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  t->coeff[i][j] = c;
166  }
167  }
168  }
169  return 0;
170 }
171 
172 static void ac_init(ArithCoder *ac, GetBitContext *gb)
173 {
174  ac->a = 4095;
175  ac->c = get_bits(gb, 12);
176  ac->overread = 0;
177 }
178 
179 static av_always_inline void ac_get(ArithCoder *ac, GetBitContext *gb, int p, int *e)
180 {
181  unsigned int k = (ac->a >> 8) | ((ac->a >> 7) & 1);
182  unsigned int q = k * p;
183  unsigned int a_q = ac->a - q;
184 
185  *e = ac->c < a_q;
186  if (*e) {
187  ac->a = a_q;
188  } else {
189  ac->a = q;
190  ac->c -= a_q;
191  }
192 
193  if (ac->a < 2048) {
194  int n = 11 - av_log2(ac->a);
195  ac->a <<= n;
196  if (get_bits_left(gb) < n)
197  ac->overread ++;
198  ac->c = (ac->c << n) | get_bits(gb, n);
199  }
200 }
201 
203 {
204  return (ff_reverse[c & 127] >> 1) + 1;
205 }
206 
207 static void build_filter(int16_t table[DST_MAX_ELEMENTS][16][256], const Table *fsets)
208 {
209  int i, j, k, l;
210 
211  for (i = 0; i < fsets->elements; i++) {
212  int length = fsets->length[i];
213 
214  for (j = 0; j < 16; j++) {
215  int total = av_clip(length - j * 8, 0, 8);
216 
217  for (k = 0; k < 256; k++) {
218  int v = 0;
219 
220  for (l = 0; l < total; l++)
221  v += (((k >> l) & 1) * 2 - 1) * fsets->coeff[i][j * 8 + l];
222  table[i][j][k] = v;
223  }
224  }
225  }
226 }
227 
228 static int decode_frame(AVCodecContext *avctx, void *data,
229  int *got_frame_ptr, AVPacket *avpkt)
230 {
231  unsigned samples_per_frame = DST_SAMPLES_PER_FRAME(avctx->sample_rate);
232  unsigned map_ch_to_felem[DST_MAX_CHANNELS];
233  unsigned map_ch_to_pelem[DST_MAX_CHANNELS];
234  unsigned i, ch, same_map, dst_x_bit;
235  unsigned half_prob[DST_MAX_CHANNELS];
236  const int channels = avctx->channels;
237  DSTContext *s = avctx->priv_data;
238  GetBitContext *gb = &s->gb;
239  ArithCoder *ac = &s->ac;
240  AVFrame *frame = data;
241  uint8_t *dsd;
242  float *pcm;
243  int ret;
244 
245  if (avpkt->size <= 1)
246  return AVERROR_INVALIDDATA;
247 
248  frame->nb_samples = samples_per_frame / 8;
249  if ((ret = ff_get_buffer(avctx, frame, 0)) < 0)
250  return ret;
251  dsd = frame->data[0];
252  pcm = (float *)frame->data[0];
253 
254  if ((ret = init_get_bits8(gb, avpkt->data, avpkt->size)) < 0)
255  return ret;
256 
257  if (!get_bits1(gb)) {
258  skip_bits1(gb);
259  if (get_bits(gb, 6))
260  return AVERROR_INVALIDDATA;
261  memcpy(frame->data[0], avpkt->data + 1, FFMIN(avpkt->size - 1, frame->nb_samples * avctx->channels));
262  goto dsd;
263  }
264 
265  /* Segmentation (10.4, 10.5, 10.6) */
266 
267  if (!get_bits1(gb)) {
268  avpriv_request_sample(avctx, "Not Same Segmentation");
269  return AVERROR_PATCHWELCOME;
270  }
271 
272  if (!get_bits1(gb)) {
273  avpriv_request_sample(avctx, "Not Same Segmentation For All Channels");
274  return AVERROR_PATCHWELCOME;
275  }
276 
277  if (!get_bits1(gb)) {
278  avpriv_request_sample(avctx, "Not End Of Channel Segmentation");
279  return AVERROR_PATCHWELCOME;
280  }
281 
282  /* Mapping (10.7, 10.8, 10.9) */
283 
284  same_map = get_bits1(gb);
285 
286  if ((ret = read_map(gb, &s->fsets, map_ch_to_felem, avctx->channels)) < 0)
287  return ret;
288 
289  if (same_map) {
290  s->probs.elements = s->fsets.elements;
291  memcpy(map_ch_to_pelem, map_ch_to_felem, sizeof(map_ch_to_felem));
292  } else {
293  avpriv_request_sample(avctx, "Not Same Mapping");
294  if ((ret = read_map(gb, &s->probs, map_ch_to_pelem, avctx->channels)) < 0)
295  return ret;
296  }
297 
298  /* Half Probability (10.10) */
299 
300  for (ch = 0; ch < avctx->channels; ch++)
301  half_prob[ch] = get_bits1(gb);
302 
303  /* Filter Coef Sets (10.12) */
304 
305  read_table(gb, &s->fsets, fsets_code_pred_coeff, 7, 9, 1, 0);
306 
307  /* Probability Tables (10.13) */
308 
309  read_table(gb, &s->probs, probs_code_pred_coeff, 6, 7, 0, 1);
310 
311  /* Arithmetic Coded Data (10.11) */
312 
313  if (get_bits1(gb))
314  return AVERROR_INVALIDDATA;
315  ac_init(ac, gb);
316 
317  build_filter(s->filter, &s->fsets);
318 
319  memset(s->status, 0xAA, sizeof(s->status));
320  memset(dsd, 0, frame->nb_samples * 4 * avctx->channels);
321 
322  ac_get(ac, gb, prob_dst_x_bit(s->fsets.coeff[0][0]), &dst_x_bit);
323 
324  for (i = 0; i < samples_per_frame; i++) {
325  for (ch = 0; ch < channels; ch++) {
326  const unsigned felem = map_ch_to_felem[ch];
327  int16_t (*filter)[256] = s->filter[felem];
328  uint8_t *status = s->status[ch];
329  int prob, residual, v;
330 
331 #define F(x) filter[(x)][status[(x)]]
332  const int16_t predict = F( 0) + F( 1) + F( 2) + F( 3) +
333  F( 4) + F( 5) + F( 6) + F( 7) +
334  F( 8) + F( 9) + F(10) + F(11) +
335  F(12) + F(13) + F(14) + F(15);
336 #undef F
337 
338  if (!half_prob[ch] || i >= s->fsets.length[felem]) {
339  unsigned pelem = map_ch_to_pelem[ch];
340  unsigned index = FFABS(predict) >> 3;
341  prob = s->probs.coeff[pelem][FFMIN(index, s->probs.length[pelem] - 1)];
342  } else {
343  prob = 128;
344  }
345 
346  if (ac->overread > 16)
347  return AVERROR_INVALIDDATA;
348 
349  ac_get(ac, gb, prob, &residual);
350  v = ((predict >> 15) ^ residual) & 1;
351  dsd[((i >> 3) * channels + ch) << 2] |= v << (7 - (i & 0x7 ));
352 
353  AV_WL64A(status + 8, (AV_RL64A(status + 8) << 1) | ((AV_RL64A(status) >> 63) & 1));
354  AV_WL64A(status, (AV_RL64A(status) << 1) | v);
355  }
356  }
357 
358 dsd:
359  for (i = 0; i < avctx->channels; i++) {
360  ff_dsd2pcm_translate(&s->dsdctx[i], frame->nb_samples, 0,
361  frame->data[0] + i * 4,
362  avctx->channels * 4, pcm + i, avctx->channels);
363  }
364 
365  *got_frame_ptr = 1;
366 
367  return avpkt->size;
368 }
369 
371  .name = "dst",
372  .long_name = NULL_IF_CONFIG_SMALL("DST (Digital Stream Transfer)"),
373  .type = AVMEDIA_TYPE_AUDIO,
374  .id = AV_CODEC_ID_DST,
375  .priv_data_size = sizeof(DSTContext),
376  .init = decode_init,
377  .decode = decode_frame,
378  .capabilities = AV_CODEC_CAP_DR1,
379  .sample_fmts = (const enum AVSampleFormat[]) { AV_SAMPLE_FMT_FLT,
381 };
#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:228
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:172
#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:370
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:202
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:179
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:207
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