FFmpeg
dstdec.c
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1 /*
2  * Direct Stream Transfer (DST) decoder
3  * Copyright (c) 2014 Peter Ross <pross@xvid.org>
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  * Direct Stream Transfer (DST) decoder
25  * ISO/IEC 14496-3 Part 3 Subpart 10: Technical description of lossless coding of oversampled audio
26  */
27 
28 #include "libavutil/avassert.h"
29 #include "libavutil/intreadwrite.h"
30 #include "internal.h"
31 #include "get_bits.h"
32 #include "avcodec.h"
33 #include "golomb.h"
34 #include "mathops.h"
35 #include "dsd.h"
36 
37 #define DST_MAX_CHANNELS 6
38 #define DST_MAX_ELEMENTS (2 * DST_MAX_CHANNELS)
39 
40 #define DSD_FS44(sample_rate) (sample_rate * 8LL / 44100)
41 
42 #define DST_SAMPLES_PER_FRAME(sample_rate) (588 * DSD_FS44(sample_rate))
43 
44 static const int8_t fsets_code_pred_coeff[3][3] = {
45  { -8 },
46  { -16, 8 },
47  { -9, -5, 6 },
48 };
49 
50 static const int8_t probs_code_pred_coeff[3][3] = {
51  { -8 },
52  { -16, 8 },
53  { -24, 24, -8 },
54 };
55 
56 typedef struct ArithCoder {
57  unsigned int a;
58  unsigned int c;
59  int overread;
60 } ArithCoder;
61 
62 typedef struct Table {
63  unsigned int elements;
64  unsigned int length[DST_MAX_ELEMENTS];
66 } Table;
67 
68 typedef struct DSTContext {
69  AVClass *class;
70 
73  Table fsets, probs;
75  DECLARE_ALIGNED(16, int16_t, filter)[DST_MAX_ELEMENTS][16][256];
77 } DSTContext;
78 
80 {
81  DSTContext *s = avctx->priv_data;
82  int i;
83 
84  if (avctx->channels > DST_MAX_CHANNELS) {
85  avpriv_request_sample(avctx, "Channel count %d", avctx->channels);
86  return AVERROR_PATCHWELCOME;
87  }
88 
90 
91  for (i = 0; i < avctx->channels; i++)
92  memset(s->dsdctx[i].buf, 0x69, sizeof(s->dsdctx[i].buf));
93 
95 
96  return 0;
97 }
98 
99 static int read_map(GetBitContext *gb, Table *t, unsigned int map[DST_MAX_CHANNELS], int channels)
100 {
101  int ch;
102  t->elements = 1;
103  map[0] = 0;
104  if (!get_bits1(gb)) {
105  for (ch = 1; ch < channels; ch++) {
106  int bits = av_log2(t->elements) + 1;
107  map[ch] = get_bits(gb, bits);
108  if (map[ch] == t->elements) {
109  t->elements++;
110  if (t->elements >= DST_MAX_ELEMENTS)
111  return AVERROR_INVALIDDATA;
112  } else if (map[ch] > t->elements) {
113  return AVERROR_INVALIDDATA;
114  }
115  }
116  } else {
117  memset(map, 0, sizeof(*map) * DST_MAX_CHANNELS);
118  }
119  return 0;
120 }
121 
123 {
124  int v = get_ur_golomb(gb, k, get_bits_left(gb), 0);
125  if (v && get_bits1(gb))
126  v = -v;
127  return v;
128 }
129 
130 static void read_uncoded_coeff(GetBitContext *gb, int *dst, unsigned int elements,
131  int coeff_bits, int is_signed, int offset)
132 {
133  int i;
134 
135  for (i = 0; i < elements; i++) {
136  dst[i] = (is_signed ? get_sbits(gb, coeff_bits) : get_bits(gb, coeff_bits)) + offset;
137  }
138 }
139 
140 static int read_table(GetBitContext *gb, Table *t, const int8_t code_pred_coeff[3][3],
141  int length_bits, int coeff_bits, int is_signed, int offset)
142 {
143  unsigned int i, j, k;
144  for (i = 0; i < t->elements; i++) {
145  t->length[i] = get_bits(gb, length_bits) + 1;
146  if (!get_bits1(gb)) {
147  read_uncoded_coeff(gb, t->coeff[i], t->length[i], coeff_bits, is_signed, offset);
148  } else {
149  int method = get_bits(gb, 2), lsb_size;
150  if (method == 3)
151  return AVERROR_INVALIDDATA;
152 
153  read_uncoded_coeff(gb, t->coeff[i], method + 1, coeff_bits, is_signed, offset);
154 
155  lsb_size = get_bits(gb, 3);
156  for (j = method + 1; j < t->length[i]; j++) {
157  int c, x = 0;
158  for (k = 0; k < method + 1; k++)
159  x += code_pred_coeff[method][k] * t->coeff[i][j - k - 1];
160  c = get_sr_golomb_dst(gb, lsb_size);
161  if (x >= 0)
162  c -= (x + 4) / 8;
163  else
164  c += (-x + 3) / 8;
165  if (!is_signed) {
166  if (c < offset || c >= offset + (1<<coeff_bits))
167  return AVERROR_INVALIDDATA;
168  }
169  t->coeff[i][j] = c;
170  }
171  }
172  }
173  return 0;
174 }
175 
176 static void ac_init(ArithCoder *ac, GetBitContext *gb)
177 {
178  ac->a = 4095;
179  ac->c = get_bits(gb, 12);
180  ac->overread = 0;
181 }
182 
183 static av_always_inline void ac_get(ArithCoder *ac, GetBitContext *gb, int p, int *e)
184 {
185  unsigned int k = (ac->a >> 8) | ((ac->a >> 7) & 1);
186  unsigned int q = k * p;
187  unsigned int a_q = ac->a - q;
188 
189  *e = ac->c < a_q;
190  if (*e) {
191  ac->a = a_q;
192  } else {
193  ac->a = q;
194  ac->c -= a_q;
195  }
196 
197  if (ac->a < 2048) {
198  int n = 11 - av_log2(ac->a);
199  ac->a <<= n;
200  if (get_bits_left(gb) < n)
201  ac->overread ++;
202  ac->c = (ac->c << n) | get_bits(gb, n);
203  }
204 }
205 
207 {
208  return (ff_reverse[c & 127] >> 1) + 1;
209 }
210 
211 static void build_filter(int16_t table[DST_MAX_ELEMENTS][16][256], const Table *fsets)
212 {
213  int i, j, k, l;
214 
215  for (i = 0; i < fsets->elements; i++) {
216  int length = fsets->length[i];
217 
218  for (j = 0; j < 16; j++) {
219  int total = av_clip(length - j * 8, 0, 8);
220 
221  for (k = 0; k < 256; k++) {
222  int v = 0;
223 
224  for (l = 0; l < total; l++)
225  v += (((k >> l) & 1) * 2 - 1) * fsets->coeff[i][j * 8 + l];
226  table[i][j][k] = v;
227  }
228  }
229  }
230 }
231 
232 static int decode_frame(AVCodecContext *avctx, void *data,
233  int *got_frame_ptr, AVPacket *avpkt)
234 {
235  unsigned samples_per_frame = DST_SAMPLES_PER_FRAME(avctx->sample_rate);
236  unsigned map_ch_to_felem[DST_MAX_CHANNELS];
237  unsigned map_ch_to_pelem[DST_MAX_CHANNELS];
238  unsigned i, ch, same_map, dst_x_bit;
239  unsigned half_prob[DST_MAX_CHANNELS];
240  const int channels = avctx->channels;
241  DSTContext *s = avctx->priv_data;
242  GetBitContext *gb = &s->gb;
243  ArithCoder *ac = &s->ac;
244  AVFrame *frame = data;
245  uint8_t *dsd;
246  float *pcm;
247  int ret;
248 
249  if (avpkt->size <= 1)
250  return AVERROR_INVALIDDATA;
251 
252  frame->nb_samples = samples_per_frame / 8;
253  if ((ret = ff_get_buffer(avctx, frame, 0)) < 0)
254  return ret;
255  dsd = frame->data[0];
256  pcm = (float *)frame->data[0];
257 
258  if ((ret = init_get_bits8(gb, avpkt->data, avpkt->size)) < 0)
259  return ret;
260 
261  if (!get_bits1(gb)) {
262  skip_bits1(gb);
263  if (get_bits(gb, 6))
264  return AVERROR_INVALIDDATA;
265  memcpy(frame->data[0], avpkt->data + 1, FFMIN(avpkt->size - 1, frame->nb_samples * avctx->channels));
266  goto dsd;
267  }
268 
269  /* Segmentation (10.4, 10.5, 10.6) */
270 
271  if (!get_bits1(gb)) {
272  avpriv_request_sample(avctx, "Not Same Segmentation");
273  return AVERROR_PATCHWELCOME;
274  }
275 
276  if (!get_bits1(gb)) {
277  avpriv_request_sample(avctx, "Not Same Segmentation For All Channels");
278  return AVERROR_PATCHWELCOME;
279  }
280 
281  if (!get_bits1(gb)) {
282  avpriv_request_sample(avctx, "Not End Of Channel Segmentation");
283  return AVERROR_PATCHWELCOME;
284  }
285 
286  /* Mapping (10.7, 10.8, 10.9) */
287 
288  same_map = get_bits1(gb);
289 
290  if ((ret = read_map(gb, &s->fsets, map_ch_to_felem, avctx->channels)) < 0)
291  return ret;
292 
293  if (same_map) {
294  s->probs.elements = s->fsets.elements;
295  memcpy(map_ch_to_pelem, map_ch_to_felem, sizeof(map_ch_to_felem));
296  } else {
297  avpriv_request_sample(avctx, "Not Same Mapping");
298  if ((ret = read_map(gb, &s->probs, map_ch_to_pelem, avctx->channels)) < 0)
299  return ret;
300  }
301 
302  /* Half Probability (10.10) */
303 
304  for (ch = 0; ch < avctx->channels; ch++)
305  half_prob[ch] = get_bits1(gb);
306 
307  /* Filter Coef Sets (10.12) */
308 
309  ret = read_table(gb, &s->fsets, fsets_code_pred_coeff, 7, 9, 1, 0);
310  if (ret < 0)
311  return ret;
312 
313  /* Probability Tables (10.13) */
314 
315  ret = read_table(gb, &s->probs, probs_code_pred_coeff, 6, 7, 0, 1);
316  if (ret < 0)
317  return ret;
318 
319  /* Arithmetic Coded Data (10.11) */
320 
321  if (get_bits1(gb))
322  return AVERROR_INVALIDDATA;
323  ac_init(ac, gb);
324 
325  build_filter(s->filter, &s->fsets);
326 
327  memset(s->status, 0xAA, sizeof(s->status));
328  memset(dsd, 0, frame->nb_samples * 4 * avctx->channels);
329 
330  ac_get(ac, gb, prob_dst_x_bit(s->fsets.coeff[0][0]), &dst_x_bit);
331 
332  for (i = 0; i < samples_per_frame; i++) {
333  for (ch = 0; ch < channels; ch++) {
334  const unsigned felem = map_ch_to_felem[ch];
335  int16_t (*filter)[256] = s->filter[felem];
336  uint8_t *status = s->status[ch];
337  int prob, residual, v;
338 
339 #define F(x) filter[(x)][status[(x)]]
340  const int16_t predict = F( 0) + F( 1) + F( 2) + F( 3) +
341  F( 4) + F( 5) + F( 6) + F( 7) +
342  F( 8) + F( 9) + F(10) + F(11) +
343  F(12) + F(13) + F(14) + F(15);
344 #undef F
345 
346  if (!half_prob[ch] || i >= s->fsets.length[felem]) {
347  unsigned pelem = map_ch_to_pelem[ch];
348  unsigned index = FFABS(predict) >> 3;
349  prob = s->probs.coeff[pelem][FFMIN(index, s->probs.length[pelem] - 1)];
350  } else {
351  prob = 128;
352  }
353 
354  if (ac->overread > 16)
355  return AVERROR_INVALIDDATA;
356 
357  ac_get(ac, gb, prob, &residual);
358  v = ((predict >> 15) ^ residual) & 1;
359  dsd[((i >> 3) * channels + ch) << 2] |= v << (7 - (i & 0x7 ));
360 
361  AV_WL64A(status + 8, (AV_RL64A(status + 8) << 1) | ((AV_RL64A(status) >> 63) & 1));
362  AV_WL64A(status, (AV_RL64A(status) << 1) | v);
363  }
364  }
365 
366 dsd:
367  for (i = 0; i < avctx->channels; i++) {
368  ff_dsd2pcm_translate(&s->dsdctx[i], frame->nb_samples, 0,
369  frame->data[0] + i * 4,
370  avctx->channels * 4, pcm + i, avctx->channels);
371  }
372 
373  *got_frame_ptr = 1;
374 
375  return avpkt->size;
376 }
377 
379  .name = "dst",
380  .long_name = NULL_IF_CONFIG_SMALL("DST (Digital Stream Transfer)"),
381  .type = AVMEDIA_TYPE_AUDIO,
382  .id = AV_CODEC_ID_DST,
383  .priv_data_size = sizeof(DSTContext),
384  .init = decode_init,
385  .decode = decode_frame,
386  .capabilities = AV_CODEC_CAP_DR1,
387  .sample_fmts = (const enum AVSampleFormat[]) { AV_SAMPLE_FMT_FLT,
389 };
#define AVERROR_INVALIDDATA
Invalid data found when processing input.
Definition: error.h:59
static av_always_inline int get_sr_golomb_dst(GetBitContext *gb, unsigned int k)
Definition: dstdec.c:122
This structure describes decoded (raw) audio or video data.
Definition: frame.h:295
int overread
Definition: dstdec.c:59
ptrdiff_t const GLvoid * data
Definition: opengl_enc.c:100
static unsigned int get_bits(GetBitContext *s, int n)
Read 1-25 bits.
Definition: get_bits.h:379
Definition: dstdec.c:62
unsigned int a
Definition: dstdec.c:57
#define DST_MAX_ELEMENTS
Definition: dstdec.c:38
static av_cold int init(AVCodecContext *avctx)
Definition: avrndec.c:35
const uint8_t ff_reverse[256]
Definition: reverse.c:23
#define avpriv_request_sample(...)
#define DST_MAX_CHANNELS
Definition: dstdec.c:37
channels
Definition: aptx.c:30
int size
Definition: avcodec.h:1481
int av_log2(unsigned v)
Definition: intmath.c:26
Table fsets
Definition: dstdec.c:73
av_cold void ff_init_dsd_data(void)
Definition: dsd.c:46
static int read_map(GetBitContext *gb, Table *t, unsigned int map[DST_MAX_CHANNELS], int channels)
Definition: dstdec.c:99
static av_always_inline void predict(PredictorState *ps, float *coef, int output_enable)
Definition: aacdec.c:174
AVCodec.
Definition: avcodec.h:3492
static int get_sbits(GetBitContext *s, int n)
Definition: get_bits.h:359
static void decode(AVCodecContext *dec_ctx, AVPacket *pkt, AVFrame *frame, FILE *outfile)
Definition: decode_audio.c:71
static int decode_frame(AVCodecContext *avctx, void *data, int *got_frame_ptr, AVPacket *avpkt)
Definition: dstdec.c:232
enum AVSampleFormat sample_fmt
audio sample format
Definition: avcodec.h:2236
uint8_t
#define av_cold
Definition: attributes.h:82
void ff_dsd2pcm_translate(DSDContext *s, size_t samples, int lsbf, const uint8_t *src, ptrdiff_t src_stride, float *dst, ptrdiff_t dst_stride)
Definition: dsd.c:55
it s the only field you need to keep assuming you have a context There is some magic you don t need to care about around this just let it vf offset
static const int8_t fsets_code_pred_coeff[3][3]
Definition: dstdec.c:44
static void ac_init(ArithCoder *ac, GetBitContext *gb)
Definition: dstdec.c:176
#define DECLARE_ALIGNED(n, t, v)
Declare a variable that is aligned in memory.
Definition: mem.h:112
uint8_t * data
Definition: avcodec.h:1480
Table probs
Definition: dstdec.c:73
bitstream reader API header.
static int get_ur_golomb(GetBitContext *gb, int k, int limit, int esc_len)
read unsigned golomb rice code (ffv1).
Definition: golomb.h:374
static const uint16_t table[]
Definition: prosumer.c:206
#define prob(name, subs,...)
Definition: cbs_vp9.c:374
static int get_bits_left(GetBitContext *gb)
Definition: get_bits.h:849
#define i(width, name, range_min, range_max)
Definition: cbs_h2645.c:259
static void read_uncoded_coeff(GetBitContext *gb, int *dst, unsigned int elements, int coeff_bits, int is_signed, int offset)
Definition: dstdec.c:130
int16_t filter[DST_MAX_ELEMENTS][16][256]
Definition: dstdec.c:75
filter_frame For filters that do not use the this method is called when a frame is pushed to the filter s input It can be called at any time except in a reentrant way If the input frame is enough to produce then the filter should push the output frames on the output link immediately As an exception to the previous rule if the input frame is enough to produce several output frames then the filter needs output only at least one per link The additional frames can be left buffered in the filter
#define NULL_IF_CONFIG_SMALL(x)
Return NULL if CONFIG_SMALL is true, otherwise the argument without modification. ...
Definition: internal.h:186
#define AV_RL64A(p)
Definition: intreadwrite.h:554
simple assert() macros that are a bit more flexible than ISO C assert().
GLsizei GLsizei * length
Definition: opengl_enc.c:114
const char * name
Name of the codec implementation.
Definition: avcodec.h:3499
uint8_t bits
Definition: vp3data.h:202
uint64_t residual
Definition: dirac_vlc.h:29
static av_cold int decode_init(AVCodecContext *avctx)
Definition: dstdec.c:79
static const int8_t probs_code_pred_coeff[3][3]
Definition: dstdec.c:50
#define FFMIN(a, b)
Definition: common.h:96
static const ElemCat * elements[ELEMENT_COUNT]
Definition: signature.h:566
ArithCoder ac
Definition: dstdec.c:72
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
Per-channel buffer.
Definition: dsd.h:42
#define FFABS(a)
Absolute value, Note, INT_MIN / INT64_MIN result in undefined behavior as they are not representable ...
Definition: common.h:72
#define s(width, name)
Definition: cbs_vp9.c:257
int n
Definition: avisynth_c.h:760
unsigned int length[DST_MAX_ELEMENTS]
Definition: dstdec.c:64
AVCodec ff_dst_decoder
Definition: dstdec.c:378
if(ret)
#define AVERROR_PATCHWELCOME
Not yet implemented in FFmpeg, patches welcome.
Definition: error.h:62
unsigned int c
Definition: dstdec.c:58
Libavcodec external API header.
AVSampleFormat
Audio sample formats.
Definition: samplefmt.h:58
int sample_rate
samples per second
Definition: avcodec.h:2228
static int init_get_bits8(GetBitContext *s, const uint8_t *buffer, int byte_size)
Initialize GetBitContext.
Definition: get_bits.h:677
static uint8_t prob_dst_x_bit(int c)
Definition: dstdec.c:206
int coeff[DST_MAX_ELEMENTS][128]
Definition: dstdec.c:65
static av_always_inline void ac_get(ArithCoder *ac, GetBitContext *gb, int p, int *e)
Definition: dstdec.c:183
main external API structure.
Definition: avcodec.h:1568
int ff_get_buffer(AVCodecContext *avctx, AVFrame *frame, int flags)
Get a buffer for a frame.
Definition: decode.c:1968
static unsigned int get_bits1(GetBitContext *s)
Definition: get_bits.h:498
static void skip_bits1(GetBitContext *s)
Definition: get_bits.h:538
Describe the class of an AVClass context structure.
Definition: log.h:67
int index
Definition: gxfenc.c:89
they must not be accessed directly The fifo field contains the frames that are queued in the input for processing by the filter The status_in and status_out fields contains the queued status(EOF or error) of the link
const VDPAUPixFmtMap * map
uint8_t pi<< 24) CONV_FUNC(AV_SAMPLE_FMT_S64, int64_t, AV_SAMPLE_FMT_U8,(uint64_t)((*(const uint8_t *) pi-0x80U))<< 56) CONV_FUNC(AV_SAMPLE_FMT_FLT, float, AV_SAMPLE_FMT_U8,(*(const uint8_t *) pi-0x80)*(1.0f/(1<< 7))) CONV_FUNC(AV_SAMPLE_FMT_DBL, double, AV_SAMPLE_FMT_U8,(*(const uint8_t *) pi-0x80)*(1.0/(1<< 7))) CONV_FUNC(AV_SAMPLE_FMT_U8, uint8_t, AV_SAMPLE_FMT_S16,(*(const int16_t *) pi >>8)+0x80) CONV_FUNC(AV_SAMPLE_FMT_S64, int64_t, AV_SAMPLE_FMT_S16,(uint64_t)(*(const int16_t *) pi)<< 48) CONV_FUNC(AV_SAMPLE_FMT_FLT, float, AV_SAMPLE_FMT_S16,*(const int16_t *) pi *(1.0f/(1<< 15))) CONV_FUNC(AV_SAMPLE_FMT_DBL, double, AV_SAMPLE_FMT_S16,*(const int16_t *) pi *(1.0/(1<< 15))) CONV_FUNC(AV_SAMPLE_FMT_U8, uint8_t, AV_SAMPLE_FMT_S32,(*(const int32_t *) pi >>24)+0x80) CONV_FUNC(AV_SAMPLE_FMT_S64, int64_t, AV_SAMPLE_FMT_S32,(uint64_t)(*(const int32_t *) pi)<< 32) CONV_FUNC(AV_SAMPLE_FMT_FLT, float, AV_SAMPLE_FMT_S32,*(const int32_t *) pi *(1.0f/(1U<< 31))) CONV_FUNC(AV_SAMPLE_FMT_DBL, double, AV_SAMPLE_FMT_S32,*(const int32_t *) pi *(1.0/(1U<< 31))) CONV_FUNC(AV_SAMPLE_FMT_U8, uint8_t, AV_SAMPLE_FMT_S64,(*(const int64_t *) pi >>56)+0x80) CONV_FUNC(AV_SAMPLE_FMT_FLT, float, AV_SAMPLE_FMT_S64,*(const int64_t *) pi *(1.0f/(UINT64_C(1)<< 63))) CONV_FUNC(AV_SAMPLE_FMT_DBL, double, AV_SAMPLE_FMT_S64,*(const int64_t *) pi *(1.0/(UINT64_C(1)<< 63))) CONV_FUNC(AV_SAMPLE_FMT_U8, uint8_t, AV_SAMPLE_FMT_FLT, av_clip_uint8(lrintf(*(const float *) pi *(1<< 7))+0x80)) CONV_FUNC(AV_SAMPLE_FMT_S16, int16_t, AV_SAMPLE_FMT_FLT, av_clip_int16(lrintf(*(const float *) pi *(1<< 15)))) CONV_FUNC(AV_SAMPLE_FMT_S32, int32_t, AV_SAMPLE_FMT_FLT, av_clipl_int32(llrintf(*(const float *) pi *(1U<< 31)))) CONV_FUNC(AV_SAMPLE_FMT_S64, int64_t, AV_SAMPLE_FMT_FLT, llrintf(*(const float *) pi *(UINT64_C(1)<< 63))) CONV_FUNC(AV_SAMPLE_FMT_U8, uint8_t, AV_SAMPLE_FMT_DBL, av_clip_uint8(lrint(*(const double *) pi *(1<< 7))+0x80)) CONV_FUNC(AV_SAMPLE_FMT_S16, int16_t, AV_SAMPLE_FMT_DBL, av_clip_int16(lrint(*(const double *) pi *(1<< 15)))) CONV_FUNC(AV_SAMPLE_FMT_S32, int32_t, AV_SAMPLE_FMT_DBL, av_clipl_int32(llrint(*(const double *) pi *(1U<< 31)))) CONV_FUNC(AV_SAMPLE_FMT_S64, int64_t, AV_SAMPLE_FMT_DBL, llrint(*(const double *) pi *(UINT64_C(1)<< 63)))#define FMT_PAIR_FUNC(out, in) static conv_func_type *const fmt_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),};static void cpy1(uint8_t **dst, const uint8_t **src, int len){memcpy(*dst,*src, len);}static void cpy2(uint8_t **dst, const uint8_t **src, int len){memcpy(*dst,*src, 2 *len);}static void cpy4(uint8_t **dst, const uint8_t **src, int len){memcpy(*dst,*src, 4 *len);}static void cpy8(uint8_t **dst, const uint8_t **src, int len){memcpy(*dst,*src, 8 *len);}AudioConvert *swri_audio_convert_alloc(enum AVSampleFormat out_fmt, enum AVSampleFormat in_fmt, int channels, const int *ch_map, int flags){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) return NULL;ctx=av_mallocz(sizeof(*ctx));if(!ctx) return NULL;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)){case 1:ctx->simd_f=cpy1;break;case 2:ctx->simd_f=cpy2;break;case 4:ctx->simd_f=cpy4;break;case 8: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);return ctx;}void swri_audio_convert_free(AudioConvert **ctx){av_freep(ctx);}int swri_audio_convert(AudioConvert *ctx, AudioData *out, AudioData *in, int len){int ch;int off=0;const int os=(out->planar?1:out->ch_count)*out->bps;unsigned misaligned=0;av_assert0(ctx->channels==out->ch_count);if(ctx->in_simd_align_mask){int planes=in->planar?in->ch_count:1;unsigned m=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){int planes=out->planar?out->ch_count:1;unsigned m=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){int planes=out->planar?out->ch_count:1;for(ch=0;ch< planes;ch++){ctx->simd_f(out-> ch ch
Definition: audioconvert.c:56
uint8_t * data[AV_NUM_DATA_POINTERS]
pointer to the picture/channel planes.
Definition: frame.h:309
unsigned int elements
Definition: dstdec.c:63
uint8_t buf[FIFOSIZE]
Definition: dsd.h:43
common internal api header.
static void build_filter(int16_t table[DST_MAX_ELEMENTS][16][256], const Table *fsets)
Definition: dstdec.c:211
uint8_t status[DST_MAX_CHANNELS][16]
Definition: dstdec.c:74
#define DST_SAMPLES_PER_FRAME(sample_rate)
Definition: dstdec.c:42
DSDContext dsdctx[DST_MAX_CHANNELS]
Definition: dstdec.c:76
void * priv_data
Definition: avcodec.h:1595
static int read_table(GetBitContext *gb, Table *t, const int8_t code_pred_coeff[3][3], int length_bits, int coeff_bits, int is_signed, int offset)
Definition: dstdec.c:140
int channels
number of audio channels
Definition: avcodec.h:2229
#define F(x)
static const double coeff[2][5]
Definition: vf_owdenoise.c:72
static enum AVSampleFormat sample_fmts[]
Definition: adpcmenc.c:701
#define av_always_inline
Definition: attributes.h:39
GetBitContext * gb
Definition: mss12.h:53
exp golomb vlc stuff
This structure stores compressed data.
Definition: avcodec.h:1457
int nb_samples
number of audio samples (per channel) described by this frame
Definition: frame.h:361
#define AV_CODEC_CAP_DR1
Codec uses get_buffer() for allocating buffers and supports custom allocators.
Definition: avcodec.h:984
GetBitContext gb
Definition: dstdec.c:71
#define AV_WL64A(p, v)
Definition: intreadwrite.h:557