FFmpeg
alac.c
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1 /*
2  * ALAC (Apple Lossless Audio Codec) decoder
3  * Copyright (c) 2005 David Hammerton
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 /**
23  * @file
24  * ALAC (Apple Lossless Audio Codec) decoder
25  * @author 2005 David Hammerton
26  * @see http://crazney.net/programs/itunes/alac.html
27  *
28  * Note: This decoder expects a 36-byte QuickTime atom to be
29  * passed through the extradata[_size] fields. This atom is tacked onto
30  * the end of an 'alac' stsd atom and has the following format:
31  *
32  * 32 bits atom size
33  * 32 bits tag ("alac")
34  * 32 bits tag version (0)
35  * 32 bits samples per frame (used when not set explicitly in the frames)
36  * 8 bits compatible version (0)
37  * 8 bits sample size
38  * 8 bits history mult (40)
39  * 8 bits initial history (10)
40  * 8 bits rice param limit (14)
41  * 8 bits channels
42  * 16 bits maxRun (255)
43  * 32 bits max coded frame size (0 means unknown)
44  * 32 bits average bitrate (0 means unknown)
45  * 32 bits samplerate
46  */
47 
48 #include <inttypes.h>
49 
51 #include "libavutil/opt.h"
52 #include "avcodec.h"
53 #include "get_bits.h"
54 #include "bytestream.h"
55 #include "internal.h"
56 #include "thread.h"
57 #include "unary.h"
58 #include "mathops.h"
59 #include "alac_data.h"
60 #include "alacdsp.h"
61 
62 #define ALAC_EXTRADATA_SIZE 36
63 
64 typedef struct ALACContext {
65  AVClass *class;
68  int channels;
69 
73 
80 
81  int extra_bits; /**< number of extra bits beyond 16-bit */
82  int nb_samples; /**< number of samples in the current frame */
83 
86 
88 } ALACContext;
89 
90 static inline unsigned int decode_scalar(GetBitContext *gb, int k, int bps)
91 {
92  unsigned int x = get_unary_0_9(gb);
93 
94  if (x > 8) { /* RICE THRESHOLD */
95  /* use alternative encoding */
96  x = get_bits_long(gb, bps);
97  } else if (k != 1) {
98  int extrabits = show_bits(gb, k);
99 
100  /* multiply x by 2^k - 1, as part of their strange algorithm */
101  x = (x << k) - x;
102 
103  if (extrabits > 1) {
104  x += extrabits - 1;
105  skip_bits(gb, k);
106  } else
107  skip_bits(gb, k - 1);
108  }
109  return x;
110 }
111 
112 static int rice_decompress(ALACContext *alac, int32_t *output_buffer,
113  int nb_samples, int bps, int rice_history_mult)
114 {
115  int i;
116  unsigned int history = alac->rice_initial_history;
117  int sign_modifier = 0;
118 
119  for (i = 0; i < nb_samples; i++) {
120  int k;
121  unsigned int x;
122 
123  if(get_bits_left(&alac->gb) <= 0)
124  return AVERROR_INVALIDDATA;
125 
126  /* calculate rice param and decode next value */
127  k = av_log2((history >> 9) + 3);
128  k = FFMIN(k, alac->rice_limit);
129  x = decode_scalar(&alac->gb, k, bps);
130  x += sign_modifier;
131  sign_modifier = 0;
132  output_buffer[i] = (x >> 1) ^ -(x & 1);
133 
134  /* update the history */
135  if (x > 0xffff)
136  history = 0xffff;
137  else
138  history += x * rice_history_mult -
139  ((history * rice_history_mult) >> 9);
140 
141  /* special case: there may be compressed blocks of 0 */
142  if ((history < 128) && (i + 1 < nb_samples)) {
143  int block_size;
144 
145  /* calculate rice param and decode block size */
146  k = 7 - av_log2(history) + ((history + 16) >> 6);
147  k = FFMIN(k, alac->rice_limit);
148  block_size = decode_scalar(&alac->gb, k, 16);
149 
150  if (block_size > 0) {
151  if (block_size >= nb_samples - i) {
152  av_log(alac->avctx, AV_LOG_ERROR,
153  "invalid zero block size of %d %d %d\n", block_size,
154  nb_samples, i);
155  block_size = nb_samples - i - 1;
156  }
157  memset(&output_buffer[i + 1], 0,
158  block_size * sizeof(*output_buffer));
159  i += block_size;
160  }
161  if (block_size <= 0xffff)
162  sign_modifier = 1;
163  history = 0;
164  }
165  }
166  return 0;
167 }
168 
169 static inline int sign_only(int v)
170 {
171  return v ? FFSIGN(v) : 0;
172 }
173 
174 static void lpc_prediction(int32_t *error_buffer, uint32_t *buffer_out,
175  int nb_samples, int bps, int16_t *lpc_coefs,
176  int lpc_order, int lpc_quant)
177 {
178  int i;
179  uint32_t *pred = buffer_out;
180 
181  /* first sample always copies */
182  *buffer_out = *error_buffer;
183 
184  if (nb_samples <= 1)
185  return;
186 
187  if (!lpc_order) {
188  memcpy(&buffer_out[1], &error_buffer[1],
189  (nb_samples - 1) * sizeof(*buffer_out));
190  return;
191  }
192 
193  if (lpc_order == 31) {
194  /* simple 1st-order prediction */
195  for (i = 1; i < nb_samples; i++) {
196  buffer_out[i] = sign_extend(buffer_out[i - 1] + error_buffer[i],
197  bps);
198  }
199  return;
200  }
201 
202  /* read warm-up samples */
203  for (i = 1; i <= lpc_order && i < nb_samples; i++)
204  buffer_out[i] = sign_extend(buffer_out[i - 1] + error_buffer[i], bps);
205 
206  /* NOTE: 4 and 8 are very common cases that could be optimized. */
207 
208  for (; i < nb_samples; i++) {
209  int j;
210  int val = 0;
211  unsigned error_val = error_buffer[i];
212  int error_sign;
213  int d = *pred++;
214 
215  /* LPC prediction */
216  for (j = 0; j < lpc_order; j++)
217  val += (pred[j] - d) * lpc_coefs[j];
218  val = (val + (1LL << (lpc_quant - 1))) >> lpc_quant;
219  val += d + error_val;
220  buffer_out[i] = sign_extend(val, bps);
221 
222  /* adapt LPC coefficients */
223  error_sign = sign_only(error_val);
224  if (error_sign) {
225  for (j = 0; j < lpc_order && (int)(error_val * error_sign) > 0; j++) {
226  int sign;
227  val = d - pred[j];
228  sign = sign_only(val) * error_sign;
229  lpc_coefs[j] -= sign;
230  val *= (unsigned)sign;
231  error_val -= (val >> lpc_quant) * (j + 1U);
232  }
233  }
234  }
235 }
236 
237 static int decode_element(AVCodecContext *avctx, AVFrame *frame, int ch_index,
238  int channels)
239 {
240  ALACContext *alac = avctx->priv_data;
241  int has_size, bps, is_compressed, decorr_shift, decorr_left_weight, ret;
242  uint32_t output_samples;
243  int i, ch;
244 
245  skip_bits(&alac->gb, 4); /* element instance tag */
246  skip_bits(&alac->gb, 12); /* unused header bits */
247 
248  /* the number of output samples is stored in the frame */
249  has_size = get_bits1(&alac->gb);
250 
251  alac->extra_bits = get_bits(&alac->gb, 2) << 3;
252  bps = alac->sample_size - alac->extra_bits + channels - 1;
253  if (bps > 32) {
254  avpriv_report_missing_feature(avctx, "bps %d", bps);
255  return AVERROR_PATCHWELCOME;
256  }
257  if (bps < 1)
258  return AVERROR_INVALIDDATA;
259 
260  /* whether the frame is compressed */
261  is_compressed = !get_bits1(&alac->gb);
262 
263  if (has_size)
264  output_samples = get_bits_long(&alac->gb, 32);
265  else
266  output_samples = alac->max_samples_per_frame;
267  if (!output_samples || output_samples > alac->max_samples_per_frame) {
268  av_log(avctx, AV_LOG_ERROR, "invalid samples per frame: %"PRIu32"\n",
269  output_samples);
270  return AVERROR_INVALIDDATA;
271  }
272  if (!alac->nb_samples) {
273  ThreadFrame tframe = { .f = frame };
274  /* get output buffer */
275  frame->nb_samples = output_samples;
276  if ((ret = ff_thread_get_buffer(avctx, &tframe, 0)) < 0)
277  return ret;
278  } else if (output_samples != alac->nb_samples) {
279  av_log(avctx, AV_LOG_ERROR, "sample count mismatch: %"PRIu32" != %d\n",
280  output_samples, alac->nb_samples);
281  return AVERROR_INVALIDDATA;
282  }
283  alac->nb_samples = output_samples;
284  if (alac->direct_output) {
285  for (ch = 0; ch < channels; ch++)
286  alac->output_samples_buffer[ch] = (int32_t *)frame->extended_data[ch_index + ch];
287  }
288 
289  if (is_compressed) {
290  int16_t lpc_coefs[2][32];
291  int lpc_order[2];
292  int prediction_type[2];
293  int lpc_quant[2];
294  int rice_history_mult[2];
295 
296  if (!alac->rice_limit) {
298  "Compression with rice limit 0");
299  return AVERROR(ENOSYS);
300  }
301 
302  decorr_shift = get_bits(&alac->gb, 8);
303  decorr_left_weight = get_bits(&alac->gb, 8);
304 
305  for (ch = 0; ch < channels; ch++) {
306  prediction_type[ch] = get_bits(&alac->gb, 4);
307  lpc_quant[ch] = get_bits(&alac->gb, 4);
308  rice_history_mult[ch] = get_bits(&alac->gb, 3);
309  lpc_order[ch] = get_bits(&alac->gb, 5);
310 
311  if (lpc_order[ch] >= alac->max_samples_per_frame || !lpc_quant[ch])
312  return AVERROR_INVALIDDATA;
313 
314  /* read the predictor table */
315  for (i = lpc_order[ch] - 1; i >= 0; i--)
316  lpc_coefs[ch][i] = get_sbits(&alac->gb, 16);
317  }
318 
319  if (alac->extra_bits) {
320  for (i = 0; i < alac->nb_samples; i++) {
321  if(get_bits_left(&alac->gb) <= 0)
322  return AVERROR_INVALIDDATA;
323  for (ch = 0; ch < channels; ch++)
324  alac->extra_bits_buffer[ch][i] = get_bits(&alac->gb, alac->extra_bits);
325  }
326  }
327  for (ch = 0; ch < channels; ch++) {
328  int ret=rice_decompress(alac, alac->predict_error_buffer[ch],
329  alac->nb_samples, bps,
330  rice_history_mult[ch] * alac->rice_history_mult / 4);
331  if(ret<0)
332  return ret;
333 
334  /* adaptive FIR filter */
335  if (prediction_type[ch] == 15) {
336  /* Prediction type 15 runs the adaptive FIR twice.
337  * The first pass uses the special-case coef_num = 31, while
338  * the second pass uses the coefs from the bitstream.
339  *
340  * However, this prediction type is not currently used by the
341  * reference encoder.
342  */
344  alac->predict_error_buffer[ch],
345  alac->nb_samples, bps, NULL, 31, 0);
346  } else if (prediction_type[ch] > 0) {
347  av_log(avctx, AV_LOG_WARNING, "unknown prediction type: %i\n",
348  prediction_type[ch]);
349  }
351  alac->output_samples_buffer[ch], alac->nb_samples,
352  bps, lpc_coefs[ch], lpc_order[ch], lpc_quant[ch]);
353  }
354  } else {
355  /* not compressed, easy case */
356  for (i = 0; i < alac->nb_samples; i++) {
357  if(get_bits_left(&alac->gb) <= 0)
358  return AVERROR_INVALIDDATA;
359  for (ch = 0; ch < channels; ch++) {
360  alac->output_samples_buffer[ch][i] =
361  get_sbits_long(&alac->gb, alac->sample_size);
362  }
363  }
364  alac->extra_bits = 0;
365  decorr_shift = 0;
366  decorr_left_weight = 0;
367  }
368 
369  if (channels == 2) {
370  if (alac->extra_bits && alac->extra_bit_bug) {
372  alac->extra_bits, channels, alac->nb_samples);
373  }
374 
375  if (decorr_left_weight) {
377  decorr_shift, decorr_left_weight);
378  }
379 
380  if (alac->extra_bits && !alac->extra_bit_bug) {
382  alac->extra_bits, channels, alac->nb_samples);
383  }
384  } else if (alac->extra_bits) {
386  alac->extra_bits, channels, alac->nb_samples);
387  }
388 
389  switch(alac->sample_size) {
390  case 16: {
391  for (ch = 0; ch < channels; ch++) {
392  int16_t *outbuffer = (int16_t *)frame->extended_data[ch_index + ch];
393  for (i = 0; i < alac->nb_samples; i++)
394  *outbuffer++ = alac->output_samples_buffer[ch][i];
395  }}
396  break;
397  case 20: {
398  for (ch = 0; ch < channels; ch++) {
399  for (i = 0; i < alac->nb_samples; i++)
400  alac->output_samples_buffer[ch][i] *= 1U << 12;
401  }}
402  break;
403  case 24: {
404  for (ch = 0; ch < channels; ch++) {
405  for (i = 0; i < alac->nb_samples; i++)
406  alac->output_samples_buffer[ch][i] *= 1U << 8;
407  }}
408  break;
409  }
410 
411  return 0;
412 }
413 
415  int *got_frame_ptr, AVPacket *avpkt)
416 {
417  ALACContext *alac = avctx->priv_data;
418  AVFrame *frame = data;
419  enum AlacRawDataBlockType element;
420  int channels;
421  int ch, ret, got_end;
422 
423  if ((ret = init_get_bits8(&alac->gb, avpkt->data, avpkt->size)) < 0)
424  return ret;
425 
426  got_end = 0;
427  alac->nb_samples = 0;
428  ch = 0;
429  while (get_bits_left(&alac->gb) >= 3) {
430  element = get_bits(&alac->gb, 3);
431  if (element == TYPE_END) {
432  got_end = 1;
433  break;
434  }
435  if (element > TYPE_CPE && element != TYPE_LFE) {
436  avpriv_report_missing_feature(avctx, "Syntax element %d", element);
437  return AVERROR_PATCHWELCOME;
438  }
439 
440  channels = (element == TYPE_CPE) ? 2 : 1;
441  if (ch + channels > alac->channels ||
442  ff_alac_channel_layout_offsets[alac->channels - 1][ch] + channels > alac->channels) {
443  av_log(avctx, AV_LOG_ERROR, "invalid element channel count\n");
444  return AVERROR_INVALIDDATA;
445  }
446 
447  ret = decode_element(avctx, frame,
449  channels);
450  if (ret < 0 && get_bits_left(&alac->gb))
451  return ret;
452 
453  ch += channels;
454  }
455  if (!got_end) {
456  av_log(avctx, AV_LOG_ERROR, "no end tag found. incomplete packet.\n");
457  return AVERROR_INVALIDDATA;
458  }
459 
460  if (avpkt->size * 8 - get_bits_count(&alac->gb) > 8) {
461  av_log(avctx, AV_LOG_ERROR, "Error : %d bits left\n",
462  avpkt->size * 8 - get_bits_count(&alac->gb));
463  }
464 
465  if (alac->channels == ch && alac->nb_samples)
466  *got_frame_ptr = 1;
467  else
468  av_log(avctx, AV_LOG_WARNING, "Failed to decode all channels\n");
469 
470  return avpkt->size;
471 }
472 
474 {
475  ALACContext *alac = avctx->priv_data;
476 
477  int ch;
478  for (ch = 0; ch < FFMIN(alac->channels, 2); ch++) {
479  av_freep(&alac->predict_error_buffer[ch]);
480  if (!alac->direct_output)
481  av_freep(&alac->output_samples_buffer[ch]);
482  av_freep(&alac->extra_bits_buffer[ch]);
483  }
484 
485  return 0;
486 }
487 
488 static int allocate_buffers(ALACContext *alac)
489 {
490  int ch;
491  unsigned buf_size = alac->max_samples_per_frame * sizeof(int32_t);
492  unsigned extra_buf_size = buf_size + AV_INPUT_BUFFER_PADDING_SIZE;
493 
494  for (ch = 0; ch < 2; ch++) {
495  alac->predict_error_buffer[ch] = NULL;
496  alac->output_samples_buffer[ch] = NULL;
497  alac->extra_bits_buffer[ch] = NULL;
498  }
499 
500  for (ch = 0; ch < FFMIN(alac->channels, 2); ch++) {
501  if (!(alac->predict_error_buffer[ch] = av_malloc(buf_size)))
502  return AVERROR(ENOMEM);
503 
504  alac->direct_output = alac->sample_size > 16;
505  if (!alac->direct_output) {
506  if (!(alac->output_samples_buffer[ch] = av_malloc(extra_buf_size)))
507  return AVERROR(ENOMEM);
508  }
509 
510  if (!(alac->extra_bits_buffer[ch] = av_malloc(extra_buf_size)))
511  return AVERROR(ENOMEM);
512  }
513  return 0;
514 }
515 
516 static int alac_set_info(ALACContext *alac)
517 {
519 
520  bytestream2_init(&gb, alac->avctx->extradata,
521  alac->avctx->extradata_size);
522 
523  bytestream2_skipu(&gb, 12); // size:4, alac:4, version:4
524 
525  alac->max_samples_per_frame = bytestream2_get_be32u(&gb);
526  if (!alac->max_samples_per_frame ||
527  alac->max_samples_per_frame > 4096 * 4096) {
528  av_log(alac->avctx, AV_LOG_ERROR,
529  "max samples per frame invalid: %"PRIu32"\n",
530  alac->max_samples_per_frame);
531  return AVERROR_INVALIDDATA;
532  }
533  bytestream2_skipu(&gb, 1); // compatible version
534  alac->sample_size = bytestream2_get_byteu(&gb);
535  alac->rice_history_mult = bytestream2_get_byteu(&gb);
536  alac->rice_initial_history = bytestream2_get_byteu(&gb);
537  alac->rice_limit = bytestream2_get_byteu(&gb);
538  alac->channels = bytestream2_get_byteu(&gb);
539  bytestream2_get_be16u(&gb); // maxRun
540  bytestream2_get_be32u(&gb); // max coded frame size
541  bytestream2_get_be32u(&gb); // average bitrate
542  alac->sample_rate = bytestream2_get_be32u(&gb);
543 
544  return 0;
545 }
546 
548 {
549  int ret;
550  ALACContext *alac = avctx->priv_data;
551  alac->avctx = avctx;
552 
553  /* initialize from the extradata */
555  av_log(avctx, AV_LOG_ERROR, "extradata is too small\n");
556  return AVERROR_INVALIDDATA;
557  }
558  if ((ret = alac_set_info(alac)) < 0) {
559  av_log(avctx, AV_LOG_ERROR, "set_info failed\n");
560  return ret;
561  }
562 
563  switch (alac->sample_size) {
564  case 16: avctx->sample_fmt = AV_SAMPLE_FMT_S16P;
565  break;
566  case 20:
567  case 24:
568  case 32: avctx->sample_fmt = AV_SAMPLE_FMT_S32P;
569  break;
570  default: avpriv_request_sample(avctx, "Sample depth %d", alac->sample_size);
571  return AVERROR_PATCHWELCOME;
572  }
573  avctx->bits_per_raw_sample = alac->sample_size;
574  avctx->sample_rate = alac->sample_rate;
575 
576  if (alac->channels < 1) {
577  av_log(avctx, AV_LOG_WARNING, "Invalid channel count\n");
578  alac->channels = avctx->channels;
579  } else {
580  if (alac->channels > ALAC_MAX_CHANNELS)
581  alac->channels = avctx->channels;
582  else
583  avctx->channels = alac->channels;
584  }
585  if (avctx->channels > ALAC_MAX_CHANNELS || avctx->channels <= 0 ) {
586  avpriv_report_missing_feature(avctx, "Channel count %d",
587  avctx->channels);
588  return AVERROR_PATCHWELCOME;
589  }
590  avctx->channel_layout = ff_alac_channel_layouts[alac->channels - 1];
591 
592  if ((ret = allocate_buffers(alac)) < 0) {
593  av_log(avctx, AV_LOG_ERROR, "Error allocating buffers\n");
594  return ret;
595  }
596 
597  ff_alacdsp_init(&alac->dsp);
598 
599  return 0;
600 }
601 
602 static const AVOption options[] = {
603  { "extra_bits_bug", "Force non-standard decoding process",
604  offsetof(ALACContext, extra_bit_bug), AV_OPT_TYPE_BOOL, { .i64 = 0 },
606  { NULL },
607 };
608 
609 static const AVClass alac_class = {
610  .class_name = "alac",
611  .item_name = av_default_item_name,
612  .option = options,
613  .version = LIBAVUTIL_VERSION_INT,
614 };
615 
617  .name = "alac",
618  .long_name = NULL_IF_CONFIG_SMALL("ALAC (Apple Lossless Audio Codec)"),
619  .type = AVMEDIA_TYPE_AUDIO,
620  .id = AV_CODEC_ID_ALAC,
621  .priv_data_size = sizeof(ALACContext),
623  .close = alac_decode_close,
626  .caps_internal = FF_CODEC_CAP_INIT_CLEANUP,
627  .priv_class = &alac_class
628 };
#define FF_CODEC_CAP_INIT_CLEANUP
The codec allows calling the close function for deallocation even if the init function returned a fai...
Definition: internal.h:48
int extra_bit_bug
Definition: alac.c:85
#define NULL
Definition: coverity.c:32
#define AVERROR_INVALIDDATA
Invalid data found when processing input.
Definition: error.h:59
int nb_samples
number of samples in the current frame
Definition: alac.c:82
This structure describes decoded (raw) audio or video data.
Definition: frame.h:308
#define ALAC_EXTRADATA_SIZE
Definition: alac.c:62
AVOption.
Definition: opt.h:248
ptrdiff_t const GLvoid * data
Definition: opengl_enc.c:100
static unsigned int get_bits(GetBitContext *s, int n)
Read 1-25 bits.
Definition: get_bits.h:379
#define AV_LOG_WARNING
Something somehow does not look correct.
Definition: log.h:200
AVFrame * f
Definition: thread.h:35
static int decode_element(AVCodecContext *avctx, AVFrame *frame, int ch_index, int channels)
Definition: alac.c:237
#define LIBAVUTIL_VERSION_INT
Definition: version.h:85
Definition: aac.h:63
static av_cold int init(AVCodecContext *avctx)
Definition: avrndec.c:35
#define avpriv_request_sample(...)
Definition: aac.h:57
#define AV_OPT_FLAG_AUDIO_PARAM
Definition: opt.h:280
int size
Definition: packet.h:356
const char * av_default_item_name(void *ptr)
Return the context name.
Definition: log.c:235
int av_log2(unsigned v)
Definition: intmath.c:26
static av_always_inline void bytestream2_init(GetByteContext *g, const uint8_t *buf, int buf_size)
Definition: bytestream.h:133
static const AVOption options[]
Definition: alac.c:602
int bits_per_raw_sample
Bits per sample/pixel of internal libavcodec pixel/sample format.
Definition: avcodec.h:1757
int32_t * extra_bits_buffer[2]
Definition: alac.c:72
AVCodec.
Definition: codec.h:190
static int get_sbits(GetBitContext *s, int n)
Definition: get_bits.h:359
static void decode(AVCodecContext *dec_ctx, AVPacket *pkt, AVFrame *frame, FILE *outfile)
Definition: decode_audio.c:71
static int get_sbits_long(GetBitContext *s, int n)
Read 0-32 bits as a signed integer.
Definition: get_bits.h:590
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
int32_t * predict_error_buffer[2]
Definition: alac.c:70
static int get_unary_0_9(GetBitContext *gb)
Definition: unary.h:64
enum AVSampleFormat sample_fmt
audio sample format
Definition: avcodec.h:1194
uint8_t
#define av_cold
Definition: attributes.h:88
#define av_malloc(s)
AVOptions.
uint8_t rice_initial_history
Definition: alac.c:77
Definition: aac.h:59
static av_cold int alac_decode_close(AVCodecContext *avctx)
Definition: alac.c:473
Multithreading support functions.
uint8_t * extradata
some codecs need / can use extradata like Huffman tables.
Definition: avcodec.h:627
uint8_t * data
Definition: packet.h:355
static int get_bits_count(const GetBitContext *s)
Definition: get_bits.h:219
static av_always_inline void bytestream2_skipu(GetByteContext *g, unsigned int size)
Definition: bytestream.h:170
static const AVClass alac_class
Definition: alac.c:609
bitstream reader API header.
int32_t * output_samples_buffer[2]
Definition: alac.c:71
static int alac_decode_frame(AVCodecContext *avctx, void *data, int *got_frame_ptr, AVPacket *avpkt)
Definition: alac.c:414
int sample_rate
Definition: alac.c:79
channels
Definition: aptx.h:33
int extra_bits
number of extra bits beyond 16-bit
Definition: alac.c:81
#define av_log(a,...)
#define U(x)
Definition: vp56_arith.h:37
static int get_bits_left(GetBitContext *gb)
Definition: get_bits.h:849
#define AV_LOG_ERROR
Something went wrong and cannot losslessly be recovered.
Definition: log.h:194
#define NULL_IF_CONFIG_SMALL(x)
Return NULL if CONFIG_SMALL is true, otherwise the argument without modification. ...
Definition: internal.h:153
static int sign_only(int v)
Definition: alac.c:169
AlacRawDataBlockType
Definition: alac_data.h:26
const char * name
Name of the codec implementation.
Definition: codec.h:197
#define AV_CODEC_CAP_FRAME_THREADS
Codec supports frame-level multithreading.
Definition: codec.h:106
uint64_t channel_layout
Audio channel layout.
Definition: avcodec.h:1237
#define ALAC_MAX_CHANNELS
Definition: alac_data.h:38
uint32_t max_samples_per_frame
Definition: alac.c:74
the pkt_dts and pkt_pts fields in AVFrame will work as usual Restrictions on codec whose streams don t reset across will not work because their bitstreams cannot be decoded in parallel *The contents of buffers must not be read before as well as code calling up to before the decode process starts Call have so the codec calls ff_thread_report set FF_CODEC_CAP_ALLOCATE_PROGRESS in AVCodec caps_internal and use ff_thread_get_buffer() to allocate frames.The frames must then be freed with ff_thread_release_buffer().Otherwise decode directly into the user-supplied frames.Call ff_thread_report_progress() after some part of the current picture has decoded.A good place to put this is where draw_horiz_band() is called-add this if it isn't called anywhere
audio channel layout utility functions
#define FFMIN(a, b)
Definition: common.h:96
signed 32 bits, planar
Definition: samplefmt.h:68
#define FFSIGN(a)
Definition: common.h:73
GetBitContext gb
Definition: alac.c:67
int32_t
static unsigned int show_bits(GetBitContext *s, int n)
Show 1-25 bits.
Definition: get_bits.h:446
ALACDSPContext dsp
Definition: alac.c:87
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
static int alac_set_info(ALACContext *alac)
Definition: alac.c:516
uint8_t rice_history_mult
Definition: alac.c:76
const uint64_t ff_alac_channel_layouts[ALAC_MAX_CHANNELS+1]
Definition: alac_data.c:35
static const float pred[4]
Definition: siprdata.h:259
#define AVERROR_PATCHWELCOME
Not yet implemented in FFmpeg, patches welcome.
Definition: error.h:62
Libavcodec external API header.
int sample_rate
samples per second
Definition: avcodec.h:1186
static int init_get_bits8(GetBitContext *s, const uint8_t *buffer, int byte_size)
Initialize GetBitContext.
Definition: get_bits.h:677
main external API structure.
Definition: avcodec.h:526
int extradata_size
Definition: avcodec.h:628
static unsigned int get_bits1(GetBitContext *s)
Definition: get_bits.h:498
Describe the class of an AVClass context structure.
Definition: log.h:67
static void skip_bits(GetBitContext *s, int n)
Definition: get_bits.h:467
static unsigned int decode_scalar(GetBitContext *gb, int k, int bps)
Definition: alac.c:90
#define AV_OPT_FLAG_DECODING_PARAM
a generic parameter which can be set by the user for demuxing or decoding
Definition: opt.h:279
void(* append_extra_bits[2])(int32_t *buffer[2], int32_t *extra_bits_buffer[2], int extra_bits, int channels, int nb_samples)
Definition: alacdsp.h:27
static unsigned int get_bits_long(GetBitContext *s, int n)
Read 0-32 bits.
Definition: get_bits.h:546
const uint8_t ff_alac_channel_layout_offsets[ALAC_MAX_CHANNELS][ALAC_MAX_CHANNELS]
Definition: alac_data.c:24
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/(UINT64_C(1)<< 63))) CONV_FUNC(AV_SAMPLE_FMT_DBL, double, AV_SAMPLE_FMT_S64,*(const int64_t *) pi *(1.0/(UINT64_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 *(UINT64_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 *(UINT64_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
static av_const int sign_extend(int val, unsigned bits)
Definition: mathops.h:130
void avpriv_report_missing_feature(void *avc, const char *msg,...) av_printf_format(2
Log a generic warning message about a missing feature.
void(* decorrelate_stereo)(int32_t *buffer[2], int nb_samples, int decorr_shift, int decorr_left_weight)
Definition: alacdsp.h:25
uint8_t sample_size
Definition: alac.c:75
int channels
Definition: alac.c:68
AVCodec ff_alac_decoder
Definition: alac.c:616
common internal api header.
static av_cold int alac_decode_init(AVCodecContext *avctx)
Definition: alac.c:547
uint8_t rice_limit
Definition: alac.c:78
unsigned bps
Definition: movenc.c:1533
#define AV_INPUT_BUFFER_PADDING_SIZE
Required number of additionally allocated bytes at the end of the input bitstream for decoding...
Definition: avcodec.h:215
void * priv_data
Definition: avcodec.h:553
static int allocate_buffers(ALACContext *alac)
Definition: alac.c:488
int channels
number of audio channels
Definition: avcodec.h:1187
AVCodecContext * avctx
Definition: alac.c:66
static void lpc_prediction(int32_t *error_buffer, uint32_t *buffer_out, int nb_samples, int bps, int16_t *lpc_coefs, int lpc_order, int lpc_quant)
Definition: alac.c:174
static int rice_decompress(ALACContext *alac, int32_t *output_buffer, int nb_samples, int bps, int rice_history_mult)
Definition: alac.c:112
#define av_freep(p)
signed 16 bits, planar
Definition: samplefmt.h:67
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:355
av_cold void ff_alacdsp_init(ALACDSPContext *c)
Definition: alacdsp.c:55
int direct_output
Definition: alac.c:84
static double val(void *priv, double ch)
Definition: aeval.c:76
This structure stores compressed data.
Definition: packet.h:332
int nb_samples
number of audio samples (per channel) described by this frame
Definition: frame.h:374
#define AV_CODEC_CAP_DR1
Codec uses get_buffer() for allocating buffers and supports custom allocators.
Definition: codec.h:50
for(j=16;j >0;--j)
int i
Definition: input.c:406