Remove useless header inclusion.
[libav.git] / 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 "dsputil.h"
25 #include "lpc.h"
26 #include "mathops.h"
27
28 #define DEFAULT_FRAME_SIZE 4096
29 #define DEFAULT_SAMPLE_SIZE 16
30 #define MAX_CHANNELS 8
31 #define ALAC_EXTRADATA_SIZE 36
32 #define ALAC_FRAME_HEADER_SIZE 55
33 #define ALAC_FRAME_FOOTER_SIZE 3
34
35 #define ALAC_ESCAPE_CODE 0x1FF
36 #define ALAC_MAX_LPC_ORDER 30
37 #define DEFAULT_MAX_PRED_ORDER 6
38 #define DEFAULT_MIN_PRED_ORDER 4
39 #define ALAC_MAX_LPC_PRECISION 9
40 #define ALAC_MAX_LPC_SHIFT 9
41
42 #define ALAC_CHMODE_LEFT_RIGHT 0
43 #define ALAC_CHMODE_LEFT_SIDE 1
44 #define ALAC_CHMODE_RIGHT_SIDE 2
45 #define ALAC_CHMODE_MID_SIDE 3
46
47 typedef struct RiceContext {
48 int history_mult;
49 int initial_history;
50 int k_modifier;
51 int rice_modifier;
52 } RiceContext;
53
54 typedef struct LPCContext {
55 int lpc_order;
56 int lpc_coeff[ALAC_MAX_LPC_ORDER+1];
57 int lpc_quant;
58 } LPCContext;
59
60 typedef struct AlacEncodeContext {
61 int compression_level;
62 int min_prediction_order;
63 int max_prediction_order;
64 int max_coded_frame_size;
65 int write_sample_size;
66 int32_t sample_buf[MAX_CHANNELS][DEFAULT_FRAME_SIZE];
67 int32_t predictor_buf[DEFAULT_FRAME_SIZE];
68 int interlacing_shift;
69 int interlacing_leftweight;
70 PutBitContext pbctx;
71 RiceContext rc;
72 LPCContext lpc[MAX_CHANNELS];
73 DSPContext dspctx;
74 AVCodecContext *avctx;
75 } AlacEncodeContext;
76
77
78 static void init_sample_buffers(AlacEncodeContext *s, int16_t *input_samples)
79 {
80 int ch, i;
81
82 for(ch=0;ch<s->avctx->channels;ch++) {
83 int16_t *sptr = input_samples + ch;
84 for(i=0;i<s->avctx->frame_size;i++) {
85 s->sample_buf[ch][i] = *sptr;
86 sptr += s->avctx->channels;
87 }
88 }
89 }
90
91 static void encode_scalar(AlacEncodeContext *s, int x, int k, int write_sample_size)
92 {
93 int divisor, q, r;
94
95 k = FFMIN(k, s->rc.k_modifier);
96 divisor = (1<<k) - 1;
97 q = x / divisor;
98 r = x % divisor;
99
100 if(q > 8) {
101 // write escape code and sample value directly
102 put_bits(&s->pbctx, 9, ALAC_ESCAPE_CODE);
103 put_bits(&s->pbctx, write_sample_size, x);
104 } else {
105 if(q)
106 put_bits(&s->pbctx, q, (1<<q) - 1);
107 put_bits(&s->pbctx, 1, 0);
108
109 if(k != 1) {
110 if(r > 0)
111 put_bits(&s->pbctx, k, r+1);
112 else
113 put_bits(&s->pbctx, k-1, 0);
114 }
115 }
116 }
117
118 static void write_frame_header(AlacEncodeContext *s, int is_verbatim)
119 {
120 put_bits(&s->pbctx, 3, s->avctx->channels-1); // No. of channels -1
121 put_bits(&s->pbctx, 16, 0); // Seems to be zero
122 put_bits(&s->pbctx, 1, 1); // Sample count is in the header
123 put_bits(&s->pbctx, 2, 0); // FIXME: Wasted bytes field
124 put_bits(&s->pbctx, 1, is_verbatim); // Audio block is verbatim
125 put_bits32(&s->pbctx, s->avctx->frame_size); // No. of samples in the frame
126 }
127
128 static void calc_predictor_params(AlacEncodeContext *s, int ch)
129 {
130 int32_t coefs[MAX_LPC_ORDER][MAX_LPC_ORDER];
131 int shift[MAX_LPC_ORDER];
132 int opt_order;
133
134 if (s->compression_level == 1) {
135 s->lpc[ch].lpc_order = 6;
136 s->lpc[ch].lpc_quant = 6;
137 s->lpc[ch].lpc_coeff[0] = 160;
138 s->lpc[ch].lpc_coeff[1] = -190;
139 s->lpc[ch].lpc_coeff[2] = 170;
140 s->lpc[ch].lpc_coeff[3] = -130;
141 s->lpc[ch].lpc_coeff[4] = 80;
142 s->lpc[ch].lpc_coeff[5] = -25;
143 } else {
144 opt_order = ff_lpc_calc_coefs(&s->dspctx, s->sample_buf[ch],
145 s->avctx->frame_size,
146 s->min_prediction_order,
147 s->max_prediction_order,
148 ALAC_MAX_LPC_PRECISION, coefs, shift, 1,
149 ORDER_METHOD_EST, ALAC_MAX_LPC_SHIFT, 1);
150
151 s->lpc[ch].lpc_order = opt_order;
152 s->lpc[ch].lpc_quant = shift[opt_order-1];
153 memcpy(s->lpc[ch].lpc_coeff, coefs[opt_order-1], opt_order*sizeof(int));
154 }
155 }
156
157 static int estimate_stereo_mode(int32_t *left_ch, int32_t *right_ch, int n)
158 {
159 int i, best;
160 int32_t lt, rt;
161 uint64_t sum[4];
162 uint64_t score[4];
163
164 /* calculate sum of 2nd order residual for each channel */
165 sum[0] = sum[1] = sum[2] = sum[3] = 0;
166 for(i=2; i<n; i++) {
167 lt = left_ch[i] - 2*left_ch[i-1] + left_ch[i-2];
168 rt = right_ch[i] - 2*right_ch[i-1] + right_ch[i-2];
169 sum[2] += FFABS((lt + rt) >> 1);
170 sum[3] += FFABS(lt - rt);
171 sum[0] += FFABS(lt);
172 sum[1] += FFABS(rt);
173 }
174
175 /* calculate score for each mode */
176 score[0] = sum[0] + sum[1];
177 score[1] = sum[0] + sum[3];
178 score[2] = sum[1] + sum[3];
179 score[3] = sum[2] + sum[3];
180
181 /* return mode with lowest score */
182 best = 0;
183 for(i=1; i<4; i++) {
184 if(score[i] < score[best]) {
185 best = i;
186 }
187 }
188 return best;
189 }
190
191 static void alac_stereo_decorrelation(AlacEncodeContext *s)
192 {
193 int32_t *left = s->sample_buf[0], *right = s->sample_buf[1];
194 int i, mode, n = s->avctx->frame_size;
195 int32_t tmp;
196
197 mode = estimate_stereo_mode(left, right, n);
198
199 switch(mode)
200 {
201 case ALAC_CHMODE_LEFT_RIGHT:
202 s->interlacing_leftweight = 0;
203 s->interlacing_shift = 0;
204 break;
205
206 case ALAC_CHMODE_LEFT_SIDE:
207 for(i=0; i<n; i++) {
208 right[i] = left[i] - right[i];
209 }
210 s->interlacing_leftweight = 1;
211 s->interlacing_shift = 0;
212 break;
213
214 case ALAC_CHMODE_RIGHT_SIDE:
215 for(i=0; i<n; i++) {
216 tmp = right[i];
217 right[i] = left[i] - right[i];
218 left[i] = tmp + (right[i] >> 31);
219 }
220 s->interlacing_leftweight = 1;
221 s->interlacing_shift = 31;
222 break;
223
224 default:
225 for(i=0; i<n; i++) {
226 tmp = left[i];
227 left[i] = (tmp + right[i]) >> 1;
228 right[i] = tmp - right[i];
229 }
230 s->interlacing_leftweight = 1;
231 s->interlacing_shift = 1;
232 break;
233 }
234 }
235
236 static void alac_linear_predictor(AlacEncodeContext *s, int ch)
237 {
238 int i;
239 LPCContext lpc = s->lpc[ch];
240
241 if(lpc.lpc_order == 31) {
242 s->predictor_buf[0] = s->sample_buf[ch][0];
243
244 for(i=1; i<s->avctx->frame_size; i++)
245 s->predictor_buf[i] = s->sample_buf[ch][i] - s->sample_buf[ch][i-1];
246
247 return;
248 }
249
250 // generalised linear predictor
251
252 if(lpc.lpc_order > 0) {
253 int32_t *samples = s->sample_buf[ch];
254 int32_t *residual = s->predictor_buf;
255
256 // generate warm-up samples
257 residual[0] = samples[0];
258 for(i=1;i<=lpc.lpc_order;i++)
259 residual[i] = samples[i] - samples[i-1];
260
261 // perform lpc on remaining samples
262 for(i = lpc.lpc_order + 1; i < s->avctx->frame_size; i++) {
263 int sum = 1 << (lpc.lpc_quant - 1), res_val, j;
264
265 for (j = 0; j < lpc.lpc_order; j++) {
266 sum += (samples[lpc.lpc_order-j] - samples[0]) *
267 lpc.lpc_coeff[j];
268 }
269
270 sum >>= lpc.lpc_quant;
271 sum += samples[0];
272 residual[i] = sign_extend(samples[lpc.lpc_order+1] - sum,
273 s->write_sample_size);
274 res_val = residual[i];
275
276 if(res_val) {
277 int index = lpc.lpc_order - 1;
278 int neg = (res_val < 0);
279
280 while(index >= 0 && (neg ? (res_val < 0):(res_val > 0))) {
281 int val = samples[0] - samples[lpc.lpc_order - index];
282 int sign = (val ? FFSIGN(val) : 0);
283
284 if(neg)
285 sign*=-1;
286
287 lpc.lpc_coeff[index] -= sign;
288 val *= sign;
289 res_val -= ((val >> lpc.lpc_quant) *
290 (lpc.lpc_order - index));
291 index--;
292 }
293 }
294 samples++;
295 }
296 }
297 }
298
299 static void alac_entropy_coder(AlacEncodeContext *s)
300 {
301 unsigned int history = s->rc.initial_history;
302 int sign_modifier = 0, i, k;
303 int32_t *samples = s->predictor_buf;
304
305 for(i=0;i < s->avctx->frame_size;) {
306 int x;
307
308 k = av_log2((history >> 9) + 3);
309
310 x = -2*(*samples)-1;
311 x ^= (x>>31);
312
313 samples++;
314 i++;
315
316 encode_scalar(s, x - sign_modifier, k, s->write_sample_size);
317
318 history += x * s->rc.history_mult
319 - ((history * s->rc.history_mult) >> 9);
320
321 sign_modifier = 0;
322 if(x > 0xFFFF)
323 history = 0xFFFF;
324
325 if((history < 128) && (i < s->avctx->frame_size)) {
326 unsigned int block_size = 0;
327
328 k = 7 - av_log2(history) + ((history + 16) >> 6);
329
330 while((*samples == 0) && (i < s->avctx->frame_size)) {
331 samples++;
332 i++;
333 block_size++;
334 }
335 encode_scalar(s, block_size, k, 16);
336
337 sign_modifier = (block_size <= 0xFFFF);
338
339 history = 0;
340 }
341
342 }
343 }
344
345 static void write_compressed_frame(AlacEncodeContext *s)
346 {
347 int i, j;
348
349 if(s->avctx->channels == 2)
350 alac_stereo_decorrelation(s);
351 put_bits(&s->pbctx, 8, s->interlacing_shift);
352 put_bits(&s->pbctx, 8, s->interlacing_leftweight);
353
354 for(i=0;i<s->avctx->channels;i++) {
355
356 calc_predictor_params(s, i);
357
358 put_bits(&s->pbctx, 4, 0); // prediction type : currently only type 0 has been RE'd
359 put_bits(&s->pbctx, 4, s->lpc[i].lpc_quant);
360
361 put_bits(&s->pbctx, 3, s->rc.rice_modifier);
362 put_bits(&s->pbctx, 5, s->lpc[i].lpc_order);
363 // predictor coeff. table
364 for(j=0;j<s->lpc[i].lpc_order;j++) {
365 put_sbits(&s->pbctx, 16, s->lpc[i].lpc_coeff[j]);
366 }
367 }
368
369 // apply lpc and entropy coding to audio samples
370
371 for(i=0;i<s->avctx->channels;i++) {
372 alac_linear_predictor(s, i);
373 alac_entropy_coder(s);
374 }
375 }
376
377 static av_cold int alac_encode_init(AVCodecContext *avctx)
378 {
379 AlacEncodeContext *s = avctx->priv_data;
380 uint8_t *alac_extradata = av_mallocz(ALAC_EXTRADATA_SIZE+1);
381
382 avctx->frame_size = DEFAULT_FRAME_SIZE;
383 avctx->bits_per_coded_sample = DEFAULT_SAMPLE_SIZE;
384
385 if(avctx->sample_fmt != SAMPLE_FMT_S16) {
386 av_log(avctx, AV_LOG_ERROR, "only pcm_s16 input samples are supported\n");
387 return -1;
388 }
389
390 // Set default compression level
391 if(avctx->compression_level == FF_COMPRESSION_DEFAULT)
392 s->compression_level = 2;
393 else
394 s->compression_level = av_clip(avctx->compression_level, 0, 2);
395
396 // Initialize default Rice parameters
397 s->rc.history_mult = 40;
398 s->rc.initial_history = 10;
399 s->rc.k_modifier = 14;
400 s->rc.rice_modifier = 4;
401
402 s->max_coded_frame_size = 8 + (avctx->frame_size*avctx->channels*avctx->bits_per_coded_sample>>3);
403
404 s->write_sample_size = avctx->bits_per_coded_sample + avctx->channels - 1; // FIXME: consider wasted_bytes
405
406 AV_WB32(alac_extradata, ALAC_EXTRADATA_SIZE);
407 AV_WB32(alac_extradata+4, MKBETAG('a','l','a','c'));
408 AV_WB32(alac_extradata+12, avctx->frame_size);
409 AV_WB8 (alac_extradata+17, avctx->bits_per_coded_sample);
410 AV_WB8 (alac_extradata+21, avctx->channels);
411 AV_WB32(alac_extradata+24, s->max_coded_frame_size);
412 AV_WB32(alac_extradata+28, avctx->sample_rate*avctx->channels*avctx->bits_per_coded_sample); // average bitrate
413 AV_WB32(alac_extradata+32, avctx->sample_rate);
414
415 // Set relevant extradata fields
416 if(s->compression_level > 0) {
417 AV_WB8(alac_extradata+18, s->rc.history_mult);
418 AV_WB8(alac_extradata+19, s->rc.initial_history);
419 AV_WB8(alac_extradata+20, s->rc.k_modifier);
420 }
421
422 s->min_prediction_order = DEFAULT_MIN_PRED_ORDER;
423 if(avctx->min_prediction_order >= 0) {
424 if(avctx->min_prediction_order < MIN_LPC_ORDER ||
425 avctx->min_prediction_order > ALAC_MAX_LPC_ORDER) {
426 av_log(avctx, AV_LOG_ERROR, "invalid min prediction order: %d\n", avctx->min_prediction_order);
427 return -1;
428 }
429
430 s->min_prediction_order = avctx->min_prediction_order;
431 }
432
433 s->max_prediction_order = DEFAULT_MAX_PRED_ORDER;
434 if(avctx->max_prediction_order >= 0) {
435 if(avctx->max_prediction_order < MIN_LPC_ORDER ||
436 avctx->max_prediction_order > ALAC_MAX_LPC_ORDER) {
437 av_log(avctx, AV_LOG_ERROR, "invalid max prediction order: %d\n", avctx->max_prediction_order);
438 return -1;
439 }
440
441 s->max_prediction_order = avctx->max_prediction_order;
442 }
443
444 if(s->max_prediction_order < s->min_prediction_order) {
445 av_log(avctx, AV_LOG_ERROR, "invalid prediction orders: min=%d max=%d\n",
446 s->min_prediction_order, s->max_prediction_order);
447 return -1;
448 }
449
450 avctx->extradata = alac_extradata;
451 avctx->extradata_size = ALAC_EXTRADATA_SIZE;
452
453 avctx->coded_frame = avcodec_alloc_frame();
454 avctx->coded_frame->key_frame = 1;
455
456 s->avctx = avctx;
457 dsputil_init(&s->dspctx, avctx);
458
459 return 0;
460 }
461
462 static int alac_encode_frame(AVCodecContext *avctx, uint8_t *frame,
463 int buf_size, void *data)
464 {
465 AlacEncodeContext *s = avctx->priv_data;
466 PutBitContext *pb = &s->pbctx;
467 int i, out_bytes, verbatim_flag = 0;
468
469 if(avctx->frame_size > DEFAULT_FRAME_SIZE) {
470 av_log(avctx, AV_LOG_ERROR, "input frame size exceeded\n");
471 return -1;
472 }
473
474 if(buf_size < 2*s->max_coded_frame_size) {
475 av_log(avctx, AV_LOG_ERROR, "buffer size is too small\n");
476 return -1;
477 }
478
479 verbatim:
480 init_put_bits(pb, frame, buf_size);
481
482 if((s->compression_level == 0) || verbatim_flag) {
483 // Verbatim mode
484 int16_t *samples = data;
485 write_frame_header(s, 1);
486 for(i=0; i<avctx->frame_size*avctx->channels; i++) {
487 put_sbits(pb, 16, *samples++);
488 }
489 } else {
490 init_sample_buffers(s, data);
491 write_frame_header(s, 0);
492 write_compressed_frame(s);
493 }
494
495 put_bits(pb, 3, 7);
496 flush_put_bits(pb);
497 out_bytes = put_bits_count(pb) >> 3;
498
499 if(out_bytes > s->max_coded_frame_size) {
500 /* frame too large. use verbatim mode */
501 if(verbatim_flag || (s->compression_level == 0)) {
502 /* still too large. must be an error. */
503 av_log(avctx, AV_LOG_ERROR, "error encoding frame\n");
504 return -1;
505 }
506 verbatim_flag = 1;
507 goto verbatim;
508 }
509
510 return out_bytes;
511 }
512
513 static av_cold int alac_encode_close(AVCodecContext *avctx)
514 {
515 av_freep(&avctx->extradata);
516 avctx->extradata_size = 0;
517 av_freep(&avctx->coded_frame);
518 return 0;
519 }
520
521 AVCodec alac_encoder = {
522 "alac",
523 AVMEDIA_TYPE_AUDIO,
524 CODEC_ID_ALAC,
525 sizeof(AlacEncodeContext),
526 alac_encode_init,
527 alac_encode_frame,
528 alac_encode_close,
529 .capabilities = CODEC_CAP_SMALL_LAST_FRAME,
530 .sample_fmts = (const enum SampleFormat[]){ SAMPLE_FMT_S16, SAMPLE_FMT_NONE},
531 .long_name = NULL_IF_CONFIG_SMALL("ALAC (Apple Lossless Audio Codec)"),
532 };