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.
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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)
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 stride = avctx->channels;
280 const int version = avctx->codec->id;
281 TrellisPath *paths = s->paths, *p;
282 TrellisNode *node_buf = s->node_buf;
283 TrellisNode **nodep_buf = s->nodep_buf;
284 TrellisNode **nodes = nodep_buf; // nodes[] is always sorted by .ssd
285 TrellisNode **nodes_next = nodep_buf + frontier;
286 int pathn = 0, froze = -1, i, j, k, generation = 0;
287 uint8_t *hash = s->trellis_hash;
288 memset(hash, 0xff, 65536 * sizeof(*hash));
290 memset(nodep_buf, 0, 2 * frontier * sizeof(*nodep_buf));
291 nodes[0] = node_buf + frontier;
294 nodes[0]->step = c->step_index;
295 nodes[0]->sample1 = c->sample1;
296 nodes[0]->sample2 = c->sample2;
297 if (version == AV_CODEC_ID_ADPCM_IMA_WAV ||
298 version == AV_CODEC_ID_ADPCM_IMA_QT ||
299 version == AV_CODEC_ID_ADPCM_SWF)
300 nodes[0]->sample1 = c->prev_sample;
301 if (version == AV_CODEC_ID_ADPCM_MS)
302 nodes[0]->step = c->idelta;
303 if (version == AV_CODEC_ID_ADPCM_YAMAHA) {
305 nodes[0]->step = 127;
306 nodes[0]->sample1 = 0;
308 nodes[0]->step = c->step;
309 nodes[0]->sample1 = c->predictor;
313 for (i = 0; i < n; i++) {
314 TrellisNode *t = node_buf + frontier*(i&1);
316 int sample = samples[i * stride];
318 memset(nodes_next, 0, frontier * sizeof(TrellisNode*));
319 for (j = 0; j < frontier && nodes[j]; j++) {
320 // higher j have higher ssd already, so they're likely
321 // to yield a suboptimal next sample too
322 const int range = (j < frontier / 2) ? 1 : 0;
323 const int step = nodes[j]->step;
325 if (version == AV_CODEC_ID_ADPCM_MS) {
326 const int predictor = ((nodes[j]->sample1 * c->coeff1) +
327 (nodes[j]->sample2 * c->coeff2)) / 64;
328 const int div = (sample - predictor) / step;
329 const int nmin = av_clip(div-range, -8, 6);
330 const int nmax = av_clip(div+range, -7, 7);
331 for (nidx = nmin; nidx <= nmax; nidx++) {
332 const int nibble = nidx & 0xf;
333 int dec_sample = predictor + nidx * step;
334 #define STORE_NODE(NAME, STEP_INDEX)\
340 dec_sample = av_clip_int16(dec_sample);\
341 d = sample - dec_sample;\
342 ssd = nodes[j]->ssd + d*d;\
343 /* Check for wraparound, skip such samples completely. \
344 * Note, changing ssd to a 64 bit variable would be \
345 * simpler, avoiding this check, but it's slower on \
346 * x86 32 bit at the moment. */\
347 if (ssd < nodes[j]->ssd)\
349 /* Collapse any two states with the same previous sample value. \
350 * One could also distinguish states by step and by 2nd to last
351 * sample, but the effects of that are negligible.
352 * Since nodes in the previous generation are iterated
353 * through a heap, they're roughly ordered from better to
354 * worse, but not strictly ordered. Therefore, an earlier
355 * node with the same sample value is better in most cases
356 * (and thus the current is skipped), but not strictly
357 * in all cases. Only skipping samples where ssd >=
358 * ssd of the earlier node with the same sample gives
359 * slightly worse quality, though, for some reason. */ \
360 h = &hash[(uint16_t) dec_sample];\
361 if (*h == generation)\
363 if (heap_pos < frontier) {\
366 /* Try to replace one of the leaf nodes with the new \
367 * one, but try a different slot each time. */\
368 pos = (frontier >> 1) +\
369 (heap_pos & ((frontier >> 1) - 1));\
370 if (ssd > nodes_next[pos]->ssd)\
375 u = nodes_next[pos];\
377 assert(pathn < FREEZE_INTERVAL << avctx->trellis);\
379 nodes_next[pos] = u;\
383 u->step = STEP_INDEX;\
384 u->sample2 = nodes[j]->sample1;\
385 u->sample1 = dec_sample;\
386 paths[u->path].nibble = nibble;\
387 paths[u->path].prev = nodes[j]->path;\
388 /* Sift the newly inserted node up in the heap to \
389 * restore the heap property. */\
391 int parent = (pos - 1) >> 1;\
392 if (nodes_next[parent]->ssd <= ssd)\
394 FFSWAP(TrellisNode*, nodes_next[parent], nodes_next[pos]);\
398 STORE_NODE(ms, FFMAX(16,
399 (ff_adpcm_AdaptationTable[nibble] * step) >> 8));
401 } else if (version == AV_CODEC_ID_ADPCM_IMA_WAV ||
402 version == AV_CODEC_ID_ADPCM_IMA_QT ||
403 version == AV_CODEC_ID_ADPCM_SWF) {
404 #define LOOP_NODES(NAME, STEP_TABLE, STEP_INDEX)\
405 const int predictor = nodes[j]->sample1;\
406 const int div = (sample - predictor) * 4 / STEP_TABLE;\
407 int nmin = av_clip(div - range, -7, 6);\
408 int nmax = av_clip(div + range, -6, 7);\
410 nmin--; /* distinguish -0 from +0 */\
413 for (nidx = nmin; nidx <= nmax; nidx++) {\
414 const int nibble = nidx < 0 ? 7 - nidx : nidx;\
415 int dec_sample = predictor +\
417 ff_adpcm_yamaha_difflookup[nibble]) / 8;\
418 STORE_NODE(NAME, STEP_INDEX);\
420 LOOP_NODES(ima, ff_adpcm_step_table[step],
421 av_clip(step + ff_adpcm_index_table[nibble], 0, 88));
422 } else { //AV_CODEC_ID_ADPCM_YAMAHA
423 LOOP_NODES(yamaha, step,
424 av_clip((step * ff_adpcm_yamaha_indexscale[nibble]) >> 8,
436 if (generation == 255) {
437 memset(hash, 0xff, 65536 * sizeof(*hash));
442 if (nodes[0]->ssd > (1 << 28)) {
443 for (j = 1; j < frontier && nodes[j]; j++)
444 nodes[j]->ssd -= nodes[0]->ssd;
448 // merge old paths to save memory
449 if (i == froze + FREEZE_INTERVAL) {
450 p = &paths[nodes[0]->path];
451 for (k = i; k > froze; k--) {
457 // other nodes might use paths that don't coincide with the frozen one.
458 // checking which nodes do so is too slow, so just kill them all.
459 // this also slightly improves quality, but I don't know why.
460 memset(nodes + 1, 0, (frontier - 1) * sizeof(TrellisNode*));
464 p = &paths[nodes[0]->path];
465 for (i = n - 1; i > froze; i--) {
470 c->predictor = nodes[0]->sample1;
471 c->sample1 = nodes[0]->sample1;
472 c->sample2 = nodes[0]->sample2;
473 c->step_index = nodes[0]->step;
474 c->step = nodes[0]->step;
475 c->idelta = nodes[0]->step;
478 static int adpcm_encode_frame(AVCodecContext *avctx, AVPacket *avpkt,
479 const AVFrame *frame, int *got_packet_ptr)
481 int n, i, st, pkt_size, ret;
482 const int16_t *samples;
484 ADPCMEncodeContext *c = avctx->priv_data;
487 samples = (const int16_t *)frame->data[0];
488 st = avctx->channels == 2;
490 if (avctx->codec_id == AV_CODEC_ID_ADPCM_SWF)
491 pkt_size = (2 + avctx->channels * (22 + 4 * (frame->nb_samples - 1)) + 7) / 8;
493 pkt_size = avctx->block_align;
494 if ((ret = ff_alloc_packet(avpkt, pkt_size))) {
495 av_log(avctx, AV_LOG_ERROR, "Error getting output packet\n");
500 switch(avctx->codec->id) {
501 case AV_CODEC_ID_ADPCM_IMA_WAV:
502 n = frame->nb_samples / 8;
503 c->status[0].prev_sample = samples[0];
504 /* c->status[0].step_index = 0;
505 XXX: not sure how to init the state machine */
506 bytestream_put_le16(&dst, c->status[0].prev_sample);
507 *dst++ = c->status[0].step_index;
508 *dst++ = 0; /* unknown */
510 if (avctx->channels == 2) {
511 c->status[1].prev_sample = samples[0];
512 /* c->status[1].step_index = 0; */
513 bytestream_put_le16(&dst, c->status[1].prev_sample);
514 *dst++ = c->status[1].step_index;
519 /* stereo: 4 bytes (8 samples) for left,
520 4 bytes for right, 4 bytes left, ... */
521 if (avctx->trellis > 0) {
522 FF_ALLOC_OR_GOTO(avctx, buf, 2 * n * 8, error);
523 adpcm_compress_trellis(avctx, samples, buf, &c->status[0], n * 8);
524 if (avctx->channels == 2)
525 adpcm_compress_trellis(avctx, samples + 1, buf + n * 8,
526 &c->status[1], n * 8);
527 for (i = 0; i < n; i++) {
528 *dst++ = buf[8 * i + 0] | (buf[8 * i + 1] << 4);
529 *dst++ = buf[8 * i + 2] | (buf[8 * i + 3] << 4);
530 *dst++ = buf[8 * i + 4] | (buf[8 * i + 5] << 4);
531 *dst++ = buf[8 * i + 6] | (buf[8 * i + 7] << 4);
532 if (avctx->channels == 2) {
533 uint8_t *buf1 = buf + n * 8;
534 *dst++ = buf1[8 * i + 0] | (buf1[8 * i + 1] << 4);
535 *dst++ = buf1[8 * i + 2] | (buf1[8 * i + 3] << 4);
536 *dst++ = buf1[8 * i + 4] | (buf1[8 * i + 5] << 4);
537 *dst++ = buf1[8 * i + 6] | (buf1[8 * i + 7] << 4);
543 *dst = adpcm_ima_compress_sample(&c->status[0], samples[0]);
544 *dst++ |= adpcm_ima_compress_sample(&c->status[0], samples[avctx->channels ]) << 4;
545 *dst = adpcm_ima_compress_sample(&c->status[0], samples[avctx->channels * 2]);
546 *dst++ |= adpcm_ima_compress_sample(&c->status[0], samples[avctx->channels * 3]) << 4;
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 * 5]) << 4;
549 *dst = adpcm_ima_compress_sample(&c->status[0], samples[avctx->channels * 6]);
550 *dst++ |= adpcm_ima_compress_sample(&c->status[0], samples[avctx->channels * 7]) << 4;
552 if (avctx->channels == 2) {
553 *dst = adpcm_ima_compress_sample(&c->status[1], samples[1 ]);
554 *dst++ |= adpcm_ima_compress_sample(&c->status[1], samples[3 ]) << 4;
555 *dst = adpcm_ima_compress_sample(&c->status[1], samples[5 ]);
556 *dst++ |= adpcm_ima_compress_sample(&c->status[1], samples[7 ]) << 4;
557 *dst = adpcm_ima_compress_sample(&c->status[1], samples[9 ]);
558 *dst++ |= adpcm_ima_compress_sample(&c->status[1], samples[11]) << 4;
559 *dst = adpcm_ima_compress_sample(&c->status[1], samples[13]);
560 *dst++ |= adpcm_ima_compress_sample(&c->status[1], samples[15]) << 4;
562 samples += 8 * avctx->channels;
566 case AV_CODEC_ID_ADPCM_IMA_QT:
570 init_put_bits(&pb, dst, pkt_size * 8);
572 for (ch = 0; ch < avctx->channels; ch++) {
573 put_bits(&pb, 9, (c->status[ch].prev_sample & 0xFFFF) >> 7);
574 put_bits(&pb, 7, c->status[ch].step_index);
575 if (avctx->trellis > 0) {
577 adpcm_compress_trellis(avctx, samples+ch, buf, &c->status[ch], 64);
578 for (i = 0; i < 64; i++)
579 put_bits(&pb, 4, buf[i ^ 1]);
581 for (i = 0; i < 64; i += 2) {
583 t1 = adpcm_ima_qt_compress_sample(&c->status[ch],
584 samples[avctx->channels * (i + 0) + ch]);
585 t2 = adpcm_ima_qt_compress_sample(&c->status[ch],
586 samples[avctx->channels * (i + 1) + ch]);
587 put_bits(&pb, 4, t2);
588 put_bits(&pb, 4, t1);
596 case AV_CODEC_ID_ADPCM_SWF:
600 init_put_bits(&pb, dst, pkt_size * 8);
602 n = frame->nb_samples - 1;
604 // store AdpcmCodeSize
605 put_bits(&pb, 2, 2); // set 4-bit flash adpcm format
607 // init the encoder state
608 for (i = 0; i < avctx->channels; i++) {
609 // clip step so it fits 6 bits
610 c->status[i].step_index = av_clip(c->status[i].step_index, 0, 63);
611 put_sbits(&pb, 16, samples[i]);
612 put_bits(&pb, 6, c->status[i].step_index);
613 c->status[i].prev_sample = samples[i];
616 if (avctx->trellis > 0) {
617 FF_ALLOC_OR_GOTO(avctx, buf, 2 * n, error);
618 adpcm_compress_trellis(avctx, samples + avctx->channels, buf,
620 if (avctx->channels == 2)
621 adpcm_compress_trellis(avctx, samples + avctx->channels + 1,
622 buf + n, &c->status[1], n);
623 for (i = 0; i < n; i++) {
624 put_bits(&pb, 4, buf[i]);
625 if (avctx->channels == 2)
626 put_bits(&pb, 4, buf[n + i]);
630 for (i = 1; i < frame->nb_samples; i++) {
631 put_bits(&pb, 4, adpcm_ima_compress_sample(&c->status[0],
632 samples[avctx->channels * i]));
633 if (avctx->channels == 2)
634 put_bits(&pb, 4, adpcm_ima_compress_sample(&c->status[1],
635 samples[2 * i + 1]));
641 case AV_CODEC_ID_ADPCM_MS:
642 for (i = 0; i < avctx->channels; i++) {
645 c->status[i].coeff1 = ff_adpcm_AdaptCoeff1[predictor];
646 c->status[i].coeff2 = ff_adpcm_AdaptCoeff2[predictor];
648 for (i = 0; i < avctx->channels; i++) {
649 if (c->status[i].idelta < 16)
650 c->status[i].idelta = 16;
651 bytestream_put_le16(&dst, c->status[i].idelta);
653 for (i = 0; i < avctx->channels; i++)
654 c->status[i].sample2= *samples++;
655 for (i = 0; i < avctx->channels; i++) {
656 c->status[i].sample1 = *samples++;
657 bytestream_put_le16(&dst, c->status[i].sample1);
659 for (i = 0; i < avctx->channels; i++)
660 bytestream_put_le16(&dst, c->status[i].sample2);
662 if (avctx->trellis > 0) {
663 int n = avctx->block_align - 7 * avctx->channels;
664 FF_ALLOC_OR_GOTO(avctx, buf, 2 * n, error);
665 if (avctx->channels == 1) {
666 adpcm_compress_trellis(avctx, samples, buf, &c->status[0], n);
667 for (i = 0; i < n; i += 2)
668 *dst++ = (buf[i] << 4) | buf[i + 1];
670 adpcm_compress_trellis(avctx, samples, buf, &c->status[0], n);
671 adpcm_compress_trellis(avctx, samples + 1, buf + n, &c->status[1], n);
672 for (i = 0; i < n; i++)
673 *dst++ = (buf[i] << 4) | buf[n + i];
677 for (i = 7 * avctx->channels; i < avctx->block_align; i++) {
679 nibble = adpcm_ms_compress_sample(&c->status[ 0], *samples++) << 4;
680 nibble |= adpcm_ms_compress_sample(&c->status[st], *samples++);
685 case AV_CODEC_ID_ADPCM_YAMAHA:
686 n = frame->nb_samples / 2;
687 if (avctx->trellis > 0) {
688 FF_ALLOC_OR_GOTO(avctx, buf, 2 * n * 2, error);
690 if (avctx->channels == 1) {
691 adpcm_compress_trellis(avctx, samples, buf, &c->status[0], n);
692 for (i = 0; i < n; i += 2)
693 *dst++ = buf[i] | (buf[i + 1] << 4);
695 adpcm_compress_trellis(avctx, samples, buf, &c->status[0], n);
696 adpcm_compress_trellis(avctx, samples + 1, buf + n, &c->status[1], n);
697 for (i = 0; i < n; i++)
698 *dst++ = buf[i] | (buf[n + i] << 4);
702 for (n *= avctx->channels; n > 0; n--) {
704 nibble = adpcm_yamaha_compress_sample(&c->status[ 0], *samples++);
705 nibble |= adpcm_yamaha_compress_sample(&c->status[st], *samples++) << 4;
710 return AVERROR(EINVAL);
713 avpkt->size = pkt_size;
717 return AVERROR(ENOMEM);
720 static const enum AVSampleFormat sample_fmts[] = {
721 AV_SAMPLE_FMT_S16, AV_SAMPLE_FMT_NONE
724 #define ADPCM_ENCODER(id_, name_, long_name_) \
725 AVCodec ff_ ## name_ ## _encoder = { \
727 .type = AVMEDIA_TYPE_AUDIO, \
729 .priv_data_size = sizeof(ADPCMEncodeContext), \
730 .init = adpcm_encode_init, \
731 .encode2 = adpcm_encode_frame, \
732 .close = adpcm_encode_close, \
733 .sample_fmts = sample_fmts, \
734 .long_name = NULL_IF_CONFIG_SMALL(long_name_), \
737 ADPCM_ENCODER(AV_CODEC_ID_ADPCM_IMA_QT, adpcm_ima_qt, "ADPCM IMA QuickTime");
738 ADPCM_ENCODER(AV_CODEC_ID_ADPCM_IMA_WAV, adpcm_ima_wav, "ADPCM IMA WAV");
739 ADPCM_ENCODER(AV_CODEC_ID_ADPCM_MS, adpcm_ms, "ADPCM Microsoft");
740 ADPCM_ENCODER(AV_CODEC_ID_ADPCM_SWF, adpcm_swf, "ADPCM Shockwave Flash");
741 ADPCM_ENCODER(AV_CODEC_ID_ADPCM_YAMAHA, adpcm_yamaha, "ADPCM Yamaha");