FFmpeg
ffwavesynth.c
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1 /*
2  * Wavesynth pseudo-codec
3  * Copyright (c) 2011 Nicolas George
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 #include "libavutil/intreadwrite.h"
23 #include "libavutil/log.h"
24 #include "avcodec.h"
25 #include "internal.h"
26 
27 
28 #define SIN_BITS 14
29 #define WS_MAX_CHANNELS 32
30 #define INF_TS 0x7FFFFFFFFFFFFFFF
31 
32 #define PINK_UNIT 128
33 
34 /*
35  Format of the extradata and packets
36 
37  THIS INFORMATION IS NOT PART OF THE PUBLIC API OR ABI.
38  IT CAN CHANGE WITHOUT NOTIFICATION.
39 
40  All numbers are in little endian.
41 
42  The codec extradata define a set of intervals with uniform content.
43  Overlapping intervals are added together.
44 
45  extradata:
46  uint32 number of intervals
47  ... intervals
48 
49  interval:
50  int64 start timestamp; time_base must be 1/sample_rate;
51  start timestamps must be in ascending order
52  int64 end timestamp
53  uint32 type
54  uint32 channels mask
55  ... additional information, depends on type
56 
57  sine interval (type fourcc "SINE"):
58  int32 start frequency, in 1/(1<<16) Hz
59  int32 end frequency
60  int32 start amplitude, 1<<16 is the full amplitude
61  int32 end amplitude
62  uint32 start phase, 0 is sin(0), 0x20000000 is sin(pi/2), etc.;
63  n | (1<<31) means to match the phase of previous channel #n
64 
65  pink noise interval (type fourcc "NOIS"):
66  int32 start amplitude
67  int32 end amplitude
68 
69  The input packets encode the time and duration of the requested segment.
70 
71  packet:
72  int64 start timestamp
73  int32 duration
74 
75 */
76 
78  WS_SINE = MKTAG('S','I','N','E'),
79  WS_NOISE = MKTAG('N','O','I','S'),
80 };
81 
82 struct ws_interval {
83  int64_t ts_start, ts_end;
84  uint64_t phi0, dphi0, ddphi;
85  uint64_t amp0, damp;
86  uint64_t phi, dphi, amp;
87  uint32_t channels;
89  int next;
90 };
91 
93  int64_t cur_ts;
94  int64_t next_ts;
96  struct ws_interval *inter;
97  uint32_t dither_state;
98  uint32_t pink_state;
99  int32_t pink_pool[PINK_UNIT];
100  unsigned pink_need, pink_pos;
101  int nb_inter;
104 };
105 
106 #define LCG_A 1284865837
107 #define LCG_C 4150755663
108 #define LCG_AI 849225893 /* A*AI = 1 [mod 1<<32] */
109 
110 static uint32_t lcg_next(uint32_t *s)
111 {
112  *s = *s * LCG_A + LCG_C;
113  return *s;
114 }
115 
116 static void lcg_seek(uint32_t *s, uint32_t dt)
117 {
118  uint32_t a, c, t = *s;
119 
120  a = LCG_A;
121  c = LCG_C;
122  while (dt) {
123  if (dt & 1)
124  t = a * t + c;
125  c *= a + 1; /* coefficients for a double step */
126  a *= a;
127  dt >>= 1;
128  }
129  *s = t;
130 }
131 
132 /* Emulate pink noise by summing white noise at the sampling frequency,
133  * white noise at half the sampling frequency (each value taken twice),
134  * etc., with a total of 8 octaves.
135  * This is known as the Voss-McCartney algorithm. */
136 
137 static void pink_fill(struct wavesynth_context *ws)
138 {
139  int32_t vt[7] = { 0 }, v = 0;
140  int i, j;
141 
142  ws->pink_pos = 0;
143  if (!ws->pink_need)
144  return;
145  for (i = 0; i < PINK_UNIT; i++) {
146  for (j = 0; j < 7; j++) {
147  if ((i >> j) & 1)
148  break;
149  v -= vt[j];
150  vt[j] = (int32_t)lcg_next(&ws->pink_state) >> 3;
151  v += vt[j];
152  }
153  ws->pink_pool[i] = v + ((int32_t)lcg_next(&ws->pink_state) >> 3);
154  }
155  lcg_next(&ws->pink_state); /* so we use exactly 256 steps */
156 }
157 
158 /**
159  * @return (1<<64) * a / b, without overflow, if a < b
160  */
161 static uint64_t frac64(uint64_t a, uint64_t b)
162 {
163  uint64_t r = 0;
164  int i;
165 
166  if (b < (uint64_t)1 << 32) { /* b small, use two 32-bits steps */
167  a <<= 32;
168  return ((a / b) << 32) | ((a % b) << 32) / b;
169  }
170  if (b < (uint64_t)1 << 48) { /* b medium, use four 16-bits steps */
171  for (i = 0; i < 4; i++) {
172  a <<= 16;
173  r = (r << 16) | (a / b);
174  a %= b;
175  }
176  return r;
177  }
178  for (i = 63; i >= 0; i--) {
179  if (a >= (uint64_t)1 << 63 || a << 1 >= b) {
180  r |= (uint64_t)1 << i;
181  a = (a << 1) - b;
182  } else {
183  a <<= 1;
184  }
185  }
186  return r;
187 }
188 
189 static uint64_t phi_at(struct ws_interval *in, int64_t ts)
190 {
191  uint64_t dt = ts - in->ts_start;
192  uint64_t dt2 = dt & 1 ? /* dt * (dt - 1) / 2 without overflow */
193  dt * ((dt - 1) >> 1) : (dt >> 1) * (dt - 1);
194  return in->phi0 + dt * in->dphi0 + dt2 * in->ddphi;
195 }
196 
197 static void wavesynth_seek(struct wavesynth_context *ws, int64_t ts)
198 {
199  int *last, i;
200  struct ws_interval *in;
201 
202  last = &ws->cur_inter;
203  for (i = 0; i < ws->nb_inter; i++) {
204  in = &ws->inter[i];
205  if (ts < in->ts_start)
206  break;
207  if (ts >= in->ts_end)
208  continue;
209  *last = i;
210  last = &in->next;
211  in->phi = phi_at(in, ts);
212  in->dphi = in->dphi0 + (ts - in->ts_start) * in->ddphi;
213  in->amp = in->amp0 + (ts - in->ts_start) * in->damp;
214  }
215  ws->next_inter = i;
216  ws->next_ts = i < ws->nb_inter ? ws->inter[i].ts_start : INF_TS;
217  *last = -1;
218  lcg_seek(&ws->dither_state, (uint32_t)ts - (uint32_t)ws->cur_ts);
219  if (ws->pink_need) {
220  int64_t pink_ts_cur = (ws->cur_ts + PINK_UNIT - 1) & ~(PINK_UNIT - 1);
221  int64_t pink_ts_next = ts & ~(PINK_UNIT - 1);
222  int pos = ts & (PINK_UNIT - 1);
223  lcg_seek(&ws->pink_state, (uint32_t)(pink_ts_next - pink_ts_cur) * 2);
224  if (pos) {
225  pink_fill(ws);
226  ws->pink_pos = pos;
227  } else {
228  ws->pink_pos = PINK_UNIT;
229  }
230  }
231  ws->cur_ts = ts;
232 }
233 
235 {
236  struct wavesynth_context *ws = avc->priv_data;
237  struct ws_interval *in;
238  uint8_t *edata, *edata_end;
239  int32_t f1, f2, a1, a2;
240  uint32_t phi;
241  int64_t dphi1, dphi2, dt, cur_ts = -0x8000000000000000;
242  int i;
243 
244  if (avc->extradata_size < 4)
245  return AVERROR(EINVAL);
246  edata = avc->extradata;
247  edata_end = edata + avc->extradata_size;
248  ws->nb_inter = AV_RL32(edata);
249  edata += 4;
250  if (ws->nb_inter < 0 || (edata_end - edata) / 24 < ws->nb_inter)
251  return AVERROR(EINVAL);
252  ws->inter = av_calloc(ws->nb_inter, sizeof(*ws->inter));
253  if (!ws->inter)
254  return AVERROR(ENOMEM);
255  for (i = 0; i < ws->nb_inter; i++) {
256  in = &ws->inter[i];
257  if (edata_end - edata < 24)
258  return AVERROR(EINVAL);
259  in->ts_start = AV_RL64(edata + 0);
260  in->ts_end = AV_RL64(edata + 8);
261  in->type = AV_RL32(edata + 16);
262  in->channels = AV_RL32(edata + 20);
263  edata += 24;
264  if (in->ts_start < cur_ts ||
265  in->ts_end <= in->ts_start ||
266  (uint64_t)in->ts_end - in->ts_start > INT64_MAX
267  )
268  return AVERROR(EINVAL);
269  cur_ts = in->ts_start;
270  dt = in->ts_end - in->ts_start;
271  switch (in->type) {
272  case WS_SINE:
273  if (edata_end - edata < 20 || avc->sample_rate <= 0)
274  return AVERROR(EINVAL);
275  f1 = AV_RL32(edata + 0);
276  f2 = AV_RL32(edata + 4);
277  a1 = AV_RL32(edata + 8);
278  a2 = AV_RL32(edata + 12);
279  phi = AV_RL32(edata + 16);
280  edata += 20;
281  dphi1 = frac64(f1, (int64_t)avc->sample_rate << 16);
282  dphi2 = frac64(f2, (int64_t)avc->sample_rate << 16);
283  in->dphi0 = dphi1;
284  in->ddphi = (dphi2 - dphi1) / dt;
285  if (phi & 0x80000000) {
286  phi &= ~0x80000000;
287  if (phi >= i)
288  return AVERROR(EINVAL);
289  in->phi0 = phi_at(&ws->inter[phi], in->ts_start);
290  } else {
291  in->phi0 = (uint64_t)phi << 33;
292  }
293  break;
294  case WS_NOISE:
295  if (edata_end - edata < 8)
296  return AVERROR(EINVAL);
297  a1 = AV_RL32(edata + 0);
298  a2 = AV_RL32(edata + 4);
299  edata += 8;
300  break;
301  default:
302  return AVERROR(EINVAL);
303  }
304  in->amp0 = (uint64_t)a1 << 32;
305  in->damp = (int64_t)(((uint64_t)a2 << 32) - ((uint64_t)a1 << 32)) / dt;
306  }
307  if (edata != edata_end)
308  return AVERROR(EINVAL);
309  return 0;
310 }
311 
313 {
314  struct wavesynth_context *ws = avc->priv_data;
315  int i, r;
316 
317  if (avc->channels > WS_MAX_CHANNELS) {
318  av_log(avc, AV_LOG_ERROR,
319  "This implementation is limited to %d channels.\n",
321  return AVERROR(EINVAL);
322  }
323  r = wavesynth_parse_extradata(avc);
324  if (r < 0) {
325  av_log(avc, AV_LOG_ERROR, "Invalid intervals definitions.\n");
326  goto fail;
327  }
328  ws->sin = av_malloc(sizeof(*ws->sin) << SIN_BITS);
329  if (!ws->sin) {
330  r = AVERROR(ENOMEM);
331  goto fail;
332  }
333  for (i = 0; i < 1 << SIN_BITS; i++)
334  ws->sin[i] = floor(32767 * sin(2 * M_PI * i / (1 << SIN_BITS)));
335  ws->dither_state = MKTAG('D','I','T','H');
336  for (i = 0; i < ws->nb_inter; i++)
337  ws->pink_need += ws->inter[i].type == WS_NOISE;
338  ws->pink_state = MKTAG('P','I','N','K');
339  ws->pink_pos = PINK_UNIT;
340  wavesynth_seek(ws, 0);
342  return 0;
343 
344 fail:
345  av_freep(&ws->inter);
346  av_freep(&ws->sin);
347  return r;
348 }
349 
350 static void wavesynth_synth_sample(struct wavesynth_context *ws, int64_t ts,
351  int32_t *channels)
352 {
353  int32_t amp, val, *cv;
354  struct ws_interval *in;
355  int i, *last, pink;
356  uint32_t c, all_ch = 0;
357 
358  i = ws->cur_inter;
359  last = &ws->cur_inter;
360  if (ws->pink_pos == PINK_UNIT)
361  pink_fill(ws);
362  pink = ws->pink_pool[ws->pink_pos++] >> 16;
363  while (i >= 0) {
364  in = &ws->inter[i];
365  i = in->next;
366  if (ts >= in->ts_end) {
367  *last = i;
368  continue;
369  }
370  last = &in->next;
371  amp = in->amp >> 32;
372  in->amp += in->damp;
373  switch (in->type) {
374  case WS_SINE:
375  val = amp * ws->sin[in->phi >> (64 - SIN_BITS)];
376  in->phi += in->dphi;
377  in->dphi += in->ddphi;
378  break;
379  case WS_NOISE:
380  val = amp * (unsigned)pink;
381  break;
382  default:
383  val = 0;
384  }
385  all_ch |= in->channels;
386  for (c = in->channels, cv = channels; c; c >>= 1, cv++)
387  if (c & 1)
388  *cv += (unsigned)val;
389  }
390  val = (int32_t)lcg_next(&ws->dither_state) >> 16;
391  for (c = all_ch, cv = channels; c; c >>= 1, cv++)
392  if (c & 1)
393  *cv += val;
394 }
395 
396 static void wavesynth_enter_intervals(struct wavesynth_context *ws, int64_t ts)
397 {
398  int *last, i;
399  struct ws_interval *in;
400 
401  last = &ws->cur_inter;
402  for (i = ws->cur_inter; i >= 0; i = ws->inter[i].next)
403  last = &ws->inter[i].next;
404  for (i = ws->next_inter; i < ws->nb_inter; i++) {
405  in = &ws->inter[i];
406  if (ts < in->ts_start)
407  break;
408  if (ts >= in->ts_end)
409  continue;
410  *last = i;
411  last = &in->next;
412  in->phi = in->phi0;
413  in->dphi = in->dphi0;
414  in->amp = in->amp0;
415  }
416  ws->next_inter = i;
417  ws->next_ts = i < ws->nb_inter ? ws->inter[i].ts_start : INF_TS;
418  *last = -1;
419 }
420 
421 static int wavesynth_decode(AVCodecContext *avc, void *rframe, int *rgot_frame,
422  AVPacket *packet)
423 {
424  struct wavesynth_context *ws = avc->priv_data;
425  AVFrame *frame = rframe;
426  int64_t ts;
427  int duration;
428  int s, c, r;
429  int16_t *pcm;
431 
432  *rgot_frame = 0;
433  if (packet->size != 12)
434  return AVERROR_INVALIDDATA;
435  ts = AV_RL64(packet->data);
436  if (ts != ws->cur_ts)
437  wavesynth_seek(ws, ts);
438  duration = AV_RL32(packet->data + 8);
439  if (duration <= 0)
440  return AVERROR(EINVAL);
441  frame->nb_samples = duration;
442  r = ff_get_buffer(avc, frame, 0);
443  if (r < 0)
444  return r;
445  pcm = (int16_t *)frame->data[0];
446  for (s = 0; s < duration; s++, ts++) {
447  memset(channels, 0, avc->channels * sizeof(*channels));
448  if (ts >= ws->next_ts)
450  wavesynth_synth_sample(ws, ts, channels);
451  for (c = 0; c < avc->channels; c++)
452  *(pcm++) = channels[c] >> 16;
453  }
454  ws->cur_ts += duration;
455  *rgot_frame = 1;
456  return packet->size;
457 }
458 
460 {
461  struct wavesynth_context *ws = avc->priv_data;
462 
463  av_freep(&ws->sin);
464  av_freep(&ws->inter);
465  return 0;
466 }
467 
469  .name = "wavesynth",
470  .long_name = NULL_IF_CONFIG_SMALL("Wave synthesis pseudo-codec"),
471  .type = AVMEDIA_TYPE_AUDIO,
473  .priv_data_size = sizeof(struct wavesynth_context),
474  .init = wavesynth_init,
475  .close = wavesynth_close,
476  .decode = wavesynth_decode,
477  .capabilities = AV_CODEC_CAP_DR1,
478 };
const char const char void * val
Definition: avisynth_c.h:863
#define AVERROR_INVALIDDATA
Invalid data found when processing input.
Definition: error.h:59
This structure describes decoded (raw) audio or video data.
Definition: frame.h:295
int64_t ts_end
Definition: ffwavesynth.c:83
uint32_t channels
Definition: ffwavesynth.c:87
struct ws_interval * inter
Definition: ffwavesynth.c:96
channels
Definition: aptx.c:30
ws_interval_type
Definition: ffwavesynth.c:77
int size
Definition: avcodec.h:1481
The reader does not expect b to be semantically here and if the code is changed by maybe adding a a division or other the signedness will almost certainly be mistaken To avoid this confusion a new type was SUINT is the C unsigned type but it holds a signed int to use the same example SUINT a
Definition: undefined.txt:36
uint64_t_TMPL AV_RL64
Definition: bytestream.h:87
AVCodec.
Definition: avcodec.h:3492
uint64_t amp
Definition: ffwavesynth.c:86
uint32_t pink_state
Definition: ffwavesynth.c:98
void * av_calloc(size_t nmemb, size_t size)
Non-inlined equivalent of av_mallocz_array().
Definition: mem.c:244
#define SIN_BITS
Definition: ffwavesynth.c:28
enum AVSampleFormat sample_fmt
audio sample format
Definition: avcodec.h:2236
uint8_t
#define av_cold
Definition: attributes.h:82
#define av_malloc(s)
int32_t f1
Definition: sbgdec.c:147
#define LCG_A
Definition: ffwavesynth.c:106
Undefined Behavior In the C some operations are like signed integer dereferencing freed accessing outside allocated Undefined Behavior must not occur in a C it is not safe even if the output of undefined operations is unused The unsafety may seem nit picking but Optimizing compilers have in fact optimized code on the assumption that no undefined Behavior occurs Optimizing code based on wrong assumptions can and has in some cases lead to effects beyond the output of computations The signed integer overflow problem in speed critical code Code which is highly optimized and works with signed integers sometimes has the problem that often the output of the computation does not c
Definition: undefined.txt:32
uint8_t * extradata
some codecs need / can use extradata like Huffman tables.
Definition: avcodec.h:1669
int64_t duration
Definition: movenc.c:63
uint8_t * data
Definition: avcodec.h:1480
enum ws_interval_type type
Definition: ffwavesynth.c:88
#define av_log(a,...)
#define i(width, name, range_min, range_max)
Definition: cbs_h2645.c:259
#define AV_LOG_ERROR
Something went wrong and cannot losslessly be recovered.
Definition: log.h:176
static void wavesynth_synth_sample(struct wavesynth_context *ws, int64_t ts, int32_t *channels)
Definition: ffwavesynth.c:350
#define LCG_C
Definition: ffwavesynth.c:107
static void pink_fill(struct wavesynth_context *ws)
Definition: ffwavesynth.c:137
#define NULL_IF_CONFIG_SMALL(x)
Return NULL if CONFIG_SMALL is true, otherwise the argument without modification. ...
Definition: internal.h:186
const char * r
Definition: vf_curves.c:114
const char * name
Name of the codec implementation.
Definition: avcodec.h:3499
int32_t a1
Definition: sbgdec.c:148
static uint32_t lcg_next(uint32_t *s)
Definition: ffwavesynth.c:110
#define fail()
Definition: checkasm.h:122
unsigned pink_need
Definition: ffwavesynth.c:100
#define b
Definition: input.c:41
uint64_t phi
Definition: ffwavesynth.c:86
static uint64_t frac64(uint64_t a, uint64_t b)
Definition: ffwavesynth.c:161
int32_t
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
#define s(width, name)
Definition: cbs_vp9.c:257
#define PINK_UNIT
Definition: ffwavesynth.c:32
int32_t pink_pool[PINK_UNIT]
Definition: ffwavesynth.c:99
static void wavesynth_enter_intervals(struct wavesynth_context *ws, int64_t ts)
Definition: ffwavesynth.c:396
sample_rate
Libavcodec external API header.
int sample_rate
samples per second
Definition: avcodec.h:2228
uint64_t damp
Definition: ffwavesynth.c:85
main external API structure.
Definition: avcodec.h:1568
static av_cold int wavesynth_init(AVCodecContext *avc)
Definition: ffwavesynth.c:312
static void lcg_seek(uint32_t *s, uint32_t dt)
Definition: ffwavesynth.c:116
int ff_get_buffer(AVCodecContext *avctx, AVFrame *frame, int flags)
Get a buffer for a frame.
Definition: decode.c:1968
int extradata_size
Definition: avcodec.h:1670
uint8_t pi<< 24) CONV_FUNC_GROUP(AV_SAMPLE_FMT_FLT, float, AV_SAMPLE_FMT_U8, uint8_t,(*(const uint8_t *) pi-0x80)*(1.0f/(1<< 7))) CONV_FUNC_GROUP(AV_SAMPLE_FMT_DBL, double, AV_SAMPLE_FMT_U8, uint8_t,(*(const uint8_t *) pi-0x80)*(1.0/(1<< 7))) CONV_FUNC_GROUP(AV_SAMPLE_FMT_U8, uint8_t, AV_SAMPLE_FMT_S16, int16_t,(*(const int16_t *) pi >> 8)+0x80) CONV_FUNC_GROUP(AV_SAMPLE_FMT_FLT, float, AV_SAMPLE_FMT_S16, int16_t,*(const int16_t *) pi *(1.0f/(1<< 15))) CONV_FUNC_GROUP(AV_SAMPLE_FMT_DBL, double, AV_SAMPLE_FMT_S16, int16_t,*(const int16_t *) pi *(1.0/(1<< 15))) CONV_FUNC_GROUP(AV_SAMPLE_FMT_U8, uint8_t, AV_SAMPLE_FMT_S32, int32_t,(*(const int32_t *) pi >> 24)+0x80) CONV_FUNC_GROUP(AV_SAMPLE_FMT_FLT, float, AV_SAMPLE_FMT_S32, int32_t,*(const int32_t *) pi *(1.0f/(1U<< 31))) CONV_FUNC_GROUP(AV_SAMPLE_FMT_DBL, double, AV_SAMPLE_FMT_S32, int32_t,*(const int32_t *) pi *(1.0/(1U<< 31))) CONV_FUNC_GROUP(AV_SAMPLE_FMT_U8, uint8_t, AV_SAMPLE_FMT_FLT, float, av_clip_uint8(lrintf(*(const float *) pi *(1<< 7))+0x80)) CONV_FUNC_GROUP(AV_SAMPLE_FMT_S16, int16_t, AV_SAMPLE_FMT_FLT, float, av_clip_int16(lrintf(*(const float *) pi *(1<< 15)))) CONV_FUNC_GROUP(AV_SAMPLE_FMT_S32, int32_t, AV_SAMPLE_FMT_FLT, float, av_clipl_int32(llrintf(*(const float *) pi *(1U<< 31)))) CONV_FUNC_GROUP(AV_SAMPLE_FMT_U8, uint8_t, AV_SAMPLE_FMT_DBL, double, av_clip_uint8(lrint(*(const double *) pi *(1<< 7))+0x80)) CONV_FUNC_GROUP(AV_SAMPLE_FMT_S16, int16_t, AV_SAMPLE_FMT_DBL, double, av_clip_int16(lrint(*(const double *) pi *(1<< 15)))) CONV_FUNC_GROUP(AV_SAMPLE_FMT_S32, int32_t, AV_SAMPLE_FMT_DBL, double, av_clipl_int32(llrint(*(const double *) pi *(1U<< 31))))#define SET_CONV_FUNC_GROUP(ofmt, ifmt) static void set_generic_function(AudioConvert *ac){}void ff_audio_convert_free(AudioConvert **ac){if(!*ac) return;ff_dither_free(&(*ac) ->dc);av_freep(ac);}AudioConvert *ff_audio_convert_alloc(AVAudioResampleContext *avr, enum AVSampleFormat out_fmt, enum AVSampleFormat in_fmt, int channels, int sample_rate, int apply_map){AudioConvert *ac;int in_planar, out_planar;ac=av_mallocz(sizeof(*ac));if(!ac) return NULL;ac->avr=avr;ac->out_fmt=out_fmt;ac->in_fmt=in_fmt;ac->channels=channels;ac->apply_map=apply_map;if(avr->dither_method!=AV_RESAMPLE_DITHER_NONE &&av_get_packed_sample_fmt(out_fmt)==AV_SAMPLE_FMT_S16 &&av_get_bytes_per_sample(in_fmt) > 2){ac->dc=ff_dither_alloc(avr, out_fmt, in_fmt, channels, sample_rate, apply_map);if(!ac->dc){av_free(ac);return NULL;}return ac;}in_planar=ff_sample_fmt_is_planar(in_fmt, channels);out_planar=ff_sample_fmt_is_planar(out_fmt, channels);if(in_planar==out_planar){ac->func_type=CONV_FUNC_TYPE_FLAT;ac->planes=in_planar?ac->channels:1;}else if(in_planar) ac->func_type=CONV_FUNC_TYPE_INTERLEAVE;else ac->func_type=CONV_FUNC_TYPE_DEINTERLEAVE;set_generic_function(ac);if(ARCH_AARCH64) ff_audio_convert_init_aarch64(ac);if(ARCH_ARM) ff_audio_convert_init_arm(ac);if(ARCH_X86) ff_audio_convert_init_x86(ac);return ac;}int ff_audio_convert(AudioConvert *ac, AudioData *out, AudioData *in){int use_generic=1;int len=in->nb_samples;int p;if(ac->dc){av_log(ac->avr, AV_LOG_TRACE,"%d samples - audio_convert: %s to %s (dithered)\n", len, av_get_sample_fmt_name(ac->in_fmt), av_get_sample_fmt_name(ac->out_fmt));return ff_convert_dither(ac-> in
int64_t ts_start
Definition: ffwavesynth.c:83
static uint64_t phi_at(struct ws_interval *in, int64_t ts)
Definition: ffwavesynth.c:189
uint64_t amp0
Definition: ffwavesynth.c:85
uint8_t * data[AV_NUM_DATA_POINTERS]
pointer to the picture/channel planes.
Definition: frame.h:309
uint64_t dphi0
Definition: ffwavesynth.c:84
AVCodec ff_ffwavesynth_decoder
Definition: ffwavesynth.c:468
common internal api header.
uint64_t ddphi
Definition: ffwavesynth.c:84
static void wavesynth_seek(struct wavesynth_context *ws, int64_t ts)
Definition: ffwavesynth.c:197
signed 16 bits
Definition: samplefmt.h:61
static int wavesynth_decode(AVCodecContext *avc, void *rframe, int *rgot_frame, AVPacket *packet)
Definition: ffwavesynth.c:421
static av_cold int wavesynth_close(AVCodecContext *avc)
Definition: ffwavesynth.c:459
int32_t f2
Definition: sbgdec.c:147
void * priv_data
Definition: avcodec.h:1595
uint64_t phi0
Definition: ffwavesynth.c:84
uint32_t dither_state
Definition: ffwavesynth.c:97
int channels
number of audio channels
Definition: avcodec.h:2229
#define WS_MAX_CHANNELS
Definition: ffwavesynth.c:29
int32_t a2
Definition: sbgdec.c:148
#define av_freep(p)
#define M_PI
Definition: mathematics.h:52
static int wavesynth_parse_extradata(AVCodecContext *avc)
Definition: ffwavesynth.c:234
Filter the word “frame” indicates either a video frame or a group of audio as stored in an AVFrame structure Format for each input and each output the list of supported formats For video that means pixel format For audio that means channel sample they are references to shared objects When the negotiation mechanism computes the intersection of the formats supported at each end of a all references to both lists are replaced with a reference to the intersection And when a single format is eventually chosen for a link amongst the remaining all references to the list are updated That means that if a filter requires that its input and output have the same format amongst a supported all it has to do is use a reference to the same list of formats query_formats can leave some formats unset and return AVERROR(EAGAIN) to cause the negotiation mechanism toagain later.That can be used by filters with complex requirements to use the format negotiated on one link to set the formats supported on another.Frame references ownership and permissions
#define MKTAG(a, b, c, d)
Definition: common.h:366
uint64_t dphi
Definition: ffwavesynth.c:86
uint64_t_TMPL AV_WL64 unsigned int_TMPL AV_RL32
Definition: bytestream.h:87
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
for(j=16;j >0;--j)
#define INF_TS
Definition: ffwavesynth.c:30