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  if (channels == 2 && decorr_left_weight && decorr_shift > 31)
306  return AVERROR_INVALIDDATA;
307 
308  for (ch = 0; ch < channels; ch++) {
309  prediction_type[ch] = get_bits(&alac->gb, 4);
310  lpc_quant[ch] = get_bits(&alac->gb, 4);
311  rice_history_mult[ch] = get_bits(&alac->gb, 3);
312  lpc_order[ch] = get_bits(&alac->gb, 5);
313 
314  if (lpc_order[ch] >= alac->max_samples_per_frame || !lpc_quant[ch])
315  return AVERROR_INVALIDDATA;
316 
317  /* read the predictor table */
318  for (i = lpc_order[ch] - 1; i >= 0; i--)
319  lpc_coefs[ch][i] = get_sbits(&alac->gb, 16);
320  }
321 
322  if (alac->extra_bits) {
323  for (i = 0; i < alac->nb_samples; i++) {
324  if(get_bits_left(&alac->gb) <= 0)
325  return AVERROR_INVALIDDATA;
326  for (ch = 0; ch < channels; ch++)
327  alac->extra_bits_buffer[ch][i] = get_bits(&alac->gb, alac->extra_bits);
328  }
329  }
330  for (ch = 0; ch < channels; ch++) {
331  int ret=rice_decompress(alac, alac->predict_error_buffer[ch],
332  alac->nb_samples, bps,
333  rice_history_mult[ch] * alac->rice_history_mult / 4);
334  if(ret<0)
335  return ret;
336 
337  /* adaptive FIR filter */
338  if (prediction_type[ch] == 15) {
339  /* Prediction type 15 runs the adaptive FIR twice.
340  * The first pass uses the special-case coef_num = 31, while
341  * the second pass uses the coefs from the bitstream.
342  *
343  * However, this prediction type is not currently used by the
344  * reference encoder.
345  */
347  alac->predict_error_buffer[ch],
348  alac->nb_samples, bps, NULL, 31, 0);
349  } else if (prediction_type[ch] > 0) {
350  av_log(avctx, AV_LOG_WARNING, "unknown prediction type: %i\n",
351  prediction_type[ch]);
352  }
354  alac->output_samples_buffer[ch], alac->nb_samples,
355  bps, lpc_coefs[ch], lpc_order[ch], lpc_quant[ch]);
356  }
357  } else {
358  /* not compressed, easy case */
359  for (i = 0; i < alac->nb_samples; i++) {
360  if(get_bits_left(&alac->gb) <= 0)
361  return AVERROR_INVALIDDATA;
362  for (ch = 0; ch < channels; ch++) {
363  alac->output_samples_buffer[ch][i] =
364  get_sbits_long(&alac->gb, alac->sample_size);
365  }
366  }
367  alac->extra_bits = 0;
368  decorr_shift = 0;
369  decorr_left_weight = 0;
370  }
371 
372  if (channels == 2) {
373  if (alac->extra_bits && alac->extra_bit_bug) {
375  alac->extra_bits, channels, alac->nb_samples);
376  }
377 
378  if (decorr_left_weight) {
380  decorr_shift, decorr_left_weight);
381  }
382 
383  if (alac->extra_bits && !alac->extra_bit_bug) {
385  alac->extra_bits, channels, alac->nb_samples);
386  }
387  } else if (alac->extra_bits) {
389  alac->extra_bits, channels, alac->nb_samples);
390  }
391 
392  switch(alac->sample_size) {
393  case 16: {
394  for (ch = 0; ch < channels; ch++) {
395  int16_t *outbuffer = (int16_t *)frame->extended_data[ch_index + ch];
396  for (i = 0; i < alac->nb_samples; i++)
397  *outbuffer++ = alac->output_samples_buffer[ch][i];
398  }}
399  break;
400  case 20: {
401  for (ch = 0; ch < channels; ch++) {
402  for (i = 0; i < alac->nb_samples; i++)
403  alac->output_samples_buffer[ch][i] *= 1U << 12;
404  }}
405  break;
406  case 24: {
407  for (ch = 0; ch < channels; ch++) {
408  for (i = 0; i < alac->nb_samples; i++)
409  alac->output_samples_buffer[ch][i] *= 1U << 8;
410  }}
411  break;
412  }
413 
414  return 0;
415 }
416 
417 static int alac_decode_frame(AVCodecContext *avctx, void *data,
418  int *got_frame_ptr, AVPacket *avpkt)
419 {
420  ALACContext *alac = avctx->priv_data;
421  AVFrame *frame = data;
422  enum AlacRawDataBlockType element;
423  int channels;
424  int ch, ret, got_end;
425 
426  if ((ret = init_get_bits8(&alac->gb, avpkt->data, avpkt->size)) < 0)
427  return ret;
428 
429  got_end = 0;
430  alac->nb_samples = 0;
431  ch = 0;
432  while (get_bits_left(&alac->gb) >= 3) {
433  element = get_bits(&alac->gb, 3);
434  if (element == TYPE_END) {
435  got_end = 1;
436  break;
437  }
438  if (element > TYPE_CPE && element != TYPE_LFE) {
439  avpriv_report_missing_feature(avctx, "Syntax element %d", element);
440  return AVERROR_PATCHWELCOME;
441  }
442 
443  channels = (element == TYPE_CPE) ? 2 : 1;
444  if (ch + channels > alac->channels ||
446  av_log(avctx, AV_LOG_ERROR, "invalid element channel count\n");
447  return AVERROR_INVALIDDATA;
448  }
449 
450  ret = decode_element(avctx, frame,
452  channels);
453  if (ret < 0 && get_bits_left(&alac->gb))
454  return ret;
455 
456  ch += channels;
457  }
458  if (!got_end) {
459  av_log(avctx, AV_LOG_ERROR, "no end tag found. incomplete packet.\n");
460  return AVERROR_INVALIDDATA;
461  }
462 
463  if (avpkt->size * 8 - get_bits_count(&alac->gb) > 8) {
464  av_log(avctx, AV_LOG_ERROR, "Error : %d bits left\n",
465  avpkt->size * 8 - get_bits_count(&alac->gb));
466  }
467 
468  if (alac->channels == ch && alac->nb_samples)
469  *got_frame_ptr = 1;
470  else
471  av_log(avctx, AV_LOG_WARNING, "Failed to decode all channels\n");
472 
473  return avpkt->size;
474 }
475 
477 {
478  ALACContext *alac = avctx->priv_data;
479 
480  int ch;
481  for (ch = 0; ch < FFMIN(alac->channels, 2); ch++) {
483  if (!alac->direct_output)
485  av_freep(&alac->extra_bits_buffer[ch]);
486  }
487 
488  return 0;
489 }
490 
491 static int allocate_buffers(ALACContext *alac)
492 {
493  int ch;
494  unsigned buf_size = alac->max_samples_per_frame * sizeof(int32_t);
495 
496  for (ch = 0; ch < 2; ch++) {
497  alac->predict_error_buffer[ch] = NULL;
498  alac->output_samples_buffer[ch] = NULL;
499  alac->extra_bits_buffer[ch] = NULL;
500  }
501 
502  for (ch = 0; ch < FFMIN(alac->channels, 2); ch++) {
504  buf_size, buf_alloc_fail);
505 
506  alac->direct_output = alac->sample_size > 16;
507  if (!alac->direct_output) {
509  buf_size + AV_INPUT_BUFFER_PADDING_SIZE, buf_alloc_fail);
510  }
511 
513  buf_size + AV_INPUT_BUFFER_PADDING_SIZE, buf_alloc_fail);
514  }
515  return 0;
516 buf_alloc_fail:
517  alac_decode_close(alac->avctx);
518  return AVERROR(ENOMEM);
519 }
520 
521 static int alac_set_info(ALACContext *alac)
522 {
523  GetByteContext gb;
524 
525  bytestream2_init(&gb, alac->avctx->extradata,
526  alac->avctx->extradata_size);
527 
528  bytestream2_skipu(&gb, 12); // size:4, alac:4, version:4
529 
530  alac->max_samples_per_frame = bytestream2_get_be32u(&gb);
531  if (!alac->max_samples_per_frame ||
532  alac->max_samples_per_frame > 4096 * 4096) {
533  av_log(alac->avctx, AV_LOG_ERROR,
534  "max samples per frame invalid: %"PRIu32"\n",
535  alac->max_samples_per_frame);
536  return AVERROR_INVALIDDATA;
537  }
538  bytestream2_skipu(&gb, 1); // compatible version
539  alac->sample_size = bytestream2_get_byteu(&gb);
540  alac->rice_history_mult = bytestream2_get_byteu(&gb);
541  alac->rice_initial_history = bytestream2_get_byteu(&gb);
542  alac->rice_limit = bytestream2_get_byteu(&gb);
543  alac->channels = bytestream2_get_byteu(&gb);
544  bytestream2_get_be16u(&gb); // maxRun
545  bytestream2_get_be32u(&gb); // max coded frame size
546  bytestream2_get_be32u(&gb); // average bitrate
547  alac->sample_rate = bytestream2_get_be32u(&gb);
548 
549  return 0;
550 }
551 
553 {
554  int ret;
555  ALACContext *alac = avctx->priv_data;
556  alac->avctx = avctx;
557 
558  /* initialize from the extradata */
560  av_log(avctx, AV_LOG_ERROR, "extradata is too small\n");
561  return AVERROR_INVALIDDATA;
562  }
563  if ((ret = alac_set_info(alac)) < 0) {
564  av_log(avctx, AV_LOG_ERROR, "set_info failed\n");
565  return ret;
566  }
567 
568  switch (alac->sample_size) {
569  case 16: avctx->sample_fmt = AV_SAMPLE_FMT_S16P;
570  break;
571  case 20:
572  case 24:
573  case 32: avctx->sample_fmt = AV_SAMPLE_FMT_S32P;
574  break;
575  default: avpriv_request_sample(avctx, "Sample depth %d", alac->sample_size);
576  return AVERROR_PATCHWELCOME;
577  }
578  avctx->bits_per_raw_sample = alac->sample_size;
579  avctx->sample_rate = alac->sample_rate;
580 
581  if (alac->channels < 1) {
582  av_log(avctx, AV_LOG_WARNING, "Invalid channel count\n");
583  alac->channels = avctx->channels;
584  } else {
585  if (alac->channels > ALAC_MAX_CHANNELS)
586  alac->channels = avctx->channels;
587  else
588  avctx->channels = alac->channels;
589  }
590  if (avctx->channels > ALAC_MAX_CHANNELS || avctx->channels <= 0 ) {
591  avpriv_report_missing_feature(avctx, "Channel count %d",
592  avctx->channels);
593  return AVERROR_PATCHWELCOME;
594  }
595  avctx->channel_layout = ff_alac_channel_layouts[alac->channels - 1];
596 
597  if ((ret = allocate_buffers(alac)) < 0) {
598  av_log(avctx, AV_LOG_ERROR, "Error allocating buffers\n");
599  return ret;
600  }
601 
602  ff_alacdsp_init(&alac->dsp);
603 
604  return 0;
605 }
606 
607 #if HAVE_THREADS
608 static int init_thread_copy(AVCodecContext *avctx)
609 {
610  ALACContext *alac = avctx->priv_data;
611  alac->avctx = avctx;
612  return allocate_buffers(alac);
613 }
614 #endif
615 
616 static const AVOption options[] = {
617  { "extra_bits_bug", "Force non-standard decoding process",
618  offsetof(ALACContext, extra_bit_bug), AV_OPT_TYPE_BOOL, { .i64 = 0 },
620  { NULL },
621 };
622 
623 static const AVClass alac_class = {
624  .class_name = "alac",
625  .item_name = av_default_item_name,
626  .option = options,
627  .version = LIBAVUTIL_VERSION_INT,
628 };
629 
631  .name = "alac",
632  .long_name = NULL_IF_CONFIG_SMALL("ALAC (Apple Lossless Audio Codec)"),
633  .type = AVMEDIA_TYPE_AUDIO,
634  .id = AV_CODEC_ID_ALAC,
635  .priv_data_size = sizeof(ALACContext),
637  .close = alac_decode_close,
641  .priv_class = &alac_class
642 };
AVCodec
AVCodec.
Definition: avcodec.h:3481
AV_LOG_WARNING
#define AV_LOG_WARNING
Something somehow does not look correct.
Definition: log.h:182
ff_alacdsp_init
av_cold void ff_alacdsp_init(ALACDSPContext *c)
Definition: alacdsp.c:55
init
static av_cold int init(AVCodecContext *avctx)
Definition: avrndec.c:35
get_bits_left
static int get_bits_left(GetBitContext *gb)
Definition: get_bits.h:849
AVERROR
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
opt.h
ALACDSPContext::decorrelate_stereo
void(* decorrelate_stereo)(int32_t *buffer[2], int nb_samples, int decorr_shift, int decorr_left_weight)
Definition: alacdsp.h:25
AVCodecContext::channel_layout
uint64_t channel_layout
Audio channel layout.
Definition: avcodec.h:2276
get_unary_0_9
static int get_unary_0_9(GetBitContext *gb)
Definition: unary.h:64
alac_data.h
lpc_prediction
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
AVCodecContext::sample_rate
int sample_rate
samples per second
Definition: avcodec.h:2225
decode_scalar
static unsigned int decode_scalar(GetBitContext *gb, int k, int bps)
Definition: alac.c:90
GetByteContext
Definition: bytestream.h:33
get_bits_long
static unsigned int get_bits_long(GetBitContext *s, int n)
Read 0-32 bits.
Definition: get_bits.h:546
bytestream2_skipu
static av_always_inline void bytestream2_skipu(GetByteContext *g, unsigned int size)
Definition: bytestream.h:170
ch
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
get_bits_count
static int get_bits_count(const GetBitContext *s)
Definition: get_bits.h:219
ALACContext::extra_bits
int extra_bits
number of extra bits beyond 16-bit
Definition: alac.c:81
alac_decode_close
static av_cold int alac_decode_close(AVCodecContext *avctx)
Definition: alac.c:476
AVFrame
This structure describes decoded (raw) audio or video data.
Definition: frame.h:295
internal.h
AVPacket::data
uint8_t * data
Definition: avcodec.h:1477
AVOption
AVOption.
Definition: opt.h:246
AV_SAMPLE_FMT_S32P
@ AV_SAMPLE_FMT_S32P
signed 32 bits, planar
Definition: samplefmt.h:68
data
const char data[16]
Definition: mxf.c:91
AV_CODEC_ID_ALAC
@ AV_CODEC_ID_ALAC
Definition: avcodec.h:580
ALACContext::rice_history_mult
uint8_t rice_history_mult
Definition: alac.c:76
channels
channels
Definition: aptx.c:30
rice_decompress
static int rice_decompress(ALACContext *alac, int32_t *output_buffer, int nb_samples, int bps, int rice_history_mult)
Definition: alac.c:112
thread.h
ThreadFrame::f
AVFrame * f
Definition: thread.h:35
alacdsp.h
sign_only
static int sign_only(int v)
Definition: alac.c:169
skip_bits
static void skip_bits(GetBitContext *s, int n)
Definition: get_bits.h:467
get_bits
static unsigned int get_bits(GetBitContext *s, int n)
Read 1-25 bits.
Definition: get_bits.h:379
U
#define U(x)
Definition: vp56_arith.h:37
FFSIGN
#define FFSIGN(a)
Definition: common.h:73
TYPE_CPE
@ TYPE_CPE
Definition: aac.h:57
GetBitContext
Definition: get_bits.h:61
AV_LOG_ERROR
#define AV_LOG_ERROR
Something went wrong and cannot losslessly be recovered.
Definition: log.h:176
av_cold
#define av_cold
Definition: attributes.h:84
ff_alac_decoder
AVCodec ff_alac_decoder
Definition: alac.c:630
init_get_bits8
static int init_get_bits8(GetBitContext *s, const uint8_t *buffer, int byte_size)
Initialize GetBitContext.
Definition: get_bits.h:677
decode
static void decode(AVCodecContext *dec_ctx, AVPacket *pkt, AVFrame *frame, FILE *outfile)
Definition: decode_audio.c:42
AVCodecContext::extradata_size
int extradata_size
Definition: avcodec.h:1667
AVMEDIA_TYPE_AUDIO
@ AVMEDIA_TYPE_AUDIO
Definition: avutil.h:202
ALACContext::channels
int channels
Definition: alac.c:68
AVCodecContext::bits_per_raw_sample
int bits_per_raw_sample
Bits per sample/pixel of internal libavcodec pixel/sample format.
Definition: avcodec.h:2796
get_sbits
static int get_sbits(GetBitContext *s, int n)
Definition: get_bits.h:359
get_bits.h
ALACContext::predict_error_buffer
int32_t * predict_error_buffer[2]
Definition: alac.c:70
AV_OPT_FLAG_AUDIO_PARAM
#define AV_OPT_FLAG_AUDIO_PARAM
Definition: opt.h:278
int32_t
int32_t
Definition: audio_convert.c:194
decode_element
static int decode_element(AVCodecContext *avctx, AVFrame *frame, int ch_index, int channels)
Definition: alac.c:237
AV_CODEC_CAP_FRAME_THREADS
#define AV_CODEC_CAP_FRAME_THREADS
Codec supports frame-level multithreading.
Definition: avcodec.h:1037
ALACContext::dsp
ALACDSPContext dsp
Definition: alac.c:87
LIBAVUTIL_VERSION_INT
#define LIBAVUTIL_VERSION_INT
Definition: version.h:85
AVClass
Describe the class of an AVClass context structure.
Definition: log.h:67
NULL
#define NULL
Definition: coverity.c:32
AVERROR_PATCHWELCOME
#define AVERROR_PATCHWELCOME
Not yet implemented in FFmpeg, patches welcome.
Definition: error.h:62
options
static const AVOption options[]
Definition: alac.c:616
av_default_item_name
const char * av_default_item_name(void *ptr)
Return the context name.
Definition: log.c:191
get_bits1
static unsigned int get_bits1(GetBitContext *s)
Definition: get_bits.h:498
mathops.h
ONLY_IF_THREADS_ENABLED
#define ONLY_IF_THREADS_ENABLED(x)
Define a function with only the non-default version specified.
Definition: internal.h:227
ff_thread_get_buffer
int ff_thread_get_buffer(AVCodecContext *avctx, ThreadFrame *f, int flags)
Wrapper around get_buffer() for frame-multithreaded codecs.
Definition: pthread_frame.c:964
ALACDSPContext
Definition: alacdsp.h:24
ALACContext::nb_samples
int nb_samples
number of samples in the current frame
Definition: alac.c:82
ALACContext::max_samples_per_frame
uint32_t max_samples_per_frame
Definition: alac.c:74
for
for(j=16;j >0;--j)
Definition: h264pred_template.c:469
ALACContext::rice_initial_history
uint8_t rice_initial_history
Definition: alac.c:77
alac_decode_init
static av_cold int alac_decode_init(AVCodecContext *avctx)
Definition: alac.c:552
AV_CODEC_CAP_DR1
#define AV_CODEC_CAP_DR1
Codec uses get_buffer() for allocating buffers and supports custom allocators.
Definition: avcodec.h:981
AVPacket::size
int size
Definition: avcodec.h:1478
NULL_IF_CONFIG_SMALL
#define NULL_IF_CONFIG_SMALL(x)
Return NULL if CONFIG_SMALL is true, otherwise the argument without modification.
Definition: internal.h:188
TYPE_END
@ TYPE_END
Definition: aac.h:63
bps
unsigned bps
Definition: movenc.c:1497
AVCodecContext::sample_fmt
enum AVSampleFormat sample_fmt
audio sample format
Definition: avcodec.h:2233
init_thread_copy
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 add an init_thread_copy() which re-allocates them for other threads. Add AV_CODEC_CAP_FRAME_THREADS to the codec capabilities. There will be very little speed gain at this point but it should work. If there are inter-frame dependencies
ALACContext::output_samples_buffer
int32_t * output_samples_buffer[2]
Definition: alac.c:71
avpriv_report_missing_feature
void avpriv_report_missing_feature(void *avc, const char *msg,...) av_printf_format(2
Log a generic warning message about a missing feature.
ALACContext
Definition: alac.c:64
ALACContext::direct_output
int direct_output
Definition: alac.c:84
val
const char const char void * val
Definition: avisynth_c.h:863
ALACContext::rice_limit
uint8_t rice_limit
Definition: alac.c:78
FFMIN
#define FFMIN(a, b)
Definition: common.h:96
ALAC_MAX_CHANNELS
#define ALAC_MAX_CHANNELS
Definition: alac_data.h:38
unary.h
AV_SAMPLE_FMT_S16P
@ AV_SAMPLE_FMT_S16P
signed 16 bits, planar
Definition: samplefmt.h:67
AVCodecContext::channels
int channels
number of audio channels
Definition: avcodec.h:2226
i
#define i(width, name, range_min, range_max)
Definition: cbs_h2645.c:259
ff_alac_channel_layout_offsets
const uint8_t ff_alac_channel_layout_offsets[ALAC_MAX_CHANNELS][ALAC_MAX_CHANNELS]
Definition: alac_data.c:24
AVCodecContext::extradata
uint8_t * extradata
some codecs need / can use extradata like Huffman tables.
Definition: avcodec.h:1666
show_bits
static unsigned int show_bits(GetBitContext *s, int n)
Show 1-25 bits.
Definition: get_bits.h:446
alac_class
static const AVClass alac_class
Definition: alac.c:623
ALACContext::avctx
AVCodecContext * avctx
Definition: alac.c:66
AV_OPT_FLAG_DECODING_PARAM
#define AV_OPT_FLAG_DECODING_PARAM
a generic parameter which can be set by the user for demuxing or decoding
Definition: opt.h:277
TYPE_LFE
@ TYPE_LFE
Definition: aac.h:59
uint8_t
uint8_t
Definition: audio_convert.c:194
AVCodec::name
const char * name
Name of the codec implementation.
Definition: avcodec.h:3488
ff_alac_channel_layouts
const uint64_t ff_alac_channel_layouts[ALAC_MAX_CHANNELS+1]
Definition: alac_data.c:35
ALACContext::extra_bit_bug
int extra_bit_bug
Definition: alac.c:85
avcodec.h
ret
ret
Definition: filter_design.txt:187
pred
static const float pred[4]
Definition: siprdata.h:259
ALACContext::sample_size
uint8_t sample_size
Definition: alac.c:75
AVClass::class_name
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
frame
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
Definition: filter_design.txt:264
AV_INPUT_BUFFER_PADDING_SIZE
#define AV_INPUT_BUFFER_PADDING_SIZE
Definition: avcodec.h:790
ALACContext::sample_rate
int sample_rate
Definition: alac.c:79
AVCodecContext
main external API structure.
Definition: avcodec.h:1565
ThreadFrame
Definition: thread.h:34
alac_decode_frame
static int alac_decode_frame(AVCodecContext *avctx, void *data, int *got_frame_ptr, AVPacket *avpkt)
Definition: alac.c:417
channel_layout.h
allocate_buffers
static int allocate_buffers(ALACContext *alac)
Definition: alac.c:491
sign_extend
static av_const int sign_extend(int val, unsigned bits)
Definition: mathops.h:130
alac_set_info
static int alac_set_info(ALACContext *alac)
Definition: alac.c:521
AlacRawDataBlockType
AlacRawDataBlockType
Definition: alac_data.h:26
FF_ALLOC_OR_GOTO
#define FF_ALLOC_OR_GOTO(ctx, p, size, label)
Definition: internal.h:140
ALACDSPContext::append_extra_bits
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
ALACContext::extra_bits_buffer
int32_t * extra_bits_buffer[2]
Definition: alac.c:72
avpriv_request_sample
#define avpriv_request_sample(...)
Definition: tableprint_vlc.h:39
AVPacket
This structure stores compressed data.
Definition: avcodec.h:1454
AVCodecContext::priv_data
void * priv_data
Definition: avcodec.h:1592
AV_OPT_TYPE_BOOL
@ AV_OPT_TYPE_BOOL
Definition: opt.h:240
av_freep
#define av_freep(p)
Definition: tableprint_vlc.h:35
bytestream.h
bytestream2_init
static av_always_inline void bytestream2_init(GetByteContext *g, const uint8_t *buf, int buf_size)
Definition: bytestream.h:133
av_log
#define av_log(a,...)
Definition: tableprint_vlc.h:28
AVERROR_INVALIDDATA
#define AVERROR_INVALIDDATA
Invalid data found when processing input.
Definition: error.h:59
get_sbits_long
static int get_sbits_long(GetBitContext *s, int n)
Read 0-32 bits as a signed integer.
Definition: get_bits.h:590
int
int
Definition: ffmpeg_filter.c:191
av_log2
int av_log2(unsigned v)
Definition: intmath.c:26
ALACContext::gb
GetBitContext gb
Definition: alac.c:67
ALAC_EXTRADATA_SIZE
#define ALAC_EXTRADATA_SIZE
Definition: alac.c:62