lavr: add a function for checking whether AVAudioResampleContext is open
[libav.git] / libavresample / 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 Libav.
6 *
7 * Libav 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 * Libav 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 Libav; if not, write to the Free Software
19 * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
20 */
21
1d9c2dc8 22#include "libavutil/common.h"
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23#include "libavutil/libm.h"
24#include "libavutil/log.h"
25#include "internal.h"
4d68269d 26#include "resample.h"
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27#include "audio_data.h"
28
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29struct ResampleContext {
30 AVAudioResampleContext *avr;
31 AudioData *buffer;
64103976 32 uint8_t *filter_bank;
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33 int filter_length;
34 int ideal_dst_incr;
35 int dst_incr;
36 int index;
37 int frac;
38 int src_incr;
39 int compensation_distance;
40 int phase_shift;
41 int phase_mask;
42 int linear;
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43 enum AVResampleFilterType filter_type;
44 int kaiser_beta;
c8af852b 45 double factor;
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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);
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50};
51
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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
3aa696e8 72/* 0th order modified bessel function of the first kind. */
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73static 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
3aa696e8 89/* Build a polyphase filterbank. */
64103976 90static int build_filter(ResampleContext *c)
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91{
92 int ph, i;
64103976 93 double x, y, w, factor;
c8af852b 94 double *tab;
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95 int tap_count = c->filter_length;
96 int phase_count = 1 << c->phase_shift;
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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 */
64103976 104 factor = FFMIN(c->factor, 1.0);
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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;
64103976 112 switch (c->filter_type) {
372647ae 113 case AV_RESAMPLE_FILTER_TYPE_CUBIC: {
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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 }
372647ae 120 case AV_RESAMPLE_FILTER_TYPE_BLACKMAN_NUTTALL:
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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;
372647ae 126 case AV_RESAMPLE_FILTER_TYPE_KAISER:
c8af852b 127 w = 2.0 * x / (factor * tap_count * M_PI);
64103976 128 y *= bessel(c->kaiser_beta * sqrt(FFMAX(1 - w * w, 0)));
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129 break;
130 }
131
132 tab[i] = y;
133 norm += y;
134 }
c8af852b 135 /* normalize so that an uniform color remains the same */
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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);
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140 }
141
142 av_free(tab);
143 return 0;
144}
145
146ResampleContext *ff_audio_resample_init(AVAudioResampleContext *avr)
147{
148 ResampleContext *c;
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;
64103976 153 int felem_size;
c8af852b 154
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155 if (avr->internal_sample_fmt != AV_SAMPLE_FMT_S16P &&
156 avr->internal_sample_fmt != AV_SAMPLE_FMT_S32P &&
157 avr->internal_sample_fmt != AV_SAMPLE_FMT_FLTP &&
158 avr->internal_sample_fmt != AV_SAMPLE_FMT_DBLP) {
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159 av_log(avr, AV_LOG_ERROR, "Unsupported internal format for "
160 "resampling: %s\n",
161 av_get_sample_fmt_name(avr->internal_sample_fmt));
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);
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174 c->filter_type = avr->filter_type;
175 c->kaiser_beta = avr->kaiser_beta;
c8af852b 176
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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);
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198 if (!c->filter_bank)
199 goto error;
200
64103976 201 if (build_filter(c) < 0)
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202 goto error;
203
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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);
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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 */
219 c->buffer = ff_audio_data_alloc(avr->resample_channels, 0,
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",
226 av_get_sample_fmt_name(avr->internal_sample_fmt),
227 avr->in_sample_rate, avr->out_sample_rate);
228
229 return c;
230
231error:
232 ff_audio_data_free(&c->buffer);
233 av_free(c->filter_bank);
234 av_free(c);
235 return NULL;
236}
237
238void ff_audio_resample_free(ResampleContext **c)
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
247int avresample_set_compensation(AVAudioResampleContext *avr, int sample_delta,
248 int compensation_distance)
249{
250 ResampleContext *c;
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
c8af852b 259 if (!avr->resample_needed) {
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260#if FF_API_RESAMPLE_CLOSE_OPEN
261 /* if resampling was not enabled previously, re-initialize the
262 AVAudioResampleContext and force resampling */
c8af852b 263 int fifo_samples;
f322b207 264 int restore_matrix = 0;
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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 */
f322b207
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280 if (avr->am) {
281 ret = avresample_get_matrix(avr, matrix, AVRESAMPLE_MAX_CHANNELS);
282 if (ret < 0)
283 goto reinit_fail;
284 restore_matrix = 1;
285 }
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286
287 /* close the AVAudioResampleContext */
288 avresample_close(avr);
289
290 avr->force_resampling = 1;
291
292 /* restore the channel mixing matrix */
f322b207
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293 if (restore_matrix) {
294 ret = avresample_set_matrix(avr, matrix, AVRESAMPLE_MAX_CHANNELS);
295 if (ret < 0)
296 goto reinit_fail;
297 }
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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 }
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312#else
313 av_log(avr, AV_LOG_ERROR, "Unable to set resampling compensation\n");
314 return AVERROR(EINVAL);
315#endif
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316 }
317 c = avr->resample;
318 c->compensation_distance = compensation_distance;
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
327reinit_fail:
328 ff_audio_data_free(&fifo_buf);
329 return ret;
330}
331
64103976 332static int resample(ResampleContext *c, void *dst, const void *src,
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333 int *consumed, int src_size, int dst_size, int update_ctx)
334{
64103976 335 int dst_index;
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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;
340 int compensation_distance = c->compensation_distance;
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++) {
64103976 355 c->resample_one(c, 1, dst, dst_index, src, 0, index2 >> 32, 0);
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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++) {
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366 int sample_index = index >> c->phase_shift;
367
64103976
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368 if (sample_index + c->filter_length > src_size ||
369 -sample_index >= src_size)
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370 break;
371
64103976
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372 if (dst)
373 c->resample_one(c, 0, dst, dst_index, src, src_size, index, frac);
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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;
403 c->compensation_distance = compensation_distance;
404 }
405
406 return dst_index;
407}
408
1d86aa8b 409int ff_audio_resample(ResampleContext *c, AudioData *dst, AudioData *src)
c8af852b 410{
1d86aa8b 411 int ch, in_samples, in_leftover, consumed = 0, out_samples = 0;
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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++) {
64103976 443 out_samples = resample(c, (void *)dst->data[ch],
1d86aa8b 444 (const void *)c->buffer->data[ch], &consumed,
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445 c->buffer->nb_samples, dst->allocated_samples,
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 */
1d86aa8b 454 ff_audio_data_drain(c->buffer, consumed);
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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
463int avresample_get_delay(AVAudioResampleContext *avr)
464{
465 if (!avr->resample_needed || !avr->resample)
466 return 0;
467
468 return avr->resample->buffer->nb_samples;
469}