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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 struct ResampleContext {
35  int dst_incr;
36  int index;
37  int frac;
38  int src_incr;
42  int linear;
45  double factor;
46  void (*set_filter)(void *filter, double *tab, int phase, int tap_count);
47  void (*resample_one)(struct ResampleContext *c, int no_filter, void *dst0,
48  int dst_index, const void *src0, int src_size,
49  int index, int frac);
50 };
51 
52 
53 /* double template */
54 #define CONFIG_RESAMPLE_DBL
55 #include "resample_template.c"
56 #undef CONFIG_RESAMPLE_DBL
57 
58 /* float template */
59 #define CONFIG_RESAMPLE_FLT
60 #include "resample_template.c"
61 #undef CONFIG_RESAMPLE_FLT
62 
63 /* s32 template */
64 #define CONFIG_RESAMPLE_S32
65 #include "resample_template.c"
66 #undef CONFIG_RESAMPLE_S32
67 
68 /* s16 template */
69 #include "resample_template.c"
70 
71 
72 /* 0th order modified bessel function of the first kind. */
73 static double bessel(double x)
74 {
75  double v = 1;
76  double lastv = 0;
77  double t = 1;
78  int i;
79 
80  x = x * x / 4;
81  for (i = 1; v != lastv; i++) {
82  lastv = v;
83  t *= x / (i * i);
84  v += t;
85  }
86  return v;
87 }
88 
89 /* Build a polyphase filterbank. */
91 {
92  int ph, i;
93  double x, y, w, factor;
94  double *tab;
95  int tap_count = c->filter_length;
96  int phase_count = 1 << c->phase_shift;
97  const int center = (tap_count - 1) / 2;
98 
99  tab = av_malloc(tap_count * sizeof(*tab));
100  if (!tab)
101  return AVERROR(ENOMEM);
102 
103  /* if upsampling, only need to interpolate, no filter */
104  factor = FFMIN(c->factor, 1.0);
105 
106  for (ph = 0; ph < phase_count; ph++) {
107  double norm = 0;
108  for (i = 0; i < tap_count; i++) {
109  x = M_PI * ((double)(i - center) - (double)ph / phase_count) * factor;
110  if (x == 0) y = 1.0;
111  else y = sin(x) / x;
112  switch (c->filter_type) {
114  const float d = -0.5; //first order derivative = -0.5
115  x = fabs(((double)(i - center) - (double)ph / phase_count) * factor);
116  if (x < 1.0) y = 1 - 3 * x*x + 2 * x*x*x + d * ( -x*x + x*x*x);
117  else y = d * (-4 + 8 * x - 5 * x*x + x*x*x);
118  break;
119  }
121  w = 2.0 * x / (factor * tap_count) + M_PI;
122  y *= 0.3635819 - 0.4891775 * cos( w) +
123  0.1365995 * cos(2 * w) -
124  0.0106411 * cos(3 * w);
125  break;
127  w = 2.0 * x / (factor * tap_count * M_PI);
128  y *= bessel(c->kaiser_beta * sqrt(FFMAX(1 - w * w, 0)));
129  break;
130  }
131 
132  tab[i] = y;
133  norm += y;
134  }
135  /* normalize so that an uniform color remains the same */
136  for (i = 0; i < tap_count; i++)
137  tab[i] = tab[i] / norm;
138 
139  c->set_filter(c->filter_bank, tab, ph, tap_count);
140  }
141 
142  av_free(tab);
143  return 0;
144 }
145 
147 {
149  int out_rate = avr->out_sample_rate;
150  int in_rate = avr->in_sample_rate;
151  double factor = FFMIN(out_rate * avr->cutoff / in_rate, 1.0);
152  int phase_count = 1 << avr->phase_shift;
153  int felem_size;
154 
159  av_log(avr, AV_LOG_ERROR, "Unsupported internal format for "
160  "resampling: %s\n",
162  return NULL;
163  }
164  c = av_mallocz(sizeof(*c));
165  if (!c)
166  return NULL;
167 
168  c->avr = avr;
169  c->phase_shift = avr->phase_shift;
170  c->phase_mask = phase_count - 1;
171  c->linear = avr->linear_interp;
172  c->factor = factor;
173  c->filter_length = FFMAX((int)ceil(avr->filter_size / factor), 1);
174  c->filter_type = avr->filter_type;
175  c->kaiser_beta = avr->kaiser_beta;
176 
177  switch (avr->internal_sample_fmt) {
178  case AV_SAMPLE_FMT_DBLP:
179  c->resample_one = resample_one_dbl;
180  c->set_filter = set_filter_dbl;
181  break;
182  case AV_SAMPLE_FMT_FLTP:
183  c->resample_one = resample_one_flt;
184  c->set_filter = set_filter_flt;
185  break;
186  case AV_SAMPLE_FMT_S32P:
187  c->resample_one = resample_one_s32;
188  c->set_filter = set_filter_s32;
189  break;
190  case AV_SAMPLE_FMT_S16P:
191  c->resample_one = resample_one_s16;
192  c->set_filter = set_filter_s16;
193  break;
194  }
195 
196  felem_size = av_get_bytes_per_sample(avr->internal_sample_fmt);
197  c->filter_bank = av_mallocz(c->filter_length * (phase_count + 1) * felem_size);
198  if (!c->filter_bank)
199  goto error;
200 
201  if (build_filter(c) < 0)
202  goto error;
203 
204  memcpy(&c->filter_bank[(c->filter_length * phase_count + 1) * felem_size],
205  c->filter_bank, (c->filter_length - 1) * felem_size);
206  memcpy(&c->filter_bank[c->filter_length * phase_count * felem_size],
207  &c->filter_bank[(c->filter_length - 1) * felem_size], felem_size);
208 
209  c->compensation_distance = 0;
210  if (!av_reduce(&c->src_incr, &c->dst_incr, out_rate,
211  in_rate * (int64_t)phase_count, INT32_MAX / 2))
212  goto error;
213  c->ideal_dst_incr = c->dst_incr;
214 
215  c->index = -phase_count * ((c->filter_length - 1) / 2);
216  c->frac = 0;
217 
218  /* allocate internal buffer */
220  avr->internal_sample_fmt,
221  "resample buffer");
222  if (!c->buffer)
223  goto error;
224 
225  av_log(avr, AV_LOG_DEBUG, "resample: %s from %d Hz to %d Hz\n",
227  avr->in_sample_rate, avr->out_sample_rate);
228 
229  return c;
230 
231 error:
233  av_free(c->filter_bank);
234  av_free(c);
235  return NULL;
236 }
237 
239 {
240  if (!*c)
241  return;
242  ff_audio_data_free(&(*c)->buffer);
243  av_free((*c)->filter_bank);
244  av_freep(c);
245 }
246 
249 {
251  AudioData *fifo_buf = NULL;
252  int ret = 0;
253 
254  if (compensation_distance < 0)
255  return AVERROR(EINVAL);
256  if (!compensation_distance && sample_delta)
257  return AVERROR(EINVAL);
258 
259  if (!avr->resample_needed) {
260 #if FF_API_RESAMPLE_CLOSE_OPEN
261  /* if resampling was not enabled previously, re-initialize the
262  AVAudioResampleContext and force resampling */
263  int fifo_samples;
264  int restore_matrix = 0;
265  double matrix[AVRESAMPLE_MAX_CHANNELS * AVRESAMPLE_MAX_CHANNELS] = { 0 };
266 
267  /* buffer any remaining samples in the output FIFO before closing */
268  fifo_samples = av_audio_fifo_size(avr->out_fifo);
269  if (fifo_samples > 0) {
270  fifo_buf = ff_audio_data_alloc(avr->out_channels, fifo_samples,
271  avr->out_sample_fmt, NULL);
272  if (!fifo_buf)
273  return AVERROR(EINVAL);
274  ret = ff_audio_data_read_from_fifo(avr->out_fifo, fifo_buf,
275  fifo_samples);
276  if (ret < 0)
277  goto reinit_fail;
278  }
279  /* save the channel mixing matrix */
280  if (avr->am) {
282  if (ret < 0)
283  goto reinit_fail;
284  restore_matrix = 1;
285  }
286 
287  /* close the AVAudioResampleContext */
288  avresample_close(avr);
289 
290  avr->force_resampling = 1;
291 
292  /* restore the channel mixing matrix */
293  if (restore_matrix) {
295  if (ret < 0)
296  goto reinit_fail;
297  }
298 
299  /* re-open the AVAudioResampleContext */
300  ret = avresample_open(avr);
301  if (ret < 0)
302  goto reinit_fail;
303 
304  /* restore buffered samples to the output FIFO */
305  if (fifo_samples > 0) {
306  ret = ff_audio_data_add_to_fifo(avr->out_fifo, fifo_buf, 0,
307  fifo_samples);
308  if (ret < 0)
309  goto reinit_fail;
310  ff_audio_data_free(&fifo_buf);
311  }
312 #else
313  av_log(avr, AV_LOG_ERROR, "Unable to set resampling compensation\n");
314  return AVERROR(EINVAL);
315 #endif
316  }
317  c = avr->resample;
319  if (compensation_distance) {
320  c->dst_incr = c->ideal_dst_incr - c->ideal_dst_incr *
321  (int64_t)sample_delta / compensation_distance;
322  } else {
323  c->dst_incr = c->ideal_dst_incr;
324  }
325  return 0;
326 
327 reinit_fail:
328  ff_audio_data_free(&fifo_buf);
329  return ret;
330 }
331 
332 static int resample(ResampleContext *c, void *dst, const void *src,
333  int *consumed, int src_size, int dst_size, int update_ctx)
334 {
335  int dst_index;
336  int index = c->index;
337  int frac = c->frac;
338  int dst_incr_frac = c->dst_incr % c->src_incr;
339  int dst_incr = c->dst_incr / c->src_incr;
341 
342  if (!dst != !src)
343  return AVERROR(EINVAL);
344 
345  if (compensation_distance == 0 && c->filter_length == 1 &&
346  c->phase_shift == 0) {
347  int64_t index2 = ((int64_t)index) << 32;
348  int64_t incr = (1LL << 32) * c->dst_incr / c->src_incr;
349  dst_size = FFMIN(dst_size,
350  (src_size-1-index) * (int64_t)c->src_incr /
351  c->dst_incr);
352 
353  if (dst) {
354  for(dst_index = 0; dst_index < dst_size; dst_index++) {
355  c->resample_one(c, 1, dst, dst_index, src, 0, index2 >> 32, 0);
356  index2 += incr;
357  }
358  } else {
359  dst_index = dst_size;
360  }
361  index += dst_index * dst_incr;
362  index += (frac + dst_index * (int64_t)dst_incr_frac) / c->src_incr;
363  frac = (frac + dst_index * (int64_t)dst_incr_frac) % c->src_incr;
364  } else {
365  for (dst_index = 0; dst_index < dst_size; dst_index++) {
366  int sample_index = index >> c->phase_shift;
367 
368  if (sample_index + c->filter_length > src_size ||
369  -sample_index >= src_size)
370  break;
371 
372  if (dst)
373  c->resample_one(c, 0, dst, dst_index, src, src_size, index, frac);
374 
375  frac += dst_incr_frac;
376  index += dst_incr;
377  if (frac >= c->src_incr) {
378  frac -= c->src_incr;
379  index++;
380  }
381  if (dst_index + 1 == compensation_distance) {
382  compensation_distance = 0;
383  dst_incr_frac = c->ideal_dst_incr % c->src_incr;
384  dst_incr = c->ideal_dst_incr / c->src_incr;
385  }
386  }
387  }
388  if (consumed)
389  *consumed = FFMAX(index, 0) >> c->phase_shift;
390 
391  if (update_ctx) {
392  if (index >= 0)
393  index &= c->phase_mask;
394 
395  if (compensation_distance) {
396  compensation_distance -= dst_index;
397  if (compensation_distance <= 0)
398  return AVERROR_BUG;
399  }
400  c->frac = frac;
401  c->index = index;
402  c->dst_incr = dst_incr_frac + c->src_incr*dst_incr;
404  }
405 
406  return dst_index;
407 }
408 
410 {
411  int ch, in_samples, in_leftover, consumed = 0, out_samples = 0;
412  int ret = AVERROR(EINVAL);
413 
414  in_samples = src ? src->nb_samples : 0;
415  in_leftover = c->buffer->nb_samples;
416 
417  /* add input samples to the internal buffer */
418  if (src) {
419  ret = ff_audio_data_combine(c->buffer, in_leftover, src, 0, in_samples);
420  if (ret < 0)
421  return ret;
422  } else if (!in_leftover) {
423  /* no remaining samples to flush */
424  return 0;
425  } else {
426  /* TODO: pad buffer to flush completely */
427  }
428 
429  /* calculate output size and reallocate output buffer if needed */
430  /* TODO: try to calculate this without the dummy resample() run */
431  if (!dst->read_only && dst->allow_realloc) {
432  out_samples = resample(c, NULL, NULL, NULL, c->buffer->nb_samples,
433  INT_MAX, 0);
434  ret = ff_audio_data_realloc(dst, out_samples);
435  if (ret < 0) {
436  av_log(c->avr, AV_LOG_ERROR, "error reallocating output\n");
437  return ret;
438  }
439  }
440 
441  /* resample each channel plane */
442  for (ch = 0; ch < c->buffer->channels; ch++) {
443  out_samples = resample(c, (void *)dst->data[ch],
444  (const void *)c->buffer->data[ch], &consumed,
446  ch + 1 == c->buffer->channels);
447  }
448  if (out_samples < 0) {
449  av_log(c->avr, AV_LOG_ERROR, "error during resampling\n");
450  return out_samples;
451  }
452 
453  /* drain consumed samples from the internal buffer */
454  ff_audio_data_drain(c->buffer, consumed);
455 
456  av_dlog(c->avr, "resampled %d in + %d leftover to %d out + %d leftover\n",
457  in_samples, in_leftover, out_samples, c->buffer->nb_samples);
458 
459  dst->nb_samples = out_samples;
460  return 0;
461 }
462 
464 {
465  if (!avr->resample_needed || !avr->resample)
466  return 0;
467 
468  return avr->resample->buffer->nb_samples;
469 }