FFmpeg
alacenc.c
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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 "libavutil/opt.h"
23 
24 #include "avcodec.h"
25 #include "put_bits.h"
26 #include "internal.h"
27 #include "lpc.h"
28 #include "mathops.h"
29 #include "alac_data.h"
30 
31 #define DEFAULT_FRAME_SIZE 4096
32 #define ALAC_EXTRADATA_SIZE 36
33 #define ALAC_FRAME_HEADER_SIZE 55
34 #define ALAC_FRAME_FOOTER_SIZE 3
35 
36 #define ALAC_ESCAPE_CODE 0x1FF
37 #define ALAC_MAX_LPC_ORDER 30
38 #define DEFAULT_MAX_PRED_ORDER 6
39 #define DEFAULT_MIN_PRED_ORDER 4
40 #define ALAC_MAX_LPC_PRECISION 9
41 #define ALAC_MIN_LPC_SHIFT 0
42 #define ALAC_MAX_LPC_SHIFT 9
43 
44 #define ALAC_CHMODE_LEFT_RIGHT 0
45 #define ALAC_CHMODE_LEFT_SIDE 1
46 #define ALAC_CHMODE_RIGHT_SIDE 2
47 #define ALAC_CHMODE_MID_SIDE 3
48 
49 typedef struct RiceContext {
54 } RiceContext;
55 
56 typedef struct AlacLPCContext {
57  int lpc_order;
58  int lpc_coeff[ALAC_MAX_LPC_ORDER+1];
59  int lpc_quant;
61 
62 typedef struct AlacEncodeContext {
63  const AVClass *class;
65  int frame_size; /**< current frame size */
66  int verbatim; /**< current frame verbatim mode flag */
73  int32_t sample_buf[2][DEFAULT_FRAME_SIZE];
74  int32_t predictor_buf[2][DEFAULT_FRAME_SIZE];
82 
83 
85  const uint8_t *samples[2])
86 {
87  int ch, i;
90 
91 #define COPY_SAMPLES(type) do { \
92  for (ch = 0; ch < channels; ch++) { \
93  int32_t *bptr = s->sample_buf[ch]; \
94  const type *sptr = (const type *)samples[ch]; \
95  for (i = 0; i < s->frame_size; i++) \
96  bptr[i] = sptr[i] >> shift; \
97  } \
98  } while (0)
99 
102  else
103  COPY_SAMPLES(int16_t);
104 }
105 
106 static void encode_scalar(AlacEncodeContext *s, int x,
107  int k, int write_sample_size)
108 {
109  int divisor, q, r;
110 
111  k = FFMIN(k, s->rc.k_modifier);
112  divisor = (1<<k) - 1;
113  q = x / divisor;
114  r = x % divisor;
115 
116  if (q > 8) {
117  // write escape code and sample value directly
119  put_bits(&s->pbctx, write_sample_size, x);
120  } else {
121  if (q)
122  put_bits(&s->pbctx, q, (1<<q) - 1);
123  put_bits(&s->pbctx, 1, 0);
124 
125  if (k != 1) {
126  if (r > 0)
127  put_bits(&s->pbctx, k, r+1);
128  else
129  put_bits(&s->pbctx, k-1, 0);
130  }
131  }
132 }
133 
135  enum AlacRawDataBlockType element,
136  int instance)
137 {
138  int encode_fs = 0;
139 
141  encode_fs = 1;
142 
143  put_bits(&s->pbctx, 3, element); // element type
144  put_bits(&s->pbctx, 4, instance); // element instance
145  put_bits(&s->pbctx, 12, 0); // unused header bits
146  put_bits(&s->pbctx, 1, encode_fs); // Sample count is in the header
147  put_bits(&s->pbctx, 2, s->extra_bits >> 3); // Extra bytes (for 24-bit)
148  put_bits(&s->pbctx, 1, s->verbatim); // Audio block is verbatim
149  if (encode_fs)
150  put_bits32(&s->pbctx, s->frame_size); // No. of samples in the frame
151 }
152 
154 {
156  int shift[MAX_LPC_ORDER];
157  int opt_order;
158 
159  if (s->compression_level == 1) {
160  s->lpc[ch].lpc_order = 6;
161  s->lpc[ch].lpc_quant = 6;
162  s->lpc[ch].lpc_coeff[0] = 160;
163  s->lpc[ch].lpc_coeff[1] = -190;
164  s->lpc[ch].lpc_coeff[2] = 170;
165  s->lpc[ch].lpc_coeff[3] = -130;
166  s->lpc[ch].lpc_coeff[4] = 80;
167  s->lpc[ch].lpc_coeff[5] = -25;
168  } else {
169  opt_order = ff_lpc_calc_coefs(&s->lpc_ctx, s->sample_buf[ch],
170  s->frame_size,
173  ALAC_MAX_LPC_PRECISION, coefs, shift,
176  ALAC_MAX_LPC_SHIFT, 1);
177 
178  s->lpc[ch].lpc_order = opt_order;
179  s->lpc[ch].lpc_quant = shift[opt_order-1];
180  memcpy(s->lpc[ch].lpc_coeff, coefs[opt_order-1], opt_order*sizeof(int));
181  }
182 }
183 
184 static int estimate_stereo_mode(int32_t *left_ch, int32_t *right_ch, int n)
185 {
186  int i, best;
187  int32_t lt, rt;
188  uint64_t sum[4];
189  uint64_t score[4];
190 
191  /* calculate sum of 2nd order residual for each channel */
192  sum[0] = sum[1] = sum[2] = sum[3] = 0;
193  for (i = 2; i < n; i++) {
194  lt = left_ch[i] - 2 * left_ch[i - 1] + left_ch[i - 2];
195  rt = right_ch[i] - 2 * right_ch[i - 1] + right_ch[i - 2];
196  sum[2] += FFABS((lt + rt) >> 1);
197  sum[3] += FFABS(lt - rt);
198  sum[0] += FFABS(lt);
199  sum[1] += FFABS(rt);
200  }
201 
202  /* calculate score for each mode */
203  score[0] = sum[0] + sum[1];
204  score[1] = sum[0] + sum[3];
205  score[2] = sum[1] + sum[3];
206  score[3] = sum[2] + sum[3];
207 
208  /* return mode with lowest score */
209  best = 0;
210  for (i = 1; i < 4; i++) {
211  if (score[i] < score[best])
212  best = i;
213  }
214  return best;
215 }
216 
218 {
219  int32_t *left = s->sample_buf[0], *right = s->sample_buf[1];
220  int i, mode, n = s->frame_size;
221  int32_t tmp;
222 
223  mode = estimate_stereo_mode(left, right, n);
224 
225  switch (mode) {
227  s->interlacing_leftweight = 0;
228  s->interlacing_shift = 0;
229  break;
231  for (i = 0; i < n; i++)
232  right[i] = left[i] - right[i];
233  s->interlacing_leftweight = 1;
234  s->interlacing_shift = 0;
235  break;
237  for (i = 0; i < n; i++) {
238  tmp = right[i];
239  right[i] = left[i] - right[i];
240  left[i] = tmp + (right[i] >> 31);
241  }
242  s->interlacing_leftweight = 1;
243  s->interlacing_shift = 31;
244  break;
245  default:
246  for (i = 0; i < n; i++) {
247  tmp = left[i];
248  left[i] = (tmp + right[i]) >> 1;
249  right[i] = tmp - right[i];
250  }
251  s->interlacing_leftweight = 1;
252  s->interlacing_shift = 1;
253  break;
254  }
255 }
256 
258 {
259  int i;
260  AlacLPCContext lpc = s->lpc[ch];
262 
263  if (lpc.lpc_order == 31) {
264  residual[0] = s->sample_buf[ch][0];
265 
266  for (i = 1; i < s->frame_size; i++) {
267  residual[i] = s->sample_buf[ch][i ] -
268  s->sample_buf[ch][i - 1];
269  }
270 
271  return;
272  }
273 
274  // generalised linear predictor
275 
276  if (lpc.lpc_order > 0) {
277  int32_t *samples = s->sample_buf[ch];
278 
279  // generate warm-up samples
280  residual[0] = samples[0];
281  for (i = 1; i <= lpc.lpc_order; i++)
282  residual[i] = sign_extend(samples[i] - samples[i-1], s->write_sample_size);
283 
284  // perform lpc on remaining samples
285  for (i = lpc.lpc_order + 1; i < s->frame_size; i++) {
286  int sum = 1 << (lpc.lpc_quant - 1), res_val, j;
287 
288  for (j = 0; j < lpc.lpc_order; j++) {
289  sum += (samples[lpc.lpc_order-j] - samples[0]) *
290  lpc.lpc_coeff[j];
291  }
292 
293  sum >>= lpc.lpc_quant;
294  sum += samples[0];
295  residual[i] = sign_extend(samples[lpc.lpc_order+1] - sum,
296  s->write_sample_size);
297  res_val = residual[i];
298 
299  if (res_val) {
300  int index = lpc.lpc_order - 1;
301  int neg = (res_val < 0);
302 
303  while (index >= 0 && (neg ? (res_val < 0) : (res_val > 0))) {
304  int val = samples[0] - samples[lpc.lpc_order - index];
305  int sign = (val ? FFSIGN(val) : 0);
306 
307  if (neg)
308  sign *= -1;
309 
310  lpc.lpc_coeff[index] -= sign;
311  val *= sign;
312  res_val -= (val >> lpc.lpc_quant) * (lpc.lpc_order - index);
313  index--;
314  }
315  }
316  samples++;
317  }
318  }
319 }
320 
322 {
323  unsigned int history = s->rc.initial_history;
324  int sign_modifier = 0, i, k;
326 
327  for (i = 0; i < s->frame_size;) {
328  int x;
329 
330  k = av_log2((history >> 9) + 3);
331 
332  x = -2 * (*samples) -1;
333  x ^= x >> 31;
334 
335  samples++;
336  i++;
337 
338  encode_scalar(s, x - sign_modifier, k, s->write_sample_size);
339 
340  history += x * s->rc.history_mult -
341  ((history * s->rc.history_mult) >> 9);
342 
343  sign_modifier = 0;
344  if (x > 0xFFFF)
345  history = 0xFFFF;
346 
347  if (history < 128 && i < s->frame_size) {
348  unsigned int block_size = 0;
349 
350  k = 7 - av_log2(history) + ((history + 16) >> 6);
351 
352  while (*samples == 0 && i < s->frame_size) {
353  samples++;
354  i++;
355  block_size++;
356  }
357  encode_scalar(s, block_size, k, 16);
358  sign_modifier = (block_size <= 0xFFFF);
359  history = 0;
360  }
361 
362  }
363 }
364 
366  enum AlacRawDataBlockType element, int instance,
367  const uint8_t *samples0, const uint8_t *samples1)
368 {
369  const uint8_t *samples[2] = { samples0, samples1 };
370  int i, j, channels;
371  int prediction_type = 0;
372  PutBitContext *pb = &s->pbctx;
373 
374  channels = element == TYPE_CPE ? 2 : 1;
375 
376  if (s->verbatim) {
377  write_element_header(s, element, instance);
378  /* samples are channel-interleaved in verbatim mode */
379  if (s->avctx->sample_fmt == AV_SAMPLE_FMT_S32P) {
380  int shift = 32 - s->avctx->bits_per_raw_sample;
381  const int32_t *samples_s32[2] = { (const int32_t *)samples0,
382  (const int32_t *)samples1 };
383  for (i = 0; i < s->frame_size; i++)
384  for (j = 0; j < channels; j++)
386  samples_s32[j][i] >> shift);
387  } else {
388  const int16_t *samples_s16[2] = { (const int16_t *)samples0,
389  (const int16_t *)samples1 };
390  for (i = 0; i < s->frame_size; i++)
391  for (j = 0; j < channels; j++)
393  samples_s16[j][i]);
394  }
395  } else {
397  channels - 1;
398 
399  init_sample_buffers(s, channels, samples);
400  write_element_header(s, element, instance);
401 
402  // extract extra bits if needed
403  if (s->extra_bits) {
404  uint32_t mask = (1 << s->extra_bits) - 1;
405  for (j = 0; j < channels; j++) {
406  int32_t *extra = s->predictor_buf[j];
407  int32_t *smp = s->sample_buf[j];
408  for (i = 0; i < s->frame_size; i++) {
409  extra[i] = smp[i] & mask;
410  smp[i] >>= s->extra_bits;
411  }
412  }
413  }
414 
415  if (channels == 2)
417  else
419  put_bits(pb, 8, s->interlacing_shift);
420  put_bits(pb, 8, s->interlacing_leftweight);
421 
422  for (i = 0; i < channels; i++) {
423  calc_predictor_params(s, i);
424 
425  put_bits(pb, 4, prediction_type);
426  put_bits(pb, 4, s->lpc[i].lpc_quant);
427 
428  put_bits(pb, 3, s->rc.rice_modifier);
429  put_bits(pb, 5, s->lpc[i].lpc_order);
430  // predictor coeff. table
431  for (j = 0; j < s->lpc[i].lpc_order; j++)
432  put_sbits(pb, 16, s->lpc[i].lpc_coeff[j]);
433  }
434 
435  // write extra bits if needed
436  if (s->extra_bits) {
437  for (i = 0; i < s->frame_size; i++) {
438  for (j = 0; j < channels; j++) {
439  put_bits(pb, s->extra_bits, s->predictor_buf[j][i]);
440  }
441  }
442  }
443 
444  // apply lpc and entropy coding to audio samples
445  for (i = 0; i < channels; i++) {
446  alac_linear_predictor(s, i);
447 
448  // TODO: determine when this will actually help. for now it's not used.
449  if (prediction_type == 15) {
450  // 2nd pass 1st order filter
452  for (j = s->frame_size - 1; j > 0; j--)
453  residual[j] -= residual[j - 1];
454  }
455  alac_entropy_coder(s, i);
456  }
457  }
458 }
459 
461  uint8_t * const *samples)
462 {
463  PutBitContext *pb = &s->pbctx;
464  const enum AlacRawDataBlockType *ch_elements = ff_alac_channel_elements[s->avctx->channels - 1];
465  const uint8_t *ch_map = ff_alac_channel_layout_offsets[s->avctx->channels - 1];
466  int ch, element, sce, cpe;
467 
468  init_put_bits(pb, avpkt->data, avpkt->size);
469 
470  ch = element = sce = cpe = 0;
471  while (ch < s->avctx->channels) {
472  if (ch_elements[element] == TYPE_CPE) {
473  write_element(s, TYPE_CPE, cpe, samples[ch_map[ch]],
474  samples[ch_map[ch + 1]]);
475  cpe++;
476  ch += 2;
477  } else {
478  write_element(s, TYPE_SCE, sce, samples[ch_map[ch]], NULL);
479  sce++;
480  ch++;
481  }
482  element++;
483  }
484 
485  put_bits(pb, 3, TYPE_END);
486  flush_put_bits(pb);
487 
488  return put_bits_count(pb) >> 3;
489 }
490 
492 {
493  int header_bits = 23 + 32 * (frame_size < DEFAULT_FRAME_SIZE);
494  return FFALIGN(header_bits + bps * ch * frame_size + 3, 8) / 8;
495 }
496 
498 {
499  AlacEncodeContext *s = avctx->priv_data;
500  ff_lpc_end(&s->lpc_ctx);
501  av_freep(&avctx->extradata);
502  avctx->extradata_size = 0;
503  return 0;
504 }
505 
507 {
508  AlacEncodeContext *s = avctx->priv_data;
509  int ret;
510  uint8_t *alac_extradata;
511 
513 
514  if (avctx->sample_fmt == AV_SAMPLE_FMT_S32P) {
515  if (avctx->bits_per_raw_sample != 24)
516  av_log(avctx, AV_LOG_WARNING, "encoding as 24 bits-per-sample\n");
517  avctx->bits_per_raw_sample = 24;
518  } else {
519  avctx->bits_per_raw_sample = 16;
520  s->extra_bits = 0;
521  }
522 
523  // Set default compression level
525  s->compression_level = 2;
526  else
527  s->compression_level = av_clip(avctx->compression_level, 0, 2);
528 
529  // Initialize default Rice parameters
530  s->rc.history_mult = 40;
531  s->rc.initial_history = 10;
532  s->rc.k_modifier = 14;
533  s->rc.rice_modifier = 4;
534 
536  avctx->channels,
537  avctx->bits_per_raw_sample);
538 
540  if (!avctx->extradata) {
541  ret = AVERROR(ENOMEM);
542  goto error;
543  }
545 
546  alac_extradata = avctx->extradata;
547  AV_WB32(alac_extradata, ALAC_EXTRADATA_SIZE);
548  AV_WB32(alac_extradata+4, MKBETAG('a','l','a','c'));
549  AV_WB32(alac_extradata+12, avctx->frame_size);
550  AV_WB8 (alac_extradata+17, avctx->bits_per_raw_sample);
551  AV_WB8 (alac_extradata+21, avctx->channels);
552  AV_WB32(alac_extradata+24, s->max_coded_frame_size);
553  AV_WB32(alac_extradata+28,
554  avctx->sample_rate * avctx->channels * avctx->bits_per_raw_sample); // average bitrate
555  AV_WB32(alac_extradata+32, avctx->sample_rate);
556 
557  // Set relevant extradata fields
558  if (s->compression_level > 0) {
559  AV_WB8(alac_extradata+18, s->rc.history_mult);
560  AV_WB8(alac_extradata+19, s->rc.initial_history);
561  AV_WB8(alac_extradata+20, s->rc.k_modifier);
562  }
563 
564 #if FF_API_PRIVATE_OPT
566  if (avctx->min_prediction_order >= 0) {
567  if (avctx->min_prediction_order < MIN_LPC_ORDER ||
569  av_log(avctx, AV_LOG_ERROR, "invalid min prediction order: %d\n",
570  avctx->min_prediction_order);
571  ret = AVERROR(EINVAL);
572  goto error;
573  }
574 
576  }
577 
578  if (avctx->max_prediction_order >= 0) {
579  if (avctx->max_prediction_order < MIN_LPC_ORDER ||
581  av_log(avctx, AV_LOG_ERROR, "invalid max prediction order: %d\n",
582  avctx->max_prediction_order);
583  ret = AVERROR(EINVAL);
584  goto error;
585  }
586 
588  }
590 #endif
591 
593  av_log(avctx, AV_LOG_ERROR,
594  "invalid prediction orders: min=%d max=%d\n",
596  ret = AVERROR(EINVAL);
597  goto error;
598  }
599 
600  s->avctx = avctx;
601 
602  if ((ret = ff_lpc_init(&s->lpc_ctx, avctx->frame_size,
604  FF_LPC_TYPE_LEVINSON)) < 0) {
605  goto error;
606  }
607 
608  return 0;
609 error:
610  alac_encode_close(avctx);
611  return ret;
612 }
613 
614 static int alac_encode_frame(AVCodecContext *avctx, AVPacket *avpkt,
615  const AVFrame *frame, int *got_packet_ptr)
616 {
617  AlacEncodeContext *s = avctx->priv_data;
618  int out_bytes, max_frame_size, ret;
619 
620  s->frame_size = frame->nb_samples;
621 
622  if (frame->nb_samples < DEFAULT_FRAME_SIZE)
623  max_frame_size = get_max_frame_size(s->frame_size, avctx->channels,
624  avctx->bits_per_raw_sample);
625  else
626  max_frame_size = s->max_coded_frame_size;
627 
628  if ((ret = ff_alloc_packet2(avctx, avpkt, 4 * max_frame_size, 0)) < 0)
629  return ret;
630 
631  /* use verbatim mode for compression_level 0 */
632  if (s->compression_level) {
633  s->verbatim = 0;
634  s->extra_bits = avctx->bits_per_raw_sample - 16;
635  } else {
636  s->verbatim = 1;
637  s->extra_bits = 0;
638  }
639 
640  out_bytes = write_frame(s, avpkt, frame->extended_data);
641 
642  if (out_bytes > max_frame_size) {
643  /* frame too large. use verbatim mode */
644  s->verbatim = 1;
645  s->extra_bits = 0;
646  out_bytes = write_frame(s, avpkt, frame->extended_data);
647  }
648 
649  avpkt->size = out_bytes;
650  *got_packet_ptr = 1;
651  return 0;
652 }
653 
654 #define OFFSET(x) offsetof(AlacEncodeContext, x)
655 #define AE AV_OPT_FLAG_AUDIO_PARAM | AV_OPT_FLAG_ENCODING_PARAM
656 static const AVOption options[] = {
657  { "min_prediction_order", NULL, OFFSET(min_prediction_order), AV_OPT_TYPE_INT, { .i64 = DEFAULT_MIN_PRED_ORDER }, MIN_LPC_ORDER, ALAC_MAX_LPC_ORDER, AE },
658  { "max_prediction_order", NULL, OFFSET(max_prediction_order), AV_OPT_TYPE_INT, { .i64 = DEFAULT_MAX_PRED_ORDER }, MIN_LPC_ORDER, ALAC_MAX_LPC_ORDER, AE },
659 
660  { NULL },
661 };
662 
663 static const AVClass alacenc_class = {
664  .class_name = "alacenc",
665  .item_name = av_default_item_name,
666  .option = options,
667  .version = LIBAVUTIL_VERSION_INT,
668 };
669 
671  .name = "alac",
672  .long_name = NULL_IF_CONFIG_SMALL("ALAC (Apple Lossless Audio Codec)"),
673  .type = AVMEDIA_TYPE_AUDIO,
674  .id = AV_CODEC_ID_ALAC,
675  .priv_data_size = sizeof(AlacEncodeContext),
676  .priv_class = &alacenc_class,
678  .encode2 = alac_encode_frame,
679  .close = alac_encode_close,
680  .capabilities = AV_CODEC_CAP_SMALL_LAST_FRAME,
682  .sample_fmts = (const enum AVSampleFormat[]){ AV_SAMPLE_FMT_S32P,
685 };
#define DEFAULT_MIN_PRED_ORDER
Definition: alacenc.c:39
static void av_unused put_bits32(PutBitContext *s, uint32_t value)
Write exactly 32 bits into a bitstream.
Definition: put_bits.h:250
#define NULL
Definition: coverity.c:32
const char const char void * val
Definition: avisynth_c.h:863
#define FF_COMPRESSION_DEFAULT
Definition: avcodec.h:1638
static int shift(int a, int b)
Definition: sonic.c:82
This structure describes decoded (raw) audio or video data.
Definition: frame.h:295
AVOption.
Definition: opt.h:246
Definition: lpc.h:52
static void put_sbits(PutBitContext *pb, int n, int32_t value)
Definition: put_bits.h:240
static void put_bits(Jpeg2000EncoderContext *s, int val, int n)
put n times val bit
Definition: j2kenc.c:208
#define AV_LOG_WARNING
Something somehow does not look correct.
Definition: log.h:182
LPCContext lpc_ctx
Definition: alacenc.c:80
#define ALAC_ESCAPE_CODE
Definition: alacenc.c:36
#define LIBAVUTIL_VERSION_INT
Definition: version.h:85
#define MAX_LPC_ORDER
Definition: lpc.h:38
Definition: aac.h:63
static av_cold int init(AVCodecContext *avctx)
Definition: avrndec.c:35
Definition: aac.h:56
Definition: aac.h:57
channels
Definition: aptx.c:30
int size
Definition: avcodec.h:1478
uint8_t pi<< 24) CONV_FUNC(AV_SAMPLE_FMT_S64, int64_t, AV_SAMPLE_FMT_U8,(uint64_t)((*(const uint8_t *) pi-0x80U))<< 56) CONV_FUNC(AV_SAMPLE_FMT_FLT, float, AV_SAMPLE_FMT_U8,(*(const uint8_t *) pi-0x80)*(1.0f/(1<< 7))) CONV_FUNC(AV_SAMPLE_FMT_DBL, double, AV_SAMPLE_FMT_U8,(*(const uint8_t *) pi-0x80)*(1.0/(1<< 7))) CONV_FUNC(AV_SAMPLE_FMT_U8, uint8_t, AV_SAMPLE_FMT_S16,(*(const int16_t *) pi >>8)+0x80) CONV_FUNC(AV_SAMPLE_FMT_S64, int64_t, AV_SAMPLE_FMT_S16,(uint64_t)(*(const int16_t *) pi)<< 48) CONV_FUNC(AV_SAMPLE_FMT_FLT, float, AV_SAMPLE_FMT_S16,*(const int16_t *) pi *(1.0f/(1<< 15))) CONV_FUNC(AV_SAMPLE_FMT_DBL, double, AV_SAMPLE_FMT_S16,*(const int16_t *) pi *(1.0/(1<< 15))) CONV_FUNC(AV_SAMPLE_FMT_U8, uint8_t, AV_SAMPLE_FMT_S32,(*(const int32_t *) pi >>24)+0x80) CONV_FUNC(AV_SAMPLE_FMT_S64, int64_t, AV_SAMPLE_FMT_S32,(uint64_t)(*(const int32_t *) pi)<< 32) CONV_FUNC(AV_SAMPLE_FMT_FLT, float, AV_SAMPLE_FMT_S32,*(const int32_t *) pi *(1.0f/(1U<< 31))) CONV_FUNC(AV_SAMPLE_FMT_DBL, double, AV_SAMPLE_FMT_S32,*(const int32_t *) pi *(1.0/(1U<< 31))) CONV_FUNC(AV_SAMPLE_FMT_U8, uint8_t, AV_SAMPLE_FMT_S64,(*(const int64_t *) pi >>56)+0x80) CONV_FUNC(AV_SAMPLE_FMT_FLT, float, AV_SAMPLE_FMT_S64,*(const int64_t *) pi *(1.0f/(INT64_C(1)<< 63))) CONV_FUNC(AV_SAMPLE_FMT_DBL, double, AV_SAMPLE_FMT_S64,*(const int64_t *) pi *(1.0/(INT64_C(1)<< 63))) CONV_FUNC(AV_SAMPLE_FMT_U8, uint8_t, AV_SAMPLE_FMT_FLT, av_clip_uint8(lrintf(*(const float *) pi *(1<< 7))+0x80)) CONV_FUNC(AV_SAMPLE_FMT_S16, int16_t, AV_SAMPLE_FMT_FLT, av_clip_int16(lrintf(*(const float *) pi *(1<< 15)))) CONV_FUNC(AV_SAMPLE_FMT_S32, int32_t, AV_SAMPLE_FMT_FLT, av_clipl_int32(llrintf(*(const float *) pi *(1U<< 31)))) CONV_FUNC(AV_SAMPLE_FMT_S64, int64_t, AV_SAMPLE_FMT_FLT, llrintf(*(const float *) pi *(INT64_C(1)<< 63))) CONV_FUNC(AV_SAMPLE_FMT_U8, uint8_t, AV_SAMPLE_FMT_DBL, av_clip_uint8(lrint(*(const double *) pi *(1<< 7))+0x80)) CONV_FUNC(AV_SAMPLE_FMT_S16, int16_t, AV_SAMPLE_FMT_DBL, av_clip_int16(lrint(*(const double *) pi *(1<< 15)))) CONV_FUNC(AV_SAMPLE_FMT_S32, int32_t, AV_SAMPLE_FMT_DBL, av_clipl_int32(llrint(*(const double *) pi *(1U<< 31)))) CONV_FUNC(AV_SAMPLE_FMT_S64, int64_t, AV_SAMPLE_FMT_DBL, llrint(*(const double *) pi *(INT64_C(1)<< 63)))#define FMT_PAIR_FUNC(out, in) static conv_func_type *const fmt_pair_to_conv_functions[AV_SAMPLE_FMT_NB *AV_SAMPLE_FMT_NB]={FMT_PAIR_FUNC(AV_SAMPLE_FMT_U8, AV_SAMPLE_FMT_U8), FMT_PAIR_FUNC(AV_SAMPLE_FMT_S16, AV_SAMPLE_FMT_U8), FMT_PAIR_FUNC(AV_SAMPLE_FMT_S32, AV_SAMPLE_FMT_U8), FMT_PAIR_FUNC(AV_SAMPLE_FMT_FLT, AV_SAMPLE_FMT_U8), FMT_PAIR_FUNC(AV_SAMPLE_FMT_DBL, AV_SAMPLE_FMT_U8), FMT_PAIR_FUNC(AV_SAMPLE_FMT_S64, AV_SAMPLE_FMT_U8), FMT_PAIR_FUNC(AV_SAMPLE_FMT_U8, AV_SAMPLE_FMT_S16), FMT_PAIR_FUNC(AV_SAMPLE_FMT_S16, AV_SAMPLE_FMT_S16), FMT_PAIR_FUNC(AV_SAMPLE_FMT_S32, AV_SAMPLE_FMT_S16), FMT_PAIR_FUNC(AV_SAMPLE_FMT_FLT, AV_SAMPLE_FMT_S16), FMT_PAIR_FUNC(AV_SAMPLE_FMT_DBL, AV_SAMPLE_FMT_S16), FMT_PAIR_FUNC(AV_SAMPLE_FMT_S64, AV_SAMPLE_FMT_S16), FMT_PAIR_FUNC(AV_SAMPLE_FMT_U8, AV_SAMPLE_FMT_S32), FMT_PAIR_FUNC(AV_SAMPLE_FMT_S16, AV_SAMPLE_FMT_S32), FMT_PAIR_FUNC(AV_SAMPLE_FMT_S32, AV_SAMPLE_FMT_S32), FMT_PAIR_FUNC(AV_SAMPLE_FMT_FLT, AV_SAMPLE_FMT_S32), FMT_PAIR_FUNC(AV_SAMPLE_FMT_DBL, AV_SAMPLE_FMT_S32), FMT_PAIR_FUNC(AV_SAMPLE_FMT_S64, AV_SAMPLE_FMT_S32), FMT_PAIR_FUNC(AV_SAMPLE_FMT_U8, AV_SAMPLE_FMT_FLT), FMT_PAIR_FUNC(AV_SAMPLE_FMT_S16, AV_SAMPLE_FMT_FLT), FMT_PAIR_FUNC(AV_SAMPLE_FMT_S32, AV_SAMPLE_FMT_FLT), FMT_PAIR_FUNC(AV_SAMPLE_FMT_FLT, AV_SAMPLE_FMT_FLT), FMT_PAIR_FUNC(AV_SAMPLE_FMT_DBL, AV_SAMPLE_FMT_FLT), FMT_PAIR_FUNC(AV_SAMPLE_FMT_S64, AV_SAMPLE_FMT_FLT), FMT_PAIR_FUNC(AV_SAMPLE_FMT_U8, AV_SAMPLE_FMT_DBL), FMT_PAIR_FUNC(AV_SAMPLE_FMT_S16, AV_SAMPLE_FMT_DBL), FMT_PAIR_FUNC(AV_SAMPLE_FMT_S32, AV_SAMPLE_FMT_DBL), FMT_PAIR_FUNC(AV_SAMPLE_FMT_FLT, AV_SAMPLE_FMT_DBL), FMT_PAIR_FUNC(AV_SAMPLE_FMT_DBL, AV_SAMPLE_FMT_DBL), FMT_PAIR_FUNC(AV_SAMPLE_FMT_S64, AV_SAMPLE_FMT_DBL), FMT_PAIR_FUNC(AV_SAMPLE_FMT_U8, AV_SAMPLE_FMT_S64), FMT_PAIR_FUNC(AV_SAMPLE_FMT_S16, AV_SAMPLE_FMT_S64), FMT_PAIR_FUNC(AV_SAMPLE_FMT_S32, AV_SAMPLE_FMT_S64), FMT_PAIR_FUNC(AV_SAMPLE_FMT_FLT, AV_SAMPLE_FMT_S64), FMT_PAIR_FUNC(AV_SAMPLE_FMT_DBL, AV_SAMPLE_FMT_S64), FMT_PAIR_FUNC(AV_SAMPLE_FMT_S64, AV_SAMPLE_FMT_S64),};static void cpy1(uint8_t **dst, const uint8_t **src, int len){memcpy(*dst,*src, len);}static void cpy2(uint8_t **dst, const uint8_t **src, int len){memcpy(*dst,*src, 2 *len);}static void cpy4(uint8_t **dst, const uint8_t **src, int len){memcpy(*dst,*src, 4 *len);}static void cpy8(uint8_t **dst, const uint8_t **src, int len){memcpy(*dst,*src, 8 *len);}AudioConvert *swri_audio_convert_alloc(enum AVSampleFormat out_fmt, enum AVSampleFormat in_fmt, int channels, const int *ch_map, int flags){AudioConvert *ctx;conv_func_type *f=fmt_pair_to_conv_functions[av_get_packed_sample_fmt(out_fmt)+AV_SAMPLE_FMT_NB *av_get_packed_sample_fmt(in_fmt)];if(!f) return NULL;ctx=av_mallocz(sizeof(*ctx));if(!ctx) return NULL;if(channels==1){in_fmt=av_get_planar_sample_fmt(in_fmt);out_fmt=av_get_planar_sample_fmt(out_fmt);}ctx->channels=channels;ctx->conv_f=f;ctx->ch_map=ch_map;if(in_fmt==AV_SAMPLE_FMT_U8||in_fmt==AV_SAMPLE_FMT_U8P) memset(ctx->silence, 0x80, sizeof(ctx->silence));if(out_fmt==in_fmt &&!ch_map){switch(av_get_bytes_per_sample(in_fmt)){case 1:ctx->simd_f=cpy1;break;case 2:ctx->simd_f=cpy2;break;case 4:ctx->simd_f=cpy4;break;case 8:ctx->simd_f=cpy8;break;}}if(HAVE_X86ASM &&1) swri_audio_convert_init_x86(ctx, out_fmt, in_fmt, channels);if(ARCH_ARM) swri_audio_convert_init_arm(ctx, out_fmt, in_fmt, channels);if(ARCH_AARCH64) swri_audio_convert_init_aarch64(ctx, out_fmt, in_fmt, channels);return ctx;}void swri_audio_convert_free(AudioConvert **ctx){av_freep(ctx);}int swri_audio_convert(AudioConvert *ctx, AudioData *out, AudioData *in, int len){int ch;int off=0;const int os=(out->planar?1:out->ch_count)*out->bps;unsigned misaligned=0;av_assert0(ctx->channels==out->ch_count);if(ctx->in_simd_align_mask){int planes=in->planar?in->ch_count:1;unsigned m=0;for(ch=0;ch< planes;ch++) m|=(intptr_t) in->ch[ch];misaligned|=m &ctx->in_simd_align_mask;}if(ctx->out_simd_align_mask){int planes=out->planar?out->ch_count:1;unsigned m=0;for(ch=0;ch< planes;ch++) m|=(intptr_t) out->ch[ch];misaligned|=m &ctx->out_simd_align_mask;}if(ctx->simd_f &&!ctx->ch_map &&!misaligned){off=len &~15;av_assert1(off >=0);av_assert1(off<=len);av_assert2(ctx->channels==SWR_CH_MAX||!in->ch[ctx->channels]);if(off >0){if(out->planar==in->planar){int planes=out->planar?out->ch_count:1;for(ch=0;ch< planes;ch++){ctx->simd_f(out-> ch ch
Definition: audioconvert.c:56
#define AV_WB8(p, d)
Definition: intreadwrite.h:396
const char * av_default_item_name(void *ptr)
Return the context name.
Definition: log.c:191
int av_log2(unsigned v)
Definition: intmath.c:26
#define COPY_SAMPLES(type)
PutBitContext pbctx
Definition: alacenc.c:77
void * av_mallocz(size_t size)
Allocate a memory block with alignment suitable for all memory accesses (including vectors if availab...
Definition: mem.c:236
static void write_element_header(AlacEncodeContext *s, enum AlacRawDataBlockType element, int instance)
Definition: alacenc.c:134
int bits_per_raw_sample
Bits per sample/pixel of internal libavcodec pixel/sample format.
Definition: avcodec.h:2796
int history_mult
Definition: alacenc.c:50
static av_cold int alac_encode_init(AVCodecContext *avctx)
Definition: alacenc.c:506
#define DEFAULT_MAX_PRED_ORDER
Definition: alacenc.c:38
AVCodec.
Definition: avcodec.h:3481
int initial_history
Definition: alacenc.c:51
static void write_element(AlacEncodeContext *s, enum AlacRawDataBlockType element, int instance, const uint8_t *samples0, const uint8_t *samples1)
Definition: alacenc.c:365
const char * class_name
The name of the class; usually it is the same name as the context structure type to which the AVClass...
Definition: log.h:72
int ff_alloc_packet2(AVCodecContext *avctx, AVPacket *avpkt, int64_t size, int64_t min_size)
Check AVPacket size and/or allocate data.
Definition: encode.c:32
int lpc_quant
Definition: alacenc.c:59
static const AVClass alacenc_class
Definition: alacenc.c:663
enum AVSampleFormat sample_fmt
audio sample format
Definition: avcodec.h:2233
uint8_t
#define av_cold
Definition: attributes.h:82
static void alac_linear_predictor(AlacEncodeContext *s, int ch)
Definition: alacenc.c:257
AVOptions.
uint8_t * extradata
some codecs need / can use extradata like Huffman tables.
Definition: avcodec.h:1666
int max_coded_frame_size
Definition: alacenc.c:70
int interlacing_leftweight
Definition: alacenc.c:76
int interlacing_shift
Definition: alacenc.c:75
AVCodecContext * avctx
Definition: alacenc.c:64
uint8_t * data
Definition: avcodec.h:1477
AVCodec ff_alac_encoder
Definition: alacenc.c:670
#define FFALIGN(x, a)
Definition: macros.h:48
#define av_log(a,...)
#define i(width, name, range_min, range_max)
Definition: cbs_h2645.c:259
#define AV_LOG_ERROR
Something went wrong and cannot losslessly be recovered.
Definition: log.h:176
static void calc_predictor_params(AlacEncodeContext *s, int ch)
Definition: alacenc.c:153
static const uint16_t mask[17]
Definition: lzw.c:38
static int alac_encode_frame(AVCodecContext *avctx, AVPacket *avpkt, const AVFrame *frame, int *got_packet_ptr)
Definition: alacenc.c:614
#define ALAC_MAX_LPC_PRECISION
Definition: alacenc.c:40
#define NULL_IF_CONFIG_SMALL(x)
Return NULL if CONFIG_SMALL is true, otherwise the argument without modification. ...
Definition: internal.h:186
const char * r
Definition: vf_curves.c:114
AlacRawDataBlockType
Definition: alac_data.h:26
const char * name
Name of the codec implementation.
Definition: avcodec.h:3488
#define AE
Definition: alacenc.c:655
static void alac_entropy_coder(AlacEncodeContext *s, int ch)
Definition: alacenc.c:321
int frame_size
current frame size
Definition: alacenc.c:65
static int put_bits_count(PutBitContext *s)
Definition: put_bits.h:85
uint64_t residual
Definition: dirac_vlc.h:29
av_cold void ff_lpc_end(LPCContext *s)
Uninitialize LPCContext.
Definition: lpc.c:322
#define AV_CODEC_CAP_SMALL_LAST_FRAME
Codec can be fed a final frame with a smaller size.
Definition: avcodec.h:1011
#define FFMIN(a, b)
Definition: common.h:96
signed 32 bits, planar
Definition: samplefmt.h:68
int verbatim
current frame verbatim mode flag
Definition: alacenc.c:66
#define FFSIGN(a)
Definition: common.h:73
int32_t
int k_modifier
Definition: alacenc.c:52
these buffered frames must be flushed immediately if a new input produces new the filter must not call request_frame to get more It must just process the frame or queue it The task of requesting more frames is left to the filter s request_frame method or the application If a filter has several the filter must be ready for frames arriving randomly on any input any filter with several inputs will most likely require some kind of queuing mechanism It is perfectly acceptable to have a limited queue and to drop frames when the inputs are too unbalanced request_frame For filters that do not use the this method is called when a frame is wanted on an output For a it should directly call filter_frame on the corresponding output For a if there are queued frames already one of these frames should be pushed If the filter should request a frame on one of its repeatedly until at least one frame has been pushed Return or at least make progress towards producing a frame
#define FFABS(a)
Absolute value, Note, INT_MIN / INT64_MIN result in undefined behavior as they are not representable ...
Definition: common.h:72
#define s(width, name)
Definition: cbs_vp9.c:257
static av_always_inline int get_max_frame_size(int frame_size, int ch, int bps)
Definition: alacenc.c:491
int n
Definition: avisynth_c.h:760
const uint64_t ff_alac_channel_layouts[ALAC_MAX_CHANNELS+1]
Definition: alac_data.c:35
int max_prediction_order
Definition: alacenc.c:69
static void error(const char *err)
static const AVOption options[]
Definition: alacenc.c:656
#define ALAC_CHMODE_LEFT_SIDE
Definition: alacenc.c:45
#define ALAC_MAX_LPC_ORDER
Definition: alacenc.c:37
AlacLPCContext lpc[2]
Definition: alacenc.c:79
attribute_deprecated int max_prediction_order
Definition: avcodec.h:2523
#define ALAC_MIN_LPC_SHIFT
Definition: alacenc.c:41
int frame_size
Number of samples per channel in an audio frame.
Definition: avcodec.h:2245
int frame_size
Definition: mxfenc.c:2215
int32_t sample_buf[2][DEFAULT_FRAME_SIZE]
Definition: alacenc.c:73
Libavcodec external API header.
AVSampleFormat
Audio sample formats.
Definition: samplefmt.h:58
int compression_level
Definition: avcodec.h:1637
int sample_rate
samples per second
Definition: avcodec.h:2225
#define MIN_LPC_ORDER
Definition: lpc.h:37
#define ALAC_MAX_LPC_SHIFT
Definition: alacenc.c:42
main external API structure.
Definition: avcodec.h:1565
int compression_level
Definition: alacenc.c:67
Levinson-Durbin recursion.
Definition: lpc.h:47
Tag MUST be and< 10hcoeff half pel interpolation filter coefficients, hcoeff[0] are the 2 middle coefficients[1] are the next outer ones and so on, resulting in a filter like:...eff[2], hcoeff[1], hcoeff[0], hcoeff[0], hcoeff[1], hcoeff[2]...the sign of the coefficients is not explicitly stored but alternates after each coeff and coeff[0] is positive, so...,+,-,+,-,+,+,-,+,-,+,...hcoeff[0] is not explicitly stored but found by subtracting the sum of all stored coefficients with signs from 32 hcoeff[0]=32-hcoeff[1]-hcoeff[2]-...a good choice for hcoeff and htaps is htaps=6 hcoeff={40,-10, 2}an alternative which requires more computations at both encoder and decoder side and may or may not be better is htaps=8 hcoeff={42,-14, 6,-2}ref_frames minimum of the number of available reference frames and max_ref_frames for example the first frame after a key frame always has ref_frames=1spatial_decomposition_type wavelet type 0 is a 9/7 symmetric compact integer wavelet 1 is a 5/3 symmetric compact integer wavelet others are reserved stored as delta from last, last is reset to 0 if always_reset||keyframeqlog quality(logarithmic quantizer scale) stored as delta from last, last is reset to 0 if always_reset||keyframemv_scale stored as delta from last, last is reset to 0 if always_reset||keyframe FIXME check that everything works fine if this changes between framesqbias dequantization bias stored as delta from last, last is reset to 0 if always_reset||keyframeblock_max_depth maximum depth of the block tree stored as delta from last, last is reset to 0 if always_reset||keyframequant_table quantization tableHighlevel bitstream structure:==============================--------------------------------------------|Header|--------------------------------------------|------------------------------------|||Block0||||split?||||yes no||||.........intra?||||:Block01:yes no||||:Block02:.................||||:Block03::y DC::ref index:||||:Block04::cb DC::motion x:||||.........:cr DC::motion y:||||.................|||------------------------------------||------------------------------------|||Block1|||...|--------------------------------------------|------------------------------------|||Y subbands||Cb subbands||Cr subbands||||------||------||------|||||LL0||HL0||||LL0||HL0||||LL0||HL0|||||------||------||------||||------||------||------|||||LH0||HH0||||LH0||HH0||||LH0||HH0|||||------||------||------||||------||------||------|||||HL1||LH1||||HL1||LH1||||HL1||LH1|||||------||------||------||||------||------||------|||||HH1||HL2||||HH1||HL2||||HH1||HL2|||||...||...||...|||------------------------------------|--------------------------------------------Decoding process:=================------------|||Subbands|------------||||------------|Intra DC||||LL0 subband prediction------------|\Dequantization-------------------\||Reference frames|\IDWT|--------------|Motion\|||Frame 0||Frame 1||Compensation.OBMC v-------|--------------|--------------.\------> Frame n output Frame Frame<----------------------------------/|...|-------------------Range Coder:============Binary Range Coder:-------------------The implemented range coder is an adapted version based upon"Range encoding: an algorithm for removing redundancy from a digitised message."by G.N.N.Martin.The symbols encoded by the Snow range coder are bits(0|1).The associated probabilities are not fix but change depending on the symbol mix seen so far.bit seen|new state---------+-----------------------------------------------0|256-state_transition_table[256-old_state];1|state_transition_table[old_state];state_transition_table={0, 0, 0, 0, 0, 0, 0, 0, 20, 21, 22, 23, 24, 25, 26, 27, 28, 29, 30, 31, 32, 33, 34, 35, 36, 37, 37, 38, 39, 40, 41, 42, 43, 44, 45, 46, 47, 48, 49, 50, 51, 52, 53, 54, 55, 56, 56, 57, 58, 59, 60, 61, 62, 63, 64, 65, 66, 67, 68, 69, 70, 71, 72, 73, 74, 75, 75, 76, 77, 78, 79, 80, 81, 82, 83, 84, 85, 86, 87, 88, 89, 90, 91, 92, 93, 94, 94, 95, 96, 97, 98, 99, 100, 101, 102, 103, 104, 105, 106, 107, 108, 109, 110, 111, 112, 113, 114, 114, 115, 116, 117, 118, 119, 120, 121, 122, 123, 124, 125, 126, 127, 128, 129, 130, 131, 132, 133, 133, 134, 135, 136, 137, 138, 139, 140, 141, 142, 143, 144, 145, 146, 147, 148, 149, 150, 151, 152, 152, 153, 154, 155, 156, 157, 158, 159, 160, 161, 162, 163, 164, 165, 166, 167, 168, 169, 170, 171, 171, 172, 173, 174, 175, 176, 177, 178, 179, 180, 181, 182, 183, 184, 185, 186, 187, 188, 189, 190, 190, 191, 192, 194, 194, 195, 196, 197, 198, 199, 200, 201, 202, 202, 204, 205, 206, 207, 208, 209, 209, 210, 211, 212, 213, 215, 215, 216, 217, 218, 219, 220, 220, 222, 223, 224, 225, 226, 227, 227, 229, 229, 230, 231, 232, 234, 234, 235, 236, 237, 238, 239, 240, 241, 242, 243, 244, 245, 246, 247, 248, 248, 0, 0, 0, 0, 0, 0, 0};FIXME Range Coding of integers:-------------------------FIXME Neighboring Blocks:===================left and top are set to the respective blocks unless they are outside of the image in which case they are set to the Null block top-left is set to the top left block unless it is outside of the image in which case it is set to the left block if this block has no larger parent block or it is at the left side of its parent block and the top right block is not outside of the image then the top right block is used for top-right else the top-left block is used Null block y, cb, cr are 128 level, ref, mx and my are 0 Motion Vector Prediction:=========================1.the motion vectors of all the neighboring blocks are scaled to compensate for the difference of reference frames scaled_mv=(mv *(256 *(current_reference+1)/(mv.reference+1))+128)> the median of the scaled left
Definition: snow.txt:206
#define ORDER_METHOD_EST
Definition: lpc.h:30
int extradata_size
Definition: avcodec.h:1667
static const uint16_t channel_layouts[7]
Definition: dca_lbr.c:113
int32_t predictor_buf[2][DEFAULT_FRAME_SIZE]
Definition: alacenc.c:74
Describe the class of an AVClass context structure.
Definition: log.h:67
#define AV_WB32(p, v)
Definition: intreadwrite.h:419
int index
Definition: gxfenc.c:89
int ff_lpc_calc_coefs(LPCContext *s, const int32_t *samples, int blocksize, int min_order, int max_order, int precision, int32_t coefs[][MAX_LPC_ORDER], int *shift, enum FFLPCType lpc_type, int lpc_passes, int omethod, int min_shift, int max_shift, int zero_shift)
Calculate LPC coefficients for multiple orders.
Definition: lpc.c:200
const uint8_t ff_alac_channel_layout_offsets[ALAC_MAX_CHANNELS][ALAC_MAX_CHANNELS]
Definition: alac_data.c:24
av_cold int ff_lpc_init(LPCContext *s, int blocksize, int max_order, enum FFLPCType lpc_type)
Initialize LPCContext.
Definition: lpc.c:300
static int write_frame(AlacEncodeContext *s, AVPacket *avpkt, uint8_t *const *samples)
Definition: alacenc.c:460
static av_const int sign_extend(int val, unsigned bits)
Definition: mathops.h:130
static void init_sample_buffers(AlacEncodeContext *s, int channels, const uint8_t *samples[2])
Definition: alacenc.c:84
enum AlacRawDataBlockType ff_alac_channel_elements[ALAC_MAX_CHANNELS][5]
Definition: alac_data.c:47
int lpc_order
Definition: alacenc.c:57
int av_get_bytes_per_sample(enum AVSampleFormat sample_fmt)
Return number of bytes per sample.
Definition: samplefmt.c:106
static int estimate_stereo_mode(int32_t *left_ch, int32_t *right_ch, int n)
Definition: alacenc.c:184
int rice_modifier
Definition: alacenc.c:53
int min_prediction_order
Definition: alacenc.c:68
#define FF_DISABLE_DEPRECATION_WARNINGS
Definition: internal.h:84
common internal api header.
static void flush_put_bits(PutBitContext *s)
Pad the end of the output stream with zeros.
Definition: put_bits.h:101
#define ALAC_CHMODE_LEFT_RIGHT
Definition: alacenc.c:44
#define ALAC_CHMODE_RIGHT_SIDE
Definition: alacenc.c:46
RiceContext rc
Definition: alacenc.c:78
static av_cold int alac_encode_close(AVCodecContext *avctx)
Definition: alacenc.c:497
static void init_put_bits(PutBitContext *s, uint8_t *buffer, int buffer_size)
Initialize the PutBitContext s.
Definition: put_bits.h:48
unsigned bps
Definition: movenc.c:1497
#define MKBETAG(a, b, c, d)
Definition: common.h:367
#define AV_INPUT_BUFFER_PADDING_SIZE
Required number of additionally allocated bytes at the end of the input bitstream for decoding...
Definition: avcodec.h:790
void * priv_data
Definition: avcodec.h:1592
int lpc_coeff[ALAC_MAX_LPC_ORDER+1]
Definition: alacenc.c:58
#define FF_ENABLE_DEPRECATION_WARNINGS
Definition: internal.h:85
attribute_deprecated int min_prediction_order
Definition: avcodec.h:2519
int channels
number of audio channels
Definition: avcodec.h:2226
static enum AVSampleFormat sample_fmts[]
Definition: adpcmenc.c:701
Filter the word “frame” indicates either a video frame or a group of audio samples
#define av_freep(p)
signed 16 bits, planar
Definition: samplefmt.h:67
#define av_always_inline
Definition: attributes.h:39
#define ALAC_EXTRADATA_SIZE
Definition: alacenc.c:32
Filter the word “frame” indicates either a video frame or a group of audio as stored in an AVFrame structure Format for each input and each output the list of supported formats For video that means pixel format For audio that means channel sample they are references to shared objects When the negotiation mechanism computes the intersection of the formats supported at each end of a all references to both lists are replaced with a reference to the intersection And when a single format is eventually chosen for a link amongst the remaining all references to the list are updated That means that if a filter requires that its input and output have the same format amongst a supported all it has to do is use a reference to the same list of formats query_formats can leave some formats unset and return AVERROR(EAGAIN) to cause the negotiation mechanism toagain later.That can be used by filters with complex requirements to use the format negotiated on one link to set the formats supported on another.Frame references ownership and permissions
uint8_t ** extended_data
pointers to the data planes/channels.
Definition: frame.h:342
#define DEFAULT_FRAME_SIZE
Definition: alacenc.c:31
This structure stores compressed data.
Definition: avcodec.h:1454
static void alac_stereo_decorrelation(AlacEncodeContext *s)
Definition: alacenc.c:217
mode
Use these values in ebur128_init (or&#39;ed).
Definition: ebur128.h:83
int nb_samples
number of audio samples (per channel) described by this frame
Definition: frame.h:361
static void encode_scalar(AlacEncodeContext *s, int x, int k, int write_sample_size)
Definition: alacenc.c:106
#define OFFSET(x)
Definition: alacenc.c:654
int write_sample_size
Definition: alacenc.c:71
static uint8_t tmp[11]
Definition: aes_ctr.c:26
bitstream writer API