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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 "libavutil/intreadwrite.h"
41 #include "avcodec.h"
42 #include "bytestream.h"
43 #include "internal.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 int16_t interplay_delta_table[] = {
53  0, 1, 2, 3, 4, 5, 6, 7,
54  8, 9, 10, 11, 12, 13, 14, 15,
55  16, 17, 18, 19, 20, 21, 22, 23,
56  24, 25, 26, 27, 28, 29, 30, 31,
57  32, 33, 34, 35, 36, 37, 38, 39,
58  40, 41, 42, 43, 47, 51, 56, 61,
59  66, 72, 79, 86, 94, 102, 112, 122,
60  133, 145, 158, 173, 189, 206, 225, 245,
61  267, 292, 318, 348, 379, 414, 452, 493,
62  538, 587, 640, 699, 763, 832, 908, 991,
63  1081, 1180, 1288, 1405, 1534, 1673, 1826, 1993,
64  2175, 2373, 2590, 2826, 3084, 3365, 3672, 4008,
65  4373, 4772, 5208, 5683, 6202, 6767, 7385, 8059,
66  8794, 9597, 10472, 11428, 12471, 13609, 14851, 16206,
67  17685, 19298, 21060, 22981, 25078, 27367, 29864, 32589,
68  -29973, -26728, -23186, -19322, -15105, -10503, -5481, -1,
69  1, 1, 5481, 10503, 15105, 19322, 23186, 26728,
70  29973, -32589, -29864, -27367, -25078, -22981, -21060, -19298,
71  -17685, -16206, -14851, -13609, -12471, -11428, -10472, -9597,
72  -8794, -8059, -7385, -6767, -6202, -5683, -5208, -4772,
73  -4373, -4008, -3672, -3365, -3084, -2826, -2590, -2373,
74  -2175, -1993, -1826, -1673, -1534, -1405, -1288, -1180,
75  -1081, -991, -908, -832, -763, -699, -640, -587,
76  -538, -493, -452, -414, -379, -348, -318, -292,
77  -267, -245, -225, -206, -189, -173, -158, -145,
78  -133, -122, -112, -102, -94, -86, -79, -72,
79  -66, -61, -56, -51, -47, -43, -42, -41,
80  -40, -39, -38, -37, -36, -35, -34, -33,
81  -32, -31, -30, -29, -28, -27, -26, -25,
82  -24, -23, -22, -21, -20, -19, -18, -17,
83  -16, -15, -14, -13, -12, -11, -10, -9,
84  -8, -7, -6, -5, -4, -3, -2, -1
85 
86 };
87 
88 static const int8_t sol_table_old[16] = {
89  0x0, 0x1, 0x2, 0x3, 0x6, 0xA, 0xF, 0x15,
90  -0x15, -0xF, -0xA, -0x6, -0x3, -0x2, -0x1, 0x0
91 };
92 
93 static const int8_t sol_table_new[16] = {
94  0x0, 0x1, 0x2, 0x3, 0x6, 0xA, 0xF, 0x15,
95  0x0, -0x1, -0x2, -0x3, -0x6, -0xA, -0xF, -0x15
96 };
97 
98 static const int16_t sol_table_16[128] = {
99  0x000, 0x008, 0x010, 0x020, 0x030, 0x040, 0x050, 0x060, 0x070, 0x080,
100  0x090, 0x0A0, 0x0B0, 0x0C0, 0x0D0, 0x0E0, 0x0F0, 0x100, 0x110, 0x120,
101  0x130, 0x140, 0x150, 0x160, 0x170, 0x180, 0x190, 0x1A0, 0x1B0, 0x1C0,
102  0x1D0, 0x1E0, 0x1F0, 0x200, 0x208, 0x210, 0x218, 0x220, 0x228, 0x230,
103  0x238, 0x240, 0x248, 0x250, 0x258, 0x260, 0x268, 0x270, 0x278, 0x280,
104  0x288, 0x290, 0x298, 0x2A0, 0x2A8, 0x2B0, 0x2B8, 0x2C0, 0x2C8, 0x2D0,
105  0x2D8, 0x2E0, 0x2E8, 0x2F0, 0x2F8, 0x300, 0x308, 0x310, 0x318, 0x320,
106  0x328, 0x330, 0x338, 0x340, 0x348, 0x350, 0x358, 0x360, 0x368, 0x370,
107  0x378, 0x380, 0x388, 0x390, 0x398, 0x3A0, 0x3A8, 0x3B0, 0x3B8, 0x3C0,
108  0x3C8, 0x3D0, 0x3D8, 0x3E0, 0x3E8, 0x3F0, 0x3F8, 0x400, 0x440, 0x480,
109  0x4C0, 0x500, 0x540, 0x580, 0x5C0, 0x600, 0x640, 0x680, 0x6C0, 0x700,
110  0x740, 0x780, 0x7C0, 0x800, 0x900, 0xA00, 0xB00, 0xC00, 0xD00, 0xE00,
111  0xF00, 0x1000, 0x1400, 0x1800, 0x1C00, 0x2000, 0x3000, 0x4000
112 };
113 
114 
116 {
117  DPCMContext *s = avctx->priv_data;
118  int i;
119 
120  if (avctx->channels < 1 || avctx->channels > 2) {
121  av_log(avctx, AV_LOG_ERROR, "invalid number of channels\n");
122  return AVERROR(EINVAL);
123  }
124 
125  s->sample[0] = s->sample[1] = 0;
126 
127  switch(avctx->codec->id) {
128 
130  /* initialize square table */
131  for (i = 0; i < 128; i++) {
132  int16_t square = i * i;
133  s->array[i ] = square;
134  s->array[i + 128] = -square;
135  }
136  break;
137 
139  switch(avctx->codec_tag){
140  case 1:
142  s->sample[0] = s->sample[1] = 0x80;
143  break;
144  case 2:
146  s->sample[0] = s->sample[1] = 0x80;
147  break;
148  case 3:
149  break;
150  default:
151  av_log(avctx, AV_LOG_ERROR, "Unknown SOL subcodec\n");
152  return -1;
153  }
154  break;
155 
157  for (i = -128; i < 128; i++) {
158  int16_t square = i * i * 2;
159  s->array[i+128] = i < 0 ? -square: square;
160  }
161  break;
162 
164  int delta = 0;
165  int code = 64;
166  int step = 45;
167 
168  s->array[0] = 0;
169  for (i = 0; i < 127; i++) {
170  delta += (code >> 5);
171  code += step;
172  step += 2;
173 
174  s->array[i*2 + 1] = delta;
175  s->array[i*2 + 2] = -delta;
176  }
177  s->array[255] = delta + (code >> 5);
178  }
179  break;
180 
181  default:
182  break;
183  }
184 
185  if (avctx->codec->id == AV_CODEC_ID_SOL_DPCM && avctx->codec_tag != 3)
186  avctx->sample_fmt = AV_SAMPLE_FMT_U8;
187  else
188  avctx->sample_fmt = AV_SAMPLE_FMT_S16;
189 
190  return 0;
191 }
192 
193 
194 static int dpcm_decode_frame(AVCodecContext *avctx, void *data,
195  int *got_frame_ptr, AVPacket *avpkt)
196 {
197  int buf_size = avpkt->size;
198  DPCMContext *s = avctx->priv_data;
199  AVFrame *frame = data;
200  int out = 0, ret;
201  int predictor[2];
202  int ch = 0;
203  int stereo = avctx->channels - 1;
204  int16_t *output_samples, *samples_end;
205  GetByteContext gb;
206 
207  if (stereo && (buf_size & 1))
208  buf_size--;
209  bytestream2_init(&gb, avpkt->data, buf_size);
210 
211  /* calculate output size */
212  switch(avctx->codec->id) {
214  out = buf_size - 8;
215  break;
217  out = buf_size - 6 - avctx->channels;
218  break;
220  out = buf_size - 2 * avctx->channels;
221  break;
223  if (avctx->codec_tag != 3)
224  out = buf_size * 2;
225  else
226  out = buf_size;
227  break;
230  out = buf_size;
231  break;
232  }
233  if (out <= 0) {
234  av_log(avctx, AV_LOG_ERROR, "packet is too small\n");
235  return AVERROR(EINVAL);
236  }
237  if (out % avctx->channels) {
238  av_log(avctx, AV_LOG_WARNING, "channels have differing number of samples\n");
239  }
240 
241  /* get output buffer */
242  frame->nb_samples = (out + avctx->channels - 1) / avctx->channels;
243  if ((ret = ff_get_buffer(avctx, frame, 0)) < 0)
244  return ret;
245  output_samples = (int16_t *)frame->data[0];
246  samples_end = output_samples + out;
247 
248  switch(avctx->codec->id) {
249 
251  bytestream2_skipu(&gb, 6);
252 
253  if (stereo) {
254  predictor[1] = sign_extend(bytestream2_get_byteu(&gb) << 8, 16);
255  predictor[0] = sign_extend(bytestream2_get_byteu(&gb) << 8, 16);
256  } else {
257  predictor[0] = sign_extend(bytestream2_get_le16u(&gb), 16);
258  }
259 
260  /* decode the samples */
261  while (output_samples < samples_end) {
262  predictor[ch] += s->array[bytestream2_get_byteu(&gb)];
263  predictor[ch] = av_clip_int16(predictor[ch]);
264  *output_samples++ = predictor[ch];
265 
266  /* toggle channel */
267  ch ^= stereo;
268  }
269  break;
270 
272  bytestream2_skipu(&gb, 6); /* skip over the stream mask and stream length */
273 
274  for (ch = 0; ch < avctx->channels; ch++) {
275  predictor[ch] = sign_extend(bytestream2_get_le16u(&gb), 16);
276  *output_samples++ = predictor[ch];
277  }
278 
279  ch = 0;
280  while (output_samples < samples_end) {
281  predictor[ch] += interplay_delta_table[bytestream2_get_byteu(&gb)];
282  predictor[ch] = av_clip_int16(predictor[ch]);
283  *output_samples++ = predictor[ch];
284 
285  /* toggle channel */
286  ch ^= stereo;
287  }
288  break;
289 
291  {
292  int shift[2] = { 4, 4 };
293 
294  for (ch = 0; ch < avctx->channels; ch++)
295  predictor[ch] = sign_extend(bytestream2_get_le16u(&gb), 16);
296 
297  ch = 0;
298  while (output_samples < samples_end) {
299  int diff = bytestream2_get_byteu(&gb);
300  int n = diff & 3;
301 
302  if (n == 3)
303  shift[ch]++;
304  else
305  shift[ch] -= (2 * n);
306  diff = sign_extend((diff &~ 3) << 8, 16);
307 
308  /* saturate the shifter to a lower limit of 0 */
309  if (shift[ch] < 0)
310  shift[ch] = 0;
311 
312  diff >>= shift[ch];
313  predictor[ch] += diff;
314 
315  predictor[ch] = av_clip_int16(predictor[ch]);
316  *output_samples++ = predictor[ch];
317 
318  /* toggle channel */
319  ch ^= stereo;
320  }
321  break;
322  }
324  if (avctx->codec_tag != 3) {
325  uint8_t *output_samples_u8 = frame->data[0],
326  *samples_end_u8 = output_samples_u8 + out;
327  while (output_samples_u8 < samples_end_u8) {
328  int n = bytestream2_get_byteu(&gb);
329 
330  s->sample[0] += s->sol_table[n >> 4];
331  s->sample[0] = av_clip_uint8(s->sample[0]);
332  *output_samples_u8++ = s->sample[0];
333 
334  s->sample[stereo] += s->sol_table[n & 0x0F];
335  s->sample[stereo] = av_clip_uint8(s->sample[stereo]);
336  *output_samples_u8++ = s->sample[stereo];
337  }
338  } else {
339  while (output_samples < samples_end) {
340  int n = bytestream2_get_byteu(&gb);
341  if (n & 0x80) s->sample[ch] -= sol_table_16[n & 0x7F];
342  else s->sample[ch] += sol_table_16[n & 0x7F];
343  s->sample[ch] = av_clip_int16(s->sample[ch]);
344  *output_samples++ = s->sample[ch];
345  /* toggle channel */
346  ch ^= stereo;
347  }
348  }
349  break;
350 
352  while (output_samples < samples_end) {
353  int8_t n = bytestream2_get_byteu(&gb);
354 
355  if (!(n & 1))
356  s->sample[ch] = 0;
357  s->sample[ch] += s->array[n + 128];
358  s->sample[ch] = av_clip_int16(s->sample[ch]);
359  *output_samples++ = s->sample[ch];
360  ch ^= stereo;
361  }
362  break;
363 
365  int idx = 0;
366 
367  while (output_samples < samples_end) {
368  uint8_t n = bytestream2_get_byteu(&gb);
369 
370  *output_samples++ = s->sample[idx] += s->array[n];
371  idx ^= 1;
372  }
373  }
374  break;
375  }
376 
377  *got_frame_ptr = 1;
378 
379  return avpkt->size;
380 }
381 
382 #define DPCM_DECODER(id_, name_, long_name_) \
383 AVCodec ff_ ## name_ ## _decoder = { \
384  .name = #name_, \
385  .long_name = NULL_IF_CONFIG_SMALL(long_name_), \
386  .type = AVMEDIA_TYPE_AUDIO, \
387  .id = id_, \
388  .priv_data_size = sizeof(DPCMContext), \
389  .init = dpcm_decode_init, \
390  .decode = dpcm_decode_frame, \
391  .capabilities = AV_CODEC_CAP_DR1, \
392 }
393 
394 DPCM_DECODER(AV_CODEC_ID_GREMLIN_DPCM, gremlin_dpcm, "DPCM Gremlin");
395 DPCM_DECODER(AV_CODEC_ID_INTERPLAY_DPCM, interplay_dpcm, "DPCM Interplay");
396 DPCM_DECODER(AV_CODEC_ID_ROQ_DPCM, roq_dpcm, "DPCM id RoQ");
397 DPCM_DECODER(AV_CODEC_ID_SDX2_DPCM, sdx2_dpcm, "DPCM Squareroot-Delta-Exact");
398 DPCM_DECODER(AV_CODEC_ID_SOL_DPCM, sol_dpcm, "DPCM Sol");
399 DPCM_DECODER(AV_CODEC_ID_XAN_DPCM, xan_dpcm, "DPCM Xan");
const struct AVCodec * codec
Definition: avcodec.h:1755
const char * s
Definition: avisynth_c.h:768
static int shift(int a, int b)
Definition: sonic.c:82
This structure describes decoded (raw) audio or video data.
Definition: frame.h:201
ptrdiff_t const GLvoid * data
Definition: opengl_enc.c:101
static const int8_t sol_table_old[16]
Definition: dpcm.c:88
#define AV_LOG_WARNING
Something somehow does not look correct.
Definition: log.h:182
int size
Definition: avcodec.h:1672
const int8_t * sol_table
delta table for SOL_DPCM
Definition: dpcm.c:49
static av_always_inline void bytestream2_init(GetByteContext *g, const uint8_t *buf, int buf_size)
Definition: bytestream.h:133
enum AVSampleFormat sample_fmt
audio sample format
Definition: avcodec.h:2516
uint8_t
#define av_cold
Definition: attributes.h:82
AV_SAMPLE_FMT_U8
float delta
uint8_t pi<< 24) CONV_FUNC(AV_SAMPLE_FMT_S64, int64_t, AV_SAMPLE_FMT_U8,(uint64_t)((*(constuint8_t *) pi-0x80U))<< 56) CONV_FUNC(AV_SAMPLE_FMT_FLT, float, AV_SAMPLE_FMT_U8,(*(constuint8_t *) pi-0x80)*(1.0f/(1<< 7))) CONV_FUNC(AV_SAMPLE_FMT_DBL, double, AV_SAMPLE_FMT_U8,(*(constuint8_t *) pi-0x80)*(1.0/(1<< 7))) CONV_FUNC(AV_SAMPLE_FMT_U8, uint8_t, AV_SAMPLE_FMT_S16,(*(constint16_t *) pi >>8)+0x80) CONV_FUNC(AV_SAMPLE_FMT_S64, int64_t, AV_SAMPLE_FMT_S16,(uint64_t)(*(constint16_t *) pi)<< 48) CONV_FUNC(AV_SAMPLE_FMT_FLT, float, AV_SAMPLE_FMT_S16,*(constint16_t *) pi *(1.0f/(1<< 15))) CONV_FUNC(AV_SAMPLE_FMT_DBL, double, AV_SAMPLE_FMT_S16,*(constint16_t *) pi *(1.0/(1<< 15))) CONV_FUNC(AV_SAMPLE_FMT_U8, uint8_t, AV_SAMPLE_FMT_S32,(*(constint32_t *) pi >>24)+0x80) CONV_FUNC(AV_SAMPLE_FMT_S64, int64_t, AV_SAMPLE_FMT_S32,(uint64_t)(*(constint32_t *) pi)<< 32) CONV_FUNC(AV_SAMPLE_FMT_FLT, float, AV_SAMPLE_FMT_S32,*(constint32_t *) pi *(1.0f/(1U<< 31))) CONV_FUNC(AV_SAMPLE_FMT_DBL, double, AV_SAMPLE_FMT_S32,*(constint32_t *) pi *(1.0/(1U<< 31))) CONV_FUNC(AV_SAMPLE_FMT_U8, uint8_t, AV_SAMPLE_FMT_S64,(*(constint64_t *) pi >>56)+0x80) CONV_FUNC(AV_SAMPLE_FMT_FLT, float, AV_SAMPLE_FMT_S64,*(constint64_t *) pi *(1.0f/(INT64_C(1)<< 63))) CONV_FUNC(AV_SAMPLE_FMT_DBL, double, AV_SAMPLE_FMT_S64,*(constint64_t *) pi *(1.0/(INT64_C(1)<< 63))) CONV_FUNC(AV_SAMPLE_FMT_U8, uint8_t, AV_SAMPLE_FMT_FLT, av_clip_uint8(lrintf(*(constfloat *) pi *(1<< 7))+0x80)) CONV_FUNC(AV_SAMPLE_FMT_S16, int16_t, AV_SAMPLE_FMT_FLT, av_clip_int16(lrintf(*(constfloat *) pi *(1<< 15)))) CONV_FUNC(AV_SAMPLE_FMT_S32, int32_t, AV_SAMPLE_FMT_FLT, av_clipl_int32(llrintf(*(constfloat *) pi *(1U<< 31)))) CONV_FUNC(AV_SAMPLE_FMT_S64, int64_t, AV_SAMPLE_FMT_FLT, llrintf(*(constfloat *) pi *(INT64_C(1)<< 63))) CONV_FUNC(AV_SAMPLE_FMT_U8, uint8_t, AV_SAMPLE_FMT_DBL, av_clip_uint8(lrint(*(constdouble *) pi *(1<< 7))+0x80)) CONV_FUNC(AV_SAMPLE_FMT_S16, int16_t, AV_SAMPLE_FMT_DBL, av_clip_int16(lrint(*(constdouble *) pi *(1<< 15)))) CONV_FUNC(AV_SAMPLE_FMT_S32, int32_t, AV_SAMPLE_FMT_DBL, av_clipl_int32(llrint(*(constdouble *) pi *(1U<< 31)))) CONV_FUNC(AV_SAMPLE_FMT_S64, int64_t, AV_SAMPLE_FMT_DBL, llrint(*(constdouble *) pi *(INT64_C(1)<< 63)))#defineFMT_PAIR_FUNC(out, in) staticconv_func_type *constfmt_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),};staticvoidcpy1(uint8_t **dst, constuint8_t **src, intlen){memcpy(*dst,*src, len);}staticvoidcpy2(uint8_t **dst, constuint8_t **src, intlen){memcpy(*dst,*src, 2 *len);}staticvoidcpy4(uint8_t **dst, constuint8_t **src, intlen){memcpy(*dst,*src, 4 *len);}staticvoidcpy8(uint8_t **dst, constuint8_t **src, intlen){memcpy(*dst,*src, 8 *len);}AudioConvert *swri_audio_convert_alloc(enumAVSampleFormatout_fmt, enumAVSampleFormatin_fmt, intchannels, constint *ch_map, intflags){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) returnNULL;ctx=av_mallocz(sizeof(*ctx));if(!ctx) returnNULL;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)){case1:ctx->simd_f=cpy1;break;case2:ctx->simd_f=cpy2;break;case4:ctx->simd_f=cpy4;break;case8: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);returnctx;}voidswri_audio_convert_free(AudioConvert **ctx){av_freep(ctx);}intswri_audio_convert(AudioConvert *ctx, AudioData *out, AudioData *in, intlen){intch;intoff=0;constintos=(out->planar?1:out->ch_count)*out->bps;unsignedmisaligned=0;av_assert0(ctx->channels==out->ch_count);if(ctx->in_simd_align_mask){intplanes=in->planar?in->ch_count:1;unsignedm=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){intplanes=out->planar?out->ch_count:1;unsignedm=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){intplanes=out->planar?out->ch_count:1;for(ch=0;ch< planes;ch++){ctx->simd_f(out-> ch ch
Definition: audioconvert.c:56
static const int16_t interplay_delta_table[]
Definition: dpcm.c:52
static AVFrame * frame
uint8_t * data
Definition: avcodec.h:1671
static av_always_inline void bytestream2_skipu(GetByteContext *g, unsigned int size)
Definition: bytestream.h:170
#define av_log(a,...)
static void predictor(uint8_t *src, int size)
Definition: exr.c:266
enum AVCodecID id
Definition: avcodec.h:3736
#define AV_LOG_ERROR
Something went wrong and cannot losslessly be recovered.
Definition: log.h:176
static const int8_t sol_table_new[16]
Definition: dpcm.c:93
#define AVERROR(e)
Definition: error.h:43
static int dpcm_decode_frame(AVCodecContext *avctx, void *data, int *got_frame_ptr, AVPacket *avpkt)
Definition: dpcm.c:194
static int square(int x)
Definition: roqvideoenc.c:113
int16_t array[256]
Definition: dpcm.c:47
int n
Definition: avisynth_c.h:684
Libavcodec external API header.
static const int16_t sol_table_16[128]
Definition: dpcm.c:98
main external API structure.
Definition: avcodec.h:1746
unsigned int codec_tag
fourcc (LSB first, so "ABCD" -> ('D'<<24) + ('C'<<16) + ('B'<<8) + 'A').
Definition: avcodec.h:1778
int ff_get_buffer(AVCodecContext *avctx, AVFrame *frame, int flags)
Get a buffer for a frame.
Definition: decode.c:1663
static av_const int sign_extend(int val, unsigned bits)
Definition: mathops.h:130
uint8_t * data[AV_NUM_DATA_POINTERS]
pointer to the picture/channel planes.
Definition: frame.h:215
#define DPCM_DECODER(id_, name_, long_name_)
Definition: dpcm.c:382
common internal api header.
if(ret< 0)
Definition: vf_mcdeint.c:282
signed 16 bits
Definition: samplefmt.h:61
static av_cold int dpcm_decode_init(AVCodecContext *avctx)
Definition: dpcm.c:115
void * priv_data
Definition: avcodec.h:1788
static av_always_inline int diff(const uint32_t a, const uint32_t b)
int channels
number of audio channels
Definition: avcodec.h:2509
FILE * out
Definition: movenc.c:54
This structure stores compressed data.
Definition: avcodec.h:1648
int sample[2]
previous sample (for SOL_DPCM)
Definition: dpcm.c:48
int nb_samples
number of audio samples (per channel) described by this frame
Definition: frame.h:267