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utils.c
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1 /*
2  * Copyright (c) 2012 Justin Ruggles <justin.ruggles@gmail.com>
3  *
4  * This file is part of FFmpeg.
5  *
6  * FFmpeg is free software; you can redistribute it and/or
7  * modify it under the terms of the GNU Lesser General Public
8  * License as published by the Free Software Foundation; either
9  * version 2.1 of the License, or (at your option) any later version.
10  *
11  * FFmpeg is distributed in the hope that it will be useful,
12  * but WITHOUT ANY WARRANTY; without even the implied warranty of
13  * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
14  * Lesser General Public License for more details.
15  *
16  * You should have received a copy of the GNU Lesser General Public
17  * License along with FFmpeg; if not, write to the Free Software
18  * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
19  */
20 
21 #include "libavutil/common.h"
22 #include "libavutil/dict.h"
23 // #include "libavutil/error.h"
24 #include "libavutil/frame.h"
25 #include "libavutil/log.h"
26 #include "libavutil/mem.h"
27 #include "libavutil/opt.h"
28 
29 #include "avresample.h"
30 #include "internal.h"
31 #include "audio_data.h"
32 #include "audio_convert.h"
33 #include "audio_mix.h"
34 #include "resample.h"
35 
37 {
38  int ret;
39 
40  if (avresample_is_open(avr)) {
41  av_log(avr, AV_LOG_ERROR, "The resampling context is already open.\n");
42  return AVERROR(EINVAL);
43  }
44 
45  /* set channel mixing parameters */
47  if (avr->in_channels <= 0 || avr->in_channels > AVRESAMPLE_MAX_CHANNELS) {
48  av_log(avr, AV_LOG_ERROR, "Invalid input channel layout: %"PRIu64"\n",
49  avr->in_channel_layout);
50  return AVERROR(EINVAL);
51  }
53  if (avr->out_channels <= 0 || avr->out_channels > AVRESAMPLE_MAX_CHANNELS) {
54  av_log(avr, AV_LOG_ERROR, "Invalid output channel layout: %"PRIu64"\n",
55  avr->out_channel_layout);
56  return AVERROR(EINVAL);
57  }
59  avr->downmix_needed = avr->in_channels > avr->out_channels;
60  avr->upmix_needed = avr->out_channels > avr->in_channels ||
61  (!avr->downmix_needed && (avr->mix_matrix ||
63  avr->mixing_needed = avr->downmix_needed || avr->upmix_needed;
64 
65  /* set resampling parameters */
66  avr->resample_needed = avr->in_sample_rate != avr->out_sample_rate ||
67  avr->force_resampling;
68 
69  /* select internal sample format if not specified by the user */
71  (avr->mixing_needed || avr->resample_needed)) {
74  int max_bps = FFMAX(av_get_bytes_per_sample(in_fmt),
75  av_get_bytes_per_sample(out_fmt));
76  if (max_bps <= 2) {
78  } else if (avr->mixing_needed) {
80  } else {
81  if (max_bps <= 4) {
82  if (in_fmt == AV_SAMPLE_FMT_S32P ||
83  out_fmt == AV_SAMPLE_FMT_S32P) {
84  if (in_fmt == AV_SAMPLE_FMT_FLTP ||
85  out_fmt == AV_SAMPLE_FMT_FLTP) {
86  /* if one is s32 and the other is flt, use dbl */
88  } else {
89  /* if one is s32 and the other is s32, s16, or u8, use s32 */
91  }
92  } else {
93  /* if one is flt and the other is flt, s16 or u8, use flt */
95  }
96  } else {
97  /* if either is dbl, use dbl */
99  }
100  }
101  av_log(avr, AV_LOG_DEBUG, "Using %s as internal sample format\n",
103  }
104 
105  /* we may need to add an extra conversion in order to remap channels if
106  the output format is not planar */
107  if (avr->use_channel_map && !avr->mixing_needed && !avr->resample_needed &&
110  }
111 
112  /* set sample format conversion parameters */
113  if (avr->resample_needed || avr->mixing_needed)
115  else
116  avr->in_convert_needed = avr->use_channel_map &&
118 
119  if (avr->resample_needed || avr->mixing_needed || avr->in_convert_needed)
121  else
122  avr->out_convert_needed = avr->in_sample_fmt != avr->out_sample_fmt;
123 
124  avr->in_copy_needed = !avr->in_convert_needed && (avr->mixing_needed ||
125  (avr->use_channel_map && avr->resample_needed));
126 
127  if (avr->use_channel_map) {
128  if (avr->in_copy_needed) {
129  avr->remap_point = REMAP_IN_COPY;
130  av_log(avr, AV_LOG_TRACE, "remap channels during in_copy\n");
131  } else if (avr->in_convert_needed) {
133  av_log(avr, AV_LOG_TRACE, "remap channels during in_convert\n");
134  } else if (avr->out_convert_needed) {
136  av_log(avr, AV_LOG_TRACE, "remap channels during out_convert\n");
137  } else {
139  av_log(avr, AV_LOG_TRACE, "remap channels during out_copy\n");
140  }
141 
142 #ifdef DEBUG
143  {
144  int ch;
145  av_log(avr, AV_LOG_TRACE, "output map: ");
146  if (avr->ch_map_info.do_remap)
147  for (ch = 0; ch < avr->in_channels; ch++)
148  av_log(avr, AV_LOG_TRACE, " % 2d", avr->ch_map_info.channel_map[ch]);
149  else
150  av_log(avr, AV_LOG_TRACE, "n/a");
151  av_log(avr, AV_LOG_TRACE, "\n");
152  av_log(avr, AV_LOG_TRACE, "copy map: ");
153  if (avr->ch_map_info.do_copy)
154  for (ch = 0; ch < avr->in_channels; ch++)
155  av_log(avr, AV_LOG_TRACE, " % 2d", avr->ch_map_info.channel_copy[ch]);
156  else
157  av_log(avr, AV_LOG_TRACE, "n/a");
158  av_log(avr, AV_LOG_TRACE, "\n");
159  av_log(avr, AV_LOG_TRACE, "zero map: ");
160  if (avr->ch_map_info.do_zero)
161  for (ch = 0; ch < avr->in_channels; ch++)
162  av_log(avr, AV_LOG_TRACE, " % 2d", avr->ch_map_info.channel_zero[ch]);
163  else
164  av_log(avr, AV_LOG_TRACE, "n/a");
165  av_log(avr, AV_LOG_TRACE, "\n");
166  av_log(avr, AV_LOG_TRACE, "input map: ");
167  for (ch = 0; ch < avr->in_channels; ch++)
168  av_log(avr, AV_LOG_TRACE, " % 2d", avr->ch_map_info.input_map[ch]);
169  av_log(avr, AV_LOG_TRACE, "\n");
170  }
171 #endif
172  } else
173  avr->remap_point = REMAP_NONE;
174 
175  /* allocate buffers */
176  if (avr->in_copy_needed || avr->in_convert_needed) {
178  0, avr->internal_sample_fmt,
179  "in_buffer");
180  if (!avr->in_buffer) {
181  ret = AVERROR(EINVAL);
182  goto error;
183  }
184  }
185  if (avr->resample_needed) {
187  1024, avr->internal_sample_fmt,
188  "resample_out_buffer");
189  if (!avr->resample_out_buffer) {
190  ret = AVERROR(EINVAL);
191  goto error;
192  }
193  }
194  if (avr->out_convert_needed) {
196  avr->out_sample_fmt, "out_buffer");
197  if (!avr->out_buffer) {
198  ret = AVERROR(EINVAL);
199  goto error;
200  }
201  }
203  1024);
204  if (!avr->out_fifo) {
205  ret = AVERROR(ENOMEM);
206  goto error;
207  }
208 
209  /* setup contexts */
210  if (avr->in_convert_needed) {
212  avr->in_sample_fmt, avr->in_channels,
213  avr->in_sample_rate,
214  avr->remap_point == REMAP_IN_CONVERT);
215  if (!avr->ac_in) {
216  ret = AVERROR(ENOMEM);
217  goto error;
218  }
219  }
220  if (avr->out_convert_needed) {
221  enum AVSampleFormat src_fmt;
222  if (avr->in_convert_needed)
223  src_fmt = avr->internal_sample_fmt;
224  else
225  src_fmt = avr->in_sample_fmt;
226  avr->ac_out = ff_audio_convert_alloc(avr, avr->out_sample_fmt, src_fmt,
227  avr->out_channels,
228  avr->out_sample_rate,
230  if (!avr->ac_out) {
231  ret = AVERROR(ENOMEM);
232  goto error;
233  }
234  }
235  if (avr->resample_needed) {
236  avr->resample = ff_audio_resample_init(avr);
237  if (!avr->resample) {
238  ret = AVERROR(ENOMEM);
239  goto error;
240  }
241  }
242  if (avr->mixing_needed) {
243  avr->am = ff_audio_mix_alloc(avr);
244  if (!avr->am) {
245  ret = AVERROR(ENOMEM);
246  goto error;
247  }
248  }
249 
250  return 0;
251 
252 error:
253  avresample_close(avr);
254  return ret;
255 }
256 
258 {
259  return !!avr->out_fifo;
260 }
261 
263 {
268  avr->out_fifo = NULL;
272  ff_audio_mix_free(&avr->am);
273  av_freep(&avr->mix_matrix);
274 
275  avr->use_channel_map = 0;
276 }
277 
279 {
280  if (!*avr)
281  return;
282  avresample_close(*avr);
283  av_opt_free(*avr);
284  av_freep(avr);
285 }
286 
288  AudioData *output, AudioData *converted)
289 {
290  int ret;
291 
292  if (!output || av_audio_fifo_size(avr->out_fifo) > 0 ||
293  (converted && output->allocated_samples < converted->nb_samples)) {
294  if (converted) {
295  /* if there are any samples in the output FIFO or if the
296  user-supplied output buffer is not large enough for all samples,
297  we add to the output FIFO */
298  av_log(avr, AV_LOG_TRACE, "[FIFO] add %s to out_fifo\n", converted->name);
299  ret = ff_audio_data_add_to_fifo(avr->out_fifo, converted, 0,
300  converted->nb_samples);
301  if (ret < 0)
302  return ret;
303  }
304 
305  /* if the user specified an output buffer, read samples from the output
306  FIFO to the user output */
307  if (output && output->allocated_samples > 0) {
308  av_log(avr, AV_LOG_TRACE, "[FIFO] read from out_fifo to output\n");
309  av_log(avr, AV_LOG_TRACE, "[end conversion]\n");
310  return ff_audio_data_read_from_fifo(avr->out_fifo, output,
311  output->allocated_samples);
312  }
313  } else if (converted) {
314  /* copy directly to output if it is large enough or there is not any
315  data in the output FIFO */
316  av_log(avr, AV_LOG_TRACE, "[copy] %s to output\n", converted->name);
317  output->nb_samples = 0;
318  ret = ff_audio_data_copy(output, converted,
319  avr->remap_point == REMAP_OUT_COPY ?
320  &avr->ch_map_info : NULL);
321  if (ret < 0)
322  return ret;
323  av_log(avr, AV_LOG_TRACE, "[end conversion]\n");
324  return output->nb_samples;
325  }
326  av_log(avr, AV_LOG_TRACE, "[end conversion]\n");
327  return 0;
328 }
329 
330 int attribute_align_arg avresample_convert(AVAudioResampleContext *avr,
331  uint8_t **output, int out_plane_size,
332  int out_samples,
333  uint8_t * const *input,
334  int in_plane_size, int in_samples)
335 {
336  AudioData input_buffer;
337  AudioData output_buffer;
338  AudioData *current_buffer;
339  int ret, direct_output;
340 
341  /* reset internal buffers */
342  if (avr->in_buffer) {
343  avr->in_buffer->nb_samples = 0;
346  }
347  if (avr->resample_out_buffer) {
351  }
352  if (avr->out_buffer) {
353  avr->out_buffer->nb_samples = 0;
356  }
357 
358  av_log(avr, AV_LOG_TRACE, "[start conversion]\n");
359 
360  /* initialize output_buffer with output data */
361  direct_output = output && av_audio_fifo_size(avr->out_fifo) == 0;
362  if (output) {
363  ret = ff_audio_data_init(&output_buffer, output, out_plane_size,
364  avr->out_channels, out_samples,
365  avr->out_sample_fmt, 0, "output");
366  if (ret < 0)
367  return ret;
368  output_buffer.nb_samples = 0;
369  }
370 
371  if (input) {
372  /* initialize input_buffer with input data */
373  ret = ff_audio_data_init(&input_buffer, input, in_plane_size,
374  avr->in_channels, in_samples,
375  avr->in_sample_fmt, 1, "input");
376  if (ret < 0)
377  return ret;
378  current_buffer = &input_buffer;
379 
380  if (avr->upmix_needed && !avr->in_convert_needed && !avr->resample_needed &&
381  !avr->out_convert_needed && direct_output && out_samples >= in_samples) {
382  /* in some rare cases we can copy input to output and upmix
383  directly in the output buffer */
384  av_log(avr, AV_LOG_TRACE, "[copy] %s to output\n", current_buffer->name);
385  ret = ff_audio_data_copy(&output_buffer, current_buffer,
386  avr->remap_point == REMAP_OUT_COPY ?
387  &avr->ch_map_info : NULL);
388  if (ret < 0)
389  return ret;
390  current_buffer = &output_buffer;
391  } else if (avr->remap_point == REMAP_OUT_COPY &&
392  (!direct_output || out_samples < in_samples)) {
393  /* if remapping channels during output copy, we may need to
394  * use an intermediate buffer in order to remap before adding
395  * samples to the output fifo */
396  av_log(avr, AV_LOG_TRACE, "[copy] %s to out_buffer\n", current_buffer->name);
397  ret = ff_audio_data_copy(avr->out_buffer, current_buffer,
398  &avr->ch_map_info);
399  if (ret < 0)
400  return ret;
401  current_buffer = avr->out_buffer;
402  } else if (avr->in_copy_needed || avr->in_convert_needed) {
403  /* if needed, copy or convert input to in_buffer, and downmix if
404  applicable */
405  if (avr->in_convert_needed) {
406  ret = ff_audio_data_realloc(avr->in_buffer,
407  current_buffer->nb_samples);
408  if (ret < 0)
409  return ret;
410  av_log(avr, AV_LOG_TRACE, "[convert] %s to in_buffer\n", current_buffer->name);
411  ret = ff_audio_convert(avr->ac_in, avr->in_buffer,
412  current_buffer);
413  if (ret < 0)
414  return ret;
415  } else {
416  av_log(avr, AV_LOG_TRACE, "[copy] %s to in_buffer\n", current_buffer->name);
417  ret = ff_audio_data_copy(avr->in_buffer, current_buffer,
418  avr->remap_point == REMAP_IN_COPY ?
419  &avr->ch_map_info : NULL);
420  if (ret < 0)
421  return ret;
422  }
424  if (avr->downmix_needed) {
425  av_log(avr, AV_LOG_TRACE, "[downmix] in_buffer\n");
426  ret = ff_audio_mix(avr->am, avr->in_buffer);
427  if (ret < 0)
428  return ret;
429  }
430  current_buffer = avr->in_buffer;
431  }
432  } else {
433  /* flush resampling buffer and/or output FIFO if input is NULL */
434  if (!avr->resample_needed)
435  return handle_buffered_output(avr, output ? &output_buffer : NULL,
436  NULL);
437  current_buffer = NULL;
438  }
439 
440  if (avr->resample_needed) {
441  AudioData *resample_out;
442 
443  if (!avr->out_convert_needed && direct_output && out_samples > 0)
444  resample_out = &output_buffer;
445  else
446  resample_out = avr->resample_out_buffer;
447  av_log(avr, AV_LOG_TRACE, "[resample] %s to %s\n",
448  current_buffer ? current_buffer->name : "null",
449  resample_out->name);
450  ret = ff_audio_resample(avr->resample, resample_out,
451  current_buffer);
452  if (ret < 0)
453  return ret;
454 
455  /* if resampling did not produce any samples, just return 0 */
456  if (resample_out->nb_samples == 0) {
457  av_log(avr, AV_LOG_TRACE, "[end conversion]\n");
458  return 0;
459  }
460 
461  current_buffer = resample_out;
462  }
463 
464  if (avr->upmix_needed) {
465  av_log(avr, AV_LOG_TRACE, "[upmix] %s\n", current_buffer->name);
466  ret = ff_audio_mix(avr->am, current_buffer);
467  if (ret < 0)
468  return ret;
469  }
470 
471  /* if we resampled or upmixed directly to output, return here */
472  if (current_buffer == &output_buffer) {
473  av_log(avr, AV_LOG_TRACE, "[end conversion]\n");
474  return current_buffer->nb_samples;
475  }
476 
477  if (avr->out_convert_needed) {
478  if (direct_output && out_samples >= current_buffer->nb_samples) {
479  /* convert directly to output */
480  av_log(avr, AV_LOG_TRACE, "[convert] %s to output\n", current_buffer->name);
481  ret = ff_audio_convert(avr->ac_out, &output_buffer, current_buffer);
482  if (ret < 0)
483  return ret;
484 
485  av_log(avr, AV_LOG_TRACE, "[end conversion]\n");
486  return output_buffer.nb_samples;
487  } else {
489  current_buffer->nb_samples);
490  if (ret < 0)
491  return ret;
492  av_log(avr, AV_LOG_TRACE, "[convert] %s to out_buffer\n", current_buffer->name);
493  ret = ff_audio_convert(avr->ac_out, avr->out_buffer,
494  current_buffer);
495  if (ret < 0)
496  return ret;
497  current_buffer = avr->out_buffer;
498  }
499  }
500 
501  return handle_buffered_output(avr, output ? &output_buffer : NULL,
502  current_buffer);
503 }
504 
506 {
507  if (avresample_is_open(avr)) {
508  avresample_close(avr);
509  }
510 
511  if (in) {
513  avr->in_sample_rate = in->sample_rate;
514  avr->in_sample_fmt = in->format;
515  }
516 
517  if (out) {
519  avr->out_sample_rate = out->sample_rate;
520  avr->out_sample_fmt = out->format;
521  }
522 
523  return 0;
524 }
525 
527  AVFrame *out, AVFrame *in)
528 {
529  int ret = 0;
530 
531  if (in) {
532  if (avr->in_channel_layout != in->channel_layout ||
533  avr->in_sample_rate != in->sample_rate ||
534  avr->in_sample_fmt != in->format) {
535  ret |= AVERROR_INPUT_CHANGED;
536  }
537  }
538 
539  if (out) {
540  if (avr->out_channel_layout != out->channel_layout ||
541  avr->out_sample_rate != out->sample_rate ||
542  avr->out_sample_fmt != out->format) {
543  ret |= AVERROR_OUTPUT_CHANGED;
544  }
545  }
546 
547  return ret;
548 }
549 
550 static inline int convert_frame(AVAudioResampleContext *avr,
551  AVFrame *out, AVFrame *in)
552 {
553  int ret;
554  uint8_t **out_data = NULL, **in_data = NULL;
555  int out_linesize = 0, in_linesize = 0;
556  int out_nb_samples = 0, in_nb_samples = 0;
557 
558  if (out) {
559  out_data = out->extended_data;
560  out_linesize = out->linesize[0];
561  out_nb_samples = out->nb_samples;
562  }
563 
564  if (in) {
565  in_data = in->extended_data;
566  in_linesize = in->linesize[0];
567  in_nb_samples = in->nb_samples;
568  }
569 
570  ret = avresample_convert(avr, out_data, out_linesize,
571  out_nb_samples,
572  in_data, in_linesize,
573  in_nb_samples);
574 
575  if (ret < 0) {
576  if (out)
577  out->nb_samples = 0;
578  return ret;
579  }
580 
581  if (out)
582  out->nb_samples = ret;
583 
584  return 0;
585 }
586 
587 static inline int available_samples(AVFrame *out)
588 {
589  int samples;
590  int bytes_per_sample = av_get_bytes_per_sample(out->format);
591  if (!bytes_per_sample)
592  return AVERROR(EINVAL);
593 
594  samples = out->linesize[0] / bytes_per_sample;
595  if (av_sample_fmt_is_planar(out->format)) {
596  return samples;
597  } else {
599  return samples / channels;
600  }
601 }
602 
604  AVFrame *out, AVFrame *in)
605 {
606  int ret, setup = 0;
607 
608  if (!avresample_is_open(avr)) {
609  if ((ret = avresample_config(avr, out, in)) < 0)
610  return ret;
611  if ((ret = avresample_open(avr)) < 0)
612  return ret;
613  setup = 1;
614  } else {
615  // return as is or reconfigure for input changes?
616  if ((ret = config_changed(avr, out, in)))
617  return ret;
618  }
619 
620  if (out) {
621  if (!out->linesize[0]) {
623  if ((ret = av_frame_get_buffer(out, 0)) < 0) {
624  if (setup)
625  avresample_close(avr);
626  return ret;
627  }
628  } else {
629  if (!out->nb_samples)
630  out->nb_samples = available_samples(out);
631  }
632  }
633 
634  return convert_frame(avr, out, in);
635 }
636 
638  int stride)
639 {
640  int in_channels, out_channels, i, o;
641 
642  if (avr->am)
643  return ff_audio_mix_get_matrix(avr->am, matrix, stride);
644 
647 
648  if ( in_channels <= 0 || in_channels > AVRESAMPLE_MAX_CHANNELS ||
649  out_channels <= 0 || out_channels > AVRESAMPLE_MAX_CHANNELS) {
650  av_log(avr, AV_LOG_ERROR, "Invalid channel layouts\n");
651  return AVERROR(EINVAL);
652  }
653 
654  if (!avr->mix_matrix) {
655  av_log(avr, AV_LOG_ERROR, "matrix is not set\n");
656  return AVERROR(EINVAL);
657  }
658 
659  for (o = 0; o < out_channels; o++)
660  for (i = 0; i < in_channels; i++)
661  matrix[o * stride + i] = avr->mix_matrix[o * in_channels + i];
662 
663  return 0;
664 }
665 
666 int avresample_set_matrix(AVAudioResampleContext *avr, const double *matrix,
667  int stride)
668 {
669  int in_channels, out_channels, i, o;
670 
671  if (avr->am)
672  return ff_audio_mix_set_matrix(avr->am, matrix, stride);
673 
676 
677  if ( in_channels <= 0 || in_channels > AVRESAMPLE_MAX_CHANNELS ||
678  out_channels <= 0 || out_channels > AVRESAMPLE_MAX_CHANNELS) {
679  av_log(avr, AV_LOG_ERROR, "Invalid channel layouts\n");
680  return AVERROR(EINVAL);
681  }
682 
683  if (avr->mix_matrix)
684  av_freep(&avr->mix_matrix);
685  avr->mix_matrix = av_malloc(in_channels * out_channels *
686  sizeof(*avr->mix_matrix));
687  if (!avr->mix_matrix)
688  return AVERROR(ENOMEM);
689 
690  for (o = 0; o < out_channels; o++)
691  for (i = 0; i < in_channels; i++)
692  avr->mix_matrix[o * in_channels + i] = matrix[o * stride + i];
693 
694  return 0;
695 }
696 
698  const int *channel_map)
699 {
700  ChannelMapInfo *info = &avr->ch_map_info;
701  int in_channels, ch, i;
702 
704  if (in_channels <= 0 || in_channels > AVRESAMPLE_MAX_CHANNELS) {
705  av_log(avr, AV_LOG_ERROR, "Invalid input channel layout\n");
706  return AVERROR(EINVAL);
707  }
708 
709  memset(info, 0, sizeof(*info));
710  memset(info->input_map, -1, sizeof(info->input_map));
711 
712  for (ch = 0; ch < in_channels; ch++) {
713  if (channel_map[ch] >= in_channels) {
714  av_log(avr, AV_LOG_ERROR, "Invalid channel map\n");
715  return AVERROR(EINVAL);
716  }
717  if (channel_map[ch] < 0) {
718  info->channel_zero[ch] = 1;
719  info->channel_map[ch] = -1;
720  info->do_zero = 1;
721  } else if (info->input_map[channel_map[ch]] >= 0) {
722  info->channel_copy[ch] = info->input_map[channel_map[ch]];
723  info->channel_map[ch] = -1;
724  info->do_copy = 1;
725  } else {
726  info->channel_map[ch] = channel_map[ch];
727  info->input_map[channel_map[ch]] = ch;
728  info->do_remap = 1;
729  }
730  }
731  /* Fill-in unmapped input channels with unmapped output channels.
732  This is used when remapping during conversion from interleaved to
733  planar format. */
734  for (ch = 0, i = 0; ch < in_channels && i < in_channels; ch++, i++) {
735  while (ch < in_channels && info->input_map[ch] >= 0)
736  ch++;
737  while (i < in_channels && info->channel_map[i] >= 0)
738  i++;
739  if (ch >= in_channels || i >= in_channels)
740  break;
741  info->input_map[ch] = i;
742  }
743 
744  avr->use_channel_map = 1;
745  return 0;
746 }
747 
749 {
750  return av_audio_fifo_size(avr->out_fifo);
751 }
752 
754 {
755  int64_t samples = avresample_get_delay(avr) + (int64_t)in_nb_samples;
756 
757  if (avr->resample_needed) {
758  samples = av_rescale_rnd(samples,
759  avr->out_sample_rate,
760  avr->in_sample_rate,
761  AV_ROUND_UP);
762  }
763 
764  samples += avresample_available(avr);
765 
766  if (samples > INT_MAX)
767  return AVERROR(EINVAL);
768 
769  return samples;
770 }
771 
772 int avresample_read(AVAudioResampleContext *avr, uint8_t **output, int nb_samples)
773 {
774  if (!output)
775  return av_audio_fifo_drain(avr->out_fifo, nb_samples);
776  return av_audio_fifo_read(avr->out_fifo, (void**)output, nb_samples);
777 }
778 
779 unsigned avresample_version(void)
780 {
782 }
783 
784 const char *avresample_license(void)
785 {
786 #define LICENSE_PREFIX "libavresample license: "
787  return LICENSE_PREFIX FFMPEG_LICENSE + sizeof(LICENSE_PREFIX) - 1;
788 }
789 
790 const char *avresample_configuration(void)
791 {
792  return FFMPEG_CONFIGURATION;
793 }
float, planar
Definition: samplefmt.h:69
int in_channels
number of input channels
Definition: internal.h:77
#define NULL
Definition: coverity.c:32
AudioConvert * ac_in
input sample format conversion context
Definition: internal.h:93
const char * name
name for debug logging
Definition: audio_data.h:55
AVAudioFifo * av_audio_fifo_alloc(enum AVSampleFormat sample_fmt, int channels, int nb_samples)
Allocate an AVAudioFifo.
Definition: audio_fifo.c:60
int av_audio_fifo_read(AVAudioFifo *af, void **data, int nb_samples)
Read data from an AVAudioFifo.
Definition: audio_fifo.c:182
int ff_audio_data_realloc(AudioData *a, int nb_samples)
Reallocate AudioData.
Definition: audio_data.c:162
This structure describes decoded (raw) audio or video data.
Definition: frame.h:184
#define LIBAVRESAMPLE_VERSION_INT
Definition: version.h:34
int input_map[AVRESAMPLE_MAX_CHANNELS]
dest index of each input channel
Definition: internal.h:50
AudioData * out_buffer
buffer for converted output
Definition: internal.h:90
Audio buffer used for intermediate storage between conversion phases.
Definition: audio_data.h:37
static int config_changed(AVAudioResampleContext *avr, AVFrame *out, AVFrame *in)
Definition: utils.c:526
int avresample_convert_frame(AVAudioResampleContext *avr, AVFrame *out, AVFrame *in)
Convert the samples in the input AVFrame and write them to the output AVFrame.
Definition: utils.c:603
int attribute_align_arg avresample_convert(AVAudioResampleContext *avr, uint8_t **output, int out_plane_size, int out_samples, uint8_t *const *input, int in_plane_size, int in_samples)
Convert input samples and write them to the output FIFO.
Definition: utils.c:330
Memory handling functions.
int ff_audio_data_add_to_fifo(AVAudioFifo *af, AudioData *a, int offset, int nb_samples)
Add samples in AudioData to an AVAudioFifo.
Definition: audio_data.c:351
int do_zero
zeroing needed
Definition: internal.h:49
AudioData * ff_audio_data_alloc(int channels, int nb_samples, enum AVSampleFormat sample_fmt, const char *name)
Allocate AudioData.
Definition: audio_data.c:119
static int convert_frame(AVAudioResampleContext *avr, AVFrame *out, AVFrame *in)
Definition: utils.c:550
int avresample_read(AVAudioResampleContext *avr, uint8_t **output, int nb_samples)
Read samples from the output FIFO.
Definition: utils.c:772
double * mix_matrix
mix matrix only used if avresample_set_matrix() is called before avresample_open() ...
Definition: internal.h:103
uint64_t out_channel_layout
output channel layout
Definition: internal.h:59
void av_audio_fifo_free(AVAudioFifo *af)
Free an AVAudioFifo.
Definition: audio_fifo.c:45
int ff_audio_convert(AudioConvert *ac, AudioData *out, AudioData *in)
Convert audio data from one sample format to another.
double, planar
Definition: samplefmt.h:70
int avresample_set_channel_mapping(AVAudioResampleContext *avr, const int *channel_map)
Set a customized input channel mapping.
Definition: utils.c:697
int ff_audio_mix(AudioMix *am, AudioData *src)
Apply channel mixing to audio data using the current mixing matrix.
Definition: audio_mix.c:428
int channel_zero[AVRESAMPLE_MAX_CHANNELS]
dest index to zero
Definition: internal.h:48
void avresample_free(AVAudioResampleContext **avr)
Free AVAudioResampleContext and associated AVOption values.
Definition: utils.c:278
int nb_samples
current number of samples
Definition: audio_data.h:43
int av_get_channel_layout_nb_channels(uint64_t channel_layout)
Return the number of channels in the channel layout.
AudioData * in_buffer
buffer for converted input
Definition: internal.h:88
Public dictionary API.
int allocated_channels
allocated channel count
Definition: audio_data.h:46
uint8_t
Round toward +infinity.
Definition: mathematics.h:83
#define av_malloc(s)
AVOptions.
int out_channels
number of output channels
Definition: internal.h:78
#define AV_LOG_TRACE
Extremely verbose debugging, useful for libav* development.
Definition: log.h:202
static int handle_buffered_output(AVAudioResampleContext *avr, AudioData *output, AudioData *converted)
Definition: utils.c:287
#define LICENSE_PREFIX
int ff_audio_mix_set_matrix(AudioMix *am, const double *matrix, int stride)
Set the current mixing matrix.
Definition: audio_mix.c:653
int ff_sample_fmt_is_planar(enum AVSampleFormat sample_fmt, int channels)
Definition: audio_data.c:51
const char * avresample_license(void)
Return the libavresample license.
Definition: utils.c:784
enum AVSampleFormat av_get_planar_sample_fmt(enum AVSampleFormat sample_fmt)
Get the planar alternative form of the given sample format.
Definition: samplefmt.c:84
AudioConvert * ff_audio_convert_alloc(AVAudioResampleContext *avr, enum AVSampleFormat out_fmt, enum AVSampleFormat in_fmt, int channels, int sample_rate, int apply_map)
Allocate and initialize AudioConvert context for sample format conversion.
#define av_log(a,...)
void avresample_close(AVAudioResampleContext *avr)
Close AVAudioResampleContext.
Definition: utils.c:262
AudioMix * am
channel mixing context
Definition: internal.h:96
int av_sample_fmt_is_planar(enum AVSampleFormat sample_fmt)
Check if the sample format is planar.
Definition: samplefmt.c:112
int ff_audio_data_set_channels(AudioData *a, int channels)
Definition: audio_data.c:59
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
int out_convert_needed
output sample format conversion is needed
Definition: internal.h:85
#define AVERROR(e)
Definition: error.h:43
int ff_audio_data_init(AudioData *a, uint8_t *const *src, int plane_size, int channels, int nb_samples, enum AVSampleFormat sample_fmt, int read_only, const char *name)
Initialize AudioData using a given source.
Definition: audio_data.c:73
#define AV_LOG_DEBUG
Stuff which is only useful for libav* developers.
Definition: log.h:197
void ff_audio_convert_free(AudioConvert **ac)
Free AudioConvert.
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
AudioConvert * ac_out
output sample format conversion context
Definition: internal.h:94
#define FFMAX(a, b)
Definition: common.h:94
int ff_audio_data_read_from_fifo(AVAudioFifo *af, AudioData *a, int nb_samples)
Read samples from an AVAudioFifo to AudioData.
Definition: audio_data.c:366
reference-counted frame API
int channel_copy[AVRESAMPLE_MAX_CHANNELS]
dest index to copy from
Definition: internal.h:46
uint64_t channel_layout
Channel layout of the audio data.
Definition: frame.h:349
int avresample_config(AVAudioResampleContext *avr, AVFrame *out, AVFrame *in)
Configure or reconfigure the AVAudioResampleContext using the information provided by the AVFrames...
Definition: utils.c:505
int upmix_needed
upmixing is needed
Definition: internal.h:81
ResampleContext * resample
resampling context
Definition: internal.h:95
int av_audio_fifo_size(AVAudioFifo *af)
Get the current number of samples in the AVAudioFifo available for reading.
Definition: audio_fifo.c:229
enum RemapPoint remap_point
Definition: internal.h:106
external API header
#define FFMIN(a, b)
Definition: common.h:96
int do_remap
remap needed
Definition: internal.h:45
signed 32 bits, planar
Definition: samplefmt.h:68
#define AVERROR_INPUT_CHANGED
Input changed between calls. Reconfiguration is required. (can be OR-ed with AVERROR_OUTPUT_CHANGED) ...
Definition: error.h:73
ChannelMapInfo ch_map_info
Definition: internal.h:107
uint64_t in_channel_layout
input channel layout
Definition: internal.h:56
int64_t av_rescale_rnd(int64_t a, int64_t b, int64_t c, enum AVRounding rnd)
Rescale a 64-bit integer with specified rounding.
Definition: mathematics.c:58
int avresample_set_matrix(AVAudioResampleContext *avr, const double *matrix, int stride)
Set channel mixing matrix.
Definition: utils.c:666
int in_sample_rate
input sample rate
Definition: internal.h:58
int format
format of the frame, -1 if unknown or unset Values correspond to enum AVPixelFormat for video frames...
Definition: frame.h:248
int avresample_get_delay(AVAudioResampleContext *avr)
Return the number of samples currently in the resampling delay buffer.
Definition: resample.c:438
int avresample_get_matrix(AVAudioResampleContext *avr, double *matrix, int stride)
Get the current channel mixing matrix.
Definition: utils.c:637
int avresample_available(AVAudioResampleContext *avr)
Return the number of available samples in the output FIFO.
Definition: utils.c:748
void ff_audio_resample_free(ResampleContext **c)
Free a ResampleContext.
Definition: resample.c:224
AVSampleFormat
Audio sample formats.
Definition: samplefmt.h:58
AVAudioFifo * out_fifo
FIFO for output samples.
Definition: internal.h:91
int linesize[AV_NUM_DATA_POINTERS]
For video, size in bytes of each picture line.
Definition: frame.h:215
#define AVRESAMPLE_MAX_CHANNELS
Definition: avresample.h:104
int avresample_get_out_samples(AVAudioResampleContext *avr, int in_nb_samples)
Provide the upper bound on the number of samples the configured conversion would output.
Definition: utils.c:753
enum AVSampleFormat internal_sample_fmt
internal sample format
Definition: internal.h:62
uint8_t pi<< 24) CONV_FUNC_GROUP(AV_SAMPLE_FMT_FLT, float, AV_SAMPLE_FMT_U8, uint8_t,(*(constuint8_t *) pi-0x80)*(1.0f/(1<< 7))) CONV_FUNC_GROUP(AV_SAMPLE_FMT_DBL, double, AV_SAMPLE_FMT_U8, uint8_t,(*(constuint8_t *) pi-0x80)*(1.0/(1<< 7))) CONV_FUNC_GROUP(AV_SAMPLE_FMT_U8, uint8_t, AV_SAMPLE_FMT_S16, int16_t,(*(constint16_t *) pi >>8)+0x80) CONV_FUNC_GROUP(AV_SAMPLE_FMT_FLT, float, AV_SAMPLE_FMT_S16, int16_t,*(constint16_t *) pi *(1.0f/(1<< 15))) CONV_FUNC_GROUP(AV_SAMPLE_FMT_DBL, double, AV_SAMPLE_FMT_S16, int16_t,*(constint16_t *) pi *(1.0/(1<< 15))) CONV_FUNC_GROUP(AV_SAMPLE_FMT_U8, uint8_t, AV_SAMPLE_FMT_S32, int32_t,(*(constint32_t *) pi >>24)+0x80) CONV_FUNC_GROUP(AV_SAMPLE_FMT_FLT, float, AV_SAMPLE_FMT_S32, int32_t,*(constint32_t *) pi *(1.0f/(1U<< 31))) CONV_FUNC_GROUP(AV_SAMPLE_FMT_DBL, double, AV_SAMPLE_FMT_S32, int32_t,*(constint32_t *) pi *(1.0/(1U<< 31))) CONV_FUNC_GROUP(AV_SAMPLE_FMT_U8, uint8_t, AV_SAMPLE_FMT_FLT, float, av_clip_uint8(lrintf(*(constfloat *) pi *(1<< 7))+0x80)) CONV_FUNC_GROUP(AV_SAMPLE_FMT_S16, int16_t, AV_SAMPLE_FMT_FLT, float, av_clip_int16(lrintf(*(constfloat *) pi *(1<< 15)))) CONV_FUNC_GROUP(AV_SAMPLE_FMT_S32, int32_t, AV_SAMPLE_FMT_FLT, float, av_clipl_int32(llrintf(*(constfloat *) pi *(1U<< 31)))) CONV_FUNC_GROUP(AV_SAMPLE_FMT_U8, uint8_t, AV_SAMPLE_FMT_DBL, double, av_clip_uint8(lrint(*(constdouble *) pi *(1<< 7))+0x80)) CONV_FUNC_GROUP(AV_SAMPLE_FMT_S16, int16_t, AV_SAMPLE_FMT_DBL, double, av_clip_int16(lrint(*(constdouble *) pi *(1<< 15)))) CONV_FUNC_GROUP(AV_SAMPLE_FMT_S32, int32_t, AV_SAMPLE_FMT_DBL, double, av_clipl_int32(llrint(*(constdouble *) pi *(1U<< 31))))#defineSET_CONV_FUNC_GROUP(ofmt, ifmt) staticvoidset_generic_function(AudioConvert *ac){}voidff_audio_convert_free(AudioConvert **ac){if(!*ac) return;ff_dither_free(&(*ac) ->dc);av_freep(ac);}AudioConvert *ff_audio_convert_alloc(AVAudioResampleContext *avr, enumAVSampleFormatout_fmt, enumAVSampleFormatin_fmt, intchannels, intsample_rate, intapply_map){AudioConvert *ac;intin_planar, out_planar;ac=av_mallocz(sizeof(*ac));if(!ac) returnNULL;ac->avr=avr;ac->out_fmt=out_fmt;ac->in_fmt=in_fmt;ac->channels=channels;ac->apply_map=apply_map;if(avr->dither_method!=AV_RESAMPLE_DITHER_NONE &&av_get_packed_sample_fmt(out_fmt)==AV_SAMPLE_FMT_S16 &&av_get_bytes_per_sample(in_fmt)>2){ac->dc=ff_dither_alloc(avr, out_fmt, in_fmt, channels, sample_rate, apply_map);if(!ac->dc){av_free(ac);returnNULL;}returnac;}in_planar=ff_sample_fmt_is_planar(in_fmt, channels);out_planar=ff_sample_fmt_is_planar(out_fmt, channels);if(in_planar==out_planar){ac->func_type=CONV_FUNC_TYPE_FLAT;ac->planes=in_planar?ac->channels:1;}elseif(in_planar) ac->func_type=CONV_FUNC_TYPE_INTERLEAVE;elseac->func_type=CONV_FUNC_TYPE_DEINTERLEAVE;set_generic_function(ac);if(ARCH_AARCH64) ff_audio_convert_init_aarch64(ac);if(ARCH_ARM) ff_audio_convert_init_arm(ac);if(ARCH_X86) ff_audio_convert_init_x86(ac);returnac;}intff_audio_convert(AudioConvert *ac, AudioData *out, AudioData *in){intuse_generic=1;intlen=in->nb_samples;intp;if(ac->dc){av_log(ac->avr, AV_LOG_TRACE,"%dsamples-audio_convert:%sto%s(dithered)\n", len, av_get_sample_fmt_name(ac->in_fmt), av_get_sample_fmt_name(ac->out_fmt));returnff_convert_dither(ac-> in
int force_resampling
force resampling
Definition: internal.h:68
void ff_audio_mix_free(AudioMix **am_p)
Free an AudioMix context.
Definition: audio_mix.c:409
int in_copy_needed
input data copy is needed
Definition: internal.h:86
const char * avresample_configuration(void)
Return the libavresample build-time configuration.
Definition: utils.c:790
int sample_rate
Sample rate of the audio data.
Definition: frame.h:344
ResampleContext * ff_audio_resample_init(AVAudioResampleContext *avr)
Allocate and initialize a ResampleContext.
Definition: resample.c:120
enum AVSampleFormat in_sample_fmt
input sample format
Definition: internal.h:57
int in_convert_needed
input sample format conversion is needed
Definition: internal.h:84
int av_frame_get_buffer(AVFrame *frame, int align)
Allocate new buffer(s) for audio or video data.
Definition: frame.c:271
int channel_map[AVRESAMPLE_MAX_CHANNELS]
source index of each output channel, -1 if not remapped
Definition: internal.h:44
int av_get_bytes_per_sample(enum AVSampleFormat sample_fmt)
Return number of bytes per sample.
Definition: samplefmt.c:106
#define AVERROR_OUTPUT_CHANGED
Output changed between calls. Reconfiguration is required. (can be OR-ed with AVERROR_INPUT_CHANGED) ...
Definition: error.h:74
enum AVSampleFormat out_sample_fmt
output sample format
Definition: internal.h:60
int av_audio_fifo_drain(AVAudioFifo *af, int nb_samples)
Drain data from an AVAudioFifo.
Definition: audio_fifo.c:202
int ff_audio_data_copy(AudioData *dst, AudioData *src, ChannelMapInfo *map)
Copy data from one AudioData to another.
Definition: audio_data.c:225
GLint GLenum GLboolean GLsizei stride
Definition: opengl_enc.c:105
void av_opt_free(void *obj)
Free all allocated objects in obj.
Definition: opt.c:1516
common internal and external API header
int resample_channels
number of channels used for resampling
Definition: internal.h:79
AudioData * resample_out_buffer
buffer for output from resampler
Definition: internal.h:89
int resample_needed
resampling is needed
Definition: internal.h:83
int do_copy
copy needed
Definition: internal.h:47
AudioMix * ff_audio_mix_alloc(AVAudioResampleContext *avr)
Allocate and initialize an AudioMix context.
Definition: audio_mix.c:341
int avresample_is_open(AVAudioResampleContext *avr)
Check whether an AVAudioResampleContext is open or closed.
Definition: utils.c:257
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 avresample_version(void)
Return the LIBAVRESAMPLE_VERSION_INT constant.
Definition: utils.c:779
int allocated_samples
number of samples the buffer can hold
Definition: audio_data.h:42
FILE * out
Definition: movenc.c:54
#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
int downmix_needed
downmixing is needed
Definition: internal.h:80
static int available_samples(AVFrame *out)
Definition: utils.c:587
uint8_t ** extended_data
pointers to the data planes/channels.
Definition: frame.h:231
int avresample_open(AVAudioResampleContext *avr)
Initialize AVAudioResampleContext.
Definition: utils.c:36
int nb_samples
number of audio samples (per channel) described by this frame
Definition: frame.h:241
int mixing_needed
either upmixing or downmixing is needed
Definition: internal.h:82
int ff_audio_mix_get_matrix(AudioMix *am, double *matrix, int stride)
Get the current mixing matrix.
Definition: audio_mix.c:483