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)
49  const int8_t *sol_table; ///< delta table for SOL_DPCM
50 } DPCMContext;
51 
52 static const int32_t derf_steps[96] = {
53  0, 1, 2, 3, 4, 5, 6, 7,
54  8, 9, 10, 11, 12, 13, 14, 16,
55  17, 19, 21, 23, 25, 28, 31, 34,
56  37, 41, 45, 50, 55, 60, 66, 73,
57  80, 88, 97, 107, 118, 130, 143, 157,
58  173, 190, 209, 230, 253, 279, 307, 337,
59  371, 408, 449, 494, 544, 598, 658, 724,
60  796, 876, 963, 1060, 1166, 1282, 1411, 1552,
61  1707, 1878, 2066, 2272, 2499, 2749, 3024, 3327,
62  3660, 4026, 4428, 4871, 5358, 5894, 6484, 7132,
63  7845, 8630, 9493, 10442, 11487, 12635, 13899, 15289,
64  16818, 18500, 20350, 22385, 24623, 27086, 29794, 32767,
65 };
66 
67 static const int16_t interplay_delta_table[] = {
68  0, 1, 2, 3, 4, 5, 6, 7,
69  8, 9, 10, 11, 12, 13, 14, 15,
70  16, 17, 18, 19, 20, 21, 22, 23,
71  24, 25, 26, 27, 28, 29, 30, 31,
72  32, 33, 34, 35, 36, 37, 38, 39,
73  40, 41, 42, 43, 47, 51, 56, 61,
74  66, 72, 79, 86, 94, 102, 112, 122,
75  133, 145, 158, 173, 189, 206, 225, 245,
76  267, 292, 318, 348, 379, 414, 452, 493,
77  538, 587, 640, 699, 763, 832, 908, 991,
78  1081, 1180, 1288, 1405, 1534, 1673, 1826, 1993,
79  2175, 2373, 2590, 2826, 3084, 3365, 3672, 4008,
80  4373, 4772, 5208, 5683, 6202, 6767, 7385, 8059,
81  8794, 9597, 10472, 11428, 12471, 13609, 14851, 16206,
82  17685, 19298, 21060, 22981, 25078, 27367, 29864, 32589,
83  -29973, -26728, -23186, -19322, -15105, -10503, -5481, -1,
84  1, 1, 5481, 10503, 15105, 19322, 23186, 26728,
85  29973, -32589, -29864, -27367, -25078, -22981, -21060, -19298,
86  -17685, -16206, -14851, -13609, -12471, -11428, -10472, -9597,
87  -8794, -8059, -7385, -6767, -6202, -5683, -5208, -4772,
88  -4373, -4008, -3672, -3365, -3084, -2826, -2590, -2373,
89  -2175, -1993, -1826, -1673, -1534, -1405, -1288, -1180,
90  -1081, -991, -908, -832, -763, -699, -640, -587,
91  -538, -493, -452, -414, -379, -348, -318, -292,
92  -267, -245, -225, -206, -189, -173, -158, -145,
93  -133, -122, -112, -102, -94, -86, -79, -72,
94  -66, -61, -56, -51, -47, -43, -42, -41,
95  -40, -39, -38, -37, -36, -35, -34, -33,
96  -32, -31, -30, -29, -28, -27, -26, -25,
97  -24, -23, -22, -21, -20, -19, -18, -17,
98  -16, -15, -14, -13, -12, -11, -10, -9,
99  -8, -7, -6, -5, -4, -3, -2, -1
100 
101 };
102 
103 static const int8_t sol_table_old[16] = {
104  0x0, 0x1, 0x2, 0x3, 0x6, 0xA, 0xF, 0x15,
105  -0x15, -0xF, -0xA, -0x6, -0x3, -0x2, -0x1, 0x0
106 };
107 
108 static const int8_t sol_table_new[16] = {
109  0x0, 0x1, 0x2, 0x3, 0x6, 0xA, 0xF, 0x15,
110  0x0, -0x1, -0x2, -0x3, -0x6, -0xA, -0xF, -0x15
111 };
112 
113 static const int16_t sol_table_16[128] = {
114  0x000, 0x008, 0x010, 0x020, 0x030, 0x040, 0x050, 0x060, 0x070, 0x080,
115  0x090, 0x0A0, 0x0B0, 0x0C0, 0x0D0, 0x0E0, 0x0F0, 0x100, 0x110, 0x120,
116  0x130, 0x140, 0x150, 0x160, 0x170, 0x180, 0x190, 0x1A0, 0x1B0, 0x1C0,
117  0x1D0, 0x1E0, 0x1F0, 0x200, 0x208, 0x210, 0x218, 0x220, 0x228, 0x230,
118  0x238, 0x240, 0x248, 0x250, 0x258, 0x260, 0x268, 0x270, 0x278, 0x280,
119  0x288, 0x290, 0x298, 0x2A0, 0x2A8, 0x2B0, 0x2B8, 0x2C0, 0x2C8, 0x2D0,
120  0x2D8, 0x2E0, 0x2E8, 0x2F0, 0x2F8, 0x300, 0x308, 0x310, 0x318, 0x320,
121  0x328, 0x330, 0x338, 0x340, 0x348, 0x350, 0x358, 0x360, 0x368, 0x370,
122  0x378, 0x380, 0x388, 0x390, 0x398, 0x3A0, 0x3A8, 0x3B0, 0x3B8, 0x3C0,
123  0x3C8, 0x3D0, 0x3D8, 0x3E0, 0x3E8, 0x3F0, 0x3F8, 0x400, 0x440, 0x480,
124  0x4C0, 0x500, 0x540, 0x580, 0x5C0, 0x600, 0x640, 0x680, 0x6C0, 0x700,
125  0x740, 0x780, 0x7C0, 0x800, 0x900, 0xA00, 0xB00, 0xC00, 0xD00, 0xE00,
126  0xF00, 0x1000, 0x1400, 0x1800, 0x1C00, 0x2000, 0x3000, 0x4000
127 };
128 
129 
131 {
132  DPCMContext *s = avctx->priv_data;
133  int i;
134 
135  if (avctx->ch_layout.nb_channels < 1 || avctx->ch_layout.nb_channels > 2) {
136  av_log(avctx, AV_LOG_ERROR, "invalid number of channels\n");
137  return AVERROR(EINVAL);
138  }
139 
140  s->sample[0] = s->sample[1] = 0;
141 
142  switch(avctx->codec->id) {
143 
145  /* initialize square table */
146  for (i = 0; i < 128; i++) {
147  int16_t square = i * i;
148  s->array[i ] = square;
149  s->array[i + 128] = -square;
150  }
151  break;
152 
154  switch(avctx->codec_tag){
155  case 1:
156  s->sol_table = sol_table_old;
157  s->sample[0] = s->sample[1] = 0x80;
158  break;
159  case 2:
160  s->sol_table = sol_table_new;
161  s->sample[0] = s->sample[1] = 0x80;
162  break;
163  case 3:
164  break;
165  default:
166  av_log(avctx, AV_LOG_ERROR, "Unknown SOL subcodec\n");
167  return -1;
168  }
169  break;
170 
172  for (i = -128; i < 128; i++) {
173  int16_t square = i * i * 2;
174  s->array[i+128] = i < 0 ? -square: square;
175  }
176  break;
177 
179  int delta = 0;
180  int code = 64;
181  int step = 45;
182 
183  s->array[0] = 0;
184  for (i = 0; i < 127; i++) {
185  delta += (code >> 5);
186  code += step;
187  step += 2;
188 
189  s->array[i*2 + 1] = delta;
190  s->array[i*2 + 2] = -delta;
191  }
192  s->array[255] = delta + (code >> 5);
193  }
194  break;
195 
196  default:
197  break;
198  }
199 
200  if (avctx->codec->id == AV_CODEC_ID_SOL_DPCM && avctx->codec_tag != 3)
201  avctx->sample_fmt = AV_SAMPLE_FMT_U8;
202  else
203  avctx->sample_fmt = AV_SAMPLE_FMT_S16;
204 
205  return 0;
206 }
207 
208 
210  int *got_frame_ptr, AVPacket *avpkt)
211 {
212  int buf_size = avpkt->size;
213  DPCMContext *s = avctx->priv_data;
214  int out = 0, ret;
215  int predictor[2];
216  int ch = 0;
217  int stereo = avctx->ch_layout.nb_channels - 1;
218  int16_t *output_samples, *samples_end;
219  GetByteContext gb;
220 
221  if (stereo && (buf_size & 1))
222  buf_size--;
223  bytestream2_init(&gb, avpkt->data, buf_size);
224 
225  /* calculate output size */
226  switch(avctx->codec->id) {
228  out = buf_size - 8;
229  break;
231  out = buf_size - 6 - avctx->ch_layout.nb_channels;
232  break;
234  out = buf_size - 2 * avctx->ch_layout.nb_channels;
235  break;
237  if (avctx->codec_tag != 3)
238  out = buf_size * 2;
239  else
240  out = buf_size;
241  break;
245  out = buf_size;
246  break;
247  }
248  if (out <= 0) {
249  av_log(avctx, AV_LOG_ERROR, "packet is too small\n");
250  return AVERROR(EINVAL);
251  }
252  if (out % avctx->ch_layout.nb_channels) {
253  av_log(avctx, AV_LOG_WARNING, "channels have differing number of samples\n");
254  }
255 
256  /* get output buffer */
257  frame->nb_samples = (out + avctx->ch_layout.nb_channels - 1) / avctx->ch_layout.nb_channels;
258  if ((ret = ff_get_buffer(avctx, frame, 0)) < 0)
259  return ret;
260  output_samples = (int16_t *)frame->data[0];
261  samples_end = output_samples + out;
262 
263  switch(avctx->codec->id) {
264 
266  bytestream2_skipu(&gb, 6);
267 
268  if (stereo) {
269  predictor[1] = sign_extend(bytestream2_get_byteu(&gb) << 8, 16);
270  predictor[0] = sign_extend(bytestream2_get_byteu(&gb) << 8, 16);
271  } else {
272  predictor[0] = sign_extend(bytestream2_get_le16u(&gb), 16);
273  }
274 
275  /* decode the samples */
276  while (output_samples < samples_end) {
277  predictor[ch] += s->array[bytestream2_get_byteu(&gb)];
278  predictor[ch] = av_clip_int16(predictor[ch]);
279  *output_samples++ = predictor[ch];
280 
281  /* toggle channel */
282  ch ^= stereo;
283  }
284  break;
285 
287  bytestream2_skipu(&gb, 6); /* skip over the stream mask and stream length */
288 
289  for (ch = 0; ch < avctx->ch_layout.nb_channels; ch++) {
290  predictor[ch] = sign_extend(bytestream2_get_le16u(&gb), 16);
291  *output_samples++ = predictor[ch];
292  }
293 
294  ch = 0;
295  while (output_samples < samples_end) {
296  predictor[ch] += interplay_delta_table[bytestream2_get_byteu(&gb)];
297  predictor[ch] = av_clip_int16(predictor[ch]);
298  *output_samples++ = predictor[ch];
299 
300  /* toggle channel */
301  ch ^= stereo;
302  }
303  break;
304 
306  {
307  int shift[2] = { 4, 4 };
308 
309  for (ch = 0; ch < avctx->ch_layout.nb_channels; ch++)
310  predictor[ch] = sign_extend(bytestream2_get_le16u(&gb), 16);
311 
312  ch = 0;
313  while (output_samples < samples_end) {
314  int diff = bytestream2_get_byteu(&gb);
315  int n = diff & 3;
316 
317  if (n == 3)
318  shift[ch]++;
319  else
320  shift[ch] -= (2 * n);
321  diff = sign_extend((diff &~ 3) << 8, 16);
322 
323  /* saturate the shifter to 0..31 */
324  shift[ch] = av_clip_uintp2(shift[ch], 5);
325 
326  diff >>= shift[ch];
327  predictor[ch] += diff;
328 
329  predictor[ch] = av_clip_int16(predictor[ch]);
330  *output_samples++ = predictor[ch];
331 
332  /* toggle channel */
333  ch ^= stereo;
334  }
335  break;
336  }
338  if (avctx->codec_tag != 3) {
339  uint8_t *output_samples_u8 = frame->data[0],
340  *samples_end_u8 = output_samples_u8 + out;
341  while (output_samples_u8 < samples_end_u8) {
342  int n = bytestream2_get_byteu(&gb);
343 
344  s->sample[0] += s->sol_table[n >> 4];
345  s->sample[0] = av_clip_uint8(s->sample[0]);
346  *output_samples_u8++ = s->sample[0];
347 
348  s->sample[stereo] += s->sol_table[n & 0x0F];
349  s->sample[stereo] = av_clip_uint8(s->sample[stereo]);
350  *output_samples_u8++ = s->sample[stereo];
351  }
352  } else {
353  while (output_samples < samples_end) {
354  int n = bytestream2_get_byteu(&gb);
355  if (n & 0x80) s->sample[ch] -= sol_table_16[n & 0x7F];
356  else s->sample[ch] += sol_table_16[n & 0x7F];
357  s->sample[ch] = av_clip_int16(s->sample[ch]);
358  *output_samples++ = s->sample[ch];
359  /* toggle channel */
360  ch ^= stereo;
361  }
362  }
363  break;
364 
366  while (output_samples < samples_end) {
367  int8_t n = bytestream2_get_byteu(&gb);
368 
369  if (!(n & 1))
370  s->sample[ch] = 0;
371  s->sample[ch] += s->array[n + 128];
372  s->sample[ch] = av_clip_int16(s->sample[ch]);
373  *output_samples++ = s->sample[ch];
374  ch ^= stereo;
375  }
376  break;
377 
379  int idx = 0;
380 
381  while (output_samples < samples_end) {
382  uint8_t n = bytestream2_get_byteu(&gb);
383 
384  *output_samples++ = s->sample[idx] += (unsigned)s->array[n];
385  idx ^= 1;
386  }
387  }
388  break;
389 
390  case AV_CODEC_ID_DERF_DPCM: {
391  int idx = 0;
392 
393  while (output_samples < samples_end) {
394  uint8_t n = bytestream2_get_byteu(&gb);
395  int index = FFMIN(n & 0x7f, 95);
396 
397  s->sample[idx] += (n & 0x80 ? -1: 1) * derf_steps[index];
398  s->sample[idx] = av_clip_int16(s->sample[idx]);
399  *output_samples++ = s->sample[idx];
400  idx ^= stereo;
401  }
402  }
403  break;
404  }
405 
406  *got_frame_ptr = 1;
407 
408  return avpkt->size;
409 }
410 
411 #define DPCM_DECODER(id_, name_, long_name_) \
412 const FFCodec ff_ ## name_ ## _decoder = { \
413  .p.name = #name_, \
414  CODEC_LONG_NAME(long_name_), \
415  .p.type = AVMEDIA_TYPE_AUDIO, \
416  .p.id = id_, \
417  .p.capabilities = AV_CODEC_CAP_DR1, \
418  .priv_data_size = sizeof(DPCMContext), \
419  .init = dpcm_decode_init, \
420  FF_CODEC_DECODE_CB(dpcm_decode_frame), \
421 }
422 
423 DPCM_DECODER(AV_CODEC_ID_DERF_DPCM, derf_dpcm, "DPCM Xilam DERF");
424 DPCM_DECODER(AV_CODEC_ID_GREMLIN_DPCM, gremlin_dpcm, "DPCM Gremlin");
425 DPCM_DECODER(AV_CODEC_ID_INTERPLAY_DPCM, interplay_dpcm, "DPCM Interplay");
426 DPCM_DECODER(AV_CODEC_ID_ROQ_DPCM, roq_dpcm, "DPCM id RoQ");
427 DPCM_DECODER(AV_CODEC_ID_SDX2_DPCM, sdx2_dpcm, "DPCM Squareroot-Delta-Exact");
428 DPCM_DECODER(AV_CODEC_ID_SOL_DPCM, sol_dpcm, "DPCM Sol");
429 DPCM_DECODER(AV_CODEC_ID_XAN_DPCM, xan_dpcm, "DPCM Xan");
sol_table_16
static const int16_t sol_table_16[128]
Definition: dpcm.c:113
AV_LOG_WARNING
#define AV_LOG_WARNING
Something somehow does not look correct.
Definition: log.h:186
DPCM_DECODER
#define DPCM_DECODER(id_, name_, long_name_)
Definition: dpcm.c:411
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:54
GetByteContext
Definition: bytestream.h:33
av_clip_uintp2
#define av_clip_uintp2
Definition: common.h:119
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:325
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:67
AVPacket::data
uint8_t * data
Definition: packet.h:374
dpcm_decode_init
static av_cold int dpcm_decode_init(AVCodecContext *avctx)
Definition: dpcm.c:130
AV_CODEC_ID_SOL_DPCM
@ AV_CODEC_ID_SOL_DPCM
Definition: codec_id.h:429
AVChannelLayout::nb_channels
int nb_channels
Number of channels in this layout.
Definition: channel_layout.h:311
sol_table_new
static const int8_t sol_table_new[16]
Definition: dpcm.c:108
AVCodecContext::codec
const struct AVCodec * codec
Definition: avcodec.h:407
AVCodecContext::ch_layout
AVChannelLayout ch_layout
Audio channel layout.
Definition: avcodec.h:2059
dpcm_decode_frame
static int dpcm_decode_frame(AVCodecContext *avctx, AVFrame *frame, int *got_frame_ptr, AVPacket *avpkt)
Definition: dpcm.c:209
AV_CODEC_ID_XAN_DPCM
@ AV_CODEC_ID_XAN_DPCM
Definition: codec_id.h:428
AV_LOG_ERROR
#define AV_LOG_ERROR
Something went wrong and cannot losslessly be recovered.
Definition: log.h:180
av_cold
#define av_cold
Definition: attributes.h:90
s
#define s(width, name)
Definition: cbs_vp9.c:256
DPCMContext::sample
int sample[2]
previous sample (for SOL_DPCM)
Definition: dpcm.c:48
decode.h
AV_CODEC_ID_DERF_DPCM
@ AV_CODEC_ID_DERF_DPCM
Definition: codec_id.h:432
sol_table_old
static const int8_t sol_table_old[16]
Definition: dpcm.c:103
if
if(ret)
Definition: filter_design.txt:179
av_clip_int16
#define av_clip_int16
Definition: common.h:110
AV_CODEC_ID_INTERPLAY_DPCM
@ AV_CODEC_ID_INTERPLAY_DPCM
Definition: codec_id.h:427
derf_steps
static const int32_t derf_steps[96]
Definition: dpcm.c:52
mathops.h
DPCMContext::sol_table
const int8_t * sol_table
delta table for SOL_DPCM
Definition: dpcm.c:49
AV_CODEC_ID_ROQ_DPCM
@ AV_CODEC_ID_ROQ_DPCM
Definition: codec_id.h:426
index
int index
Definition: gxfenc.c:89
ff_get_buffer
int ff_get_buffer(AVCodecContext *avctx, AVFrame *frame, int flags)
Get a buffer for a frame.
Definition: decode.c:1450
AVPacket::size
int size
Definition: packet.h:375
codec_internal.h
shift
static int shift(int a, int b)
Definition: bonk.c:253
AVCodecContext::sample_fmt
enum AVSampleFormat sample_fmt
audio sample format
Definition: avcodec.h:1023
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:218
i
#define i(width, name, range_min, range_max)
Definition: cbs_h2645.c:269
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
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
square
static int square(int x)
Definition: roqvideoenc.c:195
AVCodecContext
main external API structure.
Definition: avcodec.h:398
sign_extend
static av_const int sign_extend(int val, unsigned bits)
Definition: mathops.h:133
av_clip_uint8
#define av_clip_uint8
Definition: common.h:101
AV_CODEC_ID_SDX2_DPCM
@ AV_CODEC_ID_SDX2_DPCM
Definition: codec_id.h:430
diff
static av_always_inline int diff(const uint32_t a, const uint32_t b)
Definition: vf_palettegen.c:139
AVCodecContext::codec_tag
unsigned int codec_tag
fourcc (LSB first, so "ABCD" -> ('D'<<24) + ('C'<<16) + ('B'<<8) + 'A').
Definition: avcodec.h:423
AVPacket
This structure stores compressed data.
Definition: packet.h:351
AVCodecContext::priv_data
void * priv_data
Definition: avcodec.h:425
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:431