FFmpeg
dpcm.c
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1 /*
2  * Assorted DPCM codecs
3  * Copyright (c) 2003 The FFmpeg project
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  * Assorted DPCM (differential pulse code modulation) audio codecs
25  * by Mike Melanson (melanson@pcisys.net)
26  * Xan DPCM decoder by Mario Brito (mbrito@student.dei.uc.pt)
27  * for more information on the specific data formats, visit:
28  * http://www.pcisys.net/~melanson/codecs/simpleaudio.html
29  * SOL DPCMs implemented by Konstantin Shishkov
30  *
31  * Note about using the Xan DPCM decoder: Xan DPCM is used in AVI files
32  * found in the Wing Commander IV computer game. These AVI files contain
33  * WAVEFORMAT headers which report the audio format as 0x01: raw PCM.
34  * Clearly incorrect. To detect Xan DPCM, you will probably have to
35  * special-case your AVI demuxer to use Xan DPCM if the file uses 'Xxan'
36  * (Xan video) for its video codec. Alternately, such AVI files also contain
37  * the fourcc 'Axan' in the 'auds' chunk of the AVI header.
38  */
39 
40 #include "avcodec.h"
41 #include "bytestream.h"
42 #include "codec_internal.h"
43 #include "decode.h"
44 #include "mathops.h"
45 
46 #include "libavutil/attributes.h"
47 
48 typedef struct DPCMContext {
49  int16_t array[256];
50  int sample[2]; ///< previous sample (for SOL_DPCM and WADY_DPCM)
51  int scale; ///< scale for WADY_DPCM
52  const int8_t *sol_table; ///< delta table for SOL_DPCM
53 } DPCMContext;
54 
55 static const int32_t derf_steps[96] = {
56  0, 1, 2, 3, 4, 5, 6, 7,
57  8, 9, 10, 11, 12, 13, 14, 16,
58  17, 19, 21, 23, 25, 28, 31, 34,
59  37, 41, 45, 50, 55, 60, 66, 73,
60  80, 88, 97, 107, 118, 130, 143, 157,
61  173, 190, 209, 230, 253, 279, 307, 337,
62  371, 408, 449, 494, 544, 598, 658, 724,
63  796, 876, 963, 1060, 1166, 1282, 1411, 1552,
64  1707, 1878, 2066, 2272, 2499, 2749, 3024, 3327,
65  3660, 4026, 4428, 4871, 5358, 5894, 6484, 7132,
66  7845, 8630, 9493, 10442, 11487, 12635, 13899, 15289,
67  16818, 18500, 20350, 22385, 24623, 27086, 29794, 32767,
68 };
69 
70 static const int16_t interplay_delta_table[] = {
71  0, 1, 2, 3, 4, 5, 6, 7,
72  8, 9, 10, 11, 12, 13, 14, 15,
73  16, 17, 18, 19, 20, 21, 22, 23,
74  24, 25, 26, 27, 28, 29, 30, 31,
75  32, 33, 34, 35, 36, 37, 38, 39,
76  40, 41, 42, 43, 47, 51, 56, 61,
77  66, 72, 79, 86, 94, 102, 112, 122,
78  133, 145, 158, 173, 189, 206, 225, 245,
79  267, 292, 318, 348, 379, 414, 452, 493,
80  538, 587, 640, 699, 763, 832, 908, 991,
81  1081, 1180, 1288, 1405, 1534, 1673, 1826, 1993,
82  2175, 2373, 2590, 2826, 3084, 3365, 3672, 4008,
83  4373, 4772, 5208, 5683, 6202, 6767, 7385, 8059,
84  8794, 9597, 10472, 11428, 12471, 13609, 14851, 16206,
85  17685, 19298, 21060, 22981, 25078, 27367, 29864, 32589,
86  -29973, -26728, -23186, -19322, -15105, -10503, -5481, -1,
87  1, 1, 5481, 10503, 15105, 19322, 23186, 26728,
88  29973, -32589, -29864, -27367, -25078, -22981, -21060, -19298,
89  -17685, -16206, -14851, -13609, -12471, -11428, -10472, -9597,
90  -8794, -8059, -7385, -6767, -6202, -5683, -5208, -4772,
91  -4373, -4008, -3672, -3365, -3084, -2826, -2590, -2373,
92  -2175, -1993, -1826, -1673, -1534, -1405, -1288, -1180,
93  -1081, -991, -908, -832, -763, -699, -640, -587,
94  -538, -493, -452, -414, -379, -348, -318, -292,
95  -267, -245, -225, -206, -189, -173, -158, -145,
96  -133, -122, -112, -102, -94, -86, -79, -72,
97  -66, -61, -56, -51, -47, -43, -42, -41,
98  -40, -39, -38, -37, -36, -35, -34, -33,
99  -32, -31, -30, -29, -28, -27, -26, -25,
100  -24, -23, -22, -21, -20, -19, -18, -17,
101  -16, -15, -14, -13, -12, -11, -10, -9,
102  -8, -7, -6, -5, -4, -3, -2, -1
103 
104 };
105 
106 static const int8_t sol_table_old[16] = {
107  0x0, 0x1, 0x2, 0x3, 0x6, 0xA, 0xF, 0x15,
108  -0x15, -0xF, -0xA, -0x6, -0x3, -0x2, -0x1, 0x0
109 };
110 
111 static const int8_t sol_table_new[16] = {
112  0x0, 0x1, 0x2, 0x3, 0x6, 0xA, 0xF, 0x15,
113  0x0, -0x1, -0x2, -0x3, -0x6, -0xA, -0xF, -0x15
114 };
115 
116 static const int16_t sol_table_16[128] = {
117  0x000, 0x008, 0x010, 0x020, 0x030, 0x040, 0x050, 0x060, 0x070, 0x080,
118  0x090, 0x0A0, 0x0B0, 0x0C0, 0x0D0, 0x0E0, 0x0F0, 0x100, 0x110, 0x120,
119  0x130, 0x140, 0x150, 0x160, 0x170, 0x180, 0x190, 0x1A0, 0x1B0, 0x1C0,
120  0x1D0, 0x1E0, 0x1F0, 0x200, 0x208, 0x210, 0x218, 0x220, 0x228, 0x230,
121  0x238, 0x240, 0x248, 0x250, 0x258, 0x260, 0x268, 0x270, 0x278, 0x280,
122  0x288, 0x290, 0x298, 0x2A0, 0x2A8, 0x2B0, 0x2B8, 0x2C0, 0x2C8, 0x2D0,
123  0x2D8, 0x2E0, 0x2E8, 0x2F0, 0x2F8, 0x300, 0x308, 0x310, 0x318, 0x320,
124  0x328, 0x330, 0x338, 0x340, 0x348, 0x350, 0x358, 0x360, 0x368, 0x370,
125  0x378, 0x380, 0x388, 0x390, 0x398, 0x3A0, 0x3A8, 0x3B0, 0x3B8, 0x3C0,
126  0x3C8, 0x3D0, 0x3D8, 0x3E0, 0x3E8, 0x3F0, 0x3F8, 0x400, 0x440, 0x480,
127  0x4C0, 0x500, 0x540, 0x580, 0x5C0, 0x600, 0x640, 0x680, 0x6C0, 0x700,
128  0x740, 0x780, 0x7C0, 0x800, 0x900, 0xA00, 0xB00, 0xC00, 0xD00, 0xE00,
129  0xF00, 0x1000, 0x1400, 0x1800, 0x1C00, 0x2000, 0x3000, 0x4000
130 };
131 
132 static const int16_t wady_table[128] = {
133  0, 2, 4, 6, 8, 10, 12, 15,
134  18, 21, 24, 28, 32, 36, 40, 44,
135  49, 54, 59, 64, 70, 76, 82, 88,
136  95, 102, 109, 116, 124, 132, 140, 148,
137  160, 170, 180, 190, 200, 210, 220, 230,
138  240, 255, 270, 285, 300, 320, 340, 360,
139  380, 400, 425, 450, 475, 500, 525, 550,
140  580, 610, 650, 700, 750, 800, 900, 1000,
141  -0, -2, -4, -6, -8, -10, -12, -15,
142  -18, -21, -24, -28, -32, -36, -40, -44,
143  -49, -54, -59, -64, -70, -76, -82, -88,
144  -95, -102,-109,-116,-124,-132,-140,-148,
145  -160,-170,-180,-190,-200,-210,-220,-230,
146  -240,-255,-270,-285,-300,-320,-340,-360,
147  -380,-400,-425,-450,-475,-500,-525,-550,
148  -580,-610,-650,-700,-750,-800,-900,-1000,
149 };
150 
151 static av_cold int dpcm_decode_init(AVCodecContext *avctx)
152 {
153  DPCMContext *s = avctx->priv_data;
154  int i;
155 
156  if (avctx->ch_layout.nb_channels < 1 || avctx->ch_layout.nb_channels > 2) {
157  av_log(avctx, AV_LOG_ERROR, "invalid number of channels\n");
158  return AVERROR(EINVAL);
159  }
160 
161  s->sample[0] = s->sample[1] = 0;
162 
163  switch (avctx->codec->id) {
164 
165  case AV_CODEC_ID_ROQ_DPCM:
166  /* initialize square table */
167  for (i = 0; i < 128; i++) {
168  int16_t square = i * i;
169  s->array[i ] = square;
170  s->array[i + 128] = -square;
171  }
172  break;
173 
174  case AV_CODEC_ID_SOL_DPCM:
175  switch(avctx->codec_tag){
176  case 1:
177  s->sol_table = sol_table_old;
178  s->sample[0] = s->sample[1] = 0x80;
179  break;
180  case 2:
181  s->sol_table = sol_table_new;
182  s->sample[0] = s->sample[1] = 0x80;
183  break;
184  case 3:
185  break;
186  default:
187  av_log(avctx, AV_LOG_ERROR, "Unknown SOL subcodec\n");
188  return -1;
189  }
190  break;
191 
192  case AV_CODEC_ID_SDX2_DPCM:
193  for (i = -128; i < 128; i++) {
194  int16_t square = i * i * 2;
195  s->array[i+128] = i < 0 ? -square: square;
196  }
197  break;
198 
199  case AV_CODEC_ID_CBD2_DPCM:
200  for (i = -128; i < 128; i++) {
201  int16_t cube = (i * i * i) / 64;
202  s->array[i+128] = cube;
203  }
204  break;
205 
206  case AV_CODEC_ID_GREMLIN_DPCM: {
207  int delta = 0;
208  int code = 64;
209  int step = 45;
210 
211  s->array[0] = 0;
212  for (i = 0; i < 127; i++) {
213  delta += (code >> 5);
214  code += step;
215  step += 2;
216 
217  s->array[i*2 + 1] = delta;
218  s->array[i*2 + 2] = -delta;
219  }
220  s->array[255] = delta + (code >> 5);
221  }
222  break;
223 
224  case AV_CODEC_ID_WADY_DPCM:
225  s->scale = (avctx->extradata && avctx->extradata_size > 0) ? avctx->extradata[0] : 1;
226  break;
227 
228  default:
229  break;
230  }
231 
232  if (avctx->codec->id == AV_CODEC_ID_SOL_DPCM && avctx->codec_tag != 3)
233  avctx->sample_fmt = AV_SAMPLE_FMT_U8;
234  else
235  avctx->sample_fmt = AV_SAMPLE_FMT_S16;
236 
237  return 0;
238 }
239 
240 
241 static int dpcm_decode_frame(AVCodecContext *avctx, AVFrame *frame,
242  int *got_frame_ptr, AVPacket *avpkt)
243 {
244  int buf_size = avpkt->size;
245  DPCMContext *s = avctx->priv_data;
246  int out = 0, ret;
247  int predictor[2];
248  int ch = 0;
249  int stereo = avctx->ch_layout.nb_channels - 1;
250  int16_t *output_samples, *samples_end;
251  GetByteContext gb;
252 
253  if (stereo && (buf_size & 1))
254  buf_size--;
255  bytestream2_init(&gb, avpkt->data, buf_size);
256 
257  /* calculate output size */
258  switch(avctx->codec->id) {
259  case AV_CODEC_ID_ROQ_DPCM:
260  out = buf_size - 8;
261  break;
262  case AV_CODEC_ID_INTERPLAY_DPCM:
263  out = buf_size - 6 - avctx->ch_layout.nb_channels;
264  break;
265  case AV_CODEC_ID_XAN_DPCM:
266  out = buf_size - 2 * avctx->ch_layout.nb_channels;
267  break;
268  case AV_CODEC_ID_SOL_DPCM:
269  if (avctx->codec_tag != 3)
270  out = buf_size * 2;
271  else
272  out = buf_size;
273  break;
274  case AV_CODEC_ID_WADY_DPCM:
275  case AV_CODEC_ID_DERF_DPCM:
276  case AV_CODEC_ID_GREMLIN_DPCM:
277  case AV_CODEC_ID_CBD2_DPCM:
278  case AV_CODEC_ID_SDX2_DPCM:
279  out = buf_size;
280  break;
281  }
282  if (out <= 0) {
283  av_log(avctx, AV_LOG_ERROR, "packet is too small\n");
284  return AVERROR(EINVAL);
285  }
286  if (out % avctx->ch_layout.nb_channels) {
287  av_log(avctx, AV_LOG_WARNING, "channels have differing number of samples\n");
288  }
289 
290  /* get output buffer */
291  frame->nb_samples = (out + avctx->ch_layout.nb_channels - 1) / avctx->ch_layout.nb_channels;
292  if ((ret = ff_get_buffer(avctx, frame, 0)) < 0)
293  return ret;
294  output_samples = (int16_t *)frame->data[0];
295  samples_end = output_samples + out;
296 
297  switch(avctx->codec->id) {
298 
299  case AV_CODEC_ID_ROQ_DPCM:
300  bytestream2_skipu(&gb, 6);
301 
302  if (stereo) {
303  predictor[1] = sign_extend(bytestream2_get_byteu(&gb) << 8, 16);
304  predictor[0] = sign_extend(bytestream2_get_byteu(&gb) << 8, 16);
305  } else {
306  predictor[0] = sign_extend(bytestream2_get_le16u(&gb), 16);
307  }
308 
309  /* decode the samples */
310  while (output_samples < samples_end) {
311  predictor[ch] += s->array[bytestream2_get_byteu(&gb)];
312  predictor[ch] = av_clip_int16(predictor[ch]);
313  *output_samples++ = predictor[ch];
314 
315  /* toggle channel */
316  ch ^= stereo;
317  }
318  break;
319 
320  case AV_CODEC_ID_INTERPLAY_DPCM:
321  bytestream2_skipu(&gb, 6); /* skip over the stream mask and stream length */
322 
323  for (ch = 0; ch < avctx->ch_layout.nb_channels; ch++) {
324  predictor[ch] = sign_extend(bytestream2_get_le16u(&gb), 16);
325  *output_samples++ = predictor[ch];
326  }
327 
328  ch = 0;
329  while (output_samples < samples_end) {
330  predictor[ch] += interplay_delta_table[bytestream2_get_byteu(&gb)];
331  predictor[ch] = av_clip_int16(predictor[ch]);
332  *output_samples++ = predictor[ch];
333 
334  /* toggle channel */
335  ch ^= stereo;
336  }
337  break;
338 
339  case AV_CODEC_ID_XAN_DPCM:
340  {
341  int shift[2] = { 4, 4 };
342 
343  for (ch = 0; ch < avctx->ch_layout.nb_channels; ch++)
344  predictor[ch] = sign_extend(bytestream2_get_le16u(&gb), 16);
345 
346  ch = 0;
347  while (output_samples < samples_end) {
348  int diff = bytestream2_get_byteu(&gb);
349  int n = diff & 3;
350 
351  if (n == 3)
352  shift[ch]++;
353  else
354  shift[ch] -= (2 * n);
355  diff = sign_extend((diff &~ 3) << 8, 16);
356 
357  /* saturate the shifter to 0..31 */
358  shift[ch] = av_clip_uintp2(shift[ch], 5);
359 
360  diff >>= shift[ch];
361  predictor[ch] += diff;
362 
363  predictor[ch] = av_clip_int16(predictor[ch]);
364  *output_samples++ = predictor[ch];
365 
366  /* toggle channel */
367  ch ^= stereo;
368  }
369  break;
370  }
371  case AV_CODEC_ID_SOL_DPCM:
372  if (avctx->codec_tag != 3) {
373  uint8_t *output_samples_u8 = frame->data[0],
374  *samples_end_u8 = output_samples_u8 + out;
375  while (output_samples_u8 < samples_end_u8) {
376  int n = bytestream2_get_byteu(&gb);
377 
378  s->sample[0] += s->sol_table[n >> 4];
379  s->sample[0] = av_clip_uint8(s->sample[0]);
380  *output_samples_u8++ = s->sample[0];
381 
382  s->sample[stereo] += s->sol_table[n & 0x0F];
383  s->sample[stereo] = av_clip_uint8(s->sample[stereo]);
384  *output_samples_u8++ = s->sample[stereo];
385  }
386  } else {
387  while (output_samples < samples_end) {
388  int n = bytestream2_get_byteu(&gb);
389  if (n & 0x80) s->sample[ch] -= sol_table_16[n & 0x7F];
390  else s->sample[ch] += sol_table_16[n & 0x7F];
391  s->sample[ch] = av_clip_int16(s->sample[ch]);
392  *output_samples++ = s->sample[ch];
393  /* toggle channel */
394  ch ^= stereo;
395  }
396  }
397  break;
398 
399  case AV_CODEC_ID_CBD2_DPCM:
400  case AV_CODEC_ID_SDX2_DPCM:
401  while (output_samples < samples_end) {
402  int8_t n = bytestream2_get_byteu(&gb);
403 
404  if (!(n & 1))
405  s->sample[ch] = 0;
406  s->sample[ch] += s->array[n + 128];
407  s->sample[ch] = av_clip_int16(s->sample[ch]);
408  *output_samples++ = s->sample[ch];
409  ch ^= stereo;
410  }
411  break;
412 
413  case AV_CODEC_ID_GREMLIN_DPCM: {
414  int idx = 0;
415 
416  while (output_samples < samples_end) {
417  uint8_t n = bytestream2_get_byteu(&gb);
418 
419  *output_samples++ = s->sample[idx] += (unsigned)s->array[n];
420  idx ^= 1;
421  }
422  }
423  break;
424 
425  case AV_CODEC_ID_DERF_DPCM: {
426  int idx = 0;
427 
428  while (output_samples < samples_end) {
429  uint8_t n = bytestream2_get_byteu(&gb);
430  int index = FFMIN(n & 0x7f, 95);
431 
432  s->sample[idx] += (n & 0x80 ? -1: 1) * derf_steps[index];
433  s->sample[idx] = av_clip_int16(s->sample[idx]);
434  *output_samples++ = s->sample[idx];
435  idx ^= stereo;
436  }
437  }
438  break;
439 
440  case AV_CODEC_ID_WADY_DPCM: {
441  int idx = 0;
442 
443  while (output_samples < samples_end) {
444  const uint8_t n = bytestream2_get_byteu(&gb);
445 
446  if (n & 0x80)
447  s->sample[idx] = sign_extend((n & 0x7f) << 9, 16);
448  else
449  s->sample[idx] += s->scale * (unsigned)wady_table[n & 0x7f];
450  *output_samples++ = av_clip_int16(s->sample[idx]);
451  idx ^= stereo;
452  }
453  }
454  break;
455  }
456 
457  *got_frame_ptr = 1;
458 
459  return avpkt->size;
460 }
461 
462 static av_cold void dpcm_flush(AVCodecContext *avctx)
463 {
464  DPCMContext *s = avctx->priv_data;
465 
466  s->sample[0] = s->sample[1] = 0;
467 }
468 
469 #define DPCM_DECODER(id_, name_, long_name_) \
470 const FFCodec ff_ ## name_ ## _decoder = { \
471  .p.name = #name_, \
472  CODEC_LONG_NAME(long_name_), \
473  .p.type = AVMEDIA_TYPE_AUDIO, \
474  .p.id = id_, \
475  .p.capabilities = AV_CODEC_CAP_DR1, \
476  .priv_data_size = sizeof(DPCMContext), \
477  .init = dpcm_decode_init, \
478  .flush = dpcm_flush, \
479  FF_CODEC_DECODE_CB(dpcm_decode_frame), \
480 }
481 
482 DPCM_DECODER(AV_CODEC_ID_CBD2_DPCM, cbd2_dpcm, "DPCM Cuberoot-Delta-Exact");
483 DPCM_DECODER(AV_CODEC_ID_DERF_DPCM, derf_dpcm, "DPCM Xilam DERF");
484 DPCM_DECODER(AV_CODEC_ID_GREMLIN_DPCM, gremlin_dpcm, "DPCM Gremlin");
485 DPCM_DECODER(AV_CODEC_ID_INTERPLAY_DPCM, interplay_dpcm, "DPCM Interplay");
486 DPCM_DECODER(AV_CODEC_ID_ROQ_DPCM, roq_dpcm, "DPCM id RoQ");
487 DPCM_DECODER(AV_CODEC_ID_SDX2_DPCM, sdx2_dpcm, "DPCM Squareroot-Delta-Exact");
488 DPCM_DECODER(AV_CODEC_ID_SOL_DPCM, sol_dpcm, "DPCM Sol");
489 DPCM_DECODER(AV_CODEC_ID_XAN_DPCM, xan_dpcm, "DPCM Xan");
490 DPCM_DECODER(AV_CODEC_ID_WADY_DPCM, wady_dpcm, "DPCM Marble WADY");
sol_table_16
static const int16_t sol_table_16[128]
Definition: dpcm.c:116
AV_LOG_WARNING
#define AV_LOG_WARNING
Something somehow does not look correct.
Definition: log.h:216
DPCM_DECODER
#define DPCM_DECODER(id_, name_, long_name_)
Definition: dpcm.c:469
DPCMContext
Definition: dpcm.c:48
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
out
FILE * out
Definition: movenc.c:55
GetByteContext
Definition: bytestream.h:33
av_clip_uintp2
#define av_clip_uintp2
Definition: common.h:124
bytestream2_skipu
static av_always_inline void bytestream2_skipu(GetByteContext *g, unsigned int size)
Definition: bytestream.h:174
AVFrame
This structure describes decoded (raw) audio or video data.
Definition: frame.h:427
step
trying all byte sequences megabyte in length and selecting the best looking sequence will yield cases to try But a word about which is also called distortion Distortion can be quantified by almost any quality measurement one chooses the sum of squared differences is used but more complex methods that consider psychovisual effects can be used as well It makes no difference in this discussion First step
Definition: rate_distortion.txt:58
interplay_delta_table
static const int16_t interplay_delta_table[]
Definition: dpcm.c:70
DPCMContext::scale
int scale
scale for WADY_DPCM
Definition: dpcm.c:51
AVPacket::data
uint8_t * data
Definition: packet.h:558
dpcm_decode_init
static av_cold int dpcm_decode_init(AVCodecContext *avctx)
Definition: dpcm.c:151
AV_CODEC_ID_SOL_DPCM
@ AV_CODEC_ID_SOL_DPCM
Definition: codec_id.h:443
AVChannelLayout::nb_channels
int nb_channels
Number of channels in this layout.
Definition: channel_layout.h:329
sol_table_new
static const int8_t sol_table_new[16]
Definition: dpcm.c:111
AVCodecContext::codec
const struct AVCodec * codec
Definition: avcodec.h:440
AVCodecContext::ch_layout
AVChannelLayout ch_layout
Audio channel layout.
Definition: avcodec.h:1039
dpcm_decode_frame
static int dpcm_decode_frame(AVCodecContext *avctx, AVFrame *frame, int *got_frame_ptr, AVPacket *avpkt)
Definition: dpcm.c:241
AV_CODEC_ID_XAN_DPCM
@ AV_CODEC_ID_XAN_DPCM
Definition: codec_id.h:442
AV_LOG_ERROR
#define AV_LOG_ERROR
Something went wrong and cannot losslessly be recovered.
Definition: log.h:210
av_cold
#define av_cold
Definition: attributes.h:90
AVCodecContext::extradata_size
int extradata_size
Definition: avcodec.h:515
s
#define s(width, name)
Definition: cbs_vp9.c:198
AV_CODEC_ID_WADY_DPCM
@ AV_CODEC_ID_WADY_DPCM
Definition: codec_id.h:447
DPCMContext::sample
int sample[2]
previous sample (for SOL_DPCM and WADY_DPCM)
Definition: dpcm.c:50
decode.h
AV_CODEC_ID_DERF_DPCM
@ AV_CODEC_ID_DERF_DPCM
Definition: codec_id.h:446
sol_table_old
static const int8_t sol_table_old[16]
Definition: dpcm.c:106
if
if(ret)
Definition: filter_design.txt:179
av_clip_int16
#define av_clip_int16
Definition: common.h:115
AV_CODEC_ID_INTERPLAY_DPCM
@ AV_CODEC_ID_INTERPLAY_DPCM
Definition: codec_id.h:441
derf_steps
static const int32_t derf_steps[96]
Definition: dpcm.c:55
mathops.h
DPCMContext::sol_table
const int8_t * sol_table
delta table for SOL_DPCM
Definition: dpcm.c:52
AV_CODEC_ID_ROQ_DPCM
@ AV_CODEC_ID_ROQ_DPCM
Definition: codec_id.h:440
index
int index
Definition: gxfenc.c:90
ff_get_buffer
int ff_get_buffer(AVCodecContext *avctx, AVFrame *frame, int flags)
Get a buffer for a frame.
Definition: decode.c:1720
AVPacket::size
int size
Definition: packet.h:559
codec_internal.h
shift
static int shift(int a, int b)
Definition: bonk.c:261
AVCodecContext::sample_fmt
enum AVSampleFormat sample_fmt
audio sample format
Definition: avcodec.h:1031
diff
static av_always_inline int diff(const struct color_info *a, const struct color_info *b, const int trans_thresh)
Definition: vf_paletteuse.c:166
attributes.h
predictor
static void predictor(uint8_t *src, ptrdiff_t size)
Definition: exrenc.c:170
DPCMContext::array
int16_t array[256]
Definition: dpcm.c:49
dpcm_flush
static av_cold void dpcm_flush(AVCodecContext *avctx)
Definition: dpcm.c:462
AVCodec::id
enum AVCodecID id
Definition: codec.h:186
i
#define i(width, name, range_min, range_max)
Definition: cbs_h2645.c:256
code
and forward the test the status of outputs and forward it to the corresponding return FFERROR_NOT_READY If the filters stores internally one or a few frame for some it can consider them to be part of the FIFO and delay acknowledging a status change accordingly Example code
Definition: filter_design.txt:178
AVCodecContext::extradata
uint8_t * extradata
Out-of-band global headers that may be used by some codecs.
Definition: avcodec.h:514
AV_SAMPLE_FMT_U8
@ AV_SAMPLE_FMT_U8
unsigned 8 bits
Definition: samplefmt.h:57
delta
float delta
Definition: vorbis_enc_data.h:430
FFMIN
#define FFMIN(a, b)
Definition: macros.h:49
AV_SAMPLE_FMT_S16
@ AV_SAMPLE_FMT_S16
signed 16 bits
Definition: samplefmt.h:58
avcodec.h
ret
ret
Definition: filter_design.txt:187
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:265
square
static int square(int x)
Definition: roqvideoenc.c:196
AV_CODEC_ID_CBD2_DPCM
@ AV_CODEC_ID_CBD2_DPCM
Definition: codec_id.h:448
AVCodecContext
main external API structure.
Definition: avcodec.h:431
sign_extend
static av_const int sign_extend(int val, unsigned bits)
Definition: mathops.h:132
av_clip_uint8
#define av_clip_uint8
Definition: common.h:106
wady_table
static const int16_t wady_table[128]
Definition: dpcm.c:132
AV_CODEC_ID_SDX2_DPCM
@ AV_CODEC_ID_SDX2_DPCM
Definition: codec_id.h:444
AVCodecContext::codec_tag
unsigned int codec_tag
fourcc (LSB first, so "ABCD" -> ('D'<<24) + ('C'<<16) + ('B'<<8) + 'A').
Definition: avcodec.h:456
AVPacket
This structure stores compressed data.
Definition: packet.h:535
AVCodecContext::priv_data
void * priv_data
Definition: avcodec.h:458
int32_t
int32_t
Definition: audioconvert.c:56
bytestream.h
bytestream2_init
static av_always_inline void bytestream2_init(GetByteContext *g, const uint8_t *buf, int buf_size)
Definition: bytestream.h:137
av_log
#define av_log(a,...)
Definition: tableprint_vlc.h:27
AV_CODEC_ID_GREMLIN_DPCM
@ AV_CODEC_ID_GREMLIN_DPCM
Definition: codec_id.h:445
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