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_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 + AV_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);
149 adpcm_encode_close(avctx);
153 static av_cold int adpcm_encode_close(AVCodecContext *avctx)
155 ADPCMEncodeContext *s = avctx->priv_data;
157 av_freep(&s->node_buf);
158 av_freep(&s->nodep_buf);
159 av_freep(&s->trellis_hash);
165 static inline uint8_t adpcm_ima_compress_sample(ADPCMChannelStatus *c,
168 int delta = sample - c->prev_sample;
169 int nibble = FFMIN(7, abs(delta) * 4 /
170 ff_adpcm_step_table[c->step_index]) + (delta < 0) * 8;
171 c->prev_sample += ((ff_adpcm_step_table[c->step_index] *
172 ff_adpcm_yamaha_difflookup[nibble]) / 8);
173 c->prev_sample = av_clip_int16(c->prev_sample);
174 c->step_index = av_clip(c->step_index + ff_adpcm_index_table[nibble], 0, 88);
178 static inline uint8_t adpcm_ima_qt_compress_sample(ADPCMChannelStatus *c,
181 int delta = sample - c->prev_sample;
182 int diff, step = ff_adpcm_step_table[c->step_index];
183 int nibble = 8*(delta < 0);
186 diff = delta + (step >> 3);
205 c->prev_sample -= diff;
207 c->prev_sample += diff;
209 c->prev_sample = av_clip_int16(c->prev_sample);
210 c->step_index = av_clip(c->step_index + ff_adpcm_index_table[nibble], 0, 88);
215 static inline uint8_t adpcm_ms_compress_sample(ADPCMChannelStatus *c,
218 int predictor, nibble, bias;
220 predictor = (((c->sample1) * (c->coeff1)) +
221 (( c->sample2) * (c->coeff2))) / 64;
223 nibble = sample - predictor;
225 bias = c->idelta / 2;
227 bias = -c->idelta / 2;
229 nibble = (nibble + bias) / c->idelta;
230 nibble = av_clip_intp2(nibble, 3) & 0x0F;
232 predictor += ((nibble & 0x08) ? (nibble - 0x10) : nibble) * c->idelta;
234 c->sample2 = c->sample1;
235 c->sample1 = av_clip_int16(predictor);
237 c->idelta = (ff_adpcm_AdaptationTable[nibble] * c->idelta) >> 8;
244 static inline uint8_t adpcm_yamaha_compress_sample(ADPCMChannelStatus *c,
254 delta = sample - c->predictor;
256 nibble = FFMIN(7, abs(delta) * 4 / c->step) + (delta < 0) * 8;
258 c->predictor += ((c->step * ff_adpcm_yamaha_difflookup[nibble]) / 8);
259 c->predictor = av_clip_int16(c->predictor);
260 c->step = (c->step * ff_adpcm_yamaha_indexscale[nibble]) >> 8;
261 c->step = av_clip(c->step, 127, 24576);
266 static void adpcm_compress_trellis(AVCodecContext *avctx,
267 const int16_t *samples, uint8_t *dst,
268 ADPCMChannelStatus *c, int n, int stride)
270 //FIXME 6% faster if frontier is a compile-time constant
271 ADPCMEncodeContext *s = avctx->priv_data;
272 const int frontier = 1 << avctx->trellis;
273 const int version = avctx->codec->id;
274 TrellisPath *paths = s->paths, *p;
275 TrellisNode *node_buf = s->node_buf;
276 TrellisNode **nodep_buf = s->nodep_buf;
277 TrellisNode **nodes = nodep_buf; // nodes[] is always sorted by .ssd
278 TrellisNode **nodes_next = nodep_buf + frontier;
279 int pathn = 0, froze = -1, i, j, k, generation = 0;
280 uint8_t *hash = s->trellis_hash;
281 memset(hash, 0xff, 65536 * sizeof(*hash));
283 memset(nodep_buf, 0, 2 * frontier * sizeof(*nodep_buf));
284 nodes[0] = node_buf + frontier;
287 nodes[0]->step = c->step_index;
288 nodes[0]->sample1 = c->sample1;
289 nodes[0]->sample2 = c->sample2;
290 if (version == AV_CODEC_ID_ADPCM_IMA_WAV ||
291 version == AV_CODEC_ID_ADPCM_IMA_QT ||
292 version == AV_CODEC_ID_ADPCM_SWF)
293 nodes[0]->sample1 = c->prev_sample;
294 if (version == AV_CODEC_ID_ADPCM_MS)
295 nodes[0]->step = c->idelta;
296 if (version == AV_CODEC_ID_ADPCM_YAMAHA) {
298 nodes[0]->step = 127;
299 nodes[0]->sample1 = 0;
301 nodes[0]->step = c->step;
302 nodes[0]->sample1 = c->predictor;
306 for (i = 0; i < n; i++) {
307 TrellisNode *t = node_buf + frontier*(i&1);
309 int sample = samples[i * stride];
311 memset(nodes_next, 0, frontier * sizeof(TrellisNode*));
312 for (j = 0; j < frontier && nodes[j]; j++) {
313 // higher j have higher ssd already, so they're likely
314 // to yield a suboptimal next sample too
315 const int range = (j < frontier / 2) ? 1 : 0;
316 const int step = nodes[j]->step;
318 if (version == AV_CODEC_ID_ADPCM_MS) {
319 const int predictor = ((nodes[j]->sample1 * c->coeff1) +
320 (nodes[j]->sample2 * c->coeff2)) / 64;
321 const int div = (sample - predictor) / step;
322 const int nmin = av_clip(div-range, -8, 6);
323 const int nmax = av_clip(div+range, -7, 7);
324 for (nidx = nmin; nidx <= nmax; nidx++) {
325 const int nibble = nidx & 0xf;
326 int dec_sample = predictor + nidx * step;
327 #define STORE_NODE(NAME, STEP_INDEX)\
333 dec_sample = av_clip_int16(dec_sample);\
334 d = sample - dec_sample;\
335 ssd = nodes[j]->ssd + d*(unsigned)d;\
336 /* Check for wraparound, skip such samples completely. \
337 * Note, changing ssd to a 64 bit variable would be \
338 * simpler, avoiding this check, but it's slower on \
339 * x86 32 bit at the moment. */\
340 if (ssd < nodes[j]->ssd)\
342 /* Collapse any two states with the same previous sample value. \
343 * One could also distinguish states by step and by 2nd to last
344 * sample, but the effects of that are negligible.
345 * Since nodes in the previous generation are iterated
346 * through a heap, they're roughly ordered from better to
347 * worse, but not strictly ordered. Therefore, an earlier
348 * node with the same sample value is better in most cases
349 * (and thus the current is skipped), but not strictly
350 * in all cases. Only skipping samples where ssd >=
351 * ssd of the earlier node with the same sample gives
352 * slightly worse quality, though, for some reason. */ \
353 h = &hash[(uint16_t) dec_sample];\
354 if (*h == generation)\
356 if (heap_pos < frontier) {\
359 /* Try to replace one of the leaf nodes with the new \
360 * one, but try a different slot each time. */\
361 pos = (frontier >> 1) +\
362 (heap_pos & ((frontier >> 1) - 1));\
363 if (ssd > nodes_next[pos]->ssd)\
368 u = nodes_next[pos];\
370 av_assert1(pathn < FREEZE_INTERVAL << avctx->trellis);\
372 nodes_next[pos] = u;\
376 u->step = STEP_INDEX;\
377 u->sample2 = nodes[j]->sample1;\
378 u->sample1 = dec_sample;\
379 paths[u->path].nibble = nibble;\
380 paths[u->path].prev = nodes[j]->path;\
381 /* Sift the newly inserted node up in the heap to \
382 * restore the heap property. */\
384 int parent = (pos - 1) >> 1;\
385 if (nodes_next[parent]->ssd <= ssd)\
387 FFSWAP(TrellisNode*, nodes_next[parent], nodes_next[pos]);\
391 STORE_NODE(ms, FFMAX(16,
392 (ff_adpcm_AdaptationTable[nibble] * step) >> 8));
394 } else if (version == AV_CODEC_ID_ADPCM_IMA_WAV ||
395 version == AV_CODEC_ID_ADPCM_IMA_QT ||
396 version == AV_CODEC_ID_ADPCM_SWF) {
397 #define LOOP_NODES(NAME, STEP_TABLE, STEP_INDEX)\
398 const int predictor = nodes[j]->sample1;\
399 const int div = (sample - predictor) * 4 / STEP_TABLE;\
400 int nmin = av_clip(div - range, -7, 6);\
401 int nmax = av_clip(div + range, -6, 7);\
403 nmin--; /* distinguish -0 from +0 */\
406 for (nidx = nmin; nidx <= nmax; nidx++) {\
407 const int nibble = nidx < 0 ? 7 - nidx : nidx;\
408 int dec_sample = predictor +\
410 ff_adpcm_yamaha_difflookup[nibble]) / 8;\
411 STORE_NODE(NAME, STEP_INDEX);\
413 LOOP_NODES(ima, ff_adpcm_step_table[step],
414 av_clip(step + ff_adpcm_index_table[nibble], 0, 88));
415 } else { //AV_CODEC_ID_ADPCM_YAMAHA
416 LOOP_NODES(yamaha, step,
417 av_clip((step * ff_adpcm_yamaha_indexscale[nibble]) >> 8,
429 if (generation == 255) {
430 memset(hash, 0xff, 65536 * sizeof(*hash));
435 if (nodes[0]->ssd > (1 << 28)) {
436 for (j = 1; j < frontier && nodes[j]; j++)
437 nodes[j]->ssd -= nodes[0]->ssd;
441 // merge old paths to save memory
442 if (i == froze + FREEZE_INTERVAL) {
443 p = &paths[nodes[0]->path];
444 for (k = i; k > froze; k--) {
450 // other nodes might use paths that don't coincide with the frozen one.
451 // checking which nodes do so is too slow, so just kill them all.
452 // this also slightly improves quality, but I don't know why.
453 memset(nodes + 1, 0, (frontier - 1) * sizeof(TrellisNode*));
457 p = &paths[nodes[0]->path];
458 for (i = n - 1; i > froze; i--) {
463 c->predictor = nodes[0]->sample1;
464 c->sample1 = nodes[0]->sample1;
465 c->sample2 = nodes[0]->sample2;
466 c->step_index = nodes[0]->step;
467 c->step = nodes[0]->step;
468 c->idelta = nodes[0]->step;
471 static int adpcm_encode_frame(AVCodecContext *avctx, AVPacket *avpkt,
472 const AVFrame *frame, int *got_packet_ptr)
474 int n, i, ch, st, pkt_size, ret;
475 const int16_t *samples;
478 ADPCMEncodeContext *c = avctx->priv_data;
481 samples = (const int16_t *)frame->data[0];
482 samples_p = (int16_t **)frame->extended_data;
483 st = avctx->channels == 2;
485 if (avctx->codec_id == AV_CODEC_ID_ADPCM_SWF)
486 pkt_size = (2 + avctx->channels * (22 + 4 * (frame->nb_samples - 1)) + 7) / 8;
488 pkt_size = avctx->block_align;
489 if ((ret = ff_alloc_packet2(avctx, avpkt, pkt_size, 0)) < 0)
493 switch(avctx->codec->id) {
494 case AV_CODEC_ID_ADPCM_IMA_WAV:
498 blocks = (frame->nb_samples - 1) / 8;
500 for (ch = 0; ch < avctx->channels; ch++) {
501 ADPCMChannelStatus *status = &c->status[ch];
502 status->prev_sample = samples_p[ch][0];
503 /* status->step_index = 0;
504 XXX: not sure how to init the state machine */
505 bytestream_put_le16(&dst, status->prev_sample);
506 *dst++ = status->step_index;
507 *dst++ = 0; /* unknown */
510 /* stereo: 4 bytes (8 samples) for left, 4 bytes for right */
511 if (avctx->trellis > 0) {
512 FF_ALLOC_ARRAY_OR_GOTO(avctx, buf, avctx->channels, blocks * 8, error);
513 for (ch = 0; ch < avctx->channels; ch++) {
514 adpcm_compress_trellis(avctx, &samples_p[ch][1],
515 buf + ch * blocks * 8, &c->status[ch],
518 for (i = 0; i < blocks; i++) {
519 for (ch = 0; ch < avctx->channels; ch++) {
520 uint8_t *buf1 = buf + ch * blocks * 8 + i * 8;
521 for (j = 0; j < 8; j += 2)
522 *dst++ = buf1[j] | (buf1[j + 1] << 4);
527 for (i = 0; i < blocks; i++) {
528 for (ch = 0; ch < avctx->channels; ch++) {
529 ADPCMChannelStatus *status = &c->status[ch];
530 const int16_t *smp = &samples_p[ch][1 + i * 8];
531 for (j = 0; j < 8; j += 2) {
532 uint8_t v = adpcm_ima_compress_sample(status, smp[j ]);
533 v |= adpcm_ima_compress_sample(status, smp[j + 1]) << 4;
541 case AV_CODEC_ID_ADPCM_IMA_QT:
544 init_put_bits(&pb, dst, pkt_size);
546 for (ch = 0; ch < avctx->channels; ch++) {
547 ADPCMChannelStatus *status = &c->status[ch];
548 put_bits(&pb, 9, (status->prev_sample & 0xFFFF) >> 7);
549 put_bits(&pb, 7, status->step_index);
550 if (avctx->trellis > 0) {
552 adpcm_compress_trellis(avctx, &samples_p[ch][0], buf, status,
554 for (i = 0; i < 64; i++)
555 put_bits(&pb, 4, buf[i ^ 1]);
556 status->prev_sample = status->predictor;
558 for (i = 0; i < 64; i += 2) {
560 t1 = adpcm_ima_qt_compress_sample(status, samples_p[ch][i ]);
561 t2 = adpcm_ima_qt_compress_sample(status, samples_p[ch][i + 1]);
562 put_bits(&pb, 4, t2);
563 put_bits(&pb, 4, t1);
571 case AV_CODEC_ID_ADPCM_SWF:
574 init_put_bits(&pb, dst, pkt_size);
576 n = frame->nb_samples - 1;
578 // store AdpcmCodeSize
579 put_bits(&pb, 2, 2); // set 4-bit flash adpcm format
581 // init the encoder state
582 for (i = 0; i < avctx->channels; i++) {
583 // clip step so it fits 6 bits
584 c->status[i].step_index = av_clip_uintp2(c->status[i].step_index, 6);
585 put_sbits(&pb, 16, samples[i]);
586 put_bits(&pb, 6, c->status[i].step_index);
587 c->status[i].prev_sample = samples[i];
590 if (avctx->trellis > 0) {
591 FF_ALLOC_OR_GOTO(avctx, buf, 2 * n, error);
592 adpcm_compress_trellis(avctx, samples + avctx->channels, buf,
593 &c->status[0], n, avctx->channels);
594 if (avctx->channels == 2)
595 adpcm_compress_trellis(avctx, samples + avctx->channels + 1,
596 buf + n, &c->status[1], n,
598 for (i = 0; i < n; i++) {
599 put_bits(&pb, 4, buf[i]);
600 if (avctx->channels == 2)
601 put_bits(&pb, 4, buf[n + i]);
605 for (i = 1; i < frame->nb_samples; i++) {
606 put_bits(&pb, 4, adpcm_ima_compress_sample(&c->status[0],
607 samples[avctx->channels * i]));
608 if (avctx->channels == 2)
609 put_bits(&pb, 4, adpcm_ima_compress_sample(&c->status[1],
610 samples[2 * i + 1]));
616 case AV_CODEC_ID_ADPCM_MS:
617 for (i = 0; i < avctx->channels; i++) {
620 c->status[i].coeff1 = ff_adpcm_AdaptCoeff1[predictor];
621 c->status[i].coeff2 = ff_adpcm_AdaptCoeff2[predictor];
623 for (i = 0; i < avctx->channels; i++) {
624 if (c->status[i].idelta < 16)
625 c->status[i].idelta = 16;
626 bytestream_put_le16(&dst, c->status[i].idelta);
628 for (i = 0; i < avctx->channels; i++)
629 c->status[i].sample2= *samples++;
630 for (i = 0; i < avctx->channels; i++) {
631 c->status[i].sample1 = *samples++;
632 bytestream_put_le16(&dst, c->status[i].sample1);
634 for (i = 0; i < avctx->channels; i++)
635 bytestream_put_le16(&dst, c->status[i].sample2);
637 if (avctx->trellis > 0) {
638 n = avctx->block_align - 7 * avctx->channels;
639 FF_ALLOC_OR_GOTO(avctx, buf, 2 * n, error);
640 if (avctx->channels == 1) {
641 adpcm_compress_trellis(avctx, samples, buf, &c->status[0], n,
643 for (i = 0; i < n; i += 2)
644 *dst++ = (buf[i] << 4) | buf[i + 1];
646 adpcm_compress_trellis(avctx, samples, buf,
647 &c->status[0], n, avctx->channels);
648 adpcm_compress_trellis(avctx, samples + 1, buf + n,
649 &c->status[1], n, avctx->channels);
650 for (i = 0; i < n; i++)
651 *dst++ = (buf[i] << 4) | buf[n + i];
655 for (i = 7 * avctx->channels; i < avctx->block_align; i++) {
657 nibble = adpcm_ms_compress_sample(&c->status[ 0], *samples++) << 4;
658 nibble |= adpcm_ms_compress_sample(&c->status[st], *samples++);
663 case AV_CODEC_ID_ADPCM_YAMAHA:
664 n = frame->nb_samples / 2;
665 if (avctx->trellis > 0) {
666 FF_ALLOC_OR_GOTO(avctx, buf, 2 * n * 2, error);
668 if (avctx->channels == 1) {
669 adpcm_compress_trellis(avctx, samples, buf, &c->status[0], n,
671 for (i = 0; i < n; i += 2)
672 *dst++ = buf[i] | (buf[i + 1] << 4);
674 adpcm_compress_trellis(avctx, samples, buf,
675 &c->status[0], n, avctx->channels);
676 adpcm_compress_trellis(avctx, samples + 1, buf + n,
677 &c->status[1], n, avctx->channels);
678 for (i = 0; i < n; i++)
679 *dst++ = buf[i] | (buf[n + i] << 4);
683 for (n *= avctx->channels; n > 0; n--) {
685 nibble = adpcm_yamaha_compress_sample(&c->status[ 0], *samples++);
686 nibble |= adpcm_yamaha_compress_sample(&c->status[st], *samples++) << 4;
691 return AVERROR(EINVAL);
694 avpkt->size = pkt_size;
698 return AVERROR(ENOMEM);
701 static const enum AVSampleFormat sample_fmts[] = {
702 AV_SAMPLE_FMT_S16, AV_SAMPLE_FMT_NONE
705 static const enum AVSampleFormat sample_fmts_p[] = {
706 AV_SAMPLE_FMT_S16P, AV_SAMPLE_FMT_NONE
709 #define ADPCM_ENCODER(id_, name_, sample_fmts_, long_name_) \
710 AVCodec ff_ ## name_ ## _encoder = { \
712 .long_name = NULL_IF_CONFIG_SMALL(long_name_), \
713 .type = AVMEDIA_TYPE_AUDIO, \
715 .priv_data_size = sizeof(ADPCMEncodeContext), \
716 .init = adpcm_encode_init, \
717 .encode2 = adpcm_encode_frame, \
718 .close = adpcm_encode_close, \
719 .sample_fmts = sample_fmts_, \
722 ADPCM_ENCODER(AV_CODEC_ID_ADPCM_IMA_QT, adpcm_ima_qt, sample_fmts_p, "ADPCM IMA QuickTime");
723 ADPCM_ENCODER(AV_CODEC_ID_ADPCM_IMA_WAV, adpcm_ima_wav, sample_fmts_p, "ADPCM IMA WAV");
724 ADPCM_ENCODER(AV_CODEC_ID_ADPCM_MS, adpcm_ms, sample_fmts, "ADPCM Microsoft");
725 ADPCM_ENCODER(AV_CODEC_ID_ADPCM_SWF, adpcm_swf, sample_fmts, "ADPCM Shockwave Flash");
726 ADPCM_ENCODER(AV_CODEC_ID_ADPCM_YAMAHA, adpcm_yamaha, sample_fmts, "ADPCM Yamaha");