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
27 #include "bytestream.h"
29 #include "adpcm_data.h"
35 * See ADPCM decoder reference documents for codec information.
38 typedef struct TrellisPath {
43 typedef struct TrellisNode {
51 typedef struct ADPCMEncodeContext {
52 ADPCMChannelStatus status[6];
54 TrellisNode *node_buf;
55 TrellisNode **nodep_buf;
56 uint8_t *trellis_hash;
59 #define FREEZE_INTERVAL 128
61 static av_cold int adpcm_encode_init(AVCodecContext *avctx)
63 ADPCMEncodeContext *s = avctx->priv_data;
67 if (avctx->channels > 2) {
68 av_log(avctx, AV_LOG_ERROR, "only stereo or mono is supported\n");
69 return AVERROR(EINVAL);
72 if (avctx->trellis && (unsigned)avctx->trellis > 16U) {
73 av_log(avctx, AV_LOG_ERROR, "invalid trellis size\n");
74 return AVERROR(EINVAL);
77 if (avctx->trellis && avctx->codec->id == AV_CODEC_ID_ADPCM_IMA_SSI) {
79 * The current trellis implementation doesn't work for extended
80 * runs of samples without periodic resets. Disallow it.
82 av_log(avctx, AV_LOG_ERROR, "trellis not supported\n");
83 return AVERROR_PATCHWELCOME;
87 int frontier = 1 << avctx->trellis;
88 int max_paths = frontier * FREEZE_INTERVAL;
89 if (!FF_ALLOC_TYPED_ARRAY(s->paths, max_paths) ||
90 !FF_ALLOC_TYPED_ARRAY(s->node_buf, 2 * frontier) ||
91 !FF_ALLOC_TYPED_ARRAY(s->nodep_buf, 2 * frontier) ||
92 !FF_ALLOC_TYPED_ARRAY(s->trellis_hash, 65536))
93 return AVERROR(ENOMEM);
96 avctx->bits_per_coded_sample = av_get_bits_per_sample(avctx->codec->id);
98 switch (avctx->codec->id) {
99 case AV_CODEC_ID_ADPCM_IMA_WAV:
100 /* each 16 bits sample gives one nibble
101 and we have 4 bytes per channel overhead */
102 avctx->frame_size = (BLKSIZE - 4 * avctx->channels) * 8 /
103 (4 * avctx->channels) + 1;
104 /* seems frame_size isn't taken into account...
105 have to buffer the samples :-( */
106 avctx->block_align = BLKSIZE;
107 avctx->bits_per_coded_sample = 4;
109 case AV_CODEC_ID_ADPCM_IMA_QT:
110 avctx->frame_size = 64;
111 avctx->block_align = 34 * avctx->channels;
113 case AV_CODEC_ID_ADPCM_MS:
114 /* each 16 bits sample gives one nibble
115 and we have 7 bytes per channel overhead */
116 avctx->frame_size = (BLKSIZE - 7 * avctx->channels) * 2 / avctx->channels + 2;
117 avctx->bits_per_coded_sample = 4;
118 avctx->block_align = BLKSIZE;
119 if (!(avctx->extradata = av_malloc(32 + AV_INPUT_BUFFER_PADDING_SIZE)))
120 return AVERROR(ENOMEM);
121 avctx->extradata_size = 32;
122 extradata = avctx->extradata;
123 bytestream_put_le16(&extradata, avctx->frame_size);
124 bytestream_put_le16(&extradata, 7); /* wNumCoef */
125 for (i = 0; i < 7; i++) {
126 bytestream_put_le16(&extradata, ff_adpcm_AdaptCoeff1[i] * 4);
127 bytestream_put_le16(&extradata, ff_adpcm_AdaptCoeff2[i] * 4);
130 case AV_CODEC_ID_ADPCM_YAMAHA:
131 avctx->frame_size = BLKSIZE * 2 / avctx->channels;
132 avctx->block_align = BLKSIZE;
134 case AV_CODEC_ID_ADPCM_SWF:
135 if (avctx->sample_rate != 11025 &&
136 avctx->sample_rate != 22050 &&
137 avctx->sample_rate != 44100) {
138 av_log(avctx, AV_LOG_ERROR, "Sample rate must be 11025, "
140 return AVERROR(EINVAL);
142 avctx->frame_size = 512 * (avctx->sample_rate / 11025);
144 case AV_CODEC_ID_ADPCM_IMA_SSI:
145 avctx->frame_size = BLKSIZE * 2 / avctx->channels;
146 avctx->block_align = BLKSIZE;
149 return AVERROR(EINVAL);
155 static av_cold int adpcm_encode_close(AVCodecContext *avctx)
157 ADPCMEncodeContext *s = avctx->priv_data;
159 av_freep(&s->node_buf);
160 av_freep(&s->nodep_buf);
161 av_freep(&s->trellis_hash);
167 static inline uint8_t adpcm_ima_compress_sample(ADPCMChannelStatus *c,
170 int delta = sample - c->prev_sample;
171 int nibble = FFMIN(7, abs(delta) * 4 /
172 ff_adpcm_step_table[c->step_index]) + (delta < 0) * 8;
173 c->prev_sample += ((ff_adpcm_step_table[c->step_index] *
174 ff_adpcm_yamaha_difflookup[nibble]) / 8);
175 c->prev_sample = av_clip_int16(c->prev_sample);
176 c->step_index = av_clip(c->step_index + ff_adpcm_index_table[nibble], 0, 88);
180 static inline uint8_t adpcm_ima_qt_compress_sample(ADPCMChannelStatus *c,
183 int delta = sample - c->prev_sample;
184 int diff, step = ff_adpcm_step_table[c->step_index];
185 int nibble = 8*(delta < 0);
188 diff = delta + (step >> 3);
207 c->prev_sample -= diff;
209 c->prev_sample += diff;
211 c->prev_sample = av_clip_int16(c->prev_sample);
212 c->step_index = av_clip(c->step_index + ff_adpcm_index_table[nibble], 0, 88);
217 static inline uint8_t adpcm_ms_compress_sample(ADPCMChannelStatus *c,
220 int predictor, nibble, bias;
222 predictor = (((c->sample1) * (c->coeff1)) +
223 (( c->sample2) * (c->coeff2))) / 64;
225 nibble = sample - predictor;
227 bias = c->idelta / 2;
229 bias = -c->idelta / 2;
231 nibble = (nibble + bias) / c->idelta;
232 nibble = av_clip_intp2(nibble, 3) & 0x0F;
234 predictor += ((nibble & 0x08) ? (nibble - 0x10) : nibble) * c->idelta;
236 c->sample2 = c->sample1;
237 c->sample1 = av_clip_int16(predictor);
239 c->idelta = (ff_adpcm_AdaptationTable[nibble] * c->idelta) >> 8;
246 static inline uint8_t adpcm_yamaha_compress_sample(ADPCMChannelStatus *c,
256 delta = sample - c->predictor;
258 nibble = FFMIN(7, abs(delta) * 4 / c->step) + (delta < 0) * 8;
260 c->predictor += ((c->step * ff_adpcm_yamaha_difflookup[nibble]) / 8);
261 c->predictor = av_clip_int16(c->predictor);
262 c->step = (c->step * ff_adpcm_yamaha_indexscale[nibble]) >> 8;
263 c->step = av_clip(c->step, 127, 24576);
268 static void adpcm_compress_trellis(AVCodecContext *avctx,
269 const int16_t *samples, uint8_t *dst,
270 ADPCMChannelStatus *c, int n, int stride)
272 //FIXME 6% faster if frontier is a compile-time constant
273 ADPCMEncodeContext *s = avctx->priv_data;
274 const int frontier = 1 << avctx->trellis;
275 const int version = avctx->codec->id;
276 TrellisPath *paths = s->paths, *p;
277 TrellisNode *node_buf = s->node_buf;
278 TrellisNode **nodep_buf = s->nodep_buf;
279 TrellisNode **nodes = nodep_buf; // nodes[] is always sorted by .ssd
280 TrellisNode **nodes_next = nodep_buf + frontier;
281 int pathn = 0, froze = -1, i, j, k, generation = 0;
282 uint8_t *hash = s->trellis_hash;
283 memset(hash, 0xff, 65536 * sizeof(*hash));
285 memset(nodep_buf, 0, 2 * frontier * sizeof(*nodep_buf));
286 nodes[0] = node_buf + frontier;
289 nodes[0]->step = c->step_index;
290 nodes[0]->sample1 = c->sample1;
291 nodes[0]->sample2 = c->sample2;
292 if (version == AV_CODEC_ID_ADPCM_IMA_WAV ||
293 version == AV_CODEC_ID_ADPCM_IMA_QT ||
294 version == AV_CODEC_ID_ADPCM_SWF)
295 nodes[0]->sample1 = c->prev_sample;
296 if (version == AV_CODEC_ID_ADPCM_MS)
297 nodes[0]->step = c->idelta;
298 if (version == AV_CODEC_ID_ADPCM_YAMAHA) {
300 nodes[0]->step = 127;
301 nodes[0]->sample1 = 0;
303 nodes[0]->step = c->step;
304 nodes[0]->sample1 = c->predictor;
308 for (i = 0; i < n; i++) {
309 TrellisNode *t = node_buf + frontier*(i&1);
311 int sample = samples[i * stride];
313 memset(nodes_next, 0, frontier * sizeof(TrellisNode*));
314 for (j = 0; j < frontier && nodes[j]; j++) {
315 // higher j have higher ssd already, so they're likely
316 // to yield a suboptimal next sample too
317 const int range = (j < frontier / 2) ? 1 : 0;
318 const int step = nodes[j]->step;
320 if (version == AV_CODEC_ID_ADPCM_MS) {
321 const int predictor = ((nodes[j]->sample1 * c->coeff1) +
322 (nodes[j]->sample2 * c->coeff2)) / 64;
323 const int div = (sample - predictor) / step;
324 const int nmin = av_clip(div-range, -8, 6);
325 const int nmax = av_clip(div+range, -7, 7);
326 for (nidx = nmin; nidx <= nmax; nidx++) {
327 const int nibble = nidx & 0xf;
328 int dec_sample = predictor + nidx * step;
329 #define STORE_NODE(NAME, STEP_INDEX)\
335 dec_sample = av_clip_int16(dec_sample);\
336 d = sample - dec_sample;\
337 ssd = nodes[j]->ssd + d*(unsigned)d;\
338 /* Check for wraparound, skip such samples completely. \
339 * Note, changing ssd to a 64 bit variable would be \
340 * simpler, avoiding this check, but it's slower on \
341 * x86 32 bit at the moment. */\
342 if (ssd < nodes[j]->ssd)\
344 /* Collapse any two states with the same previous sample value. \
345 * One could also distinguish states by step and by 2nd to last
346 * sample, but the effects of that are negligible.
347 * Since nodes in the previous generation are iterated
348 * through a heap, they're roughly ordered from better to
349 * worse, but not strictly ordered. Therefore, an earlier
350 * node with the same sample value is better in most cases
351 * (and thus the current is skipped), but not strictly
352 * in all cases. Only skipping samples where ssd >=
353 * ssd of the earlier node with the same sample gives
354 * slightly worse quality, though, for some reason. */ \
355 h = &hash[(uint16_t) dec_sample];\
356 if (*h == generation)\
358 if (heap_pos < frontier) {\
361 /* Try to replace one of the leaf nodes with the new \
362 * one, but try a different slot each time. */\
363 pos = (frontier >> 1) +\
364 (heap_pos & ((frontier >> 1) - 1));\
365 if (ssd > nodes_next[pos]->ssd)\
370 u = nodes_next[pos];\
372 av_assert1(pathn < FREEZE_INTERVAL << avctx->trellis);\
374 nodes_next[pos] = u;\
378 u->step = STEP_INDEX;\
379 u->sample2 = nodes[j]->sample1;\
380 u->sample1 = dec_sample;\
381 paths[u->path].nibble = nibble;\
382 paths[u->path].prev = nodes[j]->path;\
383 /* Sift the newly inserted node up in the heap to \
384 * restore the heap property. */\
386 int parent = (pos - 1) >> 1;\
387 if (nodes_next[parent]->ssd <= ssd)\
389 FFSWAP(TrellisNode*, nodes_next[parent], nodes_next[pos]);\
393 STORE_NODE(ms, FFMAX(16,
394 (ff_adpcm_AdaptationTable[nibble] * step) >> 8));
396 } else if (version == AV_CODEC_ID_ADPCM_IMA_WAV ||
397 version == AV_CODEC_ID_ADPCM_IMA_QT ||
398 version == AV_CODEC_ID_ADPCM_SWF) {
399 #define LOOP_NODES(NAME, STEP_TABLE, STEP_INDEX)\
400 const int predictor = nodes[j]->sample1;\
401 const int div = (sample - predictor) * 4 / STEP_TABLE;\
402 int nmin = av_clip(div - range, -7, 6);\
403 int nmax = av_clip(div + range, -6, 7);\
405 nmin--; /* distinguish -0 from +0 */\
408 for (nidx = nmin; nidx <= nmax; nidx++) {\
409 const int nibble = nidx < 0 ? 7 - nidx : nidx;\
410 int dec_sample = predictor +\
412 ff_adpcm_yamaha_difflookup[nibble]) / 8;\
413 STORE_NODE(NAME, STEP_INDEX);\
415 LOOP_NODES(ima, ff_adpcm_step_table[step],
416 av_clip(step + ff_adpcm_index_table[nibble], 0, 88));
417 } else { //AV_CODEC_ID_ADPCM_YAMAHA
418 LOOP_NODES(yamaha, step,
419 av_clip((step * ff_adpcm_yamaha_indexscale[nibble]) >> 8,
431 if (generation == 255) {
432 memset(hash, 0xff, 65536 * sizeof(*hash));
437 if (nodes[0]->ssd > (1 << 28)) {
438 for (j = 1; j < frontier && nodes[j]; j++)
439 nodes[j]->ssd -= nodes[0]->ssd;
443 // merge old paths to save memory
444 if (i == froze + FREEZE_INTERVAL) {
445 p = &paths[nodes[0]->path];
446 for (k = i; k > froze; k--) {
452 // other nodes might use paths that don't coincide with the frozen one.
453 // checking which nodes do so is too slow, so just kill them all.
454 // this also slightly improves quality, but I don't know why.
455 memset(nodes + 1, 0, (frontier - 1) * sizeof(TrellisNode*));
459 p = &paths[nodes[0]->path];
460 for (i = n - 1; i > froze; i--) {
465 c->predictor = nodes[0]->sample1;
466 c->sample1 = nodes[0]->sample1;
467 c->sample2 = nodes[0]->sample2;
468 c->step_index = nodes[0]->step;
469 c->step = nodes[0]->step;
470 c->idelta = nodes[0]->step;
473 static int adpcm_encode_frame(AVCodecContext *avctx, AVPacket *avpkt,
474 const AVFrame *frame, int *got_packet_ptr)
476 int n, i, ch, st, pkt_size, ret;
477 const int16_t *samples;
480 ADPCMEncodeContext *c = avctx->priv_data;
483 samples = (const int16_t *)frame->data[0];
484 samples_p = (int16_t **)frame->extended_data;
485 st = avctx->channels == 2;
487 if (avctx->codec_id == AV_CODEC_ID_ADPCM_SWF)
488 pkt_size = (2 + avctx->channels * (22 + 4 * (frame->nb_samples - 1)) + 7) / 8;
489 else if (avctx->codec_id == AV_CODEC_ID_ADPCM_IMA_SSI)
490 pkt_size = (frame->nb_samples * avctx->channels) / 2;
492 pkt_size = avctx->block_align;
493 if ((ret = ff_alloc_packet2(avctx, avpkt, pkt_size, 0)) < 0)
497 switch(avctx->codec->id) {
498 case AV_CODEC_ID_ADPCM_IMA_WAV:
502 blocks = (frame->nb_samples - 1) / 8;
504 for (ch = 0; ch < avctx->channels; ch++) {
505 ADPCMChannelStatus *status = &c->status[ch];
506 status->prev_sample = samples_p[ch][0];
507 /* status->step_index = 0;
508 XXX: not sure how to init the state machine */
509 bytestream_put_le16(&dst, status->prev_sample);
510 *dst++ = status->step_index;
511 *dst++ = 0; /* unknown */
514 /* stereo: 4 bytes (8 samples) for left, 4 bytes for right */
515 if (avctx->trellis > 0) {
516 if (!FF_ALLOC_TYPED_ARRAY(buf, avctx->channels * blocks * 8))
517 return AVERROR(ENOMEM);
518 for (ch = 0; ch < avctx->channels; ch++) {
519 adpcm_compress_trellis(avctx, &samples_p[ch][1],
520 buf + ch * blocks * 8, &c->status[ch],
523 for (i = 0; i < blocks; i++) {
524 for (ch = 0; ch < avctx->channels; ch++) {
525 uint8_t *buf1 = buf + ch * blocks * 8 + i * 8;
526 for (j = 0; j < 8; j += 2)
527 *dst++ = buf1[j] | (buf1[j + 1] << 4);
532 for (i = 0; i < blocks; i++) {
533 for (ch = 0; ch < avctx->channels; ch++) {
534 ADPCMChannelStatus *status = &c->status[ch];
535 const int16_t *smp = &samples_p[ch][1 + i * 8];
536 for (j = 0; j < 8; j += 2) {
537 uint8_t v = adpcm_ima_compress_sample(status, smp[j ]);
538 v |= adpcm_ima_compress_sample(status, smp[j + 1]) << 4;
546 case AV_CODEC_ID_ADPCM_IMA_QT:
549 init_put_bits(&pb, dst, pkt_size);
551 for (ch = 0; ch < avctx->channels; ch++) {
552 ADPCMChannelStatus *status = &c->status[ch];
553 put_bits(&pb, 9, (status->prev_sample & 0xFFFF) >> 7);
554 put_bits(&pb, 7, status->step_index);
555 if (avctx->trellis > 0) {
557 adpcm_compress_trellis(avctx, &samples_p[ch][0], buf, status,
559 for (i = 0; i < 64; i++)
560 put_bits(&pb, 4, buf[i ^ 1]);
561 status->prev_sample = status->predictor;
563 for (i = 0; i < 64; i += 2) {
565 t1 = adpcm_ima_qt_compress_sample(status, samples_p[ch][i ]);
566 t2 = adpcm_ima_qt_compress_sample(status, samples_p[ch][i + 1]);
567 put_bits(&pb, 4, t2);
568 put_bits(&pb, 4, t1);
576 case AV_CODEC_ID_ADPCM_IMA_SSI:
579 init_put_bits(&pb, dst, pkt_size);
581 av_assert0(avctx->trellis == 0);
583 for (i = 0; i < frame->nb_samples; i++) {
584 for (ch = 0; ch < avctx->channels; ch++) {
585 put_bits(&pb, 4, adpcm_ima_qt_compress_sample(c->status + ch, *samples++));
592 case AV_CODEC_ID_ADPCM_SWF:
595 init_put_bits(&pb, dst, pkt_size);
597 n = frame->nb_samples - 1;
599 // store AdpcmCodeSize
600 put_bits(&pb, 2, 2); // set 4-bit flash adpcm format
602 // init the encoder state
603 for (i = 0; i < avctx->channels; i++) {
604 // clip step so it fits 6 bits
605 c->status[i].step_index = av_clip_uintp2(c->status[i].step_index, 6);
606 put_sbits(&pb, 16, samples[i]);
607 put_bits(&pb, 6, c->status[i].step_index);
608 c->status[i].prev_sample = samples[i];
611 if (avctx->trellis > 0) {
612 if (!(buf = av_malloc(2 * n)))
613 return AVERROR(ENOMEM);
614 adpcm_compress_trellis(avctx, samples + avctx->channels, buf,
615 &c->status[0], n, avctx->channels);
616 if (avctx->channels == 2)
617 adpcm_compress_trellis(avctx, samples + avctx->channels + 1,
618 buf + n, &c->status[1], n,
620 for (i = 0; i < n; i++) {
621 put_bits(&pb, 4, buf[i]);
622 if (avctx->channels == 2)
623 put_bits(&pb, 4, buf[n + i]);
627 for (i = 1; i < frame->nb_samples; i++) {
628 put_bits(&pb, 4, adpcm_ima_compress_sample(&c->status[0],
629 samples[avctx->channels * i]));
630 if (avctx->channels == 2)
631 put_bits(&pb, 4, adpcm_ima_compress_sample(&c->status[1],
632 samples[2 * i + 1]));
638 case AV_CODEC_ID_ADPCM_MS:
639 for (i = 0; i < avctx->channels; i++) {
642 c->status[i].coeff1 = ff_adpcm_AdaptCoeff1[predictor];
643 c->status[i].coeff2 = ff_adpcm_AdaptCoeff2[predictor];
645 for (i = 0; i < avctx->channels; i++) {
646 if (c->status[i].idelta < 16)
647 c->status[i].idelta = 16;
648 bytestream_put_le16(&dst, c->status[i].idelta);
650 for (i = 0; i < avctx->channels; i++)
651 c->status[i].sample2= *samples++;
652 for (i = 0; i < avctx->channels; i++) {
653 c->status[i].sample1 = *samples++;
654 bytestream_put_le16(&dst, c->status[i].sample1);
656 for (i = 0; i < avctx->channels; i++)
657 bytestream_put_le16(&dst, c->status[i].sample2);
659 if (avctx->trellis > 0) {
660 n = avctx->block_align - 7 * avctx->channels;
661 if (!(buf = av_malloc(2 * n)))
662 return AVERROR(ENOMEM);
663 if (avctx->channels == 1) {
664 adpcm_compress_trellis(avctx, samples, buf, &c->status[0], n,
666 for (i = 0; i < n; i += 2)
667 *dst++ = (buf[i] << 4) | buf[i + 1];
669 adpcm_compress_trellis(avctx, samples, buf,
670 &c->status[0], n, avctx->channels);
671 adpcm_compress_trellis(avctx, samples + 1, buf + n,
672 &c->status[1], n, avctx->channels);
673 for (i = 0; i < n; i++)
674 *dst++ = (buf[i] << 4) | buf[n + i];
678 for (i = 7 * avctx->channels; i < avctx->block_align; i++) {
680 nibble = adpcm_ms_compress_sample(&c->status[ 0], *samples++) << 4;
681 nibble |= adpcm_ms_compress_sample(&c->status[st], *samples++);
686 case AV_CODEC_ID_ADPCM_YAMAHA:
687 n = frame->nb_samples / 2;
688 if (avctx->trellis > 0) {
689 if (!(buf = av_malloc(2 * n * 2)))
690 return AVERROR(ENOMEM);
692 if (avctx->channels == 1) {
693 adpcm_compress_trellis(avctx, samples, buf, &c->status[0], n,
695 for (i = 0; i < n; i += 2)
696 *dst++ = buf[i] | (buf[i + 1] << 4);
698 adpcm_compress_trellis(avctx, samples, buf,
699 &c->status[0], n, avctx->channels);
700 adpcm_compress_trellis(avctx, samples + 1, buf + n,
701 &c->status[1], n, avctx->channels);
702 for (i = 0; i < n; i++)
703 *dst++ = buf[i] | (buf[n + i] << 4);
707 for (n *= avctx->channels; n > 0; n--) {
709 nibble = adpcm_yamaha_compress_sample(&c->status[ 0], *samples++);
710 nibble |= adpcm_yamaha_compress_sample(&c->status[st], *samples++) << 4;
715 return AVERROR(EINVAL);
718 avpkt->size = pkt_size;
723 static const enum AVSampleFormat sample_fmts[] = {
724 AV_SAMPLE_FMT_S16, AV_SAMPLE_FMT_NONE
727 static const enum AVSampleFormat sample_fmts_p[] = {
728 AV_SAMPLE_FMT_S16P, AV_SAMPLE_FMT_NONE
731 #define ADPCM_ENCODER(id_, name_, sample_fmts_, capabilities_, long_name_) \
732 AVCodec ff_ ## name_ ## _encoder = { \
734 .long_name = NULL_IF_CONFIG_SMALL(long_name_), \
735 .type = AVMEDIA_TYPE_AUDIO, \
737 .priv_data_size = sizeof(ADPCMEncodeContext), \
738 .init = adpcm_encode_init, \
739 .encode2 = adpcm_encode_frame, \
740 .close = adpcm_encode_close, \
741 .sample_fmts = sample_fmts_, \
742 .capabilities = capabilities_, \
743 .caps_internal = FF_CODEC_CAP_INIT_CLEANUP, \
746 ADPCM_ENCODER(AV_CODEC_ID_ADPCM_IMA_QT, adpcm_ima_qt, sample_fmts_p, 0, "ADPCM IMA QuickTime");
747 ADPCM_ENCODER(AV_CODEC_ID_ADPCM_IMA_SSI, adpcm_ima_ssi, sample_fmts, AV_CODEC_CAP_SMALL_LAST_FRAME, "ADPCM IMA Simon & Schuster Interactive");
748 ADPCM_ENCODER(AV_CODEC_ID_ADPCM_IMA_WAV, adpcm_ima_wav, sample_fmts_p, 0, "ADPCM IMA WAV");
749 ADPCM_ENCODER(AV_CODEC_ID_ADPCM_MS, adpcm_ms, sample_fmts, 0, "ADPCM Microsoft");
750 ADPCM_ENCODER(AV_CODEC_ID_ADPCM_SWF, adpcm_swf, sample_fmts, 0, "ADPCM Shockwave Flash");
751 ADPCM_ENCODER(AV_CODEC_ID_ADPCM_YAMAHA, adpcm_yamaha, sample_fmts, 0, "ADPCM Yamaha");