Commit | Line | Data |
---|---|---|
2ba45a60 DM |
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) { | |
2ba45a60 DM |
432 | for (i = 0; i < s->frame_size; i++) { |
433 | for (j = 0; j < channels; j++) { | |
f6fa7814 | 434 | put_bits(pb, s->extra_bits, s->predictor_buf[j][i]); |
2ba45a60 DM |
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 | |
f6fa7814 | 446 | int32_t *residual = s->predictor_buf[i]; |
2ba45a60 DM |
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 | }; |