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resample.c
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1 /*
2  * Copyright (c) 2004 Michael Niedermayer <michaelni@gmx.at>
3  * Copyright (c) 2012 Justin Ruggles <justin.ruggles@gmail.com>
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/common.h"
23 #include "libavutil/libm.h"
24 #include "libavutil/log.h"
25 #include "internal.h"
26 #include "resample.h"
27 #include "audio_data.h"
28 
29 
30 /* double template */
31 #define CONFIG_RESAMPLE_DBL
32 #include "resample_template.c"
33 #undef CONFIG_RESAMPLE_DBL
34 
35 /* float template */
36 #define CONFIG_RESAMPLE_FLT
37 #include "resample_template.c"
38 #undef CONFIG_RESAMPLE_FLT
39 
40 /* s32 template */
41 #define CONFIG_RESAMPLE_S32
42 #include "resample_template.c"
43 #undef CONFIG_RESAMPLE_S32
44 
45 /* s16 template */
46 #include "resample_template.c"
47 
48 
49 /* 0th order modified bessel function of the first kind. */
50 static double bessel(double x)
51 {
52  double v = 1;
53  double lastv = 0;
54  double t = 1;
55  int i;
56 
57  x = x * x / 4;
58  for (i = 1; v != lastv; i++) {
59  lastv = v;
60  t *= x / (i * i);
61  v += t;
62  }
63  return v;
64 }
65 
66 /* Build a polyphase filterbank. */
67 static int build_filter(ResampleContext *c, double factor)
68 {
69  int ph, i;
70  double x, y, w;
71  double *tab;
72  int tap_count = c->filter_length;
73  int phase_count = 1 << c->phase_shift;
74  const int center = (tap_count - 1) / 2;
75 
76  tab = av_malloc(tap_count * sizeof(*tab));
77  if (!tab)
78  return AVERROR(ENOMEM);
79 
80  for (ph = 0; ph < phase_count; ph++) {
81  double norm = 0;
82  for (i = 0; i < tap_count; i++) {
83  x = M_PI * ((double)(i - center) - (double)ph / phase_count) * factor;
84  if (x == 0) y = 1.0;
85  else y = sin(x) / x;
86  switch (c->filter_type) {
88  const float d = -0.5; //first order derivative = -0.5
89  x = fabs(((double)(i - center) - (double)ph / phase_count) * factor);
90  if (x < 1.0) y = 1 - 3 * x*x + 2 * x*x*x + d * ( -x*x + x*x*x);
91  else y = d * (-4 + 8 * x - 5 * x*x + x*x*x);
92  break;
93  }
95  w = 2.0 * x / (factor * tap_count) + M_PI;
96  y *= 0.3635819 - 0.4891775 * cos( w) +
97  0.1365995 * cos(2 * w) -
98  0.0106411 * cos(3 * w);
99  break;
101  w = 2.0 * x / (factor * tap_count * M_PI);
102  y *= bessel(c->kaiser_beta * sqrt(FFMAX(1 - w * w, 0)));
103  break;
104  }
105 
106  tab[i] = y;
107  norm += y;
108  }
109  /* normalize so that an uniform color remains the same */
110  for (i = 0; i < tap_count; i++)
111  tab[i] = tab[i] / norm;
112 
113  c->set_filter(c->filter_bank, tab, ph, tap_count);
114  }
115 
116  av_free(tab);
117  return 0;
118 }
119 
121 {
123  int out_rate = avr->out_sample_rate;
124  int in_rate = avr->in_sample_rate;
125  double factor = FFMIN(out_rate * avr->cutoff / in_rate, 1.0);
126  int phase_count = 1 << avr->phase_shift;
127  int felem_size;
128 
133  av_log(avr, AV_LOG_ERROR, "Unsupported internal format for "
134  "resampling: %s\n",
136  return NULL;
137  }
138  c = av_mallocz(sizeof(*c));
139  if (!c)
140  return NULL;
141 
142  c->avr = avr;
143  c->phase_shift = avr->phase_shift;
144  c->phase_mask = phase_count - 1;
145  c->linear = avr->linear_interp;
146  c->filter_length = FFMAX((int)ceil(avr->filter_size / factor), 1);
147  c->filter_type = avr->filter_type;
148  c->kaiser_beta = avr->kaiser_beta;
149 
150  switch (avr->internal_sample_fmt) {
151  case AV_SAMPLE_FMT_DBLP:
152  c->resample_one = c->linear ? resample_linear_dbl : resample_one_dbl;
153  c->resample_nearest = resample_nearest_dbl;
154  c->set_filter = set_filter_dbl;
155  break;
156  case AV_SAMPLE_FMT_FLTP:
157  c->resample_one = c->linear ? resample_linear_flt : resample_one_flt;
158  c->resample_nearest = resample_nearest_flt;
159  c->set_filter = set_filter_flt;
160  break;
161  case AV_SAMPLE_FMT_S32P:
162  c->resample_one = c->linear ? resample_linear_s32 : resample_one_s32;
163  c->resample_nearest = resample_nearest_s32;
164  c->set_filter = set_filter_s32;
165  break;
166  case AV_SAMPLE_FMT_S16P:
167  c->resample_one = c->linear ? resample_linear_s16 : resample_one_s16;
168  c->resample_nearest = resample_nearest_s16;
169  c->set_filter = set_filter_s16;
170  break;
171  }
172 
173  if (ARCH_AARCH64)
175 
176  felem_size = av_get_bytes_per_sample(avr->internal_sample_fmt);
177  c->filter_bank = av_mallocz(c->filter_length * (phase_count + 1) * felem_size);
178  if (!c->filter_bank)
179  goto error;
180 
181  if (build_filter(c, factor) < 0)
182  goto error;
183 
184  memcpy(&c->filter_bank[(c->filter_length * phase_count + 1) * felem_size],
185  c->filter_bank, (c->filter_length - 1) * felem_size);
186  memcpy(&c->filter_bank[c->filter_length * phase_count * felem_size],
187  &c->filter_bank[(c->filter_length - 1) * felem_size], felem_size);
188 
189  c->compensation_distance = 0;
190  if (!av_reduce(&c->src_incr, &c->dst_incr, out_rate,
191  in_rate * (int64_t)phase_count, INT32_MAX / 2))
192  goto error;
193  c->ideal_dst_incr = c->dst_incr;
194 
195  c->padding_size = (c->filter_length - 1) / 2;
196  c->initial_padding_filled = 0;
197  c->index = 0;
198  c->frac = 0;
199 
200  /* allocate internal buffer */
202  avr->internal_sample_fmt,
203  "resample buffer");
204  if (!c->buffer)
205  goto error;
206  c->buffer->nb_samples = c->padding_size;
208 
209  av_log(avr, AV_LOG_DEBUG, "resample: %s from %d Hz to %d Hz\n",
211  avr->in_sample_rate, avr->out_sample_rate);
212 
213  return c;
214 
215 error:
217  av_free(c->filter_bank);
218  av_free(c);
219  return NULL;
220 }
221 
223 {
224  if (!*c)
225  return;
226  ff_audio_data_free(&(*c)->buffer);
227  av_free((*c)->filter_bank);
228  av_freep(c);
229 }
230 
233 {
235  AudioData *fifo_buf = NULL;
236  int ret = 0;
237 
238  if (compensation_distance < 0)
239  return AVERROR(EINVAL);
240  if (!compensation_distance && sample_delta)
241  return AVERROR(EINVAL);
242 
243  if (!avr->resample_needed) {
244 #if FF_API_RESAMPLE_CLOSE_OPEN
245  /* if resampling was not enabled previously, re-initialize the
246  AVAudioResampleContext and force resampling */
247  int fifo_samples;
248  int restore_matrix = 0;
249  double matrix[AVRESAMPLE_MAX_CHANNELS * AVRESAMPLE_MAX_CHANNELS] = { 0 };
250 
251  /* buffer any remaining samples in the output FIFO before closing */
252  fifo_samples = av_audio_fifo_size(avr->out_fifo);
253  if (fifo_samples > 0) {
254  fifo_buf = ff_audio_data_alloc(avr->out_channels, fifo_samples,
255  avr->out_sample_fmt, NULL);
256  if (!fifo_buf)
257  return AVERROR(EINVAL);
258  ret = ff_audio_data_read_from_fifo(avr->out_fifo, fifo_buf,
259  fifo_samples);
260  if (ret < 0)
261  goto reinit_fail;
262  }
263  /* save the channel mixing matrix */
264  if (avr->am) {
266  if (ret < 0)
267  goto reinit_fail;
268  restore_matrix = 1;
269  }
270 
271  /* close the AVAudioResampleContext */
272  avresample_close(avr);
273 
274  avr->force_resampling = 1;
275 
276  /* restore the channel mixing matrix */
277  if (restore_matrix) {
279  if (ret < 0)
280  goto reinit_fail;
281  }
282 
283  /* re-open the AVAudioResampleContext */
284  ret = avresample_open(avr);
285  if (ret < 0)
286  goto reinit_fail;
287 
288  /* restore buffered samples to the output FIFO */
289  if (fifo_samples > 0) {
290  ret = ff_audio_data_add_to_fifo(avr->out_fifo, fifo_buf, 0,
291  fifo_samples);
292  if (ret < 0)
293  goto reinit_fail;
294  ff_audio_data_free(&fifo_buf);
295  }
296 #else
297  av_log(avr, AV_LOG_ERROR, "Unable to set resampling compensation\n");
298  return AVERROR(EINVAL);
299 #endif
300  }
301  c = avr->resample;
303  if (compensation_distance) {
304  c->dst_incr = c->ideal_dst_incr - c->ideal_dst_incr *
305  (int64_t)sample_delta / compensation_distance;
306  } else {
307  c->dst_incr = c->ideal_dst_incr;
308  }
309  return 0;
310 
311 reinit_fail:
312  ff_audio_data_free(&fifo_buf);
313  return ret;
314 }
315 
316 static int resample(ResampleContext *c, void *dst, const void *src,
317  int *consumed, int src_size, int dst_size, int update_ctx,
318  int nearest_neighbour)
319 {
320  int dst_index;
321  unsigned int index = c->index;
322  int frac = c->frac;
323  int dst_incr_frac = c->dst_incr % c->src_incr;
324  int dst_incr = c->dst_incr / c->src_incr;
326 
327  if (!dst != !src)
328  return AVERROR(EINVAL);
329 
330  if (nearest_neighbour) {
331  uint64_t index2 = ((uint64_t)index) << 32;
332  int64_t incr = (1LL << 32) * c->dst_incr / c->src_incr;
333  dst_size = FFMIN(dst_size,
334  (src_size-1-index) * (int64_t)c->src_incr /
335  c->dst_incr);
336 
337  if (dst) {
338  for(dst_index = 0; dst_index < dst_size; dst_index++) {
339  c->resample_nearest(dst, dst_index, src, index2 >> 32);
340  index2 += incr;
341  }
342  } else {
343  dst_index = dst_size;
344  }
345  index += dst_index * dst_incr;
346  index += (frac + dst_index * (int64_t)dst_incr_frac) / c->src_incr;
347  frac = (frac + dst_index * (int64_t)dst_incr_frac) % c->src_incr;
348  } else {
349  for (dst_index = 0; dst_index < dst_size; dst_index++) {
350  int sample_index = index >> c->phase_shift;
351 
352  if (sample_index + c->filter_length > src_size)
353  break;
354 
355  if (dst)
356  c->resample_one(c, dst, dst_index, src, index, frac);
357 
358  frac += dst_incr_frac;
359  index += dst_incr;
360  if (frac >= c->src_incr) {
361  frac -= c->src_incr;
362  index++;
363  }
364  if (dst_index + 1 == compensation_distance) {
365  compensation_distance = 0;
366  dst_incr_frac = c->ideal_dst_incr % c->src_incr;
367  dst_incr = c->ideal_dst_incr / c->src_incr;
368  }
369  }
370  }
371  if (consumed)
372  *consumed = index >> c->phase_shift;
373 
374  if (update_ctx) {
375  index &= c->phase_mask;
376 
377  if (compensation_distance) {
378  compensation_distance -= dst_index;
379  if (compensation_distance <= 0)
380  return AVERROR_BUG;
381  }
382  c->frac = frac;
383  c->index = index;
384  c->dst_incr = dst_incr_frac + c->src_incr*dst_incr;
386  }
387 
388  return dst_index;
389 }
390 
392 {
393  int ch, in_samples, in_leftover, consumed = 0, out_samples = 0;
394  int ret = AVERROR(EINVAL);
395  int nearest_neighbour = (c->compensation_distance == 0 &&
396  c->filter_length == 1 &&
397  c->phase_shift == 0);
398 
399  in_samples = src ? src->nb_samples : 0;
400  in_leftover = c->buffer->nb_samples;
401 
402  /* add input samples to the internal buffer */
403  if (src) {
404  ret = ff_audio_data_combine(c->buffer, in_leftover, src, 0, in_samples);
405  if (ret < 0)
406  return ret;
407  } else if (in_leftover <= c->final_padding_samples) {
408  /* no remaining samples to flush */
409  return 0;
410  }
411 
412  if (!c->initial_padding_filled) {
414  int i;
415 
416  if (src && c->buffer->nb_samples < 2 * c->padding_size)
417  return 0;
418 
419  for (i = 0; i < c->padding_size; i++)
420  for (ch = 0; ch < c->buffer->channels; ch++) {
421  if (c->buffer->nb_samples > 2 * c->padding_size - i) {
422  memcpy(c->buffer->data[ch] + bps * i,
423  c->buffer->data[ch] + bps * (2 * c->padding_size - i), bps);
424  } else {
425  memset(c->buffer->data[ch] + bps * i, 0, bps);
426  }
427  }
428  c->initial_padding_filled = 1;
429  }
430 
431  if (!src && !c->final_padding_filled) {
433  int i;
434 
435  ret = ff_audio_data_realloc(c->buffer, in_samples + c->padding_size);
436  if (ret < 0) {
437  av_log(c->avr, AV_LOG_ERROR, "Error reallocating resampling buffer\n");
438  return AVERROR(ENOMEM);
439  }
440 
441  for (i = 0; i < c->padding_size; i++)
442  for (ch = 0; ch < c->buffer->channels; ch++) {
443  if (in_leftover > i) {
444  memcpy(c->buffer->data[ch] + bps * (in_leftover + i),
445  c->buffer->data[ch] + bps * (in_leftover - i - 1),
446  bps);
447  } else {
448  memset(c->buffer->data[ch] + bps * (in_leftover + i),
449  0, bps);
450  }
451  }
452  c->buffer->nb_samples += c->padding_size;
454  c->final_padding_filled = 1;
455  }
456 
457 
458  /* calculate output size and reallocate output buffer if needed */
459  /* TODO: try to calculate this without the dummy resample() run */
460  if (!dst->read_only && dst->allow_realloc) {
461  out_samples = resample(c, NULL, NULL, NULL, c->buffer->nb_samples,
462  INT_MAX, 0, nearest_neighbour);
463  ret = ff_audio_data_realloc(dst, out_samples);
464  if (ret < 0) {
465  av_log(c->avr, AV_LOG_ERROR, "error reallocating output\n");
466  return ret;
467  }
468  }
469 
470  /* resample each channel plane */
471  for (ch = 0; ch < c->buffer->channels; ch++) {
472  out_samples = resample(c, (void *)dst->data[ch],
473  (const void *)c->buffer->data[ch], &consumed,
475  ch + 1 == c->buffer->channels, nearest_neighbour);
476  }
477  if (out_samples < 0) {
478  av_log(c->avr, AV_LOG_ERROR, "error during resampling\n");
479  return out_samples;
480  }
481 
482  /* drain consumed samples from the internal buffer */
483  ff_audio_data_drain(c->buffer, consumed);
485 
486  av_dlog(c->avr, "resampled %d in + %d leftover to %d out + %d leftover\n",
487  in_samples, in_leftover, out_samples, c->buffer->nb_samples);
488 
489  dst->nb_samples = out_samples;
490  return 0;
491 }
492 
494 {
495  ResampleContext *c = avr->resample;
496 
497  if (!avr->resample_needed || !avr->resample)
498  return 0;
499 
500  return FFMAX(c->buffer->nb_samples - c->padding_size, 0);
501 }