2 * Copyright (c) 2001-2003 The ffmpeg Project
4 * This file is part of Libav.
6 * Libav 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.
11 * Libav 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.
16 * You should have received a copy of the GNU Lesser General Public
17 * License along with Libav; if not, write to the Free Software
18 * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
24 #include "bytestream.h"
26 #include "adpcm_data.h"
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)
36 * See ADPCM decoder reference documents for codec information.
39 typedef struct TrellisPath {
44 typedef struct TrellisNode {
52 typedef struct ADPCMEncodeContext {
53 ADPCMChannelStatus status[6];
55 TrellisNode *node_buf;
56 TrellisNode **nodep_buf;
57 uint8_t *trellis_hash;
60 #define FREEZE_INTERVAL 128
62 static av_cold int adpcm_encode_init(AVCodecContext *avctx)
64 ADPCMEncodeContext *s = avctx->priv_data;
67 int ret = AVERROR(ENOMEM);
69 if (avctx->channels > 2) {
70 av_log(avctx, AV_LOG_ERROR, "only stereo or mono is supported\n");
71 return AVERROR(EINVAL);
74 if (avctx->trellis && (unsigned)avctx->trellis > 16U) {
75 av_log(avctx, AV_LOG_ERROR, "invalid trellis size\n");
76 return AVERROR(EINVAL);
80 int frontier = 1 << avctx->trellis;
81 int max_paths = frontier * FREEZE_INTERVAL;
82 FF_ALLOC_OR_GOTO(avctx, s->paths,
83 max_paths * sizeof(*s->paths), error);
84 FF_ALLOC_OR_GOTO(avctx, s->node_buf,
85 2 * frontier * sizeof(*s->node_buf), error);
86 FF_ALLOC_OR_GOTO(avctx, s->nodep_buf,
87 2 * frontier * sizeof(*s->nodep_buf), error);
88 FF_ALLOC_OR_GOTO(avctx, s->trellis_hash,
89 65536 * sizeof(*s->trellis_hash), error);
92 avctx->bits_per_coded_sample = av_get_bits_per_sample(avctx->codec->id);
94 switch (avctx->codec->id) {
95 case AV_CODEC_ID_ADPCM_IMA_WAV:
96 /* each 16 bits sample gives one nibble
97 and we have 4 bytes per channel overhead */
98 avctx->frame_size = (BLKSIZE - 4 * avctx->channels) * 8 /
99 (4 * avctx->channels) + 1;
100 /* seems frame_size isn't taken into account...
101 have to buffer the samples :-( */
102 avctx->block_align = BLKSIZE;
104 case AV_CODEC_ID_ADPCM_IMA_QT:
105 avctx->frame_size = 64;
106 avctx->block_align = 34 * avctx->channels;
108 case AV_CODEC_ID_ADPCM_MS:
109 /* each 16 bits sample gives one nibble
110 and we have 7 bytes per channel overhead */
111 avctx->frame_size = (BLKSIZE - 7 * avctx->channels) * 2 /
113 avctx->block_align = BLKSIZE;
114 if (!(avctx->extradata = av_malloc(32 + FF_INPUT_BUFFER_PADDING_SIZE)))
116 avctx->extradata_size = 32;
117 extradata = avctx->extradata;
118 bytestream_put_le16(&extradata, avctx->frame_size);
119 bytestream_put_le16(&extradata, 7); /* wNumCoef */
120 for (i = 0; i < 7; i++) {
121 bytestream_put_le16(&extradata, ff_adpcm_AdaptCoeff1[i] * 4);
122 bytestream_put_le16(&extradata, ff_adpcm_AdaptCoeff2[i] * 4);
125 case AV_CODEC_ID_ADPCM_YAMAHA:
126 avctx->frame_size = BLKSIZE * 2 / avctx->channels;
127 avctx->block_align = BLKSIZE;
129 case AV_CODEC_ID_ADPCM_SWF:
130 if (avctx->sample_rate != 11025 &&
131 avctx->sample_rate != 22050 &&
132 avctx->sample_rate != 44100) {
133 av_log(avctx, AV_LOG_ERROR, "Sample rate must be 11025, "
135 ret = AVERROR(EINVAL);
138 avctx->frame_size = 512 * (avctx->sample_rate / 11025);
141 ret = AVERROR(EINVAL);
145 #if FF_API_OLD_ENCODE_AUDIO
146 if (!(avctx->coded_frame = avcodec_alloc_frame()))
153 av_freep(&s->node_buf);
154 av_freep(&s->nodep_buf);
155 av_freep(&s->trellis_hash);
159 static av_cold int adpcm_encode_close(AVCodecContext *avctx)
161 ADPCMEncodeContext *s = avctx->priv_data;
162 #if FF_API_OLD_ENCODE_AUDIO
163 av_freep(&avctx->coded_frame);
166 av_freep(&s->node_buf);
167 av_freep(&s->nodep_buf);
168 av_freep(&s->trellis_hash);
174 static inline uint8_t adpcm_ima_compress_sample(ADPCMChannelStatus *c,
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);
187 static inline uint8_t adpcm_ima_qt_compress_sample(ADPCMChannelStatus *c,
190 int delta = sample - c->prev_sample;
191 int mask, step = ff_adpcm_step_table[c->step_index];
192 int diff = step >> 3;
200 for (mask = 4; mask;) {
211 c->prev_sample -= diff;
213 c->prev_sample += diff;
215 c->prev_sample = av_clip_int16(c->prev_sample);
216 c->step_index = av_clip(c->step_index + ff_adpcm_index_table[nibble], 0, 88);
221 static inline uint8_t adpcm_ms_compress_sample(ADPCMChannelStatus *c,
224 int predictor, nibble, bias;
226 predictor = (((c->sample1) * (c->coeff1)) +
227 (( c->sample2) * (c->coeff2))) / 64;
229 nibble = sample - predictor;
231 bias = c->idelta / 2;
233 bias = -c->idelta / 2;
235 nibble = (nibble + bias) / c->idelta;
236 nibble = av_clip(nibble, -8, 7) & 0x0F;
238 predictor += ((nibble & 0x08) ? (nibble - 0x10) : nibble) * c->idelta;
240 c->sample2 = c->sample1;
241 c->sample1 = av_clip_int16(predictor);
243 c->idelta = (ff_adpcm_AdaptationTable[nibble] * c->idelta) >> 8;
250 static inline uint8_t adpcm_yamaha_compress_sample(ADPCMChannelStatus *c,
260 delta = sample - c->predictor;
262 nibble = FFMIN(7, abs(delta) * 4 / c->step) + (delta < 0) * 8;
264 c->predictor += ((c->step * ff_adpcm_yamaha_difflookup[nibble]) / 8);
265 c->predictor = av_clip_int16(c->predictor);
266 c->step = (c->step * ff_adpcm_yamaha_indexscale[nibble]) >> 8;
267 c->step = av_clip(c->step, 127, 24567);
272 static void adpcm_compress_trellis(AVCodecContext *avctx,
273 const int16_t *samples, uint8_t *dst,
274 ADPCMChannelStatus *c, int n, int stride)
276 //FIXME 6% faster if frontier is a compile-time constant
277 ADPCMEncodeContext *s = avctx->priv_data;
278 const int frontier = 1 << avctx->trellis;
279 const int version = avctx->codec->id;
280 TrellisPath *paths = s->paths, *p;
281 TrellisNode *node_buf = s->node_buf;
282 TrellisNode **nodep_buf = s->nodep_buf;
283 TrellisNode **nodes = nodep_buf; // nodes[] is always sorted by .ssd
284 TrellisNode **nodes_next = nodep_buf + frontier;
285 int pathn = 0, froze = -1, i, j, k, generation = 0;
286 uint8_t *hash = s->trellis_hash;
287 memset(hash, 0xff, 65536 * sizeof(*hash));
289 memset(nodep_buf, 0, 2 * frontier * sizeof(*nodep_buf));
290 nodes[0] = node_buf + frontier;
293 nodes[0]->step = c->step_index;
294 nodes[0]->sample1 = c->sample1;
295 nodes[0]->sample2 = c->sample2;
296 if (version == AV_CODEC_ID_ADPCM_IMA_WAV ||
297 version == AV_CODEC_ID_ADPCM_IMA_QT ||
298 version == AV_CODEC_ID_ADPCM_SWF)
299 nodes[0]->sample1 = c->prev_sample;
300 if (version == AV_CODEC_ID_ADPCM_MS)
301 nodes[0]->step = c->idelta;
302 if (version == AV_CODEC_ID_ADPCM_YAMAHA) {
304 nodes[0]->step = 127;
305 nodes[0]->sample1 = 0;
307 nodes[0]->step = c->step;
308 nodes[0]->sample1 = c->predictor;
312 for (i = 0; i < n; i++) {
313 TrellisNode *t = node_buf + frontier*(i&1);
315 int sample = samples[i * stride];
317 memset(nodes_next, 0, frontier * sizeof(TrellisNode*));
318 for (j = 0; j < frontier && nodes[j]; j++) {
319 // higher j have higher ssd already, so they're likely
320 // to yield a suboptimal next sample too
321 const int range = (j < frontier / 2) ? 1 : 0;
322 const int step = nodes[j]->step;
324 if (version == AV_CODEC_ID_ADPCM_MS) {
325 const int predictor = ((nodes[j]->sample1 * c->coeff1) +
326 (nodes[j]->sample2 * c->coeff2)) / 64;
327 const int div = (sample - predictor) / step;
328 const int nmin = av_clip(div-range, -8, 6);
329 const int nmax = av_clip(div+range, -7, 7);
330 for (nidx = nmin; nidx <= nmax; nidx++) {
331 const int nibble = nidx & 0xf;
332 int dec_sample = predictor + nidx * step;
333 #define STORE_NODE(NAME, STEP_INDEX)\
339 dec_sample = av_clip_int16(dec_sample);\
340 d = sample - dec_sample;\
341 ssd = nodes[j]->ssd + d*d;\
342 /* Check for wraparound, skip such samples completely. \
343 * Note, changing ssd to a 64 bit variable would be \
344 * simpler, avoiding this check, but it's slower on \
345 * x86 32 bit at the moment. */\
346 if (ssd < nodes[j]->ssd)\
348 /* Collapse any two states with the same previous sample value. \
349 * One could also distinguish states by step and by 2nd to last
350 * sample, but the effects of that are negligible.
351 * Since nodes in the previous generation are iterated
352 * through a heap, they're roughly ordered from better to
353 * worse, but not strictly ordered. Therefore, an earlier
354 * node with the same sample value is better in most cases
355 * (and thus the current is skipped), but not strictly
356 * in all cases. Only skipping samples where ssd >=
357 * ssd of the earlier node with the same sample gives
358 * slightly worse quality, though, for some reason. */ \
359 h = &hash[(uint16_t) dec_sample];\
360 if (*h == generation)\
362 if (heap_pos < frontier) {\
365 /* Try to replace one of the leaf nodes with the new \
366 * one, but try a different slot each time. */\
367 pos = (frontier >> 1) +\
368 (heap_pos & ((frontier >> 1) - 1));\
369 if (ssd > nodes_next[pos]->ssd)\
374 u = nodes_next[pos];\
376 assert(pathn < FREEZE_INTERVAL << avctx->trellis);\
378 nodes_next[pos] = u;\
382 u->step = STEP_INDEX;\
383 u->sample2 = nodes[j]->sample1;\
384 u->sample1 = dec_sample;\
385 paths[u->path].nibble = nibble;\
386 paths[u->path].prev = nodes[j]->path;\
387 /* Sift the newly inserted node up in the heap to \
388 * restore the heap property. */\
390 int parent = (pos - 1) >> 1;\
391 if (nodes_next[parent]->ssd <= ssd)\
393 FFSWAP(TrellisNode*, nodes_next[parent], nodes_next[pos]);\
397 STORE_NODE(ms, FFMAX(16,
398 (ff_adpcm_AdaptationTable[nibble] * step) >> 8));
400 } else if (version == AV_CODEC_ID_ADPCM_IMA_WAV ||
401 version == AV_CODEC_ID_ADPCM_IMA_QT ||
402 version == AV_CODEC_ID_ADPCM_SWF) {
403 #define LOOP_NODES(NAME, STEP_TABLE, STEP_INDEX)\
404 const int predictor = nodes[j]->sample1;\
405 const int div = (sample - predictor) * 4 / STEP_TABLE;\
406 int nmin = av_clip(div - range, -7, 6);\
407 int nmax = av_clip(div + range, -6, 7);\
409 nmin--; /* distinguish -0 from +0 */\
412 for (nidx = nmin; nidx <= nmax; nidx++) {\
413 const int nibble = nidx < 0 ? 7 - nidx : nidx;\
414 int dec_sample = predictor +\
416 ff_adpcm_yamaha_difflookup[nibble]) / 8;\
417 STORE_NODE(NAME, STEP_INDEX);\
419 LOOP_NODES(ima, ff_adpcm_step_table[step],
420 av_clip(step + ff_adpcm_index_table[nibble], 0, 88));
421 } else { //AV_CODEC_ID_ADPCM_YAMAHA
422 LOOP_NODES(yamaha, step,
423 av_clip((step * ff_adpcm_yamaha_indexscale[nibble]) >> 8,
435 if (generation == 255) {
436 memset(hash, 0xff, 65536 * sizeof(*hash));
441 if (nodes[0]->ssd > (1 << 28)) {
442 for (j = 1; j < frontier && nodes[j]; j++)
443 nodes[j]->ssd -= nodes[0]->ssd;
447 // merge old paths to save memory
448 if (i == froze + FREEZE_INTERVAL) {
449 p = &paths[nodes[0]->path];
450 for (k = i; k > froze; k--) {
456 // other nodes might use paths that don't coincide with the frozen one.
457 // checking which nodes do so is too slow, so just kill them all.
458 // this also slightly improves quality, but I don't know why.
459 memset(nodes + 1, 0, (frontier - 1) * sizeof(TrellisNode*));
463 p = &paths[nodes[0]->path];
464 for (i = n - 1; i > froze; i--) {
469 c->predictor = nodes[0]->sample1;
470 c->sample1 = nodes[0]->sample1;
471 c->sample2 = nodes[0]->sample2;
472 c->step_index = nodes[0]->step;
473 c->step = nodes[0]->step;
474 c->idelta = nodes[0]->step;
477 static int adpcm_encode_frame(AVCodecContext *avctx, AVPacket *avpkt,
478 const AVFrame *frame, int *got_packet_ptr)
480 int n, i, ch, st, pkt_size, ret;
481 const int16_t *samples;
484 ADPCMEncodeContext *c = avctx->priv_data;
487 samples = (const int16_t *)frame->data[0];
488 samples_p = (int16_t **)frame->extended_data;
489 st = avctx->channels == 2;
491 if (avctx->codec_id == AV_CODEC_ID_ADPCM_SWF)
492 pkt_size = (2 + avctx->channels * (22 + 4 * (frame->nb_samples - 1)) + 7) / 8;
494 pkt_size = avctx->block_align;
495 if ((ret = ff_alloc_packet(avpkt, pkt_size))) {
496 av_log(avctx, AV_LOG_ERROR, "Error getting output packet\n");
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");