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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. */
bessel(double x)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. */
build_filter(ResampleContext * c,double factor)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) {
87             case AV_RESAMPLE_FILTER_TYPE_CUBIC: {
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             }
94             case AV_RESAMPLE_FILTER_TYPE_BLACKMAN_NUTTALL:
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;
100             case AV_RESAMPLE_FILTER_TYPE_KAISER:
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 
ff_audio_resample_init(AVAudioResampleContext * avr)120 ResampleContext *ff_audio_resample_init(AVAudioResampleContext *avr)
121 {
122     ResampleContext *c;
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 
129     if (avr->internal_sample_fmt != AV_SAMPLE_FMT_S16P &&
130         avr->internal_sample_fmt != AV_SAMPLE_FMT_S32P &&
131         avr->internal_sample_fmt != AV_SAMPLE_FMT_FLTP &&
132         avr->internal_sample_fmt != AV_SAMPLE_FMT_DBLP) {
133         av_log(avr, AV_LOG_ERROR, "Unsupported internal format for "
134                "resampling: %s\n",
135                av_get_sample_fmt_name(avr->internal_sample_fmt));
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)
174         ff_audio_resample_init_aarch64(c, avr->internal_sample_fmt);
175     if (ARCH_ARM)
176         ff_audio_resample_init_arm(c, avr->internal_sample_fmt);
177 
178     felem_size = av_get_bytes_per_sample(avr->internal_sample_fmt);
179     c->filter_bank = av_mallocz(c->filter_length * (phase_count + 1) * felem_size);
180     if (!c->filter_bank)
181         goto error;
182 
183     if (build_filter(c, factor) < 0)
184         goto error;
185 
186     memcpy(&c->filter_bank[(c->filter_length * phase_count + 1) * felem_size],
187            c->filter_bank, (c->filter_length - 1) * felem_size);
188     memcpy(&c->filter_bank[c->filter_length * phase_count * felem_size],
189            &c->filter_bank[(c->filter_length - 1) * felem_size], felem_size);
190 
191     c->compensation_distance = 0;
192     if (!av_reduce(&c->src_incr, &c->dst_incr, out_rate,
193                    in_rate * (int64_t)phase_count, INT32_MAX / 2))
194         goto error;
195     c->ideal_dst_incr = c->dst_incr;
196 
197     c->padding_size   = (c->filter_length - 1) / 2;
198     c->initial_padding_filled = 0;
199     c->index = 0;
200     c->frac  = 0;
201 
202     /* allocate internal buffer */
203     c->buffer = ff_audio_data_alloc(avr->resample_channels, c->padding_size,
204                                     avr->internal_sample_fmt,
205                                     "resample buffer");
206     if (!c->buffer)
207         goto error;
208     c->buffer->nb_samples      = c->padding_size;
209     c->initial_padding_samples = c->padding_size;
210 
211     av_log(avr, AV_LOG_DEBUG, "resample: %s from %d Hz to %d Hz\n",
212            av_get_sample_fmt_name(avr->internal_sample_fmt),
213            avr->in_sample_rate, avr->out_sample_rate);
214 
215     return c;
216 
217 error:
218     ff_audio_data_free(&c->buffer);
219     av_free(c->filter_bank);
220     av_free(c);
221     return NULL;
222 }
223 
ff_audio_resample_free(ResampleContext ** c)224 void ff_audio_resample_free(ResampleContext **c)
225 {
226     if (!*c)
227         return;
228     ff_audio_data_free(&(*c)->buffer);
229     av_free((*c)->filter_bank);
230     av_freep(c);
231 }
232 
avresample_set_compensation(AVAudioResampleContext * avr,int sample_delta,int compensation_distance)233 int avresample_set_compensation(AVAudioResampleContext *avr, int sample_delta,
234                                 int compensation_distance)
235 {
236     ResampleContext *c;
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         av_log(avr, AV_LOG_ERROR, "Unable to set resampling compensation\n");
245         return AVERROR(EINVAL);
246     }
247     c = avr->resample;
248     c->compensation_distance = compensation_distance;
249     if (compensation_distance) {
250         c->dst_incr = c->ideal_dst_incr - c->ideal_dst_incr *
251                       (int64_t)sample_delta / compensation_distance;
252     } else {
253         c->dst_incr = c->ideal_dst_incr;
254     }
255 
256     return 0;
257 }
258 
resample(ResampleContext * c,void * dst,const void * src,int * consumed,int src_size,int dst_size,int update_ctx,int nearest_neighbour)259 static int resample(ResampleContext *c, void *dst, const void *src,
260                     int *consumed, int src_size, int dst_size, int update_ctx,
261                     int nearest_neighbour)
262 {
263     int dst_index;
264     unsigned int index = c->index;
265     int frac          = c->frac;
266     int dst_incr_frac = c->dst_incr % c->src_incr;
267     int dst_incr      = c->dst_incr / c->src_incr;
268     int compensation_distance = c->compensation_distance;
269 
270     if (!dst != !src)
271         return AVERROR(EINVAL);
272 
273     if (nearest_neighbour) {
274         uint64_t index2 = ((uint64_t)index) << 32;
275         int64_t incr   = (1LL << 32) * c->dst_incr / c->src_incr;
276         dst_size       = FFMIN(dst_size,
277                                (src_size-1-index) * (int64_t)c->src_incr /
278                                c->dst_incr);
279 
280         if (dst) {
281             for(dst_index = 0; dst_index < dst_size; dst_index++) {
282                 c->resample_nearest(dst, dst_index, src, index2 >> 32);
283                 index2 += incr;
284             }
285         } else {
286             dst_index = dst_size;
287         }
288         index += dst_index * dst_incr;
289         index += (frac + dst_index * (int64_t)dst_incr_frac) / c->src_incr;
290         frac   = (frac + dst_index * (int64_t)dst_incr_frac) % c->src_incr;
291     } else {
292         for (dst_index = 0; dst_index < dst_size; dst_index++) {
293             int sample_index = index >> c->phase_shift;
294 
295             if (sample_index + c->filter_length > src_size)
296                 break;
297 
298             if (dst)
299                 c->resample_one(c, dst, dst_index, src, index, frac);
300 
301             frac  += dst_incr_frac;
302             index += dst_incr;
303             if (frac >= c->src_incr) {
304                 frac -= c->src_incr;
305                 index++;
306             }
307             if (dst_index + 1 == compensation_distance) {
308                 compensation_distance = 0;
309                 dst_incr_frac = c->ideal_dst_incr % c->src_incr;
310                 dst_incr      = c->ideal_dst_incr / c->src_incr;
311             }
312         }
313     }
314     if (consumed)
315         *consumed = index >> c->phase_shift;
316 
317     if (update_ctx) {
318         index &= c->phase_mask;
319 
320         if (compensation_distance) {
321             compensation_distance -= dst_index;
322             if (compensation_distance <= 0)
323                 return AVERROR_BUG;
324         }
325         c->frac     = frac;
326         c->index    = index;
327         c->dst_incr = dst_incr_frac + c->src_incr*dst_incr;
328         c->compensation_distance = compensation_distance;
329     }
330 
331     return dst_index;
332 }
333 
ff_audio_resample(ResampleContext * c,AudioData * dst,AudioData * src)334 int ff_audio_resample(ResampleContext *c, AudioData *dst, AudioData *src)
335 {
336     int ch, in_samples, in_leftover, consumed = 0, out_samples = 0;
337     int ret = AVERROR(EINVAL);
338     int nearest_neighbour = (c->compensation_distance == 0 &&
339                              c->filter_length == 1 &&
340                              c->phase_shift == 0);
341 
342     in_samples  = src ? src->nb_samples : 0;
343     in_leftover = c->buffer->nb_samples;
344 
345     /* add input samples to the internal buffer */
346     if (src) {
347         ret = ff_audio_data_combine(c->buffer, in_leftover, src, 0, in_samples);
348         if (ret < 0)
349             return ret;
350     } else if (in_leftover <= c->final_padding_samples) {
351         /* no remaining samples to flush */
352         return 0;
353     }
354 
355     if (!c->initial_padding_filled) {
356         int bps = av_get_bytes_per_sample(c->avr->internal_sample_fmt);
357         int i;
358 
359         if (src && c->buffer->nb_samples < 2 * c->padding_size)
360             return 0;
361 
362         for (i = 0; i < c->padding_size; i++)
363             for (ch = 0; ch < c->buffer->channels; ch++) {
364                 if (c->buffer->nb_samples > 2 * c->padding_size - i) {
365                     memcpy(c->buffer->data[ch] + bps * i,
366                            c->buffer->data[ch] + bps * (2 * c->padding_size - i), bps);
367                 } else {
368                     memset(c->buffer->data[ch] + bps * i, 0, bps);
369                 }
370             }
371         c->initial_padding_filled = 1;
372     }
373 
374     if (!src && !c->final_padding_filled) {
375         int bps = av_get_bytes_per_sample(c->avr->internal_sample_fmt);
376         int i;
377 
378         ret = ff_audio_data_realloc(c->buffer,
379                                     FFMAX(in_samples, in_leftover) +
380                                     c->padding_size);
381         if (ret < 0) {
382             av_log(c->avr, AV_LOG_ERROR, "Error reallocating resampling buffer\n");
383             return AVERROR(ENOMEM);
384         }
385 
386         for (i = 0; i < c->padding_size; i++)
387             for (ch = 0; ch < c->buffer->channels; ch++) {
388                 if (in_leftover > i) {
389                     memcpy(c->buffer->data[ch] + bps * (in_leftover + i),
390                            c->buffer->data[ch] + bps * (in_leftover - i - 1),
391                            bps);
392                 } else {
393                     memset(c->buffer->data[ch] + bps * (in_leftover + i),
394                            0, bps);
395                 }
396             }
397         c->buffer->nb_samples   += c->padding_size;
398         c->final_padding_samples = c->padding_size;
399         c->final_padding_filled  = 1;
400     }
401 
402 
403     /* calculate output size and reallocate output buffer if needed */
404     /* TODO: try to calculate this without the dummy resample() run */
405     if (!dst->read_only && dst->allow_realloc) {
406         out_samples = resample(c, NULL, NULL, NULL, c->buffer->nb_samples,
407                                INT_MAX, 0, nearest_neighbour);
408         ret = ff_audio_data_realloc(dst, out_samples);
409         if (ret < 0) {
410             av_log(c->avr, AV_LOG_ERROR, "error reallocating output\n");
411             return ret;
412         }
413     }
414 
415     /* resample each channel plane */
416     for (ch = 0; ch < c->buffer->channels; ch++) {
417         out_samples = resample(c, (void *)dst->data[ch],
418                                (const void *)c->buffer->data[ch], &consumed,
419                                c->buffer->nb_samples, dst->allocated_samples,
420                                ch + 1 == c->buffer->channels, nearest_neighbour);
421     }
422     if (out_samples < 0) {
423         av_log(c->avr, AV_LOG_ERROR, "error during resampling\n");
424         return out_samples;
425     }
426 
427     /* drain consumed samples from the internal buffer */
428     ff_audio_data_drain(c->buffer, consumed);
429     c->initial_padding_samples = FFMAX(c->initial_padding_samples - consumed, 0);
430 
431     av_log(c->avr, AV_LOG_TRACE, "resampled %d in + %d leftover to %d out + %d leftover\n",
432             in_samples, in_leftover, out_samples, c->buffer->nb_samples);
433 
434     dst->nb_samples = out_samples;
435     return 0;
436 }
437 
avresample_get_delay(AVAudioResampleContext * avr)438 int avresample_get_delay(AVAudioResampleContext *avr)
439 {
440     ResampleContext *c = avr->resample;
441 
442     if (!avr->resample_needed || !avr->resample)
443         return 0;
444 
445     return FFMAX(c->buffer->nb_samples - c->padding_size, 0);
446 }
447