2 * Copyright (c) 2001-2003 The ffmpeg Project
4 * This file is part of FFmpeg.
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.
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13 * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
14 * Lesser General Public License for more details.
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
23 #include "bytestream.h"
25 #include "adpcm_data.h"
31 * First version by Francois Revol (revol@free.fr)
32 * Fringe ADPCM codecs (e.g., DK3, DK4, Westwood)
33 * by Mike Melanson (melanson@pcisys.net)
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_close(AVCodecContext *avctx);
63 static av_cold int adpcm_encode_init(AVCodecContext *avctx)
65 ADPCMEncodeContext *s = avctx->priv_data;
68 int ret = AVERROR(ENOMEM);
70 if (avctx->channels > 2) {
71 av_log(avctx, AV_LOG_ERROR, "only stereo or mono is supported\n");
72 return AVERROR(EINVAL);
75 if (avctx->trellis && (unsigned)avctx->trellis > 16U) {
76 av_log(avctx, AV_LOG_ERROR, "invalid trellis size\n");
77 return AVERROR(EINVAL);
81 int frontier = 1 << avctx->trellis;
82 int max_paths = frontier * FREEZE_INTERVAL;
83 FF_ALLOC_OR_GOTO(avctx, s->paths,
84 max_paths * sizeof(*s->paths), error);
85 FF_ALLOC_OR_GOTO(avctx, s->node_buf,
86 2 * frontier * sizeof(*s->node_buf), error);
87 FF_ALLOC_OR_GOTO(avctx, s->nodep_buf,
88 2 * frontier * sizeof(*s->nodep_buf), error);
89 FF_ALLOC_OR_GOTO(avctx, s->trellis_hash,
90 65536 * sizeof(*s->trellis_hash), error);
93 avctx->bits_per_coded_sample = av_get_bits_per_sample(avctx->codec->id);
95 switch (avctx->codec->id) {
96 case AV_CODEC_ID_ADPCM_IMA_WAV:
97 /* each 16 bits sample gives one nibble
98 and we have 4 bytes per channel overhead */
99 avctx->frame_size = (BLKSIZE - 4 * avctx->channels) * 8 /
100 (4 * avctx->channels) + 1;
101 /* seems frame_size isn't taken into account...
102 have to buffer the samples :-( */
103 avctx->block_align = BLKSIZE;
104 avctx->bits_per_coded_sample = 4;
106 case AV_CODEC_ID_ADPCM_IMA_QT:
107 avctx->frame_size = 64;
108 avctx->block_align = 34 * avctx->channels;
110 case AV_CODEC_ID_ADPCM_MS:
111 /* each 16 bits sample gives one nibble
112 and we have 7 bytes per channel overhead */
113 avctx->frame_size = (BLKSIZE - 7 * avctx->channels) * 2 / avctx->channels + 2;
114 avctx->bits_per_coded_sample = 4;
115 avctx->block_align = BLKSIZE;
116 if (!(avctx->extradata = av_malloc(32 + FF_INPUT_BUFFER_PADDING_SIZE)))
118 avctx->extradata_size = 32;
119 extradata = avctx->extradata;
120 bytestream_put_le16(&extradata, avctx->frame_size);
121 bytestream_put_le16(&extradata, 7); /* wNumCoef */
122 for (i = 0; i < 7; i++) {
123 bytestream_put_le16(&extradata, ff_adpcm_AdaptCoeff1[i] * 4);
124 bytestream_put_le16(&extradata, ff_adpcm_AdaptCoeff2[i] * 4);
127 case AV_CODEC_ID_ADPCM_YAMAHA:
128 avctx->frame_size = BLKSIZE * 2 / avctx->channels;
129 avctx->block_align = BLKSIZE;
131 case AV_CODEC_ID_ADPCM_SWF:
132 if (avctx->sample_rate != 11025 &&
133 avctx->sample_rate != 22050 &&
134 avctx->sample_rate != 44100) {
135 av_log(avctx, AV_LOG_ERROR, "Sample rate must be 11025, "
137 ret = AVERROR(EINVAL);
140 avctx->frame_size = 512 * (avctx->sample_rate / 11025);
143 ret = AVERROR(EINVAL);
147 #if FF_API_OLD_ENCODE_AUDIO
148 if (!(avctx->coded_frame = avcodec_alloc_frame()))
154 adpcm_encode_close(avctx);
158 static av_cold int adpcm_encode_close(AVCodecContext *avctx)
160 ADPCMEncodeContext *s = avctx->priv_data;
161 #if FF_API_OLD_ENCODE_AUDIO
162 av_freep(&avctx->coded_frame);
165 av_freep(&s->node_buf);
166 av_freep(&s->nodep_buf);
167 av_freep(&s->trellis_hash);
173 static inline uint8_t adpcm_ima_compress_sample(ADPCMChannelStatus *c,
176 int delta = sample - c->prev_sample;
177 int nibble = FFMIN(7, abs(delta) * 4 /
178 ff_adpcm_step_table[c->step_index]) + (delta < 0) * 8;
179 c->prev_sample += ((ff_adpcm_step_table[c->step_index] *
180 ff_adpcm_yamaha_difflookup[nibble]) / 8);
181 c->prev_sample = av_clip_int16(c->prev_sample);
182 c->step_index = av_clip(c->step_index + ff_adpcm_index_table[nibble], 0, 88);
186 static inline uint8_t adpcm_ima_qt_compress_sample(ADPCMChannelStatus *c,
189 int delta = sample - c->prev_sample;
190 int diff, step = ff_adpcm_step_table[c->step_index];
191 int nibble = 8*(delta < 0);
194 diff = delta + (step >> 3);
213 c->prev_sample -= diff;
215 c->prev_sample += diff;
217 c->prev_sample = av_clip_int16(c->prev_sample);
218 c->step_index = av_clip(c->step_index + ff_adpcm_index_table[nibble], 0, 88);
223 static inline uint8_t adpcm_ms_compress_sample(ADPCMChannelStatus *c,
226 int predictor, nibble, bias;
228 predictor = (((c->sample1) * (c->coeff1)) +
229 (( c->sample2) * (c->coeff2))) / 64;
231 nibble = sample - predictor;
233 bias = c->idelta / 2;
235 bias = -c->idelta / 2;
237 nibble = (nibble + bias) / c->idelta;
238 nibble = av_clip(nibble, -8, 7) & 0x0F;
240 predictor += ((nibble & 0x08) ? (nibble - 0x10) : nibble) * c->idelta;
242 c->sample2 = c->sample1;
243 c->sample1 = av_clip_int16(predictor);
245 c->idelta = (ff_adpcm_AdaptationTable[nibble] * c->idelta) >> 8;
252 static inline uint8_t adpcm_yamaha_compress_sample(ADPCMChannelStatus *c,
262 delta = sample - c->predictor;
264 nibble = FFMIN(7, abs(delta) * 4 / c->step) + (delta < 0) * 8;
266 c->predictor += ((c->step * ff_adpcm_yamaha_difflookup[nibble]) / 8);
267 c->predictor = av_clip_int16(c->predictor);
268 c->step = (c->step * ff_adpcm_yamaha_indexscale[nibble]) >> 8;
269 c->step = av_clip(c->step, 127, 24567);
274 static void adpcm_compress_trellis(AVCodecContext *avctx,
275 const int16_t *samples, uint8_t *dst,
276 ADPCMChannelStatus *c, int n, int stride)
278 //FIXME 6% faster if frontier is a compile-time constant
279 ADPCMEncodeContext *s = avctx->priv_data;
280 const int frontier = 1 << avctx->trellis;
281 const int version = avctx->codec->id;
282 TrellisPath *paths = s->paths, *p;
283 TrellisNode *node_buf = s->node_buf;
284 TrellisNode **nodep_buf = s->nodep_buf;
285 TrellisNode **nodes = nodep_buf; // nodes[] is always sorted by .ssd
286 TrellisNode **nodes_next = nodep_buf + frontier;
287 int pathn = 0, froze = -1, i, j, k, generation = 0;
288 uint8_t *hash = s->trellis_hash;
289 memset(hash, 0xff, 65536 * sizeof(*hash));
291 memset(nodep_buf, 0, 2 * frontier * sizeof(*nodep_buf));
292 nodes[0] = node_buf + frontier;
295 nodes[0]->step = c->step_index;
296 nodes[0]->sample1 = c->sample1;
297 nodes[0]->sample2 = c->sample2;
298 if (version == AV_CODEC_ID_ADPCM_IMA_WAV ||
299 version == AV_CODEC_ID_ADPCM_IMA_QT ||
300 version == AV_CODEC_ID_ADPCM_SWF)
301 nodes[0]->sample1 = c->prev_sample;
302 if (version == AV_CODEC_ID_ADPCM_MS)
303 nodes[0]->step = c->idelta;
304 if (version == AV_CODEC_ID_ADPCM_YAMAHA) {
306 nodes[0]->step = 127;
307 nodes[0]->sample1 = 0;
309 nodes[0]->step = c->step;
310 nodes[0]->sample1 = c->predictor;
314 for (i = 0; i < n; i++) {
315 TrellisNode *t = node_buf + frontier*(i&1);
317 int sample = samples[i * stride];
319 memset(nodes_next, 0, frontier * sizeof(TrellisNode*));
320 for (j = 0; j < frontier && nodes[j]; j++) {
321 // higher j have higher ssd already, so they're likely
322 // to yield a suboptimal next sample too
323 const int range = (j < frontier / 2) ? 1 : 0;
324 const int step = nodes[j]->step;
326 if (version == AV_CODEC_ID_ADPCM_MS) {
327 const int predictor = ((nodes[j]->sample1 * c->coeff1) +
328 (nodes[j]->sample2 * c->coeff2)) / 64;
329 const int div = (sample - predictor) / step;
330 const int nmin = av_clip(div-range, -8, 6);
331 const int nmax = av_clip(div+range, -7, 7);
332 for (nidx = nmin; nidx <= nmax; nidx++) {
333 const int nibble = nidx & 0xf;
334 int dec_sample = predictor + nidx * step;
335 #define STORE_NODE(NAME, STEP_INDEX)\
341 dec_sample = av_clip_int16(dec_sample);\
342 d = sample - dec_sample;\
343 ssd = nodes[j]->ssd + d*d;\
344 /* Check for wraparound, skip such samples completely. \
345 * Note, changing ssd to a 64 bit variable would be \
346 * simpler, avoiding this check, but it's slower on \
347 * x86 32 bit at the moment. */\
348 if (ssd < nodes[j]->ssd)\
350 /* Collapse any two states with the same previous sample value. \
351 * One could also distinguish states by step and by 2nd to last
352 * sample, but the effects of that are negligible.
353 * Since nodes in the previous generation are iterated
354 * through a heap, they're roughly ordered from better to
355 * worse, but not strictly ordered. Therefore, an earlier
356 * node with the same sample value is better in most cases
357 * (and thus the current is skipped), but not strictly
358 * in all cases. Only skipping samples where ssd >=
359 * ssd of the earlier node with the same sample gives
360 * slightly worse quality, though, for some reason. */ \
361 h = &hash[(uint16_t) dec_sample];\
362 if (*h == generation)\
364 if (heap_pos < frontier) {\
367 /* Try to replace one of the leaf nodes with the new \
368 * one, but try a different slot each time. */\
369 pos = (frontier >> 1) +\
370 (heap_pos & ((frontier >> 1) - 1));\
371 if (ssd > nodes_next[pos]->ssd)\
376 u = nodes_next[pos];\
378 av_assert1(pathn < FREEZE_INTERVAL << avctx->trellis);\
380 nodes_next[pos] = u;\
384 u->step = STEP_INDEX;\
385 u->sample2 = nodes[j]->sample1;\
386 u->sample1 = dec_sample;\
387 paths[u->path].nibble = nibble;\
388 paths[u->path].prev = nodes[j]->path;\
389 /* Sift the newly inserted node up in the heap to \
390 * restore the heap property. */\
392 int parent = (pos - 1) >> 1;\
393 if (nodes_next[parent]->ssd <= ssd)\
395 FFSWAP(TrellisNode*, nodes_next[parent], nodes_next[pos]);\
399 STORE_NODE(ms, FFMAX(16,
400 (ff_adpcm_AdaptationTable[nibble] * step) >> 8));
402 } else if (version == AV_CODEC_ID_ADPCM_IMA_WAV ||
403 version == AV_CODEC_ID_ADPCM_IMA_QT ||
404 version == AV_CODEC_ID_ADPCM_SWF) {
405 #define LOOP_NODES(NAME, STEP_TABLE, STEP_INDEX)\
406 const int predictor = nodes[j]->sample1;\
407 const int div = (sample - predictor) * 4 / STEP_TABLE;\
408 int nmin = av_clip(div - range, -7, 6);\
409 int nmax = av_clip(div + range, -6, 7);\
411 nmin--; /* distinguish -0 from +0 */\
414 for (nidx = nmin; nidx <= nmax; nidx++) {\
415 const int nibble = nidx < 0 ? 7 - nidx : nidx;\
416 int dec_sample = predictor +\
418 ff_adpcm_yamaha_difflookup[nibble]) / 8;\
419 STORE_NODE(NAME, STEP_INDEX);\
421 LOOP_NODES(ima, ff_adpcm_step_table[step],
422 av_clip(step + ff_adpcm_index_table[nibble], 0, 88));
423 } else { //AV_CODEC_ID_ADPCM_YAMAHA
424 LOOP_NODES(yamaha, step,
425 av_clip((step * ff_adpcm_yamaha_indexscale[nibble]) >> 8,
437 if (generation == 255) {
438 memset(hash, 0xff, 65536 * sizeof(*hash));
443 if (nodes[0]->ssd > (1 << 28)) {
444 for (j = 1; j < frontier && nodes[j]; j++)
445 nodes[j]->ssd -= nodes[0]->ssd;
449 // merge old paths to save memory
450 if (i == froze + FREEZE_INTERVAL) {
451 p = &paths[nodes[0]->path];
452 for (k = i; k > froze; k--) {
458 // other nodes might use paths that don't coincide with the frozen one.
459 // checking which nodes do so is too slow, so just kill them all.
460 // this also slightly improves quality, but I don't know why.
461 memset(nodes + 1, 0, (frontier - 1) * sizeof(TrellisNode*));
465 p = &paths[nodes[0]->path];
466 for (i = n - 1; i > froze; i--) {
471 c->predictor = nodes[0]->sample1;
472 c->sample1 = nodes[0]->sample1;
473 c->sample2 = nodes[0]->sample2;
474 c->step_index = nodes[0]->step;
475 c->step = nodes[0]->step;
476 c->idelta = nodes[0]->step;
479 static int adpcm_encode_frame(AVCodecContext *avctx, AVPacket *avpkt,
480 const AVFrame *frame, int *got_packet_ptr)
482 int n, i, ch, st, pkt_size, ret;
483 const int16_t *samples;
486 ADPCMEncodeContext *c = avctx->priv_data;
489 samples = (const int16_t *)frame->data[0];
490 samples_p = (int16_t **)frame->extended_data;
491 st = avctx->channels == 2;
493 if (avctx->codec_id == AV_CODEC_ID_ADPCM_SWF)
494 pkt_size = (2 + avctx->channels * (22 + 4 * (frame->nb_samples - 1)) + 7) / 8;
496 pkt_size = avctx->block_align;
497 if ((ret = ff_alloc_packet2(avctx, avpkt, pkt_size)))
501 switch(avctx->codec->id) {
502 case AV_CODEC_ID_ADPCM_IMA_WAV:
506 blocks = (frame->nb_samples - 1) / 8;
508 for (ch = 0; ch < avctx->channels; ch++) {
509 ADPCMChannelStatus *status = &c->status[ch];
510 status->prev_sample = samples_p[ch][0];
511 /* status->step_index = 0;
512 XXX: not sure how to init the state machine */
513 bytestream_put_le16(&dst, status->prev_sample);
514 *dst++ = status->step_index;
515 *dst++ = 0; /* unknown */
518 /* stereo: 4 bytes (8 samples) for left, 4 bytes for right */
519 if (avctx->trellis > 0) {
520 FF_ALLOC_OR_GOTO(avctx, buf, avctx->channels * blocks * 8, error);
521 for (ch = 0; ch < avctx->channels; ch++) {
522 adpcm_compress_trellis(avctx, &samples_p[ch][1],
523 buf + ch * blocks * 8, &c->status[ch],
526 for (i = 0; i < blocks; i++) {
527 for (ch = 0; ch < avctx->channels; ch++) {
528 uint8_t *buf1 = buf + ch * blocks * 8 + i * 8;
529 for (j = 0; j < 8; j += 2)
530 *dst++ = buf1[j] | (buf1[j + 1] << 4);
535 for (i = 0; i < blocks; i++) {
536 for (ch = 0; ch < avctx->channels; ch++) {
537 ADPCMChannelStatus *status = &c->status[ch];
538 const int16_t *smp = &samples_p[ch][1 + i * 8];
539 for (j = 0; j < 8; j += 2) {
540 uint8_t v = adpcm_ima_compress_sample(status, smp[j ]);
541 v |= adpcm_ima_compress_sample(status, smp[j + 1]) << 4;
549 case AV_CODEC_ID_ADPCM_IMA_QT:
552 init_put_bits(&pb, dst, pkt_size * 8);
554 for (ch = 0; ch < avctx->channels; ch++) {
555 ADPCMChannelStatus *status = &c->status[ch];
556 put_bits(&pb, 9, (status->prev_sample & 0xFFFF) >> 7);
557 put_bits(&pb, 7, status->step_index);
558 if (avctx->trellis > 0) {
560 adpcm_compress_trellis(avctx, &samples_p[ch][1], buf, status,
562 for (i = 0; i < 64; i++)
563 put_bits(&pb, 4, buf[i ^ 1]);
565 for (i = 0; i < 64; i += 2) {
567 t1 = adpcm_ima_qt_compress_sample(status, samples_p[ch][i ]);
568 t2 = adpcm_ima_qt_compress_sample(status, samples_p[ch][i + 1]);
569 put_bits(&pb, 4, t2);
570 put_bits(&pb, 4, t1);
578 case AV_CODEC_ID_ADPCM_SWF:
581 init_put_bits(&pb, dst, pkt_size * 8);
583 n = frame->nb_samples - 1;
585 // store AdpcmCodeSize
586 put_bits(&pb, 2, 2); // set 4-bit flash adpcm format
588 // init the encoder state
589 for (i = 0; i < avctx->channels; i++) {
590 // clip step so it fits 6 bits
591 c->status[i].step_index = av_clip(c->status[i].step_index, 0, 63);
592 put_sbits(&pb, 16, samples[i]);
593 put_bits(&pb, 6, c->status[i].step_index);
594 c->status[i].prev_sample = samples[i];
597 if (avctx->trellis > 0) {
598 FF_ALLOC_OR_GOTO(avctx, buf, 2 * n, error);
599 adpcm_compress_trellis(avctx, samples + avctx->channels, buf,
600 &c->status[0], n, avctx->channels);
601 if (avctx->channels == 2)
602 adpcm_compress_trellis(avctx, samples + avctx->channels + 1,
603 buf + n, &c->status[1], n,
605 for (i = 0; i < n; i++) {
606 put_bits(&pb, 4, buf[i]);
607 if (avctx->channels == 2)
608 put_bits(&pb, 4, buf[n + i]);
612 for (i = 1; i < frame->nb_samples; i++) {
613 put_bits(&pb, 4, adpcm_ima_compress_sample(&c->status[0],
614 samples[avctx->channels * i]));
615 if (avctx->channels == 2)
616 put_bits(&pb, 4, adpcm_ima_compress_sample(&c->status[1],
617 samples[2 * i + 1]));
623 case AV_CODEC_ID_ADPCM_MS:
624 for (i = 0; i < avctx->channels; i++) {
627 c->status[i].coeff1 = ff_adpcm_AdaptCoeff1[predictor];
628 c->status[i].coeff2 = ff_adpcm_AdaptCoeff2[predictor];
630 for (i = 0; i < avctx->channels; i++) {
631 if (c->status[i].idelta < 16)
632 c->status[i].idelta = 16;
633 bytestream_put_le16(&dst, c->status[i].idelta);
635 for (i = 0; i < avctx->channels; i++)
636 c->status[i].sample2= *samples++;
637 for (i = 0; i < avctx->channels; i++) {
638 c->status[i].sample1 = *samples++;
639 bytestream_put_le16(&dst, c->status[i].sample1);
641 for (i = 0; i < avctx->channels; i++)
642 bytestream_put_le16(&dst, c->status[i].sample2);
644 if (avctx->trellis > 0) {
645 n = avctx->block_align - 7 * avctx->channels;
646 FF_ALLOC_OR_GOTO(avctx, buf, 2 * n, error);
647 if (avctx->channels == 1) {
648 adpcm_compress_trellis(avctx, samples, buf, &c->status[0], n,
650 for (i = 0; i < n; i += 2)
651 *dst++ = (buf[i] << 4) | buf[i + 1];
653 adpcm_compress_trellis(avctx, samples, buf,
654 &c->status[0], n, avctx->channels);
655 adpcm_compress_trellis(avctx, samples + 1, buf + n,
656 &c->status[1], n, avctx->channels);
657 for (i = 0; i < n; i++)
658 *dst++ = (buf[i] << 4) | buf[n + i];
662 for (i = 7 * avctx->channels; i < avctx->block_align; i++) {
664 nibble = adpcm_ms_compress_sample(&c->status[ 0], *samples++) << 4;
665 nibble |= adpcm_ms_compress_sample(&c->status[st], *samples++);
670 case AV_CODEC_ID_ADPCM_YAMAHA:
671 n = frame->nb_samples / 2;
672 if (avctx->trellis > 0) {
673 FF_ALLOC_OR_GOTO(avctx, buf, 2 * n * 2, error);
675 if (avctx->channels == 1) {
676 adpcm_compress_trellis(avctx, samples, buf, &c->status[0], n,
678 for (i = 0; i < n; i += 2)
679 *dst++ = buf[i] | (buf[i + 1] << 4);
681 adpcm_compress_trellis(avctx, samples, buf,
682 &c->status[0], n, avctx->channels);
683 adpcm_compress_trellis(avctx, samples + 1, buf + n,
684 &c->status[1], n, avctx->channels);
685 for (i = 0; i < n; i++)
686 *dst++ = buf[i] | (buf[n + i] << 4);
690 for (n *= avctx->channels; n > 0; n--) {
692 nibble = adpcm_yamaha_compress_sample(&c->status[ 0], *samples++);
693 nibble |= adpcm_yamaha_compress_sample(&c->status[st], *samples++) << 4;
698 return AVERROR(EINVAL);
701 avpkt->size = pkt_size;
705 return AVERROR(ENOMEM);
708 static const enum AVSampleFormat sample_fmts[] = {
709 AV_SAMPLE_FMT_S16, AV_SAMPLE_FMT_NONE
712 static const enum AVSampleFormat sample_fmts_p[] = {
713 AV_SAMPLE_FMT_S16P, AV_SAMPLE_FMT_NONE
716 #define ADPCM_ENCODER(id_, name_, sample_fmts_, long_name_) \
717 AVCodec ff_ ## name_ ## _encoder = { \
719 .type = AVMEDIA_TYPE_AUDIO, \
721 .priv_data_size = sizeof(ADPCMEncodeContext), \
722 .init = adpcm_encode_init, \
723 .encode2 = adpcm_encode_frame, \
724 .close = adpcm_encode_close, \
725 .sample_fmts = sample_fmts_, \
726 .long_name = NULL_IF_CONFIG_SMALL(long_name_), \
729 ADPCM_ENCODER(AV_CODEC_ID_ADPCM_IMA_QT, adpcm_ima_qt, sample_fmts_p, "ADPCM IMA QuickTime");
730 ADPCM_ENCODER(AV_CODEC_ID_ADPCM_IMA_WAV, adpcm_ima_wav, sample_fmts_p, "ADPCM IMA WAV");
731 ADPCM_ENCODER(AV_CODEC_ID_ADPCM_MS, adpcm_ms, sample_fmts, "ADPCM Microsoft");
732 ADPCM_ENCODER(AV_CODEC_ID_ADPCM_SWF, adpcm_swf, sample_fmts, "ADPCM Shockwave Flash");
733 ADPCM_ENCODER(AV_CODEC_ID_ADPCM_YAMAHA, adpcm_yamaha, sample_fmts, "ADPCM Yamaha");