < 0 ? 7 - nidx : nidx;\
495 int dec_sample = predictor +\
496 (STEP_TABLE *\
497 ff_adpcm_yamaha_difflookup[nibble]) / 8;\
498 STORE_NODE(NAME, STEP_INDEX);\
500 LOOP_NODES(ima, ff_adpcm_step_table[step],
501 av_clip(step + ff_adpcm_index_table[nibble], 0, 88));
502 } else { //AV_CODEC_ID_ADPCM_YAMAHA
503 LOOP_NODES(yamaha, step,
504 av_clip((step * ff_adpcm_yamaha_indexscale[nibble]) >> 8,
505 127, 24576));
506 #undef LOOP_NODES
507 #undef STORE_NODE
511 u = nodes;
512 nodes = nodes_next;
513 nodes_next = u;
515 generation++;
516 if (generation == 255) {
517 memset(hash, 0xff, 65536 * sizeof(*hash));
518 generation = 0;
521 // prevent overflow
522 if (nodes[0]->ssd > (1 << 28)) {
523 for (j = 1; j < frontier && nodes[j]; j++)
524 nodes[j]->ssd -= nodes[0]->ssd;
525 nodes[0]->ssd = 0;
528 // merge old paths to save memory
529 if (i == froze + FREEZE_INTERVAL) {
530 p = &paths[nodes[0]->path];
531 for (k = i; k > froze; k--) {
532 dst[k] = p->nibble;
533 p = &paths[p->prev];
535 froze = i;
536 pathn = 0;
537 // other nodes might use paths that don't coincide with the frozen one.
538 // checking which nodes do so is too slow, so just kill them all.
539 // this also slightly improves quality, but I don't know why.
540 memset(nodes + 1, 0, (frontier - 1) * sizeof(TrellisNode*));
544 p = &paths[nodes[0]->path];
545 for (i = n - 1; i > froze; i--) {
546 dst[i] = p->nibble;
547 p = &paths[p->prev];
550 c->predictor = nodes[0]->sample1;
551 c->sample1 = nodes[0]->sample1;
552 c->sample2 = nodes[0]->sample2;
553 c->step_index = nodes[0]->step;
554 c->step = nodes[0]->step;
555 c->idelta = nodes[0]->step;
558 #if CONFIG_ADPCM_ARGO_ENCODER
559 static inline int adpcm_argo_compress_nibble(const ADPCMChannelStatus *cs, int16_t s,
560 int shift, int flag)
562 int nibble;
564 if (flag)
565 nibble = 4 * s - 8 * cs->sample1 + 4 * cs->sample2;
566 else
567 nibble = 4 * s - 4 * cs->sample1;
569 return (nibble >> shift) & 0x0F;
572 static int64_t adpcm_argo_compress_block(ADPCMChannelStatus *cs, PutBitContext *pb,
573 const int16_t *samples, int nsamples,
574 int shift, int flag)
576 int64_t error = 0;
578 if (pb) {
579 put_bits(pb, 4, shift - 2);
580 put_bits(pb, 1, 0);
581 put_bits(pb, 1, !!flag);
582 put_bits(pb, 2, 0);
585 for (int n = 0; n < nsamples; n++) {
586 /* Compress the nibble, then expand it to see how much precision we've lost. */
587 int nibble = adpcm_argo_compress_nibble(cs, samples[n], shift, flag);
588 int16_t sample = ff_adpcm_argo_expand_nibble(cs, nibble, shift, flag);
590 error += abs(samples[n] - sample);
592 if (pb)
593 put_bits(pb, 4, nibble);
596 return error;
598 #endif
600 static int adpcm_encode_frame(AVCodecContext *avctx, AVPacket *avpkt,
601 const AVFrame *frame, int *got_packet_ptr)
603 int st, pkt_size, ret;
604 const int16_t *samples;
605 const int16_t *const *samples_p;
606 uint8_t *dst;
607 ADPCMEncodeContext *c = avctx->priv_data;
608 int channels = avctx->ch_layout.nb_channels;
610 samples = (const int16_t *)frame->data[0];
611 samples_p = (const int16_t *const *)frame->extended_data;
612 st = channels == 2;
614 if (avctx->codec_id == AV_CODEC_ID_ADPCM_IMA_SSI ||
615 avctx->codec_id == AV_CODEC_ID_ADPCM_IMA_ALP ||
616 avctx->codec_id == AV_CODEC_ID_ADPCM_IMA_APM ||
617 avctx->codec_id == AV_CODEC_ID_ADPCM_IMA_WS)
618 pkt_size = (frame->nb_samples * channels + 1) / 2;
619 else
620 pkt_size = avctx->block_align;
621 if ((ret = ff_get_encode_buffer(avctx, avpkt, pkt_size, 0)) < 0)
622 return ret;
623 dst = avpkt->data;
625 switch(avctx->codec->id) {
626 CASE(ADPCM_IMA_WAV,
627 int blocks = (frame->nb_samples - 1) / 8;
629 for (int ch = 0; ch < channels; ch++) {
630 ADPCMChannelStatus *status = &c->status[ch];
631 status->prev_sample = samples_p[ch][0];
632 /* status->step_index = 0;
633 XXX: not sure how to init the state machine */
634 bytestream_put_le16(&dst, status->prev_sample);
635 *dst++ = status->step_index;
636 *dst++ = 0; /* unknown */
639 /* stereo: 4 bytes (8 samples) for left, 4 bytes for right */
640 if (avctx->trellis > 0) {
641 uint8_t *buf;
642 if (!FF_ALLOC_TYPED_ARRAY(buf, channels * blocks * 8))
643 return AVERROR(ENOMEM);
644 for (int ch = 0; ch < channels; ch++) {
645 adpcm_compress_trellis(avctx, &samples_p[ch][1],
646 buf + ch * blocks * 8, &c->status[ch],
647 blocks * 8, 1);
649 for (int i = 0; i < blocks; i++) {
650 for (int ch = 0; ch < channels; ch++) {
651 uint8_t *buf1 = buf + ch * blocks * 8 + i * 8;
652 for (int j = 0; j < 8; j += 2)
653 *dst++ = buf1[j] | (buf1[j + 1] << 4);
656 av_free(buf);
657 } else {
658 for (int i = 0; i < blocks; i++) {
659 for (int ch = 0; ch < channels; ch++) {
660 ADPCMChannelStatus *status = &c->status[ch];
661 const int16_t *smp = &samples_p[ch][1 + i * 8];
662 for (int j = 0; j < 8; j += 2) {
663 uint8_t v = adpcm_ima_compress_sample(status, smp[j ]);
664 v |= adpcm_ima_compress_sample(status, smp[j + 1]) << 4;
665 *dst++ = v;
670 ) /* End of CASE */
671 CASE(ADPCM_IMA_QT,
672 PutBitContext pb;
673 init_put_bits(&pb, dst, pkt_size);
675 for (int ch = 0; ch < channels; ch++) {
676 ADPCMChannelStatus *status = &c->status[ch];
677 put_bits(&pb, 9, (status->prev_sample & 0xFFFF) >> 7);
678 put_bits(&pb, 7, status->step_index);
679 if (avctx->trellis > 0) {
680 uint8_t buf[64];
681 adpcm_compress_trellis(avctx, &samples_p[ch][0], buf, status,
682 64, 1);
683 for (int i = 0; i < 64; i++)
684 put_bits(&pb, 4, buf[i ^ 1]);
685 status->prev_sample = status->predictor;
686 } else {
687 for (int i = 0; i < 64; i += 2) {
688 int t1, t2;
689 t1 = adpcm_ima_qt_compress_sample(status, samples_p[ch][i ]);
690 t2 = adpcm_ima_qt_compress_sample(status, samples_p[ch][i + 1]);
691 put_bits(&pb, 4, t2);
692 put_bits(&pb, 4, t1);
697 flush_put_bits(&pb);
698 ) /* End of CASE */
699 CASE(ADPCM_IMA_SSI,
700 PutBitContext pb;
701 init_put_bits(&pb, dst, pkt_size);
703 av_assert0(avctx->trellis == 0);
705 for (int i = 0; i < frame->nb_samples; i++) {
706 for (int ch = 0; ch < channels; ch++) {
707 put_bits(&pb, 4, adpcm_ima_qt_compress_sample(c->status + ch, *samples++));
711 flush_put_bits(&pb);
712 ) /* End of CASE */
713 CASE(ADPCM_IMA_ALP,
714 PutBitContext pb;
715 init_put_bits(&pb, dst, pkt_size);
717 av_assert0(avctx->trellis == 0);
719 for (int n = frame->nb_samples / 2; n > 0; n--) {
720 for (int ch = 0; ch < channels; ch++) {
721 put_bits(&pb, 4, adpcm_ima_alp_compress_sample(c->status + ch, *samples++));
722 put_bits(&pb, 4, adpcm_ima_alp_compress_sample(c->status + ch, samples[st]));
724 samples += channels;
727 flush_put_bits(&pb);
728 ) /* End of CASE */
729 CASE(ADPCM_SWF,
730 const int n = frame->nb_samples - 1;
731 PutBitContext pb;
732 init_put_bits(&pb, dst, pkt_size);
734 /* NB: This is safe as we don't have AV_CODEC_CAP_SMALL_LAST_FRAME. */
735 av_assert0(n == 4095);
737 // store AdpcmCodeSize
738 put_bits(&pb, 2, 2); // set 4-bit flash adpcm format
740 // init the encoder state
741 for (int i = 0; i < channels; i++) {
742 // clip step so it fits 6 bits
743 c->status[i].step_index = av_clip_uintp2(c->status[i].step_index, 6);
744 put_sbits(&pb, 16, samples[i]);
745 put_bits(&pb, 6, c->status[i].step_index);
746 c->status[i].prev_sample = samples[i];
749 if (avctx->trellis > 0) {
750 uint8_t buf[8190 /* = 2 * n */];
751 adpcm_compress_trellis(avctx, samples + channels, buf,
752 &c->status[0], n, channels);
753 if (channels == 2)
754 adpcm_compress_trellis(avctx, samples + channels + 1,
755 buf + n, &c->status[1], n,
756 channels);
757 for (int i = 0; i < n; i++) {
758 put_bits(&pb, 4, buf[i]);
759 if (channels == 2)
760 put_bits(&pb, 4, buf[n + i]);
762 } else {
763 for (int i = 1; i < frame->nb_samples; i++) {
764 put_bits(&pb, 4, adpcm_ima_compress_sample(&c->status[0],
765 samples[channels * i]));
766 if (channels == 2)
767 put_bits(&pb, 4, adpcm_ima_compress_sample(&c->status[1],
768 samples[2 * i + 1]));
771 flush_put_bits(&pb);
772 ) /* End of CASE */
773 CASE(ADPCM_MS,
774 for (int i = 0; i < channels; i++) {
775 int predictor = 0;
776 *dst++ = predictor;
777 c->status[i].coeff1 = ff_adpcm_AdaptCoeff1[predictor];
778 c->status[i].coeff2 = ff_adpcm_AdaptCoeff2[predictor];
780 for (int i = 0; i < channels; i++) {
781 if (c->status[i].idelta < 16)
782 c->status[i].idelta = 16;
783 bytestream_put_le16(&dst, c->status[i].idelta);
785 for (int i = 0; i < channels; i++)
786 c->status[i].sample2= *samples++;
787 for (int i = 0; i < channels; i++) {
788 c->status[i].sample1 = *samples++;
789 bytestream_put_le16(&dst, c->status[i].sample1);
791 for (int i = 0; i < channels; i++)
792 bytestream_put_le16(&dst, c->status[i].sample2);
794 if (avctx->trellis > 0) {
795 const int n = avctx->block_align - 7 * channels;
796 uint8_t *buf = av_malloc(2 * n);
797 if (!buf)
798 return AVERROR(ENOMEM);
799 if (channels == 1) {
800 adpcm_compress_trellis(avctx, samples, buf, &c->status[0], n,
801 channels);
802 for (int i = 0; i < n; i += 2)
803 *dst++ = (buf[i] << 4) | buf[i + 1];
804 } else {
805 adpcm_compress_trellis(avctx, samples, buf,
806 &c->status[0], n, channels);
807 adpcm_compress_trellis(avctx, samples + 1, buf + n,
808 &c->status[1], n, channels);
809 for (int i = 0; i < n; i++)
810 *dst++ = (buf[i] << 4) | buf[n + i];
812 av_free(buf);
813 } else {
814 for (int i = 7 * channels; i < avctx->block_align; i++) {
815 int nibble;
816 nibble = adpcm_ms_compress_sample(&c->status[ 0], *samples++) << 4;
817 nibble |= adpcm_ms_compress_sample(&c->status[st], *samples++);
818 *dst++ = nibble;
821 ) /* End of CASE */
822 CASE(ADPCM_YAMAHA,
823 int n = frame->nb_samples / 2;
824 if (avctx->trellis > 0) {
825 uint8_t *buf = av_malloc(2 * n * 2);
826 if (!buf)
827 return AVERROR(ENOMEM);
828 n *= 2;
829 if (channels == 1) {
830 adpcm_compress_trellis(avctx, samples, buf, &c->status[0], n,
831 channels);
832 for (int i = 0; i < n; i += 2)
833 *dst++ = buf[i] | (buf[i + 1] << 4);
834 } else {
835 adpcm_compress_trellis(avctx, samples, buf,
836 &c->status[0], n, channels);
837 adpcm_compress_trellis(avctx, samples + 1, buf + n,
838 &c->status[1], n, channels);
839 for (int i = 0; i < n; i++)
840 *dst++ = buf[i] | (buf[n + i] << 4);
842 av_free(buf);
843 } else
844 for (n *= channels; n > 0; n--) {
845 int nibble;
846 nibble = adpcm_yamaha_compress_sample(&c->status[ 0], *samples++);
847 nibble |= adpcm_yamaha_compress_sample(&c->status[st], *samples++) << 4;
848 *dst++ = nibble;
850 ) /* End of CASE */
851 CASE(ADPCM_IMA_APM,
852 PutBitContext pb;
853 init_put_bits(&pb, dst, pkt_size);
855 av_assert0(avctx->trellis == 0);
857 for (int n = frame->nb_samples / 2; n > 0; n--) {
858 for (int ch = 0; ch < channels; ch++) {
859 put_bits(&pb, 4, adpcm_ima_qt_compress_sample(c->status + ch, *samples++));
860 put_bits(&pb, 4, adpcm_ima_qt_compress_sample(c->status + ch, samples[st]));
862 samples += channels;
865 flush_put_bits(&pb);
866 ) /* End of CASE */
867 CASE(ADPCM_IMA_AMV,
868 av_assert0(channels == 1);
870 c->status[0].prev_sample = *samples;
871 bytestream_put_le16(&dst, c->status[0].prev_sample);
872 bytestream_put_byte(&dst, c->status[0].step_index);
873 bytestream_put_byte(&dst, 0);
874 bytestream_put_le32(&dst, avctx->frame_size);
876 if (avctx->trellis > 0) {
877 const int n = frame->nb_samples >> 1;
878 uint8_t *buf = av_malloc(2 * n);
880 if (!buf)
881 return AVERROR(ENOMEM);
883 adpcm_compress_trellis(avctx, samples, buf, &c->status[0], 2 * n, channels);
884 for (int i = 0; i < n; i++)
885 bytestream_put_byte(&dst, (buf[2 * i] << 4) | buf[2 * i + 1]);
887 samples += 2 * n;
888 av_free(buf);
889 } else for (int n = frame->nb_samples >> 1; n > 0; n--) {
890 int nibble;
891 nibble = adpcm_ima_compress_sample(&c->status[0], *samples++) << 4;
892 nibble |= adpcm_ima_compress_sample(&c->status[0], *samples++) & 0x0F;
893 bytestream_put_byte(&dst, nibble);
896 if (avctx->frame_size & 1) {
897 int nibble = adpcm_ima_compress_sample(&c->status[0], *samples++) << 4;
898 bytestream_put_byte(&dst, nibble);
900 ) /* End of CASE */
901 CASE(ADPCM_ARGO,
902 PutBitContext pb;
903 init_put_bits(&pb, dst, pkt_size);
905 av_assert0(frame->nb_samples == 32);
907 for (int ch = 0; ch < channels; ch++) {
908 int64_t error = INT64_MAX, tmperr = INT64_MAX;
909 int shift = 2, flag = 0;
910 int saved1 = c->status[ch].sample1;
911 int saved2 = c->status[ch].sample2;
913 /* Find the optimal coefficients, bail early if we find a perfect result. */
914 for (int s = 2; s < 18 && tmperr != 0; s++) {
915 for (int f = 0; f < 2 && tmperr != 0; f++) {
916 c->status[ch].sample1 = saved1;
917 c->status[ch].sample2 = saved2;
918 tmperr = adpcm_argo_compress_block(c->status + ch, NULL, samples_p[ch],
919 frame->nb_samples, s, f);
920 if (tmperr < error) {
921 shift = s;
922 flag = f;
923 error = tmperr;
928 /* Now actually do the encode. */
929 c->status[ch].sample1 = saved1;
930 c->status[ch].sample2 = saved2;
931 adpcm_argo_compress_block(c->status + ch, &pb, samples_p[ch],
932 frame->nb_samples, shift, flag);
935 flush_put_bits(&pb);
936 ) /* End of CASE */
937 CASE(ADPCM_IMA_WS,
938 PutBitContext pb;
939 init_put_bits(&pb, dst, pkt_size);
941 av_assert0(avctx->trellis == 0);
942 for (int n = frame->nb_samples / 2; n > 0; n--) {
943 /* stereo: 1 byte (2 samples) for left, 1 byte for right */
944 for (int ch = 0; ch < channels; ch++) {
945 int t1, t2;
946 t1 = adpcm_ima_compress_sample(&c->status[ch], *samples++);
947 t2 = adpcm_ima_compress_sample(&c->status[ch], samples[st]);
948 put_bits(&pb, 4, t2);
949 put_bits(&pb, 4, t1);
951 samples += channels;
953 flush_put_bits(&pb);
954 ) /* End of CASE */
955 default:
956 return AVERROR(EINVAL);
959 *got_packet_ptr = 1;
960 return 0;
963 static const enum AVSampleFormat sample_fmts[] = {
964 AV_SAMPLE_FMT_S16, AV_SAMPLE_FMT_NONE
967 static const enum AVSampleFormat sample_fmts_p[] = {
968 AV_SAMPLE_FMT_S16P, AV_SAMPLE_FMT_NONE
971 static const AVChannelLayout ch_layouts[] = {
972 AV_CHANNEL_LAYOUT_MONO,
973 AV_CHANNEL_LAYOUT_STEREO,
974 { 0 },
977 static const AVOption options[] = {
979 .name = "block_size",
980 .help = "set the block size",
981 .offset = offsetof(ADPCMEncodeContext, block_size),
982 .type = AV_OPT_TYPE_INT,
983 .default_val = {.i64 = 1024},
984 .min = 32,
985 .max = 8192, /* Is this a reasonable upper limit? */
986 .flags = AV_OPT_FLAG_ENCODING_PARAM | AV_OPT_FLAG_AUDIO_PARAM
988 { NULL }
991 static const AVClass adpcm_encoder_class = {
992 .class_name = "ADPCM encoder",
993 .item_name = av_default_item_name,
994 .option = options,
995 .version = LIBAVUTIL_VERSION_INT,
998 #define ADPCM_ENCODER_0(id_, name_, sample_fmts_, capabilities_, long_name_)
999 #define ADPCM_ENCODER_1(id_, name_, sample_fmts_, capabilities_, long_name_) \
1000 const FFCodec ff_ ## name_ ## _encoder = { \
1001 .p.name = #name_, \
1002 CODEC_LONG_NAME(long_name_), \
1003 .p.type = AVMEDIA_TYPE_AUDIO, \
1004 .p.id = id_, \
1005 .p.sample_fmts = sample_fmts_, \
1006 .p.ch_layouts = ch_layouts, \
1007 .p.capabilities = capabilities_ | AV_CODEC_CAP_DR1 | \
1008 AV_CODEC_CAP_ENCODER_REORDERED_OPAQUE, \
1009 .p.priv_class = &adpcm_encoder_class, \
1010 .priv_data_size = sizeof(ADPCMEncodeContext), \
1011 .init = adpcm_encode_init, \
1012 FF_CODEC_ENCODE_CB(adpcm_encode_frame), \
1013 .close = adpcm_encode_close, \
1014 .caps_internal = FF_CODEC_CAP_INIT_CLEANUP, \
1016 #define ADPCM_ENCODER_2(enabled, codec_id, name, sample_fmts, capabilities, long_name) \
1017 ADPCM_ENCODER_ ## enabled(codec_id, name, sample_fmts, capabilities, long_name)
1018 #define ADPCM_ENCODER_3(config, codec_id, name, sample_fmts, capabilities, long_name) \
1019 ADPCM_ENCODER_2(config, codec_id, name, sample_fmts, capabilities, long_name)
1020 #define ADPCM_ENCODER(codec, name, sample_fmts, capabilities, long_name) \
1021 ADPCM_ENCODER_3(CONFIG_ ## codec ## _ENCODER, AV_CODEC_ID_ ## codec, \
1022 name, sample_fmts, capabilities, long_name)
1024 ADPCM_ENCODER(ADPCM_ARGO, adpcm_argo, sample_fmts_p, 0, "ADPCM Argonaut Games")
1025 ADPCM_ENCODER(ADPCM_IMA_AMV, adpcm_ima_amv, sample_fmts, 0, "ADPCM IMA AMV")
1026 ADPCM_ENCODER(ADPCM_IMA_APM, adpcm_ima_apm, sample_fmts, AV_CODEC_CAP_SMALL_LAST_FRAME, "ADPCM IMA Ubisoft APM")
1027 ADPCM_ENCODER(ADPCM_IMA_ALP, adpcm_ima_alp, sample_fmts, AV_CODEC_CAP_SMALL_LAST_FRAME, "ADPCM IMA High Voltage Software ALP")
1028 ADPCM_ENCODER(ADPCM_IMA_QT, adpcm_ima_qt, sample_fmts_p, 0, "ADPCM IMA QuickTime")
1029 ADPCM_ENCODER(ADPCM_IMA_SSI, adpcm_ima_ssi, sample_fmts, AV_CODEC_CAP_SMALL_LAST_FRAME, "ADPCM IMA Simon & Schuster Interactive")
1030 ADPCM_ENCODER(ADPCM_IMA_WAV, adpcm_ima_wav, sample_fmts_p, 0, "ADPCM IMA WAV")
1031 ADPCM_ENCODER(ADPCM_IMA_WS, adpcm_ima_ws, sample_fmts, AV_CODEC_CAP_SMALL_LAST_FRAME, "ADPCM IMA Westwood")
1032 ADPCM_ENCODER(ADPCM_MS, adpcm_ms, sample_fmts, 0, "ADPCM Microsoft")
1033 ADPCM_ENCODER(ADPCM_SWF, adpcm_swf, sample_fmts, 0, "ADPCM Shockwave Flash")
1034 ADPCM_ENCODER(ADPCM_YAMAHA, adpcm_yamaha, sample_fmts, 0, "ADPCM Yamaha")