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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  if (ARCH_ARM)
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 */
204  avr->internal_sample_fmt,
205  "resample buffer");
206  if (!c->buffer)
207  goto error;
208  c->buffer->nb_samples = c->padding_size;
210 
211  av_log(avr, AV_LOG_DEBUG, "resample: %s from %d Hz to %d Hz\n",
213  avr->in_sample_rate, avr->out_sample_rate);
214 
215  return c;
216 
217 error:
219  av_free(c->filter_bank);
220  av_free(c);
221  return NULL;
222 }
223 
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 
235 {
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;
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 
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;
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;
329  }
330 
331  return dst_index;
332 }
333 
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) {
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) {
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;
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,
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);
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 
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 }
float, planar
Definition: samplefmt.h:69
#define NULL
Definition: coverity.c:32
int initial_padding_filled
Definition: resample.h:51
int initial_padding_samples
Definition: resample.h:52
int padding_size
Definition: resample.h:50
int ff_audio_data_realloc(AudioData *a, int nb_samples)
Reallocate AudioData.
Definition: audio_data.c:162
Kaiser Windowed Sinc.
Definition: avresample.h:120
Audio buffer used for intermediate storage between conversion phases.
Definition: audio_data.h:37
void(* set_filter)(void *filter, double *tab, int phase, int tap_count)
Definition: resample.h:44
AudioData * ff_audio_data_alloc(int channels, int nb_samples, enum AVSampleFormat sample_fmt, const char *name)
Allocate AudioData.
Definition: audio_data.c:119
int allow_realloc
realloc is allowed
Definition: audio_data.h:52
double, planar
Definition: samplefmt.h:70
int avresample_set_compensation(AVAudioResampleContext *avr, int sample_delta, int compensation_distance)
Set compensation for resampling.
Definition: resample.c:233
void * av_mallocz(size_t size)
Allocate a memory block with alignment suitable for all memory accesses (including vectors if availab...
Definition: mem.c:252
double cutoff
resampling cutoff frequency.
Definition: internal.h:72
int nb_samples
current number of samples
Definition: audio_data.h:43
static int resample(ResampleContext *c, void *dst, const void *src, int *consumed, int src_size, int dst_size, int update_ctx, int nearest_neighbour)
Definition: resample.c:259
AudioData * buffer
Definition: resample.h:30
#define av_malloc(s)
AVAudioResampleContext * avr
Definition: af_resample.c:40
#define AV_LOG_TRACE
Extremely verbose debugging, useful for libav* development.
Definition: log.h:202
int read_only
data is read-only
Definition: audio_data.h:51
int compensation_distance
Definition: resample.h:38
enum AVResampleFilterType filter_type
Definition: resample.h:42
int av_reduce(int *dst_num, int *dst_den, int64_t num, int64_t den, int64_t max)
Reduce a fraction.
Definition: rational.c:35
#define av_log(a,...)
int ff_audio_resample(ResampleContext *c, AudioData *dst, AudioData *src)
Resample audio data.
Definition: resample.c:334
#define AV_LOG_ERROR
Something went wrong and cannot losslessly be recovered.
Definition: log.h:176
#define AVERROR(e)
Definition: error.h:43
int channels
channel count
Definition: audio_data.h:45
unsigned int index
Definition: resample.h:35
#define AV_LOG_DEBUG
Stuff which is only useful for libav* developers.
Definition: log.h:197
const char * av_get_sample_fmt_name(enum AVSampleFormat sample_fmt)
Return the name of sample_fmt, or NULL if sample_fmt is not recognized.
Definition: samplefmt.c:49
#define FFMAX(a, b)
Definition: common.h:94
int compensation_distance
Definition: resample2.c:71
ResampleContext * resample
resampling context
Definition: internal.h:95
av_cold void ff_audio_resample_init_arm(ResampleContext *c, enum AVSampleFormat sample_fmt)
Definition: resample_init.c:52
#define FFMIN(a, b)
Definition: common.h:96
signed 32 bits, planar
Definition: samplefmt.h:68
int phase_shift
log2 of the number of entries in the resampling polyphase filterbank
Definition: internal.h:70
int ff_audio_data_combine(AudioData *dst, int dst_offset, AudioData *src, int src_offset, int nb_samples)
Append data from one AudioData to the end of another.
Definition: audio_data.c:278
void ff_audio_data_drain(AudioData *a, int nb_samples)
Drain samples from the start of the AudioData.
Definition: audio_data.c:334
int linear_interp
if 1 then the resampling FIR filter will be linearly interpolated
Definition: internal.h:71
int kaiser_beta
beta value for Kaiser window (only applicable if filter_type == AV_FILTER_TYPE_KAISER) ...
Definition: internal.h:74
#define src
Definition: vp9dsp.c:530
int in_sample_rate
input sample rate
Definition: internal.h:58
int avresample_get_delay(AVAudioResampleContext *avr)
Return the number of samples currently in the resampling delay buffer.
Definition: resample.c:438
void ff_audio_resample_free(ResampleContext **c)
Free a ResampleContext.
Definition: resample.c:224
static int build_filter(ResampleContext *c, double factor)
Definition: resample.c:67
uint8_t * data[AVRESAMPLE_MAX_CHANNELS]
data plane pointers
Definition: audio_data.h:39
enum AVResampleFilterType filter_type
resampling filter type
Definition: internal.h:73
enum AVSampleFormat internal_sample_fmt
internal sample format
Definition: internal.h:62
Replacements for frequently missing libm functions.
#define AVERROR_BUG
Internal bug, also see AVERROR_BUG2.
Definition: error.h:50
int filter_length
Definition: resample.h:32
ResampleContext * ff_audio_resample_init(AVAudioResampleContext *avr)
Allocate and initialize a ResampleContext.
Definition: resample.c:120
int index
Definition: gxfenc.c:89
int filter_size
length of each FIR filter in the resampling filterbank relative to the cutoff frequency ...
Definition: internal.h:69
Blackman Nuttall Windowed Sinc.
Definition: avresample.h:119
static const int factor[16]
Definition: vf_pp7.c:75
int final_padding_filled
Definition: resample.h:53
void(* resample_one)(struct ResampleContext *c, void *dst0, int dst_index, const void *src0, unsigned int index, int frac)
Definition: resample.h:45
int ideal_dst_incr
Definition: resample.h:33
int av_get_bytes_per_sample(enum AVSampleFormat sample_fmt)
Return number of bytes per sample.
Definition: samplefmt.c:106
common internal and external API header
int resample_channels
number of channels used for resampling
Definition: internal.h:79
static double c[64]
int resample_needed
resampling is needed
Definition: internal.h:83
uint8_t pi<< 24) CONV_FUNC(AV_SAMPLE_FMT_S64, int64_t, AV_SAMPLE_FMT_U8,(uint64_t)((*(constuint8_t *) pi-0x80U))<< 56) CONV_FUNC(AV_SAMPLE_FMT_FLT, float, AV_SAMPLE_FMT_U8,(*(constuint8_t *) pi-0x80)*(1.0f/(1<< 7))) CONV_FUNC(AV_SAMPLE_FMT_DBL, double, AV_SAMPLE_FMT_U8,(*(constuint8_t *) pi-0x80)*(1.0/(1<< 7))) CONV_FUNC(AV_SAMPLE_FMT_U8, uint8_t, AV_SAMPLE_FMT_S16,(*(constint16_t *) pi >>8)+0x80) CONV_FUNC(AV_SAMPLE_FMT_S64, int64_t, AV_SAMPLE_FMT_S16,(uint64_t)(*(constint16_t *) pi)<< 48) CONV_FUNC(AV_SAMPLE_FMT_FLT, float, AV_SAMPLE_FMT_S16,*(constint16_t *) pi *(1.0f/(1<< 15))) CONV_FUNC(AV_SAMPLE_FMT_DBL, double, AV_SAMPLE_FMT_S16,*(constint16_t *) pi *(1.0/(1<< 15))) CONV_FUNC(AV_SAMPLE_FMT_U8, uint8_t, AV_SAMPLE_FMT_S32,(*(constint32_t *) pi >>24)+0x80) CONV_FUNC(AV_SAMPLE_FMT_S64, int64_t, AV_SAMPLE_FMT_S32,(uint64_t)(*(constint32_t *) pi)<< 32) CONV_FUNC(AV_SAMPLE_FMT_FLT, float, AV_SAMPLE_FMT_S32,*(constint32_t *) pi *(1.0f/(1U<< 31))) CONV_FUNC(AV_SAMPLE_FMT_DBL, double, AV_SAMPLE_FMT_S32,*(constint32_t *) pi *(1.0/(1U<< 31))) CONV_FUNC(AV_SAMPLE_FMT_U8, uint8_t, AV_SAMPLE_FMT_S64,(*(constint64_t *) pi >>56)+0x80) CONV_FUNC(AV_SAMPLE_FMT_FLT, float, AV_SAMPLE_FMT_S64,*(constint64_t *) pi *(1.0f/(INT64_C(1)<< 63))) CONV_FUNC(AV_SAMPLE_FMT_DBL, double, AV_SAMPLE_FMT_S64,*(constint64_t *) pi *(1.0/(INT64_C(1)<< 63))) CONV_FUNC(AV_SAMPLE_FMT_U8, uint8_t, AV_SAMPLE_FMT_FLT, av_clip_uint8(lrintf(*(constfloat *) pi *(1<< 7))+0x80)) CONV_FUNC(AV_SAMPLE_FMT_S16, int16_t, AV_SAMPLE_FMT_FLT, av_clip_int16(lrintf(*(constfloat *) pi *(1<< 15)))) CONV_FUNC(AV_SAMPLE_FMT_S32, int32_t, AV_SAMPLE_FMT_FLT, av_clipl_int32(llrintf(*(constfloat *) pi *(1U<< 31)))) CONV_FUNC(AV_SAMPLE_FMT_S64, int64_t, AV_SAMPLE_FMT_FLT, llrintf(*(constfloat *) pi *(INT64_C(1)<< 63))) CONV_FUNC(AV_SAMPLE_FMT_U8, uint8_t, AV_SAMPLE_FMT_DBL, av_clip_uint8(lrint(*(constdouble *) pi *(1<< 7))+0x80)) CONV_FUNC(AV_SAMPLE_FMT_S16, int16_t, AV_SAMPLE_FMT_DBL, av_clip_int16(lrint(*(constdouble *) pi *(1<< 15)))) CONV_FUNC(AV_SAMPLE_FMT_S32, int32_t, AV_SAMPLE_FMT_DBL, av_clipl_int32(llrint(*(constdouble *) pi *(1U<< 31)))) CONV_FUNC(AV_SAMPLE_FMT_S64, int64_t, AV_SAMPLE_FMT_DBL, llrint(*(constdouble *) pi *(INT64_C(1)<< 63)))#defineFMT_PAIR_FUNC(out, in) staticconv_func_type *constfmt_pair_to_conv_functions[AV_SAMPLE_FMT_NB *AV_SAMPLE_FMT_NB]={FMT_PAIR_FUNC(AV_SAMPLE_FMT_U8, AV_SAMPLE_FMT_U8), FMT_PAIR_FUNC(AV_SAMPLE_FMT_S16, AV_SAMPLE_FMT_U8), FMT_PAIR_FUNC(AV_SAMPLE_FMT_S32, AV_SAMPLE_FMT_U8), FMT_PAIR_FUNC(AV_SAMPLE_FMT_FLT, AV_SAMPLE_FMT_U8), FMT_PAIR_FUNC(AV_SAMPLE_FMT_DBL, AV_SAMPLE_FMT_U8), FMT_PAIR_FUNC(AV_SAMPLE_FMT_S64, AV_SAMPLE_FMT_U8), FMT_PAIR_FUNC(AV_SAMPLE_FMT_U8, AV_SAMPLE_FMT_S16), FMT_PAIR_FUNC(AV_SAMPLE_FMT_S16, AV_SAMPLE_FMT_S16), FMT_PAIR_FUNC(AV_SAMPLE_FMT_S32, AV_SAMPLE_FMT_S16), FMT_PAIR_FUNC(AV_SAMPLE_FMT_FLT, AV_SAMPLE_FMT_S16), FMT_PAIR_FUNC(AV_SAMPLE_FMT_DBL, AV_SAMPLE_FMT_S16), FMT_PAIR_FUNC(AV_SAMPLE_FMT_S64, AV_SAMPLE_FMT_S16), FMT_PAIR_FUNC(AV_SAMPLE_FMT_U8, AV_SAMPLE_FMT_S32), FMT_PAIR_FUNC(AV_SAMPLE_FMT_S16, AV_SAMPLE_FMT_S32), FMT_PAIR_FUNC(AV_SAMPLE_FMT_S32, AV_SAMPLE_FMT_S32), FMT_PAIR_FUNC(AV_SAMPLE_FMT_FLT, AV_SAMPLE_FMT_S32), FMT_PAIR_FUNC(AV_SAMPLE_FMT_DBL, AV_SAMPLE_FMT_S32), FMT_PAIR_FUNC(AV_SAMPLE_FMT_S64, AV_SAMPLE_FMT_S32), FMT_PAIR_FUNC(AV_SAMPLE_FMT_U8, AV_SAMPLE_FMT_FLT), FMT_PAIR_FUNC(AV_SAMPLE_FMT_S16, AV_SAMPLE_FMT_FLT), FMT_PAIR_FUNC(AV_SAMPLE_FMT_S32, AV_SAMPLE_FMT_FLT), FMT_PAIR_FUNC(AV_SAMPLE_FMT_FLT, AV_SAMPLE_FMT_FLT), FMT_PAIR_FUNC(AV_SAMPLE_FMT_DBL, AV_SAMPLE_FMT_FLT), FMT_PAIR_FUNC(AV_SAMPLE_FMT_S64, AV_SAMPLE_FMT_FLT), FMT_PAIR_FUNC(AV_SAMPLE_FMT_U8, AV_SAMPLE_FMT_DBL), FMT_PAIR_FUNC(AV_SAMPLE_FMT_S16, AV_SAMPLE_FMT_DBL), FMT_PAIR_FUNC(AV_SAMPLE_FMT_S32, AV_SAMPLE_FMT_DBL), FMT_PAIR_FUNC(AV_SAMPLE_FMT_FLT, AV_SAMPLE_FMT_DBL), FMT_PAIR_FUNC(AV_SAMPLE_FMT_DBL, AV_SAMPLE_FMT_DBL), FMT_PAIR_FUNC(AV_SAMPLE_FMT_S64, AV_SAMPLE_FMT_DBL), FMT_PAIR_FUNC(AV_SAMPLE_FMT_U8, AV_SAMPLE_FMT_S64), FMT_PAIR_FUNC(AV_SAMPLE_FMT_S16, AV_SAMPLE_FMT_S64), FMT_PAIR_FUNC(AV_SAMPLE_FMT_S32, AV_SAMPLE_FMT_S64), FMT_PAIR_FUNC(AV_SAMPLE_FMT_FLT, AV_SAMPLE_FMT_S64), FMT_PAIR_FUNC(AV_SAMPLE_FMT_DBL, AV_SAMPLE_FMT_S64), FMT_PAIR_FUNC(AV_SAMPLE_FMT_S64, AV_SAMPLE_FMT_S64),};staticvoidcpy1(uint8_t **dst, constuint8_t **src, intlen){memcpy(*dst,*src, len);}staticvoidcpy2(uint8_t **dst, constuint8_t **src, intlen){memcpy(*dst,*src, 2 *len);}staticvoidcpy4(uint8_t **dst, constuint8_t **src, intlen){memcpy(*dst,*src, 4 *len);}staticvoidcpy8(uint8_t **dst, constuint8_t **src, intlen){memcpy(*dst,*src, 8 *len);}AudioConvert *swri_audio_convert_alloc(enumAVSampleFormatout_fmt, enumAVSampleFormatin_fmt, intchannels, constint *ch_map, intflags){AudioConvert *ctx;conv_func_type *f=fmt_pair_to_conv_functions[av_get_packed_sample_fmt(out_fmt)+AV_SAMPLE_FMT_NB *av_get_packed_sample_fmt(in_fmt)];if(!f) returnNULL;ctx=av_mallocz(sizeof(*ctx));if(!ctx) returnNULL;if(channels==1){in_fmt=av_get_planar_sample_fmt(in_fmt);out_fmt=av_get_planar_sample_fmt(out_fmt);}ctx->channels=channels;ctx->conv_f=f;ctx->ch_map=ch_map;if(in_fmt==AV_SAMPLE_FMT_U8||in_fmt==AV_SAMPLE_FMT_U8P) memset(ctx->silence, 0x80, sizeof(ctx->silence));if(out_fmt==in_fmt &&!ch_map){switch(av_get_bytes_per_sample(in_fmt)){case1:ctx->simd_f=cpy1;break;case2:ctx->simd_f=cpy2;break;case4:ctx->simd_f=cpy4;break;case8:ctx->simd_f=cpy8;break;}}if(HAVE_YASM &&1) swri_audio_convert_init_x86(ctx, out_fmt, in_fmt, channels);if(ARCH_ARM) swri_audio_convert_init_arm(ctx, out_fmt, in_fmt, channels);if(ARCH_AARCH64) swri_audio_convert_init_aarch64(ctx, out_fmt, in_fmt, channels);returnctx;}voidswri_audio_convert_free(AudioConvert **ctx){av_freep(ctx);}intswri_audio_convert(AudioConvert *ctx, AudioData *out, AudioData *in, intlen){intch;intoff=0;constintos=(out->planar?1:out->ch_count)*out->bps;unsignedmisaligned=0;av_assert0(ctx->channels==out->ch_count);if(ctx->in_simd_align_mask){intplanes=in->planar?in->ch_count:1;unsignedm=0;for(ch=0;ch< planes;ch++) m|=(intptr_t) in->ch[ch];misaligned|=m &ctx->in_simd_align_mask;}if(ctx->out_simd_align_mask){intplanes=out->planar?out->ch_count:1;unsignedm=0;for(ch=0;ch< planes;ch++) m|=(intptr_t) out->ch[ch];misaligned|=m &ctx->out_simd_align_mask;}if(ctx->simd_f &&!ctx->ch_map &&!misaligned){off=len &~15;av_assert1(off >=0);av_assert1(off<=len);av_assert2(ctx->channels==SWR_CH_MAX||!in->ch[ctx->channels]);if(off >0){if(out->planar==in->planar){intplanes=out->planar?out->ch_count:1;for(ch=0;ch< planes;ch++){ctx->simd_f(out-> ch ch
Definition: audioconvert.c:56
unsigned bps
Definition: movenc.c:1379
#define av_free(p)
int final_padding_samples
Definition: resample.h:54
int allocated_samples
number of samples the buffer can hold
Definition: audio_data.h:42
uint8_t * filter_bank
Definition: resample.h:31
static const struct twinvq_data tab
#define av_freep(p)
int out_sample_rate
output sample rate
Definition: internal.h:61
void ff_audio_data_free(AudioData **a)
Free AudioData.
Definition: audio_data.c:217
signed 16 bits, planar
Definition: samplefmt.h:67
#define M_PI
Definition: mathematics.h:52
void(* resample_nearest)(void *dst0, int dst_index, const void *src0, unsigned int index)
Definition: resample.h:48
static double bessel(double x)
Definition: resample.c:50
av_cold void ff_audio_resample_init_aarch64(ResampleContext *c, enum AVSampleFormat sample_fmt)
Definition: resample_init.c:48