2 * Copyright (c) 2001-2003 The ffmpeg Project
4 * This file is part of FFmpeg.
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7 * modify it under the terms of the GNU Lesser General Public
8 * License as published by the Free Software Foundation; either
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13 * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
14 * Lesser General Public License for more details.
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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
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_close(AVCodecContext *avctx);
64 static av_cold int adpcm_encode_init(AVCodecContext *avctx)
66 ADPCMEncodeContext *s = avctx->priv_data;
69 int ret = AVERROR(ENOMEM);
71 if (avctx->channels > 2) {
72 av_log(avctx, AV_LOG_ERROR, "only stereo or mono is supported\n");
73 return AVERROR(EINVAL);
76 if (avctx->trellis && (unsigned)avctx->trellis > 16U) {
77 av_log(avctx, AV_LOG_ERROR, "invalid trellis size\n");
78 return AVERROR(EINVAL);
82 int frontier = 1 << avctx->trellis;
83 int max_paths = frontier * FREEZE_INTERVAL;
84 FF_ALLOC_OR_GOTO(avctx, s->paths,
85 max_paths * sizeof(*s->paths), error);
86 FF_ALLOC_OR_GOTO(avctx, s->node_buf,
87 2 * frontier * sizeof(*s->node_buf), error);
88 FF_ALLOC_OR_GOTO(avctx, s->nodep_buf,
89 2 * frontier * sizeof(*s->nodep_buf), error);
90 FF_ALLOC_OR_GOTO(avctx, s->trellis_hash,
91 65536 * sizeof(*s->trellis_hash), error);
94 avctx->bits_per_coded_sample = av_get_bits_per_sample(avctx->codec->id);
96 switch (avctx->codec->id) {
97 case CODEC_ID_ADPCM_IMA_WAV:
98 /* each 16 bits sample gives one nibble
99 and we have 4 bytes per channel overhead */
100 avctx->frame_size = (BLKSIZE - 4 * avctx->channels) * 8 /
101 (4 * avctx->channels) + 1;
102 /* seems frame_size isn't taken into account...
103 have to buffer the samples :-( */
104 avctx->block_align = BLKSIZE;
105 avctx->bits_per_coded_sample = 4;
107 case CODEC_ID_ADPCM_IMA_QT:
108 avctx->frame_size = 64;
109 avctx->block_align = 34 * avctx->channels;
111 case CODEC_ID_ADPCM_MS:
112 /* each 16 bits sample gives one nibble
113 and we have 7 bytes per channel overhead */
114 avctx->frame_size = (BLKSIZE - 7 * avctx->channels) * 2 / avctx->channels + 2;
115 avctx->bits_per_coded_sample = 4;
116 avctx->block_align = BLKSIZE;
117 if (!(avctx->extradata = av_malloc(32 + FF_INPUT_BUFFER_PADDING_SIZE)))
119 avctx->extradata_size = 32;
120 extradata = avctx->extradata;
121 bytestream_put_le16(&extradata, avctx->frame_size);
122 bytestream_put_le16(&extradata, 7); /* wNumCoef */
123 for (i = 0; i < 7; i++) {
124 bytestream_put_le16(&extradata, ff_adpcm_AdaptCoeff1[i] * 4);
125 bytestream_put_le16(&extradata, ff_adpcm_AdaptCoeff2[i] * 4);
128 case CODEC_ID_ADPCM_YAMAHA:
129 avctx->frame_size = BLKSIZE * 2 / avctx->channels;
130 avctx->block_align = BLKSIZE;
132 case CODEC_ID_ADPCM_SWF:
133 if (avctx->sample_rate != 11025 &&
134 avctx->sample_rate != 22050 &&
135 avctx->sample_rate != 44100) {
136 av_log(avctx, AV_LOG_ERROR, "Sample rate must be 11025, "
138 ret = AVERROR(EINVAL);
141 avctx->frame_size = 512 * (avctx->sample_rate / 11025);
144 ret = AVERROR(EINVAL);
148 #if FF_API_OLD_ENCODE_AUDIO
149 if (!(avctx->coded_frame = avcodec_alloc_frame()))
155 adpcm_encode_close(avctx);
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 diff, step = ff_adpcm_step_table[c->step_index];
192 int nibble = 8*(delta < 0);
195 diff = delta + (step >> 3);
214 c->prev_sample -= diff;
216 c->prev_sample += diff;
218 c->prev_sample = av_clip_int16(c->prev_sample);
219 c->step_index = av_clip(c->step_index + ff_adpcm_index_table[nibble], 0, 88);
224 static inline uint8_t adpcm_ms_compress_sample(ADPCMChannelStatus *c,
227 int predictor, nibble, bias;
229 predictor = (((c->sample1) * (c->coeff1)) +
230 (( c->sample2) * (c->coeff2))) / 64;
232 nibble = sample - predictor;
234 bias = c->idelta / 2;
236 bias = -c->idelta / 2;
238 nibble = (nibble + bias) / c->idelta;
239 nibble = av_clip(nibble, -8, 7) & 0x0F;
241 predictor += ((nibble & 0x08) ? (nibble - 0x10) : nibble) * c->idelta;
243 c->sample2 = c->sample1;
244 c->sample1 = av_clip_int16(predictor);
246 c->idelta = (ff_adpcm_AdaptationTable[nibble] * c->idelta) >> 8;
253 static inline uint8_t adpcm_yamaha_compress_sample(ADPCMChannelStatus *c,
263 delta = sample - c->predictor;
265 nibble = FFMIN(7, abs(delta) * 4 / c->step) + (delta < 0) * 8;
267 c->predictor += ((c->step * ff_adpcm_yamaha_difflookup[nibble]) / 8);
268 c->predictor = av_clip_int16(c->predictor);
269 c->step = (c->step * ff_adpcm_yamaha_indexscale[nibble]) >> 8;
270 c->step = av_clip(c->step, 127, 24567);
275 static void adpcm_compress_trellis(AVCodecContext *avctx,
276 const int16_t *samples, uint8_t *dst,
277 ADPCMChannelStatus *c, int n)
279 //FIXME 6% faster if frontier is a compile-time constant
280 ADPCMEncodeContext *s = avctx->priv_data;
281 const int frontier = 1 << avctx->trellis;
282 const int stride = avctx->channels;
283 const int version = avctx->codec->id;
284 TrellisPath *paths = s->paths, *p;
285 TrellisNode *node_buf = s->node_buf;
286 TrellisNode **nodep_buf = s->nodep_buf;
287 TrellisNode **nodes = nodep_buf; // nodes[] is always sorted by .ssd
288 TrellisNode **nodes_next = nodep_buf + frontier;
289 int pathn = 0, froze = -1, i, j, k, generation = 0;
290 uint8_t *hash = s->trellis_hash;
291 memset(hash, 0xff, 65536 * sizeof(*hash));
293 memset(nodep_buf, 0, 2 * frontier * sizeof(*nodep_buf));
294 nodes[0] = node_buf + frontier;
297 nodes[0]->step = c->step_index;
298 nodes[0]->sample1 = c->sample1;
299 nodes[0]->sample2 = c->sample2;
300 if (version == CODEC_ID_ADPCM_IMA_WAV ||
301 version == CODEC_ID_ADPCM_IMA_QT ||
302 version == CODEC_ID_ADPCM_SWF)
303 nodes[0]->sample1 = c->prev_sample;
304 if (version == CODEC_ID_ADPCM_MS)
305 nodes[0]->step = c->idelta;
306 if (version == CODEC_ID_ADPCM_YAMAHA) {
308 nodes[0]->step = 127;
309 nodes[0]->sample1 = 0;
311 nodes[0]->step = c->step;
312 nodes[0]->sample1 = c->predictor;
316 for (i = 0; i < n; i++) {
317 TrellisNode *t = node_buf + frontier*(i&1);
319 int sample = samples[i * stride];
321 memset(nodes_next, 0, frontier * sizeof(TrellisNode*));
322 for (j = 0; j < frontier && nodes[j]; j++) {
323 // higher j have higher ssd already, so they're likely
324 // to yield a suboptimal next sample too
325 const int range = (j < frontier / 2) ? 1 : 0;
326 const int step = nodes[j]->step;
328 if (version == CODEC_ID_ADPCM_MS) {
329 const int predictor = ((nodes[j]->sample1 * c->coeff1) +
330 (nodes[j]->sample2 * c->coeff2)) / 64;
331 const int div = (sample - predictor) / step;
332 const int nmin = av_clip(div-range, -8, 6);
333 const int nmax = av_clip(div+range, -7, 7);
334 for (nidx = nmin; nidx <= nmax; nidx++) {
335 const int nibble = nidx & 0xf;
336 int dec_sample = predictor + nidx * step;
337 #define STORE_NODE(NAME, STEP_INDEX)\
343 dec_sample = av_clip_int16(dec_sample);\
344 d = sample - dec_sample;\
345 ssd = nodes[j]->ssd + d*d;\
346 /* Check for wraparound, skip such samples completely. \
347 * Note, changing ssd to a 64 bit variable would be \
348 * simpler, avoiding this check, but it's slower on \
349 * x86 32 bit at the moment. */\
350 if (ssd < nodes[j]->ssd)\
352 /* Collapse any two states with the same previous sample value. \
353 * One could also distinguish states by step and by 2nd to last
354 * sample, but the effects of that are negligible.
355 * Since nodes in the previous generation are iterated
356 * through a heap, they're roughly ordered from better to
357 * worse, but not strictly ordered. Therefore, an earlier
358 * node with the same sample value is better in most cases
359 * (and thus the current is skipped), but not strictly
360 * in all cases. Only skipping samples where ssd >=
361 * ssd of the earlier node with the same sample gives
362 * slightly worse quality, though, for some reason. */ \
363 h = &hash[(uint16_t) dec_sample];\
364 if (*h == generation)\
366 if (heap_pos < frontier) {\
369 /* Try to replace one of the leaf nodes with the new \
370 * one, but try a different slot each time. */\
371 pos = (frontier >> 1) +\
372 (heap_pos & ((frontier >> 1) - 1));\
373 if (ssd > nodes_next[pos]->ssd)\
378 u = nodes_next[pos];\
380 assert(pathn < FREEZE_INTERVAL << avctx->trellis);\
382 nodes_next[pos] = u;\
386 u->step = STEP_INDEX;\
387 u->sample2 = nodes[j]->sample1;\
388 u->sample1 = dec_sample;\
389 paths[u->path].nibble = nibble;\
390 paths[u->path].prev = nodes[j]->path;\
391 /* Sift the newly inserted node up in the heap to \
392 * restore the heap property. */\
394 int parent = (pos - 1) >> 1;\
395 if (nodes_next[parent]->ssd <= ssd)\
397 FFSWAP(TrellisNode*, nodes_next[parent], nodes_next[pos]);\
401 STORE_NODE(ms, FFMAX(16,
402 (ff_adpcm_AdaptationTable[nibble] * step) >> 8));
404 } else if (version == CODEC_ID_ADPCM_IMA_WAV ||
405 version == CODEC_ID_ADPCM_IMA_QT ||
406 version == CODEC_ID_ADPCM_SWF) {
407 #define LOOP_NODES(NAME, STEP_TABLE, STEP_INDEX)\
408 const int predictor = nodes[j]->sample1;\
409 const int div = (sample - predictor) * 4 / STEP_TABLE;\
410 int nmin = av_clip(div - range, -7, 6);\
411 int nmax = av_clip(div + range, -6, 7);\
413 nmin--; /* distinguish -0 from +0 */\
416 for (nidx = nmin; nidx <= nmax; nidx++) {\
417 const int nibble = nidx < 0 ? 7 - nidx : nidx;\
418 int dec_sample = predictor +\
420 ff_adpcm_yamaha_difflookup[nibble]) / 8;\
421 STORE_NODE(NAME, STEP_INDEX);\
423 LOOP_NODES(ima, ff_adpcm_step_table[step],
424 av_clip(step + ff_adpcm_index_table[nibble], 0, 88));
425 } else { //CODEC_ID_ADPCM_YAMAHA
426 LOOP_NODES(yamaha, step,
427 av_clip((step * ff_adpcm_yamaha_indexscale[nibble]) >> 8,
439 if (generation == 255) {
440 memset(hash, 0xff, 65536 * sizeof(*hash));
445 if (nodes[0]->ssd > (1 << 28)) {
446 for (j = 1; j < frontier && nodes[j]; j++)
447 nodes[j]->ssd -= nodes[0]->ssd;
451 // merge old paths to save memory
452 if (i == froze + FREEZE_INTERVAL) {
453 p = &paths[nodes[0]->path];
454 for (k = i; k > froze; k--) {
460 // other nodes might use paths that don't coincide with the frozen one.
461 // checking which nodes do so is too slow, so just kill them all.
462 // this also slightly improves quality, but I don't know why.
463 memset(nodes + 1, 0, (frontier - 1) * sizeof(TrellisNode*));
467 p = &paths[nodes[0]->path];
468 for (i = n - 1; i > froze; i--) {
473 c->predictor = nodes[0]->sample1;
474 c->sample1 = nodes[0]->sample1;
475 c->sample2 = nodes[0]->sample2;
476 c->step_index = nodes[0]->step;
477 c->step = nodes[0]->step;
478 c->idelta = nodes[0]->step;
481 static int adpcm_encode_frame(AVCodecContext *avctx, AVPacket *avpkt,
482 const AVFrame *frame, int *got_packet_ptr)
484 int n, i, st, pkt_size, ret;
485 const int16_t *samples;
487 ADPCMEncodeContext *c = avctx->priv_data;
490 samples = (const int16_t *)frame->data[0];
491 st = avctx->channels == 2;
493 if (avctx->codec_id == 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_packet(avpkt, pkt_size))) {
498 av_log(avctx, AV_LOG_ERROR, "Error getting output packet\n");
503 switch(avctx->codec->id) {
504 case CODEC_ID_ADPCM_IMA_WAV:
505 n = frame->nb_samples / 8;
506 c->status[0].prev_sample = samples[0];
507 /* c->status[0].step_index = 0;
508 XXX: not sure how to init the state machine */
509 bytestream_put_le16(&dst, c->status[0].prev_sample);
510 *dst++ = c->status[0].step_index;
511 *dst++ = 0; /* unknown */
513 if (avctx->channels == 2) {
514 c->status[1].prev_sample = samples[0];
515 /* c->status[1].step_index = 0; */
516 bytestream_put_le16(&dst, c->status[1].prev_sample);
517 *dst++ = c->status[1].step_index;
522 /* stereo: 4 bytes (8 samples) for left,
523 4 bytes for right, 4 bytes left, ... */
524 if (avctx->trellis > 0) {
525 FF_ALLOC_OR_GOTO(avctx, buf, 2 * n * 8, error);
526 adpcm_compress_trellis(avctx, samples, buf, &c->status[0], n * 8);
527 if (avctx->channels == 2)
528 adpcm_compress_trellis(avctx, samples + 1, buf + n * 8,
529 &c->status[1], n * 8);
530 for (i = 0; i < n; i++) {
531 *dst++ = buf[8 * i + 0] | (buf[8 * i + 1] << 4);
532 *dst++ = buf[8 * i + 2] | (buf[8 * i + 3] << 4);
533 *dst++ = buf[8 * i + 4] | (buf[8 * i + 5] << 4);
534 *dst++ = buf[8 * i + 6] | (buf[8 * i + 7] << 4);
535 if (avctx->channels == 2) {
536 uint8_t *buf1 = buf + n * 8;
537 *dst++ = buf1[8 * i + 0] | (buf1[8 * i + 1] << 4);
538 *dst++ = buf1[8 * i + 2] | (buf1[8 * i + 3] << 4);
539 *dst++ = buf1[8 * i + 4] | (buf1[8 * i + 5] << 4);
540 *dst++ = buf1[8 * i + 6] | (buf1[8 * i + 7] << 4);
546 *dst = adpcm_ima_compress_sample(&c->status[0], samples[0]);
547 *dst++ |= adpcm_ima_compress_sample(&c->status[0], samples[avctx->channels ]) << 4;
548 *dst = adpcm_ima_compress_sample(&c->status[0], samples[avctx->channels * 2]);
549 *dst++ |= adpcm_ima_compress_sample(&c->status[0], samples[avctx->channels * 3]) << 4;
550 *dst = adpcm_ima_compress_sample(&c->status[0], samples[avctx->channels * 4]);
551 *dst++ |= adpcm_ima_compress_sample(&c->status[0], samples[avctx->channels * 5]) << 4;
552 *dst = adpcm_ima_compress_sample(&c->status[0], samples[avctx->channels * 6]);
553 *dst++ |= adpcm_ima_compress_sample(&c->status[0], samples[avctx->channels * 7]) << 4;
555 if (avctx->channels == 2) {
556 *dst = adpcm_ima_compress_sample(&c->status[1], samples[1 ]);
557 *dst++ |= adpcm_ima_compress_sample(&c->status[1], samples[3 ]) << 4;
558 *dst = adpcm_ima_compress_sample(&c->status[1], samples[5 ]);
559 *dst++ |= adpcm_ima_compress_sample(&c->status[1], samples[7 ]) << 4;
560 *dst = adpcm_ima_compress_sample(&c->status[1], samples[9 ]);
561 *dst++ |= adpcm_ima_compress_sample(&c->status[1], samples[11]) << 4;
562 *dst = adpcm_ima_compress_sample(&c->status[1], samples[13]);
563 *dst++ |= adpcm_ima_compress_sample(&c->status[1], samples[15]) << 4;
565 samples += 8 * avctx->channels;
569 case CODEC_ID_ADPCM_IMA_QT:
573 init_put_bits(&pb, dst, pkt_size * 8);
575 for (ch = 0; ch < avctx->channels; ch++) {
576 put_bits(&pb, 9, (c->status[ch].prev_sample + 0x10000) >> 7);
577 put_bits(&pb, 7, c->status[ch].step_index);
578 if (avctx->trellis > 0) {
580 adpcm_compress_trellis(avctx, samples+ch, buf, &c->status[ch], 64);
581 for (i = 0; i < 64; i++)
582 put_bits(&pb, 4, buf[i ^ 1]);
584 for (i = 0; i < 64; i += 2) {
586 t1 = adpcm_ima_qt_compress_sample(&c->status[ch],
587 samples[avctx->channels * (i + 0) + ch]);
588 t2 = adpcm_ima_qt_compress_sample(&c->status[ch],
589 samples[avctx->channels * (i + 1) + ch]);
590 put_bits(&pb, 4, t2);
591 put_bits(&pb, 4, t1);
599 case CODEC_ID_ADPCM_SWF:
603 init_put_bits(&pb, dst, pkt_size * 8);
605 n = frame->nb_samples - 1;
607 // store AdpcmCodeSize
608 put_bits(&pb, 2, 2); // set 4-bit flash adpcm format
610 // init the encoder state
611 for (i = 0; i < avctx->channels; i++) {
612 // clip step so it fits 6 bits
613 c->status[i].step_index = av_clip(c->status[i].step_index, 0, 63);
614 put_sbits(&pb, 16, samples[i]);
615 put_bits(&pb, 6, c->status[i].step_index);
616 c->status[i].prev_sample = samples[i];
619 if (avctx->trellis > 0) {
620 FF_ALLOC_OR_GOTO(avctx, buf, 2 * n, error);
621 adpcm_compress_trellis(avctx, samples + 2, buf, &c->status[0], n);
622 if (avctx->channels == 2)
623 adpcm_compress_trellis(avctx, samples + 3, buf + n,
625 for (i = 0; i < n; i++) {
626 put_bits(&pb, 4, buf[i]);
627 if (avctx->channels == 2)
628 put_bits(&pb, 4, buf[n + i]);
632 for (i = 1; i < frame->nb_samples; i++) {
633 put_bits(&pb, 4, adpcm_ima_compress_sample(&c->status[0],
634 samples[avctx->channels * i]));
635 if (avctx->channels == 2)
636 put_bits(&pb, 4, adpcm_ima_compress_sample(&c->status[1],
637 samples[2 * i + 1]));
643 case CODEC_ID_ADPCM_MS:
644 for (i = 0; i < avctx->channels; i++) {
647 c->status[i].coeff1 = ff_adpcm_AdaptCoeff1[predictor];
648 c->status[i].coeff2 = ff_adpcm_AdaptCoeff2[predictor];
650 for (i = 0; i < avctx->channels; i++) {
651 if (c->status[i].idelta < 16)
652 c->status[i].idelta = 16;
653 bytestream_put_le16(&dst, c->status[i].idelta);
655 for (i = 0; i < avctx->channels; i++)
656 c->status[i].sample2= *samples++;
657 for (i = 0; i < avctx->channels; i++) {
658 c->status[i].sample1 = *samples++;
659 bytestream_put_le16(&dst, c->status[i].sample1);
661 for (i = 0; i < avctx->channels; i++)
662 bytestream_put_le16(&dst, c->status[i].sample2);
664 if (avctx->trellis > 0) {
665 int n = avctx->block_align - 7 * avctx->channels;
666 FF_ALLOC_OR_GOTO(avctx, buf, 2 * n, error);
667 if (avctx->channels == 1) {
668 adpcm_compress_trellis(avctx, samples, buf, &c->status[0], n);
669 for (i = 0; i < n; i += 2)
670 *dst++ = (buf[i] << 4) | buf[i + 1];
672 adpcm_compress_trellis(avctx, samples, buf, &c->status[0], n);
673 adpcm_compress_trellis(avctx, samples + 1, buf + n, &c->status[1], n);
674 for (i = 0; i < n; i++)
675 *dst++ = (buf[i] << 4) | buf[n + i];
679 for (i = 7 * avctx->channels; i < avctx->block_align; i++) {
681 nibble = adpcm_ms_compress_sample(&c->status[ 0], *samples++) << 4;
682 nibble |= adpcm_ms_compress_sample(&c->status[st], *samples++);
687 case CODEC_ID_ADPCM_YAMAHA:
688 n = frame->nb_samples / 2;
689 if (avctx->trellis > 0) {
690 FF_ALLOC_OR_GOTO(avctx, buf, 2 * n * 2, error);
692 if (avctx->channels == 1) {
693 adpcm_compress_trellis(avctx, samples, buf, &c->status[0], n);
694 for (i = 0; i < n; i += 2)
695 *dst++ = buf[i] | (buf[i + 1] << 4);
697 adpcm_compress_trellis(avctx, samples, buf, &c->status[0], n);
698 adpcm_compress_trellis(avctx, samples + 1, buf + n, &c->status[1], n);
699 for (i = 0; i < n; i++)
700 *dst++ = buf[i] | (buf[n + i] << 4);
704 for (n *= avctx->channels; n > 0; n--) {
706 nibble = adpcm_yamaha_compress_sample(&c->status[ 0], *samples++);
707 nibble |= adpcm_yamaha_compress_sample(&c->status[st], *samples++) << 4;
712 return AVERROR(EINVAL);
715 avpkt->size = pkt_size;
719 return AVERROR(ENOMEM);
723 #define ADPCM_ENCODER(id_, name_, long_name_) \
724 AVCodec ff_ ## name_ ## _encoder = { \
726 .type = AVMEDIA_TYPE_AUDIO, \
728 .priv_data_size = sizeof(ADPCMEncodeContext), \
729 .init = adpcm_encode_init, \
730 .encode2 = adpcm_encode_frame, \
731 .close = adpcm_encode_close, \
732 .sample_fmts = (const enum AVSampleFormat[]) { AV_SAMPLE_FMT_S16, \
733 AV_SAMPLE_FMT_NONE}, \
734 .long_name = NULL_IF_CONFIG_SMALL(long_name_), \
737 ADPCM_ENCODER(CODEC_ID_ADPCM_IMA_QT, adpcm_ima_qt, "ADPCM IMA QuickTime");
738 ADPCM_ENCODER(CODEC_ID_ADPCM_IMA_WAV, adpcm_ima_wav, "ADPCM IMA WAV");
739 ADPCM_ENCODER(CODEC_ID_ADPCM_MS, adpcm_ms, "ADPCM Microsoft");
740 ADPCM_ENCODER(CODEC_ID_ADPCM_SWF, adpcm_swf, "ADPCM Shockwave Flash");
741 ADPCM_ENCODER(CODEC_ID_ADPCM_YAMAHA, adpcm_yamaha, "ADPCM Yamaha");