Imported Debian version 2.5.3~trusty1
[deb_ffmpeg.git] / ffmpeg / libavcodec / alacenc.c
1 /*
2 * ALAC audio encoder
3 * Copyright (c) 2008 Jaikrishnan Menon <realityman@gmx.net>
4 *
5 * This file is part of FFmpeg.
6 *
7 * FFmpeg is free software; you can redistribute it and/or
8 * modify it under the terms of the GNU Lesser General Public
9 * License as published by the Free Software Foundation; either
10 * version 2.1 of the License, or (at your option) any later version.
11 *
12 * FFmpeg is distributed in the hope that it will be useful,
13 * but WITHOUT ANY WARRANTY; without even the implied warranty of
14 * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
15 * Lesser General Public License for more details.
16 *
17 * You should have received a copy of the GNU Lesser General Public
18 * License along with FFmpeg; if not, write to the Free Software
19 * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
20 */
21
22 #include "avcodec.h"
23 #include "put_bits.h"
24 #include "internal.h"
25 #include "lpc.h"
26 #include "mathops.h"
27 #include "alac_data.h"
28
29 #define DEFAULT_FRAME_SIZE 4096
30 #define ALAC_EXTRADATA_SIZE 36
31 #define ALAC_FRAME_HEADER_SIZE 55
32 #define ALAC_FRAME_FOOTER_SIZE 3
33
34 #define ALAC_ESCAPE_CODE 0x1FF
35 #define ALAC_MAX_LPC_ORDER 30
36 #define DEFAULT_MAX_PRED_ORDER 6
37 #define DEFAULT_MIN_PRED_ORDER 4
38 #define ALAC_MAX_LPC_PRECISION 9
39 #define ALAC_MAX_LPC_SHIFT 9
40
41 #define ALAC_CHMODE_LEFT_RIGHT 0
42 #define ALAC_CHMODE_LEFT_SIDE 1
43 #define ALAC_CHMODE_RIGHT_SIDE 2
44 #define ALAC_CHMODE_MID_SIDE 3
45
46 typedef struct RiceContext {
47 int history_mult;
48 int initial_history;
49 int k_modifier;
50 int rice_modifier;
51 } RiceContext;
52
53 typedef struct AlacLPCContext {
54 int lpc_order;
55 int lpc_coeff[ALAC_MAX_LPC_ORDER+1];
56 int lpc_quant;
57 } AlacLPCContext;
58
59 typedef struct AlacEncodeContext {
60 int frame_size; /**< current frame size */
61 int verbatim; /**< current frame verbatim mode flag */
62 int compression_level;
63 int min_prediction_order;
64 int max_prediction_order;
65 int max_coded_frame_size;
66 int write_sample_size;
67 int extra_bits;
68 int32_t sample_buf[2][DEFAULT_FRAME_SIZE];
69 int32_t predictor_buf[2][DEFAULT_FRAME_SIZE];
70 int interlacing_shift;
71 int interlacing_leftweight;
72 PutBitContext pbctx;
73 RiceContext rc;
74 AlacLPCContext lpc[2];
75 LPCContext lpc_ctx;
76 AVCodecContext *avctx;
77 } AlacEncodeContext;
78
79
80 static void init_sample_buffers(AlacEncodeContext *s, int channels,
81 uint8_t const *samples[2])
82 {
83 int ch, i;
84 int shift = av_get_bytes_per_sample(s->avctx->sample_fmt) * 8 -
85 s->avctx->bits_per_raw_sample;
86
87 #define COPY_SAMPLES(type) do { \
88 for (ch = 0; ch < channels; ch++) { \
89 int32_t *bptr = s->sample_buf[ch]; \
90 const type *sptr = (const type *)samples[ch]; \
91 for (i = 0; i < s->frame_size; i++) \
92 bptr[i] = sptr[i] >> shift; \
93 } \
94 } while (0)
95
96 if (s->avctx->sample_fmt == AV_SAMPLE_FMT_S32P)
97 COPY_SAMPLES(int32_t);
98 else
99 COPY_SAMPLES(int16_t);
100 }
101
102 static void encode_scalar(AlacEncodeContext *s, int x,
103 int k, int write_sample_size)
104 {
105 int divisor, q, r;
106
107 k = FFMIN(k, s->rc.k_modifier);
108 divisor = (1<<k) - 1;
109 q = x / divisor;
110 r = x % divisor;
111
112 if (q > 8) {
113 // write escape code and sample value directly
114 put_bits(&s->pbctx, 9, ALAC_ESCAPE_CODE);
115 put_bits(&s->pbctx, write_sample_size, x);
116 } else {
117 if (q)
118 put_bits(&s->pbctx, q, (1<<q) - 1);
119 put_bits(&s->pbctx, 1, 0);
120
121 if (k != 1) {
122 if (r > 0)
123 put_bits(&s->pbctx, k, r+1);
124 else
125 put_bits(&s->pbctx, k-1, 0);
126 }
127 }
128 }
129
130 static void write_element_header(AlacEncodeContext *s,
131 enum AlacRawDataBlockType element,
132 int instance)
133 {
134 int encode_fs = 0;
135
136 if (s->frame_size < DEFAULT_FRAME_SIZE)
137 encode_fs = 1;
138
139 put_bits(&s->pbctx, 3, element); // element type
140 put_bits(&s->pbctx, 4, instance); // element instance
141 put_bits(&s->pbctx, 12, 0); // unused header bits
142 put_bits(&s->pbctx, 1, encode_fs); // Sample count is in the header
143 put_bits(&s->pbctx, 2, s->extra_bits >> 3); // Extra bytes (for 24-bit)
144 put_bits(&s->pbctx, 1, s->verbatim); // Audio block is verbatim
145 if (encode_fs)
146 put_bits32(&s->pbctx, s->frame_size); // No. of samples in the frame
147 }
148
149 static void calc_predictor_params(AlacEncodeContext *s, int ch)
150 {
151 int32_t coefs[MAX_LPC_ORDER][MAX_LPC_ORDER];
152 int shift[MAX_LPC_ORDER];
153 int opt_order;
154
155 if (s->compression_level == 1) {
156 s->lpc[ch].lpc_order = 6;
157 s->lpc[ch].lpc_quant = 6;
158 s->lpc[ch].lpc_coeff[0] = 160;
159 s->lpc[ch].lpc_coeff[1] = -190;
160 s->lpc[ch].lpc_coeff[2] = 170;
161 s->lpc[ch].lpc_coeff[3] = -130;
162 s->lpc[ch].lpc_coeff[4] = 80;
163 s->lpc[ch].lpc_coeff[5] = -25;
164 } else {
165 opt_order = ff_lpc_calc_coefs(&s->lpc_ctx, s->sample_buf[ch],
166 s->frame_size,
167 s->min_prediction_order,
168 s->max_prediction_order,
169 ALAC_MAX_LPC_PRECISION, coefs, shift,
170 FF_LPC_TYPE_LEVINSON, 0,
171 ORDER_METHOD_EST, ALAC_MAX_LPC_SHIFT, 1);
172
173 s->lpc[ch].lpc_order = opt_order;
174 s->lpc[ch].lpc_quant = shift[opt_order-1];
175 memcpy(s->lpc[ch].lpc_coeff, coefs[opt_order-1], opt_order*sizeof(int));
176 }
177 }
178
179 static int estimate_stereo_mode(int32_t *left_ch, int32_t *right_ch, int n)
180 {
181 int i, best;
182 int32_t lt, rt;
183 uint64_t sum[4];
184 uint64_t score[4];
185
186 /* calculate sum of 2nd order residual for each channel */
187 sum[0] = sum[1] = sum[2] = sum[3] = 0;
188 for (i = 2; i < n; i++) {
189 lt = left_ch[i] - 2 * left_ch[i - 1] + left_ch[i - 2];
190 rt = right_ch[i] - 2 * right_ch[i - 1] + right_ch[i - 2];
191 sum[2] += FFABS((lt + rt) >> 1);
192 sum[3] += FFABS(lt - rt);
193 sum[0] += FFABS(lt);
194 sum[1] += FFABS(rt);
195 }
196
197 /* calculate score for each mode */
198 score[0] = sum[0] + sum[1];
199 score[1] = sum[0] + sum[3];
200 score[2] = sum[1] + sum[3];
201 score[3] = sum[2] + sum[3];
202
203 /* return mode with lowest score */
204 best = 0;
205 for (i = 1; i < 4; i++) {
206 if (score[i] < score[best])
207 best = i;
208 }
209 return best;
210 }
211
212 static void alac_stereo_decorrelation(AlacEncodeContext *s)
213 {
214 int32_t *left = s->sample_buf[0], *right = s->sample_buf[1];
215 int i, mode, n = s->frame_size;
216 int32_t tmp;
217
218 mode = estimate_stereo_mode(left, right, n);
219
220 switch (mode) {
221 case ALAC_CHMODE_LEFT_RIGHT:
222 s->interlacing_leftweight = 0;
223 s->interlacing_shift = 0;
224 break;
225 case ALAC_CHMODE_LEFT_SIDE:
226 for (i = 0; i < n; i++)
227 right[i] = left[i] - right[i];
228 s->interlacing_leftweight = 1;
229 s->interlacing_shift = 0;
230 break;
231 case ALAC_CHMODE_RIGHT_SIDE:
232 for (i = 0; i < n; i++) {
233 tmp = right[i];
234 right[i] = left[i] - right[i];
235 left[i] = tmp + (right[i] >> 31);
236 }
237 s->interlacing_leftweight = 1;
238 s->interlacing_shift = 31;
239 break;
240 default:
241 for (i = 0; i < n; i++) {
242 tmp = left[i];
243 left[i] = (tmp + right[i]) >> 1;
244 right[i] = tmp - right[i];
245 }
246 s->interlacing_leftweight = 1;
247 s->interlacing_shift = 1;
248 break;
249 }
250 }
251
252 static void alac_linear_predictor(AlacEncodeContext *s, int ch)
253 {
254 int i;
255 AlacLPCContext lpc = s->lpc[ch];
256 int32_t *residual = s->predictor_buf[ch];
257
258 if (lpc.lpc_order == 31) {
259 residual[0] = s->sample_buf[ch][0];
260
261 for (i = 1; i < s->frame_size; i++) {
262 residual[i] = s->sample_buf[ch][i ] -
263 s->sample_buf[ch][i - 1];
264 }
265
266 return;
267 }
268
269 // generalised linear predictor
270
271 if (lpc.lpc_order > 0) {
272 int32_t *samples = s->sample_buf[ch];
273
274 // generate warm-up samples
275 residual[0] = samples[0];
276 for (i = 1; i <= lpc.lpc_order; i++)
277 residual[i] = sign_extend(samples[i] - samples[i-1], s->write_sample_size);
278
279 // perform lpc on remaining samples
280 for (i = lpc.lpc_order + 1; i < s->frame_size; i++) {
281 int sum = 1 << (lpc.lpc_quant - 1), res_val, j;
282
283 for (j = 0; j < lpc.lpc_order; j++) {
284 sum += (samples[lpc.lpc_order-j] - samples[0]) *
285 lpc.lpc_coeff[j];
286 }
287
288 sum >>= lpc.lpc_quant;
289 sum += samples[0];
290 residual[i] = sign_extend(samples[lpc.lpc_order+1] - sum,
291 s->write_sample_size);
292 res_val = residual[i];
293
294 if (res_val) {
295 int index = lpc.lpc_order - 1;
296 int neg = (res_val < 0);
297
298 while (index >= 0 && (neg ? (res_val < 0) : (res_val > 0))) {
299 int val = samples[0] - samples[lpc.lpc_order - index];
300 int sign = (val ? FFSIGN(val) : 0);
301
302 if (neg)
303 sign *= -1;
304
305 lpc.lpc_coeff[index] -= sign;
306 val *= sign;
307 res_val -= (val >> lpc.lpc_quant) * (lpc.lpc_order - index);
308 index--;
309 }
310 }
311 samples++;
312 }
313 }
314 }
315
316 static void alac_entropy_coder(AlacEncodeContext *s, int ch)
317 {
318 unsigned int history = s->rc.initial_history;
319 int sign_modifier = 0, i, k;
320 int32_t *samples = s->predictor_buf[ch];
321
322 for (i = 0; i < s->frame_size;) {
323 int x;
324
325 k = av_log2((history >> 9) + 3);
326
327 x = -2 * (*samples) -1;
328 x ^= x >> 31;
329
330 samples++;
331 i++;
332
333 encode_scalar(s, x - sign_modifier, k, s->write_sample_size);
334
335 history += x * s->rc.history_mult -
336 ((history * s->rc.history_mult) >> 9);
337
338 sign_modifier = 0;
339 if (x > 0xFFFF)
340 history = 0xFFFF;
341
342 if (history < 128 && i < s->frame_size) {
343 unsigned int block_size = 0;
344
345 k = 7 - av_log2(history) + ((history + 16) >> 6);
346
347 while (*samples == 0 && i < s->frame_size) {
348 samples++;
349 i++;
350 block_size++;
351 }
352 encode_scalar(s, block_size, k, 16);
353 sign_modifier = (block_size <= 0xFFFF);
354 history = 0;
355 }
356
357 }
358 }
359
360 static void write_element(AlacEncodeContext *s,
361 enum AlacRawDataBlockType element, int instance,
362 const uint8_t *samples0, const uint8_t *samples1)
363 {
364 uint8_t const *samples[2] = { samples0, samples1 };
365 int i, j, channels;
366 int prediction_type = 0;
367 PutBitContext *pb = &s->pbctx;
368
369 channels = element == TYPE_CPE ? 2 : 1;
370
371 if (s->verbatim) {
372 write_element_header(s, element, instance);
373 /* samples are channel-interleaved in verbatim mode */
374 if (s->avctx->sample_fmt == AV_SAMPLE_FMT_S32P) {
375 int shift = 32 - s->avctx->bits_per_raw_sample;
376 int32_t const *samples_s32[2] = { (const int32_t *)samples0,
377 (const int32_t *)samples1 };
378 for (i = 0; i < s->frame_size; i++)
379 for (j = 0; j < channels; j++)
380 put_sbits(pb, s->avctx->bits_per_raw_sample,
381 samples_s32[j][i] >> shift);
382 } else {
383 int16_t const *samples_s16[2] = { (const int16_t *)samples0,
384 (const int16_t *)samples1 };
385 for (i = 0; i < s->frame_size; i++)
386 for (j = 0; j < channels; j++)
387 put_sbits(pb, s->avctx->bits_per_raw_sample,
388 samples_s16[j][i]);
389 }
390 } else {
391 s->write_sample_size = s->avctx->bits_per_raw_sample - s->extra_bits +
392 channels - 1;
393
394 init_sample_buffers(s, channels, samples);
395 write_element_header(s, element, instance);
396
397 // extract extra bits if needed
398 if (s->extra_bits) {
399 uint32_t mask = (1 << s->extra_bits) - 1;
400 for (j = 0; j < channels; j++) {
401 int32_t *extra = s->predictor_buf[j];
402 int32_t *smp = s->sample_buf[j];
403 for (i = 0; i < s->frame_size; i++) {
404 extra[i] = smp[i] & mask;
405 smp[i] >>= s->extra_bits;
406 }
407 }
408 }
409
410 if (channels == 2)
411 alac_stereo_decorrelation(s);
412 else
413 s->interlacing_shift = s->interlacing_leftweight = 0;
414 put_bits(pb, 8, s->interlacing_shift);
415 put_bits(pb, 8, s->interlacing_leftweight);
416
417 for (i = 0; i < channels; i++) {
418 calc_predictor_params(s, i);
419
420 put_bits(pb, 4, prediction_type);
421 put_bits(pb, 4, s->lpc[i].lpc_quant);
422
423 put_bits(pb, 3, s->rc.rice_modifier);
424 put_bits(pb, 5, s->lpc[i].lpc_order);
425 // predictor coeff. table
426 for (j = 0; j < s->lpc[i].lpc_order; j++)
427 put_sbits(pb, 16, s->lpc[i].lpc_coeff[j]);
428 }
429
430 // write extra bits if needed
431 if (s->extra_bits) {
432 for (i = 0; i < s->frame_size; i++) {
433 for (j = 0; j < channels; j++) {
434 put_bits(pb, s->extra_bits, s->predictor_buf[j][i]);
435 }
436 }
437 }
438
439 // apply lpc and entropy coding to audio samples
440 for (i = 0; i < channels; i++) {
441 alac_linear_predictor(s, i);
442
443 // TODO: determine when this will actually help. for now it's not used.
444 if (prediction_type == 15) {
445 // 2nd pass 1st order filter
446 int32_t *residual = s->predictor_buf[i];
447 for (j = s->frame_size - 1; j > 0; j--)
448 residual[j] -= residual[j - 1];
449 }
450 alac_entropy_coder(s, i);
451 }
452 }
453 }
454
455 static int write_frame(AlacEncodeContext *s, AVPacket *avpkt,
456 uint8_t * const *samples)
457 {
458 PutBitContext *pb = &s->pbctx;
459 const enum AlacRawDataBlockType *ch_elements = ff_alac_channel_elements[s->avctx->channels - 1];
460 const uint8_t *ch_map = ff_alac_channel_layout_offsets[s->avctx->channels - 1];
461 int ch, element, sce, cpe;
462
463 init_put_bits(pb, avpkt->data, avpkt->size);
464
465 ch = element = sce = cpe = 0;
466 while (ch < s->avctx->channels) {
467 if (ch_elements[element] == TYPE_CPE) {
468 write_element(s, TYPE_CPE, cpe, samples[ch_map[ch]],
469 samples[ch_map[ch + 1]]);
470 cpe++;
471 ch += 2;
472 } else {
473 write_element(s, TYPE_SCE, sce, samples[ch_map[ch]], NULL);
474 sce++;
475 ch++;
476 }
477 element++;
478 }
479
480 put_bits(pb, 3, TYPE_END);
481 flush_put_bits(pb);
482
483 return put_bits_count(pb) >> 3;
484 }
485
486 static av_always_inline int get_max_frame_size(int frame_size, int ch, int bps)
487 {
488 int header_bits = 23 + 32 * (frame_size < DEFAULT_FRAME_SIZE);
489 return FFALIGN(header_bits + bps * ch * frame_size + 3, 8) / 8;
490 }
491
492 static av_cold int alac_encode_close(AVCodecContext *avctx)
493 {
494 AlacEncodeContext *s = avctx->priv_data;
495 ff_lpc_end(&s->lpc_ctx);
496 av_freep(&avctx->extradata);
497 avctx->extradata_size = 0;
498 return 0;
499 }
500
501 static av_cold int alac_encode_init(AVCodecContext *avctx)
502 {
503 AlacEncodeContext *s = avctx->priv_data;
504 int ret;
505 uint8_t *alac_extradata;
506
507 avctx->frame_size = s->frame_size = DEFAULT_FRAME_SIZE;
508
509 if (avctx->sample_fmt == AV_SAMPLE_FMT_S32P) {
510 if (avctx->bits_per_raw_sample != 24)
511 av_log(avctx, AV_LOG_WARNING, "encoding as 24 bits-per-sample\n");
512 avctx->bits_per_raw_sample = 24;
513 } else {
514 avctx->bits_per_raw_sample = 16;
515 s->extra_bits = 0;
516 }
517
518 // Set default compression level
519 if (avctx->compression_level == FF_COMPRESSION_DEFAULT)
520 s->compression_level = 2;
521 else
522 s->compression_level = av_clip(avctx->compression_level, 0, 2);
523
524 // Initialize default Rice parameters
525 s->rc.history_mult = 40;
526 s->rc.initial_history = 10;
527 s->rc.k_modifier = 14;
528 s->rc.rice_modifier = 4;
529
530 s->max_coded_frame_size = get_max_frame_size(avctx->frame_size,
531 avctx->channels,
532 avctx->bits_per_raw_sample);
533
534 avctx->extradata = av_mallocz(ALAC_EXTRADATA_SIZE + FF_INPUT_BUFFER_PADDING_SIZE);
535 if (!avctx->extradata) {
536 ret = AVERROR(ENOMEM);
537 goto error;
538 }
539 avctx->extradata_size = ALAC_EXTRADATA_SIZE;
540
541 alac_extradata = avctx->extradata;
542 AV_WB32(alac_extradata, ALAC_EXTRADATA_SIZE);
543 AV_WB32(alac_extradata+4, MKBETAG('a','l','a','c'));
544 AV_WB32(alac_extradata+12, avctx->frame_size);
545 AV_WB8 (alac_extradata+17, avctx->bits_per_raw_sample);
546 AV_WB8 (alac_extradata+21, avctx->channels);
547 AV_WB32(alac_extradata+24, s->max_coded_frame_size);
548 AV_WB32(alac_extradata+28,
549 avctx->sample_rate * avctx->channels * avctx->bits_per_raw_sample); // average bitrate
550 AV_WB32(alac_extradata+32, avctx->sample_rate);
551
552 // Set relevant extradata fields
553 if (s->compression_level > 0) {
554 AV_WB8(alac_extradata+18, s->rc.history_mult);
555 AV_WB8(alac_extradata+19, s->rc.initial_history);
556 AV_WB8(alac_extradata+20, s->rc.k_modifier);
557 }
558
559 s->min_prediction_order = DEFAULT_MIN_PRED_ORDER;
560 if (avctx->min_prediction_order >= 0) {
561 if (avctx->min_prediction_order < MIN_LPC_ORDER ||
562 avctx->min_prediction_order > ALAC_MAX_LPC_ORDER) {
563 av_log(avctx, AV_LOG_ERROR, "invalid min prediction order: %d\n",
564 avctx->min_prediction_order);
565 ret = AVERROR(EINVAL);
566 goto error;
567 }
568
569 s->min_prediction_order = avctx->min_prediction_order;
570 }
571
572 s->max_prediction_order = DEFAULT_MAX_PRED_ORDER;
573 if (avctx->max_prediction_order >= 0) {
574 if (avctx->max_prediction_order < MIN_LPC_ORDER ||
575 avctx->max_prediction_order > ALAC_MAX_LPC_ORDER) {
576 av_log(avctx, AV_LOG_ERROR, "invalid max prediction order: %d\n",
577 avctx->max_prediction_order);
578 ret = AVERROR(EINVAL);
579 goto error;
580 }
581
582 s->max_prediction_order = avctx->max_prediction_order;
583 }
584
585 if (s->max_prediction_order < s->min_prediction_order) {
586 av_log(avctx, AV_LOG_ERROR,
587 "invalid prediction orders: min=%d max=%d\n",
588 s->min_prediction_order, s->max_prediction_order);
589 ret = AVERROR(EINVAL);
590 goto error;
591 }
592
593 s->avctx = avctx;
594
595 if ((ret = ff_lpc_init(&s->lpc_ctx, avctx->frame_size,
596 s->max_prediction_order,
597 FF_LPC_TYPE_LEVINSON)) < 0) {
598 goto error;
599 }
600
601 return 0;
602 error:
603 alac_encode_close(avctx);
604 return ret;
605 }
606
607 static int alac_encode_frame(AVCodecContext *avctx, AVPacket *avpkt,
608 const AVFrame *frame, int *got_packet_ptr)
609 {
610 AlacEncodeContext *s = avctx->priv_data;
611 int out_bytes, max_frame_size, ret;
612
613 s->frame_size = frame->nb_samples;
614
615 if (frame->nb_samples < DEFAULT_FRAME_SIZE)
616 max_frame_size = get_max_frame_size(s->frame_size, avctx->channels,
617 avctx->bits_per_raw_sample);
618 else
619 max_frame_size = s->max_coded_frame_size;
620
621 if ((ret = ff_alloc_packet2(avctx, avpkt, 2 * max_frame_size)) < 0)
622 return ret;
623
624 /* use verbatim mode for compression_level 0 */
625 if (s->compression_level) {
626 s->verbatim = 0;
627 s->extra_bits = avctx->bits_per_raw_sample - 16;
628 } else {
629 s->verbatim = 1;
630 s->extra_bits = 0;
631 }
632
633 out_bytes = write_frame(s, avpkt, frame->extended_data);
634
635 if (out_bytes > max_frame_size) {
636 /* frame too large. use verbatim mode */
637 s->verbatim = 1;
638 s->extra_bits = 0;
639 out_bytes = write_frame(s, avpkt, frame->extended_data);
640 }
641
642 avpkt->size = out_bytes;
643 *got_packet_ptr = 1;
644 return 0;
645 }
646
647 AVCodec ff_alac_encoder = {
648 .name = "alac",
649 .long_name = NULL_IF_CONFIG_SMALL("ALAC (Apple Lossless Audio Codec)"),
650 .type = AVMEDIA_TYPE_AUDIO,
651 .id = AV_CODEC_ID_ALAC,
652 .priv_data_size = sizeof(AlacEncodeContext),
653 .init = alac_encode_init,
654 .encode2 = alac_encode_frame,
655 .close = alac_encode_close,
656 .capabilities = CODEC_CAP_SMALL_LAST_FRAME,
657 .channel_layouts = ff_alac_channel_layouts,
658 .sample_fmts = (const enum AVSampleFormat[]){ AV_SAMPLE_FMT_S32P,
659 AV_SAMPLE_FMT_S16P,
660 AV_SAMPLE_FMT_NONE },
661 };