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");