FFmpeg  4.0
avresample.h
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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 #ifndef AVRESAMPLE_AVRESAMPLE_H
22 #define AVRESAMPLE_AVRESAMPLE_H
23 
24 /**
25  * @file
26  * @ingroup lavr
27  * external API header
28  */
29 
30 /**
31  * @defgroup lavr libavresample
32  * @{
33  *
34  * Libavresample (lavr) is a library that handles audio resampling, sample
35  * format conversion and mixing.
36  *
37  * Interaction with lavr is done through AVAudioResampleContext, which is
38  * allocated with avresample_alloc_context(). It is opaque, so all parameters
39  * must be set with the @ref avoptions API.
40  *
41  * For example the following code will setup conversion from planar float sample
42  * format to interleaved signed 16-bit integer, downsampling from 48kHz to
43  * 44.1kHz and downmixing from 5.1 channels to stereo (using the default mixing
44  * matrix):
45  * @code
46  * AVAudioResampleContext *avr = avresample_alloc_context();
47  * av_opt_set_int(avr, "in_channel_layout", AV_CH_LAYOUT_5POINT1, 0);
48  * av_opt_set_int(avr, "out_channel_layout", AV_CH_LAYOUT_STEREO, 0);
49  * av_opt_set_int(avr, "in_sample_rate", 48000, 0);
50  * av_opt_set_int(avr, "out_sample_rate", 44100, 0);
51  * av_opt_set_int(avr, "in_sample_fmt", AV_SAMPLE_FMT_FLTP, 0);
52  * av_opt_set_int(avr, "out_sample_fmt", AV_SAMPLE_FMT_S16, 0);
53  * @endcode
54  *
55  * Once the context is initialized, it must be opened with avresample_open(). If
56  * you need to change the conversion parameters, you must close the context with
57  * avresample_close(), change the parameters as described above, then reopen it
58  * again.
59  *
60  * The conversion itself is done by repeatedly calling avresample_convert().
61  * Note that the samples may get buffered in two places in lavr. The first one
62  * is the output FIFO, where the samples end up if the output buffer is not
63  * large enough. The data stored in there may be retrieved at any time with
64  * avresample_read(). The second place is the resampling delay buffer,
65  * applicable only when resampling is done. The samples in it require more input
66  * before they can be processed. Their current amount is returned by
67  * avresample_get_delay(). At the end of conversion the resampling buffer can be
68  * flushed by calling avresample_convert() with NULL input.
69  *
70  * The following code demonstrates the conversion loop assuming the parameters
71  * from above and caller-defined functions get_input() and handle_output():
72  * @code
73  * uint8_t **input;
74  * int in_linesize, in_samples;
75  *
76  * while (get_input(&input, &in_linesize, &in_samples)) {
77  * uint8_t *output
78  * int out_linesize;
79  * int out_samples = avresample_get_out_samples(avr, in_samples);
80  *
81  * av_samples_alloc(&output, &out_linesize, 2, out_samples,
82  * AV_SAMPLE_FMT_S16, 0);
83  * out_samples = avresample_convert(avr, &output, out_linesize, out_samples,
84  * input, in_linesize, in_samples);
85  * handle_output(output, out_linesize, out_samples);
86  * av_freep(&output);
87  * }
88  * @endcode
89  *
90  * When the conversion is finished and the FIFOs are flushed if required, the
91  * conversion context and everything associated with it must be freed with
92  * avresample_free().
93  */
94 
95 #include "libavutil/avutil.h"
97 #include "libavutil/dict.h"
98 #include "libavutil/frame.h"
99 #include "libavutil/log.h"
100 #include "libavutil/mathematics.h"
101 
102 #include "libavresample/version.h"
103 
104 #define AVRESAMPLE_MAX_CHANNELS 32
105 
107 
108 /**
109  * @deprecated use libswresample
110  *
111  * Mixing Coefficient Types */
113  AV_MIX_COEFF_TYPE_Q8, /** 16-bit 8.8 fixed-point */
114  AV_MIX_COEFF_TYPE_Q15, /** 32-bit 17.15 fixed-point */
115  AV_MIX_COEFF_TYPE_FLT, /** floating-point */
116  AV_MIX_COEFF_TYPE_NB, /** Number of coeff types. Not part of ABI */
117 };
118 
119 /**
120  * @deprecated use libswresample
121  *
122  * Resampling Filter Types */
125  AV_RESAMPLE_FILTER_TYPE_BLACKMAN_NUTTALL, /**< Blackman Nuttall Windowed Sinc */
126  AV_RESAMPLE_FILTER_TYPE_KAISER, /**< Kaiser Windowed Sinc */
127 };
128 
129 /**
130  * @deprecated use libswresample
131  */
133  AV_RESAMPLE_DITHER_NONE, /**< Do not use dithering */
134  AV_RESAMPLE_DITHER_RECTANGULAR, /**< Rectangular Dither */
135  AV_RESAMPLE_DITHER_TRIANGULAR, /**< Triangular Dither*/
136  AV_RESAMPLE_DITHER_TRIANGULAR_HP, /**< Triangular Dither with High Pass */
137  AV_RESAMPLE_DITHER_TRIANGULAR_NS, /**< Triangular Dither with Noise Shaping */
138  AV_RESAMPLE_DITHER_NB, /**< Number of dither types. Not part of ABI. */
139 };
140 
141 /**
142  *
143  * @deprecated use libswresample
144  *
145  * Return the LIBAVRESAMPLE_VERSION_INT constant.
146  */
148 unsigned avresample_version(void);
149 
150 /**
151  *
152  * @deprecated use libswresample
153  *
154  * Return the libavresample build-time configuration.
155  * @return configure string
156  */
158 const char *avresample_configuration(void);
159 
160 /**
161  *
162  * @deprecated use libswresample
163  *
164  * Return the libavresample license.
165  */
167 const char *avresample_license(void);
168 
169 /**
170  *
171  * @deprecated use libswresample
172  *
173  * Get the AVClass for AVAudioResampleContext.
174  *
175  * Can be used in combination with AV_OPT_SEARCH_FAKE_OBJ for examining options
176  * without allocating a context.
177  *
178  * @see av_opt_find().
179  *
180  * @return AVClass for AVAudioResampleContext
181  */
183 const AVClass *avresample_get_class(void);
184 
185 /**
186  *
187  * @deprecated use libswresample
188  *
189  * Allocate AVAudioResampleContext and set options.
190  *
191  * @return allocated audio resample context, or NULL on failure
192  */
195 
196 /**
197  *
198  * @deprecated use libswresample
199  *
200  * Initialize AVAudioResampleContext.
201  * @note The context must be configured using the AVOption API.
202  * @note The fields "in_channel_layout", "out_channel_layout",
203  * "in_sample_rate", "out_sample_rate", "in_sample_fmt",
204  * "out_sample_fmt" must be set.
205  *
206  * @see av_opt_set_int()
207  * @see av_opt_set_dict()
208  * @see av_get_default_channel_layout()
209  *
210  * @param avr audio resample context
211  * @return 0 on success, negative AVERROR code on failure
212  */
215 
216 /**
217  *
218  * @deprecated use libswresample
219  *
220  * Check whether an AVAudioResampleContext is open or closed.
221  *
222  * @param avr AVAudioResampleContext to check
223  * @return 1 if avr is open, 0 if avr is closed.
224  */
227 
228 /**
229  *
230  * @deprecated use libswresample
231  *
232  * Close AVAudioResampleContext.
233  *
234  * This closes the context, but it does not change the parameters. The context
235  * can be reopened with avresample_open(). It does, however, clear the output
236  * FIFO and any remaining leftover samples in the resampling delay buffer. If
237  * there was a custom matrix being used, that is also cleared.
238  *
239  * @see avresample_convert()
240  * @see avresample_set_matrix()
241  *
242  * @param avr audio resample context
243  */
246 
247 /**
248  *
249  * @deprecated use libswresample
250  *
251  * Free AVAudioResampleContext and associated AVOption values.
252  *
253  * This also calls avresample_close() before freeing.
254  *
255  * @param avr audio resample context
256  */
259 
260 /**
261  *
262  * @deprecated use libswresample
263  *
264  * Generate a channel mixing matrix.
265  *
266  * This function is the one used internally by libavresample for building the
267  * default mixing matrix. It is made public just as a utility function for
268  * building custom matrices.
269  *
270  * @param in_layout input channel layout
271  * @param out_layout output channel layout
272  * @param center_mix_level mix level for the center channel
273  * @param surround_mix_level mix level for the surround channel(s)
274  * @param lfe_mix_level mix level for the low-frequency effects channel
275  * @param normalize if 1, coefficients will be normalized to prevent
276  * overflow. if 0, coefficients will not be
277  * normalized.
278  * @param[out] matrix mixing coefficients; matrix[i + stride * o] is
279  * the weight of input channel i in output channel o.
280  * @param stride distance between adjacent input channels in the
281  * matrix array
282  * @param matrix_encoding matrixed stereo downmix mode (e.g. dplii)
283  * @return 0 on success, negative AVERROR code on failure
284  */
286 int avresample_build_matrix(uint64_t in_layout, uint64_t out_layout,
287  double center_mix_level, double surround_mix_level,
288  double lfe_mix_level, int normalize, double *matrix,
290 
291 /**
292  *
293  * @deprecated use libswresample
294  *
295  * Get the current channel mixing matrix.
296  *
297  * If no custom matrix has been previously set or the AVAudioResampleContext is
298  * not open, an error is returned.
299  *
300  * @param avr audio resample context
301  * @param matrix mixing coefficients; matrix[i + stride * o] is the weight of
302  * input channel i in output channel o.
303  * @param stride distance between adjacent input channels in the matrix array
304  * @return 0 on success, negative AVERROR code on failure
305  */
307 int avresample_get_matrix(AVAudioResampleContext *avr, double *matrix,
308  int stride);
309 
310 /**
311  *
312  * @deprecated use libswresample
313  *
314  * Set channel mixing matrix.
315  *
316  * Allows for setting a custom mixing matrix, overriding the default matrix
317  * generated internally during avresample_open(). This function can be called
318  * anytime on an allocated context, either before or after calling
319  * avresample_open(), as long as the channel layouts have been set.
320  * avresample_convert() always uses the current matrix.
321  * Calling avresample_close() on the context will clear the current matrix.
322  *
323  * @see avresample_close()
324  *
325  * @param avr audio resample context
326  * @param matrix mixing coefficients; matrix[i + stride * o] is the weight of
327  * input channel i in output channel o.
328  * @param stride distance between adjacent input channels in the matrix array
329  * @return 0 on success, negative AVERROR code on failure
330  */
332 int avresample_set_matrix(AVAudioResampleContext *avr, const double *matrix,
333  int stride);
334 
335 /**
336  *
337  * @deprecated use libswresample
338  *
339  * Set a customized input channel mapping.
340  *
341  * This function can only be called when the allocated context is not open.
342  * Also, the input channel layout must have already been set.
343  *
344  * Calling avresample_close() on the context will clear the channel mapping.
345  *
346  * The map for each input channel specifies the channel index in the source to
347  * use for that particular channel, or -1 to mute the channel. Source channels
348  * can be duplicated by using the same index for multiple input channels.
349  *
350  * Examples:
351  *
352  * Reordering 5.1 AAC order (C,L,R,Ls,Rs,LFE) to FFmpeg order (L,R,C,LFE,Ls,Rs):
353  * { 1, 2, 0, 5, 3, 4 }
354  *
355  * Muting the 3rd channel in 4-channel input:
356  * { 0, 1, -1, 3 }
357  *
358  * Duplicating the left channel of stereo input:
359  * { 0, 0 }
360  *
361  * @param avr audio resample context
362  * @param channel_map customized input channel mapping
363  * @return 0 on success, negative AVERROR code on failure
364  */
367  const int *channel_map);
368 
369 /**
370  *
371  * @deprecated use libswresample
372  *
373  * Set compensation for resampling.
374  *
375  * This can be called anytime after avresample_open(). If resampling is not
376  * automatically enabled because of a sample rate conversion, the
377  * "force_resampling" option must have been set to 1 when opening the context
378  * in order to use resampling compensation.
379  *
380  * @param avr audio resample context
381  * @param sample_delta compensation delta, in samples
382  * @param compensation_distance compensation distance, in samples
383  * @return 0 on success, negative AVERROR code on failure
384  */
386 int avresample_set_compensation(AVAudioResampleContext *avr, int sample_delta,
387  int compensation_distance);
388 
389 /**
390  *
391  * @deprecated use libswresample
392  *
393  * Provide the upper bound on the number of samples the configured
394  * conversion would output.
395  *
396  * @param avr audio resample context
397  * @param in_nb_samples number of input samples
398  *
399  * @return number of samples or AVERROR(EINVAL) if the value
400  * would exceed INT_MAX
401  */
403 int avresample_get_out_samples(AVAudioResampleContext *avr, int in_nb_samples);
404 
405 /**
406  *
407  * @deprecated use libswresample
408  *
409  * Convert input samples and write them to the output FIFO.
410  *
411  * The upper bound on the number of output samples can be obtained through
412  * avresample_get_out_samples().
413  *
414  * The output data can be NULL or have fewer allocated samples than required.
415  * In this case, any remaining samples not written to the output will be added
416  * to an internal FIFO buffer, to be returned at the next call to this function
417  * or to avresample_read().
418  *
419  * If converting sample rate, there may be data remaining in the internal
420  * resampling delay buffer. avresample_get_delay() tells the number of remaining
421  * samples. To get this data as output, call avresample_convert() with NULL
422  * input.
423  *
424  * At the end of the conversion process, there may be data remaining in the
425  * internal FIFO buffer. avresample_available() tells the number of remaining
426  * samples. To get this data as output, either call avresample_convert() with
427  * NULL input or call avresample_read().
428  *
429  * @see avresample_get_out_samples()
430  * @see avresample_read()
431  * @see avresample_get_delay()
432  *
433  * @param avr audio resample context
434  * @param output output data pointers
435  * @param out_plane_size output plane size, in bytes.
436  * This can be 0 if unknown, but that will lead to
437  * optimized functions not being used directly on the
438  * output, which could slow down some conversions.
439  * @param out_samples maximum number of samples that the output buffer can hold
440  * @param input input data pointers
441  * @param in_plane_size input plane size, in bytes
442  * This can be 0 if unknown, but that will lead to
443  * optimized functions not being used directly on the
444  * input, which could slow down some conversions.
445  * @param in_samples number of input samples to convert
446  * @return number of samples written to the output buffer,
447  * not including converted samples added to the internal
448  * output FIFO
449  */
452  int out_plane_size, int out_samples,
453  uint8_t * const *input, int in_plane_size,
454  int in_samples);
455 
456 /**
457  *
458  * @deprecated use libswresample
459  *
460  * Return the number of samples currently in the resampling delay buffer.
461  *
462  * When resampling, there may be a delay between the input and output. Any
463  * unconverted samples in each call are stored internally in a delay buffer.
464  * This function allows the user to determine the current number of samples in
465  * the delay buffer, which can be useful for synchronization.
466  *
467  * @see avresample_convert()
468  *
469  * @param avr audio resample context
470  * @return number of samples currently in the resampling delay buffer
471  */
474 
475 /**
476  *
477  * @deprecated use libswresample
478  *
479  * Return the number of available samples in the output FIFO.
480  *
481  * During conversion, if the user does not specify an output buffer or
482  * specifies an output buffer that is smaller than what is needed, remaining
483  * samples that are not written to the output are stored to an internal FIFO
484  * buffer. The samples in the FIFO can be read with avresample_read() or
485  * avresample_convert().
486  *
487  * @see avresample_read()
488  * @see avresample_convert()
489  *
490  * @param avr audio resample context
491  * @return number of samples available for reading
492  */
495 
496 /**
497  *
498  * @deprecated use libswresample
499  *
500  * Read samples from the output FIFO.
501  *
502  * During conversion, if the user does not specify an output buffer or
503  * specifies an output buffer that is smaller than what is needed, remaining
504  * samples that are not written to the output are stored to an internal FIFO
505  * buffer. This function can be used to read samples from that internal FIFO.
506  *
507  * @see avresample_available()
508  * @see avresample_convert()
509  *
510  * @param avr audio resample context
511  * @param output output data pointers. May be NULL, in which case
512  * nb_samples of data is discarded from output FIFO.
513  * @param nb_samples number of samples to read from the FIFO
514  * @return the number of samples written to output
515  */
517 int avresample_read(AVAudioResampleContext *avr, uint8_t **output, int nb_samples);
518 
519 /**
520  *
521  * @deprecated use libswresample
522  *
523  * Convert the samples in the input AVFrame and write them to the output AVFrame.
524  *
525  * Input and output AVFrames must have channel_layout, sample_rate and format set.
526  *
527  * The upper bound on the number of output samples is obtained through
528  * avresample_get_out_samples().
529  *
530  * If the output AVFrame does not have the data pointers allocated the nb_samples
531  * field will be set using avresample_get_out_samples() and av_frame_get_buffer()
532  * is called to allocate the frame.
533  *
534  * The output AVFrame can be NULL or have fewer allocated samples than required.
535  * In this case, any remaining samples not written to the output will be added
536  * to an internal FIFO buffer, to be returned at the next call to this function
537  * or to avresample_convert() or to avresample_read().
538  *
539  * If converting sample rate, there may be data remaining in the internal
540  * resampling delay buffer. avresample_get_delay() tells the number of
541  * remaining samples. To get this data as output, call this function or
542  * avresample_convert() with NULL input.
543  *
544  * At the end of the conversion process, there may be data remaining in the
545  * internal FIFO buffer. avresample_available() tells the number of remaining
546  * samples. To get this data as output, either call this function or
547  * avresample_convert() with NULL input or call avresample_read().
548  *
549  * If the AVAudioResampleContext configuration does not match the output and
550  * input AVFrame settings the conversion does not take place and depending on
551  * which AVFrame is not matching AVERROR_OUTPUT_CHANGED, AVERROR_INPUT_CHANGED
552  * or AVERROR_OUTPUT_CHANGED|AVERROR_INPUT_CHANGED is returned.
553  *
554  * @see avresample_get_out_samples()
555  * @see avresample_available()
556  * @see avresample_convert()
557  * @see avresample_read()
558  * @see avresample_get_delay()
559  *
560  * @param avr audio resample context
561  * @param output output AVFrame
562  * @param input input AVFrame
563  * @return 0 on success, AVERROR on failure or nonmatching
564  * configuration.
565  */
568  AVFrame *output, AVFrame *input);
569 
570 /**
571  *
572  * @deprecated use libswresample
573  *
574  * Configure or reconfigure the AVAudioResampleContext using the information
575  * provided by the AVFrames.
576  *
577  * The original resampling context is reset even on failure.
578  * The function calls avresample_close() internally if the context is open.
579  *
580  * @see avresample_open();
581  * @see avresample_close();
582  *
583  * @param avr audio resample context
584  * @param out output AVFrame
585  * @param in input AVFrame
586  * @return 0 on success, AVERROR on failure.
587  */
590 
591 /**
592  * @}
593  */
594 
595 #endif /* AVRESAMPLE_AVRESAMPLE_H */
AV_RESAMPLE_DITHER_TRIANGULAR_HP
Triangular Dither with High Pass.
Definition: avresample.h:133
attribute_deprecated const char * avresample_license(void)
Definition: utils.c:784
attribute_deprecated int avresample_set_matrix(AVAudioResampleContext *avr, const double *matrix, int stride)
Definition: utils.c:666
AV_MIX_COEFF_TYPE_Q8
Definition: avresample.h:113
attribute_deprecated int avresample_set_compensation(AVAudioResampleContext *avr, int sample_delta, int compensation_distance)
Definition: resample.c:233
This structure describes decoded (raw) audio or video data.
Definition: frame.h:218
attribute_deprecated int avresample_open(AVAudioResampleContext *avr)
Definition: utils.c:36
attribute_deprecated int avresample_read(AVAudioResampleContext *avr, uint8_t **output, int nb_samples)
Definition: utils.c:772
attribute_deprecated int avresample_get_delay(AVAudioResampleContext *avr)
Definition: resample.c:438
enum attribute_deprecated AVResampleFilterType
Definition: avresample.h:123
attribute_deprecated int 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)
Definition: utils.c:330
Convenience header that includes libavutil&#39;s core.
enum attribute_deprecated AVMixCoeffType
Definition: avresample.h:112
int stride
Definition: mace.c:144
attribute_deprecated int avresample_convert_frame(AVAudioResampleContext *avr, AVFrame *output, AVFrame *input)
Definition: utils.c:603
double surround_mix_level
surround mix level
Definition: internal.h:65
Public dictionary API.
uint8_t
attribute_deprecated int avresample_get_out_samples(AVAudioResampleContext *avr, int in_nb_samples)
Definition: utils.c:753
AV_MIX_COEFF_TYPE_NB
floating-point
Definition: avresample.h:113
attribute_deprecated void avresample_close(AVAudioResampleContext *avr)
Definition: utils.c:262
AV_RESAMPLE_DITHER_NONE
Do not use dithering.
Definition: avresample.h:133
AV_RESAMPLE_DITHER_TRIANGULAR_NS
Triangular Dither with Noise Shaping.
Definition: avresample.h:133
double center_mix_level
center mix level
Definition: internal.h:64
attribute_deprecated int avresample_config(AVAudioResampleContext *avr, AVFrame *out, AVFrame *in)
Definition: utils.c:505
AV_RESAMPLE_FILTER_TYPE_BLACKMAN_NUTTALL
Blackman Nuttall Windowed Sinc.
Definition: avresample.h:124
attribute_deprecated int avresample_available(AVAudioResampleContext *avr)
Definition: utils.c:748
reference-counted frame API
AV_RESAMPLE_DITHER_NB
Number of dither types.
Definition: avresample.h:133
attribute_deprecated int avresample_get_matrix(AVAudioResampleContext *avr, double *matrix, int stride)
Definition: utils.c:637
audio channel layout utility functions
attribute_deprecated const char * avresample_configuration(void)
Definition: utils.c:790
attribute_deprecated unsigned avresample_version(void)
Definition: utils.c:779
attribute_deprecated const AVClass * avresample_get_class(void)
Definition: options.c:110
attribute_deprecated int avresample_is_open(AVAudioResampleContext *avr)
Definition: utils.c:257
attribute_deprecated int avresample_build_matrix(uint64_t in_layout, uint64_t out_layout, double center_mix_level, double surround_mix_level, double lfe_mix_level, int normalize, double *matrix, int stride, enum AVMatrixEncoding matrix_encoding)
attribute_deprecated int avresample_set_channel_mapping(AVAudioResampleContext *avr, const int *channel_map)
Definition: utils.c:697
enum attribute_deprecated AVResampleDitherMethod
Definition: avresample.h:132
uint8_t pi<< 24) CONV_FUNC_GROUP(AV_SAMPLE_FMT_FLT, float, AV_SAMPLE_FMT_U8, uint8_t,(*(const uint8_t *) pi - 0x80) *(1.0f/(1<< 7))) CONV_FUNC_GROUP(AV_SAMPLE_FMT_DBL, double, AV_SAMPLE_FMT_U8, uint8_t,(*(const uint8_t *) pi - 0x80) *(1.0/(1<< 7))) CONV_FUNC_GROUP(AV_SAMPLE_FMT_U8, uint8_t, AV_SAMPLE_FMT_S16, int16_t,(*(const int16_t *) pi >> 8)+0x80) CONV_FUNC_GROUP(AV_SAMPLE_FMT_FLT, float, AV_SAMPLE_FMT_S16, int16_t, *(const int16_t *) pi *(1.0f/(1<< 15))) CONV_FUNC_GROUP(AV_SAMPLE_FMT_DBL, double, AV_SAMPLE_FMT_S16, int16_t, *(const int16_t *) pi *(1.0/(1<< 15))) CONV_FUNC_GROUP(AV_SAMPLE_FMT_U8, uint8_t, AV_SAMPLE_FMT_S32, int32_t,(*(const int32_t *) pi >> 24)+0x80) CONV_FUNC_GROUP(AV_SAMPLE_FMT_FLT, float, AV_SAMPLE_FMT_S32, int32_t, *(const int32_t *) pi *(1.0f/(1U<< 31))) CONV_FUNC_GROUP(AV_SAMPLE_FMT_DBL, double, AV_SAMPLE_FMT_S32, int32_t, *(const int32_t *) pi *(1.0/(1U<< 31))) CONV_FUNC_GROUP(AV_SAMPLE_FMT_U8, uint8_t, AV_SAMPLE_FMT_FLT, float, av_clip_uint8(lrintf(*(const float *) pi *(1<< 7))+0x80)) CONV_FUNC_GROUP(AV_SAMPLE_FMT_S16, int16_t, AV_SAMPLE_FMT_FLT, float, av_clip_int16(lrintf(*(const float *) pi *(1<< 15)))) CONV_FUNC_GROUP(AV_SAMPLE_FMT_S32, int32_t, AV_SAMPLE_FMT_FLT, float, av_clipl_int32(llrintf(*(const float *) pi *(1U<< 31)))) CONV_FUNC_GROUP(AV_SAMPLE_FMT_U8, uint8_t, AV_SAMPLE_FMT_DBL, double, av_clip_uint8(lrint(*(const double *) pi *(1<< 7))+0x80)) CONV_FUNC_GROUP(AV_SAMPLE_FMT_S16, int16_t, AV_SAMPLE_FMT_DBL, double, av_clip_int16(lrint(*(const double *) pi *(1<< 15)))) CONV_FUNC_GROUP(AV_SAMPLE_FMT_S32, int32_t, AV_SAMPLE_FMT_DBL, double, av_clipl_int32(llrint(*(const double *) pi *(1U<< 31)))) #define SET_CONV_FUNC_GROUP(ofmt, ifmt) static void set_generic_function(AudioConvert *ac) { } void ff_audio_convert_free(AudioConvert **ac) { if(! *ac) return;ff_dither_free(&(*ac) ->dc);av_freep(ac);} AudioConvert *ff_audio_convert_alloc(AVAudioResampleContext *avr, enum AVSampleFormat out_fmt, enum AVSampleFormat in_fmt, int channels, int sample_rate, int apply_map) { AudioConvert *ac;int in_planar, out_planar;ac=av_mallocz(sizeof(*ac));if(!ac) return NULL;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);return NULL;} return ac;} 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;} else if(in_planar) ac->func_type=CONV_FUNC_TYPE_INTERLEAVE;else ac->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);return ac;} int ff_audio_convert(AudioConvert *ac, AudioData *out, AudioData *in) { int use_generic=1;int len=in->nb_samples;int p;if(ac->dc) { av_log(ac->avr, AV_LOG_TRACE, "%d samples - audio_convert: %s to %s (dithered)\", len, av_get_sample_fmt_name(ac->in_fmt), av_get_sample_fmt_name(ac->out_fmt));return ff_convert_dither(ac-> in
Describe the class of an AVClass context structure.
Definition: log.h:67
AV_RESAMPLE_FILTER_TYPE_CUBIC
Cubic.
Definition: avresample.h:124
attribute_deprecated AVAudioResampleContext * avresample_alloc_context(void)
Definition: options.c:96
enum AVMatrixEncoding matrix_encoding
matrixed stereo encoding
Definition: internal.h:97
AV_RESAMPLE_DITHER_RECTANGULAR
Rectangular Dither.
Definition: avresample.h:133
attribute_deprecated void avresample_free(AVAudioResampleContext **avr)
Definition: utils.c:278
AV_RESAMPLE_FILTER_TYPE_KAISER
Kaiser Windowed Sinc.
Definition: avresample.h:124
AV_MIX_COEFF_TYPE_FLT
32-bit 17.15 fixed-point
Definition: avresample.h:113
FILE * out
Definition: movenc.c:54
AV_RESAMPLE_DITHER_TRIANGULAR
Triangular Dither.
Definition: avresample.h:133
AVMatrixEncoding
Libavresample version macros.
AV_MIX_COEFF_TYPE_Q15
16-bit 8.8 fixed-point
Definition: avresample.h:113
double lfe_mix_level
lfe mix level
Definition: internal.h:66