shorten.c
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1 /*
2  * Shorten decoder
3  * Copyright (c) 2005 Jeff Muizelaar
4  *
5  * This file is part of Libav.
6  *
7  * Libav 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  * Libav 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 Libav; if not, write to the Free Software
19  * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
20  */
21 
29 #include <limits.h>
30 #include "avcodec.h"
31 #include "bytestream.h"
32 #include "get_bits.h"
33 #include "golomb.h"
34 
35 #define MAX_CHANNELS 8
36 #define MAX_BLOCKSIZE 65535
37 
38 #define OUT_BUFFER_SIZE 16384
39 
40 #define ULONGSIZE 2
41 
42 #define WAVE_FORMAT_PCM 0x0001
43 
44 #define DEFAULT_BLOCK_SIZE 256
45 
46 #define TYPESIZE 4
47 #define CHANSIZE 0
48 #define LPCQSIZE 2
49 #define ENERGYSIZE 3
50 #define BITSHIFTSIZE 2
51 
52 #define TYPE_S16HL 3
53 #define TYPE_S16LH 5
54 
55 #define NWRAP 3
56 #define NSKIPSIZE 1
57 
58 #define LPCQUANT 5
59 #define V2LPCQOFFSET (1 << LPCQUANT)
60 
61 #define FNSIZE 2
62 #define FN_DIFF0 0
63 #define FN_DIFF1 1
64 #define FN_DIFF2 2
65 #define FN_DIFF3 3
66 #define FN_QUIT 4
67 #define FN_BLOCKSIZE 5
68 #define FN_BITSHIFT 6
69 #define FN_QLPC 7
70 #define FN_ZERO 8
71 #define FN_VERBATIM 9
72 
74 static const uint8_t is_audio_command[10] = { 1, 1, 1, 1, 0, 0, 0, 1, 1, 0 };
75 
76 #define VERBATIM_CKSIZE_SIZE 5
77 #define VERBATIM_BYTE_SIZE 8
78 #define CANONICAL_HEADER_SIZE 44
79 
80 typedef struct ShortenContext {
84 
86  int channels;
87 
88  int32_t *decoded[MAX_CHANNELS];
90  int32_t *offset[MAX_CHANNELS];
91  int *coeffs;
92  uint8_t *bitstream;
98  int version;
99  int cur_chan;
100  int bitshift;
101  int nmean;
103  int nwrap;
105  int bitindex;
106  int32_t lpcqoffset;
110 
112 {
113  ShortenContext *s = avctx->priv_data;
114  s->avctx = avctx;
115  avctx->sample_fmt = AV_SAMPLE_FMT_S16;
116 
118  avctx->coded_frame = &s->frame;
119 
120  return 0;
121 }
122 
124 {
125  int i, chan;
126  int *coeffs;
127  void *tmp_ptr;
128 
129  for (chan=0; chan<s->channels; chan++) {
130  if(FFMAX(1, s->nmean) >= UINT_MAX/sizeof(int32_t)){
131  av_log(s->avctx, AV_LOG_ERROR, "nmean too large\n");
132  return -1;
133  }
134  if(s->blocksize + s->nwrap >= UINT_MAX/sizeof(int32_t) || s->blocksize + s->nwrap <= (unsigned)s->nwrap){
135  av_log(s->avctx, AV_LOG_ERROR, "s->blocksize + s->nwrap too large\n");
136  return -1;
137  }
138 
139  tmp_ptr = av_realloc(s->offset[chan], sizeof(int32_t)*FFMAX(1, s->nmean));
140  if (!tmp_ptr)
141  return AVERROR(ENOMEM);
142  s->offset[chan] = tmp_ptr;
143 
144  tmp_ptr = av_realloc(s->decoded_base[chan], (s->blocksize + s->nwrap) *
145  sizeof(s->decoded_base[0][0]));
146  if (!tmp_ptr)
147  return AVERROR(ENOMEM);
148  s->decoded_base[chan] = tmp_ptr;
149  for (i=0; i<s->nwrap; i++)
150  s->decoded_base[chan][i] = 0;
151  s->decoded[chan] = s->decoded_base[chan] + s->nwrap;
152  }
153 
154  coeffs = av_realloc(s->coeffs, s->nwrap * sizeof(*s->coeffs));
155  if (!coeffs)
156  return AVERROR(ENOMEM);
157  s->coeffs = coeffs;
158 
159  return 0;
160 }
161 
162 
163 static inline unsigned int get_uint(ShortenContext *s, int k)
164 {
165  if (s->version != 0)
167  return get_ur_golomb_shorten(&s->gb, k);
168 }
169 
170 
171 static void fix_bitshift(ShortenContext *s, int32_t *buffer)
172 {
173  int i;
174 
175  if (s->bitshift != 0)
176  for (i = 0; i < s->blocksize; i++)
177  buffer[i] <<= s->bitshift;
178 }
179 
180 
182 {
183  int32_t mean = 0;
184  int chan, i;
185  int nblock = FFMAX(1, s->nmean);
186  /* initialise offset */
187  switch (s->internal_ftype)
188  {
189  case TYPE_S16HL:
190  case TYPE_S16LH:
191  mean = 0;
192  break;
193  default:
194  av_log(s->avctx, AV_LOG_ERROR, "unknown audio type");
195  return AVERROR_INVALIDDATA;
196  }
197 
198  for (chan = 0; chan < s->channels; chan++)
199  for (i = 0; i < nblock; i++)
200  s->offset[chan][i] = mean;
201  return 0;
202 }
203 
204 static int decode_wave_header(AVCodecContext *avctx, const uint8_t *header,
205  int header_size)
206 {
207  int len;
208  short wave_format;
209 
210 
211  if (bytestream_get_le32(&header) != MKTAG('R','I','F','F')) {
212  av_log(avctx, AV_LOG_ERROR, "missing RIFF tag\n");
213  return -1;
214  }
215 
216  header += 4; /* chunk size */;
217 
218  if (bytestream_get_le32(&header) != MKTAG('W','A','V','E')) {
219  av_log(avctx, AV_LOG_ERROR, "missing WAVE tag\n");
220  return -1;
221  }
222 
223  while (bytestream_get_le32(&header) != MKTAG('f','m','t',' ')) {
224  len = bytestream_get_le32(&header);
225  header += len;
226  }
227  len = bytestream_get_le32(&header);
228 
229  if (len < 16) {
230  av_log(avctx, AV_LOG_ERROR, "fmt chunk was too short\n");
231  return -1;
232  }
233 
234  wave_format = bytestream_get_le16(&header);
235 
236  switch (wave_format) {
237  case WAVE_FORMAT_PCM:
238  break;
239  default:
240  av_log(avctx, AV_LOG_ERROR, "unsupported wave format\n");
241  return -1;
242  }
243 
244  header += 2; // skip channels (already got from shorten header)
245  avctx->sample_rate = bytestream_get_le32(&header);
246  header += 4; // skip bit rate (represents original uncompressed bit rate)
247  header += 2; // skip block align (not needed)
248  avctx->bits_per_coded_sample = bytestream_get_le16(&header);
249 
250  if (avctx->bits_per_coded_sample != 16) {
251  av_log(avctx, AV_LOG_ERROR, "unsupported number of bits per sample\n");
252  return -1;
253  }
254 
255  len -= 16;
256  if (len > 0)
257  av_log(avctx, AV_LOG_INFO, "%d header bytes unparsed\n", len);
258 
259  return 0;
260 }
261 
262 static void interleave_buffer(int16_t *samples, int nchan, int blocksize,
263  int32_t **buffer)
264 {
265  int i, chan;
266  for (i=0; i<blocksize; i++)
267  for (chan=0; chan < nchan; chan++)
268  *samples++ = av_clip_int16(buffer[chan][i]);
269 }
270 
271 static const int fixed_coeffs[3][3] = {
272  { 1, 0, 0 },
273  { 2, -1, 0 },
274  { 3, -3, 1 }
275 };
276 
277 static int decode_subframe_lpc(ShortenContext *s, int command, int channel,
278  int residual_size, int32_t coffset)
279 {
280  int pred_order, sum, qshift, init_sum, i, j;
281  const int *coeffs;
282 
283  if (command == FN_QLPC) {
284  /* read/validate prediction order */
285  pred_order = get_ur_golomb_shorten(&s->gb, LPCQSIZE);
286  if (pred_order > s->nwrap) {
287  av_log(s->avctx, AV_LOG_ERROR, "invalid pred_order %d\n", pred_order);
288  return AVERROR(EINVAL);
289  }
290  /* read LPC coefficients */
291  for (i=0; i<pred_order; i++)
292  s->coeffs[i] = get_sr_golomb_shorten(&s->gb, LPCQUANT);
293  coeffs = s->coeffs;
294 
295  qshift = LPCQUANT;
296  } else {
297  /* fixed LPC coeffs */
298  pred_order = command;
299  coeffs = fixed_coeffs[pred_order-1];
300  qshift = 0;
301  }
302 
303  /* subtract offset from previous samples to use in prediction */
304  if (command == FN_QLPC && coffset)
305  for (i = -pred_order; i < 0; i++)
306  s->decoded[channel][i] -= coffset;
307 
308  /* decode residual and do LPC prediction */
309  init_sum = pred_order ? (command == FN_QLPC ? s->lpcqoffset : 0) : coffset;
310  for (i=0; i < s->blocksize; i++) {
311  sum = init_sum;
312  for (j=0; j<pred_order; j++)
313  sum += coeffs[j] * s->decoded[channel][i-j-1];
314  s->decoded[channel][i] = get_sr_golomb_shorten(&s->gb, residual_size) + (sum >> qshift);
315  }
316 
317  /* add offset to current samples */
318  if (command == FN_QLPC && coffset)
319  for (i = 0; i < s->blocksize; i++)
320  s->decoded[channel][i] += coffset;
321 
322  return 0;
323 }
324 
326 {
327  int i, ret;
328  int maxnlpc = 0;
329  /* shorten signature */
330  if (get_bits_long(&s->gb, 32) != AV_RB32("ajkg")) {
331  av_log(s->avctx, AV_LOG_ERROR, "missing shorten magic 'ajkg'\n");
332  return -1;
333  }
334 
335  s->lpcqoffset = 0;
337  s->nmean = -1;
338  s->version = get_bits(&s->gb, 8);
340 
341  s->channels = get_uint(s, CHANSIZE);
342  if (s->channels > MAX_CHANNELS) {
343  av_log(s->avctx, AV_LOG_ERROR, "too many channels: %d\n", s->channels);
344  return -1;
345  }
346  s->avctx->channels = s->channels;
347 
348  /* get blocksize if version > 0 */
349  if (s->version > 0) {
350  int skip_bytes, blocksize;
351 
352  blocksize = get_uint(s, av_log2(DEFAULT_BLOCK_SIZE));
353  if (!blocksize || blocksize > MAX_BLOCKSIZE) {
354  av_log(s->avctx, AV_LOG_ERROR, "invalid or unsupported block size: %d\n",
355  blocksize);
356  return AVERROR(EINVAL);
357  }
358  s->blocksize = blocksize;
359 
360  maxnlpc = get_uint(s, LPCQSIZE);
361  s->nmean = get_uint(s, 0);
362 
363  skip_bytes = get_uint(s, NSKIPSIZE);
364  for (i=0; i<skip_bytes; i++) {
365  skip_bits(&s->gb, 8);
366  }
367  }
368  s->nwrap = FFMAX(NWRAP, maxnlpc);
369 
370  if ((ret = allocate_buffers(s)) < 0)
371  return ret;
372 
373  if ((ret = init_offset(s)) < 0)
374  return ret;
375 
376  if (s->version > 1)
378 
380  av_log(s->avctx, AV_LOG_ERROR, "missing verbatim section at beginning of stream\n");
381  return -1;
382  }
383 
386  av_log(s->avctx, AV_LOG_ERROR, "header is wrong size: %d\n", s->header_size);
387  return -1;
388  }
389 
390  for (i=0; i<s->header_size; i++)
392 
393  if (decode_wave_header(s->avctx, s->header, s->header_size) < 0)
394  return -1;
395 
396  s->cur_chan = 0;
397  s->bitshift = 0;
398 
399  s->got_header = 1;
400 
401  return 0;
402 }
403 
404 static int shorten_decode_frame(AVCodecContext *avctx, void *data,
405  int *got_frame_ptr, AVPacket *avpkt)
406 {
407  const uint8_t *buf = avpkt->data;
408  int buf_size = avpkt->size;
409  ShortenContext *s = avctx->priv_data;
410  int i, input_buf_size = 0;
411  int ret;
412 
413  /* allocate internal bitstream buffer */
414  if(s->max_framesize == 0){
415  void *tmp_ptr;
416  s->max_framesize= 1024; // should hopefully be enough for the first header
418  s->max_framesize);
419  if (!tmp_ptr) {
420  av_log(avctx, AV_LOG_ERROR, "error allocating bitstream buffer\n");
421  return AVERROR(ENOMEM);
422  }
423  s->bitstream = tmp_ptr;
424  }
425 
426  /* append current packet data to bitstream buffer */
427  if(1 && s->max_framesize){//FIXME truncated
428  buf_size= FFMIN(buf_size, s->max_framesize - s->bitstream_size);
429  input_buf_size= buf_size;
430 
431  if(s->bitstream_index + s->bitstream_size + buf_size > s->allocated_bitstream_size){
432  memmove(s->bitstream, &s->bitstream[s->bitstream_index], s->bitstream_size);
433  s->bitstream_index=0;
434  }
435  if (buf)
436  memcpy(&s->bitstream[s->bitstream_index + s->bitstream_size], buf, buf_size);
437  buf= &s->bitstream[s->bitstream_index];
438  buf_size += s->bitstream_size;
439  s->bitstream_size= buf_size;
440 
441  /* do not decode until buffer has at least max_framesize bytes or
442  the end of the file has been reached */
443  if (buf_size < s->max_framesize && avpkt->data) {
444  *got_frame_ptr = 0;
445  return input_buf_size;
446  }
447  }
448  /* init and position bitstream reader */
449  init_get_bits(&s->gb, buf, buf_size*8);
450  skip_bits(&s->gb, s->bitindex);
451 
452  /* process header or next subblock */
453  if (!s->got_header) {
454  if ((ret = read_header(s)) < 0)
455  return ret;
456  *got_frame_ptr = 0;
457  goto finish_frame;
458  }
459 
460  /* if quit command was read previously, don't decode anything */
461  if (s->got_quit_command) {
462  *got_frame_ptr = 0;
463  return avpkt->size;
464  }
465 
466  s->cur_chan = 0;
467  while (s->cur_chan < s->channels) {
468  int cmd;
469  int len;
470 
471  if (get_bits_left(&s->gb) < 3+FNSIZE) {
472  *got_frame_ptr = 0;
473  break;
474  }
475 
476  cmd = get_ur_golomb_shorten(&s->gb, FNSIZE);
477 
478  if (cmd > FN_VERBATIM) {
479  av_log(avctx, AV_LOG_ERROR, "unknown shorten function %d\n", cmd);
480  *got_frame_ptr = 0;
481  break;
482  }
483 
484  if (!is_audio_command[cmd]) {
485  /* process non-audio command */
486  switch (cmd) {
487  case FN_VERBATIM:
489  while (len--) {
491  }
492  break;
493  case FN_BITSHIFT:
495  break;
496  case FN_BLOCKSIZE: {
497  int blocksize = get_uint(s, av_log2(s->blocksize));
498  if (blocksize > s->blocksize) {
499  av_log(avctx, AV_LOG_ERROR, "Increasing block size is not supported\n");
500  return AVERROR_PATCHWELCOME;
501  }
502  if (!blocksize || blocksize > MAX_BLOCKSIZE) {
503  av_log(avctx, AV_LOG_ERROR, "invalid or unsupported "
504  "block size: %d\n", blocksize);
505  return AVERROR(EINVAL);
506  }
507  s->blocksize = blocksize;
508  break;
509  }
510  case FN_QUIT:
511  s->got_quit_command = 1;
512  break;
513  }
514  if (cmd == FN_BLOCKSIZE || cmd == FN_QUIT) {
515  *got_frame_ptr = 0;
516  break;
517  }
518  } else {
519  /* process audio command */
520  int residual_size = 0;
521  int channel = s->cur_chan;
522  int32_t coffset;
523 
524  /* get Rice code for residual decoding */
525  if (cmd != FN_ZERO) {
526  residual_size = get_ur_golomb_shorten(&s->gb, ENERGYSIZE);
527  /* this is a hack as version 0 differed in defintion of get_sr_golomb_shorten */
528  if (s->version == 0)
529  residual_size--;
530  }
531 
532  /* calculate sample offset using means from previous blocks */
533  if (s->nmean == 0)
534  coffset = s->offset[channel][0];
535  else {
536  int32_t sum = (s->version < 2) ? 0 : s->nmean / 2;
537  for (i=0; i<s->nmean; i++)
538  sum += s->offset[channel][i];
539  coffset = sum / s->nmean;
540  if (s->version >= 2)
541  coffset >>= FFMIN(1, s->bitshift);
542  }
543 
544  /* decode samples for this channel */
545  if (cmd == FN_ZERO) {
546  for (i=0; i<s->blocksize; i++)
547  s->decoded[channel][i] = 0;
548  } else {
549  if ((ret = decode_subframe_lpc(s, cmd, channel, residual_size, coffset)) < 0)
550  return ret;
551  }
552 
553  /* update means with info from the current block */
554  if (s->nmean > 0) {
555  int32_t sum = (s->version < 2) ? 0 : s->blocksize / 2;
556  for (i=0; i<s->blocksize; i++)
557  sum += s->decoded[channel][i];
558 
559  for (i=1; i<s->nmean; i++)
560  s->offset[channel][i-1] = s->offset[channel][i];
561 
562  if (s->version < 2)
563  s->offset[channel][s->nmean - 1] = sum / s->blocksize;
564  else
565  s->offset[channel][s->nmean - 1] = (sum / s->blocksize) << s->bitshift;
566  }
567 
568  /* copy wrap samples for use with next block */
569  for (i=-s->nwrap; i<0; i++)
570  s->decoded[channel][i] = s->decoded[channel][i + s->blocksize];
571 
572  /* shift samples to add in unused zero bits which were removed
573  during encoding */
574  fix_bitshift(s, s->decoded[channel]);
575 
576  /* if this is the last channel in the block, output the samples */
577  s->cur_chan++;
578  if (s->cur_chan == s->channels) {
579  /* get output buffer */
580  s->frame.nb_samples = s->blocksize;
581  if ((ret = avctx->get_buffer(avctx, &s->frame)) < 0) {
582  av_log(avctx, AV_LOG_ERROR, "get_buffer() failed\n");
583  return ret;
584  }
585  /* interleave output */
586  interleave_buffer((int16_t *)s->frame.data[0], s->channels,
587  s->blocksize, s->decoded);
588 
589  *got_frame_ptr = 1;
590  *(AVFrame *)data = s->frame;
591  }
592  }
593  }
594  if (s->cur_chan < s->channels)
595  *got_frame_ptr = 0;
596 
598  s->bitindex = get_bits_count(&s->gb) - 8*((get_bits_count(&s->gb))/8);
599  i= (get_bits_count(&s->gb))/8;
600  if (i > buf_size) {
601  av_log(s->avctx, AV_LOG_ERROR, "overread: %d\n", i - buf_size);
602  s->bitstream_size=0;
603  s->bitstream_index=0;
604  return -1;
605  }
606  if (s->bitstream_size) {
607  s->bitstream_index += i;
608  s->bitstream_size -= i;
609  return input_buf_size;
610  } else
611  return i;
612 }
613 
615 {
616  ShortenContext *s = avctx->priv_data;
617  int i;
618 
619  for (i = 0; i < s->channels; i++) {
620  s->decoded[i] = NULL;
621  av_freep(&s->decoded_base[i]);
622  av_freep(&s->offset[i]);
623  }
624  av_freep(&s->bitstream);
625  av_freep(&s->coeffs);
626 
627  return 0;
628 }
629 
631  .name = "shorten",
632  .type = AVMEDIA_TYPE_AUDIO,
633  .id = CODEC_ID_SHORTEN,
634  .priv_data_size = sizeof(ShortenContext),
638  .capabilities = CODEC_CAP_DELAY | CODEC_CAP_DR1,
639  .long_name= NULL_IF_CONFIG_SMALL("Shorten"),
640 };