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 typedef struct DPCMContext {
47  int16_t array[256];
48  int sample[2]; ///< previous sample (for SOL_DPCM and WADY_DPCM)
49  int scale; ///< scale for WADY_DPCM
50  const int8_t *sol_table; ///< delta table for SOL_DPCM
51 } DPCMContext;
52 
53 static const int32_t derf_steps[96] = {
54  0, 1, 2, 3, 4, 5, 6, 7,
55  8, 9, 10, 11, 12, 13, 14, 16,
56  17, 19, 21, 23, 25, 28, 31, 34,
57  37, 41, 45, 50, 55, 60, 66, 73,
58  80, 88, 97, 107, 118, 130, 143, 157,
59  173, 190, 209, 230, 253, 279, 307, 337,
60  371, 408, 449, 494, 544, 598, 658, 724,
61  796, 876, 963, 1060, 1166, 1282, 1411, 1552,
62  1707, 1878, 2066, 2272, 2499, 2749, 3024, 3327,
63  3660, 4026, 4428, 4871, 5358, 5894, 6484, 7132,
64  7845, 8630, 9493, 10442, 11487, 12635, 13899, 15289,
65  16818, 18500, 20350, 22385, 24623, 27086, 29794, 32767,
66 };
67 
68 static const int16_t interplay_delta_table[] = {
69  0, 1, 2, 3, 4, 5, 6, 7,
70  8, 9, 10, 11, 12, 13, 14, 15,
71  16, 17, 18, 19, 20, 21, 22, 23,
72  24, 25, 26, 27, 28, 29, 30, 31,
73  32, 33, 34, 35, 36, 37, 38, 39,
74  40, 41, 42, 43, 47, 51, 56, 61,
75  66, 72, 79, 86, 94, 102, 112, 122,
76  133, 145, 158, 173, 189, 206, 225, 245,
77  267, 292, 318, 348, 379, 414, 452, 493,
78  538, 587, 640, 699, 763, 832, 908, 991,
79  1081, 1180, 1288, 1405, 1534, 1673, 1826, 1993,
80  2175, 2373, 2590, 2826, 3084, 3365, 3672, 4008,
81  4373, 4772, 5208, 5683, 6202, 6767, 7385, 8059,
82  8794, 9597, 10472, 11428, 12471, 13609, 14851, 16206,
83  17685, 19298, 21060, 22981, 25078, 27367, 29864, 32589,
84  -29973, -26728, -23186, -19322, -15105, -10503, -5481, -1,
85  1, 1, 5481, 10503, 15105, 19322, 23186, 26728,
86  29973, -32589, -29864, -27367, -25078, -22981, -21060, -19298,
87  -17685, -16206, -14851, -13609, -12471, -11428, -10472, -9597,
88  -8794, -8059, -7385, -6767, -6202, -5683, -5208, -4772,
89  -4373, -4008, -3672, -3365, -3084, -2826, -2590, -2373,
90  -2175, -1993, -1826, -1673, -1534, -1405, -1288, -1180,
91  -1081, -991, -908, -832, -763, -699, -640, -587,
92  -538, -493, -452, -414, -379, -348, -318, -292,
93  -267, -245, -225, -206, -189, -173, -158, -145,
94  -133, -122, -112, -102, -94, -86, -79, -72,
95  -66, -61, -56, -51, -47, -43, -42, -41,
96  -40, -39, -38, -37, -36, -35, -34, -33,
97  -32, -31, -30, -29, -28, -27, -26, -25,
98  -24, -23, -22, -21, -20, -19, -18, -17,
99  -16, -15, -14, -13, -12, -11, -10, -9,
100  -8, -7, -6, -5, -4, -3, -2, -1
101 
102 };
103 
104 static const int8_t sol_table_old[16] = {
105  0x0, 0x1, 0x2, 0x3, 0x6, 0xA, 0xF, 0x15,
106  -0x15, -0xF, -0xA, -0x6, -0x3, -0x2, -0x1, 0x0
107 };
108 
109 static const int8_t sol_table_new[16] = {
110  0x0, 0x1, 0x2, 0x3, 0x6, 0xA, 0xF, 0x15,
111  0x0, -0x1, -0x2, -0x3, -0x6, -0xA, -0xF, -0x15
112 };
113 
114 static const int16_t sol_table_16[128] = {
115  0x000, 0x008, 0x010, 0x020, 0x030, 0x040, 0x050, 0x060, 0x070, 0x080,
116  0x090, 0x0A0, 0x0B0, 0x0C0, 0x0D0, 0x0E0, 0x0F0, 0x100, 0x110, 0x120,
117  0x130, 0x140, 0x150, 0x160, 0x170, 0x180, 0x190, 0x1A0, 0x1B0, 0x1C0,
118  0x1D0, 0x1E0, 0x1F0, 0x200, 0x208, 0x210, 0x218, 0x220, 0x228, 0x230,
119  0x238, 0x240, 0x248, 0x250, 0x258, 0x260, 0x268, 0x270, 0x278, 0x280,
120  0x288, 0x290, 0x298, 0x2A0, 0x2A8, 0x2B0, 0x2B8, 0x2C0, 0x2C8, 0x2D0,
121  0x2D8, 0x2E0, 0x2E8, 0x2F0, 0x2F8, 0x300, 0x308, 0x310, 0x318, 0x320,
122  0x328, 0x330, 0x338, 0x340, 0x348, 0x350, 0x358, 0x360, 0x368, 0x370,
123  0x378, 0x380, 0x388, 0x390, 0x398, 0x3A0, 0x3A8, 0x3B0, 0x3B8, 0x3C0,
124  0x3C8, 0x3D0, 0x3D8, 0x3E0, 0x3E8, 0x3F0, 0x3F8, 0x400, 0x440, 0x480,
125  0x4C0, 0x500, 0x540, 0x580, 0x5C0, 0x600, 0x640, 0x680, 0x6C0, 0x700,
126  0x740, 0x780, 0x7C0, 0x800, 0x900, 0xA00, 0xB00, 0xC00, 0xD00, 0xE00,
127  0xF00, 0x1000, 0x1400, 0x1800, 0x1C00, 0x2000, 0x3000, 0x4000
128 };
129 
130 static const int16_t wady_table[128] = {
131  0, 2, 4, 6, 8, 10, 12, 15,
132  18, 21, 24, 28, 32, 36, 40, 44,
133  49, 54, 59, 64, 70, 76, 82, 88,
134  95, 102, 109, 116, 124, 132, 140, 148,
135  160, 170, 180, 190, 200, 210, 220, 230,
136  240, 255, 270, 285, 300, 320, 340, 360,
137  380, 400, 425, 450, 475, 500, 525, 550,
138  580, 610, 650, 700, 750, 800, 900, 1000,
139  -0, -2, -4, -6, -8, -10, -12, -15,
140  -18, -21, -24, -28, -32, -36, -40, -44,
141  -49, -54, -59, -64, -70, -76, -82, -88,
142  -95, -102,-109,-116,-124,-132,-140,-148,
143  -160,-170,-180,-190,-200,-210,-220,-230,
144  -240,-255,-270,-285,-300,-320,-340,-360,
145  -380,-400,-425,-450,-475,-500,-525,-550,
146  -580,-610,-650,-700,-750,-800,-900,-1000,
147 };
148 
149 static av_cold int dpcm_decode_init(AVCodecContext *avctx)
150 {
151  DPCMContext *s = avctx->priv_data;
152  int i;
153 
154  if (avctx->ch_layout.nb_channels < 1 || avctx->ch_layout.nb_channels > 2) {
155  av_log(avctx, AV_LOG_ERROR, "invalid number of channels\n");
156  return AVERROR(EINVAL);
157  }
158 
159  s->sample[0] = s->sample[1] = 0;
160 
161  switch (avctx->codec->id) {
162 
163  case AV_CODEC_ID_ROQ_DPCM:
164  /* initialize square table */
165  for (i = 0; i < 128; i++) {
166  int16_t square = i * i;
167  s->array[i ] = square;
168  s->array[i + 128] = -square;
169  }
170  break;
171 
172  case AV_CODEC_ID_SOL_DPCM:
173  switch(avctx->codec_tag){
174  case 1:
175  s->sol_table = sol_table_old;
176  s->sample[0] = s->sample[1] = 0x80;
177  break;
178  case 2:
179  s->sol_table = sol_table_new;
180  s->sample[0] = s->sample[1] = 0x80;
181  break;
182  case 3:
183  break;
184  default:
185  av_log(avctx, AV_LOG_ERROR, "Unknown SOL subcodec\n");
186  return -1;
187  }
188  break;
189 
190  case AV_CODEC_ID_SDX2_DPCM:
191  for (i = -128; i < 128; i++) {
192  int16_t square = i * i * 2;
193  s->array[i+128] = i < 0 ? -square: square;
194  }
195  break;
196 
197  case AV_CODEC_ID_CBD2_DPCM:
198  for (i = -128; i < 128; i++) {
199  int16_t cube = (i * i * i) / 64;
200  s->array[i+128] = cube;
201  }
202  break;
203 
204  case AV_CODEC_ID_GREMLIN_DPCM: {
205  int delta = 0;
206  int code = 64;
207  int step = 45;
208 
209  s->array[0] = 0;
210  for (i = 0; i < 127; i++) {
211  delta += (code >> 5);
212  code += step;
213  step += 2;
214 
215  s->array[i*2 + 1] = delta;
216  s->array[i*2 + 2] = -delta;
217  }
218  s->array[255] = delta + (code >> 5);
219  }
220  break;
221 
222  case AV_CODEC_ID_WADY_DPCM:
223  s->scale = (avctx->extradata && avctx->extradata_size > 0) ? avctx->extradata[0] : 1;
224  break;
225 
226  default:
227  break;
228  }
229 
230  if (avctx->codec->id == AV_CODEC_ID_SOL_DPCM && avctx->codec_tag != 3)
231  avctx->sample_fmt = AV_SAMPLE_FMT_U8;
232  else
233  avctx->sample_fmt = AV_SAMPLE_FMT_S16;
234 
235  return 0;
236 }
237 
238 
239 static int dpcm_decode_frame(AVCodecContext *avctx, AVFrame *frame,
240  int *got_frame_ptr, AVPacket *avpkt)
241 {
242  int buf_size = avpkt->size;
243  DPCMContext *s = avctx->priv_data;
244  int out = 0, ret;
245  int predictor[2];
246  int ch = 0;
247  int stereo = avctx->ch_layout.nb_channels - 1;
248  int16_t *output_samples, *samples_end;
249  GetByteContext gb;
250 
251  if (stereo && (buf_size & 1))
252  buf_size--;
253  bytestream2_init(&gb, avpkt->data, buf_size);
254 
255  /* calculate output size */
256  switch(avctx->codec->id) {
257  case AV_CODEC_ID_ROQ_DPCM:
258  out = buf_size - 8;
259  break;
260  case AV_CODEC_ID_INTERPLAY_DPCM:
261  out = buf_size - 6 - avctx->ch_layout.nb_channels;
262  break;
263  case AV_CODEC_ID_XAN_DPCM:
264  out = buf_size - 2 * avctx->ch_layout.nb_channels;
265  break;
266  case AV_CODEC_ID_SOL_DPCM:
267  if (avctx->codec_tag != 3)
268  out = buf_size * 2;
269  else
270  out = buf_size;
271  break;
272  case AV_CODEC_ID_WADY_DPCM:
273  case AV_CODEC_ID_DERF_DPCM:
274  case AV_CODEC_ID_GREMLIN_DPCM:
275  case AV_CODEC_ID_CBD2_DPCM:
276  case AV_CODEC_ID_SDX2_DPCM:
277  out = buf_size;
278  break;
279  }
280  if (out <= 0) {
281  av_log(avctx, AV_LOG_ERROR, "packet is too small\n");
282  return AVERROR(EINVAL);
283  }
284  if (out % avctx->ch_layout.nb_channels) {
285  av_log(avctx, AV_LOG_WARNING, "channels have differing number of samples\n");
286  }
287 
288  /* get output buffer */
289  frame->nb_samples = (out + avctx->ch_layout.nb_channels - 1) / avctx->ch_layout.nb_channels;
290  if ((ret = ff_get_buffer(avctx, frame, 0)) < 0)
291  return ret;
292  output_samples = (int16_t *)frame->data[0];
293  samples_end = output_samples + out;
294 
295  switch(avctx->codec->id) {
296 
297  case AV_CODEC_ID_ROQ_DPCM:
298  bytestream2_skipu(&gb, 6);
299 
300  if (stereo) {
301  predictor[1] = sign_extend(bytestream2_get_byteu(&gb) << 8, 16);
302  predictor[0] = sign_extend(bytestream2_get_byteu(&gb) << 8, 16);
303  } else {
304  predictor[0] = sign_extend(bytestream2_get_le16u(&gb), 16);
305  }
306 
307  /* decode the samples */
308  while (output_samples < samples_end) {
309  predictor[ch] += s->array[bytestream2_get_byteu(&gb)];
310  predictor[ch] = av_clip_int16(predictor[ch]);
311  *output_samples++ = predictor[ch];
312 
313  /* toggle channel */
314  ch ^= stereo;
315  }
316  break;
317 
318  case AV_CODEC_ID_INTERPLAY_DPCM:
319  bytestream2_skipu(&gb, 6); /* skip over the stream mask and stream length */
320 
321  for (ch = 0; ch < avctx->ch_layout.nb_channels; ch++) {
322  predictor[ch] = sign_extend(bytestream2_get_le16u(&gb), 16);
323  *output_samples++ = predictor[ch];
324  }
325 
326  ch = 0;
327  while (output_samples < samples_end) {
328  predictor[ch] += interplay_delta_table[bytestream2_get_byteu(&gb)];
329  predictor[ch] = av_clip_int16(predictor[ch]);
330  *output_samples++ = predictor[ch];
331 
332  /* toggle channel */
333  ch ^= stereo;
334  }
335  break;
336 
337  case AV_CODEC_ID_XAN_DPCM:
338  {
339  int shift[2] = { 4, 4 };
340 
341  for (ch = 0; ch < avctx->ch_layout.nb_channels; ch++)
342  predictor[ch] = sign_extend(bytestream2_get_le16u(&gb), 16);
343 
344  ch = 0;
345  while (output_samples < samples_end) {
346  int diff = bytestream2_get_byteu(&gb);
347  int n = diff & 3;
348 
349  if (n == 3)
350  shift[ch]++;
351  else
352  shift[ch] -= (2 * n);
353  diff = sign_extend((diff &~ 3) << 8, 16);
354 
355  /* saturate the shifter to 0..31 */
356  shift[ch] = av_clip_uintp2(shift[ch], 5);
357 
358  diff >>= shift[ch];
359  predictor[ch] += diff;
360 
361  predictor[ch] = av_clip_int16(predictor[ch]);
362  *output_samples++ = predictor[ch];
363 
364  /* toggle channel */
365  ch ^= stereo;
366  }
367  break;
368  }
369  case AV_CODEC_ID_SOL_DPCM:
370  if (avctx->codec_tag != 3) {
371  uint8_t *output_samples_u8 = frame->data[0],
372  *samples_end_u8 = output_samples_u8 + out;
373  while (output_samples_u8 < samples_end_u8) {
374  int n = bytestream2_get_byteu(&gb);
375 
376  s->sample[0] += s->sol_table[n >> 4];
377  s->sample[0] = av_clip_uint8(s->sample[0]);
378  *output_samples_u8++ = s->sample[0];
379 
380  s->sample[stereo] += s->sol_table[n & 0x0F];
381  s->sample[stereo] = av_clip_uint8(s->sample[stereo]);
382  *output_samples_u8++ = s->sample[stereo];
383  }
384  } else {
385  while (output_samples < samples_end) {
386  int n = bytestream2_get_byteu(&gb);
387  if (n & 0x80) s->sample[ch] -= sol_table_16[n & 0x7F];
388  else s->sample[ch] += sol_table_16[n & 0x7F];
389  s->sample[ch] = av_clip_int16(s->sample[ch]);
390  *output_samples++ = s->sample[ch];
391  /* toggle channel */
392  ch ^= stereo;
393  }
394  }
395  break;
396 
397  case AV_CODEC_ID_CBD2_DPCM:
398  case AV_CODEC_ID_SDX2_DPCM:
399  while (output_samples < samples_end) {
400  int8_t n = bytestream2_get_byteu(&gb);
401 
402  if (!(n & 1))
403  s->sample[ch] = 0;
404  s->sample[ch] += s->array[n + 128];
405  s->sample[ch] = av_clip_int16(s->sample[ch]);
406  *output_samples++ = s->sample[ch];
407  ch ^= stereo;
408  }
409  break;
410 
411  case AV_CODEC_ID_GREMLIN_DPCM: {
412  int idx = 0;
413 
414  while (output_samples < samples_end) {
415  uint8_t n = bytestream2_get_byteu(&gb);
416 
417  *output_samples++ = s->sample[idx] += (unsigned)s->array[n];
418  idx ^= 1;
419  }
420  }
421  break;
422 
423  case AV_CODEC_ID_DERF_DPCM: {
424  int idx = 0;
425 
426  while (output_samples < samples_end) {
427  uint8_t n = bytestream2_get_byteu(&gb);
428  int index = FFMIN(n & 0x7f, 95);
429 
430  s->sample[idx] += (n & 0x80 ? -1: 1) * derf_steps[index];
431  s->sample[idx] = av_clip_int16(s->sample[idx]);
432  *output_samples++ = s->sample[idx];
433  idx ^= stereo;
434  }
435  }
436  break;
437 
438  case AV_CODEC_ID_WADY_DPCM: {
439  int idx = 0;
440 
441  while (output_samples < samples_end) {
442  const uint8_t n = bytestream2_get_byteu(&gb);
443 
444  if (n & 0x80)
445  s->sample[idx] = sign_extend((n & 0x7f) << 9, 16);
446  else
447  s->sample[idx] += s->scale * (unsigned)wady_table[n & 0x7f];
448  *output_samples++ = av_clip_int16(s->sample[idx]);
449  idx ^= stereo;
450  }
451  }
452  break;
453  }
454 
455  *got_frame_ptr = 1;
456 
457  return avpkt->size;
458 }
459 
460 static void dpcm_flush(AVCodecContext *avctx)
461 {
462  DPCMContext *s = avctx->priv_data;
463 
464  s->sample[0] = s->sample[1] = 0;
465 }
466 
467 #define DPCM_DECODER(id_, name_, long_name_) \
468 const FFCodec ff_ ## name_ ## _decoder = { \
469  .p.name = #name_, \
470  CODEC_LONG_NAME(long_name_), \
471  .p.type = AVMEDIA_TYPE_AUDIO, \
472  .p.id = id_, \
473  .p.capabilities = AV_CODEC_CAP_DR1, \
474  .priv_data_size = sizeof(DPCMContext), \
475  .init = dpcm_decode_init, \
476  .flush = dpcm_flush, \
477  FF_CODEC_DECODE_CB(dpcm_decode_frame), \
478 }
479 
480 DPCM_DECODER(AV_CODEC_ID_CBD2_DPCM, cbd2_dpcm, "DPCM Cuberoot-Delta-Exact");
481 DPCM_DECODER(AV_CODEC_ID_DERF_DPCM, derf_dpcm, "DPCM Xilam DERF");
482 DPCM_DECODER(AV_CODEC_ID_GREMLIN_DPCM, gremlin_dpcm, "DPCM Gremlin");
483 DPCM_DECODER(AV_CODEC_ID_INTERPLAY_DPCM, interplay_dpcm, "DPCM Interplay");
484 DPCM_DECODER(AV_CODEC_ID_ROQ_DPCM, roq_dpcm, "DPCM id RoQ");
485 DPCM_DECODER(AV_CODEC_ID_SDX2_DPCM, sdx2_dpcm, "DPCM Squareroot-Delta-Exact");
486 DPCM_DECODER(AV_CODEC_ID_SOL_DPCM, sol_dpcm, "DPCM Sol");
487 DPCM_DECODER(AV_CODEC_ID_XAN_DPCM, xan_dpcm, "DPCM Xan");
488 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:114
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:467
DPCMContext
Definition: dpcm.c:46
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:410
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:68
DPCMContext::scale
int scale
scale for WADY_DPCM
Definition: dpcm.c:49
AVPacket::data
uint8_t * data
Definition: packet.h:535
dpcm_decode_init
static av_cold int dpcm_decode_init(AVCodecContext *avctx)
Definition: dpcm.c:149
AV_CODEC_ID_SOL_DPCM
@ AV_CODEC_ID_SOL_DPCM
Definition: codec_id.h:441
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:109
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:239
AV_CODEC_ID_XAN_DPCM
@ AV_CODEC_ID_XAN_DPCM
Definition: codec_id.h:440
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:445
DPCMContext::sample
int sample[2]
previous sample (for SOL_DPCM and WADY_DPCM)
Definition: dpcm.c:48
decode.h
AV_CODEC_ID_DERF_DPCM
@ AV_CODEC_ID_DERF_DPCM
Definition: codec_id.h:444
sol_table_old
static const int8_t sol_table_old[16]
Definition: dpcm.c:104
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:439
derf_steps
static const int32_t derf_steps[96]
Definition: dpcm.c:53
mathops.h
DPCMContext::sol_table
const int8_t * sol_table
delta table for SOL_DPCM
Definition: dpcm.c:50
AV_CODEC_ID_ROQ_DPCM
@ AV_CODEC_ID_ROQ_DPCM
Definition: codec_id.h:438
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:1635
AVPacket::size
int size
Definition: packet.h:536
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
predictor
static void predictor(uint8_t *src, ptrdiff_t size)
Definition: exrenc.c:170
DPCMContext::array
int16_t array[256]
Definition: dpcm.c:47
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:264
dpcm_flush
static void dpcm_flush(AVCodecContext *avctx)
Definition: dpcm.c:460
square
static int square(int x)
Definition: roqvideoenc.c:196
AV_CODEC_ID_CBD2_DPCM
@ AV_CODEC_ID_CBD2_DPCM
Definition: codec_id.h:446
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:130
AV_CODEC_ID_SDX2_DPCM
@ AV_CODEC_ID_SDX2_DPCM
Definition: codec_id.h:442
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:512
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:443
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