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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 square_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->square_array[i ] = square;
134  s->square_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->square_array[i+128] = i < 0 ? -square: square;
160  }
161  break;
162 
163  default:
164  break;
165  }
166 
167  if (avctx->codec->id == AV_CODEC_ID_SOL_DPCM && avctx->codec_tag != 3)
168  avctx->sample_fmt = AV_SAMPLE_FMT_U8;
169  else
170  avctx->sample_fmt = AV_SAMPLE_FMT_S16;
171 
172  return 0;
173 }
174 
175 
176 static int dpcm_decode_frame(AVCodecContext *avctx, void *data,
177  int *got_frame_ptr, AVPacket *avpkt)
178 {
179  int buf_size = avpkt->size;
180  DPCMContext *s = avctx->priv_data;
181  AVFrame *frame = data;
182  int out = 0, ret;
183  int predictor[2];
184  int ch = 0;
185  int stereo = avctx->channels - 1;
186  int16_t *output_samples, *samples_end;
187  GetByteContext gb;
188 
189  if (stereo && (buf_size & 1))
190  buf_size--;
191  bytestream2_init(&gb, avpkt->data, buf_size);
192 
193  /* calculate output size */
194  switch(avctx->codec->id) {
196  out = buf_size - 8;
197  break;
199  out = buf_size - 6 - avctx->channels;
200  break;
202  out = buf_size - 2 * avctx->channels;
203  break;
205  if (avctx->codec_tag != 3)
206  out = buf_size * 2;
207  else
208  out = buf_size;
209  break;
211  out = buf_size;
212  break;
213  }
214  if (out <= 0) {
215  av_log(avctx, AV_LOG_ERROR, "packet is too small\n");
216  return AVERROR(EINVAL);
217  }
218  if (out % avctx->channels) {
219  av_log(avctx, AV_LOG_WARNING, "channels have differing number of samples\n");
220  }
221 
222  /* get output buffer */
223  frame->nb_samples = (out + avctx->channels - 1) / avctx->channels;
224  if ((ret = ff_get_buffer(avctx, frame, 0)) < 0)
225  return ret;
226  output_samples = (int16_t *)frame->data[0];
227  samples_end = output_samples + out;
228 
229  switch(avctx->codec->id) {
230 
232  bytestream2_skipu(&gb, 6);
233 
234  if (stereo) {
235  predictor[1] = sign_extend(bytestream2_get_byteu(&gb) << 8, 16);
236  predictor[0] = sign_extend(bytestream2_get_byteu(&gb) << 8, 16);
237  } else {
238  predictor[0] = sign_extend(bytestream2_get_le16u(&gb), 16);
239  }
240 
241  /* decode the samples */
242  while (output_samples < samples_end) {
243  predictor[ch] += s->square_array[bytestream2_get_byteu(&gb)];
244  predictor[ch] = av_clip_int16(predictor[ch]);
245  *output_samples++ = predictor[ch];
246 
247  /* toggle channel */
248  ch ^= stereo;
249  }
250  break;
251 
253  bytestream2_skipu(&gb, 6); /* skip over the stream mask and stream length */
254 
255  for (ch = 0; ch < avctx->channels; ch++) {
256  predictor[ch] = sign_extend(bytestream2_get_le16u(&gb), 16);
257  *output_samples++ = predictor[ch];
258  }
259 
260  ch = 0;
261  while (output_samples < samples_end) {
262  predictor[ch] += interplay_delta_table[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  {
273  int shift[2] = { 4, 4 };
274 
275  for (ch = 0; ch < avctx->channels; ch++)
276  predictor[ch] = sign_extend(bytestream2_get_le16u(&gb), 16);
277 
278  ch = 0;
279  while (output_samples < samples_end) {
280  int diff = bytestream2_get_byteu(&gb);
281  int n = diff & 3;
282 
283  if (n == 3)
284  shift[ch]++;
285  else
286  shift[ch] -= (2 * n);
287  diff = sign_extend((diff &~ 3) << 8, 16);
288 
289  /* saturate the shifter to a lower limit of 0 */
290  if (shift[ch] < 0)
291  shift[ch] = 0;
292 
293  diff >>= shift[ch];
294  predictor[ch] += diff;
295 
296  predictor[ch] = av_clip_int16(predictor[ch]);
297  *output_samples++ = predictor[ch];
298 
299  /* toggle channel */
300  ch ^= stereo;
301  }
302  break;
303  }
305  if (avctx->codec_tag != 3) {
306  uint8_t *output_samples_u8 = frame->data[0],
307  *samples_end_u8 = output_samples_u8 + out;
308  while (output_samples_u8 < samples_end_u8) {
309  int n = bytestream2_get_byteu(&gb);
310 
311  s->sample[0] += s->sol_table[n >> 4];
312  s->sample[0] = av_clip_uint8(s->sample[0]);
313  *output_samples_u8++ = s->sample[0];
314 
315  s->sample[stereo] += s->sol_table[n & 0x0F];
316  s->sample[stereo] = av_clip_uint8(s->sample[stereo]);
317  *output_samples_u8++ = s->sample[stereo];
318  }
319  } else {
320  while (output_samples < samples_end) {
321  int n = bytestream2_get_byteu(&gb);
322  if (n & 0x80) s->sample[ch] -= sol_table_16[n & 0x7F];
323  else s->sample[ch] += sol_table_16[n & 0x7F];
324  s->sample[ch] = av_clip_int16(s->sample[ch]);
325  *output_samples++ = s->sample[ch];
326  /* toggle channel */
327  ch ^= stereo;
328  }
329  }
330  break;
331 
333  while (output_samples < samples_end) {
334  int8_t n = bytestream2_get_byteu(&gb);
335 
336  if (!(n & 1))
337  s->sample[ch] = 0;
338  s->sample[ch] += s->square_array[n + 128];
339  s->sample[ch] = av_clip_int16(s->sample[ch]);
340  *output_samples++ = s->sample[ch];
341  ch ^= stereo;
342  }
343  break;
344  }
345 
346  *got_frame_ptr = 1;
347 
348  return avpkt->size;
349 }
350 
351 #define DPCM_DECODER(id_, name_, long_name_) \
352 AVCodec ff_ ## name_ ## _decoder = { \
353  .name = #name_, \
354  .long_name = NULL_IF_CONFIG_SMALL(long_name_), \
355  .type = AVMEDIA_TYPE_AUDIO, \
356  .id = id_, \
357  .priv_data_size = sizeof(DPCMContext), \
358  .init = dpcm_decode_init, \
359  .decode = dpcm_decode_frame, \
360  .capabilities = AV_CODEC_CAP_DR1, \
361 }
362 
363 DPCM_DECODER(AV_CODEC_ID_INTERPLAY_DPCM, interplay_dpcm, "DPCM Interplay");
364 DPCM_DECODER(AV_CODEC_ID_ROQ_DPCM, roq_dpcm, "DPCM id RoQ");
365 DPCM_DECODER(AV_CODEC_ID_SDX2_DPCM, sdx2_dpcm, "DPCM Squareroot-Delta-Exact");
366 DPCM_DECODER(AV_CODEC_ID_SOL_DPCM, sol_dpcm, "DPCM Sol");
367 DPCM_DECODER(AV_CODEC_ID_XAN_DPCM, xan_dpcm, "DPCM Xan");
const struct AVCodec * codec
Definition: avcodec.h:1691
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:190
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:1608
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:2452
uint8_t
#define av_cold
Definition: attributes.h:82
AV_SAMPLE_FMT_U8
static const int16_t interplay_delta_table[]
Definition: dpcm.c:52
static AVFrame * frame
uint8_t * data
Definition: avcodec.h:1607
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:254
enum AVCodecID id
Definition: avcodec.h:3621
#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:176
static int square(int x)
Definition: roqvideoenc.c:113
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:1682
int ff_get_buffer(AVCodecContext *avctx, AVFrame *frame, int flags)
Get a buffer for a frame.
Definition: utils.c:948
unsigned int codec_tag
fourcc (LSB first, so "ABCD" -> ('D'<<24) + ('C'<<16) + ('B'<<8) + 'A').
Definition: avcodec.h:1714
static av_const int sign_extend(int val, unsigned bits)
Definition: mathops.h:139
uint8_t * data[AV_NUM_DATA_POINTERS]
pointer to the picture/channel planes.
Definition: frame.h:204
#define DPCM_DECODER(id_, name_, long_name_)
Definition: dpcm.c:351
int16_t square_array[256]
Definition: dpcm.c:47
common internal api header.
if(ret< 0)
Definition: vf_mcdeint.c:282
signed 16 bits
Definition: samplefmt.h:61
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_YASM &&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 av_cold int dpcm_decode_init(AVCodecContext *avctx)
Definition: dpcm.c:115
void * priv_data
Definition: avcodec.h:1724
static av_always_inline int diff(const uint32_t a, const uint32_t b)
int channels
number of audio channels
Definition: avcodec.h:2445
FILE * out
Definition: movenc.c:54
This structure stores compressed data.
Definition: avcodec.h:1584
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:247