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 Libav.
10 * Libav 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 * Libav 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 Libav; if not, write to the Free Software
22 * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
28 #include "bytestream.h"
30 #include "adpcm_data.h"
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);
148 av_freep(&s->node_buf);
149 av_freep(&s->nodep_buf);
150 av_freep(&s->trellis_hash);
154 static av_cold int adpcm_encode_close(AVCodecContext *avctx)
156 ADPCMEncodeContext *s = avctx->priv_data;
158 av_freep(&s->node_buf);
159 av_freep(&s->nodep_buf);
160 av_freep(&s->trellis_hash);
166 static inline uint8_t adpcm_ima_compress_sample(ADPCMChannelStatus *c,
169 int delta = sample - c->prev_sample;
170 int nibble = FFMIN(7, abs(delta) * 4 /
171 ff_adpcm_step_table[c->step_index]) + (delta < 0) * 8;
172 c->prev_sample += ((ff_adpcm_step_table[c->step_index] *
173 ff_adpcm_yamaha_difflookup[nibble]) / 8);
174 c->prev_sample = av_clip_int16(c->prev_sample);
175 c->step_index = av_clip(c->step_index + ff_adpcm_index_table[nibble], 0, 88);
179 static inline uint8_t adpcm_ima_qt_compress_sample(ADPCMChannelStatus *c,
182 int delta = sample - c->prev_sample;
183 int mask, step = ff_adpcm_step_table[c->step_index];
184 int diff = step >> 3;
192 for (mask = 4; mask;) {
203 c->prev_sample -= diff;
205 c->prev_sample += diff;
207 c->prev_sample = av_clip_int16(c->prev_sample);
208 c->step_index = av_clip(c->step_index + ff_adpcm_index_table[nibble], 0, 88);
213 static inline uint8_t adpcm_ms_compress_sample(ADPCMChannelStatus *c,
216 int predictor, nibble, bias;
218 predictor = (((c->sample1) * (c->coeff1)) +
219 (( c->sample2) * (c->coeff2))) / 64;
221 nibble = sample - predictor;
223 bias = c->idelta / 2;
225 bias = -c->idelta / 2;
227 nibble = (nibble + bias) / c->idelta;
228 nibble = av_clip(nibble, -8, 7) & 0x0F;
230 predictor += ((nibble & 0x08) ? (nibble - 0x10) : nibble) * c->idelta;
232 c->sample2 = c->sample1;
233 c->sample1 = av_clip_int16(predictor);
235 c->idelta = (ff_adpcm_AdaptationTable[nibble] * c->idelta) >> 8;
242 static inline uint8_t adpcm_yamaha_compress_sample(ADPCMChannelStatus *c,
252 delta = sample - c->predictor;
254 nibble = FFMIN(7, abs(delta) * 4 / c->step) + (delta < 0) * 8;
256 c->predictor += ((c->step * ff_adpcm_yamaha_difflookup[nibble]) / 8);
257 c->predictor = av_clip_int16(c->predictor);
258 c->step = (c->step * ff_adpcm_yamaha_indexscale[nibble]) >> 8;
259 c->step = av_clip(c->step, 127, 24567);
264 static void adpcm_compress_trellis(AVCodecContext *avctx,
265 const int16_t *samples, uint8_t *dst,
266 ADPCMChannelStatus *c, int n, int stride)
268 //FIXME 6% faster if frontier is a compile-time constant
269 ADPCMEncodeContext *s = avctx->priv_data;
270 const int frontier = 1 << avctx->trellis;
271 const int version = avctx->codec->id;
272 TrellisPath *paths = s->paths, *p;
273 TrellisNode *node_buf = s->node_buf;
274 TrellisNode **nodep_buf = s->nodep_buf;
275 TrellisNode **nodes = nodep_buf; // nodes[] is always sorted by .ssd
276 TrellisNode **nodes_next = nodep_buf + frontier;
277 int pathn = 0, froze = -1, i, j, k, generation = 0;
278 uint8_t *hash = s->trellis_hash;
279 memset(hash, 0xff, 65536 * sizeof(*hash));
281 memset(nodep_buf, 0, 2 * frontier * sizeof(*nodep_buf));
282 nodes[0] = node_buf + frontier;
285 nodes[0]->step = c->step_index;
286 nodes[0]->sample1 = c->sample1;
287 nodes[0]->sample2 = c->sample2;
288 if (version == AV_CODEC_ID_ADPCM_IMA_WAV ||
289 version == AV_CODEC_ID_ADPCM_IMA_QT ||
290 version == AV_CODEC_ID_ADPCM_SWF)
291 nodes[0]->sample1 = c->prev_sample;
292 if (version == AV_CODEC_ID_ADPCM_MS)
293 nodes[0]->step = c->idelta;
294 if (version == AV_CODEC_ID_ADPCM_YAMAHA) {
296 nodes[0]->step = 127;
297 nodes[0]->sample1 = 0;
299 nodes[0]->step = c->step;
300 nodes[0]->sample1 = c->predictor;
304 for (i = 0; i < n; i++) {
305 TrellisNode *t = node_buf + frontier*(i&1);
307 int sample = samples[i * stride];
309 memset(nodes_next, 0, frontier * sizeof(TrellisNode*));
310 for (j = 0; j < frontier && nodes[j]; j++) {
311 // higher j have higher ssd already, so they're likely
312 // to yield a suboptimal next sample too
313 const int range = (j < frontier / 2) ? 1 : 0;
314 const int step = nodes[j]->step;
316 if (version == AV_CODEC_ID_ADPCM_MS) {
317 const int predictor = ((nodes[j]->sample1 * c->coeff1) +
318 (nodes[j]->sample2 * c->coeff2)) / 64;
319 const int div = (sample - predictor) / step;
320 const int nmin = av_clip(div-range, -8, 6);
321 const int nmax = av_clip(div+range, -7, 7);
322 for (nidx = nmin; nidx <= nmax; nidx++) {
323 const int nibble = nidx & 0xf;
324 int dec_sample = predictor + nidx * step;
325 #define STORE_NODE(NAME, STEP_INDEX)\
331 dec_sample = av_clip_int16(dec_sample);\
332 d = sample - dec_sample;\
333 ssd = nodes[j]->ssd + d*d;\
334 /* Check for wraparound, skip such samples completely. \
335 * Note, changing ssd to a 64 bit variable would be \
336 * simpler, avoiding this check, but it's slower on \
337 * x86 32 bit at the moment. */\
338 if (ssd < nodes[j]->ssd)\
340 /* Collapse any two states with the same previous sample value. \
341 * One could also distinguish states by step and by 2nd to last
342 * sample, but the effects of that are negligible.
343 * Since nodes in the previous generation are iterated
344 * through a heap, they're roughly ordered from better to
345 * worse, but not strictly ordered. Therefore, an earlier
346 * node with the same sample value is better in most cases
347 * (and thus the current is skipped), but not strictly
348 * in all cases. Only skipping samples where ssd >=
349 * ssd of the earlier node with the same sample gives
350 * slightly worse quality, though, for some reason. */ \
351 h = &hash[(uint16_t) dec_sample];\
352 if (*h == generation)\
354 if (heap_pos < frontier) {\
357 /* Try to replace one of the leaf nodes with the new \
358 * one, but try a different slot each time. */\
359 pos = (frontier >> 1) +\
360 (heap_pos & ((frontier >> 1) - 1));\
361 if (ssd > nodes_next[pos]->ssd)\
366 u = nodes_next[pos];\
368 assert(pathn < FREEZE_INTERVAL << avctx->trellis);\
370 nodes_next[pos] = u;\
374 u->step = STEP_INDEX;\
375 u->sample2 = nodes[j]->sample1;\
376 u->sample1 = dec_sample;\
377 paths[u->path].nibble = nibble;\
378 paths[u->path].prev = nodes[j]->path;\
379 /* Sift the newly inserted node up in the heap to \
380 * restore the heap property. */\
382 int parent = (pos - 1) >> 1;\
383 if (nodes_next[parent]->ssd <= ssd)\
385 FFSWAP(TrellisNode*, nodes_next[parent], nodes_next[pos]);\
389 STORE_NODE(ms, FFMAX(16,
390 (ff_adpcm_AdaptationTable[nibble] * step) >> 8));
392 } else if (version == AV_CODEC_ID_ADPCM_IMA_WAV ||
393 version == AV_CODEC_ID_ADPCM_IMA_QT ||
394 version == AV_CODEC_ID_ADPCM_SWF) {
395 #define LOOP_NODES(NAME, STEP_TABLE, STEP_INDEX)\
396 const int predictor = nodes[j]->sample1;\
397 const int div = (sample - predictor) * 4 / STEP_TABLE;\
398 int nmin = av_clip(div - range, -7, 6);\
399 int nmax = av_clip(div + range, -6, 7);\
401 nmin--; /* distinguish -0 from +0 */\
404 for (nidx = nmin; nidx <= nmax; nidx++) {\
405 const int nibble = nidx < 0 ? 7 - nidx : nidx;\
406 int dec_sample = predictor +\
408 ff_adpcm_yamaha_difflookup[nibble]) / 8;\
409 STORE_NODE(NAME, STEP_INDEX);\
411 LOOP_NODES(ima, ff_adpcm_step_table[step],
412 av_clip(step + ff_adpcm_index_table[nibble], 0, 88));
413 } else { //AV_CODEC_ID_ADPCM_YAMAHA
414 LOOP_NODES(yamaha, step,
415 av_clip((step * ff_adpcm_yamaha_indexscale[nibble]) >> 8,
427 if (generation == 255) {
428 memset(hash, 0xff, 65536 * sizeof(*hash));
433 if (nodes[0]->ssd > (1 << 28)) {
434 for (j = 1; j < frontier && nodes[j]; j++)
435 nodes[j]->ssd -= nodes[0]->ssd;
439 // merge old paths to save memory
440 if (i == froze + FREEZE_INTERVAL) {
441 p = &paths[nodes[0]->path];
442 for (k = i; k > froze; k--) {
448 // other nodes might use paths that don't coincide with the frozen one.
449 // checking which nodes do so is too slow, so just kill them all.
450 // this also slightly improves quality, but I don't know why.
451 memset(nodes + 1, 0, (frontier - 1) * sizeof(TrellisNode*));
455 p = &paths[nodes[0]->path];
456 for (i = n - 1; i > froze; i--) {
461 c->predictor = nodes[0]->sample1;
462 c->sample1 = nodes[0]->sample1;
463 c->sample2 = nodes[0]->sample2;
464 c->step_index = nodes[0]->step;
465 c->step = nodes[0]->step;
466 c->idelta = nodes[0]->step;
469 static int adpcm_encode_frame(AVCodecContext *avctx, AVPacket *avpkt,
470 const AVFrame *frame, int *got_packet_ptr)
472 int n, i, ch, st, pkt_size, ret;
473 const int16_t *samples;
476 ADPCMEncodeContext *c = avctx->priv_data;
479 samples = (const int16_t *)frame->data[0];
480 samples_p = (int16_t **)frame->extended_data;
481 st = avctx->channels == 2;
483 if (avctx->codec_id == AV_CODEC_ID_ADPCM_SWF)
484 pkt_size = (2 + avctx->channels * (22 + 4 * (frame->nb_samples - 1)) + 7) / 8;
486 pkt_size = avctx->block_align;
487 if ((ret = ff_alloc_packet(avpkt, pkt_size))) {
488 av_log(avctx, AV_LOG_ERROR, "Error getting output packet\n");
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_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 * 8);
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][1], buf, status,
554 for (i = 0; i < 64; i++)
555 put_bits(&pb, 4, buf[i ^ 1]);
557 for (i = 0; i < 64; i += 2) {
559 t1 = adpcm_ima_qt_compress_sample(status, samples_p[ch][i ]);
560 t2 = adpcm_ima_qt_compress_sample(status, samples_p[ch][i + 1]);
561 put_bits(&pb, 4, t2);
562 put_bits(&pb, 4, t1);
570 case AV_CODEC_ID_ADPCM_SWF:
573 init_put_bits(&pb, dst, pkt_size * 8);
575 n = frame->nb_samples - 1;
577 // store AdpcmCodeSize
578 put_bits(&pb, 2, 2); // set 4-bit flash adpcm format
580 // init the encoder state
581 for (i = 0; i < avctx->channels; i++) {
582 // clip step so it fits 6 bits
583 c->status[i].step_index = av_clip(c->status[i].step_index, 0, 63);
584 put_sbits(&pb, 16, samples[i]);
585 put_bits(&pb, 6, c->status[i].step_index);
586 c->status[i].prev_sample = samples[i];
589 if (avctx->trellis > 0) {
590 FF_ALLOC_OR_GOTO(avctx, buf, 2 * n, error);
591 adpcm_compress_trellis(avctx, samples + avctx->channels, buf,
592 &c->status[0], n, avctx->channels);
593 if (avctx->channels == 2)
594 adpcm_compress_trellis(avctx, samples + avctx->channels + 1,
595 buf + n, &c->status[1], n,
597 for (i = 0; i < n; i++) {
598 put_bits(&pb, 4, buf[i]);
599 if (avctx->channels == 2)
600 put_bits(&pb, 4, buf[n + i]);
604 for (i = 1; i < frame->nb_samples; i++) {
605 put_bits(&pb, 4, adpcm_ima_compress_sample(&c->status[0],
606 samples[avctx->channels * i]));
607 if (avctx->channels == 2)
608 put_bits(&pb, 4, adpcm_ima_compress_sample(&c->status[1],
609 samples[2 * i + 1]));
615 case AV_CODEC_ID_ADPCM_MS:
616 for (i = 0; i < avctx->channels; i++) {
619 c->status[i].coeff1 = ff_adpcm_AdaptCoeff1[predictor];
620 c->status[i].coeff2 = ff_adpcm_AdaptCoeff2[predictor];
622 for (i = 0; i < avctx->channels; i++) {
623 if (c->status[i].idelta < 16)
624 c->status[i].idelta = 16;
625 bytestream_put_le16(&dst, c->status[i].idelta);
627 for (i = 0; i < avctx->channels; i++)
628 c->status[i].sample2= *samples++;
629 for (i = 0; i < avctx->channels; i++) {
630 c->status[i].sample1 = *samples++;
631 bytestream_put_le16(&dst, c->status[i].sample1);
633 for (i = 0; i < avctx->channels; i++)
634 bytestream_put_le16(&dst, c->status[i].sample2);
636 if (avctx->trellis > 0) {
637 n = avctx->block_align - 7 * avctx->channels;
638 FF_ALLOC_OR_GOTO(avctx, buf, 2 * n, error);
639 if (avctx->channels == 1) {
640 adpcm_compress_trellis(avctx, samples, buf, &c->status[0], n,
642 for (i = 0; i < n; i += 2)
643 *dst++ = (buf[i] << 4) | buf[i + 1];
645 adpcm_compress_trellis(avctx, samples, buf,
646 &c->status[0], n, avctx->channels);
647 adpcm_compress_trellis(avctx, samples + 1, buf + n,
648 &c->status[1], n, avctx->channels);
649 for (i = 0; i < n; i++)
650 *dst++ = (buf[i] << 4) | buf[n + i];
654 for (i = 7 * avctx->channels; i < avctx->block_align; i++) {
656 nibble = adpcm_ms_compress_sample(&c->status[ 0], *samples++) << 4;
657 nibble |= adpcm_ms_compress_sample(&c->status[st], *samples++);
662 case AV_CODEC_ID_ADPCM_YAMAHA:
663 n = frame->nb_samples / 2;
664 if (avctx->trellis > 0) {
665 FF_ALLOC_OR_GOTO(avctx, buf, 2 * n * 2, error);
667 if (avctx->channels == 1) {
668 adpcm_compress_trellis(avctx, samples, buf, &c->status[0], n,
670 for (i = 0; i < n; i += 2)
671 *dst++ = buf[i] | (buf[i + 1] << 4);
673 adpcm_compress_trellis(avctx, samples, buf,
674 &c->status[0], n, avctx->channels);
675 adpcm_compress_trellis(avctx, samples + 1, buf + n,
676 &c->status[1], n, avctx->channels);
677 for (i = 0; i < n; i++)
678 *dst++ = buf[i] | (buf[n + i] << 4);
682 for (n *= avctx->channels; n > 0; n--) {
684 nibble = adpcm_yamaha_compress_sample(&c->status[ 0], *samples++);
685 nibble |= adpcm_yamaha_compress_sample(&c->status[st], *samples++) << 4;
690 return AVERROR(EINVAL);
693 avpkt->size = pkt_size;
697 return AVERROR(ENOMEM);
700 static const enum AVSampleFormat sample_fmts[] = {
701 AV_SAMPLE_FMT_S16, AV_SAMPLE_FMT_NONE
704 static const enum AVSampleFormat sample_fmts_p[] = {
705 AV_SAMPLE_FMT_S16P, AV_SAMPLE_FMT_NONE
708 #define ADPCM_ENCODER(id_, name_, sample_fmts_, long_name_) \
709 AVCodec ff_ ## name_ ## _encoder = { \
711 .long_name = NULL_IF_CONFIG_SMALL(long_name_), \
712 .type = AVMEDIA_TYPE_AUDIO, \
714 .priv_data_size = sizeof(ADPCMEncodeContext), \
715 .init = adpcm_encode_init, \
716 .encode2 = adpcm_encode_frame, \
717 .close = adpcm_encode_close, \
718 .sample_fmts = sample_fmts_, \
721 ADPCM_ENCODER(AV_CODEC_ID_ADPCM_IMA_QT, adpcm_ima_qt, sample_fmts_p, "ADPCM IMA QuickTime");
722 ADPCM_ENCODER(AV_CODEC_ID_ADPCM_IMA_WAV, adpcm_ima_wav, sample_fmts_p, "ADPCM IMA WAV");
723 ADPCM_ENCODER(AV_CODEC_ID_ADPCM_MS, adpcm_ms, sample_fmts, "ADPCM Microsoft");
724 ADPCM_ENCODER(AV_CODEC_ID_ADPCM_SWF, adpcm_swf, sample_fmts, "ADPCM Shockwave Flash");
725 ADPCM_ENCODER(AV_CODEC_ID_ADPCM_YAMAHA, adpcm_yamaha, sample_fmts, "ADPCM Yamaha");